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[Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing problem.

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jgonzalez at sqli.com
Guest

Posted: Mon Jul 27, 2009 11:12 am Post subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing prob

I'm currently trying to connect FreeSwitch to a PBX (Alcatel-Lucent), thanks to a SIP trunk. SIP trunks are available and working on the PBX thanks to a recent update. My problem is that I can't call phones linked to the PBX. When I try to call 300, I've got this message in freeswitch console : 2009-07-27 17:38:48.514105 [NOTICE] switch_channel.c:602 New Channel sofia/internal/[EMAIL PROTECTED] [93d5a10e-7ac3-11de-b456-e5e56113066d] 2009-07-27 17:38:48.516907 [INFO] mod_dialplan_xml.c:252 Processing jgonzalez jgonzalez->300 in context default 2009-07-27 17:38:48.521084 [NOTICE] switch_channel.c:602 New Channel sofia/external/[EMAIL PROTECTED] [93d69816-7ac3-11de-b456-e5e56113066d] 2009-07-27 17:38:48.636073 [NOTICE] sofia.c:3775 Hangup sofia/external/[EMAIL PROTECTED] [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-07-27 17:38:48.636073 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-07-27 17:38:48.637788 [NOTICE] mod_dptools.c:633 Hangup sofia/internal/[EMAIL PROTECTED] [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-27 17:38:48.639047 [NOTICE] switch_core_session.c:1085 Session 15 (sofia/internal/[EMAIL PROTECTED]) Ended 2009-07-27 17:38:48.639047 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/[EMAIL PROTECTED] [CS_DESTROY] 2009-07-27 17:38:48.643042 [NOTICE] switch_core_session.c:1085 Session 16 (sofia/external/[EMAIL PROTECTED]) Ended 2009-07-27 17:38:48.643042 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/[EMAIL PROTECTED] [CS_DESTROY] I've defined, in sip_profiles/external, a gateway to the PBX this way : <include> <gateway name="pbxlyon"> <param name="username" value="pbxlyon"/> <param name="realm" value="[PBX IP address]"/> <param name="password" value="pbxlyon"/> <param name="register" value="false"/> <param name="register-transport" value="udp"/> <param name="retry_seconds" value="30"/> </gateway> </include> And in the dialplan default.xml : <extension name="pbxlyon"> <condition field="destination_number" expression="300"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="sofia/gateway/pbxlyon/300"/> <action application="hangup"/> </condition> </extension> (for the moment, I'm trying only with the number 300 which is a correct number of the phone system). As you can see, I'm far from being an expert of FreeSwitch, SIP or even VoIP in general. I'm learning. I hope you can help me. Regards, Julien Gonzalez. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

brian at freeswitch.org
Guest

Posted: Mon Jul 27, 2009 11:16 am Post subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing prob

I have to guess that you put this at the bottom of the default.xml?

/b

On Jul 27, 2009, at 10:58 AM, julien wrote:

Quote:

And in the dialplan default.xml :

<extension name="pbxlyon">
<condition field="destination_number" expression="300">
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="continue_on_fail=true"/>
<action application="bridge" data="sofia/gateway/pbxlyon/300"/>
<action application="hangup"/>
</condition>
</extension>

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jgonzalez at sqli.com
Guest

Posted: Mon Jul 27, 2009 11:21 am Post subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing prob

No totally at the bottom. Before :

<X-PRE-PROCESS cmd="include" data="default/*.xml"/>

Brian West a Ã©crit :

Quote:

I have to guess that you put this at the bottom of the default.xml?

/b

On Jul 27, 2009, at 10:58 AM, julien wrote:

Quote:

jgonzalez at sqli.com
Guest

Posted: Tue Jul 28, 2009 4:39 am Post subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing prob

Hello brian,
It was not exactly at the bottom but before

<X-PRE-PROCESS cmd="include" data="default/*.xml"/>

I tried to put it higher in the dialplan but it still doesn't work (with
the same error).

Thanks for your help.

Brian West a Ã©crit :

Quote:

I have to guess that you put this at the bottom of the default.xml?

