05-08-2019 12:55 AM
Hi guys,
I am currently trying to set up a SIP trunk with T-Mobile in Czech Republic but outbound calls are not working.
At given time all numbers of the traditional PBX will be ported to the SIP trunk but ITSP said that we can already make outgoing calls for testing. They said that they mapped one number which is currently unknown to us. When we try to make an outbound call I can see our INVITE (with sdp) to ITSP, a 100 TRYING comes back, followed by 183 with sdp. Then RTP stream from ITSP to our CUBE starts. Then, a second later - I assume here is the problem - our CUBE sends an INVITE again followed by a 183. This repeats sometimes while RTP is still streamed but call is never established.. After almost 20 seconds we get a few 504 server timeouts.
The 183 contains require 100rel. I tried to set 100rel required on voice service voip under sip and at dial-peer, didn't help at all.
On SIP Profile of trunk between CUCM and CUBE SIP Rel1xx Options is set to Send PRACK if 1xx contains sdp and early offer support is set to best effort.
CUBE runs on a 4331 with 16.6.6 (first we tested 16.6.5 - same issue).
Topology looks like: Phone > CUCM > SIP Trunk > CUBE > ITSP SBC
I attached a zipped packet capture of the WAN interface pointing to ITSP.
I could post partially the config if needed.
Any idea what I could try next?
Thank you!
05-08-2019 05:09 AM
Hi,
Yes please post the config and the output of a debug ccsip messages while trying an outgoing call.
Thanks
Regards
Carlo
05-08-2019 06:58 AM
05-08-2019 08:04 AM
Hi,
Because on his side, the provider is requesting to send prack , please add these lines to your sip config
Voice service voip
sip
rel1xx supported 100rel
After you added that, please send again a debug ccsip output.
HTH
Regards
Carlo
05-09-2019 02:07 AM
Hi Carlo,
I tried this before as I wrote earlier, unfortunately no change. No PRACK was sent. Btw. it doesn't even show this command after execution when I execute show run. I assume it is already set by default.
Thanks
05-09-2019 03:06 AM
Hi,
Can you please add early-offer forced under voice service voip --> SIP and send again a debug ccsip messages while calling out?
Thanks
Regards
Carlo
05-09-2019 04:27 AM
05-09-2019 05:35 AM
Hi,
Add these lines on your VG end let see if it makes change
voice service voip
ip address trusted list
ipv4 62.141.7.0 255.255.255.0
If you can, please send the sip-ua configuration section.
Thanks
Regards
Carlo
05-10-2019 02:17 AM
Hi Carlo,
I changed to the whole subnet for ITSP SBC at IP trusted list but still same issue.
We didn't configure the sip-ua as the provider didn't request any authentication/registration to be set.
They are allowing only SIP traffic from the private IP they gave us, that seem to be their authentication.
Thank you for costantly bringing new ideas!
05-10-2019 09:35 AM
Hi,
Let's give a try to this.
under voice service voip
sip
rel1xx disable
Please let me know
Regards
Carlo
05-14-2019 12:13 AM
05-14-2019 10:21 AM
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide