From anthony.minessale at gmail.com  Tue Feb  1 01:15:35 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 31 Jan 2011 16:15:35 -0600
Subject: [Freeswitch-dev] freeswitch.com is returned to us!
Message-ID: <AANLkTi=BciYcFugPySe3QU7pK-=avKSpkOUiq119UCYP@mail.gmail.com>

http://www.freeswitch.com/

Thank you everyone who helped with this!



-- 
Anthony Minessale II

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

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

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


From msc at freeswitch.org  Tue Feb  1 02:54:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Jan 2011 15:54:38 -0800
Subject: [Freeswitch-dev] Developer Dinner Coming Up - Help Feed The
	FreeSWITCH Team!
Message-ID: <AANLkTimZjT2SQ=Go10JwLrXJEmjQt+iNZQrUKVHPBgj=@mail.gmail.com>

Well, it's that time again. The FreeSWITCH developers are gathering at a
secret location in Wisconsin next week and the community is invited to
participate in this year's "buy the devs dinner" event.

How to do it: Go to freeswitch.org and click on the Paypal link. Toss a few
bucks in the hat and send along your bon apetit message. If for some reason
you cannot use Paypal but still wish to participate then by all means call
Brian West on his SIP phone: 2000 at bkw.org. (Hint: you  can dial 9191 if you
have a default FreeSWITCH config.) Brian is well-prepared to take your
money. :)

To those who have donated recently we say, "Thank you!" If you wish to
donate again specifically for this occasion you are more than welcome to do
so. In any case, thank you in advance for your generosity. You are great
community members and the FreeSWITCH team is absolutely glad to have you
with us!

-Michael Collins
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From chat2jesse at gmail.com  Tue Feb  1 03:13:45 2011
From: chat2jesse at gmail.com (jesse)
Date: Mon, 31 Jan 2011 16:13:45 -0800
Subject: [Freeswitch-dev] Developer Dinner Coming Up - Help Feed The
 FreeSWITCH Team!
In-Reply-To: <AANLkTimZjT2SQ=Go10JwLrXJEmjQt+iNZQrUKVHPBgj=@mail.gmail.com>
References: <AANLkTimZjT2SQ=Go10JwLrXJEmjQt+iNZQrUKVHPBgj=@mail.gmail.com>
Message-ID: <AANLkTimFF8TqsyesoX4iPD=+jyw9QCo8N7yrKkNezEbe@mail.gmail.com>

- my paypal donation $20 confirmation number: 6T841976KC483020G, enjoy
dinner!
- also purchased the book from amazon.

keep the good work. just one minor request, could you guys please fix the
mail list issue. my question was posted roughly 20 hours ago on yesterday.
It still has not appeared in my inbox.

-jesse

On Mon, Jan 31, 2011 at 3:54 PM, Michael Collins <msc at freeswitch.org> wrote:

> Well, it's that time again. The FreeSWITCH developers are gathering at a
> secret location in Wisconsin next week and the community is invited to
> participate in this year's "buy the devs dinner" event.
>
> How to do it: Go to freeswitch.org and click on the Paypal link. Toss a
> few bucks in the hat and send along your bon apetit message. If for some
> reason you cannot use Paypal but still wish to participate then by all means
> call Brian West on his SIP phone: 2000 at bkw.org. (Hint: you  can dial 9191
> if you have a default FreeSWITCH config.) Brian is well-prepared to take
> your money. :)
>
> To those who have donated recently we say, "Thank you!" If you wish to
> donate again specifically for this occasion you are more than welcome to do
> so. In any case, thank you in advance for your generosity. You are great
> community members and the FreeSWITCH team is absolutely glad to have you
> with us!
>
> -Michael Collins
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Feb  1 03:23:45 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Jan 2011 16:23:45 -0800
Subject: [Freeswitch-dev] Developer Dinner Coming Up - Help Feed The
 FreeSWITCH Team!
In-Reply-To: <AANLkTimFF8TqsyesoX4iPD=+jyw9QCo8N7yrKkNezEbe@mail.gmail.com>
References: <AANLkTimZjT2SQ=Go10JwLrXJEmjQt+iNZQrUKVHPBgj=@mail.gmail.com>
	<AANLkTimFF8TqsyesoX4iPD=+jyw9QCo8N7yrKkNezEbe@mail.gmail.com>
Message-ID: <AANLkTimWrrjijgVtiaMDc6orZorFjBLp=xyjmngHX4Y9@mail.gmail.com>

Thanks!

As far as the email issue we'll check it out. I'll make sure that you're not
being moderated or something silly like that.
-MC

On Mon, Jan 31, 2011 at 4:13 PM, jesse <chat2jesse at gmail.com> wrote:

> - my paypal donation $20 confirmation number: 6T841976KC483020G, enjoy
> dinner!
> - also purchased the book from amazon.
>
> keep the good work. just one minor request, could you guys please fix the
> mail list issue. my question was posted roughly 20 hours ago on yesterday.
> It still has not appeared in my inbox.
>
> -jesse
>
> On Mon, Jan 31, 2011 at 3:54 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Well, it's that time again. The FreeSWITCH developers are gathering at a
>> secret location in Wisconsin next week and the community is invited to
>> participate in this year's "buy the devs dinner" event.
>>
>> How to do it: Go to freeswitch.org and click on the Paypal link. Toss a
>> few bucks in the hat and send along your bon apetit message. If for some
>> reason you cannot use Paypal but still wish to participate then by all means
>> call Brian West on his SIP phone: 2000 at bkw.org. (Hint: you  can dial 9191
>> if you have a default FreeSWITCH config.) Brian is well-prepared to take
>> your money. :)
>>
>> To those who have donated recently we say, "Thank you!" If you wish to
>> donate again specifically for this occasion you are more than welcome to do
>> so. In any case, thank you in advance for your generosity. You are great
>> community members and the FreeSWITCH team is absolutely glad to have you
>> with us!
>>
>> -Michael Collins
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From kevin.snow at ooma.com  Wed Feb  2 22:39:07 2011
From: kevin.snow at ooma.com (Kevin Snow)
Date: Wed, 02 Feb 2011 11:39:07 -0800
Subject: [Freeswitch-dev] Switch_core_get_variable
Message-ID: <C96EF2DB.C075%kevin.snow@ooma.com>


I found a hole in the way we handle core variables in FS.
Switch_core_get_variable does the lookup and returns the found pointer. In
Switch_core_set_variable the first step it does is look up the variable and
free it (if it exists). This would free it out from under another that has
just done a get on it. This is how I stumbled on this.

Is the right fix is to add a switch_core_get_variable_dup that dups the
string while in the mutex protection? I realize in the core case this will
require the caller to then free the returned memory, but this is better than
getting a bad pointer. This is analogous to the switch_channel_get_variable
and it?s _dup implementation, although it dups it to session memory.

Switch_channel_get_variable?s ability to peak through to the core variables
is susceptible to this. If a core variable is changed after
switch_channel_get_variable looks it up but before it dups to the session
pool, it?ll have a bad pointer.

Kevin Snow
Ooma, Inc
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From steveayre at gmail.com  Wed Feb  2 23:01:13 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 2 Feb 2011 20:01:13 +0000
Subject: [Freeswitch-dev] Switch_core_get_variable
In-Reply-To: <C96EF2DB.C075%kevin.snow@ooma.com>
References: <C96EF2DB.C075%kevin.snow@ooma.com>
Message-ID: <AANLkTi=ZJm6EM2KXM9snLkXiCCFbrKSOdV=bsBD-6sH6@mail.gmail.com>

>
> I found a hole in the way we handle core variables in FS.
> Switch_core_get_variable does the lookup and returns the found pointer. In
> Switch_core_set_variable the first step it does is look up the variable and
> free it (if it exists). This would free it out from under another that has
> just done a get on it. This is how I stumbled on this.


The correct place to report bugs is the bug tracker:
http://jira.freeswitch.org/

It's a known problem. Basically the answer is you shouldn't really be
changing core variables after FS has loaded, at least not while handling
traffic. If you're doing so as part of your application, you're doing it
wrong. mod_db is probably better suited to what you want to do.

Is the right fix is to add a switch_core_get_variable_dup that dups the
> string while in the mutex protection? I realize in the core case this will
> require the caller to then free the returned memory, but this is better than
> getting a bad pointer. This is analogous to the switch_channel_get_variable
> and it?s _dup implementation, although it dups it to session memory.


Just my 2 cents (I'm not a core developer)... That could be done. But it's
also slower because it means dynamically assigning memory and then copying
the string. And it can't be cached. And the mutex will make all calls block,
not just the single call as with the channel variable - which might possibly
introduce a risk of deadlock.

-Steve



On 2 February 2011 19:39, Kevin Snow <kevin.snow at ooma.com> wrote:

>
> I found a hole in the way we handle core variables in FS.
> Switch_core_get_variable does the lookup and returns the found pointer. In
> Switch_core_set_variable the first step it does is look up the variable and
> free it (if it exists). This would free it out from under another that has
> just done a get on it. This is how I stumbled on this.
>
> Is the right fix is to add a switch_core_get_variable_dup that dups the
> string while in the mutex protection? I realize in the core case this will
> require the caller to then free the returned memory, but this is better than
> getting a bad pointer. This is analogous to the switch_channel_get_variable
> and it?s _dup implementation, although it dups it to session memory.
>
> Switch_channel_get_variable?s ability to peak through to the core variables
> is susceptible to this. If a core variable is changed after
> switch_channel_get_variable looks it up but before it dups to the session
> pool, it?ll have a bad pointer.
>
> Kevin Snow
> Ooma, Inc
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From kevin.snow at ooma.com  Thu Feb  3 00:48:40 2011
From: kevin.snow at ooma.com (Kevin Snow)
Date: Wed, 02 Feb 2011 13:48:40 -0800
Subject: [Freeswitch-dev] Switch_core_get_variable
In-Reply-To: <AANLkTi=ZJm6EM2KXM9snLkXiCCFbrKSOdV=bsBD-6sH6@mail.gmail.com>
Message-ID: <C96F1138.C08F%kevin.snow@ooma.com>


I can enter it into jira.

I?m not sure if I agree with your assessment. There is already a mutex
lock/unlock around switch_core_get_variable?s internals today, no additional
risk of deadlock. As for switch_channel_get_variable, which also has a mutex
of it?s own, is already using switch_core_session_strdup before returning.
I?m proposing the equivalent to switch_channel_get_variable_dup be added for
switch_core_get_variable. To me it seems we must have this if you want to
reliably read a variable that might change at runtime.

