Re: CUBE SIP Trunk Outbound calls failing

Nina · ‎09-27-2017

Hello,

We can't seem to get the outbound calls to work on our SIP Trunk. The inbound calls have no issue.

Please take a look at the trace routes, need help..

003820: *Sep 27 12:34:02.125: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:915147047479@192.168.22.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.248.10:5060;branch=z9hG4bK1548106ad02277
From: <sip:9056711271@192.168.248.10>;tag=13956200~a38f0a9c-d7b3-49cb-a446-3d557da7351f-20514648
To: <sip:915147047479@192.168.22.10>
Date: Wed, 27 Sep 2017 16:31:05 GMT
Call-ID: 40c13000-9cb1d249-d91a6-af8a8c0@192.168.248.10
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM11.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP
Session-ID: 6a53137f00105000a000009e1ede3ddc;remote=00000000000000000000000000000000
Cisco-Guid: 1086402560-0000065536-0000000103-0184068288
Session-Expires: 1800
P-Asserted-Identity: <sip:9056711271@192.168.248.10>
Remote-Party-ID: <sip:9056711271@192.168.248.10>;party=calling;screen=yes;privacy=off
Contact: <sip:9056711271@192.168.248.10:5060>;+u.sip!devicename.ccm.cisco.com="SEP009E1EDE3DDC"
Max-Forwards: 69
Content-Length: 0

003821: *Sep 27 12:34:02.128: //543583/40C130000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.248.10:5060;branch=z9hG4bK1548106ad02277
From: <sip:9056711271@192.168.248.10>;tag=13956200~a38f0a9c-d7b3-49cb-a446-3d557da7351f-20514648
To: <sip:915147047479@192.168.22.10>
Date: Wed, 27 Sep 2017 12:34:02 GMT
Call-ID: 40c13000-9cb1d249-d91a6-af8a8c0@192.168.248.10
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.5.3.S4b
Content-Length: 0

003822: *Sep 27 12:34:02.131: //543584/40C130000000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:15147047479@199.188.188.3:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.10:5060;branch=z9hG4bK7D7B380
From: <sip:9056711271@hgtp.com;user=phone>;tag=21B3A136-1F4E
To: <sip:15147047479@siptrunking.bell.ca>
Date: Wed, 27 Sep 2017 12:34:02 GMT
Call-ID: 81234E75-A2D811E7-8E9C8CD2-340EA8A7@hgtp.com
Supported: 100rel,timer,resource-priority,replaces,histinfo,sdp-anat
Min-SE: 1800
Cisco-Guid: 1086402560-0000065536-0000000103-0184068288
User-Agent: Cisco-SIPGateway/IOS-15.5.3.S4b
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1506530042
Contact: <sip:9056711271;tgrp=3268488;trunk-context=siptrunking.bell.ca@192.168.22.10:5060>
History-Info: <sip:15147047479@199.188.188.3:5060>;index=1
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 68
P-Asserted-Identity: <sip:9056711271@hgtp.com;user=phone>
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 250

v=0
o=CiscoSystemsSIP-GW-UserAgent 1601 6295 IN IP4 192.168.22.10
s=SIP Call
c=IN IP4 192.168.22.10
t=0 0
m=audio 13110 RTP/AVP 0 101
c=IN IP4 192.168.22.10
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Slavik Bialik · ‎09-27-2017

Hi Nina, Please show us the full sip debug messages. You just showed us the INVITE that is outgoing to the PSTN but not the message that is returned from the PSTN.

