[Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value

Jerry Richards

31 Jan 2011 31 Jan '11

8:07 p.m.

Hello All, When I set the call_timeout variable (to 12 seconds) before a bridge to an outbound PSTN call, Freeswitch always rings for 24 seconds before going to voicemail. If I call an internal extension, then it uses the call_timeout variable. Do you know why it would do this? Here is the line that sets the call_timeout variable: switch_channel_set_variable(channel, "call_timeout", "6"); Here is the PSTN bridge log: EXECUTE sofia/internal/2003@192.168.72.79:5060 bridge({presence_id=2001@192.168.72.79}[lcr_carrier=Carrier / Location 1/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]openzap/smg_prid/a/15248142341@g1) Thanks, Jerry

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Anthony Minessale

31 Jan 31 Jan

8:13 p.m.

New subject: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value

You should use {originate_timeout=N} but that may not be the actual problem. You would probably have to look at a trace with debug level etc to be sure. On Mon, Jan 31, 2011 at 2:07 PM, Jerry Richards <jerry.richards@teotech.com> wrote:

...

Hello All,

When I set the call_timeout variable (to 12 seconds) before a bridge to an outbound PSTN call, Freeswitch always rings for 24 seconds before going to voicemail. If I call an internal extension, then it uses the call_timeout variable. Do you know why it would do this?

Here is the line that sets the call_timeout variable:

switch_channel_set_variable(channel, "call_timeout", "6");

Here is the PSTN bridge log:

EXECUTE sofia/internal/2003@192.168.72.79:5060 bridge({presence_id=2001@192.168.72.79}[lcr_carrier=Carrier / Location 1/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]openzap/smg_prid/a/15248142341@g1)

Thanks,

Jerry

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Jerry Richards

14 Feb 14 Feb

6:24 p.m.

New subject: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value

I found out why my call_timeout was not kicking in when I call a PSTN number. It is because of early media. Is there a way to play early media and also have the call_timeout kick-in according to its setting? Thanks, Jerry -----Original Message----- From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, January 31, 2011 12:13 PM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value You should use {originate_timeout=N} but that may not be the actual problem. You would probably have to look at a trace with debug level etc to be sure. On Mon, Jan 31, 2011 at 2:07 PM, Jerry Richards <jerry.richards@teotech.com> wrote:

...

Hello All,

When I set the call_timeout variable (to 12 seconds) before a bridge to an outbound PSTN call, Freeswitch always rings for 24 seconds before going to voicemail. If I call an internal extension, then it uses the call_timeout variable. Do you know why it would do this?

Here is the line that sets the call_timeout variable:

switch_channel_set_variable(channel, "call_timeout", "6");

Here is the PSTN bridge log:

EXECUTE sofia/internal/2003@192.168.72.79:5060 bridge({presence_id=2001@192.168.72.79}[lcr_carrier=Carrier / Location 1/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]openzap/smg_p rid/a/15248142341@g1)

Thanks,

Jerry

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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _______________________________________________ FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org

Anthony Minessale

6:34 p.m.

New subject: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value

set bridge_answer_timeout on the A leg On Mon, Feb 14, 2011 at 12:24 PM, Jerry Richards <jerry.richards@teotech.com> wrote:

...

I found out why my call_timeout was not kicking in when I call a PSTN number. It is because of early media. Is there a way to play early media and also have the call_timeout kick-in according to its setting?

Thanks, Jerry

-----Original Message----- From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, January 31, 2011 12:13 PM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value

You should use {originate_timeout=N} but that may not be the actual problem.

You would probably have to look at a trace with debug level etc to be sure.

On Mon, Jan 31, 2011 at 2:07 PM, Jerry Richards <jerry.richards@teotech.com> wrote:

...
Hello All,

When I set the call_timeout variable (to 12 seconds) before a bridge to an outbound PSTN call, Freeswitch always rings for 24 seconds before going to voicemail. If I call an internal extension, then it uses the call_timeout variable. Do you know why it would do this?

Here is the line that sets the call_timeout variable:

switch_channel_set_variable(channel, "call_timeout", "6");

Here is the PSTN bridge log:

EXECUTE sofia/internal/2003@192.168.72.79:5060 bridge({presence_id=2001@192.168.72.79}[lcr_carrier=Carrier / Location 1/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]openzap/smg_p rid/a/15248142341@g1)

Thanks,

Jerry

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-- Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch

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Jerry Richards

9:44 p.m.

New subject: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value

Anthony, I have more than one XML bridge statement with continue_on_fail set true. In the case of early media and the bridge_answer_timeout, it looks like it does not continue and attempt the next bridge statement in the XML dialplan. True? If so, is there a way to make Freeswitch continue with the next bridge statement in this case? Thanks, Jerry -----Original Message----- From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, February 14, 2011 10:35 AM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value set bridge_answer_timeout on the A leg On Mon, Feb 14, 2011 at 12:24 PM, Jerry Richards <jerry.richards@teotech.com> wrote:

...

I found out why my call_timeout was not kicking in when I call a PSTN number. It is because of early media. Is there a way to play early media and also have the call_timeout kick-in according to its setting?

Thanks, Jerry

-----Original Message----- From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, January 31, 2011 12:13 PM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value

You should use {originate_timeout=N} but that may not be the actual problem.

You would probably have to look at a trace with debug level etc to be sure.

On Mon, Jan 31, 2011 at 2:07 PM, Jerry Richards <jerry.richards@teotech.com> wrote:

...
Hello All,

When I set the call_timeout variable (to 12 seconds) before a bridge to an outbound PSTN call, Freeswitch always rings for 24 seconds before going to voicemail. If I call an internal extension, then it uses the call_timeout variable. Do you know why it would do this?

Here is the line that sets the call_timeout variable:

switch_channel_set_variable(channel, "call_timeout", "6");

Here is the PSTN bridge log:

EXECUTE sofia/internal/2003@192.168.72.79:5060 bridge({presence_id=2001@192.168.72.79}[lcr_carrier=Carrier / Location 1/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]openzap/smg_ p rid/a/15248142341@g1)

Thanks,

Jerry

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-- Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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Anthony Minessale

9:51 p.m.

New subject: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value

you are better off using ignore_early_media=true so the originates fail. When you allow early media, getting early media counts as success. When you are chaining many like that you can use artificial ringbak by setting the variable ringback to ${us-ring} on the A leg. On Mon, Feb 14, 2011 at 3:44 PM, Jerry Richards <jerry.richards@teotech.com> wrote:

...

Anthony,

I have more than one XML bridge statement with continue_on_fail set true. In the case of early media and the bridge_answer_timeout, it looks like it does not continue and attempt the next bridge statement in the XML dialplan. True? If so, is there a way to make Freeswitch continue with the next bridge statement in this case?

Thanks, Jerry

-----Original Message----- From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, February 14, 2011 10:35 AM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value

set bridge_answer_timeout on the A leg

On Mon, Feb 14, 2011 at 12:24 PM, Jerry Richards <jerry.richards@teotech.com> wrote:

...
I found out why my call_timeout was not kicking in when I call a PSTN number. It is because of early media. Is there a way to play early media and also have the call_timeout kick-in according to its setting?

Thanks, Jerry

-----Original Message----- From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, January 31, 2011 12:13 PM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value

You should use {originate_timeout=N} but that may not be the actual problem.

You would probably have to look at a trace with debug level etc to be sure.

On Mon, Jan 31, 2011 at 2:07 PM, Jerry Richards <jerry.richards@teotech.com> wrote:

...
Hello All,

When I set the call_timeout variable (to 12 seconds) before a bridge to an outbound PSTN call, Freeswitch always rings for 24 seconds before going to voicemail. If I call an internal extension, then it uses the call_timeout variable. Do you know why it would do this?

Here is the line that sets the call_timeout variable:

switch_channel_set_variable(channel, "call_timeout", "6");

Here is the PSTN bridge log:

EXECUTE sofia/internal/2003@192.168.72.79:5060 bridge({presence_id=2001@192.168.72.79}[lcr_carrier=Carrier / Location 1/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]openzap/smg_ p rid/a/15248142341@g1)

Thanks,

Jerry

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-- Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch

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Michael Collins

15 Feb 15 Feb

1:20 a.m.

