no reply to our critical packet

[adoman22]

Hello all, Im having a bit of trouble making calls through the vici server. Below are the specs
centos 6
vicidial-2.9
asterisk 1.8

I was able to register the phone on the server with zoiper but every time i make a test call i get the call on my phone but then it hangs up. so i ran

Code: Select all

asterisk -rvvvv

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[Sep 26 15:27:02]   == Using SIP RTP CoS mark 5
[Sep 26 15:27:02]     -- Executing [915612233457@default:1] AGI("SIP/100-0000000c", "agi://127.0.0.1:4577/call_log")
[Sep 26 15:27:02]     -- <SIP/100-0000000c>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Sep 26 15:27:02]     -- Executing [915612233457@default:2] Dial("SIP/100-0000000c", "SIP/comk/15612233457,,To") in n
[Sep 26 15:27:02]   == Using SIP RTP CoS mark 5
[Sep 26 15:27:02]     -- Called SIP/comk/15612233457
[Sep 26 15:27:06]     -- SIP/comk-0000000d is making progress passing it to SIP/100-0000000c
[Sep 26 15:27:06]   == Manager 'sendcron' logged on from 127.0.0.1
[Sep 26 15:27:06]   == Manager 'sendcron' logged off from 127.0.0.1
[Sep 26 15:27:10]     -- SIP/comk-0000000d answered SIP/100-0000000c
[Sep 26 15:27:16] WARNING[2801]: chan_sip.c:3983 retrans_pkt: Retransmission timeout reached on transmission jd7kKeTHcjOw20oUVdlGdw.. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Sep 26 15:27:16] WARNING[2801]: chan_sip.c:4012 retrans_pkt: Hanging up call jd7kKeTHcjOw20oUVdlGdw.. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[Sep 26 15:27:16]     -- Executing [h@default:1] AGI("SIP/100-0000000c", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----18-----ANSWER-----14-----6") in new stack
[Sep 26 15:27:16]     -- <SIP/100-0000000c>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----18-----ANSWER-----14-----6 completed, returning 0
[Sep 26 15:27:16]   == Spawn extension (default, 915612233457, 2) exited non-zero on 'SIP/100-0000000c'
[Sep 26 15:28:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Sep 26 15:28:02]   == Manager 'sendcron' logged on from 127.0.0.1


Thats the output. Please help

[Rogger]

Hello,

Your problem is network, try setting your carrier directly into a softphone and check firewall and NAT.

Good luck

Rogger Faioli

[adoman22]

Sorry for the late response. Would this be a network issue on my computer where i am using zoiper or on the server where vici is?

[VoIPEssential]

Looks like a NAT issue. Go to Admin > System Settings and make sure the External IP field is set to your public IP address. You will also have to make sure you have proper port forwarding rules on your router/firewall going into your dialer's private IP.

[ambiorixg12]

follow the link displayed on your logs https://wiki.asterisk.org/wiki/display/ ... nsmissions

[adoman22]

Okay so figured it out. When i setup the server i ran the script ADMIN_update_server_ip.pl and then changed the externip in admin->server but apparently it did not update the externip in sip.conf. As soon as i changed that it worked perfectly. Thanks for everyones help and pointing me in the right direction.