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