Re: [Asterisk-Users] Zaptel answering the Line
Steve Totaro wrote:
Thomas Kenyon wrote:
Steve Totaro wrote:
No, I mean specifying the group in your zapata.conf file and then
changing your dial statement to dial out like Dial(ZAP/g0/${EXTEN})
the g0 refers to group zero. If specify a channel to belong to
group=1 then it will not be used for outbound. Inbound should work as
normal as long as the channel has the same context as group 0.
I'm not trying to dial out on only a few lines (that's simple) I want
only a few of them to be bgetting answered by asterisk.
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Re: [Asterisk-Users] Zaptel answering the Line
On Sun, Jun 25, 2006 at 08:28:35PM +0100, Thomas Kenyon wrote: I have a TDM400 card with 3x FXO and 1x FXS ports on it. At the moment I'd prefer (till I can get it working more reliable with iaxmodem), for a faxmodem to answer one of the lines instead of the linecard. I've tried changing the context of that line so that the exten = s does nothing, but that stops the line from being able to receive calls (get a recorded This number is not accepting calls at the moment). So my 2 questions are... How do I set one of the channels in zaptel.conf (or elsewhere) so that it is only available for making calls (and not receiving them). What do you want to happen to incoming calls? send that specific channel (using context= ) to a context that does not even Answer() the line. in /etc/asterisk/zapata.conf: signalling=fxs_ks context=noanswer channel = 3 ; make sure you write another 'context=' after than before the next ; channel in extensions.conf: I'm not totally sure it is necessary to make this explicit, but this will reduce the warnings you'll get in the console: [noanswer] exten = s,1,Hangup() -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.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
Re: [Asterisk-Users] Signaling and media
26 jun 2006 kl. 07.10 skrev Martin Joseph: On Jun 25, 2006, at 4:11 PM, Jean-Michel Hiver wrote: Hi List, Is there a way to tell asterisk to only accept SIP streams from the same IP address that is used for signaling? SIP streams are signalling... Have you tested the ACL features in sip.conf - accept/deny ? /O --- Olle E. Johansson * Asterisk Evangelist, developer * VOOP A/S [EMAIL PROTECTED] ___ --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
Re: [Asterisk-Users] Zaptel answering the Line
On Mon, Jun 26, 2006 at 07:37:00AM +0100, Thomas Kenyon wrote:
Steve Totaro wrote:
Thomas Kenyon wrote:
Steve Totaro wrote:
No, I mean specifying the group in your zapata.conf file and then
changing your dial statement to dial out like Dial(ZAP/g0/${EXTEN})
the g0 refers to group zero. If specify a channel to belong to
group=1 then it will not be used for outbound. Inbound should work as
normal as long as the channel has the same context as group 0.
I'm not trying to dial out on only a few lines (that's simple) I want
only a few of them to be bgetting answered by asterisk.
What will be the caller ID be for calls going through those lines?
Who will people call back to?
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755 iax:[EMAIL PROTECTED]
+972-50-7952406
[EMAIL PROTECTED] http://www.xorcom.com
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[Asterisk-Users] Re: What happens if the soekris hardware is defective upon arrival? The Cortex Systems way.
Hi all, first of all thanks for your comments and ideas. I wrote because i wanted to know if i'm wrong or not and to let others khow how some companies operate. I work on technology, i work in the world we move, and i usually are in charged of handle situatios like that, and what i can tell all of you is that, if the system is faulty upon recepion, the only one common practice is open an RMA with the provider and send back the unit at the provider cost. I think, as somebody pointed in any of the lists i wrote (related to soekris technology), that the buying process didn't finish yet because i didn't receive what i bought. And my point of view seems to be different in some cases: my money or the money of my company is good, in my account and in the provider's account, so the gear i got should work fine, is a contract for both sides, not only for me. No problem at all, i will send tomorrow (i'm out of office today) the unit back to Cortex Systems and i will put cleary on the box faulty with some documents as the technical and sales consultant pointed me. I've got an invoice and a UPS delivery note so no fear at all. Thanks for all. Best regards, Jonathan GF -- si secretum tibi sit, tege illud, vel revela ___ --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
Re: [Asterisk-Users] SPA-2002 call HANGUP. May be a SIP bug.
On Friday 23 June 2006 11:07, Kevin P. Fleming wrote: - Dmytro Mishchenko [EMAIL PROTECTED] wrote: In this case during all conversation SIP packets contains Call-ID: [EMAIL PROTECTED] but the final BYE packet from adapter contains Call-ID: [EMAIL PROTECTED] Is such scenario correct from SIP protocol point of view? No, it is not valid. The Call-ID is used to uniquely identify the SIP dialog, and must remain the same. Thanks! I'll deal with Sipura support then. Dmitry. ___ --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
RE: [Asterisk-Users] TE420P/TE415P?
Can the TE406P card's VPM module be swapped for the new revision with Octasic chipset? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Sunday, June 25, 2006 8:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TE420P/TE415P? - C F [EMAIL PROTECTED] wrote: I like the TC400P card, how many T1s will that take? or is it just a Daughter card on the TE4xx ? How many channels can it transcode? Neither. It's a separate device, entirely unrelated to any TDM cards (which means it can be used for any type of channel, not just TDM). The final specs for the number of channels are not yet determined, but we expect to do at least 100 channels of G.729 and/or G.723.1 per board. ___ --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
Re: [Asterisk-Users] Signaling and media
Johansson Olle E a écrit : 26 jun 2006 kl. 07.10 skrev Martin Joseph: On Jun 25, 2006, at 4:11 PM, Jean-Michel Hiver wrote: Hi List, Is there a way to tell asterisk to only accept SIP streams from the same IP address that is used for signaling? SIP streams are signalling... Sorry, I was talking about the media. Have you tested the ACL features in sip.conf - accept/deny ? Any pointers on these ACLs? -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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
RE: [Asterisk-Users] Asterisk Startups
Well now would be a great time to come back, Doug! We miss you! 8) --Rob -Original Message- From: Douglas Garstang [mailto:[EMAIL PROTECTED] Sent: Monday, 26 June 2006 3:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Startups Paul, D'oh. The fact I left Sydney 5 years ago for the US might be a teeny complication. :P Doug. -Original Message- From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Sun 6/25/2006 11:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Asterisk Startups Douglas Garstang wrote: Does anyone know of any startups using Asterisk? What about established companies? Ones that are hiring would be nice :) Doug. -- -- ___ --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 We are always looking for good people - here in Melbourne. PaulH -- Paul Hales Technical Manager AsteriskIT www.asteriskit.com.au bus: 03 8320 8100 mob: 0434 673 529 ___ --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
Re: [Asterisk-Users] Gizmo and Asterisk analysis
On 06/25/06 19:01 Roy Sigurd Karlsbakk said the following: seem to pass all SIP and RTP traffic through their own servers... See http://karlsbakk.net/asterisk/gizmo-project.php for details interesting. but isnt Gizmo an open source client ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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
[Asterisk-Users] Agent Dump
If an agent dumps the call during an announcement (the immortal line in app_queue.c is Agent on %s hungup on the customer. They're going to be pissed) is there anyway of tracking this via a variable or hangupstatus or something - I need to be able to trap this in the dialplan Julian. ___ --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
[Asterisk-Users] Asterisk and Qsig Protocol
Hi All. Somebody works with asterisk linked in ISDN PRI with protocol QSIG with some PABX as Siemens, Philips, etc. The applications as pickup between asterisk and the PABX function? The names in the display and the number of the origin also? Which features that they can be used between the equipments? Best Regards Josué ___ --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
[Asterisk-Users] Asterisk x Siemens HiPath 4000
Hello all. I have installed and functioning asterisk-1.2.9.1 where I effected one upgradein asterisk-1.0.9, is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destinedtoSIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me? Best Regards Josué ___ --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
Re: [Asterisk-Users] TE420P/TE415P?
Hi Kevin. Where could I get more information about those boards? Thanks, D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1610A CEP: 80530-000 - Curitiba - PR +55 41 3252-2977 r 101 http://www.isolve.com.br On 25 de jun de 2006, at 07:07, Kevin P. Fleming wrote: - C F [EMAIL PROTECTED] wrote: I like the TC400P card, how many T1s will that take? or is it just a Daughter card on the TE4xx ? How many channels can it transcode? Neither. It's a separate device, entirely unrelated to any TDM cards (which means it can be used for any type of channel, not just TDM). The final specs for the number of channels are not yet determined, but we expect to do at least 100 channels of G.729 and/or G.723.1 per board. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --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
Re: [Asterisk-Users] Zaptel answering the Line
Thomas Kenyon wrote: I have a TDM400 card with 3x FXO and 1x FXS ports on it. At the moment I'd prefer (till I can get it working more reliable with iaxmodem), for a faxmodem to answer one of the lines instead of the linecard. I've tried changing the context of that line so that the exten = s does nothing, but that stops the line from being able to receive calls (get a recorded This number is not accepting calls at the moment). So my 2 questions are... How do I set one of the channels in zaptel.conf (or elsewhere) so that it is only available for making calls (and not receiving them). Does Zaptel automatically switch echo cancellation off if it detects a fax call? I'm assuming you have a fax machine bridged on the pstn line that also connects to asterisk via an fxo port. If that's the case, you have two ways to accomplish the goal. One, as others have mentioned, is to specify a context in the appropriate channel portion of zapata.conf that goes no where. E.g., it doesn't exist in extensions.conf. Second, specify a context in the appropriate channel portion of zapata.conf that goes to inbound-fax or something like that, and then in extensions.conf, do something like this: [inbound-fax] exten = s,1,Dial(SIP/3034) Notice there is no answer function and there is no timeout value associated with this. The example will ring x3034, however if no one picks up that sip phone, the call remains unanswered from an incoming pstn call perspective. (I've used the same for both a bridged fax machine and for a bridged older answering machine.) You may need to add faxdetect=no in the appropriate channel section of zapata.conf as well. ___ --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
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
I agree whole-heartedly. If I could run this on my dedicated Asterisk machine it would be perfect... On 6/28/06, Matthias Fechner [EMAIL PROTECTED] wrote: Hi Marco, Marco Mouta wrote: Please feel free to contact me if you have more ideas to improve this solution, currently i didn't test more than one simultaneous calls incoming and outgoing through Skype. get it running on unix so you can run it on the asterisk server. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --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
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Marco, bom dia. Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo externo? É freeware? Podemos seguir com o projeto Asterisk-PT? Saudações Josué 2006/6/26, John Klimek [EMAIL PROTECTED]: I agree whole-heartedly.If I could run this on my dedicated Asteriskmachine it would be perfect... On 6/28/06, Matthias Fechner [EMAIL PROTECTED] wrote: Hi Marco, Marco Mouta wrote: Please feel free to contact me if you have more ideas to improve this solution, currently i didn't test more than one simultaneous calls incoming and outgoing through Skype. get it running on unix so you can run it on the asterisk server. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --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 listTo 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
Re: [Asterisk-Users] best hardphone for Asterisk?
