Call hangup right after dial in asterisk
18,287
Check /etc/asterisk/sip_nat.conf
and make sure your LAN network is set like:
localnet=192.168.1.0/255.255.255.0
Author by
Huzoor Bux
I am a PHP developer since 2009 in Super Technologies Inc,
Updated on June 04, 2022Comments
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Huzoor Bux almost 2 years
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)