Re: [asterisk-users] polycom auto answer
At 15:06 4/14/2008, Jerry Geis wrote: I was trying to get my polycom phone to auto answer. I added this to the dialplan. Used a different phone to call 22 and the phone rang it did not auto answer. Did I miss something? exten = 22,1,SipAddHeader(Call-Info:=\;answer-after=0) exten = 22,n,SipAddHeader(Alert-Info: Ring Answer) exten = 22,n,Set(__SIPADDHEADER=Call-Info:\;answer-after=0) exten = 22,n,Set(__ALERT_INFO=Ring Answer) exten = 22,n,Set(__SIP_URI_OPTIONS=intercom=true) exten = 22,n,Dial(SIP/404) Jerry sip.cfg? voIpProt.SIP.alertInfo.2.value=Ring Answer voIpProt.SIP.alertInfo.2.class=4/ Reboot phones? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom auto answer
Jerry, Did you enable Ring Answer in the phone? Look at your sip.cfg file for: alertInfo voIpProt.SIP.alertInfo.1.value=Ring Answer voIpProt.SIP.alertInfo.1.class=4/ and ringType se.rt.enabled=1 se.rt.modification.enabled=1 DEFAULT se.rt.1.name=Default se.rt.1.type=ring se.rt.1.ringer=2 se.rt.1.callWait=6 se.rt.1.mod=1/ VISUAL_ONLY se.rt.2.name=Visual se.rt.2.type=visual/ AUTO_ANSWER se.rt.3.name=Auto Answer se.rt.3.type=answer/ RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer se.rt.4.timeout=1500 se.rt.4.ringer=13 se.rt.4.callWait=6 se.rt.4.mod=1/ INTERNAL se.rt.5.name=Internal se.rt.5.type=ring se.rt.5.ringer=2 Have a look at: http://www.voicerd.org/index.php/Auto_Pickup On Mon, Apr 14, 2008 at 4:06 PM, Jerry Geis [EMAIL PROTECTED] wrote: I was trying to get my polycom phone to auto answer. I added this to the dialplan. Used a different phone to call 22 and the phone rang it did not auto answer. Did I miss something? exten = 22,1,SipAddHeader(Call-Info:=\;answer-after=0) exten = 22,n,SipAddHeader(Alert-Info: Ring Answer) exten = 22,n,Set(__SIPADDHEADER=Call-Info:\;answer-after=0) exten = 22,n,Set(__ALERT_INFO=Ring Answer) exten = 22,n,Set(__SIP_URI_OPTIONS=intercom=true) exten = 22,n,Dial(SIP/404) Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] http://www.shift8.biz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Auto-Answer problems
Although the ipmid.cfg has been deprecated with SIP v1.5.2 (all the parameters have been moved to sip.cfg), the firmware will still parse and use the ipmid.cfg file until you specifically update your existing configuration files. If you have already updated the configuration files, then both of the parameters will be in the sip.cfg file. Regards, Derek - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, July 14, 2005 9:04 PM Subject: Re: [Asterisk-Users] Polycom Auto-Answer problems The IP-501 AFAIK comes shipped with SIP1.5.2 which does NOT use ipmid.cfg. You have to get new *.cfg files for the ip-501 or the older phones that run 1.5.2. On 7/14/05, Chad Osmond [EMAIL PROTECTED] wrote: CVS Head from 07/07/2005 I'm trying to make an IP-501 auto answer a call. exten = 301,1,SetVar(_ALERT_INFO=Ring_Ans) exten = 301,2,SetVar(ALERT_INFO=Ring_Ans) # Tried both combinations exten = 301,3,Dial(SIP/5001,15) exten = 301,4,Hangup Sip.cfg for Polycom phone alertInfo voIpProt.SIP.alertInfo.2.value=Ring_Ans voIpProt.SIP.alertInfo.1.class=4/ Ipmid.cfg RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer se.rt.4.timeout=1000 se.rt.4.ringer=2 se.rt.4.callWait=6 se.rt.4.mod=1/ Asterisk Log: -- Executing SetVar(SIP/5002-6e20, _ALERT_INFO=Ring_Ans) in new stack -- Executing SetVar(SIP/5002-6e20, ALERT_INFO=Ring_Ans) in new stack -- Executing Dial(SIP/5002-6e20, SIP/5001|15) in new stack -- Called 5001 -- SIP/5001-f735 is ringing -- Nobody picked up in 15000 ms As you can see it just rings, and then hangs up. Any one have an idea? Chad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom Auto-Answer problems
Ipmid still is being processed, sip.cfg contained the same information. I've removed it just to clean things up. Setting the class to the correct value solved the problem, I can't believe that I missed it. Thanks, -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dbruce Sent: July 15, 2005 6:39 AM To: C F; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom Auto-Answer problems Although the ipmid.cfg has been deprecated with SIP v1.5.2 (all the parameters have been moved to sip.cfg), the firmware will still parse and use the ipmid.cfg file until you specifically update your existing configuration files. If you have already updated the configuration files, then both of the parameters will be in the sip.cfg file. Regards, Derek - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, July 14, 2005 9:04 PM Subject: Re: [Asterisk-Users] Polycom Auto-Answer problems The IP-501 AFAIK comes shipped with SIP1.5.2 which does NOT use ipmid.cfg. You have to get new *.cfg files for the ip-501 or the older phones that run 1.5.2. On 7/14/05, Chad Osmond [EMAIL PROTECTED] wrote: CVS Head from 07/07/2005 I'm trying to make an IP-501 auto answer a call. exten = 301,1,SetVar(_ALERT_INFO=Ring_Ans) exten = 301,2,SetVar(ALERT_INFO=Ring_Ans) # Tried both combinations exten = 301,3,Dial(SIP/5001,15) exten = 301,4,Hangup Sip.cfg for Polycom phone alertInfo voIpProt.SIP.alertInfo.2.value=Ring_Ans voIpProt.SIP.alertInfo.1.class=4/ Ipmid.cfg RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer se.rt.4.timeout=1000 se.rt.4.ringer=2 se.rt.4.callWait=6 se.rt.4.mod=1/ Asterisk Log: -- Executing SetVar(SIP/5002-6e20, _ALERT_INFO=Ring_Ans) in new stack -- Executing SetVar(SIP/5002-6e20, ALERT_INFO=Ring_Ans) in new stack -- Executing Dial(SIP/5002-6e20, SIP/5001|15) in new stack -- Called 5001 -- SIP/5001-f735 is ringing -- Nobody picked up in 15000 ms As you can see it just rings, and then hangs up. Any one have an idea? Chad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Auto-Answer problems
C F wrote: The IP-501 AFAIK comes shipped with SIP1.5.2 which does NOT use ipmid.cfg. You have to get new *.cfg files for the ip-501 or the older phones that run 1.5.2. Sorry to tell you but that is not a correct. The IP-501 I have about 15 of them new and they came with 1.4.2 also they do use the ipmid.cfg. But it can use the newer version which is an all in one. You tell the phone which files to use via it's configuration file mac.cfg It sets up which files to load. On 7/14/05, Chad Osmond [EMAIL PROTECTED] wrote: CVS Head from 07/07/2005 I'm trying to make an IP-501 auto answer a call. exten = 301,1,SetVar(_ALERT_INFO=Ring_Ans) exten = 301,2,SetVar(ALERT_INFO=Ring_Ans) # Tried both combinations exten = 301,3,Dial(SIP/5001,15) exten = 301,4,Hangup Sip.cfg for Polycom phone alertInfo voIpProt.SIP.alertInfo.2.value=Ring_Ans voIpProt.SIP.alertInfo.1.class=4/ Ipmid.cfg RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer se.rt.4.timeout=1000 se.rt.4.ringer=2 se.rt.4.callWait=6 se.rt.4.mod=1/ Asterisk Log: -- Executing SetVar(SIP/5002-6e20, _ALERT_INFO=Ring_Ans) in new stack -- Executing SetVar(SIP/5002-6e20, ALERT_INFO=Ring_Ans) in new stack -- Executing Dial(SIP/5002-6e20, SIP/5001|15) in new stack -- Called 5001 -- SIP/5001-f735 is ringing -- Nobody picked up in 15000 ms As you can see it just rings, and then hangs up. Any one have an idea? Chad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Auto-Answer problems
On 14-Jul-05, at 2:20 PM, Chad Osmond wrote: CVS Head from 07/07/2005 I'm trying to make an IP-501 auto answer a call. exten = 301,1,SetVar(_ALERT_INFO=Ring_Ans) exten = 301,2,SetVar(ALERT_INFO=Ring_Ans) # Tried both combinations Try getting rid of the quotes: exten = 301,1,SetVar(_ALERT_INFO=Ring_Ans) exten = 301,2,SetVar(ALERT_INFO=Ring_Ans) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Auto-Answer problems