/b

On Jul 27, 2009, at 10:58 AM, julien wrote:

Quote:

brian at freeswitch.org
Guest

Posted: Tue Jul 28, 2009 4:51 am Post subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing prob

Well press F8 and increase the debug level.. then try again you'll
prob. see that its not finding it NOR matching it anywhere in your
dialplan.

/b

On Jul 28, 2009, at 4:32 AM, julien wrote:

Quote:

jason at jasonjgw.net
Guest

Posted: Tue Jul 28, 2009 5:04 am Post subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing prob

julien <jgonzalez@sqli.com> wrote:

Quote:

It was not exactly at the bottom but before

<X-PRE-PROCESS cmd="include" data="default/*.xml"/>

Why not put it in the default directory, from which it will be included by the
above line? If necessary, you could comment out any entries in default.xml
that might be matched first.

I've debugged this kind of problem before, and the best solution has always
been to read the logs carefully to see which extensions matched (or didn't
match).

Also, if necessary, check out freeswitch/log/freeswitch.xml.fsxml to see where
your extension ends up in the final dial plan.

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jgonzalez at sqli.com
Guest

Posted: Tue Jul 28, 2009 10:15 am Post subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing prob

Thanks for the tip Brian. It seems that the extension matches
successfully in the dialplan (PASS, instead of FAIL for all other
entries of the dialplan) :

Dialplan: sofia/internal/[EMAIL PROTECTED] parsing [default->pbxlyon]
continue=false
Dialplan: sofia/internal/[EMAIL PROTECTED] Regex (PASS) [pbxlyon]
destination_number(300) =~ /300/ break=on-false

But it leads nowhere. After the match the connection to the PBX fails :

2009-07-28 16:16:43.963836 [NOTICE] switch_channel.c:602 New Channel
sofia/external/300 [46fca878-7b81-11de-a9c2-0f49fee5280a]
2009-07-28 16:16:43.963836 [DEBUG] mod_sofia.c:2751 (sofia/external/300)
State Change CS_NEW -> CS_INIT
2009-07-28 16:16:43.963836 [DEBUG] switch_core_session.c:933 Send signal
sofia/external/300 [BREAK]
2009-07-28 16:16:43.973759 [DEBUG] switch_core_state_machine.c:397
(sofia/external/300) Running State Change CS_INIT
2009-07-28 16:16:43.973759 [DEBUG] switch_core_state_machine.c:480
(sofia/external/300) State INIT
2009-07-28 16:16:43.973759 [DEBUG] mod_sofia.c:83 sofia/external/300
SOFIA INIT
2009-07-28 16:16:43.975221 [DEBUG] mod_sofia.c:111 (sofia/external/300)
State Change CS_INIT -> CS_ROUTING
2009-07-28 16:16:43.975221 [DEBUG] switch_core_session.c:933 Send signal
sofia/external/300 [BREAK]
2009-07-28 16:16:43.975221 [DEBUG] sofia.c:3215 Channel
sofia/external/300 entering state [calling][0]
2009-07-28 16:16:43.975221 [DEBUG] switch_core_state_machine.c:480
(sofia/external/300) State INIT going to sleep
2009-07-28 16:16:43.975221 [DEBUG] switch_core_state_machine.c:397
(sofia/external/300) Running State Change CS_ROUTING
2009-07-28 16:16:43.975221 [DEBUG] switch_core_state_machine.c:483
(sofia/external/300) State ROUTING
2009-07-28 16:16:43.975221 [DEBUG] mod_sofia.c:130 sofia/external/300
SOFIA ROUTING
2009-07-28 16:16:43.975221 [DEBUG] switch_ivr_originate.c:63
(sofia/external/300) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2009-07-28 16:16:43.975221 [DEBUG] switch_core_session.c:933 Send signal
sofia/external/300 [BREAK]
2009-07-28 16:16:43.976763 [DEBUG] switch_core_state_machine.c:483
(sofia/external/300) State ROUTING going to sleep
2009-07-28 16:16:43.976763 [DEBUG] switch_core_state_machine.c:397
(sofia/external/300) Running State Change CS_CONSUME_MEDIA
2009-07-28 16:16:43.976763 [DEBUG] switch_core_state_machine.c:502
(sofia/external/300) State CONSUME_MEDIA
2009-07-28 16:16:44.82786 [DEBUG] sofia.c:3215 Channel
sofia/external/300 entering state [terminated][404]
2009-07-28 16:16:44.82786 [NOTICE] sofia.c:3775 Hangup
sofia/external/300 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]

