From anthony.minessale at gmail.com  Tue Mar  1 00:19:37 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Feb 2011 15:19:37 -0600
Subject: [Freeswitch-users] Polycom G722 negotiation
In-Reply-To: <5AB8CE7C-5283-4585-8DF8-856A3822F3BD@5ninesolutions.com>
References: <AE216221-5892-4337-8985-1889410A866F@5ninesolutions.com>
	<2867CE13-6EED-4FAF-9DF1-49451D164721@freeswitch.org>
	<0ABF546B-4000-4936-AD22-ED6B956277D5@5ninesolutions.com>
	<1298907264976-6073705.post@n2.nabble.com>
	<1F4FD420-3CE9-4601-9D44-80C8BEEF9707@5ninesolutions.com>
	<AANLkTikFCLzNvqpHEO4qtCyFDQOTZ0CG5pOw+SJD=2Yz@mail.gmail.com>
	<5AB8CE7C-5283-4585-8DF8-856A3822F3BD@5ninesolutions.com>
Message-ID: <AANLkTik0jy=VDy2p4XnAiAwJkT6jX=PAY1W6FgRV7cDC@mail.gmail.com>

It seems like a bug in the polycom, hold/unhold should maintain the
original SDP already negotiated.
I would try updating the firmware.
I suspect polycom is not really offering G.722 but is saying they want
to in the sdp.
Then its ignoring the G.722 because it's still expecting ULAW.

We might have to make a param that does not allow codec re-negotiation
once a call is up as a hack to deal with this.


On Mon, Feb 28, 2011 at 1:22 PM, Spencer Thomason
<spencer at 5ninesolutions.com> wrote:
> No, Freeswitch is processing the media, proxy media is not on either.
>
>
> On Feb 28, 2011, at 7:15 PM, Anthony Minessale wrote:
>
>> Are you doing this with bypass_media=true?
>>
>>
>> On Mon, Feb 28, 2011 at 9:55 AM, Spencer Thomason
>> <spencer at 5ninesolutions.com> wrote:
>>> The devices are using 20ms G722. ?The strange thing is that if the
>>> call starts out as G722, everything works totally fine. ?The problem
>>> arises when the phone has G722 first in its prefered codec, the FS
>>> profile is set to generous, and a call starts out as PCMU. ?The audio
>>> is fine until someone places a call on hold. ?After taking a call off
>>> hold, when the FS profile is set to generous, FS honors the Polycoms
>>> preference for G722 and then tries to switch to G722. ?At this point
>>> there is no audio. ?If the sophia profile is set to greedy, as
>>> expected Freeswitch does not allow the switch to G722 and there are
>>> no
>>> audio problems. ?This only happens when a call starts out in
>>> something
>>> other than G722 and then tries to switch. ?I.e. ?client ---G722--->
>>> FS
>>> ---PCMU---> Provider ?works and client ---G722---> FS ---G722--->
>>> client works but provider ---PCMU---> FS ---PCMU---> client and
>>> then a
>>> call is placed on hold and resumed a G722 does not.
>>>
>>> Spencer
>>>
>>> On Feb 28, 2011, at 3:34 PM, peely wrote:
>>>
>>>> What ptime is your device offering? FreeSWITCH only supports 20ms
>>>> G722, I get
>>>> clipped audio for example when G722 30ms is offered on Snoms, but
>>>> they work
>>>> perfectly when I force them to 20ms.
>>>>
>>>> --
>>>> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Polycom-G722-negotiation-tp6063481p6073705.html
>>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 mthakershi at gmail.com  Tue Mar  1 00:22:04 2011
From: mthakershi at gmail.com (Malay Thakershi)
Date: Mon, 28 Feb 2011 15:22:04 -0600
Subject: [Freeswitch-users] disabling ptime warning message
In-Reply-To: <AANLkTimTKLJQ+cE9r5t2fRR+riDo-DPi+7FQLJecoc9M@mail.gmail.com>
References: <AANLkTim97P5=MJgKq3ncDmbTiXkP63CBoALRD6o+b-Uw@mail.gmail.com>
	<AANLkTikw3TLW8bo39WRX6wN44vhcGSDBBTS_QffqBTwc@mail.gmail.com>
	<AANLkTi=FGWs2bs3EFvaMTTyG0=0inwe5z2UfByctX-g1@mail.gmail.com>
	<AANLkTindiPgTH65=0MwBgdviyWByQzmu1=01MYUy_fXy@mail.gmail.com>
	<AANLkTi=cFCJssphqmji_8LTXTg3ZycGfswMM77hmkie9@mail.gmail.com>
	<AANLkTimTKLJQ+cE9r5t2fRR+riDo-DPi+7FQLJecoc9M@mail.gmail.com>
Message-ID: <AANLkTi=A-qZXz23nBgFz+LgAKCsqOmFFfgs8zm8k8_Re@mail.gmail.com>

http://pastebin.com/CCwmAfZK

<http://pastebin.com/CCwmAfZK>I noticed that LPC warning only comes for
outbound calls I initiate using "Originate" API.

Malay

On Mon, Feb 28, 2011 at 2:04 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> That's L16 not LPC, you need L16 to play files.
> Why don't you just put the whole log of the call on pastebin.
>
>
> On Mon, Feb 28, 2011 at 1:53 PM, Malay Thakershi <mthakershi at gmail.com>
> wrote:
> > Following is a section from the log:
> > ---------------------------
> > 2011-02-28 13:36:08.294241 [DEBUG] sofia_glue.c:2760 Set Codec
> > sofia/sipinterface_1/##########@##.##.##.## PCMU/8000 20 ms 160 samples
> > 64000 bits
> > 2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2298 OPEN TTS
> > cepstral
> > 2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2307 Raw Codec
> > Activated
> > 2011-02-28 13:36:15.017444 [DEBUG] switch_ivr_play_say.c:1996 Speaking
> text:
> > <break strength='medium'/>We must verify your identity.
> > 2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:2188 done
> speaking
> > text
> > 2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:1244 Codec
> > Activated L16 at 8000hz 1 channels 20ms
> > 2011-02-28 13:36:25.530443 [DEBUG] switch_ivr_play_say.c:1581 done
> playing
> > file
> > 2011-02-28 13:36:25.530443 [DEBUG] switch_ivr.c:1135 Codec Activated
> > L16 at 8000hz 1 channels 20ms
> > 2011-02-28 13:36:30.549653 [DEBUG] switch_ivr_play_say.c:1244 Codec
> > Activated L16 at 8000hz 1 channels 20ms
> > ---------------------------
> > So I see when I play a file (using StreamFile / PAGD), it activates L16,
> > which the wiki pages says is not recommended. So should I deactivate it?
> If
> > so, how?
> > Now, I have not done any setting out of default / ordinary that comes
> with
> > the build. I am playing WAV file that is generated by Cepstral SWIFT
> command
> > line tool (text to WAV). The file format is "Wave PCM signed 16 bit, 8000
> > Hz, 128 kbps, mono".
> > Thank you for help so far.
> > Malay
> > On Mon, Feb 28, 2011 at 12:50 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> look at your SIP traffic and console log.
> >>
> >> enter "sofia global siptrace on" followed by "console loglevel debug"
> >> at the cli and make the call.
> >>
> >>
> >> On Mon, Feb 28, 2011 at 12:47 PM, Malay Thakershi <mthakershi at gmail.com
> >
> >> wrote:
> >> > I have no idea where to look for this setting.
> >> > This is in modules.conf.xml
> >> >     <!-- Codec Interfaces -->
> >> >     <load module="mod_spandsp"/>
> >> >     <!--<load module="mod_voipcodecs"/>-->
> >> >     <load module="mod_g723_1"/>
> >> >     <load module="mod_g729"/>
> >> >     <load module="mod_amr"/>
> >> >     <load module="mod_ilbc"/>
> >> >     <load module="mod_speex"/>
> >> >     <load module="mod_h26x"/>
> >> >     <load module="mod_siren"/>
> >> >     <!--<load module="mod_celt"/>-->
> >> >     <!--<load module="mod_opus"/>-->
> >> > Apart from settings I posted in my previous post, where else to look
> for
> >> > disabling LPC?
> >> > Malay
> >> > On Mon, Feb 28, 2011 at 12:05 PM, Anthony Minessale
> >> > <anthony.minessale at gmail.com> wrote:
> >> >>
> >> >> is your inbound call using LPC? you don't want to be using LPC and
> >> >> expect anything to sound good that's for sure.
> >> >> It would not just magically say that unless something you are doing
> has
> >> >> LPC?
> >> >>
> >> >>
> >> >> On Mon, Feb 28, 2011 at 10:24 AM, Malay Thakershi
> >> >> <mthakershi at gmail.com>
> >> >> wrote:
> >> >> > Hello,
> >> >> > I updated to the latest FS version last week.
> >> >> > I started getting the following warning when speech / sound is
> played
> >> >> > on
> >> >> > the
> >> >> > call.
> >> >> > "2011-02-28 10:15:55.985930 [WARNING] sofia_glue.c:213 Codec LPC
> >> >> > payload
> >> >> > 7
> >> >> > added to sdp wanting ptime 90 but it's already 20 (G7221:115:20),
> >> >> > disabling
> >> >> > ptime."
> >> >> > I read sections on codecs and negotiations.
> >> >> > Following are the settings from vars.xml (I have not changed them):
> >> >> >   <X-PRE-PROCESS cmd="set"
> >> >> >
> >> >> > data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h
> ,G722,PCMU,PCMA,GSM"/>
> >> >> >   <X-PRE-PROCESS cmd="set"
> >> >> > data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
> >> >> > Also, there is no codec related setting in sip_profiles files
> >> >> > and sofia.conf.xml file.
> >> >> > I am playing audio files using Cepstral TTS during the call.
> >> >> > Can someone please help me understand these settings? And if they
> are
> >> >> > appropriate?
> >> >> > Thank you.
> >> >> > Malay
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > 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-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > 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-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From spencer at 5ninesolutions.com  Tue Mar  1 00:37:11 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Mon, 28 Feb 2011 21:37:11 +0000
Subject: [Freeswitch-users] Polycom G722 negotiation
In-Reply-To: <AANLkTik0jy=VDy2p4XnAiAwJkT6jX=PAY1W6FgRV7cDC@mail.gmail.com>
References: <AE216221-5892-4337-8985-1889410A866F@5ninesolutions.com>
	<2867CE13-6EED-4FAF-9DF1-49451D164721@freeswitch.org>
	<0ABF546B-4000-4936-AD22-ED6B956277D5@5ninesolutions.com>
	<1298907264976-6073705.post@n2.nabble.com>
	<1F4FD420-3CE9-4601-9D44-80C8BEEF9707@5ninesolutions.com>
	<AANLkTikFCLzNvqpHEO4qtCyFDQOTZ0CG5pOw+SJD=2Yz@mail.gmail.com>
	<5AB8CE7C-5283-4585-8DF8-856A3822F3BD@5ninesolutions.com>
	<AANLkTik0jy=VDy2p4XnAiAwJkT6jX=PAY1W6FgRV7cDC@mail.gmail.com>
Message-ID: <C72FE323-9B29-43F3-A17E-A327DAD3B07B@5ninesolutions.com>

Yes, that what I thought.  I'm running the latest firmware, 3.3.1.   
I'll report it to them as its clearly offering a different SDP after  
the unhold.  Yes it might be nice if Freeswitch refused to switch  
codec on a call that is up even if the negotiation is set to generous.

Thanks for your help,
Spencer



On Feb 28, 2011, at 9:19 PM, Anthony Minessale wrote:

> It seems like a bug in the polycom, hold/unhold should maintain the
> original SDP already negotiated.
> I would try updating the firmware.
> I suspect polycom is not really offering G.722 but is saying they want
> to in the sdp.
> Then its ignoring the G.722 because it's still expecting ULAW.
>
> We might have to make a param that does not allow codec re-negotiation
> once a call is up as a hack to deal with this.
>
>
> On Mon, Feb 28, 2011 at 1:22 PM, Spencer Thomason
> <spencer at 5ninesolutions.com> wrote:
>> No, Freeswitch is processing the media, proxy media is not on either.
>>
>>
>> On Feb 28, 2011, at 7:15 PM, Anthony Minessale wrote:
>>
>>> Are you doing this with bypass_media=true?
>>>
>>>
>>> On Mon, Feb 28, 2011 at 9:55 AM, Spencer Thomason
>>> <spencer at 5ninesolutions.com> wrote:
>>>> The devices are using 20ms G722.  The strange thing is that if the
>>>> call starts out as G722, everything works totally fine.  The  
>>>> problem
>>>> arises when the phone has G722 first in its prefered codec, the FS
>>>> profile is set to generous, and a call starts out as PCMU.  The  
>>>> audio
>>>> is fine until someone places a call on hold.  After taking a call  
>>>> off
>>>> hold, when the FS profile is set to generous, FS honors the  
>>>> Polycoms
>>>> preference for G722 and then tries to switch to G722.  At this  
>>>> point
>>>> there is no audio.  If the sophia profile is set to greedy, as
>>>> expected Freeswitch does not allow the switch to G722 and there are
>>>> no
>>>> audio problems.  This only happens when a call starts out in
>>>> something
>>>> other than G722 and then tries to switch.  I.e.  client ---G722--->
>>>> FS
>>>> ---PCMU---> Provider  works and client ---G722---> FS ---G722--->
>>>> client works but provider ---PCMU---> FS ---PCMU---> client and
>>>> then a
>>>> call is placed on hold and resumed a G722 does not.
>>>>
>>>> Spencer
>>>>
>>>> On Feb 28, 2011, at 3:34 PM, peely wrote:
>>>>
>>>>> What ptime is your device offering? FreeSWITCH only supports 20ms
>>>>> G722, I get
>>>>> clipped audio for example when G722 30ms is offered on Snoms, but
>>>>> they work
>>>>> perfectly when I force them to 20ms.
>>>>>
>>>>> --
>>>>> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Polycom-G722-negotiation-tp6063481p6073705.html
>>>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> 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-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From anthony.minessale at gmail.com  Tue Mar  1 00:47:09 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Feb 2011 15:47:09 -0600
Subject: [Freeswitch-users] disabling ptime warning message
In-Reply-To: <AANLkTi=A-qZXz23nBgFz+LgAKCsqOmFFfgs8zm8k8_Re@mail.gmail.com>
References: <AANLkTim97P5=MJgKq3ncDmbTiXkP63CBoALRD6o+b-Uw@mail.gmail.com>
	<AANLkTikw3TLW8bo39WRX6wN44vhcGSDBBTS_QffqBTwc@mail.gmail.com>
	<AANLkTi=FGWs2bs3EFvaMTTyG0=0inwe5z2UfByctX-g1@mail.gmail.com>
	<AANLkTindiPgTH65=0MwBgdviyWByQzmu1=01MYUy_fXy@mail.gmail.com>
	<AANLkTi=cFCJssphqmji_8LTXTg3ZycGfswMM77hmkie9@mail.gmail.com>
	<AANLkTimTKLJQ+cE9r5t2fRR+riDo-DPi+7FQLJecoc9M@mail.gmail.com>
	<AANLkTi=A-qZXz23nBgFz+LgAKCsqOmFFfgs8zm8k8_Re@mail.gmail.com>
Message-ID: <AANLkTikvOhbJZfpbN9BmLs3eFFuz=uG2jRyKM4J5zYus@mail.gmail.com>

You also need "sofia global siptrace on" so we can see the SIP traffic
in the log.
We have our own pastebin at http://pastebin.freeswitch.org


On Mon, Feb 28, 2011 at 3:22 PM, Malay Thakershi <mthakershi at gmail.com> wrote:
> http://pastebin.com/CCwmAfZK
> I noticed that LPC warning only comes for outbound calls I initiate using
> "Originate" API.
> Malay
>
> On Mon, Feb 28, 2011 at 2:04 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> That's L16 not LPC, you need L16 to play files.
>> Why don't you just put the whole log of the call on pastebin.
>>
>>
>> On Mon, Feb 28, 2011 at 1:53 PM, Malay Thakershi <mthakershi at gmail.com>
>> wrote:
>> > Following is a section from the log:
>> > ---------------------------
>> > 2011-02-28 13:36:08.294241 [DEBUG] sofia_glue.c:2760 Set Codec
>> > sofia/sipinterface_1/##########@##.##.##.## PCMU/8000 20 ms 160 samples
>> > 64000 bits
>> > 2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2298 OPEN TTS
>> > cepstral
>> > 2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2307 Raw Codec
>> > Activated
>> > 2011-02-28 13:36:15.017444 [DEBUG] switch_ivr_play_say.c:1996 Speaking
>> > text:
>> > <break strength='medium'/>We must verify your identity.
>> > 2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:2188 done
>> > speaking
>> > text
>> > 2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:1244 Codec
>> > Activated L16 at 8000hz 1 channels 20ms
>> > 2011-02-28 13:36:25.530443 [DEBUG] switch_ivr_play_say.c:1581 done
>> > playing
>> > file
>> > 2011-02-28 13:36:25.530443 [DEBUG] switch_ivr.c:1135 Codec Activated
>> > L16 at 8000hz 1 channels 20ms
>> > 2011-02-28 13:36:30.549653 [DEBUG] switch_ivr_play_say.c:1244 Codec
>> > Activated L16 at 8000hz 1 channels 20ms
>> > ---------------------------
>> > So I see when I play a file (using StreamFile / PAGD), it activates L16,
>> > which the wiki pages says is not recommended. So should I deactivate it?
>> > If
>> > so, how?
>> > Now, I have not done any setting out of default / ordinary that comes
>> > with
>> > the build. I am playing WAV file that is generated by Cepstral SWIFT
>> > command
>> > line tool (text to WAV). The file format is "Wave PCM signed 16 bit,
>> > 8000
>> > Hz, 128 kbps, mono".
>> > Thank you for help so far.
>> > Malay
>> > On Mon, Feb 28, 2011 at 12:50 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> look at your SIP traffic and console log.
>> >>
>> >> enter "sofia global siptrace on" followed by "console loglevel debug"
>> >> at the cli and make the call.
>> >>
>> >>
>> >> On Mon, Feb 28, 2011 at 12:47 PM, Malay Thakershi
>> >> <mthakershi at gmail.com>
>> >> wrote:
>> >> > I have no idea where to look for this setting.
>> >> > This is in modules.conf.xml
>> >> > ?? ?<!-- Codec Interfaces -->
>> >> > ?? ?<load module="mod_spandsp"/>
>> >> > ?? ?<!--<load module="mod_voipcodecs"/>-->
>> >> > ?? ?<load module="mod_g723_1"/>
>> >> > ?? ?<load module="mod_g729"/>
>> >> > ?? ?<load module="mod_amr"/>
>> >> > ?? ?<load module="mod_ilbc"/>
>> >> > ?? ?<load module="mod_speex"/>
>> >> > ?? ?<load module="mod_h26x"/>
>> >> > ?? ?<load module="mod_siren"/>
>> >> > ?? ?<!--<load module="mod_celt"/>-->
>> >> > ?? ?<!--<load module="mod_opus"/>-->
>> >> > Apart from settings I posted in my previous post, where else to look
>> >> > for
>> >> > disabling LPC?
>> >> > Malay
>> >> > On Mon, Feb 28, 2011 at 12:05 PM, Anthony Minessale
>> >> > <anthony.minessale at gmail.com> wrote:
>> >> >>
>> >> >> is your inbound call using LPC? you don't want to be using LPC and
>> >> >> expect anything to sound good that's for sure.
>> >> >> It would not just magically say that unless something you are doing
>> >> >> has
>> >> >> LPC?
>> >> >>
>> >> >>
>> >> >> On Mon, Feb 28, 2011 at 10:24 AM, Malay Thakershi
>> >> >> <mthakershi at gmail.com>
>> >> >> wrote:
>> >> >> > Hello,
>> >> >> > I updated to the latest FS version last week.
>> >> >> > I started getting the following warning when speech / sound is
>> >> >> > played
>> >> >> > on
>> >> >> > the
>> >> >> > call.
>> >> >> > "2011-02-28 10:15:55.985930 [WARNING] sofia_glue.c:213 Codec LPC
>> >> >> > payload
>> >> >> > 7
>> >> >> > added to sdp wanting ptime?90 but it's already 20 (G7221:115:20),
>> >> >> > disabling
>> >> >> > ptime."
>> >> >> > I read sections on codecs and negotiations.
>> >> >> > Following are the settings from vars.xml (I have not changed
>> >> >> > them):
>> >> >> > ??<X-PRE-PROCESS cmd="set"
>> >> >> >
>> >> >> >
>> >> >> > data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>> >> >> > ??<X-PRE-PROCESS cmd="set"
>> >> >> > data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>> >> >> > Also, there is no codec related setting in sip_profiles files
>> >> >> > and?sofia.conf.xml file.
>> >> >> > I am playing audio files using Cepstral TTS during the call.
>> >> >> > Can someone please help me understand these settings? And if they
>> >> >> > are
>> >> >> > appropriate?
>> >> >> > Thank you.
>> >> >> > Malay
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > 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-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > 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-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From freeswitch at cartissolutions.com  Tue Mar  1 00:54:00 2011
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Mon, 28 Feb 2011 15:54:00 -0600
Subject: [Freeswitch-users] mod_dingaling and google voice
In-Reply-To: <AANLkTimUr3=w_fj_k643ChW93Q_YTm63R_qFyBO9EoQV@mail.gmail.com>
References: <AANLkTin_dFj9ygx3TAmExeN9-pDFNCuZPGZ_Uc2d7o1W@mail.gmail.com>
	<AANLkTimUr3=w_fj_k643ChW93Q_YTm63R_qFyBO9EoQV@mail.gmail.com>
Message-ID: <4D6C1978.8040702@cartissolutions.com>

What FS build are you running?  I'd like to compare notes with you on 
your config, as I'm experiencing the RTP port 0 issue with FreeSWITCH 
Version 1.0.head (git-0d8e945 2011-02-16 08-57-43 -0600).  What about 
inbound?  Do you have that set up?

Yossi Neiman
Cartis Solutions, Inc. - http://www.cartissolutions.com


On 02/28/2011 12:54 AM, envelopes envelopes wrote:
> |never mind. add this line <action application="ring_ready"/> and 
> restart FS
>
> fixed the issue.
>
>
> |
> On Sun, Feb 27, 2011 at 10:30 PM, envelopes envelopes 
> <sunwood360 at gmail.com <mailto:sunwood360 at gmail.com>> wrote:
>
>     Now I am able to use GV for outbound dialing. However, I don't
>     hear any ringback or not sure whether the other party has answered
>     the call.
>     is there any config variable to set up so that I will be notified
>     if my call is answered?
>
>     thanks!
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-------------- next part --------------
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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110228/6685542f/attachment.html 

From mauritz.lovgren at hotmail.com  Tue Mar  1 00:55:53 2011
From: mauritz.lovgren at hotmail.com (=?utf-8?Q?Mauritz_L=C3=B8vgren?=)
Date: Mon, 28 Feb 2011 22:55:53 +0100
Subject: [Freeswitch-users] mod_event_socket message pizza received from
	BSD machines
In-Reply-To: <AANLkTim81z7pa50C8VzNToTY+9=AeK8umpyBr3eeKZoF@mail.gmail.com>
References: <SNT107-DS1630BE652914A48CC2BA01E6DE0@phx.gbl>
	<549CFEF87AEDE841A38E9D15EAB4C04C58ADB727D2@cooper>
	<AANLkTim81z7pa50C8VzNToTY+9=AeK8umpyBr3eeKZoF@mail.gmail.com>
Message-ID: <BLU0-SMTP62A324D84F2EB44090EF58E6DE0@phx.gbl>

We ran the load test this morning on latest snapshot.tar.gzip on FreeBSD 8.1 64-bit and on latest nightly windows 64-bit binaries. Works well on windows, pizza on BSD / OSX. Don't think it is related to ESL-56, since we handle the case where events arrive before replies, our problem is that event and reply seems to be 'merged' corrupting the protocol.

Mauritz

28. feb. 2011 kl. 19:08 Anthony Minessale <anthony.minessale at gmail.com>:

> indeed, this is almost guaranteed to be fixed in latest HEAD.
> 
> 
> On Mon, Feb 28, 2011 at 9:36 AM, Peter Olsson
> <peter.olsson at visionutveckling.se> wrote:
>> Are you using git HEAD as of the last couple of days? I know there was some changes for this last week related to ESL-56 - at least I saw some commits related to it... Or it this not at all related to ESL lib?
>> 
>> /Peter
>> ________________________________________
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Mauritz L?vgren [mauritz.lovgren at hotmail.com]
>> Skickat: den 28 februari 2011 15:26
>> Till: FreeSWITCH Users Help
>> ?mne: [Freeswitch-users] mod_event_socket message pizza received from BSD       machines
>> 
>> Hi,
>> 
>> We are experiencing weird behaviour when using socket connection to mod_event_socket while running FS on FreeBSD and Mac OSX.
>> 
>> We send a lot of bgapi and api commands to the FreeSwitch while controlling and monitoring hundreds of sessions.
>> 
>> We have created a simple load-test that displays the problem we face.
>> The load-test simply fires the following command continuosly to freeswitch: ?bgapi sofia status?, generally as fast as freeswitch can receive it.
>> 
>> The following occurs, but only on FreeBSD or MacOSX (10.6.6) (both 64-bit) while using latest version of FreeSwitch:
>> 
>> - Replies and events are inter-mixed on the receive stream (the output from freeswitch), causing protocol errors in the receiving client.
>> 
>> This does _not_ happen on CentOS or Windows, so we wonder what could be causing this. Is there a problem with threading or socket libs here?
>> We assume that FreeSwitch protect a mod_event_socket inbound connection output stream by locking (or other means) to make sure not any two messages are written ?simultaneously? causing garbage in the receiving end?
>> 
>> Regards,
>> Mauritz Lovgren
>> System Architect
>> IPLink Inc.
>> http://www.iplink.no
>> 
>> 
>> !DSPAM:4d6bb3a032761738012832!
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From anthony.minessale at gmail.com  Tue Mar  1 01:06:07 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Feb 2011 16:06:07 -0600
Subject: [Freeswitch-users] mod_event_socket message pizza received from
 BSD machines
In-Reply-To: <BLU0-SMTP62A324D84F2EB44090EF58E6DE0@phx.gbl>
References: <SNT107-DS1630BE652914A48CC2BA01E6DE0@phx.gbl>
	<549CFEF87AEDE841A38E9D15EAB4C04C58ADB727D2@cooper>
	<AANLkTim81z7pa50C8VzNToTY+9=AeK8umpyBr3eeKZoF@mail.gmail.com>
	<BLU0-SMTP62A324D84F2EB44090EF58E6DE0@phx.gbl>
Message-ID: <AANLkTi=fNKbcxTaRTGjaQR54=RBSRC=pvn+cMNqNSGwq@mail.gmail.com>

I don't know what latest snapshot you are talking about but I am
saying to try GIT HEAD by checking it out and building.
When you say "we handle" do you mean you wrote your own client interface?

The exact symptom you describe is a bug recently fixed in the ESL
client code due to a missing null termination on packets read into the
buffer.  If you are not using libesl, there is no way the socket
itself should be messed up.

When you say only BSD or MAC do you mean running the server or the client?




2011/2/28 Mauritz L?vgren <mauritz.lovgren at hotmail.com>:
> We ran the load test this morning on latest snapshot.tar.gzip on FreeBSD 8.1 64-bit and on latest nightly windows 64-bit binaries. Works well on windows, pizza on BSD / OSX. Don't think it is related to ESL-56, since we handle the case where events arrive before replies, our problem is that event and reply seems to be 'merged' corrupting the protocol.
>
> Mauritz
>
> 28. feb. 2011 kl. 19:08 Anthony Minessale <anthony.minessale at gmail.com>:
>
>> indeed, this is almost guaranteed to be fixed in latest HEAD.
>>
>>
>> On Mon, Feb 28, 2011 at 9:36 AM, Peter Olsson
>> <peter.olsson at visionutveckling.se> wrote:
>>> Are you using git HEAD as of the last couple of days? I know there was some changes for this last week related to ESL-56 - at least I saw some commits related to it... Or it this not at all related to ESL lib?
>>>
>>> /Peter
>>> ________________________________________
>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Mauritz L?vgren [mauritz.lovgren at hotmail.com]
>>> Skickat: den 28 februari 2011 15:26
>>> Till: FreeSWITCH Users Help
>>> ?mne: [Freeswitch-users] mod_event_socket message pizza received from BSD ? ? ? machines
>>>
>>> Hi,
>>>
>>> We are experiencing weird behaviour when using socket connection to mod_event_socket while running FS on FreeBSD and Mac OSX.
>>>
>>> We send a lot of bgapi and api commands to the FreeSwitch while controlling and monitoring hundreds of sessions.
>>>
>>> We have created a simple load-test that displays the problem we face.
>>> The load-test simply fires the following command continuosly to freeswitch: ?bgapi sofia status?, generally as fast as freeswitch can receive it.
>>>
>>> The following occurs, but only on FreeBSD or MacOSX (10.6.6) (both 64-bit) while using latest version of FreeSwitch:
>>>
>>> - Replies and events are inter-mixed on the receive stream (the output from freeswitch), causing protocol errors in the receiving client.
>>>
>>> This does _not_ happen on CentOS or Windows, so we wonder what could be causing this. Is there a problem with threading or socket libs here?
>>> We assume that FreeSwitch protect a mod_event_socket inbound connection output stream by locking (or other means) to make sure not any two messages are written ?simultaneously? causing garbage in the receiving end?
>>>
>>> Regards,
>>> Mauritz Lovgren
>>> System Architect
>>> IPLink Inc.
>>> http://www.iplink.no
>>>
>>>
>>> !DSPAM:4d6bb3a032761738012832!
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Tue Mar  1 01:07:51 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Feb 2011 16:07:51 -0600
Subject: [Freeswitch-users] mod_dingaling and google voice
In-Reply-To: <4D6C1978.8040702@cartissolutions.com>
References: <AANLkTin_dFj9ygx3TAmExeN9-pDFNCuZPGZ_Uc2d7o1W@mail.gmail.com>
	<AANLkTimUr3=w_fj_k643ChW93Q_YTm63R_qFyBO9EoQV@mail.gmail.com>
	<4D6C1978.8040702@cartissolutions.com>
Message-ID: <AANLkTim5TpcQeYSC_c6ZH9WMoJnq1aNvYViwUBkZr=fv@mail.gmail.com>

The solution to the port 0 issue is to run FS with -nonat so it does
not get confused by broken upnp setup.


On Mon, Feb 28, 2011 at 3:54 PM, Yossi Neiman
<freeswitch at cartissolutions.com> wrote:
> What FS build are you running?? I'd like to compare notes with you on your
> config, as I'm experiencing the RTP port 0 issue with FreeSWITCH Version
> 1.0.head (git-0d8e945 2011-02-16 08-57-43 -0600).? What about inbound?? Do
> you have that set up?
>
> Yossi Neiman
> Cartis Solutions, Inc. - http://www.cartissolutions.com
>
> On 02/28/2011 12:54 AM, envelopes envelopes wrote:
>
> never mind. add this line <action application="ring_ready"/> and restart FS
>
> fixed the issue.
>
>
>
> On Sun, Feb 27, 2011 at 10:30 PM, envelopes envelopes <sunwood360 at gmail.com>
> wrote:
>>
>> Now I am able to use GV for outbound dialing. However, I don't hear any
>> ringback or not sure whether the other party has answered the call.
>> is there any config variable to set up so that I will be notified if my
>> call is answered?
>>
>> thanks!
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Tue Mar  1 01:37:13 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Feb 2011 16:37:13 -0600
Subject: [Freeswitch-users] freetdm with a400p segfaults
In-Reply-To: <4D6C038E.7050303@cronomagic.com>
References: <4D6C038E.7050303@cronomagic.com>
Message-ID: <AANLkTikUacToUY15VK==Njct1H4OUndOy-FXuQNVAp1y@mail.gmail.com>

The first thing you are doing wrong is reporting a bug to the mailing list.

File a jira at http://jira.freeswitch.org under FreeTDM so you have
the proper people to help you.


On Mon, Feb 28, 2011 at 2:20 PM, Jesse Cloutier <jesse at cronomagic.com> wrote:
> Im having trouble with getting freetdm to work with a openvox a400p with 4
> fxo modules using the latest freeswitch from GIT
>
> dahdi installs fine and 'dahdi_hardware' lists it as a TDM400P
>
> When I configure freetdm.conf only I can start freeswitch and do a ftdm dump
> and it displays my span with 4 channels
>
> When I configure freetdm.conf.xml freeswitch segfaults on startup
>
> I don't know if I need them but ftmod_analog.so and ftmod_zt.so have been
> built
>
> this is my freetdm.conf:
>
> [general]
>
> [span zt myFX0]
> fxo-channel => 1
> fxo-channel => 2
> fxo-channel => 3
> fxo-channel => 4
>
>
> This is my freetdm.conf.xml:
> <configuration name="freetdm.conf" description="FreeTDM Configuration">
>
> ??????? <settings>
> ??????????????? <param name="debug" value="6"/>
> ??????????????? <!--<param name="hold-music" value="$${moh_uri}"/>-->
> ??????????????? <!-- Analog global options (they apply to all spans)
> ???????????????????? Remember you can only choose between either call-swap
> ???????????????????? or 3-way, not both!
> ??????????????? -->
> ??????????????? <!--<param name="enable-analog-option"
> value="call-swap"/>-->
> ??????????????? <!--<param name="enable-analog-option" value="3-way"/>-->
> ??????? </settings>
>
> ??????? <!-- Sample analog configuration (The analog_spans tag is for
> ftmod_analog) -->
> ??????? <analog_spans>
> ??????????????? <!-- The span name must match the name in your freetdm.conf
> -->
> ???????????? <span name="myFXO">
> ?????????????? <!--<param name="hold-music" value="$${moh_uri}"/>-->
> ?????????????? <param name="dialplan" value="XML"/>
> ?????????????? <param name="context" value="Incoming-FXO"/>
> ?????????????? <param name="tonegroup" value="us"/>
> ?????????????? <param name="enable_callerid" value="true"/>
> ?????????????? <param name="enable-analog-option" value="call-swap"/>
> ?????????????? <!--<param name="enable-analog-option" value="3-way"/>-->
> ???????????? </span>
>
> ??????? </analog_spans>
>
> </configuration>
>
> Any idea what I am doing wrong?
>
> --
> Jesse Cloutier
> Network Administrator
> Cronomagic Canada
> 5143411579 x210
> jesse at cronomagic.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 ayhkor at gmail.com  Tue Mar  1 01:40:23 2011
From: ayhkor at gmail.com (deniro)
Date: Mon, 28 Feb 2011 17:40:23 -0500
Subject: [Freeswitch-users] fs_cli socket connection error
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58ADB727D7@cooper>
References: <AANLkTikCoT9dzLNoD=6TzS0jiMw4afNX94TypsLarBbz@mail.gmail.com>
	<AANLkTi=sENgBB9Cdr-EkLJD8XfmH3A8GSfPnvDHO7qrw@mail.gmail.com>
	<AANLkTin5k=2Z-=QVVzqt91172wv88P7OFkqRqTY9VDcS@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58ADB727D7@cooper>
Message-ID: <AANLkTi=aPwmnVD8ZzaBVFDxGyPsbgzDEBGaXvq_B2NbF@mail.gmail.com>

Checked the log
Nothing is logged when
fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
is issued
/opt/freeswitch/log# tail -f freeswitch.log

ls -l freeswitch.log
-rwxrwxrwx 1 freeswitch daemon 33018 2011-02-28 17:39 freeswitch.log
thx



On Mon, Feb 28, 2011 at 3:59 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> Check freeswitch.log, it probably reports some problem when loading the
> mod_event_socket module.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r deniro [
> ayhkor at gmail.com]
> Skickat: den 28 februari 2011 21:42
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] fs_cli socket connection error
>
> Steve
> Thanks for the reply
> netstat -anlp | grep 8021  is blank (nothing showing up)
>
> /opt/freeswitch/bin/fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file() Configuration
> file is /root/.fs_cli_conf.
> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file() Configuration
> file is /etc/fs_cli.conf.
> [DEBUG] libs/esl/fs_cli.c:1159 main() profile default does not exist using
> builtin profile
> [DEBUG] libs/esl/fs_cli.c:1188 main() Using profile internal
> [75.xxx.xxx.xxx]
> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
> Error]
> yes mod_event_socket is after  mod_xml_curl but I changed the order in
>  modules.conf.xml
> still getting above (restarted freeswitch)
>
> thx
> deniro--
>
>
>
> On Mon, Feb 28, 2011 at 3:35 AM, Steven Ayre <steveayre at gmail.com<mailto:
> steveayre at gmail.com>> wrote:
> Enable debug logging and you should see an error that'll tell you more.
>
> Is mod_event_socket loading before or after mod_xml_curl? Chances are
> mod_xml_curl is loading first, so it's trying to read
> event_socket.conf.xml and/or acl.conf.xml through xml_curl and either
> getting a different config to your previous local copy or the ACLs are
> different.
>
> What does "netstat -anlp | grep :8021" from the linux cli show you?
> Does it show that freeswitch is actually listening on the port? If it
> is it's probably an ACL problem, if it isn't then it's probably a
> problem with event_socket.conf.xml
>
> -Steve
>
> On 28 February 2011 00:53, deniro <ayhkor at gmail.com<mailto:
> ayhkor at gmail.com>> wrote:
> > What would be possible reasons for this and how to resolve?
> > running fs 106 on ubuntu 10.04 server
> >
> > /opt/freeswitch/bin/fs_cli -H xxx.xxx.xxx.xxx -P 8021
> > [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
> > Error]
> > was working fine before I installed  mod_xml_cdr
> > configure --prefix=/opt/freeswitch --without-libcurl
> > make mod_xml_cdr-install
> > (no errors)
> >
> > in modules.conf.xml
> > <load module="mod_event_socket"/>
> >
> > is there a packaged version of mod_xml_cdr (for fs 1.0.6)?
> >
> > thx
> > deniro--
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org<http://www.freeswitch.org/>
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org<http://www.freeswitch.org/>
>
> !DSPAM:4d6c0a1132761029518849!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Tue Mar  1 01:43:18 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Feb 2011 16:43:18 -0600
Subject: [Freeswitch-users] FreeSwithc not handling no SDP INVITE
 properly for video call
In-Reply-To: <AANLkTimvG+_vo5O0jH1nnHMaHy-a7UqwFUS2+Dsgs+kK@mail.gmail.com>
References: <AANLkTinoomTncUKH=xKLSfAaWKeTupOMv_aZuN_CVVF+@mail.gmail.com>
	<AANLkTinxTuAv0ybtguJtZh3Xkc11=EHJo6CQwdRmRphV@mail.gmail.com>
	<AANLkTimvG+_vo5O0jH1nnHMaHy-a7UqwFUS2+Dsgs+kK@mail.gmail.com>
Message-ID: <AANLkTimzvZ=0Rvq76_0GAcfcnqSj5b-J80n8wW=1OgSQ@mail.gmail.com>

try

<action application="set" data="inherit_codec=true"/>

in your dialplan before you call the phone.



On Mon, Feb 28, 2011 at 12:04 PM, George Lee <whglee at gmail.com> wrote:
> I pulled from the latest source tree and rebuilt FreeSwitch. The
> problem still persists.
> http://pastebin.freeswitch.org/15486
> Any other suggestions you could provide?
>
> Thanks,
> George
>
> On Fri, Feb 25, 2011 at 5:52 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> make sure you have tested on latest GIT (minutes ago) there was just a
>> fix to some video issues.
>>
>>
>> On Fri, Feb 25, 2011 at 3:56 PM, George Lee <whglee at gmail.com> wrote:
>>> Hi all,
>>>
>>> I am having trouble with FreeSwitch handling late codec negotiation
>>> for video calls.
>>>
>>> The call logs are here:
>>> http://pastebin.freeswitch.org/15481
>>>
>>> I have this line:
>>> <param name="enable-3pcc" value="proxy"/>
>>> added to the external sip_profile for handling late codec negotiation.
>>> I also have
>>> ?<X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,H263"/>
>>> ?<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,H263"/>
>>> in vars.xml for codec support list.
>>>
>>> The initial INVITE contains no SDP for the caller and once the callee
>>> answers the call, it sends the 200 OK with audio (pcma, pcmu) and
>>> video (H263) codec capabilities. FreeSwitch forwards the 200 OK to the
>>> caller without video codec capabilities. As a result, the video call
>>> does not establish properly.
>>>
>>> Could someone give me a pointer?
>>>
>>> Thanks,
>>> George
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 kbdfck at gmail.com  Tue Mar  1 01:43:56 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Tue, 1 Mar 2011 01:43:56 +0300
Subject: [Freeswitch-users] HANGUP_CAUSE after bridge in outbound ESL
Message-ID: <AANLkTimRzV3M4vUBQpKwyVhjBh7LxEiF=y3jJ_P+rCRK@mail.gmail.com>

Hi All

What is correct method to get bridge status after execute("bridge") in ESL?
I have inbound call that gets bridged to SIP endpoint. I need to know
whether it was ORIGINATOR_CANCEL or BUSY or something else, but if I use
sync, seems I can't determine ORIGINATOR_CANCEL status, because it is set on
a-leg which is destroyed first after hangup.

Is there a way to get bridge status without messing with events and event
source filtering by channel uuid or event type?

-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From steveayre at gmail.com  Tue Mar  1 01:46:44 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 28 Feb 2011 22:46:44 +0000
Subject: [Freeswitch-users] fs_cli socket connection error
In-Reply-To: <AANLkTi=aPwmnVD8ZzaBVFDxGyPsbgzDEBGaXvq_B2NbF@mail.gmail.com>
References: <AANLkTikCoT9dzLNoD=6TzS0jiMw4afNX94TypsLarBbz@mail.gmail.com>
	<AANLkTi=sENgBB9Cdr-EkLJD8XfmH3A8GSfPnvDHO7qrw@mail.gmail.com>
	<AANLkTin5k=2Z-=QVVzqt91172wv88P7OFkqRqTY9VDcS@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58ADB727D7@cooper>
	<AANLkTi=aPwmnVD8ZzaBVFDxGyPsbgzDEBGaXvq_B2NbF@mail.gmail.com>
Message-ID: <AANLkTikpQuxvyaU7PAGnyJuoQLQ9xn4inNJ_u5x41Anv@mail.gmail.com>

You must check the logfile for when FS starts up - if netstat shows
nothing for port 8021 then either mod_event_socket isn't being loaded
or you'll see an error when it tries to load.

Nothing from netstat means nothing's listening, so trying to connect
using fs_cli won't do anything.

-Steve


On 28 February 2011 22:40, deniro <ayhkor at gmail.com> wrote:
> Checked the log
> Nothing is logged when
> fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
> is issued
> /opt/freeswitch/log# tail -f freeswitch.log
>
> ls -l freeswitch.log
> -rwxrwxrwx 1 freeswitch daemon 33018 2011-02-28 17:39 freeswitch.log
> thx
>
>
>
> On Mon, Feb 28, 2011 at 3:59 PM, Peter Olsson
> <peter.olsson at visionutveckling.se> wrote:
>>
>> Check freeswitch.log, it probably reports some problem when loading the
>> mod_event_socket module.
>>
>> /Peter
>> ________________________________________
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r deniro
>> [ayhkor at gmail.com]
>> Skickat: den 28 februari 2011 21:42
>> Till: FreeSWITCH Users Help
>> ?mne: Re: [Freeswitch-users] fs_cli socket connection error
>>
>> Steve
>> Thanks for the reply
>> netstat -anlp | grep 8021 ?is blank (nothing showing up)
>>
>> /opt/freeswitch/bin/fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
>> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file() Configuration
>> file is /root/.fs_cli_conf.
>> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file() Configuration
>> file is /etc/fs_cli.conf.
>> [DEBUG] libs/esl/fs_cli.c:1159 main() profile default does not exist using
>> builtin profile
>> [DEBUG] libs/esl/fs_cli.c:1188 main() Using profile internal
>> [75.xxx.xxx.xxx]
>> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
>> Error]
>> yes mod_event_socket is after ?mod_xml_curl but I changed the order in
>> ?modules.conf.xml
>> still getting above (restarted freeswitch)
>>
>> thx
>> deniro--
>>
>>
>>
>> On Mon, Feb 28, 2011 at 3:35 AM, Steven Ayre
>> <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
>> Enable debug logging and you should see an error that'll tell you more.
>>
>> Is mod_event_socket loading before or after mod_xml_curl? Chances are
>> mod_xml_curl is loading first, so it's trying to read
>> event_socket.conf.xml and/or acl.conf.xml through xml_curl and either
>> getting a different config to your previous local copy or the ACLs are
>> different.
>>
>> What does "netstat -anlp | grep :8021" from the linux cli show you?
>> Does it show that freeswitch is actually listening on the port? If it
>> is it's probably an ACL problem, if it isn't then it's probably a
>> problem with event_socket.conf.xml
>>
>> -Steve
>>
>> On 28 February 2011 00:53, deniro
>> <ayhkor at gmail.com<mailto:ayhkor at gmail.com>> wrote:
>> > What would be possible reasons for this and how to resolve?
>> > running fs 106 on ubuntu 10.04 server
>> >
>> > /opt/freeswitch/bin/fs_cli -H xxx.xxx.xxx.xxx -P 8021
>> > [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
>> > Connection
>> > Error]
>> > was working fine before I installed ?mod_xml_cdr
>> > configure --prefix=/opt/freeswitch --without-libcurl
>> > make mod_xml_cdr-install
>> > (no errors)
>> >
>> > in modules.conf.xml
>> > <load module="mod_event_socket"/>
>> >
>> > is there a packaged version of mod_xml_cdr (for fs 1.0.6)?
>> >
>> > thx
>> > deniro--
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> >
>> > FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org<http://www.freeswitch.org/>
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org<http://www.freeswitch.org/>
>>
>> !DSPAM:4d6c0a1132761029518849!
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mthakershi at gmail.com  Tue Mar  1 01:53:22 2011
From: mthakershi at gmail.com (Malay Thakershi)
Date: Mon, 28 Feb 2011 16:53:22 -0600
Subject: [Freeswitch-users] disabling ptime warning message
In-Reply-To: <AANLkTikvOhbJZfpbN9BmLs3eFFuz=uG2jRyKM4J5zYus@mail.gmail.com>
References: <AANLkTim97P5=MJgKq3ncDmbTiXkP63CBoALRD6o+b-Uw@mail.gmail.com>
	<AANLkTikw3TLW8bo39WRX6wN44vhcGSDBBTS_QffqBTwc@mail.gmail.com>
	<AANLkTi=FGWs2bs3EFvaMTTyG0=0inwe5z2UfByctX-g1@mail.gmail.com>
	<AANLkTindiPgTH65=0MwBgdviyWByQzmu1=01MYUy_fXy@mail.gmail.com>
	<AANLkTi=cFCJssphqmji_8LTXTg3ZycGfswMM77hmkie9@mail.gmail.com>
	<AANLkTimTKLJQ+cE9r5t2fRR+riDo-DPi+7FQLJecoc9M@mail.gmail.com>
	<AANLkTi=A-qZXz23nBgFz+LgAKCsqOmFFfgs8zm8k8_Re@mail.gmail.com>
	<AANLkTikvOhbJZfpbN9BmLs3eFFuz=uG2jRyKM4J5zYus@mail.gmail.com>
Message-ID: <AANLkTi=9rwDSgyt5kDj8LhG+CMFAfXGjxxfcpquJxPxH@mail.gmail.com>

I can't login to FS pastebin.

I used same user / password as I use to access users group.

Malay

On Mon, Feb 28, 2011 at 3:47 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> You also need "sofia global siptrace on" so we can see the SIP traffic
> in the log.
> We have our own pastebin at http://pastebin.freeswitch.org
>
>
> On Mon, Feb 28, 2011 at 3:22 PM, Malay Thakershi <mthakershi at gmail.com>
> wrote:
> > http://pastebin.com/CCwmAfZK
> > I noticed that LPC warning only comes for outbound calls I initiate using
> > "Originate" API.
> > Malay
> >
> > On Mon, Feb 28, 2011 at 2:04 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> That's L16 not LPC, you need L16 to play files.
> >> Why don't you just put the whole log of the call on pastebin.
> >>
> >>
> >> On Mon, Feb 28, 2011 at 1:53 PM, Malay Thakershi <mthakershi at gmail.com>
> >> wrote:
> >> > Following is a section from the log:
> >> > ---------------------------
> >> > 2011-02-28 13:36:08.294241 [DEBUG] sofia_glue.c:2760 Set Codec
> >> > sofia/sipinterface_1/##########@##.##.##.## PCMU/8000 20 ms 160
> samples
> >> > 64000 bits
> >> > 2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2298 OPEN TTS
> >> > cepstral
> >> > 2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2307 Raw
> Codec
> >> > Activated
> >> > 2011-02-28 13:36:15.017444 [DEBUG] switch_ivr_play_say.c:1996 Speaking
> >> > text:
> >> > <break strength='medium'/>We must verify your identity.
> >> > 2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:2188 done
> >> > speaking
> >> > text
> >> > 2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:1244 Codec
> >> > Activated L16 at 8000hz 1 channels 20ms
> >> > 2011-02-28 13:36:25.530443 [DEBUG] switch_ivr_play_say.c:1581 done
> >> > playing
> >> > file
> >> > 2011-02-28 13:36:25.530443 [DEBUG] switch_ivr.c:1135 Codec Activated
> >> > L16 at 8000hz 1 channels 20ms
> >> > 2011-02-28 13:36:30.549653 [DEBUG] switch_ivr_play_say.c:1244 Codec
> >> > Activated L16 at 8000hz 1 channels 20ms
> >> > ---------------------------
> >> > So I see when I play a file (using StreamFile / PAGD), it activates
> L16,
> >> > which the wiki pages says is not recommended. So should I deactivate
> it?
> >> > If
> >> > so, how?
> >> > Now, I have not done any setting out of default / ordinary that comes
> >> > with
> >> > the build. I am playing WAV file that is generated by Cepstral SWIFT
> >> > command
> >> > line tool (text to WAV). The file format is "Wave PCM signed 16 bit,
> >> > 8000
> >> > Hz, 128 kbps, mono".
> >> > Thank you for help so far.
> >> > Malay
> >> > On Mon, Feb 28, 2011 at 12:50 PM, Anthony Minessale
> >> > <anthony.minessale at gmail.com> wrote:
> >> >>
> >> >> look at your SIP traffic and console log.
> >> >>
> >> >> enter "sofia global siptrace on" followed by "console loglevel debug"
> >> >> at the cli and make the call.
> >> >>
> >> >>
> >> >> On Mon, Feb 28, 2011 at 12:47 PM, Malay Thakershi
> >> >> <mthakershi at gmail.com>
> >> >> wrote:
> >> >> > I have no idea where to look for this setting.
> >> >> > This is in modules.conf.xml
> >> >> >     <!-- Codec Interfaces -->
> >> >> >     <load module="mod_spandsp"/>
> >> >> >     <!--<load module="mod_voipcodecs"/>-->
> >> >> >     <load module="mod_g723_1"/>
> >> >> >     <load module="mod_g729"/>
> >> >> >     <load module="mod_amr"/>
> >> >> >     <load module="mod_ilbc"/>
> >> >> >     <load module="mod_speex"/>
> >> >> >     <load module="mod_h26x"/>
> >> >> >     <load module="mod_siren"/>
> >> >> >     <!--<load module="mod_celt"/>-->
> >> >> >     <!--<load module="mod_opus"/>-->
> >> >> > Apart from settings I posted in my previous post, where else to
> look
> >> >> > for
> >> >> > disabling LPC?
> >> >> > Malay
> >> >> > On Mon, Feb 28, 2011 at 12:05 PM, Anthony Minessale
> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >>
> >> >> >> is your inbound call using LPC? you don't want to be using LPC and
> >> >> >> expect anything to sound good that's for sure.
> >> >> >> It would not just magically say that unless something you are
> doing
> >> >> >> has
> >> >> >> LPC?
> >> >> >>
> >> >> >>
> >> >> >> On Mon, Feb 28, 2011 at 10:24 AM, Malay Thakershi
> >> >> >> <mthakershi at gmail.com>
> >> >> >> wrote:
> >> >> >> > Hello,
> >> >> >> > I updated to the latest FS version last week.
> >> >> >> > I started getting the following warning when speech / sound is
> >> >> >> > played
> >> >> >> > on
> >> >> >> > the
> >> >> >> > call.
> >> >> >> > "2011-02-28 10:15:55.985930 [WARNING] sofia_glue.c:213 Codec LPC
> >> >> >> > payload
> >> >> >> > 7
> >> >> >> > added to sdp wanting ptime 90 but it's already 20
> (G7221:115:20),
> >> >> >> > disabling
> >> >> >> > ptime."
> >> >> >> > I read sections on codecs and negotiations.
> >> >> >> > Following are the settings from vars.xml (I have not changed
> >> >> >> > them):
> >> >> >> >   <X-PRE-PROCESS cmd="set"
> >> >> >> >
> >> >> >> >
> >> >> >> > data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h
> ,G722,PCMU,PCMA,GSM"/>
> >> >> >> >   <X-PRE-PROCESS cmd="set"
> >> >> >> > data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
> >> >> >> > Also, there is no codec related setting in sip_profiles files
> >> >> >> > and sofia.conf.xml file.
> >> >> >> > I am playing audio files using Cepstral TTS during the call.
> >> >> >> > Can someone please help me understand these settings? And if
> they
> >> >> >> > are
> >> >> >> > appropriate?
> >> >> >> > Thank you.
> >> >> >> > Malay
> >> >> >> > _______________________________________________
> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >
> >> >> >> >
> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> > 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-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > 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-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > 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-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
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From anthony.minessale at gmail.com  Tue Mar  1 01:55:37 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Feb 2011 16:55:37 -0600
Subject: [Freeswitch-users] disabling ptime warning message
In-Reply-To: <AANLkTi=9rwDSgyt5kDj8LhG+CMFAfXGjxxfcpquJxPxH@mail.gmail.com>
References: <AANLkTim97P5=MJgKq3ncDmbTiXkP63CBoALRD6o+b-Uw@mail.gmail.com>
	<AANLkTikw3TLW8bo39WRX6wN44vhcGSDBBTS_QffqBTwc@mail.gmail.com>
	<AANLkTi=FGWs2bs3EFvaMTTyG0=0inwe5z2UfByctX-g1@mail.gmail.com>
	<AANLkTindiPgTH65=0MwBgdviyWByQzmu1=01MYUy_fXy@mail.gmail.com>
	<AANLkTi=cFCJssphqmji_8LTXTg3ZycGfswMM77hmkie9@mail.gmail.com>
	<AANLkTimTKLJQ+cE9r5t2fRR+riDo-DPi+7FQLJecoc9M@mail.gmail.com>
	<AANLkTi=A-qZXz23nBgFz+LgAKCsqOmFFfgs8zm8k8_Re@mail.gmail.com>
	<AANLkTikvOhbJZfpbN9BmLs3eFFuz=uG2jRyKM4J5zYus@mail.gmail.com>
	<AANLkTi=9rwDSgyt5kDj8LhG+CMFAfXGjxxfcpquJxPxH@mail.gmail.com>
Message-ID: <AANLkTimcqpeHJN+wJ+i+DWg4ii6PYuHwF2wDf5fpyfhz@mail.gmail.com>

If you read the dialog box closely the login and password are stated
in the challenge box.
That is our way of telling who pays attention and who doesn't...

not really, its to keep spambots off the site.


On Mon, Feb 28, 2011 at 4:53 PM, Malay Thakershi <mthakershi at gmail.com> wrote:
> I can't login to FS pastebin.
> I used same user / password as I use to access users group.
> Malay
>
> On Mon, Feb 28, 2011 at 3:47 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> You also need "sofia global siptrace on" so we can see the SIP traffic
>> in the log.
>> We have our own pastebin at http://pastebin.freeswitch.org
>>
>>
>> On Mon, Feb 28, 2011 at 3:22 PM, Malay Thakershi <mthakershi at gmail.com>
>> wrote:
>> > http://pastebin.com/CCwmAfZK
>> > I noticed that LPC warning only comes for outbound calls I initiate
>> > using
>> > "Originate" API.
>> > Malay
>> >
>> > On Mon, Feb 28, 2011 at 2:04 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> That's L16 not LPC, you need L16 to play files.
>> >> Why don't you just put the whole log of the call on pastebin.
>> >>
>> >>
>> >> On Mon, Feb 28, 2011 at 1:53 PM, Malay Thakershi <mthakershi at gmail.com>
>> >> wrote:
>> >> > Following is a section from the log:
>> >> > ---------------------------
>> >> > 2011-02-28 13:36:08.294241 [DEBUG] sofia_glue.c:2760 Set Codec
>> >> > sofia/sipinterface_1/##########@##.##.##.## PCMU/8000 20 ms 160
>> >> > samples
>> >> > 64000 bits
>> >> > 2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2298 OPEN
>> >> > TTS
>> >> > cepstral
>> >> > 2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2307 Raw
>> >> > Codec
>> >> > Activated
>> >> > 2011-02-28 13:36:15.017444 [DEBUG] switch_ivr_play_say.c:1996
>> >> > Speaking
>> >> > text:
>> >> > <break strength='medium'/>We must verify your identity.
>> >> > 2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:2188 done
>> >> > speaking
>> >> > text
>> >> > 2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:1244 Codec
>> >> > Activated L16 at 8000hz 1 channels 20ms
>> >> > 2011-02-28 13:36:25.530443 [DEBUG] switch_ivr_play_say.c:1581 done
>> >> > playing
>> >> > file
>> >> > 2011-02-28 13:36:25.530443 [DEBUG] switch_ivr.c:1135 Codec Activated
>> >> > L16 at 8000hz 1 channels 20ms
>> >> > 2011-02-28 13:36:30.549653 [DEBUG] switch_ivr_play_say.c:1244 Codec
>> >> > Activated L16 at 8000hz 1 channels 20ms
>> >> > ---------------------------
>> >> > So I see when I play a file (using StreamFile / PAGD), it activates
>> >> > L16,
>> >> > which the wiki pages says is not recommended. So should I deactivate
>> >> > it?
>> >> > If
>> >> > so, how?
>> >> > Now, I have not done any setting out of default / ordinary that comes
>> >> > with
>> >> > the build. I am playing WAV file that is generated by Cepstral SWIFT
>> >> > command
>> >> > line tool (text to WAV). The file format is "Wave PCM signed 16 bit,
>> >> > 8000
>> >> > Hz, 128 kbps, mono".
>> >> > Thank you for help so far.
>> >> > Malay
>> >> > On Mon, Feb 28, 2011 at 12:50 PM, Anthony Minessale
>> >> > <anthony.minessale at gmail.com> wrote:
>> >> >>
>> >> >> look at your SIP traffic and console log.
>> >> >>
>> >> >> enter "sofia global siptrace on" followed by "console loglevel
>> >> >> debug"
>> >> >> at the cli and make the call.
>> >> >>
>> >> >>
>> >> >> On Mon, Feb 28, 2011 at 12:47 PM, Malay Thakershi
>> >> >> <mthakershi at gmail.com>
>> >> >> wrote:
>> >> >> > I have no idea where to look for this setting.
>> >> >> > This is in modules.conf.xml
>> >> >> > ?? ?<!-- Codec Interfaces -->
>> >> >> > ?? ?<load module="mod_spandsp"/>
>> >> >> > ?? ?<!--<load module="mod_voipcodecs"/>-->
>> >> >> > ?? ?<load module="mod_g723_1"/>
>> >> >> > ?? ?<load module="mod_g729"/>
>> >> >> > ?? ?<load module="mod_amr"/>
>> >> >> > ?? ?<load module="mod_ilbc"/>
>> >> >> > ?? ?<load module="mod_speex"/>
>> >> >> > ?? ?<load module="mod_h26x"/>
>> >> >> > ?? ?<load module="mod_siren"/>
>> >> >> > ?? ?<!--<load module="mod_celt"/>-->
>> >> >> > ?? ?<!--<load module="mod_opus"/>-->
>> >> >> > Apart from settings I posted in my previous post, where else to
>> >> >> > look
>> >> >> > for
>> >> >> > disabling LPC?
>> >> >> > Malay
>> >> >> > On Mon, Feb 28, 2011 at 12:05 PM, Anthony Minessale
>> >> >> > <anthony.minessale at gmail.com> wrote:
>> >> >> >>
>> >> >> >> is your inbound call using LPC? you don't want to be using LPC
>> >> >> >> and
>> >> >> >> expect anything to sound good that's for sure.
>> >> >> >> It would not just magically say that unless something you are
>> >> >> >> doing
>> >> >> >> has
>> >> >> >> LPC?
>> >> >> >>
>> >> >> >>
>> >> >> >> On Mon, Feb 28, 2011 at 10:24 AM, Malay Thakershi
>> >> >> >> <mthakershi at gmail.com>
>> >> >> >> wrote:
>> >> >> >> > Hello,
>> >> >> >> > I updated to the latest FS version last week.
>> >> >> >> > I started getting the following warning when speech / sound is
>> >> >> >> > played
>> >> >> >> > on
>> >> >> >> > the
>> >> >> >> > call.
>> >> >> >> > "2011-02-28 10:15:55.985930 [WARNING] sofia_glue.c:213 Codec
>> >> >> >> > LPC
>> >> >> >> > payload
>> >> >> >> > 7
>> >> >> >> > added to sdp wanting ptime?90 but it's already 20
>> >> >> >> > (G7221:115:20),
>> >> >> >> > disabling
>> >> >> >> > ptime."
>> >> >> >> > I read sections on codecs and negotiations.
>> >> >> >> > Following are the settings from vars.xml (I have not changed
>> >> >> >> > them):
>> >> >> >> > ??<X-PRE-PROCESS cmd="set"
>> >> >> >> >
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>> >> >> >> > ??<X-PRE-PROCESS cmd="set"
>> >> >> >> > data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>> >> >> >> > Also, there is no codec related setting in sip_profiles files
>> >> >> >> > and?sofia.conf.xml file.
>> >> >> >> > I am playing audio files using Cepstral TTS during the call.
>> >> >> >> > Can someone please help me understand these settings? And if
>> >> >> >> > they
>> >> >> >> > are
>> >> >> >> > appropriate?
>> >> >> >> > Thank you.
>> >> >> >> > Malay
>> >> >> >> > _______________________________________________
>> >> >> >> > FreeSWITCH-users mailing list
>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> > 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-users mailing list
>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > 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-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > 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-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 wstephen80 at gmail.com  Tue Mar  1 01:57:09 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Mon, 28 Feb 2011 23:57:09 +0100
Subject: [Freeswitch-users] ESL socket outbound: fail to stop tone detect
Message-ID: <AANLkTi=VnrfD+pKnG3a_T8bZnNPWQ-nAQ0Avp23V25E7@mail.gmail.com>

In an originated outbound session that is in progressing phase, I do, via
esl socket outbound, a "tone_detect".

When I receive the event "DETECTED_TONE" I do the "stop_tone_detect" but
sometimes the stop has no effect and I see in the log the warning:

"switch_core_session.c:1955 Cannot execute app 'stop_tone_detect' media
required on an outbound channel that does not have media established"

But the media is present (the tone is correctly detected!).

The event I receive on this channel are (in sequence)

CHANNEL_UUID
CHANNEL_OUTGOING
CHANNEL_ORIGINATE
CHANNEL_STATE - CS_INIT
CHANNEL_STATE - CS_ROUTING
CHANNEL_STATE - CS_CONSUME_MEDIA
CHANNEL_PROGRESS
CHANNEL_PROGRESS_MEDIA
CHANNEL_STATE - CS_EXECUTE
CHANNEL_EXECUTE
CHANNEL_PARK

then I do the "tone_detect" and I receive:

CHANNEL_EXECUTE
MEDIA_BUG_START
CHANNEL_EXECUTE_COMPLETE
DETECTED_TONE

so I do the "stop_tone_detect" that fails in the log

Please, can anyone address me to solve this issue?
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From freeswitch at cartissolutions.com  Tue Mar  1 01:58:19 2011
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Mon, 28 Feb 2011 16:58:19 -0600
Subject: [Freeswitch-users] mod_dingaling and google voice
In-Reply-To: <AANLkTim5TpcQeYSC_c6ZH9WMoJnq1aNvYViwUBkZr=fv@mail.gmail.com>
References: <AANLkTin_dFj9ygx3TAmExeN9-pDFNCuZPGZ_Uc2d7o1W@mail.gmail.com>	<AANLkTimUr3=w_fj_k643ChW93Q_YTm63R_qFyBO9EoQV@mail.gmail.com>	<4D6C1978.8040702@cartissolutions.com>
	<AANLkTim5TpcQeYSC_c6ZH9WMoJnq1aNvYViwUBkZr=fv@mail.gmail.com>
Message-ID: <4D6C288B.4050801@cartissolutions.com>

I am running with -nonat, the FS is running with a UA on a public IP 
address, etc, but both Michael Collins and I are still getting the RTP 
port 0 issue.  I figured I'd compare notes with somebody who has a 
working config before I file this as something other than a PEBKAC issue...


Yossi Neiman
Cartis Solutions, Inc. - http://www.cartissolutions.com


On 02/28/2011 04:07 PM, Anthony Minessale wrote:
> The solution to the port 0 issue is to run FS with -nonat so it does
> not get confused by broken upnp setup.
>
>
> On Mon, Feb 28, 2011 at 3:54 PM, Yossi Neiman
> <freeswitch at cartissolutions.com>  wrote:
>> What FS build are you running?  I'd like to compare notes with you on your
>> config, as I'm experiencing the RTP port 0 issue with FreeSWITCH Version
>> 1.0.head (git-0d8e945 2011-02-16 08-57-43 -0600).  What about inbound?  Do
>> you have that set up?
>>
>> Yossi Neiman
>> Cartis Solutions, Inc. - http://www.cartissolutions.com
>>
>> On 02/28/2011 12:54 AM, envelopes envelopes wrote:
>>
>> never mind. add this line<action application="ring_ready"/>  and restart FS
>>
>> fixed the issue.
>>
>>
>>
>> On Sun, Feb 27, 2011 at 10:30 PM, envelopes envelopes<sunwood360 at gmail.com>
>> wrote:
>>> Now I am able to use GV for outbound dialing. However, I don't hear any
>>> ringback or not sure whether the other party has answered the call.
>>> is there any config variable to set up so that I will be notified if my
>>> call is answered?
>>>
>>> thanks!
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>


From whglee at gmail.com  Tue Mar  1 01:57:35 2011
From: whglee at gmail.com (George Lee)
Date: Mon, 28 Feb 2011 17:57:35 -0500
Subject: [Freeswitch-users] FreeSwithc not handling no SDP INVITE
 properly for video call
In-Reply-To: <AANLkTimzvZ=0Rvq76_0GAcfcnqSj5b-J80n8wW=1OgSQ@mail.gmail.com>
References: <AANLkTinoomTncUKH=xKLSfAaWKeTupOMv_aZuN_CVVF+@mail.gmail.com>
	<AANLkTinxTuAv0ybtguJtZh3Xkc11=EHJo6CQwdRmRphV@mail.gmail.com>
	<AANLkTimvG+_vo5O0jH1nnHMaHy-a7UqwFUS2+Dsgs+kK@mail.gmail.com>
	<AANLkTimzvZ=0Rvq76_0GAcfcnqSj5b-J80n8wW=1OgSQ@mail.gmail.com>
Message-ID: <AANLkTimhRw7CXrU73yU9hs_to8DYf1disro5fidW5XYd@mail.gmail.com>

I have already had this setup in my dialplan in this failure case.
         <action application="set" data="hangup_after_bridge=true"/>
         <action application="set" data="continue_on_fail=true"/>
         <action application="set" data="inherit_codec=true"/>
         <action application="set" data="proxy_media=false"/>

- George

On Mon, Feb 28, 2011 at 5:43 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> try
>
> <action application="set" data="inherit_codec=true"/>
>
> in your dialplan before you call the phone.
>
>
>
> On Mon, Feb 28, 2011 at 12:04 PM, George Lee <whglee at gmail.com> wrote:
>> I pulled from the latest source tree and rebuilt FreeSwitch. The
>> problem still persists.
>> http://pastebin.freeswitch.org/15486
>> Any other suggestions you could provide?
>>
>> Thanks,
>> George
>>
>> On Fri, Feb 25, 2011 at 5:52 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>> make sure you have tested on latest GIT (minutes ago) there was just a
>>> fix to some video issues.
>>>
>>>
>>> On Fri, Feb 25, 2011 at 3:56 PM, George Lee <whglee at gmail.com> wrote:
>>>> Hi all,
>>>>
>>>> I am having trouble with FreeSwitch handling late codec negotiation
>>>> for video calls.
>>>>
>>>> The call logs are here:
>>>> http://pastebin.freeswitch.org/15481
>>>>
>>>> I have this line:
>>>> <param name="enable-3pcc" value="proxy"/>
>>>> added to the external sip_profile for handling late codec negotiation.
>>>> I also have
>>>> ?<X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,H263"/>
>>>> ?<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,H263"/>
>>>> in vars.xml for codec support list.
>>>>
>>>> The initial INVITE contains no SDP for the caller and once the callee
>>>> answers the call, it sends the 200 OK with audio (pcma, pcmu) and
>>>> video (H263) codec capabilities. FreeSwitch forwards the 200 OK to the
>>>> caller without video codec capabilities. As a result, the video call
>>>> does not establish properly.
>>>>
>>>> Could someone give me a pointer?
>>>>
>>>> Thanks,
>>>> George
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> 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-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mthakershi at gmail.com  Tue Mar  1 01:58:28 2011
From: mthakershi at gmail.com (Malay Thakershi)
Date: Mon, 28 Feb 2011 16:58:28 -0600
Subject: [Freeswitch-users] disabling ptime warning message
In-Reply-To: <AANLkTimcqpeHJN+wJ+i+DWg4ii6PYuHwF2wDf5fpyfhz@mail.gmail.com>
References: <AANLkTim97P5=MJgKq3ncDmbTiXkP63CBoALRD6o+b-Uw@mail.gmail.com>
	<AANLkTikw3TLW8bo39WRX6wN44vhcGSDBBTS_QffqBTwc@mail.gmail.com>
	<AANLkTi=FGWs2bs3EFvaMTTyG0=0inwe5z2UfByctX-g1@mail.gmail.com>
	<AANLkTindiPgTH65=0MwBgdviyWByQzmu1=01MYUy_fXy@mail.gmail.com>
	<AANLkTi=cFCJssphqmji_8LTXTg3ZycGfswMM77hmkie9@mail.gmail.com>
	<AANLkTimTKLJQ+cE9r5t2fRR+riDo-DPi+7FQLJecoc9M@mail.gmail.com>
	<AANLkTi=A-qZXz23nBgFz+LgAKCsqOmFFfgs8zm8k8_Re@mail.gmail.com>
	<AANLkTikvOhbJZfpbN9BmLs3eFFuz=uG2jRyKM4J5zYus@mail.gmail.com>
	<AANLkTi=9rwDSgyt5kDj8LhG+CMFAfXGjxxfcpquJxPxH@mail.gmail.com>
	<AANLkTimcqpeHJN+wJ+i+DWg4ii6PYuHwF2wDf5fpyfhz@mail.gmail.com>
Message-ID: <AANLkTik5r_qHjgSgGk-_v+9VDM-fPZmGC-4fTZzPw=b2@mail.gmail.com>

http://pastebin.freeswitch.org/15500

<http://pastebin.freeswitch.org/15500>Thank you.

Malay

On Mon, Feb 28, 2011 at 4:55 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> If you read the dialog box closely the login and password are stated
> in the challenge box.
> That is our way of telling who pays attention and who doesn't...
>
> not really, its to keep spambots off the site.
>
>
> On Mon, Feb 28, 2011 at 4:53 PM, Malay Thakershi <mthakershi at gmail.com>
> wrote:
> > I can't login to FS pastebin.
> > I used same user / password as I use to access users group.
> > Malay
> >
> > On Mon, Feb 28, 2011 at 3:47 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> You also need "sofia global siptrace on" so we can see the SIP traffic
> >> in the log.
> >> We have our own pastebin at http://pastebin.freeswitch.org
> >>
> >>
> >> On Mon, Feb 28, 2011 at 3:22 PM, Malay Thakershi <mthakershi at gmail.com>
> >> wrote:
> >> > http://pastebin.com/CCwmAfZK
> >> > I noticed that LPC warning only comes for outbound calls I initiate
> >> > using
> >> > "Originate" API.
> >> > Malay
> >> >
> >> > On Mon, Feb 28, 2011 at 2:04 PM, Anthony Minessale
> >> > <anthony.minessale at gmail.com> wrote:
> >> >>
> >> >> That's L16 not LPC, you need L16 to play files.
> >> >> Why don't you just put the whole log of the call on pastebin.
> >> >>
> >> >>
> >> >> On Mon, Feb 28, 2011 at 1:53 PM, Malay Thakershi <
> mthakershi at gmail.com>
> >> >> wrote:
> >> >> > Following is a section from the log:
> >> >> > ---------------------------
> >> >> > 2011-02-28 13:36:08.294241 [DEBUG] sofia_glue.c:2760 Set Codec
> >> >> > sofia/sipinterface_1/##########@##.##.##.## PCMU/8000 20 ms 160
> >> >> > samples
> >> >> > 64000 bits
> >> >> > 2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2298 OPEN
> >> >> > TTS
> >> >> > cepstral
> >> >> > 2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2307 Raw
> >> >> > Codec
> >> >> > Activated
> >> >> > 2011-02-28 13:36:15.017444 [DEBUG] switch_ivr_play_say.c:1996
> >> >> > Speaking
> >> >> > text:
> >> >> > <break strength='medium'/>We must verify your identity.
> >> >> > 2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:2188 done
> >> >> > speaking
> >> >> > text
> >> >> > 2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:1244 Codec
> >> >> > Activated L16 at 8000hz 1 channels 20ms
> >> >> > 2011-02-28 13:36:25.530443 [DEBUG] switch_ivr_play_say.c:1581 done
> >> >> > playing
> >> >> > file
> >> >> > 2011-02-28 13:36:25.530443 [DEBUG] switch_ivr.c:1135 Codec
> Activated
> >> >> > L16 at 8000hz 1 channels 20ms
> >> >> > 2011-02-28 13:36:30.549653 [DEBUG] switch_ivr_play_say.c:1244 Codec
> >> >> > Activated L16 at 8000hz 1 channels 20ms
> >> >> > ---------------------------
> >> >> > So I see when I play a file (using StreamFile / PAGD), it activates
> >> >> > L16,
> >> >> > which the wiki pages says is not recommended. So should I
> deactivate
> >> >> > it?
> >> >> > If
> >> >> > so, how?
> >> >> > Now, I have not done any setting out of default / ordinary that
> comes
> >> >> > with
> >> >> > the build. I am playing WAV file that is generated by Cepstral
> SWIFT
> >> >> > command
> >> >> > line tool (text to WAV). The file format is "Wave PCM signed 16
> bit,
> >> >> > 8000
> >> >> > Hz, 128 kbps, mono".
> >> >> > Thank you for help so far.
> >> >> > Malay
> >> >> > On Mon, Feb 28, 2011 at 12:50 PM, Anthony Minessale
> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >>
> >> >> >> look at your SIP traffic and console log.
> >> >> >>
> >> >> >> enter "sofia global siptrace on" followed by "console loglevel
> >> >> >> debug"
> >> >> >> at the cli and make the call.
> >> >> >>
> >> >> >>
> >> >> >> On Mon, Feb 28, 2011 at 12:47 PM, Malay Thakershi
> >> >> >> <mthakershi at gmail.com>
> >> >> >> wrote:
> >> >> >> > I have no idea where to look for this setting.
> >> >> >> > This is in modules.conf.xml
> >> >> >> >     <!-- Codec Interfaces -->
> >> >> >> >     <load module="mod_spandsp"/>
> >> >> >> >     <!--<load module="mod_voipcodecs"/>-->
> >> >> >> >     <load module="mod_g723_1"/>
> >> >> >> >     <load module="mod_g729"/>
> >> >> >> >     <load module="mod_amr"/>
> >> >> >> >     <load module="mod_ilbc"/>
> >> >> >> >     <load module="mod_speex"/>
> >> >> >> >     <load module="mod_h26x"/>
> >> >> >> >     <load module="mod_siren"/>
> >> >> >> >     <!--<load module="mod_celt"/>-->
> >> >> >> >     <!--<load module="mod_opus"/>-->
> >> >> >> > Apart from settings I posted in my previous post, where else to
> >> >> >> > look
> >> >> >> > for
> >> >> >> > disabling LPC?
> >> >> >> > Malay
> >> >> >> > On Mon, Feb 28, 2011 at 12:05 PM, Anthony Minessale
> >> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >> >>
> >> >> >> >> is your inbound call using LPC? you don't want to be using LPC
> >> >> >> >> and
> >> >> >> >> expect anything to sound good that's for sure.
> >> >> >> >> It would not just magically say that unless something you are
> >> >> >> >> doing
> >> >> >> >> has
> >> >> >> >> LPC?
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> On Mon, Feb 28, 2011 at 10:24 AM, Malay Thakershi
> >> >> >> >> <mthakershi at gmail.com>
> >> >> >> >> wrote:
> >> >> >> >> > Hello,
> >> >> >> >> > I updated to the latest FS version last week.
> >> >> >> >> > I started getting the following warning when speech / sound
> is
> >> >> >> >> > played
> >> >> >> >> > on
> >> >> >> >> > the
> >> >> >> >> > call.
> >> >> >> >> > "2011-02-28 10:15:55.985930 [WARNING] sofia_glue.c:213 Codec
> >> >> >> >> > LPC
> >> >> >> >> > payload
> >> >> >> >> > 7
> >> >> >> >> > added to sdp wanting ptime 90 but it's already 20
> >> >> >> >> > (G7221:115:20),
> >> >> >> >> > disabling
> >> >> >> >> > ptime."
> >> >> >> >> > I read sections on codecs and negotiations.
> >> >> >> >> > Following are the settings from vars.xml (I have not changed
> >> >> >> >> > them):
> >> >> >> >> >   <X-PRE-PROCESS cmd="set"
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h
> ,G722,PCMU,PCMA,GSM"/>
> >> >> >> >> >   <X-PRE-PROCESS cmd="set"
> >> >> >> >> > data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
> >> >> >> >> > Also, there is no codec related setting in sip_profiles files
> >> >> >> >> > and sofia.conf.xml file.
> >> >> >> >> > I am playing audio files using Cepstral TTS during the call.
> >> >> >> >> > Can someone please help me understand these settings? And if
> >> >> >> >> > they
> >> >> >> >> > are
> >> >> >> >> > appropriate?
> >> >> >> >> > Thank you.
> >> >> >> >> > Malay
> >> >> >> >> > _______________________________________________
> >> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> > 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-users mailing list
> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >> > _______________________________________________
> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >
> >> >> >> >
> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> > 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-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > 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-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > 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-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Tue Mar  1 02:02:10 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Feb 2011 17:02:10 -0600
Subject: [Freeswitch-users] mod_dingaling and google voice
In-Reply-To: <4D6C288B.4050801@cartissolutions.com>
References: <AANLkTin_dFj9ygx3TAmExeN9-pDFNCuZPGZ_Uc2d7o1W@mail.gmail.com>
	<AANLkTimUr3=w_fj_k643ChW93Q_YTm63R_qFyBO9EoQV@mail.gmail.com>
	<4D6C1978.8040702@cartissolutions.com>
	<AANLkTim5TpcQeYSC_c6ZH9WMoJnq1aNvYViwUBkZr=fv@mail.gmail.com>
	<4D6C288B.4050801@cartissolutions.com>
Message-ID: <AANLkTikncVxc7ebrzR4bHrzeHARKVU_Q+aRpzMTM9RwV@mail.gmail.com>

you might want to backup your configs and start over with the fresh
ones just to test and work backwards to see what's different.
its probably something to do with having to omit external-rtp-ip


On Mon, Feb 28, 2011 at 4:58 PM, Yossi Neiman
<freeswitch at cartissolutions.com> wrote:
> I am running with -nonat, the FS is running with a UA on a public IP
> address, etc, but both Michael Collins and I are still getting the RTP
> port 0 issue. ?I figured I'd compare notes with somebody who has a
> working config before I file this as something other than a PEBKAC issue...
>
>
> Yossi Neiman
> Cartis Solutions, Inc. - http://www.cartissolutions.com
>
>
> On 02/28/2011 04:07 PM, Anthony Minessale wrote:
>> The solution to the port 0 issue is to run FS with -nonat so it does
>> not get confused by broken upnp setup.
>>
>>
>> On Mon, Feb 28, 2011 at 3:54 PM, Yossi Neiman
>> <freeswitch at cartissolutions.com> ?wrote:
>>> What FS build are you running? ?I'd like to compare notes with you on your
>>> config, as I'm experiencing the RTP port 0 issue with FreeSWITCH Version
>>> 1.0.head (git-0d8e945 2011-02-16 08-57-43 -0600). ?What about inbound? ?Do
>>> you have that set up?
>>>
>>> Yossi Neiman
>>> Cartis Solutions, Inc. - http://www.cartissolutions.com
>>>
>>> On 02/28/2011 12:54 AM, envelopes envelopes wrote:
>>>
>>> never mind. add this line<action application="ring_ready"/> ?and restart FS
>>>
>>> fixed the issue.
>>>
>>>
>>>
>>> On Sun, Feb 27, 2011 at 10:30 PM, envelopes envelopes<sunwood360 at gmail.com>
>>> wrote:
>>>> Now I am able to use GV for outbound dialing. However, I don't hear any
>>>> ringback or not sure whether the other party has answered the call.
>>>> is there any config variable to set up so that I will be notified if my
>>>> call is answered?
>>>>
>>>> thanks!
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Tue Mar  1 02:06:34 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Feb 2011 17:06:34 -0600
Subject: [Freeswitch-users] FreeSwithc not handling no SDP INVITE
 properly for video call
In-Reply-To: <AANLkTimhRw7CXrU73yU9hs_to8DYf1disro5fidW5XYd@mail.gmail.com>
References: <AANLkTinoomTncUKH=xKLSfAaWKeTupOMv_aZuN_CVVF+@mail.gmail.com>
	<AANLkTinxTuAv0ybtguJtZh3Xkc11=EHJo6CQwdRmRphV@mail.gmail.com>
	<AANLkTimvG+_vo5O0jH1nnHMaHy-a7UqwFUS2+Dsgs+kK@mail.gmail.com>
	<AANLkTimzvZ=0Rvq76_0GAcfcnqSj5b-J80n8wW=1OgSQ@mail.gmail.com>
	<AANLkTimhRw7CXrU73yU9hs_to8DYf1disro5fidW5XYd@mail.gmail.com>
Message-ID: <AANLkTimuVcYo3h8YAePuqMaEwPLv6GRppukrmwjoLmnr@mail.gmail.com>

ok then file it on jira
http://jira.freeswitch.org
It's closed for maintenance so report it tomorrow.


On Mon, Feb 28, 2011 at 4:57 PM, George Lee <whglee at gmail.com> wrote:
> I have already had this setup in my dialplan in this failure case.
> ? ? ? ? <action application="set" data="hangup_after_bridge=true"/>
> ? ? ? ? <action application="set" data="continue_on_fail=true"/>
> ? ? ? ? <action application="set" data="inherit_codec=true"/>
> ? ? ? ? <action application="set" data="proxy_media=false"/>
>
> - George
>
> On Mon, Feb 28, 2011 at 5:43 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> try
>>
>> <action application="set" data="inherit_codec=true"/>
>>
>> in your dialplan before you call the phone.
>>
>>
>>
>> On Mon, Feb 28, 2011 at 12:04 PM, George Lee <whglee at gmail.com> wrote:
>>> I pulled from the latest source tree and rebuilt FreeSwitch. The
>>> problem still persists.
>>> http://pastebin.freeswitch.org/15486
>>> Any other suggestions you could provide?
>>>
>>> Thanks,
>>> George
>>>
>>> On Fri, Feb 25, 2011 at 5:52 PM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>>> make sure you have tested on latest GIT (minutes ago) there was just a
>>>> fix to some video issues.
>>>>
>>>>
>>>> On Fri, Feb 25, 2011 at 3:56 PM, George Lee <whglee at gmail.com> wrote:
>>>>> Hi all,
>>>>>
>>>>> I am having trouble with FreeSwitch handling late codec negotiation
>>>>> for video calls.
>>>>>
>>>>> The call logs are here:
>>>>> http://pastebin.freeswitch.org/15481
>>>>>
>>>>> I have this line:
>>>>> <param name="enable-3pcc" value="proxy"/>
>>>>> added to the external sip_profile for handling late codec negotiation.
>>>>> I also have
>>>>> ?<X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,H263"/>
>>>>> ?<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,H263"/>
>>>>> in vars.xml for codec support list.
>>>>>
>>>>> The initial INVITE contains no SDP for the caller and once the callee
>>>>> answers the call, it sends the 200 OK with audio (pcma, pcmu) and
>>>>> video (H263) codec capabilities. FreeSwitch forwards the 200 OK to the
>>>>> caller without video codec capabilities. As a result, the video call
>>>>> does not establish properly.
>>>>>
>>>>> Could someone give me a pointer?
>>>>>
>>>>> Thanks,
>>>>> George
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> 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-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Tue Mar  1 02:08:38 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Feb 2011 17:08:38 -0600
Subject: [Freeswitch-users] disabling ptime warning message
In-Reply-To: <AANLkTik5r_qHjgSgGk-_v+9VDM-fPZmGC-4fTZzPw=b2@mail.gmail.com>
References: <AANLkTim97P5=MJgKq3ncDmbTiXkP63CBoALRD6o+b-Uw@mail.gmail.com>
	<AANLkTikw3TLW8bo39WRX6wN44vhcGSDBBTS_QffqBTwc@mail.gmail.com>
	<AANLkTi=FGWs2bs3EFvaMTTyG0=0inwe5z2UfByctX-g1@mail.gmail.com>
	<AANLkTindiPgTH65=0MwBgdviyWByQzmu1=01MYUy_fXy@mail.gmail.com>
	<AANLkTi=cFCJssphqmji_8LTXTg3ZycGfswMM77hmkie9@mail.gmail.com>
	<AANLkTimTKLJQ+cE9r5t2fRR+riDo-DPi+7FQLJecoc9M@mail.gmail.com>
	<AANLkTi=A-qZXz23nBgFz+LgAKCsqOmFFfgs8zm8k8_Re@mail.gmail.com>
	<AANLkTikvOhbJZfpbN9BmLs3eFFuz=uG2jRyKM4J5zYus@mail.gmail.com>
	<AANLkTi=9rwDSgyt5kDj8LhG+CMFAfXGjxxfcpquJxPxH@mail.gmail.com>
	<AANLkTimcqpeHJN+wJ+i+DWg4ii6PYuHwF2wDf5fpyfhz@mail.gmail.com>
	<AANLkTik5r_qHjgSgGk-_v+9VDM-fPZmGC-4fTZzPw=b2@mail.gmail.com>
Message-ID: <AANLkTinDkJ6OEsgsrNzAuQ7nA8SyBJJ4wxLoS+4odWtr@mail.gmail.com>

You are not including the entire log going back to when the call first HITS FS.
We need to see the initial invite in the log.



On Mon, Feb 28, 2011 at 4:58 PM, Malay Thakershi <mthakershi at gmail.com> wrote:
> http://pastebin.freeswitch.org/15500
> Thank you.
> Malay
>
> On Mon, Feb 28, 2011 at 4:55 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> If you read the dialog box closely the login and password are stated
>> in the challenge box.
>> That is our way of telling who pays attention and who doesn't...
>>
>> not really, its to keep spambots off the site.
>>
>>
>> On Mon, Feb 28, 2011 at 4:53 PM, Malay Thakershi <mthakershi at gmail.com>
>> wrote:
>> > I can't login to FS pastebin.
>> > I used same user / password as I use to access users group.
>> > Malay
>> >
>> > On Mon, Feb 28, 2011 at 3:47 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> You also need "sofia global siptrace on" so we can see the SIP traffic
>> >> in the log.
>> >> We have our own pastebin at http://pastebin.freeswitch.org
>> >>
>> >>
>> >> On Mon, Feb 28, 2011 at 3:22 PM, Malay Thakershi <mthakershi at gmail.com>
>> >> wrote:
>> >> > http://pastebin.com/CCwmAfZK
>> >> > I noticed that LPC warning only comes for outbound calls I initiate
>> >> > using
>> >> > "Originate" API.
>> >> > Malay
>> >> >
>> >> > On Mon, Feb 28, 2011 at 2:04 PM, Anthony Minessale
>> >> > <anthony.minessale at gmail.com> wrote:
>> >> >>
>> >> >> That's L16 not LPC, you need L16 to play files.
>> >> >> Why don't you just put the whole log of the call on pastebin.
>> >> >>
>> >> >>
>> >> >> On Mon, Feb 28, 2011 at 1:53 PM, Malay Thakershi
>> >> >> <mthakershi at gmail.com>
>> >> >> wrote:
>> >> >> > Following is a section from the log:
>> >> >> > ---------------------------
>> >> >> > 2011-02-28 13:36:08.294241 [DEBUG] sofia_glue.c:2760 Set Codec
>> >> >> > sofia/sipinterface_1/##########@##.##.##.## PCMU/8000 20 ms 160
>> >> >> > samples
>> >> >> > 64000 bits
>> >> >> > 2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2298 OPEN
>> >> >> > TTS
>> >> >> > cepstral
>> >> >> > 2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2307 Raw
>> >> >> > Codec
>> >> >> > Activated
>> >> >> > 2011-02-28 13:36:15.017444 [DEBUG] switch_ivr_play_say.c:1996
>> >> >> > Speaking
>> >> >> > text:
>> >> >> > <break strength='medium'/>We must verify your identity.
>> >> >> > 2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:2188 done
>> >> >> > speaking
>> >> >> > text
>> >> >> > 2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:1244
>> >> >> > Codec
>> >> >> > Activated L16 at 8000hz 1 channels 20ms
>> >> >> > 2011-02-28 13:36:25.530443 [DEBUG] switch_ivr_play_say.c:1581 done
>> >> >> > playing
>> >> >> > file
>> >> >> > 2011-02-28 13:36:25.530443 [DEBUG] switch_ivr.c:1135 Codec
>> >> >> > Activated
>> >> >> > L16 at 8000hz 1 channels 20ms
>> >> >> > 2011-02-28 13:36:30.549653 [DEBUG] switch_ivr_play_say.c:1244
>> >> >> > Codec
>> >> >> > Activated L16 at 8000hz 1 channels 20ms
>> >> >> > ---------------------------
>> >> >> > So I see when I play a file (using StreamFile / PAGD), it
>> >> >> > activates
>> >> >> > L16,
>> >> >> > which the wiki pages says is not recommended. So should I
>> >> >> > deactivate
>> >> >> > it?
>> >> >> > If
>> >> >> > so, how?
>> >> >> > Now, I have not done any setting out of default / ordinary that
>> >> >> > comes
>> >> >> > with
>> >> >> > the build. I am playing WAV file that is generated by Cepstral
>> >> >> > SWIFT
>> >> >> > command
>> >> >> > line tool (text to WAV). The file format is "Wave PCM signed 16
>> >> >> > bit,
>> >> >> > 8000
>> >> >> > Hz, 128 kbps, mono".
>> >> >> > Thank you for help so far.
>> >> >> > Malay
>> >> >> > On Mon, Feb 28, 2011 at 12:50 PM, Anthony Minessale
>> >> >> > <anthony.minessale at gmail.com> wrote:
>> >> >> >>
>> >> >> >> look at your SIP traffic and console log.
>> >> >> >>
>> >> >> >> enter "sofia global siptrace on" followed by "console loglevel
>> >> >> >> debug"
>> >> >> >> at the cli and make the call.
>> >> >> >>
>> >> >> >>
>> >> >> >> On Mon, Feb 28, 2011 at 12:47 PM, Malay Thakershi
>> >> >> >> <mthakershi at gmail.com>
>> >> >> >> wrote:
>> >> >> >> > I have no idea where to look for this setting.
>> >> >> >> > This is in modules.conf.xml
>> >> >> >> > ?? ?<!-- Codec Interfaces -->
>> >> >> >> > ?? ?<load module="mod_spandsp"/>
>> >> >> >> > ?? ?<!--<load module="mod_voipcodecs"/>-->
>> >> >> >> > ?? ?<load module="mod_g723_1"/>
>> >> >> >> > ?? ?<load module="mod_g729"/>
>> >> >> >> > ?? ?<load module="mod_amr"/>
>> >> >> >> > ?? ?<load module="mod_ilbc"/>
>> >> >> >> > ?? ?<load module="mod_speex"/>
>> >> >> >> > ?? ?<load module="mod_h26x"/>
>> >> >> >> > ?? ?<load module="mod_siren"/>
>> >> >> >> > ?? ?<!--<load module="mod_celt"/>-->
>> >> >> >> > ?? ?<!--<load module="mod_opus"/>-->
>> >> >> >> > Apart from settings I posted in my previous post, where else to
>> >> >> >> > look
>> >> >> >> > for
>> >> >> >> > disabling LPC?
>> >> >> >> > Malay
>> >> >> >> > On Mon, Feb 28, 2011 at 12:05 PM, Anthony Minessale
>> >> >> >> > <anthony.minessale at gmail.com> wrote:
>> >> >> >> >>
>> >> >> >> >> is your inbound call using LPC? you don't want to be using LPC
>> >> >> >> >> and
>> >> >> >> >> expect anything to sound good that's for sure.
>> >> >> >> >> It would not just magically say that unless something you are
>> >> >> >> >> doing
>> >> >> >> >> has
>> >> >> >> >> LPC?
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> On Mon, Feb 28, 2011 at 10:24 AM, Malay Thakershi
>> >> >> >> >> <mthakershi at gmail.com>
>> >> >> >> >> wrote:
>> >> >> >> >> > Hello,
>> >> >> >> >> > I updated to the latest FS version last week.
>> >> >> >> >> > I started getting the following warning when speech / sound
>> >> >> >> >> > is
>> >> >> >> >> > played
>> >> >> >> >> > on
>> >> >> >> >> > the
>> >> >> >> >> > call.
>> >> >> >> >> > "2011-02-28 10:15:55.985930 [WARNING] sofia_glue.c:213 Codec
>> >> >> >> >> > LPC
>> >> >> >> >> > payload
>> >> >> >> >> > 7
>> >> >> >> >> > added to sdp wanting ptime?90 but it's already 20
>> >> >> >> >> > (G7221:115:20),
>> >> >> >> >> > disabling
>> >> >> >> >> > ptime."
>> >> >> >> >> > I read sections on codecs and negotiations.
>> >> >> >> >> > Following are the settings from vars.xml (I have not changed
>> >> >> >> >> > them):
>> >> >> >> >> > ??<X-PRE-PROCESS cmd="set"
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> > data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>> >> >> >> >> > ??<X-PRE-PROCESS cmd="set"
>> >> >> >> >> > data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>> >> >> >> >> > Also, there is no codec related setting in sip_profiles
>> >> >> >> >> > files
>> >> >> >> >> > and?sofia.conf.xml file.
>> >> >> >> >> > I am playing audio files using Cepstral TTS during the call.
>> >> >> >> >> > Can someone please help me understand these settings? And if
>> >> >> >> >> > they
>> >> >> >> >> > are
>> >> >> >> >> > appropriate?
>> >> >> >> >> > Thank you.
>> >> >> >> >> > Malay
>> >> >> >> >> > _______________________________________________
>> >> >> >> >> > FreeSWITCH-users mailing list
>> >> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> >> >
>> >> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> >> > 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-users mailing list
>> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> >> http://www.freeswitch.org
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > _______________________________________________
>> >> >> >> > FreeSWITCH-users mailing list
>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> > 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-users mailing list
>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > 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-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > 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-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 mthakershi at gmail.com  Tue Mar  1 02:12:35 2011
From: mthakershi at gmail.com (Malay Thakershi)
Date: Mon, 28 Feb 2011 17:12:35 -0600
Subject: [Freeswitch-users] disabling ptime warning message
In-Reply-To: <AANLkTinDkJ6OEsgsrNzAuQ7nA8SyBJJ4wxLoS+4odWtr@mail.gmail.com>
References: <AANLkTim97P5=MJgKq3ncDmbTiXkP63CBoALRD6o+b-Uw@mail.gmail.com>
	<AANLkTikw3TLW8bo39WRX6wN44vhcGSDBBTS_QffqBTwc@mail.gmail.com>
	<AANLkTi=FGWs2bs3EFvaMTTyG0=0inwe5z2UfByctX-g1@mail.gmail.com>
	<AANLkTindiPgTH65=0MwBgdviyWByQzmu1=01MYUy_fXy@mail.gmail.com>
	<AANLkTi=cFCJssphqmji_8LTXTg3ZycGfswMM77hmkie9@mail.gmail.com>
	<AANLkTimTKLJQ+cE9r5t2fRR+riDo-DPi+7FQLJecoc9M@mail.gmail.com>
	<AANLkTi=A-qZXz23nBgFz+LgAKCsqOmFFfgs8zm8k8_Re@mail.gmail.com>
	<AANLkTikvOhbJZfpbN9BmLs3eFFuz=uG2jRyKM4J5zYus@mail.gmail.com>
	<AANLkTi=9rwDSgyt5kDj8LhG+CMFAfXGjxxfcpquJxPxH@mail.gmail.com>
	<AANLkTimcqpeHJN+wJ+i+DWg4ii6PYuHwF2wDf5fpyfhz@mail.gmail.com>
	<AANLkTik5r_qHjgSgGk-_v+9VDM-fPZmGC-4fTZzPw=b2@mail.gmail.com>
	<AANLkTinDkJ6OEsgsrNzAuQ7nA8SyBJJ4wxLoS+4odWtr@mail.gmail.com>
Message-ID: <AANLkTiktR0=MxXE4frkjtG2VLRkoJkV9T6ot+eGUipms@mail.gmail.com>

I am including everything. First line - until the end.

On Mon, Feb 28, 2011 at 5:08 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> You are not including the entire log going back to when the call first HITS
> FS.
> We need to see the initial invite in the log.
>
>
>
> On Mon, Feb 28, 2011 at 4:58 PM, Malay Thakershi <mthakershi at gmail.com>
> wrote:
> > http://pastebin.freeswitch.org/15500
> > Thank you.
> > Malay
> >
> > On Mon, Feb 28, 2011 at 4:55 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> If you read the dialog box closely the login and password are stated
> >> in the challenge box.
> >> That is our way of telling who pays attention and who doesn't...
> >>
> >> not really, its to keep spambots off the site.
> >>
> >>
> >> On Mon, Feb 28, 2011 at 4:53 PM, Malay Thakershi <mthakershi at gmail.com>
> >> wrote:
> >> > I can't login to FS pastebin.
> >> > I used same user / password as I use to access users group.
> >> > Malay
> >> >
> >> > On Mon, Feb 28, 2011 at 3:47 PM, Anthony Minessale
> >> > <anthony.minessale at gmail.com> wrote:
> >> >>
> >> >> You also need "sofia global siptrace on" so we can see the SIP
> traffic
> >> >> in the log.
> >> >> We have our own pastebin at http://pastebin.freeswitch.org
> >> >>
> >> >>
> >> >> On Mon, Feb 28, 2011 at 3:22 PM, Malay Thakershi <
> mthakershi at gmail.com>
> >> >> wrote:
> >> >> > http://pastebin.com/CCwmAfZK
> >> >> > I noticed that LPC warning only comes for outbound calls I initiate
> >> >> > using
> >> >> > "Originate" API.
> >> >> > Malay
> >> >> >
> >> >> > On Mon, Feb 28, 2011 at 2:04 PM, Anthony Minessale
> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >>
> >> >> >> That's L16 not LPC, you need L16 to play files.
> >> >> >> Why don't you just put the whole log of the call on pastebin.
> >> >> >>
> >> >> >>
> >> >> >> On Mon, Feb 28, 2011 at 1:53 PM, Malay Thakershi
> >> >> >> <mthakershi at gmail.com>
> >> >> >> wrote:
> >> >> >> > Following is a section from the log:
> >> >> >> > ---------------------------
> >> >> >> > 2011-02-28 13:36:08.294241 [DEBUG] sofia_glue.c:2760 Set Codec
> >> >> >> > sofia/sipinterface_1/##########@##.##.##.## PCMU/8000 20 ms 160
> >> >> >> > samples
> >> >> >> > 64000 bits
> >> >> >> > 2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2298
> OPEN
> >> >> >> > TTS
> >> >> >> > cepstral
> >> >> >> > 2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2307
> Raw
> >> >> >> > Codec
> >> >> >> > Activated
> >> >> >> > 2011-02-28 13:36:15.017444 [DEBUG] switch_ivr_play_say.c:1996
> >> >> >> > Speaking
> >> >> >> > text:
> >> >> >> > <break strength='medium'/>We must verify your identity.
> >> >> >> > 2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:2188
> done
> >> >> >> > speaking
> >> >> >> > text
> >> >> >> > 2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:1244
> >> >> >> > Codec
> >> >> >> > Activated L16 at 8000hz 1 channels 20ms
> >> >> >> > 2011-02-28 13:36:25.530443 [DEBUG] switch_ivr_play_say.c:1581
> done
> >> >> >> > playing
> >> >> >> > file
> >> >> >> > 2011-02-28 13:36:25.530443 [DEBUG] switch_ivr.c:1135 Codec
> >> >> >> > Activated
> >> >> >> > L16 at 8000hz 1 channels 20ms
> >> >> >> > 2011-02-28 13:36:30.549653 [DEBUG] switch_ivr_play_say.c:1244
> >> >> >> > Codec
> >> >> >> > Activated L16 at 8000hz 1 channels 20ms
> >> >> >> > ---------------------------
> >> >> >> > So I see when I play a file (using StreamFile / PAGD), it
> >> >> >> > activates
> >> >> >> > L16,
> >> >> >> > which the wiki pages says is not recommended. So should I
> >> >> >> > deactivate
> >> >> >> > it?
> >> >> >> > If
> >> >> >> > so, how?
> >> >> >> > Now, I have not done any setting out of default / ordinary that
> >> >> >> > comes
> >> >> >> > with
> >> >> >> > the build. I am playing WAV file that is generated by Cepstral
> >> >> >> > SWIFT
> >> >> >> > command
> >> >> >> > line tool (text to WAV). The file format is "Wave PCM signed 16
> >> >> >> > bit,
> >> >> >> > 8000
> >> >> >> > Hz, 128 kbps, mono".
> >> >> >> > Thank you for help so far.
> >> >> >> > Malay
> >> >> >> > On Mon, Feb 28, 2011 at 12:50 PM, Anthony Minessale
> >> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >> >>
> >> >> >> >> look at your SIP traffic and console log.
> >> >> >> >>
> >> >> >> >> enter "sofia global siptrace on" followed by "console loglevel
> >> >> >> >> debug"
> >> >> >> >> at the cli and make the call.
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> On Mon, Feb 28, 2011 at 12:47 PM, Malay Thakershi
> >> >> >> >> <mthakershi at gmail.com>
> >> >> >> >> wrote:
> >> >> >> >> > I have no idea where to look for this setting.
> >> >> >> >> > This is in modules.conf.xml
> >> >> >> >> >     <!-- Codec Interfaces -->
> >> >> >> >> >     <load module="mod_spandsp"/>
> >> >> >> >> >     <!--<load module="mod_voipcodecs"/>-->
> >> >> >> >> >     <load module="mod_g723_1"/>
> >> >> >> >> >     <load module="mod_g729"/>
> >> >> >> >> >     <load module="mod_amr"/>
> >> >> >> >> >     <load module="mod_ilbc"/>
> >> >> >> >> >     <load module="mod_speex"/>
> >> >> >> >> >     <load module="mod_h26x"/>
> >> >> >> >> >     <load module="mod_siren"/>
> >> >> >> >> >     <!--<load module="mod_celt"/>-->
> >> >> >> >> >     <!--<load module="mod_opus"/>-->
> >> >> >> >> > Apart from settings I posted in my previous post, where else
> to
> >> >> >> >> > look
> >> >> >> >> > for
> >> >> >> >> > disabling LPC?
> >> >> >> >> > Malay
> >> >> >> >> > On Mon, Feb 28, 2011 at 12:05 PM, Anthony Minessale
> >> >> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >> >> >>
> >> >> >> >> >> is your inbound call using LPC? you don't want to be using
> LPC
> >> >> >> >> >> and
> >> >> >> >> >> expect anything to sound good that's for sure.
> >> >> >> >> >> It would not just magically say that unless something you
> are
> >> >> >> >> >> doing
> >> >> >> >> >> has
> >> >> >> >> >> LPC?
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >> On Mon, Feb 28, 2011 at 10:24 AM, Malay Thakershi
> >> >> >> >> >> <mthakershi at gmail.com>
> >> >> >> >> >> wrote:
> >> >> >> >> >> > Hello,
> >> >> >> >> >> > I updated to the latest FS version last week.
> >> >> >> >> >> > I started getting the following warning when speech /
> sound
> >> >> >> >> >> > is
> >> >> >> >> >> > played
> >> >> >> >> >> > on
> >> >> >> >> >> > the
> >> >> >> >> >> > call.
> >> >> >> >> >> > "2011-02-28 10:15:55.985930 [WARNING] sofia_glue.c:213
> Codec
> >> >> >> >> >> > LPC
> >> >> >> >> >> > payload
> >> >> >> >> >> > 7
> >> >> >> >> >> > added to sdp wanting ptime 90 but it's already 20
> >> >> >> >> >> > (G7221:115:20),
> >> >> >> >> >> > disabling
> >> >> >> >> >> > ptime."
> >> >> >> >> >> > I read sections on codecs and negotiations.
> >> >> >> >> >> > Following are the settings from vars.xml (I have not
> changed
> >> >> >> >> >> > them):
> >> >> >> >> >> >   <X-PRE-PROCESS cmd="set"
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> > data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h
> ,G722,PCMU,PCMA,GSM"/>
> >> >> >> >> >> >   <X-PRE-PROCESS cmd="set"
> >> >> >> >> >> > data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
> >> >> >> >> >> > Also, there is no codec related setting in sip_profiles
> >> >> >> >> >> > files
> >> >> >> >> >> > and sofia.conf.xml file.
> >> >> >> >> >> > I am playing audio files using Cepstral TTS during the
> call.
> >> >> >> >> >> > Can someone please help me understand these settings? And
> if
> >> >> >> >> >> > they
> >> >> >> >> >> > are
> >> >> >> >> >> > appropriate?
> >> >> >> >> >> > Thank you.
> >> >> >> >> >> > Malay
> >> >> >> >> >> > _______________________________________________
> >> >> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> >> >
> >> >> >> >> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> >> > 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-users mailing list
> >> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> >>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> >> http://www.freeswitch.org
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > _______________________________________________
> >> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> > 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-users mailing list
> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >> > _______________________________________________
> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >
> >> >> >> >
> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> > 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-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > 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-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > 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-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Tue Mar  1 02:12:47 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Feb 2011 17:12:47 -0600
Subject: [Freeswitch-users] ESL socket outbound: fail to stop tone detect
In-Reply-To: <AANLkTi=VnrfD+pKnG3a_T8bZnNPWQ-nAQ0Avp23V25E7@mail.gmail.com>
References: <AANLkTi=VnrfD+pKnG3a_T8bZnNPWQ-nAQ0Avp23V25E7@mail.gmail.com>
Message-ID: <AANLkTimEdf9ur=TpxScwfUTBCn7dUTqAS71wWzZQ3HRj@mail.gmail.com>

yes minor patch.
try latest git.
you are on record pace to find these strange edge cases.


On Mon, Feb 28, 2011 at 4:57 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:
> In an originated outbound session that is in progressing phase, I do, via
> esl socket outbound, a "tone_detect".
> When I receive the event "DETECTED_TONE" I do the "stop_tone_detect" but
> sometimes the stop has no effect and I see in the log the warning:
> "switch_core_session.c:1955 Cannot execute app 'stop_tone_detect' media
> required on an outbound channel that does not have media established"
> But the media is present (the tone is correctly detected!).
> The event I receive on this channel are (in sequence)
> CHANNEL_UUID
> CHANNEL_OUTGOING
> CHANNEL_ORIGINATE
> CHANNEL_STATE -?CS_INIT
> CHANNEL_STATE -?CS_ROUTING
> CHANNEL_STATE -?CS_CONSUME_MEDIA
> CHANNEL_PROGRESS
> CHANNEL_PROGRESS_MEDIA
> CHANNEL_STATE -?CS_EXECUTE
> CHANNEL_EXECUTE
> CHANNEL_PARK
> then I do the "tone_detect" and I receive:
> CHANNEL_EXECUTE
> MEDIA_BUG_START
> CHANNEL_EXECUTE_COMPLETE
> DETECTED_TONE
> so I do the "stop_tone_detect" that fails in the log
> Please, can anyone address me to solve this issue?
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 ayhkor at gmail.com  Tue Mar  1 02:18:52 2011
From: ayhkor at gmail.com (deniro)
Date: Mon, 28 Feb 2011 18:18:52 -0500
Subject: [Freeswitch-users] fs_cli socket connection error
In-Reply-To: <AANLkTikpQuxvyaU7PAGnyJuoQLQ9xn4inNJ_u5x41Anv@mail.gmail.com>
References: <AANLkTikCoT9dzLNoD=6TzS0jiMw4afNX94TypsLarBbz@mail.gmail.com>
	<AANLkTi=sENgBB9Cdr-EkLJD8XfmH3A8GSfPnvDHO7qrw@mail.gmail.com>
	<AANLkTin5k=2Z-=QVVzqt91172wv88P7OFkqRqTY9VDcS@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58ADB727D7@cooper>
	<AANLkTi=aPwmnVD8ZzaBVFDxGyPsbgzDEBGaXvq_B2NbF@mail.gmail.com>
	<AANLkTikpQuxvyaU7PAGnyJuoQLQ9xn4inNJ_u5x41Anv@mail.gmail.com>
Message-ID: <AANLkTimNYjC_G-Lq+St5VePpxEyM8iHnhaNCo=jBydi9@mail.gmail.com>

I put the log in pastebin
freeswitch.log only when starting freeswitch (after stop)

http://pastebin.freeswitch.org/15502

thx
 deniro

On Mon, Feb 28, 2011 at 5:46 PM, Steven Ayre <steveayre at gmail.com> wrote:

> You must check the logfile for when FS starts up - if netstat shows
> nothing for port 8021 then either mod_event_socket isn't being loaded
> or you'll see an error when it tries to load.
>
> Nothing from netstat means nothing's listening, so trying to connect
> using fs_cli won't do anything.
>
> -Steve
>
>
> On 28 February 2011 22:40, deniro <ayhkor at gmail.com> wrote:
> > Checked the log
> > Nothing is logged when
> > fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
> > is issued
> > /opt/freeswitch/log# tail -f freeswitch.log
> >
> > ls -l freeswitch.log
> > -rwxrwxrwx 1 freeswitch daemon 33018 2011-02-28 17:39 freeswitch.log
> > thx
> >
> >
> >
> > On Mon, Feb 28, 2011 at 3:59 PM, Peter Olsson
> > <peter.olsson at visionutveckling.se> wrote:
> >>
> >> Check freeswitch.log, it probably reports some problem when loading the
> >> mod_event_socket module.
> >>
> >> /Peter
> >> ________________________________________
> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> >> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r deniro
> >> [ayhkor at gmail.com]
> >> Skickat: den 28 februari 2011 21:42
> >> Till: FreeSWITCH Users Help
> >> ?mne: Re: [Freeswitch-users] fs_cli socket connection error
> >>
> >> Steve
> >> Thanks for the reply
> >> netstat -anlp | grep 8021  is blank (nothing showing up)
> >>
> >> /opt/freeswitch/bin/fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
> Configuration
> >> file is /root/.fs_cli_conf.
> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
> Configuration
> >> file is /etc/fs_cli.conf.
> >> [DEBUG] libs/esl/fs_cli.c:1159 main() profile default does not exist
> using
> >> builtin profile
> >> [DEBUG] libs/esl/fs_cli.c:1188 main() Using profile internal
> >> [75.xxx.xxx.xxx]
> >> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
> Connection
> >> Error]
> >> yes mod_event_socket is after  mod_xml_curl but I changed the order in
> >>  modules.conf.xml
> >> still getting above (restarted freeswitch)
> >>
> >> thx
> >> deniro--
> >>
> >>
> >>
> >> On Mon, Feb 28, 2011 at 3:35 AM, Steven Ayre
> >> <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
> >> Enable debug logging and you should see an error that'll tell you more.
> >>
> >> Is mod_event_socket loading before or after mod_xml_curl? Chances are
> >> mod_xml_curl is loading first, so it's trying to read
> >> event_socket.conf.xml and/or acl.conf.xml through xml_curl and either
> >> getting a different config to your previous local copy or the ACLs are
> >> different.
> >>
> >> What does "netstat -anlp | grep :8021" from the linux cli show you?
> >> Does it show that freeswitch is actually listening on the port? If it
> >> is it's probably an ACL problem, if it isn't then it's probably a
> >> problem with event_socket.conf.xml
> >>
> >> -Steve
> >>
> >> On 28 February 2011 00:53, deniro
> >> <ayhkor at gmail.com<mailto:ayhkor at gmail.com>> wrote:
> >> > What would be possible reasons for this and how to resolve?
> >> > running fs 106 on ubuntu 10.04 server
> >> >
> >> > /opt/freeswitch/bin/fs_cli -H xxx.xxx.xxx.xxx -P 8021
> >> > [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
> >> > Connection
> >> > Error]
> >> > was working fine before I installed  mod_xml_cdr
> >> > configure --prefix=/opt/freeswitch --without-libcurl
> >> > make mod_xml_cdr-install
> >> > (no errors)
> >> >
> >> > in modules.conf.xml
> >> > <load module="mod_event_socket"/>
> >> >
> >> > is there a packaged version of mod_xml_cdr (for fs 1.0.6)?
> >> >
> >> > thx
> >> > deniro--
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> >
> >> > FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org<http://www.freeswitch.org/>
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >>
> >> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org<http://www.freeswitch.org/>
> >>
> >> !DSPAM:4d6c0a1132761029518849!
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Tue Mar  1 02:18:52 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Feb 2011 17:18:52 -0600
Subject: [Freeswitch-users] disabling ptime warning message
In-Reply-To: <AANLkTiktR0=MxXE4frkjtG2VLRkoJkV9T6ot+eGUipms@mail.gmail.com>
References: <AANLkTim97P5=MJgKq3ncDmbTiXkP63CBoALRD6o+b-Uw@mail.gmail.com>
	<AANLkTikw3TLW8bo39WRX6wN44vhcGSDBBTS_QffqBTwc@mail.gmail.com>
	<AANLkTi=FGWs2bs3EFvaMTTyG0=0inwe5z2UfByctX-g1@mail.gmail.com>
	<AANLkTindiPgTH65=0MwBgdviyWByQzmu1=01MYUy_fXy@mail.gmail.com>
	<AANLkTi=cFCJssphqmji_8LTXTg3ZycGfswMM77hmkie9@mail.gmail.com>
	<AANLkTimTKLJQ+cE9r5t2fRR+riDo-DPi+7FQLJecoc9M@mail.gmail.com>
	<AANLkTi=A-qZXz23nBgFz+LgAKCsqOmFFfgs8zm8k8_Re@mail.gmail.com>
	<AANLkTikvOhbJZfpbN9BmLs3eFFuz=uG2jRyKM4J5zYus@mail.gmail.com>
	<AANLkTi=9rwDSgyt5kDj8LhG+CMFAfXGjxxfcpquJxPxH@mail.gmail.com>
	<AANLkTimcqpeHJN+wJ+i+DWg4ii6PYuHwF2wDf5fpyfhz@mail.gmail.com>
	<AANLkTik5r_qHjgSgGk-_v+9VDM-fPZmGC-4fTZzPw=b2@mail.gmail.com>
	<AANLkTinDkJ6OEsgsrNzAuQ7nA8SyBJJ4wxLoS+4odWtr@mail.gmail.com>
	<AANLkTiktR0=MxXE4frkjtG2VLRkoJkV9T6ot+eGUipms@mail.gmail.com>
Message-ID: <AANLkTikx4++bVqH_FASpsHgy5E2G4QLM9+7LGuhJk7eU@mail.gmail.com>

You did not mention that you are originating the call over event
socket or some script.
Check your sofia profile xml and make sure you have these set.

    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>

You are offering every single codec ever at the same time, many of
them will not go well with each other in the same offer.



On Mon, Feb 28, 2011 at 5:12 PM, Malay Thakershi <mthakershi at gmail.com> wrote:
> I am including everything. First line - until the end.
>
> On Mon, Feb 28, 2011 at 5:08 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> You are not including the entire log going back to when the call first
>> HITS FS.
>> We need to see the initial invite in the log.
>>
>>
>>
>> On Mon, Feb 28, 2011 at 4:58 PM, Malay Thakershi <mthakershi at gmail.com>
>> wrote:
>> > http://pastebin.freeswitch.org/15500
>> > Thank you.
>> > Malay
>> >
>> > On Mon, Feb 28, 2011 at 4:55 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> If you read the dialog box closely the login and password are stated
>> >> in the challenge box.
>> >> That is our way of telling who pays attention and who doesn't...
>> >>
>> >> not really, its to keep spambots off the site.
>> >>
>> >>
>> >> On Mon, Feb 28, 2011 at 4:53 PM, Malay Thakershi <mthakershi at gmail.com>
>> >> wrote:
>> >> > I can't login to FS pastebin.
>> >> > I used same user / password as I use to access users group.
>> >> > Malay
>> >> >
>> >> > On Mon, Feb 28, 2011 at 3:47 PM, Anthony Minessale
>> >> > <anthony.minessale at gmail.com> wrote:
>> >> >>
>> >> >> You also need "sofia global siptrace on" so we can see the SIP
>> >> >> traffic
>> >> >> in the log.
>> >> >> We have our own pastebin at http://pastebin.freeswitch.org
>> >> >>
>> >> >>
>> >> >> On Mon, Feb 28, 2011 at 3:22 PM, Malay Thakershi
>> >> >> <mthakershi at gmail.com>
>> >> >> wrote:
>> >> >> > http://pastebin.com/CCwmAfZK
>> >> >> > I noticed that LPC warning only comes for outbound calls I
>> >> >> > initiate
>> >> >> > using
>> >> >> > "Originate" API.
>> >> >> > Malay
>> >> >> >
>> >> >> > On Mon, Feb 28, 2011 at 2:04 PM, Anthony Minessale
>> >> >> > <anthony.minessale at gmail.com> wrote:
>> >> >> >>
>> >> >> >> That's L16 not LPC, you need L16 to play files.
>> >> >> >> Why don't you just put the whole log of the call on pastebin.
>> >> >> >>
>> >> >> >>
>> >> >> >> On Mon, Feb 28, 2011 at 1:53 PM, Malay Thakershi
>> >> >> >> <mthakershi at gmail.com>
>> >> >> >> wrote:
>> >> >> >> > Following is a section from the log:
>> >> >> >> > ---------------------------
>> >> >> >> > 2011-02-28 13:36:08.294241 [DEBUG] sofia_glue.c:2760 Set Codec
>> >> >> >> > sofia/sipinterface_1/##########@##.##.##.## PCMU/8000 20 ms 160
>> >> >> >> > samples
>> >> >> >> > 64000 bits
>> >> >> >> > 2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2298
>> >> >> >> > OPEN
>> >> >> >> > TTS
>> >> >> >> > cepstral
>> >> >> >> > 2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2307
>> >> >> >> > Raw
>> >> >> >> > Codec
>> >> >> >> > Activated
>> >> >> >> > 2011-02-28 13:36:15.017444 [DEBUG] switch_ivr_play_say.c:1996
>> >> >> >> > Speaking
>> >> >> >> > text:
>> >> >> >> > <break strength='medium'/>We must verify your identity.
>> >> >> >> > 2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:2188
>> >> >> >> > done
>> >> >> >> > speaking
>> >> >> >> > text
>> >> >> >> > 2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:1244
>> >> >> >> > Codec
>> >> >> >> > Activated L16 at 8000hz 1 channels 20ms
>> >> >> >> > 2011-02-28 13:36:25.530443 [DEBUG] switch_ivr_play_say.c:1581
>> >> >> >> > done
>> >> >> >> > playing
>> >> >> >> > file
>> >> >> >> > 2011-02-28 13:36:25.530443 [DEBUG] switch_ivr.c:1135 Codec
>> >> >> >> > Activated
>> >> >> >> > L16 at 8000hz 1 channels 20ms
>> >> >> >> > 2011-02-28 13:36:30.549653 [DEBUG] switch_ivr_play_say.c:1244
>> >> >> >> > Codec
>> >> >> >> > Activated L16 at 8000hz 1 channels 20ms
>> >> >> >> > ---------------------------
>> >> >> >> > So I see when I play a file (using StreamFile / PAGD), it
>> >> >> >> > activates
>> >> >> >> > L16,
>> >> >> >> > which the wiki pages says is not recommended. So should I
>> >> >> >> > deactivate
>> >> >> >> > it?
>> >> >> >> > If
>> >> >> >> > so, how?
>> >> >> >> > Now, I have not done any setting out of default / ordinary that
>> >> >> >> > comes
>> >> >> >> > with
>> >> >> >> > the build. I am playing WAV file that is generated by Cepstral
>> >> >> >> > SWIFT
>> >> >> >> > command
>> >> >> >> > line tool (text to WAV). The file format is "Wave PCM signed 16
>> >> >> >> > bit,
>> >> >> >> > 8000
>> >> >> >> > Hz, 128 kbps, mono".
>> >> >> >> > Thank you for help so far.
>> >> >> >> > Malay
>> >> >> >> > On Mon, Feb 28, 2011 at 12:50 PM, Anthony Minessale
>> >> >> >> > <anthony.minessale at gmail.com> wrote:
>> >> >> >> >>
>> >> >> >> >> look at your SIP traffic and console log.
>> >> >> >> >>
>> >> >> >> >> enter "sofia global siptrace on" followed by "console loglevel
>> >> >> >> >> debug"
>> >> >> >> >> at the cli and make the call.
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> On Mon, Feb 28, 2011 at 12:47 PM, Malay Thakershi
>> >> >> >> >> <mthakershi at gmail.com>
>> >> >> >> >> wrote:
>> >> >> >> >> > I have no idea where to look for this setting.
>> >> >> >> >> > This is in modules.conf.xml
>> >> >> >> >> > ?? ?<!-- Codec Interfaces -->
>> >> >> >> >> > ?? ?<load module="mod_spandsp"/>
>> >> >> >> >> > ?? ?<!--<load module="mod_voipcodecs"/>-->
>> >> >> >> >> > ?? ?<load module="mod_g723_1"/>
>> >> >> >> >> > ?? ?<load module="mod_g729"/>
>> >> >> >> >> > ?? ?<load module="mod_amr"/>
>> >> >> >> >> > ?? ?<load module="mod_ilbc"/>
>> >> >> >> >> > ?? ?<load module="mod_speex"/>
>> >> >> >> >> > ?? ?<load module="mod_h26x"/>
>> >> >> >> >> > ?? ?<load module="mod_siren"/>
>> >> >> >> >> > ?? ?<!--<load module="mod_celt"/>-->
>> >> >> >> >> > ?? ?<!--<load module="mod_opus"/>-->
>> >> >> >> >> > Apart from settings I posted in my previous post, where else
>> >> >> >> >> > to
>> >> >> >> >> > look
>> >> >> >> >> > for
>> >> >> >> >> > disabling LPC?
>> >> >> >> >> > Malay
>> >> >> >> >> > On Mon, Feb 28, 2011 at 12:05 PM, Anthony Minessale
>> >> >> >> >> > <anthony.minessale at gmail.com> wrote:
>> >> >> >> >> >>
>> >> >> >> >> >> is your inbound call using LPC? you don't want to be using
>> >> >> >> >> >> LPC
>> >> >> >> >> >> and
>> >> >> >> >> >> expect anything to sound good that's for sure.
>> >> >> >> >> >> It would not just magically say that unless something you
>> >> >> >> >> >> are
>> >> >> >> >> >> doing
>> >> >> >> >> >> has
>> >> >> >> >> >> LPC?
>> >> >> >> >> >>
>> >> >> >> >> >>
>> >> >> >> >> >> On Mon, Feb 28, 2011 at 10:24 AM, Malay Thakershi
>> >> >> >> >> >> <mthakershi at gmail.com>
>> >> >> >> >> >> wrote:
>> >> >> >> >> >> > Hello,
>> >> >> >> >> >> > I updated to the latest FS version last week.
>> >> >> >> >> >> > I started getting the following warning when speech /
>> >> >> >> >> >> > sound
>> >> >> >> >> >> > is
>> >> >> >> >> >> > played
>> >> >> >> >> >> > on
>> >> >> >> >> >> > the
>> >> >> >> >> >> > call.
>> >> >> >> >> >> > "2011-02-28 10:15:55.985930 [WARNING] sofia_glue.c:213
>> >> >> >> >> >> > Codec
>> >> >> >> >> >> > LPC
>> >> >> >> >> >> > payload
>> >> >> >> >> >> > 7
>> >> >> >> >> >> > added to sdp wanting ptime?90 but it's already 20
>> >> >> >> >> >> > (G7221:115:20),
>> >> >> >> >> >> > disabling
>> >> >> >> >> >> > ptime."
>> >> >> >> >> >> > I read sections on codecs and negotiations.
>> >> >> >> >> >> > Following are the settings from vars.xml (I have not
>> >> >> >> >> >> > changed
>> >> >> >> >> >> > them):
>> >> >> >> >> >> > ??<X-PRE-PROCESS cmd="set"
>> >> >> >> >> >> >
>> >> >> >> >> >> >
>> >> >> >> >> >> >
>> >> >> >> >> >> >
>> >> >> >> >> >> >
>> >> >> >> >> >> > data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>> >> >> >> >> >> > ??<X-PRE-PROCESS cmd="set"
>> >> >> >> >> >> > data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>> >> >> >> >> >> > Also, there is no codec related setting in sip_profiles
>> >> >> >> >> >> > files
>> >> >> >> >> >> > and?sofia.conf.xml file.
>> >> >> >> >> >> > I am playing audio files using Cepstral TTS during the
>> >> >> >> >> >> > call.
>> >> >> >> >> >> > Can someone please help me understand these settings? And
>> >> >> >> >> >> > if
>> >> >> >> >> >> > they
>> >> >> >> >> >> > are
>> >> >> >> >> >> > appropriate?
>> >> >> >> >> >> > Thank you.
>> >> >> >> >> >> > Malay
>> >> >> >> >> >> > _______________________________________________
>> >> >> >> >> >> > FreeSWITCH-users mailing list
>> >> >> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> >> >> >
>> >> >> >> >> >> >
>> >> >> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >> >> >
>> >> >> >> >> >> >
>> >> >> >> >> >> >
>> >> >> >> >> >> >
>> >> >> >> >> >> >
>> >> >> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> >> >> > 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-users mailing list
>> >> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> >> >>
>> >> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >> >>
>> >> >> >> >> >>
>> >> >> >> >> >>
>> >> >> >> >> >>
>> >> >> >> >> >>
>> >> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> >> >> http://www.freeswitch.org
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> > _______________________________________________
>> >> >> >> >> > FreeSWITCH-users mailing list
>> >> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> >> >
>> >> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> >> > 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-users mailing list
>> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> >> http://www.freeswitch.org
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > _______________________________________________
>> >> >> >> > FreeSWITCH-users mailing list
>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> > 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-users mailing list
>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > 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-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > 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-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Tue Mar  1 02:21:16 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Feb 2011 17:21:16 -0600
Subject: [Freeswitch-users] fs_cli socket connection error
In-Reply-To: <AANLkTimNYjC_G-Lq+St5VePpxEyM8iHnhaNCo=jBydi9@mail.gmail.com>
References: <AANLkTikCoT9dzLNoD=6TzS0jiMw4afNX94TypsLarBbz@mail.gmail.com>
	<AANLkTi=sENgBB9Cdr-EkLJD8XfmH3A8GSfPnvDHO7qrw@mail.gmail.com>
	<AANLkTin5k=2Z-=QVVzqt91172wv88P7OFkqRqTY9VDcS@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58ADB727D7@cooper>
	<AANLkTi=aPwmnVD8ZzaBVFDxGyPsbgzDEBGaXvq_B2NbF@mail.gmail.com>
	<AANLkTikpQuxvyaU7PAGnyJuoQLQ9xn4inNJ_u5x41Anv@mail.gmail.com>
	<AANLkTimNYjC_G-Lq+St5VePpxEyM8iHnhaNCo=jBydi9@mail.gmail.com>
Message-ID: <AANLkTimrorQz1aU0NvZ49322kWF9n1mMi_2vgpgRnBJo@mail.gmail.com>

seems to me like you solved your own problem when you say you see no activity.
It makes it pretty clear you have firewall or nat issue on your client
or server.


On Mon, Feb 28, 2011 at 5:18 PM, deniro <ayhkor at gmail.com> wrote:
> I put the log in pastebin
> freeswitch.log only?when starting freeswitch (after stop)
>
> http://pastebin.freeswitch.org/15502
> thx
> deniro
> On Mon, Feb 28, 2011 at 5:46 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> You must check the logfile for when FS starts up - if netstat shows
>> nothing for port 8021 then either mod_event_socket isn't being loaded
>> or you'll see an error when it tries to load.
>>
>> Nothing from netstat means nothing's listening, so trying to connect
>> using fs_cli won't do anything.
>>
>> -Steve
>>
>>
>> On 28 February 2011 22:40, deniro <ayhkor at gmail.com> wrote:
>> > Checked the log
>> > Nothing is logged when
>> > fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
>> > is issued
>> > /opt/freeswitch/log# tail -f freeswitch.log
>> >
>> > ls -l freeswitch.log
>> > -rwxrwxrwx 1 freeswitch daemon 33018 2011-02-28 17:39 freeswitch.log
>> > thx
>> >
>> >
>> >
>> > On Mon, Feb 28, 2011 at 3:59 PM, Peter Olsson
>> > <peter.olsson at visionutveckling.se> wrote:
>> >>
>> >> Check freeswitch.log, it probably reports some problem when loading the
>> >> mod_event_socket module.
>> >>
>> >> /Peter
>> >> ________________________________________
>> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> >> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r deniro
>> >> [ayhkor at gmail.com]
>> >> Skickat: den 28 februari 2011 21:42
>> >> Till: FreeSWITCH Users Help
>> >> ?mne: Re: [Freeswitch-users] fs_cli socket connection error
>> >>
>> >> Steve
>> >> Thanks for the reply
>> >> netstat -anlp | grep 8021 ?is blank (nothing showing up)
>> >>
>> >> /opt/freeswitch/bin/fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
>> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
>> >> Configuration
>> >> file is /root/.fs_cli_conf.
>> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
>> >> Configuration
>> >> file is /etc/fs_cli.conf.
>> >> [DEBUG] libs/esl/fs_cli.c:1159 main() profile default does not exist
>> >> using
>> >> builtin profile
>> >> [DEBUG] libs/esl/fs_cli.c:1188 main() Using profile internal
>> >> [75.xxx.xxx.xxx]
>> >> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
>> >> Connection
>> >> Error]
>> >> yes mod_event_socket is after ?mod_xml_curl but I changed the order in
>> >> ?modules.conf.xml
>> >> still getting above (restarted freeswitch)
>> >>
>> >> thx
>> >> deniro--
>> >>
>> >>
>> >>
>> >> On Mon, Feb 28, 2011 at 3:35 AM, Steven Ayre
>> >> <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
>> >> Enable debug logging and you should see an error that'll tell you more.
>> >>
>> >> Is mod_event_socket loading before or after mod_xml_curl? Chances are
>> >> mod_xml_curl is loading first, so it's trying to read
>> >> event_socket.conf.xml and/or acl.conf.xml through xml_curl and either
>> >> getting a different config to your previous local copy or the ACLs are
>> >> different.
>> >>
>> >> What does "netstat -anlp | grep :8021" from the linux cli show you?
>> >> Does it show that freeswitch is actually listening on the port? If it
>> >> is it's probably an ACL problem, if it isn't then it's probably a
>> >> problem with event_socket.conf.xml
>> >>
>> >> -Steve
>> >>
>> >> On 28 February 2011 00:53, deniro
>> >> <ayhkor at gmail.com<mailto:ayhkor at gmail.com>> wrote:
>> >> > What would be possible reasons for this and how to resolve?
>> >> > running fs 106 on ubuntu 10.04 server
>> >> >
>> >> > /opt/freeswitch/bin/fs_cli -H xxx.xxx.xxx.xxx -P 8021
>> >> > [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
>> >> > Connection
>> >> > Error]
>> >> > was working fine before I installed ?mod_xml_cdr
>> >> > configure --prefix=/opt/freeswitch --without-libcurl
>> >> > make mod_xml_cdr-install
>> >> > (no errors)
>> >> >
>> >> > in modules.conf.xml
>> >> > <load module="mod_event_socket"/>
>> >> >
>> >> > is there a packaged version of mod_xml_cdr (for fs 1.0.6)?
>> >> >
>> >> > thx
>> >> > deniro--
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> >
>> >> >
>> >> > FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org<http://www.freeswitch.org/>
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >>
>> >>
>> >> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org<http://www.freeswitch.org/>
>> >>
>> >> !DSPAM:4d6c0a1132761029518849!
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 curriegrad2004 at gmail.com  Tue Mar  1 02:24:54 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 28 Feb 2011 15:24:54 -0800
Subject: [Freeswitch-users] FreeSwitch Installation Error
In-Reply-To: <AANLkTinfmLJCs+YyERMmGLWDs4u7uOZruTRe-d+-HdWQ@mail.gmail.com>
References: <AANLkTinfmLJCs+YyERMmGLWDs4u7uOZruTRe-d+-HdWQ@mail.gmail.com>
Message-ID: <AANLkTin3aGOXL+Dmc8trq3U6PRE1Ki0JWaNGjNf-GPpA@mail.gmail.com>

probably that module is faulty, from what I can tell, it's expecting
to be installed in /usr/local but it's been configured to run from
/opt instead. I'd say configure it to run from /usr/local instead and
see what happens.

On Mon, Feb 28, 2011 at 10:14 AM, George Lee <whglee at gmail.com> wrote:
> Hi,
>
> I changed the installation directory to /opt/freeswitch
>> ./configure --prefix=/opt/freeswiDtch
>
> Then I (as root) ran make; make install and it gave me this error:
> ?/bin/bash ./libtool ? --mode=install /usr/bin/install -c
> 'ftmod_zt.la' '/opt/freeswitch/mod/ftmod_zt.la'
> libtool: install: error: cannot install `ftmod_zt.la' to a directory
> not ending in /usr/local/freeswitch/mod
> ?/bin/bash ./libtool ? --mode=install /usr/bin/install -c
> 'ftmod_skel.la' '/opt/freeswitch/mod/ftmod_skel.la'
> libtool: install: error: cannot install `ftmod_skel.la' to a directory
> not ending in /usr/local/freeswitch/mod
> ?/bin/bash ./libtool ? --mode=install /usr/bin/install -c
> 'ftmod_analog.la' '/opt/freeswitch/mod/ftmod_analog.la'
> libtool: install: error: cannot install `ftmod_analog.la' to a
> directory not ending in /usr/local/freeswitch/mod
> ?/bin/bash ./libtool ? --mode=install /usr/bin/install -c
> 'ftmod_analog_em.la' '/opt/freeswitch/mod/ftmod_analog_em.la'
> libtool: install: error: cannot install `ftmod_analog_em.la' to a
> directory not ending in /usr/local/freeswitch/mod
>
> What did I setup wrong?
>
> Thanks,
> George
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From wstephen80 at gmail.com  Tue Mar  1 02:29:03 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 1 Mar 2011 00:29:03 +0100
Subject: [Freeswitch-users] ESL socket outbound: fail to stop tone detect
In-Reply-To: <AANLkTimEdf9ur=TpxScwfUTBCn7dUTqAS71wWzZQ3HRj@mail.gmail.com>
References: <AANLkTi=VnrfD+pKnG3a_T8bZnNPWQ-nAQ0Avp23V25E7@mail.gmail.com>
	<AANLkTimEdf9ur=TpxScwfUTBCn7dUTqAS71wWzZQ3HRj@mail.gmail.com>
Message-ID: <AANLkTimNcrido9dEdqawNUqZVN8fN1=ST7LY_s2GR73L@mail.gmail.com>

Ok, thank you Anthony!

On Tue, Mar 1, 2011 at 12:12 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> yes minor patch.
> try latest git.
> you are on record pace to find these strange edge cases.
>
>
> On Mon, Feb 28, 2011 at 4:57 PM, Stephen Wilde <wstephen80 at gmail.com>
> wrote:
> > In an originated outbound session that is in progressing phase, I do, via
> > esl socket outbound, a "tone_detect".
> > When I receive the event "DETECTED_TONE" I do the "stop_tone_detect" but
> > sometimes the stop has no effect and I see in the log the warning:
> > "switch_core_session.c:1955 Cannot execute app 'stop_tone_detect' media
> > required on an outbound channel that does not have media established"
> > But the media is present (the tone is correctly detected!).
> > The event I receive on this channel are (in sequence)
> > CHANNEL_UUID
> > CHANNEL_OUTGOING
> > CHANNEL_ORIGINATE
> > CHANNEL_STATE - CS_INIT
> > CHANNEL_STATE - CS_ROUTING
> > CHANNEL_STATE - CS_CONSUME_MEDIA
> > CHANNEL_PROGRESS
> > CHANNEL_PROGRESS_MEDIA
> > CHANNEL_STATE - CS_EXECUTE
> > CHANNEL_EXECUTE
> > CHANNEL_PARK
> > then I do the "tone_detect" and I receive:
> > CHANNEL_EXECUTE
> > MEDIA_BUG_START
> > CHANNEL_EXECUTE_COMPLETE
> > DETECTED_TONE
> > so I do the "stop_tone_detect" that fails in the log
> > Please, can anyone address me to solve this issue?
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ayhkor at gmail.com  Tue Mar  1 02:37:19 2011
From: ayhkor at gmail.com (deniro)
Date: Mon, 28 Feb 2011 18:37:19 -0500
Subject: [Freeswitch-users] fs_cli socket connection error
In-Reply-To: <AANLkTimrorQz1aU0NvZ49322kWF9n1mMi_2vgpgRnBJo@mail.gmail.com>
References: <AANLkTikCoT9dzLNoD=6TzS0jiMw4afNX94TypsLarBbz@mail.gmail.com>
	<AANLkTi=sENgBB9Cdr-EkLJD8XfmH3A8GSfPnvDHO7qrw@mail.gmail.com>
	<AANLkTin5k=2Z-=QVVzqt91172wv88P7OFkqRqTY9VDcS@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58ADB727D7@cooper>
	<AANLkTi=aPwmnVD8ZzaBVFDxGyPsbgzDEBGaXvq_B2NbF@mail.gmail.com>
	<AANLkTikpQuxvyaU7PAGnyJuoQLQ9xn4inNJ_u5x41Anv@mail.gmail.com>
	<AANLkTimNYjC_G-Lq+St5VePpxEyM8iHnhaNCo=jBydi9@mail.gmail.com>
	<AANLkTimrorQz1aU0NvZ49322kWF9n1mMi_2vgpgRnBJo@mail.gmail.com>
Message-ID: <AANLkTi=EcGiLtjhTK8NY41WjqJQE8bqX6Q0kAJ6ne1B6@mail.gmail.com>

Anthony
Thanks for the quick response I appreciate the help but I am not getting
this
Again, It was working fine before I installed  mod_xml_cdr
configure --prefix=/opt/freeswitch --without-libcurl
make mod_xml_cdr-install
(no errors)

I have the tar of /opt/freeswitch  before  installing mod_xml_cdr
When I restore that tar eveything is working fine  and netstat  shows 8021
connections.
I tested again. Just restored from tar backup and I am able to run fs_cli
without any problem
my firewall settings are the same before and after tar restore.

/opt# tar -xvf freeswitch.tar.feb26
/etc/init.d/freeswitch start
/opt# /opt/freeswitch/bin/fs_cli -H 75.xxx.xxx.xxx -P 8021
            _____ ____     ____ _     ___
           |  ___/ ___|   / ___| |   |_ _|
           | |_  \___ \  | |   | |    | |
           |  _|  ___) | | |___| |___ | |
           |_|   |____/   \____|_____|___|
*******************************************************
* Anthony Minessale II, Ken Rice, Michael Jerris      *
* FreeSWITCH (http://www.freeswitch.org)              *
* Paypal Donations Appreciated: paypal at freeswitch.org *
* Brought to you by ClueCon http://www.cluecon.com/   *
*******************************************************
Type /help <enter> to see a list of commands

+OK log level  [7]
freeswitch at 75.xxx.xxx.xxx@internal>




On Mon, Feb 28, 2011 at 6:21 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> seems to me like you solved your own problem when you say you see no
> activity.
> It makes it pretty clear you have firewall or nat issue on your client
> or server.
>
>
> On Mon, Feb 28, 2011 at 5:18 PM, deniro <ayhkor at gmail.com> wrote:
> > I put the log in pastebin
> > freeswitch.log only when starting freeswitch (after stop)
> >
> > http://pastebin.freeswitch.org/15502
> > thx
> > deniro
> > On Mon, Feb 28, 2011 at 5:46 PM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >>
> >> You must check the logfile for when FS starts up - if netstat shows
> >> nothing for port 8021 then either mod_event_socket isn't being loaded
> >> or you'll see an error when it tries to load.
> >>
> >> Nothing from netstat means nothing's listening, so trying to connect
> >> using fs_cli won't do anything.
> >>
> >> -Steve
> >>
> >>
> >> On 28 February 2011 22:40, deniro <ayhkor at gmail.com> wrote:
> >> > Checked the log
> >> > Nothing is logged when
> >> > fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
> >> > is issued
> >> > /opt/freeswitch/log# tail -f freeswitch.log
> >> >
> >> > ls -l freeswitch.log
> >> > -rwxrwxrwx 1 freeswitch daemon 33018 2011-02-28 17:39 freeswitch.log
> >> > thx
> >> >
> >> >
> >> >
> >> > On Mon, Feb 28, 2011 at 3:59 PM, Peter Olsson
> >> > <peter.olsson at visionutveckling.se> wrote:
> >> >>
> >> >> Check freeswitch.log, it probably reports some problem when loading
> the
> >> >> mod_event_socket module.
> >> >>
> >> >> /Peter
> >> >> ________________________________________
> >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> >> >> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r deniro
> >> >> [ayhkor at gmail.com]
> >> >> Skickat: den 28 februari 2011 21:42
> >> >> Till: FreeSWITCH Users Help
> >> >> ?mne: Re: [Freeswitch-users] fs_cli socket connection error
> >> >>
> >> >> Steve
> >> >> Thanks for the reply
> >> >> netstat -anlp | grep 8021  is blank (nothing showing up)
> >> >>
> >> >> /opt/freeswitch/bin/fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
> >> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
> >> >> Configuration
> >> >> file is /root/.fs_cli_conf.
> >> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
> >> >> Configuration
> >> >> file is /etc/fs_cli.conf.
> >> >> [DEBUG] libs/esl/fs_cli.c:1159 main() profile default does not exist
> >> >> using
> >> >> builtin profile
> >> >> [DEBUG] libs/esl/fs_cli.c:1188 main() Using profile internal
> >> >> [75.xxx.xxx.xxx]
> >> >> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
> >> >> Connection
> >> >> Error]
> >> >> yes mod_event_socket is after  mod_xml_curl but I changed the order
> in
> >> >>  modules.conf.xml
> >> >> still getting above (restarted freeswitch)
> >> >>
> >> >> thx
> >> >> deniro--
> >> >>
> >> >>
> >> >>
> >> >> On Mon, Feb 28, 2011 at 3:35 AM, Steven Ayre
> >> >> <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
> >> >> Enable debug logging and you should see an error that'll tell you
> more.
> >> >>
> >> >> Is mod_event_socket loading before or after mod_xml_curl? Chances are
> >> >> mod_xml_curl is loading first, so it's trying to read
> >> >> event_socket.conf.xml and/or acl.conf.xml through xml_curl and either
> >> >> getting a different config to your previous local copy or the ACLs
> are
> >> >> different.
> >> >>
> >> >> What does "netstat -anlp | grep :8021" from the linux cli show you?
> >> >> Does it show that freeswitch is actually listening on the port? If it
> >> >> is it's probably an ACL problem, if it isn't then it's probably a
> >> >> problem with event_socket.conf.xml
> >> >>
> >> >> -Steve
> >> >>
> >> >> On 28 February 2011 00:53, deniro
> >> >> <ayhkor at gmail.com<mailto:ayhkor at gmail.com>> wrote:
> >> >> > What would be possible reasons for this and how to resolve?
> >> >> > running fs 106 on ubuntu 10.04 server
> >> >> >
> >> >> > /opt/freeswitch/bin/fs_cli -H xxx.xxx.xxx.xxx -P 8021
> >> >> > [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
> >> >> > Connection
> >> >> > Error]
> >> >> > was working fine before I installed  mod_xml_cdr
> >> >> > configure --prefix=/opt/freeswitch --without-libcurl
> >> >> > make mod_xml_cdr-install
> >> >> > (no errors)
> >> >> >
> >> >> > in modules.conf.xml
> >> >> > <load module="mod_event_socket"/>
> >> >> >
> >> >> > is there a packaged version of mod_xml_cdr (for fs 1.0.6)?
> >> >> >
> >> >> > thx
> >> >> > deniro--
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> >
> >> >> >
> >> >> > FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org<http://www.freeswitch.org/>
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >>
> >> >>
> >> >> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org<http://www.freeswitch.org/>
> >> >>
> >> >> !DSPAM:4d6c0a1132761029518849!
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Tue Mar  1 02:43:18 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Feb 2011 17:43:18 -0600
Subject: [Freeswitch-users] fs_cli socket connection error
In-Reply-To: <AANLkTi=EcGiLtjhTK8NY41WjqJQE8bqX6Q0kAJ6ne1B6@mail.gmail.com>
References: <AANLkTikCoT9dzLNoD=6TzS0jiMw4afNX94TypsLarBbz@mail.gmail.com>
	<AANLkTi=sENgBB9Cdr-EkLJD8XfmH3A8GSfPnvDHO7qrw@mail.gmail.com>
	<AANLkTin5k=2Z-=QVVzqt91172wv88P7OFkqRqTY9VDcS@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58ADB727D7@cooper>
	<AANLkTi=aPwmnVD8ZzaBVFDxGyPsbgzDEBGaXvq_B2NbF@mail.gmail.com>
	<AANLkTikpQuxvyaU7PAGnyJuoQLQ9xn4inNJ_u5x41Anv@mail.gmail.com>
	<AANLkTimNYjC_G-Lq+St5VePpxEyM8iHnhaNCo=jBydi9@mail.gmail.com>
	<AANLkTimrorQz1aU0NvZ49322kWF9n1mMi_2vgpgRnBJo@mail.gmail.com>
	<AANLkTi=EcGiLtjhTK8NY41WjqJQE8bqX6Q0kAJ6ne1B6@mail.gmail.com>
Message-ID: <AANLkTikYybkUJVycSwNQo7k0gz=utwZd476-0UFZ_QRy@mail.gmail.com>

that is because the module is probably not loading because that is not
a clean rebuild.

Try this:

configure --prefix=/opt/freeswitch --without-libcurl
make update-clean
make install




On Mon, Feb 28, 2011 at 5:37 PM, deniro <ayhkor at gmail.com> wrote:
>
> Anthony
> Thanks for the quick response I appreciate the help but I am not getting
> this
> Again, It was working fine before I installed? mod_xml_cdr
> configure --prefix=/opt/freeswitch --without-libcurl
> make mod_xml_cdr-install
> (no errors)
>
> I have the tar of /opt/freeswitch? before? installing mod_xml_cdr
> When I restore that tar eveything is working fine? and netstat? shows 8021
> connections.
> I tested again. Just restored?from tar backup and I am able to run fs_cli
> without any problem
> my firewall settings are the same before and after tar restore.
>
> /opt# tar -xvf freeswitch.tar.feb26
> /etc/init.d/freeswitch start
> /opt# /opt/freeswitch/bin/fs_cli -H 75.xxx.xxx.xxx -P 8021
> ??????????? _____ ____???? ____ _???? ___
> ?????????? |? ___/ ___|?? / ___| |?? |_ _|
> ?????????? | |_? \___ \? | |?? | |??? | |
> ?????????? |? _|? ___) | | |___| |___ | |
> ?????????? |_|?? |____/?? \____|_____|___|
> *******************************************************
> * Anthony Minessale II, Ken Rice, Michael Jerris????? *
> * FreeSWITCH (http://www.freeswitch.org)????????????? *
> * Paypal Donations Appreciated: paypal at freeswitch.org *
> * Brought to you by ClueCon http://www.cluecon.com/?? *
> *******************************************************
> Type /help <enter> to see a list of commands
> +OK log level? [7]
> freeswitch at 75.xxx.xxx.xxx@internal>
>
>
>
>
> On Mon, Feb 28, 2011 at 6:21 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> seems to me like you solved your own problem when you say you see no
>> activity.
>> It makes it pretty clear you have firewall or nat issue on your client
>> or server.
>>
>>
>> On Mon, Feb 28, 2011 at 5:18 PM, deniro <ayhkor at gmail.com> wrote:
>> > I put the log in pastebin
>> > freeswitch.log only?when starting freeswitch (after stop)
>> >
>> > http://pastebin.freeswitch.org/15502
>> > thx
>> > deniro
>> > On Mon, Feb 28, 2011 at 5:46 PM, Steven Ayre <steveayre at gmail.com>
>> > wrote:
>> >>
>> >> You must check the logfile for when FS starts up - if netstat shows
>> >> nothing for port 8021 then either mod_event_socket isn't being loaded
>> >> or you'll see an error when it tries to load.
>> >>
>> >> Nothing from netstat means nothing's listening, so trying to connect
>> >> using fs_cli won't do anything.
>> >>
>> >> -Steve
>> >>
>> >>
>> >> On 28 February 2011 22:40, deniro <ayhkor at gmail.com> wrote:
>> >> > Checked the log
>> >> > Nothing is logged when
>> >> > fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
>> >> > is issued
>> >> > /opt/freeswitch/log# tail -f freeswitch.log
>> >> >
>> >> > ls -l freeswitch.log
>> >> > -rwxrwxrwx 1 freeswitch daemon 33018 2011-02-28 17:39 freeswitch.log
>> >> > thx
>> >> >
>> >> >
>> >> >
>> >> > On Mon, Feb 28, 2011 at 3:59 PM, Peter Olsson
>> >> > <peter.olsson at visionutveckling.se> wrote:
>> >> >>
>> >> >> Check freeswitch.log, it probably reports some problem when loading
>> >> >> the
>> >> >> mod_event_socket module.
>> >> >>
>> >> >> /Peter
>> >> >> ________________________________________
>> >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> >> >> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r deniro
>> >> >> [ayhkor at gmail.com]
>> >> >> Skickat: den 28 februari 2011 21:42
>> >> >> Till: FreeSWITCH Users Help
>> >> >> ?mne: Re: [Freeswitch-users] fs_cli socket connection error
>> >> >>
>> >> >> Steve
>> >> >> Thanks for the reply
>> >> >> netstat -anlp | grep 8021 ?is blank (nothing showing up)
>> >> >>
>> >> >> /opt/freeswitch/bin/fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
>> >> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
>> >> >> Configuration
>> >> >> file is /root/.fs_cli_conf.
>> >> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
>> >> >> Configuration
>> >> >> file is /etc/fs_cli.conf.
>> >> >> [DEBUG] libs/esl/fs_cli.c:1159 main() profile default does not exist
>> >> >> using
>> >> >> builtin profile
>> >> >> [DEBUG] libs/esl/fs_cli.c:1188 main() Using profile internal
>> >> >> [75.xxx.xxx.xxx]
>> >> >> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
>> >> >> Connection
>> >> >> Error]
>> >> >> yes mod_event_socket is after ?mod_xml_curl but I changed the order
>> >> >> in
>> >> >> ?modules.conf.xml
>> >> >> still getting above (restarted freeswitch)
>> >> >>
>> >> >> thx
>> >> >> deniro--
>> >> >>
>> >> >>
>> >> >>
>> >> >> On Mon, Feb 28, 2011 at 3:35 AM, Steven Ayre
>> >> >> <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
>> >> >> Enable debug logging and you should see an error that'll tell you
>> >> >> more.
>> >> >>
>> >> >> Is mod_event_socket loading before or after mod_xml_curl? Chances
>> >> >> are
>> >> >> mod_xml_curl is loading first, so it's trying to read
>> >> >> event_socket.conf.xml and/or acl.conf.xml through xml_curl and
>> >> >> either
>> >> >> getting a different config to your previous local copy or the ACLs
>> >> >> are
>> >> >> different.
>> >> >>
>> >> >> What does "netstat -anlp | grep :8021" from the linux cli show you?
>> >> >> Does it show that freeswitch is actually listening on the port? If
>> >> >> it
>> >> >> is it's probably an ACL problem, if it isn't then it's probably a
>> >> >> problem with event_socket.conf.xml
>> >> >>
>> >> >> -Steve
>> >> >>
>> >> >> On 28 February 2011 00:53, deniro
>> >> >> <ayhkor at gmail.com<mailto:ayhkor at gmail.com>> wrote:
>> >> >> > What would be possible reasons for this and how to resolve?
>> >> >> > running fs 106 on ubuntu 10.04 server
>> >> >> >
>> >> >> > /opt/freeswitch/bin/fs_cli -H xxx.xxx.xxx.xxx -P 8021
>> >> >> > [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
>> >> >> > Connection
>> >> >> > Error]
>> >> >> > was working fine before I installed ?mod_xml_cdr
>> >> >> > configure --prefix=/opt/freeswitch --without-libcurl
>> >> >> > make mod_xml_cdr-install
>> >> >> > (no errors)
>> >> >> >
>> >> >> > in modules.conf.xml
>> >> >> > <load module="mod_event_socket"/>
>> >> >> >
>> >> >> > is there a packaged version of mod_xml_cdr (for fs 1.0.6)?
>> >> >> >
>> >> >> > thx
>> >> >> > deniro--
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org<http://www.freeswitch.org/>
>> >> >> >
>> >> >> >
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >>
>> >> >>
>> >> >>
>> >> >> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org<http://www.freeswitch.org/>
>> >> >>
>> >> >> !DSPAM:4d6c0a1132761029518849!
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 whglee at gmail.com  Tue Mar  1 02:52:42 2011
From: whglee at gmail.com (George Lee)
Date: Mon, 28 Feb 2011 18:52:42 -0500
Subject: [Freeswitch-users] FreeSwitch Installation Error
In-Reply-To: <AANLkTin3aGOXL+Dmc8trq3U6PRE1Ki0JWaNGjNf-GPpA@mail.gmail.com>
References: <AANLkTinfmLJCs+YyERMmGLWDs4u7uOZruTRe-d+-HdWQ@mail.gmail.com>
	<AANLkTin3aGOXL+Dmc8trq3U6PRE1Ki0JWaNGjNf-GPpA@mail.gmail.com>
Message-ID: <AANLkTinBuG1-+CVOL9tSOJns-FF+hmLtOPMQD+p1canV@mail.gmail.com>

Without using the --prefix=/opt/freeswitch, I was able to install
using the default /usr/local directory without any errors. The
question I have is that I wanted to install it to /opt/freeswitch but
it wouldn't let me. What can I do?

Thanks,
George

On Mon, Feb 28, 2011 at 6:24 PM, curriegrad2004
<curriegrad2004 at gmail.com> wrote:
> probably that module is faulty, from what I can tell, it's expecting
> to be installed in /usr/local but it's been configured to run from
> /opt instead. I'd say configure it to run from /usr/local instead and
> see what happens.
>
> On Mon, Feb 28, 2011 at 10:14 AM, George Lee <whglee at gmail.com> wrote:
>> Hi,
>>
>> I changed the installation directory to /opt/freeswitch
>>> ./configure --prefix=/opt/freeswiDtch
>>
>> Then I (as root) ran make; make install and it gave me this error:
>> ?/bin/bash ./libtool ? --mode=install /usr/bin/install -c
>> 'ftmod_zt.la' '/opt/freeswitch/mod/ftmod_zt.la'
>> libtool: install: error: cannot install `ftmod_zt.la' to a directory
>> not ending in /usr/local/freeswitch/mod
>> ?/bin/bash ./libtool ? --mode=install /usr/bin/install -c
>> 'ftmod_skel.la' '/opt/freeswitch/mod/ftmod_skel.la'
>> libtool: install: error: cannot install `ftmod_skel.la' to a directory
>> not ending in /usr/local/freeswitch/mod
>> ?/bin/bash ./libtool ? --mode=install /usr/bin/install -c
>> 'ftmod_analog.la' '/opt/freeswitch/mod/ftmod_analog.la'
>> libtool: install: error: cannot install `ftmod_analog.la' to a
>> directory not ending in /usr/local/freeswitch/mod
>> ?/bin/bash ./libtool ? --mode=install /usr/bin/install -c
>> 'ftmod_analog_em.la' '/opt/freeswitch/mod/ftmod_analog_em.la'
>> libtool: install: error: cannot install `ftmod_analog_em.la' to a
>> directory not ending in /usr/local/freeswitch/mod
>>
>> What did I setup wrong?
>>
>> Thanks,
>> George
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From curriegrad2004 at gmail.com  Tue Mar  1 02:56:11 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 28 Feb 2011 15:56:11 -0800
Subject: [Freeswitch-users] FreeSwitch Installation Error
In-Reply-To: <AANLkTinBuG1-+CVOL9tSOJns-FF+hmLtOPMQD+p1canV@mail.gmail.com>
References: <AANLkTinfmLJCs+YyERMmGLWDs4u7uOZruTRe-d+-HdWQ@mail.gmail.com>
	<AANLkTin3aGOXL+Dmc8trq3U6PRE1Ki0JWaNGjNf-GPpA@mail.gmail.com>
	<AANLkTinBuG1-+CVOL9tSOJns-FF+hmLtOPMQD+p1canV@mail.gmail.com>
Message-ID: <AANLkTikMEGOCU0C0agu0FM7zxo=HnzP2a+ZKGgVS30z2@mail.gmail.com>

To be honest, I don't really know why. It's probably the module you
were compiling was hardcoded to install in /usr/local. Try running a
git reset --hard and then pull the latest source from git again, and
then configure it to run under /opt again and see what happens.

If there's still a problem, you're more than welcome to file a bug
under JIRA for this behavior, and just out of curiosity, what distro
are you running FS under in?

On Mon, Feb 28, 2011 at 3:52 PM, George Lee <whglee at gmail.com> wrote:
> Without using the --prefix=/opt/freeswitch, I was able to install
> using the default /usr/local directory without any errors. The
> question I have is that I wanted to install it to /opt/freeswitch but
> it wouldn't let me. What can I do?
>
> Thanks,
> George
>
> On Mon, Feb 28, 2011 at 6:24 PM, curriegrad2004
> <curriegrad2004 at gmail.com> wrote:
>> probably that module is faulty, from what I can tell, it's expecting
>> to be installed in /usr/local but it's been configured to run from
>> /opt instead. I'd say configure it to run from /usr/local instead and
>> see what happens.
>>
>> On Mon, Feb 28, 2011 at 10:14 AM, George Lee <whglee at gmail.com> wrote:
>>> Hi,
>>>
>>> I changed the installation directory to /opt/freeswitch
>>>> ./configure --prefix=/opt/freeswiDtch
>>>
>>> Then I (as root) ran make; make install and it gave me this error:
>>> ?/bin/bash ./libtool ? --mode=install /usr/bin/install -c
>>> 'ftmod_zt.la' '/opt/freeswitch/mod/ftmod_zt.la'
>>> libtool: install: error: cannot install `ftmod_zt.la' to a directory
>>> not ending in /usr/local/freeswitch/mod
>>> ?/bin/bash ./libtool ? --mode=install /usr/bin/install -c
>>> 'ftmod_skel.la' '/opt/freeswitch/mod/ftmod_skel.la'
>>> libtool: install: error: cannot install `ftmod_skel.la' to a directory
>>> not ending in /usr/local/freeswitch/mod
>>> ?/bin/bash ./libtool ? --mode=install /usr/bin/install -c
>>> 'ftmod_analog.la' '/opt/freeswitch/mod/ftmod_analog.la'
>>> libtool: install: error: cannot install `ftmod_analog.la' to a
>>> directory not ending in /usr/local/freeswitch/mod
>>> ?/bin/bash ./libtool ? --mode=install /usr/bin/install -c
>>> 'ftmod_analog_em.la' '/opt/freeswitch/mod/ftmod_analog_em.la'
>>> libtool: install: error: cannot install `ftmod_analog_em.la' to a
>>> directory not ending in /usr/local/freeswitch/mod
>>>
>>> What did I setup wrong?
>>>
>>> Thanks,
>>> George
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From anthony.minessale at gmail.com  Tue Mar  1 02:57:01 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Feb 2011 17:57:01 -0600
Subject: [Freeswitch-users] FreeSwitch Installation Error
In-Reply-To: <AANLkTinBuG1-+CVOL9tSOJns-FF+hmLtOPMQD+p1canV@mail.gmail.com>
References: <AANLkTinfmLJCs+YyERMmGLWDs4u7uOZruTRe-d+-HdWQ@mail.gmail.com>
	<AANLkTin3aGOXL+Dmc8trq3U6PRE1Ki0JWaNGjNf-GPpA@mail.gmail.com>
	<AANLkTinBuG1-+CVOL9tSOJns-FF+hmLtOPMQD+p1canV@mail.gmail.com>
Message-ID: <AANLkTim7nHkz6Uk1e9G7tGVViLgN+r1js4pOBBFyA77=@mail.gmail.com>

you probably need to report a bug under FreeTDM saying that the build
system is not using the top level configure prefix.


On Mon, Feb 28, 2011 at 5:52 PM, George Lee <whglee at gmail.com> wrote:
> Without using the --prefix=/opt/freeswitch, I was able to install
> using the default /usr/local directory without any errors. The
> question I have is that I wanted to install it to /opt/freeswitch but
> it wouldn't let me. What can I do?
>
> Thanks,
> George
>
> On Mon, Feb 28, 2011 at 6:24 PM, curriegrad2004
> <curriegrad2004 at gmail.com> wrote:
>> probably that module is faulty, from what I can tell, it's expecting
>> to be installed in /usr/local but it's been configured to run from
>> /opt instead. I'd say configure it to run from /usr/local instead and
>> see what happens.
>>
>> On Mon, Feb 28, 2011 at 10:14 AM, George Lee <whglee at gmail.com> wrote:
>>> Hi,
>>>
>>> I changed the installation directory to /opt/freeswitch
>>>> ./configure --prefix=/opt/freeswiDtch
>>>
>>> Then I (as root) ran make; make install and it gave me this error:
>>> ?/bin/bash ./libtool ? --mode=install /usr/bin/install -c
>>> 'ftmod_zt.la' '/opt/freeswitch/mod/ftmod_zt.la'
>>> libtool: install: error: cannot install `ftmod_zt.la' to a directory
>>> not ending in /usr/local/freeswitch/mod
>>> ?/bin/bash ./libtool ? --mode=install /usr/bin/install -c
>>> 'ftmod_skel.la' '/opt/freeswitch/mod/ftmod_skel.la'
>>> libtool: install: error: cannot install `ftmod_skel.la' to a directory
>>> not ending in /usr/local/freeswitch/mod
>>> ?/bin/bash ./libtool ? --mode=install /usr/bin/install -c
>>> 'ftmod_analog.la' '/opt/freeswitch/mod/ftmod_analog.la'
>>> libtool: install: error: cannot install `ftmod_analog.la' to a
>>> directory not ending in /usr/local/freeswitch/mod
>>> ?/bin/bash ./libtool ? --mode=install /usr/bin/install -c
>>> 'ftmod_analog_em.la' '/opt/freeswitch/mod/ftmod_analog_em.la'
>>> libtool: install: error: cannot install `ftmod_analog_em.la' to a
>>> directory not ending in /usr/local/freeswitch/mod
>>>
>>> What did I setup wrong?
>>>
>>> Thanks,
>>> George
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 wstephen80 at gmail.com  Tue Mar  1 03:16:10 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 1 Mar 2011 01:16:10 +0100
Subject: [Freeswitch-users] FreeTDM errors in log file
Message-ID: <AANLkTinZkJPS+gywSp0EYu_3Wx=C=WEqRK_QSnfyz4dw@mail.gmail.com>

I'm using Sangoma A108 board in my FS installation and my log is full of
warnings as:

2011-03-01 00:18:14.947380 [WARNING] mod_freetdm.c:434 [s17c22][17:22] VETO
state change from PROGRESS_MEDIA to PROCEED
2011-03-01 00:18:53.151673 [WARNING] mod_freetdm.c:434 [s3c11][3:11] VETO
state change from PROGRESS_MEDIA to PROCEED
2011-03-01 00:19:23.001140 [WARNING] mod_freetdm.c:434 [s23c29][23:29] VETO
state change from RINGING to PROCEED
2011-03-01 00:19:40.862103 [WARNING] mod_freetdm.c:434 [s3c21][3:21] VETO
state change from PROGRESS_MEDIA to PROCEED
2011-03-01 00:19:46.022154 [WARNING] mod_freetdm.c:434 [s3c21][3:21] VETO
state change from PROGRESS_MEDIA to PROCEED
2011-03-01 00:19:50.608224 [WARNING] mod_freetdm.c:434 [s22c24][22:24] VETO
state change from PROGRESS_MEDIA to PROCEED
2011-03-01 00:19:52.422221 [WARNING] mod_freetdm.c:434 [s3c21][3:21] VETO
state change from PROGRESS_MEDIA to PROCEED
2011-03-01 00:21:36.461338 [WARNING] mod_freetdm.c:434 [s3c31][3:31] VETO
state change from PROGRESS_MEDIA to PROCEED
2011-03-01 00:21:36.981455 [WARNING] mod_freetdm.c:434 [s3c30][3:30] VETO
state change from PROGRESS_MEDIA to PROCEED

Another kind of messages that are in my log file are:

2011-03-01 00:53:14.427853 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:760
[SNGISDN Q931] s16: Protocol: Unknown Event Code(2): Incomp Msg(276)
2011-03-01 00:53:18.428846 [CRIT] ftmod_sangoma_isdn_stack_hndl.c:557
[s16c23][16:23] Received DISCONNECT in an invalid state (TERMINATING)
2011-03-01 00:53:18.428846 [WARNING] ftmod_sangoma_isdn_stack_hndl.c:562
[s16c23][16:23] Why bother changing state from TERMINATING to TERMINATING
2011-03-01 00:54:08.607351 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:760
[SNGISDN Q931] s17: Protocol: Unknown Event Code(2): Incomp Msg(276)
2011-03-01 00:54:12.607389 [CRIT] ftmod_sangoa_isdn_stack_hndl.c:557
[s17c20][17:20] Received DISCONNECT in an invalid state (TERMINATING)
2011-03-01 00:54:12.607389 [WARNING] ftmod_sangoma_isdn_stack_hndl.c:562
[s17c20][17:20] Why bother changing state from TERMINATING to TERMINATING
2011-03-01 00:56:45.595869 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:760
[SNGISDN Q931] s16: Protocol: Unknown Event Code(2): Incomp Msg(276)
2011-03-01 00:56:49.596913 [CRIT] ftmod_sangoma_isdn_stack_hndl.c:557
[s16c23][16:23] Received DISCONNECT in an invalid state (TERMINATING)
2011-03-01 00:56:49.596913 [WARNING] ftmod_sangoma_isdn_stack_hndl.c:562
[s16c23][16:23] Why bother changing state from TERMINATING to TERMINATING
2011-03-01 00:59:23.036314 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:760
[SNGISDN Q931] s17: Protocol: Unknown Event Code(2): Incomp Msg(276)
2011-03-01 00:59:27.037327 [CRIT] ftmod_sangoma_isdn_stack_hndl.c:557
[s17c20][17:20] Received DISCONNECT in an invalid state (TERMINATING)
2011-03-01 00:59:27.037327 [WARNING] ftmod_sangoma_isdn_stack_hndl.c:562
[s17c20][17:20] Why bother changing state from TERMINATING to TERMINATING

I'm using latest git of Freeswitch, Wanpipe 3.5.18 and lib Sangoma isdn
7.3.0.

Attached to this email the .pcap related to the call that generate the
critical error (to me it seems that is missed a CALL_PROCEEDING message).

Stephen
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From mthakershi at gmail.com  Tue Mar  1 03:34:21 2011
From: mthakershi at gmail.com (Malay Thakershi)
Date: Mon, 28 Feb 2011 18:34:21 -0600
Subject: [Freeswitch-users] disabling ptime warning message
In-Reply-To: <AANLkTikx4++bVqH_FASpsHgy5E2G4QLM9+7LGuhJk7eU@mail.gmail.com>
References: <AANLkTim97P5=MJgKq3ncDmbTiXkP63CBoALRD6o+b-Uw@mail.gmail.com>
	<AANLkTikw3TLW8bo39WRX6wN44vhcGSDBBTS_QffqBTwc@mail.gmail.com>
	<AANLkTi=FGWs2bs3EFvaMTTyG0=0inwe5z2UfByctX-g1@mail.gmail.com>
	<AANLkTindiPgTH65=0MwBgdviyWByQzmu1=01MYUy_fXy@mail.gmail.com>
	<AANLkTi=cFCJssphqmji_8LTXTg3ZycGfswMM77hmkie9@mail.gmail.com>
	<AANLkTimTKLJQ+cE9r5t2fRR+riDo-DPi+7FQLJecoc9M@mail.gmail.com>
	<AANLkTi=A-qZXz23nBgFz+LgAKCsqOmFFfgs8zm8k8_Re@mail.gmail.com>
	<AANLkTikvOhbJZfpbN9BmLs3eFFuz=uG2jRyKM4J5zYus@mail.gmail.com>
	<AANLkTi=9rwDSgyt5kDj8LhG+CMFAfXGjxxfcpquJxPxH@mail.gmail.com>
	<AANLkTimcqpeHJN+wJ+i+DWg4ii6PYuHwF2wDf5fpyfhz@mail.gmail.com>
	<AANLkTik5r_qHjgSgGk-_v+9VDM-fPZmGC-4fTZzPw=b2@mail.gmail.com>
	<AANLkTinDkJ6OEsgsrNzAuQ7nA8SyBJJ4wxLoS+4odWtr@mail.gmail.com>
	<AANLkTiktR0=MxXE4frkjtG2VLRkoJkV9T6ot+eGUipms@mail.gmail.com>
	<AANLkTikx4++bVqH_FASpsHgy5E2G4QLM9+7LGuhJk7eU@mail.gmail.com>
Message-ID: <AANLkTimpvsqxuE3DEpbeWoLtPOamtGSevvFnrGF9UBsp@mail.gmail.com>

Ok.
I am using managed esl to send originate.

I will put these params in sofia profiles.

You said I am offering all codecs at once. How do I restrict this? And what
should be the codec offered in my case?

Thank you.
On Feb 28, 2011 5:20 PM, "Anthony Minessale" <anthony.minessale at gmail.com>
wrote:
> You did not mention that you are originating the call over event
> socket or some script.
> Check your sofia profile xml and make sure you have these set.
>
> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
>
> You are offering every single codec ever at the same time, many of
> them will not go well with each other in the same offer.
>
>
>
> On Mon, Feb 28, 2011 at 5:12 PM, Malay Thakershi <mthakershi at gmail.com>
wrote:
>> I am including everything. First line - until the end.
>>
>> On Mon, Feb 28, 2011 at 5:08 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>>
>>> You are not including the entire log going back to when the call first
>>> HITS FS.
>>> We need to see the initial invite in the log.
>>>
>>>
>>>
>>> On Mon, Feb 28, 2011 at 4:58 PM, Malay Thakershi <mthakershi at gmail.com>
>>> wrote:
>>> > http://pastebin.freeswitch.org/15500
>>> > Thank you.
>>> > Malay
>>> >
>>> > On Mon, Feb 28, 2011 at 4:55 PM, Anthony Minessale
>>> > <anthony.minessale at gmail.com> wrote:
>>> >>
>>> >> If you read the dialog box closely the login and password are stated
>>> >> in the challenge box.
>>> >> That is our way of telling who pays attention and who doesn't...
>>> >>
>>> >> not really, its to keep spambots off the site.
>>> >>
>>> >>
>>> >> On Mon, Feb 28, 2011 at 4:53 PM, Malay Thakershi <
mthakershi at gmail.com>
>>> >> wrote:
>>> >> > I can't login to FS pastebin.
>>> >> > I used same user / password as I use to access users group.
>>> >> > Malay
>>> >> >
>>> >> > On Mon, Feb 28, 2011 at 3:47 PM, Anthony Minessale
>>> >> > <anthony.minessale at gmail.com> wrote:
>>> >> >>
>>> >> >> You also need "sofia global siptrace on" so we can see the SIP
>>> >> >> traffic
>>> >> >> in the log.
>>> >> >> We have our own pastebin at http://pastebin.freeswitch.org
>>> >> >>
>>> >> >>
>>> >> >> On Mon, Feb 28, 2011 at 3:22 PM, Malay Thakershi
>>> >> >> <mthakershi at gmail.com>
>>> >> >> wrote:
>>> >> >> > http://pastebin.com/CCwmAfZK
>>> >> >> > I noticed that LPC warning only comes for outbound calls I
>>> >> >> > initiate
>>> >> >> > using
>>> >> >> > "Originate" API.
>>> >> >> > Malay
>>> >> >> >
>>> >> >> > On Mon, Feb 28, 2011 at 2:04 PM, Anthony Minessale
>>> >> >> > <anthony.minessale at gmail.com> wrote:
>>> >> >> >>
>>> >> >> >> That's L16 not LPC, you need L16 to play files.
>>> >> >> >> Why don't you just put the whole log of the call on pastebin.
>>> >> >> >>
>>> >> >> >>
>>> >> >> >> On Mon, Feb 28, 2011 at 1:53 PM, Malay Thakershi
>>> >> >> >> <mthakershi at gmail.com>
>>> >> >> >> wrote:
>>> >> >> >> > Following is a section from the log:
>>> >> >> >> > ---------------------------
>>> >> >> >> > 2011-02-28 13:36:08.294241 [DEBUG] sofia_glue.c:2760 Set
Codec
>>> >> >> >> > sofia/sipinterface_1/##########@##.##.##.## PCMU/8000 20 ms
160
>>> >> >> >> > samples
>>> >> >> >> > 64000 bits
>>> >> >> >> > 2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2298
>>> >> >> >> > OPEN
>>> >> >> >> > TTS
>>> >> >> >> > cepstral
>>> >> >> >> > 2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2307
>>> >> >> >> > Raw
>>> >> >> >> > Codec
>>> >> >> >> > Activated
>>> >> >> >> > 2011-02-28 13:36:15.017444 [DEBUG] switch_ivr_play_say.c:1996
>>> >> >> >> > Speaking
>>> >> >> >> > text:
>>> >> >> >> > <break strength='medium'/>We must verify your identity.
>>> >> >> >> > 2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:2188
>>> >> >> >> > done
>>> >> >> >> > speaking
>>> >> >> >> > text
>>> >> >> >> > 2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:1244
>>> >> >> >> > Codec
>>> >> >> >> > Activated L16 at 8000hz 1 channels 20ms
>>> >> >> >> > 2011-02-28 13:36:25.530443 [DEBUG] switch_ivr_play_say.c:1581
>>> >> >> >> > done
>>> >> >> >> > playing
>>> >> >> >> > file
>>> >> >> >> > 2011-02-28 13:36:25.530443 [DEBUG] switch_ivr.c:1135 Codec
>>> >> >> >> > Activated
>>> >> >> >> > L16 at 8000hz 1 channels 20ms
>>> >> >> >> > 2011-02-28 13:36:30.549653 [DEBUG] switch_ivr_play_say.c:1244
>>> >> >> >> > Codec
>>> >> >> >> > Activated L16 at 8000hz 1 channels 20ms
>>> >> >> >> > ---------------------------
>>> >> >> >> > So I see when I play a file (using StreamFile / PAGD), it
>>> >> >> >> > activates
>>> >> >> >> > L16,
>>> >> >> >> > which the wiki pages says is not recommended. So should I
>>> >> >> >> > deactivate
>>> >> >> >> > it?
>>> >> >> >> > If
>>> >> >> >> > so, how?
>>> >> >> >> > Now, I have not done any setting out of default / ordinary
that
>>> >> >> >> > comes
>>> >> >> >> > with
>>> >> >> >> > the build. I am playing WAV file that is generated by
Cepstral
>>> >> >> >> > SWIFT
>>> >> >> >> > command
>>> >> >> >> > line tool (text to WAV). The file format is "Wave PCM signed
16
>>> >> >> >> > bit,
>>> >> >> >> > 8000
>>> >> >> >> > Hz, 128 kbps, mono".
>>> >> >> >> > Thank you for help so far.
>>> >> >> >> > Malay
>>> >> >> >> > On Mon, Feb 28, 2011 at 12:50 PM, Anthony Minessale
>>> >> >> >> > <anthony.minessale at gmail.com> wrote:
>>> >> >> >> >>
>>> >> >> >> >> look at your SIP traffic and console log.
>>> >> >> >> >>
>>> >> >> >> >> enter "sofia global siptrace on" followed by "console
loglevel
>>> >> >> >> >> debug"
>>> >> >> >> >> at the cli and make the call.
>>> >> >> >> >>
>>> >> >> >> >>
>>> >> >> >> >> On Mon, Feb 28, 2011 at 12:47 PM, Malay Thakershi
>>> >> >> >> >> <mthakershi at gmail.com>
>>> >> >> >> >> wrote:
>>> >> >> >> >> > I have no idea where to look for this setting.
>>> >> >> >> >> > This is in modules.conf.xml
>>> >> >> >> >> >     <!-- Codec Interfaces -->
>>> >> >> >> >> >     <load module="mod_spandsp"/>
>>> >> >> >> >> >     <!--<load module="mod_voipcodecs"/>-->
>>> >> >> >> >> >     <load module="mod_g723_1"/>
>>> >> >> >> >> >     <load module="mod_g729"/>
>>> >> >> >> >> >     <load module="mod_amr"/>
>>> >> >> >> >> >     <load module="mod_ilbc"/>
>>> >> >> >> >> >     <load module="mod_speex"/>
>>> >> >> >> >> >     <load module="mod_h26x"/>
>>> >> >> >> >> >     <load module="mod_siren"/>
>>> >> >> >> >> >     <!--<load module="mod_celt"/>-->
>>> >> >> >> >> >     <!--<load module="mod_opus"/>-->
>>> >> >> >> >> > Apart from settings I posted in my previous post, where
else
>>> >> >> >> >> > to
>>> >> >> >> >> > look
>>> >> >> >> >> > for
>>> >> >> >> >> > disabling LPC?
>>> >> >> >> >> > Malay
>>> >> >> >> >> > On Mon, Feb 28, 2011 at 12:05 PM, Anthony Minessale
>>> >> >> >> >> > <anthony.minessale at gmail.com> wrote:
>>> >> >> >> >> >>
>>> >> >> >> >> >> is your inbound call using LPC? you don't want to be
using
>>> >> >> >> >> >> LPC
>>> >> >> >> >> >> and
>>> >> >> >> >> >> expect anything to sound good that's for sure.
>>> >> >> >> >> >> It would not just magically say that unless something you
>>> >> >> >> >> >> are
>>> >> >> >> >> >> doing
>>> >> >> >> >> >> has
>>> >> >> >> >> >> LPC?
>>> >> >> >> >> >>
>>> >> >> >> >> >>
>>> >> >> >> >> >> On Mon, Feb 28, 2011 at 10:24 AM, Malay Thakershi
>>> >> >> >> >> >> <mthakershi at gmail.com>
>>> >> >> >> >> >> wrote:
>>> >> >> >> >> >> > Hello,
>>> >> >> >> >> >> > I updated to the latest FS version last week.
>>> >> >> >> >> >> > I started getting the following warning when speech /
>>> >> >> >> >> >> > sound
>>> >> >> >> >> >> > is
>>> >> >> >> >> >> > played
>>> >> >> >> >> >> > on
>>> >> >> >> >> >> > the
>>> >> >> >> >> >> > call.
>>> >> >> >> >> >> > "2011-02-28 10:15:55.985930 [WARNING] sofia_glue.c:213
>>> >> >> >> >> >> > Codec
>>> >> >> >> >> >> > LPC
>>> >> >> >> >> >> > payload
>>> >> >> >> >> >> > 7
>>> >> >> >> >> >> > added to sdp wanting ptime 90 but it's already 20
>>> >> >> >> >> >> > (G7221:115:20),
>>> >> >> >> >> >> > disabling
>>> >> >> >> >> >> > ptime."
>>> >> >> >> >> >> > I read sections on codecs and negotiations.
>>> >> >> >> >> >> > Following are the settings from vars.xml (I have not
>>> >> >> >> >> >> > changed
>>> >> >> >> >> >> > them):
>>> >> >> >> >> >> >   <X-PRE-PROCESS cmd="set"
>>> >> >> >> >> >> >
>>> >> >> >> >> >> >
>>> >> >> >> >> >> >
>>> >> >> >> >> >> >
>>> >> >> >> >> >> >
>>> >> >> >> >> >> > data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h
,G722,PCMU,PCMA,GSM"/>
>>> >> >> >> >> >> >   <X-PRE-PROCESS cmd="set"
>>> >> >> >> >> >> > data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>>> >> >> >> >> >> > Also, there is no codec related setting in sip_profiles
>>> >> >> >> >> >> > files
>>> >> >> >> >> >> > and sofia.conf.xml file.
>>> >> >> >> >> >> > I am playing audio files using Cepstral TTS during the
>>> >> >> >> >> >> > call.
>>> >> >> >> >> >> > Can someone please help me understand these settings?
And
>>> >> >> >> >> >> > if
>>> >> >> >> >> >> > they
>>> >> >> >> >> >> > are
>>> >> >> >> >> >> > appropriate?
>>> >> >> >> >> >> > Thank you.
>>> >> >> >> >> >> > Malay
>>> >> >> >> >> >> > _______________________________________________
>>> >> >> >> >> >> > FreeSWITCH-users mailing list
>>> >> >> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> >> >> >> >> >
>>> >> >> >> >> >> >
>>> >> >> >> >> >> >
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> >> >> >> >
>>> >> >> >> >> >> >
>>> >> >> >> >> >> >
>>> >> >> >> >> >> >
>>> >> >> >> >> >> >
>>> >> >> >> >> >> > UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> >> >> >> > 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-users mailing list
>>> >> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> >> >> >> >>
>>> >> >> >> >> >>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> >> >> >>
>>> >> >> >> >> >>
>>> >> >> >> >> >>
>>> >> >> >> >> >>
>>> >> >> >> >> >>
>>> >> >> >> >> >> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> >> >> >> http://www.freeswitch.org
>>> >> >> >> >> >
>>> >> >> >> >> >
>>> >> >> >> >> > _______________________________________________
>>> >> >> >> >> > FreeSWITCH-users mailing list
>>> >> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> >> >> >> >
>>> >> >> >> >> >
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> >> >> >
>>> >> >> >> >> >
>>> >> >> >> >> >
>>> >> >> >> >> >
>>> >> >> >> >> > UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> >> >> > 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-users mailing list
>>> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> >> >> >>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> >> >>
>>> >> >> >> >>
>>> >> >> >> >>
>>> >> >> >> >>
>>> >> >> >> >> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> >> >> http://www.freeswitch.org
>>> >> >> >> >
>>> >> >> >> >
>>> >> >> >> > _______________________________________________
>>> >> >> >> > FreeSWITCH-users mailing list
>>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> >> >
>>> >> >> >> >
>>> >> >> >> >
>>> >> >> >> > UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> >> > 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-users mailing list
>>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> >>
>>> >> >> >>
>>> >> >> >>
>>> >> >> >> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> >> http://www.freeswitch.org
>>> >> >> >
>>> >> >> >
>>> >> >> > _______________________________________________
>>> >> >> > FreeSWITCH-users mailing list
>>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> >
>>> >> >> >
>>> >> >> > UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> > 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-users mailing list
>>> >> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >>
>>> >> >>
>>> >> >> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> http://www.freeswitch.org
>>> >> >
>>> >> >
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >
>>> >> > UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> > 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-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > 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-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From msc at freeswitch.org  Tue Mar  1 05:44:35 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 28 Feb 2011 18:44:35 -0800
Subject: [Freeswitch-users] disabling ptime warning message
In-Reply-To: <AANLkTimpvsqxuE3DEpbeWoLtPOamtGSevvFnrGF9UBsp@mail.gmail.com>
References: <AANLkTim97P5=MJgKq3ncDmbTiXkP63CBoALRD6o+b-Uw@mail.gmail.com>
	<AANLkTikw3TLW8bo39WRX6wN44vhcGSDBBTS_QffqBTwc@mail.gmail.com>
	<AANLkTi=FGWs2bs3EFvaMTTyG0=0inwe5z2UfByctX-g1@mail.gmail.com>
	<AANLkTindiPgTH65=0MwBgdviyWByQzmu1=01MYUy_fXy@mail.gmail.com>
	<AANLkTi=cFCJssphqmji_8LTXTg3ZycGfswMM77hmkie9@mail.gmail.com>
	<AANLkTimTKLJQ+cE9r5t2fRR+riDo-DPi+7FQLJecoc9M@mail.gmail.com>
	<AANLkTi=A-qZXz23nBgFz+LgAKCsqOmFFfgs8zm8k8_Re@mail.gmail.com>
	<AANLkTikvOhbJZfpbN9BmLs3eFFuz=uG2jRyKM4J5zYus@mail.gmail.com>
	<AANLkTi=9rwDSgyt5kDj8LhG+CMFAfXGjxxfcpquJxPxH@mail.gmail.com>
	<AANLkTimcqpeHJN+wJ+i+DWg4ii6PYuHwF2wDf5fpyfhz@mail.gmail.com>
	<AANLkTik5r_qHjgSgGk-_v+9VDM-fPZmGC-4fTZzPw=b2@mail.gmail.com>
	<AANLkTinDkJ6OEsgsrNzAuQ7nA8SyBJJ4wxLoS+4odWtr@mail.gmail.com>
	<AANLkTiktR0=MxXE4frkjtG2VLRkoJkV9T6ot+eGUipms@mail.gmail.com>
	<AANLkTikx4++bVqH_FASpsHgy5E2G4QLM9+7LGuhJk7eU@mail.gmail.com>
	<AANLkTimpvsqxuE3DEpbeWoLtPOamtGSevvFnrGF9UBsp@mail.gmail.com>
Message-ID: <AANLkTikVmPPn_yT_yHhvi2PFZ_vmD7D1MYfQqWwHBXD2@mail.gmail.com>

On Mon, Feb 28, 2011 at 4:34 PM, Malay Thakershi <mthakershi at gmail.com>wrote:

> Ok.
> I am using managed esl to send originate.
>
> I will put these params in sofia profiles.
>
> You said I am offering all codecs at once. How do I restrict this? And what
> should be the codec offered in my case?
>

Well, assuming that what the other end sent is useful, the other end only
offered 3 codecs in the SDP coming back:
   m=audio 19624 RTP/AVP 0 3 18
   a=rtpmap:0 PCMU/8000
   a=rtpmap:3 GSM/8000
   a=rtpmap:18 G729/8000


I'm assuming you don't want the calls to sound horrible, so don't use GSM.
If you didn't pay for the G729 commercial licenses then don't use G729. That
leaves good ol' PCMU. You can add this to the beginning of your originate
dialstring:
{absolute_codec_string=PCMU}

See if that helps...

-MC
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From msc at freeswitch.org  Tue Mar  1 05:49:03 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 28 Feb 2011 18:49:03 -0800
Subject: [Freeswitch-users] t38_gateway self - no re-Invite?
In-Reply-To: <AANLkTinsvti9Q9K-fLzB8o1Xd1qther1ABRQwHeV7FpR@mail.gmail.com>
References: <1B19ABD72889C245AE8EEE08AC24103A28C1AFC00E@exmachina.office.kapper.net>
	<4D6381C0.7040408@ewetel.de>
	<1B19ABD72889C245AE8EEE08AC24103A28C1AFC015@exmachina.office.kapper.net>
	<4D63C790.7000306@ewetel.de>
	<1B19ABD72889C245AE8EEE08AC24103A28C4231F3E@exmachina.office.kapper.net>
	<AANLkTi=aVGEgz=j7rOeknPSNgqv-LvDv3HAkyLZckU3h@mail.gmail.com>
	<1B19ABD72889C245AE8EEE08AC24103A28C4231F40@exmachina.office.kapper.net>
	<AANLkTinGaJ35CZ+t13KRqpdnV3g5p=Yo1psPZY2YGmnh@mail.gmail.com>
	<1B19ABD72889C245AE8EEE08AC24103A28C4231F41@exmachina.office.kapper.net>
	<AANLkTinSoNO7Bh5rEVe=_PEOfqjBdLROVm1FB-6OZPML@mail.gmail.com>
	<1B19ABD72889C245AE8EEE08AC24103A28C4231F4F@exmachina.office.kapper.net>
	<AANLkTinsvti9Q9K-fLzB8o1Xd1qther1ABRQwHeV7FpR@mail.gmail.com>
Message-ID: <AANLkTikoUfFr5KSpR=P6FndX8kWdtBbnTRS3WnvsFj-=@mail.gmail.com>

Awesome! I'll add this to the list of things to document...

-MC

On Mon, Feb 28, 2011 at 10:45 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I added a variable to latest GIT
>
> sip_execute_on_image
>
> This is like execute_on_answer but it lets you execute an application
> as soon as you get a T.38 invite.
> This will probably allow you to change over to t38_gateway etc.
>
>
> On Sat, Feb 26, 2011 at 2:56 PM, Clemens Ebentheuer <ce at kapper.net> wrote:
> > Extra Info here:
> >
> > Just to verify that zoiper is not sending the CNG and that it is not a
> frequency issue like Helmut had, I downloaded a tonegenerator and played an
> 1100HZ tone when T38_gateway waits for it (switch_ivr_async.c:2608 Adding
> tone spec 1100.0 index 0 hits 1)
> >
> > T38 FAX was successfully gatewayed to g711.
> >
> > Thx again,
> >
> > clemens
> >
> >> Subject: RE: [Freeswitch-users] t38_gateway self - no re-Invite?
> >>
> >> Ok,
> >>
> >> I think I kick the Zoiper Softphone for testing and get my spa2102 - I
> >> think Zoiper does not send any CNG or CED...
> >>
> >> Thanks a lot - I?ll return with ATA experience,
> >>
> >>
> >> clemens
> >>
> >> >
> >> > like I said I don't know you topology but you neeed to get the
> >> gateway
> >> > to detect the CNG tone going the right way so the app will react to
> >> > that and transfer it to the data bridge where it will be ready to
> >> > handle t.38 or audio.
> >> >
> >> > if you don't see it detecting the tone you are not making it very
> >> far.
> >> >
> >> >
> >> > On Fri, Feb 25, 2011 at 5:54 PM, Clemens Ebentheuer <ce at kapper.net>
> >> > wrote:
> >> > > Yes, tried that too - no luck - both, self and peer
> >> > >
> >> > > When I try to send the fax to an extension with rxfax, all is
> >> working
> >> > fine.
> >> > >
> >> > > Zoiper invites pcma ->
> >> > > freeswitch OK ->
> >> > > Zoiper reinvites t38 ->
> >> > > freeswitch TRYING ->
> >> > > fresswitch executes rxfax - mod_spandsp_fax.c "reads" the t38 sdp->
> >> > > freeswitch sends OK with t38 sdp ->
> >> > >
> >> > > any other ideas?
> >> > >
> >> > >
> >> > >> I dont know what your topology is but did you also try setting the
> >> > app
> >> > >> on the A leg and try both peer and self
> >> > >>
> >> > >> <action application="t38_gateway" data="self"/>
> >> > >>
> >> > >> right before bridge.
> >> > >>
> >> > >>
> >> > >> On Fri, Feb 25, 2011 at 4:36 PM, Clemens Ebentheuer
> >> <ce at kapper.net>
> >> > >> wrote:
> >> > >> > Hi
> >> > >> >
> >> > >> > My way:
> >> > >> >
> >> > >> > internal:t38(zoiper) -> FS -> external:g711 (provider with no
> >> t38
> >> > >> support)
> >> > >> >
> >> > >> > I tried all versions of set and export and peer and self - but
> >> no
> >> > >> luck- for my understanding it should work with set t38_gateway
> >> self
> >> > >> >
> >> > >> > Set because of aleg and self because FS acts as the T38 fax
> >> > >> "receiver".
> >> > >> >
> >> > >> > Am I wrong here?
> >> > >> >
> >> > >> > My dialplan:
> >> > >> > <extension name="0-sil-p000016" >
> >> > >> >   <condition field="destination_number" expression="^0(.*)$">
> >> > >> >       <action application="set" data="fax_enable_t38=true"/>
> >> > >> >       <action application="set"
> >> > data="execute_on_answer=t38_gateway
> >> > >> self"/>
> >> > >> >       <action application="bridge" data="sofia/gateway/sil-
> >> > >> p000016/$1"/>
> >> > >> >   </condition>
> >> > >> > </extension>
> >> > >> > (testet with and without fax_enable_t38_request=true)
> >> > >> >
> >> > >> > Thx,
> >> > >> >
> >> > >> >
> >> > >> > Clemens
> >> > >> >
> >> > >> >
> >> > >> >
> >> > >> >
> >> > >> >> have you tried the self vs peer args to the t38 gateway app,
> >> > maybe
> >> > >> you
> >> > >> >> have it configured backwards.
> >> > >> >>
> >> > >> >>
> >> > >> >> On Fri, Feb 25, 2011 at 4:11 PM, Clemens Ebentheuer
> >> > <ce at kapper.net>
> >> > >> >> wrote:
> >> > >> >> > Hi,
> >> > >> >> >
> >> > >> >> >> It can be, that in your case the device is simply detecting
> >> > the
> >> > >> FAX
> >> > >> >> >> tones itself before FS does. When the device sends a
> >> ReINVITE
> >> > to
> >> > >> FS
> >> > >> >> FS
> >> > >> >> >> has no chance to detect CNG anymore and hence never switch
> >> to
> >> > >> t38.
> >> > >> >> >
> >> > >> >> > Maybe you?re right here - or zoiper
> >> > >> >> (http://www.zoiper.com/softphone/classic/ which is a softphone
> >> > with
> >> > >> >> only t38 fax support [no g711 fallback]) sends the reinvite
> >> every
> >> > >> time
> >> > >> >> after answer when it is sending a fax.
> >> > >> >> >
> >> > >> >> >>
> >> > >> >> >> Just trace the sending FAX-Device for SIP-Signalling and you
> >> > >> should
> >> > >> >> se
> >> > >> >> >> a
> >> > >> >> >> ReINVITE from FAX-device to FS, if this is true.
> >> > >> >> >
> >> > >> >> > Here is a debug with siptrace:
> >> > >> http://pastebin.freeswitch.org/15439
> >> > >> >> >
> >> > >> >> > If I read the logs in a right way:
> >> > >> >> > then FS executes t38_gateway on answer -
> >> > >> >> > sends a 200 OK to zoiper -
> >> > >> >> > zoiper answers with ACK and then reinvites with t38 -
> >> > >> >> > FS answers 100 Trying-
> >> > >> >> >
> >> > >> >> > And nothing is happening. I?m not sure if this is the
> >> scenario
> >> > you
> >> > >> >> describe below, but shouldn?t FS answer with a t38 sdp so
> >> zoiper
> >> > >> knows
> >> > >> >> where to send it?s t38 fax??
> >> > >> >> >
> >> > >> >> >>
> >> > >> >> >> You should check your t38-Device configuration. See if it is
> >> > able
> >> > >> to
> >> > >> >> >> let
> >> > >> >> >> the t38 fax detection job only by FS.
> >> > >> >> >>
> >> > >> >> >> Establish a call from the Fax device to a phone. Can your
> >> hear
> >> > >> the
> >> > >> >> CNG
> >> > >> >> >> signal when you pickup the phone? Does it suddenly stop? If
> >> > so,
> >> > >> the
> >> > >> >> FAX
> >> > >> >> >> device or FS/mod_spandsp has sent a t38-ReINVITE.
> >> > >> >> >>
> >> > >> >> >> Do you see something like "media Bug removed" in FS console
> >> > >> (DEBUG
> >> > >> >> >> level) after 20 seconds of listening to 1100Hz? If so, then
> >> FS
> >> > >> >> failed
> >> > >> >> >> to
> >> > >> >> >> detect CNG, hence it never received it (or for too short to
> >> > >> detect
> >> > >> >> it).
> >> > >> >> >>
> >> > >> >> >> On my Grandstream ATAs this wasn't possible, so I had to
> >> > search
> >> > >> for
> >> > >> >> a
> >> > >> >> >> hack.
> >> > >> >> >>
> >> > >> >> >> Hope this will light up your problems a little bit.
> >> > >> >> >>
> >> > >> >> >>
> >> > >> >> >> > If so, I?m wondering why rxfax in t38 mode is working fine
> >> > then
> >> > >> -
> >> > >> >> >> with re-invite.
> >> > >> >> >> Can't tell you with this, never tried that. sorry.
> >> > >> >> >
> >> > >> >> > Thx,
> >> > >> >> >
> >> > >> >> > clemens
> >> > >> >> >
> >> > >> >> > _______________________________________________
> >> > >> >> > FreeSWITCH-users mailing list
> >> > >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > >> >> >
> >> > >>
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> > >> >> users
> >> > >> >> > 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-users mailing list
> >> > >> >> FreeSWITCH-users at lists.freeswitch.org
> >> > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > >> >>
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> > >> >> users
> >> > >> >> http://www.freeswitch.org
> >> > >> >
> >> > >> > _______________________________________________
> >> > >> > FreeSWITCH-users mailing list
> >> > >> > FreeSWITCH-users at lists.freeswitch.org
> >> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > >> >
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> > >> users
> >> > >> > 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-users mailing list
> >> > >> FreeSWITCH-users at lists.freeswitch.org
> >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > >>
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> > >> users
> >> > >> http://www.freeswitch.org
> >> > >
> >> > > _______________________________________________
> >> > > FreeSWITCH-users mailing list
> >> > > FreeSWITCH-users at lists.freeswitch.org
> >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> > users
> >> > > 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-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> > users
> >> > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sunwood360 at gmail.com  Tue Mar  1 05:58:05 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Mon, 28 Feb 2011 18:58:05 -0800
Subject: [Freeswitch-users] mod_dingaling and google voice
In-Reply-To: <4D6C1978.8040702@cartissolutions.com>
References: <AANLkTin_dFj9ygx3TAmExeN9-pDFNCuZPGZ_Uc2d7o1W@mail.gmail.com>
	<AANLkTimUr3=w_fj_k643ChW93Q_YTm63R_qFyBO9EoQV@mail.gmail.com>
	<4D6C1978.8040702@cartissolutions.com>
Message-ID: <AANLkTikdossGK+FnBfmSkfmEMF4Pv0nF5UYypA1t9bc2@mail.gmail.com>

I don't use GV for direct inbound, because it is not working very well.

- Get an account from sipgate.com;
- let GV forward inbound to your sipgate PSTN #
- and your FS register as a sipgate client.



On Mon, Feb 28, 2011 at 1:54 PM, Yossi Neiman <
freeswitch at cartissolutions.com> wrote:

>  What FS build are you running?  I'd like to compare notes with you on your
> config, as I'm experiencing the RTP port 0 issue with FreeSWITCH Version
> 1.0.head (git-0d8e945 2011-02-16 08-57-43 -0600).  What about inbound?  Do
> you have that set up?
>
> Yossi Neiman
> Cartis Solutions, Inc. - http://www.cartissolutions.com
>
> On 02/28/2011 12:54 AM, envelopes envelopes wrote:
>
> never mind. add this line <action application="ring_ready"/> and restart FS
>
> fixed the issue.
>
>
>
> On Sun, Feb 27, 2011 at 10:30 PM, envelopes envelopes <
> sunwood360 at gmail.com> wrote:
>
>> Now I am able to use GV for outbound dialing. However, I don't hear any
>> ringback or not sure whether the other party has answered the call.
>> is there any config variable to set up so that I will be notified if my
>> call is answered?
>>
>> thanks!
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From thisjoy0528 at gmail.com  Tue Mar  1 06:15:51 2011
From: thisjoy0528 at gmail.com (joy this)
Date: Tue, 1 Mar 2011 11:15:51 +0800
Subject: [Freeswitch-users] problem with voice quality
In-Reply-To: <AANLkTik_zP1oX6Uwrk0MOA32ezXUp57K-DJ82BJdNeon@mail.gmail.com>
References: <AANLkTik15iq+zwivCJWZeVFOmDVvAFB70v7haeK9XE2K@mail.gmail.com>
	<1CC6662BE5B94BD9A2C7DF20A3FDD577@stor1>
	<AANLkTim5US0xk1pYHyomm-_VwA1BQtACJU4p46ahPPV8@mail.gmail.com>
	<AANLkTik_zP1oX6Uwrk0MOA32ezXUp57K-DJ82BJdNeon@mail.gmail.com>
Message-ID: <AANLkTikDK=tRczp5Dpjw55WKWy9dwpRu1hAc+RP9adQn@mail.gmail.com>

> Can you reproduce the sound problems on the very latest Git?

 I downloaded FS at 2/24 last week, but I couldn't see the version. It only
shows "FreeSWITCH Version 1.0.head (git-)" when I input "version".

Can you
> hear speech with noise, or is it just garbage?

I could hear the speech with the luod noise.

Can you get a
> debug-level log for the call with siptrace enabled so we can see
> what's going on? Also use tcpdump/wireshark/tshark to capture the RTP
> and analyze it in Wireshark to check for packet loss or jitter.
>
I have the log file and the package (by Wireshark), should I share them by
the email?
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From mthakershi at gmail.com  Tue Mar  1 09:16:42 2011
From: mthakershi at gmail.com (Malay Thakershi)
Date: Tue, 1 Mar 2011 00:16:42 -0600
Subject: [Freeswitch-users] disabling ptime warning message
In-Reply-To: <AANLkTikVmPPn_yT_yHhvi2PFZ_vmD7D1MYfQqWwHBXD2@mail.gmail.com>
References: <AANLkTim97P5=MJgKq3ncDmbTiXkP63CBoALRD6o+b-Uw@mail.gmail.com>
	<AANLkTikw3TLW8bo39WRX6wN44vhcGSDBBTS_QffqBTwc@mail.gmail.com>
	<AANLkTi=FGWs2bs3EFvaMTTyG0=0inwe5z2UfByctX-g1@mail.gmail.com>
	<AANLkTindiPgTH65=0MwBgdviyWByQzmu1=01MYUy_fXy@mail.gmail.com>
	<AANLkTi=cFCJssphqmji_8LTXTg3ZycGfswMM77hmkie9@mail.gmail.com>
	<AANLkTimTKLJQ+cE9r5t2fRR+riDo-DPi+7FQLJecoc9M@mail.gmail.com>
	<AANLkTi=A-qZXz23nBgFz+LgAKCsqOmFFfgs8zm8k8_Re@mail.gmail.com>
	<AANLkTikvOhbJZfpbN9BmLs3eFFuz=uG2jRyKM4J5zYus@mail.gmail.com>
	<AANLkTi=9rwDSgyt5kDj8LhG+CMFAfXGjxxfcpquJxPxH@mail.gmail.com>
	<AANLkTimcqpeHJN+wJ+i+DWg4ii6PYuHwF2wDf5fpyfhz@mail.gmail.com>
	<AANLkTik5r_qHjgSgGk-_v+9VDM-fPZmGC-4fTZzPw=b2@mail.gmail.com>
	<AANLkTinDkJ6OEsgsrNzAuQ7nA8SyBJJ4wxLoS+4odWtr@mail.gmail.com>
	<AANLkTiktR0=MxXE4frkjtG2VLRkoJkV9T6ot+eGUipms@mail.gmail.com>
	<AANLkTikx4++bVqH_FASpsHgy5E2G4QLM9+7LGuhJk7eU@mail.gmail.com>
	<AANLkTimpvsqxuE3DEpbeWoLtPOamtGSevvFnrGF9UBsp@mail.gmail.com>
	<AANLkTikVmPPn_yT_yHhvi2PFZ_vmD7D1MYfQqWwHBXD2@mail.gmail.com>
Message-ID: <AANLkTi=+opjZn8e1=3tFnoWoKgvDxo58bWYjrwPnu41e@mail.gmail.com>

OK. So did the following and now I do not get that warning anymore. Also
wanted to remove unnecessary codec offerings.

