From anthony.minessale at gmail.com  Wed Feb  1 00:31:42 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 31 Jan 2012 15:31:42 -0600
Subject: [Freeswitch-dev] Absolute conditions are not changing the
 proceed variable... (mod_dialplan_xml.c)
In-Reply-To: <CAKoY+k3vcob6PKU3eJFQuNqVTfy4-7=3UuGM7ckLffHPpURNOw@mail.gmail.com>
References: <CAKoY+k3vcob6PKU3eJFQuNqVTfy4-7=3UuGM7ckLffHPpURNOw@mail.gmail.com>
Message-ID: <CAKbxfG-=AH11VbVvDWzDobg0qBLBjPeDsX+hupU6TH4NLCt9dw@mail.gmail.com>

this appears to be a regression from a recent change.

Fixed:

commit e00a61c48bce26cb96f3d615bd0f08091d7143e1
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Tue Jan 31 15:31:08 2012 -0600


On Tue, Jan 31, 2012 at 2:31 PM, Rafael Vargas <vargas at khomp.com.br> wrote:
> Hi,
>
> I've noticed that when I have an extension containing only an absolute
> condition, the proceed variable in mod_dialplan_xml.c:parse_exten(...) is
> not changed and the next extension is also executed, thus ignoring the first
> extension's "continue" attribute, which was set to "false".
>
> I'm referring to this code: (mod_dialplan.c:221 in today's GIT)
> ...
> } else if (time_match == -1) {
> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG_CLEAN(session),
> SWITCH_LOG_DEBUG,
> ??"Dialplan: %s Absolute Condition [%s] match=%s\n",
> switch_channel_get_name(channel), exten_name, all ? "all" : "any");
> pass++;
> if (!all && !xor) break;
> } else if (time_match == 1) {
> ...
>
> Shouldn't this set the proceed variable to 1, just like the regex condition
> a few lines above (line 208 in today's GIT)?
>
> I just made a quick test and it worked as (I) expected.
>
> What I'm proposing is to change the code above to the following:
>
> ...
> } else if (time_match == -1) {
> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG_CLEAN(session),
> SWITCH_LOG_DEBUG,
> ??"Dialplan: %s Absolute Condition [%s] match=%s\n",
> switch_channel_get_name(channel), exten_name, all ? "all" : "any");
> proceed = 1;
> pass++;
> if (!all && !xor) break;
> } else if (time_match == 1) {
> ...
>
> Sorry if this was already discussed, I've searched this list and the users
> list to no avail...
>
> Best regards,
>
> Rafael Scheffer Vargas
> desenvolvimento? ? ? ? ? ? ? ? ? ? ? ? ?KHOMP - 15 Anos
>
> R. Joe Colla?o, 253 ? ? ? ? ? ? ? ? ?? www.khomp.com.br
> Florian?polis | SC
> +55 48 3722-2916 | 9612-0276 | USA: +1 213-943-1535
> vargas at khomp.com.br
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
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 dujinfang at gmail.com  Wed Feb  1 05:25:36 2012
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 1 Feb 2012 10:25:36 +0800
Subject: [Freeswitch-dev] switch_stun_random_string returns identical
 strings on two calls randomly
In-Reply-To: <CAKbxfG8-dpsZxe5dLvioO9Q5OHrqrEA+4=kjdkVPZPEd6TO=FA@mail.gmail.com>
References: <E8A898581AA84EAEB342AFFF27F85C16@gmail.com>
	<CAKbxfG8-dpsZxe5dLvioO9Q5OHrqrEA+4=kjdkVPZPEd6TO=FA@mail.gmail.com>
Message-ID: <AE626AA0A468412B8F989E7AF1927A0A@gmail.com>

cool, I used it and no problem been found till now.

On Wednesday, February 1, 2012 at 12:03 AM, Anthony Minessale wrote:

> ya, you're right.
> Fixed in tree....
> 
> 
> On Tue, Jan 31, 2012 at 3:51 AM, Seven Du <dujinfang at gmail.com (mailto:dujinfang at gmail.com)> wrote:
> > Hi,
> > 
> > I tried to replace the following function which I copied somewhere in msrp.c
> > with  switch_stun_random_string but I found
> > sometimes switch_stun_random_string returns identical strings on two
> > sequential calls, with the following function I never found, so I still use
> > it.
> > 
> > By looking the code I understand srand seed is time based so it's possible
> > to return duplicate random string, but I
> > found switch_stun_random_string used in some places other than stun, just
> > curious would it cause problems?
> > 
> > void random_string(char *buf, switch_size_t size)
> > {
> > long val[4];
> > int x;
> > 
> > for (x = 0; x < 4; x++)
> > val[x] = random();
> > snprintf(buf, size, "%08lx%08lx%08lx%08lx", val[0], val[1], val[2], val[3]);
> > *(buf+size) = '\0';
> > }
> > 
> > Thanks.
> > 
> > --
> > About: http://about.me/dujinfang
> > Blog: http://www.dujinfang.com
> > Proj:  http://www.freeswitch.org.cn
> > 
> > Sent with Sparrow
> > 
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at lists.freeswitch.org (mailto: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 (mailto:anthony_minessale at hotmail.com)
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com (mailto:anthony.minessale at gmail.com)
> IRC: irc.freenode.net (http://irc.freenode.net) #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org (mailto:888 at conference.freeswitch.org)
> googletalk:conf+888 at conference.freeswitch.org (mailto:conf+888 at conference.freeswitch.org)
> pstn:+19193869900
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org (mailto: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 juraj.fabo at gmail.com  Wed Feb  1 17:42:11 2012
From: juraj.fabo at gmail.com (Juraj Fabo)
Date: Wed, 1 Feb 2012 15:42:11 +0100
Subject: [Freeswitch-dev] get_trillium_val() loop termination
Message-ID: <CAAqVCyRMVGKO6g3eV7MSvXPMQNSQy6E-3H6d7admG3fYDCd2Gg@mail.gmail.com>

Hi

I'm using freetdm from git downloaded on 30.1.2012.
On windows I had no problems, but on sles 11 sp1 64bit call setup message
was missing caller and called information elements.
I've tracked down the problem to file
src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn_support.c, into function
get_trillium_val()
Particularly, when get_trillium_val(ton_codes, 0, IN_TON_UNK) is called,
0xe6 is returned (which is from the nsf_spec_codes[] table)
I think the loop should have some strict terminate condition but not sure
about the write intention.
My workaround to be able to continue development is following

diff -urN src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn_support.c
src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn_support_mod.c
--- src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn_support.c   2012-02-01
14:02:30.000000000 +0000
+++ src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn_support_mod.c
2012-02-01 14:02:08.000000000 +0000
@@ -106,10 +106,11 @@
 static uint8_t get_trillium_val(ftdm2trillium_t *vals, uint8_t ftdm_val,
uint8_t default_val)
 {
        ftdm2trillium_t *val = vals;
-       while(val++) {
+       while(val) {
                if (val->ftdm_val == ftdm_val) {
                        return val->trillium_val;
                }
+                val++;
        }
        return default_val;
 }
@@ -117,10 +118,11 @@
 static uint8_t get_ftdm_val(ftdm2trillium_t *vals, uint8_t trillium_val,
uint8_t default_val)
 {
        ftdm2trillium_t *val = vals;
-       while(val++) {
+       while(val) {
                if (val->trillium_val == trillium_val) {
                        return val->ftdm_val;
                }
+                val++;
        }
        return default_val;
 }

 Is there some other way how to go around?

 Thank you

 Juraj
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From brian at freeswitch.org  Wed Feb  1 17:48:41 2012
From: brian at freeswitch.org (Brian West)
Date: Wed, 1 Feb 2012 08:48:41 -0600
Subject: [Freeswitch-dev] get_trillium_val() loop termination
In-Reply-To: <CAAqVCyRMVGKO6g3eV7MSvXPMQNSQy6E-3H6d7admG3fYDCd2Gg@mail.gmail.com>
References: <CAAqVCyRMVGKO6g3eV7MSvXPMQNSQy6E-3H6d7admG3fYDCd2Gg@mail.gmail.com>
Message-ID: <1A5736BB-70DF-45DB-ACD5-19BA4FB9A643@freeswitch.org>

Juraj,
	I thank you for your time to track this down.  But we don't post patches to the mailing lists you'll need to open a jira at http://jira.freeswitch.org and attach the patch if possible.

Thanks,
Brian

Sent from my hospital bed.

On Feb 1, 2012, at 8:42 AM, Juraj Fabo wrote:

> Hi
> 
> I'm using freetdm from git downloaded on 30.1.2012.
> On windows I had no problems, but on sles 11 sp1 64bit call setup message
> was missing caller and called information elements.
> I've tracked down the problem to file
> src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn_support.c, into function
> get_trillium_val()
> Particularly, when get_trillium_val(ton_codes, 0, IN_TON_UNK) is called,
> 0xe6 is returned (which is from the nsf_spec_codes[] table)
> I think the loop should have some strict terminate condition but not sure
> about the write intention.
> My workaround to be able to continue development is following
> 
> diff -urN src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn_support.c
> src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn_support_mod.c
> --- src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn_support.c   2012-02-01
> 14:02:30.000000000 +0000
> +++ src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn_support_mod.c
> 2012-02-01 14:02:08.000000000 +0000
> @@ -106,10 +106,11 @@
> static uint8_t get_trillium_val(ftdm2trillium_t *vals, uint8_t ftdm_val,
> uint8_t default_val)
> {
>        ftdm2trillium_t *val = vals;
> -       while(val++) {
> +       while(val) {
>                if (val->ftdm_val == ftdm_val) {
>                        return val->trillium_val;
>                }
> +                val++;
>        }
>        return default_val;
> }
> @@ -117,10 +118,11 @@
> static uint8_t get_ftdm_val(ftdm2trillium_t *vals, uint8_t trillium_val,
> uint8_t default_val)
> {
>        ftdm2trillium_t *val = vals;
> -       while(val++) {
> +       while(val) {
>                if (val->trillium_val == trillium_val) {
>                        return val->ftdm_val;
>                }
> +                val++;
>        }
>        return default_val;
> }
> 
> Is there some other way how to go around?
> 
> Thank you
> 
> Juraj
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Brian West 
FreeSWITCH Solutions, LLC
Phone: +1 (918) 420-9266 
Fax:   +1 (918) 420-9267
brian at freeswitch.org
http://www.freeswitch.org

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From moises.silva at gmail.com  Wed Feb  1 20:20:42 2012
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 1 Feb 2012 12:20:42 -0500
Subject: [Freeswitch-dev] get_trillium_val() loop termination
In-Reply-To: <1A5736BB-70DF-45DB-ACD5-19BA4FB9A643@freeswitch.org>
References: <CAAqVCyRMVGKO6g3eV7MSvXPMQNSQy6E-3H6d7admG3fYDCd2Gg@mail.gmail.com>
	<1A5736BB-70DF-45DB-ACD5-19BA4FB9A643@freeswitch.org>
Message-ID: <CAA4nhyCKLRv-AD3jA4n53ccSoYJERvgBfRNLN0aJHXEmm2sHEg@mail.gmail.com>

> On Feb 1, 2012, at 8:42 AM, Juraj Fabo wrote:
>
> Hi
>
> I'm using freetdm from git downloaded on 30.1.2012.
> On windows I had no problems, but on sles 11 sp1 64bit call setup message
> was missing caller and called information elements.
> I've tracked down the problem to file
> src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn_support.c, into function
> get_trillium_val()
> Particularly, when get_trillium_val(ton_codes, 0, IN_TON_UNK) is called,
> 0xe6 is returned (which is from the nsf_spec_codes[] table)
> I think the loop should have some strict terminate condition but not sure
> about the write intention.
> My workaround to be able to continue development is following
>
> diff -urN src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn_support.c
> src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn_support_mod.c
> --- src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn_support.c   2012-02-01
> 14:02:30.000000000 +0000
> +++ src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn_support_mod.c
> 2012-02-01 14:02:08.000000000 +0000
> @@ -106,10 +106,11 @@
> static uint8_t get_trillium_val(ftdm2trillium_t *vals, uint8_t ftdm_val,
> uint8_t default_val)
> {
>        ftdm2trillium_t *val = vals;
> -       while(val++) {
> +       while(val) {
>                if (val->ftdm_val == ftdm_val) {
>                        return val->trillium_val;
>                }
> +                val++;
>        }
>        return default_val;
> }
> @@ -117,10 +118,11 @@
> static uint8_t get_ftdm_val(ftdm2trillium_t *vals, uint8_t trillium_val,
> uint8_t default_val)
> {
>        ftdm2trillium_t *val = vals;
> -       while(val++) {
> +       while(val) {
>                if (val->trillium_val == trillium_val) {
>                        return val->ftdm_val;
>                }
> +                val++;
>        }
>        return default_val;
> }
>
> Is there some other way how to go around?
>
>
All that looping code was f'd up pretty bad and it was fixed yesterday.
Update to see the fix.

*Moises Silva
**Software Engineer, Development Manager***

msilva at sangoma.com

Sangoma Technologies

100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada


t.   +1 800 388 2475 (N. America)

t.   +1 905 474 1990 x128

f.   +1 905 474 9223



**<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>

Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
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 | Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>`|
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From appalonovandrey at gmail.com  Wed Feb  1 08:41:41 2012
From: appalonovandrey at gmail.com (=?KOI8-R?B?4c7E0sXKIOHQ0MHMz87P1w==?=)
Date: Wed, 1 Feb 2012 11:41:41 +0600
Subject: [Freeswitch-dev] (no subject)
Message-ID: <CACJEBkyqhnn5hTJ27K3Y=jxQHPzJSCtjwkza0Bro5=ar-mDXpg@mail.gmail.com>

Hello, tell, why FS can not hear that the fax has lifted a receiver
(I use ventafax)?


From karlseacrest at yahoo.com  Wed Feb  1 19:41:03 2012
From: karlseacrest at yahoo.com (Karl Seacrest)
Date: Wed, 1 Feb 2012 08:41:03 -0800 (PST)
Subject: [Freeswitch-dev] mod_httapi
Message-ID: <1328114463.16981.YahooMailNeo@web121601.mail.ne1.yahoo.com>

Hello,
I will like to know how i can use Pre-Answer feature using the mod_httapi. This is useful to perform operations like playing?playing custom caller tunes or validate the callers phone number from database before the call is answered.

How can i achieve this using the FreeSWITCH mod_httapi?

Also does?mod_httapi send caller information (Caller ID, etc) along when requesting a URI?


Regards
Karl C.
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From vargas at khomp.com.br  Wed Feb  1 20:26:02 2012
From: vargas at khomp.com.br (Rafael Vargas)
Date: Wed, 1 Feb 2012 15:26:02 -0200
Subject: [Freeswitch-dev] Absolute conditions are not changing the
 proceed variable... (mod_dialplan_xml.c)
In-Reply-To: <CAKbxfG-=AH11VbVvDWzDobg0qBLBjPeDsX+hupU6TH4NLCt9dw@mail.gmail.com>
References: <CAKoY+k3vcob6PKU3eJFQuNqVTfy4-7=3UuGM7ckLffHPpURNOw@mail.gmail.com>
	<CAKbxfG-=AH11VbVvDWzDobg0qBLBjPeDsX+hupU6TH4NLCt9dw@mail.gmail.com>
Message-ID: <CAKoY+k3qNU=PSzt_Cd-FMU_P6=gScZ7OfCVYpZvF825q6-A=Lw@mail.gmail.com>

Hi again,

There were still cases in the function that needed to change the proceed
variable.

Please find attached the git diff output with the changes.

