Hi, everyone.
I have a router voice gateway 4331 ISR who has a SIP Trunk from a ITSP "B", but, when I try to make a call to PSTN, it doesn't work to a specifically carrier "Telcel", incomings and outgoings calls. Only to/from this carrier doesn't work. Calls to others carriers works fine.
I debug the call and show me this:
004736: Oct 20 21:19:01.326: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:8719042710@10.137.10.13:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.187.128.3:5060;branch=z9hG4bK4hdjek3018nq7pr01a90.1
Call-ID: ej898e5kjgj5leellxlx52k85il4i5lj@B.5.181.ims.enlacetpe.com
From: "8120220496"<sip:8120220496@ims.enlacetpe.com:5060;noa=international;srvattri=national;cpc=ordinary>;tag=w1wxxwgf
To: <sip:8719042710@10.137.10.13:5060;transport=udp;user=phone>
CSeq: 1 INVITE
Allow: INVITE,ACK,BYE,CANCEL,UPDATE,INFO,PRACK,NOTIFY,REFER,SUBSCRIBE,OPTIONS,MESSAGE
P-Charging-Vector: icid-value=mgcf--20221020202625-101720101810000078133;term-ioi=ims.enlacetpe.com
Max-Forwards: 62
Supported: timer,100rel,histinfo,early-session
Session-Expires: 1800
Min-SE: 600
P-Asserted-Identity: <tel:8120220496>
Contact: <sip:10.187.128.3:5060;Dpt=f73a-200;TRC=ffffffff-ffffffff;transport=udp>
Content-Length: 227
Content-Type: application/sdp
Content-Disposition: session
Route: <sip:8719042710;tgrp=tstl-e0200515767;trunk-context=ims.enlacetpe.com@10.137.10.13:5060;user=phone;lr>
P-Early-Media: supported
v=0
o=HuaweiATS9900 14205564 14205564 IN IP4 10.187.128.3
s=SipCall
c=IN IP4 10.187.128.3
t=0 0
m=audio 61720 RTP/AVP 8 18 97
a=fmtp:18 annexb=no
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=sendrecv
a=ptime:20
004737: Oct 20 21:19:01.332: //335/8FAF55C78279/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.187.128.3:5060;branch=z9hG4bK4hdjek3018nq7pr01a90.1
From: "8120220496"<sip:8120220496@ims.enlacetpe.com:5060;noa=international;srvattri=national;cpc=ordinary>;tag=w1wxxwgf
To: <sip:8719042710@10.137.10.13:5060;transport=udp;user=phone>
Date: Fri, 21 Oct 2022 02:19:01 GMT
Call-ID: ej898e5kjgj5leellxlx52k85il4i5lj@B.5.181.ims.enlacetpe.com
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.4.3.S4
Content-Length: 0
004738: Oct 20 21:19:01.336: //335/8FAF55C78279/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.187.128.3:5060;branch=z9hG4bK4hdjek3018nq7pr01a90.1
From: "8120220496"<sip:8120220496@ims.enlacetpe.com:5060;noa=international;srvattri=national;cpc=ordinary>;tag=w1wxxwgf
To: <sip:8719042710@10.137.10.13:5060;transport=udp;user=phone>;tag=7B93C9-839
Date: Fri, 21 Oct 2022 02:19:01 GMT
Call-ID: ej898e5kjgj5leellxlx52k85il4i5lj@B.5.181.ims.enlacetpe.com
CSeq: 1 INVITE
Allow-Events: telephone-event
Warning: 399 10.137.10.13 "Transcoder Not Configured"
Server: Cisco-SIPGateway/IOS-15.4.3.S4
Reason: Q.850;cause=47
Content-Length: 0
------------------------------------------------
dial-peer voice 300 voip
description SIP_TRUNK_B
translation-profile incoming Profile_IN
translation-profile outgoing Profile_OUT
destination-pattern [0-9]T
session protocol sipv2
session target ipv4:10.187.128.3
session transport udp
incoming called-number .
voice-class codec 1
dtmf-relay rtp-nte
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
codec preference 6 g722-48
codec preference 7 g722-56
codec preference 8 g722-64
codec preference 9 g723ar53
codec preference 10 g723ar63
codec preference 11 g723r53
codec preference 12 g723r63
codec preference 13 g726r16
codec preference 14 g726r24
codec preference 15 g726r32
codec preference 16 g728
codec preference 17 g729br8
I have another SIP Trunk from ITSP "A" and all calls work OK in the same router.
Any suggestions on this?