1. Commented out all <!-- Codec Interfaces --> in modules.conf.xml (this
reduced comparisons before final codec was being chosen)

2. Changed codec lines in vars.xml
  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA"/>

3. Added these to sofia.conf.xml
<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
<param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>

Thank you.

Malay



On Mon, Feb 28, 2011 at 8:44 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Feb 28, 2011 at 4:34 PM, Malay Thakershi <mthakershi at gmail.com>wrote:
>
>> Ok.
>> I am using managed esl to send originate.
>>
>> I will put these params in sofia profiles.
>>
>> You said I am offering all codecs at once. How do I restrict this? And
>> what should be the codec offered in my case?
>>
>
> Well, assuming that what the other end sent is useful, the other end only
> offered 3 codecs in the SDP coming back:
>    m=audio 19624 RTP/AVP 0 3 18
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:3 GSM/8000
>    a=rtpmap:18 G729/8000
>
>
> I'm assuming you don't want the calls to sound horrible, so don't use GSM.
> If you didn't pay for the G729 commercial licenses then don't use G729. That
> leaves good ol' PCMU. You can add this to the beginning of your originate
> dialstring:
> {absolute_codec_string=PCMU}
>
> See if that helps...
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Tue Mar  1 09:29:50 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 1 Mar 2011 11:59:50 +0530
Subject: [Freeswitch-users] Sip Info method
Message-ID: <AANLkTin=A5-vEeGHvm+tJ+epspoUvZOOey2YkeVBf1=M@mail.gmail.com>

Hello,

Can we start recording any conference bridge via sip info method  ?

It should happen for only that particular bridge users and not all the
bridges running on the application.

For that, what parameters i should set in the SIP info method to achieve
this.

Regards
Sam
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From max.clark at gmail.com  Tue Mar  1 09:56:13 2011
From: max.clark at gmail.com (Max Clark)
Date: Mon, 28 Feb 2011 22:56:13 -0800
Subject: [Freeswitch-users] Disconnect Call Timer
Message-ID: <AANLkTinwBqhyC52Apv7km4dQVSgtUO82nR5ERki3R_P5@mail.gmail.com>

Hello,

I'd like to automatically disconnect a call after x minutes (say 4
hours). Is this something that can be set in the dialplan before the
bridge? What is the best way to do this?

Thanks,
Max


From avi at avimarcus.net  Tue Mar  1 11:08:23 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 1 Mar 2011 10:08:23 +0200
Subject: [Freeswitch-users] mod_dingaling and google voice
In-Reply-To: <AANLkTikdossGK+FnBfmSkfmEMF4Pv0nF5UYypA1t9bc2@mail.gmail.com>
References: <AANLkTin_dFj9ygx3TAmExeN9-pDFNCuZPGZ_Uc2d7o1W@mail.gmail.com>
	<AANLkTimUr3=w_fj_k643ChW93Q_YTm63R_qFyBO9EoQV@mail.gmail.com>
	<4D6C1978.8040702@cartissolutions.com>
	<AANLkTikdossGK+FnBfmSkfmEMF4Pv0nF5UYypA1t9bc2@mail.gmail.com>
Message-ID: <AANLkTinyqQCqddpATA5sShtWh1QEWZH46-sU6U3mRTwi@mail.gmail.com>

Or get a gizmo5 account and then you can just do SIP forwarding to FS,
without the pstn loop. You can find them around for a fee bucks. I have a
few, not sure if I'm going to use them...
-Avi
On Mar 1, 2011 4:59 AM, "envelopes envelopes" <sunwood360 at gmail.com> wrote:
> I don't use GV for direct inbound, because it is not working very well.
>
> - Get an account from sipgate.com;
> - let GV forward inbound to your sipgate PSTN #
> - and your FS register as a sipgate client.
>
>
>
> On Mon, Feb 28, 2011 at 1:54 PM, Yossi Neiman <
> freeswitch at cartissolutions.com> wrote:
>
>> What FS build are you running? I'd like to compare notes with you on your
>> config, as I'm experiencing the RTP port 0 issue with FreeSWITCH Version
>> 1.0.head (git-0d8e945 2011-02-16 08-57-43 -0600). What about inbound? Do
>> you have that set up?
>>
>> Yossi Neiman
>> Cartis Solutions, Inc. - http://www.cartissolutions.com
>>
>> On 02/28/2011 12:54 AM, envelopes envelopes wrote:
>>
>> never mind. add this line <action application="ring_ready"/> and restart
FS
>>
>> fixed the issue.
>>
>>
>>
>> On Sun, Feb 27, 2011 at 10:30 PM, envelopes envelopes <
>> sunwood360 at gmail.com> wrote:
>>
>>> Now I am able to use GV for outbound dialing. However, I don't hear any
>>> ringback or not sure whether the other party has answered the call.
>>> is there any config variable to set up so that I will be notified if my
>>> call is answered?
>>>
>>> thanks!
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://
lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
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From sunwood360 at gmail.com  Tue Mar  1 11:37:45 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Tue, 1 Mar 2011 00:37:45 -0800
Subject: [Freeswitch-users] weird FS behavior
Message-ID: <AANLkTimDLm0qNWw=hL0pXZ6ic2McDPjwDrdb3tgx9Tfr@mail.gmail.com>

in my FS system, If i only dialed 7 digits, it automatically appended 1347.
347 is not my default area code in vars.xml
is FS crazy...???  in the following example, i only dialed :  4562345, but
it became 13474562345.




011-03-01 03:22:03.509008 [NOTICE] switch_channel.c:669 New Channel
sofia/internal/1000 at 192.168.1.115 [fcd969a4-43dc-11e0-8b62-6f20955d792b]
2011-03-01 03:22:03.518498 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1000 at 192.168.1.115) Running State Change CS_NEW
2011-03-01 03:22:03.518498 [DEBUG] switch_core_state_machine.c:320
(sofia/internal/1000 at 192.168.1.115) State NEW
2011-03-01 03:22:03.518498 [DEBUG] sofia.c:4149 Channel sofia/internal/
1000 at 192.168.1.115 entering state [received][100]
2011-03-01 03:22:03.518498 [DEBUG] sofia.c:4160 Remote SDP:
v=0
o=- 28491453 28491453 IN IP4 192.168.1.112
s=-
c=IN IP4 192.168.1.112
t=0 0
m=audio 16466 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2011-03-01 03:22:03.524230 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
[PCMU:0:8000:20]/[G722:9:8000:20]
2011-03-01 03:22:03.525384 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
[PCMU:0:8000:20]/[PCMU:0:8000:20]
2011-03-01 03:22:03.527548 [DEBUG] sofia_glue.c:2354 Set Codec
sofia/internal/1000 at 192.168.1.115 PCMU/8000 20 ms 160 samples
2011-03-01 03:22:03.534389 [DEBUG] sofia_glue.c:3524 Set 2833 dtmf send/recv
payload to 101
2011-03-01 03:22:03.534389 [DEBUG] sofia.c:4306 (sofia/internal/
1000 at 192.168.1.115) State Change CS_NEW -> CS_INIT
2011-03-01 03:22:03.536796 [DEBUG] switch_core_session.c:1021 Send signal
sofia/internal/1000 at 192.168.1.115 [BREAK]
2011-03-01 03:22:03.539167 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1000 at 192.168.1.115) Running State Change CS_INIT
2011-03-01 03:22:03.540372 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/1000 at 192.168.1.115) State INIT
2011-03-01 03:22:03.540372 [DEBUG] mod_sofia.c:83 sofia/internal/
1000 at 192.168.1.115 SOFIA INIT
2011-03-01 03:22:03.540372 [DEBUG] mod_sofia.c:117 (sofia/internal/
1000 at 192.168.1.115) State Change CS_INIT -> CS_ROUTING
2011-03-01 03:22:03.540372 [DEBUG] switch_core_session.c:1021 Send signal
sofia/internal/1000 at 192.168.1.115 [BREAK]
2011-03-01 03:22:03.540372 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/1000 at 192.168.1.115) State INIT going to sleep
2011-03-01 03:22:03.540372 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1000 at 192.168.1.115) Running State Change CS_ROUTING
2011-03-01 03:22:03.540372 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/1000 at 192.168.1.115) State ROUTING
2011-03-01 03:22:03.540372 [DEBUG] mod_sofia.c:140 sofia/internal/
1000 at 192.168.1.115 SOFIA ROUTING
2011-03-01 03:22:03.540372 [DEBUG] switch_core_state_machine.c:77
sofia/internal/1000 at 192.168.1.115 Standard ROUTING
2011-03-01 03:22:03.540372 [INFO] mod_dialplan_xml.c:418 Processing
1000->13474562345 in context default



I did a grep :
/etc/freeswitch# find . -name "*.*" | xargs grep 347
./jingle_profiles/server.xml:    <param name="server" value="
jabber.server.org:5347"/>


Where is 347 coming from? is it possible of buffer overflow? i am using FS
1.0.6.
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From peter.olsson at visionutveckling.se  Tue Mar  1 11:48:54 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 1 Mar 2011 09:48:54 +0100
Subject: [Freeswitch-users] weird FS behavior
In-Reply-To: <AANLkTimDLm0qNWw=hL0pXZ6ic2McDPjwDrdb3tgx9Tfr@mail.gmail.com>
References: <AANLkTimDLm0qNWw=hL0pXZ6ic2McDPjwDrdb3tgx9Tfr@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2BBA942@cooper>

Are you sure that your phone is not prepending this?

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r envelopes envelopes
Skickat: den 1 mars 2011 09:38
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] weird FS behavior

in my FS system, If i only dialed 7 digits, it automatically appended 1347.  347 is not my default area code in vars.xml
is FS crazy...???  in the following example, i only dialed :  4562345, but it became 13474562345.




011-03-01 03:22:03.509008 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1000 at 192.168.1.115<mailto:1000 at 192.168.1.115> [fcd969a4-43dc-11e0-8b62-6f20955d792b]
2011-03-01 03:22:03.518498 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.1.115<mailto:1000 at 192.168.1.115>) Running State Change CS_NEW
2011-03-01 03:22:03.518498 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1000 at 192.168.1.115<mailto:1000 at 192.168.1.115>) State NEW
2011-03-01 03:22:03.518498 [DEBUG] sofia.c:4149 Channel sofia/internal/1000 at 192.168.1.115<mailto:1000 at 192.168.1.115> entering state [received][100]
2011-03-01 03:22:03.518498 [DEBUG] sofia.c:4160 Remote SDP:
v=0
o=- 28491453 28491453 IN IP4 192.168.1.112
s=-
c=IN IP4 192.168.1.112
t=0 0
m=audio 16466 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2011-03-01 03:22:03.524230 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [PCMU:0:8000:20]/[G722:9:8000:20]
2011-03-01 03:22:03.525384 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20]
2011-03-01 03:22:03.527548 [DEBUG] sofia_glue.c:2354 Set Codec sofia/internal/1000 at 192.168.1.115<mailto:1000 at 192.168.1.115> PCMU/8000 20 ms 160 samples
2011-03-01 03:22:03.534389 [DEBUG] sofia_glue.c:3524 Set 2833 dtmf send/recv payload to 101
2011-03-01 03:22:03.534389 [DEBUG] sofia.c:4306 (sofia/internal/1000 at 192.168.1.115<mailto:1000 at 192.168.1.115>) State Change CS_NEW -> CS_INIT
2011-03-01 03:22:03.536796 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/1000 at 192.168.1.115<mailto:1000 at 192.168.1.115> [BREAK]
2011-03-01 03:22:03.539167 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.1.115<mailto:1000 at 192.168.1.115>) Running State Change CS_INIT
2011-03-01 03:22:03.540372 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1000 at 192.168.1.115<mailto:1000 at 192.168.1.115>) State INIT
2011-03-01 03:22:03.540372 [DEBUG] mod_sofia.c:83 sofia/internal/1000 at 192.168.1.115<mailto:1000 at 192.168.1.115> SOFIA INIT
2011-03-01 03:22:03.540372 [DEBUG] mod_sofia.c:117 (sofia/internal/1000 at 192.168.1.115<mailto:1000 at 192.168.1.115>) State Change CS_INIT -> CS_ROUTING
2011-03-01 03:22:03.540372 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/1000 at 192.168.1.115<mailto:1000 at 192.168.1.115> [BREAK]
2011-03-01 03:22:03.540372 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1000 at 192.168.1.115<mailto:1000 at 192.168.1.115>) State INIT going to sleep
2011-03-01 03:22:03.540372 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.1.115<mailto:1000 at 192.168.1.115>) Running State Change CS_ROUTING
2011-03-01 03:22:03.540372 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1000 at 192.168.1.115<mailto:1000 at 192.168.1.115>) State ROUTING
2011-03-01 03:22:03.540372 [DEBUG] mod_sofia.c:140 sofia/internal/1000 at 192.168.1.115<mailto:1000 at 192.168.1.115> SOFIA ROUTING
2011-03-01 03:22:03.540372 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1000 at 192.168.1.115<mailto:1000 at 192.168.1.115> Standard ROUTING
2011-03-01 03:22:03.540372 [INFO] mod_dialplan_xml.c:418 Processing 1000->13474562345 in context default



I did a grep :
/etc/freeswitch# find . -name "*.*" | xargs grep 347
./jingle_profiles/server.xml:    <param name="server" value="jabber.server.org:5347<http://jabber.server.org:5347>"/>


Where is 347 coming from? is it possible of buffer overflow? i am using FS 1.0.6.







!DSPAM:4d6cb1ed32761091320802!
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From steveayre at gmail.com  Tue Mar  1 12:08:27 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 1 Mar 2011 09:08:27 +0000
Subject: [Freeswitch-users] HANGUP_CAUSE after bridge in outbound ESL
In-Reply-To: <AANLkTimRzV3M4vUBQpKwyVhjBh7LxEiF=y3jJ_P+rCRK@mail.gmail.com>
References: <AANLkTimRzV3M4vUBQpKwyVhjBh7LxEiF=y3jJ_P+rCRK@mail.gmail.com>
Message-ID: <AANLkTi=BXYL3oCe9yUQNp1JyAKdbkSXURps+hqkSVT8y@mail.gmail.com>

The correct way is to use events. You can register for just the hangup event.

-Steve


On 28 February 2011 22:43, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> Hi All
>
> What is correct method to get bridge status after execute("bridge") in ESL?
> I have inbound call that gets bridged to SIP endpoint. I need to know
> whether it was ORIGINATOR_CANCEL or BUSY or something else, but if I use
> sync, seems I can't determine ORIGINATOR_CANCEL status, because it is set on
> a-leg which is destroyed first after hangup.
>
> Is there a way to get bridge status without messing with events and event
> source filtering by channel uuid or event type?
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From codecomplete at free.fr  Tue Mar  1 12:10:47 2011
From: codecomplete at free.fr (GillesToo)
Date: Tue, 1 Mar 2011 01:10:47 -0800 (PST)
Subject: [Freeswitch-users] Does FS handle three-way calling on the same
	POTS?
Message-ID: <1298970647476-6076537.post@n2.nabble.com>

Hello

My ISP offers basic PBX features and free calls through when plugging a
phone in an RJ11 on the ADSL modem. I'd like to hook up a PC to the modem
with an FXO module, and use three-way calling so that the server calls me
back on my cellphone, and then lets me call a number through the server
before switching to a conference call. Poor man's callback system :-)

I finally got Asterisk to make the first call, and put it on hold, but it
stops there: It appears that Asterisk is unable to create a second channel
on the same FXO module when a call has been put on hold ("app_dial.c:1310
dial_exec_full:Unable to create channel of type 'Dahdi' (cause 0 -
Unknown)").

I was wondering if someone had successfully used Freeswitch in this
scenario?

FWIW, here's my extensions.conf:
==============
[from_fxo]
exten => s,1,Wait(2)
exten => s,n,Set(GLOBAL(CID)=${CALLERID(num)})
exten => s,n,Hangup()

;script waits and creates callfile
exten => h,1,system(/var/tmp/test10.lua ${CID}&)

;context used by callfile
[callback]
exten => start,1,NoOp(In callback, CID is ${CID})
;how to wait until remote phone picked up?
exten => start,n,Wait(5)
exten => start,n,Answer()
exten => start,n,Playback(tt-monkeysintro)

exten => start,n,Flash()

;BAD exten => start,n,Dial(Dahdi/1/${GSM})
;does ring second number, but no actual voice channel
exten => start,n,SendDTMF(456789)
==============

Thank you.

-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Does-FS-handle-three-way-calling-on-the-same-POTS-tp6076537p6076537.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Tue Mar  1 12:16:45 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 1 Mar 2011 09:16:45 +0000
Subject: [Freeswitch-users] fs_cli socket connection error
In-Reply-To: <AANLkTimNYjC_G-Lq+St5VePpxEyM8iHnhaNCo=jBydi9@mail.gmail.com>
References: <AANLkTikCoT9dzLNoD=6TzS0jiMw4afNX94TypsLarBbz@mail.gmail.com>
	<AANLkTi=sENgBB9Cdr-EkLJD8XfmH3A8GSfPnvDHO7qrw@mail.gmail.com>
	<AANLkTin5k=2Z-=QVVzqt91172wv88P7OFkqRqTY9VDcS@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58ADB727D7@cooper>
	<AANLkTi=aPwmnVD8ZzaBVFDxGyPsbgzDEBGaXvq_B2NbF@mail.gmail.com>
	<AANLkTikpQuxvyaU7PAGnyJuoQLQ9xn4inNJ_u5x41Anv@mail.gmail.com>
	<AANLkTimNYjC_G-Lq+St5VePpxEyM8iHnhaNCo=jBydi9@mail.gmail.com>
Message-ID: <AANLkTi=QXgEb6dHU9bfb7v06WzPyWJcoNYDLi1X9oq+E@mail.gmail.com>

Ok, you really need to pay attention to any log message that logs with
the CRIT (critical) log level - they're really serious errors. You
should look at adding the critical="true" attribute to all the tags in
modules.conf.xml which you want to be sure are loaded. If a module
fails to load without that tag FS will continue to run anyway, with
that tag it'll refuse to start so you'll know instantly something is
wrong.

Most of your modules aren't loading, because they haven't been
compiled correctly. Some look like they're missing (no such file or
directory) and others haven't been built right (undefined symbols).

Here are a few of the important lines from your log:

2011-02-28 18:12:21.910738 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_event_socket.so
**/opt/freeswitch/mod/mod_event_socket.so: undefined symbol:
switch_event_serialize_json**
2011-02-28 18:12:21.910789 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_xml_curl.so
**/opt/freeswitch/mod/mod_xml_curl.so: cannot open shared object file:
No such file or directory**
2011-02-28 18:12:21.910813 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_xml_cdr.so
**/opt/freeswitch/mod/mod_xml_cdr.so: cannot open shared object file:
No such file or directory**
2011-02-28 18:12:21.911581 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_commands.so
**/opt/freeswitch/mod/mod_commands.so: undefined symbol: switch_xml_reload**
2011-02-28 18:12:21.911709 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_conference.so
**/opt/freeswitch/mod/mod_conference.so: undefined symbol:
switch_channel_test_app_flag_key**
2011-02-28 18:12:21.911842 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_dptools.so
**/opt/freeswitch/mod/mod_dptools.so: undefined symbol: switch_event_merge**
2011-02-28 18:12:21.912264 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_dialplan_xml.so
**/opt/freeswitch/mod/mod_dialplan_xml.so: undefined symbol:
switch_xml_std_datetime_check**
2011-02-28 18:12:21.912531 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_g723_1.so
**/opt/freeswitch/mod/mod_g723_1.so: cannot open shared object file:
No such file or directory**
2011-02-28 18:12:21.912573 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_g729.so
**/opt/freeswitch/mod/mod_g729.so: cannot open shared object file: No
such file or directory**
2011-02-28 18:12:21.912609 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_amr.so
**/opt/freeswitch/mod/mod_amr.so: cannot open shared object file: No
such file or directory**

Important things to note:
mod_event_socket hasn't loaded because of a undefined symbol, it
wasn't built correctly
mod_xml_curl hasn't loaded because it doesn't exist, it was either
never built or never installed

Something pretty strange has happened. Did you recompile when trying
to add mod_xml_curl?

I'd suggest you delete all the FS files, including the Git clone, make
a fresh checkout and build it from scratch.

-Steve



On 28 February 2011 23:18, deniro <ayhkor at gmail.com> wrote:
> I put the log in pastebin
> freeswitch.log only?when starting freeswitch (after stop)
>
> http://pastebin.freeswitch.org/15502
> thx
> deniro
> On Mon, Feb 28, 2011 at 5:46 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> You must check the logfile for when FS starts up - if netstat shows
>> nothing for port 8021 then either mod_event_socket isn't being loaded
>> or you'll see an error when it tries to load.
>>
>> Nothing from netstat means nothing's listening, so trying to connect
>> using fs_cli won't do anything.
>>
>> -Steve
>>
>>
>> On 28 February 2011 22:40, deniro <ayhkor at gmail.com> wrote:
>> > Checked the log
>> > Nothing is logged when
>> > fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
>> > is issued
>> > /opt/freeswitch/log# tail -f freeswitch.log
>> >
>> > ls -l freeswitch.log
>> > -rwxrwxrwx 1 freeswitch daemon 33018 2011-02-28 17:39 freeswitch.log
>> > thx
>> >
>> >
>> >
>> > On Mon, Feb 28, 2011 at 3:59 PM, Peter Olsson
>> > <peter.olsson at visionutveckling.se> wrote:
>> >>
>> >> Check freeswitch.log, it probably reports some problem when loading the
>> >> mod_event_socket module.
>> >>
>> >> /Peter
>> >> ________________________________________
>> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> >> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r deniro
>> >> [ayhkor at gmail.com]
>> >> Skickat: den 28 februari 2011 21:42
>> >> Till: FreeSWITCH Users Help
>> >> ?mne: Re: [Freeswitch-users] fs_cli socket connection error
>> >>
>> >> Steve
>> >> Thanks for the reply
>> >> netstat -anlp | grep 8021 ?is blank (nothing showing up)
>> >>
>> >> /opt/freeswitch/bin/fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
>> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
>> >> Configuration
>> >> file is /root/.fs_cli_conf.
>> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
>> >> Configuration
>> >> file is /etc/fs_cli.conf.
>> >> [DEBUG] libs/esl/fs_cli.c:1159 main() profile default does not exist
>> >> using
>> >> builtin profile
>> >> [DEBUG] libs/esl/fs_cli.c:1188 main() Using profile internal
>> >> [75.xxx.xxx.xxx]
>> >> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
>> >> Connection
>> >> Error]
>> >> yes mod_event_socket is after ?mod_xml_curl but I changed the order in
>> >> ?modules.conf.xml
>> >> still getting above (restarted freeswitch)
>> >>
>> >> thx
>> >> deniro--
>> >>
>> >>
>> >>
>> >> On Mon, Feb 28, 2011 at 3:35 AM, Steven Ayre
>> >> <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
>> >> Enable debug logging and you should see an error that'll tell you more.
>> >>
>> >> Is mod_event_socket loading before or after mod_xml_curl? Chances are
>> >> mod_xml_curl is loading first, so it's trying to read
>> >> event_socket.conf.xml and/or acl.conf.xml through xml_curl and either
>> >> getting a different config to your previous local copy or the ACLs are
>> >> different.
>> >>
>> >> What does "netstat -anlp | grep :8021" from the linux cli show you?
>> >> Does it show that freeswitch is actually listening on the port? If it
>> >> is it's probably an ACL problem, if it isn't then it's probably a
>> >> problem with event_socket.conf.xml
>> >>
>> >> -Steve
>> >>
>> >> On 28 February 2011 00:53, deniro
>> >> <ayhkor at gmail.com<mailto:ayhkor at gmail.com>> wrote:
>> >> > What would be possible reasons for this and how to resolve?
>> >> > running fs 106 on ubuntu 10.04 server
>> >> >
>> >> > /opt/freeswitch/bin/fs_cli -H xxx.xxx.xxx.xxx -P 8021
>> >> > [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
>> >> > Connection
>> >> > Error]
>> >> > was working fine before I installed ?mod_xml_cdr
>> >> > configure --prefix=/opt/freeswitch --without-libcurl
>> >> > make mod_xml_cdr-install
>> >> > (no errors)
>> >> >
>> >> > in modules.conf.xml
>> >> > <load module="mod_event_socket"/>
>> >> >
>> >> > is there a packaged version of mod_xml_cdr (for fs 1.0.6)?
>> >> >
>> >> > thx
>> >> > deniro--
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> >
>> >> >
>> >> > FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org<http://www.freeswitch.org/>
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >>
>> >>
>> >> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org<http://www.freeswitch.org/>
>> >>
>> >> !DSPAM:4d6c0a1132761029518849!
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steveayre at gmail.com  Tue Mar  1 12:19:00 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 1 Mar 2011 09:19:00 +0000
Subject: [Freeswitch-users] weird FS behavior
In-Reply-To: <AANLkTimDLm0qNWw=hL0pXZ6ic2McDPjwDrdb3tgx9Tfr@mail.gmail.com>
References: <AANLkTimDLm0qNWw=hL0pXZ6ic2McDPjwDrdb3tgx9Tfr@mail.gmail.com>
Message-ID: <AANLkTi=oACaxSnLVGps1z2+F9FnoK2P1MQbXrBr9bYHN@mail.gmail.com>

>From the cli do "sofia global siptrace on"

You'll see the raw SIP packets being sent from your phone. I think
you'll find your phone's sending that prefix even if you don't expect
it to.

It could be something like it thinks your dialing a local number and
it's trying to add the local area code. That's just a guess though.

-Steve


On 1 March 2011 08:37, envelopes envelopes <sunwood360 at gmail.com> wrote:
> in my FS system, If i only dialed 7 digits, it automatically appended 1347.
> 347 is not my default area code in vars.xml
> is FS crazy...???? in the following example, i only dialed :? 4562345, but
> it became 13474562345.
>
>
>
>
> 011-03-01 03:22:03.509008 [NOTICE] switch_channel.c:669 New Channel
> sofia/internal/1000 at 192.168.1.115 [fcd969a4-43dc-11e0-8b62-6f20955d792b]
> 2011-03-01 03:22:03.518498 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/1000 at 192.168.1.115) Running State Change CS_NEW
> 2011-03-01 03:22:03.518498 [DEBUG] switch_core_state_machine.c:320
> (sofia/internal/1000 at 192.168.1.115) State NEW
> 2011-03-01 03:22:03.518498 [DEBUG] sofia.c:4149 Channel
> sofia/internal/1000 at 192.168.1.115 entering state [received][100]
> 2011-03-01 03:22:03.518498 [DEBUG] sofia.c:4160 Remote SDP:
> v=0
> o=- 28491453 28491453 IN IP4 192.168.1.112
> s=-
> c=IN IP4 192.168.1.112
> t=0 0
> m=audio 16466 RTP/AVP 0 2 4 8 18 96 97 98 100 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:2 G726-32/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729a/8000
> a=rtpmap:96 G726-40/8000
> a=rtpmap:97 G726-24/8000
> a=rtpmap:98 G726-16/8000
> a=rtpmap:100 NSE/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
>
> 2011-03-01 03:22:03.524230 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
> [PCMU:0:8000:20]/[G722:9:8000:20]
> 2011-03-01 03:22:03.525384 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
> [PCMU:0:8000:20]/[PCMU:0:8000:20]
> 2011-03-01 03:22:03.527548 [DEBUG] sofia_glue.c:2354 Set Codec
> sofia/internal/1000 at 192.168.1.115 PCMU/8000 20 ms 160 samples
> 2011-03-01 03:22:03.534389 [DEBUG] sofia_glue.c:3524 Set 2833 dtmf send/recv
> payload to 101
> 2011-03-01 03:22:03.534389 [DEBUG] sofia.c:4306
> (sofia/internal/1000 at 192.168.1.115) State Change CS_NEW -> CS_INIT
> 2011-03-01 03:22:03.536796 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1000 at 192.168.1.115 [BREAK]
> 2011-03-01 03:22:03.539167 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/1000 at 192.168.1.115) Running State Change CS_INIT
> 2011-03-01 03:22:03.540372 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/1000 at 192.168.1.115) State INIT
> 2011-03-01 03:22:03.540372 [DEBUG] mod_sofia.c:83
> sofia/internal/1000 at 192.168.1.115 SOFIA INIT
> 2011-03-01 03:22:03.540372 [DEBUG] mod_sofia.c:117
> (sofia/internal/1000 at 192.168.1.115) State Change CS_INIT -> CS_ROUTING
> 2011-03-01 03:22:03.540372 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1000 at 192.168.1.115 [BREAK]
> 2011-03-01 03:22:03.540372 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/1000 at 192.168.1.115) State INIT going to sleep
> 2011-03-01 03:22:03.540372 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/1000 at 192.168.1.115) Running State Change CS_ROUTING
> 2011-03-01 03:22:03.540372 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/1000 at 192.168.1.115) State ROUTING
> 2011-03-01 03:22:03.540372 [DEBUG] mod_sofia.c:140
> sofia/internal/1000 at 192.168.1.115 SOFIA ROUTING
> 2011-03-01 03:22:03.540372 [DEBUG] switch_core_state_machine.c:77
> sofia/internal/1000 at 192.168.1.115 Standard ROUTING
> 2011-03-01 03:22:03.540372 [INFO] mod_dialplan_xml.c:418 Processing
> 1000->13474562345 in context default
>
>
>
> I did a grep :
> /etc/freeswitch# find . -name "*.*" | xargs grep 347
> ./jingle_profiles/server.xml:??? <param name="server"
> value="jabber.server.org:5347"/>
>
>
> Where is 347 coming from? is it possible of buffer overflow? i am using FS
> 1.0.6.
>
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steveayre at gmail.com  Tue Mar  1 12:20:42 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 1 Mar 2011 09:20:42 +0000
Subject: [Freeswitch-users] Disconnect Call Timer
In-Reply-To: <AANLkTinwBqhyC52Apv7km4dQVSgtUO82nR5ERki3R_P5@mail.gmail.com>
References: <AANLkTinwBqhyC52Apv7km4dQVSgtUO82nR5ERki3R_P5@mail.gmail.com>
Message-ID: <AANLkTin9zh3BgA95feQbWertE_WV_XGGOHLC-D1OeN_h@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_hangup

What's this for? A prepaid account? If so mod_nibblebill could be
better (it can handle when the user is making 2 calls at the same time
so they can't get more minutes than they're meant to that way).

-Steve

On 1 March 2011 06:56, Max Clark <max.clark at gmail.com> wrote:
> Hello,
>
> I'd like to automatically disconnect a call after x minutes (say 4
> hours). Is this something that can be set in the dialplan before the
> bridge? What is the best way to do this?
>
> Thanks,
> Max
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From bggoutham at gmail.com  Tue Mar  1 12:24:37 2011
From: bggoutham at gmail.com (Goutham BG)
Date: Tue, 1 Mar 2011 14:54:37 +0530
Subject: [Freeswitch-users] Global configuration to enable SRTP in the
	entire FS system
Message-ID: <AANLkTimuELApghD2yqXiJnrA9spZ0Bgy7rBSZs5aVi2W@mail.gmail.com>

Hi All,

We can enable SRTP for a particular channel/call by setting the variable
sip_secure_media in the XML dialplan or in the dial string. Is there any global
configuration to enable and enforce SRTP which would apply to all the
extensions in the dialplan and all the channels that are created.

Thanks
Goutham B G
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From tjardick at vanderkraan.net  Tue Mar  1 12:37:38 2011
From: tjardick at vanderkraan.net (Tjardick van der Kraan)
Date: Tue, 1 Mar 2011 10:37:38 +0100
Subject: [Freeswitch-users] Sip Info method
In-Reply-To: <AANLkTin=A5-vEeGHvm+tJ+epspoUvZOOey2YkeVBf1=M@mail.gmail.com>
References: <AANLkTin=A5-vEeGHvm+tJ+epspoUvZOOey2YkeVBf1=M@mail.gmail.com>
Message-ID: <AANLkTinkgcN8q7CqwzEgc5URkw9YtBrxudcTQT+Ef6nP@mail.gmail.com>

Hi Sam,

Did you check the wiki:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session#Activating_via_SIP_INFO

Has some info on this.

Regards,

Tjardick

On Tue, Mar 1, 2011 at 7:29 AM, Sam <u2nsam at gmail.com> wrote:

> Hello,
>
> Can we start recording any conference bridge via sip info method  ?
>
> It should happen for only that particular bridge users and not all the
> bridges running on the application.
>
> For that, what parameters i should set in the SIP info method to achieve
> this.
>
> Regards
> Sam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From simpot at gmail.com  Tue Mar  1 11:23:56 2011
From: simpot at gmail.com (Dmitry Saratsky)
Date: Tue, 01 Mar 2011 10:23:56 +0200
Subject: [Freeswitch-users]  How to run linux shell command from within
Message-ID: <1B58DD8DF45943A4BB4A906B4FE9DD48@wanii.local>

Thanks Christopher,


I already have found why this was not working for me (it was quotes wrong 
usage).

However I already have finished with my script with LUA?s (not FS?s) 
function: os.execute(execute_command); that also runs shell commands, but 
dose it directly from within LUA, but not through FreeSWITCH?s API.


Do you see any problems with that?


Thanks,

Dmitry.




From u2nsam at gmail.com  Tue Mar  1 13:32:04 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 1 Mar 2011 16:02:04 +0530
Subject: [Freeswitch-users] Sip Info method
In-Reply-To: <AANLkTinkgcN8q7CqwzEgc5URkw9YtBrxudcTQT+Ef6nP@mail.gmail.com>
References: <AANLkTin=A5-vEeGHvm+tJ+epspoUvZOOey2YkeVBf1=M@mail.gmail.com>
	<AANLkTinkgcN8q7CqwzEgc5URkw9YtBrxudcTQT+Ef6nP@mail.gmail.com>
Message-ID: <AANLkTikxPFHpMFzHCdvUHaiW7Xiog-o523apLFJ1Urqb@mail.gmail.com>

Yes ,

But it says that 'So as of now there seems to be no way to restrict
activating recording to one party.'

Regds
Sam



On Tue, Mar 1, 2011 at 3:07 PM, Tjardick van der Kraan <
tjardick at vanderkraan.net> wrote:

> Hi Sam,
>
> Did you check the wiki:
>
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session#Activating_via_SIP_INFO
>
> Has some info on this.
>
> Regards,
>
> Tjardick
>
> On Tue, Mar 1, 2011 at 7:29 AM, Sam <u2nsam at gmail.com> wrote:
>
>> Hello,
>>
>> Can we start recording any conference bridge via sip info method  ?
>>
>> It should happen for only that particular bridge users and not all the
>> bridges running on the application.
>>
>> For that, what parameters i should set in the SIP info method to achieve
>> this.
>>
>> Regards
>> Sam
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Tue Mar  1 15:23:54 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 1 Mar 2011 06:23:54 -0600
Subject: [Freeswitch-users] How to run linux shell command from within
In-Reply-To: <1B58DD8DF45943A4BB4A906B4FE9DD48@wanii.local>
References: <1B58DD8DF45943A4BB4A906B4FE9DD48@wanii.local>
Message-ID: <FCF92AF1-525A-42BE-9633-6B898762CEAF@freeswitch.org>

$(system()}

or

api = freeswitch.API();
something = api:executeString("system ls /");


On Mar 1, 2011, at 2:23 AM, Dmitry Saratsky wrote:

> Thanks Christopher,
> 
> 
> I already have found why this was not working for me (it was quotes wrong 
> usage).
> 
> However I already have finished with my script with LUA?s (not FS?s) 
> function: os.execute(execute_command); that also runs shell commands, but 
> dose it directly from within LUA, but not through FreeSWITCH?s API.
> 
> 
> Do you see any problems with that?
> 
> 
> Thanks,
> 
> Dmitry.



From brian at freeswitch.org  Tue Mar  1 15:24:31 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 1 Mar 2011 06:24:31 -0600
Subject: [Freeswitch-users] Global configuration to enable SRTP in the
	entire FS system
In-Reply-To: <AANLkTimuELApghD2yqXiJnrA9spZ0Bgy7rBSZs5aVi2W@mail.gmail.com>
References: <AANLkTimuELApghD2yqXiJnrA9spZ0Bgy7rBSZs5aVi2W@mail.gmail.com>
Message-ID: <48A2CED0-18E7-4183-BB6E-91B4CAAB613C@freeswitch.org>

<X-PRE-PROCESS cmd="set" data="sip_secure_media=true"/> in vars.xml or anywhere in your xml for that matter... or global_setvar sip_secure_media=true

/b


On Mar 1, 2011, at 3:24 AM, Goutham BG wrote:

> Hi All,
> 
> We can enable SRTP for a particular channel/call by setting the variable sip_secure_media in the XML dialplan or in the dial string. Is there any global configuration to enable and enforce SRTP which would apply to all the extensions in the dialplan and all the channels that are created.
> 
> Thanks
> Goutham B G
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Tue Mar  1 15:25:51 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 1 Mar 2011 06:25:51 -0600
Subject: [Freeswitch-users] Polycom G722 negotiation
In-Reply-To: <1298907264976-6073705.post@n2.nabble.com>
References: <AE216221-5892-4337-8985-1889410A866F@5ninesolutions.com>
	<2867CE13-6EED-4FAF-9DF1-49451D164721@freeswitch.org>
	<0ABF546B-4000-4936-AD22-ED6B956277D5@5ninesolutions.com>
	<1298907264976-6073705.post@n2.nabble.com>
Message-ID: <D4D47A0A-4084-4E8D-AF5A-76832ED1FFFA@freeswitch.org>

BZZZZT WRONG.  We support multiples of 20 from 20 to 120 for most codecs.

/b

On Feb 28, 2011, at 9:34 AM, peely wrote:

> What ptime is your device offering? FreeSWITCH only supports 20ms G722, I get
> clipped audio for example when G722 30ms is offered on Snoms, but they work
> perfectly when I force them to 20ms.



From steveayre at gmail.com  Tue Mar  1 15:26:31 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 1 Mar 2011 12:26:31 +0000
Subject: [Freeswitch-users] Sip Info method
In-Reply-To: <AANLkTikxPFHpMFzHCdvUHaiW7Xiog-o523apLFJ1Urqb@mail.gmail.com>
References: <AANLkTin=A5-vEeGHvm+tJ+epspoUvZOOey2YkeVBf1=M@mail.gmail.com>
	<AANLkTinkgcN8q7CqwzEgc5URkw9YtBrxudcTQT+Ef6nP@mail.gmail.com>
	<AANLkTikxPFHpMFzHCdvUHaiW7Xiog-o523apLFJ1Urqb@mail.gmail.com>
Message-ID: <AANLkTikQACVKZ30QY49NTESdRUtr6DD0+0D8Pa7bDKvG@mail.gmail.com>

The conference app mixes audio from all parties, so on your session
you can only get the combined audio. If you can trigger the recording
on the other party's leg instead perhaps that'll work because it won't
have been mixed in yet. Not sure how to do that though, or if you can.

-Steve


On 1 March 2011 10:32, Sam <u2nsam at gmail.com> wrote:
> Yes ,
>
> But it says that 'So as of now there seems to be no way to restrict
> activating recording to one party.'
>
> Regds
> Sam
>
>
>
> On Tue, Mar 1, 2011 at 3:07 PM, Tjardick van der Kraan
> <tjardick at vanderkraan.net> wrote:
>>
>> Hi Sam,
>> Did you check the wiki:
>>
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session#Activating_via_SIP_INFO
>> Has some info on this.
>> Regards,
>> Tjardick
>>
>> On Tue, Mar 1, 2011 at 7:29 AM, Sam <u2nsam at gmail.com> wrote:
>>>
>>> Hello,
>>>
>>> Can we start recording any conference bridge via sip info method? ?
>>>
>>> It should happen for only that particular bridge users and not all the
>>> bridges running on the application.
>>>
>>> For that, what parameters i should set in the SIP info method to achieve
>>> this.
>>>
>>> Regards
>>> Sam
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brian at freeswitch.org  Tue Mar  1 15:28:16 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 1 Mar 2011 06:28:16 -0600
Subject: [Freeswitch-users] Polycom G722 negotiation
In-Reply-To: <C72FE323-9B29-43F3-A17E-A327DAD3B07B@5ninesolutions.com>
References: <AE216221-5892-4337-8985-1889410A866F@5ninesolutions.com>
	<2867CE13-6EED-4FAF-9DF1-49451D164721@freeswitch.org>
	<0ABF546B-4000-4936-AD22-ED6B956277D5@5ninesolutions.com>
	<1298907264976-6073705.post@n2.nabble.com>
	<1F4FD420-3CE9-4601-9D44-80C8BEEF9707@5ninesolutions.com>
	<AANLkTikFCLzNvqpHEO4qtCyFDQOTZ0CG5pOw+SJD=2Yz@mail.gmail.com>
	<5AB8CE7C-5283-4585-8DF8-856A3822F3BD@5ninesolutions.com>
	<AANLkTik0jy=VDy2p4XnAiAwJkT6jX=PAY1W6FgRV7cDC@mail.gmail.com>
	<C72FE323-9B29-43F3-A17E-A327DAD3B07B@5ninesolutions.com>
Message-ID: <378E398F-FB0F-40D9-8DE4-F1026E77DF40@freeswitch.org>

I'm running 3.3.1 and I'm not seeing this behavior... did you happen to not install the default sip.cfg that came with 3.3.1?

/b

On Feb 28, 2011, at 3:37 PM, Spencer Thomason wrote:

> Yes, that what I thought.  I'm running the latest firmware, 3.3.1.   
> I'll report it to them as its clearly offering a different SDP after  
> the unhold.  Yes it might be nice if Freeswitch refused to switch  
> codec on a call that is up even if the negotiation is set to generous.
> 
> Thanks for your help,
> Spencer



From Nabble at slickdeals.endjunk.com  Tue Mar  1 16:18:39 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 1 Mar 2011 05:18:39 -0800 (PST)
Subject: [Freeswitch-users] mod_dingaling and google voice
In-Reply-To: <AANLkTikdossGK+FnBfmSkfmEMF4Pv0nF5UYypA1t9bc2@mail.gmail.com>
References: <AANLkTin_dFj9ygx3TAmExeN9-pDFNCuZPGZ_Uc2d7o1W@mail.gmail.com>
	<AANLkTimUr3=w_fj_k643ChW93Q_YTm63R_qFyBO9EoQV@mail.gmail.com>
	<4D6C1978.8040702@cartissolutions.com>
	<AANLkTikdossGK+FnBfmSkfmEMF4Pv0nF5UYypA1t9bc2@mail.gmail.com>
Message-ID: <1298985519600-6077162.post@n2.nabble.com>


envelopes envelopes wrote:
> I don't use GV for direct inbound, because it is not working very well.
Care to explain what problems you have encountered with GV direct inbound?


- Get an account from sipgate.com;
> - let GV forward inbound to your sipgate PSTN #
> - and your FS register as a sipgate client.
This works just fine with an additional latency.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-dingaling-and-google-voice-tp6072163p6077162.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From peter.olsson at visionutveckling.se  Tue Mar  1 16:50:24 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 1 Mar 2011 14:50:24 +0100
Subject: [Freeswitch-users] Thoughts about mod_conference enter-sound,
 when wait-moderator has been set
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2BBAB93@cooper>

Good afternoon (at least for me)!

Stupid question maybe - but here it goes :)

When I configure a conference to wait for the moderator before connecting people, I've noticed that "enter-sound" is played synchronously. Is there any special reason for this? I'm thinking about adding a new parameter to allow it to play asynchronously, even when in this mode. My reasons for this is if there are 50 people logged in, and the moderator connects to the conference, there will be lots of time spent just to play the enter-sound for each member. 1 second of sound for each member will cause 50 seconds before it has played them all. It makes sense in the case when the caller's name is read, but for a common enter-sound I think an asynchronous playback would be good enough?

/Peter

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From u2nsam at gmail.com  Tue Mar  1 17:21:59 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 1 Mar 2011 19:51:59 +0530
Subject: [Freeswitch-users] Sip Info method
In-Reply-To: <AANLkTikQACVKZ30QY49NTESdRUtr6DD0+0D8Pa7bDKvG@mail.gmail.com>
References: <AANLkTin=A5-vEeGHvm+tJ+epspoUvZOOey2YkeVBf1=M@mail.gmail.com>
	<AANLkTinkgcN8q7CqwzEgc5URkw9YtBrxudcTQT+Ef6nP@mail.gmail.com>
	<AANLkTikxPFHpMFzHCdvUHaiW7Xiog-o523apLFJ1Urqb@mail.gmail.com>
	<AANLkTikQACVKZ30QY49NTESdRUtr6DD0+0D8Pa7bDKvG@mail.gmail.com>
Message-ID: <AANLkTimg+hj8g7JYhbkLnRDbkuEnxLEqN9rvwAZQD-Nh@mail.gmail.com>

Thinking of doing it room specific, so that on the fly the moderator of a
particular room
can initiate recording or disable it.

Regards
Sam

On Tue, Mar 1, 2011 at 5:56 PM, Steven Ayre <steveayre at gmail.com> wrote:

> The conference app mixes audio from all parties, so on your session
> you can only get the combined audio. If you can trigger the recording
> on the other party's leg instead perhaps that'll work because it won't
> have been mixed in yet. Not sure how to do that though, or if you can.
>
> -Steve
>
>
> On 1 March 2011 10:32, Sam <u2nsam at gmail.com> wrote:
> > Yes ,
> >
> > But it says that 'So as of now there seems to be no way to restrict
> > activating recording to one party.'
> >
> > Regds
> > Sam
> >
> >
> >
> > On Tue, Mar 1, 2011 at 3:07 PM, Tjardick van der Kraan
> > <tjardick at vanderkraan.net> wrote:
> >>
> >> Hi Sam,
> >> Did you check the wiki:
> >>
> >>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session#Activating_via_SIP_INFO
> >> Has some info on this.
> >> Regards,
> >> Tjardick
> >>
> >> On Tue, Mar 1, 2011 at 7:29 AM, Sam <u2nsam at gmail.com> wrote:
> >>>
> >>> Hello,
> >>>
> >>> Can we start recording any conference bridge via sip info method  ?
> >>>
> >>> It should happen for only that particular bridge users and not all the
> >>> bridges running on the application.
> >>>
> >>> For that, what parameters i should set in the SIP info method to
> achieve
> >>> this.
> >>>
> >>> Regards
> >>> Sam
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From acrow at integrafin.co.uk  Tue Mar  1 17:32:06 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Tue, 01 Mar 2011 14:32:06 +0000
Subject: [Freeswitch-users] British sounds
In-Reply-To: <4D68222C.8010108@integrafin.co.uk>
References: <4D681972.7050400@integrafin.co.uk>	<AANLkTinGQk18xc8WCap6VkyTVYM1-XHcPi1PhwVTNs8B@mail.gmail.com>
	<4D68222C.8010108@integrafin.co.uk>
Message-ID: <4D6D0366.7020205@integrafin.co.uk>

On 25/02/11 21:42, Alex Crow wrote:
>
>>     2. A script to convert and rename sounds from an Asterisk install
>>     to FS?
>>
>>     We already paid for British female sounds for Asterisk so if
>>     conversion
>>     will cover at least voicemail prompts it would be a start.
>>
>>
>> There is some overlap in the prompts, especially the digits and the 
>> time, but many of the other prompts are definitely different. I'd be 
>> willing to assist you in getting the prompts you created for * 
>> converted for use w/ FS. The challenge is figuring out which prompts 
>> need to be recorded. Do you have a list of sound prompts and file 
>> names? I could take a look...
>>
>> -MC
>>
>>
>
> Michael,
>
> The ones we have for Asterisk are British English "Rachel", purchased 
> from
>
> http://www.keison.co.uk/westany/asterisk_voice_prompt.htm
>
> They do seem to originate from westany from a google search.
>
> I think I can provide a list of the directory structure from our 
> Asterisk box. Do you have a connection to the company who recorded 
> these sounds? If we could get the same voice artist it would be great, 
> especially if they were available as a product for others to buy.
>
> Cheers
>
> Alex
> -- 
> This message is intended only for the addressee and may contain
> confidential information.  Unless you are that person, you may not
> disclose its contents or use it in any way and are requested to delete
> the message along with any attachments and notify us immediately.
>
> "Transact" is operated by Integrated Financial Arrangements plc
> Domain House, 5-7 Singer Street, London  EC2A 4BQ
> Tel: (020) 7608 4900 Fax: (020) 7608 5300
> (Registered office: as above; Registered in England and Wales under number: 3727592)
> Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)
>
> -- 
> This message has been scanned for viruses and
> dangerous content by *MailScanner* <http://www.mailscanner.info/>, and is
> believed to be clean.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
Michael,

Here is a tree of the sounds on my asterisk box. They were simply 
dropped in place over the existing US sounds. Is there anything else you 
need?

Thanks

Alex

??? account_not_valid.gsm
??? activated.gsm
??? agent-alreadyon.gsm
??? agent-incorrect.gsm
??? agent-loggedoff.gsm
??? agent-loginok.gsm
??? agent-newlocation.gsm
??? agent-pass.gsm
??? agent-user.gsm
??? all-circuits-busy-now.gsm
??? AM.gsm
??? and.gsm
??? an-error-has-occured.gsm
??? at-tone-time-exactly.gsm
??? auth-incorrect.gsm
??? auth-thankyou.gsm
??? beeperr.gsm
??? beeperr.ulaw
??? beep.gsm
??? beep.ulaw
??? call-forward.gsm
??? call-forwarding.gsm
??? call_forward_notification.gsm
??? call-forward-parallel.gsm
??? call-fwd-cancelled.gsm
??? call-fwd-no-ans.gsm
??? call-fwd-on-busy.gsm
??? call-fwd-parallel.gsm
??? call-fwd-unconditional.gsm
??? call-waiting.gsm
??? callwaiting.gsm
??? channels_limit_exceeded.gsm
??? conf-adminmenu.gsm
??? conf-enteringno.gsm
??? conf-errormenu.gsm
??? conf-getchannel.gsm
??? conf-getconfno.gsm
??? conf-getpin.gsm
??? conf-hasjoin.gsm
??? conf-hasleft.gsm
??? conf-invalid.gsm
??? conf-invalidpin.gsm
??? conf-kicked.gsm
??? conf-leaderhasleft.gsm
??? conf-locked.gsm
??? conf-lockednow.gsm
??? conf-muted.gsm
??? conf-noempty.gsm
??? conf-onlyone.gsm
??? conf-onlyperson.gsm
??? conf-otherinparty.gsm
??? conf-placeintoconf.gsm
??? conf-thereare.gsm
??? conf-unlockednow.gsm
??? conf-unmuted.gsm
??? conf-usermenu.gsm
??? conf-userswilljoin.gsm
??? conf-userwilljoin.gsm
??? conf-waitforleader.gsm
??? custom
??? dash.gsm
??? de-activated.gsm
??? deactivated.gsm
??? default_auto_attendant.gsm
??? default_moh.gsm
??? demo-abouttotry.gsm
??? demo-congrats.gsm
??? demo-echodone.gsm
??? demo-echotest.gsm
??? demo-enterkeywords.gsm
??? demo-instruct.gsm
??? demo-moreinfo.gsm
??? demo-nogo.gsm
??? demo-nomatch.gsm
??? demo-thanks.gsm
??? dest_grp_not_exist.gsm
??? destination_not_allowed.gsm
??? destination_not_recognised.gsm
??? destination_not_supported.gsm
??? destination_permis_not_set.gsm
??? dest_rate_not_set.gsm
??? dictate
?   ??? en
?       ??? both_help.gsm
?       ??? enter_filename.gsm
?       ??? forhelp.gsm
?       ??? paused.gsm
?       ??? pause.gsm
?       ??? playback.gsm
?       ??? playback_mode.gsm
?       ??? play_help.gsm
?       ??? record.gsm
?       ??? record_help.gsm
?       ??? record_mode.gsm
?       ??? truncating_audio.gsm
??? digits
?   ??? en
?       ??? 0.gsm
?       ??? 10.gsm
?       ??? 11.gsm
?       ??? 12.gsm
?       ??? 13.gsm
?       ??? 14.gsm
?       ??? 15.gsm
?       ??? 16.gsm
?       ??? 17.gsm
?       ??? 18.gsm
?       ??? 19.gsm
?       ??? 1.gsm
?       ??? 20.gsm
?       ??? 2.gsm
?       ??? 30.gsm
?       ??? 3.gsm
?       ??? 40.gsm
?       ??? 4.gsm
?       ??? 50.gsm
?       ??? 5.gsm
?       ??? 60.gsm
?       ??? 6.gsm
?       ??? 70.gsm
?       ??? 7.gsm
?       ??? 80.gsm
?       ??? 8.gsm
?       ??? 90.gsm
?       ??? 9.gsm
?       ??? a-m.gsm
?       ??? at.gsm
?       ??? day-0.gsm
?       ??? day-1.gsm
?       ??? day-2.gsm
?       ??? day-3.gsm
?       ??? day-4.gsm
?       ??? day-5.gsm
?       ??? day-6.gsm
?       ??? dollar.gsm
?       ??? dollars.gsm
?       ??? euro.gsm
?       ??? euros.gsm
?       ??? h-10.gsm
?       ??? h-11.gsm
?       ??? h-12.gsm
?       ??? h-13.gsm
?       ??? h-14.gsm
?       ??? h-15.gsm
?       ??? h-16.gsm
?       ??? h-17.gsm
?       ??? h-18.gsm
?       ??? h-19.gsm
?       ??? h-1.gsm
?       ??? h-20.gsm
?       ??? h-2.gsm
?       ??? h-30.gsm
?       ??? h-3.gsm
?       ??? h-4.gsm
?       ??? h-5.gsm
?       ??? h-6.gsm
?       ??? h-7.gsm
?       ??? h-8.gsm
?       ??? h-9.gsm
?       ??? hundred.gsm
?       ??? million.gsm
?       ??? minus.gsm
?       ??? mon-0.gsm
?       ??? mon-10.gsm
?       ??? mon-11.gsm
?       ??? mon-1.gsm
?       ??? mon-2.gsm
?       ??? mon-3.gsm
?       ??? mon-4 .gsm
?       ??? mon-5.gsm
?       ??? mon-6 .gsm
?       ??? mon-7.gsm
?       ??? mon-8 .gsm
?       ??? mon-9.gsm
?       ??? oclock.gsm
?       ??? oh.gsm
?       ??? p-m.gsm
?       ??? pound.gsm
?       ??? pounds.gsm
?       ??? star.gsm
?       ??? thousand.gsm
?       ??? today.gsm
?       ??? tomorrow.gsm
?       ??? yesterday.gsm
??? dir-instr.gsm
??? dir-intro-fn.gsm
??? dir-intro-fnln.gsm
??? dir-intro-fnln-oper.gsm
??? dir-intro-fn-oper.gsm
??? dir-intro.gsm
??? dir-intro-oper.gsm
??? dir-nomatch.gsm
??? dir-nomore.gsm
??? dollar.gsm
??? dollars.gsm
??? do-not-disturb.gsm
??? dot.gsm
??? enter_account_number.gsm
??? enter-conf-pin-number.gsm
??? enter_ext_number.gsm
??? enter_ntwk_num.gsm
??? enter-num-blacklist.gsm
??? entr-num-rmv-blklist.gsm
??? ent-target-attendant.gsm
??? equals.gsm
??? exclaimation-point.gsm
??? extension.gsm
??? feature-not-avail-line.gsm
??? first-in-line.gsm
??? first-three-letters-entry.gsm
??? followme
?   ??? en
?       ??? call-from.gsm
?       ??? no-recording.gsm
?       ??? options.gsm
?       ??? sorry.gsm
?       ??? status.gsm
??? for2.gsm
??? for_accounting.gsm
??? for_accounts.gsm
??? for_customer_support.gsm
??? for_development.gsm
??? for.gsm
??? for_production.gsm
??? for_sales.gsm
??? for_technical_support.gsm
??? freshtel.gsm
??? goodbye.gsm
??? greeting-default-attendant.gsm
??? hash.gsm
??? hello-world.gsm
??? hours.gsm
??? if-correct-press.gsm
??? incoming-call-1-accept-2-decline.sln
??? incoming-call-no-longer-avail.sln
??? info-about-last-call.gsm
??? inithelp.gsm
??? insufficient_acc_bal.gsm.gsm
??? invalid.gsm
??? invalid_selection.gsm
??? is-curntly-unavail.gsm
??? is.gsm
??? is-set-to.gsm
??? is_unable_to_answer.gsm
??? last-num-to-call2.gsm
??? last-num-to-call.gsm
??? letters
?   ??? en
?       ??? a.gsm
?       ??? at.gsm
?       ??? b.gsm
?       ??? c.gsm
?       ??? dash.gsm
?       ??? d.gsm
?       ??? dollar.gsm
?       ??? dollars.gsm
?       ??? dot.gsm
?       ??? e.gsm
?       ??? equals.gsm
?       ??? euro.gsm
?       ??? euros.gsm
?       ??? exclaimation-point.gsm
?       ??? f.gsm
?       ??? g.gsm
?       ??? h.gsm
?       ??? i.gsm
?       ??? j.gsm
?       ??? k.gsm
?       ??? l.gsm
?       ??? m.gsm
?       ??? n.gsm
?       ??? o.gsm
?       ??? p.gsm
?       ??? plus.gsm
?       ??? q.gsm
?       ??? r.gsm
?       ??? s.gsm
?       ??? slash.gsm
?       ??? space.gsm
?       ??? t.gsm
?       ??? u.gsm
?       ??? v.gsm
?       ??? w.gsm
?       ??? x.gsm
?       ??? y.gsm
?       ??? zee.gsm
?       ??? z.gsm
??? minutes.gsm
??? no_active_agents.gsm
??? no_phone_support_for_acc.gsm
??? not_authorized.gsm
??? not_enabled.gsm
??? not_in_service.gsm
??? now.gsm
??? ntwk_no_not_exist.gsm
??? number2.gsm
??? number.gsm
??? one-moment-please.gsm
??? out_of_order.gsm
??? pbx-invalid.gsm
??? pbx-invalidpark.gsm
??? pbx-transfer.gsm
??? percent.gsm
??? phonetic
?   ??? en
?       ??? 9_p.gsm
?       ??? a_p.gsm
?       ??? b_p.gsm
?       ??? c_p.gsm
?       ??? d_p.gsm
?       ??? e_p.gsm
?       ??? f_p.gsm
?       ??? g_p.gsm
?       ??? h_p.gsm
?       ??? i_p.gsm
?       ??? j_p.gsm
?       ??? k_p.gsm
?       ??? l_p.gsm
?       ??? m_p.gsm
?       ??? niner.gsm
?       ??? n_p.gsm
?       ??? o_p.gsm
?       ??? p_p.gsm
?       ??? q_p.gsm
?       ??? r_p.gsm
?       ??? s_p.gsm
?       ??? t_p.gsm
?       ??? u_p.gsm
?       ??? v_p.gsm
?       ??? w_p.gsm
?       ??? x_p.gsm
?       ??? y_p.gsm
?       ??? z_p.gsm
??? please_contact_cust_support.gsm
??? please-enter-your.gsm
??? pls_enter_pin.gsm
??? pls-try-call-later.gsm
??? plus.gsm
??? PM.gsm
??? press-0.gsm
??? press-1.gsm
??? press-2.gsm
??? press-3.gsm
??? press-4.gsm
??? press-5.gsm
??? press-6.gsm
??? press-7.gsm
??? press-8.gsm
??? press-9.gsm
??? press_eight.gsm
??? press_five.gsm
??? press_four.gsm
??? press_hash.gsm
??? press_nine.gsm
??? press_one.gsm
??? press_seven.gsm
??? press_six.gsm
??? press_star.gsm
??? press-star.gsm
??? press_three.gsm
??? press_two.gsm
??? press_zero.gsm
??? privacy-blklist-last-caller.gsm
??? privacy-incorrect.gsm
??? privacy-prompt.gsm
??? privacy-thankyou.gsm
??? privacy-unident.gsm
??? priv-at.gsm
??? priv-callee-options.gsm
??? priv-callfrom.gsm
??? priv-callpending.gsm
??? priv-instruct.gsm
??? priv-introsaved.gsm
??? priv-recordintro.gsm
??? priv-sayname.gsm
??? priv-trying.gsm
??? queue-callswaiting.gsm
??? queue-holdtime.gsm
??? queue-less-than.gsm
??? queue-minutes.gsm
??? queue_number.gsm
??? queue-periodic-announce.gsm
??? queue-reporthold.gsm
??? queue-seconds.gsm
??? queue-thankyou.gsm
??? queue-thereare.gsm
??? queue-youarenext.gsm
??? screen-callee-options.gsm
??? secondary_trunk_used.gsm
??? seconds.gsm
??? security.gsm
??? slash.gsm
??? sorry-cant-let-you-do-that.gsm
??? space.gsm
??? speed-dial-empty.gsm
??? speed-dial.gsm
??? spy-agent.gsm
??? spy-h323.gsm
??? spy-iax2.gsm
??? spy-iax.gsm
??? spy-mgcp.gsm
??? spy-sip.gsm
??? spy-skinny.gsm
??? spy-zap.gsm
??? ss-noservice.gsm
??? telephone-number.gsm
??? teritary_trunk_used.gsm
??? then-press-pound2.gsm
??? then-press-pound.gsm
??? the-number-u-dialed.gsm
??? T-is-not-available.gsm
??? to-call-this-number.gsm
??? to-listen-to-it.gsm
??? to_listen_to_these_options.gsm
??? to-rerecord-it.gsm
??? to_speak_to_an_operator.gsm
??? transfer.gsm
??? tt-allbusy.gsm
??? user_account_suspended.gsm
??? user_not_activated.gsm
??? vertical_serv_not_available.gsm
??? vm-advopts.gsm
??? vm-and.gsm
??? vm-calldiffnum.gsm
??? vm-changeto.gsm
??? vm-Cust1.gsm
??? vm-Cust2.gsm
??? vm-Cust3.gsm
??? vm-Cust4.gsm
??? vm-Cust5.gsm
??? vm-deleted.gsm
??? vm-delete.gsm
??? vm-dialout.gsm
??? vm-enter-num-to-call.gsm
??? vm-extension.gsm
??? vm-Family.gsm
??? vm-first.gsm
??? vm-for.gsm
??? vm-forward.gsm
??? vm-forwardoptions.gsm
??? vm-Friends.gsm
??? vm-from-extension.gsm
??? vm-from.gsm
??? vm-from-phonenumber .gsm
??? vm-goodbye.gsm
??? vm-helpexit.gsm
??? vm-INBOX.gsm
??? vm-incorrect.gsm
??? vm-incorrect-mailbox.gsm
??? vm-instructions.gsm
??? vm-intro.gsm
??? vm-isonphone.gsm
??? vm-isunavail.gsm
??? vm-last.gsm
??? vm-leavemsg.gsm
??? vm_login.gsm
??? vm-login.gsm
??? vm-mailboxfull.gsm
??? vm-message.gsm
??? vm-messages.gsm
??? vm-minutes.gsm
??? vm-mismatch.gsm
??? vm-msginstruct.gsm
??? vm-msgsaved.gsm
??? vm-newpassword.gsm
??? vm-newuser.gsm
??? vm-next.gsm
??? vm-nobodyavail.gsm
??? vm-nobox.gsm
??? vm-no.gsm
??? vm-nomore.gsm
??? vm-nonumber.gsm
??? vm-num-i-have.gsm
??? vm-Old.gsm
??? vm-onefor.gsm
??? vm-options.gsm
??? vm-opts.gsm
??? vm-passchanged.gsm
??? vm-password.gsm
??? vm-press.gsm
??? vm-prev.gsm
??? vm-reachoper.gsm
??? vm-rec-busy.gsm
??? vm-received.gsm
??? vm-rec-name.gsm
??? vm-rec-temp.gsm
??? vm-rec-unv.gsm
??? vm-reenterpassword.gsm
??? vm-repeat.gsm
??? vm-review.gsm
??? vm-saved.gsm
??? vm-savedto.gsm
??? vm-savefolder.gsm
??? vm-savemessage.gsm
??? vm-saveoper.gsm
??? vm-sorry.gsm
??? vm-star-cancel.gsm
??? vm-starmain.gsm
??? vm-tempgreetactive.gsm
??? vm-tempgreeting2.gsm
??? vm-tempgreeting.gsm
??? vm-tempremoved.gsm
??? vm-then-hash.gsm
??? vm-then-pound.gsm
??? vm-theperson.gsm
??? vm-tocallback.gsm
??? vm-tocallnum.gsm
??? vm-tocancel.gsm
??? vm-tocancelmsg.gsm
??? vm-toenternumber.gsm
??? vm-toforward.gsm
??? vm-tohearenv.gsm
??? vm-tomakecall.gsm
??? vm-tooshort.gsm
??? vm-toreply.gsm
??? vm-torerecord.gsm
??? vm-undeleted.gsm
??? vm-undelete.gsm
??? vm-unknown-caller.gsm
??? vm-whichbox.gsm
??? vm-Work.gsm
??? vm-youhave.gsm
??? welcome-to-phonebook.gsm
??? your.gsm

11 directories, 513 files


-- 
This message is intended only for the addressee and may contain
confidential information.  Unless you are that person, you may not
disclose its contents or use it in any way and are requested to delete
the message along with any attachments and notify us immediately.

"Transact" is operated by Integrated Financial Arrangements plc
Domain House, 5-7 Singer Street, London  EC2A 4BQ
Tel: (020) 7608 4900 Fax: (020) 7608 5300
(Registered office: as above; Registered in England and Wales under number: 3727592)
Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)

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From bggoutham at gmail.com  Tue Mar  1 17:56:39 2011
From: bggoutham at gmail.com (Goutham BG)
Date: Tue, 1 Mar 2011 20:26:39 +0530
Subject: [Freeswitch-users] Global configuration to enable SRTP in the
 entire FS system
In-Reply-To: <48A2CED0-18E7-4183-BB6E-91B4CAAB613C@freeswitch.org>
References: <AANLkTimuELApghD2yqXiJnrA9spZ0Bgy7rBSZs5aVi2W@mail.gmail.com>
	<48A2CED0-18E7-4183-BB6E-91B4CAAB613C@freeswitch.org>
Message-ID: <AANLkTimxB6sxoMATUkGZQLNeAkdznzifxQXTyL5O7V2u@mail.gmail.com>

Thank you. It worked.

Thanks
Goutham B G

On Tue, Mar 1, 2011 at 5:54 PM, Brian West <brian at freeswitch.org> wrote:

> <X-PRE-PROCESS cmd="set" data="sip_secure_media=true"/> in vars.xml or
> anywhere in your xml for that matter... or global_setvar
> sip_secure_media=true
>
> /b
>
>
> On Mar 1, 2011, at 3:24 AM, Goutham BG wrote:
>
> Hi All,
>
> We can enable SRTP for a particular channel/call by setting the variable
> sip_secure_media in the XML dialplan or in the dial string. Is there any global
> configuration to enable and enforce SRTP which would apply to all the
> extensions in the dialplan and all the channels that are created.
>
> Thanks
> Goutham B G
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jjj at 3js.de  Tue Mar  1 19:27:55 2011
From: jjj at 3js.de (Johannes Jakob)
Date: Tue, 1 Mar 2011 17:27:55 +0100
Subject: [Freeswitch-users] T.38 Issues with passthrough handshaking
In-Reply-To: <AANLkTi=GmF93TkQPNuFf9ko_eQVgvqwujZOo8duN8ZDB@mail.gmail.com>
References: <640F91CF-9489-4F74-96B1-A6398CD84F07@3js.de>
	<A53FD340-169C-4DD4-8AF2-C463F9D9E3DD@3js.de>
	<AANLkTi=GmF93TkQPNuFf9ko_eQVgvqwujZOo8duN8ZDB@mail.gmail.com>
Message-ID: <64786198-FF65-49FA-BCC6-57337A412D37@3js.de>

Hi Anthony,
hi everyone,


Your help is very appreciated, thank you very much!
I'm really getting mad with all those T.38 stuff. weeks and weeks of troubleshooting and try and error - yes, I don't really understand t.38...

On 25.02.2011, at 00:04, Anthony Minessale wrote:

> <!-- you only need to set this on lastest builds of FS -->
>     <action application="export" data="t38_passthru=true"/>


Okay, thanks for this hint, the wiki suggests proxy_media... as soon as I'll get this stuff working, I'll add my experiences with my (little) insights.


> <!-- you don't mix t38_passthru with proxy media get rid of these
> completely  -->

I've done that... removed any proxy_media stuff in all profiles/dialplans.

I'm still not getting anywhere... incoming fax is still working, but outbound faxing still is not possible.

I put up a flow-chart (by wireshark) and a dump of the relevant re-invite and 200OK packets here;

	http://www.3js.de/20110301/

As far as I can see, ports get negotiated just fine, and the fax is being sent, but somehow this looks like a monolog...
Even though it looks like the fax itself is being sent fine, there is no fax received on the other side and my fax reports an error and unsent fax.

PLEASE have a quick look at those, I'm quite desperate...

> Maybe someday you'll be brave and try terminating the t38 right to FS instead.

Hmm, if I understand this right, you would suggest to terminate the outgoing fax on the FS:

Upstream <(a)> FS <(b)> asterisk <(c)> SIP ATA

The fax is being sent by ATA via plain Audio (c) to the asterisk box, which is passing it along to FS as plain audio (b). Now you'd like me to let FS send the reinvite back to the asterisk box? That means, according to the wiki I'd have to do:

<extension name="t38_reinvite">
  <condition>
    <action application="set" data="fax_enable_t38=true"/> 
    <action application="set" data="fax_enable_t38_request=true"/>
    <action application="set" data="execute_on_answer=t38_gateway self"/>
    <action application="bridge" data="sofia/external/1234 at host"/>
  </condition>
</extension>
 
But what happens for the B-leg, going out to the my upstream? There will be a t.38 reinvite as well, coming from the Upstream... won't that be a problem / a more complicated system then needed? Why would you prefer this over passthru?


> BTW it seems like whatever is generating your sip trace has a bug in
> it, there are garbage characters before each sip message unless its
> printing unfiltered network packets or something.

tcpdump -A -s0 -w file.pcap 
tcpdump -An -r file.pcap > plain.txt
That had to be done so I could replace sensible stuff with fake data....



Thanks again for your help so far!



Best regards,

  John




> On Thu, Feb 24, 2011 at 4:37 PM, Johannes Jakob <jjj at 3js.de> wrote:
>> Hi again,
>> 
>> Just a short followup:
>> 
>> originating a tif file directly on the FS (txfax) does work,
>> trying to send the same tif file from the asterisk with SendFax doesn't.
>> 
>> Full traces (sorry, just ascii, no pcap, but I had to sanitize them somehow) of all three test cases, for those having the time and willing to help, can be found here:
>> 
>>        http://www.3js.de/debug.tgz
>> 
>> 
>> According to how few people are complaining, it must be quite simple to get t38 working... am I blind or just too stupid to see my errors?
>> 
>> 
>> Thanks a lot for any hint in the right direction!
>> 
>> Best regards,
>> 
>>  John
>> 
>> 
>> 
>> 
>> On 24.02.2011, at 17:44, Johannes Jakob wrote:
>> 
>>> Hi,
>>> 
>>> I'm having some serious trouble getting outbound t38 passthru to work in the following scenario:
>>> 
>>> SIP Upstream <> FreeSWITCH <> Asterisk <> SIP ATA <> analog fax
>>> 
>>> I'm not sure if it's a plain FS problem, but I have to start somewhere and I found at least one problem (I think) with FS's behaviour when negotiating with the gateway.
>>> 
>>> Inbound T.38 faxes, when the ATA has to do the reinvite, work perfectly fine from one end to the other.
>>> 
>>> Outbound faxes on the other hand, don't work at all.
>>> 
>>> Here are the important config parts:
>>> 
>>> dialplan before bridging to gateway:
>>>      <action application="export" data="t38_passthru=true"/>
>>>      <action application="set" data="proxy_media=true"/>
>>>      <action application="set" data="bypass_media=false"/>
>>> 
>>> directory-entry for the asterisk-registration:
>>>      <param name="t38-passthru" value="true"/>
>>> and
>>>      <variable name="proxy_media" value="true"/>
>>>      <variable name="bypass_media" value="false"/>
>>> 
>>> and finally the gateway's sip_profile:
>>>  <param name="t38-passthru" value="true"/>
>>>  <param name="inbound-proxy-media" value="true"/>
>>> 
>>> So much for introduction... When trying to send a fax to 017286483798, this is what happens on the SBC:
>>> 
>>> 
>>> asterisk "client" is sending the plain audio INVITE to SBC:
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 9     37.668216       10.16.139.28    10.16.133.66    SIP/SDP 1202    Request: INVITE sip:017286483798 at sbc1.mysip.net, with session description
>>> AkE`b?FG^^B%INVITE sip:017286483798 at sbc1.mysip.net SIP/2.0
>>> Via: SIP/2.0/UDP 10.16.139.28:5060;branch=z9hG4bK3c5810ec
>>> Max-Forwards: 70
>>> From: "0692386432" <sip:0692386432 at mysip.net>;tag=as235c17b6
>>> To: <sip:017286483798 at sbc1.mysip.net>
>>> Contact: <sip:0692386432 at 10.16.139.28:5060>
>>> Call-ID: 6de3237b20a96ab010fb9b3f2436d305 at mysip.net
>>> CSeq: 103 INVITE
>>> User-Agent: FPBX-2.8.1(1.8.2.4)
>>> Proxy-Authorization: Digest username="sipuser", realm="mysip.net", algorithm=MD5, uri="sip:017286483798 at sbc1.mysip.net", nonce="aa9339fc-9e53-4017-85ff-fac5367cb733", response="96f4dea7267d436779cd106947cc25b7", qop=auth, cnonce="4e110d27", nc=00000001
>>> Date: Thu, 24 Feb 2011 07:30:27 GMT
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
>>> Supported: replaces, timer
>>> Content-Type: application/sdp
>>> Content-Length: 285
>>> 
>>> v=0
>>> o=root 1297612317 1297612318 IN IP4 10.16.139.28
>>> s=Asterisk PBX 1.8.2.4
>>> c=IN IP4 10.16.139.28
>>> t=0 0
>>> m=audio 15000 RTP/AVP 0 8 3 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:3 GSM/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>> a=sendrecv
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 
>>> SBC is forwarding the request to the correct gateway:
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 11    37.716704       10.16.133.66    10.15.12.215    SIP/SDP 1232    Request: INVITE sip:+4317286483798 at 10.15.12.215, with session description
>>> =|$E%@^BWB{INVITE sip:+4317286483798 at 10.15.12.215 SIP/2.0
>>> Via: SIP/2.0/UDP 10.16.133.66:5080;rport;branch=z9hG4bK33eQ6Na9D991D
>>> Max-Forwards: 69
>>> From: "0692386432" <sip:0692386432 at 10.16.133.66>;tag=4NH77aQN2v7ZS
>>> To: <sip:+4317286483798 at 10.15.12.215>
>>> Call-ID: e5c9ab65-ba8a-122e-21ad-863d7c249180
>>> CSeq: 8930272 INVITE
>>> Contact: <sip:gw+gateway1 at 10.16.133.66:5080;transport=udp;gw=gateway1>
>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7847289 2011-02-19 23-38-04 +0100
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
>>> Supported: timer, precondition, path, replaces
>>> Allow-Events: talk, hold, refer
>>> Content-Type: application/sdp
>>> Content-Disposition: session
>>> Content-Length: 269
>>> X-FS-Support: update_display
>>> Remote-Party-ID: "0692386432" <sip:0692386432 at 10.16.133.66>;party=calling;screen=yes;privacy=off
>>> P-Asserted-Identity: <sip:06923864320 at sip.gateway.de>
>>> 
>>> v=0
>>> o=FreeSWITCH 3022782041 3022782042 IN IP4 10.16.133.66
>>> s=FreeSWITCH
>>> c=IN IP4 10.16.133.66
>>> t=0 0
>>> m=audio 18446 RTP/AVP 0 8 3 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:3 GSM/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 
>>> Gatway says, everything OK, audio call patched through:
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 22    39.760159       10.15.12.215    10.16.133.66    SIP/SDP 750     Status: 200 OK, with session description
>>> AkEw  W^BSIP/2.0 200 OK
>>> Via: SIP/2.0/UDP 10.16.133.66:5080;branch=z9hG4bK33eQ6Na9D991D;rport=5080
>>> Call-ID: e5c9ab65-ba8a-122e-21ad-863d7c249180
>>> From: "0692386432"<sip:0692386432 at 10.16.133.66>;tag=4NH77aQN2v7ZS
>>> To: <sip:+4317286483798 at 10.15.12.215>;tag=h1hl6tsu-CC-39
>>> CSeq: 8930272 INVITE
>>> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
>>> Contact: <sip:10.15.12.215:5060;user=phone>
>>> Content-Length: 217
>>> Content-Type: application/sdp
>>> 
>>> v=0
>>> o=HuaweiSoftX3000 3014926 3014927 IN IP4 10.15.12.215
>>> s=Sip Call
>>> c=IN IP4 10.15.12.215
>>> t=0 0
>>> m=audio 15490 RTP/AVP 0 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=ptime:20
>>> a=fmtp:101 0-15
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 
>>> SBC is telling the asterisk box, everything is fine
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 28    39.770349       10.16.133.66    10.16.139.28    SIP/SDP 1185    Status: 200 OK, with session description
>>> =|$Ew at 1y^B^}aSIP/2.0 200 OK
>>> Via: SIP/2.0/UDP 10.16.139.28:5060;branch=z9hG4bK3c5810ec
>>> From: "0692386432" <sip:0692386432 at mysip.net>;tag=as235c17b6
>>> To: <sip:017286483798 at sbc1.mysip.net>;tag=crS25Ur70m0BK
>>> Call-ID: 6de3237b20a96ab010fb9b3f2436d305 at mysip.net
>>> CSeq: 103 INVITE
>>> Contact: <sip:017286483798 at 10.16.133.66:5060;transport=udp>
>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7847289 2011-02-19 23-38-04 +0100
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>> Supported: timer, precondition, path, replaces
>>> Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
>>> Content-Type: application/sdp
>>> Content-Disposition: session
>>> Content-Length: 222
>>> Remote-Party-ID: "017286483798" <sip:017286483798 at 10.16.133.66>;party=calling;privacy=off;screen=no
>>> 
>>> v=0
>>> o=FreeSWITCH 3022845683 3022845684 IN IP4 10.16.133.66
>>> s=FreeSWITCH
>>> c=IN IP4 10.16.133.66
>>> t=0 0
>>> m=audio 28850 RTP/AVP 0 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=ptime:20
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 
>>> so, after the ACK which is passed along as well, there is the typical audio stuff, ringing... in both directions the right, pre-negotiated udp ports, everything is fine so far:
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 45    39.849484       10.15.12.215    10.16.133.66    RTP     216     PT=ITU-T G.711 PCMU, SSRC=0x727E59A0, Seq=23045, Time=3032460352
>>> User Datagram Protocol, Src Port: 15490 (15490), Dst Port: 18446 (18446)
>>> -----------------------------------------------------------------------------------------------------------------------
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 46    39.849541       10.16.133.66    10.16.139.28    RTP     216     PT=ITU-T G.711 PCMU, SSRC=0x727E59A0, Seq=23045, Time=3032460352
>>> User Datagram Protocol, Src Port: 28850 (28850), Dst Port: hydap (15000)
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 
>>> Now, the magic happens, the gateway is sending it's T.38 re-INVITE to establish better fax connectivity... well, let's see what happens:
>>> 
>>> The gateway is suggesting to move the audio stuff from port 15490 to 15492, and instead speak t38/udptl on port 15490, keep that in mind.
>>> 
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 2099  50.121304       10.15.12.215    10.16.133.66    SIP/SDP 1026    Request: INVITE sip:gw+gateway1 at 10.16.133.66:5080;transport=udp;gw=gateway1, in-dialog, with session description
>>> AkEW^BwINVITE sip:gw+gateway1 at 10.16.133.66:5080;transport=udp;gw=gateway1 SIP/2.0
>>> Via: SIP/2.0/UDP 10.15.12.215:5060;branch=z9hG4bK4abf7e790396e9d932f59632f
>>> Call-ID: e5c9ab65-ba8a-122e-21ad-863d7c249180
>>> From: <sip:+4317286483798 at 10.15.12.215>;tag=h1hl6tsu-CC-39
>>> To: "0692386432"<sip:0692386432 at 10.16.133.66>;tag=4NH77aQN2v7ZS
>>> CSeq: 2 INVITE
>>> Max-Forwards: 69
>>> Contact: <sip:10.15.12.215:5060>
>>> Content-Length: 527
>>> Content-Type: application/sdp
>>> 
>>> v=0
>>> o=HuaweiSoftX3000 3014926 3014928 IN IP4 10.15.12.215
>>> s=Sip Call
>>> c=IN IP4 10.15.12.215
>>> t=0 0
>>> m=image 15490 udptl t38
>>> a=T38FaxVersion:0
>>> a=T38MaxBitRate:14400
>>> a=T38FaxRateManagement:transferredTCF
>>> a=T38FaxUdpEC:t38UDPRedundancy
>>> m=audio 15492 RTP/AVP 8 0 127 103 101
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:127 PCMU/8000
>>> a=gpmd:127 vbd=yes
>>> a=rtpmap:103 PCMA/8000
>>> a=gpmd:103 vbd=yes
>>> a=rtpmap:101 telephone-event/8000
>>> a=ptime:20
>>> a=silenceSupp:off - - - -
>>> a=ecan:fb on -
>>> a=X-fax
>>> a=fmtp:101 0-15
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 
>>> FS is handing this INVITE to asterisk (and tells asterisk, it would accept audio and/or t.38, both on the same port 28850, don't think that's a problem, but it's at least different from HuaweiSoftX3000's behavior):
>>> 
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 2101  50.122833       10.16.133.66    10.16.139.28    SIP/SDP 1398    Request: INVITE sip:0692386432 at 10.16.139.28:5060, in-dialog, with session description
>>> =|$Efw at 0^B^R6INVITE sip:0692386432 at 10.16.139.28:5060 SIP/2.0
>>> Via: SIP/2.0/UDP 10.16.133.66;rport;branch=z9hG4bKy5perv1F90acS
>>> Max-Forwards: 70
>>> From: <sip:017286483798 at sbc1.mysip.net>;tag=crS25Ur70m0BK
>>> To: "0692386432" <sip:0692386432 at mysip.net>;tag=as235c17b6
>>> Call-ID: 6de3237b20a96ab010fb9b3f2436d305 at mysip.net
>>> CSeq: 8930278 INVITE
>>> Contact: <sip:017286483798 at 10.16.133.66:5060;transport=udp>
>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7847289 2011-02-19 23-38-04 +0100
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>> Supported: timer, precondition, path, replaces
>>> Content-Type: application/sdp
>>> Content-Disposition: session
>>> Content-Length: 532
>>> X-FS-Support: update_display
>>> P-Asserted-Identity: <sip:06923864320 at sip.gateway.de>
>>> 
>>> v=0
>>> o=FreeSWITCH 3022845683 3022845685 IN IP4 10.16.133.66
>>> s=FreeSWITCH
>>> c=IN IP4 10.16.133.66
>>> t=0 0
>>> m=image 28850 udptl t38
>>> a=T38FaxVersion:0
>>> a=T38MaxBitRate:14400
>>> a=T38FaxRateManagement:transferredTCF
>>> a=T38FaxUdpEC:t38UDPRedundancy
>>> m=audio 28850 RTP/AVP 8 0 127 103 101
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:127 PCMU/8000
>>> a=rtpmap:103 PCMA/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=gpmd:127 vbd=yes
>>> a=gpmd:103 vbd=yes
>>> a=ptime:20
>>> a=silenceSupp:off - - - -
>>> a=ecan:fb on -
>>> a=X-fax
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 
>>> The asterisk box says this is fine (after of course successfully talking to the ATA, which is fine with it, too, but want's to have slower speed). The asterisk box is also changing the port it wants to get t38 data on from 15508 to 4676 and finally sets the udp port for audio 0 to disable it.
>>> Just plain t.38 in the new SDP description:
>>> 
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 2107  50.165366       10.16.139.28    10.16.133.66    SIP/SDP 920     Status: 200 OK, with session description
>>> AkE`b?G^^^BtAbSIP/2.0 200 OK
>>> Via: SIP/2.0/UDP 10.16.133.66;branch=z9hG4bKy5perv1F90acS;received=10.16.133.66;rport=5060
>>> From: <sip:017286483798 at sbc1.mysip.net>;tag=crS25Ur70m0BK
>>> To: "0692386432" <sip:0692386432 at mysip.net>;tag=as235c17b6
>>> Call-ID: 6de3237b20a96ab010fb9b3f2436d305 at mysip.net
>>> CSeq: 8930278 INVITE
>>> Server: FPBX-2.8.1(1.8.2.4)
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
>>> Supported: replaces, timer
>>> Contact: <sip:0692386432 at 10.16.139.28:5060>
>>> Content-Type: application/sdp
>>> Content-Length: 307
>>> 
>>> v=0
>>> o=root 1297612317 1297612319 IN IP4 10.16.139.28
>>> s=Asterisk PBX 1.8.2.4
>>> c=IN IP4 10.16.139.28
>>> t=0 0
>>> m=audio 0 RTP/AVP 8 0 127 103 101
>>> m=image 4676 udptl t38
>>> a=T38FaxVersion:0
>>> a=T38MaxBitRate:9600
>>> a=T38FaxRateManagement:transferredTCF
>>> a=T38FaxMaxDatagram:397
>>> a=T38FaxUdpEC:t38UDPRedundancy
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 
>>> BUT look at this! What does FreeSWITCH tell the gateway???
>>> It sends 200 OK, but suddenly wants to receive only audio data and disables comfort noise?
>>> Same udp port as before, but no sign of t.38 in the SDP description!
>>> 
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 2111  50.170760       10.16.133.66    10.15.12.215    SIP/SDP 902     Status: 200 OK, with session description
>>> =|$Ev%@^BWbA1SIP/2.0 200 OK
>>> Via: SIP/2.0/UDP 10.15.12.215:5060;branch=z9hG4bK4abf7e790396e9d932f59632f
>>> From: <sip:+4317286483798 at 10.15.12.215>;tag=h1hl6tsu-CC-39
>>> To: "0692386432" <sip:0692386432 at 10.16.133.66>;tag=4NH77aQN2v7ZS
>>> Call-ID: e5c9ab65-ba8a-122e-21ad-863d7c249180
>>> CSeq: 2 INVITE
>>> Contact: <sip:gw+gateway1 at 10.16.133.66:5080;transport=udp;gw=gateway1>
>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7847289 2011-02-19 23-38-04 +0100
>>> Accept: application/sdp
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
>>> Supported: timer, precondition, path, replaces
>>> Content-Type: application/sdp
>>> Content-Disposition: session
>>> Content-Length: 168
>>> 
>>> v=0
>>> o=FreeSWITCH 3022782041 3022782043 IN IP4 10.16.133.66
>>> s=FreeSWITCH
>>> c=IN IP4 10.16.133.66
>>> t=0 0
>>> m=audio 18446 RTP/AVP 8 0 127 103 101
>>> m=audio 0 RTP/AVP 19
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 
>>> and look at this, even though we just told the gateway to only talk audio to it, we send a t38 packet! (it's this lonely one though!)
>>> 
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 2112  50.174592       94.186.133.66   87.234.1.215    T.38    216     UDP: UDPTLPacket Seq=32768  t30ind: <unknown>[UNKNOWN PER: 10.9.3.8.1][Malformed Packet]
>>> User Datagram Protocol, Src Port: 18446 (18446), Dst Port: 15490 (15490)
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 
>>> 
>>> The gateway keeps sending normal audio to us on the specified and unchanged port, BUT from the udp port it originally told us it would only accept t.38 on...
>>> 
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 2260  51.589980       10.15.12.215    10.16.133.66    RTP     216     PT=ITU-T G.711 PCMA, SSRC=0x727E59A0, Seq=23634, Time=3032554272
>>> User Datagram Protocol, Src Port: 15490 (15490), Dst Port: 18446 (18446)
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 
>>> While the gateway is sending us plain audio, we are talking t38 to  the asterisk box (which is not responding).
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 2261  51.590063       10.16.133.66    10.16.139.28    T.38    216     UDP: UDPTLPacket Seq=32776  data:v8:[UNKNOWN PER: too long integer(per_integer)][Malformed Packet]
>>> User Datagram Protocol, Src Port: 28850 (28850), Dst Port: dhct-alerts (4676)
>>> -----------------------------------------------------------------------------------------------------------------------
>>> 
>>> 
>>> 
>>> For the record, on the asterisk (version 1.8.2.4) box I defined
>>> t38pt_udptl=yes,redundancy
>>> directmedia=no
>>> for the gateways and the ATA's extension.
>>> 
>>> On both, the sbc and the asterisk box I compiled res_fax_spandsp, mod_spandsp with spandsp-0.0.6pre18.
>>> 
>>> So... I've been spending far tooooo much time debugging this and I'm quite sure I'm just too stupid to find a solution for this.
>>> 
>>> 
>>> Is there any good pcap anonymizing utitlity, that can substitute application layer stuff as well?
>>> 
>>> Well, *any* help/hint would be appreciated very much ;)
>>> 
>>> Thanks in advance,
>>> 
>>>  John
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>> 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
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> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _______________________________________________
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> http://www.freeswitch.org



From infos at madovsky.org  Tue Mar  1 19:40:00 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 1 Mar 2011 11:40:00 -0500
Subject: [Freeswitch-users] T.38 Issues with passthrough handshaking
References: <640F91CF-9489-4F74-96B1-A6398CD84F07@3js.de><A53FD340-169C-4DD4-8AF2-C463F9D9E3DD@3js.de><AANLkTi=GmF93TkQPNuFf9ko_eQVgvqwujZOo8duN8ZDB@mail.gmail.com>
	<64786198-FF65-49FA-BCC6-57337A412D37@3js.de>
Message-ID: <1951319FA2234F89920E93B69BAC5351@e1705>

> > Okay, thanks for this hint, the wiki suggests proxy_media...
Johannes,
Anthony is one of the main developer of FS,
so you can trust him at 100%

----- Original Message ----- 
From: "Johannes Jakob" <jjj at 3js.de>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, March 01, 2011 11:27 AM
Subject: Re: [Freeswitch-users] T.38 Issues with passthrough handshaking


> Hi Anthony,
> hi everyone,
>
>
> Your help is very appreciated, thank you very much!
> I'm really getting mad with all those T.38 stuff. weeks and weeks of 
> troubleshooting and try and error - yes, I don't really understand t.38...
>
> On 25.02.2011, at 00:04, Anthony Minessale wrote:
>
>> <!-- you only need to set this on lastest builds of FS -->
>>     <action application="export" data="t38_passthru=true"/>
>
>
> Okay, thanks for this hint, the wiki suggests proxy_media... as soon as 
> I'll get this stuff working, I'll add my experiences with my (little) 
> insights.
>
>
>> <!-- you don't mix t38_passthru with proxy media get rid of these
>> completely  -->
>
> I've done that... removed any proxy_media stuff in all profiles/dialplans.
>
> I'm still not getting anywhere... incoming fax is still working, but 
> outbound faxing still is not possible.
>
> I put up a flow-chart (by wireshark) and a dump of the relevant re-invite 
> and 200OK packets here;
>
> http://www.3js.de/20110301/
>
> As far as I can see, ports get negotiated just fine, and the fax is being 
> sent, but somehow this looks like a monolog...
> Even though it looks like the fax itself is being sent fine, there is no 
> fax received on the other side and my fax reports an error and unsent fax.
>
> PLEASE have a quick look at those, I'm quite desperate...
>
>> Maybe someday you'll be brave and try terminating the t38 right to FS 
>> instead.
>
> Hmm, if I understand this right, you would suggest to terminate the 
> outgoing fax on the FS:
>
> Upstream <(a)> FS <(b)> asterisk <(c)> SIP ATA
>
> The fax is being sent by ATA via plain Audio (c) to the asterisk box, 
> which is passing it along to FS as plain audio (b). Now you'd like me to 
> let FS send the reinvite back to the asterisk box? That means, according 
> to the wiki I'd have to do:
>
> <extension name="t38_reinvite">
>  <condition>
>    <action application="set" data="fax_enable_t38=true"/>
>    <action application="set" data="fax_enable_t38_request=true"/>
>    <action application="set" data="execute_on_answer=t38_gateway self"/>
>    <action application="bridge" data="sofia/external/1234 at host"/>
>  </condition>
> </extension>
>
> But what happens for the B-leg, going out to the my upstream? There will 
> be a t.38 reinvite as well, coming from the Upstream... won't that be a 
> problem / a more complicated system then needed? Why would you prefer this 
> over passthru?
>
>
>> BTW it seems like whatever is generating your sip trace has a bug in
>> it, there are garbage characters before each sip message unless its
>> printing unfiltered network packets or something.
>
> tcpdump -A -s0 -w file.pcap
> tcpdump -An -r file.pcap > plain.txt
> That had to be done so I could replace sensible stuff with fake data....
>
>
>
> Thanks again for your help so far!
>
>
>
> Best regards,
>
>  John
>
>
>
>
>> On Thu, Feb 24, 2011 at 4:37 PM, Johannes Jakob <jjj at 3js.de> wrote:
>>> Hi again,
>>>
>>> Just a short followup:
>>>
>>> originating a tif file directly on the FS (txfax) does work,
>>> trying to send the same tif file from the asterisk with SendFax doesn't.
>>>
>>> Full traces (sorry, just ascii, no pcap, but I had to sanitize them 
>>> somehow) of all three test cases, for those having the time and willing 
>>> to help, can be found here:
>>>
>>>        http://www.3js.de/debug.tgz
>>>
>>>
>>> According to how few people are complaining, it must be quite simple to 
>>> get t38 working... am I blind or just too stupid to see my errors?
>>>
>>>
>>> Thanks a lot for any hint in the right direction!
>>>
>>> Best regards,
>>>
>>>  John
>>>
>>>
>>>
>>>
>>> On 24.02.2011, at 17:44, Johannes Jakob wrote:
>>>
>>>> Hi,
>>>>
>>>> I'm having some serious trouble getting outbound t38 passthru to work 
>>>> in the following scenario:
>>>>
>>>> SIP Upstream <> FreeSWITCH <> Asterisk <> SIP ATA <> analog fax
>>>>
>>>> I'm not sure if it's a plain FS problem, but I have to start somewhere 
>>>> and I found at least one problem (I think) with FS's behaviour when 
>>>> negotiating with the gateway.
>>>>
>>>> Inbound T.38 faxes, when the ATA has to do the reinvite, work perfectly 
>>>> fine from one end to the other.
>>>>
>>>> Outbound faxes on the other hand, don't work at all.
>>>>
>>>> Here are the important config parts:
>>>>
>>>> dialplan before bridging to gateway:
>>>>      <action application="export" data="t38_passthru=true"/>
>>>>      <action application="set" data="proxy_media=true"/>
>>>>      <action application="set" data="bypass_media=false"/>
>>>>
>>>> directory-entry for the asterisk-registration:
>>>>      <param name="t38-passthru" value="true"/>
>>>> and
>>>>      <variable name="proxy_media" value="true"/>
>>>>      <variable name="bypass_media" value="false"/>
>>>>
>>>> and finally the gateway's sip_profile:
>>>>  <param name="t38-passthru" value="true"/>
>>>>  <param name="inbound-proxy-media" value="true"/>
>>>>
>>>> So much for introduction... When trying to send a fax to 017286483798, 
>>>> this is what happens on the SBC:
>>>>
>>>>
>>>> asterisk "client" is sending the plain audio INVITE to SBC:
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>> 9     37.668216       10.16.139.28    10.16.133.66    SIP/SDP 1202 
>>>> Request: INVITE sip:017286483798 at sbc1.mysip.net, with session 
>>>> description
>>>> AkE`b?FG^^B%INVITE sip:017286483798 at sbc1.mysip.net SIP/2.0
>>>> Via: SIP/2.0/UDP 10.16.139.28:5060;branch=z9hG4bK3c5810ec
>>>> Max-Forwards: 70
>>>> From: "0692386432" <sip:0692386432 at mysip.net>;tag=as235c17b6
>>>> To: <sip:017286483798 at sbc1.mysip.net>
>>>> Contact: <sip:0692386432 at 10.16.139.28:5060>
>>>> Call-ID: 6de3237b20a96ab010fb9b3f2436d305 at mysip.net
>>>> CSeq: 103 INVITE
>>>> User-Agent: FPBX-2.8.1(1.8.2.4)
>>>> Proxy-Authorization: Digest username="sipuser", realm="mysip.net", 
>>>> algorithm=MD5, uri="sip:017286483798 at sbc1.mysip.net", 
>>>> nonce="aa9339fc-9e53-4017-85ff-fac5367cb733", 
>>>> response="96f4dea7267d436779cd106947cc25b7", qop=auth, 
>>>> cnonce="4e110d27", nc=00000001
>>>> Date: Thu, 24 Feb 2011 07:30:27 GMT
>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
>>>> INFO, PUBLISH
>>>> Supported: replaces, timer
>>>> Content-Type: application/sdp
>>>> Content-Length: 285
>>>>
>>>> v=0
>>>> o=root 1297612317 1297612318 IN IP4 10.16.139.28
>>>> s=Asterisk PBX 1.8.2.4
>>>> c=IN IP4 10.16.139.28
>>>> t=0 0
>>>> m=audio 15000 RTP/AVP 0 8 3 101
>>>> a=rtpmap:0 PCMU/8000
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:3 GSM/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=ptime:20
>>>> a=sendrecv
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>>
>>>> SBC is forwarding the request to the correct gateway:
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>> 11    37.716704       10.16.133.66    10.15.12.215    SIP/SDP 1232 
>>>> Request: INVITE sip:+4317286483798 at 10.15.12.215, with session 
>>>> description
>>>> =|$E%@^BWB{INVITE sip:+4317286483798 at 10.15.12.215 SIP/2.0
>>>> Via: SIP/2.0/UDP 10.16.133.66:5080;rport;branch=z9hG4bK33eQ6Na9D991D
>>>> Max-Forwards: 69
>>>> From: "0692386432" <sip:0692386432 at 10.16.133.66>;tag=4NH77aQN2v7ZS
>>>> To: <sip:+4317286483798 at 10.15.12.215>
>>>> Call-ID: e5c9ab65-ba8a-122e-21ad-863d7c249180
>>>> CSeq: 8930272 INVITE
>>>> Contact: <sip:gw+gateway1 at 10.16.133.66:5080;transport=udp;gw=gateway1>
>>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7847289 2011-02-19 
>>>> 23-38-04 +0100
>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
>>>> REGISTER, REFER, NOTIFY
>>>> Supported: timer, precondition, path, replaces
>>>> Allow-Events: talk, hold, refer
>>>> Content-Type: application/sdp
>>>> Content-Disposition: session
>>>> Content-Length: 269
>>>> X-FS-Support: update_display
>>>> Remote-Party-ID: "0692386432" 
>>>> <sip:0692386432 at 10.16.133.66>;party=calling;screen=yes;privacy=off
>>>> P-Asserted-Identity: <sip:06923864320 at sip.gateway.de>
>>>>
>>>> v=0
>>>> o=FreeSWITCH 3022782041 3022782042 IN IP4 10.16.133.66
>>>> s=FreeSWITCH
>>>> c=IN IP4 10.16.133.66
>>>> t=0 0
>>>> m=audio 18446 RTP/AVP 0 8 3 101
>>>> a=rtpmap:0 PCMU/8000
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:3 GSM/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=ptime:20
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>>
>>>> Gatway says, everything OK, audio call patched through:
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>> 22    39.760159       10.15.12.215    10.16.133.66    SIP/SDP 750 
>>>> Status: 200 OK, with session description
>>>> AkEw  W^BSIP/2.0 200 OK
>>>> Via: SIP/2.0/UDP 
>>>> 10.16.133.66:5080;branch=z9hG4bK33eQ6Na9D991D;rport=5080
>>>> Call-ID: e5c9ab65-ba8a-122e-21ad-863d7c249180
>>>> From: "0692386432"<sip:0692386432 at 10.16.133.66>;tag=4NH77aQN2v7ZS
>>>> To: <sip:+4317286483798 at 10.15.12.215>;tag=h1hl6tsu-CC-39
>>>> CSeq: 8930272 INVITE
>>>> Allow: 
>>>> INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
>>>> Contact: <sip:10.15.12.215:5060;user=phone>
>>>> Content-Length: 217
>>>> Content-Type: application/sdp
>>>>
>>>> v=0
>>>> o=HuaweiSoftX3000 3014926 3014927 IN IP4 10.15.12.215
>>>> s=Sip Call
>>>> c=IN IP4 10.15.12.215
>>>> t=0 0
>>>> m=audio 15490 RTP/AVP 0 101
>>>> a=rtpmap:0 PCMU/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=ptime:20
>>>> a=fmtp:101 0-15
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>>
>>>> SBC is telling the asterisk box, everything is fine
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>> 28    39.770349       10.16.133.66    10.16.139.28    SIP/SDP 1185 
>>>> Status: 200 OK, with session description
>>>> =|$Ew at 1y^B^}aSIP/2.0 200 OK
>>>> Via: SIP/2.0/UDP 10.16.139.28:5060;branch=z9hG4bK3c5810ec
>>>> From: "0692386432" <sip:0692386432 at mysip.net>;tag=as235c17b6
>>>> To: <sip:017286483798 at sbc1.mysip.net>;tag=crS25Ur70m0BK
>>>> Call-ID: 6de3237b20a96ab010fb9b3f2436d305 at mysip.net
>>>> CSeq: 103 INVITE
>>>> Contact: <sip:017286483798 at 10.16.133.66:5060;transport=udp>
>>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7847289 2011-02-19 
>>>> 23-38-04 +0100
>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
>>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>> Supported: timer, precondition, path, replaces
>>>> Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, 
>>>> include-session-description, presence.winfo, message-summary, refer
>>>> Content-Type: application/sdp
>>>> Content-Disposition: session
>>>> Content-Length: 222
>>>> Remote-Party-ID: "017286483798" 
>>>> <sip:017286483798 at 10.16.133.66>;party=calling;privacy=off;screen=no
>>>>
>>>> v=0
>>>> o=FreeSWITCH 3022845683 3022845684 IN IP4 10.16.133.66
>>>> s=FreeSWITCH
>>>> c=IN IP4 10.16.133.66
>>>> t=0 0
>>>> m=audio 28850 RTP/AVP 0 101
>>>> a=rtpmap:0 PCMU/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-15
>>>> a=ptime:20
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>>
>>>> so, after the ACK which is passed along as well, there is the typical 
>>>> audio stuff, ringing... in both directions the right, pre-negotiated 
>>>> udp ports, everything is fine so far:
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>> 45    39.849484       10.15.12.215    10.16.133.66    RTP     216 
>>>> PT=ITU-T G.711 PCMU, SSRC=0x727E59A0, Seq=23045, Time=3032460352
>>>> User Datagram Protocol, Src Port: 15490 (15490), Dst Port: 18446 
>>>> (18446)
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>> 46    39.849541       10.16.133.66    10.16.139.28    RTP     216 
>>>> PT=ITU-T G.711 PCMU, SSRC=0x727E59A0, Seq=23045, Time=3032460352
>>>> User Datagram Protocol, Src Port: 28850 (28850), Dst Port: hydap 
>>>> (15000)
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>>
>>>> Now, the magic happens, the gateway is sending it's T.38 re-INVITE to 
>>>> establish better fax connectivity... well, let's see what happens:
>>>>
>>>> The gateway is suggesting to move the audio stuff from port 15490 to 
>>>> 15492, and instead speak t38/udptl on port 15490, keep that in mind.
>>>>
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>> 2099  50.121304       10.15.12.215    10.16.133.66    SIP/SDP 1026 
>>>> Request: INVITE 
>>>> sip:gw+gateway1 at 10.16.133.66:5080;transport=udp;gw=gateway1, in-dialog, 
>>>> with session description
>>>> AkEW^BwINVITE 
>>>> sip:gw+gateway1 at 10.16.133.66:5080;transport=udp;gw=gateway1 SIP/2.0
>>>> Via: SIP/2.0/UDP 
>>>> 10.15.12.215:5060;branch=z9hG4bK4abf7e790396e9d932f59632f
>>>> Call-ID: e5c9ab65-ba8a-122e-21ad-863d7c249180
>>>> From: <sip:+4317286483798 at 10.15.12.215>;tag=h1hl6tsu-CC-39
>>>> To: "0692386432"<sip:0692386432 at 10.16.133.66>;tag=4NH77aQN2v7ZS
>>>> CSeq: 2 INVITE
>>>> Max-Forwards: 69
>>>> Contact: <sip:10.15.12.215:5060>
>>>> Content-Length: 527
>>>> Content-Type: application/sdp
>>>>
>>>> v=0
>>>> o=HuaweiSoftX3000 3014926 3014928 IN IP4 10.15.12.215
>>>> s=Sip Call
>>>> c=IN IP4 10.15.12.215
>>>> t=0 0
>>>> m=image 15490 udptl t38
>>>> a=T38FaxVersion:0
>>>> a=T38MaxBitRate:14400
>>>> a=T38FaxRateManagement:transferredTCF
>>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>> m=audio 15492 RTP/AVP 8 0 127 103 101
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:0 PCMU/8000
>>>> a=rtpmap:127 PCMU/8000
>>>> a=gpmd:127 vbd=yes
>>>> a=rtpmap:103 PCMA/8000
>>>> a=gpmd:103 vbd=yes
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=ptime:20
>>>> a=silenceSupp:off - - - -
>>>> a=ecan:fb on -
>>>> a=X-fax
>>>> a=fmtp:101 0-15
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>>
>>>> FS is handing this INVITE to asterisk (and tells asterisk, it would 
>>>> accept audio and/or t.38, both on the same port 28850, don't think 
>>>> that's a problem, but it's at least different from HuaweiSoftX3000's 
>>>> behavior):
>>>>
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>> 2101  50.122833       10.16.133.66    10.16.139.28    SIP/SDP 1398 
>>>> Request: INVITE sip:0692386432 at 10.16.139.28:5060, in-dialog, with 
>>>> session description
>>>> =|$Efw at 0^B^R6INVITE sip:0692386432 at 10.16.139.28:5060 SIP/2.0
>>>> Via: SIP/2.0/UDP 10.16.133.66;rport;branch=z9hG4bKy5perv1F90acS
>>>> Max-Forwards: 70
>>>> From: <sip:017286483798 at sbc1.mysip.net>;tag=crS25Ur70m0BK
>>>> To: "0692386432" <sip:0692386432 at mysip.net>;tag=as235c17b6
>>>> Call-ID: 6de3237b20a96ab010fb9b3f2436d305 at mysip.net
>>>> CSeq: 8930278 INVITE
>>>> Contact: <sip:017286483798 at 10.16.133.66:5060;transport=udp>
>>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7847289 2011-02-19 
>>>> 23-38-04 +0100
>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
>>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>> Supported: timer, precondition, path, replaces
>>>> Content-Type: application/sdp
>>>> Content-Disposition: session
>>>> Content-Length: 532
>>>> X-FS-Support: update_display
>>>> P-Asserted-Identity: <sip:06923864320 at sip.gateway.de>
>>>>
>>>> v=0
>>>> o=FreeSWITCH 3022845683 3022845685 IN IP4 10.16.133.66
>>>> s=FreeSWITCH
>>>> c=IN IP4 10.16.133.66
>>>> t=0 0
>>>> m=image 28850 udptl t38
>>>> a=T38FaxVersion:0
>>>> a=T38MaxBitRate:14400
>>>> a=T38FaxRateManagement:transferredTCF
>>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>> m=audio 28850 RTP/AVP 8 0 127 103 101
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:0 PCMU/8000
>>>> a=rtpmap:127 PCMU/8000
>>>> a=rtpmap:103 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-15
>>>> a=gpmd:127 vbd=yes
>>>> a=gpmd:103 vbd=yes
>>>> a=ptime:20
>>>> a=silenceSupp:off - - - -
>>>> a=ecan:fb on -
>>>> a=X-fax
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>>
>>>> The asterisk box says this is fine (after of course successfully 
>>>> talking to the ATA, which is fine with it, too, but want's to have 
>>>> slower speed). The asterisk box is also changing the port it wants to 
>>>> get t38 data on from 15508 to 4676 and finally sets the udp port for 
>>>> audio 0 to disable it.
>>>> Just plain t.38 in the new SDP description:
>>>>
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>> 2107  50.165366       10.16.139.28    10.16.133.66    SIP/SDP 920 
>>>> Status: 200 OK, with session description
>>>> AkE`b?G^^^BtAbSIP/2.0 200 OK
>>>> Via: SIP/2.0/UDP 
>>>> 10.16.133.66;branch=z9hG4bKy5perv1F90acS;received=10.16.133.66;rport=5060
>>>> From: <sip:017286483798 at sbc1.mysip.net>;tag=crS25Ur70m0BK
>>>> To: "0692386432" <sip:0692386432 at mysip.net>;tag=as235c17b6
>>>> Call-ID: 6de3237b20a96ab010fb9b3f2436d305 at mysip.net
>>>> CSeq: 8930278 INVITE
>>>> Server: FPBX-2.8.1(1.8.2.4)
>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
>>>> INFO, PUBLISH
>>>> Supported: replaces, timer
>>>> Contact: <sip:0692386432 at 10.16.139.28:5060>
>>>> Content-Type: application/sdp
>>>> Content-Length: 307
>>>>
>>>> v=0
>>>> o=root 1297612317 1297612319 IN IP4 10.16.139.28
>>>> s=Asterisk PBX 1.8.2.4
>>>> c=IN IP4 10.16.139.28
>>>> t=0 0
>>>> m=audio 0 RTP/AVP 8 0 127 103 101
>>>> m=image 4676 udptl t38
>>>> a=T38FaxVersion:0
>>>> a=T38MaxBitRate:9600
>>>> a=T38FaxRateManagement:transferredTCF
>>>> a=T38FaxMaxDatagram:397
>>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>>
>>>> BUT look at this! What does FreeSWITCH tell the gateway???
>>>> It sends 200 OK, but suddenly wants to receive only audio data and 
>>>> disables comfort noise?
>>>> Same udp port as before, but no sign of t.38 in the SDP description!
>>>>
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>> 2111  50.170760       10.16.133.66    10.15.12.215    SIP/SDP 902 
>>>> Status: 200 OK, with session description
>>>> =|$Ev%@^BWbA1SIP/2.0 200 OK
>>>> Via: SIP/2.0/UDP 
>>>> 10.15.12.215:5060;branch=z9hG4bK4abf7e790396e9d932f59632f
>>>> From: <sip:+4317286483798 at 10.15.12.215>;tag=h1hl6tsu-CC-39
>>>> To: "0692386432" <sip:0692386432 at 10.16.133.66>;tag=4NH77aQN2v7ZS
>>>> Call-ID: e5c9ab65-ba8a-122e-21ad-863d7c249180
>>>> CSeq: 2 INVITE
>>>> Contact: <sip:gw+gateway1 at 10.16.133.66:5080;transport=udp;gw=gateway1>
>>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7847289 2011-02-19 
>>>> 23-38-04 +0100
>>>> Accept: application/sdp
>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
>>>> REGISTER, REFER, NOTIFY
>>>> Supported: timer, precondition, path, replaces
>>>> Content-Type: application/sdp
>>>> Content-Disposition: session
>>>> Content-Length: 168
>>>>
>>>> v=0
>>>> o=FreeSWITCH 3022782041 3022782043 IN IP4 10.16.133.66
>>>> s=FreeSWITCH
>>>> c=IN IP4 10.16.133.66
>>>> t=0 0
>>>> m=audio 18446 RTP/AVP 8 0 127 103 101
>>>> m=audio 0 RTP/AVP 19
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>>
>>>> and look at this, even though we just told the gateway to only talk 
>>>> audio to it, we send a t38 packet! (it's this lonely one though!)
>>>>
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>> 2112  50.174592       94.186.133.66   87.234.1.215    T.38    216 
>>>> UDP: UDPTLPacket Seq=32768  t30ind: <unknown>[UNKNOWN PER: 
>>>> 10.9.3.8.1][Malformed Packet]
>>>> User Datagram Protocol, Src Port: 18446 (18446), Dst Port: 15490 
>>>> (15490)
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>>
>>>>
>>>> The gateway keeps sending normal audio to us on the specified and 
>>>> unchanged port, BUT from the udp port it originally told us it would 
>>>> only accept t.38 on...
>>>>
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>> 2260  51.589980       10.15.12.215    10.16.133.66    RTP     216 
>>>> PT=ITU-T G.711 PCMA, SSRC=0x727E59A0, Seq=23634, Time=3032554272
>>>> User Datagram Protocol, Src Port: 15490 (15490), Dst Port: 18446 
>>>> (18446)
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>>
>>>> While the gateway is sending us plain audio, we are talking t38 to  the 
>>>> asterisk box (which is not responding).
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>> 2261  51.590063       10.16.133.66    10.16.139.28    T.38    216 
>>>> UDP: UDPTLPacket Seq=32776  data:v8:[UNKNOWN PER: too long 
>>>> integer(per_integer)][Malformed Packet]
>>>> User Datagram Protocol, Src Port: 28850 (28850), Dst Port: dhct-alerts 
>>>> (4676)
>>>> -----------------------------------------------------------------------------------------------------------------------
>>>>
>>>>
>>>>
>>>> For the record, on the asterisk (version 1.8.2.4) box I defined
>>>> t38pt_udptl=yes,redundancy
>>>> directmedia=no
>>>> for the gateways and the ATA's extension.
>>>>
>>>> On both, the sbc and the asterisk box I compiled res_fax_spandsp, 
>>>> mod_spandsp with spandsp-0.0.6pre18.
>>>>
>>>> So... I've been spending far tooooo much time debugging this and I'm 
>>>> quite sure I'm just too stupid to find a solution for this.
>>>>
>>>>
>>>> Is there any good pcap anonymizing utitlity, that can substitute 
>>>> application layer stuff as well?
>>>>
>>>> Well, *any* help/hint would be appreciated very much ;)
>>>>
>>>> Thanks in advance,
>>>>
>>>>  John
>>>
>>>
>>> _______________________________________________
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>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>> 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From Nabble at slickdeals.endjunk.com  Tue Mar  1 19:45:53 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 1 Mar 2011 08:45:53 -0800 (PST)
Subject: [Freeswitch-users] freeswitch -nc -nosql
In-Reply-To: <AANLkTimxG==dquT_oeFSiPnxSN6g5mncQ5Gf1GNg_u6U@mail.gmail.com>
References: <1298688216720-6066718.post@n2.nabble.com>
	<AANLkTikjkTPk3A2FmoRo32WhKxNFaX5YL7Wxezaa6zqG@mail.gmail.com>
	<1298815506369-6070268.post@n2.nabble.com>
	<AANLkTin8ZRoFSzFoiaxNKg0VDvXi5YebMe_Z6mpyFRgJ@mail.gmail.com>
	<1298820079407-6070409.post@n2.nabble.com>
	<1298864829635-6071920.post@n2.nabble.com>
	<AANLkTimxG==dquT_oeFSiPnxSN6g5mncQ5Gf1GNg_u6U@mail.gmail.com>
Message-ID: <1298997953584-6077875.post@n2.nabble.com>


Anthony Minessale wrote:
> This means you need the sql enabled.Exactly. However, if I SQL enabled, FS
> crashes.


The entire core DB will be disabled with -nosql.
> This includes the "show" command, aliases, help, and several other
> things in the CORE.
> 
> We probably will remove this option soon because it's hard to live
> with no SQL DB in FreeSWITCH.I do agree with you that no SQL DB in FS is a
> hard time! This means no more FS for me once -nosql switch is removed. 


-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-nc-nosql-tp6066718p6077875.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Tue Mar  1 20:02:40 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 1 Mar 2011 11:02:40 -0600
Subject: [Freeswitch-users] freeswitch -nc -nosql
In-Reply-To: <1298997953584-6077875.post@n2.nabble.com>
References: <1298688216720-6066718.post@n2.nabble.com>
	<AANLkTikjkTPk3A2FmoRo32WhKxNFaX5YL7Wxezaa6zqG@mail.gmail.com>
	<1298815506369-6070268.post@n2.nabble.com>
	<AANLkTin8ZRoFSzFoiaxNKg0VDvXi5YebMe_Z6mpyFRgJ@mail.gmail.com>
	<1298820079407-6070409.post@n2.nabble.com>
	<1298864829635-6071920.post@n2.nabble.com>
	<AANLkTimxG==dquT_oeFSiPnxSN6g5mncQ5Gf1GNg_u6U@mail.gmail.com>
	<1298997953584-6077875.post@n2.nabble.com>
Message-ID: <AANLkTimuQDPXQQHLvroxd7ttWYDZxiPAxv4At_MGBqHO@mail.gmail.com>

You said it doesn't crash anymore in a previous email didn't you?
I can't keep track of 200 people's issues via email.

Did you update to HEAD and reproduce it again?

Please.... Stop reporting bugs on the mailing list.  The mailing list
is NOT for bug reports its for discussion.
I can understand someone may want to discuss if something is a bug or
not first but a segfault is always a bug, it's up to the JIRA process
to decide if its something we can fix or if there is a solution.

Update to the very current right now GIT HEAD, reproduce it, and open
a bug on JIRA and attach the relevant info.



On Tue, Mar 1, 2011 at 10:45 AM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
> Anthony Minessale wrote:
>> This means you need the sql enabled.Exactly. However, if I SQL enabled, FS
>> crashes.
>
>
> The entire core DB will be disabled with -nosql.
>> This includes the "show" command, aliases, help, and several other
>> things in the CORE.
>>
>> We probably will remove this option soon because it's hard to live
>> with no SQL DB in FreeSWITCH.I do agree with you that no SQL DB in FS is a
>> hard time! This means no more FS for me once -nosql switch is removed.
>
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-nc-nosql-tp6066718p6077875.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 ayhkor at gmail.com  Tue Mar  1 20:02:53 2011
From: ayhkor at gmail.com (deniro)
Date: Tue, 1 Mar 2011 12:02:53 -0500
Subject: [Freeswitch-users] fs_cli socket connection error
In-Reply-To: <AANLkTi=QXgEb6dHU9bfb7v06WzPyWJcoNYDLi1X9oq+E@mail.gmail.com>
References: <AANLkTikCoT9dzLNoD=6TzS0jiMw4afNX94TypsLarBbz@mail.gmail.com>
	<AANLkTi=sENgBB9Cdr-EkLJD8XfmH3A8GSfPnvDHO7qrw@mail.gmail.com>
	<AANLkTin5k=2Z-=QVVzqt91172wv88P7OFkqRqTY9VDcS@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58ADB727D7@cooper>
	<AANLkTi=aPwmnVD8ZzaBVFDxGyPsbgzDEBGaXvq_B2NbF@mail.gmail.com>
	<AANLkTikpQuxvyaU7PAGnyJuoQLQ9xn4inNJ_u5x41Anv@mail.gmail.com>
	<AANLkTimNYjC_G-Lq+St5VePpxEyM8iHnhaNCo=jBydi9@mail.gmail.com>
	<AANLkTi=QXgEb6dHU9bfb7v06WzPyWJcoNYDLi1X9oq+E@mail.gmail.com>
Message-ID: <AANLkTikEDgpAJppNUnLac73ip7qsk+8AK-N-pTW7NBm6@mail.gmail.com>

Thank you all for advice  and direction
Will re-install from scratch at a later time.
deniro--

On Tue, Mar 1, 2011 at 4:16 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Ok, you really need to pay attention to any log message that logs with
> the CRIT (critical) log level - they're really serious errors. You
> should look at adding the critical="true" attribute to all the tags in
> modules.conf.xml which you want to be sure are loaded. If a module
> fails to load without that tag FS will continue to run anyway, with
> that tag it'll refuse to start so you'll know instantly something is
> wrong.
>
> Most of your modules aren't loading, because they haven't been
> compiled correctly. Some look like they're missing (no such file or
> directory) and others haven't been built right (undefined symbols).
>
> Here are a few of the important lines from your log:
>
> 2011-02-28 18:12:21.910738 [CRIT] switch_loadable_module.c:882 Error
> Loading module /opt/freeswitch/mod/mod_event_socket.so
> **/opt/freeswitch/mod/mod_event_socket.so: undefined symbol:
> switch_event_serialize_json**
> 2011-02-28 18:12:21.910789 [CRIT] switch_loadable_module.c:882 Error
> Loading module /opt/freeswitch/mod/mod_xml_curl.so
> **/opt/freeswitch/mod/mod_xml_curl.so: cannot open shared object file:
> No such file or directory**
> 2011-02-28 18:12:21.910813 [CRIT] switch_loadable_module.c:882 Error
> Loading module /opt/freeswitch/mod/mod_xml_cdr.so
> **/opt/freeswitch/mod/mod_xml_cdr.so: cannot open shared object file:
> No such file or directory**
> 2011-02-28 18:12:21.911581 [CRIT] switch_loadable_module.c:882 Error
> Loading module /opt/freeswitch/mod/mod_commands.so
> **/opt/freeswitch/mod/mod_commands.so: undefined symbol:
> switch_xml_reload**
> 2011-02-28 18:12:21.911709 [CRIT] switch_loadable_module.c:882 Error
> Loading module /opt/freeswitch/mod/mod_conference.so
> **/opt/freeswitch/mod/mod_conference.so: undefined symbol:
> switch_channel_test_app_flag_key**
> 2011-02-28 18:12:21.911842 [CRIT] switch_loadable_module.c:882 Error
> Loading module /opt/freeswitch/mod/mod_dptools.so
> **/opt/freeswitch/mod/mod_dptools.so: undefined symbol:
> switch_event_merge**
> 2011-02-28 18:12:21.912264 [CRIT] switch_loadable_module.c:882 Error
> Loading module /opt/freeswitch/mod/mod_dialplan_xml.so
> **/opt/freeswitch/mod/mod_dialplan_xml.so: undefined symbol:
> switch_xml_std_datetime_check**
> 2011-02-28 18:12:21.912531 [CRIT] switch_loadable_module.c:882 Error
> Loading module /opt/freeswitch/mod/mod_g723_1.so
> **/opt/freeswitch/mod/mod_g723_1.so: cannot open shared object file:
> No such file or directory**
> 2011-02-28 18:12:21.912573 [CRIT] switch_loadable_module.c:882 Error
> Loading module /opt/freeswitch/mod/mod_g729.so
> **/opt/freeswitch/mod/mod_g729.so: cannot open shared object file: No
> such file or directory**
> 2011-02-28 18:12:21.912609 [CRIT] switch_loadable_module.c:882 Error
> Loading module /opt/freeswitch/mod/mod_amr.so
> **/opt/freeswitch/mod/mod_amr.so: cannot open shared object file: No
> such file or directory**
>
> Important things to note:
> mod_event_socket hasn't loaded because of a undefined symbol, it
> wasn't built correctly
> mod_xml_curl hasn't loaded because it doesn't exist, it was either
> never built or never installed
>
> Something pretty strange has happened. Did you recompile when trying
> to add mod_xml_curl?
>
> I'd suggest you delete all the FS files, including the Git clone, make
> a fresh checkout and build it from scratch.
>
> -Steve
>
>
>
> On 28 February 2011 23:18, deniro <ayhkor at gmail.com> wrote:
> > I put the log in pastebin
> > freeswitch.log only when starting freeswitch (after stop)
> >
> > http://pastebin.freeswitch.org/15502
> > thx
> > deniro
> > On Mon, Feb 28, 2011 at 5:46 PM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >>
> >> You must check the logfile for when FS starts up - if netstat shows
> >> nothing for port 8021 then either mod_event_socket isn't being loaded
> >> or you'll see an error when it tries to load.
> >>
> >> Nothing from netstat means nothing's listening, so trying to connect
> >> using fs_cli won't do anything.
> >>
> >> -Steve
> >>
> >>
> >> On 28 February 2011 22:40, deniro <ayhkor at gmail.com> wrote:
> >> > Checked the log
> >> > Nothing is logged when
> >> > fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
> >> > is issued
> >> > /opt/freeswitch/log# tail -f freeswitch.log
> >> >
> >> > ls -l freeswitch.log
> >> > -rwxrwxrwx 1 freeswitch daemon 33018 2011-02-28 17:39 freeswitch.log
> >> > thx
> >> >
> >> >
> >> >
> >> > On Mon, Feb 28, 2011 at 3:59 PM, Peter Olsson
> >> > <peter.olsson at visionutveckling.se> wrote:
> >> >>
> >> >> Check freeswitch.log, it probably reports some problem when loading
> the
> >> >> mod_event_socket module.
> >> >>
> >> >> /Peter
> >> >> ________________________________________
> >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> >> >> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r deniro
> >> >> [ayhkor at gmail.com]
> >> >> Skickat: den 28 februari 2011 21:42
> >> >> Till: FreeSWITCH Users Help
> >> >> ?mne: Re: [Freeswitch-users] fs_cli socket connection error
> >> >>
> >> >> Steve
> >> >> Thanks for the reply
> >> >> netstat -anlp | grep 8021  is blank (nothing showing up)
> >> >>
> >> >> /opt/freeswitch/bin/fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
> >> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
> >> >> Configuration
> >> >> file is /root/.fs_cli_conf.
> >> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
> >> >> Configuration
> >> >> file is /etc/fs_cli.conf.
> >> >> [DEBUG] libs/esl/fs_cli.c:1159 main() profile default does not exist
> >> >> using
> >> >> builtin profile
> >> >> [DEBUG] libs/esl/fs_cli.c:1188 main() Using profile internal
> >> >> [75.xxx.xxx.xxx]
> >> >> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
> >> >> Connection
> >> >> Error]
> >> >> yes mod_event_socket is after  mod_xml_curl but I changed the order
> in
> >> >>  modules.conf.xml
> >> >> still getting above (restarted freeswitch)
> >> >>
> >> >> thx
> >> >> deniro--
> >> >>
> >> >>
> >> >>
> >> >> On Mon, Feb 28, 2011 at 3:35 AM, Steven Ayre
> >> >> <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
> >> >> Enable debug logging and you should see an error that'll tell you
> more.
> >> >>
> >> >> Is mod_event_socket loading before or after mod_xml_curl? Chances are
> >> >> mod_xml_curl is loading first, so it's trying to read
> >> >> event_socket.conf.xml and/or acl.conf.xml through xml_curl and either
> >> >> getting a different config to your previous local copy or the ACLs
> are
> >> >> different.
> >> >>
> >> >> What does "netstat -anlp | grep :8021" from the linux cli show you?
> >> >> Does it show that freeswitch is actually listening on the port? If it
> >> >> is it's probably an ACL problem, if it isn't then it's probably a
> >> >> problem with event_socket.conf.xml
> >> >>
> >> >> -Steve
> >> >>
> >> >> On 28 February 2011 00:53, deniro
> >> >> <ayhkor at gmail.com<mailto:ayhkor at gmail.com>> wrote:
> >> >> > What would be possible reasons for this and how to resolve?
> >> >> > running fs 106 on ubuntu 10.04 server
> >> >> >
> >> >> > /opt/freeswitch/bin/fs_cli -H xxx.xxx.xxx.xxx -P 8021
> >> >> > [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
> >> >> > Connection
> >> >> > Error]
> >> >> > was working fine before I installed  mod_xml_cdr
> >> >> > configure --prefix=/opt/freeswitch --without-libcurl
> >> >> > make mod_xml_cdr-install
> >> >> > (no errors)
> >> >> >
> >> >> > in modules.conf.xml
> >> >> > <load module="mod_event_socket"/>
> >> >> >
> >> >> > is there a packaged version of mod_xml_cdr (for fs 1.0.6)?
> >> >> >
> >> >> > thx
> >> >> > deniro--
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> >
> >> >> >
> >> >> > FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org<http://www.freeswitch.org/>
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >>
> >> >>
> >> >> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org<http://www.freeswitch.org/>
> >> >>
> >> >> !DSPAM:4d6c0a1132761029518849!
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jjj at 3js.de  Tue Mar  1 20:09:12 2011
From: jjj at 3js.de (Johannes Jakob)
Date: Tue, 1 Mar 2011 18:09:12 +0100
Subject: [Freeswitch-users] T.38 Issues with passthrough handshaking
In-Reply-To: <1951319FA2234F89920E93B69BAC5351@e1705>
References: <640F91CF-9489-4F74-96B1-A6398CD84F07@3js.de><A53FD340-169C-4DD4-8AF2-C463F9D9E3DD@3js.de><AANLkTi=GmF93TkQPNuFf9ko_eQVgvqwujZOo8duN8ZDB@mail.gmail.com>
	<64786198-FF65-49FA-BCC6-57337A412D37@3js.de>
	<1951319FA2234F89920E93B69BAC5351@e1705>
Message-ID: <7FD8D98C-266D-4CA4-8342-E6B953B9DEDC@3js.de>

>>> Okay, thanks for this hint, the wiki suggests proxy_media...
> Johannes,
> Anthony is one of the main developer of FS,
> so you can trust him at 100%

hehe, thank you, but I'm fully aware of who Anthony is and I *do* in fact fully trust him in all things FreeSWITCH ;) sorry if my badly chosen words imply I wouldn't!

All I wanted to say is, that the setup I had done and described in my first posting has been done after reading Anthony's, Michael's and Darren's book and lots and lots of wiki pages and mailing list postings. In the past it seems to have been necessary to use proxy_media and disable bypass manually, so I had done that, not knowing this is deprecated.

So, if anybody of you FreeSWITCH wizards has ANY idea on how to get my setup working, I'd be very very *very* happy....

I love FreeSWITCH for it's clean and organized structure, it's flexibility and speed.... but t.38 drives me crazy... and that's of course NOT FreeSWITCH's fault!


Thanks to everybody who has time and willing to read my debugging stuff!

And of course sorry to Anthony, if I by any means disrespected you with my bad english! ;)

Greetings,
  John

From infos at madovsky.org  Tue Mar  1 20:18:14 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 1 Mar 2011 12:18:14 -0500
Subject: [Freeswitch-users] T.38 Issues with passthrough handshaking
References: <640F91CF-9489-4F74-96B1-A6398CD84F07@3js.de><A53FD340-169C-4DD4-8AF2-C463F9D9E3DD@3js.de><AANLkTi=GmF93TkQPNuFf9ko_eQVgvqwujZOo8duN8ZDB@mail.gmail.com><64786198-FF65-49FA-BCC6-57337A412D37@3js.de><1951319FA2234F89920E93B69BAC5351@e1705>
	<7FD8D98C-266D-4CA4-8342-E6B953B9DEDC@3js.de>
Message-ID: <00DD4259DF544C7390702EF7B024CE11@e1705>

since the implementation of T38 you should
set proxy_media to false and bypass_media to false also....

----- Original Message ----- 
From: "Johannes Jakob" <jjj at 3js.de>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, March 01, 2011 12:09 PM
Subject: Re: [Freeswitch-users] T.38 Issues with passthrough handshaking


>>>> Okay, thanks for this hint, the wiki suggests proxy_media...
>> Johannes,
>> Anthony is one of the main developer of FS,
>> so you can trust him at 100%
>
> hehe, thank you, but I'm fully aware of who Anthony is and I *do* in fact 
> fully trust him in all things FreeSWITCH ;) sorry if my badly chosen words 
> imply I wouldn't!
>
> All I wanted to say is, that the setup I had done and described in my 
> first posting has been done after reading Anthony's, Michael's and 
> Darren's book and lots and lots of wiki pages and mailing list postings. 
> In the past it seems to have been necessary to use proxy_media and disable 
> bypass manually, so I had done that, not knowing this is deprecated.
>
> So, if anybody of you FreeSWITCH wizards has ANY idea on how to get my 
> setup working, I'd be very very *very* happy....
>
> I love FreeSWITCH for it's clean and organized structure, it's flexibility 
> and speed.... but t.38 drives me crazy... and that's of course NOT 
> FreeSWITCH's fault!
>
>
> Thanks to everybody who has time and willing to read my debugging stuff!
>
> And of course sorry to Anthony, if I by any means disrespected you with my 
> bad english! ;)
>
> Greetings,
>  John
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From anthony.minessale at gmail.com  Tue Mar  1 20:18:29 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 1 Mar 2011 11:18:29 -0600
Subject: [Freeswitch-users] [ANNOUNCEMENT] EVERYONE PLEASE READ
Message-ID: <AANLkTi=rODUOed8dyo41au8Rf2a6UQkn+rPr2y=7Ru3K@mail.gmail.com>

Hi,

I just want to give a reminder that we have a growing community and we
want to put our resources to their best use.

Here are some important things to consider.  Many of you are already
doing these things so if you already are, thank you, please help
encourage others to follow your example.

1) We have a bug/issue tracker site using JIRA: http://jira.freeswitch.org
This site is for reporting and discussing issues.
Please use this tool to keep track of issues you have.  Even if it's
not a bug we can easily close it tagged "not a bug".
Also please help out when you can by reviewing this site to see if you
can provide answers to other people.

2) If you see someone complain about the WIKI please remind them that
anyone can edit the WIKI and if it's wrong we can clean it up. It
takes longer to complain about the wiki than it does to edit it to be
correct or add the missing info.  That is the idea behind wikis.

3) Please take the time to provide answers to any questions you may
know when perusing the list.  The more people who help the less work
it is to keep everyone informed.

4) Take the time to test the latest development build.  Its better to
find bugs before we release than after.  People tend to anchor
themselves to the "official" releases and never try the BETA versions.
 The sad truth is the tagged release will begin to build a large list
of known bugs starting 30 seconds after its tagged.  You don't have to
run HEAD in production every morning but you should stay up to speed
with what's going on with the software and help to produce a stable
release that truly is stable.

5) We will be branching 1.0 so it will slow down and taper off from
upstream patches from HEAD.  Anyone who wants to help maintain that
via JIRA and GIT should let us know.  Once we branch, most of our
attention will still be focused on HEAD and only critical patches or
bug fixes will be shared between 1.0 and 1.2 and we will rely on those
interested in slower paced development to help keep that code base in
order.











-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Tue Mar  1 20:30:56 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 1 Mar 2011 11:30:56 -0600
Subject: [Freeswitch-users] T.38 Issues with passthrough handshaking
In-Reply-To: <00DD4259DF544C7390702EF7B024CE11@e1705>
References: <640F91CF-9489-4F74-96B1-A6398CD84F07@3js.de>
	<A53FD340-169C-4DD4-8AF2-C463F9D9E3DD@3js.de>
	<AANLkTi=GmF93TkQPNuFf9ko_eQVgvqwujZOo8duN8ZDB@mail.gmail.com>
	<64786198-FF65-49FA-BCC6-57337A412D37@3js.de>
	<1951319FA2234F89920E93B69BAC5351@e1705>
	<7FD8D98C-266D-4CA4-8342-E6B953B9DEDC@3js.de>
	<00DD4259DF544C7390702EF7B024CE11@e1705>
Message-ID: <AANLkTikDSKqf0TeeWjXUzAd3KLqsw7_wK1TwJ3sOZmZf@mail.gmail.com>

You need to get traces from the FreeSWITCH console more than a pcap.

Execute these commands on the cli: (don't forget any I often have to
tell people to take traces more than once)

sofia global siptrace on
console loglevel debug

Reproduce the problem situation then paste that log.
Every new time something changes, repeat this step.


On Tue, Mar 1, 2011 at 11:18 AM, Madovsky <infos at madovsky.org> wrote:
> since the implementation of T38 you should
> set proxy_media to false and bypass_media to false also....
>
> ----- Original Message -----
> From: "Johannes Jakob" <jjj at 3js.de>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, March 01, 2011 12:09 PM
> Subject: Re: [Freeswitch-users] T.38 Issues with passthrough handshaking
>
>
>>>>> Okay, thanks for this hint, the wiki suggests proxy_media...
>>> Johannes,
>>> Anthony is one of the main developer of FS,
>>> so you can trust him at 100%
>>
>> hehe, thank you, but I'm fully aware of who Anthony is and I *do* in fact
>> fully trust him in all things FreeSWITCH ;) sorry if my badly chosen words
>> imply I wouldn't!
>>
>> All I wanted to say is, that the setup I had done and described in my
>> first posting has been done after reading Anthony's, Michael's and
>> Darren's book and lots and lots of wiki pages and mailing list postings.
>> In the past it seems to have been necessary to use proxy_media and disable
>> bypass manually, so I had done that, not knowing this is deprecated.
>>
>> So, if anybody of you FreeSWITCH wizards has ANY idea on how to get my
>> setup working, I'd be very very *very* happy....
>>
>> I love FreeSWITCH for it's clean and organized structure, it's flexibility
>> and speed.... but t.38 drives me crazy... and that's of course NOT
>> FreeSWITCH's fault!
>>
>>
>> Thanks to everybody who has time and willing to read my debugging stuff!
>>
>> And of course sorry to Anthony, if I by any means disrespected you with my
>> bad english! ;)
>>
>> Greetings,
>> ?John
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 jjj at 3js.de  Tue Mar  1 20:36:16 2011
From: jjj at 3js.de (Johannes Jakob)
Date: Tue, 1 Mar 2011 18:36:16 +0100
Subject: [Freeswitch-users] T.38 Issues with passthrough handshaking
In-Reply-To: <00DD4259DF544C7390702EF7B024CE11@e1705>
References: <640F91CF-9489-4F74-96B1-A6398CD84F07@3js.de><A53FD340-169C-4DD4-8AF2-C463F9D9E3DD@3js.de><AANLkTi=GmF93TkQPNuFf9ko_eQVgvqwujZOo8duN8ZDB@mail.gmail.com><64786198-FF65-49FA-BCC6-57337A412D37@3js.de><1951319FA2234F89920E93B69BAC5351@e1705>
	<7FD8D98C-266D-4CA4-8342-E6B953B9DEDC@3js.de>
	<00DD4259DF544C7390702EF7B024CE11@e1705>
Message-ID: <FFC7F56A-8627-4655-83FD-F0D276785767@3js.de>

On 01.03.2011, at 18:18, Madovsky wrote:

> since the implementation of T38 you should
> set proxy_media to false and bypass_media to false also....


Okay, didn't know that, but do since Anthony told me.
But as I said, I removed all of the proxy stuff from my xml config, but T.38 passthru still doesn't work the way I believe it has to ;)




> ----- Original Message ----- 
> From: "Johannes Jakob" <jjj at 3js.de>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, March 01, 2011 12:09 PM
> Subject: Re: [Freeswitch-users] T.38 Issues with passthrough handshaking
> 
> 
>>>>> Okay, thanks for this hint, the wiki suggests proxy_media...
>>> Johannes,
>>> Anthony is one of the main developer of FS,
>>> so you can trust him at 100%
>> 
>> hehe, thank you, but I'm fully aware of who Anthony is and I *do* in fact 
>> fully trust him in all things FreeSWITCH ;) sorry if my badly chosen words 
>> imply I wouldn't!
>> 
>> All I wanted to say is, that the setup I had done and described in my 
>> first posting has been done after reading Anthony's, Michael's and 
>> Darren's book and lots and lots of wiki pages and mailing list postings. 
>> In the past it seems to have been necessary to use proxy_media and disable 
>> bypass manually, so I had done that, not knowing this is deprecated.
>> 
>> So, if anybody of you FreeSWITCH wizards has ANY idea on how to get my 
>> setup working, I'd be very very *very* happy....
>> 
>> I love FreeSWITCH for it's clean and organized structure, it's flexibility 
>> and speed.... but t.38 drives me crazy... and that's of course NOT 
>> FreeSWITCH's fault!
>> 
>> 
>> Thanks to everybody who has time and willing to read my debugging stuff!
>> 
>> And of course sorry to Anthony, if I by any means disrespected you with my 
>> bad english! ;)
>> 
>> Greetings,
>> John
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From ayhkor at gmail.com  Tue Mar  1 20:53:48 2011
From: ayhkor at gmail.com (deniro)
Date: Tue, 1 Mar 2011 12:53:48 -0500
Subject: [Freeswitch-users] conference dialing and montly billing
In-Reply-To: <4D6AFF45.50108@communicatefreely.net>
References: <AANLkTinY_-7zPySJMQ915hKY0RTHZtycXoszw3K3LE0u@mail.gmail.com>
	<4D6AFF45.50108@communicatefreely.net>
Message-ID: <AANLkTintF2OWr__D=0N-ViQWxwHeURRib_1W8NNegEX6@mail.gmail.com>

Thanks Tim for your direction
Will study and try this. I am relatively new to this stuff and some
concepts still ambigious in my mind.
At this time will re-install freeswitch from scracth as I am having
problems  instaling and loading mod_xml_cdr , mod_xml_curl on my existing
setup.
deniro--

On Sun, Feb 27, 2011 at 8:49 PM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:

> HI Deniro,
>
> This is a pretty simple scenario.  I can't build it for you (too busy
> doing my own!), but here's what you need:
>
> A basic, working FreeSWITCH installation that takes care of routing the
> calls, asking for the PIN, setting up the conferences.  You can do the
> PIN IVR in lua pretty easily.  All it has to do is set a variable that
> contains your user account number, as well as any other details that are
> billing specific (DID number call came in on, toll-free vs. local, etc.)
>
> Use xml_curl_cdr to post the call records to a web server.
>
> Using a PHP script, or another language that you are comfortable with,
> parse the XML record that is posted for things like:
> -User ID (set as a variable)
> -Call Start
> -Call duration
> -Conference room
>
> Insert these values into your favorite database, along with a cost
> column that gets calculated by the script, based on the duration and
> other relevant parameters.
>
> Have another php script that runs monthly, rendering the database call
> records as a nice looking PDF invoice.  You may want to have another
> database table that keeps track of monthly invoices and payments to
> track balance due, etc.
>
> Most of the work is done externally by web server scripts that manage
> the billing data.  Freeswitch just has to ask the caller the right
> questions, and put them in the right room.
>
> You may also want to use xml_curl to dynamically generate dialplan as
> well as the conference config XML so that you can have custom
> per-conference settings, as well as easily manage your users by updating
> their information in your database.
>
> Good luck!
>
> -Tim
>
> deniro wrote:
> > Hi All
> > I would like to write some type of billing program  that will collect
> > the charges for each account monthly
> >
> > When someone calls into the  (freeswitch) conference by dialing  toll
> > free number or local number and enters PIN number
> > the  program will recognize that and start collecting number of
> > minutes and number of  persons dialed in
> > and calculate  the  amount of  dollars.
> >
> > Lets say conference is  10cent/per minute /per person for a tool free
> > number
> > Each time people  dial into conference it will calculate total amount
> > by person and by minutes, and generate monthly billing.
> >
> > The PIN numbers may be different  for conferences that belong to same
> > account.
> >
> > How would I do such thing? Where do I start from?
> > Is there any sample programs like it somewhere out there
> > Which language would be best to it with
> >
> > I would pay for any professional services
> >
> > Thanks in advance
> > deniro--
> >
> >
> > ------------------------------------------------------------------------
>  >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dftoro at yahoo.com  Tue Mar  1 17:04:22 2011
From: dftoro at yahoo.com (Diego Toro)
Date: Tue, 1 Mar 2011 06:04:22 -0800 (PST)
Subject: [Freeswitch-users] HANGUP_CAUSE after bridge in outbound ESL
In-Reply-To: <AANLkTi=BXYL3oCe9yUQNp1JyAKdbkSXURps+hqkSVT8y@mail.gmail.com>
Message-ID: <262550.10523.qm@web33503.mail.mud.yahoo.com>

Hi, check originate_disposition variable

http://wiki.freeswitch.org/wiki/Extension_Status_Example
http://wiki.freeswitch.org/wiki/Variable_originate_disposition

Diego Toro

http://voipensando.blogspot.com/

--- On Tue, 3/1/11, Steven Ayre <steveayre at gmail.com> wrote:

From: Steven Ayre <steveayre at gmail.com>
Subject: Re: [Freeswitch-users] HANGUP_CAUSE after bridge in outbound ESL
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Tuesday, March 1, 2011, 4:08 AM

The correct way is to use events. You can register for just the hangup event.

-Steve


On 28 February 2011 22:43, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> Hi All
>
> What is correct method to get bridge status after execute("bridge") in ESL?
> I have inbound call that gets bridged to SIP endpoint. I need to know
> whether it was ORIGINATOR_CANCEL or BUSY or something else, but if I use
> sync, seems I can't determine ORIGINATOR_CANCEL status, because it is set on
> a-leg which is destroyed first after hangup.
>
> Is there a way to get bridge status without messing with events and event
> source filtering by channel uuid or event type?
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

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



      
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From julf at julf.com  Tue Mar  1 19:08:17 2011
From: julf at julf.com (Johan Helsingius)
Date: Tue, 01 Mar 2011 17:08:17 +0100
Subject: [Freeswitch-users] Changing voicemail greeting to pronounce
 (instead of spelling out) id/number
Message-ID: <4D6D19F1.4040105@julf.com>

Hi!

I am using actual names of users instead of numbers (numbers
are so POTS), but there doesn't seem to be a way to change
the way the voicemail user id is announced - I know you can
use vm-alternate-greet-id to change the ID, but if it is
not numerical, it gets spelled out letter by letter instead
of pronounced. Is there any way to change this behavior
without changing the source code?

	Julf


From anthony.minessale at gmail.com  Tue Mar  1 21:11:08 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 1 Mar 2011 12:11:08 -0600
Subject: [Freeswitch-users] fs_cli socket connection error
In-Reply-To: <AANLkTikEDgpAJppNUnLac73ip7qsk+8AK-N-pTW7NBm6@mail.gmail.com>
References: <AANLkTikCoT9dzLNoD=6TzS0jiMw4afNX94TypsLarBbz@mail.gmail.com>
	<AANLkTi=sENgBB9Cdr-EkLJD8XfmH3A8GSfPnvDHO7qrw@mail.gmail.com>
	<AANLkTin5k=2Z-=QVVzqt91172wv88P7OFkqRqTY9VDcS@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58ADB727D7@cooper>
	<AANLkTi=aPwmnVD8ZzaBVFDxGyPsbgzDEBGaXvq_B2NbF@mail.gmail.com>
	<AANLkTikpQuxvyaU7PAGnyJuoQLQ9xn4inNJ_u5x41Anv@mail.gmail.com>
	<AANLkTimNYjC_G-Lq+St5VePpxEyM8iHnhaNCo=jBydi9@mail.gmail.com>
	<AANLkTi=QXgEb6dHU9bfb7v06WzPyWJcoNYDLi1X9oq+E@mail.gmail.com>
	<AANLkTikEDgpAJppNUnLac73ip7qsk+8AK-N-pTW7NBm6@mail.gmail.com>
Message-ID: <AANLkTi=NP_Kw-Hrb6+XXtuid3YWJUM2ehFaeoaV-UEaC@mail.gmail.com>

like I said you re-installed with a new version of libcurl but did not
fully rebuild.
So most of the modules that depend on curl were broken torn between
the old and new version it was linked to..

had you executed the 4 commands I posted yesterday it would be fine.


On Tue, Mar 1, 2011 at 11:02 AM, deniro <ayhkor at gmail.com> wrote:
> Thank you all for advice? and direction
> Will re-install from scratch at a later time.
> deniro--
>
> On Tue, Mar 1, 2011 at 4:16 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> Ok, you really need to pay attention to any log message that logs with
>> the CRIT (critical) log level - they're really serious errors. You
>> should look at adding the critical="true" attribute to all the tags in
>> modules.conf.xml which you want to be sure are loaded. If a module
>> fails to load without that tag FS will continue to run anyway, with
>> that tag it'll refuse to start so you'll know instantly something is
>> wrong.
>>
>> Most of your modules aren't loading, because they haven't been
>> compiled correctly. Some look like they're missing (no such file or
>> directory) and others haven't been built right (undefined symbols).
>>
>> Here are a few of the important lines from your log:
>>
>> 2011-02-28 18:12:21.910738 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_event_socket.so
>> **/opt/freeswitch/mod/mod_event_socket.so: undefined symbol:
>> switch_event_serialize_json**
>> 2011-02-28 18:12:21.910789 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_xml_curl.so
>> **/opt/freeswitch/mod/mod_xml_curl.so: cannot open shared object file:
>> No such file or directory**
>> 2011-02-28 18:12:21.910813 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_xml_cdr.so
>> **/opt/freeswitch/mod/mod_xml_cdr.so: cannot open shared object file:
>> No such file or directory**
>> 2011-02-28 18:12:21.911581 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_commands.so
>> **/opt/freeswitch/mod/mod_commands.so: undefined symbol:
>> switch_xml_reload**
>> 2011-02-28 18:12:21.911709 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_conference.so
>> **/opt/freeswitch/mod/mod_conference.so: undefined symbol:
>> switch_channel_test_app_flag_key**
>> 2011-02-28 18:12:21.911842 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_dptools.so
>> **/opt/freeswitch/mod/mod_dptools.so: undefined symbol:
>> switch_event_merge**
>> 2011-02-28 18:12:21.912264 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_dialplan_xml.so
>> **/opt/freeswitch/mod/mod_dialplan_xml.so: undefined symbol:
>> switch_xml_std_datetime_check**
>> 2011-02-28 18:12:21.912531 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_g723_1.so
>> **/opt/freeswitch/mod/mod_g723_1.so: cannot open shared object file:
>> No such file or directory**
>> 2011-02-28 18:12:21.912573 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_g729.so
>> **/opt/freeswitch/mod/mod_g729.so: cannot open shared object file: No
>> such file or directory**
>> 2011-02-28 18:12:21.912609 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_amr.so
>> **/opt/freeswitch/mod/mod_amr.so: cannot open shared object file: No
>> such file or directory**
>>
>> Important things to note:
>> mod_event_socket hasn't loaded because of a undefined symbol, it
>> wasn't built correctly
>> mod_xml_curl hasn't loaded because it doesn't exist, it was either
>> never built or never installed
>>
>> Something pretty strange has happened. Did you recompile when trying
>> to add mod_xml_curl?
>>
>> I'd suggest you delete all the FS files, including the Git clone, make
>> a fresh checkout and build it from scratch.
>>
>> -Steve
>>
>>
>>
>> On 28 February 2011 23:18, deniro <ayhkor at gmail.com> wrote:
>> > I put the log in pastebin
>> > freeswitch.log only?when starting freeswitch (after stop)
>> >
>> > http://pastebin.freeswitch.org/15502
>> > thx
>> > deniro
>> > On Mon, Feb 28, 2011 at 5:46 PM, Steven Ayre <steveayre at gmail.com>
>> > wrote:
>> >>
>> >> You must check the logfile for when FS starts up - if netstat shows
>> >> nothing for port 8021 then either mod_event_socket isn't being loaded
>> >> or you'll see an error when it tries to load.
>> >>
>> >> Nothing from netstat means nothing's listening, so trying to connect
>> >> using fs_cli won't do anything.
>> >>
>> >> -Steve
>> >>
>> >>
>> >> On 28 February 2011 22:40, deniro <ayhkor at gmail.com> wrote:
>> >> > Checked the log
>> >> > Nothing is logged when
>> >> > fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
>> >> > is issued
>> >> > /opt/freeswitch/log# tail -f freeswitch.log
>> >> >
>> >> > ls -l freeswitch.log
>> >> > -rwxrwxrwx 1 freeswitch daemon 33018 2011-02-28 17:39 freeswitch.log
>> >> > thx
>> >> >
>> >> >
>> >> >
>> >> > On Mon, Feb 28, 2011 at 3:59 PM, Peter Olsson
>> >> > <peter.olsson at visionutveckling.se> wrote:
>> >> >>
>> >> >> Check freeswitch.log, it probably reports some problem when loading
>> >> >> the
>> >> >> mod_event_socket module.
>> >> >>
>> >> >> /Peter
>> >> >> ________________________________________
>> >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> >> >> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r deniro
>> >> >> [ayhkor at gmail.com]
>> >> >> Skickat: den 28 februari 2011 21:42
>> >> >> Till: FreeSWITCH Users Help
>> >> >> ?mne: Re: [Freeswitch-users] fs_cli socket connection error
>> >> >>
>> >> >> Steve
>> >> >> Thanks for the reply
>> >> >> netstat -anlp | grep 8021 ?is blank (nothing showing up)
>> >> >>
>> >> >> /opt/freeswitch/bin/fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
>> >> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
>> >> >> Configuration
>> >> >> file is /root/.fs_cli_conf.
>> >> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
>> >> >> Configuration
>> >> >> file is /etc/fs_cli.conf.
>> >> >> [DEBUG] libs/esl/fs_cli.c:1159 main() profile default does not exist
>> >> >> using
>> >> >> builtin profile
>> >> >> [DEBUG] libs/esl/fs_cli.c:1188 main() Using profile internal
>> >> >> [75.xxx.xxx.xxx]
>> >> >> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
>> >> >> Connection
>> >> >> Error]
>> >> >> yes mod_event_socket is after ?mod_xml_curl but I changed the order
>> >> >> in
>> >> >> ?modules.conf.xml
>> >> >> still getting above (restarted freeswitch)
>> >> >>
>> >> >> thx
>> >> >> deniro--
>> >> >>
>> >> >>
>> >> >>
>> >> >> On Mon, Feb 28, 2011 at 3:35 AM, Steven Ayre
>> >> >> <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
>> >> >> Enable debug logging and you should see an error that'll tell you
>> >> >> more.
>> >> >>
>> >> >> Is mod_event_socket loading before or after mod_xml_curl? Chances
>> >> >> are
>> >> >> mod_xml_curl is loading first, so it's trying to read
>> >> >> event_socket.conf.xml and/or acl.conf.xml through xml_curl and
>> >> >> either
>> >> >> getting a different config to your previous local copy or the ACLs
>> >> >> are
>> >> >> different.
>> >> >>
>> >> >> What does "netstat -anlp | grep :8021" from the linux cli show you?
>> >> >> Does it show that freeswitch is actually listening on the port? If
>> >> >> it
>> >> >> is it's probably an ACL problem, if it isn't then it's probably a
>> >> >> problem with event_socket.conf.xml
>> >> >>
>> >> >> -Steve
>> >> >>
>> >> >> On 28 February 2011 00:53, deniro
>> >> >> <ayhkor at gmail.com<mailto:ayhkor at gmail.com>> wrote:
>> >> >> > What would be possible reasons for this and how to resolve?
>> >> >> > running fs 106 on ubuntu 10.04 server
>> >> >> >
>> >> >> > /opt/freeswitch/bin/fs_cli -H xxx.xxx.xxx.xxx -P 8021
>> >> >> > [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
>> >> >> > Connection
>> >> >> > Error]
>> >> >> > was working fine before I installed ?mod_xml_cdr
>> >> >> > configure --prefix=/opt/freeswitch --without-libcurl
>> >> >> > make mod_xml_cdr-install
>> >> >> > (no errors)
>> >> >> >
>> >> >> > in modules.conf.xml
>> >> >> > <load module="mod_event_socket"/>
>> >> >> >
>> >> >> > is there a packaged version of mod_xml_cdr (for fs 1.0.6)?
>> >> >> >
>> >> >> > thx
>> >> >> > deniro--
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org<http://www.freeswitch.org/>
>> >> >> >
>> >> >> >
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >>
>> >> >>
>> >> >>
>> >> >> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org<http://www.freeswitch.org/>
>> >> >>
>> >> >> !DSPAM:4d6c0a1132761029518849!
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From msc at freeswitch.org  Tue Mar  1 21:15:19 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 1 Mar 2011 10:15:19 -0800
Subject: [Freeswitch-users] problem with voice quality
In-Reply-To: <AANLkTikDK=tRczp5Dpjw55WKWy9dwpRu1hAc+RP9adQn@mail.gmail.com>
References: <AANLkTik15iq+zwivCJWZeVFOmDVvAFB70v7haeK9XE2K@mail.gmail.com>
	<1CC6662BE5B94BD9A2C7DF20A3FDD577@stor1>
	<AANLkTim5US0xk1pYHyomm-_VwA1BQtACJU4p46ahPPV8@mail.gmail.com>
	<AANLkTik_zP1oX6Uwrk0MOA32ezXUp57K-DJ82BJdNeon@mail.gmail.com>
	<AANLkTikDK=tRczp5Dpjw55WKWy9dwpRu1hAc+RP9adQn@mail.gmail.com>
Message-ID: <AANLkTimQ6Df2LUb0aEQ9Ds+Taj9_v4ryc4EPGFioAw8d@mail.gmail.com>

On Mon, Feb 28, 2011 at 7:15 PM, joy this <thisjoy0528 at gmail.com> wrote:

>
> Can you reproduce the sound problems on the very latest Git?
>
>  I downloaded FS at 2/24 last week, but I couldn't see the version. It only
> shows "FreeSWITCH Version 1.0.head (git-)" when I input "version".
>
> You have an older version of git. See if you can get up to at least git 1.7


> Can you
>> hear speech with noise, or is it just garbage?
>
> I could hear the speech with the luod noise.
>

> Can you get a
>> debug-level log for the call with siptrace enabled so we can see
>> what's going on? Also use tcpdump/wireshark/tshark to capture the RTP
>> and analyze it in Wireshark to check for packet loss or jitter.
>>
> I have the log file and the package (by Wireshark), should I share them by
> the email?
>

Put this out on a web server somewhere so that people can download.
(Something like a drop box or a web server that you have.)

-MC
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From peter.olsson at visionutveckling.se  Tue Mar  1 21:15:33 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 1 Mar 2011 19:15:33 +0100
Subject: [Freeswitch-users] Changing voicemail greeting to pronounce
 (instead of spelling out) id/number
In-Reply-To: <4D6D19F1.4040105@julf.com>
References: <4D6D19F1.4040105@julf.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2C494CC@cooper>

You will need some kind of TTS engine to get this working, however, I'm not sure if any changes must be made to mod_voicemail as well.
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Johan Helsingius [julf at julf.com]
Skickat: den 1 mars 2011 17:08
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] Changing voicemail greeting to pronounce (instead of spelling out) id/number

Hi!

I am using actual names of users instead of numbers (numbers
are so POTS), but there doesn't seem to be a way to change
the way the voicemail user id is announced - I know you can
use vm-alternate-greet-id to change the ID, but if it is
not numerical, it gets spelled out letter by letter instead
of pronounced. Is there any way to change this behavior
without changing the source code?

        Julf

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

!DSPAM:4d6d34bf32762095167759!



From msc at freeswitch.org  Tue Mar  1 21:17:39 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 1 Mar 2011 10:17:39 -0800
Subject: [Freeswitch-users] Does FS handle three-way calling on the same
	POTS?
In-Reply-To: <1298970647476-6076537.post@n2.nabble.com>
References: <1298970647476-6076537.post@n2.nabble.com>
Message-ID: <AANLkTikyTQU1JiA1mhEVWxz=GXk_Ragy6U3nZnbNBVwB@mail.gmail.com>

What are the instructions from the provider to create the second call?
Usually with an analog line you have to do a hook flash, then dial some more
digits, and then do another hook flash.

-MC

On Tue, Mar 1, 2011 at 1:10 AM, GillesToo <codecomplete at free.fr> wrote:

> Hello
>
> My ISP offers basic PBX features and free calls through when plugging a
> phone in an RJ11 on the ADSL modem. I'd like to hook up a PC to the modem
> with an FXO module, and use three-way calling so that the server calls me
> back on my cellphone, and then lets me call a number through the server
> before switching to a conference call. Poor man's callback system :-)
>
> I finally got Asterisk to make the first call, and put it on hold, but it
> stops there: It appears that Asterisk is unable to create a second channel
> on the same FXO module when a call has been put on hold ("app_dial.c:1310
> dial_exec_full:Unable to create channel of type 'Dahdi' (cause 0 -
> Unknown)").
>
> I was wondering if someone had successfully used Freeswitch in this
> scenario?
>
> FWIW, here's my extensions.conf:
> ==============
> [from_fxo]
> exten => s,1,Wait(2)
> exten => s,n,Set(GLOBAL(CID)=${CALLERID(num)})
> exten => s,n,Hangup()
>
> ;script waits and creates callfile
> exten => h,1,system(/var/tmp/test10.lua ${CID}&)
>
> ;context used by callfile
> [callback]
> exten => start,1,NoOp(In callback, CID is ${CID})
> ;how to wait until remote phone picked up?
> exten => start,n,Wait(5)
> exten => start,n,Answer()
> exten => start,n,Playback(tt-monkeysintro)
>
> exten => start,n,Flash()
>
> ;BAD exten => start,n,Dial(Dahdi/1/${GSM})
> ;does ring second number, but no actual voice channel
> exten => start,n,SendDTMF(456789)
> ==============
>
> Thank you.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Does-FS-handle-three-way-calling-on-the-same-POTS-tp6076537p6076537.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Mar  1 21:20:05 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 1 Mar 2011 10:20:05 -0800
Subject: [Freeswitch-users] weird FS behavior
In-Reply-To: <AANLkTimDLm0qNWw=hL0pXZ6ic2McDPjwDrdb3tgx9Tfr@mail.gmail.com>
References: <AANLkTimDLm0qNWw=hL0pXZ6ic2McDPjwDrdb3tgx9Tfr@mail.gmail.com>
Message-ID: <AANLkTin-cCDbiFAQ+H02zUkNHb+y_eC479dCyTw4rfoH@mail.gmail.com>

> 2011-03-01 03:22:03.540372 [INFO] mod_dialplan_xml.c:418 Processing
> 1000->13474562345 in context default
>
>
I'm with Peter on this one. It looks like FreeSWITCH is receiving the
dialstring with the "1347" already pre-pended. Time to check the phone.

-MC
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From jjj at 3js.de  Tue Mar  1 21:21:44 2011
From: jjj at 3js.de (Johannes Jakob)
Date: Tue, 1 Mar 2011 19:21:44 +0100
Subject: [Freeswitch-users] T.38 Issues with passthrough handshaking
In-Reply-To: <AANLkTikDSKqf0TeeWjXUzAd3KLqsw7_wK1TwJ3sOZmZf@mail.gmail.com>
References: <640F91CF-9489-4F74-96B1-A6398CD84F07@3js.de>
	<A53FD340-169C-4DD4-8AF2-C463F9D9E3DD@3js.de>
	<AANLkTi=GmF93TkQPNuFf9ko_eQVgvqwujZOo8duN8ZDB@mail.gmail.com>
	<64786198-FF65-49FA-BCC6-57337A412D37@3js.de>
	<1951319FA2234F89920E93B69BAC5351@e1705>
	<7FD8D98C-266D-4CA4-8342-E6B953B9DEDC@3js.de>
	<00DD4259DF544C7390702EF7B024CE11@e1705>
	<AANLkTikDSKqf0TeeWjXUzAd3KLqsw7_wK1TwJ3sOZmZf@mail.gmail.com>
Message-ID: <DEA386DE-4A4B-45FE-9454-53A14B591C40@3js.de>

Hi,


On 01.03.2011, at 18:30, Anthony Minessale wrote:

> You need to get traces from the FreeSWITCH console more than a pcap.

Et voila: http://pastebin.freeswitch.org/15509

Same situation, same result, like in the other debug stuff.

If you need anything else.... let me know.

Thanks,

  John




> Execute these commands on the cli: (don't forget any I often have to
> tell people to take traces more than once)
> 
> sofia global siptrace on
> console loglevel debug
> 
> Reproduce the problem situation then paste that log.
> Every new time something changes, repeat this step.
> 
> 
> On Tue, Mar 1, 2011 at 11:18 AM, Madovsky <infos at madovsky.org> wrote:
>> since the implementation of T38 you should
>> set proxy_media to false and bypass_media to false also....
>> 
>> ----- Original Message -----
>> From: "Johannes Jakob" <jjj at 3js.de>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Tuesday, March 01, 2011 12:09 PM
>> Subject: Re: [Freeswitch-users] T.38 Issues with passthrough handshaking
>> 
>> 
>>>>>> Okay, thanks for this hint, the wiki suggests proxy_media...
>>>> Johannes,
>>>> Anthony is one of the main developer of FS,
>>>> so you can trust him at 100%
>>> 
>>> hehe, thank you, but I'm fully aware of who Anthony is and I *do* in fact
>>> fully trust him in all things FreeSWITCH ;) sorry if my badly chosen words
>>> imply I wouldn't!
>>> 
>>> All I wanted to say is, that the setup I had done and described in my
>>> first posting has been done after reading Anthony's, Michael's and
>>> Darren's book and lots and lots of wiki pages and mailing list postings.
>>> In the past it seems to have been necessary to use proxy_media and disable
>>> bypass manually, so I had done that, not knowing this is deprecated.
>>> 
>>> So, if anybody of you FreeSWITCH wizards has ANY idea on how to get my
>>> setup working, I'd be very very *very* happy....
>>> 
>>> I love FreeSWITCH for it's clean and organized structure, it's flexibility
>>> and speed.... but t.38 drives me crazy... and that's of course NOT
>>> FreeSWITCH's fault!
>>> 
>>> 
>>> Thanks to everybody who has time and willing to read my debugging stuff!
>>> 
>>> And of course sorry to Anthony, if I by any means disrespected you with my
>>> bad english! ;)
>>> 
>>> Greetings,
>>>  John
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Tue Mar  1 21:22:44 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 1 Mar 2011 10:22:44 -0800
Subject: [Freeswitch-users] Thoughts about mod_conference enter-sound,
 when wait-moderator has been set
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B2BBAB93@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C58B2BBAB93@cooper>
Message-ID: <AANLkTi=pCWWarPvpeemzphhNNDGFZg84om7F2Txsjez3@mail.gmail.com>

Or a really short sound, like a 100ms ding sound.
-MC

On Tue, Mar 1, 2011 at 5:50 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> Good afternoon (at least for me)!
>
>
>
> Stupid question maybe ? but here it goes :)
>
>
>
> When I configure a conference to wait for the moderator before connecting
> people, I?ve noticed that ?enter-sound? is played synchronously. Is there
> any special reason for this? I?m thinking about adding a new parameter to
> allow it to play asynchronously, even when in this mode. My reasons for this
> is if there are 50 people logged in, and the moderator connects to the
> conference, there will be lots of time spent just to play the enter-sound
> for each member. 1 second of sound for each member will cause 50 seconds
> before it has played them all. It makes sense in the case when the caller?s
> name is read, but for a common enter-sound I think an asynchronous playback
> would be good enough?
>
>
>
> /Peter
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Mar  1 21:27:13 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 1 Mar 2011 10:27:13 -0800
Subject: [Freeswitch-users] British sounds
In-Reply-To: <4D6D0366.7020205@integrafin.co.uk>
References: <4D681972.7050400@integrafin.co.uk>
	<AANLkTinGQk18xc8WCap6VkyTVYM1-XHcPi1PhwVTNs8B@mail.gmail.com>
	<4D68222C.8010108@integrafin.co.uk>
	<4D6D0366.7020205@integrafin.co.uk>
Message-ID: <AANLkTi=pz1GgEz9OLvW_7=QtO=B=nWiHS+4G6D7MY=-N@mail.gmail.com>

Okay, this is good. I recommend getting a spreadsheet together and creating
columns for Ast file name, Ast prompt, FS file name, FS prompt. The prompts
would contain the actual text of what is voiced in the audio file. On the FS
side look in $FS_SOURCE/docs/phrase/phrase_en.xml for the phrases and file
names. In Asterisk there usually is a text file that has the filename and
the text of the prompt.

Once we have that together we can figure out what prompts we can re-use
versus what needs to be re-recorded.

-MC

On Tue, Mar 1, 2011 at 6:32 AM, Alex Crow <acrow at integrafin.co.uk> wrote:

>  On 25/02/11 21:42, Alex Crow wrote:
>
>
>   2. A script to convert and rename sounds from an Asterisk install to FS?
>>
>> We already paid for British female sounds for Asterisk so if conversion
>> will cover at least voicemail prompts it would be a start.
>>
>
>  There is some overlap in the prompts, especially the digits and the time,
> but many of the other prompts are definitely different. I'd be willing to
> assist you in getting the prompts you created for * converted for use w/ FS.
> The challenge is figuring out which prompts need to be recorded. Do you have
> a list of sound prompts and file names? I could take a look...
>
>  -MC
>
>
>
> Michael,
>
> The ones we have for Asterisk are British English "Rachel", purchased from
>
> http://www.keison.co.uk/westany/asterisk_voice_prompt.htm
>
> They do seem to originate from westany from a google search.
>
> I think I can provide a list of the directory structure from our Asterisk
> box. Do you have a connection to the company who recorded these sounds? If
> we could get the same voice artist it would be great, especially if they
> were available as a product for others to buy.
>
> Cheers
>
> Alex
>
> --
> This message is intended only for the addressee and may contain
> confidential information.  Unless you are that person, you may not
> disclose its contents or use it in any way and are requested to delete
> the message along with any attachments and notify us immediately.
>
> "Transact" is operated by Integrated Financial Arrangements plc
> Domain House, 5-7 Singer Street, London  EC2A 4BQ
> Tel: (020) 7608 4900 Fax: (020) 7608 5300
> (Registered office: as above; Registered in England and Wales under number: 3727592)
> Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)
>
>
> --
> This message has been scanned for viruses and
> dangerous content by *MailScanner* <http://www.mailscanner.info/>, and is
> believed to be clean.
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>  Michael,
>
> Here is a tree of the sounds on my asterisk box. They were simply dropped
> in place over the existing US sounds. Is there anything else you need?
>
> Thanks
>
> Alex
>
> ??? account_not_valid.gsm
> ??? activated.gsm
> ??? agent-alreadyon.gsm
> ??? agent-incorrect.gsm
> ??? agent-loggedoff.gsm
> ??? agent-loginok.gsm
> ??? agent-newlocation.gsm
> ??? agent-pass.gsm
> ??? agent-user.gsm
> ??? all-circuits-busy-now.gsm
> ??? AM.gsm
> ??? and.gsm
> ??? an-error-has-occured.gsm
> ??? at-tone-time-exactly.gsm
> ???
> auth-incorrect.gsm                                                                                                                                                                 &n
> bsp;
> ???
> auth-thankyou.gsm                                                                                                                                                                 &nb
> sp;
> ???
> beeperr.gsm                                                                                                                                                                  &nb
> sp;
> ???
> beeperr.ulaw                                                                                                                                                                  &n
> bsp;
> ???
> beep.gsm
>
> ???
> beep.ulaw                                                                                                                                                                  &nbsp
> ;
> ???
> call-forward.gsm                                                                                                                                                                 &nbs
> p;
> ???
> call-forwarding.gsm                                                                                                                                                                 &
> nbsp;
> ???
> call_forward_notification.gsm
>
> ???
> call-forward-parallel.gsm                                                                                                                                                                &
> nbsp;
> ???
> call-fwd-cancelled.gsm                                                                                                                                                                &nbs
> p;
> ???
> call-fwd-no-ans.gsm                                                                                                                                                                 &
> nbsp;
> ???
> call-fwd-on-busy.gsm
>
> ???
> call-fwd-parallel.gsm                                                                                                                                                                &nbsp
> ;
> ???
> call-fwd-unconditional.gsm
>
> ??? call-waiting.gsm
> ??? callwaiting.gsm
> ??? channels_limit_exceeded.gsm
> ??? conf-adminmenu.gsm
> ??? conf-enteringno.gsm
> ??? conf-errormenu.gsm
> ??? conf-getchannel.gsm
> ??? conf-getconfno.gsm
> ??? conf-getpin.gsm
> ??? conf-hasjoin.gsm
> ??? conf-hasleft.gsm
> ??? conf-invalid.gsm
> ??? conf-invalidpin.gsm
> ??? conf-kicked.gsm
> ??? conf-leaderhasleft.gsm
> ??? conf-locked.gsm
> ??? conf-lockednow.gsm
> ??? conf-muted.gsm
> ??? conf-noempty.gsm
> ??? conf-onlyone.gsm
> ??? conf-onlyperson.gsm
> ??? conf-otherinparty.gsm
> ??? conf-placeintoconf.gsm
> ??? conf-thereare.gsm
> ??? conf-unlockednow.gsm
> ??? conf-unmuted.gsm
> ??? conf-usermenu.gsm
> ??? conf-userswilljoin.gsm
> ??? conf-userwilljoin.gsm
> ??? conf-waitforleader.gsm
> ??? custom
> ??? dash.gsm
> ??? de-activated.gsm
> ??? deactivated.gsm
> ??? default_auto_attendant.gsm
> ??? default_moh.gsm
> ??? demo-abouttotry.gsm
> ??? demo-congrats.gsm
> ??? demo-echodone.gsm
> ??? demo-echotest.gsm
> ??? demo-enterkeywords.gsm
> ??? demo-instruct.gsm
> ??? demo-moreinfo.gsm
> ??? demo-nogo.gsm
> ??? demo-nomatch.gsm
> ??? demo-thanks.gsm
> ??? dest_grp_not_exist.gsm
> ??? destination_not_allowed.gsm
> ??? destination_not_recognised.gsm
> ??? destination_not_supported.gsm
> ??? destination_permis_not_set.gsm
> ??? dest_rate_not_set.gsm
> ??? dictate
> ?   ??? en
> ?       ??? both_help.gsm
> ?       ??? enter_filename.gsm
> ?       ??? forhelp.gsm
> ?       ??? paused.gsm
> ?       ??? pause.gsm
> ?       ??? playback.gsm
> ?       ??? playback_mode.gsm
> ?       ??? play_help.gsm
> ?       ??? record.gsm
> ?       ??? record_help.gsm
> ?       ??? record_mode.gsm
> ?       ??? truncating_audio.gsm
> ??? digits
> ?   ??? en
> ?       ??? 0.gsm
> ?       ??? 10.gsm
> ?       ??? 11.gsm
> ?       ??? 12.gsm
> ?       ??? 13.gsm
> ?       ??? 14.gsm
> ?       ??? 15.gsm
> ?       ??? 16.gsm
> ?       ??? 17.gsm
> ?       ??? 18.gsm
> ?       ??? 19.gsm
> ?       ??? 1.gsm
> ?       ??? 20.gsm
> ?       ??? 2.gsm
> ?       ??? 30.gsm
> ?       ??? 3.gsm
> ?       ??? 40.gsm
> ?       ??? 4.gsm
> ?       ??? 50.gsm
> ?       ??? 5.gsm
> ?       ??? 60.gsm
> ?       ??? 6.gsm
> ?       ??? 70.gsm
> ?       ??? 7.gsm
> ?       ??? 80.gsm
> ?       ??? 8.gsm
> ?       ??? 90.gsm
> ?       ??? 9.gsm
> ?       ??? a-m.gsm
> ?       ??? at.gsm
> ?       ??? day-0.gsm
> ?       ??? day-1.gsm
> ?       ??? day-2.gsm
> ?       ??? day-3.gsm
> ?       ??? day-4.gsm
> ?       ??? day-5.gsm
> ?       ??? day-6.gsm
> ?       ??? dollar.gsm
> ?       ??? dollars.gsm
> ?       ??? euro.gsm
> ?       ??? euros.gsm
> ?       ??? h-10.gsm
> ?       ??? h-11.gsm
> ?       ??? h-12.gsm
> ?       ??? h-13.gsm
> ?       ??? h-14.gsm
> ?       ??? h-15.gsm
> ?       ??? h-16.gsm
> ?       ??? h-17.gsm
> ?       ??? h-18.gsm
> ?       ??? h-19.gsm
> ?       ??? h-1.gsm
> ?       ??? h-20.gsm
> ?       ??? h-2.gsm
> ?       ??? h-30.gsm
> ?       ??? h-3.gsm
> ?       ??? h-4.gsm
> ?       ??? h-5.gsm
> ?       ??? h-6.gsm
> ?       ??? h-7.gsm
> ?       ??? h-8.gsm
> ?       ??? h-9.gsm
> ?       ??? hundred.gsm
> ?       ??? million.gsm
> ?       ??? minus.gsm
> ?       ??? mon-0.gsm
> ?       ??? mon-10.gsm
> ?       ??? mon-11.gsm
> ?       ??? mon-1.gsm
> ?       ??? mon-2.gsm
> ?       ??? mon-3.gsm
> ?       ??? mon-4 .gsm
> ?       ??? mon-5.gsm
> ?       ??? mon-6 .gsm
> ?       ??? mon-7.gsm
> ?       ??? mon-8 .gsm
> ?       ??? mon-9.gsm
> ?       ??? oclock.gsm
> ?       ??? oh.gsm
> ?       ??? p-m.gsm
> ?       ??? pound.gsm
> ?       ??? pounds.gsm
> ?       ??? star.gsm
> ?       ??? thousand.gsm
> ?       ??? today.gsm
> ?       ??? tomorrow.gsm
> ?       ??? yesterday.gsm
> ??? dir-instr.gsm
> ??? dir-intro-fn.gsm
> ??? dir-intro-fnln.gsm
> ??? dir-intro-fnln-oper.gsm
> ??? dir-intro-fn-oper.gsm
> ??? dir-intro.gsm
> ??? dir-intro-oper.gsm
> ??? dir-nomatch.gsm
> ??? dir-nomore.gsm
> ??? dollar.gsm
> ??? dollars.gsm
> ??? do-not-disturb.gsm
> ??? dot.gsm
> ??? enter_account_number.gsm
> ??? enter-conf-pin-number.gsm
> ??? enter_ext_number.gsm
> ??? enter_ntwk_num.gsm
> ??? enter-num-blacklist.gsm
> ??? entr-num-rmv-blklist.gsm
> ??? ent-target-attendant.gsm
> ??? equals.gsm
> ??? exclaimation-point.gsm
> ??? extension.gsm
> ??? feature-not-avail-line.gsm
> ??? first-in-line.gsm
> ??? first-three-letters-entry.gsm
> ??? followme
> ?   ??? en
> ?       ??? call-from.gsm
> ?       ??? no-recording.gsm
> ?       ??? options.gsm
> ?       ??? sorry.gsm
> ?       ??? status.gsm
> ??? for2.gsm
> ??? for_accounting.gsm
> ??? for_accounts.gsm
> ??? for_customer_support.gsm
> ??? for_development.gsm
> ??? for.gsm
> ??? for_production.gsm
> ??? for_sales.gsm
> ??? for_technical_support.gsm
> ??? freshtel.gsm
> ??? goodbye.gsm
> ??? greeting-default-attendant.gsm
> ??? hash.gsm
> ??? hello-world.gsm
> ??? hours.gsm
> ??? if-correct-press.gsm
> ??? incoming-call-1-accept-2-decline.sln
> ??? incoming-call-no-longer-avail.sln
> ??? info-about-last-call.gsm
> ??? inithelp.gsm
> ??? insufficient_acc_bal.gsm.gsm
> ??? invalid.gsm
> ??? invalid_selection.gsm
> ??? is-curntly-unavail.gsm
> ??? is.gsm
> ??? is-set-to.gsm
> ??? is_unable_to_answer.gsm
> ??? last-num-to-call2.gsm
> ??? last-num-to-call.gsm
> ??? letters
> ?   ??? en
> ?       ??? a.gsm
> ?       ??? at.gsm
> ?       ??? b.gsm
> ?       ??? c.gsm
> ?       ??? dash.gsm
> ?       ??? d.gsm
> ?       ??? dollar.gsm
> ?       ??? dollars.gsm
> ?       ??? dot.gsm
> ?       ??? e.gsm
> ?       ??? equals.gsm
> ?       ??? euro.gsm
> ?       ??? euros.gsm
> ?       ??? exclaimation-point.gsm
> ?       ??? f.gsm
> ?       ??? g.gsm
> ?       ??? h.gsm
> ?       ??? i.gsm
> ?       ??? j.gsm
> ?       ??? k.gsm
> ?       ??? l.gsm
> ?       ??? m.gsm
> ?       ??? n.gsm
> ?       ??? o.gsm
> ?       ??? p.gsm
> ?       ??? plus.gsm
> ?       ??? q.gsm
> ?       ??? r.gsm
> ?       ??? s.gsm
> ?       ??? slash.gsm
> ?       ??? space.gsm
> ?       ??? t.gsm
> ?       ??? u.gsm
> ?       ??? v.gsm
> ?       ??? w.gsm
> ?       ??? x.gsm
> ?       ??? y.gsm
> ?       ??? zee.gsm
> ?       ??? z.gsm
> ??? minutes.gsm
> ??? no_active_agents.gsm
> ??? no_phone_support_for_acc.gsm
> ??? not_authorized.gsm
> ??? not_enabled.gsm
> ??? not_in_service.gsm
> ??? now.gsm
> ??? ntwk_no_not_exist.gsm
> ??? number2.gsm
> ??? number.gsm
> ??? one-moment-please.gsm
> ??? out_of_order.gsm
> ??? pbx-invalid.gsm
> ??? pbx-invalidpark.gsm
> ??? pbx-transfer.gsm
> ??? percent.gsm
> ??? phonetic
> ?   ??? en
> ?       ??? 9_p.gsm
> ?       ??? a_p.gsm
> ?       ??? b_p.gsm
> ?       ??? c_p.gsm
> ?       ??? d_p.gsm
> ?       ??? e_p.gsm
> ?       ??? f_p.gsm
> ?       ??? g_p.gsm
> ?       ??? h_p.gsm
> ?       ??? i_p.gsm
> ?       ??? j_p.gsm
> ?       ??? k_p.gsm
> ?       ??? l_p.gsm
> ?       ??? m_p.gsm
> ?       ??? niner.gsm
> ?       ??? n_p.gsm
> ?       ??? o_p.gsm
> ?       ??? p_p.gsm
> ?       ??? q_p.gsm
> ?       ??? r_p.gsm
> ?       ??? s_p.gsm
> ?       ??? t_p.gsm
> ?       ??? u_p.gsm
> ?       ??? v_p.gsm
> ?       ??? w_p.gsm
> ?       ??? x_p.gsm
> ?       ??? y_p.gsm
> ?       ??? z_p.gsm
> ??? please_contact_cust_support.gsm
> ??? please-enter-your.gsm
> ??? pls_enter_pin.gsm
> ??? pls-try-call-later.gsm
> ??? plus.gsm
> ??? PM.gsm
> ??? press-0.gsm
> ??? press-1.gsm
> ??? press-2.gsm
> ??? press-3.gsm
> ??? press-4.gsm
> ??? press-5.gsm
> ??? press-6.gsm
> ??? press-7.gsm
> ??? press-8.gsm
> ??? press-9.gsm
> ??? press_eight.gsm
> ??? press_five.gsm
> ??? press_four.gsm
> ??? press_hash.gsm
> ??? press_nine.gsm
> ??? press_one.gsm
> ??? press_seven.gsm
> ??? press_six.gsm
> ??? press_star.gsm
> ??? press-star.gsm
> ??? press_three.gsm
> ??? press_two.gsm
> ??? press_zero.gsm
> ??? privacy-blklist-last-caller.gsm
> ??? privacy-incorrect.gsm
> ??? privacy-prompt.gsm
> ??? privacy-thankyou.gsm
> ??? privacy-unident.gsm
> ??? priv-at.gsm
> ??? priv-callee-options.gsm
> ??? priv-callfrom.gsm
> ??? priv-callpending.gsm
> ??? priv-instruct.gsm
> ??? priv-introsaved.gsm
> ??? priv-recordintro.gsm
> ??? priv-sayname.gsm
> ??? priv-trying.gsm
> ??? queue-callswaiting.gsm
> ??? queue-holdtime.gsm
> ??? queue-less-than.gsm
> ??? queue-minutes.gsm
> ??? queue_number.gsm
> ??? queue-periodic-announce.gsm
> ??? queue-reporthold.gsm
> ??? queue-seconds.gsm
> ??? queue-thankyou.gsm
> ??? queue-thereare.gsm
> ??? queue-youarenext.gsm
> ??? screen-callee-options.gsm
> ??? secondary_trunk_used.gsm
> ??? seconds.gsm
> ??? security.gsm
> ??? slash.gsm
> ??? sorry-cant-let-you-do-that.gsm
> ??? space.gsm
> ??? speed-dial-empty.gsm
> ??? speed-dial.gsm
> ??? spy-agent.gsm
> ??? spy-h323.gsm
> ??? spy-iax2.gsm
> ??? spy-iax.gsm
> ??? spy-mgcp.gsm
> ??? spy-sip.gsm
> ??? spy-skinny.gsm
> ??? spy-zap.gsm
> ??? ss-noservice.gsm
> ??? telephone-number.gsm
> ??? teritary_trunk_used.gsm
> ??? then-press-pound2.gsm
> ??? then-press-pound.gsm
> ??? the-number-u-dialed.gsm
> ??? T-is-not-available.gsm
> ??? to-call-this-number.gsm
> ??? to-listen-to-it.gsm
> ??? to_listen_to_these_options.gsm
> ??? to-rerecord-it.gsm
> ??? to_speak_to_an_operator.gsm
> ??? transfer.gsm
> ??? tt-allbusy.gsm
> ??? user_account_suspended.gsm
> ??? user_not_activated.gsm
> ??? vertical_serv_not_available.gsm
> ??? vm-advopts.gsm
> ??? vm-and.gsm
> ??? vm-calldiffnum.gsm
> ??? vm-changeto.gsm
> ??? vm-Cust1.gsm
> ??? vm-Cust2.gsm
> ??? vm-Cust3.gsm
> ??? vm-Cust4.gsm
> ??? vm-Cust5.gsm
> ??? vm-deleted.gsm
> ??? vm-delete.gsm
> ??? vm-dialout.gsm
> ??? vm-enter-num-to-call.gsm
> ??? vm-extension.gsm
> ??? vm-Family.gsm
> ??? vm-first.gsm
> ??? vm-for.gsm
> ??? vm-forward.gsm
> ??? vm-forwardoptions.gsm
> ??? vm-Friends.gsm
> ??? vm-from-extension.gsm
> ??? vm-from.gsm
> ??? vm-from-phonenumber .gsm
> ??? vm-goodbye.gsm
> ??? vm-helpexit.gsm
> ??? vm-INBOX.gsm
> ??? vm-incorrect.gsm
> ??? vm-incorrect-mailbox.gsm
> ??? vm-instructions.gsm
> ??? vm-intro.gsm
> ??? vm-isonphone.gsm
> ??? vm-isunavail.gsm
> ??? vm-last.gsm
> ??? vm-leavemsg.gsm
> ??? vm_login.gsm
> ??? vm-login.gsm
> ??? vm-mailboxfull.gsm
> ??? vm-message.gsm
> ??? vm-messages.gsm
> ??? vm-minutes.gsm
> ??? vm-mismatch.gsm
> ??? vm-msginstruct.gsm
> ??? vm-msgsaved.gsm
> ??? vm-newpassword.gsm
> ??? vm-newuser.gsm
> ??? vm-next.gsm
> ??? vm-nobodyavail.gsm
> ??? vm-nobox.gsm
> ??? vm-no.gsm
> ??? vm-nomore.gsm
> ??? vm-nonumber.gsm
> ??? vm-num-i-have.gsm
> ??? vm-Old.gsm
> ??? vm-onefor.gsm
> ??? vm-options.gsm
> ??? vm-opts.gsm
> ??? vm-passchanged.gsm
> ??? vm-password.gsm
> ??? vm-press.gsm
> ??? vm-prev.gsm
> ??? vm-reachoper.gsm
> ??? vm-rec-busy.gsm
> ??? vm-received.gsm
> ??? vm-rec-name.gsm
> ??? vm-rec-temp.gsm
> ??? vm-rec-unv.gsm
> ??? vm-reenterpassword.gsm
> ??? vm-repeat.gsm
> ??? vm-review.gsm
> ??? vm-saved.gsm
> ??? vm-savedto.gsm
> ??? vm-savefolder.gsm
> ??? vm-savemessage.gsm
> ??? vm-saveoper.gsm
> ??? vm-sorry.gsm
> ??? vm-star-cancel.gsm
> ??? vm-starmain.gsm
> ??? vm-tempgreetactive.gsm
> ??? vm-tempgreeting2.gsm
> ??? vm-tempgreeting.gsm
> ??? vm-tempremoved.gsm
> ??? vm-then-hash.gsm
> ??? vm-then-pound.gsm
> ??? vm-theperson.gsm
> ??? vm-tocallback.gsm
> ??? vm-tocallnum.gsm
> ??? vm-tocancel.gsm
> ??? vm-tocancelmsg.gsm
> ??? vm-toenternumber.gsm
> ??? vm-toforward.gsm
> ??? vm-tohearenv.gsm
> ??? vm-tomakecall.gsm
> ??? vm-tooshort.gsm
> ??? vm-toreply.gsm
> ??? vm-torerecord.gsm
> ??? vm-undeleted.gsm
> ??? vm-undelete.gsm
> ??? vm-unknown-caller.gsm
> ??? vm-whichbox.gsm
> ??? vm-Work.gsm
> ??? vm-youhave.gsm
> ??? welcome-to-phonebook.gsm
> ??? your.gsm
>
> 11 directories, 513 files
>
>
>
> --
> This message is intended only for the addressee and may contain
> confidential information.  Unless you are that person, you may not
> disclose its contents or use it in any way and are requested to delete
> the message along with any attachments and notify us immediately.
>
> "Transact" is operated by Integrated Financial Arrangements plc
> Domain House, 5-7 Singer Street, London  EC2A 4BQ
> Tel: (020) 7608 4900 Fax: (020) 7608 5300
> (Registered office: as above; Registered in England and Wales under number: 3727592)
> Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From freeswitch at cartissolutions.com  Tue Mar  1 22:10:47 2011
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Tue, 01 Mar 2011 13:10:47 -0600
Subject: [Freeswitch-users] mod_dingaling and google voice
In-Reply-To: <AANLkTikncVxc7ebrzR4bHrzeHARKVU_Q+aRpzMTM9RwV@mail.gmail.com>
References: <AANLkTin_dFj9ygx3TAmExeN9-pDFNCuZPGZ_Uc2d7o1W@mail.gmail.com>	<AANLkTimUr3=w_fj_k643ChW93Q_YTm63R_qFyBO9EoQV@mail.gmail.com>	<4D6C1978.8040702@cartissolutions.com>	<AANLkTim5TpcQeYSC_c6ZH9WMoJnq1aNvYViwUBkZr=fv@mail.gmail.com>	<4D6C288B.4050801@cartissolutions.com>
	<AANLkTikncVxc7ebrzR4bHrzeHARKVU_Q+aRpzMTM9RwV@mail.gmail.com>
Message-ID: <4D6D44B7.6070900@cartissolutions.com>

This is a completely fresh rebuild of my configurations from the ground up.

jingle configs:
http://pastebin.freeswitch.org/15510

Inbound call:
http://pastebin.freeswitch.org/15512

Outbound call:
http://pastebin.freeswitch.org/15513

FreeSWITCH Version 1.0.head (git-0d8e945 2011-02-16 08-57-43 -0600), 
launched with this:

#!/bin/sh
ulimit -s 240
exec /usr/local/freeswitch/bin/freeswitch -u SOME_FS_USER -g 
SOME_FS_USERS_GROUP -conf SOME_FS_USER_HOMEDIR/instance_00/conf -log 
SOME_FS_USER_HOMEDIR/instance_00/log -run 
SOME_FS_USER_HOMEDIR/instance_00/run -db 
SOME_FS_USER_HOMEDIR/instance_00/db -scripts 
SOME_FS_USER_HOMEDIR/instance_00/scripts -nonat

Sofia UA's running on both 99.X.X.X:5060 and 192.X.X.X:5060.  Machine FS 
runs on is an iptables NAT/Firewall/router.  SIP calls to/from the 
outside world work fine otherwise.

-Yossi

On 02/28/2011 05:02 PM, Anthony Minessale wrote:
> you might want to backup your configs and start over with the fresh
> ones just to test and work backwards to see what's different.
> its probably something to do with having to omit external-rtp-ip
>
>
> On Mon, Feb 28, 2011 at 4:58 PM, Yossi Neiman
> <freeswitch at cartissolutions.com>  wrote:
>> I am running with -nonat, the FS is running with a UA on a public IP
>> address, etc, but both Michael Collins and I are still getting the RTP
>> port 0 issue.  I figured I'd compare notes with somebody who has a
>> working config before I file this as something other than a PEBKAC issue...
>>
>>
>> Yossi Neiman
>> Cartis Solutions, Inc. - http://www.cartissolutions.com
>>
>>
>> On 02/28/2011 04:07 PM, Anthony Minessale wrote:
>>> The solution to the port 0 issue is to run FS with -nonat so it does
>>> not get confused by broken upnp setup.
>>>
>>>
>>> On Mon, Feb 28, 2011 at 3:54 PM, Yossi Neiman
>>> <freeswitch at cartissolutions.com>    wrote:
>>>> What FS build are you running?  I'd like to compare notes with you on your
>>>> config, as I'm experiencing the RTP port 0 issue with FreeSWITCH Version
>>>> 1.0.head (git-0d8e945 2011-02-16 08-57-43 -0600).  What about inbound?  Do
>>>> you have that set up?
>>>>
>>>> Yossi Neiman
>>>> Cartis Solutions, Inc. - http://www.cartissolutions.com
>>>>
>>>> On 02/28/2011 12:54 AM, envelopes envelopes wrote:
>>>>
>>>> never mind. add this line<action application="ring_ready"/>    and restart FS
>>>>
>>>> fixed the issue.
>>>>
>>>>
>>>>
>>>> On Sun, Feb 27, 2011 at 10:30 PM, envelopes envelopes<sunwood360 at gmail.com>
>>>> wrote:
>>>>> Now I am able to use GV for outbound dialing. However, I don't hear any
>>>>> ringback or not sure whether the other party has answered the call.
>>>>> is there any config variable to set up so that I will be notified if my
>>>>> call is answered?
>>>>>
>>>>> thanks!
>>>>>



From curriegrad2004 at gmail.com  Tue Mar  1 22:13:45 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 1 Mar 2011 11:13:45 -0800
Subject: [Freeswitch-users] Compilation with -march options
In-Reply-To: <AANLkTinJ2Qf06vcrCK+=uwBX+7NNwLd-SxJrz-fi=6rP@mail.gmail.com>
References: <AANLkTi=civOgvpz55hxEifdkcmKWmZABsitj290cCkxN@mail.gmail.com>
	<1296444815434-5976090.post@n2.nabble.com>
	<AANLkTinc=5cmFngN5uJbWPGLQHBjQ65+rNh6cnYNnX-E@mail.gmail.com>
	<AANLkTinJ2Qf06vcrCK+=uwBX+7NNwLd-SxJrz-fi=6rP@mail.gmail.com>
Message-ID: <AANLkTi=QjJ+q7rvyYppum18QtaLMx0MZ2DF5t78D2xSA@mail.gmail.com>

Update on this issue, it was definitely the way how samba-winbind
handles user home directories that caused issues on compiling FS with
a -march option specified.

On Sun, Feb 27, 2011 at 12:43 AM, curriegrad2004
<curriegrad2004 at gmail.com> wrote:
> And I probably forgot to mention that I use winbind for logging on to
> the compilation F13 vm to do this.
>
> On Sun, Feb 27, 2011 at 12:40 AM, curriegrad2004
> <curriegrad2004 at gmail.com> wrote:
>> Sorry to dig up something this old... But last week compiling
>> freeswitch with a march option on a non-root account worked. Git
>> revisions around 24 of this month seemed not to work at all with
>> non-root users compiling the switch as a non-root user. Specific
>> details wise is it stops the compilation at mod_conference with an
>> error 1. Don't have the error message with me as it's pretty late when
>> I'm writing this.
>>
>> Was there a change that caused this to happen or is Fedora 13 the one
>> to blame? Haven't tried re-compiling this on a RHEL 5 or a SL 6 RC1
>> box yet to iron out the possibility of a regression bug.
>>
>> On Sun, Jan 30, 2011 at 7:33 PM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>>>
>>>
>>> curriegrad2004 wrote:
>>>> ..., but I was
>>>> wondering if anybody else out there is also compiling FS with a march
>>>> option and saw some performance increase or adverse effects...
>>> I use -march=armv5te switch to cross-compile FS for an ARM platform sans any
>>> problem.
>>>
>>> -----
>>> don't and stop are the ONLY two 4-letter words considered offensive to men,
>>> but not when used together.
>>> --
>>> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Compilation-with-march-options-tp5975263p5976090.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>


From covici at ccs.covici.com  Tue Mar  1 22:30:55 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 01 Mar 2011 14:30:55 -0500
Subject: [Freeswitch-users] Does FS handle three-way calling on the same
	POTS?
In-Reply-To: <AANLkTikyTQU1JiA1mhEVWxz=GXk_Ragy6U3nZnbNBVwB@mail.gmail.com>
References: <1298970647476-6076537.post@n2.nabble.com>
	<AANLkTikyTQU1JiA1mhEVWxz=GXk_Ragy6U3nZnbNBVwB@mail.gmail.com>
Message-ID: <339.1299007855@ccs.covici.com>

Sadly, fs will not do a hook flash.

Michael Collins <msc at freeswitch.org> wrote:

> What are the instructions from the provider to create the second call?
> Usually with an analog line you have to do a hook flash, then dial some more
> digits, and then do another hook flash.
> 
> -MC
> 
> On Tue, Mar 1, 2011 at 1:10 AM, GillesToo <codecomplete at free.fr> wrote:
> 
> > Hello
> >
> > My ISP offers basic PBX features and free calls through when plugging a
> > phone in an RJ11 on the ADSL modem. I'd like to hook up a PC to the modem
> > with an FXO module, and use three-way calling so that the server calls me
> > back on my cellphone, and then lets me call a number through the server
> > before switching to a conference call. Poor man's callback system :-)
> >
> > I finally got Asterisk to make the first call, and put it on hold, but it
> > stops there: It appears that Asterisk is unable to create a second channel
> > on the same FXO module when a call has been put on hold ("app_dial.c:1310
> > dial_exec_full:Unable to create channel of type 'Dahdi' (cause 0 -
> > Unknown)").
> >
> > I was wondering if someone had successfully used Freeswitch in this
> > scenario?
> >
> > FWIW, here's my extensions.conf:
> > ==============
> > [from_fxo]
> > exten => s,1,Wait(2)
> > exten => s,n,Set(GLOBAL(CID)=${CALLERID(num)})
> > exten => s,n,Hangup()
> >
> > ;script waits and creates callfile
> > exten => h,1,system(/var/tmp/test10.lua ${CID}&)
> >
> > ;context used by callfile
> > [callback]
> > exten => start,1,NoOp(In callback, CID is ${CID})
> > ;how to wait until remote phone picked up?
> > exten => start,n,Wait(5)
> > exten => start,n,Answer()
> > exten => start,n,Playback(tt-monkeysintro)
> >
> > exten => start,n,Flash()
> >
> > ;BAD exten => start,n,Dial(Dahdi/1/${GSM})
> > ;does ring second number, but no actual voice channel
> > exten => start,n,SendDTMF(456789)
> > ==============
> >
> > Thank you.
> >
> > --
> > View this message in context:
> > http://freeswitch-users.2379917.n2.nabble.com/Does-FS-handle-three-way-calling-on-the-same-POTS-tp6076537p6076537.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From Nabble at slickdeals.endjunk.com  Tue Mar  1 22:52:57 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 1 Mar 2011 11:52:57 -0800 (PST)
Subject: [Freeswitch-users] freeswitch -nc -nosql
In-Reply-To: <AANLkTimuQDPXQQHLvroxd7ttWYDZxiPAxv4At_MGBqHO@mail.gmail.com>
References: <1298688216720-6066718.post@n2.nabble.com>
	<AANLkTikjkTPk3A2FmoRo32WhKxNFaX5YL7Wxezaa6zqG@mail.gmail.com>
	<1298815506369-6070268.post@n2.nabble.com>
	<AANLkTin8ZRoFSzFoiaxNKg0VDvXi5YebMe_Z6mpyFRgJ@mail.gmail.com>
	<1298820079407-6070409.post@n2.nabble.com>
	<1298864829635-6071920.post@n2.nabble.com>
	<AANLkTimxG==dquT_oeFSiPnxSN6g5mncQ5Gf1GNg_u6U@mail.gmail.com>
	<1298997953584-6077875.post@n2.nabble.com>
	<AANLkTimuQDPXQQHLvroxd7ttWYDZxiPAxv4At_MGBqHO@mail.gmail.com>
Message-ID: <1299009177605-6078480.post@n2.nabble.com>


Anthony Minessale wrote:
> You said it doesn't crash anymore in a previous email didn't you?
> I can't keep track of 200 people's issues via email.Yes and it works only
> if I use the -nosql switch to disable SQL. Unfortunately, disabling SQL
> also disables the help menu and among other things that depends on SQL,
> let alone your plan to remove the -nosql switch in the future.


Did you update to HEAD and reproduce it again?
Yes. Have a look below:
2011-03-01 14:43:15.513489 [CONSOLE] switch_core.c:1765 
   _____              ______        _____ _____ ____ _   _  
  |  ___| __ ___  ___/ ___\ \      / /_ _|_   _/ ___| | | | 
  | |_ | '__/ _ \/ _ \___ \\ \ /\ / / | |  | || |   | |_| | 
  |  _|| | |  __/  __/___) |\ V  V /  | |  | || |___|  _  | 
  |_|  |_|  \___|\___|____/  \_/\_/  |___| |_| \____|_| |_| 

************************************************************
* Anthony Minessale II, Michael Jerris, Brian West, Others *
* FreeSWITCH (http://www.freeswitch.org)                   *
* Paypal Donations Appreciated: paypal at freeswitch.org      *
* Brought to you by ClueCon http://www.cluecon.com/        *
************************************************************

2011-03-01 14:43:15.513639 [CONSOLE] switch_core.c:1768 
FreeSWITCH Version 1.0.head (git-2044a74 2011-03-01 10-52-21 -0600) Started.
Max Sessions[1000]
Session Rate[30]
SQL [Enabled]
Segmentation fault
@DockStar:/#

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-nc-nosql-tp6066718p6078480.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Tue Mar  1 23:03:52 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 1 Mar 2011 12:03:52 -0800 (PST)
Subject: [Freeswitch-users] Thoughts about mod_conference enter-sound,
 when wait-moderator has been set
In-Reply-To: <AANLkTi=pCWWarPvpeemzphhNNDGFZg84om7F2Txsjez3@mail.gmail.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C58B2BBAB93@cooper>
	<AANLkTi=pCWWarPvpeemzphhNNDGFZg84om7F2Txsjez3@mail.gmail.com>
Message-ID: <1299009832210-6078512.post@n2.nabble.com>


mercutioviz wrote:
> Or a really short sound, like a 100ms ding sound.
With lots of participants in the conference room, it will start to feel like
sitting in front of a jackpot machine in Las Vegas! DING DING DING DING DING
DING ...   

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Thoughts-about-mod-conference-enter-sound-when-wait-moderator-has-been-set-tp6077264p6078512.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Tue Mar  1 23:04:28 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 1 Mar 2011 20:04:28 +0000
Subject: [Freeswitch-users] HANGUP_CAUSE after bridge in outbound ESL
In-Reply-To: <262550.10523.qm@web33503.mail.mud.yahoo.com>
References: <262550.10523.qm@web33503.mail.mud.yahoo.com>
Message-ID: <E14CC11E-6F0A-4C8D-8860-F7B2063C0C5B@gmail.com>

Yes... Problem is how to get it. One way is the CDR, but that's not through ESL. Events are another - it'll be sent as a header. There's the uuid_getvar api but that won't work if the session has been destroyed.

Steve on iPhone

On 1 Mar 2011, at 14:04, Diego Toro <dftoro at yahoo.com> wrote:

> Hi, check originate_disposition variable
> 
> http://wiki.freeswitch.org/wiki/Extension_Status_Example
> http://wiki.freeswitch.org/wiki/Variable_originate_disposition
> 
> Diego Toro
> http://voipensando.blogspot.com/
> 
> --- On Tue, 3/1/11, Steven Ayre <steveayre at gmail.com> wrote:
> 
> From: Steven Ayre <steveayre at gmail.com>
> Subject: Re: [Freeswitch-users] HANGUP_CAUSE after bridge in outbound ESL
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Tuesday, March 1, 2011, 4:08 AM
> 
> The correct way is to use events. You can register for just the hangup event.
> 
> -Steve
> 
> 
> On 28 February 2011 22:43, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> > Hi All
> >
> > What is correct method to get bridge status after execute("bridge") in ESL?
> > I have inbound call that gets bridged to SIP endpoint. I need to know
> > whether it was ORIGINATOR_CANCEL or BUSY or something else, but if I use
> > sync, seems I can't determine ORIGINATOR_CANCEL status, because it is set on
> > a-leg which is destroyed first after hangup.
> >
> > Is there a way to get bridge status without messing with events and event
> > source filtering by channel uuid or event type?
> >
> > --
> > Best regards,
> >
> > Dmitry Sytchev,
> > IT Engineer
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From peter.olsson at visionutveckling.se  Tue Mar  1 23:19:40 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 1 Mar 2011 21:19:40 +0100
Subject: [Freeswitch-users] Thoughts about mod_conference enter-sound,
 when wait-moderator has been set
In-Reply-To: <1299009832210-6078512.post@n2.nabble.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C58B2BBAB93@cooper>
	<AANLkTi=pCWWarPvpeemzphhNNDGFZg84om7F2Txsjez3@mail.gmail.com>,
	<1299009832210-6078512.post@n2.nabble.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2C494D2@cooper>

Yeah, might be fun :)

I think I'll add a patch for this, so it's possible to override current behaviour.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r mazilo [Nabble at slickdeals.endjunk.com]
Skickat: den 1 mars 2011 21:03
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Thoughts about mod_conference enter-sound, when wait-moderator has been set

mercutioviz wrote:
> Or a really short sound, like a 100ms ding sound.
With lots of participants in the conference room, it will start to feel like
sitting in front of a jackpot machine in Las Vegas! DING DING DING DING DING
DING ...

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Thoughts-about-mod-conference-enter-sound-when-wait-moderator-has-been-set-tp6077264p6078512.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

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

!DSPAM:4d6d522e32761009827340!



From andreas at tuerpe-net.de  Tue Mar  1 22:16:00 2011
From: andreas at tuerpe-net.de (Andreas Tuerpe)
Date: Tue, 01 Mar 2011 20:16:00 +0100
Subject: [Freeswitch-users] route inbound - based on sip account
Message-ID: <4D6D45F0.5060205@tuerpe-net.de>

Hallo FreeSWITCH Users,

I use FS V.0.9.6 on pfSense
see on -> [http://doc.pfsense.org/index.php/FreeSWITCH]

Symptom:
German ISP "Portunity" forward inbound calls without any destination_number.


ISP solution tip:
FS to register over a second sip account to the provider twice.
Any account has a separate number.
Based on the channel over which the call come in, I have to decide which
number is called.

So I need help, which condition assignment must use - howto ???
- which fields can I use?
- which syntax I have to use?


<!--  here my intention ... ???

   <extension name="test_did">
     <condition field="channel over which the call come in"
expression="sip-account-1">
       <action application="set" data="called_number=111111"/>
       <action application="transfer" data="1001 XML default"/>
     <condition field="channel over which the call come in"
expression="sip-account-2">
       <action application="set" data="called_number=222222"/>
       <action application="transfer" data="1002 XML default"/>
     </condition>
   </extension>

??? -->

thanks in advance
   tuerpean



From msc at freeswitch.org  Wed Mar  2 00:07:37 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 1 Mar 2011 13:07:37 -0800
Subject: [Freeswitch-users] Does FS handle three-way calling on the same
	POTS?
In-Reply-To: <339.1299007855@ccs.covici.com>
References: <1298970647476-6076537.post@n2.nabble.com>
	<AANLkTikyTQU1JiA1mhEVWxz=GXk_Ragy6U3nZnbNBVwB@mail.gmail.com>
	<339.1299007855@ccs.covici.com>
Message-ID: <AANLkTik_1_1vuqqgOYfijb4vnRS6cop52SvZsuDskSEE@mail.gmail.com>

On Tue, Mar 1, 2011 at 11:30 AM, <covici at ccs.covici.com> wrote:

> Sadly, fs will not do a hook flash.
>

FreeSWITCH won't or FreeTDM won't?
-MC
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From frankie.k.yiu at gmail.com  Wed Mar  2 00:01:26 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Tue, 1 Mar 2011 13:01:26 -0800
Subject: [Freeswitch-users] How to create IVR application in C#?
Message-ID: <AANLkTi=E4uTFjExb7jbJUJ8nmG_7UdxDvO=tuMmwT5OZ@mail.gmail.com>

Hi there,

I am newbie to FreeSwitch and I have question about creating an IVR
application in C#, with a possibly of using VoiceXML.  Could someone please
points me to how I can get started or any example that I can look at?

Thanks a lot!!!

Frankie
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From gmaruzz at gmail.com  Wed Mar  2 00:21:24 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 1 Mar 2011 22:21:24 +0100
Subject: [Freeswitch-users] freeswitch -nc -nosql
In-Reply-To: <1299009177605-6078480.post@n2.nabble.com>
References: <1298688216720-6066718.post@n2.nabble.com>
	<AANLkTikjkTPk3A2FmoRo32WhKxNFaX5YL7Wxezaa6zqG@mail.gmail.com>
	<1298815506369-6070268.post@n2.nabble.com>
	<AANLkTin8ZRoFSzFoiaxNKg0VDvXi5YebMe_Z6mpyFRgJ@mail.gmail.com>
	<1298820079407-6070409.post@n2.nabble.com>
	<1298864829635-6071920.post@n2.nabble.com>
	<AANLkTimxG==dquT_oeFSiPnxSN6g5mncQ5Gf1GNg_u6U@mail.gmail.com>
	<1298997953584-6077875.post@n2.nabble.com>
	<AANLkTimuQDPXQQHLvroxd7ttWYDZxiPAxv4At_MGBqHO@mail.gmail.com>
	<1299009177605-6078480.post@n2.nabble.com>
Message-ID: <AANLkTikBZimcGkTyoKiQN8mFeXsgwrftCN3Z1KhSq_0-@mail.gmail.com>

Hi Mazilo,

so, please open a Jira issue, and try to add it all the info required,
and all the info you can gather.

Please have a look at the wiki page about how to report a bug, it will
show you how to gather all the info required in case of a segfault.

-giovanni

On Tue, Mar 1, 2011 at 8:52 PM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
> Anthony Minessale wrote:
>> You said it doesn't crash anymore in a previous email didn't you?
>> I can't keep track of 200 people's issues via email.Yes and it works only
>> if I use the -nosql switch to disable SQL. Unfortunately, disabling SQL
>> also disables the help menu and among other things that depends on SQL,
>> let alone your plan to remove the -nosql switch in the future.
>
>
> Did you update to HEAD and reproduce it again?
> Yes. Have a look below:
> 2011-03-01 14:43:15.513489 [CONSOLE] switch_core.c:1765
> ? _____ ? ? ? ? ? ? ?______ ? ? ? ?_____ _____ ____ _ ? _
> ?| ?___| __ ___ ?___/ ___\ \ ? ? ?/ /_ _|_ ? _/ ___| | | |
> ?| |_ | '__/ _ \/ _ \___ \\ \ /\ / / | | ?| || | ? | |_| |
> ?| ?_|| | | ?__/ ?__/___) |\ V ?V / ?| | ?| || |___| ?_ ?|
> ?|_| ?|_| ?\___|\___|____/ ?\_/\_/ ?|___| |_| \____|_| |_|
>
> ************************************************************
> * Anthony Minessale II, Michael Jerris, Brian West, Others *
> * FreeSWITCH (http://www.freeswitch.org) ? ? ? ? ? ? ? ? ? *
> * Paypal Donations Appreciated: paypal at freeswitch.org ? ? ?*
> * Brought to you by ClueCon http://www.cluecon.com/ ? ? ? ?*
> ************************************************************
>
> 2011-03-01 14:43:15.513639 [CONSOLE] switch_core.c:1768
> FreeSWITCH Version 1.0.head (git-2044a74 2011-03-01 10-52-21 -0600) Started.
> Max Sessions[1000]
> Session Rate[30]
> SQL [Enabled]
> Segmentation fault
> @DockStar:/#
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-nc-nosql-tp6066718p6078480.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From mthakershi at gmail.com  Wed Mar  2 01:42:47 2011
From: mthakershi at gmail.com (Malay Thakershi)
Date: Tue, 1 Mar 2011 16:42:47 -0600
Subject: [Freeswitch-users] How to create IVR application in C#?
In-Reply-To: <AANLkTi=E4uTFjExb7jbJUJ8nmG_7UdxDvO=tuMmwT5OZ@mail.gmail.com>
References: <AANLkTi=E4uTFjExb7jbJUJ8nmG_7UdxDvO=tuMmwT5OZ@mail.gmail.com>
Message-ID: <AANLkTimHLyTh38G+zyLj7L-3ieEBPJ+3tB7vSAzHMXQ0@mail.gmail.com>

mod_managed could be an option.

http://wiki.freeswitch.org/wiki/Mod_managed

<http://wiki.freeswitch.org/wiki/Mod_managed>It allows you to use most
native FS features from C# managed code.

Malay

On Tue, Mar 1, 2011 at 3:01 PM, Frankie Yiu <frankie.k.yiu at gmail.com> wrote:

> Hi there,
>
> I am newbie to FreeSwitch and I have question about creating an IVR
> application in C#, with a possibly of using VoiceXML.  Could someone please
> points me to how I can get started or any example that I can look at?
>
> Thanks a lot!!!
>
> Frankie
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From codecomplete at free.fr  Wed Mar  2 01:52:28 2011
From: codecomplete at free.fr (GillesToo)
Date: Tue, 1 Mar 2011 14:52:28 -0800 (PST)
Subject: [Freeswitch-users] Does FS handle three-way calling on the same
	POTS?
In-Reply-To: <AANLkTikyTQU1JiA1mhEVWxz=GXk_Ragy6U3nZnbNBVwB@mail.gmail.com>
References: <1298970647476-6076537.post@n2.nabble.com>
	<AANLkTikyTQU1JiA1mhEVWxz=GXk_Ragy6U3nZnbNBVwB@mail.gmail.com>
Message-ID: <1299019948759-6079027.post@n2.nabble.com>


mercutioviz wrote:
> What are the instructions from the provider to create the second call?
> Usually with an analog line you have to do a hook flash, then dial some
> more digits, and then do another hook flash

Thanks for the help. Asterisk doesn't allow using Dial() to call a second
number after the original call was put on hold. A work-around is using
SendDTMF() to dial the second number, but it appears there's no way to get
call progress and know if the remote party has answered.

Does FS provide a better way than this hack? I really need to use three-way
calling on the same line since all calls are free this way.

Thank you.


-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Does-FS-handle-three-way-calling-on-the-same-POTS-tp6076537p6079027.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Wed Mar  2 02:07:28 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 1 Mar 2011 15:07:28 -0800
Subject: [Freeswitch-users] route inbound - based on sip account
In-Reply-To: <4D6D45F0.5060205@tuerpe-net.de>
References: <4D6D45F0.5060205@tuerpe-net.de>
Message-ID: <AANLkTimmuZSR5pPLDd9B6cXjgE9jQ7TDHibWQ_T4Fjx-@mail.gmail.com>

The FS on pfSense is pretty old, but if all you are working on is a simple
routing issue your best bet is to add the "info" app in the public context.
Somewhere near the top of public.xml just add this:

<extension name="quick info dump" continue="true">
  <condition>
    <action application="info"/>
  </condition>
</extension>

Save that, press F6 (or do reloadxml) and then make a test inbound call.
Watch the console - you'll see a TON of information. Look through the pieces
of data that are displayed. You should be able to find something to key off
of. Once you've done that then go read up on creating your dialplan here:

http://wiki.freeswitch.org/wiki/Dialplan_XML#Condition

In fact, that whole page is important in understanding the XML dialplan. I
would read it more than once.

-MC

On Tue, Mar 1, 2011 at 11:16 AM, Andreas Tuerpe <andreas at tuerpe-net.de>wrote:

> Hallo FreeSWITCH Users,
>
> I use FS V.0.9.6 on pfSense
> see on -> [http://doc.pfsense.org/index.php/FreeSWITCH]
>
> Symptom:
> German ISP "Portunity" forward inbound calls without any
> destination_number.
>
>
> ISP solution tip:
> FS to register over a second sip account to the provider twice.
> Any account has a separate number.
> Based on the channel over which the call come in, I have to decide which
> number is called.
>
> So I need help, which condition assignment must use - howto ???
> - which fields can I use?
> - which syntax I have to use?
>
>
> <!--  here my intention ... ???
>
>   <extension name="test_did">
>     <condition field="channel over which the call come in"
> expression="sip-account-1">
>       <action application="set" data="called_number=111111"/>
>       <action application="transfer" data="1001 XML default"/>
>     <condition field="channel over which the call come in"
> expression="sip-account-2">
>       <action application="set" data="called_number=222222"/>
>       <action application="transfer" data="1002 XML default"/>
>     </condition>
>   </extension>
>
> ??? -->
>
> thanks in advance
>   tuerpean
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Nabble at slickdeals.endjunk.com  Wed Mar  2 04:48:49 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 1 Mar 2011 17:48:49 -0800 (PST)
Subject: [Freeswitch-users] mod_dingaling and google voice
In-Reply-To: <4D6D44B7.6070900@cartissolutions.com>
References: <AANLkTin_dFj9ygx3TAmExeN9-pDFNCuZPGZ_Uc2d7o1W@mail.gmail.com>
	<AANLkTimUr3=w_fj_k643ChW93Q_YTm63R_qFyBO9EoQV@mail.gmail.com>
	<4D6C1978.8040702@cartissolutions.com>
	<AANLkTim5TpcQeYSC_c6ZH9WMoJnq1aNvYViwUBkZr=fv@mail.gmail.com>
	<4D6C288B.4050801@cartissolutions.com>
	<AANLkTikncVxc7ebrzR4bHrzeHARKVU_Q+aRpzMTM9RwV@mail.gmail.com>
	<4D6D44B7.6070900@cartissolutions.com>
Message-ID: <1299030529099-6079399.post@n2.nabble.com>


Yossi Neiman wrote:
> Sofia UA's running on both 99.X.X.X:5060 and 192.X.X.X:5060.  Machine FS 
> runs on is an iptables NAT/Firewall/router.  SIP calls to/from the 
> outside world work fine otherwise.
I am lost here. You have two FS machines and one on a public IP Address
while the other on a private IP Address? Both are having problems with GV
incoming calls?

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-dingaling-and-google-voice-tp6072163p6079399.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From moises.silva at gmail.com  Wed Mar  2 05:12:16 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 1 Mar 2011 21:12:16 -0500
Subject: [Freeswitch-users] FreeTDM errors in log file
In-Reply-To: <AANLkTinZkJPS+gywSp0EYu_3Wx=C=WEqRK_QSnfyz4dw@mail.gmail.com>
References: <AANLkTinZkJPS+gywSp0EYu_3Wx=C=WEqRK_QSnfyz4dw@mail.gmail.com>
Message-ID: <AANLkTinYXEwnDZcKvibuPb5n6FT=38EjwCmnUVR5riLY@mail.gmail.com>

Stephen,

Can you open a bug report in jira for this and assign it to me?

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 Mon, Feb 28, 2011 at 7:16 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> I'm using Sangoma A108 board in my FS installation and my log is full of
> warnings as:
>
> 2011-03-01 00:18:14.947380 [WARNING] mod_freetdm.c:434 [s17c22][17:22] VETO
> state change from PROGRESS_MEDIA to PROCEED
> 2011-03-01 00:18:53.151673 [WARNING] mod_freetdm.c:434 [s3c11][3:11] VETO
> state change from PROGRESS_MEDIA to PROCEED
> 2011-03-01 00:19:23.001140 [WARNING] mod_freetdm.c:434 [s23c29][23:29] VETO
> state change from RINGING to PROCEED
> 2011-03-01 00:19:40.862103 [WARNING] mod_freetdm.c:434 [s3c21][3:21] VETO
> state change from PROGRESS_MEDIA to PROCEED
> 2011-03-01 00:19:46.022154 [WARNING] mod_freetdm.c:434 [s3c21][3:21] VETO
> state change from PROGRESS_MEDIA to PROCEED
> 2011-03-01 00:19:50.608224 [WARNING] mod_freetdm.c:434 [s22c24][22:24] VETO
> state change from PROGRESS_MEDIA to PROCEED
> 2011-03-01 00:19:52.422221 [WARNING] mod_freetdm.c:434 [s3c21][3:21] VETO
> state change from PROGRESS_MEDIA to PROCEED
> 2011-03-01 00:21:36.461338 [WARNING] mod_freetdm.c:434 [s3c31][3:31] VETO
> state change from PROGRESS_MEDIA to PROCEED
> 2011-03-01 00:21:36.981455 [WARNING] mod_freetdm.c:434 [s3c30][3:30] VETO
> state change from PROGRESS_MEDIA to PROCEED
>
> Another kind of messages that are in my log file are:
>
> 2011-03-01 00:53:14.427853 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:760
> [SNGISDN Q931] s16: Protocol: Unknown Event Code(2): Incomp Msg(276)
> 2011-03-01 00:53:18.428846 [CRIT] ftmod_sangoma_isdn_stack_hndl.c:557
> [s16c23][16:23] Received DISCONNECT in an invalid state (TERMINATING)
> 2011-03-01 00:53:18.428846 [WARNING] ftmod_sangoma_isdn_stack_hndl.c:562
> [s16c23][16:23] Why bother changing state from TERMINATING to TERMINATING
> 2011-03-01 00:54:08.607351 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:760
> [SNGISDN Q931] s17: Protocol: Unknown Event Code(2): Incomp Msg(276)
> 2011-03-01 00:54:12.607389 [CRIT] ftmod_sangoa_isdn_stack_hndl.c:557
> [s17c20][17:20] Received DISCONNECT in an invalid state (TERMINATING)
> 2011-03-01 00:54:12.607389 [WARNING] ftmod_sangoma_isdn_stack_hndl.c:562
> [s17c20][17:20] Why bother changing state from TERMINATING to TERMINATING
> 2011-03-01 00:56:45.595869 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:760
> [SNGISDN Q931] s16: Protocol: Unknown Event Code(2): Incomp Msg(276)
> 2011-03-01 00:56:49.596913 [CRIT] ftmod_sangoma_isdn_stack_hndl.c:557
> [s16c23][16:23] Received DISCONNECT in an invalid state (TERMINATING)
> 2011-03-01 00:56:49.596913 [WARNING] ftmod_sangoma_isdn_stack_hndl.c:562
> [s16c23][16:23] Why bother changing state from TERMINATING to TERMINATING
> 2011-03-01 00:59:23.036314 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:760
> [SNGISDN Q931] s17: Protocol: Unknown Event Code(2): Incomp Msg(276)
> 2011-03-01 00:59:27.037327 [CRIT] ftmod_sangoma_isdn_stack_hndl.c:557
> [s17c20][17:20] Received DISCONNECT in an invalid state (TERMINATING)
> 2011-03-01 00:59:27.037327 [WARNING] ftmod_sangoma_isdn_stack_hndl.c:562
> [s17c20][17:20] Why bother changing state from TERMINATING to TERMINATING
>
> I'm using latest git of Freeswitch, Wanpipe 3.5.18 and lib Sangoma isdn
> 7.3.0.
>
> Attached to this email the .pcap related to the call that generate the
> critical error (to me it seems that is missed a CALL_PROCEEDING message).
>
> Stephen
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bobc at devassert.com  Wed Mar  2 08:34:14 2011
From: bobc at devassert.com (Bob Coleman)
Date: Wed, 2 Mar 2011 18:34:14 +1300
Subject: [Freeswitch-users] How to create IVR application in C#?
In-Reply-To: <AANLkTi=E4uTFjExb7jbJUJ8nmG_7UdxDvO=tuMmwT5OZ@mail.gmail.com>
References: <AANLkTi=E4uTFjExb7jbJUJ8nmG_7UdxDvO=tuMmwT5OZ@mail.gmail.com>
Message-ID: <AANLkTingyqfiHtgw6NSbrKGguwxgzsHk3KyHf+oN_WNc@mail.gmail.com>

Another easy way is to use mod event socket

When you download the source, there is a libs/esl/managed folder that
has the ESL project for .NET

You can use this wrapper to talk to FreeSWITCH and write an IVR app
without much trouble.

The event socket route is really nice, I have built a complete windows
ivr stack using FreeSWITCH and help from the guys here, very reliable
and no maintenance.

Bob

On Wed, Mar 2, 2011 at 10:01 AM, Frankie Yiu <frankie.k.yiu at gmail.com> wrote:
> Hi there,
>
> I am newbie to FreeSwitch and I have question about creating an IVR
> application in C#, with a possibly of using VoiceXML.? Could someone please
> points me to how I can get started or any example that I can look at?
>


From kbdfck at gmail.com  Wed Mar  2 10:30:42 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Wed, 2 Mar 2011 10:30:42 +0300
Subject: [Freeswitch-users] HANGUP_CAUSE after bridge in outbound ESL
In-Reply-To: <E14CC11E-6F0A-4C8D-8860-F7B2063C0C5B@gmail.com>
References: <262550.10523.qm@web33503.mail.mud.yahoo.com>
	<E14CC11E-6F0A-4C8D-8860-F7B2063C0C5B@gmail.com>
Message-ID: <AANLkTimWUnAsxRPGG5CbjKd7wsymN-xtYHPMfGMCa6U-@mail.gmail.com>

Yes, this was exactly my problem with session variables. Seems I've been
using freeswitch git version where perl ESL event filtering was broken. I
saw a thread about this on mailing list. I'll try with latest git.

2011/3/1 Steven Ayre <steveayre at gmail.com>

> Yes... Problem is how to get it. One way is the CDR, but that's not through
> ESL. Events are another - it'll be sent as a header. There's the uuid_getvar
> api but that won't work if the session has been destroyed.
>
> Steve on iPhone
>
> On 1 Mar 2011, at 14:04, Diego Toro <dftoro at yahoo.com> wrote:
>
> Hi, check originate_disposition variable
>
> http://wiki.freeswitch.org/wiki/Extension_Status_Example
> http://wiki.freeswitch.org/wiki/Variable_originate_disposition
>
> Diego Toro
> http://voipensando.blogspot.com/
>
> --- On *Tue, 3/1/11, Steven Ayre <steveayre at gmail.com>* wrote:
>
>
> From: Steven Ayre <steveayre at gmail.com>
> Subject: Re: [Freeswitch-users] HANGUP_CAUSE after bridge in outbound ESL
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Tuesday, March 1, 2011, 4:08 AM
>
> The correct way is to use events. You can register for just the hangup
> event.
>
> -Steve
>
>
> On 28 February 2011 22:43, Dmitry Sytchev <kbdfck at gmail.com<http://mc/compose?to=kbdfck at gmail.com>>
> wrote:
> > Hi All
> >
> > What is correct method to get bridge status after execute("bridge") in
> ESL?
> > I have inbound call that gets bridged to SIP endpoint. I need to know
> > whether it was ORIGINATOR_CANCEL or BUSY or something else, but if I use
> > sync, seems I can't determine ORIGINATOR_CANCEL status, because it is set
> on
> > a-leg which is destroyed first after hangup.
> >
> > Is there a way to get bridge status without messing with events and event
> > source filtering by channel uuid or event type?
> >
> > --
> > Best regards,
> >
> > Dmitry Sytchev,
> > IT Engineer
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From mayamatakeshi at gmail.com  Wed Mar  2 11:08:10 2011
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Wed, 2 Mar 2011 17:08:10 +0900
Subject: [Freeswitch-users] How to use uuid_chat and uuid_display?
Message-ID: <AANLkTinJuea69oME32J7q==jRPBLWK9pnU1pVkGmbtEE@mail.gmail.com>

What are the conditions for uuid_chat and uuid_display to work? I have two
channels bridged (using intercept) but when i try with them in the FS cli,
although I get a positive reply, nothing is sent to the SIP agents (i have
enabled siptrace but also confirmed using tcpdump):


freeswitch at IPX050> sofia loglevel nua 9

Sofia log level for component [nua] has been set to [9]
freeswitch at IPX050> sofia profile internal siptrace on
nua: nua_set_params: entering
nua((nil)): sent signal r_set_params

Enabled sip debugging on internal
nua((nil)): recv signal r_set_params
nua: nua_stack_set_params: entering
nua((nil)): event r_set_params 200 OK
nua: nua_application_event: entering
freeswitch at IPX050> uuid_chat 19d300fa-3d95-4b09-a2a7-2ffefa708b3c test

+OK

2011-03-02 16:59:23.665054 [DEBUG] switch_core_session.c:876 Send signal
sofia/internal/term6 at dev2.basix5.ne.jp [BREAK]
freeswitch at IPX050> uuid_display 19d300fa-3d95-4b09-a2a7-2ffefa708b3c test

+OK Success



The header Allow in the INVITE from the terminals is like this:
Allow: SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK,
INFO, MESSAGE

I have also tried application send_display in the dialplan but nothing
happens.

br,
takeshi
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From julf at julf.com  Wed Mar  2 11:54:31 2011
From: julf at julf.com (Johan Helsingius)
Date: Wed, 02 Mar 2011 09:54:31 +0100
Subject: [Freeswitch-users] Changing voicemail greeting to pronounce
 (instead of spelling out) id/number
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B2C494CC@cooper>
References: <4D6D19F1.4040105@julf.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2C494CC@cooper>
Message-ID: <4D6E05C7.7090304@julf.com>

Peter,

> You will need some kind of TTS engine to get this working, however, I'm not sure if any changes must be made to mod_voicemail as well.

So I guess per-user custom recorded messages is pretty much the way to go :(

	Julf



From tayeb.meftah at gmail.com  Wed Mar  2 12:42:27 2011
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Wed, 02 Mar 2011 10:42:27 +0100
Subject: [Freeswitch-users] route inbound - based on sip account
In-Reply-To: <4D6D45F0.5060205@tuerpe-net.de>
References: <4D6D45F0.5060205@tuerpe-net.de>
Message-ID: <4D6E1103.9070600@gmail.com>

please try providing a siptrace
sofia global siptrace on
Le 01/03/2011 20:16, Andreas Tuerpe a ?crit :
> Hallo FreeSWITCH Users,
>
> I use FS V.0.9.6 on pfSense
> see on ->  [http://doc.pfsense.org/index.php/FreeSWITCH]
>
> Symptom:
> German ISP "Portunity" forward inbound calls without any destination_number.
>
>
> ISP solution tip:
> FS to register over a second sip account to the provider twice.
> Any account has a separate number.
> Based on the channel over which the call come in, I have to decide which
> number is called.
>
> So I need help, which condition assignment must use - howto ???
> - which fields can I use?
> - which syntax I have to use?
>
>
> <!--  here my intention ... ???
>
>     <extension name="test_did">
>       <condition field="channel over which the call come in"
> expression="sip-account-1">
>         <action application="set" data="called_number=111111"/>
>         <action application="transfer" data="1001 XML default"/>
>       <condition field="channel over which the call come in"
> expression="sip-account-2">
>         <action application="set" data="called_number=222222"/>
>         <action application="transfer" data="1002 XML default"/>
>       </condition>
>     </extension>
>
> ??? -->
>
> thanks in advance
>     tuerpean
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
phone: +13477595883



From u2nsam at gmail.com  Wed Mar  2 12:48:02 2011
From: u2nsam at gmail.com (Sam)
Date: Wed, 2 Mar 2011 15:18:02 +0530
Subject: [Freeswitch-users] API command
Message-ID: <AANLkTim-dEY_DOLQPUf9apOwFyntFsNVPhnvUXOp9_2d@mail.gmail.com>

HI,


Suppose if i want to place an external caller in the conference so i should
use the below command ?

originate freetdm/wp1/a/9322273640  67287006 context &bridge 7050

where 7050 leads to conference .


Regds
Sam
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From mbsip at gazeta.pl  Wed Mar  2 14:30:35 2011
From: mbsip at gazeta.pl (Maciej Bylica)
Date: Wed, 2 Mar 2011 12:30:35 +0100
Subject: [Freeswitch-users] Conference - control actions problem
Message-ID: <AANLkTimq9dYbq8ztbQ5dJmY2Y+2mK5mKzM78UirEzzYn@mail.gmail.com>

Hello,

I am running FreeSWITCH Version 1.0.head (git-06988e1 2011-02-20
15-10-02 -0600) and have some problem with conference module and
respective control actions digits.
To give you an overview of how it is configured, there is lua script
fired in public.xml.
The script checks pin code, conf room and at the end run
"session:execute("conference", string.format("%s at fsconf", conf_num))".
Now coming back to the problem, there are few scenations i would like to cover.

1. When two or more people are in conference and one of them is trying
to push # button (RFC2833) then that person is leaving the conference
and is hanged up (not released, no BYE). The person who left the
conference cannot join again or perform other actions without making
the same phone call again.

2011-03-02 11:42:08.863139 [DEBUG] mod_conference.c:2414 Setup timer
soft success interval: 20  samples: 160
2011-03-02 11:42:12.283051 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:480
2011-03-02 11:42:12.323050 [DEBUG] mod_conference.c:2659 Channel
leaving conference, cause: NONE

2. The same as above, but * digit is entered. The person is muted and
then unmuted. Next try with * fires muted but this is the point where
the problem manifest itself.
The procedure of unmuting UA fails. Below you may find relevant logs.
The UA is pressing * but without success (no RTP RECV DTMF *).

freeswitch at internal> 2011-03-02 11:51:13.423651 [DEBUG]
switch_rtp.c:3237 RTP RECV DTMF *:504
2011-03-02 11:51:13.443650 [DEBUG] mod_conference.c:3125 Queueing file
'/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav'
for play
2011-03-02 11:51:18.163532 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF *:544
2011-03-02 11:51:18.183531 [DEBUG] mod_conference.c:3125 Queueing file
'/usr/local/freeswitch/sounds/en/us/callie/conference/conf-unmuted.wav'
for play
2011-03-02 11:51:23.763391 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF *:504
2011-03-02 11:51:23.783390 [DEBUG] mod_conference.c:3125 Queueing file
'/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav'
for play

3. The same as above but with 0 digit

2011-03-02 12:04:21.064862 [DEBUG] mod_conference.c:2414 Setup timer
soft success interval: 20  samples: 160
2011-03-02 12:04:22.863736 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 0:880
2011-03-02 12:04:22.883736 [DEBUG] mod_conference.c:3125 Queueing file
'/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav'
for play

then i cannot get back to the conference


4. The as above but with 6 digit

2011-03-02 12:15:03.244609 [DEBUG] mod_conference.c:2414 Setup timer
soft success interval: 20  samples: 160
2011-03-02 12:15:21.602919 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 6:504
2011-03-02 12:15:22.602893 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 6:800
2011-03-02 12:15:24.263851 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 6:864


I have disabled all control digits, but the problem still persist. No
action is fired but is looks like UA is being muted (micro is off, but
i can hear the other person).
2011-03-02 12:24:05.322935 [DEBUG] mod_conference.c:2414 Setup timer
soft success interval: 20  samples: 160
2011-03-02 12:24:08.663851 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:480
2011-03-02 12:24:11.223787 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:504
2011-03-02 12:24:13.703725 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:480
2011-03-02 12:24:16.043666 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:520


Has anybody similar experience or know where the problem may be located.

Regards,
Maciej


From rajesh.npnr at yahoo.com  Wed Mar  2 14:51:19 2011
From: rajesh.npnr at yahoo.com (rex.alex)
Date: Wed, 2 Mar 2011 03:51:19 -0800 (PST)
Subject: [Freeswitch-users] API command
In-Reply-To: <AANLkTim-dEY_DOLQPUf9apOwFyntFsNVPhnvUXOp9_2d@mail.gmail.com>
References: <AANLkTim-dEY_DOLQPUf9apOwFyntFsNVPhnvUXOp9_2d@mail.gmail.com>
Message-ID: <1299066679731-6080531.post@n2.nabble.com>

I used the following command to place an external caller directly to the
conference and it worked. Try this..

originate sofia/gateway/gw_name/9322273640 &conference(7050)

Regards,
Rex

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/API-command-tp6080264p6080531.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From peter.olsson at visionutveckling.se  Wed Mar  2 15:01:02 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 2 Mar 2011 13:01:02 +0100
Subject: [Freeswitch-users] Conference - control actions problem
In-Reply-To: <AANLkTimq9dYbq8ztbQ5dJmY2Y+2mK5mKzM78UirEzzYn@mail.gmail.com>
References: <AANLkTimq9dYbq8ztbQ5dJmY2Y+2mK5mKzM78UirEzzYn@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2BBAF1D@cooper>

If you exit the conference (#) the user will still be in your lua script, so you would have to do whatever you want it to (hangup maybe). I think the recommended way is to transfer away the call to the conference, that way the Lua scripts won't need to do any more processing.

Also check this Javascript example on the wiki: http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR

About DTMF's - if you don't see them in your log, then FS havn't received any digits - you would have to do a packet capture and check the RTP packets.

/Peter

-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Maciej Bylica
Skickat: den 2 mars 2011 12:31
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] Conference - control actions problem

Hello,

I am running FreeSWITCH Version 1.0.head (git-06988e1 2011-02-20
15-10-02 -0600) and have some problem with conference module and
respective control actions digits.
To give you an overview of how it is configured, there is lua script
fired in public.xml.
The script checks pin code, conf room and at the end run
"session:execute("conference", string.format("%s at fsconf", conf_num))".
Now coming back to the problem, there are few scenations i would like to cover.

1. When two or more people are in conference and one of them is trying
to push # button (RFC2833) then that person is leaving the conference
and is hanged up (not released, no BYE). The person who left the
conference cannot join again or perform other actions without making
the same phone call again.

2011-03-02 11:42:08.863139 [DEBUG] mod_conference.c:2414 Setup timer
soft success interval: 20  samples: 160
2011-03-02 11:42:12.283051 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:480
2011-03-02 11:42:12.323050 [DEBUG] mod_conference.c:2659 Channel
leaving conference, cause: NONE

2. The same as above, but * digit is entered. The person is muted and
then unmuted. Next try with * fires muted but this is the point where
the problem manifest itself.
The procedure of unmuting UA fails. Below you may find relevant logs.
The UA is pressing * but without success (no RTP RECV DTMF *).

freeswitch at internal> 2011-03-02 11:51:13.423651 [DEBUG]
switch_rtp.c:3237 RTP RECV DTMF *:504
2011-03-02 11:51:13.443650 [DEBUG] mod_conference.c:3125 Queueing file
'/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav'
for play
2011-03-02 11:51:18.163532 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF *:544
2011-03-02 11:51:18.183531 [DEBUG] mod_conference.c:3125 Queueing file
'/usr/local/freeswitch/sounds/en/us/callie/conference/conf-unmuted.wav'
for play
2011-03-02 11:51:23.763391 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF *:504
2011-03-02 11:51:23.783390 [DEBUG] mod_conference.c:3125 Queueing file
'/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav'
for play

3. The same as above but with 0 digit

2011-03-02 12:04:21.064862 [DEBUG] mod_conference.c:2414 Setup timer
soft success interval: 20  samples: 160
2011-03-02 12:04:22.863736 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 0:880
2011-03-02 12:04:22.883736 [DEBUG] mod_conference.c:3125 Queueing file
'/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav'
for play

then i cannot get back to the conference


4. The as above but with 6 digit

2011-03-02 12:15:03.244609 [DEBUG] mod_conference.c:2414 Setup timer
soft success interval: 20  samples: 160
2011-03-02 12:15:21.602919 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 6:504
2011-03-02 12:15:22.602893 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 6:800
2011-03-02 12:15:24.263851 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 6:864


I have disabled all control digits, but the problem still persist. No
action is fired but is looks like UA is being muted (micro is off, but
i can hear the other person).
2011-03-02 12:24:05.322935 [DEBUG] mod_conference.c:2414 Setup timer
soft success interval: 20  samples: 160
2011-03-02 12:24:08.663851 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:480
2011-03-02 12:24:11.223787 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:504
2011-03-02 12:24:13.703725 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:480
2011-03-02 12:24:16.043666 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:520


Has anybody similar experience or know where the problem may be located.

Regards,
Maciej

_______________________________________________
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From linux4michelle at tamay-dogan.net  Wed Mar  2 14:14:37 2011
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Wed, 2 Mar 2011 12:14:37 +0100
Subject: [Freeswitch-users] Does someone has a Huawei K3765 runing with
	FreeSWITCH?
Message-ID: <20110302111437.GO3248@michelle1>

Hello,

I have a Vodafone DSL EasyBox 803A (Astoria Networks)  with  the  Huawei
K3765 runing and it works, but since the EasyBoy 803A does  not  support
SIP, I want to drop it and use the Huawei K3765 directly with FreeSWITCH

Does someone has the Huawei K3765 runing with FreeSWITCH?

And how canI configure it without lossing of the Internet connection?

Thanks, Greetings and nice Day/Evening
    Michelle Konzack

-- 
##################### Debian GNU/Linux Consultant ######################
   Development of Intranet and Embedded Systems with Debian GNU/Linux

itsystems at tdnet France EURL       itsystems at tdnet UG (limited liability)
Owner Michelle Konzack            Owner Michelle Konzack

Apt. 917 (homeoffice)
50, rue de Soultz                 Kinzigstra?e 17
67100 Strasbourg/France           77694 Kehl/Germany
Tel: +33-6-61925193 mobil         Tel: +49-177-9351947 mobil
Tel: +33-9-52705884 fix

<http://www.itsystems.tamay-dogan.net/>  <http://www.flexray4linux.org/>
<http://www.debian.tamay-dogan.net/>         <http://www.can4linux.org/>

Jabber linux4michelle at jabber.ccc.de

Linux-User #280138 with the Linux Counter, http://counter.li.org/
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From julf at julf.com  Wed Mar  2 17:34:07 2011
From: julf at julf.com (Johan Helsingius)
Date: Wed, 02 Mar 2011 15:34:07 +0100
Subject: [Freeswitch-users] checking voicemail without ID?
Message-ID: <4D6E555F.4000003@julf.com>

Apologies for what probably is a FAQ, but even with a fair bit
of googling I couldn't find a resolution.

I am trying to make it possible for users to check their
voicemail without knowing their voicemail id (because I
want to use a common mailbox for multiple accounts).

The standard way seems to be
<action application="set" data="voicemail_authorized=${sip_authorized}"/>
but it doesn't do the trick for me...

	Julf


From brian at freeswitch.org  Wed Mar  2 18:12:46 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 2 Mar 2011 09:12:46 -0600
Subject: [Freeswitch-users] checking voicemail without ID?
In-Reply-To: <4D6E555F.4000003@julf.com>
References: <4D6E555F.4000003@julf.com>
Message-ID: <649CD6B6-AB32-41B0-8C9D-2BB5F3DEA9FD@freeswitch.org>

I'm going to guess that sip_authorized is not true?

/b

On Mar 2, 2011, at 8:34 AM, Johan Helsingius wrote:

> The standard way seems to be
> <action application="set" data="voicemail_authorized=${sip_authorized}"/>
> but it doesn't do the trick for me...
> 
> 	Julf



From acrow at integrafin.co.uk  Wed Mar  2 18:25:44 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Wed, 02 Mar 2011 15:25:44 +0000
Subject: [Freeswitch-users] British sounds
In-Reply-To: <AANLkTi=pz1GgEz9OLvW_7=QtO=B=nWiHS+4G6D7MY=-N@mail.gmail.com>
References: <4D681972.7050400@integrafin.co.uk>	<AANLkTinGQk18xc8WCap6VkyTVYM1-XHcPi1PhwVTNs8B@mail.gmail.com>	<4D68222C.8010108@integrafin.co.uk>	<4D6D0366.7020205@integrafin.co.uk>
	<AANLkTi=pz1GgEz9OLvW_7=QtO=B=nWiHS+4G6D7MY=-N@mail.gmail.com>
Message-ID: <4D6E6178.8040205@integrafin.co.uk>

On 01/03/11 18:27, Michael Collins wrote:
> Okay, this is good. I recommend getting a spreadsheet together and 
> creating columns for Ast file name, Ast prompt, FS file name, FS 
> prompt. The prompts would contain the actual text of what is voiced in 
> the audio file. On the FS side look in 
> $FS_SOURCE/docs/phrase/phrase_en.xml for the phrases and file names. 
> In Asterisk there usually is a text file that has the filename and the 
> text of the prompt.
>
> Once we have that together we can figure out what prompts we can 
> re-use versus what needs to be re-recorded.
>
> -MC
>

Michael,

I have attached (hope the list allows it) an ods with the info you have 
asked for. The * names appear to be unique, but the FS one are not 
always so I've inserted a type column for these.

Cheers

Alex

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From julf at julf.com  Wed Mar  2 18:36:30 2011
From: julf at julf.com (Johan Helsingius)
Date: Wed, 02 Mar 2011 16:36:30 +0100
Subject: [Freeswitch-users] checking voicemail without ID?
In-Reply-To: <649CD6B6-AB32-41B0-8C9D-2BB5F3DEA9FD@freeswitch.org>
References: <4D6E555F.4000003@julf.com>
	<649CD6B6-AB32-41B0-8C9D-2BB5F3DEA9FD@freeswitch.org>
Message-ID: <4D6E63FE.1020302@julf.com>

> I'm going to guess that sip_authorized is not true?

A good guess, but changing the line to
<action application="set" data="voicemail_authorized=true"/>
still doesn't help.

	Julf

>> The standard way seems to be
>> <action application="set" data="voicemail_authorized=${sip_authorized}"/>
>> but it doesn't do the trick for me...


From brian at freeswitch.org  Wed Mar  2 18:40:11 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 2 Mar 2011 09:40:11 -0600
Subject: [Freeswitch-users] checking voicemail without ID?
In-Reply-To: <4D6E63FE.1020302@julf.com>
References: <4D6E555F.4000003@julf.com>
	<649CD6B6-AB32-41B0-8C9D-2BB5F3DEA9FD@freeswitch.org>
	<4D6E63FE.1020302@julf.com>
Message-ID: <518984EC-1B52-4631-B3D5-2DB6776255C5@freeswitch.org>

Show us the whole dialplan.

/b

On Mar 2, 2011, at 9:36 AM, Johan Helsingius wrote:

> A good guess, but changing the line to
> <action application="set" data="voicemail_authorized=true"/>
> still doesn't help.
> 
> 	Julf



From keithmsn at laaks.com  Wed Mar  2 18:40:18 2011
From: keithmsn at laaks.com (keithmsn at laaks.com)
Date: Wed, 2 Mar 2011 17:40:18 +0200 (SAST)
Subject: [Freeswitch-users] Eavesdrop - How to get out of the application ?
Message-ID: <52315.41.146.72.147.1299080418.squirrel@mail.laaks.com>

Hi,

I am calling the eavesdrop application as follows:
   session.setVariable("eavesdrop_enable_dtmf","true");
   session.execute("eavesdrop","all");

Works 100%, with the application responding to the keys for talking to the
one, other or both sides of the channel, resetting and also moving onto
the next channel (*).

But there is just no way to get out of eavesdrop other than hanging up the
call and calling back in. Maybe I am missing something?

I looked into:
   mod_dptools.c at 'eavesdrop_function',
   switch_ivr_async.c at 'switch_ivr_eavesdrop_session'
but no code for some escape key (i.e. #) in 'switch_ivr_eavesdrop_session'
which will break the permanent loop in 'eavesdrop_function'.

Maybe I should not be using 'all' and instead go the db route?

I also spotted two variables: "eavesdrop_announce_id" and
"eavesdrop_annnounce_macro" used in switch_ivr_eavesdrop_session. Looks
like this allows for a macro to be played on switching to a new session to
verbally identify it. Don't see this in the wiki. Anybody tried this
before?

Regards

Keith






From julf at julf.com  Wed Mar  2 18:58:25 2011
From: julf at julf.com (Johan Helsingius)
Date: Wed, 02 Mar 2011 16:58:25 +0100
Subject: [Freeswitch-users] checking voicemail without ID?
In-Reply-To: <518984EC-1B52-4631-B3D5-2DB6776255C5@freeswitch.org>
References: <4D6E555F.4000003@julf.com>	<649CD6B6-AB32-41B0-8C9D-2BB5F3DEA9FD@freeswitch.org>	<4D6E63FE.1020302@julf.com>
	<518984EC-1B52-4631-B3D5-2DB6776255C5@freeswitch.org>
Message-ID: <4D6E6921.9000604@julf.com>

Brian,

> Show us the whole dialplan.

Here is the voicemail part:

  <!-- voicemail main extension -->
    <extension name="vmain">
      <condition field="destination_number" expression="^vmain$|^\*98$">
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="set" data="voicemail_authorized=${sip_authorized}"/>
        <action application="set" data="voicemail_authorized=true"/>
        <action application="voicemail" data="check default ${domain_name}
julf_mailbox"/>
      </condition>
    </extension>


	Julf


From me at nevian.org  Wed Mar  2 19:01:31 2011
From: me at nevian.org (Serge S. Yuriev)
Date: Wed, 02 Mar 2011 19:01:31 +0300
Subject: [Freeswitch-users] LCR troubles
Message-ID: <37641299081691@web132.yandex.ru>

Hello,

I have created JIRA about this FS-3109.
I post it here to wide it's coverage.

In short:
in LCR mode FS forwards (negative) GW answer to caller but continues on next GW so useless one legged call created.
If I'm doing this via list of gateways - callflow continues normal.

Pls advice.

-- 
wbr,
Serge


From kawarod at laposte.net  Wed Mar  2 19:07:02 2011
From: kawarod at laposte.net (kawarod)
Date: Wed, 2 Mar 2011 20:07:02 +0400
Subject: [Freeswitch-users] Play a file based on SIP error code
Message-ID: <0AC82A20-57D9-47E6-AEA7-A2EE53D661F6@laposte.net>

Hi List,

I'd like to know how I can play a file based on the SIP error code returned by a peer.

For example, I'd like to play the following announce if I receive a SIP 404: 
	" The number you dialed is unreachable"

Thanks for your help.

rod.



From mbsip at gazeta.pl  Wed Mar  2 19:15:07 2011
From: mbsip at gazeta.pl (Maciej Bylica)
Date: Wed, 2 Mar 2011 17:15:07 +0100
Subject: [Freeswitch-users] Conference - control actions problem
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B2BBAF1D@cooper>
References: <AANLkTimq9dYbq8ztbQ5dJmY2Y+2mK5mKzM78UirEzzYn@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2BBAF1D@cooper>
Message-ID: <AANLkTikPa1E9o3-NgWBjd7P8HWfr1vRxAnh+AH0UpqW1@mail.gmail.com>

Thx Peter for prompt reply.

I've just made one more configuration change. There are no lua script
fired anymore, just pure conference application like following:
       <action application="conference" data="1000 at fsconf"/>

The effect is the same (after pressing * or # key just one time).
2011-03-02 16:55:32.677018 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:520
or later
2011-03-02 16:58:13.645012 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF *:620

As you can see there are DTMF detected on Freeswitch interpreted as
'RTP RECV DTMF'
Is git-06988e1 rel stable one?


Regards,
Maciej

2011/3/2 Peter Olsson <peter.olsson at visionutveckling.se>:
> If you exit the conference (#) the user will still be in your lua script, so you would have to do whatever you want it to (hangup maybe). I think the recommended way is to transfer away the call to the conference, that way the Lua scripts won't need to do any more processing.
>
> Also check this Javascript example on the wiki: http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR
>
> About DTMF's - if you don't see them in your log, then FS havn't received any digits - you would have to do a packet capture and check the RTP packets.
>
> /Peter
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Maciej Bylica
> Skickat: den 2 mars 2011 12:31
> Till: FreeSWITCH Users Help
> ?mne: [Freeswitch-users] Conference - control actions problem
>
> Hello,
>
> I am running FreeSWITCH Version 1.0.head (git-06988e1 2011-02-20
> 15-10-02 -0600) and have some problem with conference module and
> respective control actions digits.
> To give you an overview of how it is configured, there is lua script
> fired in public.xml.
> The script checks pin code, conf room and at the end run
> "session:execute("conference", string.format("%s at fsconf", conf_num))".
> Now coming back to the problem, there are few scenations i would like to cover.
>
> 1. When two or more people are in conference and one of them is trying
> to push # button (RFC2833) then that person is leaving the conference
> and is hanged up (not released, no BYE). The person who left the
> conference cannot join again or perform other actions without making
> the same phone call again.
>
> 2011-03-02 11:42:08.863139 [DEBUG] mod_conference.c:2414 Setup timer
> soft success interval: 20 ?samples: 160
> 2011-03-02 11:42:12.283051 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:480
> 2011-03-02 11:42:12.323050 [DEBUG] mod_conference.c:2659 Channel
> leaving conference, cause: NONE
>
> 2. The same as above, but * digit is entered. The person is muted and
> then unmuted. Next try with * fires muted but this is the point where
> the problem manifest itself.
> The procedure of unmuting UA fails. Below you may find relevant logs.
> The UA is pressing * but without success (no RTP RECV DTMF *).
>
> freeswitch at internal> 2011-03-02 11:51:13.423651 [DEBUG]
> switch_rtp.c:3237 RTP RECV DTMF *:504
> 2011-03-02 11:51:13.443650 [DEBUG] mod_conference.c:3125 Queueing file
> '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav'
> for play
> 2011-03-02 11:51:18.163532 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF *:544
> 2011-03-02 11:51:18.183531 [DEBUG] mod_conference.c:3125 Queueing file
> '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-unmuted.wav'
> for play
> 2011-03-02 11:51:23.763391 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF *:504
> 2011-03-02 11:51:23.783390 [DEBUG] mod_conference.c:3125 Queueing file
> '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav'
> for play
>
> 3. The same as above but with 0 digit
>
> 2011-03-02 12:04:21.064862 [DEBUG] mod_conference.c:2414 Setup timer
> soft success interval: 20 ?samples: 160
> 2011-03-02 12:04:22.863736 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 0:880
> 2011-03-02 12:04:22.883736 [DEBUG] mod_conference.c:3125 Queueing file
> '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav'
> for play
>
> then i cannot get back to the conference
>
>
> 4. The as above but with 6 digit
>
> 2011-03-02 12:15:03.244609 [DEBUG] mod_conference.c:2414 Setup timer
> soft success interval: 20 ?samples: 160
> 2011-03-02 12:15:21.602919 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 6:504
> 2011-03-02 12:15:22.602893 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 6:800
> 2011-03-02 12:15:24.263851 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 6:864
>
>
> I have disabled all control digits, but the problem still persist. No
> action is fired but is looks like UA is being muted (micro is off, but
> i can hear the other person).
> 2011-03-02 12:24:05.322935 [DEBUG] mod_conference.c:2414 Setup timer
> soft success interval: 20 ?samples: 160
> 2011-03-02 12:24:08.663851 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:480
> 2011-03-02 12:24:11.223787 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:504
> 2011-03-02 12:24:13.703725 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:480
> 2011-03-02 12:24:16.043666 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:520
>
>
> Has anybody similar experience or know where the problem may be located.
>
> Regards,
> Maciej
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4d6e2bb232761664219435!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From steveayre at gmail.com  Wed Mar  2 19:22:53 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 2 Mar 2011 16:22:53 +0000
Subject: [Freeswitch-users] Play a file based on SIP error code
In-Reply-To: <0AC82A20-57D9-47E6-AEA7-A2EE53D661F6@laposte.net>
References: <0AC82A20-57D9-47E6-AEA7-A2EE53D661F6@laposte.net>
Message-ID: <AANLkTikdtd41i=gmuEyVX1NFVaThJ=8AOpou=+ya7Xgt@mail.gmail.com>

To sketch our how to do it...

In dialplan:
<action application="set" data="hangup_on_bridge=true"/>
<action application="set" data="continue_on_fail=true"/>
<action application="bridge" data="..."/>
<action application="lua" data="playerror.lua"/>

That'll run the lua script after the bridge (it'll only get there if
the bridge fails)

In the lua script:
   cause = session:getVariable("proto_specific_hangup_cause")
   if (cause == "sip:486") then
      session:streamFile("/path/to/busy.wav");
   elseif (cause == "sip:404") then
      session:streamFile("/path/to/unknown_user.wav");
   elseif (cause == "sip:480") then
      session:streamFile("/path/to/unavailable.wav");
   else
      session:streamFile("/path/to/unknown_error.wav");
   end

You can also use the ISDN clearing cause from session:hangupCause() if you wish.

-Steve


On 2 March 2011 16:07, kawarod <kawarod at laposte.net> wrote:
> Hi List,
>
> I'd like to know how I can play a file based on the SIP error code returned by a peer.
>
> For example, I'd like to play the following announce if I receive a SIP 404:
> ? ? ? ?" The number you dialed is unreachable"
>
> Thanks for your help.
>
> rod.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From msc at freeswitch.org  Wed Mar  2 19:40:13 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 2 Mar 2011 08:40:13 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <AANLkTi=tT4gT_95v5G4gpy7qSUJDw6F2no5=F1Agv69=@mail.gmail.com>

Hello all!

Today's agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2011_03_02

We are going to have an update from Mark Crane on the FusionPBX project.
Bring your questions!

-MC
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From ephipps at gmail.com  Wed Mar  2 19:21:23 2011
From: ephipps at gmail.com (Eric Phipps)
Date: Wed, 2 Mar 2011 11:21:23 -0500
Subject: [Freeswitch-users] Remaking default config files in Centos
Message-ID: <AANLkTikr8z3T7tB4bKbzQS2=HqvRTd0peGkh8oUC4wdJ@mail.gmail.com>

Hello,

I'm working with Freeswitch and have managed to place an extra "<"
somewhere in my xml files.  Unfortunately, I have been unable to
locate it and it's preventing the starting of Freeswitch.

I was curious if there was an easy way to overwrite the configs which
exist there now and replace them with the default configurations for
CentOS.

-Eric Phipps


From wstephen80 at gmail.com  Wed Mar  2 19:51:47 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Wed, 2 Mar 2011 17:51:47 +0100
Subject: [Freeswitch-users] FreeTDM errors in log file
In-Reply-To: <AANLkTinYXEwnDZcKvibuPb5n6FT=38EjwCmnUVR5riLY@mail.gmail.com>
References: <AANLkTinZkJPS+gywSp0EYu_3Wx=C=WEqRK_QSnfyz4dw@mail.gmail.com>
	<AANLkTinYXEwnDZcKvibuPb5n6FT=38EjwCmnUVR5riLY@mail.gmail.com>
Message-ID: <AANLkTin2kqSCGZ1oL_E7ru-TStcVQWFWP8bDviNss1cB@mail.gmail.com>

Ok, thank you Moises, I have opened a bug report:

http://jira.freeswitch.org/browse/OPENZAP-143

and assigned to you.

Stephen

On Wed, Mar 2, 2011 at 3:12 AM, Moises Silva <moises.silva at gmail.com> wrote:

> Stephen,
>
> Can you open a bug report in jira for this and assign it to me?
>
> 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 Mon, Feb 28, 2011 at 7:16 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> I'm using Sangoma A108 board in my FS installation and my log is full of
>> warnings as:
>>
>> 2011-03-01 00:18:14.947380 [WARNING] mod_freetdm.c:434 [s17c22][17:22]
>> VETO state change from PROGRESS_MEDIA to PROCEED
>> 2011-03-01 00:18:53.151673 [WARNING] mod_freetdm.c:434 [s3c11][3:11] VETO
>> state change from PROGRESS_MEDIA to PROCEED
>> 2011-03-01 00:19:23.001140 [WARNING] mod_freetdm.c:434 [s23c29][23:29]
>> VETO state change from RINGING to PROCEED
>> 2011-03-01 00:19:40.862103 [WARNING] mod_freetdm.c:434 [s3c21][3:21] VETO
>> state change from PROGRESS_MEDIA to PROCEED
>> 2011-03-01 00:19:46.022154 [WARNING] mod_freetdm.c:434 [s3c21][3:21] VETO
>> state change from PROGRESS_MEDIA to PROCEED
>> 2011-03-01 00:19:50.608224 [WARNING] mod_freetdm.c:434 [s22c24][22:24]
>> VETO state change from PROGRESS_MEDIA to PROCEED
>> 2011-03-01 00:19:52.422221 [WARNING] mod_freetdm.c:434 [s3c21][3:21] VETO
>> state change from PROGRESS_MEDIA to PROCEED
>> 2011-03-01 00:21:36.461338 [WARNING] mod_freetdm.c:434 [s3c31][3:31] VETO
>> state change from PROGRESS_MEDIA to PROCEED
>> 2011-03-01 00:21:36.981455 [WARNING] mod_freetdm.c:434 [s3c30][3:30] VETO
>> state change from PROGRESS_MEDIA to PROCEED
>>
>> Another kind of messages that are in my log file are:
>>
>> 2011-03-01 00:53:14.427853 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:760
>> [SNGISDN Q931] s16: Protocol: Unknown Event Code(2): Incomp Msg(276)
>> 2011-03-01 00:53:18.428846 [CRIT] ftmod_sangoma_isdn_stack_hndl.c:557
>> [s16c23][16:23] Received DISCONNECT in an invalid state (TERMINATING)
>> 2011-03-01 00:53:18.428846 [WARNING] ftmod_sangoma_isdn_stack_hndl.c:562
>> [s16c23][16:23] Why bother changing state from TERMINATING to TERMINATING
>> 2011-03-01 00:54:08.607351 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:760
>> [SNGISDN Q931] s17: Protocol: Unknown Event Code(2): Incomp Msg(276)
>> 2011-03-01 00:54:12.607389 [CRIT] ftmod_sangoa_isdn_stack_hndl.c:557
>> [s17c20][17:20] Received DISCONNECT in an invalid state (TERMINATING)
>> 2011-03-01 00:54:12.607389 [WARNING] ftmod_sangoma_isdn_stack_hndl.c:562
>> [s17c20][17:20] Why bother changing state from TERMINATING to TERMINATING
>> 2011-03-01 00:56:45.595869 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:760
>> [SNGISDN Q931] s16: Protocol: Unknown Event Code(2): Incomp Msg(276)
>> 2011-03-01 00:56:49.596913 [CRIT] ftmod_sangoma_isdn_stack_hndl.c:557
>> [s16c23][16:23] Received DISCONNECT in an invalid state (TERMINATING)
>> 2011-03-01 00:56:49.596913 [WARNING] ftmod_sangoma_isdn_stack_hndl.c:562
>> [s16c23][16:23] Why bother changing state from TERMINATING to TERMINATING
>> 2011-03-01 00:59:23.036314 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:760
>> [SNGISDN Q931] s17: Protocol: Unknown Event Code(2): Incomp Msg(276)
>> 2011-03-01 00:59:27.037327 [CRIT] ftmod_sangoma_isdn_stack_hndl.c:557
>> [s17c20][17:20] Received DISCONNECT in an invalid state (TERMINATING)
>> 2011-03-01 00:59:27.037327 [WARNING] ftmod_sangoma_isdn_stack_hndl.c:562
>> [s17c20][17:20] Why bother changing state from TERMINATING to TERMINATING
>>
>> I'm using latest git of Freeswitch, Wanpipe 3.5.18 and lib Sangoma isdn
>> 7.3.0.
>>
>> Attached to this email the .pcap related to the call that generate the
>> critical error (to me it seems that is missed a CALL_PROCEEDING message).
>>
>> Stephen
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From djbinter at gmail.com  Wed Mar  2 20:09:42 2011
From: djbinter at gmail.com (DJB International)
Date: Wed, 2 Mar 2011 09:09:42 -0800
Subject: [Freeswitch-users] checking voicemail without ID?
In-Reply-To: <4D6E6921.9000604@julf.com>
References: <4D6E555F.4000003@julf.com>
	<649CD6B6-AB32-41B0-8C9D-2BB5F3DEA9FD@freeswitch.org>
	<4D6E63FE.1020302@julf.com>
	<518984EC-1B52-4631-B3D5-2DB6776255C5@freeswitch.org>
	<4D6E6921.9000604@julf.com>
Message-ID: <AANLkTimqvK-f-b43Pm3Ws6cEm7ZcdD2cv78OPv7oxycN@mail.gmail.com>

Try it with "auth"

<action application="voicemail" data="check auth $${voicemail_profile}
$${domain} $1"/>



On Wed, Mar 2, 2011 at 7:58 AM, Johan Helsingius <julf at julf.com> wrote:

> Brian,
>
> > Show us the whole dialplan.
>
> Here is the voicemail part:
>
>  <!-- voicemail main extension -->
>    <extension name="vmain">
>      <condition field="destination_number" expression="^vmain$|^\*98$">
>        <action application="answer"/>
>        <action application="sleep" data="1000"/>
>        <action application="set"
> data="voicemail_authorized=${sip_authorized}"/>
>         <action application="set" data="voicemail_authorized=true"/>
>         <action application="voicemail" data="check default ${domain_name}
> julf_mailbox"/>
>      </condition>
>    </extension>
>
>
>        Julf
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mario_fs at mgtech.com  Wed Mar  2 21:59:18 2011
From: mario_fs at mgtech.com (Mario G)
Date: Wed, 2 Mar 2011 10:59:18 -0800
Subject: [Freeswitch-users] checking voicemail without ID?
In-Reply-To: <4D6E555F.4000003@julf.com>
References: <4D6E555F.4000003@julf.com>
Message-ID: <8A592B05-0D5F-4EFA-87A0-730F6770B58E@mgtech.com>

I do this so they can just press the envelope key, in my extension definition for the user id, I have the following in the variable section:

      <variable name="mailbox" value="100"/>

I also have:
      <variable name="voicemail_domain" value="mgtech.com"/>

I also have some phone that dont need to even put a password in.


On Mar 2, 2011, at 6:34 AM, Johan Helsingius wrote:

> Apologies for what probably is a FAQ, but even with a fair bit
> of googling I couldn't find a resolution.
> 
> I am trying to make it possible for users to check their
> voicemail without knowing their voicemail id (because I
> want to use a common mailbox for multiple accounts).
> 
> The standard way seems to be
> <action application="set" data="voicemail_authorized=${sip_authorized}"/>
> but it doesn't do the trick for me...
> 
> 	Julf
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From ephipps at gmail.com  Wed Mar  2 21:08:39 2011
From: ephipps at gmail.com (Eric Phipps)
Date: Wed, 2 Mar 2011 13:08:39 -0500
Subject: [Freeswitch-users] Remaking default config files in Centos
In-Reply-To: <AANLkTikr8z3T7tB4bKbzQS2=HqvRTd0peGkh8oUC4wdJ@mail.gmail.com>
References: <AANLkTikr8z3T7tB4bKbzQS2=HqvRTd0peGkh8oUC4wdJ@mail.gmail.com>
Message-ID: <AANLkTimTtOmvspUOJ_d3dp68eQ_ucGmhrYWsgetgfuLM@mail.gmail.com>

I actually found the error thanks to logging and was able to correct it.

Thank you!

On Wed, Mar 2, 2011 at 11:21 AM, Eric Phipps <ephipps at gmail.com> wrote:
> Hello,
>
> I'm working with Freeswitch and have managed to place an extra "<"
> somewhere in my xml files. ?Unfortunately, I have been unable to
> locate it and it's preventing the starting of Freeswitch.
>
> I was curious if there was an easy way to overwrite the configs which
> exist there now and replace them with the default configurations for
> CentOS.
>
> -Eric Phipps
>


From freeswitch at cartissolutions.com  Wed Mar  2 22:17:10 2011
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Wed, 02 Mar 2011 13:17:10 -0600
Subject: [Freeswitch-users] mod_dingaling and google voice
In-Reply-To: <1299030529099-6079399.post@n2.nabble.com>
References: <AANLkTin_dFj9ygx3TAmExeN9-pDFNCuZPGZ_Uc2d7o1W@mail.gmail.com>	<AANLkTimUr3=w_fj_k643ChW93Q_YTm63R_qFyBO9EoQV@mail.gmail.com>	<4D6C1978.8040702@cartissolutions.com>	<AANLkTim5TpcQeYSC_c6ZH9WMoJnq1aNvYViwUBkZr=fv@mail.gmail.com>	<4D6C288B.4050801@cartissolutions.com>	<AANLkTikncVxc7ebrzR4bHrzeHARKVU_Q+aRpzMTM9RwV@mail.gmail.com>	<4D6D44B7.6070900@cartissolutions.com>
	<1299030529099-6079399.post@n2.nabble.com>
Message-ID: <4D6E97B6.3040805@cartissolutions.com>

Negative.  I am working with a single instance of FS in this scenario.  
This single instance just happens to have a SIP UA that runs on 
99.X.X.X:5060 and one on 192.168.X.X:5060.

Updating to:
FreeSWITCH Version 1.0.head (git-9a7dbfb 2011-03-01 13-44-53 -0600)

Fixed the outbound calling.  But inbound is still not dialplan hunting.  
http://pastebin.freeswitch.org/15525

Tony, if you have a moment I'm curious to know more about the edge case 
here.  I'm using arno's iptables firewall on that box, and using 
iptables 1.4.7, kernel 2.6.37.

Yossi Neiman
Cartis Solutions, Inc. - http://www.cartissolutions.com


On 03/01/2011 07:48 PM, mazilo wrote:
> Yossi Neiman wrote:
>> Sofia UA's running on both 99.X.X.X:5060 and 192.X.X.X:5060.  Machine FS
>> runs on is an iptables NAT/Firewall/router.  SIP calls to/from the
>> outside world work fine otherwise.
> I am lost here. You have two FS machines and one on a public IP Address
> while the other on a private IP Address? Both are having problems with GV
> incoming calls?
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.


From andreas at tuerpe-net.de  Wed Mar  2 23:46:52 2011
From: andreas at tuerpe-net.de (Andreas Tuerpe)
Date: Wed, 02 Mar 2011 21:46:52 +0100
Subject: [Freeswitch-users] route inbound - based on sip account
In-Reply-To: <AANLkTimmuZSR5pPLDd9B6cXjgE9jQ7TDHibWQ_T4Fjx-@mail.gmail.com>
References: <4D6D45F0.5060205@tuerpe-net.de>
	<AANLkTimmuZSR5pPLDd9B6cXjgE9jQ7TDHibWQ_T4Fjx-@mail.gmail.com>
Message-ID: <4D6EACBC.7040509@tuerpe-net.de>

Hallo Michael, Johannes and Meftah

here is...
-------------------------------
--- start trace log extract ---

INVITE sip:gw+sip69250 at 188.246.xxx.xxx:5080;transport=udp SIP/2.0
Via: SIP/2.0/UDP 82.139.xxx.x:5060;branch=z9hG4bK2fa2cd41;rport
From: "+491758301234" <sip:+491758301234 at 82.139.xxx.x>;tag=as20cd750d
To: <sip:gw+sip69250 at 188.246.xxx.xxx:5080;transport=udp>
Contact: <sip:+491758301234 at 82.139.xxx.x>
Call-ID: 2637b63a2411697f7af5f089732a68eb at 82.139.xxx.x
CSeq: 102 INVITE
User-Agent: PTY SIPPort

...

# the only advice to the inbound channel
# -> 'sip69250' is the 2. gateway name
[INFO] mod_dialplan_xml.c:252 Processing +491758301234->sip69250 in 
context public


Dialplan: sofia/external/+491758301234 at 82.139.xxx.x parsing 
[public->unloop] continue=false
...
# 'public_extensions' is the 1. extension name in public context
Dialplan: sofia/external/+491758302577 at 82.139.223.1 parsing 
[public->public_extensions] continue=false
# my test of condition assignment with ${variable_sip_gateway} fails
Dialplan: ... ... Regex (FAIL) [public_extensions] 
${variable_sip_gateway}() =~ /sofia/gateway/sip69250/ break=on-false
...
# '4932229982781' is the 2. extension name in public context
Dialplan: sofia/external/+491758301234 at 82.139.xxx.x parsing 
[public->4932229982781] continue=false
# empty condition assignment run's without any errors
Dialplan: sofia/external/+491758301234 at 82.139.xxx.x Regex (PASS) 
[4932229982781] () =~ // break=on-false
Dialplan: sofia/external/+491758301234 at 82.139.xxx.x Action transfer(1001 
XML default)
...
#all others run's without any errors
...
---- end trace log extract ----
-------------------------------


# code of first extension name
<extension name="public_extensions">
    <condition field="${variable_sip_gateway}" 
expression="sofia/gateway/sip69250">
        <action application="transfer" data="1004 XML default"/>
    </condition>
</extension>

# code of second extension name
<extension name="4932229982781">
    <condition field="" expression="">
        <action application="transfer" data="1001 XML default"/>
    </condition>
</extension>


I'm unsure with syntax of condition variable esp. expression="?"

Thanks
    Andreas



Am 02.03.2011 00:07, schrieb Michael Collins:
> The FS on pfSense is pretty old, but if all you are working on is a
> simple routing issue your best bet is to add the "info" app in the
> public context. Somewhere near the top of public.xml just add this:
>
> <extension name="quick info dump" continue="true">
> <condition>
> <action application="info"/>
> </condition>
> </extension>
>
> Save that, press F6 (or do reloadxml) and then make a test inbound call.
> Watch the console - you'll see a TON of information. Look through the
> pieces of data that are displayed. You should be able to find something
> to key off of. Once you've done that then go read up on creating your
> dialplan here:
>
> http://wiki.freeswitch.org/wiki/Dialplan_XML#Condition
>
> In fact, that whole page is important in understanding the XML dialplan.
> I would read it more than once.
>
> -MC
>
> On Tue, Mar 1, 2011 at 11:16 AM, Andreas Tuerpe <andreas at tuerpe-net.de
> <mailto:andreas at tuerpe-net.de>> wrote:
>
>     Hallo FreeSWITCH Users,
>
>     I use FS V.0.9.6 on pfSense
>     see on -> [http://doc.pfsense.org/index.php/FreeSWITCH]
>
>     Symptom:
>     German ISP "Portunity" forward inbound calls without any
>     destination_number.
>
>
>     ISP solution tip:
>     FS to register over a second sip account to the provider twice.
>     Any account has a separate number.
>     Based on the channel over which the call come in, I have to decide which
>     number is called.
>
>     So I need help, which condition assignment must use - howto ???
>     - which fields can I use?
>     - which syntax I have to use?
>
>
>     <!--  here my intention ... ???
>
>     <extension name="test_did">
>     <condition field="channel over which the call come in"
>     expression="sip-account-1">
>     <action application="set" data="called_number=111111"/>
>     <action application="transfer" data="1001 XML default"/>
>     <condition field="channel over which the call come in"
>     expression="sip-account-2">
>     <action application="set" data="called_number=222222"/>
>     <action application="transfer" data="1002 XML default"/>
>     </condition>
>     </extension>
>
>     ??? -->
>
>     thanks in advance
>        tuerpean
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From simpot at simpot.com  Thu Mar  3 00:05:34 2011
From: simpot at simpot.com (Dmitry Saratsky)
Date: Wed, 2 Mar 2011 23:05:34 +0200
Subject: [Freeswitch-users] How to run linux shell command from within LUA
 script in Freeswitch?
Message-ID: <52114D605A462A4E9A50E588EE7D0686012E575D9EDB@mail.forest.simpot.com>

Hi Christopher,

Thanks you for your reply!

I already found why this didn't work for me (syntax issue - quotes), however while I tried to understand this, I already finished my lua script with oter command: os.execute()
This is pure lua script command, but not FS-API's Lua command, so I run my command not from within FS, but from within LUA script itself.

Do you see any disadvantages in this way?

Thanks,
Dmitry.

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From cmrienzo at gmail.com  Thu Mar  3 00:19:15 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Wed, 2 Mar 2011 16:19:15 -0500
Subject: [Freeswitch-users] How to run linux shell command from within
 LUA script in Freeswitch?
In-Reply-To: <52114D605A462A4E9A50E588EE7D0686012E575D9EDB@mail.forest.simpot.com>
References: <52114D605A462A4E9A50E588EE7D0686012E575D9EDB@mail.forest.simpot.com>
Message-ID: <AANLkTikbeeTFhYT22B03XwTm7yutj_WQRSvDS4Lz3v=3@mail.gmail.com>

No, your way is probably better.


On Wed, Mar 2, 2011 at 4:05 PM, Dmitry Saratsky <simpot at simpot.com> wrote:

> Hi Christopher,
>
>
>
> Thanks you for your reply!
>
>
>
> I already found why this didn?t work for me (syntax issue - quotes),
> however while I tried to understand this, I already finished my lua script
> with oter command: os.execute()
>
> This is pure lua script command, but not FS-API?s Lua command, so I run my
> command not from within FS, but from within LUA script itself.
>
>
>
> Do you see any disadvantages in this way?
>
>
>
> Thanks,
>
> Dmitry.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From frankie.k.yiu at gmail.com  Thu Mar  3 01:00:18 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Wed, 2 Mar 2011 14:00:18 -0800
Subject: [Freeswitch-users] How to create IVR application in C#?
Message-ID: <AANLkTi=ynpjSF0G43o6a7GG8TOHntZJZ0erO42XFhC8B@mail.gmail.com>

Thanks!

I am using C# managed function through swig.cs.  And I am calling the
function:

switch_ivr_menu_init(SWIGTYPE_
p_p_switch_ivr_menu new_menu,.......)

With the type "SWIGTYPE_p_p_switch_ivr_menu", do they type ever work?  It
has a run time error saying "Attempted to read or write protected memory.
This is often an indication that other memory is corrupt." for this variable
basically it is dereferencing a Null pointer.

How do I resolve this if I want to call this in C#?

Thanks,

Frankie

---------- Forwarded message ----------
> From: Malay Thakershi <mthakershi at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 1 Mar 2011 16:42:47 -0600
> Subject: Re: [Freeswitch-users] How to create IVR application in C#?
> mod_managed could be an option.
>
> http://wiki.freeswitch.org/wiki/Mod_managed
>
> <http://wiki.freeswitch.org/wiki/Mod_managed>It allows you to use most
> native FS features from C# managed code.
>
> Malay
>
> On Tue, Mar 1, 2011 at 3:01 PM, Frankie Yiu <frankie.k.yiu at gmail.com>wrote:
>
>> Hi there,
>>
>> I am newbie to FreeSwitch and I have question about creating an IVR
>> application in C#, with a possibly of using VoiceXML.  Could someone please
>> points me to how I can get started or any example that I can look at?
>>
>> Thanks a lot!!!
>>
>> Frankie
>>
>>
>>
>
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From ayhkor at gmail.com  Thu Mar  3 01:25:24 2011
From: ayhkor at gmail.com (deniro)
Date: Wed, 2 Mar 2011 17:25:24 -0500
Subject: [Freeswitch-users] fs_cli socket connection error
In-Reply-To: <AANLkTi=NP_Kw-Hrb6+XXtuid3YWJUM2ehFaeoaV-UEaC@mail.gmail.com>
References: <AANLkTikCoT9dzLNoD=6TzS0jiMw4afNX94TypsLarBbz@mail.gmail.com>
	<AANLkTi=sENgBB9Cdr-EkLJD8XfmH3A8GSfPnvDHO7qrw@mail.gmail.com>
	<AANLkTin5k=2Z-=QVVzqt91172wv88P7OFkqRqTY9VDcS@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58ADB727D7@cooper>
	<AANLkTi=aPwmnVD8ZzaBVFDxGyPsbgzDEBGaXvq_B2NbF@mail.gmail.com>
	<AANLkTikpQuxvyaU7PAGnyJuoQLQ9xn4inNJ_u5x41Anv@mail.gmail.com>
	<AANLkTimNYjC_G-Lq+St5VePpxEyM8iHnhaNCo=jBydi9@mail.gmail.com>
	<AANLkTi=QXgEb6dHU9bfb7v06WzPyWJcoNYDLi1X9oq+E@mail.gmail.com>
	<AANLkTikEDgpAJppNUnLac73ip7qsk+8AK-N-pTW7NBm6@mail.gmail.com>
	<AANLkTi=NP_Kw-Hrb6+XXtuid3YWJUM2ehFaeoaV-UEaC@mail.gmail.com>
Message-ID: <AANLkTinQ7g2YFCGNRtxWr-apHuByNFdF3Z_ph8J7K01e@mail.gmail.com>

I typed the 4 commands but still unable to load mod_xml_cdr and mod_xml_curl


2011-03-02 17:21:32.139672 [ERR] mod_xml_cdr.c:479 Open of xml_cdr.conf
failed
/opt# find . -name xml_cdr.conf
nothing appears

 here is errors from fs_cli -d 7

freeswitch at 75.xxx.xxx.xx@internal> load mod_xml_cdr
[DEBUG] libs/esl/src/esl.c:1063 esl_send() SEND
api load mod_xml_cdr
console_execute: true

[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type] =
[api/response]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Length]
= [50]
-ERR [module load file routine returned an error]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type] =
[log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Length]
= [79]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
[3]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel] =
[0]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
[mod_xml_cdr.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
[mod_xml_cdr_load]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
[479]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] = []
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 79
Log-Level: 3
Text-Channel: 0
Log-File: mod_xml_cdr.c
Log-Func: mod_xml_cdr_load
Log-Line: 479
User-Data: _undef_

2011-03-02 17:21:32.139672 [ERR] mod_xml_cdr.c:479 Open of xml_cdr.conf
failed
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type] =
[log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Length]
= [161]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
[2]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel] =
[0]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
[switch_loadable_module.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
[switch_loadable_module_load_file]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
[882]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] = []
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 161
Log-Level: 2
Text-Channel: 0
Log-File: switch_loadable_module.c
Log-Func: switch_loadable_module_load_file
Log-Line: 882
User-Data: _undef_

2011-03-02 17:21:32.139672 [CRIT] switch_loadable_module.c:882 Error Loading
module /opt/freeswitch/mod/mod_xml_cdr.so
**Module load routine returned an error**






On Tue, Mar 1, 2011 at 1:11 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> like I said you re-installed with a new version of libcurl but did not
> fully rebuild.
> So most of the modules that depend on curl were broken torn between
> the old and new version it was linked to..
>
> had you executed the 4 commands I posted yesterday it would be fine.
>
>
> On Tue, Mar 1, 2011 at 11:02 AM, deniro <ayhkor at gmail.com> wrote:
> > Thank you all for advice  and direction
> > Will re-install from scratch at a later time.
> > deniro--
> >
> > On Tue, Mar 1, 2011 at 4:16 AM, Steven Ayre <steveayre at gmail.com> wrote:
> >>
> >> Ok, you really need to pay attention to any log message that logs with
> >> the CRIT (critical) log level - they're really serious errors. You
> >> should look at adding the critical="true" attribute to all the tags in
> >> modules.conf.xml which you want to be sure are loaded. If a module
> >> fails to load without that tag FS will continue to run anyway, with
> >> that tag it'll refuse to start so you'll know instantly something is
> >> wrong.
> >>
> >> Most of your modules aren't loading, because they haven't been
> >> compiled correctly. Some look like they're missing (no such file or
> >> directory) and others haven't been built right (undefined symbols).
> >>
> >> Here are a few of the important lines from your log:
> >>
> >> 2011-02-28 18:12:21.910738 [CRIT] switch_loadable_module.c:882 Error
> >> Loading module /opt/freeswitch/mod/mod_event_socket.so
> >> **/opt/freeswitch/mod/mod_event_socket.so: undefined symbol:
> >> switch_event_serialize_json**
> >> 2011-02-28 18:12:21.910789 [CRIT] switch_loadable_module.c:882 Error
> >> Loading module /opt/freeswitch/mod/mod_xml_curl.so
> >> **/opt/freeswitch/mod/mod_xml_curl.so: cannot open shared object file:
> >> No such file or directory**
> >> 2011-02-28 18:12:21.910813 [CRIT] switch_loadable_module.c:882 Error
> >> Loading module /opt/freeswitch/mod/mod_xml_cdr.so
> >> **/opt/freeswitch/mod/mod_xml_cdr.so: cannot open shared object file:
> >> No such file or directory**
> >> 2011-02-28 18:12:21.911581 [CRIT] switch_loadable_module.c:882 Error
> >> Loading module /opt/freeswitch/mod/mod_commands.so
> >> **/opt/freeswitch/mod/mod_commands.so: undefined symbol:
> >> switch_xml_reload**
> >> 2011-02-28 18:12:21.911709 [CRIT] switch_loadable_module.c:882 Error
> >> Loading module /opt/freeswitch/mod/mod_conference.so
> >> **/opt/freeswitch/mod/mod_conference.so: undefined symbol:
> >> switch_channel_test_app_flag_key**
> >> 2011-02-28 18:12:21.911842 [CRIT] switch_loadable_module.c:882 Error
> >> Loading module /opt/freeswitch/mod/mod_dptools.so
> >> **/opt/freeswitch/mod/mod_dptools.so: undefined symbol:
> >> switch_event_merge**
> >> 2011-02-28 18:12:21.912264 [CRIT] switch_loadable_module.c:882 Error
> >> Loading module /opt/freeswitch/mod/mod_dialplan_xml.so
> >> **/opt/freeswitch/mod/mod_dialplan_xml.so: undefined symbol:
> >> switch_xml_std_datetime_check**
> >> 2011-02-28 18:12:21.912531 [CRIT] switch_loadable_module.c:882 Error
> >> Loading module /opt/freeswitch/mod/mod_g723_1.so
> >> **/opt/freeswitch/mod/mod_g723_1.so: cannot open shared object file:
> >> No such file or directory**
> >> 2011-02-28 18:12:21.912573 [CRIT] switch_loadable_module.c:882 Error
> >> Loading module /opt/freeswitch/mod/mod_g729.so
> >> **/opt/freeswitch/mod/mod_g729.so: cannot open shared object file: No
> >> such file or directory**
> >> 2011-02-28 18:12:21.912609 [CRIT] switch_loadable_module.c:882 Error
> >> Loading module /opt/freeswitch/mod/mod_amr.so
> >> **/opt/freeswitch/mod/mod_amr.so: cannot open shared object file: No
> >> such file or directory**
> >>
> >> Important things to note:
> >> mod_event_socket hasn't loaded because of a undefined symbol, it
> >> wasn't built correctly
> >> mod_xml_curl hasn't loaded because it doesn't exist, it was either
> >> never built or never installed
> >>
> >> Something pretty strange has happened. Did you recompile when trying
> >> to add mod_xml_curl?
> >>
> >> I'd suggest you delete all the FS files, including the Git clone, make
> >> a fresh checkout and build it from scratch.
> >>
> >> -Steve
> >>
> >>
> >>
> >> On 28 February 2011 23:18, deniro <ayhkor at gmail.com> wrote:
> >> > I put the log in pastebin
> >> > freeswitch.log only when starting freeswitch (after stop)
> >> >
> >> > http://pastebin.freeswitch.org/15502
> >> > thx
> >> > deniro
> >> > On Mon, Feb 28, 2011 at 5:46 PM, Steven Ayre <steveayre at gmail.com>
> >> > wrote:
> >> >>
> >> >> You must check the logfile for when FS starts up - if netstat shows
> >> >> nothing for port 8021 then either mod_event_socket isn't being loaded
> >> >> or you'll see an error when it tries to load.
> >> >>
> >> >> Nothing from netstat means nothing's listening, so trying to connect
> >> >> using fs_cli won't do anything.
> >> >>
> >> >> -Steve
> >> >>
> >> >>
> >> >> On 28 February 2011 22:40, deniro <ayhkor at gmail.com> wrote:
> >> >> > Checked the log
> >> >> > Nothing is logged when
> >> >> > fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
> >> >> > is issued
> >> >> > /opt/freeswitch/log# tail -f freeswitch.log
> >> >> >
> >> >> > ls -l freeswitch.log
> >> >> > -rwxrwxrwx 1 freeswitch daemon 33018 2011-02-28 17:39
> freeswitch.log
> >> >> > thx
> >> >> >
> >> >> >
> >> >> >
> >> >> > On Mon, Feb 28, 2011 at 3:59 PM, Peter Olsson
> >> >> > <peter.olsson at visionutveckling.se> wrote:
> >> >> >>
> >> >> >> Check freeswitch.log, it probably reports some problem when
> loading
> >> >> >> the
> >> >> >> mod_event_socket module.
> >> >> >>
> >> >> >> /Peter
> >> >> >> ________________________________________
> >> >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> >> >> >> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r deniro
> >> >> >> [ayhkor at gmail.com]
> >> >> >> Skickat: den 28 februari 2011 21:42
> >> >> >> Till: FreeSWITCH Users Help
> >> >> >> ?mne: Re: [Freeswitch-users] fs_cli socket connection error
> >> >> >>
> >> >> >> Steve
> >> >> >> Thanks for the reply
> >> >> >> netstat -anlp | grep 8021  is blank (nothing showing up)
> >> >> >>
> >> >> >> /opt/freeswitch/bin/fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
> >> >> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
> >> >> >> Configuration
> >> >> >> file is /root/.fs_cli_conf.
> >> >> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
> >> >> >> Configuration
> >> >> >> file is /etc/fs_cli.conf.
> >> >> >> [DEBUG] libs/esl/fs_cli.c:1159 main() profile default does not
> exist
> >> >> >> using
> >> >> >> builtin profile
> >> >> >> [DEBUG] libs/esl/fs_cli.c:1188 main() Using profile internal
> >> >> >> [75.xxx.xxx.xxx]
> >> >> >> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
> >> >> >> Connection
> >> >> >> Error]
> >> >> >> yes mod_event_socket is after  mod_xml_curl but I changed the
> order
> >> >> >> in
> >> >> >>  modules.conf.xml
> >> >> >> still getting above (restarted freeswitch)
> >> >> >>
> >> >> >> thx
> >> >> >> deniro--
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >> On Mon, Feb 28, 2011 at 3:35 AM, Steven Ayre
> >> >> >> <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
> >> >> >> Enable debug logging and you should see an error that'll tell you
> >> >> >> more.
> >> >> >>
> >> >> >> Is mod_event_socket loading before or after mod_xml_curl? Chances
> >> >> >> are
> >> >> >> mod_xml_curl is loading first, so it's trying to read
> >> >> >> event_socket.conf.xml and/or acl.conf.xml through xml_curl and
> >> >> >> either
> >> >> >> getting a different config to your previous local copy or the ACLs
> >> >> >> are
> >> >> >> different.
> >> >> >>
> >> >> >> What does "netstat -anlp | grep :8021" from the linux cli show
> you?
> >> >> >> Does it show that freeswitch is actually listening on the port? If
> >> >> >> it
> >> >> >> is it's probably an ACL problem, if it isn't then it's probably a
> >> >> >> problem with event_socket.conf.xml
> >> >> >>
> >> >> >> -Steve
> >> >> >>
> >> >> >> On 28 February 2011 00:53, deniro
> >> >> >> <ayhkor at gmail.com<mailto:ayhkor at gmail.com>> wrote:
> >> >> >> > What would be possible reasons for this and how to resolve?
> >> >> >> > running fs 106 on ubuntu 10.04 server
> >> >> >> >
> >> >> >> > /opt/freeswitch/bin/fs_cli -H xxx.xxx.xxx.xxx -P 8021
> >> >> >> > [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
> >> >> >> > Connection
> >> >> >> > Error]
> >> >> >> > was working fine before I installed  mod_xml_cdr
> >> >> >> > configure --prefix=/opt/freeswitch --without-libcurl
> >> >> >> > make mod_xml_cdr-install
> >> >> >> > (no errors)
> >> >> >> >
> >> >> >> > in modules.conf.xml
> >> >> >> > <load module="mod_event_socket"/>
> >> >> >> >
> >> >> >> > is there a packaged version of mod_xml_cdr (for fs 1.0.6)?
> >> >> >> >
> >> >> >> > thx
> >> >> >> > deniro--
> >> >> >> > _______________________________________________
> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> >
> >> >> >> >
> >> >> >> >
> >> >> >> > FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >
> >> >> >> >
> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> > http://www.freeswitch.org<http://www.freeswitch.org/>
> >> >> >> >
> >> >> >> >
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org<http://www.freeswitch.org/>
> >> >> >>
> >> >> >> !DSPAM:4d6c0a1132761029518849!
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From frankie.k.yiu at gmail.com  Thu Mar  3 00:48:35 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Wed, 2 Mar 2011 13:48:35 -0800
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 57, Issue 25
In-Reply-To: <mailman.4677.1299030530.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.4677.1299030530.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <AANLkTikidzEqktWupHkO_O6MrHVPTXGg5L=4gVvNWaYt@mail.gmail.com>

Thanks!

I am using C# managed function through swig.cs.  And I am calling the
function:

switch_ivr_menu_init(SWIGTYPE_p_p_switch_ivr_menu new_menu,.......)

With the type "SWIGTYPE_p_p_switch_ivr_menu", do they type ever work?  It
has a run time error saying "Attempted to read or write protected memory.
This is often an indication that other memory is corrupt." for this variable
basically it is dereferencing a Null pointer.

How do I resolve this if I want to call this in C#?

Thanks,

Frankie


>
> ---------- Forwarded message ----------
> From: Malay Thakershi <mthakershi at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 1 Mar 2011 16:42:47 -0600
> Subject: Re: [Freeswitch-users] How to create IVR application in C#?
> mod_managed could be an option.
>
> http://wiki.freeswitch.org/wiki/Mod_managed
>
> <http://wiki.freeswitch.org/wiki/Mod_managed>It allows you to use most
> native FS features from C# managed code.
>
> Malay
>
> On Tue, Mar 1, 2011 at 3:01 PM, Frankie Yiu <frankie.k.yiu at gmail.com>wrote:
>
>> Hi there,
>>
>> I am newbie to FreeSwitch and I have question about creating an IVR
>> application in C#, with a possibly of using VoiceXML.  Could someone please
>> points me to how I can get started or any example that I can look at?
>>
>> Thanks a lot!!!
>>
>> Frankie
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> ---------- Forwarded message ----------
> From: GillesToo <codecomplete at free.fr>
> To: freeswitch-users at lists.freeswitch.org
> Date: Tue, 1 Mar 2011 14:52:28 -0800 (PST)
> Subject: Re: [Freeswitch-users] Does FS handle three-way calling on the
> same POTS?
>
> mercutioviz wrote:
> > What are the instructions from the provider to create the second call?
> > Usually with an analog line you have to do a hook flash, then dial some
> > more digits, and then do another hook flash
>
> Thanks for the help. Asterisk doesn't allow using Dial() to call a second
> number after the original call was put on hold. A work-around is using
> SendDTMF() to dial the second number, but it appears there's no way to get
> call progress and know if the remote party has answered.
>
> Does FS provide a better way than this hack? I really need to use three-way
> calling on the same line since all calls are free this way.
>
> Thank you.
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Does-FS-handle-three-way-calling-on-the-same-POTS-tp6076537p6079027.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
>
>
> ---------- Forwarded message ----------
> From: Michael Collins <msc at freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 1 Mar 2011 15:07:28 -0800
> Subject: Re: [Freeswitch-users] route inbound - based on sip account
> The FS on pfSense is pretty old, but if all you are working on is a simple
> routing issue your best bet is to add the "info" app in the public context.
> Somewhere near the top of public.xml just add this:
>
> <extension name="quick info dump" continue="true">
>   <condition>
>     <action application="info"/>
>   </condition>
> </extension>
>
> Save that, press F6 (or do reloadxml) and then make a test inbound call.
> Watch the console - you'll see a TON of information. Look through the pieces
> of data that are displayed. You should be able to find something to key off
> of. Once you've done that then go read up on creating your dialplan here:
>
> http://wiki.freeswitch.org/wiki/Dialplan_XML#Condition
>
> In fact, that whole page is important in understanding the XML dialplan. I
> would read it more than once.
>
> -MC
>
> On Tue, Mar 1, 2011 at 11:16 AM, Andreas Tuerpe <andreas at tuerpe-net.de>wrote:
>
>> Hallo FreeSWITCH Users,
>>
>> I use FS V.0.9.6 on pfSense
>> see on -> [http://doc.pfsense.org/index.php/FreeSWITCH]
>>
>> Symptom:
>> German ISP "Portunity" forward inbound calls without any
>> destination_number.
>>
>>
>> ISP solution tip:
>> FS to register over a second sip account to the provider twice.
>> Any account has a separate number.
>> Based on the channel over which the call come in, I have to decide which
>> number is called.
>>
>> So I need help, which condition assignment must use - howto ???
>> - which fields can I use?
>> - which syntax I have to use?
>>
>>
>> <!--  here my intention ... ???
>>
>>   <extension name="test_did">
>>     <condition field="channel over which the call come in"
>> expression="sip-account-1">
>>       <action application="set" data="called_number=111111"/>
>>       <action application="transfer" data="1001 XML default"/>
>>     <condition field="channel over which the call come in"
>> expression="sip-account-2">
>>       <action application="set" data="called_number=222222"/>
>>       <action application="transfer" data="1002 XML default"/>
>>     </condition>
>>   </extension>
>>
>> ??? -->
>>
>> thanks in advance
>>   tuerpean
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> ---------- Forwarded message ----------
> From: mazilo <Nabble at slickdeals.endjunk.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Tue, 1 Mar 2011 17:48:49 -0800 (PST)
> Subject: Re: [Freeswitch-users] mod_dingaling and google voice
>
> Yossi Neiman wrote:
> > Sofia UA's running on both 99.X.X.X:5060 and 192.X.X.X:5060.  Machine FS
> > runs on is an iptables NAT/Firewall/router.  SIP calls to/from the
> > outside world work fine otherwise.
> I am lost here. You have two FS machines and one on a public IP Address
> while the other on a private IP Address? Both are having problems with GV
> incoming calls?
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/mod-dingaling-and-google-voice-tp6072163p6079399.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Thu Mar  3 02:25:01 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 2 Mar 2011 23:25:01 +0000
Subject: [Freeswitch-users] fs_cli socket connection error
In-Reply-To: <AANLkTinQ7g2YFCGNRtxWr-apHuByNFdF3Z_ph8J7K01e@mail.gmail.com>
References: <AANLkTikCoT9dzLNoD=6TzS0jiMw4afNX94TypsLarBbz@mail.gmail.com>
	<AANLkTi=sENgBB9Cdr-EkLJD8XfmH3A8GSfPnvDHO7qrw@mail.gmail.com>
	<AANLkTin5k=2Z-=QVVzqt91172wv88P7OFkqRqTY9VDcS@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58ADB727D7@cooper>
	<AANLkTi=aPwmnVD8ZzaBVFDxGyPsbgzDEBGaXvq_B2NbF@mail.gmail.com>
	<AANLkTikpQuxvyaU7PAGnyJuoQLQ9xn4inNJ_u5x41Anv@mail.gmail.com>
	<AANLkTimNYjC_G-Lq+St5VePpxEyM8iHnhaNCo=jBydi9@mail.gmail.com>
	<AANLkTi=QXgEb6dHU9bfb7v06WzPyWJcoNYDLi1X9oq+E@mail.gmail.com>
	<AANLkTikEDgpAJppNUnLac73ip7qsk+8AK-N-pTW7NBm6@mail.gmail.com>
	<AANLkTi=NP_Kw-Hrb6+XXtuid3YWJUM2ehFaeoaV-UEaC@mail.gmail.com>
	<AANLkTinQ7g2YFCGNRtxWr-apHuByNFdF3Z_ph8J7K01e@mail.gmail.com>
Message-ID: <AANLkTin_Y-azNb2sT-U9HQsjVbL3JqPiBamTiX5a_HC+@mail.gmail.com>

You need to create its config file. It won't load without it.

"find . -name xml_cdr.conf"

Actually it's xml_cdr.conf.xml, all config files will show .conf as
their logical name but on the hard drive they're stored in .conf.xml.
It should be in conf/autoload_configs

-Steve


On 2 March 2011 22:25, deniro <ayhkor at gmail.com> wrote:
> I typed the 4 commands but still unable to load mod_xml_cdr and mod_xml_curl
>
>
> 2011-03-02 17:21:32.139672 [ERR] mod_xml_cdr.c:479 Open of xml_cdr.conf
> failed
> /opt# find . -name xml_cdr.conf
> nothing appears
>
> here is errors from fs_cli -d 7
>
> freeswitch at 75.xxx.xxx.xx@internal> load mod_xml_cdr
> [DEBUG] libs/esl/src/esl.c:1063 esl_send() SEND
> api load mod_xml_cdr
> console_execute: true
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type] =
> [api/response]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Length]
> = [50]
> -ERR [module load file routine returned an error]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type] =
> [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Length]
> = [79]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [3]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel] =
> [0]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [mod_xml_cdr.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [mod_xml_cdr_load]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [479]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] = []
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 79
> Log-Level: 3
> Text-Channel: 0
> Log-File: mod_xml_cdr.c
> Log-Func: mod_xml_cdr_load
> Log-Line: 479
> User-Data: _undef_
> 2011-03-02 17:21:32.139672 [ERR] mod_xml_cdr.c:479 Open of xml_cdr.conf
> failed
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type] =
> [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Length]
> = [161]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [2]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel] =
> [0]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [switch_loadable_module.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_loadable_module_load_file]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [882]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] = []
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 161
> Log-Level: 2
> Text-Channel: 0
> Log-File: switch_loadable_module.c
> Log-Func: switch_loadable_module_load_file
> Log-Line: 882
> User-Data: _undef_
> 2011-03-02 17:21:32.139672 [CRIT] switch_loadable_module.c:882 Error Loading
> module /opt/freeswitch/mod/mod_xml_cdr.so
> **Module load routine returned an error**
>
>
>
>
>
> On Tue, Mar 1, 2011 at 1:11 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> like I said you re-installed with a new version of libcurl but did not
>> fully rebuild.
>> So most of the modules that depend on curl were broken torn between
>> the old and new version it was linked to..
>>
>> had you executed the 4 commands I posted yesterday it would be fine.
>>
>>
>> On Tue, Mar 1, 2011 at 11:02 AM, deniro <ayhkor at gmail.com> wrote:
>> > Thank you all for advice? and direction
>> > Will re-install from scratch at a later time.
>> > deniro--
>> >
>> > On Tue, Mar 1, 2011 at 4:16 AM, Steven Ayre <steveayre at gmail.com> wrote:
>> >>
>> >> Ok, you really need to pay attention to any log message that logs with
>> >> the CRIT (critical) log level - they're really serious errors. You
>> >> should look at adding the critical="true" attribute to all the tags in
>> >> modules.conf.xml which you want to be sure are loaded. If a module
>> >> fails to load without that tag FS will continue to run anyway, with
>> >> that tag it'll refuse to start so you'll know instantly something is
>> >> wrong.
>> >>
>> >> Most of your modules aren't loading, because they haven't been
>> >> compiled correctly. Some look like they're missing (no such file or
>> >> directory) and others haven't been built right (undefined symbols).
>> >>
>> >> Here are a few of the important lines from your log:
>> >>
>> >> 2011-02-28 18:12:21.910738 [CRIT] switch_loadable_module.c:882 Error
>> >> Loading module /opt/freeswitch/mod/mod_event_socket.so
>> >> **/opt/freeswitch/mod/mod_event_socket.so: undefined symbol:
>> >> switch_event_serialize_json**
>> >> 2011-02-28 18:12:21.910789 [CRIT] switch_loadable_module.c:882 Error
>> >> Loading module /opt/freeswitch/mod/mod_xml_curl.so
>> >> **/opt/freeswitch/mod/mod_xml_curl.so: cannot open shared object file:
>> >> No such file or directory**
>> >> 2011-02-28 18:12:21.910813 [CRIT] switch_loadable_module.c:882 Error
>> >> Loading module /opt/freeswitch/mod/mod_xml_cdr.so
>> >> **/opt/freeswitch/mod/mod_xml_cdr.so: cannot open shared object file:
>> >> No such file or directory**
>> >> 2011-02-28 18:12:21.911581 [CRIT] switch_loadable_module.c:882 Error
>> >> Loading module /opt/freeswitch/mod/mod_commands.so
>> >> **/opt/freeswitch/mod/mod_commands.so: undefined symbol:
>> >> switch_xml_reload**
>> >> 2011-02-28 18:12:21.911709 [CRIT] switch_loadable_module.c:882 Error
>> >> Loading module /opt/freeswitch/mod/mod_conference.so
>> >> **/opt/freeswitch/mod/mod_conference.so: undefined symbol:
>> >> switch_channel_test_app_flag_key**
>> >> 2011-02-28 18:12:21.911842 [CRIT] switch_loadable_module.c:882 Error
>> >> Loading module /opt/freeswitch/mod/mod_dptools.so
>> >> **/opt/freeswitch/mod/mod_dptools.so: undefined symbol:
>> >> switch_event_merge**
>> >> 2011-02-28 18:12:21.912264 [CRIT] switch_loadable_module.c:882 Error
>> >> Loading module /opt/freeswitch/mod/mod_dialplan_xml.so
>> >> **/opt/freeswitch/mod/mod_dialplan_xml.so: undefined symbol:
>> >> switch_xml_std_datetime_check**
>> >> 2011-02-28 18:12:21.912531 [CRIT] switch_loadable_module.c:882 Error
>> >> Loading module /opt/freeswitch/mod/mod_g723_1.so
>> >> **/opt/freeswitch/mod/mod_g723_1.so: cannot open shared object file:
>> >> No such file or directory**
>> >> 2011-02-28 18:12:21.912573 [CRIT] switch_loadable_module.c:882 Error
>> >> Loading module /opt/freeswitch/mod/mod_g729.so
>> >> **/opt/freeswitch/mod/mod_g729.so: cannot open shared object file: No
>> >> such file or directory**
>> >> 2011-02-28 18:12:21.912609 [CRIT] switch_loadable_module.c:882 Error
>> >> Loading module /opt/freeswitch/mod/mod_amr.so
>> >> **/opt/freeswitch/mod/mod_amr.so: cannot open shared object file: No
>> >> such file or directory**
>> >>
>> >> Important things to note:
>> >> mod_event_socket hasn't loaded because of a undefined symbol, it
>> >> wasn't built correctly
>> >> mod_xml_curl hasn't loaded because it doesn't exist, it was either
>> >> never built or never installed
>> >>
>> >> Something pretty strange has happened. Did you recompile when trying
>> >> to add mod_xml_curl?
>> >>
>> >> I'd suggest you delete all the FS files, including the Git clone, make
>> >> a fresh checkout and build it from scratch.
>> >>
>> >> -Steve
>> >>
>> >>
>> >>
>> >> On 28 February 2011 23:18, deniro <ayhkor at gmail.com> wrote:
>> >> > I put the log in pastebin
>> >> > freeswitch.log only?when starting freeswitch (after stop)
>> >> >
>> >> > http://pastebin.freeswitch.org/15502
>> >> > thx
>> >> > deniro
>> >> > On Mon, Feb 28, 2011 at 5:46 PM, Steven Ayre <steveayre at gmail.com>
>> >> > wrote:
>> >> >>
>> >> >> You must check the logfile for when FS starts up - if netstat shows
>> >> >> nothing for port 8021 then either mod_event_socket isn't being
>> >> >> loaded
>> >> >> or you'll see an error when it tries to load.
>> >> >>
>> >> >> Nothing from netstat means nothing's listening, so trying to connect
>> >> >> using fs_cli won't do anything.
>> >> >>
>> >> >> -Steve
>> >> >>
>> >> >>
>> >> >> On 28 February 2011 22:40, deniro <ayhkor at gmail.com> wrote:
>> >> >> > Checked the log
>> >> >> > Nothing is logged when
>> >> >> > fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
>> >> >> > is issued
>> >> >> > /opt/freeswitch/log# tail -f freeswitch.log
>> >> >> >
>> >> >> > ls -l freeswitch.log
>> >> >> > -rwxrwxrwx 1 freeswitch daemon 33018 2011-02-28 17:39
>> >> >> > freeswitch.log
>> >> >> > thx
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > On Mon, Feb 28, 2011 at 3:59 PM, Peter Olsson
>> >> >> > <peter.olsson at visionutveckling.se> wrote:
>> >> >> >>
>> >> >> >> Check freeswitch.log, it probably reports some problem when
>> >> >> >> loading
>> >> >> >> the
>> >> >> >> mod_event_socket module.
>> >> >> >>
>> >> >> >> /Peter
>> >> >> >> ________________________________________
>> >> >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> >> >> >> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r deniro
>> >> >> >> [ayhkor at gmail.com]
>> >> >> >> Skickat: den 28 februari 2011 21:42
>> >> >> >> Till: FreeSWITCH Users Help
>> >> >> >> ?mne: Re: [Freeswitch-users] fs_cli socket connection error
>> >> >> >>
>> >> >> >> Steve
>> >> >> >> Thanks for the reply
>> >> >> >> netstat -anlp | grep 8021 ?is blank (nothing showing up)
>> >> >> >>
>> >> >> >> /opt/freeswitch/bin/fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
>> >> >> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
>> >> >> >> Configuration
>> >> >> >> file is /root/.fs_cli_conf.
>> >> >> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
>> >> >> >> Configuration
>> >> >> >> file is /etc/fs_cli.conf.
>> >> >> >> [DEBUG] libs/esl/fs_cli.c:1159 main() profile default does not
>> >> >> >> exist
>> >> >> >> using
>> >> >> >> builtin profile
>> >> >> >> [DEBUG] libs/esl/fs_cli.c:1188 main() Using profile internal
>> >> >> >> [75.xxx.xxx.xxx]
>> >> >> >> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
>> >> >> >> Connection
>> >> >> >> Error]
>> >> >> >> yes mod_event_socket is after ?mod_xml_curl but I changed the
>> >> >> >> order
>> >> >> >> in
>> >> >> >> ?modules.conf.xml
>> >> >> >> still getting above (restarted freeswitch)
>> >> >> >>
>> >> >> >> thx
>> >> >> >> deniro--
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> On Mon, Feb 28, 2011 at 3:35 AM, Steven Ayre
>> >> >> >> <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
>> >> >> >> Enable debug logging and you should see an error that'll tell you
>> >> >> >> more.
>> >> >> >>
>> >> >> >> Is mod_event_socket loading before or after mod_xml_curl? Chances
>> >> >> >> are
>> >> >> >> mod_xml_curl is loading first, so it's trying to read
>> >> >> >> event_socket.conf.xml and/or acl.conf.xml through xml_curl and
>> >> >> >> either
>> >> >> >> getting a different config to your previous local copy or the
>> >> >> >> ACLs
>> >> >> >> are
>> >> >> >> different.
>> >> >> >>
>> >> >> >> What does "netstat -anlp | grep :8021" from the linux cli show
>> >> >> >> you?
>> >> >> >> Does it show that freeswitch is actually listening on the port?
>> >> >> >> If
>> >> >> >> it
>> >> >> >> is it's probably an ACL problem, if it isn't then it's probably a
>> >> >> >> problem with event_socket.conf.xml
>> >> >> >>
>> >> >> >> -Steve
>> >> >> >>
>> >> >> >> On 28 February 2011 00:53, deniro
>> >> >> >> <ayhkor at gmail.com<mailto:ayhkor at gmail.com>> wrote:
>> >> >> >> > What would be possible reasons for this and how to resolve?
>> >> >> >> > running fs 106 on ubuntu 10.04 server
>> >> >> >> >
>> >> >> >> > /opt/freeswitch/bin/fs_cli -H xxx.xxx.xxx.xxx -P 8021
>> >> >> >> > [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
>> >> >> >> > Connection
>> >> >> >> > Error]
>> >> >> >> > was working fine before I installed ?mod_xml_cdr
>> >> >> >> > configure --prefix=/opt/freeswitch --without-libcurl
>> >> >> >> > make mod_xml_cdr-install
>> >> >> >> > (no errors)
>> >> >> >> >
>> >> >> >> > in modules.conf.xml
>> >> >> >> > <load module="mod_event_socket"/>
>> >> >> >> >
>> >> >> >> > is there a packaged version of mod_xml_cdr (for fs 1.0.6)?
>> >> >> >> >
>> >> >> >> > thx
>> >> >> >> > deniro--
>> >> >> >> > _______________________________________________
>> >> >> >> > FreeSWITCH-users mailing list
>> >> >> >> >
>> >> >> >> >
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> > http://www.freeswitch.org<http://www.freeswitch.org/>
>> >> >> >> >
>> >> >> >> >
>> >> >> >>
>> >> >> >> _______________________________________________
>> >> >> >> FreeSWITCH-users mailing list
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> http://www.freeswitch.org<http://www.freeswitch.org/>
>> >> >> >>
>> >> >> >> !DSPAM:4d6c0a1132761029518849!
>> >> >> >>
>> >> >> >> _______________________________________________
>> >> >> >> FreeSWITCH-users mailing list
>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From ayhkor at gmail.com  Thu Mar  3 03:28:38 2011
From: ayhkor at gmail.com (deniro)
Date: Wed, 2 Mar 2011 19:28:38 -0500
Subject: [Freeswitch-users] fs_cli socket connection error
In-Reply-To: <AANLkTin_Y-azNb2sT-U9HQsjVbL3JqPiBamTiX5a_HC+@mail.gmail.com>
References: <AANLkTikCoT9dzLNoD=6TzS0jiMw4afNX94TypsLarBbz@mail.gmail.com>
	<AANLkTi=sENgBB9Cdr-EkLJD8XfmH3A8GSfPnvDHO7qrw@mail.gmail.com>
	<AANLkTin5k=2Z-=QVVzqt91172wv88P7OFkqRqTY9VDcS@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58ADB727D7@cooper>
	<AANLkTi=aPwmnVD8ZzaBVFDxGyPsbgzDEBGaXvq_B2NbF@mail.gmail.com>
	<AANLkTikpQuxvyaU7PAGnyJuoQLQ9xn4inNJ_u5x41Anv@mail.gmail.com>
	<AANLkTimNYjC_G-Lq+St5VePpxEyM8iHnhaNCo=jBydi9@mail.gmail.com>
	<AANLkTi=QXgEb6dHU9bfb7v06WzPyWJcoNYDLi1X9oq+E@mail.gmail.com>
	<AANLkTikEDgpAJppNUnLac73ip7qsk+8AK-N-pTW7NBm6@mail.gmail.com>
	<AANLkTi=NP_Kw-Hrb6+XXtuid3YWJUM2ehFaeoaV-UEaC@mail.gmail.com>
	<AANLkTinQ7g2YFCGNRtxWr-apHuByNFdF3Z_ph8J7K01e@mail.gmail.com>
	<AANLkTin_Y-azNb2sT-U9HQsjVbL3JqPiBamTiX5a_HC+@mail.gmail.com>
Message-ID: <AANLkTi=d18FUPX_mWDO4eLEW5Uf+PYFfmtLfZ9C=LTKR@mail.gmail.com>

ok, I see one under src. I copied that to conf/autoload_configs and looks
like I am able to load mod_xml_cdr this way
freeswitch at xxxx@internal>
load mod_xml_cdr
module_exists mod_xml_cdr
true
same method did not work for mod_xml_curl
below is error message after I copied file
cp /usr/src/freeswitch-1.0.6/conf/autoload_configs/xml_curl.conf.xml
/opt/freeswitch/conf/autoload_configs/xml_curl.conf.xml

freeswitch at xxx@internal> load mod_xml_curl
[DEBUG] libs/esl/src/esl.c:1140 esl_send() SEND
api load mod_xml_curl
console_execute: true

[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Type] =
[api/response]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Length]
= [68]
+OK Reloading XML
-ERR [module load file routine returned an error]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Type] =
[log/data]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Length]
= [63]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Level] =
[6]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Text-Channel] =
[0]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-File] =
[mod_enum.c]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Func] =
[event_handler]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Line] =
[808]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [User-Data] = []
[DEBUG] libs/esl/src/esl.c:1114 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 63
Log-Level: 6
Text-Channel: 0
Log-File: mod_enum.c
Log-Func: event_handler
Log-Line: 808
User-Data: _undef_
2011-03-02 19:23:34.416231 [INFO] mod_enum.c:808 ENUM Reloaded
2011-03-02 19:23:34.416231 [INFO] mod_enum.c:808 ENUM Reloaded
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Type] =
[log/data]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Length]
= [72]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Level] =
[3]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Text-Channel] =
[0]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-File] =
[mod_xml_curl.c]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Func] =
[do_config]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Line] =
[444]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [User-Data] = []
[DEBUG] libs/esl/src/esl.c:1114 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 72
Log-Level: 3
Text-Channel: 0
Log-File: mod_xml_curl.c
Log-Func: do_config
Log-Line: 444
User-Data: _undef_
2011-03-02 19:23:34.416231 [ERR] mod_xml_curl.c:444 Binding has no url!
2011-03-02 19:23:34.416231 [ERR] mod_xml_curl.c:444 Binding has no url!
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Type] =
[log/data]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Length]
= [162]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Level] =
[2]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Text-Channel] =
[0]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-File] =
[switch_loadable_module.c]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Func] =
[switch_loadable_module_load_file]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Line] =
[926]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [User-Data] = []
[DEBUG] libs/esl/src/esl.c:1114 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 162
Log-Level: 2
Text-Channel: 0
Log-File: switch_loadable_module.c
Log-Func: switch_loadable_module_load_file
Log-Line: 926
User-Data: _undef_
2011-03-02 19:23:34.416231 [CRIT] switch_loadable_module.c:926 Error Loading
module /opt/freeswitch/mod/mod_xml_curl.so
**Module load routine returned an error**
2011-03-02 19:23:34.416231 [CRIT] switch_loadable_module.c:926 Error Loading
module /opt/freeswitch/mod/mod_xml_curl.so
**Module load routine returned an error**
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Type] =
[log/data]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Length]
= [86]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Level] =
[6]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Text-Channel] =
[0]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-File] =
[switch_time.c]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Func] =
[switch_load_timezones]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Line] =
[950]
[DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [User-Data] = []
[DEBUG] libs/esl/src/esl.c:1114 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 86
Log-Level: 6
Text-Channel: 0
Log-File: switch_time.c
Log-Func: switch_load_timezones
Log-Line: 950
User-Data: _undef_
2011-03-02 19:23:34.417121 [INFO] switch_time.c:950 Timezone reloaded 530
definitions
2011-03-02 19:23:34.417121 [INFO] switch_time.c:950 Timezone reloaded 530
definitions











On Wed, Mar 2, 2011 at 6:25 PM, Steven Ayre <steveayre at gmail.com> wrote:

> You need to create its config file. It won't load without it.
>
> "find . -name xml_cdr.conf"
>
> Actually it's xml_cdr.conf.xml, all config files will show .conf as
> their logical name but on the hard drive they're stored in .conf.xml.
> It should be in conf/autoload_configs
>
> -Steve
>
>
> On 2 March 2011 22:25, deniro <ayhkor at gmail.com> wrote:
> > I typed the 4 commands but still unable to load mod_xml_cdr and
> mod_xml_curl
> >
> >
> > 2011-03-02 17:21:32.139672 [ERR] mod_xml_cdr.c:479 Open of xml_cdr.conf
> > failed
> > /opt# find . -name xml_cdr.conf
> > nothing appears
> >
> > here is errors from fs_cli -d 7
> >
> > freeswitch at 75.xxx.xxx.xx@internal> load mod_xml_cdr
> > [DEBUG] libs/esl/src/esl.c:1063 esl_send() SEND
> > api load mod_xml_cdr
> > console_execute: true
> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Type] =
> > [api/response]
> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length]
> > = [50]
> > -ERR [module load file routine returned an error]
> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Type] =
> > [log/data]
> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length]
> > = [79]
> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> > [3]
> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Text-Channel] =
> > [0]
> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> > [mod_xml_cdr.c]
> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> > [mod_xml_cdr_load]
> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> > [479]
> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> []
> > [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> > Event-Name: SOCKET_DATA
> > Content-Type: log/data
> > Content-Length: 79
> > Log-Level: 3
> > Text-Channel: 0
> > Log-File: mod_xml_cdr.c
> > Log-Func: mod_xml_cdr_load
> > Log-Line: 479
> > User-Data: _undef_
> > 2011-03-02 17:21:32.139672 [ERR] mod_xml_cdr.c:479 Open of xml_cdr.conf
> > failed
> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Type] =
> > [log/data]
> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length]
> > = [161]
> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> > [2]
> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Text-Channel] =
> > [0]
> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> > [switch_loadable_module.c]
> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> > [switch_loadable_module_load_file]
> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> > [882]
> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> []
> > [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> > Event-Name: SOCKET_DATA
> > Content-Type: log/data
> > Content-Length: 161
> > Log-Level: 2
> > Text-Channel: 0
> > Log-File: switch_loadable_module.c
> > Log-Func: switch_loadable_module_load_file
> > Log-Line: 882
> > User-Data: _undef_
> > 2011-03-02 17:21:32.139672 [CRIT] switch_loadable_module.c:882 Error
> Loading
> > module /opt/freeswitch/mod/mod_xml_cdr.so
> > **Module load routine returned an error**
> >
> >
> >
> >
> >
> > On Tue, Mar 1, 2011 at 1:11 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> like I said you re-installed with a new version of libcurl but did not
> >> fully rebuild.
> >> So most of the modules that depend on curl were broken torn between
> >> the old and new version it was linked to..
> >>
> >> had you executed the 4 commands I posted yesterday it would be fine.
> >>
> >>
> >> On Tue, Mar 1, 2011 at 11:02 AM, deniro <ayhkor at gmail.com> wrote:
> >> > Thank you all for advice  and direction
> >> > Will re-install from scratch at a later time.
> >> > deniro--
> >> >
> >> > On Tue, Mar 1, 2011 at 4:16 AM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >> >>
> >> >> Ok, you really need to pay attention to any log message that logs
> with
> >> >> the CRIT (critical) log level - they're really serious errors. You
> >> >> should look at adding the critical="true" attribute to all the tags
> in
> >> >> modules.conf.xml which you want to be sure are loaded. If a module
> >> >> fails to load without that tag FS will continue to run anyway, with
> >> >> that tag it'll refuse to start so you'll know instantly something is
> >> >> wrong.
> >> >>
> >> >> Most of your modules aren't loading, because they haven't been
> >> >> compiled correctly. Some look like they're missing (no such file or
> >> >> directory) and others haven't been built right (undefined symbols).
> >> >>
> >> >> Here are a few of the important lines from your log:
> >> >>
> >> >> 2011-02-28 18:12:21.910738 [CRIT] switch_loadable_module.c:882 Error
> >> >> Loading module /opt/freeswitch/mod/mod_event_socket.so
> >> >> **/opt/freeswitch/mod/mod_event_socket.so: undefined symbol:
> >> >> switch_event_serialize_json**
> >> >> 2011-02-28 18:12:21.910789 [CRIT] switch_loadable_module.c:882 Error
> >> >> Loading module /opt/freeswitch/mod/mod_xml_curl.so
> >> >> **/opt/freeswitch/mod/mod_xml_curl.so: cannot open shared object
> file:
> >> >> No such file or directory**
> >> >> 2011-02-28 18:12:21.910813 [CRIT] switch_loadable_module.c:882 Error
> >> >> Loading module /opt/freeswitch/mod/mod_xml_cdr.so
> >> >> **/opt/freeswitch/mod/mod_xml_cdr.so: cannot open shared object file:
> >> >> No such file or directory**
> >> >> 2011-02-28 18:12:21.911581 [CRIT] switch_loadable_module.c:882 Error
> >> >> Loading module /opt/freeswitch/mod/mod_commands.so
> >> >> **/opt/freeswitch/mod/mod_commands.so: undefined symbol:
> >> >> switch_xml_reload**
> >> >> 2011-02-28 18:12:21.911709 [CRIT] switch_loadable_module.c:882 Error
> >> >> Loading module /opt/freeswitch/mod/mod_conference.so
> >> >> **/opt/freeswitch/mod/mod_conference.so: undefined symbol:
> >> >> switch_channel_test_app_flag_key**
> >> >> 2011-02-28 18:12:21.911842 [CRIT] switch_loadable_module.c:882 Error
> >> >> Loading module /opt/freeswitch/mod/mod_dptools.so
> >> >> **/opt/freeswitch/mod/mod_dptools.so: undefined symbol:
> >> >> switch_event_merge**
> >> >> 2011-02-28 18:12:21.912264 [CRIT] switch_loadable_module.c:882 Error
> >> >> Loading module /opt/freeswitch/mod/mod_dialplan_xml.so
> >> >> **/opt/freeswitch/mod/mod_dialplan_xml.so: undefined symbol:
> >> >> switch_xml_std_datetime_check**
> >> >> 2011-02-28 18:12:21.912531 [CRIT] switch_loadable_module.c:882 Error
> >> >> Loading module /opt/freeswitch/mod/mod_g723_1.so
> >> >> **/opt/freeswitch/mod/mod_g723_1.so: cannot open shared object file:
> >> >> No such file or directory**
> >> >> 2011-02-28 18:12:21.912573 [CRIT] switch_loadable_module.c:882 Error
> >> >> Loading module /opt/freeswitch/mod/mod_g729.so
> >> >> **/opt/freeswitch/mod/mod_g729.so: cannot open shared object file: No
> >> >> such file or directory**
> >> >> 2011-02-28 18:12:21.912609 [CRIT] switch_loadable_module.c:882 Error
> >> >> Loading module /opt/freeswitch/mod/mod_amr.so
> >> >> **/opt/freeswitch/mod/mod_amr.so: cannot open shared object file: No
> >> >> such file or directory**
> >> >>
> >> >> Important things to note:
> >> >> mod_event_socket hasn't loaded because of a undefined symbol, it
> >> >> wasn't built correctly
> >> >> mod_xml_curl hasn't loaded because it doesn't exist, it was either
> >> >> never built or never installed
> >> >>
> >> >> Something pretty strange has happened. Did you recompile when trying
> >> >> to add mod_xml_curl?
> >> >>
> >> >> I'd suggest you delete all the FS files, including the Git clone,
> make
> >> >> a fresh checkout and build it from scratch.
> >> >>
> >> >> -Steve
> >> >>
> >> >>
> >> >>
> >> >> On 28 February 2011 23:18, deniro <ayhkor at gmail.com> wrote:
> >> >> > I put the log in pastebin
> >> >> > freeswitch.log only when starting freeswitch (after stop)
> >> >> >
> >> >> > http://pastebin.freeswitch.org/15502
> >> >> > thx
> >> >> > deniro
> >> >> > On Mon, Feb 28, 2011 at 5:46 PM, Steven Ayre <steveayre at gmail.com>
> >> >> > wrote:
> >> >> >>
> >> >> >> You must check the logfile for when FS starts up - if netstat
> shows
> >> >> >> nothing for port 8021 then either mod_event_socket isn't being
> >> >> >> loaded
> >> >> >> or you'll see an error when it tries to load.
> >> >> >>
> >> >> >> Nothing from netstat means nothing's listening, so trying to
> connect
> >> >> >> using fs_cli won't do anything.
> >> >> >>
> >> >> >> -Steve
> >> >> >>
> >> >> >>
> >> >> >> On 28 February 2011 22:40, deniro <ayhkor at gmail.com> wrote:
> >> >> >> > Checked the log
> >> >> >> > Nothing is logged when
> >> >> >> > fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
> >> >> >> > is issued
> >> >> >> > /opt/freeswitch/log# tail -f freeswitch.log
> >> >> >> >
> >> >> >> > ls -l freeswitch.log
> >> >> >> > -rwxrwxrwx 1 freeswitch daemon 33018 2011-02-28 17:39
> >> >> >> > freeswitch.log
> >> >> >> > thx
> >> >> >> >
> >> >> >> >
> >> >> >> >
> >> >> >> > On Mon, Feb 28, 2011 at 3:59 PM, Peter Olsson
> >> >> >> > <peter.olsson at visionutveckling.se> wrote:
> >> >> >> >>
> >> >> >> >> Check freeswitch.log, it probably reports some problem when
> >> >> >> >> loading
> >> >> >> >> the
> >> >> >> >> mod_event_socket module.
> >> >> >> >>
> >> >> >> >> /Peter
> >> >> >> >> ________________________________________
> >> >> >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> >> >> >> >> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r
> deniro
> >> >> >> >> [ayhkor at gmail.com]
> >> >> >> >> Skickat: den 28 februari 2011 21:42
> >> >> >> >> Till: FreeSWITCH Users Help
> >> >> >> >> ?mne: Re: [Freeswitch-users] fs_cli socket connection error
> >> >> >> >>
> >> >> >> >> Steve
> >> >> >> >> Thanks for the reply
> >> >> >> >> netstat -anlp | grep 8021  is blank (nothing showing up)
> >> >> >> >>
> >> >> >> >> /opt/freeswitch/bin/fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
> >> >> >> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
> >> >> >> >> Configuration
> >> >> >> >> file is /root/.fs_cli_conf.
> >> >> >> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
> >> >> >> >> Configuration
> >> >> >> >> file is /etc/fs_cli.conf.
> >> >> >> >> [DEBUG] libs/esl/fs_cli.c:1159 main() profile default does not
> >> >> >> >> exist
> >> >> >> >> using
> >> >> >> >> builtin profile
> >> >> >> >> [DEBUG] libs/esl/fs_cli.c:1188 main() Using profile internal
> >> >> >> >> [75.xxx.xxx.xxx]
> >> >> >> >> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
> >> >> >> >> Connection
> >> >> >> >> Error]
> >> >> >> >> yes mod_event_socket is after  mod_xml_curl but I changed the
> >> >> >> >> order
> >> >> >> >> in
> >> >> >> >>  modules.conf.xml
> >> >> >> >> still getting above (restarted freeswitch)
> >> >> >> >>
> >> >> >> >> thx
> >> >> >> >> deniro--
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> On Mon, Feb 28, 2011 at 3:35 AM, Steven Ayre
> >> >> >> >> <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
> >> >> >> >> Enable debug logging and you should see an error that'll tell
> you
> >> >> >> >> more.
> >> >> >> >>
> >> >> >> >> Is mod_event_socket loading before or after mod_xml_curl?
> Chances
> >> >> >> >> are
> >> >> >> >> mod_xml_curl is loading first, so it's trying to read
> >> >> >> >> event_socket.conf.xml and/or acl.conf.xml through xml_curl and
> >> >> >> >> either
> >> >> >> >> getting a different config to your previous local copy or the
> >> >> >> >> ACLs
> >> >> >> >> are
> >> >> >> >> different.
> >> >> >> >>
> >> >> >> >> What does "netstat -anlp | grep :8021" from the linux cli show
> >> >> >> >> you?
> >> >> >> >> Does it show that freeswitch is actually listening on the port?
> >> >> >> >> If
> >> >> >> >> it
> >> >> >> >> is it's probably an ACL problem, if it isn't then it's probably
> a
> >> >> >> >> problem with event_socket.conf.xml
> >> >> >> >>
> >> >> >> >> -Steve
> >> >> >> >>
> >> >> >> >> On 28 February 2011 00:53, deniro
> >> >> >> >> <ayhkor at gmail.com<mailto:ayhkor at gmail.com>> wrote:
> >> >> >> >> > What would be possible reasons for this and how to resolve?
> >> >> >> >> > running fs 106 on ubuntu 10.04 server
> >> >> >> >> >
> >> >> >> >> > /opt/freeswitch/bin/fs_cli -H xxx.xxx.xxx.xxx -P 8021
> >> >> >> >> > [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting
> [Socket
> >> >> >> >> > Connection
> >> >> >> >> > Error]
> >> >> >> >> > was working fine before I installed  mod_xml_cdr
> >> >> >> >> > configure --prefix=/opt/freeswitch --without-libcurl
> >> >> >> >> > make mod_xml_cdr-install
> >> >> >> >> > (no errors)
> >> >> >> >> >
> >> >> >> >> > in modules.conf.xml
> >> >> >> >> > <load module="mod_event_socket"/>
> >> >> >> >> >
> >> >> >> >> > is there a packaged version of mod_xml_cdr (for fs 1.0.6)?
> >> >> >> >> >
> >> >> >> >> > thx
> >> >> >> >> > deniro--
> >> >> >> >> > _______________________________________________
> >> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> >> >> >> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> > http://www.freeswitch.org<http://www.freeswitch.org/>
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >>
> >> >> >> >> _______________________________________________
> >> >> >> >> FreeSWITCH-users mailing list
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> http://www.freeswitch.org<http://www.freeswitch.org/>
> >> >> >> >>
> >> >> >> >> !DSPAM:4d6c0a1132761029518849!
> >> >> >> >>
> >> >> >> >> _______________________________________________
> >> >> >> >> FreeSWITCH-users mailing list
> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >> > _______________________________________________
> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >
> >> >> >> >
> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> > http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > 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-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From u2nsam at gmail.com  Thu Mar  3 07:47:33 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 3 Mar 2011 10:17:33 +0530
Subject: [Freeswitch-users] count
Message-ID: <AANLkTi=oKxBtho8q5kQq52ZxrU=r++SzFXkVzjq=P8B1@mail.gmail.com>

Hello,

Whats the command to get the participants count in the conference ?

Rwgards
Sam
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From frankie.k.yiu at gmail.com  Thu Mar  3 08:03:38 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Wed, 2 Mar 2011 21:03:38 -0800
Subject: [Freeswitch-users] How to create IVR application in C#?
Message-ID: <AANLkTi=GBZp5cbDbGmEya8jDhTTDF=LH4apNogBLAzkf@mail.gmail.com>

Hi Bob or anyone can help,

I would like to know if you have used the function
"switch_ivr_menu_init(*SWIGTYPE_ p_p_switch_ivr_menu*  new_menu,.......) "
in your C# code?  If so, how did you call this function specially on the
first parameter?  It crashes on me because on the C code side, it is a
pointer to pointer type and if I just passed in a :

*SWIGTYPE_ p_p_switch_ivr_menu*  new_menu = null;

and it does not work, and there is no other way to initialize it, as I see
it right now.

Please let me know how to resolve this kind of problem (related to "*SWIGTYPE_
p_p"* type).  Or is there other functions that I can use and get a new
*switch_ivr_menu
*type back?

Thank you!!!

Frankie

---------- Forwarded message ----------

> From: Bob Coleman <bobc at devassert.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Wed, 2 Mar 2011 18:34:14 +1300
> Subject: Re: [Freeswitch-users] How to create IVR application in C#?
> Another easy way is to use mod event socket
>
> When you download the source, there is a libs/esl/managed folder that
> has the ESL project for .NET
>
> You can use this wrapper to talk to FreeSWITCH and write an IVR app
> without much trouble.
>
> The event socket route is really nice, I have built a complete windows
> ivr stack using FreeSWITCH and help from the guys here, very reliable
> and no maintenance.
>
> Bob
>
> On Wed, Mar 2, 2011 at 10:01 AM, Frankie Yiu <frankie.k.yiu at gmail.com>
> wrote:
> > Hi there,
> >
> > I am newbie to FreeSwitch and I have question about creating an IVR
> > application in C#, with a possibly of using VoiceXML.  Could someone
> please
> > points me to how I can get started or any example that I can look at?
> >
>
>
>
>
>
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From u2nsam at gmail.com  Thu Mar  3 08:14:08 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 3 Mar 2011 10:44:08 +0530
Subject: [Freeswitch-users] count
In-Reply-To: <AANLkTi=oKxBtho8q5kQq52ZxrU=r++SzFXkVzjq=P8B1@mail.gmail.com>
References: <AANLkTi=oKxBtho8q5kQq52ZxrU=r++SzFXkVzjq=P8B1@mail.gmail.com>
Message-ID: <AANLkTinyzQ4gOm6JZvMzP66pkW9nODqRuE6Pc5+ps2d+@mail.gmail.com>

And how to specify the threshold for the conference on the fly when the
someone hits the conference dial-plan.

Regards
Sam

On Thu, Mar 3, 2011 at 10:17 AM, Sam <u2nsam at gmail.com> wrote:

> Hello,
>
> Whats the command to get the participants count in the conference ?
>
> Rwgards
> Sam
>
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From adminjew at gmail.com  Thu Mar  3 08:18:16 2011
From: adminjew at gmail.com (Yitzchok)
Date: Thu, 3 Mar 2011 00:18:16 -0500
Subject: [Freeswitch-users] How to create IVR application in C#?
In-Reply-To: <AANLkTi=GBZp5cbDbGmEya8jDhTTDF=LH4apNogBLAzkf@mail.gmail.com>
References: <AANLkTi=GBZp5cbDbGmEya8jDhTTDF=LH4apNogBLAzkf@mail.gmail.com>
Message-ID: <AANLkTimb2RdXGHt0eFchQ1i0TOT_mbaJZYpHj2q2h9HM@mail.gmail.com>

I uploaded a while ago some helpers for Freeswitch in .net you can find it
on github @ https://github.com/Yitzchok/FreeSwitchUtilitiesDotNet


<https://github.com/Yitzchok/FreeSwitchUtilitiesDotNet>

public class VoiceQuestionnaireServiceSample : IAppPlugin
    {
        public ManagedSession Session { get; set; }
        private const string PhraseStart = "phrase:sample_ivr_";
        private const string InvalidAudioFile = PhraseStart + "invalid";
        private const string RecordLocation =
"/usr/local/freeswitch/recordings/";

        public void Run(AppContext context)
        {
            Session = context.Session;
            Session.Answer();
            Session.SetTtsParameters("flite", "kal");
            Session.sleep(1000, 0);

            var questionnaireService = new VoiceQuestionnaireService(
                                            Session, InvalidAudioFile,
                                            PhraseStart, RecordLocation);

            var phoneNumber = questionnaireService
                .AskAndVerifyQuestion(
                9, 11, 3, 10000, "#", "enter_phone_number",
                x => "you_entered_phone_number:" + x, "\\d+", _ => true);

            var nameRecordingFile = questionnaireService
                    .AskRecordAndVerifyQuestion(
                    "record_name", x => "verify_recording:" + x,
                    string.Format("{0}_{1}_{2}", Session.Uuid,
DateTime.UtcNow.ToString("MMddyyyy"), phoneNumber));

        }
    }

Hope this helps.

Yitzchok


On Thu, Mar 3, 2011 at 12:03 AM, Frankie Yiu <frankie.k.yiu at gmail.com>wrote:

> Hi Bob or anyone can help,
>
> I would like to know if you have used the function
> "switch_ivr_menu_init(*SWIGTYPE_ p_p_switch_ivr_menu*  new_menu,.......) "
> in your C# code?  If so, how did you call this function specially on the
> first parameter?  It crashes on me because on the C code side, it is a
> pointer to pointer type and if I just passed in a :
>
> *SWIGTYPE_ p_p_switch_ivr_menu*  new_menu = null;
>
> and it does not work, and there is no other way to initialize it, as I see
> it right now.
>
> Please let me know how to resolve this kind of problem (related to "*SWIGTYPE_
> p_p"* type).  Or is there other functions that I can use and get a new *switch_ivr_menu
> *type back?
>
> Thank you!!!
>
> Frankie
>
> ---------- Forwarded message ----------
>
>> From: Bob Coleman <bobc at devassert.com>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Date: Wed, 2 Mar 2011 18:34:14 +1300
>> Subject: Re: [Freeswitch-users] How to create IVR application in C#?
>> Another easy way is to use mod event socket
>>
>> When you download the source, there is a libs/esl/managed folder that
>> has the ESL project for .NET
>>
>> You can use this wrapper to talk to FreeSWITCH and write an IVR app
>> without much trouble.
>>
>> The event socket route is really nice, I have built a complete windows
>> ivr stack using FreeSWITCH and help from the guys here, very reliable
>> and no maintenance.
>>
>> Bob
>>
>> On Wed, Mar 2, 2011 at 10:01 AM, Frankie Yiu <frankie.k.yiu at gmail.com>
>> wrote:
>> > Hi there,
>> >
>> > I am newbie to FreeSwitch and I have question about creating an IVR
>> > application in C#, with a possibly of using VoiceXML.  Could someone
>> please
>> > points me to how I can get started or any example that I can look at?
>> >
>>
>>
>>
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From spencer at 5ninesolutions.com  Thu Mar  3 09:03:59 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Thu, 3 Mar 2011 06:03:59 +0000
Subject: [Freeswitch-users] Polycom G722 negotiation
In-Reply-To: <378E398F-FB0F-40D9-8DE4-F1026E77DF40@freeswitch.org>
References: <AE216221-5892-4337-8985-1889410A866F@5ninesolutions.com>
	<2867CE13-6EED-4FAF-9DF1-49451D164721@freeswitch.org>
	<0ABF546B-4000-4936-AD22-ED6B956277D5@5ninesolutions.com>
	<1298907264976-6073705.post@n2.nabble.com>
	<1F4FD420-3CE9-4601-9D44-80C8BEEF9707@5ninesolutions.com>
	<AANLkTikFCLzNvqpHEO4qtCyFDQOTZ0CG5pOw+SJD=2Yz@mail.gmail.com>
	<5AB8CE7C-5283-4585-8DF8-856A3822F3BD@5ninesolutions.com>
	<AANLkTik0jy=VDy2p4XnAiAwJkT6jX=PAY1W6FgRV7cDC@mail.gmail.com>
	<C72FE323-9B29-43F3-A17E-A327DAD3B07B@5ninesolutions.com>
	<378E398F-FB0F-40D9-8DE4-F1026E77DF40@freeswitch.org>
Message-ID: <F24EF3DA-830E-4D9C-A491-C29F822B6BDA@5ninesolutions.com>

If you'd like I can pastebin my config files.  But with 3.3.1 its my understanding that there is no sip.cfg file anymore since Polycom changed the config format.  I did however base my config files off the ones in the /configs directory in the firmware zip file and specifically everything codec related has not been changed from the defaults.




On Mar 1, 2011, at 12:28 PM, Brian West wrote:

> I'm running 3.3.1 and I'm not seeing this behavior... did you happen to not install the default sip.cfg that came with 3.3.1?
> 
> /b
> 
> On Feb 28, 2011, at 3:37 PM, Spencer Thomason wrote:
> 
>> Yes, that what I thought.  I'm running the latest firmware, 3.3.1.   
>> I'll report it to them as its clearly offering a different SDP after  
>> the unhold.  Yes it might be nice if Freeswitch refused to switch  
>> codec on a call that is up even if the negotiation is set to generous.
>> 
>> Thanks for your help,
>> Spencer
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From msc at freeswitch.org  Thu Mar  3 09:43:24 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 2 Mar 2011 22:43:24 -0800
Subject: [Freeswitch-users] checking voicemail without ID?
In-Reply-To: <4D6E6921.9000604@julf.com>
References: <4D6E555F.4000003@julf.com>
	<649CD6B6-AB32-41B0-8C9D-2BB5F3DEA9FD@freeswitch.org>
	<4D6E63FE.1020302@julf.com>
	<518984EC-1B52-4631-B3D5-2DB6776255C5@freeswitch.org>
	<4D6E6921.9000604@julf.com>
Message-ID: <AANLkTi=gj4QzuR2AAdMphwxpzOfSu9B1Fy4HHWyA8-YN@mail.gmail.com>

On Wed, Mar 2, 2011 at 7:58 AM, Johan Helsingius <julf at julf.com> wrote:

> Brian,
>
> > Show us the whole dialplan.
>
> Here is the voicemail part:
>
>  <!-- voicemail main extension -->
>    <extension name="vmain">
>      <condition field="destination_number" expression="^vmain$|^\*98$">
>        <action application="answer"/>
>        <action application="sleep" data="1000"/>
>        <action application="set"
> data="voicemail_authorized=${sip_authorized}"/>
>

Don't you need to execute this one inline?

>         <action application="set" data="voicemail_authorized=true"/>
>
Should be
 <action application="set" data="voicemail_authorized=true" inline="true"/>

Try that and let us know what happens.
-MC
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From msc at freeswitch.org  Thu Mar  3 09:56:52 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 2 Mar 2011 22:56:52 -0800
Subject: [Freeswitch-users] API command
In-Reply-To: <AANLkTim-dEY_DOLQPUf9apOwFyntFsNVPhnvUXOp9_2d@mail.gmail.com>
References: <AANLkTim-dEY_DOLQPUf9apOwFyntFsNVPhnvUXOp9_2d@mail.gmail.com>
Message-ID: <AANLkTinkxPVyTAQDFnoatZzJv1imWQWE5zxUfX4vNY76@mail.gmail.com>

On Wed, Mar 2, 2011 at 1:48 AM, Sam <u2nsam at gmail.com> wrote:

> HI,
>
>
> Suppose if i want to place an external caller in the conference so i should
> use the below command ?
>
> originate freetdm/wp1/a/9322273640  67287006 context &bridge 7050
>

What are 67287006 and "context"? if you want to transfer to ext 7050 in a
context named "67287006" then do this:
originate freetdm/wp1/a/9322273640  7050 XML 67287006

remember that the second argument to originate is a dialplan destination
number OR an &application(arg)

-MC


> where 7050 leads to conference .
>
>
> Regds
> Sam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Thu Mar  3 11:32:55 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 3 Mar 2011 08:32:55 +0000
Subject: [Freeswitch-users] fs_cli socket connection error
In-Reply-To: <AANLkTi=d18FUPX_mWDO4eLEW5Uf+PYFfmtLfZ9C=LTKR@mail.gmail.com>
References: <AANLkTikCoT9dzLNoD=6TzS0jiMw4afNX94TypsLarBbz@mail.gmail.com>
	<AANLkTi=sENgBB9Cdr-EkLJD8XfmH3A8GSfPnvDHO7qrw@mail.gmail.com>
	<AANLkTin5k=2Z-=QVVzqt91172wv88P7OFkqRqTY9VDcS@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58ADB727D7@cooper>
	<AANLkTi=aPwmnVD8ZzaBVFDxGyPsbgzDEBGaXvq_B2NbF@mail.gmail.com>
	<AANLkTikpQuxvyaU7PAGnyJuoQLQ9xn4inNJ_u5x41Anv@mail.gmail.com>
	<AANLkTimNYjC_G-Lq+St5VePpxEyM8iHnhaNCo=jBydi9@mail.gmail.com>
	<AANLkTi=QXgEb6dHU9bfb7v06WzPyWJcoNYDLi1X9oq+E@mail.gmail.com>
	<AANLkTikEDgpAJppNUnLac73ip7qsk+8AK-N-pTW7NBm6@mail.gmail.com>
	<AANLkTi=NP_Kw-Hrb6+XXtuid3YWJUM2ehFaeoaV-UEaC@mail.gmail.com>
	<AANLkTinQ7g2YFCGNRtxWr-apHuByNFdF3Z_ph8J7K01e@mail.gmail.com>
	<AANLkTin_Y-azNb2sT-U9HQsjVbL3JqPiBamTiX5a_HC+@mail.gmail.com>
	<AANLkTi=d18FUPX_mWDO4eLEW5Uf+PYFfmtLfZ9C=LTKR@mail.gmail.com>
Message-ID: <AANLkTim8HxbSxoVnfFHWmG5WxdWsvHF4v7rX-ZonuW6k@mail.gmail.com>

freeswitch at xxx@internal> load mod_xml_curl
-ERR [module load file routine returned an error]

There's still an error while loading. You need to look at the logs for
freeswitch, not the fs_cli ones.

Either run 'fs_cli -l debug' or do '/log 7' within fs_cli

-Steve

On 3 March 2011 00:28, deniro <ayhkor at gmail.com> wrote:
> ok, I see one under src. I copied that to conf/autoload_configs and looks
> like I am able to load mod_xml_cdr this way
> freeswitch at xxxx@internal>
> load mod_xml_cdr
> module_exists mod_xml_cdr
> true
> same method did not work for mod_xml_curl
> below is error message after I copied file
> cp /usr/src/freeswitch-1.0.6/conf/autoload_configs/xml_curl.conf.xml
> /opt/freeswitch/conf/autoload_configs/xml_curl.conf.xml
>
> freeswitch at xxx@internal> load mod_xml_curl
> [DEBUG] libs/esl/src/esl.c:1140 esl_send() SEND
> api load mod_xml_curl
> console_execute: true
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Type] =
> [api/response]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Length]
> = [68]
> +OK Reloading XML
> -ERR [module load file routine returned an error]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Type] =
> [log/data]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Length]
> = [63]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Level] =
> [6]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Text-Channel] =
> [0]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-File] =
> [mod_enum.c]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Func] =
> [event_handler]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Line] =
> [808]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [User-Data] = []
> [DEBUG] libs/esl/src/esl.c:1114 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 63
> Log-Level: 6
> Text-Channel: 0
> Log-File: mod_enum.c
> Log-Func: event_handler
> Log-Line: 808
> User-Data: _undef_
> 2011-03-02 19:23:34.416231 [INFO] mod_enum.c:808 ENUM Reloaded
> 2011-03-02 19:23:34.416231 [INFO] mod_enum.c:808 ENUM Reloaded
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Type] =
> [log/data]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Length]
> = [72]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Level] =
> [3]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Text-Channel] =
> [0]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-File] =
> [mod_xml_curl.c]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Func] =
> [do_config]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Line] =
> [444]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [User-Data] = []
> [DEBUG] libs/esl/src/esl.c:1114 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 72
> Log-Level: 3
> Text-Channel: 0
> Log-File: mod_xml_curl.c
> Log-Func: do_config
> Log-Line: 444
> User-Data: _undef_
> 2011-03-02 19:23:34.416231 [ERR] mod_xml_curl.c:444 Binding has no url!
> 2011-03-02 19:23:34.416231 [ERR] mod_xml_curl.c:444 Binding has no url!
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Type] =
> [log/data]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Length]
> = [162]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Level] =
> [2]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Text-Channel] =
> [0]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-File] =
> [switch_loadable_module.c]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_loadable_module_load_file]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Line] =
> [926]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [User-Data] = []
> [DEBUG] libs/esl/src/esl.c:1114 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 162
> Log-Level: 2
> Text-Channel: 0
> Log-File: switch_loadable_module.c
> Log-Func: switch_loadable_module_load_file
> Log-Line: 926
> User-Data: _undef_
> 2011-03-02 19:23:34.416231 [CRIT] switch_loadable_module.c:926 Error Loading
> module /opt/freeswitch/mod/mod_xml_curl.so
> **Module load routine returned an error**
> 2011-03-02 19:23:34.416231 [CRIT] switch_loadable_module.c:926 Error Loading
> module /opt/freeswitch/mod/mod_xml_curl.so
> **Module load routine returned an error**
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Type] =
> [log/data]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Length]
> = [86]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Level] =
> [6]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Text-Channel] =
> [0]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-File] =
> [switch_time.c]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_load_timezones]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Line] =
> [950]
> [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [User-Data] = []
> [DEBUG] libs/esl/src/esl.c:1114 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 86
> Log-Level: 6
> Text-Channel: 0
> Log-File: switch_time.c
> Log-Func: switch_load_timezones
> Log-Line: 950
> User-Data: _undef_
> 2011-03-02 19:23:34.417121 [INFO] switch_time.c:950 Timezone reloaded 530
> definitions
> 2011-03-02 19:23:34.417121 [INFO] switch_time.c:950 Timezone reloaded 530
> definitions
>
>
>
>
>
>
>
>
>
>
> On Wed, Mar 2, 2011 at 6:25 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> You need to create its config file. It won't load without it.
>>
>> "find . -name xml_cdr.conf"
>>
>> Actually it's xml_cdr.conf.xml, all config files will show .conf as
>> their logical name but on the hard drive they're stored in .conf.xml.
>> It should be in conf/autoload_configs
>>
>> -Steve
>>
>>
>> On 2 March 2011 22:25, deniro <ayhkor at gmail.com> wrote:
>> > I typed the 4 commands but still unable to load mod_xml_cdr and
>> > mod_xml_curl
>> >
>> >
>> > 2011-03-02 17:21:32.139672 [ERR] mod_xml_cdr.c:479 Open of xml_cdr.conf
>> > failed
>> > /opt# find . -name xml_cdr.conf
>> > nothing appears
>> >
>> > here is errors from fs_cli -d 7
>> >
>> > freeswitch at 75.xxx.xxx.xx@internal> load mod_xml_cdr
>> > [DEBUG] libs/esl/src/esl.c:1063 esl_send() SEND
>> > api load mod_xml_cdr
>> > console_execute: true
>> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>> > [Content-Type] =
>> > [api/response]
>> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>> > [Content-Length]
>> > = [50]
>> > -ERR [module load file routine returned an error]
>> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>> > [Content-Type] =
>> > [log/data]
>> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>> > [Content-Length]
>> > = [79]
>> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level]
>> > =
>> > [3]
>> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>> > [Text-Channel] =
>> > [0]
>> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
>> > [mod_xml_cdr.c]
>> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
>> > [mod_xml_cdr_load]
>> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
>> > [479]
>> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data]
>> > = []
>> > [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>> > Event-Name: SOCKET_DATA
>> > Content-Type: log/data
>> > Content-Length: 79
>> > Log-Level: 3
>> > Text-Channel: 0
>> > Log-File: mod_xml_cdr.c
>> > Log-Func: mod_xml_cdr_load
>> > Log-Line: 479
>> > User-Data: _undef_
>> > 2011-03-02 17:21:32.139672 [ERR] mod_xml_cdr.c:479 Open of xml_cdr.conf
>> > failed
>> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>> > [Content-Type] =
>> > [log/data]
>> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>> > [Content-Length]
>> > = [161]
>> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level]
>> > =
>> > [2]
>> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>> > [Text-Channel] =
>> > [0]
>> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
>> > [switch_loadable_module.c]
>> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
>> > [switch_loadable_module_load_file]
>> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
>> > [882]
>> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data]
>> > = []
>> > [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>> > Event-Name: SOCKET_DATA
>> > Content-Type: log/data
>> > Content-Length: 161
>> > Log-Level: 2
>> > Text-Channel: 0
>> > Log-File: switch_loadable_module.c
>> > Log-Func: switch_loadable_module_load_file
>> > Log-Line: 882
>> > User-Data: _undef_
>> > 2011-03-02 17:21:32.139672 [CRIT] switch_loadable_module.c:882 Error
>> > Loading
>> > module /opt/freeswitch/mod/mod_xml_cdr.so
>> > **Module load routine returned an error**
>> >
>> >
>> >
>> >
>> >
>> > On Tue, Mar 1, 2011 at 1:11 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> like I said you re-installed with a new version of libcurl but did not
>> >> fully rebuild.
>> >> So most of the modules that depend on curl were broken torn between
>> >> the old and new version it was linked to..
>> >>
>> >> had you executed the 4 commands I posted yesterday it would be fine.
>> >>
>> >>
>> >> On Tue, Mar 1, 2011 at 11:02 AM, deniro <ayhkor at gmail.com> wrote:
>> >> > Thank you all for advice? and direction
>> >> > Will re-install from scratch at a later time.
>> >> > deniro--
>> >> >
>> >> > On Tue, Mar 1, 2011 at 4:16 AM, Steven Ayre <steveayre at gmail.com>
>> >> > wrote:
>> >> >>
>> >> >> Ok, you really need to pay attention to any log message that logs
>> >> >> with
>> >> >> the CRIT (critical) log level - they're really serious errors. You
>> >> >> should look at adding the critical="true" attribute to all the tags
>> >> >> in
>> >> >> modules.conf.xml which you want to be sure are loaded. If a module
>> >> >> fails to load without that tag FS will continue to run anyway, with
>> >> >> that tag it'll refuse to start so you'll know instantly something is
>> >> >> wrong.
>> >> >>
>> >> >> Most of your modules aren't loading, because they haven't been
>> >> >> compiled correctly. Some look like they're missing (no such file or
>> >> >> directory) and others haven't been built right (undefined symbols).
>> >> >>
>> >> >> Here are a few of the important lines from your log:
>> >> >>
>> >> >> 2011-02-28 18:12:21.910738 [CRIT] switch_loadable_module.c:882 Error
>> >> >> Loading module /opt/freeswitch/mod/mod_event_socket.so
>> >> >> **/opt/freeswitch/mod/mod_event_socket.so: undefined symbol:
>> >> >> switch_event_serialize_json**
>> >> >> 2011-02-28 18:12:21.910789 [CRIT] switch_loadable_module.c:882 Error
>> >> >> Loading module /opt/freeswitch/mod/mod_xml_curl.so
>> >> >> **/opt/freeswitch/mod/mod_xml_curl.so: cannot open shared object
>> >> >> file:
>> >> >> No such file or directory**
>> >> >> 2011-02-28 18:12:21.910813 [CRIT] switch_loadable_module.c:882 Error
>> >> >> Loading module /opt/freeswitch/mod/mod_xml_cdr.so
>> >> >> **/opt/freeswitch/mod/mod_xml_cdr.so: cannot open shared object
>> >> >> file:
>> >> >> No such file or directory**
>> >> >> 2011-02-28 18:12:21.911581 [CRIT] switch_loadable_module.c:882 Error
>> >> >> Loading module /opt/freeswitch/mod/mod_commands.so
>> >> >> **/opt/freeswitch/mod/mod_commands.so: undefined symbol:
>> >> >> switch_xml_reload**
>> >> >> 2011-02-28 18:12:21.911709 [CRIT] switch_loadable_module.c:882 Error
>> >> >> Loading module /opt/freeswitch/mod/mod_conference.so
>> >> >> **/opt/freeswitch/mod/mod_conference.so: undefined symbol:
>> >> >> switch_channel_test_app_flag_key**
>> >> >> 2011-02-28 18:12:21.911842 [CRIT] switch_loadable_module.c:882 Error
>> >> >> Loading module /opt/freeswitch/mod/mod_dptools.so
>> >> >> **/opt/freeswitch/mod/mod_dptools.so: undefined symbol:
>> >> >> switch_event_merge**
>> >> >> 2011-02-28 18:12:21.912264 [CRIT] switch_loadable_module.c:882 Error
>> >> >> Loading module /opt/freeswitch/mod/mod_dialplan_xml.so
>> >> >> **/opt/freeswitch/mod/mod_dialplan_xml.so: undefined symbol:
>> >> >> switch_xml_std_datetime_check**
>> >> >> 2011-02-28 18:12:21.912531 [CRIT] switch_loadable_module.c:882 Error
>> >> >> Loading module /opt/freeswitch/mod/mod_g723_1.so
>> >> >> **/opt/freeswitch/mod/mod_g723_1.so: cannot open shared object file:
>> >> >> No such file or directory**
>> >> >> 2011-02-28 18:12:21.912573 [CRIT] switch_loadable_module.c:882 Error
>> >> >> Loading module /opt/freeswitch/mod/mod_g729.so
>> >> >> **/opt/freeswitch/mod/mod_g729.so: cannot open shared object file:
>> >> >> No
>> >> >> such file or directory**
>> >> >> 2011-02-28 18:12:21.912609 [CRIT] switch_loadable_module.c:882 Error
>> >> >> Loading module /opt/freeswitch/mod/mod_amr.so
>> >> >> **/opt/freeswitch/mod/mod_amr.so: cannot open shared object file: No
>> >> >> such file or directory**
>> >> >>
>> >> >> Important things to note:
>> >> >> mod_event_socket hasn't loaded because of a undefined symbol, it
>> >> >> wasn't built correctly
>> >> >> mod_xml_curl hasn't loaded because it doesn't exist, it was either
>> >> >> never built or never installed
>> >> >>
>> >> >> Something pretty strange has happened. Did you recompile when trying
>> >> >> to add mod_xml_curl?
>> >> >>
>> >> >> I'd suggest you delete all the FS files, including the Git clone,
>> >> >> make
>> >> >> a fresh checkout and build it from scratch.
>> >> >>
>> >> >> -Steve
>> >> >>
>> >> >>
>> >> >>
>> >> >> On 28 February 2011 23:18, deniro <ayhkor at gmail.com> wrote:
>> >> >> > I put the log in pastebin
>> >> >> > freeswitch.log only?when starting freeswitch (after stop)
>> >> >> >
>> >> >> > http://pastebin.freeswitch.org/15502
>> >> >> > thx
>> >> >> > deniro
>> >> >> > On Mon, Feb 28, 2011 at 5:46 PM, Steven Ayre <steveayre at gmail.com>
>> >> >> > wrote:
>> >> >> >>
>> >> >> >> You must check the logfile for when FS starts up - if netstat
>> >> >> >> shows
>> >> >> >> nothing for port 8021 then either mod_event_socket isn't being
>> >> >> >> loaded
>> >> >> >> or you'll see an error when it tries to load.
>> >> >> >>
>> >> >> >> Nothing from netstat means nothing's listening, so trying to
>> >> >> >> connect
>> >> >> >> using fs_cli won't do anything.
>> >> >> >>
>> >> >> >> -Steve
>> >> >> >>
>> >> >> >>
>> >> >> >> On 28 February 2011 22:40, deniro <ayhkor at gmail.com> wrote:
>> >> >> >> > Checked the log
>> >> >> >> > Nothing is logged when
>> >> >> >> > fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
>> >> >> >> > is issued
>> >> >> >> > /opt/freeswitch/log# tail -f freeswitch.log
>> >> >> >> >
>> >> >> >> > ls -l freeswitch.log
>> >> >> >> > -rwxrwxrwx 1 freeswitch daemon 33018 2011-02-28 17:39
>> >> >> >> > freeswitch.log
>> >> >> >> > thx
>> >> >> >> >
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > On Mon, Feb 28, 2011 at 3:59 PM, Peter Olsson
>> >> >> >> > <peter.olsson at visionutveckling.se> wrote:
>> >> >> >> >>
>> >> >> >> >> Check freeswitch.log, it probably reports some problem when
>> >> >> >> >> loading
>> >> >> >> >> the
>> >> >> >> >> mod_event_socket module.
>> >> >> >> >>
>> >> >> >> >> /Peter
>> >> >> >> >> ________________________________________
>> >> >> >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> >> >> >> >> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r
>> >> >> >> >> deniro
>> >> >> >> >> [ayhkor at gmail.com]
>> >> >> >> >> Skickat: den 28 februari 2011 21:42
>> >> >> >> >> Till: FreeSWITCH Users Help
>> >> >> >> >> ?mne: Re: [Freeswitch-users] fs_cli socket connection error
>> >> >> >> >>
>> >> >> >> >> Steve
>> >> >> >> >> Thanks for the reply
>> >> >> >> >> netstat -anlp | grep 8021 ?is blank (nothing showing up)
>> >> >> >> >>
>> >> >> >> >> /opt/freeswitch/bin/fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
>> >> >> >> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
>> >> >> >> >> Configuration
>> >> >> >> >> file is /root/.fs_cli_conf.
>> >> >> >> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
>> >> >> >> >> Configuration
>> >> >> >> >> file is /etc/fs_cli.conf.
>> >> >> >> >> [DEBUG] libs/esl/fs_cli.c:1159 main() profile default does not
>> >> >> >> >> exist
>> >> >> >> >> using
>> >> >> >> >> builtin profile
>> >> >> >> >> [DEBUG] libs/esl/fs_cli.c:1188 main() Using profile internal
>> >> >> >> >> [75.xxx.xxx.xxx]
>> >> >> >> >> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
>> >> >> >> >> Connection
>> >> >> >> >> Error]
>> >> >> >> >> yes mod_event_socket is after ?mod_xml_curl but I changed the
>> >> >> >> >> order
>> >> >> >> >> in
>> >> >> >> >> ?modules.conf.xml
>> >> >> >> >> still getting above (restarted freeswitch)
>> >> >> >> >>
>> >> >> >> >> thx
>> >> >> >> >> deniro--
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> On Mon, Feb 28, 2011 at 3:35 AM, Steven Ayre
>> >> >> >> >> <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
>> >> >> >> >> Enable debug logging and you should see an error that'll tell
>> >> >> >> >> you
>> >> >> >> >> more.
>> >> >> >> >>
>> >> >> >> >> Is mod_event_socket loading before or after mod_xml_curl?
>> >> >> >> >> Chances
>> >> >> >> >> are
>> >> >> >> >> mod_xml_curl is loading first, so it's trying to read
>> >> >> >> >> event_socket.conf.xml and/or acl.conf.xml through xml_curl and
>> >> >> >> >> either
>> >> >> >> >> getting a different config to your previous local copy or the
>> >> >> >> >> ACLs
>> >> >> >> >> are
>> >> >> >> >> different.
>> >> >> >> >>
>> >> >> >> >> What does "netstat -anlp | grep :8021" from the linux cli show
>> >> >> >> >> you?
>> >> >> >> >> Does it show that freeswitch is actually listening on the
>> >> >> >> >> port?
>> >> >> >> >> If
>> >> >> >> >> it
>> >> >> >> >> is it's probably an ACL problem, if it isn't then it's
>> >> >> >> >> probably a
>> >> >> >> >> problem with event_socket.conf.xml
>> >> >> >> >>
>> >> >> >> >> -Steve
>> >> >> >> >>
>> >> >> >> >> On 28 February 2011 00:53, deniro
>> >> >> >> >> <ayhkor at gmail.com<mailto:ayhkor at gmail.com>> wrote:
>> >> >> >> >> > What would be possible reasons for this and how to resolve?
>> >> >> >> >> > running fs 106 on ubuntu 10.04 server
>> >> >> >> >> >
>> >> >> >> >> > /opt/freeswitch/bin/fs_cli -H xxx.xxx.xxx.xxx -P 8021
>> >> >> >> >> > [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting
>> >> >> >> >> > [Socket
>> >> >> >> >> > Connection
>> >> >> >> >> > Error]
>> >> >> >> >> > was working fine before I installed ?mod_xml_cdr
>> >> >> >> >> > configure --prefix=/opt/freeswitch --without-libcurl
>> >> >> >> >> > make mod_xml_cdr-install
>> >> >> >> >> > (no errors)
>> >> >> >> >> >
>> >> >> >> >> > in modules.conf.xml
>> >> >> >> >> > <load module="mod_event_socket"/>
>> >> >> >> >> >
>> >> >> >> >> > is there a packaged version of mod_xml_cdr (for fs 1.0.6)?
>> >> >> >> >> >
>> >> >> >> >> > thx
>> >> >> >> >> > deniro--
>> >> >> >> >> > _______________________________________________
>> >> >> >> >> > FreeSWITCH-users mailing list
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> > FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >> >> >> >> >
>> >> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> >> > http://www.freeswitch.org<http://www.freeswitch.org/>
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >>
>> >> >> >> >> _______________________________________________
>> >> >> >> >> FreeSWITCH-users mailing list
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> >> http://www.freeswitch.org<http://www.freeswitch.org/>
>> >> >> >> >>
>> >> >> >> >> !DSPAM:4d6c0a1132761029518849!
>> >> >> >> >>
>> >> >> >> >> _______________________________________________
>> >> >> >> >> FreeSWITCH-users mailing list
>> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> >> http://www.freeswitch.org
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > _______________________________________________
>> >> >> >> > FreeSWITCH-users mailing list
>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> > http://www.freeswitch.org
>> >> >> >> >
>> >> >> >> >
>> >> >> >>
>> >> >> >> _______________________________________________
>> >> >> >> FreeSWITCH-users mailing list
>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > 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-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From jrichey at itltd.net  Thu Mar  3 01:56:15 2011
From: jrichey at itltd.net (JRichey)
Date: Wed, 2 Mar 2011 14:56:15 -0800 
Subject: [Freeswitch-users] fs_cli socket connection error
Message-ID: <6ECAF1527329364583AB525CF34ABF950B31A527@ms.kallback.com>

The file you should be looking for is "xml_cdr.conf.xml".  You can copy the
sample file from conf/autoload_configs/ in the freeswitch source directory.
 
-Justin
 
 

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of deniro
Sent: Wednesday, March 02, 2011 2:25 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] fs_cli socket connection error


I typed the 4 commands but still unable to load mod_xml_cdr and mod_xml_curl
 
 
2011-03-02 17:21:32.139672 [ERR] mod_xml_cdr.c:479 Open of xml_cdr.conf
failed

/opt# find . -name xml_cdr.conf
nothing appears
 
here is errors from fs_cli -d 7
 
freeswitch at 75.xxx.xxx.xx@internal <mailto:freeswitch at 75.xxx.xxx.xx@internal>
> load mod_xml_cdr
[DEBUG] libs/esl/src/esl.c:1063 esl_send() SEND
api load mod_xml_cdr
console_execute: true

[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type] =
[api/response]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Length]
= [50]
-ERR [module load file routine returned an error]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type] =
[log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Length]
= [79]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
[3]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel] =
[0]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
[mod_xml_cdr.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
[mod_xml_cdr_load]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
[479]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] = []
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 79
Log-Level: 3
Text-Channel: 0
Log-File: mod_xml_cdr.c
Log-Func: mod_xml_cdr_load
Log-Line: 479
User-Data: _undef_

2011-03-02 17:21:32.139672 [ERR] mod_xml_cdr.c:479 Open of xml_cdr.conf
failed
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type] =
[log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Length]
= [161]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
[2]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel] =
[0]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
[switch_loadable_module.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
[switch_loadable_module_load_file]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
[882]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] = []
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 161
Log-Level: 2
Text-Channel: 0
Log-File: switch_loadable_module.c
Log-Func: switch_loadable_module_load_file
Log-Line: 882
User-Data: _undef_

2011-03-02 17:21:32.139672 [CRIT] switch_loadable_module.c:882 Error Loading
module /opt/freeswitch/mod/mod_xml_cdr.so
**Module load routine returned an error**
 
 
 


 
On Tue, Mar 1, 2011 at 1:11 PM, Anthony Minessale <
anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com> > wrote:


like I said you re-installed with a new version of libcurl but did not
fully rebuild.
So most of the modules that depend on curl were broken torn between
the old and new version it was linked to..

had you executed the 4 commands I posted yesterday it would be fine.



On Tue, Mar 1, 2011 at 11:02 AM, deniro < ayhkor at gmail.com
<mailto:ayhkor at gmail.com> > wrote:
> Thank you all for advice  and direction
> Will re-install from scratch at a later time.
> deniro--
>
> On Tue, Mar 1, 2011 at 4:16 AM, Steven Ayre < steveayre at gmail.com
<mailto:steveayre at gmail.com> > wrote:
>>
>> Ok, you really need to pay attention to any log message that logs with
>> the CRIT (critical) log level - they're really serious errors. You
>> should look at adding the critical="true" attribute to all the tags in
>> modules.conf.xml which you want to be sure are loaded. If a module
>> fails to load without that tag FS will continue to run anyway, with
>> that tag it'll refuse to start so you'll know instantly something is
>> wrong.
>>
>> Most of your modules aren't loading, because they haven't been
>> compiled correctly. Some look like they're missing (no such file or
>> directory) and others haven't been built right (undefined symbols).
>>
>> Here are a few of the important lines from your log:
>>
>> 2011-02-28 18:12:21.910738 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_event_socket.so
>> **/opt/freeswitch/mod/mod_event_socket.so: undefined symbol:
>> switch_event_serialize_json**
>> 2011-02-28 18:12:21.910789 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_xml_curl.so
>> **/opt/freeswitch/mod/mod_xml_curl.so: cannot open shared object file:
>> No such file or directory**
>> 2011-02-28 18:12:21.910813 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_xml_cdr.so
>> **/opt/freeswitch/mod/mod_xml_cdr.so: cannot open shared object file:
>> No such file or directory**
>> 2011-02-28 18:12:21.911581 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_commands.so
>> **/opt/freeswitch/mod/mod_commands.so: undefined symbol:
>> switch_xml_reload**
>> 2011-02-28 18:12:21.911709 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_conference.so
>> **/opt/freeswitch/mod/mod_conference.so: undefined symbol:
>> switch_channel_test_app_flag_key**
>> 2011-02-28 18:12:21.911842 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_dptools.so
>> **/opt/freeswitch/mod/mod_dptools.so: undefined symbol:
>> switch_event_merge**
>> 2011-02-28 18:12:21.912264 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_dialplan_xml.so
>> **/opt/freeswitch/mod/mod_dialplan_xml.so: undefined symbol:
>> switch_xml_std_datetime_check**
>> 2011-02-28 18:12:21.912531 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_g723_1.so
>> **/opt/freeswitch/mod/mod_g723_1.so: cannot open shared object file:
>> No such file or directory**
>> 2011-02-28 18:12:21.912573 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_g729.so
>> **/opt/freeswitch/mod/mod_g729.so: cannot open shared object file: No
>> such file or directory**
>> 2011-02-28 18:12:21.912609 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /opt/freeswitch/mod/mod_amr.so
>> **/opt/freeswitch/mod/mod_amr.so: cannot open shared object file: No
>> such file or directory**
>>
>> Important things to note:
>> mod_event_socket hasn't loaded because of a undefined symbol, it
>> wasn't built correctly
>> mod_xml_curl hasn't loaded because it doesn't exist, it was either
>> never built or never installed
>>
>> Something pretty strange has happened. Did you recompile when trying
>> to add mod_xml_curl?
>>
>> I'd suggest you delete all the FS files, including the Git clone, make
>> a fresh checkout and build it from scratch.
>>
>> -Steve
>>
>>
>>
>> On 28 February 2011 23:18, deniro < ayhkor at gmail.com
<mailto:ayhkor at gmail.com> > wrote:
>> > I put the log in pastebin
>> > freeswitch.log only when starting freeswitch (after stop)
>> >
>> > http://pastebin.freeswitch.org/15502
<http://pastebin.freeswitch.org/15502> 
>> > thx
>> > deniro
>> > On Mon, Feb 28, 2011 at 5:46 PM, Steven Ayre < steveayre at gmail.com
<mailto:steveayre at gmail.com> >
>> > wrote:
>> >>
>> >> You must check the logfile for when FS starts up - if netstat shows
>> >> nothing for port 8021 then either mod_event_socket isn't being loaded
>> >> or you'll see an error when it tries to load.
>> >>
>> >> Nothing from netstat means nothing's listening, so trying to connect
>> >> using fs_cli won't do anything.
>> >>
>> >> -Steve
>> >>
>> >>
>> >> On 28 February 2011 22:40, deniro < ayhkor at gmail.com
<mailto:ayhkor at gmail.com> > wrote:
>> >> > Checked the log
>> >> > Nothing is logged when
>> >> > fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
>> >> > is issued
>> >> > /opt/freeswitch/log# tail -f freeswitch.log
>> >> >
>> >> > ls -l freeswitch.log
>> >> > -rwxrwxrwx 1 freeswitch daemon 33018 2011-02-28 17:39 freeswitch.log
>> >> > thx
>> >> >
>> >> >
>> >> >
>> >> > On Mon, Feb 28, 2011 at 3:59 PM, Peter Olsson
>> >> > < peter.olsson at visionutveckling.se
<mailto:peter.olsson at visionutveckling.se> > wrote:
>> >> >>
>> >> >> Check freeswitch.log, it probably reports some problem when loading
>> >> >> the
>> >> >> mod_event_socket module.
>> >> >>
>> >> >> /Peter
>> >> >> ________________________________________
>> >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
<mailto:freeswitch-users-bounces at lists.freeswitch.org> 
>> >> >> [ freeswitch-users-bounces at lists.freeswitch.org
<mailto:freeswitch-users-bounces at lists.freeswitch.org> ] f&#246;r deniro
>> >> >> [ ayhkor at gmail.com <mailto:ayhkor at gmail.com> ]
>> >> >> Skickat: den 28 februari 2011 21:42
>> >> >> Till: FreeSWITCH Users Help
>> >> >> ?mne: Re: [Freeswitch-users] fs_cli socket connection error
>> >> >>
>> >> >> Steve
>> >> >> Thanks for the reply
>> >> >> netstat -anlp | grep 8021  is blank (nothing showing up)
>> >> >>
>> >> >> /opt/freeswitch/bin/fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
>> >> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
>> >> >> Configuration
>> >> >> file is /root/.fs_cli_conf.
>> >> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
>> >> >> Configuration
>> >> >> file is /etc/fs_cli.conf.
>> >> >> [DEBUG] libs/esl/fs_cli.c:1159 main() profile default does not
exist
>> >> >> using
>> >> >> builtin profile
>> >> >> [DEBUG] libs/esl/fs_cli.c:1188 main() Using profile internal
>> >> >> [75.xxx.xxx.xxx]
>> >> >> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
>> >> >> Connection
>> >> >> Error]
>> >> >> yes mod_event_socket is after  mod_xml_curl but I changed the order
>> >> >> in
>> >> >>  modules.conf.xml
>> >> >> still getting above (restarted freeswitch)
>> >> >>
>> >> >> thx
>> >> >> deniro--
>> >> >>
>> >> >>
>> >> >>
>> >> >> On Mon, Feb 28, 2011 at 3:35 AM, Steven Ayre
>> >> >> < steveayre at gmail.com <mailto:steveayre at gmail.com> <mailto:
steveayre at gmail.com <mailto:steveayre at gmail.com> >> wrote:
>> >> >> Enable debug logging and you should see an error that'll tell you
>> >> >> more.
>> >> >>
>> >> >> Is mod_event_socket loading before or after mod_xml_curl? Chances
>> >> >> are
>> >> >> mod_xml_curl is loading first, so it's trying to read
>> >> >> event_socket.conf.xml and/or acl.conf.xml through xml_curl and
>> >> >> either
>> >> >> getting a different config to your previous local copy or the ACLs
>> >> >> are
>> >> >> different.
>> >> >>
>> >> >> What does "netstat -anlp | grep :8021" from the linux cli show you?
>> >> >> Does it show that freeswitch is actually listening on the port? If
>> >> >> it
>> >> >> is it's probably an ACL problem, if it isn't then it's probably a
>> >> >> problem with event_socket.conf.xml
>> >> >>
>> >> >> -Steve
>> >> >>
>> >> >> On 28 February 2011 00:53, deniro
>> >> >> < ayhkor at gmail.com <mailto:ayhkor at gmail.com> <mailto:
ayhkor at gmail.com <mailto:ayhkor at gmail.com> >> wrote:
>> >> >> > What would be possible reasons for this and how to resolve?
>> >> >> > running fs 106 on ubuntu 10.04 server
>> >> >> >
>> >> >> > /opt/freeswitch/bin/fs_cli -H xxx.xxx.xxx.xxx -P 8021
>> >> >> > [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
>> >> >> > Connection
>> >> >> > Error]
>> >> >> > was working fine before I installed  mod_xml_cdr
>> >> >> > configure --prefix=/opt/freeswitch --without-libcurl
>> >> >> > make mod_xml_cdr-install
>> >> >> > (no errors)
>> >> >> >
>> >> >> > in modules.conf.xml
>> >> >> > <load module="mod_event_socket"/>
>> >> >> >
>> >> >> > is there a packaged version of mod_xml_cdr (for fs 1.0.6)?
>> >> >> >
>> >> >> > thx
>> >> >> > deniro--
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
<mailto:FreeSWITCH-users at lists.freeswitch.org> <mailto:
FreeSWITCH-users at lists.freeswitch.org
<mailto:FreeSWITCH-users at lists.freeswitch.org> >
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> 
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
<http://lists.freeswitch.org/mailman/options/freeswitch-users> 
>> >> >> > http://www.freeswitch.org <http://www.freeswitch.org/> <
http://www.freeswitch.org/ <http://www.freeswitch.org/> >
>> >> >> >
>> >> >> >
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >>
>> >> >>
>> >> >>
>> >> >> FreeSWITCH-users at lists.freeswitch.org
<mailto:FreeSWITCH-users at lists.freeswitch.org> <mailto:
FreeSWITCH-users at lists.freeswitch.org
<mailto:FreeSWITCH-users at lists.freeswitch.org> >
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> 
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
<http://lists.freeswitch.org/mailman/options/freeswitch-users> 
>> >> >> http://www.freeswitch.org <http://www.freeswitch.org/> <
http://www.freeswitch.org/ <http://www.freeswitch.org/> >
>> >> >>
>> >> >> !DSPAM:4d6c0a1132761029518849!
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
<mailto:FreeSWITCH-users at lists.freeswitch.org> 
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> 
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
<http://lists.freeswitch.org/mailman/options/freeswitch-users> 
>> >> >> http://www.freeswitch.org <http://www.freeswitch.org/> 
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
<mailto:FreeSWITCH-users at lists.freeswitch.org> 
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> 
>> >> >
>> >> > UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
<http://lists.freeswitch.org/mailman/options/freeswitch-users> 
>> >> > http://www.freeswitch.org <http://www.freeswitch.org/> 
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
<mailto:FreeSWITCH-users at lists.freeswitch.org> 
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> 
>> >>
>> >> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
<http://lists.freeswitch.org/mailman/options/freeswitch-users> 
>> >> http://www.freeswitch.org <http://www.freeswitch.org/> 
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
<mailto:FreeSWITCH-users at lists.freeswitch.org> 
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> 
>> > UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
<http://lists.freeswitch.org/mailman/options/freeswitch-users> 
>> > http://www.freeswitch.org <http://www.freeswitch.org/> 
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
<mailto:FreeSWITCH-users at lists.freeswitch.org> 
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> 
>> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users
<http://lists.freeswitch.org/mailman/options/freeswitch-users> 
>> http://www.freeswitch.org <http://www.freeswitch.org/> 
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
<mailto:FreeSWITCH-users at lists.freeswitch.org> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> 
> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users
<http://lists.freeswitch.org/mailman/options/freeswitch-users> 
> http://www.freeswitch.org <http://www.freeswitch.org/> 
>
>




--

Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net <http://irc.freenode.net/>  #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

_______________________________________________

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
<mailto:FreeSWITCH-users at lists.freeswitch.org> 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users
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From eric at bmcrministries.org  Thu Mar  3 01:49:45 2011
From: eric at bmcrministries.org (Eric Michel)
Date: Wed, 2 Mar 2011 15:49:45 -0700
Subject: [Freeswitch-users] Segfault on startup when mod_freetdm is enabled.
Message-ID: <AANLkTinFixtmr_A29JFm1TDyTUYs4QeBC8gJFg5OncEw@mail.gmail.com>

I updated to the latest git last night, and started having issues.  I
assumed I'd screwed up and (not being very adept at unistalling things from
source) decided to do a clean install today (including the OS).

When I enable mod_freetdm in modules.conf.xml, freeswitch segfaults on
startup.  If I go back and disable mod_freetdm it starts fine.  As this is a
clean install the only thing I've changed is modules.conf.xml.  Is any one
else having this problem?  Do I need to file a bug report or am I just
missing something really simple?

I'm running FreeSWITCH Version 1.0.head(git-64806d2 2011-03-02 18-23-19
+0100).
I installed the latest non-beta version of Wanpipe which is 3.5.18, and used
"make freetdm" when I compiled it.
I've got the Sagnoma A200 card.
The OS is CentOS 5.5 64bit, AMD Athalon 7850 Dual-core processor, 2Gb RAM, 2
500Gb HD in a software RAID1.

Before I compiled freeswitch I enabled "../../libs/freetdm/mod_freetdm" and
"asr_tts/mod_flite" in modules.conf.
I've run wancfg_fs.




Eric
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From mattzerah+freeswitch at gmail.com  Thu Mar  3 08:13:41 2011
From: mattzerah+freeswitch at gmail.com (Matt Paine)
Date: Thu, 3 Mar 2011 15:13:41 +1000
Subject: [Freeswitch-users] FreeTDM and Libpri with a Digium TE122 ... some
	calls not being answered...
Message-ID: <AANLkTimb3rLWvfZ=dTBM9XZeLa=OenJoz=zjRjdoD6Ym@mail.gmail.com>

Hi Guys.

Current setup is a FS box with a Digium TE122 installed, with DAHDI kernel
drivers and using FreeTDM with a libpri span...

== autoload_configs/freetdm.conf.xml ==

<configuration name="freetdm.conf" description="FreeTDM Configuration">
    <settings>
        <param name="debug" value="0"/>
        <param name="sip_headers" value="true"/>
    </settings>
    <libpri_spans>
        <span name="ISDN">
            <param name="mode" value="user"/>
            <param name="dialect" value="q931"/>
             <param name="dialplan" value="XML"/>
            <param name="context" value="public"/>
        </span>
    </libpri_spans>
</configuration>

== freetdm.conf ==

[general]
cpu_monitor => yes
cpu_monitoring_interval => 1000
cpu_set_alarm_threshold => 80
cpu_reset_alarm_threshold => 70
cpu_alarm_action => warn

[span zt ISDN]
trunk_type => E1
b-channel => 1-15
d-channel => 16
b-channel => 17-21

== zt.conf ==
[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750
echo_cancel_level => 128
rxgain => 0.0
txgain => 0.0

== /etc/dahdi/system.conf ==
span=1,1,0,ccs,hdb3,crc4
# termtype: te
bchan=1-15,17-31
dchan=16
echocanceller=mg2,1-15,17-31
# Global data
loadzone    = au
defaultzone = au


=========
And finally some logs from two calls, a successful one, and a failed one...
 http://pastebin.freeswitch.org/15533


If anyone could offer any suggestions as to what I can do to answer every
incoming call into the box that would very appreciated. Of course if I need
to provide more information I am willing to, and any suggestions for
settings that need changing to test out will also be great.

Thank you in advance
Matt
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From kbdfck at gmail.com  Thu Mar  3 11:58:27 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Thu, 3 Mar 2011 11:58:27 +0300
Subject: [Freeswitch-users] xml_curl and domain aliases
Message-ID: <AANLkTimS2ZbzaVaVGzgraGjNF5MZd=909VBD3dg7yNSP@mail.gmail.com>

Hi All

I'm trying to setup xml_curl-based directory to provide registration and
bridging information from database.

The thing I can't still reliably handle is domain aliases. I have some
domain like 'access.freeswitch.test', but I need to allow users to register
via ip address, for example, 1.1.1.1. My scripts answer on REGISTER curl
request with directory entry where domain is set to domain user trying to
register at, and this works.

But then I need to dial that user. I'm trying to use
user/${userpart}%{domainpart} syntax, but I don't know exactly domain user
was registered at. How to tell freeswitch to take in account domain aliases
when looking up for sofia contact?
Or, instead, can I make freeswitch to store user in registration database
with some primary domain, which I can then specify to bridge(user/) command?

Thanks in advance!

-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From peter.olsson at visionutveckling.se  Thu Mar  3 12:00:26 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 3 Mar 2011 10:00:26 +0100
Subject: [Freeswitch-users] Conference - control actions problem
In-Reply-To: <AANLkTikPa1E9o3-NgWBjd7P8HWfr1vRxAnh+AH0UpqW1@mail.gmail.com>
References: <AANLkTimq9dYbq8ztbQ5dJmY2Y+2mK5mKzM78UirEzzYn@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2BBAF1D@cooper>
	<AANLkTikPa1E9o3-NgWBjd7P8HWfr1vRxAnh+AH0UpqW1@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2BBB2F5@cooper>

Hmm - I'm not really sure what your problem is. You receive DTMF, but nothing happens anyway, or what?

Latest git HEAD is stable, so you should be ok.

Could you describe exactly what the problem is after sending a DTMF?

/Peter


-----Ursprungligt meddelande-----
Fr?n: Maciej Bylica [mailto:mbsip at gazeta.pl] 
Skickat: den 2 mars 2011 17:15
Till: FreeSWITCH Users Help
Kopia: Peter Olsson
?mne: Re: [Freeswitch-users] Conference - control actions problem

Thx Peter for prompt reply.

I've just made one more configuration change. There are no lua script
fired anymore, just pure conference application like following:
       <action application="conference" data="1000 at fsconf"/>

The effect is the same (after pressing * or # key just one time).
2011-03-02 16:55:32.677018 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:520
or later
2011-03-02 16:58:13.645012 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF *:620

As you can see there are DTMF detected on Freeswitch interpreted as
'RTP RECV DTMF'
Is git-06988e1 rel stable one?


Regards,
Maciej

2011/3/2 Peter Olsson <peter.olsson at visionutveckling.se>:
> If you exit the conference (#) the user will still be in your lua script, so you would have to do whatever you want it to (hangup maybe). I think the recommended way is to transfer away the call to the conference, that way the Lua scripts won't need to do any more processing.
>
> Also check this Javascript example on the wiki: http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR
>
> About DTMF's - if you don't see them in your log, then FS havn't received any digits - you would have to do a packet capture and check the RTP packets.
>
> /Peter
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Maciej Bylica
> Skickat: den 2 mars 2011 12:31
> Till: FreeSWITCH Users Help
> ?mne: [Freeswitch-users] Conference - control actions problem
>
> Hello,
>
> I am running FreeSWITCH Version 1.0.head (git-06988e1 2011-02-20
> 15-10-02 -0600) and have some problem with conference module and
> respective control actions digits.
> To give you an overview of how it is configured, there is lua script
> fired in public.xml.
> The script checks pin code, conf room and at the end run
> "session:execute("conference", string.format("%s at fsconf", conf_num))".
> Now coming back to the problem, there are few scenations i would like to cover.
>
> 1. When two or more people are in conference and one of them is trying
> to push # button (RFC2833) then that person is leaving the conference
> and is hanged up (not released, no BYE). The person who left the
> conference cannot join again or perform other actions without making
> the same phone call again.
>
> 2011-03-02 11:42:08.863139 [DEBUG] mod_conference.c:2414 Setup timer
> soft success interval: 20 ?samples: 160
> 2011-03-02 11:42:12.283051 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:480
> 2011-03-02 11:42:12.323050 [DEBUG] mod_conference.c:2659 Channel
> leaving conference, cause: NONE
>
> 2. The same as above, but * digit is entered. The person is muted and
> then unmuted. Next try with * fires muted but this is the point where
> the problem manifest itself.
> The procedure of unmuting UA fails. Below you may find relevant logs.
> The UA is pressing * but without success (no RTP RECV DTMF *).
>
> freeswitch at internal> 2011-03-02 11:51:13.423651 [DEBUG]
> switch_rtp.c:3237 RTP RECV DTMF *:504
> 2011-03-02 11:51:13.443650 [DEBUG] mod_conference.c:3125 Queueing file
> '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav'
> for play
> 2011-03-02 11:51:18.163532 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF *:544
> 2011-03-02 11:51:18.183531 [DEBUG] mod_conference.c:3125 Queueing file
> '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-unmuted.wav'
> for play
> 2011-03-02 11:51:23.763391 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF *:504
> 2011-03-02 11:51:23.783390 [DEBUG] mod_conference.c:3125 Queueing file
> '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav'
> for play
>
> 3. The same as above but with 0 digit
>
> 2011-03-02 12:04:21.064862 [DEBUG] mod_conference.c:2414 Setup timer
> soft success interval: 20 ?samples: 160
> 2011-03-02 12:04:22.863736 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 0:880
> 2011-03-02 12:04:22.883736 [DEBUG] mod_conference.c:3125 Queueing file
> '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav'
> for play
>
> then i cannot get back to the conference
>
>
> 4. The as above but with 6 digit
>
> 2011-03-02 12:15:03.244609 [DEBUG] mod_conference.c:2414 Setup timer
> soft success interval: 20 ?samples: 160
> 2011-03-02 12:15:21.602919 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 6:504
> 2011-03-02 12:15:22.602893 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 6:800
> 2011-03-02 12:15:24.263851 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 6:864
>
>
> I have disabled all control digits, but the problem still persist. No
> action is fired but is looks like UA is being muted (micro is off, but
> i can hear the other person).
> 2011-03-02 12:24:05.322935 [DEBUG] mod_conference.c:2414 Setup timer
> soft success interval: 20 ?samples: 160
> 2011-03-02 12:24:08.663851 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:480
> 2011-03-02 12:24:11.223787 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:504
> 2011-03-02 12:24:13.703725 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:480
> 2011-03-02 12:24:16.043666 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:520
>
>
> Has anybody similar experience or know where the problem may be located.
>
> Regards,
> Maciej
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> 
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

!DSPAM:4d6e6c8a32761758178328!



From oa at estation.dk  Thu Mar  3 12:03:40 2011
From: oa at estation.dk (=?ISO-8859-1?Q?=D8yvind_Albrigtsen?=)
Date: Thu, 03 Mar 2011 10:03:40 +0100
Subject: [Freeswitch-users] Snom and Aastra one-way audio after picking up
 call that has been on hold (NAT)
Message-ID: <4D6F596C.7010403@estation.dk>

Hi,

We have some Snom and Aastra phones that work well for normal calls, but 
when we call them from another location and then put the call on hold 
and pick it up again it only has one-way audio.

Does anyone know how to solve this issue? It works fine when we call 
between 2 phones behind the same NAT, so I guess they send the local IP, 
but I havent found an option to make them behave otherwise.

The calls get in like this:
PSTN-gw -> FS -> NAT -> Snom/Aastra

It works fine when I call out from the phones and set the call on hold 
(on the Snom/Aastra phones), though.

This is on FreeSWITCH Version 1.0.head (git-cb6f1ed 2011-02-22 20-25-16 
-0500)


Regards
Oyvind


From mbsip at gazeta.pl  Thu Mar  3 13:07:02 2011
From: mbsip at gazeta.pl (Maciej Bylica)
Date: Thu, 3 Mar 2011 11:07:02 +0100
Subject: [Freeswitch-users] Conference - control actions problem
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B2BBB2F5@cooper>
References: <AANLkTimq9dYbq8ztbQ5dJmY2Y+2mK5mKzM78UirEzzYn@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2BBAF1D@cooper>
	<AANLkTikPa1E9o3-NgWBjd7P8HWfr1vRxAnh+AH0UpqW1@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2BBB2F5@cooper>
Message-ID: <AANLkTim=VDq1ztB7Bn=_E0eq_8KeRnZceLapmT8jz2DS@mail.gmail.com>

Hi,

As stated in first email control action digits does not behave as they
should. Then it turned out that I have the same issue when these
actions are disabled.
Once again, lets assume that UA1 and UA2 are in conf room. UA1 is
pressing any digit (let say '5') and as a aftereffect it looks like
UA1 has microphone muted - so cannot be heard by UA2. Furthermore no
consecutive dtmfs coming from UA1 are collected.
The only wayout is to hang up the phone and dial again.

Thanks,
Maciej

> Hmm - I'm not really sure what your problem is. You receive DTMF, but nothing happens anyway, or what?
>
> Latest git HEAD is stable, so you should be ok.
>
> Could you describe exactly what the problem is after sending a DTMF?
>
> /Peter
>
>
> -----Ursprungligt meddelande-----
> Fr?n: Maciej Bylica [mailto:mbsip at gazeta.pl]
> Skickat: den 2 mars 2011 17:15
> Till: FreeSWITCH Users Help
> Kopia: Peter Olsson
> ?mne: Re: [Freeswitch-users] Conference - control actions problem
>
> Thx Peter for prompt reply.
>
> I've just made one more configuration change. There are no lua script
> fired anymore, just pure conference application like following:
> ? ? ? <action application="conference" data="1000 at fsconf"/>
>
> The effect is the same (after pressing * or # key just one time).
> 2011-03-02 16:55:32.677018 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:520
> or later
> 2011-03-02 16:58:13.645012 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF *:620
>
> As you can see there are DTMF detected on Freeswitch interpreted as
> 'RTP RECV DTMF'
> Is git-06988e1 rel stable one?
>
>
> Regards,
> Maciej
>
> 2011/3/2 Peter Olsson <peter.olsson at visionutveckling.se>:
>> If you exit the conference (#) the user will still be in your lua script, so you would have to do whatever you want it to (hangup maybe). I think the recommended way is to transfer away the call to the conference, that way the Lua scripts won't need to do any more processing.
>>
>> Also check this Javascript example on the wiki: http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR
>>
>> About DTMF's - if you don't see them in your log, then FS havn't received any digits - you would have to do a packet capture and check the RTP packets.
>>
>> /Peter
>>
>> -----Ursprungligt meddelande-----
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Maciej Bylica
>> Skickat: den 2 mars 2011 12:31
>> Till: FreeSWITCH Users Help
>> ?mne: [Freeswitch-users] Conference - control actions problem
>>
>> Hello,
>>
>> I am running FreeSWITCH Version 1.0.head (git-06988e1 2011-02-20
>> 15-10-02 -0600) and have some problem with conference module and
>> respective control actions digits.
>> To give you an overview of how it is configured, there is lua script
>> fired in public.xml.
>> The script checks pin code, conf room and at the end run
>> "session:execute("conference", string.format("%s at fsconf", conf_num))".
>> Now coming back to the problem, there are few scenations i would like to cover.
>>
>> 1. When two or more people are in conference and one of them is trying
>> to push # button (RFC2833) then that person is leaving the conference
>> and is hanged up (not released, no BYE). The person who left the
>> conference cannot join again or perform other actions without making
>> the same phone call again.
>>
>> 2011-03-02 11:42:08.863139 [DEBUG] mod_conference.c:2414 Setup timer
>> soft success interval: 20 ?samples: 160
>> 2011-03-02 11:42:12.283051 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:480
>> 2011-03-02 11:42:12.323050 [DEBUG] mod_conference.c:2659 Channel
>> leaving conference, cause: NONE
>>
>> 2. The same as above, but * digit is entered. The person is muted and
>> then unmuted. Next try with * fires muted but this is the point where
>> the problem manifest itself.
>> The procedure of unmuting UA fails. Below you may find relevant logs.
>> The UA is pressing * but without success (no RTP RECV DTMF *).
>>
>> freeswitch at internal> 2011-03-02 11:51:13.423651 [DEBUG]
>> switch_rtp.c:3237 RTP RECV DTMF *:504
>> 2011-03-02 11:51:13.443650 [DEBUG] mod_conference.c:3125 Queueing file
>> '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav'
>> for play
>> 2011-03-02 11:51:18.163532 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF *:544
>> 2011-03-02 11:51:18.183531 [DEBUG] mod_conference.c:3125 Queueing file
>> '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-unmuted.wav'
>> for play
>> 2011-03-02 11:51:23.763391 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF *:504
>> 2011-03-02 11:51:23.783390 [DEBUG] mod_conference.c:3125 Queueing file
>> '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav'
>> for play
>>
>> 3. The same as above but with 0 digit
>>
>> 2011-03-02 12:04:21.064862 [DEBUG] mod_conference.c:2414 Setup timer
>> soft success interval: 20 ?samples: 160
>> 2011-03-02 12:04:22.863736 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 0:880
>> 2011-03-02 12:04:22.883736 [DEBUG] mod_conference.c:3125 Queueing file
>> '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav'
>> for play
>>
>> then i cannot get back to the conference
>>
>>
>> 4. The as above but with 6 digit
>>
>> 2011-03-02 12:15:03.244609 [DEBUG] mod_conference.c:2414 Setup timer
>> soft success interval: 20 ?samples: 160
>> 2011-03-02 12:15:21.602919 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 6:504
>> 2011-03-02 12:15:22.602893 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 6:800
>> 2011-03-02 12:15:24.263851 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 6:864
>>
>>
>> I have disabled all control digits, but the problem still persist. No
>> action is fired but is looks like UA is being muted (micro is off, but
>> i can hear the other person).
>> 2011-03-02 12:24:05.322935 [DEBUG] mod_conference.c:2414 Setup timer
>> soft success interval: 20 ?samples: 160
>> 2011-03-02 12:24:08.663851 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:480
>> 2011-03-02 12:24:11.223787 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:504
>> 2011-03-02 12:24:13.703725 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:480
>> 2011-03-02 12:24:16.043666 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:520
>>
>>
>> Has anybody similar experience or know where the problem may be located.
>>
>> Regards,
>> Maciej
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> !DSPAM:4d6e6c8a32761758178328!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From julf at julf.com  Thu Mar  3 13:30:04 2011
From: julf at julf.com (Johan Helsingius)
Date: Thu, 03 Mar 2011 11:30:04 +0100
Subject: [Freeswitch-users] checking voicemail without ID?
In-Reply-To: <AANLkTimqvK-f-b43Pm3Ws6cEm7ZcdD2cv78OPv7oxycN@mail.gmail.com>
References: <4D6E555F.4000003@julf.com>	<649CD6B6-AB32-41B0-8C9D-2BB5F3DEA9FD@freeswitch.org>	<4D6E63FE.1020302@julf.com>	<518984EC-1B52-4631-B3D5-2DB6776255C5@freeswitch.org>	<4D6E6921.9000604@julf.com>
	<AANLkTimqvK-f-b43Pm3Ws6cEm7ZcdD2cv78OPv7oxycN@mail.gmail.com>
Message-ID: <4D6F6DAC.9090505@julf.com>

> Try it with "auth"

Doesn't seem to make any difference.

	Julf


From julf at julf.com  Thu Mar  3 13:31:16 2011
From: julf at julf.com (Johan Helsingius)
Date: Thu, 03 Mar 2011 11:31:16 +0100
Subject: [Freeswitch-users] checking voicemail without ID?
In-Reply-To: <8A592B05-0D5F-4EFA-87A0-730F6770B58E@mgtech.com>
References: <4D6E555F.4000003@julf.com>
	<8A592B05-0D5F-4EFA-87A0-730F6770B58E@mgtech.com>
Message-ID: <4D6F6DF4.7070801@julf.com>

Mario,

> I do this so they can just press the envelope key, in my extension definition
> for the user id, I have the following in the variable section:
> 
>       <variable name="mailbox" value="100"/>
> 
> I also have:
>       <variable name="voicemail_domain"value="mgtech.com <http://mgtech.com>"/>
> 
> I also have some phone that dont need to even put a password in.

Precisely what I am trying to do. So how does your dialplan
look for checking the mailbox?

	Julf


From julf at julf.com  Thu Mar  3 13:32:52 2011
From: julf at julf.com (Johan Helsingius)
Date: Thu, 03 Mar 2011 11:32:52 +0100
Subject: [Freeswitch-users] checking voicemail without ID?
In-Reply-To: <AANLkTi=gj4QzuR2AAdMphwxpzOfSu9B1Fy4HHWyA8-YN@mail.gmail.com>
References: <4D6E555F.4000003@julf.com>	<649CD6B6-AB32-41B0-8C9D-2BB5F3DEA9FD@freeswitch.org>	<4D6E63FE.1020302@julf.com>	<518984EC-1B52-4631-B3D5-2DB6776255C5@freeswitch.org>	<4D6E6921.9000604@julf.com>
	<AANLkTi=gj4QzuR2AAdMphwxpzOfSu9B1Fy4HHWyA8-YN@mail.gmail.com>
Message-ID: <4D6F6E54.1000303@julf.com>

Michael,

> Don't you need to execute this one inline? 
> 
>            <action application="set" data="voicemail_authorized=true"/>
> 
> Should be
>  <action application="set" data="voicemail_authorized=true" inline="true"/>

I would have thought that the voicemail app only checks the
authorization at execute time, not when hunting.

> Try that and let us know what happens.

Unfortunately doesn't change the situation.

	Julf


From peter.olsson at visionutveckling.se  Thu Mar  3 13:39:51 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 3 Mar 2011 11:39:51 +0100
Subject: [Freeswitch-users] Conference - control actions problem
In-Reply-To: <AANLkTim=VDq1ztB7Bn=_E0eq_8KeRnZceLapmT8jz2DS@mail.gmail.com>
References: <AANLkTimq9dYbq8ztbQ5dJmY2Y+2mK5mKzM78UirEzzYn@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2BBAF1D@cooper>
	<AANLkTikPa1E9o3-NgWBjd7P8HWfr1vRxAnh+AH0UpqW1@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2BBB2F5@cooper>
	<AANLkTim=VDq1ztB7Bn=_E0eq_8KeRnZceLapmT8jz2DS@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2BBB395@cooper>

Sounds really strange. This stuff has worked forever, and I'm not able to reproduce it in any way. Could you pastebin relevant parts of the config, conference.conf.xml and the dialplan + plus logs and examples for what goes wrong.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Maciej Bylica
Skickat: den 3 mars 2011 11:07
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Conference - control actions problem

Hi,

As stated in first email control action digits does not behave as they
should. Then it turned out that I have the same issue when these
actions are disabled.
Once again, lets assume that UA1 and UA2 are in conf room. UA1 is
pressing any digit (let say '5') and as a aftereffect it looks like
UA1 has microphone muted - so cannot be heard by UA2. Furthermore no
consecutive dtmfs coming from UA1 are collected.
The only wayout is to hang up the phone and dial again.

Thanks,
Maciej

> Hmm - I'm not really sure what your problem is. You receive DTMF, but nothing happens anyway, or what?
>
> Latest git HEAD is stable, so you should be ok.
>
> Could you describe exactly what the problem is after sending a DTMF?
>
> /Peter
>
>
> -----Ursprungligt meddelande-----
> Fr?n: Maciej Bylica [mailto:mbsip at gazeta.pl]
> Skickat: den 2 mars 2011 17:15
> Till: FreeSWITCH Users Help
> Kopia: Peter Olsson
> ?mne: Re: [Freeswitch-users] Conference - control actions problem
>
> Thx Peter for prompt reply.
>
> I've just made one more configuration change. There are no lua script
> fired anymore, just pure conference application like following:
> ? ? ? <action application="conference" data="1000 at fsconf"/>
>
> The effect is the same (after pressing * or # key just one time).
> 2011-03-02 16:55:32.677018 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:520
> or later
> 2011-03-02 16:58:13.645012 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF *:620
>
> As you can see there are DTMF detected on Freeswitch interpreted as
> 'RTP RECV DTMF'
> Is git-06988e1 rel stable one?
>
>
> Regards,
> Maciej
>
> 2011/3/2 Peter Olsson <peter.olsson at visionutveckling.se>:
>> If you exit the conference (#) the user will still be in your lua script, so you would have to do whatever you want it to (hangup maybe). I think the recommended way is to transfer away the call to the conference, that way the Lua scripts won't need to do any more processing.
>>
>> Also check this Javascript example on the wiki: http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR
>>
>> About DTMF's - if you don't see them in your log, then FS havn't received any digits - you would have to do a packet capture and check the RTP packets.
>>
>> /Peter
>>
>> -----Ursprungligt meddelande-----
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Maciej Bylica
>> Skickat: den 2 mars 2011 12:31
>> Till: FreeSWITCH Users Help
>> ?mne: [Freeswitch-users] Conference - control actions problem
>>
>> Hello,
>>
>> I am running FreeSWITCH Version 1.0.head (git-06988e1 2011-02-20
>> 15-10-02 -0600) and have some problem with conference module and
>> respective control actions digits.
>> To give you an overview of how it is configured, there is lua script
>> fired in public.xml.
>> The script checks pin code, conf room and at the end run
>> "session:execute("conference", string.format("%s at fsconf", conf_num))".
>> Now coming back to the problem, there are few scenations i would like to cover.
>>
>> 1. When two or more people are in conference and one of them is trying
>> to push # button (RFC2833) then that person is leaving the conference
>> and is hanged up (not released, no BYE). The person who left the
>> conference cannot join again or perform other actions without making
>> the same phone call again.
>>
>> 2011-03-02 11:42:08.863139 [DEBUG] mod_conference.c:2414 Setup timer
>> soft success interval: 20 ?samples: 160
>> 2011-03-02 11:42:12.283051 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:480
>> 2011-03-02 11:42:12.323050 [DEBUG] mod_conference.c:2659 Channel
>> leaving conference, cause: NONE
>>
>> 2. The same as above, but * digit is entered. The person is muted and
>> then unmuted. Next try with * fires muted but this is the point where
>> the problem manifest itself.
>> The procedure of unmuting UA fails. Below you may find relevant logs.
>> The UA is pressing * but without success (no RTP RECV DTMF *).
>>
>> freeswitch at internal> 2011-03-02 11:51:13.423651 [DEBUG]
>> switch_rtp.c:3237 RTP RECV DTMF *:504
>> 2011-03-02 11:51:13.443650 [DEBUG] mod_conference.c:3125 Queueing file
>> '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav'
>> for play
>> 2011-03-02 11:51:18.163532 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF *:544
>> 2011-03-02 11:51:18.183531 [DEBUG] mod_conference.c:3125 Queueing file
>> '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-unmuted.wav'
>> for play
>> 2011-03-02 11:51:23.763391 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF *:504
>> 2011-03-02 11:51:23.783390 [DEBUG] mod_conference.c:3125 Queueing file
>> '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav'
>> for play
>>
>> 3. The same as above but with 0 digit
>>
>> 2011-03-02 12:04:21.064862 [DEBUG] mod_conference.c:2414 Setup timer
>> soft success interval: 20 ?samples: 160
>> 2011-03-02 12:04:22.863736 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 0:880
>> 2011-03-02 12:04:22.883736 [DEBUG] mod_conference.c:3125 Queueing file
>> '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav'
>> for play
>>
>> then i cannot get back to the conference
>>
>>
>> 4. The as above but with 6 digit
>>
>> 2011-03-02 12:15:03.244609 [DEBUG] mod_conference.c:2414 Setup timer
>> soft success interval: 20 ?samples: 160
>> 2011-03-02 12:15:21.602919 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 6:504
>> 2011-03-02 12:15:22.602893 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 6:800
>> 2011-03-02 12:15:24.263851 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 6:864
>>
>>
>> I have disabled all control digits, but the problem still persist. No
>> action is fired but is looks like UA is being muted (micro is off, but
>> i can hear the other person).
>> 2011-03-02 12:24:05.322935 [DEBUG] mod_conference.c:2414 Setup timer
>> soft success interval: 20 ?samples: 160
>> 2011-03-02 12:24:08.663851 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:480
>> 2011-03-02 12:24:11.223787 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:504
>> 2011-03-02 12:24:13.703725 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:480
>> 2011-03-02 12:24:16.043666 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:520
>>
>>
>> Has anybody similar experience or know where the problem may be located.
>>
>> Regards,
>> Maciej
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> 
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4d6f6a0732761929213956!