On Mon, Jun 26, 2006 at 12:08:48AM -0400, Doug Crompton wrote: Still awfully pricey for home use and the styling is not there for a bedroom or many other areas of a modern home. What we need is a wireless sip phone modeled like the panasonic or uniden which allow multiple extension off of one base. The base would connect to the internet. The other problem is many of these phones require power, so even if you have backup for your central system the phone still needs to be on it. Power over ethernet would help. 1. If you have *, you don't necessarily need multiple handsets off of one base. 2. Cordless phones also require power 3. If the multi-handset cordless phone does suit your needs best, then get a SIP ATA device like a Sipura or IAXy and you should have your needs met. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Bom dia, On 6/26/06, Josué Conti [EMAIL PROTECTED] wrote: Marco, bom dia. Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo externo? Sim é um software da Uplink, disponível para download gratuitamente, n garanto q seja freeware (talvez tenha limitações esta versao free) Podes ver a demo no site: http://asteriskpt.blogspot.com Se tudo estiver ok, deverás ouvir Musiconhold de um asterisk q tenho em casa ( n estou la agora). É freeware? Podemos seguir com o projeto Asterisk-PT? Claro que sim! http://asteriskpt.blogspot.com Podes por posts la, vou criar contas para podermos cooperar no blog. Se preferirem um site ou outra solução, estou aberto a sugestões. Saudações Josué 2006/6/26, John Klimek [EMAIL PROTECTED]: I agree whole-heartedly. If I could run this on my dedicated Asterisk machine it would be perfect... On 6/28/06, Matthias Fechner [EMAIL PROTECTED] wrote: Hi Marco, Marco Mouta wrote: Please feel free to contact me if you have more ideas to improve this solution, currently i didn't test more than one simultaneous calls incoming and outgoing through Skype. get it running on unix so you can run it on the asterisk server. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --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 ___ --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 -- Com os melhores cumprimentos, Marco Mouta ___ --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
[Asterisk-Users] This is getting really annoying - re: POSTFIX
What on earth is going on with the list?!?! Some of my messages never make it... then days later I get something like this back: Final-Recipient: rfc822; asterisk-users@lists.digium.com Action: failed Status: 5.0.0 Diagnostic-Code: X-Postfix; mail forwarding loop for asterisk-users@lists.digium.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
Re: [Asterisk-Users] SE Michigan asterisk users group
Our main office is near Lansing, but we have a person who lives in the AA are that would like to attend such a group. On Thu, Jun 22, 2006 at 04:27:02PM -0400, BerkHolz, Steven wrote: I am thinking of getting an asterisk user group together for either SE Michigan or just Metro-Detroit. How much interest in asterisk in Michigan is there on this list? I am already on the board of glimasoutheast, with is a group for technology professionals. (very broad range) It is a spin-off from Automation Alley, which is SE Michigan's version of Silicone Valley. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
OK Marco, irei efetuar os testes. Se você quiser, posso lhe ajudar no forum, estou a disposição. Assim que você criar as contas avise para podermos já ir colaborando. Saudações Josué 2006/6/26, Marco Mouta [EMAIL PROTECTED]: Bom dia,On 6/26/06, Josué Conti [EMAIL PROTECTED] wrote: Marco, bom dia. Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo externo?Sim é um software da Uplink, disponível para download gratuitamente, n garanto q seja freeware (talvez tenha limitações esta versao free)Podes ver a demo no site:http://asteriskpt.blogspot.comSe tudo estiver ok, deverás ouvir Musiconhold de um asterisk q tenho em casa ( n estou la agora). É freeware? Podemos seguir com o projeto Asterisk-PT?Claro que sim! http://asteriskpt.blogspot.comPodes por posts la, vou criar contas para podermos cooperar no blog. Se preferirem um site ou outra solução, estou aberto a sugestões. Saudações Josué 2006/6/26, John Klimek [EMAIL PROTECTED] : I agree whole-heartedly.If I could run this on my dedicated Asterisk machine it would be perfect...On 6/28/06, Matthias Fechner [EMAIL PROTECTED] wrote: Hi Marco, Marco Mouta wrote:Please feel free to contact me if you have more ideas to improve thissolution, currently i didn't test more than one simultaneous calls incoming and outgoing through Skype. get it running on unix so you can run it on the asterisk server. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --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 ___ --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 --Com os melhores cumprimentos,Marco Mouta___--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
Re: [Asterisk-Users] This is getting really annoying - re: POSTFIX
Matt wrote: What on earth is going on with the list?!?! Some of my messages never make it... then days later I get something like this back: Final-Recipient: rfc822; asterisk-users@lists.digium.com Action: failed Status: 5.0.0 Diagnostic-Code: X-Postfix; mail forwarding loop for asterisk-users@lists.digium.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 And I thought it was just me, or maybe gmail. I've seen very little traffic since last Wednesday or so. Bob... Bob... ___ --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
Re: [Asterisk-Users] Gizmo and Asterisk analysis
On Mon, Jun 26, 2006 at 06:06:01PM +0800, Dinesh Nair wrote: On 06/25/06 19:01 Roy Sigurd Karlsbakk said the following: seem to pass all SIP and RTP traffic through their own servers... See http://karlsbakk.net/asterisk/gizmo-project.php for details interesting. but isnt Gizmo an open source client ? No. As with the good tradition of Michael Robertson: lots of pretty words and much proprietary software. It's based on x-lite. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.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
[Asterisk-Users] chan_sip.c: Insufficient information for SDP
Hi! I'am often see this WARNINGs in messages file. What does it mean? Jun 26 16:59:00 WARNING[62792] chan_sip.c: Insufficient information for SDP (m = '', c = '') Jun 26 16:59:01 WARNING[62792] chan_sip.c: Insufficient information for SDP (m = '', c = '') And it seems that at this time I can't hear my peer correspondent. Thanks! -- DSS5-RIPE DSS-RIPN 2:550/[EMAIL PROTECTED] 2:550/[EMAIL PROTECTED] xmpp:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://neva.vlink.ru/~dsh/ ___ --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
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote: Marco, bom dia. Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo externo? É freeware? Podemos seguir com o projeto Asterisk-PT? English, please, folks. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.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
[Asterisk-Users] Asterisk ACD with Polycom IP501
Hi, Has anybody got the polycom acd function to work? I have the following setup: Debian 3.1 - 2.6.8 linux zlib-1.1.4 libpri-1.2.3 zaptel- 1.2.6 Asterisk - the bweschke/polycom_acd_funtions branch version - I get one error when doing a make install about needing a newer version of libpri and zaptel, I got the above versions from asterisk.org, are there newer version anywhere else? In the sip.conf file I have set the agentlogin=yes and agentcbcontext=demo (demo as from extensions.conf context) I have setup an agent in agents.conf as ,1234,Name I have changed in the sip.cfg of the polycom phone: feature.15.name=acd-login-logout feature.15.enabled=1 feature.16.name=acd-agent-availability feature.16.enabled=1 and in the phone1.cfg of the polycom I'm only using line1 so made the changes below: reg.1.acd-login-logout=1 reg.1.acd-agent-available=1 I get the login button on the phone, and when I try and login with the agent it just goes back to login. Any help would be appreciated. Thanks, Dean Bath ___ --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
Re: [Asterisk-Users] This is getting really annoying - re: POSTFIX
On 08:51, Mon 26 Jun 06, Matt wrote: What on earth is going on with the list?!?! Some of my messages never make it... then days later I get something like this back: Final-Recipient: rfc822; asterisk-users@lists.digium.com Action: failed Status: 5.0.0 Diagnostic-Code: X-Postfix; mail forwarding loop for asterisk-users@lists.digium.com Thank god! I have been looking and grepping and reconfiguring my postfix during the last week because of this messages. I now know it's really not me. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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
Re: [Asterisk-Users] PCI or MiniPCI Hardware DSP for G.729, G.723.1 and/or GSM
On Saturday 24 June 2006 09:44, Paul Hewlett wrote: I would imagine that this would not solve any problems - the extra overhead of piping the data over the PCI bus would very quickly negate any speed gains of the DSP over the native Intel FPU. Additionally you would probably introduce extra latency. I did quite a lot of work on DSP co-processor boards and there was always a considerable startup time when all the data pipes had to be filled. Talk to Digium, they're releasing a PCI DSP board that does exactly this. I guess that the PCI overhead isn't as great as was first thought? Interesting times... -A. ___ --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
[Asterisk-Users] struggling with the g flag
If I have in my dialplan
[AgentQ]
exten = _XX.,1,Dial(Sip/{$exten},120,g)
exten = _XX.,2,NoOP(here we are)
where [AgentQ] is called by the queue command to a member added by
addqueuemember(Local/[EMAIL PROTECTED])
why don't I get to the NoOp if the agent hangs up during the
announcement message (to the agent) ?
I see in the app_dial.c program that the g flag is tested thus:
if ((ast_test_flag(peerflags, OPT_GO_ON)) (!chan-_softhangup)
(res != AST_PBX_KEEPALIVE))
res = 0;
So this would indicate that if all three of these conditions are met
then res would be set to 0, and things would behave how I want.
In chan_agent.c, the following line is where the agent has hung up
if (peer-_softhangup) {
/* Agent must have hung up */
ast_log(LOG_WARNING, Agent on %s hungup on the customer. They're
going to be pissed.\n, peer-name);
I see that in chan_agent the peer-_softhangup is true (I get the
message on the console) but the test in app_dial specifically tests to
see if chan-_softhangup is *not* true.
Why is that ?
Julian
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Re: [Asterisk-Users] SE Michigan asterisk users group
Count me in, my office is in Livonia, but I currently reside in the D. Someone should set up a mailing list for this.--Tom HaydenOn 6/26/06, Michael George [EMAIL PROTECTED] wrote:Our main office is near Lansing, but we have a person who lives in the AA are that would like to attend such a group.On Thu, Jun 22, 2006 at 04:27:02PM -0400, BerkHolz, Steven wrote: I am thinking of getting an asterisk user group together for either SE Michigan or just Metro-Detroit. How much interest in asterisk in Michigan is there on this list? I am already on the board of glimasoutheast, with is a group for technology professionals. (very broad range) It is a spin-off from Automation Alley, which is SE Michigan's version of Silicone Valley.---MThere are 10 kinds of people in this world:Those who can count in binary and those who cannot.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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
Re: [Asterisk-Users] This is getting really annoying - re: POSTFIX
Hello All. Accurately, my messages also are not being received for the list and the traffic of messages really is very low. It will be a problem of the list? Best Regards Josué 2006/6/26, Michiel van Baak [EMAIL PROTECTED]: On 08:51, Mon 26 Jun 06, Matt wrote: What on earth is going on with the list?!?! Some of my messages never make it... then days later I get something like this back: Final-Recipient: rfc822; asterisk-users@lists.digium.com Action: failed Status: 5.0.0 Diagnostic-Code: X-Postfix; mail forwarding loop for asterisk-users@lists.digium.comThank god!I have been looking and grepping and reconfiguring my postfix during the last week because of this messages.I now know it's really not me.--Michiel van Baak[EMAIL PROTECTED] http://michiel.vanbaak.euGnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BDWhy is it drug addicts and computer afficionados are both called users? ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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
Re: [Asterisk-Users] This is getting really annoying - re: POSTFIX
Michiel van Baak wrote: On 08:51, Mon 26 Jun 06, Matt wrote: What on earth is going on with the list?!?! Some of my messages never make it... then days later I get something like this back: Final-Recipient: rfc822; asterisk-users@lists.digium.com Action: failed Status: 5.0.0 Diagnostic-Code: X-Postfix; mail forwarding loop for asterisk-users@lists.digium.com Thank god! I have been looking and grepping and reconfiguring my postfix during the last week because of this messages. I now know it's really not me. Nope - me too. Some of my messages make it, but all get bounced, even the ones that do make it. W ___ --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
Re: [Asterisk-Users] Asterisk-1.2.9.1 with Siemens HiPath 4000
Hi Josué if the Siemens phone calls Asterisk, it didn't get a dial tone from Asterisk? Is it correct? if yes, this is depending of Asterisk which didn't generates a ringback messages as it expexts dial ton generation localy. So try this workaround for HiPath local dial ton generation: - Add option TR6Q(TRGT) to the class of trunk (COT) parameters hope it will help... rich --- Josué Conti [EMAIL PROTECTED] wrote: Hello all. I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9, is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me? Best Regards Josué ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
Re: [Asterisk-Users] best hardphone for Asterisk?
