Inbound Retransmittion Timeout

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Inbound Retransmittion Timeout

Postby Keyfin » Sat Nov 12, 2016 12:14 pm

All Twilio carrier inbound numbers are dropping after 6-7 seconds.
Code: Select all
[Nov 12 12:08:44]     -- Executing [+17272050085@trunkinbound:1] Goto("SIP/Twili                                                                             o1-0000014e", "trunkinbound,7272050085,1") in new stack
[Nov 12 12:08:44]     -- Goto (trunkinbound,7272050085,1)
[Nov 12 12:08:44]     -- Executing [7272050085@trunkinbound:1] AGI("SIP/Twilio1-                                                                             0000014e", "agi-DID_route.agi") in new stack
[Nov 12 12:08:44]     -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID                                                                             _route.agi
[Nov 12 12:08:44]     -- <SIP/Twilio1-0000014e>AGI Script agi-DID_route.agi comp                                                                             leted, returning 0
[Nov 12 12:08:44]     -- Executing [99909*13***DID@default:1] Answer("SIP/Twilio                                                                             1-0000014e", "") in new stack
[Nov 12 12:08:45]     -- Executing [99909*13***DID@default:2] AGI("SIP/Twilio1-0                                                                             000014e", "agi-VDAD_ALL_inbound.agi") in new stack
[Nov 12 12:08:45]     -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDA                                                                             D_ALL_inbound.agi
[Nov 12 12:08:45]     -- <SIP/Twilio1-0000014e> Playing 'sip-silence.gsm' (escap                                                                             e_digits=) (sample_offset 0) (language 'en')
[Nov 12 12:08:45]     -- <SIP/Twilio1-0000014e> Playing 'sip-silence.gsm' (escap                                                                             e_digits=) (sample_offset 0) (language 'en')
[Nov 12 12:08:45]        > 0x7fc2f8001b40 -- Probation passed - setting RTP sour                                                                             ce address to 54.172.61.234:19060
[Nov 12 12:08:47]     -- Started music on hold, class 'default', on SIP/Twilio1-                                                                             0000014e
[Nov 12 12:08:50]     -- Stopped music on hold on SIP/Twilio1-0000014e
[Nov 12 12:08:50]     -- <SIP/Twilio1-0000014e> Playing 'sip-silence.gsm' (escap                                                                             e_digits=) (sample_offset 0) (language 'en')
[Nov 12 12:08:50]     -- <SIP/Twilio1-0000014e> Playing 'sip-silence.gsm' (escap                                                                             e_digits=) (sample_offset 0) (language 'en')
[Nov 12 12:08:50]     -- <SIP/Twilio1-0000014e> Playing 'sip-silence.gsm' (escap                                                                             e_digits=) (sample_offset 0) (language 'en')
[Nov 12 12:08:50]     -- <SIP/Twilio1-0000014e> Playing 'generic_hold.gsm' (esca                                                                             pe_digits=) (sample_offset 0) (language 'en')
[Nov 12 12:08:51] WARNING[1887]: chan_sip.c:4038 retrans_pkt: Retransmission tim                                                                             eout reached on transmission 53d5aa88673f35e08da00f79564f283c@0.0.0.0 for seqno                                                                              524156 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP                                                                             +Retransmissions
Packet timed out after 6399ms with no response
[Nov 12 12:08:51] WARNING[1887]: chan_sip.c:4067 retrans_pkt: Hanging up call 53                                                                             d5aa88673f35e08da00f79564f283c@0.0.0.0 - no reply to our critical packet (see ht                                                                             tps://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[Nov 12 12:08:51]     -- <SIP/Twilio1-0000014e>AGI Script agi-VDAD_ALL_inbound.a                                                                             gi completed, returning 4
[Nov 12 12:08:51]   == Spawn extension (default, 99909*13***DID, 2) exited non-z                                                                             ero on 'SIP/Twilio1-0000014e'
[Nov 12 12:08:51]     -- Executing [h@default:1] AGI("SIP/Twilio1-0000014e", "ag                                                                             i://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----18---------------") i                                                                             n new stack
[Nov 12 12:08:51]     -- <SIP/Twilio1-0000014e>AGI Script agi://127.0.0.1:4577/c                                                                             all_log--HVcauses--PRI-----NODEBUG-----18--------------- completed, returning 0


Carrier settings
Code: Select all
[Twilio1]
type=peer
secret=[twilio secret]
username=[twilio username]
host=[name].pstn.twilio.com
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw
insecure=port,invite
fromuser=+17272050075
fromdomain=[name].pstn.twilio.com
protocol: sip
global string: DIAL8TRUNK = SIP/Twilio1
exten => _81NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _81NXXNXXXXXX,n,Dial(${DIAL8TRUNK}/+1${EXTEN:2},,To)
exten => _81NXXNXXXXXX,n,Hangup

exten => _8NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _8NXXNXXXXXX,n,Dial(${DIAL8TRUNK}/+1${EXTEN:1},,tTo)
exten => _8NXXNXXXXXX,n,Hangup


I have already tried turning off the firewall on the server, no change, any suggestions?
ViciBox: 7.0.3 | VERSION: 2.14-585a BUILD: 170114-1356 | SVN Version: 2661 |Single Server | DGG installed
Keyfin
 
Posts: 60
Joined: Tue Feb 23, 2016 8:27 pm

Re: Inbound Retransmittion Timeout

Postby ambiorixg12 » Sun Nov 13, 2016 1:25 am

the logs are throwing the cause of your issue
+Retransmissions
Packet timed out after 6399ms with no response
[Nov 12 12:08:51] WARNING[1887]: chan_sip.c:4067 retrans_pkt: Hanging up call 53 d5aa88673f35e08da00f79564f283c@0.0.0.0 - no reply to our critical packet (see ht tps://wiki.asterisk.org/wiki/display/AS ... nsmissions).


This means NAT issue and you havent defined any option for NAT support on your sip.conf file
ambiorixg12
 
Posts: 453
Joined: Tue Sep 17, 2013 10:35 pm

Re: Inbound Retransmittion Timeout

Postby Keyfin » Mon Nov 14, 2016 10:42 am

yep, thanks, i had to add these three lines to the sip.conf general section, manually

udpenable=yes
tcpenable=yes
preferred_codec_only=yes

also for some reason,

externip = was commented out,

;externip =
changed to
externip =
ViciBox: 7.0.3 | VERSION: 2.14-585a BUILD: 170114-1356 | SVN Version: 2661 |Single Server | DGG installed
Keyfin
 
Posts: 60
Joined: Tue Feb 23, 2016 8:27 pm


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