2
votes

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/[email protected]: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 [email protected]: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 [email protected]: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 [email protected]: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 [email protected]: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)
4

4 Answers

4
votes

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

localnet=192.168.1.0/255.255.255.0
0
votes

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

0
votes

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

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

0
votes

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