I guess I did not make my point clearly enough. I already do have just that. An spa-3000 with ALL internal analog phones on it's on FXO. But this gives just ONE extension for all phones. Yes I could get more FXS's and run seperate wires. So with that background what would be nice is a wireless device like the Panasonic cordless with one base and multiple phone capability that connected via ethernet and serves the phones. Just wishful thinking. I will stick with what I have until something useful, sylish, and less expensive arrives. Doug On Mon, 26 Jun 2006, Michael George wrote: On Mon, Jun 26, 2006 at 12:08:48AM -0400, Doug Crompton wrote: Still awfully pricey for home use and the styling is not there for a bedroom or many other areas of a modern home. What we need is a wireless sip phone modeled like the panasonic or uniden which allow multiple extension off of one base. The base would connect to the internet. The other problem is many of these phones require power, so even if you have backup for your central system the phone still needs to be on it. Power over ethernet would help. 1. If you have *, you don't necessarily need multiple handsets off of one base. 2. Cordless phones also require power 3. If the multi-handset cordless phone does suit your needs best, then get a SIP ATA device like a Sipura or IAXy and you should have your needs met. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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 Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.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
RE: [Asterisk-Users] Asterisk Startups
Yeah, that's what I like about Oz. Everyone knows everyone... miss you guys too! -Original Message- From: Rob Thomas [mailto:[EMAIL PROTECTED] Sent: Monday, June 26, 2006 2:58 AM To: asterisk-users Subject: RE: [Asterisk-Users] Asterisk Startups Well now would be a great time to come back, Doug! We miss you! 8) --Rob -Original Message- From: Douglas Garstang [mailto:[EMAIL PROTECTED] Sent: Monday, 26 June 2006 3:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Startups Paul, D'oh. The fact I left Sydney 5 years ago for the US might be a teeny complication. :P Doug. -Original Message- From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Sun 6/25/2006 11:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Asterisk Startups Douglas Garstang wrote: Does anyone know of any startups using Asterisk? What about established companies? Ones that are hiring would be nice :) Doug. -- -- ___ --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 We are always looking for good people - here in Melbourne. PaulH -- Paul Hales Technical Manager AsteriskIT www.asteriskit.com.au bus: 03 8320 8100 mob: 0434 673 529 ___ --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 ___ --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
Re: [Asterisk-Users] Asterisk ACD with Polycom IP501
Hi Dean - It should be working. If not, please email me a sip debug trace along with your /etc/asterisk/agents.conf and your /etc/asterisk/sip.conf. Thanks. BJ On 6/26/06, Dean @ INKnBITs [EMAIL PROTECTED] wrote: Hi, Has anybody got the polycom acd function to work? I have the following setup: Debian 3.1 - 2.6.8 linux zlib-1.1.4 libpri-1.2.3 zaptel- 1.2.6 Asterisk - the bweschke/polycom_acd_funtions branch version - I get one error when doing a make install about needing a newer version of libpri and zaptel, I got the above versions from asterisk.org, are there newer version anywhere else? In the sip.conf file I have set the agentlogin=yes and agentcbcontext=demo (demo as from extensions.conf context) I have setup an agent in agents.conf as ,1234,Name I have changed in the sip.cfg of the polycom phone: feature.15.name=acd-login-logout feature.15.enabled=1 feature.16.name=acd-agent-availability feature.16.enabled=1 and in the phone1.cfg of the polycom I'm only using line1 so made the changes below: reg.1.acd-login-logout=1 reg.1.acd-agent-available=1 I get the login button on the phone, and when I try and login with the agent it just goes back to login. Any help would be appreciated. Thanks, Dean Bath ___ --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 -- Bird's The Word Technologies, Inc. http://www.btwtech.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
[Asterisk-Users] 1.2.9.1 SIP/Local/Queue behaviours weird
Hi, Does any one experience that SIP phone to SIP phone (Polycom phone) calls cant hear each other, but Monitor application records both ends voices. It also happens in group pickup calls. Zap calls to queue (Local channel) also experience this problem (sometimes, our SIP phone cant hear any voice from incoming Zap calls when pickup, sometimes this happens after 10-50 seconds talk). It is weird. Jun 26 16:53:35 VERBOSE[8290] logger.c: -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/7188|30|trWwT) in new stack Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Setting NAT on RTP to 0 Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Setting NAT on VRTP to 0 Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Outgoing Call for 7188 Jun 26 16:53:35 VERBOSE[8290] logger.c: -- Called 7188 Jun 26 16:53:35 VERBOSE[8287] logger.c: -- Local/[EMAIL PROTECTED],1 is ringing Jun 26 16:53:35 DEBUG[2966] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Jun 26 16:53:35 DEBUG[2966] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Jun 26 16:53:35 DEBUG[2957] channel.c: Avoiding initial deadlock for 'SIP/7188-6b1f' Jun 26 16:53:35 VERBOSE[8290] logger.c: -- SIP/7188-6b1f is ringing Jun 26 16:53:37 DEBUG[2966] chan_sip.c: Acked pending invite 102 Jun 26 16:53:37 DEBUG[2966] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jun 26 16:53:37 DEBUG[2966] chan_sip.c: build_route: Contact hop: Jun 26 16:53:37 VERBOSE[8290] logger.c: -- SIP/7188-6b1f answered Local/[EMAIL PROTECTED],2 Jun 26 16:53:37 DEBUG[8287] app_queue.c: Dunno what to do with control type -1 Jun 26 16:53:37 VERBOSE[8287] logger.c: -- Local/[EMAIL PROTECTED],1 answered Zap/13-1 Jun 26 16:53:37 DEBUG[8287] chan_zap.c: Set option TONE VERIFY, mode: MUTECONF(1) on Zap/13-1 Jun 26 16:53:37 VERBOSE[8287] logger.c: -- Stopped music on hold on Zap/13-1 Jun 26 16:53:37 DEBUG[8287] channel.c: Scheduling timer at 0 sample intervals Jun 26 16:54:02 DEBUG[8290] channel.c: Didn't get a frame from channel: SIP/7188-6b1f Jun 26 16:54:02 DEBUG[8290] channel.c: Bridge stops bridging channels Local/[EMAIL PROTECTED],2 and SIP/7188-6b1f Jun 26 16:54:02 DEBUG[8290] chan_sip.c: update_call_counter(7188) - decrement call limit counter Jun 26 16:54:02 DEBUG[8290] app_dial.c: Exiting with DIALSTATUS=ANSWER. Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'dial' Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'exten-vm' Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'Local/[EMAIL PROTECTED],2' Jun 26 16:54:02 DEBUG[8290] res_monitor.c: monitor executing ( nice -n 19 soxmix /var/spool/asterisk/monitor/20060626-165333-1151304813.901-in.gsm /var/spool/asterisk/monitor/20060626-165333-1151304813.901-out.gsm /var/spool/asterisk/monitor/20060626-165333-1151304813.901.gsm rm -f /var/spool/asterisk/monitor/20060626-165333-1151304813.901-* ) Jun 26 16:54:02 DEBUG[8287] channel.c: Didn't get a frame from channel: Local/[EMAIL PROTECTED],1 Jun 26 16:54:02 DEBUG[8287] channel.c: Bridge stops bridging channels Zap/13-1 and Local/[EMAIL PROTECTED],1 Jun 26 16:54:02 VERBOSE[8287] logger.c: == Spawn extension (ext-queues, 7141, 6) exited non-zero on 'Zap/13-1' Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/13-1 Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Hangup: channel: 13 index = 0, normal = 27, callwait = -1, thirdcall = -1 Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Not yet hungup... Calling hangup once with icause, and clearing call Jun 26 16:54:02 DEBUG[8287] chan_zap.c: disabled echo cancellation on channel 13 Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/13-1 Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Updated conferencing on 13, with 0 conference users Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/13-1 Jun 26 16:54:02 DEBUG[8287] chan_zap.c: disabled echo cancellation on channel 13 Jun 26 16:54:02 VERBOSE[8287] logger.c: -- Hungup 'Zap/13-1' Isaac Xiao ___ --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
RE: [Asterisk-Users] SE Michigan asterisk users group
I live in Southfield, our main office is in Pontiac, but our Colo is in Southfield. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael George Sent: Monday, June 26, 2006 8:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SE Michigan asterisk users group Our main office is near Lansing, but we have a person who lives in the AA are that would like to attend such a group. On Thu, Jun 22, 2006 at 04:27:02PM -0400, BerkHolz, Steven wrote: I am thinking of getting an asterisk user group together for either SE Michigan or just Metro-Detroit. How much interest in asterisk in Michigan is there on this list? I am already on the board of glimasoutheast, with is a group for technology professionals. (very broad range) It is a spin-off from Automation Alley, which is SE Michigan's version of Silicone Valley. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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
Re: [Asterisk-Users] 1.2.9.1 SIP/Local/Queue behaviours weird
I have seen this when Polycom has to communicate with none polycom phones and a transfer is initiated to a polycom, unless the Polycom presses Hold and then unhold, there is only one way audio, this is without NAT involved. There might also be other cases when this happens. My workaround is to add canreinvite=no On 6/26/06, Isaac Xiao [EMAIL PROTECTED] wrote: Hi, Does any one experience that SIP phone to SIP phone (Polycom phone) calls can't hear each other, but Monitor application records both end's voices. It also happens in group pickup calls. Zap calls to queue (Local channel) also experience this problem (sometimes, our SIP phone can't hear any voice from incoming Zap calls when pickup, sometimes this happens after 10-50 seconds' talk). It is weird. Jun 26 16:53:35 VERBOSE[8290] logger.c: -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/7188|30|trWwT) in new stack Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Setting NAT on RTP to 0 Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Setting NAT on VRTP to 0 Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Outgoing Call for 7188 Jun 26 16:53:35 VERBOSE[8290] logger.c: -- Called 7188 Jun 26 16:53:35 VERBOSE[8287] logger.c: -- Local/[EMAIL PROTECTED],1 is ringing Jun 26 16:53:35 DEBUG[2966] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Jun 26 16:53:35 DEBUG[2966] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Jun 26 16:53:35 DEBUG[2957] channel.c: Avoiding initial deadlock for 'SIP/7188-6b1f' Jun 26 16:53:35 VERBOSE[8290] logger.c: -- SIP/7188-6b1f is ringing Jun 26 16:53:37 DEBUG[2966] chan_sip.c: Acked pending invite 102 Jun 26 16:53:37 DEBUG[2966] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jun 26 16:53:37 DEBUG[2966] chan_sip.c: build_route: Contact hop: Jun 26 16:53:37 VERBOSE[8290] logger.c: -- SIP/7188-6b1f answered Local/[EMAIL PROTECTED],2 Jun 26 16:53:37 DEBUG[8287] app_queue.c: Dunno what to do with control type -1 Jun 26 16:53:37 VERBOSE[8287] logger.c: -- Local/[EMAIL PROTECTED],1 answered Zap/13-1 Jun 26 16:53:37 DEBUG[8287] chan_zap.c: Set option TONE VERIFY, mode: MUTECONF(1) on Zap/13-1 Jun 26 16:53:37 VERBOSE[8287] logger.c: -- Stopped music on hold on Zap/13-1 Jun 26 16:53:37 DEBUG[8287] channel.c: Scheduling timer at 0 sample intervals Jun 26 16:54:02 DEBUG[8290] channel.c: Didn't get a frame from channel: SIP/7188-6b1f Jun 26 16:54:02 DEBUG[8290] channel.c: Bridge stops bridging channels Local/[EMAIL PROTECTED],2 and SIP/7188-6b1f Jun 26 16:54:02 DEBUG[8290] chan_sip.c: update_call_counter(7188) - decrement call limit counter Jun 26 16:54:02 DEBUG[8290] app_dial.c: Exiting with DIALSTATUS=ANSWER. Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'dial' Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'exten-vm' Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'Local/[EMAIL PROTECTED],2' Jun 26 16:54:02 DEBUG[8290] res_monitor.c: monitor executing ( nice -n 19 soxmix /var/spool/asterisk/monitor/20060626-165333-1151304813.901-in.gsm /var/spool/asterisk/monitor/20060626-165333-1151304813.901-out.gsm /var/spool/asterisk/monitor/20060626-165333-1151304813.901.gsm rm -f /var/spool/asterisk/monitor/20060626-165333-1151304813.901-* ) Jun 26 16:54:02 DEBUG[8287] channel.c: Didn't get a frame from channel: Local/[EMAIL PROTECTED],1 Jun 26 16:54:02 DEBUG[8287] channel.c: Bridge stops bridging channels Zap/13-1 and Local/[EMAIL PROTECTED],1 Jun 26 16:54:02 VERBOSE[8287] logger.c: == Spawn extension (ext-queues, 7141, 6) exited non-zero on 'Zap/13-1' Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/13-1 Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Hangup: channel: 13 index = 0, normal = 27, callwait = -1, thirdcall = -1 Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Not yet hungup... Calling hangup once with icause, and clearing call Jun 26 16:54:02 DEBUG[8287] chan_zap.c: disabled echo cancellation on channel 13 Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/13-1 Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Updated conferencing on 13, with 0 conference users Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/13-1 Jun 26 16:54:02 DEBUG[8287] chan_zap.c: disabled echo cancellation on channel 13 Jun 26 16:54:02 VERBOSE[8287] logger.c: -- Hungup 'Zap/13-1' Isaac Xiao ___ --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
