Call forward not working with SIP trunk and CUCM

Unanswered Question
Jun 20th, 2012
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We have a branch site with SIP trunk which connects back to HQ CUCM (6.0.1.3000-7) via SIP. (Upgrading to CUCM 8.3 btw )


There are also four PSTN lines at the site as a backup to be used in SRST fail-over mode. All Cisco phones on the site are registered back to HQ CUCM. Customers can make outbound calls and receive inbound calls via SIP trunk without any issue. However, forwarding calls are going out via backup PSTN lines matching one of those dial-peers although they have preference 2.


dial-peer voice 10 voip

description Inbound - SIP from CUCM

session protocol sipv2

session target sip-server

incoming called-number .T

voice-class codec 1

no voice-class sip outbound-proxy

dtmf-relay rtp-nte

fax protocol none

no vad

!

dial-peer voice 22 pots

trunkgroup ALL_FXO

description Outbound - POTS to PSTN

translation-profile outgoing OUTBOUND_STRIPZERO

preference 2

destination-pattern .T

!

dial-peer voice 20 voip

description Outbound - SIP to PSTN

preference 1

destination-pattern .T

session protocol sipv2

session target sip-server

voice-class codec 1

no voice-class sip outbound-proxy

voice-class sip profiles 1

dtmf-relay rtp-nte

fax protocol none

!

I got the following if I call to an extension on the site where I have CForwardAll on that extension to a mobile 0417171234.


SiteAr001#sh voice call status

CallID     CID  ccVdb      Port        Slot/DSP:Ch  Called #   Codec    MLPP Dial-peers

0x7C282    3C2  0x3118B310 0/3/2            0/1:1  *417171234  g711ulaw 10/22

1 active call found

!

Can someone please advise if this can be dial-peer matching issue due to my limited knowledge on dial-peers or possible SIP provider issue on diversion? Thanks much in advance for your sharing.


Regards,

Lay

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Chris Deren Wed, 06/20/2012 - 06:49
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Please post "debug ccsip messages"


Make sure the carrier allows you to send calls with caller ID not belonging to the SIP trunk.


Chris

Ayodeji Okanlawon Wed, 06/20/2012 - 06:52
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This is most likely as issue on diversion to your SIP provider, hence when that failed, the gateway routed the call through the other dial-peer.


Can you send a


debug ccsip messages.


Please let me know what the calling and called number is and possible the diverted number.


With diversion to sip providers, if the Mask of the diverted number is not in your DDI range your provider will reject the call. Funny enough I resolved a similar problem a few hours ago



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"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"

layhlaing Wed, 06/20/2012 - 20:11
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Thanks guys for your replies. Yes, it makes sense to me that SIP provider is not allowing the diverted calls.


Here is output:


SiteAr001#debug ccsip message

SIP Call messages tracing is enabled



SiteAr001#

Jun 21 09:41:15 AWST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP 10.10.1.50:5060;branch=z9hG4bK23a727214430d

Remote-Party-ID: "Alan Rhodes" ;party=calling;screen=yes;privacy=off

From: "Alan Rhodes" ;tag=3cf9bdf1-32ba-4e0c-9392-584cf24ea6f2-27139830

To:

Date: Thu, 21 Jun 2012 01:41:15 GMT

Call-ID: [email protected]

Supported: timer,replaces

Min-SE:  1800

User-Agent: Cisco-CCM6.0

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

CSeq: 101 INVITE

Contact:

Expires: 180

Allow-Events: presence

Session-Expires:  1800

Max-Forwards: 70

Content-Length: 0



Jun 21 09:41:15 AWST: //509764/05E5B4B0B776/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 10.10.1.50:5060;branch=z9hG4bK23a727214430d

From: "Alan Rhodes" ;tag=3cf9bdf1-32ba-4e0c-9392-584cf24ea6f2-27139830

To:

Date: Thu, 21 Jun 2012 01:41:15 GMT

Call-ID: [email protected]

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0



Jun 21 09:41:15 AWST: //509765/05E5B4B0B776/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP 172.17.83.2:5060;branch=z9hG4bK838B91EA4

Remote-Party-ID: "Alan Rhodes" [email protected]>;party=calling;screen=yes;privacy=off

From: "Alan Rhodes" [email protected]>;tag=F8E756B0-11E7

To: [email protected]>

Date: Thu, 21 Jun 2012 01:41:15 GMT

Call-ID: [email protected]

Supported: timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 0098940080-3128496609-3077998146-2929939867

User-Agent: Cisco-SIPGateway/IOS-12.x

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

CSeq: 101 INVITE

Timestamp: 1340242875

Contact:

Expires: 300

Allow-Events: telephone-event

Max-Forwards: 69

Session-Expires:  1800

Content-Length: 0



Jun 21 09:41:15 AWST: //509765/05E5B4B0B776/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP 172.17.83.2:5060;branch=z9hG4bK838B91EA4

Remote-Party-ID: "Alan Rhodes" [email protected]>;party=calling;screen=yes;privacy=off

From: "Alan Rhodes" [email protected]>;tag=F8E756B0-11E7

To: [email protected]>

Date: Thu, 21 Jun 2012 01:41:15 GMT

Call-ID: [email protected]

Supported: timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 0098940080-3128496609-3077998146-2929939867

User-Agent: Cisco-SIPGateway/IOS-12.x

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

CSeq: 101 INVITE

Timestamp: 1340242875

Contact:

Expires: 300

Allow-Events: telephone-event

Max-Forwards: 69

Session-Expires:  1800

Content-Length: 0



Jun 21 09:41:16 AWST: //509765/05E5B4B0B776/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP 172.17.83.2:5060;branch=z9hG4bK838B91EA4

Remote-Party-ID: "Alan Rhodes" [email protected]>;party=calling;screen=yes;privacy=off

From: "Alan Rhodes" [email protected]>;tag=F8E756B0-11E7

To: [email protected]>

Date: Thu, 21 Jun 2012 01:41:16 GMT

Call-ID: [email protected]

Supported: timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 0098940080-3128496609-3077998146-2929939867

User-Agent: Cisco-SIPGateway/IOS-12.x

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

CSeq: 101 INVITE

Timestamp: 1340242876

Contact:

Expires: 300

Allow-Events: telephone-event

Max-Forwards: 69

Session-Expires:  1800

Content-Length: 0



Jun 21 09:41:18 AWST: //509765/05E5B4B0B776/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP 172.17.83.2:5060;branch=z9hG4bK838B91EA4

Remote-Party-ID: "Alan Rhodes" [email protected]>;party=calling;screen=yes;privacy=off

From: "Alan Rhodes" [email protected]>;tag=F8E7645C-2386

To: [email protected]>

Date: Thu, 21 Jun 2012 01:41:18 GMT

Call-ID: [email protected]

Supported: timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 0098940080-3128496609-3077998146-2929939867

User-Agent: Cisco-SIPGateway/IOS-12.x

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

CSeq: 101 INVITE

Timestamp: 1340242878

Contact:

Expires: 300

Allow-Events: telephone-event

Max-Forwards: 69

Session-Expires:  1800

Content-Length: 0



Jun 21 09:41:19 AWST: //509765/05E5B4B0B776/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP 172.17.83.2:5060;branch=z9hG4bK838B91EA4

Remote-Party-ID: "Alan Rhodes" [email protected]>;party=calling;screen=yes;privacy=off

From: "Alan Rhodes" [email protected]>;tag=F8E7645C-2386

To: [email protected]>

Date: Thu, 21 Jun 2012 01:41:19 GMT

Call-ID: [email protected]

Supported: timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 0098940080-3128496609-3077998146-2929939867

User-Agent: Cisco-SIPGateway/IOS-12.x

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

CSeq: 101 INVITE

Timestamp: 1340242879

Contact:

Expires: 300

Allow-Events: telephone-event

Max-Forwards: 69

Session-Expires:  1800

Content-Length: 0



Jun 21 09:41:20 AWST: //509765/05E5B4B0B776/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP 172.17.83.2:5060;branch=z9hG4bK838B91EA4

