Internal calls dropping

Hi,
I have a problem with the internal calls. I can call to another extension but as soon as somebody answering on that side call dropping.
External calls are working well.
The only one error which I can see is

[2016-09-21 12:09:26] WARNING[25670][C-000093cc] chan_iax2.c: Resyncing the jb. last_delay 0, this delay -478823262, threshold 1000, new offset 478823262
[2016-09-21 12:09:26] WARNING[25712][C-000093cc] chan_iax2.c: Resyncing the jb. last_delay 0, this delay -4907, threshold 1000, new offset 4907

Generally speaking, Jitter Buffer problems are indications of some local network issue.

Try disabling the Jitter Buffer stuff and see if the problem gets better or worse.

Nope. That didn’t help.

It looks less of system resource and result in the jitter buffer issue.

Any ideas how can I fix it or what should I check?

I made a few test:
When I’m calling from the extension to the extension then everything fine.
When I’m calling from the extension to our main number and somebody answer then there is no sound.

[2016-09-23 15:30:05] WARNING[25817][C-00000426] app_dial.c: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
[2016-09-23 15:30:05] WARNING[25817][C-00000426] app_dial.c: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)

Ports 5060 and 5061 and RTP 10000-20000 are opend.
RTP debug showing

Sent RTP packet to 50.123.456.789:16766 (type 00, seq 029269, ts 001280, len 000160)
Sent RTP packet to 50.123.456.789:16766 (type 00, seq 029270, ts 001440, len 000160)
Got RTP packet from 192.123.456.789:2236 (type 00, seq 018092, ts 3325527802, len 000160)
Sent RTP packet to 50.123.456.789:16702 (type 00, seq 004985, ts 3325527800, len 000160)

Got RTP packet from 192.123.456.789:2236 (type 00, seq 018466, ts 3325587642, len 000160)
Sent RTP packet to 50.123.456.789:16702 (type 00, seq 005359, ts 3325587640, len 000160)
Got RTP packet from 192.123.456.789:2236 (type 00, seq 018467, ts 3325587802, len 000160)
Sent RTP packet to 50.123.456.789:16702 (type 00, seq 005360, ts 3325587800, len 000160)
Got RTP packet from 192.123.456.789:2236 (type 00, seq 018468, ts 3325587962, len 000160)
Sent RTP packet to 50.123.456.789:16702 (type 00, seq 005361, ts 3325587960, len 000160)

Make sure the subnet is set correctly in sip settings I believe is where it is at

OK - this I might not make a lot of sense or seem like it might apply, but hang in there. There might be something that helps.

I was playing around with our conference number last night (testing out the new “Welcome to Conferencing” prompt). If I call if from outside the network, it works fine. If I call the conference “extension”, it works fine. If I call the inbound DID from an extension on the same server, I get no sound. This sounds like a description of your situation. The difference is that I don’t get the “unable to create channel” message.

If I let the calls go out to my “normal” service provider and route the calls back to my main number, the audio will fail. If I let the traffic stay local, or come in from any other source, the calls will work fine,

What I had to do is set up a second trunk (which I already did for cell phone forwarding to allow for the foreign Caller ID) that calls back to my numbers. This way, the outbound traffic comes from an external source, instead of coming from the same service my calls are going out on.

So, I set up the DID for the conference line in my “cell phone redirect” outgoing so that calls to that number are routed out through the second service. After that, every call from every device worked flawlessly.

Now, I’m certain there’s a simpler explanation for my problem, but this is what works for me. I’ve changed my system so that anytime I call one of the numbers that terminates in my system (from an extension connected to the server), I route that call out onto the “cell phone redirect” service.

Did you make 2 inbound or outbound trunks? Because I have 2 outbound trunks.

One inbound, two outbound.

My inbound is actually bidirectional, in that most of my calls go out through my normal inbound trunk. The only things that go out on the Cell Trunk are things where I lose two-way audio or need to use a foreign Caller ID.