[Asterisk-Users] RE: Voice calls sent to fax extension
I thought it might be an inadvertent button press, but none of the keys (on my phone at least) are recognized by Asterisk as fax tones. This has happened to two different users getting calls from different people using different equipment. Does anyone else see this behavior occasionally? Paul -Original Message- Date: Fri, 23 Jun 2006 15:29:31 -0400 From: Bill Gibbs [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Voice calls sent to fax extension To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Maybe their fat jowls hit a few buttons on the keypad and sent the fax tone down the line and they didn't realize it? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul A. Pringle Sent: Friday, June 23, 2006 2:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voice calls sent to fax extension I have a situation that has repeated itself a few times. Someone calls into Asterisk and is connected with a voice extension. At some point during the call, the log shows chan_zap.c: DTMF digit: f on Zap/2-1. At this point, the call is redirected to receive a fax and the Asterisk voice extension is hung up. The users report that there were no noticable tones heard just before the cutover, so I'm not sure what's going on. Is there a way to disable detection of faxes after the voicecall is initiated? We're running a Digium card to convert our analog trunks if that makes any difference. Thanks! Paul ___ --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
[Asterisk-Users] Is there a way to reinstall the AMP
Hi all. Today I have tried to connect to the AMP with http://myserverip but I can not connect to the AMP (it sends me out of my network). What would be happening?. The last thing I did is to try to change the digital receptionist manually. Is there a way to re-install the amp? Thanks __ Correo Yahoo! Espacio para todos tus mensajes, antivirus y antispam ¡gratis! Regístrate ya - http://correo.yahoo.com.mx/ ___ --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
Re: [Asterisk-Users] SE Michigan asterisk users group
I'm also in the area, near Southfield. I'd be interested as well. On 6/22/06, BerkHolz, Steven [EMAIL PROTECTED] wrote: I am thinking of getting an asterisk user group together for either SEMichigan or just Metro-Detroit. How much interest in asterisk in Michigan is there on this list?I am already on the board of glimasoutheast, with is a group fortechnology professionals. (very broad range)It is a spin-off from Automation Alley, which is SE Michigan's version of Silicone Valley.--Stevenhttp://www.glimasoutheast.org___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users -- Is it something someone said, was it something someone said? ___ --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
Re: [Asterisk-Users] caller id
I am not sure, I will check. If I dont', and get it started, will it just start working? If not, what do I need to do?ThanksJoshua West [EMAIL PROTECTED] wrote: Do you have the Caller ID feature with your telephone service package?sdgesa gaeharth wrote: How can I get the external caller id to show on the polycom 501 phones. Currently, when someone calls our office, we only see the word "asterisk" in the caller id. This is our set up: VOIP(polycom)---Asterisk 1.2.4---PSTN Thanks Yahoo! Groups gets better. Check out the new email design. Plus theres much more to come. ___ --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 -- Joshua WestLinux Infrastructure EngineerBoston Engineering Corporationhttp://www.boston-engineering.com___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Next-gen email? Have it all with the all-new Yahoo! Mail Beta.___ --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
[Asterisk-Users] Re: Voice calls sent to fax extension
I do get random DTMF tones. They have been to sparse to diagnose if there was anything common with those calls. When it is noticed and I look it up in the logs, it may be any digits. We see this on zap(PRI) to zap(PRI) bridged calls too. We are using a TE411P. -- -- Steven http://www.glimasoutheast.org Paul A. Pringle [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] I thought it might be an inadvertent button press, but none of the keys (on my phone at least) are recognized by Asterisk as fax tones. This has happened to two different users getting calls from different people using different equipment. Does anyone else see this behavior occasionally? Paul -Original Message- Date: Fri, 23 Jun 2006 15:29:31 -0400 From: Bill Gibbs [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Voice calls sent to fax extension To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Maybe their fat jowls hit a few buttons on the keypad and sent the fax tone down the line and they didn't realize it? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul A. Pringle Sent: Friday, June 23, 2006 2:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voice calls sent to fax extension I have a situation that has repeated itself a few times. Someone calls into Asterisk and is connected with a voice extension. At some point during the call, the log shows chan_zap.c: DTMF digit: f on Zap/2-1. At this point, the call is redirected to receive a fax and the Asterisk voice extension is hung up. The users report that there were no noticable tones heard just before the cutover, so I'm not sure what's going on. Is there a way to disable detection of faxes after the voicecall is initiated? We're running a Digium card to convert our analog trunks if that makes any difference. Thanks! Paul ___ --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
Re: [Asterisk-Users] RE: Voice calls sent to fax extension
Paul A. Pringle wrote: I thought it might be an inadvertent button press, but none of the keys (on my phone at least) are recognized by Asterisk as fax tones. This has happened to two different users getting calls from different people using different equipment. Does anyone else see this behavior occasionally? Yes, which is why I disable faxdetect entirely. My sister-in-law was constantly being detected as a fax machine several minutes into conversations with my wife. As funny as that may seem at first ... those two eventually make it a not-so-funny situation for me. The fax detection should, in theory, only be looking for CNG tones and should, in theory, only be looking for them during the first several seconds of a call. Analogue DTMF tones should not be detectable as CNG, in theory. Lee. ___ --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
RE: [Asterisk-Users] RE: Voice calls sent to fax extension
yes. Wind whistling in a car, female voices at a particular pitch and volume, fax machine running in the background of a voice call with the speaker on. It happens. Whether this is a problem or not depends on your pain threshold. I get a couple reports a week, which means that it actually happens ten times a couple times a week, so twenty times a week, and I process ~20K calls a week, so it happens to me .1 % of the time. Is this a problem for me? Nah. Is it a problem for you? Maybe - what's your pain threshold? ps fwiw, this behavior will happen with any device that listens inline for a CNG tone, so it's not just an Asterisk thing -Original Message- From: Paul A. Pringle [mailto:[EMAIL PROTECTED] Sent: Monday, June 26, 2006 8:54 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RE: Voice calls sent to fax extension I thought it might be an inadvertent button press, but none of the keys (on my phone at least) are recognized by Asterisk as fax tones. This has happened to two different users getting calls from different people using different equipment. Does anyone else see this behavior occasionally? Paul -Original Message- Date: Fri, 23 Jun 2006 15:29:31 -0400 From: Bill Gibbs [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Voice calls sent to fax extension To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Maybe their fat jowls hit a few buttons on the keypad and sent the fax tone down the line and they didn't realize it? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul A. Pringle Sent: Friday, June 23, 2006 2:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voice calls sent to fax extension I have a situation that has repeated itself a few times. Someone calls into Asterisk and is connected with a voice extension. At some point during the call, the log shows chan_zap.c: DTMF digit: f on Zap/2-1. At this point, the call is redirected to receive a fax and the Asterisk voice extension is hung up. The users report that there were no noticable tones heard just before the cutover, so I'm not sure what's going on. Is there a way to disable detection of faxes after the voicecall is initiated? We're running a Digium card to convert our analog trunks if that makes any difference. Thanks! Paul ___ --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
RE: [Asterisk-Users] RE: Voice calls sent to fax extension
I also had an intermittent problem, on average one or two faxes a week, that were not recongnzed as a fax. Then I switched phone companies and have not had that problem since. It has been over 2 months. I addition, my echo problem has been practically eliminated and overall voice quality is better. I believe that it is because of better levels of RX gain for fax recognition, but don't know for sure. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paul A. Pringle Sent: Monday, June 26, 2006 10:54 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RE: Voice calls sent to fax extension I thought it might be an inadvertent button press, but none of the keys (on my phone at least) are recognized by Asterisk as fax tones. This has happened to two different users getting calls from different people using different equipment. Does anyone else see this behavior occasionally? Paul -Original Message- Date: Fri, 23 Jun 2006 15:29:31 -0400 From: Bill Gibbs [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Voice calls sent to fax extension To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Maybe their fat jowls hit a few buttons on the keypad and sent the fax tone down the line and they didn't realize it? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul A. Pringle Sent: Friday, June 23, 2006 2:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voice calls sent to fax extension I have a situation that has repeated itself a few times. Someone calls into Asterisk and is connected with a voice extension. At some point during the call, the log shows chan_zap.c: DTMF digit: f on Zap/2-1. At this point, the call is redirected to receive a fax and the Asterisk voice extension is hung up. The users report that there were no noticable tones heard just before the cutover, so I'm not sure what's going on. Is there a way to disable detection of faxes after the voicecall is initiated? We're running a Digium card to convert our analog trunks if that makes any difference. Thanks! Paul ___ --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
[Asterisk-Users] AEL scripting, CUT use and string concatenation
Hi to all,
i'm wondering to realize a dynamic macro that can take the number of
extensions to RING,the ring type and all the parameter in a dynamic way.