It looks to me that it's more a problem from the gateway than from the
dialplan? Don't you think so?
Do you think the way I defined my gateway is good for a connexion to a PBX ?

Thanks for your replies Brian and Jason.

Brian West a Ã©crit :

Quote:

Well press F8 and increase the debug level.. then try again you'll
prob. see that its not finding it NOR matching it anywhere in your
dialplan.

/b

On Jul 28, 2009, at 4:32 AM, julien wrote:

Quote:

brian at freeswitch.org
Guest

Posted: Tue Jul 28, 2009 10:22 am Post subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing prob

The remote end said 404

/b

On Jul 28, 2009, at 10:00 AM, julien wrote:

Quote:

2009-07-28 16:16:44.82786 [DEBUG] sofia.c:3215 Channel
sofia/external/300 entering state [terminated][404]

jgonzalez at sqli.com
Guest

Posted: Wed Jul 29, 2009 3:08 am Post subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing prob

According to http://wiki.freeswitch.org/wiki/Sofia.conf.xml, if the
remote end say 404, it can be reached. So i guess the problem is that
the remote number 300 can't be found on the otherside? Or could it be
that the PBX is reachable but there's a problem of communication between
FS and the PBX?

Brian West a écrit :

Quote:

The remote end said 404

/b

On Jul 28, 2009, at 10:00 AM, julien wrote:

Quote:

2009-07-28 16:16:44.82786 [DEBUG] sofia.c:3215 Channel
sofia/external/300 entering state [terminated][404]

msc at freeswitch.org
Guest

Posted: Wed Jul 29, 2009 11:34 am Post subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing prob

On Wed, Jul 29, 2009 at 1:05 AM, julien <jgonzalez@sqli.com (jgonzalez@sqli.com)> wrote:

Quote:

A 404 in SIP is just like a 404 when web surfing: the target server can't find whatever it is that you're looking for. In other words, your FS server made contact with the server at the far end, told it what endpoint you're looking for, and the server there said, "I can't find this endpoint, sorry." You'll need to see what's happening at the far end, like whether there really is a 300 or not.
-MC

jgonzalez at sqli.com
Guest

Posted: Thu Jul 30, 2009 3:16 am Post subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing prob

Michael Collins a écrit :

Quote:

A 404 in SIP is just like a 404 when web surfing: the target server
can't find whatever it is that you're looking for. In other words,
your FS server made contact with the server at the far end, told it
what endpoint you're looking for, and the server there said, "I can't
find this endpoint, sorry." You'll need to see what's happening at the
far end, like whether there really is a 300 or not.
-MC

Thanks for your help Michael.
Thanks to the SIP trace mode, I saw that, as you said, the remote server
can be contacted but, it can't find the phone with the number 300 :

send 1135 bytes to udp/[PBX IP ADDRESS]:5060 at 07:36:50.305204:
------------------------------------------------------------------------
INVITE sip:300@[PBX IP ADDRESS] SIP/2.0

recv 463 bytes from udp/[FS IP ADDRESS]:5060 at 07:11:33.165472:
------------------------------------------------------------------------
SIP/2.0 404 Not Found
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, REFER, NOTIFY
User-Agent: OxO_GW_710/022.001
To: <sip:300@[PBX IP ADDRESS]>;tag=f75e19901fe0d4a3eed1295e30a93b53

But the private number 300 exist and work very well. So i guess the way
to join the endpoint (an Alcatel IP phone) is not the good one. I even
tried with the public number. I don't think the pbx can find 300@[PBX IP
ADDRESS], it can only find 300 as this phone already has an IP address.

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