Rafael Scheffer Vargas
desenvolvimento                         KHOMP - 15 Anos

R. Joe Colla?o, 253                    www.khomp.com.br
Florian?polis | SC
+55 48 3722-2916 | 9612-0276 | USA: +1 213-943-1535
vargas at khomp.com.br



On Tue, Jan 31, 2012 at 19:31, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> this appears to be a regression from a recent change.
>
> Fixed:
>
> commit e00a61c48bce26cb96f3d615bd0f08091d7143e1
> Author: Anthony Minessale <anthm at freeswitch.org>
> Date:   Tue Jan 31 15:31:08 2012 -0600
>
>
> On Tue, Jan 31, 2012 at 2:31 PM, Rafael Vargas <vargas at khomp.com.br>
> wrote:
> > Hi,
> >
> > I've noticed that when I have an extension containing only an absolute
> > condition, the proceed variable in mod_dialplan_xml.c:parse_exten(...) is
> > not changed and the next extension is also executed, thus ignoring the
> first
> > extension's "continue" attribute, which was set to "false".
> >
> > I'm referring to this code: (mod_dialplan.c:221 in today's GIT)
> > ...
> > } else if (time_match == -1) {
> > switch_log_printf(SWITCH_CHANNEL_SESSION_LOG_CLEAN(session),
> > SWITCH_LOG_DEBUG,
> >   "Dialplan: %s Absolute Condition [%s] match=%s\n",
> > switch_channel_get_name(channel), exten_name, all ? "all" : "any");
> > pass++;
> > if (!all && !xor) break;
> > } else if (time_match == 1) {
> > ...
> >
> > Shouldn't this set the proceed variable to 1, just like the regex
> condition
> > a few lines above (line 208 in today's GIT)?
> >
> > I just made a quick test and it worked as (I) expected.
> >
> > What I'm proposing is to change the code above to the following:
> >
> > ...
> > } else if (time_match == -1) {
> > switch_log_printf(SWITCH_CHANNEL_SESSION_LOG_CLEAN(session),
> > SWITCH_LOG_DEBUG,
> >   "Dialplan: %s Absolute Condition [%s] match=%s\n",
> > switch_channel_get_name(channel), exten_name, all ? "all" : "any");
> > proceed = 1;
> > pass++;
> > if (!all && !xor) break;
> > } else if (time_match == 1) {
> > ...
> >
> > Sorry if this was already discussed, I've searched this list and the
> users
> > list to no avail...
> >
> > Best regards,
> >
> > Rafael Scheffer Vargas
> > desenvolvimento                         KHOMP - 15 Anos
> >
> > R. Joe Colla?o, 253                    www.khomp.com.br
> > Florian?polis | SC
> > +55 48 3722-2916 | 9612-0276 | USA: +1 213-943-1535
> > vargas at khomp.com.br
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> >
>
>
>
> --
> 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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 anthony.minessale at gmail.com  Wed Feb  1 21:05:19 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 1 Feb 2012 12:05:19 -0600
Subject: [Freeswitch-dev] Absolute conditions are not changing the
 proceed variable... (mod_dialplan_xml.c)
In-Reply-To: <CAKoY+k3qNU=PSzt_Cd-FMU_P6=gScZ7OfCVYpZvF825q6-A=Lw@mail.gmail.com>
References: <CAKoY+k3vcob6PKU3eJFQuNqVTfy4-7=3UuGM7ckLffHPpURNOw@mail.gmail.com>
	<CAKbxfG-=AH11VbVvDWzDobg0qBLBjPeDsX+hupU6TH4NLCt9dw@mail.gmail.com>
	<CAKoY+k3qNU=PSzt_Cd-FMU_P6=gScZ7OfCVYpZvF825q6-A=Lw@mail.gmail.com>
Message-ID: <CAKbxfG8ZDZYRJuj1Gr-CMjngX+VV9BDAY9yKQo23-aQpAp6d8Q@mail.gmail.com>

can you use jira for this going forward its against our policy to do
patches on the mailing list.


On Wed, Feb 1, 2012 at 11:26 AM, Rafael Vargas <vargas at khomp.com.br> wrote:
> Hi again,
>
> There were still cases in the function that needed to change the proceed
> variable.
>
> Please find attached the git diff output with the changes.
>
> Rafael Scheffer Vargas
> desenvolvimento? ? ? ? ? ? ? ? ? ? ? ? ?KHOMP - 15 Anos
>
> R. Joe Colla?o, 253 ? ? ? ? ? ? ? ? ?? www.khomp.com.br
> Florian?polis | SC
> +55 48 3722-2916 | 9612-0276 | USA: +1 213-943-1535
> vargas at khomp.com.br
>
>
>
> On Tue, Jan 31, 2012 at 19:31, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> this appears to be a regression from a recent change.
>>
>> Fixed:
>>
>> commit e00a61c48bce26cb96f3d615bd0f08091d7143e1
>> Author: Anthony Minessale <anthm at freeswitch.org>
>> Date: ? Tue Jan 31 15:31:08 2012 -0600
>>
>>
>> On Tue, Jan 31, 2012 at 2:31 PM, Rafael Vargas <vargas at khomp.com.br>
>> wrote:
>> > Hi,
>> >
>> > I've noticed that when I have an extension containing only an absolute
>> > condition, the proceed variable in mod_dialplan_xml.c:parse_exten(...)
>> > is
>> > not changed and the next extension is also executed, thus ignoring the
>> > first
>> > extension's "continue" attribute, which was set to "false".
>> >
>> > I'm referring to this code: (mod_dialplan.c:221 in today's GIT)
>> > ...
>> > } else if (time_match == -1) {
>> > switch_log_printf(SWITCH_CHANNEL_SESSION_LOG_CLEAN(session),
>> > SWITCH_LOG_DEBUG,
>> > ??"Dialplan: %s Absolute Condition [%s] match=%s\n",
>> > switch_channel_get_name(channel), exten_name, all ? "all" : "any");
>> > pass++;
>> > if (!all && !xor) break;
>> > } else if (time_match == 1) {
>> > ...
>> >
>> > Shouldn't this set the proceed variable to 1, just like the regex
>> > condition
>> > a few lines above (line 208 in today's GIT)?
>> >
>> > I just made a quick test and it worked as (I) expected.
>> >
>> > What I'm proposing is to change the code above to the following:
>> >
>> > ...
>> > } else if (time_match == -1) {
>> > switch_log_printf(SWITCH_CHANNEL_SESSION_LOG_CLEAN(session),
>> > SWITCH_LOG_DEBUG,
>> > ??"Dialplan: %s Absolute Condition [%s] match=%s\n",
>> > switch_channel_get_name(channel), exten_name, all ? "all" : "any");
>> > proceed = 1;
>> > pass++;
>> > if (!all && !xor) break;
>> > } else if (time_match == 1) {
>> > ...
>> >
>> > Sorry if this was already discussed, I've searched this list and the
>> > users
>> > list to no avail...
>> >
>> > Best regards,
>> >
>> > Rafael Scheffer Vargas
>> > desenvolvimento? ? ? ? ? ? ? ? ? ? ? ? ?KHOMP - 15 Anos
>> >
>> > R. Joe Colla?o, 253 ? ? ? ? ? ? ? ? ?? www.khomp.com.br
>> > Florian?polis | SC
>> > +55 48 3722-2916 | 9612-0276 | USA: +1 213-943-1535
>> > vargas at khomp.com.br
>> >
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.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
>> >
>>
>>
>>
>> --
>> 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
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
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 vargas at khomp.com.br  Wed Feb  1 21:08:32 2012
From: vargas at khomp.com.br (Rafael Vargas)
Date: Wed, 1 Feb 2012 16:08:32 -0200
Subject: [Freeswitch-dev] Absolute conditions are not changing the
 proceed variable... (mod_dialplan_xml.c)
In-Reply-To: <CAKbxfG8ZDZYRJuj1Gr-CMjngX+VV9BDAY9yKQo23-aQpAp6d8Q@mail.gmail.com>
References: <CAKoY+k3vcob6PKU3eJFQuNqVTfy4-7=3UuGM7ckLffHPpURNOw@mail.gmail.com>
	<CAKbxfG-=AH11VbVvDWzDobg0qBLBjPeDsX+hupU6TH4NLCt9dw@mail.gmail.com>
	<CAKoY+k3qNU=PSzt_Cd-FMU_P6=gScZ7OfCVYpZvF825q6-A=Lw@mail.gmail.com>
	<CAKbxfG8ZDZYRJuj1Gr-CMjngX+VV9BDAY9yKQo23-aQpAp6d8Q@mail.gmail.com>
Message-ID: <CAKoY+k1Q0N+uKHe_bNQm6K2p5M3fXAhBcsdBWRBkLhBmqzDDsg@mail.gmail.com>

Will do! Thanks!

Rafael Scheffer Vargas
desenvolvimento                         KHOMP - 15 Anos

R. Joe Colla?o, 253                    www.khomp.com.br
Florian?polis | SC
+55 48 3722-2916 | 9612-0276 | USA: +1 213-943-1535
vargas at khomp.com.br



On Wed, Feb 1, 2012 at 16:05, Anthony Minessale <anthony.minessale at gmail.com
> wrote:

> can you use jira for this going forward its against our policy to do
> patches on the mailing list.
>
>
> On Wed, Feb 1, 2012 at 11:26 AM, Rafael Vargas <vargas at khomp.com.br>
> wrote:
> > Hi again,
> >
> > There were still cases in the function that needed to change the proceed
> > variable.
> >
> > Please find attached the git diff output with the changes.
> >
> > Rafael Scheffer Vargas
> > desenvolvimento                         KHOMP - 15 Anos
> >
> > R. Joe Colla?o, 253                    www.khomp.com.br
> > Florian?polis | SC
> > +55 48 3722-2916 | 9612-0276 | USA: +1 213-943-1535
> > vargas at khomp.com.br
> >
> >
> >
> > On Tue, Jan 31, 2012 at 19:31, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> this appears to be a regression from a recent change.
> >>
> >> Fixed:
> >>
> >> commit e00a61c48bce26cb96f3d615bd0f08091d7143e1
> >> Author: Anthony Minessale <anthm at freeswitch.org>
> >> Date:   Tue Jan 31 15:31:08 2012 -0600
> >>
> >>
> >> On Tue, Jan 31, 2012 at 2:31 PM, Rafael Vargas <vargas at khomp.com.br>
> >> wrote:
> >> > Hi,
> >> >
> >> > I've noticed that when I have an extension containing only an absolute
> >> > condition, the proceed variable in mod_dialplan_xml.c:parse_exten(...)
> >> > is
> >> > not changed and the next extension is also executed, thus ignoring the
> >> > first
> >> > extension's "continue" attribute, which was set to "false".
> >> >
> >> > I'm referring to this code: (mod_dialplan.c:221 in today's GIT)
> >> > ...
> >> > } else if (time_match == -1) {
> >> > switch_log_printf(SWITCH_CHANNEL_SESSION_LOG_CLEAN(session),
> >> > SWITCH_LOG_DEBUG,
> >> >   "Dialplan: %s Absolute Condition [%s] match=%s\n",
> >> > switch_channel_get_name(channel), exten_name, all ? "all" : "any");
> >> > pass++;
> >> > if (!all && !xor) break;
> >> > } else if (time_match == 1) {
> >> > ...
> >> >
> >> > Shouldn't this set the proceed variable to 1, just like the regex
> >> > condition
> >> > a few lines above (line 208 in today's GIT)?
> >> >
> >> > I just made a quick test and it worked as (I) expected.
> >> >
> >> > What I'm proposing is to change the code above to the following:
> >> >
> >> > ...
> >> > } else if (time_match == -1) {
> >> > switch_log_printf(SWITCH_CHANNEL_SESSION_LOG_CLEAN(session),
> >> > SWITCH_LOG_DEBUG,
> >> >   "Dialplan: %s Absolute Condition [%s] match=%s\n",
> >> > switch_channel_get_name(channel), exten_name, all ? "all" : "any");
> >> > proceed = 1;
> >> > pass++;
> >> > if (!all && !xor) break;
> >> > } else if (time_match == 1) {
> >> > ...
> >> >
> >> > Sorry if this was already discussed, I've searched this list and the
> >> > users
> >> > list to no avail...
> >> >
> >> > Best regards,
> >> >
> >> > Rafael Scheffer Vargas
> >> > desenvolvimento                         KHOMP - 15 Anos
> >> >
> >> > R. Joe Colla?o, 253                    www.khomp.com.br
> >> > Florian?polis | SC
> >> > +55 48 3722-2916 | 9612-0276 | USA: +1 213-943-1535
> >> > vargas at khomp.com.br
> >> >
> >> >
> >> >
> >> >
> _________________________________________________________________________
> >> > Professional FreeSWITCH Consulting Services:
> >> > consulting at freeswitch.org
> >> > http://www.freeswitchsolutions.com
> >> >
> >> > 
> >> > 
> >> >
> >> > Official FreeSWITCH Sites
> >> > http://www.freeswitch.org
> >> > http://wiki.freeswitch.org
> >> > http://www.cluecon.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
> >> >
> >>
> >>
> >>
> >> --
> >> 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
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.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
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> >
>
>
>
> --
> 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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 anthony.minessale at gmail.com  Wed Feb  1 21:45:55 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 1 Feb 2012 12:45:55 -0600
Subject: [Freeswitch-dev] mod_httapi
In-Reply-To: <1328114463.16981.YahooMailNeo@web121601.mail.ne1.yahoo.com>
References: <1328114463.16981.YahooMailNeo@web121601.mail.ne1.yahoo.com>
Message-ID: <CAKbxfG-pz7Fvs-=xwoar0nGr4VEytR5G23JFasVKD5iP3t0SXQ@mail.gmail.com>

if you try to play a file it will pre-answer by itself

you can send anything you want via the dialplan inside the {foo=bar}
var expressions and if you enable the permissions you can get the full
call data

On Wed, Feb 1, 2012 at 10:41 AM, Karl Seacrest <karlseacrest at yahoo.com> wrote:
> Hello,
> I will like to know how i can use Pre-Answer feature using the mod_httapi.
> This is useful to perform operations like playing?playing custom caller
> tunes or validate the callers phone number from database before the call is
> answered.
>
> How can i achieve this using the FreeSWITCH mod_httapi?
>
> Also does?mod_httapi send caller information (Caller ID, etc) along when
> requesting a URI?
>
>
> Regards
> Karl C.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
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 b2m at a-cti.com  Thu Feb  2 10:52:26 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Thu, 2 Feb 2012 13:22:26 +0530
Subject: [Freeswitch-dev] Fwd: help
In-Reply-To: <mailman.2106.1328169025.13443.freeswitch-users@lists.freeswitch.org>
References: <mailman.2106.1328169025.13443.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAP2fbbd-y3RkTuSCjysoRrxkprzndq=R-kt0jY664hk00Ms2QA@mail.gmail.com>

Hi team,

I am trying to get these values from the lua script for custom CDR report,
I am able to get few values but many variables are missing or getting null
value. Any help would be appreciated.

local aa=session:getVariable("context");
local bb=session:getVariable("destination_number");
local cc=session:getVariable("caller_id_name");
local dd=session:getVariable("caller_id_number");
local ee=session:getVariable("network_addr");
local ff=session:getVariable("ani");
local ii=session:getVariable("source");
local jj=session:getVariable("chan_name");
local kk=session:getVariable("uuid");
local ll=session:getVariable("created_time");
local aa1=session:getVariable("direction");
local bb1=session:getVariable("username");
local cc1=session:getVariable("dialplan");
local dd1=session:getVariable("caller_id_number");
local ee1=session:getVariable("unique_id");

session:execute("bridge","user/601");
session:hangup();

local ff1=session:getVariable("answered_time");
local hh1=session:getVariable("created_time");
local gg1=session:getVariable("hangup_time");
local gg2=session:getVariable("end_stamp");

I am getting values for all the above variables except end_stamp,
hangup_time, duration, billsec,hangup_cause.


Thanks for your time.

Thanks,
Bala
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From karlseacrest at yahoo.com  Thu Feb  2 12:15:53 2012
From: karlseacrest at yahoo.com (Karl Seacrest)
Date: Thu, 2 Feb 2012 01:15:53 -0800 (PST)
Subject: [Freeswitch-dev] mod_httapi
In-Reply-To: <mailman.2109.1328171759.13443.freeswitch-dev@lists.freeswitch.org>
References: <mailman.2109.1328171759.13443.freeswitch-dev@lists.freeswitch.org>
Message-ID: <1328174153.43955.YahooMailNeo@web121601.mail.ne1.yahoo.com>

Thanks for your reply?Anthony,?

But i think you didn't quite understand my question.?Custom Ring Back Tones (Early Media) is one of the features listed in FreeSWITCH features page and details of??FreeSWITCH?Early Media mode is detailed in?FreeSWITCH wiki?:?http://wiki.freeswitch.org/wiki/Early_Media

By pre-answer i mean picking a call inEarly Media?mode (Early media is the exchange of information before establishment of a connection) which mean on early media mode, even though an audio is playing like Custom Ring Back Tones, the call is not connected so the caller is not billed.

The question again is how can the?FreeSWITCH?Early Media feature be used from mod_httapi?

Also, is there any?performance?benefit for using??mod_httapi over a framework like Plivo that provide RESTFul features to FreeSWITCH?

Regards

Karl C.







________________________________
 
On Wed, Feb 1, 2012 at 16:05, Anthony Minessale <anthony.minessale at gmail.com>wrote:



if you try to play a file it will pre-answer by itselfyou can send anything you want via the dialplan inside the {foo=bar}
var expressions and if you enable the permissions you can get the full
call data

On Wed, Feb 1, 2012 at 10:41 AM, Karl Seacrest <karlseacrest at yahoo.com> wrote:
> Hello,
> I will like to know how i can use Pre-Answer feature using the mod_httapi.
> This is useful to perform operations like playing?playing custom caller
> tunes or validate the callers phone number from database before the call is
> answered.
>
> How can i achieve this using the FreeSWITCH mod_httapi?
>
> Also does?mod_httapi send caller information (Caller ID, etc) along when
> requesting a URI?
>
>
> Regards
> Karl C.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
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



Hi team,

I am trying to get these values from the lua script for?custom?CDR report, I am able to get few values but many variables are missing or getting null value. Any help would be appreciated.

local aa=session:getVariable("context");
local bb=session:getVariable("destination_number");
local cc=session:getVariable("caller_id_name");
local dd=session:getVariable("caller_id_number");
local ee=session:getVariable("network_addr");
local ff=session:getVariable("ani");
local ii=session:getVariable("source");
local jj=session:getVariable("chan_name");
local kk=session:getVariable("uuid");
local ll=session:getVariable("created_time");
local aa1=session:getVariable("direction");
local bb1=session:getVariable("username");
local cc1=session:getVariable("dialplan");
local dd1=session:getVariable("caller_id_number");
local ee1=session:getVariable("unique_id");

session:execute("bridge","user/601");
session:hangup();

local ff1=session:getVariable("answered_time");
local hh1=session:getVariable("created_time");
local gg1=session:getVariable("hangup_time");
local gg2=session:getVariable("end_stamp");

I am getting values for all the above variables except end_stamp, hangup_time, duration, billsec,hangup_cause.


