hi all, how to set the caller id facility for the TDM400p card. Please help me thanks, sandeep.s ----- Original Message ----- From: <asterisk-users-request at lists.digium.com> To: <asterisk-users at lists.digium.com> Sent: Tuesday, January 15, 2008 3:09 PM Subject: asterisk-users Digest, Vol 42, Issue 51> Send asterisk-users mailing list submissions to > asterisk-users at 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 at lists.digium.com > > You can reach the person managing the list at > asterisk-users-owner at 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: app_voicemail for spanish (Andrew Joakimsen) > 2. SVN servers down for maintenance (Russell Bryant) > 3. Re: Asterisk 1.4.17 crashing more (Steve Totaro) > 4. Zaptel 1.2.23 and 1.4.8 released (The Asterisk Development Team) > 5. Re: AGISTATUS is SUCCESS even though my PHP script returned > -1 (Matt Riddell) > 6. Re: Video Call and Asterisk (Matt Riddell) > 7. Re: app_voicemail for spanish (Anton Krall) > 8. Re: G.729 pre-compiled binaries and Asterisk 1.2.x. (Steve Totaro) > 9. Re: Asterisk RFC2833 to SIP INFO DTMF conversion erros. (Mayur) > 10. Re: AGISTATUS is SUCCESS even though my PHP script returned > -1 (Steve Edwards) > 11. Re: G.729 pre-compiled binaries and Asterisk 1.2.x. > (Tzafrir Cohen) > 12. Park() help, extension not heard (Rob) > 13. Re: AGISTATUS is SUCCESS even though my PHP script returned > -1 (Brian Hutchinson) > 14. Re: Asterisk 1.4.17 crashing more (Brian Hutchinson) > 15. Re: app_voicemail for spanish (Andrew Joakimsen) > 16. Re: G.729 pre-compiled binaries and Asterisk 1.2.x. > (Andrew Joakimsen) > 17. Re: Park() help, extension not heard (Rob) > 18. pickupchan without bristuffed version? (Stefan Guenther) > 19. Re: G.729 pre-compiled binaries and Asterisk 1.2.x. > (Bruce McAlister) > 20. Re: G.729 pre-compiled binaries and Asterisk 1.2.x. > (Thomas Kenyon) > 21. Re: G.729 pre-compiled binaries and Asterisk 1.2.x. > (Andrew Joakimsen) > 22. Re: G.729 pre-compiled binaries and Asterisk 1.2.x. > (Thomas Kenyon) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 14 Jan 2008 18:57:34 -0500 > From: "Andrew Joakimsen" <joakimsen at gmail.com> > Subject: Re: [asterisk-users] app_voicemail for spanish > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <23fd749a0801141557o7c84fa5ah3545781a978e230e at mail.gmail.com> > Content-Type: text/plain; charset=UTF-8 > > The language support is supposed to be there I know I've played with > it and there are at least SOME grammatical changes (don't recall which > right now) > > But if further language support is needed you should file a bugreport. > > > > On Jan 14, 2008 5:04 PM, Anton Krall <akrall at intruder.com.mx> wrote: >> Guys, anybody has a 1.2.x compatible app_voicemail patched for Spanish >> prompts that can handle for example, instead of saying "trabajo mensjes" >> would say "mensajes de trabajo o mensajes trabajo" (inverse)? Also can >> handle singular and plural (mensaje vs. mensajes)? >> >> Anton >> >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > ------------------------------ > > Message: 2 > Date: Mon, 14 Jan 2008 17:59:51 -0600 > From: Russell Bryant <russell at digium.com> > Subject: [asterisk-users] SVN servers down for maintenance > To: undisclosed-recipients:; > Message-ID: <478BF777.5030903 at digium.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > The Digium svn servers are down, and will likely be down for the rest of > the > evening, as I perform some system maintenance. I apologize for any > inconvenience that this may cause. > > -- > Russell Bryant > Senior Software Engineer > Open Source Team Lead > Digium, Inc. > > > > ------------------------------ > > Message: 3 > Date: Mon, 14 Jan 2008 19:03:21 -0500 > From: "Steve Totaro" <stotaro at totarotechnologies.com> > Subject: Re: [asterisk-users] Asterisk 1.4.17 crashing more > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <ea18e54a0801141603i2569a9d2i58011e5000fcfec at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > On Jan 14, 2008 6:23 PM, Abdul <abdul_zu at yahoo.com> wrote: > >> Hi All, >> >> We updated with Asterisk 1.4.17 but it seems unstable. 