From anthony.minessale at gmail.com  Thu Nov  1 00:50:39 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 31 Oct 2012 16:50:39 -0500
Subject: [Freeswitch-dev] freeswitch as sbc , upper register how
In-Reply-To: <CAAB3jWvoyr-CBYO1dRTfAm5g-ygLpAYzdHOq73X-sFkET0vZ8Q@mail.gmail.com>
References: <e589620.3ad3.13ab7482ede.Coremail.openser@yeah.net>
	<CAAB3jWvoyr-CBYO1dRTfAm5g-ygLpAYzdHOq73X-sFkET0vZ8Q@mail.gmail.com>
Message-ID: <CAKbxfG_JhVBGNG+HFjWFUSeBjzFnsKvhbxsm6bjpktMSSs20aQ@mail.gmail.com>

see
http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#User-Specific_Gateways


On Wed, Oct 31, 2012 at 10:27 AM, DJB International <djbinter at gmail.com>wrote:

> If you are talking about mirror proxy function, then I don't think it can
> be done since FreeSWITCH is a B2BUA, not a proxy.
>
> -djbinter
>
>
> On Wed, Oct 31, 2012 at 7:44 AM, openser <openser at yeah.net> wrote:
>
>> hi guys,
>>
>> I know freeswitch can  configure to be a sbc, but for one feature, the
>> upper register , i do not know how to configure? this is a common feature
>> most sbc support, is freeswitch support this ? or if does not support, how
>> can we achive this ?  any  suggest ?
>>
>> thanks.
>>
>>
>>
>> _________________________________________________________________________
>> 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
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From mahesoftengg at gmail.com  Fri Nov  2 12:19:04 2012
From: mahesoftengg at gmail.com (Mahesh R)
Date: Fri, 2 Nov 2012 14:49:04 +0530
Subject: [Freeswitch-dev] DTMF Error
Message-ID: <CANZX-89mwjnsqtGW4TEv-B2mjtA35D9NnXbkEricbdKoUZc5+g@mail.gmail.com>

Hi,

I'm suffering from dtmf "double digit" problem for incoming calls that
destined to IVR on my FS. Can any one please help this issue..

-- 
--
Regards,
Mahesh R
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From dave at 3c.co.uk  Fri Nov  2 17:33:21 2012
From: dave at 3c.co.uk (David Knell)
Date: Fri, 2 Nov 2012 14:33:21 -0000
Subject: [Freeswitch-dev] DTMF Error
In-Reply-To: <CANZX-89mwjnsqtGW4TEv-B2mjtA35D9NnXbkEricbdKoUZc5+g@mail.gmail.com>
References: <CANZX-89mwjnsqtGW4TEv-B2mjtA35D9NnXbkEricbdKoUZc5+g@mail.gmail.com>
Message-ID: <00a701cdb907$0298f260$07cad720$@co.uk>

Best guess is that you've got more than one sort of DTMF detection enabled -
first suggestion would be to make sure that you've inband detection turned
off and that you're just using RFC2833.

 

--Dave

 

From: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Mahesh R
Sent: 02 November 2012 09:19
To: freeswitch-dev at lists.freeswitch.org
Subject: [Freeswitch-dev] DTMF Error

 

Hi,

I'm suffering from dtmf "double digit" problem for incoming calls that
destined to IVR on my FS. Can any one please help this issue..

-- 
--

Regards,

Mahesh R

 

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From NuwanW at unifybusiness.co.uk  Mon Nov  5 12:25:02 2012
From: NuwanW at unifybusiness.co.uk (Nuwan Wijerathne)
Date: Mon, 5 Nov 2012 09:25:02 +0000
Subject: [Freeswitch-dev] [Confidential] - uuid_broadcast
Message-ID: <78990CE7CC964442A7C2CA5F4689695E99BC8079@BARXB0003.UnifyBusiness.local>

Hello,

I'm trying to broadcast audio on a bridged call. My requirement is to play audio on both legs at the same time. I used uuid_broadcast in following order,

Uuid_broadcast uuid 'path' both

Please note that I'm sending uuid_broadcast through an esl connection. So my actual request to freeswitch is as follows,

eslWriteConnection.Send("bgapi uuid_broadcast uuid 'path to audio file' both");

(eslWriteConnection is an object of .Net ESLConnection)

The  issue I'm having is, freeswitch not playing the audio on both channels at the same time. FreeSwitch plays the audio on one leg first, then plays on the second leg (After it finished playing on first leg).

I don't have this issue with FreeSwitch 1.0.6, where it plays audio on both legs at the same time. I'm having this issue with FreeSwtich 1.2.3.

Could anyone please suggest any solution.

Thank you,


This e-mail and any attachments are for the intended addressee(s) only and may contain confidential and/or privileged material. If you are not a named addressee, do not use, retain or disclose such information. This email is not guaranteed to be free from viruses and does not bind Unify in any contract or obligation. Unify Business Solutions Ltd. Registered in England and Wales. No: 4749638 Registered Office: Ambassador House, 5 Midland Way, Barlborough, S43 4XA United Kingdom.
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From anthonynovatsis at gmail.com  Thu Nov  8 06:11:33 2012
From: anthonynovatsis at gmail.com (Anthony Novatsis)
Date: Thu, 8 Nov 2012 14:11:33 +1100
Subject: [Freeswitch-dev] Connecting Two Freeswitch Boxes and Presence
Message-ID: <CAHH0AN+-=BPvjE4njbaK5yXR80oOXtbYsVRG6nYuJk=b_6t9Sg@mail.gmail.com>

Dear All,

I have successfully connected two Freeswitch boxes together using IP
authentication (as described on the wiki page
http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes).

Calls are working as expected.  But I would also like
publish/subscribe to work across the two boxes so that clients
registered on BoxB can subscribe to the presence of devices that are
registered on BoxA.

Is this possible?  And if yes, could anybody tell me what changes I
need to make to the config to enable this behaviour?

Any help would be appreciated.

Thanks,
Anthony


From 2005xj at gmail.com  Thu Nov  8 11:59:08 2012
From: 2005xj at gmail.com (jun xie)
Date: Thu, 8 Nov 2012 16:59:08 +0800
Subject: [Freeswitch-dev] About Rxfax function of Freeswitch
Message-ID: <CAM1pmq4zWYBzFYRUMdA+cbRvcbbToPOLD50XaxO_7wcvTAQKug@mail.gmail.com>

HI, every body.
Recently,i tested a case to use FreeSwitch to receive T38 fax from a As5400
(Cisco voice GW).According with Google and Freeswitch.org useful
information,the procession is smooth.Rxfax could receive almost  all papers.
But the fax images received have some problems,that's the images were all
re sized,were shorter and broader.
I will show you one,it should be Fax paper type's problem.but i don't know
how to revise.Anybody can give me some suggestions? thanks a lot.
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From anthony.minessale at gmail.com  Thu Nov  8 23:57:33 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 8 Nov 2012 14:57:33 -0600
Subject: [Freeswitch-dev] Connecting Two Freeswitch Boxes and Presence
In-Reply-To: <CAHH0AN+-=BPvjE4njbaK5yXR80oOXtbYsVRG6nYuJk=b_6t9Sg@mail.gmail.com>
References: <CAHH0AN+-=BPvjE4njbaK5yXR80oOXtbYsVRG6nYuJk=b_6t9Sg@mail.gmail.com>
Message-ID: <CAKbxfG-acFqm0oXw5eaXDHAj0_mxSXdyrYuPWQcKfQ=e8XYYzA@mail.gmail.com>

It might work if the 2 boxes were sharing an external db and were
configured identically and the phones accepted notifies from other boxes
but I would not count on it.





On Wed, Nov 7, 2012 at 9:11 PM, Anthony Novatsis
<anthonynovatsis at gmail.com>wrote:

> Dear All,
>
> I have successfully connected two Freeswitch boxes together using IP
> authentication (as described on the wiki page
> http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes).
>
> Calls are working as expected.  But I would also like
> publish/subscribe to work across the two boxes so that clients
> registered on BoxB can subscribe to the presence of devices that are
> registered on BoxA.
>
> Is this possible?  And if yes, could anybody tell me what changes I
> need to make to the config to enable this behaviour?
>
> Any help would be appreciated.
>
> Thanks,
> Anthony
>
> _________________________________________________________________________
> 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
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From steveu at coppice.org  Fri Nov  9 14:54:28 2012
From: steveu at coppice.org (Steve Underwood)
Date: Fri, 09 Nov 2012 19:54:28 +0800
Subject: [Freeswitch-dev] About Rxfax function of Freeswitch
In-Reply-To: <CAM1pmq4zWYBzFYRUMdA+cbRvcbbToPOLD50XaxO_7wcvTAQKug@mail.gmail.com>
References: <CAM1pmq4zWYBzFYRUMdA+cbRvcbbToPOLD50XaxO_7wcvTAQKug@mail.gmail.com>
Message-ID: <509CEEF4.7010003@coppice.org>

On 11/08/2012 04:59 PM, jun xie wrote:
> HI, every body.
> Recently,i tested a case to use FreeSwitch to receive T38 fax from a 
> As5400 (Cisco voice GW).According with Google and Freeswitch.org 
> useful information,the procession is smooth.Rxfax could receive almost 
>  all papers.
> But the fax images received have some problems,that's the images were 
> all re sized,were shorter and broader.
> I will show you one,it should be Fax paper type's problem.but i don't 
> know how to revise.Anybody can give me some suggestions? thanks a lot.
>
Try using a viewer which isn't broken.

Steve


From kheimerl at cs.berkeley.edu  Mon Nov 12 04:20:21 2012
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Sun, 11 Nov 2012 17:20:21 -0800
Subject: [Freeswitch-dev] Patch adding system call support to mod_sms
Message-ID: <CACyT-3=-WZDTCKB8Uz+7SOXaEtwompTtR4hmhq-NZoAGFdXpVA@mail.gmail.com>

We needed to make system calls from a mod_sms chatplan, here's a patch
supporting that feature written by my colleague Shaddi Hassan.
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From dujinfang at gmail.com  Mon Nov 12 08:09:02 2012
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 12 Nov 2012 13:09:02 +0800
Subject: [Freeswitch-dev] Patch adding system call support to mod_sms
In-Reply-To: <CACyT-3=-WZDTCKB8Uz+7SOXaEtwompTtR4hmhq-NZoAGFdXpVA@mail.gmail.com>
References: <CACyT-3=-WZDTCKB8Uz+7SOXaEtwompTtR4hmhq-NZoAGFdXpVA@mail.gmail.com>
Message-ID: <02428AC8035A418EBC7A9F9DEF3F9195@gmail.com>

It would be good if you submit to jira.freeswitch.org


On Monday, November 12, 2012 at 9:20 AM, Kurtis Heimerl wrote:

> We needed to make system calls from a mod_sms chatplan, here's a patch supporting that feature written by my colleague Shaddi Hassan.  
> _________________________________________________________________________
> 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
> 
> 
> 
> 
> Attachments: 
> - mod_sms-system.patch
> 


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From kheimerl at cs.berkeley.edu  Mon Nov 12 08:19:00 2012
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Sun, 11 Nov 2012 21:19:00 -0800
Subject: [Freeswitch-dev] Patch adding system call support to mod_sms
In-Reply-To: <02428AC8035A418EBC7A9F9DEF3F9195@gmail.com>
References: <CACyT-3=-WZDTCKB8Uz+7SOXaEtwompTtR4hmhq-NZoAGFdXpVA@mail.gmail.com>
	<02428AC8035A418EBC7A9F9DEF3F9195@gmail.com>
Message-ID: <CACyT-3=WLiFMkBb15iX5g-TPgKhivYKjv42Os6RO+uNTBCpQGw@mail.gmail.com>

Thanks for the suggestion.

Done here: http://jira.freeswitch.org/browse/FS-4825

On Sunday, November 11, 2012, Seven Du wrote:

> It would be good if you submit to jira.freeswitch.org
>
> On Monday, November 12, 2012 at 9:20 AM, Kurtis Heimerl wrote:
>
> We needed to make system calls from a mod_sms chatplan, here's a patch
> supporting that feature written by my colleague Shaddi Hassan.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:_e({}, 'cvml',
> '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 <javascript:_e({}, 'cvml',
> '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
>
> Attachments:
>  - mod_sms-system.patch
>
>
>
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From msc at freeswitch.org  Wed Nov 14 19:33:32 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Nov 2012 08:33:32 -0800
Subject: [Freeswitch-dev] FreeSWITCH Conf Call Today
Message-ID: <CAKzWOxV3MyaH-FYE0UKPKQ-PraieBWDE4Kayt1RkR0rKy5DN=w@mail.gmail.com>

Hello All!

Today's conference call agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_14

Ken Rice and a few of our FS long time users will be presenting some tips
and tricks that you may not be aware of or may have forgotten about. Talk
to you soon!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From jerry.richards at teotech.com  Fri Nov 16 19:57:23 2012
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 16 Nov 2012 16:57:23 +0000
Subject: [Freeswitch-dev] Gateway Presence Subscription Storage
Message-ID: <1545146083A72C4DB7B66584B7E5D9842090DC67@CH1PRD0410MB381.namprd04.prod.outlook.com>

I added the XML to the gateway.xml file for one of my sip_profiles so that it subscribes to presence (and it works fine).  My question is, I cannot find where in the sqlite3 DB that Freeswitch stores the subscription data (the sip_subscriptions table is empty).  Is this subscription only saved/maintained in RAM on the heap?

        <subscriptions name="x">
            <subscription event="presence">
                <param name="retry-seconds" value="50"/>
                <param name="expire-seconds" value="60"/>
                <param name="content-type" value="application/pidf+xml"/>
            </subscription>
        </subscriptions>

Thanks,
Jerry
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From msc at freeswitch.org  Sat Nov 17 00:21:09 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 16 Nov 2012 13:21:09 -0800
Subject: [Freeswitch-dev] Friday Free-for-all!
Message-ID: <CAKzWOxVepyZ-puUBM7ikZBCEjvQFpTXMUEWaUoH7-URTrykuyA@mail.gmail.com>

Come join us! We're having a nice chat in the main conf room

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From anthony.minessale at gmail.com  Sat Nov 17 03:14:43 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 16 Nov 2012 18:14:43 -0600
Subject: [Freeswitch-dev] Gateway Presence Subscription Storage
In-Reply-To: <1545146083A72C4DB7B66584B7E5D9842090DC67@CH1PRD0410MB381.namprd04.prod.outlook.com>
References: <1545146083A72C4DB7B66584B7E5D9842090DC67@CH1PRD0410MB381.namprd04.prod.outlook.com>
Message-ID: <CAKbxfG9zibbq=bjpLeFNb-48X9yB-yeaSSCCkfYgoYwf0sKyug@mail.gmail.com>

sip subscriptions is only for inbound, the outbound is in memory only and
is not heavily used or tested.



On Fri, Nov 16, 2012 at 10:57 AM, Jerry Richards <jerry.richards at teotech.com
> wrote:

>  I added the XML to the gateway.xml file for one of my sip_profiles so
> that it subscribes to presence (and it works fine).  My question is, I
> cannot find where in the sqlite3 DB that Freeswitch stores the subscription
> data (the sip_subscriptions table is empty).  Is this subscription only
> saved/maintained in RAM on the heap?****
>
> ** **
>
>         <subscriptions name="x">****
>
>             <subscription event="presence">****
>
>                 <param name="retry-seconds" value="50"/>****
>
>                 <param name="expire-seconds" value="60"/>****
>
>                 <param name="content-type" value="application/pidf+xml"/>*
> ***
>
>             </subscription>****
>
>         </subscriptions>****
>
> ** **
>
> Thanks,****
>
> Jerry****
>
> _________________________________________________________________________
> 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
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From janvb at live.com  Mon Nov 19 20:39:08 2012
From: janvb at live.com (Jan Berger)
Date: Mon, 19 Nov 2012 18:39:08 +0100
Subject: [Freeswitch-dev] H.264 / RTSP
Message-ID: <SNT135-W14E7068F3B5748F5FFA2F9D2560@phx.gbl>





hi, Have anyone considered adding support for Video streams in FreeSWITCH? Particulary RTP+RTSP supporting Mpeg4/H.264. /J  		 	   		  
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From marketing at cluecon.com  Mon Nov 19 21:13:57 2012
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 19 Nov 2012 10:13:57 -0800
Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxVB-kjZ-s1ESuQfhkQ8QhCULymdZvD9DDNEeHjdh0PZOg@mail.gmail.com>

Happy short week to those of you in North America!

The weekly FreeSWITCH news and notes took a hiatus while I was out on a
medical leave. I am happy to report that I am back to work and recovering
nicely. Many thanks to those who sent their well-wishes and happy thoughts.
We have a great community and I am glad to be a part of it!

On last week's conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_14>
 we covered some Linux/FreeSWITCH install and configuration tips. A special
thanks to Ken Rice for giving us some practical information on many of the
useful files and utility items that are available in the FreeSWITCH source
tree and how to implement them, including FreeSWITCH init
scripts<http://fisheye.freeswitch.org/browse/freeswitch.git/build/freeswitch.init.redhat?hb=true>,
Anthony's .emacs
file<http://fisheye.freeswitch.org/browse/freeswitch.git/support-d/.emacs?hb=true>,
and even a monit <http://mmonit.com/monit/> configuration
example<http://fisheye.freeswitch.org/browse/freeswitch.git/build/freeswitch.monitrc?hb=true>.
I hope you found these items as useful as I did.

We recently released FreeSWITCH 1.2.4 and Ken Rice tells me that more
updates are in the works. More information will be available on this week's
conference call <http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_21>.
This week I will be presenting a Wiki how-to: adding a channel
variable page<http://wiki.freeswitch.org/wiki/Variable_skeleton>.
This will be especially useful because it illustrates a number of
Mediawiki<http://www.mediawiki.org/>
 concepts. Also, we have a lot of missing channel variables so if everyone
picks one or two to add we'll be able to expand the wiki coverage.

Have a great week!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From steveayre at gmail.com  Tue Nov 20 02:30:42 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 19 Nov 2012 23:30:42 +0000
Subject: [Freeswitch-dev] H.264 / RTSP
In-Reply-To: <SNT135-W14E7068F3B5748F5FFA2F9D2560@phx.gbl>
References: <SNT135-W14E7068F3B5748F5FFA2F9D2560@phx.gbl>
Message-ID: <CAFiqYukVREHHuMWHCpyoAXBN1stkz9CrFMfEexrSkTxPy+7uJQ@mail.gmail.com>

These are supported, but you will need to adjust your config to enable them.

Look at mod_h26X and mod_mp4v.

-Steve




On 19 November 2012 17:39, Jan Berger <janvb at live.com> wrote:

>  hi,
>
> Have anyone considered adding support for Video streams in FreeSWITCH?
>
> Particulary RTP+RTSP supporting Mpeg4/H.264.
>
> /J
>
>
> _________________________________________________________________________
> 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 dujinfang at gmail.com  Tue Nov 20 03:41:22 2012
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 20 Nov 2012 08:41:22 +0800
Subject: [Freeswitch-dev] H.264 / RTSP
In-Reply-To: <SNT135-W14E7068F3B5748F5FFA2F9D2560@phx.gbl>
References: <SNT135-W14E7068F3B5748F5FFA2F9D2560@phx.gbl>
Message-ID: <1FF69EBC2C3D44DF806A50B8727D3DF8@gmail.com>

We tried this on : https://github.com/seven1240/FreeSWITCH/tree/rtsp

Also we have a patch on mod_vlc that can play any video (including rtsp) ... 

-- 
Seven Du
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Tuesday, November 20, 2012 at 1:39 AM, Jan Berger wrote:

> hi,
>  
> Have anyone considered adding support for Video streams in FreeSWITCH?
>  
> Particulary RTP+RTSP supporting Mpeg4/H.264.
>  
> /J
>  
> _________________________________________________________________________
> 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 debasish.chandra at telemune.net  Wed Nov 21 11:58:08 2012
From: debasish.chandra at telemune.net (Debasish Chandra)
Date: Wed, 21 Nov 2012 14:28:08 +0530
Subject: [Freeswitch-dev] Failed DTMF sanity check.
Message-ID: <CAE8KqXy6cSj87vCr2x-46CL5KsxH=+JQPEuwkvqShuk80ZUL-w@mail.gmail.com>

Hi,

I am getting "Failed DTMF sanity check" randomly (not on every call, 2 - 3
calls out of 10). Application is running using Plivo REST API, while FS
receiving call from Sangoma NSG TDM gateway. I have read on FS wiki
that "Failed DTMF sanity check" means FS never received RFC2833 DTMF 'end'
packet. However call tcpdump shows that DTMF digits has been sent to FS
from Sangoma NSG, FS showing error "Failed DTMF sanity check" and ignores
the digit.

Please let me know if I need to do any extra settings like editing dtmf
duration or type.

Best Regards,
Debasish Chandra
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From msc at freeswitch.org  Wed Nov 21 19:23:03 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Nov 2012 08:23:03 -0800
Subject: [Freeswitch-dev] Failed DTMF sanity check.
In-Reply-To: <CAE8KqXy6cSj87vCr2x-46CL5KsxH=+JQPEuwkvqShuk80ZUL-w@mail.gmail.com>
References: <CAE8KqXy6cSj87vCr2x-46CL5KsxH=+JQPEuwkvqShuk80ZUL-w@mail.gmail.com>
Message-ID: <CAKzWOxWZGKkGMO+Rx8OWEcy54FBni7Va+S7B7+WeTYDU7dZg5A@mail.gmail.com>

Is there any noticeable delay in the amount of time between the middle
packet(s) and the end packet(s) for the DTMF that failed? How does that
compare to a working DTMF capture?

-MC

On Wed, Nov 21, 2012 at 12:58 AM, Debasish Chandra <
debasish.chandra at telemune.net> wrote:

> Hi,
>
> I am getting "Failed DTMF sanity check" randomly (not on every call, 2 - 3
> calls out of 10). Application is running using Plivo REST API, while FS
> receiving call from Sangoma NSG TDM gateway. I have read on FS wiki
> that "Failed DTMF sanity check" means FS never received RFC2833 DTMF 'end'
> packet. However call tcpdump shows that DTMF digits has been sent to FS
> from Sangoma NSG, FS showing error "Failed DTMF sanity check" and ignores
> the digit.
>
> Please let me know if I need to do any extra settings like editing dtmf
> duration or type.
>
> Best Regards,
> Debasish Chandra
>
>
>
> _________________________________________________________________________
> 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Wed Nov 21 19:42:53 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Nov 2012 08:42:53 -0800
Subject: [Freeswitch-dev] FreeSWITCH Community Conference Call Today
Message-ID: <CAKzWOxU=eiWfZE=WqvG3d09stvwq98R65_DAJ64ts7C=7OAkXA@mail.gmail.com>

Hello all,

Today's agenda page is here:
http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_21

We are going to do a wiki tutorial, specifically how to add a channel
variable to the wiki. We'll also go over script I wrote that produces an
HTML page that lists all the channel variables.

Talk to you soon!


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From brian at freeswitch.org  Wed Nov 21 20:00:00 2012
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Nov 2012 11:00:00 -0600
Subject: [Freeswitch-dev] Failed DTMF sanity check.
In-Reply-To: <CAE8KqXy6cSj87vCr2x-46CL5KsxH=+JQPEuwkvqShuk80ZUL-w@mail.gmail.com>
References: <CAE8KqXy6cSj87vCr2x-46CL5KsxH=+JQPEuwkvqShuk80ZUL-w@mail.gmail.com>
Message-ID: <F186EF15-43FB-4F66-AEA9-A29C9CE14B96@freeswitch.org>

This means what ever device is sending the dtmf is FAILING at life.  AKA Not sending the proper end bits so we give up and accept the digit anyway.  But you should complain to your device manufacture and have them read and understand RFC2833 properly.  Seems to be a great source of pain in SIP.
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 21, 2012, at 2:58 AM, Debasish Chandra <debasish.chandra at telemune.net> wrote:

> Hi,
> 
> I am getting "Failed DTMF sanity check" randomly (not on every call, 2 - 3 calls out of 10). Application is running using Plivo REST API, while FS receiving call from Sangoma NSG TDM gateway. I have read on FS wiki that "Failed DTMF sanity check" means FS never received RFC2833 DTMF 'end' packet. However call tcpdump shows that DTMF digits has been sent to FS from Sangoma NSG, FS showing error "Failed DTMF sanity check" and ignores the digit.
> 
> Please let me know if I need to do any extra settings like editing dtmf duration or type. 
> 
> Best Regards,
> Debasish Chandra



From jon2718 at gmail.com  Wed Nov 21 21:25:11 2012
From: jon2718 at gmail.com (Jon Lederman)
Date: Wed, 21 Nov 2012 13:25:11 -0500
Subject: [Freeswitch-dev] Freeswitch Hangup
Message-ID: <394E6730-1B2B-4408-BB56-BC941E133212@gmail.com>

Hi,

I am able to make calls and connect.  However after about 30 seconds the call is always disconnected.  I am using a public IP address on my server.  Everything was working perfectly before.  The only thing new is that I upgraded to the new version of linphone.  Here is my log file.  Any thoughts?  

Thanks

2012-11-21 00:40:31.418129 [DEBUG] switch_core_session.c:905 Send signalsofia/internal/1000 at 54.245.109.196[BREAK]
2012-11-21 00:40:31.418129 [DEBUG] sofia.c:6285 Channelsofia/internal/1000 at 54.245.109.196 entering state [completed][200]
2012-11-21 00:40:31.418129 [DEBUG] switch_core_session.c:759 Send signalsofia/internal/1000 at 54.245.109.196[BREAK]
2012-11-21 00:40:31.418129 [DEBUG] switch_channel.c:3351 (sofia/internal/1000 at 54.245.109.196) Callstate Change EARLY -> ACTIVE
2012-11-21 00:40:31.418129 [NOTICE] switch_ivr_originate.c:3357 Channel [sofia/internal/1000 at 54.245.109.196] has been answered
2012-11-21 00:40:31.418129 [DEBUG] switch_ivr_originate.c:3414 Originate Resulted in Success: [sofia/internal/sip:1001 at 75.83.124.194:35948]
2012-11-21 00:40:31.418129 [DEBUG] switch_ivr_originate.c:3414 Originate Resulted in Success: [sofia/internal/sip:1001 at 75.83.124.194:35948]
2012-11-21 00:40:31.418129 [DEBUG] switch_core_session.c:759 Send signal sofia/internal/sip:1001 at 75.83.124.194:35948 [BREAK]
2012-11-21 00:40:31.418129 [DEBUG] switch_core_session.c:759 Send signalsofia/internal/1000 at 54.245.109.196[BREAK]
2012-11-21 00:40:31.418129 [DEBUG] switch_ivr_bridge.c:1361 (sofia/internal/sip:1001 at 75.83.124.194:35948) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2012-11-21 00:40:31.418129 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1001 at 75.83.124.194:35948) Running State Change CS_EXCHANGE_MEDIA
2012-11-21 00:40:31.418129 [DEBUG] switch_core_session.c:1210 Send signal sofia/internal/sip:1001 at 75.83.124.194:35948 [BREAK]
2012-11-21 00:40:31.418129 [DEBUG] switch_core_state_machine.c:456 (sofia/internal/sip:1001 at 75.83.124.194:35948) State EXCHANGE_MEDIA
2012-11-21 00:40:31.418129 [DEBUG] mod_sofia.c:661 SOFIA EXCHANGE_MEDIA
2012-11-21 00:40:31.518129 [DEBUG] switch_core_session.c:905 Send signalsofia/internal/1000 at 54.245.109.196[BREAK]
2012-11-21 00:40:31.538130 [DEBUG] switch_core_session.c:821 Send signal sofia/internal/sip:1001 at 75.83.124.194:35948 [BREAK]
2012-11-21 00:40:31.538130 [DEBUG] switch_core_session.c:821 Send signalsofia/internal/1000 at 54.245.109.196[BREAK]
2012-11-21 00:40:31.538130 [DEBUG] switch_core_session.c:905 Send signalsofia/internal/1000 at 54.245.109.196[BREAK]
2012-11-21 00:40:31.538130 [DEBUG] switch_core_session.c:905 Send signalsofia/internal/1000 at 54.245.109.196[BREAK]
2012-11-21 00:40:31.558132 [DEBUG] sofia.c:6285 Channelsofia/internal/1000 at 54.245.109.196 entering state [ready][200]
2012-11-21 00:40:33.198130 [INFO] switch_rtp.c:3579 Auto Changing port from 10.0.1.15:7076 to 75.83.124.194:37261
2012-11-21 00:41:07.718130 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/sip:1001 at 75.83.124.194:35948 [BREAK]
2012-11-21 00:41:07.738130 [DEBUG] switch_channel.c:2950 (sofia/internal/sip:1001 at 75.83.124.194:35948) Callstate Change ACTIVE -> HANGUP
2012-11-21 00:41:07.738130 [DEBUG] switch_ivr_bridge.c:501 sofia/internal/sip:1001 at 75.83.124.194:35948 ending bridge by request from write function
2012-11-21 00:41:07.738130 [DEBUG] switch_ivr_bridge.c:588 BRIDGE THREAD DONE [sofia/internal/1000 at 54.245.109.196]
2012-11-21 00:41:07.738130 [DEBUG] switch_ivr_bridge.c:613 Send signal sofia/internal/sip:1001 at 75.83.124.194:35948 [BREAK]


Sent from my iPhone

Sent from my iPhone
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From anthony.minessale at gmail.com  Wed Nov 21 21:56:29 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Nov 2012 12:56:29 -0600
Subject: [Freeswitch-dev] Failed DTMF sanity check.
In-Reply-To: <CAKzWOxWZGKkGMO+Rx8OWEcy54FBni7Va+S7B7+WeTYDU7dZg5A@mail.gmail.com>
References: <CAE8KqXy6cSj87vCr2x-46CL5KsxH=+JQPEuwkvqShuk80ZUL-w@mail.gmail.com>
	<CAKzWOxWZGKkGMO+Rx8OWEcy54FBni7Va+S7B7+WeTYDU7dZg5A@mail.gmail.com>
Message-ID: <CAKbxfG9fpzYgWL3nmZr5VwHyK0cRXZTFa6MX-2zrKTueqQyAUQ@mail.gmail.com>

edit src/switch_rtp.c
uncomment this line
//#define DEBUG_2833

make install_core

now you can debug it.






On Wed, Nov 21, 2012 at 10:23 AM, Michael Collins <msc at freeswitch.org>wrote:

> Is there any noticeable delay in the amount of time between the middle
> packet(s) and the end packet(s) for the DTMF that failed? How does that
> compare to a working DTMF capture?
>
> -MC
>
> On Wed, Nov 21, 2012 at 12:58 AM, Debasish Chandra <
> debasish.chandra at telemune.net> wrote:
>
>> Hi,
>>
>> I am getting "Failed DTMF sanity check" randomly (not on every call, 2 -
>> 3 calls out of 10). Application is running using Plivo REST API, while FS
>> receiving call from Sangoma NSG TDM gateway. I have read on FS wiki
>> that "Failed DTMF sanity check" means FS never received RFC2833 DTMF 'end'
>> packet. However call tcpdump shows that DTMF digits has been sent to FS
>> from Sangoma NSG, FS showing error "Failed DTMF sanity check" and ignores
>> the digit.
>>
>> Please let me know if I need to do any extra settings like editing dtmf
>> duration or type.
>>
>> Best Regards,
>> Debasish Chandra
>>
>>
>>
>> _________________________________________________________________________
>> 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
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.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
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From debasish.chandra at telemune.net  Thu Nov 22 11:16:36 2012
From: debasish.chandra at telemune.net (Debasish Chandra)
Date: Thu, 22 Nov 2012 13:46:36 +0530
Subject: [Freeswitch-dev] Failed DTMF sanity check.
In-Reply-To: <CAKzWOxWZGKkGMO+Rx8OWEcy54FBni7Va+S7B7+WeTYDU7dZg5A@mail.gmail.com>
References: <CAE8KqXy6cSj87vCr2x-46CL5KsxH=+JQPEuwkvqShuk80ZUL-w@mail.gmail.com>
	<CAKzWOxWZGKkGMO+Rx8OWEcy54FBni7Va+S7B7+WeTYDU7dZg5A@mail.gmail.com>
Message-ID: <CAE8KqXzOauH7eKuvDYLMWGeRx8SY6VGqwAEhxj_Ou6s9KE=KYg@mail.gmail.com>

I couldn't see any significant delays between DTMF packets. Please find
attached DTMF pcap dump. It has 2 traces. 1st one was working one, while
2nd one was failed.

Best Regards,
Debasish Chandra

On Wed, Nov 21, 2012 at 9:53 PM, Michael Collins <msc at freeswitch.org> wrote:

> Is there any noticeable delay in the amount of time between the middle
> packet(s) and the end packet(s) for the DTMF that failed? How does that
> compare to a working DTMF capture?
>
> -MC
>
> On Wed, Nov 21, 2012 at 12:58 AM, Debasish Chandra <
> debasish.chandra at telemune.net> wrote:
>
>> Hi,
>>
>> I am getting "Failed DTMF sanity check" randomly (not on every call, 2 -
>> 3 calls out of 10). Application is running using Plivo REST API, while FS
>> receiving call from Sangoma NSG TDM gateway. I have read on FS wiki
>> that "Failed DTMF sanity check" means FS never received RFC2833 DTMF 'end'
>> packet. However call tcpdump shows that DTMF digits has been sent to FS
>> from Sangoma NSG, FS showing error "Failed DTMF sanity check" and ignores
>> the digit.
>>
>> Please let me know if I need to do any extra settings like editing dtmf
>> duration or type.
>>
>> Best Regards,
>> Debasish Chandra
>>
>>
>>
>> _________________________________________________________________________
>> 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
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.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 kheimerl at cs.berkeley.edu  Mon Nov 26 02:40:23 2012
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Mon, 26 Nov 2012 08:40:23 +0900
Subject: [Freeswitch-dev] Return code from ESL Message Sending
In-Reply-To: <9E235C2C-3884-45CF-AAD0-03D422C3BD28@edge-net.net>
References: <mailman.2378.1353837561.576.freeswitch-users@lists.freeswitch.org>
	<9E235C2C-3884-45CF-AAD0-03D422C3BD28@edge-net.net>
Message-ID: <CACyT-3kAN4-TgEATOdbuKE4-6f-Ygpb5DG6SDWdRJRwyeY4S6w@mail.gmail.com>

Sure. I'm sending a message through event creation, as detailed here on the
wiki:

http://wiki.freeswitch.org/wiki/Mod_sms#Sending_a_Message_from_a_script

The exact script i'm using to do this is here:
https://github.com/kheimerl/libvbts/blob/master/libvbts/FreeSwitchMessenger.py

On Monday, November 26, 2012, Eli Burke wrote:

> Kurtis,
>
> I wouldn't be surprised if it was caused by those patches. Unfortunately
> (or perhaps fortunately) I didn't write them. And sending messages via
> direct event creation wasn't part of our test matrix. Can you give me some
> more detail on exactly what you are doing?
> FWIW, we tested from handsets, from ESL scripts, and from fs_cli.
>
> -ELi
>
> On Nov 25, 2012, at 4:59 AM, freeswitch-users-request at lists.freeswitch.org<javascript:_e({}, 'cvml',
> 'freeswitch-users-request at lists.freeswitch.org');> wrote:
>
> *From: *Kurtis Heimerl <kheimerl at cs.berkeley.edu <javascript:_e({},
> 'cvml', 'kheimerl at cs.berkeley.edu');>>
> *Subject: **Re: [Freeswitch-users] Return code from ESL Message Sending*
> *Date: *November 25, 2012 1:00:08 AM EST
> *To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<javascript:_e({}, 'cvml', 'freeswitch-users at lists.freeswitch.org');>
> >
> *Reply-To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<javascript:_e({}, 'cvml', 'freeswitch-users at lists.freeswitch.org');>
> >
>
>
> Hi Eli,
>
> I recently updated my installation, and now my old scripts for sending
> messages through event creation aren't working. Basically, FS is sending
> tens to hundreds of MESSAGEs a second after I send a SMS::SEND_MESSAGE event.
> This was working before your patch (and I'm sure a bunch of other patches),
> so I'm hoping you have some intuition as to what might be broken. I'll
> start tracking it down myself here in a little bit.
>
>
>
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From brian at freeswitch.org  Mon Nov 26 02:57:54 2012
From: brian at freeswitch.org (Brian West)
Date: Sun, 25 Nov 2012 17:57:54 -0600
Subject: [Freeswitch-dev] Return code from ESL Message Sending
In-Reply-To: <CACyT-3kAN4-TgEATOdbuKE4-6f-Ygpb5DG6SDWdRJRwyeY4S6w@mail.gmail.com>
References: <mailman.2378.1353837561.576.freeswitch-users@lists.freeswitch.org>
	<9E235C2C-3884-45CF-AAD0-03D422C3BD28@edge-net.net>
	<CACyT-3kAN4-TgEATOdbuKE4-6f-Ygpb5DG6SDWdRJRwyeY4S6w@mail.gmail.com>
Message-ID: <E2231006-6F79-4743-8874-B75BE21F0878@freeswitch.org>

And if its really a bug you should be opening a jira.
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 25, 2012, at 5:40 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:

> Sure. I'm sending a message through event creation, as detailed here on the wiki:
> 
> http://wiki.freeswitch.org/wiki/Mod_sms#Sending_a_Message_from_a_script
> 
> The exact script i'm using to do this is here:
> https://github.com/kheimerl/libvbts/blob/master/libvbts/FreeSwitchMessenger.py
> 
> On Monday, November 26, 2012, Eli Burke wrote:
> Kurtis,
> 
> I wouldn't be surprised if it was caused by those patches. Unfortunately (or perhaps fortunately) I didn't write them. And sending messages via direct event creation wasn't part of our test matrix. Can you give me some more detail on exactly what you are doing?
> FWIW, we tested from handsets, from ESL scripts, and from fs_cli.
> 
> -ELi
>  
> On Nov 25, 2012, at 4:59 AM, freeswitch-users-request at lists.freeswitch.org wrote:
> 
>> From: Kurtis Heimerl <kheimerl at cs.berkeley.edu>
>> Subject: Re: [Freeswitch-users] Return code from ESL Message Sending
>> Date: November 25, 2012 1:00:08 AM EST
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> 
>> 
>> Hi Eli,
>> 
>> I recently updated my installation, and now my old scripts for sending messages through event creation aren't working. Basically, FS is sending tens to hundreds of MESSAGEs a second after I send a SMS::SEND_MESSAGE event. This was working before your patch (and I'm sure a bunch of other patches), so I'm hoping you have some intuition as to what might be broken. I'll start tracking it down myself here in a little bit. 
> 
> _________________________________________________________________________
> 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



From kheimerl at cs.berkeley.edu  Mon Nov 26 03:17:13 2012
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Mon, 26 Nov 2012 09:17:13 +0900
Subject: [Freeswitch-dev] Return code from ESL Message Sending
In-Reply-To: <E2231006-6F79-4743-8874-B75BE21F0878@freeswitch.org>
References: <mailman.2378.1353837561.576.freeswitch-users@lists.freeswitch.org>
	<9E235C2C-3884-45CF-AAD0-03D422C3BD28@edge-net.net>
	<CACyT-3kAN4-TgEATOdbuKE4-6f-Ygpb5DG6SDWdRJRwyeY4S6w@mail.gmail.com>
	<E2231006-6F79-4743-8874-B75BE21F0878@freeswitch.org>
Message-ID: <CACyT-3=1o8xW3s8VRHzxopegFECxhd+71ymsRjv5oAS48HM8zg@mail.gmail.com>

Here: http://jira.freeswitch.org/browse/FS-4872

Though someone should add mod_sms to the components list. I had to use
freeswitch-core.

On Monday, November 26, 2012, Brian West wrote:

> And if its really a bug you should be opening a jira.
> --
> Brian West
> brian at freeswitch.org <javascript:;>
> FreeSWITCH Solutions, LLC
> PO BOX PO BOX 2531
> Brookfield, WI 53008-2531
> Twitter: @FreeSWITCH_Wire
> T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
> iNUM: +883 5100 1420 9266
> UK: +44 20 3298 4900
> ISN: 410*543
>
>
>
>
>
> On Nov 25, 2012, at 5:40 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu<javascript:;>>
> wrote:
>
> > Sure. I'm sending a message through event creation, as detailed here on
> the wiki:
> >
> > http://wiki.freeswitch.org/wiki/Mod_sms#Sending_a_Message_from_a_script
> >
> > The exact script i'm using to do this is here:
> >
> https://github.com/kheimerl/libvbts/blob/master/libvbts/FreeSwitchMessenger.py
> >
> > On Monday, November 26, 2012, Eli Burke wrote:
> > Kurtis,
> >
> > I wouldn't be surprised if it was caused by those patches. Unfortunately
> (or perhaps fortunately) I didn't write them. And sending messages via
> direct event creation wasn't part of our test matrix. Can you give me some
> more detail on exactly what you are doing?
> > FWIW, we tested from handsets, from ESL scripts, and from fs_cli.
> >
> > -ELi
> >
> > On Nov 25, 2012, at 4:59 AM,
> freeswitch-users-request at lists.freeswitch.org <javascript:;> wrote:
> >
> >> From: Kurtis Heimerl <kheimerl at cs.berkeley.edu <javascript:;>>
> >> Subject: Re: [Freeswitch-users] Return code from ESL Message Sending
> >> Date: November 25, 2012 1:00:08 AM EST
> >> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<javascript:;>
> >
> >> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<javascript:;>
> >
> >>
> >>
> >> Hi Eli,
> >>
> >> I recently updated my installation, and now my old scripts for sending
> messages through event creation aren't working. Basically, FS is sending
> tens to hundreds of MESSAGEs a second after I send a SMS::SEND_MESSAGE
> event. This was working before your patch (and I'm sure a bunch of other
> patches), so I'm hoping you have some intuition as to what might be broken.
> I'll start tracking it down myself here in a little bit.
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org <javascript:;>
> > 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 <javascript:;>
> > 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 <javascript:;>
> 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 <javascript:;>
> 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 rodnet at gmail.com  Mon Nov 26 06:52:23 2012
From: rodnet at gmail.com (Rodney)
Date: Mon, 26 Nov 2012 14:52:23 +1100
Subject: [Freeswitch-dev] Redis Module Implementation
Message-ID: <CAA0Jt6-obx7GVzFto32m-zNHAO3MehajvAUE+wERaC9C6uhsZg@mail.gmail.com>

I'm currently looking at the redis module as a way of implementing limits
that are shared across multiple servers and noticed that for every limit I
create, two keys are set in redis.

One key is what I'm expecting, with the name derived from the values I've
supplied, however the other key is the same thing but with the hostname
prefixed.

Looking at the code, this could be either the hostname or the switchname
that is being prefixed (depending on if a switchname is set).

My question is basically why is it doing this ? I would have assumed that
if a dialplan author wanted the key in question to have the hostname or
switch name, couldn't they just include it in the call to limit in the dial
plan ?

It also looks to me like there could be a bug where the hostname prefixed
key only gets incremented if there is a max set but will always get
decremented.

I'm just trying to work out if this is intended behaviour and I have just
misunderstood the purpose of the module, or if it is worth trying to put a
patch together.

Thanks,
Rodney
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From openser at yeah.net  Mon Nov 26 13:46:27 2012
From: openser at yeah.net (openser)
Date: Mon, 26 Nov 2012 18:46:27 +0800 (CST)
Subject: [Freeswitch-dev] rtcp-mux support
Message-ID: <16cd812.5669.13b3c53434a.Coremail.openser@yeah.net>

Hi all,


Does freeswitch support rtcp-mux feature ? if it support , freeswitch should send rtcp packet by rtp port but not rtp +1  for given audio/video session,
my client support rtcp-mux , but freeswitch send rtcp to port rtp +1 .



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From DAVIDROT at synel.co.il  Mon Nov 26 14:53:32 2012
From: DAVIDROT at synel.co.il (David Rot)
Date: Mon, 26 Nov 2012 13:53:32 +0200
Subject: [Freeswitch-dev] FreeSWITCH-dev Digest, Vol 77, Issue 6
In-Reply-To: <mailman.2413.1353917568.576.freeswitch-dev@lists.freeswitch.org>
References: <mailman.2413.1353917568.576.freeswitch-dev@lists.freeswitch.org>
Message-ID: <1A07FEDC778C3E4AA417F089511F859C9C94C42362@Mail-yoq.SYNEL-COMPANY.LTD>

Dears 


 1.	In some occasion I get the caller-id with an extension, and in other occasions I get the caller-id without an extension.
    How do I make sure that I always get the caller-id with an extension?
 2.	Soon after I perform a recording, I hear a tick sound (like hanging up) and then the call continues without any problems. This situation is very confusing for our customers as they believe that the call was disconnected (because of the tick sound). How can I avoid the tick sound that causes the confusion?


Thanks  


From sparklezou at 163.com  Mon Nov 26 17:58:46 2012
From: sparklezou at 163.com (sparklezou)
Date: Mon, 26 Nov 2012 22:58:46 +0800
Subject: [Freeswitch-dev] About module "Mod_pandsp"
Message-ID: <71a1dca6.4c4e.13b3d3cd561.Coremail.sparklezou@163.com>

Hi Mod_spandsp developers,

Is it possible to develop Mod_spandsp like "Mod_voicemail"?

I means, set all of the related parameters in the user configure file. such as  "vm-enabled", "vm-mailto", "vm-mailfrom", and so on. Also the mail sender configuration, such as SMTP.

Something seemd the same, send "voice message" via eamil. Send "fax file" via email.

Thanks!

2012-11-26



sparklezou
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From krice at freeswitch.org  Mon Nov 26 19:35:54 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 26 Nov 2012 10:35:54 -0600
Subject: [Freeswitch-dev] About module "Mod_pandsp"
In-Reply-To: <71a1dca6.4c4e.13b3d3cd561.Coremail.sparklezou@163.com>
Message-ID: <CCD8F68A.6113C%krice@freeswitch.org>

Huh what? Mod_spandsp is a module that provides underlying fax and codec
support, theres not really anything in there to ?develop? ....

If you are talking about fax routing you can still put the settings int he
user directory, you just need to handle the routing of the faxes in your
dailplan or other mechanism...