Nina · ‎09-28-2017

Hello Slavik,

Here is the full sip debug message:

003820: *Sep 27 12:34:02.125: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:915147047479@192.168.22.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.248.10:5060;branch=z9hG4bK1548106ad02277
From: <sip:9056711271@192.168.248.10>;tag=13956200~a38f0a9c-d7b3-49cb-a446-3d557da7351f-20514648
To: <sip:915147047479@192.168.22.10>
Date: Wed, 27 Sep 2017 16:31:05 GMT
Call-ID: 40c13000-9cb1d249-d91a6-af8a8c0@192.168.248.10
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM11.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP
Session-ID: 6a53137f00105000a000009e1ede3ddc;remote=00000000000000000000000000000000
Cisco-Guid: 1086402560-0000065536-0000000103-0184068288
Session-Expires: 1800
P-Asserted-Identity: <sip:9056711271@192.168.248.10>
Remote-Party-ID: <sip:9056711271@192.168.248.10>;party=calling;screen=yes;privacy=off
Contact: <sip:9056711271@192.168.248.10:5060>;+u.sip!devicename.ccm.cisco.com="SEP009E1EDE3DDC"
Max-Forwards: 69
Content-Length: 0

003821: *Sep 27 12:34:02.128: //543583/40C130000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.248.10:5060;branch=z9hG4bK1548106ad02277
From: <sip:9056711271@192.168.248.10>;tag=13956200~a38f0a9c-d7b3-49cb-a446-3d557da7351f-20514648
To: <sip:915147047479@192.168.22.10>
Date: Wed, 27 Sep 2017 12:34:02 GMT
Call-ID: 40c13000-9cb1d249-d91a6-af8a8c0@192.168.248.10
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.5.3.S4b
Content-Length: 0

003822: *Sep 27 12:34:02.131: //543584/40C130000000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:15147047479@199.188.188.3:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.10:5060;branch=z9hG4bK7D7B380
From: <sip:9056711271@hgtp.com;user=phone>;tag=21B3A136-1F4E
To: <sip:15147047479@siptrunking.bell.ca>
Date: Wed, 27 Sep 2017 12:34:02 GMT
Call-ID: 81234E75-A2D811E7-8E9C8CD2-340EA8A7@hgtp.com
Supported: 100rel,timer,resource-priority,replaces,histinfo,sdp-anat
Min-SE: 1800
Cisco-Guid: 1086402560-0000065536-0000000103-0184068288
User-Agent: Cisco-SIPGateway/IOS-15.5.3.S4b
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1506530042
Contact: <sip:9056711271;tgrp=3268488;trunk-context=siptrunking.bell.ca@192.168.22.10:5060>
History-Info: <sip:15147047479@199.188.188.3:5060>;index=1
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 68
P-Asserted-Identity: <sip:9056711271@hgtp.com;user=phone>
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 250

v=0
o=CiscoSystemsSIP-GW-UserAgent 1601 6295 IN IP4 192.168.22.10
s=SIP Call
c=IN IP4 192.168.22.10
t=0 0
m=audio 13110 RTP/AVP 0 101
c=IN IP4 192.168.22.10
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

003823: *Sep 27 12:34:02.150: //543584/40C130000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 205.204.70.71;branch=z9hG4bK7D7B380;received=205.204.70.71;rport=54080
From: <sip:9056711271@hgtp.com;user=phone>;tag=21B3A136-1F4E
To: <sip:15147047479@siptrunking.bell.ca>
Call-ID: 81234E75-A2D811E7-8E9C8CD2-340EA8A7@hgtp.com
CSeq: 101 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:15147047479@199.188.188.3:5060>
Content-Length: 0

003824: *Sep 27 12:34:02.185: //543584/40C130000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 205.204.70.71;branch=z9hG4bK7D7B380;received=205.204.70.71;rport=54080
From: <sip:9056711271@hgtp.com;user=phone>;tag=21B3A136-1F4E
To: <sip:15147047479@siptrunking.bell.ca>;tag=as5eac6e5c
Call-ID: 81234E75-A2D811E7-8E9C8CD2-340EA8A7@hgtp.com
CSeq: 101 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:15147047479@199.188.188.3:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 237

v=0
o=root 548566970 548566970 IN IP4 199.188.188.3
s=Asterisk PBX 1.8.21.0
c=IN IP4 199.188.188.3
t=0 0
m=audio 19138 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