I have done this code to test it:
macro pbx-ring-group-ael(pbx_id,num_int,ring_type,timeout,ext_string) {
//; pbx_id = Id of PBX in the DB
//; num_int = Quantity of extensions to ring
//; ring_type = Kind of RING (C=contemporaneous S=sequential)
//; timeout = Amount of time to ring
//; ext_string = String with extension numbers like 101-102-103-104-105
if(${ring_type}=C) {
for (x=1 ; ${x} = ${num_int} ; x=${x} + 1) {
int=${CUT(ext_string,,${x})};
if(${x} = 1) {
dialstring=SIP/${pbx_id}-${int};
} else {
dialstring=${dialstring}SIP/${pbx_id}-${int};
};
if(${x} = ${num_int}) {
dialstring=${dialstring}|${timeout};
};
NoOp(STRING ${dialstring});
};
};
Hangup();
};
I'm getting problems both in the CUT expression and the concatenation of
strings due to the presence of ,/,- in it. I think something can be
done with double quote but it will be inserted as part of the string, so
the concatenation will fail.
For the CUT i don't know what is the problem. I tried with
CUT(int=(ext_string,,${x}) too but without success.
This is the dialplan resulting from the expansion of the ael script:
show dialplan macro-pbx-ring-group-ael
[ Context 'macro-pbx-ring-group-ael' created by 'pbx_ael' ]
's' =1. Set(pbx_id=${ARG1})[pbx_ael]
2. Set(num_int=${ARG2}) [pbx_ael]
3. Set(ring_type=${ARG3}) [pbx_ael]
4. Set(timeout=${ARG4}) [pbx_ael]
5. Set(ext_string=${ARG5})[pbx_ael]
6. GotoIf($[ ${ring_type}=C ]?7:22) [pbx_ael]
7. Set(x=$[ 1 ]) [pbx_ael]
8. GotoIf($[ ${x} = ${num_int} ]?9:21) [pbx_ael]
9. Set(x=$[ ${x} + 1 ]) [pbx_ael]
10. Set(int=$[ ${CUT(ext_string,,${x})} ]) [pbx_ael]
11. GotoIf($[ ${x} = 1 ]?12:14) [pbx_ael]
12. Set(dialstring=$[ SIP/${pbx_id}-${int} ]) [pbx_ael]
13. Goto(15) [pbx_ael]
14. Set(dialstring=$[
${dialstring}SIP/${pbx_id}-${int} ]) [pbx_ael]
15. NoOp(Finish
if-for-if-pbx-ring-group-ael-6-7-11) [pbx_ael]
16. GotoIf($[ ${x} = ${num_int} ]?17:18) [pbx_ael]
17. Set(dialstring=$[ ${dialstring}|${timeout} ])
[pbx_ael]
18. NoOp(Finish
if-for-if-pbx-ring-group-ael-6-7-16) [pbx_ael]
19. NoOp(STRING ${dialstring}) [pbx_ael]
20. Goto(8)[pbx_ael]
21. NoOp(Finish for-if-pbx-ring-group-ael-6-7)
[pbx_ael]
22. NoOp(Finish if-pbx-ring-group-ael-6) [pbx_ael]
23. Hangup()[pbx_ael]
-= 1 extension (23 priorities) in 1 context. =-
This is the log of errors:
-- Executing Macro(SIP/1234-100-b263,
pbx-ring-group-ael|1234|5|C|20|101-102-103-104-105) in new stack
-- Executing Set(SIP/1234-100-b263, pbx_id=1234) in new stack
-- Executing Set(SIP/1234-100-b263, num_int=5) in new stack
-- Executing Set(SIP/1234-100-b263, ring_type=C) in new stack
-- Executing Set(SIP/1234-100-b263, timeout=20) in new stack
-- Executing Set(SIP/1234-100-b263,
ext_string=101-102-103-104-105) in new stack
-- Executing GotoIf(SIP/1234-100-b263, 1?7:22) in new stack
-- Goto (macro-pbx-ring-group-ael,s,7)
-- Executing Set(SIP/1234-100-b263, x=1) in new stack
-- Executing GotoIf(SIP/1234-100-b263, 1?9:21) in new stack
-- Goto (macro-pbx-ring-group-ael,s,9)
-- Executing Set(SIP/1234-100-b263, x=2) in new stack
Jun 26 17:17:24 WARNING[31282]: ast_expr2.fl:176 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected $end, expecting
TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:
^
Jun 26 17:17:24 WARNING[31282]: ast_expr2.fl:180 ast_yyerror: If you
have questions, please refer to doc/README.variables in the asterisk source.
-- Executing Set(SIP/1234-100-b263, int=0) in new stack
-- Executing GotoIf(SIP/1234-100-b263, 0?12:14) in new stack
-- Goto (macro-pbx-ring-group-ael,s,14)
Jun 26 17:17:24 WARNING[31282]: ast_expr2.fl:176 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected TOK_AND, expecting
TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:
SIP/1234-0
^
Jun 26 17:17:24 WARNING[31282]: ast_expr2.fl:180 ast_yyerror: If you
have questions, please refer to
Re: [Asterisk-Users] Asterisk-1.2.9.1 with Siemens HiPath 4000
Hi Richard. Thank you very much for its attention. In the reality what is occurring is that in some originated calls of the HiPath with destination to the Asterisk they are being without the dumb andrings. I do not have this parameter in my HiPath 4000, what I have seemed in the COT is TR6T (1tr6 isdn tie link) would be this parameter? Best Regards Josué 2006/6/26, richard Coco [EMAIL PROTECTED]: Hi Josuéif the Siemens phone calls Asterisk, it didn't get adial tone from Asterisk? Is it correct? if yes, this is depending of Asterisk which didn'tgenerates a ringback messages as it expexts dial tongeneration localy. So try this workaround for HiPathlocal dial ton generation:- Add option TR6Q(TRGT) to the class of trunk (COT) parametershope it will help...rich--- Josué Conti [EMAIL PROTECTED] wrote: Hello all.I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9, is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me? Best Regards Josué ___ --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__Do You Yahoo!? Tired of spam?Yahoo! Mail has the best spam protection aroundhttp://mail.yahoo.com___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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
RE: [Asterisk-Users] RE: Voice calls sent to fax extension
Yes, which is why I disable faxdetect entirely. My sister-in-law was constantly being detected as a fax machine several minutes into conversations with my wife. As funny as that may seem at first ... those two eventually make it a not-so-funny situation for me. lol, Spousal Acceptance Factor, I have found, is the cornerstone of any Asterisk feature. Seriously, if I have a great idea and I am going to introduce it to my users, I put it on my Asterisk box at home and let my wife use it. If she says: That doesn't suck then I am golden with my users. Note that she doesn't say: Wow thats a great feature she simply says: That doesn't suck which to her is high praise. ___ --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
Re: [Asterisk-Users] RE: Voice calls sent to fax extension
Surely once the call has been bridged the fax detection should turn off ? Julian Colin Anderson wrote: yes. Wind whistling in a car, female voices at a particular pitch and volume, fax machine running in the background of a voice call with the speaker on. It happens. Whether this is a problem or not depends on your pain threshold. I get a couple reports a week, which means that it actually happens ten times a couple times a week, so twenty times a week, and I process ~20K calls a week, so it happens to me .1 % of the time. Is this a problem for me? Nah. Is it a problem for you? Maybe - what's your pain threshold? ps fwiw, this behavior will happen with any device that listens inline for a CNG tone, so it's not just an Asterisk thing -Original Message- From: Paul A. Pringle [mailto:[EMAIL PROTECTED] Sent: Monday, June 26, 2006 8:54 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RE: Voice calls sent to fax extension I thought it might be an inadvertent button press, but none of the keys (on my phone at least) are recognized by Asterisk as fax tones. This has happened to two different users getting calls from different people using different equipment. Does anyone else see this behavior occasionally? Paul -Original Message- Date: Fri, 23 Jun 2006 15:29:31 -0400 From: Bill Gibbs [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Voice calls sent to fax extension To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Maybe their fat jowls hit a few buttons on the keypad and sent the fax tone down the line and they didn't realize it? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul A. Pringle Sent: Friday, June 23, 2006 2:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voice calls sent to fax extension I have a situation that has repeated itself a few times. Someone calls into Asterisk and is connected with a voice extension. At some point during the call, the log shows chan_zap.c: DTMF digit: f on Zap/2-1. At this point, the call is redirected to receive a fax and the Asterisk voice extension is hung up. The users report that there were no noticable tones heard just before the cutover, so I'm not sure what's going on. Is there a way to disable detection of faxes after the voicecall is initiated? We're running a Digium card to convert our analog trunks if that makes any difference. Thanks! Paul ___ --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 ___ --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
[Asterisk-Users] Pickup zap issue
Hi, As you can see in this log the problem is very new to me.. Connected to Asterisk 1.2.5 currently running on volcano (pid = 7874) volcano*CLI set verbose 4 Verbosity was 0 and is now 4 -- Starting simple switch on 'Zap/2-1' -- Executing Dial(Zap/2-1, SIP/180|60) in new stack -- Called 180 -- SIP/180-d11c is ringing -- SIP/180-d11c answered Zap/2-1 Now everything looks good, BUT when I pickup the handset on extension 180 nothing happens as you can see in the log Asterisk notices that 180 answers but the person calling in on the ZAP interface still hears the ringing tone, it seems that Asterisk does not bridge the call or physically answers it on the ZAP interface. Anybody had issues like this or could have an idea where to start looking for the problem? Fred ___ --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
RE: [Asterisk-Users] SE Michigan asterisk users group
Ok, I count at least 4. Just lets propose when where for the first meeting group, and start to think about issues discussion. Tom, what to we need for the mailing list? I can do something about that. Carlos Alperin From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon RadonSent: Monday, June 26, 2006 11:12 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] SE Michigan asterisk users group I'm also in the area, near Southfield. I'd be interested as well. On 6/22/06, BerkHolz, Steven [EMAIL PROTECTED] wrote: I am thinking of getting an asterisk user group together for either SEMichigan or just Metro-Detroit.How much interest in asterisk in Michigan is there on this list?I am already on the board of glimasoutheast, with is a group fortechnology professionals. (very broad range)It is a spin-off from Automation Alley, which is SE Michigan's version of Silicone Valley.--Stevenhttp://www.glimasoutheast.org___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users -- Is it something someone said, was it something someone said? ___ --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
[Asterisk-Users] MeetMe Volume Issues
Hi, I'm using the latest 1.2 release of Asterisk and I've noticed that one of the releases of Zaptel or Asterisk in the past few months seems to have introduced a problem with MeetMe. Here are the symptoms: - Very high volume for internal IP phone users - Very low volume for incoming analog callers - Analog callers can not hear each other in conference This seems to happen with the 4-port and 24-port TDM cards sold by Digium. Has anyone else experienced similar problems? Thanks! -- Justin Tunney ___ --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
Re: [Asterisk-Users] Asterisk-1.2.9.1 with Siemens HiPath 4000
Jun 26 12:43:16 WARNING[31148]: chan_zap.c:8386 pri_dchannel: Ring requested on unconfigured channel 0/16 span 1Inoticed this message in the CLI, when I tried to effect one call of HiPath 4000 for asterisk. Ring occurred, however when the voicemail of asterisk took care of call it was dumb, without no sound. I thank the attention RegardsJosué 2006/6/26, Josué Conti [EMAIL PROTECTED]: Hi Richard. Thank you very much for its attention. In the reality what is occurring is that in some originated calls of the HiPath with destination to the Asterisk they are being without the dumb andrings. I do not have this parameter in my HiPath 4000, what I have seemed in the COT is TR6T (1tr6 isdn tie link) would be this parameter? Best Regards Josué 2006/6/26, richard Coco [EMAIL PROTECTED]: Hi Josuéif the Siemens phone calls Asterisk, it didn't get adial tone from Asterisk? Is it correct? if yes, this is depending of Asterisk which didn'tgenerates a ringback messages as it expexts dial tongeneration localy. So try this workaround for HiPathlocal dial ton generation:- Add option TR6Q(TRGT) to the class of trunk (COT) parametershope it will help...rich--- Josué Conti [EMAIL PROTECTED] wrote: Hello all.I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9, is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me? Best Regards Josué ___ --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__Do You Yahoo!? Tired of spam?Yahoo! Mail has the best spam protection around http://mail.yahoo.com___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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
[Asterisk-Users] asterisk-stat display problems
Hey all, having a terrible time with asterisk-stat -- it runs, server is fine, but some of the pages don't display properly/at all --- I think this is a code problem with them, but not sure. I thought everyone loved the asterisk-stat package? See below problems. Any ideas? Areski hasn't replied to me since -- Chris - Original Message - From: Chris Earle (CBL) To: Areski Sent: Tuesday, June 13, 2006 6:15 PM Subject: Re: CDR-Analyser version question Thank you for the reply; I see now that the main file cdr.php does work with argument ?s=1, 2, etc but when s=0, does not load I get an Apache error : relocation error: /usr/lib/php4/20020429/gd.so: undefined symbol: gdFontCacheShutdown Not sure if that means anything important; Also, in the new Asterisk-Stat feature pages like Calls Compare (s=2), the pages do not complete their output -- no search button displayed, stops outputting radio buttons for UserField row etc So at this point, only the main Call-log page (s=1) works. I am using Debian with php 4.4.1 Mysql ver 12.22, Distrib 4.0.24 GD Library is 2.0.33 I think Any input you can pass along would be much appreciated! I am comfortable with php so if you want me to modify sourcecode that is fine Thanks! - Original Message - From: Areski To: Chris Earle (CBL) Sent: Sunday, May 28, 2006 7:11 PM Subject: Re: CDR-Analyser version question No there is no asterisk requirement to make asterisk-stat. Indeed the soft is only based on the cdr database. If you have an error you can give me more info, I may help you. Rgds, Areski On 5/25/06, Chris Earle (CBL) wrote: Hi there, quick question: Does asterisk-stat v2.0.1 require Asterisk 1.2+ ? I am using Asterisk 1.0.x and can't get it to load the cdr.php properly so I downgraded to v1.3 and it works... Let me know if there's an asterisk version requirement for each version of the CDR Analyser Thanks! -- Chris Earle snip -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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
RE: [Asterisk-Users] RE: Voice calls sent to fax extension
Surely once the call has been bridged the fax detection should turn off ? I'd like to find out a way it can be done, can anyone else comment? ___ --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
Re: [Asterisk-Users] Signaling and media
26 jun 2006 kl. 10.54 skrev Jean-Michel Hiver: Johansson Olle E a écrit : 26 jun 2006 kl. 07.10 skrev Martin Joseph: On Jun 25, 2006, at 4:11 PM, Jean-Michel Hiver wrote: Hi List, Is there a way to tell asterisk to only accept SIP streams from the same IP address that is used for signaling? SIP streams are signalling... Sorry, I was talking about the media. Have you tested the ACL features in sip.conf - accept/deny ? Any pointers on these ACLs? Check permit and deny in sip.conf. /O --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --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
[Asterisk-Users] registering a Motorola vt1005
Has anyone successfully registered a moto vt1005 to asterisk. If so, how? Brandon Warner Assistant Director of NOC Services Dark Fiber Solutions 600 1/2 Grant Ave. York, NE 68467 Office: 402-362-3334 Cell:402-366-2087 "The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from all computers." ___ --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
[Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY
Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? I called Polycom tech support, who where utterly useless. Of course Polycom won't officially support it anyway, as they only support Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite some time. We have about 35 phones and it's happening on most (also on the few running SIP software 1.6.6). SIP Software version: 1.6.3.0067 BootROM version: 2.6.2.0032 Reliably Transmitting (no NAT) to xxx.187.128.95:5060: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport From: sip:[EMAIL PROTECTED];tag=as6fd80d1b To: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 114 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 371 ?xml version=1.0? !DOCTYPE presence PUBLIC -//IETF//DTD RFC XPIDF 1.0//EN xpidf.dtd presence presentity uri=sip:[EMAIL PROTECTED];method=SUBSCRIBE / atom id=2944026 address uri=sip:[EMAIL PROTECTED];user=ip priority=0.80 status status=open / msnsubstatus substatus=online / /address /atom /presence -- SIP read from xxx.187.128.95:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport From: sip:[EMAIL PROTECTED];tag=as6fd80d1b To: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007 CSeq: 114 NOTIFY Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Event: presence User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Content-Length: 0 Doug. ___ --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
[Asterisk-Users] STUN?
Hi all, Could someone point at resources for running Asterisk behind a firewall. STUN keeps coming up but, alas, Im easily confused. J Ray ___ --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
Re: [Asterisk-Users] Meetme max users
thanks we are planing to have around 50-60 users in 1 room. We've had over 100 participants spread across 30 meetme rooms on a single server before, and the most we've had in a single meetme room is 46. I don't know of a hard limit for meetme participants and I haven't seen a limit in the code. You would most likely be limited by the resources on your server I would guess. MATT--- On 6/23/06, Bartosz Wegrzyn - asterisk [EMAIL PROTECTED] wrote: Does anyone knows what is the max of users that meefme can handle. I am using Iax2 clients to connect to the conference. 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 ___ --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
Re: [Asterisk-Users] This is getting really annoying - re: POSTFIX
On Jun 26, 2006, at 5:51 AM, Matt wrote: What on earth is going on with the list?!?! Some of my messages never make it... then days later I get something like this back: I'm hoping this was a transient issue. I saw this too with a couple of posts, but it's been ok since. Marty ___ --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
[Asterisk-Users] Soekris net4801-50 + IAXY
Hi, I'm having an issue with a soekris net4801 board and a S101i IAXy device. When I connect a successfully provisioned IAXy directly via a crossover cable into an ethernet port of the soekris, the link led turns on orange so i'ts 10Mb and the activity led blinks like if there is some action going on but when I try 'tcpdump -nettti sis1' I see nothing going on, no received packets. When I plug a regular PC on the same ethernet port there I can see all the traffic going on. I'm really stuck on this one. Help me please! Regards. Juan Luis Moyano ___ --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
[Asterisk-Users] EuroISDN and Sangoma Card
Hi List, Just a little question about a notice from asterisk I don't understand: Here is what I have as soon as I place a call on a E1 line with an a104D Sangoma Card ( asterisk 1.2.9.1 ) : Jun 26 18:57:24 NOTICE[16489] channel.c: Don't know what to do with control frame 15 Does Anyone has a clue of how to get rid of that ? May it's because I use the HDLC decoding in hardware ? Thanks in advance for your help !!! Cheers, Tristan ___ --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
Re: [Asterisk-Users] best hardphone for Asterisk?
On Jun 26, 2006, at 7:14 AM, Doug Crompton wrote: I guess I did not make my point clearly enough. I already do have just that. An spa-3000 with ALL internal analog phones on it's on FXO. That's wrong, phones hook to an FXS. But this gives just ONE extension for all phones. Yes I could get more FXS's and run seperate wires. I am using a HT-488 as my secondary FXS, which works ok, but still has problems with DTMF unless I use inband through the gateway a wellgate 3701a in my case. So with that background what would be nice is a wireless device like the Panasonic cordless with one base and multiple phone capability that connected via ethernet and serves the phones. Just wishful thinking. I will stick with what I have until something useful, sylish, and less expensive arrives. Yeah, that does sound nice... I have a panasonic cordless hooked to the HT-488, this gives me mobility with my preferred fxs, and also allows for multiple calls to occur which is very slick. ie I can call out long distance and calls can still arrive through the fxo to the other house phones. This is kind of disorienting to housemates who are used to standard phone systems ;~) Marty ___ --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
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Tzafrir Cohen wrote: On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote: Marco, bom dia. Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo externo? É freeware? Podemos seguir com o projeto Asterisk-PT? English, please, folks. I don't know Portuguese and my Spanish is terrible, but I understood that Josue wanted to know if he needed any external modules. Marco pointed him to the right place to get skype-to-sip and now they're going to collaborate. So, please guys English please or you'll get more of my bad translations. ;) Mike ___ --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
Re: [Asterisk-Users] SE Michigan asterisk users group
On 6/22/06, BerkHolz, Steven [EMAIL PROTECTED] wrote: I am thinking of getting an asterisk user group together for either SE Michigan or just Metro-Detroit. I'm in Ann Arbor and would be interested in such a group; if you create a mailing list for it, could you please add me? Thanks, Rusty ___ --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
Re: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY
Douglas Garstang wrote: Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? Yes, for quite a while. Happens for us, when you do a transfer via the Polycom's transfer button. Doesn't seem to cause any problems though. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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
Re: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY
Yes. It does not seem to cause any problems. Douglas Garstang wrote: Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? I called Polycom tech support, who where utterly useless. Of course Polycom won't officially support it anyway, as they only support Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite some time. We have about 35 phones and it's happening on most (also on the few running SIP software 1.6.6). SIP Software version: 1.6.3.0067 BootROM version: 2.6.2.0032 Reliably Transmitting (no NAT) to xxx.187.128.95:5060: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport From: sip:[EMAIL PROTECTED];tag=as6fd80d1b To: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 114 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 371 ?xml version=1.0? !DOCTYPE presence PUBLIC -//IETF//DTD RFC XPIDF 1.0//EN xpidf.dtd presence presentity uri=sip:[EMAIL PROTECTED];method=SUBSCRIBE / atom id=2944026 address DEFANGED_uri=sip:[EMAIL PROTECTED];user=ip DEFANGED_priority=0.80 status status=open / msnsubstatus substatus=online / /address /atom /presence -- SIP read from xxx.187.128.95:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport From: sip:[EMAIL PROTECTED];tag=as6fd80d1b To: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007 CSeq: 114 NOTIFY Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Event: presence User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Content-Length: 0 Doug. ___ --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 -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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
Re: [Asterisk-Users] STUN?