3, 4 times in one >> day it stop to response to the SIP Clinets so they cannot make call or >> register. But safe_asterisk not restarting it back because asterisk >> running >> without any response to the sip clients. >> >> When we try to do 'core show channels' using Manager it returns only >> >> Action: Command >> Command: show channels >> >> That time asterisk not responding anything for clients for registration >> either for invitation. >> >> Please advice us how we can fix this issue. >> > > > Upgrade to Asterisk 1.2.X unless you need the features in 1.4. > > Thanks, > Steve Totaro > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20080114/2985e65a/attachment-0001.htm > > ------------------------------ > > Message: 4 > Date: Mon, 14 Jan 2008 18:03:28 -0600 > From: The Asterisk Development Team <asteriskteam at digium.com> > Subject: [asterisk-users] Zaptel 1.2.23 and 1.4.8 released > To: undisclosed-recipients:; > Message-ID: <478BF850.7020702 at digium.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > The Asterisk.org development team has released Zaptel versions 1.2.23 and > 1.4.8. > > These releases contain a number of bug fixes as well as new features, > including: > > * New and greatly improved fxotune utility > - > http://lists.digium.com/pipermail/asterisk-users/2008-January/203778.html > * Full support for new Digium cards, TE120P, TE121P, TE122P > * DTMF generator updates allow tones to be generated at runtime, as well > as support for a DTMF "twist", on a per-zone basis. The tones for > Brazil > have been updated to include a 2 dB DTMF twist. > > These releases are available for immediate download from > http://downloads.digium.com/. > > Thank you for your support! > > > > ------------------------------ > > Message: 5 > Date: Tue, 15 Jan 2008 13:21:56 +1300 > From: Matt Riddell <matt at venturevoip.com> > Subject: Re: [asterisk-users] AGISTATUS is SUCCESS even though my PHP > script returned -1 > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <478BFCA4.4010606 at venturevoip.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Brian Hutchinson wrote: > | Hi, > | > | Using Asterisk 1.4.17. I'm calling a PHP script through AGI. No matter > | what my script returns (0 or -1), AGISTATUS always appears to be 0 > | SUCCESS. > | > | I was wanting my script to be able to return a value to the dialplan and > | then test AGISTATUS but it looks like I'm going down the wrong path. > | > | Any suggestions? > > Why don't you just set a variable from the AGI and then test for it in > the dialplan > > - -- > Kind Regards, > > Matt Riddell > Director > _______________________________________________ > > http://www.venturevoip.com (Great new VoIP end to end solution) > http://www.venturevoip.com/news.php (Daily Asterisk News - html) > http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org > > iD8DBQFHi/ykDQNt8rg0Kp4RAh93AJ9BOV/+IjAqte/coiOTCAciRzI25wCZAYW3 > BW/ubpchpy2KUQROsmPnonQ> =4oUM > -----END PGP SIGNATURE----- > > > > ------------------------------ > > Message: 6 > Date: Tue, 15 Jan 2008 13:26:37 +1300 > From: Matt Riddell <matt at venturevoip.com> > Subject: Re: [asterisk-users] Video Call and Asterisk > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <478BFDBD.4080701 at venturevoip.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > bilal ghayyad wrote: > | Hi List; > | > | With new technolgy, alot of mobiles now support Video > | Call, so what is the possibility to have Asterisk > | supporting Video so it support Video call at theie > | Phones? > > Have a look at sip.fontventa.com as well as the Asterisk-Video mailing > list. > > - -- > Kind Regards, > > Matt Riddell > Director > _______________________________________________ > > http://www.venturevoip.com (Great new VoIP end to end solution) > http://www.venturevoip.com/news.php (Daily Asterisk News - html) > http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org > > iD8DBQFHi/29DQNt8rg0Kp4RAmNTAJwNylh0kXzAVeRKXfrkmi9KPjeTrQCeISQN > ognhVpn4zXNu+QR+Rp3quPA> =yhjW > -----END PGP SIGNATURE----- > > > > ------------------------------ > > Message: 7 > Date: Mon, 14 Jan 2008 18:47:52 -0600 > From: "Anton Krall" <akrall at intruder.com.mx> > Subject: Re: [asterisk-users] app_voicemail for spanish > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <330417F4B6BCA34C917684152699E21405556A at mail.exchange.intruder.com.mx> > Content-Type: text/plain; charset="us-ascii" > > Im looking at app_voicemail (remember, this is on 1.2.x) and there seems > to be some syntax changes for Spanish but doesn't seem to have all > that's required... Ill file a bug report on mantis. > > AK > > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Andrew > Joakimsen > Sent: lunes, 14 de enero de 2008 05:58 p.m. > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] app_voicemail for spanish > > The language support is supposed to be there I know I've played with > it and there are at least SOME grammatical changes (don't recall which > right now) > > But if further language support is needed you should file a bugreport. > > > > On Jan 14, 2008 5:04 PM, Anton Krall <akrall at intruder.com.mx> wrote: >> Guys, anybody has a 1.2.x compatible app_voicemail patched for Spanish >> prompts that can handle for example, instead of saying "trabajo > mensjes" >> would say "mensajes de trabajo o mensajes trabajo" (inverse)? Also can >> handle singular and plural (mensaje vs. mensajes)? >> >> Anton >> >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ------------------------------ > > Message: 8 > Date: Mon, 14 Jan 2008 19:50:22 -0500 > From: "Steve Totaro" <stotaro at totarotechnologies.com> > Subject: Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk > 1.2.x. > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <ea18e54a0801141650o242877b7te9e68882b5d05237 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > On Jan 14, 2008 6:54 PM, Andrew Joakimsen <joakimsen at gmail.com> wrote: > >> On Jan 14, 2008 5:51 PM, Steve Totaro <stotaro at totarotechnologies.com> >> wrote: >> > >> > Either that or pay for the legal licensing of G729 and get support >> through >> > the appropriate channels. Using the code for anything other than >> learning >> > purposes is illegal, not to mention that licensing is quite >> > inexpensive. >> > >> >> Using the code period in a country which recognizes software patents >> is an infringement of the patentholder rights. It is not illegal >> anywhere but it does open you up to a great deal of legal liability. >> It does not matter if its in production use or not it is still >> infringement on the patent. Of course unless you have a large >> operation, say the size of Vonage, noone's really going to care.. but >> why are you going to start small with that sort of thinking? You'll >> never get anywhere. >> > > I would argue that it is illegal. The main definition of illegal is > "1. *against > law: *contravening a specific law, especially a criminal law". > http://encarta.msn.com/dictionary_/illegal.html > > While it may not be against criminal law in the US it can be in France and > Austria, in the US it is certainly "against a specific law". > http://en.wikipedia.org/wiki/Patent_law#Law > > Anyways, buying the license is the right thing to do unless you live where > software patent laws are not applicable. > > > >> >> I wonder how many Chinese VoIP phones with G729 & G723 codecs have >> actually licensed the codec? >> >> > Probably none. > > > Thanks, > Steve Totaro > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20080114/3f3aca82/attachment-0001.htm > > ------------------------------ > > Message: 9 > Date: Tue, 15 Jan 2008 06:22:27 +0530 > From: "Mayur" <mninama at varaha.com> > Subject: Re: [asterisk-users] Asterisk RFC2833 to SIP INFO DTMF > conversion erros. > To: <david.cantera at IBSOneCall.com>, "'Asterisk Users Mailing List - > Non-Commercial Discussion'" <asterisk-users at lists.digium.com> > Message-ID: > <mailman.10082.1200389991.10646.asterisk-users at lists.digium.com> > Content-Type: text/plain; charset="us-ascii" > > Hi David, > > Thank you for suggestion. It seems to work well. So asterisk does inband > dtmf to SIP INFO dtmf conversion well. I am curious to know why there is > no > consistency with 2833 to INFO DTMF conversion. Is it a known issue with > asterisk? > > Regards, > > Mayur > > > > _____ > > From: dave cantera [mailto:david.cantera at iacnet.net] > Sent: Sunday, January 13, 2008 11:15 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion; > mninama at varaha.com > Subject: Re: [asterisk-users] Asterisk RFC2833 to SIP INFO DTMF conversion > erros. > > > > mayur, > did you try inband? with sip? > daveC > ;dtmfmode=inband ; Choices are inband, rfc2833, or info > ;allow=ulaw ; dtmfmode=inband only works with ulaw or > alaw! > ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info > ;allow=ulaw ; dtmfmode=inband only works with ulaw or > alaw! > > Mayur wrote: > > Hi, > > I am using asterisk 1.4.17 which is connected to a SIP trunk supporting > rfc2833 dtmf events. Asterisk stays in the media path. In sip.conf I have > set dtmfmode=rfc2833 for the outbound sip proxy (SIP Trunk account) and > for > SIP clients I have set dtmfmode=info. So when I make a call to a cell > number > using the sip trunk and then press digits I can see the 2833 dtmf