Call hangup right after dial in asterisk

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)

score 4 · Accepted Answer · answered Aug 29 '13 at 01:46

4

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

localnet=192.168.1.0/255.255.255.0

answered Aug 29 '13 at 01:46

Nicolas

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score 0 · Answer 2 · answered Apr 25 '13 at 01:54

0

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

answered Apr 25 '13 at 01:54

MichelV69

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NAT enable but not fixed :( – Huzoor Bux Apr 25 '13 at 06:05
Ok. Do you have firewall rules in place to allow SIP & RTP incoming and outgoing? – MichelV69 Apr 25 '13 at 22:31

score 0 · Answer 3 · answered Apr 25 '13 at 05:48

0

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

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

answered Apr 25 '13 at 05:48

arheops

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score 0 · Answer 4 · answered Sep 09 '16 at 07:44

0

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

answered Sep 09 '16 at 07:44

sofs1

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Call hangup right after dial in asterisk

4 Answers4