From peter.olsson at visionutveckling.se  Thu Mar  3 13:42:11 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 3 Mar 2011 11:42:11 +0100
Subject: [Freeswitch-users] checking voicemail without ID?
In-Reply-To: <4D6F6E54.1000303@julf.com>
References: <4D6E555F.4000003@julf.com>
	<649CD6B6-AB32-41B0-8C9D-2BB5F3DEA9FD@freeswitch.org>
	<4D6E63FE.1020302@julf.com>
	<518984EC-1B52-4631-B3D5-2DB6776255C5@freeswitch.org>
	<4D6E6921.9000604@julf.com>
	<AANLkTi=gj4QzuR2AAdMphwxpzOfSu9B1Fy4HHWyA8-YN@mail.gmail.com>
	<4D6F6E54.1000303@julf.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2BBB396@cooper>

Just a control question - you do use "reloadxml" after you changes right? :)

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Johan Helsingius
Skickat: den 3 mars 2011 11:33
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] checking voicemail without ID?

Michael,

> Don't you need to execute this one inline? 
> 
>            <action application="set" data="voicemail_authorized=true"/>
> 
> Should be
>  <action application="set" data="voicemail_authorized=true" inline="true"/>

I would have thought that the voicemail app only checks the
authorization at execute time, not when hunting.

> Try that and let us know what happens.

Unfortunately doesn't change the situation.

	Julf

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

!DSPAM:4d6f6f0032763122217987!



From julf at julf.com  Thu Mar  3 14:03:58 2011
From: julf at julf.com (Johan Helsingius)
Date: Thu, 03 Mar 2011 12:03:58 +0100
Subject: [Freeswitch-users] checking voicemail without ID?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B2BBB396@cooper>
References: <4D6E555F.4000003@julf.com>	<649CD6B6-AB32-41B0-8C9D-2BB5F3DEA9FD@freeswitch.org>	<4D6E63FE.1020302@julf.com>	<518984EC-1B52-4631-B3D5-2DB6776255C5@freeswitch.org>	<4D6E6921.9000604@julf.com>	<AANLkTi=gj4QzuR2AAdMphwxpzOfSu9B1Fy4HHWyA8-YN@mail.gmail.com>	<4D6F6E54.1000303@julf.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2BBB396@cooper>
Message-ID: <4D6F759E.8040401@julf.com>

> Just a control question - you do use "reloadxml" after you changes right? :)

Well, just to be sure, I restart freeswitch completely after every change.

	Julf


From kawarod at laposte.net  Thu Mar  3 14:55:47 2011
From: kawarod at laposte.net (kawarod)
Date: Thu, 3 Mar 2011 15:55:47 +0400
Subject: [Freeswitch-users] Play a file based on SIP error code
In-Reply-To: <AANLkTikdtd41i=gmuEyVX1NFVaThJ=8AOpou=+ya7Xgt@mail.gmail.com>
References: <0AC82A20-57D9-47E6-AEA7-A2EE53D661F6@laposte.net>
	<AANLkTikdtd41i=gmuEyVX1NFVaThJ=8AOpou=+ya7Xgt@mail.gmail.com>
Message-ID: <B35A9CE6-30AC-455F-8D17-364EC8773CB2@laposte.net>

Thanks Steven.

I found this useful and found also this page on wiki:
http://wiki.freeswitch.org/wiki/Variable_bridge_hangup_cause

I will give it a try.

regards,
rod

Le 2 mars 2011 ? 20:22, Steven Ayre a ?crit :

> 
> To sketch our how to do it...
> 
> In dialplan:
> <action application="set" data="hangup_on_bridge=true"/>
> <action application="set" data="continue_on_fail=true"/>
> <action application="bridge" data="..."/>
> <action application="lua" data="playerror.lua"/>
> 
> That'll run the lua script after the bridge (it'll only get there if
> the bridge fails)
> 
> In the lua script:
>   cause = session:getVariable("proto_specific_hangup_cause")
>   if (cause == "sip:486") then
>      session:streamFile("/path/to/busy.wav");
>   elseif (cause == "sip:404") then
>      session:streamFile("/path/to/unknown_user.wav");
>   elseif (cause == "sip:480") then
>      session:streamFile("/path/to/unavailable.wav");
>   else
>      session:streamFile("/path/to/unknown_error.wav");
>   end
> 
> You can also use the ISDN clearing cause from session:hangupCause() if you wish.
> 
> -Steve
> 
> 
> On 2 March 2011 16:07, kawarod <kawarod at laposte.net> wrote:
>> Hi List,
>> 
>> I'd like to know how I can play a file based on the SIP error code returned by a peer.
>> 
>> For example, I'd like to play the following announce if I receive a SIP 404:
>>        " The number you dialed is unreachable"
>> 
>> Thanks for your help.
>> 
>> rod.
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From dnotivol at gmail.com  Thu Mar  3 15:19:28 2011
From: dnotivol at gmail.com (David Notivol)
Date: Thu, 3 Mar 2011 13:19:28 +0100
Subject: [Freeswitch-users] mod_say in different codecs
Message-ID: <AANLkTikb_8mv+L1PSqRk-d2QMi1sNSJN-6_5WFmWDrKW@mail.gmail.com>

Hi all,

I'm trying to setup an IVR server, and I'm using the session:say function
from a LUA script.

My question is if there's any way of having mod_say playing audios different
than audio files; I mean audios encoded in G729, G711ulaw, G711alaw, etc. to
avoid having FS making transcoding every time I run a say command.

Checking the folders tree for the sounds, I can see the place for the
different languages, voices, and wav qualities (8k, 16k...) is clearly
specified; but is it a way to place files encoded in different codecs?

Thanks in advance.

Regards,
David Notivol
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From kris at livecall.com  Thu Mar  3 15:56:49 2011
From: kris at livecall.com (Kris)
Date: Thu, 3 Mar 2011 04:56:49 -0800
Subject: [Freeswitch-users] Conference - control actions problem
References: <AANLkTimq9dYbq8ztbQ5dJmY2Y+2mK5mKzM78UirEzzYn@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C58B2BBAF1D@cooper><AANLkTikPa1E9o3-NgWBjd7P8HWfr1vRxAnh+AH0UpqW1@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C58B2BBB2F5@cooper>
	<AANLkTim=VDq1ztB7Bn=_E0eq_8KeRnZceLapmT8jz2DS@mail.gmail.com>
Message-ID: <265B34F688F74462B0B6B54969AFB913@stor1>

I am also having strange problem with DTMFs after transfering last member 
out of conference back to my C# managed App- no more DTMFCallback to detect 
digits, but only on the last member out of the conf.. Seems like the last 
one keeps getting it's DTMF events trapped by mod_conference or FS. I just 
did a clean GIT clone from scratch a few hours ago. The transfer action 
control also hangs up the last caller as it has always done, so I have been 
using the Api. Something must have changed in the git in the past 1-2 weeks.



----- Original Message ----- 
From: "Maciej Bylica" <mbsip at gazeta.pl>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, March 03, 2011 2:07 AM
Subject: Re: [Freeswitch-users] Conference - control actions problem


Hi,

As stated in first email control action digits does not behave as they
should. Then it turned out that I have the same issue when these
actions are disabled.
Once again, lets assume that UA1 and UA2 are in conf room. UA1 is
pressing any digit (let say '5') and as a aftereffect it looks like
UA1 has microphone muted - so cannot be heard by UA2. Furthermore no
consecutive dtmfs coming from UA1 are collected.
The only wayout is to hang up the phone and dial again.

Thanks,
Maciej

> Hmm - I'm not really sure what your problem is. You receive DTMF, but 
> nothing happens anyway, or what?
>
> Latest git HEAD is stable, so you should be ok.
>
> Could you describe exactly what the problem is after sending a DTMF?
>
> /Peter
>
>
> -----Ursprungligt meddelande-----
> Fr?n: Maciej Bylica [mailto:mbsip at gazeta.pl]
> Skickat: den 2 mars 2011 17:15
> Till: FreeSWITCH Users Help
> Kopia: Peter Olsson
> ?mne: Re: [Freeswitch-users] Conference - control actions problem
>
> Thx Peter for prompt reply.
>
> I've just made one more configuration change. There are no lua script
> fired anymore, just pure conference application like following:
> <action application="conference" data="1000 at fsconf"/>
>
> The effect is the same (after pressing * or # key just one time).
> 2011-03-02 16:55:32.677018 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:520
> or later
> 2011-03-02 16:58:13.645012 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF *:620
>
> As you can see there are DTMF detected on Freeswitch interpreted as
> 'RTP RECV DTMF'
> Is git-06988e1 rel stable one?
>
>
> Regards,
> Maciej
>
> 2011/3/2 Peter Olsson <peter.olsson at visionutveckling.se>:
>> If you exit the conference (#) the user will still be in your lua script, 
>> so you would have to do whatever you want it to (hangup maybe). I think 
>> the recommended way is to transfer away the call to the conference, that 
>> way the Lua scripts won't need to do any more processing.
>>
>> Also check this Javascript example on the wiki: 
>> http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR
>>
>> About DTMF's - if you don't see them in your log, then FS havn't received 
>> any digits - you would have to do a packet capture and check the RTP 
>> packets.
>>
>> /Peter
>>
>> -----Ursprungligt meddelande-----
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org 
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Maciej Bylica
>> Skickat: den 2 mars 2011 12:31
>> Till: FreeSWITCH Users Help
>> ?mne: [Freeswitch-users] Conference - control actions problem
>>
>> Hello,
>>
>> I am running FreeSWITCH Version 1.0.head (git-06988e1 2011-02-20
>> 15-10-02 -0600) and have some problem with conference module and
>> respective control actions digits.
>> To give you an overview of how it is configured, there is lua script
>> fired in public.xml.
>> The script checks pin code, conf room and at the end run
>> "session:execute("conference", string.format("%s at fsconf", conf_num))".
>> Now coming back to the problem, there are few scenations i would like to 
>> cover.
>>
>> 1. When two or more people are in conference and one of them is trying
>> to push # button (RFC2833) then that person is leaving the conference
>> and is hanged up (not released, no BYE). The person who left the
>> conference cannot join again or perform other actions without making
>> the same phone call again.
>>
>> 2011-03-02 11:42:08.863139 [DEBUG] mod_conference.c:2414 Setup timer
>> soft success interval: 20 samples: 160
>> 2011-03-02 11:42:12.283051 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:480
>> 2011-03-02 11:42:12.323050 [DEBUG] mod_conference.c:2659 Channel
>> leaving conference, cause: NONE
>>
>> 2. The same as above, but * digit is entered. The person is muted and
>> then unmuted. Next try with * fires muted but this is the point where
>> the problem manifest itself.
>> The procedure of unmuting UA fails. Below you may find relevant logs.
>> The UA is pressing * but without success (no RTP RECV DTMF *).
>>
>> freeswitch at internal> 2011-03-02 11:51:13.423651 [DEBUG]
>> switch_rtp.c:3237 RTP RECV DTMF *:504
>> 2011-03-02 11:51:13.443650 [DEBUG] mod_conference.c:3125 Queueing file
>> '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav'
>> for play
>> 2011-03-02 11:51:18.163532 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF *:544
>> 2011-03-02 11:51:18.183531 [DEBUG] mod_conference.c:3125 Queueing file
>> '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-unmuted.wav'
>> for play
>> 2011-03-02 11:51:23.763391 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF *:504
>> 2011-03-02 11:51:23.783390 [DEBUG] mod_conference.c:3125 Queueing file
>> '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav'
>> for play
>>
>> 3. The same as above but with 0 digit
>>
>> 2011-03-02 12:04:21.064862 [DEBUG] mod_conference.c:2414 Setup timer
>> soft success interval: 20 samples: 160
>> 2011-03-02 12:04:22.863736 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 0:880
>> 2011-03-02 12:04:22.883736 [DEBUG] mod_conference.c:3125 Queueing file
>> '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav'
>> for play
>>
>> then i cannot get back to the conference
>>
>>
>> 4. The as above but with 6 digit
>>
>> 2011-03-02 12:15:03.244609 [DEBUG] mod_conference.c:2414 Setup timer
>> soft success interval: 20 samples: 160
>> 2011-03-02 12:15:21.602919 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 6:504
>> 2011-03-02 12:15:22.602893 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 6:800
>> 2011-03-02 12:15:24.263851 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 6:864
>>
>>
>> I have disabled all control digits, but the problem still persist. No
>> action is fired but is looks like UA is being muted (micro is off, but
>> i can hear the other person).
>> 2011-03-02 12:24:05.322935 [DEBUG] mod_conference.c:2414 Setup timer
>> soft success interval: 20 samples: 160
>> 2011-03-02 12:24:08.663851 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:480
>> 2011-03-02 12:24:11.223787 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:504
>> 2011-03-02 12:24:13.703725 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:480
>> 2011-03-02 12:24:16.043666 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:520
>>
>>
>> Has anybody similar experience or know where the problem may be located.
>>
>> Regards,
>> Maciej
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> !DSPAM:4d6e6c8a32761758178328!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>





From steveayre at gmail.com  Thu Mar  3 17:26:01 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 3 Mar 2011 14:26:01 +0000
Subject: [Freeswitch-users] mod_say in different codecs
In-Reply-To: <AANLkTikb_8mv+L1PSqRk-d2QMi1sNSJN-6_5WFmWDrKW@mail.gmail.com>
References: <AANLkTikb_8mv+L1PSqRk-d2QMi1sNSJN-6_5WFmWDrKW@mail.gmail.com>
Message-ID: <375B3BB7-DC39-4C80-A32B-DF1941904291@gmail.com>

Look at mod_nativefile - does that suit your needs?

Steve on iPhone

On 3 Mar 2011, at 12:19, David Notivol <dnotivol at gmail.com> wrote:

> Hi all,
> 
> I'm trying to setup an IVR server, and I'm using the session:say function from a LUA script.
> 
> My question is if there's any way of having mod_say playing audios different than audio files; I mean audios encoded in G729, G711ulaw, G711alaw, etc. to avoid having FS making transcoding every time I run a say command.
> 
> Checking the folders tree for the sounds, I can see the place for the different languages, voices, and wav qualities (8k, 16k...) is clearly specified; but is it a way to place files encoded in different codecs?
> 
> Thanks in advance.
> 
> Regards,
> David Notivol
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From misi at niif.hu  Thu Mar  3 17:42:14 2011
From: misi at niif.hu (=?ISO-8859-1?Q?M=C9SZ=C1ROS_Mih=E1ly?=)
Date: Thu, 03 Mar 2011 15:42:14 +0100
Subject: [Freeswitch-users] Freeswtich as a media proxy between
 ipv4<=>ipv6 using Polycom HDX8006 SIP UA-s
In-Reply-To: <AANLkTinqtxZ94zKXiwFBWpyKkQPiOqzAPWt6_3EP5pqH@mail.gmail.com>
References: <4D650823.5050305@niif.hu>
	<AANLkTinqtxZ94zKXiwFBWpyKkQPiOqzAPWt6_3EP5pqH@mail.gmail.com>
Message-ID: <4D6FA8C6.7070605@niif.hu>

Hi Steven et al.

2011-02-25 16:01 keltez?ssel, Steven Ayre ?rta:
>> Is it possible to create ipv6<=>  ipv4 media proxy from FreeSwitch?
>> So i need to mangle SDP to replace ipv6 to ipv4 and vice-versa. (but i use
>> fnacy things like BFCP,FECC(H.224),secondary video)
>>      
> If aleg is on IPv4 and bleg is on IPv6 then FS will allow that and
> it'll handle putting the correct IPs in the SDPs itself. No extra work
> required. A sip profile for each class, and bridge between them in
> dialplan.
>    
It is working but i need extra features like secondary video stream for 
content and Application protocol (H.224 for Far End Camera Control) and 
BFCP for content.

So in B2BUA mode with bridged media the advanced protocols will not work.
If i use FreeSwitch media_bypass mode, so SDP and the media is flowing 
directly then i can't connect ipv4 only with ipv6 only media.

So i need media proxy mode, but i am experiencing that advanced 
features, don't work.
So what i need to replace only ipv6 and ipv4 address in sdp, and connect 
all streams regardless they are UDP or TCP or otherelse to FreeSwitch 
internal media proxy.
I miss application media streams proxy capability etc.
>    
>> Further more I need to know that FreeSwitch can function as a real media
>> proxy?
>>      
> Yes, it does.
>
>    
Ok I am experiencing FS media proxy has limitations.  AFAIK only one 
audio and one video stream, no application protocols.
>> So can it PROXY as MEDIA like  BFCP(TCP!), FECC and secondary video?
>>      
> No support for BFCP.
> FECC isn't supported natively, but should work in proxy_media mode
> where FS doesn't need to know the codec, just forwards the RTP
> straight through.
> AFAIK, secondary video isn't supported. It might work in proxy_media
> but probably not...
>
> However, you might find you have a problem even with proxy_media...
>    
>> m=application
>>      
> I've only ever seen FS handling m=audio and m=video streams. I don't
> know whether it would handle m=application ones.
>
>    
My experience is that it is not handling m=application . :-(

Thank you for your reply!

Thanks,
Misi
> -Steve
>
>
>
> 2011/2/23 M?SZ?ROS Mih?ly<misi at niif.hu>:
>    
>> Hi,
>>
>> Is it possible to create ipv6<=>  ipv4 media proxy from FreeSwitch?
>> So i need to mangle SDP to replace ipv6 to ipv4 and vice-versa. (but i use
>> fnacy things like BFCP,FECC(H.224),secondary video)
>> Further more I need to know that FreeSwitch can function as a real media
>> proxy?
>> So can it PROXY as MEDIA like  BFCP(TCP!), FECC and secondary video?
>> Can i use more than one stream so more than 1 audio + 1 video stream in a
>> sip call in proxy media mode?
>> For example 1 audio + 2 video stream (people+presentation)
>>
>> Example SDP piece for BFCP, and FECC(H.224):
>>
>> m=application 49158 RTP/SAVP 100
>> a=rtpmap:100 H224/4800
>> a=crypto:1 AES_CM_128_HMAC_SHA1_80
>> inline:lv6Y5eO/RSDDOFNvbFmM0q9tfrt3/ZxXqPJx4Pj5|2^31
>> m=application 0 TCP/BFCP *
>> a=floorctrl:c-s
>> a=setup:actpass
>> a=connection:new
>>
>>
>>
>> Any help highly appreciated!
>>
>> Thanks,
>> Misi
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>      
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    



From benkokakao at gmail.com  Thu Mar  3 17:59:17 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Thu, 3 Mar 2011 15:59:17 +0100
Subject: [Freeswitch-users] FS Webinterfaces - Opinions on wikiPBX, blue.box,
	fusionPBX
Message-ID: <AANLkTik-J2z-P7EvghBH1i-EPL-_G7FqDGX0PCvqYu+g@mail.gmail.com>

Hi!

I'm currently working on a PBX solution for customers of my
company(Small to medium businesses) and since i have to make sure my
colleagues and customers don't have to dig into the depths of
Freeswitch i need a graphical interface for the most mundane tasks.

I've spent the last week evaluating FusionPBX and while i was excited
at first, i stumbled over a few bugs(Which i could partially solve
myself) and had to dig a bit in the code, so i'm not that thrilled
anymore. Before i stay with FusionPBX and try to learn some more php
and work my way through the code(Is there no ORM in php or did Mark
just leave it out?) and add what i'm missing, i'd like to spend
another few days trying other projects.

Since it takes a while to get to understand a new webinterface and its
quirks, i wanted to ask here first if the other options are even worth
the time. I think the alternatives are blue.box and WikiPBX?

What i'm specifically looking for are:

- Proper AAA/groups
I want to be able to have a "root"-admin(SuperAdmin in FusionPBX) that
has access to the full GUI, a local admin(Admin in FusionPBX) that can
change call-relevant features like conferences, dialplans, follow me,
voicemail, CDRs and such and finally users, who can only change
settings for their own extension and have access to their voicemail
and CDRs. Ideally the root-admin should be able to modify what
options/features the "lower" groups are able to manipulate(Switch
on/off GUI-elements). FusionPBX offers that partially - but the "local
admin" has too much rights and i can't change its rights dynamically.

- A layer between GUI and configfiles(e.g. a database)
I'll have to do some bulk-provisioning with a script from csv-files
and if i can write user-information to a DB instead of creating
xml-files it would make my life easier

- Feature configuration
Dialplans are complex and individual and i can comprehend that i won't
find an interface that supports all the features i need.
FusionPBX was satisfying in this regard, i didn't miss too much and
while i couldn't trust a customer to setup his PBX with a complex
configuration with Time Conditions, Hunt Groups, Follow-Me etc. i
think at least our 2nd level support would be happy to use the gui
instead of tinkering with the flatfiles. And i hope my time allows me
to implement some of the needed features to FusionPBX(If i don't find
a better alternative).
Some of them are: Voicemail(Setup and file-access),
Huntgroups/FollowMe, Recording, Click2Call, Conferences, CDRs,
AutoAttendant/IVR, MOH, TimeConditions, Directory, BossSec/ChefSek and
other dubious features asked for by upper mgmt

- Phone provisioning
It would be nice to be able to add a MAC-adress to a newly created
extension and the rest is taken care of by the GUI-scripts. FusionPBX
has this feature, but i didn't have time to try it yet(And i'll have
to adapt the templates to fit the phones i'm going to use, hope it
really works). Nice to have, if not it shouldn't be too much work to
write a custom script that can take care of that, given provisioning
data from the GUI.

- Written in a language i don't have to learn(Python)
WikiPBX is based on Django - yes! "http://www.wikipbx.org/ -> Current
Status -> CDR issues improved -> 11 Feb 2010" - Crap! Is this project
as dead as it seems or is it just so mature there's no further
development needed? The screenshots look a bit minimalistic.


That's all i need in a perfect world. I guess none of the projects
mentioned above can do that for me - but i'd still love to hear your
opinions and experiences if it's even worth the time to install/try
any of them or stick with FusionPBX!

Cheers,
Christian


From avi at avimarcus.net  Thu Mar  3 18:31:25 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 3 Mar 2011 17:31:25 +0200
Subject: [Freeswitch-users] FS Webinterfaces - Opinions on wikiPBX,
 blue.box, fusionPBX
In-Reply-To: <AANLkTik-J2z-P7EvghBH1i-EPL-_G7FqDGX0PCvqYu+g@mail.gmail.com>
References: <AANLkTik-J2z-P7EvghBH1i-EPL-_G7FqDGX0PCvqYu+g@mail.gmail.com>
Message-ID: <AANLkTikVTwM3FfsuxYfH5D3HTM5LnYJq9E5eCEkA4LEm@mail.gmail.com>

Hey. Not sure what you really want - FusionPBX's next project is a
permission rewrite - multi-tenant was just added.
Depending on what you want to change though, you might be able to just
change a line or two in the actual php files.
I don't recall hearing any questions on #fusionpbx in freenode. I haven't
heard of any any outstanding bugs...

And yeah, wikipbx seems to be dead. As opposed to FusionPBX who was on the
conference yesterday about the roadmap and he gets sponsorships to add new
stuff.

There's blue.box also as you mentioned. Haven't touched it in a while.

-Avi

On Thu, Mar 3, 2011 at 4:59 PM, Christian Benke <benkokakao at gmail.com>wrote:

> Hi!
>
> I'm currently working on a PBX solution for customers of my
> company(Small to medium businesses) and since i have to make sure my
> colleagues and customers don't have to dig into the depths of
> Freeswitch i need a graphical interface for the most mundane tasks.
>
> I've spent the last week evaluating FusionPBX and while i was excited
> at first, i stumbled over a few bugs(Which i could partially solve
> myself) and had to dig a bit in the code, so i'm not that thrilled
> anymore. Before i stay with FusionPBX and try to learn some more php
> and work my way through the code(Is there no ORM in php or did Mark
> just leave it out?) and add what i'm missing, i'd like to spend
> another few days trying other projects.
>
> Since it takes a while to get to understand a new webinterface and its
> quirks, i wanted to ask here first if the other options are even worth
> the time. I think the alternatives are blue.box and WikiPBX?
>
> What i'm specifically looking for are:
>
> - Proper AAA/groups
> I want to be able to have a "root"-admin(SuperAdmin in FusionPBX) that
> has access to the full GUI, a local admin(Admin in FusionPBX) that can
> change call-relevant features like conferences, dialplans, follow me,
> voicemail, CDRs and such and finally users, who can only change
> settings for their own extension and have access to their voicemail
> and CDRs. Ideally the root-admin should be able to modify what
> options/features the "lower" groups are able to manipulate(Switch
> on/off GUI-elements). FusionPBX offers that partially - but the "local
> admin" has too much rights and i can't change its rights dynamically.
>
> - A layer between GUI and configfiles(e.g. a database)
> I'll have to do some bulk-provisioning with a script from csv-files
> and if i can write user-information to a DB instead of creating
> xml-files it would make my life easier
>
> - Feature configuration
> Dialplans are complex and individual and i can comprehend that i won't
> find an interface that supports all the features i need.
> FusionPBX was satisfying in this regard, i didn't miss too much and
> while i couldn't trust a customer to setup his PBX with a complex
> configuration with Time Conditions, Hunt Groups, Follow-Me etc. i
> think at least our 2nd level support would be happy to use the gui
> instead of tinkering with the flatfiles. And i hope my time allows me
> to implement some of the needed features to FusionPBX(If i don't find
> a better alternative).
> Some of them are: Voicemail(Setup and file-access),
> Huntgroups/FollowMe, Recording, Click2Call, Conferences, CDRs,
> AutoAttendant/IVR, MOH, TimeConditions, Directory, BossSec/ChefSek and
> other dubious features asked for by upper mgmt
>
> - Phone provisioning
> It would be nice to be able to add a MAC-adress to a newly created
> extension and the rest is taken care of by the GUI-scripts. FusionPBX
> has this feature, but i didn't have time to try it yet(And i'll have
> to adapt the templates to fit the phones i'm going to use, hope it
> really works). Nice to have, if not it shouldn't be too much work to
> write a custom script that can take care of that, given provisioning
> data from the GUI.
>
> - Written in a language i don't have to learn(Python)
> WikiPBX is based on Django - yes! "http://www.wikipbx.org/ -> Current
> Status -> CDR issues improved -> 11 Feb 2010" - Crap! Is this project
> as dead as it seems or is it just so mature there's no further
> development needed? The screenshots look a bit minimalistic.
>
>
> That's all i need in a perfect world. I guess none of the projects
> mentioned above can do that for me - but i'd still love to hear your
> opinions and experiences if it's even worth the time to install/try
> any of them or stick with FusionPBX!
>
> Cheers,
> Christian
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From benkokakao at gmail.com  Thu Mar  3 19:40:06 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Thu, 3 Mar 2011 17:40:06 +0100
Subject: [Freeswitch-users] FS Webinterfaces - Opinions on wikiPBX,
 blue.box, fusionPBX
In-Reply-To: <AANLkTikVTwM3FfsuxYfH5D3HTM5LnYJq9E5eCEkA4LEm@mail.gmail.com>
References: <AANLkTik-J2z-P7EvghBH1i-EPL-_G7FqDGX0PCvqYu+g@mail.gmail.com>
	<AANLkTikVTwM3FfsuxYfH5D3HTM5LnYJq9E5eCEkA4LEm@mail.gmail.com>
Message-ID: <AANLkTiktKMFdEtknKJzC=h5g76HVBPEyOeOchRGdRXMm@mail.gmail.com>

On 3 March 2011 16:31, Avi Marcus <avi at avimarcus.net> wrote:
> Hey. Not sure what you really want - FusionPBX's next project is a
> permission?rewrite?- multi-tenant was just added.

The main feature i'm missing is exactly that, giving granulated
permissions to groups - beeing able to dynamically modify the access
and permissions of users - which is currently hard-coded for
admin/superadmin as far as i've grasped.
If my managers agree on having the same set of permissions for our
customers though, i could remove some of the admin-permission in the
code and live without a dynamic solution for a while(Until we need to
release a 2nd version of our product next year).

Another important thing i miss is a proper debug mode - FusionPBX does
not report errors and acts as if everything went fine, while postgres
actually fills logs with syntax-errors and such. I've been spoiled by
the error-handling of python-frameworks i'm afraid.

> Depending on what you want to change though, you might be able to just
> change a line or two in the actual php files.

I should have followed through reading that PHP-book i bought 8 years
ago, i've not much indulged in the pleasures of working with PHP :-)
As i wrote i can fix small issues which i can debug without effort,
but to dig deeper into the code and really understand the project will
take more time than i can afford right now(Which i'll have to do if i
want to add features like Blacklisting or Boss/Sec as "easy to
configure" GUI-elements).

> I don't recall hearing any questions on #fusionpbx in freenode. I haven't
> heard of any any outstanding bugs...

Didn't really seek support as i've a tight schedule and if i can't get
something to work within a few hours i move on to the next.
Small bugs like http://code.google.com/p/fusionpbx/issues/detail?id=57
and the lack of proper error-reporting pester me - i've checked out
the latest revision from trunk though, should i've taken the regular
packed release?

> And yeah, wikipbx seems to be dead. As opposed to FusionPBX who was on the
> conference yesterday about the roadmap and he gets sponsorships to add new
> stuff.

Sounds great :-)

Cheers,
Christian


From lloydie.t at gmail.com  Thu Mar  3 20:03:59 2011
From: lloydie.t at gmail.com (lloyd thomas)
Date: Thu, 3 Mar 2011 17:03:59 +0000
Subject: [Freeswitch-users] Dialplan in a muddle
Message-ID: <AANLkTi=GE8X_FC7UOfxLvjP6EFzttjdsocLcqNNvRA2t@mail.gmail.com>

Maybe getting a bit ahead of myself but I am trying to set up a
multi-company dial plan. Things are working fine internally, but I am not
getting the grasp for external calls.

The first company xml file is conf/dialplan/companya.tele.co.uk.xml

which has the line
<X-PRE-PROCESS cmd="include" data="companya.tele.co.uk/*.xml"/>
to include xml files in that conf/dialplan/companya.tele.co.uk folder

In this folder I have the xml file soho66.co.uk.xml which includes the regex
to remove the initial 9 when dialling and extenal number. I cant seem to be
able to get a call to go anywhere near connecting to the soho gateway.

I think problem is to do with this

2011-03-03 17:00:23.089153 [NOTICE] switch_ivr.c:1606 Transfer
sofia/internal/200 at companya.tele.co.uk to enum[
907973323260 at companya.tele.co.uk]

help me please!

----------------- soho66.co.uk.xml----------------------------
<include>
<extension name="Local Dial">
   <condition field="destination_number" expression="^9?([2-9]{6})$">
     <action application="set" data="effective_caller_id_name=12345678"/>
     <action application="set" data="effective_caller_id_number=12345678"/>
     <action application="bridge" data="sofia/gateway/
sip.soho66.co.uk/01869$1 at sip.soho66.co.uk"/>
   </condition>
</extension>

<extension name="Regional 01">
  <condition field="destination_number" expression="^9?(01\d{9})$">
     <action application="set" data="effective_caller_id_name=12345678"/>
     <action application="set" data="effective_caller_id_number=12345678"/>
     <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1"/>
  </condition>
</extension>

<extension name="Regional 02">
  <condition field="destination_number" expression="^9?(02\d{9})$$">
     <action application="set" data="effective_caller_id_name=12345678"/>
     <action application="set" data="effective_caller_id_number=123456781"/>
     <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1"/>
  </condition>
</extension>

<extension name="Mobile">
  <condition field="destination_number" expression="^9?(07\d{9})$">
     <action application="set" data="effective_caller_id_name=12345678"/>
     <action application="set" data="effective_caller_id_number=12345678"/>
     <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1"/>
  </condition>
</extension>

<extension name="International">
  <condition field="destination_number" expression="^9?(00\d*))$">
     <action application="set" data="effective_caller_id_name=12345678"/>
     <action application="set" data="effective_caller_id_number=12345678"/>
     <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1"/>
  </condition>
</extension>
</include>
-----------------eof soho66.co.uk.xml-------------------------

-------------conf/sip_profiles/external/companya_soho66.xml
-----------------------
<include>
      <gateway name="sip.soho66.co.uk">
      <param name="username" value="12345678"/>
      <param name="from-user" value="12345678"/>
      <param name="password" value="ABCDEFGH"/>
      <param name="expire-seconds" value="600"/>
      <param name="register" value="true"/>
      <param name="retry-seconds" value="30"/>
    </gateway>
</include>
------------- eof conf/sip_profiles/external/companya_soho66.xml
-----------------------
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From frank at telonium.com  Thu Mar  3 20:17:54 2011
From: frank at telonium.com (Frank Park)
Date: Thu, 3 Mar 2011 12:17:54 -0500
Subject: [Freeswitch-users] xml_curl and domain aliases
In-Reply-To: <AANLkTimS2ZbzaVaVGzgraGjNF5MZd=909VBD3dg7yNSP@mail.gmail.com>
References: <AANLkTimS2ZbzaVaVGzgraGjNF5MZd=909VBD3dg7yNSP@mail.gmail.com>
Message-ID: <AANLkTik8NCD4bUsLYt=K16z6HygdQKWjLAzK_fcK6sXV@mail.gmail.com>

We use domain aliases to do multi-tenancy and we have SIP clients to specify
the realm that matches the alias you have, and outbound proxy to match the
actual proxy server.
This will use the profile alias to register, (which then you should be able
to see in the sip_registrations table (in columns sip_host and mwi_host).
You should then be able to use user/internal/${userpart}@{domain_name}.

As for registering users via IP address, you should look into mod_acl

Frank

On Thu, Mar 3, 2011 at 3:58 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:

> Hi All
>
> I'm trying to setup xml_curl-based directory to provide registration and
> bridging information from database.
>
> The thing I can't still reliably handle is domain aliases. I have some
> domain like 'access.freeswitch.test', but I need to allow users to register
> via ip address, for example, 1.1.1.1. My scripts answer on REGISTER curl
> request with directory entry where domain is set to domain user trying to
> register at, and this works.
>
> But then I need to dial that user. I'm trying to use
> user/${userpart}%{domainpart} syntax, but I don't know exactly domain user
> was registered at. How to tell freeswitch to take in account domain aliases
> when looking up for sofia contact?
> Or, instead, can I make freeswitch to store user in registration database
> with some primary domain, which I can then specify to bridge(user/) command?
>
> Thanks in advance!
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 

----=======================----
Frank Park
Telonium Communications, LLC
frank at telonium.com
http://www.telonium.com
Follow Us on Twitter: @GetTelonium
404-566-8888 x1001 Office
404-939-4242 Cell
----=======================----
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From lloyd.aloysius at gmail.com  Thu Mar  3 20:25:46 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 3 Mar 2011 12:25:46 -0500
Subject: [Freeswitch-users] Dialplan in a muddle
In-Reply-To: <AANLkTi=GE8X_FC7UOfxLvjP6EFzttjdsocLcqNNvRA2t@mail.gmail.com>
References: <AANLkTi=GE8X_FC7UOfxLvjP6EFzttjdsocLcqNNvRA2t@mail.gmail.com>
Message-ID: <AANLkTimvk0VbS5r8nEKzTGi5GPykHj5KOa17m+pFEROQ@mail.gmail.com>

Can you post your conf/dialplan/companya.tele.co.uk.xml

I think there is a problem in the above file. Do you have a

 <extension name="enum">
      <condition field="${module_exists(mod_enum)}" expression="true"/>
      <condition field="destination_number" expression="^(.*)$">
        <action application="transfer" data="$1 enum"/>
      </condition>
    </extension>

and

the line <X-PRE-PROCESS cmd="include" data="companya.tele.co.uk/*.xml"/>
below?

THanks
Lloyd

On Thu, Mar 3, 2011 at 12:03 PM, lloyd thomas <lloydie.t at gmail.com> wrote:

> Maybe getting a bit ahead of myself but I am trying to set up a
> multi-company dial plan. Things are working fine internally, but I am not
> getting the grasp for external calls.
>
> The first company xml file is conf/dialplan/companya.tele.co.uk.xml
>
> which has the line
> <X-PRE-PROCESS cmd="include" data="companya.tele.co.uk/*.xml"/>
> to include xml files in that conf/dialplan/companya.tele.co.uk folder
>
> In this folder I have the xml file soho66.co.uk.xml which includes the
> regex to remove the initial 9 when dialling and extenal number. I cant seem
> to be able to get a call to go anywhere near connecting to the soho gateway.
>
> I think problem is to do with this
>
> 2011-03-03 17:00:23.089153 [NOTICE] switch_ivr.c:1606 Transfer
> sofia/internal/200 at companya.tele.co.uk to enum[
> 907973323260 at companya.tele.co.uk]
>
> help me please!
>
> ----------------- soho66.co.uk.xml----------------------------
> <include>
> <extension name="Local Dial">
>    <condition field="destination_number" expression="^9?([2-9]{6})$">
>      <action application="set" data="effective_caller_id_name=12345678"/>
>      <action application="set" data="effective_caller_id_number=12345678"/>
>      <action application="bridge" data="sofia/gateway/
> sip.soho66.co.uk/01869$1 at sip.soho66.co.uk"/>
>    </condition>
> </extension>
>
> <extension name="Regional 01">
>   <condition field="destination_number" expression="^9?(01\d{9})$">
>      <action application="set" data="effective_caller_id_name=12345678"/>
>      <action application="set" data="effective_caller_id_number=12345678"/>
>      <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1
> "/>
>   </condition>
> </extension>
>
> <extension name="Regional 02">
>   <condition field="destination_number" expression="^9?(02\d{9})$$">
>      <action application="set" data="effective_caller_id_name=12345678"/>
>      <action application="set"
> data="effective_caller_id_number=123456781"/>
>      <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1
> "/>
>   </condition>
> </extension>
>
> <extension name="Mobile">
>   <condition field="destination_number" expression="^9?(07\d{9})$">
>      <action application="set" data="effective_caller_id_name=12345678"/>
>      <action application="set" data="effective_caller_id_number=12345678"/>
>      <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1
> "/>
>   </condition>
> </extension>
>
> <extension name="International">
>   <condition field="destination_number" expression="^9?(00\d*))$">
>      <action application="set" data="effective_caller_id_name=12345678"/>
>      <action application="set" data="effective_caller_id_number=12345678"/>
>      <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1
> "/>
>   </condition>
> </extension>
> </include>
> -----------------eof soho66.co.uk.xml-------------------------
>
> -------------conf/sip_profiles/external/companya_soho66.xml
> -----------------------
> <include>
>       <gateway name="sip.soho66.co.uk">
>       <param name="username" value="12345678"/>
>       <param name="from-user" value="12345678"/>
>       <param name="password" value="ABCDEFGH"/>
>       <param name="expire-seconds" value="600"/>
>       <param name="register" value="true"/>
>       <param name="retry-seconds" value="30"/>
>     </gateway>
> </include>
> ------------- eof conf/sip_profiles/external/companya_soho66.xml
> -----------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dftoro at yahoo.com  Thu Mar  3 20:30:07 2011
From: dftoro at yahoo.com (Diego Toro)
Date: Thu, 3 Mar 2011 09:30:07 -0800 (PST)
Subject: [Freeswitch-users] How to create IVR application in C#?
In-Reply-To: <AANLkTi=ynpjSF0G43o6a7GG8TOHntZJZ0erO42XFhC8B@mail.gmail.com>
Message-ID: <184962.13646.qm@web33505.mail.mud.yahoo.com>

Hi, use ESL Managed is better.

Diego 

http://voipensando.blogspot.com/

--- On Wed, 3/2/11, Frankie Yiu <frankie.k.yiu at gmail.com> wrote:

From: Frankie Yiu <frankie.k.yiu at gmail.com>
Subject: Re: [Freeswitch-users] How to create IVR application in C#?
To: freeswitch-users at lists.freeswitch.org
Date: Wednesday, March 2, 2011, 5:00 PM

Thanks!

I am using C# managed function through swig.cs.? And I am calling the function:

switch_ivr_menu_init(SWIGTYPE_p_p_switch_ivr_menu new_menu,.......)

With the type "SWIGTYPE_p_p_switch_ivr_menu",
 do they type ever work?? It has a run time error saying "Attempted to 
read or write protected memory. This is often an indication that other 
memory is corrupt." for this variable basically it is dereferencing a 
Null pointer.


How do I resolve this if I want to call this in C#?

Thanks,

Frankie


---------- Forwarded message ----------
From:?Malay Thakershi <mthakershi at gmail.com>
To:?FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>

Date:?Tue, 1 Mar 2011 16:42:47 -0600
Subject:?Re: [Freeswitch-users] How to create IVR application in C#?
mod_managed could be an option.
http://wiki.freeswitch.org/wiki/Mod_managed

It allows you to use most native FS features from C# managed code.

Malay

On Tue, Mar 1, 2011 at 3:01 PM, Frankie Yiu <frankie.k.yiu at gmail.com> wrote:


Hi there,

I am newbie to FreeSwitch and I have question about creating an IVR application in C#, with a possibly of using VoiceXML.? Could someone please points me to how I can get started or any example that I can look at?




Thanks a lot!!!

Frankie









-----Inline Attachment Follows-----

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From lloyd.aloysius at gmail.com  Thu Mar  3 20:30:06 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 3 Mar 2011 12:30:06 -0500
Subject: [Freeswitch-users] Cisco SPA504G - BLF Issues
Message-ID: <AANLkTinoVeHcJUOG05CNxBmOLbgi5Y1y1RUSs0ojN18x@mail.gmail.com>

Hi All,

Is there any know issues with Cisco504G + BLF and Firmware Version 7.4.7

I configure the Line4 key to monitor the Extension 201. The following
configuration I used.

Line 4

Extenion : Disabled
Shared Call Appearance: Shared
Extended Function : fnc=blf+sd+cp;sub=201 at mydomain.com;nme=201
Short Name: Ext 201


When Ext 201 on the Phone BLF works then after few second the BLF key stop
working and the LED lights off. But Ext 201 still on the phone.

Any help is appreciated.

Thanks
Lloyd
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From avi at avimarcus.net  Thu Mar  3 20:32:36 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 3 Mar 2011 19:32:36 +0200
Subject: [Freeswitch-users] FS Webinterfaces - Opinions on wikiPBX,
 blue.box, fusionPBX
In-Reply-To: <AANLkTiktKMFdEtknKJzC=h5g76HVBPEyOeOchRGdRXMm@mail.gmail.com>
References: <AANLkTik-J2z-P7EvghBH1i-EPL-_G7FqDGX0PCvqYu+g@mail.gmail.com>
	<AANLkTikVTwM3FfsuxYfH5D3HTM5LnYJq9E5eCEkA4LEm@mail.gmail.com>
	<AANLkTiktKMFdEtknKJzC=h5g76HVBPEyOeOchRGdRXMm@mail.gmail.com>
Message-ID: <AANLkTimKYJWyHP2aUgMOqRvvU-6dPyLNy-+iN7HPf8yS@mail.gmail.com>

On Thu, Mar 3, 2011 at 6:40 PM, Christian Benke <benkokakao at gmail.com>wrote:

> On 3 March 2011 16:31, Avi Marcus <avi at avimarcus.net> wrote:
> > Hey. Not sure what you really want - FusionPBX's next project is a
> > permission rewrite - multi-tenant was just added.
>
> The main feature i'm missing is exactly that, giving granulated
> permissions to groups - beeing able to dynamically modify the access
> and permissions of users - which is currently hard-coded for
> admin/superadmin as far as i've grasped.
> If my managers agree on having the same set of permissions for our
> customers though, i could remove some of the admin-permission in the
> code and live without a dynamic solution for a while(Until we need to
> release a 2nd version of our product next year).
>
If this is for business, perhaps you can sponsor it.


>
> Another important thing i miss is a proper debug mode - FusionPBX does
> not report errors and acts as if everything went fine, while postgres
> actually fills logs with syntax-errors and such. I've been spoiled by
> the error-handling of python-frameworks i'm afraid.
>
Yeah, I have noticed that. I usually make stuff show errors. An error
handler should be able to be written into the sql classes if someone puts in
the time.

>
> > Depending on what you want to change though, you might be able to just
> > change a line or two in the actual php files.
>
> I should have followed through reading that PHP-book i bought 8 years
> ago, i've not much indulged in the pleasures of working with PHP :-)
> As i wrote i can fix small issues which i can debug without effort,
> but to dig deeper into the code and really understand the project will
> take more time than i can afford right now(Which i'll have to do if i
> want to add features like Blacklisting or Boss/Sec as "easy to
> configure" GUI-elements).
>
> > I don't recall hearing any questions on #fusionpbx in freenode. I haven't
> > heard of any any outstanding bugs...
>
> Didn't really seek support as i've a tight schedule and if i can't get
> something to work within a few hours i move on to the next.
> Small bugs like http://code.google.com/p/fusionpbx/issues/detail?id=57
> and the lack of proper error-reporting pester me - i've checked out
> the latest revision from trunk though, should i've taken the regular
> packed release?
>
Not really a point to using a release, we work on the latest SVN. Just some
big changes tend to make for a slight upgrade hiccup that wouldn't happen on
a fresh install (e.g. the last multi-tenant on gateways).

-Avi
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From benkokakao at gmail.com  Thu Mar  3 20:49:48 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Thu, 3 Mar 2011 18:49:48 +0100
Subject: [Freeswitch-users] FS Webinterfaces - Opinions on wikiPBX,
 blue.box, fusionPBX
In-Reply-To: <AANLkTimKYJWyHP2aUgMOqRvvU-6dPyLNy-+iN7HPf8yS@mail.gmail.com>
References: <AANLkTik-J2z-P7EvghBH1i-EPL-_G7FqDGX0PCvqYu+g@mail.gmail.com>
	<AANLkTikVTwM3FfsuxYfH5D3HTM5LnYJq9E5eCEkA4LEm@mail.gmail.com>
	<AANLkTiktKMFdEtknKJzC=h5g76HVBPEyOeOchRGdRXMm@mail.gmail.com>
	<AANLkTimKYJWyHP2aUgMOqRvvU-6dPyLNy-+iN7HPf8yS@mail.gmail.com>
Message-ID: <AANLkTikHHbF-5Rbg4RhrRX9wY6eaZcgLVDK=BBmiQYUR@mail.gmail.com>

On 3 March 2011 18:32, Avi Marcus <avi at avimarcus.net> wrote:
> If this is for business, perhaps you can sponsor it.

Yeah, i'll definitely push for sponsoring the involved OSS-projects
when we're finished and the project is a commercial success.
This is the first time this company uses an open source solution and
i've been hired specifically for this task(One month ago), so i'm
going to give them a slow introduction to how the OSS-world works ;-)
If we absolutely need a feature in the GUI _now_ which i can't
implement, i guess i will convince them to send out a bounty.

I've also not yet told them yet that we're not going to use
Asterisk(Which they refer to when they talk about "my" project) :-P

Cheers,
Christian


From msc at freeswitch.org  Thu Mar  3 21:16:03 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 3 Mar 2011 10:16:03 -0800
Subject: [Freeswitch-users] Conference - control actions problem
In-Reply-To: <265B34F688F74462B0B6B54969AFB913@stor1>
References: <AANLkTimq9dYbq8ztbQ5dJmY2Y+2mK5mKzM78UirEzzYn@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2BBAF1D@cooper>
	<AANLkTikPa1E9o3-NgWBjd7P8HWfr1vRxAnh+AH0UpqW1@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2BBB2F5@cooper>
	<AANLkTim=VDq1ztB7Bn=_E0eq_8KeRnZceLapmT8jz2DS@mail.gmail.com>
	<265B34F688F74462B0B6B54969AFB913@stor1>
Message-ID: <AANLkTinJD756oM8cQDi+=nPPjYNXH8Qid4DVygMbv_Xp@mail.gmail.com>

On Thu, Mar 3, 2011 at 4:56 AM, Kris <kris at livecall.com> wrote:

> I am also having strange problem with DTMFs after transfering last member
> out of conference back to my C# managed App- no more DTMFCallback to detect
> digits, but only on the last member out of the conf.. Seems like the last
> one keeps getting it's DTMF events trapped by mod_conference or FS. I just
> did a clean GIT clone from scratch a few hours ago. The transfer action
> control also hangs up the last caller as it has always done, so I have been
> using the Api. Something must have changed in the git in the past 1-2
> weeks.
>

Please find the last known git release where it worked and/or the first git
release where it broke. Then open a Jira and supply a sample
config/script/etc. that will allow the devs to recreate this issue.

-MC
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From azatek0 at gmail.com  Thu Mar  3 21:22:24 2011
From: azatek0 at gmail.com (Aza Tek)
Date: Thu, 3 Mar 2011 20:22:24 +0200
Subject: [Freeswitch-users] FS Webinterfaces - Opinions on wikiPBX,
 blue.box, fusionPBX
In-Reply-To: <AANLkTikHHbF-5Rbg4RhrRX9wY6eaZcgLVDK=BBmiQYUR@mail.gmail.com>
References: <AANLkTik-J2z-P7EvghBH1i-EPL-_G7FqDGX0PCvqYu+g@mail.gmail.com>
	<AANLkTikVTwM3FfsuxYfH5D3HTM5LnYJq9E5eCEkA4LEm@mail.gmail.com>
	<AANLkTiktKMFdEtknKJzC=h5g76HVBPEyOeOchRGdRXMm@mail.gmail.com>
	<AANLkTimKYJWyHP2aUgMOqRvvU-6dPyLNy-+iN7HPf8yS@mail.gmail.com>
	<AANLkTikHHbF-5Rbg4RhrRX9wY6eaZcgLVDK=BBmiQYUR@mail.gmail.com>
Message-ID: <AANLkTinvCSBNew-K=OAqwiSRQEzB4TxQaEgpKiHhFDUb@mail.gmail.com>

Going back to PHP?? Not sure that's a very wise move.
WikiPBX is great, it just needs more people.
If you have a bounty to spend on additional features, it'd be better spent
on wikipbx.



On Thu, Mar 3, 2011 at 7:49 PM, Christian Benke <benkokakao at gmail.com>wrote:

> On 3 March 2011 18:32, Avi Marcus <avi at avimarcus.net> wrote:
> > If this is for business, perhaps you can sponsor it.
>
> Yeah, i'll definitely push for sponsoring the involved OSS-projects
> when we're finished and the project is a commercial success.
> This is the first time this company uses an open source solution and
> i've been hired specifically for this task(One month ago), so i'm
> going to give them a slow introduction to how the OSS-world works ;-)
> If we absolutely need a feature in the GUI _now_ which i can't
> implement, i guess i will convince them to send out a bounty.
>
> I've also not yet told them yet that we're not going to use
> Asterisk(Which they refer to when they talk about "my" project) :-P
>
> Cheers,
> Christian
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Mar  3 21:25:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 3 Mar 2011 10:25:36 -0800
Subject: [Freeswitch-users] route inbound - based on sip account
In-Reply-To: <4D6EACBC.7040509@tuerpe-net.de>
References: <4D6D45F0.5060205@tuerpe-net.de>
	<AANLkTimmuZSR5pPLDd9B6cXjgE9jQ7TDHibWQ_T4Fjx-@mail.gmail.com>
	<4D6EACBC.7040509@tuerpe-net.de>
Message-ID: <AANLkTi=zU0YBsYTwJcf3N1JaKtxxUyygAj7JyRDOGk31@mail.gmail.com>

You've cut out a fair amount of data, but I think I see at least one issue:

<condition
field="${variable_sip_gateway}" expression="sofia/gateway/sip69250">

Do you have a channel variable that is named "variable_sip_gateway"? More
likely you have a channel variable named "sip_gateway". Try rewriting the
above line like this:

<condition field="${sip_gateway}" expression="sip69250$">

The only thing I'm not sure about is whether or not "sip_gateway" is
populated at the time the dialplan is parsed. Without seeing the info dump I
can't offer any other suggestions.

-MC

On Wed, Mar 2, 2011 at 12:46 PM, Andreas Tuerpe <andreas at tuerpe-net.de>wrote:

> Hallo Michael, Johannes and Meftah
>
> here is...
> -------------------------------
> --- start trace log extract ---
>
> INVITE sip:gw+sip69250 at 188.246.xxx.xxx:5080;transport=udp SIP/2.0
> Via: SIP/2.0/UDP 82.139.xxx.x:5060;branch=z9hG4bK2fa2cd41;rport
> From: "+491758301234" <sip:+491758301234 at 82.139.xxx.x>;tag=as20cd750d
> To: <sip:gw+sip69250 at 188.246.xxx.xxx:5080;transport=udp>
> Contact: <sip:+491758301234 at 82.139.xxx.x>
> Call-ID: 2637b63a2411697f7af5f089732a68eb at 82.139.xxx.x
> CSeq: 102 INVITE
> User-Agent: PTY SIPPort
>
> ...
>
> # the only advice to the inbound channel
> # -> 'sip69250' is the 2. gateway name
> [INFO] mod_dialplan_xml.c:252 Processing +491758301234->sip69250 in
> context public
>
>
> Dialplan: sofia/external/+491758301234 at 82.139.xxx.x parsing
> [public->unloop] continue=false
> ...
> # 'public_extensions' is the 1. extension name in public context
> Dialplan: sofia/external/+491758302577 at 82.139.223.1 parsing
> [public->public_extensions] continue=false
> # my test of condition assignment with ${variable_sip_gateway} fails
> Dialplan: ... ... Regex (FAIL) [public_extensions]
> ${variable_sip_gateway}() =~ /sofia/gateway/sip69250/ break=on-false
> ...
> # '4932229982781' is the 2. extension name in public context
> Dialplan: sofia/external/+491758301234 at 82.139.xxx.x parsing
> [public->4932229982781] continue=false
> # empty condition assignment run's without any errors
> Dialplan: sofia/external/+491758301234 at 82.139.xxx.x Regex (PASS)
> [4932229982781] () =~ // break=on-false
> Dialplan: sofia/external/+491758301234 at 82.139.xxx.x Action transfer(1001
> XML default)
> ...
> #all others run's without any errors
> ...
> ---- end trace log extract ----
> -------------------------------
>
>
> # code of first extension name
> <extension name="public_extensions">
>    <condition field="${variable_sip_gateway}"
> expression="sofia/gateway/sip69250">
>        <action application="transfer" data="1004 XML default"/>
>    </condition>
> </extension>
>
> # code of second extension name
> <extension name="4932229982781">
>    <condition field="" expression="">
>        <action application="transfer" data="1001 XML default"/>
>    </condition>
> </extension>
>
>
> I'm unsure with syntax of condition variable esp. expression="?"
>
> Thanks
>    Andreas
>
>
>
> Am 02.03.2011 00:07, schrieb Michael Collins:
> > The FS on pfSense is pretty old, but if all you are working on is a
> > simple routing issue your best bet is to add the "info" app in the
> > public context. Somewhere near the top of public.xml just add this:
> >
> > <extension name="quick info dump" continue="true">
> > <condition>
> > <action application="info"/>
> > </condition>
> > </extension>
> >
> > Save that, press F6 (or do reloadxml) and then make a test inbound call.
> > Watch the console - you'll see a TON of information. Look through the
> > pieces of data that are displayed. You should be able to find something
> > to key off of. Once you've done that then go read up on creating your
> > dialplan here:
> >
> > http://wiki.freeswitch.org/wiki/Dialplan_XML#Condition
> >
> > In fact, that whole page is important in understanding the XML dialplan.
> > I would read it more than once.
> >
> > -MC
> >
> > On Tue, Mar 1, 2011 at 11:16 AM, Andreas Tuerpe <andreas at tuerpe-net.de
> > <mailto:andreas at tuerpe-net.de>> wrote:
> >
> >     Hallo FreeSWITCH Users,
> >
> >     I use FS V.0.9.6 on pfSense
> >     see on -> [http://doc.pfsense.org/index.php/FreeSWITCH]
> >
> >     Symptom:
> >     German ISP "Portunity" forward inbound calls without any
> >     destination_number.
> >
> >
> >     ISP solution tip:
> >     FS to register over a second sip account to the provider twice.
> >     Any account has a separate number.
> >     Based on the channel over which the call come in, I have to decide
> which
> >     number is called.
> >
> >     So I need help, which condition assignment must use - howto ???
> >     - which fields can I use?
> >     - which syntax I have to use?
> >
> >
> >     <!--  here my intention ... ???
> >
> >     <extension name="test_did">
> >     <condition field="channel over which the call come in"
> >     expression="sip-account-1">
> >     <action application="set" data="called_number=111111"/>
> >     <action application="transfer" data="1001 XML default"/>
> >     <condition field="channel over which the call come in"
> >     expression="sip-account-2">
> >     <action application="set" data="called_number=222222"/>
> >     <action application="transfer" data="1002 XML default"/>
> >     </condition>
> >     </extension>
> >
> >     ??? -->
> >
> >     thanks in advance
> >        tuerpean
> >
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From avi at avimarcus.net  Thu Mar  3 21:49:29 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 3 Mar 2011 20:49:29 +0200
Subject: [Freeswitch-users] FS Webinterfaces - Opinions on wikiPBX,
 blue.box, fusionPBX
In-Reply-To: <AANLkTinvCSBNew-K=OAqwiSRQEzB4TxQaEgpKiHhFDUb@mail.gmail.com>
References: <AANLkTik-J2z-P7EvghBH1i-EPL-_G7FqDGX0PCvqYu+g@mail.gmail.com>
	<AANLkTikVTwM3FfsuxYfH5D3HTM5LnYJq9E5eCEkA4LEm@mail.gmail.com>
	<AANLkTiktKMFdEtknKJzC=h5g76HVBPEyOeOchRGdRXMm@mail.gmail.com>
	<AANLkTimKYJWyHP2aUgMOqRvvU-6dPyLNy-+iN7HPf8yS@mail.gmail.com>
	<AANLkTikHHbF-5Rbg4RhrRX9wY6eaZcgLVDK=BBmiQYUR@mail.gmail.com>
	<AANLkTinvCSBNew-K=OAqwiSRQEzB4TxQaEgpKiHhFDUb@mail.gmail.com>
Message-ID: <AANLkTik_+mMCabYhqjLohxiC=WL=8bLhsx8irncL+Gyr@mail.gmail.com>

Does wikipbx have ANY people? I don't think I've ever heard it mentioned on
IRC and as stated before, the last status change is from 11 Feb 2010, 19:54
GMT+0200 - over a year ago.

On Thu, Mar 3, 2011 at 8:22 PM, Aza Tek <azatek0 at gmail.com> wrote:

> Going back to PHP?? Not sure that's a very wise move.
> WikiPBX is great, it just needs more people.
> If you have a bounty to spend on additional features, it'd be better spent
> on wikipbx.
>
>
>
> On Thu, Mar 3, 2011 at 7:49 PM, Christian Benke <benkokakao at gmail.com>wrote:
>
>> On 3 March 2011 18:32, Avi Marcus <avi at avimarcus.net> wrote:
>> > If this is for business, perhaps you can sponsor it.
>>
>> Yeah, i'll definitely push for sponsoring the involved OSS-projects
>> when we're finished and the project is a commercial success.
>> This is the first time this company uses an open source solution and
>> i've been hired specifically for this task(One month ago), so i'm
>> going to give them a slow introduction to how the OSS-world works ;-)
>> If we absolutely need a feature in the GUI _now_ which i can't
>> implement, i guess i will convince them to send out a bounty.
>>
>> I've also not yet told them yet that we're not going to use
>> Asterisk(Which they refer to when they talk about "my" project) :-P
>>
>> Cheers,
>> Christian
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Mar  3 21:45:20 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 3 Mar 2011 10:45:20 -0800
Subject: [Freeswitch-users] FS Webinterfaces - Opinions on wikiPBX,
 blue.box, fusionPBX
In-Reply-To: <AANLkTinvCSBNew-K=OAqwiSRQEzB4TxQaEgpKiHhFDUb@mail.gmail.com>
References: <AANLkTik-J2z-P7EvghBH1i-EPL-_G7FqDGX0PCvqYu+g@mail.gmail.com>
	<AANLkTikVTwM3FfsuxYfH5D3HTM5LnYJq9E5eCEkA4LEm@mail.gmail.com>
	<AANLkTiktKMFdEtknKJzC=h5g76HVBPEyOeOchRGdRXMm@mail.gmail.com>
	<AANLkTimKYJWyHP2aUgMOqRvvU-6dPyLNy-+iN7HPf8yS@mail.gmail.com>
	<AANLkTikHHbF-5Rbg4RhrRX9wY6eaZcgLVDK=BBmiQYUR@mail.gmail.com>
	<AANLkTinvCSBNew-K=OAqwiSRQEzB4TxQaEgpKiHhFDUb@mail.gmail.com>
Message-ID: <AANLkTikaGbQ+u-p=WjWLC0Hbw4PeZs0Aur4K33_Y9Rev@mail.gmail.com>

On Thu, Mar 3, 2011 at 10:22 AM, Aza Tek <azatek0 at gmail.com> wrote:

> Going back to PHP?? Not sure that's a very wise move.
> WikiPBX is great, it just needs more people.
> If you have a bounty to spend on additional features, it'd be better spent
> on wikipbx.
>

Unlikely. WikiPBX has been in existence for 4+ years and has stagnated over
and over again. This is one reason why people have created their own GUIs.
The new projects have momentum and are under active development. They have
long since surpassed WikiPBX in functionality, support, and documentation
even though they are much younger. WikiPBX was the first real effort at a
true FS front-end GUI and it failed to deliver. The site at wikipbx.org has
cobwebs. The most recent forum post is 112 days old. The last announcement
is over a year old.

As far as PHP vs. Python most of us just *shrug*. If you want to see someone
doing stuff with Python then go check out Gabe Gundy's stuff. (
http://gundy.org/)

Of course, the *best* GUI for FreeSWITCH is this one: ;)

-MC
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From lloydie.t at gmail.com  Thu Mar  3 21:58:53 2011
From: lloydie.t at gmail.com (lloyd thomas)
Date: Thu, 3 Mar 2011 18:58:53 +0000
Subject: [Freeswitch-users] Fwd:  Dialplan in a muddle
In-Reply-To: <F05761CEA50FF14094E4F2A86C1517A504D4538025@sha-exch14.shared.ifeltd.com>
References: <AANLkTi=GE8X_FC7UOfxLvjP6EFzttjdsocLcqNNvRA2t@mail.gmail.com>
	<F05761CEA50FF14094E4F2A86C1517A504D453801B@sha-exch14.shared.ifeltd.com>
	<AANLkTimtrZzLUAjK2SG720N46PvvROJah_tWLfj07R6P@mail.gmail.com>
	<F05761CEA50FF14094E4F2A86C1517A504D4538025@sha-exch14.shared.ifeltd.com>
Message-ID: <AANLkTin+DKWPWsANLJCnZOgW0pFRE+cQc2MZt5u92ebu@mail.gmail.com>

No Joy. Looks like the may be something wrong with the initial
companya.tele.co.uk.xml file.
Have a look

http://pastebin.freeswitch.org/15539


---------- Forwarded message ----------
From: Garry Matthews <garry.matthews at ethermagic.co.uk>
Date: 3 March 2011 18:08
Subject: RE: [Freeswitch-users] Dialplan in a muddle
To: lloyd thomas <lloydie.t at gmail.com>


*-* <include>

*-* <gateway name="*soho66*">

* * <param name="*username*" value="*1000014877*" />

* * <param name="*realm*" value="*sip.soho66.co.uk:8060*" />

* * <param name="*password*" value="*C9J6U0LO*" />

* * </gateway>

* * </include>



I?ve made a few changes for security reasons, but it should give you the
idea !



If you type >sofia status



Then you should see REGED to show that you have registered.



Any luck?



Garry.

*From:* lloyd thomas [mailto:lloydie.t at gmail.com]
*Sent:* 03 March 2011 18:02
*To:* Garry Matthews
*Subject:* Re: [Freeswitch-users] Dialplan in a muddle



Ah Ha, what is your gateway settings (roughly)?

On 3 March 2011 17:51, Garry Matthews <garry.matthews at ethermagic.co.uk>
wrote:

I?m not sure if this helps ? I just started with Freeswicth and Soho 66
today. I?m a total beginner and have been working from the book.

However, I can get calls out using the following file -



conf/dialplan/default/01_custom.xml



<include>

                <extension name="Dial Out Custom Gateway">

                <condition field="destination_number"

                                expression="^9(0\d{10})$">

                                <action application="bridge"

                                data="sofia/gateway/soho66/$1"/>

                </condition>

                </extension>

</include>



You then need to -



reloadxml



This may give you something to compare with.



I hope this helps.



Garry.



*From:* lloyd thomas [mailto:lloydie.t at gmail.com]
*Sent:* 03 March 2011 17:04
*To:* freeswitch-users
*Subject:* [Freeswitch-users] Dialplan in a muddle



Maybe getting a bit ahead of myself but I am trying to set up a
multi-company dial plan. Things are working fine internally, but I am not
getting the grasp for external calls.

The first company xml file is conf/dialplan/companya.tele.co.uk.xml

which has the line
<X-PRE-PROCESS cmd="include" data="companya.tele.co.uk/*.xml"/>
to include xml files in that conf/dialplan/companya.tele.co.uk folder

In this folder I have the xml file soho66.co.uk.xml which includes the regex
to remove the initial 9 when dialling and extenal number. I cant seem to be
able to get a call to go anywhere near connecting to the soho gateway.

I think problem is to do with this

2011-03-03 17:00:23.089153 [NOTICE] switch_ivr.c:1606 Transfer
sofia/internal/200 at companya.tele.co.uk to enum[
907973323260 at companya.tele.co.uk]

help me please!

----------------- soho66.co.uk.xml----------------------------
<include>
<extension name="Local Dial">
   <condition field="destination_number" expression="^9?([2-9]{6})$">
     <action application="set" data="effective_caller_id_name=12345678"/>
     <action application="set" data="effective_caller_id_number=12345678"/>
     <action application="bridge" data="sofia/gateway/
sip.soho66.co.uk/01869$1 at sip.soho66.co.uk"/>
   </condition>
</extension>

<extension name="Regional 01">
  <condition field="destination_number" expression="^9?(01\d{9})$">
     <action application="set" data="effective_caller_id_name=12345678"/>
     <action application="set" data="effective_caller_id_number=12345678"/>
     <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1"/>
  </condition>
</extension>

<extension name="Regional 02">
  <condition field="destination_number" expression="^9?(02\d{9})$$">
     <action application="set" data="effective_caller_id_name=12345678"/>
     <action application="set" data="effective_caller_id_number=123456781"/>
     <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1"/>
  </condition>
</extension>

<extension name="Mobile">
  <condition field="destination_number" expression="^9?(07\d{9})$">
     <action application="set" data="effective_caller_id_name=12345678"/>
     <action application="set" data="effective_caller_id_number=12345678"/>
     <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1"/>
  </condition>
</extension>

<extension name="International">
  <condition field="destination_number" expression="^9?(00\d*))$">
     <action application="set" data="effective_caller_id_name=12345678"/>
     <action application="set" data="effective_caller_id_number=12345678"/>
     <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1"/>
  </condition>
</extension>
</include>
-----------------eof soho66.co.uk.xml-------------------------

-------------conf/sip_profiles/external/companya_soho66.xml
-----------------------
<include>
      <gateway name="sip.soho66.co.uk">
      <param name="username" value="12345678"/>
      <param name="from-user" value="12345678"/>
      <param name="password" value="ABCDEFGH"/>
      <param name="expire-seconds" value="600"/>
      <param name="register" value="true"/>
      <param name="retry-seconds" value="30"/>
    </gateway>
</include>
------------- eof conf/sip_profiles/external/companya_soho66.xml
-----------------------
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From lloydie.t at gmail.com  Thu Mar  3 23:06:42 2011
From: lloydie.t at gmail.com (lloyd thomas)
Date: Thu, 3 Mar 2011 20:06:42 +0000
Subject: [Freeswitch-users] Dialplan in a muddle
In-Reply-To: <AANLkTin+DKWPWsANLJCnZOgW0pFRE+cQc2MZt5u92ebu@mail.gmail.com>
References: <AANLkTi=GE8X_FC7UOfxLvjP6EFzttjdsocLcqNNvRA2t@mail.gmail.com>
	<F05761CEA50FF14094E4F2A86C1517A504D453801B@sha-exch14.shared.ifeltd.com>
	<AANLkTimtrZzLUAjK2SG720N46PvvROJah_tWLfj07R6P@mail.gmail.com>
	<F05761CEA50FF14094E4F2A86C1517A504D4538025@sha-exch14.shared.ifeltd.com>
	<AANLkTin+DKWPWsANLJCnZOgW0pFRE+cQc2MZt5u92ebu@mail.gmail.com>
Message-ID: <AANLkTi=41H_=eLX=qu88WqZV4H5vt_unjkevxMiU0-Mw@mail.gmail.com>

EXECUTE sofia/internal/200 at companya.tele.co.ukbridge(sofia/gateway/soho66/07973323260)
2011-03-03 19:57:22.189028 [ERR] mod_sofia.c:3891 Invalid Gateway

So it looks as though my gateway does not register. The file is in
conf/sip_profiles/external/01_companya_soho66.xml
Don't understand what the problem is

Help

<include>
  <gateway name="soho66">
    <param name="username" value="1000022541" />
    <param name="realm" value="sip.soho66.co.uk:8060" />
    <param name="password" value="5J8D139O" />
  </gateway>
</include>


sofia status
                     Name
Type                                       Data      State
=================================================================================================
                 internal       profile
sip:mod_sofia at 188.222.84.32:5060      RUNNING (0)
                 external       profile
sip:mod_sofia at 188.222.84.32:5080      RUNNING (0)
    external::example.com       gateway
sip:joeuser at example.com      NOREG
      companyb.tele.co.uk         alias
internal      ALIASED
            internal-ipv6       profile
sip:mod_sofia@[::1]:5060
RUNNING (0)
    companya.tele.co.uk         alias
internal      ALIASED
            188.222.85.33         alias
internal      ALIASED
=================================================================================================
3 profiles 3 aliases




On 3 March 2011 18:58, lloyd thomas <lloydie.t at gmail.com> wrote:

> No Joy. Looks like the may be something wrong with the initial
> companya.tele.co.uk.xml file.
> Have a look
>
>
> http://pastebin.freeswitch.org/15539
>
>
> ---------- Forwarded message ----------
> From: Garry Matthews <garry.matthews at ethermagic.co.uk>
> Date: 3 March 2011 18:08
> Subject: RE: [Freeswitch-users] Dialplan in a muddle
> To: lloyd thomas <lloydie.t at gmail.com>
>
>
> *-* <include>
>
> *-* <gateway name="*soho66*">
>
> * * <param name="*username*" value="*1000014877*" />
>
> * * <param name="*realm*" value="*sip.soho66.co.uk:8060*" />
>
> * * <param name="*password*" value="*C9J6U0LO*" />
>
> * * </gateway>
>
> * * </include>
>
>
>
> I?ve made a few changes for security reasons, but it should give you the
> idea !
>
>
>
> If you type >sofia status
>
>
>
> Then you should see REGED to show that you have registered.
>
>
>
> Any luck?
>
>
>
> Garry.
>
> *From:* lloyd thomas [mailto:lloydie.t at gmail.com]
> *Sent:* 03 March 2011 18:02
> *To:* Garry Matthews
> *Subject:* Re: [Freeswitch-users] Dialplan in a muddle
>
>
>
> Ah Ha, what is your gateway settings (roughly)?
>
> On 3 March 2011 17:51, Garry Matthews <garry.matthews at ethermagic.co.uk>
> wrote:
>
> I?m not sure if this helps ? I just started with Freeswicth and Soho 66
> today. I?m a total beginner and have been working from the book.
>
> However, I can get calls out using the following file -
>
>
>
> conf/dialplan/default/01_custom.xml
>
>
>
> <include>
>
>                 <extension name="Dial Out Custom Gateway">
>
>                 <condition field="destination_number"
>
>                                 expression="^9(0\d{10})$">
>
>                                 <action application="bridge"
>
>                                 data="sofia/gateway/soho66/$1"/>
>
>                 </condition>
>
>                 </extension>
>
> </include>
>
>
>
> You then need to -
>
>
>
> reloadxml
>
>
>
> This may give you something to compare with.
>
>
>
> I hope this helps.
>
>
>
> Garry.
>
>
>
> *From:* lloyd thomas [mailto:lloydie.t at gmail.com]
> *Sent:* 03 March 2011 17:04
> *To:* freeswitch-users
> *Subject:* [Freeswitch-users] Dialplan in a muddle
>
>
>
> Maybe getting a bit ahead of myself but I am trying to set up a
> multi-company dial plan. Things are working fine internally, but I am not
> getting the grasp for external calls.
>
> The first company xml file is conf/dialplan/companya.tele.co.uk.xml
>
> which has the line
> <X-PRE-PROCESS cmd="include" data="companya.tele.co.uk/*.xml"/>
> to include xml files in that conf/dialplan/companya.tele.co.uk folder
>
> In this folder I have the xml file soho66.co.uk.xml which includes the
> regex to remove the initial 9 when dialling and extenal number. I cant seem
> to be able to get a call to go anywhere near connecting to the soho gateway.
>
> I think problem is to do with this
>
> 2011-03-03 17:00:23.089153 [NOTICE] switch_ivr.c:1606 Transfer
> sofia/internal/200 at companya.tele.co.uk to enum[
> 907973323260 at companya.tele.co.uk]
>
> help me please!
>
> ----------------- soho66.co.uk.xml----------------------------
> <include>
> <extension name="Local Dial">
>    <condition field="destination_number" expression="^9?([2-9]{6})$">
>      <action application="set" data="effective_caller_id_name=12345678"/>
>      <action application="set" data="effective_caller_id_number=12345678"/>
>      <action application="bridge" data="sofia/gateway/
> sip.soho66.co.uk/01869$1 at sip.soho66.co.uk"/>
>    </condition>
> </extension>
>
> <extension name="Regional 01">
>   <condition field="destination_number" expression="^9?(01\d{9})$">
>      <action application="set" data="effective_caller_id_name=12345678"/>
>      <action application="set" data="effective_caller_id_number=12345678"/>
>      <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1
> "/>
>   </condition>
> </extension>
>
> <extension name="Regional 02">
>   <condition field="destination_number" expression="^9?(02\d{9})$$">
>      <action application="set" data="effective_caller_id_name=12345678"/>
>      <action application="set"
> data="effective_caller_id_number=123456781"/>
>      <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1
> "/>
>   </condition>
> </extension>
>
> <extension name="Mobile">
>   <condition field="destination_number" expression="^9?(07\d{9})$">
>      <action application="set" data="effective_caller_id_name=12345678"/>
>      <action application="set" data="effective_caller_id_number=12345678"/>
>      <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1
> "/>
>   </condition>
> </extension>
>
> <extension name="International">
>   <condition field="destination_number" expression="^9?(00\d*))$">
>      <action application="set" data="effective_caller_id_name=12345678"/>
>      <action application="set" data="effective_caller_id_number=12345678"/>
>      <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1
> "/>
>   </condition>
> </extension>
> </include>
> -----------------eof soho66.co.uk.xml-------------------------
>
> -------------conf/sip_profiles/external/companya_soho66.xml
> -----------------------
> <include>
>       <gateway name="sip.soho66.co.uk">
>       <param name="username" value="12345678"/>
>       <param name="from-user" value="12345678"/>
>       <param name="password" value="ABCDEFGH"/>
>       <param name="expire-seconds" value="600"/>
>       <param name="register" value="true"/>
>       <param name="retry-seconds" value="30"/>
>     </gateway>
> </include>
> ------------- eof conf/sip_profiles/external/companya_soho66.xml
> -----------------------
>
>
>
>
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From dnotivol at gmail.com  Thu Mar  3 22:02:29 2011
From: dnotivol at gmail.com (David Notivol)
Date: Thu, 3 Mar 2011 20:02:29 +0100
Subject: [Freeswitch-users] mod_say in different codecs
In-Reply-To: <375B3BB7-DC39-4C80-A32B-DF1941904291@gmail.com>
References: <AANLkTikb_8mv+L1PSqRk-d2QMi1sNSJN-6_5WFmWDrKW@mail.gmail.com>
	<375B3BB7-DC39-4C80-A32B-DF1941904291@gmail.com>
Message-ID: <AANLkTik4b1-iHrdA8cOtzvb6797ZpoYWRA6CyThru6vz@mail.gmail.com>

Thanks Steve for your prompt reply.

Yes, mod_native_file does suit my needs for regular audio files; but I think
it doesn't for numbers. Actually, I'm using say to pronounce numbers, but it
seems mod_say forces FS to use the wav files.

I was having a look at the source code for the mod_say_en, and I removed
some of the ".wav" strings for the digits and recompiled it; and  then the
native_file was triggered and played the .PCMU file (or any other needed
codec) instead of the .wav file.
But I'm not sure this is the way to proceed, or if that can get to other
problems...

Is there any way of having mod_say not forcing to use always .wav files and
relying on mod_native_file ?

--
David

2011/3/3 Steven Ayre <steveayre at gmail.com>

> Look at mod_nativefile - does that suit your needs?
>
> Steve on iPhone
>
> On 3 Mar 2011, at 12:19, David Notivol <dnotivol at gmail.com> wrote:
>
> > Hi all,
> >
> > I'm trying to setup an IVR server, and I'm using the session:say function
> from a LUA script.
> >
> > My question is if there's any way of having mod_say playing audios
> different than audio files; I mean audios encoded in G729, G711ulaw,
> G711alaw, etc. to avoid having FS making transcoding every time I run a say
> command.
> >
> > Checking the folders tree for the sounds, I can see the place for the
> different languages, voices, and wav qualities (8k, 16k...) is clearly
> specified; but is it a way to place files encoded in different codecs?
> >
> > Thanks in advance.
> >
> > Regards,
> > David Notivol
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kbdfck at gmail.com  Fri Mar  4 00:06:31 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Fri, 4 Mar 2011 00:06:31 +0300
Subject: [Freeswitch-users] xml_curl and domain aliases
In-Reply-To: <AANLkTik8NCD4bUsLYt=K16z6HygdQKWjLAzK_fcK6sXV@mail.gmail.com>
References: <AANLkTimS2ZbzaVaVGzgraGjNF5MZd=909VBD3dg7yNSP@mail.gmail.com>
	<AANLkTik8NCD4bUsLYt=K16z6HygdQKWjLAzK_fcK6sXV@mail.gmail.com>
Message-ID: <AANLkTi=BuS+UG4V2f4V4oXp3oERv4bB4svTRcevAOMAc@mail.gmail.com>

Seems I didn't explain clean what I'm trying to achieve.

I have a domain, say, access.test.com. My users can use srv records to reach
it, or specify server's address directly as IP.
Default configuration with domain="all" works like a charm.

Then I moved from static files to xml_curl directory in database. Now every
time user registers I should respond with directory record, containing
domain in which user is trying to register, so I need to respond with IP if
user registers with IP, and with name if user registers with name. I do this
having DomainAlias table in my DB.

After user registration, I need to bridge calls to it by my routing rules.
But I don't know exactly how to instruct freeswitch to lookup user's
contact, since I don't know in which domain user is registered. I don't want
to interact with Freeswitch registration database, as it decreases
flexibility of solution.

At present I use force-register-domain set to my domain name, but I want to
use several names with different aliases, so looking for another solution =)




2011/3/3 Frank Park <frank at telonium.com>

> We use domain aliases to do multi-tenancy and we have SIP clients to
> specify the realm that matches the alias you have, and outbound proxy to
> match the actual proxy server.
> This will use the profile alias to register, (which then you should be able
> to see in the sip_registrations table (in columns sip_host and mwi_host).
> You should then be able to use user/internal/${userpart}@{domain_name}.
>
> As for registering users via IP address, you should look into mod_acl
>
> Frank
>
> On Thu, Mar 3, 2011 at 3:58 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>
>> Hi All
>>
>> I'm trying to setup xml_curl-based directory to provide registration and
>> bridging information from database.
>>
>> The thing I can't still reliably handle is domain aliases. I have some
>> domain like 'access.freeswitch.test', but I need to allow users to register
>> via ip address, for example, 1.1.1.1. My scripts answer on REGISTER curl
>> request with directory entry where domain is set to domain user trying to
>> register at, and this works.
>>
>> But then I need to dial that user. I'm trying to use
>> user/${userpart}%{domainpart} syntax, but I don't know exactly domain user
>> was registered at. How to tell freeswitch to take in account domain aliases
>> when looking up for sofia contact?
>> Or, instead, can I make freeswitch to store user in registration database
>> with some primary domain, which I can then specify to bridge(user/) command?
>>
>> Thanks in advance!
>>
>> --
>> Best regards,
>>
>> Dmitry Sytchev,
>> IT Engineer
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
>
> ----=======================----
> Frank Park
> Telonium Communications, LLC
> frank at telonium.com
> http://www.telonium.com
> Follow Us on Twitter: @GetTelonium
> 404-566-8888 x1001 Office
> 404-939-4242 Cell
> ----=======================----
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From brian at freeswitch.org  Fri Mar  4 00:11:15 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 3 Mar 2011 15:11:15 -0600
Subject: [Freeswitch-users] xml_curl and domain aliases
In-Reply-To: <AANLkTi=BuS+UG4V2f4V4oXp3oERv4bB4svTRcevAOMAc@mail.gmail.com>
References: <AANLkTimS2ZbzaVaVGzgraGjNF5MZd=909VBD3dg7yNSP@mail.gmail.com>
	<AANLkTik8NCD4bUsLYt=K16z6HygdQKWjLAzK_fcK6sXV@mail.gmail.com>
	<AANLkTi=BuS+UG4V2f4V4oXp3oERv4bB4svTRcevAOMAc@mail.gmail.com>
Message-ID: <573EDC16-1C96-4C16-81FD-3DA85CD60BE9@freeswitch.org>

Well you have to make some decisions here... either use domain or IP not either.  So your users should NOT be allowed to register to the IP and only use the DNS name.  Thats my opinion... else you have to dive into the registration db... and it gets messy.

/b

On Mar 3, 2011, at 3:06 PM, Dmitry Sytchev wrote:

> Seems I didn't explain clean what I'm trying to achieve.
> 
> I have a domain, say, access.test.com. My users can use srv records to reach
> it, or specify server's address directly as IP.
> Default configuration with domain="all" works like a charm.
> 
> Then I moved from static files to xml_curl directory in database. Now every
> time user registers I should respond with directory record, containing
> domain in which user is trying to register, so I need to respond with IP if
> user registers with IP, and with name if user registers with name. I do this
> having DomainAlias table in my DB.
> 
> After user registration, I need to bridge calls to it by my routing rules.
> But I don't know exactly how to instruct freeswitch to lookup user's
> contact, since I don't know in which domain user is registered. I don't want
> to interact with Freeswitch registration database, as it decreases
> flexibility of solution.
> 
> At present I use force-register-domain set to my domain name, but I want to
> use several names with different aliases, so looking for another solution =)

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From kbdfck at gmail.com  Fri Mar  4 00:17:26 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Fri, 4 Mar 2011 00:17:26 +0300
Subject: [Freeswitch-users] xml_curl and domain aliases
In-Reply-To: <573EDC16-1C96-4C16-81FD-3DA85CD60BE9@freeswitch.org>
References: <AANLkTimS2ZbzaVaVGzgraGjNF5MZd=909VBD3dg7yNSP@mail.gmail.com>
	<AANLkTik8NCD4bUsLYt=K16z6HygdQKWjLAzK_fcK6sXV@mail.gmail.com>
	<AANLkTi=BuS+UG4V2f4V4oXp3oERv4bB4svTRcevAOMAc@mail.gmail.com>
	<573EDC16-1C96-4C16-81FD-3DA85CD60BE9@freeswitch.org>
Message-ID: <AANLkTi=bBKaxuTcLLxQOSEoK3A+6a68J+1jSKrqOuXcH@mail.gmail.com>

There could be an option to force registration domain which I can return
with data retrieved with xml_curl ;)
I know exactly which user is trying to register, all information is linked
in database, but I can't find him later when I need to create bridge :)

2011/3/4 Brian West <brian at freeswitch.org>

> Well you have to make some decisions here... either use domain or IP not
> either.  So your users should NOT be allowed to register to the IP and only
> use the DNS name.  Thats my opinion... else you have to dive into the
> registration db... and it gets messy.
>
> /b
>
> On Mar 3, 2011, at 3:06 PM, Dmitry Sytchev wrote:
>
> Seems I didn't explain clean what I'm trying to achieve.
>
> I have a domain, say, access.test.com. My users can use srv records to
> reach
> it, or specify server's address directly as IP.
> Default configuration with domain="all" works like a charm.
>
> Then I moved from static files to xml_curl directory in database. Now every
> time user registers I should respond with directory record, containing
> domain in which user is trying to register, so I need to respond with IP if
> user registers with IP, and with name if user registers with name. I do
> this
> having DomainAlias table in my DB.
>
> After user registration, I need to bridge calls to it by my routing rules.
> But I don't know exactly how to instruct freeswitch to lookup user's
> contact, since I don't know in which domain user is registered. I don't
> want
> to interact with Freeswitch registration database, as it decreases
> flexibility of solution.
>
> At present I use force-register-domain set to my domain name, but I want to
> use several names with different aliases, so looking for another solution
> =)
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From ce at kapper.net  Fri Mar  4 00:21:07 2011
From: ce at kapper.net (Clemens Ebentheuer)
Date: Thu, 3 Mar 2011 22:21:07 +0100
Subject: [Freeswitch-users] Dialplan in a muddle
In-Reply-To: <AANLkTi=41H_=eLX=qu88WqZV4H5vt_unjkevxMiU0-Mw@mail.gmail.com>
References: <AANLkTi=GE8X_FC7UOfxLvjP6EFzttjdsocLcqNNvRA2t@mail.gmail.com>
	<F05761CEA50FF14094E4F2A86C1517A504D453801B@sha-exch14.shared.ifeltd.com>
	<AANLkTimtrZzLUAjK2SG720N46PvvROJah_tWLfj07R6P@mail.gmail.com>
	<F05761CEA50FF14094E4F2A86C1517A504D4538025@sha-exch14.shared.ifeltd.com>
	<AANLkTin+DKWPWsANLJCnZOgW0pFRE+cQc2MZt5u92ebu@mail.gmail.com>
	<AANLkTi=41H_=eLX=qu88WqZV4H5vt_unjkevxMiU0-Mw@mail.gmail.com>
Message-ID: <1B19ABD72889C245AE8EEE08AC24103A28C4231FB5@exmachina.office.kapper.net>

Hi,

i?m no expert but you should see something like in sofia status:

external::soho66       gateway       sip:1000022541 at sip.soho66.co.uk:8060      TRYING (retry: 22s)

maybe you need a:  sofia profile external rescan
and/or: sofia profile external restart

clemens

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of lloyd thomas
Sent: Thursday, March 03, 2011 9:07 PM
To: freeswitch-users
Subject: Re: [Freeswitch-users] Dialplan in a muddle

EXECUTE sofia/internal/200 at companya.tele.co.uk<mailto:200 at companya.tele.co.uk> bridge(sofia/gateway/soho66/07973323260)
2011-03-03 19:57:22.189028 [ERR] mod_sofia.c:3891 Invalid Gateway

So it looks as though my gateway does not register. The file is in conf/sip_profiles/external/01_companya_soho66.xml
Don't understand what the problem is

Help

<include>
  <gateway name="soho66">
    <param name="username" value="1000022541" />
    <param name="realm" value="sip.soho66.co.uk:8060<http://sip.soho66.co.uk:8060>" />
    <param name="password" value="5J8D139O" />
  </gateway>
</include>


sofia status
                     Name          Type                                       Data      State
=================================================================================================
                 internal       profile           sip:mod_sofia at 188.222.84.32:5060<http://sip:mod_sofia at 188.222.84.32:5060>      RUNNING (0)
                 external       profile           sip:mod_sofia at 188.222.84.32:5080<http://sip:mod_sofia at 188.222.84.32:5080>      RUNNING (0)
    external::example.com<http://example.com>       gateway                    sip:joeuser at example.com<mailto:sip%3Ajoeuser at example.com>      NOREG
      companyb.tele.co.uk<http://companyb.tele.co.uk>         alias                                   internal      ALIASED
            internal-ipv6       profile                   sip:mod_sofia@[::1]:5060      RUNNING (0)
    companya.tele.co.uk<http://companya.tele.co.uk>         alias                                   internal      ALIASED
            188.222.85.33         alias                                   internal      ALIASED
=================================================================================================
3 profiles 3 aliases



On 3 March 2011 18:58, lloyd thomas <lloydie.t at gmail.com<mailto:lloydie.t at gmail.com>> wrote:
No Joy. Looks like the may be something wrong with the initial companya.tele.co.uk.xml file.
Have a look


http://pastebin.freeswitch.org/15539

---------- Forwarded message ----------
From: Garry Matthews <garry.matthews at ethermagic.co.uk<mailto:garry.matthews at ethermagic.co.uk>>
Date: 3 March 2011 18:08
Subject: RE: [Freeswitch-users] Dialplan in a muddle
To: lloyd thomas <lloydie.t at gmail.com<mailto:lloydie.t at gmail.com>>

- <include>
- <gateway name="soho66">
  <param name="username" value="1000014877" />
  <param name="realm" value="sip.soho66.co.uk:8060<http://sip.soho66.co.uk:8060>" />
  <param name="password" value="C9J6U0LO" />
  </gateway>
  </include>

I've made a few changes for security reasons, but it should give you the idea !

If you type >sofia status

Then you should see REGED to show that you have registered.

Any luck?

Garry.
From: lloyd thomas [mailto:lloydie.t at gmail.com<mailto:lloydie.t at gmail.com>]
Sent: 03 March 2011 18:02
To: Garry Matthews
Subject: Re: [Freeswitch-users] Dialplan in a muddle

Ah Ha, what is your gateway settings (roughly)?
On 3 March 2011 17:51, Garry Matthews <garry.matthews at ethermagic.co.uk<mailto:garry.matthews at ethermagic.co.uk>> wrote:
I'm not sure if this helps - I just started with Freeswicth and Soho 66 today. I'm a total beginner and have been working from the book.
However, I can get calls out using the following file -

conf/dialplan/default/01_custom.xml

<include>
                <extension name="Dial Out Custom Gateway">
                <condition field="destination_number"
                                expression="^9(0\d{10})$">
                                <action application="bridge"
                                data="sofia/gateway/soho66/$1"/>
                </condition>
                </extension>
</include>

You then need to -

reloadxml

This may give you something to compare with.

I hope this helps.

Garry.

From: lloyd thomas [mailto:lloydie.t at gmail.com<mailto:lloydie.t at gmail.com>]
Sent: 03 March 2011 17:04
To: freeswitch-users
Subject: [Freeswitch-users] Dialplan in a muddle

Maybe getting a bit ahead of myself but I am trying to set up a multi-company dial plan. Things are working fine internally, but I am not getting the grasp for external calls.

The first company xml file is conf/dialplan/companya.tele.co.uk.xml

which has the line
<X-PRE-PROCESS cmd="include" data="companya.tele.co.uk/*.xml<http://companya.tele.co.uk/*.xml>"/>
to include xml files in that conf/dialplan/companya.tele.co.uk<http://companya.tele.co.uk> folder

In this folder I have the xml file soho66.co.uk.xml which includes the regex to remove the initial 9 when dialling and extenal number. I cant seem to be able to get a call to go anywhere near connecting to the soho gateway.

I think problem is to do with this

2011-03-03 17:00:23.089153 [NOTICE] switch_ivr.c:1606 Transfer sofia/internal/200 at companya.tele.co.uk<mailto:200 at companya.tele.co.uk> to enum[907973323260 at companya.tele.co.uk<mailto:907973323260 at companya.tele.co.uk>]

help me please!

----------------- soho66.co.uk.xml----------------------------
<include>
<extension name="Local Dial">
   <condition field="destination_number" expression="^9?([2-9]{6})$">
     <action application="set" data="effective_caller_id_name=12345678"/>
     <action application="set" data="effective_caller_id_number=12345678"/>
     <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/01869$1 at sip.soho66.co.uk<http://sip.soho66.co.uk/01869$1 at sip.soho66.co.uk>"/>
   </condition>
</extension>

<extension name="Regional 01">
  <condition field="destination_number" expression="^9?(01\d{9})$">
     <action application="set" data="effective_caller_id_name=12345678"/>
     <action application="set" data="effective_caller_id_number=12345678"/>
     <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1<http://sip.soho66.co.uk/$1>"/>
  </condition>
</extension>

<extension name="Regional 02">
  <condition field="destination_number" expression="^9?(02\d{9})$$">
     <action application="set" data="effective_caller_id_name=12345678"/>
     <action application="set" data="effective_caller_id_number=123456781"/>
     <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1<http://sip.soho66.co.uk/$1>"/>
  </condition>
</extension>

<extension name="Mobile">
  <condition field="destination_number" expression="^9?(07\d{9})$">
     <action application="set" data="effective_caller_id_name=12345678"/>
     <action application="set" data="effective_caller_id_number=12345678"/>
     <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1<http://sip.soho66.co.uk/$1>"/>
  </condition>
</extension>

<extension name="International">
  <condition field="destination_number" expression="^9?(00\d*))$">
     <action application="set" data="effective_caller_id_name=12345678"/>
     <action application="set" data="effective_caller_id_number=12345678"/>
     <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1<http://sip.soho66.co.uk/$1>"/>
  </condition>
</extension>
</include>
-----------------eof soho66.co.uk.xml-------------------------

-------------conf/sip_profiles/external/companya_soho66.xml -----------------------
<include>
      <gateway name="sip.soho66.co.uk<http://sip.soho66.co.uk>">
      <param name="username" value="12345678"/>
      <param name="from-user" value="12345678"/>
      <param name="password" value="ABCDEFGH"/>
      <param name="expire-seconds" value="600"/>
      <param name="register" value="true"/>
      <param name="retry-seconds" value="30"/>
    </gateway>
</include>
------------- eof conf/sip_profiles/external/companya_soho66.xml -----------------------



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From kris at livecall.com  Fri Mar  4 00:38:11 2011
From: kris at livecall.com (Kris)
Date: Thu, 3 Mar 2011 13:38:11 -0800
Subject: [Freeswitch-users] Conference - control actions problem
References: <AANLkTimq9dYbq8ztbQ5dJmY2Y+2mK5mKzM78UirEzzYn@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C58B2BBAF1D@cooper><AANLkTikPa1E9o3-NgWBjd7P8HWfr1vRxAnh+AH0UpqW1@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C58B2BBB2F5@cooper><AANLkTim=VDq1ztB7Bn=_E0eq_8KeRnZceLapmT8jz2DS@mail.gmail.com><265B34F688F74462B0B6B54969AFB913@stor1>
	<AANLkTinJD756oM8cQDi+=nPPjYNXH8Qid4DVygMbv_Xp@mail.gmail.com>
Message-ID: <7A8E28E22D9B41DFAD4E17B869EA074A@stor1>

I am not sure..I think I did an update around 1/24 which was working then 
about 2/19 and 2/28, but my own code was butchered at the time and I thought 
the git updates were also butchering FS, that's why I did the clean clone 
about midnight. Sometime1/24 to now...
It seems if I let the prompt finish and go into GetDigits, the digit comes 
through. If the digit is pressed during play is when the callback is not 
called by FS--BUT...if I press really fast..2-3 times a second, I get the 
digit most likely because it goes into GetDigit().
I will probably forget about the callback and try to get the digit by 
subscribing to the DTMF event.
----- Original Message ----- 
From: "Michael Collins" <msc at freeswitch.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, March 03, 2011 10:16 AM
Subject: Re: [Freeswitch-users] Conference - control actions problem


> On Thu, Mar 3, 2011 at 4:56 AM, Kris <kris at livecall.com> wrote:
>
>> I am also having strange problem with DTMFs after transfering last member
>> out of conference back to my C# managed App- no more DTMFCallback to 
>> detect
>> digits, but only on the last member out of the conf.. Seems like the last
>> one keeps getting it's DTMF events trapped by mod_conference or FS. I 
>> just
>> did a clean GIT clone from scratch a few hours ago. The transfer action
>> control also hangs up the last caller as it has always done, so I have 
>> been
>> using the Api. Something must have changed in the git in the past 1-2
>> weeks.
>>
>
> Please find the last known git release where it worked and/or the first 
> git
> release where it broke. Then open a Jira and supply a sample
> config/script/etc. that will allow the devs to recreate this issue.
>
> -MC
> 



From aligzaidi at gmail.com  Fri Mar  4 00:44:12 2011
From: aligzaidi at gmail.com (Ali Zaidi)
Date: Thu, 3 Mar 2011 16:44:12 -0500
Subject: [Freeswitch-users] Dialing extensions behind NAT
Message-ID: <AANLkTik_Q9DsUNeb7ogkgkt5ykSWv+4rE98oZu_OtRdv@mail.gmail.com>

Hi,



Not sure if this was previously asked, my scenario as following.



UA1,UA2,UA3===(NAT) ====(Internet)====(FreeSwitch)===(PSTN)



When UA1 dial UA2 the traffic go over the internet connection to FreeSwitch
and then comes back to UA2. Can UA1 dial UA2 without going to internet?

My both phones are Linksys SPA942, please let me know any good phones that
can work best with free switch over wan connection.



Thanks,



AGZ
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From brian at freeswitch.org  Fri Mar  4 01:34:05 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 3 Mar 2011 16:34:05 -0600
Subject: [Freeswitch-users] Snom and Aastra one-way audio after picking
	up call that has been on hold (NAT)
In-Reply-To: <4D6F596C.7010403@estation.dk>
References: <4D6F596C.7010403@estation.dk>
Message-ID: <30915FCC-D7B4-4BEC-850F-BA321615A5D6@freeswitch.org>

sip trace please.

/b

On Mar 3, 2011, at 3:03 AM, ?yvind Albrigtsen wrote:

> 
> The calls get in like this:
> PSTN-gw -> FS -> NAT -> Snom/Aastra
> 
> It works fine when I call out from the phones and set the call on hold 
> (on the Snom/Aastra phones), though.
> 
> This is on FreeSWITCH Version 1.0.head (git-cb6f1ed 2011-02-22 20-25-16 
> -0500)
> 
> 
> Regards
> Oyvind



From jjj at 3js.de  Fri Mar  4 01:50:29 2011
From: jjj at 3js.de (Johannes Jakob)
Date: Thu, 3 Mar 2011 23:50:29 +0100
Subject: [Freeswitch-users] DTMF not passed through to client
Message-ID: <BFA68905-4F2E-4782-95CE-7C0D2CB01328@3js.de>

Hi,

what do I have to do to get RFC2833 DTMF passed to the client in an inbound scenario like this:

Upstream >rfc2833> FreeSWITCH >rfc2833> Asterisk

?

The Upstream is sending the DTMF tones correctly:

---
329	2.114845	UPSTREAMIP	FS_IP	RTP EVENT	62	Payload type=RTP Event, DTMF Eight 8
---

FreeSWITCH isn't logging much of the dtmf stuff, even with loglevel debug:

http://pastebin.freeswitch.org/15541

...
Set 2833 dtmf receive payload to 101
...
2011-03-03 23:29:59.498466 [ERR] switch_rtp.c:282 Failed DTMF sanity check.
...


User's directory entry includes:

      <param name="dtmf-type" value="rfc2833"/>
      <param name="pass-rfc2833" value="true"/>

I tried adding those in the gateway profile as well, no change.


On the asterisk box I don't see those rtpevents in the pcap trace anymore, they just vanish on the FS box.


Can somebody help me out here? Or point me to the right wiki page?
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#DTMF doesn't really tell me, what the options do and http://wiki.freeswitch.org/wiki/DTMF is something different.


Thanks and bye,

  Johannes

From brian at freeswitch.org  Fri Mar  4 01:58:36 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 3 Mar 2011 16:58:36 -0600
Subject: [Freeswitch-users] Dialing extensions behind NAT
In-Reply-To: <AANLkTik_Q9DsUNeb7ogkgkt5ykSWv+4rE98oZu_OtRdv@mail.gmail.com>
References: <AANLkTik_Q9DsUNeb7ogkgkt5ykSWv+4rE98oZu_OtRdv@mail.gmail.com>
Message-ID: <FBFB4BE1-4856-4C93-8CE4-044D051AF6C6@freeswitch.org>

give each phone a different client side sip port.

/b

On Mar 3, 2011, at 3:44 PM, Ali Zaidi wrote:

> My both phones are Linksys SPA942, please let me know any good phones that
> can work best with free switch over wan connection.



From avi at avimarcus.net  Fri Mar  4 02:45:02 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 4 Mar 2011 01:45:02 +0200
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
Message-ID: <AANLkTim0vpXhg16p6LTwbYa7ng5Ki-5-aw=MpfZOT3pY@mail.gmail.com>

Hi - I didn't notice this in my latest git contrib pull today. Did you get
the access worked out?
-Avi

On Sun, Feb 27, 2011 at 10:06 PM, Saeed Ahmed <saeedahmad1981 at gmail.com>wrote:

> press sent too quick..
>
> what did you use for routing? curl? esl?
>
> did you use nibble bill for prepaid app?
>
>
> On Sun, Feb 27, 2011 at 9:05 PM, Saeed Ahmed <saeedahmad1981 at gmail.com>wrote:
>
>> Great!
>>
>> want to see it soon.
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ayhkor at gmail.com  Fri Mar  4 07:08:15 2011
From: ayhkor at gmail.com (deniro)
Date: Thu, 3 Mar 2011 23:08:15 -0500
Subject: [Freeswitch-users] fs_cli socket connection error
In-Reply-To: <AANLkTim8HxbSxoVnfFHWmG5WxdWsvHF4v7rX-ZonuW6k@mail.gmail.com>
References: <AANLkTikCoT9dzLNoD=6TzS0jiMw4afNX94TypsLarBbz@mail.gmail.com>
	<AANLkTi=sENgBB9Cdr-EkLJD8XfmH3A8GSfPnvDHO7qrw@mail.gmail.com>
	<AANLkTin5k=2Z-=QVVzqt91172wv88P7OFkqRqTY9VDcS@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58ADB727D7@cooper>
	<AANLkTi=aPwmnVD8ZzaBVFDxGyPsbgzDEBGaXvq_B2NbF@mail.gmail.com>
	<AANLkTikpQuxvyaU7PAGnyJuoQLQ9xn4inNJ_u5x41Anv@mail.gmail.com>
	<AANLkTimNYjC_G-Lq+St5VePpxEyM8iHnhaNCo=jBydi9@mail.gmail.com>
	<AANLkTi=QXgEb6dHU9bfb7v06WzPyWJcoNYDLi1X9oq+E@mail.gmail.com>
	<AANLkTikEDgpAJppNUnLac73ip7qsk+8AK-N-pTW7NBm6@mail.gmail.com>
	<AANLkTi=NP_Kw-Hrb6+XXtuid3YWJUM2ehFaeoaV-UEaC@mail.gmail.com>
	<AANLkTinQ7g2YFCGNRtxWr-apHuByNFdF3Z_ph8J7K01e@mail.gmail.com>
	<AANLkTin_Y-azNb2sT-U9HQsjVbL3JqPiBamTiX5a_HC+@mail.gmail.com>
	<AANLkTi=d18FUPX_mWDO4eLEW5Uf+PYFfmtLfZ9C=LTKR@mail.gmail.com>
	<AANLkTim8HxbSxoVnfFHWmG5WxdWsvHF4v7rX-ZonuW6k@mail.gmail.com>
Message-ID: <AANLkTi=0xgVmox-Px1QSu27h6uxSStTRRWKbn2Utsmfs@mail.gmail.com>

Alright,  I did some googling  around  "[ERR] mod_xml_curl.c:444 Binding has
no url "  then
uncommented below line in conf/autoload_configs/xml_curl.conf.xml
<param name="gateway-url" value="http://www.freeswitch.org/gateway.xml"
bindings="dialplan"/>
though I don't know what this is for..

Now I am able to load mod_xml_curl
no need to re-install everything
thanks everyone
deniro--


On Thu, Mar 3, 2011 at 3:32 AM, Steven Ayre <steveayre at gmail.com> wrote:

> freeswitch at xxx@internal> load mod_xml_curl
> -ERR [module load file routine returned an error]
>
> There's still an error while loading. You need to look at the logs for
> freeswitch, not the fs_cli ones.
>
> Either run 'fs_cli -l debug' or do '/log 7' within fs_cli
>
> -Steve
>
> On 3 March 2011 00:28, deniro <ayhkor at gmail.com> wrote:
> > ok, I see one under src. I copied that to conf/autoload_configs and looks
> > like I am able to load mod_xml_cdr this way
> > freeswitch at xxxx@internal>
> > load mod_xml_cdr
> > module_exists mod_xml_cdr
> > true
> > same method did not work for mod_xml_curl
> > below is error message after I copied file
> > cp /usr/src/freeswitch-1.0.6/conf/autoload_configs/xml_curl.conf.xml
> > /opt/freeswitch/conf/autoload_configs/xml_curl.conf.xml
> >
> > freeswitch at xxx@internal> load mod_xml_curl
> > [DEBUG] libs/esl/src/esl.c:1140 esl_send() SEND
> > api load mod_xml_curl
> > console_execute: true
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER
> [Content-Type] =
> > [api/response]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER
> [Content-Length]
> > = [68]
> > +OK Reloading XML
> > -ERR [module load file routine returned an error]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER
> [Content-Type] =
> > [log/data]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER
> [Content-Length]
> > = [63]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Level] =
> > [6]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER
> [Text-Channel] =
> > [0]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-File] =
> > [mod_enum.c]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Func] =
> > [event_handler]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Line] =
> > [808]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [User-Data] =
> []
> > [DEBUG] libs/esl/src/esl.c:1114 esl_recv_event() RECV MESSAGE
> > Event-Name: SOCKET_DATA
> > Content-Type: log/data
> > Content-Length: 63
> > Log-Level: 6
> > Text-Channel: 0
> > Log-File: mod_enum.c
> > Log-Func: event_handler
> > Log-Line: 808
> > User-Data: _undef_
> > 2011-03-02 19:23:34.416231 [INFO] mod_enum.c:808 ENUM Reloaded
> > 2011-03-02 19:23:34.416231 [INFO] mod_enum.c:808 ENUM Reloaded
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER
> [Content-Type] =
> > [log/data]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER
> [Content-Length]
> > = [72]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Level] =
> > [3]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER
> [Text-Channel] =
> > [0]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-File] =
> > [mod_xml_curl.c]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Func] =
> > [do_config]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Line] =
> > [444]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [User-Data] =
> []
> > [DEBUG] libs/esl/src/esl.c:1114 esl_recv_event() RECV MESSAGE
> > Event-Name: SOCKET_DATA
> > Content-Type: log/data
> > Content-Length: 72
> > Log-Level: 3
> > Text-Channel: 0
> > Log-File: mod_xml_curl.c
> > Log-Func: do_config
> > Log-Line: 444
> > User-Data: _undef_
> > 2011-03-02 19:23:34.416231 [ERR] mod_xml_curl.c:444 Binding has no url!
> > 2011-03-02 19:23:34.416231 [ERR] mod_xml_curl.c:444 Binding has no url!
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER
> [Content-Type] =
> > [log/data]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER
> [Content-Length]
> > = [162]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Level] =
> > [2]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER
> [Text-Channel] =
> > [0]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-File] =
> > [switch_loadable_module.c]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Func] =
> > [switch_loadable_module_load_file]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Line] =
> > [926]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [User-Data] =
> []
> > [DEBUG] libs/esl/src/esl.c:1114 esl_recv_event() RECV MESSAGE
> > Event-Name: SOCKET_DATA
> > Content-Type: log/data
> > Content-Length: 162
> > Log-Level: 2
> > Text-Channel: 0
> > Log-File: switch_loadable_module.c
> > Log-Func: switch_loadable_module_load_file
> > Log-Line: 926
> > User-Data: _undef_
> > 2011-03-02 19:23:34.416231 [CRIT] switch_loadable_module.c:926 Error
> Loading
> > module /opt/freeswitch/mod/mod_xml_curl.so
> > **Module load routine returned an error**
> > 2011-03-02 19:23:34.416231 [CRIT] switch_loadable_module.c:926 Error
> Loading
> > module /opt/freeswitch/mod/mod_xml_curl.so
> > **Module load routine returned an error**
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER
> [Content-Type] =
> > [log/data]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER
> [Content-Length]
> > = [86]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Level] =
> > [6]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER
> [Text-Channel] =
> > [0]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-File] =
> > [switch_time.c]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Func] =
> > [switch_load_timezones]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Line] =
> > [950]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [User-Data] =
> []
> > [DEBUG] libs/esl/src/esl.c:1114 esl_recv_event() RECV MESSAGE
> > Event-Name: SOCKET_DATA
> > Content-Type: log/data
> > Content-Length: 86
> > Log-Level: 6
> > Text-Channel: 0
> > Log-File: switch_time.c
> > Log-Func: switch_load_timezones
> > Log-Line: 950
> > User-Data: _undef_
> > 2011-03-02 19:23:34.417121 [INFO] switch_time.c:950 Timezone reloaded 530
> > definitions
> > 2011-03-02 19:23:34.417121 [INFO] switch_time.c:950 Timezone reloaded 530
> > definitions
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > On Wed, Mar 2, 2011 at 6:25 PM, Steven Ayre <steveayre at gmail.com> wrote:
> >>
> >> You need to create its config file. It won't load without it.
> >>
> >> "find . -name xml_cdr.conf"
> >>
> >> Actually it's xml_cdr.conf.xml, all config files will show .conf as
> >> their logical name but on the hard drive they're stored in .conf.xml.
> >> It should be in conf/autoload_configs
> >>
> >> -Steve
> >>
> >>
> >> On 2 March 2011 22:25, deniro <ayhkor at gmail.com> wrote:
> >> > I typed the 4 commands but still unable to load mod_xml_cdr and
> >> > mod_xml_curl
> >> >
> >> >
> >> > 2011-03-02 17:21:32.139672 [ERR] mod_xml_cdr.c:479 Open of
> xml_cdr.conf
> >> > failed
> >> > /opt# find . -name xml_cdr.conf
> >> > nothing appears
> >> >
> >> > here is errors from fs_cli -d 7
> >> >
> >> > freeswitch at 75.xxx.xxx.xx@internal> load mod_xml_cdr
> >> > [DEBUG] libs/esl/src/esl.c:1063 esl_send() SEND
> >> > api load mod_xml_cdr
> >> > console_execute: true
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> >> > [Content-Type] =
> >> > [api/response]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> >> > [Content-Length]
> >> > = [50]
> >> > -ERR [module load file routine returned an error]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> >> > [Content-Type] =
> >> > [log/data]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> >> > [Content-Length]
> >> > = [79]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Log-Level]
> >> > =
> >> > [3]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> >> > [Text-Channel] =
> >> > [0]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File]
> =
> >> > [mod_xml_cdr.c]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func]
> =
> >> > [mod_xml_cdr_load]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line]
> =
> >> > [479]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [User-Data]
> >> > = []
> >> > [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> >> > Event-Name: SOCKET_DATA
> >> > Content-Type: log/data
> >> > Content-Length: 79
> >> > Log-Level: 3
> >> > Text-Channel: 0
> >> > Log-File: mod_xml_cdr.c
> >> > Log-Func: mod_xml_cdr_load
> >> > Log-Line: 479
> >> > User-Data: _undef_
> >> > 2011-03-02 17:21:32.139672 [ERR] mod_xml_cdr.c:479 Open of
> xml_cdr.conf
> >> > failed
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> >> > [Content-Type] =
> >> > [log/data]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> >> > [Content-Length]
> >> > = [161]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Log-Level]
> >> > =
> >> > [2]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> >> > [Text-Channel] =
> >> > [0]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File]
> =
> >> > [switch_loadable_module.c]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func]
> =
> >> > [switch_loadable_module_load_file]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line]
> =
> >> > [882]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [User-Data]
> >> > = []
> >> > [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> >> > Event-Name: SOCKET_DATA
> >> > Content-Type: log/data
> >> > Content-Length: 161
> >> > Log-Level: 2
> >> > Text-Channel: 0
> >> > Log-File: switch_loadable_module.c
> >> > Log-Func: switch_loadable_module_load_file
> >> > Log-Line: 882
> >> > User-Data: _undef_
> >> > 2011-03-02 17:21:32.139672 [CRIT] switch_loadable_module.c:882 Error
> >> > Loading
> >> > module /opt/freeswitch/mod/mod_xml_cdr.so
> >> > **Module load routine returned an error**
> >> >
> >> >
> >> >
> >> >
> >> >
> >> > On Tue, Mar 1, 2011 at 1:11 PM, Anthony Minessale
> >> > <anthony.minessale at gmail.com> wrote:
> >> >>
> >> >> like I said you re-installed with a new version of libcurl but did
> not
> >> >> fully rebuild.
> >> >> So most of the modules that depend on curl were broken torn between
> >> >> the old and new version it was linked to..
> >> >>
> >> >> had you executed the 4 commands I posted yesterday it would be fine.
> >> >>
> >> >>
> >> >> On Tue, Mar 1, 2011 at 11:02 AM, deniro <ayhkor at gmail.com> wrote:
> >> >> > Thank you all for advice  and direction
> >> >> > Will re-install from scratch at a later time.
> >> >> > deniro--
> >> >> >
> >> >> > On Tue, Mar 1, 2011 at 4:16 AM, Steven Ayre <steveayre at gmail.com>
> >> >> > wrote:
> >> >> >>
> >> >> >> Ok, you really need to pay attention to any log message that logs
> >> >> >> with
> >> >> >> the CRIT (critical) log level - they're really serious errors. You
> >> >> >> should look at adding the critical="true" attribute to all the
> tags
> >> >> >> in
> >> >> >> modules.conf.xml which you want to be sure are loaded. If a module
> >> >> >> fails to load without that tag FS will continue to run anyway,
> with
> >> >> >> that tag it'll refuse to start so you'll know instantly something
> is
> >> >> >> wrong.
> >> >> >>
> >> >> >> Most of your modules aren't loading, because they haven't been
> >> >> >> compiled correctly. Some look like they're missing (no such file
> or
> >> >> >> directory) and others haven't been built right (undefined
> symbols).
> >> >> >>
> >> >> >> Here are a few of the important lines from your log:
> >> >> >>
> >> >> >> 2011-02-28 18:12:21.910738 [CRIT] switch_loadable_module.c:882
> Error
> >> >> >> Loading module /opt/freeswitch/mod/mod_event_socket.so
> >> >> >> **/opt/freeswitch/mod/mod_event_socket.so: undefined symbol:
> >> >> >> switch_event_serialize_json**
> >> >> >> 2011-02-28 18:12:21.910789 [CRIT] switch_loadable_module.c:882
> Error
> >> >> >> Loading module /opt/freeswitch/mod/mod_xml_curl.so
> >> >> >> **/opt/freeswitch/mod/mod_xml_curl.so: cannot open shared object
> >> >> >> file:
> >> >> >> No such file or directory**
> >> >> >> 2011-02-28 18:12:21.910813 [CRIT] switch_loadable_module.c:882
> Error
> >> >> >> Loading module /opt/freeswitch/mod/mod_xml_cdr.so
> >> >> >> **/opt/freeswitch/mod/mod_xml_cdr.so: cannot open shared object
> >> >> >> file:
> >> >> >> No such file or directory**
> >> >> >> 2011-02-28 18:12:21.911581 [CRIT] switch_loadable_module.c:882
> Error
> >> >> >> Loading module /opt/freeswitch/mod/mod_commands.so
> >> >> >> **/opt/freeswitch/mod/mod_commands.so: undefined symbol:
> >> >> >> switch_xml_reload**
> >> >> >> 2011-02-28 18:12:21.911709 [CRIT] switch_loadable_module.c:882
> Error
> >> >> >> Loading module /opt/freeswitch/mod/mod_conference.so
> >> >> >> **/opt/freeswitch/mod/mod_conference.so: undefined symbol:
> >> >> >> switch_channel_test_app_flag_key**
> >> >> >> 2011-02-28 18:12:21.911842 [CRIT] switch_loadable_module.c:882
> Error
> >> >> >> Loading module /opt/freeswitch/mod/mod_dptools.so
> >> >> >> **/opt/freeswitch/mod/mod_dptools.so: undefined symbol:
> >> >> >> switch_event_merge**
> >> >> >> 2011-02-28 18:12:21.912264 [CRIT] switch_loadable_module.c:882
> Error
> >> >> >> Loading module /opt/freeswitch/mod/mod_dialplan_xml.so
> >> >> >> **/opt/freeswitch/mod/mod_dialplan_xml.so: undefined symbol:
> >> >> >> switch_xml_std_datetime_check**
> >> >> >> 2011-02-28 18:12:21.912531 [CRIT] switch_loadable_module.c:882
> Error
> >> >> >> Loading module /opt/freeswitch/mod/mod_g723_1.so
> >> >> >> **/opt/freeswitch/mod/mod_g723_1.so: cannot open shared object
> file:
> >> >> >> No such file or directory**
> >> >> >> 2011-02-28 18:12:21.912573 [CRIT] switch_loadable_module.c:882
> Error
> >> >> >> Loading module /opt/freeswitch/mod/mod_g729.so
> >> >> >> **/opt/freeswitch/mod/mod_g729.so: cannot open shared object file:
> >> >> >> No
> >> >> >> such file or directory**
> >> >> >> 2011-02-28 18:12:21.912609 [CRIT] switch_loadable_module.c:882
> Error
> >> >> >> Loading module /opt/freeswitch/mod/mod_amr.so
> >> >> >> **/opt/freeswitch/mod/mod_amr.so: cannot open shared object file:
> No
> >> >> >> such file or directory**
> >> >> >>
> >> >> >> Important things to note:
> >> >> >> mod_event_socket hasn't loaded because of a undefined symbol, it
> >> >> >> wasn't built correctly
> >> >> >> mod_xml_curl hasn't loaded because it doesn't exist, it was either
> >> >> >> never built or never installed
> >> >> >>
> >> >> >> Something pretty strange has happened. Did you recompile when
> trying
> >> >> >> to add mod_xml_curl?
> >> >> >>
> >> >> >> I'd suggest you delete all the FS files, including the Git clone,
> >> >> >> make
> >> >> >> a fresh checkout and build it from scratch.
> >> >> >>
> >> >> >> -Steve
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >> On 28 February 2011 23:18, deniro <ayhkor at gmail.com> wrote:
> >> >> >> > I put the log in pastebin
> >> >> >> > freeswitch.log only when starting freeswitch (after stop)
> >> >> >> >
> >> >> >> > http://pastebin.freeswitch.org/15502
> >> >> >> > thx
> >> >> >> > deniro
> >> >> >> > On Mon, Feb 28, 2011 at 5:46 PM, Steven Ayre <
> steveayre at gmail.com>
> >> >> >> > wrote:
> >> >> >> >>
> >> >> >> >> You must check the logfile for when FS starts up - if netstat
> >> >> >> >> shows
> >> >> >> >> nothing for port 8021 then either mod_event_socket isn't being
> >> >> >> >> loaded
> >> >> >> >> or you'll see an error when it tries to load.
> >> >> >> >>
> >> >> >> >> Nothing from netstat means nothing's listening, so trying to
> >> >> >> >> connect
> >> >> >> >> using fs_cli won't do anything.
> >> >> >> >>
> >> >> >> >> -Steve
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> On 28 February 2011 22:40, deniro <ayhkor at gmail.com> wrote:
> >> >> >> >> > Checked the log
> >> >> >> >> > Nothing is logged when
> >> >> >> >> > fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
> >> >> >> >> > is issued
> >> >> >> >> > /opt/freeswitch/log# tail -f freeswitch.log
> >> >> >> >> >
> >> >> >> >> > ls -l freeswitch.log
> >> >> >> >> > -rwxrwxrwx 1 freeswitch daemon 33018 2011-02-28 17:39
> >> >> >> >> > freeswitch.log
> >> >> >> >> > thx
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > On Mon, Feb 28, 2011 at 3:59 PM, Peter Olsson
> >> >> >> >> > <peter.olsson at visionutveckling.se> wrote:
> >> >> >> >> >>
> >> >> >> >> >> Check freeswitch.log, it probably reports some problem when
> >> >> >> >> >> loading
> >> >> >> >> >> the
> >> >> >> >> >> mod_event_socket module.
> >> >> >> >> >>
> >> >> >> >> >> /Peter
> >> >> >> >> >> ________________________________________
> >> >> >> >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> >> >> >> >> >> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r
> >> >> >> >> >> deniro
> >> >> >> >> >> [ayhkor at gmail.com]
> >> >> >> >> >> Skickat: den 28 februari 2011 21:42
> >> >> >> >> >> Till: FreeSWITCH Users Help
> >> >> >> >> >> ?mne: Re: [Freeswitch-users] fs_cli socket connection error
> >> >> >> >> >>
> >> >> >> >> >> Steve
> >> >> >> >> >> Thanks for the reply
> >> >> >> >> >> netstat -anlp | grep 8021  is blank (nothing showing up)
> >> >> >> >> >>
> >> >> >> >> >> /opt/freeswitch/bin/fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
> >> >> >> >> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
> >> >> >> >> >> Configuration
> >> >> >> >> >> file is /root/.fs_cli_conf.
> >> >> >> >> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
> >> >> >> >> >> Configuration
> >> >> >> >> >> file is /etc/fs_cli.conf.
> >> >> >> >> >> [DEBUG] libs/esl/fs_cli.c:1159 main() profile default does
> not
> >> >> >> >> >> exist
> >> >> >> >> >> using
> >> >> >> >> >> builtin profile
> >> >> >> >> >> [DEBUG] libs/esl/fs_cli.c:1188 main() Using profile internal
> >> >> >> >> >> [75.xxx.xxx.xxx]
> >> >> >> >> >> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting
> [Socket
> >> >> >> >> >> Connection
> >> >> >> >> >> Error]
> >> >> >> >> >> yes mod_event_socket is after  mod_xml_curl but I changed
> the
> >> >> >> >> >> order
> >> >> >> >> >> in
> >> >> >> >> >>  modules.conf.xml
> >> >> >> >> >> still getting above (restarted freeswitch)
> >> >> >> >> >>
> >> >> >> >> >> thx
> >> >> >> >> >> deniro--
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >> On Mon, Feb 28, 2011 at 3:35 AM, Steven Ayre
> >> >> >> >> >> <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
> >> >> >> >> >> Enable debug logging and you should see an error that'll
> tell
> >> >> >> >> >> you
> >> >> >> >> >> more.
> >> >> >> >> >>
> >> >> >> >> >> Is mod_event_socket loading before or after mod_xml_curl?
> >> >> >> >> >> Chances
> >> >> >> >> >> are
> >> >> >> >> >> mod_xml_curl is loading first, so it's trying to read
> >> >> >> >> >> event_socket.conf.xml and/or acl.conf.xml through xml_curl
> and
> >> >> >> >> >> either
> >> >> >> >> >> getting a different config to your previous local copy or
> the
> >> >> >> >> >> ACLs
> >> >> >> >> >> are
> >> >> >> >> >> different.
> >> >> >> >> >>
> >> >> >> >> >> What does "netstat -anlp | grep :8021" from the linux cli
> show
> >> >> >> >> >> you?
> >> >> >> >> >> Does it show that freeswitch is actually listening on the
> >> >> >> >> >> port?
> >> >> >> >> >> If
> >> >> >> >> >> it
> >> >> >> >> >> is it's probably an ACL problem, if it isn't then it's
> >> >> >> >> >> probably a
> >> >> >> >> >> problem with event_socket.conf.xml
> >> >> >> >> >>
> >> >> >> >> >> -Steve
> >> >> >> >> >>
> >> >> >> >> >> On 28 February 2011 00:53, deniro
> >> >> >> >> >> <ayhkor at gmail.com<mailto:ayhkor at gmail.com>> wrote:
> >> >> >> >> >> > What would be possible reasons for this and how to
> resolve?
> >> >> >> >> >> > running fs 106 on ubuntu 10.04 server
> >> >> >> >> >> >
> >> >> >> >> >> > /opt/freeswitch/bin/fs_cli -H xxx.xxx.xxx.xxx -P 8021
> >> >> >> >> >> > [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting
> >> >> >> >> >> > [Socket
> >> >> >> >> >> > Connection
> >> >> >> >> >> > Error]
> >> >> >> >> >> > was working fine before I installed  mod_xml_cdr
> >> >> >> >> >> > configure --prefix=/opt/freeswitch --without-libcurl
> >> >> >> >> >> > make mod_xml_cdr-install
> >> >> >> >> >> > (no errors)
> >> >> >> >> >> >
> >> >> >> >> >> > in modules.conf.xml
> >> >> >> >> >> > <load module="mod_event_socket"/>
> >> >> >> >> >> >
> >> >> >> >> >> > is there a packaged version of mod_xml_cdr (for fs 1.0.6)?
> >> >> >> >> >> >
> >> >> >> >> >> > thx
> >> >> >> >> >> > deniro--
> >> >> >> >> >> > _______________________________________________
> >> >> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> > FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> >> >> >> >> >> >
> >> >> >> >> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> >> > http://www.freeswitch.org<http://www.freeswitch.org/>
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >>
> >> >> >> >> >> _______________________________________________
> >> >> >> >> >> FreeSWITCH-users mailing list
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> >> >> >> >> >>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> >> http://www.freeswitch.org<http://www.freeswitch.org/>
> >> >> >> >> >>
> >> >> >> >> >> !DSPAM:4d6c0a1132761029518849!
> >> >> >> >> >>
> >> >> >> >> >> _______________________________________________
> >> >> >> >> >> FreeSWITCH-users mailing list
> >> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> >>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> >> http://www.freeswitch.org
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > _______________________________________________
> >> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> > http://www.freeswitch.org
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >>
> >> >> >> >> _______________________________________________
> >> >> >> >> FreeSWITCH-users mailing list
> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >> > _______________________________________________
> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >
> >> >> >> >
> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> > http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > 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-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sunwood360 at gmail.com  Fri Mar  4 07:13:13 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Thu, 3 Mar 2011 20:13:13 -0800
Subject: [Freeswitch-users] weird FS behavior
In-Reply-To: <AANLkTin-cCDbiFAQ+H02zUkNHb+y_eC479dCyTw4rfoH@mail.gmail.com>
References: <AANLkTimDLm0qNWw=hL0pXZ6ic2McDPjwDrdb3tgx9Tfr@mail.gmail.com>
	<AANLkTin-cCDbiFAQ+H02zUkNHb+y_eC479dCyTw4rfoH@mail.gmail.com>
Message-ID: <AANLkTi=CSDJiAuJKprdjD3Y0qchuwnPYbp7npoP7+80B@mail.gmail.com>

yes, my fault. I totally forgot my phone is connected to a linksys PAP.


On Tue, Mar 1, 2011 at 10:20 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
>> 2011-03-01 03:22:03.540372 [INFO] mod_dialplan_xml.c:418 Processing
>> 1000->13474562345 in context default
>>
>>
> I'm with Peter on this one. It looks like FreeSWITCH is receiving the
> dialstring with the "1347" already pre-pended. Time to check the phone.
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From aligzaidi at gmail.com  Fri Mar  4 05:11:25 2011
From: aligzaidi at gmail.com (Ali G. Zaidi)
Date: Thu, 3 Mar 2011 21:11:25 -0500
Subject: [Freeswitch-users] Dialing extensions behind NAT
In-Reply-To: <FBFB4BE1-4856-4C93-8CE4-044D051AF6C6@freeswitch.org>
References: <AANLkTik_Q9DsUNeb7ogkgkt5ykSWv+4rE98oZu_OtRdv@mail.gmail.com>
	<FBFB4BE1-4856-4C93-8CE4-044D051AF6C6@freeswitch.org>
Message-ID: <3BA5114D-BF0A-44EE-9229-8BFA3FE78CCE@gmail.com>

Sorry my question wasn't clear enough, so I ask again.
UA1,UA2,UA3===(NAT) ====(Internet)====(FreeSwitch)===(PSTN)

 

When UA1 dial UA2 the traffic go over the internet connection to FreeSwitch and then comes back to UA2. Can UA1 dial UA2 without going to internet? In other words, after SIP signaling, RTP traffic flow between UAs without FreeSwitch in between? I know in Asterisk you can configure UA with "caninvite" enable, wondering if FS also able to this and how?

Sent from my iPhone


On Mar 3, 2011, at 5:58 PM, Brian West <brian at freeswitch.org> wrote:

> give each phone a different client side sip port.
> 
> /b
> 
> On Mar 3, 2011, at 3:44 PM, Ali Zaidi wrote:
> 
>> My both phones are Linksys SPA942, please let me know any good phones that
>> can work best with free switch over wan connection.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From steveayre at gmail.com  Fri Mar  4 07:28:00 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 4 Mar 2011 04:28:00 +0000
Subject: [Freeswitch-users] fs_cli socket connection error
In-Reply-To: <AANLkTi=0xgVmox-Px1QSu27h6uxSStTRRWKbn2Utsmfs@mail.gmail.com>
References: <AANLkTikCoT9dzLNoD=6TzS0jiMw4afNX94TypsLarBbz@mail.gmail.com>
	<AANLkTi=sENgBB9Cdr-EkLJD8XfmH3A8GSfPnvDHO7qrw@mail.gmail.com>
	<AANLkTin5k=2Z-=QVVzqt91172wv88P7OFkqRqTY9VDcS@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58ADB727D7@cooper>
	<AANLkTi=aPwmnVD8ZzaBVFDxGyPsbgzDEBGaXvq_B2NbF@mail.gmail.com>
	<AANLkTikpQuxvyaU7PAGnyJuoQLQ9xn4inNJ_u5x41Anv@mail.gmail.com>
	<AANLkTimNYjC_G-Lq+St5VePpxEyM8iHnhaNCo=jBydi9@mail.gmail.com>
	<AANLkTi=QXgEb6dHU9bfb7v06WzPyWJcoNYDLi1X9oq+E@mail.gmail.com>
	<AANLkTikEDgpAJppNUnLac73ip7qsk+8AK-N-pTW7NBm6@mail.gmail.com>
	<AANLkTi=NP_Kw-Hrb6+XXtuid3YWJUM2ehFaeoaV-UEaC@mail.gmail.com>
	<AANLkTinQ7g2YFCGNRtxWr-apHuByNFdF3Z_ph8J7K01e@mail.gmail.com>
	<AANLkTin_Y-azNb2sT-U9HQsjVbL3JqPiBamTiX5a_HC+@mail.gmail.com>
	<AANLkTi=d18FUPX_mWDO4eLEW5Uf+PYFfmtLfZ9C=LTKR@mail.gmail.com>
	<AANLkTim8HxbSxoVnfFHWmG5WxdWsvHF4v7rX-ZonuW6k@mail.gmail.com>
	<AANLkTi=0xgVmox-Px1QSu27h6uxSStTRRWKbn2Utsmfs@mail.gmail.com>
Message-ID: <C908E234-FAE2-4268-82EC-322FD5BBFDB4@gmail.com>

If you don't know what it's for, you've just broken your fs setup.

The mod_XML_curl module replaces certain bindings with URLs - dialplan, configuration, directory. Any request to access those generates a http request to the URL you give in the config, so that you can build a script that generates the XML dynamically.

Every binding is optional - you do not have to have any of them. The error probably means you had defined a binding but with no URL. Can't say for sure without seeing your old config file though.

I'd suggest you read up on the wiki pages if you haven't already done so - http://wiki.freeswitch.org/wiki/Mod_xml_curl.

If you don't need the module then I'd not bother loading it.

Steve on iPhone

On 4 Mar 2011, at 04:08, deniro <ayhkor at gmail.com> wrote:

> Alright,  I did some googling  around  "[ERR] mod_xml_curl.c:444 Binding has no url "  then
> uncommented below line in conf/autoload_configs/xml_curl.conf.xml
> <param name="gateway-url" value="http://www.freeswitch.org/gateway.xml" bindings="dialplan"/>
> though I don't know what this is for..
> 
> Now I am able to load mod_xml_curl
> no need to re-install everything
> thanks everyone
> deniro--
>  
>  
> On Thu, Mar 3, 2011 at 3:32 AM, Steven Ayre <steveayre at gmail.com> wrote:
> freeswitch at xxx@internal> load mod_xml_curl
> -ERR [module load file routine returned an error]
> 
> There's still an error while loading. You need to look at the logs for
> freeswitch, not the fs_cli ones.
> 
> Either run 'fs_cli -l debug' or do '/log 7' within fs_cli
> 
> -Steve
> 
> On 3 March 2011 00:28, deniro <ayhkor at gmail.com> wrote:
> > ok, I see one under src. I copied that to conf/autoload_configs and looks
> > like I am able to load mod_xml_cdr this way
> > freeswitch at xxxx@internal>
> > load mod_xml_cdr
> > module_exists mod_xml_cdr
> > true
> > same method did not work for mod_xml_curl
> > below is error message after I copied file
> > cp /usr/src/freeswitch-1.0.6/conf/autoload_configs/xml_curl.conf.xml
> > /opt/freeswitch/conf/autoload_configs/xml_curl.conf.xml
> >
> > freeswitch at xxx@internal> load mod_xml_curl
> > [DEBUG] libs/esl/src/esl.c:1140 esl_send() SEND
> > api load mod_xml_curl
> > console_execute: true
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Type] =
> > [api/response]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Length]
> > = [68]
> > +OK Reloading XML
> > -ERR [module load file routine returned an error]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Type] =
> > [log/data]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Length]
> > = [63]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Level] =
> > [6]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Text-Channel] =
> > [0]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-File] =
> > [mod_enum.c]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Func] =
> > [event_handler]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Line] =
> > [808]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [User-Data] = []
> > [DEBUG] libs/esl/src/esl.c:1114 esl_recv_event() RECV MESSAGE
> > Event-Name: SOCKET_DATA
> > Content-Type: log/data
> > Content-Length: 63
> > Log-Level: 6
> > Text-Channel: 0
> > Log-File: mod_enum.c
> > Log-Func: event_handler
> > Log-Line: 808
> > User-Data: _undef_
> > 2011-03-02 19:23:34.416231 [INFO] mod_enum.c:808 ENUM Reloaded
> > 2011-03-02 19:23:34.416231 [INFO] mod_enum.c:808 ENUM Reloaded
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Type] =
> > [log/data]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Length]
> > = [72]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Level] =
> > [3]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Text-Channel] =
> > [0]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-File] =
> > [mod_xml_curl.c]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Func] =
> > [do_config]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Line] =
> > [444]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [User-Data] = []
> > [DEBUG] libs/esl/src/esl.c:1114 esl_recv_event() RECV MESSAGE
> > Event-Name: SOCKET_DATA
> > Content-Type: log/data
> > Content-Length: 72
> > Log-Level: 3
> > Text-Channel: 0
> > Log-File: mod_xml_curl.c
> > Log-Func: do_config
> > Log-Line: 444
> > User-Data: _undef_
> > 2011-03-02 19:23:34.416231 [ERR] mod_xml_curl.c:444 Binding has no url!
> > 2011-03-02 19:23:34.416231 [ERR] mod_xml_curl.c:444 Binding has no url!
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Type] =
> > [log/data]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Length]
> > = [162]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Level] =
> > [2]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Text-Channel] =
> > [0]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-File] =
> > [switch_loadable_module.c]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Func] =
> > [switch_loadable_module_load_file]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Line] =
> > [926]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [User-Data] = []
> > [DEBUG] libs/esl/src/esl.c:1114 esl_recv_event() RECV MESSAGE
> > Event-Name: SOCKET_DATA
> > Content-Type: log/data
> > Content-Length: 162
> > Log-Level: 2
> > Text-Channel: 0
> > Log-File: switch_loadable_module.c
> > Log-Func: switch_loadable_module_load_file
> > Log-Line: 926
> > User-Data: _undef_
> > 2011-03-02 19:23:34.416231 [CRIT] switch_loadable_module.c:926 Error Loading
> > module /opt/freeswitch/mod/mod_xml_curl.so
> > **Module load routine returned an error**
> > 2011-03-02 19:23:34.416231 [CRIT] switch_loadable_module.c:926 Error Loading
> > module /opt/freeswitch/mod/mod_xml_curl.so
> > **Module load routine returned an error**
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Type] =
> > [log/data]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Content-Length]
> > = [86]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Level] =
> > [6]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Text-Channel] =
> > [0]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-File] =
> > [switch_time.c]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Func] =
> > [switch_load_timezones]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [Log-Line] =
> > [950]
> > [DEBUG] libs/esl/src/esl.c:988 esl_recv_event() RECV HEADER [User-Data] = []
> > [DEBUG] libs/esl/src/esl.c:1114 esl_recv_event() RECV MESSAGE
> > Event-Name: SOCKET_DATA
> > Content-Type: log/data
> > Content-Length: 86
> > Log-Level: 6
> > Text-Channel: 0
> > Log-File: switch_time.c
> > Log-Func: switch_load_timezones
> > Log-Line: 950
> > User-Data: _undef_
> > 2011-03-02 19:23:34.417121 [INFO] switch_time.c:950 Timezone reloaded 530
> > definitions
> > 2011-03-02 19:23:34.417121 [INFO] switch_time.c:950 Timezone reloaded 530
> > definitions
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > On Wed, Mar 2, 2011 at 6:25 PM, Steven Ayre <steveayre at gmail.com> wrote:
> >>
> >> You need to create its config file. It won't load without it.
> >>
> >> "find . -name xml_cdr.conf"
> >>
> >> Actually it's xml_cdr.conf.xml, all config files will show .conf as
> >> their logical name but on the hard drive they're stored in .conf.xml.
> >> It should be in conf/autoload_configs
> >>
> >> -Steve
> >>
> >>
> >> On 2 March 2011 22:25, deniro <ayhkor at gmail.com> wrote:
> >> > I typed the 4 commands but still unable to load mod_xml_cdr and
> >> > mod_xml_curl
> >> >
> >> >
> >> > 2011-03-02 17:21:32.139672 [ERR] mod_xml_cdr.c:479 Open of xml_cdr.conf
> >> > failed
> >> > /opt# find . -name xml_cdr.conf
> >> > nothing appears
> >> >
> >> > here is errors from fs_cli -d 7
> >> >
> >> > freeswitch at 75.xxx.xxx.xx@internal> load mod_xml_cdr
> >> > [DEBUG] libs/esl/src/esl.c:1063 esl_send() SEND
> >> > api load mod_xml_cdr
> >> > console_execute: true
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> >> > [Content-Type] =
> >> > [api/response]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> >> > [Content-Length]
> >> > = [50]
> >> > -ERR [module load file routine returned an error]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> >> > [Content-Type] =
> >> > [log/data]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> >> > [Content-Length]
> >> > = [79]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level]
> >> > =
> >> > [3]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> >> > [Text-Channel] =
> >> > [0]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> >> > [mod_xml_cdr.c]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> >> > [mod_xml_cdr_load]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> >> > [479]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data]
> >> > = []
> >> > [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> >> > Event-Name: SOCKET_DATA
> >> > Content-Type: log/data
> >> > Content-Length: 79
> >> > Log-Level: 3
> >> > Text-Channel: 0
> >> > Log-File: mod_xml_cdr.c
> >> > Log-Func: mod_xml_cdr_load
> >> > Log-Line: 479
> >> > User-Data: _undef_
> >> > 2011-03-02 17:21:32.139672 [ERR] mod_xml_cdr.c:479 Open of xml_cdr.conf
> >> > failed
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> >> > [Content-Type] =
> >> > [log/data]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> >> > [Content-Length]
> >> > = [161]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level]
> >> > =
> >> > [2]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> >> > [Text-Channel] =
> >> > [0]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> >> > [switch_loadable_module.c]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> >> > [switch_loadable_module_load_file]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> >> > [882]
> >> > [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data]
> >> > = []
> >> > [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> >> > Event-Name: SOCKET_DATA
> >> > Content-Type: log/data
> >> > Content-Length: 161
> >> > Log-Level: 2
> >> > Text-Channel: 0
> >> > Log-File: switch_loadable_module.c
> >> > Log-Func: switch_loadable_module_load_file
> >> > Log-Line: 882
> >> > User-Data: _undef_
> >> > 2011-03-02 17:21:32.139672 [CRIT] switch_loadable_module.c:882 Error
> >> > Loading
> >> > module /opt/freeswitch/mod/mod_xml_cdr.so
> >> > **Module load routine returned an error**
> >> >
> >> >
> >> >
> >> >
> >> >
> >> > On Tue, Mar 1, 2011 at 1:11 PM, Anthony Minessale
> >> > <anthony.minessale at gmail.com> wrote:
> >> >>
> >> >> like I said you re-installed with a new version of libcurl but did not
> >> >> fully rebuild.
> >> >> So most of the modules that depend on curl were broken torn between
> >> >> the old and new version it was linked to..
> >> >>
> >> >> had you executed the 4 commands I posted yesterday it would be fine.
> >> >>
> >> >>
> >> >> On Tue, Mar 1, 2011 at 11:02 AM, deniro <ayhkor at gmail.com> wrote:
> >> >> > Thank you all for advice  and direction
> >> >> > Will re-install from scratch at a later time.
> >> >> > deniro--
> >> >> >
> >> >> > On Tue, Mar 1, 2011 at 4:16 AM, Steven Ayre <steveayre at gmail.com>
> >> >> > wrote:
> >> >> >>
> >> >> >> Ok, you really need to pay attention to any log message that logs
> >> >> >> with
> >> >> >> the CRIT (critical) log level - they're really serious errors. You
> >> >> >> should look at adding the critical="true" attribute to all the tags
> >> >> >> in
> >> >> >> modules.conf.xml which you want to be sure are loaded. If a module
> >> >> >> fails to load without that tag FS will continue to run anyway, with
> >> >> >> that tag it'll refuse to start so you'll know instantly something is
> >> >> >> wrong.
> >> >> >>
> >> >> >> Most of your modules aren't loading, because they haven't been
> >> >> >> compiled correctly. Some look like they're missing (no such file or
> >> >> >> directory) and others haven't been built right (undefined symbols).
> >> >> >>
> >> >> >> Here are a few of the important lines from your log:
> >> >> >>
> >> >> >> 2011-02-28 18:12:21.910738 [CRIT] switch_loadable_module.c:882 Error
> >> >> >> Loading module /opt/freeswitch/mod/mod_event_socket.so
> >> >> >> **/opt/freeswitch/mod/mod_event_socket.so: undefined symbol:
> >> >> >> switch_event_serialize_json**
> >> >> >> 2011-02-28 18:12:21.910789 [CRIT] switch_loadable_module.c:882 Error
> >> >> >> Loading module /opt/freeswitch/mod/mod_xml_curl.so
> >> >> >> **/opt/freeswitch/mod/mod_xml_curl.so: cannot open shared object
> >> >> >> file:
> >> >> >> No such file or directory**
> >> >> >> 2011-02-28 18:12:21.910813 [CRIT] switch_loadable_module.c:882 Error
> >> >> >> Loading module /opt/freeswitch/mod/mod_xml_cdr.so
> >> >> >> **/opt/freeswitch/mod/mod_xml_cdr.so: cannot open shared object
> >> >> >> file:
> >> >> >> No such file or directory**
> >> >> >> 2011-02-28 18:12:21.911581 [CRIT] switch_loadable_module.c:882 Error
> >> >> >> Loading module /opt/freeswitch/mod/mod_commands.so
> >> >> >> **/opt/freeswitch/mod/mod_commands.so: undefined symbol:
> >> >> >> switch_xml_reload**
> >> >> >> 2011-02-28 18:12:21.911709 [CRIT] switch_loadable_module.c:882 Error
> >> >> >> Loading module /opt/freeswitch/mod/mod_conference.so
> >> >> >> **/opt/freeswitch/mod/mod_conference.so: undefined symbol:
> >> >> >> switch_channel_test_app_flag_key**
> >> >> >> 2011-02-28 18:12:21.911842 [CRIT] switch_loadable_module.c:882 Error
> >> >> >> Loading module /opt/freeswitch/mod/mod_dptools.so
> >> >> >> **/opt/freeswitch/mod/mod_dptools.so: undefined symbol:
> >> >> >> switch_event_merge**
> >> >> >> 2011-02-28 18:12:21.912264 [CRIT] switch_loadable_module.c:882 Error
> >> >> >> Loading module /opt/freeswitch/mod/mod_dialplan_xml.so
> >> >> >> **/opt/freeswitch/mod/mod_dialplan_xml.so: undefined symbol:
> >> >> >> switch_xml_std_datetime_check**
> >> >> >> 2011-02-28 18:12:21.912531 [CRIT] switch_loadable_module.c:882 Error
> >> >> >> Loading module /opt/freeswitch/mod/mod_g723_1.so
> >> >> >> **/opt/freeswitch/mod/mod_g723_1.so: cannot open shared object file:
> >> >> >> No such file or directory**
> >> >> >> 2011-02-28 18:12:21.912573 [CRIT] switch_loadable_module.c:882 Error
> >> >> >> Loading module /opt/freeswitch/mod/mod_g729.so
> >> >> >> **/opt/freeswitch/mod/mod_g729.so: cannot open shared object file:
> >> >> >> No
> >> >> >> such file or directory**
> >> >> >> 2011-02-28 18:12:21.912609 [CRIT] switch_loadable_module.c:882 Error
> >> >> >> Loading module /opt/freeswitch/mod/mod_amr.so
> >> >> >> **/opt/freeswitch/mod/mod_amr.so: cannot open shared object file: No
> >> >> >> such file or directory**
> >> >> >>
> >> >> >> Important things to note:
> >> >> >> mod_event_socket hasn't loaded because of a undefined symbol, it
> >> >> >> wasn't built correctly
> >> >> >> mod_xml_curl hasn't loaded because it doesn't exist, it was either
> >> >> >> never built or never installed
> >> >> >>
> >> >> >> Something pretty strange has happened. Did you recompile when trying
> >> >> >> to add mod_xml_curl?
> >> >> >>
> >> >> >> I'd suggest you delete all the FS files, including the Git clone,
> >> >> >> make
> >> >> >> a fresh checkout and build it from scratch.
> >> >> >>
> >> >> >> -Steve
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >> On 28 February 2011 23:18, deniro <ayhkor at gmail.com> wrote:
> >> >> >> > I put the log in pastebin
> >> >> >> > freeswitch.log only when starting freeswitch (after stop)
> >> >> >> >
> >> >> >> > http://pastebin.freeswitch.org/15502
> >> >> >> > thx
> >> >> >> > deniro
> >> >> >> > On Mon, Feb 28, 2011 at 5:46 PM, Steven Ayre <steveayre at gmail.com>
> >> >> >> > wrote:
> >> >> >> >>
> >> >> >> >> You must check the logfile for when FS starts up - if netstat
> >> >> >> >> shows
> >> >> >> >> nothing for port 8021 then either mod_event_socket isn't being
> >> >> >> >> loaded
> >> >> >> >> or you'll see an error when it tries to load.
> >> >> >> >>
> >> >> >> >> Nothing from netstat means nothing's listening, so trying to
> >> >> >> >> connect
> >> >> >> >> using fs_cli won't do anything.
> >> >> >> >>
> >> >> >> >> -Steve
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> On 28 February 2011 22:40, deniro <ayhkor at gmail.com> wrote:
> >> >> >> >> > Checked the log
> >> >> >> >> > Nothing is logged when
> >> >> >> >> > fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
> >> >> >> >> > is issued
> >> >> >> >> > /opt/freeswitch/log# tail -f freeswitch.log
> >> >> >> >> >
> >> >> >> >> > ls -l freeswitch.log
> >> >> >> >> > -rwxrwxrwx 1 freeswitch daemon 33018 2011-02-28 17:39
> >> >> >> >> > freeswitch.log
> >> >> >> >> > thx
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > On Mon, Feb 28, 2011 at 3:59 PM, Peter Olsson
> >> >> >> >> > <peter.olsson at visionutveckling.se> wrote:
> >> >> >> >> >>
> >> >> >> >> >> Check freeswitch.log, it probably reports some problem when
> >> >> >> >> >> loading
> >> >> >> >> >> the
> >> >> >> >> >> mod_event_socket module.
> >> >> >> >> >>
> >> >> >> >> >> /Peter
> >> >> >> >> >> ________________________________________
> >> >> >> >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> >> >> >> >> >> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r
> >> >> >> >> >> deniro
> >> >> >> >> >> [ayhkor at gmail.com]
> >> >> >> >> >> Skickat: den 28 februari 2011 21:42
> >> >> >> >> >> Till: FreeSWITCH Users Help
> >> >> >> >> >> ?mne: Re: [Freeswitch-users] fs_cli socket connection error
> >> >> >> >> >>
> >> >> >> >> >> Steve
> >> >> >> >> >> Thanks for the reply
> >> >> >> >> >> netstat -anlp | grep 8021  is blank (nothing showing up)
> >> >> >> >> >>
> >> >> >> >> >> /opt/freeswitch/bin/fs_cli -d 7 -H 75.xxx.xxx.xxx -P 8021
> >> >> >> >> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
> >> >> >> >> >> Configuration
> >> >> >> >> >> file is /root/.fs_cli_conf.
> >> >> >> >> >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file()
> >> >> >> >> >> Configuration
> >> >> >> >> >> file is /etc/fs_cli.conf.
> >> >> >> >> >> [DEBUG] libs/esl/fs_cli.c:1159 main() profile default does not
> >> >> >> >> >> exist
> >> >> >> >> >> using
> >> >> >> >> >> builtin profile
> >> >> >> >> >> [DEBUG] libs/esl/fs_cli.c:1188 main() Using profile internal
> >> >> >> >> >> [75.xxx.xxx.xxx]
> >> >> >> >> >> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
> >> >> >> >> >> Connection
> >> >> >> >> >> Error]
> >> >> >> >> >> yes mod_event_socket is after  mod_xml_curl but I changed the
> >> >> >> >> >> order
> >> >> >> >> >> in
> >> >> >> >> >>  modules.conf.xml
> >> >> >> >> >> still getting above (restarted freeswitch)
> >> >> >> >> >>
> >> >> >> >> >> thx
> >> >> >> >> >> deniro--
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >> On Mon, Feb 28, 2011 at 3:35 AM, Steven Ayre
> >> >> >> >> >> <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
> >> >> >> >> >> Enable debug logging and you should see an error that'll tell
> >> >> >> >> >> you
> >> >> >> >> >> more.
> >> >> >> >> >>
> >> >> >> >> >> Is mod_event_socket loading before or after mod_xml_curl?
> >> >> >> >> >> Chances
> >> >> >> >> >> are
> >> >> >> >> >> mod_xml_curl is loading first, so it's trying to read
> >> >> >> >> >> event_socket.conf.xml and/or acl.conf.xml through xml_curl and
> >> >> >> >> >> either
> >> >> >> >> >> getting a different config to your previous local copy or the
> >> >> >> >> >> ACLs
> >> >> >> >> >> are
> >> >> >> >> >> different.
> >> >> >> >> >>
> >> >> >> >> >> What does "netstat -anlp | grep :8021" from the linux cli show
> >> >> >> >> >> you?
> >> >> >> >> >> Does it show that freeswitch is actually listening on the
> >> >> >> >> >> port?
> >> >> >> >> >> If
> >> >> >> >> >> it
> >> >> >> >> >> is it's probably an ACL problem, if it isn't then it's
> >> >> >> >> >> probably a
> >> >> >> >> >> problem with event_socket.conf.xml
> >> >> >> >> >>
> >> >> >> >> >> -Steve
> >> >> >> >> >>
> >> >> >> >> >> On 28 February 2011 00:53, deniro
> >> >> >> >> >> <ayhkor at gmail.com<mailto:ayhkor at gmail.com>> wrote:
> >> >> >> >> >> > What would be possible reasons for this and how to resolve?
> >> >> >> >> >> > running fs 106 on ubuntu 10.04 server
> >> >> >> >> >> >
> >> >> >> >> >> > /opt/freeswitch/bin/fs_cli -H xxx.xxx.xxx.xxx -P 8021
> >> >> >> >> >> > [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting
> >> >> >> >> >> > [Socket
> >> >> >> >> >> > Connection
> >> >> >> >> >> > Error]
> >> >> >> >> >> > was working fine before I installed  mod_xml_cdr
> >> >> >> >> >> > configure --prefix=/opt/freeswitch --without-libcurl
> >> >> >> >> >> > make mod_xml_cdr-install
> >> >> >> >> >> > (no errors)
> >> >> >> >> >> >
> >> >> >> >> >> > in modules.conf.xml
> >> >> >> >> >> > <load module="mod_event_socket"/>
> >> >> >> >> >> >
> >> >> >> >> >> > is there a packaged version of mod_xml_cdr (for fs 1.0.6)?
> >> >> >> >> >> >
> >> >> >> >> >> > thx
> >> >> >> >> >> > deniro--
> >> >> >> >> >> > _______________________________________________
> >> >> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> > FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> >> >> >> >> >> >
> >> >> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> >> > http://www.freeswitch.org<http://www.freeswitch.org/>
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >>
> >> >> >> >> >> _______________________________________________
> >> >> >> >> >> FreeSWITCH-users mailing list
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> >> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> >> http://www.freeswitch.org<http://www.freeswitch.org/>
> >> >> >> >> >>
> >> >> >> >> >> !DSPAM:4d6c0a1132761029518849!
> >> >> >> >> >>
> >> >> >> >> >> _______________________________________________
> >> >> >> >> >> FreeSWITCH-users mailing list
> >> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> >> http://www.freeswitch.org
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > _______________________________________________
> >> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> > http://www.freeswitch.org
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >>
> >> >> >> >> _______________________________________________
> >> >> >> >> FreeSWITCH-users mailing list
> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >> > _______________________________________________
> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >
> >> >> >> >
> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> > http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > 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-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From steveayre at gmail.com  Fri Mar  4 07:31:14 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 4 Mar 2011 04:31:14 +0000
Subject: [Freeswitch-users] Dialing extensions behind NAT
In-Reply-To: <3BA5114D-BF0A-44EE-9229-8BFA3FE78CCE@gmail.com>
References: <AANLkTik_Q9DsUNeb7ogkgkt5ykSWv+4rE98oZu_OtRdv@mail.gmail.com>
	<FBFB4BE1-4856-4C93-8CE4-044D051AF6C6@freeswitch.org>
	<3BA5114D-BF0A-44EE-9229-8BFA3FE78CCE@gmail.com>
Message-ID: <0FD8DCDC-EDD4-42BA-AE05-C8A7773F1552@gmail.com>

<action application="set" data="bypass_media=true"/> before bridge.

You'll need to be sure they aren't behind different nat routers though.

Steve on iPhone

On 4 Mar 2011, at 02:11, "Ali G. Zaidi" <aligzaidi at gmail.com> wrote:

> Sorry my question wasn't clear enough, so I ask again.
> UA1,UA2,UA3===(NAT) ====(Internet)====(FreeSwitch)===(PSTN)
> 
>  
> 
> When UA1 dial UA2 the traffic go over the internet connection to FreeSwitch and then comes back to UA2. Can UA1 dial UA2 without going to internet? In other words, after SIP signaling, RTP traffic flow between UAs without FreeSwitch in between? I know in Asterisk you can configure UA with "caninvite" enable, wondering if FS also able to this and how?
> 
> Sent from my iPhone
> 
> 
> On Mar 3, 2011, at 5:58 PM, Brian West <brian at freeswitch.org> wrote:
> 
>> give each phone a different client side sip port.
>> 
>> /b
>> 
>> On Mar 3, 2011, at 3:44 PM, Ali Zaidi wrote:
>> 
>>> My both phones are Linksys SPA942, please let me know any good phones that
>>> can work best with free switch over wan connection.
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From u2nsam at gmail.com  Fri Mar  4 07:39:31 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 4 Mar 2011 10:09:31 +0530
Subject: [Freeswitch-users] API command
In-Reply-To: <AANLkTinkxPVyTAQDFnoatZzJv1imWQWE5zxUfX4vNY76@mail.gmail.com>
References: <AANLkTim-dEY_DOLQPUf9apOwFyntFsNVPhnvUXOp9_2d@mail.gmail.com>
	<AANLkTinkxPVyTAQDFnoatZzJv1imWQWE5zxUfX4vNY76@mail.gmail.com>
Message-ID: <AANLkTimL1hHZPFS49iKFhgLiBcQthBV1UiOQpfsbudpD@mail.gmail.com>

Thanks,

I got it, but calls disconnects when other person enters the conference,i
guess it could be because of codec negotiation @ conference room.

http://pastebin.freeswitch.org/15545


Rgards
Sam

On Thu, Mar 3, 2011 at 12:26 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Mar 2, 2011 at 1:48 AM, Sam <u2nsam at gmail.com> wrote:
>
>> HI,
>>
>>
>> Suppose if i want to place an external caller in the conference so i
>> should use the below command ?
>>
>> originate freetdm/wp1/a/9322273640  67287006 context &bridge 7050
>>
>
> What are 67287006 and "context"? if you want to transfer to ext 7050 in a
> context named "67287006" then do this:
> originate freetdm/wp1/a/9322273640  7050 XML 67287006
>
> remember that the second argument to originate is a dialplan destination
> number OR an &application(arg)
>
> -MC
>
>
>> where 7050 leads to conference .
>>
>>
>> Regds
>> Sam
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Fri Mar  4 07:43:06 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 4 Mar 2011 04:43:06 +0000
Subject: [Freeswitch-users] DTMF not passed through to client
In-Reply-To: <BFA68905-4F2E-4782-95CE-7C0D2CB01328@3js.de>
References: <BFA68905-4F2E-4782-95CE-7C0D2CB01328@3js.de>
Message-ID: <C9A6E059-78EE-4438-81CA-650A772E84CA@gmail.com>

Are you running the latest version?

It appears the dtmf is failing a sanity check and getting ignored as a result, possibly as a result of the device sending the wrong rtp timestamps. Can you get a tcpdump of the rtp stream?

Steve on iPhone

On 3 Mar 2011, at 22:50, Johannes Jakob <jjj at 3js.de> wrote:

> Hi,
> 
> what do I have to do to get RFC2833 DTMF passed to the client in an inbound scenario like this:
> 
> Upstream >rfc2833> FreeSWITCH >rfc2833> Asterisk
> 
> ?
> 
> The Upstream is sending the DTMF tones correctly:
> 
> ---
> 329    2.114845    UPSTREAMIP    FS_IP    RTP EVENT    62    Payload type=RTP Event, DTMF Eight 8
> ---
> 
> FreeSWITCH isn't logging much of the dtmf stuff, even with loglevel debug:
> 
> http://pastebin.freeswitch.org/15541
> 
> ...
> Set 2833 dtmf receive payload to 101
> ...
> 2011-03-03 23:29:59.498466 [ERR] switch_rtp.c:282 Failed DTMF sanity check.
> ...
> 
> 
> User's directory entry includes:
> 
>      <param name="dtmf-type" value="rfc2833"/>
>      <param name="pass-rfc2833" value="true"/>
> 
> I tried adding those in the gateway profile as well, no change.
> 
> 
> On the asterisk box I don't see those rtpevents in the pcap trace anymore, they just vanish on the FS box.
> 
> 
> Can somebody help me out here? Or point me to the right wiki page?
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#DTMF doesn't really tell me, what the options do and http://wiki.freeswitch.org/wiki/DTMF is something different.
> 
> 
> Thanks and bye,
> 
>  Johannes
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From u2nsam at gmail.com  Fri Mar  4 07:48:56 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 4 Mar 2011 10:18:56 +0530
Subject: [Freeswitch-users] API command
In-Reply-To: <AANLkTimL1hHZPFS49iKFhgLiBcQthBV1UiOQpfsbudpD@mail.gmail.com>
References: <AANLkTim-dEY_DOLQPUf9apOwFyntFsNVPhnvUXOp9_2d@mail.gmail.com>
	<AANLkTinkxPVyTAQDFnoatZzJv1imWQWE5zxUfX4vNY76@mail.gmail.com>
	<AANLkTimL1hHZPFS49iKFhgLiBcQthBV1UiOQpfsbudpD@mail.gmail.com>
Message-ID: <AANLkTimFJNBzt6_n-LAxpH+Rjozojd_W2yEO0eLJ5bTn@mail.gmail.com>

It disconnected because unknowingly # was pressed and got
disconnected,thanks.

How a moderator can ( mute and kick ) a particular person in the conference
room, is there any method

Regards
Sam




On Fri, Mar 4, 2011 at 10:09 AM, Sam <u2nsam at gmail.com> wrote:

> Thanks,
>
> I got it, but calls disconnects when other person enters the conference,i
> guess it could be because of codec negotiation @ conference room.
>
> http://pastebin.freeswitch.org/15545
>
>
> Rgards
> Sam
>
>
> On Thu, Mar 3, 2011 at 12:26 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Wed, Mar 2, 2011 at 1:48 AM, Sam <u2nsam at gmail.com> wrote:
>>
>>> HI,
>>>
>>>
>>> Suppose if i want to place an external caller in the conference so i
>>> should use the below command ?
>>>
>>> originate freetdm/wp1/a/9322273640  67287006 context &bridge 7050
>>>
>>
>> What are 67287006 and "context"? if you want to transfer to ext 7050 in a
>> context named "67287006" then do this:
>> originate freetdm/wp1/a/9322273640  7050 XML 67287006
>>
>> remember that the second argument to originate is a dialplan destination
>> number OR an &application(arg)
>>
>> -MC
>>
>>
>>> where 7050 leads to conference .
>>>
>>>
>>> Regds
>>> Sam
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From jjj at 3js.de  Fri Mar  4 12:02:26 2011
From: jjj at 3js.de (Johannes Jakob)
Date: Fri, 4 Mar 2011 10:02:26 +0100
Subject: [Freeswitch-users] DTMF not passed through to client
In-Reply-To: <C9A6E059-78EE-4438-81CA-650A772E84CA@gmail.com>
References: <BFA68905-4F2E-4782-95CE-7C0D2CB01328@3js.de>
	<C9A6E059-78EE-4438-81CA-650A772E84CA@gmail.com>
Message-ID: <4FB4579E-8C22-42F4-A579-3E68C37B93BC@3js.de>

Hi Steven,

thanks for helping!

What "device" do you think is the problem? 
Are you talking about the Upstream? I don't think he can be the problem since it is a very large ISP in Germany with lots of clients not complaining ;)

>> Upstream >rfc2833> FreeSWITCH >rfc2833> Asterisk


I can't give you the raw pcap files, but I hope the information put up here: http://3js.de/20110304/ contains the information you need.


freeswitch at internal> version 
FreeSWITCH Version 1.0.head (git-7847289 2011-02-19 23-38-04 +0100)


BTW: this freeswitch box is a virtual machine, running on citrix xen dom0. could this be a problem?


Best regards,

  Johannes



On 04.03.2011, at 05:43, Steven Ayre wrote:

> Are you running the latest version?
> 
> It appears the dtmf is failing a sanity check and getting ignored as a result, possibly as a result of the device sending the wrong rtp timestamps. Can you get a tcpdump of the rtp stream?
> 
> Steve on iPhone
> 
> On 3 Mar 2011, at 22:50, Johannes Jakob <jjj at 3js.de> wrote:
> 
>> Hi,
>> 
>> what do I have to do to get RFC2833 DTMF passed to the client in an inbound scenario like this:
>> 
>> Upstream >rfc2833> FreeSWITCH >rfc2833> Asterisk
>> 
>> ?
>> 
>> The Upstream is sending the DTMF tones correctly:
>> 
>> ---
>> 329    2.114845    UPSTREAMIP    FS_IP    RTP EVENT    62    Payload type=RTP Event, DTMF Eight 8
>> ---
>> 
>> FreeSWITCH isn't logging much of the dtmf stuff, even with loglevel debug:
>> 
>> http://pastebin.freeswitch.org/15541
>> 
>> ...
>> Set 2833 dtmf receive payload to 101
>> ...
>> 2011-03-03 23:29:59.498466 [ERR] switch_rtp.c:282 Failed DTMF sanity check.
>> ...
>> 
>> 
>> User's directory entry includes:
>> 
>>     <param name="dtmf-type" value="rfc2833"/>
>>     <param name="pass-rfc2833" value="true"/>
>> 
>> I tried adding those in the gateway profile as well, no change.
>> 
>> 
>> On the asterisk box I don't see those rtpevents in the pcap trace anymore, they just vanish on the FS box.
>> 
>> 
>> Can somebody help me out here? Or point me to the right wiki page?
>> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#DTMF doesn't really tell me, what the options do and http://wiki.freeswitch.org/wiki/DTMF is something different.
>> 
>> 
>> Thanks and bye,
>> 
>> Johannes
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From benkokakao at gmail.com  Fri Mar  4 12:23:44 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Fri, 4 Mar 2011 10:23:44 +0100
Subject: [Freeswitch-users] FS Webinterfaces - Opinions on wikiPBX,
 blue.box, fusionPBX
In-Reply-To: <AANLkTikaGbQ+u-p=WjWLC0Hbw4PeZs0Aur4K33_Y9Rev@mail.gmail.com>
References: <AANLkTik-J2z-P7EvghBH1i-EPL-_G7FqDGX0PCvqYu+g@mail.gmail.com>
	<AANLkTikVTwM3FfsuxYfH5D3HTM5LnYJq9E5eCEkA4LEm@mail.gmail.com>
	<AANLkTiktKMFdEtknKJzC=h5g76HVBPEyOeOchRGdRXMm@mail.gmail.com>
	<AANLkTimKYJWyHP2aUgMOqRvvU-6dPyLNy-+iN7HPf8yS@mail.gmail.com>
	<AANLkTikHHbF-5Rbg4RhrRX9wY6eaZcgLVDK=BBmiQYUR@mail.gmail.com>
	<AANLkTinvCSBNew-K=OAqwiSRQEzB4TxQaEgpKiHhFDUb@mail.gmail.com>
	<AANLkTikaGbQ+u-p=WjWLC0Hbw4PeZs0Aur4K33_Y9Rev@mail.gmail.com>
Message-ID: <AANLkTimEs24fc8b6XJnTp_FTgD1oE3g5y=aGzzbjG43R@mail.gmail.com>

On 3 March 2011 19:45, Michael Collins <msc at freeswitch.org> wrote:
> As far as PHP vs. Python most of us just *shrug*. If you want to see someone
> doing stuff with Python then go check out Gabe Gundy's stuff.
> (http://gundy.org/)

Looks promising - looking forward to more tech- and in-depth-posts by
Gabriel! Any other FS-related blogs i should take a look at?
I'm still new to FreeSWITCH having only used it as the company-PBX for
my previous employer(Which was a breeze to implement and automatize),
but have a 3 year history with pre-fork OpenSER and Asterisk(And i'm
not so fond of the latter so i hope i can fulfill all requirements
with FS).

> Of course, the *best* GUI for FreeSWITCH is this one: 

Hehe, yeah, had already checked out your project before. The
screenshots look promising - so where can i find the sources? :-P

Regards,
Christian


From benkokakao at gmail.com  Fri Mar  4 12:40:16 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Fri, 4 Mar 2011 10:40:16 +0100
Subject: [Freeswitch-users] FS Webinterfaces - Opinions on wikiPBX,
 blue.box, fusionPBX
In-Reply-To: <AANLkTinvCSBNew-K=OAqwiSRQEzB4TxQaEgpKiHhFDUb@mail.gmail.com>
References: <AANLkTik-J2z-P7EvghBH1i-EPL-_G7FqDGX0PCvqYu+g@mail.gmail.com>
	<AANLkTikVTwM3FfsuxYfH5D3HTM5LnYJq9E5eCEkA4LEm@mail.gmail.com>
	<AANLkTiktKMFdEtknKJzC=h5g76HVBPEyOeOchRGdRXMm@mail.gmail.com>
	<AANLkTimKYJWyHP2aUgMOqRvvU-6dPyLNy-+iN7HPf8yS@mail.gmail.com>
	<AANLkTikHHbF-5Rbg4RhrRX9wY6eaZcgLVDK=BBmiQYUR@mail.gmail.com>
	<AANLkTinvCSBNew-K=OAqwiSRQEzB4TxQaEgpKiHhFDUb@mail.gmail.com>
Message-ID: <AANLkTim10UtuRrZZ20wAXk+MmuaHc9BY1jp56+bU1PgD@mail.gmail.com>

On 3 March 2011 19:22, Aza Tek <azatek0 at gmail.com> wrote:
> Going back to PHP?? Not sure that's a very wise move.
> WikiPBX is great, it just needs more people.
> If you have a bounty to spend on additional features, it'd be better spent
> on wikipbx.

If i had my way, i would hire a webdev and let him/her implement a GUI
which exactly fits our needs. But not doable yet, as explained above.

If there's a bounty, it would be for quickly hacking together a visual
representation of minor features which are not yet in the existing
GUIs - since it would take me four times as long as an experienced dev
to do it in PHP/FusionPBX, i mentioned the bounty(Where i don't even
know if i get a budget).

I'm still thinking to take a look at wikiPBX though, as there
unfortunately where no answers besides "this is better than that" ;-)
(Hope that's not because my initial mail was too fuzzy)

blue.box has released a virtualbox-image, so i'm going to try that today :-)

Regards,
Christian


From codecomplete at free.fr  Fri Mar  4 13:26:32 2011
From: codecomplete at free.fr (GillesToo)
Date: Fri, 4 Mar 2011 02:26:32 -0800 (PST)
Subject: [Freeswitch-users] [Dahdi/FreeTDM] Does FS provide call supervision?
Message-ID: <1299234392809-6087967.post@n2.nabble.com>

Hello,

Suprisingly, Asterisk doesn't provide call supervision when used with PCI
cards + FXO module + Zaptel/Dahdi.
http://www.voip-info.org/wiki/view/Asterisk+func+channel CHANNEL() , the
only function that is supposed to provide information, always returns "Up",
no matter if Asterisk is still dialing or the callee has actually answered
the phone.

I assume Freeswitch will have the same issue since FreeTDM relies on Dahdi
to work with this type of hardware, but I'd like confirmation of this.

Thank you.

PS: For those interested, here's the code I naively thought would let
Asterisk pause until the callee has gone off-hook:

======
;Down, Rsrvd, OffHook, Dialing, Ring, Ringing, Up, Busy, Dialing Offhook,
Pre-ring, 
Unknown
exten => start,n,Set(INDEX=0)

;tried "Up", "OffHook", and "Rsrvd", to no avail
exten => start,n,While($["${CHANNEL(state)}" != "Up" & ${INDEX} < 10])

exten => start,n,NoOp(Channel still ringing: ${CHANNEL(state)})
exten => start,n,Wait(1)
exten => start,n,Set(INDEX=$[${INDEX} + 1])

exten => start,n,EndWhile()
======

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dahdi-FreeTDM-Does-FS-provide-call-supervision-tp6087967p6087967.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From lloydie.t at gmail.com  Fri Mar  4 13:56:10 2011
From: lloydie.t at gmail.com (lloyd thomas)
Date: Fri, 4 Mar 2011 10:56:10 +0000
Subject: [Freeswitch-users] Dialplan in a muddle
In-Reply-To: <1B19ABD72889C245AE8EEE08AC24103A28C4231FB5@exmachina.office.kapper.net>
References: <AANLkTi=GE8X_FC7UOfxLvjP6EFzttjdsocLcqNNvRA2t@mail.gmail.com>
	<F05761CEA50FF14094E4F2A86C1517A504D453801B@sha-exch14.shared.ifeltd.com>
	<AANLkTimtrZzLUAjK2SG720N46PvvROJah_tWLfj07R6P@mail.gmail.com>
	<F05761CEA50FF14094E4F2A86C1517A504D4538025@sha-exch14.shared.ifeltd.com>
	<AANLkTin+DKWPWsANLJCnZOgW0pFRE+cQc2MZt5u92ebu@mail.gmail.com>
	<AANLkTi=41H_=eLX=qu88WqZV4H5vt_unjkevxMiU0-Mw@mail.gmail.com>
	<1B19ABD72889C245AE8EEE08AC24103A28C4231FB5@exmachina.office.kapper.net>
Message-ID: <AANLkTi=ym3+jHeNO-0qWX=t2kxD6q3bjZvgEY+zfHDc0@mail.gmail.com>

I had to reboot the FS box anyway and that seemed to sort it all out. I
suspect a profile rescan may have sorted it out

Thanks for all your help.

On 3 March 2011 21:21, Clemens Ebentheuer <ce at kapper.net> wrote:

> Hi,
>
>
>
> i?m no expert but you should see something like in sofia status:
>
>
>
> external::soho66       gateway       sip:1000022541 at sip.soho66.co.uk:8060
> TRYING (retry: 22s)
>
>
>
> maybe you need a:  sofia profile external rescan
>
> and/or: sofia profile external restart
>
>
>
> clemens
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *lloyd thomas
> *Sent:* Thursday, March 03, 2011 9:07 PM
> *To:* freeswitch-users
>
> *Subject:* Re: [Freeswitch-users] Dialplan in a muddle
>
>
>
> EXECUTE sofia/internal/200 at companya.tele.co.ukbridge(sofia/gateway/soho66/07973323260)
> 2011-03-03 19:57:22.189028 [ERR] mod_sofia.c:3891 Invalid Gateway
>
> So it looks as though my gateway does not register. The file is in
> conf/sip_profiles/external/01_companya_soho66.xml
> Don't understand what the problem is
>
> Help
>
> <include>
>   <gateway name="soho66">
>     <param name="username" value="1000022541" />
>     <param name="realm" value="sip.soho66.co.uk:8060" />
>     <param name="password" value="5J8D139O" />
>   </gateway>
> </include>
>
>
> sofia status
>                      Name
> Type                                       Data      State
>
> =================================================================================================
>                  internal       profile
> sip:mod_sofia at 188.222.84.32:5060      RUNNING (0)
>                  external       profile
> sip:mod_sofia at 188.222.84.32:5080      RUNNING (0)
>     external::example.com       gateway
> sip:joeuser at example.com      NOREG
>       companyb.tele.co.uk         alias
> internal      ALIASED
>             internal-ipv6       profile                   sip:mod_sofia@[::1]:5060
> RUNNING (0)
>     companya.tele.co.uk         alias
> internal      ALIASED
>             188.222.85.33         alias
> internal      ALIASED
>
> =================================================================================================
> 3 profiles 3 aliases
>
>
>
> On 3 March 2011 18:58, lloyd thomas <lloydie.t at gmail.com> wrote:
>
> No Joy. Looks like the may be something wrong with the initial
> companya.tele.co.uk.xml file.
> Have a look
>
>
>
> http://pastebin.freeswitch.org/15539
>
> ---------- Forwarded message ----------
> From: *Garry Matthews* <garry.matthews at ethermagic.co.uk>
> Date: 3 March 2011 18:08
> Subject: RE: [Freeswitch-users] Dialplan in a muddle
> To: lloyd thomas <lloydie.t at gmail.com>
>
> *-* <include>
>
> *-* <gateway name="*soho66*">
>
> * * <param name="*username*" value="*1000014877*" />
>
> * * <param name="*realm*" value="*sip.soho66.co.uk:8060*" />
>
> * * <param name="*password*" value="*C9J6U0LO*" />
>
> * * </gateway>
>
> * * </include>
>
>
>
> I?ve made a few changes for security reasons, but it should give you the
> idea !
>
>
>
> If you type >sofia status
>
>
>
> Then you should see REGED to show that you have registered.
>
>
>
> Any luck?
>
>
>
> Garry.
>
> *From:* lloyd thomas [mailto:lloydie.t at gmail.com]
> *Sent:* 03 March 2011 18:02
> *To:* Garry Matthews
> *Subject:* Re: [Freeswitch-users] Dialplan in a muddle
>
>
>
> Ah Ha, what is your gateway settings (roughly)?
>
> On 3 March 2011 17:51, Garry Matthews <garry.matthews at ethermagic.co.uk>
> wrote:
>
> I?m not sure if this helps ? I just started with Freeswicth and Soho 66
> today. I?m a total beginner and have been working from the book.
>
> However, I can get calls out using the following file -
>
>
>
> conf/dialplan/default/01_custom.xml
>
>
>
> <include>
>
>                 <extension name="Dial Out Custom Gateway">
>
>                 <condition field="destination_number"
>
>                                 expression="^9(0\d{10})$">
>
>                                 <action application="bridge"
>
>                                 data="sofia/gateway/soho66/$1"/>
>
>                 </condition>
>
>                 </extension>
>
> </include>
>
>
>
> You then need to -
>
>
>
> reloadxml
>
>
>
> This may give you something to compare with.
>
>
>
> I hope this helps.
>
>
>
> Garry.
>
>
>
> *From:* lloyd thomas [mailto:lloydie.t at gmail.com]
> *Sent:* 03 March 2011 17:04
> *To:* freeswitch-users
> *Subject:* [Freeswitch-users] Dialplan in a muddle
>
>
>
> Maybe getting a bit ahead of myself but I am trying to set up a
> multi-company dial plan. Things are working fine internally, but I am not
> getting the grasp for external calls.
>
> The first company xml file is conf/dialplan/companya.tele.co.uk.xml
>
> which has the line
> <X-PRE-PROCESS cmd="include" data="companya.tele.co.uk/*.xml"/>
> to include xml files in that conf/dialplan/companya.tele.co.uk folder
>
> In this folder I have the xml file soho66.co.uk.xml which includes the
> regex to remove the initial 9 when dialling and extenal number. I cant seem
> to be able to get a call to go anywhere near connecting to the soho gateway.
>
> I think problem is to do with this
>
> 2011-03-03 17:00:23.089153 [NOTICE] switch_ivr.c:1606 Transfer
> sofia/internal/200 at companya.tele.co.uk to enum[
> 907973323260 at companya.tele.co.uk]
>
> help me please!
>
> ----------------- soho66.co.uk.xml----------------------------
> <include>
> <extension name="Local Dial">
>    <condition field="destination_number" expression="^9?([2-9]{6})$">
>      <action application="set" data="effective_caller_id_name=12345678"/>
>      <action application="set" data="effective_caller_id_number=12345678"/>
>      <action application="bridge" data="sofia/gateway/
> sip.soho66.co.uk/01869$1 at sip.soho66.co.uk"/>
>    </condition>
> </extension>
>
> <extension name="Regional 01">
>   <condition field="destination_number" expression="^9?(01\d{9})$">
>      <action application="set" data="effective_caller_id_name=12345678"/>
>      <action application="set" data="effective_caller_id_number=12345678"/>
>      <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1
> "/>
>   </condition>
> </extension>
>
> <extension name="Regional 02">
>   <condition field="destination_number" expression="^9?(02\d{9})$$">
>      <action application="set" data="effective_caller_id_name=12345678"/>
>      <action application="set"
> data="effective_caller_id_number=123456781"/>
>      <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1
> "/>
>   </condition>
> </extension>
>
> <extension name="Mobile">
>   <condition field="destination_number" expression="^9?(07\d{9})$">
>      <action application="set" data="effective_caller_id_name=12345678"/>
>      <action application="set" data="effective_caller_id_number=12345678"/>
>      <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1
> "/>
>   </condition>
> </extension>
>
> <extension name="International">
>   <condition field="destination_number" expression="^9?(00\d*))$">
>      <action application="set" data="effective_caller_id_name=12345678"/>
>      <action application="set" data="effective_caller_id_number=12345678"/>
>      <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1
> "/>
>   </condition>
> </extension>
> </include>
> -----------------eof soho66.co.uk.xml-------------------------
>
> -------------conf/sip_profiles/external/companya_soho66.xml
> -----------------------
> <include>
>       <gateway name="sip.soho66.co.uk">
>       <param name="username" value="12345678"/>
>       <param name="from-user" value="12345678"/>
>       <param name="password" value="ABCDEFGH"/>
>       <param name="expire-seconds" value="600"/>
>       <param name="register" value="true"/>
>       <param name="retry-seconds" value="30"/>
>     </gateway>
> </include>
> ------------- eof conf/sip_profiles/external/companya_soho66.xml
> -----------------------
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From boris at tagnet.ru  Fri Mar  4 14:00:11 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Fri, 04 Mar 2011 16:00:11 +0500
Subject: [Freeswitch-users] What is wrong with my extension?
Message-ID: <4D70C63B.7080005@tagnet.ru>

Hello!

May someone tell me why the 73435 is stripped ???  And if I doing 
something wrong what is the right way. I'm useing FreeSWITCH Version 
1.0.head (git-1c95ad9 2011-01-20 22-43-50 -0300)


     Here is my extension:

<!--
             Transfer call to the TOP context
         -->
<extension name="top.ctx">
<condition field="${v_tagnet_dst}" expression="^([0-9]+)$">
<action application="log" data="ALERT [top.ctx] - transfer to top.ctx"/>
<action application="log" data="ALERT [top.ctx] ORIG: ${v_tagnet_dst} 
REGEX: $1"/>

<!-- Caller ID normalization -->
<action application="execute_extension" data="ext_translate_src XML 
features"/>

<action application="log" data="ALERT [top.ctx] - transfer to top.ctx $1"/>
<action application="transfer" data="$1 XML top.ctx"/>
</condition>
</extension>


Here is the output:
2011-03-04 15:56:42.127837 [ALERT] mod_dptools.c:1174 [top.ctx] - 
transfer to top.ctx
2011-03-04 15:56:42.129862 [ALERT] mod_dptools.c:1174 [top.ctx] ORIG: 
73435230020 REGEX: 230020
-- 

? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From julf at julf.com  Fri Mar  4 14:42:04 2011
From: julf at julf.com (Johan Helsingius)
Date: Fri, 04 Mar 2011 12:42:04 +0100
Subject: [Freeswitch-users] checking voicemail without ID?
In-Reply-To: <4D6F759E.8040401@julf.com>
References: <4D6E555F.4000003@julf.com>	<649CD6B6-AB32-41B0-8C9D-2BB5F3DEA9FD@freeswitch.org>	<4D6E63FE.1020302@julf.com>	<518984EC-1B52-4631-B3D5-2DB6776255C5@freeswitch.org>	<4D6E6921.9000604@julf.com>	<AANLkTi=gj4QzuR2AAdMphwxpzOfSu9B1Fy4HHWyA8-YN@mail.gmail.com>	<4D6F6E54.1000303@julf.com>	<549CFEF87AEDE841A38E9D15EAB4C04C58B2BBB396@cooper>
	<4D6F759E.8040401@julf.com>
Message-ID: <4D70D00C.6030801@julf.com>

OK, found the problem. Egg on my face, as expected...
Hopefully my error will be a warning for others
that might fall for the same trap.

As I modified the default configuration, I moved away
the original conf/dialplan/default/examples.xml by
moving it up to the conf/dialplan directory, to still
have it close at hand for reference, but thinking it
would not be included  (based on the include of
default/*.xml in default.xml), but I did not
realize that conf/freeswitch.xml does an include
of dialplan/*.xml...

Mea culpa!

	Julf


From oa at estation.dk  Fri Mar  4 11:29:26 2011
From: oa at estation.dk (=?ISO-8859-1?Q?=D8yvind_Albrigtsen?=)
Date: Fri, 04 Mar 2011 09:29:26 +0100
Subject: [Freeswitch-users] Snom and Aastra one-way audio after picking
 up call that has been on hold (NAT)
In-Reply-To: <30915FCC-D7B4-4BEC-850F-BA321615A5D6@freeswitch.org>
References: <4D6F596C.7010403@estation.dk>
	<30915FCC-D7B4-4BEC-850F-BA321615A5D6@freeswitch.org>
Message-ID: <4D70A2E6.4030703@estation.dk>

Here you go.

5328XXXX is the number the phone is called from (coming from a PSTN 
gateway without any NAT).


On 03/03/2011 11:34 PM, Brian West wrote:
> sip trace please.
>
> /b
>
> On Mar 3, 2011, at 3:03 AM, ?yvind Albrigtsen wrote:
>
>    
>> The calls get in like this:
>> PSTN-gw ->  FS ->  NAT ->  Snom/Aastra
>>
>> It works fine when I call out from the phones and set the call on hold
>> (on the Snom/Aastra phones), though.
>>
>> This is on FreeSWITCH Version 1.0.head (git-cb6f1ed 2011-02-22 20-25-16
>> -0500)
>>
>>
>> Regards
>> Oyvind
>>      
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>    

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From hkalyoncu at gmail.com  Fri Mar  4 16:14:29 2011
From: hkalyoncu at gmail.com (hkalyoncu)
Date: Fri, 4 Mar 2011 05:14:29 -0800 (PST)
Subject: [Freeswitch-users] mod_python
Message-ID: <1299244469418-6088430.post@n2.nabble.com>

hello

im trying to write a dynamic dialplan module with mod_python. 
i did the dialplan binding of my module in python configuration file.
and im using xml_fetch in my module. everything is ok up to that.

But i cannot find how i can access channel variables from my module?
(absolutely no docs for this) 
can someone help how can i dump all channel variables to a file from my
module?


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-python-tp6088430p6088430.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Fri Mar  4 17:56:23 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 4 Mar 2011 14:56:23 +0000
Subject: [Freeswitch-users] mod_python
In-Reply-To: <1299244469418-6088430.post@n2.nabble.com>
References: <1299244469418-6088430.post@n2.nabble.com>
Message-ID: <AANLkTikQ4NVFE2BLNYxiY05zFaYeq8TJjqbuZJKp3OgG@mail.gmail.com>

There's documentation at http://wiki.freeswitch.org/wiki/Mod_python

If you're running a script as an app from the dialplan you get a
session object and can do session.getVariable("varname").

For the dialplan binding, you don't have a session object so there's
no way to get the variables that way. I assume a session has been
created by then, but it's simply not passed through the binding API.

What do you get for param1 & param in "def xml_fetch(param1, param2)"
- perhaps they contain a UUID that you can use?

-Steve



On 4 March 2011 13:14, hkalyoncu <hkalyoncu at gmail.com> wrote:
> hello
>
> im trying to write a dynamic dialplan module with mod_python.
> i did the dialplan binding of my module in python configuration file.
> and im using xml_fetch in my module. everything is ok up to that.
>
> But i cannot find how i can access channel variables from my module?
> (absolutely no docs for this)
> can someone help how can i dump all channel variables to a file from my
> module?
>
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-python-tp6088430p6088430.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From tculjaga at gmail.com  Fri Mar  4 18:41:51 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 4 Mar 2011 16:41:51 +0100
Subject: [Freeswitch-users] mod_say and bargein
Message-ID: <AANLkTina6BuY5fHvOTYWtNqM4s0FuKisOWHVVLCq2SkM@mail.gmail.com>

hello, is there any way to barge in for things being played by mod_say ?


lets say .... you are executing a msgcount macro ...


<action application="set" data="playback_terminators=#"/>
<action application="phrase" data="msgcount:1234567890"/>


when i press # during execution it just suppress 1 prompt than the next one
kicks in ...

is there any way to do it for entire msgcount macro ?
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From aligzaidi at gmail.com  Fri Mar  4 18:51:55 2011
From: aligzaidi at gmail.com (Ali G. Zaidi)
Date: Fri, 4 Mar 2011 10:51:55 -0500
Subject: [Freeswitch-users] Dialing extensions behind NAT
In-Reply-To: <0FD8DCDC-EDD4-42BA-AE05-C8A7773F1552@gmail.com>
References: <AANLkTik_Q9DsUNeb7ogkgkt5ykSWv+4rE98oZu_OtRdv@mail.gmail.com>
	<FBFB4BE1-4856-4C93-8CE4-044D051AF6C6@freeswitch.org>
	<3BA5114D-BF0A-44EE-9229-8BFA3FE78CCE@gmail.com>
	<0FD8DCDC-EDD4-42BA-AE05-C8A7773F1552@gmail.com>
Message-ID: <BCB5ED82-5737-4D1A-8949-BCAD8A833035@gmail.com>

Steve, thanks for your reply... I'm planning to add more UA's in behind different NATs. For example 5 UAs in each location. California, UK and New York while FreeSwitch sitting in Atlanta. 

Sent from my iPhone

On Mar 3, 2011, at 11:31 PM, Steven Ayre <steveayre at gmail.com> wrote:

> <action application="set" data="bypass_media=true"/> before bridge.
> 
> You'll need to be sure they aren't behind different nat routers though.
> 
> Steve on iPhone
> 
> On 4 Mar 2011, at 02:11, "Ali G. Zaidi" <aligzaidi at gmail.com> wrote:
> 
>> Sorry my question wasn't clear enough, so I ask again.
>> UA1,UA2,UA3===(NAT) ====(Internet)====(FreeSwitch)===(PSTN)
>> 
>>  
>> 
>> When UA1 dial UA2 the traffic go over the internet connection to FreeSwitch and then comes back to UA2. Can UA1 dial UA2 without going to internet? In other words, after SIP signaling, RTP traffic flow between UAs without FreeSwitch in between? I know in Asterisk you can configure UA with "caninvite" enable, wondering if FS also able to this and how?
>> 
>> Sent from my iPhone
>> 
>> 
>> On Mar 3, 2011, at 5:58 PM, Brian West <brian at freeswitch.org> wrote:
>> 
>>> give each phone a different client side sip port.
>>> 
>>> /b
>>> 
>>> On Mar 3, 2011, at 3:44 PM, Ali Zaidi wrote:
>>> 
>>>> My both phones are Linksys SPA942, please let me know any good phones that
>>>> can work best with free switch over wan connection.
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From david.villasmil.work at gmail.com  Fri Mar  4 20:44:09 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 4 Mar 2011 18:44:09 +0100
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
Message-ID: <AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>

Hello,

I do the route selecting with a lua script. overflow and load distribution
with mod_distributor loaded via curl.

the prepaid side is done with nibble, yes. But i don't like it too much, i
might just deduct the balance when the call disconnects and let the
authorization block new calls, so the balance might go under 0 a little...




On Sun, Feb 27, 2011 at 9:06 PM, Saeed Ahmed <saeedahmad1981 at gmail.com>wrote:

> press sent too quick..
>
> what did you use for routing? curl? esl?
>
> did you use nibble bill for prepaid app?
>
>
> On Sun, Feb 27, 2011 at 9:05 PM, Saeed Ahmed <saeedahmad1981 at gmail.com>wrote:
>
>> Great!
>>
>> want to see it soon.
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.villasmil.work at gmail.com  Fri Mar  4 20:45:30 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 4 Mar 2011 18:45:30 +0100
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <AANLkTim0vpXhg16p6LTwbYa7ng5Ki-5-aw=MpfZOT3pY@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTim0vpXhg16p6LTwbYa7ng5Ki-5-aw=MpfZOT3pY@mail.gmail.com>
Message-ID: <AANLkTimoAGNBvi4mDjU7_DsoTsyanNVVjSDfwOWUr1eJ@mail.gmail.com>

Hello,

No, i haven't uploaded it yet (i still don't know how); as it is still not
finished. I might be doing so in a week or so.

Cheers

david

On Fri, Mar 4, 2011 at 12:45 AM, Avi Marcus <avi at avimarcus.net> wrote:

> Hi - I didn't notice this in my latest git contrib pull today. Did you get
> the access worked out?
> -Avi
>
> On Sun, Feb 27, 2011 at 10:06 PM, Saeed Ahmed <saeedahmad1981 at gmail.com>wrote:
>
>> press sent too quick..
>>
>> what did you use for routing? curl? esl?
>>
>> did you use nibble bill for prepaid app?
>>
>>
>> On Sun, Feb 27, 2011 at 9:05 PM, Saeed Ahmed <saeedahmad1981 at gmail.com>wrote:
>>
>>> Great!
>>>
>>> want to see it soon.
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From simpot at simpot.com  Fri Mar  4 20:24:43 2011
From: simpot at simpot.com (Dmitry Saratsky)
Date: Fri, 4 Mar 2011 19:24:43 +0200
Subject: [Freeswitch-users] LUA FS's API function "freeswitch.email" do not
	work
Message-ID: <52114D605A462A4E9A50E588EE7D0686012E575D9EDF@mail.forest.simpot.com>

Hi all,

According to: http://wiki.freeswitch.org/wiki/Lua#freeswitch.email
I should successfully send emails from within FS with following script:
19:19 fs:/usr/local/freeswitch/scripts# cat test.lua
freeswitch.email("user at domain.com",
                 "user at domain.com",
                 "subject: Voicemail from 1234\n",
                 "Hello,\n\nYou've got a voicemail, click the attachment to listen to it.")
19:19 fs:/usr/local/freeswitch/scripts#

When I run script, I can see in console:
freeswitch at default> lua test.lua
-ERR no reply

2011-03-04 19:20:46.783430 [DEBUG] switch_utils.c:709 Emailed data to [user at domain.com]
freeswitch at default>

I have the following configured in my autoload_configs/switch.conf.xml:
    <param name="mailer-app" value="sendmail"/>
    <param name="mailer-app-args" value="-t"/>

However nothing passed to sendmail... it is no records in /var/log/maillog.

I run:
CentOS release 5.5 (Final)
FreeSWITCH Version 1.0.head (git-53fc3f7 2011-02-28 12-43-05 -0600)

Any ideas?
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From leon at scarlet-internet.nl  Fri Mar  4 21:06:24 2011
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Fri, 4 Mar 2011 19:06:24 +0100
Subject: [Freeswitch-users] static rtpmap entries are not shown in outbound
	invites
Message-ID: <DD73E3C2-BFA2-406E-B3B5-0078AECCD329@scarlet-internet.nl>

Hi,

When I'm using late negotiation and in my dialplan do the following:

<action application="export" data="codec_string=${ep_codec_string}"/>
<action application="set" data="inherit_codec=true"/>

(I also tried with absolute_codec_string instead of codec_string)

Then the codecs (or more precisely said, the rtpmap lines) that are offered in the SDP of the A leg are not offered to the b-leg. Only the RTP payload numbers as defined by IANA are there in the media (m) header.

For example:


a-leg SDP in INVITE:

m=audio 16864 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.


b-leg SDP in INVITE:

m=audio 24148 RTP/AVP 18 101 13.
a=rtpmap:101 telephone-event/8000.


Is there a reason why FS removes the G729 rtpmap, even though I exported all into the codec_string chan var ?

Now, this would normally be fine because G729 being 18 is a static rtpmap, but I have one endpoint that has the behavior that if I send an invite with RTP payload 18 and omit the rtpmap being G729, then it responds with a 183 session progress containing rtpmap G.729 (notice the dot) - this is wrong as far as I can tell.

Apparently they respond with the correct G729 (without the dot) in their 183 sess progress if I include the rtpmap in my invite.

I believe I read somewhere there was an option settable (on a sip profile?) that enables to always send all payloads as defined in the media (m) sdp header as a=rtpmap lines, but I cannot find the option anymore (or perhaps I'm mistaken and saw it somewhere else ?)

Can someone help me with this ?

Thanks & kind regards,

Leon

From brian at freeswitch.org  Fri Mar  4 21:09:24 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 4 Mar 2011 12:09:24 -0600
Subject: [Freeswitch-users] static rtpmap entries are not shown in
	outbound invites
In-Reply-To: <DD73E3C2-BFA2-406E-B3B5-0078AECCD329@scarlet-internet.nl>
References: <DD73E3C2-BFA2-406E-B3B5-0078AECCD329@scarlet-internet.nl>
Message-ID: <0D61D75E-A552-49F3-8A76-F3010BBE32B7@freeswitch.org>

verbose_sdp=true

docs/ChangeLog: mod_sofia: apparently some sip device vendors did not read the RFC (who knew?) adding verbose_sdp=true var to add needless a= lines for standard iana codecs that explicitly do not require them (r:6c4f49a8)

/b

On Mar 4, 2011, at 12:06 PM, Leon de Rooij wrote:

> Can someone help me with this ?



From msc at freeswitch.org  Fri Mar  4 21:49:22 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 4 Mar 2011 10:49:22 -0800
Subject: [Freeswitch-users] Segfault on startup when mod_freetdm is
	enabled.
In-Reply-To: <AANLkTinFixtmr_A29JFm1TDyTUYs4QeBC8gJFg5OncEw@mail.gmail.com>
References: <AANLkTinFixtmr_A29JFm1TDyTUYs4QeBC8gJFg5OncEw@mail.gmail.com>
Message-ID: <AANLkTimV68sLsDY6azRXxxasMo63sto=h3yqEyhz8XRh@mail.gmail.com>

Definitely open a bug report. You can safely assume that a segfault is
worthy of opening a Jira. Gather all the relevant info and open the bug. Moy
should be able to have a look relatively soon.

-MC

On Wed, Mar 2, 2011 at 2:49 PM, Eric Michel <eric at bmcrministries.org> wrote:

> I updated to the latest git last night, and started having issues.  I
> assumed I'd screwed up and (not being very adept at unistalling things from
> source) decided to do a clean install today (including the OS).
>
> When I enable mod_freetdm in modules.conf.xml, freeswitch segfaults on
> startup.  If I go back and disable mod_freetdm it starts fine.  As this is a
> clean install the only thing I've changed is modules.conf.xml.  Is any one
> else having this problem?  Do I need to file a bug report or am I just
> missing something really simple?
>
> I'm running FreeSWITCH Version 1.0.head(git-64806d2 2011-03-02 18-23-19
> +0100).
> I installed the latest non-beta version of Wanpipe which is 3.5.18, and
> used "make freetdm" when I compiled it.
> I've got the Sagnoma A200 card.
> The OS is CentOS 5.5 64bit, AMD Athalon 7850 Dual-core processor, 2Gb RAM,
> 2 500Gb HD in a software RAID1.
>
> Before I compiled freeswitch I enabled "../../libs/freetdm/mod_freetdm" and
> "asr_tts/mod_flite" in modules.conf.
> I've run wancfg_fs.
>
>
>
>
> Eric
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From aligzaidi at gmail.com  Fri Mar  4 21:56:49 2011
From: aligzaidi at gmail.com (Ali G. Zaidi)
Date: Fri, 4 Mar 2011 13:56:49 -0500
Subject: [Freeswitch-users] Good Colo
Message-ID: <235EF63C-9AE0-458F-9EAD-1B180EB03566@gmail.com>

Hi,

Can someone recommend me a provider that can rent me physical Linux server for voice traffic. My end-points will be allover the USA.  

Thanks,

AGZ

Sent from my iPhone

From msc at freeswitch.org  Fri Mar  4 22:06:06 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 4 Mar 2011 11:06:06 -0800
Subject: [Freeswitch-users] API command
In-Reply-To: <AANLkTimFJNBzt6_n-LAxpH+Rjozojd_W2yEO0eLJ5bTn@mail.gmail.com>
References: <AANLkTim-dEY_DOLQPUf9apOwFyntFsNVPhnvUXOp9_2d@mail.gmail.com>
	<AANLkTinkxPVyTAQDFnoatZzJv1imWQWE5zxUfX4vNY76@mail.gmail.com>
	<AANLkTimL1hHZPFS49iKFhgLiBcQthBV1UiOQpfsbudpD@mail.gmail.com>
	<AANLkTimFJNBzt6_n-LAxpH+Rjozojd_W2yEO0eLJ5bTn@mail.gmail.com>
Message-ID: <AANLkTinraH9cxjJpYEHUBbzgcp3WKZG1eyoU6cAdvXn3@mail.gmail.com>

On Thu, Mar 3, 2011 at 8:48 PM, Sam <u2nsam at gmail.com> wrote:

> It disconnected because unknowingly # was pressed and got
> disconnected,thanks.
>
> How a moderator can ( mute and kick ) a particular person in the conference
> room, is there any method
>

Well, you can do this from the fs_cli and/or event socket. You could, in
theory, do it with moderator controls, but you would have to create some
method for the moderator to be able to specify which user to
kick/mute/unmute using only DTMFs. Not impossible, but not something for a
beginner.

-MC
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From msc at freeswitch.org  Fri Mar  4 22:13:01 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 4 Mar 2011 11:13:01 -0800
Subject: [Freeswitch-users] What is wrong with my extension?
In-Reply-To: <4D70C63B.7080005@tagnet.ru>
References: <4D70C63B.7080005@tagnet.ru>
Message-ID: <AANLkTin4ZVJGNbMmVwCtrc0Po-G7xpzP8qHADe4RmRAq@mail.gmail.com>

Boris,

I recommend that you pastebin the entire debug output of the call, including
all the dialplan matching stuff. That will help narrow down what is
happening. Use pastebin.freeswitch.org and put the link in this thread.

-MC

On Fri, Mar 4, 2011 at 3:00 AM, Boris Kovalenko <boris at tagnet.ru> wrote:

> Hello!
>
> May someone tell me why the 73435 is stripped ???  And if I doing
> something wrong what is the right way. I'm useing FreeSWITCH Version
> 1.0.head (git-1c95ad9 2011-01-20 22-43-50 -0300)
>
>
>     Here is my extension:
>
> <!--
>             Transfer call to the TOP context
>         -->
> <extension name="top.ctx">
> <condition field="${v_tagnet_dst}" expression="^([0-9]+)$">
> <action application="log" data="ALERT [top.ctx] - transfer to top.ctx"/>
> <action application="log" data="ALERT [top.ctx] ORIG: ${v_tagnet_dst}
> REGEX: $1"/>
>
> <!-- Caller ID normalization -->
> <action application="execute_extension" data="ext_translate_src XML
> features"/>
>
> <action application="log" data="ALERT [top.ctx] - transfer to top.ctx $1"/>
> <action application="transfer" data="$1 XML top.ctx"/>
> </condition>
> </extension>
>
>
> Here is the output:
> 2011-03-04 15:56:42.127837 [ALERT] mod_dptools.c:1174 [top.ctx] -
> transfer to top.ctx
> 2011-03-04 15:56:42.129862 [ALERT] mod_dptools.c:1174 [top.ctx] ORIG:
> 73435230020 REGEX: 230020
> --
>
>
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From msc at freeswitch.org  Fri Mar  4 22:22:57 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 4 Mar 2011 11:22:57 -0800
Subject: [Freeswitch-users] FreeTDM and Libpri with a Digium TE122 ...
 some calls not being answered...
In-Reply-To: <AANLkTimb3rLWvfZ=dTBM9XZeLa=OenJoz=zjRjdoD6Ym@mail.gmail.com>
References: <AANLkTimb3rLWvfZ=dTBM9XZeLa=OenJoz=zjRjdoD6Ym@mail.gmail.com>
Message-ID: <AANLkTikuBBd2UqfdASK+7ChizOu=5c7xhQYaHnf2LpYJ@mail.gmail.com>

Well...

