From anthony.minessale at gmail.com  Mon Nov  1 09:19:01 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Nov 2010 11:19:01 -0500
Subject: [Freeswitch-dev] playback in background
In-Reply-To: <AANLkTinzcbE0zzt7386F22DOSqjqiAVS4asVVnOUBJsm@mail.gmail.com>
References: <AANLkTikYX_vdnnTZyViBVHavrY1Qd3fzkw2DjzvL=KLc@mail.gmail.com>
	<AANLkTim=OF3-Y5r0AWTnfuifObdxoR89M3n-jO-EBP50@mail.gmail.com>
	<AANLkTinzcbE0zzt7386F22DOSqjqiAVS4asVVnOUBJsm@mail.gmail.com>
Message-ID: <AANLkTikck5nPP7c1hQ4RcLkZDvJYVapGjgohh2WA7X2g@mail.gmail.com>

it doesn't work with sleep because sleep does not write any packets.

send_silence_when_idle=400


On Sun, Oct 31, 2010 at 1:23 AM, Seven Du <dujinfang at gmail.com> wrote:
> Cool. Thank you Anthony. ?Sorry I didn't look much hard.
>
> And, I found a few problems:
>
> 1) instead of displace, it's called displace_session.
>
> 2) the calling party hear nothing before echo, with or without the m
> flag. I want to hear MOH?while waiting in the sleep.
>
> <condition>
> ? ? ? ?<action application="answer"/>
> ? ? ? ?<action application="playback" data="tone_stream://%(100,1000,800);loops=1"/>
> ? ? ? ?<action application="displace_session" data="local_stream://moh m"/>
> ? ? ? ?<action application="sleep" data="5000"/>
> ? ? ? ?<action application="echo"/>
> </condition>
>
> 3) inconsistent syntax: ?limit mux vs flags limit, with or without +
>
> uuid_displace,<uuid> [start|stop] <path> [<limit>] [mux],session
> displace,mod_commands
> displace_session: <path> [<flags>] [+time_limit_ms]
>
> I understand fix it might beak backward compatibilities, however, as
> they go to the same switch_ivr function it would be better if they
> looks like consistent.
>
> On Sun, Oct 31, 2010 at 11:16 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> Lookup displace and uuid_displace
>>
>> On Oct 30, 2010 8:14 PM, "Seven Du" <dujinfang at gmail.com> wrote:
>>> Hi,
>>>
>>> I didn't find a way to playback in background in normal dialplan/lua
>>> scripts. I know how to use event socket to do async, but I still think
>>> it's useful in the following scenario and don't need a socket.
>>>
>>> - start playback in background
>>> - long time query (http or db query)
>>> - explicitly stop background playback, or implicitly stop when any
>>> further app executes(or when new data writes to the channel)
>>> - do other things using the query result
>>>
>>> Is there any existing solutions? or how hart to make a patch? if I
>>> make a patch to achieve that, would it be better to make in
>>> mod_commands or should I make a new module?
>>>
>>> What I'm thinking is to make a BUG in another thread and feed sound
>>> data to the channel. Is that the right approach?
>>>
>>> Thanks.
>>>
>>> --
>>> Blog: http://www.dujinfang.com
>>> Proj:? http://www.freeswitch.org.cn
>>>
>>> _______________________________________________
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> 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
>>
>>
>
>
>
> --
> Blog: http://www.dujinfang.com
> Proj:? http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From dome at tel.co.th  Mon Nov  1 09:31:23 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Mon, 1 Nov 2010 23:31:23 +0700
Subject: [Freeswitch-dev] CRM Screen Popup
Message-ID: <AANLkTimVDS+YRnoi1fP=eJryPJ2R6Cbi_enVd_zbbuWX@mail.gmail.com>

Dear All,

        I'm looking for software like http://adm.hamnett.org/  for
freeswitch. client running on windows. i need simple feature when call
incoming open browser and go to url + callerid

BG

Dome C.


From steveayre at gmail.com  Mon Nov  1 09:40:15 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 1 Nov 2010 16:40:15 +0000
Subject: [Freeswitch-dev] CRM Screen Popup
In-Reply-To: <AANLkTimVDS+YRnoi1fP=eJryPJ2R6Cbi_enVd_zbbuWX@mail.gmail.com>
References: <AANLkTimVDS+YRnoi1fP=eJryPJ2R6Cbi_enVd_zbbuWX@mail.gmail.com>
Message-ID: <AANLkTimeWVs6gOqKEvo3xDhNPrnKgtvR5mUGmN7H0gjC@mail.gmail.com>

If you're willing to write it yourself, the ESL interface would allow
you to do it.

Regards,
-Steve


On 1 November 2010 16:31, Dome Charoenyost <dome at tel.co.th> wrote:
> Dear All,
>
> ? ? ? ?I'm looking for software like http://adm.hamnett.org/ ?for
> freeswitch. client running on windows. i need simple feature when call
> incoming open browser and go to url + callerid
>
> BG
>
> Dome C.
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>


From dome at tel.co.th  Mon Nov  1 09:48:33 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Mon, 1 Nov 2010 23:48:33 +0700
Subject: [Freeswitch-dev] CRM Screen Popup
In-Reply-To: <AANLkTimeWVs6gOqKEvo3xDhNPrnKgtvR5mUGmN7H0gjC@mail.gmail.com>
References: <AANLkTimVDS+YRnoi1fP=eJryPJ2R6Cbi_enVd_zbbuWX@mail.gmail.com>
	<AANLkTimeWVs6gOqKEvo3xDhNPrnKgtvR5mUGmN7H0gjC@mail.gmail.com>
Message-ID: <AANLkTimT5w=C8wp=E+qUHRBgyq2LWyVcjBqRQSNm__cM@mail.gmail.com>

2010/11/1 Steven Ayre <steveayre at gmail.com>:
> If you're willing to write it yourself, the ESL interface would allow
> you to do it.
Installing windows 7 :)  I'll use C# .net start with simple feature first.

Dome C.

>
> Regards,
> -Steve
>
>
> On 1 November 2010 16:31, Dome Charoenyost <dome at tel.co.th> wrote:
>> Dear All,
>>
>> ? ? ? ?I'm looking for software like http://adm.hamnett.org/ ?for
>> freeswitch. client running on windows. i need simple feature when call
>> incoming open browser and go to url + callerid
>>
>> BG
>>
>> Dome C.
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> 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 william.suffill at gmail.com  Mon Nov  1 10:02:41 2010
From: william.suffill at gmail.com (William Suffill)
Date: Mon, 1 Nov 2010 13:02:41 -0400
Subject: [Freeswitch-dev] CRM Screen Popup
In-Reply-To: <AANLkTimT5w=C8wp=E+qUHRBgyq2LWyVcjBqRQSNm__cM@mail.gmail.com>
References: <AANLkTimVDS+YRnoi1fP=eJryPJ2R6Cbi_enVd_zbbuWX@mail.gmail.com>
	<AANLkTimeWVs6gOqKEvo3xDhNPrnKgtvR5mUGmN7H0gjC@mail.gmail.com>
	<AANLkTimT5w=C8wp=E+qUHRBgyq2LWyVcjBqRQSNm__cM@mail.gmail.com>
Message-ID: <AANLkTi=LCcxviFn_inki3zNVhE8B2cS2pK9MSKs42GRN@mail.gmail.com>

Last time I needed to do this we kept it browser based vs custom
software needed to be installed on the individual clients. If I had it
to do it over again I'd lean in this direction as well unless I had
someone who can focus on a client based service for the pcs.

Look forward to see how this concept progresses in this community.

-- W


From dujinfang at gmail.com  Mon Nov  1 15:42:16 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 2 Nov 2010 06:42:16 +0800
Subject: [Freeswitch-dev] playback in background
In-Reply-To: <AANLkTikck5nPP7c1hQ4RcLkZDvJYVapGjgohh2WA7X2g@mail.gmail.com>
References: <AANLkTikYX_vdnnTZyViBVHavrY1Qd3fzkw2DjzvL=KLc@mail.gmail.com>
	<AANLkTim=OF3-Y5r0AWTnfuifObdxoR89M3n-jO-EBP50@mail.gmail.com>
	<AANLkTinzcbE0zzt7386F22DOSqjqiAVS4asVVnOUBJsm@mail.gmail.com>
	<AANLkTikck5nPP7c1hQ4RcLkZDvJYVapGjgohh2WA7X2g@mail.gmail.com>
Message-ID: <AANLkTi=03Lj65XermQfO=fUeNws1OUfT-d8+COh8e=ek@mail.gmail.com>

That works, thanks a lot.

On Tue, Nov 2, 2010 at 12:19 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> it doesn't work with sleep because sleep does not write any packets.
>
> send_silence_when_idle=400
>
>
> On Sun, Oct 31, 2010 at 1:23 AM, Seven Du <dujinfang at gmail.com> wrote:
>> Cool. Thank you Anthony. ?Sorry I didn't look much hard.
>>
>> And, I found a few problems:
>>
>> 1) instead of displace, it's called displace_session.
>>
>> 2) the calling party hear nothing before echo, with or without the m
>> flag. I want to hear MOH?while waiting in the sleep.
>>
>> <condition>
>> ? ? ? ?<action application="answer"/>
>> ? ? ? ?<action application="playback" data="tone_stream://%(100,1000,800);loops=1"/>
>> ? ? ? ?<action application="displace_session" data="local_stream://moh m"/>
>> ? ? ? ?<action application="sleep" data="5000"/>
>> ? ? ? ?<action application="echo"/>
>> </condition>
>>
>> 3) inconsistent syntax: ?limit mux vs flags limit, with or without +
>>
>> uuid_displace,<uuid> [start|stop] <path> [<limit>] [mux],session
>> displace,mod_commands
>> displace_session: <path> [<flags>] [+time_limit_ms]
>>
>> I understand fix it might beak backward compatibilities, however, as
>> they go to the same switch_ivr function it would be better if they
>> looks like consistent.
>>
>> On Sun, Oct 31, 2010 at 11:16 AM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>> Lookup displace and uuid_displace
>>>
>>> On Oct 30, 2010 8:14 PM, "Seven Du" <dujinfang at gmail.com> wrote:
>>>> Hi,
>>>>
>>>> I didn't find a way to playback in background in normal dialplan/lua
>>>> scripts. I know how to use event socket to do async, but I still think
>>>> it's useful in the following scenario and don't need a socket.
>>>>
>>>> - start playback in background
>>>> - long time query (http or db query)
>>>> - explicitly stop background playback, or implicitly stop when any
>>>> further app executes(or when new data writes to the channel)
>>>> - do other things using the query result
>>>>
>>>> Is there any existing solutions? or how hart to make a patch? if I
>>>> make a patch to achieve that, would it be better to make in
>>>> mod_commands or should I make a new module?
>>>>
>>>> What I'm thinking is to make a BUG in another thread and feed sound
>>>> data to the channel. Is that the right approach?
>>>>
>>>> Thanks.
>>>>
>>>> --
>>>> Blog: http://www.dujinfang.com
>>>> Proj:? http://www.freeswitch.org.cn
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-dev mailing list
>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>>
>>
>> --
>> Blog: http://www.dujinfang.com
>> Proj:? http://www.freeswitch.org.cn
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> 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
>



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


From bernhard.suttner at winet.ch  Tue Nov  2 06:36:47 2010
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Tue, 2 Nov 2010 14:36:47 +0100
Subject: [Freeswitch-dev] mod_loopback no ringback tone
In-Reply-To: <AANLkTimgfszvmyfmvpt0OA8ayew5NSiONhcbjJCSKd6b@mail.gmail.com>
References: <20101025204118.4e281b00@mail.winet.ch>
	<AANLkTinQ3tZBiLJ2OsvtKbJU=ZCP562Fb8qPsXjVUC6u@mail.gmail.com>
	<6e4039f4-6577-4191-9215-3ea0ef3d7bc5@winet.ch>
	<AANLkTi=ghxTv3bKu9ZpEpZPJAW5GQbzx7aLj7P5xU2RD@mail.gmail.com>
	<3451ded7-c923-4cb5-b200-7c24e886e5fe@winet.ch>
	<AANLkTimibSveBYi2chhKK78LgvpChibPbvYdK5mM4-Lu@mail.gmail.com>
	<c42268c8-a069-4fd9-9974-d97994cd64af@winet.ch>
	<da1fa293-29df-4c45-8a80-cd802e45872e@winet.ch>
	<AANLkTimgfszvmyfmvpt0OA8ayew5NSiONhcbjJCSKd6b@mail.gmail.com>
Message-ID: <3bfa8bcf-80e1-4059-9ccf-f474ccc20279@winet.ch>

Hi,

I have not set ignore_early_media. I have uploaded to pastebin 14381 the call-setup.

Best regards,
Bernhard Suttner

-----Urspr?ngliche Nachricht-----
Von: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] Im Auftrag von Anthony Minessale
Gesendet: Freitag, 29. Oktober 2010 17:00
An: freeswitch-dev at lists.freeswitch.org
Betreff: Re: [Freeswitch-dev] mod_loopback no ringback tone

A is already answered or you have ignore_early_media=true somewhere.

type these CLI commands:

console loglevel debug
sofia global siptrace on

make the call and the debug will spell it out, if you can't find it
post the log on pastebin.


On Fri, Oct 29, 2010 at 4:59 AM, Bernhard Suttner
<bernhard.suttner at winet.ch> wrote:
> Hi,
>
> I get the ringback correctly, but from the signaling point of view, I get no register:
>
> A ----(SIP) ---> FS ---(loopback)---> FS ----(SIP)---> B
>
> B does send the ringing to FS. Loopback will set the flag to send the ringback to A but A does not get a RINGING.
>
> Is there maybe something other to set to get that working?
>
> Best regards,
> Bernhard
>
> -----Urspr?ngliche Nachricht-----
> Von: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] Im Auftrag von Bernhard Suttner
> Gesendet: Dienstag, 26. Oktober 2010 19:07
> An: freeswitch-dev at lists.freeswitch.org
> Betreff: Re: [Freeswitch-dev] mod_loopback no ringback tone
>
> Fixed. Problem was within mod_loopback but anthm did fix it. Thanks a lot again!
>
> -----Urspr?ngliche Nachricht-----
> Von: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] Im Auftrag von Anthony Minessale
> Gesendet: Dienstag, 26. Oktober 2010 18:03
> An: freeswitch-dev at lists.freeswitch.org
> Betreff: Re: [Freeswitch-dev] mod_loopback no ringback tone
>
> you still did not update to latest git then
>
> On Tue, Oct 26, 2010 at 10:48 AM, Bernhard Suttner
> <bernhard.suttner at winet.ch> wrote:
>> Hi,
>>
>> I have not activated inbound-bypass-media and I tried with ignore-eary-media set to true and to false. Both times, I have no ringback. I added some debug stuff within mod_loopback.c and saw, that the code after switch_core_session_get_partner will not be called for msg->message_id = 7.
>>
>> Best regards,
>> Bernhard
>>
>> -----Urspr?ngliche Nachricht-----
>> Von: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] Im Auftrag von Anthony Minessale
>> Gesendet: Dienstag, 26. Oktober 2010 16:22
>> An: freeswitch-dev at lists.freeswitch.org
>> Betreff: Re: [Freeswitch-dev] mod_loopback no ringback tone
>>
>> do you have inbound-bypass-media on?
>> or are you not setting {ignore_early_media=true} when you call loopback
>>
>> The default config ext 9181 demonstrates this functionality and is
>> confirmed to work.
>>
>>
>>
>> On Tue, Oct 26, 2010 at 3:09 AM, Bernhard Suttner
>> <bernhard.suttner at winet.ch> wrote:
>>> Hi,
>>>
>>> it does not work. I think the switch_core_get_partner function does not return the required session - so, the BOND_VARIABLE for sofia-/loopback-a is not set.
>>>
>>> Best regards,
>>> Bernhard
>>>
>>> -----Urspr?ngliche Nachricht-----
>>> Von: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] Im Auftrag von Anthony Minessale
>>> Gesendet: Montag, 25. Oktober 2010 21:22
>>> An: freeswitch-dev at lists.freeswitch.org
>>> Betreff: Re: [Freeswitch-dev] mod_loopback no ringback tone
>>>
>>> We try. We're not always on top of things as much as we'd like but we
>>> sure try =p
>>>
>>>
>>> On Mon, Oct 25, 2010 at 1:41 PM, Bernhard Suttner
>>> <bernhard.suttner at winet.ch> wrote:
>>>> Hi,
>>>>
>>>> thanks for your fast answer. I tried the version about 4 hours ago. I will re-check it tomorrow.
>>>>
>>>> Thanks a lot for your VERY FAST RESPONSES. Within the asterisk project it does need more than X days to get a response if the whole asterisk does crash :-)
>>>>
>>>> Best regards,
>>>> Bernhard
>>>>
>>>>
>>>> ----- Original Message -----
>>>> From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
>>>> To: freeswitch-dev at lists.freeswitch.org
>>>> Sent: Mon, 25 Oct 2010 19:00:57 +0200
>>>> Subject: Re: [Freeswitch-dev] mod_loopback no ringback tone
>>>>
>>>>
>>>>> try again on latest HEAD that may have been broken over the weekend.
>>>>>
>>>>>
>>>>> On Mon, Oct 25, 2010 at 11:11 AM, Bernhard Suttner
>>>>> <bernhard.suttner at winet.ch> wrote:
>>>>> > Hi,
>>>>> >
>>>>> > I have the following scenario
>>>>> >
>>>>> > A ----(SIP) ---> FS ---(loopback)---> FS ----(SIP)---> B
>>>>> >
>>>>> > FS is of course the same Freeswitch Sever. If B sends RINGING to FS the
>>>>> RINGING will not be passed through to A because of the loopback module.
>>>>> (That is was the trace does say). What has to happen on mod_loopback if it
>>>>> does receive the RINGING? Does it have to do something like
>>>>> switch_channel_mark_ring_ready, switch_channel_ring_ready or something else?
>>>>> I think it has maybe something to do with the missing INDICATE_RINGING
>>>>> section in mod_loopback.
>>>>> >
>>>>> > Best regards,
>>>>> > Bernhard Suttner
>>>>> >
>>>>> > _______________________________________________
>>>>> > FreeSWITCH-dev mailing list
>>>>> > FreeSWITCH-dev at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>>> > http://www.freeswitch.org
>>>>> >
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org
>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-dev mailing list
>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-dev mailing list
>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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

_______________________________________________
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 msc at freeswitch.org  Wed Nov  3 08:15:09 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Nov 2010 08:15:09 -0700
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today - DRK Discusses
	.NET Fun
Message-ID: <AANLkTi=SS_BP-C5R+g6-kdypunZDzJM7K9gaEtCc9tkt@mail.gmail.com>

Hello all,

The FreeSWITCH conference call agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_11_03

Dave Kompel will be showing off some more .NET stuff that he's got cooking
plus we have a few other things to discuss.

Talk to you soon,
Michael
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From bernhard.suttner at winet.ch  Thu Nov  4 08:25:10 2010
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Thu, 4 Nov 2010 16:25:10 +0100
Subject: [Freeswitch-dev] Session Progress and RINGING
Message-ID: <0949bdad-547b-4734-8231-b7a1d4bd9915@winet.ch>

Hi,

A ---> FS ---> B

B does send a Session Progress to FS. FS will forward the Session Progress to A.
B does send a RINGING to FS. FS does _not_ forward this to A.

Could it be, that the check the switch_test_flag(channel, CF_EARLY_MEDIA) in mod_sofia.c within the INDICATE_RINGING section is wrong. I think it is allowed to send the Session Progress first and later the RINGING. The ringing is for example for SIP/ISDN gateways necessary.

Any hint is appreciated.

Best regards,
Bernhard Suttner




From anthony.minessale at gmail.com  Thu Nov  4 08:54:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Nov 2010 10:54:30 -0500
Subject: [Freeswitch-dev] Session Progress and RINGING
In-Reply-To: <0949bdad-547b-4734-8231-b7a1d4bd9915@winet.ch>
References: <0949bdad-547b-4734-8231-b7a1d4bd9915@winet.ch>
Message-ID: <AANLkTin6S5VQ-x_s1VR2FXB7L_5vZnhu+3880m=ncKa8@mail.gmail.com>

no, once it gets session progress it will not send any ringing.
The sip side 180 ringing is used to tell the phone to generate its own
inband ringing.

does your 183 session progress contain a sdp?


On Thu, Nov 4, 2010 at 10:25 AM, Bernhard Suttner
<bernhard.suttner at winet.ch> wrote:
> Hi,
>
> A ---> FS ---> B
>
> B does send a Session Progress to FS. FS will forward the Session Progress to A.
> B does send a RINGING to FS. FS does _not_ forward this to A.
>
> Could it be, that the check the switch_test_flag(channel, CF_EARLY_MEDIA) in mod_sofia.c within the INDICATE_RINGING section is wrong. I think it is allowed to send the Session Progress first and later the RINGING. The ringing is for example for SIP/ISDN gateways necessary.
>
> Any hint is appreciated.
>
> Best regards,
> Bernhard Suttner
>
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From bernhard.suttner at winet.ch  Thu Nov  4 13:51:09 2010
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Thu, 04 Nov 2010 21:51:09 +0100
Subject: [Freeswitch-dev] Session Progress and RINGING
Message-ID: <20101104215109.e83f1ba5@mail.winet.ch>

Hi,

session progress does have SDP which will then go through to A. But on the display of A it will only display "session progress" and not Ringing. If A is now a ISDN Gateway and has to signalize RINGING into the "ISDN world". I am not sure, if RINGING will be used for A to generate the ringback tone for itself on every device. This behaves on the client/device configuration. 

There was a nice discussion about this "issue" on the sip-implementors list:

http://www.mail-archive.com/sip-implementors at lists.cs.columbia.edu/msg06400.html

There is a RFC for this case (Section: 3.2.  Ringing Tone Generation)
http://www.rfc-editor.org/rfc/rfc3960.txt

What do you think?

Best regards,
Bernhard Suttner



----- Original Message -----
From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
To: freeswitch-dev at lists.freeswitch.org
Sent: Thu, 04 Nov 2010 16:54:30 +0100
Subject: Re: [Freeswitch-dev] Session Progress and RINGING


> no, once it gets session progress it will not send any ringing.
> The sip side 180 ringing is used to tell the phone to generate its own
> inband ringing.
> 
> does your 183 session progress contain a sdp?
> 
> 
> On Thu, Nov 4, 2010 at 10:25 AM, Bernhard Suttner
> <bernhard.suttner at winet.ch> wrote:
> > Hi,
> >
> > A ---> FS ---> B
> >
> > B does send a Session Progress to FS. FS will forward the Session Progress
> to A.
> > B does send a RINGING to FS. FS does _not_ forward this to A.
> >
> > Could it be, that the check the switch_test_flag(channel, CF_EARLY_MEDIA)
> in mod_sofia.c within the INDICATE_RINGING section is wrong. I think it is
> allowed to send the Session Progress first and later the RINGING. The
> ringing is for example for SIP/ISDN gateways necessary.
> >
> > Any hint is appreciated.
> >
> > Best regards,
> > Bernhard Suttner
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
> >
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
> 


From anthony.minessale at gmail.com  Thu Nov  4 14:09:48 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Nov 2010 16:09:48 -0500
Subject: [Freeswitch-dev] Session Progress and RINGING
In-Reply-To: <20101104215109.e83f1ba5@mail.winet.ch>
References: <20101104215109.e83f1ba5@mail.winet.ch>
Message-ID: <AANLkTik4U6MJFC6TOYFh6wvsVh41xYU9qSkwVmiSwacV@mail.gmail.com>

My conclusion from reading that discussion is they have no idea what to do.
There are votes both ways and 2 attempts to go of onto a tangent into
another topic.

We have chosen on our implementation to ignore 180 once we have
already established a media path from a 183.  The RFC is just vague
enough that this decision falls to us.

I understand that ISDN has more precise signaling in this regard.
PROGRESS or ALERTING both with or without media as a flag on the packet.

But the vast majority of SIP devices in the wild will ignore the 180
once early media has been established.
So even if we make changes to support this, it will be ignored by the
next guy in line.

Do you hear media during this early media phase?

This is why we have the originate param
{ignore_early_media=ring_ready} which will, in the case of sip,
translate 183 or 180 with and without SDP into a 180

To change it to do what you are asking for could have extreme negative
side effects and not even work in most devices so this is why I am
reluctant to change it.

Are you trying to cross connect 2 ISDN lines over SIP and preserve the
signalling?
If so, this is why SIP is flawed to begin with because it is lossy in
telephony signaling data.
This is why they now try to embed ss7 messages in the sip packets. =p



On Thu, Nov 4, 2010 at 3:51 PM, Bernhard Suttner
<bernhard.suttner at winet.ch> wrote:
> Hi,
>
> session progress does have SDP which will then go through to A. But on the display of A it will only display "session progress" and not Ringing. If A is now a ISDN Gateway and has to signalize RINGING into the "ISDN world". I am not sure, if RINGING will be used for A to generate the ringback tone for itself on every device. This behaves on the client/device configuration.
>
> There was a nice discussion about this "issue" on the sip-implementors list:
>
> http://www.mail-archive.com/sip-implementors at lists.cs.columbia.edu/msg06400.html
>
> There is a RFC for this case (Section: 3.2. ?Ringing Tone Generation)
> http://www.rfc-editor.org/rfc/rfc3960.txt
>
> What do you think?
>
> Best regards,
> Bernhard Suttner
>
>
>
> ----- Original Message -----
> From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
> To: freeswitch-dev at lists.freeswitch.org
> Sent: Thu, 04 Nov 2010 16:54:30 +0100
> Subject: Re: [Freeswitch-dev] Session Progress and RINGING
>
>
>> no, once it gets session progress it will not send any ringing.
>> The sip side 180 ringing is used to tell the phone to generate its own
>> inband ringing.
>>
>> does your 183 session progress contain a sdp?
>>
>>
>> On Thu, Nov 4, 2010 at 10:25 AM, Bernhard Suttner
>> <bernhard.suttner at winet.ch> wrote:
>> > Hi,
>> >
>> > A ---> FS ---> B
>> >
>> > B does send a Session Progress to FS. FS will forward the Session Progress
>> to A.
>> > B does send a RINGING to FS. FS does _not_ forward this to A.
>> >
>> > Could it be, that the check the switch_test_flag(channel, CF_EARLY_MEDIA)
>> in mod_sofia.c within the INDICATE_RINGING section is wrong. I think it is
>> allowed to send the Session Progress first and later the RINGING. The
>> ringing is for example for SIP/ISDN gateways necessary.
>> >
>> > Any hint is appreciated.
>> >
>> > Best regards,
>> > Bernhard Suttner
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-dev mailing list
>> > FreeSWITCH-dev at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From bernhard.suttner at winet.ch  Thu Nov  4 14:29:42 2010
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Thu, 04 Nov 2010 22:29:42 +0100
Subject: [Freeswitch-dev] Session Progress and RINGING
Message-ID: <20101104222942.c241368d@mail.winet.ch>

Hi,

yes, they are not really sure but I think that the given RFC does specify this correctly (this does not mean, that all the SIP devices work like that). I could understand that a change will maybe result in big troubles. Perhaps a option for mod_sofia to "support-rfc3960" would be a good solution. If you want I will write the patch because it should be "really simple" to ignore the CF_EARLY_MEDIA if the option is set, or?

We get the Session Progress with SDP and later the 180 Ringing from a session border controller (I am not 100% sure, but I think its a a Audiocodes).  The 180 Ringing has to be sent towards A because A is a ISDN gateway.

Would do you prefer? Thanks for your investigation.

