Sending SIP calls via outbound proxy
i forgot to give you the pastebin URL http://pastebin.freeswitch.org/6379
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 at 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 at ejjohnson.org and the SIP packets are logged to console.
Thanks guys.
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 at junctionnetworks.com> wrote:
Both:
originate sofia/external/'erick at ejjohnson.org;fs_path=proxybeta.foo.net' &echo() originate sofia/external/erick at ejjohnson.org;fs_path=proxybeta.foo.net &echo()
produce the exact same result & log
:(
* I think you need to '' the sofia uri /b
-- Erick Johnson sip/email erick@junctionnetworks.com 1-800-801-3381 x7006 Software Engineer Junction Networks
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=<sip:@11.22.33.44;lr>" /> <action application="bridge" data="sofia/gateway/gw/$1"/> Cheers -- Dave
i forgot to give you the pastebin URL http://pastebin.freeswitch.org/6379
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 at 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 at ejjohnson.org and the SIP packets are logged to console.
Thanks guys.
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 at junctionnetworks.com> wrote:
Both:
originate sofia/external/'erick at
ejjohnson.org;fs_path=proxybeta.foo.net'
&echo() originate sofia/external/erick at
ejjohnson.org;fs_path=proxybeta.foo.net
&echo()
produce the exact same result & log
:(
* I think you need to '' the sofia uri /b
-- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk
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
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=<sip:@11.22.33.44;lr>" /> <action application="bridge" data="sofia/gateway/gw/$1"/>
Cheers --
Dave
On Thu, Dec 11, 2008 at 12:52 PM, Erick Johnson <erick@junctionnetworks.com> wrote:
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.
Erick, thanks for the clarification! I'll get it put on the wiki right away. -MC
Hope this helps somebody who gets stuck like I did.
Cheers,
Erick
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=<sip:@11.22.33.44;lr>" /> <action application="bridge" data="sofia/gateway/gw/$1"/>
Cheers --
Dave
_______________________________________________ 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
Is there a mechanism to configure FS to work with an edge proxy? What I am attempting to achieve is a front end proxy that all the clients connect to which simply forwards all messages to FS. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, December 11, 2008 11:00 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Sending SIP calls via outbound proxy On Thu, Dec 11, 2008 at 12:52 PM, Erick Johnson <erick@junctionnetworks.com> wrote:
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.
Erick, thanks for the clarification! I'll get it put on the wiki right away. -MC
Hope this helps somebody who gets stuck like I did.
Cheers,
Erick
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=<sip:@11.22.33.44;lr>" /> <action application="bridge" data="sofia/gateway/gw/$1"/>
Cheers --
Dave
_______________________________________________ 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
_______________________________________________ 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
From the Wiki..
Specifying SIP Proxy With fs_path You can route a call through a specific SIP proxy by using the "fs_path" directive. Example: sofia/foo/user@that.domain;fs_path=sip:proxy.this.domain Regards, -Adam -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Simon Shaw Sent: Wednesday, January 28, 2009 9:52 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Sending SIP calls via outbound proxy Is there a mechanism to configure FS to work with an edge proxy? What I am attempting to achieve is a front end proxy that all the clients connect to which simply forwards all messages to FS. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, December 11, 2008 11:00 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Sending SIP calls via outbound proxy On Thu, Dec 11, 2008 at 12:52 PM, Erick Johnson <erick@junctionnetworks.com> wrote:
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.
Erick, thanks for the clarification! I'll get it put on the wiki right away. -MC
Hope this helps somebody who gets stuck like I did.
Cheers,
Erick
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=<sip:@11.22.33.44;lr>" /> <action application="bridge" data="sofia/gateway/gw/$1"/>
Cheers --
Dave
_______________________________________________ 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
_______________________________________________ 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 _______________________________________________ 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
participants (5)
-
Adam Long
-
David Knell
-
Erick Johnson
-
Michael Collins
-
Simon Shaw