New subject: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value

Or if you're truly masochistic you can try the monitor_early_media_fail_xxx chan vars. :) -MC On Mon, Feb 14, 2011 at 3:51 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:

...

you are better off using ignore_early_media=true so the originates fail. When you allow early media, getting early media counts as success. When you are chaining many like that you can use artificial ringbak by setting the variable ringback to ${us-ring} on the A leg.

Jerry Richards

6:57 p.m.

New subject: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value

I tried setting ignore_early_media=true and ringback=${us-ring} but I still do not hear artificial ringback (see http://pastebin.freeswitch.org/15392). I also tried enable and disable of bypass-media but it didn't make any difference. Do you see what could be wrong? Thanks, Jerry -----Original Message----- From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, February 14, 2011 1:51 PM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value you are better off using ignore_early_media=true so the originates fail. When you allow early media, getting early media counts as success. When you are chaining many like that you can use artificial ringbak by setting the variable ringback to ${us-ring} on the A leg. On Mon, Feb 14, 2011 at 3:44 PM, Jerry Richards <jerry.richards@teotech.com> wrote:

...

Anthony,

I have more than one XML bridge statement with continue_on_fail set true. In the case of early media and the bridge_answer_timeout, it looks like it does not continue and attempt the next bridge statement in the XML dialplan. True? If so, is there a way to make Freeswitch continue with the next bridge statement in this case?

Thanks, Jerry

-----Original Message----- From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, February 14, 2011 10:35 AM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value

set bridge_answer_timeout on the A leg

On Mon, Feb 14, 2011 at 12:24 PM, Jerry Richards <jerry.richards@teotech.com> wrote:

...
I found out why my call_timeout was not kicking in when I call a PSTN number. It is because of early media. Is there a way to play early media and also have the call_timeout kick-in according to its setting?

Thanks, Jerry

-----Original Message----- From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, January 31, 2011 12:13 PM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value

You should use {originate_timeout=N} but that may not be the actual problem.

You would probably have to look at a trace with debug level etc to be sure.

On Mon, Jan 31, 2011 at 2:07 PM, Jerry Richards <jerry.richards@teotech.com> wrote:

...
Hello All,

When I set the call_timeout variable (to 12 seconds) before a bridge to an outbound PSTN call, Freeswitch always rings for 24 seconds before going to voicemail. If I call an internal extension, then it uses the call_timeout variable. Do you know why it would do this?

Here is the line that sets the call_timeout variable:

switch_channel_set_variable(channel, "call_timeout", "6");

Here is the PSTN bridge log:

EXECUTE sofia/internal/2003@192.168.72.79:5060 bridge({presence_id=2001@192.168.72.79}[lcr_carrier=Carrier / Location 1/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]openzap/smg _ p rid/a/15248142341@g1)

Thanks,

Jerry

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-- Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch

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AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch

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Anthony Minessale

8:50 p.m.

New subject: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value

outbound leg is openzap? try putting the instant_ringback=true in the {} along with ignore_early_media On Tue, Feb 15, 2011 at 12:57 PM, Jerry Richards <jerry.richards@teotech.com> wrote:

...

I tried setting ignore_early_media=true and ringback=${us-ring} but I still do not hear artificial ringback (see http://pastebin.freeswitch.org/15392). I also tried enable and disable of bypass-media but it didn't make any difference. Do you see what could be wrong?

Thanks, Jerry

-----Original Message----- From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, February 14, 2011 1:51 PM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value

you are better off using ignore_early_media=true so the originates fail. When you allow early media, getting early media counts as success. When you are chaining many like that you can use artificial ringbak by setting the variable ringback to ${us-ring} on the A leg.

On Mon, Feb 14, 2011 at 3:44 PM, Jerry Richards <jerry.richards@teotech.com> wrote:

...
Anthony,

I have more than one XML bridge statement with continue_on_fail set true. In the case of early media and the bridge_answer_timeout, it looks like it does not continue and attempt the next bridge statement in the XML dialplan. True? If so, is there a way to make Freeswitch continue with the next bridge statement in this case?

