Asterisk and Ericsson Integration issue

josephalwary
Joined: Mon 20 of Feb, 2012

Asterisk and Ericsson Integration issue

Posted:Tue 21 of Feb, 2012 (09:05 UTC)
In my firm, we use for telephony a PABX Ericsson Consono MD110 integrated with an Asterisk version 1.6.1.10 server. We have almost 600 analog lines connected to our PABX. We have 1000 IP phone subscribers on Asterisk connected through Mediatirx 4100 series gateways. We also have 200 softphone users on the Asterisk server.
For calls from asterisk to PABX, we use 2 Mediatrix 1204 FXO gateways. For calls from PABX to Asterisk, we use one Mediatrix 4108 FXS gateway. Below is a schematic of our telephony network.

IP phones call to analog phones:
IP phones ---->Mediatrix 4100 series gateways ----> Asterisk Server ---->Mediatrix 1204 FXO gateway ----> PABX Ericsson MD110 ----> Analog phones

Analog phones call to IP phones:
Analog phones ----> PABX Ericsson MD110 ----> Mediatrix 4108 FXS gateway ----> Asterisk Server ----> Mediatrix 4100 series gateways ----> IP phones

For each Mediatrix 1204 FXO gateway, we defined a VOIP trunk on Asterisk user.conf.

users.conf snippet:
[trunk_1]
host=172.35.0.218
username=
secret=
trunkname=FXO_1
context=DID_trunk_1
hasexten=no
hasiax=no
hassip=yes
registeriax=no
registersip=yes
trunk
insecure=no
disallow=all
allow=ulaw


We do not have any problem calling from analog phones to IP phones, but we have problem calling from IP phones to analog phones. Several times a day, both Mediatrix 1204 FXO gateways are congested and we have to reboot them to make them functional again.

Here is sip debug when we call from an IP phone to an analog phone while FXO gateways are congested:

[root@asterisk ~]# asterisk -r
Asterisk 1.6.1.10, Copyright (C) 1999 - 2009 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 1.6.1.10 currently running on asterisk (pid = 2249)
Verbosity is at least 3
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
-- Executing [33353@DLPN_local:1] Macro("SIP/67365-00025418", "trunkdial-failover-0.3,SIP/trunk_1/33353,SIP/trunk_2/33353,trunk_1,trunk_2") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:1] GotoIf("SIP/67365-00025418", "0?1-fmsetcid,1") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:2] GotoIf("SIP/67365-00025418", "0?1-setgbobname,1") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:3] Set("SIP/67365-00025418", "CALLERID(num)=") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:4] Set("SIP/67365-00025418", "CALLERID(all)=") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:5] GotoIf("SIP/67365-00025418", "0?1-dial,1") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:6] Set("SIP/67365-00025418", "CALLERID(all)=") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:7] Set("SIP/67365-00025418", "CALLERID(all)=") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:8] Goto("SIP/67365-00025418", "1-dial,1") in new stack
-- Goto (macro-trunkdial-failover-0.3,1-dial,1)
-- Executing [1-dial@macro-trunkdial-failover-0.3:1] Dial("SIP/67365-00025418", "SIP/trunk_1/33353") in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
-- Called trunk_1/33353
-- Got SIP response 480 "Temporarily Unavailable" back from 172.35.0.218
-- SIP/trunk_1-00025419 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [1-dial@macro-trunkdial-failover-0.3:2] GotoIf("SIP/67365-00025418", "16 > 0 ?1-CONGESTION,1:1-out,1") in new stack
-- Goto (macro-trunkdial-failover-0.3,1-CONGESTION,1)
-- Executing [1-CONGESTION@macro-trunkdial-failover-0.3:1] Dial("SIP/67365-00025418", "SIP/trunk_2/33353") in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
-- Called trunk_2/33353
-- Got SIP response 480 "Temporarily Unavailable" back from 172.35.0.219
-- SIP/trunk_2-0002541a is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [1-CONGESTION@macro-trunkdial-failover-0.3:2] Hangup("SIP/67365-00025418", "") in new stack
== Spawn extension (macro-trunkdial-failover-0.3, 1-CONGESTION, 2) exited non-zero on 'SIP/67365-00025418' in macro 'trunkdial-failover-0.3'
== Spawn extension (DLPN_local, 33353, 1) exited non-zero on 'SIP/67365-00025418'




I would appreciate if somebody can help me to fix this issue. I need to know what the problem is and what should I do to fix it.