No audio between two SIP softphones on 3G calling via Asterisk

Hello,

We are new to Asterisk and have issues making successful calls between SIP softphones connected via Asterisk. We would like to be able to communicate using cell phones connected on different 3G networks with different operators. We installed SIP phone software (such as linphone and others) and tried to make calls using many different client-side configuration settings (STUN/ICE enabled/disabled, etc.). The phone rings but there is no audio when call is answered. We are aware that it is most probably a NAT issue because everything works fine internally, inside LAN.

Asterisk is installed on a public IP (no LAN) without firewall with “nat=yes” (since clients are behind NAT) set in sip.conf

Any help would be greatly appreciated.

Pat

You would need to provide the console output with SIP logging (sip set debug on for chan_sip or pjsip set logger on for chan_pjsip) so the negotiation can be seen. As well for media “rtp set debug on” would be useful. Finally the configuration in use for each side.

Thank you for your message.

I am not sure which steps and commands I should execute to get the console output, in a file I assume. Should I redirect the output of the command? AFAIK, I should access the console using command asterisk -r (maybe with a verbose level?) and then type “sip set debug on” and “rtp set debug on” before making a call and log everything. Is this correct? Should I copy/paste the log or what? I am also assuming we must be using chan_sip because this is the default, correct?

Also which config files do you need? sip.conf, extensions.conf, users.conf? All or maybe anything else? I can also provide you with the softphone settings: network, audio codecs, etc.

I appreciate your being patient as I am very new here.

Thanks,
Pat

The wiki has details[1] on the best way to collect information. The sip.conf file would be the one to provide, minus passwords.

[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

1 Like

I changed the STUN server and all worked fine. Thanks a lot.

Pat