Asterisk and Ericsson Integration issue

Asterisk Asterisk General
#1

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
trunkstyle=voip
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:

[i][root@asterisk ~]# asterisk -r
Asterisk 1.6.1.10, Copyright © 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]

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.

#2

This is the wrong forum. If you are prepared to work at raw configuration file level, you want Asterisk Support, however you are clearly using a GUI, so a GUI related forum may be better. Asterisk General is for discussing the philosophy and future of Asterisk.

Also, it is very difficult to debug SIP problem with sip debugging traces, including the actual SIP packet data. As this, on a quick skim, is about failures to close sessions, sip show channels and core show channels may be useful.

You should note that 1.6.1.10 is no longer supported, along with the whole 1.6.1.x branch, even for security fixes, and is not the most recent version on the 1.6.1 branch. That is 1.6.1.25. There will be dozens of bug fixes between those versions.

#3

I would suspect that this is a Mediatirx gateway issue. The trace cleary states that the gateway refused the SIP Invite with SIP 480 “Temporarily Unavailable”.

#4

Thanks david55 for your reply.
What forum do you recommend?
Would you also let me know how can I debug the issue? what command should I use and how to proceed?
For upgrade my Asterisk server, do you have any helpful instruction or How to?

#5

Thank you dejanst. I could not find any forum for Mediatrix. I also have problem contacting them. Do you know any helpful community so I can use to follow this issue?

#6

Asterisk Support, if you don’t mind dealing directly with the Asterisk configurations normally maintained by the GUI.

AsteriskNow Support if the GUI was the result of installing Asterisk Now.

Otherwise, you need to identify the GUI and search for a forum for that GUI.

#7

I would try contacting Mediatrix support (e-mail). No forum applys when you are using a non-popular gateway.

#8

We use mediatrix alot to integrate with our Siemens Hipath PBX’s, and let me tell you, thats a harware that you can’t rely on!

#9

I just wonder why you didnt trunk Asterisk to MD110 through E1 ? you can use PCI cards in asterisk for that and they are working very well . Im using Asterisk instead of AXD to transfer MD110 Calls between each other ( E1 Trunk ) and they are working without any problem .