[Freeswitch-users] Resend: Issues with SIP + TCP?

[Peter Olsson]

I'm also having problems with this. When running FS compiled about 10 days ago it works fine (don't remember exact revision), but when using latest SVN it doesn't work anymore. I seems like it's trying to use UDP when it should use TCP.

My setup is this: Avaya Communication Manager PBX -> Talks TLS to Avaya SIP SES Server -> Talks TCP to FreeSwitch.

The replies from FS seems to be sent using UDP instead of TCP, and when I keep the config and revert to the 10 day old version it starts working again, so there is definately something wrong.

I'll try to do some more testing, and get back with some SIP-traces as well.

/Peter


-----Ursprungligt meddelande-----
Från: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] För Kristian Kielhofner
Skickat: den 26 oktober 2009 16:47
Till: freeswitch-users at lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

Tony,

  It seemed strange to me too (I'm using TCP in other places).

  I'll take another look at this with your suggestions for debugging.

  Thanks!

On Mon, Oct 26, 2009 at 11:25 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> i cant seem to reproduce it.
>
> originate sofia/internal/1235 at conference.freeswitch.org;transport=tcp 9998
>
> I get a working call and trace.
>
> Could you possibly have a dns error?  I know it's an ip but it may still
> fail if it has no dns.
>
> try
>
> sofia loglevel all 9
>
> and look for other errors.
>
>
>
>
> On Mon, Oct 26, 2009 at 9:56 AM, Kristian Kielhofner
> <kristian.kielhofner at gmail.com> wrote:
>>
>> I originally sent this last Friday but I've been unable to confirm it
>> ever made it to the list.
>>
>> Hello everyone,
>>
>>  I'm having some issues with SIP and TCP.  I've used it before with
>> success but I'm seeing some strange behavior...
>>
>>  Level 7 debugs with siptrace on both profiles.  UDP invite from
>> softphone comes in on port 5062, it's supposed to bridge to
>> 10.70.0.62.  When configured to use UDP FS sends an INVITE (nothing
>> currently answers) while TCP doesn't send anything (confirmed with
>> siptrace and packet sniffer).  I confirmed this behavior with a
>> gateway configured for TCP and appending ;transport=tcp to a bridge
>> line.
>>
>>  This is trunk rev 15211 on an Intel Mac running Snow Leopard.  I've
>> also confirmed this behavior on an Intel Linux machine running Ubuntu
>> (not sure of version ATM).
>>
>> TCP:
>>
>> http://pastebin.freeswitch.org/10825
>>
>> UDP:
>>
>> http://pastebin.freeswitch.org/10826
>>
>> dialplan (UDP):
>>
>>   <extension name="smhpbx">
>>     <condition field="destination_number" expression="^(7887)$">
>>       <action application="set" data="call_timeout=60"/>
>>       <action application="set" data="effective_caller_id_name=Voalte
>> Test"/>
>>       <action application="set"
>> data="effective_caller_id_number=19412848354"/>
>>       <action application="bridge" data="sofia/avaya/7887 at 10.70.0.62"/>
>>     </condition>
>>   </extension>
>>
>> dialplan (TCP):
>>
>>   <extension name="smhpbx">
>>     <condition field="destination_number" expression="^(7887)$">
>>       <action application="set" data="call_timeout=60"/>
>>       <action application="set" data="effective_caller_id_name=Voalte
>> Test"/>
>>       <action application="set"
>> data="effective_caller_id_number=19412848354"/>
>>       <action application="bridge"
>> data="sofia/avaya/7887 at 10.70.0.62;transport=tcp"/>
>>     </condition>
>>   </extension>
>>
>>  Any thoughts?
>>
>> Thanks!
>>
>> --
>> Kristian Kielhofner
>> http://www.astlinux.org
>> http://blog.krisk.org
>> http://www.star2star.com
>> http://www.submityoursip.com
>> http://www.voalte.com
>>
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>
>
>
> --
> Anthony Minessale II
>
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>
> AIM: anthm
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>
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>
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>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com

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