cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
2624
Views
0
Helpful
8
Replies
galinvelkov
Beginner

CUBE SIP TRUNK

Hi,

We have a problem with a SIP trunk realization to SP Proxy. We have a cisco 2901, CUCME 8.6, mode CUBE. The problem is with outgoing call to SP - the SDP message body contains the IP address of CUCME not WAN side IP address of CUBE.

Can you help us how to resolve this problem - when we make outgoing calls the SDP message must contain WAN IP address of CUBE!!!

3 ACCEPTED SOLUTIONS

Accepted Solutions

For traces, use  "debug ccsip message" from router.. No need to use attachments.

View solution in original post

casanavep
Participant

You can also bind signaling and media at the dial-peer level or globally. I agree with the above, voice translations is the way to do that manipulation of digits. Usually normalization standards are per ITSP requirements, so each ITSP dial-peer can have its own associated outbound translations for the appropriate calling and called party presentation. Let's say you were in Washington DC, USA and used +1-703-555-897X numbers. Internally your extensions were just the 897X remainder and the First ITSP wanted you to present your calling party without the e.164 +1 country code, but wanted the NANP area code followed by the remaining seven digits. You would build "rules", "translation-profiles" that call on those rules and and then apply the profiles under the dial-peer.

voice translation-rule 1

rule 1 /^897\(.\)$/ /703555897\1/

voice translation-profile Normalization-To-ITSP-A

translate calling 1

dial-peer voice XXXXX voip

translation-profile outgoing Normalization-To-ITSP-A

voice−class sip bind control source−interface GigabitEthernet0/0

voice−class sip bind media source−interface GigabitEthernet0/0

Caveats:

1. SIP Binding current IOS releases has issues when used in conjunction with HSRP. This does not impact the CUBE-on-a-Stick deployments:

http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?method=fetchBugDetails&bugId=CSCua28559

2. If binding both media and signaling at the dial-peer level, the dial-peer must not be in use for an active call.

3. SIP binding at the dial-peer level not available until 15.1(2)

View solution in original post

You don't necessarily need separate dial-peers for voice and fax. Here are a few things that I would verify:

- Ensure the SPA is nailed up to g711uLaw under its voice register pool configuration

- Ensure VAD is disabled under the SPA's voice register pool configuration

- nail up DTMF as RTP-NTE under the SPA's voice register pool configuration

- Web into your SPA and under Connections/Devices, Set Advanced settings/Fax Mode to G.711 Passthrough

- let's try nailing up fax capabilities at the dial-peer level ( I have had issues without doing it there). If you want to define it under the voice service voip area, make sure you define "fax protocol system" under the dial-peers to call on the global setting. But for a test, let's try the following to add the following to dial-peer voice 3:

     - fax rate 14400

     - fax protocol pass-through g711ulaw

     - fax-relay ecm disable

I suggest trying g711uLaw instead of T.38 simply because you have a SPA vs an FSX. G.711uLaw has the highest rate of comparability with thrid-party devices. Yes, I count a SPA as third-party.... If ever possible, drop the SPA for a HWIC for POTS FAX on small deployments. FXS interoperability provides a much more reliable outcome and more rapid integration. That is my personal opinion. Let me know how this works out. If it is still failing, do you mind posting the rest of your voice service voip, voice class codec, and sip-profile configurations.

View solution in original post

8 REPLIES 8
paolo bevilacqua
Hall of Fame Master

Likely configuration error, impossible to say more without seeing config and traces.

For traces, use  "debug ccsip message" from router.. No need to use attachments.

View solution in original post

paolo bevilacqua
Hall of Fame Master

Yoi don;t need to use SIP profile to manipulate number, just voice translation-profile and rule.

Thank you for the nice rating and good luck!

Hi Paolo,

Thanks for your reply!

I tried it last week but I did not bring success and therefore did sip normalization. I know that the translation rules and profiles are right practice.

Between with the sip profile normalization outgoing calls work but today I had problems twice - outgoing call stoped work correct -  and did not understand why!

I will do outgoing translation rules and translation profile applied to outgoing dial-peer and will post the result.

Best regards!

casanavep
Participant

You can also bind signaling and media at the dial-peer level or globally. I agree with the above, voice translations is the way to do that manipulation of digits. Usually normalization standards are per ITSP requirements, so each ITSP dial-peer can have its own associated outbound translations for the appropriate calling and called party presentation. Let's say you were in Washington DC, USA and used +1-703-555-897X numbers. Internally your extensions were just the 897X remainder and the First ITSP wanted you to present your calling party without the e.164 +1 country code, but wanted the NANP area code followed by the remaining seven digits. You would build "rules", "translation-profiles" that call on those rules and and then apply the profiles under the dial-peer.

voice translation-rule 1

rule 1 /^897\(.\)$/ /703555897\1/

voice translation-profile Normalization-To-ITSP-A

translate calling 1

dial-peer voice XXXXX voip

translation-profile outgoing Normalization-To-ITSP-A

voice−class sip bind control source−interface GigabitEthernet0/0

voice−class sip bind media source−interface GigabitEthernet0/0

Caveats:

1. SIP Binding current IOS releases has issues when used in conjunction with HSRP. This does not impact the CUBE-on-a-Stick deployments:

http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?method=fetchBugDetails&bugId=CSCua28559

2. If binding both media and signaling at the dial-peer level, the dial-peer must not be in use for an active call.

3. SIP binding at the dial-peer level not available until 15.1(2)

View solution in original post

I have a question:

Should I use another dial-peer for FAX to make and receive fax messages?

Now for outgoing calls I use dial-peer 3!

You don't necessarily need separate dial-peers for voice and fax. Here are a few things that I would verify:

- Ensure the SPA is nailed up to g711uLaw under its voice register pool configuration

- Ensure VAD is disabled under the SPA's voice register pool configuration

- nail up DTMF as RTP-NTE under the SPA's voice register pool configuration

- Web into your SPA and under Connections/Devices, Set Advanced settings/Fax Mode to G.711 Passthrough

- let's try nailing up fax capabilities at the dial-peer level ( I have had issues without doing it there). If you want to define it under the voice service voip area, make sure you define "fax protocol system" under the dial-peers to call on the global setting. But for a test, let's try the following to add the following to dial-peer voice 3:

     - fax rate 14400

     - fax protocol pass-through g711ulaw

     - fax-relay ecm disable

I suggest trying g711uLaw instead of T.38 simply because you have a SPA vs an FSX. G.711uLaw has the highest rate of comparability with thrid-party devices. Yes, I count a SPA as third-party.... If ever possible, drop the SPA for a HWIC for POTS FAX on small deployments. FXS interoperability provides a much more reliable outcome and more rapid integration. That is my personal opinion. Let me know how this works out. If it is still failing, do you mind posting the rest of your voice service voip, voice class codec, and sip-profile configurations.

View solution in original post

Hi casanavep,

Thanks for your replay about FAX config! Your answer is comprehensive and deep in details.

Thanks again for the help, time and detailed answers!

Best regards