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:1117@10.60.10.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.60.10.109:5060;branch=z9hG4bK6a9be462793F1867 From: "1111" <sip:1111@10.60.10.1>;tag=AFBEA619-6B2C56BC To: <sip:1117@10.60.10.1;user=phone> CSeq: 2 CANCEL Call-ID: 878bee4d-19a7593b-986d6546@10.60.10.109 Contact: <sip:1111@10.60.10.109:5060> 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:1117@10.60.10.1;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:1111@10.60.10.1>;tag=AFBEA619-6B2C56BC To: <sip:1117@10.60.10.1;user=phone>;tag=as54df4909 Call-ID: 878bee4d-19a7593b-986d6546@10.60.10.109 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:1117@10.60.10.1> 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:1111@10.60.10.1>;tag=AFBEA619-6B2C56BC To: <sip:1117@10.60.10.1;user=phone>;tag=as54df4909 Call-ID: 878bee4d-19a7593b-986d6546@10.60.10.109 CSeq: 2 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:1117@10.60.10.1> Content-Length: 0 <-- SIP read from 10.60.10.109:5060: ACK sip:1117@10.60.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.60.10.109:5060;branch=z9hG4bK6a9be462793F1867 From: "1111" <sip:1111@10.60.10.1>;tag=AFBEA619-6B2C56BC To: <sip:1117@10.60.10.1;user=phone>;tag=as54df4909 CSeq: 2 ACK Call-ID: 878bee4d-19a7593b-986d6546@10.60.10.109 Contact: <sip:1111@10.60.10.109:5060> 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)---
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: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] 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 tospeak> 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:1117@10.60.10.1;user=phone SIP/2.0 > Via: SIP/2.0/UDP 10.60.10.109:5060;branch=z9hG4bK6a9be462793F1867 > From: "1111" <sip:1111@10.60.10.1>;tag=AFBEA619-6B2C56BC > To: <sip:1117@10.60.10.1;user=phone> > CSeq: 2 CANCEL > Call-ID: 878bee4d-19a7593b-986d6546@10.60.10.109 > Contact: <sip:1111@10.60.10.109:5060> > 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:1117@10.60.10.1;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:1111@10.60.10.1>;tag=AFBEA619-6B2C56BC > To: <sip:1117@10.60.10.1;user=phone>;tag=as54df4909 > Call-ID: 878bee4d-19a7593b-986d6546@10.60.10.109 > CSeq: 2 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:1117@10.60.10.1> > 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:1111@10.60.10.1>;tag=AFBEA619-6B2C56BC > To: <sip:1117@10.60.10.1;user=phone>;tag=as54df4909 > Call-ID: 878bee4d-19a7593b-986d6546@10.60.10.109 > CSeq: 2 CANCEL > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:1117@10.60.10.1> > Content-Length: 0 > > <-- SIP read from 10.60.10.109:5060: > ACK sip:1117@10.60.10.1 SIP/2.0 > Via: SIP/2.0/UDP 10.60.10.109:5060;branch=z9hG4bK6a9be462793F1867 > From: "1111" <sip:1111@10.60.10.1>;tag=AFBEA619-6B2C56BC > To: <sip:1117@10.60.10.1;user=phone>;tag=as54df4909 > CSeq: 2 ACK > Call-ID: 878bee4d-19a7593b-986d6546@10.60.10.109 > Contact: <sip:1111@10.60.10.109:5060> > 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
>>Do you have audio running during the hold (MOH), or silence? >>Could the Polycom (or asterisk) be dropping the call due to inactivity?Yes is running... I can listen to the music (MOH) and then suddenly I get disconnected.> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] 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 tospeak> 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:1117@10.60.10.1;user=phone SIP/2.0 > Via: SIP/2.0/UDP 10.60.10.109:5060;branch=z9hG4bK6a9be462793F1867 > From: "1111" <sip:1111@10.60.10.1>;tag=AFBEA619-6B2C56BC > To: <sip:1117@10.60.10.1;user=phone> > CSeq: 2 CANCEL > Call-ID: 878bee4d-19a7593b-986d6546@10.60.10.109 > Contact: <sip:1111@10.60.10.109:5060> > 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:1117@10.60.10.1;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:1111@10.60.10.1>;tag=AFBEA619-6B2C56BC > To: <sip:1117@10.60.10.1;user=phone>;tag=as54df4909 > Call-ID: 878bee4d-19a7593b-986d6546@10.60.10.109 > CSeq: 2 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:1117@10.60.10.1> > 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:1111@10.60.10.1>;tag=AFBEA619-6B2C56BC > To: <sip:1117@10.60.10.1;user=phone>;tag=as54df4909 > Call-ID: 878bee4d-19a7593b-986d6546@10.60.10.109 > CSeq: 2 CANCEL > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:1117@10.60.10.1> > Content-Length: 0 > > <-- SIP read from 10.60.10.109:5060: > ACK sip:1117@10.60.10.1 SIP/2.0 > Via: SIP/2.0/UDP 10.60.10.109:5060;branch=z9hG4bK6a9be462793F1867 > From: "1111" <sip:1111@10.60.10.1>;tag=AFBEA619-6B2C56BC > To: <sip:1117@10.60.10.1;user=phone>;tag=as54df4909 > CSeq: 2 ACK > Call-ID: 878bee4d-19a7593b-986d6546@10.60.10.109 > Contact: <sip:1111@10.60.10.109:5060> > 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