On 11/26/12 8:58 AM, "sparklezou" <sparklezou at 163.com> wrote:

> Hi Mod_spandsp developers,
>  
> Is it possible to develop Mod_spandsp like "Mod_voicemail"?
>  
> I means, set all of the related parameters in the user configure file. such as
> "vm-enabled", "vm-mailto", "vm-mailfrom", and so on. Also the mail sender
> configuration, such as SMTP.
>  
> Something seemd the same, send "voice message" via eamil. Send "fax file" via
> email.
>  
> Thanks!
>  
> 2012-11-26
> 
> sparklezou
> 
> 
> _________________________________________________________________________
> 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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From msc at freeswitch.org  Mon Nov 26 19:58:04 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 26 Nov 2012 08:58:04 -0800
Subject: [Freeswitch-dev] Failed DTMF sanity check.
In-Reply-To: <CAE8KqXzOauH7eKuvDYLMWGeRx8SY6VGqwAEhxj_Ou6s9KE=KYg@mail.gmail.com>
References: <CAE8KqXy6cSj87vCr2x-46CL5KsxH=+JQPEuwkvqShuk80ZUL-w@mail.gmail.com>
	<CAKzWOxWZGKkGMO+Rx8OWEcy54FBni7Va+S7B7+WeTYDU7dZg5A@mail.gmail.com>
	<CAE8KqXzOauH7eKuvDYLMWGeRx8SY6VGqwAEhxj_Ou6s9KE=KYg@mail.gmail.com>
Message-ID: <CAKzWOxX60WX=j_d0UkC2oQgD8w+r3GQY4svattDMzn0TM11Huw@mail.gmail.com>

That pcap only contains a small UDP stream. It would be better if you had
the whole call, including signaling. It's much easier to see what's going
on that way. I would use pcapsipdump.
-MC

On Thu, Nov 22, 2012 at 12:16 AM, Debasish Chandra <
debasish.chandra at telemune.net> wrote:

> I couldn't see any significant delays between DTMF packets. Please find
> attached DTMF pcap dump. It has 2 traces. 1st one was working one, while
> 2nd one was failed.
>
> Best Regards,
> Debasish Chandra
>
> On Wed, Nov 21, 2012 at 9:53 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Is there any noticeable delay in the amount of time between the middle
>> packet(s) and the end packet(s) for the DTMF that failed? How does that
>> compare to a working DTMF capture?
>>
>> -MC
>>
>> On Wed, Nov 21, 2012 at 12:58 AM, Debasish Chandra <
>> debasish.chandra at telemune.net> wrote:
>>
>>> Hi,
>>>
>>> I am getting "Failed DTMF sanity check" randomly (not on every call, 2 -
>>> 3 calls out of 10). Application is running using Plivo REST API, while FS
>>> receiving call from Sangoma NSG TDM gateway. I have read on FS wiki
>>> that "Failed DTMF sanity check" means FS never received RFC2833 DTMF 'end'
>>> packet. However call tcpdump shows that DTMF digits has been sent to FS
>>> from Sangoma NSG, FS showing error "Failed DTMF sanity check" and ignores
>>> the digit.
>>>
>>> Please let me know if I need to do any extra settings like editing dtmf
>>> duration or type.
>>>
>>> Best Regards,
>>> Debasish Chandra
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> 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
>>>
>>>
>>
>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.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
>>
>>
>
> _________________________________________________________________________
> 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From steveayre at gmail.com  Mon Nov 26 20:15:04 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 26 Nov 2012 17:15:04 +0000
Subject: [Freeswitch-dev] About module "Mod_pandsp"
In-Reply-To: <71a1dca6.4c4e.13b3d3cd561.Coremail.sparklezou@163.com>
References: <71a1dca6.4c4e.13b3d3cd561.Coremail.sparklezou@163.com>
Message-ID: <CAFiqYumi247K8MeYHZT0ALDybUnM2KuGxpYmN_6G0z0ejE1dSw@mail.gmail.com>

You can setting variables in a user directory entry (<variable> entries in
<variables> section). Examples are on the Wiki.

They'll be set for incoming calls and would affect the behaviour.

If you're wanting to set them on the outbound channels based on
destination, I'd suggest configuring the destinations as sofia gateways. In
the gateway definitions you can set variables to be set on
incoming/outgoing/both calls via that gateway... setting outbound variables
there should affect the bleg. Again, examples on the wiki.


On 26 November 2012 14:58, sparklezou <sparklezou at 163.com> wrote:

> **
> **
> Hi Mod_spandsp developers,
>
> Is it possible to develop Mod_spandsp like "Mod_voicemail"?
>
> I means, set all of the related parameters in the user configure file.
> such as "vm-enabled", "vm-mailto", "vm-mailfrom", and so on. Also the mail
> sender configuration, such as SMTP.
>
> Something seemd the same, send "voice message" via eamil. Send "fax file"
> via email.
>
> Thanks!
>
> 2012-11-26
> ------------------------------
>  sparklezou
> **
>
> _________________________________________________________________________
> 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  Mon Nov 26 21:06:55 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 26 Nov 2012 10:06:55 -0800
Subject: [Freeswitch-dev] FreeSWITCH-dev Digest, Vol 77, Issue 6
In-Reply-To: <1A07FEDC778C3E4AA417F089511F859C9C94C42362@Mail-yoq.SYNEL-COMPANY.LTD>
References: <mailman.2413.1353917568.576.freeswitch-dev@lists.freeswitch.org>
	<1A07FEDC778C3E4AA417F089511F859C9C94C42362@Mail-yoq.SYNEL-COMPANY.LTD>
Message-ID: <CAKzWOxXa=Wwe2FS7kAzuHSgm1Vyy1vcZiybkMOipdFikyRtnXQ@mail.gmail.com>

1. Do you have SIP traces of calls that do and do not get the extension? If
so, post to pastebin.freeswitch.org and the gang here will take a look.
2. It depends on where the "tick" sound comes from. It may be best to play
a brief audio call right after the hangup/tick that says "call continuing"
or something like that.

-MC

On Mon, Nov 26, 2012 at 3:53 AM, David Rot <DAVIDROT at synel.co.il> wrote:

> Dears
>
>
>  1.     In some occasion I get the caller-id with an extension, and in
> other occasions I get the caller-id without an extension.
>     How do I make sure that I always get the caller-id with an extension?
>  2.     Soon after I perform a recording, I hear a tick sound (like
> hanging up) and then the call continues without any problems. This
> situation is very confusing for our customers as they believe that the call
> was disconnected (because of the tick sound). How can I avoid the tick
> sound that causes the confusion?
>
>
> Thanks
>
> _________________________________________________________________________
> 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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From krice at freeswitch.org  Mon Nov 26 21:31:52 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 26 Nov 2012 12:31:52 -0600
Subject: [Freeswitch-dev] FreeSWITCH-dev Digest, Vol 77, Issue 6
In-Reply-To: <CAKzWOxXa=Wwe2FS7kAzuHSgm1Vyy1vcZiybkMOipdFikyRtnXQ@mail.gmail.com>
Message-ID: <CCD911B8.61151%krice@freeswitch.org>

Also, please do not hijack threads, please start a new email chain and set a
reasonable subject...
Yes I know its usually easier then clicking reply and maybe changing the
subject line, but it really helps the list management software separate out
replys from new threads, and it doesn?t cause many peoples email clients
(like mine) to throw your new thread under an existing one.

Thanks
K


On 11/26/12 12:06 PM, "Michael Collins" <msc at freeswitch.org> wrote:

> 1. Do you have SIP traces of calls that do and do not get the extension? If
> so, post to pastebin.freeswitch.org <http://pastebin.freeswitch.org>  and the
> gang here will take a look.
> 2. It depends on where the "tick" sound comes from. It may be best to play a
> brief audio call right after the hangup/tick that says "call continuing" or
> something like that.
> 
> -MC
> 
> On Mon, Nov 26, 2012 at 3:53 AM, David Rot <DAVIDROT at synel.co.il> wrote:
>> Dears
>> 
>> 
>> ?1. ? ? In some occasion I get the caller-id with an extension, and in other
>> occasions I get the caller-id without an extension.
>> ? ? How do I make sure that I always get the caller-id with an extension?
>> ?2. ? ? Soon after I perform a recording, I hear a tick sound (like hanging
>> up) and then the call continues without any problems. This situation is very
>> confusing for our customers as they believe that the call was disconnected
>> (because of the tick sound). How can I avoid the tick sound that causes the
>> confusion?
>> 
>> 
>> Thanks
>> 
>> _________________________________________________________________________
>> 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
> 
> 

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From abaci64 at gmail.com  Mon Nov 26 21:57:07 2012
From: abaci64 at gmail.com (Abaci)
Date: Mon, 26 Nov 2012 13:57:07 -0500
Subject: [Freeswitch-dev] About module "Mod_pandsp"
In-Reply-To: <71a1dca6.4c4e.13b3d3cd561.Coremail.sparklezou@163.com>
References: <71a1dca6.4c4e.13b3d3cd561.Coremail.sparklezou@163.com>
Message-ID: <50B3BB83.60700@gmail.com>

I think what he's looking for is automatic fax to email like Voicemail 
to email is handled. this is not something FreeSWITCH has, not sure if 
it's because of related patents or something else.

On 11/26/2012 9:58 AM, sparklezou wrote:
> Hi Mod_spandsp developers,
> Is it possible to develop Mod_spandsp like "Mod_voicemail"?
> I means, set all of the related parameters in the user configure file. 
> such as "vm-enabled", "vm-mailto", "vm-mailfrom", and so on. Also the 
> mail sender configuration, such as SMTP.
> Something seemd the same, send "voice message" via eamil. Send "fax 
> file" via email.
> Thanks!
> 2012-11-26
> ------------------------------------------------------------------------
> sparklezou
>
>
> _________________________________________________________________________
> 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  Mon Nov 26 22:35:01 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 26 Nov 2012 11:35:01 -0800
Subject: [Freeswitch-dev] About module "Mod_pandsp"
In-Reply-To: <50B3BB83.60700@gmail.com>
References: <71a1dca6.4c4e.13b3d3cd561.Coremail.sparklezou@163.com>
	<50B3BB83.60700@gmail.com>
Message-ID: <CAKzWOxUdFmF9kZ+p43n6Ft=RsLzF7jdU4NAd0QLykwVuy2uyOQ@mail.gmail.com>

That would be a nice feature to have but I don't know if mod_spandsp is the
right place for it. A better approach might be to have one of our intrepid
community members write, debug, test, test, test (and test some more) a
script that we could include in the vanilla configs. It would have the same
net effect and would avoid adding potentially unnecessary code to
mod_spandsp.

Any takers?

