Calls auto disconnect within 90 seconds

FreePBX Providers
1

Hello Team,

We are facing call auto disconnect within 90 seconds with the telecom provider Airtel and intercom is working fine.

2

Please use the below link to check the logs.

https://pastebin.freepbx.org/view/fdd7a954

3

That link does not work.

4

Hello,

Please use the below one.

https://pastebin.freepbx.org/view/e63337a5

And please help us on priority as we are facing production issue from last 5 days.

And when asterisk disconnect the call after some time, that call was already disconnected from the caller end we are getting bellow Notice.

[2024-12-05 16:09:58] NOTICE[8005]: chan_sip.c:29987 check_rtp_timeout: Disconnecting call ‘SIP/Airtel_SIP-00000043’ for lack of RTP activity in 31 seconds

Also we are able to hear both side voice.

5

This is a volunteer based peer support forum; there are no service level agreements. There is a Jobs section where you can ask for paid support, but there is no guarantee that anyone will accept.

See also How To Ask Questions The Smart Way

Your logs don’t contain time stamps, but timing is part of the symptom. You have probably screen scraped them, rather than used the log file.

Both legs of the call appear to be still up at the point where the log end.

90 seconds is a relatively uncommon time; a session timer failure is about hte only cause I can think of.

6

On pastebin trace log (…e63337a5) it doesn’t appear any log about; “Disconnecting call” or RTP lack activity.

7

Hello Mcgrathr,

We also checked that call get disconnect firstly from caller end and after a long time asterisk send the bye for the call.

8

That sounds like a router has stopped forwarding the media streams. As far as Asterisk is concerned, the call is still up at the point at which the user has decided it has ended.

1 Like
9

Hello David,

As i checked with vendor directly, its working as expected and with asterisk facing this issue.

10

Definitely not going to be Asterisk. The only part of the FreePBX that might cause something like that is fail2ban, if the address is also generating security alerts. Personally I don’t think it is either.

11

Side FreePBX you can have an issue with the sip option.
You need to send it often.
For chan_sip that was defaultexpiry=120

12

Although chan_sip hasn’t been supported for some time, I don’t believe a registration expiry should have any effect on calls in progress, and even more so that it could impact the flow of RTP.

13

You will have to trace the point where the media stops flowing, by using rtp set debug on, inside the Linux firewall, tcpdump or wireshark, outside, it and the equivalent on your routers, on both their interfaces.

14

That was an example only.
That’s it.
It’s like a qualify=yes on chan_sip

15

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.