Thanks for your time.

Thanks,
Bala






_______________________________________________
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
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From ustcorporation at yahoo.com  Thu Feb  2 13:22:55 2012
From: ustcorporation at yahoo.com (Darren C.)
Date: Thu, 2 Feb 2012 02:22:55 -0800 (PST)
Subject: [Freeswitch-dev] switch_ivr_bridge.c question
Message-ID: <1328178175.80384.YahooMailNeo@web161003.mail.bf1.yahoo.com>

OS is Centos 5.7.? Most recent GIT has same code.
?
In switch_ivr_bridge.c on line 1474, we have the following lines of code:
?
???????????????????? switch_channel_clear_state_handler(originator_channel, NULL);
???????????????????? switch_channel_clear_state_handler(originatee_channel, NULL);
???????????????????? switch_channel_set_state_flag(originator_channel, CF_BRIDGE_ORIGINATOR);
???????????????????? switch_channel_clear_flag_recursive(originatee_channel, CF_BRIDGE_ORIGINATOR);
???????????????????? switch_channel_add_state_handler(originator_channel, &uuid_bridge_state_handlers);
???????????????????? switch_channel_add_state_handler(originatee_channel, &uuid_bridge_state_handlers);
?
Is the call to ?switch_channel_set_state_flag? a typo? Since it? triggers an asynchronous flag change on the channel thread, and at least in the scenario when the asynchronous change immediately after the next ?switch_channel_clear_flag_recursive?, the clearing of the flag will not do anything.
?
Thanks in advance.
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From msc at freeswitch.org  Thu Feb  2 19:31:36 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 2 Feb 2012 08:31:36 -0800
Subject: [Freeswitch-dev] Fwd: help
In-Reply-To: <CAP2fbbd-y3RkTuSCjysoRrxkprzndq=R-kt0jY664hk00Ms2QA@mail.gmail.com>
References: <mailman.2106.1328169025.13443.freeswitch-users@lists.freeswitch.org>
	<CAP2fbbd-y3RkTuSCjysoRrxkprzndq=R-kt0jY664hk00Ms2QA@mail.gmail.com>
Message-ID: <CAKzWOxXPzChFq5Tx9THtTeezotV3-wWvnax=Y=Dv=4XriKvhSQ@mail.gmail.com>

If this is a hangup hook then you need the special "env" object:
http://wiki.freeswitch.org/wiki/Mod_lua#Special_Case:_env_object

If this is a dialplan script then those values will, of course, not be
populated because the call is in progress.

-MC

On Wed, Feb 1, 2012 at 11:52 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:

> Hi team,
>
> I am trying to get these values from the lua script for custom CDR report,
> I am able to get few values but many variables are missing or getting null
> value. Any help would be appreciated.
>
> local aa=session:getVariable("context");
> local bb=session:getVariable("destination_number");
> local cc=session:getVariable("caller_id_name");
> local dd=session:getVariable("caller_id_number");
> local ee=session:getVariable("network_addr");
> local ff=session:getVariable("ani");
> local ii=session:getVariable("source");
> local jj=session:getVariable("chan_name");
> local kk=session:getVariable("uuid");
> local ll=session:getVariable("created_time");
> local aa1=session:getVariable("direction");
> local bb1=session:getVariable("username");
> local cc1=session:getVariable("dialplan");
> local dd1=session:getVariable("caller_id_number");
> local ee1=session:getVariable("unique_id");
>
> session:execute("bridge","user/601");
> session:hangup();
>
> local ff1=session:getVariable("answered_time");
> local hh1=session:getVariable("created_time");
> local gg1=session:getVariable("hangup_time");
> local gg2=session:getVariable("end_stamp");
>
> I am getting values for all the above variables except end_stamp,
> hangup_time, duration, billsec,hangup_cause.
>
>
> Thanks for your time.
>
> Thanks,
> Bala
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Thu Feb  2 19:44:16 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 2 Feb 2012 08:44:16 -0800
Subject: [Freeswitch-dev] mod_httapi
In-Reply-To: <1328174153.43955.YahooMailNeo@web121601.mail.ne1.yahoo.com>
References: <mailman.2109.1328171759.13443.freeswitch-dev@lists.freeswitch.org>
	<1328174153.43955.YahooMailNeo@web121601.mail.ne1.yahoo.com>
Message-ID: <CAKzWOxXF7NRmrEYbaaJCaYKL6Vui7F4HJWxg3juU8NrvL0bkrA@mail.gmail.com>

Actually, Anthony understood your question and gave you a solution with
bonus information. I especially liked the part where you tell Anthony what
early media is. That's kind of like telling a neurosurgeon that a person's
brain is inside the cranium. :P

If you issue a playback without actually answering the call then the net
effect is that you are pre_answering the call and the file you are playing
is in early media, i.e. you have not actually "answered" the call from a
billing perspective. From the caller perspective the file you play is the
ringback that they will hear.

Is there something you were curious about beyond sending audio to the
caller in early media mode?

-MC

On Thu, Feb 2, 2012 at 1:15 AM, Karl Seacrest <karlseacrest at yahoo.com>wrote:

> Thanks for your reply Anthony,
> But i think you didn't quite understand my question. Custom Ring Back
> Tones (Early Media) is one of the features listed in FreeSWITCH features
> page and details of  FreeSWITCH Early Media mode is detailed in FreeSWITCH
> wiki : http://wiki.freeswitch.org/wiki/Early_Media
>
> By pre-answer i mean picking a call in Early Media mode (Early media is
> the exchange of information before establishment of a connection) which
> mean on early media mode, even though an audio is playing like Custom Ring
> Back Tones, the call is not connected so the caller is not billed.
>
> The question again is how can the FreeSWITCH Early Media feature be used
> from mod_httapi?
>
> Also, is there any performance benefit for using  mod_httapi over a
> framework like Plivo that provide RESTFul features to FreeSWITCH?
>
> Regards
>
> Karl C.
>
>
>
>
>
>
>   ------------------------------
> On Wed, Feb 1, 2012 at 16:05, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
> if you try to play a file it will pre-answer by itself
> you can send anything you want via the dialplan inside the {foo=bar}
> var expressions and if you enable the permissions you can get the full
> call data
>
> On Wed, Feb 1, 2012 at 10:41 AM, Karl Seacrest <karlseacrest at yahoo.com>
> wrote:
> > Hello,
> > I will like to know how i can use Pre-Answer feature using the
> mod_httapi.
> > This is useful to perform operations like playing playing custom caller
> > tunes or validate the callers phone number from database before the call
> is
> > answered.
> >
> > How can i achieve this using the FreeSWITCH mod_httapi?
> >
> > Also does mod_httapi send caller information (Caller ID, etc) along when
> > requesting a URI?
> >
> >
> > Regards
> > Karl C.
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> >
>
>
>
> --
> 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
>
>
> Hi team,
>
> I am trying to get these values from the lua script for custom CDR report,
> I am able to get few values but many variables are missing or getting null
> value. Any help would be appreciated.
>
> local aa=session:getVariable("context");
> local bb=session:getVariable("destination_number");
> local cc=session:getVariable("caller_id_name");
> local dd=session:getVariable("caller_id_number");
> local ee=session:getVariable("network_addr");
> local ff=session:getVariable("ani");
> local ii=session:getVariable("source");
> local jj=session:getVariable("chan_name");
> local kk=session:getVariable("uuid");
> local ll=session:getVariable("created_time");
> local aa1=session:getVariable("direction");
> local bb1=session:getVariable("username");
> local cc1=session:getVariable("dialplan");
> local dd1=session:getVariable("caller_id_number");
> local ee1=session:getVariable("unique_id");
>
> session:execute("bridge","user/601");
> session:hangup();
>
> local ff1=session:getVariable("answered_time");
> local hh1=session:getVariable("created_time");
> local gg1=session:getVariable("hangup_time");
> local gg2=session:getVariable("end_stamp");
>
> I am getting values for all the above variables except end_stamp,
> hangup_time, duration, billsec,hangup_cause.
>
>
> Thanks for your time.
>
> Thanks,
> Bala
>
>
>
>
>
>
>
> _______________________________________________
> 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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 ustcorporation at yahoo.com  Thu Feb  2 20:43:03 2012
From: ustcorporation at yahoo.com (teldev)
Date: Thu, 2 Feb 2012 09:43:03 -0800 (PST)
Subject: [Freeswitch-dev] switch_ivr_bridge.c question
Message-ID: <1328204583380-4359635.post@n6.nabble.com>

Re-worded my original post a bit, sorry if it becomes a new thread...having
Nabble issues.

In switch_ivr_bridge.c on line 1474, we have the following lines of code:

 1474                   
switch_channel_clear_state_handler(originator_channel, NULL);
 1475                   
switch_channel_clear_state_handler(originatee_channel, NULL);
 1476                    switch_channel_set_state_flag(originator_channel,
CF_BRIDGE_ORIGINATOR);
 1477                   
switch_channel_clear_flag_recursive(originatee_channel,
CF_BRIDGE_ORIGINATOR);
 1478                   
switch_channel_add_state_handler(originator_channel,
&uuid_bridge_state_handlers);
 1479                   
switch_channel_add_state_handler(originatee_channel,
&uuid_bridge_state_handlers);

Is the call to ?switch_channel_set_state_flag? a typo? Since line 1476
triggers an asynchronous flag change, and line 1477 changes the same flag
synchronously, on line 1478 the value of the flag becomes unpredictable
since the asynchronous change from line 1476 can happen before or after line
1477 (mostly after, of course).

Thanks again.

--
View this message in context: http://freeswitch-dev.4355.n6.nabble.com/switch-ivr-bridge-c-question-tp4359635p4359635.html
Sent from the FreeSWITCH-dev mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Thu Feb  2 23:20:22 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 2 Feb 2012 14:20:22 -0600
Subject: [Freeswitch-dev] switch_ivr_bridge.c question
In-Reply-To: <1328204583380-4359635.post@n6.nabble.com>
References: <1328204583380-4359635.post@n6.nabble.com>
Message-ID: <CAKbxfG-G7JN6-1hUUsxXnh8Pw_bxXjgcXZw8P6EzN=feTDcJwQ@mail.gmail.com>

The code is not the same on the latest.
Look closer ;)

You should probably keep yourself on the latest for this deep of an examination.



On Thu, Feb 2, 2012 at 11:43 AM, teldev <ustcorporation at yahoo.com> wrote:
> Re-worded my original post a bit, sorry if it becomes a new thread...having
> Nabble issues.
>
> In switch_ivr_bridge.c on line 1474, we have the following lines of code:
>
> ?1474
> switch_channel_clear_state_handler(originator_channel, NULL);
> ?1475
> switch_channel_clear_state_handler(originatee_channel, NULL);
> ?1476 ? ? ? ? ? ? ? ? ? ?switch_channel_set_state_flag(originator_channel,
> CF_BRIDGE_ORIGINATOR);
> ?1477
> switch_channel_clear_flag_recursive(originatee_channel,
> CF_BRIDGE_ORIGINATOR);
> ?1478
> switch_channel_add_state_handler(originator_channel,
> &uuid_bridge_state_handlers);
> ?1479
> switch_channel_add_state_handler(originatee_channel,
> &uuid_bridge_state_handlers);
>
> Is the call to ?switch_channel_set_state_flag? a typo? Since line 1476
> triggers an asynchronous flag change, and line 1477 changes the same flag
> synchronously, on line 1478 the value of the flag becomes unpredictable
> since the asynchronous change from line 1476 can happen before or after line
> 1477 (mostly after, of course).
>
> Thanks again.
>
> --
> View this message in context: http://freeswitch-dev.4355.n6.nabble.com/switch-ivr-bridge-c-question-tp4359635p4359635.html
> Sent from the FreeSWITCH-dev mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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



-- 
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 karlseacrest at yahoo.com  Thu Feb  2 23:42:55 2012
From: karlseacrest at yahoo.com (Karl Seacrest)
Date: Thu, 2 Feb 2012 12:42:55 -0800 (PST)
Subject: [Freeswitch-dev] mod_httapi
In-Reply-To: <mailman.2135.1328200329.13443.freeswitch-dev@lists.freeswitch.org>
References: <mailman.2135.1328200329.13443.freeswitch-dev@lists.freeswitch.org>
Message-ID: <1328215375.43945.YahooMailNeo@web121603.mail.ne1.yahoo.com>

Thanks for your reply?Anthony,?

But i think you didn't quite understand my question.?Custom Ring Back Tones (Early Media) is one of the features listed in FreeSWITCH features page and details of??FreeSWITCH?Early Media mode is detailed in?FreeSWITCH wiki?:?http://wiki.freeswitch.org/wiki/Early_Media

By pre-answer i mean picking a call in Early Media?mode (Early media is the exchange of information before establishment of a connection) which mean on early media mode, even though an audio is playing like Custom Ring Back Tones, the call is not connected so the caller is not billed.

The question again is how can the?FreeSWITCH?Early Media feature be used from mod_httapi?

Also, is there any?performance?benefit for using??mod_httapi over a framework like Plivo that provide RESTFul features to FreeSWITCH?

Regards

Karl C.
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From krice at freeswitch.org  Thu Feb  2 23:55:49 2012
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 02 Feb 2012 14:55:49 -0600
Subject: [Freeswitch-dev] mod_httapi
In-Reply-To: <1328215375.43945.YahooMailNeo@web121603.mail.ne1.yahoo.com>
Message-ID: <CB505475.2B659%krice@freeswitch.org>

On the performance thing... Whats faster... FS asking directly or FS going
thru a translation layer to ask?


On 2/2/12 2:42 PM, "Karl Seacrest" <karlseacrest at yahoo.com> wrote:

> Thanks for your reply Anthony,
> But i think you didn't quite understand my question. Custom Ring Back Tones
> (Early Media) is one of the features listed in FreeSWITCH features page and
> details of  FreeSWITCH Early Media mode is detailed in FreeSWITCH wiki :
> http://wiki.freeswitch.org/wiki/Early_Media
> 
> By pre-answer i mean picking a call in Early Media mode (Early media is the
> exchange of information before establishment of a connection) which mean on
> early media mode, even though an audio is playing like Custom Ring Back Tones,
> the call is not connected so the caller is not billed.
> 
> The question again is how can the FreeSWITCH Early Media feature be used from
> mod_httapi?
> 
> Also, is there any performance benefit for using  mod_httapi over a framework
> like Plivo that provide RESTFul features to FreeSWITCH?
> 
> Regards
> 
> Karl C.
>   
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 karlseacrest at yahoo.com  Fri Feb  3 00:08:34 2012
From: karlseacrest at yahoo.com (Karl Seacrest)
Date: Thu, 2 Feb 2012 13:08:34 -0800 (PST)
Subject: [Freeswitch-dev] mod_httapi
In-Reply-To: <mailman.2156.1328215407.13443.freeswitch-dev@lists.freeswitch.org>
References: <mailman.2156.1328215407.13443.freeswitch-dev@lists.freeswitch.org>
Message-ID: <1328216914.70737.YahooMailNeo@web121603.mail.ne1.yahoo.com>

Thanks?Michael Collins,
I am still laughing about your point: Me?you telling Anthony what early media is kind of like telling a neurosurgeon that a person's brain is inside the cranium. :P
Seeing the way you put it makes me laugh.?sorry??Anthony, i am not trying to educated you about?Early Media.

Michael??as regards your question, i am developing a premium customer support application whereby registered members can call-in and the customer support staffs will pick the call. But before the call will connect with to live support. I want the caller id to be checked from the database if it's registered and active or not. if registered, call is connect to live support but if not, an audio will be played to the caller telling them they are not registered for the service and disconnect without the caller being charged.


The only time the call will be charged is when the call connected to live support. so i am thinking using the early media mode will help me achieve this but i am not sure how to go about it using the??mod_httapi.

My second question is, is there any?performance?benefit for using??mod_httapi over a framework like Plivo that provide RESTFul features to FreeSWITCH?

Karl C.




________________________________
 From: "freeswitch-dev-request at lists.freeswitch.org" <freeswitch-dev-request at lists.freeswitch.org>
To: freeswitch-dev at lists.freeswitch.org 
Sent: Thursday, February 2, 2012 9:43 PM
Subject: FreeSWITCH-dev Digest, Vol 68, Issue 6
 
----- Forwarded Message -----

Send FreeSWITCH-dev mailing list submissions to
??? freeswitch-dev at lists.freeswitch.org

To subscribe or unsubscribe via the World Wide Web, visit
??? http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
or, via email, send a message with subject or body 'help' to
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You can reach the person managing the list at
??? freeswitch-dev-owner at lists.freeswitch.org

When replying, please edit your Subject line so it is more specific
than "Re: Contents of FreeSWITCH-dev digest..."

Today's Topics:

?  1. Re: mod_httapi (Michael Collins)
?  2. switch_ivr_bridge.c question (teldev)
?  3. Re: switch_ivr_bridge.c question (Anthony Minessale)
?  4. mod_httapi (Karl Seacrest)

Actually, Anthony understood your question and gave you a solution with bonus information. I especially liked the part where you tell Anthony what early media is. That's kind of like telling a neurosurgeon that a person's brain is inside the cranium. :P

If you issue a playback without actually answering the call then the net effect is that you are pre_answering the call and the file you are playing is in early media, i.e. you have not actually "answered" the call from a billing perspective. From the caller perspective the file you play is the ringback that they will hear.

Is there something you were curious about beyond sending audio to the caller in early media mode?

-MC


On Thu, Feb 2, 2012 at 1:15 AM, Karl Seacrest <karlseacrest at yahoo.com> wrote:

Thanks for your reply?Anthony,?
>
>But i think you didn't quite understand my question.?Custom Ring Back Tones (Early Media) is one of the features listed in FreeSWITCH features page and details of??FreeSWITCH?Early Media mode is detailed in?FreeSWITCH wiki?:?http://wiki.freeswitch.org/wiki/Early_Media
>
>
>By pre-answer i mean picking a call in
Early Media?mode (Early media is the exchange of information before establishment of a connection) which mean on early media mode, even though an audio is playing like Custom Ring Back Tones, the call is not connected so the caller is not billed.
>
>
>The question again is how can the?FreeSWITCH?Early Media feature be used from mod_httapi?
>
>
>Also, is there any?performance?benefit for using??mod_httapi over a framework like Plivo that provide RESTFul features to FreeSWITCH?
>
>
>Regards
>
>
>Karl C.
>
>
>
>
>
>
>
>
>
>
>
>
>
>________________________________
> 
>On Wed, Feb 1, 2012 at 16:05, Anthony Minessale <anthony.minessale at gmail.com>wrote:
>
>
>
>if you try to play a file it will pre-answer by itselfyou can send anything you want via the dialplan inside the {foo=bar}
>var expressions and if you enable the permissions you can get the full
>call data
>
>On Wed, Feb 1, 2012 at 10:41 AM, Karl Seacrest <karlseacrest at yahoo.com> wrote:
>> Hello,
>> I will like to know how i can use Pre-Answer feature using the
 mod_httapi.
>> This is useful to perform operations like playing?playing custom caller
>> tunes or validate the callers phone number from database before the call is
>> answered.
>>
>> How can i achieve this using the FreeSWITCH mod_httapi?
>>
>> Also does?mod_httapi send caller information (Caller ID, etc) along when
>> requesting a URI?
>>
>>
>> Regards
>> Karl C.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
>
>-- 
>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
>
>
>
>Hi team,
>
>
>I am trying to get these values from the lua script for?custom?CDR report, I am able to get few values but many variables are missing or getting null value. Any help would be appreciated.
>
>
>local aa=session:getVariable("context");
>local bb=session:getVariable("destination_number");
>local cc=session:getVariable("caller_id_name");
>local dd=session:getVariable("caller_id_number");
>local ee=session:getVariable("network_addr");
>local ff=session:getVariable("ani");
>local ii=session:getVariable("source");
>local jj=session:getVariable("chan_name");
>local kk=session:getVariable("uuid");
>local ll=session:getVariable("created_time");
>local aa1=session:getVariable("direction");
>local bb1=session:getVariable("username");
>local cc1=session:getVariable("dialplan");
>local dd1=session:getVariable("caller_id_number");
>local ee1=session:getVariable("unique_id");
>
>
>session:execute("bridge","user/601");
>session:hangup();
>
>
>local ff1=session:getVariable("answered_time");
>local hh1=session:getVariable("created_time");
>local gg1=session:getVariable("hangup_time");
>local gg2=session:getVariable("end_stamp");
>
>
>I am getting values for all the above variables except end_stamp, hangup_time, duration, billsec,hangup_cause.
>
>
>
>
>Thanks for your time.
>
>
>Thanks,
>Bala
>
>
>
>
>
>
>_______________________________________________
>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
>
>
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.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
>
>
Re-worded my original post a bit, sorry if it becomes a new thread...having
Nabble issues.

In switch_ivr_bridge.c on line 1474, we have the following lines of code:

1474? ? ? ? ? ? ? ? ? 
switch_channel_clear_state_handler(originator_channel, NULL);
1475? ? ? ? ? ? ? ? ? 
switch_channel_clear_state_handler(originatee_channel, NULL);
1476? ? ? ? ? ? ? ? ? ? switch_channel_set_state_flag(originator_channel,
CF_BRIDGE_ORIGINATOR);
1477? ? ? ? ? ? ? ? ? 
switch_channel_clear_flag_recursive(originatee_channel,
CF_BRIDGE_ORIGINATOR);
1478? ? ? ? ? ? ? ? ? 
switch_channel_add_state_handler(originator_channel,
&uuid_bridge_state_handlers);
1479? ? ? ? ? ? ? ? ? 
switch_channel_add_state_handler(originatee_channel,
&uuid_bridge_state_handlers);

Is the call to ?switch_channel_set_state_flag? a typo? Since line 1476
triggers an asynchronous flag change, and line 1477 changes the same flag
synchronously, on line 1478 the value of the flag becomes unpredictable
since the asynchronous change from line 1476 can happen before or after line
1477 (mostly after, of course).

Thanks again.

--
View this message in context: http://freeswitch-dev.4355.n6.nabble.com/switch-ivr-bridge-c-question-tp4359635p4359635.html
Sent from the FreeSWITCH-dev mailing list archive at Nabble.com.


The code is not the same on the latest.
Look closer ;)

You should probably keep yourself on the latest for this deep of an examination.



On Thu, Feb 2, 2012 at 11:43 AM, teldev <ustcorporation at yahoo.com> wrote:
> Re-worded my original post a bit, sorry if it becomes a new thread...having
> Nabble issues.
>
> In switch_ivr_bridge.c on line 1474, we have the following lines of code:
>
> ?1474
> switch_channel_clear_state_handler(originator_channel, NULL);
> ?1475
> switch_channel_clear_state_handler(originatee_channel, NULL);
> ?1476 ? ? ? ? ? ? ? ? ? ?switch_channel_set_state_flag(originator_channel,
> CF_BRIDGE_ORIGINATOR);
> ?1477
> switch_channel_clear_flag_recursive(originatee_channel,
> CF_BRIDGE_ORIGINATOR);
> ?1478
> switch_channel_add_state_handler(originator_channel,
> &uuid_bridge_state_handlers);
> ?1479
> switch_channel_add_state_handler(originatee_channel,
> &uuid_bridge_state_handlers);
>
> Is the call to ?switch_channel_set_state_flag? a typo? Since line 1476
> triggers an asynchronous flag change, and line 1477 changes the same flag
> synchronously, on line 1478 the value of the flag becomes unpredictable
> since the asynchronous change from line 1476 can happen before or after line
> 1477 (mostly after, of course).
>
> Thanks again.
>
> --
> View this message in context: http://freeswitch-dev.4355.n6.nabble.com/switch-ivr-bridge-c-question-tp4359635p4359635.html
> Sent from the FreeSWITCH-dev mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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



-- 
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



Thanks for your reply?Anthony,?

But i think you didn't quite understand my question.?Custom Ring Back Tones (Early Media) is one of the features listed in FreeSWITCH features page and details of??FreeSWITCH?Early Media mode is detailed in?FreeSWITCH wiki?:?http://wiki.freeswitch.org/wiki/Early_Media

By pre-answer i mean picking a call in Early Media?mode (Early media is the exchange of information before establishment of a connection) which mean on early media mode, even though an audio is playing like Custom Ring Back Tones, the call is not connected so the caller is not billed.

The question again is how can the?FreeSWITCH?Early Media feature be used from mod_httapi?

Also, is there any?performance?benefit for using??mod_httapi over a framework like Plivo that provide RESTFul features to FreeSWITCH?

Regards

Karl C.
_______________________________________________
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 ustcorporation at yahoo.com  Fri Feb  3 01:08:11 2012
From: ustcorporation at yahoo.com (teldev)
Date: Thu, 2 Feb 2012 14:08:11 -0800 (PST)
Subject: [Freeswitch-dev] switch_ivr_bridge.c question
In-Reply-To: <CAKbxfG-G7JN6-1hUUsxXnh8Pw_bxXjgcXZw8P6EzN=feTDcJwQ@mail.gmail.com>
References: <1328204583380-4359635.post@n6.nabble.com>
	<CAKbxfG-G7JN6-1hUUsxXnh8Pw_bxXjgcXZw8P6EzN=feTDcJwQ@mail.gmail.com>
Message-ID: <1328220491160-4360149.post@n6.nabble.com>

Hello Anthony,

Yes, my line numbers were off by 4 after getting latest GIT from the master
branch.  The code is the same there.  If you can take a look at it now, here
is the refreshed version with correct line numbers:

1470  switch_channel_clear_state_handler(originator_channel, NULL);
1471  switch_channel_clear_state_handler(originatee_channel, NULL);
1472  switch_channel_set_state_flag(originator_channel,
CF_BRIDGE_ORIGINATOR);
1473  switch_channel_clear_flag_recursive(originatee_channel,
CF_BRIDGE_ORIGINATOR);
1474  switch_channel_add_state_handler(originator_channel,
&uuid_bridge_state_handlers);
1475  switch_channel_add_state_handler(originatee_channel,
&uuid_bridge_state_handlers);

Thanks in advance.


--
View this message in context: http://freeswitch-dev.4355.n6.nabble.com/switch-ivr-bridge-c-question-tp4359635p4360149.html
Sent from the FreeSWITCH-dev mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Fri Feb  3 01:19:55 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 2 Feb 2012 16:19:55 -0600
Subject: [Freeswitch-dev] switch_ivr_bridge.c question
In-Reply-To: <1328220491160-4360149.post@n6.nabble.com>
References: <1328204583380-4359635.post@n6.nabble.com>
	<CAKbxfG-G7JN6-1hUUsxXnh8Pw_bxXjgcXZw8P6EzN=feTDcJwQ@mail.gmail.com>
	<1328220491160-4360149.post@n6.nabble.com>
Message-ID: <CAKbxfG_ZKe9MFBxGcyoy4LuEu4qPZ5iQn18ZuYQHs+KRiA_HoA@mail.gmail.com>

not even close?

Where are you getting this from?
you need GIT HEAD from our standard git repo.


On Thu, Feb 2, 2012 at 4:08 PM, teldev <ustcorporation at yahoo.com> wrote:
> Hello Anthony,
>
> Yes, my line numbers were off by 4 after getting latest GIT from the master
> branch. ?The code is the same there. ?If you can take a look at it now, here
> is the refreshed version with correct line numbers:
>
> 1470 ?switch_channel_clear_state_handler(originator_channel, NULL);
> 1471 ?switch_channel_clear_state_handler(originatee_channel, NULL);
> 1472 ?switch_channel_set_state_flag(originator_channel,
> CF_BRIDGE_ORIGINATOR);
> 1473 ?switch_channel_clear_flag_recursive(originatee_channel,
> CF_BRIDGE_ORIGINATOR);
> 1474 ?switch_channel_add_state_handler(originator_channel,
> &uuid_bridge_state_handlers);
> 1475 ?switch_channel_add_state_handler(originatee_channel,
> &uuid_bridge_state_handlers);
>
> Thanks in advance.
>
>
> --
> View this message in context: http://freeswitch-dev.4355.n6.nabble.com/switch-ivr-bridge-c-question-tp4359635p4360149.html
> Sent from the FreeSWITCH-dev mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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



-- 
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  Sat Feb  4 02:55:04 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 3 Feb 2012 15:55:04 -0800
Subject: [Freeswitch-dev] It's that time of year: Buy the devs dinner!
Message-ID: <CAKzWOxVUhUagmqbePx8L=S5z4-jSsyTa0B=tw+hag7fo_-Mvmw@mail.gmail.com>

Hello all!

Each February the FreeSWITCH team converges on a small Milwaukee suburb to
plot and plan how we will continue to conquer the world of open source
telephony. From personal experience I can attest to the fact that it's
really difficult to conquer the world on an empty stomach. :)

We'd like to invite everyone to throw a few bucks into the hat to buy
dinner for the FreeSWITCH core development team. The easiest way to donate
is to hit the gold "donate" button on the right-hand side of the main
FreeSWITCH page. Be sure to include a memo line, something like "bon
appetit," so that we can keep track.

Thanks again for supporting the FreeSWITCH team! We have an awesome
community and you continue to show it on a daily basis.

-Michael
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From brian at freeswitch.org  Mon Feb  6 20:25:01 2012
From: brian at freeswitch.org (Brian West)
Date: Mon, 6 Feb 2012 11:25:01 -0600
Subject: [Freeswitch-dev] I'm Alive!
Message-ID: <E6B42012-A660-4F56-8A18-2EDAAA3EABC4@freeswitch.org>

I would like to thank everyone that has donated money to help me pay for all the expense of having a stay in the hospital.  It was a pretty scary week for me last week.  If it were not for the morphine last week would have sucked even more!  I have missed out on the annual Developer meeting in Milwaukee this year due to this event.  So please make sure you guys donate some money for the dinner for those guys. 

Again Thank you all for your support.

-- 
Brian West 
FreeSWITCH Solutions, LLC
Phone: +1 (918) 420-9266 
Fax:   +1 (918) 420-9267
brian at freeswitch.org
http://www.freeswitch.org

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From msc at freeswitch.org  Tue Feb  7 23:17:25 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 7 Feb 2012 12:17:25 -0800
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Tomorrow
Message-ID: <CAKzWOxWVP7ZFwk_7_AdiCTC5AMw1UDFnaQpEb-o7UEANze1_fw@mail.gmail.com>

Hello all!

Just a reminder that we are having Travis Cross come back and talk to us
about git again. I think we've all had some time to digest the information
from his first presentation and hopefully we have some intelligent
questions for him. I'm wondering if he can show us an example of git
bisect. :)

Here's the agenda page:
http://wiki.freeswitch.org/wiki/FS_weekly_2012_02_08

FYI, I am trying to get caught up on the release notes so I've found a few
new FreeSWITCH things that I'd like to have you all help me get documented.
:)

Thanks,
Michael
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From appalonovandrey at gmail.com  Wed Feb  8 09:41:05 2012
From: appalonovandrey at gmail.com (Andrey Appalonov)
Date: Wed, 8 Feb 2012 12:41:05 +0600
Subject: [Freeswitch-dev] (no subject)
Message-ID: <CACJEBkye0ri9En4KOYhRyqaA6cuQbE2Job=WhkR9epJVaA6c+g@mail.gmail.com>

Hello, please tell me why the FS can not hear
 soft-fax (VentaFax Minioffice)? Fax automatically picks it up,
 but in FS it is not visible ..

PS: If someone has worked with a soft-fax, what do you
 used?

Best regards,
Appalonov Andrey.


From dujinfang at gmail.com  Wed Feb  8 16:55:22 2012
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 8 Feb 2012 21:55:22 +0800
Subject: [Freeswitch-dev] backslash in call-id
Message-ID: <DA69EC1B2C7D4B47921EC2CCC80EDE61@gmail.com>

Hi,

I have a sip client which generates call-id with backslashes which causing error:

ERR: [insert into sip_registrations ............. values ('<c)X/Iz\'


I looked into

http://sofia-sip.sourceforge.net/refdocs/sip/group__sip__call__id.html

looks like backslash '\' is valid in call-id

so is it ok to to let switch_mprintf %q also quote backslash ?

should I report a jira?

Thanks. 

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

Sent with Sparrow (http://www.sparrowmailapp.com)

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From krice at freeswitch.org  Wed Feb  8 17:58:13 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 08 Feb 2012 08:58:13 -0600
Subject: [Freeswitch-dev] backslash in call-id
In-Reply-To: <DA69EC1B2C7D4B47921EC2CCC80EDE61@gmail.com>
Message-ID: <CB57E9A5.2BCCC%krice@freeswitch.org>

Yes you should open a Jira, attach a proposed patch if you have one...