Best regards,
Bernhard Suttner

----- Original Message -----
From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
To: freeswitch-dev at lists.freeswitch.org
Sent: Thu, 04 Nov 2010 22:09:48 +0100
Subject: Re: [Freeswitch-dev] Session Progress and RINGING


> My conclusion from reading that discussion is they have no idea what to do.
> There are votes both ways and 2 attempts to go of onto a tangent into
> another topic.
> 
> We have chosen on our implementation to ignore 180 once we have
> already established a media path from a 183.  The RFC is just vague
> enough that this decision falls to us.
> 
> I understand that ISDN has more precise signaling in this regard.
> PROGRESS or ALERTING both with or without media as a flag on the packet.
> 
> But the vast majority of SIP devices in the wild will ignore the 180
> once early media has been established.
> So even if we make changes to support this, it will be ignored by the
> next guy in line.
> 
> Do you hear media during this early media phase?
> 
> This is why we have the originate param
> {ignore_early_media=ring_ready} which will, in the case of sip,
> translate 183 or 180 with and without SDP into a 180
> 
> To change it to do what you are asking for could have extreme negative
> side effects and not even work in most devices so this is why I am
> reluctant to change it.
> 
> Are you trying to cross connect 2 ISDN lines over SIP and preserve the
> signalling?
> If so, this is why SIP is flawed to begin with because it is lossy in
> telephony signaling data.
> This is why they now try to embed ss7 messages in the sip packets. =p
> 
> 
> 
> On Thu, Nov 4, 2010 at 3:51 PM, Bernhard Suttner
> <bernhard.suttner at winet.ch> wrote:
> > Hi,
> >
> > session progress does have SDP which will then go through to A. But on the
> display of A it will only display "session progress" and not Ringing. If A
> is now a ISDN Gateway and has to signalize RINGING into the "ISDN world". I
> am not sure, if RINGING will be used for A to generate the ringback tone for
> itself on every device. This behaves on the client/device configuration.
> >
> > There was a nice discussion about this "issue" on the sip-implementors
> list:
> >
> >
> http://www.mail-archive.com/sip-implementors at lists.cs.columbia.edu/msg06400.html
> >
> > There is a RFC for this case (Section: 3.2. ?Ringing Tone Generation)
> > http://www.rfc-editor.org/rfc/rfc3960.txt
> >
> > What do you think?
> >
> > Best regards,
> > Bernhard Suttner
> >
> >
> >
> > ----- Original Message -----
> > From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
> > To: freeswitch-dev at lists.freeswitch.org
> > Sent: Thu, 04 Nov 2010 16:54:30 +0100
> > Subject: Re: [Freeswitch-dev] Session Progress and RINGING
> >
> >
> >> no, once it gets session progress it will not send any ringing.
> >> The sip side 180 ringing is used to tell the phone to generate its own
> >> inband ringing.
> >>
> >> does your 183 session progress contain a sdp?
> >>
> >>
> >> On Thu, Nov 4, 2010 at 10:25 AM, Bernhard Suttner
> >> <bernhard.suttner at winet.ch> wrote:
> >> > Hi,
> >> >
> >> > A ---> FS ---> B
> >> >
> >> > B does send a Session Progress to FS. FS will forward the Session
> Progress
> >> to A.
> >> > B does send a RINGING to FS. FS does _not_ forward this to A.
> >> >
> >> > Could it be, that the check the switch_test_flag(channel,
> CF_EARLY_MEDIA)
> >> in mod_sofia.c within the INDICATE_RINGING section is wrong. I think it
> is
> >> allowed to send the Session Progress first and later the RINGING. The
> >> ringing is for example for SIP/ISDN gateways necessary.
> >> >
> >> > Any hint is appreciated.
> >> >
> >> > Best regards,
> >> > Bernhard Suttner
> >> >
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-dev mailing list
> >> > FreeSWITCH-dev at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> >> > http://www.freeswitch.org
> >> >
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-dev mailing list
> >> FreeSWITCH-dev at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> >> http://www.freeswitch.org
> >>
> >
> > _______________________________________________
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
> >
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
> 


From anthony.minessale at gmail.com  Thu Nov  4 14:54:34 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Nov 2010 16:54:34 -0500
Subject: [Freeswitch-dev] Session Progress and RINGING
In-Reply-To: <20101104222942.c241368d@mail.winet.ch>
References: <20101104222942.c241368d@mail.winet.ch>
Message-ID: <AANLkTinyvnFmqtDq5ke3zv-3auUJr_G_ZHao7t-y9ccd@mail.gmail.com>

how about what I said:
{ignore_early_media=ring_ready}



On Thu, Nov 4, 2010 at 4:29 PM, Bernhard Suttner
<bernhard.suttner at winet.ch> wrote:
> Hi,
>
> yes, they are not really sure but I think that the given RFC does specify this correctly (this does not mean, that all the SIP devices work like that). I could understand that a change will maybe result in big troubles. Perhaps a option for mod_sofia to "support-rfc3960" would be a good solution. If you want I will write the patch because it should be "really simple" to ignore the CF_EARLY_MEDIA if the option is set, or?
>
> We get the Session Progress with SDP and later the 180 Ringing from a session border controller (I am not 100% sure, but I think its a a Audiocodes). ?The 180 Ringing has to be sent towards A because A is a ISDN gateway.
>
> Would do you prefer? Thanks for your investigation.
>
> Best regards,
> Bernhard Suttner
>
> ----- Original Message -----
> From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
> To: freeswitch-dev at lists.freeswitch.org
> Sent: Thu, 04 Nov 2010 22:09:48 +0100
> Subject: Re: [Freeswitch-dev] Session Progress and RINGING
>
>
>> My conclusion from reading that discussion is they have no idea what to do.
>> There are votes both ways and 2 attempts to go of onto a tangent into
>> another topic.
>>
>> We have chosen on our implementation to ignore 180 once we have
>> already established a media path from a 183. ?The RFC is just vague
>> enough that this decision falls to us.
>>
>> I understand that ISDN has more precise signaling in this regard.
>> PROGRESS or ALERTING both with or without media as a flag on the packet.
>>
>> But the vast majority of SIP devices in the wild will ignore the 180
>> once early media has been established.
>> So even if we make changes to support this, it will be ignored by the
>> next guy in line.
>>
>> Do you hear media during this early media phase?
>>
>> This is why we have the originate param
>> {ignore_early_media=ring_ready} which will, in the case of sip,
>> translate 183 or 180 with and without SDP into a 180
>>
>> To change it to do what you are asking for could have extreme negative
>> side effects and not even work in most devices so this is why I am
>> reluctant to change it.
>>
>> Are you trying to cross connect 2 ISDN lines over SIP and preserve the
>> signalling?
>> If so, this is why SIP is flawed to begin with because it is lossy in
>> telephony signaling data.
>> This is why they now try to embed ss7 messages in the sip packets. =p
>>
>>
>>
>> On Thu, Nov 4, 2010 at 3:51 PM, Bernhard Suttner
>> <bernhard.suttner at winet.ch> wrote:
>> > Hi,
>> >
>> > session progress does have SDP which will then go through to A. But on the
>> display of A it will only display "session progress" and not Ringing. If A
>> is now a ISDN Gateway and has to signalize RINGING into the "ISDN world". I
>> am not sure, if RINGING will be used for A to generate the ringback tone for
>> itself on every device. This behaves on the client/device configuration.
>> >
>> > There was a nice discussion about this "issue" on the sip-implementors
>> list:
>> >
>> >
>> http://www.mail-archive.com/sip-implementors at lists.cs.columbia.edu/msg06400.html
>> >
>> > There is a RFC for this case (Section: 3.2. ?Ringing Tone Generation)
>> > http://www.rfc-editor.org/rfc/rfc3960.txt
>> >
>> > What do you think?
>> >
>> > Best regards,
>> > Bernhard Suttner
>> >
>> >
>> >
>> > ----- Original Message -----
>> > From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
>> > To: freeswitch-dev at lists.freeswitch.org
>> > Sent: Thu, 04 Nov 2010 16:54:30 +0100
>> > Subject: Re: [Freeswitch-dev] Session Progress and RINGING
>> >
>> >
>> >> no, once it gets session progress it will not send any ringing.
>> >> The sip side 180 ringing is used to tell the phone to generate its own
>> >> inband ringing.
>> >>
>> >> does your 183 session progress contain a sdp?
>> >>
>> >>
>> >> On Thu, Nov 4, 2010 at 10:25 AM, Bernhard Suttner
>> >> <bernhard.suttner at winet.ch> wrote:
>> >> > Hi,
>> >> >
>> >> > A ---> FS ---> B
>> >> >
>> >> > B does send a Session Progress to FS. FS will forward the Session
>> Progress
>> >> to A.
>> >> > B does send a RINGING to FS. FS does _not_ forward this to A.
>> >> >
>> >> > Could it be, that the check the switch_test_flag(channel,
>> CF_EARLY_MEDIA)
>> >> in mod_sofia.c within the INDICATE_RINGING section is wrong. I think it
>> is
>> >> allowed to send the Session Progress first and later the RINGING. The
>> >> ringing is for example for SIP/ISDN gateways necessary.
>> >> >
>> >> > Any hint is appreciated.
>> >> >
>> >> > Best regards,
>> >> > Bernhard Suttner
>> >> >
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-dev mailing list
>> >> > FreeSWITCH-dev at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> >> > http://www.freeswitch.org
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org
>> >> googletalk:conf+888 at conference.freeswitch.org
>> >> pstn:+19193869900
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-dev mailing list
>> >> FreeSWITCH-dev at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> >> http://www.freeswitch.org
>> >>
>> >
>> > _______________________________________________
>> > FreeSWITCH-dev mailing list
>> > FreeSWITCH-dev at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From bernhard.suttner at winet.ch  Thu Nov  4 15:05:38 2010
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Thu, 04 Nov 2010 23:05:38 +0100
Subject: [Freeswitch-dev] Session Progress and RINGING
Message-ID: <20101104230538.7fc24d16@mail.winet.ch>

We will try that out but its not "100%" the functionality as the sbc does signalize. If I understand you correct, the 183 with SDP will be forwarded to A with 180 with SDP. But this does change the functionality to A (first of all on SIP/ISDN gateways).

Therefore I would prefer the "clean" way with RFC 3960.

Would you include a patch like "support-rfc3960" configuration option to sofia config which will ignore the CF_EARLY_MEDIA test?



----- Original Message -----
From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
To: freeswitch-dev at lists.freeswitch.org
Sent: Thu, 04 Nov 2010 22:54:34 +0100
Subject: Re: [Freeswitch-dev] Session Progress and RINGING


> how about what I said:
> {ignore_early_media=ring_ready}
> 
> 
> 
> On Thu, Nov 4, 2010 at 4:29 PM, Bernhard Suttner
> <bernhard.suttner at winet.ch> wrote:
> > Hi,
> >
> > yes, they are not really sure but I think that the given RFC does specify
> this correctly (this does not mean, that all the SIP devices work like
> that). I could understand that a change will maybe result in big troubles.
> Perhaps a option for mod_sofia to "support-rfc3960" would be a good
> solution. If you want I will write the patch because it should be "really
> simple" to ignore the CF_EARLY_MEDIA if the option is set, or?
> >
> > We get the Session Progress with SDP and later the 180 Ringing from a
> session border controller (I am not 100% sure, but I think its a a
> Audiocodes). ?The 180 Ringing has to be sent towards A because A is a ISDN
> gateway.
> >
> > Would do you prefer? Thanks for your investigation.
> >
> > Best regards,
> > Bernhard Suttner
> >
> > ----- Original Message -----
> > From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
> > To: freeswitch-dev at lists.freeswitch.org
> > Sent: Thu, 04 Nov 2010 22:09:48 +0100
> > Subject: Re: [Freeswitch-dev] Session Progress and RINGING
> >
> >
> >> My conclusion from reading that discussion is they have no idea what to
> do.
> >> There are votes both ways and 2 attempts to go of onto a tangent into
> >> another topic.
> >>
> >> We have chosen on our implementation to ignore 180 once we have
> >> already established a media path from a 183. ?The RFC is just vague
> >> enough that this decision falls to us.
> >>
> >> I understand that ISDN has more precise signaling in this regard.
> >> PROGRESS or ALERTING both with or without media as a flag on the packet.
> >>
> >> But the vast majority of SIP devices in the wild will ignore the 180
> >> once early media has been established.
> >> So even if we make changes to support this, it will be ignored by the
> >> next guy in line.
> >>
> >> Do you hear media during this early media phase?
> >>
> >> This is why we have the originate param
> >> {ignore_early_media=ring_ready} which will, in the case of sip,
> >> translate 183 or 180 with and without SDP into a 180
> >>
> >> To change it to do what you are asking for could have extreme negative
> >> side effects and not even work in most devices so this is why I am
> >> reluctant to change it.
> >>
> >> Are you trying to cross connect 2 ISDN lines over SIP and preserve the
> >> signalling?
> >> If so, this is why SIP is flawed to begin with because it is lossy in
> >> telephony signaling data.
> >> This is why they now try to embed ss7 messages in the sip packets. =p
> >>
> >>
> >>
> >> On Thu, Nov 4, 2010 at 3:51 PM, Bernhard Suttner
> >> <bernhard.suttner at winet.ch> wrote:
> >> > Hi,
> >> >
> >> > session progress does have SDP which will then go through to A. But on
> the
> >> display of A it will only display "session progress" and not Ringing. If
> A
> >> is now a ISDN Gateway and has to signalize RINGING into the "ISDN world".
> I
> >> am not sure, if RINGING will be used for A to generate the ringback tone
> for
> >> itself on every device. This behaves on the client/device configuration.
> >> >
> >> > There was a nice discussion about this "issue" on the sip-implementors
> >> list:
> >> >
> >> >
> >>
> http://www.mail-archive.com/sip-implementors at lists.cs.columbia.edu/msg06400.html
> >> >
> >> > There is a RFC for this case (Section: 3.2. ?Ringing Tone Generation)
> >> > http://www.rfc-editor.org/rfc/rfc3960.txt
> >> >
> >> > What do you think?
> >> >
> >> > Best regards,
> >> > Bernhard Suttner
> >> >
> >> >
> >> >
> >> > ----- Original Message -----
> >> > From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
> >> > To: freeswitch-dev at lists.freeswitch.org
> >> > Sent: Thu, 04 Nov 2010 16:54:30 +0100
> >> > Subject: Re: [Freeswitch-dev] Session Progress and RINGING
> >> >
> >> >
> >> >> no, once it gets session progress it will not send any ringing.
> >> >> The sip side 180 ringing is used to tell the phone to generate its own
> >> >> inband ringing.
> >> >>
> >> >> does your 183 session progress contain a sdp?
> >> >>
> >> >>
> >> >> On Thu, Nov 4, 2010 at 10:25 AM, Bernhard Suttner
> >> >> <bernhard.suttner at winet.ch> wrote:
> >> >> > Hi,
> >> >> >
> >> >> > A ---> FS ---> B
> >> >> >
> >> >> > B does send a Session Progress to FS. FS will forward the Session
> >> Progress
> >> >> to A.
> >> >> > B does send a RINGING to FS. FS does _not_ forward this to A.
> >> >> >
> >> >> > Could it be, that the check the switch_test_flag(channel,
> >> CF_EARLY_MEDIA)
> >> >> in mod_sofia.c within the INDICATE_RINGING section is wrong. I think
> it
> >> is
> >> >> allowed to send the Session Progress first and later the RINGING. The
> >> >> ringing is for example for SIP/ISDN gateways necessary.
> >> >> >
> >> >> > Any hint is appreciated.
> >> >> >
> >> >> > Best regards,
> >> >> > Bernhard Suttner
> >> >> >
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-dev mailing list
> >> >> > FreeSWITCH-dev at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >> >> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >>
> >> >>
> >> >>
> >> >> --
> >> >> Anthony Minessale II
> >> >>
> >> >> FreeSWITCH http://www.freeswitch.org/
> >> >> ClueCon http://www.cluecon.com/
> >> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >> >>
> >> >> AIM: anthm
> >> >> MSN:anthony_minessale at hotmail.com
> >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> >> IRC: irc.freenode.net #freeswitch
> >> >>
> >> >> FreeSWITCH Developer Conference
> >> >> sip:888 at conference.freeswitch.org
> >> >> googletalk:conf+888 at conference.freeswitch.org
> >> >> pstn:+19193869900
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-dev mailing list
> >> >> FreeSWITCH-dev at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> >> >> http://www.freeswitch.org
> >> >>
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-dev mailing list
> >> > FreeSWITCH-dev at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> >> > http://www.freeswitch.org
> >> >
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-dev mailing list
> >> FreeSWITCH-dev at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> >> http://www.freeswitch.org
> >>
> >
> > _______________________________________________
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
> >
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
> 


From anthony.minessale at gmail.com  Thu Nov  4 15:17:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Nov 2010 17:17:38 -0500
Subject: [Freeswitch-dev] Session Progress and RINGING
In-Reply-To: <20101104230538.7fc24d16@mail.winet.ch>
References: <20101104230538.7fc24d16@mail.winet.ch>
Message-ID: <AANLkTinOsit+_BHVxG9GHTu9nLHV93TkH=aZPS9Tzmg0@mail.gmail.com>

try to patch it i guess.
I still say the RFC is to vague to call this behavior complying with the RFC.
if you can make a clean patch that we can wrap in a pram like
support-one-of-many-interpretation-of-rfc3960=true





On Thu, Nov 4, 2010 at 5:05 PM, Bernhard Suttner
<bernhard.suttner at winet.ch> wrote:
> We will try that out but its not "100%" the functionality as the sbc does signalize. If I understand you correct, the 183 with SDP will be forwarded to A with 180 with SDP. But this does change the functionality to A (first of all on SIP/ISDN gateways).
>
> Therefore I would prefer the "clean" way with RFC 3960.
>
> Would you include a patch like "support-rfc3960" configuration option to sofia config which will ignore the CF_EARLY_MEDIA test?
>
>
>
> ----- Original Message -----
> From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
> To: freeswitch-dev at lists.freeswitch.org
> Sent: Thu, 04 Nov 2010 22:54:34 +0100
> Subject: Re: [Freeswitch-dev] Session Progress and RINGING
>
>
>> how about what I said:
>> {ignore_early_media=ring_ready}
>>
>>
>>
>> On Thu, Nov 4, 2010 at 4:29 PM, Bernhard Suttner
>> <bernhard.suttner at winet.ch> wrote:
>> > Hi,
>> >
>> > yes, they are not really sure but I think that the given RFC does specify
>> this correctly (this does not mean, that all the SIP devices work like
>> that). I could understand that a change will maybe result in big troubles.
>> Perhaps a option for mod_sofia to "support-rfc3960" would be a good
>> solution. If you want I will write the patch because it should be "really
>> simple" to ignore the CF_EARLY_MEDIA if the option is set, or?
>> >
>> > We get the Session Progress with SDP and later the 180 Ringing from a
>> session border controller (I am not 100% sure, but I think its a a
>> Audiocodes). ?The 180 Ringing has to be sent towards A because A is a ISDN
>> gateway.
>> >
>> > Would do you prefer? Thanks for your investigation.
>> >
>> > Best regards,
>> > Bernhard Suttner
>> >
>> > ----- Original Message -----
>> > From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
>> > To: freeswitch-dev at lists.freeswitch.org
>> > Sent: Thu, 04 Nov 2010 22:09:48 +0100
>> > Subject: Re: [Freeswitch-dev] Session Progress and RINGING
>> >
>> >
>> >> My conclusion from reading that discussion is they have no idea what to
>> do.
>> >> There are votes both ways and 2 attempts to go of onto a tangent into
>> >> another topic.
>> >>
>> >> We have chosen on our implementation to ignore 180 once we have
>> >> already established a media path from a 183. ?The RFC is just vague
>> >> enough that this decision falls to us.
>> >>
>> >> I understand that ISDN has more precise signaling in this regard.
>> >> PROGRESS or ALERTING both with or without media as a flag on the packet.
>> >>
>> >> But the vast majority of SIP devices in the wild will ignore the 180
>> >> once early media has been established.
>> >> So even if we make changes to support this, it will be ignored by the
>> >> next guy in line.
>> >>
>> >> Do you hear media during this early media phase?
>> >>
>> >> This is why we have the originate param
>> >> {ignore_early_media=ring_ready} which will, in the case of sip,
>> >> translate 183 or 180 with and without SDP into a 180
>> >>
>> >> To change it to do what you are asking for could have extreme negative
>> >> side effects and not even work in most devices so this is why I am
>> >> reluctant to change it.
>> >>
>> >> Are you trying to cross connect 2 ISDN lines over SIP and preserve the
>> >> signalling?
>> >> If so, this is why SIP is flawed to begin with because it is lossy in
>> >> telephony signaling data.
>> >> This is why they now try to embed ss7 messages in the sip packets. =p
>> >>
>> >>
>> >>
>> >> On Thu, Nov 4, 2010 at 3:51 PM, Bernhard Suttner
>> >> <bernhard.suttner at winet.ch> wrote:
>> >> > Hi,
>> >> >
>> >> > session progress does have SDP which will then go through to A. But on
>> the
>> >> display of A it will only display "session progress" and not Ringing. If
>> A
>> >> is now a ISDN Gateway and has to signalize RINGING into the "ISDN world".
>> I
>> >> am not sure, if RINGING will be used for A to generate the ringback tone
>> for
>> >> itself on every device. This behaves on the client/device configuration.
>> >> >
>> >> > There was a nice discussion about this "issue" on the sip-implementors
>> >> list:
>> >> >
>> >> >
>> >>
>> http://www.mail-archive.com/sip-implementors at lists.cs.columbia.edu/msg06400.html
>> >> >
>> >> > There is a RFC for this case (Section: 3.2. ?Ringing Tone Generation)
>> >> > http://www.rfc-editor.org/rfc/rfc3960.txt
>> >> >
>> >> > What do you think?
>> >> >
>> >> > Best regards,
>> >> > Bernhard Suttner
>> >> >
>> >> >
>> >> >
>> >> > ----- Original Message -----
>> >> > From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
>> >> > To: freeswitch-dev at lists.freeswitch.org
>> >> > Sent: Thu, 04 Nov 2010 16:54:30 +0100
>> >> > Subject: Re: [Freeswitch-dev] Session Progress and RINGING
>> >> >
>> >> >
>> >> >> no, once it gets session progress it will not send any ringing.
>> >> >> The sip side 180 ringing is used to tell the phone to generate its own
>> >> >> inband ringing.
>> >> >>
>> >> >> does your 183 session progress contain a sdp?
>> >> >>
>> >> >>
>> >> >> On Thu, Nov 4, 2010 at 10:25 AM, Bernhard Suttner
>> >> >> <bernhard.suttner at winet.ch> wrote:
>> >> >> > Hi,
>> >> >> >
>> >> >> > A ---> FS ---> B
>> >> >> >
>> >> >> > B does send a Session Progress to FS. FS will forward the Session
>> >> Progress
>> >> >> to A.
>> >> >> > B does send a RINGING to FS. FS does _not_ forward this to A.
>> >> >> >
>> >> >> > Could it be, that the check the switch_test_flag(channel,
>> >> CF_EARLY_MEDIA)
>> >> >> in mod_sofia.c within the INDICATE_RINGING section is wrong. I think
>> it
>> >> is
>> >> >> allowed to send the Session Progress first and later the RINGING. The
>> >> >> ringing is for example for SIP/ISDN gateways necessary.
>> >> >> >
>> >> >> > Any hint is appreciated.
>> >> >> >
>> >> >> > Best regards,
>> >> >> > Bernhard Suttner
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-dev mailing list
>> >> >> > FreeSWITCH-dev at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >> >> >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> >> >> > http://www.freeswitch.org
>> >> >> >
>> >> >>
>> >> >>
>> >> >>
>> >> >> --
>> >> >> Anthony Minessale II
>> >> >>
>> >> >> FreeSWITCH http://www.freeswitch.org/
>> >> >> ClueCon http://www.cluecon.com/
>> >> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >>
>> >> >> AIM: anthm
>> >> >> MSN:anthony_minessale at hotmail.com
>> >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> >> IRC: irc.freenode.net #freeswitch
>> >> >>
>> >> >> FreeSWITCH Developer Conference
>> >> >> sip:888 at conference.freeswitch.org
>> >> >> googletalk:conf+888 at conference.freeswitch.org
>> >> >> pstn:+19193869900
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-dev mailing list
>> >> >> FreeSWITCH-dev at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> >> >> http://www.freeswitch.org
>> >> >>
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-dev mailing list
>> >> > FreeSWITCH-dev at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> >> > http://www.freeswitch.org
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org
>> >> googletalk:conf+888 at conference.freeswitch.org
>> >> pstn:+19193869900
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-dev mailing list
>> >> FreeSWITCH-dev at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> >> http://www.freeswitch.org
>> >>
>> >
>> > _______________________________________________
>> > FreeSWITCH-dev mailing list
>> > FreeSWITCH-dev at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From bernhard.suttner at winet.ch  Fri Nov  5 06:08:06 2010
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Fri, 5 Nov 2010 14:08:06 +0100
Subject: [Freeswitch-dev] Session Progress and RINGING
In-Reply-To: <AANLkTinOsit+_BHVxG9GHTu9nLHV93TkH=aZPS9Tzmg0@mail.gmail.com>
References: <20101104230538.7fc24d16@mail.winet.ch>
	<AANLkTinOsit+_BHVxG9GHTu9nLHV93TkH=aZPS9Tzmg0@mail.gmail.com>
Message-ID: <6d8efc90-97f4-4177-8cd6-0ae47678590c@winet.ch>

Hi Anthony,

please find attached the patch.  It will add a variable called "forward_180_after_183". If this is set (and exported!) to true, it will pass-through the Ringing from B to A. 

I had another idea why the behavior is necessary:
B could send a session progress but B (a SBC) does not really know, if the call will be ever in RINGING state towards the other party. Maybe the other party on B is not reachable and therefore the SBC does forward a informational message over RTP of the pre-answered (Session Progress 183) call to FS that the member is not reachable. In another case the call party of B is reachable and therefore the SBC does forward RINGING. 

I know that the main problem of this, is the flexible interpretation of the SIP protocol where all the different device manufacture try to read it different.  

Best regards,
Bernhard Suttner

-----Urspr?ngliche Nachricht-----
Von: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] Im Auftrag von Anthony Minessale
Gesendet: Donnerstag, 4. November 2010 23:18
An: freeswitch-dev at lists.freeswitch.org
Betreff: Re: [Freeswitch-dev] Session Progress and RINGING

try to patch it i guess.
I still say the RFC is to vague to call this behavior complying with the RFC.
if you can make a clean patch that we can wrap in a pram like
support-one-of-many-interpretation-of-rfc3960=true





On Thu, Nov 4, 2010 at 5:05 PM, Bernhard Suttner
<bernhard.suttner at winet.ch> wrote:
> We will try that out but its not "100%" the functionality as the sbc does signalize. If I understand you correct, the 183 with SDP will be forwarded to A with 180 with SDP. But this does change the functionality to A (first of all on SIP/ISDN gateways).

>
> Therefore I would prefer the "clean" way with RFC 3960.
>
> Would you include a patch like "support-rfc3960" configuration option to sofia config which will ignore the CF_EARLY_MEDIA test?
>
>
>
> ----- Original Message -----
> From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
> To: freeswitch-dev at lists.freeswitch.org
> Sent: Thu, 04 Nov 2010 22:54:34 +0100
> Subject: Re: [Freeswitch-dev] Session Progress and RINGING
>
>
>> how about what I said:
>> {ignore_early_media=ring_ready}
>>
>>
>>
>> On Thu, Nov 4, 2010 at 4:29 PM, Bernhard Suttner
>> <bernhard.suttner at winet.ch> wrote:
>> > Hi,
>> >
>> > yes, they are not really sure but I think that the given RFC does specify
>> this correctly (this does not mean, that all the SIP devices work like
>> that). I could understand that a change will maybe result in big troubles.
>> Perhaps a option for mod_sofia to "support-rfc3960" would be a good
>> solution. If you want I will write the patch because it should be "really
>> simple" to ignore the CF_EARLY_MEDIA if the option is set, or?
>> >
>> > We get the Session Progress with SDP and later the 180 Ringing from a
>> session border controller (I am not 100% sure, but I think its a a
>> Audiocodes). ?The 180 Ringing has to be sent towards A because A is a ISDN
>> gateway.
>> >
>> > Would do you prefer? Thanks for your investigation.
>> >
>> > Best regards,
>> > Bernhard Suttner
>> >
>> > ----- Original Message -----
>> > From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
>> > To: freeswitch-dev at lists.freeswitch.org
>> > Sent: Thu, 04 Nov 2010 22:09:48 +0100
>> > Subject: Re: [Freeswitch-dev] Session Progress and RINGING
>> >
>> >
>> >> My conclusion from reading that discussion is they have no idea what to
>> do.
>> >> There are votes both ways and 2 attempts to go of onto a tangent into
>> >> another topic.
>> >>
>> >> We have chosen on our implementation to ignore 180 once we have
>> >> already established a media path from a 183. ?The RFC is just vague
>> >> enough that this decision falls to us.
>> >>
>> >> I understand that ISDN has more precise signaling in this regard.
>> >> PROGRESS or ALERTING both with or without media as a flag on the packet.
>> >>
>> >> But the vast majority of SIP devices in the wild will ignore the 180
>> >> once early media has been established.
>> >> So even if we make changes to support this, it will be ignored by the
>> >> next guy in line.
>> >>
>> >> Do you hear media during this early media phase?
>> >>
>> >> This is why we have the originate param
>> >> {ignore_early_media=ring_ready} which will, in the case of sip,
>> >> translate 183 or 180 with and without SDP into a 180
>> >>
>> >> To change it to do what you are asking for could have extreme negative
>> >> side effects and not even work in most devices so this is why I am
>> >> reluctant to change it.
>> >>
>> >> Are you trying to cross connect 2 ISDN lines over SIP and preserve the
>> >> signalling?
>> >> If so, this is why SIP is flawed to begin with because it is lossy in
>> >> telephony signaling data.
>> >> This is why they now try to embed ss7 messages in the sip packets. =p
>> >>
>> >>
>> >>
>> >> On Thu, Nov 4, 2010 at 3:51 PM, Bernhard Suttner
>> >> <bernhard.suttner at winet.ch> wrote:
>> >> > Hi,
>> >> >
>> >> > session progress does have SDP which will then go through to A. But on
>> the
>> >> display of A it will only display "session progress" and not Ringing. If
>> A
>> >> is now a ISDN Gateway and has to signalize RINGING into the "ISDN world".
>> I
>> >> am not sure, if RINGING will be used for A to generate the ringback tone
>> for
>> >> itself on every device. This behaves on the client/device configuration.
>> >> >
>> >> > There was a nice discussion about this "issue" on the sip-implementors
>> >> list:
>> >> >
>> >> >
>> >>
>> http://www.mail-archive.com/sip-implementors at lists.cs.columbia.edu/msg06400.html
>> >> >
>> >> > There is a RFC for this case (Section: 3.2. ?Ringing Tone Generation)
>> >> > http://www.rfc-editor.org/rfc/rfc3960.txt
>> >> >
>> >> > What do you think?
>> >> >
>> >> > Best regards,
>> >> > Bernhard Suttner
>> >> >
>> >> >
>> >> >
>> >> > ----- Original Message -----
>> >> > From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
>> >> > To: freeswitch-dev at lists.freeswitch.org
>> >> > Sent: Thu, 04 Nov 2010 16:54:30 +0100
>> >> > Subject: Re: [Freeswitch-dev] Session Progress and RINGING
>> >> >
>> >> >
>> >> >> no, once it gets session progress it will not send any ringing.
>> >> >> The sip side 180 ringing is used to tell the phone to generate its own
>> >> >> inband ringing.
>> >> >>
>> >> >> does your 183 session progress contain a sdp?
>> >> >>
>> >> >>
>> >> >> On Thu, Nov 4, 2010 at 10:25 AM, Bernhard Suttner
>> >> >> <bernhard.suttner at winet.ch> wrote:
>> >> >> > Hi,
>> >> >> >
>> >> >> > A ---> FS ---> B
>> >> >> >
>> >> >> > B does send a Session Progress to FS. FS will forward the Session
>> >> Progress
>> >> >> to A.
>> >> >> > B does send a RINGING to FS. FS does _not_ forward this to A.
>> >> >> >
>> >> >> > Could it be, that the check the switch_test_flag(channel,
>> >> CF_EARLY_MEDIA)
>> >> >> in mod_sofia.c within the INDICATE_RINGING section is wrong. I think
>> it
>> >> is
>> >> >> allowed to send the Session Progress first and later the RINGING. The
>> >> >> ringing is for example for SIP/ISDN gateways necessary.
>> >> >> >
>> >> >> > Any hint is appreciated.
>> >> >> >
>> >> >> > Best regards,
>> >> >> > Bernhard Suttner
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-dev mailing list
>> >> >> > FreeSWITCH-dev at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >> >> >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> >> >> > http://www.freeswitch.org
>> >> >> >
>> >> >>
>> >> >>
>> >> >>
>> >> >> --
>> >> >> Anthony Minessale II
>> >> >>
>> >> >> FreeSWITCH http://www.freeswitch.org/
>> >> >> ClueCon http://www.cluecon.com/
>> >> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >>
>> >> >> AIM: anthm
>> >> >> MSN:anthony_minessale at hotmail.com
>> >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> >> IRC: irc.freenode.net #freeswitch
>> >> >>
>> >> >> FreeSWITCH Developer Conference
>> >> >> sip:888 at conference.freeswitch.org
>> >> >> googletalk:conf+888 at conference.freeswitch.org
>> >> >> pstn:+19193869900
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-dev mailing list
>> >> >> FreeSWITCH-dev at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> >> >> http://www.freeswitch.org
>> >> >>
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-dev mailing list
>> >> > FreeSWITCH-dev at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> >> > http://www.freeswitch.org
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org
>> >> googletalk:conf+888 at conference.freeswitch.org
>> >> pstn:+19193869900
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-dev mailing list
>> >> FreeSWITCH-dev at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> >> http://www.freeswitch.org
>> >>
>> >
>> > _______________________________________________
>> > FreeSWITCH-dev mailing list
>> > FreeSWITCH-dev at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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

_______________________________________________
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 paulo at voicetechnology.com.br  Tue Nov  9 13:40:11 2010
From: paulo at voicetechnology.com.br (=?ISO-8859-1?Q?Paulo_Rog=E9rio_Panhoto?=)
Date: Tue, 09 Nov 2010 19:40:11 -0200
Subject: [Freeswitch-dev] Getting DTMF from channel [SOLVED]
In-Reply-To: <AANLkTi=pKpLE25RiHLmYThmgd09WS_y1JdSMorPFgC8i@mail.gmail.com>
References: <4CC1EF79.7040409@voicetechnology.com.br>	<AANLkTineqLijmVO3zf-K+RR_suQKSDcPX7eAjgByCFAP@mail.gmail.com>	<4CC1FD91.8020000@voicetechnology.com.br>
	<AANLkTi=pKpLE25RiHLmYThmgd09WS_y1JdSMorPFgC8i@mail.gmail.com>
Message-ID: <4CD9BFBB.4030007@voicetechnology.com.br>

The audio thread was lacking a call to switch_core_session_read_frame().
After I placed the call, DTMF worked OK.

    About the contribution process, I was asked to send a ssh public key
and a jira user name. I sent the required information and got no reply
since.

    Kind Regards

        -- Paulo

On 22/10/10 19:47, Anthony Minessale wrote:
> Ok let's work on that.
> email consulting at freeswitch.org and ask for GIT commit to a dir for
> that module and we can get you a dir.
>
> Once its building on latest GIT, we can dig into it and get DTMF working.
>
>
>
> 2010/10/22 Paulo Rog?rio Panhoto <paulo at voicetechnology.com.br>:
>   
>>    I really want to post it as contribution but I don't know how the
>> process works. So, in the meantine, it is already published on github
>>
>> git://github.com/ppanhoto/Freeswitch-mod_mp4.git
>>
>>    Regards,
>>
>>    Paulo.
>>
>> On 22/10/10 18:38, Anthony Minessale wrote:
>>     
>>> maybe if you post the module for contribution we can add it to FS and
>>> figure it out by looking in the code to make it do what you want.
>>>
>>>
>>> 2010/10/22 Paulo Rog?rio Panhoto <paulo at voicetechnology.com.br>:
>>>
>>>       
>>>>    Hi,
>>>>
>>>>    I'm writing a module that allows playback of MP4 video files (with
>>>> libmp4v2). The playback itself is made by two functions ready to run on
>>>> separate threads (though, audio runs on current thread and video runs on
>>>> a separate one). At this point, I'm trying to implement dtmf cut-through.
>>>>
>>>>    After some research (I checked out mod_dptools.c and
>>>> switch_ivr_play_say.c) and this code was my best guess -- it runs on the
>>>> audio stream:
>>>>
>>>>        if(switch_channel_test_flag(pt->channel, CF_BREAK))
>>>>        {
>>>>            switch_channel_clear_flag(pt->channel, CF_BREAK);
>>>>            break;
>>>>        }
>>>>
>>>>        switch_ivr_parse_all_events(pt->session);
>>>>
>>>>        if(switch_channel_has_dtmf(pt->channel))
>>>>        {
>>>>            switch_channel_dequeue_dtmf(pt->channel, &dtmf);
>>>>            const char * terminators =
>>>> switch_channel_get_variable(pt->channel,
>>>> SWITCH_PLAYBACK_TERMINATORS_VARIABLE);
>>>>            if(terminators && !strcasecmp(terminators, "none"))
>>>> terminators = NULL;
>>>>            switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(pt->session),
>>>> SWITCH_LOG_DEBUG, "Digit %c\n", dtmf.digit);
>>>>            if(terminators && strchr(terminators, dtmf.digit))
>>>>            {
>>>>                std::string digit(&dtmf.digit, 0, 1);
>>>>                switch_channel_set_variable(pt->channel,
>>>> SWITCH_PLAYBACK_TERMINATOR_USED, digit.c_str());
>>>>                break;
>>>>            }
>>>>        }
>>>>
>>>>    Which didn't work. I'm asking if anyone has any idea.
>>>>
>>>>    Any help is appreciated
>>>>
>>>>    Regards,
>>>>
>>>>    Paulo R. Panhoto
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-dev mailing list
>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>         
>>>
>>>
>>>       
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>     
>
>
>   



From paulo at voicetechnology.com.br  Tue Nov  9 13:40:11 2010
From: paulo at voicetechnology.com.br (=?ISO-8859-1?Q?Paulo_Rog=E9rio_Panhoto?=)
Date: Tue, 09 Nov 2010 19:40:11 -0200
Subject: [Freeswitch-dev] Getting DTMF from channel [SOLVED]
In-Reply-To: <AANLkTi=pKpLE25RiHLmYThmgd09WS_y1JdSMorPFgC8i@mail.gmail.com>
References: <4CC1EF79.7040409@voicetechnology.com.br>	<AANLkTineqLijmVO3zf-K+RR_suQKSDcPX7eAjgByCFAP@mail.gmail.com>	<4CC1FD91.8020000@voicetechnology.com.br>
	<AANLkTi=pKpLE25RiHLmYThmgd09WS_y1JdSMorPFgC8i@mail.gmail.com>
Message-ID: <4CD9BFBB.4030007@voicetechnology.com.br>

The audio thread was lacking a call to switch_core_session_read_frame().
After I placed the call, DTMF worked OK.

    About the contribution process, I was asked to send a ssh public key
and a jira user name. I sent the required information and got no reply
since.

    Kind Regards

        -- Paulo

On 22/10/10 19:47, Anthony Minessale wrote:
> Ok let's work on that.
> email consulting at freeswitch.org and ask for GIT commit to a dir for
> that module and we can get you a dir.
>
> Once its building on latest GIT, we can dig into it and get DTMF working.
>
>
>
> 2010/10/22 Paulo Rog?rio Panhoto <paulo at voicetechnology.com.br>:
>   
>>    I really want to post it as contribution but I don't know how the
>> process works. So, in the meantine, it is already published on github
>>
>> git://github.com/ppanhoto/Freeswitch-mod_mp4.git
>>
>>    Regards,
>>
>>    Paulo.
>>
>> On 22/10/10 18:38, Anthony Minessale wrote:
>>     
>>> maybe if you post the module for contribution we can add it to FS and
>>> figure it out by looking in the code to make it do what you want.
>>>
>>>
>>> 2010/10/22 Paulo Rog?rio Panhoto <paulo at voicetechnology.com.br>:
>>>
>>>       
>>>>    Hi,
>>>>
>>>>    I'm writing a module that allows playback of MP4 video files (with
>>>> libmp4v2). The playback itself is made by two functions ready to run on
>>>> separate threads (though, audio runs on current thread and video runs on
>>>> a separate one). At this point, I'm trying to implement dtmf cut-through.
>>>>
>>>>    After some research (I checked out mod_dptools.c and
>>>> switch_ivr_play_say.c) and this code was my best guess -- it runs on the
>>>> audio stream:
>>>>
>>>>        if(switch_channel_test_flag(pt->channel, CF_BREAK))
>>>>        {
>>>>            switch_channel_clear_flag(pt->channel, CF_BREAK);
>>>>            break;
>>>>        }
>>>>
>>>>        switch_ivr_parse_all_events(pt->session);
>>>>
>>>>        if(switch_channel_has_dtmf(pt->channel))
>>>>        {
>>>>            switch_channel_dequeue_dtmf(pt->channel, &dtmf);
>>>>            const char * terminators =
>>>> switch_channel_get_variable(pt->channel,
>>>> SWITCH_PLAYBACK_TERMINATORS_VARIABLE);
>>>>            if(terminators && !strcasecmp(terminators, "none"))
>>>> terminators = NULL;
>>>>            switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(pt->session),
>>>> SWITCH_LOG_DEBUG, "Digit %c\n", dtmf.digit);
>>>>            if(terminators && strchr(terminators, dtmf.digit))
>>>>            {
>>>>                std::string digit(&dtmf.digit, 0, 1);
>>>>                switch_channel_set_variable(pt->channel,
>>>> SWITCH_PLAYBACK_TERMINATOR_USED, digit.c_str());
>>>>                break;
>>>>            }
>>>>        }
>>>>
>>>>    Which didn't work. I'm asking if anyone has any idea.
>>>>
>>>>    Any help is appreciated
>>>>
>>>>    Regards,
>>>>
>>>>    Paulo R. Panhoto
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-dev mailing list
>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>         
>>>
>>>
>>>       
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>     
>
>
>   



From msc at freeswitch.org  Wed Nov 10 07:02:10 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 10 Nov 2010 07:02:10 -0800
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today
Message-ID: <AANLkTimSd9b+tJnSKRVx_wGcNgarGbC7DzCGb=LEki3d@mail.gmail.com>

Hey folks,
Today's agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_11_10

We have a special announcement from Chad Philips and Kristian Kielhofner so
be sure to join us today!
-Michael
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From cbeeton at avaya.com  Wed Nov 10 11:41:42 2010
From: cbeeton at avaya.com (Beeton, Carolyn (Carolyn))
Date: Wed, 10 Nov 2010 14:41:42 -0500
Subject: [Freeswitch-dev] Question about presence_id
Message-ID: <E151F470685C15418AEA609E825A1710223F6B8C51@DC-US1MBEX4.global.avaya.com>

I am trying to understand the PRESENCE_IN events that are sent on call init, answer, and hangup.  I have two endpoints registered to Freeswitch, configured in the dialplan with presence_id set.  One set calls the other.

There is one event for each leg of the call, inbound and outbound.  The events contain the Channel-Name and the Channel-Presence-ID, and I would expect these to match (i.e. each event contains the presence_id of that leg), but it seems that for the inbound leg, they only match on the "ringing" event, not the "answered" and "hangup" event.  The two events for "answered" and "hangup" both have the presence_id of the callee, regardless of the Call-Direction.  Is this intentional? (I think it is the reason that lights stick on in certain scenarios - for example, if there are subscribers to the caller but not the callee then no notifications are sent on hangup)

Carolyn
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From anthony.minessale at gmail.com  Wed Nov 10 13:05:35 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Nov 2010 15:05:35 -0600
Subject: [Freeswitch-dev] Question about presence_id
In-Reply-To: <E151F470685C15418AEA609E825A1710223F6B8C51@DC-US1MBEX4.global.avaya.com>
References: <E151F470685C15418AEA609E825A1710223F6B8C51@DC-US1MBEX4.global.avaya.com>
Message-ID: <AANLkTi=_gqSJqsdJ=LknBq_tK9mDLro6t6m-LaNw9idf@mail.gmail.com>

do you have a trace?


On Wed, Nov 10, 2010 at 1:41 PM, Beeton, Carolyn (Carolyn)
<cbeeton at avaya.com> wrote:
> I am trying to understand the PRESENCE_IN events that are sent on call init,
> answer, and hangup.??I have two endpoints registered to Freeswitch,
> configured in the dialplan with presence_id set.? One set calls the other.
>
> There is one event for each leg of the call, inbound and outbound.? The
> events contain the Channel-Name and the Channel-Presence-ID, and I would
> expect these to match (i.e. each event contains the presence_id of that
> leg), but it seems that for the inbound leg, they only match on the
> "ringing" event, not the "answered" and "hangup" event.? The two events for
> "answered" and "hangup" both have the presence_id of the callee, regardless
> of the Call-Direction.? Is this intentional? (I think it is the reason that
> lights stick on in certain scenarios - for example, if there are subscribers
> to the caller but not the callee then no notifications are sent on hangup)
>
> Carolyn
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From chad at apartmentlines.com  Wed Nov 10 16:16:02 2010
From: chad at apartmentlines.com (Chad Phillips -- Apartment Lines)
Date: Wed, 10 Nov 2010 16:16:02 -0800
Subject: [Freeswitch-dev] Announcing the Jester Mail project
In-Reply-To: <mailman.0.1289433845.16624.freeswitch-dev@lists.freeswitch.org>
References: <mailman.0.1289433845.16624.freeswitch-dev@lists.freeswitch.org>
Message-ID: <E397DC81-923E-4C11-BBCC-9584ADC39D60@apartmentlines.com>

With funding and support from Star2Star Communications, and in conjunction with FreeSWITCH Solutions, I will soon be embarking on a project to bring a drop-in replacement for Asterisk's Comedian Mail to the FreeSWITCH platform.

As I learned in my own move from Asterisk to FreeSWITCH, one of the bigger challenges was finding a transparent way to transition my user's voicemail experience.  Since FreeSWITCH's native voicemail module did not adequately meet this need, I elected to write my own voicemail system, leveraging Lua -- the recommended scripting language for integrating complex tasks into the dialplan.

The underlying concept in Jester Mail is to provide a flexible, extensible, and easily customizable voicemail plugin for FreeSWITCH.  Unlike mod_voicemail, the code/configuration will be written in either XML or Lua, so that it will be easily hackable should the need arise.  Every effort will be made to make the components modular and configurable, to avoid the need to rewrite any of the core functionality.

Initial alpha releases will be based on my existing company's code, adding more and more of the standard Comedian Mail features, culminating in the 1.0 release, which will be a complete drop in replacement for Comedian Mail.  Future releases may extend functionality even beyond that provided in Comedian Mail -- we'll see about that when we get there.  :)

We'd like to encourage people to offer feedback and insight on the project as it progresses, and we'll do our best (as it fits into our budget, timeline, and larger goals) to incorporate this into our final product in a way that is most beneficial for all in the community.

Within the next 2-3 weeks, I'll be posting a strategy/architecture battle plan on the FreeSWITCH wiki.  A few weeks after that I'll commit the initial pre-alpha code to the contributions repository for community feedback.

There should be two nice side effects of this project:

  a) More people should be willing to make the transition from Asterisk to FreeSWITCH, as this removes another significant barrier for some.

  b) We will have excellent new resource to point people to for learning how to leverage Lua in FreeSWITCH.


Chad Phillips aka hunmonk

From cbeeton at avaya.com  Thu Nov 11 05:57:00 2010
From: cbeeton at avaya.com (Beeton, Carolyn (Carolyn))
Date: Thu, 11 Nov 2010 08:57:00 -0500
Subject: [Freeswitch-dev] Question about presence_id
In-Reply-To: <AANLkTi=_gqSJqsdJ=LknBq_tK9mDLro6t6m-LaNw9idf@mail.gmail.com>
References: <E151F470685C15418AEA609E825A1710223F6B8C51@DC-US1MBEX4.global.avaya.com>
	<AANLkTi=_gqSJqsdJ=LknBq_tK9mDLro6t6m-LaNw9idf@mail.gmail.com>
Message-ID: <E151F470685C15418AEA609E825A1710223F6F5042@DC-US1MBEX4.global.avaya.com>

Freeswitch log with EVENT DUMPs is attached.  47 and 49 are registered to freeswitch and both have presence_id set in the dialplan.  Another set is SUBSCRIBEd to 49 (for presence events) but nobody is SUBSCRIBEd to 47. 49 calls 47, 47 answers, then hangs up.  The LED for 49 sticks on at the monitoring set.  The "hangup" PRESENCE_IN event for both legs of the call has 47 as the presence_id.

Carolyn

> -----Original Message-----
> From: freeswitch-dev-bounces at lists.freeswitch.org 
> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On 
> Behalf Of Anthony Minessale
> Sent: Wednesday, November 10, 2010 4:06 PM
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] Question about presence_id
> 
> do you have a trace?
> 
> 
> On Wed, Nov 10, 2010 at 1:41 PM, Beeton, Carolyn (Carolyn) 
> <cbeeton at avaya.com> wrote:
> > I am trying to understand the PRESENCE_IN events that are 
> sent on call 
> > init, answer, and hangup.??I have two endpoints registered to 
> > Freeswitch, configured in the dialplan with presence_id 
> set.? One set calls the other.
> >
> > There is one event for each leg of the call, inbound and outbound.? 
> > The events contain the Channel-Name and the 
> Channel-Presence-ID, and I 
> > would expect these to match (i.e. each event contains the 
> presence_id 
> > of that leg), but it seems that for the inbound leg, they 
> only match 
> > on the "ringing" event, not the "answered" and "hangup" event.? The 
> > two events for "answered" and "hangup" both have the presence_id of 
> > the callee, regardless of the Call-Direction.? Is this 
> intentional? (I 
> > think it is the reason that lights stick on in certain 
> scenarios - for 
> > example, if there are subscribers to the caller but not the callee 
> > then no notifications are sent on hangup)
> >
> > Carolyn
> > _______________________________________________
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
> >
> >
> 
> 
> 
> --
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
> 
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From cbeeton at avaya.com  Thu Nov 11 08:13:10 2010
From: cbeeton at avaya.com (Beeton, Carolyn (Carolyn))
Date: Thu, 11 Nov 2010 11:13:10 -0500
Subject: [Freeswitch-dev] Minor log cleanup in sofia_reg.c
Message-ID: <E151F470685C15418AEA609E825A1710223F6F52F4@DC-US1MBEX4.global.avaya.com>

Here's a patch for a minor log cleanup in sofia_reg.c.   The "forbidden" variable is never set, so the logs keep showing a "challenge" being sent instead of "forbidden", when a registration is rejected.
 
diff --git a/src/mod/endpoints/mod_sofia/sofia_reg.c b/src/mod/endpoints/mod_sofia/sofia_reg.c
index 67641ae..61bb4aa 100644
--- a/src/mod/endpoints/mod_sofia/sofia_reg.c
+++ b/src/mod/endpoints/mod_sofia/sofia_reg.c
@@ -816,7 +816,7 @@ uint8_t sofia_reg_handle_register(nua_t *nua, sofia_profile_t *profile, nua_hand
        char contact_str[1024] = "";
        int nat_hack = 0;
        uint8_t multi_reg = 0, multi_reg_contact = 0, avoid_multi_reg = 0;
-       uint8_t stale = 0, forbidden = 0;
+       uint8_t stale = 0;
        auth_res_t auth_res;
        long exptime = 300;
        switch_event_t *event;
@@ -1115,7 +1115,7 @@ uint8_t sofia_reg_handle_register(nua_t *nua, sofia_profile_t *profile, nua_hand
 
                if (auth_res != AUTH_OK && !stale) {
                        if (profile->debug) {
-                               switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Send %s for [%s@%s]\n", forbidden ? "forbidden" : "challenge", to_user, to_host);
+                               switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Send %s for [%s@%s]\n", (auth_res == AUTH_FORBIDDEN) ? "forbidden" : "challenge", to_user, to_host);
                        }
                        if (auth_res == AUTH_FORBIDDEN) {
                                nua_respond(nh, SIP_403_FORBIDDEN, NUTAG_WITH_THIS(nua), TAG_END());

From william.suffill at gmail.com  Thu Nov 11 08:22:19 2010
From: william.suffill at gmail.com (William Suffill)
Date: Thu, 11 Nov 2010 11:22:19 -0500
Subject: [Freeswitch-dev] Minor log cleanup in sofia_reg.c
In-Reply-To: <E151F470685C15418AEA609E825A1710223F6F52F4@DC-US1MBEX4.global.avaya.com>
References: <E151F470685C15418AEA609E825A1710223F6F52F4@DC-US1MBEX4.global.avaya.com>
Message-ID: <AANLkTi=Ahtk=ir1CBwH5LgGMg1qpTS2C2TM3ZJVqr8kV@mail.gmail.com>

Best to open an issue in JIRA with the patch attached so it's easier
to keep track of.
http://jira.freeswitch.org


From anthony.minessale at gmail.com  Thu Nov 11 10:54:09 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Nov 2010 12:54:09 -0600
Subject: [Freeswitch-dev] Question about presence_id
In-Reply-To: <E151F470685C15418AEA609E825A1710223F6F5042@DC-US1MBEX4.global.avaya.com>
References: <E151F470685C15418AEA609E825A1710223F6B8C51@DC-US1MBEX4.global.avaya.com>
	<AANLkTi=_gqSJqsdJ=LknBq_tK9mDLro6t6m-LaNw9idf@mail.gmail.com>
	<E151F470685C15418AEA609E825A1710223F6F5042@DC-US1MBEX4.global.avaya.com>
Message-ID: <AANLkTikbur9-jhEZOtEpNOV3ovNSj_WgwBV6O23K3dmW@mail.gmail.com>

oh,

Dialplan: sofia/cbeetonscs.ca.nortel.com/49 at cbeetonscs.ca.nortel.com
Action set(presence_id=47 at cbeetonscs.ca.nortel.com)

That's cos you are setting it to 47 manually.

I think the problem you may be having with the paradigm in freeswitch
is the each leg of the call is it's own call.  There is not proxy
concept in FS where the bridge is actually 2 legs of one call.

When you call into FS to a 1 legged app like a conference then you can
set the var to 47, sure.
But if you are bridging and you want to monitor what you are bridged
to, you set the var on THAT LEG.

When you bridge to 47 use this as the data attr

{presence_id=47}sofia/cbeetonscs.ca.nortel.com/47%cbeetonscs.ca.nortel.com




On Thu, Nov 11, 2010 at 7:57 AM, Beeton, Carolyn (Carolyn)
<cbeeton at avaya.com> wrote:
> Freeswitch log with EVENT DUMPs is attached. ?47 and 49 are registered to freeswitch and both have presence_id set in the dialplan. ?Another set is SUBSCRIBEd to 49 (for presence events) but nobody is SUBSCRIBEd to 47. 49 calls 47, 47 answers, then hangs up. ?The LED for 49 sticks on at the monitoring set. ?The "hangup" PRESENCE_IN event for both legs of the call has 47 as the presence_id.
>
> Carolyn
>
>> -----Original Message-----
>> From: freeswitch-dev-bounces at lists.freeswitch.org
>> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On
>> Behalf Of Anthony Minessale
>> Sent: Wednesday, November 10, 2010 4:06 PM
>> To: freeswitch-dev at lists.freeswitch.org
>> Subject: Re: [Freeswitch-dev] Question about presence_id
>>
>> do you have a trace?
>>
>>
>> On Wed, Nov 10, 2010 at 1:41 PM, Beeton, Carolyn (Carolyn)
>> <cbeeton at avaya.com> wrote:
>> > I am trying to understand the PRESENCE_IN events that are
>> sent on call
>> > init, answer, and hangup.??I have two endpoints registered to
>> > Freeswitch, configured in the dialplan with presence_id
>> set.? One set calls the other.
>> >
>> > There is one event for each leg of the call, inbound and outbound.
>> > The events contain the Channel-Name and the
>> Channel-Presence-ID, and I
>> > would expect these to match (i.e. each event contains the
>> presence_id
>> > of that leg), but it seems that for the inbound leg, they
>> only match
>> > on the "ringing" event, not the "answered" and "hangup" event.? The
>> > two events for "answered" and "hangup" both have the presence_id of
>> > the callee, regardless of the Call-Direction.? Is this
>> intentional? (I
>> > think it is the reason that lights stick on in certain
>> scenarios - for
>> > example, if there are subscribers to the caller but not the callee
>> > then no notifications are sent on hangup)
>> >
>> > Carolyn
>> > _______________________________________________
>> > FreeSWITCH-dev mailing list
>> > FreeSWITCH-dev at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Thu Nov 11 10:55:29 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Nov 2010 12:55:29 -0600
Subject: [Freeswitch-dev] Question about presence_id
In-Reply-To: <AANLkTikbur9-jhEZOtEpNOV3ovNSj_WgwBV6O23K3dmW@mail.gmail.com>
References: <E151F470685C15418AEA609E825A1710223F6B8C51@DC-US1MBEX4.global.avaya.com>
	<AANLkTi=_gqSJqsdJ=LknBq_tK9mDLro6t6m-LaNw9idf@mail.gmail.com>
	<E151F470685C15418AEA609E825A1710223F6F5042@DC-US1MBEX4.global.avaya.com>
	<AANLkTikbur9-jhEZOtEpNOV3ovNSj_WgwBV6O23K3dmW@mail.gmail.com>
Message-ID: <AANLkTinBNK3u6H63v=R9z1n_7D3uq9QNQt4z18EqHNg5@mail.gmail.com>

P.S.

This is the reason we based it on registration because when devices
register this all works out of the box.


On Thu, Nov 11, 2010 at 12:54 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> oh,
>
> Dialplan: sofia/cbeetonscs.ca.nortel.com/49 at cbeetonscs.ca.nortel.com
> Action set(presence_id=47 at cbeetonscs.ca.nortel.com)
>
> That's cos you are setting it to 47 manually.
>
> I think the problem you may be having with the paradigm in freeswitch
> is the each leg of the call is it's own call. ?There is not proxy
> concept in FS where the bridge is actually 2 legs of one call.
>
> When you call into FS to a 1 legged app like a conference then you can
> set the var to 47, sure.
> But if you are bridging and you want to monitor what you are bridged
> to, you set the var on THAT LEG.
>
> When you bridge to 47 use this as the data attr
>
> {presence_id=47}sofia/cbeetonscs.ca.nortel.com/47%cbeetonscs.ca.nortel.com
>
>
>
>
> On Thu, Nov 11, 2010 at 7:57 AM, Beeton, Carolyn (Carolyn)
> <cbeeton at avaya.com> wrote:
>> Freeswitch log with EVENT DUMPs is attached. ?47 and 49 are registered to freeswitch and both have presence_id set in the dialplan. ?Another set is SUBSCRIBEd to 49 (for presence events) but nobody is SUBSCRIBEd to 47. 49 calls 47, 47 answers, then hangs up. ?The LED for 49 sticks on at the monitoring set. ?The "hangup" PRESENCE_IN event for both legs of the call has 47 as the presence_id.
>>
>> Carolyn
>>
>>> -----Original Message-----
>>> From: freeswitch-dev-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On
>>> Behalf Of Anthony Minessale
>>> Sent: Wednesday, November 10, 2010 4:06 PM
>>> To: freeswitch-dev at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-dev] Question about presence_id
>>>
>>> do you have a trace?
>>>
>>>
>>> On Wed, Nov 10, 2010 at 1:41 PM, Beeton, Carolyn (Carolyn)
>>> <cbeeton at avaya.com> wrote:
>>> > I am trying to understand the PRESENCE_IN events that are
>>> sent on call
>>> > init, answer, and hangup.??I have two endpoints registered to
>>> > Freeswitch, configured in the dialplan with presence_id
>>> set.? One set calls the other.
>>> >
>>> > There is one event for each leg of the call, inbound and outbound.
>>> > The events contain the Channel-Name and the
>>> Channel-Presence-ID, and I
>>> > would expect these to match (i.e. each event contains the
>>> presence_id
>>> > of that leg), but it seems that for the inbound leg, they
>>> only match
>>> > on the "ringing" event, not the "answered" and "hangup" event.? The
>>> > two events for "answered" and "hangup" both have the presence_id of
>>> > the callee, regardless of the Call-Direction.? Is this
>>> intentional? (I
>>> > think it is the reason that lights stick on in certain
>>> scenarios - for
>>> > example, if there are subscribers to the caller but not the callee
>>> > then no notifications are sent on hangup)
>>> >
>>> > Carolyn
>>> > _______________________________________________
>>> > FreeSWITCH-dev mailing list
>>> > FreeSWITCH-dev at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> >
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>



-- 
Anthony Minessale II

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

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

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


From cbeeton at avaya.com  Thu Nov 11 11:33:33 2010
From: cbeeton at avaya.com (Beeton, Carolyn (Carolyn))
Date: Thu, 11 Nov 2010 14:33:33 -0500
Subject: [Freeswitch-dev] Question about presence_id
In-Reply-To: <AANLkTikbur9-jhEZOtEpNOV3ovNSj_WgwBV6O23K3dmW@mail.gmail.com>
References: <E151F470685C15418AEA609E825A1710223F6B8C51@DC-US1MBEX4.global.avaya.com>
	<AANLkTi=_gqSJqsdJ=LknBq_tK9mDLro6t6m-LaNw9idf@mail.gmail.com>
	<E151F470685C15418AEA609E825A1710223F6F5042@DC-US1MBEX4.global.avaya.com>
	<AANLkTikbur9-jhEZOtEpNOV3ovNSj_WgwBV6O23K3dmW@mail.gmail.com>
Message-ID: <E151F470685C15418AEA609E825A1710223F6F55FA@DC-US1MBEX4.global.avaya.com>

I have 47's presence set to 47 and 49's presence set to 49.  Here's my dialplan:

   <extension name="47">
      <condition field="destination_number" expression="^47$">
         <action application="set" data="presence_id=47@$${domain}"/>
         <action application="bridge" data="sofia/cbeetonscs.ca.nortel.com/47%cbeetonscs.ca.nortel.com"/>
      </condition>
   </extension>
   <extension name="49">
      <condition field="destination_number" expression="^49$">
         <action application="set" data="presence_id=49@$${domain}"/>
         <action application="bridge" data="sofia/cbeetonscs.ca.nortel.com/49%cbeetonscs.ca.nortel.com"/>
      </condition>
   </extension>

Should I not be setting the presence_id?  I thought I was told to at one point... Maybe I misunderstood.

Carolyn

> -----Original Message-----
> From: freeswitch-dev-bounces at lists.freeswitch.org 
> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On 
> Behalf Of Anthony Minessale
> Sent: Thursday, November 11, 2010 1:54 PM
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] Question about presence_id
> 
> oh,
> 
> Dialplan: sofia/cbeetonscs.ca.nortel.com/49 at cbeetonscs.ca.nortel.com
> Action set(presence_id=47 at cbeetonscs.ca.nortel.com)
> 
> That's cos you are setting it to 47 manually.
> 
> I think the problem you may be having with the paradigm in 
> freeswitch is the each leg of the call is it's own call.  
> There is not proxy concept in FS where the bridge is actually 
> 2 legs of one call.
> 
> When you call into FS to a 1 legged app like a conference 
> then you can set the var to 47, sure.
> But if you are bridging and you want to monitor what you are 
> bridged to, you set the var on THAT LEG.
> 
> When you bridge to 47 use this as the data attr
> 
> {presence_id=47}sofia/cbeetonscs.ca.nortel.com/47%cbeetonscs.c
> a.nortel.com
> 
> 
> 
> 
> On Thu, Nov 11, 2010 at 7:57 AM, Beeton, Carolyn (Carolyn) 
> <cbeeton at avaya.com> wrote:
> > Freeswitch log with EVENT DUMPs is attached. ?47 and 49 are 
> registered to freeswitch and both have presence_id set in the 
> dialplan. ?Another set is SUBSCRIBEd to 49 (for presence 
> events) but nobody is SUBSCRIBEd to 47. 49 calls 47, 47 
> answers, then hangs up. ?The LED for 49 sticks on at the 
> monitoring set. ?The "hangup" PRESENCE_IN event for both legs 
> of the call has 47 as the presence_id.
> >
> > Carolyn
> >
> >> -----Original Message-----
> >> From: freeswitch-dev-bounces at lists.freeswitch.org
> >> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of 
> >> Anthony Minessale
> >> Sent: Wednesday, November 10, 2010 4:06 PM
> >> To: freeswitch-dev at lists.freeswitch.org
> >> Subject: Re: [Freeswitch-dev] Question about presence_id
> >>
> >> do you have a trace?
> >>
> >>
> >> On Wed, Nov 10, 2010 at 1:41 PM, Beeton, Carolyn (Carolyn) 
> >> <cbeeton at avaya.com> wrote:
> >> > I am trying to understand the PRESENCE_IN events that are
> >> sent on call
> >> > init, answer, and hangup.??I have two endpoints registered to 
> >> > Freeswitch, configured in the dialplan with presence_id
> >> set.? One set calls the other.
> >> >
> >> > There is one event for each leg of the call, inbound and 
> outbound.
> >> > The events contain the Channel-Name and the
> >> Channel-Presence-ID, and I
> >> > would expect these to match (i.e. each event contains the
> >> presence_id
> >> > of that leg), but it seems that for the inbound leg, they
> >> only match
> >> > on the "ringing" event, not the "answered" and "hangup" 
> event.? The 
> >> > two events for "answered" and "hangup" both have the 
> presence_id of 
> >> > the callee, regardless of the Call-Direction.? Is this
> >> intentional? (I
> >> > think it is the reason that lights stick on in certain
> >> scenarios - for
> >> > example, if there are subscribers to the caller but not 
> the callee 
> >> > then no notifications are sent on hangup)
> >> >
> >> > Carolyn
> >> > _______________________________________________
> >> > FreeSWITCH-dev mailing list
> >> > FreeSWITCH-dev at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >> >
> >> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-de
> >> v
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/ ClueCon 
> http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-dev mailing list
> >> FreeSWITCH-dev at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-de
> >> v
> >> http://www.freeswitch.org
> >>
> > _______________________________________________
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
> >
> >
> 
> 
> 
> --
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
> 

From anthony.minessale at gmail.com  Thu Nov 11 11:54:58 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Nov 2010 13:54:58 -0600
Subject: [Freeswitch-dev] Question about presence_id
In-Reply-To: <E151F470685C15418AEA609E825A1710223F6F55FA@DC-US1MBEX4.global.avaya.com>
References: <E151F470685C15418AEA609E825A1710223F6B8C51@DC-US1MBEX4.global.avaya.com>
	<AANLkTi=_gqSJqsdJ=LknBq_tK9mDLro6t6m-LaNw9idf@mail.gmail.com>
	<E151F470685C15418AEA609E825A1710223F6F5042@DC-US1MBEX4.global.avaya.com>
	<AANLkTikbur9-jhEZOtEpNOV3ovNSj_WgwBV6O23K3dmW@mail.gmail.com>
	<E151F470685C15418AEA609E825A1710223F6F55FA@DC-US1MBEX4.global.avaya.com>
Message-ID: <AANLkTinMWVSZdUQJHmVvYtmA-fiOGo9EMaZNtujgLKR7@mail.gmail.com>

you want to set the presence_id var on the leg that actually is
connected to the phone.
We set vars in oubound calls with {var=val} blocks before the channel
name in the dial string.
like so:

  <extension name="47">
     <condition field="destination_number" expression="^47$">
        <action application="bridge"
data="{presence_id=47@$${domain}}sofia/cbeetonscs.ca.nortel.com/47%cbeetonscs.ca.nortel.com"/>
     </condition>
  </extension>
  <extension name="49">
     <condition field="destination_number" expression="^49$">
        <action application="bridge"
data="{presence_id=49@$${domain}}sofia/cbeetonscs.ca.nortel.com/49%cbeetonscs.ca.nortel.com"/>
     </condition>
  </extension>



On Thu, Nov 11, 2010 at 1:33 PM, Beeton, Carolyn (Carolyn)
<cbeeton at avaya.com> wrote:
> I have 47's presence set to 47 and 49's presence set to 49. ?Here's my dialplan:
>
> ? <extension name="47">
> ? ? ?<condition field="destination_number" expression="^47$">
> ? ? ? ? <action application="set" data="presence_id=47@$${domain}"/>
> ? ? ? ? <action application="bridge" data="sofia/cbeetonscs.ca.nortel.com/47%cbeetonscs.ca.nortel.com"/>
> ? ? ?</condition>
> ? </extension>
> ? <extension name="49">
> ? ? ?<condition field="destination_number" expression="^49$">
> ? ? ? ? <action application="set" data="presence_id=49@$${domain}"/>
> ? ? ? ? <action application="bridge" data="sofia/cbeetonscs.ca.nortel.com/49%cbeetonscs.ca.nortel.com"/>
> ? ? ?</condition>
> ? </extension>
>
> Should I not be setting the presence_id? ?I thought I was told to at one point... Maybe I misunderstood.
>
> Carolyn
>
>> -----Original Message-----
>> From: freeswitch-dev-bounces at lists.freeswitch.org
>> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On
>> Behalf Of Anthony Minessale
>> Sent: Thursday, November 11, 2010 1:54 PM
>> To: freeswitch-dev at lists.freeswitch.org
>> Subject: Re: [Freeswitch-dev] Question about presence_id
>>
>> oh,
>>
>> Dialplan: sofia/cbeetonscs.ca.nortel.com/49 at cbeetonscs.ca.nortel.com
>> Action set(presence_id=47 at cbeetonscs.ca.nortel.com)
>>
>> That's cos you are setting it to 47 manually.
>>
>> I think the problem you may be having with the paradigm in
>> freeswitch is the each leg of the call is it's own call.
>> There is not proxy concept in FS where the bridge is actually
>> 2 legs of one call.
>>
>> When you call into FS to a 1 legged app like a conference
>> then you can set the var to 47, sure.
>> But if you are bridging and you want to monitor what you are
>> bridged to, you set the var on THAT LEG.
>>
>> When you bridge to 47 use this as the data attr
>>
>> {presence_id=47}sofia/cbeetonscs.ca.nortel.com/47%cbeetonscs.c
>> a.nortel.com
>>
>>
>>
>>
>> On Thu, Nov 11, 2010 at 7:57 AM, Beeton, Carolyn (Carolyn)
>> <cbeeton at avaya.com> wrote:
>> > Freeswitch log with EVENT DUMPs is attached. ?47 and 49 are
>> registered to freeswitch and both have presence_id set in the
>> dialplan. ?Another set is SUBSCRIBEd to 49 (for presence
>> events) but nobody is SUBSCRIBEd to 47. 49 calls 47, 47
>> answers, then hangs up. ?The LED for 49 sticks on at the
>> monitoring set. ?The "hangup" PRESENCE_IN event for both legs
>> of the call has 47 as the presence_id.
>> >
>> > Carolyn
>> >
>> >> -----Original Message-----
>> >> From: freeswitch-dev-bounces at lists.freeswitch.org
>> >> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of
>> >> Anthony Minessale
>> >> Sent: Wednesday, November 10, 2010 4:06 PM
>> >> To: freeswitch-dev at lists.freeswitch.org
>> >> Subject: Re: [Freeswitch-dev] Question about presence_id
>> >>
>> >> do you have a trace?
>> >>
>> >>
>> >> On Wed, Nov 10, 2010 at 1:41 PM, Beeton, Carolyn (Carolyn)
>> >> <cbeeton at avaya.com> wrote:
>> >> > I am trying to understand the PRESENCE_IN events that are
>> >> sent on call
>> >> > init, answer, and hangup.??I have two endpoints registered to
>> >> > Freeswitch, configured in the dialplan with presence_id
>> >> set.? One set calls the other.
>> >> >
>> >> > There is one event for each leg of the call, inbound and
>> outbound.
>> >> > The events contain the Channel-Name and the
>> >> Channel-Presence-ID, and I
>> >> > would expect these to match (i.e. each event contains the
>> >> presence_id
>> >> > of that leg), but it seems that for the inbound leg, they
>> >> only match
>> >> > on the "ringing" event, not the "answered" and "hangup"
>> event.? The
>> >> > two events for "answered" and "hangup" both have the
>> presence_id of
>> >> > the callee, regardless of the Call-Direction.? Is this
>> >> intentional? (I
>> >> > think it is the reason that lights stick on in certain
>> >> scenarios - for
>> >> > example, if there are subscribers to the caller but not
>> the callee
>> >> > then no notifications are sent on hangup)
>> >> >
>> >> > Carolyn
>> >> > _______________________________________________
>> >> > FreeSWITCH-dev mailing list
>> >> > FreeSWITCH-dev at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >> >
>> >>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-de
>> >> v
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/ ClueCon
>> http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org
>> >> googletalk:conf+888 at conference.freeswitch.org
>> >> pstn:+19193869900
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-dev mailing list
>> >> FreeSWITCH-dev at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-de
>> >> v
>> >> http://www.freeswitch.org
>> >>
>> > _______________________________________________
>> > FreeSWITCH-dev mailing list
>> > FreeSWITCH-dev at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From cbeeton at avaya.com  Thu Nov 11 12:24:43 2010
From: cbeeton at avaya.com (Beeton, Carolyn (Carolyn))
Date: Thu, 11 Nov 2010 15:24:43 -0500
Subject: [Freeswitch-dev] Question about presence_id
In-Reply-To: <E151F470685C15418AEA609E825A1710223F6F55FA@DC-US1MBEX4.global.avaya.com>
References: <E151F470685C15418AEA609E825A1710223F6B8C51@DC-US1MBEX4.global.avaya.com>
	<AANLkTi=_gqSJqsdJ=LknBq_tK9mDLro6t6m-LaNw9idf@mail.gmail.com>
	<E151F470685C15418AEA609E825A1710223F6F5042@DC-US1MBEX4.global.avaya.com>
	<AANLkTikbur9-jhEZOtEpNOV3ovNSj_WgwBV6O23K3dmW@mail.gmail.com>
	<E151F470685C15418AEA609E825A1710223F6F55FA@DC-US1MBEX4.global.avaya.com>
Message-ID: <E151F470685C15418AEA609E825A1710223F6F56D0@DC-US1MBEX4.global.avaya.com>

Ok - I think I see now.  Changed my dialplan to this:

   <extension name="47">
      <condition field="destination_number" expression="^47$">
         <action application="bridge" data="{presence_id=47@$${domain}}sofia/cbeetonscs.ca.nortel.com/47%cbeetonscs.ca.nortel.com"/>
      </condition>
   </extension> 
   <extension name="49">
      <condition field="destination_number" expression="^49$">
         <action application="bridge" data="{presence_id=49@$${domain}}sofia/cbeetonscs.ca.nortel.com/49%cbeetonscs.ca.nortel.com"/>
      </condition>
   </extension>

That sets the presence ID for the outgoing legs properly.  The presence ID for the incoming leg is set by default by registration.  And events are generated as expected and lights work as expected.  I just didn't find this bit of info anywhere in the docs!

Carolyn

From msc at freeswitch.org  Fri Nov 12 15:14:55 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 12 Nov 2010 15:14:55 -0800
Subject: [Freeswitch-dev] FreeSWITCH Snapshot Information
Message-ID: <AANLkTikbuoq9cvLU4kaB5HAn6E6MM5fbzS7ucnYUbrLP@mail.gmail.com>

Hello all,

It was recently brought to my attention that the freeswitch-snapshot files
were not getting updated properly. It turns out that the cron process was
still using the SVN repo! Our bad.

I have fixed the snapshot to use the git repo. I manually ran the script and
it created new snapshot files which can be downloaded now. I will monitor
the cron job this weekend to confirm if it is properly generating new files
each night. Note: it can take up to 24 hours from the time the cron job runs
to the time you see the files over at files.freeswitch.org so be patient.

The new files are:
freeswitch-snapshot.tar.bz2
freeswitch-snapshot.tar.bz2.md5
freeswitch-snapshot.tar.bz2.sha1
freeswitch-snapshot.tar.gz
freeswitch-snapshot.tar.gz.md5
freeswitch-snapshot.tar.gz.sha1

The files should be dated 2010-Nov-12. Any files earlier than this should be
considered unreliable. Feel free to try the snapshot on your system.
Remember that the snapshots are pre-bootstrapped so you can go straight to
the ./configure && make install process.

Email me off list if you have any questions.

Thanks,
Michael
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From math.parent at gmail.com  Sat Nov 13 05:22:26 2010
From: math.parent at gmail.com (Mathieu Parent)
Date: Sat, 13 Nov 2010 14:22:26 +0100
Subject: [Freeswitch-dev] How to compile FS debian packages?
Message-ID: <AANLkTinCit4AS2nbj0Zxr9HasWnpOJjOW3tThefWzzZL@mail.gmail.com>

Hello,

There was some changes in the debian dir and I am not able to compile
FS anymore using dpkg-buildpackage.

What is the new correct way to build the package?

Mathieu


From brian at freeswitch.org  Sat Nov 13 09:59:12 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Nov 2010 11:59:12 -0600
Subject: [Freeswitch-dev] How to compile FS debian packages?
In-Reply-To: <AANLkTinCit4AS2nbj0Zxr9HasWnpOJjOW3tThefWzzZL@mail.gmail.com>
References: <AANLkTinCit4AS2nbj0Zxr9HasWnpOJjOW3tThefWzzZL@mail.gmail.com>
Message-ID: <3FFBD02D-9798-4C99-AA4D-B72EE133FD45@freeswitch.org>

What is the error and why isn't it compiling?  

/b

On Nov 13, 2010, at 7:22 AM, Mathieu Parent wrote:

> Hello,
> 
> There was some changes in the debian dir and I am not able to compile
> FS anymore using dpkg-buildpackage.
> 
> What is the new correct way to build the package?
> 
> Mathieu



From math.parent at gmail.com  Sat Nov 13 11:02:08 2010
From: math.parent at gmail.com (Mathieu Parent)
Date: Sat, 13 Nov 2010 20:02:08 +0100
Subject: [Freeswitch-dev] How to compile FS debian packages?
In-Reply-To: <3FFBD02D-9798-4C99-AA4D-B72EE133FD45@freeswitch.org>
References: <AANLkTinCit4AS2nbj0Zxr9HasWnpOJjOW3tThefWzzZL@mail.gmail.com>
	<3FFBD02D-9798-4C99-AA4D-B72EE133FD45@freeswitch.org>
Message-ID: <AANLkTiniSFhBcgYe7qcqfPKR46p-GVzw8+JQ7BrO-6sN@mail.gmail.com>

2010/11/13 Brian West <brian at freeswitch.org>:
> What is the error and why isn't it compiling?

dpkg-source: error: can't build with source format '3.0 (quilt)': no
orig.tar file found


One step further: I have run:

./debian/rules upstream-convert
cd /home/mathieu/apps/freeswitch/freeswitch-1.0.head~git.master.20101112.0
dpkg-buildpackage

And now I have:
...
quiet_libtool: link: (cd ".libs" && rm -f "libfreeswitch.so.1" && ln
-s "libfreeswitch.so.1.0.0" "libfreeswitch.so.1")
quiet_libtool: link: (cd ".libs" && rm -f "libfreeswitch.so" && ln -s
"libfreeswitch.so.1.0.0" "libfreeswitch.so")
quiet_libtool: link: ar cru .libs/libfreeswitch.a
libs/libedit/src/.libs/libedit.a  libfreeswitch_la-switch_apr.o
libfreeswitch_la-switch_buffer.o libfreeswitch_la-switch_caller.o
libfreeswitch_la-switch_channel.o libfreeswitch_la-switch_console.o
libfreeswitch_la-switch_mprintf.o
libfreeswitch_la-switch_core_media_bug.o
libfreeswitch_la-switch_core_timer.o
libfreeswitch_la-switch_core_asr.o
libfreeswitch_la-switch_core_event_hook.o
libfreeswitch_la-switch_core_speech.o
libfreeswitch_la-switch_core_memory.o
libfreeswitch_la-switch_core_codec.o
libfreeswitch_la-switch_core_file.o
libfreeswitch_la-switch_core_hash.o
libfreeswitch_la-switch_core_sqldb.o
libfreeswitch_la-switch_core_session.o
libfreeswitch_la-switch_core_directory.o
libfreeswitch_la-switch_core_state_machine.o
libfreeswitch_la-switch_core_io.o
libfreeswitch_la-switch_core_rwlock.o
libfreeswitch_la-switch_core_port_allocator.o
libfreeswitch_la-switch_core.o libfreeswitch_la-switch_scheduler.o
libfreeswitch_la-switch_core_db.o libfreeswitch_la-switch_dso.o
libfreeswitch_la-switch_loadable_module.o
libfreeswitch_la-switch_utils.o libfreeswitch_la-switch_event.o
libfreeswitch_la-switch_resample.o libfreeswitch_la-switch_regex.o
libfreeswitch_la-switch_rtp.o libfreeswitch_la-switch_ivr_bridge.o
libfreeswitch_la-switch_ivr_originate.o
libfreeswitch_la-switch_ivr_async.o
libfreeswitch_la-switch_ivr_play_say.o
libfreeswitch_la-switch_ivr_say.o libfreeswitch_la-switch_ivr_menu.o
libfreeswitch_la-switch_ivr.o libfreeswitch_la-switch_stun.o
libfreeswitch_la-switch_nat.o libfreeswitch_la-switch_log.o
libfreeswitch_la-switch_xml.o libfreeswitch_la-switch_xml_config.o
libfreeswitch_la-switch_config.o libfreeswitch_la-switch_time.o
libfreeswitch_la-switch_odbc.o libfreeswitch_la-switch_limit.o
libfreeswitch_la-g711.o libfreeswitch_la-switch_pcm.o
libfreeswitch_la-switch_profile.o libfreeswitch_la-switch_json.o
libfreeswitch_la-stfu.o libfreeswitch_la-libteletone_detect.o
libfreeswitch_la-libteletone_generate.o libfreeswitch_la-miniwget.o
libfreeswitch_la-minixml.o libfreeswitch_la-igd_desc_parse.o
libfreeswitch_la-minisoap.o libfreeswitch_la-miniupnpc.o
libfreeswitch_la-upnpreplyparse.o libfreeswitch_la-upnpcommands.o
libfreeswitch_la-minissdpc.o libfreeswitch_la-upnperrors.o
libfreeswitch_la-natpmp.o libfreeswitch_la-getgateway.o switch_cpp.o
quiet_libtool: link: ranlib .libs/libfreeswitch.a
/bin/sed: can't read
/home/mathieu/apps/freeswitch/freeswitch.git/libs/apr-util/xml/expat/lib/libexpat.la:
No such file or directory
quiet_libtool: link:
`/home/mathieu/apps/freeswitch/freeswitch.git/libs/apr-util/xml/expat/lib/libexpat.la'
is not a valid libtool archive
make[2]: *** [libfreeswitch.la] Erreur 1
make[2]: quittant le r?pertoire ?
/home/mathieu/apps/freeswitch/freeswitch-1.0.head~git.master.20101112.0
?
make[1]: *** [all] Erreur 2
make[1]: quittant le r?pertoire ?
/home/mathieu/apps/freeswitch/freeswitch-1.0.head~git.master.20101112.0
?
make: *** [build-stamp] Erreur 2
dpkg-buildpackage: erreur: debian/rules build a produit une erreur de
sortie de type 2



-- 
Mathieu


From lakindia89 at gmail.com  Mon Nov 15 01:13:24 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Mon, 15 Nov 2010 14:43:24 +0530
Subject: [Freeswitch-dev] ftmod_libpri - RDNIS patch
Message-ID: <AANLkTik1K9=G-mtqrUP2FR-mYp-H6T_8RL8E8WY+f8zu@mail.gmail.com>

Hi all,
Please find the attached patch file which contain a small change to handle
RDNIS in ftmod_libpri.
Somebody can review it and find ok, do commit it.


regards,
Lakshmanan G.
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From steve.kurzeja at gmail.com  Wed Nov 10 00:54:45 2010
From: steve.kurzeja at gmail.com (Steve Kurzeja)
Date: Wed, 10 Nov 2010 21:54:45 +1300
Subject: [Freeswitch-dev] VLAN support in sofia module
In-Reply-To: <AANLkTinWgbmE-B7C=2Y4xsvTBD9UzLbpQ62kzc2gksno@mail.gmail.com>
References: <AANLkTi=9RFk3dfj2ZKgphguNT6S+cuwGh6qiqS7r78Vj@mail.gmail.com>
	<AANLkTiks+SOQZ2Kc+nzXwkwuVaBEAk-Ja1CfmTO9dz-G@mail.gmail.com>
	<AANLkTikkCOu1MvMOEyBXBXcMX+svKNKLyi5Oa=v+fNdF@mail.gmail.com>
	<AANLkTinWgbmE-B7C=2Y4xsvTBD9UzLbpQ62kzc2gksno@mail.gmail.com>
Message-ID: <AANLkTin0chVQT-RAv-7w7EJb4fwRhsB7MBMorwOR=7Ce@mail.gmail.com>

I think what the original poster is saying is that his existing architecture
uses "overlapping" IP addresses ("one ip address for all clients").
Clients/devices are on separate vlans and the SBC sits in each vlan
listening on the same IP address. Client <--> SBC IP is then on the same
broadcast domain and not IP routed. The clients are all configured to talk
to the same SBC address which makes provisioning simple.

So he would want something like this:

sip profile 1 listening on 10.0.0.1 vlan1
sip profile 2 listening on 10.0.0.1 vlan2

The likes of ACME packet SBCs/GENBAND etc support this but it is not
possible in freeswitch currently.

In linux this can be done using the SO_BINDDEVICE socket option to force
packets out a particular interface rather than use the linux IP routing
table. A good description is here:
http://codingrelic.geekhold.com/2009/10/code-snippet-sobindtodevice.html

Regards,
Steve


On Wed, Oct 27, 2010 at 6:22 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> its not possible to listen on 0.0.0.0 you can only guess at best on
> which ip goes with what packets and you cannot choose which source
> addr to put in outbound packets in that case.  This is a limitation
> caused by the design of the SIP spec.
>
>
> You must create a distinct profile for every interface with the
> desired IP from each vlan.
>
>
>
> On Tue, Oct 26, 2010 at 11:49 AM, Steven Ayre <steveayre at gmail.com> wrote:
> > Thinking about it, it won't work. ext-sip-ip will be inside the
> > packets but they'll be sent from sip-ip so the clients won't see it
> > coming from the IP you want.
> >
> > On 26 October 2010 17:47, Steven Ayre <steveayre at gmail.com> wrote:
> >> Possibly your best bet would be to set sip-ip and rtp-ip to the
> >> specific interface and ext-sip-ip and ext-rtp-ip to the global IP.
> >>
> >> I don't know how well this'll work though. Sofia might get very
> >> confused about where to send stuff to, since the same IP is used on
> >> essentially separate networks.
> >>
> >> -Steve
> >>
> >>
> >> On 23 October 2010 19:47, Adam Ku?mirek <amkusmirek at gmail.com> wrote:
> >>> Hi All
> >>>
> >>> I would like to run FreeSwitch as SBC in my network. I have
> >>> architecture with all clients connected via separate vlans.
> >>> It gives me possibility to use one ip address for all clients, but I
> >>> need FS sofia profile to bind to specific interface (eg eth0:20 vlan
> >>> 20) and global ip address.
> >>> As I see, for now sofia module allows to bind to ip address only. I
> >>> looked quickly to sourcees and it seems that this is limited by sofia
> >>> library.
> >>> Of course i may be wrong because I'm new to FS and didn't have time to
> >>> analyze sources.
> >>> Please, write some words on this topic.
> >>> Do you have plans to implement vlan support in FS.
> >>> I know that AcmePacket Net-Net gives such feature. Don't know any open
> >>> voip solution with this iplemented.
> >>>
> >>> Regards Adam
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-dev mailing list
> >>> FreeSWITCH-dev at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> >>> http://www.freeswitch.org
> >>>
> >>
> >
> > _______________________________________________
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
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From bjohns at digium.com  Thu Nov  4 09:56:00 2010
From: bjohns at digium.com (Bryan M. Johns)
Date: Thu, 4 Nov 2010 11:56:00 -0500 (CDT)
Subject: [Freeswitch-dev] CFP - Open Source Telephony Devroom - FOSDEM 2011
In-Reply-To: <1198051859.134562.1288889622968.JavaMail.root@zimbra>
Message-ID: <337828871.134577.1288889760688.JavaMail.root@zimbra>

Greetings,

This is a call for presentations for the open source telephony devroom at FOSDEM 2011 - http://www.fosdem.org/.

We will be holding a day full of presentations on development topics in the area of open source telephony on Sunday, February 6th.  The schedule allows for presentations from 9:00 to 17:00.  The room we have available will have a projector, wifi, and 59 seats.

Please submit all proposals no later than 2010-12-10.  Notification of accepted speakers will be provided by 2010-12-17.  We will then work to have a schedule finalized by 2011-01-07.  Talks should be submitted by fulling out the following form:

    http://goo.gl/9SWlz

Please direct discussion about this devroom to the telephony-devroom mailing list hosted on http://lists.fosdem.org/.  If you would like to contact the devroom organizer directly, please contact Russell Bryant <russell at digium.com>.

Feel free to forward this along to any people or mailing lists that you think would be interested in this event.

Thank you!

Bryan M. Johns 
Digium, Inc. | Community Director 
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 
direct: +1 256-428-6007 
Check us out at : www.asterisk.org or www.digium.com 



From gabe at gundy.org  Sat Nov 13 12:09:32 2010
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 13 Nov 2010 13:09:32 -0700
Subject: [Freeswitch-dev] How to compile FS debian packages?
In-Reply-To: <AANLkTinCit4AS2nbj0Zxr9HasWnpOJjOW3tThefWzzZL@mail.gmail.com>
References: <AANLkTinCit4AS2nbj0Zxr9HasWnpOJjOW3tThefWzzZL@mail.gmail.com>
Message-ID: <AANLkTino-2iKoKW63EYLq3yxhFM4JRsguxv-1Y=NNCcx@mail.gmail.com>

This is the script that I've been using to build my debs.  I just ran
it on the latest git and it products debs without errors:

################################
#!/bin/bash

sudo apt-get update

sudo apt-get -y install \
autoconf \
automake \
debhelper \
devscripts \
git-core \
libasound2-dev \
libcurl4-openssl-dev \
libdb-dev \
libgdbm-dev \
libgnutls-dev \
libogg-dev \
libperl-dev \
libssl-dev \
libtiff4-dev \
libtool \
libvorbis-dev \
libx11-dev \
ncurses-dev \
python-dev \
unixodbc-dev \
uuid-dev

git clone git://git.freeswitch.org/freeswitch.git freeswitch_git

cd freeswitch_git

debuild -i -us -uc -b

echo "Here are your debs:"
ls -l ../*.deb
################################


Give that a try and let us know what you find.


Gabe

BTW,

gabe at gabe-desktop:~/Downloads$ uname -a
Linux gabe-desktop 2.6.35-22-generic #35-Ubuntu SMP Sat Oct 16
20:36:48 UTC 2010 i686 GNU/Linux

gabe at gabe-desktop:~/Downloads$ cat /etc/issue
Ubuntu 10.10 \n \l



On Sat, Nov 13, 2010 at 6:22 AM, Mathieu Parent <math.parent at gmail.com> wrote:
> Hello,
>
> There was some changes in the debian dir and I am not able to compile
> FS anymore using dpkg-buildpackage.
>
> What is the new correct way to build the package?
>
> Mathieu


From mirceac at ezuce.com  Tue Nov  9 10:16:10 2010
From: mirceac at ezuce.com (Mircea Carasel)
Date: Tue, 9 Nov 2010 20:16:10 +0200
Subject: [Freeswitch-dev] Conference commands through Java API
Message-ID: <AANLkTin+trnbLuxME6i-G-nEeH1dB+L-tJ-V-O8wzbyn@mail.gmail.com>

Hi,

I am sending some conference commands, like : conference list; conference
<name_conf> lock through java client
I am using inbound socket for this and I am listening to response events.
The client is listening until it founds something like
in the response:
"OK"
"Non-Existant"
"Conference" "not found\n"
"-ERR"

The thing is that when the call is successful, I never get "OK" in the
response, and due to this, the java client gets "hanged"

Here are some successful responses that I get using freeswitch CLI:

freeswitch at internal> conference list
No active conferences.

so there is no "OK" message there

or, when I have one active conference:

conference list
Conference mirceaConf (0 members rate: 8000)

the same, no "OK" message

I don't know if there is a problem in the java client (we should never
assume that we get "OK" in the response) or, maybe there is a problem in
freeswitch

Please advice,

Thanks,
Mircea
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From stkn at freeswitch.org  Mon Nov 15 02:49:51 2010
From: stkn at freeswitch.org (Stefan Knoblich)
Date: Mon, 15 Nov 2010 11:49:51 +0100
Subject: [Freeswitch-dev] ftmod_libpri - RDNIS patch
In-Reply-To: <AANLkTik1K9=G-mtqrUP2FR-mYp-H6T_8RL8E8WY+f8zu@mail.gmail.com>
References: <AANLkTik1K9=G-mtqrUP2FR-mYp-H6T_8RL8E8WY+f8zu@mail.gmail.com>
Message-ID: <201011151149.51797.stkn@freeswitch.org>

Am Monday 15 November 2010 schrieb lakshmanan ganapathy:
> Hi all,
> Please find the attached patch file which contain a small change to handle
> RDNIS in ftmod_libpri.
> Somebody can review it and find ok, do commit it.
> 
> 
> regards,
> Lakshmanan G.

reviewed + committed:

http://oss.axsentis.de/gitweb/?p=freeswitch.git;a=commitdiff;h=e98b4a6b8dccbd22e2bcccb04ae992633f7fee3c


thanks,
stkn

-- 
-------------------------------------------------------------------------------
Stefan Knoblich                        | Web:   http://www.axsentis.de/
axsentis GmbH                          |        http://oss.axsentis.de/
Eupener Str. 74, 50933 Koeln, Germany  |
Amtsgericht Koeln: HR B 56238          | Email: s.knoblich at axsentis.de
UST-ID: DE244977565                    | JID:   s.knoblich at jabber.axsentis.de
-------------------------------------------------------------------------------
Web:   http://stkn.techmage.de/
Email: stkn at freeswitch.org
IRC:   #freeswitch-de @ irc.freenode.net


From steveayre at gmail.com  Mon Nov 15 04:14:15 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 15 Nov 2010 12:14:15 +0000
Subject: [Freeswitch-dev] VLAN support in sofia module
In-Reply-To: <AANLkTin0chVQT-RAv-7w7EJb4fwRhsB7MBMorwOR=7Ce@mail.gmail.com>
References: <AANLkTi=9RFk3dfj2ZKgphguNT6S+cuwGh6qiqS7r78Vj@mail.gmail.com>
	<AANLkTiks+SOQZ2Kc+nzXwkwuVaBEAk-Ja1CfmTO9dz-G@mail.gmail.com>
	<AANLkTikkCOu1MvMOEyBXBXcMX+svKNKLyi5Oa=v+fNdF@mail.gmail.com>
	<AANLkTinWgbmE-B7C=2Y4xsvTBD9UzLbpQ62kzc2gksno@mail.gmail.com>
	<AANLkTin0chVQT-RAv-7w7EJb4fwRhsB7MBMorwOR=7Ce@mail.gmail.com>
Message-ID: <AANLkTikNSZTNA+ydyNd1vLAsTRvg3FYY6Ogob5Vos6ga@mail.gmail.com>

Yes, it would be possible to bind signalling to the device, but the
RTP stack would likely get confused since it wouldn't be bound to the
same device. To get RTP to bind to the same device would involve a
pretty major rewrite of the RTP stack I suspect. And bypass media
would simply not be possible.

-Steve


On 10 November 2010 08:54, Steve Kurzeja <steve.kurzeja at gmail.com> wrote:
> I think what the original poster is saying is that his existing architecture
> uses "overlapping" IP addresses ("one ip address for all clients").
> Clients/devices are on separate vlans and the SBC sits in each vlan
> listening on the same IP address. Client <--> SBC IP is then on the same
> broadcast domain and not IP routed. The clients are all configured to talk
> to the same SBC address which makes provisioning simple.
>
> So he would want something like this:
>
> sip profile 1 listening on 10.0.0.1 vlan1
> sip profile 2 listening on 10.0.0.1 vlan2
>
> The likes of ACME packet SBCs/GENBAND etc support this but it is not
> possible in freeswitch currently.
>
> In linux this can be done using the SO_BINDDEVICE socket option to force
> packets out a particular interface rather than use the linux IP routing
> table. A good description is here:
> http://codingrelic.geekhold.com/2009/10/code-snippet-sobindtodevice.html
>
> Regards,
> Steve
>
>
> On Wed, Oct 27, 2010 at 6:22 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> its not possible to listen on 0.0.0.0 you can only guess at best on
>> which ip goes with what packets and you cannot choose which source
>> addr to put in outbound packets in that case. ?This is a limitation
>> caused by the design of the SIP spec.
>>
>>
>> You must create a distinct profile for every interface with the
>> desired IP from each vlan.
>>
>>
>>
>> On Tue, Oct 26, 2010 at 11:49 AM, Steven Ayre <steveayre at gmail.com> wrote:
>> > Thinking about it, it won't work. ext-sip-ip will be inside the
>> > packets but they'll be sent from sip-ip so the clients won't see it
>> > coming from the IP you want.
>> >
>> > On 26 October 2010 17:47, Steven Ayre <steveayre at gmail.com> wrote:
>> >> Possibly your best bet would be to set sip-ip and rtp-ip to the
>> >> specific interface and ext-sip-ip and ext-rtp-ip to the global IP.
>> >>
>> >> I don't know how well this'll work though. Sofia might get very
>> >> confused about where to send stuff to, since the same IP is used on
>> >> essentially separate networks.
>> >>
>> >> -Steve
>> >>
>> >>
>> >> On 23 October 2010 19:47, Adam Ku?mirek <amkusmirek at gmail.com> wrote:
>> >>> Hi All
>> >>>
>> >>> I would like to run FreeSwitch as SBC in my network. I have
>> >>> architecture with all clients connected via separate vlans.
>> >>> It gives me possibility to use one ip address for all clients, but I
>> >>> need FS sofia profile to bind to specific interface (eg eth0:20 vlan
>> >>> 20) and global ip address.
>> >>> As I see, for now sofia module allows to bind to ip address only. I
>> >>> looked quickly to sourcees and it seems that this is limited by sofia
>> >>> library.
>> >>> Of course i may be wrong because I'm new to FS and didn't have time to
>> >>> analyze sources.
>> >>> Please, write some words on this topic.
>> >>> Do you have plans to implement vlan support in FS.
>> >>> I know that AcmePacket Net-Net gives such feature. Don't know any open
>> >>> voip solution with this iplemented.
>> >>>
>> >>> Regards Adam
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-dev mailing list
>> >>> FreeSWITCH-dev at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >
>> > _______________________________________________
>> > FreeSWITCH-dev mailing list
>> > FreeSWITCH-dev at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


From steveayre at gmail.com  Mon Nov 15 04:15:39 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 15 Nov 2010 12:15:39 +0000
Subject: [Freeswitch-dev] VLAN support in sofia module
In-Reply-To: <AANLkTin0chVQT-RAv-7w7EJb4fwRhsB7MBMorwOR=7Ce@mail.gmail.com>
References: <AANLkTi=9RFk3dfj2ZKgphguNT6S+cuwGh6qiqS7r78Vj@mail.gmail.com>
	<AANLkTiks+SOQZ2Kc+nzXwkwuVaBEAk-Ja1CfmTO9dz-G@mail.gmail.com>
	<AANLkTikkCOu1MvMOEyBXBXcMX+svKNKLyi5Oa=v+fNdF@mail.gmail.com>
	<AANLkTinWgbmE-B7C=2Y4xsvTBD9UzLbpQ62kzc2gksno@mail.gmail.com>
	<AANLkTin0chVQT-RAv-7w7EJb4fwRhsB7MBMorwOR=7Ce@mail.gmail.com>
Message-ID: <AANLkTi=6gwTDkLax7w9fWdMHjE8U5f8vUGBnaYzM=8Wu@mail.gmail.com>

Plus the sofia library does not currently support SO_BINDDEVICE so it
would mean patching the library as well as FS.

