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Hello I am using the asterisk of the new version. Please give me some help to configure the IAXsoftphone to IAXsoftphone with asterisk. Thanks in advanced dipak -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040818/24320004/attachment.htm
Dipak wrote:> I am using the asterisk of the new version. Please give me some help to > configure the IAXsoftphone to IAXsoftphone with asterisk. > >Can you at least give more details? What is not working? What is working? What are you trying to do? Any samples of your extensions.conf and iax.conf files??? Flynn
Please unsubscribe me from the distribution list. Thanks. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Friday, August 20, 2004 8:11 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users digest, Vol 1 #4999 - 13 msgs Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to asterisk-users-request@lists.digium.com You can reach the person managing the list at asterisk-users-admin@lists.digium.com When replying, please edit your Subject line so it is more specific than "Re: Contents of Asterisk-Users digest..." Today's Topics: 1. Re: spandsp (administrator tootai) 2. Number unavailable (Thomas Kuepper) 3. Problem with asterisk and pgsql (DIPAK PAUL) 4. Re: telnet and Root (Thomas Kuepper) 5. Re: telnet and Root (Roberto Piola) 6. x100p won't answer (Imran Akbar) 7. Multi-bitrate codecs (Simone Ricci) 8. Re: problem in with mysql modules.conf to load cdr_addon_mysql.so (Lerale Erwan) 9. IAXY S100I noise (Conrad Vermeulen) 10. Re: Request for help designing an unusual * application (Rich Adamson) 11. Problem in with mysql modules.conf to load cdr_addon_mysql.so (DIPAK PAUL) 12. Re: Problem in with mysql modules.conf to load cdr_addon_mysql.so (Lerale Erwan) 13. Re: How to run different codecs between the same endpoints on an IAX trunk? (Andrew Kohlsmith) --__--__-- Message: 1 Date: Fri, 20 Aug 2004 12:01:37 +0200 From: administrator tootai <admin@tootai.net> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] spandsp Reply-To: asterisk-users@lists.digium.com Simone Ricci a ?crit :> [...] > Bingo! Maybe spandsp isn't SMP friendly...thus it doesn't work with > multiprocessor/hyperthreading systems? Can anyone confirm this? > Eventually, try booting your system with a non-SMP kernel, just to see > what happens.If nobody do it this days I will do it next week. -- Daniel --__--__-- Message: 2 To: asterisk-users@lists.digium.com From: Thomas Kuepper <tk@teldafax.de> Subject: [Asterisk-Users] Number unavailable Date: Fri, 20 Aug 2004 11:57:25 +0200 Reply-To: asterisk-users@lists.digium.com hi list, no call into pstn and from pstn to asterisk endpoints displays the=20 calling party number. I see in my sip phones number unavailable. in the=20 asterisk cdrs i see the number which is calling but at the endpoints=20 its unavailable. any hints how i can activate to display the calling from number? thx -- Thomas K=FCpper 01063 Telecom GmbH & Co. KG Mottmannstr. 2 53842 Troisdorf Telefon: 02241-9434-506 Telefax: 02241-9434-846 E-Mail: thomas.kuepper@01063telecom.de E-Mail: tk@teldafax.de Homepage: http://www.01063telecom.de --------------------------------------- Diese Nachricht ist vertraulich. Sie ist ausschliesslich fuer den im Adressfeld ausgewiesenen Adressaten bestimmt. Sollten Sie nicht der vorgesehene Empfaenger sein, so bitten wir um eine kurze Nachricht. Jede unbefugte Weiterleitung oder Fertigung einer Kopie ist unzulaessig. Da=20 wir nicht die Echtheit oder Vollstaendigkeit der in dieser Nachricht=20 enthaltenen Informationen garantieren koennen, schliessen wir die rechtliche Verbindlichkeit der vorstehenden Erklaerungen und Aeusserungen aus. Wir verweisen in diesem Zusammenhang auch auf die fuer uns geltenden=20 Regelungen ueber die Verbindlichkeit von Willenserklaerungen mit verpflichtendem Inhalt, die in den bank- bzw. unternehmensueblichen Unterschriftenverzeichnissen bekannt gemacht werden. --------------------------------------- This message is confidential and may be privileged. It is intended=20 solely for the named addressee. If you are not the intended recipient please=20 inform us. Any unauthorised dissemination, distribution or copying hereof is prohibited. As we cannot guarantee the genuineness or completeness of=20 the information contained in this message, the statements set forth above=20 are not legally binding. In connection therewith, we also refer to our=20 governing regulations of concerning signatory authority published in the standard=20 bank or company signature lists with regard to the legally binding effect of statements made with the intent to obligate us. --------------------------------------- --__--__-- Message: 3 From: "DIPAK PAUL" <dipak_kr_paul@hotmail.com> To: asterisk-users@lists.digium.com Cc: asterisk-perl-subscribe@lists.gnuinter.net Date: Fri, 20 Aug 2004 15:32:13 +0530 Subject: [Asterisk-Users] Problem with asterisk and pgsql Reply-To: asterisk-users@lists.digium.com Hi I am a asterisk user. My asterisk is working fine. But I want to store all the billing information ot the postgradesql. I had installed properly in my linux box. But when I have used createdb mydb I faced this type of problem "createdb: could not connect to database template1: could not connect to server: No such file or directory Is the server running locally and accepting connections on Unix domain socket "/tmp/.s.PGSQL.5432"? " I have already start pgsql from the services Please help me. Thanks in advanced. Dipak Kumar Paul _________________________________________________________________ Marriage? http://www.bharatmatrimony.com/cgi-bin/bmclicks1.cgi?74 Join BharatMatrimony.com for free. --__--__-- Message: 4 From: Thomas Kuepper <tk@teldafax.de> Subject: Re: [Asterisk-Users] telnet and Root Date: Fri, 20 Aug 2004 12:02:37 +0200 To: asterisk-users@lists.digium.com Reply-To: asterisk-users@lists.digium.com --Apple-Mail-2-872324320 Content-Transfer-Encoding: quoted-printable Content-Type: text/plain; charset=ISO-8859-1; format=flowed use ssh instead of telnet. telnet is a bad idea. Am 20.08.2004 um 11:39 schrieb neil:> Sorry if this is posted to the wrong forum but as it is related to a=20 > problem I have with Asterisk it may just scrape through!! > > =A0 > > I am running Fedora 1 and I can telnet in to my asterisk box as any=20 > user except root and am using the same credentials as logging in=20 > locally. I am new to Linux and any help would be gratefully=20 > appreciated. > > =A0 > > Thanks > > =A0 > > Neil >-- Thomas K=FCpper 01063 Telecom GmbH & Co. KG Mottmannstr. 2 53842 Troisdorf Telefon: 02241-9434-506 Telefax: 02241-9434-846 E-Mail: thomas.kuepper@01063telecom.de E-Mail: tk@teldafax.de Homepage: http://www.01063telecom.de --------------------------------------- Diese Nachricht ist vertraulich. Sie ist ausschliesslich fuer den im Adressfeld ausgewiesenen Adressaten bestimmt. Sollten Sie nicht der vorgesehene Empfaenger sein, so bitten wir um eine kurze Nachricht. Jede unbefugte Weiterleitung oder Fertigung einer Kopie ist unzulaessig. Da=20 wir nicht die Echtheit oder Vollstaendigkeit der in dieser Nachricht=20 enthaltenen Informationen garantieren koennen, schliessen wir die rechtliche Verbindlichkeit der vorstehenden Erklaerungen und Aeusserungen aus. Wir verweisen in diesem Zusammenhang auch auf die fuer uns geltenden=20 Regelungen ueber die Verbindlichkeit von Willenserklaerungen mit verpflichtendem Inhalt, die in den bank- bzw. unternehmensueblichen Unterschriftenverzeichnissen bekannt gemacht werden. --------------------------------------- This message is confidential and may be privileged. It is intended=20 solely for the named addressee. If you are not the intended recipient please=20 inform us. Any unauthorised dissemination, distribution or copying hereof is prohibited. As we cannot guarantee the genuineness or completeness of=20 the information contained in this message, the statements set forth above=20 are not legally binding. In connection therewith, we also refer to our=20 governing regulations of concerning signatory authority published in the standard=20 bank or company signature lists with regard to the legally binding effect of statements made with the intent to obligate us. --------------------------------------- --Apple-Mail-2-872324320 Content-Transfer-Encoding: quoted-printable Content-Type: text/enriched; charset=ISO-8859-1 use ssh instead of telnet. telnet is a bad idea. Am 20.08.2004 um 11:39 schrieb neil: <excerpt><fontfamily><param>Arial</param><x-tad-bigger>Sorry if this is posted to the wrong forum but as it is related to a problem I have with Asterisk it may just scrape through!!</x-tad-bigger></fontfamily> <fontfamily><param>Arial</param><x-tad-bigger>=A0</x-tad-bigger></fontfamily> <fontfamily><param>Arial</param><x-tad-bigger>I am running Fedora 1 and I can telnet in to my asterisk box as any user except root and am using the same credentials as logging in locally. I am new to Linux and any help would be gratefully appreciated.</x-tad-bigger></fontfamily> <fontfamily><param>Arial</param><x-tad-bigger>=A0</x-tad-bigger></fontfamily> <fontfamily><param>Arial</param><x-tad-bigger>Thanks</x-tad-bigger></fontfamily> <fontfamily><param>Arial</param><x-tad-bigger>=A0</x-tad-bigger></fontfamily> <fontfamily><param>Arial</param><x-tad-bigger>Neil</x-tad-bigger></fontfamily> </excerpt>-- Thomas K=FCpper 01063 Telecom GmbH & Co. KG Mottmannstr. 2 53842 Troisdorf Telefon: 02241-9434-506 Telefax: 02241-9434-846 E-Mail: thomas.kuepper@01063telecom.de E-Mail: tk@teldafax.de Homepage: http://www.01063telecom.de --------------------------------------- Diese Nachricht ist vertraulich. Sie ist ausschliesslich fuer den im Adressfeld ausgewiesenen Adressaten bestimmt. Sollten Sie nicht der vorgesehene Empfaenger sein, so bitten wir um eine kurze Nachricht. Jede unbefugte Weiterleitung oder Fertigung einer Kopie ist unzulaessig. Da wir nicht die Echtheit oder Vollstaendigkeit der in dieser Nachricht enthaltenen Informationen garantieren koennen, schliessen wir die rechtliche Verbindlichkeit der vorstehenden Erklaerungen und Aeusserungen aus. Wir verweisen in diesem Zusammenhang auch auf die fuer uns geltenden Regelungen ueber die Verbindlichkeit von Willenserklaerungen mit verpflichtendem Inhalt, die in den bank- bzw. unternehmensueblichen Unterschriftenverzeichnissen bekannt gemacht werden. --------------------------------------- This message is confidential and may be privileged. It is intended solely for the named addressee. If you are not the intended recipient please inform us. Any unauthorised dissemination, distribution or copying hereof is prohibited. As we cannot guarantee the genuineness or completeness of the information contained in this message, the statements set forth above are not legally binding. In connection therewith, we also refer to our governing regulations of concerning signatory authority published in the standard bank or company signature lists with regard to the legally binding effect of statements made with the intent to obligate us. --------------------------------------- --Apple-Mail-2-872324320-- --__--__-- Message: 5 From: Roberto Piola <Roberto.Piola@gruppoih.it> To: "'asterisk-users@lists.digium.com'" <asterisk-users@lists.digium.com> Cc: "'neil@3tech.biz'" <neil@3tech.biz> Date: Fri, 20 Aug 2004 12:31:28 +0200 Subject: [Asterisk-Users] Re: telnet and Root Reply-To: asterisk-users@lists.digium.com you can login as root only on the console or on the lines listed in /etc/securetty if you want to log in remotely as root, you can either: - log in as a regular user and then issue the "su -" command in order to become root - use a ssh client (secure shell) instead of telnet (well, you can disable root access in ssh as well) ------ Original message: From: "neil" <neil@3tech.biz> To: <asterisk-users@lists.digium.com> Date: Fri, 20 Aug 2004 10:39:03 +0100 Subject: [Asterisk-Users] telnet and Root Sorry if this is posted to the wrong forum but as it is related to a problem I have with Asterisk it may just scrape through!! I am running Fedora 1 and I can telnet in to my asterisk box as any user except root and am using the same credentials as logging in locally. I am new to Linux and any help would be gratefully appreciated. Thanks --__--__-- Message: 6 Date: Fri, 20 Aug 2004 15:35:31 +0500 From: Imran Akbar <mail@akbars.net> To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] x100p won't answer Reply-To: asterisk-users@lists.digium.com Hi, I just got two digium x100p clones and installed asterisk on fedora core 2 which took some tweaking. After getting asterisk up I installed the zaptel stuff - then modprobed zaptel, wcfxs (for the fxo cards), which worked fine. ztcfg is showing two channels configured, but when I start asterisk and do show channels, i see no active channels. zapata.conf has: signalling = fxs_ks context = line1 channel => 1 signalling = fxs_ks context = line1 channel => 2 zaptel.conf has: loadzone=us defaultzone=us fxsks=1-2 extensions.conf has: [line1] exten => s,1,Answer exten => s,2,DigitTimeout,5 exten => s,3,ResponseTimeout,10 exten => s,4,BackGround(demo-congrats) exten => s,5,BackGround(demo-instruct) [line2] exten => s,1,Answer exten => s,2,DigitTimeout,5 exten => s,3,ResponseTimeout,10 exten => s,4,BackGround(demo-congrats) exten => s,5,BackGround(demo-instruct) I have no idea why it's not working would appreciate any help Thanks, Imran --__--__-- Message: 7 Date: Fri, 20 Aug 2004 12:42:31 +0200 From: Simone Ricci <simone.ricci@cwnet.it> To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Multi-bitrate codecs Reply-To: asterisk-users@lists.digium.com Anyone knows if there's a way to select the bitrate of those codecs supporting multiple bitrates (eg. g.726)? I've tried searching and googling a lot, but without useful results... Cheers, Simone. --__--__-- Message: 8 Date: Fri, 20 Aug 2004 12:54:59 +0200 From: Lerale Erwan <erwan@fr.clara.net> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] problem in with mysql modules.conf to load cdr_addon_mysql.so Reply-To: asterisk-users@lists.digium.com On Fri, Aug 20, 2004 at 03:25:31PM +0530, DIPAK PAUL wrote:> > [cdr_addon_mysql.so]Aug 20 15:18:38 WARNING[1076245120]: loader.c:242 > ast_load_resource: /usr/lib/asterisk/modules/cdr_addon_mysql.so: cannot > open shared object file: No such file or directory > Aug 20 15:18:38 WARNING[1076245120]: loader.c:374 load_modules: Loading > module cdr_addon_mysql.so failed!Hello, Do you really have cdr_addon_mysql.so in /usr/lib/asterisk/modules/ ? did you run : www# make install for x in cdr_addon_mysql.so; do install -m 755 $x /usr/lib/asterisk/modules ; done www# ? Cheers r1 -- Manager Support Pro - www.claranet.fr --__--__-- Message: 9 From: Conrad Vermeulen <conrad@novagrove.com> To: asterisk-users@lists.digium.com Date: Fri, 20 Aug 2004 12:42:43 +0200 Subject: [Asterisk-Users] IAXY S100I noise Reply-To: asterisk-users@lists.digium.com Hi, What noise should I expect to hear once I've plugged an IAXY into my network and connected to Asterisk? I suspect I should hear a dialtone? I'm not sure if I have a faulty device or have a configuration issue. Currently,it is making a seriously strange noise - machine gun like with static. Any comments appreciated. Thanks, Conrad --__--__-- Message: 10 Date: Fri, 20 Aug 2004 05:59:09 -0600 From: Rich Adamson <radamson@routers.com> Subject: Re: [Asterisk-Users] Request for help designing an unusual * application To: asterisk-users@lists.digium.com Reply-To: asterisk-users@lists.digium.com> I have been reading asterisk doc's for the past couple weeks, and > monitoring this list. I have to implement an unusual (I think) > application of asterisk. I have the beginnings of a plan, and I would > like to throw it up here for comments. > > The application: > An after-hours emergency support "hotline" for our technology company. > > We have 5 different support people that take turns on 24-hour call (at > any time, one support person is "on call"). We may have 3 or more > contact numbers for each person, eg: > - office phone > - cell phone > - home phone > > The support people, and their contact numbers, would ideally be stored > in a database. > > When a customer calls in, they get a canned greeting, and then they > leave a message. Asterisk records it. > > Asterisk then tries to reach the current "on call" person. It startsTechnically, all of that's very possible. Having been through similar on-call arrangements in a previous life, you might want to consider a slightly different management approach. Whoever is the primary on-call person, allow that person to call into asterisk and enter their on-call number. Store it, and use it for delivery of the calls. Same for a secondary (or backup) on call person. The management problem with your approach essentially is one that says the 'system' will dictate the calling numbers. In practical cases, if the on-call person knowingly is at a location where cell phones (etc) don't work, the system breaks down. Whereas, if you allow the on-call person to program a "can-be-reached-at" variable, the on-call person has the freedom to move around and it becomes that person's responsibility to provide the contact number. Accountability is shifted to the on-call person, which then avoids the finger pointing that happens when the 'system' can't properly dispatch the call. Same technical requirements, slightly different way to handle the called numbers. Using an approach similar to the above, its trivial to build an escalation process into the call delivery mechanism. (E.g, if the primary on-call person doesn't respond within xx minutes, then dispatch the secondary. If that person doesn't respond, dispatch same message to a manager, etc, etc.) Typical problems go something like: - was in church/mass, turned my cell phone off - guess its marginal coverage at my girlfriend's house - I was at xyz's house and didn't receive the call - etc Rich --__--__-- Message: 11 From: "DIPAK PAUL" <dipak_kr_paul@hotmail.com> To: asterisk-users@lists.digium.com Date: Fri, 20 Aug 2004 17:02:02 +0530 Subject: [Asterisk-Users] Problem in with mysql modules.conf to load cdr_addon_mysql.so Reply-To: asterisk-users@lists.digium.com Hi Lerale Erwan I have used "make install" after downloaded the asterisk and also i have used another time "make install" after using cd /usr/src cvs checkout asterisk-addons There are no cdr_addon_mysql.so file not created in the /usr/lib/asterisk/modules/ and also /usr/src/asterisk/cdr directory. The sorce directories of my machine are /usr/src/asterisk /usr/src/asterisk-addons In this directory present the files: cdr_addon_mysql.c Makefile mkdep mysql-vm-routines.h and three directory configs,CVS,doc. My previous mail is I have installed asterisk from cvs properly. I have try to store the billing information in the mysql database. I had used this cd /usr/src cvs checkout asterisk-addons And also copy sample configuration file to /etc/asterisk/cdr_mysql.conf & edit it according to my requirements. I had edit the modules.conf to load cdr_addon_mysql.so my cdr_mysql.conf file is like [global] hostname=localhost dbname=asteriskcdrdb password=password user=asteriskcdruser port=3306 sock=/var/lib/mysql/mysql.sock And also I had added one line in the module.conf file load => cdr_addon_mysql.so and again I make and make install in the asterisk But when I have using this command "./asterisk -vvvvc" I got an error message and stop [cdr_addon_mysql.so]Aug 20 15:18:38 WARNING[1076245120]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/cdr_addon_mysql.so: cannot open shared object file: No such file or directory Aug 20 15:18:38 WARNING[1076245120]: loader.c:374 load_modules: Loading module cdr_addon_mysql.so failed! Please help me Thanks in advanced Dipak Kumar Paul _________________________________________________________________ Sports, sports and more sports! Keep up with all the action! http://www.msn.co.in/sports/ Stay connected with MSN Sports! --__--__-- Message: 12 Date: Fri, 20 Aug 2004 13:40:44 +0200 From: Lerale Erwan <erwan@fr.clara.net> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Problem in with mysql modules.conf to load cdr_addon_mysql.so Reply-To: asterisk-users@lists.digium.com On Fri, Aug 20, 2004 at 05:02:02PM +0530, DIPAK PAUL wrote:> > There are no cdr_addon_mysql.so file not created in the > /usr/lib/asterisk/modules/ and also /usr/src/asterisk/cdr directory.So... that's definitly your problem :) Did you run "make install" here : /usr/local/src/ASTERISK/asterisk-addons ? -- Manager Support Pro - www.claranet.fr --__--__-- Message: 13 From: Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com> Organization: Benshaw Canada To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] How to run different codecs between the same endpoints on an IAX trunk? Date: Fri, 20 Aug 2004 07:52:44 -0400 Reply-To: asterisk-users@lists.digium.com On Thursday 19 August 2004 18:11, Kris Boutilier wrote:> I have a situation where I'm using G.729A as my IAX trunking codec. Now I > need to push some short duration, low bitrate modem traffic over the link > (a credit card terminal). Obviously the modem audio isn't going to survive > the G.729 codec process intact, so for the times the device is used I'd > like to service calls from that device (and only that device) with a > higher-data rate codec.Am I missing something? I do this with my Office Asterisk and Colocation Asterisk boxes all the time. On the colocation asterisk box (office asterisk box is pretty much identical to allow outgoing calls and outgoing faxes): [officeasterisk] type=peer host=192.168.2.2 qualify=500 notransfer=yes [officephone] type=user context=officephone host=192.168.2.2 qualify=500 disallow=ulaw notransfer=yes [officefax] type=user context=faxout host=192.168.2.2 qualify=500 disallow=all allow=ulaw notransfer=yes When a fax call comes in from the PRI, it calls officefax. When a phone call comes in, it calls officephone. Similarly when someone calls out, it calls colophone (same as officephone but with different host), and likewise, outgoing faxes use colofax. Seems to be working fine here, and has since June when we moved. :-) Now I have turned off trunking between officeasterisk and coloasterisk -- the 20040806 CVS HEAD seemed to not like to trunk between two boxes running that code, but would trunk just fine to RC1 (with the trunking fixes that Steve did)... Is this the kind of thing you want? I was trunking between office and colocation up until the last source update I did and it seemed to work just fine. -A. --__--__-- _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest
My server is generating a bunch of these channels from different parts. How can I avoid them? Thank you jmoura Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format 24.53.209.102 (None) 00515/29458 00002/00002 00000ms 0000ms 0000ms UNKN 24.53.209.102 (None) 00583/29362 00002/00002 00000ms 0000ms 0000ms UNKN 24.53.209.102 (None) 00584/29058 00002/00002 00000ms 0000ms 0000ms UNKN 24.53.209.102 (None) 00585/29596 00002/00002 00000ms 0000ms 0000ms UNKN 24.53.209.102 (None) 00586/29216 00002/00002 00000ms 0000ms 0000ms UNKN 24.53.209.102 (None) 00587/29217 00002/00002 00000ms 0000ms 0000ms UNKN 24.53.209.102 (None) 00588/29464 00002/00002 00000ms 0000ms 0000ms UNKN 24.53.209.102 (None) 00589/29465 00002/00002 00000ms 0000ms 0000ms UNKN 24.53.209.102 (None) 00590/28804 00002/00002 00000ms 0000ms 0000ms UNKN 24.53.209.102 (None) 00591/29059 00001/00001 00000ms 0000ms 0000ms UNKN 24.53.209.102 (None) 00592/28805 00001/00001 00000ms 0000ms 0000ms UNKN 24.53.209.102 (None) 00593/29597 00001/00001 00000ms 0000ms 0000ms UNKN 24.53.209.102 (None) 00594/29466 00001/00001 00000ms 0000ms 0000ms UNKN 24.53.209.102 (None) 00595/29218 00001/00001 00000ms 0000ms 0000ms UNKN
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Citando asterisk-users-request@lists.digium.com:> Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > > You can reach the person managing the list at > asterisk-users-owner@lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Asterisk-Users digest..." > > > Today's Topics: > > 1. Re: append # to dial string (Eric Wieling) > 2. Re: VAD/DTX implementation through zaptel cards (Eric Wieling) > 3. Re: CDR and TDS (Eric Wieling) > 4. RE: Zaptel Compile on a virtual dedicated host. > (vgrskovic@optonline.net) > 5. Re: TE110P/Hipath3750 - Yellow Alarm (Henry Jensen) > 6. Re: Re: PTSN POTS Differences SOLVED (Robert Keller) > 7. Re: Can you comment on this Qos script? How does one shape > RTP? (Sean Kennedy) > 8. Interface bonding + asterisk (Jesus Mogollon) > 9. Re: Can you comment on this Qos script? How doesone shape > RTP? (Henry) > 10. Re: Can you comment on this Qos script? How does one shape > RTP? (trixter http://www.0xdecafbad.com) > 11. RE: Sangoma A101 + Rhino channelbank (mattf) > 12. Re: Can you comment on this Qos script? How does one shape > RTP? (Andrew Kohlsmith) > 13. Re: TDM400P power supply (Ricardo Peironcely) > 14. Problem with X101P (Yusuf Iqbal) > 15. Re: Can you comment on this Qos script? How does one shape > RTP? (trixter http://www.0xdecafbad.com) > 16. wcfxo problem (Dave Weis) > 17. (no subject) (Robert Webb) > 18. Re: Sipura SPA-841 Phone Review (Doug Millsaps) > 19. Re: From OH323 to SIP or OH323 without gatekeeper (Bruno Hertz) > 20. Re: wcfxo problem (Sahil Gupta) > 21. Re: TDM400P Revision question. (Robert Webb) > 22. Intercom with Aastra 480e? (Bobby Lacey) > 23. Manipulate Asterisk Database from manager? (Matt) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 11 Apr 2005 08:39:00 -0500 > From: Eric Wieling <eric@fnords.org> > Subject: Re: [Asterisk-Users] append # to dial string > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <425A7DF4.3060306@fnords.org> > Content-Type: text/plain; charset=us-ascii; format=flowed > > John Breeden wrote: > > > Been there, done that - no joy :-) > > > > It appears the modifier only excepts a numeric, anyone know if/how you > > can feed it adecimal/hex for ascii #? > > > > Rich Adamson wrote: > > > >>> Is there anyway to append the '#' symbol to a dial string? - > >>> hex/octal whatever? I'm surprised that I can't find anything > >>> searching the wiki or google. > >>> > >> > >> > >> Try something like this: > >> > >> exten => _9XXXXXXX,1,Dial(Zap/4/${EXTEN}#) > > Then you are doing something wrong. The above syntax is correct. > > -- > Always do right. This will gratify some people and astonish the rest. > Mark Twain > > > ------------------------------ > > Message: 2 > Date: Mon, 11 Apr 2005 08:40:53 -0500 > From: Eric Wieling <eric@fnords.org> > Subject: Re: [Asterisk-Users] VAD/DTX implementation through zaptel > cards > To: parijat@varaha.com, Asterisk Users Mailing List - Non-Commercial > Discussion <asterisk-users@lists.digium.com> > Message-ID: <425A7E65.2040400@fnords.org> > Content-Type: text/plain; charset=us-ascii; format=flowed > > parijat@varaha.com wrote: > > > Hi, > > How can i implement VAD/DTX using zaptel with asterisk towards PSTN. > > TDM (PSTN/telcos) do not support VAD. The entire idea of VAD is not > even a valid idea. > > > ------------------------------ > > Message: 3 > Date: Mon, 11 Apr 2005 08:44:00 -0500 > From: Eric Wieling <eric@fnords.org> > Subject: Re: [Asterisk-Users] CDR and TDS > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <425A7F20.9070701@fnords.org> > Content-Type: text/plain; charset=us-ascii; format=flowed > > David Masure wrote: > > > > > Hi, > > > > I want to use the cdr to record the call log to my Microsoft SQL Server > > using unixodbc and freetds.... > > > > but when I compile, I've got this message.... > > > > Does anyone have the same problem and/or know how to solve it ? > > > Update of /usr/cvsroot/asterisk/doc > In directory mongoose.digium.com:/tmp/cvs-serv24936/doc > > Added Files: > README.tds > Log Message: > Add documentation for TDS noting compilation problem on 0.63+ > > > --- NEW FILE: README.tds --- > PLEASE NOTE > > The cdr_tds module is NOT compatible with version 0.63 of FreeTDS. > > The cdr_tds module is known to work with FreeTDS version 0.62.1; > it should also work with 0.62.2, 0.62.3 and 0.62.4, which are bug > fix releases. > > The cdr_tds module uses the raw "libtds" API of FreeTDS. It appears > that from 0.63 onwards, this is not considered a published API > of FreeTDS and is subject to change without notice. > > Between 0.62.x and 0.63 of FreeTDS, many incompatible changes > have been made to the libtds API. > > For newer versions of FreeTDS, it is recommended that you use the > ODBC driver. > > > > -- > Always do right. This will gratify some people and astonish the rest. > Mark Twain > > > ------------------------------ > > Message: 4 > Date: Mon, 11 Apr 2005 09:45:54 -0400 > From: vgrskovic@optonline.net > Subject: RE: [Asterisk-Users] Zaptel Compile on a virtual dedicated > host. > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > <asterisk-users@lists.digium.com> > Message-ID: <007601c53e9c$c85a2b60$0302a8c0@zeus> > Content-Type: text/plain; charset="us-ascii" > > It appears to be Virtuozzo.. > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Henry > Sent: Monday, April 11, 2005 9:34 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Zaptel Compile on a virtual dedicated > host. > > Hi, > > Do you happen to know what VPS system your host uses (e.g. UML, > Virtuozzo, VMWare, FreeVPS, etc.)? It could make a lot of difference, as > some platforms will allow changes that others will not. > > -- Henry Owens. > > > On 11/4/05 2:20 pm, "vgrskovic@optonline.net" <vgrskovic@optonline.net> > wrote: > Giles thank you for getting back so quickly, "dmesg" doesn't output > anything, but even if it did, I am not sure that I could recompile the > kernel. > > The server I am using is in a virtual dedicated hosting environment, I > do not have access to recompile the kernel, nor can I replace it. The > server prevents me from doing so. I do not have access to the "real" > /boot and don't have access as far as I can tell to the .config for the > kernel source. ("make oldconfig" seems to work) > > After a few more days of tech support, google searches and etc, I have > found that my provider is using kernel 2.24.21.4.0.1.elsmp. Of course, > cat /proc/version doesn't think so!! It thinks I am running Kernel > 2.4.20-021stab022.11.777-enterprise. I am able to use rpmfind to source > the corresponding rpm which installs without incident. The interesting > part is "rpm -qa kernel" doesn't see it :-(. I even tried to "rpm > -rebuilddb" > > Zaptel appears to compile fine, but when I run "modprobe zaptel" I get > the following: > > ----> > /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o: > kernel-module version mismatch > /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o > was compiled for kernel version 2.4.21-4.0.1.EL > while this kernel is version 2.4.20-021stab022.11.777-enterp. > /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o: insmod > /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o failed > /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o: insmod > zaptel failed > <--- > > Is there a way to override zaptel's kernel check or have linux fool it > into thinking the kernel is 2.4.21-4.0.1.EL? > > thanks! > > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] > <mailto:asterisk-users-bounces@lists.digium.com%5d> On Behalf Of Giles > Coochey > Sent: Wednesday, April 06, 2005 9:01 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: > > > >Anyone have any ideas on where I can find the right kernel source? I > have look at > > rpmfind.net and google'd with no avail! > > You could always download the Vanilla kernel source from > http://www.kernel.org and compile a kernel from source. I tend to always > use the Vanilla source, it's what everything has been tested against and > it tastes better. > > You should probably print out the "dmesg" output to help you configure > the kernel options prior to compilation so that your "hardware" is > correctly detected. > > I would also urge you to use a bootloader such as grub or lilo to ensure > that you can revert to the original kernel should it panic on boot, I > suspect Redhat already uses one of those anyway. > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _____ > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20050411/a0f7f809/attachment-0001.htm> > ------------------------------ > > Message: 5 > Date: Mon, 11 Apr 2005 15:54:44 +0200 > From: Henry Jensen <hjensen@gmx.de> > Subject: Re: [Asterisk-Users] TE110P/Hipath3750 - Yellow Alarm > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <20050411135444.GA1282@jensen.local> > Content-Type: text/plain; charset=iso-8859-15 > > > On Tue, Apr 05, 2005 at 09:06:33PM +0200, Peter Svensson wrote: > > A yellow alarm means the remote end is sensing some error condition. Try > > looking for an error message at the remote end. It may be as easy as a > > broken cable (where the Hipath does not hear the Asterisk box). > > The problem is, that the TMS2-Card in the HiPath is not activated, > it says, that the line is dead. According to the Siemens-People the > Card should activate itself as soon as a signal reaches the card. > But it appears, that Asterisk sends no signal. > > > This is what the layout looks like: > > Asterisk|TE110P - TMS2|HiPath|TMS2 - PSTN > > > The cable is functional and the wiring is correct. But I'm not sure how I > must configure the TMS2 card. > > Regards, > Henry > > > > > > > ------------------------------ > > Message: 6 > Date: Mon, 11 Apr 2005 07:01:03 -0700 > From: Robert Keller <rkeller@ferndale.wednet.edu> > Subject: Re: [Asterisk-Users] Re: PTSN POTS Differences SOLVED > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <BE7FD12F.1A2CD%rkeller@ferndale.wednet.edu> > Content-Type: text/plain; charset="US-ASCII" > > Tony, I don't see "${EXTEN}" anywhere in the [macro-dialout-trunk] context. > Am I missing something? > > Robert Andrew Keller > Ferndale School District #502 > rkeller@ferndale.wednet.edu > 360-383-9228 PH. > 360-383-9218 FAX > "Paving the way for tomorrows genius." > > > From: tony@softins.clara.co.uk (Tony Mountifield) > > Organization: Software Insight Ltd., Winchester, UK > > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > > <asterisk-users@lists.digium.com> > > Date: Mon, 11 Apr 2005 07:48:19 +0000 (UTC) > > To: asterisk-users@lists.digium.com > > Subject: [Asterisk-Users] Re: PTSN POTS Differences SOLVED > > > > In article <BE7F361F.1A279%rkeller@ferndale.wednet.edu>, > > Robert Keller <rkeller@ferndale.wednet.edu> wrote: > >> Thanks Rich, I wasn't sure where to find that context. I found the > outbound > >> context in the extensions_additional.conf and added w's in the following > >> manner: > >> > >> [outrt-001-Out1] > >> include => outrt-001-Out1-custom > >> exten => _1NXXNXXXXXX,1,Macro(dialout-trunk,1,w${EXTEN}) > >> exten => _1NXXNXXXXXX,2,Macro(outisbusy) ; No available circuits > >> exten => _9.,1,Macro(dialout-trunk,1,w${EXTEN:1}) > >> exten => _9.,2,Macro(outisbusy) ; No available circuits > >> exten => _NXXNXXXXXX,1,Macro(dialout-trunk,1,w${EXTEN}) > >> exten => _NXXNXXXXXX,2,Macro(outisbusy) ; No available circuits > >> exten => _NXXXXXX,1,Macro(dialout-trunk,1,w${EXTEN}) > >> exten => _NXXXXXX,2,Macro(outisbusy) ; No available circuits > > > > Couldn't you have just put the w in once, in the Dial command that > > is inside [macro-dialout-trunk] ? > > > > Cheers > > Tony > > -- > > Tony Mountifield > > Work: tony@softins.co.uk - http://www.softins.co.uk > > Play: tony@mountifield.org - http://tony.mountifield.org > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ------------------------------ > > Message: 7 > Date: Mon, 11 Apr 2005 07:08:54 -0700 > From: Sean Kennedy <skennedy@tpno-co.org> > Subject: Re: [Asterisk-Users] Can you comment on this Qos script? How > does one shape RTP? > To: asterisk-users@lists.digium.com > Message-ID: <425A84F6.1050005@tpno-co.org> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Honestly, the best script I've ever found is the wondershaper script ( > google it ). I tried the correct one posted in this thread, tried > modifying it, but in the end I just used wondershaper. > > Does a great job. My only fear is it doesn't specifically target IAX2 > traffic as high priority, but I can modify it later to do so if needed. > > On a 192 line I am able to get 4 ulaw ( IAX2 ) calls out with no > noticable problems. Along with someone streaming a shoutcast station ( > sigh ). The station broke up, but the calls didn't. > > cmisip wrote: > > >I got this from the voip wiki but the original script didn't seem to > >work right so I fiddled with it a little bit. I am no expert so maybe > >someone can look at it for errors. This is for my cable connection. So > >far asterisk seems to use 1:10 while all other traffic uses 1:102. How > >does one packet shape RTP? > > > >Thanks for any help. > > > > > ------------------------------ > > Message: 8 > Date: Mon, 11 Apr 2005 07:15:30 -0700 > From: Jesus Mogollon <gocho26@gmail.com> > Subject: [Asterisk-Users] Interface bonding + asterisk > To: asterisk-users@lists.digium.com > Message-ID: <d18206f705041107157958062f@mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi all > > I installed asterisk on a dual PIII 700 with two NICs. I then proceeded to > configure both NICs with bonding enable (bonding miimon=100 mode=1). I know > certain features (like load balancing) under a bonded configuration is not > understood by some switches, so I configured it using mode=1 (Failover > only). The problem I'm having is that, sometimes, calls start fine but then > one of the parties loses audio (it could be the caller of the callee who > loses audio, there is no pattern). I was wondering if someone has hit the > same wall as me. There are people using this server right now, so I haven't > tried the no-bonding option as it means downtime. Any help would be > appreciated. > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20050411/e427fb81/attachment-0001.htm> > ------------------------------ > > Message: 9 > Date: Mon, 11 Apr 2005 15:19:38 +0100 > From: Henry <henry@adiungo.com> > Subject: Re: [Asterisk-Users] Can you comment on this Qos script? How > doesone shape RTP? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <BE80460A.DE62%henry@adiungo.com> > Content-Type: text/plain; charset="US-ASCII" > > I agree that Wondershaper is a great script; prior to using it in an office > where I set up asterisk, there were some major problems with call quality, > but it seems to have helped hugely (the same DSL line is used for both VoIP > and everyday 'net usage for seven people - not ideal, but I didn't set the > budget :-) ). > > If you happen to modify it to to prioritize IAX2, drop me a copy! > > -- Henry Owens. > > > On 11/4/05 3:08 pm, "Sean Kennedy" <skennedy@tpno-co.org> wrote: > > > Honestly, the best script I've ever found is the wondershaper script ( > > google it ). I tried the correct one posted in this thread, tried > > modifying it, but in the end I just used wondershaper. > > > > Does a great job. My only fear is it doesn't specifically target IAX2 > > traffic as high priority, but I can modify it later to do so if needed. > > > > On a 192 line I am able to get 4 ulaw ( IAX2 ) calls out with no > > noticable problems. Along with someone streaming a shoutcast station ( > > sigh ). The station broke up, but the calls didn't. > > > > cmisip wrote: > > > >> I got this from the voip wiki but the original script didn't seem to > >> work right so I fiddled with it a little bit. I am no expert so maybe > >> someone can look at it for errors. This is for my cable connection. So > >> far asterisk seems to use 1:10 while all other traffic uses 1:102. How > >> does one packet shape RTP? > >> > >> Thanks for any help. > >> > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ------------------------------ > > Message: 10 > Date: Mon, 11 Apr 2005 07:21:43 -0700 > From: "trixter http://www.0xdecafbad.com" <trixter@0xdecafbad.com> > Subject: Re: [Asterisk-Users] Can you comment on this Qos script? How > does one shape RTP? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <1113229303.6647.14.camel@rufus.home.tld> > Content-Type: text/plain; charset="us-ascii" > > I used the one posted to this list and for a test did a > speedtest.dslreports.com bandwidth test duringa call, no loss in > quality. > > I set ports 10000-11024 to RTP in rtp.conf, I dont need 10k ports for > that as I have few calls being processed. I also added sip to the queue > although that prolly doesnt matter becuase its such a low bandwidth > protocol comparitevly speaking. > > > # udp/5060 is SIP > tc filter add dev $DSLDEV parent 1:0 protocol ip prio 1 u32 match ip > dport 506 > 0 0xffff match ip protocol 17 0xff flowid 1:0 > tc filter add dev $DSLDEV parent 1:0 protocol ip prio 2 u32 match ip > sport 506 > 0 0xffff match ip protocol 17 0xff flowid 1:0 > > # udp/10000-11024 is RTP > tc filter add dev $DSLDEV parent 1:0 protocol ip prio 1 u32 match ip > dport 100 > 00 0xf670 match ip protocol 17 0xff flowid 1:0 > tc filter add dev $DSLDEV parent 1:0 protocol ip prio 2 u32 match ip > sport 100 > 00 0xf670 match ip protocol 17 0xff flowid 1:0 > > > > On Mon, 2005-04-11 at 07:08 -0700, Sean Kennedy wrote: > > Honestly, the best script I've ever found is the wondershaper script ( > > google it ). I tried the correct one posted in this thread, tried > > modifying it, but in the end I just used wondershaper. > > > > Does a great job. My only fear is it doesn't specifically target IAX2 > > traffic as high priority, but I can modify it later to do so if needed. > > > > On a 192 line I am able to get 4 ulaw ( IAX2 ) calls out with no > > noticable problems. Along with someone streaming a shoutcast station ( > > sigh ). The station broke up, but the calls didn't. > > > > cmisip wrote: > > > > >I got this from the voip wiki but the original script didn't seem to > > >work right so I fiddled with it a little bit. I am no expert so maybe > > >someone can look at it for errors. This is for my cable connection. So > > >far asterisk seems to use 1:10 while all other traffic uses 1:102. How > > >does one packet shape RTP? > > > > > >Thanks for any help. > > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > Trixter http://www.0xdecafbad.com > -------------- next part -------------- > A non-text attachment was scrubbed... > Name: not available > Type: application/pgp-signature > Size: 189 bytes > Desc: This is a digitally signed message part > Url : > http://lists.digium.com/pipermail/asterisk-users/attachments/20050411/b94c9559/attachment-0001.pgp> > ------------------------------ > > Message: 11 > Date: Mon, 11 Apr 2005 10:34:51 -0400 > From: mattf <mattf@vicimarketing.com> > Subject: RE: [Asterisk-Users] Sangoma A101 + Rhino channelbank > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > <asterisk-users@lists.digium.com> > Message-ID: ><DB43F516702AAF4392AA45573F18181901267879@vicimail.vicimarketinggroup.co m>> > Content-Type: text/plain; charset="iso-8859-1" > > Keep on bugging the Sangoma guys, I know they are working on several RBS T1 > issues right now(They called me Friday to go over a few things) They just > need help from users like you and I to find the bugs in their drivers. > > Have you tried any other signalling types other than LOOP? > > MATT--- > > > -----Original Message----- > From: Felician CHELU [mailto:tehnic@intertel.ro] > Sent: Monday, April 11, 2005 9:52 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Sangoma A101 + Rhino channelbank > > > Hello, > > I have Asterisk 1.0.6 - I try to setup Sangoma A101 T1 board together with > the Rhino fxs chanelbank. > Things done: > - T1 cross cable = I have carrier, signalling and framnig leds on > the channelbank green. > - channelbank configuration: > t1 - Proto: LOOP Frame: esf Clock: slave Coding: > b8zs > channels(analog) : Function:A-fxs Mode:loop > - zaptel.conf > span=2,1,0,esf,b8zs > fxols=32-55 > (i have a span 1 with a digium e1) > - zapata.conf > .... signalling=fxo_ls > - wanpipe1.conf > > [devices] > wanpipe1 = WAN_AFT, Comment > > [interfaces] > w1g1 = wanpipe1, , TDM_VOICE, Comment > > [wanpipe1] > CARD_TYPE = AFT > S514CPU = A > CommPort = PRI > AUTO_PCISLOT = NO > PCISLOT = 10 > PCIBUS = 2 > FE_MEDIA = T1 > FE_LCODE = B8ZS > FE_FRAME = ESF > FE_LINE = 1 > TE_CLOCK = MASTER > ACTIVE_CH = ALL > TE_HIGHIMPEDANCE = NO > LBO = 0DB > INTERFACE = V35 > CLOCKING = EXTERNAL > BaudRate = 0 > MTU = 1500 > UDPPORT = 9000 > TTL = 255 > IGNORE_FRONT_END = NO > > [w1g1] > PROTOCOL = HDLC > HDLC_STREAMING = YES > ACTIVE_CH = ALL > IDLE_FLAG = 0x7E > MTU = 1500 > MRU = 1500 > TDMV_SPAN = 2 > TDMV_ECHO_OFF = NO > MULTICAST = NO > TRUE_ENCODING_TYPE = NO > > > I already called Sangoma and Rhino support, but after hours of long distance > call conversation the problem is still not solved. Finnaly, a guy from Rhino > told me that their "asterisk expert" (which was not avaliable) knows about > this problem and that it is that the sangoma driver is not communicating > with asterisk. > > The wanrouter starts ok, after ztcfg I see the channels configured. > The problem: i don't have dialtone on phones. > > Question: When i enter zttoll, if i go to the sangoma span and I make "loop" > then it freezes. Is it normal? > > If someone has experienced this combination and made it work please give me > a sign. > > Thank you. > > PS: > > Felician > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ------------------------------ > > Message: 12 > Date: Mon, 11 Apr 2005 10:32:26 -0400 > From: Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com> > Subject: Re: [Asterisk-Users] Can you comment on this Qos script? How > does one shape RTP? > To: asterisk-users@lists.digium.com > Message-ID: <200504111032.27090.akohlsmith-asterisk@benshaw.com> > Content-Type: text/plain; charset="iso-8859-1" > > On April 11, 2005 10:08 am, Sean Kennedy wrote: > > Honestly, the best script I've ever found is the wondershaper script ( > > google it ). I tried the correct one posted in this thread, tried > > modifying it, but in the end I just used wondershaper. > > :-) I started out with wshaper and just didn't like it, which is where rc.tc > > came from. > > -A. > > > ------------------------------ > > Message: 13 > Date: Mon, 11 Apr 2005 16:48:27 +0200 > From: Ricardo Peironcely <rpr_listas@telefonica.net> > Subject: Re: [Asterisk-Users] TDM400P power supply > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <425A8E3B.2010104@telefonica.net> > Content-Type: text/plain; charset="iso-8859-1" > > Thanks, > > I will try with external power supply. > > Rpr > > Rich Adamson escribi?: > > >>I've a problem with a TDM400P digium card. > >> > >>My box has no molex connectors for power supply. Simply has no any power > >>connector, because is not a normal PC) And I need to know if i can use a > >>external supply. But I've several questions: > >> > >>1.- Are both circuits (PCI-power and Phone-line-power) electrically > >>separated? > >>2.- A little voltage difference can create an undesired internal current? > >>3.- What are the current needs for this supply? > >> > >>I need the power supply because I want to use both FXS and FXO ports. > >>And I can't use a Y cable, because I've no molex connectors. > >> > >> > > > >Been discussed several times before and you should have found the > >answer using google. > > > >The TDM connector is only used for the fxs modules, and then only the > >+12 volt lead on that connector (and ground) is actually wired to > >anything on the TDM board. So, there is no conflict with internal > >system voltages. > > > >Yes you can use an external 12 volt power supply. > > > >The 12 volts is only used on the card to generate ringing voltage to > >the fxs modules. No ringing, no significant current draw. Just about > >any 12 volt supply should do, however I think I'd be looking for > >one that is at least somewhat regulated. No other idea on the power > >supply specs. > > > > > >_______________________________________________ > >Asterisk-Users mailing list > >Asterisk-Users@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20050411/aa53ab12/attachment-0001.htm> > ------------------------------ > > Message: 14 > Date: Mon, 11 Apr 2005 20:52:03 +0600 > From: "Yusuf Iqbal" <yusii_bd@hotmail.com> > Subject: [Asterisk-Users] Problem with X101P > To: asterisk-users@lists.digium.com > Message-ID: <BAY15-F4152F83923D0C75DB42882E9320@phx.gbl> > Content-Type: text/plain; charset="us-ascii" > > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20050411/4930a523/attachment-0001.htm> > ------------------------------ > > Message: 15 > Date: Mon, 11 Apr 2005 07:52:07 -0700 > From: "trixter http://www.0xdecafbad.com" <trixter@0xdecafbad.com> > Subject: Re: [Asterisk-Users] Can you comment on this Qos script? How > does one shape RTP? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <1113231127.6650.16.camel@rufus.home.tld> > Content-Type: text/plain; charset="us-ascii" > > On Mon, 2005-04-11 at 10:32 -0400, Andrew Kohlsmith wrote: > > On April 11, 2005 10:08 am, Sean Kennedy wrote: > > > Honestly, the best script I've ever found is the wondershaper script ( > > > google it ). I tried the correct one posted in this thread, tried > > > modifying it, but in the end I just used wondershaper. > > > > :-) I started out with wshaper and just didn't like it, which is where > rc.tc > > came from. > you may want to pull at least the RTP lines I just posted and add them > to your rc.tc since that is what I got and tweaked since I use RTP :) > > -- > Trixter http://www.0xdecafbad.com > -------------- next part -------------- > A non-text attachment was scrubbed... > Name: not available > Type: application/pgp-signature > Size: 189 bytes > Desc: This is a digitally signed message part > Url : > http://lists.digium.com/pipermail/asterisk-users/attachments/20050411/c5e30ef3/attachment-0001.pgp> > ------------------------------ > > Message: 16 > Date: Mon, 11 Apr 2005 09:49:17 -0500 (CDT) > From: Dave Weis <djweis@sjdjweis.com> > Subject: [Asterisk-Users] wcfxo problem > To: asterisk-users@lists.digium.com > Message-ID: > <Pine.LNX.4.62.0504110947390.27314@charmed.internetsolver.com> > Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed > > > I've got a X100P in a compaq proliant 3000. My system stops taking calls > and making calls. I had been getting the FXO PCI Master abort before > updating, I am now running a cvs head checkout from a week or so ago. Now > I still have the problem but get more error messages: > > Found a Wildcard FXO: Wildcard X101P > Registered tone zone 0 (United States / North America) > Registered tone zone 0 (United States / North America) > FXO PCI Master abort > wcfxo: Out of space to write register 05 with 02 > wcfxo: Out of space to write register 05 with 03 > wcfxo: Out of space to write register 05 with 0a > wcfxo: Out of space to write register 05 with 0a > wcfxo: Out of space to write register 05 with 0a > wcfxo: Out of space to write register 05 with 0a > > Any solution? > > -- > Dave Weis "I believe there are more instances of the abridgment > djweis@sjdjweis.com of the freedom of the people by gradual and silent > encroachments of those in power than by violent > and sudden usurpations."- James Madison > > > ------------------------------ > > Message: 17 > Date: Mon, 11 Apr 2005 10:54:30 -0400 > From: "Robert Webb" <asterisk@ropeguru.com> > Subject: [Asterisk-Users] (no subject) > To: Asterisk-Users@lists.digium.com > Message-ID: <web-1103722@ropeguru.com> > Content-Type: text/plain; charset="ISO-8859-1"; format="flowed" > > > Good morning all.. > > I was following a discussion on this list about the > TDM400P revisions. It is my understanding that the current > revision that one should have is the Rev. H and not the > E/F. I have not yet been able to verify the rev stamped on > the board, but zaptel is reporting that I have the Rev. > E/F. I just bought this card in January direct from Digium > and was wondering if I got the wrong Rev. somehow?? I have > been having some intermittent problems but only thought it > was my setup. > > > > ------------------------------ > > Message: 18 > Date: Mon, 11 Apr 2005 09:57:16 -0500 > From: Doug Millsaps <asterisk@txpe.net> > Subject: Re: [Asterisk-Users] Sipura SPA-841 Phone Review > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <6.1.2.0.2.20050411095336.033c5b20@mail.txpe.net> > Content-Type: text/plain; charset="us-ascii"; format=flowed > > I use a headset w/out any problems, except for if my cell phone is close by > and rings. Otherwise, volume is ok and no humming. Could it be your > headset? > > At 01:56 PM 4/10/2005, you wrote: > > >Just make sure you don't have a cordless or cell phone near by or the > >headset jack will "receive" a considerable amount of interference into > >your conversation (when NOT using a headset). > > > >Also don't even try using a headset... volume is low and there is a loud > >humming noise. > > > > ------------------------------ > > Message: 19 > Date: Mon, 11 Apr 2005 17:03:32 +0200 > From: "Bruno Hertz" <brrhtz@yahoo.de> > Subject: Re: [Asterisk-Users] From OH323 to SIP or OH323 without > gatekeeper > To: asterisk-users@lists.digium.com > Message-ID: <m3is2ttkxn.fsf@caruso.quasi.local> > Content-Type: text/plain; charset=us-ascii > > "Joe S" <printingfoot@hotmail.com> writes: > > > Hi, > > > > I am new with asterisk. I was wondering if there is a way to call a > > OH323 user or SIP user using Netmeeting/SJPhone with H323 as the > > default protocol without having a gatekeeper. > > > > I can make a call from SIP to OH323 by specifying it in the > > extensions.conf file, like: > > > > exten=>1001, 1, Dial(OH323/10.10.10.1) > > > > so I was wondering if there was a way to call from OH323 to SIP or OH323. > > Sure. Just specify in oh323.conf the context where incoming calls > should go. That context then can include dial statements for any > protocol, SIP, H323, IAX, whatever. See the Wiki for details on how to > setup dial plans. > > Finally, instruct your H323 phone to use asterisk as a gateway > resp. proxy, not a gatekeeper. Any calls will then go through > asterisk, and to the context you specified. > > I'm doing that with Gnomemeeting all the time, and it works without > problems. > > Regards, Bruno. > > > > ------------------------------ > > Message: 20 > Date: Tue, 12 Apr 2005 01:03:51 +1000 (EST) > From: Sahil Gupta <sgupta@voicevalley.com.au> > Subject: Re: [Asterisk-Users] wcfxo problem > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <Pine.LNX.4.60.0504120103280.6587@asterisk.in.com.au> > Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed > > I'm having similar issues using an X100P Ambient Chipset Clone Card.... > any ideas? > > Regards, > > > Sahil Gupta > VoiceValley > > On Mon, 11 Apr 2005, Dave Weis wrote: > > > > > I've got a X100P in a compaq proliant 3000. My system stops taking calls > and > > making calls. I had been getting the FXO PCI Master abort before updating, > I > > am now running a cvs head checkout from a week or so ago. Now I still have > > > the problem but get more error messages: > > > > Found a Wildcard FXO: Wildcard X101P > > Registered tone zone 0 (United States / North America) > > Registered tone zone 0 (United States / North America) > > FXO PCI Master abort > > wcfxo: Out of space to write register 05 with 02 > > wcfxo: Out of space to write register 05 with 03 > > wcfxo: Out of space to write register 05 with 0a > > wcfxo: Out of space to write register 05 with 0a > > wcfxo: Out of space to write register 05 with 0a > > wcfxo: Out of space to write register 05 with 0a > > > > Any solution? > > > > -- > > Dave Weis "I believe there are more instances of the > abridgment > > djweis@sjdjweis.com of the freedom of the people by gradual and silent > > encroachments of those in power than by violent > > and sudden usurpations."- James Madison > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ------------------------------ > > Message: 21 > Date: Mon, 11 Apr 2005 11:11:45 -0400 > From: "Robert Webb" <asterisk@ropeguru.com> > Subject: Re: [Asterisk-Users] TDM400P Revision question. > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <web-1103721@ropeguru.com> > Content-Type: text/plain; charset="ISO-8859-1"; format="flowed" > > Sorry for the initial no subject line. Was in a hurry when > I typed this and somehow missed putting it in. > > Please accept my apologies.... > > On Mon, 11 Apr 2005 10:54:30 -0400 > "Robert Webb" <asterisk@ropeguru.com> wrote: > > > > Good morning all.. > > > > I was following a discussion on this list about the > >TDM400P revisions. It is my understanding that the > >current revision that one should have is the Rev. H and > >not the E/F. I have not yet been able to verify the rev > >stamped on the board, but zaptel is reporting that I have > >the Rev. E/F. I just bought this card in January direct > >from Digium and was wondering if I got the wrong Rev. > >somehow?? I have been having some intermittent problems > >but only thought it was my setup. > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ------------------------------ > > Message: 22 > Date: Mon, 11 Apr 2005 11:13:45 -0400 > From: "Bobby Lacey" <asterisk@duaneallman.net> > Subject: [Asterisk-Users] Intercom with Aastra 480e? > To: <asterisk-users@lists.digium.com> > Message-ID: <006e01c53ea9$0dfe4550$0e00000a@rosehill> > Content-Type: text/plain; charset="us-ascii" > > Hello list, > > I have been successful in setting up my first * box with a pair of > x100p's, Cisco 7960, and a Digium iAXy. > > I would like to incorporate an Aastra 480e using my iAXy and ADSI. I > want to be able to answer phone calls with my 7960 in the back of the > house and park the call, then in turn call the intercom on the 480e in > the front (using two way audio) to announce that there is a call that > needs to be picked up on 701. > > Also, by using the Aastra 480e, can I see my Zap line status to see what > lines are available and also if extensions are in use? > > Thanks in advance. > > B. Lacey > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20050411/2f2fcc73/attachment-0001.htm> > ------------------------------ > > Message: 23 > Date: Mon, 11 Apr 2005 11:16:56 -0400 > From: Matt <mhoppes@gmail.com> > Subject: [Asterisk-Users] Manipulate Asterisk Database from manager? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <c11d025305041108165384be0d@mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > Hi, > Is there anyway to manipulate the asterisk internal database from the > manager (the one you can telnet to)? And if so.. how does one do it? > (ie for enabling call forwarding, etc) > > > ------------------------------ > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > > End of Asterisk-Users Digest, Vol 9, Issue 93 > ********************************************* >
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
asterisk-users-request@lists.digium.com
Sent: Thursday, January 19, 2006 12:00 PM
To: asterisk-users@lists.digium.com
Subject: Asterisk-Users Digest, Vol 18, Issue 121
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Today's Topics:
1. Re: DTMF # ? (Mojo with Horan & Company, LLC)
2. Re: Brief silences during calls (Mojo with Horan & Company, LLC)
3. Problem with rxfax - Dropping incompatible voice frame?
(Micha?l Gaudette)
4. transfer and zap (Marcel Pennewi?)
5. Sound issue with Asterisk (Kevin)
6. Re: Brief silences during calls (Rob Lith)
7. Re: MeetMe Listen Only flag (|m) (Tony Mountifield)
8. Re: SAN Devices (Jared Watkins)
9. Disabling zap echo cancellor from dialplan (Massimo De Nadal)
----------------------------------------------------------------------
Message: 1
Date: Thu, 19 Jan 2006 08:22:22 -0900
From: "Mojo with Horan & Company, LLC"
<mojo@horanappraisals.com>
Subject: Re: [Asterisk-Users] DTMF # ?
To: chris.songer@getblaze.com, Asterisk Users Mailing List -
Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <43CFCACE.1090800@horanappraisals.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
It's mapped to blind transfer in features.conf -- If you want to use the
blind transfer feature, which I find easier than my phones' transfer
features, remap it to ## in features.conf. That way if you hit # it
dtmfs through to the target IVR, but you can hit ## real quick to get
the transfer function.
Or maybe I could refer you to the notes in the wiki: ;)
http://www.voip-info.org/wiki-Asterisk+config+features.conf
Using the blindxfer in [featuremap] section you can redefine the
transfer key. For example, if the blindxfer is set to "##", transfer
only happens when you press the "#" key twice very quickly. This
solves
a problem using Asterisk phones to call IVR systems such as those used
by banks and credit card companies - "Enter you account number followed
by the # key".
Moj
chris songer wrote:> Can the # be used as a valid key press for a user in a dial plan?
> if so how can the asterisk recognize it as a valid key press?
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Mojo <mojo@horanappraisals.com>
Office Manger, Horan & Company, LLC
(907) 747-6666 x112
------------------------------
Message: 2
Date: Thu, 19 Jan 2006 08:31:11 -0900
From: "Mojo with Horan & Company, LLC"
<mojo@horanappraisals.com>
Subject: Re: [Asterisk-Users] Brief silences during calls
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <43CFCCDF.5090205@horanappraisals.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
check irq supply on the * server -- When you run zttest do you maintain
over 98%? or like Steve suggested, the network may have congestion or
other errors ethereal may help you figure out.
I had a polycom 500 that was doing this to my user, 301s and 501s
wouldn't do it. Not sure if that was network issues or something with
the polycom itself.
Moj
Mimmus wrote:> Where can I investigate the origin of brief silences during calls
from/to my> SIP phone?
> Only rare pauses of 0.5-1.0 sec when I'm not able to hear anything.
>
> Thanks
> Mimmus
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Mojo <mojo@horanappraisals.com>
Office Manger, Horan & Company, LLC
(907) 747-6666 x112
------------------------------
Message: 3
Date: Thu, 19 Jan 2006 12:40:26 -0500
From: Micha?l Gaudette <michael.gaudette@virtutel.ca>
Subject: [Asterisk-Users] Problem with rxfax - Dropping incompatible
voice frame?
To: asterisk-users@lists.digium.com
Message-ID: <003201c61d1f$6ea88f50$0a01a8c0@mike>
Content-Type: text/plain; charset=iso-8859-1
Hi,
I'm having problems with the rxFax app. One of the messages that appear
in
my console is:
Executing Set("SIP/something",
"FAXFILE=/var/spool/asterisk-fax/1137692307.5.tif") in new stack
-- Executing RxFAX("SIP/something",
"/var/spool/asterisk-fax/1137692307.5.tif") in new stack
Jan 19 12:38:30 NOTICE[12008]: channel.c:1906 ast_read: Dropping
incompatible voice frame on SIP/something of format slin since our
native
format has changed to ulaw
"Dropping incompatible voice frame on SIP/something of format slin since
our
native format has changed to ulaw"
This seemed particularly important, but I can't really say why....Could
this
be why my faxes are often interrupting during transmission and giving me
errors on my PSTN fax machine that is used for sending the fax?
Mike
------------------------------
Message: 4
Date: Thu, 19 Jan 2006 18:40:19 +0100
From: Marcel Pennewi? <mape2k@gmail.com>
Subject: [Asterisk-Users] transfer and zap
To: asterisk-users@lists.digium.com
Message-ID: <1445813335.20060119184019@gmail.com>
Content-Type: text/plain; charset=ISO-8859-15
Hello,
some problems with transfer and zap...
one hfc-card in NT mode and one fritz isdn-card in server.
there is one gigaset SX353 isdn phone on the hfc-card.
anybody calls from external via capi and the call is bridged
to the zap-device. if you want to transfer the call via R-button on
the isdn-phone the caller get the music-on-hold. you get a dialtone
and dial - if the called person gets on phone - i will hang up the
phone. but the call did'nt transfer - the moh to the first caller
will not stop. how can i transfer the call?
i want to transfer back the origin call to the asterisk-server in an
extension for faxtransfer, so if anyone call and a faxtone is there i
want transfer it back, so that asterisk answer the call.
any ideas?
sorry for my bad english ;-)
Marcel Pennewiss
------------------------------
Message: 5
Date: Thu, 19 Jan 2006 12:45:50 -0500
From: Kevin <kevin@locker70.com>
Subject: [Asterisk-Users] Sound issue with Asterisk
To: asterisk-users@lists.digium.com
Cc: steve@daviesfam.org
Message-ID: <43CFD04E.8080608@locker70.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hey Steve and everyone,
I looked at the configuration, and unless I am missing something I don't
think they are configured
# ztcfg -vv
Zaptel Configuration
=====================Channel map:
0 channels configured.
In the zapata.conf file, it is the sample version, but I didn't notice
anything in there that related to what you said. Or is it in a
different file or location?
I am in the office now so I am able to provide some more information
about the issue that I am having.
Here is the kernel if this helps Fedora core 4 -- 2.6.11-1.1369_FC4smp
I know that ztdummy is at least loaded now. Also as stated before there
is nothing plugged into the T1 card. So I wasn't sure if that was
causing a problem or not which is why I enabled ztdummy but it was not
the first time I e-mailed you.
# lsmod | grep ztdummy
ztdummy 7748 0
zaptel 192516 6 ztdummy,wct4xxp
If I look at the connections from tcpdump, I see my phone call coming
in, but no traffic is being sent back to the phone. With an Echo() test,
I see the traffic going back and forth, but when I call into a menu,
then there is nothing.
