11-10-2015 12:18 AM - edited 03-17-2019 04:51 AM
Hello,
I have a issue with sip trunk outgoing calls. The incoming calls work well.
I my network the path to join provider sip server is like this: phone--firewall--CUCM--CME srst router--firewall--sip server.
phone ip address: 10.19.51.X
CUCM ip address: 158.113.41.84
CME router ip address: 158.113.41.80
SIP server ip address: 172.20.254.252
You can see in attached the running configuration of the CME router and the debug ccsip messages output.
Please share your experience.
Regards,
Aristide
11-10-2015 12:48 AM
Your SIP provider might not be supporting SIP delayed offer. Enable SIP early offer in CUBE and then share the results.
voice service voip
early-offer forced
- Vivek
11-10-2015 02:15 AM
11-10-2015 02:28 AM
Well so other issue could be which I see frequently on forum, you're sending simply extension number 1980 in From field of INVITE message. That is what you want (??) or would like to send agreed PSTN number in From field as per instructions from service provider.
- Vivek
Please rate useful posts.
11-10-2015 02:49 AM
11-10-2015 02:57 AM
Add G711 codec also under the respective outbound dial-peer. If still having the same issue and as suggested by Carlo, please check whether calls are even going out from your network or not (any firewall issue...).
- Vivek
Please rate useful posts.
11-10-2015 02:59 AM
Hi Vivek,
No problem, I will do it, and I will back to you.
Regards,
Aristide
11-10-2015 03:15 AM
Hi Vivek,
Firewall have a diagnostic tool.
Does I have to check the traffic between the ip phone: ip address 10.19.51.75 and sip server: 172.20.254.252
or traffic between the CCME router(ip address 158.113.41.80) which establish trunk link with the sip server: 172.20.254.252
Regards,
Aristide
11-10-2015 03:31 AM
First we need to check the signaling part hence check the traffic between CCME and sip server.
- Vivek
Please rate useful posts.
11-10-2015 04:58 AM
Hi Vivek,
We don't see anything in a firewall, it just put connected. Does I have to check now with the sip provider?
Regards,
Aristide
11-10-2015 05:01 AM
Hi Aristide,
Yes i think so that is the only option left unless someone else on forum would have any other feedback.
- Vivek
Please rate useful posts.
11-10-2015 10:27 AM
11-10-2015 10:34 AM
Yes the sip session is natted in the firewall.
Regards
11-10-2015 10:43 AM
Ok...
so I need to look at your actual configuration after suggested changes.
Before posting the config please add under sip-ua configuration what follows:
nat symmetric role passive
nat symmetric check-media-src
Thanks
Regards
Carlo
11-10-2015 11:31 AM
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