Hi Ashish,

PFA, the compiled file which I used.

Thanks & Regards,
-
Akash Deep Verma


On Wed, Apr 17, 2013 at 1:14 PM, <freeswitch-users-request@lists.freeswitch.org> wrote:
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Today's Topics:

   1. Re: PRI channels suspended (Ashish gautam)
   2. Re: Respect the request URI domain and To URI domain when
      bridging (Jos? Luis Mill?n)


---------- Forwarded message ----------
From: Ashish gautam <ashish@nms.co.in>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc: 
Date: Wed, 17 Apr 2013 12:33:23 +0530
Subject: Re: [Freeswitch-users] PRI channels suspended
Hi Akash,

I made these changes to the source and recompiled. Still the same this happening. Channels 5 to 30 are suspended.

Thanks.

On Tue, Apr 16, 2013 at 4:39 PM, Akash Deep Verma <akashdeep.co@gmail.com> wrote:
Hi Ashish,

This is a hack but you get what you need.

This is because, whenever retry attempt is made 1 channel per pri get started, as there are 30 channels and retry attempts are 3 so the channels that freeswitch load is in between 3-6, so I increased retry attempt and complied it again. As I got rid of this error but it takes some time to load the pri channels (30*10 = 300seconds(5 seconds) max).

Also, remember if you dial a call in between restarting phase you may face some error. To avoid this, dial call only after all pri channel get started.

Here are the steps:

In freeswitch source:
Edit libs/freetdm/src/ftmod/ftmod_libpri/ftmod_libpri.h

1. Decrease timeout of channels.
#define T316_TIMEOUT_MS_DEFAULT     30000   /* 30 sec */

change this to 

#define T316_TIMEOUT_MS_DEFAULT     10000   /* 10 sec */

2. Increase attempts to 30.
#define T316_ATTEMPT_LIMIT_DEFAULT  3

change this to

#define T316_ATTEMPT_LIMIT_DEFAULT  30

3. Increase maximum limit as according to default(set more than that of default)
#define T316_ATTEMPT_LIMIT_MAX      10

change this to

#define T316_ATTEMPT_LIMIT_MAX      35

Thanks & Regards,
-
Akash Deep Verma

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---------- Forwarded message ----------
From: "José Luis Millán" <jmillan@aliax.net>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc: 
Date: Wed, 17 Apr 2013 09:43:31 +0200
Subject: Re: [Freeswitch-users] Respect the request URI domain and To URI domain when bridging
Hi,


2013/4/16 João Mesquita <jmesquita@freeswitch.org>
Ain't that what fs_path is for? I am not sure, maybe someone can chime-in here...

Yes, 'fs_path' is the directive I was looking for. It specifies to SIP proxy to which the message is being routed.

Thank you.
 

João Mesquita
FreeSWITCH™ Solutions


On Tue, Apr 16, 2013 at 5:10 PM, José Luis Millán <jmillan@aliax.net> wrote:
Hi,

It was easy to define the sip_invite_full_to for legB given the information exposed. I'm still unable to send the SIP request to a different host that the one specified in the request URI.

Can I send a request to a host that differs from the request URI domain and To header URI domain of legB? If it is possible, any reference to the doc will be much appreciated.

Regards.


2013/4/16 Michael Collins <msc@freeswitch.org>
For the most part I'd say yes. I think just having them on the wiki will be useful, even if it's just the self-explanatory part.
-MC


On Tue, Apr 16, 2013 at 11:56 AM, João Mesquita <jmesquita@freeswitch.org> wrote:
MC, will do, although they are pretty self-explanatory, ain't it?

João Mesquita
FreeSWITCH™ Solutions


On Tue, Apr 16, 2013 at 2:29 PM, Michael Collins <msc@freeswitch.org> wrote:
J,

Thanks for this information. It looks like none of these are on the wiki. Any volunteers out there willing to add these? Even if you don't know what they do you could add placeholders for them. See http://wiki.freeswitch.org/wiki/Variable_skeleton for details.

Thanks,
MC


On Tue, Apr 16, 2013 at 9:32 AM, João Mesquita <jmesquita@freeswitch.org> wrote:
From sofia_glue.c:

const char *invite_record_route = switch_channel_get_variable(tech_pvt->channel, "sip_invite_record_route");
const char *invite_route_uri = switch_channel_get_variable(tech_pvt->channel, "sip_invite_route_uri");
const char *invite_full_from = switch_channel_get_variable(tech_pvt->channel, "sip_invite_full_from");
const char *invite_full_to = switch_channel_get_variable(tech_pvt->channel, "sip_invite_full_to");
const char *handle_full_from = switch_channel_get_variable(tech_pvt->channel, "sip_handle_full_from");
const char *handle_full_to = switch_channel_get_variable(tech_pvt->channel, "sip_handle_full_to");
const char *force_full_from = switch_channel_get_variable(tech_pvt->channel, "sip_force_full_from");
const char *force_full_to = switch_channel_get_variable(tech_pvt->channel, "sip_force_full_to");


Glad to see you using FS José Luis! :-D Welcome.

João Mesquita


On Tue, Apr 16, 2013 at 5:19 AM, José Luis Millán <jmillan@aliax.net> wrote:
Let me fix an error:


This is the message I would like to send to the IP 2.2.2.2 (my proxy)
```
INVITE sip:jmillan@1.2.3.4 SIP/2.0 Via: SIP/2.0/TCP bl23sl8lmku7.invalid;branch=z9hG4bK9420964 To: <sip:jmillan@1.2.3.4>```
```

Regards.



2013/4/16 José Luis Millán <jmillan@aliax.net>
Hi all,

Every call bridged though a specific 'gateway' does rewrite the SIP request URI domain and To URI domain values to the gateway's 'proxy' parameter value.

I would like to respect such values and receive in my proxy the SIP messages with their original request URI domain and To URI domain. 

This is, I would like to be able to send a SIP message with the following request URI and To headers, to the IP address 2.2.2.2

```
INVITE sip:jmillan@1.2.3.4 SIP/2.0 Via: SIP/2.0/TCP bl23sl8lmku7.invalid;branch=z9hG4bK9420964 To: <sip:d@1.2.3.4>```
```

Is it possible to handle this with FreeSwitch?

I have followed the FS wiki but I've been unable to handle this.

Thanks in advance.

--
José Luis Millán



--
José Luis Millán

_________________________________________________________________________
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--
Michael S Collins
Twitter: @mercutioviz
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_________________________________________________________________________
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_________________________________________________________________________
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--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
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http://www.cluecon.com

FreeSWITCH-users mailing list
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--
José Luis Millán

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
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_________________________________________________________________________
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consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
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Official FreeSWITCH Sites
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--
José Luis Millán

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