IAX2 does not hang up

[caspar]

I have 2x servers:
1) Mercury (10.0.0.161) runs Asterisk + VICIDIAL + MySQL + Apache
2) Beta (10.0.0.149) runs Asterisk only

I have some SIP phones that registers on Mercury which works fine.
I have some SIP phones that registers on Beta that is connected through an IAX2 trunk to Mercury, which seems to work fine except that it will not hang up.

A typical phone configuration looks like this in the VICIDIAL Admin page:

Phone extension: mercury:test@10.0.0.149/1000@default
Dialplan Number: 1000
...
Server IP: 10.0.0.149 - Beta
...
Status: Active
Active Account: Y
...
Client Protocol: IAX2
...
Manager Login: cron
Manager Secret: 1234
...
Exten Context: default
...

The Beta server configuration looks like this in the VICIDIAL Admin page:

...
Server IP Address: 10.0.0.149
Active: Y
...
Telnet Host: 10.0.0.149
Telnet Port: 5038
Manager User: cron
Manager Secret: 1234
Manager Update User: updatecron
Manager Listen User: listencron
Manager Send User: sendcron
...
Default Context: default

Does my configuration look okay? When a Beta-agent login to VICIDIAL, everything seems to work fine except that when the agent logout, VICIDIAL does not hang up the line. Any ideas how I could fix this?

[dlapitan]

Hi,

You have 2 server. Single box(Mercury (10.0.0.161) and separate server Beta (10.0.0.149). Which one of the server you having problem? Are the two servers connected each other?
Can you please post your /etc/astguiclient.conf.

[caspar]

No problems with SIP phones on Mercury.

The problem is with the SIP phones that registers on Beta and connects through IAX2 to Mercury.

This is the /etc/astguiclient.conf on Mercury.

Code: Select all

# astguiclient.conf - configuration elements for the astguiclient package
# this is the astguiclient configuration file
# all comments will be lost if you run install.pl again

# Paths used by astGUIclient
PATHhome => /usr/share/astguiclient
PATHlogs => /var/log/astguiclient
PATHagi => /var/lib/asterisk/agi-bin
PATHweb => /var/www/html
PATHsounds => /var/lib/asterisk/sounds
PATHmonitor => /var/spool/asterisk/monitor

# The IP address of this machine
VARserver_ip => 10.0.0.161

# Database connection information
VARDB_server => localhost
VARDB_database => asterisk
VARDB_user => cron
VARDB_pass => 1234
VARDB_port => 3306

# Alpha-Numeric list of the astGUIclient processes to be kept running
# (value should be listing of characters with no spaces: 123456)
#  X - NO KEEPALIVE PROCESSES (use only if you want none to be keepalive)
#  1 - AST_update
#  2 - AST_send_listen
#  3 - AST_VDauto_dial
#  4 - AST_VDremote_agents
#  5 - AST_VDadapt (If multi-server system, this must only be on one server)
#  6 - FastAGI_log
#  7 - AST_VDauto_dial_FILL (only for multi-server, this must only be on one server)
VARactive_keepalives => 123456

# Settings for FastAGI logging server
VARfastagi_log_min_servers => 3
VARfastagi_log_max_servers => 16
VARfastagi_log_min_spare_servers => 2
VARfastagi_log_max_spare_servers => 8
VARfastagi_log_max_requests => 1000
VARfastagi_log_checkfordead => 30
VARfastagi_log_checkforwait => 60


[aster1]

this could be a iax issue . i had same problem with one of my providers .Try sip instead

[capmicro]

I have the same probleme.
I have a serveur with Asterisk (1.2.18) and Vicidial.
And a second with Asterisk(1.4.2) whit ip phones in sip.

The two servers connected throught IAX2 trunk.

The lougout doesn't work.
I've tried with a SIP trunk, no changes.

[mflorell]

I have experienced issues with VOIP trunks between 1.2.x and 1.4.X and would not recommend using different trees of Asterisk together.

[capmicro]

i have installed another 1.2.18 to replace the 1.4.2 but i have always the same problem

[mflorell]

You must have an Asterisk configuration issue somewhere then.

[caspar]

Both my servers run Asterisk 1.2.17

[caspar]

SOLVED!

I also had another application that was also using the IAX trunks so I added a separate additional IAX connection for VICIDIAL with the "context=default". I assumed that by default the "default" context will be used and that you can override this with in the extensions.conf file as well, but in reality it does not work like that.

I also added @default to each IAX extension in the dial plan.