Thanks,
Kevin
I ran a sip debug as well but I felt it was better at the end of the
e-mail:
<-- SIP read from 64.7.189.14:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.7.161.26:5060;branch=z9hG4bK2b87d1ab;rport
From: "asterisk" <sip:asterisk@64.7.161.26>;tag=as4a36d77b
To: <sip:budgeTone-PubIP@64.7.189.14>;tag=a0efbf44ecab5900
Call-ID: 286e70e60596cc1b34a1fcac4e4c5337@64.7.161.26
CSeq: 102 OPTIONS
User-Agent: Grandstream BT100 1.0.6.7
Contact: <sip:budgeTone-PubIP@64.7.189.14>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Supported: replaces
Content-Length: 0
--- (11 headers 0 lines)---
Destroying call '286e70e60596cc1b34a1fcac4e4c5337@64.7.161.26'
<-- SIP read from 64.7.189.14:5060:
INVITE sip:500@64.7.161.26 SIP/2.0
Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bKa4b619218b6ad43c
From: "Budge Tone"
<sip:budgeTone-PubIP@64.7.161.26>;tag=b72941c93fe74588
To: <sip:500@64.7.161.26>
Contact: <sip:budgeTone-PubIP@64.7.189.14>
Supported: replaces
Call-ID: 2143389df4877360@64.7.189.14
CSeq: 19606 INVITE
User-Agent: Grandstream BT100 1.0.6.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 337
v=0
o=budgeTone-PubIP 8000 8000 IN IP4 64.7.189.14
s=SIP Call
c=IN IP4 64.7.189.14
t=0 0
m=audio 5004 RTP/AVP 2 8 4 18 15 97 9
a=sendrecv
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:15 G728/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/16000
a=ptime:20
--- (13 headers 16 lines)---
Using INVITE request as basis request - 2143389df4877360@64.7.189.14
Sending to 64.7.189.14 : 5060 (non-NAT)
Found peer 'budgeTone-PubIP'
Reliably Transmitting (no NAT) to 64.7.189.14:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
64.7.189.14;branch=z9hG4bKa4b619218b6ad43c;received=64.7.189.14
From: "Budge Tone"
<sip:budgeTone-PubIP@64.7.161.26>;tag=b72941c93fe74588
To: <sip:500@64.7.161.26>;tag=as6f00184d
Call-ID: 2143389df4877360@64.7.189.14
CSeq: 19606 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:500@64.7.161.26>
Proxy-Authenticate: Digest realm="asterisk",
nonce="351ca5f6"
Content-Length: 0
---
Scheduling destruction of call '2143389df4877360@64.7.189.14' in 15000
ms
<-- SIP read from 64.7.189.14:5060:
ACK sip:500@64.7.161.26 SIP/2.0
Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bKa4b619218b6ad43c
From: "Budge Tone"
<sip:budgeTone-PubIP@64.7.161.26>;tag=b72941c93fe74588
To: <sip:500@64.7.161.26>;tag=as6f00184d
Contact: <sip:budgeTone-PubIP@64.7.189.14>
Call-ID: 2143389df4877360@64.7.189.14
CSeq: 19606 ACK
User-Agent: Grandstream BT100 1.0.6.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
--- (11 headers 0 lines)---
<-- SIP read from 64.7.189.14:5060:
INVITE sip:500@64.7.161.26 SIP/2.0
Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bK1b2220ace977c3a7
From: "Budge Tone"
<sip:budgeTone-PubIP@64.7.161.26>;tag=b72941c93fe74588
To: <sip:500@64.7.161.26>
Contact: <sip:budgeTone-PubIP@64.7.189.14>
Supported: replaces
Proxy-Authorization: Digest username="budgeTone-PubIP",
realm="asterisk", algorithm=MD5, uri="sip:500@64.7.161.26",
nonce="351ca5f6",
response="3748b6120c7f4ecc4873cbdaf178d507"
Call-ID: 2143389df4877360@64.7.189.14
CSeq: 19607 INVITE
User-Agent: Grandstream BT100 1.0.6.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 337
v=0
o=budgeTone-PubIP 8000 8001 IN IP4 64.7.189.14
s=SIP Call
c=IN IP4 64.7.189.14
t=0 0
m=audio 5004 RTP/AVP 2 8 4 18 15 97 9
a=sendrecv
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:15 G728/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/16000
a=ptime:20
--- (14 headers 16 lines)---
Using INVITE request as basis request - 2143389df4877360@64.7.189.14
Sending to 64.7.189.14 : 5060 (non-NAT)
Found peer 'budgeTone-PubIP'
Found RTP audio format 2
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 15
Found RTP audio format 97
Found RTP audio format 9
Peer audio RTP is at port 64.7.189.14:5004
Found description format G726-32
Found description format PCMA
Found description format G723
Found description format G729
Found description format G728
Found description format iLBC
Found description format G722
Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x519
(g723|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x400 (ilbc)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0 (nothing)
Looking for 500 in local (domain 64.7.161.26)
list_route: hop: <sip:budgeTone-PubIP@64.7.189.14>
Transmitting (no NAT) to 64.7.189.14:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
64.7.189.14;branch=z9hG4bK1b2220ace977c3a7;received=64.7.189.14
From: "Budge Tone"
<sip:budgeTone-PubIP@64.7.161.26>;tag=b72941c93fe74588
To: <sip:500@64.7.161.26>
Call-ID: 2143389df4877360@64.7.189.14
CSeq: 19607 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:500@64.7.161.26>
Content-Length: 0
---
-- Executing Playback("SIP/budgeTone-PubIP-7e44",
"demo-abouttotry")
in new stack
We're at 64.7.161.26 port 13648
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x400 (ilbc) to SDP
Reliably Transmitting (no NAT) to 64.7.189.14:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
64.7.189.14;branch=z9hG4bK1b2220ace977c3a7;received=64.7.189.14
From: "Budge Tone"
<sip:budgeTone-PubIP@64.7.161.26>;tag=b72941c93fe74588
To: <sip:500@64.7.161.26>;tag=as324cfd6f
Call-ID: 2143389df4877360@64.7.189.14
CSeq: 19607 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:500@64.7.161.26>
Content-Type: application/sdp
Content-Length: 182
v=0
o=root 3026 3026 IN IP4 64.7.161.26
s=session
c=IN IP4 64.7.161.26
t=0 0
m=audio 13648 RTP/AVP 0 97
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=silenceSupp:off - - - -
---
-- Playing 'demo-abouttotry' (language 'en')
<-- SIP read from 64.7.189.14:5060:
ACK sip:500@64.7.161.26 SIP/2.0
Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bKb9dc124752ee52c9
From: "Budge Tone"
<sip:budgeTone-PubIP@64.7.161.26>;tag=b72941c93fe74588
To: <sip:500@64.7.161.26>;tag=as324cfd6f
Contact: <sip:budgeTone-PubIP@64.7.189.14>
Proxy-Authorization: Digest username="budgeTone-PubIP",
realm="asterisk", algorithm=MD5, uri="sip:500@64.7.161.26",
nonce="351ca5f6",
response="eb1f3ad109804e62f1ba1edcc3706bf9"
Call-ID: 2143389df4877360@64.7.189.14
CSeq: 19607 ACK
User-Agent: Grandstream BT100 1.0.6.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
--- (12 headers 0 lines)---
<-- SIP read from 64.7.189.14:5060:
BYE sip:500@64.7.161.26 SIP/2.0
Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bKd592db0d7856fb18
From: "Budge Tone"
<sip:budgeTone-PubIP@64.7.161.26>;tag=b72941c93fe74588
To: <sip:500@64.7.161.26>;tag=as324cfd6f
Proxy-Authorization: Digest username="budgeTone-PubIP",
realm="asterisk", algorithm=MD5, uri="sip:500@64.7.161.26",
nonce="351ca5f6",
response="677ef75492337c32f50f3a936f4c0919"
Call-ID: 2143389df4877360@64.7.189.14
CSeq: 19608 BYE
User-Agent: Grandstream BT100 1.0.6.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
--- (11 headers 0 lines)---
Sending to 64.7.189.14 : 5060 (non-NAT)
Transmitting (no NAT) to 64.7.189.14:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
64.7.189.14;branch=z9hG4bKd592db0d7856fb18;received=64.7.189.14
From: "Budge Tone"
<sip:budgeTone-PubIP@64.7.161.26>;tag=b72941c93fe74588
To: <sip:500@64.7.161.26>;tag=as324cfd6f
Call-ID: 2143389df4877360@64.7.189.14
CSeq: 19608 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:500@64.7.161.26>
Content-Length: 0
---
== Spawn extension (local, 500, 1) exited non-zero on
'SIP/budgeTone-PubIP-7e44'
Destroying call '2143389df4877360@64.7.189.14'
------------------------------
Message: 6
Date: Thu, 19 Jan 2006 19:47:19 +0200
From: Rob Lith <rob@connection-telecom.com>
Subject: Re: [Asterisk-Users] Brief silences during calls
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID:
<32b7e7b80601190947v808f974la6225f06afc9ed6c@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Just look through the devices settings for suppress silence or transmit
silence and don't supress or prevent transmission... this is a common
problem inX-Lite
Rob
On 1/19/06, Mojo with Horan & Company, LLC <mojo@horanappraisals.com>
wrote:>
> check irq supply on the * server -- When you run zttest do you
maintain> over 98%? or like Steve suggested, the network may have congestion or
> other errors ethereal may help you figure out.
>
> I had a polycom 500 that was doing this to my user, 301s and 501s
> wouldn't do it. Not sure if that was network issues or something with
> the polycom itself.
>
> Moj
>
> Mimmus wrote:
> > Where can I investigate the origin of brief silences during calls
> from/to my
> > SIP phone?
> > Only rare pauses of 0.5-1.0 sec when I'm not able to hear
anything.
> >
> > Thanks
> > Mimmus
> >
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> --
> Mojo <mojo@horanappraisals.com>
> Office Manger, Horan & Company, LLC
> (907) 747-6666 x112
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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------------------------------
Message: 7
Date: Thu, 19 Jan 2006 17:47:29 +0000 (UTC)
From: tony@softins.clara.co.uk (Tony Mountifield)
Subject: [Asterisk-Users] Re: MeetMe Listen Only flag (|m)
To: asterisk-users@lists.digium.com
Message-ID: <dqojbh$i17$1@softins.clara.co.uk>
In article
<B0CF4196F21DC0448367514774331AB7E831BF@scl-exch2k3.phoenix.com>,
Dan Austin <Dan_Austin@Phoenix.com> wrote:> Tony wrote:
> > I should tidy it up and submit it, but haven't got round to it :-(
>
> Let us know if you can. I'm already maintaining a grocery list
> of patches to make MeetMe viable in my orginization, so one more
> won't kill me.
I should be able do so this weekend. That's the plan, anyway :-)
I'll post the Mantis bug# when I've submitted it.
Cheers
Tony
--
Tony Mountifield
Work: tony@softins.co.uk - http://www.softins.co.uk
Play: tony@mountifield.org - http://tony.mountifield.org
------------------------------
Message: 8
Date: Thu, 19 Jan 2006 12:52:36 -0500
From: Jared Watkins <jared@watkins.net>
Subject: Re: [Asterisk-Users] SAN Devices
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <43CFD1E4.3070302@watkins.net>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Adam Robins wrote:
>Anyone out there using small-midsized (2-4 TB) SAN solution among
>multiple Asterisk systems? I don't have the budget for an EMC-caliber
>solution, and can't seem to find much else out there.
>
>
I designed a virtualized san and have been running it in production for
the last two years... Speaking from experience... stay away from EMC!
We have several storage systems in production.. from multiple vendors...
and I've had nothing but problems with the CX line of emc systems.
Performance problems... hardware/crashing problems.. (they run embedded
xp you know) and dead fibre port problems. If I didn't have two of
everything.. mirroring across cabinets with IpStor we would have had
serious problems.
Just my two cents on the issue of 'EMC-caliber' storage...
Jared
------------------------------
Message: 9
Date: Thu, 19 Jan 2006 18:52:35 +0100
From: Massimo De Nadal <maxx@digital-system.it>
Subject: [Asterisk-Users] Disabling zap echo cancellor from dialplan
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <43CFD1E3.6020108@digital-system.it>
Content-Type: text/plain; charset=ISO-8859-15; format=flowed
Anybody knows if it's possible to disable zap echo cancellor from
dialplan only for certain outbound calls ??
I share the same phone lines for voice calls and faxes. Iaxmodem works
fine for me only turning off the echo cancellor, but I need it for
voice calls.
Any ideas ?
maxx
------------------------------
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Asterisk-Users mailing list
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http://lists.digium.com/mailman/listinfo/asterisk-users
End of Asterisk-Users Digest, Vol 18, Issue 121
***********************************************
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
asterisk-users-request@lists.digium.com
Sent: Wednesday, October 04, 2006 11:03 AM
To: asterisk-users@lists.digium.com
Subject: asterisk-users Digest, Vol 27, Issue 16
Send asterisk-users mailing list submissions to
asterisk-users@lists.digium.com
To subscribe or unsubscribe via the World Wide Web, visit
http://lists.digium.com/mailman/listinfo/asterisk-users
or, via email, send a message with subject or body 'help' to
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You can reach the person managing the list at
asterisk-users-owner@lists.digium.com
When replying, please edit your Subject line so it is more specific
than "Re: Contents of asterisk-users digest..."
Today's Topics:
1. Re: Spandsp and tif (Steve Underwood)
2. Zaptel problems (Shea, Matt)
3. Call Interception (Delca)
4. RE: Zaptel problems (Colin Anderson)
5. Asterisk 1.4 moh - mohsuggest (Douglas Garstang)
6. Re: Zaptel problems (Bernardo Vieira)
7. Re: Call Interception (Bernardo Vieira)
8. Re: Digium TDM or SPA-3000? (Jay R. Ashworth)
9. Oneway audio (Giordano Grandis)
10. digium compatibility notes (marek cervenka)
11. Re: Call Interception (Jay R. Ashworth)
12. Transfer feature - howto? (Mike)
13. Re: Call Interception (Time Bandit)
14. New tutorial - peering two * servers using IAX (lenz)
15. Re: Call Interception (Steve Edwards)
16. snom 360: how to make record button working ? (noro kamen)
17. Re: Call Interception (Don)
18. SIP client that runs on Linux or Solaris through X Windows?
(Joe)
19. Re: Where is the PlayDTMF command? (Frank Church)
20. Wouldn't Tri-tone detection in Dial() be cool? (Steve Murphy)
21. Re: T1 incoming connects, but no sound (Nathan Bell)
22. Intel Chipset 945p compatible? (R.R. Libera)
23. RE: New tutorial - peering two * servers using IAX
(Douglas Garstang)
24. Need USA DID + trunk provider (R.R. Libera)
25. Need USA DID + trunk provider (R.R. Libera)
----------------------------------------------------------------------
Message: 1
Date: Thu, 05 Oct 2006 00:22:15 +0800
From: Steve Underwood <steveu@coppice.org>
Subject: Re: [asterisk-users] Spandsp and tif
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <4523DFB7.2040605@coppice.org>
Content-Type: text/plain; charset=UTF-8; format=flowed
Giedrius Augys wrote:
> Hi,
> Now I'm testing faxes with spandsp. I have problems that spandsp do
> not add headers to fax page: LOCALHEADERINFO.
> Please help me.
There is a bug in adding page header with spandsp-0.0.2pre26. I have
fixed this in the development code, but I haven't yet put the fix into
the 0.0.2prexx series.
Steve
------------------------------
Message: 2
Date: Wed, 4 Oct 2006 12:27:02 -0400
From: "Shea, Matt" <Matt.Shea@ONSTAR.com>
Subject: [asterisk-users] Zaptel problems
To: <asterisk-users@lists.digium.com>
Message-ID:
<E251C4244455254CBF8F7F03465B43A30175772B@USRN4EX0ONS01.onstar.ad.gm.com>
Content-Type: text/plain; charset="us-ascii"
I'm running Asterisk/Zaptel on a Fedora Core 4 machine. The software
runs ok with one exception. Zaptel appears to load OK on bootup, but
when you check it on login, zttool still shows red/nop alarms on the T1
lines. I have to manually start it again for the alarms to disappear
and the T1 lines to function properly. I've updated the drivers to
1.2.9.1 and double checked my configuration files to no effect. Any
suggestions will be much appreciated.
Matt
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------------------------------
Message: 3
Date: Wed, 4 Oct 2006 16:31:51 +0000
From: Delca <delcas@gmail.com>
Subject: [asterisk-users] Call Interception
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID:
<9d0acd5b0610040931t357d6bbel438f680cbd6cd08a@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi,
I'm deploying an asterisk PBX for a Call Center and i was ordered to
check if the Customer Support Supervisor could intercept the calls so
they can check how they employees work with Asterisk.
I read http://www.voip-info.org/wiki/view/Intercepting+SIP+Calls and i
got a clue about intercepting calls. But actually i wanted to know if
someone have experience with this sort of things.
Cheers!