events > coming to asterisk in the rtp captures. Asterisk seems to detect those and > give SIP INFO to the SIP client. However it fails to detect some of the > digits (which is random) hence the correct sequence of digits is not > received at the SIP client. > > I have tried setting relaxdtmf=yes in sip.conf but that does not seem to > help. Can anyone help me out here? > > > > Regards, > > Mayur > > > > > > > > _____ > > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > _____ > > > > > Internal Virus Database is out-of-date. > Checked by AVG Free Edition. > Version: 7.5.516 / Virus Database: 269.17.13/1209 - Release Date: > 01/04/2008 > 12:05 PM > > > > > > > -- > My wife's sister is in California. > I should buy her a Videophone2008! > > Truly, The Next Best Thing to Being There! > -- > > WorldWideVideoPhones.com > 856.380.0894 > > > > > > __________ NOD32 2786 (20080112) Information __________ > > This message was checked by NOD32 antivirus system. > http://www.eset.com > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20080115/d820b864/attachment.htm > > ------------------------------ > > Message: 10 > Date: Mon, 14 Jan 2008 16:58:13 -0800 (PST) > From: Steve Edwards <asterisk.org at sedwards.com> > Subject: Re: [asterisk-users] AGISTATUS is SUCCESS even though my PHP > script returned -1 > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <Pine.LNX.4.64.0801141647310.21507 at fs.sedwards.com> > Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed > > On Tue, 15 Jan 2008, Matt Riddell wrote: > >> Brian Hutchinson wrote: >> | >> | Using Asterisk 1.4.17. I'm calling a PHP script through AGI. No >> matter >> | what my script returns (0 or -1), AGISTATUS always appears to be 0 >> | SUCCESS. >> >> Why don't you just set a variable from the AGI and then test for it in >> the dialplan > >>From UPGRADE.txt: > > * The exit behavior of the AGI applications has changed. Previously, when > a connection to an AGI server failed, the application would cause the > channel > to immediately stop dialplan execution and hangup. Now, the only time > that > the AGI applications will cause the channel to stop dialplan execution > is > when the channel itself requests hangup. The AGI applications now set an > AGISTATUS variable which will allow you to find out whether running the > AGI > was successful or not. > > Previously, there was no way to handle the case where Asterisk was > unable to > locally execute an AGI script for some reason. In this case, dialplan > execution will continue as it did before, but the AGISTATUS variable > will be > set to "FAILURE". > > A locally executed AGI script can now exit with a non-zero exit code and > this > failure will be detected by Asterisk. If an AGI script exits with a > non-zero > exit code, the AGISTATUS variable will be set to "FAILURE" as opposed to > "SUCCESS". > > I find the idea of a proliferation of inconsistently implemented AGI > failure or success variables undesirable. As I read the above, if > returning a non-zero exit code does not set AGISTATUS to "FAILURE," it's a > bug that needs to be reported. > > Thanks in advance, > ------------------------------------------------------------------------ > Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > > > ------------------------------ > > Message: 11 > Date: Tue, 15 Jan 2008 03:06:46 +0200 > From: Tzafrir Cohen <tzafrir.cohen at xorcom.com> > Subject: Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk > 1.2.x. > To: asterisk-users at lists.digium.com > Message-ID: <20080115010646.GV32205 at xorcom.com> > Content-Type: text/plain; charset=us-ascii > > On Mon, Jan 14, 2008 at 07:50:22PM -0500, Steve Totaro wrote: >> On Jan 14, 2008 6:54 PM, Andrew Joakimsen <joakimsen at gmail.com> wrote: >> > I wonder how many Chinese VoIP phones with G729 & G723 codecs have >> > actually licensed the codec? >> > >> Probably none. > > Well, they sell in the US and in other countries. I suspect that if > licensing requirements were not satisfied, their reselers would have to > pay the licensing fees instead. > > -- > Tzafrir Cohen > icq#16849755 jabber:tzafrir.cohen at xorcom.com > +972-50-7952406 mailto:tzafrir.cohen at xorcom.com > http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir > > > > ------------------------------ > > Message: 12 > Date: Mon, 14 Jan 2008 21:02:59 -0800 > From: Rob <asterisk at private.eklhq.com> > Subject: [asterisk-users] Park() help, extension not heard > To: asterisk-users at lists.digium.com > Message-ID: <478C3E83.9060100 at private.eklhq.com> > Content-Type: text/plain; charset="iso-8859-1" > > I'm trying to get call parking to work, but I've run out of things to try. > > I can place a call between two internal extensions, then on one > extension transfer the call to extension 700, and the call gets parked > on 701 but I don't hear the extension number when I do the transfer. I > can hangup and call 701 and get the call back. > > Here's what I see: (comments added on lines starting with !!) > > !! Start call from desktop to phone > -- Executing [*00 at internal:1] Macro("SIP/rob_desktop-007fbcb0", > "ring-all") in new stack > -- Executing [s at macro-ring-all:1] Dial("SIP/rob_desktop-007fbcb0", > "SIP/gs100|20") in new stack > -- Called gs100 > -- SIP/gs100-00816bf0 is ringing > !! Answer the call > -- SIP/gs100-00816bf0 answered SIP/rob_desktop-007fbcb0 > !! Press "transfer" button on phone > -- Started music on hold, class 'default', on SIP/rob_desktop-007fbcb0 > == Spawn extension (macro-ring-all, s, 1) exited non-zero on > 'SIP/rob_desktop-007fbcb0' > !! Dial "700" and "send" on phone > -- Started music on hold, class 'default', on SIP/rob_desktop-007fbcb0 > == Parked SIP/rob_desktop-007fbcb0 on 701 at parkedcalls. Will timeout > back to extension [macro-ring-all] s, 1 in 45 seconds > -- <SIP/gs100-00816bf0> Playing 'digits/7' (language 'en') > -- <SIP/gs100-00816bf0> Playing 'digits/0' (language 'en') > -- <SIP/gs100-00816bf0> Playing 'digits/1' (language 'en') > !! Hear "beep" on phone > -- Added extension '701' priority 1 to parkedcalls > -- Stopped music on hold on SIP/rob_desktop-007fbcb0 > == SIP/rob_desktop-007fbcb0 got tired of being parked > > > > > It looks like it's doing the right thing, but I never hear "7" "0" "1". > I hear a "beep" after the "1" message is logged. > > > I added an extension that does Answer(), SayDigits(123), and Hangup(), > and I hear "one" "two" "three" perfectly. > > > What do I need to do to hear the extension where the call gets parked? > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20080114/3c34584f/attachment-0001.htm > > ------------------------------ > > Message: 13 > Date: Tue, 15 Jan 2008 08:10:37 +0300 > From: "Brian Hutchinson" <b.hutchman at gmail.com> > Subject: Re: [asterisk-users] AGISTATUS is SUCCESS even though my PHP > script returned -1 > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Cc: matt at venturevoip.com > Message-ID: > <3d1967ab0801142110x4147fb64yda1cafe765a373d7 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > >> >> >> >> Why don't you just set a variable from the AGI and then test for it in > > > That is what I ended up doing and that worked. Just thought I'd post to > the > list since from what I read it sounds like the script return value should > be > reflected in AGISTATUS and it wasn't. Didn't know if it was a bug that > should be reported or not. > > Thanks for your help. > > Regards, > > Brian > > >> the dialplan >> >> - -- >> Kind Regards, >> >> Matt Riddell >> Director >> _______________________________________________ >> >> http://www.venturevoip.com (Great new VoIP end to end solution) >> http://www.venturevoip.com/news.php (Daily Asterisk News - html) >> http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) >> -----BEGIN PGP SIGNATURE----- >> Version: GnuPG v1.4.7 (MingW32) >> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org >> >> iD8DBQFHi/ykDQNt8rg0Kp4RAh93AJ9BOV/+IjAqte/coiOTCAciRzI25wCZAYW3 >> BW/ubpchpy2KUQROsmPnonQ>> =4oUM >> -----END PGP SIGNATURE----- >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> 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/20080115/c4120762/attachment-0001.htm > > ------------------------------ > > Message: 14 > Date: Tue, 15 Jan 2008 08:16:04 +0300 > From: "Brian Hutchinson" <b.hutchman at gmail.com> > Subject: Re: [asterisk-users] Asterisk 1.4.17 crashing more > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Cc: abdul_zu at yahoo.com > Message-ID: > <3d1967ab0801142116h52205ae0t3d08c5acdfd8e65e at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > I'm running 1.4.17. I've been running that version plus an addition of > Unicall MFC/R2 and the only time I have seen it die is right away on > startup > due to something in one of the .conf files not being right. It has not > died > during normal operation. I'm running two TE420B cards on a large Dell > 2950. Not doing SIP so I'm not exercising that portion of the code. > > Regards, > > Brian > > On Jan 15, 2008 2:23 AM, Abdul <abdul_zu at yahoo.com> wrote: > >> Hi All, >> >> We updated with Asterisk 1.4.17 but it seems unstable. 