All, Working through a SIP integration between UCME and UC using SIP, and I've got a couple of issues versus how the solutions guide lays out the solution.
Item 1.4 states "Make sure that calls within the HQ or BR1 site use g711ulaw and calls between sites use g729." (1 pt) But in item 6.2, the solution shows the SIP dial-peer to the UC server with codec g711ulaw. Calls from BR2 to voicemail at HQ are now using g711ulaw, so haven't we just cost ourselves a point? (Better 1 point than the 4 points for the SIP UC integration I guess). I have been trying to work around this issue by various means, but haven't come up with a solution yet. I have hardcoded the UC dial-peer to g729 Because BR2 has SIP phones, I have registered a transcoder to UCME at BR2, and the call to voicemail from the SIP phones invokes the transcoder correctly. The specific issue that I'm facing is if I dial by extension within the Main Greeting, UC transfers the call back to BR2, but the call is rejected. I can get around this specific limitation by configuring voice-class codec on the UC dial-peer, but then the SIP phone uses g711u straight to UC. Here's the debug ccsip messages from the call when UC tries to send the call back to 3001. BR2# BR2#sh sccp conn sess_id conn_id stype mode codec sport rport ripaddr 1 1 xcode sendrecv g729 17442 2000 10.10.202.1 1 2 xcode sendrecv g711u 17630 2000 10.10.202.1 Total number of active session(s) 1, and connection(s) 2 BR2# BR2# BR2# BR2# BR2# BR2# BR2# Aug 31 03:14:11.319: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:3...@10.10.110.3:5060 SIP/2.0 From: <sip:3...@10.10.210.12 <sip%3a3...@10.10.210.12> >;tag=b479d639c3b54f96995687a535357a46 To: "BR2 Phone3" <sip:3...@10.10.110.3 <sip%3a3...@10.10.110.3> >;tag=91A708-25BC Via: SIP/2.0/UDP 10.10.210.12:5060 ;branch=z9hG4bKb9702a2ec4174a48916a263f9434aa8b Max-Forwards: 70 User-Agent: Cisco-UnityConnection/7.0 Contact: <sip:10.10.210.12:5060;transport=tcp> Call-ID: a01a55e6-b3e411df-81e98f0a-3d3c3...@10.10.110.3 CSeq: 2 INVITE Allow-Events: kpml Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,SUBSCRIBE Content-Length: 256 Content-Type: application/sdp v=0 o=10.10.210.12 3324197134 3324197134 IN IP4 10.10.210.12 s=No Subject c=IN IP4 0.0.0.0 t=0 0 m=audio 16702 RTP/AVP 0 18 101 a=rtpmap:0 pcmu/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Aug 31 03:14:11.331: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Date: Tue, 31 Aug 2010 03:14:11 GMT From: <sip:3...@10.10.210.12 <sip%3a3...@10.10.210.12> >;tag=b479d639c3b54f96995687a535357a46 Allow-Events: telephone-event Content-Length: 0 To: "BR2 Phone3" <sip:3...@10.10.110.3 <sip%3a3...@10.10.110.3> >;tag=91A708-25BC Call-ID: a01a55e6-b3e411df-81e98f0a-3d3c3...@10.10.110.3 Via: SIP/2.0/UDP 10.10.210.12:5060 ;branch=z9hG4bKb9702a2ec4174a48916a263f9434aa8b CSeq: 2 INVITE Server: Cisco-SIPGateway/IOS-12.x Aug 31 03:14:11.331: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 488 Not Acceptable Media Date: Tue, 31 Aug 2010 03:14:11 GMT From: <sip:3...@10.10.210.12 <sip%3a3...@10.10.210.12> >;tag=b479d639c3b54f96995687a535357a46 Allow-Events: telephone-event Content-Length: 0 To: "BR2 Phone3" <sip:3...@10.10.110.3 <sip%3a3...@10.10.110.3> >;tag=91A708-25BC Call-ID: a01a55e6-b3e411df-81e98f0a-3d3c3...@10.10.110.3 Via: SIP/2.0/UDP 10.10.210.12:5060 ;branch=z9hG4bKb9702a2ec4174a48916a263f9434aa8b CSeq: 2 INVITE Server: Cisco-SIPGateway/IOS-12.x Aug 31 03:14:11.343: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: ACK sip:3...@10.10.110.3:5060 SIP/2.0 From: <sip:3...@10.10.210.12 <sip%3a3...@10.10.210.12> >;tag=b479d639c3b54f96995687a535357a46 To: "BR2 Phone3" <sip:3...@10.10.110.3 <sip%3a3...@10.10.110.3> >;tag=91A708-25BC Via: SIP/2.0/UDP 10.10.210.12:5060 ;branch=z9hG4bKb9702a2ec4174a48916a263f9434aa8b Max-Forwards: 70 User-Agent: Cisco-UnityConnection/7.0 Call-ID: a01a55e6-b3e411df-81e98f0a-3d3c3...@10.10.110.3 CSeq: 2 ACK Content-Length: 0 BR2# Is there a way to make this work? mike
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