Call hangup right after dial in asterisk

StackOverflow https://stackoverflow.com/questions/16186607

  •  11-04-2022
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Question

I am using asterisk 11 and my call hangup right after dial command and shows bellow error

Retransmission timeout reached on transmission

Mydial command is

AGI Script Executing Application: (DIAL) Options: (SIP/112233@202.174.211.30:8060)

Call works fine on default port (5060) in this case not work on given port 8060.

Complete Debug:

Everyone is busy/congested at this time (1:0/0/1)
[Apr 23 17:27:42] WARNING[9213]: chan_sip.c:4169 retrans_pkt: Retransmission timeout reached on transmission 03993f2a3d90ec4f7260711836681fd0@88.208.208.34:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
[Apr 23 17:27:42] WARNING[9213]: chan_sip.c:4198 retrans_pkt: Hanging up call 03993f2a3d90ec4f7260711836681fd0@88.208.208.34:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  == Everyone is busy/congested at this time (1:0/0/1)
[Apr 23 17:27:43] WARNING[9213]: chan_sip.c:4169 retrans_pkt: Retransmission timeout reached on transmission 2b2effa966c193e32440ebd945173521@88.208.208.34:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Apr 23 17:27:43] WARNING[9213]: chan_sip.c:4198 retrans_pkt: Hanging up call 2b2effa966c193e32440ebd945173521@88.208.208.34:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
    -- SIP/80.231.23.240-000000e4 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
Was it helpful?

Solution

Check /etc/asterisk/sip_nat.conf and make sure your LAN network is set like:

localnet=192.168.1.0/255.255.255.0

OTHER TIPS

If you do not have NAT enabled on your SIP trunk, turn it on and let me know if that fixes it.

Very likly you have nat or firewall issue. For nat check this article:

http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions

make sure you have correct ip address in 'externip=' in sip.conf under /etc/asterisk.

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