ā11-10-2015 12:18 AM - edited ā03-17-2019 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
ā11-10-2015 11:42 AM
Hi Aristide.
Can you try to let your Cube to contact the provider with its real public IP and send a debug ccsip message output?
Let us know
Regards
Carlo
ā11-10-2015 11:48 AM
Hi Carlo,
do you want me to bypass my CUCM server?
Regards,
Aristide
ā11-10-2015 11:52 AM
No.
Sip trunk is between your Voice Gateway and your provider.
Now you are natting your provider's IP Address.
I'm asking you to point to your provider with its real Public IP Address.
Thanks
Carlo
ā11-10-2015 11:57 AM
Ok Carlo, I understand now, but now I want to make a direct sip trunk with a CUCM server to make a test, because I have three location which will use SIP trunk.
Regards,
Aristide
ā11-10-2015 12:29 PM
Can you please describe your call flow?
Are you putting CUCM behind nat?
Carlo
ā11-10-2015 01:05 PM
Carlo,
finally I will use the same configuration and the call flow will the same as I describe:
phone--firewall-CUCM--CME router--firewall--Sip server.
I have new information. I can ping sip server with a CUCM but the sip server can't ping the CUCM.
Do you think it can be the issue?
Regards,
Aristide
ā11-10-2015 01:21 PM
Hi Aristide,
this kind of layout could lead to many issues if firewall are not correctly configured.
Because you are communicating through Voice gateway to Sip server, it should reach your VG on the source port that it uses to register to Sip proxy.
I suggest you to investigate further on your network reachability and also to avoid to put a firewall between CUCM and ip phones
HTH
Regards
Carlo
ā11-10-2015 12:48 AM
Hi Aristide.
Please enable sip user agent on your gateway adding this:
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 2
After that please post another debug ccsip message
Thanks
Regards
Carlo
ā11-10-2015 02:10 AM
ā11-10-2015 02:49 AM
Hi Aristide,
did you check your firewall if SIP inspection is enabled or there are some rules blocking SIP traffic?
Please let us know
Regards
Carlo
ā11-10-2015 02:55 AM
Hi Carlo,
I will check and I will back to you.
Regards,
Aristide
ā11-10-2015 03:16 AM
Hi Carlo,
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
ā05-01-2024 12:21 PM
Hi Aristides,
Were you able to find a solution? I'm curious to know!
BR,
NURLAN
ā05-02-2024 07:31 AM
As this is a 9-year-old-post, I encourage you to create your own new post with a description of your problem, and your config and debug and we'll be happy to help.
Maren
ā05-03-2024 02:48 AM
Thanks a lot, Maren!
I'll definitely get to it when I have some free time. Your help is much appreciated!"Thank you, Maren.
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