Outgoing Calls with SIP Trunk fail

[Solomon Rufus Kavala]

I have a SIP trunk from a TSP terminated on the CUBE. - c3925 gateway connected via H323 & SIP to a CUCM 8.5 cluster. Incoming calls are working fine, but outgoing calls fail with "SIP/2.0 488 Not Acceptable Media" and also with Warning: 304 10.180.174.1 "Media Type(s) Unavailable" from CCSIP Debugs. I have attached Voice configuration, Dial-peers configured in the gateway and also debug of Outgoing call. Any general advise would be much appreciated.

Thanks,

Solomon.

[Ayodeji Okanlawon]

Solomon,

There are a few things wrong with your config.

1. From the trace....you have a sip trunk from CUCM to voice gateway...CUCM sends an invite with early offer and G711alaw codec

Received:

INVITE sip:13304903845@10.180.174.1:5060 SIP/2.0

Via: SIP/2.0/TCP 10.180.174.21:5060;branch=z9hG4bK175d77e524

From: "Solomon Kavala 80226" <19725380226>;tag=30898~66cfeef1-fb6d-46b8-9b3a-8ad21c4b6307-23209181

To: <13304903845>

v=0
o=CiscoSystemsCCM-SIP 30898 1 IN IP4 10.180.174.21
s=SIP Call
c=IN IP4 10.180.174.21
t=0 0
m=audio 24976 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000

2. Your inbound dial-peer routing calls from CUCM is however configured for

a. h323 protocol and b. g729 codec

dial-peer voice 4444 voip

destination-pattern 7...

session target ipv4:10.180.174.21

incoming called-number .

dtmf-relay h245-alphanumeric

ip qos dscp cs3 signaling

no vad

To resolve your issues you will need to adjust your config as follows:

1. Configure a inbound dial-peer that matches calls to your ITSP to use sip protocol and G711a as the codec..

e.g

dial-peer voice x voip-------------where X is any number you want to assign

incoming called number 13304903845

session protocol sipv2

codec g711alaw

NB: this config will only work for the configured number..You can define a range of patterns so as to allow all outboubd calls to your sip provider match this dial-peer

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[Solomon Rufus Kavala]

Hello Sir,

I have added inbound dial-peer 4000 which use SIP Protocol andG711alaw as codec. But still got fastbusy.

dial-peer voice 4000 voip

session protocol sipv2

incoming called-number 13304903845

codec g711alaw

Thanks,

Solomon.

[Ayodeji Okanlawon]

Solomon,

So that looks much better than before. But now we see your ITSP telling us that the request timed out

*Jan  3 14:15:26 UTC: //3617/93F69D800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.180.174.1:5060;branch=z9hG4bK4EB5
From: "Solomon Kavala 80226" <1972539725380226>;tag=3C658BE8-89D
To: <13304903845>
Call-ID: DAD8AFE0-54E611E2-9A0C99C3-5100E663@10.180.174.1
CSeq: 101 INVITE
Timestamp: 1357222526

us-tx-coppell-rtr#
*Jan  3 14:15:35 UTC: //3617/93F69D800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 10.180.174.1:5060;branch=z9hG4bK4EB5
From: "Solomon Kavala 80226" <1972539725380226>;tag=3C658BE8-89D
To: <13304903845>;tag=aprqngfrt-01uh9430000a6
Call-ID: DAD8AFE0-54E611E2-9A0C99C3-5100E663@10.180.174.1
CSeq: 101 INVITE
Timestamp: 1357222526
Content-Length: 0

You need to open a ticket with your service provider and find out why they cant route the number you dialled

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

[Ayodeji Okanlawon]

On another note is this number part of your DDI allocated by your sip provider..."1972539725380226"

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

[Solomon Rufus Kavala]

Thank you for your response.Verizon provided 9725380226 as a test DID,whcih I have configured in my CIPC.I have also configured my CIPC with

External Phone Number Mask set as 19725380226 and enabled the Calling Party Transformations to use External Phone Number Mask in Route Patterns. Today when I ran SIP debugs, got different error "SIP/2.0 503 Service Unavailable". Still I need to check with ITSP on this Issue or Do I need to tweak my configuration further.Debugs were attached blow. 

Thanks

Solomon.

[Ayodeji Okanlawon]

Solomon,

For a start, your external mask should be the test DDI Verizon provided (by the way I work for VzB) So configure your mask to as the DDI and remove the leading 1.