Reporting bugs like this on the mailing list almost always get lost in the
shuffle



On 2/8/12 7:55 AM, "Seven Du" <dujinfang at gmail.com> wrote:

>  
> Hi,
> 
> I have a sip client which generates call-id with backslashes which causing
> error:
> 
> ERR: [insert into sip_registrations ............. values ('<c)X/Iz\'
> 
> 
> I looked into
> 
> http://sofia-sip.sourceforge.net/refdocs/sip/group__sip__call__id.html
> 
> looks like backslash '\' is valid in call-id
> 
> so is it ok to to let switch_mprintf %q also quote backslash ?
> 
> should I report a jira?
> 
> Thanks.
>  

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From dujinfang at gmail.com  Thu Feb  9 08:38:09 2012
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 9 Feb 2012 13:38:09 +0800
Subject: [Freeswitch-dev] backslash in call-id
In-Reply-To: <CB57E9A5.2BCCC%krice@freeswitch.org>
References: <CB57E9A5.2BCCC%krice@freeswitch.org>
Message-ID: <54FF4A22161D48E8B36202BD171D0EE8@gmail.com>

Thanks Ken, I don't have a patch, ira opened http://jira.freeswitch.org/browse/FS-3893


On Wednesday, February 8, 2012 at 10:58 PM, Ken Rice wrote:

> Re: [Freeswitch-dev] backslash in call-id Yes you should open a Jira, attach a proposed patch if you have one...
> 
> Reporting bugs like this on the mailing list almost always get lost in the shuffle
> 
> 
> 
> On 2/8/12 7:55 AM, "Seven Du" <dujinfang at gmail.com> wrote:
> 
> > 
> > Hi,
> > 
> > I have a sip client which generates call-id with backslashes which causing error:
> > 
> > ERR: [insert into sip_registrations ............. values ('<c)X/Iz\'
> > 
> > 
> > I looked into
> > 
> > http://sofia-sip.sourceforge.net/refdocs/sip/group__sip__call__id.html
> > 
> > looks like backslash '\' is valid in call-id
> > 
> > so is it ok to to let switch_mprintf %q also quote backslash ?
> > 
> > should I report a jira?
> > 
> > Thanks.
> >  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org (mailto: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 dujinfang at gmail.com  Thu Feb  9 17:32:11 2012
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 9 Feb 2012 22:32:11 +0800
Subject: [Freeswitch-dev] no video in mod_fifo
Message-ID: <1C2E6F90C3044A9BB5CDC6FC31C5992B@gmail.com>

Hi, 


There's no video when bridged calls through fifo, I found no video support in the following code.


I made a work around by setting sip_use_codec_name="PCMU,H264"

question is will this causing problem? what's the purpose of the following code? how can I add video support ?

Thanks.



if ((codec = switch_event_get_header(pop, "variable_sip_use_codec_name"))) {
const char *rate = switch_event_get_header(pop, "variable_sip_use_codec_rate");
const char *ptime = switch_event_get_header(pop, "variable_sip_use_codec_ptime");
char nstr[256] = "";

if (strcasecmp(codec, "PCMU") && strcasecmp(codec, "PCMA")) {
switch_snprintf(nstr, sizeof(nstr), "%s@%si@%sh,PCMU@%si,PCMA@%si", codec, ptime, rate, ptime, ptime);
} else {
switch_snprintf(nstr, sizeof(nstr), "%s@%si@%sh", codec, ptime, rate);
}

switch_event_add_header_string(ovars, SWITCH_STACK_BOTTOM, "absolute_codec_string", nstr);
}



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

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From msc at freeswitch.org  Thu Feb  9 22:34:59 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 9 Feb 2012 11:34:59 -0800
Subject: [Freeswitch-dev] mod_httapi
In-Reply-To: <1328216914.70737.YahooMailNeo@web121603.mail.ne1.yahoo.com>
References: <mailman.2156.1328215407.13443.freeswitch-dev@lists.freeswitch.org>
	<1328216914.70737.YahooMailNeo@web121603.mail.ne1.yahoo.com>
Message-ID: <CAKzWOxUXa6sh72FsSR3GHNv3gscq6H3D2sDyuPZVW_=RpEcmTQ@mail.gmail.com>

Karl,

Have you done any more work on this? I started working on it, found a bug,
reported it, and tinkered a bit after Tony fixed the bug. However I've had
a lot going on with the day job, FreeSWITCH cookbook, etc. so I haven't
done much more.

Let us know where you stand on this.

-MC

On Thu, Feb 2, 2012 at 1:08 PM, Karl Seacrest <karlseacrest at yahoo.com>wrote:

> Thanks Michael Collins,
> I am still laughing about your point: Me you telling Anthony what early
> media is kind of like telling a neurosurgeon that a person's brain is
> inside the cranium. :P
> Seeing the way you put it makes me laugh. sorry  Anthony, i am not trying
> to educated you about Early Media.
>
> Michael  as regards your question, i am developing a premium customer
> support application whereby registered members can call-in and the customer
> support staffs will pick the call. But before the call will connect with to
> live support. I want the caller id to be checked from the database if it's
> registered and active or not. if registered, call is connect to live
> support but if not, an audio will be played to the caller telling them they
> are not registered for the service and disconnect without the caller being
> charged.
>
> The only time the call will be charged is when the call connected to live
> support. so i am thinking using the early media mode will help me achieve
> this but i am not sure how to go about it using the  mod_httapi.
>
> My second question is, is there any performance benefit for using  mod_httapi
> over a framework like Plivo that provide RESTFul features to FreeSWITCH?
>
> Karl C.
>
>
>
>   ------------------------------
> *From:* "freeswitch-dev-request at lists.freeswitch.org" <
> freeswitch-dev-request at lists.freeswitch.org>
> *To:* freeswitch-dev at lists.freeswitch.org
> *Sent:* Thursday, February 2, 2012 9:43 PM
> *Subject:* FreeSWITCH-dev Digest, Vol 68, Issue 6
>
> ----- Forwarded Message -----
>
> Send FreeSWITCH-dev mailing list submissions to
>     freeswitch-dev at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> or, via email, send a message with subject or body 'help' to
>     freeswitch-dev-request at lists.freeswitch.org
>
> You can reach the person managing the list at
>     freeswitch-dev-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-dev digest..."
>
> Today's Topics:
>
>   1. Re: mod_httapi (Michael Collins)
>   2. switch_ivr_bridge.c question (teldev)
>   3. Re: switch_ivr_bridge.c question (Anthony Minessale)
>   4. mod_httapi (Karl Seacrest)
> Actually, Anthony understood your question and gave you a solution with
> bonus information. I especially liked the part where you tell Anthony what
> early media is. That's kind of like telling a neurosurgeon that a person's
> brain is inside the cranium. :P
>
> If you issue a playback without actually answering the call then the net
> effect is that you are pre_answering the call and the file you are playing
> is in early media, i.e. you have not actually "answered" the call from a
> billing perspective. From the caller perspective the file you play is the
> ringback that they will hear.
>
> Is there something you were curious about beyond sending audio to the
> caller in early media mode?
>
> -MC
>
> On Thu, Feb 2, 2012 at 1:15 AM, Karl Seacrest <karlseacrest at yahoo.com>wrote:
>
> Thanks for your reply Anthony,
> But i think you didn't quite understand my question. Custom Ring Back
> Tones (Early Media) is one of the features listed in FreeSWITCH features
> page and details of  FreeSWITCH Early Media mode is detailed in FreeSWITCH
> wiki : http://wiki.freeswitch.org/wiki/Early_Media
>
> By pre-answer i mean picking a call in Early Media mode (Early media is
> the exchange of information before establishment of a connection) which
> mean on early media mode, even though an audio is playing like Custom Ring
> Back Tones, the call is not connected so the caller is not billed.
>
> The question again is how can the FreeSWITCH Early Media feature be used
> from mod_httapi?
>
> Also, is there any performance benefit for using  mod_httapi over a
> framework like Plivo that provide RESTFul features to FreeSWITCH?
>
> Regards
>
> Karl C.
>
>
>
>
>
>
>   ------------------------------
> On Wed, Feb 1, 2012 at 16:05, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
> if you try to play a file it will pre-answer by itself
> you can send anything you want via the dialplan inside the {foo=bar}
> var expressions and if you enable the permissions you can get the full
> call data
>
> On Wed, Feb 1, 2012 at 10:41 AM, Karl Seacrest <karlseacrest at yahoo.com>
> wrote:
> > Hello,
> > I will like to know how i can use Pre-Answer feature using the
> mod_httapi.
> > This is useful to perform operations like playing playing custom caller
> > tunes or validate the callers phone number from database before the call
> is
> > answered.
> >
> > How can i achieve this using the FreeSWITCH mod_httapi?
> >
> > Also does mod_httapi send caller information (Caller ID, etc) along when
> > requesting a URI?
> >
> >
> > Regards
> > Karl C.
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> >
>
>
>
> --
> 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
>
>
> Hi team,
>
> I am trying to get these values from the lua script for custom CDR report,
> I am able to get few values but many variables are missing or getting null
> value. Any help would be appreciated.
>
> local aa=session:getVariable("context");
> local bb=session:getVariable("destination_number");
> local cc=session:getVariable("caller_id_name");
> local dd=session:getVariable("caller_id_number");
> local ee=session:getVariable("network_addr");
> local ff=session:getVariable("ani");
> local ii=session:getVariable("source");
> local jj=session:getVariable("chan_name");
> local kk=session:getVariable("uuid");
> local ll=session:getVariable("created_time");
> local aa1=session:getVariable("direction");
> local bb1=session:getVariable("username");
> local cc1=session:getVariable("dialplan");
> local dd1=session:getVariable("caller_id_number");
> local ee1=session:getVariable("unique_id");
>
> session:execute("bridge","user/601");
> session:hangup();
>
> local ff1=session:getVariable("answered_time");
> local hh1=session:getVariable("created_time");
> local gg1=session:getVariable("hangup_time");
> local gg2=session:getVariable("end_stamp");
>
> I am getting values for all the above variables except end_stamp,
> hangup_time, duration, billsec,hangup_cause.
>
>
> Thanks for your time.
>
> Thanks,
> Bala
>
>
>
>
>
>
>
> _______________________________________________
> 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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> Re-worded my original post a bit, sorry if it becomes a new thread...having
> Nabble issues.
>
> In switch_ivr_bridge.c on line 1474, we have the following lines of code:
>
> 1474
> switch_channel_clear_state_handler(originator_channel, NULL);
> 1475
> switch_channel_clear_state_handler(originatee_channel, NULL);
> 1476                    switch_channel_set_state_flag(originator_channel,
> CF_BRIDGE_ORIGINATOR);
> 1477
> switch_channel_clear_flag_recursive(originatee_channel,
> CF_BRIDGE_ORIGINATOR);
> 1478
> switch_channel_add_state_handler(originator_channel,
> &uuid_bridge_state_handlers);
> 1479
> switch_channel_add_state_handler(originatee_channel,
> &uuid_bridge_state_handlers);
>
> Is the call to ?switch_channel_set_state_flag? a typo? Since line 1476
> triggers an asynchronous flag change, and line 1477 changes the same flag
> synchronously, on line 1478 the value of the flag becomes unpredictable
> since the asynchronous change from line 1476 can happen before or after
> line
> 1477 (mostly after, of course).
>
> Thanks again.
>
> --
> View this message in context:
> http://freeswitch-dev.4355.n6.nabble.com/switch-ivr-bridge-c-question-tp4359635p4359635.html
> Sent from the FreeSWITCH-dev mailing list archive at Nabble.com.
>
>
> The code is not the same on the latest.
> Look closer ;)
>
> You should probably keep yourself on the latest for this deep of an
> examination.
>
>
>
> On Thu, Feb 2, 2012 at 11:43 AM, teldev <ustcorporation at yahoo.com> wrote:
> > Re-worded my original post a bit, sorry if it becomes a new
> thread...having
> > Nabble issues.
> >
> > In switch_ivr_bridge.c on line 1474, we have the following lines of code:
> >
> >  1474
> > switch_channel_clear_state_handler(originator_channel, NULL);
> >  1475
> > switch_channel_clear_state_handler(originatee_channel, NULL);
> >  1476
>  switch_channel_set_state_flag(originator_channel,
> > CF_BRIDGE_ORIGINATOR);
> >  1477
> > switch_channel_clear_flag_recursive(originatee_channel,
> > CF_BRIDGE_ORIGINATOR);
> >  1478
> > switch_channel_add_state_handler(originator_channel,
> > &uuid_bridge_state_handlers);
> >  1479
> > switch_channel_add_state_handler(originatee_channel,
> > &uuid_bridge_state_handlers);
> >
> > Is the call to ?switch_channel_set_state_flag? a typo? Since line 1476
> > triggers an asynchronous flag change, and line 1477 changes the same flag
> > synchronously, on line 1478 the value of the flag becomes unpredictable
> > since the asynchronous change from line 1476 can happen before or after
> line
> > 1477 (mostly after, of course).
> >
> > Thanks again.
> >
> > --
> > View this message in context:
> http://freeswitch-dev.4355.n6.nabble.com/switch-ivr-bridge-c-question-tp4359635p4359635.html
> > Sent from the FreeSWITCH-dev mailing list archive at Nabble.com.
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
>
>
>
> --
> 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
>
>
> Thanks for your reply Anthony,
> But i think you didn't quite understand my question. Custom Ring Back
> Tones (Early Media) is one of the features listed in FreeSWITCH features
> page and details of  FreeSWITCH Early Media mode is detailed in FreeSWITCH
> wiki : http://wiki.freeswitch.org/wiki/Early_Media
>
> By pre-answer i mean picking a call in Early Media mode (Early media is
> the exchange of information before establishment of a connection) which
> mean on early media mode, even though an audio is playing like Custom Ring
> Back Tones, the call is not connected so the caller is not billed.
>
> The question again is how can the FreeSWITCH Early Media feature be used
> from mod_httapi?
>
> Also, is there any performance benefit for using  mod_httapi over a
> framework like Plivo that provide RESTFul features to FreeSWITCH?
>
> Regards
>
> Karl C.
>
> _______________________________________________
> 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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 ustcorporation at yahoo.com  Sat Feb 11 04:53:39 2012
From: ustcorporation at yahoo.com (teldev)
Date: Fri, 10 Feb 2012 17:53:39 -0800 (PST)
Subject: [Freeswitch-dev] switch_ivr_bridge.c question
In-Reply-To: <CAKbxfG_ZKe9MFBxGcyoy4LuEu4qPZ5iQn18ZuYQHs+KRiA_HoA@mail.gmail.com>
References: <1328204583380-4359635.post@n6.nabble.com>
	<CAKbxfG-G7JN6-1hUUsxXnh8Pw_bxXjgcXZw8P6EzN=feTDcJwQ@mail.gmail.com>
	<1328220491160-4360149.post@n6.nabble.com>
	<CAKbxfG_ZKe9MFBxGcyoy4LuEu4qPZ5iQn18ZuYQHs+KRiA_HoA@mail.gmail.com>
Message-ID: <1328925219958-4385158.post@n6.nabble.com>

Anthony,

Sorry, I have several versions and I was indeed looking at the wrong one!

Thanks for pointing this out.

--
View this message in context: http://freeswitch-dev.4355.n6.nabble.com/switch-ivr-bridge-c-question-tp4359635p4385158.html
Sent from the FreeSWITCH-dev mailing list archive at Nabble.com.


From mgg at giagnocavo.net  Tue Feb 14 19:40:57 2012
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Tue, 14 Feb 2012 11:40:57 -0500
Subject: [Freeswitch-dev] backslash in call-id
In-Reply-To: <DA69EC1B2C7D4B47921EC2CCC80EDE61@gmail.com>
References: <DA69EC1B2C7D4B47921EC2CCC80EDE61@gmail.com>
Message-ID: <83FF8D7C9F526E44B77C97DD2891652A12EA5906@mse17be1.mse17.exchange.ms>

If you quote backslash, you corrupt data for databases that follow the SQL standard and don?t use backslash. <snarky MySQL comment elided>

This is really more a problem with using a ?quote? function instead of proper parameterized queries. I don?t believe it?s possible to accommodate both escaping methods with the same function.

If parameterized queries aren?t possible to do in FS (or too much work), there should probably be some switch that controls backslash handling.

-Michael


From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Seven Du
Sent: Wednesday, February 08, 2012 6:55 AM
To: freeswitch-dev
Subject: [Freeswitch-dev] backslash in call-id

Hi,

I have a sip client which generates call-id with backslashes which causing error:

ERR: [insert into sip_registrations ............. values ('<c)X/Iz\'


I looked into

http://sofia-sip.sourceforge.net/refdocs/sip/group__sip__call__id.html

looks like backslash '\' is valid in call-id

so is it ok to to let switch_mprintf %q also quote backslash ?

should I report a jira?

Thanks.

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

Sent with Sparrow<http://www.sparrowmailapp.com>

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From anthony.minessale at gmail.com  Tue Feb 14 20:16:51 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 14 Feb 2012 11:16:51 -0600
Subject: [Freeswitch-dev] backslash in call-id
In-Reply-To: <83FF8D7C9F526E44B77C97DD2891652A12EA5906@mse17be1.mse17.exchange.ms>
References: <DA69EC1B2C7D4B47921EC2CCC80EDE61@gmail.com>
	<83FF8D7C9F526E44B77C97DD2891652A12EA5906@mse17be1.mse17.exchange.ms>
Message-ID: <CAKbxfG8VOABYT1stAYv1smE9DoP2x-Z6RRyDQtnCbCN9dR--XA@mail.gmail.com>

This turned out to be postgres defaulting to respecting to
non-stansard quoted backslash for back-compat or something.
There is a param to disable it which fixes the problem.