On Jun 26, 2006, at 9:32 AM, Raymond Tant wrote: x-tad-smallerHi all,/x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerCould someone point at resources for running Asterisk behind a firewall./x-tad-smallerx-tad-smallerSTUN keeps coming up but, alas, I’m easily confused. /x-tad-smallerx-tad-smallerJ /x-tad-smaller STUN is just a way to discover the true address of a machine behind a NAT. Firewalls aren't really an issue per se, other then needing to open particular ports for asterisk to use. For example, udp port 4569 for IAX2 traffic, and 5060 for SIP signaling, as well as ports in the 1-2 range RTP traffic. ___ --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
[Asterisk-Users] Oh oh. Micro$oft just noticed VoIP
It will be interesting to see how many standards get broken, and how many proprietary hooks get thrown into the pot. The bean counters smell some money, and their OS franchise is waning: http://www.nytimes.com/2006/06/26/technology/26soft.html -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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
Re: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY
I have been seeing the same errors here with Polycom 501 and 601 phones. Asterisk version is 1.2.9.1 and Polycom SIP version 1.6.3On 6/26/06, Douglas Garstang [EMAIL PROTECTED] wrote: Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them?I called Polycom tech support, who where utterly useless.Of course Polycom won't officially support it anyway, as they only support Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite some time. We have about 35 phones and it's happening on most (also on the few running SIP software 1.6.6).SIP Software version: 1.6.3.0067 BootROM version: 2.6.2.0032Reliably Transmitting (no NAT) to xxx.187.128.95:5060:NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rportFrom: sip:[EMAIL PROTECTED];tag=as6fd80d1bTo: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: presenceContent-Type: application/xpidf+xml Subscription-State: activeContent-Length: 371?xml version=1.0?!DOCTYPE presence PUBLIC -//IETF//DTD RFC XPIDF 1.0//EN xpidf.dtdpresence presentity uri=sip:[EMAIL PROTECTED];method=SUBSCRIBE /atom id=2944026address uri= sip:[EMAIL PROTECTED];user=ip priority=0.80status status=open /msnsubstatus substatus=online //address/atom /presence-- SIP read from xxx.187.128.95:5060:SIP/2.0 500 Internal Server ErrorVia: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rportFrom: sip:[EMAIL PROTECTED];tag=as6fd80d1bTo: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007CSeq: 114 NOTIFY Call-ID: [EMAIL PROTECTED]Contact: sip:[EMAIL PROTECTED]Event: presence User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036Content-Length: 0Doug.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- BruceNortex Networks ___ --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
[Asterisk-Users] registering a Motorola VT1005
I am trying to register a motorola VT1005. I have many supura ata's that work fine. Anyhelp, would be great. Brandon ___ --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
Re: [Asterisk-Users] asterisk-stat display problems
do you have the php-gd package installed on your * server? Chris Earle (CBL) wrote: Hey all, having a terrible time with asterisk-stat -- it runs, server is fine, but some of the pages don't display properly/at all --- I think this is a code problem with them, but not sure. I thought everyone loved the asterisk-stat package? See below problems. Any ideas? Areski hasn't replied to me since -- Chris - Original Message - From: Chris Earle (CBL) To: Areski Sent: Tuesday, June 13, 2006 6:15 PM Subject: Re: CDR-Analyser version question Thank you for the reply; I see now that the main file cdr.php does work with argument ?s=1, 2, etc but when s=0, does not load I get an Apache error : relocation error: /usr/lib/php4/20020429/gd.so: undefined symbol: gdFontCacheShutdown Not sure if that means anything important; Also, in the new Asterisk-Stat feature pages like Calls Compare (s=2), the pages do not complete their output -- no search button displayed, stops outputting radio buttons for UserField row etc So at this point, only the main Call-log page (s=1) works. I am using Debian with php 4.4.1 Mysql ver 12.22, Distrib 4.0.24 GD Library is 2.0.33 I think Any input you can pass along would be much appreciated! I am comfortable with php so if you want me to modify sourcecode that is fine Thanks! - Original Message - From: Areski To: Chris Earle (CBL) Sent: Sunday, May 28, 2006 7:11 PM Subject: Re: CDR-Analyser version question No there is no asterisk requirement to make asterisk-stat. Indeed the soft is only based on the cdr database. If you have an error you can give me more info, I may help you. Rgds, Areski On 5/25/06, Chris Earle (CBL) wrote: Hi there, quick question: Does asterisk-stat v2.0.1 require Asterisk 1.2+ ? I am using Asterisk 1.0.x and can't get it to load the cdr.php properly so I downgraded to v1.3 and it works... Let me know if there's an asterisk version requirement for each version of the CDR Analyser Thanks! -- Chris Earle snip -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- 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
RE: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY
-Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Monday, June 26, 2006 11:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY Douglas Garstang wrote: Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? Yes, for quite a while. Happens for us, when you do a transfer via the Polycom's transfer button. Doesn't seem to cause any problems though. It's bloody annoying though, especially for those type-A's that don't like to see the console cluttered up with junk. :) Doug ___ --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
Re: [Asterisk-Users] asterisk-stat display problems
yep I don't know exactly which things the php-gd is used for, but like I said, someof the pages work, like the main record page, the little red bars showing call volume work fine Really annoying, cause it looks so good at that point, then you go to use the other pages/features and it's broken Thanks for the reply, -- Chris - Original Message - From: Mojo with Horan Company, LLC [EMAIL PROTECTED] To: Chris Earle (CBL) [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, June 26, 2006 1:49 PM Subject: Re: [Asterisk-Users] asterisk-stat display problems do you have the php-gd package installed on your * server? Chris Earle (CBL) wrote: Hey all, having a terrible time with asterisk-stat -- it runs, server is fine, but some of the pages don't display properly/at all --- I think this is a code problem with them, but not sure. I thought everyone loved the asterisk-stat package? See below problems. Any ideas? Areski hasn't replied to me since -- Chris - Original Message - From: Chris Earle (CBL) To: Areski Sent: Tuesday, June 13, 2006 6:15 PM Subject: Re: CDR-Analyser version question Thank you for the reply; I see now that the main file cdr.php does work with argument ?s=1, 2, etc but when s=0, does not load I get an Apache error : relocation error: /usr/lib/php4/20020429/gd.so: undefined symbol: gdFontCacheShutdown Not sure if that means anything important; Also, in the new Asterisk-Stat feature pages like Calls Compare (s=2), the pages do not complete their output -- no search button displayed, stops outputting radio buttons for UserField row etc So at this point, only the main Call-log page (s=1) works. I am using Debian with php 4.4.1 Mysql ver 12.22, Distrib 4.0.24 GD Library is 2.0.33 I think Any input you can pass along would be much appreciated! I am comfortable with php so if you want me to modify sourcecode that is fine Thanks! - Original Message - From: Areski To: Chris Earle (CBL) Sent: Sunday, May 28, 2006 7:11 PM Subject: Re: CDR-Analyser version question No there is no asterisk requirement to make asterisk-stat. Indeed the soft is only based on the cdr database. If you have an error you can give me more info, I may help you. Rgds, Areski On 5/25/06, Chris Earle (CBL) wrote: Hi there, quick question: Does asterisk-stat v2.0.1 require Asterisk 1.2+ ? I am using Asterisk 1.0.x and can't get it to load the cdr.php properly so I downgraded to v1.3 and it works... Let me know if there's an asterisk version requirement for each version of the CDR Analyser Thanks! -- Chris Earle snip -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Tzafrir Cohen wrote: On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote: Marco, bom dia. Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo externo? É freeware? Podemos seguir com o projeto Asterisk-PT? English, please, folks. Let them talk. What's it hurt the rest of us? We have seen the wages of tortured English sometimes unleashed on the list. If they're getting the job done, I say hit the Delete button and get on with your life. If 80% of the list traffic were in foreign languages, then I would say we would have an issue. MO. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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
[Asterisk-Users] Email notification
Is there a way to get asterisk to send you a email when it looses or an extension doesn’t re-register Roger Workman Business Development Upperclassman/Universal Holdings LLC Voice: 304.324.3800 Fax: 304.324.3801 ICQ: 4447584 Website: http://www.upperclassman.net Billing Questions: billing at upperclassman.net Rental Questions: rentals at upperclassman.net Maintenance: help at upperclassman.net This e-mail and any of its attachments may contain sensitive information, which is privileged, confidential, or subject to copyright belonging to RW Management Inc, Universal Holdings LLC or Upperclassman LLC. This e-mail is intended solely for the use of the individual or entity to which it is addressed. If you are not the intended recipient of this e-mail, you are hereby notified that any dissemination, distribution, copying, or action taken in relation to the contents of and attachments to this e-mail is strictly prohibited and may be unlawful. If you have received this e-mail in error, please notify the sender immediately and permanently delete the original and any copy of or printout of this e-mail. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Monday, June 26, 2006 1:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY Yes. It does not seem to cause any problems. Douglas Garstang wrote: Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? I called Polycom tech support, who where utterly useless. Of course Polycom won't officially support it anyway, as they only support Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite some time. We have about 35 phones and it's happening on most (also on the few running SIP software 1.6.6). SIP Software version: 1.6.3.0067 BootROM version: 2.6.2.0032 Reliably Transmitting (no NAT) to xxx.187.128.95:5060: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport From: sip:[EMAIL PROTECTED];tag=as6fd80d1b To: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 114 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 371 ?xml version=1.0? !DOCTYPE presence PUBLIC -//IETF//DTD RFC XPIDF 1.0//EN xpidf.dtd presence presentity uri=sip:[EMAIL PROTECTED];method=SUBSCRIBE / atom id=2944026 address DEFANGED_uri=sip:[EMAIL PROTECTED];user=ip DEFANGED_priority=0.80 status status=open / msnsubstatus substatus=online / /address /atom /presence -- SIP read from xxx.187.128.95:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport From: sip:[EMAIL PROTECTED];tag=as6fd80d1b To: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007 CSeq: 114 NOTIFY Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Event: presence User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Content-Length: 0 Doug. ___ --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 -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Sorry to all, Now only English speaking :) Your translation was perfect. Thanks once more On 6/26/06, Mike Fedyk [EMAIL PROTECTED] wrote: Tzafrir Cohen wrote: On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote: Marco, bom dia. Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo externo? É freeware? Podemos seguir com o projeto Asterisk-PT? English, please, folks. I don't know Portuguese and my Spanish is terrible, but I understood that Josue wanted to know if he needed any external modules. Marco pointed him to the right place to get skype-to-sip and now they're going to collaborate. So, please guys English please or you'll get more of my bad translations. ;) Mike ___ --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 -- Com os melhores cumprimentos, Marco Mouta ___ --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
Re: [Asterisk-Users] Oh oh. Micro$oft just noticed VoIP
On Mon, 2006-06-26 at 13:16 -0400, Brian Capouch wrote: It will be interesting to see how many standards get broken, and how many proprietary hooks get thrown into the pot. The bean counters smell some money, and their OS franchise is waning: http://www.nytimes.com/2006/06/26/technology/26soft.html and they have been working with cisco on ice (which is standards based, although ice is more of an extension to sip than anything else). But shhh that doesnt help the people that want to bash for no other reason than they can! -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --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
Re: [Asterisk-Users] STUN?
