01-15-2019 02:17 AM
Hi everyone.
Our ITSP only permits SIP Traffic with Source UDP port equal 5060. By default, Cisco CUBE uses dynamic port. Please help.
01-15-2019 04:15 AM
01-15-2019 07:01 AM
Have I need to restart sip service on CUBE or reboot whole system?
01-15-2019 07:31 AM
01-16-2019 02:48 AM
Is it possible to make changes per dial-peer?
01-15-2019 05:37 AM
Are you taking about SIP signaling port? CUBE will use 5060 by default, perhaps you need to use UDP vs. TCP which sometimes needs to be changed depending on ITSP requirements, but you should not need to change signaling port. Only RTP media ports are dynamic as you need different port for each call and these can be adjusted if needed.
01-15-2019 06:59 AM
Nope, in tcdump file i can see that CUCM listens on 5060 port, but in outgoing SIP messages at Layer4/UDP it sets dynamic port.
01-15-2019 07:08 AM
Can you provide snippet of "debug ccsip messages" and your configuration?
01-16-2019 08:49 PM
08-21-2019 11:57 AM
Hello All,
I was having similar issue and I used connection reuse command on sip-ua config and capture packets using wireshark and I see src and des port as 5060.
Thank you
08-22-2019 06:54 AM
Hi All,
I have also observed this issue where ITSP provider was not accepting the outgoing call , when R Port in outgoing invite section was different from the first invite message. It SIP message while checking I found the SIP R Port was dynamic due to which ITSP was rejecting the call.
I have used below command under sip-ua which fixed the issue , I checked sip debugs again and found the all sip message were using sip r port as 5060 which was coming in first invite.
It is basically used to use the same port for the sip message communication.
connection-reuse via-port
Thanks
08-11-2021 11:25 PM
The same issue on Cisco Phone 7965G.
Can anyone help?
08-12-2021 05:02 AM
A few things, number one as your question is off topic to the OP it would be recommended to create your own post. Secondly it would be helpful if you gave a little more detailed information than just the phone model. For one are you using the phone with CM or are you using it with a 3:rd party SIP service?
08-12-2021 11:01 PM
Thank you? Ill create new post.
To answer your question, Im using Cisco PI Phone 7965G with Asterisk, but I think the problem is in phone settings, in traffic dump I see that packets from phone have high source port (49561 for ex.), which I want to change to 5060.
Is there the correct setting key to do that?
voip_control_port does not do the thing.
08-13-2021 12:15 AM
Sorry, but I have no experience what so ever with phones on 3:rd party SIP services. There are a few folks here in the community that has good knowledge in this area, hopefully one of them can chime in on this.
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