Kevin




On 2/2/11 12:01 PM, "Steven Ayre" <steveayre at gmail.com> wrote:

>> I found a hole in the way we handle core variables in FS.
>> Switch_core_get_variable does the lookup and returns the found pointer. In
>> Switch_core_set_variable the first step it does is look up the variable and
>> free it (if it exists). This would free it out from under another that has
>> just done a get on it. This is how I stumbled on this.?
> 
> The correct place to report bugs is the bug
> tracker:?http://jira.freeswitch.org/
> 
> It's a known problem. Basically the answer is you shouldn't really be changing
> core variables after FS has loaded, at least not while handling traffic. If
> you're doing so as part of your application, you're doing it wrong. mod_db is
> probably better suited to what you want to do.
> 
>> Is the right fix is to add a switch_core_get_variable_dup that dups the
>> string while in the mutex protection? I realize in the core case this will
>> require the caller to then free the returned memory, but this is better than
>> getting a bad pointer. This is analogous to the switch_channel_get_variable
>> and it?s _dup implementation, although it dups it to session memory.
> 
> Just my 2 cents (I'm not a core developer)... That could be done. But it's
> also slower because it means dynamically assigning memory and then copying the
> string. And it can't be cached. And the mutex will make all calls block, not
> just the single call as with the channel variable - which might possibly
> introduce a risk of deadlock.
> 
> -Steve
> 
> 
> 
> On 2 February 2011 19:39, Kevin Snow <kevin.snow at ooma.com> wrote:
>> 
>> I found a hole in the way we handle core variables in FS.
>> Switch_core_get_variable does the lookup and returns the found pointer. In
>> Switch_core_set_variable the first step it does is look up the variable and
>> free it (if it exists). This would free it out from under another that has
>> just done a get on it. This is how I stumbled on this.
>> 
>> Is the right fix is to add a switch_core_get_variable_dup that dups the
>> string while in the mutex protection? I realize in the core case this will
>> require the caller to then free the returned memory, but this is better than
>> getting a bad pointer. This is analogous to the switch_channel_get_variable
>> and it?s _dup implementation, although it dups it to session memory.
>> 
>> Switch_channel_get_variable?s ability to peak through to the core variables
>> is susceptible to this. If a core variable is changed after
>> switch_channel_get_variable looks it up but before it dups to the session
>> pool, it?ll have a bad pointer.
>> 
>> Kevin Snow
>> Ooma, Inc 
>> 
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org


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From anthony.minessale at gmail.com  Thu Feb  3 00:57:32 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 2 Feb 2011 15:57:32 -0600
Subject: [Freeswitch-dev] Switch_core_get_variable
In-Reply-To: <C96F1138.C08F%kevin.snow@ooma.com>
References: <AANLkTi=ZJm6EM2KXM9snLkXiCCFbrKSOdV=bsBD-6sH6@mail.gmail.com>
	<C96F1138.C08F%kevin.snow@ooma.com>
Message-ID: <AANLkTi=X+a_cqe2dP8QbM8af3y8JaZLXMpafH-N_pY+Z@mail.gmail.com>

I read this email then I went and solved it.
I see some replies came in.

Leaning on global vars is a bit of abuse but we can't afford to be
susceptible to a crash.
I have modified 26 files and changed all places to use one of the
following new functions:

char *switch_core_get_variable_dup(const char *var)
char *switch_core_get_variable_pdup(const char *var, switch_memory_pool_t *pool)

one strdups the value, the other pool dups it.

All that I ask is that you now review it to make sure I did not mess
up anything.



On Wed, Feb 2, 2011 at 3:48 PM, Kevin Snow <kevin.snow at ooma.com> wrote:
>
> I can enter it into jira.
>
> I?m not sure if I agree with your assessment. There is already a mutex
> lock/unlock around switch_core_get_variable?s internals today, no additional
> risk of deadlock. As for switch_channel_get_variable, which also has a mutex
> of it?s own, is already using switch_core_session_strdup before returning.
> I?m proposing the equivalent to switch_channel_get_variable_dup be added for
> switch_core_get_variable. To me it seems we must have this if you want to
> reliably read a variable that might change at runtime.
>
> Kevin
>
>
>
>
> On 2/2/11 12:01 PM, "Steven Ayre" <steveayre at gmail.com> wrote:
>
> I found a hole in the way we handle core variables in FS.
> Switch_core_get_variable does the lookup and returns the found pointer. In
> Switch_core_set_variable the first step it does is look up the variable and
> free it (if it exists). This would free it out from under another that has
> just done a get on it. This is how I stumbled on this.
>
> The correct place to report bugs is the bug
> tracker:?http://jira.freeswitch.org/
>
> It's a known problem. Basically the answer is you shouldn't really be
> changing core variables after FS has loaded, at least not while handling
> traffic. If you're doing so as part of your application, you're doing it
> wrong. mod_db is probably better suited to what you want to do.
>
> Is the right fix is to add a switch_core_get_variable_dup that dups the
> string while in the mutex protection? I realize in the core case this will
> require the caller to then free the returned memory, but this is better than
> getting a bad pointer. This is analogous to the switch_channel_get_variable
> and it?s _dup implementation, although it dups it to session memory.
>
> Just my 2 cents (I'm not a core developer)... That could be done. But it's
> also slower because it means dynamically assigning memory and then copying
> the string. And it can't be cached. And the mutex will make all calls block,
> not just the single call as with the channel variable - which might possibly
> introduce a risk of deadlock.
>
> -Steve
>
>
>
> On 2 February 2011 19:39, Kevin Snow <kevin.snow at ooma.com> wrote:
>
> I found a hole in the way we handle core variables in FS.
> Switch_core_get_variable does the lookup and returns the found pointer. In
> Switch_core_set_variable the first step it does is look up the variable and
> free it (if it exists). This would free it out from under another that has
> just done a get on it. This is how I stumbled on this.
>
> Is the right fix is to add a switch_core_get_variable_dup that dups the
> string while in the mutex protection? I realize in the core case this will
> require the caller to then free the returned memory, but this is better than
> getting a bad pointer. This is analogous to the switch_channel_get_variable
> and it?s _dup implementation, although it dups it to session memory.
>
> Switch_channel_get_variable?s ability to peak through to the core variables
> is susceptible to this. If a core variable is changed after
> switch_channel_get_variable looks it up but before it dups to the session
> pool, it?ll have a bad pointer.
>
> Kevin Snow
> Ooma, Inc
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
>
> ________________________________
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at 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 II

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

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

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


From kevin.snow at ooma.com  Thu Feb  3 01:04:15 2011
From: kevin.snow at ooma.com (Kevin Snow)
Date: Wed, 02 Feb 2011 14:04:15 -0800
Subject: [Freeswitch-dev] Switch_core_get_variable
In-Reply-To: <AANLkTi=X+a_cqe2dP8QbM8af3y8JaZLXMpafH-N_pY+Z@mail.gmail.com>
Message-ID: <C96F14DF.C098%kevin.snow@ooma.com>

Anthony - You're awesome!

I'll sync up and take a look.

Thanks!

Kevin



On 2/2/11 1:57 PM, "Anthony Minessale" <anthony.minessale at gmail.com> wrote:

> I read this email then I went and solved it.
> I see some replies came in.
> 
> Leaning on global vars is a bit of abuse but we can't afford to be
> susceptible to a crash.
> I have modified 26 files and changed all places to use one of the
> following new functions:
> 
> char *switch_core_get_variable_dup(const char *var)
> char *switch_core_get_variable_pdup(const char *var, switch_memory_pool_t
> *pool)
> 
> one strdups the value, the other pool dups it.
> 
> All that I ask is that you now review it to make sure I did not mess
> up anything.
> 
> 
> 
> On Wed, Feb 2, 2011 at 3:48 PM, Kevin Snow <kevin.snow at ooma.com> wrote:
>> 
>> I can enter it into jira.
>> 
>> I?m not sure if I agree with your assessment. There is already a mutex
>> lock/unlock around switch_core_get_variable?s internals today, no additional
>> risk of deadlock. As for switch_channel_get_variable, which also has a mutex
>> of it?s own, is already using switch_core_session_strdup before returning.
>> I?m proposing the equivalent to switch_channel_get_variable_dup be added for
>> switch_core_get_variable. To me it seems we must have this if you want to
>> reliably read a variable that might change at runtime.
>> 
>> Kevin
>> 
>> 
>> 
>> 
>> On 2/2/11 12:01 PM, "Steven Ayre" <steveayre at gmail.com> wrote:
>> 
>> I found a hole in the way we handle core variables in FS.
>> Switch_core_get_variable does the lookup and returns the found pointer. In
>> Switch_core_set_variable the first step it does is look up the variable and
>> free it (if it exists). This would free it out from under another that has
>> just done a get on it. This is how I stumbled on this.
>> 
>> The correct place to report bugs is the bug
>> tracker:?http://jira.freeswitch.org/
>> 
>> It's a known problem. Basically the answer is you shouldn't really be
>> changing core variables after FS has loaded, at least not while handling
>> traffic. If you're doing so as part of your application, you're doing it
>> wrong. mod_db is probably better suited to what you want to do.
>> 
>> Is the right fix is to add a switch_core_get_variable_dup that dups the
>> string while in the mutex protection? I realize in the core case this will
>> require the caller to then free the returned memory, but this is better than
>> getting a bad pointer. This is analogous to the switch_channel_get_variable
>> and it?s _dup implementation, although it dups it to session memory.
>> 
>> Just my 2 cents (I'm not a core developer)... That could be done. But it's
>> also slower because it means dynamically assigning memory and then copying
>> the string. And it can't be cached. And the mutex will make all calls block,
>> not just the single call as with the channel variable - which might possibly
>> introduce a risk of deadlock.
>> 
>> -Steve
>> 
>> 
>> 
>> On 2 February 2011 19:39, Kevin Snow <kevin.snow at ooma.com> wrote:
>> 
>> I found a hole in the way we handle core variables in FS.
>> Switch_core_get_variable does the lookup and returns the found pointer. In
>> Switch_core_set_variable the first step it does is look up the variable and
>> free it (if it exists). This would free it out from under another that has
>> just done a get on it. This is how I stumbled on this.
>> 
>> Is the right fix is to add a switch_core_get_variable_dup that dups the
>> string while in the mutex protection? I realize in the core case this will
>> require the caller to then free the returned memory, but this is better than
>> getting a bad pointer. This is analogous to the switch_channel_get_variable
>> and it?s _dup implementation, although it dups it to session memory.
>> 
>> Switch_channel_get_variable?s ability to peak through to the core variables
>> is susceptible to this. If a core variable is changed after
>> switch_channel_get_variable looks it up but before it dups to the session
>> pool, it?ll have a bad pointer.
>> 
>> Kevin Snow
>> Ooma, Inc
>> 
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>> 
>> 
>> 
>> ________________________________
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>> 
>> 
> 
> 



From kevin.snow at ooma.com  Thu Feb  3 03:54:45 2011
From: kevin.snow at ooma.com (Kevin Snow)
Date: Wed, 02 Feb 2011 16:54:45 -0800
Subject: [Freeswitch-dev] Switch_core_get_variable
In-Reply-To: <AANLkTi=X+a_cqe2dP8QbM8af3y8JaZLXMpafH-N_pY+Z@mail.gmail.com>
Message-ID: <C96F3CD5.C0A3%kevin.snow@ooma.com>


I haven't merged it into our system yet but I did download and review the
change. Looks good.

I'll let you know if I have issues.

Thanks again.

Kevin





On 2/2/11 1:57 PM, "Anthony Minessale" <anthony.minessale at gmail.com> wrote:

> I read this email then I went and solved it.
> I see some replies came in.
> 
> Leaning on global vars is a bit of abuse but we can't afford to be
> susceptible to a crash.
> I have modified 26 files and changed all places to use one of the
> following new functions:
> 
> char *switch_core_get_variable_dup(const char *var)
> char *switch_core_get_variable_pdup(const char *var, switch_memory_pool_t
> *pool)
> 
> one strdups the value, the other pool dups it.
> 
> All that I ask is that you now review it to make sure I did not mess
> up anything.
> 
> 
> 
> On Wed, Feb 2, 2011 at 3:48 PM, Kevin Snow <kevin.snow at ooma.com> wrote:
>> 
>> I can enter it into jira.
>> 
>> I?m not sure if I agree with your assessment. There is already a mutex
>> lock/unlock around switch_core_get_variable?s internals today, no additional
>> risk of deadlock. As for switch_channel_get_variable, which also has a mutex
>> of it?s own, is already using switch_core_session_strdup before returning.
>> I?m proposing the equivalent to switch_channel_get_variable_dup be added for
>> switch_core_get_variable. To me it seems we must have this if you want to
>> reliably read a variable that might change at runtime.
>> 
>> Kevin
>> 
>> 
>> 
>> 
>> On 2/2/11 12:01 PM, "Steven Ayre" <steveayre at gmail.com> wrote:
>> 
>> I found a hole in the way we handle core variables in FS.
>> Switch_core_get_variable does the lookup and returns the found pointer. In
>> Switch_core_set_variable the first step it does is look up the variable and
>> free it (if it exists). This would free it out from under another that has
>> just done a get on it. This is how I stumbled on this.
>> 
>> The correct place to report bugs is the bug
>> tracker:?http://jira.freeswitch.org/
>> 
>> It's a known problem. Basically the answer is you shouldn't really be
>> changing core variables after FS has loaded, at least not while handling
>> traffic. If you're doing so as part of your application, you're doing it
>> wrong. mod_db is probably better suited to what you want to do.
>> 
>> Is the right fix is to add a switch_core_get_variable_dup that dups the
>> string while in the mutex protection? I realize in the core case this will
>> require the caller to then free the returned memory, but this is better than
>> getting a bad pointer. This is analogous to the switch_channel_get_variable
>> and it?s _dup implementation, although it dups it to session memory.
>> 
>> Just my 2 cents (I'm not a core developer)... That could be done. But it's
>> also slower because it means dynamically assigning memory and then copying
>> the string. And it can't be cached. And the mutex will make all calls block,
>> not just the single call as with the channel variable - which might possibly
>> introduce a risk of deadlock.
>> 
>> -Steve
>> 
>> 
>> 
>> On 2 February 2011 19:39, Kevin Snow <kevin.snow at ooma.com> wrote:
>> 
>> I found a hole in the way we handle core variables in FS.
>> Switch_core_get_variable does the lookup and returns the found pointer. In
>> Switch_core_set_variable the first step it does is look up the variable and
>> free it (if it exists). This would free it out from under another that has
>> just done a get on it. This is how I stumbled on this.
>> 
>> Is the right fix is to add a switch_core_get_variable_dup that dups the
>> string while in the mutex protection? I realize in the core case this will
>> require the caller to then free the returned memory, but this is better than
>> getting a bad pointer. This is analogous to the switch_channel_get_variable
>> and it?s _dup implementation, although it dups it to session memory.
>> 
>> Switch_channel_get_variable?s ability to peak through to the core variables
>> is susceptible to this. If a core variable is changed after
>> switch_channel_get_variable looks it up but before it dups to the session
>> pool, it?ll have a bad pointer.
>> 
>> Kevin Snow
>> Ooma, Inc
>> 
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>> 
>> 
>> 
>> ________________________________
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>> 
>> 
> 
> 



From anthony.minessale at gmail.com  Fri Feb  4 00:37:00 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 3 Feb 2011 15:37:00 -0600
Subject: [Freeswitch-dev] ANNOUNCEMENT CudaTel IP PBX (powered by
	FreeSWITCH) Reaches Critical Mass
Message-ID: <AANLkTin7xCfqk1kyyyQMW-VZQo2t9MJ9c++_Zjc=FgER@mail.gmail.com>

I have been splitting my time developing not only FreeSWITCH but the
CudaTel Phone system by Barracuda Networks that uses FreeSWITCH as the
telephony engine.  I'm proud to announce that not only has the product
reached a 2.0 status, it's now 100% deployed within Barracuda Networks
running the entire company's phone services across multiple locations.
 The CudaTel is using FreeSWITCH right from GIT with no special
modifications at all.  The best part is it works great as a PBX you
can then feed out to your existing FreeSWITCH setup to route the
traffic to custom applications or whatever you can think of.

The next step is to develop more exciting and impressive PBX features
that can be added for free with an energize update package available
at the time of purchase.  CudaTel comes with the same great support
you receive today on FreeSWITCH.

This is also a great opportunity for those of your out there looking
to get into the reseller market.  Contact me at the consulting link at
the top of this page or call 408-588-3633 to learn how you can become
a reseller of CudaTel.  Use your existing VoIP expertise to support
and deploy CudaTel to others.



Check it out at 


-- 
Anthony Minessale II

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

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

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


From freeswitch-list at puzzled.xs4all.nl  Fri Feb  4 01:47:21 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Thu, 03 Feb 2011 23:47:21 +0100
Subject: [Freeswitch-dev] ANNOUNCEMENT CudaTel IP PBX (powered
 by	FreeSWITCH) Reaches Critical Mass
In-Reply-To: <AANLkTin7xCfqk1kyyyQMW-VZQo2t9MJ9c++_Zjc=FgER@mail.gmail.com>
References: <AANLkTin7xCfqk1kyyyQMW-VZQo2t9MJ9c++_Zjc=FgER@mail.gmail.com>
Message-ID: <4D4B3079.7030005@puzzled.xs4all.nl>

On 02/03/2011 10:37 PM, Anthony Minessale wrote:
> I have been splitting my time developing not only FreeSWITCH but the
> CudaTel Phone system by Barracuda Networks that uses FreeSWITCH as the
> telephony engine.  I'm proud to announce that not only has the product
> reached a 2.0 status, it's now 100% deployed within Barracuda Networks
> running the entire company's phone services across multiple locations.

Congratulations Anthony (and team)!

>   The CudaTel is using FreeSWITCH right from GIT with no special
> modifications at all.  The best part is it works great as a PBX you
> can then feed out to your existing FreeSWITCH setup to route the
> traffic to custom applications or whatever you can think of.

Just let your imagination get creative.

> The next step is to develop more exciting and impressive PBX features
> that can be added for free with an energize update package available
> at the time of purchase.  CudaTel comes with the same great support
> you receive today on FreeSWITCH.

I watched the Asterisk SCF demo video earlier this week. It simply can 
not be that FreeSWITCH does not have (will not have) that capability :)

Regards,
Patrick


From anthony.minessale at gmail.com  Fri Feb  4 02:03:20 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 3 Feb 2011 17:03:20 -0600
Subject: [Freeswitch-dev] ANNOUNCEMENT CudaTel IP PBX (powered by
 FreeSWITCH) Reaches Critical Mass
In-Reply-To: <4D4B3079.7030005@puzzled.xs4all.nl>
References: <AANLkTin7xCfqk1kyyyQMW-VZQo2t9MJ9c++_Zjc=FgER@mail.gmail.com>
	<4D4B3079.7030005@puzzled.xs4all.nl>
Message-ID: <AANLkTim4eX_LunQv59Knx0mMrT_yYQ3CQ=89x_RiNP6j@mail.gmail.com>

Good news, Actually it already does.
Added in Dec 2009.
Demo'd at ClueCon 2010 (1 month sooner than the demo mentioned)
See http://wiki.freeswitch.org/wiki/Freeswitch_HA


On Thu, Feb 3, 2011 at 4:47 PM, Patrick Lists
<freeswitch-list at puzzled.xs4all.nl> wrote:
> On 02/03/2011 10:37 PM, Anthony Minessale wrote:
>> I have been splitting my time developing not only FreeSWITCH but the
>> CudaTel Phone system by Barracuda Networks that uses FreeSWITCH as the
>> telephony engine. ?I'm proud to announce that not only has the product
>> reached a 2.0 status, it's now 100% deployed within Barracuda Networks
>> running the entire company's phone services across multiple locations.
>
> Congratulations Anthony (and team)!
>
>> ? The CudaTel is using FreeSWITCH right from GIT with no special
>> modifications at all. ?The best part is it works great as a PBX you
>> can then feed out to your existing FreeSWITCH setup to route the
>> traffic to custom applications or whatever you can think of.
>
> Just let your imagination get creative.
>
>> The next step is to develop more exciting and impressive PBX features
>> that can be added for free with an energize update package available
>> at the time of purchase. ?CudaTel comes with the same great support
>> you receive today on FreeSWITCH.
>
> I watched the Asterisk SCF demo video earlier this week. It simply can
> not be that FreeSWITCH does not have (will not have) that capability :)
>
> Regards,
> Patrick
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at 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 II

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

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

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


From freeswitch-list at puzzled.xs4all.nl  Fri Feb  4 02:23:03 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Fri, 04 Feb 2011 00:23:03 +0100
Subject: [Freeswitch-dev] ANNOUNCEMENT CudaTel IP PBX (powered by
 FreeSWITCH) Reaches Critical Mass
In-Reply-To: <AANLkTim4eX_LunQv59Knx0mMrT_yYQ3CQ=89x_RiNP6j@mail.gmail.com>
References: <AANLkTin7xCfqk1kyyyQMW-VZQo2t9MJ9c++_Zjc=FgER@mail.gmail.com>	<4D4B3079.7030005@puzzled.xs4all.nl>
	<AANLkTim4eX_LunQv59Knx0mMrT_yYQ3CQ=89x_RiNP6j@mail.gmail.com>
Message-ID: <4D4B38D7.3000600@puzzled.xs4all.nl>

On 02/04/2011 12:03 AM, Anthony Minessale wrote:
> Good news, Actually it already does.
> Added in Dec 2009.
> Demo'd at ClueCon 2010 (1 month sooner than the demo mentioned)
> See http://wiki.freeswitch.org/wiki/Freeswitch_HA

Heh I should have known. Glad to see that "it can't be" is indeed the 
case in a positive way. Tip of the hat.

Regards,
Patrick


From dujinfang at gmail.com  Fri Feb  4 05:03:01 2011
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 4 Feb 2011 10:03:01 +0800
Subject: [Freeswitch-dev] ANNOUNCEMENT CudaTel IP PBX (powered by
 FreeSWITCH) Reaches Critical Mass
In-Reply-To: <4D4B38D7.3000600@puzzled.xs4all.nl>
References: <AANLkTin7xCfqk1kyyyQMW-VZQo2t9MJ9c++_Zjc=FgER@mail.gmail.com>
	<4D4B3079.7030005@puzzled.xs4all.nl>
	<AANLkTim4eX_LunQv59Knx0mMrT_yYQ3CQ=89x_RiNP6j@mail.gmail.com>
	<4D4B38D7.3000600@puzzled.xs4all.nl>
Message-ID: <AANLkTi=cCTbuey+_6+RgYXypERW_3z0SmXurhjnRsKBL@mail.gmail.com>

Awsome.

Is customers free to extend functions with lua, event_socket as FS
does? does't it has room for a erlang VM or other technologies on the
PBX? Actually I'd like to see a full linux server :)

Cannot wait to see one. Will it available on China market? I'd like to
do some customization (like translation and extend functions) and make
it available for China customers.

On Fri, Feb 4, 2011 at 7:23 AM, Patrick Lists
<freeswitch-list at puzzled.xs4all.nl> wrote:
> On 02/04/2011 12:03 AM, Anthony Minessale wrote:
>> Good news, Actually it already does.
>> Added in Dec 2009.
>> Demo'd at ClueCon 2010 (1 month sooner than the demo mentioned)
>> See http://wiki.freeswitch.org/wiki/Freeswitch_HA
>
> Heh I should have known. Glad to see that "it can't be" is indeed the
> case in a positive way. Tip of the hat.
>
> Regards,
> Patrick
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From anthony.minessale at gmail.com  Fri Feb  4 05:56:38 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 3 Feb 2011 20:56:38 -0600
Subject: [Freeswitch-dev] ANNOUNCEMENT CudaTel IP PBX (powered by
 FreeSWITCH) Reaches Critical Mass
In-Reply-To: <AANLkTi=cCTbuey+_6+RgYXypERW_3z0SmXurhjnRsKBL@mail.gmail.com>
References: <AANLkTin7xCfqk1kyyyQMW-VZQo2t9MJ9c++_Zjc=FgER@mail.gmail.com>
	<4D4B3079.7030005@puzzled.xs4all.nl>
	<AANLkTim4eX_LunQv59Knx0mMrT_yYQ3CQ=89x_RiNP6j@mail.gmail.com>
	<4D4B38D7.3000600@puzzled.xs4all.nl>
	<AANLkTi=cCTbuey+_6+RgYXypERW_3z0SmXurhjnRsKBL@mail.gmail.com>
Message-ID: <AANLkTin73rv7bXRX7Gtq8hrWyeqs-D6daTm==nLrfB4v@mail.gmail.com>

you cant use lua or event socket on the cudatel itself. Its too
difficult to secure and ensure stability, you just have another
FreeSWITCH box alongside it and do whatever you want from there.

We will, however, be adding more and more REST api's you can talk to
over http as we go.



On Thu, Feb 3, 2011 at 8:03 PM, Seven Du <dujinfang at gmail.com> wrote:
> Awsome.
>
> Is customers free to extend functions with lua, event_socket as FS
> does? does't it has room for a erlang VM or other technologies on the
> PBX? Actually I'd like to see a full linux server :)
>
> Cannot wait to see one. Will it available on China market? I'd like to
> do some customization (like translation and extend functions) and make
> it available for China customers.
>
> On Fri, Feb 4, 2011 at 7:23 AM, Patrick Lists
> <freeswitch-list at puzzled.xs4all.nl> wrote:
>> On 02/04/2011 12:03 AM, Anthony Minessale wrote:
>>> Good news, Actually it already does.
>>> Added in Dec 2009.
>>> Demo'd at ClueCon 2010 (1 month sooner than the demo mentioned)
>>> See http://wiki.freeswitch.org/wiki/Freeswitch_HA
>>
>> Heh I should have known. Glad to see that "it can't be" is indeed the
>> case in a positive way. Tip of the hat.
>>
>> Regards,
>> Patrick
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>
>
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:? http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at 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 II

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

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

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


From d.mordovin at dwide.com  Fri Feb  4 09:44:20 2011
From: d.mordovin at dwide.com (Dmitry Mordovin)
Date: Fri, 04 Feb 2011 09:44:20 +0300
Subject: [Freeswitch-dev] FreeSwitch and simple encryption
Message-ID: <4D4BA044.9090206@dwide.com>

Hello,

I wish to implement some simple encryption stuff into FS.

My Idea:
Imagine, some clients who have SIP softphone with encryption (My own 
method and only SIP signaling packets will affected) will interconnect 
with FS.
So, FS must decrypt this packets before proceed it and encrypt before 
sent to this client.

Could anyone direct me to easy way to implement it?

Thanks
Dmitry

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From steveayre at gmail.com  Fri Feb  4 11:41:09 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 4 Feb 2011 08:41:09 +0000
Subject: [Freeswitch-dev] FreeSwitch and simple encryption
In-Reply-To: <4D4BA044.9090206@dwide.com>
References: <4D4BA044.9090206@dwide.com>
Message-ID: <AANLkTik5MJM5vEoF-O3cYto_v2XSCqG0yjP_nG-5aVwm@mail.gmail.com>

For signalling:
http://wiki.freeswitch.org/wiki/SIP_TLS

For media:
http://wiki.freeswitch.org/wiki/SRTP

The key used to encrypt the media is passed through the signalling, so you
should make sure the signalling is encrypted otherwise everyone will be able
to see the key and read the media.

-Steve



On 4 February 2011 06:44, Dmitry Mordovin <d.mordovin at dwide.com> wrote:

>  Hello,
>
> I wish to implement some simple encryption stuff into FS.
>
> My Idea:
> Imagine, some clients who have SIP softphone with encryption (My own method
> and only SIP signaling packets will affected) will interconnect with FS.
> So, FS must decrypt this packets before proceed it and encrypt before sent
> to this client.
>
> Could anyone direct me to easy way to implement it?
>
> Thanks
> Dmitry
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From mrene_lists at avgs.ca  Sun Feb  6 09:41:37 2011
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sun, 6 Feb 2011 01:41:37 -0500
Subject: [Freeswitch-dev] switch_xml_config.c
In-Reply-To: <2146B97617A64E0985101460B7E0FBC4@mapleworks.com>
References: <2146B97617A64E0985101460B7E0FBC4@mapleworks.com>
Message-ID: <ADDC679F-B835-4BB2-ABA4-D50E287EC163@avgs.ca>

Hi,

It indeed seems appropriate to change those to be inclusive.

Fixed in 0d5fcf65a0fef932f32874da6f4bdddb69279c53

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2011-01-26, at 3:24 PM, Gilles Depatie wrote:

> Hi
>  
> I?m running into what appears to be a questionable range interpretation issue in switch_xml_config_parse_event function.
>  
> When enforce_min and enforce_max are TRUE should the value not be inclusive of MIN and MAX?
>  
> Say a value range of min = ?0? and max = ?1000?, should values of 0 and 1000 not be acceptable?
>  
> The current statement:
>                         if ((int_options->enforce_min && !(intval > int_options->min)) || (int_options->enforce_max && !(intval < int_options->max)))
> excludes those values presently
>  
> The error message seems to imply that the range is inclusive of the min and max values
>  
> I?ve had a look at the latest git version or the file and in this respect nothing has changed from the version I am currently using
> GILLES
> Gilles Depatie, Senior S/W Designer, MapleWorks Technology
> O: 819-776-6066 x353  |  gilles.depatie at mapleworks.com
>  
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org

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From msc at freeswitch.org  Tue Feb  8 19:42:35 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 8 Feb 2011 10:42:35 -0600
Subject: [Freeswitch-dev] UPDATE: Buy the devs dinner
Message-ID: <AANLkTimqurw-1TdPyYNXw7C6y3tHy9kufqJf9AhP03yD@mail.gmail.com>

Hello all!

Just an update: all the FreeSWITCH developers are together this week and
will be going to dinner. It's still not too late for those who wish to pitch
in and help feed them. Just hit the Paypal link on the main page and drop a
few dollars!

Thanks,
MC
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From yky1628 at yahoo.com  Tue Feb  8 06:46:32 2011
From: yky1628 at yahoo.com (Frankie Yiu)
Date: Mon, 7 Feb 2011 19:46:32 -0800 (PST)
Subject: [Freeswitch-dev] C# problem for calling
	"freeswitch.switch_core_session_read_frame()"
Message-ID: <453609.53187.qm@web30507.mail.mud.yahoo.com>

Hi there,


I would like to?dial a phone number, and read the?RTP package back when 
connected?so that we can analyze the data; (to determine when we should play?an 
audio?at the right time--human or answer machine.)

We found a code?for IVR 
test?(http://docs.freeswitch.org/switch__ivr_8c-source.html) Function name: 
switch_ivr_sound_test

We would like to do?the same but with C# code,?but we encountered a problem when 
calling the?function "freeswitch.switch_core_session_read_frame(??)?" 


< in swig.cs

switch_core_session_read_frame(SWIGTYPE_p_switch_core_session session, 
SWIGTYPE_p_p_switch_frame frame, uint flags, int stream_id)?? >

where the second parameter--frame is a pointer to pointer of switch_frame 
type?and in C# code,?it?is?having a difficulty passing?an object to the C++ side 
and keep the pointer place holder before going deeper into the C code 
(switch_core_io.c)

1) So is there any way I can call this function in C#?
2) Is there another function or routine that you can suggest me to for reading 
RTP package?

Thanks,

?Frank 



      
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From mbodbg at gmx.net  Tue Feb  8 22:36:42 2011
From: mbodbg at gmx.net (mbo)
Date: Tue, 8 Feb 2011 20:36:42 +0100
Subject: [Freeswitch-dev] Error using gpgme in mod_python
Message-ID: <25CB2B05-A925-4381-B526-B15D7AD7B67D@gmx.net>

Hello NG,

we are launching a python script which is accessing the gpgme libary for encryption/decryption : 

import os
import gpgme
from freeswitch import *

ctx=gpgme.Context()
ctx.get_key(?B9F2747A?)


If we run this python script direcly in the command line (without "from freeswitch import *"), everything works fine. If we call the script within freeswitch using the mod_python module, we receive the following debug output, when the environment variable GPGME_DEBUG is set to ?9:/tmp?:

...
gpgme_op_keylist_start: enter: ctx=0x7fcd8c2078c0, pattern=B9F2747A, secret_only=0
GPGME 2011-02-07 21:40:58 <0x3b06>      _gpgme_io_pipe: enter: filedes=0x7fcd8c1e5bc8, inherit_idx=1 (GPGME uses it for reading)
GPGME 2011-02-07 21:40:58 <0x3b06>      _gpgme_io_pipe: leave: read=0x1a5, write=0x1a6
GPGME 2011-02-07 21:40:58 <0x3b06>      _gpgme_io_set_close_notify: enter: fd=0x1a5, close_handler=0x7fcd84074420/0x7fcd8c1e5ba0
GPGME 2011-02-07 21:40:58 <0x3b06>      _gpgme_io_set_close_notify: error: Invalid argument
GPGME 2011-02-07 21:40:58 <0x3b06>      _gpgme_io_close: enter: fd=0x1a5
GPGME 2011-02-07 21:40:58 <0x3b06>      _gpgme_io_close: leave: result=0
GPGME 2011-02-07 21:40:58 <0x3b06>      _gpgme_io_close: enter: fd=0x1a6
GPGME 2011-02-07 21:40:58 <0x3b06>      _gpgme_io_close: leave: result=0
GPGME 2011-02-07 21:40:58 <0x3b06>    gpgme_op_keylist_start: error: General error <GPGME>
GPGME 2011-02-07 21:40:58 <0x3b06>    gpgme_release: call: ctx=0x7fcd8c2078c0
GPGME 2011-02-07 21:40:58 <0x3b06>  gpgme_get_key: error: General error <GPGME>
... 

Full trace can be seen here: http://pastebin.freeswitch.org/15285  

Here are the versions we are using:

Debian 5.5, Kernel 2.6.26-2-amd64 #1 SMP Tue Jan 25 05:59:43 UTC 2011 x86_64 GNU/Linux
FreeSWITCH Version 1.0.head (git-2ec2a9b 2011-02-04 09-40-04 -0600)
gpgme-1.3.0
libassuan-2.0.1
libgpg-error-1.10
pygpgme-0.1+bzr20090820



From yky1628 at yahoo.com  Wed Feb  9 13:45:54 2011
From: yky1628 at yahoo.com (Frankie Yiu)
Date: Wed, 9 Feb 2011 02:45:54 -0800 (PST)
Subject: [Freeswitch-dev] How to read RTP package/data?
Message-ID: <68103.31487.qm@web30504.mail.mud.yahoo.com>

Hi there,

I am new to FreeSwitch, and I?have a question to ask.

If I want to read the?RTP package (data from a person picking up a phone 
and?speak?or an answer machine) in C++ or C#,?what function?should?I call to get 
it???Currently I can make a call?to a?phone and play an audio.

Thanks?in advance.
?Frankie 


 
____________________________________________________________________________________
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From steveayre at gmail.com  Wed Feb  9 13:50:56 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 9 Feb 2011 10:50:56 +0000
Subject: [Freeswitch-dev] How to read RTP package/data?
In-Reply-To: <68103.31487.qm@web30504.mail.mud.yahoo.com>
References: <68103.31487.qm@web30504.mail.mud.yahoo.com>
Message-ID: <AANLkTikxv=ruVm3vUqhUvf5Sg8t5LcbmGrLdvNYy+_MD@mail.gmail.com>

You want a media bug. It registers a callback to your module where FS will
decode the data and pass it to your module. You can then process it, and
even modify it if you wish.

API Documentation is on http://docs.freeswitch.org/ at Modules -> Core
Library -> Media Bugs

-Steve



On 9 February 2011 10:45, Frankie Yiu <yky1628 at yahoo.com> wrote:

>  Hi there,
>
> I am new to FreeSwitch, and I have a question to ask.
>
> If I want to read the RTP package (data from a person picking up a phone
> and speak or an answer machine) in C++ or C#, what function should I call to
> get it?  Currently I can make a call to a phone and play an audio.
>
> Thanks in advance.
>
> Frankie
>
> ------------------------------
> Food fight?<http://answers.yahoo.com/dir/index;_ylc=X3oDMTFvbGNhMGE3BF9TAzM5NjU0NTEwOARfcwMzOTY1NDUxMDMEc2VjA21haWxfdGFnbGluZQRzbGsDbWFpbF90YWcx?link=ask&sid=396545367>Enjoy some healthy debate
> in the Yahoo! Answers Food & Drink Q&A.<http://answers.yahoo.com/dir/index;_ylc=X3oDMTFvbGNhMGE3BF9TAzM5NjU0NTEwOARfcwMzOTY1NDUxMDMEc2VjA21haWxfdGFnbGluZQRzbGsDbWFpbF90YWcx?link=ask&sid=396545367>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Wed Feb  9 13:51:27 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 9 Feb 2011 10:51:27 +0000
Subject: [Freeswitch-dev] How to read RTP package/data?
In-Reply-To: <AANLkTikxv=ruVm3vUqhUvf5Sg8t5LcbmGrLdvNYy+_MD@mail.gmail.com>
References: <68103.31487.qm@web30504.mail.mud.yahoo.com>
	<AANLkTikxv=ruVm3vUqhUvf5Sg8t5LcbmGrLdvNYy+_MD@mail.gmail.com>
Message-ID: <AANLkTimpemK1aeB5yNnLuHTTcXE_sj+zqaSfHaiEUkuk@mail.gmail.com>

The buffer will contain audio in L16 format.

-Steve


On 9 February 2011 10:50, Steven Ayre <steveayre at gmail.com> wrote:

> You want a media bug. It registers a callback to your module where FS will
> decode the data and pass it to your module. You can then process it, and
> even modify it if you wish.
>
> API Documentation is on http://docs.freeswitch.org/ at Modules -> Core
> Library -> Media Bugs
>
> -Steve
>
>
>
> On 9 February 2011 10:45, Frankie Yiu <yky1628 at yahoo.com> wrote:
>
>>  Hi there,
>>
>> I am new to FreeSwitch, and I have a question to ask.
>>
>> If I want to read the RTP package (data from a person picking up a phone
>> and speak or an answer machine) in C++ or C#, what function should I call to
>> get it?  Currently I can make a call to a phone and play an audio.
>>
>> Thanks in advance.
>>
>> Frankie
>>
>> ------------------------------
>> Food fight?<http://answers.yahoo.com/dir/index;_ylc=X3oDMTFvbGNhMGE3BF9TAzM5NjU0NTEwOARfcwMzOTY1NDUxMDMEc2VjA21haWxfdGFnbGluZQRzbGsDbWFpbF90YWcx?link=ask&sid=396545367>Enjoy some healthy debate
>> in the Yahoo! Answers Food & Drink Q&A.<http://answers.yahoo.com/dir/index;_ylc=X3oDMTFvbGNhMGE3BF9TAzM5NjU0NTEwOARfcwMzOTY1NDUxMDMEc2VjA21haWxfdGFnbGluZQRzbGsDbWFpbF90YWcx?link=ask&sid=396545367>
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
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From williams.owen at gmail.com  Wed Feb  9 15:22:26 2011
From: williams.owen at gmail.com (Owen Williams)
Date: Wed, 9 Feb 2011 12:22:26 +0000
Subject: [Freeswitch-dev] C# problem for calling
	"freeswitch.switch_core_session_read_frame()"
In-Reply-To: <453609.53187.qm@web30507.mail.mud.yahoo.com>
References: <453609.53187.qm@web30507.mail.mud.yahoo.com>
Message-ID: <AANLkTin0J3s-LYx+KMXOeysAHZH6fgPp9gtN_iJ_C7+z@mail.gmail.com>

Have you looked at http://wiki.freeswitch.org/wiki/Mod_avmd

Perhaps it could solve your problem.  i.e. you could start playing your
message (initially making the assumption that a human has answered) and if
mod_avmd detects a beep then interrupt and replay your message.

If this doesn't fit your use case the mod_avmd source code may give you
somewhere to start.

Regards,

Owen

On 8 February 2011 03:46, Frankie Yiu <yky1628 at yahoo.com> wrote:

>  Hi there,
>
> I would like to dial a phone number, and read the RTP package back when
> connected so that we can analyze the data; (to determine when we should
> play an audio at the right time--human or answer machine.)
>
> We found a code for IVR test (
> http://docs.freeswitch.org/switch__ivr_8c-source.html) Function name:
> switch_ivr_sound_test
>
> We would like to do the same but with C# code, but we encountered a problem
> when calling the function "freeswitch.switch_core_session_read_frame(  ) "
>
> < in swig.cs
>
> switch_core_session_read_frame(SWIGTYPE_p_switch_core_session session,
> SWIGTYPE_p_p_switch_frame frame, uint flags, int stream_id)   >
>
> where the second parameter--frame is a pointer to pointer of switch_frame
> type and in C# code, it is having a difficulty passing an object to the C++
> side and keep the pointer place holder before going deeper into the C code
> (switch_core_io.c)
>
> 1) So is there any way I can call this function in C#?
> 2) Is there another function or routine that you can suggest me to for
> reading RTP package?
>
> Thanks,
>
>
> Frank
>
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Wed Feb  9 19:10:22 2011
From: mike at jerris.com (Michael Jerris)
Date: Wed, 9 Feb 2011 11:10:22 -0500
Subject: [Freeswitch-dev] offering two "m=" lines from FreeSWITCH
In-Reply-To: <AANLkTi=q3hYnsVrD8NCX7DXqyoQ=prHLZcyfXA5fo4nu@mail.gmail.com>
References: <AANLkTi=q3hYnsVrD8NCX7DXqyoQ=prHLZcyfXA5fo4nu@mail.gmail.com>
Message-ID: <084E082D-CB69-49D3-8BDE-8DF5ECFDC39A@jerris.com>

http://wiki.freeswitch.org/wiki/Secure_RTP

On Jan 28, 2011, at 2:22 AM, Ujjwal SIngh wrote:

> Hi,
> 
> I am new to FreeSWITCH, I was just wondering that can we initiate Best Effort SRTP offer from FreeSWITCH in the same way as Polycom phones do, 
> i.e sending two "m=" lines for the same media, the first line having a SAVP profile, while the second one having the normal AVP profile.

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From mike at jerris.com  Wed Feb  9 19:17:57 2011
From: mike at jerris.com (Michael Jerris)
Date: Wed, 9 Feb 2011 11:17:57 -0500
Subject: [Freeswitch-dev] SIP INFO carrying MSML
In-Reply-To: <AANLkTikUR4-xdjCzdzaFfYqxa7cbx2DM=BNnqsBeBzu9@mail.gmail.com>
References: <AANLkTikUR4-xdjCzdzaFfYqxa7cbx2DM=BNnqsBeBzu9@mail.gmail.com>
Message-ID: <06EEB7D7-F71F-4302-A7B1-8B2B5A4BE195@jerris.com>

see mod_sofia.c:4512

    case SWITCH_EVENT_SEND_INFO:
...

On Jan 28, 2011, at 6:31 AM, Owen Williams wrote:

> Hi all,
> 
> I'm looking into the possibility of getting basic msml support into freeswitch.  We have a few applications hosted on a remote SIP Application server that already use MSML against another media server.  The application acts as a back2back between a caller and the media server and sends info messages (inside an invite dialog) to the media server with the INFO sdp containing MSML (e.g. play file, join conf, etc).  
> 
> I'm a bit new to freeswitch but I'm guessing mod_sofia would need to modified to send info events to a new module or to the ESL, and mod_sofia also needs to be capable of receiving events to send INFOs back to the remote B2B.
> 
> There was a bounty completed about 18 months ago (Generate and be notified of SIP INFO messages) however this may have been only for out of dialog INFOs.
> http://wiki.freeswitch.org/wiki/Bounty#Generate_and_be_notified_of_SIP_INFO_messages
> 
> Any information as to whether the above bounty made it into trunk and how it can be enabled would be appreciated.  I'd like to have a play with writing a MSML processor through ESL as I've far more experience with ESL than writing c modules.