On Tue, Feb 14, 2012 at 10:40 AM, Michael Giagnocavo <mgg at giagnocavo.net> wrote:
> If you quote backslash, you corrupt data for databases that follow the SQL
> standard and don?t use backslash. <snarky MySQL comment elided>
>
> This is really more a problem with using a ?quote? function instead of
> proper parameterized queries. I don?t believe it?s possible to accommodate
> both escaping methods with the same function.
>
>
>
> If parameterized queries aren?t possible to do in FS (or too much work),
> there should probably be some switch that controls backslash handling.
>
>
>
> -Michael
>
>
>
>
>
> From: freeswitch-dev-bounces at lists.freeswitch.org
> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Seven Du
> Sent: Wednesday, February 08, 2012 6:55 AM
> To: freeswitch-dev
> Subject: [Freeswitch-dev] backslash in call-id
>
>
>
> Hi,
>
>
>
> I have a sip client which generates call-id with backslashes which causing
> error:
>
>
>
> ERR: [insert into sip_registrations ............. values ('<c)X/Iz\'
>
>
>
>
>
> I looked into
>
>
>
> http://sofia-sip.sourceforge.net/refdocs/sip/group__sip__call__id.html
>
>
>
> looks like backslash '\' is valid in call-id
>
>
>
> so is it ok to to let switch_mprintf %q also quote backslash ?
>
>
>
> should I report a jira?
>
>
>
> Thanks.
>
>
>
> --
>
> About: http://about.me/dujinfang
>
> Blog: http://www.dujinfang.com
>
> Proj: ?http://www.freeswitch.org.cn
>
>
>
> Sent with Sparrow
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
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 freefall1986 at gmail.com  Tue Feb 14 15:30:28 2012
From: freefall1986 at gmail.com (Arif Hossain)
Date: Tue, 14 Feb 2012 18:30:28 +0600
Subject: [Freeswitch-dev] Increasing Packetization time for g729 codec
Message-ID: <1329222628.3185.0.camel@arifLaptop>

I have tested that the packetization time (ptime) for g729 codec does go
beyond 50ms. I want to modify the code so freeswitch can repacketize the
rtp stream upto 200 ms.

What i'm trying to achieve is :


client-----150ms------>FREESWITCH------20m-------->IPTDM Terminations

		AND

IPTDM termination------20ms---->FREESWITCH-------->200ms--->client.


First one can be done through configuration. But second one is not yet
possible for my test environment.

I'm using sangoma codec card. It supports packetization time upto 200ms.

So please help me with some pointers about the freeswitch codebase to
achieve this. After modifying i will submit the patches. I have not
filed any bug report yet. I'm going to do that tod




From msc at freeswitch.org  Wed Feb 15 20:51:43 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 15 Feb 2012 09:51:43 -0800
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today: 2600Hz Team
Message-ID: <CAKzWOxXBOhLKyW22WmTUFKdy4pVLjSs8Wz1jL-EySgMqbLsrAg@mail.gmail.com>

Hello all!

Apologies for the late notice. When Brian West isn't at my beck and call I
just can't function. :)

Today's conference call will start shortly. Here's the agenda page:
http://wiki.freeswitch.org/wiki/FS_weekly_2012_02_15

We are looking forward to having Darren Schreiber (IRC: pyite) and the rest
of the 2600Hz team come talk to us about the state of their projects.

-MC
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From e.tylepov at btcom.kz  Wed Feb 15 12:30:46 2012
From: e.tylepov at btcom.kz (Erjan Tylepov)
Date: Wed, 15 Feb 2012 09:30:46 +0000
Subject: [Freeswitch-dev] mod_nibblebill
Message-ID: <20120215093046.e03a3597@mail.btcom.kz>

Hi all,


I'm using mod_nibblebill, in which there is a line of code "if (profile->times->answered < 1) {", that determines whether a call has been answered.



The abovesaid worked fine with a commit 883de14f3068d8cc269cccce3895effc750ef108. But after pulling a commit 303de52bad2d43efcfe41520e4afbae919cf4864, the module doesn't work as before, namely, after a call has been answered the field "times->answered" in a structure "switch_caller_profile" is still 0. Does anybody know what has changed?


Regards,
Yerzhan.
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From miha at softnet.si  Thu Feb 16 12:36:25 2012
From: miha at softnet.si (Miha Zoubek)
Date: Thu, 16 Feb 2012 10:36:25 +0100
Subject: [Freeswitch-dev] radius_cdr
Message-ID: <4F3CCE19.4040905@softnet.si>

Hi,

just one information about radius_cdr modul.

For A leg it sends stop&start packet and also same for B leg.
I have also one another softswitch that behaviour is just like on FS 
(stop&start for A and B leg).
The difference that I have noticed is that FS for A leg sends different 
session ID that for B leg. So, this mean that I will have two entries in 
my sql table for one call.
Other softswith send same session ID for a and b leg, so this mean one 
entry in sql table for one call.

So, what is right? Is it possible to change this behaviour on FS?

Thanks!

-- 
Best regards / Lep Pozdrav
Miha Zoubek
Softnet d.o.o.



From freefall1986 at gmail.com  Thu Feb 16 12:41:29 2012
From: freefall1986 at gmail.com (Arif Hossain)
Date: Thu, 16 Feb 2012 15:41:29 +0600
Subject: [Freeswitch-dev] How to understand codebase
Message-ID: <1329385289.4913.9.camel@arifLaptop>

I'm quite new to freeswitch. I have some experience in asterisk. But
Mostly about how to configure asterisk. The codebase of freeswitch is
quite large and seems overhelming at first sight. I'm a moderately good
c programmer but i don't have a lot of experience understanding a huge
codebase only by reading code. 

I saw freeswitch does not come with a "developer manual" or something
like that. Which would be very helpful. Because i do have a moderate
knowledge about sip and telephony technologies. 

So any other ways to understand freeswitch codebase(the core, and
important modules like media session handling etc) efficiently? I'm
willing to "learn" and contribute to the project. 

I know "knowledge" is itself a "resource". But any help regarding this
would be wonderful.

At this time i have checked out freeswitch code from git, loaded it in
eclipse, and trying to understand some core data structures. But I've
realized it would be very difficult to understand everything by myself.

Thanks in advance.
-- 
"You have a voice"
                                        ----The King's Speech

Public Key : $ gpg --keyserver keyserver.ubuntu.com --recv-key C88CFC23
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From fdelawarde at wirelessmundi.com  Thu Feb 16 13:00:37 2012
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Thu, 16 Feb 2012 11:00:37 +0100
Subject: [Freeswitch-dev] How to understand codebase
In-Reply-To: <1329385289.4913.9.camel@arifLaptop>
References: <1329385289.4913.9.camel@arifLaptop>
Message-ID: <1329386437.6226.233.camel@luna.madrid.commsmundi.com>

I would start by using FS extensively and looking at console logs or
playing with ESL to understand the logic (channel states, events, etc.).

Then try to understand well the code of small modules like
mod_blacklist, and slowly going to the core stuff from there.

FS code is in general really well put together and, compared to
Asterisk, I find it very easy to read and understand without
documentation. Well, maybe a few drawings could help to learn faster...

Have fun,
Fran?ois.

On Thu, 2012-02-16 at 15:41 +0600, Arif Hossain wrote: 
> I'm quite new to freeswitch. I have some experience in asterisk. But
> Mostly about how to configure asterisk. The codebase of freeswitch is
> quite large and seems overhelming at first sight. I'm a moderately good
> c programmer but i don't have a lot of experience understanding a huge
> codebase only by reading code. 
> 
> I saw freeswitch does not come with a "developer manual" or something
> like that. Which would be very helpful. Because i do have a moderate
> knowledge about sip and telephony technologies. 
> 
> So any other ways to understand freeswitch codebase(the core, and
> important modules like media session handling etc) efficiently? I'm
> willing to "learn" and contribute to the project. 
> 
> I know "knowledge" is itself a "resource". But any help regarding this
> would be wonderful.
> 
> At this time i have checked out freeswitch code from git, loaded it in
> eclipse, and trying to understand some core data structures. But I've
> realized it would be very difficult to understand everything by myself.
> 
> Thanks in advance.



From janvb at live.com  Thu Feb 16 13:10:31 2012
From: janvb at live.com (Jan Berger)
Date: Thu, 16 Feb 2012 11:10:31 +0100
Subject: [Freeswitch-dev] How to understand codebase
In-Reply-To: <1329386437.6226.233.camel@luna.madrid.commsmundi.com>
References: <1329385289.4913.9.camel@arifLaptop>,
	<1329386437.6226.233.camel@luna.madrid.commsmundi.com>
Message-ID: <SNT135-W8CEDAEDEC044FC97DDB97D2620@phx.gbl>


Reading a large code-base like FS is not that difficult if you use divide & conquer techniques.
 
Firstly learn to understand FS modularity and get a brief understanding about how the switch core works. That is always nice to know.
 
Once you have done that you only need to focus on the modules you actually want to work with.
 
And the advice below is important - it's a bit hard to read and understand code before you know how to use the system. I will advice you to move usage questions to the user list - simply because you get more actual users responding in there.
 
have fun
 
Jan
 

> From: fdelawarde at wirelessmundi.com
> To: freeswitch-dev at lists.freeswitch.org
> Date: Thu, 16 Feb 2012 11:00:37 +0100
> Subject: Re: [Freeswitch-dev] How to understand codebase
> 
> I would start by using FS extensively and looking at console logs or
> playing with ESL to understand the logic (channel states, events, etc.).
> 
> Then try to understand well the code of small modules like
> mod_blacklist, and slowly going to the core stuff from there.
> 
> FS code is in general really well put together and, compared to
> Asterisk, I find it very easy to read and understand without
> documentation. Well, maybe a few drawings could help to learn faster...
> 
> Have fun,
> Fran?ois.
> 
> On Thu, 2012-02-16 at 15:41 +0600, Arif Hossain wrote: 
> > I'm quite new to freeswitch. I have some experience in asterisk. But
> > Mostly about how to configure asterisk. The codebase of freeswitch is
> > quite large and seems overhelming at first sight. I'm a moderately good
> > c programmer but i don't have a lot of experience understanding a huge
> > codebase only by reading code. 
> > 
> > I saw freeswitch does not come with a "developer manual" or something
> > like that. Which would be very helpful. Because i do have a moderate
> > knowledge about sip and telephony technologies. 
> > 
> > So any other ways to understand freeswitch codebase(the core, and
> > important modules like media session handling etc) efficiently? I'm
> > willing to "learn" and contribute to the project. 
> > 
> > I know "knowledge" is itself a "resource". But any help regarding this
> > would be wonderful.
> > 
> > At this time i have checked out freeswitch code from git, loaded it in
> > eclipse, and trying to understand some core data structures. But I've
> > realized it would be very difficult to understand everything by myself.
> > 
> > Thanks in advance.
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 freefall1986 at gmail.com  Thu Feb 16 13:16:36 2012
From: freefall1986 at gmail.com (Arif Hossain)
Date: Thu, 16 Feb 2012 16:16:36 +0600
Subject: [Freeswitch-dev] How to understand codebase
In-Reply-To: <1329386437.6226.233.camel@luna.madrid.commsmundi.com>
References: <1329385289.4913.9.camel@arifLaptop>
	<1329386437.6226.233.camel@luna.madrid.commsmundi.com>
Message-ID: <1329387396.4913.13.camel@arifLaptop>

On Thu, 2012-02-16 at 11:00 +0100, Fran?ois Delawarde wrote:
> I would start by using FS extensively and looking at console logs or
> playing with ESL to understand the logic (channel states, events, etc.).
> 
> Then try to understand well the code of small modules like
> mod_blacklist, and slowly going to the core stuff from there.
> 
> FS code is in general really well put together and, compared to
> Asterisk, I find it very easy to read and understand without
> documentation. Well, maybe a few drawings could help to learn faster...
> 
> Have fun,
> Fran?ois.


Thank you for this suggestion. I would certainly do that. Do i need to
write my own esl connector or freeswitch distributes  some samples?(I
see some perl/java/ruby samples in the wiki)?



From chat2jesse at gmail.com  Thu Feb 16 19:26:24 2012
From: chat2jesse at gmail.com (jesse)
Date: Thu, 16 Feb 2012 08:26:24 -0800
Subject: [Freeswitch-dev] How to understand codebase
In-Reply-To: <1329387396.4913.13.camel@arifLaptop>
References: <1329385289.4913.9.camel@arifLaptop>
	<1329386437.6226.233.camel@luna.madrid.commsmundi.com>
	<1329387396.4913.13.camel@arifLaptop>
Message-ID: <CAE6kErgPzdH6gqFrLGVryg_i=zacxd9W3o4Sry-yGEXhWsj4Pw@mail.gmail.com>

One tip: build freeswitch in vs2008 express. You can easily debug line by
line.
On Feb 16, 2012 2:17 AM, "Arif Hossain" <freefall1986 at gmail.com> wrote:

> On Thu, 2012-02-16 at 11:00 +0100, Fran?ois Delawarde wrote:
> > I would start by using FS extensively and looking at console logs or
> > playing with ESL to understand the logic (channel states, events, etc.).
> >
> > Then try to understand well the code of small modules like
> > mod_blacklist, and slowly going to the core stuff from there.
> >
> > FS code is in general really well put together and, compared to
> > Asterisk, I find it very easy to read and understand without
> > documentation. Well, maybe a few drawings could help to learn faster...
> >
> > Have fun,
> > Fran?ois.
>
>
> Thank you for this suggestion. I would certainly do that. Do i need to
> write my own esl connector or freeswitch distributes  some samples?(I
> see some perl/java/ruby samples in the wiki)?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 james.gledhill at sipstorm.com  Mon Feb 20 20:05:01 2012
From: james.gledhill at sipstorm.com (Gledhill, James)
Date: Mon, 20 Feb 2012 12:05:01 -0500
Subject: [Freeswitch-dev] conference play / stop / PAUSE/RESUME
Message-ID: <A671FCA12875564E8CC11878981732BB14E1F7C1@mse2be2.mse2.exchange.ms>

I have been trying to do the same thing as was asked in an earlier
question, but it was asked 10 months ago and I did not want to use the
same thread, as it went a different direction than my question. I
understood the concept of the answer, but I am not quite sure how to
implement it.  

 

I would like to make a call to another person (via FS), be able to
record parts of the call, and then play it back to both participants -
being able to use the features of uuid_fileman (pause, seek +/-, etc),
then record another part of the conversation, playback, etc.  

 

>From other searches, it was made clear that this had to be done with a
conference call.

 

===== From the previous thread Apr 18, 2011; 11:30am Re: conference play
/ stop / PAUSE/RESUME ====

 

Anthony Minessale II - wrote 

 

if you need this level of granularity, I suggest instead of the 

conference play stuff that is built in, that you use the originate 

command to call a new channel into the conference to play the file 

then use the uuid_fileman api command to manipulate the playback. 

 

 

On Sat, Apr 16, 2011 at 1:33 PM, Dale Trub <[hidden email]> wrote:

 

> We let the conference host play a file to the conference and want to
give 

> them a control to "pause" such a file, and then resume it. 

> Has anyone found a way to do this? 

> Would someone create something like that for a modest bounty?  We
could 

> incorp in the code we use, as well as submit to the project overall. 

> -Dale

 

======= End Previous thread quoted =========

 

Does anyone have an example of this?  I would like to implement it is
either dailplan or LUA, but Javascript or anything else would be good
examples.  

 

I have tried a number of different ways and have not gotten it to work.

 

*       I may not be doing "use the originate command to call a new
channel into the conference to play the file" properly.  Suggestion on
this?  

 

One thing I tried was in LUA.  Session1 is the original call-ER,
Session2 is the call-ED party - both sessions are (1&2) are
uuid_transer-ed to a conference bridge.

*           api:execute("uuid_transfer", session1:get_uuid() .. "
conf_11_111 XML callRecPlayback" );

*           api:execute("uuid_transfer", session2:get_uuid() .. "
conf_11_111 XML callRecPlayback" );

*        

*           session3 =
freeswitch.Session("loopback/conf_11_111/callRecPlayback/XML");

*           session3:setAutoHangup(false);

*           freeswitch.consoleLog("info","\n\n    SEssion 3 == "..
session3:get_uuid() .."  \n\n");

*        

*           session:execute("bind_digit_action",
"cool,444,api:uuid_broadcast," .. session3:get_uuid() .. "
ivr/ivr-welcome_to_freeswitch.wav both");

 

 

The bind_digit_action works and calls the api, but fs_cli reports
CHANNEL_KILL on the loopback channel, and I do not hear anything

 

2012-02-20 10:19:09.381934 [DEBUG] switch_rtp.c:3405 RTP RECV DTMF 4:640

2012-02-20 10:19:09.681942 [DEBUG] switch_rtp.c:3405 RTP RECV DTMF 4:640

2012-02-20 10:19:09.981952 [DEBUG] switch_rtp.c:3405 RTP RECV DTMF
4:1120

2012-02-20 10:19:09.981952 [DEBUG] mod_dptools.c:185
sofia/callRecPlayback/1004 at 172.30.110.112:5083 Digit match binding
[api:uuid_broadcast][8d48d97a-5c74-4104-bc66-e424a9a1d155
ivr/ivr-welcome_to_freeswitch.wav both]

2012-02-20 10:19:09.981952 [DEBUG] switch_core_session.c:1012 Send
signal loopback/conf_11_111-a [BREAK]

2012-02-20 10:19:09.981952 [DEBUG] mod_loopback.c:475
loopback/conf_11_111-a CHANNEL KILL

2012-02-20 10:19:09.981952 [DEBUG] mod_dptools.c:244
sofia/callRecPlayback/1004 at 172.30.110.112:5083 Digit match binding
[api:uuid_broadcast][8d48d97a-5c74-4104-bc66-e424a9a1d155
ivr/ivr-welcome_to_freeswitch.wav both] api executed, +OK Message Sent

 

 

 

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From james.gledhill at sipstorm.com  Tue Feb 21 01:37:16 2012
From: james.gledhill at sipstorm.com (Gledhill, James)
Date: Mon, 20 Feb 2012 17:37:16 -0500
Subject: [Freeswitch-dev] conference play / stop / PAUSE/RESUME
In-Reply-To: <A671FCA12875564E8CC11878981732BB14E1F7C1@mse2be2.mse2.exchange.ms>
Message-ID: <A671FCA12875564E8CC11878981732BB14E1F928@mse2be2.mse2.exchange.ms>

After sending this email, I finally figured it out.  

