Bug #5590
closed
Please HELP! Agent disconnected, calls don't work, voip timeout
0%
Description
Hi, please help me!!!
I'm a newbie and I've made a Goautodial installation server (v.3.3) for a custumer that has 8 agents, just follow your guide for the configuration but I got this problems:
1) Agent take this message after few seconds of login: "You have been disconnected"
2) When agent do login, XLite rings like an incoming call with a telephone number that is a campaign code and when we answer, the call goes closed;
3) Calls don't work manually.
4) Our Voip provider is Voicetrading.com and it works if I try it on XLite without Goautodial carriers configuration.
5) I've put a static lan ip on server and the adsl is a business one, with static public ip.
I've follow every step of your guide for the configuration of the server and made a nat on router but it doesn't work properly! I've also try to read forums solution but they don't work for my situation.
Can you tell me please what do you need to understand and solve those problems? Please is urgent, I can't figure it out and I have to solve as soon as possible!
Thank's a lot and wish you a good work!
Serena
Updated by Serena Se over 6 years ago
[root@go ~]# asterisk -vvvvvvvvvr
Asterisk 1.8.23.0-1_centos5.go RPM by demian@goautodial.com, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details. =========================================================================
Connected to Asterisk 1.8.23.0-1_centos5.go RPM by demian@goautodial.com currently running on go (pid = 3155)
Verbosity is at least 9
[Sep 7 15:36:46] WARNING3306: chan_sip.c:3714 __sip_xmit: sip_xmit of 0x2aaed05a4880 (len 459) to (null) returned 1: Invalid argument
[Sep 7 15:36:50] WARNING3306: chan_sip.c:3714 __sip_xmit: sip_xmit of 0x2aaed05a4880 (len 459) to (null) returned -1: Invalid argument
[Sep 7 15:36:54] WARNING3306: chan_sip.c:3714 __sip_xmit: sip_xmit of 0x2aaed05a4880 (len 459) to (null) returned -1: Invalid argument
[Sep 7 15:36:55] ERROR3306: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("sip.voicetrading.com", "(null)", ...): Temporary failure in name resolution
[Sep 7 15:36:55] WARNING3306: acl.c:590 resolve_first: Unable to lookup 'sip.voicetrading.com'
[Sep 7 15:36:55] WARNING3306: acl.c:719 ast_ouraddrfor: Cannot connect
[Sep 7 15:36:55] WARNING3306: chan_sip.c:3714 __sip_xmit: sip_xmit of 0x2aaed05a3700 (len 459) to (null) returned -1: Invalid argument
[Sep 7 15:36:55] NOTICE3306: chan_sip.c:13795 sip_reg_timeout: - Registration for 'Voiceservice@sip.voicetrading.com' timed out, trying again (Attempt #621)
[Sep 7 15:36:56] WARNING3306: chan_sip.c:3714 __sip_xmit: sip_xmit of 0x2aaed05a3700 (len 459) to (null) returned -1: Invalid argument
[Sep 7 15:36:57] WARNING3306: chan_sip.c:3714 __sip_xmit: sip_xmit of 0x2aaed05a3700 (len 459) to (null) returned -1: Invalid argument
[Sep 7 15:36:59] WARNING3306: chan_sip.c:3714 __sip_xmit: sip_xmit of 0x2aaed05a3700 (len 459) to (null) returned -1: Invalid argument
Manager 'sendcron' logged on from 127.0.0.1
Manager 'sendcron' logged off from 127.0.0.1
Manager 'sendcron' logged on from 127.0.0.1
[Sep 7 15:37:03] WARNING3306: chan_sip.c:3714 __sip_xmit: sip_xmit of 0x2aaed05a3700 (len 459) to (null) returned -1: Invalid argument
Manager 'sendcron' logged off from 127.0.0.1
Manager 'sendcron' logged on from 127.0.0.1
Manager 'sendcron' logged off from 127.0.0.1
[Sep 7 15:37:07] WARNING3306: chan_sip.c:3714 __sip_xmit: sip_xmit of 0x2aaed05a3700 (len 459) to (null) returned -1: Invalid argument
[Sep 7 15:37:11] WARNING3306: chan_sip.c:3714 __sip_xmit: sip_xmit of 0x2aaed05a3700 (len 459) to (null) returned -1: Invalid argument
go*CLI>
Updated by Serena Se over 6 years ago
[Voicetrading]
disallow=all
allow=g729
allow=ulaw
type=peer
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=very
nat=yes
host=sip.voicetrading.com
username=xxxxxxxx
secret=xxxxxxxx
allow=gsm
Here is the dial plan:
exten => _6067754251.,1,AGI
exten => _6067754251.,2,Dial(SIP/${EXTEN:10}@Voicetrading,,tTo)
exten => _6067754251.,3,Hangup
Please we need halp to solve it as soon as possible!
Thank's
Regards
Serena
Updated by Onil Jimenez Artiles over 6 years ago
Hello,
Are the other agents also disconnected after several seconds of logging in?
In case the answer is positive, verify that you do not have a conflict of extensions.
If other agents can make your calls without problems, discard a problem with the provider unless you got fewer channels available than the number of your agents, to test it just put the agent to make a call when the others are disconnected.
Regards
Onil
Updated by