Second this present issue has nothing to do with VzB. Its your gateway sending this error. Can you send your sh run again. It seems something has changed.

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

[Solomon Rufus Kavala]

         It is really great to know that you are working with VzB.As per your comments,I have removed leading 1 on the Test DID under External number mask.Plaese find my running configuration of CUBE.

CUCM - 10.180.174.21

CUBE - 10.180.174.1

Thanks,

Solomon.

[Ayodeji Okanlawon]

Why are you using port 5170 on your dial-peers? Can you change your config and remove this port..as ff: (NB I have removed 5170 fro your dial-peer 100)

dial-peer voice 100 voip

description OUTBOUND G711 Voice SIP calls to VzB

translation-profile outgoing Outgoing-Translate

destination-pattern .T

session protocol sipv2

session target ipv4:172.31.134.49

voice-class codec 1 

voice-class sip early-offer forced

dtmf-relay rtp-nte

no vad

Do another test call with the new config and send a new debug ccsip messages. Remove the other debugs and send only the sip debugs

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

[Solomon Rufus Kavala]

5170 is the VzB SBC Signaling Port and 172.31.134.49 is VzB SBC Signaling IP. As per your suggestion,I have removed VzB SBC Signaling Port 5170 on both Dial-peers 100 and 101. Ran SIP debug,while calling out.Got the same error "SIP/2.0 503 Service Unavailable".

[Ayodeji Okanlawon]

Ok,

So here is what I see. You are doing early offer from cucm with g711alaw, and you have voice class codec configured with g711ulaw as te preferred codec...Hence creating a codec mismatch.

Yesterday you configured codec g711alaw.

dial-peer voice 4000 voip

session protocol sipv2

incoming called-number 13304903845

codec g711alaw

We can see the reason for service unavailable..cause code 47 (which means codec related issue)

Sent:

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/TCP 10.180.174.21:5060;branch=z9hG4bK2430f868c2

From: "Solomon Kavala 80226" <19725380226>;tag=33652~66cfeef1-fb6d-46b8-9b3a-8ad21c4b6307-23211303

To: <13304903845>;tag=40741C68-0

Date: Fri, 04 Jan 2013 09:09:49 GMT

Call-ID:

cac3700-e619d4f-25-15aeb40a@10.180.174.21

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=47

There is no point using voice class codec when you have already set the codec from cucm to use g711alaw. So remove the voice class codec and set the codec to use g711alaw as you did yesterday

Pls send debugs again if it doesnt work

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

[Solomon Rufus Kavala]

      I have removed Voice Class Codec 1 from the cofiguration and ran test call,but no luck. Please find the debugs of outbound call.

[Ayodeji Okanlawon]

You need to add codec g711alaw after you remove voice class codec.

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

[Solomon Rufus Kavala]

        My bad,I have added Codec g711alaw on both incoming and outgoing dial-peers (101 & 100) to ITSP and this time, after I dial the digits got Request Timed Out from ITSP. Please find the debugs.

[Ayodeji Okanlawon]

OK. Your transation rule is modifying the CLI you are sending to VzB.

The call come in from CUCM with a te correct CLI 972580226

INVITE sip:19784255217@10.180.174.1:5060 SIP/2.0

Via: SIP/2.0/TCP 10.180.174.21:5060;branch=z9hG4bK2d5bd0e0bc

From: "Solomon Kavala 80226" <>9725380226@10.180.174.21>;tag=33921~66cfeef1-fb6d-46b8-9b3a-8ad21c4b6307-23211330

But when the call is sent to VzB the translatio rule is changing it to this 972539725380226.

Remote-Party-ID: "Solomon Kavala 80226" <>972539725380226@10.180.174.1>;party=calling;screen=yes;privacy=off

From: "Solomon Kavala 80226" <>972539725380226@10.180.174.1>;tag=40E12648-237

voice translation-rule 3

rule 1 /80226/ /9725380226/

rule 2 /^7\(...\)/ /4695497\1/

You need to modify this tranlsation rule...

voice translation-rule 3
rule 1 /^80226/ /9725380226/----------------------NB I have added a ^ to the beginning of 80226.

This will ensure that its only numbers that start with 80226 that will be xlated not any where 80226 appears in a call, because what is happening is that in the CLI 9725380226, the gateway is changing 80226 to 9725380226 and you end up with 972539725380226

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson