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[kristian.kielhofner at...]

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@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@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 at g...]

i cant seem to reproduce it.

originate sofia/internal/1235@conference.freeswitch.org (1235@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@gmail.com (kristian.kielhofner@gmail.com)> wrote:

[kristian.kielhofner at...]

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@gmail.com> wrote:

[peter.olsson at vision...]

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@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Kristian Kielhofner
Skickat: den 26 oktober 2009 16:47
Till: freeswitch-users@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@gmail.com> wrote:

[peter.olsson at vision...]

Hmm... I remembered incorrectly about my setup The Avaya PBX talks TLS to the Avaya SES Server, and then UDP to FS, not TCP - sorry, my bad!

However, something that has changed the last 10 days seems to affect my setup so it doesn't work anymore. I'll do some more SIP tracing, and get back when I know more about it.

/Peter


-----Ursprungligt meddelande-----
Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Kristian Kielhofner
Skickat: den 26 oktober 2009 16:47
Till: freeswitch-users@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@gmail.com> wrote:

[brian at freeswitch.org]

Finding the exact rev that broke it would be helpful.

/b

On Oct 26, 2009, at 12:06 PM, Peter Olsson wrote:

[peter.olsson at vision...]

Yes, I know However, now I think this is related to the new headers introduced, it's probably not a TCP issue.

Everything seems to work just fine until the 200 OK is sent, the Avaya PBX doesn't seem to accept that reply anymore.

The only differences I've found between a working revision, and a non-working is this;

In the working 200-OK transaction method UPDATE is listed in the Allow-header, and there is a header called "X-Actually-Support: UPDATE".

In the non-working one I don't have these, and instead I have these headers;
X-FS-Display-Name: 9099
X-FS-Display-Number: 9099
X-FS-Support: update_display
P-Asserted-Identity: "9099" <9099>

So I guess this could be related to the other thread going on right now "Downloaded tar vs latest SVN - 200 OK has more headers", and not a TCP issue. It's probably the Avaya doing something wrong (not the first time), but still it seems these changes affect more systems than mine.

I'm just using it as a lab setup for now, so if anyone want me to test something, I can do it immediately.

/Peter


-----Ursprungligt meddelande-----
Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Brian West
Skickat: den 26 oktober 2009 18:23
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

Finding the exact rev that broke it would be helpful.

/b

On Oct 26, 2009, at 12:06 PM, Peter Olsson wrote:

[brian at freeswitch.org]

Bet your hardware just barfs on those like others have... I mean
really I HATE SIP. This is stupid.

/b

On Oct 26, 2009, at 12:39 PM, Peter Olsson wrote:

[anthony.minessale at g...]

try r15230

add the profile param

<param name="pass-callee-id" value="false"/>


On Mon, Oct 26, 2009 at 12:46 PM, Brian West <brian@freeswitch.org (brian@freeswitch.org)> wrote:

[peter.olsson at vision...]

I understand your frustration We deal with SIP integration with about 10 different PBX vendors today, And it's always something that doesn't work as it should. Right now I don't have anything more connected to FS though.

/Peter


-----Ursprungligt meddelande-----
Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Brian West
Skickat: den 26 oktober 2009 18:46
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

Bet your hardware just barfs on those like others have... I mean
really I HATE SIP. This is stupid.

/b

On Oct 26, 2009, at 12:39 PM, Peter Olsson wrote:

[brian at freeswitch.org]

At some point we'll have to NO NO NO fix your broken crap. The
reason we have sip so broken now is NOBODY CAN SAY NO!... Say it with
me... "NO!"

/b

On Oct 26, 2009, at 1:27 PM, Peter Olsson wrote:

[msc at freeswitch.org]

On Mon, Oct 26, 2009 at 11:40 AM, Brian West <brian@freeswitch.org (brian@freeswitch.org)> wrote:

[brian at freeswitch.org]

You have SIPit, which was the SIP Backoff till Pillsbury got their
panties in a wad.

/b

On Oct 26, 2009, at 3:03 PM, Michael Collins wrote:

[woof at iwoof.org]

Woof!

On Mon, 26 Oct 2009 16:03:59 -0400, Michael Collins <msc@freeswitch.org>
wrote:

[peter.olsson at vision...]

Thanks Brian, it works now. I'll try to learn to say NO next time

/Peter


-----Ursprungligt meddelande-----
Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Brian West
Skickat: den 26 oktober 2009 19:40
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

At some point we'll have to NO NO NO fix your broken crap. The
reason we have sip so broken now is NOBODY CAN SAY NO!... Say it with
me... "NO!"

/b

On Oct 26, 2009, at 1:27 PM, Peter Olsson wrote: