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[erick at junctionnetwo...]

Hi There,

I'm trying to get freeswitch to originate all SIP calls through
an outbound proxy. When I use the originate API command to create a call
to a telephone number I see the SIP packets getting to my proxy just fine.
However if I originate a call to a SIP address then proxy server is
bypassed,
instead FS is directly messaging the addressee.

Here is the command that I'm trying to use that behaves unexpectedly:
originate sofia/gateway/proxy/alice@bar.com &echo()

However this command produces the results I'm expecting:
originate sofia/gateway/proxy/15551234 &echo()

Here is the result of my sofia status:

freeswitch> sofia status
API CALL [sofia(status)] output:
Name Type
Data State
=================================================================================================
external profile
sip:mod_sofia@X.X.X.X:5070 RUNNING (0)
proxy gateway
sip:ejjohnson_ippx@ejjohnson.org NOREG
=================================================================================================
1 profile 0 aliases

I have also tried setting the sip_invite_domain channel var through the {}

Could you let me know what I'm doing wrong?

Much appreciated,

Erick J


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[brian at freeswitch.org]

First example is WRONG you don't dial via a gateway that way. If you wish to dial alice@bar.com (alice@bar.com) then try sofia/internal/alice@bar.com ([email]sofia/internal/alice@bar.com[/email]) as you don't require a gateway to call alice right?

/b

On Dec 8, 2008, at 7:14 PM, Erick Johnson wrote:

[brian at freeswitch.org]

Can you clarify why you need a gateway? Is the far side going to challenge us and request authentication credentials?

So you want us to not resolve the domain of the target at all in any way? That kinda breaks the rules because you should always check the NAPTR's and SRV and resolve to the target in that manner its a requirement. If you want to force things to a proxy and let the proxy on the far side do the work then you do this:


sofia/profile/alice@bar.com ([email]sofia/profile/alice@bar.com[/email]);fs_path=proxy.foo.com


/b





On Dec 9, 2008, at 12:06 PM, Erick Johnson wrote:

[brian at freeswitch.org]

I think you need to '' the sofia uri

/b

On Dec 9, 2008, at 1:11 PM, Erick Johnson wrote:

[msc at freeswitch.org]

What SVN rev are you running? Also, could you do a SIP trace?
TPORT_LOG=1 && /usr/local/freeswitch/bin/freeswitch
Pastebin the output of that and we'll take it from there.
-MC

On Tue, Dec 9, 2008 at 11:41 AM, Erick Johnson
<erick@junctionnetworks.com> wrote:

[brian at freeswitch.org]

originate 'sofia/internal/brian@bkw.org;fs_path=bob.com' &echo()

/b

On Dec 9, 2008, at 1:52 PM, Michael Collins wrote:

[erick at junctionnetwo...]

I'm running latest trunk - Revision: 10682

I've been doing an ngrep on my external freeswitch SIP port and FS
is not sending any SIP packets anywhere when I run the following command.
Bumping up TPORT_LOG to 9 also confirms this, as no SIP packets are logged.

originate
'sofia/external/erick@ejjohnson.org;fs_path=proxybeta.jnctn.net' &echo()

Also, just to be clear, when I remove ";fs_path=..." from the command
above a call
is set up normally to erick@ejjohnson.org and the SIP packets are logged
to console.

Thanks guys.

[dave at 3c.co.uk]

Hi Erick,

Not sure if you've tried this (or if it'll help), but you can force routing in the dialplan like so:
<action application="set" data="sip_h_Route=[url=sip:@11.22.33.44;lr]<sip:@11.22.33.44;lr>[/url]" />
<action application="bridge" data="sofia/gateway/gw/$1"/>

Cheers --

Dave

[erick at junctionnetwo...]

Thanks Dave,

Actually I realized my problem (stupid mistake of course). For anyone else
trying to use the fs_path variable the value needs to be a fully
qualified SIP
URI, e.g. "bob@bar.com;fs_path=sip:host.domain.net", notice it being
prefaced
with the "sip:", my problem was that I was only entering
the host name. Then somewhere down in mod_sofia it must have decided that
it didn't like that and just closed the channel.

Hope this helps somebody who gets stuck like I did.

Cheers,

Erick

[msc at freeswitch.org]

On Thu, Dec 11, 2008 at 12:52 PM, Erick Johnson
<erick@junctionnetworks.com> wrote: