SIP Trunk, Incoming works, outgoing doesn't

[acollard83]

What am I missing on this. The relevant config is below as well as some SIP logs. It doesn't look like it's even trying to hit the SBC/ITSP gateway.

dial-peer voice 100 voip
description Incoming dialpeer and 1+10 digits to SBC
translation-profile incoming in_rmv_plus1
destination-pattern ^1[2-9]..[2-9]......$
voice-class codec 1
voice-class sip early-offer forced
session protocol sipv2
session target ipv4:162.210.240.XX
session transport tcp
incoming called-number .T
dtmf-relay rtp-nte
!
dial-peer voice 101 voip
description 10-digit local calls to SBC
destination-pattern ^[2-9]..[2-9]......$
voice-class codec 1
voice-class sip early-offer forced
session protocol sipv2
session target ipv4:162.210.240.XX
session transport tcp
dtmf-relay rtp-nte
!
dial-peer voice 102 voip
description International calls to SBC
destination-pattern ^011T
voice-class codec 1
voice-class sip early-offer forced
session protocol sipv2
session target ipv4:162.210.240.XX
session transport tcp
dtmf-relay rtp-nte
!
dial-peer voice 103 voip
description N11 calls to SBC
destination-pattern ^[2-9]11$
voice-class codec 1
voice-class sip early-offer forced
session protocol sipv2
session target ipv4:162.210.240.XX
session transport tcp
dtmf-relay rtp-nte
!
dial-peer voice 300 voip
description 5178363XXX calls to CUCM
huntstop
preference 1
destination-pattern ^5178363...$
voice-class codec 1
voice-class sip early-offer forced
session protocol sipv2
session target ipv4:10.1.20.10
dtmf-relay rtp-nte
!
dial-peer voice 302 voip
description Incoming calls from CUCM
huntstop
voice-class codec 1
voice-class sip early-offer forced
session protocol sipv2
session target ipv4:10.1.20.10
incoming called-number .%
dtmf-relay rtp-nte

voice rtp send-recv
!
voice service voip
allow-connections sip to sip
sip
rel1xx disable
header-passing error-passthru
early-offer forced
midcall-signaling passthru
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
!
voice translation-rule 10
rule 1 /^\+1\(..........\)$/ /\1/
!
voice translation-profile in_rmv_plus1
translate called 10


Logs

*Nov 19 16:02:51.298: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:1517441XXXX@10.1.20.1:5060 SIP/2.0

Date: Tue, 19 Nov 2013 16:00:42 GMT

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH

From: "XXX" <sip:5178363XXX@10.1.20.10>;tag=d7bc92a1-20f8-419a-a426-c52b3bebb57e-19575454

Allow-Events: presence

Supported: timer,replaces

Min-SE: 1800

Remote-Party-ID: "XXX" <sip:5178363XXX@10.1.20.10>;party=calling;screen=yes;privacy=off

Content-Length: 208

User-Agent: Cisco-CUCM6.1

To: <sip:1517441XXXX@10.1.20.1>

Contact: <sip:5178363XXX@10.1.20.10:5060;transport=tcp>

Expires: 180

Content-Type: application/sdp

Call-ID: bc897680-28b18b2a-1fb-a14010a@10.1.20.10

Via: SIP/2.0/TCP 10.1.20.10:5060;branch=z9hG4bK257e205c3a

CSeq: 101 INVITE

Session-Expires: 1800

Max-Forwards: 70



v=0

o=CiscoSystemsCCM-SIP 2000 1 IN IP4 10.1.20.10

s=SIP Call

c=IN IP4 10.1.20.10

t=0 0

m=audio 24644 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=ptime:20

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15


*Nov 19 16:02:51.310: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying

Via: SIP/2.0/TCP 10.1.20.10:5060;branch=z9hG4bK257e205c3a

From: "XXX" <sip:5178363XXX@10.1.20.10>;tag=d7bc92a1-20f8-419a-a426-c52b3bebb57e-19575454

To: <sip:1517441XXXX@10.1.20.1>

Date: Tue, 19 Nov 2013 16:02:51 GMT

Call-ID: bc897680-28b18b2a-1fb-a14010a@10.1.20.10

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 101 INVITE

Allow-Events: telephone-event

Content-Length: 0




*Nov 19 16:02:56.322: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 503 Service Unavailable

Via: SIP/2.0/TCP 10.1.20.10:5060;branch=z9hG4bK257e205c3a

From: "XXX" <sip:5178363XXX@10.1.20.10>;tag=d7bc92a1-20f8-419a-a426-c52b3bebb57e-19575454

To: <sip:1517441XXXX@10.1.20.1>;tag=87A778C-157E

Date: Tue, 19 Nov 2013 16:02:51 GMT

Call-ID: bc897680-28b18b2a-1fb-a14010a@10.1.20.10

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 101 INVITE

Allow-Events: telephone-event

Reason: Q.850;cause=38

Content-Length: 0




*Nov 19 16:02:56.326: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:15174411031@10.1.20.1:5060 SIP/2.0

Date: Tue, 19 Nov 2013 16:00:42 GMT

From: "Adam Collard" <sip:5178363721@10.1.20.10>;tag=d7bc92a1-20f8-419a-a426-c52b3bebb57e-19575454

Allow-Events: presence

Content-Length: 0

To: <sip:15174411031@10.1.20.1>;tag=87A778C-157E

Call-ID: bc897680-28b18b2a-1fb-a14010a@10.1.20.10

Via: SIP/2.0/TCP 10.1.20.10:5060;branch=z9hG4bK257e205c3a

CSeq: 101 ACK

Max-Forwards: 70

 

[gnrslash4life]

Do you have patterns set up in CUCM to route out the gateway? Have you looked at DNA to see what that is showing?

 

[whykap]

Also can you share the rest of the SIP config on you router? If that's all you got (dial-peers) it's partial.
Who's the provider?
Do you have a basic topology of call routing?

 

[phrk]

These are all very valid questions. Some clarification/restatement of the issue may definitely help. I think I see a problem tho.

This is what your router received:

Received:
INVITE sip:1517441XXXX@10.1.20.1:5060 SIP/2.0

Perhaps you sanitized the log to hide your phone numbers with the XXXXs? If not, the string following the INVITE sip: method should usually be a string of digits that would match one of your destination patterns, correct? XXXX won't match.

Then your router SENT this message, so it never got to the ITSP.

Sent:
SIP/2.0 503 Service Unavailable
--snip
Reason: Q.850;cause=38


I am assuming, perhaps incorrectly, that the gateway did so because it didn't have a matching destination pattern to forward the invite (because of the XXXXs?)

Do you have a "called party transform mask" configured on the Route Pattern/Translation Pattern you are using to match the Outgoing call in CUCM which maybe masking the last four digits?