Olger Merlos V.
2006-Mar-27 05:21 UTC
[Asterisk-Users] Question about Polycom 601 and expansion module.
Hi, I have questions about the Polycom 601 and side card.... 1) In the side card the lights all time off... But all functions it's ok. I need help with extension module of polycom... All works fine... But lights not work.... So... I don't know when any person or extension is busy... Any ideas? , Olger On 3/27/06 11:34 PM, "asterisk-users-request@lists.digium.com" <asterisk-users-request@lists.digium.com> wrote:> 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. Alarm on Unicall (acriollo) > 2. Ability to put call on hold via manager? (Steve Totaro) > 3. Re: Receptionist Phones (was 3Com Phones) (Daniel Hazelbaker) > 4. Call Waiting Issues (Brad Glonka) > 5. Wanted: Cd-bootable Fedora+Asterisk (Bruce Komito) > 6. Master.csv Shell Script (Jeremy) > 7. Re: Ability to put call on hold via manager? (Alberto Sagredo) > 8. TE 205P/A102 fit in hp dc7600? (JOSE MANUEL CORTES DAVID) > 9. Re: * Meetme Freeze patch found (Brent Torrenga) > 10. Re: RE : [Asterisk-Users] Stability of Asterisk with 2 x > TDM400P cards (6analogue lines) (Krzysztof Drewicz) > 11. queue caveats (asterisk@anime.net) > 12. RE: Bluetooth headset in handsfree modewith SJPhoneor X-lite > (wendell hamilton) > 13. Re: Config File Management (Giovanni Miano) > 14. RE: Ability to put call on hold via manager? (Steve Totaro) > 15. Re: Authorization by ip (Giovanni Miano) > 16. Re: Call Simulator (Giovanni Miano) > 17. Re: Alarm on Unicall (Melcon Moraes) > 18. Re: Receptionist Phones (was 3Com Phones) (Justin Moore) > 19. Re: Master.csv Shell Script (Mojo with Horan & Company, LLC) > 20. Re: On site installtion Tech. wanted (Richard Amerman) > 21. Re: Receptionist Phones (Daniel Hazelbaker) > 22. FXO without answer supervision (Dan Austin) > 23. Re: Call Waiting Issues (C F) > 24. Re: Caller ID length (C F) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 27 Mar 2006 13:06:03 -0600 > From: acriollo <crmeae@gmail.com> > Subject: [Asterisk-Users] Alarm on Unicall > To: asterisk-users@lists.digium.com > Message-ID: > <e5ff64dc0603271106q3c43b113u17078eeb50deb1ae@mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi all, > any body can tell me why i am receiving this message in my sever ? > > I have running * with 10 Digital Lines, but i am receiving a lot of times > this message . > Is a software issue or is a hardware issue ? > > Regards. > > Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event: > Unicall/5 event Alarm > Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event: > Unicall/5 Alarm masks 0x0000 0x0004 > Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3096 handle_uc_event: > Unicall/5 Alarm No Alarm raised, Yellow Alarm cleared > Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event: > Unicall/6 event Alarm > Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event: > Unicall/6 Alarm masks 0x0000 0x0004 > Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3096 handle_uc_event: > Unicall/6 Alarm No Alarm raised, Yellow Alarm cleared > Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event: > Unicall/7 event Alarm > Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event: > Unicall/7 Alarm masks 0x0000 0x0004 > Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3096 handle_uc_event: > Unicall/7 Alarm No Alarm raised, Yellow Alarm cleared > Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event: > Unicall/8 event Alarm > Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event: > Unicall/8 Alarm masks 0x0000 0x0004 > Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3096 handle_uc_event: > Unicall/8 Alarm No Alarm raised, Yellow Alarm cleared > Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event: > Unicall/9 event Alarm > Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event: > Unicall/9 Alarm masks 0x0000 0x0004 > Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3096 handle_uc_event: > Unicall/9 Alarm No Alarm raised, Yellow Alarm cleared > Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event: > Unicall/10 event Alarm > Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event: > Unicall/10 Alarm masks 0x0000 0x0004 > Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3096 handle_uc_event: > Unicall/10 Alarm No Alarm raised, Yellow Alarm cleared > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20060327/07581018 > /attachment-0001.htm > > ------------------------------ > > Message: 2 > Date: Mon, 27 Mar 2006 14:20:49 -0400 > From: "Steve Totaro" <stotaro@asteriskhelpdesk.com> > Subject: [Asterisk-Users] Ability to put call on hold via manager? > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: > <DFB93BD730105941BD1A782A1EE9E95CCC07@1-0fa9e300af524.asteriskhelpdesk.com> > > Content-Type: text/plain; charset="us-ascii" > > Does anyone know if there is built in ability to put call on hold via > the manager interface? > > Thanks, > Steve Totaro > http://www.asteriskhelpdesk.com > > > > > > ------------------------------ > > Message: 3 > Date: Mon, 27 Mar 2006 11:18:10 -0800 > From: Daniel Hazelbaker <daniel@highdesertchurch.com> > Subject: Re: [Asterisk-Users] Receptionist Phones (was 3Com Phones) > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: > <124F4DD7-FDE4-4156-BCC8-897837AFF38A@highdesertchurch.com> > Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed > > We may end up using a software solution, but there are two main > issues with a software solution (for us at least): > > 1) For us in particular, our receptionists have ALWAYS (for the past > 15 years at least) used a physical switchboard style for "routing" > and seeing availability. From past hardware->software changes we > know that it will be very frustrating for them. For us, it is much > more worth it to spend $1,000 to buy each of the two receptionists a > really nice phone that supports these features rather than get a > cheap software (though very nice) solution. > > 2) Having a software solution can cause grief and frustration to an > already overworked receptionist. Just a few examples (these are not > as uncommon as one might think): User quits web browser after > finishing looking something up on-line, doesn't realize they just > closed out their switchboard until they need it and it is not there. > User "gets lost" trying to find the right window while trying to not > sound like an idiot to the person on the phone. Computer has frozen, > or otherwise has problems, and must be rebooted. > > I do like the look of Asternic, it is very "old-style" and easy to > get used to, but we would still prefer a hardware solution if > possible. We may end up having to say, "sorry but you need to deal > with this for a while until some bugs in the system are resolved > (i.e. the 7 line problem), but as soon as a hardware solution is > available we will switch you back to it." Hopefully we can find > something before we switch, but if not it is good to know that > software solutions are a viable alternative. > > >> Have you looked that the flash operator panel? >> >> http://www.asternic.org/demo.html >> >> I know you mentioned not wanting a software solution because of >> confusion >> but I think that would be pretty easy to understand. >> >> Curt >> >> _______________________________________________ >> --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: 4 > Date: Mon, 27 Mar 2006 14:18:54 -0500 > From: "Brad Glonka" <glonka@gmail.com> > Subject: [Asterisk-Users] Call Waiting Issues > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: > <70d6d74b0603271118w2fec495dhc9091ff1c8e776c7@mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > I have two call waiting problems. > > I have a POTS line into and FXO port > and telephones on an FXS port > > 1) I can't seem to use the flash button(on the phone) to answer a call > waiting call. > I see the callerid coming though and here the call waiting tone, > but I just can't seem to answer it. The flash button seems to have no > effect. > > I have: > callwaiting=yes in zapata.conf > > > 2) When the PSTN line is in use and a call comes though via call waiting. > I don't think it hits my exten => s > Instead it rings the phone (but as I mentioned above I > can't seem to answer it) > > Thanks for any suggestions. > > > ------------------------------ > > Message: 5 > Date: Mon, 27 Mar 2006 11:09:45 -0800 (PST) > From: Bruce Komito <brucek@bagel.com> > Subject: [Asterisk-Users] Wanted: Cd-bootable Fedora+Asterisk > To: asterisk-users@lists.digium.com > Message-ID: <20060327110636.P70224-100000@mustang.bagel.com> > Content-Type: TEXT/PLAIN; charset=US-ASCII > > I'm in search someone who would be interested in developing a Fedora-baed > Asterisk system that is bootable from a CD or possible flash. I am aware > of the various commercial and free solutions out there, but none I have > found suit our needs...mainly because they are not easily extensible > and/or upgradeable. > > If you are interested in working on such a project, please contact me > off-list. > > Thanks > > Bruce Komito > High Sierra Networks, Inc. > www.servers-r-us.com > (775) 236-5815 > > > > > ------------------------------ > > Message: 6 > Date: Mon, 27 Mar 2006 14:25:17 -0500 > From: "Jeremy" <thezerogroup@gmail.com> > Subject: [Asterisk-Users] Master.csv Shell Script > To: <asterisk-users@lists.digium.com> > Message-ID: <44283c23.4b9614b5.2e6e.08d4@mx.gmail.com> > Content-Type: text/plain; charset="us-ascii" > > Im not looking for anything super detailed, just something to run through > the master.csv file and give total time per account code. . . .does anyone > out there have a script like this I could work from? > > > > ------------------------------ > > Message: 7 > Date: Mon, 27 Mar 2006 21:35:38 +0200 > From: Alberto Sagredo <asagredo@peoplecall.com> > Subject: Re: [Asterisk-Users] Ability to put call on hold via manager? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <44283E8A.30506@peoplecall.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > You could park it to parking extensiones. > > Does it help you? > > Steve Totaro escribi?: >> Does anyone know if there is built in ability to put call on hold via >> the manager interface? >> >> Thanks, >> Steve Totaro >> http://www.asteriskhelpdesk.com >> >> >> >> _______________________________________________ >> --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: 8 > Date: Mon, 27 Mar 2006 14:43:35 -0500 > From: "JOSE MANUEL CORTES DAVID" <jmcortes@puj.edu.co> > Subject: [Asterisk-Users] TE 205P/A102 fit in hp dc7600? > To: <asterisk-users@lists.digium.com> > Message-ID: > <ECCE1C17FAB85140A69454A40B443FCA46251E@CORREOWEB.puj.edu.co> > Content-Type: text/plain; charset="iso-8859-1" > > Hi > > I would like to know if the TE 205 fit in a hp dc7600? what about the A 102 > from Sangoma? > > Thanks > > Jose Manuel Cortes > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20060327/11de5468 > /attachment-0001.htm > > ------------------------------ > > Message: 9 > Date: Mon, 27 Mar 2006 13:41:37 -0600 > From: "Brent Torrenga" <lists@torrenga.com> > Subject: [Asterisk-Users] Re: * Meetme Freeze patch found > To: <asterisk-users@lists.digium.com> > Message-ID: <004f01c651d6$76d4da30$7200a8c0@oscar> > Content-Type: text/plain; charset="US-ASCII" > > Forgoe the patch, just upgrade to 1.2.6. The changelog lists it as a fix > from 1.2.5 to 1.2.6. > > >> I'm a bit newbie, could you tell me how to i apply the patch? >> >> Thanks in advance >> Marco Mouta >> >> On 3/27/06, Benoit Panizzon <benoit.panizzon@imp.ch> wrote: >>> On Friday 24 March 2006 16:05, Benoit Panizzon wrote: >>>> Hi all >>>> >>>> Apparently there is a patch for those 1.2.4/5 MeetMe Freezes: >>>> >>>> http://bugs.digium.com/view.php?id=5884 >>>> >>>> Haven't tried it out yet. >>> >>> I can now confirm: No freezes/crashes anymore since I applied the patch. >>> >>> -Benoit- > > Sincerely, > > Brent A. Torrenga > brent.torrenga@torrenga.com > > Torrenga Engineering, Inc. > 907 Ridge Road > Munster, Indiana 46321-1771 > > 219.836.8918x325 Voice > 219.836.1138 Facsimile > www.torrenga.com > > > > ------------------------------ > > Message: 10 > Date: Mon, 27 Mar 2006 21:44:53 +0200 > From: Krzysztof Drewicz <drewicz@citicom.pl> > Subject: Re: RE : [Asterisk-Users] Stability of Asterisk with 2 x > TDM400P cards (6analogue lines) > To: f6hqz-m@hamwlan.net, Asterisk Users Mailing List - Non-Commercial > Discussion <asterisk-users@lists.digium.com> > Message-ID: <442840B5.2090205@citicom.pl> > Content-Type: text/plain; charset=ISO-8859-2 > > f6hqz-m@hamwlan.net wrote: >> Hi, >> >> Jump to a TDM2402E for 6 POTS lines with hardware echocan. >> Only one IRQ used, and easy future extensions by adding modules. >> > > Have anyone here used a clone i.e. A1200P-01 (A1200P + 1 FXO100 module) ? > > > > ------------------------------ > > Message: 11 > Date: Mon, 27 Mar 2006 11:55:39 -0800 (PST) > From: asterisk@anime.net > Subject: [Asterisk-Users] queue caveats > To: Asterisk-Users@lists.digium.com > Message-ID: <Pine.LNX.4.63.0603271153340.18525@sasami.anime.net> > Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed > > According to http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue, under > the "Notes" section: > > "Transfers of calls that are answered out of a queue must be done using > Asterisk '#' transfers (enabled with the 't' option above). SIP transfers > result in the Agent remaining affiliated with the call until its eventual > termination, preventing that agent from being offered another call." > > Is this still true in asterisk 1.2.6? > > -Dan > > > ------------------------------ > > Message: 12 > Date: Mon, 27 Mar 2006 11:59:49 -0800 > From: "wendell hamilton" <routerguy@rightsolve.com> > Subject: RE: [Asterisk-Users] Bluetooth headset in handsfree modewith > SJPhoneor X-lite > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: > <88C40680C3714444B9BF371990DE880535877C@mail.rightsolve.com> > Content-Type: text/plain; charset="us-ascii" > > Hi, > > You need to have completely replaced the Microsoft driver, because