I examined these Q931 traces and I don't see anything different between the
failed call and the successful call. I would hop into #freetdm on
irc.freenode.net and see if moy is around. He might be able to shed some
light on this. If it's not occurring at the libpri level then he might be
able to help figure out if FreeTDM is not behaving properly.

-MC

On Wed, Mar 2, 2011 at 9:13 PM, Matt Paine
<mattzerah+freeswitch at gmail.com>wrote:

> Hi Guys.
>
> Current setup is a FS box with a Digium TE122 installed, with DAHDI kernel
> drivers and using FreeTDM with a libpri span...
>
> == autoload_configs/freetdm.conf.xml ==
>
> <configuration name="freetdm.conf" description="FreeTDM Configuration">
>     <settings>
>         <param name="debug" value="0"/>
>         <param name="sip_headers" value="true"/>
>     </settings>
>     <libpri_spans>
>         <span name="ISDN">
>             <param name="mode" value="user"/>
>             <param name="dialect" value="q931"/>
>              <param name="dialplan" value="XML"/>
>             <param name="context" value="public"/>
>         </span>
>     </libpri_spans>
> </configuration>
>
> == freetdm.conf ==
>
> [general]
> cpu_monitor => yes
> cpu_monitoring_interval => 1000
> cpu_set_alarm_threshold => 80
> cpu_reset_alarm_threshold => 70
> cpu_alarm_action => warn
>
> [span zt ISDN]
> trunk_type => E1
> b-channel => 1-15
> d-channel => 16
> b-channel => 17-21
>
> == zt.conf ==
> [defaults]
> codec_ms => 20
> wink_ms => 150
> flash_ms => 750
> echo_cancel_level => 128
> rxgain => 0.0
> txgain => 0.0
>
> == /etc/dahdi/system.conf ==
> span=1,1,0,ccs,hdb3,crc4
> # termtype: te
> bchan=1-15,17-31
> dchan=16
> echocanceller=mg2,1-15,17-31
> # Global data
> loadzone    = au
> defaultzone = au
>
>
> =========
> And finally some logs from two calls, a successful one, and a failed one...
>  http://pastebin.freeswitch.org/15533
>
>
> If anyone could offer any suggestions as to what I can do to answer every
> incoming call into the box that would very appreciated. Of course if I need
> to provide more information I am willing to, and any suggestions for
> settings that need changing to test out will also be great.
>
> Thank you in advance
> Matt
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fs-list at communicatefreely.net  Fri Mar  4 22:36:07 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 04 Mar 2011 14:36:07 -0500
Subject: [Freeswitch-users] count
In-Reply-To: <AANLkTinyzQ4gOm6JZvMzP66pkW9nODqRuE6Pc5+ps2d+@mail.gmail.com>
References: <AANLkTi=oKxBtho8q5kQq52ZxrU=r++SzFXkVzjq=P8B1@mail.gmail.com>
	<AANLkTinyzQ4gOm6JZvMzP66pkW9nODqRuE6Pc5+ps2d+@mail.gmail.com>
Message-ID: <4D713F27.20907@communicatefreely.net>

conference_list (api) will give you a list of members, (use the cli help 
to figure these out).  You would have to count them in your script logic.
You can also use mod_limit, set to an arbitrarily high value, then use 
limit_usage to see the current count.
To have an adjustable maximum user limit after the conference starts, 
you would have to have that threshold set in a database somewhere.  Set 
up your dialplan so the threshold is compared against the current number 
of participants whenever someone calls into the conference room.  If you 
reduced the limit, it wouldn't kick anybody though, it would just keep 
more people from coming in until the count got back down below the 
threshold.

Hope that's helpful.

-Tim

Sam wrote:
> And how to specify the threshold for the conference on the fly when 
> the someone hits the conference dial-plan.
>
> Regards
> Sam
>
> On Thu, Mar 3, 2011 at 10:17 AM, Sam <u2nsam at gmail.com 
> <mailto:u2nsam at gmail.com>> wrote:
>
>     Hello,
>
>     Whats the command to get the participants count in the conference ?
>
>     Rwgards
>     Sam
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From jjj at 3js.de  Fri Mar  4 23:33:57 2011
From: jjj at 3js.de (Johannes Jakob)
Date: Fri, 4 Mar 2011 21:33:57 +0100
Subject: [Freeswitch-users] DTMF not passed through to client
In-Reply-To: <4FB4579E-8C22-42F4-A579-3E68C37B93BC@3js.de>
References: <BFA68905-4F2E-4782-95CE-7C0D2CB01328@3js.de>
	<C9A6E059-78EE-4438-81CA-650A772E84CA@gmail.com>
	<4FB4579E-8C22-42F4-A579-3E68C37B93BC@3js.de>
Message-ID: <23361B69-E44D-4A65-A770-24F08A53661E@3js.de>

Hi,

Just to give feedback:

problem solved by updating to last git head, although I didn't find a significant change in the revision history about dtmf stuff except this one:
http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=b53a68484361c9581a26f6b090a58b0803bd412d

FreeSWITCH Version 1.0.head (git-7847289 2011-02-19 23-38-04 +0100)
=>
FreeSWITCH Version 1.0.head (git-8fe24a2 2011-03-04 12-28-41 -0600)


Thanks anyways for your thoughts!

Greets,
  Johannes


On 04.03.2011, at 10:02, Johannes Jakob wrote:

> Hi Steven,
> 
> thanks for helping!
> 
> What "device" do you think is the problem? 
> Are you talking about the Upstream? I don't think he can be the problem since it is a very large ISP in Germany with lots of clients not complaining ;)
> 
>>> Upstream >rfc2833> FreeSWITCH >rfc2833> Asterisk
> 
> 
> I can't give you the raw pcap files, but I hope the information put up here: http://3js.de/20110304/ contains the information you need.
> 
> 
> freeswitch at internal> version 
> FreeSWITCH Version 1.0.head (git-7847289 2011-02-19 23-38-04 +0100)
> 
> 
> BTW: this freeswitch box is a virtual machine, running on citrix xen dom0. could this be a problem?
> 
> 
> Best regards,
> 
>  Johannes
> 
> 
> 
> On 04.03.2011, at 05:43, Steven Ayre wrote:
> 
>> Are you running the latest version?
>> 
>> It appears the dtmf is failing a sanity check and getting ignored as a result, possibly as a result of the device sending the wrong rtp timestamps. Can you get a tcpdump of the rtp stream?
>> 
>> Steve on iPhone
>> 
>> On 3 Mar 2011, at 22:50, Johannes Jakob <jjj at 3js.de> wrote:
>> 
>>> Hi,
>>> 
>>> what do I have to do to get RFC2833 DTMF passed to the client in an inbound scenario like this:
>>> 
>>> Upstream >rfc2833> FreeSWITCH >rfc2833> Asterisk
>>> 
>>> ?
>>> 
>>> The Upstream is sending the DTMF tones correctly:
>>> 
>>> ---
>>> 329    2.114845    UPSTREAMIP    FS_IP    RTP EVENT    62    Payload type=RTP Event, DTMF Eight 8
>>> ---
>>> 
>>> FreeSWITCH isn't logging much of the dtmf stuff, even with loglevel debug:
>>> 
>>> http://pastebin.freeswitch.org/15541
>>> 
>>> ...
>>> Set 2833 dtmf receive payload to 101
>>> ...
>>> 2011-03-03 23:29:59.498466 [ERR] switch_rtp.c:282 Failed DTMF sanity check.
>>> ...
>>> 
>>> 
>>> User's directory entry includes:
>>> 
>>>    <param name="dtmf-type" value="rfc2833"/>
>>>    <param name="pass-rfc2833" value="true"/>
>>> 
>>> I tried adding those in the gateway profile as well, no change.
>>> 
>>> 
>>> On the asterisk box I don't see those rtpevents in the pcap trace anymore, they just vanish on the FS box.
>>> 
>>> 
>>> Can somebody help me out here? Or point me to the right wiki page?
>>> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#DTMF doesn't really tell me, what the options do and http://wiki.freeswitch.org/wiki/DTMF is something different.
>>> 
>>> 
>>> Thanks and bye,
>>> 
>>> Johannes
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 



From brian at freeswitch.org  Fri Mar  4 23:42:55 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 4 Mar 2011 14:42:55 -0600
Subject: [Freeswitch-users] DTMF not passed through to client
In-Reply-To: <23361B69-E44D-4A65-A770-24F08A53661E@3js.de>
References: <BFA68905-4F2E-4782-95CE-7C0D2CB01328@3js.de>
	<C9A6E059-78EE-4438-81CA-650A772E84CA@gmail.com>
	<4FB4579E-8C22-42F4-A579-3E68C37B93BC@3js.de>
	<23361B69-E44D-4A65-A770-24F08A53661E@3js.de>
Message-ID: <66EBA09B-5A1F-4645-9642-DA7DA522631F@freeswitch.org>

There was a one day period in the past two weeks that DTMF was a bit busted.  So always git pull.

/b

On Mar 4, 2011, at 2:33 PM, Johannes Jakob wrote:

> Hi,
> 
> Just to give feedback:
> 
> problem solved by updating to last git head, although I didn't find a significant change in the revision history about dtmf stuff except this one:
> http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=b53a68484361c9581a26f6b090a58b0803bd412d
> 
> FreeSWITCH Version 1.0.head (git-7847289 2011-02-19 23-38-04 +0100)
> =>
> FreeSWITCH Version 1.0.head (git-8fe24a2 2011-03-04 12-28-41 -0600)
> 
> 
> Thanks anyways for your thoughts!
> 
> Greets,
>  Johannes

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From fs-list at communicatefreely.net  Sat Mar  5 01:21:27 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 04 Mar 2011 17:21:27 -0500
Subject: [Freeswitch-users] Sip Info method
In-Reply-To: <AANLkTimg+hj8g7JYhbkLnRDbkuEnxLEqN9rvwAZQD-Nh@mail.gmail.com>
References: <AANLkTin=A5-vEeGHvm+tJ+epspoUvZOOey2YkeVBf1=M@mail.gmail.com>	<AANLkTinkgcN8q7CqwzEgc5URkw9YtBrxudcTQT+Ef6nP@mail.gmail.com>	<AANLkTikxPFHpMFzHCdvUHaiW7Xiog-o523apLFJ1Urqb@mail.gmail.com>	<AANLkTikQACVKZ30QY49NTESdRUtr6DD0+0D8Pa7bDKvG@mail.gmail.com>
	<AANLkTimg+hj8g7JYhbkLnRDbkuEnxLEqN9rvwAZQD-Nh@mail.gmail.com>
Message-ID: <4D7165E7.5050507@communicatefreely.net>

Maybe not quite what you are looking for, but do your phones have XML 
programmable buttons?

We are working on an XML (via http) -> ESL -> FreeSWITCH conference 
admin tool.
In addition to recording, you can also list / kick / mute people.

-Tim

Sam wrote:
> Thinking of doing it room specific, so that on the fly the moderator 
> of a particular room
> can initiate recording or disable it.
>
> Regards
> Sam
>
> On Tue, Mar 1, 2011 at 5:56 PM, Steven Ayre <steveayre at gmail.com 
> <mailto:steveayre at gmail.com>> wrote:
>
>     The conference app mixes audio from all parties, so on your session
>     you can only get the combined audio. If you can trigger the recording
>     on the other party's leg instead perhaps that'll work because it won't
>     have been mixed in yet. Not sure how to do that though, or if you can.
>
>     -Steve
>
>
>     On 1 March 2011 10:32, Sam <u2nsam at gmail.com
>     <mailto:u2nsam at gmail.com>> wrote:
>     > Yes ,
>     >
>     > But it says that 'So as of now there seems to be no way to restrict
>     > activating recording to one party.'
>     >
>     > Regds
>     > Sam
>     >
>     >
>     >
>     > On Tue, Mar 1, 2011 at 3:07 PM, Tjardick van der Kraan
>     > <tjardick at vanderkraan.net <mailto:tjardick at vanderkraan.net>> wrote:
>     >>
>     >> Hi Sam,
>     >> Did you check the wiki:
>     >>
>     >>
>     http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session#Activating_via_SIP_INFO
>     >> Has some info on this.
>     >> Regards,
>     >> Tjardick
>     >>
>     >> On Tue, Mar 1, 2011 at 7:29 AM, Sam <u2nsam at gmail.com
>     <mailto:u2nsam at gmail.com>> wrote:
>     >>>
>     >>> Hello,
>     >>>
>     >>> Can we start recording any conference bridge via sip info
>     method  ?
>     >>>
>     >>> It should happen for only that particular bridge users and not
>     all the
>     >>> bridges running on the application.
>     >>>
>     >>> For that, what parameters i should set in the SIP info method
>     to achieve
>     >>> this.
>     >>>
>     >>> Regards
>     >>> Sam
>     >>>
>     >>> _______________________________________________
>     >>> FreeSWITCH-users mailing list
>     >>> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >>> http://www.freeswitch.org
>     >>>
>     >>
>     >>
>     >> _______________________________________________
>     >> FreeSWITCH-users mailing list
>     >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >> http://www.freeswitch.org
>     >>
>     >
>     >
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     >
>     >
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From norm at voicenetwork.ca  Sat Mar  5 01:44:42 2011
From: norm at voicenetwork.ca (Norman Tomlins)
Date: Fri, 4 Mar 2011 17:44:42 -0500
Subject: [Freeswitch-users] Good Colo
In-Reply-To: <235EF63C-9AE0-458F-9EAD-1B180EB03566@gmail.com>
References: <235EF63C-9AE0-458F-9EAD-1B180EB03566@gmail.com>
Message-ID: <AANLkTimm6Mca9S6f7nnpST5ce86ei7woDySFCZ+z+e1L@mail.gmail.com>

Hi,

We offer some great co-location space, and we are currently offering a Dell
R210, Quad Core 2.93 Ghz X3470, 4GB DDR RAM, 500GB Hard Drive, and 10 TB of
monthly Transfer for $119.95 per month.   If you are interested just email
me at norm at voicenetwork.ca or PM me on the  IRC.

These server work great with FreeSwitch!


Norman Tomlins
Voice Network Inc. - Proud Sponsor of FreeSwitch !
http://www.VoiceNetwork.ca


On Fri, Mar 4, 2011 at 1:56 PM, Ali G. Zaidi <aligzaidi at gmail.com> wrote:

> Hi,
>
> Can someone recommend me a provider that can rent me physical Linux server
> for voice traffic. My end-points will be allover the USA.
>
> Thanks,
>
> AGZ
>
> Sent from my iPhone
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From cliff at develix.com  Sat Mar  5 02:14:38 2011
From: cliff at develix.com (Cliff Wells)
Date: Fri, 04 Mar 2011 15:14:38 -0800
Subject: [Freeswitch-users] mod_cdr_pg_csv
Message-ID: <1299280478.2640.220.camel@portable-evil>

Is this module still supported?   I added it to my modules.conf but
during compilation I get the following: 

Compiling /usr/local/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c ...
cc1: warnings being treated as errors
/usr/local/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: In function 'save_cdr':
/usr/local/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:225: warning: ISO C90 forbids mixed declarations and code
/usr/local/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:235: warning: ISO C90 forbids mixed declarations and code
/usr/local/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:248: warning: ISO C90 forbids mixed declarations and code
/usr/local/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:275: warning: ISO C90 forbids mixed declarations and code
make[5]: *** [mod_cdr_pg_csv.lo] Error 1

This seems odd since the Makefile includes modmake.rules which has
-std=c99.   I guess I could disable -pedantic, but I feel I'm starting
down a road that ends somewhere over my head.

Regards,
Cliff



From renjian at gmail.com  Sat Mar  5 02:41:56 2011
From: renjian at gmail.com (Jian Ren)
Date: Fri, 4 Mar 2011 18:41:56 -0500
Subject: [Freeswitch-users] How could I forward the SIP incoming call to one
	of the available extensions?
Message-ID: <AANLkTi=A-NSss214Qe=aiofh-qhofDKZMFBx3v8=DgDi@mail.gmail.com>

Hi,
Right now I only have one extension in the SIP configuration as the
destination. Could I put a list of extensions then FS forwards the call to
the first available one?
Thanks!
Jian
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From msc at freeswitch.org  Sat Mar  5 03:21:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 4 Mar 2011 16:21:36 -0800
Subject: [Freeswitch-users] mod_cdr_pg_csv
In-Reply-To: <1299280478.2640.220.camel@portable-evil>
References: <1299280478.2640.220.camel@portable-evil>
Message-ID: <AANLkTintN-qmFj0dy0V5nWbE-75U0rGcQe-TD8vS79HW@mail.gmail.com>

I just did a git pull and compiled it with no problem. Get the latest git
and try again, just to be sure it wasn't already fixed.
-MC

On Fri, Mar 4, 2011 at 3:14 PM, Cliff Wells <cliff at develix.com> wrote:

> Is this module still supported?   I added it to my modules.conf but
> during compilation I get the following:
>
> Compiling
> /usr/local/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c
> ...
> cc1: warnings being treated as errors
> /usr/local/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:
> In function 'save_cdr':
> /usr/local/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:225:
> warning: ISO C90 forbids mixed declarations and code
> /usr/local/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:235:
> warning: ISO C90 forbids mixed declarations and code
> /usr/local/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:248:
> warning: ISO C90 forbids mixed declarations and code
> /usr/local/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:275:
> warning: ISO C90 forbids mixed declarations and code
> make[5]: *** [mod_cdr_pg_csv.lo] Error 1
>
> This seems odd since the Makefile includes modmake.rules which has
> -std=c99.   I guess I could disable -pedantic, but I feel I'm starting
> down a road that ends somewhere over my head.
>
> Regards,
> Cliff
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From cliff at develix.com  Sat Mar  5 04:40:20 2011
From: cliff at develix.com (Cliff Wells)
Date: Fri, 04 Mar 2011 17:40:20 -0800
Subject: [Freeswitch-users] mod_cdr_pg_csv
In-Reply-To: <AANLkTintN-qmFj0dy0V5nWbE-75U0rGcQe-TD8vS79HW@mail.gmail.com>
References: <1299280478.2640.220.camel@portable-evil>
	<AANLkTintN-qmFj0dy0V5nWbE-75U0rGcQe-TD8vS79HW@mail.gmail.com>
Message-ID: <1299289220.2640.470.camel@portable-evil>

Yes, that fixed it.  I should have updated first, but I was trying to
avoid it since that also required updating the Sangoma drivers.

Thanks,
Cliff

On Fri, 2011-03-04 at 16:21 -0800, Michael Collins wrote:
> I just did a git pull and compiled it with no problem. Get the latest
> git and try again, just to be sure it wasn't already fixed.
> -MC
> 
> On Fri, Mar 4, 2011 at 3:14 PM, Cliff Wells <cliff at develix.com> wrote:
>         Is this module still supported?   I added it to my
>         modules.conf but
>         during compilation I get the following:
>         
>         Compiling /usr/local/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c ...
>         cc1: warnings being treated as errors
>         /usr/local/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: In function 'save_cdr':
>         /usr/local/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:225: warning: ISO C90 forbids mixed declarations and code
>         /usr/local/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:235: warning: ISO C90 forbids mixed declarations and code
>         /usr/local/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:248: warning: ISO C90 forbids mixed declarations and code
>         /usr/local/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:275: warning: ISO C90 forbids mixed declarations and code
>         make[5]: *** [mod_cdr_pg_csv.lo] Error 1
>         
>         This seems odd since the Makefile includes modmake.rules which
>         has
>         -std=c99.   I guess I could disable -pedantic, but I feel I'm
>         starting
>         down a road that ends somewhere over my head.
>         
>         Regards,
>         Cliff
>         
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
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From moises.silva at gmail.com  Sat Mar  5 05:24:23 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 4 Mar 2011 21:24:23 -0500
Subject: [Freeswitch-users] [Dahdi/FreeTDM] Does FS provide call
	supervision?
In-Reply-To: <1299234392809-6087967.post@n2.nabble.com>
References: <1299234392809-6087967.post@n2.nabble.com>
Message-ID: <AANLkTinnCCWPcHo62W=fn6-M-4BP=2VZPm0PFTMNBsY0@mail.gmail.com>

On Fri, Mar 4, 2011 at 5:26 AM, GillesToo <codecomplete at free.fr> wrote:

> I assume Freeswitch will have the same issue since FreeTDM relies on Dahdi
> to work with this type of hardware, but I'd like confirmation of this.
>
>
FreeTDM currently has the same limitation unless your telco/switch sends
polarity reverse on answer. If polarity reversal is enabled, then FreeTDM
will not put the channel in state UP until the telco reverses the polarity
to signal answer.

This has nothing to do with the dependency on DAHDI for Digium cards btw.

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
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From xyangni at gmail.com  Sat Mar  5 06:43:26 2011
From: xyangni at gmail.com (Yihui Li)
Date: Sat, 5 Mar 2011 03:43:26 +0000
Subject: [Freeswitch-users] skype to x-lite birdge hangup imediately when
	answered
Message-ID: <AANLkTik3De=vPS8VfZfcJCdQNjHc7EvjxW=uej33JFjT@mail.gmail.com>

Dear all,

I am tring skypopen with latest git-head.
There are just 1 skype instance. When forward skype call to 5000, it works
fine. It also works fine with originate skypopen/interface1/*****
&bridge(user/1001)

 But when I try to forward skype to 1001 with the following dial plan, call
hang up imediately when answered.

  <extension name="skypetest">
    <condition field="destination_number" expression="^(5100)$">
      <action application="bridge" data="user/1001"/>
    </condition>
  </extension>

I am working on l ubuntu 10.10 server edition and skype static 2.0.0.72

here is the skype configure
=============full mod_skypopen.conf.xml=====================
<configuration name="skypopen.conf" description="Skypopen Configuration">
  <global_settings>
    <param name="debug" value="8"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="default"/>
    <param name="destination" value="5100"/>
    <param name="skype_user" value="user1"/>
    <param name="report_incoming_chatmessages" value="true"/>
    <param name="silent_mode" value="false"/>
    <param name="write_silence_when_idle" value="true"/>
  </global_settings>
  <!-- one entry here per each skypopen interface -->
  <per_interface_settings>
    <interface id="1" name="interface1">
        <param name="X11-display" value="localhost:10.0"/>
        <param name="skype_user" value="niqizhi"/>
    </interface>
  </per_interface_settings>
</configuration>




the debug message of the call is listed below. May I ask what is the
possiable cause? Thanks.



2011-03-05 03:15:13.353532 [DEBUG] skypopen_protocol.c:173
[32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE] READING:
|||CALL 142 CONF_ID 0|||
2011-03-05 03:15:13.438727 [DEBUG] skypopen_protocol.c:173
[32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE] READING:
|||CALL 142 STATUS RINGING|||
2011-03-05 03:15:13.438727 [DEBUG] skypopen_protocol.c:595
[32b8f10|9350fb9] [DEBUG_SKYPE  595  ][interface1     ][IDLE,IDLE] NO ACTIVE
calls in this moment, skype_call 142 is RINGING, to ask PARTNER_DISPNAME and
PARTNER_HANDLE
2011-03-05 03:15:13.438727 [DEBUG] skypopen_protocol.c:1619
 [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][IDLE,IDLE] SENDING:
|||GET CALL 142 PARTNER_DISPNAME||||
2011-03-05 03:15:13.438727 [DEBUG] skypopen_protocol.c:1619
 [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][IDLE,IDLE] SENDING:
|||GET CALL 142 PARTNER_HANDLE||||
2011-03-05 03:15:13.576818 [DEBUG] skypopen_protocol.c:173
[32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE] READING:
|||CALL 142 PARTNER_DISPNAME Eric Ni|||
2011-03-05 03:15:13.609592 [DEBUG] skypopen_protocol.c:173
[32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE] READING:
|||CALL 142 PARTNER_HANDLE eric.nqz|||
2011-03-05 03:15:13.609592 [DEBUG] skypopen_protocol.c:506
[32b8f10|9350fb9] [DEBUG_SKYPE  506  ][interface1     ][IDLE,IDLE] Call 142
go to skypopen_partner_handle_ring
2011-03-05 03:15:13.609592 [DEBUG] mod_skypopen.c:2626
[32b8f10|9350fb9] [DEBUG_SKYPE  2626 ][interface1     ][IDLE,IDLE] NOT FOUND
2011-03-05 03:15:13.609592 [DEBUG] mod_skypopen.c:2244
[32b8f10|9350fb9] [DEBUG_SKYPE  2244 ][interface1     ][PRERING,IDLE] 2
SESSION_REQUEST c958d9be-46d6-11e0-bb71-d9b9e6428344
2011-03-05 03:15:13.609592 [DEBUG] mod_skypopen.c:244
 [32b8f10|9350fb9] [DEBUG_SKYPE  244  ][interface1     ][PRERING,IDLE]
codecs UP
2011-03-05 03:15:13.609592 [DEBUG] mod_skypopen.c:273
 [32b8f10|9350fb9] [DEBUG_SKYPE  273  ][interface1     ][PRERING,IDLE]
skypopen_tech_init SUCCESS
2011-03-05 03:15:13.609592 [NOTICE] switch_channel.c:812 New Channel
skypopen/interface1 [c958d9be-46d6-11e0-bb71-d9b9e6428344]
2011-03-05 03:15:13.609592 [DEBUG] mod_skypopen.c:2268 (skypopen/interface1)
State Change CS_NEW -> CS_INIT
2011-03-05 03:15:13.609592 [DEBUG] switch_core_session.c:1116 Send signal
skypopen/interface1 [BREAK]
2011-03-05 03:15:13.609592 [DEBUG] mod_skypopen.c:727
 [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PRERING,IDLE]
skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
2011-03-05 03:15:13.609592 [DEBUG] mod_skypopen.c:2280
[32b8f10|9350fb9] [DEBUG_SKYPE  2280 ][interface1     ][PRERING,IDLE]
new_inbound_channel
2011-03-05 03:15:13.610977 [DEBUG] switch_core_state_machine.c:320
(skypopen/interface1) Running State Change CS_INIT
2011-03-05 03:15:13.610977 [DEBUG] switch_core_state_machine.c:356
(skypopen/interface1) State INIT
2011-03-05 03:15:13.610977 [DEBUG] mod_skypopen.c:463 (skypopen/interface1)
State Change CS_INIT -> CS_ROUTING
2011-03-05 03:15:13.610977 [DEBUG] switch_core_session.c:1116 Send signal
skypopen/interface1 [BREAK]
2011-03-05 03:15:13.610977 [DEBUG] mod_skypopen.c:727
 [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PRERING,IDLE]
skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
2011-03-05 03:15:13.610977 [DEBUG] mod_skypopen.c:464
 [32b8f10|9350fb9] [DEBUG_SKYPE  464  ][interface1     ][PRERING,IDLE]
interface1 CHANNEL INIT c958d9be-46d6-11e0-bb71-d9b9e6428344
2011-03-05 03:15:13.610977 [DEBUG] switch_core_state_machine.c:356
(skypopen/interface1) State INIT going to sleep
2011-03-05 03:15:13.610977 [DEBUG] switch_core_state_machine.c:320
(skypopen/interface1) Running State Change CS_ROUTING
2011-03-05 03:15:13.610977 [DEBUG] switch_channel.c:1664
(skypopen/interface1) Callstate Change DOWN -> RINGING
2011-03-05 03:15:13.610977 [DEBUG] switch_core_state_machine.c:359
(skypopen/interface1) State ROUTING
2011-03-05 03:15:13.610977 [DEBUG] mod_skypopen.c:664
 [32b8f10|9350fb9] [DEBUG_SKYPE  664  ][interface1     ][PRERING,IDLE]
interface1 CHANNEL ROUTING
2011-03-05 03:15:13.610977 [DEBUG] switch_core_state_machine.c:77
skypopen/interface1 Standard ROUTING
2011-03-05 03:15:13.610977 [INFO] mod_dialplan_xml.c:331 Processing Eric Ni
<eric.nqz>->5100 in context default
Dialplan: skypopen/interface1 parsing [default->unloop] continue=false
Dialplan: skypopen/interface1 Regex (PASS) [unloop] ${unroll_loops}(true) =~
/^true$/ break=on-false
Dialplan: skypopen/interface1 Regex (FAIL) [unloop] ${sip_looped_call}() =~
/^true$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->tod_example] continue=true
Dialplan: skypopen/interface1 Date/Time Match (FAIL) [tod_example]
break=on-false
Dialplan: skypopen/interface1 parsing [default->holiday_example]
continue=true
Dialplan: skypopen/interface1 Date/Time Match (FAIL) [holiday_example]
break=on-false
Dialplan: skypopen/interface1 parsing [default->global-intercept]
continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [global-intercept]
destination_number(5100) =~ /^886$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->group-intercept]
continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [group-intercept]
destination_number(5100) =~ /^\*8$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->intercept-ext]
continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [intercept-ext]
destination_number(5100) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->redial] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [redial] destination_number(5100)
=~ /^(redial|870)$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->global] continue=true
Dialplan: skypopen/interface1 Regex (FAIL) [global] ${call_debug}(false) =~
/^true$/ break=never
Dialplan: skypopen/interface1 Regex (FAIL) [global] ${sip_has_crypto}() =~
/^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
Dialplan: skypopen/interface1 Absolute Condition [global]
Dialplan: skypopen/interface1 Action
hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
Dialplan: skypopen/interface1 Action
hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
Dialplan: skypopen/interface1 Action
hash(insert/${domain_name}-last_dial/global/${uuid})
Dialplan: skypopen/interface1 Action set(RFC2822_DATE=${strftime(%a, %d %b
%Y %T %z)})
Dialplan: skypopen/interface1 parsing [default->snom-demo-2] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [snom-demo-2]
destination_number(5100) =~ /^9001$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->snom-demo-1] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [snom-demo-1]
destination_number(5100) =~ /^9000$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->eavesdrop] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [eavesdrop]
destination_number(5100) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->eavesdrop] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [eavesdrop]
destination_number(5100) =~ /^779$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->call_return] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [call_return]
destination_number(5100) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->del-group] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [del-group]
destination_number(5100) =~ /^80(\d{2})$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->add-group] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [add-group]
destination_number(5100) =~ /^81(\d{2})$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->call-group-simo]
continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [call-group-simo]
destination_number(5100) =~ /^82(\d{2})$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->call-group-order]
continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [call-group-order]
destination_number(5100) =~ /^83(\d{2})$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->extension-intercom]
continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [extension-intercom]
destination_number(5100) =~ /^8(10[01][0-9])$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->Local_Extension]
continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [Local_Extension]
destination_number(5100) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->Local_Extension_Skinny]
continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [Local_Extension_Skinny]
destination_number(5100) =~ /^(11[01][0-9])$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->group_dial_sales]
continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [group_dial_sales]
destination_number(5100) =~ /^2000$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->group_dial_support]
continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [group_dial_support]
destination_number(5100) =~ /^2001$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->group_dial_billing]
continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [group_dial_billing]
destination_number(5100) =~ /^2002$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->operator] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [operator]
destination_number(5100) =~ /^(operator|0)$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->vmain] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [vmain] destination_number(5100)
=~ /^vmain$|^4000$|^\*98$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->sip_uri] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [sip_uri]
destination_number(5100) =~ /^sip:(.*)$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->nb_conferences]
continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [nb_conferences]
destination_number(5100) =~ /^(30\d{2})$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->wb_conferences]
continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [wb_conferences]
destination_number(5100) =~ /^(31\d{2})$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->uwb_conferences]
continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [uwb_conferences]
destination_number(5100) =~ /^(32\d{2})$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->cdquality_conferences]
continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [cdquality_conferences]
destination_number(5100) =~ /^(33\d{2})$/ break=on-false
Dialplan: skypopen/interface1 parsing
[default->freeswitch_public_conf_via_sip] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [freeswitch_public_conf_via_sip]
destination_number(5100) =~ /^9(888|8888|1616|3232)$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->mad_boss_intercom]
continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [mad_boss_intercom]
destination_number(5100) =~ /^0911$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->mad_boss_intercom]
continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [mad_boss_intercom]
destination_number(5100) =~ /^0912$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->mad_boss] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [mad_boss]
destination_number(5100) =~ /^0913$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->ivr_demo] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [ivr_demo]
destination_number(5100) =~ /^5000$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->dynamic_conference]
continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [dynamic_conference]
destination_number(5100) =~ /^5001$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->rtp_multicast_page]
continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [rtp_multicast_page]
destination_number(5100) =~ /^pagegroup$|^7243$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->park] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [park] destination_number(5100)
=~ /^5900$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->unpark] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [unpark] destination_number(5100)
=~ /^5901$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->valet_park] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [valet_park]
destination_number(5100) =~ /^(6000)$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->valet_park] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [valet_park]
destination_number(5100) =~ /^(60\d[1-9])$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->park] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [park] source(mod_skypopen) =~
/mod_sofia/ break=on-false
Dialplan: skypopen/interface1 parsing [default->unpark] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [unpark] source(mod_skypopen) =~
/mod_sofia/ break=on-false
Dialplan: skypopen/interface1 parsing [default->park] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [park] source(mod_skypopen) =~
/mod_sofia/ break=on-false
Dialplan: skypopen/interface1 parsing [default->unpark] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [unpark] source(mod_skypopen) =~
/mod_sofia/ break=on-false
Dialplan: skypopen/interface1 parsing [default->wait] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [wait] destination_number(5100)
=~ /^wait$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->fax_receive] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [fax_receive]
destination_number(5100) =~ /^9178$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->fax_transmit] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [fax_transmit]
destination_number(5100) =~ /^9179$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->ringback_180] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [ringback_180]
destination_number(5100) =~ /^9180$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->ringback_183_uk_ring]
continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [ringback_183_uk_ring]
destination_number(5100) =~ /^9181$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->ringback_183_music_ring]
continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [ringback_183_music_ring]
destination_number(5100) =~ /^9182$/ break=on-false
Dialplan: skypopen/interface1 parsing
[default->ringback_post_answer_uk_ring] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [ringback_post_answer_uk_ring]
destination_number(5100) =~ /^9183$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->ringback_post_answer_music]
continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [ringback_post_answer_music]
destination_number(5100) =~ /^9184$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->ClueCon] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [ClueCon]
destination_number(5100) =~ /^9191$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->show_info] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [show_info]
destination_number(5100) =~ /^9192$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->video_record] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [video_record]
destination_number(5100) =~ /^9193$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->video_playback]
continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [video_playback]
destination_number(5100) =~ /^9194$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->delay_echo] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [delay_echo]
destination_number(5100) =~ /^9195$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->echo] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [echo] destination_number(5100)
=~ /^9196$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->milliwatt] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [milliwatt]
destination_number(5100) =~ /^9197$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->tone_stream] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [tone_stream]
destination_number(5100) =~ /^9198$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->zrtp_enrollement]
continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [zrtp_enrollement]
destination_number(5100) =~ /^9787$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->hold_music] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [hold_music]
destination_number(5100) =~ /^9664$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->pizza_demo] continue=false
Dialplan: skypopen/interface1 Regex (FAIL) [pizza_demo]
destination_number(5100) =~ /^(pizza|74992)$/ break=on-false
Dialplan: skypopen/interface1 parsing [default->skypetest] continue=false
Dialplan: skypopen/interface1 Regex (PASS) [skypetest]
destination_number(5100) =~ /^(5100)$/ break=on-false
Dialplan: skypopen/interface1 Action bridge(user/1001)
2011-03-05 03:15:13.612058 [DEBUG] switch_core_state_machine.c:119
(skypopen/interface1) State Change CS_ROUTING -> CS_EXECUTE
2011-03-05 03:15:13.612058 [DEBUG] switch_core_session.c:1116 Send signal
skypopen/interface1 [BREAK]
2011-03-05 03:15:13.612058 [DEBUG] mod_skypopen.c:727
 [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PRERING,IDLE]
skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
2011-03-05 03:15:13.612058 [DEBUG] switch_core_state_machine.c:359
(skypopen/interface1) State ROUTING going to sleep
2011-03-05 03:15:13.612058 [DEBUG] switch_core_state_machine.c:320
(skypopen/interface1) Running State Change CS_EXECUTE
2011-03-05 03:15:13.612058 [DEBUG] switch_core_state_machine.c:366
(skypopen/interface1) State EXECUTE
2011-03-05 03:15:13.612058 [DEBUG] mod_skypopen.c:681
 [32b8f10|9350fb9] [DEBUG_SKYPE  681  ][interface1     ][PRERING,IDLE]
interface1 CHANNEL EXECUTE
2011-03-05 03:15:13.612058 [DEBUG] switch_core_state_machine.c:157
skypopen/interface1 Standard EXECUTE
EXECUTE skypopen/interface1
hash(insert/178.79.156.136-spymap/eric.nqz/c958d9be-46d6-11e0-bb71-d9b9e6428344)
2011-03-05 03:15:13.612058 [DEBUG] mod_skypopen.c:1201
[32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
MSG_ID=26
2011-03-05 03:15:13.612058 [DEBUG] mod_skypopen.c:1201
[32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
MSG_ID=27
EXECUTE skypopen/interface1
hash(insert/178.79.156.136-last_dial/eric.nqz/5100)
2011-03-05 03:15:13.613142 [DEBUG] mod_skypopen.c:1201
[32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
MSG_ID=26
2011-03-05 03:15:13.613142 [DEBUG] mod_skypopen.c:1201
[32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
MSG_ID=27
EXECUTE skypopen/interface1
hash(insert/178.79.156.136-last_dial/global/c958d9be-46d6-11e0-bb71-d9b9e6428344)
2011-03-05 03:15:13.613142 [DEBUG] mod_skypopen.c:1201
[32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
MSG_ID=26
2011-03-05 03:15:13.613142 [DEBUG] mod_skypopen.c:1201
[32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
MSG_ID=27
EXECUTE skypopen/interface1 set(RFC2822_DATE=Sat, 05 Mar 2011 03:15:13
+0000)
2011-03-05 03:15:13.613142 [DEBUG] mod_skypopen.c:1201
[32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
MSG_ID=26
2011-03-05 03:15:13.613142 [DEBUG] mod_dptools.c:1059 skypopen/interface1
SET [RFC2822_DATE]=[Sat, 05 Mar 2011 03:15:13 +0000]
2011-03-05 03:15:13.613142 [DEBUG] mod_skypopen.c:1201
[32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
MSG_ID=27
EXECUTE skypopen/interface1 bridge(user/1001)
2011-03-05 03:15:13.613142 [DEBUG] mod_skypopen.c:1201
[32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
MSG_ID=26
2011-03-05 03:15:13.614232 [DEBUG] switch_ivr_originate.c:1971 variable
string 0 = [sip_invite_domain=178.79.156.136]
2011-03-05 03:15:13.614232 [DEBUG] switch_ivr_originate.c:1971 variable
string 1 = [presence_id=1001 at 178.79.156.136]
2011-03-05 03:15:13.614232 [NOTICE] switch_channel.c:812 New Channel
sofia/internal/sip:1001 at 90.198.86.129:62954[c959849a-46d6-11e0-bb72-d9b9e6428344]
2011-03-05 03:15:13.614232 [DEBUG] mod_sofia.c:4151 (sofia/internal/
sip:1001 at 90.198.86.129:62954) State Change CS_NEW -> CS_INIT
2011-03-05 03:15:13.614232 [DEBUG] switch_core_session.c:1116 Send signal
sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
2011-03-05 03:15:13.614232 [DEBUG] switch_core_state_machine.c:320
(sofia/internal/sip:1001 at 90.198.86.129:62954) Running State Change CS_INIT
2011-03-05 03:15:13.615308 [DEBUG] switch_core_state_machine.c:356
(sofia/internal/sip:1001 at 90.198.86.129:62954) State INIT
2011-03-05 03:15:13.615308 [DEBUG] mod_sofia.c:84 sofia/internal/
sip:1001 at 90.198.86.129:62954 SOFIA INIT
2011-03-05 03:15:13.615308 [DEBUG] mod_sofia.c:124 (sofia/internal/
sip:1001 at 90.198.86.129:62954) State Change CS_INIT -> CS_ROUTING
2011-03-05 03:15:13.615308 [DEBUG] switch_core_session.c:1116 Send signal
sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
2011-03-05 03:15:13.615308 [DEBUG] switch_core_state_machine.c:356
(sofia/internal/sip:1001 at 90.198.86.129:62954) State INIT going to sleep
2011-03-05 03:15:13.615308 [DEBUG] switch_core_state_machine.c:320
(sofia/internal/sip:1001 at 90.198.86.129:62954) Running State Change
CS_ROUTING
2011-03-05 03:15:13.615308 [DEBUG] switch_channel.c:1664 (sofia/internal/
sip:1001 at 90.198.86.129:62954) Callstate Change DOWN -> RINGING
2011-03-05 03:15:13.615308 [DEBUG] switch_core_state_machine.c:359
(sofia/internal/sip:1001 at 90.198.86.129:62954) State ROUTING
2011-03-05 03:15:13.615308 [DEBUG] mod_sofia.c:147 sofia/internal/
sip:1001 at 90.198.86.129:62954 SOFIA ROUTING
2011-03-05 03:15:13.615308 [DEBUG] switch_ivr_originate.c:66
(sofia/internal/sip:1001 at 90.198.86.129:62954) State Change CS_ROUTING ->
CS_CONSUME_MEDIA
2011-03-05 03:15:13.615308 [DEBUG] switch_core_session.c:1116 Send signal
sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
2011-03-05 03:15:13.615308 [DEBUG] switch_core_state_machine.c:359
(sofia/internal/sip:1001 at 90.198.86.129:62954) State ROUTING going to sleep
2011-03-05 03:15:13.615308 [DEBUG] switch_core_state_machine.c:320
(sofia/internal/sip:1001 at 90.198.86.129:62954) Running State Change
CS_CONSUME_MEDIA
2011-03-05 03:15:13.615308 [DEBUG] switch_core_state_machine.c:378
(sofia/internal/sip:1001 at 90.198.86.129:62954) State CONSUME_MEDIA
2011-03-05 03:15:13.615308 [DEBUG] sofia.c:4683 Channel sofia/internal/
sip:1001 at 90.198.86.129:62954 entering state [calling][0]
2011-03-05 03:15:13.615308 [DEBUG] switch_core_state_machine.c:378
(sofia/internal/sip:1001 at 90.198.86.129:62954) State CONSUME_MEDIA going to
sleep
2011-03-05 03:15:13.758128 [INFO] sofia.c:729 sofia/internal/
sip:1001 at 90.198.86.129:62954 Update Callee ID to "Outbound Call" <1001>
2011-03-05 03:15:13.759171 [DEBUG] sofia.c:4683 Channel sofia/internal/
sip:1001 at 90.198.86.129:62954 entering state [proceeding][180]
2011-03-05 03:15:13.759171 [NOTICE] sofia.c:4761 Ring-Ready sofia/internal/
sip:1001 at 90.198.86.129:62954!
2011-03-05 03:15:13.759171 [DEBUG] mod_skypopen.c:1201
[32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
MSG_ID=7
2011-03-05 03:15:13.759171 [DEBUG] switch_core_session.c:709 Send signal
skypopen/interface1 [BREAK]
2011-03-05 03:15:13.759171 [DEBUG] mod_skypopen.c:727
 [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PRERING,IDLE]
skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
2011-03-05 03:15:13.759171 [NOTICE] switch_ivr_originate.c:479 Ring Ready
skypopen/interface1!
2011-03-05 03:15:13.759171 [NOTICE] switch_ivr_originate.c:479 Ring-Ready
skypopen/interface1!
2011-03-05 03:15:15.372653 [DEBUG] sofia.c:4683 Channel sofia/internal/
sip:1001 at 90.198.86.129:62954 entering state [completing][200]
2011-03-05 03:15:15.372653 [DEBUG] sofia.c:4694 Remote SDP:
v=0
o=- 1 2 IN IP4 90.198.86.129
s=CounterPath eyeBeam 1.5
c=IN IP4 90.198.86.129
t=0 0
m=audio 56446 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

2011-03-05 03:15:15.373722 [DEBUG] sofia.c:4683 Channel sofia/internal/
sip:1001 at 90.198.86.129:62954 entering state [ready][200]
2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:4529 Audio Codec Compare
[PCMA:8:8000:20:64000]/[L16:10:16000:20:256000]
2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:4529 Audio Codec Compare
[PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000]
2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:4529 Audio Codec Compare
[PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000]
2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:4529 Audio Codec Compare
[PCMA:8:8000:20:64000]/[G722:9:8000:20:64000]
2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:4529 Audio Codec Compare
[PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:4529 Audio Codec Compare
[PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:2760 Set Codec
sofia/internal/sip:1001 at 90.198.86.129:62954 PCMA/8000 20 ms 160 samples
64000 bits
2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:4635 Set 2833 dtmf send
payload to 101
2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:2990 AUDIO RTP
[sofia/internal/sip:1001 at 90.198.86.129:62954] 178.79.156.136 port 32054 ->
90.198.86.129 port 56446 codec: 8 ms: 20
2011-03-05 03:15:15.373722 [DEBUG] switch_rtp.c:1621 Starting timer [soft]
160 bytes per 20ms
2011-03-05 03:15:15.374777 [DEBUG] sofia_glue.c:3235 Set 2833 dtmf send
payload to 101
2011-03-05 03:15:15.374777 [DEBUG] sofia_glue.c:3240 Set 2833 dtmf receive
payload to 101
2011-03-05 03:15:15.374777 [DEBUG] switch_channel.c:2795 (sofia/internal/
sip:1001 at 90.198.86.129:62954) Callstate Change RINGING -> ACTIVE
2011-03-05 03:15:15.374777 [DEBUG] switch_channel.c:2807 Send signal
skypopen/interface1 [BREAK]
2011-03-05 03:15:15.374777 [DEBUG] mod_skypopen.c:727
 [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PRERING,IDLE]
skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
2011-03-05 03:15:15.374777 [NOTICE] sofia.c:5267 Channel [sofia/internal/
sip:1001 at 90.198.86.129:62954] has been answered
2011-03-05 03:15:15.375852 [DEBUG] mod_skypopen.c:1115
[32b8f10|9350fb9] [DEBUG_SKYPE  1115 ][interface1     ][PRERING,IDLE]
skypopen/interface1 CHANNEL got SWITCH_MESSAGE_INDICATE_ANSWER
2011-03-05 03:15:15.375852 [DEBUG] mod_skypopen.c:2723
[32b8f10|9350fb9] [DEBUG_SKYPE  2723 ][interface1     ][PRERING,IDLE] NOT
FOUND
2011-03-05 03:15:15.375852 [DEBUG] skypopen_protocol.c:1619
 [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][PREANSW,IDLE]
SENDING: |||ALTER CALL 142 ANSWER||||
2011-03-05 03:15:15.375852 [DEBUG] mod_skypopen.c:2734
[32b8f10|9350fb9] [DEBUG_SKYPE  2734 ][interface1     ][PREANSW,IDLE] We
answered a Skype RING on skype_call 142
2011-03-05 03:15:15.375852 [DEBUG] mod_skypopen.c:2742
[32b8f10|9350fb9] [DEBUG_SKYPE  2742 ][interface1     ][PREANSW,IDLE] NEW!
 name: interface1, state: 11, value=eric.nqz,
tech_pvt->callid_number=eric.nqz, tech_pvt->skype_user=niqizhi
2011-03-05 03:15:15.472676 [DEBUG] skypopen_protocol.c:173
[32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][PREANSW,IDLE]
READING: |||ALTER CALL 142 ANSWER|||
2011-03-05 03:15:15.884213 [DEBUG] mod_skypopen.c:1142
[32b8f10|9350fb9] [DEBUG_SKYPE  1142 ][interface1     ][PREANSW,IDLE]
Synching audio
2011-03-05 03:15:15.884213 [DEBUG] switch_core_session.c:709 Send signal
skypopen/interface1 [BREAK]
2011-03-05 03:15:15.884213 [DEBUG] mod_skypopen.c:727
 [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PREANSW,IDLE]
skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
2011-03-05 03:15:15.884213 [DEBUG] switch_channel.c:2795
(skypopen/interface1) Callstate Change RINGING -> ACTIVE
2011-03-05 03:15:15.884213 [NOTICE] switch_ivr_originate.c:3363 Channel
[skypopen/interface1] has been answered
2011-03-05 03:15:15.884213 [DEBUG] mod_skypopen.c:1148
[32b8f10|9350fb9] [DEBUG_SKYPE  1148 ][interface1     ][PREANSW,IDLE]
skypopen/interface1 CHANNEL got SWITCH_MESSAGE_INDICATE_AUDIO_SYNC
2011-03-05 03:15:15.884213 [DEBUG] mod_skypopen.c:1170
[32b8f10|9350fb9] [DEBUG_SKYPE  1170 ][interface1     ][PREANSW,IDLE]
Synching audio
2011-03-05 03:15:15.884213 [DEBUG] switch_ivr_originate.c:3408 Originate
Resulted in Success: [sofia/internal/sip:1001 at 90.198.86.129:62954]
2011-03-05 03:15:15.884213 [DEBUG] mod_skypopen.c:1148
[32b8f10|9350fb9] [DEBUG_SKYPE  1148 ][interface1     ][PREANSW,IDLE]
skypopen/interface1 CHANNEL got SWITCH_MESSAGE_INDICATE_AUDIO_SYNC
2011-03-05 03:15:15.884213 [DEBUG] mod_skypopen.c:1170
[32b8f10|9350fb9] [DEBUG_SKYPE  1170 ][interface1     ][PREANSW,IDLE]
Synching audio
2011-03-05 03:15:15.885293 [DEBUG] switch_ivr_originate.c:3408 Originate
Resulted in Success: [sofia/internal/sip:1001 at 90.198.86.129:62954]
2011-03-05 03:15:15.885293 [DEBUG] mod_skypopen.c:1148
[32b8f10|9350fb9] [DEBUG_SKYPE  1148 ][interface1     ][PREANSW,IDLE]
skypopen/interface1 CHANNEL got SWITCH_MESSAGE_INDICATE_AUDIO_SYNC
2011-03-05 03:15:15.885293 [DEBUG] mod_skypopen.c:1170
[32b8f10|9350fb9] [DEBUG_SKYPE  1170 ][interface1     ][PREANSW,IDLE]
Synching audio
2011-03-05 03:15:15.885293 [DEBUG] switch_core_session.c:709 Send signal
sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
2011-03-05 03:15:15.885293 [DEBUG] mod_skypopen.c:1173
[32b8f10|9350fb9] [DEBUG_SKYPE  1173 ][interface1     ][PREANSW,IDLE]
skypopen/interface1 CHANNEL got SWITCH_MESSAGE_INDICATE_BRIDGE
2011-03-05 03:15:15.885293 [DEBUG] mod_skypopen.c:1195
[32b8f10|9350fb9] [DEBUG_SKYPE  1195 ][interface1     ][PREANSW,IDLE]
Synching audio
2011-03-05 03:15:15.885293 [DEBUG] switch_core_session.c:709 Send signal
skypopen/interface1 [BREAK]
2011-03-05 03:15:15.885293 [DEBUG] mod_skypopen.c:727
 [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PREANSW,IDLE]
skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
2011-03-05 03:15:15.885293 [DEBUG] switch_ivr_bridge.c:1234 (sofia/internal/
sip:1001 at 90.198.86.129:62954) State Change CS_CONSUME_MEDIA ->
CS_EXCHANGE_MEDIA
2011-03-05 03:15:15.885293 [DEBUG] switch_core_session.c:1116 Send signal
sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
2011-03-05 03:15:15.885293 [DEBUG] switch_core_state_machine.c:320
(sofia/internal/sip:1001 at 90.198.86.129:62954) Running State Change
CS_EXCHANGE_MEDIA
2011-03-05 03:15:15.885293 [DEBUG] switch_core_state_machine.c:369
(sofia/internal/sip:1001 at 90.198.86.129:62954) State EXCHANGE_MEDIA
2011-03-05 03:15:15.885293 [DEBUG] mod_sofia.c:552 SOFIA EXCHANGE_MEDIA
2011-03-05 03:15:15.886388 [DEBUG] switch_core_session.c:771 Send signal
sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
2011-03-05 03:15:15.886388 [DEBUG] switch_core_session.c:771 Send signal
skypopen/interface1 [BREAK]
2011-03-05 03:15:15.886388 [DEBUG] mod_skypopen.c:727
 [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PREANSW,IDLE]
skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
2011-03-05 03:15:15.906103 [DEBUG] switch_ivr_bridge.c:500
skypopen/interface1 ending bridge by request from read function
2011-03-05 03:15:15.906103 [DEBUG] switch_ivr_bridge.c:581 BRIDGE THREAD
DONE [skypopen/interface1]
2011-03-05 03:15:15.906103 [DEBUG] switch_ivr_bridge.c:601 Send signal
sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
2011-03-05 03:15:15.909404 [DEBUG] switch_ivr_bridge.c:581 BRIDGE THREAD
DONE [sofia/internal/sip:1001 at 90.198.86.129:62954]
2011-03-05 03:15:15.909404 [DEBUG] switch_ivr_bridge.c:601 Send signal
skypopen/interface1 [BREAK]
2011-03-05 03:15:15.909404 [DEBUG] mod_skypopen.c:727
 [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PREANSW,IDLE]
skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
2011-03-05 03:15:15.909404 [DEBUG] switch_channel.c:2545 (sofia/internal/
sip:1001 at 90.198.86.129:62954) Callstate Change ACTIVE -> HANGUP
2011-03-05 03:15:15.909404 [NOTICE] switch_ivr_bridge.c:653 Hangup
sofia/internal/sip:1001 at 90.198.86.129:62954 [CS_EXCHANGE_MEDIA]
[NORMAL_CLEARING]
2011-03-05 03:15:15.909404 [DEBUG] switch_channel.c:2561 Send signal
sofia/internal/sip:1001 at 90.198.86.129:62954 [KILL]
2011-03-05 03:15:15.909404 [DEBUG] switch_core_session.c:1116 Send signal
sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
2011-03-05 03:15:15.909404 [DEBUG] switch_core_state_machine.c:369
(sofia/internal/sip:1001 at 90.198.86.129:62954) State EXCHANGE_MEDIA going to
sleep
2011-03-05 03:15:15.909404 [DEBUG] switch_core_state_machine.c:320
(sofia/internal/sip:1001 at 90.198.86.129:62954) Running State Change CS_HANGUP
2011-03-05 03:15:15.910537 [DEBUG] switch_core_state_machine.c:560
(sofia/internal/sip:1001 at 90.198.86.129:62954) State HANGUP
2011-03-05 03:15:15.910537 [DEBUG] mod_sofia.c:457 Channel sofia/internal/
sip:1001 at 90.198.86.129:62954 hanging up, cause: NORMAL_CLEARING
2011-03-05 03:15:15.910537 [DEBUG] switch_ivr_bridge.c:1305 sofia/internal/
sip:1001 at 90.198.86.129:62954 skip receive message [UNBRIDGE] (channel is
hungup already)
2011-03-05 03:15:15.910537 [DEBUG] mod_skypopen.c:1201
[32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PREANSW,IDLE]
MSG_ID=5
2011-03-05 03:15:15.910537 [DEBUG] switch_core_session.c:709 Send signal
skypopen/interface1 [BREAK]
2011-03-05 03:15:15.910537 [DEBUG] mod_skypopen.c:727
 [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PREANSW,IDLE]
skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
2011-03-05 03:15:15.910537 [DEBUG] mod_skypopen.c:1201
[32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PREANSW,IDLE]
MSG_ID=27
2011-03-05 03:15:15.910537 [NOTICE] switch_core_state_machine.c:189
skypopen/interface1 has executed the last dialplan instruction, hanging up.
2011-03-05 03:15:15.910537 [DEBUG] switch_channel.c:2545
(skypopen/interface1) Callstate Change ACTIVE -> HANGUP
2011-03-05 03:15:15.910537 [NOTICE] switch_core_state_machine.c:191 Hangup
skypopen/interface1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-05 03:15:15.910537 [DEBUG] switch_channel.c:2561 Send signal
skypopen/interface1 [KILL]
2011-03-05 03:15:15.910537 [DEBUG] mod_skypopen.c:700
 [32b8f10|9350fb9] [DEBUG_SKYPE  700  ][interface1     ][PREANSW,IDLE]
skypopen/interface1 CHANNEL got SWITCH_SIG_KILL
2011-03-05 03:15:15.910537 [DEBUG] switch_core_session.c:1116 Send signal
skypopen/interface1 [BREAK]
2011-03-05 03:15:15.910537 [DEBUG] mod_skypopen.c:727
 [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][HANG_RQ,IDLE]
skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
2011-03-05 03:15:15.910537 [DEBUG] switch_core_state_machine.c:366
(skypopen/interface1) State EXECUTE going to sleep
2011-03-05 03:15:15.910537 [DEBUG] mod_sofia.c:500 Sending BYE to
sofia/internal/sip:1001 at 90.198.86.129:62954
2011-03-05 03:15:15.910537 [DEBUG] switch_core_state_machine.c:320
(skypopen/interface1) Running State Change CS_HANGUP
2011-03-05 03:15:15.910537 [DEBUG] switch_core_state_machine.c:560
(skypopen/interface1) State HANGUP
2011-03-05 03:15:15.910537 [DEBUG] mod_skypopen.c:623
 [32b8f10|9350fb9] [DEBUG_SKYPE  623  ][interface1     ][HANG_RQ,IDLE]
hanging up skype call: 142
2011-03-05 03:15:15.911571 [DEBUG] skypopen_protocol.c:1619
 [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][HANG_RQ,IDLE]
SENDING: |||ALTER CALL 142 END HANGUP||||
2011-03-05 03:15:15.911571 [DEBUG] skypopen_protocol.c:1619
 [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][HANG_RQ,IDLE]
SENDING: |||ALTER CALL 142 HANGUP||||
2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:629
 [32b8f10|9350fb9] [DEBUG_SKYPE  629  ][interface1     ][HANG_RQ,IDLE]
interface1 CHANNEL HANGUP
2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:648 (skypopen/interface1)
State Change CS_HANGUP -> CS_DESTROY
2011-03-05 03:15:15.911571 [DEBUG] switch_core_session.c:1116 Send signal
skypopen/interface1 [BREAK]
2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:727
 [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][DOWN,IDLE]
skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:560
(skypopen/interface1) State HANGUP going to sleep
2011-03-05 03:15:15.911571 [DEBUG] switch_core_session.c:1288 Session 24
(skypopen/interface1) Locked, Waiting on external entities
2011-03-05 03:15:15.911571 [NOTICE] switch_core_session.c:1306 Session 24
(skypopen/interface1) Ended
2011-03-05 03:15:15.911571 [NOTICE] switch_core_session.c:1308 Close Channel
skypopen/interface1 [CS_DESTROY]
2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:46
sofia/internal/sip:1001 at 90.198.86.129:62954 Standard HANGUP, cause:
NORMAL_CLEARING
2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:560
(sofia/internal/sip:1001 at 90.198.86.129:62954) State HANGUP going to sleep
2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:351
(sofia/internal/sip:1001 at 90.198.86.129:62954) State Change CS_HANGUP ->
CS_REPORTING
2011-03-05 03:15:15.911571 [DEBUG] switch_core_session.c:1116 Send signal
sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:320
(sofia/internal/sip:1001 at 90.198.86.129:62954) Running State Change
CS_REPORTING
2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:620
(sofia/internal/sip:1001 at 90.198.86.129:62954) State REPORTING
2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:1001 at 90.198.86.129:62954 Standard REPORTING, cause:
NORMAL_CLEARING
2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:620
(sofia/internal/sip:1001 at 90.198.86.129:62954) State REPORTING going to sleep
2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:449
(skypopen/interface1) Callstate Change HANGUP -> DOWN
2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:452
(skypopen/interface1) Running State Change CS_DESTROY
2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:462
(skypopen/interface1) State DESTROY
2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:479
 [32b8f10|9350fb9] [DEBUG_SKYPE  479  ][interface1     ][DOWN,IDLE]
interface1 CHANNEL DESTROY c958d9be-46d6-11e0-bb71-d9b9e6428344
2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:493
 [32b8f10|9350fb9] [DEBUG_SKYPE  493  ][interface1     ][DOWN,IDLE] audio
tcp threads to DIE
2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:503
 [32b8f10|9350fb9] [DEBUG_SKYPE  503  ][interface1     ][DOWN,IDLE] audio
tcp srv thread DEAD 0
2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:513
 [32b8f10|9350fb9] [DEBUG_SKYPE  513  ][interface1     ][DOWN,IDLE] audio
tcp cli thread DEAD 0
2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:345
(sofia/internal/sip:1001 at 90.198.86.129:62954) State Change CS_REPORTING ->
CS_DESTROY
2011-03-05 03:15:15.911571 [DEBUG] switch_core_session.c:1116 Send signal
sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:524
 [32b8f10|9350fb9] [DEBUG_SKYPE  524  ][interface1     ][DOWN,IDLE] codecs
DOWN
2011-03-05 03:15:15.911571 [DEBUG] switch_core_session.c:1288 Session 25
(sofia/internal/sip:1001 at 90.198.86.129:62954) Locked, Waiting on external
entities
2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:579
 [32b8f10|9350fb9] [DEBUG_SKYPE  579  ][interface1     ][IDLE,IDLE] CHANNEL
DESTROYED c958d9be-46d6-11e0-bb71-d9b9e6428344
2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:60
skypopen/interface1 Standard DESTROY
2011-03-05 03:15:15.911571 [NOTICE] switch_core_session.c:1306 Session 25
(sofia/internal/sip:1001 at 90.198.86.129:62954) Ended
2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:462
(skypopen/interface1) State DESTROY going to sleep
2011-03-05 03:15:15.911571 [NOTICE] switch_core_session.c:1308 Close Channel
sofia/internal/sip:1001 at 90.198.86.129:62954 [CS_DESTROY]
2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:449
(sofia/internal/sip:1001 at 90.198.86.129:62954) Callstate Change HANGUP ->
DOWN
2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:452
(sofia/internal/sip:1001 at 90.198.86.129:62954) Running State Change
CS_DESTROY
2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:462
(sofia/internal/sip:1001 at 90.198.86.129:62954) State DESTROY
2011-03-05 03:15:15.911571 [DEBUG] mod_sofia.c:362 sofia/internal/
sip:1001 at 90.198.86.129:62954 SOFIA DESTROY
2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:60
sofia/internal/sip:1001 at 90.198.86.129:62954 Standard DESTROY
2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:462
(sofia/internal/sip:1001 at 90.198.86.129:62954) State DESTROY going to sleep
2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:173
[32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE] READING:
|||CALL 142 STATUS INPROGRESS|||
2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:719
[32b8f10|9350fb9] [DEBUG_SKYPE  719  ][interface1     ][IDLE,IDLE] no
tech_pvt->session_uuid_str
2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:725
[32b8f10|9350fb9] [DEBUG_SKYPE  725  ][interface1     ][IDLE,IDLE]
skype_call: 142 is now active
2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:732
[32b8f10|9350fb9] [DEBUG_SKYPE  732  ][interface1     ][UP,INPROGRS] START
start_audio_threads
2011-03-05 03:15:16.149772 [DEBUG] mod_skypopen.c:2203
[32b8f10|9350fb9] [DEBUG_SKYPE  2203 ][interface1     ][UP,INPROGRS] started
tcp_srv_thread thread.
2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:89
 [32b8f10|9350fb9] [DEBUG_SKYPE  89   ][interface1     ][UP,INPROGRS]
Binded! *which_port=32783, tech_pvt->tcp_cli_port=32782,
tech_pvt->tcp_srv_port=32783
2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:94
 [32b8f10|9350fb9] [DEBUG_SKYPE  94   ][interface1     ][UP,INPROGRS] 1
SO_RCVBUF is 87380, size is 4
2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:98
 [32b8f10|9350fb9] [DEBUG_SKYPE  98   ][interface1     ][UP,INPROGRS] 1
SO_SNDBUF is 16384, size is 4
2011-03-05 03:15:16.149772 [DEBUG] mod_skypopen.c:2219
[32b8f10|9350fb9] [DEBUG_SKYPE  2219 ][interface1     ][UP,INPROGRS] started
tcp_cli_thread thread.
2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:115
[32b8f10|9350fb9] [DEBUG_SKYPE  115  ][interface1     ][UP,INPROGRS] 2
SO_RCVBUF is 87380, size is 4
2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:130
[32b8f10|9350fb9] [DEBUG_SKYPE  130  ][interface1     ][UP,INPROGRS] 2
SO_SNDBUF is 16384, size is 4
2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:89
 [32b8f10|9350fb9] [DEBUG_SKYPE  89   ][interface1     ][UP,INPROGRS]
Binded! *which_port=32784, tech_pvt->tcp_cli_port=32784,
tech_pvt->tcp_srv_port=32783
2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:134
[32b8f10|9350fb9] [DEBUG_SKYPE  134  ][interface1     ][UP,INPROGRS]
TCP_NODELAY is 0
2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:94
 [32b8f10|9350fb9] [DEBUG_SKYPE  94   ][interface1     ][UP,INPROGRS] 1
SO_RCVBUF is 87380, size is 4
2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:141
[32b8f10|9350fb9] [DEBUG_SKYPE  141  ][interface1     ][UP,INPROGRS]
TCP_NODELAY is 0
2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:98
 [32b8f10|9350fb9] [DEBUG_SKYPE  98   ][interface1     ][UP,INPROGRS] 1
SO_SNDBUF is 16384, size is 4
2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:833
[32b8f10|9350fb9] [DEBUG_SKYPE  833  ][interface1     ][UP,INPROGRS] started
tcp_srv_thread thread.
2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:115
[32b8f10|9350fb9] [DEBUG_SKYPE  115  ][interface1     ][UP,INPROGRS] 2
SO_RCVBUF is 87380, size is 4
2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:130
[32b8f10|9350fb9] [DEBUG_SKYPE  130  ][interface1     ][UP,INPROGRS] 2
SO_SNDBUF is 16384, size is 4
2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:134
[32b8f10|9350fb9] [DEBUG_SKYPE  134  ][interface1     ][UP,INPROGRS]
TCP_NODELAY is 0
2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:141
[32b8f10|9350fb9] [DEBUG_SKYPE  141  ][interface1     ][UP,INPROGRS]
TCP_NODELAY is 0
2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:997
[32b8f10|9350fb9] [DEBUG_SKYPE  997  ][interface1     ][UP,INPROGRS] started
tcp_cli_thread thread.
2011-03-05 03:15:16.250297 [DEBUG] skypopen_protocol.c:1619
 [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][UP,INPROGRS]
SENDING: |||ALTER CALL 142 SET_INPUT PORT="32784"||||
2011-03-05 03:15:16.250297 [DEBUG] skypopen_protocol.c:1619
 [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][UP,INPROGRS]
SENDING: |||#output ALTER CALL 142 SET_OUTPUT PORT="32783"||||
2011-03-05 03:15:16.250297 [WARNING] skypopen_protocol.c:2036
 [32b8f10|9350fb9] [WARNINGA     2036 ][interface1     ][UP,INPROGRS] no
tech_pvt->session_uuid_str after INPROGRESS, let's hangup
2011-03-05 03:15:16.250297 [DEBUG] skypopen_protocol.c:1619
 [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][UP,INPROGRS]
SENDING: |||ALTER CALL 142 HANGUP||||
2011-03-05 03:15:16.250297 [DEBUG] skypopen_protocol.c:173
[32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][UP,INPROGRS]
READING: |||CALL 142 VIDEO_SEND_STATUS AVAILABLE|||
2011-03-05 03:15:16.290473 [DEBUG] skypopen_protocol.c:173
[32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][UP,INPROGRS]
READING: |||CALL 142 VIDEO_STATUS VIDEO_BOTH_ENABLED|||
2011-03-05 03:15:16.363839 [DEBUG] skypopen_protocol.c:173
[32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][UP,INPROGRS]
READING: |||CALL 142 DURATION 1|||
2011-03-05 03:15:16.432605 [DEBUG] skypopen_protocol.c:173
[32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][UP,INPROGRS]
READING: |||ALTER CALL 142 END HANGUP|||
2011-03-05 03:15:16.494506 [DEBUG] skypopen_protocol.c:173
[32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][UP,INPROGRS]
READING: |||ERROR 559 CALL: Action failed|||
2011-03-05 03:15:16.495584 [DEBUG] skypopen_protocol.c:228
[32b8f10|9350fb9] [DEBUG_SKYPE  228  ][interface1     ][UP,INPROGRS] Skype
got ERROR about a failed action (probably TRYING to HANGUP A CALL), no
problem: |||ERROR 559 CALL: Action failed|||
2011-03-05 03:15:16.710271 [DEBUG] skypopen_protocol.c:173
[32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][UP,INPROGRS]
READING: |||ERROR 589 ALTER CALL: unable to alter input/output|||
2011-03-05 03:15:16.711346 [DEBUG] skypopen_protocol.c:247
[32b8f10|9350fb9] [DEBUG_SKYPE  247  ][interface1     ][UP,INPROGRS] Skype
client was not able to correctly manage tcp audio sockets, probably got a
local or remote hangup: |||ERROR 589 ALTER CALL: unable to alter
input/output|||
2011-03-05 03:15:16.711346 [DEBUG] skypopen_protocol.c:1619
 [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][UP,INPROGRS]
SENDING: |||ALTER CALL 142 HANGUP||||
2011-03-05 03:15:16.711346 [DEBUG] skypopen_protocol.c:1619
 [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][UP,INPROGRS]
SENDING: |||ALTER CALL 142 END HANGUP||||
2011-03-05 03:15:16.711346 [DEBUG] mod_skypopen.c:1413
[32b8f10|9350fb9] [DEBUG_SKYPE  1413 ][interface1     ][DOWN,INPROGRS] skype
call ended
2011-03-05 03:15:16.711346 [DEBUG] mod_skypopen.c:1433
[32b8f10|9350fb9] [DEBUG_SKYPE  1433 ][interface1     ][DOWN,INPROGRS] no
session
2011-03-05 03:15:16.711346 [DEBUG] mod_skypopen.c:1436
[32b8f10|9350fb9] [DEBUG_SKYPE  1436 ][interface1     ][DOWN,INPROGRS] audio
tcp threads to DIE
2011-03-05 03:15:16.712434 [DEBUG] skypopen_protocol.c:960
[32b8f10|9350fb9] [DEBUG_SKYPE  960  ][interface1     ][DOWN,INPROGRS]
incoming audio (read) server (I am it) EXITING
2011-03-05 03:15:16.715692 [DEBUG] mod_skypopen.c:1446
[32b8f10|9350fb9] [DEBUG_SKYPE  1446 ][interface1     ][DOWN,INPROGRS] audio
tcp srv thread DEAD 1
2011-03-05 03:15:16.750763 [DEBUG] skypopen_protocol.c:1112
 [32b8f10|9350fb9] [DEBUG_SKYPE  1112 ][interface1     ][DOWN,INPROGRS]
outbound audio server (I am it) EXITING
2011-03-05 03:15:16.751849 [DEBUG] mod_skypopen.c:1456
[32b8f10|9350fb9] [DEBUG_SKYPE  1456 ][interface1     ][DOWN,INPROGRS] audio
tcp cli thread DEAD 7
2011-03-05 03:15:16.751849 [DEBUG] skypopen_protocol.c:173
[32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE] READING:
|||#output ERROR 589 ALTER CALL: unable to alter input/output|||
2011-03-05 03:15:16.779660 [DEBUG] skypopen_protocol.c:173
[32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE] READING:
|||ERROR 559 CALL: Action failed|||
2011-03-05 03:15:16.780762 [DEBUG] skypopen_protocol.c:228
[32b8f10|9350fb9] [DEBUG_SKYPE  228  ][interface1     ][IDLE,IDLE] Skype got
ERROR about a failed action (probably TRYING to HANGUP A CALL), no problem:
|||ERROR 559 CALL: Action failed|||
2011-03-05 03:15:16.828877 [DEBUG] skypopen_protocol.c:173
[32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE] READING:
|||ERROR 559 CALL: Action failed|||
2011-03-05 03:15:16.828877 [DEBUG] skypopen_protocol.c:228
[32b8f10|9350fb9] [DEBUG_SKYPE  228  ][interface1     ][IDLE,IDLE] Skype got
ERROR about a failed action (probably TRYING to HANGUP A CALL), no problem:
|||ERROR 559 CALL: Action failed|||
2011-03-05 03:15:16.869183 [DEBUG] skypopen_protocol.c:173
[32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE] READING:
|||ERROR 559 CALL: Action failed|||
2011-03-05 03:15:16.869183 [DEBUG] skypopen_protocol.c:228
[32b8f10|9350fb9] [DEBUG_SKYPE  228  ][interface1     ][IDLE,IDLE] Skype got
ERROR about a failed action (probably TRYING to HANGUP A CALL), no problem:
|||ERROR 559 CALL: Action failed|||
2011-03-05 03:15:16.911668 [DEBUG] skypopen_protocol.c:173
[32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE] READING:
|||CALL 142 STATUS FINISHED|||
2011-03-05 03:15:16.911668 [DEBUG] skypopen_protocol.c:661
[32b8f10|9350fb9] [DEBUG_SKYPE  661  ][interface1     ][IDLE,IDLE]
skype_call 142 is MY call, now I'm going DOWN
2011-03-05 03:15:16.911668 [DEBUG] mod_skypopen.c:1413
[32b8f10|9350fb9] [DEBUG_SKYPE  1413 ][interface1     ][IDLE,IDLE] skype
call ended
2011-03-05 03:15:16.911668 [DEBUG] mod_skypopen.c:1433
[32b8f10|9350fb9] [DEBUG_SKYPE  1433 ][interface1     ][IDLE,IDLE] no
session
2011-03-05 03:15:16.911668 [DEBUG] mod_skypopen.c:1436
[32b8f10|9350fb9] [DEBUG_SKYPE  1436 ][interface1     ][DOWN,IDLE] audio tcp
threads to DIE
2011-03-05 03:15:16.911668 [DEBUG] mod_skypopen.c:1446
[32b8f10|9350fb9] [DEBUG_SKYPE  1446 ][interface1     ][DOWN,IDLE] audio tcp
srv thread DEAD 0
2011-03-05 03:15:16.911668 [DEBUG] mod_skypopen.c:1456
[32b8f10|9350fb9] [DEBUG_SKYPE  1456 ][interface1     ][DOWN,IDLE] audio tcp
cli thread DEAD 0
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From boris at tagnet.ru  Sat Mar  5 07:42:48 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 05 Mar 2011 09:42:48 +0500
Subject: [Freeswitch-users] What is wrong with my extension?
In-Reply-To: <AANLkTin4ZVJGNbMmVwCtrc0Po-G7xpzP8qHADe4RmRAq@mail.gmail.com>
References: <4D70C63B.7080005@tagnet.ru>
	<AANLkTin4ZVJGNbMmVwCtrc0Po-G7xpzP8qHADe4RmRAq@mail.gmail.com>
Message-ID: <4D71BF48.4000602@tagnet.ru>

Hello!

     Here it is: http://pastebin.freeswitch.org/15564
> Boris,
>
> I recommend that you pastebin the entire debug output of the call, 
> including all the dialplan matching stuff. That will help narrow down 
> what is happening. Use pastebin.freeswitch.org 
> <http://pastebin.freeswitch.org> and put the link in this thread.
>
> -MC
>
> On Fri, Mar 4, 2011 at 3:00 AM, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     Hello!
>
>     May someone tell me why the 73435 is stripped ???  And if I doing
>     something wrong what is the right way. I'm useing FreeSWITCH Version
>     1.0.head (git-1c95ad9 2011-01-20 22-43-50 -0300)
>
>
>         Here is my extension:
>
>     <!--
>                 Transfer call to the TOP context
>             -->
>     <extension name="top.ctx">
>     <condition field="${v_tagnet_dst}" expression="^([0-9]+)$">
>     <action application="log" data="ALERT [top.ctx] - transfer to
>     top.ctx"/>
>     <action application="log" data="ALERT [top.ctx] ORIG: ${v_tagnet_dst}
>     REGEX: $1"/>
>
>     <!-- Caller ID normalization -->
>     <action application="execute_extension" data="ext_translate_src XML
>     features"/>
>
>     <action application="log" data="ALERT [top.ctx] - transfer to
>     top.ctx $1"/>
>     <action application="transfer" data="$1 XML top.ctx"/>
>     </condition>
>     </extension>
>
>
>     Here is the output:
>     2011-03-04 15:56:42.127837 [ALERT] mod_dptools.c:1174 [top.ctx] -
>     transfer to top.ctx
>     2011-03-04 15:56:42.129862 [ALERT] mod_dptools.c:1174 [top.ctx] ORIG:
>     73435230020 REGEX: 230020
>     --
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991


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From gmaruzz at gmail.com  Sat Mar  5 08:01:43 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 5 Mar 2011 06:01:43 +0100
Subject: [Freeswitch-users] skype to x-lite birdge hangup imediately
	when answered
In-Reply-To: <AANLkTik3De=vPS8VfZfcJCdQNjHc7EvjxW=uej33JFjT@mail.gmail.com>
References: <AANLkTik3De=vPS8VfZfcJCdQNjHc7EvjxW=uej33JFjT@mail.gmail.com>
Message-ID: <AANLkTimiNY193=vVer4AbCy2h7kLmqwbQxhNiM3ydnwL@mail.gmail.com>

Hi Yi,

could you please test your extension inserting an action "answer"
before the action "bridge" ?

-giovanni

On 3/5/11, Yihui Li <xyangni at gmail.com> wrote:
> Dear all,
>
> I am tring skypopen with latest git-head.
> There are just 1 skype instance. When forward skype call to 5000, it works
> fine. It also works fine with originate skypopen/interface1/*****
> &bridge(user/1001)
>
>  But when I try to forward skype to 1001 with the following dial plan, call
> hang up imediately when answered.
>
>   <extension name="skypetest">
>     <condition field="destination_number" expression="^(5100)$">
>       <action application="bridge" data="user/1001"/>
>     </condition>
>   </extension>
>
> I am working on l ubuntu 10.10 server edition and skype static 2.0.0.72
>
> here is the skype configure
> =============full mod_skypopen.conf.xml=====================
> <configuration name="skypopen.conf" description="Skypopen Configuration">
>   <global_settings>
>     <param name="debug" value="8"/>
>     <param name="dialplan" value="XML"/>
>     <param name="context" value="default"/>
>     <param name="destination" value="5100"/>
>     <param name="skype_user" value="user1"/>
>     <param name="report_incoming_chatmessages" value="true"/>
>     <param name="silent_mode" value="false"/>
>     <param name="write_silence_when_idle" value="true"/>
>   </global_settings>
>   <!-- one entry here per each skypopen interface -->
>   <per_interface_settings>
>     <interface id="1" name="interface1">
>         <param name="X11-display" value="localhost:10.0"/>
>         <param name="skype_user" value="niqizhi"/>
>     </interface>
>   </per_interface_settings>
> </configuration>
>
>
>
>
> the debug message of the call is listed below. May I ask what is the
> possiable cause? Thanks.
>
>
>
> 2011-03-05 03:15:13.353532 [DEBUG] skypopen_protocol.c:173
> [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE] READING:
> |||CALL 142 CONF_ID 0|||
> 2011-03-05 03:15:13.438727 [DEBUG] skypopen_protocol.c:173
> [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE] READING:
> |||CALL 142 STATUS RINGING|||
> 2011-03-05 03:15:13.438727 [DEBUG] skypopen_protocol.c:595
> [32b8f10|9350fb9] [DEBUG_SKYPE  595  ][interface1     ][IDLE,IDLE] NO ACTIVE
> calls in this moment, skype_call 142 is RINGING, to ask PARTNER_DISPNAME and
> PARTNER_HANDLE
> 2011-03-05 03:15:13.438727 [DEBUG] skypopen_protocol.c:1619
>  [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][IDLE,IDLE] SENDING:
> |||GET CALL 142 PARTNER_DISPNAME||||
> 2011-03-05 03:15:13.438727 [DEBUG] skypopen_protocol.c:1619
>  [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][IDLE,IDLE] SENDING:
> |||GET CALL 142 PARTNER_HANDLE||||
> 2011-03-05 03:15:13.576818 [DEBUG] skypopen_protocol.c:173
> [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE] READING:
> |||CALL 142 PARTNER_DISPNAME Eric Ni|||
> 2011-03-05 03:15:13.609592 [DEBUG] skypopen_protocol.c:173
> [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE] READING:
> |||CALL 142 PARTNER_HANDLE eric.nqz|||
> 2011-03-05 03:15:13.609592 [DEBUG] skypopen_protocol.c:506
> [32b8f10|9350fb9] [DEBUG_SKYPE  506  ][interface1     ][IDLE,IDLE] Call 142
> go to skypopen_partner_handle_ring
> 2011-03-05 03:15:13.609592 [DEBUG] mod_skypopen.c:2626
> [32b8f10|9350fb9] [DEBUG_SKYPE  2626 ][interface1     ][IDLE,IDLE] NOT FOUND
> 2011-03-05 03:15:13.609592 [DEBUG] mod_skypopen.c:2244
> [32b8f10|9350fb9] [DEBUG_SKYPE  2244 ][interface1     ][PRERING,IDLE] 2
> SESSION_REQUEST c958d9be-46d6-11e0-bb71-d9b9e6428344
> 2011-03-05 03:15:13.609592 [DEBUG] mod_skypopen.c:244
>  [32b8f10|9350fb9] [DEBUG_SKYPE  244  ][interface1     ][PRERING,IDLE]
> codecs UP
> 2011-03-05 03:15:13.609592 [DEBUG] mod_skypopen.c:273
>  [32b8f10|9350fb9] [DEBUG_SKYPE  273  ][interface1     ][PRERING,IDLE]
> skypopen_tech_init SUCCESS
> 2011-03-05 03:15:13.609592 [NOTICE] switch_channel.c:812 New Channel
> skypopen/interface1 [c958d9be-46d6-11e0-bb71-d9b9e6428344]
> 2011-03-05 03:15:13.609592 [DEBUG] mod_skypopen.c:2268 (skypopen/interface1)
> State Change CS_NEW -> CS_INIT
> 2011-03-05 03:15:13.609592 [DEBUG] switch_core_session.c:1116 Send signal
> skypopen/interface1 [BREAK]
> 2011-03-05 03:15:13.609592 [DEBUG] mod_skypopen.c:727
>  [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PRERING,IDLE]
> skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
> 2011-03-05 03:15:13.609592 [DEBUG] mod_skypopen.c:2280
> [32b8f10|9350fb9] [DEBUG_SKYPE  2280 ][interface1     ][PRERING,IDLE]
> new_inbound_channel
> 2011-03-05 03:15:13.610977 [DEBUG] switch_core_state_machine.c:320
> (skypopen/interface1) Running State Change CS_INIT
> 2011-03-05 03:15:13.610977 [DEBUG] switch_core_state_machine.c:356
> (skypopen/interface1) State INIT
> 2011-03-05 03:15:13.610977 [DEBUG] mod_skypopen.c:463 (skypopen/interface1)
> State Change CS_INIT -> CS_ROUTING
> 2011-03-05 03:15:13.610977 [DEBUG] switch_core_session.c:1116 Send signal
> skypopen/interface1 [BREAK]
> 2011-03-05 03:15:13.610977 [DEBUG] mod_skypopen.c:727
>  [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PRERING,IDLE]
> skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
> 2011-03-05 03:15:13.610977 [DEBUG] mod_skypopen.c:464
>  [32b8f10|9350fb9] [DEBUG_SKYPE  464  ][interface1     ][PRERING,IDLE]
> interface1 CHANNEL INIT c958d9be-46d6-11e0-bb71-d9b9e6428344
> 2011-03-05 03:15:13.610977 [DEBUG] switch_core_state_machine.c:356
> (skypopen/interface1) State INIT going to sleep
> 2011-03-05 03:15:13.610977 [DEBUG] switch_core_state_machine.c:320
> (skypopen/interface1) Running State Change CS_ROUTING
> 2011-03-05 03:15:13.610977 [DEBUG] switch_channel.c:1664
> (skypopen/interface1) Callstate Change DOWN -> RINGING
> 2011-03-05 03:15:13.610977 [DEBUG] switch_core_state_machine.c:359
> (skypopen/interface1) State ROUTING
> 2011-03-05 03:15:13.610977 [DEBUG] mod_skypopen.c:664
>  [32b8f10|9350fb9] [DEBUG_SKYPE  664  ][interface1     ][PRERING,IDLE]
> interface1 CHANNEL ROUTING
> 2011-03-05 03:15:13.610977 [DEBUG] switch_core_state_machine.c:77
> skypopen/interface1 Standard ROUTING
> 2011-03-05 03:15:13.610977 [INFO] mod_dialplan_xml.c:331 Processing Eric Ni
> <eric.nqz>->5100 in context default
> Dialplan: skypopen/interface1 parsing [default->unloop] continue=false
> Dialplan: skypopen/interface1 Regex (PASS) [unloop] ${unroll_loops}(true) =~
> /^true$/ break=on-false
> Dialplan: skypopen/interface1 Regex (FAIL) [unloop] ${sip_looped_call}() =~
> /^true$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->tod_example] continue=true
> Dialplan: skypopen/interface1 Date/Time Match (FAIL) [tod_example]
> break=on-false
> Dialplan: skypopen/interface1 parsing [default->holiday_example]
> continue=true
> Dialplan: skypopen/interface1 Date/Time Match (FAIL) [holiday_example]
> break=on-false
> Dialplan: skypopen/interface1 parsing [default->global-intercept]
> continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [global-intercept]
> destination_number(5100) =~ /^886$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->group-intercept]
> continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [group-intercept]
> destination_number(5100) =~ /^\*8$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->intercept-ext]
> continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [intercept-ext]
> destination_number(5100) =~ /^\*\*(\d+)$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->redial] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [redial] destination_number(5100)
> =~ /^(redial|870)$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->global] continue=true
> Dialplan: skypopen/interface1 Regex (FAIL) [global] ${call_debug}(false) =~
> /^true$/ break=never
> Dialplan: skypopen/interface1 Regex (FAIL) [global] ${sip_has_crypto}() =~
> /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
> Dialplan: skypopen/interface1 Absolute Condition [global]
> Dialplan: skypopen/interface1 Action
> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
> Dialplan: skypopen/interface1 Action
> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
> Dialplan: skypopen/interface1 Action
> hash(insert/${domain_name}-last_dial/global/${uuid})
> Dialplan: skypopen/interface1 Action set(RFC2822_DATE=${strftime(%a, %d %b
> %Y %T %z)})
> Dialplan: skypopen/interface1 parsing [default->snom-demo-2] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [snom-demo-2]
> destination_number(5100) =~ /^9001$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->snom-demo-1] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [snom-demo-1]
> destination_number(5100) =~ /^9000$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->eavesdrop] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [eavesdrop]
> destination_number(5100) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->eavesdrop] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [eavesdrop]
> destination_number(5100) =~ /^779$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->call_return] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [call_return]
> destination_number(5100) =~ /^\*69$|^869$|^lcr$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->del-group] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [del-group]
> destination_number(5100) =~ /^80(\d{2})$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->add-group] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [add-group]
> destination_number(5100) =~ /^81(\d{2})$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->call-group-simo]
> continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [call-group-simo]
> destination_number(5100) =~ /^82(\d{2})$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->call-group-order]
> continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [call-group-order]
> destination_number(5100) =~ /^83(\d{2})$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->extension-intercom]
> continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [extension-intercom]
> destination_number(5100) =~ /^8(10[01][0-9])$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->Local_Extension]
> continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [Local_Extension]
> destination_number(5100) =~ /^(10[01][0-9])$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->Local_Extension_Skinny]
> continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [Local_Extension_Skinny]
> destination_number(5100) =~ /^(11[01][0-9])$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->group_dial_sales]
> continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [group_dial_sales]
> destination_number(5100) =~ /^2000$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->group_dial_support]
> continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [group_dial_support]
> destination_number(5100) =~ /^2001$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->group_dial_billing]
> continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [group_dial_billing]
> destination_number(5100) =~ /^2002$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->operator] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [operator]
> destination_number(5100) =~ /^(operator|0)$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->vmain] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [vmain] destination_number(5100)
> =~ /^vmain$|^4000$|^\*98$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->sip_uri] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [sip_uri]
> destination_number(5100) =~ /^sip:(.*)$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->nb_conferences]
> continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [nb_conferences]
> destination_number(5100) =~ /^(30\d{2})$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->wb_conferences]
> continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [wb_conferences]
> destination_number(5100) =~ /^(31\d{2})$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->uwb_conferences]
> continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [uwb_conferences]
> destination_number(5100) =~ /^(32\d{2})$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->cdquality_conferences]
> continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [cdquality_conferences]
> destination_number(5100) =~ /^(33\d{2})$/ break=on-false
> Dialplan: skypopen/interface1 parsing
> [default->freeswitch_public_conf_via_sip] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [freeswitch_public_conf_via_sip]
> destination_number(5100) =~ /^9(888|8888|1616|3232)$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->mad_boss_intercom]
> continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [mad_boss_intercom]
> destination_number(5100) =~ /^0911$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->mad_boss_intercom]
> continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [mad_boss_intercom]
> destination_number(5100) =~ /^0912$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->mad_boss] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [mad_boss]
> destination_number(5100) =~ /^0913$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->ivr_demo] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [ivr_demo]
> destination_number(5100) =~ /^5000$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->dynamic_conference]
> continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [dynamic_conference]
> destination_number(5100) =~ /^5001$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->rtp_multicast_page]
> continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [rtp_multicast_page]
> destination_number(5100) =~ /^pagegroup$|^7243$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->park] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [park] destination_number(5100)
> =~ /^5900$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->unpark] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [unpark] destination_number(5100)
> =~ /^5901$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->valet_park] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [valet_park]
> destination_number(5100) =~ /^(6000)$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->valet_park] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [valet_park]
> destination_number(5100) =~ /^(60\d[1-9])$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->park] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [park] source(mod_skypopen) =~
> /mod_sofia/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->unpark] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [unpark] source(mod_skypopen) =~
> /mod_sofia/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->park] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [park] source(mod_skypopen) =~
> /mod_sofia/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->unpark] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [unpark] source(mod_skypopen) =~
> /mod_sofia/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->wait] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [wait] destination_number(5100)
> =~ /^wait$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->fax_receive] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [fax_receive]
> destination_number(5100) =~ /^9178$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->fax_transmit] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [fax_transmit]
> destination_number(5100) =~ /^9179$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->ringback_180] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [ringback_180]
> destination_number(5100) =~ /^9180$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->ringback_183_uk_ring]
> continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [ringback_183_uk_ring]
> destination_number(5100) =~ /^9181$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->ringback_183_music_ring]
> continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [ringback_183_music_ring]
> destination_number(5100) =~ /^9182$/ break=on-false
> Dialplan: skypopen/interface1 parsing
> [default->ringback_post_answer_uk_ring] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [ringback_post_answer_uk_ring]
> destination_number(5100) =~ /^9183$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->ringback_post_answer_music]
> continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [ringback_post_answer_music]
> destination_number(5100) =~ /^9184$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->ClueCon] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [ClueCon]
> destination_number(5100) =~ /^9191$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->show_info] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [show_info]
> destination_number(5100) =~ /^9192$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->video_record] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [video_record]
> destination_number(5100) =~ /^9193$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->video_playback]
> continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [video_playback]
> destination_number(5100) =~ /^9194$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->delay_echo] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [delay_echo]
> destination_number(5100) =~ /^9195$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->echo] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [echo] destination_number(5100)
> =~ /^9196$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->milliwatt] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [milliwatt]
> destination_number(5100) =~ /^9197$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->tone_stream] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [tone_stream]
> destination_number(5100) =~ /^9198$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->zrtp_enrollement]
> continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [zrtp_enrollement]
> destination_number(5100) =~ /^9787$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->hold_music] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [hold_music]
> destination_number(5100) =~ /^9664$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->pizza_demo] continue=false
> Dialplan: skypopen/interface1 Regex (FAIL) [pizza_demo]
> destination_number(5100) =~ /^(pizza|74992)$/ break=on-false
> Dialplan: skypopen/interface1 parsing [default->skypetest] continue=false
> Dialplan: skypopen/interface1 Regex (PASS) [skypetest]
> destination_number(5100) =~ /^(5100)$/ break=on-false
> Dialplan: skypopen/interface1 Action bridge(user/1001)
> 2011-03-05 03:15:13.612058 [DEBUG] switch_core_state_machine.c:119
> (skypopen/interface1) State Change CS_ROUTING -> CS_EXECUTE
> 2011-03-05 03:15:13.612058 [DEBUG] switch_core_session.c:1116 Send signal
> skypopen/interface1 [BREAK]
> 2011-03-05 03:15:13.612058 [DEBUG] mod_skypopen.c:727
>  [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PRERING,IDLE]
> skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
> 2011-03-05 03:15:13.612058 [DEBUG] switch_core_state_machine.c:359
> (skypopen/interface1) State ROUTING going to sleep
> 2011-03-05 03:15:13.612058 [DEBUG] switch_core_state_machine.c:320
> (skypopen/interface1) Running State Change CS_EXECUTE
> 2011-03-05 03:15:13.612058 [DEBUG] switch_core_state_machine.c:366
> (skypopen/interface1) State EXECUTE
> 2011-03-05 03:15:13.612058 [DEBUG] mod_skypopen.c:681
>  [32b8f10|9350fb9] [DEBUG_SKYPE  681  ][interface1     ][PRERING,IDLE]
> interface1 CHANNEL EXECUTE
> 2011-03-05 03:15:13.612058 [DEBUG] switch_core_state_machine.c:157
> skypopen/interface1 Standard EXECUTE
> EXECUTE skypopen/interface1
> hash(insert/178.79.156.136-spymap/eric.nqz/c958d9be-46d6-11e0-bb71-d9b9e6428344)
> 2011-03-05 03:15:13.612058 [DEBUG] mod_skypopen.c:1201
> [32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
> MSG_ID=26
> 2011-03-05 03:15:13.612058 [DEBUG] mod_skypopen.c:1201
> [32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
> MSG_ID=27
> EXECUTE skypopen/interface1
> hash(insert/178.79.156.136-last_dial/eric.nqz/5100)
> 2011-03-05 03:15:13.613142 [DEBUG] mod_skypopen.c:1201
> [32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
> MSG_ID=26
> 2011-03-05 03:15:13.613142 [DEBUG] mod_skypopen.c:1201
> [32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
> MSG_ID=27
> EXECUTE skypopen/interface1
> hash(insert/178.79.156.136-last_dial/global/c958d9be-46d6-11e0-bb71-d9b9e6428344)
> 2011-03-05 03:15:13.613142 [DEBUG] mod_skypopen.c:1201
> [32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
> MSG_ID=26
> 2011-03-05 03:15:13.613142 [DEBUG] mod_skypopen.c:1201
> [32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
> MSG_ID=27
> EXECUTE skypopen/interface1 set(RFC2822_DATE=Sat, 05 Mar 2011 03:15:13
> +0000)
> 2011-03-05 03:15:13.613142 [DEBUG] mod_skypopen.c:1201
> [32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
> MSG_ID=26
> 2011-03-05 03:15:13.613142 [DEBUG] mod_dptools.c:1059 skypopen/interface1
> SET [RFC2822_DATE]=[Sat, 05 Mar 2011 03:15:13 +0000]
> 2011-03-05 03:15:13.613142 [DEBUG] mod_skypopen.c:1201
> [32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
> MSG_ID=27
> EXECUTE skypopen/interface1 bridge(user/1001)
> 2011-03-05 03:15:13.613142 [DEBUG] mod_skypopen.c:1201
> [32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
> MSG_ID=26
> 2011-03-05 03:15:13.614232 [DEBUG] switch_ivr_originate.c:1971 variable
> string 0 = [sip_invite_domain=178.79.156.136]
> 2011-03-05 03:15:13.614232 [DEBUG] switch_ivr_originate.c:1971 variable
> string 1 = [presence_id=1001 at 178.79.156.136]
> 2011-03-05 03:15:13.614232 [NOTICE] switch_channel.c:812 New Channel
> sofia/internal/sip:1001 at 90.198.86.129:62954[c959849a-46d6-11e0-bb72-d9b9e6428344]
> 2011-03-05 03:15:13.614232 [DEBUG] mod_sofia.c:4151 (sofia/internal/
> sip:1001 at 90.198.86.129:62954) State Change CS_NEW -> CS_INIT
> 2011-03-05 03:15:13.614232 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
> 2011-03-05 03:15:13.614232 [DEBUG] switch_core_state_machine.c:320
> (sofia/internal/sip:1001 at 90.198.86.129:62954) Running State Change CS_INIT
> 2011-03-05 03:15:13.615308 [DEBUG] switch_core_state_machine.c:356
> (sofia/internal/sip:1001 at 90.198.86.129:62954) State INIT
> 2011-03-05 03:15:13.615308 [DEBUG] mod_sofia.c:84 sofia/internal/
> sip:1001 at 90.198.86.129:62954 SOFIA INIT
> 2011-03-05 03:15:13.615308 [DEBUG] mod_sofia.c:124 (sofia/internal/
> sip:1001 at 90.198.86.129:62954) State Change CS_INIT -> CS_ROUTING
> 2011-03-05 03:15:13.615308 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
> 2011-03-05 03:15:13.615308 [DEBUG] switch_core_state_machine.c:356
> (sofia/internal/sip:1001 at 90.198.86.129:62954) State INIT going to sleep
> 2011-03-05 03:15:13.615308 [DEBUG] switch_core_state_machine.c:320
> (sofia/internal/sip:1001 at 90.198.86.129:62954) Running State Change
> CS_ROUTING
> 2011-03-05 03:15:13.615308 [DEBUG] switch_channel.c:1664 (sofia/internal/
> sip:1001 at 90.198.86.129:62954) Callstate Change DOWN -> RINGING
> 2011-03-05 03:15:13.615308 [DEBUG] switch_core_state_machine.c:359
> (sofia/internal/sip:1001 at 90.198.86.129:62954) State ROUTING
> 2011-03-05 03:15:13.615308 [DEBUG] mod_sofia.c:147 sofia/internal/
> sip:1001 at 90.198.86.129:62954 SOFIA ROUTING
> 2011-03-05 03:15:13.615308 [DEBUG] switch_ivr_originate.c:66
> (sofia/internal/sip:1001 at 90.198.86.129:62954) State Change CS_ROUTING ->
> CS_CONSUME_MEDIA
> 2011-03-05 03:15:13.615308 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
> 2011-03-05 03:15:13.615308 [DEBUG] switch_core_state_machine.c:359
> (sofia/internal/sip:1001 at 90.198.86.129:62954) State ROUTING going to sleep
> 2011-03-05 03:15:13.615308 [DEBUG] switch_core_state_machine.c:320
> (sofia/internal/sip:1001 at 90.198.86.129:62954) Running State Change
> CS_CONSUME_MEDIA
> 2011-03-05 03:15:13.615308 [DEBUG] switch_core_state_machine.c:378
> (sofia/internal/sip:1001 at 90.198.86.129:62954) State CONSUME_MEDIA
> 2011-03-05 03:15:13.615308 [DEBUG] sofia.c:4683 Channel sofia/internal/
> sip:1001 at 90.198.86.129:62954 entering state [calling][0]
> 2011-03-05 03:15:13.615308 [DEBUG] switch_core_state_machine.c:378
> (sofia/internal/sip:1001 at 90.198.86.129:62954) State CONSUME_MEDIA going to
> sleep
> 2011-03-05 03:15:13.758128 [INFO] sofia.c:729 sofia/internal/
> sip:1001 at 90.198.86.129:62954 Update Callee ID to "Outbound Call" <1001>
> 2011-03-05 03:15:13.759171 [DEBUG] sofia.c:4683 Channel sofia/internal/
> sip:1001 at 90.198.86.129:62954 entering state [proceeding][180]
> 2011-03-05 03:15:13.759171 [NOTICE] sofia.c:4761 Ring-Ready sofia/internal/
> sip:1001 at 90.198.86.129:62954!
> 2011-03-05 03:15:13.759171 [DEBUG] mod_skypopen.c:1201
> [32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
> MSG_ID=7
> 2011-03-05 03:15:13.759171 [DEBUG] switch_core_session.c:709 Send signal
> skypopen/interface1 [BREAK]
> 2011-03-05 03:15:13.759171 [DEBUG] mod_skypopen.c:727
>  [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PRERING,IDLE]
> skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
> 2011-03-05 03:15:13.759171 [NOTICE] switch_ivr_originate.c:479 Ring Ready
> skypopen/interface1!
> 2011-03-05 03:15:13.759171 [NOTICE] switch_ivr_originate.c:479 Ring-Ready
> skypopen/interface1!
> 2011-03-05 03:15:15.372653 [DEBUG] sofia.c:4683 Channel sofia/internal/
> sip:1001 at 90.198.86.129:62954 entering state [completing][200]
> 2011-03-05 03:15:15.372653 [DEBUG] sofia.c:4694 Remote SDP:
> v=0
> o=- 1 2 IN IP4 90.198.86.129
> s=CounterPath eyeBeam 1.5
> c=IN IP4 90.198.86.129
> t=0 0
> m=audio 56446 RTP/AVP 8 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
> 2011-03-05 03:15:15.373722 [DEBUG] sofia.c:4683 Channel sofia/internal/
> sip:1001 at 90.198.86.129:62954 entering state [ready][200]
> 2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:4529 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[L16:10:16000:20:256000]
> 2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:4529 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000]
> 2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:4529 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000]
> 2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:4529 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000]
> 2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:4529 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000]
> 2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:4529 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
> 2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:2760 Set Codec
> sofia/internal/sip:1001 at 90.198.86.129:62954 PCMA/8000 20 ms 160 samples
> 64000 bits
> 2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:4635 Set 2833 dtmf send
> payload to 101
> 2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:2990 AUDIO RTP
> [sofia/internal/sip:1001 at 90.198.86.129:62954] 178.79.156.136 port 32054 ->
> 90.198.86.129 port 56446 codec: 8 ms: 20
> 2011-03-05 03:15:15.373722 [DEBUG] switch_rtp.c:1621 Starting timer [soft]
> 160 bytes per 20ms
> 2011-03-05 03:15:15.374777 [DEBUG] sofia_glue.c:3235 Set 2833 dtmf send
> payload to 101
> 2011-03-05 03:15:15.374777 [DEBUG] sofia_glue.c:3240 Set 2833 dtmf receive
> payload to 101
> 2011-03-05 03:15:15.374777 [DEBUG] switch_channel.c:2795 (sofia/internal/
> sip:1001 at 90.198.86.129:62954) Callstate Change RINGING -> ACTIVE
> 2011-03-05 03:15:15.374777 [DEBUG] switch_channel.c:2807 Send signal
> skypopen/interface1 [BREAK]
> 2011-03-05 03:15:15.374777 [DEBUG] mod_skypopen.c:727
>  [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PRERING,IDLE]
> skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
> 2011-03-05 03:15:15.374777 [NOTICE] sofia.c:5267 Channel [sofia/internal/
> sip:1001 at 90.198.86.129:62954] has been answered
> 2011-03-05 03:15:15.375852 [DEBUG] mod_skypopen.c:1115
> [32b8f10|9350fb9] [DEBUG_SKYPE  1115 ][interface1     ][PRERING,IDLE]
> skypopen/interface1 CHANNEL got SWITCH_MESSAGE_INDICATE_ANSWER
> 2011-03-05 03:15:15.375852 [DEBUG] mod_skypopen.c:2723
> [32b8f10|9350fb9] [DEBUG_SKYPE  2723 ][interface1     ][PRERING,IDLE] NOT
> FOUND
> 2011-03-05 03:15:15.375852 [DEBUG] skypopen_protocol.c:1619
>  [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][PREANSW,IDLE]
> SENDING: |||ALTER CALL 142 ANSWER||||
> 2011-03-05 03:15:15.375852 [DEBUG] mod_skypopen.c:2734
> [32b8f10|9350fb9] [DEBUG_SKYPE  2734 ][interface1     ][PREANSW,IDLE] We
> answered a Skype RING on skype_call 142
> 2011-03-05 03:15:15.375852 [DEBUG] mod_skypopen.c:2742
> [32b8f10|9350fb9] [DEBUG_SKYPE  2742 ][interface1     ][PREANSW,IDLE] NEW!
>  name: interface1, state: 11, value=eric.nqz,
> tech_pvt->callid_number=eric.nqz, tech_pvt->skype_user=niqizhi
> 2011-03-05 03:15:15.472676 [DEBUG] skypopen_protocol.c:173
> [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][PREANSW,IDLE]
> READING: |||ALTER CALL 142 ANSWER|||
> 2011-03-05 03:15:15.884213 [DEBUG] mod_skypopen.c:1142
> [32b8f10|9350fb9] [DEBUG_SKYPE  1142 ][interface1     ][PREANSW,IDLE]
> Synching audio
> 2011-03-05 03:15:15.884213 [DEBUG] switch_core_session.c:709 Send signal
> skypopen/interface1 [BREAK]
> 2011-03-05 03:15:15.884213 [DEBUG] mod_skypopen.c:727
>  [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PREANSW,IDLE]
> skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
> 2011-03-05 03:15:15.884213 [DEBUG] switch_channel.c:2795
> (skypopen/interface1) Callstate Change RINGING -> ACTIVE
> 2011-03-05 03:15:15.884213 [NOTICE] switch_ivr_originate.c:3363 Channel
> [skypopen/interface1] has been answered
> 2011-03-05 03:15:15.884213 [DEBUG] mod_skypopen.c:1148
> [32b8f10|9350fb9] [DEBUG_SKYPE  1148 ][interface1     ][PREANSW,IDLE]
> skypopen/interface1 CHANNEL got SWITCH_MESSAGE_INDICATE_AUDIO_SYNC
> 2011-03-05 03:15:15.884213 [DEBUG] mod_skypopen.c:1170
> [32b8f10|9350fb9] [DEBUG_SKYPE  1170 ][interface1     ][PREANSW,IDLE]
> Synching audio
> 2011-03-05 03:15:15.884213 [DEBUG] switch_ivr_originate.c:3408 Originate
> Resulted in Success: [sofia/internal/sip:1001 at 90.198.86.129:62954]
> 2011-03-05 03:15:15.884213 [DEBUG] mod_skypopen.c:1148
> [32b8f10|9350fb9] [DEBUG_SKYPE  1148 ][interface1     ][PREANSW,IDLE]
> skypopen/interface1 CHANNEL got SWITCH_MESSAGE_INDICATE_AUDIO_SYNC
> 2011-03-05 03:15:15.884213 [DEBUG] mod_skypopen.c:1170
> [32b8f10|9350fb9] [DEBUG_SKYPE  1170 ][interface1     ][PREANSW,IDLE]
> Synching audio
> 2011-03-05 03:15:15.885293 [DEBUG] switch_ivr_originate.c:3408 Originate
> Resulted in Success: [sofia/internal/sip:1001 at 90.198.86.129:62954]
> 2011-03-05 03:15:15.885293 [DEBUG] mod_skypopen.c:1148
> [32b8f10|9350fb9] [DEBUG_SKYPE  1148 ][interface1     ][PREANSW,IDLE]
> skypopen/interface1 CHANNEL got SWITCH_MESSAGE_INDICATE_AUDIO_SYNC
> 2011-03-05 03:15:15.885293 [DEBUG] mod_skypopen.c:1170
> [32b8f10|9350fb9] [DEBUG_SKYPE  1170 ][interface1     ][PREANSW,IDLE]
> Synching audio
> 2011-03-05 03:15:15.885293 [DEBUG] switch_core_session.c:709 Send signal
> sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
> 2011-03-05 03:15:15.885293 [DEBUG] mod_skypopen.c:1173
> [32b8f10|9350fb9] [DEBUG_SKYPE  1173 ][interface1     ][PREANSW,IDLE]
> skypopen/interface1 CHANNEL got SWITCH_MESSAGE_INDICATE_BRIDGE
> 2011-03-05 03:15:15.885293 [DEBUG] mod_skypopen.c:1195
> [32b8f10|9350fb9] [DEBUG_SKYPE  1195 ][interface1     ][PREANSW,IDLE]
> Synching audio
> 2011-03-05 03:15:15.885293 [DEBUG] switch_core_session.c:709 Send signal
> skypopen/interface1 [BREAK]
> 2011-03-05 03:15:15.885293 [DEBUG] mod_skypopen.c:727
>  [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PREANSW,IDLE]
> skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
> 2011-03-05 03:15:15.885293 [DEBUG] switch_ivr_bridge.c:1234 (sofia/internal/
> sip:1001 at 90.198.86.129:62954) State Change CS_CONSUME_MEDIA ->
> CS_EXCHANGE_MEDIA
> 2011-03-05 03:15:15.885293 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
> 2011-03-05 03:15:15.885293 [DEBUG] switch_core_state_machine.c:320
> (sofia/internal/sip:1001 at 90.198.86.129:62954) Running State Change
> CS_EXCHANGE_MEDIA
> 2011-03-05 03:15:15.885293 [DEBUG] switch_core_state_machine.c:369
> (sofia/internal/sip:1001 at 90.198.86.129:62954) State EXCHANGE_MEDIA
> 2011-03-05 03:15:15.885293 [DEBUG] mod_sofia.c:552 SOFIA EXCHANGE_MEDIA
> 2011-03-05 03:15:15.886388 [DEBUG] switch_core_session.c:771 Send signal
> sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
> 2011-03-05 03:15:15.886388 [DEBUG] switch_core_session.c:771 Send signal
> skypopen/interface1 [BREAK]
> 2011-03-05 03:15:15.886388 [DEBUG] mod_skypopen.c:727
>  [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PREANSW,IDLE]
> skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
> 2011-03-05 03:15:15.906103 [DEBUG] switch_ivr_bridge.c:500
> skypopen/interface1 ending bridge by request from read function
> 2011-03-05 03:15:15.906103 [DEBUG] switch_ivr_bridge.c:581 BRIDGE THREAD
> DONE [skypopen/interface1]
> 2011-03-05 03:15:15.906103 [DEBUG] switch_ivr_bridge.c:601 Send signal
> sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
> 2011-03-05 03:15:15.909404 [DEBUG] switch_ivr_bridge.c:581 BRIDGE THREAD
> DONE [sofia/internal/sip:1001 at 90.198.86.129:62954]
> 2011-03-05 03:15:15.909404 [DEBUG] switch_ivr_bridge.c:601 Send signal
> skypopen/interface1 [BREAK]
> 2011-03-05 03:15:15.909404 [DEBUG] mod_skypopen.c:727
>  [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PREANSW,IDLE]
> skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
> 2011-03-05 03:15:15.909404 [DEBUG] switch_channel.c:2545 (sofia/internal/
> sip:1001 at 90.198.86.129:62954) Callstate Change ACTIVE -> HANGUP
> 2011-03-05 03:15:15.909404 [NOTICE] switch_ivr_bridge.c:653 Hangup
> sofia/internal/sip:1001 at 90.198.86.129:62954 [CS_EXCHANGE_MEDIA]
> [NORMAL_CLEARING]
> 2011-03-05 03:15:15.909404 [DEBUG] switch_channel.c:2561 Send signal
> sofia/internal/sip:1001 at 90.198.86.129:62954 [KILL]
> 2011-03-05 03:15:15.909404 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
> 2011-03-05 03:15:15.909404 [DEBUG] switch_core_state_machine.c:369
> (sofia/internal/sip:1001 at 90.198.86.129:62954) State EXCHANGE_MEDIA going to
> sleep
> 2011-03-05 03:15:15.909404 [DEBUG] switch_core_state_machine.c:320
> (sofia/internal/sip:1001 at 90.198.86.129:62954) Running State Change CS_HANGUP
> 2011-03-05 03:15:15.910537 [DEBUG] switch_core_state_machine.c:560
> (sofia/internal/sip:1001 at 90.198.86.129:62954) State HANGUP
> 2011-03-05 03:15:15.910537 [DEBUG] mod_sofia.c:457 Channel sofia/internal/
> sip:1001 at 90.198.86.129:62954 hanging up, cause: NORMAL_CLEARING
> 2011-03-05 03:15:15.910537 [DEBUG] switch_ivr_bridge.c:1305 sofia/internal/
> sip:1001 at 90.198.86.129:62954 skip receive message [UNBRIDGE] (channel is
> hungup already)
> 2011-03-05 03:15:15.910537 [DEBUG] mod_skypopen.c:1201
> [32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PREANSW,IDLE]
> MSG_ID=5
> 2011-03-05 03:15:15.910537 [DEBUG] switch_core_session.c:709 Send signal
> skypopen/interface1 [BREAK]
> 2011-03-05 03:15:15.910537 [DEBUG] mod_skypopen.c:727
>  [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PREANSW,IDLE]
> skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
> 2011-03-05 03:15:15.910537 [DEBUG] mod_skypopen.c:1201
> [32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PREANSW,IDLE]
> MSG_ID=27
> 2011-03-05 03:15:15.910537 [NOTICE] switch_core_state_machine.c:189
> skypopen/interface1 has executed the last dialplan instruction, hanging up.
> 2011-03-05 03:15:15.910537 [DEBUG] switch_channel.c:2545
> (skypopen/interface1) Callstate Change ACTIVE -> HANGUP
> 2011-03-05 03:15:15.910537 [NOTICE] switch_core_state_machine.c:191 Hangup
> skypopen/interface1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-05 03:15:15.910537 [DEBUG] switch_channel.c:2561 Send signal
> skypopen/interface1 [KILL]
> 2011-03-05 03:15:15.910537 [DEBUG] mod_skypopen.c:700
>  [32b8f10|9350fb9] [DEBUG_SKYPE  700  ][interface1     ][PREANSW,IDLE]
> skypopen/interface1 CHANNEL got SWITCH_SIG_KILL
> 2011-03-05 03:15:15.910537 [DEBUG] switch_core_session.c:1116 Send signal
> skypopen/interface1 [BREAK]
> 2011-03-05 03:15:15.910537 [DEBUG] mod_skypopen.c:727
>  [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][HANG_RQ,IDLE]
> skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
> 2011-03-05 03:15:15.910537 [DEBUG] switch_core_state_machine.c:366
> (skypopen/interface1) State EXECUTE going to sleep
> 2011-03-05 03:15:15.910537 [DEBUG] mod_sofia.c:500 Sending BYE to
> sofia/internal/sip:1001 at 90.198.86.129:62954
> 2011-03-05 03:15:15.910537 [DEBUG] switch_core_state_machine.c:320
> (skypopen/interface1) Running State Change CS_HANGUP
> 2011-03-05 03:15:15.910537 [DEBUG] switch_core_state_machine.c:560
> (skypopen/interface1) State HANGUP
> 2011-03-05 03:15:15.910537 [DEBUG] mod_skypopen.c:623
>  [32b8f10|9350fb9] [DEBUG_SKYPE  623  ][interface1     ][HANG_RQ,IDLE]
> hanging up skype call: 142
> 2011-03-05 03:15:15.911571 [DEBUG] skypopen_protocol.c:1619
>  [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][HANG_RQ,IDLE]
> SENDING: |||ALTER CALL 142 END HANGUP||||
> 2011-03-05 03:15:15.911571 [DEBUG] skypopen_protocol.c:1619
>  [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][HANG_RQ,IDLE]
> SENDING: |||ALTER CALL 142 HANGUP||||
> 2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:629
>  [32b8f10|9350fb9] [DEBUG_SKYPE  629  ][interface1     ][HANG_RQ,IDLE]
> interface1 CHANNEL HANGUP
> 2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:648 (skypopen/interface1)
> State Change CS_HANGUP -> CS_DESTROY
> 2011-03-05 03:15:15.911571 [DEBUG] switch_core_session.c:1116 Send signal
> skypopen/interface1 [BREAK]
> 2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:727
>  [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][DOWN,IDLE]
> skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
> 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:560
> (skypopen/interface1) State HANGUP going to sleep
> 2011-03-05 03:15:15.911571 [DEBUG] switch_core_session.c:1288 Session 24
> (skypopen/interface1) Locked, Waiting on external entities
> 2011-03-05 03:15:15.911571 [NOTICE] switch_core_session.c:1306 Session 24
> (skypopen/interface1) Ended
> 2011-03-05 03:15:15.911571 [NOTICE] switch_core_session.c:1308 Close Channel
> skypopen/interface1 [CS_DESTROY]
> 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/sip:1001 at 90.198.86.129:62954 Standard HANGUP, cause:
> NORMAL_CLEARING
> 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:560
> (sofia/internal/sip:1001 at 90.198.86.129:62954) State HANGUP going to sleep
> 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:351
> (sofia/internal/sip:1001 at 90.198.86.129:62954) State Change CS_HANGUP ->
> CS_REPORTING
> 2011-03-05 03:15:15.911571 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
> 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:320
> (sofia/internal/sip:1001 at 90.198.86.129:62954) Running State Change
> CS_REPORTING
> 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:620
> (sofia/internal/sip:1001 at 90.198.86.129:62954) State REPORTING
> 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/sip:1001 at 90.198.86.129:62954 Standard REPORTING, cause:
> NORMAL_CLEARING
> 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:620
> (sofia/internal/sip:1001 at 90.198.86.129:62954) State REPORTING going to sleep
> 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:449
> (skypopen/interface1) Callstate Change HANGUP -> DOWN
> 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:452
> (skypopen/interface1) Running State Change CS_DESTROY
> 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:462
> (skypopen/interface1) State DESTROY
> 2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:479
>  [32b8f10|9350fb9] [DEBUG_SKYPE  479  ][interface1     ][DOWN,IDLE]
> interface1 CHANNEL DESTROY c958d9be-46d6-11e0-bb71-d9b9e6428344
> 2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:493
>  [32b8f10|9350fb9] [DEBUG_SKYPE  493  ][interface1     ][DOWN,IDLE] audio
> tcp threads to DIE
> 2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:503
>  [32b8f10|9350fb9] [DEBUG_SKYPE  503  ][interface1     ][DOWN,IDLE] audio
> tcp srv thread DEAD 0
> 2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:513
>  [32b8f10|9350fb9] [DEBUG_SKYPE  513  ][interface1     ][DOWN,IDLE] audio
> tcp cli thread DEAD 0
> 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:345
> (sofia/internal/sip:1001 at 90.198.86.129:62954) State Change CS_REPORTING ->
> CS_DESTROY
> 2011-03-05 03:15:15.911571 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
> 2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:524
>  [32b8f10|9350fb9] [DEBUG_SKYPE  524  ][interface1     ][DOWN,IDLE] codecs
> DOWN
> 2011-03-05 03:15:15.911571 [DEBUG] switch_core_session.c:1288 Session 25
> (sofia/internal/sip:1001 at 90.198.86.129:62954) Locked, Waiting on external
> entities
> 2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:579
>  [32b8f10|9350fb9] [DEBUG_SKYPE  579  ][interface1     ][IDLE,IDLE] CHANNEL
> DESTROYED c958d9be-46d6-11e0-bb71-d9b9e6428344
> 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:60
> skypopen/interface1 Standard DESTROY
> 2011-03-05 03:15:15.911571 [NOTICE] switch_core_session.c:1306 Session 25
> (sofia/internal/sip:1001 at 90.198.86.129:62954) Ended
> 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:462
> (skypopen/interface1) State DESTROY going to sleep
> 2011-03-05 03:15:15.911571 [NOTICE] switch_core_session.c:1308 Close Channel
> sofia/internal/sip:1001 at 90.198.86.129:62954 [CS_DESTROY]
> 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:449
> (sofia/internal/sip:1001 at 90.198.86.129:62954) Callstate Change HANGUP ->
> DOWN
> 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:452
> (sofia/internal/sip:1001 at 90.198.86.129:62954) Running State Change
> CS_DESTROY
> 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:462
> (sofia/internal/sip:1001 at 90.198.86.129:62954) State DESTROY
> 2011-03-05 03:15:15.911571 [DEBUG] mod_sofia.c:362 sofia/internal/
> sip:1001 at 90.198.86.129:62954 SOFIA DESTROY
> 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/sip:1001 at 90.198.86.129:62954 Standard DESTROY
> 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:462
> (sofia/internal/sip:1001 at 90.198.86.129:62954) State DESTROY going to sleep
> 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:173
> [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE] READING:
> |||CALL 142 STATUS INPROGRESS|||
> 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:719
> [32b8f10|9350fb9] [DEBUG_SKYPE  719  ][interface1     ][IDLE,IDLE] no
> tech_pvt->session_uuid_str
> 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:725
> [32b8f10|9350fb9] [DEBUG_SKYPE  725  ][interface1     ][IDLE,IDLE]
> skype_call: 142 is now active
> 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:732
> [32b8f10|9350fb9] [DEBUG_SKYPE  732  ][interface1     ][UP,INPROGRS] START
> start_audio_threads
> 2011-03-05 03:15:16.149772 [DEBUG] mod_skypopen.c:2203
> [32b8f10|9350fb9] [DEBUG_SKYPE  2203 ][interface1     ][UP,INPROGRS] started
> tcp_srv_thread thread.
> 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:89
>  [32b8f10|9350fb9] [DEBUG_SKYPE  89   ][interface1     ][UP,INPROGRS]
> Binded! *which_port=32783, tech_pvt->tcp_cli_port=32782,
> tech_pvt->tcp_srv_port=32783
> 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:94
>  [32b8f10|9350fb9] [DEBUG_SKYPE  94   ][interface1     ][UP,INPROGRS] 1
> SO_RCVBUF is 87380, size is 4
> 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:98
>  [32b8f10|9350fb9] [DEBUG_SKYPE  98   ][interface1     ][UP,INPROGRS] 1
> SO_SNDBUF is 16384, size is 4
> 2011-03-05 03:15:16.149772 [DEBUG] mod_skypopen.c:2219
> [32b8f10|9350fb9] [DEBUG_SKYPE  2219 ][interface1     ][UP,INPROGRS] started
> tcp_cli_thread thread.
> 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:115
> [32b8f10|9350fb9] [DEBUG_SKYPE  115  ][interface1     ][UP,INPROGRS] 2
> SO_RCVBUF is 87380, size is 4
> 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:130
> [32b8f10|9350fb9] [DEBUG_SKYPE  130  ][interface1     ][UP,INPROGRS] 2
> SO_SNDBUF is 16384, size is 4
> 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:89
>  [32b8f10|9350fb9] [DEBUG_SKYPE  89   ][interface1     ][UP,INPROGRS]
> Binded! *which_port=32784, tech_pvt->tcp_cli_port=32784,
> tech_pvt->tcp_srv_port=32783
> 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:134
> [32b8f10|9350fb9] [DEBUG_SKYPE  134  ][interface1     ][UP,INPROGRS]
> TCP_NODELAY is 0
> 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:94
>  [32b8f10|9350fb9] [DEBUG_SKYPE  94   ][interface1     ][UP,INPROGRS] 1
> SO_RCVBUF is 87380, size is 4
> 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:141
> [32b8f10|9350fb9] [DEBUG_SKYPE  141  ][interface1     ][UP,INPROGRS]
> TCP_NODELAY is 0
> 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:98
>  [32b8f10|9350fb9] [DEBUG_SKYPE  98   ][interface1     ][UP,INPROGRS] 1
> SO_SNDBUF is 16384, size is 4
> 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:833
> [32b8f10|9350fb9] [DEBUG_SKYPE  833  ][interface1     ][UP,INPROGRS] started
> tcp_srv_thread thread.
> 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:115
> [32b8f10|9350fb9] [DEBUG_SKYPE  115  ][interface1     ][UP,INPROGRS] 2
> SO_RCVBUF is 87380, size is 4
> 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:130
> [32b8f10|9350fb9] [DEBUG_SKYPE  130  ][interface1     ][UP,INPROGRS] 2
> SO_SNDBUF is 16384, size is 4
> 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:134
> [32b8f10|9350fb9] [DEBUG_SKYPE  134  ][interface1     ][UP,INPROGRS]
> TCP_NODELAY is 0
> 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:141
> [32b8f10|9350fb9] [DEBUG_SKYPE  141  ][interface1     ][UP,INPROGRS]
> TCP_NODELAY is 0
> 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:997
> [32b8f10|9350fb9] [DEBUG_SKYPE  997  ][interface1     ][UP,INPROGRS] started
> tcp_cli_thread thread.
> 2011-03-05 03:15:16.250297 [DEBUG] skypopen_protocol.c:1619
>  [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][UP,INPROGRS]
> SENDING: |||ALTER CALL 142 SET_INPUT PORT="32784"||||
> 2011-03-05 03:15:16.250297 [DEBUG] skypopen_protocol.c:1619
>  [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][UP,INPROGRS]
> SENDING: |||#output ALTER CALL 142 SET_OUTPUT PORT="32783"||||
> 2011-03-05 03:15:16.250297 [WARNING] skypopen_protocol.c:2036
>  [32b8f10|9350fb9] [WARNINGA     2036 ][interface1     ][UP,INPROGRS] no
> tech_pvt->session_uuid_str after INPROGRESS, let's hangup
> 2011-03-05 03:15:16.250297 [DEBUG] skypopen_protocol.c:1619
>  [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][UP,INPROGRS]
> SENDING: |||ALTER CALL 142 HANGUP||||
> 2011-03-05 03:15:16.250297 [DEBUG] skypopen_protocol.c:173
> [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][UP,INPROGRS]
> READING: |||CALL 142 VIDEO_SEND_STATUS AVAILABLE|||
> 2011-03-05 03:15:16.290473 [DEBUG] skypopen_protocol.c:173
> [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][UP,INPROGRS]
> READING: |||CALL 142 VIDEO_STATUS VIDEO_BOTH_ENABLED|||
> 2011-03-05 03:15:16.363839 [DEBUG] skypopen_protocol.c:173
> [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][UP,INPROGRS]
> READING: |||CALL 142 DURATION 1|||
> 2011-03-05 03:15:16.432605 [DEBUG] skypopen_protocol.c:173
> [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][UP,INPROGRS]
> READING: |||ALTER CALL 142 END HANGUP|||
> 2011-03-05 03:15:16.494506 [DEBUG] skypopen_protocol.c:173
> [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][UP,INPROGRS]
> READING: |||ERROR 559 CALL: Action failed|||
> 2011-03-05 03:15:16.495584 [DEBUG] skypopen_protocol.c:228
> [32b8f10|9350fb9] [DEBUG_SKYPE  228  ][interface1     ][UP,INPROGRS] Skype
> got ERROR about a failed action (probably TRYING to HANGUP A CALL), no
> problem: |||ERROR 559 CALL: Action failed|||
> 2011-03-05 03:15:16.710271 [DEBUG] skypopen_protocol.c:173
> [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][UP,INPROGRS]
> READING: |||ERROR 589 ALTER CALL: unable to alter input/output|||
> 2011-03-05 03:15:16.711346 [DEBUG] skypopen_protocol.c:247
> [32b8f10|9350fb9] [DEBUG_SKYPE  247  ][interface1     ][UP,INPROGRS] Skype
> client was not able to correctly manage tcp audio sockets, probably got a
> local or remote hangup: |||ERROR 589 ALTER CALL: unable to alter
> input/output|||
> 2011-03-05 03:15:16.711346 [DEBUG] skypopen_protocol.c:1619
>  [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][UP,INPROGRS]
> SENDING: |||ALTER CALL 142 HANGUP||||
> 2011-03-05 03:15:16.711346 [DEBUG] skypopen_protocol.c:1619
>  [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][UP,INPROGRS]
> SENDING: |||ALTER CALL 142 END HANGUP||||
> 2011-03-05 03:15:16.711346 [DEBUG] mod_skypopen.c:1413
> [32b8f10|9350fb9] [DEBUG_SKYPE  1413 ][interface1     ][DOWN,INPROGRS] skype
> call ended
> 2011-03-05 03:15:16.711346 [DEBUG] mod_skypopen.c:1433
> [32b8f10|9350fb9] [DEBUG_SKYPE  1433 ][interface1     ][DOWN,INPROGRS] no
> session
> 2011-03-05 03:15:16.711346 [DEBUG] mod_skypopen.c:1436
> [32b8f10|9350fb9] [DEBUG_SKYPE  1436 ][interface1     ][DOWN,INPROGRS] audio
> tcp threads to DIE
> 2011-03-05 03:15:16.712434 [DEBUG] skypopen_protocol.c:960
> [32b8f10|9350fb9] [DEBUG_SKYPE  960  ][interface1     ][DOWN,INPROGRS]
> incoming audio (read) server (I am it) EXITING
> 2011-03-05 03:15:16.715692 [DEBUG] mod_skypopen.c:1446
> [32b8f10|9350fb9] [DEBUG_SKYPE  1446 ][interface1     ][DOWN,INPROGRS] audio
> tcp srv thread DEAD 1
> 2011-03-05 03:15:16.750763 [DEBUG] skypopen_protocol.c:1112
>  [32b8f10|9350fb9] [DEBUG_SKYPE  1112 ][interface1     ][DOWN,INPROGRS]
> outbound audio server (I am it) EXITING
> 2011-03-05 03:15:16.751849 [DEBUG] mod_skypopen.c:1456
> [32b8f10|9350fb9] [DEBUG_SKYPE  1456 ][interface1     ][DOWN,INPROGRS] audio
> tcp cli thread DEAD 7
> 2011-03-05 03:15:16.751849 [DEBUG] skypopen_protocol.c:173
> [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE] READING:
> |||#output ERROR 589 ALTER CALL: unable to alter input/output|||
> 2011-03-05 03:15:16.779660 [DEBUG] skypopen_protocol.c:173
> [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE] READING:
> |||ERROR 559 CALL: Action failed|||
> 2011-03-05 03:15:16.780762 [DEBUG] skypopen_protocol.c:228
> [32b8f10|9350fb9] [DEBUG_SKYPE  228  ][interface1     ][IDLE,IDLE] Skype got
> ERROR about a failed action (probably TRYING to HANGUP A CALL), no problem:
> |||ERROR 559 CALL: Action failed|||
> 2011-03-05 03:15:16.828877 [DEBUG] skypopen_protocol.c:173
> [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE] READING:
> |||ERROR 559 CALL: Action failed|||
> 2011-03-05 03:15:16.828877 [DEBUG] skypopen_protocol.c:228
> [32b8f10|9350fb9] [DEBUG_SKYPE  228  ][interface1     ][IDLE,IDLE] Skype got
> ERROR about a failed action (probably TRYING to HANGUP A CALL), no problem:
> |||ERROR 559 CALL: Action failed|||
> 2011-03-05 03:15:16.869183 [DEBUG] skypopen_protocol.c:173
> [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE] READING:
> |||ERROR 559 CALL: Action failed|||
> 2011-03-05 03:15:16.869183 [DEBUG] skypopen_protocol.c:228
> [32b8f10|9350fb9] [DEBUG_SKYPE  228  ][interface1     ][IDLE,IDLE] Skype got
> ERROR about a failed action (probably TRYING to HANGUP A CALL), no problem:
> |||ERROR 559 CALL: Action failed|||
> 2011-03-05 03:15:16.911668 [DEBUG] skypopen_protocol.c:173
> [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE] READING:
> |||CALL 142 STATUS FINISHED|||
> 2011-03-05 03:15:16.911668 [DEBUG] skypopen_protocol.c:661
> [32b8f10|9350fb9] [DEBUG_SKYPE  661  ][interface1     ][IDLE,IDLE]
> skype_call 142 is MY call, now I'm going DOWN
> 2011-03-05 03:15:16.911668 [DEBUG] mod_skypopen.c:1413
> [32b8f10|9350fb9] [DEBUG_SKYPE  1413 ][interface1     ][IDLE,IDLE] skype
> call ended
> 2011-03-05 03:15:16.911668 [DEBUG] mod_skypopen.c:1433
> [32b8f10|9350fb9] [DEBUG_SKYPE  1433 ][interface1     ][IDLE,IDLE] no
> session
> 2011-03-05 03:15:16.911668 [DEBUG] mod_skypopen.c:1436
> [32b8f10|9350fb9] [DEBUG_SKYPE  1436 ][interface1     ][DOWN,IDLE] audio tcp
> threads to DIE
> 2011-03-05 03:15:16.911668 [DEBUG] mod_skypopen.c:1446
> [32b8f10|9350fb9] [DEBUG_SKYPE  1446 ][interface1     ][DOWN,IDLE] audio tcp
> srv thread DEAD 0
> 2011-03-05 03:15:16.911668 [DEBUG] mod_skypopen.c:1456
> [32b8f10|9350fb9] [DEBUG_SKYPE  1456 ][interface1     ][DOWN,IDLE] audio tcp
> cli thread DEAD 0
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From benkokakao at gmail.com  Sat Mar  5 10:19:07 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Sat, 5 Mar 2011 09:19:07 +0200
Subject: [Freeswitch-users] How could I forward the SIP incoming call to
 one of the available extensions?
In-Reply-To: <AANLkTi=A-NSss214Qe=aiofh-qhofDKZMFBx3v8=DgDi@mail.gmail.com>
References: <AANLkTi=A-NSss214Qe=aiofh-qhofDKZMFBx3v8=DgDi@mail.gmail.com>
Message-ID: <AANLkTikEwOW2KJz3z-of86wb4a8w8=SxoPCo0E3avVqY@mail.gmail.com>

> Right now I only have one extension in the SIP configuration as the
> destination. Could I put a list of extensions then FS forwards the call to
> the first available one?

I'm not sure if i understand you exactly. By "extension" do you mean a
SIP-endpoint like a phone or a gateway? Take a look at this simple
example, is this good enough for your situation?:
http://wiki.freeswitch.org/wiki/Extension_Status_Example

If not, please let us now in more detail what you want to achieve(Is
this a Call Center solution? Take a look at
http://wiki.freeswitch.org/wiki/Mod_fifo)

Best regards,
Christian


From codecomplete at free.fr  Sat Mar  5 12:31:17 2011
From: codecomplete at free.fr (GillesToo)
Date: Sat, 5 Mar 2011 01:31:17 -0800 (PST)
Subject: [Freeswitch-users] [Dahdi/FreeTDM] Does FS provide call
	supervision?
In-Reply-To: <AANLkTinnCCWPcHo62W=fn6-M-4BP=2VZPm0PFTMNBsY0@mail.gmail.com>
References: <1299234392809-6087967.post@n2.nabble.com>
	<AANLkTinnCCWPcHo62W=fn6-M-4BP=2VZPm0PFTMNBsY0@mail.gmail.com>
Message-ID: <1299317477152-6091429.post@n2.nabble.com>


Moises Silva wrote:
> FreeTDM currently has the same limitation unless your telco/switch sends
> polarity reverse on answer. If polarity reversal is enabled, then FreeTDM
> will not put the channel in state UP until the telco reverses the polarity
> to signal answer.

Thanks Moises for the information.


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dahdi-FreeTDM-Does-FS-provide-call-supervision-tp6087967p6091429.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kbdfck at gmail.com  Sat Mar  5 13:18:13 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Sat, 5 Mar 2011 13:18:13 +0300
Subject: [Freeswitch-users] Perl script or ESL memory leaks, how to check?
Message-ID: <AANLkTik+i8CNDH90TKp2wNYhXrwpS8DyOzYmSma3rtqw@mail.gmail.com>

Hi All,

I'm using Net::Server::Prefork togethere with perl ESL to listen for
outbound socket connections from Freeswitch in sync mode.
Due to use of Prefork, one script instance processes many requests
(max_requests).
I'm doing database request, finding user to dial and bridge.
I noticed that if channel hungs up by initiator, memory consumed by my
script is never returned to OS, but if bridge finished by remote side (or
freeswitch), memory returns to OS and everything works ok.

I thought there are some circular references in my script or something like
that, but that would affect both cases. In my case when call is finished by
freeswitch by reaching end of script and then end of dialplan, everything
works fine, but if call is finished by originating channel, memory leak
occurs.

How can I trace these leaks in my script or ESL library? What should I start
from?

-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From xyangni at gmail.com  Sat Mar  5 14:06:55 2011
From: xyangni at gmail.com (Yihui Li)
Date: Sat, 5 Mar 2011 11:06:55 +0000
Subject: [Freeswitch-users] skype to x-lite birdge hangup imediately
	when answered
In-Reply-To: <AANLkTimiNY193=vVer4AbCy2h7kLmqwbQxhNiM3ydnwL@mail.gmail.com>
References: <AANLkTik3De=vPS8VfZfcJCdQNjHc7EvjxW=uej33JFjT@mail.gmail.com>
	<AANLkTimiNY193=vVer4AbCy2h7kLmqwbQxhNiM3ydnwL@mail.gmail.com>
Message-ID: <AANLkTi=XYp5c5F2AT84nxw9wvXVC_Kv4rJVtOc+9f05j@mail.gmail.com>

Hi Giovanni,

I have tested your suggestion, the call works fine, when answered in dial
plan before bridge.


On Sat, Mar 5, 2011 at 5:01 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> Hi Yi,
>
> could you please test your extension inserting an action "answer"
> before the action "bridge" ?
>
> -giovanni
>
> On 3/5/11, Yihui Li <xyangni at gmail.com> wrote:
> > Dear all,
> >
> > I am tring skypopen with latest git-head.
> > There are just 1 skype instance. When forward skype call to 5000, it
> works
> > fine. It also works fine with originate skypopen/interface1/*****
> > &bridge(user/1001)
> >
> >  But when I try to forward skype to 1001 with the following dial plan,
> call
> > hang up imediately when answered.
> >
> >   <extension name="skypetest">
> >     <condition field="destination_number" expression="^(5100)$">
> >       <action application="bridge" data="user/1001"/>
> >     </condition>
> >   </extension>
> >
> > I am working on l ubuntu 10.10 server edition and skype static 2.0.0.72
> >
> > here is the skype configure
> > =============full mod_skypopen.conf.xml=====================
> > <configuration name="skypopen.conf" description="Skypopen Configuration">
> >   <global_settings>
> >     <param name="debug" value="8"/>
> >     <param name="dialplan" value="XML"/>
> >     <param name="context" value="default"/>
> >     <param name="destination" value="5100"/>
> >     <param name="skype_user" value="user1"/>
> >     <param name="report_incoming_chatmessages" value="true"/>
> >     <param name="silent_mode" value="false"/>
> >     <param name="write_silence_when_idle" value="true"/>
> >   </global_settings>
> >   <!-- one entry here per each skypopen interface -->
> >   <per_interface_settings>
> >     <interface id="1" name="interface1">
> >         <param name="X11-display" value="localhost:10.0"/>
> >         <param name="skype_user" value="niqizhi"/>
> >     </interface>
> >   </per_interface_settings>
> > </configuration>
> >
> >
> >
> >
> > the debug message of the call is listed below. May I ask what is the
> > possiable cause? Thanks.
> >
> >
> >
> > 2011-03-05 03:15:13.353532 [DEBUG] skypopen_protocol.c:173
> > [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE]
> READING:
> > |||CALL 142 CONF_ID 0|||
> > 2011-03-05 03:15:13.438727 [DEBUG] skypopen_protocol.c:173
> > [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE]
> READING:
> > |||CALL 142 STATUS RINGING|||
> > 2011-03-05 03:15:13.438727 [DEBUG] skypopen_protocol.c:595
> > [32b8f10|9350fb9] [DEBUG_SKYPE  595  ][interface1     ][IDLE,IDLE] NO
> ACTIVE
> > calls in this moment, skype_call 142 is RINGING, to ask PARTNER_DISPNAME
> and
> > PARTNER_HANDLE
> > 2011-03-05 03:15:13.438727 [DEBUG] skypopen_protocol.c:1619
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][IDLE,IDLE]
> SENDING:
> > |||GET CALL 142 PARTNER_DISPNAME||||
> > 2011-03-05 03:15:13.438727 [DEBUG] skypopen_protocol.c:1619
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][IDLE,IDLE]
> SENDING:
> > |||GET CALL 142 PARTNER_HANDLE||||
> > 2011-03-05 03:15:13.576818 [DEBUG] skypopen_protocol.c:173
> > [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE]
> READING:
> > |||CALL 142 PARTNER_DISPNAME Eric Ni|||
> > 2011-03-05 03:15:13.609592 [DEBUG] skypopen_protocol.c:173
> > [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE]
> READING:
> > |||CALL 142 PARTNER_HANDLE eric.nqz|||
> > 2011-03-05 03:15:13.609592 [DEBUG] skypopen_protocol.c:506
> > [32b8f10|9350fb9] [DEBUG_SKYPE  506  ][interface1     ][IDLE,IDLE] Call
> 142
> > go to skypopen_partner_handle_ring
> > 2011-03-05 03:15:13.609592 [DEBUG] mod_skypopen.c:2626
> > [32b8f10|9350fb9] [DEBUG_SKYPE  2626 ][interface1     ][IDLE,IDLE] NOT
> FOUND
> > 2011-03-05 03:15:13.609592 [DEBUG] mod_skypopen.c:2244
> > [32b8f10|9350fb9] [DEBUG_SKYPE  2244 ][interface1     ][PRERING,IDLE] 2
> > SESSION_REQUEST c958d9be-46d6-11e0-bb71-d9b9e6428344
> > 2011-03-05 03:15:13.609592 [DEBUG] mod_skypopen.c:244
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  244  ][interface1     ][PRERING,IDLE]
> > codecs UP
> > 2011-03-05 03:15:13.609592 [DEBUG] mod_skypopen.c:273
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  273  ][interface1     ][PRERING,IDLE]
> > skypopen_tech_init SUCCESS
> > 2011-03-05 03:15:13.609592 [NOTICE] switch_channel.c:812 New Channel
> > skypopen/interface1 [c958d9be-46d6-11e0-bb71-d9b9e6428344]
> > 2011-03-05 03:15:13.609592 [DEBUG] mod_skypopen.c:2268
> (skypopen/interface1)
> > State Change CS_NEW -> CS_INIT
> > 2011-03-05 03:15:13.609592 [DEBUG] switch_core_session.c:1116 Send signal
> > skypopen/interface1 [BREAK]
> > 2011-03-05 03:15:13.609592 [DEBUG] mod_skypopen.c:727
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PRERING,IDLE]
> > skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
> > 2011-03-05 03:15:13.609592 [DEBUG] mod_skypopen.c:2280
> > [32b8f10|9350fb9] [DEBUG_SKYPE  2280 ][interface1     ][PRERING,IDLE]
> > new_inbound_channel
> > 2011-03-05 03:15:13.610977 [DEBUG] switch_core_state_machine.c:320
> > (skypopen/interface1) Running State Change CS_INIT
> > 2011-03-05 03:15:13.610977 [DEBUG] switch_core_state_machine.c:356
> > (skypopen/interface1) State INIT
> > 2011-03-05 03:15:13.610977 [DEBUG] mod_skypopen.c:463
> (skypopen/interface1)
> > State Change CS_INIT -> CS_ROUTING
> > 2011-03-05 03:15:13.610977 [DEBUG] switch_core_session.c:1116 Send signal
> > skypopen/interface1 [BREAK]
> > 2011-03-05 03:15:13.610977 [DEBUG] mod_skypopen.c:727
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PRERING,IDLE]
> > skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
> > 2011-03-05 03:15:13.610977 [DEBUG] mod_skypopen.c:464
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  464  ][interface1     ][PRERING,IDLE]
> > interface1 CHANNEL INIT c958d9be-46d6-11e0-bb71-d9b9e6428344
> > 2011-03-05 03:15:13.610977 [DEBUG] switch_core_state_machine.c:356
> > (skypopen/interface1) State INIT going to sleep
> > 2011-03-05 03:15:13.610977 [DEBUG] switch_core_state_machine.c:320
> > (skypopen/interface1) Running State Change CS_ROUTING
> > 2011-03-05 03:15:13.610977 [DEBUG] switch_channel.c:1664
> > (skypopen/interface1) Callstate Change DOWN -> RINGING
> > 2011-03-05 03:15:13.610977 [DEBUG] switch_core_state_machine.c:359
> > (skypopen/interface1) State ROUTING
> > 2011-03-05 03:15:13.610977 [DEBUG] mod_skypopen.c:664
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  664  ][interface1     ][PRERING,IDLE]
> > interface1 CHANNEL ROUTING
> > 2011-03-05 03:15:13.610977 [DEBUG] switch_core_state_machine.c:77
> > skypopen/interface1 Standard ROUTING
> > 2011-03-05 03:15:13.610977 [INFO] mod_dialplan_xml.c:331 Processing Eric
> Ni
> > <eric.nqz>->5100 in context default
> > Dialplan: skypopen/interface1 parsing [default->unloop] continue=false
> > Dialplan: skypopen/interface1 Regex (PASS) [unloop] ${unroll_loops}(true)
> =~
> > /^true$/ break=on-false
> > Dialplan: skypopen/interface1 Regex (FAIL) [unloop] ${sip_looped_call}()
> =~
> > /^true$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->tod_example]
> continue=true
> > Dialplan: skypopen/interface1 Date/Time Match (FAIL) [tod_example]
> > break=on-false
> > Dialplan: skypopen/interface1 parsing [default->holiday_example]
> > continue=true
> > Dialplan: skypopen/interface1 Date/Time Match (FAIL) [holiday_example]
> > break=on-false
> > Dialplan: skypopen/interface1 parsing [default->global-intercept]
> > continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [global-intercept]
> > destination_number(5100) =~ /^886$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->group-intercept]
> > continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [group-intercept]
> > destination_number(5100) =~ /^\*8$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->intercept-ext]
> > continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [intercept-ext]
> > destination_number(5100) =~ /^\*\*(\d+)$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->redial] continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [redial]
> destination_number(5100)
> > =~ /^(redial|870)$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->global] continue=true
> > Dialplan: skypopen/interface1 Regex (FAIL) [global] ${call_debug}(false)
> =~
> > /^true$/ break=never
> > Dialplan: skypopen/interface1 Regex (FAIL) [global] ${sip_has_crypto}()
> =~
> > /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
> > Dialplan: skypopen/interface1 Absolute Condition [global]
> > Dialplan: skypopen/interface1 Action
> > hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
> > Dialplan: skypopen/interface1 Action
> >
> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
> > Dialplan: skypopen/interface1 Action
> > hash(insert/${domain_name}-last_dial/global/${uuid})
> > Dialplan: skypopen/interface1 Action set(RFC2822_DATE=${strftime(%a, %d
> %b
> > %Y %T %z)})
> > Dialplan: skypopen/interface1 parsing [default->snom-demo-2]
> continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [snom-demo-2]
> > destination_number(5100) =~ /^9001$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->snom-demo-1]
> continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [snom-demo-1]
> > destination_number(5100) =~ /^9000$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->eavesdrop] continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [eavesdrop]
> > destination_number(5100) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->eavesdrop] continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [eavesdrop]
> > destination_number(5100) =~ /^779$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->call_return]
> continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [call_return]
> > destination_number(5100) =~ /^\*69$|^869$|^lcr$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->del-group] continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [del-group]
> > destination_number(5100) =~ /^80(\d{2})$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->add-group] continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [add-group]
> > destination_number(5100) =~ /^81(\d{2})$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->call-group-simo]
> > continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [call-group-simo]
> > destination_number(5100) =~ /^82(\d{2})$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->call-group-order]
> > continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [call-group-order]
> > destination_number(5100) =~ /^83(\d{2})$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->extension-intercom]
> > continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [extension-intercom]
> > destination_number(5100) =~ /^8(10[01][0-9])$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->Local_Extension]
> > continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [Local_Extension]
> > destination_number(5100) =~ /^(10[01][0-9])$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->Local_Extension_Skinny]
> > continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [Local_Extension_Skinny]
> > destination_number(5100) =~ /^(11[01][0-9])$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->group_dial_sales]
> > continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [group_dial_sales]
> > destination_number(5100) =~ /^2000$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->group_dial_support]
> > continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [group_dial_support]
> > destination_number(5100) =~ /^2001$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->group_dial_billing]
> > continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [group_dial_billing]
> > destination_number(5100) =~ /^2002$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->operator] continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [operator]
> > destination_number(5100) =~ /^(operator|0)$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->vmain] continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [vmain]
> destination_number(5100)
> > =~ /^vmain$|^4000$|^\*98$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->sip_uri] continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [sip_uri]
> > destination_number(5100) =~ /^sip:(.*)$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->nb_conferences]
> > continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [nb_conferences]
> > destination_number(5100) =~ /^(30\d{2})$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->wb_conferences]
> > continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [wb_conferences]
> > destination_number(5100) =~ /^(31\d{2})$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->uwb_conferences]
> > continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [uwb_conferences]
> > destination_number(5100) =~ /^(32\d{2})$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->cdquality_conferences]
> > continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [cdquality_conferences]
> > destination_number(5100) =~ /^(33\d{2})$/ break=on-false
> > Dialplan: skypopen/interface1 parsing
> > [default->freeswitch_public_conf_via_sip] continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL)
> [freeswitch_public_conf_via_sip]
> > destination_number(5100) =~ /^9(888|8888|1616|3232)$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->mad_boss_intercom]
> > continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [mad_boss_intercom]
> > destination_number(5100) =~ /^0911$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->mad_boss_intercom]
> > continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [mad_boss_intercom]
> > destination_number(5100) =~ /^0912$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->mad_boss] continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [mad_boss]
> > destination_number(5100) =~ /^0913$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->ivr_demo] continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [ivr_demo]
> > destination_number(5100) =~ /^5000$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->dynamic_conference]
> > continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [dynamic_conference]
> > destination_number(5100) =~ /^5001$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->rtp_multicast_page]
> > continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [rtp_multicast_page]
> > destination_number(5100) =~ /^pagegroup$|^7243$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->park] continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [park]
> destination_number(5100)
> > =~ /^5900$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->unpark] continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [unpark]
> destination_number(5100)
> > =~ /^5901$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->valet_park]
> continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [valet_park]
> > destination_number(5100) =~ /^(6000)$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->valet_park]
> continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [valet_park]
> > destination_number(5100) =~ /^(60\d[1-9])$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->park] continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [park] source(mod_skypopen) =~
> > /mod_sofia/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->unpark] continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [unpark] source(mod_skypopen)
> =~
> > /mod_sofia/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->park] continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [park] source(mod_skypopen) =~
> > /mod_sofia/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->unpark] continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [unpark] source(mod_skypopen)
> =~
> > /mod_sofia/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->wait] continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [wait]
> destination_number(5100)
> > =~ /^wait$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->fax_receive]
> continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [fax_receive]
> > destination_number(5100) =~ /^9178$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->fax_transmit]
> continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [fax_transmit]
> > destination_number(5100) =~ /^9179$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->ringback_180]
> continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [ringback_180]
> > destination_number(5100) =~ /^9180$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->ringback_183_uk_ring]
> > continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [ringback_183_uk_ring]
> > destination_number(5100) =~ /^9181$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->ringback_183_music_ring]
> > continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [ringback_183_music_ring]
> > destination_number(5100) =~ /^9182$/ break=on-false
> > Dialplan: skypopen/interface1 parsing
> > [default->ringback_post_answer_uk_ring] continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [ringback_post_answer_uk_ring]
> > destination_number(5100) =~ /^9183$/ break=on-false
> > Dialplan: skypopen/interface1 parsing
> [default->ringback_post_answer_music]
> > continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [ringback_post_answer_music]
> > destination_number(5100) =~ /^9184$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->ClueCon] continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [ClueCon]
> > destination_number(5100) =~ /^9191$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->show_info] continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [show_info]
> > destination_number(5100) =~ /^9192$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->video_record]
> continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [video_record]
> > destination_number(5100) =~ /^9193$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->video_playback]
> > continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [video_playback]
> > destination_number(5100) =~ /^9194$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->delay_echo]
> continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [delay_echo]
> > destination_number(5100) =~ /^9195$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->echo] continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [echo]
> destination_number(5100)
> > =~ /^9196$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->milliwatt] continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [milliwatt]
> > destination_number(5100) =~ /^9197$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->tone_stream]
> continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [tone_stream]
> > destination_number(5100) =~ /^9198$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->zrtp_enrollement]
> > continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [zrtp_enrollement]
> > destination_number(5100) =~ /^9787$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->hold_music]
> continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [hold_music]
> > destination_number(5100) =~ /^9664$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->pizza_demo]
> continue=false
> > Dialplan: skypopen/interface1 Regex (FAIL) [pizza_demo]
> > destination_number(5100) =~ /^(pizza|74992)$/ break=on-false
> > Dialplan: skypopen/interface1 parsing [default->skypetest] continue=false
> > Dialplan: skypopen/interface1 Regex (PASS) [skypetest]
> > destination_number(5100) =~ /^(5100)$/ break=on-false
> > Dialplan: skypopen/interface1 Action bridge(user/1001)
> > 2011-03-05 03:15:13.612058 [DEBUG] switch_core_state_machine.c:119
> > (skypopen/interface1) State Change CS_ROUTING -> CS_EXECUTE
> > 2011-03-05 03:15:13.612058 [DEBUG] switch_core_session.c:1116 Send signal
> > skypopen/interface1 [BREAK]
> > 2011-03-05 03:15:13.612058 [DEBUG] mod_skypopen.c:727
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PRERING,IDLE]
> > skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
> > 2011-03-05 03:15:13.612058 [DEBUG] switch_core_state_machine.c:359
> > (skypopen/interface1) State ROUTING going to sleep
> > 2011-03-05 03:15:13.612058 [DEBUG] switch_core_state_machine.c:320
> > (skypopen/interface1) Running State Change CS_EXECUTE
> > 2011-03-05 03:15:13.612058 [DEBUG] switch_core_state_machine.c:366
> > (skypopen/interface1) State EXECUTE
> > 2011-03-05 03:15:13.612058 [DEBUG] mod_skypopen.c:681
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  681  ][interface1     ][PRERING,IDLE]
> > interface1 CHANNEL EXECUTE
> > 2011-03-05 03:15:13.612058 [DEBUG] switch_core_state_machine.c:157
> > skypopen/interface1 Standard EXECUTE
> > EXECUTE skypopen/interface1
> >
> hash(insert/178.79.156.136-spymap/eric.nqz/c958d9be-46d6-11e0-bb71-d9b9e6428344)
> > 2011-03-05 03:15:13.612058 [DEBUG] mod_skypopen.c:1201
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
> > MSG_ID=26
> > 2011-03-05 03:15:13.612058 [DEBUG] mod_skypopen.c:1201
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
> > MSG_ID=27
> > EXECUTE skypopen/interface1
> > hash(insert/178.79.156.136-last_dial/eric.nqz/5100)
> > 2011-03-05 03:15:13.613142 [DEBUG] mod_skypopen.c:1201
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
> > MSG_ID=26
> > 2011-03-05 03:15:13.613142 [DEBUG] mod_skypopen.c:1201
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
> > MSG_ID=27
> > EXECUTE skypopen/interface1
> >
> hash(insert/178.79.156.136-last_dial/global/c958d9be-46d6-11e0-bb71-d9b9e6428344)
> > 2011-03-05 03:15:13.613142 [DEBUG] mod_skypopen.c:1201
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
> > MSG_ID=26
> > 2011-03-05 03:15:13.613142 [DEBUG] mod_skypopen.c:1201
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
> > MSG_ID=27
> > EXECUTE skypopen/interface1 set(RFC2822_DATE=Sat, 05 Mar 2011 03:15:13
> > +0000)
> > 2011-03-05 03:15:13.613142 [DEBUG] mod_skypopen.c:1201
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
> > MSG_ID=26
> > 2011-03-05 03:15:13.613142 [DEBUG] mod_dptools.c:1059 skypopen/interface1
> > SET [RFC2822_DATE]=[Sat, 05 Mar 2011 03:15:13 +0000]
> > 2011-03-05 03:15:13.613142 [DEBUG] mod_skypopen.c:1201
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
> > MSG_ID=27
> > EXECUTE skypopen/interface1 bridge(user/1001)
> > 2011-03-05 03:15:13.613142 [DEBUG] mod_skypopen.c:1201
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
> > MSG_ID=26
> > 2011-03-05 03:15:13.614232 [DEBUG] switch_ivr_originate.c:1971 variable
> > string 0 = [sip_invite_domain=178.79.156.136]
> > 2011-03-05 03:15:13.614232 [DEBUG] switch_ivr_originate.c:1971 variable
> > string 1 = [presence_id=1001 at 178.79.156.136]
> > 2011-03-05 03:15:13.614232 [NOTICE] switch_channel.c:812 New Channel
> > sofia/internal/sip:1001 at 90.198.86.129:62954
> [c959849a-46d6-11e0-bb72-d9b9e6428344]
> > 2011-03-05 03:15:13.614232 [DEBUG] mod_sofia.c:4151 (sofia/internal/
> > sip:1001 at 90.198.86.129:62954) State Change CS_NEW -> CS_INIT
> > 2011-03-05 03:15:13.614232 [DEBUG] switch_core_session.c:1116 Send signal
> > sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
> > 2011-03-05 03:15:13.614232 [DEBUG] switch_core_state_machine.c:320
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) Running State Change
> CS_INIT
> > 2011-03-05 03:15:13.615308 [DEBUG] switch_core_state_machine.c:356
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) State INIT
> > 2011-03-05 03:15:13.615308 [DEBUG] mod_sofia.c:84 sofia/internal/
> > sip:1001 at 90.198.86.129:62954 SOFIA INIT
> > 2011-03-05 03:15:13.615308 [DEBUG] mod_sofia.c:124 (sofia/internal/
> > sip:1001 at 90.198.86.129:62954) State Change CS_INIT -> CS_ROUTING
> > 2011-03-05 03:15:13.615308 [DEBUG] switch_core_session.c:1116 Send signal
> > sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
> > 2011-03-05 03:15:13.615308 [DEBUG] switch_core_state_machine.c:356
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) State INIT going to sleep
> > 2011-03-05 03:15:13.615308 [DEBUG] switch_core_state_machine.c:320
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) Running State Change
> > CS_ROUTING
> > 2011-03-05 03:15:13.615308 [DEBUG] switch_channel.c:1664 (sofia/internal/
> > sip:1001 at 90.198.86.129:62954) Callstate Change DOWN -> RINGING
> > 2011-03-05 03:15:13.615308 [DEBUG] switch_core_state_machine.c:359
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) State ROUTING
> > 2011-03-05 03:15:13.615308 [DEBUG] mod_sofia.c:147 sofia/internal/
> > sip:1001 at 90.198.86.129:62954 SOFIA ROUTING
> > 2011-03-05 03:15:13.615308 [DEBUG] switch_ivr_originate.c:66
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) State Change CS_ROUTING ->
> > CS_CONSUME_MEDIA
> > 2011-03-05 03:15:13.615308 [DEBUG] switch_core_session.c:1116 Send signal
> > sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
> > 2011-03-05 03:15:13.615308 [DEBUG] switch_core_state_machine.c:359
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) State ROUTING going to
> sleep
> > 2011-03-05 03:15:13.615308 [DEBUG] switch_core_state_machine.c:320
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) Running State Change
> > CS_CONSUME_MEDIA
> > 2011-03-05 03:15:13.615308 [DEBUG] switch_core_state_machine.c:378
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) State CONSUME_MEDIA
> > 2011-03-05 03:15:13.615308 [DEBUG] sofia.c:4683 Channel sofia/internal/
> > sip:1001 at 90.198.86.129:62954 entering state [calling][0]
> > 2011-03-05 03:15:13.615308 [DEBUG] switch_core_state_machine.c:378
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) State CONSUME_MEDIA going
> to
> > sleep
> > 2011-03-05 03:15:13.758128 [INFO] sofia.c:729 sofia/internal/
> > sip:1001 at 90.198.86.129:62954 Update Callee ID to "Outbound Call" <1001>
> > 2011-03-05 03:15:13.759171 [DEBUG] sofia.c:4683 Channel sofia/internal/
> > sip:1001 at 90.198.86.129:62954 entering state [proceeding][180]
> > 2011-03-05 03:15:13.759171 [NOTICE] sofia.c:4761 Ring-Ready
> sofia/internal/
> > sip:1001 at 90.198.86.129:62954!
> > 2011-03-05 03:15:13.759171 [DEBUG] mod_skypopen.c:1201
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PRERING,IDLE]
> > MSG_ID=7
> > 2011-03-05 03:15:13.759171 [DEBUG] switch_core_session.c:709 Send signal
> > skypopen/interface1 [BREAK]
> > 2011-03-05 03:15:13.759171 [DEBUG] mod_skypopen.c:727
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PRERING,IDLE]
> > skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
> > 2011-03-05 03:15:13.759171 [NOTICE] switch_ivr_originate.c:479 Ring Ready
> > skypopen/interface1!
> > 2011-03-05 03:15:13.759171 [NOTICE] switch_ivr_originate.c:479 Ring-Ready
> > skypopen/interface1!
> > 2011-03-05 03:15:15.372653 [DEBUG] sofia.c:4683 Channel sofia/internal/
> > sip:1001 at 90.198.86.129:62954 entering state [completing][200]
> > 2011-03-05 03:15:15.372653 [DEBUG] sofia.c:4694 Remote SDP:
> > v=0
> > o=- 1 2 IN IP4 90.198.86.129
> > s=CounterPath eyeBeam 1.5
> > c=IN IP4 90.198.86.129
> > t=0 0
> > m=audio 56446 RTP/AVP 8 101
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-15
> >
> > 2011-03-05 03:15:15.373722 [DEBUG] sofia.c:4683 Channel sofia/internal/
> > sip:1001 at 90.198.86.129:62954 entering state [ready][200]
> > 2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:4529 Audio Codec Compare
> > [PCMA:8:8000:20:64000]/[L16:10:16000:20:256000]
> > 2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:4529 Audio Codec Compare
> > [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000]
> > 2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:4529 Audio Codec Compare
> > [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000]
> > 2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:4529 Audio Codec Compare
> > [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000]
> > 2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:4529 Audio Codec Compare
> > [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000]
> > 2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:4529 Audio Codec Compare
> > [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
> > 2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:2760 Set Codec
> > sofia/internal/sip:1001 at 90.198.86.129:62954 PCMA/8000 20 ms 160 samples
> > 64000 bits
> > 2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:4635 Set 2833 dtmf send
> > payload to 101
> > 2011-03-05 03:15:15.373722 [DEBUG] sofia_glue.c:2990 AUDIO RTP
> > [sofia/internal/sip:1001 at 90.198.86.129:62954] 178.79.156.136 port 32054
> ->
> > 90.198.86.129 port 56446 codec: 8 ms: 20
> > 2011-03-05 03:15:15.373722 [DEBUG] switch_rtp.c:1621 Starting timer
> [soft]
> > 160 bytes per 20ms
> > 2011-03-05 03:15:15.374777 [DEBUG] sofia_glue.c:3235 Set 2833 dtmf send
> > payload to 101
> > 2011-03-05 03:15:15.374777 [DEBUG] sofia_glue.c:3240 Set 2833 dtmf
> receive
> > payload to 101
> > 2011-03-05 03:15:15.374777 [DEBUG] switch_channel.c:2795 (sofia/internal/
> > sip:1001 at 90.198.86.129:62954) Callstate Change RINGING -> ACTIVE
> > 2011-03-05 03:15:15.374777 [DEBUG] switch_channel.c:2807 Send signal
> > skypopen/interface1 [BREAK]
> > 2011-03-05 03:15:15.374777 [DEBUG] mod_skypopen.c:727
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PRERING,IDLE]
> > skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
> > 2011-03-05 03:15:15.374777 [NOTICE] sofia.c:5267 Channel [sofia/internal/
> > sip:1001 at 90.198.86.129:62954] has been answered
> > 2011-03-05 03:15:15.375852 [DEBUG] mod_skypopen.c:1115
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1115 ][interface1     ][PRERING,IDLE]
> > skypopen/interface1 CHANNEL got SWITCH_MESSAGE_INDICATE_ANSWER
> > 2011-03-05 03:15:15.375852 [DEBUG] mod_skypopen.c:2723
> > [32b8f10|9350fb9] [DEBUG_SKYPE  2723 ][interface1     ][PRERING,IDLE] NOT
> > FOUND
> > 2011-03-05 03:15:15.375852 [DEBUG] skypopen_protocol.c:1619
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][PREANSW,IDLE]
> > SENDING: |||ALTER CALL 142 ANSWER||||
> > 2011-03-05 03:15:15.375852 [DEBUG] mod_skypopen.c:2734
> > [32b8f10|9350fb9] [DEBUG_SKYPE  2734 ][interface1     ][PREANSW,IDLE] We
> > answered a Skype RING on skype_call 142
> > 2011-03-05 03:15:15.375852 [DEBUG] mod_skypopen.c:2742
> > [32b8f10|9350fb9] [DEBUG_SKYPE  2742 ][interface1     ][PREANSW,IDLE]
> NEW!
> >  name: interface1, state: 11, value=eric.nqz,
> > tech_pvt->callid_number=eric.nqz, tech_pvt->skype_user=niqizhi
> > 2011-03-05 03:15:15.472676 [DEBUG] skypopen_protocol.c:173
> > [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][PREANSW,IDLE]
> > READING: |||ALTER CALL 142 ANSWER|||
> > 2011-03-05 03:15:15.884213 [DEBUG] mod_skypopen.c:1142
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1142 ][interface1     ][PREANSW,IDLE]
> > Synching audio
> > 2011-03-05 03:15:15.884213 [DEBUG] switch_core_session.c:709 Send signal
> > skypopen/interface1 [BREAK]
> > 2011-03-05 03:15:15.884213 [DEBUG] mod_skypopen.c:727
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PREANSW,IDLE]
> > skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
> > 2011-03-05 03:15:15.884213 [DEBUG] switch_channel.c:2795
> > (skypopen/interface1) Callstate Change RINGING -> ACTIVE
> > 2011-03-05 03:15:15.884213 [NOTICE] switch_ivr_originate.c:3363 Channel
> > [skypopen/interface1] has been answered
> > 2011-03-05 03:15:15.884213 [DEBUG] mod_skypopen.c:1148
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1148 ][interface1     ][PREANSW,IDLE]
> > skypopen/interface1 CHANNEL got SWITCH_MESSAGE_INDICATE_AUDIO_SYNC
> > 2011-03-05 03:15:15.884213 [DEBUG] mod_skypopen.c:1170
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1170 ][interface1     ][PREANSW,IDLE]
> > Synching audio
> > 2011-03-05 03:15:15.884213 [DEBUG] switch_ivr_originate.c:3408 Originate
> > Resulted in Success: [sofia/internal/sip:1001 at 90.198.86.129:62954]
> > 2011-03-05 03:15:15.884213 [DEBUG] mod_skypopen.c:1148
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1148 ][interface1     ][PREANSW,IDLE]
> > skypopen/interface1 CHANNEL got SWITCH_MESSAGE_INDICATE_AUDIO_SYNC
> > 2011-03-05 03:15:15.884213 [DEBUG] mod_skypopen.c:1170
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1170 ][interface1     ][PREANSW,IDLE]
> > Synching audio
> > 2011-03-05 03:15:15.885293 [DEBUG] switch_ivr_originate.c:3408 Originate
> > Resulted in Success: [sofia/internal/sip:1001 at 90.198.86.129:62954]
> > 2011-03-05 03:15:15.885293 [DEBUG] mod_skypopen.c:1148
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1148 ][interface1     ][PREANSW,IDLE]
> > skypopen/interface1 CHANNEL got SWITCH_MESSAGE_INDICATE_AUDIO_SYNC
> > 2011-03-05 03:15:15.885293 [DEBUG] mod_skypopen.c:1170
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1170 ][interface1     ][PREANSW,IDLE]
> > Synching audio
> > 2011-03-05 03:15:15.885293 [DEBUG] switch_core_session.c:709 Send signal
> > sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
> > 2011-03-05 03:15:15.885293 [DEBUG] mod_skypopen.c:1173
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1173 ][interface1     ][PREANSW,IDLE]
> > skypopen/interface1 CHANNEL got SWITCH_MESSAGE_INDICATE_BRIDGE
> > 2011-03-05 03:15:15.885293 [DEBUG] mod_skypopen.c:1195
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1195 ][interface1     ][PREANSW,IDLE]
> > Synching audio
> > 2011-03-05 03:15:15.885293 [DEBUG] switch_core_session.c:709 Send signal
> > skypopen/interface1 [BREAK]
> > 2011-03-05 03:15:15.885293 [DEBUG] mod_skypopen.c:727
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PREANSW,IDLE]
> > skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
> > 2011-03-05 03:15:15.885293 [DEBUG] switch_ivr_bridge.c:1234
> (sofia/internal/
> > sip:1001 at 90.198.86.129:62954) State Change CS_CONSUME_MEDIA ->
> > CS_EXCHANGE_MEDIA
> > 2011-03-05 03:15:15.885293 [DEBUG] switch_core_session.c:1116 Send signal
> > sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
> > 2011-03-05 03:15:15.885293 [DEBUG] switch_core_state_machine.c:320
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) Running State Change
> > CS_EXCHANGE_MEDIA
> > 2011-03-05 03:15:15.885293 [DEBUG] switch_core_state_machine.c:369
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) State EXCHANGE_MEDIA
> > 2011-03-05 03:15:15.885293 [DEBUG] mod_sofia.c:552 SOFIA EXCHANGE_MEDIA
> > 2011-03-05 03:15:15.886388 [DEBUG] switch_core_session.c:771 Send signal
> > sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
> > 2011-03-05 03:15:15.886388 [DEBUG] switch_core_session.c:771 Send signal
> > skypopen/interface1 [BREAK]
> > 2011-03-05 03:15:15.886388 [DEBUG] mod_skypopen.c:727
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PREANSW,IDLE]
> > skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
> > 2011-03-05 03:15:15.906103 [DEBUG] switch_ivr_bridge.c:500
> > skypopen/interface1 ending bridge by request from read function
> > 2011-03-05 03:15:15.906103 [DEBUG] switch_ivr_bridge.c:581 BRIDGE THREAD
> > DONE [skypopen/interface1]
> > 2011-03-05 03:15:15.906103 [DEBUG] switch_ivr_bridge.c:601 Send signal
> > sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
> > 2011-03-05 03:15:15.909404 [DEBUG] switch_ivr_bridge.c:581 BRIDGE THREAD
> > DONE [sofia/internal/sip:1001 at 90.198.86.129:62954]
> > 2011-03-05 03:15:15.909404 [DEBUG] switch_ivr_bridge.c:601 Send signal
> > skypopen/interface1 [BREAK]
> > 2011-03-05 03:15:15.909404 [DEBUG] mod_skypopen.c:727
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PREANSW,IDLE]
> > skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
> > 2011-03-05 03:15:15.909404 [DEBUG] switch_channel.c:2545 (sofia/internal/
> > sip:1001 at 90.198.86.129:62954) Callstate Change ACTIVE -> HANGUP
> > 2011-03-05 03:15:15.909404 [NOTICE] switch_ivr_bridge.c:653 Hangup
> > sofia/internal/sip:1001 at 90.198.86.129:62954 [CS_EXCHANGE_MEDIA]
> > [NORMAL_CLEARING]
> > 2011-03-05 03:15:15.909404 [DEBUG] switch_channel.c:2561 Send signal
> > sofia/internal/sip:1001 at 90.198.86.129:62954 [KILL]
> > 2011-03-05 03:15:15.909404 [DEBUG] switch_core_session.c:1116 Send signal
> > sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
> > 2011-03-05 03:15:15.909404 [DEBUG] switch_core_state_machine.c:369
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) State EXCHANGE_MEDIA going
> to
> > sleep
> > 2011-03-05 03:15:15.909404 [DEBUG] switch_core_state_machine.c:320
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) Running State Change
> CS_HANGUP
> > 2011-03-05 03:15:15.910537 [DEBUG] switch_core_state_machine.c:560
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) State HANGUP
> > 2011-03-05 03:15:15.910537 [DEBUG] mod_sofia.c:457 Channel
> sofia/internal/
> > sip:1001 at 90.198.86.129:62954 hanging up, cause: NORMAL_CLEARING
> > 2011-03-05 03:15:15.910537 [DEBUG] switch_ivr_bridge.c:1305
> sofia/internal/
> > sip:1001 at 90.198.86.129:62954 skip receive message [UNBRIDGE] (channel is
> > hungup already)
> > 2011-03-05 03:15:15.910537 [DEBUG] mod_skypopen.c:1201
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PREANSW,IDLE]
> > MSG_ID=5
> > 2011-03-05 03:15:15.910537 [DEBUG] switch_core_session.c:709 Send signal
> > skypopen/interface1 [BREAK]
> > 2011-03-05 03:15:15.910537 [DEBUG] mod_skypopen.c:727
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][PREANSW,IDLE]
> > skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
> > 2011-03-05 03:15:15.910537 [DEBUG] mod_skypopen.c:1201
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1201 ][interface1     ][PREANSW,IDLE]
> > MSG_ID=27
> > 2011-03-05 03:15:15.910537 [NOTICE] switch_core_state_machine.c:189
> > skypopen/interface1 has executed the last dialplan instruction, hanging
> up.
> > 2011-03-05 03:15:15.910537 [DEBUG] switch_channel.c:2545
> > (skypopen/interface1) Callstate Change ACTIVE -> HANGUP
> > 2011-03-05 03:15:15.910537 [NOTICE] switch_core_state_machine.c:191
> Hangup
> > skypopen/interface1 [CS_EXECUTE] [NORMAL_CLEARING]
> > 2011-03-05 03:15:15.910537 [DEBUG] switch_channel.c:2561 Send signal
> > skypopen/interface1 [KILL]
> > 2011-03-05 03:15:15.910537 [DEBUG] mod_skypopen.c:700
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  700  ][interface1     ][PREANSW,IDLE]
> > skypopen/interface1 CHANNEL got SWITCH_SIG_KILL
> > 2011-03-05 03:15:15.910537 [DEBUG] switch_core_session.c:1116 Send signal
> > skypopen/interface1 [BREAK]
> > 2011-03-05 03:15:15.910537 [DEBUG] mod_skypopen.c:727
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][HANG_RQ,IDLE]
> > skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
> > 2011-03-05 03:15:15.910537 [DEBUG] switch_core_state_machine.c:366
> > (skypopen/interface1) State EXECUTE going to sleep
> > 2011-03-05 03:15:15.910537 [DEBUG] mod_sofia.c:500 Sending BYE to
> > sofia/internal/sip:1001 at 90.198.86.129:62954
> > 2011-03-05 03:15:15.910537 [DEBUG] switch_core_state_machine.c:320
> > (skypopen/interface1) Running State Change CS_HANGUP
> > 2011-03-05 03:15:15.910537 [DEBUG] switch_core_state_machine.c:560
> > (skypopen/interface1) State HANGUP
> > 2011-03-05 03:15:15.910537 [DEBUG] mod_skypopen.c:623
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  623  ][interface1     ][HANG_RQ,IDLE]
> > hanging up skype call: 142
> > 2011-03-05 03:15:15.911571 [DEBUG] skypopen_protocol.c:1619
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][HANG_RQ,IDLE]
> > SENDING: |||ALTER CALL 142 END HANGUP||||
> > 2011-03-05 03:15:15.911571 [DEBUG] skypopen_protocol.c:1619
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][HANG_RQ,IDLE]
> > SENDING: |||ALTER CALL 142 HANGUP||||
> > 2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:629
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  629  ][interface1     ][HANG_RQ,IDLE]
> > interface1 CHANNEL HANGUP
> > 2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:648
> (skypopen/interface1)
> > State Change CS_HANGUP -> CS_DESTROY
> > 2011-03-05 03:15:15.911571 [DEBUG] switch_core_session.c:1116 Send signal
> > skypopen/interface1 [BREAK]
> > 2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:727
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  727  ][interface1     ][DOWN,IDLE]
> > skypopen/interface1 CHANNEL got SWITCH_SIG_BREAK
> > 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:560
> > (skypopen/interface1) State HANGUP going to sleep
> > 2011-03-05 03:15:15.911571 [DEBUG] switch_core_session.c:1288 Session 24
> > (skypopen/interface1) Locked, Waiting on external entities
> > 2011-03-05 03:15:15.911571 [NOTICE] switch_core_session.c:1306 Session 24
> > (skypopen/interface1) Ended
> > 2011-03-05 03:15:15.911571 [NOTICE] switch_core_session.c:1308 Close
> Channel
> > skypopen/interface1 [CS_DESTROY]
> > 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:46
> > sofia/internal/sip:1001 at 90.198.86.129:62954 Standard HANGUP, cause:
> > NORMAL_CLEARING
> > 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:560
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) State HANGUP going to
> sleep
> > 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:351
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) State Change CS_HANGUP ->
> > CS_REPORTING
> > 2011-03-05 03:15:15.911571 [DEBUG] switch_core_session.c:1116 Send signal
> > sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
> > 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:320
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) Running State Change
> > CS_REPORTING
> > 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:620
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) State REPORTING
> > 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:53
> > sofia/internal/sip:1001 at 90.198.86.129:62954 Standard REPORTING, cause:
> > NORMAL_CLEARING
> > 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:620
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) State REPORTING going to
> sleep
> > 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:449
> > (skypopen/interface1) Callstate Change HANGUP -> DOWN
> > 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:452
> > (skypopen/interface1) Running State Change CS_DESTROY
> > 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:462
> > (skypopen/interface1) State DESTROY
> > 2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:479
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  479  ][interface1     ][DOWN,IDLE]
> > interface1 CHANNEL DESTROY c958d9be-46d6-11e0-bb71-d9b9e6428344
> > 2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:493
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  493  ][interface1     ][DOWN,IDLE] audio
> > tcp threads to DIE
> > 2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:503
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  503  ][interface1     ][DOWN,IDLE] audio
> > tcp srv thread DEAD 0
> > 2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:513
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  513  ][interface1     ][DOWN,IDLE] audio
> > tcp cli thread DEAD 0
> > 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:345
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) State Change CS_REPORTING
> ->
> > CS_DESTROY
> > 2011-03-05 03:15:15.911571 [DEBUG] switch_core_session.c:1116 Send signal
> > sofia/internal/sip:1001 at 90.198.86.129:62954 [BREAK]
> > 2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:524
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  524  ][interface1     ][DOWN,IDLE]
> codecs
> > DOWN
> > 2011-03-05 03:15:15.911571 [DEBUG] switch_core_session.c:1288 Session 25
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) Locked, Waiting on
> external
> > entities
> > 2011-03-05 03:15:15.911571 [DEBUG] mod_skypopen.c:579
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  579  ][interface1     ][IDLE,IDLE]
> CHANNEL
> > DESTROYED c958d9be-46d6-11e0-bb71-d9b9e6428344
> > 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:60
> > skypopen/interface1 Standard DESTROY
> > 2011-03-05 03:15:15.911571 [NOTICE] switch_core_session.c:1306 Session 25
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) Ended
> > 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:462
> > (skypopen/interface1) State DESTROY going to sleep
> > 2011-03-05 03:15:15.911571 [NOTICE] switch_core_session.c:1308 Close
> Channel
> > sofia/internal/sip:1001 at 90.198.86.129:62954 [CS_DESTROY]
> > 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:449
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) Callstate Change HANGUP ->
> > DOWN
> > 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:452
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) Running State Change
> > CS_DESTROY
> > 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:462
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) State DESTROY
> > 2011-03-05 03:15:15.911571 [DEBUG] mod_sofia.c:362 sofia/internal/
> > sip:1001 at 90.198.86.129:62954 SOFIA DESTROY
> > 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:60
> > sofia/internal/sip:1001 at 90.198.86.129:62954 Standard DESTROY
> > 2011-03-05 03:15:15.911571 [DEBUG] switch_core_state_machine.c:462
> > (sofia/internal/sip:1001 at 90.198.86.129:62954) State DESTROY going to
> sleep
> > 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:173
> > [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE]
> READING:
> > |||CALL 142 STATUS INPROGRESS|||
> > 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:719
> > [32b8f10|9350fb9] [DEBUG_SKYPE  719  ][interface1     ][IDLE,IDLE] no
> > tech_pvt->session_uuid_str
> > 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:725
> > [32b8f10|9350fb9] [DEBUG_SKYPE  725  ][interface1     ][IDLE,IDLE]
> > skype_call: 142 is now active
> > 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:732
> > [32b8f10|9350fb9] [DEBUG_SKYPE  732  ][interface1     ][UP,INPROGRS]
> START
> > start_audio_threads
> > 2011-03-05 03:15:16.149772 [DEBUG] mod_skypopen.c:2203
> > [32b8f10|9350fb9] [DEBUG_SKYPE  2203 ][interface1     ][UP,INPROGRS]
> started
> > tcp_srv_thread thread.
> > 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:89
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  89   ][interface1     ][UP,INPROGRS]
> > Binded! *which_port=32783, tech_pvt->tcp_cli_port=32782,
> > tech_pvt->tcp_srv_port=32783
> > 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:94
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  94   ][interface1     ][UP,INPROGRS] 1
> > SO_RCVBUF is 87380, size is 4
> > 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:98
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  98   ][interface1     ][UP,INPROGRS] 1
> > SO_SNDBUF is 16384, size is 4
> > 2011-03-05 03:15:16.149772 [DEBUG] mod_skypopen.c:2219
> > [32b8f10|9350fb9] [DEBUG_SKYPE  2219 ][interface1     ][UP,INPROGRS]
> started
> > tcp_cli_thread thread.
> > 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:115
> > [32b8f10|9350fb9] [DEBUG_SKYPE  115  ][interface1     ][UP,INPROGRS] 2
> > SO_RCVBUF is 87380, size is 4
> > 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:130
> > [32b8f10|9350fb9] [DEBUG_SKYPE  130  ][interface1     ][UP,INPROGRS] 2
> > SO_SNDBUF is 16384, size is 4
> > 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:89
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  89   ][interface1     ][UP,INPROGRS]
> > Binded! *which_port=32784, tech_pvt->tcp_cli_port=32784,
> > tech_pvt->tcp_srv_port=32783
> > 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:134
> > [32b8f10|9350fb9] [DEBUG_SKYPE  134  ][interface1     ][UP,INPROGRS]
> > TCP_NODELAY is 0
> > 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:94
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  94   ][interface1     ][UP,INPROGRS] 1
> > SO_RCVBUF is 87380, size is 4
> > 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:141
> > [32b8f10|9350fb9] [DEBUG_SKYPE  141  ][interface1     ][UP,INPROGRS]
> > TCP_NODELAY is 0
> > 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:98
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  98   ][interface1     ][UP,INPROGRS] 1
> > SO_SNDBUF is 16384, size is 4
> > 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:833
> > [32b8f10|9350fb9] [DEBUG_SKYPE  833  ][interface1     ][UP,INPROGRS]
> started
> > tcp_srv_thread thread.
> > 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:115
> > [32b8f10|9350fb9] [DEBUG_SKYPE  115  ][interface1     ][UP,INPROGRS] 2
> > SO_RCVBUF is 87380, size is 4
> > 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:130
> > [32b8f10|9350fb9] [DEBUG_SKYPE  130  ][interface1     ][UP,INPROGRS] 2
> > SO_SNDBUF is 16384, size is 4
> > 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:134
> > [32b8f10|9350fb9] [DEBUG_SKYPE  134  ][interface1     ][UP,INPROGRS]
> > TCP_NODELAY is 0
> > 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:141
> > [32b8f10|9350fb9] [DEBUG_SKYPE  141  ][interface1     ][UP,INPROGRS]
> > TCP_NODELAY is 0
> > 2011-03-05 03:15:16.149772 [DEBUG] skypopen_protocol.c:997
> > [32b8f10|9350fb9] [DEBUG_SKYPE  997  ][interface1     ][UP,INPROGRS]
> started
> > tcp_cli_thread thread.
> > 2011-03-05 03:15:16.250297 [DEBUG] skypopen_protocol.c:1619
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][UP,INPROGRS]
> > SENDING: |||ALTER CALL 142 SET_INPUT PORT="32784"||||
> > 2011-03-05 03:15:16.250297 [DEBUG] skypopen_protocol.c:1619
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][UP,INPROGRS]
> > SENDING: |||#output ALTER CALL 142 SET_OUTPUT PORT="32783"||||
> > 2011-03-05 03:15:16.250297 [WARNING] skypopen_protocol.c:2036
> >  [32b8f10|9350fb9] [WARNINGA     2036 ][interface1     ][UP,INPROGRS] no
> > tech_pvt->session_uuid_str after INPROGRESS, let's hangup
> > 2011-03-05 03:15:16.250297 [DEBUG] skypopen_protocol.c:1619
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][UP,INPROGRS]
> > SENDING: |||ALTER CALL 142 HANGUP||||
> > 2011-03-05 03:15:16.250297 [DEBUG] skypopen_protocol.c:173
> > [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][UP,INPROGRS]
> > READING: |||CALL 142 VIDEO_SEND_STATUS AVAILABLE|||
> > 2011-03-05 03:15:16.290473 [DEBUG] skypopen_protocol.c:173
> > [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][UP,INPROGRS]
> > READING: |||CALL 142 VIDEO_STATUS VIDEO_BOTH_ENABLED|||
> > 2011-03-05 03:15:16.363839 [DEBUG] skypopen_protocol.c:173
> > [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][UP,INPROGRS]
> > READING: |||CALL 142 DURATION 1|||
> > 2011-03-05 03:15:16.432605 [DEBUG] skypopen_protocol.c:173
> > [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][UP,INPROGRS]
> > READING: |||ALTER CALL 142 END HANGUP|||
> > 2011-03-05 03:15:16.494506 [DEBUG] skypopen_protocol.c:173
> > [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][UP,INPROGRS]
> > READING: |||ERROR 559 CALL: Action failed|||
> > 2011-03-05 03:15:16.495584 [DEBUG] skypopen_protocol.c:228
> > [32b8f10|9350fb9] [DEBUG_SKYPE  228  ][interface1     ][UP,INPROGRS]
> Skype
> > got ERROR about a failed action (probably TRYING to HANGUP A CALL), no
> > problem: |||ERROR 559 CALL: Action failed|||
> > 2011-03-05 03:15:16.710271 [DEBUG] skypopen_protocol.c:173
> > [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][UP,INPROGRS]
> > READING: |||ERROR 589 ALTER CALL: unable to alter input/output|||
> > 2011-03-05 03:15:16.711346 [DEBUG] skypopen_protocol.c:247
> > [32b8f10|9350fb9] [DEBUG_SKYPE  247  ][interface1     ][UP,INPROGRS]
> Skype
> > client was not able to correctly manage tcp audio sockets, probably got a
> > local or remote hangup: |||ERROR 589 ALTER CALL: unable to alter
> > input/output|||
> > 2011-03-05 03:15:16.711346 [DEBUG] skypopen_protocol.c:1619
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][UP,INPROGRS]
> > SENDING: |||ALTER CALL 142 HANGUP||||
> > 2011-03-05 03:15:16.711346 [DEBUG] skypopen_protocol.c:1619
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  1619 ][interface1     ][UP,INPROGRS]
> > SENDING: |||ALTER CALL 142 END HANGUP||||
> > 2011-03-05 03:15:16.711346 [DEBUG] mod_skypopen.c:1413
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1413 ][interface1     ][DOWN,INPROGRS]
> skype
> > call ended
> > 2011-03-05 03:15:16.711346 [DEBUG] mod_skypopen.c:1433
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1433 ][interface1     ][DOWN,INPROGRS] no
> > session
> > 2011-03-05 03:15:16.711346 [DEBUG] mod_skypopen.c:1436
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1436 ][interface1     ][DOWN,INPROGRS]
> audio
> > tcp threads to DIE
> > 2011-03-05 03:15:16.712434 [DEBUG] skypopen_protocol.c:960
> > [32b8f10|9350fb9] [DEBUG_SKYPE  960  ][interface1     ][DOWN,INPROGRS]
> > incoming audio (read) server (I am it) EXITING
> > 2011-03-05 03:15:16.715692 [DEBUG] mod_skypopen.c:1446
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1446 ][interface1     ][DOWN,INPROGRS]
> audio
> > tcp srv thread DEAD 1
> > 2011-03-05 03:15:16.750763 [DEBUG] skypopen_protocol.c:1112
> >  [32b8f10|9350fb9] [DEBUG_SKYPE  1112 ][interface1     ][DOWN,INPROGRS]
> > outbound audio server (I am it) EXITING
> > 2011-03-05 03:15:16.751849 [DEBUG] mod_skypopen.c:1456
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1456 ][interface1     ][DOWN,INPROGRS]
> audio
> > tcp cli thread DEAD 7
> > 2011-03-05 03:15:16.751849 [DEBUG] skypopen_protocol.c:173
> > [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE]
> READING:
> > |||#output ERROR 589 ALTER CALL: unable to alter input/output|||
> > 2011-03-05 03:15:16.779660 [DEBUG] skypopen_protocol.c:173
> > [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE]
> READING:
> > |||ERROR 559 CALL: Action failed|||
> > 2011-03-05 03:15:16.780762 [DEBUG] skypopen_protocol.c:228
> > [32b8f10|9350fb9] [DEBUG_SKYPE  228  ][interface1     ][IDLE,IDLE] Skype
> got
> > ERROR about a failed action (probably TRYING to HANGUP A CALL), no
> problem:
> > |||ERROR 559 CALL: Action failed|||
> > 2011-03-05 03:15:16.828877 [DEBUG] skypopen_protocol.c:173
> > [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE]
> READING:
> > |||ERROR 559 CALL: Action failed|||
> > 2011-03-05 03:15:16.828877 [DEBUG] skypopen_protocol.c:228
> > [32b8f10|9350fb9] [DEBUG_SKYPE  228  ][interface1     ][IDLE,IDLE] Skype
> got
> > ERROR about a failed action (probably TRYING to HANGUP A CALL), no
> problem:
> > |||ERROR 559 CALL: Action failed|||
> > 2011-03-05 03:15:16.869183 [DEBUG] skypopen_protocol.c:173
> > [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE]
> READING:
> > |||ERROR 559 CALL: Action failed|||
> > 2011-03-05 03:15:16.869183 [DEBUG] skypopen_protocol.c:228
> > [32b8f10|9350fb9] [DEBUG_SKYPE  228  ][interface1     ][IDLE,IDLE] Skype
> got
> > ERROR about a failed action (probably TRYING to HANGUP A CALL), no
> problem:
> > |||ERROR 559 CALL: Action failed|||
> > 2011-03-05 03:15:16.911668 [DEBUG] skypopen_protocol.c:173
> > [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE]
> READING:
> > |||CALL 142 STATUS FINISHED|||
> > 2011-03-05 03:15:16.911668 [DEBUG] skypopen_protocol.c:661
> > [32b8f10|9350fb9] [DEBUG_SKYPE  661  ][interface1     ][IDLE,IDLE]
> > skype_call 142 is MY call, now I'm going DOWN
> > 2011-03-05 03:15:16.911668 [DEBUG] mod_skypopen.c:1413
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1413 ][interface1     ][IDLE,IDLE] skype
> > call ended
> > 2011-03-05 03:15:16.911668 [DEBUG] mod_skypopen.c:1433
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1433 ][interface1     ][IDLE,IDLE] no
> > session
> > 2011-03-05 03:15:16.911668 [DEBUG] mod_skypopen.c:1436
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1436 ][interface1     ][DOWN,IDLE] audio
> tcp
> > threads to DIE
> > 2011-03-05 03:15:16.911668 [DEBUG] mod_skypopen.c:1446
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1446 ][interface1     ][DOWN,IDLE] audio
> tcp
> > srv thread DEAD 0
> > 2011-03-05 03:15:16.911668 [DEBUG] mod_skypopen.c:1456
> > [32b8f10|9350fb9] [DEBUG_SKYPE  1456 ][interface1     ][DOWN,IDLE] audio
> tcp
> > cli thread DEAD 0
> >
>
> --
> Sent from my mobile device
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From renjian at gmail.com  Sat Mar  5 18:14:44 2011
From: renjian at gmail.com (Jian Ren)
Date: Sat, 5 Mar 2011 10:14:44 -0500
Subject: [Freeswitch-users] How could I forward the SIP incoming call to
 one of the available extensions?
In-Reply-To: <AANLkTikEwOW2KJz3z-of86wb4a8w8=SxoPCo0E3avVqY@mail.gmail.com>
References: <AANLkTi=A-NSss214Qe=aiofh-qhofDKZMFBx3v8=DgDi@mail.gmail.com>
	<AANLkTikEwOW2KJz3z-of86wb4a8w8=SxoPCo0E3avVqY@mail.gmail.com>
Message-ID: <AANLkTik5DuoUavfyhfg69wnp_72zqiz1FKXSo9Rcp8zU@mail.gmail.com>

Yes, my "extension" means the physical ATA connected to phones. I have 3 at
home all registered to the FS server. I want all of them ring, or one by one
for incoming calls from an external SIP connection. Then when two of them
are being used to call outside I could still  answer the incoming call.
Thanks!
Jian

On Sat, Mar 5, 2011 at 2:19 AM, Christian Benke <benkokakao at gmail.com>wrote:

> > Right now I only have one extension in the SIP configuration as the
> > destination. Could I put a list of extensions then FS forwards the call
> to
> > the first available one?
>
> I'm not sure if i understand you exactly. By "extension" do you mean a
> SIP-endpoint like a phone or a gateway? Take a look at this simple
> example, is this good enough for your situation?:
> http://wiki.freeswitch.org/wiki/Extension_Status_Example
>
> If not, please let us now in more detail what you want to achieve(Is
> this a Call Center solution? Take a look at
> http://wiki.freeswitch.org/wiki/Mod_fifo)
>
> Best regards,
> Christian
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From andreas at tuerpe-net.de  Sat Mar  5 12:12:17 2011
From: andreas at tuerpe-net.de (Andreas Tuerpe)
Date: Sat, 05 Mar 2011 10:12:17 +0100
Subject: [Freeswitch-users] route inbound - based on sip account (solved)
In-Reply-To: <AANLkTi=zU0YBsYTwJcf3N1JaKtxxUyygAj7JyRDOGk31@mail.gmail.com>
References: <4D6D45F0.5060205@tuerpe-net.de>	<AANLkTimmuZSR5pPLDd9B6cXjgE9jQ7TDHibWQ_T4Fjx-@mail.gmail.com>	<4D6EACBC.7040509@tuerpe-net.de>
	<AANLkTi=zU0YBsYTwJcf3N1JaKtxxUyygAj7JyRDOGk31@mail.gmail.com>
Message-ID: <4D71FE71.60806@tuerpe-net.de>

Hallo Michael,
thank you very much!

This is the solution in inbound dialplan:

<extension name="4932xxxxxx782">
   <condition field="${sip_gateway}" expression="sip69250$">
       <action application="set" data="domain_name=$${domain}"/>
       <action application="transfer" data="1004 XML default"/>
   </condition>
</extension>

<extension name="4932xxxxxx781">
   <condition field="${sip_gateway}" expression="sip68962$">
       <action application="set" data="domain_name=$${domain}"/>
       <action application="transfer" data="1001 XML default"/>
   </condition>
</extension>

Now I'm hapy with FreeSWITCH...

Many thanks to all folks.
   Andreas


Am 03.03.2011 19:25, schrieb Michael Collins:
> You've cut out a fair amount of data, but I think I see at least one issue:
> 
> <condition
> field="${variable_sip_gateway}" expression="sofia/gateway/sip69250">
> 
> Do you have a channel variable that is named "variable_sip_gateway"?
> More likely you have a channel variable named "sip_gateway". Try
> rewriting the above line like this:
> 
> <condition field="${sip_gateway}" expression="sip69250$">
> 
> The only thing I'm not sure about is whether or not "sip_gateway" is
> populated at the time the dialplan is parsed. Without seeing the info
> dump I can't offer any other suggestions.
> 
> -MC
> 
> On Wed, Mar 2, 2011 at 12:46 PM, Andreas Tuerpe <andreas at tuerpe-net.de
> <mailto:andreas at tuerpe-net.de>> wrote:
> 
>     Hallo Michael, Johannes and Meftah
> 
>     here is...
>     -------------------------------
>     --- start trace log extract ---
> 
>     INVITE sip:gw+sip69250 at 188.246.xxx.xxx:5080;transport=udp SIP/2.0
>     Via: SIP/2.0/UDP 82.139.xxx.x:5060;branch=z9hG4bK2fa2cd41;rport
>     From: "+491758301234" <sip:+491758301234 at 82.139.xxx.x>;tag=as20cd750d
>     To: <sip:gw+sip69250 at 188.246.xxx.xxx:5080;transport=udp>
>     Contact: <sip:+491758301234 at 82.139.xxx.x>
>     Call-ID: 2637b63a2411697f7af5f089732a68eb at 82.139.xxx.x
>     CSeq: 102 INVITE
>     User-Agent: PTY SIPPort
> 
>     ...
> 
>     # the only advice to the inbound channel
>     # -> 'sip69250' is the 2. gateway name
>     [INFO] mod_dialplan_xml.c:252 Processing +491758301234->sip69250 in
>     context public
> 
> 
>     Dialplan: sofia/external/+491758301234 at 82.139.xxx.x parsing
>     [public->unloop] continue=false
>     ...
>     # 'public_extensions' is the 1. extension name in public context
>     Dialplan: sofia/external/+491758302577 at 82.139.223.1
>     <mailto:491758302577 at 82.139.223.1> parsing
>     [public->public_extensions] continue=false
>     # my test of condition assignment with ${variable_sip_gateway} fails
>     Dialplan: ... ... Regex (FAIL) [public_extensions]
>     ${variable_sip_gateway}() =~ /sofia/gateway/sip69250/ break=on-false
>     ...
>     # '4932229982781' is the 2. extension name in public context
>     Dialplan: sofia/external/+491758301234 at 82.139.xxx.x parsing
>     [public->4932229982781] continue=false
>     # empty condition assignment run's without any errors
>     Dialplan: sofia/external/+491758301234 at 82.139.xxx.x Regex (PASS)
>     [4932229982781] () =~ // break=on-false
>     Dialplan: sofia/external/+491758301234 at 82.139.xxx.x Action transfer(1001
>     XML default)
>     ...
>     #all others run's without any errors
>     ...
>     ---- end trace log extract ----
>     -------------------------------
> 
> 
>     # code of first extension name
>     <extension name="public_extensions">
>        <condition field="${variable_sip_gateway}"
>     expression="sofia/gateway/sip69250">
>            <action application="transfer" data="1004 XML default"/>
>        </condition>
>     </extension>
> 
>     # code of second extension name
>     <extension name="4932229982781">
>        <condition field="" expression="">
>            <action application="transfer" data="1001 XML default"/>
>        </condition>
>     </extension>
> 
> 
>     I'm unsure with syntax of condition variable esp. expression="?"
> 
>     Thanks
>        Andreas
> 
> 
> 
>     Am 02.03.2011 00:07, schrieb Michael Collins:
>     > The FS on pfSense is pretty old, but if all you are working on is a
>     > simple routing issue your best bet is to add the "info" app in the
>     > public context. Somewhere near the top of public.xml just add this:
>     >
>     > <extension name="quick info dump" continue="true">
>     > <condition>
>     > <action application="info"/>
>     > </condition>
>     > </extension>
>     >
>     > Save that, press F6 (or do reloadxml) and then make a test inbound
>     call.
>     > Watch the console - you'll see a TON of information. Look through the
>     > pieces of data that are displayed. You should be able to find
>     something
>     > to key off of. Once you've done that then go read up on creating your
>     > dialplan here:
>     >
>     > http://wiki.freeswitch.org/wiki/Dialplan_XML#Condition
>     >
>     > In fact, that whole page is important in understanding the XML
>     dialplan.
>     > I would read it more than once.
>     >
>     > -MC
>     >
>     > On Tue, Mar 1, 2011 at 11:16 AM, Andreas Tuerpe
>     <andreas at tuerpe-net.de <mailto:andreas at tuerpe-net.de>
>     > <mailto:andreas at tuerpe-net.de <mailto:andreas at tuerpe-net.de>>> wrote:
>     >
>     >     Hallo FreeSWITCH Users,
>     >
>     >     I use FS V.0.9.6 on pfSense
>     >     see on -> [http://doc.pfsense.org/index.php/FreeSWITCH]
>     >
>     >     Symptom:
>     >     German ISP "Portunity" forward inbound calls without any
>     >     destination_number.
>     >
>     >
>     >     ISP solution tip:
>     >     FS to register over a second sip account to the provider twice.
>     >     Any account has a separate number.
>     >     Based on the channel over which the call come in, I have to
>     decide which
>     >     number is called.
>     >
>     >     So I need help, which condition assignment must use - howto ???
>     >     - which fields can I use?
>     >     - which syntax I have to use?
>     >
>     >
>     >     <!--  here my intention ... ???
>     >
>     >     <extension name="test_did">
>     >     <condition field="channel over which the call come in"
>     >     expression="sip-account-1">
>     >     <action application="set" data="called_number=111111"/>
>     >     <action application="transfer" data="1001 XML default"/>
>     >     <condition field="channel over which the call come in"
>     >     expression="sip-account-2">
>     >     <action application="set" data="called_number=222222"/>
>     >     <action application="transfer" data="1002 XML default"/>
>     >     </condition>
>     >     </extension>
>     >
>     >     ??? -->
>     >
>     >     thanks in advance
>     >        tuerpean
>     >
>     >
>     >     _______________________________________________
>     >     FreeSWITCH-users mailing list
>     >     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >    
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >     http://www.freeswitch.org
>     >
>     >
>     >
>     >
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From philtrick123 at hotmail.com  Sat Mar  5 15:11:56 2011
From: philtrick123 at hotmail.com (Phil T)
Date: Sat, 5 Mar 2011 12:11:56 +0000
Subject: [Freeswitch-users] Standard Functionality of Freeswitch as
 Conference Server? Automatic Listen only conference.
Message-ID: <COL124-W233179629C8AC5F32450BF85C50@phx.gbl>


Hi,

 

I'm trying to setup a conference server for a local charity but I wanted to check on the standard conference functionality of Freeswitch before I started.

The requirements are quite simple- Conference call only (no extra pbx features) for up to 10 people (via Public SIP trunks), but the critical part is there will be one presenter & 9 people listening only. So ideally I want to have two pin codes- one for use by the presenter, and the other pin code for listen only attendees. The listen only people should be automatically muted when they join, and obviously the presenter should have full duplex audio. There might be the requirement to allow individual listen only attendee to opt in/out to speak on the call (via a feature code i.e. *6).

 

This type of feature is common among commercial conference solutions, is it available on Freeswitch via standard scripting without having to have a custom programmed module?

 

If the answer is yes- I'll get started on setting it up. Any tips greatly appreciated!

 

Finally- I have very little linux command line experience, so I'd like to do the xml script setup/ testing on a Windows 7 pc, then move over to Linux if the performance is required. Is that a sensible approach? Is there a GUI to help with the Freeswitch setup?

 

 

Thanks Phil
 		 	   		  
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From david.villasmil.work at gmail.com  Sat Mar  5 19:43:25 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 5 Mar 2011 17:43:25 +0100
Subject: [Freeswitch-users] Pastebin
Message-ID: <AANLkTimWhdnqbf4Oa6jVCTg4SNauv-Sz1rFQ4cifBUfx@mail.gmail.com>

Hello people,

Why is it that even though I'm logged into freeswitch.org, when I click on
the pastebin link, it asks for user/pass and doesn't recognize my
credentials? it's very annoying.

Thanks

David
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From steveayre at gmail.com  Sat Mar  5 19:48:20 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 5 Mar 2011 16:48:20 +0000
Subject: [Freeswitch-users] Pastebin
In-Reply-To: <AANLkTimWhdnqbf4Oa6jVCTg4SNauv-Sz1rFQ4cifBUfx@mail.gmail.com>
References: <AANLkTimWhdnqbf4Oa6jVCTg4SNauv-Sz1rFQ4cifBUfx@mail.gmail.com>
Message-ID: <138CCB3D-E1A2-4843-BF07-DC8C3A5D6BCD@gmail.com>

They don't use the same details.



