Make these changes on your gateway.
voice class codec 3 no codec preference 1 no codec preference 2 no codec preference 3 codec preference 1 g711alaw codec preference 2 g711ulaw codec preference 3 g729r8 ! dial-peer voice 200 voip no codec g711ulaw voice-class codec 3 ! dial-peer voice 300 voip no codec g711ulaw voice-class codec 3 ! dial-peer voice 600 voip no codec g711ulaw voice-class codec 3 ! dial-peer voice 701 voip no codec g711ulaw voice-class codec 3 ! dial-peer voice 400 voip no codec g711ulaw voice-class codec 3 ! dial-peer voice 100 voip no codec g711ulaw voice-class codec 3 ! dial-peer voice 500 voip no codec g711ulaw voice-class codec 3 !
Apart from this I would recommend you to look over your dial-peer configuration as it is quite messy. Normally you'd need 4 dial-peers, one pair for in- and outbound to/from CM and another pair for in- and outbound to/from your service provider. This could be combined into just two dial-peers, one for CM for both directions and another for the SP in both directions. Personally I'm not a big fan of this, as it makes it harder to see the path or the call and it gives you less flexibility IMHO.
Post the output from these debugs running in parallel when making a call in an attached text file.
- debug ccsip message
- debug voip ccapi inout
From what I can tell the shared debug does not contain any information for ccsip message. Per the debug shared the call comes in on dial peer 100 and then first uses dial peer 403 outbound, where it disconnects with cause value 47 and then dial peer 401 outbound, where it disconnects with cause value 63. AFAIK the debug does not contain what you initially mentioned, ie there is no error 500 or 488.
Please share the asked for debugs where both of them are running at the same time.
The call is sent to your service provider with g711ulaw as the codec.
*Mar 21 11:08:08.623: //295/8CDCB7800000/SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:08028800563@41.138.163.34:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.15.43:5060;branch=z9hG4bKA91AC7 Remote-Party-ID: "KAY" <sip:015150200@10.10.15.43>;party=calling;screen=yes;privacy=off From: "KAY" <sip:015150200@10.10.15.43>;tag=B49394-542 To: <sip:08028800563@41.138.163.34> Date: Mon, 21 Mar 2022 11:08:08 GMT Call-ID: 625F1DF-A83E11EC-834A90F9-9731C464@10.10.15.43 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 2363275136-0000065536-0000015091-1343162890 User-Agent: Cisco-SIPGateway/IOS-15.4.3.M3 Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1647860888 Contact: <sip:015150200@10.10.15.43:5060> Expires: 180 Allow-Events: telephone-event Max-Forwards: 68 Session-Expires: 1800 Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 268 v=0 o=CiscoSystemsSIP-GW-UserAgent 2614 4665 IN IP4 10.10.15.43 s=SIP Call c=IN IP4 10.10.15.43 t=0 0 m=audio 16908 RTP/AVP 0 101 19 c=IN IP4 10.10.15.43 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:19 CN/8000 a=ptime:20 *Mar 21 11:08:08.627: //295/8CDCB7800000/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.15.43:5060;branch=z9hG4bKA91AC7 CSeq: 101 INVITE Content-Length: 0 Call-ID: 625F1DF-A83E11EC-834A90F9-9731C464@10.10.15.43 To: <sip:08028800563@41.138.163.34> From: "KAY" <sip:015150200@10.10.15.43>;tag=B49394-542 *Mar 21 11:08:08.831: //295/8CDCB7800000/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 10.10.15.43:5060;branch=z9hG4bKA91AC7 CSeq: 101 INVITE Content-Length: 0 Call-ID: 625F1DF-A83E11EC-834A90F9-9731C464@10.10.15.43 To: <sip:08028800563@41.138.163.34>;tag=1301172986076015 From: "KAY" <sip:015150200@10.10.15.43>;tag=B49394-542 *Mar 21 11:08:08.835: //293/8CDCB7800000/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 500 Internal Server Error Via: SIP/2.0/TCP 10.10.15.80:5060;branch=z9hG4bK58c747cf77be7 From: "KAY" <sip:015150200@10.10.15.80>;tag=10284120~aa90adaa-cf9d-4579-9e7f-c8715f9da470-18462389 To: <sip:08028800563@10.10.15.43>;tag=B49468-7A7 Date: Mon, 21 Mar 2022 11:08:08 GMT Call-ID: 8cdcb780-238168f4-43fb4-500f0a0a@10.10.15.80 CSeq: 101 INVITE Allow-Events: telephone-event Warning: 399 10.10.15.43 "Transcoder Not Configured" Server: Cisco-SIPGateway/IOS-15.4.3.M3 Reason: Q.850;cause=63 Content-Length: 0
Then the service provider answers with a 503 Service Unavailable / 500 Internal Server Error and the SBC states Transcoder Not Configured. You'll have to check with the SP why they are doing this and you should check why the call is not using G711 alaw as you'd want it to do.