 

I will send a new post after I get things cleaned up and share the code
so others can use it later.

 

 

 

________________________________

From: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of
Gledhill, James
Sent: Monday, February 20, 2012 11:05 AM
To: freeswitch-dev at lists.freeswitch.org
Subject: [Freeswitch-dev] conference play / stop / PAUSE/RESUME

 

I have been trying to do the same thing as was asked in an earlier
question, but it was asked 10 months ago and I did not want to use the
same thread, as it went a different direction than my question. I
understood the concept of the answer, but I am not quite sure how to
implement it.  

 

I would like to make a call to another person (via FS), be able to
record parts of the call, and then play it back to both participants -
being able to use the features of uuid_fileman (pause, seek +/-, etc),
then record another part of the conversation, playback, etc.  

 

>From other searches, it was made clear that this had to be done with a
conference call.

 

===== From the previous thread Apr 18, 2011; 11:30am Re: conference play
/ stop / PAUSE/RESUME ====

 

Anthony Minessale II - wrote 

 

if you need this level of granularity, I suggest instead of the 

conference play stuff that is built in, that you use the originate 

command to call a new channel into the conference to play the file 

then use the uuid_fileman api command to manipulate the playback. 

 

 

On Sat, Apr 16, 2011 at 1:33 PM, Dale Trub <[hidden email]> wrote:

 

> We let the conference host play a file to the conference and want to
give 

> them a control to "pause" such a file, and then resume it. 

> Has anyone found a way to do this? 

> Would someone create something like that for a modest bounty?  We
could 

> incorp in the code we use, as well as submit to the project overall. 

> -Dale

 

======= End Previous thread quoted =========

 

Does anyone have an example of this?  I would like to implement it is
either dailplan or LUA, but Javascript or anything else would be good
examples.  

 

I have tried a number of different ways and have not gotten it to work.

 

*       I may not be doing "use the originate command to call a new
channel into the conference to play the file" properly.  Suggestion on
this?  

 

One thing I tried was in LUA.  Session1 is the original call-ER,
Session2 is the call-ED party - both sessions are (1&2) are
uuid_transer-ed to a conference bridge.

*           api:execute("uuid_transfer", session1:get_uuid() .. "
conf_11_111 XML callRecPlayback" );

*           api:execute("uuid_transfer", session2:get_uuid() .. "
conf_11_111 XML callRecPlayback" );

*        

*           session3 =
freeswitch.Session("loopback/conf_11_111/callRecPlayback/XML");

*           session3:setAutoHangup(false);

*           freeswitch.consoleLog("info","\n\n    SEssion 3 == "..
session3:get_uuid() .."  \n\n");

*        

*           session:execute("bind_digit_action",
"cool,444,api:uuid_broadcast," .. session3:get_uuid() .. "
ivr/ivr-welcome_to_freeswitch.wav both");

 

 

The bind_digit_action works and calls the api, but fs_cli reports
CHANNEL_KILL on the loopback channel, and I do not hear anything

 

2012-02-20 10:19:09.381934 [DEBUG] switch_rtp.c:3405 RTP RECV DTMF 4:640

2012-02-20 10:19:09.681942 [DEBUG] switch_rtp.c:3405 RTP RECV DTMF 4:640

2012-02-20 10:19:09.981952 [DEBUG] switch_rtp.c:3405 RTP RECV DTMF
4:1120

2012-02-20 10:19:09.981952 [DEBUG] mod_dptools.c:185
sofia/callRecPlayback/1004 at 172.30.110.112:5083 Digit match binding
[api:uuid_broadcast][8d48d97a-5c74-4104-bc66-e424a9a1d155
ivr/ivr-welcome_to_freeswitch.wav both]

2012-02-20 10:19:09.981952 [DEBUG] switch_core_session.c:1012 Send
signal loopback/conf_11_111-a [BREAK]

2012-02-20 10:19:09.981952 [DEBUG] mod_loopback.c:475
loopback/conf_11_111-a CHANNEL KILL

2012-02-20 10:19:09.981952 [DEBUG] mod_dptools.c:244
sofia/callRecPlayback/1004 at 172.30.110.112:5083 Digit match binding
[api:uuid_broadcast][8d48d97a-5c74-4104-bc66-e424a9a1d155
ivr/ivr-welcome_to_freeswitch.wav both] api executed, +OK Message Sent

 

 

 

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From eli at netspectrum.com  Tue Feb 21 13:25:13 2012
From: eli at netspectrum.com (Erjian Li)
Date: Tue, 21 Feb 2012 18:25:13 +0800
Subject: [Freeswitch-dev] Look for some advice on how to implement a
	FreeSWITCH controlling module
Message-ID: <CADBCHNSwZssQ-nHS+gHs=wtpV-Cu=xe=hkZSe2eBHZ8vdDfpcA@mail.gmail.com>

Hi everyone,

I am developing software for a conference call product with FreeSWITCH as
IP PBX software, and I have some troubles on a module implementation.  So I
come here for some advice. Thanks in advance.

My requirement of the module:
I want to implement a controlling module (or separate program) for
FreeSWITCH, this module (or separate program) receives the admin messages
from a so-called admin server , parses the messages, and generates
FreeSWITCH commands according to the message data, then send these commands
to FreeSWITCH to execute (e.g. send a "conference dial ..." command to add
a member into the conference call). Also, this module (or separate program)
is required to listen the CHANNEL related events fired by FreeSWITCH, and
report the FreeSWITCH channel status to admin server.

My questions:
 I want to implement above functionality in a separate program. When
running this program communicates with FreeSWITCH via FreeSWITCH's event
socket library, is it applicable?  I don't have much knowledge about
FreeSWITCH's loadable module, is it better to implement it as a
FreeSWITCH's loadable module, like the modules in src/mod/applications?

-- 
Best Regards

Erjian
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From jmesquita at freeswitch.org  Tue Feb 21 17:19:37 2012
From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=)
Date: Tue, 21 Feb 2012 11:19:37 -0300
Subject: [Freeswitch-dev] Look for some advice on how to implement a
 FreeSWITCH controlling module
In-Reply-To: <CADBCHNSwZssQ-nHS+gHs=wtpV-Cu=xe=hkZSe2eBHZ8vdDfpcA@mail.gmail.com>
References: <CADBCHNSwZssQ-nHS+gHs=wtpV-Cu=xe=hkZSe2eBHZ8vdDfpcA@mail.gmail.com>
Message-ID: <40248E8B913848F19CCCA20C171B487E@freeswitch.org>

ESL is your best choice unless you are trying to really mess with the audio streams. From what you describe, that's not what you are trying to do.

I normally use inbound ESL for that.

Regards,  

--  
Jo?o Mesquita
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Tuesday, February 21, 2012 at 7:25 AM, Erjian Li wrote:

> Hi everyone,
>  
> I am developing software for a conference call product with FreeSWITCH as IP PBX software, and I have some troubles on a module implementation.  So I come here for some advice. Thanks in advance.
>  
> My requirement of the module:
> I want to implement a controlling module (or separate program) for FreeSWITCH, this module (or separate program) receives the admin messages from a so-called admin server , parses the messages, and generates FreeSWITCH commands according to the message data, then send these commands to FreeSWITCH to execute (e.g. send a "conference dial ..." command to add a member into the conference call). Also, this module (or separate program) is required to listen the CHANNEL related events fired by FreeSWITCH, and report the FreeSWITCH channel status to admin server.
>  
> My questions:
>  I want to implement above functionality in a separate program. When running this program communicates with FreeSWITCH via FreeSWITCH's event socket library, is it applicable?  I don't have much knowledge about FreeSWITCH's loadable module, is it better to implement it as a FreeSWITCH's loadable module, like the modules in src/mod/applications?      
>  
> --  
> Best Regards
>  
> Erjian
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
>  
> 
> 
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>  
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org (mailto: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 peter.olsson at visionutveckling.se  Tue Feb 21 17:19:46 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 21 Feb 2012 14:19:46 +0000
Subject: [Freeswitch-dev] Look for some advice on how to implement
	a	FreeSWITCH controlling module
In-Reply-To: <CADBCHNSwZssQ-nHS+gHs=wtpV-Cu=xe=hkZSe2eBHZ8vdDfpcA@mail.gmail.com>
References: <CADBCHNSwZssQ-nHS+gHs=wtpV-Cu=xe=hkZSe2eBHZ8vdDfpcA@mail.gmail.com>
Message-ID: <1FFF97C269757C458224B7C895F35F15042D0F@cantor.std.visionutv.se>

In my experience, ESL (mod_event_socket) is the perfect solution for this kind of implementation.

I've done some simple conference handling, and I'm using ESL for it all.

/Peter

Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r Erjian Li
Skickat: den 21 februari 2012 11:25
Till: freeswitch-dev at lists.freeswitch.org
?mne: [Freeswitch-dev] Look for some advice on how to implement a FreeSWITCH controlling module

Hi everyone,

I am developing software for a conference call product with FreeSWITCH as IP PBX software, and I have some troubles on a module implementation.  So I come here for some advice. Thanks in advance.

My requirement of the module:
I want to implement a controlling module (or separate program) for FreeSWITCH, this module (or separate program) receives the admin messages from a so-called admin server , parses the messages, and generates FreeSWITCH commands according to the message data, then send these commands to FreeSWITCH to execute (e.g. send a "conference dial ..." command to add a member into the conference call). Also, this module (or separate program) is required to listen the CHANNEL related events fired by FreeSWITCH, and report the FreeSWITCH channel status to admin server.

My questions:
 I want to implement above functionality in a separate program. When running this program communicates with FreeSWITCH via FreeSWITCH's event socket library, is it applicable?  I don't have much knowledge about FreeSWITCH's loadable module, is it better to implement it as a FreeSWITCH's loadable module, like the modules in src/mod/applications?

--
Best Regards

Erjian

!DSPAM:4f43a50432761362825821!
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From msc at freeswitch.org  Tue Feb 21 22:52:05 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 21 Feb 2012 11:52:05 -0800
Subject: [Freeswitch-dev] Call For Community Assistance
Message-ID: <CAKzWOxWH8d18dOOqtF2t=Fj0PqpAgZMnWWvQOXLke2ve5NhVYA@mail.gmail.com>

Hello all!

As you may have noticed, I've been rather absent from the IRC channel and
mailing lists for the past week or so. Like others on the FreeSWITCH and
CudaTel team I have learned first hand just how valuable Brian is and how
much work he produces. We are all trying to pitch in and lighten the load.
To that end I wanted to say thank you to Ken Rice, Jeff Lenk, Avi Marcus,
and all the rest of the community members who have been stepping up to help
out on IRC and the mailing list.

We definitely can use more help in this regard. I'd like to ask everyone
subscribed to this mailing list to review the recent threads and look for
questions which you can answer quickly and then do so. If there are any
subjects that you cannot answer right away but could research, I'd ask that
you send a reply to the thread indicating your willingness to do some
research on the subject at hand. If everyone can do just a little bit then
we'll get a lot accomplished.

One other thing you can help with: testing the new sounds batch. I've
rolled sounds version 1.0.18 and put it up on files.freeswitch.org. I've
run it on several of my test machines and now I'd like to have a lot of
others do so as well. Just be sure to do this on a test machine before
trying it on a production machine. :D

Edit fs_src/build/sounds_version.txt and set the en-callie version to
"1.0.18"
Save the file, then issue "make cd-sounds-install"
It should download the latest sound tarballs

Once the downloads are done do some basic testing like calling in to a
conference, sending and retrieving voicemails, etc. Let me know if you run
into any issues.

Thanks again for all your help!
-Michael
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From msc at freeswitch.org  Wed Feb 22 02:43:50 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 21 Feb 2012 15:43:50 -0800
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Tomorrow (Wed Feb 22)
Message-ID: <CAKzWOxV+9fOcs=50SSoXXhjzQRJpJUz6D+Kyxct3jmFq0UWTKQ@mail.gmail.com>

Hey all!

Yes, I promise that I will come up for air and hang out with the community
tomorrow. :)

We have a special guest joining us tomorrow: Stefan Wintermeyer from
Amooma. He will be talking about Gemeinschaft v4 and all the interesting
things that lead to going with FreeSWITCH instead of Asterisk.

The agenda page is here:
http://wiki.freeswitch.org/wiki/FS_weekly_2012_02_22

Look forward to speaking with you all tomorrow!

-Michael
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From gabe at gundy.org  Wed Feb 22 10:26:37 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Wed, 22 Feb 2012 00:26:37 -0700
Subject: [Freeswitch-dev] Look for some advice on how to implement a
 FreeSWITCH controlling module
In-Reply-To: <CADBCHNSwZssQ-nHS+gHs=wtpV-Cu=xe=hkZSe2eBHZ8vdDfpcA@mail.gmail.com>
References: <CADBCHNSwZssQ-nHS+gHs=wtpV-Cu=xe=hkZSe2eBHZ8vdDfpcA@mail.gmail.com>
Message-ID: <CAMwZ-tKQjW0F45x0OJ3fPgoatzRTZoDffYrxymR7fdrTcVDEuA@mail.gmail.com>

On Tue, Feb 21, 2012 at 3:25 AM, Erjian Li <eli at netspectrum.com> wrote:
> I want to implement above functionality in a separate program. When running
> this program communicates with FreeSWITCH via FreeSWITCH's event socket
> library, is it applicable?? I don't have much knowledge about FreeSWITCH's
> loadable module, is it better to implement it as a FreeSWITCH's loadable
> module, like the modules in src/mod/applications?

Both writing a module and using the Event Socket will work. If you
don't know which one to use, chances are, you want to start with Event
Socket. Event Socket also has the added benefit of allowing you to use
whatever language you like.

BTW, there is already a mod_callcenter loadable module in the source
code. Perhaps you can start with improving it.

Good luck,
Gabe


From eli at netspectrum.com  Thu Feb 23 13:27:51 2012
From: eli at netspectrum.com (Erjian Li)
Date: Thu, 23 Feb 2012 18:27:51 +0800
Subject: [Freeswitch-dev] my problems on originating a new call and saying
	digits
Message-ID: <CADBCHNS8RPT1ND7GeQa+HJ+CzWnKGxNnHeeC8s-mvwWJF9r5aQ@mail.gmail.com>

Hi All,

I  have a problem about he API command <originate>, so I'm here for some
help. Thanks in advance.

 In FreeSWITCH, I want to originate a call to a remote user, when the user
answers the call, FreeSWITCH will say some digits to the user. My command
is as following:
=======================================
  freeswitch at internal>originate sofia/internal/8613302033450 at xx.xx.xx.xx:5060
'&say(en number iterated 12345678912345)'
========================================

After I execute this command, the destination cellphone rings, but when I
answer the call, the  digit string "12345678912345" has almost finished
saying, so I can't hear the complete digit string. In other words,  the
'say' application has been executed before the call is answered. This is
not what I expect. Could any one  give me some advice on what I should do ?
Or some other commands to meet my requirement? Any advice would be
appreciated.

BTW, if I originate a call to local user (e.g. 1005, which is the
preconfigured user account of FreeSWITCH), and say some digits, the
behavior is what I expect. For example, if I execute the below command:
=======================================
freeswitch at internal>originate sofia/internal/1005%10.20.1.158 '&say(en
number iterated 12345678912345)'
=======================================

 The digits "12345678912345" begin to be said right after I answer the call
(I use X-Lite as the soft phone). This is what I expect. But why the
behavior when with a remote user is different?