003825: *Sep 27 12:34:02.188: //543583/40C130000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.248.10:5060;branch=z9hG4bK1548106ad02277
From: <sip:9056711271@192.168.248.10>;tag=13956200~a38f0a9c-d7b3-49cb-a446-3d557da7351f-20514648
To: <sip:915147047479@192.168.22.10>;tag=21B3A16F-2571
Date: Wed, 27 Sep 2017 12:34:02 GMT
Call-ID: 40c13000-9cb1d249-d91a6-af8a8c0@192.168.248.10
CSeq: 101 INVITE
Require: 100rel
RSeq: 2124
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
P-Asserted-Identity: <sip:15147047479@hgtp.com>
Contact: <sip:915147047479@192.168.22.10:5060>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.5.3.S4b
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 249

v=0
o=CiscoSystemsSIP-GW-UserAgent 974 4461 IN IP4 192.168.22.10
s=SIP Call
c=IN IP4 192.168.22.10
t=0 0
m=audio 13108 RTP/AVP 0 101
c=IN IP4 192.168.22.10
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

003826: *Sep 27 12:34:02.208: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
PRACK sip:915147047479@192.168.22.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.248.10:5060;branch=z9hG4bK154811322b3701
From: <sip:9056711271@192.168.248.10>;tag=13956200~a38f0a9c-d7b3-49cb-a446-3d557da7351f-20514648
To: <sip:915147047479@192.168.22.10>;tag=21B3A16F-2571
Date: Wed, 27 Sep 2017 16:31:05 GMT
Call-ID: 40c13000-9cb1d249-d91a6-af8a8c0@192.168.248.10
User-Agent: Cisco-CUCM11.5
CSeq: 102 PRACK
RAck: 2124 101 INVITE
Allow-Events: presence
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 231

v=0
o=CiscoSystemsCCM-SIP 13956200 1 IN IP4 192.168.248.10
s=SIP Call
c=IN IP4 192.168.64.101
b=TIAS:64000
b=AS:64
t=0 0
m=audio 18568 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

003827: *Sep 27 12:34:02.210: //543583/40C130000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.248.10:5060;branch=z9hG4bK154811322b3701
From: <sip:9056711271@192.168.248.10>;tag=13956200~a38f0a9c-d7b3-49cb-a446-3d557da7351f-20514648
To: <sip:915147047479@192.168.22.10>;tag=21B3A16F-2571
Date: Wed, 27 Sep 2017 12:34:02 GMT
Call-ID: 40c13000-9cb1d249-d91a6-af8a8c0@192.168.248.10
Server: Cisco-SIPGateway/IOS-15.5.3.S4b
CSeq: 102 PRACK
Content-Length: 0

HOL-VG01#u a
003832: *Sep 27 12:34:05.099: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
CANCEL sip:915147047479@192.168.22.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.248.10:5060;branch=z9hG4bK1548106ad02277
From: <sip:9056711271@192.168.248.10>;tag=13956200~a38f0a9c-d7b3-49cb-a446-3d557da7351f-20514648
To: <sip:915147047479@192.168.22.10>
Date: Wed, 27 Sep 2017 16:31:05 GMT
Call-ID: 40c13000-9cb1d249-d91a6-af8a8c0@192.168.248.10
User-Agent: Cisco-CUCM11.5
CSeq: 101 CANCEL
Max-Forwards: 70
Session-ID: 6a53137f00105000a000009e1ede3ddc;remote=206acbdcdf9249ea5cfd75ab13956200
Content-Length: 0

003833: *Sep 27 12:34:05.100: //543583/40C130000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.248.10:5060;branch=z9hG4bK1548106ad02277
From: <sip:9056711271@192.168.248.10>;tag=13956200~a38f0a9c-d7b3-49cb-a446-3d557da7351f-20514648
To: <sip:915147047479@192.168.22.10>
Date: Wed, 27 Sep 2017 12:34:05 GMT
Call-ID: 40c13000-9cb1d249-d91a6-af8a8c0@192.168.248.10
CSeq: 101 CANCEL
Content-Length: 0