-MC

On Mon, Nov 26, 2012 at 10:57 AM, Abaci <abaci64 at gmail.com> wrote:

>  I think what he's looking for is automatic fax to email like Voicemail
> to email is handled. this is not something FreeSWITCH has, not sure if it's
> because of related patents or something else.
>
> On 11/26/2012 9:58 AM, sparklezou wrote:
>
> **
> Hi Mod_spandsp developers,
>
> Is it possible to develop Mod_spandsp like "Mod_voicemail"?
>
> I means, set all of the related parameters in the user configure file.
> such as "vm-enabled", "vm-mailto", "vm-mailfrom", and so on. Also the mail
> sender configuration, such as SMTP.
>
> Something seemd the same, send "voice message" via eamil. Send "fax file"
> via email.
>
> Thanks!
>
> 2012-11-26
>  ------------------------------
>  sparklezou
> **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-dev mailing listFreeSWITCH-dev at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-devhttp://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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From marketing at cluecon.com  Mon Nov 26 22:39:32 2012
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 26 Nov 2012 11:39:32 -0800
Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxW1uQwkvZN=-oQoYCX63oHQiC31ydSuPiey_=2FkkJnMQ@mail.gmail.com>

Hello all!

We are all back to a full week after many of us enjoyed some well-deserved
time off last week. However, even though there was a holiday here in the
US, the intrepid FreeSWITCH development team was working hard on your
behalf. As Ken Rice previously mentioned, Anthony spotted a potential issue
in the recently released 1.2.5 version. Therefore, this past Saturday they
made 1.2.5.1 available<http://files.freeswitch.org/freeswitch-1.2.5.1.tar.bz2>for
us. Many thanks to those who work so hard to make sure that FreeSWITCH
is running smoothly for us all.

On last week's conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_21>we spent
some time getting everyone up to speed on how to edit the FreeSWITCH
wiki <http://wiki.freeswitch.org>, specifically focusing on channel
variables <http://wiki.freeswitch.org/wiki/Channel_Variables> pages.
Updating documentation is one of the least glamorous aspects of maintaining
an open source project. Many thanks to those who've stepped up over the
past weeks and months to help us out. With the end of the year upon us we
are slowing down a bit in our speaking schedule for the weekly community
conference call. We have a few things in the works but nothing yet
scheduled. On this week's
call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_28>we will be
doing a community scrum. Be sure to bring your questions and
topics for discussion. If you have a tip or trick that you'd like to share
with the group that would be most welcomed. If time permits we will
crowdsource a few selected questions from the mailing list.

Have a great week and we'll talk to you on Wednesday.

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From anthony.minessale at gmail.com  Tue Nov 27 01:38:04 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Nov 2012 16:38:04 -0600
Subject: [Freeswitch-dev] rtcp-mux support
In-Reply-To: <16cd812.5669.13b3c53434a.Coremail.openser@yeah.net>
References: <16cd812.5669.13b3c53434a.Coremail.openser@yeah.net>
Message-ID: <CAKbxfG8nBWQPxCZbevUD2rNSO8mSAbro=AX9xWFvq9REqVe3ZA@mail.gmail.com>

no, it does not.  We will probably support it in the future though.



On Mon, Nov 26, 2012 at 4:46 AM, openser <openser at yeah.net> wrote:

> Hi all,
>
> Does freeswitch support rtcp-mux feature ? if it support , freeswitch
> should send rtcp packet by rtp port but not rtp +1  for given audio/video
> session,
> my client support rtcp-mux , but freeswitch send rtcp to port rtp +1 .
>
>
>
>
>
> _________________________________________________________________________
> 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
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From sparklezou at 163.com  Tue Nov 27 02:52:25 2012
From: sparklezou at 163.com (sparklezou)
Date: Tue, 27 Nov 2012 07:52:25 +0800
Subject: [Freeswitch-dev] About module "Mod_pandsp"
In-Reply-To: <50B3BB83.60700@gmail.com>
References: <50B3BB83.60700@gmail.com>
Message-ID: <56435cdd.20c.13b3f25698d.Coremail.sparklezou@163.com>

Hi All,

Correct! 

I hope the emaill function for voice-message and fax file coulde be combind.

Both "Mod_spandsp" and "Mod_voicemail" could get the usre email configuration from the user profile. such as the current "vm-enabled", "vm-mailto", "vm-mailfrom" parameters in voice-message.

It will be much easy to implement fax2email function. And much more coupling.

Most of the email fuctions should be the same for both modules.

Thanks!

2012-11-27



sparklezou



????Abaci
?????2012-11-27 02:57
???Re: [Freeswitch-dev] About module "Mod_pandsp"
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

I think what he's looking for is automatic fax to email like Voicemail to email is handled. this is not something FreeSWITCH has, not sure if it's because of related patents or something else.

On 11/26/2012 9:58 AM, sparklezou wrote:

Hi Mod_spandsp developers,

Is it possible to develop Mod_spandsp like "Mod_voicemail"?

I means, set all of the related parameters in the user configure file. such as "vm-enabled", "vm-mailto", "vm-mailfrom", and so on. Also the mail sender configuration, such as SMTP.

Something seemd the same, send "voice message" via eamil. Send "fax file" via email.

Thanks!

2012-11-26



sparklezou

 

_________________________________________________________________________
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 daggelinckxmichel at gmail.com  Tue Nov 27 05:10:47 2012
From: daggelinckxmichel at gmail.com (Michel Daggelinckx)
Date: Tue, 27 Nov 2012 03:10:47 +0100
Subject: [Freeswitch-dev] rpm problem
Message-ID: <50B42127.3010404@gmail.com>

Just updated my test system and got

Package freeswitch-application-sms-1.2.5.1-1.el6.i386.rpm is not signed


Michel



From krice at freeswitch.org  Tue Nov 27 05:27:02 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 26 Nov 2012 20:27:02 -0600
Subject: [Freeswitch-dev] rpm problem
In-Reply-To: <50B42127.3010404@gmail.com>
Message-ID: <CCD98116.61199%krice@freeswitch.org>

Ack! Let me check that out...




On 11/26/12 8:10 PM, "Michel Daggelinckx" <daggelinckxmichel at gmail.com>
wrote:

> Just updated my test system and got
> 
> Package freeswitch-application-sms-1.2.5.1-1.el6.i386.rpm is not signed
> 
> 
> Michel
> 
> 
> _________________________________________________________________________
> 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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From sparklezou at 163.com  Wed Nov 28 12:16:28 2012
From: sparklezou at 163.com (sparklezou)
Date: Wed, 28 Nov 2012 17:16:28 +0800
Subject: [Freeswitch-dev] About the "transfer" function
Message-ID: <f996992.43dc.13b46500e9d.Coremail.sparklezou@163.com>

Hi All,

When dial an internal number "1234", it will "bridge" or "transfer" to an external number "87654321".

And on the caller phone will disply "Outbound Call", also show the number "87654321".

I checked the sip message, the info is got from the 180 or 183 message.

Sometime, don't want to show the real external number "87654321" to the enduser.

How to keep the caller phone displaying "1234".

Thanks in advance!

2012-11-28



sparklezou
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From brian at freeswitch.org  Thu Nov 29 04:41:37 2012
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Nov 2012 19:41:37 -0600
Subject: [Freeswitch-dev] About the "transfer" function
In-Reply-To: <f996992.43dc.13b46500e9d.Coremail.sparklezou@163.com>
References: <f996992.43dc.13b46500e9d.Coremail.sparklezou@163.com>
Message-ID: <B7CEC52B-11E6-4DDF-A395-CF1BCDD63B7E@freeswitch.org>

search wiki for ignore_display_updates. 
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 28, 2012, at 3:16 AM, sparklezou <sparklezou at 163.com> wrote:

> Hi All,
>  
> When dial an internal number "1234", it will "bridge" or "transfer" to an external number "87654321".
>  
> And on the caller phone will disply "Outbound Call", also show the number "87654321".
>  
> I checked the sip message, the info is got from the 180 or 183 message.
>  
> Sometime, don't want to show the real external number "87654321" to the enduser.
>  
> How to keep the caller phone displaying "1234".
>  
> Thanks in advance!
>  
> 2012-11-28
> sparklezou
> _________________________________________________________________________
> 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



From gabe at gundy.org  Tue Nov 27 07:19:48 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 26 Nov 2012 21:19:48 -0700
Subject: [Freeswitch-dev] rpm problem
In-Reply-To: <50B42127.3010404@gmail.com>
References: <50B42127.3010404@gmail.com>
Message-ID: <CAMwZ-tJRJ61DHdYU0hee9nfLusps4-yktnjx76KEQUUOY3R=6A@mail.gmail.com>

On Mon, Nov 26, 2012 at 7:10 PM, Michel Daggelinckx
<daggelinckxmichel at gmail.com> wrote:
> Just updated my test system and got
>
> Package freeswitch-application-sms-1.2.5.1-1.el6.i386.rpm is not signed

You should include a link to the Jira you opened so that Ken knows
where he should log information about this bug. You did open one,
right? ;)


Best
Gabe


From brian at freeswitch.org  Thu Nov 29 04:58:54 2012
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Nov 2012 19:58:54 -0600
Subject: [Freeswitch-dev] rpm problem
In-Reply-To: <CAMwZ-tJRJ61DHdYU0hee9nfLusps4-yktnjx76KEQUUOY3R=6A@mail.gmail.com>
References: <50B42127.3010404@gmail.com>
	<CAMwZ-tJRJ61DHdYU0hee9nfLusps4-yktnjx76KEQUUOY3R=6A@mail.gmail.com>
Message-ID: <E3B72169-595E-49F5-BF5E-D4505C78F225@freeswitch.org>

You can now if you try.  Everything is ALIVE!
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 26, 2012, at 10:19 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> You should include a link to the Jira you opened so that Ken knows
> where he should log information about this bug. You did open one,
> right? ;)
> 
> 
> Best
> Gabe



From sparklezou at 163.com  Thu Nov 29 05:25:49 2012
From: sparklezou at 163.com (sparklezou)
Date: Thu, 29 Nov 2012 10:25:49 +0800
Subject: [Freeswitch-dev] [Freeswitch-users] About the "transfer"
	function
In-Reply-To: <B7CEC52B-11E6-4DDF-A395-CF1BCDD63B7E@freeswitch.org>
References: <B7CEC52B-11E6-4DDF-A395-CF1BCDD63B7E@freeswitch.org>
Message-ID: <2b32dd13.12a6.13b49fe985e.Coremail.sparklezou@163.com>

Hi All,

Thanks! I have fixed it.