-- 
Best Regards

Erjian
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From prashant.lamba at gmail.com  Thu Feb 23 13:33:48 2012
From: prashant.lamba at gmail.com (Prashant Lamba)
Date: Thu, 23 Feb 2012 16:03:48 +0530
Subject: [Freeswitch-dev] my problems on originating a new call and
	saying digits
In-Reply-To: <CADBCHNS8RPT1ND7GeQa+HJ+CzWnKGxNnHeeC8s-mvwWJF9r5aQ@mail.gmail.com>
References: <CADBCHNS8RPT1ND7GeQa+HJ+CzWnKGxNnHeeC8s-mvwWJF9r5aQ@mail.gmail.com>
Message-ID: <CAPHrvXEx6b5vx_+xuQX372kYKf1D61ZO1vSC0_evxtksZaCpGA@mail.gmail.com>

You need to use 'ignore_early_media=true' in the originate command. Like
this

originate {ignore_early_media=true} sofia ....
 On Feb 23, 2012 3:59 PM, "Erjian Li" <eli at netspectrum.com> wrote:

> Hi All,
>
> I  have a problem about he API command <originate>, so I'm here for some
> help. Thanks in advance.
>
>  In FreeSWITCH, I want to originate a call to a remote user, when the user
> answers the call, FreeSWITCH will say some digits to the user. My command
> is as following:
> =======================================
>   freeswitch at internal>originate sofia/internal/8613302033450 at xx.xx.xx.xx:5060
> '&say(en number iterated 12345678912345)'
> ========================================
>
> After I execute this command, the destination cellphone rings, but when I
> answer the call, the  digit string "12345678912345" has almost finished
> saying, so I can't hear the complete digit string. In other words,  the
> 'say' application has been executed before the call is answered. This is
> not what I expect. Could any one  give me some advice on what I should do ?
> Or some other commands to meet my requirement? Any advice would be
> appreciated.
>
> BTW, if I originate a call to local user (e.g. 1005, which is the
> preconfigured user account of FreeSWITCH), and say some digits, the
> behavior is what I expect. For example, if I execute the below command:
> =======================================
> freeswitch at internal>originate sofia/internal/1005%10.20.1.158 '&say(en
> number iterated 12345678912345)'
> =======================================
>
>  The digits "12345678912345" begin to be said right after I answer the
> call (I use X-Lite as the soft phone). This is what I expect. But why the
> behavior when with a remote user is different?
>
>
> --
> Best Regards
>
> Erjian
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Fri Feb 24 00:00:53 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Feb 2012 13:00:53 -0800
Subject: [Freeswitch-dev] FreeSWITCH Cookbook Is Published!
Message-ID: <CAKzWOxW08PCGpwqsfq0OWxfasLJPKGYAUBKtC6VWPQQrqNOwWA@mail.gmail.com>

Hey all,

Just wanted to mention a little story I wrote up on the main FreeSWITCH
website:
http://www.freeswitch.org/node/381

Packt has officially published the the cookbook, so get your copy today!

-Michael
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From e.tylepov at btcom.kz  Fri Feb 24 08:54:23 2012
From: e.tylepov at btcom.kz (Erjan Tylepov)
Date: Fri, 24 Feb 2012 05:54:23 +0000
Subject: [Freeswitch-dev] Write access permission to repository
Message-ID: <20120224055423.91419940@mail.btcom.kz>

Hi all,


I've generated rsa keys in openssh, sent a public one to the git-access at freeswitch.org yesterday, hadn't recieved any acknowledgement from then yet. Today I tried to:
"git clone ssh://git at git.freeswitch.org:222/freeswitch.git"
and got:
"git at git.freeswitch.org's password:"
What else I need to do to get write access?


Regards,
Yerzhan.
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From krice at freeswitch.org  Fri Feb 24 22:36:38 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 24 Feb 2012 13:36:38 -0600
Subject: [Freeswitch-dev] Write access permission to repository
In-Reply-To: <20120224055423.91419940@mail.btcom.kz>
References: <20120224055423.91419940@mail.btcom.kz>
Message-ID: <CAPcNkdccLTdaUpc1Xwepkc=ezYHgNcLpZHAem5icMLOkHQnFVw@mail.gmail.com>

you should start by showing patches to the development team, commit access
is not something thats just handed out on a whim...

>From there, your request will be evaluated for access...

K


On Thu, Feb 23, 2012 at 11:54 PM, Erjan Tylepov <e.tylepov at btcom.kz> wrote:

> **
> Hi all,
>
> I've generated rsa keys in openssh, sent a public one to the
> git-access at freeswitch.org yesterday, hadn't recieved any acknowledgement
> from then yet. Today I tried to:
> "git clone ssh://git at git.freeswitch.org:222/freeswitch.git"
> and got:
> "git at git.freeswitch.org's password:"
> What else I need to do to get write access?
>
> Regards,
> Yerzhan.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 william.suffill at gmail.com  Sat Feb 25 19:11:10 2012
From: william.suffill at gmail.com (William Suffill)
Date: Sat, 25 Feb 2012 11:11:10 -0500
Subject: [Freeswitch-dev] Write access permission to repository
In-Reply-To: <CAPcNkdccLTdaUpc1Xwepkc=ezYHgNcLpZHAem5icMLOkHQnFVw@mail.gmail.com>
References: <20120224055423.91419940@mail.btcom.kz>
	<CAPcNkdccLTdaUpc1Xwepkc=ezYHgNcLpZHAem5icMLOkHQnFVw@mail.gmail.com>
Message-ID: <CAF-ZooTegNR+d99VvpznViRkSku35eurRvZAsaDRdNJwO3LDig@mail.gmail.com>

Besides with the nature of git it's just as easily to generate a diff
patch and put it in Jira for review and if approve someone will merge
it for you. This is to ensure the quality of what makes it to the
master tree.

-- W


From wlin121443 at yahoo.com  Sat Feb 25 20:55:50 2012
From: wlin121443 at yahoo.com (Warren Lin)
Date: Sat, 25 Feb 2012 09:55:50 -0800 (PST)
Subject: [Freeswitch-dev] Testing mod_sms
In-Reply-To: <1330189095.54572.YahooMailNeo@web124904.mail.ne1.yahoo.com>
References: <1329798347.23072.YahooMailNeo@web124911.mail.ne1.yahoo.com>
	<CAKbxfG8FHPRbUGQW0x8GvNdjpzGGn-e5GnCcEg1vLY5NM-HKFA@mail.gmail.com>
	<1330189095.54572.YahooMailNeo@web124904.mail.ne1.yahoo.com>
Message-ID: <1330192550.4176.YahooMailNeo@web124906.mail.ne1.yahoo.com>





________________________________
From: Warren Lin <wlin121443 at yahoo.com>
To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org> 
Cc: Anthony Minessale <anthony.minessale at gmail.com> 
Sent: Saturday, February 25, 2012 11:58 AM
Subject: Re: Testing mod_sms


Hi, Anthony,
?
????Sorry for the late reply, ?I can only do the test over weekend... 
?
???? to re-cap, I was testing "a message was sent while target cell phone was off" , hopefully to get a notice when such non-delivery happens. Or?someway , somehow 
???? to tell if a message was successfuly sent to target cell phone? or not (not to the freeswitch).
?
??? it looks like your patch did send out an event in this situation. but that causes another problem. freeswitch is trying to re-send the message again and
??? again without any waiting in between....that?took over?whole CPU and hung my machine....
?
???I would like to make following suggestion with my narrow view scope (I am using ESL) . see if it can be used to fit a broader scope.
?
? (1) mod_sms is?mostly used?by external programs such java client or php.... Is that possible let the application programer to decide to do the "re-send" or not.
????????in the case of?progammer decide to do the re-send, they should be able to define the re-send frequency and how many time it re-sends. At the end of 
??????? re-trying. there will be an event to tell the?re-try fail or not.
?
???????Or make life easier, just leave the logic to the application programmer. "do not re-send at all". but do issue an Event. (this event needs some id, pls see?#2
?????? as follows..
?
? (2)? I could not examine the detail of the event you send back for the non-delivery ( since my machine hung, need to reboot. lost all the message on the console).
??????? but anyhow, in the event, hopefully there is some correlation-id. to help programmer to see which?message was sent?sucessfully which one is not....
??????? in this case, there will be one id to return from two different places.? one id return right on the call
?
?????????????????????? returnEvt=con.sendEvent(evt);
?
??????? the other is when you generates non-delivery event. so that they can be related.
?
?????? Hmm....I briefly traced the code. this implemention "might" need big surgery on existing code...(just maybe since I?don't know too well about the code...)
???? 
?????? but least, on the non-delivery event, if you can keep ORIGINAL "to" "from" "event-timestamp" in the header. so a programmer can co-relate back the original msg.
?
?
this is just my 2 cents.
?
??????? BTW, I am also looking into? "clustering" freeswtich without using iptel.org.? could you give me?some heads-up. since I google this topic. but not many good
solutions...
?
Thanks
?
Warren
?
?
??


________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
To: Warren Lin <wlin121443 at yahoo.com> 
Sent: Tuesday, February 21, 2012 11:01 AM
Subject: Re: Testing mod_sms

hi you can find out how to use the lists at http://lists.freeswitch.org

try latest GIT, i put in a patch to fire the events with the heder
Failed-Delivery true
if you are listening for MESSAGE events you should catch them.


On Mon, Feb 20, 2012 at 10:25 PM, Warren Lin <wlin121443 at yahoo.com> wrote:
> Hi Anthony,
>
> ??? Found your e-mail address on the one of freeswitch discussion thread. do
> not know how to post a msg there.
> so, I send this? mail. hopefully you can get this.
>
> ?? I have used you mod_sms with ESL and Xlite, that works very well. but it
> looks like
> there is no way to get the message delivery confirmation back from
> freeswitch.
>
> ?on this senario, a message were sent out while target cell phone was off.
>
> ?I used ESL in java to test this senario. I do see something from the log...
>
> ....
> 2012-02-20 16:55:03.144870 [WARNING] sofia_presence.c:194 Not sending to
> local
> box for 2001 at xxx.xxx.xxx
> 2012-02-20 16:55:03.144870 [ERR] sofia_presence.c:243 Chat proto [global]
> ....
> ...
> Nobody to send to: Profile internal
> ...
> ? but ESL can not see any "Event" for this non-delivery... Is there any way
> to trap this event on ESL side?
>
> Warren
>



-- 
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
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From e.tylepov at btcom.kz  Mon Feb 27 07:56:04 2012
From: e.tylepov at btcom.kz (Erjan Tylepov)
Date: Mon, 27 Feb 2012 04:56:04 +0000
Subject: [Freeswitch-dev] Write access permission to repository
In-Reply-To: CAF-ZooTegNR+d99VvpznViRkSku35eurRvZAsaDRdNJwO3LDig@mail.gmail.com
Message-ID: <20120227045604.278d3e76@mail.btcom.kz>

Ok, I see. Thanks.   I've added FS-3950 improvement request.


I guess,   http://wiki.freeswitch.org/wiki/Git_Tips#Read.2FWrite should be rewritten.


Regards,
YT.

  _____  

From: William Suffill [mailto:william.suffill at gmail.com]
To: freeswitch-dev at lists.freeswitch.org
Sent: Sat, 25 Feb 2012 16:11:10 +0000
Subject: Re: [Freeswitch-dev] Write access permission to repository

Besides with the nature of git it's just as easily to generate a diff
  patch and put it in Jira for review and if approve someone will merge
  it for you. This is to ensure the quality of what makes it to the
  master tree.
  
  -- W
  
  _________________________________________________________________________
  Professional FreeSWITCH Consulting Services:
  consulting at freeswitch.org
  http://www.freeswitchsolutions.com
  
  
  
  
  Official FreeSWITCH Sites
  http://www.freeswitch.org
  http://wiki.freeswitch.org
  http://www.cluecon.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 william.suffill at gmail.com  Mon Feb 27 08:04:54 2012
From: william.suffill at gmail.com (William Suffill)
Date: Mon, 27 Feb 2012 00:04:54 -0500
Subject: [Freeswitch-dev] Write access permission to repository
In-Reply-To: <20120227045604.278d3e76@mail.btcom.kz>
References: <20120227045604.278d3e76@mail.btcom.kz>
Message-ID: <CAF-ZooQxzxFFX1ecyODntGVpHXscM+H=EYk6ndAXF3-e07pKqQ@mail.gmail.com>

The project has grown very quickly and it's a learning process to
adapt things to scale properly. I appreciate you taking the time to
opening a jira for your contribution. If you don't mind can you also
attach a diff of your changes against the current git revision. I see
you attached the whole modified file when only the diff would be
necessary.

http://wiki.freeswitch.org/wiki/Submitting_Patches covers how to
generate a patch. Just attach it to the JIRA with a .txt extension so
anyone reading the JIRA can pull up the diff and quickly see the
changes.

-- W


From e.tylepov at btcom.kz  Mon Feb 27 11:12:19 2012
From: e.tylepov at btcom.kz (Erjan Tylepov)
Date: Mon, 27 Feb 2012 08:12:19 +0000
Subject: [Freeswitch-dev] Write access permission to repository
In-Reply-To: CAF-ZooQxzxFFX1ecyODntGVpHXscM+H=EYk6ndAXF3-e07pKqQ@mail.gmail.com
Message-ID: <20120227081219.b2ce9ce8@mail.btcom.kz>

Actually the diff was in   http://pastebin.freeswitch.org/18527. But anyway attached the diff against the latest revision.


YT.
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From eli at netspectrum.com  Wed Feb 29 13:02:24 2012
From: eli at netspectrum.com (Erjian Li)
Date: Wed, 29 Feb 2012 18:02:24 +0800
Subject: [Freeswitch-dev] a problem about freeswitch conference caller
	control
Message-ID: <CADBCHNR_tEHzFwPVdKvB+zf4mzF_8esGV=RS+rF_Dxb+U0Gjqg@mail.gmail.com>

Hi All,

Firstly, I want to give my thanks to every one in the mailing list,
especially the ones who helped me on my questions. My two questions about
freeswitch has been solved due to the helps provided here.

Today I meet a new question about freeswitch:
My freeswitch server has been connected to a SIP provider's server (IDT
server), and can dial out to  normal cellphone numbers.  Now I want to
start a conference call using "conference <conf-name> dial ..." command,
when the destination phones join the conference, the phone callers can't
control the conference by pressing the keys specified in <caller-controls>
section of conference.conf.xml. (In other words, if I press key '0', it
can't mute myself; press '#', it can't hang up, etc. I use the default
caller-controls group.)  I want to know in this situation, whether the
cellphone's key tone can't be transferred to freeswitch server? Can
freeswitch server only receive DTMF encapsulated in RTP packet transferred
over IP?  I can control the conference when I use X-Lite as client, and
freeswitch outputs following log when I press key '0' in X-Lite interface:
==================================
freeswitch at eli-desktop> 2012-02-29 17:02:59.349272 [DEBUG]
switch_rtp.c:2428 RTP RECV DTMF 0:960
2012-02-29 17:02:59.367880 [DEBUG] mod_conference.c:2919 Queueing file
'/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav' for
play
=================================

In the case of cellphone, when users press key, the DTMF is generated and
transferred via PSTN channel to the IDT server, is this correct? and if so,
is it up to IDT server to encapsulate the DTMF in RTP packets and send it
to freeswitch server?

-- 
Best Regards

Erjian
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From msc at freeswitch.org  Wed Feb 29 20:47:30 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 29 Feb 2012 09:47:30 -0800
Subject: [Freeswitch-dev] Mailing List Reminders
Message-ID: <CAKzWOxW4ZSMwbKs_cpiozE5iUK==oe1MuXPMSYm9VwL=j3qZ0Q@mail.gmail.com>

Hello All!

It seems we've had a little drama on the mailing list (ML) lately so I
thought I would take this opportunity to offer a few reminders...


   - This is a public list, so anything you say will be forever etched into
   the annals of Internet lore. Please think before you click Send.
   - This is a public list, so be aware that others may say things that are
   offensive to you personally but not to the subscribers in general. Please
   be ready to "deal with it" on occasion.
   - This list is moderated to a degree. We keep only a loose reign on the
   users here. However, if after several warnings a user repeatedly engages in
   trollish behavior then he or she WILL get moderated. Those who are
   incorrigibly trollish will be banned.

I appreciate that most of our readers don't need these reminders, so thank
you for indulging us. Keep up the good work!

-Michael
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From msc at freeswitch.org  Wed Feb 29 20:48:16 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 29 Feb 2012 09:48:16 -0800
Subject: [Freeswitch-dev] FreeSWITCH Community Conference Call Today
Message-ID: <CAKzWOxX4N6ckA-DXehSZWkdzToON21vD2w3jr9-s-G_fKP3mQA@mail.gmail.com>

Hey all!

Don't forget our call in about 15 minutes. Agenda page is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2012_02_29

Talk to you soon!

-Michael
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From msc at freeswitch.org  Wed Feb 29 22:41:54 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 29 Feb 2012 11:41:54 -0800
Subject: [Freeswitch-dev] Subject: ClueCon 2012 - Call For Speaking Proposals
Message-ID: <CAKzWOxUy_3Fq4rw=ZFU6uU5dab5D3vvgn5c0Kz1Yez3jJaZpWA@mail.gmail.com>

Greetings!

The ClueCon team is gearing up for this summer's event, which will be held
August 7-9, 2012 at the Wyndham in downtown Chicago. We want to hear your
proposals for presentations to be given at ClueCon '12. Here's what we're
looking for:


   -     Discussions on technology or software that relate directly or
   indirectly to open source telephony
   -     Working title or very brief description of the talk
   -     A two or three sentence abstract giving more details
   -     Speaker's name, company, and biographical information
   -     All presentations should be 30 minutes in length, including
   question and answer time
   -     ClueCon sponsors who wish to have a presentation always get first
   priority when it comes to scheduling


Please send all proposals to marketing at cluecon.com. If you spoke at ClueCon
last year and your bio has not changed then please make note of this fact.
We look forward to hearing from you!

The ClueCon Team
http://www.cluecon.com
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