Thanks, Jerry

-----Original Message----- From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, February 14, 2011 10:35 AM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value

set bridge_answer_timeout on the A leg

On Mon, Feb 14, 2011 at 12:24 PM, Jerry Richards <jerry.richards@teotech.com> wrote:

...
I found out why my call_timeout was not kicking in when I call a PSTN number. It is because of early media. Is there a way to play early media and also have the call_timeout kick-in according to its setting?

Thanks, Jerry

-----Original Message----- From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, January 31, 2011 12:13 PM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value

You should use {originate_timeout=N} but that may not be the actual problem.

You would probably have to look at a trace with debug level etc to be sure.

On Mon, Jan 31, 2011 at 2:07 PM, Jerry Richards <jerry.richards@teotech.com> wrote:

...
Hello All,

When I set the call_timeout variable (to 12 seconds) before a bridge to an outbound PSTN call, Freeswitch always rings for 24 seconds before going to voicemail. If I call an internal extension, then it uses the call_timeout variable. Do you know why it would do this?

Here is the line that sets the call_timeout variable:

switch_channel_set_variable(channel, "call_timeout", "6");

Here is the PSTN bridge log:

EXECUTE sofia/internal/2003@192.168.72.79:5060 bridge({presence_id=2001@192.168.72.79}[lcr_carrier=Carrier / Location 1/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]openzap/smg _ p rid/a/15248142341@g1)

Thanks,

Jerry

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Michael Collins

9:01 p.m.

New subject: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value

Pastebin your relevant configs and a console log including SIP trace. I'm sure there's a clue in there somewhere. -MC On Tue, Feb 15, 2011 at 12:57 PM, Jerry Richards <jerry.richards@teotech.com

...

wrote:

...

I tried setting ignore_early_media=true and ringback=${us-ring} but I still do not hear artificial ringback (see http://pastebin.freeswitch.org/15392). I also tried enable and disable of bypass-media but it didn't make any difference. Do you see what could be wrong?

Thanks, Jerry

-----Original Message----- From: freeswitch-dev-bounces@lists.freeswitch.org [mailto: freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, February 14, 2011 1:51 PM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value

you are better off using ignore_early_media=true so the originates fail. When you allow early media, getting early media counts as success. When you are chaining many like that you can use artificial ringbak by setting the variable ringback to ${us-ring} on the A leg.

On Mon, Feb 14, 2011 at 3:44 PM, Jerry Richards < jerry.richards@teotech.com> wrote:

...
Anthony,

I have more than one XML bridge statement with continue_on_fail set true. In the case of early media and the bridge_answer_timeout, it looks like it does not continue and attempt the next bridge statement in the XML dialplan. True? If so, is there a way to make Freeswitch continue with the next bridge statement in this case?

Thanks, Jerry

-----Original Message----- From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, February 14, 2011 10:35 AM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value

set bridge_answer_timeout on the A leg

On Mon, Feb 14, 2011 at 12:24 PM, Jerry Richards < jerry.richards@teotech.com> wrote:

...
I found out why my call_timeout was not kicking in when I call a PSTN number. It is because of early media. Is there a way to play early media and also have the call_timeout kick-in according to its setting?

Thanks, Jerry

-----Original Message----- From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, January 31, 2011 12:13 PM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value

You should use {originate_timeout=N} but that may not be the actual problem.

You would probably have to look at a trace with debug level etc to be sure.

On Mon, Jan 31, 2011 at 2:07 PM, Jerry Richards < jerry.richards@teotech.com> wrote:

...
Hello All,

When I set the call_timeout variable (to 12 seconds) before a bridge to an outbound PSTN call, Freeswitch always rings for 24 seconds before going to voicemail. If I call an internal extension, then it uses the call_timeout variable. Do you know why it would do this?

Here is the line that sets the call_timeout variable:

switch_channel_set_variable(channel, "call_timeout", "6");

Here is the PSTN bridge log:

EXECUTE sofia/internal/2003@192.168.72.79:5060 bridge({presence_id=2001@192.168.72.79}[lcr_carrier=Carrier / Location 1/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]openzap/smg _ p rid/a/15248142341@g1)

Thanks,

Jerry

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Anthony Minessale
Jerry Richards
Michael Collins