3, 4 times in one >> day it stop to response to the SIP Clinets so they cannot make call or >> register. But safe_asterisk not restarting it back because asterisk >> running >> without any response to the sip clients. >> >> When we try to do 'core show channels' using Manager it returns only >> >> Action: Command >> Command: show channels >> >> That time asterisk not responding anything for clients for registration >> either for invitation. >> >> Please advice us how we can fix this issue. >> >> >> ------------------------------ >> Looking for last minute shopping deals? Find them fast with Yahoo! >> Search.<http://us.rd.yahoo.com/evt=51734/*http://tools.search.yahoo.com/newsearch/category.php?category=shopping> >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> 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/20080115/c72612ca/attachment-0001.htm > > ------------------------------ > > Message: 15 > Date: Tue, 15 Jan 2008 00:48:11 -0500 > From: "Andrew Joakimsen" <joakimsen at gmail.com> > Subject: Re: [asterisk-users] app_voicemail for spanish > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <23fd749a0801142148i1ee66cc6o64411b43de0c7a4f at mail.gmail.com> > Content-Type: text/plain; charset=UTF-8 > > No features are being added for 1.2 so I'd check to see if 1.4 has the > changes you need before filing a bugreport. > > > > On Jan 14, 2008 7:47 PM, Anton Krall <akrall at intruder.com.mx> wrote: >> Im looking at app_voicemail (remember, this is on 1.2.x) and there seems >> to be some syntax changes for Spanish but doesn't seem to have all >> that's required... Ill file a bug report on mantis. >> >> AK >> >> >> >> -----Original Message----- >> From: asterisk-users-bounces at lists.digium.com >> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Andrew >> Joakimsen >> Sent: lunes, 14 de enero de 2008 05:58 p.m. >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] app_voicemail for spanish >> >> The language support is supposed to be there I know I've played with >> it and there are at least SOME grammatical changes (don't recall which >> right now) >> >> But if further language support is needed you should file a bugreport. >> >> >> >> On Jan 14, 2008 5:04 PM, Anton Krall <akrall at intruder.com.mx> wrote: >> > Guys, anybody has a 1.2.x compatible app_voicemail patched for Spanish >> > prompts that can handle for example, instead of saying "trabajo >> mensjes" >> > would say "mensajes de trabajo o mensajes trabajo" (inverse)? Also can >> > handle singular and plural (mensaje vs. mensajes)? >> > >> > Anton >> > >> > >> > _______________________________________________ >> > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> > >> > asterisk-users mailing list >> > To UNSUBSCRIBE or update options visit: >> > http://lists.digium.com/mailman/listinfo/asterisk-users >> > >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > ------------------------------ > > Message: 16 > Date: Tue, 15 Jan 2008 00:57:13 -0500 > From: "Andrew Joakimsen" <joakimsen at gmail.com> > Subject: Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk > 1.2.x. > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <23fd749a0801142157p69ab6be2i53b38a9dfdc8eb3d at mail.gmail.com> > Content-Type: text/plain; charset=UTF-8 > > On Jan 14, 2008 7:50 PM, Steve Totaro <stotaro at totarotechnologies.com> > wrote: >> >> > >> I would argue that it is illegal. The main definition of illegal is " 1. >> against law: contravening a specific law, especially a criminal law". >> http://encarta.msn.com/dictionary_/illegal.html > > Illegal means that something violates a criminal law. You linked to a > page that describe the law in the US regarding patentholders > registration of said patents. I'm not saying we should infringe on the > patentholder's right I am simply saying it is not a criminal act, at > least in the US. > >> While it may not be against criminal law in the US it can be in France >> and >> Austria, in the US it is certainly "against a specific law". >> http://en.wikipedia.org/wiki/Patent_law#Law > > Software is generally not patentable in the European Union (and > probably in the countries that are pseudo-EU members) > >> Anyways, buying the license is the right thing to do unless you live >> where >> software patent laws are not applicable. > > Totally agree. > > > > ------------------------------ > > Message: 17 > Date: Mon, 14 Jan 2008 22:34:05 -0800 > From: Rob <asterisk at private.eklhq.com> > Subject: Re: [asterisk-users] Park() help, extension not heard > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <478C53DD.6080008 at private.eklhq.com> > Content-Type: text/plain; charset=ISO-8859-1 > > 1.4.17. > > > Rob wrote: >> I'm trying to get call parking to work, but I've run out of things to >> try. >> >> I can place a call between two internal extensions, then on one >> extension transfer the call to extension 700, and the call gets parked >> on 701 but I don't hear the extension number when I do the transfer. >> I can hangup and call 701 and get the call back. >> >> Here's what I see: (comments added on lines starting with !!) >> >> !! Start call from desktop to phone >> -- Executing [*00 at internal:1] Macro("SIP/rob_desktop-007fbcb0", >> "ring-all") in new stack >> -- Executing [s at macro-ring-all:1] Dial("SIP/rob_desktop-007fbcb0", >> "SIP/gs100|20") in new stack >> -- Called gs100 >> -- SIP/gs100-00816bf0 is ringing >> !! Answer the call >> -- SIP/gs100-00816bf0 answered SIP/rob_desktop-007fbcb0 >> !! Press "transfer" button on phone >> -- Started music on hold, class 'default', on >> SIP/rob_desktop-007fbcb0 >> == Spawn extension (macro-ring-all, s, 1) exited non-zero on >> 'SIP/rob_desktop-007fbcb0' >> !! Dial "700" and "send" on phone >> -- Started music on hold, class 'default', on >> SIP/rob_desktop-007fbcb0 >> == Parked SIP/rob_desktop-007fbcb0 on 701 at parkedcalls. Will timeout >> back to extension [macro-ring-all] s, 1 in 45 seconds >> -- <SIP/gs100-00816bf0> Playing 'digits/7' (language 'en') >> -- <SIP/gs100-00816bf0> Playing 'digits/0' (language 'en') >> -- <SIP/gs100-00816bf0> Playing 'digits/1' (language 'en') >> !! Hear "beep" on phone >> -- Added extension '701' priority 1 to parkedcalls >> -- Stopped music on hold on SIP/rob_desktop-007fbcb0 >> == SIP/rob_desktop-007fbcb0 got tired of being parked >> >> >> >> >> It looks like it's doing the right thing, but I never hear "7" "0" >> "1". I hear a "beep" after the "1" message is logged. >> >> >> I added an extension that does Answer(), SayDigits(123), and Hangup(), >> and I hear "one" "two" "three" perfectly. >> >> >> What do I need to do to hear the extension where the call gets parked? >> > > > > ------------------------------ > > Message: 18 > Date: Tue, 15 Jan 2008 08:17:19 +0100 > From: Stefan Guenther <asterisk01 at in-put.de> > Subject: [asterisk-users] pickupchan without bristuffed version? > To: asterisk-users at lists.digium.com > Message-ID: <478C5DFF.7090008 at in-put.de> > Content-Type: text/plain; charset=ISO-8859-15; format=flowed > > Hello, > > following the description in the wiki > (http://www.voip-info.org/wiki/view/Asterisk+phone+snom) > > I have set up a number of SNOM phones to monitor extensions with hints. > The lights on the phones flash when a call on another phone comes in. > > According to the article in the wiki I need the application PickUpChan > to catch on of the calls which causes the light to flash. But PickUpchan > is only available in the bristuff version of asterisk > > Is there another way to get a specific call and not just press *8 to get > a random call out of the callgroup? > > Thanks for your help, > > Stefan > -- > > ******************************************** > in-put GbR - Das Linux-Systemhaus > Stefan-Michael Guenther > Geschaeftsfuehrer > Moltkestrasse 49 D-76133 Karlsruhe > Tel./Fax : +49 (0)721 / 83044 - 98/93 > http://www.in-put.de > ******************************************** > Schulungen Installationen > Beratung Support > Voice-over-IP-Loesungen > ******************************************** > > > > > ------------------------------ > > Message: 19 > Date: Tue, 15 Jan 2008 09:01:25 +0000 > From: Bruce McAlister <bruce.mcalister at blueface.ie> > Subject: Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk > 1.2.x. > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <478C7665.1000404 at blueface.ie> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Steve Totaro wrote: > >> >> I would suggest building it yourself >> (http://www.readytechnology.co.uk/open/ipp-codecs/doc-svn6.txt >> <http://www.readytechnology.co.uk/open/ipp-codecs/doc-svn6.txt>). It is >> not that difficult and ensures that it "should" be compatible with your >> machine. Just a little work. >> > > Has anyone tried building this on Solaris, I just had a look at the link > and it looks like the Intel IPP stuff is only released