I am running a call center with Asterisk doing the queueing but routing calls through Cisco Call Manager 6.1 to SCCP IP phones.
I am seeing this issue where the phone will ring an agent, the agent will pickup but no one will be there and the same call will ring the next agent until it eventually gets answered. This seems to be happening only during peak hours. Looking through the console I see a sip Server Error Response. At the moment I am under the assumption that there is some problem between asterisk and callmanager handoff.
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— (9 headers 0 lines) —
HQASX04*CLI>
<— SIP read from 10.11.220.2:5060 —>
SIP/2.0 180 Ringing
Date: Wed, 07 Jan 2009 19:41:43 GMT
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH
From: “CS Event Queue” sip:4049932296@10.11.220.18;tag=as0cacd96b
Allow-Events: presence
Remote-Party-ID: “agent name” sip:2018@10.11.220.2;party=called;screen=yes;privacy=off
Content-Length: 0
To: sip:2018@10.11.220.2;tag=e3ee7c3b-31b8-47c3-997a-7c3386f263d2-49733052
Contact: sip:2018@10.11.220.2:5060
Call-ID: 30c573f44382a7157eea80551024ce8c@10.11.220.18
Via: SIP/2.0/UDP 10.11.220.18:5060;branch=z9hG4bK684f93f8;rport
CSeq: 102 INVITE
<------------->
— (12 headers 0 lines) —
– SIP/callman02-08277880 is ringing
– Agent/2821 is ringing
Really destroying SIP dialog ‘5677df0758a1988b5b91699973749c36@10.11.220.18’ Method: INVITE
HQASX04*CLI>
<— SIP read from 10.11.220.2:5060 —>
SIP/2.0 503 Service Unavailable
Reason: Q.850;cause=47
Date: Wed, 07 Jan 2009 19:41:43 GMT
From: “CS Event Queue” sip:4049932296@10.11.220.18;tag=as0cacd96b
Allow-Events: presence
Content-Length: 0
To: sip:2018@10.11.220.2;tag=e3ee7c3b-31b8-47c3-997a-7c3386f263d2-49733052
Call-ID: 30c573f44382a7157eea80551024ce8c@10.11.220.18
Via: SIP/2.0/UDP 10.11.220.18:5060;branch=z9hG4bK684f93f8;rport
CSeq: 102 INVITE