[asterisk-users] Polycom just disconnects

[Bartosz Jozwiak]

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)---