003834: *Sep 27 12:34:05.101: //543584/40C130000000/SIP/Msg/ccsipDisplayMsg:
Sent:
CANCEL sip:15147047479@199.188.188.3:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.10:5060;branch=z9hG4bK7D7B380
From: <sip:9056711271@199.188.188.3>;tag=21B3A136-1F4E
To: <sip:15147047479@199.188.188.3>
Date: Wed, 27 Sep 2017 12:34:02 GMT
Call-ID: 81234E75-A2D811E7-8E9C8CD2-340EA8A7@hgtp.com
CSeq: 101 CANCEL
Max-Forwards: 70
Timestamp: 1506530045
Reason: Q.850;cause=16
Content-Length: 0

003835: *Sep 27 12:34:05.102: //543583/40C130000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.168.248.10:5060;branch=z9hG4bK1548106ad02277
From: <sip:9056711271@192.168.248.10>;tag=13956200~a38f0a9c-d7b3-49cb-a446-3d557da7351f-20514648
To: <sip:915147047479@192.168.22.10>;tag=21B3A16F-2571
Date: Wed, 27 Sep 2017 12:34:05 GMT
Call-ID: 40c13000-9cb1d249-d91a6-af8a8c0@192.168.248.10
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.5.3.S4b
Reason: Q.850;cause=16
Content-Length: 0

003836: *Sep 27 12:34:05.120: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:915147047479@192.168.22.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.248.10:5060;branch=z9hG4bK1548106ad02277
From: <sip:9056711271@192.168.248.10>;tag=13956200~a38f0a9c-d7b3-49cb-a446-3d557da7351f-20514648
To: <sip:915147047479@192.168.22.10>;tag=21B3A16F-2571
Date: Wed, 27 Sep 2017 16:31:05 GMT
Call-ID: 40c13000-9cb1d249-d91a6-af8a8c0@192.168.248.10
User-Agent: Cisco-CUCM11.5
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0

003837: *Sep 27 12:34:05.122: //543584/40C130000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 205.204.70.71;branch=z9hG4bK7D7B380;received=205.204.70.71;rport=54080
From: <sip:9056711271@hgtp.com;user=phone>;tag=21B3A136-1F4E
To: <sip:15147047479@siptrunking.bell.ca>;tag=as5eac6e5c
Call-ID: 81234E75-A2D811E7-8E9C8CD2-340EA8A7@hgtp.com
CSeq: 101 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

003838: *Sep 27 12:34:05.123: //543584/40C130000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 205.204.70.71;branch=z9hG4bK7D7B380;received=205.204.70.71;rport=54080
From: <sip:9056711271@199.188.188.3>;tag=21B3A136-1F4E
To: <sip:15147047479@199.188.188.3>;tag=as5eac6e5c
Call-ID: 81234E75-A2D811E7-8E9C8CD2-340EA8A7@hgtp.com
CSeq: 101 CANCEL
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

003839: *Sep 27 12:34:05.123: //543584/40C130000000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:15147047479@199.188.188.3:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.10:5060;branch=z9hG4bK7D7B380
From: <sip:9056711271@199.188.188.3>;tag=21B3A136-1F4E
To: <sip:15147047479@siptrunking.bell.ca>;tag=as5eac6e5c
Date: Wed, 27 Sep 2017 12:34:02 GMT
Call-ID: 81234E75-A2D811E7-8E9C8CD2-340EA8A7@hgtp.com
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

Dennis Mink · ‎09-27-2017

can you send all debug info?

also what I noticed, and what could cause issues, is the fact that CUC>CUBE uses SIP delayed offer and from CUBE>Provider uses Early offer.

Please remember to rate useful posts, by clicking on the stars below.

Andrew West · ‎09-27-2017

Also does your ITSP require any sort of validation to place calls to the PSTN?

MM · ‎09-28-2017

What is happening when you attempt to place an outbound call? Is this a new implementation, or did this just start happening?

Nina · ‎09-28-2017

This is a new implementation, the call cannot be completed, the message is "The number you dialed does not exist".

MM · ‎09-28-2017

Can you post your SIP Dial Peers and any translation profiles/rules that you have configured on your CUBE?

Nina · ‎09-28-2017

Here are the dial-peer and the translation I'm using for the outbound to SIP Trunk. However, it's more the sip-profiles (see below) I think is not properly configured.

voice translation-rule 3
rule 1 /^9\(.*\)/ /\1/

voice translation-profile STRIP_9
translate called 3

dial-peer voice 101 voip
description WAN Outgoing to SIP Trunk
translation-profile outgoing STRIP_9
destination-pattern 9[2-9]..[2-9]......
session protocol sipv2
session target ipv4:199.188.189.3
voice-class sip profiles 101
voice-class sip options-keepalive down-interval 20
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 102 voip
description WAN Outgoing to SIP Trunk
translation-profile outgoing STRIP_9
destination-pattern 91[2-9]..[2-9]......
session protocol sipv2
session target ipv4:199.188.189.3
voice-class sip profiles 101
voice-class sip options-keepalive down-interval 20
dtmf-relay rtp-nte
codec g711ulaw
no vad

voice class sip-profiles 101

request INVITE sip-header SIP-Req-URI modify "@.*:" "199.188.189.3"

request INVITE sip-header Diversion modify "(@.*)>" "@205.204.70.71>"

request INVITE sip-header Contact modify "@" ";tgrp=3268488;trunk-context=199.188.189.3@"

request INVITE sip-header From modify "(@.*)>" "@205.204.70.71>"

request INVITE sip-header P-Asserted-Identity modify "(@.*)>" "@205.204.70.71;user=phone>"

request INVITE sip-header To modify "@.*>" "@199.188.189.3>"

request INVITE sip-header Diversion modify "reason=unconditional" ""

request INVITE sip-header Diversion modify "reason=user-busy" ""

request INVITE sip-header Diversion modify "reason=no-answer" ""

request REINVITE sip-header From modify "(@.*)>" "@205.204.70.71>"

request REINVITE sip-header P-Asserted-Identity modify "(@.*)>" "@205.204.70.71;user=phone>"

request REINVITE sip-header Diversion modify "(@.*)>" "@205.204.70.71>"

MM · ‎09-29-2017

Do you know how many digits your carrier is expecting to recieve from you? Are you using an E.164 dial plan?

For testing, I'd strip down your dial-peers and get rid of the sip profiles, digit strip, etc. Here is sample config from my lab which is working, if this is in a lab enviroment for you try to paste these dial-peers and test.

dial-peer voice 200 voip

description outbound dial-peer to SP
destination-pattern 9[2-9]..[2-9]......
session protocol sipv2
session target ipv4 sip-server
codec g711ulaw

dtmf-relay rtp-nte

no vad

dial-peer voice 201 voip

description outbound dial-peer to SP
destination-pattern 91[2-9]..[2-9]......
session protocol sipv2
session target ipv4 sip-server
codec g711ulaw

dtmf-relay rtp-nte
no vad

Nina · ‎09-29-2017

The carrier receives 10 digits

Nina · ‎09-29-2017

Didn't work

see attached trace

Slavik Bialik · ‎09-29-2017

What are the chances that your ITSP require SIP registration before sending SIP INVITEs to him? If so, you'll have to add the relevant commands for it.

Anyway, did you try to reach your ITSP? Maybe the issue is on their end.

Olayinka Popoola · ‎04-12-2019

I'm experiencing the exact issue..did you get a workaround?

Nina · ‎04-12-2019

The issue was solved it was the customer's phone system.

the CUCM was sending a PAI field in his invite but asterisk didn't not recognize it. It was removed and all worked fine.