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From msc at freeswitch.org  Wed Feb  9 20:02:28 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 9 Feb 2011 11:02:28 -0600
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today
Message-ID: <AANLkTikkXQaP5Y5p75ou0HE39hNk4areMsAhhyLEtnY1@mail.gmail.com>

Hey gang,

We decided that we'd have an open call today. No real agenda items, but
we're happy to hang out and maybe answer a few questions.

-Michael
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From bernhard.suttner at winet.ch  Wed Feb  9 22:36:44 2011
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Wed, 9 Feb 2011 20:36:44 +0100
Subject: [Freeswitch-dev] Wrong log message
Message-ID: <9be88e11-bb38-4ff1-8cc7-0ff5d3e71d5c@winet.ch>

Hi,

Just want to let you know, that the following log message is wrong:

                if (!strcmp(tech_pvt->remote_sdp_video_ip, rip) && atoi(rvp) == tech_pvt->remote_sdp_video_port) {
                        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(tech_pvt->session), SWITCH_LOG_DEBUG, "Remote video address:port [%s:%d] has not changed.\n",
                                                          tech_pvt->remote_sdp_audio_ip, tech_pvt->remote_sdp_audio_port);

Should be remote_sdp_video_port I guess.

Best regards,
Bernhard



From williams.owen at gmail.com  Wed Feb  9 23:09:53 2011
From: williams.owen at gmail.com (Owen Williams)
Date: Wed, 9 Feb 2011 20:09:53 +0000
Subject: [Freeswitch-dev] SIP INFO carrying MSML
In-Reply-To: <06EEB7D7-F71F-4302-A7B1-8B2B5A4BE195@jerris.com>
References: <AANLkTikUR4-xdjCzdzaFfYqxa7cbx2DM=BNnqsBeBzu9@mail.gmail.com>
	<06EEB7D7-F71F-4302-A7B1-8B2B5A4BE195@jerris.com>
Message-ID: <AANLkTinYvck_q6Q-S5dmFuE8dFW5L_v=GHYF7VYVhmNn@mail.gmail.com>

Cheers Michael,

I've since done a bit of digging and have found the commits associated with
the SIP INFO bounty (thanks Anthony!).  This bounty doesn't go far enough to
make a msml parser using existing ESL interface.  Specifically it is not
(currently) possible to put xml into an outbound 200 response to INFO and it
is also not possible to find out if xml has been placed in an inbound 200
response to INFO.

Whilst we haven't made a final decision, it is likely that we will use the
existing ESL interface rather than trying to port MSML to freeswitch
(sorry!).  I'm hoping that we will be able to make a positive contribution,
as and when we commit to using Freeswitch.

Thanks all!

Owen


On 9 February 2011 16:17, Michael Jerris <mike at jerris.com> wrote:

> see mod_sofia.c:4512
>
>     case SWITCH_EVENT_SEND_INFO:
> ...
>
> On Jan 28, 2011, at 6:31 AM, Owen Williams wrote:
>
> Hi all,
>
> I'm looking into the possibility of getting basic msml support into
> freeswitch.  We have a few applications hosted on a remote SIP Application
> server that already use MSML against another media server.  The application
> acts as a back2back between a caller and the media server and sends info
> messages (inside an invite dialog) to the media server with the INFO sdp
> containing MSML (e.g. play file, join conf, etc).
>
> I'm a bit new to freeswitch but I'm guessing mod_sofia would need to
> modified to send info events to a new module or to the ESL, and mod_sofia
> also needs to be capable of receiving events to send INFOs back to the
> remote B2B.
>
> There was a bounty completed about 18 months ago (Generate and be notified
> of SIP INFO messages) however this may have been only for out of dialog
> INFOs.
>
> http://wiki.freeswitch.org/wiki/Bounty#Generate_and_be_notified_of_SIP_INFO_messages
>
> Any information as to whether the above bounty made it into trunk and how
> it can be enabled would be appreciated.  I'd like to have a play with
> writing a MSML processor through ESL as I've far more experience with ESL
> than writing c modules.
>
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed Feb  9 23:48:46 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 9 Feb 2011 14:48:46 -0600
Subject: [Freeswitch-dev] offering two "m=" lines from FreeSWITCH
In-Reply-To: <084E082D-CB69-49D3-8BDE-8DF5ECFDC39A@jerris.com>
References: <AANLkTi=q3hYnsVrD8NCX7DXqyoQ=prHLZcyfXA5fo4nu@mail.gmail.com>
	<084E082D-CB69-49D3-8BDE-8DF5ECFDC39A@jerris.com>
Message-ID: <A6E015C7-100C-4ED2-8E09-FD8E426AD689@freeswitch.org>

Yes but we do not send an outbound invite to a device with two m lines... the AVP and SAVP lines.  A bounty/patch would be needed to enable this.

/b

On Feb 9, 2011, at 10:10 AM, Michael Jerris wrote:

> http://wiki.freeswitch.org/wiki/Secure_RTP

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From brian at freeswitch.org  Wed Feb  9 23:55:14 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 9 Feb 2011 14:55:14 -0600
Subject: [Freeswitch-dev] Wrong log message
In-Reply-To: <9be88e11-bb38-4ff1-8cc7-0ff5d3e71d5c@winet.ch>
References: <9be88e11-bb38-4ff1-8cc7-0ff5d3e71d5c@winet.ch>
Message-ID: <B4C90CEB-BA61-43A0-A4E0-6419DBC935E6@freeswitch.org>

Stuff like this goes on Jira.  http://jira.freeswitch.org so we don't lose track of them.

/b

On Feb 9, 2011, at 1:36 PM, Bernhard Suttner wrote:

> Hi,
> 
> Just want to let you know, that the following log message is wrong:
> 
>                if (!strcmp(tech_pvt->remote_sdp_video_ip, rip) && atoi(rvp) == tech_pvt->remote_sdp_video_port) {
>                        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(tech_pvt->session), SWITCH_LOG_DEBUG, "Remote video address:port [%s:%d] has not changed.\n",
>                                                          tech_pvt->remote_sdp_audio_ip, tech_pvt->remote_sdp_audio_port);
> 
> Should be remote_sdp_video_port I guess.
> 
> Best regards,
> Bernhard
> 
> 
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org



From cmrienzo at gmail.com  Thu Feb 10 05:58:36 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Wed, 9 Feb 2011 21:58:36 -0500
Subject: [Freeswitch-dev] SIP INFO carrying MSML
In-Reply-To: <AANLkTinYvck_q6Q-S5dmFuE8dFW5L_v=GHYF7VYVhmNn@mail.gmail.com>
References: <AANLkTikUR4-xdjCzdzaFfYqxa7cbx2DM=BNnqsBeBzu9@mail.gmail.com>
	<06EEB7D7-F71F-4302-A7B1-8B2B5A4BE195@jerris.com>
	<AANLkTinYvck_q6Q-S5dmFuE8dFW5L_v=GHYF7VYVhmNn@mail.gmail.com>
Message-ID: <AANLkTi=FFaz5YtQnNpMvypCMB7dup-2fbkjNeB5B+_Yu@mail.gmail.com>

If you were to contribute mod_msml with an msml parser, I bet you could get
assistance with making changes to mod_sofia to support it.


On Wed, Feb 9, 2011 at 3:09 PM, Owen Williams <williams.owen at gmail.com>wrote:

> Cheers Michael,
>
> I've since done a bit of digging and have found the commits associated with
> the SIP INFO bounty (thanks Anthony!).  This bounty doesn't go far enough to
> make a msml parser using existing ESL interface.  Specifically it is not
> (currently) possible to put xml into an outbound 200 response to INFO and it
> is also not possible to find out if xml has been placed in an inbound 200
> response to INFO.
>
> Whilst we haven't made a final decision, it is likely that we will use the
> existing ESL interface rather than trying to port MSML to freeswitch
> (sorry!).  I'm hoping that we will be able to make a positive contribution,
> as and when we commit to using Freeswitch.
>
> Thanks all!
>
> Owen
>
>
> On 9 February 2011 16:17, Michael Jerris <mike at jerris.com> wrote:
>
>> see mod_sofia.c:4512
>>
>>     case SWITCH_EVENT_SEND_INFO:
>> ...
>>
>> On Jan 28, 2011, at 6:31 AM, Owen Williams wrote:
>>
>> Hi all,
>>
>> I'm looking into the possibility of getting basic msml support into
>> freeswitch.  We have a few applications hosted on a remote SIP Application
>> server that already use MSML against another media server.  The application
>> acts as a back2back between a caller and the media server and sends info
>> messages (inside an invite dialog) to the media server with the INFO sdp
>> containing MSML (e.g. play file, join conf, etc).
>>
>> I'm a bit new to freeswitch but I'm guessing mod_sofia would need to
>> modified to send info events to a new module or to the ESL, and mod_sofia
>> also needs to be capable of receiving events to send INFOs back to the
>> remote B2B.
>>
>> There was a bounty completed about 18 months ago (Generate and be notified
>> of SIP INFO messages) however this may have been only for out of dialog
>> INFOs.
>>
>> http://wiki.freeswitch.org/wiki/Bounty#Generate_and_be_notified_of_SIP_INFO_messages
>>
>> Any information as to whether the above bounty made it into trunk and how
>> it can be enabled would be appreciated.  I'd like to have a play with
>> writing a MSML processor through ESL as I've far more experience with ESL
>> than writing c modules.
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri Feb 11 01:48:43 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 10 Feb 2011 16:48:43 -0600
Subject: [Freeswitch-dev] OSTAG FB Page
Message-ID: <AANLkTinv0am7kVzjJ6VW9d+hTqOE1aem4Q4QBGB_A91W@mail.gmail.com>

Hello all!

For those of you on Facebook please go make a new friend:

http://www.facebook.com/pages/Open-Source-Telephony-Advancement-Group-Inc/188675947819805

This is our official OSTAG Facebook page. :)

-Michael
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From patrick.plattes at niemann-frey.info  Fri Feb 11 16:28:50 2011
From: patrick.plattes at niemann-frey.info (Patrick Plattes)
Date: Fri, 11 Feb 2011 14:28:50 +0100
Subject: [Freeswitch-dev] Always true if statement (sofia_presence.c:837 )
Message-ID: <AANLkTikz8CSrwidAU8u2FdO8G1HMnF-KhckC=1-kZ2xf@mail.gmail.com>

Hi,

on sofia_presence.c:837, isn't it always be true?

Bye,
?Patrick


From steveayre at gmail.com  Fri Feb 11 20:10:48 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 11 Feb 2011 17:10:48 +0000
Subject: [Freeswitch-dev] Always true if statement (sofia_presence.c:837
	)
In-Reply-To: <AANLkTikz8CSrwidAU8u2FdO8G1HMnF-KhckC=1-kZ2xf@mail.gmail.com>
References: <AANLkTikz8CSrwidAU8u2FdO8G1HMnF-KhckC=1-kZ2xf@mail.gmail.com>
Message-ID: <AANLkTikJ0W4B1LdBJA5SNADe08WLS7Vp_BFN-x9_3_7O@mail.gmail.com>

Bug reports should go on jira.freeswitch.org

On 11 February 2011 13:28, Patrick Plattes <
patrick.plattes at niemann-frey.info> wrote:

> Hi,
>
> on sofia_presence.c:837, isn't it always be true?
>
> Bye,
>  Patrick
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
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From mrene_lists at avgs.ca  Fri Feb 11 22:10:42 2011
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Fri, 11 Feb 2011 14:10:42 -0500
Subject: [Freeswitch-dev] Always true if statement (sofia_presence.c:837
	)
In-Reply-To: <AANLkTikz8CSrwidAU8u2FdO8G1HMnF-KhckC=1-kZ2xf@mail.gmail.com>
References: <AANLkTikz8CSrwidAU8u2FdO8G1HMnF-KhckC=1-kZ2xf@mail.gmail.com>
Message-ID: <9122295A-1013-4A11-83B8-5D8A70704021@avgs.ca>

My line 837 is ",'%q','%q','%q',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'%q','%q',"

If I assume you mean line 832  if ((sql = switch_mprintf("select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,"
It'll be true if switch_mprintf() succeeds, false if it fails.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2011-02-11, at 8:28 AM, Patrick Plattes wrote:

> Hi,
> 
> on sofia_presence.c:837, isn't it always be true?
> 
> Bye,
>  Patrick
> 
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org



From lconroy at insensate.co.uk  Mon Feb 14 21:05:21 2011
From: lconroy at insensate.co.uk (Lawrence Conroy)
Date: Mon, 14 Feb 2011 18:05:21 +0000
Subject: [Freeswitch-dev] DB update of 11th February -- is it just me?
Message-ID: <BDDBE4A9-A376-461C-9243-1D50593731DD@insensate.co.uk>

Hi there,
 Just wondered -- has anyone had problems since the registration db commit of Friday 11th?

I'm running fS on a Mac TiBook (PPC, running Mac OS X 10.5.8), so this is waaaay off the standard target.

Having updated to head on a cleaned system, the new fS dumps a core on startup.

Did a git revert -n a2c0da53f368f0b11340c3a72814c93b182753b7, followed by make sure.
Problem went away.

----

As a minor issue, the new code seems to generate an error report well before fS segs.
This error says that the registration db doesn't have a field "real".

Looking at the patch for the a2c0da53f368f0b11340c3a72814c93b182753b7 commit,
this error seems to tie in with the change to switch_core_sqldb.c, and the
static char create_registrations_sql[]=
bit, where there's an added line (1638):
 create index regindex1 on registrations (user, real, hostname)\n";

Was that intended?

Putting this as:
  create index regindex1 on registrations (user, realm, hostname)\n";
removes the error report.

Sorry for the delay in reporting, but it takes ages to do anything on this old
machine and it is locked away without a reliable Internet connection.

all the best,
  Lawrence





From jerry.richards at teotech.com  Mon Feb 14 21:24:38 2011
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 14 Feb 2011 10:24:38 -0800
Subject: [Freeswitch-dev] Bridge to PSTN Number Not Using Call
	Timeout	Value
In-Reply-To: <AANLkTinBOwwUdtReEyHwbwOpFtGSgG46vzpiLLSzRYja@mail.gmail.com>
References: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F2C52EF@VA3DIAXVS351.RED001.local>
	<AANLkTinBOwwUdtReEyHwbwOpFtGSgG46vzpiLLSzRYja@mail.gmail.com>
Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F823C71@VA3DIAXVS351.RED001.local>

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 at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Monday, January 31, 2011 12:13 PM
To: freeswitch-dev at 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 at 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 at 192.168.72.79:5060
> bridge({presence_id=2001 at 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 at g1)
>
>
>
> Thanks,
>
> Jerry
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at 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 II

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

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

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

_______________________________________________
FreeSWITCH-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
http://www.freeswitch.org




From anthony.minessale at gmail.com  Mon Feb 14 21:34:56 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 14 Feb 2011 12:34:56 -0600
Subject: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout
	Value
In-Reply-To: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F823C71@VA3DIAXVS351.RED001.local>
References: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F2C52EF@VA3DIAXVS351.RED001.local>
	<AANLkTinBOwwUdtReEyHwbwOpFtGSgG46vzpiLLSzRYja@mail.gmail.com>
	<2BF7FB90DF25EA4485949F3AF2B9D6960B3F823C71@VA3DIAXVS351.RED001.local>
Message-ID: <AANLkTikbG2VPVqEJJ_Ep+T8yCOS8x13rZfqQWaK9tRT8@mail.gmail.com>

set bridge_answer_timeout on the A leg


On Mon, Feb 14, 2011 at 12:24 PM, Jerry Richards
<jerry.richards at 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 at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
> Sent: Monday, January 31, 2011 12:13 PM
> To: freeswitch-dev at 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 at 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 at 192.168.72.79:5060
>> bridge({presence_id=2001 at 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 at g1)
>>
>>
>>
>> Thanks,
>>
>> Jerry
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at 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 II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at 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 II

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

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

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


From anthony.minessale at gmail.com  Tue Feb 15 00:21:11 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 14 Feb 2011 15:21:11 -0600
Subject: [Freeswitch-dev] DB update of 11th February -- is it just me?
In-Reply-To: <BDDBE4A9-A376-461C-9243-1D50593731DD@insensate.co.uk>
References: <BDDBE4A9-A376-461C-9243-1D50593731DD@insensate.co.uk>
Message-ID: <AANLkTinHa-jXwXcgbzyBpiAhphPmXGBUF129=oHLWBG7@mail.gmail.com>

already fixed

On Mon, Feb 14, 2011 at 12:05 PM, Lawrence Conroy
<lconroy at insensate.co.uk> wrote:
> Hi there,
> ?Just wondered -- has anyone had problems since the registration db commit of Friday 11th?
>
> I'm running fS on a Mac TiBook (PPC, running Mac OS X 10.5.8), so this is waaaay off the standard target.
>
> Having updated to head on a cleaned system, the new fS dumps a core on startup.
>
> Did a git revert -n a2c0da53f368f0b11340c3a72814c93b182753b7, followed by make sure.
> Problem went away.
>
> ----
>
> As a minor issue, the new code seems to generate an error report well before fS segs.
> This error says that the registration db doesn't have a field "real".
>
> Looking at the patch for the a2c0da53f368f0b11340c3a72814c93b182753b7 commit,
> this error seems to tie in with the change to switch_core_sqldb.c, and the
> static char create_registrations_sql[]=
> bit, where there's an added line (1638):
> ?create index regindex1 on registrations (user, real, hostname)\n";
>
> Was that intended?
>
> Putting this as:
> ?create index regindex1 on registrations (user, realm, hostname)\n";
> removes the error report.
>
> Sorry for the delay in reporting, but it takes ages to do anything on this old
> machine and it is locked away without a reliable Internet connection.
>
> all the best,
> ?Lawrence
>
>
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at 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 II

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

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

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


From jerry.richards at teotech.com  Tue Feb 15 00:44:40 2011
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 14 Feb 2011 13:44:40 -0800
Subject: [Freeswitch-dev] Bridge to PSTN Number Not Using Call
	Timeout	Value
In-Reply-To: <AANLkTikbG2VPVqEJJ_Ep+T8yCOS8x13rZfqQWaK9tRT8@mail.gmail.com>
References: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F2C52EF@VA3DIAXVS351.RED001.local>
	<AANLkTinBOwwUdtReEyHwbwOpFtGSgG46vzpiLLSzRYja@mail.gmail.com>
	<2BF7FB90DF25EA4485949F3AF2B9D6960B3F823C71@VA3DIAXVS351.RED001.local>
	<AANLkTikbG2VPVqEJJ_Ep+T8yCOS8x13rZfqQWaK9tRT8@mail.gmail.com>
Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F823DA9@VA3DIAXVS351.RED001.local>

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 at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Monday, February 14, 2011 10:35 AM
To: freeswitch-dev at 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 at 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 at lists.freeswitch.org 
> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of 
> Anthony Minessale
> Sent: Monday, January 31, 2011 12:13 PM
> To: freeswitch-dev at 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 at 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 at 192.168.72.79:5060
>> bridge({presence_id=2001 at 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 at g1)
>>
>>
>>
>> Thanks,
>>
>> Jerry
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-de
>> v
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at 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 II

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

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

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

_______________________________________________
FreeSWITCH-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
http://www.freeswitch.org




From anthony.minessale at gmail.com  Tue Feb 15 00:51:01 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 14 Feb 2011 15:51:01 -0600
Subject: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout
	Value
In-Reply-To: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F823DA9@VA3DIAXVS351.RED001.local>
References: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F2C52EF@VA3DIAXVS351.RED001.local>
	<AANLkTinBOwwUdtReEyHwbwOpFtGSgG46vzpiLLSzRYja@mail.gmail.com>
	<2BF7FB90DF25EA4485949F3AF2B9D6960B3F823C71@VA3DIAXVS351.RED001.local>
	<AANLkTikbG2VPVqEJJ_Ep+T8yCOS8x13rZfqQWaK9tRT8@mail.gmail.com>
	<2BF7FB90DF25EA4485949F3AF2B9D6960B3F823DA9@VA3DIAXVS351.RED001.local>
Message-ID: <AANLkTi=7PoSRhpSi967UpVUGksYRZm5Khmdm_So6QUqG@mail.gmail.com>

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 at 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 at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
> Sent: Monday, February 14, 2011 10:35 AM
> To: freeswitch-dev at 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 at 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 at lists.freeswitch.org
>> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of
>> Anthony Minessale
>> Sent: Monday, January 31, 2011 12:13 PM
>> To: freeswitch-dev at 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 at 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 at 192.168.72.79:5060
>>> bridge({presence_id=2001 at 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 at g1)
>>>
>>>
>>>
>>> Thanks,
>>>
>>> Jerry
>>>
>>> _______________________________________________
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-de
>>> v
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at 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 II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at 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 II

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

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

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


From msc at freeswitch.org  Tue Feb 15 03:42:47 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 14 Feb 2011 18:42:47 -0600
Subject: [Freeswitch-dev] INFORMATION: FreeSWITCH Training, West Coast
Message-ID: <AANLkTinDFXwz+9Ln3xPvFxQbTfYKDOLn=fU2pE5_1wOD@mail.gmail.com>

This is a reminder for those who are interested in the FreeSWITCH Bootcamp
put on by the 2600hz project. There are two upcoming training sessions, both
on the west coast of the USA:
Mar 9-11, San Francisco
May 11-13, Seattle

The complete list is available at:
http://www.voipkb.com/

There are a few international training sessions that will be happening later
in the year, so check the VoIPKB site for updates.

Thanks to all those who are promoting the use of FreeSWITCH around the
world!

-Michael
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From msc at freeswitch.org  Tue Feb 15 04:20:19 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 14 Feb 2011 19:20:19 -0600
Subject: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout
	Value
In-Reply-To: <AANLkTi=7PoSRhpSi967UpVUGksYRZm5Khmdm_So6QUqG@mail.gmail.com>
References: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F2C52EF@VA3DIAXVS351.RED001.local>
	<AANLkTinBOwwUdtReEyHwbwOpFtGSgG46vzpiLLSzRYja@mail.gmail.com>
	<2BF7FB90DF25EA4485949F3AF2B9D6960B3F823C71@VA3DIAXVS351.RED001.local>
	<AANLkTikbG2VPVqEJJ_Ep+T8yCOS8x13rZfqQWaK9tRT8@mail.gmail.com>
	<2BF7FB90DF25EA4485949F3AF2B9D6960B3F823DA9@VA3DIAXVS351.RED001.local>
	<AANLkTi=7PoSRhpSi967UpVUGksYRZm5Khmdm_So6QUqG@mail.gmail.com>
Message-ID: <AANLkTi=X0RN6BH3nHWNF7k4nEEH5SigV-1KX5uvTDC=K@mail.gmail.com>

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 at 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.
>
>
-------------- next part --------------
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From jerry.richards at teotech.com  Tue Feb 15 21:57:43 2011
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 15 Feb 2011 10:57:43 -0800
Subject: [Freeswitch-dev] Bridge to PSTN Number Not Using Call
	Timeout	Value
In-Reply-To: <AANLkTi=7PoSRhpSi967UpVUGksYRZm5Khmdm_So6QUqG@mail.gmail.com>
References: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F2C52EF@VA3DIAXVS351.RED001.local>
	<AANLkTinBOwwUdtReEyHwbwOpFtGSgG46vzpiLLSzRYja@mail.gmail.com>
	<2BF7FB90DF25EA4485949F3AF2B9D6960B3F823C71@VA3DIAXVS351.RED001.local>
	<AANLkTikbG2VPVqEJJ_Ep+T8yCOS8x13rZfqQWaK9tRT8@mail.gmail.com>
	<2BF7FB90DF25EA4485949F3AF2B9D6960B3F823DA9@VA3DIAXVS351.RED001.local>
	<AANLkTi=7PoSRhpSi967UpVUGksYRZm5Khmdm_So6QUqG@mail.gmail.com>
Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F8240F0@VA3DIAXVS351.RED001.local>

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 at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Monday, February 14, 2011 1:51 PM
To: freeswitch-dev at 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 at 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 at lists.freeswitch.org 
> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of 
> Anthony Minessale
> Sent: Monday, February 14, 2011 10:35 AM
> To: freeswitch-dev at 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 at 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 at lists.freeswitch.org
>> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of 
>> Anthony Minessale
>> Sent: Monday, January 31, 2011 12:13 PM
>> To: freeswitch-dev at 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 at 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 at 192.168.72.79:5060
>>> bridge({presence_id=2001 at 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 at g1)
>>>
>>>
>>>
>>> Thanks,
>>>
>>> Jerry
>>>
>>> _______________________________________________
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-d
>>> e
>>> v
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-de
>> v
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-de
>> v
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at 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 II

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

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

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

_______________________________________________
FreeSWITCH-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
http://www.freeswitch.org




From anthony.minessale at gmail.com  Tue Feb 15 23:50:28 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 15 Feb 2011 14:50:28 -0600
Subject: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout
	Value
In-Reply-To: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F8240F0@VA3DIAXVS351.RED001.local>
References: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F2C52EF@VA3DIAXVS351.RED001.local>
	<AANLkTinBOwwUdtReEyHwbwOpFtGSgG46vzpiLLSzRYja@mail.gmail.com>
	<2BF7FB90DF25EA4485949F3AF2B9D6960B3F823C71@VA3DIAXVS351.RED001.local>
	<AANLkTikbG2VPVqEJJ_Ep+T8yCOS8x13rZfqQWaK9tRT8@mail.gmail.com>
	<2BF7FB90DF25EA4485949F3AF2B9D6960B3F823DA9@VA3DIAXVS351.RED001.local>
	<AANLkTi=7PoSRhpSi967UpVUGksYRZm5Khmdm_So6QUqG@mail.gmail.com>
	<2BF7FB90DF25EA4485949F3AF2B9D6960B3F8240F0@VA3DIAXVS351.RED001.local>
Message-ID: <AANLkTimE5Edn6H_RaaO-1cZEigbXF6-5pAOJsJYLx9X9@mail.gmail.com>

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 at 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 at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
> Sent: Monday, February 14, 2011 1:51 PM
> To: freeswitch-dev at 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 at 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 at lists.freeswitch.org
>> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of
>> Anthony Minessale
>> Sent: Monday, February 14, 2011 10:35 AM
>> To: freeswitch-dev at 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 at 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 at lists.freeswitch.org
>>> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of
>>> Anthony Minessale
>>> Sent: Monday, January 31, 2011 12:13 PM
>>> To: freeswitch-dev at 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 at 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 at 192.168.72.79:5060
>>>> bridge({presence_id=2001 at 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 at g1)
>>>>
>>>>
>>>>
>>>> Thanks,
>>>>
>>>> Jerry
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-dev mailing list
>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-d
>>>> e
>>>> v
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-de
>>> v
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-de
>>> v
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at 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 II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at 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 II

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

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

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


From msc at freeswitch.org  Wed Feb 16 00:01:32 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 15 Feb 2011 15:01:32 -0600
Subject: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout
	Value
In-Reply-To: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F8240F0@VA3DIAXVS351.RED001.local>
References: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F2C52EF@VA3DIAXVS351.RED001.local>
	<AANLkTinBOwwUdtReEyHwbwOpFtGSgG46vzpiLLSzRYja@mail.gmail.com>
	<2BF7FB90DF25EA4485949F3AF2B9D6960B3F823C71@VA3DIAXVS351.RED001.local>
	<AANLkTikbG2VPVqEJJ_Ep+T8yCOS8x13rZfqQWaK9tRT8@mail.gmail.com>
	<2BF7FB90DF25EA4485949F3AF2B9D6960B3F823DA9@VA3DIAXVS351.RED001.local>
	<AANLkTi=7PoSRhpSi967UpVUGksYRZm5Khmdm_So6QUqG@mail.gmail.com>
	<2BF7FB90DF25EA4485949F3AF2B9D6960B3F8240F0@VA3DIAXVS351.RED001.local>
Message-ID: <AANLkTimSsHa+G3jXmiBqBnrYqN-e-fDMbitVM7VJxZtk@mail.gmail.com>

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 at 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 at lists.freeswitch.org [mailto:
> freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anthony
> Minessale
> Sent: Monday, February 14, 2011 1:51 PM
> To: freeswitch-dev at 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 at 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 at lists.freeswitch.org
> > [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of
> > Anthony Minessale
> > Sent: Monday, February 14, 2011 10:35 AM
> > To: freeswitch-dev at 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 at 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 at lists.freeswitch.org
> >> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of
> >> Anthony Minessale
> >> Sent: Monday, January 31, 2011 12:13 PM
> >> To: freeswitch-dev at 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 at 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 at 192.168.72.79:5060
> >>> bridge({presence_id=2001 at 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 at g1)
> >>>
> >>>
> >>>
> >>> Thanks,
> >>>
> >>> Jerry
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-dev mailing list
> >>> FreeSWITCH-dev at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-d
> >>> e
> >>> v
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-dev mailing list
> >> FreeSWITCH-dev at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-de
> >> v
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-dev mailing list
> >> FreeSWITCH-dev at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-de
> >> v
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> > googletalk:conf+888 at conference.freeswitch.org
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at 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 II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed Feb 16 20:34:40 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 16 Feb 2011 09:34:40 -0800
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today
Message-ID: <AANLkTinWchNsK7HCoe71BC+XjU84AiKJP-JXcRZW3A+B@mail.gmail.com>

Hey folks,

We are having a "get caught up with the latest in FreeSWITCH" session today.
Here's the agenda:

http://wiki.freeswitch.org/wiki/FS_weekly_2011_02_16

I will be reviewing a lot of the stuff that went into FreeSWITCH in the
month of January. If you are familiar with some of the new things please
feel free to add your input to the call today and/or update the wiki page.
For example, if you are familiar with the mod_snmp or mod_fsk we would
definitely welcome your observations.

Talk to you soon,
Michael
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From msc at freeswitch.org  Fri Feb 18 03:29:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 17 Feb 2011 16:29:38 -0800
Subject: [Freeswitch-dev] Skype 2.0.72 for Linux
Message-ID: <AANLkTinYLAuoB9wy_vjbZwbO38PthoFzSia-2bS9yCwf@mail.gmail.com>

Does anyone know where to find this? It's the only one that the wiki says
works with mod_skypopen but not even Google seems to be able to find one. (I
find "2.0.72" but it links to the latest beta on the skype.com site which is
not what we want.)

Any suggestions are appreciated.

-MC
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From msc at freeswitch.org  Wed Feb 23 20:56:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 23 Feb 2011 09:56:38 -0800
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Starting Shortly
Message-ID: <AANLkTinz1f_o08j976ORyR+GSHPg3CZyO2Q-Zo4WZ4DS@mail.gmail.com>

Hey folks, feel free to hop on!

http://wiki.freeswitch.org/wiki/FS_weekly_2011_02_23

Talk to you shortly,
MC
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From michelhabib at gmail.com  Fri Feb 18 13:42:40 2011
From: michelhabib at gmail.com (Michel Habib)
Date: Fri, 18 Feb 2011 10:42:40 -0000
Subject: [Freeswitch-dev] ASR from Freeswitch to Loquendo - No Audio passed
	to MRCP Server
Message-ID: <AANLkTiknFyESH2kuj2erYAEp1tKN6eoDRrFGcvQ-drvC@mail.gmail.com>

Dears, I am seeking expert advice on setting up Freeswitch to Loquendo MRCP
Connection.
I have freeswitch on one computer, configured for MRCP V1 - Loquendo Speech
Suite 7.
I have Loquendo set-up on the other computer - with MRCP V1 Profile.
Using a simple lua script driving my extension, I can successfully:-
- establish the connection to Loquendo Server.
- define grammar
- compile grammar successfully.
- Begin Recognition
Then it waits forever until i hangup the call.

Looking at the Wireshark Traces, the RTP packets are transferred succesfully
from softphone to freeswitch [i could hear my voice on the playback]
But, when playing back the RTP packets between freeswitch and Loquendo,
there is only deep silence [same thing with audio dump created by Loquendo].

Can you direct me where to troubleshoot this on the Freeswitch side? I will
be more than happy to provide any logs you need to resolve this.

Thank you,
Michel.
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