Santiago
------------------------------
Message: 4
Date: Wed, 4 Oct 2006 10:36:07 -0600
From: Colin Anderson <ColinA@landmarkmasterbuilder.com>
Subject: RE: [asterisk-users] Zaptel problems
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
<asterisk-users@lists.digium.com>
Message-ID:
<E251506D3758AA4882130317D52ACD324A9D1D@land-edm-hs2.landmarkhomes.net>
Content-Type: text/plain; charset="iso-8859-1"
Had the same problem in fc2. Solution was to chkconfig zaptel off chkconfig
asterisk off then in rc.local modprobe wct1xxp (i think) then ztcfg then
start safe_asterisk. Dunno why.
Hey, is OnStar using Asterisk? Details, please.
-----Original Message-----
From: Shea, Matt [mailto:Matt.Shea@ONSTAR.com]
Sent: Wednesday, October 04, 2006 10:27 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Zaptel problems
I'm running Asterisk/Zaptel on a Fedora Core 4 machine. The software runs
ok with one exception. Zaptel appears to load OK on bootup, but when you
check it on login, zttool still shows red/nop alarms on the T1 lines. I
have to manually start it again for the alarms to disappear and the T1 lines
to function properly. I've updated the drivers to 1.2.9.1 and double
checked my configuration files to no effect. Any suggestions will be much
appreciated.
Matt
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------------------------------
Message: 5
Date: Wed, 4 Oct 2006 10:37:00 -0600
From: "Douglas Garstang" <dgarstang@oneeighty.com>
Subject: [asterisk-users] Asterisk 1.4 moh - mohsuggest
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID:
<645FEC31A18FE54A8721500CDD55A7B6035D0B45@mail.oneeighty.com>
Content-Type: text/plain; charset="iso-8859-1"
I'm trying to get moh working correctly in Asterisk 1.4. A complete lack of
documentation isn't helping much.
I have this in sip.conf:
[3254101]
type=friend
...
mohsuggest=class1
[3254102]
type=friend
...
mohsuggest=class2
A call is bridged between the two extensions. When 3254102 puts 3254101 on
hold, 3254101 hears moh class 'class2' which is correct. However, when
3254101 puts 3254102 on hold, the 3254102 hears the default music class.
Why?
Doug.
------------------------------
Message: 6
Date: Wed, 04 Oct 2006 13:39:47 -0300
From: Bernardo Vieira <bernardo.vieira@terra.com.br>
Subject: Re: [asterisk-users] Zaptel problems
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <4523E3D3.5040603@terra.com.br>
Content-Type: text/plain; charset=ISO-8859-1
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Is ztcfg running at boot after the zaptel modules have been loaded?
What's the output of ztcfg?
Shea, Matt wrote:> I'm running Asterisk/Zaptel on a Fedora Core 4 machine. The software
> runs ok with one exception. Zaptel appears to load OK on bootup, but
> when you check it on login, zttool still shows red/nop alarms on the T1
> lines. I have to manually start it again for the alarms to disappear
> and the T1 lines to function properly. I've updated the drivers to
> 1.2.9.1 and double checked my configuration files to no effect. Any
> suggestions will be much appreciated.
>
>
>
> Matt
>
>
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
- --
"What most profoundly divides two men is a different sense and degree of
cleanliness. What help is all honesty and mutual utility, what help is
all the good will for each other: in the end the fact remains-they can't
stand each other?s smell!"
- - Nietzsche
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------------------------------
Message: 7
Date: Wed, 04 Oct 2006 13:45:21 -0300
From: Bernardo Vieira <bernardo.vieira@terra.com.br>
Subject: Re: [asterisk-users] Call Interception
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <4523E521.3070900@terra.com.br>
Content-Type: text/plain; charset=ISO-8859-1
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Do you need to actively intercept the call (i.e. participate in the
conversation) or just listen in the channel? For the latter you can just
use the ChanSpy application.
Delca wrote:> Hi,
>
> I'm deploying an asterisk PBX for a Call Center and i was ordered to
> check if the Customer Support Supervisor could intercept the calls so
> they can check how they employees work with Asterisk.
>
> I read http://www.voip-info.org/wiki/view/Intercepting+SIP+Calls and i
> got a clue about intercepting calls. But actually i wanted to know if
> someone have experience with this sort of things.
>
>
> Cheers!
> Santiago
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
- --
"What most profoundly divides two men is a different sense and degree of
cleanliness. What help is all honesty and mutual utility, what help is
all the good will for each other: in the end the fact remains-they can't
stand each other?s smell!"
- - Nietzsche
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------------------------------
Message: 8
Date: Wed, 4 Oct 2006 12:48:10 -0400
From: "Jay R. Ashworth" <jra@baylink.com>
Subject: Re: [asterisk-users] Digium TDM or SPA-3000?
To: asterisk-users@lists.digium.com
Message-ID: <20061004164810.GD17956@cgi.jachomes.com>
Content-Type: text/plain; charset=us-ascii
On Tue, Oct 03, 2006 at 09:41:04PM -0600, Joseph wrote:> Since you are just planning it, keep in mind to select something that
> will be IPv6 ready.
I don't know that this is necessary, actually.
If I understood the OP correctly, he's terminating line/trunk
appearances which arrive at his switch analog, so the IP side of a
media gateway would be on a private LAN, and therefore IPv6 would be
entirely unnecessary, no?
Cheers,
-- jra
--
Jay R. Ashworth
jra@baylink.com
Designer Baylink RFC
2100
Ashworth & Associates The Things I Think
'87
e24
St Petersburg FL USA http://baylink.pitas.com +1 727 647
1274
"That's women for you; you divorce them, and 10 years later,
they stop having sex with you." -- Jennifer Crusie; _Fast_Women_
------------------------------
Message: 9
Date: Wed, 4 Oct 2006 18:46:53 +0200
From: "Giordano Grandis" <g.grandis@invidea.it>
Subject: [asterisk-users] Oneway audio
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID:
<62A244761070314C8DC5860B0ED131FD0232AB@homer.tecnojest.it>
Content-Type: text/plain; charset="us-ascii"
Hi list,
I'm testing transfer with sip re-invite and bristuff-0.0.8-RCn using an
HFC pci card connetced directly to telco; this is what happen:
1. SIP phone calls a mobile phone (or another residential phone)
2. The called party answers the call
3. Now the sip phone puts on hold the call and calls another sip phone
4. They speak normally
5. Now hte phone that called the mobile transfer the session to the
second one phone
6. The sip phone can hear the mobile phone, but not viceversa.
This works perfectly if i try a blind transfer.
Whaere could be the problem? On the phone....on asterisk ?
Anyone can help me?
Thanks in advance
Giordano
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Message: 10
Date: Wed, 4 Oct 2006 18:53:38 +0200 (CEST)
From: marek cervenka <cervajs@fpf.slu.cz>
Subject: [asterisk-users] digium compatibility notes
To: asterisk-users@lists.digium.com
Message-ID: <Pine.LNX.4.61.0610041847220.28620@axpsu.fpf.slu.cz>
Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
hi,
what is mean by "partially incompatible" in
http://www.digium.com/en/docs/misc/compatibility_notes.php
i have server with E7221+te110p mobo and i think i dont have any problems
thanks
---------------------------------------
Marek Cervenka
======================================
------------------------------
Message: 11
Date: Wed, 4 Oct 2006 12:57:19 -0400
From: "Jay R. Ashworth" <jra@baylink.com>
Subject: Re: [asterisk-users] Call Interception
To: asterisk-users@lists.digium.com
Message-ID: <20061004165719.GF17956@cgi.jachomes.com>
Content-Type: text/plain; charset=us-ascii
On Wed, Oct 04, 2006 at 04:31:51PM +0000, Delca wrote:> I'm deploying an asterisk PBX for a Call Center and i was ordered to
> check if the Customer Support Supervisor could intercept the calls so
> they can check how they employees work with Asterisk.
The call center bix calls that "Service Observing", and I believe
that yeah, you can do that with *. I base that thought on some
things I've read on the mailing list this week and last; if you've
just subscribed, you might want to scan the archives.
Cheers,
-- jra
--
Jay R. Ashworth
jra@baylink.com
Designer Baylink RFC
2100
Ashworth & Associates The Things I Think
'87
e24
St Petersburg FL USA http://baylink.pitas.com +1 727 647
1274
"That's women for you; you divorce them, and 10 years later,
they stop having sex with you." -- Jennifer Crusie; _Fast_Women_
------------------------------
Message: 12
Date: Wed, 4 Oct 2006 13:01:50 -0400
From: "Mike" <list@virtutel.ca>
Subject: [asterisk-users] Transfer feature - howto?
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
<asterisk-users@lists.digium.com>
Message-ID: <003001c6e7d6$ca15ed10$0a01a8c0@MIKE>
Content-Type: text/plain; charset="iso-8859-1"
Hi,
My setup is the following: Voip provider-----------(SIP
DID)----------->Asterisk box----(SIP through a termination
provider)--------------->multiple cell phones.
The cell phones each have their extension (201,202,203,204) and I'd like to
be able to have them transfer a call to somebody else. Ex: Prospect calls
extension 201, talks to the salesgy, who forwards him to the tech guru
somehow.
My guess is I have to use the transfer feature found in feature.conf. I
tried, no success. What Wiki page do I need to look at to get details on
this?
Mike
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------------------------------
Message: 13
Date: Wed, 4 Oct 2006 13:07:21 -0400
From: "Time Bandit" <timebandit001@gmail.com>
Subject: Re: [asterisk-users] Call Interception
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID:
<1e2050d50610041007q6195362eh5b4047612caa56df@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> I'm deploying an asterisk PBX for a Call Center and i was ordered to
> check if the Customer Support Supervisor could intercept the calls so
> they can check how they employees work with Asterisk.
have a look at these :
http://www.voip-info.org/wiki-Asterisk+cmd+ZapBarge
and
http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy
hth
------------------------------
Message: 14
Date: Wed, 04 Oct 2006 19:10:35 +0200
From: lenz <lenz-ml@oinko.net>
Subject: [asterisk-users] New tutorial - peering two * servers using
IAX
To: "asterisk-users@lists.digium.com"
<asterisk-users@lists.digium.com>
Message-ID: <op.tgwpzxz5uxa8ts@smtp.ngi.it>
Content-Type: text/plain; format=flowed; delsp=yes;
charset=iso-8859-15
Hi list,
today I have been teaching a class on * and have found that many students
find it quite hard to understand how setting up IAX peering between two
servers may work. So I prepared a little step by step tutorial hoping it
might be useful to someone in the future.
See it at http://astrecipes.net/index.php?n=204
Comments and corrections are welcome. The site is a wiki, so feel free to
modify and improve.
l.
--
Home of QueueMetrics - http://queuemetrics.loway.it
------------------------------
Message: 15
Date: Wed, 4 Oct 2006 10:20:39 -0700 (PDT)
From: Steve Edwards <asterisk.org@sedwards.com>
Subject: Re: [asterisk-users] Call Interception
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <Pine.LNX.4.64.0610040959230.16990@fs.sedwards.com>
Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
Check out meetme.
We create a meetme conference for each agent when the agent logs in. As
customer's call in, the call is matched (by DNIS and IVR) to the
"longest
idle" agent with the required skill (or any agent if no agent with the
matching skill is available).
The supervisors can join any conference "pre-muted" by entering the
agent
ID. If needed, they can "un-mute" and contribute to the call or kick
the
agent and take the call.
It took a couple of AGI's and some tweaks to app_meetme.c for custom
whispers at the start of the call to tell the agent the type of call while
the customer hears "ring" and kicking the agents, but we're pretty
happy
at this point.
On Wed, 4 Oct 2006, Delca wrote:
> Hi,
>
> I'm deploying an asterisk PBX for a Call Center and i was ordered to
> check if the Customer Support Supervisor could intercept the calls so
> they can check how they employees work with Asterisk.
>
> I read http://www.voip-info.org/wiki/view/Intercepting+SIP+Calls and i
> got a clue about intercepting calls. But actually i wanted to know if
> someone have experience with this sort of things.
>
>
> Cheers!
> Santiago
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
Thanks in advance,
------------------------------------------------------------------------
Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
------------------------------
Message: 16
Date: Wed, 4 Oct 2006 19:33:50 +0200
From: "noro kamen" <noroast@gmail.com>
Subject: [asterisk-users] snom 360: how to make record button working
?
To: asterisk-users@lists.digium.com
Message-ID:
<6f66c2f50610041033m39c64d5cibaee7b46232b5d50@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi,
I'd like to make record button working on snom 320/360 + asterisk.
As I learned from wireshark output, the phone produces SIP info
message "Record: on", while record button pressed.
Can anybody give me an advice, how to teach asterisk to understand
that SIP info message and start recording ?
TIA
noro
------------------------------
Message: 17
Date: Wed, 4 Oct 2006 13:34:30 -0400
From: "Don" <sales@xwebfactor.com>
Subject: Re: [asterisk-users] Call Interception
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID: <004301c6e7db$5a6dfed0$1d01a8c0@shizznit2000>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
reply-type=original
If they are just trying to listen in you can use zapbarge
----- Original Message -----
From: "Jay R. Ashworth" <jra@baylink.com>
To: <asterisk-users@lists.digium.com>
Sent: Wednesday, October 04, 2006 12:57 PM
Subject: Re: [asterisk-users] Call Interception
> On Wed, Oct 04, 2006 at 04:31:51PM +0000, Delca wrote:
>> I'm deploying an asterisk PBX for a Call Center and i was ordered
to
>> check if the Customer Support Supervisor could intercept the calls so
>> they can check how they employees work with Asterisk.
>
> The call center bix calls that "Service Observing", and I believe
> that yeah, you can do that with *. I base that thought on some
> things I've read on the mailing list this week and last; if you've
> just subscribed, you might want to scan the archives.