it > doesn't support the headset or ctp Bluetooth profiles. This gave me > fits! I followed the instructions at > http://www.windowsdevcenter.com/pub/a/windows/2005/07/05/bluetooth.html > and it works with both a Plantronics and a Motorola Headset, and I can > answer calls with idefisk, eyebeam, x-lite, and kapanga. > > If you end up not having both of these in the Bluetooth service > selection, you won't end up with the results you're looking for. > > HTH > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chuck Bunn > Sent: Monday, March 27, 2006 9:46 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Bluetooth headset in handsfree modewith > SJPhoneor X-lite > > Hi, > > I am not having trouble with the bluetooth stack since the Toshiba stack > > has the headset profile which supports a subset of AT commands > <http://en.wikipedia.org/wiki/AT_command> from GSM 07.07 for minimal > controls including the ability to ring, answer a call, hang up and > adjust the volume. The problem is getting the softphone to work with > these AT commands so that the answer/hangup function will work from the > bluetooth headset. > > Thanks > > wendell hamilton wrote: >> Try replacing the XP Bluetooth stack with the widcomm drivers...google >> is your friend! >> >> >> -----Original Message----- >> From: asterisk-users-bounces@lists.digium.com >> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chuck > Bunn >> Sent: Monday, March 27, 2006 6:21 AM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: [Asterisk-Users] Bluetooth headset in handsfree mode with >> SJPhoneor X-lite >> >> Hi, >> >> After much searching I have found that it might be possible to get a >> bluetooth headset to answer/hangup with SJPhone or Xlite if the > headset >> supports handsfree mode. My Toshiba bluetooth stack supports this but > I >> have not been able to figure out how to enable it. Also Windows XP >> desktop bluetooth stack does not support handsfree but Windows CE does > >> (go figure). Has anyone got handsfree mode working with a bluetooth >> headset? How about working with SJPhone or Xlite or some other SIP >> phone? For some reason the SJPhone when used with a bluetooth headset >> disconnects/reconnects bluetooth when the answer/hangup button is used > >> on the headset (how the hell did that come about). Using a bluetooth >> headset with a SIP phone and asterisk would really help me by removing > >> those pesky wires.... >> >> 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 >> >> This message is confidential. It may also be privileged or otherwise > protected by work product immunity or other legal rules. If you have > received it by mistake, please let us know by e-mail reply and delete it > from your system; you may not copy this message or disclose its contents > to anyone. Please send us by fax any message containing deadlines as > incoming e-mails are not screened for response deadlines. The integrity > and security of this message cannot be guaranteed on the Internet. >> >> _______________________________________________ >> --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 >> >> >> >> >> > > _______________________________________________ > --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: 13 > Date: Mon, 27 Mar 2006 22:00:29 +0200 > From: "Giovanni Miano" <giomiano@gmail.com> > Subject: Re: [Asterisk-Users] Config File Management > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: > <d75be1ca0603271200l3536e7cbp9ffe8a269c51b8ca@mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > You can use FastAGI > > See http://www.asteriskjava.org > > 2006/3/27, David Gomillion <dgomillion@eyecarenow.com>: >> >> Sorry for thread breaking... I'm on digest. >> >>>> I'm curious (ok, well I admit it - it's for perosnal gain) what >>>> methods people are using to manage asterisk config files when they >>>> have multiple asterisk systems? >>> >>> I'm using CVS. I only have one server right now. I use it on other >>> clusters to sync files and it works for me.. >> >> Instead of doing this, I ended up creating a MySQL database and a few >> scripts to generate the config files for each of my servers. All I have >> to >> do is update the database, and the correct server pulls the information >> from >> the DB, generates the file, reloads, and sends reboot messages to the >> proper >> phones. Very specific to my needs, but extremely fast and effective. And >> all it requires on each Asterisk server is cron, PHP, and php-mysql. >> >> I had to customize a few of the variables inside the PHP scripts for each >> server, but by putting them close to the top, it's not a real big deal >> when >> I update the scripts to customize them for my