Remote-Party-ID: "Alan Rhodes" [email protected]>;party=calling;screen=yes;privacy=off

From: "Alan Rhodes" [email protected]>;tag=F8E7645C-2386

To: [email protected]>

Date: Thu, 21 Jun 2012 01:41:20 GMT

Call-ID: [email protected]

Supported: timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 0098940080-3128496609-3077998146-2929939867

User-Agent: Cisco-SIPGateway/IOS-12.x

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

CSeq: 101 INVITE

Timestamp: 1340242880

Contact:

Expires: 300

Allow-Events: telephone-event

Max-Forwards: 69

Session-Expires:  1800

Content-Length: 0



Jun 21 09:41:23 AWST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

REGISTER sip:as.nipt.telstra.com:5060 SIP/2.0

Via: SIP/2.0/UDP 172.17.83.2:5060;branch=z9hG4bK838BA1AC3

From: [email protected]>;tag=F8E776EC-A56

To: [email protected]>

Date: Thu, 21 Jun 2012 01:41:23 GMT

Call-ID: B102CC6A-B9E111E1-B5048642-AEA3559B

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 70

Timestamp: 1340242883

CSeq: 363 REGISTER

Contact:

Expires:  3600

Supported: path

Content-Length: 0



Jun 21 09:41:23 AWST: //509767/000000000000/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 404 Not found

Via: SIP/2.0/UDP 172.17.83.2:5060;branch=z9hG4bK838BA1AC3

From: [email protected]>;tag=F8E776EC-A56

To: [email protected]>;tag=454442420-1340242883452

Call-ID: B102CC6A-B9E111E1-B5048642-AEA3559B

Timestamp: 1340242883

CSeq: 363 REGISTER

Content-Length: 0



Jun 21 09:41:25 AWST: //509764/05E5B4B0B776/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 10.10.1.50:5060;branch=z9hG4bK23a727214430d

From: "Alan Rhodes" ;tag=3cf9bdf1-32ba-4e0c-9392-584cf24ea6f2-27139830

To: ;tag=F8E77F70-1C98

Date: Thu, 21 Jun 2012 01:41:15 GMT

Call-ID: [email protected]

CSeq: 101 INVITE

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

Allow-Events: telephone-event

Remote-Party-ID: ;party=called;screen=no;privacy=off

Contact:

Supported: sdp-anat

Server: Cisco-SIPGateway/IOS-12.x

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 303


v=0

o=CiscoSystemsSIP-GW-UserAgent 3809 6909 IN IP4 172.17.83.2

s=SIP Call

c=IN IP4 172.17.83.2

t=0 0

m=audio 23036 RTP/AVP 8 0 18 101

c=IN IP4 172.17.83.2

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16


Jun 21 09:41:25 AWST: //509764/05E5B4B0B776/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.10.1.50:5060;branch=z9hG4bK23a727214430d

From: "Alan Rhodes" ;tag=3cf9bdf1-32ba-4e0c-9392-584cf24ea6f2-27139830

To: ;tag=F8E77F70-1C98

Date: Thu, 21 Jun 2012 01:41:15 GMT

Call-ID: [email protected]

CSeq: 101 INVITE

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

Allow-Events: telephone-event

Remote-Party-ID: ;party=called;screen=no;privacy=off

Contact:

Supported: replaces

Supported: sdp-anat

Server: Cisco-SIPGateway/IOS-12.x

Require: timer

Session-Expires:  1800;refresher=uac

Supported: timer

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 303


v=0

o=CiscoSystemsSIP-GW-UserAgent 3809 6909 IN IP4 172.17.83.2

s=SIP Call

c=IN IP4 172.17.83.2

t=0 0

m=audio 23036 RTP/AVP 8 0 18 101

c=IN IP4 172.17.83.2

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16


Jun 21 09:41:25 AWST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP 10.10.1.50:5060;branch=z9hG4bK23a735933bd05

From: "Alan Rhodes" ;tag=3cf9bdf1-32ba-4e0c-9392-584cf24ea6f2-27139830

To: ;tag=F8E77F70-1C98

Date: Thu, 21 Jun 2012 01:41:15 GMT

Call-ID: [email protected]

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: presence

Content-Type: application/sdp

Content-Length: 209


v=0

o=CiscoSystemsCCM-SIP 2000 1 IN IP4 10.10.1.50

s=SIP Call

c=IN IP4 10.6.120.17

t=0 0

m=audio 26048 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=ptime:20

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15


Jun 21 09:41:36 AWST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

BYE sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP 10.10.1.50:5060;branch=z9hG4bK23a7451c27407

From: "Alan Rhodes" ;tag=3cf9bdf1-32ba-4e0c-9392-584cf24ea6f2-27139830

To: ;tag=F8E77F70-1C98

Date: Thu, 21 Jun 2012 01:41:15 GMT

Call-ID: [email protected]

User-Agent: Cisco-CCM6.0

Max-Forwards: 70

CSeq: 102 BYE

Content-Length: 0



Jun 21 09:41:36 AWST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.10.1.50:5060;branch=z9hG4bK23a7451c27407

From: "Alan Rhodes" ;tag=3cf9bdf1-32ba-4e0c-9392-584cf24ea6f2-27139830

To: ;tag=F8E77F70-1C98

Date: Thu, 21 Jun 2012 01:41:36 GMT

Call-ID: [email protected]

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 102 BYE

Reason: Q.850;cause=16

P-RTP-Stat: PS=534,OS=85440,PR=534,OR=85440,PL=0,JI=0,LA=0,DU=10

Content-Length: 0


But I have tried

voice class sip-profiles 1

   response ANY sip-header Diversion remove

   request ANY sip-header Diversion remove


it didn't make any difference?


Thanks.

layhlaing Thu, 06/21/2012 - 00:20
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Looks like I am having a bigger issue, customers cannot make outbound calls via SIP trunk failing over to backup PSTN lines. Can someone please share what you think?

Ayodeji Okanlawon Thu, 06/21/2012 - 01:01
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From your trace..Here is the analysis..


+++CUBE receives an Invite for 0417171234+++


INVITE sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP 10.10.1.50:5060;branch=z9hG4bK23a727214430d

Remote-Party-ID: "Alan Rhodes" ;party=calling;screen=yes;privacy=off

From: "Alan Rhodes" ;tag=3cf9bdf1-32ba-4e0c-9392-584cf24ea6f2-27139830

To:

Date: Thu, 21 Jun 2012 01:41:15 GMT

Call-ID: [email protected]


+++After cube sent a trying to your ITSP++++


++CUBE sends an Invite to a device called as.nipt.telstra.com++


Sent:

INVITE sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP 172.17.83.2:5060;branch=z9hG4bK838B91EA4

Remote-Party-ID: "Alan Rhodes" [email protected]>;party=calling;screen=yes;privacy=off

From: "Alan Rhodes" [email protected]>;tag=F8E756B0-11E7

To: [email protected]>

Date: Thu, 21 Jun 2012 01:41:15 GMT

Call-ID: [email protected]


This invite was sent 6 times and CUBE never got a response.


after the invite sent a registration was also sent and there was an error 404 not found was received for the registration request


Sent:

REGISTER sip:as.nipt.telstra.com:5060 SIP/2.0

Via: SIP/2.0/UDP 172.17.83.2:5060;branch=z9hG4bK838BA1AC3

From: [email protected]>;tag=F8E776EC-A56

To: [email protected]>

Date: Thu, 21 Jun 2012 01:41:23 GMT


Received:

SIP/2.0 404 Not found

Via: SIP/2.0/UDP 172.17.83.2:5060;branch=z9hG4bK838BA1AC3

From: [email protected]>;tag=F8E776EC-A56


Question is this what is  as.nipt.telstra.com? From the trace it looks like its your ITSP.


is 0417171234 not a device on your cucm? Why are you sending the call back to your ITSP? For calls to your internal extension shouldnt you be sending it to your CUCM?


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"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"

layhlaing Thu, 06/21/2012 - 02:02
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Thanks for your reply. Correction: Customers can make outbound calls via SIP trunk, I was having "

no voice-class sip outbound-proxy" on TPIT dial-peer.


