[asterisk-users] Polycom just disconnects

[Bartosz Jozwiak]

>>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 at lists.digium.com [mailto:asterisk-users-
> bounces at 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 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 at 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 at 10.60.10.1>;tag=AFBEA619-6B2C56BC
> To: <sip:1117 at 10.60.10.1;user=phone>
> CSeq: 2 CANCEL
> Call-ID: 878bee4d-19a7593b-986d6546 at 10.60.10.109
> Contact: <sip:1111 at 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 at 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 at 10.60.10.1>;tag=AFBEA619-6B2C56BC
> To: <sip:1117 at 10.60.10.1;user=phone>;tag=as54df4909
> Call-ID: 878bee4d-19a7593b-986d6546 at 10.60.10.109
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:1117 at 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 at 10.60.10.1>;tag=AFBEA619-6B2C56BC
> To: <sip:1117 at 10.60.10.1;user=phone>;tag=as54df4909
> Call-ID: 878bee4d-19a7593b-986d6546 at 10.60.10.109
> CSeq: 2 CANCEL
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:1117 at 10.60.10.1>
> Content-Length: 0
> 
> <-- SIP read from 10.60.10.109:5060:
> ACK sip:1117 at 10.60.10.1 SIP/2.0
> Via: SIP/2.0/UDP 10.60.10.109:5060;branch=z9hG4bK6a9be462793F1867
> From: "1111" <sip:1111 at 10.60.10.1>;tag=AFBEA619-6B2C56BC
> To: <sip:1117 at 10.60.10.1;user=phone>;tag=as54df4909
> CSeq: 2 ACK
> Call-ID: 878bee4d-19a7593b-986d6546 at 10.60.10.109
> Contact: <sip:1111 at 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)---
> 
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