Steve on iPhone

On 5 Mar 2011, at 16:43, David Villasmil <david.villasmil.work at gmail.com> wrote:

> Hello people,
> 
> Why is it that even though I'm logged into freeswitch.org, when I click on the pastebin link, it asks for user/pass and doesn't recognize my credentials? it's very annoying.
> 
> Thanks
> 
> David
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From steveayre at gmail.com  Sat Mar  5 19:49:14 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 5 Mar 2011 16:49:14 +0000
Subject: [Freeswitch-users] Pastebin
In-Reply-To: <AANLkTimWhdnqbf4Oa6jVCTg4SNauv-Sz1rFQ4cifBUfx@mail.gmail.com>
References: <AANLkTimWhdnqbf4Oa6jVCTg4SNauv-Sz1rFQ4cifBUfx@mail.gmail.com>
Message-ID: <8132EFE9-A73E-4FAF-862D-A079C4D6387C@gmail.com>

Sorry, knocked send early.

Read the pastebin password prompt carefully....

Steve on iPhone

On 5 Mar 2011, at 16:43, David Villasmil <david.villasmil.work at gmail.com> wrote:

> Hello people,
> 
> Why is it that even though I'm logged into freeswitch.org, when I click on the pastebin link, it asks for user/pass and doesn't recognize my credentials? it's very annoying.
> 
> Thanks
> 
> David
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From Nabble at slickdeals.endjunk.com  Sat Mar  5 19:56:59 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sat, 5 Mar 2011 08:56:59 -0800 (PST)
Subject: [Freeswitch-users] Pastebin
In-Reply-To: <AANLkTimWhdnqbf4Oa6jVCTg4SNauv-Sz1rFQ4cifBUfx@mail.gmail.com>
References: <AANLkTimWhdnqbf4Oa6jVCTg4SNauv-Sz1rFQ4cifBUfx@mail.gmail.com>
Message-ID: <1299344219611-6092170.post@n2.nabble.com>


David Villasmil wrote:
> Why is it that even though I'm logged into freeswitch.org, when I click on
> the pastebin link, it asks for user/pass and doesn't recognize my
> credentials? it's very annoying.
The user/pass requirement is to ward out any spam bots. Without the
user/pass, it will become an annoyance, if not a nightmare, to the mods to
clean out the spam posts by spam bots.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Pastebin-tp6092153p6092170.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From hkalyoncu at gmail.com  Sat Mar  5 20:46:18 2011
From: hkalyoncu at gmail.com (hkalyoncu)
Date: Sat, 5 Mar 2011 09:46:18 -0800 (PST)
Subject: [Freeswitch-users] mod_python
In-Reply-To: <AANLkTikQ4NVFE2BLNYxiY05zFaYeq8TJjqbuZJKp3OgG@mail.gmail.com>
References: <1299244469418-6088430.post@n2.nabble.com>
	<AANLkTikQ4NVFE2BLNYxiY05zFaYeq8TJjqbuZJKp3OgG@mail.gmail.com>
Message-ID: <1299347178939-6092296.post@n2.nabble.com>

thanks for your reply
that documentation you pointed contains not much info for my case.

im not running a script from inside dialplan.
as i wrote before im trying to serve dynamic dialplans with mod_python

it seems that param1 has a type of some object
but i have absolutely no idea what it contains.
there is no documentation explaining those params.

could anyone help?
may the writer of the module give some hints?  


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-python-tp6088430p6092296.html
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From djbinter at gmail.com  Sat Mar  5 20:49:31 2011
From: djbinter at gmail.com (DJB International)
Date: Sat, 5 Mar 2011 09:49:31 -0800
Subject: [Freeswitch-users] Old hung calls
Message-ID: <AANLkTinumTOTJUNJmhazH-ZRTh=jH_412PHV2EG6Wsew@mail.gmail.com>

I am running FS git-deec244 2011-02-24 10-22-47 -0600.  I ran into a problem
the other day when our ISP was down badly for couple hours (fortunately
after midnight).   The problem was that after the internet went back up;
there were so many old hung calls before the internet got cut off in the
core.db (calls & channels table) database.

Here are my questions:
- What is the proper way to clean those calls up, or will FS clean those
calls out from database automatically?
- If I decided to restart FS which I know it will clean up those calls, will
those calls get written to mod_cdr_csv?

Here is my sip profile:
  <settings>
    <param name="debug" value="0"/>
    <param name="sip-trace" value="no"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="sip-port" value="5060"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="routing"/>
    <param name="dtmf-duration" value="2000"/>
    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="rtp-timer-name" value="soft"/>
    <param name="local-network-acl" value="localnet.auto"/>
    <param name="manage-presence" value="false"/>
    <param name="inbound-codec-negotiation" value="generous"/>
    <param name="nonce-ttl" value="60"/>
    <param name="auth-calls" value="$${internal_auth_calls}"/>
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="auto-nat"/>
    <param name="rtp-timeout-sec" value="300"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>
    <param name="auth-all-packets" value="false"/>
    <param name="disable-transfer" value="true"/>
    <param name="disable-register" value="true"/>
    <param name="inbound-late-negotiation" value="true"/>
    <param name="inbound-bypass-media" value="true"/>
    <param name="apply-inbound-acl" value="domains"/>
  </settings>

Thank you,
-djbinter
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From david.villasmil.work at gmail.com  Sat Mar  5 21:11:14 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 5 Mar 2011 19:11:14 +0100
Subject: [Freeswitch-users] Pastebin
In-Reply-To: <1299344219611-6092170.post@n2.nabble.com>
References: <AANLkTimWhdnqbf4Oa6jVCTg4SNauv-Sz1rFQ4cifBUfx@mail.gmail.com>
	<1299344219611-6092170.post@n2.nabble.com>
Message-ID: <AANLkTik60rneBM41RM++roemQ9_GJe2DznTCzOyLJaJw@mail.gmail.com>

Hello,

I read very carefully... I'm in.

The wording is a little confusing, btw.

Thanks!

David

On Sat, Mar 5, 2011 at 5:56 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
> David Villasmil wrote:
> > Why is it that even though I'm logged into freeswitch.org, when I click
> on
> > the pastebin link, it asks for user/pass and doesn't recognize my
> > credentials? it's very annoying.
> The user/pass requirement is to ward out any spam bots. Without the
> user/pass, it will become an annoyance, if not a nightmare, to the mods to
> clean out the spam posts by spam bots.
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Pastebin-tp6092153p6092170.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gilles.gerlinger at free.fr  Sat Mar  5 20:10:25 2011
From: gilles.gerlinger at free.fr (gigerlin)
Date: Sat, 5 Mar 2011 09:10:25 -0800 (PST)
Subject: [Freeswitch-users] writing to file in javascript
Message-ID: <1299345025947-6092204.post@n2.nabble.com>

Dear Freeswitch users,

I am developing some javascript applications that run within Freeswitch. I
can read without difficulty from files located on the same machine as
Freeswitch, but I can't write or create new files. The following code issues
an error when trying to open a file in "write mode".

var fd = new File("test.txt");
fd.open("write");
fd.write("Hello\n");
fd.close();

The error I got is:
2011-02-18 11:07:25.893733 [ERR] test.js:7  Error: File operation open
failed

I thought it was a sort of sandbox model issue. However I could generate a
file using the following system command:

session.execute("system", "echo Hello > test.txt");

The same error occurs when I tried to use the javascript function:

var tmp = fetchURLFile("http://myserver/test.html", "test.txt");

This creates an empty test.txt file.

I tried then to run the following system command:

session.execute("system", "wget -O test.txt http://myserver/test.html");

This as well creates an empty test.txt file. If I run "wget -O test.txt
http://myserver/test.html" directly from the system prompt this correctly
downloads and creates the test.txt file.

So my questions are: what did I miss? Is there some parameter to configure
that I haven't found?

Thank you in advance for your help,
Gilles Gerlinger

PS: Freeswitch is running over Ubuntu server 10.10 

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/writing-to-file-in-javascript-tp6092204p6092204.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Sat Mar  5 22:17:36 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 5 Mar 2011 19:17:36 +0000
Subject: [Freeswitch-users] Pastebin
In-Reply-To: <AANLkTik60rneBM41RM++roemQ9_GJe2DznTCzOyLJaJw@mail.gmail.com>
References: <AANLkTimWhdnqbf4Oa6jVCTg4SNauv-Sz1rFQ4cifBUfx@mail.gmail.com>
	<1299344219611-6092170.post@n2.nabble.com>
	<AANLkTik60rneBM41RM++roemQ9_GJe2DznTCzOyLJaJw@mail.gmail.com>
Message-ID: <AANLkTimwzc8E-47pJr+zdRrYHwgaQzCw8gtYJTXyRn2h@mail.gmail.com>

"the login and password is pastebin/freeswitch"

What's confusing there?

-Steve


On 5 March 2011 18:11, David Villasmil <david.villasmil.work at gmail.com> wrote:
> Hello,
> I read very carefully... I'm in.
> The wording is a little confusing, btw.
> Thanks!
> David
>
> On Sat, Mar 5, 2011 at 5:56 PM, mazilo <Nabble at slickdeals.endjunk.com>
> wrote:
>>
>> David Villasmil wrote:
>> > Why is it that even though I'm logged into freeswitch.org, when I click
>> > on
>> > the pastebin link, it asks for user/pass and doesn't recognize my
>> > credentials? it's very annoying.
>> The user/pass requirement is to ward out any spam bots. Without the
>> user/pass, it will become an annoyance, if not a nightmare, to the mods to
>> clean out the spam posts by spam bots.
>>
>> -----
>> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Pastebin-tp6092153p6092170.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From peter.olsson at visionutveckling.se  Sat Mar  5 22:19:45 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sat, 5 Mar 2011 20:19:45 +0100
Subject: [Freeswitch-users] writing to file in javascript
In-Reply-To: <1299345025947-6092204.post@n2.nabble.com>
References: <1299345025947-6092204.post@n2.nabble.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2C494E9@cooper>

Googled on this, and found this - does that solve the issue?

file.open("write,create", "text");

________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r gigerlin [gilles.gerlinger at free.fr]
Skickat: den 5 mars 2011 18:10
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] writing to file in javascript

Dear Freeswitch users,

I am developing some javascript applications that run within Freeswitch. I
can read without difficulty from files located on the same machine as
Freeswitch, but I can't write or create new files. The following code issues
an error when trying to open a file in "write mode".

var fd = new File("test.txt");
fd.open("write");
fd.write("Hello\n");
fd.close();

The error I got is:
2011-02-18 11:07:25.893733 [ERR] test.js:7  Error: File operation open
failed

I thought it was a sort of sandbox model issue. However I could generate a
file using the following system command:

session.execute("system", "echo Hello > test.txt");

The same error occurs when I tried to use the javascript function:

var tmp = fetchURLFile("http://myserver/test.html", "test.txt");

This creates an empty test.txt file.

I tried then to run the following system command:

session.execute("system", "wget -O test.txt http://myserver/test.html");

This as well creates an empty test.txt file. If I run "wget -O test.txt
http://myserver/test.html" directly from the system prompt this correctly
downloads and creates the test.txt file.

So my questions are: what did I miss? Is there some parameter to configure
that I haven't found?

Thank you in advance for your help,
Gilles Gerlinger

PS: Freeswitch is running over Ubuntu server 10.10

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/writing-to-file-in-javascript-tp6092204p6092204.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

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

!DSPAM:4d72869b32762024114202!



From peter.olsson at visionutveckling.se  Sat Mar  5 22:21:24 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sat, 5 Mar 2011 20:21:24 +0100
Subject: [Freeswitch-users] Old hung calls
In-Reply-To: <AANLkTinumTOTJUNJmhazH-ZRTh=jH_412PHV2EG6Wsew@mail.gmail.com>
References: <AANLkTinumTOTJUNJmhazH-ZRTh=jH_412PHV2EG6Wsew@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EA@cooper>

Since RTP timeout is set to 5 minutes (300 sec) FS should clear the call after this (and of course the record in core.db as well).

You probably need to pastebin some logs with examples when this happens, something seems strange here...

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r DJB International [djbinter at gmail.com]
Skickat: den 5 mars 2011 18:49
Till: FREESWITCH-USERS MAILING LIST
?mne: [Freeswitch-users] Old hung calls

I am running FS git-deec244 2011-02-24 10-22-47 -0600.  I ran into a problem the other day when our ISP was down badly for couple hours (fortunately after midnight).   The problem was that after the internet went back up; there were so many old hung calls before the internet got cut off in the core.db (calls & channels table) database.

Here are my questions:
- What is the proper way to clean those calls up, or will FS clean those calls out from database automatically?
- If I decided to restart FS which I know it will clean up those calls, will those calls get written to mod_cdr_csv?

Here is my sip profile:
  <settings>
    <param name="debug" value="0"/>
    <param name="sip-trace" value="no"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="sip-port" value="5060"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="routing"/>
    <param name="dtmf-duration" value="2000"/>
    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="rtp-timer-name" value="soft"/>
    <param name="local-network-acl" value="localnet.auto"/>
    <param name="manage-presence" value="false"/>
    <param name="inbound-codec-negotiation" value="generous"/>
    <param name="nonce-ttl" value="60"/>
    <param name="auth-calls" value="$${internal_auth_calls}"/>
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="auto-nat"/>
    <param name="rtp-timeout-sec" value="300"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>
    <param name="auth-all-packets" value="false"/>
    <param name="disable-transfer" value="true"/>
    <param name="disable-register" value="true"/>
    <param name="inbound-late-negotiation" value="true"/>
    <param name="inbound-bypass-media" value="true"/>
    <param name="apply-inbound-acl" value="domains"/>
  </settings>

Thank you,
-djbinter
!DSPAM:4d7278a232769675984080!


From david.villasmil.work at gmail.com  Sat Mar  5 22:39:28 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 5 Mar 2011 20:39:28 +0100
Subject: [Freeswitch-users] Pastebin
In-Reply-To: <AANLkTimwzc8E-47pJr+zdRrYHwgaQzCw8gtYJTXyRn2h@mail.gmail.com>
References: <AANLkTimWhdnqbf4Oa6jVCTg4SNauv-Sz1rFQ4cifBUfx@mail.gmail.com>
	<1299344219611-6092170.post@n2.nabble.com>
	<AANLkTik60rneBM41RM++roemQ9_GJe2DznTCzOyLJaJw@mail.gmail.com>
	<AANLkTimwzc8E-47pJr+zdRrYHwgaQzCw8gtYJTXyRn2h@mail.gmail.com>
Message-ID: <AANLkTinqz8-e-UuD6qYtAJQUM4Nx+LAvykOzLw8fFCmy@mail.gmail.com>

Maybe i'm just having a dumb year :(


Thanks!

On Sat, Mar 5, 2011 at 8:17 PM, Steven Ayre <steveayre at gmail.com> wrote:

> "the login and password is pastebin/freeswitch"
>
> What's confusing there?
>
> -Steve
>
>
> On 5 March 2011 18:11, David Villasmil <david.villasmil.work at gmail.com>
> wrote:
> > Hello,
> > I read very carefully... I'm in.
> > The wording is a little confusing, btw.
> > Thanks!
> > David
> >
> > On Sat, Mar 5, 2011 at 5:56 PM, mazilo <Nabble at slickdeals.endjunk.com>
> > wrote:
> >>
> >> David Villasmil wrote:
> >> > Why is it that even though I'm logged into freeswitch.org, when I
> click
> >> > on
> >> > the pastebin link, it asks for user/pass and doesn't recognize my
> >> > credentials? it's very annoying.
> >> The user/pass requirement is to ward out any spam bots. Without the
> >> user/pass, it will become an annoyance, if not a nightmare, to the mods
> to
> >> clean out the spam posts by spam bots.
> >>
> >> -----
> >> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> >> --
> >> View this message in context:
> >>
> http://freeswitch-users.2379917.n2.nabble.com/Pastebin-tp6092153p6092170.html
> >> Sent from the freeswitch-users mailing list archive at Nabble.com.
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From simpot at simpot.com  Sat Mar  5 22:53:47 2011
From: simpot at simpot.com (Dmitry Saratsky)
Date: Sat, 5 Mar 2011 21:53:47 +0200
Subject: [Freeswitch-users] How do I originiate new session from within LUA
	script?
Message-ID: <52114D605A462A4E9A50E588EE7D0686012E575D9EE4@mail.forest.simpot.com>

Hello all,

According to wiki:
http://wiki.freeswitch.org/wiki/Mod_lua#session:originate

I can use: "session:originate" command in LUA script to originate new session, but I can't understand how should I do it and can't find any example how I use this command.

Can you help, guys?

Thanks,
Dmitry.
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From mthakershi at gmail.com  Sat Mar  5 23:01:36 2011
From: mthakershi at gmail.com (Malay Thakershi)
Date: Sat, 5 Mar 2011 14:01:36 -0600
Subject: [Freeswitch-users] avmd from mod managed
Message-ID: <AANLkTimq1xPh2SBezytcaj17KXiwmSXsPNEwbY9x16Ee@mail.gmail.com>

Hello,

Is there a way to do AVMD from mod_managed?

I see JavaScript example technique from
http://wiki.freeswitch.org/wiki/Mod_avmd but it seems signature of
StreamFile is different in mod_managed.

In JavaScript, second argument to StreamFile is a function (OnInput) where
detection is done while is mod_managed, it is sample start number.

So is AVMD possible from mod_managed?

Thank you.

Malay
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From freeswitch at peely.com  Sat Mar  5 23:07:26 2011
From: freeswitch at peely.com (peely)
Date: Sat, 5 Mar 2011 12:07:26 -0800 (PST)
Subject: [Freeswitch-users] Recent changes to fs_cli
Message-ID: <1299355646956-6092631.post@n2.nabble.com>

Hi,

It looks like there's been a recent change to fs_cli which catches CTRL +
C's and forces them to be CTRL + D's or specific commands.

I quite often do an fs_cli > output.txt for example, then use nano to
investigate test calls. It seems that now a CTRL + C doesn't end the command
and I have to kill my connection.

THis used to be quite useful so I wonder if it would be possible to have
this functionality back?


Cheers,



Neil.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Recent-changes-to-fs-cli-tp6092631p6092631.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From djbinter at gmail.com  Sat Mar  5 23:46:35 2011
From: djbinter at gmail.com (DJB International)
Date: Sat, 5 Mar 2011 12:46:35 -0800
Subject: [Freeswitch-users] Old hung calls
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EA@cooper>
References: <AANLkTinumTOTJUNJmhazH-ZRTh=jH_412PHV2EG6Wsew@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EA@cooper>
Message-ID: <AANLkTinTM7-p_2eNz1_V4APnbe_1mSph-xCWjqc-ai5w@mail.gmail.com>

Just for your info, I am running in media bypass mode.

In my sip profile:
<param name="inbound-bypass-media" value="true"/>

Thank you,
-djbinter


On Sat, Mar 5, 2011 at 11:21 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> Since RTP timeout is set to 5 minutes (300 sec) FS should clear the call
> after this (and of course the record in core.db as well).
>
> You probably need to pastebin some logs with examples when this happens,
> something seems strange here...
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r DJB International
> [djbinter at gmail.com]
> Skickat: den 5 mars 2011 18:49
> Till: FREESWITCH-USERS MAILING LIST
> ?mne: [Freeswitch-users] Old hung calls
>
> I am running FS git-deec244 2011-02-24 10-22-47 -0600.  I ran into a
> problem the other day when our ISP was down badly for couple hours
> (fortunately after midnight).   The problem was that after the internet went
> back up; there were so many old hung calls before the internet got cut off
> in the core.db (calls & channels table) database.
>
> Here are my questions:
> - What is the proper way to clean those calls up, or will FS clean those
> calls out from database automatically?
> - If I decided to restart FS which I know it will clean up those calls,
> will those calls get written to mod_cdr_csv?
>
> Here is my sip profile:
>  <settings>
>    <param name="debug" value="0"/>
>    <param name="sip-trace" value="no"/>
>    <param name="rfc2833-pt" value="101"/>
>    <param name="sip-port" value="5060"/>
>    <param name="dialplan" value="XML"/>
>    <param name="context" value="routing"/>
>    <param name="dtmf-duration" value="2000"/>
>    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
>    <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
>    <param name="hold-music" value="$${hold_music}"/>
>    <param name="rtp-timer-name" value="soft"/>
>    <param name="local-network-acl" value="localnet.auto"/>
>    <param name="manage-presence" value="false"/>
>    <param name="inbound-codec-negotiation" value="generous"/>
>    <param name="nonce-ttl" value="60"/>
>    <param name="auth-calls" value="$${internal_auth_calls}"/>
>    <param name="rtp-ip" value="$${local_ip_v4}"/>
>    <param name="sip-ip" value="$${local_ip_v4}"/>
>    <param name="ext-rtp-ip" value="auto-nat"/>
>    <param name="ext-sip-ip" value="auto-nat"/>
>    <param name="rtp-timeout-sec" value="300"/>
>    <param name="rtp-hold-timeout-sec" value="1800"/>
>    <param name="auth-all-packets" value="false"/>
>    <param name="disable-transfer" value="true"/>
>    <param name="disable-register" value="true"/>
>    <param name="inbound-late-negotiation" value="true"/>
>    <param name="inbound-bypass-media" value="true"/>
>    <param name="apply-inbound-acl" value="domains"/>
>  </settings>
>
> Thank you,
> -djbinter
> !DSPAM:4d7278a232769675984080!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From peter.olsson at visionutveckling.se  Sat Mar  5 23:48:04 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sat, 5 Mar 2011 21:48:04 +0100
Subject: [Freeswitch-users] How do I originiate new session from within
 LUA	script?
In-Reply-To: <52114D605A462A4E9A50E588EE7D0686012E575D9EE4@mail.forest.simpot.com>
References: <52114D605A462A4E9A50E588EE7D0686012E575D9EE4@mail.forest.simpot.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EB@cooper>

This should work, last parameter is the originate timeout in seconds.

s = freeswitch.Session();
s:originate(session, "sofia/internal/1000 at 192.168.1.1", 10);

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Dmitry Saratsky [simpot at simpot.com]
Skickat: den 5 mars 2011 20:53
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] How do I originiate new session from within LUA        script?

Hello all,

According to wiki:
http://wiki.freeswitch.org/wiki/Mod_lua#session:originate

I can use: ?session:originate? command in LUA script to originate new session, but I can?t understand how should I do it and can?t find any example how I use this command.

Can you help, guys?

Thanks,
Dmitry.
!DSPAM:4d72960532761307510204!


From peter.olsson at visionutveckling.se  Sun Mar  6 00:06:02 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sat, 5 Mar 2011 22:06:02 +0100
Subject: [Freeswitch-users] Old hung calls
In-Reply-To: <AANLkTinTM7-p_2eNz1_V4APnbe_1mSph-xCWjqc-ai5w@mail.gmail.com>
References: <AANLkTinumTOTJUNJmhazH-ZRTh=jH_412PHV2EG6Wsew@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EA@cooper>,
	<AANLkTinTM7-p_2eNz1_V4APnbe_1mSph-xCWjqc-ai5w@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EC@cooper>

Sorry, totally missed this :)

You should probably enable SIP Session timers, check out http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r DJB International [djbinter at gmail.com]
Skickat: den 5 mars 2011 21:46
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Old hung calls

Just for your info, I am running in media bypass mode.

In my sip profile:
<param name="inbound-bypass-media" value="true"/>

Thank you,
-djbinter


On Sat, Mar 5, 2011 at 11:21 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
Since RTP timeout is set to 5 minutes (300 sec) FS should clear the call after this (and of course the record in core.db as well).

You probably need to pastebin some logs with examples when this happens, something seems strange here...

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] f&#246;r DJB International [djbinter at gmail.com<mailto:djbinter at gmail.com>]
Skickat: den 5 mars 2011 18:49
Till: FREESWITCH-USERS MAILING LIST
?mne: [Freeswitch-users] Old hung calls

I am running FS git-deec244 2011-02-24 10-22-47 -0600.  I ran into a problem the other day when our ISP was down badly for couple hours (fortunately after midnight).   The problem was that after the internet went back up; there were so many old hung calls before the internet got cut off in the core.db (calls & channels table) database.

Here are my questions:
- What is the proper way to clean those calls up, or will FS clean those calls out from database automatically?
- If I decided to restart FS which I know it will clean up those calls, will those calls get written to mod_cdr_csv?

Here is my sip profile:
 <settings>
   <param name="debug" value="0"/>
   <param name="sip-trace" value="no"/>
   <param name="rfc2833-pt" value="101"/>
   <param name="sip-port" value="5060"/>
   <param name="dialplan" value="XML"/>
   <param name="context" value="routing"/>
   <param name="dtmf-duration" value="2000"/>
   <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
   <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
   <param name="hold-music" value="$${hold_music}"/>
   <param name="rtp-timer-name" value="soft"/>
   <param name="local-network-acl" value="localnet.auto"/>
   <param name="manage-presence" value="false"/>
   <param name="inbound-codec-negotiation" value="generous"/>
   <param name="nonce-ttl" value="60"/>
   <param name="auth-calls" value="$${internal_auth_calls}"/>
   <param name="rtp-ip" value="$${local_ip_v4}"/>
   <param name="sip-ip" value="$${local_ip_v4}"/>
   <param name="ext-rtp-ip" value="auto-nat"/>
   <param name="ext-sip-ip" value="auto-nat"/>
   <param name="rtp-timeout-sec" value="300"/>
   <param name="rtp-hold-timeout-sec" value="1800"/>
   <param name="auth-all-packets" value="false"/>
   <param name="disable-transfer" value="true"/>
   <param name="disable-register" value="true"/>
   <param name="inbound-late-negotiation" value="true"/>
   <param name="inbound-bypass-media" value="true"/>
   <param name="apply-inbound-acl" value="domains"/>
 </settings>

Thank you,
-djbinter


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4d72a28032761152210852!


From djbinter at gmail.com  Sun Mar  6 00:46:10 2011
From: djbinter at gmail.com (DJB International)
Date: Sat, 5 Mar 2011 13:46:10 -0800
Subject: [Freeswitch-users] Old hung calls
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EC@cooper>
References: <AANLkTinumTOTJUNJmhazH-ZRTh=jH_412PHV2EG6Wsew@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EA@cooper>
	<AANLkTinTM7-p_2eNz1_V4APnbe_1mSph-xCWjqc-ai5w@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EC@cooper>
Message-ID: <AANLkTinnG-7+WHTwMxY=v+38SqshbKNeXQoiAeAR9Qgt@mail.gmail.com>

Peter,

Thank you.  Do I also need <param name="session-timeout" value="1800"/> as
well; otherwise, was it the default session-timeout is 0 as I can see it in
sofia status: SESSION-TO 0

-djbinter


On Sat, Mar 5, 2011 at 1:06 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> Sorry, totally missed this :)
>
> You should probably enable SIP Session timers, check out
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r DJB International
> [djbinter at gmail.com]
> Skickat: den 5 mars 2011 21:46
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] Old hung calls
>
> Just for your info, I am running in media bypass mode.
>
> In my sip profile:
> <param name="inbound-bypass-media" value="true"/>
>
> Thank you,
> -djbinter
>
>
> On Sat, Mar 5, 2011 at 11:21 AM, Peter Olsson <
> peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>>
> wrote:
> Since RTP timeout is set to 5 minutes (300 sec) FS should clear the call
> after this (and of course the record in core.db as well).
>
> You probably need to pastebin some logs with examples when this happens,
> something seems strange here...
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:
> freeswitch-users-bounces at lists.freeswitch.org> [
> freeswitch-users-bounces at lists.freeswitch.org<mailto:
> freeswitch-users-bounces at lists.freeswitch.org>] f&#246;r DJB International
> [djbinter at gmail.com<mailto:djbinter at gmail.com>]
> Skickat: den 5 mars 2011 18:49
> Till: FREESWITCH-USERS MAILING LIST
> ?mne: [Freeswitch-users] Old hung calls
>
> I am running FS git-deec244 2011-02-24 10-22-47 -0600.  I ran into a
> problem the other day when our ISP was down badly for couple hours
> (fortunately after midnight).   The problem was that after the internet went
> back up; there were so many old hung calls before the internet got cut off
> in the core.db (calls & channels table) database.
>
> Here are my questions:
> - What is the proper way to clean those calls up, or will FS clean those
> calls out from database automatically?
> - If I decided to restart FS which I know it will clean up those calls,
> will those calls get written to mod_cdr_csv?
>
> Here is my sip profile:
>  <settings>
>   <param name="debug" value="0"/>
>   <param name="sip-trace" value="no"/>
>   <param name="rfc2833-pt" value="101"/>
>   <param name="sip-port" value="5060"/>
>   <param name="dialplan" value="XML"/>
>   <param name="context" value="routing"/>
>   <param name="dtmf-duration" value="2000"/>
>   <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
>   <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
>   <param name="hold-music" value="$${hold_music}"/>
>   <param name="rtp-timer-name" value="soft"/>
>   <param name="local-network-acl" value="localnet.auto"/>
>   <param name="manage-presence" value="false"/>
>   <param name="inbound-codec-negotiation" value="generous"/>
>   <param name="nonce-ttl" value="60"/>
>   <param name="auth-calls" value="$${internal_auth_calls}"/>
>   <param name="rtp-ip" value="$${local_ip_v4}"/>
>   <param name="sip-ip" value="$${local_ip_v4}"/>
>   <param name="ext-rtp-ip" value="auto-nat"/>
>   <param name="ext-sip-ip" value="auto-nat"/>
>   <param name="rtp-timeout-sec" value="300"/>
>   <param name="rtp-hold-timeout-sec" value="1800"/>
>   <param name="auth-all-packets" value="false"/>
>   <param name="disable-transfer" value="true"/>
>   <param name="disable-register" value="true"/>
>   <param name="inbound-late-negotiation" value="true"/>
>   <param name="inbound-bypass-media" value="true"/>
>   <param name="apply-inbound-acl" value="domains"/>
>  </settings>
>
> Thank you,
> -djbinter
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4d72a28032761152210852!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From simpot at simpot.com  Sun Mar  6 00:51:23 2011
From: simpot at simpot.com (Dmitry Saratsky)
Date: Sat, 5 Mar 2011 23:51:23 +0200
Subject: [Freeswitch-users] How do I originiate new session from within
 LUA	script?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EB@cooper>
References: <52114D605A462A4E9A50E588EE7D0686012E575D9EE4@mail.forest.simpot.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EB@cooper>
Message-ID: <52114D605A462A4E9A50E588EE7D0686012E575D9EE5@mail.forest.simpot.com>

Hi Peter,

Thanks for your replay.
It works in this way, thanks.

Can you please also explain (or point me to right documentation) what does mean "10" as 3rd variable in your example?
Also I'm looking the way to set session variables before origination and I also failed with this...

For example I have tried this (the call itself was succeeded, but caller_id was not passed):
   s = freeswitch.Session();
   s:setVariable("effective_caller_id_number", "999");
   s:setVariable("effective_caller_id_name", "999");
   s:originate(session, "sofia/gateway/mygw/1000", 10);

and I got the following in console:

2011-03-05 23:43:05.754621 [ERR] switch_cpp.cpp:634 session is not initalized
2011-03-05 23:43:05.754621 [ERR] switch_cpp.cpp:634 session is not initialized

Thanks,
Dmitry.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson
Sent: 05 Mar 2011 22:48
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How do I originiate new session from within LUA script?

This should work, last parameter is the originate timeout in seconds.

s = freeswitch.Session();
s:originate(session, "sofia/internal/1000 at 192.168.1.1", 10);

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Dmitry Saratsky [simpot at simpot.com]
Skickat: den 5 mars 2011 20:53
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] How do I originiate new session from within LUA        script?

Hello all,

According to wiki:
http://wiki.freeswitch.org/wiki/Mod_lua#session:originate

I can use: "session:originate" command in LUA script to originate new session, but I can't understand how should I do it and can't find any example how I use this command.

Can you help, guys?

Thanks,
Dmitry.
!DSPAM:4d72960532761307510204!

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


From peter.olsson at visionutveckling.se  Sun Mar  6 01:18:01 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sat, 5 Mar 2011 23:18:01 +0100
Subject: [Freeswitch-users] Old hung calls
In-Reply-To: <AANLkTinnG-7+WHTwMxY=v+38SqshbKNeXQoiAeAR9Qgt@mail.gmail.com>
References: <AANLkTinumTOTJUNJmhazH-ZRTh=jH_412PHV2EG6Wsew@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EA@cooper>
	<AANLkTinTM7-p_2eNz1_V4APnbe_1mSph-xCWjqc-ai5w@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EC@cooper>,
	<AANLkTinnG-7+WHTwMxY=v+38SqshbKNeXQoiAeAR9Qgt@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2C494ED@cooper>

It should be default 30 minutes (1800), but to be on the safe side you could set the param as well. Also make sure to restart the profile (or FS) after the change.
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r DJB International [djbinter at gmail.com]
Skickat: den 5 mars 2011 22:46
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Old hung calls

Peter,

Thank you.  Do I also need <param name="session-timeout" value="1800"/> as well; otherwise, was it the default session-timeout is 0 as I can see it in sofia status: SESSION-TO 0

-djbinter


On Sat, Mar 5, 2011 at 1:06 PM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
Sorry, totally missed this :)

You should probably enable SIP Session timers, check out http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] f&#246;r DJB International [djbinter at gmail.com<mailto:djbinter at gmail.com>]
Skickat: den 5 mars 2011 21:46
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Old hung calls

Just for your info, I am running in media bypass mode.

In my sip profile:
<param name="inbound-bypass-media" value="true"/>

Thank you,
-djbinter


On Sat, Mar 5, 2011 at 11:21 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se><mailto:peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>>> wrote:
Since RTP timeout is set to 5 minutes (300 sec) FS should clear the call after this (and of course the record in core.db as well).

You probably need to pastebin some logs with examples when this happens, something seems strange here...

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org><mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>> [freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org><mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>>] f&#246;r DJB International [djbinter at gmail.com<mailto:djbinter at gmail.com><mailto:djbinter at gmail.com<mailto:djbinter at gmail.com>>]
Skickat: den 5 mars 2011 18:49
Till: FREESWITCH-USERS MAILING LIST
?mne: [Freeswitch-users] Old hung calls

I am running FS git-deec244 2011-02-24 10-22-47 -0600.  I ran into a problem the other day when our ISP was down badly for couple hours (fortunately after midnight).   The problem was that after the internet went back up; there were so many old hung calls before the internet got cut off in the core.db (calls & channels table) database.

Here are my questions:
- What is the proper way to clean those calls up, or will FS clean those calls out from database automatically?
- If I decided to restart FS which I know it will clean up those calls, will those calls get written to mod_cdr_csv?

Here is my sip profile:
 <settings>
  <param name="debug" value="0"/>
  <param name="sip-trace" value="no"/>
  <param name="rfc2833-pt" value="101"/>
  <param name="sip-port" value="5060"/>
  <param name="dialplan" value="XML"/>
  <param name="context" value="routing"/>
  <param name="dtmf-duration" value="2000"/>
  <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
  <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
  <param name="hold-music" value="$${hold_music}"/>
  <param name="rtp-timer-name" value="soft"/>
  <param name="local-network-acl" value="localnet.auto"/>
  <param name="manage-presence" value="false"/>
  <param name="inbound-codec-negotiation" value="generous"/>
  <param name="nonce-ttl" value="60"/>
  <param name="auth-calls" value="$${internal_auth_calls}"/>
  <param name="rtp-ip" value="$${local_ip_v4}"/>
  <param name="sip-ip" value="$${local_ip_v4}"/>
  <param name="ext-rtp-ip" value="auto-nat"/>
  <param name="ext-sip-ip" value="auto-nat"/>
  <param name="rtp-timeout-sec" value="300"/>
  <param name="rtp-hold-timeout-sec" value="1800"/>
  <param name="auth-all-packets" value="false"/>
  <param name="disable-transfer" value="true"/>
  <param name="disable-register" value="true"/>
  <param name="inbound-late-negotiation" value="true"/>
  <param name="inbound-bypass-media" value="true"/>
  <param name="apply-inbound-acl" value="domains"/>
 </settings>

Thank you,
-djbinter


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From peter.olsson at visionutveckling.se  Sun Mar  6 01:29:17 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sat, 5 Mar 2011 23:29:17 +0100
Subject: [Freeswitch-users] How do I originiate new session from within
 LUA	script?
In-Reply-To: <52114D605A462A4E9A50E588EE7D0686012E575D9EE5@mail.forest.simpot.com>
References: <52114D605A462A4E9A50E588EE7D0686012E575D9EE4@mail.forest.simpot.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EB@cooper>,
	<52114D605A462A4E9A50E588EE7D0686012E575D9EE5@mail.forest.simpot.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EE@cooper>

10 is the origination timeout in seconds. This part is missing in the documentation right now, so I can't really tell you where to look - except in the source :)

To set variables you should be able to do it as you do in the dialplan, changing the dial string to something like "{var1=value,var2=value}sofia/internal/test at test.com". By the way - for originate variables origination_caller_id_xxxx must be used.

If you're successful with these commands, I would really appreciate if this could be updated in the Wiki as well.

/Peter

________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Dmitry Saratsky [simpot at simpot.com]
Skickat: den 5 mars 2011 22:51
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] How do I originiate new session from within LUA    script?

Hi Peter,

Thanks for your replay.
It works in this way, thanks.

Can you please also explain (or point me to right documentation) what does mean "10" as 3rd variable in your example?
Also I'm looking the way to set session variables before origination and I also failed with this...

For example I have tried this (the call itself was succeeded, but caller_id was not passed):
   s = freeswitch.Session();
   s:setVariable("effective_caller_id_number", "999");
   s:setVariable("effective_caller_id_name", "999");
   s:originate(session, "sofia/gateway/mygw/1000", 10);

and I got the following in console:

2011-03-05 23:43:05.754621 [ERR] switch_cpp.cpp:634 session is not initalized
2011-03-05 23:43:05.754621 [ERR] switch_cpp.cpp:634 session is not initialized

Thanks,
Dmitry.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson
Sent: 05 Mar 2011 22:48
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How do I originiate new session from within LUA script?

This should work, last parameter is the originate timeout in seconds.

s = freeswitch.Session();
s:originate(session, "sofia/internal/1000 at 192.168.1.1", 10);

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Dmitry Saratsky [simpot at simpot.com]
Skickat: den 5 mars 2011 20:53
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] How do I originiate new session from within LUA        script?

Hello all,

According to wiki:
http://wiki.freeswitch.org/wiki/Mod_lua#session:originate

I can use: "session:originate" command in LUA script to originate new session, but I can't understand how should I do it and can't find any example how I use this command.

Can you help, guys?

Thanks,
Dmitry.


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From sunwood360 at gmail.com  Sun Mar  6 03:08:40 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Sat, 5 Mar 2011 16:08:40 -0800
Subject: [Freeswitch-users] send fax via google voice.
Message-ID: <AANLkTin4uo+oKr_Nzw6qVzf72E7n896j_5xyoB6nFcnk@mail.gmail.com>

Google Voice integration with FS works fine. but sending fax seems not
working.  12068235374 is a K7 number, If I dial it , I can hear fax machine
answering sound.
In fact, people has used GV to send fax via fax machine. (
http://www.magicjacksupport.com/faxing-via-google-voice-t7412.html)
Is this problem related with mod_dingaling?

please take a look!

thanks



itch at internal> originate
dingaling/xmppc/+12068235374 at voice.google.com&txfax(/tmp/test.tiff)
+OK c20e3fc4-4784-11e0-8d1d-eda21484fd55

2011-03-05 16:00:33.808039 [NOTICE] switch_channel.c:808 New Channel
dingaling/xmppc/+12068235374 at voice.google.com[c20e3fc4-4784-11e0-8d1d-eda21484fd55]

freeswitch at internal> 2011-03-05 16:00:33.809595 [DEBUG] mod_dingaling.c:1824
(dingaling/xmppc/+12068235374 at voice.google.com) State Change CS_NEW ->
CS_INIT

2011-03-05 16:00:33.809595 [DEBUG] switch_core_session.c:1116 Send signal
dingaling/xmppc/+12068235374 at voice.google.com [BREAK]
2011-03-05 16:00:33.809595 [DEBUG] mod_dingaling.c:1348 dingaling/xmppc/+
12068235374 at voice.google.com CHANNEL KILL
2011-03-05 16:00:33.810718 [DEBUG] switch_core_state_machine.c:320
(dingaling/xmppc/+12068235374 at voice.google.com) Running State Change
CS_INIT

2011-03-05 16:00:33.810718 [DEBUG] switch_core_state_machine.c:356
(dingaling/xmppc/+12068235374 at voice.google.com) State INIT
2011-03-05 16:00:33.810718 [NOTICE] mod_dingaling.c:1110 Ring-Ready
dingaling/xmppc/+12068235374 at voice.google.com!
2011-03-05 16:00:33.811202 [DEBUG] mod_dingaling.c:1063 Don't have my codec
yet here's one
2011-03-05 16:00:33.811202 [DEBUG] mod_dingaling.c:1083 Send Describe
[PCMU at 8000]
2011-03-05 16:00:34.000304 [DEBUG] mod_dingaling.c:2941 using Existing
session for 6100480476
2011-03-05 16:00:34.000304 [DEBUG] mod_dingaling.c:1083 Send Describe
[PCMU at 8000]
2011-03-05 16:00:34.311385 [DEBUG] mod_dingaling.c:2941 using Existing
session for 6100480476
2011-03-05 16:00:34.311385 [DEBUG] mod_dingaling.c:1008 Send Candidate
192.168.1.146:17960 [9CsLT8VWsB1yDqnv]
2011-03-05 16:00:34.707433 [DEBUG] mod_dingaling.c:2941 using Existing
session for 6100480476
2011-03-05 16:00:34.707433 [DEBUG] mod_dingaling.c:3279 3
candidates

2011-03-05 16:00:34.707433 [DEBUG] mod_dingaling.c:3299 candidate
74.125.53.126:19295 PASS ACL wan.auto
2011-03-05 16:00:34.707433 [DEBUG] mod_dingaling.c:3351 Acceptable Candidate
74.125.53.126:19295
2011-03-05 16:00:40.334509 [DEBUG] mod_dingaling.c:2941 using Existing
session for 6100480476
2011-03-05 16:00:40.334509 [DEBUG] mod_dingaling.c:3193 Already decided on a
codec
2011-03-05 16:00:40.336590 [DEBUG] mod_dingaling.c:865 Set Read Codec to
PCMU at 8000
2011-03-05 16:00:40.336590 [DEBUG] mod_dingaling.c:880 Set Write Codec to
PCMU at 8000
2011-03-05 16:00:40.337483 [DEBUG] mod_dingaling.c:892 SETUP RTP
192.168.1.146:17960 -> 74.125.53.126:19295
2011-03-05 16:00:40.337483 [DEBUG] switch_rtp.c:1429 Starting timer [soft]
160 bytes per 20ms
2011-03-05 16:00:40.339652 [DEBUG] switch_rtp.c:3775 Activate VAD codec PCMU
20ms
2011-03-05 16:00:40.340311 [DEBUG] switch_channel.c:2782 (dingaling/xmppc/+
12068235374 at voice.google.com) Callstate Change DOWN -> ACTIVE
2011-03-05 16:00:40.340311 [NOTICE] mod_dingaling.c:1202 Channel
[dingaling/xmppc/+12068235374 at voice.google.com] has been answered
2011-03-05 16:00:40.340311 [DEBUG] mod_dingaling.c:1205 (dingaling/xmppc/+
12068235374 at voice.google.com) State Change CS_INIT -> CS_ROUTING
2011-03-05 16:00:40.340311 [DEBUG] switch_core_session.c:1116 Send signal
dingaling/xmppc/+12068235374 at voice.google.com [BREAK]
2011-03-05 16:00:40.340311 [DEBUG] mod_dingaling.c:1348 dingaling/xmppc/+
12068235374 at voice.google.com CHANNEL KILL
2011-03-05 16:00:40.340311 [DEBUG] switch_core_state_machine.c:356
(dingaling/xmppc/+12068235374 at voice.google.com) State INIT going to sleep
2011-03-05 16:00:40.340311 [DEBUG] switch_core_state_machine.c:320
(dingaling/xmppc/+12068235374 at voice.google.com) Running State Change
CS_ROUTING

2011-03-05 16:00:40.340311 [DEBUG] switch_channel.c:1657 (dingaling/xmppc/+
12068235374 at voice.google.com) Callstate Change ACTIVE -> RINGING
2011-03-05 16:00:40.340311 [DEBUG] switch_core_state_machine.c:359
(dingaling/xmppc/+12068235374 at voice.google.com) State ROUTING
2011-03-05 16:00:40.340311 [DEBUG] mod_dingaling.c:1219 dingaling/xmppc/+
12068235374 at voice.google.com CHANNEL ROUTING
2011-03-05 16:00:40.340311 [DEBUG] switch_ivr_originate.c:66
(dingaling/xmppc/+12068235374 at voice.google.com) State Change CS_ROUTING ->
CS_CONSUME_MEDIA

2011-03-05 16:00:40.340311 [DEBUG] switch_core_session.c:1116 Send signal
dingaling/xmppc/+12068235374 at voice.google.com [BREAK]
2011-03-05 16:00:40.340311 [DEBUG] switch_ivr_originate.c:3388 Originate
Resulted in Success: [dingaling/xmppc/+12068235374 at voice.google.com]

2011-03-05 16:00:40.340311 [DEBUG] mod_dingaling.c:1348 dingaling/xmppc/+
12068235374 at voice.google.com CHANNEL KILL
2011-03-05 16:00:40.340311 [DEBUG] mod_commands.c:3194 (dingaling/xmppc/+
12068235374 at voice.google.com) State Change CS_CONSUME_MEDIA ->
CS_EXECUTE

2011-03-05 16:00:40.340311 [DEBUG] switch_core_session.c:1116 Send signal
dingaling/xmppc/+12068235374 at voice.google.com [BREAK]
2011-03-05 16:00:40.340311 [DEBUG] mod_dingaling.c:1348 dingaling/xmppc/+
12068235374 at voice.google.com CHANNEL KILL
2011-03-05 16:00:40.340311 [DEBUG] switch_core_state_machine.c:359
(dingaling/xmppc/+12068235374 at voice.google.com) State ROUTING going to
sleep

2011-03-05 16:00:40.340311 [DEBUG] switch_core_state_machine.c:320
(dingaling/xmppc/+12068235374 at voice.google.com) Running State Change
CS_EXECUTE

2011-03-05 16:00:40.340311 [DEBUG] switch_channel.c:1659 (dingaling/xmppc/+
12068235374 at voice.google.com) Callstate Change RINGING -> ACTIVE
2011-03-05 16:00:40.342729 [DEBUG] switch_core_state_machine.c:366
(dingaling/xmppc/+12068235374 at voice.google.com) State EXECUTE
2011-03-05 16:00:40.342729 [DEBUG] mod_dingaling.c:1236 dingaling/xmppc/+
12068235374 at voice.google.com CHANNEL EXECUTE
2011-03-05 16:00:40.342729 [DEBUG] switch_core_state_machine.c:157
dingaling/xmppc/+12068235374 at voice.google.com Standard EXECUTE
EXECUTE dingaling/xmppc/+12068235374 at voice.google.comtxfax(/tmp/test.tiff)

2011-03-05 16:00:40.344059 [DEBUG] mod_spandsp_fax.c:1092 Raw read codec
activation Success L16 20000
2011-03-05 16:00:40.344059 [DEBUG] switch_core_codec.c:116 dingaling/xmppc/+
12068235374 at voice.google.com Push codec L16:10
2011-03-05 16:00:40.344059 [DEBUG] mod_spandsp_fax.c:1108 Raw write codec
activation Success L16
2011-03-05 16:00:42.649258 [DEBUG] mod_dingaling.c:2941 using Existing
session for 6100480476
2011-03-05 16:00:42.649258 [DEBUG] mod_dingaling.c:3411 hungup
dingaling/xmppc/+12068235374 at voice.google.com

2011-03-05 16:00:42.649258 [DEBUG] mod_dingaling.c:710 Terminate called from
line 3412 state=CS_EXECUTE
2011-03-05 16:00:42.649258 [DEBUG] switch_channel.c:2535 (dingaling/xmppc/+
12068235374 at voice.google.com) Callstate Change ACTIVE -> HANGUP
2011-03-05 16:00:42.649258 [NOTICE] mod_dingaling.c:731 Hangup
dingaling/xmppc/+12068235374 at voice.google.com [CS_EXECUTE] [NORMAL_CLEARING]

2011-03-05 16:00:42.649258 [DEBUG] switch_channel.c:2551 Send signal
dingaling/xmppc/+12068235374 at voice.google.com [KILL]
2011-03-05 16:00:42.649258 [DEBUG] mod_dingaling.c:1348 dingaling/xmppc/+
12068235374 at voice.google.com CHANNEL KILL
2011-03-05 16:00:42.649258 [DEBUG] switch_core_session.c:1116 Send signal
dingaling/xmppc/+12068235374 at voice.google.com [BREAK]
2011-03-05 16:00:42.649258 [DEBUG] mod_dingaling.c:1348 dingaling/xmppc/+
12068235374 at voice.google.com CHANNEL KILL
2011-03-05 16:00:42.649258 [DEBUG] mod_dingaling.c:3413 End
Call

2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:323
==============================================================================

2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:335 Fax processing not
successful - result (49) The call dropped prematurely.
2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:340 Remote station
id:
2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:341 Local station id:
SpanDSP Fax Ident
2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:342 Pages transferred:
0
2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:344 Total fax pages:
0
2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:345 Image resolution:
0x0
2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:346 Transfer Rate:
14400
2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:348 ECM status
off
2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:349 remote
country:
2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:350 remote
vendor:
2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:351 remote
model:
2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:353
==============================================================================

2011-03-05 16:00:42.658379 [DEBUG] switch_core_codec.c:141 dingaling/xmppc/+
12068235374 at voice.google.com Restore previous codec PCMU:0.
2011-03-05 16:00:42.658379 [DEBUG] switch_core_session.c:2045
dingaling/xmppc/+12068235374 at voice.google.com skip receive message
[APPLICATION_EXEC_COMPLETE] (channel is hungup
already)

2011-03-05 16:00:42.658379 [DEBUG] switch_core_state_machine.c:366
(dingaling/xmppc/+12068235374 at voice.google.com) State EXECUTE going to
sleep

2011-03-05 16:00:42.658379 [DEBUG] switch_core_state_machine.c:320
(dingaling/xmppc/+12068235374 at voice.google.com) Running State Change
CS_HANGUP

2011-03-05 16:00:42.659331 [DEBUG] switch_core_state_machine.c:557
(dingaling/xmppc/+12068235374 at voice.google.com) State HANGUP
2011-03-05 16:00:42.659331 [DEBUG] mod_dingaling.c:1317 dingaling/xmppc/+
12068235374 at voice.google.com CHANNEL HANGUP
-------------- next part --------------
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From simpot at simpot.com  Sun Mar  6 04:03:03 2011
From: simpot at simpot.com (Dmitry Saratsky)
Date: Sun, 6 Mar 2011 03:03:03 +0200
Subject: [Freeswitch-users] How do I originiate new session from within
 LUA	script?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EE@cooper>
References: <52114D605A462A4E9A50E588EE7D0686012E575D9EE4@mail.forest.simpot.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EB@cooper>,
	<52114D605A462A4E9A50E588EE7D0686012E575D9EE5@mail.forest.simpot.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EE@cooper>
Message-ID: <52114D605A462A4E9A50E588EE7D0686012E575D9EE6@mail.forest.simpot.com>

Hi Peter,

Thanks again for your help, this way works well for me.
I will update wiki, when I fix all puzzles of my script.

Now I have another problem... Sorry...

I actually hoped also to read session variables after my new originated sessions complete in same way, I set them.
However in way you have suggested me to set vars, I have no idea how to read those variables now...

I have tried to use the following after origination command:
   local fax_result_code = session:getVariable("fax_result_code"); local fax_result_text = session:getVariable("fax_result_text");
   freeswitch.consoleLog("info", "FAX - Error code: (" .. fax_result_code .. ") " .. fax_result_text .. "\n");

I got the following in console:
2011-03-06 02:54:23.311416 [ERR] mod_lua.cpp:182 /usr/local/freeswitch/scripts/faxoutservice.lua:53: attempt to concatenate global 'fax_success' (a nil value)
stack traceback:
        /usr/local/freeswitch/scripts/faxoutservice.lua:53: in function 'send_fax'
        /usr/local/freeswitch/scripts/faxoutservice.lua:74: in main chunk

According to docs (http://wiki.freeswitch.org/wiki/Mod_spandsp), those vars should be set on hangup both for tx and rx faxes...
Anyway, I have lua script I have wrote for receiving faxes - I can read those variables successfully in the way I wrote above.

So, any ideas how can I read session (that I start in that way - origination) variables after session hangup?

Thanks again,
Dmitry.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson
Sent: 06 Mar 2011 00:29
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How do I originiate new session from within LUA script?

10 is the origination timeout in seconds. This part is missing in the documentation right now, so I can't really tell you where to look - except in the source :)

To set variables you should be able to do it as you do in the dialplan, changing the dial string to something like "{var1=value,var2=value}sofia/internal/test at test.com". By the way - for originate variables origination_caller_id_xxxx must be used.

If you're successful with these commands, I would really appreciate if this could be updated in the Wiki as well.

/Peter

________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Dmitry Saratsky [simpot at simpot.com]
Skickat: den 5 mars 2011 22:51
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] How do I originiate new session from within LUA    script?

Hi Peter,

Thanks for your replay.
It works in this way, thanks.

Can you please also explain (or point me to right documentation) what does mean "10" as 3rd variable in your example?
Also I'm looking the way to set session variables before origination and I also failed with this...

For example I have tried this (the call itself was succeeded, but caller_id was not passed):
   s = freeswitch.Session();
   s:setVariable("effective_caller_id_number", "999");
   s:setVariable("effective_caller_id_name", "999");
   s:originate(session, "sofia/gateway/mygw/1000", 10);

and I got the following in console:

2011-03-05 23:43:05.754621 [ERR] switch_cpp.cpp:634 session is not initalized
2011-03-05 23:43:05.754621 [ERR] switch_cpp.cpp:634 session is not initialized

Thanks,
Dmitry.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson
Sent: 05 Mar 2011 22:48
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How do I originiate new session from within LUA script?

This should work, last parameter is the originate timeout in seconds.

s = freeswitch.Session();
s:originate(session, "sofia/internal/1000 at 192.168.1.1", 10);

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Dmitry Saratsky [simpot at simpot.com]
Skickat: den 5 mars 2011 20:53
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] How do I originiate new session from within LUA        script?

Hello all,

According to wiki:
http://wiki.freeswitch.org/wiki/Mod_lua#session:originate

I can use: "session:originate" command in LUA script to originate new session, but I can't understand how should I do it and can't find any example how I use this command.

Can you help, guys?

Thanks,
Dmitry.


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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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http://www.freeswitch.org

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!DSPAM:4d72b11b32769028195376!


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From curriegrad2004 at gmail.com  Sun Mar  6 05:33:09 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 5 Mar 2011 18:33:09 -0800
Subject: [Freeswitch-users] send fax via google voice.
In-Reply-To: <AANLkTin4uo+oKr_Nzw6qVzf72E7n896j_5xyoB6nFcnk@mail.gmail.com>
References: <AANLkTin4uo+oKr_Nzw6qVzf72E7n896j_5xyoB6nFcnk@mail.gmail.com>
Message-ID: <AANLkTimDfszKNxgugnSECrtCE+ByKgYV-sjK0tjXppfK@mail.gmail.com>

I'd personally not go there if you need to use GV to fax. Faxing with
uLaw isn't reliable at all. You're just playing with fire if you do
decide to use uLaw to fax stuff. Besides, google voice's quality is
just horrible anyways.

On Sat, Mar 5, 2011 at 4:08 PM, envelopes envelopes
<sunwood360 at gmail.com> wrote:
> Google Voice integration with FS works fine. but sending fax seems not
> working.? 12068235374 is a K7 number, If I dial it , I can hear fax machine
> answering sound.
> In fact, people has used GV to send fax via fax machine.
> (http://www.magicjacksupport.com/faxing-via-google-voice-t7412.html)
> Is this problem related with mod_dingaling?
>
> please take a look!
>
> thanks
>
>
>
> itch at internal> originate dingaling/xmppc/+12068235374 at voice.google.com
> &txfax(/tmp/test.tiff)
> +OK c20e3fc4-4784-11e0-8d1d-eda21484fd55
>
> 2011-03-05 16:00:33.808039 [NOTICE] switch_channel.c:808 New Channel
> dingaling/xmppc/+12068235374 at voice.google.com
> [c20e3fc4-4784-11e0-8d1d-eda21484fd55]
> freeswitch at internal> 2011-03-05 16:00:33.809595 [DEBUG] mod_dingaling.c:1824
> (dingaling/xmppc/+12068235374 at voice.google.com) State Change CS_NEW ->
> CS_INIT
> 2011-03-05 16:00:33.809595 [DEBUG] switch_core_session.c:1116 Send signal
> dingaling/xmppc/+12068235374 at voice.google.com [BREAK]
> 2011-03-05 16:00:33.809595 [DEBUG] mod_dingaling.c:1348
> dingaling/xmppc/+12068235374 at voice.google.com CHANNEL
> KILL
> 2011-03-05 16:00:33.810718 [DEBUG] switch_core_state_machine.c:320
> (dingaling/xmppc/+12068235374 at voice.google.com) Running State Change
> CS_INIT
> 2011-03-05 16:00:33.810718 [DEBUG] switch_core_state_machine.c:356
> (dingaling/xmppc/+12068235374 at voice.google.com) State INIT
> 2011-03-05 16:00:33.810718 [NOTICE] mod_dingaling.c:1110 Ring-Ready
> dingaling/xmppc/+12068235374 at voice.google.com!
> 2011-03-05 16:00:33.811202 [DEBUG] mod_dingaling.c:1063 Don't have my codec
> yet here's one
> 2011-03-05 16:00:33.811202 [DEBUG] mod_dingaling.c:1083 Send Describe
> [PCMU at 8000]
> 2011-03-05 16:00:34.000304 [DEBUG] mod_dingaling.c:2941 using Existing
> session for 6100480476
> 2011-03-05 16:00:34.000304 [DEBUG] mod_dingaling.c:1083 Send Describe
> [PCMU at 8000]
> 2011-03-05 16:00:34.311385 [DEBUG] mod_dingaling.c:2941 using Existing
> session for 6100480476
> 2011-03-05 16:00:34.311385 [DEBUG] mod_dingaling.c:1008 Send Candidate
> 192.168.1.146:17960 [9CsLT8VWsB1yDqnv]
> 2011-03-05 16:00:34.707433 [DEBUG] mod_dingaling.c:2941 using Existing
> session for 6100480476
> 2011-03-05 16:00:34.707433 [DEBUG] mod_dingaling.c:3279 3
> candidates
> 2011-03-05 16:00:34.707433 [DEBUG] mod_dingaling.c:3299 candidate
> 74.125.53.126:19295 PASS ACL wan.auto
> 2011-03-05 16:00:34.707433 [DEBUG] mod_dingaling.c:3351 Acceptable Candidate
> 74.125.53.126:19295
> 2011-03-05 16:00:40.334509 [DEBUG] mod_dingaling.c:2941 using Existing
> session for 6100480476
> 2011-03-05 16:00:40.334509 [DEBUG] mod_dingaling.c:3193 Already decided on a
> codec
> 2011-03-05 16:00:40.336590 [DEBUG] mod_dingaling.c:865 Set Read Codec to
> PCMU at 8000
> 2011-03-05 16:00:40.336590 [DEBUG] mod_dingaling.c:880 Set Write Codec to
> PCMU at 8000
> 2011-03-05 16:00:40.337483 [DEBUG] mod_dingaling.c:892 SETUP RTP
> 192.168.1.146:17960 -> 74.125.53.126:19295
> 2011-03-05 16:00:40.337483 [DEBUG] switch_rtp.c:1429 Starting timer [soft]
> 160 bytes per 20ms
> 2011-03-05 16:00:40.339652 [DEBUG] switch_rtp.c:3775 Activate VAD codec PCMU
> 20ms
> 2011-03-05 16:00:40.340311 [DEBUG] switch_channel.c:2782
> (dingaling/xmppc/+12068235374 at voice.google.com) Callstate Change DOWN ->
> ACTIVE
> 2011-03-05 16:00:40.340311 [NOTICE] mod_dingaling.c:1202 Channel
> [dingaling/xmppc/+12068235374 at voice.google.com] has been answered
> 2011-03-05 16:00:40.340311 [DEBUG] mod_dingaling.c:1205
> (dingaling/xmppc/+12068235374 at voice.google.com) State Change CS_INIT ->
> CS_ROUTING
> 2011-03-05 16:00:40.340311 [DEBUG] switch_core_session.c:1116 Send signal
> dingaling/xmppc/+12068235374 at voice.google.com [BREAK]
> 2011-03-05 16:00:40.340311 [DEBUG] mod_dingaling.c:1348
> dingaling/xmppc/+12068235374 at voice.google.com CHANNEL
> KILL
> 2011-03-05 16:00:40.340311 [DEBUG] switch_core_state_machine.c:356
> (dingaling/xmppc/+12068235374 at voice.google.com) State INIT going to sleep
> 2011-03-05 16:00:40.340311 [DEBUG] switch_core_state_machine.c:320
> (dingaling/xmppc/+12068235374 at voice.google.com) Running State Change
> CS_ROUTING
> 2011-03-05 16:00:40.340311 [DEBUG] switch_channel.c:1657
> (dingaling/xmppc/+12068235374 at voice.google.com) Callstate Change ACTIVE ->
> RINGING
> 2011-03-05 16:00:40.340311 [DEBUG] switch_core_state_machine.c:359
> (dingaling/xmppc/+12068235374 at voice.google.com) State ROUTING
> 2011-03-05 16:00:40.340311 [DEBUG] mod_dingaling.c:1219
> dingaling/xmppc/+12068235374 at voice.google.com CHANNEL
> ROUTING
> 2011-03-05 16:00:40.340311 [DEBUG] switch_ivr_originate.c:66
> (dingaling/xmppc/+12068235374 at voice.google.com) State Change CS_ROUTING ->
> CS_CONSUME_MEDIA
> 2011-03-05 16:00:40.340311 [DEBUG] switch_core_session.c:1116 Send signal
> dingaling/xmppc/+12068235374 at voice.google.com [BREAK]
> 2011-03-05 16:00:40.340311 [DEBUG] switch_ivr_originate.c:3388 Originate
> Resulted in Success:
> [dingaling/xmppc/+12068235374 at voice.google.com]
> 2011-03-05 16:00:40.340311 [DEBUG] mod_dingaling.c:1348
> dingaling/xmppc/+12068235374 at voice.google.com CHANNEL
> KILL
> 2011-03-05 16:00:40.340311 [DEBUG] mod_commands.c:3194
> (dingaling/xmppc/+12068235374 at voice.google.com) State Change
> CS_CONSUME_MEDIA ->
> CS_EXECUTE
> 2011-03-05 16:00:40.340311 [DEBUG] switch_core_session.c:1116 Send signal
> dingaling/xmppc/+12068235374 at voice.google.com [BREAK]
> 2011-03-05 16:00:40.340311 [DEBUG] mod_dingaling.c:1348
> dingaling/xmppc/+12068235374 at voice.google.com CHANNEL
> KILL
> 2011-03-05 16:00:40.340311 [DEBUG] switch_core_state_machine.c:359
> (dingaling/xmppc/+12068235374 at voice.google.com) State ROUTING going to
> sleep
> 2011-03-05 16:00:40.340311 [DEBUG] switch_core_state_machine.c:320
> (dingaling/xmppc/+12068235374 at voice.google.com) Running State Change
> CS_EXECUTE
> 2011-03-05 16:00:40.340311 [DEBUG] switch_channel.c:1659
> (dingaling/xmppc/+12068235374 at voice.google.com) Callstate Change RINGING ->
> ACTIVE
> 2011-03-05 16:00:40.342729 [DEBUG] switch_core_state_machine.c:366
> (dingaling/xmppc/+12068235374 at voice.google.com) State EXECUTE
> 2011-03-05 16:00:40.342729 [DEBUG] mod_dingaling.c:1236
> dingaling/xmppc/+12068235374 at voice.google.com CHANNEL
> EXECUTE
> 2011-03-05 16:00:40.342729 [DEBUG] switch_core_state_machine.c:157
> dingaling/xmppc/+12068235374 at voice.google.com Standard EXECUTE
> EXECUTE dingaling/xmppc/+12068235374 at voice.google.com
> txfax(/tmp/test.tiff)
> 2011-03-05 16:00:40.344059 [DEBUG] mod_spandsp_fax.c:1092 Raw read codec
> activation Success L16 20000
> 2011-03-05 16:00:40.344059 [DEBUG] switch_core_codec.c:116
> dingaling/xmppc/+12068235374 at voice.google.com Push codec
> L16:10
> 2011-03-05 16:00:40.344059 [DEBUG] mod_spandsp_fax.c:1108 Raw write codec
> activation Success L16
> 2011-03-05 16:00:42.649258 [DEBUG] mod_dingaling.c:2941 using Existing
> session for 6100480476
> 2011-03-05 16:00:42.649258 [DEBUG] mod_dingaling.c:3411 hungup
> dingaling/xmppc/+12068235374 at voice.google.com
> 2011-03-05 16:00:42.649258 [DEBUG] mod_dingaling.c:710 Terminate called from
> line 3412 state=CS_EXECUTE
> 2011-03-05 16:00:42.649258 [DEBUG] switch_channel.c:2535
> (dingaling/xmppc/+12068235374 at voice.google.com) Callstate Change ACTIVE ->
> HANGUP
> 2011-03-05 16:00:42.649258 [NOTICE] mod_dingaling.c:731 Hangup
> dingaling/xmppc/+12068235374 at voice.google.com [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-05 16:00:42.649258 [DEBUG] switch_channel.c:2551 Send signal
> dingaling/xmppc/+12068235374 at voice.google.com [KILL]
> 2011-03-05 16:00:42.649258 [DEBUG] mod_dingaling.c:1348
> dingaling/xmppc/+12068235374 at voice.google.com CHANNEL
> KILL
> 2011-03-05 16:00:42.649258 [DEBUG] switch_core_session.c:1116 Send signal
> dingaling/xmppc/+12068235374 at voice.google.com [BREAK]
> 2011-03-05 16:00:42.649258 [DEBUG] mod_dingaling.c:1348
> dingaling/xmppc/+12068235374 at voice.google.com CHANNEL
> KILL
> 2011-03-05 16:00:42.649258 [DEBUG] mod_dingaling.c:3413 End
> Call
> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:323
> ==============================================================================
> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:335 Fax processing not
> successful - result (49) The call dropped prematurely.
> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:340 Remote station
> id:
> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:341 Local station id:
> SpanDSP Fax Ident
> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:342 Pages transferred:
> 0
> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:344 Total fax pages:
> 0
> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:345 Image resolution:
> 0x0
> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:346 Transfer Rate:
> 14400
> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:348 ECM status
> off
> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:349 remote
> country:
> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:350 remote
> vendor:
> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:351 remote
> model:
> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:353
> ==============================================================================
> 2011-03-05 16:00:42.658379 [DEBUG] switch_core_codec.c:141
> dingaling/xmppc/+12068235374 at voice.google.com Restore previous codec
> PCMU:0.
> 2011-03-05 16:00:42.658379 [DEBUG] switch_core_session.c:2045
> dingaling/xmppc/+12068235374 at voice.google.com skip receive message
> [APPLICATION_EXEC_COMPLETE] (channel is hungup
> already)
> 2011-03-05 16:00:42.658379 [DEBUG] switch_core_state_machine.c:366
> (dingaling/xmppc/+12068235374 at voice.google.com) State EXECUTE going to
> sleep
> 2011-03-05 16:00:42.658379 [DEBUG] switch_core_state_machine.c:320
> (dingaling/xmppc/+12068235374 at voice.google.com) Running State Change
> CS_HANGUP
> 2011-03-05 16:00:42.659331 [DEBUG] switch_core_state_machine.c:557
> (dingaling/xmppc/+12068235374 at voice.google.com) State HANGUP
> 2011-03-05 16:00:42.659331 [DEBUG] mod_dingaling.c:1317
> dingaling/xmppc/+12068235374 at voice.google.com CHANNEL HANGUP
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From xyangni at gmail.com  Sun Mar  6 05:40:13 2011
From: xyangni at gmail.com (Yihui Li)
Date: Sun, 6 Mar 2011 02:40:13 +0000
Subject: [Freeswitch-users] Voice quality of bridged call comparing to
	direct connect
Message-ID: <AANLkTi=HmP1aGPGbQ33BpctXv_JcvBzMZ9Banp80fZoO@mail.gmail.com>

Dear all,

I have a accounts with sipgate.co.uk. When I connect directly form X-lite to
sipgate with only PCMA codec. The voice quality is good.
Recently I set a FS server on Linode VPS and use FS to bridge call from
sipgate to X-lite on my computer, The voice quality become much worse with
lots of noise.
If the call is bridged through skypopen to my skype account, the quality is
much better.
If the noise is cause by my network speed or PCMA codec, how can sipgate's
server avoid it?  Is there any way to reduce noise of the bridged call?
Thanks.
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From curriegrad2004 at gmail.com  Sun Mar  6 06:11:49 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 5 Mar 2011 19:11:49 -0800
Subject: [Freeswitch-users] Pastebin
In-Reply-To: <AANLkTinqz8-e-UuD6qYtAJQUM4Nx+LAvykOzLw8fFCmy@mail.gmail.com>
References: <AANLkTimWhdnqbf4Oa6jVCTg4SNauv-Sz1rFQ4cifBUfx@mail.gmail.com>
	<1299344219611-6092170.post@n2.nabble.com>
	<AANLkTik60rneBM41RM++roemQ9_GJe2DznTCzOyLJaJw@mail.gmail.com>
	<AANLkTimwzc8E-47pJr+zdRrYHwgaQzCw8gtYJTXyRn2h@mail.gmail.com>
	<AANLkTinqz8-e-UuD6qYtAJQUM4Nx+LAvykOzLw8fFCmy@mail.gmail.com>
Message-ID: <AANLkTimELT_+0oPDiw5=_we=XAzDrD18FTuyD2D4kOHQ@mail.gmail.com>

Or maybe a lack of coffee is occuring

On Sat, Mar 5, 2011 at 11:39 AM, David Villasmil
<david.villasmil.work at gmail.com> wrote:
> Maybe i'm just having a dumb year :(
>
> Thanks!
>
> On Sat, Mar 5, 2011 at 8:17 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> "the login and password is pastebin/freeswitch"
>>
>> What's confusing there?
>>
>> -Steve
>>
>>
>> On 5 March 2011 18:11, David Villasmil <david.villasmil.work at gmail.com>
>> wrote:
>> > Hello,
>> > I read very carefully... I'm in.
>> > The wording is a little confusing, btw.
>> > Thanks!
>> > David
>> >
>> > On Sat, Mar 5, 2011 at 5:56 PM, mazilo <Nabble at slickdeals.endjunk.com>
>> > wrote:
>> >>
>> >> David Villasmil wrote:
>> >> > Why is it that even though I'm logged into freeswitch.org, when I
>> >> > click
>> >> > on
>> >> > the pastebin link, it asks for user/pass and doesn't recognize my
>> >> > credentials? it's very annoying.
>> >> The user/pass requirement is to ward out any spam bots. Without the
>> >> user/pass, it will become an annoyance, if not a nightmare, to the mods
>> >> to
>> >> clean out the spam posts by spam bots.
>> >>
>> >> -----
>> >> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
>> >> --
>> >> View this message in context:
>> >>
>> >> http://freeswitch-users.2379917.n2.nabble.com/Pastebin-tp6092153p6092170.html
>> >> Sent from the freeswitch-users mailing list archive at Nabble.com.
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From sunwood360 at gmail.com  Sun Mar  6 06:21:36 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Sat, 5 Mar 2011 19:21:36 -0800
Subject: [Freeswitch-users] send fax via google voice.
In-Reply-To: <AANLkTimDfszKNxgugnSECrtCE+ByKgYV-sjK0tjXppfK@mail.gmail.com>
References: <AANLkTin4uo+oKr_Nzw6qVzf72E7n896j_5xyoB6nFcnk@mail.gmail.com>
	<AANLkTimDfszKNxgugnSECrtCE+ByKgYV-sjK0tjXppfK@mail.gmail.com>
Message-ID: <AANLkTi=bZ0geQnP5BT37D7udeSPXmzwe=nN5FuDFgbiP@mail.gmail.com>

Why is PCMU codec not reliable?
 On Mar 5, 2011 6:34 PM, "curriegrad2004" <curriegrad2004 at gmail.com> wrote:
> I'd personally not go there if you need to use GV to fax. Faxing with
> uLaw isn't reliable at all. You're just playing with fire if you do
> decide to use uLaw to fax stuff. Besides, google voice's quality is
> just horrible anyways.
>
> On Sat, Mar 5, 2011 at 4:08 PM, envelopes envelopes
> <sunwood360 at gmail.com> wrote:
>> Google Voice integration with FS works fine. but sending fax seems not
>> working.  12068235374 is a K7 number, If I dial it , I can hear fax
machine
>> answering sound.
>> In fact, people has used GV to send fax via fax machine.
>> (http://www.magicjacksupport.com/faxing-via-google-voice-t7412.html)
>> Is this problem related with mod_dingaling?
>>
>> please take a look!
>>
>> thanks
>>
>>
>>
>> itch at internal> originate dingaling/xmppc/+12068235374 at voice.google.com
>> &txfax(/tmp/test.tiff)
>> +OK c20e3fc4-4784-11e0-8d1d-eda21484fd55
>>
>> 2011-03-05 16:00:33.808039 [NOTICE] switch_channel.c:808 New Channel
>> dingaling/xmppc/+12068235374 at voice.google.com
>> [c20e3fc4-4784-11e0-8d1d-eda21484fd55]
>> freeswitch at internal> 2011-03-05 16:00:33.809595 [DEBUG]
mod_dingaling.c:1824
>> (dingaling/xmppc/+12068235374 at voice.google.com) State Change CS_NEW ->
>> CS_INIT
>> 2011-03-05 16:00:33.809595 [DEBUG] switch_core_session.c:1116 Send signal
>> dingaling/xmppc/+12068235374 at voice.google.com [BREAK]
>> 2011-03-05 16:00:33.809595 [DEBUG] mod_dingaling.c:1348
>> dingaling/xmppc/+12068235374 at voice.google.com CHANNEL
>> KILL
>> 2011-03-05 16:00:33.810718 [DEBUG] switch_core_state_machine.c:320
>> (dingaling/xmppc/+12068235374 at voice.google.com) Running State Change
>> CS_INIT
>> 2011-03-05 16:00:33.810718 [DEBUG] switch_core_state_machine.c:356
>> (dingaling/xmppc/+12068235374 at voice.google.com) State INIT
>> 2011-03-05 16:00:33.810718 [NOTICE] mod_dingaling.c:1110 Ring-Ready
>> dingaling/xmppc/+12068235374 at voice.google.com!
>> 2011-03-05 16:00:33.811202 [DEBUG] mod_dingaling.c:1063 Don't have my
codec
>> yet here's one
>> 2011-03-05 16:00:33.811202 [DEBUG] mod_dingaling.c:1083 Send Describe
>> [PCMU at 8000]
>> 2011-03-05 16:00:34.000304 [DEBUG] mod_dingaling.c:2941 using Existing
>> session for 6100480476
>> 2011-03-05 16:00:34.000304 [DEBUG] mod_dingaling.c:1083 Send Describe
>> [PCMU at 8000]
>> 2011-03-05 16:00:34.311385 [DEBUG] mod_dingaling.c:2941 using Existing
>> session for 6100480476
>> 2011-03-05 16:00:34.311385 [DEBUG] mod_dingaling.c:1008 Send Candidate
>> 192.168.1.146:17960 [9CsLT8VWsB1yDqnv]
>> 2011-03-05 16:00:34.707433 [DEBUG] mod_dingaling.c:2941 using Existing
>> session for 6100480476
>> 2011-03-05 16:00:34.707433 [DEBUG] mod_dingaling.c:3279 3
>> candidates
>> 2011-03-05 16:00:34.707433 [DEBUG] mod_dingaling.c:3299 candidate
>> 74.125.53.126:19295 PASS ACL wan.auto
>> 2011-03-05 16:00:34.707433 [DEBUG] mod_dingaling.c:3351 Acceptable
Candidate
>> 74.125.53.126:19295
>> 2011-03-05 16:00:40.334509 [DEBUG] mod_dingaling.c:2941 using Existing
>> session for 6100480476
>> 2011-03-05 16:00:40.334509 [DEBUG] mod_dingaling.c:3193 Already decided
on a
>> codec
>> 2011-03-05 16:00:40.336590 [DEBUG] mod_dingaling.c:865 Set Read Codec to
>> PCMU at 8000
>> 2011-03-05 16:00:40.336590 [DEBUG] mod_dingaling.c:880 Set Write Codec to
>> PCMU at 8000
>> 2011-03-05 16:00:40.337483 [DEBUG] mod_dingaling.c:892 SETUP RTP
>> 192.168.1.146:17960 -> 74.125.53.126:19295
>> 2011-03-05 16:00:40.337483 [DEBUG] switch_rtp.c:1429 Starting timer
[soft]
>> 160 bytes per 20ms
>> 2011-03-05 16:00:40.339652 [DEBUG] switch_rtp.c:3775 Activate VAD codec
PCMU
>> 20ms
>> 2011-03-05 16:00:40.340311 [DEBUG] switch_channel.c:2782
>> (dingaling/xmppc/+12068235374 at voice.google.com) Callstate Change DOWN ->
>> ACTIVE
>> 2011-03-05 16:00:40.340311 [NOTICE] mod_dingaling.c:1202 Channel
>> [dingaling/xmppc/+12068235374 at voice.google.com] has been answered
>> 2011-03-05 16:00:40.340311 [DEBUG] mod_dingaling.c:1205
>> (dingaling/xmppc/+12068235374 at voice.google.com) State Change CS_INIT ->
>> CS_ROUTING
>> 2011-03-05 16:00:40.340311 [DEBUG] switch_core_session.c:1116 Send signal
>> dingaling/xmppc/+12068235374 at voice.google.com [BREAK]
>> 2011-03-05 16:00:40.340311 [DEBUG] mod_dingaling.c:1348
>> dingaling/xmppc/+12068235374 at voice.google.com CHANNEL
>> KILL
>> 2011-03-05 16:00:40.340311 [DEBUG] switch_core_state_machine.c:356
>> (dingaling/xmppc/+12068235374 at voice.google.com) State INIT going to sleep
>> 2011-03-05 16:00:40.340311 [DEBUG] switch_core_state_machine.c:320
>> (dingaling/xmppc/+12068235374 at voice.google.com) Running State Change
>> CS_ROUTING
>> 2011-03-05 16:00:40.340311 [DEBUG] switch_channel.c:1657
>> (dingaling/xmppc/+12068235374 at voice.google.com) Callstate Change ACTIVE
->
>> RINGING
>> 2011-03-05 16:00:40.340311 [DEBUG] switch_core_state_machine.c:359
>> (dingaling/xmppc/+12068235374 at voice.google.com) State ROUTING
>> 2011-03-05 16:00:40.340311 [DEBUG] mod_dingaling.c:1219
>> dingaling/xmppc/+12068235374 at voice.google.com CHANNEL
>> ROUTING
>> 2011-03-05 16:00:40.340311 [DEBUG] switch_ivr_originate.c:66
>> (dingaling/xmppc/+12068235374 at voice.google.com) State Change CS_ROUTING
->
>> CS_CONSUME_MEDIA
>> 2011-03-05 16:00:40.340311 [DEBUG] switch_core_session.c:1116 Send signal
>> dingaling/xmppc/+12068235374 at voice.google.com [BREAK]
>> 2011-03-05 16:00:40.340311 [DEBUG] switch_ivr_originate.c:3388 Originate
>> Resulted in Success:
>> [dingaling/xmppc/+12068235374 at voice.google.com]
>> 2011-03-05 16:00:40.340311 [DEBUG] mod_dingaling.c:1348
>> dingaling/xmppc/+12068235374 at voice.google.com CHANNEL
>> KILL
>> 2011-03-05 16:00:40.340311 [DEBUG] mod_commands.c:3194
>> (dingaling/xmppc/+12068235374 at voice.google.com) State Change
>> CS_CONSUME_MEDIA ->
>> CS_EXECUTE
>> 2011-03-05 16:00:40.340311 [DEBUG] switch_core_session.c:1116 Send signal
>> dingaling/xmppc/+12068235374 at voice.google.com [BREAK]
>> 2011-03-05 16:00:40.340311 [DEBUG] mod_dingaling.c:1348
>> dingaling/xmppc/+12068235374 at voice.google.com CHANNEL
>> KILL
>> 2011-03-05 16:00:40.340311 [DEBUG] switch_core_state_machine.c:359
>> (dingaling/xmppc/+12068235374 at voice.google.com) State ROUTING going to
>> sleep
>> 2011-03-05 16:00:40.340311 [DEBUG] switch_core_state_machine.c:320
>> (dingaling/xmppc/+12068235374 at voice.google.com) Running State Change
>> CS_EXECUTE
>> 2011-03-05 16:00:40.340311 [DEBUG] switch_channel.c:1659
>> (dingaling/xmppc/+12068235374 at voice.google.com) Callstate Change RINGING
->
>> ACTIVE
>> 2011-03-05 16:00:40.342729 [DEBUG] switch_core_state_machine.c:366
>> (dingaling/xmppc/+12068235374 at voice.google.com) State EXECUTE
>> 2011-03-05 16:00:40.342729 [DEBUG] mod_dingaling.c:1236
>> dingaling/xmppc/+12068235374 at voice.google.com CHANNEL
>> EXECUTE
>> 2011-03-05 16:00:40.342729 [DEBUG] switch_core_state_machine.c:157
>> dingaling/xmppc/+12068235374 at voice.google.com Standard EXECUTE
>> EXECUTE dingaling/xmppc/+12068235374 at voice.google.com
>> txfax(/tmp/test.tiff)
>> 2011-03-05 16:00:40.344059 [DEBUG] mod_spandsp_fax.c:1092 Raw read codec
>> activation Success L16 20000
>> 2011-03-05 16:00:40.344059 [DEBUG] switch_core_codec.c:116
>> dingaling/xmppc/+12068235374 at voice.google.com Push codec
>> L16:10
>> 2011-03-05 16:00:40.344059 [DEBUG] mod_spandsp_fax.c:1108 Raw write codec
>> activation Success L16
>> 2011-03-05 16:00:42.649258 [DEBUG] mod_dingaling.c:2941 using Existing
>> session for 6100480476
>> 2011-03-05 16:00:42.649258 [DEBUG] mod_dingaling.c:3411 hungup
>> dingaling/xmppc/+12068235374 at voice.google.com
>> 2011-03-05 16:00:42.649258 [DEBUG] mod_dingaling.c:710 Terminate called
from
>> line 3412 state=CS_EXECUTE
>> 2011-03-05 16:00:42.649258 [DEBUG] switch_channel.c:2535
>> (dingaling/xmppc/+12068235374 at voice.google.com) Callstate Change ACTIVE
->
>> HANGUP
>> 2011-03-05 16:00:42.649258 [NOTICE] mod_dingaling.c:731 Hangup
>> dingaling/xmppc/+12068235374 at voice.google.com [CS_EXECUTE]
[NORMAL_CLEARING]
>> 2011-03-05 16:00:42.649258 [DEBUG] switch_channel.c:2551 Send signal
>> dingaling/xmppc/+12068235374 at voice.google.com [KILL]
>> 2011-03-05 16:00:42.649258 [DEBUG] mod_dingaling.c:1348
>> dingaling/xmppc/+12068235374 at voice.google.com CHANNEL
>> KILL
>> 2011-03-05 16:00:42.649258 [DEBUG] switch_core_session.c:1116 Send signal
>> dingaling/xmppc/+12068235374 at voice.google.com [BREAK]
>> 2011-03-05 16:00:42.649258 [DEBUG] mod_dingaling.c:1348
>> dingaling/xmppc/+12068235374 at voice.google.com CHANNEL
>> KILL
>> 2011-03-05 16:00:42.649258 [DEBUG] mod_dingaling.c:3413 End
>> Call
>> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:323
>>
==============================================================================
>> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:335 Fax processing
not
>> successful - result (49) The call dropped prematurely.
>> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:340 Remote station
>> id:
>> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:341 Local station
id:
>> SpanDSP Fax Ident
>> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:342 Pages
transferred:
>> 0
>> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:344 Total fax pages:
>> 0
>> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:345 Image
resolution:
>> 0x0
>> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:346 Transfer Rate:
>> 14400
>> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:348 ECM status
>> off
>> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:349 remote
>> country:
>> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:350 remote
>> vendor:
>> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:351 remote
>> model:
>> 2011-03-05 16:00:42.658379 [DEBUG] mod_spandsp_fax.c:353
>>
==============================================================================
>> 2011-03-05 16:00:42.658379 [DEBUG] switch_core_codec.c:141
>> dingaling/xmppc/+12068235374 at voice.google.com Restore previous codec
>> PCMU:0.
>> 2011-03-05 16:00:42.658379 [DEBUG] switch_core_session.c:2045
>> dingaling/xmppc/+12068235374 at voice.google.com skip receive message
>> [APPLICATION_EXEC_COMPLETE] (channel is hungup
>> already)
>> 2011-03-05 16:00:42.658379 [DEBUG] switch_core_state_machine.c:366
>> (dingaling/xmppc/+12068235374 at voice.google.com) State EXECUTE going to
>> sleep
>> 2011-03-05 16:00:42.658379 [DEBUG] switch_core_state_machine.c:320
>> (dingaling/xmppc/+12068235374 at voice.google.com) Running State Change
>> CS_HANGUP
>> 2011-03-05 16:00:42.659331 [DEBUG] switch_core_state_machine.c:557
>> (dingaling/xmppc/+12068235374 at voice.google.com) State HANGUP
>> 2011-03-05 16:00:42.659331 [DEBUG] mod_dingaling.c:1317
>> dingaling/xmppc/+12068235374 at voice.google.com CHANNEL HANGUP
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From steveu at coppice.org  Sun Mar  6 06:29:14 2011
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 06 Mar 2011 11:29:14 +0800
Subject: [Freeswitch-users] send fax via google voice.
In-Reply-To: <AANLkTi=bZ0geQnP5BT37D7udeSPXmzwe=nN5FuDFgbiP@mail.gmail.com>
References: <AANLkTin4uo+oKr_Nzw6qVzf72E7n896j_5xyoB6nFcnk@mail.gmail.com>	<AANLkTimDfszKNxgugnSECrtCE+ByKgYV-sjK0tjXppfK@mail.gmail.com>
	<AANLkTi=bZ0geQnP5BT37D7udeSPXmzwe=nN5FuDFgbiP@mail.gmail.com>
Message-ID: <4D72FF8A.4080209@coppice.org>

On 03/06/2011 11:21 AM, envelopes envelopes wrote:
>
> Why is PCMU codec not reliable?
>
>
Unsynchronised clocks, and packet loss.

Steve



From mitch.johnson7 at gmail.com  Sun Mar  6 08:15:13 2011
From: mitch.johnson7 at gmail.com (Mitch Johnson)
Date: Sun, 6 Mar 2011 00:15:13 -0500
Subject: [Freeswitch-users] tls/srtp goes straight to voicemail
Message-ID: <2160E963-E7A7-48D3-A715-DD3C02F06FB0@gmail.com>

I'm trying to setup a secure phone call across two eyebeam clients running on Apple iPhones.

I am pretty new to Freeswitch.

The problem is that when I make a call using SRTP between the two phones it goes straight to voicemail.  I took some snippets of the debug.  Any help would be greatly appreciated.

2011-03-05 23:21:08.502797 [DEBUG] sofia.c:6451 IP 192.168.60.130 Rejected by acl "domains". Falling back to Digest auth.
2011-03-05 23:21:08.502797 [WARNING] sofia_reg.c:1246 SIP auth challenge (INVITE) on sofia profile 'internal' for [1005 at 172.16.200.60] from ip 192.168.60.130
2011-03-05 23:21:08.527274 [DEBUG] sofia.c:6451 IP 192.168.60.130 Rejected by acl "domains". Falling back to Digest auth.
2011-03-05 23:21:08.527274 [NOTICE] switch_channel.c:812 New Channel sofia/internal/1006 at 172.16.200.60 [d9976fea-1e9e-4d77-8c26-fa3bd32c4979]
2011-03-05 23:21:08.527274 [DEBUG] sofia.c:4683 Channel sofia/internal/1006 at 172.16.200.60 entering state [received][100]
2011-03-05 23:21:08.527274 [DEBUG] sofia.c:4694 Remote SDP:
v=0
o=- 3508374466 3508374466 IN IP4 192.168.60.130
s=cpc_med
c=IN IP4 192.168.60.130
t=0 0
a=X-nat:0
m=audio 4004 RTP/SAVP 0 8 104 3 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:znq1mGYq9Dz+5G2GY3tKNJ0gTMGfvBNeeIYKuIKZ
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:iRs0wouqlzmT6T06/K3CS9HUJmwnwkP+w1rhaDre

2011-03-05 23:21:08.527274 [DEBUG] sofia_glue.c:4415 Set Remote Key [1 AES_CM_128_HMAC_SHA1_80 inline:znq1mGYq9Dz+5G2GY3tKNJ0gTMGfvBNeeIYKuIKZ]
2011-03-05 23:21:08.527274 [DEBUG] sofia_glue.c:2821 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:zXMFICKxSVhKLofieTRxrWH5I35i/Bu6lkit2BbE]

......

2011-03-05 23:21:08.535483 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1006 at 172.16.200.60 Standard EXECUTE
EXECUTE sofia/internal/1006 at 172.16.200.60 set(sip_secure_media=true)
2011-03-05 23:21:08.535483 [DEBUG] mod_dptools.c:1059 sofia/internal/1006 at 172.16.200.60 SET [sip_secure_media]=[true]
EXECUTE sofia/internal/1006 at 172.16.200.60 export(sip_secure_media=true)
2011-03-05 23:21:08.536478 [DEBUG] switch_channel.c:961 EXPORT (export_vars) [sip_secure_media]=[true]
EXECUTE sofia/internal/1006 at 172.16.200.60 hash(insert/172.16.200.60-spymap/1006/d9976fea-1e9e-4d77-8c26-fa3bd32c4979)
EXECUTE sofia/internal/1006 at 172.16.200.60 hash(insert/172.16.200.60-last_dial/1006/1005)
EXECUTE sofia/internal/1006 at 172.16.200.60 hash(insert/172.16.200.60-last_dial/global/d9976fea-1e9e-4d77-8c26-fa3bd32c4979)
EXECUTE sofia/internal/1006 at 172.16.200.60 set(RFC2822_DATE=Sat, 05 Mar 2011 23:21:08 -0500)
2011-03-05 23:21:08.537477 [DEBUG] mod_dptools.c:1059 sofia/internal/1006 at 172.16.200.60 SET [RFC2822_DATE]=[Sat, 05 Mar 2011 23:21:08 -0500]
EXECUTE sofia/internal/1006 at 172.16.200.60 set(dialed_extension=1005)
2011-03-05 23:21:08.538480 [DEBUG] mod_dptools.c:1059 sofia/internal/1006 at 172.16.200.60 SET [dialed_extension]=[1005]
EXECUTE sofia/internal/1006 at 172.16.200.60 export(dialed_extension=1005)
2011-03-05 23:21:08.538480 [DEBUG] switch_channel.c:961 EXPORT (export_vars) [dialed_extension]=[1005]
EXECUTE sofia/internal/1006 at 172.16.200.60 bind_meta_app(1 b s execute_extension::dx XML features)
2011-03-05 23:21:08.539474 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *1 execute_extension::dx XML features
EXECUTE sofia/internal/1006 at 172.16.200.60 bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/1006.2011-03-05-23-21-08.wav)

.....

EXECUTE sofia/internal/1006 at 172.16.200.60 hash(insert/172.16.200.60-last_dial/techsupport/d9976fea-1e9e-4d77-8c26-fa3bd32c4979)
EXECUTE sofia/internal/1006 at 172.16.200.60 bridge(user/1005 at 172.16.200.60)
2011-03-05 23:21:08.556511 [DEBUG] switch_channel.c:918 sofia/internal/1006 at 172.16.200.60 EXPORTING[export_vars] [sip_secure_media]=[true] to event
2011-03-05 23:21:08.556511 [DEBUG] switch_channel.c:918 sofia/internal/1006 at 172.16.200.60 EXPORTING[export_vars] [dialed_extension]=[1005] to event
2011-03-05 23:21:08.558529 [DEBUG] switch_channel.c:918 sofia/internal/1006 at 172.16.200.60 EXPORTING[export_vars] [sip_secure_media]=[true] to event
2011-03-05 23:21:08.558529 [DEBUG] switch_channel.c:918 sofia/internal/1006 at 172.16.200.60 EXPORTING[export_vars] [dialed_extension]=[1005] to event
2011-03-05 23:21:08.558529 [DEBUG] switch_ivr_originate.c:1971 variable string 0 = [sip_secure_media=false]
2011-03-05 23:21:08.558529 [DEBUG] switch_ivr_originate.c:1971 variable string 1 = [presence_id=1005 at 172.16.200.60]
2011-03-05 23:21:08.558529 [NOTICE] switch_channel.c:812 New Channel sofia/internal/sip:1005 at 192.168.60.135:53664 [aed041ea-4355-46aa-ad11-e58389f8e6d9]
2011-03-05 23:21:08.558529 [DEBUG] mod_sofia.c:4151 (sofia/internal/sip:1005 at 192.168.60.135:53664) State Change CS_NEW -> CS_INIT
2011-03-05 23:21:08.558529 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/sip:1005 at 192.168.60.135:53664 [BREAK]
2011-03-05 23:21:08.565722 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/sip:1005 at 192.168.60.135:53664) Running State Change CS_INIT
2011-03-05 23:21:08.566543 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/sip:1005 at 192.168.60.135:53664) State INIT
2011-03-05 23:21:08.566543 [DEBUG] mod_sofia.c:84 sofia/internal/sip:1005 at 192.168.60.135:53664 SOFIA INIT
2011-03-05 23:21:08.570553 [DEBUG] sofia_glue.c:2327 sip:1005 at 192.168.60.135:53710;transport=TLS Setting proxy route to sofia/internal/sip:1005 at 192.168.60.135:53664
2011-03-05 23:21:08.572076 [DEBUG] mod_sofia.c:124 (sofia/internal/sip:1005 at 192.168.60.135:53664) State Change CS_INIT -> CS_ROUTING
2011-03-05 23:21:08.572076 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/sip:1005 at 192.168.60.135:53664 [BREAK]
2011-03-05 23:21:08.572076 [DEBUG] sofia.c:4683 Channel sofia/internal/sip:1005 at 192.168.60.135:53664 entering state [calling][0]
2011-03-05 23:21:08.572076 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/sip:1005 at 192.168.60.135:53664) State INIT going to sleep
2011-03-05 23:21:08.572076 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/sip:1005 at 192.168.60.135:53664) Running State Change CS_ROUTING
2011-03-05 23:21:08.572076 [DEBUG] switch_channel.c:1664 (sofia/internal/sip:1005 at 192.168.60.135:53664) Callstate Change DOWN -> RINGING
2011-03-05 23:21:08.573709 [DEBUG] switch_core_state_machine.c:359 (sofia/internal/sip:1005 at 192.168.60.135:53664) State ROUTING
2011-03-05 23:21:08.573709 [DEBUG] mod_sofia.c:147 sofia/internal/sip:1005 at 192.168.60.135:53664 SOFIA ROUTING
2011-03-05 23:21:08.574575 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/sip:1005 at 192.168.60.135:53664) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2011-03-05 23:21:08.574575 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/sip:1005 at 192.168.60.135:53664 [BREAK]
2011-03-05 23:21:08.574575 [DEBUG] switch_core_state_machine.c:359 (sofia/internal/sip:1005 at 192.168.60.135:53664) State ROUTING going to sleep
2011-03-05 23:21:08.574575 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/sip:1005 at 192.168.60.135:53664) Running State Change CS_CONSUME_MEDIA
2011-03-05 23:21:08.574575 [DEBUG] switch_core_state_machine.c:378 (sofia/internal/sip:1005 at 192.168.60.135:53664) State CONSUME_MEDIA
2011-03-05 23:21:08.574575 [DEBUG] switch_core_state_machine.c:378 (sofia/internal/sip:1005 at 192.168.60.135:53664) State CONSUME_MEDIA going to sleep
2011-03-05 23:21:08.650486 [DEBUG] sofia.c:4683 Channel sofia/internal/sip:1005 at 192.168.60.135:53664 entering state [terminated][406]
2011-03-05 23:21:08.650486 [DEBUG] switch_channel.c:2545 (sofia/internal/sip:1005 at 192.168.60.135:53664) Callstate Change RINGING -> HANGUP
2011-03-05 23:21:08.650486 [NOTICE] sofia.c:5323 Hangup sofia/internal/sip:1005 at 192.168.60.135:53664 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
2011-03-05 23:21:08.650486 [DEBUG] switch_channel.c:2561 Send signal sofia/internal/sip:1005 at 192.168.60.135:53664 [KILL]
2011-03-05 23:21:08.650486 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/sip:1005 at 192.168.60.135:53664 [BREAK]
2011-03-05 23:21:08.652479 [DEBUG] switch_ivr_originate.c:3502 Originate Resulted in Error Cause: 79 [SERVICE_NOT_IMPLEMENTED]
2011-03-05 23:21:08.652479 [ERR] switch_ivr_originate.c:2638 Cannot create outgoing channel of type [user] cause: [SERVICE_NOT_IMPLEMENTED]
2011-03-05 23:21:08.652479 [DEBUG] switch_ivr_originate.c:3502 Originate Resulted in Error Cause: 79 [SERVICE_NOT_IMPLEMENTED]
2011-03-05 23:21:08.653662 [INFO] mod_dptools.c:2623 Originate Failed.  Cause: SERVICE_NOT_IMPLEMENTED
EXECUTE sofia/internal/1006 at 172.16.200.60 answer()
2011-03-05 23:21:08.655518 [DEBUG] sofia_glue.c:2990 AUDIO RTP [sofia/internal/1006 at 172.16.200.60] 172.16.200.60 port 22870 -> 192.168.60.130 port 4004 codec: 0 ms: 20
2011-03-05 23:21:08.656829 [DEBUG] switch_rtp.c:1621 Starting timer [soft] 160 bytes per 20ms
2011-03-05 23:21:08.658299 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/sip:1005 at 192.168.60.135:53664) Running State Change CS_HANGUP
2011-03-05 23:21:08.658299 [DEBUG] sofia_glue.c:3235 Set 2833 dtmf send payload to 96
2011-03-05 23:21:08.658299 [DEBUG] sofia_glue.c:3240 Set 2833 dtmf receive payload to 96
2011-03-05 23:21:08.658299 [INFO] switch_rtp.c:1450 Activating Secure RTP SEND
2011-03-05 23:21:08.659547 [DEBUG] switch_core_sqldb.c:1422 Secure Type: srtp:AES_CM_128_HMAC_SHA1_80
2011-03-05 23:21:08.659547 [INFO] switch_rtp.c:1430 Activating Secure RTP RECV
2011-03-05 23:21:08.659547 [DEBUG] switch_core_sqldb.c:1422 Secure Type: srtp:AES_CM_128_HMAC_SHA1_80
2011-03-05 23:21:08.659547 [DEBUG] mod_sofia.c:681 Local SDP sofia/internal/1006 at 172.16.200.60:
v=0
o=FreeSWITCH 1299362398 1299362399 IN IP4 172.16.200.60
s=FreeSWITCH
c=IN IP4 172.16.200.60
t=0 0
m=audio 22870 RTP/SAVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:zXMFICKxSVhKLofieTRxrWH5I35i/Bu6lkit2BbE

2011-03-05 23:21:08.661306 [DEBUG] sofia.c:4683 Channel sofia/internal/1006 at 172.16.200.60 entering state [completed][200]
2011-03-05 23:21:08.661306 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/1006 at 172.16.200.60 [BREAK]
2011-03-05 23:21:08.661306 [DEBUG] switch_channel.c:2795 (sofia/internal/1006 at 172.16.200.60) Callstate Change RINGING -> ACTIVE
2011-03-05 23:21:08.662543 [NOTICE] mod_dptools.c:929 Channel [sofia/internal/1006 at 172.16.200.60] has been answered
EXECUTE sofia/internal/1006 at 172.16.200.60 sleep(1000)
2011-03-05 23:21:08.664527 [DEBUG] switch_core_state_machine.c:560 (sofia/internal/sip:1005 at 192.168.60.135:53664) State HANGUP
2011-03-05 23:21:08.665590 [DEBUG] mod_sofia.c:451 sofia/internal/sip:1005 at 192.168.60.135:53664 Overriding SIP cause 501 with 406 from the other leg
2011-03-05 23:21:08.665590 [DEBUG] mod_sofia.c:457 Channel sofia/internal/sip:1005 at 192.168.60.135:53664 hanging up, cause: SERVICE_NOT_IMPLEMENTED
2011-03-05 23:21:08.666556 [DEBUG] switch_core_state_machine.c:46 sofia/internal/sip:1005 at 192.168.60.135:53664 Standard HANGUP, cause: SERVICE_NOT_IMPLEMENTED
2011-03-05 23:21:08.666556 [DEBUG] switch_core_state_machine.c:560 (sofia/internal/sip:1005 at 192.168.60.135:53664) State HANGUP going to sleep
2011-03-05 23:21:08.666556 [DEBUG] switch_core_state_machine.c:351 (sofia/internal/sip:1005 at 192.168.60.135:53664) State Change CS_HANGUP -> CS_REPORTING
2011-03-05 23:21:08.666556 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/sip:1005 at 192.168.60.135:53664 [BREAK]
2011-03-05 23:21:08.666556 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/sip:1005 at 192.168.60.135:53664) Running State Change CS_REPORTING
2011-03-05 23:21:08.667827 [DEBUG] switch_core_state_machine.c:620 (sofia/internal/sip:1005 at 192.168.60.135:53664) State REPORTING
2011-03-05 23:21:08.667827 [DEBUG] switch_core_state_machine.c:53 sofia/internal/sip:1005 at 192.168.60.135:53664 Standard REPORTING, cause: SERVICE_NOT_IMPLEMENTED
2011-03-05 23:21:08.667827 [DEBUG] switch_core_state_machine.c:620 (sofia/internal/sip:1005 at 192.168.60.135:53664) State REPORTING going to sleep
2011-03-05 23:21:08.669333 [DEBUG] switch_core_state_machine.c:345 (sofia/internal/sip:1005 at 192.168.60.135:53664) State Change CS_REPORTING -> CS_DESTROY
2011-03-05 23:21:08.669333 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/sip:1005 at 192.168.60.135:53664 [BREAK]
2011-03-05 23:21:08.669333 [DEBUG] switch_core_session.c:1288 Session 162 (sofia/internal/sip:1005 at 192.168.60.135:53664) Locked, Waiting on external entities
2011-03-05 23:21:08.669333 [NOTICE] switch_core_session.c:1306 Session 162 (sofia/internal/sip:1005 at 192.168.60.135:53664) Ended
2011-03-05 23:21:08.669333 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/sip:1005 at 192.168.60.135:53664 [CS_DESTROY]
2011-03-05 23:21:08.670544 [DEBUG] switch_core_state_machine.c:449 (sofia/internal/sip:1005 at 192.168.60.135:53664) Callstate Change HANGUP -> DOWN
2011-03-05 23:21:08.670544 [DEBUG] switch_core_state_machine.c:452 (sofia/internal/sip:1005 at 192.168.60.135:53664) Running State Change CS_DESTROY
2011-03-05 23:21:08.671619 [DEBUG] switch_core_state_machine.c:462 (sofia/internal/sip:1005 at 192.168.60.135:53664) State DESTROY
2011-03-05 23:21:08.671619 [DEBUG] mod_sofia.c:362 sofia/internal/sip:1005 at 192.168.60.135:53664 SOFIA DESTROY
2011-03-05 23:21:08.671619 [DEBUG] switch_core_state_machine.c:60 sofia/internal/sip:1005 at 192.168.60.135:53664 Standard DESTROY
2011-03-05 23:21:08.671619 [DEBUG] switch_core_state_machine.c:462 (sofia/internal/sip:1005 at 192.168.60.135:53664) State DESTROY going to sleep
2011-03-05 23:21:08.742675 [DEBUG] switch_rtp.c:2989 Correct ip/port confirmed.
2011-03-05 23:21:08.742675 [DEBUG] sofia.c:4683 Channel sofia/internal/1006 at 172.16.200.60 entering state [ready][200]
EXECUTE sofia/internal/1006 at 172.16.200.60 bridge(loopback/app=voicemail:default 172.16.200.60 1005)
2011-03-05 23:21:09.682417 [DEBUG] switch_channel.c:918 sofia/internal/1006 at 172.16.200.60 EXPORTING[export_vars] [sip_secure_media]=[true] to event
2011-03-05 23:21:09.682417 [DEBUG] switch_channel.c:918 sofia/internal/1006 at 172.16.200.60 EXPORTING[export_vars] [dialed_extension]=[1005] to event
2011-03-05 23:21:09.682417 [NOTICE] switch_channel.c:812 New Channel loopback/app=voicemail:default 172.16.200.60 1005-a [f2016cad-3669-4b54-95c2-7f3a958ddca0]
2011-03-05 23:21:09.682417 [DEBUG] mod_loopback.c:131 loopback/app=voicemail:default 172.16.200.60 1005-a setup codec PCMU/8000/20
2011-03-05 23:21:09.682417 [NOTICE] switch_channel.c:810 Rename Channel loopback/app=voicemail:default 172.16.200.60 1005-a->loopback/voicemail-a [f2016cad-3669-4b54-95c2-7f3a958ddca0]
2011-03-05 23:21:09.682417 [DEBUG] mod_loopback.c:939 (loopback/voicemail-a) State Change CS_NEW -> CS_INIT
2011-03-05 23:21:09.682417 [DEBUG] switch_core_session.c:1116 Send signal loopback/voicemail-a [BREAK]
2011-03-05 23:21:09.682417 [DEBUG] mod_loopback.c:469 loopback/voicemail-a CHANNEL KILL
2011-03-05 23:21:09.687524 [DEBUG] switch_core_state_machine.c:320 (loopback/voicemail-a) Running State Change CS_INIT
2011-03-05 23:21:09.687524 [DEBUG] switch_core_state_machine.c:356 (loopback/voicemail-a) State INIT
2011-03-05 23:21:09.687524 [NOTICE] switch_channel.c:812 New Channel loopback/voicemail-b [23eaf856-1bf2-4a84-94cc-e71178199468]
2011-03-05 23:21:09.688656 [DEBUG] mod_loopback.c:131 loopback/voicemail-b setup codec PCMU/8000/20
2011-03-05 23:21:09.689557 [DEBUG] mod_loopback.c:244 (loopback/voicemail-b) State Change CS_NEW -> CS_INIT
2011-03-05 23:21:09.689557 [DEBUG] switch_core_session.c:1116 Send signal loopback/voicemail-b [BREAK]
2011-03-05 23:21:09.689557 [DEBUG] mod_loopback.c:469 loopback/voicemail-b CHANNEL KILL
2011-03-05 23:21:09.690598 [DEBUG] mod_loopback.c:290 (loopback/voicemail-a) State Change CS_INIT -> CS_ROUTING
2011-03-05 23:21:09.690598 [DEBUG] switch_core_session.c:1116 Send signal loopback/voicemail-a [BREAK]
2011-03-05 23:21:09.690598 [DEBUG] mod_loopback.c:469 loopback/voicemail-a CHANNEL KILL
2011-03-05 23:21:09.690598 [DEBUG] switch_core_state_machine.c:356 (loopback/voicemail-a) State INIT going to sleep
2011-03-05 23:21:09.690598 [DEBUG] switch_core_state_machine.c:320 (loopback/voicemail-a) Running State Change CS_ROUTING
2011-03-05 23:21:09.690598 [DEBUG] switch_channel.c:1664 (loopback/voicemail-a) Callstate Change DOWN -> RINGING
2011-03-05 23:21:09.691794 [DEBUG] switch_core_state_machine.c:359 (loopback/voicemail-a) State ROUTING
2011-03-05 23:21:09.691794 [DEBUG] mod_loopback.c:324 loopback/voicemail-a CHANNEL ROUTING
2011-03-05 23:21:09.691794 [DEBUG] switch_ivr_originate.c:66 (loopback/voicemail-a) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2011-03-05 23:21:09.691794 [DEBUG] switch_core_session.c:1116 Send signal loopback/voicemail-a [BREAK]
2011-03-05 23:21:09.691794 [DEBUG] mod_loopback.c:469 loopback/voicemail-a CHANNEL KILL
2011-03-05 23:21:09.691794 [DEBUG] switch_core_state_machine.c:359 (loopback/voicemail-a) State ROUTING going to sleep
2011-03-05 23:21:09.691794 [DEBUG] switch_core_state_machine.c:320 (loopback/voicemail-a) Running State Change CS_CONSUME_MEDIA
2011-03-05 23:21:09.691794 [DEBUG] switch_core_state_machine.c:378 (loopback/voicemail-a) State CONSUME_MEDIA
2011-03-05 23:21:09.691794 [DEBUG] mod_loopback.c:529 CHANNEL CONSUME_MEDIA
2011-03-05 23:21:09.693484 [DEBUG] switch_core_state_machine.c:378 (loopback/voicemail-a) State CONSUME_MEDIA going to sleep
2011-03-05 23:21:09.693484 [DEBUG] switch_core_state_machine.c:320 (loopback/voicemail-b) Running State Change CS_INIT
2011-03-05 23:21:09.693484 [DEBUG] switch_core_state_machine.c:356 (loopback/voicemail-b) State INIT
2011-03-05 23:21:09.693484 [DEBUG] mod_loopback.c:290 (loopback/voicemail-b) State Change CS_INIT -> CS_ROUTING
2011-03-05 23:21:09.693484 [DEBUG] switch_core_session.c:1116 Send signal loopback/voicemail-b [BREAK]
2011-03-05 23:21:09.693484 [DEBUG] mod_loopback.c:469 loopback/voicemail-b CHANNEL KILL
2011-03-05 23:21:09.693484 [DEBUG] switch_core_state_machine.c:356 (loopback/voicemail-b) State INIT going to sleep
2011-03-05 23:21:09.693484 [DEBUG] switch_core_state_machine.c:320 (loopback/voicemail-b) Running State Change CS_ROUTING
2011-03-05 23:21:09.693484 [DEBUG] switch_channel.c:1664 (loopback/voicemail-b) Callstate Change DOWN -> RINGING
2011-03-05 23:21:09.695557 [DEBUG] switch_core_state_machine.c:359 (loopback/voicemail-b) State ROUTING
2011-03-05 23:21:09.695557 [DEBUG] mod_loopback.c:324 loopback/voicemail-b CHANNEL ROUTING
2011-03-05 23:21:09.695557 [DEBUG] mod_loopback.c:343 (loopback/voicemail-b) State Change CS_ROUTING -> CS_EXECUTE
2011-03-05 23:21:09.695557 [DEBUG] switch_core_session.c:1116 Send signal loopback/voicemail-b [BREAK]
2011-03-05 23:21:09.695557 [DEBUG] mod_loopback.c:469 loopback/voicemail-b CHANNEL KILL
2011-03-05 23:21:09.695557 [DEBUG] switch_core_state_machine.c:359 (loopback/voicemail-b) State ROUTING going to sleep
2011-03-05 23:21:09.695557 [DEBUG] switch_core_state_machine.c:320 (loopback/voicemail-b) Running State Change CS_EXECUTE
2011-03-05 23:21:09.695557 [DEBUG] switch_core_state_machine.c:366 (loopback/voicemail-b) State EXECUTE
2011-03-05 23:21:09.695557 [DEBUG] mod_loopback.c:363 loopback/voicemail-b CHANNEL EXECUTE
2011-03-05 23:21:09.696643 [DEBUG] switch_core_state_machine.c:157 loopback/voicemail-b Standard EXECUTE
EXECUTE loopback/voicemail-b pre_answer()
2011-03-05 23:21:09.696643 [NOTICE] mod_loopback.c:722 Pre-Answer loopback/voicemail-a!
2011-03-05 23:21:09.696643 [DEBUG] switch_channel.c:2638 (loopback/voicemail-a) Callstate Change RINGING -> EARLY
2011-03-05 23:21:09.697671 [DEBUG] switch_channel.c:2689 Send signal sofia/internal/1006 at 172.16.200.60 [BREAK]
2011-03-05 23:21:09.698642 [DEBUG] switch_core_session.c:709 Send signal loopback/voicemail-b [BREAK]
2011-03-05 23:21:09.698642 [DEBUG] mod_loopback.c:469 loopback/voicemail-b CHANNEL KILL
2011-03-05 23:21:09.698642 [NOTICE] mod_dptools.c:955 Pre-Answer loopback/voicemail-b!
2011-03-05 23:21:09.698642 [DEBUG] switch_channel.c:2638 (loopback/voicemail-b) Callstate Change RINGING -> EARLY
2011-03-05 23:21:09.700393 [DEBUG] switch_channel.c:2689 Send signal sofia/internal/1006 at 172.16.200.60 [BREAK]
EXECUTE loopback/voicemail-b voicemail(default 172.16.200.60 1005)
2011-03-05 23:21:09.701645 [DEBUG] switch_ivr_originate.c:3408 Originate Resulted in Success: [loopback/voicemail-a]
2011-03-05 23:21:09.701645 [DEBUG] switch_core_session.c:709 Send signal loopback/voicemail-a [BREAK]
2011-03-05 23:21:09.701645 [DEBUG] mod_loopback.c:469 loopback/voicemail-a CHANNEL KILL
2011-03-05 23:21:09.701645 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/1006 at 172.16.200.60 [BREAK]
2011-03-05 23:21:09.701645 [DEBUG] switch_ivr_bridge.c:1234 (loopback/voicemail-a) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2011-03-05 23:21:09.701645 [DEBUG] switch_core_session.c:1116 Send signal loopback/voicemail-a [BREAK]
2011-03-05 23:21:09.701645 [DEBUG] mod_loopback.c:469 loopback/voicemail-a CHANNEL KILL
2011-03-05 23:21:09.703337 [DEBUG] switch_core_state_machine.c:320 (loopback/voicemail-a) Running State Change CS_EXCHANGE_MEDIA
2011-03-05 23:21:09.703337 [DEBUG] switch_core_state_machine.c:369 (loopback/voicemail-a) State EXCHANGE_MEDIA
2011-03-05 23:21:09.703337 [DEBUG] mod_loopback.c:491 CHANNEL LOOPBACK
2011-03-05 23:21:09.862580 [DEBUG] switch_ivr_play_say.c:63 No language specified - Using [en]
2011-03-05 23:21:09.874639 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-person.wav] (en:en)
2011-03-05 23:21:09.875636 [DEBUG] switch_ivr_play_say.c:1291 Codec Activated L16 at 8000hz 1 channels 20ms
2011-03-05 23:21:11.262410 [DEBUG] switch_ivr_play_say.c:1635 done playing file
2011-03-05 23:21:11.362431 [DEBUG] switch_ivr_play_say.c:244 Handle say:[1005] (en:en)
2011-03-05 23:21:11.362431 [DEBUG] switch_ivr_play_say.c:1291 Codec Activated L16 at 8000hz 1 channels 20ms
2011-03-05 23:21:11.370005 [DEBUG] switch_channel.c:2545 (sofia/internal/1006 at 172.16.200.60) Callstate Change ACTIVE -> HANGUP
2011-03-05 23:21:11.370005 [NOTICE] sofia.c:537 Hangup sofia/internal/1006 at 172.16.200.60 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-05 23:21:11.370005 [DEBUG] switch_channel.c:2561 Send signal sofia/internal/1006 at 172.16.200.60 [KILL]
2011-03-05 23:21:11.370005 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1006 at 172.16.200.60 [BREAK]
2011-03-05 23:21:11.381511 [DEBUG] switch_ivr_bridge.c:500 sofia/internal/1006 at 172.16.200.60 ending bridge by request from read function
2011-03-05 23:21:11.381511 [DEBUG] switch_ivr_bridge.c:581 BRIDGE THREAD DONE [sofia/internal/1006 at 172.16.200.60]
2011-03-05 23:21:11.381511 [DEBUG] switch_ivr_bridge.c:601 Send signal loopback/voicemail-a [BREAK]
2011-03-05 23:21:11.381511 [DEBUG] mod_loopback.c:469 loopback/voicemail-a CHANNEL KILL
2011-03-05 23:21:11.381511 [DEBUG] switch_ivr_bridge.c:581 BRIDGE THREAD DONE [loopback/voicemail-a]
2011-03-05 23:21:11.381511 [DEBUG] switch_ivr_bridge.c:601 Send signal sofia/internal/1006 at 172.16.200.60 [BREAK]
2011-03-05 23:21:11.381511 [DEBUG] switch_channel.c:2545 (loopback/voicemail-a) Callstate Change EARLY -> HANGUP
2011-03-05 23:21:11.381511 [NOTICE] switch_ivr_bridge.c:653 Hangup loopback/voicemail-a [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2011-03-05 23:21:11.381511 [DEBUG] switch_channel.c:2561 Send signal loopback/voicemail-a [KILL]
2011-03-05 23:21:11.381511 [DEBUG] mod_loopback.c:469 loopback/voicemail-a CHANNEL KILL
2011-03-05 23:21:11.381511 [DEBUG] switch_core_session.c:1116 Send signal loopback/voicemail-a [BREAK]
2011-03-05 23:21:11.381511 [DEBUG] mod_loopback.c:469 loopback/voicemail-a CHANNEL KILL
2011-03-05 23:21:11.381511 [DEBUG] switch_core_state_machine.c:369 (loopback/voicemail-a) State EXCHANGE_MEDIA going to sleep
2011-03-05 23:21:11.381511 [DEBUG] switch_core_state_machine.c:320 (loopback/voicemail-a) Running State Change CS_HANGUP
2011-03-05 23:21:11.381511 [DEBUG] switch_core_state_machine.c:560 (loopback/voicemail-a) State HANGUP
2011-03-05 23:21:11.381511 [DEBUG] mod_loopback.c:415 loopback/voicemail-a CHANNEL HANGUP
2011-03-05 23:21:11.381511 [DEBUG] switch_channel.c:2545 (loopback/voicemail-b) Callstate Change EARLY -> HANGUP
2011-03-05 23:21:11.381511 [NOTICE] mod_loopback.c:426 Hangup loopback/voicemail-b [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-05 23:21:11.381511 [DEBUG] switch_channel.c:2561 Send signal loopback/voicemail-b [KILL]
2011-03-05 23:21:11.381511 [DEBUG] mod_loopback.c:469 loopback/voicemail-b CHANNEL KILL
2011-03-05 23:21:11.381511 [DEBUG] switch_core_session.c:1116 Send signal loopback/voicemail-b [BREAK]
2011-03-05 23:21:11.381511 [DEBUG] mod_loopback.c:469 loopback/voicemail-b CHANNEL KILL
2011-03-05 23:21:11.381511 [DEBUG] switch_core_state_machine.c:46 loopback/voicemail-a Standard HANGUP, cause: NORMAL_CLEARING
2011-03-05 23:21:11.381511 [DEBUG] switch_core_state_machine.c:560 (loopback/voicemail-a) State HANGUP going to sleep
2011-03-05 23:21:11.381511 [DEBUG] switch_core_state_machine.c:351 (loopback/voicemail-a) State Change CS_HANGUP -> CS_REPORTING
2011-03-05 23:21:11.381511 [DEBUG] switch_core_session.c:1116 Send signal loopback/voicemail-a [BREAK]
2011-03-05 23:21:11.381511 [DEBUG] mod_loopback.c:469 loopback/voicemail-a CHANNEL KILL
2011-03-05 23:21:11.381511 [DEBUG] switch_core_state_machine.c:320 (loopback/voicemail-a) Running State Change CS_REPORTING
2011-03-05 23:21:11.381511 [DEBUG] switch_core_state_machine.c:620 (loopback/voicemail-a) State REPORTING
2011-03-05 23:21:11.381511 [DEBUG] switch_core_state_machine.c:53 loopback/voicemail-a Standard REPORTING, cause: NORMAL_CLEARING
2011-03-05 23:21:11.381511 [DEBUG] switch_core_state_machine.c:620 (loopback/voicemail-a) State REPORTING going to sleep
2011-03-05 23:21:11.381511 [DEBUG] switch_core_state_machine.c:345 (loopback/voicemail-a) State Change CS_REPORTING -> CS_DESTROY
2011-03-05 23:21:11.381511 [DEBUG] switch_core_session.c:1116 Send signal loopback/voicemail-a [BREAK]
2011-03-05 23:21:11.381511 [DEBUG] mod_loopback.c:469 loopback/voicemail-a CHANNEL KILL
2011-03-05 23:21:11.381511 [DEBUG] switch_core_session.c:1288 Session 163 (loopback/voicemail-a) Locked, Waiting on external entities
2011-03-05 23:21:11.384116 [DEBUG] switch_ivr_bridge.c:1308 sofia/internal/1006 at 172.16.200.60 skip receive message [UNBRIDGE] (channel is hungup already)
2011-03-05 23:21:11.384116 [DEBUG] switch_core_session.c:2060 sofia/internal/1006 at 172.16.200.60 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2011-03-05 23:21:11.384116 [DEBUG] switch_core_state_machine.c:366 (sofia/internal/1006 at 172.16.200.60) State EXECUTE going to sleep
2011-03-05 23:21:11.384116 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1006 at 172.16.200.60) Running State Change CS_HANGUP
2011-03-05 23:21:11.384116 [DEBUG] switch_core_state_machine.c:560 (sofia/internal/1006 at 172.16.200.60) State HANGUP
2011-03-05 23:21:11.384116 [DEBUG] mod_sofia.c:451 sofia/internal/1006 at 172.16.200.60 Overriding SIP cause 480 with 200 from the other leg
2011-03-05 23:21:11.384116 [DEBUG] mod_sofia.c:457 Channel sofia/internal/1006 at 172.16.200.60 hanging up, cause: NORMAL_CLEARING
2011-03-05 23:21:11.384116 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1006 at 172.16.200.60 Standard HANGUP, cause: NORMAL_CLEARING
2011-03-05 23:21:11.384116 [DEBUG] switch_core_state_machine.c:560 (sofia/internal/1006 at 172.16.200.60) State HANGUP going to sleep
2011-03-05 23:21:11.384116 [DEBUG] switch_core_state_machine.c:351 (sofia/internal/1006 at 172.16.200.60) State Change CS_HANGUP -> CS_REPORTING
2011-03-05 23:21:11.384116 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1006 at 172.16.200.60 [BREAK]
2011-03-05 23:21:11.384116 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1006 at 172.16.200.60) Running State Change CS_REPORTING
2011-03-05 23:21:11.384116 [DEBUG] switch_core_state_machine.c:620 (sofia/internal/1006 at 172.16.200.60) State REPORTING
2011-03-05 23:21:11.384116 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1006 at 172.16.200.60 Standard REPORTING, cause: NORMAL_CLEARING
2011-03-05 23:21:11.384116 [DEBUG] switch_core_state_machine.c:620 (sofia/internal/1006 at 172.16.200.60) State REPORTING going to sleep
2011-03-05 23:21:11.384116 [DEBUG] switch_core_state_machine.c:345 (sofia/internal/1006 at 172.16.200.60) State Change CS_REPORTING -> CS_DESTROY
2011-03-05 23:21:11.384116 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1006 at 172.16.200.60 [BREAK]
2011-03-05 23:21:11.384116 [DEBUG] switch_core_session.c:1288 Session 161 (sofia/internal/1006 at 172.16.200.60) Locked, Waiting on external entities
2011-03-05 23:21:11.384116 [NOTICE] switch_core_session.c:1306 Session 161 (sofia/internal/1006 at 172.16.200.60) Ended
2011-03-05 23:21:11.384116 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/1006 at 172.16.200.60 [CS_DESTROY]
2011-03-05 23:21:11.384116 [DEBUG] switch_core_state_machine.c:449 (sofia/internal/1006 at 172.16.200.60) Callstate Change HANGUP -> DOWN
2011-03-05 23:21:11.384116 [DEBUG] switch_core_state_machine.c:452 (sofia/internal/1006 at 172.16.200.60) Running State Change CS_DESTROY
2011-03-05 23:21:11.384116 [DEBUG] switch_core_state_machine.c:462 (sofia/internal/1006 at 172.16.200.60) State DESTROY
2011-03-05 23:21:11.384116 [DEBUG] mod_sofia.c:362 sofia/internal/1006 at 172.16.200.60 SOFIA DESTROY
2011-03-05 23:21:11.384116 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1006 at 172.16.200.60 Standard DESTROY
2011-03-05 23:21:11.384116 [DEBUG] switch_core_state_machine.c:462 (sofia/internal/1006 at 172.16.200.60) State DESTROY going to sleep 


I'm not sure how to avoid so much output, but I hope it's enough information to sort this out.

Thanks,

Mitch	




From curriegrad2004 at gmail.com  Sun Mar  6 08:18:59 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 5 Mar 2011 21:18:59 -0800
Subject: [Freeswitch-users] send fax via google voice.
In-Reply-To: <4D72FF8A.4080209@coppice.org>
References: <AANLkTin4uo+oKr_Nzw6qVzf72E7n896j_5xyoB6nFcnk@mail.gmail.com>
	<AANLkTimDfszKNxgugnSECrtCE+ByKgYV-sjK0tjXppfK@mail.gmail.com>
	<AANLkTi=bZ0geQnP5BT37D7udeSPXmzwe=nN5FuDFgbiP@mail.gmail.com>
	<4D72FF8A.4080209@coppice.org>
Message-ID: <AANLkTinAWMUULZHDDeBNrRJtPyLY7VGp96qwJDg2cUsk@mail.gmail.com>

And relativity comes into play too.

Also you do need to realize that the packets sent via UDP isn't
guaranteed to be in order as the fax machine expects the data to be
sent in order, not out of order. That's the beauty of packet switched
networks.

On Sat, Mar 5, 2011 at 7:29 PM, Steve Underwood <steveu at coppice.org> wrote:
> On 03/06/2011 11:21 AM, envelopes envelopes wrote:
>>
>> Why is PCMU codec not reliable?
>>
>>
> Unsynchronised clocks, and packet loss.
>
> Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From sunwood360 at gmail.com  Sun Mar  6 08:35:08 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Sat, 5 Mar 2011 21:35:08 -0800
Subject: [Freeswitch-users] send fax via google voice.
In-Reply-To: <AANLkTinAWMUULZHDDeBNrRJtPyLY7VGp96qwJDg2cUsk@mail.gmail.com>
References: <AANLkTin4uo+oKr_Nzw6qVzf72E7n896j_5xyoB6nFcnk@mail.gmail.com>
	<AANLkTimDfszKNxgugnSECrtCE+ByKgYV-sjK0tjXppfK@mail.gmail.com>
	<AANLkTi=bZ0geQnP5BT37D7udeSPXmzwe=nN5FuDFgbiP@mail.gmail.com>
	<4D72FF8A.4080209@coppice.org>
	<AANLkTinAWMUULZHDDeBNrRJtPyLY7VGp96qwJDg2cUsk@mail.gmail.com>
Message-ID: <AANLkTim0x9nwLCr3-KSXuekJmvBLKBt0JxQqt+tZhBfG@mail.gmail.com>

ok,

Reliability is a different issue. I am just wondering whether mod_dingaling
supports fax via GV. because someone proved sending fax via GV from fax
machine did work
.(http://www.magicjacksupport.com/faxing-via-google-voice-t7412.html).



On Sat, Mar 5, 2011 at 9:18 PM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:

> And relativity comes into play too.
>
> Also you do need to realize that the packets sent via UDP isn't
> guaranteed to be in order as the fax machine expects the data to be
> sent in order, not out of order. That's the beauty of packet switched
> networks.
>
> On Sat, Mar 5, 2011 at 7:29 PM, Steve Underwood <steveu at coppice.org>
> wrote:
> > On 03/06/2011 11:21 AM, envelopes envelopes wrote:
> >>
> >> Why is PCMU codec not reliable?
> >>
> >>
> > Unsynchronised clocks, and packet loss.
> >
> > Steve
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From curriegrad2004 at gmail.com  Sun Mar  6 08:42:57 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 5 Mar 2011 21:42:57 -0800
Subject: [Freeswitch-users] send fax via google voice.
In-Reply-To: <AANLkTim0x9nwLCr3-KSXuekJmvBLKBt0JxQqt+tZhBfG@mail.gmail.com>
References: <AANLkTin4uo+oKr_Nzw6qVzf72E7n896j_5xyoB6nFcnk@mail.gmail.com>
	<AANLkTimDfszKNxgugnSECrtCE+ByKgYV-sjK0tjXppfK@mail.gmail.com>
	<AANLkTi=bZ0geQnP5BT37D7udeSPXmzwe=nN5FuDFgbiP@mail.gmail.com>
	<4D72FF8A.4080209@coppice.org>
	<AANLkTinAWMUULZHDDeBNrRJtPyLY7VGp96qwJDg2cUsk@mail.gmail.com>
	<AANLkTim0x9nwLCr3-KSXuekJmvBLKBt0JxQqt+tZhBfG@mail.gmail.com>
Message-ID: <AANLkTimH8c8BfSU54HV4=LqnZug8nFWZiCKpijLfGgiY@mail.gmail.com>

Just because it worked for somebody, it doesn't means it can work for
you. In theory, yes uLaw is supposed to work with faxing, but the
chances of sending a fax successfully is quite slim in the real world.

Google voice was intended for voice in the first place, not for
faxing. Yes I know, somebody did manage to get the fax through mainly
because he was plain lucky or he had an ISP that can actually handle
VoIP correctly.

To sum it up, the simple answer to GV faxing: If you like to play with
fire, go ahead and try. I'm telling you this because I've been there
and done that, faxing over uLaw isn't pretty at all.

On Sat, Mar 5, 2011 at 9:35 PM, envelopes envelopes
<sunwood360 at gmail.com> wrote:
> ok,
>
> Reliability is a different issue. I am just wondering whether mod_dingaling
> supports fax via GV. because someone proved sending fax via GV from fax
> machine did work
> .(http://www.magicjacksupport.com/faxing-via-google-voice-t7412.html).
>
>
>
> On Sat, Mar 5, 2011 at 9:18 PM, curriegrad2004 <curriegrad2004 at gmail.com>
> wrote:
>>
>> And relativity comes into play too.
>>
>> Also you do need to realize that the packets sent via UDP isn't
>> guaranteed to be in order as the fax machine expects the data to be
>> sent in order, not out of order. That's the beauty of packet switched
>> networks.
>>
>> On Sat, Mar 5, 2011 at 7:29 PM, Steve Underwood <steveu at coppice.org>
>> wrote:
>> > On 03/06/2011 11:21 AM, envelopes envelopes wrote:
>> >>
>> >> Why is PCMU codec not reliable?
>> >>
>> >>
>> > Unsynchronised clocks, and packet loss.
>> >
>> > Steve
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steveu at coppice.org  Sun Mar  6 09:45:29 2011
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 06 Mar 2011 14:45:29 +0800
Subject: [Freeswitch-users] send fax via google voice.
In-Reply-To: <AANLkTinAWMUULZHDDeBNrRJtPyLY7VGp96qwJDg2cUsk@mail.gmail.com>
References: <AANLkTin4uo+oKr_Nzw6qVzf72E7n896j_5xyoB6nFcnk@mail.gmail.com>	<AANLkTimDfszKNxgugnSECrtCE+ByKgYV-sjK0tjXppfK@mail.gmail.com>	<AANLkTi=bZ0geQnP5BT37D7udeSPXmzwe=nN5FuDFgbiP@mail.gmail.com>	<4D72FF8A.4080209@coppice.org>
	<AANLkTinAWMUULZHDDeBNrRJtPyLY7VGp96qwJDg2cUsk@mail.gmail.com>
Message-ID: <4D732D89.2060101@coppice.org>

On 03/06/2011 01:18 PM, curriegrad2004 wrote:
> And relativity comes into play too.
Not unless you travel pretty fast. :-)
> Also you do need to realize that the packets sent via UDP isn't
> guaranteed to be in order as the fax machine expects the data to be
> sent in order, not out of order. That's the beauty of packet switched
> networks.
You will also see a small number of packets arrive twice for UDP over 
the internet, presumably due to router bugs.

Out of order packets are not, in themselves, a problem. A jitter buffer 
will reorder them, if they are not excessively delayed (and the jitter 
buffer is not of a broken design). The real issue is excessive jitter in 
the packet delivery times, which is functionally equivalent to packet 
loss - packets have to be dropped if they arrive too late for the real 
time deadline of the analogue channel.
> On Sat, Mar 5, 2011 at 7:29 PM, Steve Underwood<steveu at coppice.org>  wrote:
>> On 03/06/2011 11:21 AM, envelopes envelopes wrote:
>>> Why is PCMU codec not reliable?
>>>
>>>
>> Unsynchronised clocks, and packet loss.
>>
>> Steve
Steve



From krice at freeswitch.org  Sun Mar  6 09:56:59 2011
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 06 Mar 2011 00:56:59 -0600
Subject: [Freeswitch-users] send fax via google voice.
In-Reply-To: <AANLkTim0x9nwLCr3-KSXuekJmvBLKBt0JxQqt+tZhBfG@mail.gmail.com>
Message-ID: <C9988C5B.12692%krice@freeswitch.org>

Ok
1. will sending a fax over it work... The answer is yes it will work..
2. will it work reliably, no
Faxing just needs to die already...
The problem you have here with Faxing over G711 (PCMU or PCMA) is that
faxing uses a modem to encode a digital signal into an analog carrier meant
for a circuit switch voice network with a constant amount of delay and
guaranteed delivery of the entire analog stream

VoIP by definition is Voice Over Internet Protocol... IP is a packet
switched network, where you can no guarantee a constant amount delay
(latency) ... Not to mention it is not common to see a udp packet or 2
missing along the way.

Where as 1 20ms packet may not seem like much to you, that?s roughtly
equivalent to 575bytes of data in the fax and it will screw up the fax
either causing a corruption of the data, or a drop in the analog modem
carrier which requires a retrain...




On 3/5/11 11:35 PM, "envelopes envelopes" <sunwood360 at gmail.com> wrote:

> ok,? 
> 
> Reliability is a different issue. I am just wondering whether mod_dingaling
> supports fax via GV. because someone proved sending fax via GV from fax
> machine did work
> .(http://www.magicjacksupport.com/faxing-via-google-voice-t7412.html).
> 
> 
> 
> On Sat, Mar 5, 2011 at 9:18 PM, curriegrad2004 <curriegrad2004 at gmail.com>
> wrote:
>> And relativity comes into play too.
>> 
>> Also you do need to realize that the packets sent via UDP isn't
>> guaranteed to be in order as the fax machine expects the data to be
>> sent in order, not out of order. That's the beauty of packet switched
>> networks.
>> 
>> On Sat, Mar 5, 2011 at 7:29 PM, Steve Underwood <steveu at coppice.org> wrote:
>>> > On 03/06/2011 11:21 AM, envelopes envelopes wrote:
>>>> >>
>>>> >> Why is PCMU codec not reliable?
>>>> >>
>>>> >>
>>> > Unsynchronised clocks, and packet loss.
>>> >
>>> > Steve
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From curriegrad2004 at gmail.com  Sun Mar  6 10:34:49 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 5 Mar 2011 23:34:49 -0800
Subject: [Freeswitch-users] send fax via google voice.
In-Reply-To: <C9988C5B.12692%krice@freeswitch.org>
References: <AANLkTim0x9nwLCr3-KSXuekJmvBLKBt0JxQqt+tZhBfG@mail.gmail.com>
	<C9988C5B.12692%krice@freeswitch.org>
Message-ID: <AANLkTi=DfGVQPrGgSc_Z7mUpH3Pq=J93pFY=UAhncjJM@mail.gmail.com>

That's why PDF's/XPS/tiff's have replaced fax completely. Not many
people out there still do have a fax machine :)

On Sat, Mar 5, 2011 at 10:56 PM, Ken Rice <krice at freeswitch.org> wrote:
> Ok
>
> will sending a fax over it work... The answer is yes it will work..
> will it work reliably, no
>
> Faxing just needs to die already...
> The problem you have here with Faxing over G711 (PCMU or PCMA) is that
> faxing uses a modem to encode a digital signal into an analog carrier meant
> for a circuit switch voice network with a constant amount of delay and
> guaranteed delivery of the entire analog stream
>
> VoIP by definition is Voice Over Internet Protocol... IP is a packet
> switched network, where you can no guarantee a constant amount delay
> (latency) ... Not to mention it is not common to see a udp packet or 2
> missing along the way.
>
> Where as 1 20ms packet may not seem like much to you, that?s roughtly
> equivalent to 575bytes of data in the fax and it will screw up the fax
> either causing a corruption of the data, or a drop in the analog modem
> carrier which requires a retrain...
>
>
>
>
> On 3/5/11 11:35 PM, "envelopes envelopes" <sunwood360 at gmail.com> wrote:
>
> ok,
>
> Reliability is a different issue. I am just wondering whether mod_dingaling
> supports fax via GV. because someone proved sending fax via GV from fax
> machine did work
> .(http://www.magicjacksupport.com/faxing-via-google-voice-t7412.html).
>
>
>
> On Sat, Mar 5, 2011 at 9:18 PM, curriegrad2004 <curriegrad2004 at gmail.com>
> wrote:
>
> And relativity comes into play too.
>
> Also you do need to realize that the packets sent via UDP isn't
> guaranteed to be in order as the fax machine expects the data to be
> sent in order, not out of order. That's the beauty of packet switched
> networks.
>
> On Sat, Mar 5, 2011 at 7:29 PM, Steve Underwood <steveu at coppice.org> wrote:
>> On 03/06/2011 11:21 AM, envelopes envelopes wrote:
>>>
>>> Why is PCMU codec not reliable?
>>>
>>>
>> Unsynchronised clocks, and packet loss.
>>
>> Steve
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ________________________________
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From sunwood360 at gmail.com  Sun Mar  6 10:55:45 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Sat, 5 Mar 2011 23:55:45 -0800
Subject: [Freeswitch-users] send fax via google voice.
In-Reply-To: <AANLkTi=DfGVQPrGgSc_Z7mUpH3Pq=J93pFY=UAhncjJM@mail.gmail.com>
References: <AANLkTim0x9nwLCr3-KSXuekJmvBLKBt0JxQqt+tZhBfG@mail.gmail.com>
	<C9988C5B.12692%krice@freeswitch.org>
	<AANLkTi=DfGVQPrGgSc_Z7mUpH3Pq=J93pFY=UAhncjJM@mail.gmail.com>
Message-ID: <AANLkTimMTzkjwKy-MaXdojtovEL6vcc6ZSg7iZM4v4qJ@mail.gmail.com>

that is not true.

Real estator/law firm/accounting offices/travel agents...many of them still
prefer fax for security reasons.
I am always worried about sending my driver license pdf file via e-mail.



On Sat, Mar 5, 2011 at 11:34 PM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:

> That's why PDF's/XPS/tiff's have replaced fax completely. Not many
> people out there still do have a fax machine :)
>
> On Sat, Mar 5, 2011 at 10:56 PM, Ken Rice <krice at freeswitch.org> wrote:
> > Ok
> >
> > will sending a fax over it work... The answer is yes it will work..
> > will it work reliably, no
> >
> > Faxing just needs to die already...
> > The problem you have here with Faxing over G711 (PCMU or PCMA) is that
> > faxing uses a modem to encode a digital signal into an analog carrier
> meant
> > for a circuit switch voice network with a constant amount of delay and
> > guaranteed delivery of the entire analog stream
> >
> > VoIP by definition is Voice Over Internet Protocol... IP is a packet
> > switched network, where you can no guarantee a constant amount delay
> > (latency) ... Not to mention it is not common to see a udp packet or 2
> > missing along the way.
> >
> > Where as 1 20ms packet may not seem like much to you, that?s roughtly
> > equivalent to 575bytes of data in the fax and it will screw up the fax
> > either causing a corruption of the data, or a drop in the analog modem
> > carrier which requires a retrain...
> >
> >
> >
> >
> > On 3/5/11 11:35 PM, "envelopes envelopes" <sunwood360 at gmail.com> wrote:
> >
> > ok,
> >
> > Reliability is a different issue. I am just wondering whether
> mod_dingaling
> > supports fax via GV. because someone proved sending fax via GV from fax
> > machine did work
> > .(http://www.magicjacksupport.com/faxing-via-google-voice-t7412.html).
> >
> >
> >
> > On Sat, Mar 5, 2011 at 9:18 PM, curriegrad2004 <curriegrad2004 at gmail.com
> >
> > wrote:
> >
> > And relativity comes into play too.
> >
> > Also you do need to realize that the packets sent via UDP isn't
> > guaranteed to be in order as the fax machine expects the data to be
> > sent in order, not out of order. That's the beauty of packet switched
> > networks.
> >
> > On Sat, Mar 5, 2011 at 7:29 PM, Steve Underwood <steveu at coppice.org>
> wrote:
> >> On 03/06/2011 11:21 AM, envelopes envelopes wrote:
> >>>
> >>> Why is PCMU codec not reliable?
> >>>
> >>>
> >> Unsynchronised clocks, and packet loss.
> >>
> >> Steve
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > ________________________________
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveu at coppice.org  Sun Mar  6 11:02:43 2011
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 06 Mar 2011 16:02:43 +0800
Subject: [Freeswitch-users] send fax via google voice.
In-Reply-To: <AANLkTi=DfGVQPrGgSc_Z7mUpH3Pq=J93pFY=UAhncjJM@mail.gmail.com>
References: <AANLkTim0x9nwLCr3-KSXuekJmvBLKBt0JxQqt+tZhBfG@mail.gmail.com>	<C9988C5B.12692%krice@freeswitch.org>
	<AANLkTi=DfGVQPrGgSc_Z7mUpH3Pq=J93pFY=UAhncjJM@mail.gmail.com>
Message-ID: <4D733FA3.6090907@coppice.org>

On 03/06/2011 03:34 PM, curriegrad2004 wrote:
> That's why PDF's/XPS/tiff's have replaced fax completely. Not many
> people out there still do have a fax machine :)
I suspect if you make that comment again in 20 years, it will still have 
the smiley face at the end, but probably won't have XPS in it.

Steve



From curriegrad2004 at gmail.com  Sun Mar  6 11:14:05 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 6 Mar 2011 00:14:05 -0800
Subject: [Freeswitch-users] send fax via google voice.
In-Reply-To: <4D733FA3.6090907@coppice.org>
References: <AANLkTim0x9nwLCr3-KSXuekJmvBLKBt0JxQqt+tZhBfG@mail.gmail.com>
	<C9988C5B.12692%krice@freeswitch.org>
	<AANLkTi=DfGVQPrGgSc_Z7mUpH3Pq=J93pFY=UAhncjJM@mail.gmail.com>
	<4D733FA3.6090907@coppice.org>
Message-ID: <AANLkTimrB18Ct5XbvA6vsXMpQRkh4LthyQjRVe-Sgyka@mail.gmail.com>

> I suspect if you make that comment again in 20 years, it will still have
> the smiley face at the end, but probably won't have XPS in it.

That was just an example there :)

> that is not true.
>
> Real estator/law firm/accounting offices/travel agents...many of them still
> prefer fax for security reasons.
> I am always worried about sending my driver license pdf file via e-mail.
>
Faxing stuff over the PSTN network is no different than sending an
unencrypted PDF file via e-mails. You do realize that somebody inside
the VoIP provider could possibly tap on to the "line" and record
everything. Now when I mean everything, they can record the entire
handshake of both fax machines. Your good old POTS provider might just
be doing that without letting you know, they're called "soft taps".

Now faxing stuff over VoIP makes fax interception much much easier. So
no, faxing stuff over the PSTN isn't technically secure, but it
'feels' secure because of the level of difficulty for the average
person out there to intercept fax calls over the PSTN network. But
then again your telco might be able to do this without letting you
know.

On Sun, Mar 6, 2011 at 12:02 AM, Steve Underwood <steveu at coppice.org> wrote:
> On 03/06/2011 03:34 PM, curriegrad2004 wrote:
>> That's why PDF's/XPS/tiff's have replaced fax completely. Not many
>> people out there still do have a fax machine :)
> I suspect if you make that comment again in 20 years, it will still have
> the smiley face at the end, but probably won't have XPS in it.
>
> Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From peter.olsson at visionutveckling.se  Sun Mar  6 11:39:42 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 6 Mar 2011 09:39:42 +0100
Subject: [Freeswitch-users] How do I originiate new session from within
 LUA	script?
In-Reply-To: <52114D605A462A4E9A50E588EE7D0686012E575D9EE6@mail.forest.simpot.com>
References: <52114D605A462A4E9A50E588EE7D0686012E575D9EE4@mail.forest.simpot.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EB@cooper>,
	<52114D605A462A4E9A50E588EE7D0686012E575D9EE5@mail.forest.simpot.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EE@cooper>,
	<52114D605A462A4E9A50E588EE7D0686012E575D9EE6@mail.forest.simpot.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EF@cooper>

It doesn't matter how you set the variables, getVariable() should always return then anyway, the "{var=val}" trick is just to make it possible to set vars when originating a new channel, since the actual channel will be created at the same time.

However, if you want to check variables on the originated leg, it would be something like this;

local s = freeswitch.Session();
s:originate(session, "{var1=val1,var2=val2}sofia/internal/1000 at x.x.x.x", 30);

-- Do whatever you want with this channel.

-- Get variables from the originated channel
local var1 = s:getVariable("whatever_variable");

That is, you must read the var from the correct object. In this case "s" is the new originated channel, and "session" is the channel that was created when dialing into the lua script.

PS! I've not written so much in Lua myself, so it's not really something I'm good at - but I guess it should be like this :)

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Dmitry Saratsky [simpot at simpot.com]
Skickat: den 6 mars 2011 02:03
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] How do I originiate new session from within LUA    script?

Hi Peter,

Thanks again for your help, this way works well for me.
I will update wiki, when I fix all puzzles of my script.

Now I have another problem... Sorry...

I actually hoped also to read session variables after my new originated sessions complete in same way, I set them.
However in way you have suggested me to set vars, I have no idea how to read those variables now...

I have tried to use the following after origination command:
   local fax_result_code = session:getVariable("fax_result_code"); local fax_result_text = session:getVariable("fax_result_text");
   freeswitch.consoleLog("info", "FAX - Error code: (" .. fax_result_code .. ") " .. fax_result_text .. "\n");

I got the following in console:
2011-03-06 02:54:23.311416 [ERR] mod_lua.cpp:182 /usr/local/freeswitch/scripts/faxoutservice.lua:53: attempt to concatenate global 'fax_success' (a nil value)
stack traceback:
        /usr/local/freeswitch/scripts/faxoutservice.lua:53: in function 'send_fax'
        /usr/local/freeswitch/scripts/faxoutservice.lua:74: in main chunk

According to docs (http://wiki.freeswitch.org/wiki/Mod_spandsp), those vars should be set on hangup both for tx and rx faxes...
Anyway, I have lua script I have wrote for receiving faxes - I can read those variables successfully in the way I wrote above.

So, any ideas how can I read session (that I start in that way - origination) variables after session hangup?

Thanks again,
Dmitry.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson
Sent: 06 Mar 2011 00:29
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How do I originiate new session from within LUA script?

10 is the origination timeout in seconds. This part is missing in the documentation right now, so I can't really tell you where to look - except in the source :)

To set variables you should be able to do it as you do in the dialplan, changing the dial string to something like "{var1=value,var2=value}sofia/internal/test at test.com". By the way - for originate variables origination_caller_id_xxxx must be used.

If you're successful with these commands, I would really appreciate if this could be updated in the Wiki as well.

/Peter

________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Dmitry Saratsky [simpot at simpot.com]
Skickat: den 5 mars 2011 22:51
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] How do I originiate new session from within LUA    script?

Hi Peter,

Thanks for your replay.
It works in this way, thanks.

Can you please also explain (or point me to right documentation) what does mean "10" as 3rd variable in your example?
Also I'm looking the way to set session variables before origination and I also failed with this...

For example I have tried this (the call itself was succeeded, but caller_id was not passed):
   s = freeswitch.Session();
   s:setVariable("effective_caller_id_number", "999");
   s:setVariable("effective_caller_id_name", "999");
   s:originate(session, "sofia/gateway/mygw/1000", 10);

and I got the following in console:

2011-03-05 23:43:05.754621 [ERR] switch_cpp.cpp:634 session is not initalized
2011-03-05 23:43:05.754621 [ERR] switch_cpp.cpp:634 session is not initialized

Thanks,
Dmitry.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson
Sent: 05 Mar 2011 22:48
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How do I originiate new session from within LUA script?

This should work, last parameter is the originate timeout in seconds.

s = freeswitch.Session();
s:originate(session, "sofia/internal/1000 at 192.168.1.1", 10);

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Dmitry Saratsky [simpot at simpot.com]
Skickat: den 5 mars 2011 20:53
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] How do I originiate new session from within LUA        script?

Hello all,

According to wiki:
http://wiki.freeswitch.org/wiki/Mod_lua#session:originate

I can use: "session:originate" command in LUA script to originate new session, but I can't understand how should I do it and can't find any example how I use this command.

Can you help, guys?

Thanks,
Dmitry.


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

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




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

_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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!DSPAM:4d72dea332761110315127!



From gilles.gerlinger at free.fr  Sun Mar  6 14:25:38 2011
From: gilles.gerlinger at free.fr (gigerlin)
Date: Sun, 6 Mar 2011 03:25:38 -0800 (PST)
Subject: [Freeswitch-users] writing to file in javascript
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B2C494E9@cooper>
References: <1299345025947-6092204.post@n2.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2C494E9@cooper>
Message-ID: <1299410738667-6094014.post@n2.nabble.com>

Thanks Peter,

That works fine now!

Regards,
Gilles Gerlinger

--
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From gilles.gerlinger at free.fr  Sun Mar  6 14:30:47 2011
From: gilles.gerlinger at free.fr (gigerlin)
Date: Sun, 6 Mar 2011 03:30:47 -0800 (PST)
Subject: [Freeswitch-users] Deactivating write resampler
Message-ID: <1299411047906-6094024.post@n2.nabble.com>

Dear Freeswitch users,

When playing a wav file inside a session with:

session.execute("playback", sounds + "bip.wav");

where bip.wav is a mono 22050 Hz 16-bit PCM file

I get the following message:
2011-02-18 11:37:23.079177 [NOTICE] switch_core_io.c:865 Deactivating write
resampler

What does that means exactly? Should I do something about it?

Kind regards,
Gilles Gerlinger

PS: I have no notice with mono 44100 Hz 16-bit PCM files. 

--
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From mitch.johnson7 at gmail.com  Sun Mar  6 19:43:10 2011
From: mitch.johnson7 at gmail.com (Mitch Johnson)
Date: Sun, 6 Mar 2011 11:43:10 -0500
Subject: [Freeswitch-users] srtp clarification
Message-ID: <14A16D06-02BF-444F-AE26-6478B6A5AEEF@gmail.com>

My previous post may have suggested that the TLS/SRTP was not working.  Where in fact, the TLS works like a charm.

The problem comes in when I require SRTP only on the phones.  When SRTP s turned off it works great, and so does TLS.  

I've been trying to understand how the voice part of the call is setup using SRTP.  When I go through the logs, I don't see anything that says that SRTP failed anywhere.  I'm pretty sure it's somewhere in my configuration.  In Asterisk I had to define the transport mechanism of tls and encryption=yes to make it supposed to work.  But then I never got it working there either, the difference with Asterisk is that it was showing SRTP as failing, but there's a bug causing that so it was pretty much a brick wall for me.

Am I supposed to do something under the user profile or somewhere else where that call is encrypted using SRTP?  I followed the TLS and SRTP guides to do the setup.

Any help on this would be greatly appreciated.  As with any problem, it's consuming my life until I can sort it out.

Thanks so much ahead of time,

Mitch

From krice at freeswitch.org  Sun Mar  6 19:55:32 2011
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 06 Mar 2011 10:55:32 -0600
Subject: [Freeswitch-users] send fax via google voice.
In-Reply-To: <AANLkTimMTzkjwKy-MaXdojtovEL6vcc6ZSg7iZM4v4qJ@mail.gmail.com>
Message-ID: <C99918A4.126A8%krice@freeswitch.org>

What security reasons? Its just as easy to tap a PSTN line as it is to sniff
a network connection, the only difference is you might need a little bit
different hardware...

If it were about security, fax machines would actually encrypt a fax but
they don?t... 


On 3/6/11 1:55 AM, "envelopes envelopes" <sunwood360 at gmail.com> wrote:

> that is not true.
> 
> Real estator/law firm/accounting offices/travel agents...many of them still
> prefer fax for security reasons.
> I am always worried about sending my driver license pdf file via e-mail.
> 
> 
> 
> On Sat, Mar 5, 2011 at 11:34 PM, curriegrad2004 <curriegrad2004 at gmail.com>
> wrote:
>> That's why PDF's/XPS/tiff's have replaced fax completely. Not many
>> people out there still do have a fax machine :)
>> 
>> On Sat, Mar 5, 2011 at 10:56 PM, Ken Rice <krice at freeswitch.org> wrote:
>>> > Ok
>>> >
>>> > will sending a fax over it work... The answer is yes it will work..
>>> > will it work reliably, no
>>> >
>>> > Faxing just needs to die already...
>>> > The problem you have here with Faxing over G711 (PCMU or PCMA) is that
>>> > faxing uses a modem to encode a digital signal into an analog carrier
>>> meant
>>> > for a circuit switch voice network with a constant amount of delay and
>>> > guaranteed delivery of the entire analog stream
>>> >
>>> > VoIP by definition is Voice Over Internet Protocol... IP is a packet
>>> > switched network, where you can no guarantee a constant amount delay
>>> > (latency) ... Not to mention it is not common to see a udp packet or 2
>>> > missing along the way.
>>> >
>>> > Where as 1 20ms packet may not seem like much to you, that?s roughtly
>>> > equivalent to 575bytes of data in the fax and it will screw up the fax
>>> > either causing a corruption of the data, or a drop in the analog modem
>>> > carrier which requires a retrain...
>>> >
>>> >
>>> >
>>> >
>>> > On 3/5/11 11:35 PM, "envelopes envelopes" <sunwood360 at gmail.com> wrote:
>>> >
>>> > ok,
>>> >
>>> > Reliability is a different issue. I am just wondering whether
>>> mod_dingaling
>>> > supports fax via GV. because someone proved sending fax via GV from fax
>>> > machine did work
>>> > .(http://www.magicjacksupport.com/faxing-via-google-voice-t7412.html).
>>> >
>>> >
>>> >
>>> > On Sat, Mar 5, 2011 at 9:18 PM, curriegrad2004 <curriegrad2004 at gmail.com>
>>> > wrote:
>>> >
>>> > And relativity comes into play too.
>>> >
>>> > Also you do need to realize that the packets sent via UDP isn't
>>> > guaranteed to be in order as the fax machine expects the data to be
>>> > sent in order, not out of order. That's the beauty of packet switched
>>> > networks.
>>> >
>>> > On Sat, Mar 5, 2011 at 7:29 PM, Steve Underwood <steveu at coppice.org>
>>> wrote:
>>>> >> On 03/06/2011 11:21 AM, envelopes envelopes wrote:
>>>>> >>>
>>>>> >>> Why is PCMU codec not reliable?
>>>>> >>>
>>>>> >>>
>>>> >> Unsynchronised clocks, and packet loss.
>>>> >>
>>>> >> Steve
>>>> >>
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >>
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>> > ________________________________
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jeff at jefflenk.com  Sun Mar  6 20:07:06 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sun, 6 Mar 2011 09:07:06 -0800 (PST)
Subject: [Freeswitch-users] avmd from mod managed
In-Reply-To: <AANLkTimq1xPh2SBezytcaj17KXiwmSXsPNEwbY9x16Ee@mail.gmail.com>
References: <AANLkTimq1xPh2SBezytcaj17KXiwmSXsPNEwbY9x16Ee@mail.gmail.com>
Message-ID: <1299431226666-6094692.post@n2.nabble.com>

Sure,

Modify this sample out of demo.csx from the source tree. maybe enough here
to get you started.

using System;
using FreeSWITCH;
using FreeSWITCH.Native;

public class AppDemo : IAppPlugin {

    ManagedSession Session;
    public void Run(AppContext context) {
        Session = context.Session;
        Session.Answer();
        Session.DtmfReceivedFunction = (d, t) => {
            Log.WriteLine(LogLevel.Notice, "Received {0} for {1}.", d, t);
            return "";
        };

        Session.EventReceivedFunction = (ev, s) => {
            return "";
        };

// add your stuff here

        Log.WriteLine(LogLevel.Notice, "AppDemo is finishing its run and
will now hang up.");
        Session.Hangup("USER_BUSY");
    }

}

--
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From infos at madovsky.org  Sun Mar  6 20:08:19 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 6 Mar 2011 12:08:19 -0500
Subject: [Freeswitch-users] send fax via google voice.
References: <C99918A4.126A8%krice@freeswitch.org>
Message-ID: <843CF6A82C2E44EC91515E847BCD422E@e1705>

Re: [Freeswitch-users] send fax via google voice.I think it's more hard for a guy to hack your US pstn line if he lives
in Australia... fax on internet can be hacked from everywhere...
  ----- Original Message ----- 
  From: Ken Rice 
  To: FreeSWITCH Users Help 
  Sent: Sunday, March 06, 2011 11:55 AM
  Subject: Re: [Freeswitch-users] send fax via google voice.


  What security reasons? Its just as easy to tap a PSTN line as it is to sniff a network connection, the only difference is you might need a little bit different hardware...

  If it were about security, fax machines would actually encrypt a fax but they don't... 


  On 3/6/11 1:55 AM, "envelopes envelopes" <sunwood360 at gmail.com> wrote:


    that is not true.

    Real estator/law firm/accounting offices/travel agents...many of them still prefer fax for security reasons.
    I am always worried about sending my driver license pdf file via e-mail.



    On Sat, Mar 5, 2011 at 11:34 PM, curriegrad2004 <curriegrad2004 at gmail.com> wrote:

      That's why PDF's/XPS/tiff's have replaced fax completely. Not many
      people out there still do have a fax machine :)

      On Sat, Mar 5, 2011 at 10:56 PM, Ken Rice <krice at freeswitch.org> wrote:
      > Ok
      >
      > will sending a fax over it work... The answer is yes it will work..
      > will it work reliably, no
      >
      > Faxing just needs to die already...
      > The problem you have here with Faxing over G711 (PCMU or PCMA) is that
      > faxing uses a modem to encode a digital signal into an analog carrier meant
      > for a circuit switch voice network with a constant amount of delay and
      > guaranteed delivery of the entire analog stream
      >
      > VoIP by definition is Voice Over Internet Protocol... IP is a packet
      > switched network, where you can no guarantee a constant amount delay
      > (latency) ... Not to mention it is not common to see a udp packet or 2
      > missing along the way.
      >
      > Where as 1 20ms packet may not seem like much to you, that's roughtly
      > equivalent to 575bytes of data in the fax and it will screw up the fax
      > either causing a corruption of the data, or a drop in the analog modem
      > carrier which requires a retrain...
      >
      >
      >
      >
      > On 3/5/11 11:35 PM, "envelopes envelopes" <sunwood360 at gmail.com> wrote:
      >
      > ok,
      >
      > Reliability is a different issue. I am just wondering whether mod_dingaling
      > supports fax via GV. because someone proved sending fax via GV from fax
      > machine did work
      > .(http://www.magicjacksupport.com/faxing-via-google-voice-t7412.html).
      >
      >
      >
      > On Sat, Mar 5, 2011 at 9:18 PM, curriegrad2004 <curriegrad2004 at gmail.com>
      > wrote:
      >
      > And relativity comes into play too.
      >
      > Also you do need to realize that the packets sent via UDP isn't
      > guaranteed to be in order as the fax machine expects the data to be
      > sent in order, not out of order. That's the beauty of packet switched
      > networks.
      >
      > On Sat, Mar 5, 2011 at 7:29 PM, Steve Underwood <steveu at coppice.org> wrote:
      >> On 03/06/2011 11:21 AM, envelopes envelopes wrote:
      >>>
      >>> Why is PCMU codec not reliable?
      >>>
      >>>
      >> Unsynchronised clocks, and packet loss.
      >>
      >> Steve
      >>
      >>
      >> _______________________________________________
      >> FreeSWITCH-users mailing list
      >> FreeSWITCH-users at lists.freeswitch.org
      >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      >> http://www.freeswitch.org
      >>
      >
      > _______________________________________________
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      > FreeSWITCH-users at lists.freeswitch.org
      > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      > http://www.freeswitch.org
      >
      >
      > ________________________________
      > _______________________________________________
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      > FreeSWITCH-users at lists.freeswitch.org
      > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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      >
      > _______________________________________________
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      > FreeSWITCH-users at lists.freeswitch.org
      > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      > http://www.freeswitch.org
      >
      >

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From krice at freeswitch.org  Sun Mar  6 20:23:53 2011
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 06 Mar 2011 11:23:53 -0600
Subject: [Freeswitch-users] send fax via google voice.
In-Reply-To: <843CF6A82C2E44EC91515E847BCD422E@e1705>
Message-ID: <C9991F49.126AE%krice@freeswitch.org>

Geography doesn?t stop a determined attacker...

Just because its a little more difficult, that doesn?t mean it is secure...
If that were the case then Classified US Government Faxes wouldn?t require
crypto gear in line (see STU-III for an example of this that?s over 20 years
old)

K


On 3/6/11 11:08 AM, "Madovsky" <infos at madovsky.org> wrote:

> I think it's more hard for a guy to hack your US pstn line if he lives
> in Australia... fax on internet can be hacked from everywhere...
>>  
>> ----- Original Message -----
>>  
>> From:  Ken Rice <mailto:krice at freeswitch.org>
>>  
>> To: FreeSWITCH Users Help <mailto:freeswitch-users at lists.freeswitch.org>
>>  
>> Sent: Sunday, March 06, 2011 11:55  AM
>>  
>> Subject: Re: [Freeswitch-users] send fax  via google voice.
>>  
>> 
>> What security reasons? Its just as easy to tap a PSTN  line as it is to sniff
>> a network connection, the only difference is you might  need a little bit
>> different hardware...
>> 
>> If it were about security, fax  machines would actually encrypt a fax but
>> they don?t... 
>> 
>> 
>> On 3/6/11  1:55 AM, "envelopes envelopes" <sunwood360 at gmail.com>  wrote:
>> 
>>  
>>> that is not true.
>>> 
>>> Real estator/law  firm/accounting offices/travel agents...many of them still
>>> prefer fax for  security reasons.
>>> I am always worried about sending my driver license pdf  file via e-mail.
>>> 
>>> 
>>> 
>>> On Sat, Mar 5, 2011 at 11:34 PM,  curriegrad2004 <curriegrad2004 at gmail.com>
>>> wrote:
>>>  
>>>> That's why PDF's/XPS/tiff's have replaced fax  completely. Not many
>>>> people out there still do have a fax machine  :)
>>>> 
>>>> On Sat, Mar 5, 2011 at 10:56 PM, Ken Rice <krice at freeswitch.org> wrote:
>>>>> >  Ok
>>>>> >
>>>>> > will sending a fax over it work... The answer is yes it  will work..
>>>>> > will it work reliably, no
>>>>> >
>>>>> > Faxing just  needs to die already...
>>>>> > The problem you have here with Faxing over  G711 (PCMU or PCMA) is that
>>>>> > faxing uses a modem to encode a  digital signal into an analog carrier
>>>>> meant
>>>>> > for a circuit switch  voice network with a constant amount of delay and
>>>>> > guaranteed  delivery of the entire analog stream
>>>>> >
>>>>> > VoIP by definition is  Voice Over Internet Protocol... IP is a packet
>>>>> > switched network,  where you can no guarantee a constant amount delay
>>>>> > (latency) ...  Not to mention it is not common to see a udp packet or 2
>>>>> > missing  along the way.
>>>>> >
>>>>> > Where as 1 20ms packet may not seem like  much to you, that?s roughtly
>>>>> > equivalent to 575bytes of data in the  fax and it will screw up the fax
>>>>> > either causing a corruption of  the data, or a drop in the analog modem
>>>>> > carrier which requires a  retrain...
>>>>> >
>>>>> >
>>>>> >
>>>>> >
>>>>> > On 3/5/11 11:35 PM,  "envelopes envelopes" <sunwood360 at gmail.com>
>>>>> wrote:
>>>>> >
>>>>> > ok,
>>>>> >
>>>>> > Reliability is a different  issue. I am just wondering whether
>>>>> mod_dingaling
>>>>> > supports fax via  GV. because someone proved sending fax via GV from fax
>>>>> > machine did  work
>>>>> > .(http://www.magicjacksupport.com/faxing-via-google-voice-t7412.html).
>>>>> >
>>>>> >
>>>>> >
>>>>> >  On Sat, Mar 5, 2011 at 9:18 PM, curriegrad2004
>>>>> <curriegrad2004 at gmail.com>
>>>>> >  wrote:
>>>>> >
>>>>> > And relativity comes into play too.
>>>>> >
>>>>> >  Also you do need to realize that the packets sent via UDP isn't
>>>>> >  guaranteed to be in order as the fax machine expects the data to  be
>>>>> > sent in order, not out of order. That's the beauty of packet  switched
>>>>> > networks.
>>>>> >
>>>>> > On Sat, Mar 5, 2011 at 7:29 PM,  Steve Underwood <steveu at coppice.org>
>>>>> wrote:
>>>>>> >> On  03/06/2011 11:21 AM, envelopes envelopes  wrote:
>>>>>>> >>>
>>>>>>> >>> Why is PCMU codec not  reliable?
>>>>>>> >>>
>>>>>>> >>>
>>>>>> >> Unsynchronised  clocks, and packet loss.
>>>>>> >>
>>>>>> >>  Steve
>>>>>> >>
>>>>>> >>
>>>>>> >>  _______________________________________________
>>>>>> >>  FreeSWITCH-users mailing list
>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>> >>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >>  
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> >>  http://www.freeswitch.org
>>>>>> >>
>>>>> >
>>>>> >  _______________________________________________
>>>>> > FreeSWITCH-users  mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> >  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >  
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >  http://www.freeswitch.org
>>>>> >
>>>>> >
>>>>> >  ________________________________
>>>>> >  _______________________________________________
>>>>> > FreeSWITCH-users  mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> >  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >  
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >  http://www.freeswitch.org
>>>>> >
>>>>> >  _______________________________________________
>>>>> > FreeSWITCH-users  mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> >  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >  
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >  http://www.freeswitch.org
>>>>> >
>>>>> >
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users  mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>>  
>>> 
>>>  _______________________________________________
>>> FreeSWITCH-users  mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>  
>> 
>>  
>> 
>>  
>> 
>> _______________________________________________
>> FreeSWITCH-users  mailing  list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Sun Mar  6 20:33:15 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 6 Mar 2011 12:33:15 -0500
Subject: [Freeswitch-users] send fax via google voice.
References: <C9991F49.126AE%krice@freeswitch.org>
Message-ID: <1811F3578978423DB3DB7E6695601915@e1705>

Re: [Freeswitch-users] send fax via google voice.because crypto was and is not allowed in some countries also...
  ----- Original Message ----- 
  From: Ken Rice 
  To: FreeSWITCH Users Help 
  Sent: Sunday, March 06, 2011 12:23 PM
  Subject: Re: [Freeswitch-users] send fax via google voice.


  Geography doesn't stop a determined attacker... 

  Just because its a little more difficult, that doesn't mean it is secure... If that were the case then Classified US Government Faxes wouldn't require crypto gear in line (see STU-III for an example of this that's over 20 years old)

  K


  On 3/6/11 11:08 AM, "Madovsky" <infos at madovsky.org> wrote:


    I think it's more hard for a guy to hack your US pstn line if he lives
    in Australia... fax on internet can be hacked from everywhere...


      ----- Original Message ----- 
       
      From:  Ken Rice <mailto:krice at freeswitch.org>   
       
      To: FreeSWITCH Users Help <mailto:freeswitch-users at lists.freeswitch.org>   
       
      Sent: Sunday, March 06, 2011 11:55  AM
       
      Subject: Re: [Freeswitch-users] send fax  via google voice.
       

      What security reasons? Its just as easy to tap a PSTN  line as it is to sniff a network connection, the only difference is you might  need a little bit different hardware...

      If it were about security, fax  machines would actually encrypt a fax but they don't... 


      On 3/6/11  1:55 AM, "envelopes envelopes" <sunwood360 at gmail.com>  wrote:

       

        that is not true.

        Real estator/law  firm/accounting offices/travel agents...many of them still prefer fax for  security reasons.
        I am always worried about sending my driver license pdf  file via e-mail.



        On Sat, Mar 5, 2011 at 11:34 PM,  curriegrad2004 <curriegrad2004 at gmail.com>  wrote:
         

          That's why PDF's/XPS/tiff's have replaced fax  completely. Not many
          people out there still do have a fax machine  :)

          On Sat, Mar 5, 2011 at 10:56 PM, Ken Rice <krice at freeswitch.org> wrote:
          >  Ok
          >
          > will sending a fax over it work... The answer is yes it  will work..
          > will it work reliably, no
          >
          > Faxing just  needs to die already...
          > The problem you have here with Faxing over  G711 (PCMU or PCMA) is that
          > faxing uses a modem to encode a  digital signal into an analog carrier meant
          > for a circuit switch  voice network with a constant amount of delay and
          > guaranteed  delivery of the entire analog stream
          >
          > VoIP by definition is  Voice Over Internet Protocol... IP is a packet
          > switched network,  where you can no guarantee a constant amount delay
          > (latency) ...  Not to mention it is not common to see a udp packet or 2
          > missing  along the way.
          >
          > Where as 1 20ms packet may not seem like  much to you, that's roughtly
          > equivalent to 575bytes of data in the  fax and it will screw up the fax
          > either causing a corruption of  the data, or a drop in the analog modem
          > carrier which requires a  retrain...
          >
          >
          >
          >
          > On 3/5/11 11:35 PM,  "envelopes envelopes" <sunwood360 at gmail.com>  wrote:
          >
          > ok,
          >
          > Reliability is a different  issue. I am just wondering whether mod_dingaling
          > supports fax via  GV. because someone proved sending fax via GV from fax
          > machine did  work
          > .(http://www.magicjacksupport.com/faxing-via-google-voice-t7412.html).
          >
          >
          >
          >  On Sat, Mar 5, 2011 at 9:18 PM, curriegrad2004 <curriegrad2004 at gmail.com>
          >  wrote:
          >
          > And relativity comes into play too.
          >
          >  Also you do need to realize that the packets sent via UDP isn't
          >  guaranteed to be in order as the fax machine expects the data to  be
          > sent in order, not out of order. That's the beauty of packet  switched
          > networks.
          >
          > On Sat, Mar 5, 2011 at 7:29 PM,  Steve Underwood <steveu at coppice.org> wrote:
          >> On  03/06/2011 11:21 AM, envelopes envelopes  wrote:
          >>>
          >>> Why is PCMU codec not  reliable?
          >>>
          >>>
          >> Unsynchronised  clocks, and packet loss.
          >>
          >>  Steve
          >>
          >>
          >>  _______________________________________________
          >>  FreeSWITCH-users mailing list
          >> FreeSWITCH-users at lists.freeswitch.org
          >>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          >>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
          >>  http://www.freeswitch.org
          >>
          >
          >  _______________________________________________
          > FreeSWITCH-users  mailing list
          > FreeSWITCH-users at lists.freeswitch.org
          >  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          >  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
          >  http://www.freeswitch.org
          >
          >
          >  ________________________________
          >  _______________________________________________
          > FreeSWITCH-users  mailing list
          > FreeSWITCH-users at lists.freeswitch.org
          >  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          >  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
          >  http://www.freeswitch.org
          >
          >  _______________________________________________
          > FreeSWITCH-users  mailing list
          > FreeSWITCH-users at lists.freeswitch.org
          >  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          >  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
          >  http://www.freeswitch.org
          >
          >

          _______________________________________________
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          UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From krice at freeswitch.org  Sun Mar  6 20:44:53 2011
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 06 Mar 2011 11:44:53 -0600
Subject: [Freeswitch-users] send fax via google voice.
In-Reply-To: <1811F3578978423DB3DB7E6695601915@e1705>
Message-ID: <C9992435.126B3%krice@freeswitch.org>

That makes it even easier in those countries... The whole point is anywhere
along the line and then some on the PSTN there are multiple points in which
a call can be compromised just as easy if not easier then cracking a network
along the way. Any 10 year old with 5 minutes of instruction can build a
pstn line tap and listen to phone calls... Its only a step up from that to
tap fax line... That being said, just because its perceived by luddites to
be more secure then say an email doesn?t make it so...




On 3/6/11 11:33 AM, "Madovsky" <infos at madovsky.org> wrote:

> because crypto was and is not allowed in some countries also...
>>  
>> ----- Original Message -----
>>  
>> From:  Ken Rice <mailto:krice at freeswitch.org>
>>  
>> To: FreeSWITCH Users Help <mailto:freeswitch-users at lists.freeswitch.org>
>>  
>> Sent: Sunday, March 06, 2011 12:23  PM
>>  
>> Subject: Re: [Freeswitch-users] send fax  via google voice.
>>  
>> 
>> Geography doesn?t stop a determined attacker...
>> 
>> Just because its a little more difficult, that doesn?t mean it is  secure...
>> If that were the case then Classified US Government Faxes wouldn?t  require
>> crypto gear in line (see STU-III for an example of this that?s over 20  years
>> old)
>> 
>> K
>> 
>> 
>> On 3/6/11 11:08 AM, "Madovsky" <infos at madovsky.org>  wrote:
>> 
>>  
>>> I think it's more hard for a guy to hack your US  pstn line if he lives
>>> in Australia... fax on internet can be hacked from  everywhere...
>>>  
>>>> 
>>>> ----- Original Message -----
>>>>  
>>>> From:  Ken Rice <mailto:krice at freeswitch.org>
>>>>  
>>>> To: FreeSWITCH Users Help <mailto:freeswitch-users at lists.freeswitch.org>
>>>>  
>>>> Sent: Sunday, March 06, 2011 11:55   AM
>>>>  
>>>> Subject: Re: [Freeswitch-users] send fax   via google voice.
>>>>  
>>>> 
>>>> What security reasons? Its just  as easy to tap a PSTN  line as it is to
>>>> sniff a network connection,  the only difference is you might  need a
>>>> little bit different  hardware...
>>>> 
>>>> If it were about security, fax  machines would  actually encrypt a fax but
>>>> they don?t... 
>>>> 
>>>> 
>>>> On 3/6/11  1:55  AM, "envelopes envelopes" <sunwood360 at gmail.com>   wrote:
>>>> 
>>>>  
>>>>  
>>>>> that is not true.
>>>>> 
>>>>> Real estator/law   firm/accounting offices/travel agents...many of them
>>>>> still prefer  fax for  security reasons.
>>>>> I am always worried about sending my  driver license pdf  file via e-mail.
>>>>> 
>>>>> 
>>>>> 
>>>>> On Sat, Mar 5,  2011 at 11:34 PM,  curriegrad2004
>>>>> <curriegrad2004 at gmail.com>   wrote:
>>>>>  
>>>>>  
>>>>>> That's why PDF's/XPS/tiff's have replaced fax   completely. Not many
>>>>>> people out there still do have a fax  machine  :)
>>>>>> 
>>>>>> On Sat, Mar 5, 2011 at 10:56 PM, Ken Rice  <krice at freeswitch.org>  wrote:
>>>>>>> >  Ok
>>>>>>> >
>>>>>>> > will sending a fax over it  work... The answer is yes it  will work..
>>>>>>> > will it work  reliably, no
>>>>>>> >
>>>>>>> > Faxing just  needs to die  already...
>>>>>>> > The problem you have here with Faxing over   G711 (PCMU or PCMA) is
>>>>>>> that
>>>>>>> > faxing uses a modem to  encode a  digital signal into an analog
>>>>>>> carrier meant
>>>>>>> > for  a circuit switch  voice network with a constant amount of delay
>>>>>>> and
>>>>>>> > guaranteed  delivery of the entire analog  stream
>>>>>>> >
>>>>>>> > VoIP by definition is  Voice Over Internet  Protocol... IP is a packet
>>>>>>> > switched network,  where you  can no guarantee a constant amount delay
>>>>>>> > (latency) ...   Not to mention it is not common to see a udp packet or
2
>>>>>>> >  missing  along the way.
>>>>>>> >
>>>>>>> > Where as 1 20ms packet  may not seem like  much to you, that?s
>>>>>>> roughtly
>>>>>>> >  equivalent to 575bytes of data in the  fax and it will screw up  the
>>>>>>> fax
>>>>>>> > either causing a corruption of  the data, or a  drop in the analog
>>>>>>> modem
>>>>>>> > carrier which requires a   retrain...
>>>>>>> >
>>>>>>> >
>>>>>>> >
>>>>>>> >
>>>>>>> > On 3/5/11  11:35 PM,  "envelopes envelopes" <sunwood360 at gmail.com>
>>>>>>> wrote:
>>>>>>> >
>>>>>>> > ok,
>>>>>>> >
>>>>>>> > Reliability is a  different  issue. I am just wondering whether
>>>>>>> mod_dingaling
>>>>>>> > supports fax via  GV. because someone  proved sending fax via GV from
>>>>>>> fax
>>>>>>> > machine did   work
>>>>>>> > .(http://www.magicjacksupport.com/faxing-via-google-voice-t7412.html).
>>>>>>> >
>>>>>>> >
>>>>>>> >
>>>>>>> >   On Sat, Mar 5, 2011 at 9:18 PM, curriegrad2004
>>>>>>> <curriegrad2004 at gmail.com>
>>>>>>> >   wrote:
>>>>>>> >
>>>>>>> > And relativity comes into play  too.
>>>>>>> >
>>>>>>> >  Also you do need to realize that the  packets sent via UDP isn't
>>>>>>> >  guaranteed to be in order as  the fax machine expects the data to  be
>>>>>>> > sent in order,  not out of order. That's the beauty of packet
>>>>>>> switched
>>>>>>> >  networks.
>>>>>>> >
>>>>>>> > On Sat, Mar 5, 2011 at 7:29 PM,  Steve  Underwood <steveu at coppice.org>
>>>>>>> wrote:
>>>>>>>> >> On  03/06/2011 11:21 AM, envelopes envelopes   wrote:
>>>>>>>>> >>>
>>>>>>>>> >>> Why is PCMU codec not   reliable?
>>>>>>>>> >>>
>>>>>>>>> >>>
>>>>>>>> >>  Unsynchronised  clocks, and packet loss.
>>>>>>>> >>
>>>>>>>> >>   Steve
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> >>   _______________________________________________
>>>>>>>> >>   FreeSWITCH-users mailing list
>>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >>   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >>   
>>>>>>>> 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> >>   http://www.freeswitch.org
>>>>>>>> >>
>>>>>>> >
>>>>>>> >   _______________________________________________
>>>>>>> >  FreeSWITCH-users  mailing list
>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >   
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> >   http://www.freeswitch.org
>>>>>>> >
>>>>>>> >
>>>>>>> >   ________________________________
>>>>>>> >   _______________________________________________
>>>>>>> >  FreeSWITCH-users  mailing list
>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >   
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> >   http://www.freeswitch.org
>>>>>>> >
>>>>>>> >   _______________________________________________
>>>>>>> >  FreeSWITCH-users  mailing list
>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >   
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> >   http://www.freeswitch.org
>>>>>>> >
>>>>>>> >
>>>>>> 
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users   mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>>  
>>>>>  
>>>>> 
>>>>>  _______________________________________________
>>>>> FreeSWITCH-users   mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>>  
>>>>  
>>>> 
>>>>  
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users   mailing  list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>>  
>>>> 
>>>>  
>>> _______________________________________________
>>> FreeSWITCH-users  mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>  
>> 
>>  
>> 
>>  
>> 
>> _______________________________________________
>> FreeSWITCH-users  mailing  list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From steveayre at gmail.com  Sun Mar  6 22:05:17 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 6 Mar 2011 19:05:17 +0000
Subject: [Freeswitch-users] srtp clarification
In-Reply-To: <14A16D06-02BF-444F-AE26-6478B6A5AEEF@gmail.com>
References: <14A16D06-02BF-444F-AE26-6478B6A5AEEF@gmail.com>
Message-ID: <AANLkTinCCB60C2Vssi+PNhqyeVrnCsXXvphMbMJsJMBB@mail.gmail.com>

> The problem comes in when I require SRTP only on the phones.

If you use SRTP without TLS, you get no security at all. The
encryption key used for the SRTP is passed within the SIP signalling.
Unless you encrypt that then anyone intercepting the call can get the
key from the signalling and then decrypt the media at will.

-Steve



On 6 March 2011 16:43, Mitch Johnson <mitch.johnson7 at gmail.com> wrote:
> My previous post may have suggested that the TLS/SRTP was not working. ?Where in fact, the TLS works like a charm.
>
> The problem comes in when I require SRTP only on the phones. ?When SRTP s turned off it works great, and so does TLS.
>
> I've been trying to understand how the voice part of the call is setup using SRTP. ?When I go through the logs, I don't see anything that says that SRTP failed anywhere. ?I'm pretty sure it's somewhere in my configuration. ?In Asterisk I had to define the transport mechanism of tls and encryption=yes to make it supposed to work. ?But then I never got it working there either, the difference with Asterisk is that it was showing SRTP as failing, but there's a bug causing that so it was pretty much a brick wall for me.
>
> Am I supposed to do something under the user profile or somewhere else where that call is encrypted using SRTP? ?I followed the TLS and SRTP guides to do the setup.
>
> Any help on this would be greatly appreciated. ?As with any problem, it's consuming my life until I can sort it out.
>
> Thanks so much ahead of time,
>
> Mitch
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From infos at madovsky.org  Sun Mar  6 23:34:00 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 6 Mar 2011 15:34:00 -0500
Subject: [Freeswitch-users] trunk ping
Message-ID: <4C4B74E7DAB04F5E825C161C2B4E1497@e1705>

I have a trunk that auth by IP so
in FS I set auth like

 <param name="register" value="false" />

butt ping in log show 
[WARNING] sofia.c:4028 Ping failed trunk_3 with code 408 - count -1/-1/1, state DOWN

however FS is using this trunk without any problem.
is it a normal message behavior ?

Thanks
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From anthony.minessale at gmail.com  Sun Mar  6 23:47:28 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 6 Mar 2011 14:47:28 -0600
Subject: [Freeswitch-users] Deactivating write resampler
In-Reply-To: <1299411047906-6094024.post@n2.nabble.com>
References: <1299411047906-6094024.post@n2.nabble.com>
Message-ID: <AANLkTikJcUnKihxEh8yXXV5SD2GHg3hydFNUM+z9G4gO@mail.gmail.com>

its just telling you when it no longer needs to resample, because you
are playing files at different sample rate from your call.


On Sun, Mar 6, 2011 at 5:30 AM, gigerlin <gilles.gerlinger at free.fr> wrote:
> Dear Freeswitch users,
>
> When playing a wav file inside a session with:
>
> session.execute("playback", sounds + "bip.wav");
>
> where bip.wav is a mono 22050 Hz 16-bit PCM file
>
> I get the following message:
> 2011-02-18 11:37:23.079177 [NOTICE] switch_core_io.c:865 Deactivating write
> resampler
>
> What does that means exactly? Should I do something about it?
>
> Kind regards,
> Gilles Gerlinger
>
> PS: I have no notice with mono 44100 Hz 16-bit PCM files.
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Deactivating-write-resampler-tp6094024p6094024.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 infos at madovsky.org  Mon Mar  7 00:06:01 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 6 Mar 2011 16:06:01 -0500
Subject: [Freeswitch-users] native default sounds
Message-ID: <2102851715E74577A63C93A5180F122A@e1705>

are default us callie sounds in speex already exist somewhere ?
or should I need to convert them on my own ?

Thanks

Franck
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From mitch.johnson7 at gmail.com  Mon Mar  7 00:18:04 2011
From: mitch.johnson7 at gmail.com (Mitch Johnson)
Date: Sun, 6 Mar 2011 16:18:04 -0500
Subject: [Freeswitch-users] srtp clarification
In-Reply-To: <mailman.4942.1299443642.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.4942.1299443642.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <F5DB05D0-2D8E-4D91-8713-C03414CF3F43@gmail.com>

I do understand the need for tls, I have no issues with tls, it works fine, it's the srtp I haven't managed to get working.

Thanks for your reply.

Mitch

> From: Steven Ayre <steveayre at gmail.com>
> Date: March 6, 2011 2:05:17 PM EST
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] srtp clarification
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
>> The problem comes in when I require SRTP only on the phones.
> 
> If you use SRTP without TLS, you get no security at all. The
> encryption key used for the SRTP is passed within the SIP signalling.
> Unless you encrypt that then anyone intercepting the call can get the
> key from the signalling and then decrypt the media at will.
> 
> -Steve
> 
> 
> 
> On 6 March 2011 16:43, Mitch Johnson <mitch.johnson7 at gmail.com> wrote:
>> My previous post may have suggested that the TLS/SRTP was not working.  Where in fact, the TLS works like a charm.
>> 
>> The problem comes in when I require SRTP only on the phones.  When SRTP s turned off it works great, and so does TLS.
>> 
>> I've been trying to understand how the voice part of the call is setup using SRTP.  When I go through the logs, I don't see anything that says that SRTP failed anywhere.  I'm pretty sure it's somewhere in my configuration.  In Asterisk I had to define the transport mechanism of tls and encryption=yes to make it supposed to work.  But then I never got it working there either, the difference with Asterisk is that it was showing SRTP as failing, but there's a bug causing that so it was pretty much a brick wall for me.
>> 
>> Am I supposed to do something under the user profile or somewhere else where that call is encrypted using SRTP?  I followed the TLS and SRTP guides to do the setup.
>> 
>> Any help on this would be greatly appreciated.  As with any problem, it's consuming my life until I can sort it out.
>> 
>> Thanks so much ahead of time,
>> 
>> Mitch
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 

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From steveayre at gmail.com  Mon Mar  7 00:57:45 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 6 Mar 2011 21:57:45 +0000
Subject: [Freeswitch-users] trunk ping
In-Reply-To: <4C4B74E7DAB04F5E825C161C2B4E1497@e1705>
References: <4C4B74E7DAB04F5E825C161C2B4E1497@e1705>
Message-ID: <AANLkTin0HkBxabnrkCfxfXHQkg9_5uXA=_VK2wry6ki2@mail.gmail.com>

"408 Request timed out" - It means the OPTIONS ping didn't get a reply.

FS should stop sending to the gateway while the gateway is DOWN. I
think the bleg fails with NETWORK_OUT_OF_ORDER.

Does the gateway still show as DOWN? When the next OPTIONS is sent
it'd mark it as UP again and continue sending traffic, your gateway
might have come back up which is why FS can still send to it.
"sofia status gateway xxx"

-Steve


On 6 March 2011 20:34, Madovsky <infos at madovsky.org> wrote:
> I have a trunk that auth by IP so
> in FS I set auth like
>
> ?<param name="register" value="false" />
>
> butt ping in log show
> [WARNING] sofia.c:4028 Ping failed trunk_3 with code 408 - count -1/-1/1,
> state DOWN
>
> however FS is using this trunk without any problem.
> is it a normal message behavior ?
>
> Thanks
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From infos at madovsky.org  Mon Mar  7 01:31:19 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 6 Mar 2011 17:31:19 -0500
Subject: [Freeswitch-users] trunk ping
References: <4C4B74E7DAB04F5E825C161C2B4E1497@e1705>
	<AANLkTin0HkBxabnrkCfxfXHQkg9_5uXA=_VK2wry6ki2@mail.gmail.com>
Message-ID: <3813972AB4ED4E96916766041BB36264@e1705>

...
Context         public
Expires         3600
Freq            3600
Ping            1299450234
PingFreq        30
PingState       -1/-1/1
State           NOREG
Status          DOWN (ping)
CallsIN         0
CallsOUT        0
FailedCallsIN   0
FailedCallsOUT  16
...
the trunk support said that they can ping my server...

I think the problem is not FS.

Thanks

----- Original Message ----- 
From: "Steven Ayre" <steveayre at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Sunday, March 06, 2011 4:57 PM
Subject: Re: [Freeswitch-users] trunk ping


"408 Request timed out" - It means the OPTIONS ping didn't get a reply.

FS should stop sending to the gateway while the gateway is DOWN. I
think the bleg fails with NETWORK_OUT_OF_ORDER.

Does the gateway still show as DOWN? When the next OPTIONS is sent
it'd mark it as UP again and continue sending traffic, your gateway
might have come back up which is why FS can still send to it.
"sofia status gateway xxx"

-Steve


On 6 March 2011 20:34, Madovsky <infos at madovsky.org> wrote:
> I have a trunk that auth by IP so
> in FS I set auth like
>
> <param name="register" value="false" />
>
> butt ping in log show
> [WARNING] sofia.c:4028 Ping failed trunk_3 with code 408 - count -1/-1/1,
> state DOWN
>
> however FS is using this trunk without any problem.
> is it a normal message behavior ?
>
> Thanks
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

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


From simpot at simpot.com  Mon Mar  7 01:48:26 2011
From: simpot at simpot.com (Dmitry Saratsky)
Date: Mon, 7 Mar 2011 00:48:26 +0200
Subject: [Freeswitch-users] What is so dangeres in "freeswitch.msleep()" in
	LUA?
Message-ID: <52114D605A462A4E9A50E588EE7D0686012E575D9EEB@mail.forest.simpot.com>

Hi All,

According to: http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.msleep
It is not good to use: "freeswitch.msleep()" in session-based script:
NOTE: Do not use this on a session-based script or bad things will happen.

Can anyone explain please why not and what does author mean with "session-based script".

I want to use it inside some function between I originate session and hangup it in the following way:

   repeat
--      os.execute("sleep 1");
        freeswitch.msleep(1000);
        until ((not new_session:ready() == true) or (new_session:getVariable("fax_result_code") >= "0"))
   new_session:hangup();

Is it danger for me to use this?
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From infos at madovsky.org  Mon Mar  7 02:17:53 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 6 Mar 2011 18:17:53 -0500
Subject: [Freeswitch-users] tell FS use native files
Message-ID: <9C45A99498D142449CDA503E2D0A9F72@e1705>

I'd like to use spx files for voicemail and all hold music
I tried to change voicemail.conf.xml "wav" to "" or "spx"
but no effect.
which settings I need to change to force FS use native files

Thanks
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From dujinfang at gmail.com  Mon Mar  7 02:24:09 2011
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 7 Mar 2011 07:24:09 +0800
Subject: [Freeswitch-users] What is so dangeres in "freeswitch.msleep()"
 in LUA?
In-Reply-To: <52114D605A462A4E9A50E588EE7D0686012E575D9EEB@mail.forest.simpot.com>
References: <52114D605A462A4E9A50E588EE7D0686012E575D9EEB@mail.forest.simpot.com>
Message-ID: <AANLkTimU9hgaYrkmFMezXafduxCYadic6Ku5NHkPKu=v@mail.gmail.com>

session-based means you have a session, so you can just use

if session:ready() ....

session:sleep(1000)

On Mon, Mar 7, 2011 at 6:48 AM, Dmitry Saratsky <simpot at simpot.com> wrote:
> Hi All,
>
>
>
> According to: http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.msleep
>
> It is not good to use: ?freeswitch.msleep()? in session-based script:
>
> NOTE: Do not use this on a session-based script or bad things will happen.
>
>
>
> Can anyone explain please why not and what does author mean with
> ?session-based script?.
>
>
>
> I want to use it inside some function between I originate session and hangup
> it in the following way:
>
>
>
> ?? repeat
>
> --????? os.execute("sleep 1");
>
> ??????? freeswitch.msleep(1000);
>
> ??????? until ((not new_session:ready() == true) or
> (new_session:getVariable("fax_result_code") >= "0"))
>
> ?? new_session:hangup();
>
>
>
> Is it danger for me to use this?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



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


From lloydie.t at gmail.com  Mon Mar  7 03:27:55 2011
From: lloydie.t at gmail.com (lloyd thomas)
Date: Mon, 7 Mar 2011 00:27:55 +0000
Subject: [Freeswitch-users] continual registers
Message-ID: <AANLkTim6RnrbuYmBbACOjZeBeoD5mrvrQGcCPjBVgZT6@mail.gmail.com>

Just notice that my polycom 550 seems to register every 15 secs. Thought
reg.1.server.1.expires="600" on polycom config would fix it, seems not.
Also tried my xlite client, which also displays the same tendency. Both are
behind NAT'ed networks but FS box is on public IP address.
Can I extend the register timeout and how? Is it wise to?
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From anthony.minessale at gmail.com  Mon Mar  7 04:41:47 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 6 Mar 2011 19:41:47 -0600
Subject: [Freeswitch-users] continual registers
In-Reply-To: <AANLkTim6RnrbuYmBbACOjZeBeoD5mrvrQGcCPjBVgZT6@mail.gmail.com>
References: <AANLkTim6RnrbuYmBbACOjZeBeoD5mrvrQGcCPjBVgZT6@mail.gmail.com>
Message-ID: <AANLkTi=BCgr86EecrLARJ-U34pVqvJ=ZFMcuu3L9Vh4a@mail.gmail.com>

FS changes the reg freq when called from behind nat to keep the mapping open.


On Sun, Mar 6, 2011 at 6:27 PM, lloyd thomas <lloydie.t at gmail.com> wrote:
> Just notice that my polycom 550 seems to register every 15 secs. Thought
> reg.1.server.1.expires="600" on polycom config would fix it, seems not.
> Also tried my xlite client, which also displays the same tendency. Both are
> behind NAT'ed networks but FS box is on public IP address.
> Can I extend the register timeout and how? Is it wise to?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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  Mon Mar  7 04:58:31 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 6 Mar 2011 19:58:31 -0600
Subject: [Freeswitch-users] Recent changes to fs_cli
In-Reply-To: <1299355646956-6092631.post@n2.nabble.com>
References: <1299355646956-6092631.post@n2.nabble.com>
Message-ID: <AANLkTi=qRMHiRgCjc=VB+TMA+4xT5ppy_JAQP416OG1S@mail.gmail.com>

no, but as a compromise try latest version where -i flag will give you
the desired behavior.


On Sat, Mar 5, 2011 at 2:07 PM, peely <freeswitch at peely.com> wrote:
> Hi,
>
> It looks like there's been a recent change to fs_cli which catches CTRL +
> C's and forces them to be CTRL + D's or specific commands.
>
> I quite often do an fs_cli > output.txt for example, then use nano to
> investigate test calls. It seems that now a CTRL + C doesn't end the command
> and I have to kill my connection.
>
> THis used to be quite useful so I wonder if it would be possible to have
> this functionality back?
>
>
> Cheers,
>
>
>
> Neil.
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Recent-changes-to-fs-cli-tp6092631p6092631.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 mitch.johnson7 at gmail.com  Mon Mar  7 05:49:32 2011
From: mitch.johnson7 at gmail.com (Mitch Johnson)
Date: Sun, 6 Mar 2011 21:49:32 -0500
Subject: [Freeswitch-users] SRTP
Message-ID: <C9354FF4-6636-4912-9A0B-DDE8050C22EC@gmail.com>

When I dial 9664 to test the tls/srtp configuration it says that the call is secure, however, when I dial another phone configured for tls/srtp the call doesn't go through, the automated attendant comes online to say that the extension is not available and then puts the call to voicemail.

I read the Secure RTP wiki, I do see a similar entries in the dialplan under the extension name global, for both inbound and outbound:

<condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never">
        <action application="set" data="sip_secure_media=true"/>
        <!-- Offer SRTP on outbound legs if we have it on inbound. -->
        <action application="export" data="sip_secure_media=true"/>
      </condition>

So if I can make the test call to 9664 on both phones, which I assume is using the inbound part by connecting the call.  

Any help in figuring this out would be greatly appreciated.

BTW, I did buy the book, but there's no mention of SRTP/TLS in there.

Thanks,

Mitch

From witt at cylogistics.com  Mon Mar  7 07:45:09 2011
From: witt at cylogistics.com (Don WItt)
Date: Sun, 6 Mar 2011 20:45:09 -0800
Subject: [Freeswitch-users] SRTP
In-Reply-To: <C9354FF4-6636-4912-9A0B-DDE8050C22EC@gmail.com>
References: <C9354FF4-6636-4912-9A0B-DDE8050C22EC@gmail.com>
Message-ID: <000001cbdc82$70878560$51969020$@com>

Is your dad's name Jerry?

Don Witt

Cylogistics
809B Cuesta Dr. #2149
Mountain View, CA 94040
650-694-4949 X 9102
Fax: 650-694-4953
witt at cylogistics.com
http://www.cylogistics.com
---------------------------------------------------
????????????????? Suppliers of award winning 
????????????????????????????? Ring Carrier
?? ???????????

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitch
Johnson
Sent: Sunday, March 06, 2011 6:50 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] SRTP

When I dial 9664 to test the tls/srtp configuration it says that the call is
secure, however, when I dial another phone configured for tls/srtp the call
doesn't go through, the automated attendant comes online to say that the
extension is not available and then puts the call to voicemail.

I read the Secure RTP wiki, I do see a similar entries in the dialplan under
the extension name global, for both inbound and outbound:

<condition field="${sip_has_crypto}"
expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"
break="never">
        <action application="set" data="sip_secure_media=true"/>
        <!-- Offer SRTP on outbound legs if we have it on inbound. -->
        <action application="export" data="sip_secure_media=true"/>
      </condition>

So if I can make the test call to 9664 on both phones, which I assume is
using the inbound part by connecting the call.  

Any help in figuring this out would be greatly appreciated.

BTW, I did buy the book, but there's no mention of SRTP/TLS in there.

Thanks,

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



From u2nsam at gmail.com  Mon Mar  7 09:20:09 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 7 Mar 2011 11:50:09 +0530
Subject: [Freeswitch-users] incompatible destination
Message-ID: <AANLkTincX-zpfnqx4dRk2jLCPf6HDiGN+cuPDA7+75WX@mail.gmail.com>

Hello,


The extension 7006 was working few days ago and now it not working,rest all
the phone are working properly.


U 192.168.2.190:5060 -> 192.168.2.14:5060
  INVITE sip:7006 at 192.168.2.14:5060 SIP/2.0..Via: SIP/2.0/UDP
192.168.2.190;rport;branch=z9hG4bK8mmBB5e8Z2mKr..Max-Forwards: 70..From: ""
<sip:9322273640 at 192.168.2.190>;tag=4QggZ1XcNZD3F..To: <
sip:7006 at 192.168.2.14:5060>..Call-ID: 7bf
  ae496-c324-122e-bc85-0026b976aef7..CSeq: 9403059 INVITE..Contact: <
sip:mod_sofia at 192.168.2.190:5060>..User-Agent: NOVANET..Allow: INVITE, ACK,
BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY,
PUBLISH, SUBSCRIBE.
  .Supported: timer, precondition, path, replaces..Allow-Events: talk, hold,
presence, dialog, line-seize, call-info, sla, include-session-description,
presence.winfo, message-summary, refer..Content-Type:
application/sdp..Content-Dis
  position: session..Content-Length: 203..X-FS-Support:
update_display..Remote-Party-ID:
<sip:9322273640 at 192.168.2.190>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH
1299446773 1299446774 IN IP4 192.168.2.190..s=FreeSWITCH
  ..c=IN IP4 192.168.2.190..t=0 0..m=audio 31474 RTP/AVP 8 101
13..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=ptime:20..

U 192.168.2.14:5060 -> 192.168.2.190:5060
  SIP/2.0 488 Not Acceptable Here..Via: SIP/2.0/UDP
192.168.2.190;rport;branch=z9hG4bK8mmBB5e8Z2mKr..From: <
sip:9322273640 at 192.168.2.190>;tag=4QggZ1XcNZD3F..To: "7006" <
sip:7006 at 192.168.2.14:5060>;tag=983B5A59-EB953B22..CSeq: 9403059
  INVITE..Call-ID: 7bfae496-c324-122e-bc85-0026b976aef7..Contact: <
sip:7006 at 192.168.2.14:5060>..User-Agent:
PolycomSoundPointIP-SPIP_335-UA/3.2.2.0477..Accept-Language: en..Warning:
488 SDP "NotAcceptableHere"..Content-Length: 0....



I tried all the possibilities with codec priorities on U/A .

http://pastebin.freeswitch.org/15577

Any help.

Regards
Sam
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From benkokakao at gmail.com  Mon Mar  7 11:11:01 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Mon, 7 Mar 2011 09:11:01 +0100
Subject: [Freeswitch-users] How could I forward the SIP incoming call to
 one of the available extensions?
In-Reply-To: <AANLkTik5DuoUavfyhfg69wnp_72zqiz1FKXSo9Rcp8zU@mail.gmail.com>
References: <AANLkTi=A-NSss214Qe=aiofh-qhofDKZMFBx3v8=DgDi@mail.gmail.com>
	<AANLkTikEwOW2KJz3z-of86wb4a8w8=SxoPCo0E3avVqY@mail.gmail.com>
	<AANLkTik5DuoUavfyhfg69wnp_72zqiz1FKXSo9Rcp8zU@mail.gmail.com>
Message-ID: <AANLkTi=v4fNSbMAcjY5hePCfgzAg2oxcuDeYBpMUTJ-M@mail.gmail.com>

On 5 March 2011 16:14, Jian Ren <renjian at gmail.com> wrote:
> Yes, my "extension" means the physical ATA connected to phones. I have 3 at
> home all registered to the FS server. I want all of them ring, or one by one
> for incoming calls from an external SIP connection. Then when two of them
> are being used to call outside I could still ?answer the incoming call.

Take a look at http://wiki.freeswitch.org/wiki/Dialplan_XML and the
link i've posted
earlier(http://wiki.freeswitch.org/wiki/Extension_Status_Example) -
this is easy to accomplish. If you are still stuck after reading the
wiki, let us know.

Best regards
Christian


From steveayre at gmail.com  Mon Mar  7 12:16:33 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 7 Mar 2011 09:16:33 +0000
Subject: [Freeswitch-users] What is so dangeres in "freeswitch.msleep()"
 in LUA?
In-Reply-To: <52114D605A462A4E9A50E588EE7D0686012E575D9EEB@mail.forest.simpot.com>
References: <52114D605A462A4E9A50E588EE7D0686012E575D9EEB@mail.forest.simpot.com>
Message-ID: <AANLkTimhk603FH-ABd=ju4EgBNsgO3rUw1DqCePxK+2o@mail.gmail.com>

AFAIK it stops all processing of the call during the sleep, you should
use session:sleep()

-Steve


On 6 March 2011 22:48, Dmitry Saratsky <simpot at simpot.com> wrote:
> Hi All,
>
>
>
> According to: http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.msleep
>
> It is not good to use: ?freeswitch.msleep()? in session-based script:
>
> NOTE: Do not use this on a session-based script or bad things will happen.
>
>
>
> Can anyone explain please why not and what does author mean with
> ?session-based script?.
>
>
>
> I want to use it inside some function between I originate session and hangup
> it in the following way:
>
>
>
> ?? repeat
>
> --????? os.execute("sleep 1");
>
> ??????? freeswitch.msleep(1000);
>
> ??????? until ((not new_session:ready() == true) or
> (new_session:getVariable("fax_result_code") >= "0"))
>
> ?? new_session:hangup();
>
>
>
> Is it danger for me to use this?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From michal.bielicki at seventhsignal.de  Mon Mar  7 12:37:41 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Mon, 7 Mar 2011 10:37:41 +0100
Subject: [Freeswitch-users] send fax via google voice.
In-Reply-To: <C9992435.126B3%krice@freeswitch.org>
References: <C9992435.126B3%krice@freeswitch.org>
Message-ID: <C187C712-671A-4524-8C2C-DD734D220485@seventhsignal.de>

I do not think it makes any difference if it is secure or not. In a court in Germany, a fax is a document, an email is not, at least not yet. So contracts need to be faxed, not emailed.
We can discuss the security of fax as much as we want, as long as the regulations make fax a requirement, it will stay around.