However diversion is still an issue. as.nipt.telstra.com is our ITSP provider. 0417171234 is a mobile number that the customer want the incoming call forward to. Provider confirmed that they won't allow to send calls with caller ID not belonging to the SIP trunk. Can you pls advise how the caller ID can be set/modify on the diversion?


Cheers,

Lay

Ayodeji Okanlawon Thu, 06/21/2012 - 04:29
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We cna use sip profiles to modify that. But before we do that the trace you sent me does not show the original number called. The trace with the 047 number does not show that this is a diversion as the diverison  header is missing. Did you send this trace after you configured the headers to be removed. Can you please remove the config from the sip profile.


Please send a full trace. Let me know what the calling and called number is. For us to modify the diversion headers I also need to know what your DDI is and your internal DN range.


Finally can you send a sh run. Please put it in a text file and attach here. What type of gateway do you have configured in cucm? Is it a sip trunk or h323 gateway?

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"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"

layhlaing Sun, 06/24/2012 - 17:53
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Thanks aoknlawon for your sharing. I have done debug ccsip message again without modifying anything. Diversion happens matching dial-peer that uses backup PSTN lines. Please have a look into attached outputs if it makes sense to you.


Calling Number: 04 6778 4687 (Mobile number - third party)

Called Number: 08 9290 6198 (One of DID belonged to this SIP Trunk - range 08 9290 61XX)

Forwarded Number: 04 4857 4470 (Mobile number - third party)


I have SIP trunk in CUCM for this site. Thanks again for your sharing.


Cheers,

Lay

Ayodeji Okanlawon Mon, 06/25/2012 - 03:34
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Hi,


Add this to your voice class sip-profles 1 and then test again


voice class sip-profile 1

request INVITE sip-header From modify "[email protected]>" ""


Send me a debug ccsip messages after adding the command. Let me know if it works. I belive it should work with this. If it works we will then need to configure a profile that will match all your DN for diverted calls.


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"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"

layhlaing Tue, 06/26/2012 - 05:56
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Thanks very much for that, aokanlawon. We have recently resolved this with an additional translation rule in the dial-peer.


voice translation-rule 3

rule 1 /^\(61..\)$/ /89290\1/

rule 2 /^\(....\)$/ /892906100/


voice translation-profile INBOUND_FXS_N_DIVERSION

translate calling 3


dial-peer voice 20 voip

description Outbound - SIP to PSTN

translation-profile outgoing INBOUND_FXS_N_DIVERSION

preference 1

destination-pattern .T

session protocol sipv2

session target sip-server

voice-class codec 1

voice-class sip profiles 1

dtmf-relay rtp-nte

fax protocol none


I have also tried what you advised adding request INVITE sip-header From modify "[email protected]>" "" in voice class sip-profile  since it seems to be one of better method to resolve this. It wasn't still working, please see attached output, I wonder if you have experienced a case the provider expects something elese, other than sip-header From? Thanks again for your sharing.

Cheers, Lay

Ayodeji Okanlawon Tue, 06/26/2012 - 06:30
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Hi,


From your xaltion rule it showed that the number you send to your ITSP is 892906198 (without a 0 in front)


So to use my method change your profile config to the one below and test again


request INVITE sip-header From modify "[email protected]>" ""


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"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"

nickeoannidis Mon, 09/17/2012 - 00:00
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Hi Guys,


i had the same issue and we are also on Telstras TIPT network. The TIPT network was just seeing our 4 digit extensions for calls forwarded from internal CUCM phones


In the end the translation rule worked for me, its funny in there deployment guide they dont tell you these things


//snip



voice translation-rule 1
  rule 1 /^\(72..\)$/ /89218\1/



voice translation-profile Call_Forward_From_CUCM
  translate calling 1



dial-peer voice 30 voip (on the outgoing dial peer if just copied the relevant bit)
  description ## outgoing tipt ##
  translation-profile outgoing Call_Forward_From_CUCM
 
//snip



Cheers,

Nick

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