Chad Osmond wrote: CVS Head from 07/07/2005 I'm trying to make an IP-501 auto answer a call. exten = 301,1,SetVar(_ALERT_INFO=Ring_Ans) exten = 301,2,SetVar(ALERT_INFO=Ring_Ans) # Tried both combinations exten = 301,3,Dial(SIP/5001,15) exten = 301,4,Hangup Sip.cfg for Polycom phone alertInfo voIpProt.SIP.alertInfo.2.value=Ring_Ans voIpProt.SIP.alertInfo.1.class=4/ Ipmid.cfg RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer se.rt.4.timeout=1000 se.rt.4.ringer=2 se.rt.4.callWait=6 se.rt.4.mod=1/ you um, rebooted the phone, right? (and are sure the new configuration was loaded?) Asterisk Log: -- Executing SetVar(SIP/5002-6e20, _ALERT_INFO=Ring_Ans) in new stack -- Executing SetVar(SIP/5002-6e20, ALERT_INFO=Ring_Ans) in new stack -- Executing Dial(SIP/5002-6e20, SIP/5001|15) in new stack -- Called 5001 -- SIP/5001-f735 is ringing -- Nobody picked up in 15000 ms As you can see it just rings, and then hangs up. Any one have an idea? Chad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Auto-Answer problems
Although the suggestions from the previous responders are goo suggestions, you have overlooked an obvious mis-configuration... In your ipmid.cfg, you have: alertInfo voIpProt.SIP.alertInfo.2.value=Ring_Ans voIpProt.SIP.alertInfo.1.class=4/ when you should have: alertInfo voIpProt.SIP.alertInfo.2.value=Ring_Ans voIpProt.SIP.alertInfo.2.class=4/ What you have done is assign the RING_ANSWER ringtype to the previous entry. You also need to make sure that each entry in alertInfo section is sequential. Make the correction in your ipmid.cfg, reboot the phones and all should work as expected. Regards, Derek Bruce - Original Message - From: Adam Dobrin [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, July 14, 2005 4:07 PM Subject: Re: [Asterisk-Users] Polycom Auto-Answer problems Chad Osmond wrote: CVS Head from 07/07/2005 I'm trying to make an IP-501 auto answer a call. exten = 301,1,SetVar(_ALERT_INFO=Ring_Ans) exten = 301,2,SetVar(ALERT_INFO=Ring_Ans) # Tried both combinations exten = 301,3,Dial(SIP/5001,15) exten = 301,4,Hangup Sip.cfg for Polycom phone alertInfo voIpProt.SIP.alertInfo.2.value=Ring_Ans voIpProt.SIP.alertInfo.1.class=4/ Ipmid.cfg RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer se.rt.4.timeout=1000 se.rt.4.ringer=2 se.rt.4.callWait=6 se.rt.4.mod=1/ you um, rebooted the phone, right? (and are sure the new configuration was loaded?) Asterisk Log: -- Executing SetVar(SIP/5002-6e20, _ALERT_INFO=Ring_Ans) in new stack -- Executing SetVar(SIP/5002-6e20, ALERT_INFO=Ring_Ans) in new stack -- Executing Dial(SIP/5002-6e20, SIP/5001|15) in new stack -- Called 5001 -- SIP/5001-f735 is ringing -- Nobody picked up in 15000 ms As you can see it just rings, and then hangs up. Any one have an idea? Chad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Auto-Answer problems
The IP-501 AFAIK comes shipped with SIP1.5.2 which does NOT use ipmid.cfg. You have to get new *.cfg files for the ip-501 or the older phones that run 1.5.2. On 7/14/05, Chad Osmond [EMAIL PROTECTED] wrote: CVS Head from 07/07/2005 I'm trying to make an IP-501 auto answer a call. exten = 301,1,SetVar(_ALERT_INFO=Ring_Ans) exten = 301,2,SetVar(ALERT_INFO=Ring_Ans) # Tried both combinations exten = 301,3,Dial(SIP/5001,15) exten = 301,4,Hangup Sip.cfg for Polycom phone alertInfo voIpProt.SIP.alertInfo.2.value=Ring_Ans voIpProt.SIP.alertInfo.1.class=4/ Ipmid.cfg RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer se.rt.4.timeout=1000 se.rt.4.ringer=2 se.rt.4.callWait=6 se.rt.4.mod=1/ Asterisk Log: -- Executing SetVar(SIP/5002-6e20, _ALERT_INFO=Ring_Ans) in new stack -- Executing SetVar(SIP/5002-6e20, ALERT_INFO=Ring_Ans) in new stack -- Executing Dial(SIP/5002-6e20, SIP/5001|15) in new stack -- Called 5001 -- SIP/5001-f735 is ringing -- Nobody picked up in 15000 ms As you can see it just rings, and then hangs up. Any one have an idea? Chad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Auto-Answer