-Steve


On 10 November 2010 08:54, Steve Kurzeja <steve.kurzeja at gmail.com> wrote:
> I think what the original poster is saying is that his existing architecture
> uses "overlapping" IP addresses ("one ip address for all clients").
> Clients/devices are on separate vlans and the SBC sits in each vlan
> listening on the same IP address. Client <--> SBC IP is then on the same
> broadcast domain and not IP routed. The clients are all configured to talk
> to the same SBC address which makes provisioning simple.
>
> So he would want something like this:
>
> sip profile 1 listening on 10.0.0.1 vlan1
> sip profile 2 listening on 10.0.0.1 vlan2
>
> The likes of ACME packet SBCs/GENBAND etc support this but it is not
> possible in freeswitch currently.
>
> In linux this can be done using the SO_BINDDEVICE socket option to force
> packets out a particular interface rather than use the linux IP routing
> table. A good description is here:
> http://codingrelic.geekhold.com/2009/10/code-snippet-sobindtodevice.html
>
> Regards,
> Steve
>
>
> On Wed, Oct 27, 2010 at 6:22 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> its not possible to listen on 0.0.0.0 you can only guess at best on
>> which ip goes with what packets and you cannot choose which source
>> addr to put in outbound packets in that case. ?This is a limitation
>> caused by the design of the SIP spec.
>>
>>
>> You must create a distinct profile for every interface with the
>> desired IP from each vlan.
>>
>>
>>
>> On Tue, Oct 26, 2010 at 11:49 AM, Steven Ayre <steveayre at gmail.com> wrote:
>> > Thinking about it, it won't work. ext-sip-ip will be inside the
>> > packets but they'll be sent from sip-ip so the clients won't see it
>> > coming from the IP you want.
>> >
>> > On 26 October 2010 17:47, Steven Ayre <steveayre at gmail.com> wrote:
>> >> Possibly your best bet would be to set sip-ip and rtp-ip to the
>> >> specific interface and ext-sip-ip and ext-rtp-ip to the global IP.
>> >>
>> >> I don't know how well this'll work though. Sofia might get very
>> >> confused about where to send stuff to, since the same IP is used on
>> >> essentially separate networks.
>> >>
>> >> -Steve
>> >>
>> >>
>> >> On 23 October 2010 19:47, Adam Ku?mirek <amkusmirek at gmail.com> wrote:
>> >>> Hi All
>> >>>
>> >>> I would like to run FreeSwitch as SBC in my network. I have
>> >>> architecture with all clients connected via separate vlans.
>> >>> It gives me possibility to use one ip address for all clients, but I
>> >>> need FS sofia profile to bind to specific interface (eg eth0:20 vlan
>> >>> 20) and global ip address.
>> >>> As I see, for now sofia module allows to bind to ip address only. I
>> >>> looked quickly to sourcees and it seems that this is limited by sofia
>> >>> library.
>> >>> Of course i may be wrong because I'm new to FS and didn't have time to
>> >>> analyze sources.
>> >>> Please, write some words on this topic.
>> >>> Do you have plans to implement vlan support in FS.
>> >>> I know that AcmePacket Net-Net gives such feature. Don't know any open
>> >>> voip solution with this iplemented.
>> >>>
>> >>> Regards Adam
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-dev mailing list
>> >>> FreeSWITCH-dev at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >
>> > _______________________________________________
>> > FreeSWITCH-dev mailing list
>> > FreeSWITCH-dev at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


From anthony.minessale at gmail.com  Mon Nov 15 07:48:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Nov 2010 09:48:51 -0600
Subject: [Freeswitch-dev] stfu jitterbuffer in FS
In-Reply-To: <92332.76985.qm@web110303.mail.gq1.yahoo.com>
References: <92332.76985.qm@web110303.mail.gq1.yahoo.com>
Message-ID: <AANLkTi=CrX-Rbht-U4cTZRunas2QHJgtD45fox6aOVcT@mail.gmail.com>

try the latest build

On Wed, Oct 27, 2010 at 2:25 PM, Jyotshna Cherukuri
<jcherukuri_necc at yahoo.com> wrote:
> Hi,
> I enabled Jitterbuffer in FS for A leg of the call and when I set up packet
> loss simulation(4%) on my end client ?I expected FS to tag the lost packets
> as SFF_PLC but its not .I rather see the below code being executed . Could
> you please explain what this code does in stfu.c?
> i->miss_count++;
> if (i->miss_count > 10 || (i->in_queue->array_len ==
> i->in_queue->array_size) || tried >= i->in_queue->array_size) {
> i->running = 0;
> i->interval = 0;
> i->out_queue->wr_len = i->out_queue->array_size;
> return NULL;
> }
> Any help is greatly appreciated
> Thanks
> Jyotshna
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From jerry.richards at teotech.com  Mon Nov 15 08:56:51 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 15 Nov 2010 08:56:51 -0800
Subject: [Freeswitch-dev] Trouble Accessing DB Table in C-module Via ODBC
Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D69603EB7F73E5@VA3DIAXVS351.RED001.local>

Hello,

I need to access a database using two different schemas.  During initialization the default schema is set as follows:

set schema 'ucm';

Then later, I need to get data from a table in another schema 'ts_sofia_internal', so I tried the following, but it returns an error as shown:

teo=# select * from sip_presence('ts_sofia_internal') where sip_presence.sip_user='1003';
ERROR:  function sip_presence(unknown) does not exist
LINE 1: select * from sip_presence('ts_sofia_internal') where sip_pr...
                      ^
HINT:  No function matches the given name and argument types. You might need to add explicit type casts.
teo=#

Does anyone know how I can do this?  By the way, I will actually implement the query in C and connect to the DB via ODBC.

Thanks,
Jerry



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From cbeeton at avaya.com  Mon Nov 15 09:27:10 2010
From: cbeeton at avaya.com (Beeton, Carolyn (Carolyn))
Date: Mon, 15 Nov 2010 12:27:10 -0500
Subject: [Freeswitch-dev] Question about presence_id
In-Reply-To: <E151F470685C15418AEA609E825A1710223F6F56D0@DC-US1MBEX4.global.avaya.com>
References: <E151F470685C15418AEA609E825A1710223F6B8C51@DC-US1MBEX4.global.avaya.com>
	<AANLkTi=_gqSJqsdJ=LknBq_tK9mDLro6t6m-LaNw9idf@mail.gmail.com>
	<E151F470685C15418AEA609E825A1710223F6F5042@DC-US1MBEX4.global.avaya.com>
	<AANLkTikbur9-jhEZOtEpNOV3ovNSj_WgwBV6O23K3dmW@mail.gmail.com>
	<E151F470685C15418AEA609E825A1710223F6F55FA@DC-US1MBEX4.global.avaya.com>
	<E151F470685C15418AEA609E825A1710223F6F56D0@DC-US1MBEX4.global.avaya.com>
Message-ID: <E151F470685C15418AEA609E825A1710223F7417B9@DC-US1MBEX4.global.avaya.com>

Now this leads to another question.  If I am monitoring an extension which is running a socket application or playback or something like that,  I can't do this:

<action application="set" data="presence_id=44@$${domain}"/>

because that really "hijacks" the channel from whoever it actually belongs to, who might have a valid presence_id of their own (i.e. might be a registered set which others are monitoring).  (That also might explain why I was seeing the "direction" backwards when monitoring a FS app).

Is the complete solution then to use a bridge application with data="loopback/app=socket...", and set the presence_id of the application extension on the far side of the bridge? i.e. something like this:

<action application="bridge" data="presence_id=44@$${domain};loopback/app=playback..."/>

On a somewhat related note, how does monitoring of conference and park extensions work?  Does the presence_id have to be set in the dialplan, or is it done automatically for these applications?  When a registered user calls into a conference, would PRESENCE_IN events be generated for both ends of the call?

Carolyn

> -----Original Message-----
> From: freeswitch-dev-bounces at lists.freeswitch.org 
> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On 
> Behalf Of Beeton, Carolyn (Carolyn)
> Sent: Thursday, November 11, 2010 3:25 PM
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] Question about presence_id
> 
> Ok - I think I see now.  Changed my dialplan to this:
> 
>    <extension name="47">
>       <condition field="destination_number" expression="^47$">
>          <action application="bridge" 
> data="{presence_id=47@$${domain}}sofia/cbeetonscs.ca.nortel.co
> m/47%cbeetonscs.ca.nortel.com"/>
>       </condition>
>    </extension> 
>    <extension name="49">
>       <condition field="destination_number" expression="^49$">
>          <action application="bridge" 
> data="{presence_id=49@$${domain}}sofia/cbeetonscs.ca.nortel.co
> m/49%cbeetonscs.ca.nortel.com"/>
>       </condition>
>    </extension>
> 
> That sets the presence ID for the outgoing legs properly.  
> The presence ID for the incoming leg is set by default by 
> registration.  And events are generated as expected and 
> lights work as expected.  I just didn't find this bit of info 
> anywhere in the docs!
> 
> Carolyn
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
> 

From anthony.minessale at gmail.com  Mon Nov 15 09:52:26 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Nov 2010 11:52:26 -0600
Subject: [Freeswitch-dev] Question about presence_id
In-Reply-To: <E151F470685C15418AEA609E825A1710223F7417B9@DC-US1MBEX4.global.avaya.com>
References: <E151F470685C15418AEA609E825A1710223F6B8C51@DC-US1MBEX4.global.avaya.com>
	<AANLkTi=_gqSJqsdJ=LknBq_tK9mDLro6t6m-LaNw9idf@mail.gmail.com>
	<E151F470685C15418AEA609E825A1710223F6F5042@DC-US1MBEX4.global.avaya.com>
	<AANLkTikbur9-jhEZOtEpNOV3ovNSj_WgwBV6O23K3dmW@mail.gmail.com>
	<E151F470685C15418AEA609E825A1710223F6F55FA@DC-US1MBEX4.global.avaya.com>
	<E151F470685C15418AEA609E825A1710223F6F56D0@DC-US1MBEX4.global.avaya.com>
	<E151F470685C15418AEA609E825A1710223F7417B9@DC-US1MBEX4.global.avaya.com>
Message-ID: <AANLkTimq3kx0Sw+FcvqcYw7VBK5_WD3bZQXz9mEhx+sm@mail.gmail.com>

the problem is since loopback is not a real channel type there would
not be anything in the sip presence tables for active calls even if
you used it.  You would see the events change from state to state but
you would not have any persistance to pick up on if you started
monitoring when the calls was already up.  You're only choice would be
to override the real presence_id with the var or to add more code to
let one leg have multiple presence identities which may be possible
but also could prove challenging.

The conference uses it's own special distributed namespace inside FS
which is delimited by +



conf+3001 at my.box would allow you to subscribe to conference 3001 at my.box

you set the my.box domain name in the domain field of conference.conf.xml

The default config shows conference 3001 advertised in this manner
from the config.





On Mon, Nov 15, 2010 at 11:27 AM, Beeton, Carolyn (Carolyn)
<cbeeton at avaya.com> wrote:
> Now this leads to another question. ?If I am monitoring an extension which is running a socket application or playback or something like that, ?I can't do this:
>
> <action application="set" data="presence_id=44@$${domain}"/>
>
> because that really "hijacks" the channel from whoever it actually belongs to, who might have a valid presence_id of their own (i.e. might be a registered set which others are monitoring). ?(That also might explain why I was seeing the "direction" backwards when monitoring a FS app).
>
> Is the complete solution then to use a bridge application with data="loopback/app=socket...", and set the presence_id of the application extension on the far side of the bridge? i.e. something like this:
>
> <action application="bridge" data="presence_id=44@$${domain};loopback/app=playback..."/>
>
> On a somewhat related note, how does monitoring of conference and park extensions work? ?Does the presence_id have to be set in the dialplan, or is it done automatically for these applications? ?When a registered user calls into a conference, would PRESENCE_IN events be generated for both ends of the call?
>
> Carolyn
>
>> -----Original Message-----
>> From: freeswitch-dev-bounces at lists.freeswitch.org
>> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On
>> Behalf Of Beeton, Carolyn (Carolyn)
>> Sent: Thursday, November 11, 2010 3:25 PM
>> To: freeswitch-dev at lists.freeswitch.org
>> Subject: Re: [Freeswitch-dev] Question about presence_id
>>
>> Ok - I think I see now. ?Changed my dialplan to this:
>>
>> ? ?<extension name="47">
>> ? ? ? <condition field="destination_number" expression="^47$">
>> ? ? ? ? ?<action application="bridge"
>> data="{presence_id=47@$${domain}}sofia/cbeetonscs.ca.nortel.co
>> m/47%cbeetonscs.ca.nortel.com"/>
>> ? ? ? </condition>
>> ? ?</extension>
>> ? ?<extension name="49">
>> ? ? ? <condition field="destination_number" expression="^49$">
>> ? ? ? ? ?<action application="bridge"
>> data="{presence_id=49@$${domain}}sofia/cbeetonscs.ca.nortel.co
>> m/49%cbeetonscs.ca.nortel.com"/>
>> ? ? ? </condition>
>> ? ?</extension>
>>
>> That sets the presence ID for the outgoing legs properly.
>> The presence ID for the incoming leg is set by default by
>> registration. ?And events are generated as expected and
>> lights work as expected. ?I just didn't find this bit of info
>> anywhere in the docs!
>>
>> Carolyn
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
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FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From jerry.richards at teotech.com  Mon Nov 15 10:05:22 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 15 Nov 2010 10:05:22 -0800
Subject: [Freeswitch-dev] "intercept_unbridged_only=true" Not Working
Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D69603EB7F744B@VA3DIAXVS351.RED001.local>

Hello,

The <action application="set" data="intercept_unbridged_only=true"/> statement does not work for me.  I added it to the <extension name="intercept_ext"> of the default dialplan, but it still unconditionally bridges to a call even after it has already been answered.  Am I missing a tag?

    <extension name="intercept-ext">
      <condition field="destination_number" expression="^\*\*(\d+)$">
      <action application="set" data="intercept_unbridged_only=true"/>
       <action application="answer"/>
       <action application="intercept" data="${hash(select/${domain_name}-last_dial_ext/$1)}"/>
       <action application="sleep" data="2000"/>
      </condition>
    </extension>

Thanks,
Jerry
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From anthony.minessale at gmail.com  Mon Nov 15 10:17:16 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Nov 2010 12:17:16 -0600
Subject: [Freeswitch-dev] "intercept_unbridged_only=true" Not Working
In-Reply-To: <2BF7FB90DF25EA4485949F3AF2B9D69603EB7F744B@VA3DIAXVS351.RED001.local>
References: <2BF7FB90DF25EA4485949F3AF2B9D69603EB7F744B@VA3DIAXVS351.RED001.local>
Message-ID: <AANLkTik8fHbFZ6XaeSyoug0C--+OWX2oz+bxwtdrmhUA@mail.gmail.com>

according to the code in switch_ivr_intercept_session

    if ((var = switch_channel_get_variable(channel,
"intercept_unbridged_only")) && switch_true(var)) {
        if ((switch_channel_test_flag(rchannel, CF_BRIDGED))) {
            switch_core_session_rwunlock(rsession);
            return;
        }
    }

so is your channel not bridged?
its not calls which are not answered its calls that are not bridged.



On Mon, Nov 15, 2010 at 12:05 PM, Jerry Richards
<jerry.richards at teotech.com> wrote:
> Hello,
>
>
>
> The <action application="set" data="intercept_unbridged_only=true"/>
> statement does not work for me.? I added it to the <extension
> name="intercept_ext"> of the default dialplan, but it still unconditionally
> bridges to a call even after it has already been answered.? Am I missing a
> tag?
>
>
>
> ??? <extension name="intercept-ext">
>
> ????? <condition field="destination_number" expression="^\*\*(\d+)$">
>
> ????? <action application="set" data="intercept_unbridged_only=true"/>
>
> ?????? <action application="answer"/>
>
> ?????? <action application="intercept"
> data="${hash(select/${domain_name}-last_dial_ext/$1)}"/>
>
> ?????? <action application="sleep" data="2000"/>
>
> ????? </condition>
>
> ??? </extension>
>
>
>
> Thanks,
>
> Jerry
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From rupa at rupa.com  Mon Nov 15 11:05:49 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 15 Nov 2010 13:05:49 -0600
Subject: [Freeswitch-dev] Trouble Accessing DB Table in C-module Via ODBC
In-Reply-To: <2BF7FB90DF25EA4485949F3AF2B9D69603EB7F73E5@VA3DIAXVS351.RED001.local>
References: <2BF7FB90DF25EA4485949F3AF2B9D69603EB7F73E5@VA3DIAXVS351.RED001.local>
Message-ID: <AANLkTikyGj-0wL-_H=gh3bXxaB1c3++NFuQE2SwdadnS@mail.gmail.com>

What database engine?  That looks like postgresql.  If so, add the schema
you are interested in working with into the search path or explicitly use
the schema in your query.

schema.function(args) rather than just function(args)

You can set the search path using alter user ...

On Mon, Nov 15, 2010 at 10:56 AM, Jerry Richards <jerry.richards at teotech.com
> wrote:

> Hello,
>
>
>
> I need to access a database using two different schemas.  During
> initialization the default schema is set as follows:
>
>
>
> set schema 'ucm';
>
>
>
> Then later, I need to get data from a table in another schema
> 'ts_sofia_internal', so I tried the following, but it returns an error as
> shown:
>
>
>
> teo=# select * from sip_presence('ts_sofia_internal') where
> sip_presence.sip_user='1003';
>
> ERROR:  function sip_presence(unknown) does not exist
>
> LINE 1: select * from sip_presence('ts_sofia_internal') where sip_pr...
>
>                       ^
>
> HINT:  No function matches the given name and argument types. You might
> need to add explicit type casts.
>
> teo=#
>
>
>
> Does anyone know how I can do this?  By the way, I will actually implement
> the query in C and connect to the DB via ODBC.
>
>
>
> Thanks,
>
> Jerry
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From jerry.richards at teotech.com  Mon Nov 15 11:05:54 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 15 Nov 2010 11:05:54 -0800
Subject: [Freeswitch-dev] "intercept_unbridged_only=true" Not Working
In-Reply-To: <AANLkTik8fHbFZ6XaeSyoug0C--+OWX2oz+bxwtdrmhUA@mail.gmail.com>
References: <2BF7FB90DF25EA4485949F3AF2B9D69603EB7F744B@VA3DIAXVS351.RED001.local>
	<AANLkTik8fHbFZ6XaeSyoug0C--+OWX2oz+bxwtdrmhUA@mail.gmail.com>
Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D69603EB7F7497@VA3DIAXVS351.RED001.local>

Okay, I thought bridged and answered were the same thing.  What would I need to do if I want it to only intercept unanswered calls?


-----Original Message-----
From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Monday, November 15, 2010 10:17 AM
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] "intercept_unbridged_only=true" Not Working

according to the code in switch_ivr_intercept_session

    if ((var = switch_channel_get_variable(channel,
"intercept_unbridged_only")) && switch_true(var)) {
        if ((switch_channel_test_flag(rchannel, CF_BRIDGED))) {
            switch_core_session_rwunlock(rsession);
            return;
        }
    }

so is your channel not bridged?
its not calls which are not answered its calls that are not bridged.



On Mon, Nov 15, 2010 at 12:05 PM, Jerry Richards <jerry.richards at teotech.com> wrote:
> Hello,
>
>
>
> The <action application="set" data="intercept_unbridged_only=true"/>
> statement does not work for me.? I added it to the <extension 
> name="intercept_ext"> of the default dialplan, but it still 
> unconditionally bridges to a call even after it has already been 
> answered.? Am I missing a tag?
>
>
>
> ??? <extension name="intercept-ext">
>
> ????? <condition field="destination_number" expression="^\*\*(\d+)$">
>
> ????? <action application="set" data="intercept_unbridged_only=true"/>
>
> ?????? <action application="answer"/>
>
> ?????? <action application="intercept"
> data="${hash(select/${domain_name}-last_dial_ext/$1)}"/>
>
> ?????? <action application="sleep" data="2000"/>
>
> ????? </condition>
>
> ??? </extension>
>
>
>
> Thanks,
>
> Jerry
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>



--
Anthony Minessale II

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

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

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

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




From anthony.minessale at gmail.com  Mon Nov 15 11:59:32 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Nov 2010 13:59:32 -0600
Subject: [Freeswitch-dev] "intercept_unbridged_only=true" Not Working
In-Reply-To: <2BF7FB90DF25EA4485949F3AF2B9D69603EB7F7497@VA3DIAXVS351.RED001.local>
References: <2BF7FB90DF25EA4485949F3AF2B9D69603EB7F744B@VA3DIAXVS351.RED001.local>
	<AANLkTik8fHbFZ6XaeSyoug0C--+OWX2oz+bxwtdrmhUA@mail.gmail.com>
	<2BF7FB90DF25EA4485949F3AF2B9D69603EB7F7497@VA3DIAXVS351.RED001.local>
Message-ID: <AANLkTi==HhdocU1vMVKvU_WWGSVunikRQeOcHUwp-_t5@mail.gmail.com>

commit 36ba0f24625291c85cd1a82e295238ad05fbdf67
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Mon Nov 15 13:51:27 2010 -0600

    add intercept_unanswered_only var akin to intercept_unbridged_only

try this patch
if you could document it on the wiki i would appreciate it.


On Mon, Nov 15, 2010 at 1:05 PM, Jerry Richards
<jerry.richards at teotech.com> wrote:
> Okay, I thought bridged and answered were the same thing. ?What would I need to do if I want it to only intercept unanswered calls?
>
>
> -----Original Message-----
> From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
> Sent: Monday, November 15, 2010 10:17 AM
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] "intercept_unbridged_only=true" Not Working
>
> according to the code in switch_ivr_intercept_session
>
> ? ?if ((var = switch_channel_get_variable(channel,
> "intercept_unbridged_only")) && switch_true(var)) {
> ? ? ? ?if ((switch_channel_test_flag(rchannel, CF_BRIDGED))) {
> ? ? ? ? ? ?switch_core_session_rwunlock(rsession);
> ? ? ? ? ? ?return;
> ? ? ? ?}
> ? ?}
>
> so is your channel not bridged?
> its not calls which are not answered its calls that are not bridged.
>
>
>
> On Mon, Nov 15, 2010 at 12:05 PM, Jerry Richards <jerry.richards at teotech.com> wrote:
>> Hello,
>>
>>
>>
>> The <action application="set" data="intercept_unbridged_only=true"/>
>> statement does not work for me.? I added it to the <extension
>> name="intercept_ext"> of the default dialplan, but it still
>> unconditionally bridges to a call even after it has already been
>> answered.? Am I missing a tag?
>>
>>
>>
>> ??? <extension name="intercept-ext">
>>
>> ????? <condition field="destination_number" expression="^\*\*(\d+)$">
>>
>> ????? <action application="set" data="intercept_unbridged_only=true"/>
>>
>> ?????? <action application="answer"/>
>>
>> ?????? <action application="intercept"
>> data="${hash(select/${domain_name}-last_dial_ext/$1)}"/>
>>
>> ?????? <action application="sleep" data="2000"/>
>>
>> ????? </condition>
>>
>> ??? </extension>
>>
>>
>>
>> Thanks,
>>
>> Jerry
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From jerry.richards at teotech.com  Mon Nov 15 12:01:14 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 15 Nov 2010 12:01:14 -0800
Subject: [Freeswitch-dev] Trouble Accessing DB Table in C-module Via ODBC
In-Reply-To: <AANLkTikyGj-0wL-_H=gh3bXxaB1c3++NFuQE2SwdadnS@mail.gmail.com>
References: <2BF7FB90DF25EA4485949F3AF2B9D69603EB7F73E5@VA3DIAXVS351.RED001.local>
	<AANLkTikyGj-0wL-_H=gh3bXxaB1c3++NFuQE2SwdadnS@mail.gmail.com>
Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D69603EB7F74FF@VA3DIAXVS351.RED001.local>

Hi Rupa,

Yes, I'm using postgresql.  When I run the C-module using the schema.table format it gets this error:

2010-11-15 11:11:18.431977 [ERR] switch_core_sqldb.c:806 SQL ERR: [select sip_presence.status, sip_presence.rpid from ts_sofia_internal.sip_presence where sip_presence.sip_user='1003'] no such table: ts_sofia_internal.sip_presence

If I execute the select statement from the psql command line, it works as shown:

teo=# select sip_presence.status, sip_presence.rpid from ts_sofia_internal.sip_presence where sip_presence.sip_user='1003';
status | rpid
--------+------
Away   | away
(1 row)

Could this have something to do with accessing the DB via ODBC?  Do you have another suggestion?

I did try adding 'ts_sofia_internal' to the search path (i.e. set search_path to ucm, ts_sofia_internal), but it also did not work.

Thanks,
Jerry



From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker
Sent: Monday, November 15, 2010 11:06 AM
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] Trouble Accessing DB Table in C-module Via ODBC

What database engine?  That looks like postgresql.  If so, add the schema you are interested in working with into the search path or explicitly use the schema in your query.

schema.function(args) rather than just function(args)

You can set the search path using alter user ...
On Mon, Nov 15, 2010 at 10:56 AM, Jerry Richards <jerry.richards at teotech.com<mailto:jerry.richards at teotech.com>> wrote:
Hello,

I need to access a database using two different schemas.  During initialization the default schema is set as follows:

set schema 'ucm';

Then later, I need to get data from a table in another schema 'ts_sofia_internal', so I tried the following, but it returns an error as shown:

teo=# select * from sip_presence('ts_sofia_internal') where sip_presence.sip_user='1003';
ERROR:  function sip_presence(unknown) does not exist
LINE 1: select * from sip_presence('ts_sofia_internal') where sip_pr...
                      ^
HINT:  No function matches the given name and argument types. You might need to add explicit type casts.
teo=#

Does anyone know how I can do this?  By the way, I will actually implement the query in C and connect to the DB via ODBC.

Thanks,
Jerry




_______________________________________________
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
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--
-Rupa
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From msc at freeswitch.org  Mon Nov 15 12:13:26 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Nov 2010 12:13:26 -0800
Subject: [Freeswitch-dev] "intercept_unbridged_only=true" Not Working
In-Reply-To: <AANLkTi==HhdocU1vMVKvU_WWGSVunikRQeOcHUwp-_t5@mail.gmail.com>
References: <2BF7FB90DF25EA4485949F3AF2B9D69603EB7F744B@VA3DIAXVS351.RED001.local>
	<AANLkTik8fHbFZ6XaeSyoug0C--+OWX2oz+bxwtdrmhUA@mail.gmail.com>
	<2BF7FB90DF25EA4485949F3AF2B9D69603EB7F7497@VA3DIAXVS351.RED001.local>
	<AANLkTi==HhdocU1vMVKvU_WWGSVunikRQeOcHUwp-_t5@mail.gmail.com>
Message-ID: <AANLkTikx3fkJVC4d6GuWu9xK-R-DqwRfWdE4VDBCDjqh@mail.gmail.com>

On Mon, Nov 15, 2010 at 11:59 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> commit 36ba0f24625291c85cd1a82e295238ad05fbdf67
> Author: Anthony Minessale <anthm at freeswitch.org>
> Date:   Mon Nov 15 13:51:27 2010 -0600
>
>    add intercept_unanswered_only var akin to intercept_unbridged_only
>
> try this patch
> if you could document it on the wiki i would appreciate it.
>

FYI I created the necessary links, stub, etc. so that all you have to do is
confirm that the information on this page is correct:

http://wiki.freeswitch.org/wiki/Variable_intercept_unanswered_only

Also, if there are any other caveats please add them to this page.
-MC
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From kevin.snow at ooma.com  Mon Nov 15 14:15:25 2010
From: kevin.snow at ooma.com (Kevin Snow)
Date: Mon, 15 Nov 2010 14:15:25 -0800
Subject: [Freeswitch-dev] Repeat 180 RINGING messages
Message-ID: <C906F0FD.ACFA%kevin.snow@ooma.com>


I?d like to be able to make 180 RINGING messages be repeated to the caller
every X seconds until the 200 OK (or other error). I dug through the source
& FreeSWITCH book but didn?t see a mechanism for doing this. Does FS have
such a thing?

Thanks

Kevin

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From anthony.minessale at gmail.com  Mon Nov 15 14:25:14 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Nov 2010 16:25:14 -0600
Subject: [Freeswitch-dev] Repeat 180 RINGING messages
In-Reply-To: <C906F0FD.ACFA%kevin.snow@ooma.com>
References: <C906F0FD.ACFA%kevin.snow@ooma.com>
Message-ID: <AANLkTin7mS=O7ovoKwnGJraA4Eu0d2_j3_EVh2H5KjZ6@mail.gmail.com>

We don't support that. Its completely non standard.
you could however send 183 early media and play the ringing inband.


On Mon, Nov 15, 2010 at 4:15 PM, Kevin Snow <kevin.snow at ooma.com> wrote:
>
> I?d like to be able to make 180 RINGING messages be repeated to the caller
> every X seconds until the 200 OK (or other error). I dug through the source
> & FreeSWITCH book but didn?t see a mechanism for doing this. Does FS have
> such a thing?
>
> Thanks
>
> Kevin
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From steveayre at gmail.com  Mon Nov 15 14:29:52 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 15 Nov 2010 22:29:52 +0000
Subject: [Freeswitch-dev] Repeat 180 RINGING messages
In-Reply-To: <C906F0FD.ACFA%kevin.snow@ooma.com>
References: <C906F0FD.ACFA%kevin.snow@ooma.com>
Message-ID: <3569839F-448D-4291-8070-96647A8D3902@gmail.com>

You might be looking for PRACK. That resends the 18x until an ACK for it is received, thereby ensuring its delivered. There is a sofia profile parameter that enables it, but it used to have warnings that it was unstable.

You could also use the tcp transport, which would guarantee the 180 was received.

Steve on iPhone



On 15 Nov 2010, at 22:15, Kevin Snow <kevin.snow at ooma.com> wrote:

> 
> I?d like to be able to make 180 RINGING messages be repeated to the caller every X seconds until the 200 OK (or other error). I dug through the source & FreeSWITCH book but didn?t see a mechanism for doing this. Does FS have such a thing?
> 
> Thanks
> 
> Kevin
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
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From anthony.minessale at gmail.com  Mon Nov 15 14:38:39 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Nov 2010 16:38:39 -0600
Subject: [Freeswitch-dev] Repeat 180 RINGING messages
In-Reply-To: <3569839F-448D-4291-8070-96647A8D3902@gmail.com>
References: <C906F0FD.ACFA%kevin.snow@ooma.com>
	<3569839F-448D-4291-8070-96647A8D3902@gmail.com>
Message-ID: <AANLkTikewfGUyZkr_bxe=hDFiM3C1sufVhoNfXrYsViP@mail.gmail.com>

he does not want to guarantee it's received. he wants to keep sending
them because some broken device only rings once per 180 not forever
until it's answered like the spec says.



