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: [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
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