Call from SIP Trunk Disconnects in about 30s

[zscore]

Hello, I’m aware that this is a frequently asked question and happens for Asterisk instance behind a NAT. I am also aware of the page Configuring res_pjsip to work through NAT but I wasn’t able to figure out the correct settings.

Issue, any calls to and from the SIP are being disconnect in about 30 seconds. I believe this is due to call setup (Invite, 200OK, Ack) not happening correctly.

PJSIP Settings,

[airtel]
type = wizard
sends_auth = yes
sends_registration = yes
remote_hosts = ims.airtel.in
outbound_auth/username = +918046******@ims.airtel.in
outbound_auth/password = hunter2
endpoint/context = from-external
endpoint/allow = !all,ulaw
endpoint/from_user = +918046******
endpoint/from_domain = ims.airtel.in
aor/qualify_frequency

Here’s the SIP logs from Asterisk,
https://pastebin.freepbx.org/view/ae9220a9

In the logs above, I guess there is something wrong with the ACK request sent by the SIP. Can someone please help me in debugging this?

[jcolp]

PJSIP does not support the use of tel URIs. Packets of such will be ignored, or responded to with a 416. The ACK contains a tel URI, and thus is ignored.

[zscore]

Thanks for the insight! I read other topics that discuss this here and here along with your posts also discussing the blockers.

I did find PJSIP documentation on tel: URI at a few places [0] [1] (the last ticket is 12 years ago )

At this point, I guess my best bet is to try out sip.conf instead. Please let me know if you have any other ideas, I do not think I will be able to convince my ISP to not use tel: URI

[zscore]

Is there a way I can disable call setup ACKs?

[jcolp]

No, because that’s a fundamental part of the SIP protocol.

[zscore]

Quick update, I switched to chan_sip.so a few days ago and haven’t faced issues since then even though the ACK packet is still the same and contains a tel URI.

[david551]

The difference is that code for tel: URIs has been enable in chan_sip, but it has not been tested thoroughly, so may present a denial of service risk. In chan_pjsip, the code has been more carefully security vetted, and tel: will not be incorporated until someone provides an extensive test suite to prove that edge cases don’t cause a crash. It’s therefore not officially supported in either version.

[jcolp]

Even for chan_sip it’s iffy, it can spit out warning and errors at you and may or may not work.

[voip.com.vn]

You should check session-timer parameter.