servers. Mind you, I only >> have 4 servers on this system, but we don't anticipate growing beyond one >> more server for a while. >> >> One thing to mention that I have found: use lots of macros. Some of my >> macros require 6 or 7 arguments, but they are extremely flexible and >> trivial >> to generate on the fly through these tools. Each extension fits in only >> one >> line in the dialplan (calls a macro). Entries in the DB turn on and off >> features, sets the timeout, forwards to another extension or sends to >> voicemail, etc. >> >> Just what I'm doing. Hope it helps. >> >> David >> >> >> _______________________________________________ >> --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 >> > > > > -- > Giovanni Miano > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20060327/f85da612 > /attachment-0001.htm > > ------------------------------ > > Message: 14 > Date: Mon, 27 Mar 2006 15:10:36 -0400 > From: "Steve Totaro" <stotaro@asteriskhelpdesk.com> > Subject: RE: [Asterisk-Users] Ability to put call on hold via manager? > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: > <DFB93BD730105941BD1A782A1EE9E95CCC09@1-0fa9e300af524.asteriskhelpdesk.com> > > Content-Type: text/plain; charset="iso-8859-1" > > I was thinking about that as an option. > > Basically I am integrating a CRM call center app with * and want the agents to > be able to click a radio button to put callers on hold. They only have analog > headsets with on-hook and off-hook. > > It seems like parking and un-parking the call would be pretty complicated. > > Thanks, > Steve Totaro > > >> -----Original Message----- >> From: Alberto Sagredo [mailto:asagredo@peoplecall.com] >> Sent: Monday, March 27, 2006 2:36 PM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [Asterisk-Users] Ability to put call on hold via manager? >> >> You could park it to parking extensiones. >> >> Does it help you? >> >> Steve Totaro escribi?: >>> Does anyone know if there is built in ability to put call on hold via >>> the manager interface? >>> >>> Thanks, >>> Steve Totaro >>> http://www.asteriskhelpdesk.com >>> >>> >>> >>> _______________________________________________ >>> --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 >>> >> >> _______________________________________________ >> --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: 15 > Date: Mon, 27 Mar 2006 22:02:47 +0200 > From: "Giovanni Miano" <giomiano@gmail.com> > Subject: Re: [Asterisk-Users] Authorization by ip > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: > <d75be1ca0603271202i45a61826i8e30288d23535fe3@mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > You can use in sip.conf tag "host" > > host=192.168.1.1 > > 2006/3/27, Sam Tam <sam@netenable.co.uk>: >> >> Can somebody send me a config of how to authorize SIP client by IP? >> >> Sam >> >> >> >> _______________________________________________ >> --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 >> > > > > -- > Giovanni Miano > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20060327/5e050ef1 > /attachment-0001.htm > > ------------------------------ > > Message: 16 > Date: Mon, 27 Mar 2006 22:04:52 +0200 > From: "Giovanni Miano" <giomiano@gmail.com> > Subject: Re: [Asterisk-Users] Call Simulator > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: > <d75be1ca0603271204s4c6f9cat5d41624c1d0c7635@mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > You can use dialout file > > http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+auto-dial+out&prev > iew=20 > > 2006/3/27, Steve Totaro <stotaro@asteriskhelpdesk.com>: >> >> SIPPS is one, I would like to hear of others. >> >> >> >> Of course you could create a dialplan that loops calls in and out. >> >> >> >> Thanks, >> Steve Totaro >> http://www.asteriskhelpdesk.com >> >> ------------------------------ >> >> *From:* voipman [mailto:emeyeem@gmail.com] >> *Sent:* Monday, March 27, 2006 6:39 AM >> *To:* asterisk-users@lists.digium.com >> *Subject:* [Asterisk-Users] Call Simulator >> >> >> >> Guyz, >> >> >> >> I wanna test my asterisk load capability before going to production, >> anyone know is there any call simulator to test this thing? >> >> >> >> Thanks in advance, >> >> >> >> Voipman >> >> _______________________________________________ >> --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 >> >> >> > > > -- > Giovanni Miano > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20060327/02e6026e > /attachment-0001.htm > > ------------------------------ > > Message: 17 > Date: Mon, 27 Mar 2006 17:05:05 -0300 > From: Melcon Moraes <melcon@principaltelecom.com.br> > Subject: Re: [Asterisk-Users] Alarm on Unicall > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <44284571.4040706@principaltelecom.com.br> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > What about some unicall.conf and zaptel.conf lines? > > []'s > MM > > acriollo wrote: >> Hi all, >> any body can tell me why i am receiving this message in my sever ? >> >> I have running * with 10 Digital Lines, but i am receiving a lot of >> times this message . >> Is a software issue or is a hardware issue ? >> >> Regards. >> >> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event: >> Unicall/5 event Alarm >> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event: >> Unicall/5 Alarm masks 0x0000 0x0004 >> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3096 handle_uc_event: >> Unicall/5 Alarm No Alarm raised, Yellow Alarm cleared >> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event: >> Unicall/6 event Alarm >> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event: >> Unicall/6 Alarm masks 0x0000 0x0004 >> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3096 handle_uc_event: >> Unicall/6 Alarm No Alarm raised, Yellow Alarm cleared >> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event: >> Unicall/7 event Alarm >> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event: >> Unicall/7 Alarm masks 0x0000 0x0004 >> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3096 handle_uc_event: >> Unicall/7 Alarm No Alarm raised, Yellow Alarm cleared >> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event: >> Unicall/8 event Alarm >> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event: >> Unicall/8 Alarm masks 0x0000 0x0004 >> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3096 handle_uc_event: >> Unicall/8 Alarm No Alarm raised, Yellow Alarm cleared >> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event: >> Unicall/9 event Alarm >> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event: >> Unicall/9 Alarm masks 0x0000 0x0004 >> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3096 handle_uc_event: >> Unicall/9 Alarm No Alarm raised, Yellow Alarm cleared >> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event: >> Unicall/10 event Alarm >> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event: >> Unicall/10 Alarm masks 0x0000 0x0004 >> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3096 handle_uc_event: >> Unicall/10 Alarm No Alarm raised, Yellow Alarm cleared >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> --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: 18 > Date: Mon, 27 Mar 2006 15:08:16 -0500 > From: "Justin Moore" <wantmoore@gmail.com> > Subject: Re: [Asterisk-Users] Receptionist Phones (was 3Com Phones) > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: > <75835fe10603271208v45c1d8b2kbfbffb829bfdab7a@mail.gmail.com> > Content-Type: text/plain; charset=UTF-8 > > On 3/27/06, Daniel Hazelbaker <daniel@highdesertchurch.com> wrote: >> I have seen that the polycom setup (601+sidecar) works but only for up to 7 >> phones > >> From what I've seen, each sidecar supports up to 14 additional > stations. Three of those along with the 5 buttons on the 601 comes up > to 47 on my calculator. Is there a known problem with the 601+sidecars > and * that prevents the user from being able to monitor more than 7 > extensions? > > Just curious as I've been leaning toward this for our receptionist as > well (only 12 extensions to monitor...) > > -- > Justin Moore > aka wantmoore > --------------------------------------- > www.wantmoore.com > > ------------------------------ > > Message: 19 > Date: Mon, 27 Mar 2006 11:22:21 -0900 > From: "Mojo with Horan & Company, LLC" <mojo@horanappraisals.com> > Subject: Re: [Asterisk-Users] Master.csv Shell Script > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <4428497D.1030907@horanappraisals.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > If you've got PHP installed, here's one I made for our office: > > http://horanappraisals.com/asterisk/total_account_codes/ > > Run it with no parameters to check Master.csv in the current directory, > or pass the filename to parse as the first parameter. > > # ./total_account_codes /var/log/asterisk/cdr-csv/Master.csv > "test" total is 310 seconds or 5.17 minutes or 0.09 hours > "" total is 33130 seconds or 552.17 minutes or 9.2 hours > > # > > The second line totals all lines with no account code specified. > > hth moj > > Jeremy wrote: >> Im not looking for anything super detailed, just something to run through >> the master.csv file and give total time per account code. . . .does anyone >> out there have a script like this I could work from? >> >> _______________________________________________ >> --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 >>