Retrans_pkt: Hanging up call XX77yy - no reply to our critical packet

When i make a test call ringing well but when i receive call after 20 second call automatically hangup. I am not find out the problem. Can any one help me.

Asterisk version: 16
Server: Google cloud
Port: All UDP and TCP port open
OS: Ubuntu 18
Change port 5060 to 5160.

Here is the sip.conf file
[general]
context=public
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
transport=udp
srvlookup=yes
qualify=yes
port=5160
bindport=5160
[authentication]
basic-options
dtmfmode=rfc2833
context=from-office
type=friend
natted-phone
directmedia=no
host=dynamic
public-phone
directmedia=yes
my-codecs
disallow=all
allow=ilbc
allow=g729
allow=gsm
allow=g723
allow=ulaw
ulaw-phone
disallow=all
allow=ulaw

[1001]
type=friend
context=phones
host=dynamic
allow=ulaw,alaw
secret=12345678

[1002]
type=friend
context=phones
host=dynamic
allow=ulaw,alaw
secret=12345678

Here is the error:
Using SIP RTP CoS mark 5
> 0x7f7ec003eda0 – Strict RTP learning after remote address set to: 103.94.217.14:4000
– Executing [1002@phones:1] NoOp(“SIP/1001-0000003f”, “First Line”) in new stack
– Executing [1002@phones:2] NoOp(“SIP/1001-0000003f”, “Second Line”) in new stack
– Executing [1002@phones:3] Dial(“SIP/1001-0000003f”, “SIP/1002”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/1002
– SIP/1002-00000040 is ringing
– SIP/1002-00000040 is ringing
> 0x7f7ed4009680 – Strict RTP learning after remote address set to: 192.168.0.2:4000
– SIP/1002-00000040 answered SIP/1001-0000003f
– Channel SIP/1002-00000040 joined ‘simple_bridge’ basic-bridge <305bf707-4b95-4a39-a7f6-f465b2bbaf19>
– Channel SIP/1001-0000003f joined ‘simple_bridge’ basic-bridge <305bf707-4b95-4a39-a7f6-f465b2bbaf19>
> Bridge 305bf707-4b95-4a39-a7f6-f465b2bbaf19: switching from simple_bridge technology to native_rtp
> Remotely bridged ‘SIP/1001-0000003f’ and ‘SIP/1002-00000040’ - media will flow directly between them
> 0x7f7ed4009680 – Strict RTP learning after remote address set to: 192.168.0.2:4000
[Sep 19 00:44:21] WARNING[16758]: chan_sip.c:4142 retrans_pkt: Retransmission timeout reached on transmission f4dwzXNM-2F1wgHpr7-LMpBBjMqCAvWG for seqno 28597 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 15551ms with no response
[Sep 19 00:44:21] WARNING[16758]: chan_sip.c:4166 retrans_pkt: Hanging up call f4dwzXNM-2F1wgHpr7-LMpBBjMqCAvWG - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
– Channel SIP/1001-0000003f left ‘native_rtp’ basic-bridge <305bf707-4b95-4a39-a7f6-f465b2bbaf19>
– Channel SIP/1002-00000040 left ‘native_rtp’ basic-bridge <305bf707-4b95-4a39-a7f6-f465b2bbaf19>
== Spawn extension (phones, 1002, 3) exited non-zero on ‘SIP/1001-0000003f’
> 0x7f7ec003eda0 – Strict RTP learning after remote address set to: 103.94.217.14:4000
> 0x7f7ed4009680 – Strict RTP learning after remote address set to: 192.168.0.2:4000

Packet timed out after 15551ms with no response

  • you are not receiving the ACK on your SIP/OK

I had such problem - and ACK was sent back to Contact: header , where port was not set, and default 5060 was used, but was rejected, since no any listeners on that port.
It was a FreePBX, where pjsip was on 5060, and SIP was on 5160.

If you set back port to 5060, it will work (try just to ensure) .

Try to put port in fromdomain option as well

Lost ACKs, on incoming calls, will also happen if Asterisk is behind NAT but hasn’t been told that. I don’t know about the former, but there is nothing in your configuration that would make Asterisk think that it was behiind NAT (no extern*** lines).

Although not relevant here, I would advise using type=peer for everything, unless you have two peers with the same IP address.

problem solve. Thanks @david551 , @a4business

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