On Mon, Nov 15, 2010 at 4:29 PM, Steven Ayre <steveayre at gmail.com> wrote:
> You might be looking for PRACK. That resends the 18x until an ACK for it is
> received, thereby ensuring its delivered. There is a sofia profile parameter
> that enables it, but it used to have warnings that it was unstable.
>
> You could also use the tcp transport, which would guarantee the 180 was
> received.
> Steve on iPhone
>
>
> On 15 Nov 2010, at 22:15, Kevin Snow <kevin.snow at ooma.com> wrote:
>
>
> I?d like to be able to make 180 RINGING messages be repeated to the caller
> every X seconds until the 200 OK (or other error). I dug through the source
> & FreeSWITCH book but didn?t see a mechanism for doing this. Does FS have
> such a thing?
>
> Thanks
>
> Kevin
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From jerry.richards at teotech.com  Mon Nov 15 14:41:05 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 15 Nov 2010 14:41:05 -0800
Subject: [Freeswitch-dev] "intercept_unbridged_only=true" Not Working
In-Reply-To: <AANLkTikx3fkJVC4d6GuWu9xK-R-DqwRfWdE4VDBCDjqh@mail.gmail.com>
References: <2BF7FB90DF25EA4485949F3AF2B9D69603EB7F744B@VA3DIAXVS351.RED001.local>
	<AANLkTik8fHbFZ6XaeSyoug0C--+OWX2oz+bxwtdrmhUA@mail.gmail.com>
	<2BF7FB90DF25EA4485949F3AF2B9D69603EB7F7497@VA3DIAXVS351.RED001.local>
	<AANLkTi==HhdocU1vMVKvU_WWGSVunikRQeOcHUwp-_t5@mail.gmail.com>
	<AANLkTikx3fkJVC4d6GuWu9xK-R-DqwRfWdE4VDBCDjqh@mail.gmail.com>
Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D69603EB7F75F3@VA3DIAXVS351.RED001.local>

Okay, maybe something is wrong with my configuration?  FS is connecting the call when I dial "**<extension>" from another phone after the call was answered at extension.  I posted a trace to the pastebin: http://pastebin.freeswitch.org/14499.

CONFIGURATION:
Freeswitch Server:  FreeSWITCH Version 1.0.head (git-97c65a0 2010-11-15 12-22-09 -0600))
Hardphone Ext 1003
Hardphone Ext 1004
Hardphone Ext 1005
conf/dialplan/default.xml:
    <extension name="intercept-ext">
      <condition field="destination_number" expression="^\*\*(\d+)$">
      <action application="set" data="intercept_unanswered_only=true"/>
       <action application="answer"/>
       <action application="intercept" data="${hash(select/${domain_name}-last_dial_ext/$1)}"/>
       <action application="sleep" data="2000"/>
      </condition>
    </extension>

SCENARIO:
1) 1004 calls 1003
2) 1003 answers call
3) 1005 calls **1003
4) Freeswitch takes call from 1003 and establishes talk path between 1004 and 1005

Thanks,
Jerry



From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Monday, November 15, 2010 12:13 PM
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] "intercept_unbridged_only=true" Not Working

On Mon, Nov 15, 2010 at 11:59 AM, Anthony Minessale <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>> wrote:
commit 36ba0f24625291c85cd1a82e295238ad05fbdf67
Author: Anthony Minessale <anthm at freeswitch.org<mailto:anthm at freeswitch.org>>
Date:   Mon Nov 15 13:51:27 2010 -0600

   add intercept_unanswered_only var akin to intercept_unbridged_only

try this patch
if you could document it on the wiki i would appreciate it.

FYI I created the necessary links, stub, etc. so that all you have to do is confirm that the information on this page is correct:

http://wiki.freeswitch.org/wiki/Variable_intercept_unanswered_only

Also, if there are any other caveats please add them to this page.
-MC
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From mrene_lists at avgs.ca  Mon Nov 15 17:01:39 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 15 Nov 2010 20:01:39 -0500
Subject: [Freeswitch-dev] Repeat 180 RINGING messages
In-Reply-To: <C906F0FD.ACFA%kevin.snow@ooma.com>
References: <C906F0FD.ACFA%kevin.snow@ooma.com>
Message-ID: <50E1EAEC-12E1-4679-9439-000696A0CDA6@avgs.ca>

You would need to hack libsofiasip in the transport layer (nta). FS only deals with the messages themselves, the library takes care of retransmissions.

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




On 2010-11-15, at 5:15 PM, Kevin Snow wrote:

> 
> I?d like to be able to make 180 RINGING messages be repeated to the caller every X seconds until the 200 OK (or other error). I dug through the source & FreeSWITCH book but didn?t see a mechanism for doing this. Does FS have such a thing?
> 
> Thanks
> 
> Kevin
> _______________________________________________
> 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 David.Brazier at 360crm.co.uk  Tue Nov 16 01:36:39 2010
From: David.Brazier at 360crm.co.uk (David Brazier)
Date: Tue, 16 Nov 2010 01:36:39 -0800
Subject: [Freeswitch-dev] Verbose Events
Message-ID: <C948D46A3126C344A5EF84C5FD1D49464E0B256D68@IE2RD2XVS021.red002.local>

Hi All

I want to get session variables on all events.  I have tried using the "verbose-events" dialplan app but that isn't convenient to use when bridging.  So I have set 

 <param name="verbose-channel-events" value="yes"/>

in switch.conf.xml & restarted to make sure.  But I still don't see variables on all events, for example CHANNEL_CALLSTATE.  In the source code, there are 3 related flags:

CF_VERBOSE_EVENTS - the "channel flag" set by the "verbose-events" dialpan app
SCF_VERBOSE_EVENTS - the "switch control flag", set by "verbose-channel-events" in switch.conf.xml
SCSC_VERBOSE_EVENTS - used by "fsctl verbose_events" - actually SCSC_VERBOSE_EVENTS is only an internal name for the command - using "fsctl verbose_events" sets or clears SCF_VERBOSE_EVENTS

In switch_core.c the SCSC_ flags are accessed via switch_core_session_ctl: 


SWITCH_DECLARE(int32_t) switch_core_session_ctl(switch_session_ctl_t cmd, void *val)
{
        int *intval = (int *) val;
        int oldintval = 0, newintval = 0;

        if (intval) {
                oldintval = *intval;
        }

        if (switch_test_flag((&runtime), SCF_SHUTTING_DOWN)) {
                return -1;
        }

        switch (cmd) {
        case SCSC_VERBOSE_EVENTS:
                if (intval) {
                        if (oldintval > -1) {
                                if (oldintval) {
                                        switch_set_flag((&runtime), SCF_VERBOSE_EVENTS);
                                } else {
                                        switch_clear_flag((&runtime), SCF_VERBOSE_EVENTS);
                                }
                        }
                        newintval = switch_test_flag((&runtime), SCF_VERBOSE_EVENTS);
                }
                break;
        ...
        }

        if (intval) {
                *intval = newintval;
        }


        return 0;
}

If I read this right, the caller can use this to just get the flag without setting it by setting val to -1 initially.  However, in switch_channel.c where it is deciding whether to do a verbose event:

SWITCH_DECLARE(void) switch_channel_event_set_extended_data(switch_channel_t *channel, switch_event_t *event)
{
        switch_event_header_t *hi;
        int x, global_verbose_events = 0;

        switch_mutex_lock(channel->profile_mutex);

        switch_core_session_ctl(SCSC_VERBOSE_EVENTS, &global_verbose_events);

        if (global_verbose_events ||
                switch_channel_test_flag(channel, CF_VERBOSE_EVENTS) ||
                switch_event_get_header(event, "presence-data-cols") ||
                event->event_id == SWITCH_EVENT_CHANNEL_CREATE ||
                event->event_id == SWITCH_EVENT_CHANNEL_ORIGINATE || ...)
        // then add the variables to the event

Because this does "global_verbose_events = 0" it always clears SCF_VERBOSE_EVENTS.

I could easily have misuderstood, and if anyone else has had success with "verbose-channel-events" please let me know.

Cheers

David


From David.Brazier at 360crm.co.uk  Tue Nov 16 02:37:34 2010
From: David.Brazier at 360crm.co.uk (David Brazier)
Date: Tue, 16 Nov 2010 02:37:34 -0800
Subject: [Freeswitch-dev] Verbose Events
In-Reply-To: <C948D46A3126C344A5EF84C5FD1D49464E0B256D68@IE2RD2XVS021.red002.local>
References: <C948D46A3126C344A5EF84C5FD1D49464E0B256D68@IE2RD2XVS021.red002.local>
Message-ID: <C948D46A3126C344A5EF84C5FD1D49464E0B66E71C@IE2RD2XVS021.red002.local>

[same as last message, better formatting!]

Hi All

I want to get session variables on all events.  I have tried using the "verbose-events" dialplan app but that isn't convenient to use when bridging.  So I have set 

 <param name="verbose-channel-events" value="yes"/>

in switch.conf.xml & restarted to make sure.  But I still don't see variables on all events, for example CHANNEL_CALLSTATE.  In the source code, there are 3 related flags:

1.  CF_VERBOSE_EVENTS - the "channel flag" set by the "verbose-events" dialpan app 

2.  SCF_VERBOSE_EVENTS - the "switch control flag", set by "verbose-channel-events" in switch.conf.xml 

3.  SCSC_VERBOSE_EVENTS - used by "fsctl verbose_events" - actually SCSC_VERBOSE_EVENTS is only an internal name for the command - using "fsctl verbose_events" sets or clears SCF_VERBOSE_EVENTS

In switch_core.c the SCSC_ flags are accessed via switch_core_session_ctl: 

SWITCH_DECLARE(int32_t) switch_core_session_ctl(switch_session_ctl_t cmd, void *val) {
        int *intval = (int *) val;
        int oldintval = 0, newintval = 0;

        if (intval) {
                oldintval = *intval;
        }

        if (switch_test_flag((&runtime), SCF_SHUTTING_DOWN)) {
                return -1;
        }

        switch (cmd) {
        case SCSC_VERBOSE_EVENTS:
                if (intval) {
                        if (oldintval > -1) {
                                if (oldintval) {
                                        switch_set_flag((&runtime), SCF_VERBOSE_EVENTS);
                                } else {
                                        switch_clear_flag((&runtime), SCF_VERBOSE_EVENTS);
                                }
                        }
                        newintval = switch_test_flag((&runtime), SCF_VERBOSE_EVENTS);
                }
                break;
        ...
        }

        if (intval) {
                *intval = newintval;
        }


        return 0;
}

If I read this right, the caller can use this to just get the flag without setting it by setting val to -1 initially.  However, in switch_channel.c where it is deciding whether to do a verbose event:

SWITCH_DECLARE(void) switch_channel_event_set_extended_data(switch_channel_t *channel, switch_event_t *event) {
        switch_event_header_t *hi;
        int x, global_verbose_events = 0;

        switch_mutex_lock(channel->profile_mutex);

        switch_core_session_ctl(SCSC_VERBOSE_EVENTS, &global_verbose_events);

        if (global_verbose_events ||
                switch_channel_test_flag(channel, CF_VERBOSE_EVENTS) ||
                switch_event_get_header(event, "presence-data-cols") ||
                event->event_id == SWITCH_EVENT_CHANNEL_CREATE ||
                event->event_id == SWITCH_EVENT_CHANNEL_ORIGINATE || ...)
        // then add the variables to the event

Because this does "global_verbose_events = 0" it always clears SCF_VERBOSE_EVENTS.

I could easily have misuderstood, and if anyone else has had success with "verbose-channel-events" please let me know.

Cheers

David


From anthony.minessale at gmail.com  Tue Nov 16 08:00:03 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Nov 2010 10:00:03 -0600
Subject: [Freeswitch-dev] "intercept_unbridged_only=true" Not Working
In-Reply-To: <2BF7FB90DF25EA4485949F3AF2B9D69603EB7F75F3@VA3DIAXVS351.RED001.local>
References: <2BF7FB90DF25EA4485949F3AF2B9D69603EB7F744B@VA3DIAXVS351.RED001.local>
	<AANLkTik8fHbFZ6XaeSyoug0C--+OWX2oz+bxwtdrmhUA@mail.gmail.com>
	<2BF7FB90DF25EA4485949F3AF2B9D69603EB7F7497@VA3DIAXVS351.RED001.local>
	<AANLkTi==HhdocU1vMVKvU_WWGSVunikRQeOcHUwp-_t5@mail.gmail.com>
	<AANLkTikx3fkJVC4d6GuWu9xK-R-DqwRfWdE4VDBCDjqh@mail.gmail.com>
	<2BF7FB90DF25EA4485949F3AF2B9D69603EB7F75F3@VA3DIAXVS351.RED001.local>
Message-ID: <AANLkTi=JPMGCO3to6ORY2DSjkZQMb0SCY6CXH6XfE9Be@mail.gmail.com>

I think the patch didn't push confirm you have the revision i posted.


On Mon, Nov 15, 2010 at 4:41 PM, Jerry Richards
<jerry.richards at teotech.com> wrote:
> Okay, maybe something is wrong with my configuration?? FS is connecting the
> call when I dial "**<extension>" from another phone after the call was
> answered at extension.? I posted a trace to the pastebin:
> http://pastebin.freeswitch.org/14499.
>
>
>
> CONFIGURATION:
>
> Freeswitch Server:? FreeSWITCH Version 1.0.head (git-97c65a0 2010-11-15
> 12-22-09 -0600))
>
> Hardphone Ext 1003
>
> Hardphone Ext 1004
>
> Hardphone Ext 1005
>
> conf/dialplan/default.xml:
>
> ??? <extension name="intercept-ext">
>
> ????? <condition field="destination_number" expression="^\*\*(\d+)$">
>
> ????? <action application="set" data="intercept_unanswered_only=true"/>
>
> ?????? <action application="answer"/>
>
> ?????? <action application="intercept"
> data="${hash(select/${domain_name}-last_dial_ext/$1)}"/>
>
> ?????? <action application="sleep" data="2000"/>
>
> ????? </condition>
>
> ??? </extension>
>
>
>
> SCENARIO:
>
> 1) 1004 calls 1003
>
> 2) 1003 answers call
>
> 3) 1005 calls **1003
>
> 4) Freeswitch takes call from 1003 and establishes talk path between 1004
> and 1005
>
>
>
> Thanks,
>
> Jerry
>
>
>
>
>
>
>
> From: freeswitch-dev-bounces at lists.freeswitch.org
> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael
> Collins
> Sent: Monday, November 15, 2010 12:13 PM
>
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] "intercept_unbridged_only=true" Not Working
>
>
>
> On Mon, Nov 15, 2010 at 11:59 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>
> commit 36ba0f24625291c85cd1a82e295238ad05fbdf67
> Author: Anthony Minessale <anthm at freeswitch.org>
> Date: ? Mon Nov 15 13:51:27 2010 -0600
>
> ? ?add intercept_unanswered_only var akin to intercept_unbridged_only
>
> try this patch
> if you could document it on the wiki i would appreciate it.
>
>
>
> FYI I created the necessary links, stub, etc. so that all you have to do is
> confirm that the information on this page is correct:
>
>
>
> http://wiki.freeswitch.org/wiki/Variable_intercept_unanswered_only
>
>
>
> Also, if there are any other caveats please add them to this page.
>
> -MC
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From math.parent at gmail.com  Tue Nov 16 13:27:06 2010
From: math.parent at gmail.com (Mathieu Parent)
Date: Tue, 16 Nov 2010 22:27:06 +0100
Subject: [Freeswitch-dev] How to compile FS debian packages?
In-Reply-To: <AANLkTino-2iKoKW63EYLq3yxhFM4JRsguxv-1Y=NNCcx@mail.gmail.com>
References: <AANLkTinCit4AS2nbj0Zxr9HasWnpOJjOW3tThefWzzZL@mail.gmail.com>
	<AANLkTino-2iKoKW63EYLq3yxhFM4JRsguxv-1Y=NNCcx@mail.gmail.com>
Message-ID: <AANLkTinY52rr29nwQkSjuV1mdrEHhLyMEQBRaZ+s7+hf@mail.gmail.com>

2010/11/13 Gabriel Gunderson <gabe at gundy.org>:
> This is the script that I've been using to build my debs. ?I just ran
> it on the latest git and it products debs without errors:

Thanks. I will test and report back once I have some time.
(this can take some days)


Mathieu


From lakindia89 at gmail.com  Tue Nov 16 20:49:47 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Wed, 17 Nov 2010 10:19:47 +0530
Subject: [Freeswitch-dev] ftmod_libpri - Channel Selection patch
Message-ID: <AANLkTikgFD4rSU3cL-8dFfwxFtqfBB7muUaxzvqXFRHW@mail.gmail.com>

Hi all,
When using ftmod_libpri, if a incoming call comes without "Channel
Identification" ( in SETUP packet ), then the call didn't get proceeded.
I've taken some code snippets from ftmod_isdn and I applied it in
ftmod_libpri. I've tested it and in my setup it works fine.

I've attached the patch here. Please verify it and found ok commit it.

regards,
Lakshmanan G.
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From moises.silva at gmail.com  Tue Nov 16 21:13:52 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 16 Nov 2010 23:13:52 -0600
Subject: [Freeswitch-dev] ftmod_libpri - Channel Selection patch
In-Reply-To: <AANLkTikgFD4rSU3cL-8dFfwxFtqfBB7muUaxzvqXFRHW@mail.gmail.com>
References: <AANLkTikgFD4rSU3cL-8dFfwxFtqfBB7muUaxzvqXFRHW@mail.gmail.com>
Message-ID: <AANLkTin7oT6RqNoEEtwEir3Nqn1YA0g520_OhqEZ5hZO@mail.gmail.com>

Please put this in Jira (jira.freeswitch.org)

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


On Tue, Nov 16, 2010 at 10:49 PM, lakshmanan ganapathy <lakindia89 at gmail.com
> wrote:

> Hi all,
> When using ftmod_libpri, if a incoming call comes without "Channel
> Identification" ( in SETUP packet ), then the call didn't get proceeded.
> I've taken some code snippets from ftmod_isdn and I applied it in
> ftmod_libpri. I've tested it and in my setup it works fine.
>
> I've attached the patch here. Please verify it and found ok commit it.
>
> regards,
> Lakshmanan G.
>
>
> _______________________________________________
> 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 lakindia89 at gmail.com  Wed Nov 17 03:04:41 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Wed, 17 Nov 2010 16:34:41 +0530
Subject: [Freeswitch-dev] ftmod_libpri - Channel Selection patch
In-Reply-To: <AANLkTin7oT6RqNoEEtwEir3Nqn1YA0g520_OhqEZ5hZO@mail.gmail.com>
References: <AANLkTikgFD4rSU3cL-8dFfwxFtqfBB7muUaxzvqXFRHW@mail.gmail.com>
	<AANLkTin7oT6RqNoEEtwEir3Nqn1YA0g520_OhqEZ5hZO@mail.gmail.com>
Message-ID: <AANLkTi=oY32-AHvmBFXrMkfwff3yMx85UmZv9Cyh+iRF@mail.gmail.com>

I've posted a bug and given the patch as resolution.

http://jira.freeswitch.org/browse/OPENZAP-118

On Wed, Nov 17, 2010 at 10:43 AM, Moises Silva <moises.silva at gmail.com>wrote:

> Please put this in Jira (jira.freeswitch.org)
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R
> 9R6 Canada
> t. 1 905 474 1990 x128 | e. moy at sangoma.com
>
>
> On Tue, Nov 16, 2010 at 10:49 PM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> Hi all,
>> When using ftmod_libpri, if a incoming call comes without "Channel
>> Identification" ( in SETUP packet ), then the call didn't get proceeded.
>> I've taken some code snippets from ftmod_isdn and I applied it in
>> ftmod_libpri. I've tested it and in my setup it works fine.
>>
>> I've attached the patch here. Please verify it and found ok commit it.
>>
>> regards,
>> Lakshmanan G.
>>
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From david.varnes at gmail.com  Wed Nov 17 03:07:46 2010
From: david.varnes at gmail.com (david varnes)
Date: Wed, 17 Nov 2010 22:07:46 +1100
Subject: [Freeswitch-dev] Conference commands through Java API
In-Reply-To: <AANLkTin+trnbLuxME6i-G-nEeH1dB+L-tJ-V-O8wzbyn@mail.gmail.com>
References: <AANLkTin+trnbLuxME6i-G-nEeH1dB+L-tJ-V-O8wzbyn@mail.gmail.com>
Message-ID: <AANLkTi=24gZ5FRp4y-edGhmazDYMN9zSFJxb0L8m-cWN@mail.gmail.com>

Hi Mircea,

Which java 'api' are you using ?  [1]

I have never used the conference commands, but I just did a
quick test using the Java ESL Client [2] and sent the
'conference list' command, and got back a response
object with the 'No active conferences' content you
describe below, ie it seemed to work just fine.

Hope this helps?

davidv

[1] http://wiki.freeswitch.org/wiki/Java_ESL
[2] http://wiki.freeswitch.org/wiki/Java_ESL_Client


On 10 November 2010 05:16, Mircea Carasel <mirceac at ezuce.com> wrote:
> Hi,
>
> I am sending some conference commands, like : conference list; conference
> <name_conf> lock through java client
> I am using inbound socket for this and I am listening to response events.
> The client is listening until it founds something like
> in the response:
> "OK"
> "Non-Existant"
> "Conference" "not found\n"
> "-ERR"
>
> The thing is that when the call is successful, I never get "OK" in the
> response, and due to this, the java client gets "hanged"
>
> Here are some successful responses that I get using freeswitch CLI:
>
> freeswitch at internal> conference list
> No active conferences.
>
> so there is no "OK" message there
>
> or, when I have one active conference:
>
> conference list
> Conference mirceaConf (0 members rate: 8000)
>
> the same, no "OK" message
>
> I don't know if there is a problem in the java client (we should never
> assume that we get "OK" in the response) or, maybe there is a problem in
> freeswitch
>
> Please advice,
>
> Thanks,
> Mircea
>



-- 
david varnes


From rentmycoder at gmail.com  Wed Nov 17 08:14:17 2010
From: rentmycoder at gmail.com (rentmycoder rentmycoder)
Date: Wed, 17 Nov 2010 17:14:17 +0100
Subject: [Freeswitch-dev] compiling on Amazon EC2
Message-ID: <AANLkTi=pbt0tYGbtPzrqpqwxSQT6vhvPXGcPwMgjqM9x@mail.gmail.com>

Hi all,

I have a problem installing FS on amazon EC2 micro instance (actually
free) using SuseLinux Enterprise Server 11SP1 x86.
Only this is the supported free trial OS, so I can't use another distro...

Freeswitch compiles, but make install hangs at:
ldconfig -n /usr/local/freeswitch/lib


I've tried 1.0.4, 1.0.6, current, all have the same problem...
I've stripped down modules.conf before configure, same problem..

Any hint?
Are there any precompiled packages for  SuseLinux Enterprise Server 11SP1 x86?


 +---------- FreeSWITCH Build Complete ----------+
 + FreeSWITCH has been successfully built.       +
 + Install by running:                           +
 +                                               +
 +                make install                   +
 +-----------------------------------------------+
ip-10-234-215-206:/usr/src/cc/freeswitch-1.0.6 # make install
make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#"
/usr/src/cc/freeswitch-1.0.6/modules.conf | sed -e "s|^.*/||" | sort |
uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo
$i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ;
then tmp_mods="$(grep -v "#" /usr/src/cc/freeswitch-1.0.6/modules.conf
| sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ;
mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )"
"OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#"
/usr/src/cc/freeswitch-1.0.6/modules.conf | sed -e "s|^.*/||" | sort |
uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo
$i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z
"" ; then tmp_mods="$(grep -v "#"
/usr/src/cc/freeswitch-1.0.6/modules.conf | sed -e "s|^.*/||" | sort |
uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo
$i-uninstall ; done)"; echo $mods )"
"OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#"
/usr/src/cc/freeswitch-1.0.6/modules.conf | grep -v "##" | sed -e
"s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo
$i-all ; done )"; echo $mods )"
"OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#"
/usr/src/cc/freeswitch-1.0.6/modules.conf | grep -v "##" | sed -e
"s|^.*/||" | sort | uniq )";  mods="$(for i in $tmp_mods ; do echo
$i-clean ; done )"; echo $mods )"
"OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#"
/usr/src/cc/freeswitch-1.0.6/modules.conf | grep -v "##" | sed -e
"s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo
$i-install ; done)"; echo $mods )"
"OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#"
/usr/src/cc/freeswitch-1.0.6/modules.conf | grep -v "##" | sed -e
"s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo
$i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s`
install-recursive
 /bin/sh /usr/src/cc/freeswitch-1.0.6/quiet_libtool --mode=install
/usr/bin/install -c  'libfreeswitch.la'
'/usr/local/freeswitch/lib/libfreeswitch.la'
/usr/bin/install -c .libs/libfreeswitch.so.1.0.0
/usr/local/freeswitch/lib/libfreeswitch.so.1.0.0
(cd /usr/local/freeswitch/lib && { ln -s -f libfreeswitch.so.1.0.0
libfreeswitch.so.1 || { rm -f libfreeswitch.so.1 && ln -s
libfreeswitch.so.1.0.0 libfreeswitch.so.1; }; })
(cd /usr/local/freeswitch/lib && { ln -s -f libfreeswitch.so.1.0.0
libfreeswitch.so || { rm -f libfreeswitch.so && ln -s
libfreeswitch.so.1.0.0 libfreeswitch.so; }; })
/usr/bin/install -c .libs/libfreeswitch.lai
/usr/local/freeswitch/lib/libfreeswitch.la
/usr/bin/install -c .libs/libfreeswitch.a
/usr/local/freeswitch/lib/libfreeswitch.a
chmod 644 /usr/local/freeswitch/lib/libfreeswitch.a
ranlib /usr/local/freeswitch/lib/libfreeswitch.a
PATH="$PATH:/sbin" ldconfig -n /usr/local/freeswitch/lib


From brian at freeswitch.org  Wed Nov 17 08:17:25 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Nov 2010 10:17:25 -0600
Subject: [Freeswitch-dev] compiling on Amazon EC2
In-Reply-To: <AANLkTi=pbt0tYGbtPzrqpqwxSQT6vhvPXGcPwMgjqM9x@mail.gmail.com>
References: <AANLkTi=pbt0tYGbtPzrqpqwxSQT6vhvPXGcPwMgjqM9x@mail.gmail.com>
Message-ID: <F6DA0A0F-9AF8-402F-9C96-5BA8BBB1B924@freeswitch.org>

This contains NO useful info.

/b

On Nov 17, 2010, at 10:14 AM, rentmycoder rentmycoder wrote:

> Hi all,
> 
> I have a problem installing FS on amazon EC2 micro instance (actually
> free) using SuseLinux Enterprise Server 11SP1 x86.
> Only this is the supported free trial OS, so I can't use another distro...
> 
> Freeswitch compiles, but make install hangs at:
> ldconfig -n /usr/local/freeswitch/lib
> 
> 
> I've tried 1.0.4, 1.0.6, current, all have the same problem...
> I've stripped down modules.conf before configure, same problem..
> 
> Any hint?
> Are there any precompiled packages for  SuseLinux Enterprise Server 11SP1 x86?



From william.suffill at gmail.com  Wed Nov 17 08:28:38 2010
From: william.suffill at gmail.com (William Suffill)
Date: Wed, 17 Nov 2010 11:28:38 -0500
Subject: [Freeswitch-dev] compiling on Amazon EC2
In-Reply-To: <AANLkTi=pbt0tYGbtPzrqpqwxSQT6vhvPXGcPwMgjqM9x@mail.gmail.com>
References: <AANLkTi=pbt0tYGbtPzrqpqwxSQT6vhvPXGcPwMgjqM9x@mail.gmail.com>
Message-ID: <AANLkTimgNJNXpPzHQ6w03DZez2H4Pkn5N9+suKVK7J1O@mail.gmail.com>

Can you pastebin the output so it won't get wrapped?
http://pastebin.freeswitch.org


From rupa at rupa.com  Wed Nov 17 09:00:48 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 17 Nov 2010 11:00:48 -0600
Subject: [Freeswitch-dev] Trouble Accessing DB Table in C-module Via ODBC
In-Reply-To: <2BF7FB90DF25EA4485949F3AF2B9D69603EB7F74FF@VA3DIAXVS351.RED001.local>
References: <2BF7FB90DF25EA4485949F3AF2B9D69603EB7F73E5@VA3DIAXVS351.RED001.local>
	<AANLkTikyGj-0wL-_H=gh3bXxaB1c3++NFuQE2SwdadnS@mail.gmail.com>
	<2BF7FB90DF25EA4485949F3AF2B9D69603EB7F74FF@VA3DIAXVS351.RED001.local>
Message-ID: <AANLkTinCLVhW3xf9B+wWh=H+54LcTeSkwu1GkLG7t7ck@mail.gmail.com>

I don't see how this is an ODBC issue.  Are you using the same user in your
module as with psql?  Have you tried the same query with isql (odbc equiv to
psql)?   Are you SURE you are connecting to the same db instance?

On Mon, Nov 15, 2010 at 2:01 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

> Hi Rupa,
>
>
>
> Yes, I'm using postgresql.  When I run the C-module using the schema.table
> format it gets this error:
>
>
>
> 2010-11-15 11:11:18.431977 [ERR] switch_core_sqldb.c:806 SQL ERR: [select
> sip_presence.status, sip_presence.rpid from ts_sofia_internal.sip_presence
> where sip_presence.sip_user='1003'] no such table:
> ts_sofia_internal.sip_presence
>
>
>
> If I execute the select statement from the psql command line, it works as
> shown:
>
>
>
> teo=# select sip_presence.status, sip_presence.rpid from
> ts_sofia_internal.sip_presence where sip_presence.sip_user='1003';
>
> status | rpid
>
> --------+------
>
> Away   | away
>
> (1 row)
>
>
>
> Could this have something to do with accessing the DB via ODBC?  Do you
> have another suggestion?
>
>
>
> I did try adding 'ts_sofia_internal' to the search path (i.e. set
> search_path to ucm, ts_sofia_internal), but it also did not work.
>
>
>
> Thanks,
>
> Jerry
>
>
>
>
>
>
>
> *From:* freeswitch-dev-bounces at lists.freeswitch.org [mailto:
> freeswitch-dev-bounces at lists.freeswitch.org] *On Behalf Of *Rupa Schomaker
> *Sent:* Monday, November 15, 2010 11:06 AM
> *To:* freeswitch-dev at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-dev] Trouble Accessing DB Table in C-module Via
> ODBC
>
>
>
> What database engine?  That looks like postgresql.  If so, add the schema
> you are interested in working with into the search path or explicitly use
> the schema in your query.
>
>
>
> schema.function(args) rather than just function(args)
>
>
>
> You can set the search path using alter user ...
>
> On Mon, Nov 15, 2010 at 10:56 AM, Jerry Richards <
> jerry.richards at teotech.com> wrote:
>
> Hello,
>
>
>
> I need to access a database using two different schemas.  During
> initialization the default schema is set as follows:
>
>
>
> set schema 'ucm';
>
>
>
> Then later, I need to get data from a table in another schema
> 'ts_sofia_internal', so I tried the following, but it returns an error as
> shown:
>
>
>
> teo=# select * from sip_presence('ts_sofia_internal') where
> sip_presence.sip_user='1003';
>
> ERROR:  function sip_presence(unknown) does not exist
>
> LINE 1: select * from sip_presence('ts_sofia_internal') where sip_pr...
>
>                       ^
>
> HINT:  No function matches the given name and argument types. You might
> need to add explicit type casts.
>
> teo=#
>
>
>
> Does anyone know how I can do this?  By the way, I will actually implement
> the query in C and connect to the DB via ODBC.
>
>
>
> Thanks,
>
> Jerry
>
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
>
>
> --
> -Rupa
>
> _______________________________________________
> 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
>
>


-- 
-Rupa
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From msc at freeswitch.org  Wed Nov 17 09:24:48 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Nov 2010 09:24:48 -0800
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today
Message-ID: <AANLkTikinmrr=RGwAYjcEBGCo9rVueFgqua2cSvv4+dj@mail.gmail.com>

Hello all,

It's Wednesday so that means FreeSWITCH conference call! Here's the agenda:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_11_17

We have a lot of little things to discuss, plus DRK is planning on stopping
by to talk more about his fun .NET stuff! Talk to you soon.

-MC
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From rentmycoder at gmail.com  Wed Nov 17 13:23:08 2010
From: rentmycoder at gmail.com (rentmycoder rentmycoder)
Date: Wed, 17 Nov 2010 22:23:08 +0100
Subject: [Freeswitch-dev] compiling on Amazon EC2
Message-ID: <AANLkTikgr7z=cRxbS0-6umXPuNG-CxL9iUWff093F6Am@mail.gmail.com>

http://pastebin.freeswitch.org/14516

Is this enough info?
Do you need something else to analyse?
Any hint?
Thanks a lot...


From brian at freeswitch.org  Wed Nov 17 13:39:31 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Nov 2010 15:39:31 -0600
Subject: [Freeswitch-dev] compiling on Amazon EC2
In-Reply-To: <AANLkTikgr7z=cRxbS0-6umXPuNG-CxL9iUWff093F6Am@mail.gmail.com>
References: <AANLkTikgr7z=cRxbS0-6umXPuNG-CxL9iUWff093F6Am@mail.gmail.com>
Message-ID: <2466BEAE-43B2-4123-B1DA-6927F250881A@freeswitch.org>

I'm not sure if you're just not understanding us or what... but this has ZERO information that would indicate any sort of failure... was this the whole output or did you pick the parts to pastebin?

/b

On Nov 17, 2010, at 3:23 PM, rentmycoder rentmycoder wrote:

> http://pastebin.freeswitch.org/14516
> 
> Is this enough info?
> Do you need something else to analyse?
> Any hint?
> Thanks a lot...
> 
> _______________________________________________
> 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 paulh at instruments.com  Wed Nov 17 14:47:44 2010
From: paulh at instruments.com (Paul Herring)
Date: Wed, 17 Nov 2010 16:47:44 -0600
Subject: [Freeswitch-dev] Add me to the dev list.
Message-ID: <3AC6176675CB60409DF572405E36FA0D0A90902539@ciexch>

Add me to the dev list.  Thanks.  Paul
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From msc at freeswitch.org  Wed Nov 17 14:56:28 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Nov 2010 14:56:28 -0800
Subject: [Freeswitch-dev] Add me to the dev list.
In-Reply-To: <3AC6176675CB60409DF572405E36FA0D0A90902539@ciexch>
References: <3AC6176675CB60409DF572405E36FA0D0A90902539@ciexch>
Message-ID: <AANLkTind_JQiskO7W7Nxct+QXTPfAzb4N2E7L73MpbGc@mail.gmail.com>

I will, but only if you're coming to next year's ClueCon! ;)
-MC


On Wed, Nov 17, 2010 at 2:47 PM, Paul Herring <paulh at instruments.com> wrote:

> Add me to the dev list.  Thanks.  Paul
>
> _______________________________________________
> 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 rentmycoder at gmail.com  Thu Nov 18 04:13:09 2010
From: rentmycoder at gmail.com (rentmycoder rentmycoder)
Date: Thu, 18 Nov 2010 13:13:09 +0100
Subject: [Freeswitch-dev] compiling on Amazon EC2
Message-ID: <AANLkTimPJ+vfWyc+MF2rz0zuogWLDVw3Xm80hqyS+TpX@mail.gmail.com>

Brian,

Maybe you misunderstood me...
make runs well, as you can see in the pastebin: FreeSWITCH has been
successfully built.

When I type make install, the script just hangs on, cpu load becomes
100% and thats all... I can't even connect to the image using ssh, I
can only restart the vm.

Tha last command cause the hang was: ldconfig -n /usr/local/freeswitch/lib
as you can see at pastebin...

>but this has ZERO information

Ok, but WHAT kind of information do you need?
Sorry, I'm not a linux expert, please describe what other output do you need...

Please help...

Thhanks in advance...

>I'm not sure if you're just not understanding us or what... but this has ZERO information that would indicate any sort of failure... was this the whole output or did you pick the parts >to pastebin?
>/b

On Nov 17, 2010, at 3:23 PM, rentmycoder rentmycoder wrote:

> http://pastebin.freeswitch.org/14516
>
> Is this enough info?
> Do you need something else to analyse?
> Any hint?
> Thanks a lot...
>
> _______________________________________________
> 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 mike at jerris.com  Thu Nov 18 09:19:05 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 18 Nov 2010 12:19:05 -0500
Subject: [Freeswitch-dev] Verbose Events
In-Reply-To: <C948D46A3126C344A5EF84C5FD1D49464E0B66E71C@IE2RD2XVS021.red002.local>
References: <C948D46A3126C344A5EF84C5FD1D49464E0B256D68@IE2RD2XVS021.red002.local>
	<C948D46A3126C344A5EF84C5FD1D49464E0B66E71C@IE2RD2XVS021.red002.local>
Message-ID: <9FC5B55F-DBE2-419C-A400-0285E285FF30@jerris.com>

A patch went in earlier this week on this issue.

Mike

On Nov 16, 2010, at 5:37 AM, David Brazier wrote:

> [same as last message, better formatting!]
> 
> Hi All
> 
> I want to get session variables on all events.  I have tried using the "verbose-events" dialplan app but that isn't convenient to use when bridging.  So I have set 
> 
> <param name="verbose-channel-events" value="yes"/>
> 
> in switch.conf.xml & restarted to make sure.  But I still don't see variables on all events, for example CHANNEL_CALLSTATE.  In the source code, there are 3 related flags:
> 
> 1.  CF_VERBOSE_EVENTS - the "channel flag" set by the "verbose-events" dialpan app 
> 
> 2.  SCF_VERBOSE_EVENTS - the "switch control flag", set by "verbose-channel-events" in switch.conf.xml 
> 
> 3.  SCSC_VERBOSE_EVENTS - used by "fsctl verbose_events" - actually SCSC_VERBOSE_EVENTS is only an internal name for the command - using "fsctl verbose_events" sets or clears SCF_VERBOSE_EVENTS
> 
> In switch_core.c the SCSC_ flags are accessed via switch_core_session_ctl: 
> 
> SWITCH_DECLARE(int32_t) switch_core_session_ctl(switch_session_ctl_t cmd, void *val) {
>        int *intval = (int *) val;
>        int oldintval = 0, newintval = 0;
> 
>        if (intval) {
>                oldintval = *intval;
>        }
> 
>        if (switch_test_flag((&runtime), SCF_SHUTTING_DOWN)) {
>                return -1;
>        }
> 
>        switch (cmd) {
>        case SCSC_VERBOSE_EVENTS:
>                if (intval) {
>                        if (oldintval > -1) {
>                                if (oldintval) {
>                                        switch_set_flag((&runtime), SCF_VERBOSE_EVENTS);
>                                } else {
>                                        switch_clear_flag((&runtime), SCF_VERBOSE_EVENTS);
>                                }
>                        }
>                        newintval = switch_test_flag((&runtime), SCF_VERBOSE_EVENTS);
>                }
>                break;
>        ...
>        }
> 
>        if (intval) {
>                *intval = newintval;
>        }
> 
> 
>        return 0;
> }
> 
> If I read this right, the caller can use this to just get the flag without setting it by setting val to -1 initially.  However, in switch_channel.c where it is deciding whether to do a verbose event:
> 
> SWITCH_DECLARE(void) switch_channel_event_set_extended_data(switch_channel_t *channel, switch_event_t *event) {
>        switch_event_header_t *hi;
>        int x, global_verbose_events = 0;
> 
>        switch_mutex_lock(channel->profile_mutex);
> 
>        switch_core_session_ctl(SCSC_VERBOSE_EVENTS, &global_verbose_events);
> 
>        if (global_verbose_events ||
>                switch_channel_test_flag(channel, CF_VERBOSE_EVENTS) ||
>                switch_event_get_header(event, "presence-data-cols") ||
>                event->event_id == SWITCH_EVENT_CHANNEL_CREATE ||
>                event->event_id == SWITCH_EVENT_CHANNEL_ORIGINATE || ...)
>        // then add the variables to the event
> 
> Because this does "global_verbose_events = 0" it always clears SCF_VERBOSE_EVENTS.
> 
> I could easily have misuderstood, and if anyone else has had success with "verbose-channel-events" please let me know.
> 
> Cheers
> 
> David
> 
> _______________________________________________
> 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 msc at freeswitch.org  Thu Nov 18 11:33:00 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Nov 2010 11:33:00 -0800
Subject: [Freeswitch-dev] compiling on Amazon EC2
In-Reply-To: <AANLkTimPJ+vfWyc+MF2rz0zuogWLDVw3Xm80hqyS+TpX@mail.gmail.com>
References: <AANLkTimPJ+vfWyc+MF2rz0zuogWLDVw3Xm80hqyS+TpX@mail.gmail.com>
Message-ID: <AANLkTin1hWvbJod15zk84iQ8uGSoNcpf1YpjAMrsgDU7@mail.gmail.com>

Brian is saying that EC2 isn't showing us any useful information. You may
need to have EC2 support look at this to see if they can diagnose what is
going on. ldconfig shouldn't be going nuts like this and most likely this
isn't an issue specifically with the FS install.

-MC

On Thu, Nov 18, 2010 at 4:13 AM, rentmycoder rentmycoder <
rentmycoder at gmail.com> wrote:

> Brian,
>
> Maybe you misunderstood me...
> make runs well, as you can see in the pastebin: FreeSWITCH has been
> successfully built.
>
> When I type make install, the script just hangs on, cpu load becomes
> 100% and thats all... I can't even connect to the image using ssh, I
> can only restart the vm.
>
> Tha last command cause the hang was: ldconfig -n /usr/local/freeswitch/lib
> as you can see at pastebin...
>
> >but this has ZERO information
>
> Ok, but WHAT kind of information do you need?
> Sorry, I'm not a linux expert, please describe what other output do you
> need...
>
> Please help...
>
> Thhanks in advance...
>
> >I'm not sure if you're just not understanding us or what... but this has
> ZERO information that would indicate any sort of failure... was this the
> whole output or did you pick the parts >to pastebin?
> >/b
>
> On Nov 17, 2010, at 3:23 PM, rentmycoder rentmycoder wrote:
>
> > http://pastebin.freeswitch.org/14516
> >
> > Is this enough info?
> > Do you need something else to analyse?
> > Any hint?
> > Thanks a lot...
> >
> > _______________________________________________
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Nov 18 11:50:44 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Nov 2010 11:50:44 -0800
Subject: [Freeswitch-dev] IMPORTANT: If you use "verbose events" then read
	this
Message-ID: <AANLkTikgCD-VR12A8HABRyk_0xtgm=dBeyApzK7xOBGx@mail.gmail.com>

Guys,

Just an FYI for those of you who use verbose events in your event
programming... Tony fixed a bug recently:

http://jira.freeswitch.org/browse/FS-2851

The commit is here:

http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=180f58a677a2ef49f71cbbe377f9eb177e6574e5

If you have no clue what verbose events are for then you can probably ignore
this. :)

-MC
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From farhan.husain at csebuet.org  Thu Nov 18 23:48:08 2010
From: farhan.husain at csebuet.org (Farhan Husain)
Date: Fri, 19 Nov 2010 01:48:08 -0600
Subject: [Freeswitch-dev] Dynamic Conference creation
Message-ID: <AANLkTin-Z_AwkYkhh4Qz_6u=WqhNPojqtK8RYFZAu1yf@mail.gmail.com>

Hello,

I could successfully create audio conference by adding extension in
dialplan. However, how can I create conferences dynamically? Can I use
mod_event_socket for that? Is there any command to create a conference
dynamically?

Thanks,
Farhan
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From cstomi.levlist at gmail.com  Fri Nov 19 08:34:40 2010
From: cstomi.levlist at gmail.com (Tamas Cseke)
Date: Fri, 19 Nov 2010 17:34:40 +0100
Subject: [Freeswitch-dev] media bug remove codec destroy crash
Message-ID: <4CE6A720.1050506@gmail.com>

Hello,

We have a segfault. It's not the newest code, so I didn't opened a jira yet.

commit 84d897cb05079a4bfb3ed4ff98df0b5f666e0f70
Date:   Thu Jul 22 01:23:06 2010 -0500

I advised to upgrade but it's not going so fast, and I think I found 
something so I ask before we do the upgarde
I saw there were commits in media bug remove code, but I'm not sure it 
fixes it.

http://pastebin.freeswitch.org/14554

The codec is destroyed in switch_core_media_bug_remove_all although 
switch_codec_ready said it's ready.
I found that it could be removed in bug_remove or bug_prune too.
In those functions the write lock is locked, but in remove_all it's 
unlocked before the codec destroy.
So I think it somehow the bug remove and session hangup happend in the 
same time (I'm not 100% sure it's possible)
and remove_all tried to destroy the codec that was already destroyed.

I think the rwunlock sould go after the codec destroy in remove_all.
Please check, and let me know if my theory make sense :)

Thanks,
Tamas











From anthony.minessale at gmail.com  Fri Nov 19 09:24:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Nov 2010 11:24:24 -0600
Subject: [Freeswitch-dev] media bug remove codec destroy crash
In-Reply-To: <4CE6A720.1050506@gmail.com>
References: <4CE6A720.1050506@gmail.com>
Message-ID: <AANLkTim_1Do2ir-HcdrsvkkhrpfN4nPejCeyay0HW0sS@mail.gmail.com>

try latest or newest patch to switch_core_codec.c
I changed it to eliminate any destroy races


On Fri, Nov 19, 2010 at 10:34 AM, Tamas Cseke <cstomi.levlist at gmail.com> wrote:
> Hello,
>
> We have a segfault. It's not the newest code, so I didn't opened a jira yet.
>
> commit 84d897cb05079a4bfb3ed4ff98df0b5f666e0f70
> Date: ? Thu Jul 22 01:23:06 2010 -0500
>
> I advised to upgrade but it's not going so fast, and I think I found
> something so I ask before we do the upgarde
> I saw there were commits in media bug remove code, but I'm not sure it
> fixes it.
>
> http://pastebin.freeswitch.org/14554
>
> The codec is destroyed in switch_core_media_bug_remove_all although
> switch_codec_ready said it's ready.
> I found that it could be removed in bug_remove or bug_prune too.
> In those functions the write lock is locked, but in remove_all it's
> unlocked before the codec destroy.
> So I think it somehow the bug remove and session hangup happend in the
> same time (I'm not 100% sure it's possible)
> and remove_all tried to destroy the codec that was already destroyed.
>
> I think the rwunlock sould go after the codec destroy in remove_all.
> Please check, and let me know if my theory make sense :)
>
> Thanks,
> Tamas
>
>
>
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From msc at freeswitch.org  Fri Nov 19 10:45:19 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 19 Nov 2010 10:45:19 -0800
Subject: [Freeswitch-dev] IMPORTANT NEWS: blacklist.pl script not working
 because of infiltrated.net expiration
Message-ID: <AANLkTimRcSyKvhfsweqUv9mS6TXzXFd4ZKbmBs6Dzuah@mail.gmail.com>

Hello all,

It seems that infiltrated.net expired yesterday. This is the source of the
data for our blacklist.pl script. The script pulls two files:

http://www.infiltrated.net/voipabuse/addresses.txt
http://www.infiltrated.net/voipabuse/netblocks.txt

It uses these two files for building a better ACL that rejects known bad
guys. Since these files are not currently available the script will receive
bogus information and pollute the ACL. If you have run the
blacklist.plscript prior to the failure and your system is working
properly then please
make a backup copy of your freeswitch.xml.fsxml file. This will contain the
proper ACL nodes. DO NOT RESTART FREESWITCH OR RELOAD XML BEFORE BACKING UP
THIS FILE! If you do then you will lose the information.

I am researching this right now and I will update you as soon as I have some
information. In the meantime if any of you have a freeswitch.xml.fsxml file
with the blacklist info please zip and email me offlist a copy of that file
and I will see if I can't extract the IPs and create a temporary workaround.
I will also be updating the blacklist.pl script to do some error checking so
that it doesn't pollute the ACL.

Last item: if you have inside knowledge of what's happening with the VoIP
Abuse project please contact me off list.

Thanks,
Michael S Collins
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From alex.deiter at gmail.com  Mon Nov 22 13:20:00 2010
From: alex.deiter at gmail.com (Alex Deiter)
Date: Tue, 23 Nov 2010 00:20:00 +0300
Subject: [Freeswitch-dev] issue ESL-52: event nitifications filters
Message-ID: <AANLkTik9Nm68gupvjCThcPqPxtUqH4QJqN2t8Zguqxka@mail.gmail.com>

Hi,

Could you please review issue ESL-52 ?
We are used FreeSWITCH with BigBlueButton and found problem with ESL events.
Each new filter loses the previous filter. Example:

% ./fs_cli -H 127.0.0.1 -p ClueCon
...
freeswitch at 127.0.0.1@internal> /event all
+OK event listener enabled plain

===> now received ALL events
Event-Name: HEARTBEAT
Event-Name: RE_SCHEDULE
Event-Name: HEARTBEAT
Event-Name: RE_SCHEDULE

===> add filter
===> now received ONLY HEARTBEAT events
freeswitch at 127.0.0.1@internal> /filter Event-Name HEARTBEAT
+OK filter added. [Event-Name]=[HEARTBEAT]

===> add another filter
===> now FreeSWITCH forgotten about the previous filter and used only
last filter
freeswitch at 127.0.0.1@internal> /filter Event-Name RE_SCHEDULE
+OK filter added. [Event-Name]=[RE_SCHEDULE]
Event-Name: RE_SCHEDULE
Event-Name: RE_SCHEDULE
Event-Name: RE_SCHEDULE

This behavior was noted after FreeSWITCH snapshot 12-Nov-2010

FreeSWITCH snapshot 15-Oct-2010 works as it should: each new filter
was added to the existing filters.

Could you please explain this ?

Thanks a lot!

--
Alex Deiter


From andy.pyles at gmail.com  Tue Nov 16 07:08:08 2010
From: andy.pyles at gmail.com (Andy Pyles)
Date: Tue, 16 Nov 2010 10:08:08 -0500
Subject: [Freeswitch-dev] VAD/ CNG question
Message-ID: <AANLkTi=Jc6f-6XeVZiz0MMGGq2bnrFnjmYY1jnppcWY5@mail.gmail.com>

I setup the following scenario:  A calls FS, FS plays back a wav file.
I turned on the following channel variables:

rtp_enable_vad_in=true
rtp_enable_vad_out=true

My question is where in the code are these channel variables defined?
Specifically, I'd like to modify the silence threshold that is used.
For my purposes, I'd like to tune the threshold to something more
agressive than the default value.

I'm using g.711U codec.

thanks,
Andy


From chat2jesse at gmail.com  Sat Nov 20 22:48:08 2010
From: chat2jesse at gmail.com (jesse)
Date: Sat, 20 Nov 2010 22:48:08 -0800
Subject: [Freeswitch-dev] please shed me light on code reading
In-Reply-To: <AANLkTinoBQJwiy_h0ygY+RsA6zU0EfzJBLCPhVmD_ved@mail.gmail.com>
References: <AANLkTinoBQJwiy_h0ygY+RsA6zU0EfzJBLCPhVmD_ved@mail.gmail.com>
Message-ID: <AANLkTi=MG0PbjT1s8A=LCAdYOpGeqM0TJj2xSONT+QmL@mail.gmail.com>

hi,

 Any expert can shed me some light on quick code understanding?
thanks in advance!

suppose I dial 88<1001> to spy on extension user 1001. and 5 minutes
later 1001 calls 1002.

<extension name="userspy">
    <condition field="destination_number" expression="^88(.*)$|^\*0(.*)$">
      <action application="answer"/>
      <action application="userspy" data="$1 at 192.168.1.1"/>
    </condition>
  </extension>

 in event_handle(...) of mod_spy.c.
 -- Whats is the reason that we need to call process_event for both event and
peer_event?
    is not processing event for call leg-a 's bridge enough? it seems
mod_spy.c in 1.0.6 version
doesn't process peer_event.

static void event_handler(switch_event_t *event)
{
	if (process_event(event) != SWITCH_STATUS_SUCCESS) {
		const char *peer_uuid = switch_event_get_header(event,
"variable_signal_bond");
                ...
		if (peer_event) {
			process_event(peer_event);
		}
	}
}


http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/applications/mod_spy/mod_spy.c


From chat2jesse at gmail.com  Sat Nov 20 23:36:02 2010
From: chat2jesse at gmail.com (jesse)
Date: Sat, 20 Nov 2010 23:36:02 -0800
Subject: [Freeswitch-dev] please shed me light on code reading
In-Reply-To: <AANLkTi=MG0PbjT1s8A=LCAdYOpGeqM0TJj2xSONT+QmL@mail.gmail.com>
References: <AANLkTinoBQJwiy_h0ygY+RsA6zU0EfzJBLCPhVmD_ved@mail.gmail.com>
	<AANLkTi=MG0PbjT1s8A=LCAdYOpGeqM0TJj2xSONT+QmL@mail.gmail.com>
Message-ID: <AANLkTikp-hYSVwq0G5KYZfkn+z1baYmuabUWU2j52teT@mail.gmail.com>

one more question.

switch_channel_set_variable(channel, "spy_uuid", my_uuid);
switch_channel_set_private(channel, "_userspy_", (void *) argv[0]);


why one is set via rugular variable and one is set via private?  is
that necessary?

-jesse


On Sat, Nov 20, 2010 at 10:48 PM, jesse <chat2jesse at gmail.com> wrote:
> hi,
>
> ?Any expert can shed me some light on quick code understanding?
> thanks in advance!
>
> suppose I dial 88<1001> to spy on extension user 1001. and 5 minutes
> later 1001 calls 1002.
>
> <extension name="userspy">
> ? ?<condition field="destination_number" expression="^88(.*)$|^\*0(.*)$">
> ? ? ?<action application="answer"/>
> ? ? ?<action application="userspy" data="$1 at 192.168.1.1"/>
> ? ?</condition>
> ?</extension>
>
> ?in event_handle(...) of mod_spy.c.
> ?-- Whats is the reason that we need to call process_event for both event and
> peer_event?
> ? ?is not processing event for call leg-a 's bridge enough? it seems
> mod_spy.c in 1.0.6 version
> doesn't process peer_event.
>
> static void event_handler(switch_event_t *event)
> {
> ? ? ? ?if (process_event(event) != SWITCH_STATUS_SUCCESS) {
> ? ? ? ? ? ? ? ?const char *peer_uuid = switch_event_get_header(event,
> "variable_signal_bond");
> ? ? ? ? ? ? ? ?...
> ? ? ? ? ? ? ? ?if (peer_event) {
> ? ? ? ? ? ? ? ? ? ? ? ?process_event(peer_event);
> ? ? ? ? ? ? ? ?}
> ? ? ? ?}
> }
>
>
> http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/applications/mod_spy/mod_spy.c
>


From prashant.lamba at gmail.com  Tue Nov 16 06:12:57 2010
From: prashant.lamba at gmail.com (Prashant Lamba)
Date: Tue, 16 Nov 2010 19:42:57 +0530
Subject: [Freeswitch-dev] sending custom SDP in 183 Session Progress
Message-ID: <AANLkTi=HpRxXLLgBQW64Hym4rFrfC1wm_nYZ+ykA2PNB@mail.gmail.com>

>> On Fri, Apr 23, 2010 at 12:28 AM, Raj Kiran Talusani <
>> rajkiran.talusani at gmail.com> wrote:
>>
>> I have written a application called mod_custom. In this module i would like
>> to set a user-defined RTP port and IP adress and then send the 200 OK
>> (answer the call). But i see that, before my application is invoked, 183
>> message is sent with SDP. And i didnot find a way to either disable sending
>> 183 and just send 180 OR modify the SDP to be sent in 200 OK.
>> please help.


> Anthony Minessale anthony.minessale at gmail.com
> Mon May 10 09:51:51 PDT 2010
>
> if you use anything that needs media before you send 200 it will have to
> send 183


I have a similar problem to the one posted above, which I got by
searching the archieves:
http://lists.freeswitch.org/pipermail/freeswitch-dev/2010-May/003653.html

We have integrated our own IVR with FS. Our IVR handles all media
(RTP) directly and FS handles only SIP.

When a call comes to FS with INVITE, in response we send 183 Session
Progress with FS generated SDP (and RTP IP and port). In 200 OK, FS
sends SDP with our application generated RTP IP and port. This works
in most cases.

Is there any way to invoke my application before sending 183 Session
Progress, so that our application generated SDP can be sent in the 183
Session Progress?

Prashant
prashant.lamba at gmail.com


From raul2r2 at gmail.com  Sat Nov 20 11:24:54 2010
From: raul2r2 at gmail.com (raul2r2 raul)
Date: Sat, 20 Nov 2010 20:24:54 +0100
Subject: [Freeswitch-dev] Commercial module development?
Message-ID: <AANLkTin+qs2oJG0B9AhY0hGL=Mx9SYhUFzpqsEtMsQdH@mail.gmail.com>

Hi...
      I'm thinking in developing a new module for Freeswitch for commercial
use and distribution. It would be a new end point module based in a
propietary binary protocol. And here the problem raise, because I'm a bit
confusing about MPL license and its implications. Would this module be
considered as a derivate work?.
      If it's considered as a derivate work, is it any kind of commercial
license for freeswitch?.
      and the last question, if any of this is possible, could I statically
link Freeswitch core or some of its modules in my application if I don't
modify its source code?.
Regards,
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From rodrigoferreiralang at gmail.com  Wed Nov 17 04:29:43 2010
From: rodrigoferreiralang at gmail.com (Rodrigo Lang)
Date: Wed, 17 Nov 2010 10:29:43 -0200
Subject: [Freeswitch-dev] How to compile FS debian packages?
In-Reply-To: <AANLkTinY52rr29nwQkSjuV1mdrEHhLyMEQBRaZ+s7+hf@mail.gmail.com>
References: <AANLkTinCit4AS2nbj0Zxr9HasWnpOJjOW3tThefWzzZL@mail.gmail.com>
	<AANLkTino-2iKoKW63EYLq3yxhFM4JRsguxv-1Y=NNCcx@mail.gmail.com>
	<AANLkTinY52rr29nwQkSjuV1mdrEHhLyMEQBRaZ+s7+hf@mail.gmail.com>
Message-ID: <AANLkTim1PU31+FPy7pBRbRO9Vfjij0kYtEoP_Hgb_nGo@mail.gmail.com>

>
> 2010/11/13 Gabriel Gunderson <gabe at gundy.org>:
> > This is the script that I've been using to build my debs.  I just ran
> > it on the latest git and it products debs without errors:
>
> Thanks. I will test and report back once I have some time.
> (this can take some days)
>


Hi, i made this script [1] for me, but i made available in the pastebin.
Works for Ubuntu and Debian, install FreeSWITCH-1.0.6.


[1] http://pastebin.com/2Jyh5dxS



Best regards,
-- 
Rodrigo Lang
Opening your mind - Just another Open Source
site<http://openingyourmind.wordpress.com/>
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From anthony.minessale at gmail.com  Mon Nov 22 14:37:49 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Nov 2010 16:37:49 -0600
Subject: [Freeswitch-dev] Commercial module development?
In-Reply-To: <AANLkTin+qs2oJG0B9AhY0hGL=Mx9SYhUFzpqsEtMsQdH@mail.gmail.com>
References: <AANLkTin+qs2oJG0B9AhY0hGL=Mx9SYhUFzpqsEtMsQdH@mail.gmail.com>
Message-ID: <AANLkTi=09TA7DichUdhXiXN4A_G0KZ2gYBHvC60xNULv@mail.gmail.com>

Commercial modules are acceptable.

You can dist your module as binary only but you may not use the
trademark of FreeSWITCH to promote it without license.
We prefer that companies who develop commercial add-ons do something
to help the community but we do not demand it.





On Sat, Nov 20, 2010 at 1:24 PM, raul2r2 raul <raul2r2 at gmail.com> wrote:
> Hi...
> ?? ? ?I'm thinking in developing a new module for Freeswitch for commercial
> use and distribution. It would be a new end point module based in a
> propietary binary protocol. And here the problem raise, because I'm a bit
> confusing about MPL license and its implications. Would this module be
> considered as a derivate work?.
> ?? ? ?If it's considered as a derivate work, is it any kind of commercial
> license for freeswitch?.
> ?? ? ?and the last question, if any of this is possible, could I statically
> link Freeswitch core or some of its modules in my application if I don't
> modify its source code?.
> Regards,
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From paulo at voicetechnology.com.br  Mon Nov 22 21:10:48 2010
From: paulo at voicetechnology.com.br (=?ISO-8859-1?Q?Paulo_Rog=E9rio_Panhoto?=)
Date: Tue, 23 Nov 2010 00:10:48 -0500
Subject: [Freeswitch-dev] please shed me light on code reading
In-Reply-To: <AANLkTi=MG0PbjT1s8A=LCAdYOpGeqM0TJj2xSONT+QmL@mail.gmail.com>
References: <AANLkTinoBQJwiy_h0ygY+RsA6zU0EfzJBLCPhVmD_ved@mail.gmail.com>
	<AANLkTi=MG0PbjT1s8A=LCAdYOpGeqM0TJj2xSONT+QmL@mail.gmail.com>
Message-ID: <4CEB4CD8.3000406@voicetechnology.com.br>

    From a quick peek at the code, I can state the obvious:
process(peer_event) is only called when process(event) has failed. So,
process_event will be called for event or peer_event (not both).

    I can think of a scenario where this might happen (this is an
educated guess, though):
    If channels A and B bridged and A hangs up. A receives a
CHANNEL_HANGUP and channel B must receive a CHANNEL_UNBRIDGE (from
channel A) event, so that it can take appropriate actions (like hang up
too or go back to an IVR menu).

    Regards,

    Paulo.

On 21/11/10 01:48, jesse wrote:
> hi,
>
>  Any expert can shed me some light on quick code understanding?
> thanks in advance!
>
> suppose I dial 88<1001> to spy on extension user 1001. and 5 minutes
> later 1001 calls 1002.
>
> <extension name="userspy">
>     <condition field="destination_number" expression="^88(.*)$|^\*0(.*)$">
>       <action application="answer"/>
>       <action application="userspy" data="$1 at 192.168.1.1"/>
>     </condition>
>   </extension>
>
>  in event_handle(...) of mod_spy.c.
>  -- Whats is the reason that we need to call process_event for both event and
> peer_event?
>     is not processing event for call leg-a 's bridge enough? it seems
> mod_spy.c in 1.0.6 version
> doesn't process peer_event.
>
> static void event_handler(switch_event_t *event)
> {
> 	if (process_event(event) != SWITCH_STATUS_SUCCESS) {
> 		const char *peer_uuid = switch_event_get_header(event,
> "variable_signal_bond");
>                 ...
> 		if (peer_event) {
> 			process_event(peer_event);
> 		}
> 	}
> }
>
>
> http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/applications/mod_spy/mod_spy.c
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>   


From mike at jerris.com  Tue Nov 23 06:25:35 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 23 Nov 2010 09:25:35 -0500
Subject: [Freeswitch-dev] sending custom SDP in 183 Session Progress
In-Reply-To: <AANLkTi=HpRxXLLgBQW64Hym4rFrfC1wm_nYZ+ykA2PNB@mail.gmail.com>
References: <AANLkTi=HpRxXLLgBQW64Hym4rFrfC1wm_nYZ+ykA2PNB@mail.gmail.com>
Message-ID: <83D1914E-F2D1-400D-B98E-7FA34348F18E@jerris.com>

make sure to set the flags on your application to say it supports no media, you should find some examples in mod_dptools.  Then you can set channel vars for either codecs, or the entire sdp string before calling switch_channel_pre_answer.

Mike

> I have a similar problem to the one posted above, which I got by
> searching the archieves:
> http://lists.freeswitch.org/pipermail/freeswitch-dev/2010-May/003653.html
> 
> We have integrated our own IVR with FS. Our IVR handles all media
> (RTP) directly and FS handles only SIP.
> 
> When a call comes to FS with INVITE, in response we send 183 Session
> Progress with FS generated SDP (and RTP IP and port). In 200 OK, FS
> sends SDP with our application generated RTP IP and port. This works
> in most cases.
> 
> Is there any way to invoke my application before sending 183 Session
> Progress, so that our application generated SDP can be sent in the 183
> Session Progress?
> 



From mike at jerris.com  Tue Nov 23 06:27:00 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 23 Nov 2010 09:27:00 -0500
Subject: [Freeswitch-dev] VAD/ CNG question
In-Reply-To: <AANLkTi=Jc6f-6XeVZiz0MMGGq2bnrFnjmYY1jnppcWY5@mail.gmail.com>
References: <AANLkTi=Jc6f-6XeVZiz0MMGGq2bnrFnjmYY1jnppcWY5@mail.gmail.com>
Message-ID: <A5281DF8-8D02-47BF-9E22-831847741B99@jerris.com>


On Nov 16, 2010, at 10:08 AM, Andy Pyles wrote:

> I setup the following scenario:  A calls FS, FS plays back a wav file.
> I turned on the following channel variables:
> 
> rtp_enable_vad_in=true
> rtp_enable_vad_out=true
> 
> My question is where in the code are these channel variables defined?
> Specifically, I'd like to modify the silence threshold that is used.
> For my purposes, I'd like to tune the threshold to something more
> agressive than the default value.
> 
> I'm using g.711U codec.
> 

>:git grep -n rtp_enable_vad_
mod/endpoints/mod_sofia/sofia_glue.c:3073:              if ((val = switch_channel_get_variable(tech_pvt->channel, "rtp_enable_vad_in")) && switch_true(val)) {
mod/endpoints/mod_sofia/sofia_glue.c:3076:              if ((val = switch_channel_get_variable(tech_pvt->channel, "rtp_enable_vad_out")) && switch_true(val)) {





From mike at jerris.com  Tue Nov 23 06:30:07 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 23 Nov 2010 09:30:07 -0500
Subject: [Freeswitch-dev] Commercial module development?
In-Reply-To: <AANLkTin+qs2oJG0B9AhY0hGL=Mx9SYhUFzpqsEtMsQdH@mail.gmail.com>
References: <AANLkTin+qs2oJG0B9AhY0hGL=Mx9SYhUFzpqsEtMsQdH@mail.gmail.com>
Message-ID: <641D033D-7185-4566-A4EC-14A00A1407B0@jerris.com>

as for static linking, you can with MPL (we have not done anything in the build system to support this in a sane way, but patches welcome), but be aware that there is LGPL dependencies as well, so you need to think of this when doing static linking and how that effects your distribution requirements.  For the MPL code, linking shouldn't be an issue.

DISCLAIMER: I am not a lawyer, these are my opinions only, if you have legal questions, you should ask a lawyer.

Mike

On Nov 20, 2010, at 2:24 PM, raul2r2 raul wrote:

> Hi...
>       I'm thinking in developing a new module for Freeswitch for commercial use and distribution. It would be a new end point module based in a propietary binary protocol. And here the problem raise, because I'm a bit confusing about MPL license and its implications. Would this module be considered as a derivate work?. 
>       If it's considered as a derivate work, is it any kind of commercial license for freeswitch?.
>       and the last question, if any of this is possible, could I statically link Freeswitch core or some of its modules in my application if I don't modify its source code?.
> Regards,
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org

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From mike at jerris.com  Tue Nov 23 06:39:17 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 23 Nov 2010 09:39:17 -0500
Subject: [Freeswitch-dev] please shed me light on code reading
In-Reply-To: <AANLkTikp-hYSVwq0G5KYZfkn+z1baYmuabUWU2j52teT@mail.gmail.com>
References: <AANLkTinoBQJwiy_h0ygY+RsA6zU0EfzJBLCPhVmD_ved@mail.gmail.com>
	<AANLkTi=MG0PbjT1s8A=LCAdYOpGeqM0TJj2xSONT+QmL@mail.gmail.com>
	<AANLkTikp-hYSVwq0G5KYZfkn+z1baYmuabUWU2j52teT@mail.gmail.com>
Message-ID: <C9E0C9E5-7805-42D3-8B5E-7424A6E77242@jerris.com>

a variable is a channel variable that you can access in the dial-plan, in events, etc.  A private is a void pointer that we use to store data that is later accessed by name in the application.  The code you were talking about below was added to support loopback channel:

commit d0a74dd5c445e8a1540ba0a3f465ad394e033193
Date:   Mon Apr 26 04:30:10 2010 -0400
    mod_spy: add support for loopback endpoint (MODAPP-416)



Mike

On Nov 21, 2010, at 2:36 AM, jesse wrote:

> one more question.
> 
> switch_channel_set_variable(channel, "spy_uuid", my_uuid);
> switch_channel_set_private(channel, "_userspy_", (void *) argv[0]);
> 
> 
> why one is set via rugular variable and one is set via private?  is
> that necessary?
> 
> -jesse
> 
> 
> On Sat, Nov 20, 2010 at 10:48 PM, jesse <chat2jesse at gmail.com> wrote:
>> hi,
>> 
>>  Any expert can shed me some light on quick code understanding?
>> thanks in advance!
>> 
>> suppose I dial 88<1001> to spy on extension user 1001. and 5 minutes
>> later 1001 calls 1002.
>> 
>> <extension name="userspy">
>>    <condition field="destination_number" expression="^88(.*)$|^\*0(.*)$">
>>      <action application="answer"/>
>>      <action application="userspy" data="$1 at 192.168.1.1"/>
>>    </condition>
>>  </extension>
>> 
>>  in event_handle(...) of mod_spy.c.
>>  -- Whats is the reason that we need to call process_event for both event and
>> peer_event?
>>    is not processing event for call leg-a 's bridge enough? it seems
>> mod_spy.c in 1.0.6 version
>> doesn't process peer_event.
>> 
>> static void event_handler(switch_event_t *event)
>> {
>>        if (process_event(event) != SWITCH_STATUS_SUCCESS) {
>>                const char *peer_uuid = switch_event_get_header(event,
>> "variable_signal_bond");
>>                ...
>>                if (peer_event) {
>>                        process_event(peer_event);
>>                }
>>        }
>> }
>> 
>> 
>> http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/applications/mod_spy/mod_spy.c
>> 
> 
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org



From mike at jerris.com  Tue Nov 23 06:42:26 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 23 Nov 2010 09:42:26 -0500
Subject: [Freeswitch-dev] issue ESL-52: event nitifications filters
In-Reply-To: <AANLkTik9Nm68gupvjCThcPqPxtUqH4QJqN2t8Zguqxka@mail.gmail.com>
References: <AANLkTik9Nm68gupvjCThcPqPxtUqH4QJqN2t8Zguqxka@mail.gmail.com>
Message-ID: <28A35C47-BB97-4687-9B8B-EE0CFA786F20@jerris.com>

please see commit 89dbe0b from yesterday.

On Nov 22, 2010, at 4:20 PM, Alex Deiter wrote:

> Hi,
> 
> Could you please review issue ESL-52 ?
> We are used FreeSWITCH with BigBlueButton and found problem with ESL events.
> Each new filter loses the previous filter. Example:
> 
> % ./fs_cli -H 127.0.0.1 -p ClueCon
> ...
> freeswitch at 127.0.0.1@internal> /event all
> +OK event listener enabled plain
> 
> ===> now received ALL events
> Event-Name: HEARTBEAT
> Event-Name: RE_SCHEDULE
> Event-Name: HEARTBEAT
> Event-Name: RE_SCHEDULE
> 
> ===> add filter
> ===> now received ONLY HEARTBEAT events
> freeswitch at 127.0.0.1@internal> /filter Event-Name HEARTBEAT
> +OK filter added. [Event-Name]=[HEARTBEAT]
> 
> ===> add another filter
> ===> now FreeSWITCH forgotten about the previous filter and used only
> last filter
> freeswitch at 127.0.0.1@internal> /filter Event-Name RE_SCHEDULE
> +OK filter added. [Event-Name]=[RE_SCHEDULE]
> Event-Name: RE_SCHEDULE
> Event-Name: RE_SCHEDULE
> Event-Name: RE_SCHEDULE
> 
> This behavior was noted after FreeSWITCH snapshot 12-Nov-2010
> 
> FreeSWITCH snapshot 15-Oct-2010 works as it should: each new filter
> was added to the existing filters.



From math.parent at gmail.com  Tue Nov 23 13:20:20 2010
From: math.parent at gmail.com (Mathieu Parent)
Date: Tue, 23 Nov 2010 22:20:20 +0100
Subject: [Freeswitch-dev] How to compile FS debian packages?
In-Reply-To: <AANLkTim1PU31+FPy7pBRbRO9Vfjij0kYtEoP_Hgb_nGo@mail.gmail.com>
References: <AANLkTinCit4AS2nbj0Zxr9HasWnpOJjOW3tThefWzzZL@mail.gmail.com>
	<AANLkTino-2iKoKW63EYLq3yxhFM4JRsguxv-1Y=NNCcx@mail.gmail.com>
	<AANLkTinY52rr29nwQkSjuV1mdrEHhLyMEQBRaZ+s7+hf@mail.gmail.com>
	<AANLkTim1PU31+FPy7pBRbRO9Vfjij0kYtEoP_Hgb_nGo@mail.gmail.com>
Message-ID: <AANLkTik=dCf0BrAVqH9izn=MGbWkih8f++zP1JVVATp_@mail.gmail.com>

Hello,

2010/11/17 Rodrigo Lang <rodrigoferreiralang at gmail.com>:
>> 2010/11/13 Gabriel Gunderson <gabe at gundy.org>:
>> > This is the script that I've been using to build my debs. ?I just ran
>> > it on the latest git and it products debs without errors:
>>
>> Thanks. I will test and report back once I have some time.
>> (this can take some days)
>
>
> Hi, i made this script [1] for me, but i made available in the pastebin.
> Works for Ubuntu and Debian, install FreeSWITCH-1.0.6.
>
>
> [1] http://pastebin.com/2Jyh5dxS

Thanks, but I want a debian package (which is easier to uninstall and upgrade).

Mathieu


From alex.deiter at gmail.com  Tue Nov 23 13:54:59 2010
From: alex.deiter at gmail.com (Alex Deiter)
Date: Wed, 24 Nov 2010 00:54:59 +0300
Subject: [Freeswitch-dev] issue ESL-52: event nitifications filters
In-Reply-To: <28A35C47-BB97-4687-9B8B-EE0CFA786F20@jerris.com>
References: <AANLkTik9Nm68gupvjCThcPqPxtUqH4QJqN2t8Zguqxka@mail.gmail.com>
	<28A35C47-BB97-4687-9B8B-EE0CFA786F20@jerris.com>
Message-ID: <AANLkTiktNOBak0Kbbga9JGEM0QDo4RxgC6v2ydkyyq=T@mail.gmail.com>

Hi Michael,

Now FreeSWITCH works fine for me!

Thanks a lot!

2010/11/23 Michael Jerris <mike at jerris.com>:
> please see commit 89dbe0b from yesterday.
>
> On Nov 22, 2010, at 4:20 PM, Alex Deiter wrote:
>
>> Hi,
>>
>> Could you please review issue ESL-52 ?
>> We are used FreeSWITCH with BigBlueButton and found problem with ESL events.
>> Each new filter loses the previous filter. Example:
>>
>> % ./fs_cli -H 127.0.0.1 -p ClueCon
>> ...
>> freeswitch at 127.0.0.1@internal> /event all
>> +OK event listener enabled plain
>>
>> ===> now received ALL events
>> Event-Name: HEARTBEAT
>> Event-Name: RE_SCHEDULE
>> Event-Name: HEARTBEAT
>> Event-Name: RE_SCHEDULE
>>
>> ===> add filter
>> ===> now received ONLY HEARTBEAT events
>> freeswitch at 127.0.0.1@internal> /filter Event-Name HEARTBEAT
>> +OK filter added. [Event-Name]=[HEARTBEAT]
>>
>> ===> add another filter
>> ===> now FreeSWITCH forgotten about the previous filter and used only
>> last filter
>> freeswitch at 127.0.0.1@internal> /filter Event-Name RE_SCHEDULE
>> +OK filter added. [Event-Name]=[RE_SCHEDULE]
>> Event-Name: RE_SCHEDULE
>> Event-Name: RE_SCHEDULE
>> Event-Name: RE_SCHEDULE
>>
>> This behavior was noted after FreeSWITCH snapshot 12-Nov-2010
>>
>> FreeSWITCH snapshot 15-Oct-2010 works as it should: each new filter
>> was added to the existing filters.
>
>
> _______________________________________________
> 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
>



-- 
--
Alex Deiter


From chat2jesse at gmail.com  Tue Nov 23 19:56:35 2010
From: chat2jesse at gmail.com (jesse)
Date: Tue, 23 Nov 2010 19:56:35 -0800
Subject: [Freeswitch-dev] please shed me light on code reading
In-Reply-To: <C9E0C9E5-7805-42D3-8B5E-7424A6E77242@jerris.com>
References: <AANLkTinoBQJwiy_h0ygY+RsA6zU0EfzJBLCPhVmD_ved@mail.gmail.com>
	<AANLkTi=MG0PbjT1s8A=LCAdYOpGeqM0TJj2xSONT+QmL@mail.gmail.com>
	<AANLkTikp-hYSVwq0G5KYZfkn+z1baYmuabUWU2j52teT@mail.gmail.com>
	<C9E0C9E5-7805-42D3-8B5E-7424A6E77242@jerris.com>
Message-ID: <AANLkTin7tfXnGFEOhd3MycxnaEYVARXvNEhn2ssJOU5o@mail.gmail.com>

Michael,

   my question is more more specific like :
    Can we use   : switch_channel_set_variable(channel, "_userspy_",
(void *) argv[0]);  instead of using private?
   It seems this should serve the same purpose. thus,   I don't see a
reason why one is using private, the other is using variable.

   thanks for pointing out the code process_event is used to support
loopback. I still couldn't figure out why the process_event won't work
on call leg-a, but it will work on call leg-b in the case of loop
back? please illustrate me.

   just one general observation, FS source code really lacks code
comments, it takes tremendous effort to understand the code.  FS dev
community should enforce developer to add more comments for business
logic for readability. This is going to help FS down the road in terms
of sustaining, adapting, and feature enriching.

  -jesse



On Tue, Nov 23, 2010 at 6:39 AM, Michael Jerris <mike at jerris.com> wrote:
> a variable is a channel variable that you can access in the dial-plan, in events, etc. ?A private is a void pointer that we use to store data that is later accessed by name in the application. ?The code you were talking about below was added to support loopback channel:
>
> commit d0a74dd5c445e8a1540ba0a3f465ad394e033193
> Date: ? Mon Apr 26 04:30:10 2010 -0400
> ? ?mod_spy: add support for loopback endpoint (MODAPP-416)
>
>
>
> Mike
>
> On Nov 21, 2010, at 2:36 AM, jesse wrote:
>
>> one more question.
>>
>> switch_channel_set_variable(channel, "spy_uuid", my_uuid);
>> switch_channel_set_private(channel, "_userspy_", (void *) argv[0]);
>>
>>
>> why one is set via rugular variable and one is set via private? ?is
>> that necessary?
>>
>> -jesse
>>
>>
>> On Sat, Nov 20, 2010 at 10:48 PM, jesse <chat2jesse at gmail.com> wrote:
>>> hi,
>>>
>>> ?Any expert can shed me some light on quick code understanding?
>>> thanks in advance!
>>>
>>> suppose I dial 88<1001> to spy on extension user 1001. and 5 minutes
>>> later 1001 calls 1002.
>>>
>>> <extension name="userspy">
>>> ? ?<condition field="destination_number" expression="^88(.*)$|^\*0(.*)$">
>>> ? ? ?<action application="answer"/>
>>> ? ? ?<action application="userspy" data="$1 at 192.168.1.1"/>
>>> ? ?</condition>
>>> ?</extension>
>>>
>>> ?in event_handle(...) of mod_spy.c.
>>> ?-- Whats is the reason that we need to call process_event for both event and
>>> peer_event?
>>> ? ?is not processing event for call leg-a 's bridge enough? it seems
>>> mod_spy.c in 1.0.6 version
>>> doesn't process peer_event.
>>>
>>> static void event_handler(switch_event_t *event)
>>> {
>>> ? ? ? ?if (process_event(event) != SWITCH_STATUS_SUCCESS) {
>>> ? ? ? ? ? ? ? ?const char *peer_uuid = switch_event_get_header(event,
>>> "variable_signal_bond");
>>> ? ? ? ? ? ? ? ?...
>>> ? ? ? ? ? ? ? ?if (peer_event) {
>>> ? ? ? ? ? ? ? ? ? ? ? ?process_event(peer_event);
>>> ? ? ? ? ? ? ? ?}
>>> ? ? ? ?}
>>> }
>>>
>>>
>>> http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/applications/mod_spy/mod_spy.c
>>>
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>
>
> _______________________________________________
> 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 steveayre at gmail.com  Wed Nov 24 04:11:44 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 24 Nov 2010 12:11:44 +0000
Subject: [Freeswitch-dev] How to compile FS debian packages?
In-Reply-To: <AANLkTik=dCf0BrAVqH9izn=MGbWkih8f++zP1JVVATp_@mail.gmail.com>
References: <AANLkTinCit4AS2nbj0Zxr9HasWnpOJjOW3tThefWzzZL@mail.gmail.com>
	<AANLkTino-2iKoKW63EYLq3yxhFM4JRsguxv-1Y=NNCcx@mail.gmail.com>
	<AANLkTinY52rr29nwQkSjuV1mdrEHhLyMEQBRaZ+s7+hf@mail.gmail.com>
	<AANLkTim1PU31+FPy7pBRbRO9Vfjij0kYtEoP_Hgb_nGo@mail.gmail.com>
	<AANLkTik=dCf0BrAVqH9izn=MGbWkih8f++zP1JVVATp_@mail.gmail.com>
Message-ID: <AANLkTinYc6X=15JJWK0qoCdesP0UeqOuoZ43TQVriOuP@mail.gmail.com>

I just use:
dpkg-buildpackage -uc -us
and it works fine.

-Steve


On 23 November 2010 21:20, Mathieu Parent <math.parent at gmail.com> wrote:
> Hello,
>
> 2010/11/17 Rodrigo Lang <rodrigoferreiralang at gmail.com>:
>>> 2010/11/13 Gabriel Gunderson <gabe at gundy.org>:
>>> > This is the script that I've been using to build my debs. ?I just ran
>>> > it on the latest git and it products debs without errors:
>>>
>>> Thanks. I will test and report back once I have some time.
>>> (this can take some days)
>>
>>
>> Hi, i made this script [1] for me, but i made available in the pastebin.
>> Works for Ubuntu and Debian, install FreeSWITCH-1.0.6.
>>
>>
>> [1] http://pastebin.com/2Jyh5dxS
>
> Thanks, but I want a debian package (which is easier to uninstall and upgrade).
>
> Mathieu
>
> _______________________________________________
> 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 msc at freeswitch.org  Wed Nov 24 09:51:32 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 24 Nov 2010 09:51:32 -0800
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Starting Shortly
Message-ID: <AANLkTi=AK7SOu68ZBo+KrW-dviFAWoow6tKpQUbg-x2j@mail.gmail.com>

Hello all,

Here's the agenda for today:
http://wiki.freeswitch.org/wiki/FS_weekly_2010_11_24

It's a bit light so we'll focus on community talk, answering questions and
tag-teaming the documenting of the items that are still outstanding.

-MC
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From leon at scarlet-internet.nl  Thu Nov 25 02:38:43 2010
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Thu, 25 Nov 2010 11:38:43 +0100
Subject: [Freeswitch-dev] nolocal exports from dial plan
In-Reply-To: <80D38AC5-3C8B-46A3-8BD3-380BC1060BD7@avgs.ca>
References: <AANLkTilNniAENoPV8txtfcR7y8A8v5OZvDqAJ04Y8CPd@mail.gmail.com>
	<B4D95578-CD1E-4D4C-8957-55DDAAD2261A@freeswitch.org>
	<AANLkTildaE7QOt_bx-D0Z67l9MT14Ix5c70GEQPHm-mK@mail.gmail.com>
	<AANLkTikRkLWOXMdQ8sWp79PcbSCdxR9N1V9wWHU_RCj_@mail.gmail.com>
	<80D38AC5-3C8B-46A3-8BD3-380BC1060BD7@avgs.ca>
Message-ID: <23AF5AFA-6B96-402A-B4A8-47C0CEEDFE63@scarlet-internet.nl>

Hi Mathieu,

Is there any update on this issue ? I stumbled onto it just now.

I'm doing something like this:

1 - inbound context: set a-leg identity_id as channel variable  
identity_id
2 - lookup context: set b-leg identity_id as channel variable  
nolocal:identity_id
3 - outbound context: use nolocal:identity_id in condition to do some  
actions and eventually bridge
4 - generate separate a- and b-leg xml cdr's

My xml cdr parser breaks on the nolocal: prefix on some variables in  
the a-leg cdr. (4)

One other issue I'm having is that it seems I can't use nolocal:  
channel variables in xml dialplan conditions. (3) This is probably  
because the colon is used in for example ${foo:5} to get the 5 first  
characters, so I can't do something like this:

<extension name="foo">
	<condition field="${nolocal:identity_id}" expression="bar">
		<action etc....

Is there any solution to doing this ?

Thanks a lot,

Leon




On Jun 23, 2010, at 5:49 AM, Mathieu Rene wrote:

> That is completely unrelated. I added that function so one can all  
> switch_channel_export_variable() instead of  
> switch_channel_set_variable() followed by getting "exports" and  
> appending the name of the new variable.
>
> As for nolocal, it actually saves the variable with a prefix so it  
> has no effect on the A-leg, what would be the proper fix for the xml  
> without completely wiping the entry away? Its sometimes useful to  
> know something was exported for the B-leg. Can we escape the : ?
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 2010-06-22, at 7:27 PM, Rupa Schomaker wrote:
>
>> commit 2c830f84aa8e5a5b05c55c5f02ae0025c4986bd2
>> Author: Mathieu Rene <mrene at avgs.ca>
>> Date:   Wed Apr 7 15:02:00 2010 -0400
>>
>>     remove switch_channel_export_variable's nolocal argument, more  
>> confusing than anything else
>>
>>
>> On Tue, Jun 22, 2010 at 5:26 PM, Dino Korah <dckorah at gmail.com>  
>> wrote:
>> Thanks Brian,
>>
>> Then by the looks, there is something broken.
>>
>> When a variable (say MyAppName) is exported as "nolocal", it  
>> appears in A leg CDR as "<nolocal:MyAppName>HelloWorldApp</ 
>> nolocal:MyAppName>", which then breaks the XML (those strict  
>> parsers).
>>
>> In the line I pointed earlier; as I understand, if the variable  
>> being exported from dial_plan is set "nolocal" should be ignored  
>> for A.
>>
>> I can fix and checkin or send a patch; but not sure if it will  
>> break anything else.
>>
>> "dk"
>>
>>
>>
>> On 22 June 2010 17:25, Brian West <brian at freeswitch.org> wrote:
>> nolocal is just like setting inside {} so its not on the A leg but  
>> only on the B leg.
>>
>> /b
>>
>> On Jun 16, 2010, at 4:49 AM, Dino Korah wrote:
>>
>> > Guys,
>> >
>> > When I export a variable as nolocal, is it intended to prefix  
>> "nolocal:" for that variable on A, when that appears in the CDR?
>> >
>> > Or is it a glitch?
>> >
>> > Ref: src/mod/applications/mod_dptools/mod_dptools.c line 889
>> >
>> > Here if you check, the export seems to ignore the local variable  
>> altogether.
>> >
>> > I am quite new to freeswitch; any help much appreciated.
>> >
>> > Many thanks,
>> > "dk"
>>
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> dev
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> dev
>> http://www.freeswitch.org
>>
>>
>>
>>
>> -- 
>> -Rupa
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> dev
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org

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From mrene_lists at avgs.ca  Thu Nov 25 07:30:48 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 25 Nov 2010 10:30:48 -0500
Subject: [Freeswitch-dev] please shed me light on code reading
In-Reply-To: <AANLkTin7tfXnGFEOhd3MycxnaEYVARXvNEhn2ssJOU5o@mail.gmail.com>
References: <AANLkTinoBQJwiy_h0ygY+RsA6zU0EfzJBLCPhVmD_ved@mail.gmail.com>
	<AANLkTi=MG0PbjT1s8A=LCAdYOpGeqM0TJj2xSONT+QmL@mail.gmail.com>
	<AANLkTikp-hYSVwq0G5KYZfkn+z1baYmuabUWU2j52teT@mail.gmail.com>
	<C9E0C9E5-7805-42D3-8B5E-7424A6E77242@jerris.com>
	<AANLkTin7tfXnGFEOhd3MycxnaEYVARXvNEhn2ssJOU5o@mail.gmail.com>
Message-ID: <04F7E6A3-0569-4C26-B5FC-4A2655936D33@avgs.ca>

Hi,

switch_channel_set_variable() is limited to saving strings, if you want to save other data structures, you have to use switch_channel_set_private().

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




On 2010-11-23, at 10:56 PM, jesse wrote:

> Michael,
> 
>   my question is more more specific like :
>    Can we use   : switch_channel_set_variable(channel, "_userspy_",
> (void *) argv[0]);  instead of using private?
>   It seems this should serve the same purpose. thus,   I don't see a
> reason why one is using private, the other is using variable.
> 
>   thanks for pointing out the code process_event is used to support
> loopback. I still couldn't figure out why the process_event won't work
> on call leg-a, but it will work on call leg-b in the case of loop
> back? please illustrate me.
> 
>   just one general observation, FS source code really lacks code
> comments, it takes tremendous effort to understand the code.  FS dev
> community should enforce developer to add more comments for business
> logic for readability. This is going to help FS down the road in terms
> of sustaining, adapting, and feature enriching.
> 
>  -jesse
> 
> 
> 
> On Tue, Nov 23, 2010 at 6:39 AM, Michael Jerris <mike at jerris.com> wrote:
>> a variable is a channel variable that you can access in the dial-plan, in events, etc.  A private is a void pointer that we use to store data that is later accessed by name in the application.  The code you were talking about below was added to support loopback channel:
>> 
>> commit d0a74dd5c445e8a1540ba0a3f465ad394e033193
>> Date:   Mon Apr 26 04:30:10 2010 -0400
>>    mod_spy: add support for loopback endpoint (MODAPP-416)
>> 
>> 
>> 
>> Mike
>> 
>> On Nov 21, 2010, at 2:36 AM, jesse wrote:
>> 
>>> one more question.
>>> 
>>> switch_channel_set_variable(channel, "spy_uuid", my_uuid);
>>> switch_channel_set_private(channel, "_userspy_", (void *) argv[0]);
>>> 
>>> 
>>> why one is set via rugular variable and one is set via private?  is
>>> that necessary?
>>> 
>>> -jesse
>>> 
>>> 
>>> On Sat, Nov 20, 2010 at 10:48 PM, jesse <chat2jesse at gmail.com> wrote:
>>>> hi,
>>>> 
>>>>  Any expert can shed me some light on quick code understanding?
>>>> thanks in advance!
>>>> 
>>>> suppose I dial 88<1001> to spy on extension user 1001. and 5 minutes
>>>> later 1001 calls 1002.
>>>> 
>>>> <extension name="userspy">
>>>>    <condition field="destination_number" expression="^88(.*)$|^\*0(.*)$">
>>>>      <action application="answer"/>
>>>>      <action application="userspy" data="$1 at 192.168.1.1"/>
>>>>    </condition>
>>>>  </extension>
>>>> 
>>>>  in event_handle(...) of mod_spy.c.
>>>>  -- Whats is the reason that we need to call process_event for both event and
>>>> peer_event?
>>>>    is not processing event for call leg-a 's bridge enough? it seems
>>>> mod_spy.c in 1.0.6 version
>>>> doesn't process peer_event.
>>>> 
>>>> static void event_handler(switch_event_t *event)
>>>> {
>>>>        if (process_event(event) != SWITCH_STATUS_SUCCESS) {
>>>>                const char *peer_uuid = switch_event_get_header(event,
>>>> "variable_signal_bond");
>>>>                ...
>>>>                if (peer_event) {
>>>>                        process_event(peer_event);
>>>>                }
>>>>        }
>>>> }
>>>> 
>>>> 
>>>> http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/applications/mod_spy/mod_spy.c
>>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>> 
> 
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org



From ktk at netlabs.org  Thu Nov 25 02:29:52 2010
From: ktk at netlabs.org (Adrian Gschwend)
Date: Thu, 25 Nov 2010 11:29:52 +0100
Subject: [Freeswitch-dev] Compiling on FreeBSD
Message-ID: <icldr0$e9s$1@dough.gmane.org>

Hi group,

I use Freeswitch on FreeBSD for ages now, always trunk which I compile
every few weeks. Unfortunately it doesn't compile anymore for a while,
it fails in mod_spandsp

I checked the docs according to:

http://wiki.freeswitch.org/wiki/Installation_Guide#FreeBSD
http://wiki.freeswitch.org/wiki/Mod_spandsp

so I compiled both graphics/tiff-4.0.0 and graphics/jpeg-8_3, I'm on
8.1-RELEASE

But it doesn't find jpeg, like the possible error stated on Mod_spandsp
page. But unless I completely miss something here it *should* be
installed (in terms of the port is definitely there).

--
[...]
Creating mod_spandsp.la
/usr/bin/ld: cannot find -ljpeg
quiet_libtool: link: gcc -shared  .libs/mod_spandsp_la-mod_spandsp.o
.libs/mod_spandsp_la-udptl.o .libs/mod_spandsp_la-mod_spandsp_fax.o
.libs/mod_spandsp_la-mod_spandsp_dsp.o
.libs/mod_spandsp_la-mod_spandsp_codecs.o   -Wl,-rpath
-Wl,/usr/local/src/freeswitch/.libs -Wl,-rpath
-Wl,/usr/local/freeswitch/lib
-L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs
-L/usr/local/src/freeswitch/libs/apr/.libs
-L/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs -ljpeg
/usr/local/src/freeswitch/.libs/libfreeswitch.so
-L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib
/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a
/usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a -lcrypt -lpthread
/usr/local/src/freeswitch/libs/spandsp/src/.libs/libspandsp.a
-L/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff
/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -lz
-lm -lssl -lcrypto -lncurses    -Wl,-soname -Wl,mod_spandsp.so -o
.libs/mod_spandsp.so
gmake[4]: *** [mod_spandsp.la] Error 1
gmake[3]: *** [mod_spandsp-all] Error 1
gmake[2]: *** [all-recursive] Error 1
--

Any FreeBSD user which can compile that ATM? If so, what did I miss?

Will update the wiki once I get it to work.

cu

Adrian

-- 
Adrian Gschwend
@ netlabs.org

ktk [a t] netlabs.org
-------
Open Source Project
http://www.netlabs.org