Call drops after 25-30 secs when transfered to another extension

Hi,

One of our FreePBX Sever is having a weird issue.

We can receive / make calls, no problems there. But whenever an already answered inbound call is transferred to another extension on the same pbx the call drops after 25-30 secs. This does not happen when the original party which have answered the call is in a 3-way conference call with another extension.

I have tried changing the RTP timers (rtptimeout 300, rtpkeepalive 30) , checked all my NAT settings, SIP & RTP ports.

FreePBX 12.0.76.2
Asterisk (Ver. 11.16.0)
Grandstream IP Phones (have reset them also).
All phones are on the same network.
All Inbound calls destinations are ring groups (hunt).

Can anyone suggest where should we look further.

Thanks.

What do the logs say?

The issue is resolved. Thanks for sending us in the right direction.

Here is an excerpt from the log:

 WARNING[1979] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32001ms with no response

WARNING[1979] chan_sip.c: Hanging up call [email protected] - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).`

The time stamp on the above log was exactly 25 secs after the call transfer.

This looked like a NAT or firewall issue.
(We don’t have any ip from the network 192.168.201.0)

We ran a wireshark packet capture, it shows that RTP packets are flowing only from the Freepbx server to the IP Phone and no reply from the Freepbx server to the SIP provider.

So we checked our trunks and it turned out that our SIP provider has a different ip for the trunks we have. After changing that we see the RTP packets from the Freepbx server to the SIP provider and transferred calls do not drop.

Correct me if I’am wrong in the diagnosis.

I’m going to guess that your Asterisk SIP Settings were not correctly configured. Am I right?

No. The PEER details in the Incoming Trunk settings namely host and fromdomain options were not exactly what our SIP Provider expected from the type of trunks we were using. They have different IPs for the type of trunk you purchase.