I am trying to get a SIP trunk from Cbeyond connected to our CUCM7.1 via a Cisco CUBE.
I have registred the account and on calls in and out, I can see signaling but the call gets rejected.
After some discussion with Cbeyond support the tell me that the headers are changed during the callsetup and therefore the call is in the end rejected.
Does any one have a running config with this setup??
I am working on getting a test account from Cbeyond to some more indepth debugging, but if some one had some inputs that would be much appreciated.
Attached is a debug ccsip messages.
When I try to make an outbound call I get a prerecorded message from Cbeyond that the call cannot be completed. So the sip trunk is working, but Cbeyond is telling me that they see an IP-address, where they expect a name; sipconnect....
Inbound call is working after I changed the signaling between the UCM and CUBE to H323.
So the callflow is: Phone - UCM - H323 - CUBE - SIP - Cbeyond
First and foremost, I will suggest you do sip all the way. That is instead of using H323 between CCM and CUBE, use a sip trunk. So you flow ill look like this..
A sample of the trace you sent..
INVITE sip:email@example.com:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.150:5060;branch=z9hG4bKBB5F8E
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.1.150:5060;branch=z9hG4bKBB5F8E
There is no IP address in there, its the FQDN that is been used. I will go back to the Service provider and let them know that there is no IP in the trace...They should investgate further on their part and send you a trace that shows the headers are modified.
the issue was the sip-proxy.sfo.cbeyond.net, should be sipconnect.
So now I have a working setup with the use of H323 between UCM and CUBE.
My initial setup was with SIP, but this failed and moved to H323, where I have success.
I tried to switch back to SIP and did some troubleshooting and have discovered the following line is giving the issues:
voice service voip
If I remove the line, I can make inbound calls from the provider to UCM, but outbound calls receive:
SIP/2.0 604 Does not exist anywhere
If I add the line, outbound calls succeeds but indbound calls never reaches the UCM.
SO it looks like there is a configuration issue. Can you send the trace for the succesful inbound calls and the trace for the failed outbound calls. debug ccsip messages
can you also send me a sh run please. I need to see your full config
After digging in and looking through the traces and a few research, I still cant say whats going on, but I have observed a few things.
The error 604 Does Not Exist Anywhere implies that the server has authoritative information that the user indicated in the Request-URI does not exist anywhere. In simple words, there is something you are sending in your headers that the sip server does not like.
In the two traces, here is my observation..
The failed call had the following format for the remote-party id:
the succesful call has this..
Remote-Party-ID: "Torsten Thomsen" <1293>;party=called;screen=yes;privacy=off1293>
You can see that the format for the failed call is missing the actual calling or called party name. I will suggest that you involve your SP, let them take a look at the log, see what you are sending that their server does not like.
Can you also check that on your sip trunk you have the remote-party-id checked...
can you also confirm what is the called and calling number for clarification. Is it the same number you have been dialling, have you tried another number?
the inbound call is made from 2818172060 to 5108994949 which is then translated to 1293
The outbound call is made from 5108994949 to 4085264000
The remote-party-id is set on the trunk in the UCM
The UCM Ip is 10.1.1.51 and the CUBE is 10.102.43.10 on inside IP and 10.0.1.150 on outside towards Cbeyond.
Do I gain any thing form using SIP instead of H323? else I would shift to H323....
I think you gain a lot using SIP. H323 is old and there is no development on it. Going back to it, is like looking backwards rather than moving forward. Moreso SIP to SIP on CUBE is much more desirable and recommended by cisco. Question is what do you gain by going back to H323 in this case? Will it solve your problems?
If I shift back to H323, it works.
but yes, agreed SIP would be the way, but I have previously had issues with SIP-trunks that was solved by shifting to H323 between the CUBE and UCM.
Right. I will look more at solving the sip issue while I can. I didnt know you had it working with h323. It didnt work yesterday with it. I think there ate issues with the sip profiles and headers you have configured. Why dont you want to engage your SP? The sip service provider. They may tell you right away what they want.
Sorry if my post this morning: Feb 7, 2012 2:26 AM wasn't clear.
I have no problems contacting the SP although they would proberly say that it is the missing sip-proxy statement..
they only have experience with the CME...
If it is possible, I would greatly appreciate your 2nd view on this matter.
Ok if this issue with the sip proxy is the from field, then we can use your sip profile rules to change it so that its showing sip connect. so add this rule to your sip profile
request INVITE sip-header To modify "<>" "<>">>
You can then use the sip-proxy command and try again. Send the traces so we know that the sip profile worked and what is sent to then is sipconnect
I can see I gave wrong info last time.
I have narrowed it down to issue with the Outbound-proxy command. For some reason Cbeyond require this command set to:
In the cbeyond documentation they add the command to:
voice service voip - SIP
but with this command my indbound calls does not work, because it tries to use the proxy.
I moved the command to the dialpeer and then I had successfull calls both in and out with SIP both between SP and CUBE and CUCM.
But now my register with the SP fails...
If I add the command to Voice Service Voip - SIP again, the register succeed.
As if the Sip-UA configuration only uses the global config, not the config under the Dial-peer.
Does any one know how to make SIP-UA use outbound-proxy under dialpeer, or add the command to SIP-UA config?
Thank you for your help.
I have solved the issue.
The Outbound-Proxy needs to be configured globally on voice service voip -> SIP.
Then on the dialpeers where they should not use the outbound-Proxy configure the following:
no voice-class sip outbound-proxy
this solves the issue.