SIP trunk with 6.0

Unanswered Question
Jan 26th, 2008

We have CM 6.0 running in a lab and we have a SIP provider. I was curious is it possible to configure CM and configure it to use the SIP provider? We have tried but cant seem to get it to work. Xlite works fine so I know the firewall is not blocking anything. Thanks.

I have this problem too.
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lffrwatson Tue, 01/29/2008 - 18:36

Could you provide detail instructions on how you got this to work? I have tried everything and CCM is not authenticating with the ISP. I get a fast busy when trying to dial out. Thanks.

kelvin.blair Tue, 01/29/2008 - 19:17

First you need to create a SIP Realm.

Realm = Sip Provider IP Address

User = user account provided by Sip Provider

Digist Credentials = password provided to you from SIP provider

Confirm Digest Credentials.

Create a SIP trunk Profile in CM.

Create a SIP Trunk Security Profile

Enable Accept Out-of-dialog Refer &

Accept Unsolicited Notication

Create your Trunk and add the above features. The SIP Realm is not a selection to add in the trunk don't worry it will still be used in the authenication process.

Let me know if you're still having problems.

lffrwatson Wed, 01/30/2008 - 12:13

I created the Realm with all the information provided from my ISP.

Not sure what you mean by SIP trunk profile in CM. I left the default SIP profile with all the defaults.

Created SIP trunk security with Enable digest checked as well as Accept-out-of-dialog refer and accept unsolicited notification.

Applied this to my sip trunk, media termination point was unchecked. SIP destination ip is pointing to my ISP. Applied sip trunk profile and sip trunk security profile.

Finally created a route pattern 9.@ classified as Offnet and applied this to the sip trunk.

When i dial 9 and then the number i get a fast busy.

Thanks again for all your help.

kelvin.blair Wed, 01/30/2008 - 12:33

In my case. I did not have to enable Digest Authenication. Try unchecking that and see if it resolve the problem. Pull the trace files. These will help in identify where the problem lies.

lffrwatson Wed, 01/30/2008 - 16:08

Lol im thinking of giving up. I tried everything and it still fast busys when dialing an external number. Im fixing to wireshark the line and see if sip is authenticating etc.

nelson-serrao Fri, 02/15/2008 - 15:52

Hi. I tried the exact steps listed in the post and have the same problems. I get fast/busy tone as well.

Just to be sure, is the process detailed above going to eliminate the need for the border element (CUBE) incase it works??

kelvin.blair Fri, 02/15/2008 - 18:55

What do your traces say in call manager? Make a call and post the trace.

lffrwatson Tue, 03/18/2008 - 11:51

Whats the best way to set up a trace? What im seeing is just graphs and does not show much. I also am able to use digit analyzer but everything appears right in it.


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