Eric Rees wrote: I am having a problem with Polycom auto-answer. I have the auto-answer working between PhoneA and PhoneB, but when I try to use the intercom between more then one phone I start having problems. PhoneA dials *3 which calls PhoneB, PhoneC, and PhoneD. All the phones ring, but only one will pick up, the rest will hang up and I get this error on Asterisk: Got SIP response 500 Internal Server Error. U Yeah. What did you think was going to happen, Asterisk was going to magically bridge four phones together because they all answered? As soon as one phone answers, the call is complete and the remaining phones will not be able to answer (because the calls going out to them will have been destroyed). If you need more than two parties in a call, you need to use MeetMe or one of the other conferencing applications. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom Auto-Answer
That's kind of what I thought, but I am trying to put together a phone to multi-phone paging system. I all ready have and overhead paging systems, but the powers-at-be want a phone paging system. -Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 01, 2005 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom Auto-Answer Eric Rees wrote: I am having a problem with Polycom auto-answer. I have the auto-answer working between PhoneA and PhoneB, but when I try to use the intercom between more then one phone I start having problems. PhoneA dials *3 which calls PhoneB, PhoneC, and PhoneD. All the phones ring, but only one will pick up, the rest will hang up and I get this error on Asterisk: Got SIP response 500 Internal Server Error. U Yeah. What did you think was going to happen, Asterisk was going to magically bridge four phones together because they all answered? As soon as one phone answers, the call is complete and the remaining phones will not be able to answer (because the calls going out to them will have been destroyed). If you need more than two parties in a call, you need to use MeetMe or one of the other conferencing applications. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom Auto-Answer
Take a look at http://www.voip-info.org/wiki-Cisco+7940-7960+auto-answer+config. With a few small modifications it should work like a champ on the Polycom phones. B. J. -Original Message- From: Eric Rees [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 01, 2005 10:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom Auto-Answer That's kind of what I thought, but I am trying to put together a phone to multi-phone paging system. I all ready have and overhead paging systems, but the powers-at-be want a phone paging system. -Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 01, 2005 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom Auto-Answer Eric Rees wrote: I am having a problem with Polycom auto-answer. I have the auto-answer working between PhoneA and PhoneB, but when I try to use the intercom between more then one phone I start having problems. PhoneA dials *3 which calls PhoneB, PhoneC, and PhoneD. All the phones ring, but only one will pick up, the rest will hang up and I get this error on Asterisk: Got SIP response 500 Internal Server Error. U Yeah. What did you think was going to happen, Asterisk was going to magically bridge four phones together because they all answered? As soon as one phone answers, the call is complete and the remaining phones will not be able to answer (because the calls going out to them will have been destroyed). If you need more than two parties in a call, you need to use MeetMe or one of the other conferencing applications. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Auto-Answer
B. J. Bomar wrote: Take a look at http://www.voip-info.org/wiki-Cisco+7940-7960+auto-answer+config. With a few small modifications it should work like a champ on the Polycom phones. B. J. Polycom has a much better way to do auto-answer using SIP_INFO. My sample configs have both a AutoAnswer and a Ring_Answer (for intercom): http://www.kriscompanies.com/modules.php?name=Downloadsd_op=viewdownloadcid=1 -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Auto-Answer