>
> Cheers,
> -- jra
> --
> Jay R. Ashworth
> jra@baylink.com
> Designer Baylink RFC
> 2100
> Ashworth & Associates The Things I Think
'87
> e24
> St Petersburg FL USA http://baylink.pitas.com +1 727 647
> 1274
>
> "That's women for you; you divorce them, and 10 years later,
> they stop having sex with you." -- Jennifer Crusie; _Fast_Women_
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> No virus found in this incoming message.
> Checked by AVG Free Edition.
> Version: 7.1.407 / Virus Database: 268.12.12/462 - Release Date: 10/3/2006
>
>
------------------------------
Message: 18
Date: Wed, 4 Oct 2006 12:35:42 -0500
From: Joe <joe.uelk@gmail.com>
Subject: [asterisk-users] SIP client that runs on Linux or Solaris
through X Windows?
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID:
<9d6632e0610041035p2d34fac4g5412041ab050af0c@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hello,
I'm looking for a SIP client that work with Asterisk that will run on
Linux or Solaris and will work with X Windows. I know X won't all the
media to work but I'm really only interested in SIP signaling.
Thanks!
Joe
------------------------------
Message: 19
Date: Wed, 4 Oct 2006 18:36:04 +0100
From: "Frank Church" <voipfc@googlemail.com>
Subject: Re: [asterisk-users] Where is the PlayDTMF command?
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID:
<84b7c6460610041036n693e9f97l1fdf7ff386f22d9b@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Moises, do you know if the DTMF event in bug 6082 made it into version 1.4?
When I last tried to compile that branch it needed the latest version
of make 3.81, which trunk did not, and caused me to wonder if it had
been committed to trunk.
The DTMF detection events in trunk did not also function, and made we
wonder if they had been taken out or required some additional post
install configuration, as they worked well before
That bug thread seems to has gone rather quiet now.
On 10/4/06, Moises Silva <moises.silva@gmail.com>
wrote:> You are just not loading the module. Connect to Asterisk terminal
>
> # asterisk -vr
>
> and load the module
>
> CLI> load app_senddtmf.so
>
>
> Best Regards.
>
> On 10/4/06, Jan du Toit <jan.du.toit@decisionworx.com> wrote:
> > Hi all.
> >
> > I'm confused. On
http://www.voip-info.org/wiki-Asterisk+manager+API it
> > says that the PlayDTMF command is available since version 1.2.8. I
> > upgraded to version 1.2.12.1 but I cant find it if I type in
"show
> > manager commands" there is no PlayDTMF command. According to
resources
> > on the internet this action links to the send dtmf application. I
> > checked the source code under the apps folder and it is their!
> > |apps/app_senddtmf.c
> >
> > |Is it not compiling? Why is this function not available to me?
> >
> > Please help.
> > Thanks.||
> >
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> --
> "Su nombre es GNU/Linux, no solamente Linux, mas info en
http://www.gnu.org"> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
------------------------------
Message: 20
Date: Wed, 04 Oct 2006 11:35:53 -0600
From: Steve Murphy <murf@digium.com>
Subject: [asterisk-users] Wouldn't Tri-tone detection in Dial() be
cool?
To: asterisk-users@lists.digium.com
Message-ID: <1159983353.3638.58.camel@monster>
Content-Type: text/plain; charset="us-ascii"
To: Whom it may Concern:
Well, it hit me last night as I was falling asleep... Asterisk (in the
app Zapateller)
can emit the tri-tone (you know beep-Beep-BEEP... The number you have
dialed is no longer
in service. Please check the number and...blah, blah)
Well, it occurred to me that, for the sake of orthogonality, wouldn't it
be cool if
Asterisk's Dial function also detected that tone, with an option to
immediately
hang up if it occurred, with a result code of WRONGNUMBER or NOSERVICE
or whatever?
It also occurred to me that this **might** only be useful to the hated
and dreaded
autodialers that telemarketers use. Even so, it wouldn't hurt me any
more than normal
to have asterisk-based autodialers detect that and get me off their call
lists!
Hah, I'm not trying to imply that I have the skill set right now to
implement this,
nor am I trying to convince anyone right now to do it. The idea just hit
me, and
I wonder if it has already been done somewhere?
murf
--
Steve Murphy
Software Developer
Digium
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Message: 21
Date: Wed, 04 Oct 2006 11:45:03 -0600
From: Nathan Bell <nathanb@actarg.com>
Subject: Re: [asterisk-users] T1 incoming connects, but no sound
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <4523F31F.5080806@actarg.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Mark Farver wrote:
> Nathan Bell wrote:
>
>> extensions.conf:
>> [from-ptsn]
>> exten => s,1,Answer()
>> exten => s,2,Playback(vm-goodbye)
>> exten => s,3,Hangup()
>>
> You might try adding a "wait(3)" command after the answer. Some
> analog lines do not pass audio immediately after being answered.
> (Something to do with how toll processing is handled)
>
> Mark
>
After adding in a "wait(3)" to the extensions.conf, at s,2, I still
get
no audo be passed to me. However, I noticed that each time I place a
call, asterisk thinks that two calls are happening. Here's the log
output of what happens (all with no audio):
Oct 3 17:16:48 NOTICE[10763] chan_zap.c: Got event 18 (Ring Begin)...
Oct 3 17:16:48 VERBOSE[10763] logger.c: -- Executing
Answer("Zap/1-1", "") in new stack
Oct 3 17:16:48 DEBUG[10763] chan_zap.c: Took Zap/1-1 off hook
Oct 3 17:16:48 DEBUG[10763] chan_zap.c: Enabled echo cancellation on
channel 1
Oct 3 17:16:48 DEBUG[10763] chan_zap.c: Engaged echo training on channel 1
Oct 3 17:16:48 VERBOSE[10763] logger.c: -- Executing
Wait("Zap/1-1", "3") in new stack
Oct 3 17:16:48 VERBOSE[10766] logger.c: -- Starting simple switch
on 'Zap/2-1'
Oct 3 17:16:51 VERBOSE[10763] logger.c: -- Executing
Playback("Zap/1-1", "vm-goodbye") in new stack
Oct 3 17:16:51 DEBUG[10763] channel.c: Scheduling timer at 160 sample
intervals
Oct 3 17:16:51 VERBOSE[10763] logger.c: -- Playing 'vm-goodbye'
(language 'en')
Oct 3 17:16:52 DEBUG[10763] channel.c: Scheduling timer at 0 sample
intervals
Oct 3 17:16:52 DEBUG[10763] channel.c: Scheduling timer at 0 sample
intervals
Oct 3 17:16:52 VERBOSE[10763] logger.c: -- Executing
Hangup("Zap/1-1", "") in new stack
Oct 3 17:16:52 VERBOSE[10763] logger.c: == Spawn extension
(from-ptsn, s, 4) exited non-zero on 'Zap/1-1'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is '(null)'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is '(null)'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is 's'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is 'from-ptsn'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is 'Zap/1-1'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is '(null)'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is 'Hangup'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is '(null)'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is '2006-10-03
17:16:48'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is '2006-10-03
17:16:48'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is '2006-10-03
17:16:52'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is '4'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is '4'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is 'ANSWERED'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is 'DOCUMENTATION'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is '(null)'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is '1159917403.6'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is '(null)'
Oct 3 17:16:52 DEBUG[10763] chan_zap.c: Hangup: channel: 1 index = 0,
normal = 17, callwait = -1, thirdcall = -1
Oct 3 17:16:52 DEBUG[10763] chan_zap.c: disabled echo cancellation on
channel 1
Oct 3 17:16:52 DEBUG[10763] chan_zap.c: Set option TDD MODE, value:
OFF(0) on Zap/1-1
Oct 3 17:16:52 DEBUG[10763] chan_zap.c: Updated conferencing on 1, with
0 conference users
Oct 3 17:16:52 VERBOSE[10763] logger.c: -- Hungup 'Zap/1-1'
Oct 3 17:16:53 NOTICE[10766] chan_zap.c: Got event 18 (Ring Begin)...
Oct 3 17:16:53 VERBOSE[10766] logger.c: -- Executing
Answer("Zap/2-1", "") in new stack
Oct 3 17:16:53 DEBUG[10766] chan_zap.c: Took Zap/2-1 off hook
Oct 3 17:16:53 DEBUG[10766] chan_zap.c: Enabled echo cancellation on
channel 2
Oct 3 17:16:53 DEBUG[10766] chan_zap.c: Engaged echo training on channel 2
Oct 3 17:16:53 VERBOSE[10766] logger.c: -- Executing
Wait("Zap/2-1", "3") in new stack
Oct 3 17:16:56 VERBOSE[10766] logger.c: -- Executing
Playback("Zap/2-1", "vm-goodbye") in new stack
Oct 3 17:16:56 DEBUG[10766] channel.c: Scheduling timer at 160 sample
intervals
Oct 3 17:16:56 VERBOSE[10766] logger.c: -- Playing 'vm-goodbye'
(language 'en')
Oct 3 17:16:57 DEBUG[10766] channel.c: Scheduling timer at 0 sample
intervals
Oct 3 17:16:57 DEBUG[10766] channel.c: Scheduling timer at 0 sample
intervals
Oct 3 17:16:57 VERBOSE[10766] logger.c: -- Executing
Hangup("Zap/2-1", "") in new stack
Oct 3 17:16:57 VERBOSE[10766] logger.c: == Spawn extension
(from-ptsn, s, 4) exited non-zero on 'Zap/2-1'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is '(null)'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is '(null)'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is 's'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is 'from-ptsn'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is 'Zap/2-1'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is '(null)'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is 'Hangup'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is '(null)'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is '2006-10-03
17:16:53'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is '2006-10-03
17:16:53'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is '2006-10-03
17:16:57'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is '4'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is '4'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is 'ANSWERED'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is 'DOCUMENTATION'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is '(null)'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is '1159917408.7'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is '(null)'
Oct 3 17:16:57 DEBUG[10766] chan_zap.c: Hangup: channel: 2 index = 0,
normal = 18, callwait = -1, thirdcall = -1
Oct 3 17:16:57 DEBUG[10766] chan_zap.c: disabled echo cancellation on
channel 2
Oct 3 17:16:57 DEBUG[10766] chan_zap.c: Set option TDD MODE, value:
OFF(0) on Zap/2-1
Oct 3 17:16:57 DEBUG[10766] chan_zap.c: Updated conferencing on 2, with
0 conference users
Oct 3 17:16:57 VERBOSE[10766] logger.c: -- Hungup 'Zap/2-1'
Notice that "Starting simple switch on 'Zap/2-1'" happens
almost
immediatly after it starts on Zap/1-1. Caling in multiple times results
in nearly identical output, with the only difference being which line
Zap/2-1 starts at.
>
>> Testing my setup from a channel bank seems to work just fine
>> (slightly different zaptel.conf and zapata.conf).
>> _______________________________________________
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>
>
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------------------------------
Message: 22
Date: Wed, 04 Oct 2006 15:00:23 -0300
From: "R.R. Libera" <astecomm@gmail.com>
Subject: [asterisk-users] Intel Chipset 945p compatible?
To: Asterisk-Users List <asterisk-users@lists.digium.com>
Message-ID: <4523F6B7.5060603@gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hello,
I had recently install an Asterisk PBX into a brand new PC: Intel
Pentium D 3.4GHz Dual-Core + P5LD2 motherboard + SATA HDD.
I?m planning to handle one E1 with a TE110P interface and I want to know
the compatibility between TE110P and Intel 945P chipset. I already buy
the hardware and the only thing I got into account was the compatibility
between the hardware selected and Debian Sarge (the distro I selected).
I?ll accept any suggestion, advice or comment. Thanks in advance.
R.R. Libera
------------------------------
Message: 23
Date: Wed, 4 Oct 2006 12:01:20 -0600
From: "Douglas Garstang" <dgarstang@oneeighty.com>
Subject: RE: [asterisk-users] New tutorial - peering two * servers
using IAX
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID:
<645FEC31A18FE54A8721500CDD55A7B6035D0B47@mail.oneeighty.com>
Content-Type: text/plain; charset="iso-8859-15"
How about preparing a step by step guide to DUNDi? Good luck with that though
because base DUNDi docs are rarer than periodic element #114 in the known
universe.
Doug.
> -----Original Message-----
> From: lenz [mailto:lenz-ml@oinko.net]
> Sent: Wednesday, October 04, 2006 11:11 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] New tutorial - peering two *
> servers using IAX
>
>
>
> Hi list,
> today I have been teaching a class on * and have found that
> many students
> find it quite hard to understand how setting up IAX peering
> between two
> servers may work. So I prepared a little step by step
> tutorial hoping it
> might be useful to someone in the future.
>
> See it at http://astrecipes.net/index.php?n=204
>
> Comments and corrections are welcome. The site is a wiki, so
> feel free to
> modify and improve.
> l.
>
>
>
>
> --
> Home of QueueMetrics - http://queuemetrics.loway.it
>
> _______________________________________________
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
------------------------------
Message: 24
Date: Wed, 04 Oct 2006 15:03:59 -0300
From: "R.R. Libera" <astecomm@gmail.com>
Subject: [asterisk-users] Need USA DID + trunk provider
To: Asterisk-Users List <asterisk-users@lists.digium.com>
Message-ID: <4523F78F.3010506@gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hello,
I need an USA DID + 15 b-channels. The only option I already have is
OpenVox and I want to see some alternatives. Sound quality is my
priority. Thanks in advance.
R.R Libera
------------------------------
Message: 25
Date: Wed, 04 Oct 2006 15:05:26 -0300
From: "R.R. Libera" <astecomm@gmail.com>
Subject: [asterisk-users] Need USA DID + trunk provider
To: Asterisk-Users List <asterisk-users@lists.digium.com>
Message-ID: <4523F7E6.4040401@gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Sorry, when I said "OpenVox" I should say "VoxBone".
Regards,
------------------------------
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End of asterisk-users Digest, Vol 27, Issue 16
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