for Windows, > Linux and MAC. And the v32 G729 codec from Digium does not load within > asterisk on Solaris, sooo, the Solaris users out there dont have much > support when it comes to G729 codecs, a real pity really, this stops > some large scale roll-outs. > > > > ------------------------------ > > Message: 20 > Date: Tue, 15 Jan 2008 09:05:35 +0000 > From: Thomas Kenyon <digium at sanguinarius.co.uk> > Subject: Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk > 1.2.x. > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <478C775F.4010002 at sanguinarius.co.uk> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Andrew Joakimsen wrote: >> On Jan 14, 2008 7:50 PM, Steve Totaro <stotaro at totarotechnologies.com> >> wrote: >>> >> >>> Anyways, buying the license is the right thing to do unless you live >>> where >>> software patent laws are not applicable. >> >> Totally agree. >> > I have bought many more licenses from asterisk than I've ever used, and > mostly use the asterisk.hosting.lv codecs. > > Twice now while using the digium codec, upon upgrading asterisk, it > stopped working. > > The Beta codec (based on IPP5), is much much faster than either the > digium or the older codec, and at home (only place I run beta software), > there hasn't been a problem. > > Mind you, according to show translation, the older codec (based on > IPP4), is faster than the digium codec too. > > > > ------------------------------ > > Message: 21 > Date: Tue, 15 Jan 2008 04:23:04 -0500 > From: "Andrew Joakimsen" <joakimsen at gmail.com> > Subject: Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk > 1.2.x. > To: bruce.mcalister at blueface.ie, "Asterisk Users Mailing List - > Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Message-ID: > <23fd749a0801150123u66de469dm14fd4b31cf0fddfa at mail.gmail.com> > Content-Type: text/plain; charset=UTF-8 > > They used to have solaris on the Digium FTP site but they seem to be gone > now :( > > On the "free" codec site they have some complied with icc and others > with gcc4 so I don't see why you can't get this working with gcc on > solaris. > > On Jan 15, 2008 4:01 AM, Bruce McAlister <bruce.mcalister at blueface.ie> > wrote: >> Steve Totaro wrote: >> >> > >> > I would suggest building it yourself >> > (http://www.readytechnology.co.uk/open/ipp-codecs/doc-svn6.txt >> > <http://www.readytechnology.co.uk/open/ipp-codecs/doc-svn6.txt>). It >> > is >> > not that difficult and ensures that it "should" be compatible with your >> > machine. Just a little work. >> > >> >> Has anyone tried building this on Solaris, I just had a look at the link >> and it looks like the Intel IPP stuff is only released for Windows, >> Linux and MAC. And the v32 G729 codec from Digium does not load within >> asterisk on Solaris, sooo, the Solaris users out there dont have much >> support when it comes to G729 codecs, a real pity really, this stops >> some large scale roll-outs. >> >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > ------------------------------ > > Message: 22 > Date: Tue, 15 Jan 2008 09:39:16 +0000 > From: Thomas Kenyon <digium at sanguinarius.co.uk> > Subject: Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk > 1.2.x. > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <478C7F44.1020406 at sanguinarius.co.uk> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Andrew Joakimsen wrote: >> They used to have solaris on the Digium FTP site but they seem to be gone >> now :( >> >> On the "free" codec site they have some complied with icc and others >> with gcc4 so I don't see why you can't get this working with gcc on >> solaris. >> > If you can, be sure to submit it to arkadi at kvin.lv , I'm sure he'll be > happy to receive it. > > > > ------------------------------ > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > End of asterisk-users Digest, Vol 42, Issue 51 > ********************************************** >
Guilherme Loch Waltrick Góes
2008-Jan-17 11:03 UTC
[asterisk-users] asterisk-users Digest, Vol 42, Issue 51
On zapata.conf use the parameter callerid. On Jan 17, 2008 3:33 AM, sandeep <sandeep.s at briotelecom.com> wrote:> hi all, > how to set the caller id facility for > the TDM400p card. > > Please help me > > thanks, > sandeep.s > > --Guilherme Loch G?es Visite nossa loja virtual: http://www.shopvoip.com.br Not?cias e F?rum sobre VoIP com software livre: http://www.asteriskexperts.com.br -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080117/32de7271/attachment.htm