Kristian Kielhofner wrote: B. J. Bomar wrote: Take a look at http://www.voip-info.org/wiki-Cisco+7940-7960+auto-answer+config. With a few small modifications it should work like a champ on the Polycom phones. B. J. Polycom has a much better way to do auto-answer using SIP_INFO. My sample configs have both a AutoAnswer and a Ring_Answer (for intercom): http://www.kriscompanies.com/modules.php?name=Downloadsd_op=viewdownloadcid=1 That's only for station to station paging, not all station paging. Calling a group of phones with auto-answer will connect the caller to the FIRST phone that responds, not all phones. For all station paging you really need a MeetMe. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom Auto-Answer
That worked great. -Original Message- From: B. J. Bomar [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 01, 2005 11:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom Auto-Answer Take a look at http://www.voip-info.org/wiki-Cisco+7940-7960+auto-answer+config. With a few small modifications it should work like a champ on the Polycom phones. B. J. -Original Message- From: Eric Rees [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 01, 2005 10:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom Auto-Answer That's kind of what I thought, but I am trying to put together a phone to multi-phone paging system. I all ready have and overhead paging systems, but the powers-at-be want a phone paging system. -Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 01, 2005 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom Auto-Answer Eric Rees wrote: I am having a problem with Polycom auto-answer. I have the auto-answer working between PhoneA and PhoneB, but when I try to use the intercom between more then one phone I start having problems. PhoneA dials *3 which calls PhoneB, PhoneC, and PhoneD. All the phones ring, but only one will pick up, the rest will hang up and I get this error on Asterisk: Got SIP response 500 Internal Server Error. U Yeah. What did you think was going to happen, Asterisk was going to magically bridge four phones together because they all answered? As soon as one phone answers, the call is complete and the remaining phones will not be able to answer (because the calls going out to them will have been destroyed). If you need more than two parties in a call, you need to use MeetMe or one of the other conferencing applications. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Auto-Answer
Eric Wieling wrote: Polycom has a much better way to do auto-answer using SIP_INFO. My sample configs have both a AutoAnswer and a Ring_Answer (for intercom): http://www.kriscompanies.com/modules.php?name=Downloadsd_op=viewdownloadcid=1 That's only for station to station paging, not all station paging. Calling a group of phones with auto-answer will connect the caller to the FIRST phone that responds, not all phones. For all station paging you really need a MeetMe. I know. But there is a better way to do it than to setup an additional line just for paging. That's why I like the ALERT_INFO variable. -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom Auto-Answer
Yes they do, but you can use the dialplan as an example to setup a global phone page. That is all I was pointing out. B. J. -Original Message- From: Kristian Kielhofner [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 01, 2005 12:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom Auto-Answer B. J. Bomar wrote: Take a look at http://www.voip-info.org/wiki-Cisco+7940-7960+auto-answer+config. With a few small modifications it should work like a champ on the Polycom phones. B. J. Polycom has a much better way to do auto-answer using SIP_INFO. My sample configs have both a AutoAnswer and a Ring_Answer (for intercom): http://www.kriscompanies.com/modules.php?name=Downloadsd_op=viewdownloadci d=1 -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