The "update" info will be notify from "180", "183", "UPDATE" sip message.

BR,
Zou Yu

2012-11-29



sparklezou



????Brian West
?????2012-11-29 09:41
???Re: [Freeswitch-users] [Freeswitch-dev] About the "transfer" function
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???"freeswitch-users"<freeswitch-users at lists.freeswitch.org>

search wiki for ignore_display_updates.  
-- 
Brian West 
brian at freeswitch.org 
FreeSWITCH Solutions, LLC 
PO BOX PO BOX 2531 
Brookfield, WI 53008-2531 
Twitter: @FreeSWITCH_Wire 
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST 
iNUM: +883 5100 1420 9266 
UK: +44 20 3298 4900 
ISN: 410*543 





On Nov 28, 2012, at 3:16 AM, sparklezou <sparklezou at 163.com> wrote: 

> Hi All, 
>   
> When dial an internal number "1234", it will "bridge" or "transfer" to an external number "87654321". 
>   
> And on the caller phone will disply "Outbound Call", also show the number "87654321". 
>   
> I checked the sip message, the info is got from the 180 or 183 message. 
>   
> Sometime, don't want to show the real external number "87654321" to the enduser. 
>   
> How to keep the caller phone displaying "1234". 
>   
> Thanks in advance! 
>   
> 2012-11-28 
> sparklezou 
> _________________________________________________________________________ 
> 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 sparklezou at 163.com  Fri Nov 30 06:19:48 2012
From: sparklezou at 163.com (sparklezou)
Date: Fri, 30 Nov 2012 11:19:48 +0800
Subject: [Freeswitch-dev] About the display problem during calling
Message-ID: <5c994372.3147.13b4f55d827.Coremail.sparklezou@163.com>

Hi All,

I facing a strange problem.

Before, when calling Internal number, the name and number will both be displayed on the caller phone. When connected, only the name will display on the phone. At called phone, display the name and number. Seems fine.

But yesterday afternoon, when calling Internal number, only the number is displayed on the caller phone. When connected, display "Outbound Call" on the caller phone. At the called phone, everthing is the same. Display both the name and number.

I think, mabybe something configured wrong. So I try to restore back the configuration. seems the same. I have to re-install the last release stable 1.2.5.1 version on another server. total fresh setting. But found the same problem.

I checked the sip info.

The "Outbound Call" is in the 200 SDP message. the called is set to 

""Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

Not the correct called name.

Is there anything wrong?

Please help me.

Thanks!
 

2012-11-30



sparklezou
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From sparklezou at 163.com  Fri Nov 30 10:53:03 2012
From: sparklezou at 163.com (sparklezou)
Date: Fri, 30 Nov 2012 15:53:03 +0800
Subject: [Freeswitch-dev] [Freeswitch-users] About the display problem
	during calling
In-Reply-To: <5c994372.3147.13b4f55d827.Coremail.sparklezou@163.com>
References: <5c994372.3147.13b4f55d827.Coremail.sparklezou@163.com>
Message-ID: <79e311e1.6599.13b504fd0fa.Coremail.sparklezou@163.com>

Hi All,

Before, I saved the user name in the phone. So it will show the name when ringing.

From the FreeSwitch side, there is NO difference.

@ Develop Team,

For Internal call, or any sip call from gateway, it should be better show the Name during ring.  And also keep show the name in the call, in stead of "Outbound Call".

Usually the phone will update the display name from the sip response message.

So in the 100, 180, 183, 200 sip/sdp message. Respons with the called name, like "ABC"<sip: 1234 at a.b.c.d>.

Currently in the inital 100,180,183 message, only <sip: 1234 at a.b.c.d>. In the 200 message "Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

If it could be updated, then the called name could be displayed on the caller phone.

Thanks!

2012-11-30



sparklezou



????sparklezou
?????2012-11-30 11:19
???[Freeswitch-users] About the display problem during calling
????"freeswitch-users"<freeswitch-users at lists.freeswitch.org>,"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi All,

I facing a strange problem.

Before, when calling Internal number, the name and number will both be displayed on the caller phone. When connected, only the name will display on the phone. At called phone, display the name and number. Seems fine.

But yesterday afternoon, when calling Internal number, only the number is displayed on the caller phone. When connected, display "Outbound Call" on the caller phone. At the called phone, everthing is the same. Display both the name and number.

I think, mabybe something configured wrong. So I try to restore back the configuration. seems the same. I have to re-install the last release stable 1.2.5.1 version on another server. total fresh setting. But found the same problem.

I checked the sip info.

The "Outbound Call" is in the 200 SDP message. the called is set to 

""Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

Not the correct called name.

Is there anything wrong?

Please help me.

Thanks!


2012-11-30



sparklezou
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From sparklezou at 163.com  Fri Nov 30 16:11:45 2012
From: sparklezou at 163.com (sparklezou)
Date: Fri, 30 Nov 2012 21:11:45 +0800
Subject: [Freeswitch-dev] [Freeswitch-users] About the display problem
	during calling
In-Reply-To: <79e311e1.6599.13b504fd0fa.Coremail.sparklezou@163.com>
References: <5c994372.3147.13b4f55d827.Coremail.sparklezou@163.com><79e311e1.6599.13b504fd0fa.Coremail.sparklezou@163.com>
Message-ID: <35a86017.4737.13b51737748.Coremail.sparklezou@163.com>

Hi Develop team,

Just simlify this question.

In the message 100, 180,183, 200 response to the caller side from Freeswitch side, the called part Name should be include in the sip address of "To" & SDP part. It should be  "ABC"<sip: 1234 at a.b.c.d>, NOT only <sip: 1234 at a.b.c.d>.

Currently, at the called part, the caller info is correct with Name. So the display at called side is OK.

Could you please let me know where is the source code for such process? I also would like to review it.

Thanks!

2012-11-30



sparklezou



????sparklezou
?????2012-11-30 15:55
???Re: [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???"freeswitch-users"<freeswitch-users at lists.freeswitch.org>

Hi All,

Before, I saved the user name in the phone. So it will show the name when ringing.

From the FreeSwitch side, there is NO difference.

@ Develop Team,

For Internal call, or any sip call from gateway, it should be better show the Name during ring.  And also keep show the name in the call, in stead of "Outbound Call".

Usually the phone will update the display name from the sip response message.

So in the 100, 180, 183, 200 sip/sdp message. Respons with the called name, like "ABC"<sip: 1234 at a.b.c.d>.

Currently in the inital 100,180,183 message, only <sip: 1234 at a.b.c.d>. In the 200 message "Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

If it could be updated, then the called name could be displayed on the caller phone.

Thanks!

2012-11-30



sparklezou



????sparklezou
?????2012-11-30 11:19
???[Freeswitch-users] About the display problem during calling
????"freeswitch-users"<freeswitch-users at lists.freeswitch.org>,"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi All,

I facing a strange problem.

Before, when calling Internal number, the name and number will both be displayed on the caller phone. When connected, only the name will display on the phone. At called phone, display the name and number. Seems fine.

But yesterday afternoon, when calling Internal number, only the number is displayed on the caller phone. When connected, display "Outbound Call" on the caller phone. At the called phone, everthing is the same. Display both the name and number.

I think, mabybe something configured wrong. So I try to restore back the configuration. seems the same. I have to re-install the last release stable 1.2.5.1 version on another server. total fresh setting. But found the same problem.

I checked the sip info.

The "Outbound Call" is in the 200 SDP message. the called is set to 

""Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

Not the correct called name.

Is there anything wrong?

Please help me.

Thanks!


2012-11-30



sparklezou
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From brian at freeswitch.org  Fri Nov 30 23:50:20 2012
From: brian at freeswitch.org (Brian West)
Date: Fri, 30 Nov 2012 14:50:20 -0600
Subject: [Freeswitch-dev] [Freeswitch-users] About the display problem
	during calling
In-Reply-To: <35a86017.4737.13b51737748.Coremail.sparklezou@163.com>
References: <5c994372.3147.13b4f55d827.Coremail.sparklezou@163.com><79e311e1.6599.13b504fd0fa.Coremail.sparklezou@163.com>
	<35a86017.4737.13b51737748.Coremail.sparklezou@163.com>
Message-ID: <66CC55C6-9532-4528-888D-AD9E0A9AA218@freeswitch.org>

How are you placing this call?  Give me dialplan.
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 30, 2012, at 7:11 AM, sparklezou <sparklezou at 163.com> wrote:

> Hi Develop team,
>  
> Just simlify this question.
>  
> In the message 100, 180,183, 200 response to the caller side from Freeswitch side, the called part Name should be include in the sip address of "To" & SDP part. It should be  "ABC"<sip: 1234 at a.b.c.d>, NOT only <sip: 1234 at a.b.c.d>.
>  
> Currently, at the called part, the caller info is correct with Name. So the display at called side is OK.
>  
> Could you please let me know where is the source code for such process? I also would like to review it.
>  
> Thanks!