please type in google.com: STUN server ALG The fourth result is a good and small explanation. On 6/26/06, Martin Joseph [EMAIL PROTECTED] wrote: On Jun 26, 2006, at 9:32 AM, Raymond Tant wrote: Hi all, Could someone point at resources for running Asterisk behind a firewall. STUN keeps coming up but, alas, I'm easily confused. J STUN is just a way to discover the true address of a machine behind a NAT. Firewalls aren't really an issue per se, other then needing to open particular ports for asterisk to use. For example, udp port 4569 for IAX2 traffic, and 5060 for SIP signaling, as well as ports in the 1-2 range RTP traffic. ___ --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
Re: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY
Yes we have been getting this error message '500 Internal Server' errors back from their Polycom IP-601 (normally IP address). Do not know why. Was able to regenerate the same issue some times, but not all the time. It is not consistent. Symptom: If you have several phones online (10 extns)if for some reason all the phones start to sent the message because several people in the office are transferring and answering new calls and existing calls in a certain manor, after a while the Asterisk reboots, and if at that instance, if you have any lines on park or on hold, all those lines gets dropped, andthenlight gets stuck on the Polycom IP601 phone. The only way you could get rid of this light on the Polycom phone is by rebooting the phones where the lights are stuck (Almost all phones). Symptom regeneration: It happens when a person is talking, then multiple calls come in and then the person tries to transfer the call to some one. If only one or two error message is coming from the IP601 it will not cause any problem. Solution: We do not have any solutions for it yet. Hope that Asterisk or Polycom will come up with a solution/ Patch/ Firmware upgrade soon. If you do find a solution please let us know. Thanks, Ben K. Chennat On 6/26/06, Doug Lytle [EMAIL PROTECTED] wrote: Douglas Garstang wrote: Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? Yes, for quite a while.Happens for us, when you do a transfer via thePolycom's transfer button.Doesn't seem to cause any problems though.Doug--Ben Franklin quote:Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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
Re: [Asterisk-Users] GXP-2000 and Shared Line Appearances
Daniel Salama wrote: Dustin, any updates on this? Thanks, Daniel Hey Daniel! Yes - just posted the link. I appologize for the delay. Here's the link to the forum as well, if anyone is interested. This should compile and run on Asterisk-1.2.4 and higher. http://www.vecsector.com/phonecall/valet/ Enjoy! Dustin Wildes VecSector, LLC 1.912.422.7082 x101 ___ --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
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Sorry to all. Speaking English only. Regards Josué 2006/6/26, Marco Mouta [EMAIL PROTECTED]: Sorryto all,Now only English speaking :)Your translation was perfect.Thanks once more On 6/26/06, Mike Fedyk [EMAIL PROTECTED] wrote: Tzafrir Cohen wrote: On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote: Marco, bom dia. Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo externo? É freeware? Podemos seguir com o projeto Asterisk-PT? English, please, folks. I don't know Portuguese and my Spanish is terrible, but I understood that Josue wanted to know if he needed any external modules.Marco pointed him to the right place to get skype-to-sip and now they're going to collaborate. So, please guys English please or you'll get more of my bad translations. ;) Mike ___ --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--Com os melhores cumprimentos,Marco Mouta___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
On Mon, June 26, 2006 20:06, Brian Capouch said: Tzafrir Cohen wrote: On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote: Marco, bom dia. Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo externo? É freeware? Podemos seguir com o projeto Asterisk-PT? English, please, folks. Let them talk. What's it hurt the rest of us? It is more a question of netiquette... If you're on an English mailinglist, you should speak English (Not attacking Josué and Marco, just answering the question here). It is not only more productive (If you keep to English, more people understand and can contribute to *and* profit from the discussion), but speaking a different language not spoken by the majority on list is generally considered akin whispering in company: not quite rude, but also not-done... We have seen the wages of tortured English sometimes unleashed on the list. If they're getting the job done, I say hit the Delete button and get on with your life. You can hit the delete button for bad English too, you know! ;-) If 80% of the list traffic were in foreign languages, then I would say we would have an issue. Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig Engels praten! ;-) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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
Re: [Asterisk-Users] GXP-2000 and Shared Line Appearances
Beautiful. Will test and give you comments. Nice work. - Daniel On Jun 26, 2006, at 2:55 PM, Dustin Wildes wrote: Daniel Salama wrote: Dustin, any updates on this? Thanks, Daniel Hey Daniel! Yes - just posted the link. I appologize for the delay. Here's the link to the forum as well, if anyone is interested. This should compile and run on Asterisk-1.2.4 and higher. http://www.vecsector.com/phonecall/valet/ Enjoy! Dustin Wildes VecSector, LLC 1.912.422.7082 x101 ___ --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
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Francesco Peeters (Asterisk) wrote: Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig Engels praten! ;-) Pues my punto fue que un poquito de correo en otro idioma no hace daño, y si ayuda mucho y molesta poco, ¿por qué quejarse? B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
On 6/26/06, Josué Conti [EMAIL PROTECTED] wrote: OK Marco, irei efetuar os testes. Se você quiser, posso lhe ajudar no forum, estou a disposição. Assim que você criar as contas avise para podermos já ir colaborando. Saudações JosuéThe differences of licenses are here: https://www.nch.com.au/cgi-bin/register.exe?software=uplink The site only says that support is different.-- Ralph LiebessohnICQ: 74835911Skype: liebessohn ___ --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
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
On Mon, June 26, 2006 21:39, Brian Capouch said: Francesco Peeters (Asterisk) wrote: Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig Engels praten! ;-) Pues my punto fue que un poquito de correo en otro idioma no hace daño, y si ayuda mucho y molesta poco, ¿por qué quejarse? B. Ningunas quejas aquí... Apenas una explicación en el 'netiquette' --FP ___ --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
[Asterisk-Users] AGI script can not print out error message to console
Hi, guys, I used /usr/src/asterisk/agi/eagi-test.c script to test AGI API, but that script could not print out message tostderr. any ideas? ___ --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
Re: [Asterisk-Users] AGI script can not print out error message to console
what do you mean by could not print out message to stderr??? Try being more descriptive about your problem. Error messages, how have you tried etc. On 6/26/06, Zichao Wu [EMAIL PROTECTED] wrote: Hi, guys, I used /usr/src/asterisk/agi/eagi-test.c script to test AGI API, but that script could not print out message to stderr. any ideas? ___ --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
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Hi All. Please, we need to have more respect with the list. Regards Josué 2006/6/26, Francesco Peeters [EMAIL PROTECTED]: On Mon, June 26, 2006 21:39, Brian Capouch said: Francesco Peeters (Asterisk) wrote: Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig Engels praten!;-) Pues my punto fue que un poquito de correo en otro idioma no hace daño, y si ayuda mucho y molesta poco, ¿por qué quejarse? B.Ningunas quejas aquí... Apenas una explicación en el 'netiquette'--FP___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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
[Asterisk-Users] Say Applications fail
All of the Asterisk Say applications have stopped working. Example: SayDigits(), SayNumber(), etc... CLI output: -- Executing SayDigits(SIP/209.247.17.5-b7901508, 12356) in new stack == Spawn extension (facloc-english, 12356, 2) exited non-zero on 'SIP/209.247.17.5-b7901508' This is driving me crazy. ___ --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
RE: [Asterisk-Users] AGI script can not print out error message toconsole
-Original Message- From: Moises Silva [mailto:[EMAIL PROTECTED] Sent: Monday, June 26, 2006 2:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AGI script can not print out error message toconsole what do you mean by could not print out message to stderr??? Try being more descriptive about your problem. Error messages, how have you tried etc. On 6/26/06, Zichao Wu [EMAIL PROTECTED] wrote: Hi, guys, I used /usr/src/asterisk/agi/eagi-test.c script to test AGI API, but that script could not print out message to stderr. any ideas? He may be referring to the fact that when you run asterisk in non-console mode, stderr goes nowhere (not even /var/log/asterisk/messages). Considering that in a production environment, your going to want to run it like this, it means that if, say, an AGI script encounters a syntax error, you can't see what the problem was, unless you shut asterisk run, re-run it in console mode, debug, and restart it again. Not very convenient! Doug. ___ --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
Re: [Asterisk-Users] asterisk-stat display problems
Check /var/log/http/error.log Usually, asterisk-stat fails because it tries to use more memory than allowed in php.ini. Julian J. M. On 6/26/06, Chris Earle (CBL) [EMAIL PROTECTED] wrote: yep I don't know exactly which things the php-gd is used for, but like I said, someof the pages work, like the main record page, the little red bars showing call volume work fine Really annoying, cause it looks so good at that point, then you go to use the other pages/features and it's broken Thanks for the reply, -- Chris - Original Message - From: Mojo with Horan Company, LLC [EMAIL PROTECTED] To: Chris Earle (CBL) [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, June 26, 2006 1:49 PM Subject: Re: [Asterisk-Users] asterisk-stat display problems do you have the php-gd package installed on your * server? Chris Earle (CBL) wrote: Hey all, having a terrible time with asterisk-stat -- it runs, server is fine, but some of the pages don't display properly/at all --- I think this is a code problem with them, but not sure. I thought everyone loved the asterisk-stat package? See below problems. Any ideas? Areski hasn't replied to me since -- Chris - Original Message - From: Chris Earle (CBL) To: Areski Sent: Tuesday, June 13, 2006 6:15 PM Subject: Re: CDR-Analyser version question Thank you for the reply; I see now that the main file cdr.php does work with argument ?s=1, 2, etc but when s=0, does not load I get an Apache error : relocation error: /usr/lib/php4/20020429/gd.so: undefined symbol: gdFontCacheShutdown Not sure if that means anything important; Also, in the new Asterisk-Stat feature pages like Calls Compare (s=2), the pages do not complete their output -- no search button displayed, stops outputting radio buttons for UserField row etc So at this point, only the main Call-log page (s=1) works. I am using Debian with php 4.4.1 Mysql ver 12.22, Distrib 4.0.24 GD Library is 2.0.33 I think Any input you can pass along would be much appreciated! I am comfortable with php so if you want me to modify sourcecode that is fine Thanks! - Original Message - From: Areski To: Chris Earle (CBL) Sent: Sunday, May 28, 2006 7:11 PM Subject: Re: CDR-Analyser version question No there is no asterisk requirement to make asterisk-stat. Indeed the soft is only based on the cdr database. If you have an error you can give me more info, I may help you. Rgds, Areski On 5/25/06, Chris Earle (CBL) wrote: Hi there, quick question: Does asterisk-stat v2.0.1 require Asterisk 1.2+ ? I am using Asterisk 1.0.x and can't get it to load the cdr.php properly so I downgraded to v1.3 and it works... Let me know if there's an asterisk version requirement for each version of the CDR Analyser Thanks! -- Chris Earle snip -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig Engels praten! ;-) Pues my punto fue que un poquito de correo en otro idioma no hace daño, y si ayuda mucho y molesta poco, ¿por qué quejarse? Quel bordel, sacrebleu! -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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
Re: [Asterisk-Users] SE Michigan asterisk users group
Carlos Alperin wrote: I live in Southfield, our main office is in Pontiac, but our Colo is in Southfield. I'm here in Sterling Heights, have a call center in Clinton Twp that's 100% Cisco/Linksys phones (7940s and SPA942s) and rent in a colo down in Southfield as well where I connect to the PSTN and run other type of calling services based on Asterisk. I've been working on building diskless Asterisk servers to improve reliability in some of my applications. I've been in all kinds of user groups in the past, MCUG, AACS, WCAU, etc, as well as First Tuesdays. It was through these groups I have gained quite a few contacts at Michigan Bell/Ameritech/SBC/ATT and other fortune 500 companies. Depends on where an Asterisk user group meets, I might be interested. -Ron ___ --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
Re: [Asterisk-Users] ASTCC: How to reset periodically all card in use flag back?
Nicolás Gudiño wrote: Hi Ronald, If a user calls and hangs up before the destination party rings, than the in-use flag remains set! This is one case, but maybe there are many other cases. You should install php-pcntl (or compile php to add support for process control functions). The inuse problem will be fixed then. Regards, Can you please give us more info about that? What is php-pcntl? What should it do? How can it be used to be a solution? bye Ronald Wiplinger ___ --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
