Ask the Expert:Cisco Unified Border Element for PSTN SIP Trunks

[ciscomoderator]

With Randy Wu

Welcome to the Cisco Support Community Ask the Expert conversation. This is an opportunity to learn from Cisco expert Randy Wu  best practices on how to configure and troubleshoot Cisco UBE for the public switched telephone network Session Initiation Protocol trunks.

Randy Wu is a senior customer support engineer in the Multiservice Voice team at Cisco in Sydney. He has vast experience and knowledge configuring, troubleshooting, and designing Cisco UBE, gateways, and gatekeepers, working with H323, MGCP, and SIP protocols. He joined Cisco as a systems engineer in 1999. He holds CCIE certification (#8550) in Service Provider, Routing, and Switching and Voice.

Remember to use the rating system to let Randy know if you have received an adequate response. 

Randy might not be able to answer each question due to the volume expected during this event. Remember that you can continue the conversation on the  Collaboration, Voice and Video sub-community discussion forum shortly after the event. This event lasts through June 29, 2012. Visit this forum often to view responses to your questions and the questions of other community members.

[Ayodeji Okanlawon]

Randy,

I want to know if its possible to make sip profiles work lke xlation rules. This is what I mean..

If I have an DN sending a divert to cube like this..

request INVITE sip-header Diversion modify "<56345408>;privacy=off;reason=follow-me;screen=yes"

Can I modify the divert header such that i the DN is split like 5634*

So I can have a sip profile like this..

request INVITE sip-header Diversion modify "<5634>;privacy=off;reason=follow-me;screen=yes"

"<44121424>;privacy=off;reason=follow-me;screen=yes"

This is to enable me present different DDI based on the last 4 digit extension. Can this be done?

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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"

[yuanwu]

Hi, Aokanlawon

Thanks for your question.

From your description,  you want to replace the user infor in the diversion header from 56345408 to 441214245408 ?

Rgds/Randy

[Ayodeji Okanlawon]

Yes, but I also want it to be used for 56344000, 56345000 without having to configure a new diversion header for each  number such that  56344000 is replaced with 441214245000 etc.

Can I use a single sip profile to achieve this?

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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"

[yuanwu]

Hi, Aokanlawon

Please try this sip-profile,

request INVITE sip-header Diversion modify "<5634>" "<44121424>"

Rgds/Randy

[Ayodeji Okanlawon]

Thanks Randy! This is what I was looking for

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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"

[INHO CHOI]

Hi Randy,

I hope you are very well and thanks for all your answers here. That's really helpful. Thanks again.

I am in the middle of testing BE3K with 881 CUBE router. It's working fine so far except for one thing: MOH.

Internally MOH is working fine but if I make a call to a mobile phone and put on hold then MOH is played by our ITSP.

After this happening I can't get it off hold by pressing RESUME button. My phone looks back to off hold but the mobile phone user is still listening ITSP's MOH. I have to just hang up to finish the MOH playing on the mobile phone.

Does this happen because our CUBE is not sending right information to ITSP? I am very confusing with this issue. :-)

Please help.

Thanks.

[Ayodeji Okanlawon]

Randy,

I have a burning question that I've needed help on for years now regarding properly binding media and signaling for SIP in CUBE.  Growing up in the Cisco world I've always been told and followed the best practice of binding media and signalling for H323, MGCP, CME and SRST to a loopback on the router and some features simply won't work unless you bind to a loopback.  This has worked great until I came across utilizing CUBE for SIP integration with CUCM and a PSTN provider such as AT&T, Verizon or Global Crossing.  What's my issue?  I can no longer bind my traffic to that loopback address and have a succesful CUBE integration with an external entity or with CUCM.  You ask how that's an issue?  When I bind to that loopback address which has an internal non public routable IP address CUBE sends out of all interfaces that Loopback interface as the source address so when my SIP traffic gets sent to the PSTN provider they have no way of getting back to that router but obviously it works for talking to CUCM.

A cisco staff responded to this by saying "I don't recommend you bind SIP on a SIPto-
SIP CUBE, since it removes the listener from the external interface. 90% of installs
shouldn't need a SIP bind on SIP to SIP since we'll source from the interfaces closest to the
destination of the SIP packet"

Can you please clarify this. In a scenario where I have an Internal IP and a public IP to talking to the provider. How should I go about binding sip signalling? It looks obvious that I cant bind to my local interface.

Secondly what does the phrase mean "we'll source from the interfaces closest to the destination of the SIP packet"  Does that suggest the Public interface?

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[Gordon Ross]

aokanlawon - just bind a public IP address to the loopback interface. That's what I do.

GTG

[yuanwu]

Hi, Aokanlawon

Thanks for your questions.

1.  the CUBE is a natural De-Marcation point, by using address hiding, you can have internal ip address facing your internal network, and public ip address facing Provider

2. the phrase means if you don't use bind command, it will use the source address for signaling by your routing table.

Rgds/Randy

[Ayodeji Okanlawon]

Thanks Randy.

Yes I understand CUBE to be a demarcation point, however  in this scenario what will be the best practise? Not to use the bind command? Or where do we use the bind command?

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[yuanwu]

Hi, Aokanlawon

Thanks for your feedback.

1.  In general, the bind interface needs to be routable to the destination.

2. Usually we configure the bind command at dial-peer level pointing to internal or external, in which it will be quite logical for the media and signaling interface used for the traffic.

Rgds/Randy

[yuanwu]

Hi, InHo Choi

Thanks for your question.

From your description, I think when you put your ip phone under BE3k on hold, then you can't resume it,right?

Usually the MOH will be played by BE3k when you put the mobile phone on hold from your ip phone while in your setup the ITSP will play the MOH.

I need to check the "debug ccsip message" from CUBE and "show running" from 881 to see what happens to the RE-INVITE messages when you resume the call.

Rgds/Randy

[INHO CHOI]

Hi Yuan,

I have just uploaded the files you asked.

Thanks.

[INHO CHOI]

Hello Randy,

I have just fixed this issue by putting "midcall-signalling block" under the VOICE SERVICE VOIP > SIP.

Could you explain if that's the right thing for my issue?

Cheers,

[yuanwu]

Hi, INHO

Thanks for your update.

This command will block all the mid-call signaling,  normally it will used with another command "midcall-signaling passthru media-change" to support fax, video escalation from audio.

If you only configure this command, which is to block all mid-call signaling , then fax, video escalation calls will be failed.  You also need to verify other call flow the CUBE will have, which will be used in some specific environment.

For this ITSP environment, it can be a workaround for this issue, but it is still a bit rare for this kind of Sip implemenation.

Rgds/Randy