Do you have audio running during the hold (MOH), or silence? Could the Polycom (or asterisk) be dropping the call due to inactivity?
> -----Original Message----- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak > Sent: Friday, August 11, 2006 6:04 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Polycom just disconnects > > Hello, > > I have a polycom 500 phone. While testing our queue and waiting to speak > with operator my phone after about > 2 minutes just disconnects. > Here is sip debug. > I cannot find out what the problem might be. > Does anybody can see something strange in it : > > <-- SIP read from 10.60.10.109:5060: > CANCEL sip:[EMAIL PROTECTED];user=phone SIP/2.0 > Via: SIP/2.0/UDP 10.60.10.109:5060;branch=z9hG4bK6a9be462793F1867 > From: "1111" <sip:[EMAIL PROTECTED]>;tag=AFBEA619-6B2C56BC > To: <sip:[EMAIL PROTECTED];user=phone> > CSeq: 2 CANCEL > Call-ID: [EMAIL PROTECTED] > Contact: <sip:[EMAIL PROTECTED]:5060> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, > NOTIFY, > PRACK, UPDATE, REFER > User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054 > Proxy-Authorization: Digest username="1111", realm="asterisk", > nonce="54dd123c", uri="sip:[EMAIL PROTECTED];user=phone", > response="607995a40ae6b4e1e061c6ac1d0fbf1d", algorithm=MD5 > Max-Forwards: 70 > Content-Length: 0 > > > --- (12 headers 0 lines)--- > Sending to 10.60.10.109 : 5060 (non-NAT) > Reliably Transmitting (no NAT) to 10.60.10.109:5060: > SIP/2.0 487 Request Terminated > Via: SIP/2.0/UDP > 10.60.10.109:5060;branch=z9hG4bK6a9be462793F1867;received=10.60.10.109 > From: "1111" <sip:[EMAIL PROTECTED]>;tag=AFBEA619-6B2C56BC > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as54df4909 > Call-ID: [EMAIL PROTECTED] > CSeq: 2 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:[EMAIL PROTECTED]> > Content-Length: 0 > > > --- > Transmitting (no NAT) to 10.60.10.109:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 10.60.10.109:5060;branch=z9hG4bK6a9be462793F1867;received=10.60.10.109 > From: "1111" <sip:[EMAIL PROTECTED]>;tag=AFBEA619-6B2C56BC > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as54df4909 > Call-ID: [EMAIL PROTECTED] > CSeq: 2 CANCEL > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:[EMAIL PROTECTED]> > Content-Length: 0 > > <-- SIP read from 10.60.10.109:5060: > ACK sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 10.60.10.109:5060;branch=z9hG4bK6a9be462793F1867 > From: "1111" <sip:[EMAIL PROTECTED]>;tag=AFBEA619-6B2C56BC > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as54df4909 > CSeq: 2 ACK > Call-ID: [EMAIL PROTECTED] > Contact: <sip:[EMAIL PROTECTED]:5060> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, > NOTIFY, > PRACK, UPDATE, REFER > User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054 > Max-Forwards: 70 > Content-Length: 0 > > > --- (11 headers 0 lines)--- > > _______________________________________________ > --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
