I am developing a server to communicate with Asterisk 13 thru ARI. Everything is working fine so far.
I just have one issue pending, which is when I make an inbound call. I put the channels on a bridge and it seems ok (just like the outbound calls), but in this scenario there is no audio on any of the sides of the call.
To accomplish that I created an inbound trunk on pjsip.conf
I created the context [sc_blending_for_test] which calls the Stasis(app); Then I handle it on my server, where i put that call to the same bridge as my available agent is waiting.
To test it, I register on that inbound_trunk using ZOIPER and make the call. It gets to the dialplan as expected and I can check both channels on the bridge.
- Asterisk LOG: shows my agent’s channel __00014 and the inbound channel 00015 on the same bridge named 24. There is also a channel for the call recording
– Channel PJSIP/clickproxytrunk-00000014 joined ‘simple_bridge’ stasis-bridge <24>
** – Channel PJSIP/inbound_trunk-00000015 joined ‘simple_bridge’ stasis-bridge <24>**
** – Channel Recorder/ARI-00000003;2 joined ‘simple_bridge’ stasis-bridge <24>**
** – x=0, open writing: /var/sc_recordings/client_24/2017-12-19/recording_inbound_3068.ulaw format: ulaw, 0x7f136400c0f0**
- Bridge details
localhost*CLI> bridge show 24
I am not using the native_rtp technology. I had the same issue “NO AUDIO” when I implemented the outbound calls and I disabled the native_rtp to make it work.
module unload bridge_native_rtp.so
I read that audio problems can be related to the NAT configuration.
Do you know what might be wrong ?