[SOLVED] One Way Voice only in certain outgoing calls

Dear community

I am currently running an Asterisk version 13.1.0 on a Ubuntu 16.04 server.

We are facing one way voice only in certain outgoing calls.

I was able to reproduce the issue and it appears only in the following scenario:

  • I configured on my VoIP provider a number that transfers incoming calls to 3 numbers(simultaneously)

  • Then I called this number with my extension, the VoIP provider sends “183 session progress” ergo ring back was send via early media(negotiated codec and port for RTP)

  • One of the ring group members picks up the phone, source port from the RTP proxy of my VoIP provider for transmitting RTP packets changes(in 200 OK SIP packet)

  • Asterisk does not send the RTP packets to the in the 200 OK transmitted destination port

I retested the same scenario with my test VoIP Phone Yealink T4G VoIP without the PBX bewtween and the issue does not appear.
Respectively the VoIP phone was able to change the destination port after the call receiver has responded.

First of all, I attempted to search an option in the Asterisk config that could help out, but wasn’t able to find a matching setting.

Is it possible that this issues seems to be an Asterisk bug? If yes, could someone post a link(didn’t find a bug report).
If no, can someone give me a hint what I could try to solve this issue?

Any helpful input would be greatly appreciated.

Kind regards
Michel Rios

We’d need to see the SIP traffic (sip set debug on) and configuration for this.

Dear jcolp

Thank you for your feedback,

Is it possible to send you these details directly and not post it in this thread?

Thank you and kind regards
Michel Rios

I don’t provide direct support, and it would not allow anyone else in the community to help.

Do you have nat=set? nat=yes or nat=comedia, will cause Asterisk to wait for incoming media and use the address from that media, rather than information in the SDP, as the destination of its outgoing media.

1 Like

Hi jcolp

That makes perfect sense, sorry I didn’t consider it.

I will approximately post the necessary files tomorrow.

Thank you and kind regards
Michel

Hi David

Thank you for your input.

According to the general section in “sip.conf” file, the NAT flag is set to “nat=no”. “nat=comedia” could help out, but since there is already a RTP stream running because of the early media(183 session progress, ring back), I’m not sure if it will react like described in the documentation.

As far as I know I should be able to set a different setting in a specific SIP trunk context. I will configure the NAT flag as proposed and will give you a feedback.

Thank you and kind regards
Michel

Hi David

I’ve changed the “nat=” flag as proposed and it appears that it did the trick on my test trunk :slight_smile:
Nevertheless I need to change this flag on our productive system and will give you a definite feedback afterwards.

Kind regards
Michel Rios

Hi jcolp

The solution provided by david551 may have solved my reported issue.
In case it didn’t solved it, I will provide the necessary data.

Kind regards
Michel Rios

Hi David

The setting the NAT flag to “nat=comedia” did the trick and solved my issue.

Thank you for your helping out in this issue, I appreciate the support very much.

Have a nice day.

Kind regards
Michel Rios