Am 06.03.2011 um 18:44 schrieb Ken Rice:

> That makes it even easier in those countries... The whole point is anywhere along the line and then some on the PSTN there are multiple points in which a call can be compromised just as easy if not easier then cracking a network along the way. Any 10 year old with 5 minutes of instruction can build a pstn line tap and listen to phone calls... Its only a step up from that to tap fax line... That being said, just because its perceived by luddites to be more secure then say an email doesn?t make it so... 
> 
> 
> 
> 
> On 3/6/11 11:33 AM, "Madovsky" <infos at madovsky.org> wrote:
> 
>> because crypto was and is not allowed in some countries also...
>>> 
>>> ----- Original Message ----- 
>>>  
>>> From:  Ken Rice <mailto:krice at freeswitch.org>   
>>>  
>>> To: FreeSWITCH Users Help <mailto:freeswitch-users at lists.freeswitch.org>   
>>>  
>>> Sent: Sunday, March 06, 2011 12:23  PM
>>>  
>>> Subject: Re: [Freeswitch-users] send fax  via google voice.
>>>  
>>> 
>>> Geography doesn?t stop a determined attacker...  
>>> 
>>> Just because its a little more difficult, that doesn?t mean it is  secure... If that were the case then Classified US Government Faxes wouldn?t  require crypto gear in line (see STU-III for an example of this that?s over 20  years old)
>>> 
>>> K
>>> 
>>> 
>>> On 3/6/11 11:08 AM, "Madovsky" <infos at madovsky.org>  wrote:
>>> 
>>>  
>>>> I think it's more hard for a guy to hack your US  pstn line if he lives
>>>> in Australia... fax on internet can be hacked from  everywhere...
>>>>  
>>>>> 
>>>>> ----- Original Message -----  
>>>>>  
>>>>> From:  Ken Rice <mailto:krice at freeswitch.org>    
>>>>>  
>>>>> To: FreeSWITCH Users Help <mailto:freeswitch-users at lists.freeswitch.org>    
>>>>>  
>>>>> Sent: Sunday, March 06, 2011 11:55   AM
>>>>>  
>>>>> Subject: Re: [Freeswitch-users] send fax   via google voice.
>>>>>  
>>>>> 
>>>>> What security reasons? Its just  as easy to tap a PSTN  line as it is to sniff a network connection,  the only difference is you might  need a little bit different  hardware...
>>>>> 
>>>>> If it were about security, fax  machines would  actually encrypt a fax but they don?t... 
>>>>> 
>>>>> 
>>>>> On 3/6/11  1:55  AM, "envelopes envelopes" <sunwood360 at gmail.com>   wrote:
>>>>> 
>>>>>  
>>>>>  
>>>>>> that is not true.
>>>>>> 
>>>>>> Real estator/law   firm/accounting offices/travel agents...many of them still prefer  fax for  security reasons.
>>>>>> I am always worried about sending my  driver license pdf  file via e-mail.
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> On Sat, Mar 5,  2011 at 11:34 PM,  curriegrad2004 <curriegrad2004 at gmail.com>   wrote:
>>>>>>  
>>>>>>  
>>>>>>> That's why PDF's/XPS/tiff's have replaced fax   completely. Not many
>>>>>>> people out there still do have a fax  machine  :)
>>>>>>> 
>>>>>>> On Sat, Mar 5, 2011 at 10:56 PM, Ken Rice  <krice at freeswitch.org>  wrote:
>>>>>>> >  Ok
>>>>>>> >
>>>>>>> > will sending a fax over it  work... The answer is yes it  will work..
>>>>>>> > will it work  reliably, no
>>>>>>> >
>>>>>>> > Faxing just  needs to die  already...
>>>>>>> > The problem you have here with Faxing over   G711 (PCMU or PCMA) is that
>>>>>>> > faxing uses a modem to  encode a  digital signal into an analog carrier meant
>>>>>>> > for  a circuit switch  voice network with a constant amount of delay  and
>>>>>>> > guaranteed  delivery of the entire analog  stream
>>>>>>> >
>>>>>>> > VoIP by definition is  Voice Over Internet  Protocol... IP is a packet
>>>>>>> > switched network,  where you  can no guarantee a constant amount delay
>>>>>>> > (latency) ...   Not to mention it is not common to see a udp packet or 2
>>>>>>> >  missing  along the way.
>>>>>>> >
>>>>>>> > Where as 1 20ms packet  may not seem like  much to you, that?s roughtly
>>>>>>> >  equivalent to 575bytes of data in the  fax and it will screw up  the fax
>>>>>>> > either causing a corruption of  the data, or a  drop in the analog modem
>>>>>>> > carrier which requires a   retrain...
>>>>>>> >
>>>>>>> >
>>>>>>> >
>>>>>>> >
>>>>>>> > On 3/5/11  11:35 PM,  "envelopes envelopes" <sunwood360 at gmail.com>   wrote:
>>>>>>> >
>>>>>>> > ok,
>>>>>>> >
>>>>>>> > Reliability is a  different  issue. I am just wondering whether  mod_dingaling
>>>>>>> > supports fax via  GV. because someone  proved sending fax via GV from fax
>>>>>>> > machine did   work
>>>>>>> > .(http://www.magicjacksupport.com/faxing-via-google-voice-t7412.html).
>>>>>>> >
>>>>>>> >
>>>>>>> >
>>>>>>> >   On Sat, Mar 5, 2011 at 9:18 PM, curriegrad2004 <curriegrad2004 at gmail.com>
>>>>>>> >   wrote:
>>>>>>> >
>>>>>>> > And relativity comes into play  too.
>>>>>>> >
>>>>>>> >  Also you do need to realize that the  packets sent via UDP isn't
>>>>>>> >  guaranteed to be in order as  the fax machine expects the data to  be
>>>>>>> > sent in order,  not out of order. That's the beauty of packet  switched
>>>>>>> >  networks.
>>>>>>> >
>>>>>>> > On Sat, Mar 5, 2011 at 7:29 PM,  Steve  Underwood <steveu at coppice.org>  wrote:
>>>>>>> >> On  03/06/2011 11:21 AM, envelopes envelopes   wrote:
>>>>>>> >>>
>>>>>>> >>> Why is PCMU codec not   reliable?
>>>>>>> >>>
>>>>>>> >>>
>>>>>>> >>  Unsynchronised  clocks, and packet loss.
>>>>>>> >>
>>>>>>> >>   Steve
>>>>>>> >>
>>>>>>> >>
>>>>>>> >>   _______________________________________________
>>>>>>> >>   FreeSWITCH-users mailing list
>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >>   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >>   UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> >>   http://www.freeswitch.org
>>>>>>> >>
>>>>>>> >
>>>>>>> >   _______________________________________________
>>>>>>> >  FreeSWITCH-users  mailing list
>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >   UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> >   http://www.freeswitch.org
>>>>>>> >
>>>>>>> >
>>>>>>> >   ________________________________
>>>>>>> >   _______________________________________________
>>>>>>> >  FreeSWITCH-users  mailing list
>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >   UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> >   http://www.freeswitch.org
>>>>>>> >
>>>>>>> >   _______________________________________________
>>>>>>> >  FreeSWITCH-users  mailing list
>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >   UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> >   http://www.freeswitch.org
>>>>>>> >
>>>>>>> >
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users   mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>>  
>>>>>>  
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users   mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>>  
>>>>>  
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users   mailing  list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>> 
>>>>>  
>>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users  mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>>  
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users  mailing  list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de

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From michelhabib at gmail.com  Mon Mar  7 12:55:32 2011
From: michelhabib at gmail.com (Michel Habib)
Date: Mon, 7 Mar 2011 11:55:32 +0200
Subject: [Freeswitch-users] ASR from Freeswitch to MS Speech Server [using
 MRCP Connector] - Audio Problem
Message-ID: <AANLkTikp0yM7Mpr3nQOjVT4_P_EoXeZZyFpiVo+CGNdf@mail.gmail.com>

Hello All,
I have MS OCS Speech Server 2007 [working correctly, as i can make SIP calls
and use its ASR and TTS Services successfully]
I am also using MRCP Connector from AumTech - which allows me to use ASR and
TTS Services through an MRCP Client .
Now, i am using Freeswitch mod unimrcp to use ASR and TTS.

for TTS, I can successfully make the call, the Audio RTP of the TTS voice is
transferred succesfully from Speech Server [through MRCP Connector] back to
the Freeswitch Server.
However, Freeswitch is not sending back the Audio RTP to the SIP client.

for ASR, I can successfully define the grammar and start recognition, but
the audio RTP sent to speech server [through MRCP Connector] is silent
[empty].

I am suspecting something is wrong with the RTP Configuration - can you help
me?

Let me now if you need any specific logs/scripts/configuration?

Thank you,
Michel.
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From simpot at simpot.com  Mon Mar  7 13:37:34 2011
From: simpot at simpot.com (Dmitry Saratsky)
Date: Mon, 7 Mar 2011 12:37:34 +0200
Subject: [Freeswitch-users] What is so dangeres in "freeswitch.msleep()"
 in LUA?
In-Reply-To: <AANLkTimhk603FH-ABd=ju4EgBNsgO3rUw1DqCePxK+2o@mail.gmail.com>
References: <52114D605A462A4E9A50E588EE7D0686012E575D9EEB@mail.forest.simpot.com>
	<AANLkTimhk603FH-ABd=ju4EgBNsgO3rUw1DqCePxK+2o@mail.gmail.com>
Message-ID: <52114D605A462A4E9A50E588EE7D0686012E575D9EEE@mail.forest.simpot.com>

Hi Steve,

Thank you for your reply.

The problem is that I need script to sleep, while session ready and transmits fax in same time until session will not be ready or fax transmitted. So I mean that I want script to sleep, not the session, just because my session may work in that time (transmit a fax). I think if I use session:sleep() in my case - my fax transmission will fail... Do you think I'm right? Do you think I can use freeswitch.msleep() in my case? And do you know which (if it will at all) troubles should I get?

Thanks,
Dmitry.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: 07 Mar 2011 11:17
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] What is so dangeres in "freeswitch.msleep()" in LUA?

AFAIK it stops all processing of the call during the sleep, you should
use session:sleep()

-Steve


On 6 March 2011 22:48, Dmitry Saratsky <simpot at simpot.com> wrote:
> Hi All,
>
>
>
> According to: http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.msleep
>
> It is not good to use: "freeswitch.msleep()" in session-based script:
>
> NOTE: Do not use this on a session-based script or bad things will happen.
>
>
>
> Can anyone explain please why not and what does author mean with
> "session-based script".
>
>
>
> I want to use it inside some function between I originate session and hangup
> it in the following way:
>
>
>
> ?? repeat
>
> --????? os.execute("sleep 1");
>
> ??????? freeswitch.msleep(1000);
>
> ??????? until ((not new_session:ready() == true) or
> (new_session:getVariable("fax_result_code") >= "0"))
>
> ?? new_session:hangup();
>
>
>
> Is it danger for me to use this?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

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


From marcdecorny at gmail.com  Mon Mar  7 15:36:51 2011
From: marcdecorny at gmail.com (Marc de Corny)
Date: Mon, 7 Mar 2011 12:36:51 +0000
Subject: [Freeswitch-users] Limit the size of queue in mod_fifo
Message-ID: <AANLkTinU_j8zRqKWvVfp=DXedPP83dSx09KRUbgo-rKb@mail.gmail.com>

Hi All,

does anyone know of a way of limiting the numbers of calls that are held in
the mod_fifo queue so that we can deny more calls after having reacded that
limit ?

thanks
Marc
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From peter.olsson at visionutveckling.se  Mon Mar  7 15:48:52 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 7 Mar 2011 13:48:52 +0100
Subject: [Freeswitch-users] What is so dangeres in "freeswitch.msleep()"
 in LUA?
In-Reply-To: <52114D605A462A4E9A50E588EE7D0686012E575D9EEE@mail.forest.simpot.com>
References: <52114D605A462A4E9A50E588EE7D0686012E575D9EEB@mail.forest.simpot.com>
	<AANLkTimhk603FH-ABd=ju4EgBNsgO3rUw1DqCePxK+2o@mail.gmail.com>
	<52114D605A462A4E9A50E588EE7D0686012E575D9EEE@mail.forest.simpot.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2E3EFAF@cooper>

My understanding is that Session:sleep() continues to handle the loop thread within the sleep. msleep() will stop the thread totally during the sleep, so I think you should use session:sleep().

/Peter

-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Dmitry Saratsky
Skickat: den 7 mars 2011 11:38
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] What is so dangeres in "freeswitch.msleep()" in LUA?

Hi Steve,

Thank you for your reply.

The problem is that I need script to sleep, while session ready and transmits fax in same time until session will not be ready or fax transmitted. So I mean that I want script to sleep, not the session, just because my session may work in that time (transmit a fax). I think if I use session:sleep() in my case - my fax transmission will fail... Do you think I'm right? Do you think I can use freeswitch.msleep() in my case? And do you know which (if it will at all) troubles should I get?

Thanks,
Dmitry.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: 07 Mar 2011 11:17
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] What is so dangeres in "freeswitch.msleep()" in LUA?

AFAIK it stops all processing of the call during the sleep, you should
use session:sleep()

-Steve


On 6 March 2011 22:48, Dmitry Saratsky <simpot at simpot.com> wrote:
> Hi All,
>
>
>
> According to: http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.msleep
>
> It is not good to use: "freeswitch.msleep()" in session-based script:
>
> NOTE: Do not use this on a session-based script or bad things will happen.
>
>
>
> Can anyone explain please why not and what does author mean with
> "session-based script".
>
>
>
> I want to use it inside some function between I originate session and hangup
> it in the following way:
>
>
>
> ?? repeat
>
> --????? os.execute("sleep 1");
>
> ??????? freeswitch.msleep(1000);
>
> ??????? until ((not new_session:ready() == true) or
> (new_session:getVariable("fax_result_code") >= "0"))
>
> ?? new_session:hangup();
>
>
>
> Is it danger for me to use this?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

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

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

!DSPAM:4d74b6b532762805214069!



From hkalyoncu at gmail.com  Mon Mar  7 15:52:38 2011
From: hkalyoncu at gmail.com (hkalyoncu)
Date: Mon, 7 Mar 2011 04:52:38 -0800 (PST)
Subject: [Freeswitch-users] mod_python
In-Reply-To: <1299347178939-6092296.post@n2.nabble.com>
References: <1299244469418-6088430.post@n2.nabble.com>
	<AANLkTikQ4NVFE2BLNYxiY05zFaYeq8TJjqbuZJKp3OgG@mail.gmail.com>
	<1299347178939-6092296.post@n2.nabble.com>
Message-ID: <1299502358077-6097179.post@n2.nabble.com>

ok i made some progress.

i can get the serialized string of all variables with

Event.serialize(params)  method

But when i try to access directly each channel variables like

params["Caller-Caller-ID-Name"] 

it returns error.

is there any way to access each channel variable?

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-python-tp6088430p6097179.html
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From mitch.johnson7 at gmail.com  Mon Mar  7 16:10:06 2011
From: mitch.johnson7 at gmail.com (mitch Johnson)
Date: Mon, 7 Mar 2011 08:10:06 -0500
Subject: [Freeswitch-users] SRTP
Message-ID: <AANLkTi=zyQpYdvg=ZXYmtrDn7-SVwYArfHKOAAstjxAr@mail.gmail.com>

No.

From: "Don WItt" <witt at cylogistics.com>
To: "'FreeSWITCH Users Help'" <freeswitch-users at lists.freeswitch.org>
Date: Sun, 6 Mar 2011 20:45:09 -0800
Subject: Re: [Freeswitch-users] SRTP
Is your dad's name Jerry?

Don Witt

Cylogistics
809B Cuesta Dr. #2149
Mountain View, CA 94040
650-694-4949 X 9102
Fax: 650-694-4953
witt at cylogistics.com
http://www.cylogistics.com
---------------------------------------------------
                  Suppliers of award winning
                              Ring Carrier


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitch
Johnson
Sent: Sunday, March 06, 2011 6:50 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] SRTP

When I dial 9664 to test the tls/srtp configuration it says that the call is
secure, however, when I dial another phone configured for tls/srtp the call
doesn't go through, the automated attendant comes online to say that the
extension is not available and then puts the call to voicemail.

I read the Secure RTP wiki, I do see a similar entries in the dialplan under
the extension name global, for both inbound and outbound:

<condition field="${sip_has_crypto}"
expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"
break="never">
       <action application="set" data="sip_secure_media=true"/>
       <!-- Offer SRTP on outbound legs if we have it on inbound. -->
       <action application="export" data="sip_secure_media=true"/>
     </condition>

So if I can make the test call to 9664 on both phones, which I assume is
using the inbound part by connecting the call.

Any help in figuring this out would be greatly appreciated.

BTW, I did buy the book, but there's no mention of SRTP/TLS in there.

Thanks,

Mitch
_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From leon at scarlet-internet.nl  Mon Mar  7 16:59:47 2011
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Mon, 7 Mar 2011 14:59:47 +0100
Subject: [Freeswitch-users] static rtpmap entries are not shown in
	outbound invites
In-Reply-To: <0D61D75E-A552-49F3-8A76-F3010BBE32B7@freeswitch.org>
References: <DD73E3C2-BFA2-406E-B3B5-0078AECCD329@scarlet-internet.nl>
	<0D61D75E-A552-49F3-8A76-F3010BBE32B7@freeswitch.org>
Message-ID: <5363DF5E-9569-4FEA-8C68-BB87CB87EBA2@scarlet-internet.nl>

Hi Brian,

Thanks, that works :-)

I got one last question about sdp, would you know why chan var ep_codec_string isn't set after an incoming invite with the following sdp:

v=0.
o=FOOBAR 123456 654321 IN IP4 1.2.3.4.
s=-.
c=IN IP4 1.2.3.4.
t=0 0.
m=audio 22262 RTP/AVP 18 101.
a=rtpmap:18 G729/8000/1.
a=rtpmap:101 telephone-event/8000/1.
a=fmtp:101 0-15.

Can it be the trailing /1 (encoding parameters) on both rtpmaps ? - According to parse_rtpmap() in libsofia it should be parsed alright, but still, ep_codec_string gets filled when I receive an SDP like this (from another endpoint):

v=0.
o=BARFOO 20494030 20494030 IN IP4 4.3.2.1.
s=-.
c=IN IP4 4.3.2.1.
t=0 0.
m=audio 50000 RTP/AVP 18 100.
a=rtpmap:18 G729/8000.
a=sendrecv.
a=ptime:20.
a=rtpmap:100 telephone-event/8000.
a=fmtp:100 0-15.


(btw, I'm having inbound- and outbound-codec-prefs on the incoming profile set as: "G7221 at 32000h,G7221 at 16000h,G722,PCMA,PCMU,G729,GSM" and am doing inbound-late-negotiation="true")


Thanks,

Leon




On Mar 4, 2011, at 7:09 PM, Brian West wrote:

> verbose_sdp=true
> 
> docs/ChangeLog: mod_sofia: apparently some sip device vendors did not read the RFC (who knew?) adding verbose_sdp=true var to add needless a= lines for standard iana codecs that explicitly do not require them (r:6c4f49a8)
> 
> /b
> 
> On Mar 4, 2011, at 12:06 PM, Leon de Rooij wrote:
> 
>> Can someone help me with this ?
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From cmrienzo at gmail.com  Mon Mar  7 17:32:51 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Mon, 7 Mar 2011 09:32:51 -0500
Subject: [Freeswitch-users] ASR from Freeswitch to MS Speech Server
 [using MRCP Connector] - Audio Problem
In-Reply-To: <AANLkTikp0yM7Mpr3nQOjVT4_P_EoXeZZyFpiVo+CGNdf@mail.gmail.com>
References: <AANLkTikp0yM7Mpr3nQOjVT4_P_EoXeZZyFpiVo+CGNdf@mail.gmail.com>
Message-ID: <AANLkTi=dhToGKtySAzT6GhfD026hhCw7edEKzoKgYqE=@mail.gmail.com>

Do you get audio between FS and your SIP client when not using ASR/TTS?

Show me the MRCP profile configuration and your FreeSWITCH logs during the
call.



On Mon, Mar 7, 2011 at 4:55 AM, Michel Habib <michelhabib at gmail.com> wrote:

> Hello All,
> I have MS OCS Speech Server 2007 [working correctly, as i can make SIP
> calls and use its ASR and TTS Services successfully]
> I am also using MRCP Connector from AumTech - which allows me to use ASR
> and TTS Services through an MRCP Client .
> Now, i am using Freeswitch mod unimrcp to use ASR and TTS.
>
> for TTS, I can successfully make the call, the Audio RTP of the TTS voice
> is transferred succesfully from Speech Server [through MRCP Connector] back
> to the Freeswitch Server.
> However, Freeswitch is not sending back the Audio RTP to the SIP client.
>
> for ASR, I can successfully define the grammar and start recognition, but
> the audio RTP sent to speech server [through MRCP Connector] is silent
> [empty].
>
> I am suspecting something is wrong with the RTP Configuration - can you
> help me?
>
> Let me now if you need any specific logs/scripts/configuration?
>
> Thank you,
> Michel.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mitch.capper at gmail.com  Mon Mar  7 19:05:17 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Mon, 7 Mar 2011 08:05:17 -0800
Subject: [Freeswitch-users] srtp clarification
In-Reply-To: <F5DB05D0-2D8E-4D91-8713-C03414CF3F43@gmail.com>
References: <mailman.4942.1299443642.24555.freeswitch-users@lists.freeswitch.org>
	<F5DB05D0-2D8E-4D91-8713-C03414CF3F43@gmail.com>
Message-ID: <AANLkTi=fP+WF=pdNbb-L7hcZLztvK5NM6ZQ8Et_MJk4q@mail.gmail.com>

Hi Mitch,
We will be updating the SRTP/TLS guides shortly with some additional
information.  Please stop by the IRC channel we can certainly
troubleshoot a bit faster there  (I am MitchCapper on irc).   What
error are you having with SRTP?  Make sure you have the jitter buffer
disabled as right now it will break SRTP.  Finally you can use sofia
loglevel tport 9 to turn most of the encryption layer debugging log
messages up and see if you see anything there.  Also make sure you are
running trunk if you can, there were some other bugs with TLS/SRTP
that were only fixed recently.

~Mitch

On Sun, Mar 6, 2011 at 1:18 PM, Mitch Johnson <mitch.johnson7 at gmail.com> wrote:
> I do understand the need for tls, I have no issues with tls, it works fine,
> it's the srtp I haven't managed to get working.
> Thanks for your reply.
> Mitch
>
> From: Steven Ayre <steveayre at gmail.com>
> Date: March 6, 2011 2:05:17 PM EST
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] srtp clarification
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>
>
> The problem comes in when I require SRTP only on the phones.
>
> If you use SRTP without TLS, you get no security at all. The
> encryption key used for the SRTP is passed within the SIP signalling.
> Unless you encrypt that then anyone intercepting the call can get the
> key from the signalling and then decrypt the media at will.
>
> -Steve
>
>
>
> On 6 March 2011 16:43, Mitch Johnson <mitch.johnson7 at gmail.com> wrote:
>
> My previous post may have suggested that the TLS/SRTP was not working.
> ?Where in fact, the TLS works like a charm.
>
> The problem comes in when I require SRTP only on the phones. ?When SRTP s
> turned off it works great, and so does TLS.
>
> I've been trying to understand how the voice part of the call is setup using
> SRTP. ?When I go through the logs, I don't see anything that says that SRTP
> failed anywhere. ?I'm pretty sure it's somewhere in my configuration. ?In
> Asterisk I had to define the transport mechanism of tls and encryption=yes
> to make it supposed to work. ?But then I never got it working there either,
> the difference with Asterisk is that it was showing SRTP as failing, but
> there's a bug causing that so it was pretty much a brick wall for me.
>
> Am I supposed to do something under the user profile or somewhere else where
> that call is encrypted using SRTP? ?I followed the TLS and SRTP guides to do
> the setup.
>
> Any help on this would be greatly appreciated. ?As with any problem, it's
> consuming my life until I can sort it out.
>
> Thanks so much ahead of time,
>
> Mitch
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From julf at julf.com  Mon Mar  7 19:12:37 2011
From: julf at julf.com (Johan Helsingius)
Date: Mon, 07 Mar 2011 17:12:37 +0100
Subject: [Freeswitch-users] tones.conf parameters?
Message-ID: <4D7503F5.6010606@julf.com>

Hi!

Trying to use freeTDM with FreeSWITCH, but the supplied tones.conf
only has setups for US, SG and RU. I found the ones for India
on the wiki, but haven't found anything for NL (that I need),
nor any other European country. Any pointers to how to configure
tones.conf for NL? Any way to use stuff from asterisk?

	Julf


From gilles.gerlinger at free.fr  Mon Mar  7 19:21:24 2011
From: gilles.gerlinger at free.fr (gigerlin)
Date: Mon, 7 Mar 2011 08:21:24 -0800 (PST)
Subject: [Freeswitch-users] Deactivating write resampler
In-Reply-To: <AANLkTikJcUnKihxEh8yXXV5SD2GHg3hydFNUM+z9G4gO@mail.gmail.com>
References: <1299411047906-6094024.post@n2.nabble.com>
	<AANLkTikJcUnKihxEh8yXXV5SD2GHg3hydFNUM+z9G4gO@mail.gmail.com>
Message-ID: <1299514884432-6097900.post@n2.nabble.com>

Ok, understood!

Thank you Anthony,
Gilles Gerlinger

--
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From wstephen80 at gmail.com  Mon Mar  7 19:24:51 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Mon, 7 Mar 2011 17:24:51 +0100
Subject: [Freeswitch-users] Avoid Loops
Message-ID: <AANLkTi=dza0zK4kRuT8WsPhA1xiTa3SQO2FRLmkOMGqw@mail.gmail.com>

In some situation there are calls that loops among some providers and I want
avoid that situation.
For example, I receive the call from my client A, I send this call to my
provider B but for some reason the call return to me (for example because B
sends the call to A and A resends the call to me).
In case of loop for me the solution can be simply to drop the call so my
client can reroute to a different provider.
There is anyone that can suggest me a way to drop looped calls?


Stephen
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From grsingh750 at gmail.com  Mon Mar  7 19:33:20 2011
From: grsingh750 at gmail.com (guru singh)
Date: Mon, 7 Mar 2011 22:03:20 +0530
Subject: [Freeswitch-users] tones.conf parameters?
In-Reply-To: <4D7503F5.6010606@julf.com>
References: <4D7503F5.6010606@julf.com>
Message-ID: <AANLkTin0N+79ZOQc8pMHw4U+CRS0Rc+LQ27YiPEuo0Bp@mail.gmail.com>

Hi Johan,

Another way would be to use the tone_detect application.
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect

I found this handy as well
http://www.3amsystems.com/wireline/tone-search.htm

guru

On Mon, Mar 7, 2011 at 9:42 PM, Johan Helsingius <julf at julf.com> wrote:
> Hi!
>
> Trying to use freeTDM with FreeSWITCH, but the supplied tones.conf
> only has setups for US, SG and RU. I found the ones for India
> on the wiki, but haven't found anything for NL (that I need),
> nor any other European country. Any pointers to how to configure
> tones.conf for NL? Any way to use stuff from asterisk?
>
> ? ? ? ?Julf
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From krice at freeswitch.org  Mon Mar  7 19:37:55 2011
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 07 Mar 2011 10:37:55 -0600
Subject: [Freeswitch-users] Avoid Loops
In-Reply-To: <AANLkTi=dza0zK4kRuT8WsPhA1xiTa3SQO2FRLmkOMGqw@mail.gmail.com>
Message-ID: <C99A6603.126FA%krice@freeswitch.org>

I hate this problem... Mod_limit matching ani+dnis doing rate limiting (say
5 calls with matchig ani+dnis in 2 seconds) can kill the loop and either
allow you to re-route or allow then to re-route... This happens a lot more
than you thing between 2nd and 3rd tier aggregators



On 3/7/11 10:24 AM, "Stephen Wilde" <wstephen80 at gmail.com> wrote:

> In some situation there are calls that loops among some providers and I want
> avoid that situation.
> For example, I receive the call from my client A, I send this call to my
> provider B but for some reason the call return to me (for example because B
> sends the call to A and A resends the call to me).
> In case of loop for me the solution can be simply to drop the call so my
> client can reroute to a different provider.
> There is anyone that can suggest me a way to drop looped calls?
> 
> 
> Stephen
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From wstephen80 at gmail.com  Mon Mar  7 20:01:21 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Mon, 7 Mar 2011 18:01:21 +0100
Subject: [Freeswitch-users] Avoid Loops
In-Reply-To: <C99A6603.126FA%krice@freeswitch.org>
References: <AANLkTi=dza0zK4kRuT8WsPhA1xiTa3SQO2FRLmkOMGqw@mail.gmail.com>
	<C99A6603.126FA%krice@freeswitch.org>
Message-ID: <AANLkTinV1QO1=44vyaoeALnzJQFowAcnGY_yqLyN3BJ1@mail.gmail.com>

Thank you for your suggestion! Good idea! I was searching for more
complicated solutions but mod_limit is perfect!

You say to use rate limiting but why not simply limit the calls number
instead of rate?

For example a row in dialplan as:

<action application="limit" data="hash inbound ${destination_number} 1" />

can avoid loops?

Stephen


On Mon, Mar 7, 2011 at 5:37 PM, Ken Rice <krice at freeswitch.org> wrote:

>  I hate this problem... Mod_limit matching ani+dnis doing rate limiting
> (say 5 calls with matchig ani+dnis in 2 seconds) can kill the loop and
> either allow you to re-route or allow then to re-route... This happens a lot
> more than you thing between 2nd and 3rd tier aggregators
>
>
>
>
> On 3/7/11 10:24 AM, "Stephen Wilde" <wstephen80 at gmail.com> wrote:
>
> In some situation there are calls that loops among some providers and I
> want avoid that situation.
> For example, I receive the call from my client A, I send this call to my
> provider B but for some reason the call return to me (for example because B
> sends the call to A and A resends the call to me).
> In case of loop for me the solution can be simply to drop the call so my
> client can reroute to a different provider.
> There is anyone that can suggest me a way to drop looped calls?
>
>
> Stephen
>
>
> ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From krice at freeswitch.org  Mon Mar  7 20:14:02 2011
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 07 Mar 2011 11:14:02 -0600
Subject: [Freeswitch-users] Avoid Loops
In-Reply-To: <AANLkTinV1QO1=44vyaoeALnzJQFowAcnGY_yqLyN3BJ1@mail.gmail.com>
Message-ID: <C99A6E7A.126FE%krice@freeswitch.org>

Well you could do that, but lets say you have a client that sends you
several calls to a conf number... Also when you loop calls they usually loop
fairly fast, by using a rate limit based on the ani and dnis you can still
let additional calls pass while reducing the chances of getting loop due to
call_id rewrites that are happening at the b2buas that are causing the loop

K


On 3/7/11 11:01 AM, "Stephen Wilde" <wstephen80 at gmail.com> wrote:

> Thank you for your suggestion!?Good idea! I was searching for more complicated
> solutions but mod_limit is perfect!
> 
> You say to use rate limiting but why not simply limit the calls number instead
> of rate?
> 
> For example a row in dialplan as:
> 
> <action application="limit" data="hash inbound ${destination_number} 1" />
> 
> can avoid loops?
> 
> Stephen
> 
> 
> On Mon, Mar 7, 2011 at 5:37 PM, Ken Rice <krice at freeswitch.org> wrote:
>> I hate this problem... Mod_limit matching ani+dnis doing rate limiting (say 5
>> calls with matchig ani+dnis in 2 seconds) can kill the loop and either allow
>> you to re-route or allow then to re-route... This happens a lot more than you
>> thing between 2nd and 3rd tier aggregators
>> 
>> 
>> 
>> 
>> On 3/7/11 10:24 AM, "Stephen Wilde" <wstephen80 at gmail.com
>> <http://wstephen80 at gmail.com> > wrote:
>> 
>>> In some situation there are calls that loops among some providers and I want
>>> avoid that situation.
>>> For example, I receive the call from my client A, I send this call to my
>>> provider B but for some reason the call return to me (for example because B
>>> sends the call to A and A resends the call to me).
>>> In case of loop for me the solution can be simply to drop the call so my
>>> client can reroute to a different provider.
>>> There is anyone that can suggest me a way to drop looped calls?
>>> 
>>> 
>>> Stephen
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From krice at freeswitch.org  Mon Mar  7 20:14:44 2011
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 07 Mar 2011 11:14:44 -0600
Subject: [Freeswitch-users] Polycom Phones
Message-ID: <C99A6EA4.126FE%krice@freeswitch.org>

Anyone out there selling polycom phones? Looking for 40 to 60 of them. Drop
me a note off list

K
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From sunwood360 at gmail.com  Mon Mar  7 20:24:12 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Mon, 7 Mar 2011 09:24:12 -0800
Subject: [Freeswitch-users] send fax via google voice.
Message-ID: <AANLkTik58RREBEXwzMsUL6djbFMQzyVt=ddJZc0rJO0B@mail.gmail.com>

>From the perspective of end user,  it is pretty easier to hacker an end
user' inbox globally.  However, you need physical presence to grab fax
documents.

Interesting to see the view from IT people is so different from
non-technical people. After, any existence has its reason.
On Mar 6, 2011 8:56 AM, "Ken Rice" <krice at freeswitch.org> wrote:
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From krice at freeswitch.org  Mon Mar  7 20:38:18 2011
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 07 Mar 2011 11:38:18 -0600
Subject: [Freeswitch-users] send fax via google voice.
In-Reply-To: <AANLkTik58RREBEXwzMsUL6djbFMQzyVt=ddJZc0rJO0B@mail.gmail.com>
Message-ID: <C99A742A.12703%krice@freeswitch.org>

Whats the old saying? Ignorance is bliss? Heh

Just because and end user perceives it as being more secure doesn?t make it
so... Even faxing over voip you just tcpdump out the rtp and process it thru
and you can recover then entire exchange... Same thing with a laptop, a
sound card, and a wav recorder with a cheap passive phone tap... Only
difference with legacy PSTN vs VoIP is its easier to hack a box somewhere
along the way and intercept the traffic vs finding the trunk line carrying
the traffic, however, it doesn?t take long to figure out that there are tons
of unsecure access points to the wire between your office and the CO...
(even the fiber these days can be compromised relatively easily compared to
what it was 10 years ago)

My whole point here is I am not malicious when it comes to this sort of
thing but I know what can be done and how cheap the technology to pull off
these things really is. I smart but I am far from the smartest guy and if I
can figure out how to do this stuff many people can figure it out...

I?m still say that the only secure computer is one fired into the sun on the
tip of a rocket with no comms back to earth.. Then that?s only secure until
someone figures out how to retrieve it.

K



On 3/7/11 11:24 AM, "envelopes envelopes" <sunwood360 at gmail.com> wrote:

> From the perspective of end user,? it is pretty easier to hacker an end user'
> inbox globally.? However, you need physical presence to grab fax documents.
> 
> Interesting to see the view from IT people is so different from non-technical
> people. After, any existence has its reason.
> 
> On Mar 6, 2011 8:56 AM, "Ken Rice" <krice at freeswitch.org> wrote:
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From wstephen80 at gmail.com  Mon Mar  7 20:41:15 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Mon, 7 Mar 2011 18:41:15 +0100
Subject: [Freeswitch-users] Avoid Loops
In-Reply-To: <C99A6E7A.126FE%krice@freeswitch.org>
References: <AANLkTinV1QO1=44vyaoeALnzJQFowAcnGY_yqLyN3BJ1@mail.gmail.com>
	<C99A6E7A.126FE%krice@freeswitch.org>
Message-ID: <AANLkTimX4aXh1XxyNuvC9qgmiu9pCC_CYkcFWmV+0OiS@mail.gmail.com>

Ok, so a line like:

<action application="limit" data="hash inbound ${destination_number} 1/5
!NORMAL_CIRCUIT_CONGESTION" />

avoid loops because Freeswitch doesn't accept more than 1 call each 5
seconds (the loops are very fast) but allow to have concurrent calls to the
same destination number.
Thank you Ken!


Stephen


On Mon, Mar 7, 2011 at 6:14 PM, Ken Rice <krice at freeswitch.org> wrote:

>  Well you could do that, but lets say you have a client that sends you
> several calls to a conf number... Also when you loop calls they usually loop
> fairly fast, by using a rate limit based on the ani and dnis you can still
> let additional calls pass while reducing the chances of getting loop due to
> call_id rewrites that are happening at the b2buas that are causing the loop
>
> K
>
>
>
> On 3/7/11 11:01 AM, "Stephen Wilde" <wstephen80 at gmail.com> wrote:
>
> Thank you for your suggestion! Good idea! I was searching for more
> complicated solutions but mod_limit is perfect!
>
> You say to use rate limiting but why not simply limit the calls number
> instead of rate?
>
> For example a row in dialplan as:
>
> <action application="limit" data="hash inbound ${destination_number} 1" />
>
> can avoid loops?
>
> Stephen
>
>
> On Mon, Mar 7, 2011 at 5:37 PM, Ken Rice <krice at freeswitch.org> wrote:
>
> I hate this problem... Mod_limit matching ani+dnis doing rate limiting (say
> 5 calls with matchig ani+dnis in 2 seconds) can kill the loop and either
> allow you to re-route or allow then to re-route... This happens a lot more
> than you thing between 2nd and 3rd tier aggregators
>
>
>
>
> On 3/7/11 10:24 AM, "Stephen Wilde" <wstephen80 at gmail.com <
> http://wstephen80 at gmail.com> > wrote:
>
> In some situation there are calls that loops among some providers and I
> want avoid that situation.
> For example, I receive the call from my client A, I send this call to my
> provider B but for some reason the call return to me (for example because B
> sends the call to A and A resends the call to me).
> In case of loop for me the solution can be simply to drop the call so my
> client can reroute to a different provider.
> There is anyone that can suggest me a way to drop looped calls?
>
>
> Stephen
>
>
> ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <
> http://FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From curriegrad2004 at gmail.com  Mon Mar  7 20:43:13 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 7 Mar 2011 09:43:13 -0800
Subject: [Freeswitch-users] send fax via google voice.
In-Reply-To: <C99A742A.12703%krice@freeswitch.org>
References: <AANLkTik58RREBEXwzMsUL6djbFMQzyVt=ddJZc0rJO0B@mail.gmail.com>
	<C99A742A.12703%krice@freeswitch.org>
Message-ID: <AANLkTiks9kco2ZvQxR8k-spJJ0TONwmdTgxPOD8O2HZ4@mail.gmail.com>

And you forgot to mention that telco's might not have a well secured
system at all. Soft taps have been around for quite some time in the
POTS world... So anybody can tap on to the conversations effortlessly.

On Mon, Mar 7, 2011 at 9:38 AM, Ken Rice <krice at freeswitch.org> wrote:
> Whats the old saying? Ignorance is bliss? Heh
>
> Just because and end user perceives it as being more secure doesn?t make it
> so... Even faxing over voip you just tcpdump out the rtp and process it thru
> and you can recover then entire exchange... Same thing with a laptop, a
> sound card, and a wav recorder with a cheap passive phone tap... Only
> difference with legacy PSTN vs VoIP is its easier to hack a box somewhere
> along the way and intercept the traffic vs finding the trunk line carrying
> the traffic, however, it doesn?t take long to figure out that there are tons
> of unsecure access points to the wire between your office and the CO...
> (even the fiber these days can be compromised relatively easily compared to
> what it was 10 years ago)
>
> My whole point here is I am not malicious when it comes to this sort of
> thing but I know what can be done and how cheap the technology to pull off
> these things really is. I smart but I am far from the smartest guy and if I
> can figure out how to do this stuff many people can figure it out...
>
> I?m still say that the only secure computer is one fired into the sun on the
> tip of a rocket with no comms back to earth.. Then that?s only secure until
> someone figures out how to retrieve it.
>
> K
>
>
>
> On 3/7/11 11:24 AM, "envelopes envelopes" <sunwood360 at gmail.com> wrote:
>
> From the perspective of end user,? it is pretty easier to hacker an end
> user' inbox globally.? However, you need physical presence to grab fax
> documents.
>
> Interesting to see the view from IT people is so different from
> non-technical people. After, any existence has its reason.
>
> On Mar 6, 2011 8:56 AM, "Ken Rice" <krice at freeswitch.org> wrote:
>
> ________________________________
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From curriegrad2004 at gmail.com  Mon Mar  7 20:46:53 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 7 Mar 2011 09:46:53 -0800
Subject: [Freeswitch-users] Scientific Linux 6.0 - Released
Message-ID: <AANLkTik0PiZivuHtTxdwK6-uQrU6=_muxV=+dd_eu5WL@mail.gmail.com>

As some of you may already know, Scientific Linux 6.0 has been finally
released. (March 3 2011 to be precise.)

Scientific Linux 6 is a RHEL 6 clone. Surprisingly CentOS 6 should
also a RHEL 6 clone. If anyone wants to have a go to see how
FreeSwitch runs on a RHEL/CentOS 6 like box, then by all means,
finally there is a RHEL 6 clone that should be fully compatible with
the upstream sources.


From steveayre at gmail.com  Mon Mar  7 20:57:59 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 7 Mar 2011 17:57:59 +0000
Subject: [Freeswitch-users] static rtpmap entries are not shown in
 outbound invites
In-Reply-To: <5363DF5E-9569-4FEA-8C68-BB87CB87EBA2@scarlet-internet.nl>
References: <DD73E3C2-BFA2-406E-B3B5-0078AECCD329@scarlet-internet.nl>
	<0D61D75E-A552-49F3-8A76-F3010BBE32B7@freeswitch.org>
	<5363DF5E-9569-4FEA-8C68-BB87CB87EBA2@scarlet-internet.nl>
Message-ID: <AANLkTimF+rofWtwTN7PaRY5t9_xbgc6qYfMz4s8i6se6@mail.gmail.com>

> Can it be the trailing /1 (encoding parameters) on both rtpmaps ? - According to parse_rtpmap() in libsofia it should be parsed alright, but still, ep_codec_string gets filled when I receive an SDP like this (from another endpoint):

That means 1 channel (mono), which is the default. Although optional
it is correct and *should* be handled fine...

-Steve



On 7 March 2011 13:59, Leon de Rooij <leon at scarlet-internet.nl> wrote:
> Hi Brian,
>
> Thanks, that works :-)
>
> I got one last question about sdp, would you know why chan var ep_codec_string isn't set after an incoming invite with the following sdp:
>
> v=0.
> o=FOOBAR 123456 654321 IN IP4 1.2.3.4.
> s=-.
> c=IN IP4 1.2.3.4.
> t=0 0.
> m=audio 22262 RTP/AVP 18 101.
> a=rtpmap:18 G729/8000/1.
> a=rtpmap:101 telephone-event/8000/1.
> a=fmtp:101 0-15.
>
> Can it be the trailing /1 (encoding parameters) on both rtpmaps ? - According to parse_rtpmap() in libsofia it should be parsed alright, but still, ep_codec_string gets filled when I receive an SDP like this (from another endpoint):
>
> v=0.
> o=BARFOO 20494030 20494030 IN IP4 4.3.2.1.
> s=-.
> c=IN IP4 4.3.2.1.
> t=0 0.
> m=audio 50000 RTP/AVP 18 100.
> a=rtpmap:18 G729/8000.
> a=sendrecv.
> a=ptime:20.
> a=rtpmap:100 telephone-event/8000.
> a=fmtp:100 0-15.
>
>
> (btw, I'm having inbound- and outbound-codec-prefs on the incoming profile set as: "G7221 at 32000h,G7221 at 16000h,G722,PCMA,PCMU,G729,GSM" and am doing inbound-late-negotiation="true")
>
>
> Thanks,
>
> Leon
>
>
>
>
> On Mar 4, 2011, at 7:09 PM, Brian West wrote:
>
>> verbose_sdp=true
>>
>> docs/ChangeLog: mod_sofia: apparently some sip device vendors did not read the RFC (who knew?) adding verbose_sdp=true var to add needless a= lines for standard iana codecs that explicitly do not require them (r:6c4f49a8)
>>
>> /b
>>
>> On Mar 4, 2011, at 12:06 PM, Leon de Rooij wrote:
>>
>>> Can someone help me with this ?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From julf at julf.com  Mon Mar  7 21:01:14 2011
From: julf at julf.com (Johan Helsingius)
Date: Mon, 07 Mar 2011 19:01:14 +0100
Subject: [Freeswitch-users] tones.conf parameters?
In-Reply-To: <AANLkTin0N+79ZOQc8pMHw4U+CRS0Rc+LQ27YiPEuo0Bp@mail.gmail.com>
References: <4D7503F5.6010606@julf.com>
	<AANLkTin0N+79ZOQc8pMHw4U+CRS0Rc+LQ27YiPEuo0Bp@mail.gmail.com>
Message-ID: <4D751D6A.70605@julf.com>

Hi, guru,

> Another way would be to use the tone_detect application.
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect

The problem is that I don't seem to get my DECT phone
to generate anything before it gets a dial tone.

> I found this handy as well
> http://www.3amsystems.com/wireline/tone-search.htm

That is helpful - but is the tones.conf format documented anywhere?

	Julf


From ovvenkatesan at gmail.com  Mon Mar  7 21:01:25 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Mon, 7 Mar 2011 23:31:25 +0530
Subject: [Freeswitch-users] not able to view ESL output on the web browser
Message-ID: <AANLkTind+xyiEHvO_tfrhymG_K5RHYMZpU0Z9KFPTNvJ@mail.gmail.com>

Hi,

I have installed ESL phpmod module on fedora 13.
even though it was painful to installing it on fedora, some how i did it .

I can run and view the output of test.php on shell command.
like,
# *php test.php*

When, I copied test.php and ESL.php files into my web server  root
directory,
trying to to run it from web browser, it shows only blank page.
I am trying it figure it out, no luck.

Anyone please shed some light on this.


-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
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From oa at estation.dk  Mon Mar  7 11:46:32 2011
From: oa at estation.dk (=?ISO-8859-1?Q?=D8yvind_Albrigtsen?=)
Date: Mon, 07 Mar 2011 09:46:32 +0100
Subject: [Freeswitch-users] Snom and Aastra one-way audio after picking
 up call that has been on hold (NAT)
In-Reply-To: <4D70A2E6.4030703@estation.dk>
References: <4D6F596C.7010403@estation.dk>
	<30915FCC-D7B4-4BEC-850F-BA321615A5D6@freeswitch.org>
	<4D70A2E6.4030703@estation.dk>
Message-ID: <4D749B68.2040102@estation.dk>

Have you had the chance to investigate this? It seems to be using the 
local IP (from behind NAT), but it's working fine with regular calls so 
the issue is specific to putting calls on Hold.


Thanks in advance.

Regards
Oyvind

On 03/04/2011 09:29 AM, ?yvind Albrigtsen wrote:
> Here you go.
>
> 5328XXXX is the number the phone is called from (coming from a PSTN 
> gateway without any NAT).
>
>
> On 03/03/2011 11:34 PM, Brian West wrote:
>> sip trace please.
>>
>> /b
>>
>> On Mar 3, 2011, at 3:03 AM, ?yvind Albrigtsen wrote:
>>
>>> The calls get in like this:
>>> PSTN-gw ->  FS ->  NAT ->  Snom/Aastra
>>>
>>> It works fine when I call out from the phones and set the call on hold
>>> (on the Snom/Aastra phones), though.
>>>
>>> This is on FreeSWITCH Version 1.0.head (git-cb6f1ed 2011-02-22 20-25-16
>>> -0500)
>>>
>>>
>>> Regards
>>> Oyvind
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>



From johns1433 at gmail.com  Mon Mar  7 20:42:06 2011
From: johns1433 at gmail.com (John Smith)
Date: Mon, 7 Mar 2011 18:42:06 +0100
Subject: [Freeswitch-users] DTMF not taken into account within a conference
Message-ID: <AANLkTikrUYJAdEYjMOnhJmH9iiH87XdHNAumUEkdZUgK@mail.gmail.com>

Hi all,

I have an issue with DTMF within a conference with the latest versions of
FS.
When I enter a DTMF within a conference it only works the first time. For
example if I press 0 I?m muted but afterwards my DTMF commands are not taken
into account whatever I press.
I checked the logs and I have just mention for the first digit received:
switch_rtp.c:3237 RTP RECV DTMF 0:800

This issue seems only appear within a conference. I tested IVR demo and it
worked fine.
Before entering the conference, everything works also fine: all the digits
are received by FS.

I tried many recent builds and I always got the issue. I made tests today
with the latest git HEAD and I still have the problem.
I don?t have this issue with FS V1.0.6: all DTMF entered are taken into
account. I use exactly the same configuration and the same SIP client.

Has anyone an idea of what is going on?

Thanks

John
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From brian at freeswitch.org  Mon Mar  7 21:23:49 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 7 Mar 2011 12:23:49 -0600
Subject: [Freeswitch-users] DTMF not taken into account within a
	conference
In-Reply-To: <AANLkTikrUYJAdEYjMOnhJmH9iiH87XdHNAumUEkdZUgK@mail.gmail.com>
References: <AANLkTikrUYJAdEYjMOnhJmH9iiH87XdHNAumUEkdZUgK@mail.gmail.com>
Message-ID: <77DF4C66-1FCE-4034-A65F-14853076B3B5@freeswitch.org>

Well I have a feeling you haven't really updated to the VERY latest GIT HEAD have you?  We had one day this past month that had this EXACT bug in it and has since been fixed.

/b

On Mar 7, 2011, at 11:42 AM, John Smith wrote:

> When I enter a DTMF within a conference it only works the first time. For
> example if I press 0 I?m muted but afterwards my DTMF commands are not taken
> into account whatever I press.



From steveayre at gmail.com  Mon Mar  7 21:23:46 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 7 Mar 2011 18:23:46 +0000
Subject: [Freeswitch-users] not able to view ESL output on the web
	browser
In-Reply-To: <AANLkTind+xyiEHvO_tfrhymG_K5RHYMZpU0Z9KFPTNvJ@mail.gmail.com>
References: <AANLkTind+xyiEHvO_tfrhymG_K5RHYMZpU0Z9KFPTNvJ@mail.gmail.com>
Message-ID: <AANLkTikYvO0ioh8tXEfYtS3TJYW=_4cvbhpyvgZ1bAv6@mail.gmail.com>

What web server? Apache? Check the error log, perhaps there's a clue there.

-Steve


On 7 March 2011 18:01, ovvenkat <ovvenkatesan at gmail.com> wrote:
> Hi,
>
> I have installed ESL phpmod module on fedora 13.
> even though it was painful to installing it on fedora, some how i did it .
>
> I can run and view the output of test.php on shell command.
> like,
> # php test.php
>
> When, I copied test.php and ESL.php files into my web server? root
> directory,
> trying to to run it from web browser, it shows only blank page.
> I am trying it figure it out, no luck.
>
> Anyone please shed some light on this.
>
>
> --
>
> If you have come to help me, you are wasting your time.
> If you have come to because your liberation is bound up in mine, we can work
> together.
>
>
> Regards
> Venkatesan OV.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From infos at madovsky.org  Mon Mar  7 21:26:30 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 7 Mar 2011 13:26:30 -0500
Subject: [Freeswitch-users] not able to view ESL output on the web
	browser
References: <AANLkTind+xyiEHvO_tfrhymG_K5RHYMZpU0Z9KFPTNvJ@mail.gmail.com>
Message-ID: <3A7744BAF6A3453996F3F3AD2DF8BDE1@e1705>

activate php ERR message first

error_reporting(E_ERROR);

  ----- Original Message ----- 
  From: ovvenkat 
  To: FreeSWITCH Users Help 
  Sent: Monday, March 07, 2011 1:01 PM
  Subject: [Freeswitch-users] not able to view ESL output on the web browser


  Hi, 

  I have installed ESL phpmod module on fedora 13. 
  even though it was painful to installing it on fedora, some how i did it .

  I can run and view the output of test.php on shell command. 
  like, 
  # php test.php

  When, I copied test.php and ESL.php files into my web server  root directory, 
  trying to to run it from web browser, it shows only blank page.  
  I am trying it figure it out, no luck. 

  Anyone please shed some light on this. 


  -- 


  If you have come to help me, you are wasting your time. 
  If you have come to because your liberation is bound up in mine, we can work together.

   
  Regards
  Venkatesan OV.




------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From Nabble at slickdeals.endjunk.com  Mon Mar  7 22:09:48 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 7 Mar 2011 11:09:48 -0800 (PST)
Subject: [Freeswitch-users] freeswitch -nc -nosql
In-Reply-To: <1298688216720-6066718.post@n2.nabble.com>
References: <1298688216720-6066718.post@n2.nabble.com>
Message-ID: <1299524988820-6098582.post@n2.nabble.com>

Finaly, I found out that FreeSWITCH Version 1.0.head (git-88d410d 2011-02-11
20-15-06 -0600) is the last version that works with SQL. Starting from the
version after this one with a commit
a2c0da53f368f0b11340c3a72814c93b182753b7 crashes if -nosql switch is called
when freeswitch is launched. Here is the git log pertaining to the two
commits for your perusal and hope FS developers will be able to localize why
adding centralized registration db to core db and use it from mod_sofia
causes SQL to crash on ARM platform:

commit a2c0da53f368f0b11340c3a72814c93b182753b7
Author: Anthony Minessale 
Date:   Fri Feb 11 23:10:12 2011 -0600

    add centralized registration db to core db and use it from mod_sofia

commit 88d410d31485d13911f0958af5a73f1f6f49a454
Author: Anthony Minessale 
Date:   Fri Feb 11 20:15:06 2011 -0600

    fix uuid_jitterbuffer edge case debugging a non-existant jb causing a
seg


-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-nc-nosql-tp6066718p6098582.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Mon Mar  7 22:23:06 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 7 Mar 2011 13:23:06 -0600
Subject: [Freeswitch-users] freeswitch -nc -nosql
In-Reply-To: <1299524988820-6098582.post@n2.nabble.com>
References: <1298688216720-6066718.post@n2.nabble.com>
	<1299524988820-6098582.post@n2.nabble.com>
Message-ID: <AANLkTik6c5rHYnr6cxmsBzRCcc94VL=jmYtud29TVBzc@mail.gmail.com>

Will you please explain to me why you keep ignoring my request for backtraces?
There is a bug open already http://jira.freeswitch.org/browse/FS-3126
There is even a backtrace which can be very useful , now if you posted
your findings there we might even figure it out.....

Are you not receiving my many pleas to use JIRA for working on bugs?


On Mon, Mar 7, 2011 at 1:09 PM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
> Finaly, I found out that FreeSWITCH Version 1.0.head (git-88d410d 2011-02-11
> 20-15-06 -0600) is the last version that works with SQL. Starting from the
> version after this one with a commit
> a2c0da53f368f0b11340c3a72814c93b182753b7 crashes if -nosql switch is called
> when freeswitch is launched. Here is the git log pertaining to the two
> commits for your perusal and hope FS developers will be able to localize why
> adding centralized registration db to core db and use it from mod_sofia
> causes SQL to crash on ARM platform:
>
> commit a2c0da53f368f0b11340c3a72814c93b182753b7
> Author: Anthony Minessale
> Date: ? Fri Feb 11 23:10:12 2011 -0600
>
> ? ?add centralized registration db to core db and use it from mod_sofia
>
> commit 88d410d31485d13911f0958af5a73f1f6f49a454
> Author: Anthony Minessale
> Date: ? Fri Feb 11 20:15:06 2011 -0600
>
> ? ?fix uuid_jitterbuffer edge case debugging a non-existant jb causing a
> seg
>
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-nc-nosql-tp6066718p6098582.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 elijah at crankenstein.com  Mon Mar  7 22:28:24 2011
From: elijah at crankenstein.com (elijah)
Date: Mon, 7 Mar 2011 11:28:24 -0800
Subject: [Freeswitch-users] database is locked
Message-ID: <AANLkTimHPa50yukZfq9P5emxMP=5uGizdPiTVeiV4N0C@mail.gmail.com>

I updated (via Git) today an installation of FreeSWITCH I've been running
without incident for a few months. Upon restarting the FreeSWITCH process I
receive at a constant rate these errors:

...switch_core_sqldb.c xxx SQL ERR [database is locked]
...switch_core_sqldb.c xxx SQL ERR [database is locked]
...switch_core_sqldb.c xxx SQL ERR [database is locked]

How should I resolve?
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From ovvenkatesan at gmail.com  Mon Mar  7 22:31:50 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Tue, 8 Mar 2011 01:01:50 +0530
Subject: [Freeswitch-users] not able to view ESL output on the web
	browser
In-Reply-To: <AANLkTikYvO0ioh8tXEfYtS3TJYW=_4cvbhpyvgZ1bAv6@mail.gmail.com>
References: <AANLkTind+xyiEHvO_tfrhymG_K5RHYMZpU0Z9KFPTNvJ@mail.gmail.com>
	<AANLkTikYvO0ioh8tXEfYtS3TJYW=_4cvbhpyvgZ1bAv6@mail.gmail.com>
Message-ID: <AANLkTin7mU7oM0kAY5Wu+Vn-PxZ6ZLnZnLf38EavsFJs@mail.gmail.com>

>
> What web server? Apache? Check the error log, perhaps there's a clue there.
>
>
Thanks Steven for your replay,

When, I checked in the log file, I found fatal error, It says that

[Tue Mar 08 00:49:21 2011] [error] [client 127.0.0.1] PHP Fatal error:  Call
to undefined function dl() in /var/www/html/ESL/ESL.php on line 23

But, when I run the same file through shell command, its works fine.


Regards,
Venkat.




-Steve
>
>
> On 7 March 2011 18:01, ovvenkat <ovvenkatesan at gmail.com> wrote:
> > Hi,
> >
> > I have installed ESL phpmod module on fedora 13.
> > even though it was painful to installing it on fedora, some how i did it
> .
> >
> > I can run and view the output of test.php on shell command.
> > like,
> > # php test.php
> >
> > When, I copied test.php and ESL.php files into my web server  root
> > directory,
> > trying to to run it from web browser, it shows only blank page.
> > I am trying it figure it out, no luck.
> >
> > Anyone please shed some light on this.
> >
> >
> > --
> >
> > If you have come to help me, you are wasting your time.
> > If you have come to because your liberation is bound up in mine, we can
> work
> > together.
> >
> >
> > Regards
> > Venkatesan OV.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
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From anthony.minessale at gmail.com  Mon Mar  7 22:36:31 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 7 Mar 2011 13:36:31 -0600
Subject: [Freeswitch-users] database is locked
In-Reply-To: <AANLkTimHPa50yukZfq9P5emxMP=5uGizdPiTVeiV4N0C@mail.gmail.com>
References: <AANLkTimHPa50yukZfq9P5emxMP=5uGizdPiTVeiV4N0C@mail.gmail.com>
Message-ID: <AANLkTik3TC1ALNzYfM+xnP7CNR40EBUWT6N_6D8xsavE@mail.gmail.com>

why did you feel the need to xxx out the line numbers?

This is what exact revision, was it literally from today?
Do you have the exact excerpt from the logs not a paraphrased on?


On Mon, Mar 7, 2011 at 1:28 PM, elijah <elijah at crankenstein.com> wrote:
> I updated (via Git) today an installation of FreeSWITCH I've been running
> without incident for a few months. Upon restarting the FreeSWITCH process I
> receive at a constant rate these errors:
> ...switch_core_sqldb.c xxx SQL ERR [database is locked]
> ...switch_core_sqldb.c xxx SQL ERR [database is locked]
> ...switch_core_sqldb.c xxx SQL ERR [database is locked]
> How should I resolve?
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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  Mon Mar  7 22:42:31 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 7 Mar 2011 13:42:31 -0600
Subject: [Freeswitch-users] database is locked
In-Reply-To: <AANLkTik3TC1ALNzYfM+xnP7CNR40EBUWT6N_6D8xsavE@mail.gmail.com>
References: <AANLkTimHPa50yukZfq9P5emxMP=5uGizdPiTVeiV4N0C@mail.gmail.com>
	<AANLkTik3TC1ALNzYfM+xnP7CNR40EBUWT6N_6D8xsavE@mail.gmail.com>
Message-ID: <AANLkTi=kJpAT3Fia8TqiXatE7AORW4Zt508RMjoTJ_a=@mail.gmail.com>

ideally run

fsctl debug_level 10
console loglevel debug

then capture it and send it to me.


On Mon, Mar 7, 2011 at 1:36 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> why did you feel the need to xxx out the line numbers?
>
> This is what exact revision, was it literally from today?
> Do you have the exact excerpt from the logs not a paraphrased on?
>
>
> On Mon, Mar 7, 2011 at 1:28 PM, elijah <elijah at crankenstein.com> wrote:
>> I updated (via Git) today an installation of FreeSWITCH I've been running
>> without incident for a few months. Upon restarting the FreeSWITCH process I
>> receive at a constant rate these errors:
>> ...switch_core_sqldb.c xxx SQL ERR [database is locked]
>> ...switch_core_sqldb.c xxx SQL ERR [database is locked]
>> ...switch_core_sqldb.c xxx SQL ERR [database is locked]
>> How should I resolve?
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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 msc at freeswitch.org  Mon Mar  7 22:43:15 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 7 Mar 2011 11:43:15 -0800
Subject: [Freeswitch-users] not able to view ESL output on the web
	browser
In-Reply-To: <AANLkTin7mU7oM0kAY5Wu+Vn-PxZ6ZLnZnLf38EavsFJs@mail.gmail.com>
References: <AANLkTind+xyiEHvO_tfrhymG_K5RHYMZpU0Z9KFPTNvJ@mail.gmail.com>
	<AANLkTikYvO0ioh8tXEfYtS3TJYW=_4cvbhpyvgZ1bAv6@mail.gmail.com>
	<AANLkTin7mU7oM0kAY5Wu+Vn-PxZ6ZLnZnLf38EavsFJs@mail.gmail.com>
Message-ID: <AANLkTino-+XhpTOEADJTiG8jFPgca8Z_iWSxnTLSYu0j@mail.gmail.com>

did you do the phpmod-install command in the libs/esl/ directory?
-MC

On Mon, Mar 7, 2011 at 11:31 AM, ovvenkat <ovvenkatesan at gmail.com> wrote:

> What web server? Apache? Check the error log, perhaps there's a clue there.
>>
>>
> Thanks Steven for your replay,
>
> When, I checked in the log file, I found fatal error, It says that
>
> [Tue Mar 08 00:49:21 2011] [error] [client 127.0.0.1] PHP Fatal error:
> Call to undefined function dl() in /var/www/html/ESL/ESL.php on line 23
>
> But, when I run the same file through shell command, its works fine.
>
>
> Regards,
> Venkat.
>
>
>
>
> -Steve
>>
>>
>> On 7 March 2011 18:01, ovvenkat <ovvenkatesan at gmail.com> wrote:
>> > Hi,
>> >
>> > I have installed ESL phpmod module on fedora 13.
>> > even though it was painful to installing it on fedora, some how i did it
>> .
>> >
>> > I can run and view the output of test.php on shell command.
>> > like,
>> > # php test.php
>> >
>> > When, I copied test.php and ESL.php files into my web server  root
>> > directory,
>> > trying to to run it from web browser, it shows only blank page.
>> > I am trying it figure it out, no luck.
>> >
>> > Anyone please shed some light on this.
>> >
>> >
>> > --
>> >
>> > If you have come to help me, you are wasting your time.
>> > If you have come to because your liberation is bound up in mine, we can
>> work
>> > together.
>> >
>> >
>> > Regards
>> > Venkatesan OV.
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> If you have come to help me, you are wasting your time.
> If you have come to because your liberation is bound up in mine, we can
> work together.
>
>
> Regards
> Venkatesan OV.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From elijah at crankenstein.com  Mon Mar  7 22:50:29 2011
From: elijah at crankenstein.com (elijah)
Date: Mon, 7 Mar 2011 11:50:29 -0800
Subject: [Freeswitch-users] database is locked
In-Reply-To: <AANLkTik3TC1ALNzYfM+xnP7CNR40EBUWT6N_6D8xsavE@mail.gmail.com>
References: <AANLkTimHPa50yukZfq9P5emxMP=5uGizdPiTVeiV4N0C@mail.gmail.com>
	<AANLkTik3TC1ALNzYfM+xnP7CNR40EBUWT6N_6D8xsavE@mail.gmail.com>
Message-ID: <AANLkTikwupXBjumOXN7sVoeJH4hQ-1nuPMaB70oj_-iF@mail.gmail.com>

Of course. Sorry (I hastily thought I was replacing a counter):

2011-03-07 11:46:23.406465 [ERR] switch_core_sqldb.c:432 SQL ERR [database
is locked]
DELETE FROM members WHERE system = 'single_box' AND uuid = '' AND
(abandoned_epoch = '1299525498' OR joined_epoch = '1299525427')
2011-03-07 11:46:53.320898 [ERR] switch_core_sqldb.c:432 SQL ERR [database
is locked]
DELETE FROM members WHERE system = 'single_box' AND uuid = '' AND
(abandoned_epoch = '1299525498' OR joined_epoch = '1299525427')
2011-03-07 11:47:23.230337 [ERR] switch_core_sqldb.c:432 SQL ERR [database
is locked]
DELETE FROM members WHERE system = 'single_box' AND uuid = '' AND
(abandoned_epoch = '1299525498' OR joined_epoch = '1299525427')
2011-03-07 11:47:53.139777 [ERR] switch_core_sqldb.c:432 SQL ERR [database
is locked]
DELETE FROM members WHERE system = 'single_box' AND uuid = '' AND
(abandoned_epoch = '1299525498' OR joined_epoch = '1299525427')

The version that is producing this error for me is *1.0.head (git-bfd0ba9
2011-03-07 13-02-41 -0600)*
Additionally, I am running a previous version (*1.0.head (git-c6f044d
2011-02-28 10-02-11 -0600)*) that is working without this error.

Thank you



On Mon, Mar 7, 2011 at 11:36 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> why did you feel the need to xxx out the line numbers?
>
> This is what exact revision, was it literally from today?
> Do you have the exact excerpt from the logs not a paraphrased on?
>
>
> On Mon, Mar 7, 2011 at 1:28 PM, elijah <elijah at crankenstein.com> wrote:
> > I updated (via Git) today an installation of FreeSWITCH I've been running
> > without incident for a few months. Upon restarting the FreeSWITCH process
> I
> > receive at a constant rate these errors:
> > ...switch_core_sqldb.c xxx SQL ERR [database is locked]
> > ...switch_core_sqldb.c xxx SQL ERR [database is locked]
> > ...switch_core_sqldb.c xxx SQL ERR [database is locked]
> > How should I resolve?
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From peter.olsson at visionutveckling.se  Mon Mar  7 22:50:40 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 7 Mar 2011 20:50:40 +0100
Subject: [Freeswitch-users] database is locked
In-Reply-To: <AANLkTi=kJpAT3Fia8TqiXatE7AORW4Zt508RMjoTJ_a=@mail.gmail.com>
References: <AANLkTimHPa50yukZfq9P5emxMP=5uGizdPiTVeiV4N0C@mail.gmail.com>
	<AANLkTik3TC1ALNzYfM+xnP7CNR40EBUWT6N_6D8xsavE@mail.gmail.com>,
	<AANLkTi=kJpAT3Fia8TqiXatE7AORW4Zt508RMjoTJ_a=@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2C494F5@cooper>

There was also a similar issue posted at Jira today, related to mod_callcenter - FS-3127.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Anthony Minessale [anthony.minessale at gmail.com]
Skickat: den 7 mars 2011 20:42
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] database is locked

ideally run

fsctl debug_level 10
console loglevel debug

then capture it and send it to me.


On Mon, Mar 7, 2011 at 1:36 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> why did you feel the need to xxx out the line numbers?
>
> This is what exact revision, was it literally from today?
> Do you have the exact excerpt from the logs not a paraphrased on?
>
>
> On Mon, Mar 7, 2011 at 1:28 PM, elijah <elijah at crankenstein.com> wrote:
>> I updated (via Git) today an installation of FreeSWITCH I've been running
>> without incident for a few months. Upon restarting the FreeSWITCH process I
>> receive at a constant rate these errors:
>> ...switch_core_sqldb.c xxx SQL ERR [database is locked]
>> ...switch_core_sqldb.c xxx SQL ERR [database is locked]
>> ...switch_core_sqldb.c xxx SQL ERR [database is locked]
>> How should I resolve?
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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

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

!DSPAM:4d7535d532761229612182!



From steveayre at gmail.com  Mon Mar  7 22:55:02 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 7 Mar 2011 19:55:02 +0000
Subject: [Freeswitch-users] not able to view ESL output on the web
	browser
In-Reply-To: <AANLkTin7mU7oM0kAY5Wu+Vn-PxZ6ZLnZnLf38EavsFJs@mail.gmail.com>
References: <AANLkTind+xyiEHvO_tfrhymG_K5RHYMZpU0Z9KFPTNvJ@mail.gmail.com>
	<AANLkTikYvO0ioh8tXEfYtS3TJYW=_4cvbhpyvgZ1bAv6@mail.gmail.com>
	<AANLkTin7mU7oM0kAY5Wu+Vn-PxZ6ZLnZnLf38EavsFJs@mail.gmail.com>
Message-ID: <AANLkTikDkWFYFq_5d4u=rsb0KfAtUeFvkk+RB8=afLc2@mail.gmail.com>

>From the PHP documentation:
1) "This function has been removed from some SAPI's in PHP 5.3"
2) "dl() is not supported when PHP is built with ZTS support. Use the
Extension Loading Directives instead."
3) Note: This function is disabled when PHP is running in safe mode.

It's possible it's one of those 3 things...

Re: "make phpmod-install" - It's complaining the dl function doesn't
exist, not that it can't find the module... so it probably is
installed ok, just can't be used because of one of the above.

-Steve



On 7 March 2011 19:31, ovvenkat <ovvenkatesan at gmail.com> wrote:
>> What web server? Apache? Check the error log, perhaps there's a clue
>> there.
>>
>
> Thanks Steven for your replay,
>
> When, I checked in the log file, I found fatal error, It says that
>
> [Tue Mar 08 00:49:21 2011] [error] [client 127.0.0.1] PHP Fatal error:? Call
> to undefined function dl() in /var/www/html/ESL/ESL.php on line 23
>
> But, when I run the same file through shell command, its works fine.
>
>
> Regards,
> Venkat.
>
>
>
>
>> -Steve
>>
>>
>> On 7 March 2011 18:01, ovvenkat <ovvenkatesan at gmail.com> wrote:
>> > Hi,
>> >
>> > I have installed ESL phpmod module on fedora 13.
>> > even though it was painful to installing it on fedora, some how i did it
>> > .
>> >
>> > I can run and view the output of test.php on shell command.
>> > like,
>> > # php test.php
>> >
>> > When, I copied test.php and ESL.php files into my web server? root
>> > directory,
>> > trying to to run it from web browser, it shows only blank page.
>> > I am trying it figure it out, no luck.
>> >
>> > Anyone please shed some light on this.
>> >
>> >
>> > --
>> >
>> > If you have come to help me, you are wasting your time.
>> > If you have come to because your liberation is bound up in mine, we can
>> > work
>> > together.
>> >
>> >
>> > Regards
>> > Venkatesan OV.
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
>
> If you have come to help me, you are wasting your time.
> If you have come to because your liberation is bound up in mine, we can work
> together.
>
>
> Regards
> Venkatesan OV.
>


From elijah at crankenstein.com  Mon Mar  7 22:56:22 2011
From: elijah at crankenstein.com (elijah)
Date: Mon, 7 Mar 2011 11:56:22 -0800
Subject: [Freeswitch-users] database is locked
In-Reply-To: <AANLkTi=kJpAT3Fia8TqiXatE7AORW4Zt508RMjoTJ_a=@mail.gmail.com>
References: <AANLkTimHPa50yukZfq9P5emxMP=5uGizdPiTVeiV4N0C@mail.gmail.com>
	<AANLkTik3TC1ALNzYfM+xnP7CNR40EBUWT6N_6D8xsavE@mail.gmail.com>
	<AANLkTi=kJpAT3Fia8TqiXatE7AORW4Zt508RMjoTJ_a=@mail.gmail.com>
Message-ID: <AANLkTi=xM2B6ubsJuoDzRD+TxgZkp+gBhebyUUa3CmD-@mail.gmail.com>

here's the extra debug output:

2011-03-07 11:53:04.747232 [DEBUG] switch_core_sqldb.c:1685 Reuse Unused
Cached DB handle db="core";user="";pass="" [CORE_DB]
2011-03-07 11:53:19.744948 [DEBUG] switch_core_sqldb.c:1685 Reuse Unused
Cached DB handle db="core";user="";pass="" [CORE_DB]
2011-03-07 11:53:22.161581 [ERR] switch_core_sqldb.c:432 SQL ERR [database
is locked]
DELETE FROM members WHERE system = 'single_box' AND uuid = '' AND
(abandoned_epoch = '1299525498' OR joined_epoch = '1299525427')
2011-03-07 11:53:22.161581 [DEBUG] mod_callcenter.c:744 [
7 at live001.voice.telifi.com] rwlock
2011-03-07 11:53:22.261564 [DEBUG] mod_callcenter.c:511 Reuse Unused Cached
DB handle db="callcenter";user="";pass="" [CORE_DB]
2011-03-07 11:53:22.261564 [DEBUG] mod_callcenter.c:744 [
7 at live001.voice.telifi.com] rwlock
2011-03-07 11:53:22.261564 [DEBUG] mod_callcenter.c:511 Reuse Unused Cached
DB handle db="callcenter";user="";pass="" [CORE_DB]
2011-03-07 11:53:23.546367 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal-ipv6";user="";pass="" [CORE_DB]
2011-03-07 11:53:23.546367 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal-ipv6";user="";pass="" [CORE_DB]
2011-03-07 11:53:23.546367 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal-ipv6";user="";pass="" [CORE_DB]
2011-03-07 11:53:23.546367 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal-ipv6";user="";pass="" [CORE_DB]
2011-03-07 11:53:23.546367 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal-ipv6";user="";pass="" [CORE_DB]
2011-03-07 11:53:23.546367 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal-ipv6";user="";pass="" [CORE_DB]
2011-03-07 11:53:23.546367 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal-ipv6";user="";pass="" [CORE_DB]
2011-03-07 11:53:23.546367 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal-ipv6";user="";pass="" [CORE_DB]
2011-03-07 11:53:23.546367 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal-ipv6";user="";pass="" [CORE_DB]
2011-03-07 11:53:24.546215 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_external";user="";pass="" [CORE_DB]
2011-03-07 11:53:24.546215 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_external";user="";pass="" [CORE_DB]
2011-03-07 11:53:24.546215 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_external";user="";pass="" [CORE_DB]
2011-03-07 11:53:24.546215 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_external";user="";pass="" [CORE_DB]
2011-03-07 11:53:24.546215 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_external";user="";pass="" [CORE_DB]
2011-03-07 11:53:24.546215 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_external";user="";pass="" [CORE_DB]
2011-03-07 11:53:24.546215 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_external";user="";pass="" [CORE_DB]
2011-03-07 11:53:24.546215 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_external";user="";pass="" [CORE_DB]
2011-03-07 11:53:24.546215 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_external";user="";pass="" [CORE_DB]
2011-03-07 11:53:24.546215 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_external";user="";pass="" [CORE_DB]
2011-03-07 11:53:24.550215 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal";user="";pass="" [CORE_DB]
2011-03-07 11:53:24.550215 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal";user="";pass="" [CORE_DB]
2011-03-07 11:53:24.550215 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal";user="";pass="" [CORE_DB]
2011-03-07 11:53:24.550215 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal";user="";pass="" [CORE_DB]
2011-03-07 11:53:24.550215 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal";user="";pass="" [CORE_DB]
2011-03-07 11:53:24.550215 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal";user="";pass="" [CORE_DB]
2011-03-07 11:53:24.550215 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal";user="";pass="" [CORE_DB]
2011-03-07 11:53:24.550215 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal";user="";pass="" [CORE_DB]
2011-03-07 11:53:24.550215 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal";user="";pass="" [CORE_DB]
2011-03-07 11:53:34.742661 [DEBUG] switch_core_sqldb.c:1685 Reuse Unused
Cached DB handle db="core";user="";pass="" [CORE_DB]
2011-03-07 11:53:49.740377 [DEBUG] switch_core_sqldb.c:198 Dropping idle DB
connection db="voicemail_telifi";user="";pass=""
2011-03-07 11:53:49.740377 [DEBUG] switch_core_sqldb.c:1685 Reuse Unused
Cached DB handle db="core";user="";pass="" [CORE_DB]
2011-03-07 11:53:52.063018 [ERR] switch_core_sqldb.c:432 SQL ERR [database
is locked]
DELETE FROM members WHERE system = 'single_box' AND uuid = '' AND
(abandoned_epoch = '1299525498' OR joined_epoch = '1299525427')
2011-03-07 11:53:52.063018 [DEBUG] mod_callcenter.c:744 [
7 at live001.voice.telifi.com] rwlock
2011-03-07 11:53:52.063018 [DEBUG] mod_callcenter.c:511 Reuse Unused Cached
DB handle db="callcenter";user="";pass="" [CORE_DB]
2011-03-07 11:53:53.601788 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal-ipv6";user="";pass="" [CORE_DB]
2011-03-07 11:53:53.601788 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal-ipv6";user="";pass="" [CORE_DB]
2011-03-07 11:53:53.601788 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal-ipv6";user="";pass="" [CORE_DB]
2011-03-07 11:53:53.601788 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal-ipv6";user="";pass="" [CORE_DB]
2011-03-07 11:53:53.601788 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal-ipv6";user="";pass="" [CORE_DB]
2011-03-07 11:53:53.601788 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal-ipv6";user="";pass="" [CORE_DB]
2011-03-07 11:53:53.601788 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal-ipv6";user="";pass="" [CORE_DB]
2011-03-07 11:53:53.601788 [DEBUG] sofia_glue.c:5894 Reuse Unused Cached DB
handle db="sofia_reg_internal-ipv6";user="";pass="" [CORE_DB]



On Mon, Mar 7, 2011 at 11:42 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> ideally run
>
> fsctl debug_level 10
> console loglevel debug
>
> then capture it and send it to me.
>
>
> On Mon, Mar 7, 2011 at 1:36 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > why did you feel the need to xxx out the line numbers?
> >
> > This is what exact revision, was it literally from today?
> > Do you have the exact excerpt from the logs not a paraphrased on?
> >
> >
> > On Mon, Mar 7, 2011 at 1:28 PM, elijah <elijah at crankenstein.com> wrote:
> >> I updated (via Git) today an installation of FreeSWITCH I've been
> running
> >> without incident for a few months. Upon restarting the FreeSWITCH
> process I
> >> receive at a constant rate these errors:
> >> ...switch_core_sqldb.c xxx SQL ERR [database is locked]
> >> ...switch_core_sqldb.c xxx SQL ERR [database is locked]
> >> ...switch_core_sqldb.c xxx SQL ERR [database is locked]
> >> How should I resolve?
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ovvenkatesan at gmail.com  Mon Mar  7 23:04:41 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Tue, 8 Mar 2011 01:34:41 +0530
Subject: [Freeswitch-users] not able to view ESL output on the web
	browser
In-Reply-To: <AANLkTino-+XhpTOEADJTiG8jFPgca8Z_iWSxnTLSYu0j@mail.gmail.com>
References: <AANLkTind+xyiEHvO_tfrhymG_K5RHYMZpU0Z9KFPTNvJ@mail.gmail.com>
	<AANLkTikYvO0ioh8tXEfYtS3TJYW=_4cvbhpyvgZ1bAv6@mail.gmail.com>
	<AANLkTin7mU7oM0kAY5Wu+Vn-PxZ6ZLnZnLf38EavsFJs@mail.gmail.com>
	<AANLkTino-+XhpTOEADJTiG8jFPgca8Z_iWSxnTLSYu0j@mail.gmail.com>
Message-ID: <AANLkTi=GBPLW0BW08rfzOLzynq5kTA3y0eKUsDWGxGZ2@mail.gmail.com>

On Tue, Mar 8, 2011 at 1:13 AM, Michael Collins <msc at freeswitch.org> wrote:

> did you do the phpmod-install command in the libs/esl/ directory?
> -MC
>
>
oops. Thanks Michel, Now Its fine.
Thank you very much . you guys rocks !! :)



>
> On Mon, Mar 7, 2011 at 11:31 AM, ovvenkat <ovvenkatesan at gmail.com> wrote:
>
>> What web server? Apache? Check the error log, perhaps there's a clue
>>> there.
>>>
>>>
>> Thanks Steven for your replay,
>>
>> When, I checked in the log file, I found fatal error, It says that
>>
>> [Tue Mar 08 00:49:21 2011] [error] [client 127.0.0.1] PHP Fatal error:
>> Call to undefined function dl() in /var/www/html/ESL/ESL.php on line 23
>>
>> But, when I run the same file through shell command, its works fine.
>>
>>
>> Regards,
>> Venkat.
>>
>>
>>
>>
>> -Steve
>>>
>>>
>>> On 7 March 2011 18:01, ovvenkat <ovvenkatesan at gmail.com> wrote:
>>> > Hi,
>>> >
>>> > I have installed ESL phpmod module on fedora 13.
>>> > even though it was painful to installing it on fedora, some how i did
>>> it .
>>> >
>>> > I can run and view the output of test.php on shell command.
>>> > like,
>>> > # php test.php
>>> >
>>> > When, I copied test.php and ESL.php files into my web server  root
>>> > directory,
>>> > trying to to run it from web browser, it shows only blank page.
>>> > I am trying it figure it out, no luck.
>>> >
>>> > Anyone please shed some light on this.
>>> >
>>> >
>>> > --
>>> >
>>> > If you have come to help me, you are wasting your time.
>>> > If you have come to because your liberation is bound up in mine, we can
>>> work
>>> > together.
>>> >
>>> >
>>> > Regards
>>> > Venkatesan OV.
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> If you have come to help me, you are wasting your time.
>> If you have come to because your liberation is bound up in mine, we can
>> work together.
>>
>>
>> Regards
>> Venkatesan OV.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
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From grsingh750 at gmail.com  Mon Mar  7 23:22:45 2011
From: grsingh750 at gmail.com (guru singh)
Date: Tue, 8 Mar 2011 01:52:45 +0530
Subject: [Freeswitch-users] tones.conf parameters?
In-Reply-To: <4D751D6A.70605@julf.com>
References: <4D7503F5.6010606@julf.com>
	<AANLkTin0N+79ZOQc8pMHw4U+CRS0Rc+LQ27YiPEuo0Bp@mail.gmail.com>
	<4D751D6A.70605@julf.com>
Message-ID: <AANLkTinAzspPWKX5k2ZDizoXAb6L8+XxDkTHjpRf8f41@mail.gmail.com>

On Mon, Mar 7, 2011 at 11:31 PM, Johan Helsingius <julf at julf.com> wrote:
> Hi, guru,
>
>> Another way would be to use the tone_detect application.
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
>
> The problem is that I don't seem to get my DECT phone
> to generate anything before it gets a dial tone.

This is mostly useful to detect busy tone for incoming calls I guess.
I was lucky to have tones.conf for India on the wiki :)
I can't help you beyond that, maybe somebody else would chime in.

>> I found this handy as well
>> http://www.3amsystems.com/wireline/tone-search.htm
>
> That is helpful - but is the tones.conf format documented anywhere?

>
> ? ? ? ?Julf
guru
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From sunwood360 at gmail.com  Tue Mar  8 00:06:22 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Mon, 7 Mar 2011 13:06:22 -0800
Subject: [Freeswitch-users] send fax via google voice.
In-Reply-To: <C99A742A.12703%krice@freeswitch.org>
References: <AANLkTik58RREBEXwzMsUL6djbFMQzyVt=ddJZc0rJO0B@mail.gmail.com>
	<C99A742A.12703%krice@freeswitch.org>
Message-ID: <AANLkTinS7tg2iar53-AciETJeuXbGcnpmM33V-mEiZ3c@mail.gmail.com>

Great!

please go ahead claim lawyers' ignorance and lobby german that email pdf is
a document as equivalent as fax. We are looking forward to you next great
accomplish.  ?
On Mar 7, 2011 9:41 AM, "Ken Rice" <krice at freeswitch.org> wrote:
> Whats the old saying? Ignorance is bliss? Heh
>
> Just because and end user perceives it as being more secure doesn?t make
it
> so... Even faxing over voip you just tcpdump out the rtp and process it
thru
> and you can recover then entire exchange... Same thing with a laptop, a
> sound card, and a wav recorder with a cheap passive phone tap... Only
> difference with legacy PSTN vs VoIP is its easier to hack a box somewhere
> along the way and intercept the traffic vs finding the trunk line carrying
> the traffic, however, it doesn?t take long to figure out that there are
tons
> of unsecure access points to the wire between your office and the CO...
> (even the fiber these days can be compromised relatively easily compared
to
> what it was 10 years ago)
>
> My whole point here is I am not malicious when it comes to this sort of
> thing but I know what can be done and how cheap the technology to pull off
> these things really is. I smart but I am far from the smartest guy and if
I
> can figure out how to do this stuff many people can figure it out...
>
> I?m still say that the only secure computer is one fired into the sun on
the
> tip of a rocket with no comms back to earth.. Then that?s only secure
until
> someone figures out how to retrieve it.
>
> K
>
>
>
> On 3/7/11 11:24 AM, "envelopes envelopes" <sunwood360 at gmail.com> wrote:
>
>> From the perspective of end user,  it is pretty easier to hacker an end
user'
>> inbox globally.  However, you need physical presence to grab fax
documents.
>>
>> Interesting to see the view from IT people is so different from
non-technical
>> people. After, any existence has its reason.
>>
>> On Mar 6, 2011 8:56 AM, "Ken Rice" <krice at freeswitch.org> wrote:
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
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From wstephen80 at gmail.com  Tue Mar  8 01:09:22 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Mon, 7 Mar 2011 23:09:22 +0100
Subject: [Freeswitch-users] Avoid Loops
In-Reply-To: <AANLkTimX4aXh1XxyNuvC9qgmiu9pCC_CYkcFWmV+0OiS@mail.gmail.com>
References: <AANLkTinV1QO1=44vyaoeALnzJQFowAcnGY_yqLyN3BJ1@mail.gmail.com>
	<C99A6E7A.126FE%krice@freeswitch.org>
	<AANLkTimX4aXh1XxyNuvC9qgmiu9pCC_CYkcFWmV+0OiS@mail.gmail.com>
Message-ID: <AANLkTi=7fXabo1VYmMtiMJtggH3VdRyHZJmBb6fw1B0i@mail.gmail.com>

A doubt: if the customer sends the first call that for some reason doesn't
go up (i.e. congestion) and it send a second try before 5 seconds, with this
line in dialplan, the second call is dropped? Also if this isn't a loop?

Stephen

On Mon, Mar 7, 2011 at 6:41 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> Ok, so a line like:
>
> <action application="limit" data="hash inbound ${destination_number} 1/5
> !NORMAL_CIRCUIT_CONGESTION" />
>
> avoid loops because Freeswitch doesn't accept more than 1 call each 5
> seconds (the loops are very fast) but allow to have concurrent calls to the
> same destination number.
> Thank you Ken!
>
>
> Stephen
>
>
> On Mon, Mar 7, 2011 at 6:14 PM, Ken Rice <krice at freeswitch.org> wrote:
>
>>  Well you could do that, but lets say you have a client that sends you
>> several calls to a conf number... Also when you loop calls they usually loop
>> fairly fast, by using a rate limit based on the ani and dnis you can still
>> let additional calls pass while reducing the chances of getting loop due to
>> call_id rewrites that are happening at the b2buas that are causing the loop
>>
>> K
>>
>>
>>
>> On 3/7/11 11:01 AM, "Stephen Wilde" <wstephen80 at gmail.com> wrote:
>>
>> Thank you for your suggestion! Good idea! I was searching for more
>> complicated solutions but mod_limit is perfect!
>>
>> You say to use rate limiting but why not simply limit the calls number
>> instead of rate?
>>
>> For example a row in dialplan as:
>>
>> <action application="limit" data="hash inbound ${destination_number} 1" />
>>
>> can avoid loops?
>>
>> Stephen
>>
>>
>> On Mon, Mar 7, 2011 at 5:37 PM, Ken Rice <krice at freeswitch.org> wrote:
>>
>> I hate this problem... Mod_limit matching ani+dnis doing rate limiting
>> (say 5 calls with matchig ani+dnis in 2 seconds) can kill the loop and
>> either allow you to re-route or allow then to re-route... This happens a lot
>> more than you thing between 2nd and 3rd tier aggregators
>>
>>
>>
>>
>> On 3/7/11 10:24 AM, "Stephen Wilde" <wstephen80 at gmail.com <
>> http://wstephen80 at gmail.com> > wrote:
>>
>> In some situation there are calls that loops among some providers and I
>> want avoid that situation.
>> For example, I receive the call from my client A, I send this call to my
>> provider B but for some reason the call return to me (for example because B
>> sends the call to A and A resends the call to me).
>> In case of loop for me the solution can be simply to drop the call so my
>> client can reroute to a different provider.
>> There is anyone that can suggest me a way to drop looped calls?
>>
>>
>> Stephen
>>
>>
>> ------------------------------
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <
>> http://FreeSWITCH-users at lists.freeswitch.org>
>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> ------------------------------
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From simpot at simpot.com  Tue Mar  8 02:17:53 2011
From: simpot at simpot.com (Dmitry Saratsky)
Date: Tue, 8 Mar 2011 01:17:53 +0200
Subject: [Freeswitch-users] What is so dangeres in "freeswitch.msleep()"
 in LUA?
In-Reply-To: <AANLkTimhk603FH-ABd=ju4EgBNsgO3rUw1DqCePxK+2o@mail.gmail.com>
References: <52114D605A462A4E9A50E588EE7D0686012E575D9EEB@mail.forest.simpot.com>
	<AANLkTimhk603FH-ABd=ju4EgBNsgO3rUw1DqCePxK+2o@mail.gmail.com>
Message-ID: <52114D605A462A4E9A50E588EE7D0686012E575D9EF4@mail.forest.simpot.com>

Thanks Steve,

Session:sleep works like a harm!


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: 07 Mar 2011 11:17
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] What is so dangeres in "freeswitch.msleep()" in LUA?

AFAIK it stops all processing of the call during the sleep, you should
use session:sleep()

-Steve


On 6 March 2011 22:48, Dmitry Saratsky <simpot at simpot.com> wrote:
> Hi All,
>
>
>
> According to: http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.msleep
>
> It is not good to use: "freeswitch.msleep()" in session-based script:
>
> NOTE: Do not use this on a session-based script or bad things will happen.
>
>
>
> Can anyone explain please why not and what does author mean with
> "session-based script".
>
>
>
> I want to use it inside some function between I originate session and hangup
> it in the following way:
>
>
>
> ?? repeat
>
> --????? os.execute("sleep 1");
>
> ??????? freeswitch.msleep(1000);
>
> ??????? until ((not new_session:ready() == true) or
> (new_session:getVariable("fax_result_code") >= "0"))
>
> ?? new_session:hangup();
>
>
>
> Is it danger for me to use this?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

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


From simpot at simpot.com  Tue Mar  8 02:18:50 2011
From: simpot at simpot.com (Dmitry Saratsky)
Date: Tue, 8 Mar 2011 01:18:50 +0200
Subject: [Freeswitch-users] What is so dangeres in "freeswitch.msleep()"
 in LUA?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B2E3EFAF@cooper>
References: <52114D605A462A4E9A50E588EE7D0686012E575D9EEB@mail.forest.simpot.com>
	<AANLkTimhk603FH-ABd=ju4EgBNsgO3rUw1DqCePxK+2o@mail.gmail.com>
	<52114D605A462A4E9A50E588EE7D0686012E575D9EEE@mail.forest.simpot.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2E3EFAF@cooper>
Message-ID: <52114D605A462A4E9A50E588EE7D0686012E575D9EF5@mail.forest.simpot.com>

Hi Peter,

You are right, session:sleep works well.
Thanks a lot!

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson
Sent: 07 Mar 2011 14:49
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] What is so dangeres in "freeswitch.msleep()" in LUA?

My understanding is that Session:sleep() continues to handle the loop thread within the sleep. msleep() will stop the thread totally during the sleep, so I think you should use session:sleep().

/Peter

-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Dmitry Saratsky
Skickat: den 7 mars 2011 11:38
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] What is so dangeres in "freeswitch.msleep()" in LUA?

Hi Steve,

Thank you for your reply.

The problem is that I need script to sleep, while session ready and transmits fax in same time until session will not be ready or fax transmitted. So I mean that I want script to sleep, not the session, just because my session may work in that time (transmit a fax). I think if I use session:sleep() in my case - my fax transmission will fail... Do you think I'm right? Do you think I can use freeswitch.msleep() in my case? And do you know which (if it will at all) troubles should I get?

Thanks,
Dmitry.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: 07 Mar 2011 11:17
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] What is so dangeres in "freeswitch.msleep()" in LUA?

AFAIK it stops all processing of the call during the sleep, you should
use session:sleep()

-Steve


On 6 March 2011 22:48, Dmitry Saratsky <simpot at simpot.com> wrote:
> Hi All,
>
>
>
> According to: http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.msleep
>
> It is not good to use: "freeswitch.msleep()" in session-based script:
>
> NOTE: Do not use this on a session-based script or bad things will happen.
>
>
>
> Can anyone explain please why not and what does author mean with
> "session-based script".
>
>
>
> I want to use it inside some function between I originate session and hangup
> it in the following way:
>
>
>
> ?? repeat
>
> --????? os.execute("sleep 1");
>
> ??????? freeswitch.msleep(1000);
>
> ??????? until ((not new_session:ready() == true) or
> (new_session:getVariable("fax_result_code") >= "0"))
>
> ?? new_session:hangup();
>
>
>
> Is it danger for me to use this?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

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From simpot at simpot.com  Tue Mar  8 02:21:48 2011
From: simpot at simpot.com (Dmitry Saratsky)
Date: Tue, 8 Mar 2011 01:21:48 +0200
Subject: [Freeswitch-users] How do I originiate new session from within
 LUA	script?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EF@cooper>
References: <52114D605A462A4E9A50E588EE7D0686012E575D9EE4@mail.forest.simpot.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EB@cooper>,
	<52114D605A462A4E9A50E588EE7D0686012E575D9EE5@mail.forest.simpot.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EE@cooper>,
	<52114D605A462A4E9A50E588EE7D0686012E575D9EE6@mail.forest.simpot.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2C494EF@cooper>
Message-ID: <52114D605A462A4E9A50E588EE7D0686012E575D9EF6@mail.forest.simpot.com>

Hi Peter,

Thanks again, it is exactly the syntax!


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson
Sent: 06 Mar 2011 10:40
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How do I originiate new session from within LUA script?

It doesn't matter how you set the variables, getVariable() should always return then anyway, the "{var=val}" trick is just to make it possible to set vars when originating a new channel, since the actual channel will be created at the same time.

However, if you want to check variables on the originated leg, it would be something like this;

local s = freeswitch.Session();
s:originate(session, "{var1=val1,var2=val2}sofia/internal/1000 at x.x.x.x", 30);

-- Do whatever you want with this channel.

-- Get variables from the originated channel
local var1 = s:getVariable("whatever_variable");

That is, you must read the var from the correct object. In this case "s" is the new originated channel, and "session" is the channel that was created when dialing into the lua script.

PS! I've not written so much in Lua myself, so it's not really something I'm good at - but I guess it should be like this :)

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Dmitry Saratsky [simpot at simpot.com]
Skickat: den 6 mars 2011 02:03
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] How do I originiate new session from within LUA    script?

Hi Peter,

Thanks again for your help, this way works well for me.
I will update wiki, when I fix all puzzles of my script.

Now I have another problem... Sorry...

I actually hoped also to read session variables after my new originated sessions complete in same way, I set them.
However in way you have suggested me to set vars, I have no idea how to read those variables now...

I have tried to use the following after origination command:
   local fax_result_code = session:getVariable("fax_result_code"); local fax_result_text = session:getVariable("fax_result_text");
   freeswitch.consoleLog("info", "FAX - Error code: (" .. fax_result_code .. ") " .. fax_result_text .. "\n");

I got the following in console:
2011-03-06 02:54:23.311416 [ERR] mod_lua.cpp:182 /usr/local/freeswitch/scripts/faxoutservice.lua:53: attempt to concatenate global 'fax_success' (a nil value)
stack traceback:
        /usr/local/freeswitch/scripts/faxoutservice.lua:53: in function 'send_fax'
        /usr/local/freeswitch/scripts/faxoutservice.lua:74: in main chunk

According to docs (http://wiki.freeswitch.org/wiki/Mod_spandsp), those vars should be set on hangup both for tx and rx faxes...
Anyway, I have lua script I have wrote for receiving faxes - I can read those variables successfully in the way I wrote above.

So, any ideas how can I read session (that I start in that way - origination) variables after session hangup?

Thanks again,
Dmitry.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson
Sent: 06 Mar 2011 00:29
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How do I originiate new session from within LUA script?

10 is the origination timeout in seconds. This part is missing in the documentation right now, so I can't really tell you where to look - except in the source :)

To set variables you should be able to do it as you do in the dialplan, changing the dial string to something like "{var1=value,var2=value}sofia/internal/test at test.com". By the way - for originate variables origination_caller_id_xxxx must be used.

If you're successful with these commands, I would really appreciate if this could be updated in the Wiki as well.

/Peter

________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Dmitry Saratsky [simpot at simpot.com]
Skickat: den 5 mars 2011 22:51
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] How do I originiate new session from within LUA    script?

Hi Peter,

Thanks for your replay.
It works in this way, thanks.

Can you please also explain (or point me to right documentation) what does mean "10" as 3rd variable in your example?
Also I'm looking the way to set session variables before origination and I also failed with this...

For example I have tried this (the call itself was succeeded, but caller_id was not passed):
   s = freeswitch.Session();
   s:setVariable("effective_caller_id_number", "999");
   s:setVariable("effective_caller_id_name", "999");
   s:originate(session, "sofia/gateway/mygw/1000", 10);

and I got the following in console:

2011-03-05 23:43:05.754621 [ERR] switch_cpp.cpp:634 session is not initalized
2011-03-05 23:43:05.754621 [ERR] switch_cpp.cpp:634 session is not initialized

Thanks,
Dmitry.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson
Sent: 05 Mar 2011 22:48
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How do I originiate new session from within LUA script?

This should work, last parameter is the originate timeout in seconds.

s = freeswitch.Session();
s:originate(session, "sofia/internal/1000 at 192.168.1.1", 10);

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Dmitry Saratsky [simpot at simpot.com]
Skickat: den 5 mars 2011 20:53
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] How do I originiate new session from within LUA        script?

Hello all,

According to wiki:
http://wiki.freeswitch.org/wiki/Mod_lua#session:originate

I can use: "session:originate" command in LUA script to originate new session, but I can't understand how should I do it and can't find any example how I use this command.

Can you help, guys?

Thanks,
Dmitry.


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From jrichey at itltd.net  Tue Mar  8 00:47:38 2011
From: jrichey at itltd.net (JRichey)
Date: Mon, 7 Mar 2011 13:47:38 -0800 
Subject: [Freeswitch-users] send fax via google voice.
Message-ID: <6ECAF1527329364583AB525CF34ABF950B31A52A@ms.kallback.com>

I'm with Ken, faxing needs to die.

Of faxes being sent now, I wonder how many originate and/or terminate via a
fax to email gateway. These and the possibility that your call will be sent
via VoIP somewhere along the way should strip away the false idea that
faxing is secure because it travels over the PSTN. The OP is testing to K7,
which is in part a fax to email service.

Incidentally, the system that terminates K7 faxes is a FreeSwitch box. :-)

 

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of envelopes
envelopes
Sent: Monday, March 07, 2011 1:06 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] send fax via google voice.



Great!  

please go ahead claim lawyers' ignorance and lobby german that email pdf is
a document as equivalent as fax. We are looking forward to you next great
accomplish.  ?

On Mar 7, 2011 9:41 AM, "Ken Rice" < krice at freeswitch.org
<mailto:krice at freeswitch.org> > wrote:
> Whats the old saying? Ignorance is bliss? Heh
> 
> Just because and end user perceives it as being more secure doesn?t make
it
> so... Even faxing over voip you just tcpdump out the rtp and process it
thru
> and you can recover then entire exchange... Same thing with a laptop, a
> sound card, and a wav recorder with a cheap passive phone tap... Only
> difference with legacy PSTN vs VoIP is its easier to hack a box somewhere
> along the way and intercept the traffic vs finding the trunk line carrying
> the traffic, however, it doesn?t take long to figure out that there are
tons
> of unsecure access points to the wire between your office and the CO...
> (even the fiber these days can be compromised relatively easily compared
to
> what it was 10 years ago)
> 
> My whole point here is I am not malicious when it comes to this sort of
> thing but I know what can be done and how cheap the technology to pull off
> these things really is. I smart but I am far from the smartest guy and if
I
> can figure out how to do this stuff many people can figure it out...
> 
> I?m still say that the only secure computer is one fired into the sun on
the
> tip of a rocket with no comms back to earth.. Then that?s only secure
until
> someone figures out how to retrieve it.
> 
> K
> 
> 
> 
> On 3/7/11 11:24 AM, "envelopes envelopes" < sunwood360 at gmail.com
<mailto:sunwood360 at gmail.com> > wrote:
> 
>> From the perspective of end user,  it is pretty easier to hacker an end
user'
>> inbox globally.  However, you need physical presence to grab fax
documents.
>> 
>> Interesting to see the view from IT people is so different from
non-technical
>> people. After, any existence has its reason.
>> 
>> On Mar 6, 2011 8:56 AM, "Ken Rice" < krice at freeswitch.org
<mailto:krice at freeswitch.org> > wrote:
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
<mailto:FreeSWITCH-users at lists.freeswitch.org> 
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> 
>> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users
<http://lists.freeswitch.org/mailman/options/freeswitch-users> 
>> http://www.freeswitch.org <http://www.freeswitch.org> 
> 


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From krice at freeswitch.org  Tue Mar  8 02:29:38 2011
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 07 Mar 2011 17:29:38 -0600
Subject: [Freeswitch-users] send fax via google voice.
In-Reply-To: <AANLkTinS7tg2iar53-AciETJeuXbGcnpmM33V-mEiZ3c@mail.gmail.com>
Message-ID: <C99AC682.12738%krice@freeswitch.org>

I dunno about there but a digitally signed document has had the same weight
as a faxed document in the USA for 10 years

K


On 3/7/11 3:06 PM, "envelopes envelopes" <sunwood360 at gmail.com> wrote:

> Great!? 
> 
> please go ahead claim lawyers' ignorance and lobby german that email pdf is a
> document as equivalent as fax. We are looking forward to you next great
> accomplish.? ?
> 
> On Mar 7, 2011 9:41 AM, "Ken Rice" <krice at freeswitch.org> wrote:
>> > Whats the old saying? Ignorance is bliss? Heh
>> > 
>> > Just because and end user perceives it as being more secure doesn?t make it
>> > so... Even faxing over voip you just tcpdump out the rtp and process it
>> thru
>> > and you can recover then entire exchange... Same thing with a laptop, a
>> > sound card, and a wav recorder with a cheap passive phone tap... Only
>> > difference with legacy PSTN vs VoIP is its easier to hack a box somewhere
>> > along the way and intercept the traffic vs finding the trunk line carrying
>> > the traffic, however, it doesn?t take long to figure out that there are
>> tons
>> > of unsecure access points to the wire between your office and the CO...
>> > (even the fiber these days can be compromised relatively easily compared to
>> > what it was 10 years ago)
>> > 
>> > My whole point here is I am not malicious when it comes to this sort of
>> > thing but I know what can be done and how cheap the technology to pull off
>> > these things really is. I smart but I am far from the smartest guy and if I
>> > can figure out how to do this stuff many people can figure it out...
>> > 
>> > I?m still say that the only secure computer is one fired into the sun on >>
the
>> > tip of a rocket with no comms back to earth.. Then that?s only secure until
>> > someone figures out how to retrieve it.
>> > 
>> > K
>> > 
>> > 
>> > 
>> > On 3/7/11 11:24 AM, "envelopes envelopes" <sunwood360 at gmail.com> wrote:
>> > 
>>> >> From the perspective of end user,? it is pretty easier to hacker an end
>>> user'
>>> >> inbox globally.? However, you need physical presence to grab fax
>>> documents.
>>> >> 
>>> >> Interesting to see the view from IT people is so different from
>>> non-technical
>>> >> people. After, any existence has its reason.
>>> >> 
>>> >> On Mar 6, 2011 8:56 AM, "Ken Rice" <krice at freeswitch.org> wrote:
>>> >> 
>>> >> 
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>> > 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From infos at madovsky.org  Tue Mar  8 02:38:04 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 7 Mar 2011 18:38:04 -0500
Subject: [Freeswitch-users] send fax via google voice.
References: <6ECAF1527329364583AB525CF34ABF950B31A52A@ms.kallback.com>
Message-ID: <DEF66296BCC742AEBF693A649C2C360E@e1705>

old tools is still existing in poor countries and very useful...
  ----- Original Message ----- 
  From: JRichey 
  To: 'FreeSWITCH Users Help' 
  Sent: Monday, March 07, 2011 4:47 PM
  Subject: Re: [Freeswitch-users] send fax via google voice.


  I'm with Ken, faxing needs to die.

  Of faxes being sent now, I wonder how many originate and/or terminate via a fax to email gateway. These and the possibility that your call will be sent via VoIP somewhere along the way should strip away the false idea that faxing is secure because it travels over the PSTN. The OP is testing to K7, which is in part a fax to email service.

  Incidentally, the system that terminates K7 faxes is a FreeSwitch box. :-)



    -----Original Message-----
    From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of envelopes envelopes
    Sent: Monday, March 07, 2011 1:06 PM
    To: FreeSWITCH Users Help
    Subject: Re: [Freeswitch-users] send fax via google voice.


    Great!  

    please go ahead claim lawyers' ignorance and lobby german that email pdf is a document as equivalent as fax. We are looking forward to you next great accomplish.  ?

    On Mar 7, 2011 9:41 AM, "Ken Rice" <krice at freeswitch.org> wrote:
    > Whats the old saying? Ignorance is bliss? Heh
    > 
    > Just because and end user perceives it as being more secure doesn?t make it
    > so... Even faxing over voip you just tcpdump out the rtp and process it thru
    > and you can recover then entire exchange... Same thing with a laptop, a
    > sound card, and a wav recorder with a cheap passive phone tap... Only
    > difference with legacy PSTN vs VoIP is its easier to hack a box somewhere
    > along the way and intercept the traffic vs finding the trunk line carrying
    > the traffic, however, it doesn?t take long to figure out that there are tons
    > of unsecure access points to the wire between your office and the CO...
    > (even the fiber these days can be compromised relatively easily compared to
    > what it was 10 years ago)
    > 
    > My whole point here is I am not malicious when it comes to this sort of
    > thing but I know what can be done and how cheap the technology to pull off
    > these things really is. I smart but I am far from the smartest guy and if I
    > can figure out how to do this stuff many people can figure it out...
    > 
    > I?m still say that the only secure computer is one fired into the sun on the
    > tip of a rocket with no comms back to earth.. Then that?s only secure until
    > someone figures out how to retrieve it.
    > 
    > K
    > 
    > 
    > 
    > On 3/7/11 11:24 AM, "envelopes envelopes" <sunwood360 at gmail.com> wrote:
    > 
    >> From the perspective of end user,  it is pretty easier to hacker an end user'
    >> inbox globally.  However, you need physical presence to grab fax documents.
    >> 
    >> Interesting to see the view from IT people is so different from non-technical
    >> people. After, any existence has its reason.
    >> 
    >> On Mar 6, 2011 8:56 AM, "Ken Rice" <krice at freeswitch.org> wrote:
    >> 
    >> 
    >> _______________________________________________
    >> FreeSWITCH-users mailing list
    >> FreeSWITCH-users at lists.freeswitch.org
    >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    >> http://www.freeswitch.org
    > 



------------------------------------------------------------------------------


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From simpot at simpot.com  Tue Mar  8 02:40:45 2011
From: simpot at simpot.com (Dmitry Saratsky)
Date: Tue, 8 Mar 2011 01:40:45 +0200
Subject: [Freeswitch-users] txfax bug?
Message-ID: <52114D605A462A4E9A50E588EE7D0686012E575D9EF7@mail.forest.simpot.com>

Hi All,

I have some LUA script that designed to send faxes.

If I originate session from within lua in the following way, all ok (fax can be sent successfully), but I see annoying error log in console... why?
Relevant part of LUA script:

local new_session = freeswitch.Session();
new_session:originate(session, "{autohangup=false,sip_cid_type=rpid,absolute_codec_string='PCMU,PCMA',origination_caller_id_name=" .. fax_caller_id .. ",origination_caller_id_number=" .. fax_caller_id .. "}sofia/gateway/sip1.myprovider.com/" .. fax_dst_num .. "|sofia/gateway/ sip2.myprovider.com /" .. fax_dst_num, 20);
new_session:execute("txfax", outgoing_fax_path .. fax_tiff_file .. ".tiff");
repeat
   new_session:sleep(1000);
   until ((new_session:ready() == false) or (new_session:getVariable("fax_result_code") >= "0"))
new_session:hangup();

Error log:
2011-03-08 01:24:02.289821 [ERR] switch_ivr.c:481 Invalid Command!

According to my investigation, I can say for sure, that reason for error log in console is following line in my LUA script:
new_session:execute("txfax", outgoing_fax_path .. fax_tiff_file .. ".tiff");


Yes, I know that all docs says to use txfax function in lay like: "originate sofia/external/100 at 10.10.10.10 &txfax(/path_to_fax_file)", but from other examples for "rxfax" I decided to do it in my way. It is even works perfect... So, why I still see error for this?

Thanks,
Dmitry.

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From msc at freeswitch.org  Tue Mar  8 02:58:58 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 7 Mar 2011 15:58:58 -0800
Subject: [Freeswitch-users] tones.conf parameters?
In-Reply-To: <4D751D6A.70605@julf.com>
References: <4D7503F5.6010606@julf.com>
	<AANLkTin0N+79ZOQc8pMHw4U+CRS0Rc+LQ27YiPEuo0Bp@mail.gmail.com>
	<4D751D6A.70605@julf.com>
Message-ID: <AANLkTikV1oG9-45b5mhsk5Xk3vXq+WGuFeOoZWPg8166@mail.gmail.com>

On Mon, Mar 7, 2011 at 10:01 AM, Johan Helsingius <julf at julf.com> wrote:

> Hi, guru,
>
> > Another way would be to use the tone_detect application.
> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
>
> The problem is that I don't seem to get my DECT phone
> to generate anything before it gets a dial tone.
>
> > I found this handy as well
> > http://www.3amsystems.com/wireline/tone-search.htm
>
> That is helpful - but is the tones.conf format documented anywhere?
>

The format is not "documented" per se, but the format can be inferred by
looking at the existing entries:

generate-dial => v=-7;%(1000,0,350,440)
detect-dial => 350,440

generate-ring => v=-7;%(2000,4000,440,480)
detect-ring => 440,480

generate-busy => v=-7;%(500,500,480,620)
detect-busy => 480,620

"generate-dial" means "generate dial tone" and uses the TGML. (Look up TGML
on the wiki)
"detect-dial" means the frequenc(y|ies) to detect a dial tone.

If I read the interwebs correct then NL uses 150Hz + 450Hz for dialtone,
425Hz (.5 sec on, .5 sec off) for busy, and 425Hz (1 sec on, 4 secs off) for
ringing.

Try these values:

[nl]
generate-dial => v=-7;%(1000,0,150,450)
detect-dial => 150,450

generate-ring => v=-7;%(1000,4000,425)
detect-ring => 425

generate-busy => v=-7;%(500,500,425)
detect-busy => 425

If they work then let us know and we'll add them to the tones.conf file.
FYI, I'll let you research the "attn" "call-waiting" and "detect-failX"
tones for your country. :)

-MC
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From msc at freeswitch.org  Tue Mar  8 03:02:11 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 7 Mar 2011 16:02:11 -0800
Subject: [Freeswitch-users] send fax via google voice.
In-Reply-To: <DEF66296BCC742AEBF693A649C2C360E@e1705>
References: <6ECAF1527329364583AB525CF34ABF950B31A52A@ms.kallback.com>
	<DEF66296BCC742AEBF693A649C2C360E@e1705>
Message-ID: <AANLkTi=jiCfJggf4YgdvNag_xWeLNuBjhhPXXcJLghyh@mail.gmail.com>

At the rate we're going this crazy thread will be around longer than
faxes...

On Mon, Mar 7, 2011 at 3:38 PM, Madovsky <infos at madovsky.org> wrote:

>  old tools is still existing in poor countries and very useful...
>
> ----- Original Message -----
> *From:* JRichey <jrichey at itltd.net>
> *To:* 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, March 07, 2011 4:47 PM
> *Subject:* Re: [Freeswitch-users] send fax via google voice.
>
>  I'm with Ken, faxing needs to die.
>
> Of faxes being sent now, I wonder how many originate and/or terminate via a
> fax to email gateway. These and the possibility that your call will be sent
> via VoIP somewhere along the way should strip away the false idea that
> faxing is secure because it travels over the PSTN. The OP is testing to K7,
> which is in part a fax to email service.
>
> Incidentally, the system that terminates K7 faxes is a FreeSwitch box. :-)
>
>
>
> -----Original Message-----
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org]*On Behalf Of *envelopes
> envelopes
> *Sent:* Monday, March 07, 2011 1:06 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] send fax via google voice.
>
> Great!
>
> please go ahead claim lawyers' ignorance and lobby german that email pdf is
> a document as equivalent as fax. We are looking forward to you next great
> accomplish.  ?
> On Mar 7, 2011 9:41 AM, "Ken Rice" <krice at freeswitch.org> wrote:
> > Whats the old saying? Ignorance is bliss? Heh
> >
> > Just because and end user perceives it as being more secure doesn?t make
> it
> > so... Even faxing over voip you just tcpdump out the rtp and process it
> thru
> > and you can recover then entire exchange... Same thing with a laptop, a
> > sound card, and a wav recorder with a cheap passive phone tap... Only
> > difference with legacy PSTN vs VoIP is its easier to hack a box somewhere
> > along the way and intercept the traffic vs finding the trunk line
> carrying
> > the traffic, however, it doesn?t take long to figure out that there are
> tons
> > of unsecure access points to the wire between your office and the CO...
> > (even the fiber these days can be compromised relatively easily compared
> to
> > what it was 10 years ago)
> >
> > My whole point here is I am not malicious when it comes to this sort of
> > thing but I know what can be done and how cheap the technology to pull
> off
> > these things really is. I smart but I am far from the smartest guy and if
> I
> > can figure out how to do this stuff many people can figure it out...
> >
> > I?m still say that the only secure computer is one fired into the sun on
> the
> > tip of a rocket with no comms back to earth.. Then that?s only secure
> until
> > someone figures out how to retrieve it.
> >
> > K
> >
> >
> >
> > On 3/7/11 11:24 AM, "envelopes envelopes" <sunwood360 at gmail.com> wrote:
> >
> >> From the perspective of end user,  it is pretty easier to hacker an end
> user'
> >> inbox globally.  However, you need physical presence to grab fax
> documents.
> >>
> >> Interesting to see the view from IT people is so different from
> non-technical
> >> people. After, any existence has its reason.
> >>
> >> On Mar 6, 2011 8:56 AM, "Ken Rice" <krice at freeswitch.org> wrote:
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Mar  8 03:06:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 7 Mar 2011 16:06:02 -0800
Subject: [Freeswitch-users] mod_python
In-Reply-To: <1299502358077-6097179.post@n2.nabble.com>
References: <1299244469418-6088430.post@n2.nabble.com>
	<AANLkTikQ4NVFE2BLNYxiY05zFaYeq8TJjqbuZJKp3OgG@mail.gmail.com>
	<1299347178939-6092296.post@n2.nabble.com>
	<1299502358077-6097179.post@n2.nabble.com>
Message-ID: <AANLkTim39o1o5Hw2REhmG4+8zqZGHhcmDXRcrd4NKk6F@mail.gmail.com>

On Mon, Mar 7, 2011 at 4:52 AM, hkalyoncu <hkalyoncu at gmail.com> wrote:

> ok i made some progress.
>
> i can get the serialized string of all variables with
>
> Event.serialize(params)  method
>
> But when i try to access directly each channel variables like
>
> params["Caller-Caller-ID-Name"]
>
> it returns error.
>
> is there any way to access each channel variable?
>
> Try the variable names listed here:
http://wiki.freeswitch.org/wiki/Channel_Variables#Info_Application_Variable_Names_.28variable_xxxx.29

so for that var try
params["caller_id_name']

Let us know.
-MC
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From curriegrad2004 at gmail.com  Tue Mar  8 03:21:07 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 7 Mar 2011 16:21:07 -0800
Subject: [Freeswitch-users] send fax via google voice.
In-Reply-To: <AANLkTi=jiCfJggf4YgdvNag_xWeLNuBjhhPXXcJLghyh@mail.gmail.com>
References: <6ECAF1527329364583AB525CF34ABF950B31A52A@ms.kallback.com>
	<DEF66296BCC742AEBF693A649C2C360E@e1705>
	<AANLkTi=jiCfJggf4YgdvNag_xWeLNuBjhhPXXcJLghyh@mail.gmail.com>
Message-ID: <AANLkTinWjMfHV=SemVb-sbA2gshk995d+32rBWK5UQNZ@mail.gmail.com>

Okay, to sum things up here, Yes it will work. But are you going to
fax over google voice? It may have worked for a few, but definitely
not for me.

On Mon, Mar 7, 2011 at 4:02 PM, Michael Collins <msc at freeswitch.org> wrote:
> At the rate we're going this crazy thread will be around longer than
> faxes...
>
> On Mon, Mar 7, 2011 at 3:38 PM, Madovsky <infos at madovsky.org> wrote:
>>
>> old tools is still existing in poor countries and very useful...
>>
>> ----- Original Message -----
>> From: JRichey
>> To: 'FreeSWITCH Users Help'
>> Sent: Monday, March 07, 2011 4:47 PM
>> Subject: Re: [Freeswitch-users] send fax via google voice.
>>
>> I'm with Ken, faxing needs to die.
>>
>> Of faxes being sent now, I wonder how many originate and/or terminate via
>> a fax to email gateway. These and the possibility that your call will be
>> sent via VoIP somewhere along the way should strip away the false idea that
>> faxing is secure because it travels over the PSTN. The OP is testing to K7,
>> which is in part?a fax to email?service.
>>
>> Incidentally, the system that terminates K7 faxes is a FreeSwitch box. :-)
>>
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of envelopes
>> envelopes
>> Sent: Monday, March 07, 2011 1:06 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] send fax via google voice.
>>
>> Great!
>>
>> please go ahead claim lawyers' ignorance and lobby german that email pdf
>> is a document as equivalent as fax. We are looking forward to you next great
>> accomplish.? ?
>>
>> On Mar 7, 2011 9:41 AM, "Ken Rice" <krice at freeswitch.org> wrote:
>> > Whats the old saying? Ignorance is bliss? Heh
>> >
>> > Just because and end user perceives it as being more secure doesn?t make
>> > it
>> > so... Even faxing over voip you just tcpdump out the rtp and process it
>> > thru
>> > and you can recover then entire exchange... Same thing with a laptop, a
>> > sound card, and a wav recorder with a cheap passive phone tap... Only
>> > difference with legacy PSTN vs VoIP is its easier to hack a box
>> > somewhere
>> > along the way and intercept the traffic vs finding the trunk line
>> > carrying
>> > the traffic, however, it doesn?t take long to figure out that there are
>> > tons
>> > of unsecure access points to the wire between your office and the CO...
>> > (even the fiber these days can be compromised relatively easily compared
>> > to
>> > what it was 10 years ago)
>> >
>> > My whole point here is I am not malicious when it comes to this sort of
>> > thing but I know what can be done and how cheap the technology to pull
>> > off
>> > these things really is. I smart but I am far from the smartest guy and
>> > if I
>> > can figure out how to do this stuff many people can figure it out...
>> >
>> > I?m still say that the only secure computer is one fired into the sun on
>> > the
>> > tip of a rocket with no comms back to earth.. Then that?s only secure
>> > until
>> > someone figures out how to retrieve it.
>> >
>> > K
>> >
>> >
>> >
>> > On 3/7/11 11:24 AM, "envelopes envelopes" <sunwood360 at gmail.com> wrote:
>> >
>> >> From the perspective of end user,? it is pretty easier to hacker an end
>> >> user'
>> >> inbox globally.? However, you need physical presence to grab fax
>> >> documents.
>> >>
>> >> Interesting to see the view from IT people is so different from
>> >> non-technical
>> >> people. After, any existence has its reason.
>> >>
>> >> On Mar 6, 2011 8:56 AM, "Ken Rice" <krice at freeswitch.org> wrote:
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>>
>> ________________________________
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From s.spiliadis at gmail.com  Tue Mar  8 03:23:43 2011
From: s.spiliadis at gmail.com (Stavros Spiliadis)
Date: Mon, 7 Mar 2011 19:23:43 -0500
Subject: [Freeswitch-users] whistlephone on freeswitch
Message-ID: <AANLkTi=pJSvrFVC_8Mb5BOmmnPDPUk=H-LJEx-zStr7v@mail.gmail.com>

Ok,
Here is what I did. I copied the 5 (five) whistelphone proxys to the
freeswitch SIP_Accounts config file as explained here ?
wiki.freeswitch.org/wiki/Provide???tlePhone<http://wiki.freeswitch.org/wiki/Provider_Configuration:_WhistlePhone>.
All 5 instances are reported as registered (REGED) under
Status--SIP_Status. So far so good. Incoming calls work fine and quality of
sound is good. Outgoing calls give the "... I'm sorry but I cannot complete
your call." Any ideas on what I'm doing wrong, Suggestions? Thank
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From curriegrad2004 at gmail.com  Tue Mar  8 03:34:12 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 7 Mar 2011 16:34:12 -0800
Subject: [Freeswitch-users] whistlephone on freeswitch
In-Reply-To: <AANLkTi=pJSvrFVC_8Mb5BOmmnPDPUk=H-LJEx-zStr7v@mail.gmail.com>
References: <AANLkTi=pJSvrFVC_8Mb5BOmmnPDPUk=H-LJEx-zStr7v@mail.gmail.com>
Message-ID: <AANLkTimOPxPfW9HDA9w+92m+AjnQex=cuJne9FQ18mau@mail.gmail.com>

did you try the ${sip_to_user} variable yet with the 10 digit DID they
assigned you?

On Mon, Mar 7, 2011 at 4:23 PM, Stavros Spiliadis <s.spiliadis at gmail.com> wrote:
> Ok,
> Here is what I did. I copied the 5 (five) whistelphone proxys to the
> freeswitch SIP_Accounts config file as explained here
> ?wiki.freeswitch.org/wiki/Provide???tlePhone . All 5 instances are reported
> as registered (REGED) under Status--SIP_Status. So far so good. Incoming
> calls work fine and quality of sound is good. Outgoing calls give the "...
> I'm sorry but I cannot complete your call." Any ideas on what I'm doing
> wrong, Suggestions? Thank
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From hkalyoncu at gmail.com  Tue Mar  8 10:58:00 2011
From: hkalyoncu at gmail.com (hkalyoncu)
Date: Mon, 7 Mar 2011 23:58:00 -0800 (PST)
Subject: [Freeswitch-users] mod_python
In-Reply-To: <AANLkTim39o1o5Hw2REhmG4+8zqZGHhcmDXRcrd4NKk6F@mail.gmail.com>
References: <1299244469418-6088430.post@n2.nabble.com>
	<AANLkTikQ4NVFE2BLNYxiY05zFaYeq8TJjqbuZJKp3OgG@mail.gmail.com>
	<1299347178939-6092296.post@n2.nabble.com>
	<1299502358077-6097179.post@n2.nabble.com>
	<AANLkTim39o1o5Hw2REhmG4+8zqZGHhcmDXRcrd4NKk6F@mail.gmail.com>
Message-ID: <1299571080264-6112761.post@n2.nabble.com>

thanks for your kind reply but im afraid its not working either.

anyway i have to do it with serialize method. not a nice solution but at
least working :).

here is my solution for anybody who needs it :

        s = Event.serialize(params)
        s2 = s.replace("\'", "")[:-2]
        d = dict(item.split(":") for item in s2.split("\n"))
        caller = d["Caller-Caller-ID-Number"]
        callee = d["Caller-Destination-Number"]


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-python-tp6088430p6112761.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveu at coppice.org  Tue Mar  8 11:04:38 2011
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 08 Mar 2011 16:04:38 +0800
Subject: [Freeswitch-users] send fax via google voice.
In-Reply-To: <843CF6A82C2E44EC91515E847BCD422E@e1705>
References: <C99918A4.126A8%krice@freeswitch.org>
	<843CF6A82C2E44EC91515E847BCD422E@e1705>
Message-ID: <4D75E316.5020603@coppice.org>

On 03/07/2011 01:08 AM, Madovsky wrote:
> I think it's more hard for a guy to hack your US pstn line if he lives
> in Australia... fax on internet can be hacked from everywhere...
There used to be some truth in that, a long time ago. It ended when 
arbitrary hops of ordinary PSTN calls started to be carried over IP 
networks. When you call from a PSTN FAX to a PSTN FAX you have little 
idea how the call travels. Its the same with phone banking by DTMF, and 
other services where people regularly send sensitive information. Its 
hard to mount a focussed attack on a particular person by trawling 
through fairly arbitrary IP communications, but you can sure harvest a 
lot of interesting material from a long distance these days.

Interestingly, the FAX standards do include security feature options, 
but I've never seen them used. The documentation lacks clarity, and I'm 
not sure how well most T.38 systems would handle them, but they are there.

Steve



From erkan at speedingtrade.com  Tue Mar  8 13:39:14 2011
From: erkan at speedingtrade.com (=?iso-8859-9?B?RXJrYW4g3G5s/A==?=)
Date: Tue, 8 Mar 2011 12:39:14 +0200
Subject: [Freeswitch-users] sip auth challenge HACKING ???
Message-ID: <81C2CEF80046FB4F863A60D4347DD33A0C56EC@server1.st.local>

Hi FS Users,

 

in last time i see in my console of FS this kind of error messages.

 

[WARNING] sofia_reg.c:1246 SIP auth challenge (INVITE) on sofia profile 'internal' for [5828@?????????] from ip ???7?.1??.7??.???

 

i check my config files ever again and again, but today the console is only given this kind of messages. Maybe 20 messages per second.

 

i see the ip that given in the console "from ip xx.xx.xx.xx" i block this ip in my firewall and everything is fine.

Now i understand that this a trying to hacking my server. The blocking of the ip is a solution but can not handle this in Freeswitch, because i see this problem sometimes on different FS servers also and in normally the FS server maybe must can handle this problem. For example with automatic black lists if an ip trys more than 20 times with wrong login. So the ip will be banned for 1 hour or so.

 

i'm interesting in if other users have the same problems and ideas in how we can handle this.

 

Kind regards

Erkan

 

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From shamun.toha at gmail.com  Tue Mar  8 13:50:52 2011
From: shamun.toha at gmail.com (Shamun toha md)
Date: Tue, 8 Mar 2011 11:50:52 +0100
Subject: [Freeswitch-users] Skype - video and audio can it be done?
Message-ID: <AANLkTinL1ALv5pBT+pEMODYmMkQzdw8sGjQsziDgQHy+@mail.gmail.com>

Dear Experts,

I have FS Skype running for audio.

- But how can i send a SIP video/audio calls to Skype video/audio? or
is that impossible for video?

Thanks


From jaybinks at gmail.com  Tue Mar  8 13:55:24 2011
From: jaybinks at gmail.com (jay binks)
Date: Tue, 8 Mar 2011 20:55:24 +1000
Subject: [Freeswitch-users] sip auth challenge HACKING ???
In-Reply-To: <81C2CEF80046FB4F863A60D4347DD33A0C56EC@server1.st.local>
References: <81C2CEF80046FB4F863A60D4347DD33A0C56EC@server1.st.local>
Message-ID: <AANLkTi=BTOEPqLEtmaqNrONVC8tX-5tOp-oKxMsA7EOT@mail.gmail.com>

your after Fail2Ban ( which is a separate script )
but it inspects these log messages and adds them to iptables.

http://wiki.freeswitch.org/wiki/Fail2ban

<http://wiki.freeswitch.org/wiki/Fail2ban>Jay



2011/3/8 Erkan ?nl? <erkan at speedingtrade.com>

> Hi FS Users,
>
>
>
> in last time i see in my console of FS this kind of error messages.
>
>
>
> [WARNING] sofia_reg.c:1246 SIP auth challenge (INVITE) on sofia profile
> 'internal' for [5828@?????????] from ip ???7?.1??.7??.???
>
>
>
> i check my config files ever again and again, but today the console is only
> given this kind of messages. Maybe 20 messages per second.
>
>
>
> i see the ip that given in the console ?from ip xx.xx.xx.xx? i block this
> ip in my firewall and everything is fine.
>
> Now i understand that this a trying to hacking my server. The blocking of
> the ip is a solution but can not handle this in Freeswitch, because i see
> this problem sometimes on different FS servers also and in normally the FS
> server maybe must can handle this problem. For example with automatic black
> lists if an ip trys more than 20 times with wrong login. So the ip will be
> banned for 1 hour or so.
>
>
>
> i?m interesting in if other users have the same problems and ideas in how
> we can handle this.
>
>
>
> Kind regards
>
> Erkan
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely

Jay
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From simpot at simpot.com  Tue Mar  8 14:00:13 2011
From: simpot at simpot.com (Dmitry Saratsky)
Date: Tue, 8 Mar 2011 13:00:13 +0200
Subject: [Freeswitch-users] sip auth challenge HACKING ???
In-Reply-To: <81C2CEF80046FB4F863A60D4347DD33A0C56EC@server1.st.local>
References: <81C2CEF80046FB4F863A60D4347DD33A0C56EC@server1.st.local>
Message-ID: <52114D605A462A4E9A50E588EE7D0686012E575D9EF8@mail.forest.simpot.com>

I'm blocking it with: http://wiki.freeswitch.org/wiki/Fail2ban



From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Erkan ?nl?
Sent: 08 Mar 2011 12:39
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] sip auth challenge HACKING ???

Hi FS Users,

in last time i see in my console of FS this kind of error messages.

[WARNING] sofia_reg.c:1246 SIP auth challenge (INVITE) on sofia profile 'internal' for [5828@?????????] from ip ???7?.1??.7??.???

i check my config files ever again and again, but today the console is only given this kind of messages. Maybe 20 messages per second.

i see the ip that given in the console "from ip xx.xx.xx.xx" i block this ip in my firewall and everything is fine.
Now i understand that this a trying to hacking my server. The blocking of the ip is a solution but can not handle this in Freeswitch, because i see this problem sometimes on different FS servers also and in normally the FS server maybe must can handle this problem. For example with automatic black lists if an ip trys more than 20 times with wrong login. So the ip will be banned for 1 hour or so.

i'm interesting in if other users have the same problems and ideas in how we can handle this.

Kind regards
Erkan

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From julf at julf.com  Tue Mar  8 14:05:50 2011
From: julf at julf.com (Johan Helsingius)
Date: Tue, 08 Mar 2011 12:05:50 +0100
Subject: [Freeswitch-users] tones.conf parameters?
In-Reply-To: <AANLkTikV1oG9-45b5mhsk5Xk3vXq+WGuFeOoZWPg8166@mail.gmail.com>
References: <4D7503F5.6010606@julf.com>	<AANLkTin0N+79ZOQc8pMHw4U+CRS0Rc+LQ27YiPEuo0Bp@mail.gmail.com>	<4D751D6A.70605@julf.com>
	<AANLkTikV1oG9-45b5mhsk5Xk3vXq+WGuFeOoZWPg8166@mail.gmail.com>
Message-ID: <4D760D8E.50505@julf.com>

Michael,

> The format is not "documented" per se, but the format can be inferred by looking
> at the existing entries:

Well, it is pretty easy to guess what the names of tones are,
but "-7;%(1000,0,350,440)" was not exactly clear... :)

> Look up TGML on the wiki

Ah, yes, that helps!

> Try these values:

Well, seems my problem is also related to not being able to
get freetdm to generate a dialtone in the first place. About
to dash off on a 3-day trip, but will investigate as soon as
I am back.

> If they work then let us know and we'll add them to the tones.conf file. FYI,
> I'll let you research the "attn" "call-waiting" and "detect-failX" tones for
> your country. :)

Will do! Thanks!

	Julf


From michelhabib at gmail.com  Tue Mar  8 15:18:46 2011
From: michelhabib at gmail.com (Michel Habib)
Date: Tue, 8 Mar 2011 14:18:46 +0200
Subject: [Freeswitch-users] ASR from Freeswitch to MS Speech Server
 [using MRCP Connector] - Audio Problem
Message-ID: <AANLkTim6u-77TfSJ6aypfNxNLjHrDwnW9EYNin-L40w7@mail.gmail.com>

Yes, I get the Audio from FS in regular calls - I already disabled all
possible firewalls - all 3 machines [softphone, freeswitch, Speech Server
(and mrcp connector) ] are on a switch.
192.168.5.107 is the freeswitch server
192.168.5.110 is the MRCP-Connector/Speech Server 2007 server
I made too many iterations on the configuration below:

<include>
  <profile name="mrcp-connector" version="2">
    <param name="client-ip" value="192.168.5.107"/>
    <param name="client-port" value="5090"/>
    <param name="server-ip" value="192.168.5.110"/>
    <param name="server-port" value="5070"/>
    <!--param name="force-destination" value="1"/-->
    <param name="sip-transport" value="udp"/>
    <!--param name="ua-name" value="FreeSWITCH"/-->
    <!--param name="sdp-origin" value="FreeSWITCH"/-->
    <param name="rtp-ext-ip" value="192.168.5.107"/>
    <param name="rtp-ip" value="192.168.5.107"/>
    <param name="rtp-port-min" value="4000"/>
    <param name="rtp-port-max" value="5000"/>
    <!-- enable/disable rtcp support -->
    <param name="rtcp" value="1"/>
    <!-- rtcp bye policies (rtcp must be enabled first)
             0 - disable rtcp bye
             1 - send rtcp bye at the end of session
             2 - send rtcp bye also at the end of each talkspurt (input)
    -->
    <param name="rtcp-bye" value="2"/>
    <!-- rtcp transmission interval in msec (set 0 to disable) -->
    <param name="rtcp-tx-interval" value="5000"/>
    <!-- period (timeout) to check for new rtcp messages in msec (set 0 to
disable) -->
    <param name="rtcp-rx-resolution" value="1000"/>

    <!--param name="playout-delay" value="50"/-->
    <!--param name="max-playout-delay" value="200"/-->
    <!--param name="ptime" value="20"/-->
    <param name="codecs" value="PCMU PCMA L16/96/8000"/>

    <!-- Add any default MRCP params for SPEAK requests here -->
    <synthparams>
    </synthparams>

    <!-- Add any default MRCP params for RECOGNIZE requests here -->
    <recogparams>
      <!--param name="start-input-timers" value="false"/-->
    </recogparams>
  </profile>
</include>


---------- Forwarded message ----------

> From: Christopher Rienzo <cmrienzo at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Mon, 7 Mar 2011 09:32:51 -0500
> Subject: Re: [Freeswitch-users] ASR from Freeswitch to MS Speech Server
> [using MRCP Connector] - Audio Problem
> Do you get audio between FS and your SIP client when not using ASR/TTS?
>
> Show me the MRCP profile configuration and your FreeSWITCH logs during the
> call.
>
>
>
> On Mon, Mar 7, 2011 at 4:55 AM, Michel Habib <michelhabib at gmail.com>wrote:
>
>> Hello All,
>> I have MS OCS Speech Server 2007 [working correctly, as i can make SIP
>> calls and use its ASR and TTS Services successfully]
>> I am also using MRCP Connector from AumTech - which allows me to use ASR
>> and TTS Services through an MRCP Client .
>> Now, i am using Freeswitch mod unimrcp to use ASR and TTS.
>>
>> for TTS, I can successfully make the call, the Audio RTP of the TTS voice
>> is transferred succesfully from Speech Server [through MRCP Connector] back
>> to the Freeswitch Server.
>> However, Freeswitch is not sending back the Audio RTP to the SIP client.
>>
>> for ASR, I can successfully define the grammar and start recognition, but
>> the audio RTP sent to speech server [through MRCP Connector] is silent
>> [empty].
>>
>> I am suspecting something is wrong with the RTP Configuration - can you
>> help me?
>>
>> Let me now if you need any specific logs/scripts/configuration?
>>
>> Thank you,
>> Michel.
>>
>>
>>
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From leon at scarlet-internet.nl  Tue Mar  8 16:07:44 2011
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Tue, 8 Mar 2011 14:07:44 +0100
Subject: [Freeswitch-users] static rtpmap entries are not shown in
	outbound invites
In-Reply-To: <AANLkTimF+rofWtwTN7PaRY5t9_xbgc6qYfMz4s8i6se6@mail.gmail.com>
References: <DD73E3C2-BFA2-406E-B3B5-0078AECCD329@scarlet-internet.nl>
	<0D61D75E-A552-49F3-8A76-F3010BBE32B7@freeswitch.org>
	<5363DF5E-9569-4FEA-8C68-BB87CB87EBA2@scarlet-internet.nl>
	<AANLkTimF+rofWtwTN7PaRY5t9_xbgc6qYfMz4s8i6se6@mail.gmail.com>
Message-ID: <27C6F133-1B03-42F7-96FE-03044F4817BC@scarlet-internet.nl>

Hi Steven,

Ah yes, I knew it meant mono. But I found the problem..

* The call that worked properly with G729/8000 was on a sip profile that had G729 allowed in inbound-codec-prefs
* The call that didn't work was on a second profile that apparently didn't have G729 set for inbound-codec-prefs (I did specify it in the xml, but only did a reload on the profile which isn't sufficient - I should've restarted it -- doh)

Thanks all & kind regards,

Leon






On Mar 7, 2011, at 6:57 PM, Steven Ayre wrote:

>> Can it be the trailing /1 (encoding parameters) on both rtpmaps ? - According to parse_rtpmap() in libsofia it should be parsed alright, but still, ep_codec_string gets filled when I receive an SDP like this (from another endpoint):
> 
> That means 1 channel (mono), which is the default. Although optional
> it is correct and *should* be handled fine...
> 
> -Steve
> 
> 
> 
> On 7 March 2011 13:59, Leon de Rooij <leon at scarlet-internet.nl> wrote:
>> Hi Brian,
>> 
>> Thanks, that works :-)
>> 
>> I got one last question about sdp, would you know why chan var ep_codec_string isn't set after an incoming invite with the following sdp:
>> 
>> v=0.
>> o=FOOBAR 123456 654321 IN IP4 1.2.3.4.
>> s=-.
>> c=IN IP4 1.2.3.4.
>> t=0 0.
>> m=audio 22262 RTP/AVP 18 101.
>> a=rtpmap:18 G729/8000/1.
>> a=rtpmap:101 telephone-event/8000/1.
>> a=fmtp:101 0-15.
>> 
>> Can it be the trailing /1 (encoding parameters) on both rtpmaps ? - According to parse_rtpmap() in libsofia it should be parsed alright, but still, ep_codec_string gets filled when I receive an SDP like this (from another endpoint):
>> 
>> v=0.
>> o=BARFOO 20494030 20494030 IN IP4 4.3.2.1.
>> s=-.
>> c=IN IP4 4.3.2.1.
>> t=0 0.
>> m=audio 50000 RTP/AVP 18 100.
>> a=rtpmap:18 G729/8000.
>> a=sendrecv.
>> a=ptime:20.
>> a=rtpmap:100 telephone-event/8000.
>> a=fmtp:100 0-15.
>> 
>> 
>> (btw, I'm having inbound- and outbound-codec-prefs on the incoming profile set as: "G7221 at 32000h,G7221 at 16000h,G722,PCMA,PCMU,G729,GSM" and am doing inbound-late-negotiation="true")
>> 
>> 
>> Thanks,
>> 
>> Leon
>> 
>> 
>> 
>> 
>> On Mar 4, 2011, at 7:09 PM, Brian West wrote:
>> 
>>> verbose_sdp=true
>>> 
>>> docs/ChangeLog: mod_sofia: apparently some sip device vendors did not read the RFC (who knew?) adding verbose_sdp=true var to add needless a= lines for standard iana codecs that explicitly do not require them (r:6c4f49a8)
>>> 
>>> /b
>>> 
>>> On Mar 4, 2011, at 12:06 PM, Leon de Rooij wrote:
>>> 
>>>> Can someone help me with this ?
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From gmaruzz at gmail.com  Tue Mar  8 16:22:27 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 8 Mar 2011 14:22:27 +0100
Subject: [Freeswitch-users] Skype - video and audio can it be done?
In-Reply-To: <AANLkTinL1ALv5pBT+pEMODYmMkQzdw8sGjQsziDgQHy+@mail.gmail.com>
References: <AANLkTinL1ALv5pBT+pEMODYmMkQzdw8sGjQsziDgQHy+@mail.gmail.com>
Message-ID: <AANLkTimxa8LhPAF9_0sSAS5jkv1xea7k94jahFjM4Mu3@mail.gmail.com>

no way to interact with skype video.

It's just not in the Skype API.

-giovanni

On Tue, Mar 8, 2011 at 11:50 AM, Shamun toha md <shamun.toha at gmail.com> wrote:
> Dear Experts,
>
> I have FS Skype running for audio.
>
> - But how can i send a SIP video/audio calls to Skype video/audio? or
> is that impossible for video?
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From johns1433 at gmail.com  Tue Mar  8 17:20:19 2011
From: johns1433 at gmail.com (John Smith)
Date: Tue, 8 Mar 2011 15:20:19 +0100
Subject: [Freeswitch-users] DTMF not taken into account within a
	conference
Message-ID: <AANLkTikKEgDAO-j7nf0dYvPqn+5ZXY5c74tbvkDdDuXR@mail.gmail.com>

Yes, you are right. There were an error in my scripts and I used a version
released the 20th of February.
I tested with the latest git (git-b3a2fa1 2011-03-03 20-07-43 -0600) and
everything is working fine.
Thanks a lot for your help.
John


2011/3/7 Brian West <brian at freeswitch.org>

> Well I have a feeling you haven't really updated to the VERY latest GIT
> HEAD have you?  We had one day this past month that had this EXACT bug in it
> and has since been fixed.
>
> /b
>
> On Mar 7, 2011, at 11:42 AM, John Smith wrote:
>
> > When I enter a DTMF within a conference it only works the first time. For
> > example if I press 0 I?m muted but afterwards my DTMF commands are not
> taken
> > into account whatever I press.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From hkalyoncu at gmail.com  Tue Mar  8 17:51:44 2011
From: hkalyoncu at gmail.com (hkalyoncu)
Date: Tue, 8 Mar 2011 06:51:44 -0800 (PST)
Subject: [Freeswitch-users] mod_python
In-Reply-To: <1299571080264-6112761.post@n2.nabble.com>
References: <1299244469418-6088430.post@n2.nabble.com>
	<AANLkTikQ4NVFE2BLNYxiY05zFaYeq8TJjqbuZJKp3OgG@mail.gmail.com>
	<1299347178939-6092296.post@n2.nabble.com>
	<1299502358077-6097179.post@n2.nabble.com>
	<AANLkTim39o1o5Hw2REhmG4+8zqZGHhcmDXRcrd4NKk6F@mail.gmail.com>
	<1299571080264-6112761.post@n2.nabble.com>
Message-ID: <1299595904276-6126290.post@n2.nabble.com>

u dont have to serialize!
all u have to do is this:

        caller = Event.getHeader(params, 'Caller-Caller-ID-Number')
        callee = Event.getHeader(params, 'Caller-Destination-Number')


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-python-tp6088430p6126290.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From curriegrad2004 at gmail.com  Tue Mar  8 18:32:06 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 8 Mar 2011 07:32:06 -0800
Subject: [Freeswitch-users] sip auth challenge HACKING ???
In-Reply-To: <52114D605A462A4E9A50E588EE7D0686012E575D9EF8@mail.forest.simpot.com>
References: <81C2CEF80046FB4F863A60D4347DD33A0C56EC@server1.st.local>
	<52114D605A462A4E9A50E588EE7D0686012E575D9EF8@mail.forest.simpot.com>
Message-ID: <AANLkTi=o-5oFhnBcgA=KX9Zo+i1AgisusyH88zw067Gk@mail.gmail.com>

Or just for fun, you can set up a honeypot with all extensions routing
to nowhere or to a very very nasty extension ;)

On Tue, Mar 8, 2011 at 3:00 AM, Dmitry Saratsky <simpot at simpot.com> wrote:
> I?m blocking it with: http://wiki.freeswitch.org/wiki/Fail2ban
>
>
>
>
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Erkan
> ?nl?
> Sent: 08 Mar 2011 12:39
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] sip auth challenge HACKING ???
>
>
>
> Hi FS Users,
>
>
>
> in last time i see in my console of FS this kind of error messages.
>
>
>
> [WARNING] sofia_reg.c:1246 SIP auth challenge (INVITE) on sofia profile
> 'internal' for [5828@?????????] from ip ???7?.1??.7??.???
>
>
>
> i check my config files ever again and again, but today the console is only
> given this kind of messages. Maybe 20 messages per second.
>
>
>
> i see the ip that given in the console ?from ip xx.xx.xx.xx? i block this ip
> in my firewall and everything is fine.
>
> Now i understand that this a trying to hacking my server. The blocking of
> the ip is a solution but can not handle this in Freeswitch, because i see
> this problem sometimes on different FS servers also and in normally the FS
> server maybe must can handle this problem. For example with automatic black
> lists if an ip trys more than 20 times with wrong login. So the ip will be
> banned for 1 hour or so.
>
>
>
> i?m interesting in if other users have the same problems and ideas in how we
> can handle this.
>
>
>
> Kind regards
>
> Erkan
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From a.afzali2003 at gmail.com  Tue Mar  8 18:58:42 2011
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Tue, 8 Mar 2011 19:28:42 +0330
Subject: [Freeswitch-users] Limit the size of queue in mod_fifo
In-Reply-To: <AANLkTinU_j8zRqKWvVfp=DXedPP83dSx09KRUbgo-rKb@mail.gmail.com>
References: <AANLkTinU_j8zRqKWvVfp=DXedPP83dSx09KRUbgo-rKb@mail.gmail.com>
Message-ID: <AANLkTimFGo0xrb07a1b9SOoije-rkamGua8tViUuwWz9@mail.gmail.com>

Maybe mod_limit be a solution

-- afshin

On Mon, Mar 7, 2011 at 4:06 PM, Marc de Corny <marcdecorny at gmail.com> wrote:

> Hi All,
>
> does anyone know of a way of limiting the numbers of calls that are held in
> the mod_fifo queue so that we can deny more calls after having reacded that
> limit ?
>
> thanks
> Marc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jeff at jefflenk.com  Tue Mar  8 21:42:25 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 8 Mar 2011 10:42:25 -0800 (PST)
Subject: [Freeswitch-users] database is locked
In-Reply-To: <AANLkTi=xM2B6ubsJuoDzRD+TxgZkp+gBhebyUUa3CmD-@mail.gmail.com>
References: <AANLkTimHPa50yukZfq9P5emxMP=5uGizdPiTVeiV4N0C@mail.gmail.com>
	<AANLkTik3TC1ALNzYfM+xnP7CNR40EBUWT6N_6D8xsavE@mail.gmail.com>
	<AANLkTi=kJpAT3Fia8TqiXatE7AORW4Zt508RMjoTJ_a=@mail.gmail.com>
	<AANLkTi=xM2B6ubsJuoDzRD+TxgZkp+gBhebyUUa3CmD-@mail.gmail.com>
Message-ID: <1299609745053-6134748.post@n2.nabble.com>

update to git head and try again - changes were commited that fix this

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/database-is-locked-tp6098671p6134748.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mishraspecific at gmail.com  Tue Mar  8 07:43:33 2011
From: mishraspecific at gmail.com (Manish Mishra)
Date: Tue, 8 Mar 2011 04:43:33 +0000 (UTC)
Subject: [Freeswitch-users] Invitation to connect on LinkedIn
Message-ID: <1138856109.8669932.1299559413267.JavaMail.app@ela4-bed80.prod>

LinkedIn
------------

   
I'd like to add you to my professional network on LinkedIn.

- Manish

Manish Mishra
Mobile App Developer at Techshastra (I) Pvt Ltd Mumbai 
Mumbai Area, India

Confirm that you know Manish Mishra
https://www.linkedin.com/e/xbphn8-gl0c4cun-4t/isd/2460946473/THWHKqcI/


 
-- 
(c) 2011, LinkedIn Corporation
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From jeroeneeuwes at gmail.com  Tue Mar  8 14:41:45 2011
From: jeroeneeuwes at gmail.com (Jeroen Eeuwes)
Date: Tue, 8 Mar 2011 12:41:45 +0100
Subject: [Freeswitch-users] tones.conf parameters?
In-Reply-To: <4D760D8E.50505@julf.com>
References: <4D7503F5.6010606@julf.com>
	<AANLkTin0N+79ZOQc8pMHw4U+CRS0Rc+LQ27YiPEuo0Bp@mail.gmail.com>
	<4D751D6A.70605@julf.com>
	<AANLkTikV1oG9-45b5mhsk5Xk3vXq+WGuFeOoZWPg8166@mail.gmail.com>
	<4D760D8E.50505@julf.com>
Message-ID: <AANLkTiksxcOwtm4Dz-6HEqFtfNbwsNdta1shwfqgHgOt@mail.gmail.com>

Hi Michael and Johan,

> If they work then let us know and we'll add them to the tones.conf file. FYI,
> I'll let you research the "attn" "call-waiting" and "detect-failX" tones for
> your country. :)

I'm not sure I understand correclty, but does this help with the needed tones?

http://vxlabs.com/2011/02/05/sipura-linksys-cisco-spa3102-voice-gateway-in-the-netherlands/

Best regards,
Jeroen Eeuwes


From sascha.daniels at amooma.de  Tue Mar  8 16:30:35 2011
From: sascha.daniels at amooma.de (Sascha Daniels)
Date: Tue, 08 Mar 2011 14:30:35 +0100
Subject: [Freeswitch-users] Compile error on Suse 11.3
Message-ID: <4D762F7B.4060108@amooma.de>

Hi together.

I did build FS on different systems without any problems.

On Suse 11.3 (32bit) I get the following error:

Compiling src/switch_core_sqldb.c ...
cc1: warnings being treated as errors
src/switch_core_sqldb.c: In function '_switch_cache_db_get_db_handle':
src/switch_core_sqldb.c:314:47: error: comparison between
'switch_odbc_status_t' and 'enum <anonymous>'
src/switch_core_sqldb.c: In function 'switch_cache_db_execute_sql_real':
src/switch_core_sqldb.c:389:90: error: comparison between
'switch_status_t' and 'enum <anonymous>'
make[1]: *** [libfreeswitch_la-switch_core_sqldb.lo] Fehler 1
make: *** [all] Fehler 2

Fehler = Error

I did not find a solution anywhere in the internet.

Any hints?

Regards

                    Sascha

-- 
AMOOMA GmbH - Bachstr. 124 - 56566 Neuwied  -->  http://www.amooma.de
Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister Montabaur B14998

B?cher: http://das-asterisk-buch.de - http://ruby-auf-schienen.de



From mh at markholloway.com  Tue Mar  8 18:52:10 2011
From: mh at markholloway.com (Mark Holloway)
Date: Tue, 8 Mar 2011 08:52:10 -0700
Subject: [Freeswitch-users] sip auth challenge HACKING ???
In-Reply-To: <AANLkTi=o-5oFhnBcgA=KX9Zo+i1AgisusyH88zw067Gk@mail.gmail.com>
References: <81C2CEF80046FB4F863A60D4347DD33A0C56EC@server1.st.local>
	<52114D605A462A4E9A50E588EE7D0686012E575D9EF8@mail.forest.simpot.com>
	<AANLkTi=o-5oFhnBcgA=KX9Zo+i1AgisusyH88zw067Gk@mail.gmail.com>
Message-ID: <07A18C43-F674-4B54-A383-4A1A623D98D2@markholloway.com>

A session border controller would solve your problem.  You can configure it so any IP that fails SIP authentication X number of times within X number of seconds/minutes will get "demoted" and no additional SIP messages will get through to Freeswtich.  You can set the hold-down timer on the SBC so the IP isn't demoted forever. Some refer to it as SIP DoS, DDoS, call it what you want, but this functionality is best handled by the SBC.

On Mar 8, 2011, at 8:32 AM, curriegrad2004 wrote:

> Or just for fun, you can set up a honeypot with all extensions routing
> to nowhere or to a very very nasty extension ;)
> 
> On Tue, Mar 8, 2011 at 3:00 AM, Dmitry Saratsky <simpot at simpot.com> wrote:
>> I?m blocking it with: http://wiki.freeswitch.org/wiki/Fail2ban
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Erkan
>> ?nl?
>> Sent: 08 Mar 2011 12:39
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: [Freeswitch-users] sip auth challenge HACKING ???
>> 
>> 
>> 
>> Hi FS Users,
>> 
>> 
>> 
>> in last time i see in my console of FS this kind of error messages.
>> 
>> 
>> 
>> [WARNING] sofia_reg.c:1246 SIP auth challenge (INVITE) on sofia profile
>> 'internal' for [5828@?????????] from ip ???7?.1??.7??.???
>> 
>> 
>> 
>> i check my config files ever again and again, but today the console is only
>> given this kind of messages. Maybe 20 messages per second.
>> 
>> 
>> 
>> i see the ip that given in the console ?from ip xx.xx.xx.xx? i block this ip
>> in my firewall and everything is fine.
>> 
>> Now i understand that this a trying to hacking my server. The blocking of
>> the ip is a solution but can not handle this in Freeswitch, because i see
>> this problem sometimes on different FS servers also and in normally the FS
>> server maybe must can handle this problem. For example with automatic black
>> lists if an ip trys more than 20 times with wrong login. So the ip will be
>> banned for 1 hour or so.
>> 
>> 
>> 
>> i?m interesting in if other users have the same problems and ideas in how we
>> can handle this.
>> 
>> 
>> 
>> Kind regards
>> 
>> Erkan
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Tue Mar  8 23:26:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 8 Mar 2011 12:26:38 -0800
Subject: [Freeswitch-users] Standard Functionality of Freeswitch as
 Conference Server? Automatic Listen only conference.
In-Reply-To: <COL124-W233179629C8AC5F32450BF85C50@phx.gbl>
References: <COL124-W233179629C8AC5F32450BF85C50@phx.gbl>
Message-ID: <AANLkTi=fQhxHryh3uwSWv-zAQBXBdStC1CFQPjtXcvTM@mail.gmail.com>

Phil,

First off, welcome to FreeSWITCH! And yes, this kind of functionality is
completely doable in FreeSWITCH. You'd need to spend a little time learning
the conference application and the dialplan but it's totally within reach.

As far as GUIs go, I believe that FusionPBX has some level of conference
controls but I can't speak to the specifics. They have an ISO that you can
download and throw into a virtual box and try out.

Hope this helps.
-MC

On Sat, Mar 5, 2011 at 4:11 AM, Phil T <philtrick123 at hotmail.com> wrote:

>  Hi,
>
> I'm trying to setup a conference server for a local charity but I wanted to
> check on the standard conference functionality of Freeswitch before I
> started.
> The requirements are quite simple- Conference call only (no extra pbx
> features) for up to 10 people (via Public SIP trunks), but the critical part
> is there will be one presenter & 9 people listening only. So ideally I want
> to have two pin codes- one for use by the presenter, and the other pin code
> for listen only attendees. The listen only people should be automatically
> muted when they join, and obviously the presenter should have full duplex
> audio. There might be the requirement to allow individual listen only
> attendee to opt in/out to speak on the call (via a feature code i.e. *6).
>
> This type of feature is common among commercial conference solutions, is it
> available on Freeswitch via standard scripting without having to have a
> custom programmed module?
>
> If the answer is yes- I'll get started on setting it up. Any tips greatly
> appreciated!
>
> Finally- I have very little linux command line experience, so I'd like to
> do the xml script setup/ testing on a Windows 7 pc, then move over to Linux
> if the performance is required. Is that a sensible approach? Is there a GUI
> to help with the Freeswitch setup?
>
>
> Thanks Phil
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jjj at 3js.de  Wed Mar  9 00:19:02 2011
From: jjj at 3js.de (Johannes Jakob)
Date: Tue, 08 Mar 2011 22:19:02 +0100
Subject: [Freeswitch-users] Virtualisation and freeswitch
In-Reply-To: <AANLkTikiQUXWnrWZxoLHWOb3DL6MvXsYBmTdCUkcgjBK@mail.gmail.com>
References: <AANLkTimZTwZfUYMjC1ddAQMWHHWSLH=BLajv0cgSux4B@mail.gmail.com>
	<AANLkTikQbXLdCb4nNFEbfte-EAP8ytJyfURMALC6kwtK@mail.gmail.com>
	<AANLkTinB17uNUKtEQAtuTgCGPf_61jnErGmv7+vXV_n+@mail.gmail.com>
	<AANLkTimtnMqFyrXFEYirmhLgqPJxaWsYWowYyiLi_k_=@mail.gmail.com>
	<AANLkTikhDhRL_hXcosBQzq9_vD75_=G0Ju8+3nJ9jyhn@mail.gmail.com>
	<AANLkTikiQUXWnrWZxoLHWOb3DL6MvXsYBmTdCUkcgjBK@mail.gmail.com>
Message-ID: <8E9CD4A5-871D-4082-809D-D684B2FC633F@3js.de>

Hi,

since I'm running a Debian Squeeze domU in a Citrix Xen dom0 with kernel 
2.6.32-4-amd64 and having serious latency issues and audio hick-ups in 
longer calls, I've to look closer at timing issues.

I didn't see any error messages or warnings, but maybe I don't have the 
right debug options set.

# cat /sys/devices/system/clocksource/clocksource0/available_clocksource
xen

# cat /sys/devices/system/clocksource/clocksource0/current_clocksource
xen


The box is running chrony to provide accurate date&time.

Below you'll find the output of timer_test, but I need some help 
interpreting this. In comparison to a bare metal box, the last column 
shows more "jitter" in the virtual box, but how bad are my values?

According to a little program I found here: 
http://www.advenage.com/topics/linux-timer-interrupt-frequency.php :

kernel timer interrupt frequency is approx. 4016 Hz or higher

But how do I check the kernel timer resolution?


All of this leaves me with the following questions:
1) Can I tweak the system to get better results?
2) Would it be better when using CentOS? Would I still have this 
problem, or would it be completely gone?
3) Is my latency problem (up to 30seconds of latency(!!!)) timer 
related, or do I have an other serious problem?

There aren't any networking issues! Gigabit connections, private 
interconnect to the upstream provider and no packet loss at all.



freeswitch at internal> timer_test
Avg: 19.947ms Total Time: 999.762ms

2011-03-08 21:45:42.287865 [CONSOLE] mod_commands.c:461 Timer Test: 
samplecount after init: 1
freeswitch at internal> 2011-03-08 21:45:42.293298 [CONSOLE] 
mod_commands.c:466 Timer Test: samplecount after first step: 2
2011-03-08 21:45:42.312859 [CONSOLE] mod_commands.c:475 Timer Test: 1 
sleep 20 19520
2011-03-08 21:45:42.333572 [CONSOLE] mod_commands.c:475 Timer Test: 2 
sleep 20 20644
2011-03-08 21:45:42.353187 [CONSOLE] mod_commands.c:475 Timer Test: 3 
sleep 20 19573
2011-03-08 21:45:42.373144 [CONSOLE] mod_commands.c:475 Timer Test: 4 
sleep 20 19994
2011-03-08 21:45:42.393090 [CONSOLE] mod_commands.c:475 Timer Test: 5 
sleep 20 19900
2011-03-08 21:45:42.412943 [CONSOLE] mod_commands.c:475 Timer Test: 6 
sleep 20 19782
2011-03-08 21:45:42.432720 [CONSOLE] mod_commands.c:475 Timer Test: 7 
sleep 20 19731
2011-03-08 21:45:42.452532 [CONSOLE] mod_commands.c:475 Timer Test: 8 
sleep 20 19835
2011-03-08 21:45:42.473430 [CONSOLE] mod_commands.c:475 Timer Test: 9 
sleep 20 20857
2011-03-08 21:45:42.493169 [CONSOLE] mod_commands.c:475 Timer Test: 10 
sleep 20 19706
2011-03-08 21:45:42.512776 [CONSOLE] mod_commands.c:475 Timer Test: 11 
sleep 20 19563
2011-03-08 21:45:42.533521 [CONSOLE] mod_commands.c:475 Timer Test: 12 
sleep 20 20700
2011-03-08 21:45:42.553103 [CONSOLE] mod_commands.c:475 Timer Test: 13 
sleep 20 19531
2011-03-08 21:45:42.572708 [CONSOLE] mod_commands.c:475 Timer Test: 14 
sleep 20 19536
2011-03-08 21:45:42.593020 [CONSOLE] mod_commands.c:475 Timer Test: 15 
sleep 20 20352
2011-03-08 21:45:42.612628 [CONSOLE] mod_commands.c:475 Timer Test: 16 
sleep 20 19525
2011-03-08 21:45:42.633322 [CONSOLE] mod_commands.c:475 Timer Test: 17 
sleep 20 20686
2011-03-08 21:45:42.652949 [CONSOLE] mod_commands.c:475 Timer Test: 18 
sleep 20 19701
2011-03-08 21:45:42.672995 [CONSOLE] mod_commands.c:475 Timer Test: 19 
sleep 20 19891
2011-03-08 21:45:42.693032 [CONSOLE] mod_commands.c:475 Timer Test: 20 
sleep 20 19990
2011-03-08 21:45:42.713020 [CONSOLE] mod_commands.c:475 Timer Test: 21 
sleep 20 19870
2011-03-08 21:45:42.732986 [CONSOLE] mod_commands.c:475 Timer Test: 22 
sleep 20 19896
2011-03-08 21:45:42.752899 [CONSOLE] mod_commands.c:475 Timer Test: 23 
sleep 20 19839
2011-03-08 21:45:42.772596 [CONSOLE] mod_commands.c:475 Timer Test: 24 
sleep 20 19631
2011-03-08 21:45:42.793190 [CONSOLE] mod_commands.c:475 Timer Test: 25 
sleep 20 20596
2011-03-08 21:45:42.812808 [CONSOLE] mod_commands.c:475 Timer Test: 26 
sleep 20 19542
2011-03-08 21:45:42.833619 [CONSOLE] mod_commands.c:475 Timer Test: 27 
sleep 20 20719
2011-03-08 21:45:42.853174 [CONSOLE] mod_commands.c:475 Timer Test: 28 
sleep 20 19651
2011-03-08 21:45:42.872883 [CONSOLE] mod_commands.c:475 Timer Test: 29 
sleep 20 19567
2011-03-08 21:45:42.893592 [CONSOLE] mod_commands.c:475 Timer Test: 30 
sleep 20 20591
2011-03-08 21:45:42.913173 [CONSOLE] mod_commands.c:475 Timer Test: 31 
sleep 20 19613
2011-03-08 21:45:42.932796 [CONSOLE] mod_commands.c:475 Timer Test: 32 
sleep 20 19547
2011-03-08 21:45:42.953478 [CONSOLE] mod_commands.c:475 Timer Test: 33 
sleep 20 20651
2011-03-08 21:45:42.973076 [CONSOLE] mod_commands.c:475 Timer Test: 34 
sleep 20 19537
2011-03-08 21:45:42.992682 [CONSOLE] mod_commands.c:475 Timer Test: 35 
sleep 20 19558
2011-03-08 21:45:43.013388 [CONSOLE] mod_commands.c:475 Timer Test: 36 
sleep 20 20648
2011-03-08 21:45:43.033012 [CONSOLE] mod_commands.c:475 Timer Test: 37 
sleep 20 19558
2011-03-08 21:45:43.052626 [CONSOLE] mod_commands.c:475 Timer Test: 38 
sleep 20 19545
2011-03-08 21:45:43.073304 [CONSOLE] mod_commands.c:475 Timer Test: 39 
sleep 20 20579
2011-03-08 21:45:43.092915 [CONSOLE] mod_commands.c:475 Timer Test: 40 
sleep 20 19631
2011-03-08 21:45:43.112525 [CONSOLE] mod_commands.c:475 Timer Test: 41 
sleep 20 19515
2011-03-08 21:45:43.133227 [CONSOLE] mod_commands.c:475 Timer Test: 42 
sleep 20 20660
2011-03-08 21:45:43.152862 [CONSOLE] mod_commands.c:475 Timer Test: 43 
sleep 20 19646
2011-03-08 21:45:43.173575 [CONSOLE] mod_commands.c:475 Timer Test: 44 
sleep 20 20647
2011-03-08 21:45:43.193157 [CONSOLE] mod_commands.c:475 Timer Test: 45 
sleep 20 19537
2011-03-08 21:45:43.212762 [CONSOLE] mod_commands.c:475 Timer Test: 46 
sleep 20 19535
2011-03-08 21:45:43.233524 [CONSOLE] mod_commands.c:475 Timer Test: 47 
sleep 20 20669
2011-03-08 21:45:43.252520 [CONSOLE] mod_commands.c:475 Timer Test: 48 
sleep 20 19003
2011-03-08 21:45:43.273352 [CONSOLE] mod_commands.c:475 Timer Test: 49 
sleep 20 20722
2011-03-08 21:45:43.293062 [CONSOLE] mod_commands.c:475 Timer Test: 50 
sleep 20 19676

freeswitch at internal> timer_test 20 3
Avg: 19.792ms Total Time: 59.610ms

2011-03-08 21:48:34.467320 [CONSOLE] mod_commands.c:461 Timer Test: 
samplecount after init: 1
freeswitch at internal> 2011-03-08 21:48:34.473424 [CONSOLE] 
mod_commands.c:466 Timer Test: samplecount after first step: 2
2011-03-08 21:48:34.493299 [CONSOLE] mod_commands.c:475 Timer Test: 1 
sleep 20 19738
2011-03-08 21:48:34.513203 [CONSOLE] mod_commands.c:475 Timer Test: 2 
sleep 20 19833
2011-03-08 21:48:34.533075 [CONSOLE] mod_commands.c:475 Timer Test: 3 
sleep 20 19805


On 29 Jan 2011, at 21:06, Chris Burns wrote:

> Many Xen kernels are compiled with a lower resolution kernel timer. 
> Most
> applications will not notice this, but a VoIP application doing async 
> RTP
> probably will. It can make audio quality poor, especially if something 
> other
> than FreeSWITCH is trying to get CPU time while you are handling 
> media. That
> is the main reason that I found. I have had OK results with Xen with 
> the
> proper configuration.


From msc at freeswitch.org  Wed Mar  9 00:50:06 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 8 Mar 2011 13:50:06 -0800
Subject: [Freeswitch-users] native default sounds
In-Reply-To: <2102851715E74577A63C93A5180F122A@e1705>
References: <2102851715E74577A63C93A5180F122A@e1705>
Message-ID: <AANLkTinKe1OhFQ6oPwEnmHtx2y9L9+ochy8P70h9dzMM@mail.gmail.com>

On Sun, Mar 6, 2011 at 1:06 PM, Madovsky <infos at madovsky.org> wrote:

>  are default us callie sounds in speex already exist somewhere ?
> or should I need to convert them on my own ?
>
On yer own, buddy! :)
-MC
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From msc at freeswitch.org  Wed Mar  9 00:49:26 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 8 Mar 2011 13:49:26 -0800
Subject: [Freeswitch-users] tls/srtp goes straight to voicemail
In-Reply-To: <2160E963-E7A7-48D3-A715-DD3C02F06FB0@gmail.com>
References: <2160E963-E7A7-48D3-A715-DD3C02F06FB0@gmail.com>
Message-ID: <AANLkTinpfSR4ht-=4Bq+RjXUmKHHtG1KHEXbKO7i1vtB@mail.gmail.com>

Are you sure that the Eyebeam clients support SRTP? It looks like you're
getting a 406:

2011-03-05 23:21:08.650486 [NOTICE] sofia.c:5323 Hangup sofia/internal/
sip:1005 at 192.168.60.135:53664 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

You probably need to look directly at the SIP traces and see if there is any
more information as to what's going on.

-MC
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From infos at madovsky.org  Wed Mar  9 02:24:23 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 8 Mar 2011 18:24:23 -0500
Subject: [Freeswitch-users] incoming call and callee busy
Message-ID: <CF955E05ED054AE3A8635355005DC7B1@e1705>

What's the A leg SIP return status of the callee if
the calle is already in early media status with another call ? 

Thanks
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From bwibowo at gmail.com  Wed Mar  9 03:17:12 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Wed, 9 Mar 2011 07:17:12 +0700
Subject: [Freeswitch-users] nibblebill cdr
Message-ID: <AANLkTikqy8O50YJZxDUBiE=qSjHZ7JvHOm5F6cZMOVvZ@mail.gmail.com>

hi
is there any possibility that mod_nibblebill create cdr with call pricing?
right now i just see the balance deducted, in fs_cli console i can see the
call info


regards

budi
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From sos at sokhapkin.dyndns.org  Wed Mar  9 03:22:47 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 8 Mar 2011 19:22:47 -0500
Subject: [Freeswitch-users] nibblebill cdr
In-Reply-To: <AANLkTikqy8O50YJZxDUBiE=qSjHZ7JvHOm5F6cZMOVvZ@mail.gmail.com>
References: <AANLkTikqy8O50YJZxDUBiE=qSjHZ7JvHOm5F6cZMOVvZ@mail.gmail.com>
Message-ID: <201103081922.47958.sos@sokhapkin.dyndns.org>

Nibblebil is not related to CDRs, use mod_xml_cdr to store CDRs.

On Tuesday 08 March 2011, budi wibowo wrote:
> hi
> is there any possibility that mod_nibblebill create cdr with call pricing?
> right now i just see the balance deducted, in fs_cli console i can see the
> call info
> 
> 
> regards
> 
> budi



From msc at freeswitch.org  Wed Mar  9 08:06:10 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 8 Mar 2011 21:06:10 -0800
Subject: [Freeswitch-users] incompatible destination
In-Reply-To: <AANLkTincX-zpfnqx4dRk2jLCPf6HDiGN+cuPDA7+75WX@mail.gmail.com>
References: <AANLkTincX-zpfnqx4dRk2jLCPf6HDiGN+cuPDA7+75WX@mail.gmail.com>
Message-ID: <AANLkTi=LwtDEF6-RumWCw2dLThTjcZXwET9_O_OiMN96@mail.gmail.com>

On Sun, Mar 6, 2011 at 10:20 PM, Sam <u2nsam at gmail.com> wrote:

> Hello,
>
>
> The extension 7006 was working few days ago and now it not working,rest all
> the phone are working properly.
>

Sam,

Is 192.168.2.190 your FreeSWITCH box? If so then the issue is with the phone
not accepting the PCMA codec. Go into your Polycom config and make sure that
PCMA is enabled on the phone itself. The other option would be to let the
Polycom speak in G722 and let FS do transcoding. In any case, if I were you
I would find another working Polycom phone and mimic the configs on this
non-working Poly 335.

-MC


>
> U 192.168.2.190:5060 -> 192.168.2.14:5060
>   INVITE sip:7006 at 192.168.2.14:5060 SIP/2.0..Via: SIP/2.0/UDP
> 192.168.2.190;rport;branch=z9hG4bK8mmBB5e8Z2mKr..Max-Forwards: 70..From: ""
> <sip:9322273640 at 192.168.2.190>;tag=4QggZ1XcNZD3F..To: <
> sip:7006 at 192.168.2.14:5060>..Call-ID: 7bf
>   ae496-c324-122e-bc85-0026b976aef7..CSeq: 9403059 INVITE..Contact: <
> sip:mod_sofia at 192.168.2.190:5060>..User-Agent: NOVANET..Allow: INVITE,
> ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY,
> PUBLISH, SUBSCRIBE.
>   .Supported: timer, precondition, path, replaces..Allow-Events: talk,
> hold, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary,
> refer..Content-Type: application/sdp..Content-Dis
>   position: session..Content-Length: 203..X-FS-Support:
> update_display..Remote-Party-ID: <sip:9322273640 at 192.168.2.190>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH
> 1299446773 1299446774 IN IP4 192.168.2.190..s=FreeSWITCH
>   ..c=IN IP4 192.168.2.190..t=0 0..m=audio 31474 RTP/AVP 8 101
> 13..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=ptime:20..
>
> U 192.168.2.14:5060 -> 192.168.2.190:5060
>   SIP/2.0 488 Not Acceptable Here..Via: SIP/2.0/UDP
> 192.168.2.190;rport;branch=z9hG4bK8mmBB5e8Z2mKr..From: <
> sip:9322273640 at 192.168.2.190>;tag=4QggZ1XcNZD3F..To: "7006" <
> sip:7006 at 192.168.2.14:5060>;tag=983B5A59-EB953B22..CSeq: 9403059
>   INVITE..Call-ID: 7bfae496-c324-122e-bc85-0026b976aef7..Contact: <
> sip:7006 at 192.168.2.14:5060>..User-Agent:
> PolycomSoundPointIP-SPIP_335-UA/3.2.2.0477..Accept-Language: en..Warning:
> 488 SDP "NotAcceptableHere"..Content-Length: 0....
>
>
>
> I tried all the possibilities with codec priorities on U/A .
>
> http://pastebin.freeswitch.org/15577
>
> Any help.
>
> Regards
> Sam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Wed Mar  9 09:00:35 2011
From: u2nsam at gmail.com (Sam)
Date: Wed, 9 Mar 2011 11:30:35 +0530
Subject: [Freeswitch-users] incompatible destination
In-Reply-To: <AANLkTi=LwtDEF6-RumWCw2dLThTjcZXwET9_O_OiMN96@mail.gmail.com>
References: <AANLkTincX-zpfnqx4dRk2jLCPf6HDiGN+cuPDA7+75WX@mail.gmail.com>
	<AANLkTi=LwtDEF6-RumWCw2dLThTjcZXwET9_O_OiMN96@mail.gmail.com>
Message-ID: <AANLkTimghyn8KeN_PETXtqkkKgHQvyNBktw1jAEnav6Y@mail.gmail.com>

Thanks,

Yes 192.168.2.190 is FS and I only have 1 polycom phone and rest are cisco
7960s .

I tried with g711alaw in third priority and its working, but i want to have
the codec preference on polycom like g722,g711ulaw,g711alaw.

I have 2 debugs when i removed G711alaw and just kept G722 ,G711,G729 .

Made a call from polycom to cisco the call disconnected after pick up.
http://pastebin.freeswitch.org/15621

Made an incoming call to polycom and it did not transcode.
http://pastebin.freeswitch.org/15622

I am having this settings in profile:
<param name="inbound-codec-negotiation" value="generous"/>
    <param name="disable-transcoding" value="true"/>
also tried :-
<param name="inbound-codec-negotiation" value="generous"/>
    <param name="disable-transcoding" value="false"/>

Am i correct with those , if not do correct me.

Regards
Sam




On Wed, Mar 9, 2011 at 10:36 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Sun, Mar 6, 2011 at 10:20 PM, Sam <u2nsam at gmail.com> wrote:
>
>> Hello,
>>
>>
>> The extension 7006 was working few days ago and now it not working,rest
>> all the phone are working properly.
>>
>
> Sam,
>
> Is 192.168.2.190 your FreeSWITCH box? If so then the issue is with the
> phone not accepting the PCMA codec. Go into your Polycom config and make
> sure that PCMA is enabled on the phone itself. The other option would be to
> let the Polycom speak in G722 and let FS do transcoding. In any case, if I
> were you I would find another working Polycom phone and mimic the configs on
> this non-working Poly 335.
>
> -MC
>
>
>>
>> U 192.168.2.190:5060 -> 192.168.2.14:5060
>>   INVITE sip:7006 at 192.168.2.14:5060 SIP/2.0..Via: SIP/2.0/UDP
>> 192.168.2.190;rport;branch=z9hG4bK8mmBB5e8Z2mKr..Max-Forwards: 70..From: ""
>> <sip:9322273640 at 192.168.2.190>;tag=4QggZ1XcNZD3F..To: <
>> sip:7006 at 192.168.2.14:5060>..Call-ID: 7bf
>>   ae496-c324-122e-bc85-0026b976aef7..CSeq: 9403059 INVITE..Contact: <
>> sip:mod_sofia at 192.168.2.190:5060>..User-Agent: NOVANET..Allow: INVITE,
>> ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY,
>> PUBLISH, SUBSCRIBE.
>>   .Supported: timer, precondition, path, replaces..Allow-Events: talk,
>> hold, presence, dialog, line-seize, call-info, sla,
>> include-session-description, presence.winfo, message-summary,
>> refer..Content-Type: application/sdp..Content-Dis
>>   position: session..Content-Length: 203..X-FS-Support:
>> update_display..Remote-Party-ID: <sip:9322273640 at 192.168.2.190>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH
>> 1299446773 1299446774 IN IP4 192.168.2.190..s=FreeSWITCH
>>   ..c=IN IP4 192.168.2.190..t=0 0..m=audio 31474 RTP/AVP 8 101
>> 13..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=ptime:20..
>>
>> U 192.168.2.14:5060 -> 192.168.2.190:5060
>>   SIP/2.0 488 Not Acceptable Here..Via: SIP/2.0/UDP
>> 192.168.2.190;rport;branch=z9hG4bK8mmBB5e8Z2mKr..From: <
>> sip:9322273640 at 192.168.2.190>;tag=4QggZ1XcNZD3F..To: "7006" <
>> sip:7006 at 192.168.2.14:5060>;tag=983B5A59-EB953B22..CSeq: 9403059
>>   INVITE..Call-ID: 7bfae496-c324-122e-bc85-0026b976aef7..Contact: <
>> sip:7006 at 192.168.2.14:5060>..User-Agent:
>> PolycomSoundPointIP-SPIP_335-UA/3.2.2.0477..Accept-Language: en..Warning:
>> 488 SDP "NotAcceptableHere"..Content-Length: 0....
>>
>>
>>
>> I tried all the possibilities with codec priorities on U/A .
>>
>> http://pastebin.freeswitch.org/15577
>>
>> Any help.
>>
>> Regards
>> Sam
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Wed Mar  9 12:43:45 2011
From: u2nsam at gmail.com (Sam)
Date: Wed, 9 Mar 2011 15:13:45 +0530
Subject: [Freeswitch-users] count
In-Reply-To: <AANLkTi=oKxBtho8q5kQq52ZxrU=r++SzFXkVzjq=P8B1@mail.gmail.com>
References: <AANLkTi=oKxBtho8q5kQq52ZxrU=r++SzFXkVzjq=P8B1@mail.gmail.com>
Message-ID: <AANLkTimWrn1HHbhQj5CK2Mhk64h-d5fGbGH4E-Mpj0i_@mail.gmail.com>

Hello,

What is the command to fetch the member id of the members in conference ?

Regards
Sam

On Thu, Mar 3, 2011 at 10:17 AM, Sam <u2nsam at gmail.com> wrote:

> Hello,
>
> Whats the command to get the participants count in the conference ?
>
> Rwgards
> Sam
>
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From steveayre at gmail.com  Wed Mar  9 12:46:34 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 9 Mar 2011 09:46:34 +0000
Subject: [Freeswitch-users] incoming call and callee busy
In-Reply-To: <CF955E05ED054AE3A8635355005DC7B1@e1705>
References: <CF955E05ED054AE3A8635355005DC7B1@e1705>
Message-ID: <AANLkTimXs-8ovLxRoPyzqiHoyJt7vLoPAr7rM=dGkb3_@mail.gmail.com>

Unless you're using mod_limit to detect if there is already a call to the
callee, it'll send a new INVITE to the callee. It's up to them what they
return. It *may* be 486 Busy Here, but they can also handle it in other
ways, for example with phones with multiple lines.

-Steve


On 8 March 2011 23:24, Madovsky <infos at madovsky.org> wrote:

>  What's the A leg SIP return status of the callee if
> the calle is already in early media status with another call ?
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gchen00 at insightbb.com  Wed Mar  9 16:27:23 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Wed, 09 Mar 2011 08:27:23 -0500
Subject: [Freeswitch-users] send and receive call for non registered
	endpoint.
Message-ID: <fc6fe08794f0.4d7739eb@insightbb.com>

I have a IAD with PRI interface and ?multiple ?SIP users. I do not want to register each user on Freeswitch.
I do still want each user to be able to send call and receive call.?
For sending call from IAD to FS, since all the sip calls have to go through SIP-router (Kamailio) and then routed to FS, I am thinking using X-AUTH-IP in ACL so no auth required.?
For ? receiveing call from FS, I am thinking either to setup each user on FS and ?manually create contact information and make them never expire or add cidr to each user and modify dial-string to use cidr instead of sip contact. Are these approaches workable? ?What is the best way to do this?

Gary ?
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From mitch.johnson7 at gmail.com  Wed Mar  9 16:30:59 2011
From: mitch.johnson7 at gmail.com (Mitch Johnson)
Date: Wed, 9 Mar 2011 08:30:59 -0500
Subject: [Freeswitch-users] tls/srtp goes straight to voicemail
In-Reply-To: <mailman.5071.1299647203.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.5071.1299647203.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <659E8C7B-B97A-4A9D-998B-435CE7664E2B@gmail.com>

It turns out that the Eyebeam app was the cause of the issue.  Using the Acrobits client showed no issues whatsoever.  This was discovered with a lot of help from another Mitch.

> From: Michael Collins <msc at freeswitch.org>
> Date: March 8, 2011 4:49:26 PM EST
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] tls/srtp goes straight to voicemail
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> Are you sure that the Eyebeam clients support SRTP? It looks like you're getting a 406:
> 
> 2011-03-05 23:21:08.650486 [NOTICE] sofia.c:5323 Hangup sofia/internal/sip:1005 at 192.168.60.135:53664 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
> 
> You probably need to look directly at the SIP traces and see if there is any more information as to what's going on.
> 
> -MC
> 
> 
> 
> 

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From max.clark at gmail.com  Wed Mar  9 18:52:04 2011
From: max.clark at gmail.com (Max Clark)
Date: Wed, 9 Mar 2011 07:52:04 -0800
Subject: [Freeswitch-users] To Proxy Media or Not
Message-ID: <AANLkTi=73tghHw5Usvgb82VuHUDj_=MXGQ=MvLHhqK3r@mail.gmail.com>

Hello,

I have a couple of FreeSWITCH boxes functioning as SBCs on our
network. We need them to:

- Route Calls
- Function as B2BUAs for call logging
- Hide topology from the endpoints

They do not need to:

- Intercept/play audio
- Manipulate the SDP
- Transcode
- Process anything other than SIP

So these machines have been running with Proxy Media enabled in the
dialplan. Is the the proper/recommended way of configuring for our
environment/needs? Is the minimal gain of resources by not processing
the RTP worth it?

Thanks,
Max


From max.clark at gmail.com  Wed Mar  9 18:53:57 2011
From: max.clark at gmail.com (Max Clark)
Date: Wed, 9 Mar 2011 07:53:57 -0800
Subject: [Freeswitch-users] Pre-Answer / Sending early media Command not
	found
Message-ID: <AANLkTikE69MyqsxST8DNo2tLkuwUF8OuTFm=O+jREZGC@mail.gmail.com>

I upgraded to current last night, since then I've noticed these errors
on the console:

-ERR 2011-03-09 15:49:15.629671 [NOTICE] sofia.c:4794 Pre-Answer
sofia/external/xxxx at y.y.y.y:5060! Command not found!

freeswitch at internal> 2011-03-09 15:49:15.630671 [INFO]
switch_ivr_originate.c:3365 Sending early media
-ERR 2011-03-09 15:49:15.630671 [INFO] switch_ivr_originate.c:3365
Sending early media Command not found!

Do I need to add/adjust configuration on my side?

Thanks,
Max


From steveayre at gmail.com  Wed Mar  9 19:04:25 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 9 Mar 2011 16:04:25 +0000
Subject: [Freeswitch-users] Pre-Answer / Sending early media Command not
	found
In-Reply-To: <AANLkTikE69MyqsxST8DNo2tLkuwUF8OuTFm=O+jREZGC@mail.gmail.com>
References: <AANLkTikE69MyqsxST8DNo2tLkuwUF8OuTFm=O+jREZGC@mail.gmail.com>
Message-ID: <AANLkTi=Xj7BpZwQr9gCC6LFryX1kExDsE0Mg+59Ra4dk@mail.gmail.com>

"freeswitch at internal> 2011-03-09 15:49:15.630671 [INFO]
switch_ivr_originate.c:3365 Sending early media
-ERR 2011-03-09 15:49:15.630671 [INFO] switch_ivr_originate.c:3365
Sending early media Command not found!"

You're pasting log output at the cli, which of course isn't a valid command.

"Command not found" is an error message that's only ever generated by
mod_event_socket, the command is the text between "-ERR" and "Command not
found"

-Steve


On 9 March 2011 15:53, Max Clark <max.clark at gmail.com> wrote:

> I upgraded to current last night, since then I've noticed these errors
> on the console:
>
> -ERR 2011-03-09 15:49:15.629671 [NOTICE] sofia.c:4794 Pre-Answer
> sofia/external/xxxx at y.y.y.y:5060! Command not found!
>
> freeswitch at internal> 2011-03-09 15:49:15.630671 [INFO]
> switch_ivr_originate.c:3365 Sending early media
> -ERR 2011-03-09 15:49:15.630671 [INFO] switch_ivr_originate.c:3365
> Sending early media Command not found!
>
> Do I need to add/adjust configuration on my side?
>
> Thanks,
> Max
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From max.clark at gmail.com  Wed Mar  9 19:07:21 2011
From: max.clark at gmail.com (Max Clark)
Date: Wed, 9 Mar 2011 08:07:21 -0800
Subject: [Freeswitch-users] sofia.c:5172 Codec Error! v=0
Message-ID: <AANLkTi=3uea+vjqs2A+hw9V_NdH2UYAXFOZ0KsFSjjBy@mail.gmail.com>

How do I identify the source of this error and fix it?

2011-03-09 16:02:42.845215 [ERR] sofia.c:5172 Codec Error! v=0
o=MxSIP 0 28211 IN IP4 x.x.x.x
s=SIP Call
c=IN IP4 x.x.x.x
t=0 0
m=audio 5238 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

I'm actually surprised that I'm even seeing this with proxy_media=true
enabled. The codec list is correct and what are supported by both
endpoints.

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/>

Thanks,
Max


From max.clark at gmail.com  Wed Mar  9 19:10:40 2011
From: max.clark at gmail.com (Max Clark)
Date: Wed, 9 Mar 2011 08:10:40 -0800
Subject: [Freeswitch-users] Pre-Answer / Sending early media Command not
	found
In-Reply-To: <AANLkTi=Xj7BpZwQr9gCC6LFryX1kExDsE0Mg+59Ra4dk@mail.gmail.com>
References: <AANLkTikE69MyqsxST8DNo2tLkuwUF8OuTFm=O+jREZGC@mail.gmail.com>
	<AANLkTi=Xj7BpZwQr9gCC6LFryX1kExDsE0Mg+59Ra4dk@mail.gmail.com>
Message-ID: <AANLkTikqm+PQexUninQZ0-5KSUh8qjk_ZSPuJb3ECHBX@mail.gmail.com>

Steve,

Okay strange - I don't see how I pasted back into the console without
realizing it, or how I had copied text to paste without realizing it.
It's too early in the morning to be working on this obviously.

Thanks,
Max

On Wed, Mar 9, 2011 at 8:04 AM, Steven Ayre <steveayre at gmail.com> wrote:
> "freeswitch at internal> 2011-03-09 15:49:15.630671 [INFO]
> switch_ivr_originate.c:3365 Sending early media
> -ERR 2011-03-09 15:49:15.630671 [INFO] switch_ivr_originate.c:3365
> Sending early media Command not found!"
>
> You're pasting log output at the cli, which of course isn't a valid command.
>
> "Command not found" is an error message that's only ever generated by
> mod_event_socket, the command is the text between "-ERR" and "Command not
> found"
>
> -Steve
>
>
> On 9 March 2011 15:53, Max Clark <max.clark at gmail.com> wrote:
>>
>> I upgraded to current last night, since then I've noticed these errors
>> on the console:
>>
>> -ERR 2011-03-09 15:49:15.629671 [NOTICE] sofia.c:4794 Pre-Answer
>> sofia/external/xxxx at y.y.y.y:5060! Command not found!
>>
>> freeswitch at internal> 2011-03-09 15:49:15.630671 [INFO]
>> switch_ivr_originate.c:3365 Sending early media
>> -ERR 2011-03-09 15:49:15.630671 [INFO] switch_ivr_originate.c:3365
>> Sending early media Command not found!
>>
>> Do I need to add/adjust configuration on my side?
>>
>> Thanks,
>> Max
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Wed Mar  9 19:12:11 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 9 Mar 2011 08:12:11 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today: Special
	Presentation
Message-ID: <AANLkTi=4H6QykVimWiUVj=FBJgNiFyQncL1NaR7EgSSX@mail.gmail.com>

Hello all!

Today we are having a special presentation by Jim Gettys and Dave Taht of
the Bufferbloat project. Please join us and learn more about this subject -
you'll be glad you did.

The official agenda page is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2011_03_09

Talk to you soon!

-Michael
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From max.clark at gmail.com  Wed Mar  9 19:19:05 2011
From: max.clark at gmail.com (Max Clark)
Date: Wed, 9 Mar 2011 08:19:05 -0800
Subject: [Freeswitch-users] How does mod_lcr handle invalid ANIs?
Message-ID: <AANLkTin3TaT=fV_mJhq0Hq4_X+GFQ4aDT46hSJdiFNfH@mail.gmail.com>

Hello,

I am calling the lcr module from the dialplan like this:

<action application="lcr" data="$2 default ${caller_id_number} intrastate"/>

So I can get interstate vs. intrastate routing (I dropped the
intralata column from the database). How does mod_lcr handle an
invalid ANI? Which rate is used to route for this condition?

Thanks,
Max


From kris at kriskinc.com  Wed Mar  9 20:02:14 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 9 Mar 2011 12:02:14 -0500
Subject: [Freeswitch-users] To Proxy Media or Not
In-Reply-To: <AANLkTi=73tghHw5Usvgb82VuHUDj_=MXGQ=MvLHhqK3r@mail.gmail.com>
References: <AANLkTi=73tghHw5Usvgb82VuHUDj_=MXGQ=MvLHhqK3r@mail.gmail.com>
Message-ID: <AANLkTikM_pXZ=jFDJt5i_bkR3-rFP1ZAZdLXrWMn7y5H@mail.gmail.com>

If your intention is to *strictly* hide topology proxy media will
rewrite the IP address in the SDP but that's about it.  Technically
that counts as "topology hiding".  Other SDP parameters, however, will
pass through and leak various bits of information about the endpoints
behind the SBC:

- Codec support
- Payload types
- Other "random" various SDP parameters

Most real SBCs will rewrite the entire SDP (as FS does by default).

Another thing to consider - the vast majority of FS users do not use
proxy media (and most that do use it incorrectly).  This leaves you
open to the possibility that you'll experience strange edge case bugs
using a media mode different from the majority of the rest of the FS
user base.

On Wed, Mar 9, 2011 at 10:52 AM, Max Clark <max.clark at gmail.com> wrote:
> Hello,
>
> I have a couple of FreeSWITCH boxes functioning as SBCs on our
> network. We need them to:
>
> - Route Calls
> - Function as B2BUAs for call logging
> - Hide topology from the endpoints
>
> They do not need to:
>
> - Intercept/play audio
> - Manipulate the SDP
> - Transcode
> - Process anything other than SIP
>
> So these machines have been running with Proxy Media enabled in the
> dialplan. Is the the proper/recommended way of configuring for our
> environment/needs? Is the minimal gain of resources by not processing
> the RTP worth it?
>
> Thanks,
> Max



-- 
Kristian Kielhofner


From avi at avimarcus.net  Wed Mar  9 20:09:14 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 9 Mar 2011 19:09:14 +0200
Subject: [Freeswitch-users] To Proxy Media or Not
In-Reply-To: <AANLkTikM_pXZ=jFDJt5i_bkR3-rFP1ZAZdLXrWMn7y5H@mail.gmail.com>
References: <AANLkTi=73tghHw5Usvgb82VuHUDj_=MXGQ=MvLHhqK3r@mail.gmail.com>
	<AANLkTikM_pXZ=jFDJt5i_bkR3-rFP1ZAZdLXrWMn7y5H@mail.gmail.com>
Message-ID: <AANLkTinLY4VMNd9ckWcU-QUWH4C=AK9fh9Rez0rqpkaq@mail.gmail.com>

Yes, you probably don't need "media proxy", just the "default" to hide the
remote IP.
See: http://wiki.freeswitch.org/wiki/Proxy_Media

<http://wiki.freeswitch.org/wiki/Proxy_Media>-Avi

On Wed, Mar 9, 2011 at 7:02 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> If your intention is to *strictly* hide topology proxy media will
> rewrite the IP address in the SDP but that's about it.  Technically
> that counts as "topology hiding".  Other SDP parameters, however, will
> pass through and leak various bits of information about the endpoints
> behind the SBC:
>
> - Codec support
> - Payload types
> - Other "random" various SDP parameters
>
> Most real SBCs will rewrite the entire SDP (as FS does by default).
>
> Another thing to consider - the vast majority of FS users do not use
> proxy media (and most that do use it incorrectly).  This leaves you
> open to the possibility that you'll experience strange edge case bugs
> using a media mode different from the majority of the rest of the FS
> user base.
>
> On Wed, Mar 9, 2011 at 10:52 AM, Max Clark <max.clark at gmail.com> wrote:
> > Hello,
> >
> > I have a couple of FreeSWITCH boxes functioning as SBCs on our
> > network. We need them to:
> >
> > - Route Calls
> > - Function as B2BUAs for call logging
> > - Hide topology from the endpoints
> >
> > They do not need to:
> >
> > - Intercept/play audio
> > - Manipulate the SDP
> > - Transcode
> > - Process anything other than SIP
> >
> > So these machines have been running with Proxy Media enabled in the
> > dialplan. Is the the proper/recommended way of configuring for our
> > environment/needs? Is the minimal gain of resources by not processing
> > the RTP worth it?
> >
> > Thanks,
> > Max
>
>
>
> --
> Kristian Kielhofner
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From max.clark at gmail.com  Wed Mar  9 20:13:02 2011
From: max.clark at gmail.com (Max Clark)
Date: Wed, 9 Mar 2011 09:13:02 -0800
Subject: [Freeswitch-users] To Proxy Media or Not
In-Reply-To: <AANLkTikM_pXZ=jFDJt5i_bkR3-rFP1ZAZdLXrWMn7y5H@mail.gmail.com>
References: <AANLkTi=73tghHw5Usvgb82VuHUDj_=MXGQ=MvLHhqK3r@mail.gmail.com>
	<AANLkTikM_pXZ=jFDJt5i_bkR3-rFP1ZAZdLXrWMn7y5H@mail.gmail.com>
Message-ID: <AANLkTimXRm9or4H2A6PKaeG+9zuO_wF36OsTga_9tKPF@mail.gmail.com>

Advantages of Proxy Media:

1) Support codecs not natively supported by FreeSWITCH
2) Lower CPU resources on FreeSWITCH

So for 1) I'm really only concerned with t38/fax over g711. With our
current configuration faxing (although unsupported) is working just
fine. What should I expect if I revert to media mode?

2) If I disable transcoding, what kind of difference in resource
utilization should I expect? Nominal (less than 10%), Moderate
(11-30%), or more?

Thanks,
Max

On Wed, Mar 9, 2011 at 9:02 AM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> If your intention is to *strictly* hide topology proxy media will
> rewrite the IP address in the SDP but that's about it. ?Technically
> that counts as "topology hiding". ?Other SDP parameters, however, will
> pass through and leak various bits of information about the endpoints
> behind the SBC:
>
> - Codec support
> - Payload types
> - Other "random" various SDP parameters
>
> Most real SBCs will rewrite the entire SDP (as FS does by default).
>
> Another thing to consider - the vast majority of FS users do not use
> proxy media (and most that do use it incorrectly). ?This leaves you
> open to the possibility that you'll experience strange edge case bugs
> using a media mode different from the majority of the rest of the FS
> user base.
>
> On Wed, Mar 9, 2011 at 10:52 AM, Max Clark <max.clark at gmail.com> wrote:
>> Hello,
>>
>> I have a couple of FreeSWITCH boxes functioning as SBCs on our
>> network. We need them to:
>>
>> - Route Calls
>> - Function as B2BUAs for call logging
>> - Hide topology from the endpoints
>>
>> They do not need to:
>>
>> - Intercept/play audio
>> - Manipulate the SDP
>> - Transcode
>> - Process anything other than SIP
>>
>> So these machines have been running with Proxy Media enabled in the
>> dialplan. Is the the proper/recommended way of configuring for our
>> environment/needs? Is the minimal gain of resources by not processing
>> the RTP worth it?
>>
>> Thanks,
>> Max
>
>
>
> --
> Kristian Kielhofner
>


From steveayre at gmail.com  Wed Mar  9 20:12:42 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 9 Mar 2011 17:12:42 +0000
Subject: [Freeswitch-users] Pre-Answer / Sending early media Command not
	found
In-Reply-To: <AANLkTikqm+PQexUninQZ0-5KSUh8qjk_ZSPuJb3ECHBX@mail.gmail.com>
References: <AANLkTikE69MyqsxST8DNo2tLkuwUF8OuTFm=O+jREZGC@mail.gmail.com>
	<AANLkTi=Xj7BpZwQr9gCC6LFryX1kExDsE0Mg+59Ra4dk@mail.gmail.com>
	<AANLkTikqm+PQexUninQZ0-5KSUh8qjk_ZSPuJb3ECHBX@mail.gmail.com>
Message-ID: <AANLkTim5NgaGn1LrG7A=MdbsvNTiGO7+LEUqxJGWhfBK@mail.gmail.com>

Some terminals let you copy and paste by selecting text and right clicking,
it could be easy to accidentally do so on those. Just a guess. :)

-Steve


On 9 March 2011 16:10, Max Clark <max.clark at gmail.com> wrote:

> Steve,
>
> Okay strange - I don't see how I pasted back into the console without
> realizing it, or how I had copied text to paste without realizing it.
> It's too early in the morning to be working on this obviously.
>
> Thanks,
> Max
>
> On Wed, Mar 9, 2011 at 8:04 AM, Steven Ayre <steveayre at gmail.com> wrote:
> > "freeswitch at internal> 2011-03-09 15:49:15.630671 [INFO]
> > switch_ivr_originate.c:3365 Sending early media
> > -ERR 2011-03-09 15:49:15.630671 [INFO] switch_ivr_originate.c:3365
> > Sending early media Command not found!"
> >
> > You're pasting log output at the cli, which of course isn't a valid
> command.
> >
> > "Command not found" is an error message that's only ever generated by
> > mod_event_socket, the command is the text between "-ERR" and "Command not
> > found"
> >
> > -Steve
> >
> >
> > On 9 March 2011 15:53, Max Clark <max.clark at gmail.com> wrote:
> >>
> >> I upgraded to current last night, since then I've noticed these errors
> >> on the console:
> >>
> >> -ERR 2011-03-09 15:49:15.629671 [NOTICE] sofia.c:4794 Pre-Answer
> >> sofia/external/xxxx at y.y.y.y:5060! Command not found!
> >>
> >> freeswitch at internal> 2011-03-09 15:49:15.630671 [INFO]
> >> switch_ivr_originate.c:3365 Sending early media
> >> -ERR 2011-03-09 15:49:15.630671 [INFO] switch_ivr_originate.c:3365
> >> Sending early media Command not found!
> >>
> >> Do I need to add/adjust configuration on my side?
> >>
> >> Thanks,
> >> Max
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Wed Mar  9 20:19:25 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 9 Mar 2011 17:19:25 +0000
Subject: [Freeswitch-users] sofia.c:5172 Codec Error! v=0
In-Reply-To: <AANLkTi=3uea+vjqs2A+hw9V_NdH2UYAXFOZ0KsFSjjBy@mail.gmail.com>
References: <AANLkTi=3uea+vjqs2A+hw9V_NdH2UYAXFOZ0KsFSjjBy@mail.gmail.com>
Message-ID: <AANLkTi=0TLtzFTsz8Tjy-7iFsDo9NYaQ54WA+Ha+DM=t@mail.gmail.com>

What are the codec preferences of the SIP profile?

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/>
That just sets a variable that's can be used in the SIP profile
configuration, it's not necessarily what's being used.

You can check with "sofia status profile <name>", the CODECS IN and CODECS
OUT lines.

Can you also show us the SDP from both legs?

-Steve



On 9 March 2011 16:07, Max Clark <max.clark at gmail.com> wrote:

> How do I identify the source of this error and fix it?
>
> 2011-03-09 16:02:42.845215 [ERR] sofia.c:5172 Codec Error! v=0
> o=MxSIP 0 28211 IN IP4 x.x.x.x
> s=SIP Call
> c=IN IP4 x.x.x.x
> t=0 0
> m=audio 5238 RTP/AVP 0 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
> I'm actually surprised that I'm even seeing this with proxy_media=true
> enabled. The codec list is correct and what are supported by both
> endpoints.
>
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/>
>
> Thanks,
> Max
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Wed Mar  9 20:20:59 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 9 Mar 2011 17:20:59 +0000
Subject: [Freeswitch-users] To Proxy Media or Not
In-Reply-To: <AANLkTimXRm9or4H2A6PKaeG+9zuO_wF36OsTga_9tKPF@mail.gmail.com>
References: <AANLkTi=73tghHw5Usvgb82VuHUDj_=MXGQ=MvLHhqK3r@mail.gmail.com>
	<AANLkTikM_pXZ=jFDJt5i_bkR3-rFP1ZAZdLXrWMn7y5H@mail.gmail.com>
	<AANLkTimXRm9or4H2A6PKaeG+9zuO_wF36OsTga_9tKPF@mail.gmail.com>
Message-ID: <AANLkTingEN7Kusm+ryoObxzAHzHpDJoJC5=7Ez2PnAUc@mail.gmail.com>

"Nominal (less than 10%), Moderate (11-30%), or more?"

I think it's far less than even that 10%.

-Steve


On 9 March 2011 17:13, Max Clark <max.clark at gmail.com> wrote:

> Advantages of Proxy Media:
>
> 1) Support codecs not natively supported by FreeSWITCH
> 2) Lower CPU resources on FreeSWITCH
>
> So for 1) I'm really only concerned with t38/fax over g711. With our
> current configuration faxing (although unsupported) is working just
> fine. What should I expect if I revert to media mode?
>
> 2) If I disable transcoding, what kind of difference in resource
> utilization should I expect? Nominal (less than 10%), Moderate
> (11-30%), or more?
>
> Thanks,
> Max
>
> On Wed, Mar 9, 2011 at 9:02 AM, Kristian Kielhofner <kris at kriskinc.com>
> wrote:
> > If your intention is to *strictly* hide topology proxy media will
> > rewrite the IP address in the SDP but that's about it.  Technically
> > that counts as "topology hiding".  Other SDP parameters, however, will
> > pass through and leak various bits of information about the endpoints
> > behind the SBC:
> >
> > - Codec support
> > - Payload types
> > - Other "random" various SDP parameters
> >
> > Most real SBCs will rewrite the entire SDP (as FS does by default).
> >
> > Another thing to consider - the vast majority of FS users do not use
> > proxy media (and most that do use it incorrectly).  This leaves you
> > open to the possibility that you'll experience strange edge case bugs
> > using a media mode different from the majority of the rest of the FS
> > user base.
> >
> > On Wed, Mar 9, 2011 at 10:52 AM, Max Clark <max.clark at gmail.com> wrote:
> >> Hello,
> >>
> >> I have a couple of FreeSWITCH boxes functioning as SBCs on our
> >> network. We need them to:
> >>
> >> - Route Calls
> >> - Function as B2BUAs for call logging
> >> - Hide topology from the endpoints
> >>
> >> They do not need to:
> >>
> >> - Intercept/play audio
> >> - Manipulate the SDP
> >> - Transcode
> >> - Process anything other than SIP
> >>
> >> So these machines have been running with Proxy Media enabled in the
> >> dialplan. Is the the proper/recommended way of configuring for our
> >> environment/needs? Is the minimal gain of resources by not processing
> >> the RTP worth it?
> >>
> >> Thanks,
> >> Max
> >
> >
> >
> > --
> > Kristian Kielhofner
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Wed Mar  9 20:21:44 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 9 Mar 2011 17:21:44 +0000
Subject: [Freeswitch-users] sofia.c:5172 Codec Error! v=0
In-Reply-To: <AANLkTi=0TLtzFTsz8Tjy-7iFsDo9NYaQ54WA+Ha+DM=t@mail.gmail.com>
References: <AANLkTi=3uea+vjqs2A+hw9V_NdH2UYAXFOZ0KsFSjjBy@mail.gmail.com>
	<AANLkTi=0TLtzFTsz8Tjy-7iFsDo9NYaQ54WA+Ha+DM=t@mail.gmail.com>
Message-ID: <AANLkTinMHtgb9O1tiPqAwqTt96cc7BR=BeG3gc0tfnnP@mail.gmail.com>

Also, does this happen only proxy_media mode, or in both? Just to eliminate
the possibility of it being a bug in the proxy media code?

-Steve


On 9 March 2011 17:19, Steven Ayre <steveayre at gmail.com> wrote:

> What are the codec preferences of the SIP profile?
>
>
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/>
> That just sets a variable that's can be used in the SIP profile
> configuration, it's not necessarily what's being used.
>
> You can check with "sofia status profile <name>", the CODECS IN and CODECS
> OUT lines.
>
> Can you also show us the SDP from both legs?
>
> -Steve
>
>
>
>
> On 9 March 2011 16:07, Max Clark <max.clark at gmail.com> wrote:
>
>> How do I identify the source of this error and fix it?
>>
>> 2011-03-09 16:02:42.845215 [ERR] sofia.c:5172 Codec Error! v=0
>> o=MxSIP 0 28211 IN IP4 x.x.x.x
>> s=SIP Call
>> c=IN IP4 x.x.x.x
>> t=0 0
>> m=audio 5238 RTP/AVP 0 18 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:18 G729/8000
>> a=fmtp:18 annexb=no
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>>
>> I'm actually surprised that I'm even seeing this with proxy_media=true
>> enabled. The codec list is correct and what are supported by both
>> endpoints.
>>
>> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/>
>>
>> Thanks,
>> Max
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From mh at markholloway.com  Wed Mar  9 20:36:12 2011
From: mh at markholloway.com (Mark Holloway)
Date: Wed, 9 Mar 2011 10:36:12 -0700
Subject: [Freeswitch-users] SIP Trunk Registration
Message-ID: <6CF1CE22-FF54-46FC-A159-202CA04542BE@markholloway.com>

Does FreeSwitch support single phone number registration for a Trunk Group?  For example, in Broadsoft Broadworks I can build a Trunk Group with 100 numbers but I identify one number as the Pilot Number.  I have a SIP endpoint (such as a Cisco ISR or Adtran TA900) that registers with Broadworks using the main Pilot Number only (Business Contact) so I don't have to register all 100 numbers.  It's like a parent-child relationship where the other 99 numbers are a child of the main parent (pilot) number. 

Thanks,
Mark




From Nabble at slickdeals.endjunk.com  Wed Mar  9 20:48:32 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 9 Mar 2011 09:48:32 -0800 (PST)
Subject: [Freeswitch-users] To Proxy Media or Not
In-Reply-To: <AANLkTinLY4VMNd9ckWcU-QUWH4C=AK9fh9Rez0rqpkaq@mail.gmail.com>
References: <AANLkTi=73tghHw5Usvgb82VuHUDj_=MXGQ=MvLHhqK3r@mail.gmail.com>
	<AANLkTikM_pXZ=jFDJt5i_bkR3-rFP1ZAZdLXrWMn7y5H@mail.gmail.com>
	<AANLkTinLY4VMNd9ckWcU-QUWH4C=AK9fh9Rez0rqpkaq@mail.gmail.com>
Message-ID: <1299692912185-6154695.post@n2.nabble.com>


Avi Marcus-2 wrote:
> Yes, you probably don't need "media proxy", just the "default" to hide the
> remote IP.
> See: http://wiki.freeswitch.org/wiki/Proxy_Media
I took a look at the above link in the section on 
http://wiki.freeswitch.org/wiki/Proxy_Media#How_to_enable_it How to enable
it  which states to set proxy_media=true before the bridge. When I take a
look at the conf/dialplan/public/00_inbound_did.xml file, there is no bridge
except transfer as the application. So, will proxy media work on transfer?

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/To-Proxy-Media-or-Not-tp6154192p6154695.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Wed Mar  9 21:56:07 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 9 Mar 2011 18:56:07 +0000
Subject: [Freeswitch-users] To Proxy Media or Not
In-Reply-To: <1299692912185-6154695.post@n2.nabble.com>
References: <AANLkTi=73tghHw5Usvgb82VuHUDj_=MXGQ=MvLHhqK3r@mail.gmail.com>
	<AANLkTikM_pXZ=jFDJt5i_bkR3-rFP1ZAZdLXrWMn7y5H@mail.gmail.com>
	<AANLkTinLY4VMNd9ckWcU-QUWH4C=AK9fh9Rez0rqpkaq@mail.gmail.com>
	<1299692912185-6154695.post@n2.nabble.com>
Message-ID: <AANLkTikX_bf+4U6kaPp8JOokDSDH5+E5TXJiiVFOjmBx@mail.gmail.com>

The transfer application doesn't do a bridge and doesn't create any other
legs, it just moves the call processing to another point in the dialplan.
Think of it as a GoTo (albeit one that can change the destination_number).
As such proxy_media doesn't apply.

That set, once channel variables are set the variable remains set, even
after a transfer. So you can set proxy_media=true, transfer to another
extension, bridge in that extension and proxy media will be enabled for that
bridge because proxy_media is still true after the transfer.

-Steve


On 9 March 2011 17:48, mazilo <Nabble at slickdeals.endjunk.com> wrote:

>
> Avi Marcus-2 wrote:
> > Yes, you probably don't need "media proxy", just the "default" to hide
> the
> > remote IP.
> > See: http://wiki.freeswitch.org/wiki/Proxy_Media
> I took a look at the above link in the section on
> http://wiki.freeswitch.org/wiki/Proxy_Media#How_to_enable_it How to enable
> it  which states to set proxy_media=true before the bridge. When I take a
> look at the conf/dialplan/public/00_inbound_did.xml file, there is no
> bridge
> except transfer as the application. So, will proxy media work on transfer?
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/To-Proxy-Media-or-Not-tp6154192p6154695.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kris at kriskinc.com  Wed Mar  9 22:07:17 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 9 Mar 2011 14:07:17 -0500
Subject: [Freeswitch-users] To Proxy Media or Not
In-Reply-To: <AANLkTimXRm9or4H2A6PKaeG+9zuO_wF36OsTga_9tKPF@mail.gmail.com>
References: <AANLkTi=73tghHw5Usvgb82VuHUDj_=MXGQ=MvLHhqK3r@mail.gmail.com>
	<AANLkTikM_pXZ=jFDJt5i_bkR3-rFP1ZAZdLXrWMn7y5H@mail.gmail.com>
	<AANLkTimXRm9or4H2A6PKaeG+9zuO_wF36OsTga_9tKPF@mail.gmail.com>
Message-ID: <AANLkTi=6Uosi+OzVKanOiWMTQnn1xxHbdZSZkU6cBN4V@mail.gmail.com>

Fax over g711 (if it's going to work at all) should not be affected by
proxy media or default media handling as long as you allow PCMU/PCMA
when using the default media handling.  However, T38 and fax over 711
are completely different.  Using T38 through FreeSWITCH with the
default media handling will require some extra configuration (nothing
drastic).

No one said anything about transcoding...  Proxy media mode does not
have the ability to transcode at all as FreeSWITCH is completely
unaware what codec you are using (internally it actually reports
"proxy" as the codec).

Just because you're using default media mode DOES NOT mean you will be
transcoding.  Using some Sofia params and late-negotiation dialplan
logic you can have an incredible amount of control over what codecs
your end points are using.  As long as they agree on at least one
codec and codec parameters you are the one who decides which codecs
take precedence and whether or not you'll be transcoding.

On Wed, Mar 9, 2011 at 12:13 PM, Max Clark <max.clark at gmail.com> wrote:
> Advantages of Proxy Media:
>
> 1) Support codecs not natively supported by FreeSWITCH
> 2) Lower CPU resources on FreeSWITCH
>
> So for 1) I'm really only concerned with t38/fax over g711. With our
> current configuration faxing (although unsupported) is working just
> fine. What should I expect if I revert to media mode?
>
> 2) If I disable transcoding, what kind of difference in resource
> utilization should I expect? Nominal (less than 10%), Moderate
> (11-30%), or more?
>
> Thanks,
> Max
>

-- 
Kristian Kielhofner


From max.clark at gmail.com  Wed Mar  9 22:21:17 2011
From: max.clark at gmail.com (Max Clark)
Date: Wed, 9 Mar 2011 11:21:17 -0800
Subject: [Freeswitch-users] To Proxy Media or Not
In-Reply-To: <AANLkTi=6Uosi+OzVKanOiWMTQnn1xxHbdZSZkU6cBN4V@mail.gmail.com>
References: <AANLkTi=73tghHw5Usvgb82VuHUDj_=MXGQ=MvLHhqK3r@mail.gmail.com>
	<AANLkTikM_pXZ=jFDJt5i_bkR3-rFP1ZAZdLXrWMn7y5H@mail.gmail.com>
	<AANLkTimXRm9or4H2A6PKaeG+9zuO_wF36OsTga_9tKPF@mail.gmail.com>
	<AANLkTi=6Uosi+OzVKanOiWMTQnn1xxHbdZSZkU6cBN4V@mail.gmail.com>
Message-ID: <AANLkTi=i3_iRvHVMogtLdfcpN8GFx3ZHAYu4R3OcGYG_@mail.gmail.com>

So this is an interesting side effect of removing proxy_media=true
from the dialplan:

2011-03-09 19:17:31.055236 [ERR] mod_g729.c:145 This codec is only
usable in passthrough mode!
2011-03-09 19:17:31.055236 [ERR] switch_core_io.c:897 Codec G.729 decoder error!

Am I reading this correctly that FreeSWITCH attempted to transcode a
G.729 audio stream?

On Wed, Mar 9, 2011 at 11:07 AM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> Fax over g711 (if it's going to work at all) should not be affected by
> proxy media or default media handling as long as you allow PCMU/PCMA
> when using the default media handling. ?However, T38 and fax over 711
> are completely different. ?Using T38 through FreeSWITCH with the
> default media handling will require some extra configuration (nothing
> drastic).
>
> No one said anything about transcoding... ?Proxy media mode does not
> have the ability to transcode at all as FreeSWITCH is completely
> unaware what codec you are using (internally it actually reports
> "proxy" as the codec).
>
> Just because you're using default media mode DOES NOT mean you will be
> transcoding. ?Using some Sofia params and late-negotiation dialplan
> logic you can have an incredible amount of control over what codecs
> your end points are using. ?As long as they agree on at least one
> codec and codec parameters you are the one who decides which codecs
> take precedence and whether or not you'll be transcoding.
>
> On Wed, Mar 9, 2011 at 12:13 PM, Max Clark <max.clark at gmail.com> wrote:
>> Advantages of Proxy Media:
>>
>> 1) Support codecs not natively supported by FreeSWITCH
>> 2) Lower CPU resources on FreeSWITCH
>>
>> So for 1) I'm really only concerned with t38/fax over g711. With our
>> current configuration faxing (although unsupported) is working just
>> fine. What should I expect if I revert to media mode?
>>
>> 2) If I disable transcoding, what kind of difference in resource
>> utilization should I expect? Nominal (less than 10%), Moderate
>> (11-30%), or more?
>>
>> Thanks,
>> Max
>>
>
> --
> Kristian Kielhofner
>


From freeswitch at tlainvestments.com  Wed Mar  9 23:06:55 2011
From: freeswitch at tlainvestments.com (Troy Anderson)
Date: Wed, 9 Mar 2011 13:06:55 -0700
Subject: [Freeswitch-users] Duplicate UUIDs in voicemail_msgs
Message-ID: <560287E1-3FB4-4B58-8221-46A7231BB233@tlainvestments.com>

Hi All,

I am regularly getting multiple copies of voicemails when checking voicemail on one of the extensions of a system, and, trying to diagnose the situation, seeing duplicate UUIDs in voicemail_msgs database.  I suspect that's why I get multiple "copies" of the message.  I'm trying to figure out what would cause multiple entries in the database for the same message.  I checked the version of mod_voicemail.c I'm using on this machine (from Nov 18, 2010) against the latest version and none of the few changes seem remotely related.  I would love to update to the latest and see if the issue is resolved, but this is on a client's production system, so that's not possible ATM.

Any ideas on what may be causing this, or how to diagnose further?


Thanks!
Troy

From mario_fs at mgtech.com  Wed Mar  9 23:16:23 2011
From: mario_fs at mgtech.com (Mario G)
Date: Wed, 9 Mar 2011 12:16:23 -0800
Subject: [Freeswitch-users] Freeswitch / Zywall USG100 problems
Message-ID: <E04386CF-006E-4F87-A2B0-FA6C810A34B7@mgtech.com>

Just wondering if anyone here can help. I replaced a Linksys/Cisco RV042 router with a Zywall USG100. It solved a lot but broke Freeswitch badly. Yes I ran traces, etc. Both routers are set to use a static wan and no natting, ext-ip-sip/rtp are the wan static address.  Just checking of anyone knows of a solution for one of these:

1. If I turn off ALG I cannot make a call  or receive calls, the traces shows time outs. No firewall issues, nothing in any log to show the issue,

2. If I do have ALG on I can make calls but the HOLD button no longer works. Phone (Linksys SPA962) says "Trying to Hold" then times out.

3. With ALG on (so I can make calls) when I call out the call drops in the middle of a conversation between 30 seconds and 5 minutes.

Been battling this and hope someone has already solved one of these. Thanks!
Mario

From mario_fs at mgtech.com  Wed Mar  9 23:20:08 2011
From: mario_fs at mgtech.com (Mario G)
Date: Wed, 9 Mar 2011 12:20:08 -0800
Subject: [Freeswitch-users] Freeswitch / Zywall USG100 problems
In-Reply-To: <E04386CF-006E-4F87-A2B0-FA6C810A34B7@mgtech.com>
References: <E04386CF-006E-4F87-A2B0-FA6C810A34B7@mgtech.com>
Message-ID: <022B01F7-0577-457B-94C8-D6F47B14E1F0@mgtech.com>

The router is using natting, I meant freeswitch uses -nonat

On Mar 9, 2011, at 12:16 PM, Mario G wrote:

> Just wondering if anyone here can help. I replaced a Linksys/Cisco RV042 router with a Zywall USG100. It solved a lot but broke Freeswitch badly. Yes I ran traces, etc. Both routers are set to use a static wan and no natting, ext-ip-sip/rtp are the wan static address.  Just checking of anyone knows of a solution for one of these:
> 
> 1. If I turn off ALG I cannot make a call  or receive calls, the traces shows time outs. No firewall issues, nothing in any log to show the issue,
> 
> 2. If I do have ALG on I can make calls but the HOLD button no longer works. Phone (Linksys SPA962) says "Trying to Hold" then times out.
> 
> 3. With ALG on (so I can make calls) when I call out the call drops in the middle of a conversation between 30 seconds and 5 minutes.
> 
> Been battling this and hope someone has already solved one of these. Thanks!
> Mario
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From fernando.berretta at gmail.com  Thu Mar 10 00:38:12 2011
From: fernando.berretta at gmail.com (Fernando Berretta)
Date: Wed, 09 Mar 2011 18:38:12 -0300
Subject: [Freeswitch-users] FreeSwitch for Windows
In-Reply-To: <022B01F7-0577-457B-94C8-D6F47B14E1F0@mgtech.com>
References: <E04386CF-006E-4F87-A2B0-FA6C810A34B7@mgtech.com>
	<022B01F7-0577-457B-94C8-D6F47B14E1F0@mgtech.com>
Message-ID: <4D77F344.3050207@gmail.com>

Hi,

I'm newbie in FreeSwitch and would like to know if freeswitch could be 
used for production under Windows platforms ? telephony boards like 
Digium / OpenVox, Sangoma, etc are going to work ok under Windows ? is 
there any problem / limitation ? E1 MFCR2 Trunk works ?

Thanks for your time,
Best Regards,
Fernando


From kris at kriskinc.com  Thu Mar 10 00:40:03 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 9 Mar 2011 16:40:03 -0500
Subject: [Freeswitch-users] To Proxy Media or Not
In-Reply-To: <AANLkTi=i3_iRvHVMogtLdfcpN8GFx3ZHAYu4R3OcGYG_@mail.gmail.com>
References: <AANLkTi=73tghHw5Usvgb82VuHUDj_=MXGQ=MvLHhqK3r@mail.gmail.com>
	<AANLkTikM_pXZ=jFDJt5i_bkR3-rFP1ZAZdLXrWMn7y5H@mail.gmail.com>
	<AANLkTimXRm9or4H2A6PKaeG+9zuO_wF36OsTga_9tKPF@mail.gmail.com>
	<AANLkTi=6Uosi+OzVKanOiWMTQnn1xxHbdZSZkU6cBN4V@mail.gmail.com>
	<AANLkTi=i3_iRvHVMogtLdfcpN8GFx3ZHAYu4R3OcGYG_@mail.gmail.com>
Message-ID: <AANLkTi=8OCMHo_jH7rhYxGj8Ki2_6Mn_ENbfNcdMvDTL@mail.gmail.com>

It appears so.  What are your codec settings on the profile(s)?  Do
you have late negotiation enabled?  If so, what does your dialplan
look like?  What are the SDP offers of each of your endpoints?

On Wed, Mar 9, 2011 at 2:21 PM, Max Clark <max.clark at gmail.com> wrote:
> So this is an interesting side effect of removing proxy_media=true
> from the dialplan:
>
> 2011-03-09 19:17:31.055236 [ERR] mod_g729.c:145 This codec is only
> usable in passthrough mode!
> 2011-03-09 19:17:31.055236 [ERR] switch_core_io.c:897 Codec G.729 decoder error!
>
> Am I reading this correctly that FreeSWITCH attempted to transcode a
> G.729 audio stream?
>

-- 
Kristian Kielhofner


From steveayre at gmail.com  Thu Mar 10 00:59:36 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 9 Mar 2011 21:59:36 +0000
Subject: [Freeswitch-users] To Proxy Media or Not
In-Reply-To: <AANLkTi=i3_iRvHVMogtLdfcpN8GFx3ZHAYu4R3OcGYG_@mail.gmail.com>
References: <AANLkTi=73tghHw5Usvgb82VuHUDj_=MXGQ=MvLHhqK3r@mail.gmail.com>
	<AANLkTikM_pXZ=jFDJt5i_bkR3-rFP1ZAZdLXrWMn7y5H@mail.gmail.com>
	<AANLkTimXRm9or4H2A6PKaeG+9zuO_wF36OsTga_9tKPF@mail.gmail.com>
	<AANLkTi=6Uosi+OzVKanOiWMTQnn1xxHbdZSZkU6cBN4V@mail.gmail.com>
	<AANLkTi=i3_iRvHVMogtLdfcpN8GFx3ZHAYu4R3OcGYG_@mail.gmail.com>
Message-ID: <AANLkTimNsb7vy8sAQu2eDBdNN1SwsMaTOArUGLkq-GuW@mail.gmail.com>

Yes.

You said you have PCMU and G729 enabled? Is that both inbound and outbound?

If so, it means that either your caller negotiated PCMU with FS, then FS
offers PCMU+G729 to the callee and they pick G729, or visa versa.

That's because FS will offer all the enabled codecs on the bleg. That leaves
the choice up to the callee on what they use and it might not be the same as
the aleg. That's only a problem with passthrough codecs such as G729, or if
you want to avoid transcoding because of CPU.

The way around it is to set either disable-transcoding or late-negotiation
parameters on the SIP profile.
disable-transcoding will only offer on the bleg the codec which has already
been negotiated for the aleg.
late-negotiation approaches it from the other end - it doesn't pick an aleg
codec straight away but instead offers all the choices to the bleg and then
uses for the aleg what the bleg selected.

-Steve



On 9 March 2011 19:21, Max Clark <max.clark at gmail.com> wrote:

> So this is an interesting side effect of removing proxy_media=true
> from the dialplan:
>
> 2011-03-09 19:17:31.055236 [ERR] mod_g729.c:145 This codec is only
> usable in passthrough mode!
> 2011-03-09 19:17:31.055236 [ERR] switch_core_io.c:897 Codec G.729 decoder
> error!
>
> Am I reading this correctly that FreeSWITCH attempted to transcode a
> G.729 audio stream?
>
> On Wed, Mar 9, 2011 at 11:07 AM, Kristian Kielhofner <kris at kriskinc.com>
> wrote:
> > Fax over g711 (if it's going to work at all) should not be affected by
> > proxy media or default media handling as long as you allow PCMU/PCMA
> > when using the default media handling.  However, T38 and fax over 711
> > are completely different.  Using T38 through FreeSWITCH with the
> > default media handling will require some extra configuration (nothing
> > drastic).
> >
> > No one said anything about transcoding...  Proxy media mode does not
> > have the ability to transcode at all as FreeSWITCH is completely
> > unaware what codec you are using (internally it actually reports
> > "proxy" as the codec).
> >
> > Just because you're using default media mode DOES NOT mean you will be
> > transcoding.  Using some Sofia params and late-negotiation dialplan
> > logic you can have an incredible amount of control over what codecs
> > your end points are using.  As long as they agree on at least one
> > codec and codec parameters you are the one who decides which codecs
> > take precedence and whether or not you'll be transcoding.
> >
> > On Wed, Mar 9, 2011 at 12:13 PM, Max Clark <max.clark at gmail.com> wrote:
> >> Advantages of Proxy Media:
> >>
> >> 1) Support codecs not natively supported by FreeSWITCH
> >> 2) Lower CPU resources on FreeSWITCH
> >>
> >> So for 1) I'm really only concerned with t38/fax over g711. With our
> >> current configuration faxing (although unsupported) is working just
> >> fine. What should I expect if I revert to media mode?
> >>
> >> 2) If I disable transcoding, what kind of difference in resource
> >> utilization should I expect? Nominal (less than 10%), Moderate
> >> (11-30%), or more?
> >>
> >> Thanks,
> >> Max
> >>
> >
> > --
> > Kristian Kielhofner
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mario_fs at mgtech.com  Thu Mar 10 01:12:28 2011
From: mario_fs at mgtech.com (Mario G)
Date: Wed, 9 Mar 2011 14:12:28 -0800
Subject: [Freeswitch-users] Freeswitch / Zywall USG100 problems
In-Reply-To: <022B01F7-0577-457B-94C8-D6F47B14E1F0@mgtech.com>
References: <E04386CF-006E-4F87-A2B0-FA6C810A34B7@mgtech.com>
	<022B01F7-0577-457B-94C8-D6F47B14E1F0@mgtech.com>
Message-ID: <7560DFC1-76DD-466C-9D50-33C3A521B6D5@mgtech.com>

Was this thread hijacked?
On Mar 9, 2011, at 12:20 PM, Mario G wrote:

> The router is using natting, I meant freeswitch uses -nonat
> 
> On Mar 9, 2011, at 12:16 PM, Mario G wrote:
> 
>> Just wondering if anyone here can help. I replaced a Linksys/Cisco RV042 router with a Zywall USG100. It solved a lot but broke Freeswitch badly. Yes I ran traces, etc. Both routers are set to use a static wan and no natting, ext-ip-sip/rtp are the wan static address.  Just checking of anyone knows of a solution for one of these:
>> 
>> 1. If I turn off ALG I cannot make a call  or receive calls, the traces shows time outs. No firewall issues, nothing in any log to show the issue,
>> 
>> 2. If I do have ALG on I can make calls but the HOLD button no longer works. Phone (Linksys SPA962) says "Trying to Hold" then times out.
>> 
>> 3. With ALG on (so I can make calls) when I call out the call drops in the middle of a conversation between 30 seconds and 5 minutes.
>> 
>> Been battling this and hope someone has already solved one of these. Thanks!
>> Mario
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Thu Mar 10 01:24:50 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 9 Mar 2011 14:24:50 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Followup - Recordings
	Online
Message-ID: <AANLkTi=RE=XtJwORA1AeEapqUBFMz8-E-O4=itWg8OaC@mail.gmail.com>

FYI,

We had a nice conference call with Jim Gettys and Dave T?ht from the
Bufferbloat project. I've made the recordings available in their usual spot:

http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls

Use the links next to "FS_weekly_2011_03_09" to download the audio format of
your choice.

Thanks to Jim, Dave, and all those who participated today!

-MC
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From mattzerah+freeswitch at gmail.com  Wed Mar  9 08:27:26 2011
From: mattzerah+freeswitch at gmail.com (Matt Paine)
Date: Wed, 9 Mar 2011 15:27:26 +1000
Subject: [Freeswitch-users] FreeTDM and Libpri with a Digium TE122 ...
 some calls not being answered...
In-Reply-To: <AANLkTikuBBd2UqfdASK+7ChizOu=5c7xhQYaHnf2LpYJ@mail.gmail.com>
References: <AANLkTimb3rLWvfZ=dTBM9XZeLa=OenJoz=zjRjdoD6Ym@mail.gmail.com>
	<AANLkTikuBBd2UqfdASK+7ChizOu=5c7xhQYaHnf2LpYJ@mail.gmail.com>
Message-ID: <AANLkTimbFveqqYzMN-FdXboKtw_16UhMf7pCugsG07oO@mail.gmail.com>

Hi Michael, thankyou for your reply. I havn't had a chance to catch Moy on
#freetdm yet, but I'll keep an eye out so I can get to the bottom of this...

Further information perhaps, a snippet from a failed log is at
http://pastebin.freeswitch.org/15620 which I've highlighted a few lines
which may or may not be the cause.

If anyone notices anything out of the ordinary that will be fantastic,
otherwise I'm back to square one.

I've also noticed it runs perfectly fine, up until a point (usually around
1:30 in the afternoon). I'm doing a reload mod_freetdm in a cron job each
night, yet still it gets to the point where calls are not being accepted.

Thank you again.

Matt.




On 5 March 2011 05:22, Michael Collins <msc at freeswitch.org> wrote:

> Well...
>
> I examined these Q931 traces and I don't see anything different between the
> failed call and the successful call. I would hop into #freetdm on
> irc.freenode.net and see if moy is around. He might be able to shed some
> light on this. If it's not occurring at the libpri level then he might be
> able to help figure out if FreeTDM is not behaving properly.
>
> -MC
>
> On Wed, Mar 2, 2011 at 9:13 PM, Matt Paine <mattzerah+freeswitch at gmail.com
> > wrote:
>
>> Hi Guys.
>>
>> Current setup is a FS box with a Digium TE122 installed, with DAHDI kernel
>> drivers and using FreeTDM with a libpri span...
>>
>> == autoload_configs/freetdm.conf.xml ==
>>
>> <configuration name="freetdm.conf" description="FreeTDM Configuration">
>>     <settings>
>>         <param name="debug" value="0"/>
>>         <param name="sip_headers" value="true"/>
>>     </settings>
>>     <libpri_spans>
>>         <span name="ISDN">
>>             <param name="mode" value="user"/>
>>             <param name="dialect" value="q931"/>
>>              <param name="dialplan" value="XML"/>
>>             <param name="context" value="public"/>
>>         </span>
>>     </libpri_spans>
>> </configuration>
>>
>> == freetdm.conf ==
>>
>> [general]
>> cpu_monitor => yes
>> cpu_monitoring_interval => 1000
>> cpu_set_alarm_threshold => 80
>> cpu_reset_alarm_threshold => 70
>> cpu_alarm_action => warn
>>
>> [span zt ISDN]
>> trunk_type => E1
>> b-channel => 1-15
>> d-channel => 16
>> b-channel => 17-21
>>
>> == zt.conf ==
>> [defaults]
>> codec_ms => 20
>> wink_ms => 150
>> flash_ms => 750
>> echo_cancel_level => 128
>> rxgain => 0.0
>> txgain => 0.0
>>
>> == /etc/dahdi/system.conf ==
>> span=1,1,0,ccs,hdb3,crc4
>> # termtype: te
>> bchan=1-15,17-31
>> dchan=16
>> echocanceller=mg2,1-15,17-31
>> # Global data
>> loadzone    = au
>> defaultzone = au
>>
>>
>> =========
>> And finally some logs from two calls, a successful one, and a failed
>> one...
>>  http://pastebin.freeswitch.org/15533
>>
>>
>> If anyone could offer any suggestions as to what I can do to answer every
>> incoming call into the box that would very appreciated. Of course if I need
>> to provide more information I am willing to, and any suggestions for
>> settings that need changing to test out will also be great.
>>
>> Thank you in advance
>> Matt
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From laurentiu.ceausescu at gmail.com  Thu Mar 10 00:48:32 2011
From: laurentiu.ceausescu at gmail.com (Laurentiu Ceausescu)
Date: Wed, 9 Mar 2011 23:48:32 +0200
Subject: [Freeswitch-users] Error! cannot get read lock
Message-ID: <AANLkTimKdKafaD4tzko68jb4zXd_djT+ZiaK4Jz_YCzT@mail.gmail.com>

Hi,

I have the following dialplan:
  <extension name="line1">
    <condition field="destination_number" expression="^333$">
      <action application="answer"/>
      <action application="set" data="allow_voicemail=true"/>
      <action application="set" data="queue=Queue"/>
      <action application="set" data="brand=110011"/>
      <action application="playback" data="/usr/local/audio/welcome.wav"/>
      <action application="set" data="domain_name=$${domain}"/>
      <action application="erlang_sendmsg"
data="freeswitch_media_manager toor at 127.0.0.1 inivr ${uuid}"/>
      <action application="erlang" data="freeswitch_media_manager:!
toor at 127.0.0.1"/>
    </condition>
  </extension>

When I call 333 I can hear welcome message but then the call is dropped.
FS logs shows this error: [ERR] handle_msg.c:56 Error! cannot get read lock.

[NOTICE] switch_channel.c:812 New Channel
sofia/toor.mydomain.org/401 at toor.mydomain.org
[03ecb2ca-24df-4903-917a-c9720b43a511]
[NOTICE] sofia_glue.c:3754 Pre-Answer
sofia/toor.mydomain.org/401 at toor.mydomain.org!
[NOTICE] mod_dptools.c:929 Channel
[sofia/toor.mydomain.org/401 at toor.mydomain.org] has been answered
[ERR] handle_msg.c:56 Error! cannot get read lock.
[WARNING] mod_erlang_event.c:1365 Timed out when waiting for outbound
pid 1.0.0 at freeswitch@127.0.0.1 03ecb2ca-24df-4903-917a-c
[NOTICE] switch_core_state_machine.c:189
sofia/toor.mydomain.org/401 at toor.mydomain.org has executed the last
dialplan instruct
[NOTICE] switch_core_state_machine.c:191 Hangup
sofia/toor.mydomain.org/401 at toor.mydomain.org [CS_EXECUTE]
[NORMAL_CLEARING]
[NOTICE] switch_core_session.c:1306 Session 4
(sofia/toor.mydomain.org/401 at toor.mydomain.org) Ended
[NOTICE] switch_core_session.c:1308 Close Channel
sofia/toor.mydomain.org/401 at toor.mydomain.org [CS_DESTROY]
[ERR] handle_msg.c:56 Error! cannot get read lock.


This thing is happend only after I updated to last git head
(419d7e2335042e2d86c678c0daf86c2d9c867036 /Mar 9).
With a older revision (e88b9639624cef4f35901146241f515730b3b118 /Jan
31) everything works fine.
Any idea what has been changed?

Thanks,
Laurentiu


From msc at freeswitch.org  Thu Mar 10 01:44:33 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 9 Mar 2011 14:44:33 -0800
Subject: [Freeswitch-users] incompatible destination
In-Reply-To: <AANLkTimghyn8KeN_PETXtqkkKgHQvyNBktw1jAEnav6Y@mail.gmail.com>
References: <AANLkTincX-zpfnqx4dRk2jLCPf6HDiGN+cuPDA7+75WX@mail.gmail.com>
	<AANLkTi=LwtDEF6-RumWCw2dLThTjcZXwET9_O_OiMN96@mail.gmail.com>
	<AANLkTimghyn8KeN_PETXtqkkKgHQvyNBktw1jAEnav6Y@mail.gmail.com>
Message-ID: <AANLkTimgmHsVNzyONXkMCH12-9G=de0quW=NjHFjhVZx@mail.gmail.com>

Without seeing the whole config or the SIP trace I can't really say much
about what is happening. If I were you I would backup my configs and start
from scratch. My guess is that you changed something that you didn't mean to
change. If you restart from a known I'll bet you can get it working again.

-MC

On Tue, Mar 8, 2011 at 10:00 PM, Sam <u2nsam at gmail.com> wrote:

> Thanks,
>
> Yes 192.168.2.190 is FS and I only have 1 polycom phone and rest are cisco
> 7960s .
>
> I tried with g711alaw in third priority and its working, but i want to have
> the codec preference on polycom like g722,g711ulaw,g711alaw.
>
> I have 2 debugs when i removed G711alaw and just kept G722 ,G711,G729 .
>
> Made a call from polycom to cisco the call disconnected after pick up.
> http://pastebin.freeswitch.org/15621
>
> Made an incoming call to polycom and it did not transcode.
> http://pastebin.freeswitch.org/15622
>
> I am having this settings in profile:
> <param name="inbound-codec-negotiation" value="generous"/>
>     <param name="disable-transcoding" value="true"/>
> also tried :-
> <param name="inbound-codec-negotiation" value="generous"/>
>     <param name="disable-transcoding" value="false"/>
>
> Am i correct with those , if not do correct me.
>
> Regards
> Sam
>
>
>
>
>
> On Wed, Mar 9, 2011 at 10:36 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Sun, Mar 6, 2011 at 10:20 PM, Sam <u2nsam at gmail.com> wrote:
>>
>>> Hello,
>>>
>>>
>>> The extension 7006 was working few days ago and now it not working,rest
>>> all the phone are working properly.
>>>
>>
>> Sam,
>>
>> Is 192.168.2.190 your FreeSWITCH box? If so then the issue is with the
>> phone not accepting the PCMA codec. Go into your Polycom config and make
>> sure that PCMA is enabled on the phone itself. The other option would be to
>> let the Polycom speak in G722 and let FS do transcoding. In any case, if I
>> were you I would find another working Polycom phone and mimic the configs on
>> this non-working Poly 335.
>>
>> -MC
>>
>>
>>>
>>> U 192.168.2.190:5060 -> 192.168.2.14:5060
>>>   INVITE sip:7006 at 192.168.2.14:5060 SIP/2.0..Via: SIP/2.0/UDP
>>> 192.168.2.190;rport;branch=z9hG4bK8mmBB5e8Z2mKr..Max-Forwards: 70..From: ""
>>> <sip:9322273640 at 192.168.2.190>;tag=4QggZ1XcNZD3F..To: <
>>> sip:7006 at 192.168.2.14:5060>..Call-ID: 7bf
>>>   ae496-c324-122e-bc85-0026b976aef7..CSeq: 9403059 INVITE..Contact: <
>>> sip:mod_sofia at 192.168.2.190:5060>..User-Agent: NOVANET..Allow: INVITE,
>>> ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY,
>>> PUBLISH, SUBSCRIBE.
>>>   .Supported: timer, precondition, path, replaces..Allow-Events: talk,
>>> hold, presence, dialog, line-seize, call-info, sla,
>>> include-session-description, presence.winfo, message-summary,
>>> refer..Content-Type: application/sdp..Content-Dis
>>>   position: session..Content-Length: 203..X-FS-Support:
>>> update_display..Remote-Party-ID: <sip:9322273640 at 192.168.2.190>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH
>>> 1299446773 1299446774 IN IP4 192.168.2.190..s=FreeSWITCH
>>>   ..c=IN IP4 192.168.2.190..t=0 0..m=audio 31474 RTP/AVP 8 101
>>> 13..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=ptime:20..
>>>
>>> U 192.168.2.14:5060 -> 192.168.2.190:5060
>>>   SIP/2.0 488 Not Acceptable Here..Via: SIP/2.0/UDP
>>> 192.168.2.190;rport;branch=z9hG4bK8mmBB5e8Z2mKr..From: <
>>> sip:9322273640 at 192.168.2.190>;tag=4QggZ1XcNZD3F..To: "7006" <
>>> sip:7006 at 192.168.2.14:5060>;tag=983B5A59-EB953B22..CSeq: 9403059
>>>   INVITE..Call-ID: 7bfae496-c324-122e-bc85-0026b976aef7..Contact: <
>>> sip:7006 at 192.168.2.14:5060>..User-Agent:
>>> PolycomSoundPointIP-SPIP_335-UA/3.2.2.0477..Accept-Language: en..Warning:
>>> 488 SDP "NotAcceptableHere"..Content-Length: 0....
>>>
>>>
>>>
>>> I tried all the possibilities with codec priorities on U/A .
>>>
>>> http://pastebin.freeswitch.org/15577
>>>
>>> Any help.
>>>
>>> Regards
>>> Sam
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Thu Mar 10 05:26:01 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 9 Mar 2011 20:26:01 -0600
Subject: [Freeswitch-users] Error! cannot get read lock
In-Reply-To: <AANLkTimKdKafaD4tzko68jb4zXd_djT+ZiaK4Jz_YCzT@mail.gmail.com>
References: <AANLkTimKdKafaD4tzko68jb4zXd_djT+ZiaK4Jz_YCzT@mail.gmail.com>
Message-ID: <AANLkTimwfAFbJu8HxLxp2eoAznoR3bYzoB+75eea4+AX@mail.gmail.com>

you should open a JIRA under mod_erlang_event to find out.


On Wed, Mar 9, 2011 at 3:48 PM, Laurentiu Ceausescu
<laurentiu.ceausescu at gmail.com> wrote:
> Hi,
>
> I have the following dialplan:
> ?<extension name="line1">
> ? ?<condition field="destination_number" expression="^333$">
> ? ? ?<action application="answer"/>
> ? ? ?<action application="set" data="allow_voicemail=true"/>
> ? ? ?<action application="set" data="queue=Queue"/>
> ? ? ?<action application="set" data="brand=110011"/>
> ? ? ?<action application="playback" data="/usr/local/audio/welcome.wav"/>
> ? ? ?<action application="set" data="domain_name=$${domain}"/>
> ? ? ?<action application="erlang_sendmsg"
> data="freeswitch_media_manager toor at 127.0.0.1 inivr ${uuid}"/>
> ? ? ?<action application="erlang" data="freeswitch_media_manager:!
> toor at 127.0.0.1"/>
> ? ?</condition>
> ?</extension>
>
> When I call 333 I can hear welcome message but then the call is dropped.
> FS logs shows this error: [ERR] handle_msg.c:56 Error! cannot get read lock.
>
> [NOTICE] switch_channel.c:812 New Channel
> sofia/toor.mydomain.org/401 at toor.mydomain.org
> [03ecb2ca-24df-4903-917a-c9720b43a511]
> [NOTICE] sofia_glue.c:3754 Pre-Answer
> sofia/toor.mydomain.org/401 at toor.mydomain.org!
> [NOTICE] mod_dptools.c:929 Channel
> [sofia/toor.mydomain.org/401 at toor.mydomain.org] has been answered
> [ERR] handle_msg.c:56 Error! cannot get read lock.
> [WARNING] mod_erlang_event.c:1365 Timed out when waiting for outbound
> pid 1.0.0 at freeswitch@127.0.0.1 03ecb2ca-24df-4903-917a-c
> [NOTICE] switch_core_state_machine.c:189
> sofia/toor.mydomain.org/401 at toor.mydomain.org has executed the last
> dialplan instruct
> [NOTICE] switch_core_state_machine.c:191 Hangup
> sofia/toor.mydomain.org/401 at toor.mydomain.org [CS_EXECUTE]
> [NORMAL_CLEARING]
> [NOTICE] switch_core_session.c:1306 Session 4
> (sofia/toor.mydomain.org/401 at toor.mydomain.org) Ended
> [NOTICE] switch_core_session.c:1308 Close Channel
> sofia/toor.mydomain.org/401 at toor.mydomain.org [CS_DESTROY]
> [ERR] handle_msg.c:56 Error! cannot get read lock.
>
>
> This thing is happend only after I updated to last git head
> (419d7e2335042e2d86c678c0daf86c2d9c867036 /Mar 9).
> With a older revision (e88b9639624cef4f35901146241f515730b3b118 /Jan
> 31) everything works fine.
> Any idea what has been changed?
>
> Thanks,
> Laurentiu
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From dujinfang at gmail.com  Thu Mar 10 06:24:53 2011
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 10 Mar 2011 11:24:53 +0800
Subject: [Freeswitch-users] Error! cannot get read lock
In-Reply-To: <AANLkTimwfAFbJu8HxLxp2eoAznoR3bYzoB+75eea4+AX@mail.gmail.com>
References: <AANLkTimKdKafaD4tzko68jb4zXd_djT+ZiaK4Jz_YCzT@mail.gmail.com>
	<AANLkTimwfAFbJu8HxLxp2eoAznoR3bYzoB+75eea4+AX@mail.gmail.com>
Message-ID: <AANLkTi=nVN4+b1vi__rSU5PeWhU6vHKip5c4Eh1vZV+F@mail.gmail.com>

I run into this short time ago and rebuilt mod_erlang_event fixed.

cd freeswitch-source-tree
make mod_erlang_event-install

fs_cli> reload mod_erlang_event

On Thu, Mar 10, 2011 at 10:26 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> you should open a JIRA under mod_erlang_event to find out.
>
>
> On Wed, Mar 9, 2011 at 3:48 PM, Laurentiu Ceausescu
> <laurentiu.ceausescu at gmail.com> wrote:
>> Hi,
>>
>> I have the following dialplan:
>> ?<extension name="line1">
>> ? ?<condition field="destination_number" expression="^333$">
>> ? ? ?<action application="answer"/>
>> ? ? ?<action application="set" data="allow_voicemail=true"/>
>> ? ? ?<action application="set" data="queue=Queue"/>
>> ? ? ?<action application="set" data="brand=110011"/>
>> ? ? ?<action application="playback" data="/usr/local/audio/welcome.wav"/>
>> ? ? ?<action application="set" data="domain_name=$${domain}"/>
>> ? ? ?<action application="erlang_sendmsg"
>> data="freeswitch_media_manager toor at 127.0.0.1 inivr ${uuid}"/>
>> ? ? ?<action application="erlang" data="freeswitch_media_manager:!
>> toor at 127.0.0.1"/>
>> ? ?</condition>
>> ?</extension>
>>
>> When I call 333 I can hear welcome message but then the call is dropped.
>> FS logs shows this error: [ERR] handle_msg.c:56 Error! cannot get read lock.
>>
>> [NOTICE] switch_channel.c:812 New Channel
>> sofia/toor.mydomain.org/401 at toor.mydomain.org
>> [03ecb2ca-24df-4903-917a-c9720b43a511]
>> [NOTICE] sofia_glue.c:3754 Pre-Answer
>> sofia/toor.mydomain.org/401 at toor.mydomain.org!
>> [NOTICE] mod_dptools.c:929 Channel
>> [sofia/toor.mydomain.org/401 at toor.mydomain.org] has been answered
>> [ERR] handle_msg.c:56 Error! cannot get read lock.
>> [WARNING] mod_erlang_event.c:1365 Timed out when waiting for outbound
>> pid 1.0.0 at freeswitch@127.0.0.1 03ecb2ca-24df-4903-917a-c
>> [NOTICE] switch_core_state_machine.c:189
>> sofia/toor.mydomain.org/401 at toor.mydomain.org has executed the last
>> dialplan instruct
>> [NOTICE] switch_core_state_machine.c:191 Hangup
>> sofia/toor.mydomain.org/401 at toor.mydomain.org [CS_EXECUTE]
>> [NORMAL_CLEARING]
>> [NOTICE] switch_core_session.c:1306 Session 4
>> (sofia/toor.mydomain.org/401 at toor.mydomain.org) Ended
>> [NOTICE] switch_core_session.c:1308 Close Channel
>> sofia/toor.mydomain.org/401 at toor.mydomain.org [CS_DESTROY]
>> [ERR] handle_msg.c:56 Error! cannot get read lock.
>>
>>
>> This thing is happend only after I updated to last git head
>> (419d7e2335042e2d86c678c0daf86c2d9c867036 /Mar 9).
>> With a older revision (e88b9639624cef4f35901146241f515730b3b118 /Jan
>> 31) everything works fine.
>> Any idea what has been changed?
>>
>> Thanks,
>> Laurentiu
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



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


From cmrienzo at gmail.com  Thu Mar 10 06:25:34 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Wed, 9 Mar 2011 22:25:34 -0500
Subject: [Freeswitch-users] ASR from Freeswitch to MS Speech Server
 [using MRCP Connector] - Audio Problem
In-Reply-To: <AANLkTim6u-77TfSJ6aypfNxNLjHrDwnW9EYNin-L40w7@mail.gmail.com>
References: <AANLkTim6u-77TfSJ6aypfNxNLjHrDwnW9EYNin-L40w7@mail.gmail.com>
Message-ID: <AANLkTimgt3idavgB1Pq_EYtVFCNSmnhHgRsOZYrLtJnW@mail.gmail.com>

I still would like to see the logs for your call.


On Tue, Mar 8, 2011 at 7:18 AM, Michel Habib <michelhabib at gmail.com> wrote:

> Yes, I get the Audio from FS in regular calls - I already disabled all
> possible firewalls - all 3 machines [softphone, freeswitch, Speech Server
> (and mrcp connector) ] are on a switch.
> 192.168.5.107 is the freeswitch server
> 192.168.5.110 is the MRCP-Connector/Speech Server 2007 server
> I made too many iterations on the configuration below:
>
> <include>
>   <profile name="mrcp-connector" version="2">
>     <param name="client-ip" value="192.168.5.107"/>
>     <param name="client-port" value="5090"/>
>     <param name="server-ip" value="192.168.5.110"/>
>     <param name="server-port" value="5070"/>
>     <!--param name="force-destination" value="1"/-->
>     <param name="sip-transport" value="udp"/>
>     <!--param name="ua-name" value="FreeSWITCH"/-->
>     <!--param name="sdp-origin" value="FreeSWITCH"/-->
>     <param name="rtp-ext-ip" value="192.168.5.107"/>
>     <param name="rtp-ip" value="192.168.5.107"/>
>     <param name="rtp-port-min" value="4000"/>
>     <param name="rtp-port-max" value="5000"/>
>     <!-- enable/disable rtcp support -->
>     <param name="rtcp" value="1"/>
>     <!-- rtcp bye policies (rtcp must be enabled first)
>              0 - disable rtcp bye
>              1 - send rtcp bye at the end of session
>              2 - send rtcp bye also at the end of each talkspurt (input)
>     -->
>     <param name="rtcp-bye" value="2"/>
>     <!-- rtcp transmission interval in msec (set 0 to disable) -->
>     <param name="rtcp-tx-interval" value="5000"/>
>     <!-- period (timeout) to check for new rtcp messages in msec (set 0 to
> disable) -->
>     <param name="rtcp-rx-resolution" value="1000"/>
>
>     <!--param name="playout-delay" value="50"/-->
>     <!--param name="max-playout-delay" value="200"/-->
>     <!--param name="ptime" value="20"/-->
>     <param name="codecs" value="PCMU PCMA L16/96/8000"/>
>
>     <!-- Add any default MRCP params for SPEAK requests here -->
>     <synthparams>
>     </synthparams>
>
>     <!-- Add any default MRCP params for RECOGNIZE requests here -->
>     <recogparams>
>       <!--param name="start-input-timers" value="false"/-->
>     </recogparams>
>   </profile>
> </include>
>
>
>
> ---------- Forwarded message ----------
>
>> From: Christopher Rienzo <cmrienzo at gmail.com>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Date: Mon, 7 Mar 2011 09:32:51 -0500
>> Subject: Re: [Freeswitch-users] ASR from Freeswitch to MS Speech Server
>> [using MRCP Connector] - Audio Problem
>> Do you get audio between FS and your SIP client when not using ASR/TTS?
>>
>> Show me the MRCP profile configuration and your FreeSWITCH logs during the
>> call.
>>
>>
>>
>> On Mon, Mar 7, 2011 at 4:55 AM, Michel Habib <michelhabib at gmail.com>wrote:
>>
>>> Hello All,
>>> I have MS OCS Speech Server 2007 [working correctly, as i can make SIP
>>> calls and use its ASR and TTS Services successfully]
>>> I am also using MRCP Connector from AumTech - which allows me to use ASR
>>> and TTS Services through an MRCP Client .
>>> Now, i am using Freeswitch mod unimrcp to use ASR and TTS.
>>>
>>> for TTS, I can successfully make the call, the Audio RTP of the TTS voice
>>> is transferred succesfully from Speech Server [through MRCP Connector] back
>>> to the Freeswitch Server.
>>> However, Freeswitch is not sending back the Audio RTP to the SIP client.
>>>
>>> for ASR, I can successfully define the grammar and start recognition, but
>>> the audio RTP sent to speech server [through MRCP Connector] is silent
>>> [empty].
>>>
>>> I am suspecting something is wrong with the RTP Configuration - can you
>>> help me?
>>>
>>> Let me now if you need any specific logs/scripts/configuration?
>>>
>>> Thank you,
>>> Michel.
>>>
>>>
>>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Thu Mar 10 07:02:58 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 10 Mar 2011 09:32:58 +0530
Subject: [Freeswitch-users] mod callcenter
In-Reply-To: <AANLkTikTRte3_848LiNH0=ZjqL7Atmzx_wBaLziHzUnJ@mail.gmail.com>
References: <AANLkTikdRNMo4CgFcuon0X4KkyZSzZYHK6RECaDxrNwa@mail.gmail.com>
	<AANLkTin0_cUT1Wb-SanDj_sD_JWnrzJ_QC1YpUMbBDc0@mail.gmail.com>
	<AANLkTikTRte3_848LiNH0=ZjqL7Atmzx_wBaLziHzUnJ@mail.gmail.com>
Message-ID: <AANLkTikb0D5vu78dZmLuVvocoGA2_aDXQKA6OKVD+fU7@mail.gmail.com>

Hello,

I have updated to the latest git and still every week i need to reload mod
callcenter ,
is there any problem with the module as it shows invalid application and
after mod reload
it works fine for 4-5 day.

Regards
Sam



On Tue, Jan 4, 2011 at 1:33 PM, Sam <u2nsam at gmail.com> wrote:

> i have pasted more logs for the error on jira as it showed up again.
> FS-2952
>
> Regds
> Sam
>
>
>
>
> On Thu, Dec 30, 2010 at 3:54 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Open a tick on jira.freeswitch.org and Moc will take a look. Be sure to
>> provide as much info as possible.
>> -MC
>>
>> On Wed, Dec 29, 2010 at 6:48 AM, Sam <u2nsam at gmail.com> wrote:
>>
>>> Hello,
>>>
>>> Was testing callcenter module and found out that at times it gives error
>>> " invalid application callcenter " and after
>>> reloading the module it works fine.
>>> Some time also happens that if I reload the module it do not reloads the
>>> parameters of the callcenter like the agents & tires.
>>> It just unload & reloads even if there are changes to the specifications
>>> of agents.
>>>
>>> Regds
>>> Sam
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From kris at kriskinc.com  Thu Mar 10 07:36:09 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 9 Mar 2011 23:36:09 -0500
Subject: [Freeswitch-users] SIP Trunk Registration
In-Reply-To: <6CF1CE22-FF54-46FC-A159-202CA04542BE@markholloway.com>
References: <6CF1CE22-FF54-46FC-A159-202CA04542BE@markholloway.com>
Message-ID: <AANLkTin8hk42StWU_3rxFEX9imDw8-=OdUw3zQfs3ExT@mail.gmail.com>

Mark,

  FreeSWITCH will register providing the contact in the "extension"
field for the gateway entry.  Broadsoft (most likely) just strips the
username portion of the contact URI and inserts the destination number
for the remaining 99 dids.

  In short, populate the extension field in the gateway entry with the
"pilot number" and see what happens.

On Wed, Mar 9, 2011 at 12:36 PM, Mark Holloway <mh at markholloway.com> wrote:
> Does FreeSwitch support single phone number registration for a Trunk Group? ?For example, in Broadsoft Broadworks I can build a Trunk Group with 100 numbers but I identify one number as the Pilot Number. ?I have a SIP endpoint (such as a Cisco ISR or Adtran TA900) that registers with Broadworks using the main Pilot Number only (Business Contact) so I don't have to register all 100 numbers. ?It's like a parent-child relationship where the other 99 numbers are a child of the main parent (pilot) number.
>
> Thanks,
> Mark
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner


From ovvenkatesan at gmail.com  Thu Mar 10 12:32:54 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Thu, 10 Mar 2011 15:02:54 +0530
Subject: [Freeswitch-users] Not able to install gsmopen module on FC 13
Message-ID: <AANLkTiksCJBxnVB9-U2HefWdT8LuTwH1bry4KozBZ2yW@mail.gmail.com>

Hi to all,

I have followed this wiki page to intall gsmopen module

http://wiki.freeswitch.org/wiki/GSMopen#Linux

Installed following prerequisites packages

# yum -y install alsa-lib-devel alsa-utils

then, go to gsmopen directory

#/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/gsmlib/gsmlib-1.10-patched-12ubuntu1

I wanted to compile only SMS support, no voice call.

When, I compiled the source code , I am getting following error ,

#make
......
......
gsm_unix_serial.cc:89: error: 'strerror' was not declared in this scope
make[1]: *** [gsm_unix_serial.lo] Error 1
make[1]: Leaving directory `/usr/local/src/freeswitch/
src/mod/endpoints/mod_gsmopen/gsmlib/gsmlib-1.10-patched-12ubuntu1/gsmlib'
make: *** [install-recursive] Error 1

When,  I enter into following directory and trying the compile the source
code,

#/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen

# make

I am getting following error

gsmopen_protocol.cpp:5:28: error: gsmlib/gsm_sms.h: No such file or
directory
gsmopen_protocol.cpp:9:36: error: gsmlib/gsm_unix_serial.h: No such file or
directory
gsmopen_protocol.cpp:11:30: error: gsmlib/gsm_me_ta.h: No such file or
directory
gsmopen_protocol.cpp:16: error: ?gsmlib? is not a namespace-name
...
...
gsmopen_protocol.cpp:3223: error: ?DCS_DEFAULT_ALPHABET? was not declared in
this scope
gsmopen_protocol.cpp:3225: error: ?DCS_SIXTEEN_BIT_ALPHABET? was not
declared in this scope
gsmopen_protocol.cpp:3226: error: ?bufToHex? was not declared in this scope
make[1]: *** [gsmopen_protocol.o] Error 1
make: *** [all] Error 1


I do not know, what I am doing wrong.
Anyone please help me regarding this.


--

Regards
Venkatesan OV.
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From erkan at speedingtrade.com  Thu Mar 10 14:29:09 2011
From: erkan at speedingtrade.com (=?utf-8?B?RXJrYW4gw5xubMO8?=)
Date: Thu, 10 Mar 2011 13:29:09 +0200
Subject: [Freeswitch-users] sip auth challenge HACKING ???
References: <81C2CEF80046FB4F863A60D4347DD33A0C56EC@server1.st.local><52114D605A462A4E9A50E588EE7D0686012E575D9EF8@mail.forest.simpot.com>
	<AANLkTi=o-5oFhnBcgA=KX9Zo+i1AgisusyH88zw067Gk@mail.gmail.com>
Message-ID: <81C2CEF80046FB4F863A60D4347DD33A0C56F6@server1.st.local>

:-)  good idea my friend.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of curriegrad2004
Sent: Tuesday, March 08, 2011 5:32 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sip auth challenge HACKING ???

Or just for fun, you can set up a honeypot with all extensions routing to nowhere or to a very very nasty extension ;)

On Tue, Mar 8, 2011 at 3:00 AM, Dmitry Saratsky <simpot at simpot.com> wrote:
> I?m blocking it with: http://wiki.freeswitch.org/wiki/Fail2ban
>
>
>
>
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of 
> Erkan ?nl?
> Sent: 08 Mar 2011 12:39
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] sip auth challenge HACKING ???
>
>
>
> Hi FS Users,
>
>
>
> in last time i see in my console of FS this kind of error messages.
>
>
>
> [WARNING] sofia_reg.c:1246 SIP auth challenge (INVITE) on sofia 
> profile 'internal' for [5828@?????????] from ip ???7?.1??.7??.???
>
>
>
> i check my config files ever again and again, but today the console is 
> only given this kind of messages. Maybe 20 messages per second.
>
>
>
> i see the ip that given in the console ?from ip xx.xx.xx.xx? i block 
> this ip in my firewall and everything is fine.
>
> Now i understand that this a trying to hacking my server. The blocking 
> of the ip is a solution but can not handle this in Freeswitch, because 
> i see this problem sometimes on different FS servers also and in 
> normally the FS server maybe must can handle this problem. For example 
> with automatic black lists if an ip trys more than 20 times with wrong 
> login. So the ip will be banned for 1 hour or so.
>
>
>
> i?m interesting in if other users have the same problems and ideas in 
> how we can handle this.
>
>
>
> Kind regards
>
> Erkan
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>
>

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

From dnotivol at gmail.com  Thu Mar 10 14:53:05 2011
From: dnotivol at gmail.com (David Notivol)
Date: Thu, 10 Mar 2011 12:53:05 +0100
Subject: [Freeswitch-users] mod_say in different codecs
In-Reply-To: <AANLkTik4b1-iHrdA8cOtzvb6797ZpoYWRA6CyThru6vz@mail.gmail.com>
References: <AANLkTikb_8mv+L1PSqRk-d2QMi1sNSJN-6_5WFmWDrKW@mail.gmail.com>
	<375B3BB7-DC39-4C80-A32B-DF1941904291@gmail.com>
	<AANLkTik4b1-iHrdA8cOtzvb6797ZpoYWRA6CyThru6vz@mail.gmail.com>
Message-ID: <AANLkTineyBkZ-e_d9vMzCnOHoy2E5aMWU3KeT9Rdb=8b@mail.gmail.com>

Hi all,

Does anyone know how to have mod_say working using mod_native_file for the
pronounced numbers? It seems mod_say has the .wav extension included in the
source code...

Thanks.
-- 
David.

2011/3/3 David Notivol <dnotivol at gmail.com>

> Thanks Steve for your prompt reply.
>
> Yes, mod_native_file does suit my needs for regular audio files; but I
> think it doesn't for numbers. Actually, I'm using say to pronounce numbers,
> but it seems mod_say forces FS to use the wav files.
>
> I was having a look at the source code for the mod_say_en, and I removed
> some of the ".wav" strings for the digits and recompiled it; and  then the
> native_file was triggered and played the .PCMU file (or any other needed
> codec) instead of the .wav file.
> But I'm not sure this is the way to proceed, or if that can get to other
> problems...
>
> Is there any way of having mod_say not forcing to use always .wav files and
> relying on mod_native_file ?
>
> --
> David
>
> 2011/3/3 Steven Ayre <steveayre at gmail.com>
>
> Look at mod_nativefile - does that suit your needs?
>>
>> Steve on iPhone
>>
>> On 3 Mar 2011, at 12:19, David Notivol <dnotivol at gmail.com> wrote:
>>
>> > Hi all,
>> >
>> > I'm trying to setup an IVR server, and I'm using the session:say
>> function from a LUA script.
>> >
>> > My question is if there's any way of having mod_say playing audios
>> different than audio files; I mean audios encoded in G729, G711ulaw,
>> G711alaw, etc. to avoid having FS making transcoding every time I run a say
>> command.
>> >
>> > Checking the folders tree for the sounds, I can see the place for the
>> different languages, voices, and wav qualities (8k, 16k...) is clearly
>> specified; but is it a way to place files encoded in different codecs?
>> >
>> > Thanks in advance.
>> >
>> > Regards,
>> > David Notivol
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
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From curriegrad2004 at gmail.com  Thu Mar 10 18:48:11 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 10 Mar 2011 07:48:11 -0800
Subject: [Freeswitch-users] Not able to install gsmopen module on FC 13
In-Reply-To: <AANLkTiksCJBxnVB9-U2HefWdT8LuTwH1bry4KozBZ2yW@mail.gmail.com>
References: <AANLkTiksCJBxnVB9-U2HefWdT8LuTwH1bry4KozBZ2yW@mail.gmail.com>
Message-ID: <AANLkTinVODFX_X8657jJLjCf9fy+WCXr9YxgwqKO5p9s@mail.gmail.com>

seems like you haven't installed gsmlib correctly. you'll need to get
a copy of the gsmlib headers or preferrably the source code and re-try
the compilation again

On Thu, Mar 10, 2011 at 1:32 AM, ovvenkat <ovvenkatesan at gmail.com> wrote:
> Hi to all,
>
> I have followed this wiki page to intall gsmopen module
>
> http://wiki.freeswitch.org/wiki/GSMopen#Linux
>
> Installed following prerequisites packages
>
> # yum -y install alsa-lib-devel alsa-utils
>
> then, go to gsmopen directory
>
> #/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/gsmlib/gsmlib-1.10-patched-12ubuntu1
>
> I wanted to compile only SMS support, no voice call.
>
> When, I compiled the source code , I am getting following error ,
>
> #make
> ......
> ......
> gsm_unix_serial.cc:89: error: 'strerror' was not declared in this scope
> make[1]: *** [gsm_unix_serial.lo] Error 1
> make[1]: Leaving directory
> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/gsmlib/gsmlib-1.10-patched-12ubuntu1/gsmlib'
> make: *** [install-recursive] Error 1
>
> When,? I enter into following directory and trying the compile the source
> code,
>
> #/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen
>
> # make
>
> I am getting following error
>
> gsmopen_protocol.cpp:5:28: error: gsmlib/gsm_sms.h: No such file or
> directory
> gsmopen_protocol.cpp:9:36: error: gsmlib/gsm_unix_serial.h: No such file or
> directory
> gsmopen_protocol.cpp:11:30: error: gsmlib/gsm_me_ta.h: No such file or
> directory
> gsmopen_protocol.cpp:16: error: ?gsmlib? is not a namespace-name
> ...
> ...
> gsmopen_protocol.cpp:3223: error: ?DCS_DEFAULT_ALPHABET? was not declared in
> this scope
> gsmopen_protocol.cpp:3225: error: ?DCS_SIXTEEN_BIT_ALPHABET? was not
> declared in this scope
> gsmopen_protocol.cpp:3226: error: ?bufToHex? was not declared in this scope
> make[1]: *** [gsmopen_protocol.o] Error 1
> make: *** [all] Error 1
>
>
> I do not know, what I am doing wrong.
> Anyone please help me regarding this.
>
>
> --
>
> Regards
> Venkatesan OV.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From gmaruzz at gmail.com  Thu Mar 10 18:55:28 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 10 Mar 2011 16:55:28 +0100
Subject: [Freeswitch-users] Not able to install gsmopen module on FC 13
In-Reply-To: <AANLkTinVODFX_X8657jJLjCf9fy+WCXr9YxgwqKO5p9s@mail.gmail.com>
References: <AANLkTiksCJBxnVB9-U2HefWdT8LuTwH1bry4KozBZ2yW@mail.gmail.com>
	<AANLkTinVODFX_X8657jJLjCf9fy+WCXr9YxgwqKO5p9s@mail.gmail.com>
Message-ID: <AANLkTikq5v1nTWCU1uN1vHJh2jGDGxF3F-c7z=kqxO9i@mail.gmail.com>

yep

you fail on gsmlib compilation (so it cannot find it installed)

Problem is, that the version of gsmlib that is in gsmopen sources do
not compile on newer g++ (eg: on Fedora 13, recent Ubuntus).

So, try to see if you can find gsmlib for fedora ("yum search gsmlib")
and be sure to install the development package too (Ubuntu has it, I
think Fedora has it too).

-giovanni

On Thu, Mar 10, 2011 at 4:48 PM, curriegrad2004
<curriegrad2004 at gmail.com> wrote:
> seems like you haven't installed gsmlib correctly. you'll need to get
> a copy of the gsmlib headers or preferrably the source code and re-try
> the compilation again
>
> On Thu, Mar 10, 2011 at 1:32 AM, ovvenkat <ovvenkatesan at gmail.com> wrote:
>> Hi to all,
>>
>> I have followed this wiki page to intall gsmopen module
>>
>> http://wiki.freeswitch.org/wiki/GSMopen#Linux
>>
>> Installed following prerequisites packages
>>
>> # yum -y install alsa-lib-devel alsa-utils
>>
>> then, go to gsmopen directory
>>
>> #/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/gsmlib/gsmlib-1.10-patched-12ubuntu1
>>
>> I wanted to compile only SMS support, no voice call.
>>
>> When, I compiled the source code , I am getting following error ,
>>
>> #make
>> ......
>> ......
>> gsm_unix_serial.cc:89: error: 'strerror' was not declared in this scope
>> make[1]: *** [gsm_unix_serial.lo] Error 1
>> make[1]: Leaving directory
>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/gsmlib/gsmlib-1.10-patched-12ubuntu1/gsmlib'
>> make: *** [install-recursive] Error 1
>>
>> When,? I enter into following directory and trying the compile the source
>> code,
>>
>> #/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen
>>
>> # make
>>
>> I am getting following error
>>
>> gsmopen_protocol.cpp:5:28: error: gsmlib/gsm_sms.h: No such file or
>> directory
>> gsmopen_protocol.cpp:9:36: error: gsmlib/gsm_unix_serial.h: No such file or
>> directory
>> gsmopen_protocol.cpp:11:30: error: gsmlib/gsm_me_ta.h: No such file or
>> directory
>> gsmopen_protocol.cpp:16: error: ?gsmlib? is not a namespace-name
>> ...
>> ...
>> gsmopen_protocol.cpp:3223: error: ?DCS_DEFAULT_ALPHABET? was not declared in
>> this scope
>> gsmopen_protocol.cpp:3225: error: ?DCS_SIXTEEN_BIT_ALPHABET? was not
>> declared in this scope
>> gsmopen_protocol.cpp:3226: error: ?bufToHex? was not declared in this scope
>> make[1]: *** [gsmopen_protocol.o] Error 1
>> make: *** [all] Error 1
>>
>>
>> I do not know, what I am doing wrong.
>> Anyone please help me regarding this.
>>
>>
>> --
>>
>> Regards
>> Venkatesan OV.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From anthony.minessale at gmail.com  Thu Mar 10 20:37:14 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 10 Mar 2011 11:37:14 -0600
Subject: [Freeswitch-users] freeswitch -nc -nosql
In-Reply-To: <AANLkTik6c5rHYnr6cxmsBzRCcc94VL=jmYtud29TVBzc@mail.gmail.com>
References: <1298688216720-6066718.post@n2.nabble.com>
	<1299524988820-6098582.post@n2.nabble.com>
	<AANLkTik6c5rHYnr6cxmsBzRCcc94VL=jmYtud29TVBzc@mail.gmail.com>
Message-ID: <AANLkTindLhrtsbVmifg699TyQtv+JOYa52_Tj1CqkwVs@mail.gmail.com>

try latest it'd fixed now.

integer boundary issue, switch_time_t vs regular time_t

On Mon, Mar 7, 2011 at 1:23 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Will you please explain to me why you keep ignoring my request for backtraces?
> There is a bug open already http://jira.freeswitch.org/browse/FS-3126
> There is even a backtrace which can be very useful , now if you posted
> your findings there we might even figure it out.....
>
> Are you not receiving my many pleas to use JIRA for working on bugs?
>
>
> On Mon, Mar 7, 2011 at 1:09 PM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>> Finaly, I found out that FreeSWITCH Version 1.0.head (git-88d410d 2011-02-11
>> 20-15-06 -0600) is the last version that works with SQL. Starting from the
>> version after this one with a commit
>> a2c0da53f368f0b11340c3a72814c93b182753b7 crashes if -nosql switch is called
>> when freeswitch is launched. Here is the git log pertaining to the two
>> commits for your perusal and hope FS developers will be able to localize why
>> adding centralized registration db to core db and use it from mod_sofia
>> causes SQL to crash on ARM platform:
>>
>> commit a2c0da53f368f0b11340c3a72814c93b182753b7
>> Author: Anthony Minessale
>> Date: ? Fri Feb 11 23:10:12 2011 -0600
>>
>> ? ?add centralized registration db to core db and use it from mod_sofia
>>
>> commit 88d410d31485d13911f0958af5a73f1f6f49a454
>> Author: Anthony Minessale
>> Date: ? Fri Feb 11 20:15:06 2011 -0600
>>
>> ? ?fix uuid_jitterbuffer edge case debugging a non-existant jb causing a
>> seg
>>
>>
>> -----
>> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
>> --
>> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-nc-nosql-tp6066718p6098582.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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 freeswitch at tlainvestments.com  Thu Mar 10 20:51:57 2011
From: freeswitch at tlainvestments.com (Troy Anderson)
Date: Thu, 10 Mar 2011 10:51:57 -0700
Subject: [Freeswitch-users] Call back from voice mail creates loop
In-Reply-To: <AANLkTimg5iRzRSYKpY4uT3O9R+-ifYU09jCCyHDYpmq-@mail.gmail.com>
References: <4CA9ECBB.2000609@communicatefreely.net>
	<AANLkTik24EaqTsfazwBPBZ4UG8eFPTZF-8XFMQ6xNFkb@mail.gmail.com>
	<4CAA313E.3070008@communicatefreely.net>
	<AANLkTimg5iRzRSYKpY4uT3O9R+-ifYU09jCCyHDYpmq-@mail.gmail.com>
Message-ID: <52BE5DAC-1F51-48CA-ABF3-FE65F7EA175F@tlainvestments.com>

I hope this isn't considered "hijacking" a thread as I think this issue is related.  The version of fs I'm using is from Feb 10, 2011, and I looked at the diff for mod_voicemail.c between trunk and the version I have installed, and it doesn't appear that anything related to this issue was modified.  However, I am in the process recompiling with latest to be sure.  In the mean time, perhaps you can help me understand what's going on:

I have been experiencing something odd related to voicemail and pressing 5 to reply.  While researching the problem, this thread seems to suggest the reason, but not an obvious answer.

When you press 5 after listening to a voicemail, the mod_voicemail.c executes switch_core_session_execute_exten(session, cbt->cid_number, profile->callback_dialplan, profile->callback_context);, which I assume is equivalent to execute_extnesion.  The context I'm using is the same context that handles all of my internal extensions, so I expected that when one extension leaves a message for another, pressing 5 after listening to the message would ring back the original extension.  This is not working.

My dialplan has the following condition required before trying any internal extensions :

      <condition field="${user_exists(id ${destination_number} ${domain_name})}" expression="^true$" />

An example may make my question more clear:

I dial from 119 to 105.  119 leaves a message.  Later, 105 dials ** to retrieve its voicemails and listens to the message from 119.  After the message, he dials 5 to return the call.  mod_voicemail runs execute_extension to 119,XML,my_context.  In my_context, I have the following to see what's up:

<extension name="test" continue="true">
<condition break="never" field="destination_number" expression="^119$">
	<action application="log" data="ERR Extension 119 matches"/>
</condition>
<condition break="never" field="${user_exists(id ${destination_number} ${domain_name})}" expression="^true$">
	<action application="log" data="ERR User Exists when using variables"/>
</condition>
<condition break="never" field="${user_exists(id 119 ${domain_name})}" expression="^true$">
	<action application="log" data="ERR User Exists when using plugged value"/>
</condition>
</extension>

The output is:

2011-03-10 10:42:54.720596 [NOTICE] switch_core_session.c:2152 Execute log(ERR Extension 119 matches)
2011-03-10 10:42:54.720596 [ERR] mod_dptools.c:1183 Extension 119 matches
2011-03-10 10:42:54.720596 [NOTICE] switch_core_session.c:2152 Execute log(ERR User Exists when using plugged value)
2011-03-10 10:42:54.720596 [ERR] mod_dptools.c:1183 User Exists when using plugged value

Notice that the second condition fails.  Why is that?  Is it related to the issue identified in this thread?  That execute_extension is somehow setting destination_number differently than transfer?

Also (not shown here), I have applicaiton="info" as the very next condition, and it shows dialed_extension as **, not 119.  I'm very confused about that.

How do I remedy this since mod_voicemail is using execute_extension?  Can I somehow determine this and execute my own transfer?  Or should I somehow modify my user_exists expression?

Thanks for any guidance!

-Troy


On Oct 4, 2010, at 1:55 PM, Michael Collins wrote:

> Yes, transfer is your friend in this scenario. :)
> -MC
> 
> On Mon, Oct 4, 2010 at 12:55 PM, Tim St. Pierre <fs-list at communicatefreely.net> wrote:
> Michael Collins wrote:
> > Are you trying to bridge the current leg (user <--> voicemail) to
> > another endpoint? If so, how are you doing that? Are you transferring
> > the leg back into the dialplan for processing?
> >
> I may have just answered my own question - quite by accident while working on another problem.
> 
> I was sending the call back to the dial plan for processing, but I was using the execute_extension
> application instead of transfer
> 
> It looks like execute_extension doesn't affect the destination_number variable, whereas transfer
> changes the destination number, and sets the previously dialed number as RDNIS.
> 
> Changing my logic to use transfer instead of execute_extension seems to have solved things.
> 
> -Tim
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From 29prime at gmail.com  Thu Mar 10 12:25:42 2011
From: 29prime at gmail.com (Prime)
Date: Thu, 10 Mar 2011 11:25:42 +0200
Subject: [Freeswitch-users] Problem with mapping new DIDs
Message-ID: <AANLkTimJY4OYsGx_znMy-pUUA19oqXT_-qxf=adygLx7@mail.gmail.com>

Hey!

This most likely is a newb mistake but I'm stuck at the moment.

Recently I had to add another provider to FS to handle incoming calls.
And now I have a problem with calling my new provider's DID number on
my FS.
I looked at FS logs.
Usually FS log says something like this " [INFO]
mod_dialplan_xml.c:418 Processing caller_number->My_DID in context
public" and then routes DID number to local extension,
but now I see something like this " [INFO] mod_dialplan_xml.c:418
Processing caller_number->caller_number in context public ".
Why new numbers does not match the regex?

The number are configured exactly as old ones.

Thanks in advance

Best regards


From mitja.thomas1 at ewetel.de  Thu Mar 10 13:19:27 2011
From: mitja.thomas1 at ewetel.de (Mitja Thomas)
Date: Thu, 10 Mar 2011 11:19:27 +0100
Subject: [Freeswitch-users] [freeswitch] <defunct> when using cmd="exec" in
	freeswitch.xml
Message-ID: <4D78A5AF.1000808@ewetel.de>

Hi there,

we tried to set up the FreeSWITCH and other Applications, so that we can 
configure them easier and more centralised.
Thus we defined some Environment Variables (using CentOS) which hold 
often used Configuration Parameter like MySQL IP or FS Event Socket IP.
We tried to integrate these Env Variables into the FS conf files by 
executing a shell Skript in freeswitch.xml via cmd="exec" which prepares 
an conf file which we include into freeswitch.xml:

<X-PRE-PROCESS cmd="exec" data="sh conf/make_my_vars.sh"/>
<X-PRE-PROCESS cmd="include" data="my_vars.xml"/>

This works as expected and the pre defined variables in my_vars can be 
accessed from the other config Files, except that when we start our 
FreeSWITCH  a zombie child process is spawned.

# ps -eaf | grep free
ippbx    22191 22190  4 09:41 pts/1    00:00:01 
/opt/app/voip/ippbx/bin/freeswitch -waste -nonat -hp
ippbx    22197 22191  0 09:41 pts/1    00:00:00 [freeswitch] <defunct>

What I wanna know is: Is this a FS missbehaviour or do we use this in a 
wrong way?

make_my_vars.sh:
F="conf/my_vars.xml"
echo "<!-- Auto generated file for getting *NIX evironment variables 
-->" > $F
echo "<include>" >> $F

fs_ip=`printenv MY_FS_IP`
if test -n "$fs_ip"
then
         echo '<X-PRE-PROCESS cmd="set" data="my_fs_ip='$fs_ip'"/>' >> $F
fi
...
echo "</include>" >> $F

my_vars.xml (after FS startup):
<!-- Auto generated file for getting *NIX evironment variables -->
<include>
<X-PRE-PROCESS cmd="set" data="my_fs_ip=***.***.***.***"/>
...
</include>

Regards
Mitja
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From msc at freeswitch.org  Thu Mar 10 22:51:04 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 10 Mar 2011 11:51:04 -0800
Subject: [Freeswitch-users] count
In-Reply-To: <AANLkTimWrn1HHbhQj5CK2Mhk64h-d5fGbGH4E-Mpj0i_@mail.gmail.com>
References: <AANLkTi=oKxBtho8q5kQq52ZxrU=r++SzFXkVzjq=P8B1@mail.gmail.com>
	<AANLkTimWrn1HHbhQj5CK2Mhk64h-d5fGbGH4E-Mpj0i_@mail.gmail.com>
Message-ID: <AANLkTinWKn8jEUt11m3oiiOzxqB9oLtavOsp9WOXkaDX@mail.gmail.com>

conference list <conference_name> will list them all
You can also get the list in XML format:
conference xml_list <conference_name>

-MC

On Wed, Mar 9, 2011 at 1:43 AM, Sam <u2nsam at gmail.com> wrote:

> Hello,
>
> What is the command to fetch the member id of the members in conference ?
>
> Regards
> Sam
>
>
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From kapil.rastogi at telemune.net  Thu Mar 10 23:40:04 2011
From: kapil.rastogi at telemune.net (kapil.rastogi)
Date: Thu, 10 Mar 2011 12:40:04 -0800 (PST)
Subject: [Freeswitch-users] Conference silent issue for more than 2 callers
Message-ID: <1299789604046-6159247.post@n2.nabble.com>

Hi,

Login with 5 subscribers. 1st subscriber logs in, 2nd Subscriber log in and
listen each other, 3rd subscribe and it is not listened and can?t listen
others. 4th as 3rd and 5th as 3rd and 4th.

Currently i m using freeswitch with diastar, that create a bridge between
dialogic and freeswitch. Please help me to resolve the issue.

Thanks

-----
Regards,
Kapil Rastogi
Telemune Software Solutions P Ltd.
kapil.rastogi at telemune.net
+919013204760
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Conference-silent-issue-for-more-than-2-callers-tp6159247p6159247.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brian at freeswitch.org  Thu Mar 10 23:49:09 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 10 Mar 2011 14:49:09 -0600
Subject: [Freeswitch-users] Conference silent issue for more than 2
	callers
In-Reply-To: <1299789604046-6159247.post@n2.nabble.com>
References: <1299789604046-6159247.post@n2.nabble.com>
Message-ID: <21117897-2827-4B9F-AE55-42DADA92D69A@freeswitch.org>

what rev of FreeSWITCH are you using?

/b

On Mar 10, 2011, at 2:40 PM, kapil.rastogi wrote:

> 
> Currently i m using freeswitch with diastar, that create a bridge between
> dialogic and freeswitch. Please help me to resolve the issue.



From mario_fs at mgtech.com  Fri Mar 11 00:13:03 2011
From: mario_fs at mgtech.com (Mario G)
Date: Thu, 10 Mar 2011 13:13:03 -0800
Subject: [Freeswitch-users] Internal extensions placing call on hold drops
	immediately
Message-ID: <5FD88FAF-068C-4DF8-BEC3-736268A05940@mgtech.com>

I don't know when this started but was working fine at one time. If a call is made between extensions and either extension places the call on hold, it states "trying to hold". I installed the git from 3/9/11 before posting here. Could this be a Sofia problem (see FREEPBX comment) or something that FS needs to "setup" the phone to work since it works fine using a different PBX? Any help appreciated!

* The phones are Linksys SPA962
* If I plug the old SPA9000 in as a PBX hold works fine for the SPA9000 extensions, only FS has the issue
* All connected via HP Procurve 2520G switch
* Outside calls are also affected but this is still a problem even with the router  disconnected
* I can ping (ICMP) from FS to the phone and from anywhere to the phone

FS TRACE:
I started several FS traces: sip global,internal, log 9 console debug, etc. It looked like NOTHING was sent when the hold button was pressed.

WIRESHARK showed these messages stacking up when the hold is pressed
src 1.123.1.23 (phone)   dest 1.123.1.7 (FS)  Proto ICMP  Destination unreachable (Port unreachable)

Someone updated FREEPBX in FEB 2011 and had the same issue with SPA phones.
http://www.freepbx.org/forum/freepbx/users/spa-941-942-drop-call-when-placed-on-hold-also-have-to-click-submit-before-syste

From anthony.minessale at gmail.com  Fri Mar 11 01:49:09 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 10 Mar 2011 16:49:09 -0600
Subject: [Freeswitch-users] Internal extensions placing call on hold
	drops immediately
In-Reply-To: <5FD88FAF-068C-4DF8-BEC3-736268A05940@mgtech.com>
References: <5FD88FAF-068C-4DF8-BEC3-736268A05940@mgtech.com>
Message-ID: <AANLkTimW=zaLRsiLqBF_r6jHLzXeC9sns_xqk6rHTduR@mail.gmail.com>

try latest git again to be sure.  There was a patch in that area of
the code last night.


On Thu, Mar 10, 2011 at 3:13 PM, Mario G <mario_fs at mgtech.com> wrote:
> I don't know when this started but was working fine at one time. If a call is made between extensions and either extension places the call on hold, it states "trying to hold". I installed the git from 3/9/11 before posting here. Could this be a Sofia problem (see FREEPBX comment) or something that FS needs to "setup" the phone to work since it works fine using a different PBX? Any help appreciated!
>
> * The phones are Linksys SPA962
> * If I plug the old SPA9000 in as a PBX hold works fine for the SPA9000 extensions, only FS has the issue
> * All connected via HP Procurve 2520G switch
> * Outside calls are also affected but this is still a problem even with the router ?disconnected
> * I can ping (ICMP) from FS to the phone and from anywhere to the phone
>
> FS TRACE:
> I started several FS traces: sip global,internal, log 9 console debug, etc. It looked like NOTHING was sent when the hold button was pressed.
>
> WIRESHARK showed these messages stacking up when the hold is pressed
> src 1.123.1.23 (phone) ? dest 1.123.1.7 (FS) ?Proto ICMP ?Destination unreachable (Port unreachable)
>
> Someone updated FREEPBX in FEB 2011 and had the same issue with SPA phones.
> http://www.freepbx.org/forum/freepbx/users/spa-941-942-drop-call-when-placed-on-hold-also-have-to-click-submit-before-syste
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 sunwood360 at gmail.com  Fri Mar 11 02:34:27 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Thu, 10 Mar 2011 15:34:27 -0800
Subject: [Freeswitch-users] freeswitch -nc -nosql
In-Reply-To: <AANLkTindLhrtsbVmifg699TyQtv+JOYa52_Tj1CqkwVs@mail.gmail.com>
References: <1298688216720-6066718.post@n2.nabble.com>
	<1299524988820-6098582.post@n2.nabble.com>
	<AANLkTik6c5rHYnr6cxmsBzRCcc94VL=jmYtud29TVBzc@mail.gmail.com>
	<AANLkTindLhrtsbVmifg699TyQtv+JOYa52_Tj1CqkwVs@mail.gmail.com>
Message-ID: <AANLkTi=O=ZpYt_3MDwB2t0jHDG3DQ0qgx4apA+1WUGyM@mail.gmail.com>

Awesome, I will try my debian box tonight.
On Mar 10, 2011 9:38 AM, "Anthony Minessale" <anthony.minessale at gmail.com>
wrote:
> try latest it'd fixed now.
>
> integer boundary issue, switch_time_t vs regular time_t
>
> On Mon, Mar 7, 2011 at 1:23 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> Will you please explain to me why you keep ignoring my request for
backtraces?
>> There is a bug open already http://jira.freeswitch.org/browse/FS-3126
>> There is even a backtrace which can be very useful , now if you posted
>> your findings there we might even figure it out.....
>>
>> Are you not receiving my many pleas to use JIRA for working on bugs?
>>
>>
>> On Mon, Mar 7, 2011 at 1:09 PM, mazilo <Nabble at slickdeals.endjunk.com>
wrote:
>>> Finaly, I found out that FreeSWITCH Version 1.0.head (git-88d410d
2011-02-11
>>> 20-15-06 -0600) is the last version that works with SQL. Starting from
the
>>> version after this one with a commit
>>> a2c0da53f368f0b11340c3a72814c93b182753b7 crashes if -nosql switch is
called
>>> when freeswitch is launched. Here is the git log pertaining to the two
>>> commits for your perusal and hope FS developers will be able to localize
why
>>> adding centralized registration db to core db and use it from mod_sofia
>>> causes SQL to crash on ARM platform:
>>>
>>> commit a2c0da53f368f0b11340c3a72814c93b182753b7
>>> Author: Anthony Minessale
>>> Date:   Fri Feb 11 23:10:12 2011 -0600
>>>
>>>    add centralized registration db to core db and use it from mod_sofia
>>>
>>> commit 88d410d31485d13911f0958af5a73f1f6f49a454
>>> Author: Anthony Minessale
>>> Date:   Fri Feb 11 20:15:06 2011 -0600
>>>
>>>    fix uuid_jitterbuffer edge case debugging a non-existant jb causing a
>>> seg
>>>
>>>
>>> -----
>>> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
>>> --
>>> View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/freeswitch-nc-nosql-tp6066718p6098582.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From marcdecorny at gmail.com  Fri Mar 11 11:08:24 2011
From: marcdecorny at gmail.com (Marc de Corny)
Date: Fri, 11 Mar 2011 08:08:24 +0000
Subject: [Freeswitch-users] Limit the size of queue in mod_fifo
In-Reply-To: <AANLkTimFGo0xrb07a1b9SOoije-rkamGua8tViUuwWz9@mail.gmail.com>
References: <AANLkTinU_j8zRqKWvVfp=DXedPP83dSx09KRUbgo-rKb@mail.gmail.com>
	<AANLkTimFGo0xrb07a1b9SOoije-rkamGua8tViUuwWz9@mail.gmail.com>
Message-ID: <AANLkTikHKbvjZv+x9M4ththG5=w--=-FACtLFKNAV6hQ@mail.gmail.com>

Thanks afshin,

I had actually looked at that, and it seems to limit the number of calls to
a destination, which is fine. Im loooking specifically to limit the number
of simultaneous calls, so I would need that figure to increase and decrease
as calls are initiated and cleared.

I'm looking at this specifically as the depth of the queue in mod_fifo, so
that I can stop a user having 100 calls all queuing.

Any ideas?

thanks
Marc

On Tue, Mar 8, 2011 at 3:58 PM, afshin afzali <a.afzali2003 at gmail.com>wrote:

> Maybe mod_limit be a solution
>
> -- afshin
>
>   On Mon, Mar 7, 2011 at 4:06 PM, Marc de Corny <marcdecorny at gmail.com>wrote:
>
>>   Hi All,
>>
>> does anyone know of a way of limiting the numbers of calls that are held
>> in the mod_fifo queue so that we can deny more calls after having reacded
>> that limit ?
>>
>> thanks
>> Marc
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From pkelly at gmail.com  Fri Mar 11 12:04:58 2011
From: pkelly at gmail.com (Pete Kelly)
Date: Fri, 11 Mar 2011 09:04:58 +0000
Subject: [Freeswitch-users] exec app on originate
In-Reply-To: <AANLkTikmsGve+DMf9RzyyB3wBH0dgpT_1gkG4hNuq7du@mail.gmail.com>
References: <748093B885AD48CE86278841FBEA1159@e1705>
	<AANLkTikmsGve+DMf9RzyyB3wBH0dgpT_1gkG4hNuq7du@mail.gmail.com>
Message-ID: <AANLkTinoN7tdVDA8e+XLcjFGV-Z1KhP1a3+gLSGNaGOn@mail.gmail.com>

On 25 February 2011 01:04, Anthony Minessale <anthony.minessale at gmail.com>wrote:

> early media
> if you want after answer do
>
> originate {ignore_early_media=true}sofia/gateway/blabla/9999999999
> &app(hahaha)
>

Is there any way to get the app launched even earlier? I need to be able to
act upon a 4XX/5XX class response to the INVITE within the app.


>
>
> On Thu, Feb 24, 2011 at 7:02 PM, Madovsky <infos at madovsky.org> wrote:
> > when I do this :
> > originate /sofia/gateway/blabla/9999999999 &app(hahaha)
> >
> > is the ap executed at early media or after answer ?
> >
> > Thanks
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From laurentiu.ceausescu at gmail.com  Fri Mar 11 12:25:28 2011
From: laurentiu.ceausescu at gmail.com (Laurentiu Ceausescu)
Date: Fri, 11 Mar 2011 11:25:28 +0200
Subject: [Freeswitch-users] Error! cannot get read lock
In-Reply-To: <AANLkTi=nVN4+b1vi__rSU5PeWhU6vHKip5c4Eh1vZV+F@mail.gmail.com>
References: <AANLkTimKdKafaD4tzko68jb4zXd_djT+ZiaK4Jz_YCzT@mail.gmail.com>
	<AANLkTimwfAFbJu8HxLxp2eoAznoR3bYzoB+75eea4+AX@mail.gmail.com>
	<AANLkTi=nVN4+b1vi__rSU5PeWhU6vHKip5c4Eh1vZV+F@mail.gmail.com>
Message-ID: <AANLkTindVPUAeoquS5TKcD-znYRCoOuA4B+XN=YBOaqH@mail.gmail.com>

On Thu, Mar 10, 2011 at 5:24 AM, Seven Du <dujinfang at gmail.com> wrote:
> I run into this short time ago and rebuilt mod_erlang_event fixed.
>
> cd freeswitch-source-tree
> make mod_erlang_event-install
>
> fs_cli> reload mod_erlang_event

Unfortunately it doesn't work for me ...
I've opened http://jira.freeswitch.org/browse/FS-3145

Thanks,
Laurentiu


> On Thu, Mar 10, 2011 at 10:26 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> you should open a JIRA under mod_erlang_event to find out.
>>
>>
>> On Wed, Mar 9, 2011 at 3:48 PM, Laurentiu Ceausescu
>> <laurentiu.ceausescu at gmail.com> wrote:
>>> Hi,
>>>
>>> I have the following dialplan:
>>> ?<extension name="line1">
>>> ? ?<condition field="destination_number" expression="^333$">
>>> ? ? ?<action application="answer"/>
>>> ? ? ?<action application="set" data="allow_voicemail=true"/>
>>> ? ? ?<action application="set" data="queue=Queue"/>
>>> ? ? ?<action application="set" data="brand=110011"/>
>>> ? ? ?<action application="playback" data="/usr/local/audio/welcome.wav"/>
>>> ? ? ?<action application="set" data="domain_name=$${domain}"/>
>>> ? ? ?<action application="erlang_sendmsg"
>>> data="freeswitch_media_manager toor at 127.0.0.1 inivr ${uuid}"/>
>>> ? ? ?<action application="erlang" data="freeswitch_media_manager:!
>>> toor at 127.0.0.1"/>
>>> ? ?</condition>
>>> ?</extension>
>>>
>>> When I call 333 I can hear welcome message but then the call is dropped.
>>> FS logs shows this error: [ERR] handle_msg.c:56 Error! cannot get read lock.
>>>
>>> [NOTICE] switch_channel.c:812 New Channel
>>> sofia/toor.mydomain.org/401 at toor.mydomain.org
>>> [03ecb2ca-24df-4903-917a-c9720b43a511]
>>> [NOTICE] sofia_glue.c:3754 Pre-Answer
>>> sofia/toor.mydomain.org/401 at toor.mydomain.org!
>>> [NOTICE] mod_dptools.c:929 Channel
>>> [sofia/toor.mydomain.org/401 at toor.mydomain.org] has been answered
>>> [ERR] handle_msg.c:56 Error! cannot get read lock.
>>> [WARNING] mod_erlang_event.c:1365 Timed out when waiting for outbound
>>> pid 1.0.0 at freeswitch@127.0.0.1 03ecb2ca-24df-4903-917a-c
>>> [NOTICE] switch_core_state_machine.c:189
>>> sofia/toor.mydomain.org/401 at toor.mydomain.org has executed the last
>>> dialplan instruct
>>> [NOTICE] switch_core_state_machine.c:191 Hangup
>>> sofia/toor.mydomain.org/401 at toor.mydomain.org [CS_EXECUTE]
>>> [NORMAL_CLEARING]
>>> [NOTICE] switch_core_session.c:1306 Session 4
>>> (sofia/toor.mydomain.org/401 at toor.mydomain.org) Ended
>>> [NOTICE] switch_core_session.c:1308 Close Channel
>>> sofia/toor.mydomain.org/401 at toor.mydomain.org [CS_DESTROY]
>>> [ERR] handle_msg.c:56 Error! cannot get read lock.
>>>
>>>
>>> This thing is happend only after I updated to last git head
>>> (419d7e2335042e2d86c678c0daf86c2d9c867036 /Mar 9).
>>> With a older revision (e88b9639624cef4f35901146241f515730b3b118 /Jan
>>> 31) everything works fine.
>>> Any idea what has been changed?
>>>
>>> Thanks,
>>> Laurentiu
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:? http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From kbdfck at gmail.com  Fri Mar 11 13:05:58 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Fri, 11 Mar 2011 13:05:58 +0300
Subject: [Freeswitch-users] Already hung channels in 'show channels' output
	never go away
Message-ID: <AANLkTi=jiL-aSmkDusoH_9EYRzPZ1ePXkWAQHXn53Np5@mail.gmail.com>

Hi All
I'm using Perl ESL outbound script to bridge incoming call to sip endpoint.
I'm doing execute("bridge","sofia/user/somebody"), then processing events
like CHANNEL_HANGUP and EXECUTE_COMPLETE to determine bridge result.
Everything works fine, but original incoming call channel is never removed
from list:
After few calls I see original incoming channels in 'show channels' output:

7ae28028-4bc2-11e0-a92f-452f3d3d66ea,inbound,2011-03-11
12:32:27,1299835947,sofia/external/xxxx at 85.x.x.199
,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,
xxx.xxx.ru,,,HANGUP,,,,
883c9eb6-4bc2-11e0-a935-452f3d3d66ea,inbound,2011-03-11
12:32:50,1299835970,sofia/external/xxxx at 85.x.x.199
,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,
xxx.xxx.ru,,,HANGUP,,,,
9f6a0f92-4bc2-11e0-a93b-452f3d3d66ea,inbound,2011-03-11
12:33:29,1299836009,sofia/external/xxxx at 85.x.x.199
,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,
xxx.xxx.ru,,,HANGUP,,,

Also, when I try to stop freeswitch i see these messages on console:

2011-03-11 15:28:22.517063 [CRIT] sofia.c:1632 Waiting for 13 session(s)
2011-03-11 15:28:22.701034 [CRIT] sofia.c:1632 Waiting for 3 session(s)
2011-03-11 15:28:23.517064 [CRIT] sofia.c:1632 Waiting for 13 session(s)
2011-03-11 15:28:23.701033 [CRIT] sofia.c:1632 Waiting for 3 session(s)
2011-03-11 15:28:24.517062 [CRIT] sofia.c:1632 Waiting for 13 session(s)
2011-03-11 15:28:24.705036 [CRIT] sofia.c:1632 Waiting for 3 session(s)
2011-03-11 15:28:25.517064 [CRIT] sofia.c:1632 Waiting for 13 session(s)
2011-03-11 15:28:25.709031 [CRIT] sofia.c:1632 Waiting for 3 session(s)
2011-03-11 15:28:26.517065 [CRIT] sofia.c:1632 Waiting for 13 session(s)

Why these channels are not removed from list? I also noticed that memory
consumption by freeswitch process constantly grows call by call.

-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From kbdfck at gmail.com  Fri Mar 11 13:33:28 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Fri, 11 Mar 2011 13:33:28 +0300
Subject: [Freeswitch-users] Already hung channels in 'show channels'
	output never go away
In-Reply-To: <AANLkTi=jiL-aSmkDusoH_9EYRzPZ1ePXkWAQHXn53Np5@mail.gmail.com>
References: <AANLkTi=jiL-aSmkDusoH_9EYRzPZ1ePXkWAQHXn53Np5@mail.gmail.com>
Message-ID: <AANLkTinUg76_ncbjCwf=W8T5BPSS3FdstGQkkptv-xme@mail.gmail.com>

BTW, uuid_dump 7ae28028-4bc2-11e0-a92f-452f3d3d66ea returns with -ERR No
such channel.

2011/3/11 Dmitry Sytchev <kbdfck at gmail.com>

> Hi All
> I'm using Perl ESL outbound script to bridge incoming call to sip endpoint.
> I'm doing execute("bridge","sofia/user/somebody"), then processing events
> like CHANNEL_HANGUP and EXECUTE_COMPLETE to determine bridge result.
> Everything works fine, but original incoming call channel is never removed
> from list:
> After few calls I see original incoming channels in 'show channels' output:
>
> 7ae28028-4bc2-11e0-a92f-452f3d3d66ea,inbound,2011-03-11
> 12:32:27,1299835947,sofia/external/xxxx at 85.x.x.199
> ,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,
> xxx.xxx.ru,,,HANGUP,,,,
> 883c9eb6-4bc2-11e0-a935-452f3d3d66ea,inbound,2011-03-11
> 12:32:50,1299835970,sofia/external/xxxx at 85.x.x.199
> ,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,
> xxx.xxx.ru,,,HANGUP,,,,
> 9f6a0f92-4bc2-11e0-a93b-452f3d3d66ea,inbound,2011-03-11
> 12:33:29,1299836009,sofia/external/xxxx at 85.x.x.199
> ,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,
> xxx.xxx.ru,,,HANGUP,,,
>
> Also, when I try to stop freeswitch i see these messages on console:
>
> 2011-03-11 15:28:22.517063 [CRIT] sofia.c:1632 Waiting for 13 session(s)
> 2011-03-11 15:28:22.701034 [CRIT] sofia.c:1632 Waiting for 3 session(s)
> 2011-03-11 15:28:23.517064 [CRIT] sofia.c:1632 Waiting for 13 session(s)
> 2011-03-11 15:28:23.701033 [CRIT] sofia.c:1632 Waiting for 3 session(s)
> 2011-03-11 15:28:24.517062 [CRIT] sofia.c:1632 Waiting for 13 session(s)
> 2011-03-11 15:28:24.705036 [CRIT] sofia.c:1632 Waiting for 3 session(s)
> 2011-03-11 15:28:25.517064 [CRIT] sofia.c:1632 Waiting for 13 session(s)
> 2011-03-11 15:28:25.709031 [CRIT] sofia.c:1632 Waiting for 3 session(s)
> 2011-03-11 15:28:26.517065 [CRIT] sofia.c:1632 Waiting for 13 session(s)
>
> Why these channels are not removed from list? I also noticed that memory
> consumption by freeswitch process constantly grows call by call.
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>



-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From sid.kshatriya at gmail.com  Thu Mar 10 23:21:55 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Fri, 11 Mar 2011 01:51:55 +0530
Subject: [Freeswitch-users] Strange Behavior with Channel Variable
 [session:setVariable() vs <action application="set" ... /> ]
Message-ID: <AANLkTin2+5TAVHEbTTQDKwoqqfWFJLruoyD_FiffptaP@mail.gmail.com>

Dear All,

I am using a little known but very useful channel variable called
"playback_timeout_sec"

In Lua if I do: session:setVariable("playback_timeout_sec", "10")

and subsequently in a macro I have:

<action function="play-file" date="/some/filename.mp3" />

Then the filename.mp3 stops playing automatically after 10 seconds. This is
very useful if you want to play short excerpts of a file that could be very
long.

But if I do:

<action application="set" date="playback_timeout_sec=10" />
<action function="play-file" date="/some/filename.mp3" />

My file does NOT stop playing after 10 seconds. This is weird given that
<action application="set" .. /> is doing exactly the same thing as
session:setVariable() ... right?

My question
1. What could be the problem with application set?
2. Is there a way to automatically stop playback after a certain number of
seconds for a file? (without doing tedious things like create a separate
10second version of an mp3 using ffmpeg etc.)

Thanks,

Sidharth
-- 
Sidharth Kshatriya
www.sidk.info
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From avi at avimarcus.net  Fri Mar 11 14:16:56 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 11 Mar 2011 13:16:56 +0200
Subject: [Freeswitch-users] Already hung channels in 'show channels'
 output never go away
In-Reply-To: <AANLkTinUg76_ncbjCwf=W8T5BPSS3FdstGQkkptv-xme@mail.gmail.com>
References: <AANLkTi=jiL-aSmkDusoH_9EYRzPZ1ePXkWAQHXn53Np5@mail.gmail.com>
	<AANLkTinUg76_ncbjCwf=W8T5BPSS3FdstGQkkptv-xme@mail.gmail.com>
Message-ID: <AANLkTimzvf-ZaiaVKCprbHh6JcFx3dXnbaU54T8KL=O6@mail.gmail.com>

Regarding ram usage, I'd imagine this is the case:
http://www.linuxatemyram.com/
 <http://www.linuxatemyram.com/>-Avi

On Fri, Mar 11, 2011 at 12:33 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:

> BTW, uuid_dump 7ae28028-4bc2-11e0-a92f-452f3d3d66ea returns with -ERR No
> such channel.
>
> 2011/3/11 Dmitry Sytchev <kbdfck at gmail.com>
>
> Hi All
>> I'm using Perl ESL outbound script to bridge incoming call to sip
>> endpoint. I'm doing execute("bridge","sofia/user/somebody"), then processing
>> events like CHANNEL_HANGUP and EXECUTE_COMPLETE to determine bridge result.
>> Everything works fine, but original incoming call channel is never removed
>> from list:
>> After few calls I see original incoming channels in 'show channels'
>> output:
>>
>> 7ae28028-4bc2-11e0-a92f-452f3d3d66ea,inbound,2011-03-11
>> 12:32:27,1299835947,sofia/external/xxxx at 85.x.x.199
>> ,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,
>> xxx.xxx.ru,,,HANGUP,,,,
>> 883c9eb6-4bc2-11e0-a935-452f3d3d66ea,inbound,2011-03-11
>> 12:32:50,1299835970,sofia/external/xxxx at 85.x.x.199
>> ,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,
>> xxx.xxx.ru,,,HANGUP,,,,
>> 9f6a0f92-4bc2-11e0-a93b-452f3d3d66ea,inbound,2011-03-11
>> 12:33:29,1299836009,sofia/external/xxxx at 85.x.x.199
>> ,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,
>> xxx.xxx.ru,,,HANGUP,,,
>>
>> Also, when I try to stop freeswitch i see these messages on console:
>>
>> 2011-03-11 15:28:22.517063 [CRIT] sofia.c:1632 Waiting for 13 session(s)
>> 2011-03-11 15:28:22.701034 [CRIT] sofia.c:1632 Waiting for 3 session(s)
>> 2011-03-11 15:28:23.517064 [CRIT] sofia.c:1632 Waiting for 13 session(s)
>> 2011-03-11 15:28:23.701033 [CRIT] sofia.c:1632 Waiting for 3 session(s)
>> 2011-03-11 15:28:24.517062 [CRIT] sofia.c:1632 Waiting for 13 session(s)
>> 2011-03-11 15:28:24.705036 [CRIT] sofia.c:1632 Waiting for 3 session(s)
>> 2011-03-11 15:28:25.517064 [CRIT] sofia.c:1632 Waiting for 13 session(s)
>> 2011-03-11 15:28:25.709031 [CRIT] sofia.c:1632 Waiting for 3 session(s)
>> 2011-03-11 15:28:26.517065 [CRIT] sofia.c:1632 Waiting for 13 session(s)
>>
>> Why these channels are not removed from list? I also noticed that memory
>> consumption by freeswitch process constantly grows call by call.
>>
>> --
>> Best regards,
>>
>> Dmitry Sytchev,
>> IT Engineer
>>
>
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peter.olsson at visionutveckling.se  Fri Mar 11 14:23:24 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 11 Mar 2011 12:23:24 +0100
Subject: [Freeswitch-users] Strange Behavior with Channel Variable
 [session:setVariable() vs <action application="set" ... /> ]
In-Reply-To: <AANLkTin2+5TAVHEbTTQDKwoqqfWFJLruoyD_FiffptaP@mail.gmail.com>
References: <AANLkTin2+5TAVHEbTTQDKwoqqfWFJLruoyD_FiffptaP@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2F6BF36@cooper>

Is this a delayed email? I think this was resolved yesterday, right?

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Sidharth Kshatriya
Skickat: den 10 mars 2011 21:22
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] Strange Behavior with Channel Variable [session:setVariable() vs <action application="set" ... /> ]

Dear All,

I am using a little known but very useful channel variable called "playback_timeout_sec"

In Lua if I do: session:setVariable("playback_timeout_sec", "10")

and subsequently in a macro I have:

<action function="play-file" date="/some/filename.mp3" />

Then the filename.mp3 stops playing automatically after 10 seconds. This is very useful if you want to play short excerpts of a file that could be very long.

But if I do:

<action application="set" date="playback_timeout_sec=10" />
<action function="play-file" date="/some/filename.mp3" />

My file does NOT stop playing after 10 seconds. This is weird given that <action application="set" .. /> is doing exactly the same thing as session:setVariable() ... right?

My question
1. What could be the problem with application set?
2. Is there a way to automatically stop playback after a certain number of seconds for a file? (without doing tedious things like create a separate 10second version of an mp3 using ffmpeg etc.)

Thanks,

Sidharth
--
Sidharth Kshatriya
www.sidk.info<http://www.sidk.info>
!DSPAM:4d7a03ab32764871212382!
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From julf at julf.com  Fri Mar 11 14:34:55 2011
From: julf at julf.com (Johan Helsingius)
Date: Fri, 11 Mar 2011 12:34:55 +0100
Subject: [Freeswitch-users] tones.conf parameters?
In-Reply-To: <AANLkTiksxcOwtm4Dz-6HEqFtfNbwsNdta1shwfqgHgOt@mail.gmail.com>
References: <4D7503F5.6010606@julf.com>	<AANLkTin0N+79ZOQc8pMHw4U+CRS0Rc+LQ27YiPEuo0Bp@mail.gmail.com>	<4D751D6A.70605@julf.com>	<AANLkTikV1oG9-45b5mhsk5Xk3vXq+WGuFeOoZWPg8166@mail.gmail.com>	<4D760D8E.50505@julf.com>
	<AANLkTiksxcOwtm4Dz-6HEqFtfNbwsNdta1shwfqgHgOt@mail.gmail.com>
Message-ID: <4D7A08DF.5020901@julf.com>

Jeroen,

> I'm not sure I understand correclty, but does this help with the needed tones?
> 
> http://vxlabs.com/2011/02/05/sipura-linksys-cisco-spa3102-voice-gateway-in-the-netherlands/

Definitely! Many thanks!

	Julf


From julf at julf.com  Fri Mar 11 14:45:33 2011
From: julf at julf.com (Johan Helsingius)
Date: Fri, 11 Mar 2011 12:45:33 +0100
Subject: [Freeswitch-users] Another beginner question: garbled voicemail
Message-ID: <4D7A0B5D.6030803@julf.com>

Hi!

For some reason the recorded voicemail messages come out as random noise.
All the system messages sound fine, it's just that the recorded message
is pretty much random garbage.

Any hints for where to start looking?

	Julf


From acrow at integrafin.co.uk  Fri Mar 11 14:52:57 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Fri, 11 Mar 2011 11:52:57 +0000
Subject: [Freeswitch-users] British sounds
In-Reply-To: <4D6E6178.8040205@integrafin.co.uk>
References: <4D681972.7050400@integrafin.co.uk>	<AANLkTinGQk18xc8WCap6VkyTVYM1-XHcPi1PhwVTNs8B@mail.gmail.com>	<4D68222C.8010108@integrafin.co.uk>	<4D6D0366.7020205@integrafin.co.uk>	<AANLkTi=pz1GgEz9OLvW_7=QtO=B=nWiHS+4G6D7MY=-N@mail.gmail.com>
	<4D6E6178.8040205@integrafin.co.uk>
Message-ID: <4D7A0D19.3090608@integrafin.co.uk>

On 02/03/11 15:25, Alex Crow wrote:
> On 01/03/11 18:27, Michael Collins wrote:
>> Okay, this is good. I recommend getting a spreadsheet together and 
>> creating columns for Ast file name, Ast prompt, FS file name, FS 
>> prompt. The prompts would contain the actual text of what is voiced 
>> in the audio file. On the FS side look in 
>> $FS_SOURCE/docs/phrase/phrase_en.xml for the phrases and file names. 
>> In Asterisk there usually is a text file that has the filename and 
>> the text of the prompt.
>>
>> Once we have that together we can figure out what prompts we can 
>> re-use versus what needs to be re-recorded.
>>
>> -MC
>>
>
> Michael,
>
> I have attached (hope the list allows it) an ods with the info you 
> have asked for. The * names appear to be unique, but the FS one are 
> not always so I've inserted a type column for these.
>
> Cheers
>
> Alex
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

I've pasted the two sets on the same sheet, sadly it looks like most of 
them do not match.

It is looking like it might be easier just to get all of these re-recorded.

Cheers

Alex

-- 
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From kbdfck at gmail.com  Fri Mar 11 14:58:37 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Fri, 11 Mar 2011 14:58:37 +0300
Subject: [Freeswitch-users] Already hung channels in 'show channels'
 output never go away
In-Reply-To: <AANLkTimzvf-ZaiaVKCprbHh6JcFx3dXnbaU54T8KL=O6@mail.gmail.com>
References: <AANLkTi=jiL-aSmkDusoH_9EYRzPZ1ePXkWAQHXn53Np5@mail.gmail.com>
	<AANLkTinUg76_ncbjCwf=W8T5BPSS3FdstGQkkptv-xme@mail.gmail.com>
	<AANLkTimzvf-ZaiaVKCprbHh6JcFx3dXnbaU54T8KL=O6@mail.gmail.com>
Message-ID: <AANLkTinTApDm-YL5UjqPnc8vaE-e+xpdAXiwaD+DTZhX@mail.gmail.com>

I think this is not the case, but anyway, what to do with these hung
channels?
Maybe I'm doing something wrong while bridging or processing events? Maybe
unprocessed events can affect channel destroy procedure?

2011/3/11 Avi Marcus <avi at avimarcus.net>

> Regarding ram usage, I'd imagine this is the case:
> http://www.linuxatemyram.com/
> -Avi
>
> On Fri, Mar 11, 2011 at 12:33 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>
>> BTW, uuid_dump 7ae28028-4bc2-11e0-a92f-452f3d3d66ea returns with -ERR No
>> such channel.
>>
>> 2011/3/11 Dmitry Sytchev <kbdfck at gmail.com>
>>
>> Hi All
>>> I'm using Perl ESL outbound script to bridge incoming call to sip
>>> endpoint. I'm doing execute("bridge","sofia/user/somebody"), then processing
>>> events like CHANNEL_HANGUP and EXECUTE_COMPLETE to determine bridge result.
>>> Everything works fine, but original incoming call channel is never removed
>>> from list:
>>> After few calls I see original incoming channels in 'show channels'
>>> output:
>>>
>>> 7ae28028-4bc2-11e0-a92f-452f3d3d66ea,inbound,2011-03-11
>>> 12:32:27,1299835947,sofia/external/xxxx at 85.x.x.199
>>> ,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,
>>> xxx.xxx.ru,,,HANGUP,,,,
>>> 883c9eb6-4bc2-11e0-a935-452f3d3d66ea,inbound,2011-03-11
>>> 12:32:50,1299835970,sofia/external/xxxx at 85.x.x.199
>>> ,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,
>>> xxx.xxx.ru,,,HANGUP,,,,
>>> 9f6a0f92-4bc2-11e0-a93b-452f3d3d66ea,inbound,2011-03-11
>>> 12:33:29,1299836009,sofia/external/xxxx at 85.x.x.199
>>> ,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,
>>> xxx.xxx.ru,,,HANGUP,,,
>>>
>>> Also, when I try to stop freeswitch i see these messages on console:
>>>
>>> 2011-03-11 15:28:22.517063 [CRIT] sofia.c:1632 Waiting for 13 session(s)
>>> 2011-03-11 15:28:22.701034 [CRIT] sofia.c:1632 Waiting for 3 session(s)
>>> 2011-03-11 15:28:23.517064 [CRIT] sofia.c:1632 Waiting for 13 session(s)
>>> 2011-03-11 15:28:23.701033 [CRIT] sofia.c:1632 Waiting for 3 session(s)
>>> 2011-03-11 15:28:24.517062 [CRIT] sofia.c:1632 Waiting for 13 session(s)
>>> 2011-03-11 15:28:24.705036 [CRIT] sofia.c:1632 Waiting for 3 session(s)
>>> 2011-03-11 15:28:25.517064 [CRIT] sofia.c:1632 Waiting for 13 session(s)
>>> 2011-03-11 15:28:25.709031 [CRIT] sofia.c:1632 Waiting for 3 session(s)
>>> 2011-03-11 15:28:26.517065 [CRIT] sofia.c:1632 Waiting for 13 session(s)
>>>
>>> Why these channels are not removed from list? I also noticed that memory
>>> consumption by freeswitch process constantly grows call by call.
>>>
>>> --
>>> Best regards,
>>>
>>> Dmitry Sytchev,
>>> IT Engineer
>>>
>>
>>
>>
>> --
>> Best regards,
>>
>> Dmitry Sytchev,
>> IT Engineer
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From peter.olsson at visionutveckling.se  Fri Mar 11 15:03:39 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 11 Mar 2011 13:03:39 +0100
Subject: [Freeswitch-users] Another beginner question: garbled voicemail
In-Reply-To: <4D7A0B5D.6030803@julf.com>
References: <4D7A0B5D.6030803@julf.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2F6BF79@cooper>

Check codec negotiations, network issues etc. Also, are you running in a virtual machine?

However, random noise should probably have something to do with codecs.

/Peter



-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Johan Helsingius
Skickat: den 11 mars 2011 12:46
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] Another beginner question: garbled voicemail

Hi!

For some reason the recorded voicemail messages come out as random noise.
All the system messages sound fine, it's just that the recorded message
is pretty much random garbage.

Any hints for where to start looking?

	Julf

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

!DSPAM:4d7a0c4d32761581579636!



From codecomplete at free.fr  Fri Mar 11 15:23:29 2011
From: codecomplete at free.fr (GillesToo)
Date: Fri, 11 Mar 2011 04:23:29 -0800 (PST)
Subject: [Freeswitch-users] Working configuration for FS behind NAT?
Message-ID: <1299846209646-6161235.post@n2.nabble.com>

Hello

I'd like to set things up, where...
- FS and local SIP extensions are on a private LAN behind a NAT firewall,
- remote employees are allowed to register with Freeswitch so they're part
of Freeswitch,
- none of the routers support UPnP or NAT-PMP, and
- to lower load on the Freeswitch server, I'd like to have RTP packets flow
directly between extensions instead of going through Freeswitch, like
Asterisk does with its "canreinvite=no" option. However, Freeswitch should
remain in the loop somehow so that users can access PBX features during
their call.

Here's the picture:

http://img577.imageshack.us/img577/1795/freeswitchnat.png

Apparently, changes to Freeswitch made 
http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios this page 
obsolete. If Freeswitch can do this, does someone have a basic, working
configuration that I could use to get started?

Thank you.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Working-configuration-for-FS-behind-NAT-tp6161235p6161235.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Fri Mar 11 15:34:42 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 11 Mar 2011 12:34:42 +0000
Subject: [Freeswitch-users] Strange Behavior with Channel Variable
 [session:setVariable() vs <action application="set" ... /> ]
In-Reply-To: <AANLkTin2+5TAVHEbTTQDKwoqqfWFJLruoyD_FiffptaP@mail.gmail.com>
References: <AANLkTin2+5TAVHEbTTQDKwoqqfWFJLruoyD_FiffptaP@mail.gmail.com>
Message-ID: <AANLkTikGuaKchNQe7N63nfKnhxmqGDM-MpBN0TroSfzY@mail.gmail.com>

<action application="set" date="playback_timeout_sec=10" />

I assume you mean data="..." not date="..."?

-Steve



On 10 March 2011 20:21, Sidharth Kshatriya <sid.kshatriya at gmail.com> wrote:

> Dear All,
>
> I am using a little known but very useful channel variable called
> "playback_timeout_sec"
>
> In Lua if I do: session:setVariable("playback_timeout_sec", "10")
>
> and subsequently in a macro I have:
>
> <action function="play-file" date="/some/filename.mp3" />
>
> Then the filename.mp3 stops playing automatically after 10 seconds. This is
> very useful if you want to play short excerpts of a file that could be very
> long.
>
> But if I do:
>
> <action application="set" date="playback_timeout_sec=10" />
> <action function="play-file" date="/some/filename.mp3" />
>
> My file does NOT stop playing after 10 seconds. This is weird given that
> <action application="set" .. /> is doing exactly the same thing as
> session:setVariable() ... right?
>
> My question
> 1. What could be the problem with application set?
> 2. Is there a way to automatically stop playback after a certain number of
> seconds for a file? (without doing tedious things like create a separate
> 10second version of an mp3 using ffmpeg etc.)
>
> Thanks,
>
> Sidharth
> --
> Sidharth Kshatriya
> www.sidk.info
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From julf at julf.com  Fri Mar 11 15:38:42 2011
From: julf at julf.com (Johan Helsingius)
Date: Fri, 11 Mar 2011 13:38:42 +0100
Subject: [Freeswitch-users] Another beginner question: garbled voicemail
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B2F6BF79@cooper>
References: <4D7A0B5D.6030803@julf.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2F6BF79@cooper>
Message-ID: <4D7A17D2.5000307@julf.com>

Peter,

> Check codec negotiations, network issues etc. Also, are you running in a virtual machine?

No, native on Kubuntu 10.10.

> However, random noise should probably have something to do with codecs.

Agree. Any links to help in debugging the codecs?

Here are my current codec settings:

vars.xml:  <X-PRE-PROCESS cmd="set"
data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
vars.xml:  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
sip_profiles/internal.xml:    <param name="inbound-codec-prefs"
value="$${global_codec_prefs}"/>
sip_profiles/internal.xml:    <param name="outbound-codec-prefs"
value="$${global_codec_prefs}"/>
sip_profiles/internal.xml:    <param name="inbound-codec-negotiation"
value="generous"/>
sip_profiles/external.xml:    <param name="inbound-codec-prefs"
value="$${global_codec_prefs}"/>
sip_profiles/external.xml:    <param name="outbound-codec-prefs"
value="$${outbound_codec_prefs}"/>
sip_profiles/external.xml:    <param name="inbound-codec-negotiation"
value="generous"/>

	Julf


From hkalyoncu at gmail.com  Fri Mar 11 16:02:43 2011
From: hkalyoncu at gmail.com (hkalyoncu)
Date: Fri, 11 Mar 2011 05:02:43 -0800 (PST)
Subject: [Freeswitch-users] strange errors when using mod_python
Message-ID: <1299848563170-6161361.post@n2.nabble.com>

hello,

im doing dynamic routing with mod_python.
my python script returns a dialplan after checking db record in terms of
destination number.

the problem is:
although python script returns correct result, freeswitch cannot process
returned dialplan and gives strange errors. This is not happening in all
calls but i can say 3-4 out of 10 calls drop like this.

i rebuilt freeswitch with latest repository but there is no change.   

here is a sample output from console(sorry for censored fields):

2011-03-11 14:20:49.073690 [INFO] mod_dialplan_xml.c:331 Processing
xxxxxxxxxxxx ->xxxxxxxxxxxx in context context_3
2011-03-11 14:20:49.074747 [NOTICE] mod_python.c:118 Invoking py module: dp
2011-03-11 14:20:49.074747 [DEBUG] mod_python.c:188 Call python script
2011-03-11 14:20:49.074747 [INFO] switch_cpp.cpp:1197 calling number :
xxxxxxxxxx
2011-03-11 14:20:49.074747 [INFO] switch_cpp.cpp:1197 called number :
xxxxxxxxxx
2011-03-11 14:20:49.075833 [DEBUG] mod_python.c:191 Finished calling python
script
Dialplan: sofia/sipinterface_4/xxxxxxxxxxxx at xxx.22.9.195 parsing
[context_3->generated] continue=false
Dialplan: sofia/sipinterface_4/xxxxxxxxxxxx at xxx.22.9.195 Absolute Condition
[generated]
Dialplan: sofia/sipinterface_4/xxxxxxxxxxxx at xxx.22.9.195 Action answer()
Dialplan: sofia/sipinterface_4/xxxxxxxxxxxx at xxx.22.9.195 Action
bridge(sofia/sipinterface_5/xxxxxxxxxx at 10.10.1.5:5060)
Dialplan: sofia/sipinterface_4/xxxxxxxxxxxx at xxx.22.9.195 Action ()
2011-03-11 14:20:49.075833 [DEBUG] switch_core_state_machine.c:119
(sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195) State Change CS_ROUTING ->
CS_EXECUTE
2011-03-11 14:20:49.075833 [DEBUG] switch_core_session.c:1116 Send signal
sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195 [BREAK]
2011-03-11 14:20:49.075833 [DEBUG] switch_core_state_machine.c:359
(sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195) State ROUTING going to sleep
2011-03-11 14:20:49.075833 [DEBUG] switch_core_state_machine.c:320
(sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195) Running State Change
CS_EXECUTE
2011-03-11 14:20:49.075833 [DEBUG] switch_core_state_machine.c:366
(sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195) State EXECUTE
2011-03-11 14:20:49.075833 [DEBUG] mod_sofia.c:240
sofia/sipinterface_4/xxxxxxxxxxxxx at xxx.22.9.195 SOFIA EXECUTE
2011-03-11 14:20:49.075833 [DEBUG] switch_core_state_machine.c:157
sofia/sipinterface_4/xxxxxxxxxx at xxx.22.9.195 Standard EXECUTE
EXECUTE sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195 answer()
2011-03-11 14:20:49.075833 [DEBUG] sofia_glue.c:2990 AUDIO RTP
[sofia/sipinterface_4/xxxxxxxxxxxx at xxx.22.9.195] 10.xxx.xxx.xxx port 31882
-> xxx.xxx.xxx.xxx port 35892 codec: 18 ms: 20
2011-03-11 14:20:49.075833 [DEBUG] switch_rtp.c:1623 Starting timer [soft]
160 bytes per 20ms
2011-03-11 14:20:49.076915 [DEBUG] sofia_glue.c:3245 Set 2833 dtmf send
payload to 101
2011-03-11 14:20:49.076915 [DEBUG] sofia_glue.c:3250 Set 2833 dtmf receive
payload to 101
2011-03-11 14:20:49.076915 [DEBUG] mod_sofia.c:681 Local SDP
sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195:
v=0
o=FreeSWITCH 1299814167 1299814168 IN IP4 10.xxx.xxx.xxx
s=FreeSWITCH
c=IN IP4 10.xxx.xxx.xxx
t=0 0
m=audio 31882 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2011-03-11 14:20:49.076915 [DEBUG] switch_core_session.c:709 Send signal
sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195 [BREAK]
2011-03-11 14:20:49.076915 [DEBUG] switch_channel.c:2796
(sofia/sipinterface_4/xxxxxxxxxxxxx at xxx.22.9.195) Callstate Change RINGING
-> ACTIVE
2011-03-11 14:20:49.077991 [NOTICE] mod_dptools.c:929 Channel
[sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195] has been answered
2011-03-11 14:20:49.077991 [DEBUG] sofia.c:4725 Channel
sofia/sipinterface_4/xxxxxxxxxxxx at xxx.22.9.195 entering state
[completed][200]
2011-03-11 14:20:49.077991 [ERR] switch_core_session.c:1918 Invalid
Application .22.9.195 Action
bridge(sofia/sipinterface_5/xxxxxxxxxxxx at 10.10.1.5:5060)

2011-03-11 14:20:49.077991 [DEBUG] switch_channel.c:2546
(sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195) Callstate Change ACTIVE ->
HANGUP
2011-03-11 14:20:49.077991 [NOTICE] switch_core_session.c:1919 Hangup
sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195 [CS_EXECUTE]
[DESTINATION_OUT_OF_ORDER]
2011-03-11 14:20:49.077991 [DEBUG] switch_channel.c:2562 Send signal
sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195 [KILL]
2011-03-11 14:20:49.077991 [DEBUG] switch_core_session.c:1116 Send signal
sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195 [BREAK]
2011-03-11 14:20:49.077991 [DEBUG] switch_core_state_machine.c:366
(sofia/sipinterface_4/xxxxxxxxxxxx at xxx.22.9.195) State EXECUTE going to
sleep
2011-03-11 14:20:49.077991 [DEBUG] switch_core_state_machine.c:320
(sofia/sipinterface_4/xxxxxxxxxxxx at xxx.22.9.195) Running State Change
CS_HANGUP
2011-03-11 14:20:49.077991 [DEBUG] switch_core_state_machine.c:560
(sofia/sipinterface_4/xxxxxxxxxxxx at xxx.22.9.195) State HANGUP
2011-03-11 14:20:49.077991 [DEBUG] mod_sofia.c:457 Channel
sofia/sipinterface_4/xxxxxxxxxxxx at xxx.22.9.195 hanging up, cause:
DESTINATION_OUT_OF_ORDER
2011-03-11 14:20:49.077991 [DEBUG] mod_sofia.c:500 Sending BYE to
sofia/sipinterface_4/xxxxxxxxxxxx at xxx.22.9.195

The text in error message (Invalid Application xxxxx) changes in every call.
i recheck the python script several times, each time  it gives expected xml
output.
But freeswitch somehow cannot get (or process?)  that output.

Any idea? 

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/strange-errors-when-using-mod-python-tp6161361p6161361.html
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From peter.olsson at visionutveckling.se  Fri Mar 11 16:18:08 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 11 Mar 2011 14:18:08 +0100
Subject: [Freeswitch-users] strange errors when using mod_python
In-Reply-To: <1299848563170-6161361.post@n2.nabble.com>
References: <1299848563170-6161361.post@n2.nabble.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2F6C044@cooper>

This seems suspect;

Invalid Application .22.9.195 Action
bridge(sofia/sipinterface_5/xxxxxxxxxxxx at 10.10.1.5:5060)

/Peter

-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r hkalyoncu
Skickat: den 11 mars 2011 14:03
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] strange errors when using mod_python

hello,

im doing dynamic routing with mod_python.
my python script returns a dialplan after checking db record in terms of
destination number.

the problem is:
although python script returns correct result, freeswitch cannot process
returned dialplan and gives strange errors. This is not happening in all
calls but i can say 3-4 out of 10 calls drop like this.

i rebuilt freeswitch with latest repository but there is no change.   

here is a sample output from console(sorry for censored fields):

2011-03-11 14:20:49.073690 [INFO] mod_dialplan_xml.c:331 Processing
xxxxxxxxxxxx ->xxxxxxxxxxxx in context context_3
2011-03-11 14:20:49.074747 [NOTICE] mod_python.c:118 Invoking py module: dp
2011-03-11 14:20:49.074747 [DEBUG] mod_python.c:188 Call python script
2011-03-11 14:20:49.074747 [INFO] switch_cpp.cpp:1197 calling number :
xxxxxxxxxx
2011-03-11 14:20:49.074747 [INFO] switch_cpp.cpp:1197 called number :
xxxxxxxxxx
2011-03-11 14:20:49.075833 [DEBUG] mod_python.c:191 Finished calling python
script
Dialplan: sofia/sipinterface_4/xxxxxxxxxxxx at xxx.22.9.195 parsing
[context_3->generated] continue=false
Dialplan: sofia/sipinterface_4/xxxxxxxxxxxx at xxx.22.9.195 Absolute Condition
[generated]
Dialplan: sofia/sipinterface_4/xxxxxxxxxxxx at xxx.22.9.195 Action answer()
Dialplan: sofia/sipinterface_4/xxxxxxxxxxxx at xxx.22.9.195 Action
bridge(sofia/sipinterface_5/xxxxxxxxxx at 10.10.1.5:5060)
Dialplan: sofia/sipinterface_4/xxxxxxxxxxxx at xxx.22.9.195 Action ()
2011-03-11 14:20:49.075833 [DEBUG] switch_core_state_machine.c:119
(sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195) State Change CS_ROUTING ->
CS_EXECUTE
2011-03-11 14:20:49.075833 [DEBUG] switch_core_session.c:1116 Send signal
sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195 [BREAK]
2011-03-11 14:20:49.075833 [DEBUG] switch_core_state_machine.c:359
(sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195) State ROUTING going to sleep
2011-03-11 14:20:49.075833 [DEBUG] switch_core_state_machine.c:320
(sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195) Running State Change
CS_EXECUTE
2011-03-11 14:20:49.075833 [DEBUG] switch_core_state_machine.c:366
(sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195) State EXECUTE
2011-03-11 14:20:49.075833 [DEBUG] mod_sofia.c:240
sofia/sipinterface_4/xxxxxxxxxxxxx at xxx.22.9.195 SOFIA EXECUTE
2011-03-11 14:20:49.075833 [DEBUG] switch_core_state_machine.c:157
sofia/sipinterface_4/xxxxxxxxxx at xxx.22.9.195 Standard EXECUTE
EXECUTE sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195 answer()
2011-03-11 14:20:49.075833 [DEBUG] sofia_glue.c:2990 AUDIO RTP
[sofia/sipinterface_4/xxxxxxxxxxxx at xxx.22.9.195] 10.xxx.xxx.xxx port 31882
-> xxx.xxx.xxx.xxx port 35892 codec: 18 ms: 20
2011-03-11 14:20:49.075833 [DEBUG] switch_rtp.c:1623 Starting timer [soft]
160 bytes per 20ms
2011-03-11 14:20:49.076915 [DEBUG] sofia_glue.c:3245 Set 2833 dtmf send
payload to 101
2011-03-11 14:20:49.076915 [DEBUG] sofia_glue.c:3250 Set 2833 dtmf receive
payload to 101
2011-03-11 14:20:49.076915 [DEBUG] mod_sofia.c:681 Local SDP
sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195:
v=0
o=FreeSWITCH 1299814167 1299814168 IN IP4 10.xxx.xxx.xxx
s=FreeSWITCH
c=IN IP4 10.xxx.xxx.xxx
t=0 0
m=audio 31882 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2011-03-11 14:20:49.076915 [DEBUG] switch_core_session.c:709 Send signal
sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195 [BREAK]
2011-03-11 14:20:49.076915 [DEBUG] switch_channel.c:2796
(sofia/sipinterface_4/xxxxxxxxxxxxx at xxx.22.9.195) Callstate Change RINGING
-> ACTIVE
2011-03-11 14:20:49.077991 [NOTICE] mod_dptools.c:929 Channel
[sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195] has been answered
2011-03-11 14:20:49.077991 [DEBUG] sofia.c:4725 Channel
sofia/sipinterface_4/xxxxxxxxxxxx at xxx.22.9.195 entering state
[completed][200]
2011-03-11 14:20:49.077991 [ERR] switch_core_session.c:1918 Invalid
Application .22.9.195 Action
bridge(sofia/sipinterface_5/xxxxxxxxxxxx at 10.10.1.5:5060)

2011-03-11 14:20:49.077991 [DEBUG] switch_channel.c:2546
(sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195) Callstate Change ACTIVE ->
HANGUP
2011-03-11 14:20:49.077991 [NOTICE] switch_core_session.c:1919 Hangup
sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195 [CS_EXECUTE]
[DESTINATION_OUT_OF_ORDER]
2011-03-11 14:20:49.077991 [DEBUG] switch_channel.c:2562 Send signal
sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195 [KILL]
2011-03-11 14:20:49.077991 [DEBUG] switch_core_session.c:1116 Send signal
sofia/sipinterface_4/xxxxxxxxxxx at xxx.22.9.195 [BREAK]
2011-03-11 14:20:49.077991 [DEBUG] switch_core_state_machine.c:366
(sofia/sipinterface_4/xxxxxxxxxxxx at xxx.22.9.195) State EXECUTE going to
sleep
2011-03-11 14:20:49.077991 [DEBUG] switch_core_state_machine.c:320
(sofia/sipinterface_4/xxxxxxxxxxxx at xxx.22.9.195) Running State Change
CS_HANGUP
2011-03-11 14:20:49.077991 [DEBUG] switch_core_state_machine.c:560
(sofia/sipinterface_4/xxxxxxxxxxxx at xxx.22.9.195) State HANGUP
2011-03-11 14:20:49.077991 [DEBUG] mod_sofia.c:457 Channel
sofia/sipinterface_4/xxxxxxxxxxxx at xxx.22.9.195 hanging up, cause:
DESTINATION_OUT_OF_ORDER
2011-03-11 14:20:49.077991 [DEBUG] mod_sofia.c:500 Sending BYE to
sofia/sipinterface_4/xxxxxxxxxxxx at xxx.22.9.195

The text in error message (Invalid Application xxxxx) changes in every call.
i recheck the python script several times, each time  it gives expected xml
output.
But freeswitch somehow cannot get (or process?)  that output.

Any idea? 

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/strange-errors-when-using-mod-python-tp6161361p6161361.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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!DSPAM:4d7a1ea232768243425244!



From sid.kshatriya at gmail.com  Fri Mar 11 14:56:20 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Fri, 11 Mar 2011 17:26:20 +0530
Subject: [Freeswitch-users] Strange Behavior with Channel Variable
 [session:setVariable() vs <action application="set" ... /> ]
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B2F6BF36@cooper>
References: <AANLkTin2+5TAVHEbTTQDKwoqqfWFJLruoyD_FiffptaP@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2F6BF36@cooper>
Message-ID: <AANLkTins9NDYKw6y2bEfW-EnDxu_gS1S4KvsKd+Z4YP4@mail.gmail.com>

Yes, this was resolved!

On Fri, Mar 11, 2011 at 4:53 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> Is this a delayed email? I think this was resolved yesterday, right?
>
>
>
> /Peter
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Sidharth Kshatriya
> *Skickat:* den 10 mars 2011 21:22
> *Till:* freeswitch-users at lists.freeswitch.org
> *?mne:* [Freeswitch-users] Strange Behavior with Channel Variable
> [session:setVariable() vs <action application="set" ... /> ]
>
>
>
> Dear All,
>
>
>
> I am using a little known but very useful channel variable called
> "playback_timeout_sec"
>
>
>
> In Lua if I do: session:setVariable("playback_timeout_sec", "10")
>
>
>
> and subsequently in a macro I have:
>
>
>
> <action function="play-file" date="/some/filename.mp3" />
>
>
>
> Then the filename.mp3 stops playing automatically after 10 seconds. This is
> very useful if you want to play short excerpts of a file that could be very
> long.
>
>
>
> But if I do:
>
>
>
> <action application="set" date="playback_timeout_sec=10" />
>
> <action function="play-file" date="/some/filename.mp3" />
>
>
>
> My file does NOT stop playing after 10 seconds. This is weird given that
> <action application="set" .. /> is doing exactly the same thing as
> session:setVariable() ... right?
>
>
>
> My question
>
> 1. What could be the problem with application set?
>
> 2. Is there a way to automatically stop playback after a certain number of
> seconds for a file? (without doing tedious things like create a separate
> 10second version of an mp3 using ffmpeg etc.)
>
>
>
> Thanks,
>
>
>
> Sidharth
>
> --
> Sidharth Kshatriya
> www.sidk.info
>
> !DSPAM:4d7a03ab32764871212382!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sidharth Kshatriya
www.sidk.info
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From hkalyoncu at gmail.com  Fri Mar 11 16:32:48 2011
From: hkalyoncu at gmail.com (hkalyoncu)
Date: Fri, 11 Mar 2011 05:32:48 -0800 (PST)
Subject: [Freeswitch-users] strange errors when using mod_python
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B2F6C044@cooper>
References: <1299848563170-6161361.post@n2.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2F6C044@cooper>
Message-ID: <1299850368472-6161441.post@n2.nabble.com>

yeah 
i wrote in my first message that partial text changing in every other error
message.
as you can see that is the part of the ip adress. but never used it in my
script. 
it somehow gets part of channels name and shows it like invalid application.
so strange.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/strange-errors-when-using-mod-python-tp6161361p6161441.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From julf at julf.com  Fri Mar 11 16:44:01 2011
From: julf at julf.com (Johan Helsingius)
Date: Fri, 11 Mar 2011 14:44:01 +0100
Subject: [Freeswitch-users] Another beginner question: garbled voicemail
In-Reply-To: <4D7A17D2.5000307@julf.com>
References: <4D7A0B5D.6030803@julf.com>	<549CFEF87AEDE841A38E9D15EAB4C04C58B2F6BF79@cooper>
	<4D7A17D2.5000307@julf.com>
Message-ID: <4D7A2721.6020003@julf.com>

> However, random noise should probably have something to do with codecs.

Well, looks like the incoming call that gets directed to voicemail
is PCMA, and the connection to check voicemail is PCMU, but shouldn't
freeswitch "do the right thing"?

	Julf


From peter.olsson at visionutveckling.se  Fri Mar 11 17:03:12 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 11 Mar 2011 15:03:12 +0100
Subject: [Freeswitch-users] Another beginner question: garbled voicemail
In-Reply-To: <4D7A2721.6020003@julf.com>
References: <4D7A0B5D.6030803@julf.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2F6BF79@cooper>
	<4D7A17D2.5000307@julf.com> <4D7A2721.6020003@julf.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2F6C097@cooper>

Send us some logs and example from you dialplan.

/Peter



-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Johan Helsingius
Skickat: den 11 mars 2011 14:44
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Another beginner question: garbled voicemail

> However, random noise should probably have something to do with codecs.

Well, looks like the incoming call that gets directed to voicemail
is PCMA, and the connection to check voicemail is PCMU, but shouldn't
freeswitch "do the right thing"?

	Julf

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

!DSPAM:4d7a27fd32769505921137!



From steveayre at gmail.com  Fri Mar 11 17:12:43 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 11 Mar 2011 14:12:43 +0000
Subject: [Freeswitch-users] strange errors when using mod_python
In-Reply-To: <1299850368472-6161441.post@n2.nabble.com>
References: <1299848563170-6161361.post@n2.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2F6C044@cooper>
	<1299850368472-6161441.post@n2.nabble.com>
Message-ID: <AANLkTinLJLU3+fc=L3EwAAJe_U-kpTi7HvB40wUu78zD@mail.gmail.com>

Can you write the generated XML to a file on disk, to verify the XML given
to FS is correct?

You said it's ok manually, but it's possible that it's different when a FS
session runs the script.

-Steve

On 11 March 2011 13:32, hkalyoncu <hkalyoncu at gmail.com> wrote:

> yeah
> i wrote in my first message that partial text changing in every other error
> message.
> as you can see that is the part of the ip adress. but never used it in my
> script.
> it somehow gets part of channels name and shows it like invalid
> application.
> so strange.
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/strange-errors-when-using-mod-python-tp6161361p6161441.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From julf at julf.com  Fri Mar 11 17:37:28 2011
From: julf at julf.com (Johan Helsingius)
Date: Fri, 11 Mar 2011 15:37:28 +0100
Subject: [Freeswitch-users] Another beginner question: garbled voicemail
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B2F6C097@cooper>
References: <4D7A0B5D.6030803@julf.com>	<549CFEF87AEDE841A38E9D15EAB4C04C58B2F6BF79@cooper>	<4D7A17D2.5000307@julf.com>
	<4D7A2721.6020003@julf.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2F6C097@cooper>
Message-ID: <4D7A33A8.1000008@julf.com>

Peter,

> Send us some logs and example from you dialplan.

Rather annoyingly, as I tried again, in order to get a nice, clean
log, I git a perfectly fine voice recording. Will have to try again
until I get one that captures things going wrong...

	Julf


From anthony.minessale at gmail.com  Fri Mar 11 18:46:23 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 11 Mar 2011 09:46:23 -0600
Subject: [Freeswitch-users] Another beginner question: garbled voicemail
In-Reply-To: <4D7A2721.6020003@julf.com>
References: <4D7A0B5D.6030803@julf.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2F6BF79@cooper>
	<4D7A17D2.5000307@julf.com> <4D7A2721.6020003@julf.com>
Message-ID: <AANLkTik2C95ooHuC2_ux=LfXQXR3ZGC3i6LNQde1rqeO@mail.gmail.com>

are you using x-lite by any chance? they have a long standing
ulaw/alaw bug where they mess up in a transcoding situation.


On Fri, Mar 11, 2011 at 7:44 AM, Johan Helsingius <julf at julf.com> wrote:
>> However, random noise should probably have something to do with codecs.
>
> Well, looks like the incoming call that gets directed to voicemail
> is PCMA, and the connection to check voicemail is PCMU, but shouldn't
> freeswitch "do the right thing"?
>
> ? ? ? ?Julf
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 sid.kshatriya at gmail.com  Fri Mar 11 19:12:50 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Fri, 11 Mar 2011 21:42:50 +0530
Subject: [Freeswitch-users] Strange Behavior with Channel Variable
 [session:setVariable() vs <action application="set" ... /> ]
In-Reply-To: <AANLkTikGuaKchNQe7N63nfKnhxmqGDM-MpBN0TroSfzY@mail.gmail.com>
References: <AANLkTin2+5TAVHEbTTQDKwoqqfWFJLruoyD_FiffptaP@mail.gmail.com>
	<AANLkTikGuaKchNQe7N63nfKnhxmqGDM-MpBN0TroSfzY@mail.gmail.com>
Message-ID: <AANLkTimDDe2LL=kM_MSS=SB_jBdzvAHviX0XOSFKucFC@mail.gmail.com>

Yes data :-)

checkout http://jira.freeswitch.org/browse/FS-3142 for resolution...

On Fri, Mar 11, 2011 at 6:04 PM, Steven Ayre <steveayre at gmail.com> wrote:

> <action application="set" date="playback_timeout_sec=10" />
>
> I assume you mean data="..." not date="..."?
>
> -Steve
>
>
>
> On 10 March 2011 20:21, Sidharth Kshatriya <sid.kshatriya at gmail.com>wrote:
>
>>  Dear All,
>>
>> I am using a little known but very useful channel variable called
>> "playback_timeout_sec"
>>
>> In Lua if I do: session:setVariable("playback_timeout_sec", "10")
>>
>> and subsequently in a macro I have:
>>
>> <action function="play-file" date="/some/filename.mp3" />
>>
>> Then the filename.mp3 stops playing automatically after 10 seconds. This
>> is very useful if you want to play short excerpts of a file that could be
>> very long.
>>
>> But if I do:
>>
>> <action application="set" date="playback_timeout_sec=10" />
>> <action function="play-file" date="/some/filename.mp3" />
>>
>> My file does NOT stop playing after 10 seconds. This is weird given that
>> <action application="set" .. /> is doing exactly the same thing as
>> session:setVariable() ... right?
>>
>> My question
>> 1. What could be the problem with application set?
>> 2. Is there a way to automatically stop playback after a certain number of
>> seconds for a file? (without doing tedious things like create a separate
>> 10second version of an mp3 using ffmpeg etc.)
>>
>> Thanks,
>>
>> Sidharth
>> --
>> Sidharth Kshatriya
>> www.sidk.info
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sidharth Kshatriya
www.sidk.info
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From julf at julf.com  Fri Mar 11 19:24:03 2011
From: julf at julf.com (Johan Helsingius)
Date: Fri, 11 Mar 2011 17:24:03 +0100
Subject: [Freeswitch-users] Another beginner question: garbled voicemail
In-Reply-To: <AANLkTik2C95ooHuC2_ux=LfXQXR3ZGC3i6LNQde1rqeO@mail.gmail.com>
References: <4D7A0B5D.6030803@julf.com>	<549CFEF87AEDE841A38E9D15EAB4C04C58B2F6BF79@cooper>	<4D7A17D2.5000307@julf.com>
	<4D7A2721.6020003@julf.com>
	<AANLkTik2C95ooHuC2_ux=LfXQXR3ZGC3i6LNQde1rqeO@mail.gmail.com>
Message-ID: <4D7A4CA3.9030701@julf.com>

> are you using x-lite by any chance? they have a long standing
> ulaw/alaw bug where they mess up in a transcoding situation.

No, cisco 7960's, nokia N900 (running sofia, I assume), and
a pots-to-voip service.

	Julf


From anthony.minessale at gmail.com  Fri Mar 11 19:25:54 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 11 Mar 2011 10:25:54 -0600
Subject: [Freeswitch-users] Another beginner question: garbled voicemail
In-Reply-To: <4D7A4CA3.9030701@julf.com>
References: <4D7A0B5D.6030803@julf.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2F6BF79@cooper>
	<4D7A17D2.5000307@julf.com> <4D7A2721.6020003@julf.com>
	<AANLkTik2C95ooHuC2_ux=LfXQXR3ZGC3i6LNQde1rqeO@mail.gmail.com>
	<4D7A4CA3.9030701@julf.com>
Message-ID: <AANLkTi=_XwM09bHXuEctmW9Zzts2+tSvWnp-D3Z4_AZX@mail.gmail.com>

try pcap of the call so you can look at the packets and the trace and
even extract the audio to see what is going on.


On Fri, Mar 11, 2011 at 10:24 AM, Johan Helsingius <julf at julf.com> wrote:
>> are you using x-lite by any chance? they have a long standing
>> ulaw/alaw bug where they mess up in a transcoding situation.
>
> No, cisco 7960's, nokia N900 (running sofia, I assume), and
> a pots-to-voip service.
>
> ? ? ? ?Julf
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 mario_fs at mgtech.com  Fri Mar 11 19:42:29 2011
From: mario_fs at mgtech.com (Mario G)
Date: Fri, 11 Mar 2011 08:42:29 -0800
Subject: [Freeswitch-users] Internal extensions placing call on hold
	drops immediately
In-Reply-To: <AANLkTimW=zaLRsiLqBF_r6jHLzXeC9sns_xqk6rHTduR@mail.gmail.com>
References: <5FD88FAF-068C-4DF8-BEC3-736268A05940@mgtech.com>
	<AANLkTimW=zaLRsiLqBF_r6jHLzXeC9sns_xqk6rHTduR@mail.gmail.com>
Message-ID: <978D4DCB-85BE-4190-9411-3238AD4D130D@mgtech.com>

It now works, I guess I had bad timing pulling the git. Thanks!

On Mar 10, 2011, at 2:49 PM, Anthony Minessale wrote:

> try latest git again to be sure.  There was a patch in that area of
> the code last night.
> 
> 
> On Thu, Mar 10, 2011 at 3:13 PM, Mario G <mario_fs at mgtech.com> wrote:
>> I don't know when this started but was working fine at one time. If a call is made between extensions and either extension places the call on hold, it states "trying to hold". I installed the git from 3/9/11 before posting here. Could this be a Sofia problem (see FREEPBX comment) or something that FS needs to "setup" the phone to work since it works fine using a different PBX? Any help appreciated!
>> 
>> * The phones are Linksys SPA962
>> * If I plug the old SPA9000 in as a PBX hold works fine for the SPA9000 extensions, only FS has the issue
>> * All connected via HP Procurve 2520G switch
>> * Outside calls are also affected but this is still a problem even with the router  disconnected
>> * I can ping (ICMP) from FS to the phone and from anywhere to the phone
>> 
>> FS TRACE:
>> I started several FS traces: sip global,internal, log 9 console debug, etc. It looked like NOTHING was sent when the hold button was pressed.
>> 
>> WIRESHARK showed these messages stacking up when the hold is pressed
>> src 1.123.1.23 (phone)   dest 1.123.1.7 (FS)  Proto ICMP  Destination unreachable (Port unreachable)
>> 
>> Someone updated FREEPBX in FEB 2011 and had the same issue with SPA phones.
>> http://www.freepbx.org/forum/freepbx/users/spa-941-942-drop-call-when-placed-on-hold-also-have-to-click-submit-before-syste
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From Nabble at slickdeals.endjunk.com  Fri Mar 11 20:55:12 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 11 Mar 2011 09:55:12 -0800 (PST)
Subject: [Freeswitch-users] freeswitch -nc -nosql
In-Reply-To: <AANLkTindLhrtsbVmifg699TyQtv+JOYa52_Tj1CqkwVs@mail.gmail.com>
References: <1298688216720-6066718.post@n2.nabble.com>
	<1299524988820-6098582.post@n2.nabble.com>
	<AANLkTik6c5rHYnr6cxmsBzRCcc94VL=jmYtud29TVBzc@mail.gmail.com>
	<AANLkTindLhrtsbVmifg699TyQtv+JOYa52_Tj1CqkwVs@mail.gmail.com>
Message-ID: <1299866112249-6162295.post@n2.nabble.com>


Anthony Minessale wrote:
> try latest it'd fixed now.
I just saw this in the morning. After recompiled/reinstalled, my Seagate
DockStar is up and smoothly running with FreeSWITCH Version 1.0.head
(git-59f6654 2011-03-10 22-02-45 -0600). Many thanks.


integer boundary issue, switch_time_t vs regular time_t
I noticed some other codes also depend on this switch_time_t and am
wondering if it may affect them on all 32-bit platforms.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-nc-nosql-tp6066718p6162295.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Fri Mar 11 21:23:39 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 11 Mar 2011 12:23:39 -0600
Subject: [Freeswitch-users] freeswitch -nc -nosql
In-Reply-To: <1299866112249-6162295.post@n2.nabble.com>
References: <1298688216720-6066718.post@n2.nabble.com>
	<1299524988820-6098582.post@n2.nabble.com>
	<AANLkTik6c5rHYnr6cxmsBzRCcc94VL=jmYtud29TVBzc@mail.gmail.com>
	<AANLkTindLhrtsbVmifg699TyQtv+JOYa52_Tj1CqkwVs@mail.gmail.com>
	<1299866112249-6162295.post@n2.nabble.com>
Message-ID: <AANLkTinpvkr+_D-06Umkdj+97DUiGxQ038HKMP0cusjD@mail.gmail.com>

only when you get them mixed up in the code.

On Fri, Mar 11, 2011 at 11:55 AM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
> Anthony Minessale wrote:
>> try latest it'd fixed now.
> I just saw this in the morning. After recompiled/reinstalled, my Seagate
> DockStar is up and smoothly running with FreeSWITCH Version 1.0.head
> (git-59f6654 2011-03-10 22-02-45 -0600). Many thanks.
>
>
> integer boundary issue, switch_time_t vs regular time_t
> I noticed some other codes also depend on this switch_time_t and am
> wondering if it may affect them on all 32-bit platforms.
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-nc-nosql-tp6066718p6162295.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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  Fri Mar 11 22:15:55 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 11 Mar 2011 13:15:55 -0600
Subject: [Freeswitch-users] Already hung channels in 'show channels'
 output never go away
In-Reply-To: <AANLkTinTApDm-YL5UjqPnc8vaE-e+xpdAXiwaD+DTZhX@mail.gmail.com>
References: <AANLkTi=jiL-aSmkDusoH_9EYRzPZ1ePXkWAQHXn53Np5@mail.gmail.com>
	<AANLkTinUg76_ncbjCwf=W8T5BPSS3FdstGQkkptv-xme@mail.gmail.com>
	<AANLkTimzvf-ZaiaVKCprbHh6JcFx3dXnbaU54T8KL=O6@mail.gmail.com>
	<AANLkTinTApDm-YL5UjqPnc8vaE-e+xpdAXiwaD+DTZhX@mail.gmail.com>
Message-ID: <AANLkTik_CE66S0FToNMycX+U_WV3m12T=8D4t88BdcKP@mail.gmail.com>

that looks like they are all stuck in the CDR module.
What module are you using?



On Fri, Mar 11, 2011 at 5:58 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> I think this is not the case, but anyway, what to do with these hung
> channels?
> Maybe I'm doing something wrong while bridging or processing events? Maybe
> unprocessed events can affect channel destroy procedure?
> 2011/3/11 Avi Marcus <avi at avimarcus.net>
>>
>> Regarding ram usage, I'd imagine this is the case:
>> http://www.linuxatemyram.com/
>> -Avi
>>
>> On Fri, Mar 11, 2011 at 12:33 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>>
>>> BTW, uuid_dump?7ae28028-4bc2-11e0-a92f-452f3d3d66ea returns with -ERR No
>>> such channel.
>>>
>>> 2011/3/11 Dmitry Sytchev <kbdfck at gmail.com>
>>>>
>>>> Hi All
>>>> I'm using Perl ESL outbound script to bridge incoming call to sip
>>>> endpoint. I'm doing execute("bridge","sofia/user/somebody"), then processing
>>>> events like CHANNEL_HANGUP and EXECUTE_COMPLETE to determine bridge result.
>>>> Everything works fine, but original incoming call channel is never removed
>>>> from list:
>>>> After few calls I see original incoming channels in 'show channels'
>>>> output:
>>>> 7ae28028-4bc2-11e0-a92f-452f3d3d66ea,inbound,2011-03-11
>>>> 12:32:27,1299835947,sofia/external/xxxx at 85.x.x.199,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,xxx.xxx.ru,,,HANGUP,,,,
>>>> 883c9eb6-4bc2-11e0-a935-452f3d3d66ea,inbound,2011-03-11
>>>> 12:32:50,1299835970,sofia/external/xxxx at 85.x.x.199,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,xxx.xxx.ru,,,HANGUP,,,,
>>>> 9f6a0f92-4bc2-11e0-a93b-452f3d3d66ea,inbound,2011-03-11
>>>> 12:33:29,1299836009,sofia/external/xxxx at 85.x.x.199,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,xxx.xxx.ru,,,HANGUP,,,
>>>> Also, when I try to stop freeswitch i see these messages on console:
>>>> 2011-03-11 15:28:22.517063 [CRIT] sofia.c:1632 Waiting for 13 session(s)
>>>> 2011-03-11 15:28:22.701034 [CRIT] sofia.c:1632 Waiting for 3 session(s)
>>>> 2011-03-11 15:28:23.517064 [CRIT] sofia.c:1632 Waiting for 13 session(s)
>>>> 2011-03-11 15:28:23.701033 [CRIT] sofia.c:1632 Waiting for 3 session(s)
>>>> 2011-03-11 15:28:24.517062 [CRIT] sofia.c:1632 Waiting for 13 session(s)
>>>> 2011-03-11 15:28:24.705036 [CRIT] sofia.c:1632 Waiting for 3 session(s)
>>>> 2011-03-11 15:28:25.517064 [CRIT] sofia.c:1632 Waiting for 13 session(s)
>>>> 2011-03-11 15:28:25.709031 [CRIT] sofia.c:1632 Waiting for 3 session(s)
>>>> 2011-03-11 15:28:26.517065 [CRIT] sofia.c:1632 Waiting for 13 session(s)
>>>>
>>>> Why these channels are not removed from list? I also noticed that memory
>>>> consumption by freeswitch process constantly grows call by call.
>>>>
>>>> --
>>>> Best regards,
>>>>
>>>> Dmitry Sytchev,
>>>> IT Engineer
>>>
>>>
>>>
>>> --
>>> Best regards,
>>>
>>> Dmitry Sytchev,
>>> IT Engineer
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From msc at freeswitch.org  Sat Mar 12 00:23:41 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 11 Mar 2011 13:23:41 -0800
Subject: [Freeswitch-users] Call back from voice mail creates loop
In-Reply-To: <52BE5DAC-1F51-48CA-ABF3-FE65F7EA175F@tlainvestments.com>
References: <4CA9ECBB.2000609@communicatefreely.net>
	<AANLkTik24EaqTsfazwBPBZ4UG8eFPTZF-8XFMQ6xNFkb@mail.gmail.com>
	<4CAA313E.3070008@communicatefreely.net>
	<AANLkTimg5iRzRSYKpY4uT3O9R+-ifYU09jCCyHDYpmq-@mail.gmail.com>
	<52BE5DAC-1F51-48CA-ABF3-FE65F7EA175F@tlainvestments.com>
Message-ID: <AANLkTi=Mt2bikUESq7sTjoSvcekYSWyTAf6OEO2czdSU@mail.gmail.com>

FWIW I just tested this on latest git and it worked perfectly fine for me.
Let us know what happens when you get updated to latest.
-MC

On Thu, Mar 10, 2011 at 9:51 AM, Troy Anderson <
freeswitch at tlainvestments.com> wrote:

> I hope this isn't considered "hijacking" a thread as I think this issue is
> related.  The version of fs I'm using is from Feb 10, 2011, and I looked at
> the diff for mod_voicemail.c between trunk and the version I have installed,
> and it doesn't appear that anything related to this issue was modified.
>  However, I am in the process recompiling with latest to be sure.  In the
> mean time, perhaps you can help me understand what's going on:
>
> I have been experiencing something odd related to voicemail and pressing 5
> to reply.  While researching the problem, this thread seems to suggest the
> reason, but not an obvious answer.
>
> When you press 5 after listening to a voicemail, the mod_voicemail.c
> executes switch_core_session_execute_exten(session, cbt->cid_number,
> profile->callback_dialplan, profile->callback_context);, which I assume is
> equivalent to execute_extnesion.  The context I'm using is the same context
> that handles all of my internal extensions, so I expected that when one
> extension leaves a message for another, pressing 5 after listening to the
> message would ring back the original extension.  This is not working.
>
> My dialplan has the following condition required before trying any internal
> extensions :
>
>       <condition field="${user_exists(id ${destination_number}
> ${domain_name})}" expression="^true$" />
>
> An example may make my question more clear:
>
> I dial from 119 to 105.  119 leaves a message.  Later, 105 dials ** to
> retrieve its voicemails and listens to the message from 119.  After the
> message, he dials 5 to return the call.  mod_voicemail runs
> execute_extension to 119,XML,my_context.  In my_context, I have the
> following to see what's up:
>
> <extension name="test" continue="true">
>
> <condition break="never" field="*destination_number*" expression="^*119*
> $">
> <action application="log" data="*ERR Extension 119 matches*"/>
> </condition>
> <condition break="never" field="*${user_exists(id ${destination_number}
> ${domain_name})}*" expression="^*true*$">
> <action application="log" data="*ERR User Exists when using variables*"/>
> </condition>
> <condition break="never" field="*${user_exists(id 119 ${domain_name})}*"
> expression="^*true*$">
> <action application="log" data="*ERR User Exists when using plugged value*
> "/>
> </condition>
>
> </extension>
>
> The output is:
>
> 2011-03-10 10:42:54.720596 [NOTICE] switch_core_session.c:2152 Execute
> log(ERR Extension 119 matches)
> 2011-03-10 10:42:54.720596 [ERR] mod_dptools.c:1183 *Extension 119 matches
> *
> 2011-03-10 10:42:54.720596 [NOTICE] switch_core_session.c:2152 Execute
> log(ERR User Exists when using plugged value)
> 2011-03-10 10:42:54.720596 [ERR] mod_dptools.c:1183 *User Exists when
> using plugged value*
>
> Notice that the second condition fails.  Why is that?  Is it related to the
> issue identified in this thread?  That execute_extension is somehow setting
> destination_number differently than transfer?
>
> Also (not shown here), I have applicaiton="info" as the very next
> condition, and it shows dialed_extension as **, not 119.  I'm very confused
> about that.
>
> How do I remedy this since mod_voicemail is using execute_extension?  Can I
> somehow determine this and execute my own transfer?  Or should I somehow
> modify my user_exists expression?
>
> Thanks for any guidance!
>
> -Troy
>
>
> On Oct 4, 2010, at 1:55 PM, Michael Collins wrote:
>
> Yes, transfer is your friend in this scenario. :)
> -MC
>
> On Mon, Oct 4, 2010 at 12:55 PM, Tim St. Pierre <
> fs-list at communicatefreely.net> wrote:
>
>> Michael Collins wrote:
>> > Are you trying to bridge the current leg (user <--> voicemail) to
>> > another endpoint? If so, how are you doing that? Are you transferring
>> > the leg back into the dialplan for processing?
>> >
>> I may have just answered my own question - quite by accident while working
>> on another problem.
>>
>> I was sending the call back to the dial plan for processing, but I was
>> using the execute_extension
>> application instead of transfer
>>
>> It looks like execute_extension doesn't affect the destination_number
>> variable, whereas transfer
>> changes the destination number, and sets the previously dialed number as
>> RDNIS.
>>
>> Changing my logic to use transfer instead of execute_extension seems to
>> have solved things.
>>
>> -Tim
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Sat Mar 12 00:38:00 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 11 Mar 2011 13:38:00 -0800
Subject: [Freeswitch-users] exec app on originate
In-Reply-To: <AANLkTinoN7tdVDA8e+XLcjFGV-Z1KhP1a3+gLSGNaGOn@mail.gmail.com>
References: <748093B885AD48CE86278841FBEA1159@e1705>
	<AANLkTikmsGve+DMf9RzyyB3wBH0dgpT_1gkG4hNuq7du@mail.gmail.com>
	<AANLkTinoN7tdVDA8e+XLcjFGV-Z1KhP1a3+gLSGNaGOn@mail.gmail.com>
Message-ID: <AANLkTinbgUe0M_MCFZKA96zfErht3pmpw-UhC+sX3fwF@mail.gmail.com>

On Fri, Mar 11, 2011 at 1:04 AM, Pete Kelly <pkelly at gmail.com> wrote:

>
>
> On 25 February 2011 01:04, Anthony Minessale <anthony.minessale at gmail.com>wrote:
>
>> early media
>> if you want after answer do
>>
>> originate {ignore_early_media=true}sofia/gateway/blabla/9999999999
>> &app(hahaha)
>>
>
> Is there any way to get the app launched even earlier? I need to be able to
> act upon a 4XX/5XX class response to the INVITE within the app.
>

The originate API does not work that way. The &app(foo) won't be executed
unless the originate is successful. If you get a 4xx/5xx then the originate
will fail and &app(foo) won't ever get executed. There is no notion of
"execute_on_fail" since a failed call means there is no channel/session to
send through the dialplan or through an application.

Depending on exactly what you're doing you probably need to be using "bgapi
originate ..." and then checking the BACKGROUND_JOB event to see what
happened. Alternatively you could be listening/watching for CDRs and then
acting upon them. Alternatively using mod_xml_cdr is an easy way to receive
notification of failed calls. The channel variable
"proto_specific_hangup_cause" will have the "SIP:4xx" value, and you can
react to it accordingly.

-MC
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From kbdfck at gmail.com  Sat Mar 12 00:38:47 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Sat, 12 Mar 2011 00:38:47 +0300
Subject: [Freeswitch-users] Already hung channels in 'show channels'
 output never go away
In-Reply-To: <AANLkTik_CE66S0FToNMycX+U_WV3m12T=8D4t88BdcKP@mail.gmail.com>
References: <AANLkTi=jiL-aSmkDusoH_9EYRzPZ1ePXkWAQHXn53Np5@mail.gmail.com>
	<AANLkTinUg76_ncbjCwf=W8T5BPSS3FdstGQkkptv-xme@mail.gmail.com>
	<AANLkTimzvf-ZaiaVKCprbHh6JcFx3dXnbaU54T8KL=O6@mail.gmail.com>
	<AANLkTinTApDm-YL5UjqPnc8vaE-e+xpdAXiwaD+DTZhX@mail.gmail.com>
	<AANLkTik_CE66S0FToNMycX+U_WV3m12T=8D4t88BdcKP@mail.gmail.com>
Message-ID: <AANLkTikTQpMHe67cSHr3T9Xr7UFVTu=RWXnNthxaBqCV@mail.gmail.com>

I use cdr_xml only. If the call fails before entering ESL script, seems
there are no stuck records, but they appear after some actions in ESL even
if there was no bridge attempt.

2011/3/11 Anthony Minessale <anthony.minessale at gmail.com>

> that looks like they are all stuck in the CDR module.
> What module are you using?
>
>
>
> On Fri, Mar 11, 2011 at 5:58 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> > I think this is not the case, but anyway, what to do with these hung
> > channels?
> > Maybe I'm doing something wrong while bridging or processing events?
> Maybe
> > unprocessed events can affect channel destroy procedure?
> > 2011/3/11 Avi Marcus <avi at avimarcus.net>
> >>
> >> Regarding ram usage, I'd imagine this is the case:
> >> http://www.linuxatemyram.com/
> >> -Avi
> >>
> >> On Fri, Mar 11, 2011 at 12:33 PM, Dmitry Sytchev <kbdfck at gmail.com>
> wrote:
> >>>
> >>> BTW, uuid_dump 7ae28028-4bc2-11e0-a92f-452f3d3d66ea returns with -ERR
> No
> >>> such channel.
> >>>
> >>> 2011/3/11 Dmitry Sytchev <kbdfck at gmail.com>
> >>>>
> >>>> Hi All
> >>>> I'm using Perl ESL outbound script to bridge incoming call to sip
> >>>> endpoint. I'm doing execute("bridge","sofia/user/somebody"), then
> processing
> >>>> events like CHANNEL_HANGUP and EXECUTE_COMPLETE to determine bridge
> result.
> >>>> Everything works fine, but original incoming call channel is never
> removed
> >>>> from list:
> >>>> After few calls I see original incoming channels in 'show channels'
> >>>> output:
> >>>> 7ae28028-4bc2-11e0-a92f-452f3d3d66ea,inbound,2011-03-11
> >>>> 12:32:27,1299835947,sofia/external/xxxx at 85.x.x.199
> ,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,
> xxx.xxx.ru,,,HANGUP,,,,
> >>>> 883c9eb6-4bc2-11e0-a935-452f3d3d66ea,inbound,2011-03-11
> >>>> 12:32:50,1299835970,sofia/external/xxxx at 85.x.x.199
> ,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,
> xxx.xxx.ru,,,HANGUP,,,,
> >>>> 9f6a0f92-4bc2-11e0-a93b-452f3d3d66ea,inbound,2011-03-11
> >>>> 12:33:29,1299836009,sofia/external/xxxx at 85.x.x.199
> ,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,
> xxx.xxx.ru,,,HANGUP,,,
> >>>> Also, when I try to stop freeswitch i see these messages on console:
> >>>> 2011-03-11 15:28:22.517063 [CRIT] sofia.c:1632 Waiting for 13
> session(s)
> >>>> 2011-03-11 15:28:22.701034 [CRIT] sofia.c:1632 Waiting for 3
> session(s)
> >>>> 2011-03-11 15:28:23.517064 [CRIT] sofia.c:1632 Waiting for 13
> session(s)
> >>>> 2011-03-11 15:28:23.701033 [CRIT] sofia.c:1632 Waiting for 3
> session(s)
> >>>> 2011-03-11 15:28:24.517062 [CRIT] sofia.c:1632 Waiting for 13
> session(s)
> >>>> 2011-03-11 15:28:24.705036 [CRIT] sofia.c:1632 Waiting for 3
> session(s)
> >>>> 2011-03-11 15:28:25.517064 [CRIT] sofia.c:1632 Waiting for 13
> session(s)
> >>>> 2011-03-11 15:28:25.709031 [CRIT] sofia.c:1632 Waiting for 3
> session(s)
> >>>> 2011-03-11 15:28:26.517065 [CRIT] sofia.c:1632 Waiting for 13
> session(s)
> >>>>
> >>>> Why these channels are not removed from list? I also noticed that
> memory
> >>>> consumption by freeswitch process constantly grows call by call.
> >>>>
> >>>> --
> >>>> Best regards,
> >>>>
> >>>> Dmitry Sytchev,
> >>>> IT Engineer
> >>>
> >>>
> >>>
> >>> --
> >>> Best regards,
> >>>
> >>> Dmitry Sytchev,
> >>> IT Engineer
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > Best regards,
> >
> > Dmitry Sytchev,
> > IT Engineer
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From mario_fs at mgtech.com  Sat Mar 12 00:39:42 2011
From: mario_fs at mgtech.com (Mario G)
Date: Fri, 11 Mar 2011 13:39:42 -0800
Subject: [Freeswitch-users] Can you use ext-ip.. autonat: with -nonat ?
Message-ID: <2DC8F9F6-253E-47A3-9651-D652A911FB80@mgtech.com>

Changed router now only 1 out of 5 calls come in. I previously had "ext-sip/rtp-ip:1.2.3.4" and -nonat on another router but having major issues on new one. Wiki does not answer this question:

Can you use "exp-rtp/sip-ip"autonat:1.2.3.4" when freeswitch is started with -nonat ? Outbound is fine, inbound fails 80 percent if the time.

I really wanted use ALG but when I did all worked fine except call dropped while conversing.



From msc at freeswitch.org  Sat Mar 12 00:41:58 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 11 Mar 2011 13:41:58 -0800
Subject: [Freeswitch-users] FreeSwitch for Windows
In-Reply-To: <4D77F344.3050207@gmail.com>
References: <E04386CF-006E-4F87-A2B0-FA6C810A34B7@mgtech.com>
	<022B01F7-0577-457B-94C8-D6F47B14E1F0@mgtech.com>
	<4D77F344.3050207@gmail.com>
Message-ID: <AANLkTim=fsgSuVci4wn_hwc96DOD+6qF0RfE0mNMpxpZ@mail.gmail.com>

If you plan on using TDM then I'd recommend you stay away from Windows.
Sangoma and Digium/clone cards are all geared toward a Linux environment.
Stay away from the exotic OSes and use something simple like CentOS 5.5 or
Debian Lenny.

-MC

On Wed, Mar 9, 2011 at 1:38 PM, Fernando Berretta <
fernando.berretta at gmail.com> wrote:

> Hi,
>
> I'm newbie in FreeSwitch and would like to know if freeswitch could be
> used for production under Windows platforms ? telephony boards like
> Digium / OpenVox, Sangoma, etc are going to work ok under Windows ? is
> there any problem / limitation ? E1 MFCR2 Trunk works ?
>
> Thanks for your time,
> Best Regards,
> Fernando
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fernando.berretta at gmail.com  Sat Mar 12 00:54:58 2011
From: fernando.berretta at gmail.com (Fernando Berretta)
Date: Fri, 11 Mar 2011 18:54:58 -0300
Subject: [Freeswitch-users] FreeSwitch for Windows
In-Reply-To: <AANLkTim=fsgSuVci4wn_hwc96DOD+6qF0RfE0mNMpxpZ@mail.gmail.com>
References: <E04386CF-006E-4F87-A2B0-FA6C810A34B7@mgtech.com>	<022B01F7-0577-457B-94C8-D6F47B14E1F0@mgtech.com>	<4D77F344.3050207@gmail.com>
	<AANLkTim=fsgSuVci4wn_hwc96DOD+6qF0RfE0mNMpxpZ@mail.gmail.com>
Message-ID: <4D7A9A32.9090508@gmail.com>

Michael,

Thanks for your valuable recommendation

Best Regards,
Fernando

On 3/11/2011 6:41 PM, Michael Collins wrote:
> If you plan on using TDM then I'd recommend you stay away from 
> Windows. Sangoma and Digium/clone cards are all geared toward a Linux 
> environment. Stay away from the exotic OSes and use something simple 
> like CentOS 5.5 or Debian Lenny.
>
> -MC
>
> On Wed, Mar 9, 2011 at 1:38 PM, Fernando Berretta 
> <fernando.berretta at gmail.com <mailto:fernando.berretta at gmail.com>> wrote:
>
>     Hi,
>
>     I'm newbie in FreeSwitch and would like to know if freeswitch could be
>     used for production under Windows platforms ? telephony boards like
>     Digium / OpenVox, Sangoma, etc are going to work ok under Windows ? is
>     there any problem / limitation ? E1 MFCR2 Trunk works ?
>
>     Thanks for your time,
>     Best Regards,
>     Fernando
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From anthony.minessale at gmail.com  Sat Mar 12 02:02:48 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 11 Mar 2011 17:02:48 -0600
Subject: [Freeswitch-users] Already hung channels in 'show channels'
 output never go away
In-Reply-To: <AANLkTikTQpMHe67cSHr3T9Xr7UFVTu=RWXnNthxaBqCV@mail.gmail.com>
References: <AANLkTi=jiL-aSmkDusoH_9EYRzPZ1ePXkWAQHXn53Np5@mail.gmail.com>
	<AANLkTinUg76_ncbjCwf=W8T5BPSS3FdstGQkkptv-xme@mail.gmail.com>
	<AANLkTimzvf-ZaiaVKCprbHh6JcFx3dXnbaU54T8KL=O6@mail.gmail.com>
	<AANLkTinTApDm-YL5UjqPnc8vaE-e+xpdAXiwaD+DTZhX@mail.gmail.com>
	<AANLkTik_CE66S0FToNMycX+U_WV3m12T=8D4t88BdcKP@mail.gmail.com>
	<AANLkTikTQpMHe67cSHr3T9Xr7UFVTu=RWXnNthxaBqCV@mail.gmail.com>
Message-ID: <AANLkTimAJ2ULQZk55B63zBoVnKK+E9nrQeCR8tP5CuFQ@mail.gmail.com>

this sounds odd,
are you on the latest GIT build?

try doing a gcore on the box when its like that and run gdb on it and
do thread apply all bt


On Fri, Mar 11, 2011 at 3:38 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> I use cdr_xml only. If the call fails before entering ESL script, seems
> there are no stuck records, but they appear after some actions in ESL even
> if there was no bridge attempt.
>
> 2011/3/11 Anthony Minessale <anthony.minessale at gmail.com>
>>
>> that looks like they are all stuck in the CDR module.
>> What module are you using?
>>
>>
>>
>> On Fri, Mar 11, 2011 at 5:58 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>> > I think this is not the case, but anyway, what to do with these hung
>> > channels?
>> > Maybe I'm doing something wrong while bridging or processing events?
>> > Maybe
>> > unprocessed events can affect channel destroy procedure?
>> > 2011/3/11 Avi Marcus <avi at avimarcus.net>
>> >>
>> >> Regarding ram usage, I'd imagine this is the case:
>> >> http://www.linuxatemyram.com/
>> >> -Avi
>> >>
>> >> On Fri, Mar 11, 2011 at 12:33 PM, Dmitry Sytchev <kbdfck at gmail.com>
>> >> wrote:
>> >>>
>> >>> BTW, uuid_dump?7ae28028-4bc2-11e0-a92f-452f3d3d66ea returns with -ERR
>> >>> No
>> >>> such channel.
>> >>>
>> >>> 2011/3/11 Dmitry Sytchev <kbdfck at gmail.com>
>> >>>>
>> >>>> Hi All
>> >>>> I'm using Perl ESL outbound script to bridge incoming call to sip
>> >>>> endpoint. I'm doing execute("bridge","sofia/user/somebody"), then
>> >>>> processing
>> >>>> events like CHANNEL_HANGUP and EXECUTE_COMPLETE to determine bridge
>> >>>> result.
>> >>>> Everything works fine, but original incoming call channel is never
>> >>>> removed
>> >>>> from list:
>> >>>> After few calls I see original incoming channels in 'show channels'
>> >>>> output:
>> >>>> 7ae28028-4bc2-11e0-a92f-452f3d3d66ea,inbound,2011-03-11
>> >>>>
>> >>>> 12:32:27,1299835947,sofia/external/xxxx at 85.x.x.199,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,xxx.xxx.ru,,,HANGUP,,,,
>> >>>> 883c9eb6-4bc2-11e0-a935-452f3d3d66ea,inbound,2011-03-11
>> >>>>
>> >>>> 12:32:50,1299835970,sofia/external/xxxx at 85.x.x.199,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,xxx.xxx.ru,,,HANGUP,,,,
>> >>>> 9f6a0f92-4bc2-11e0-a93b-452f3d3d66ea,inbound,2011-03-11
>> >>>>
>> >>>> 12:33:29,1299836009,sofia/external/xxxx at 85.x.x.199,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,xxx.xxx.ru,,,HANGUP,,,
>> >>>> Also, when I try to stop freeswitch i see these messages on console:
>> >>>> 2011-03-11 15:28:22.517063 [CRIT] sofia.c:1632 Waiting for 13
>> >>>> session(s)
>> >>>> 2011-03-11 15:28:22.701034 [CRIT] sofia.c:1632 Waiting for 3
>> >>>> session(s)
>> >>>> 2011-03-11 15:28:23.517064 [CRIT] sofia.c:1632 Waiting for 13
>> >>>> session(s)
>> >>>> 2011-03-11 15:28:23.701033 [CRIT] sofia.c:1632 Waiting for 3
>> >>>> session(s)
>> >>>> 2011-03-11 15:28:24.517062 [CRIT] sofia.c:1632 Waiting for 13
>> >>>> session(s)
>> >>>> 2011-03-11 15:28:24.705036 [CRIT] sofia.c:1632 Waiting for 3
>> >>>> session(s)
>> >>>> 2011-03-11 15:28:25.517064 [CRIT] sofia.c:1632 Waiting for 13
>> >>>> session(s)
>> >>>> 2011-03-11 15:28:25.709031 [CRIT] sofia.c:1632 Waiting for 3
>> >>>> session(s)
>> >>>> 2011-03-11 15:28:26.517065 [CRIT] sofia.c:1632 Waiting for 13
>> >>>> session(s)
>> >>>>
>> >>>> Why these channels are not removed from list? I also noticed that
>> >>>> memory
>> >>>> consumption by freeswitch process constantly grows call by call.
>> >>>>
>> >>>> --
>> >>>> Best regards,
>> >>>>
>> >>>> Dmitry Sytchev,
>> >>>> IT Engineer
>> >>>
>> >>>
>> >>>
>> >>> --
>> >>> Best regards,
>> >>>
>> >>> Dmitry Sytchev,
>> >>> IT Engineer
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> > --
>> > Best regards,
>> >
>> > Dmitry Sytchev,
>> > IT Engineer
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 mstockton at harqen.com  Sat Mar 12 01:19:46 2011
From: mstockton at harqen.com (Matt Stockton)
Date: Fri, 11 Mar 2011 16:19:46 -0600
Subject: [Freeswitch-users] session:record / prevent DTMF tones from being
	recorded
Message-ID: <AANLkTik7SxDLCq0QYnp6twJCsajxrcDP2p0qpZuWJdtZ@mail.gmail.com>

Hi all,

I have a freeswitch lua script that is recording and I want the recording to
be terminated when the user presses the '#' key. That is all working fine
for me, but the recording always has the DTMF tone at the end of it.

I am wondering, is there a way to instruct freeswitch to not record the DTMF
tones? Any help is appreciated. For reference, here are some excerpts from
my lua script:

function inputHook(s, type, obj)

  freeswitch.consoleLog("INFO", "On input");

  if (inRecording and type == "dtmf" and obj['digit'] == '#') then

      inRecording = false;

      return "break";

  end

end

.....

session:setInputCallback("inputHook", "");

.....

session:recordFile(recordFilePath, 202, 500, 25);


Thanks in advance
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From raison at chatsubo.net  Sat Mar 12 01:44:49 2011
From: raison at chatsubo.net (Kevin Raison)
Date: Fri, 11 Mar 2011 14:44:49 -0800
Subject: [Freeswitch-users] inbound dtmf issue after upgrade
Message-ID: <4D7AA5E1.1050402@chatsubo.net>

I upgraded a freeswitch setup from 1.0.4 to 1.0.6 yesterday, and also
upgraded my wanpipe drivers from 3.4.9 to 3.5.10.  I am using openzap
with a sangoma card.  Before the upgrade, the following stanza would
correctly send a caller to a mod_event_socket-based application that I
have running on another machine and that caller's dtmf would flow
through to the app:

<extension name="Kevin_test">
        <condition field="destination_number" expression="^XXXXXXXXXX$">
                <action application="set" data="ringback=${us-ring}"/>
                <action application="set"
data="transfer_ringback=$${hold_music}"/>
                <action application="bridge"
data="sofia/internal/5000 at prod.XXXXXXX.com"/>
        </condition>
</extension>

After the upgrade, everything but inbound dtmf works as it should, but
the only way that I can get inbound dtmf to work is by adding
	<action application="start_dtmf"/>
to the stanza.  This is not ideal and I would like to go back to not
using in-band dtmf if possible.

In my log files, I see absolutely no dtmf being registered when a caller
calls in.  However, if someone calls out from a desk phone and then
transfers the person that they called to
sofia/internal/5000 at prod.XXXXXXX.com, dtmf works just fine.

Any ideas?

Thanks!
Kevin


From cmrienzo at gmail.com  Sat Mar 12 02:31:41 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Fri, 11 Mar 2011 18:31:41 -0500
Subject: [Freeswitch-users] session:record / prevent DTMF tones from
	being recorded
In-Reply-To: <AANLkTik7SxDLCq0QYnp6twJCsajxrcDP2p0qpZuWJdtZ@mail.gmail.com>
References: <AANLkTik7SxDLCq0QYnp6twJCsajxrcDP2p0qpZuWJdtZ@mail.gmail.com>
Message-ID: <AANLkTikWgK8WnhriRMz7rDokW-HbXPGcviowH5Mr8iJg@mail.gmail.com>

Trim some of the audio off the end of the file after the recording is done.



On Fri, Mar 11, 2011 at 5:19 PM, Matt Stockton <mstockton at harqen.com> wrote:

> Hi all,
>
> I have a freeswitch lua script that is recording and I want the recording
> to be terminated when the user presses the '#' key. That is all working fine
> for me, but the recording always has the DTMF tone at the end of it.
>
> I am wondering, is there a way to instruct freeswitch to not record the
> DTMF tones? Any help is appreciated. For reference, here are some excerpts
> from my lua script:
>
> function inputHook(s, type, obj)
>
>   freeswitch.consoleLog("INFO", "On input");
>
>   if (inRecording and type == "dtmf" and obj['digit'] == '#') then
>
>       inRecording = false;
>
>       return "break";
>
>   end
>
> end
>
> .....
>
> session:setInputCallback("inputHook", "");
>
> .....
>
> session:recordFile(recordFilePath, 202, 500, 25);
>
>
> Thanks in advance
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sunwood360 at gmail.com  Sat Mar 12 07:52:28 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Fri, 11 Mar 2011 20:52:28 -0800
Subject: [Freeswitch-users] freeswitch -nc -nosql
In-Reply-To: <AANLkTi=O=ZpYt_3MDwB2t0jHDG3DQ0qgx4apA+1WUGyM@mail.gmail.com>
References: <1298688216720-6066718.post@n2.nabble.com>
	<1299524988820-6098582.post@n2.nabble.com>
	<AANLkTik6c5rHYnr6cxmsBzRCcc94VL=jmYtud29TVBzc@mail.gmail.com>
	<AANLkTindLhrtsbVmifg699TyQtv+JOYa52_Tj1CqkwVs@mail.gmail.com>
	<AANLkTi=O=ZpYt_3MDwB2t0jHDG3DQ0qgx4apA+1WUGyM@mail.gmail.com>
Message-ID: <AANLkTimXwMcXyZW=C+A09ynezyR3fQApU0Jj_RKzOsPC@mail.gmail.com>

Horray, works beautifully.

now I can forget the lowsy openwrt platform and run fs on my dockstar
debian.
On Mar 10, 2011 3:34 PM, "envelopes envelopes" <sunwood360 at gmail.com> wrote:
> Awesome, I will try my debian box tonight.
> On Mar 10, 2011 9:38 AM, "Anthony Minessale" <anthony.minessale at gmail.com>
> wrote:
>> try latest it'd fixed now.
>>
>> integer boundary issue, switch_time_t vs regular time_t
>>
>> On Mon, Mar 7, 2011 at 1:23 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>> Will you please explain to me why you keep ignoring my request for
> backtraces?
>>> There is a bug open already http://jira.freeswitch.org/browse/FS-3126
>>> There is even a backtrace which can be very useful , now if you posted
>>> your findings there we might even figure it out.....
>>>
>>> Are you not receiving my many pleas to use JIRA for working on bugs?
>>>
>>>
>>> On Mon, Mar 7, 2011 at 1:09 PM, mazilo <Nabble at slickdeals.endjunk.com>
> wrote:
>>>> Finaly, I found out that FreeSWITCH Version 1.0.head (git-88d410d
> 2011-02-11
>>>> 20-15-06 -0600) is the last version that works with SQL. Starting from
> the
>>>> version after this one with a commit
>>>> a2c0da53f368f0b11340c3a72814c93b182753b7 crashes if -nosql switch is
> called
>>>> when freeswitch is launched. Here is the git log pertaining to the two
>>>> commits for your perusal and hope FS developers will be able to
localize
> why
>>>> adding centralized registration db to core db and use it from mod_sofia
>>>> causes SQL to crash on ARM platform:
>>>>
>>>> commit a2c0da53f368f0b11340c3a72814c93b182753b7
>>>> Author: Anthony Minessale
>>>> Date: Fri Feb 11 23:10:12 2011 -0600
>>>>
>>>> add centralized registration db to core db and use it from mod_sofia
>>>>
>>>> commit 88d410d31485d13911f0958af5a73f1f6f49a454
>>>> Author: Anthony Minessale
>>>> Date: Fri Feb 11 20:15:06 2011 -0600
>>>>
>>>> fix uuid_jitterbuffer edge case debugging a non-existant jb causing a
>>>> seg
>>>>
>>>>
>>>> -----
>>>> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
>>>> --
>>>> View this message in context:
>
http://freeswitch-users.2379917.n2.nabble.com/freeswitch-nc-nosql-tp6066718p6098582.html
>>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> 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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
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From grsingh750 at gmail.com  Sat Mar 12 10:23:58 2011
From: grsingh750 at gmail.com (guru singh)
Date: Sat, 12 Mar 2011 12:53:58 +0530
Subject: [Freeswitch-users] inbound dtmf issue after upgrade
In-Reply-To: <4D7AA5E1.1050402@chatsubo.net>
References: <4D7AA5E1.1050402@chatsubo.net>
Message-ID: <AANLkTikTjx52euT=P7O-eXC+PCigyQGWLCHxjkQyOyL6@mail.gmail.com>

Hi Kevin,

You should look at mod_freetdm. It replaces openzap.
http://wiki.freeswitch.org/wiki/FreeTDM

guru

On Sat, Mar 12, 2011 at 4:14 AM, Kevin Raison <raison at chatsubo.net> wrote:
> I upgraded a freeswitch setup from 1.0.4 to 1.0.6 yesterday, and also
> upgraded my wanpipe drivers from 3.4.9 to 3.5.10. ?I am using openzap
> with a sangoma card. ?Before the upgrade, the following stanza would
> correctly send a caller to a mod_event_socket-based application that I
> have running on another machine and that caller's dtmf would flow
> through to the app:
>
> <extension name="Kevin_test">
> ? ? ? ?<condition field="destination_number" expression="^XXXXXXXXXX$">
> ? ? ? ? ? ? ? ?<action application="set" data="ringback=${us-ring}"/>
> ? ? ? ? ? ? ? ?<action application="set"
> data="transfer_ringback=$${hold_music}"/>
> ? ? ? ? ? ? ? ?<action application="bridge"
> data="sofia/internal/5000 at prod.XXXXXXX.com"/>
> ? ? ? ?</condition>
> </extension>
>
> After the upgrade, everything but inbound dtmf works as it should, but
> the only way that I can get inbound dtmf to work is by adding
> ? ? ? ?<action application="start_dtmf"/>
> to the stanza. ?This is not ideal and I would like to go back to not
> using in-band dtmf if possible.
>
> In my log files, I see absolutely no dtmf being registered when a caller
> calls in. ?However, if someone calls out from a desk phone and then
> transfers the person that they called to
> sofia/internal/5000 at prod.XXXXXXX.com, dtmf works just fine.
>
> Any ideas?
>
> Thanks!
> Kevin
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From raison at chatsubo.net  Sat Mar 12 11:19:37 2011
From: raison at chatsubo.net (Kevin Raison)
Date: Sat, 12 Mar 2011 00:19:37 -0800
Subject: [Freeswitch-users] inbound dtmf issue after upgrade
In-Reply-To: <AANLkTikTjx52euT=P7O-eXC+PCigyQGWLCHxjkQyOyL6@mail.gmail.com>
References: <4D7AA5E1.1050402@chatsubo.net>
	<AANLkTikTjx52euT=P7O-eXC+PCigyQGWLCHxjkQyOyL6@mail.gmail.com>
Message-ID: <4D7B2C99.2020401@chatsubo.net>

Thanks for the pointer, guru;  and while I will definitely move toward
that in the future, I am hoping to remedy my current situation without
introducing too much risk to my customer, and unfortunately changing out
oz for freetdm is a little risky in the eyes of the customer.  So, are
there any other possibilities that don't involve switching out oz?  I
know that wanpipe is setup to handle dtmf properly:
  TDMV_HW_DTMF    = YES

Is there something specific in the freeswitch config that needs enabling
(other than using the "start_dtmf" option)?

Thanks,
Kevin


On 03/11/2011 11:23 PM, guru singh wrote:
> Hi Kevin,
> 
> You should look at mod_freetdm. It replaces openzap.
> http://wiki.freeswitch.org/wiki/FreeTDM
> 
> guru
> 
> On Sat, Mar 12, 2011 at 4:14 AM, Kevin Raison <raison at chatsubo.net> wrote:
>> I upgraded a freeswitch setup from 1.0.4 to 1.0.6 yesterday, and also
>> upgraded my wanpipe drivers from 3.4.9 to 3.5.10.  I am using openzap
>> with a sangoma card.  Before the upgrade, the following stanza would
>> correctly send a caller to a mod_event_socket-based application that I
>> have running on another machine and that caller's dtmf would flow
>> through to the app:
>>
>> <extension name="Kevin_test">
>>        <condition field="destination_number" expression="^XXXXXXXXXX$">
>>                <action application="set" data="ringback=${us-ring}"/>
>>                <action application="set"
>> data="transfer_ringback=$${hold_music}"/>
>>                <action application="bridge"
>> data="sofia/internal/5000 at prod.XXXXXXX.com"/>
>>        </condition>
>> </extension>
>>
>> After the upgrade, everything but inbound dtmf works as it should, but
>> the only way that I can get inbound dtmf to work is by adding
>>        <action application="start_dtmf"/>
>> to the stanza.  This is not ideal and I would like to go back to not
>> using in-band dtmf if possible.
>>
>> In my log files, I see absolutely no dtmf being registered when a caller
>> calls in.  However, if someone calls out from a desk phone and then
>> transfers the person that they called to
>> sofia/internal/5000 at prod.XXXXXXX.com, dtmf works just fine.
>>
>> Any ideas?
>>
>> Thanks!
>> Kevin
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From boris at tagnet.ru  Sat Mar 12 14:40:08 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 12 Mar 2011 16:40:08 +0500
Subject: [Freeswitch-users] How to timeout att_xfer?
Message-ID: <4D7B5B98.3060608@tagnet.ru>

Hello!

     Is there a possibility to timeout att_xfer application? For example 
-> A calls B, B does att_xfer to C, but C doesn't answer for a 10 
seconds so B is bridged again to A (of course if B is still off hook).

-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From anthony.minessale at gmail.com  Sat Mar 12 16:54:26 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 12 Mar 2011 07:54:26 -0600
Subject: [Freeswitch-users] How to timeout att_xfer?
In-Reply-To: <4D7B5B98.3060608@tagnet.ru>
References: <4D7B5B98.3060608@tagnet.ru>
Message-ID: <AANLkTimehsD42iPNByQ1zq0YCRwE44ct+TgDMvG9mSeA@mail.gmail.com>

add {origination_timeout=N} before the dial string you pass into it.


On Sat, Mar 12, 2011 at 5:40 AM, Boris Kovalenko <boris at tagnet.ru> wrote:
> Hello!
>
> ? ? Is there a possibility to timeout att_xfer application? For example
> -> A calls B, B does att_xfer to C, but C doesn't answer for a 10
> seconds so B is bridged again to A (of course if B is still off hook).
>
> --
> ? ?????????,
> ? ????? ?????????
> ? ??? "??????"
> ? (3435) 494991
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 xyangni at gmail.com  Sat Mar 12 16:58:09 2011
From: xyangni at gmail.com (Yihui Li)
Date: Sat, 12 Mar 2011 13:58:09 +0000
Subject: [Freeswitch-users] Multiple Registrations not working
Message-ID: <AANLkTinKQ673wyyK-BwKpBv1JN287ssa17sLPKs=vb3o@mail.gmail.com>

Hi,

I am trying to use Multiple Registrations feature in bridge, but found it
not always working when there are more than 1 bridge targets like below:

<action application="bridge" data="user/1015,user/1016" />

where 1015 is registered on 2 PC.

To simplify my case, I have also tested it with CLI command.
Test 1,
originate user/1015 &parks
both PC are ringing,

Test 2,
originate user/1015,user/1016 &parks
only 1 PC is ringing

in internal.xml, I added
<param name="multiple-registrations" value="true"/>

I am running git-head on ubuntu.
Can anyone help with this issue? Thanks.

Eric
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From boris at tagnet.ru  Sat Mar 12 17:05:28 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 12 Mar 2011 19:05:28 +0500
Subject: [Freeswitch-users] How to timeout att_xfer?
In-Reply-To: <AANLkTimehsD42iPNByQ1zq0YCRwE44ct+TgDMvG9mSeA@mail.gmail.com>
References: <4D7B5B98.3060608@tagnet.ru>
	<AANLkTimehsD42iPNByQ1zq0YCRwE44ct+TgDMvG9mSeA@mail.gmail.com>
Message-ID: <4D7B7DA8.7080506@tagnet.ru>

Hello!

     origination_timeout or originate_timeout? Indeed, have tried both 
without success. Freeswitch FreeSWITCH Version 1.0.head (git-1c95ad9 
2011-01-20 22-43-50 -0300)
My extension:
<extension name="attented_xfer">
<condition field="destination_number" expression="^attented_xfer$">
<action application="read" data="3 3 
$${sound_prefix}/ivr/8000/ivr-enter_ext.wav attxfer
_digits 10000 #"/>
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="origination_cancel_key=#"/>
<action application="set" data="transfer_ringback=$${hold_music}"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="att_xfer" 
data="{originate_timeout=5}loopback/${attxfer_digits}/${c
ontext}/${dialplan}"/>
<!--<action application="info" data="ALERT"/>-->
</condition>
</extension>

> add {origination_timeout=N} before the dial string you pass into it.
>
>
> On Sat, Mar 12, 2011 at 5:40 AM, Boris Kovalenko<boris at tagnet.ru>  wrote:
>> Hello!
>>
>>      Is there a possibility to timeout att_xfer application? For example
>> ->  A calls B, B does att_xfer to C, but C doesn't answer for a 10
>> seconds so B is bridged again to A (of course if B is still off hook).
>>
>> --
>> ? ?????????,
>>    ????? ?????????
>>    ??? "??????"
>>    (3435) 494991
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From anthony.minessale at gmail.com  Sat Mar 12 17:07:18 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 12 Mar 2011 08:07:18 -0600
Subject: [Freeswitch-users] How to timeout att_xfer?
In-Reply-To: <4D7B7DA8.7080506@tagnet.ru>
References: <4D7B5B98.3060608@tagnet.ru>
	<AANLkTimehsD42iPNByQ1zq0YCRwE44ct+TgDMvG9mSeA@mail.gmail.com>
	<4D7B7DA8.7080506@tagnet.ru>
Message-ID: <AANLkTi=wX=Jt=PuokT=ZP9xYY9vLi4zvX4Sq9V5XQyFc@mail.gmail.com>

orignation_timeout inside {} not by set app


On Sat, Mar 12, 2011 at 8:05 AM, Boris Kovalenko <boris at tagnet.ru> wrote:
> Hello!
>
> ? ? origination_timeout or originate_timeout? Indeed, have tried both
> without success. Freeswitch FreeSWITCH Version 1.0.head (git-1c95ad9
> 2011-01-20 22-43-50 -0300)
> My extension:
> <extension name="attented_xfer">
> <condition field="destination_number" expression="^attented_xfer$">
> <action application="read" data="3 3
> $${sound_prefix}/ivr/8000/ivr-enter_ext.wav attxfer
> _digits 10000 #"/>
> <action application="set" data="continue_on_fail=true"/>
> <action application="set" data="origination_cancel_key=#"/>
> <action application="set" data="transfer_ringback=$${hold_music}"/>
> <action application="set" data="hangup_after_bridge=true"/>
> <action application="att_xfer"
> data="{originate_timeout=5}loopback/${attxfer_digits}/${c
> ontext}/${dialplan}"/>
> <!--<action application="info" data="ALERT"/>-->
> </condition>
> </extension>
>
>> add {origination_timeout=N} before the dial string you pass into it.
>>
>>
>> On Sat, Mar 12, 2011 at 5:40 AM, Boris Kovalenko<boris at tagnet.ru> ?wrote:
>>> Hello!
>>>
>>> ? ? ?Is there a possibility to timeout att_xfer application? For example
>>> -> ?A calls B, B does att_xfer to C, but C doesn't answer for a 10
>>> seconds so B is bridged again to A (of course if B is still off hook).
>>>
>>> --
>>> ? ?????????,
>>> ? ?????? ?????????
>>> ? ???? "??????"
>>> ? ?(3435) 494991
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
>
> --
> ? ?????????,
> ? ????? ?????????
> ? ??? "??????"
> ? (3435) 494991
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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  Sat Mar 12 17:10:21 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 12 Mar 2011 08:10:21 -0600
Subject: [Freeswitch-users] How to timeout att_xfer?
In-Reply-To: <AANLkTi=wX=Jt=PuokT=ZP9xYY9vLi4zvX4Sq9V5XQyFc@mail.gmail.com>
References: <4D7B5B98.3060608@tagnet.ru>
	<AANLkTimehsD42iPNByQ1zq0YCRwE44ct+TgDMvG9mSeA@mail.gmail.com>
	<4D7B7DA8.7080506@tagnet.ru>
	<AANLkTi=wX=Jt=PuokT=ZP9xYY9vLi4zvX4Sq9V5XQyFc@mail.gmail.com>
Message-ID: <AANLkTinkJA6eEvJ3V9Vr7V9BhMTMqeyw=RLem8wKDjUv@mail.gmail.com>

frick,

yes {originate_timeout=N}

its early still here ....

It will work, but you need to use ignore_early_media=true with it
otherwise you also need to set bridge_answer_timeout on the inbound
leg to the same value.

{ignore_early_media=true,originate_timeout=N}


On Sat, Mar 12, 2011 at 8:07 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> orignation_timeout inside {} not by set app
>
>
> On Sat, Mar 12, 2011 at 8:05 AM, Boris Kovalenko <boris at tagnet.ru> wrote:
>> Hello!
>>
>> ? ? origination_timeout or originate_timeout? Indeed, have tried both
>> without success. Freeswitch FreeSWITCH Version 1.0.head (git-1c95ad9
>> 2011-01-20 22-43-50 -0300)
>> My extension:
>> <extension name="attented_xfer">
>> <condition field="destination_number" expression="^attented_xfer$">
>> <action application="read" data="3 3
>> $${sound_prefix}/ivr/8000/ivr-enter_ext.wav attxfer
>> _digits 10000 #"/>
>> <action application="set" data="continue_on_fail=true"/>
>> <action application="set" data="origination_cancel_key=#"/>
>> <action application="set" data="transfer_ringback=$${hold_music}"/>
>> <action application="set" data="hangup_after_bridge=true"/>
>> <action application="att_xfer"
>> data="{originate_timeout=5}loopback/${attxfer_digits}/${c
>> ontext}/${dialplan}"/>
>> <!--<action application="info" data="ALERT"/>-->
>> </condition>
>> </extension>
>>
>>> add {origination_timeout=N} before the dial string you pass into it.
>>>
>>>
>>> On Sat, Mar 12, 2011 at 5:40 AM, Boris Kovalenko<boris at tagnet.ru> ?wrote:
>>>> Hello!
>>>>
>>>> ? ? ?Is there a possibility to timeout att_xfer application? For example
>>>> -> ?A calls B, B does att_xfer to C, but C doesn't answer for a 10
>>>> seconds so B is bridged again to A (of course if B is still off hook).
>>>>
>>>> --
>>>> ? ?????????,
>>>> ? ?????? ?????????
>>>> ? ???? "??????"
>>>> ? ?(3435) 494991
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>
>>
>> --
>> ? ?????????,
>> ? ????? ?????????
>> ? ??? "??????"
>> ? (3435) 494991
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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 boris at tagnet.ru  Sat Mar 12 17:17:03 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 12 Mar 2011 19:17:03 +0500
Subject: [Freeswitch-users] How to timeout att_xfer?
In-Reply-To: <AANLkTinkJA6eEvJ3V9Vr7V9BhMTMqeyw=RLem8wKDjUv@mail.gmail.com>
References: <4D7B5B98.3060608@tagnet.ru>	<AANLkTimehsD42iPNByQ1zq0YCRwE44ct+TgDMvG9mSeA@mail.gmail.com>	<4D7B7DA8.7080506@tagnet.ru>	<AANLkTi=wX=Jt=PuokT=ZP9xYY9vLi4zvX4Sq9V5XQyFc@mail.gmail.com>
	<AANLkTinkJA6eEvJ3V9Vr7V9BhMTMqeyw=RLem8wKDjUv@mail.gmail.com>
Message-ID: <4D7B805F.1020700@tagnet.ru>

Hello!

     Yes, it works now! But there is another problem -> while A is 
hearing MOH, the B is hearing nothing... Is there a way for B to hear 
MOH or progress tone?
> frick,
>
> yes {originate_timeout=N}
>
> its early still here ....
>
> It will work, but you need to use ignore_early_media=true with it
> otherwise you also need to set bridge_answer_timeout on the inbound
> leg to the same value.
>
> {ignore_early_media=true,originate_timeout=N}
>
>
> On Sat, Mar 12, 2011 at 8:07 AM, Anthony Minessale
> <anthony.minessale at gmail.com>  wrote:
>> orignation_timeout inside {} not by set app
>>
>>
>> On Sat, Mar 12, 2011 at 8:05 AM, Boris Kovalenko<boris at tagnet.ru>  wrote:
>>> Hello!
>>>
>>>      origination_timeout or originate_timeout? Indeed, have tried both
>>> without success. Freeswitch FreeSWITCH Version 1.0.head (git-1c95ad9
>>> 2011-01-20 22-43-50 -0300)
>>> My extension:
>>> <extension name="attented_xfer">
>>> <condition field="destination_number" expression="^attented_xfer$">
>>> <action application="read" data="3 3
>>> $${sound_prefix}/ivr/8000/ivr-enter_ext.wav attxfer
>>> _digits 10000 #"/>
>>> <action application="set" data="continue_on_fail=true"/>
>>> <action application="set" data="origination_cancel_key=#"/>
>>> <action application="set" data="transfer_ringback=$${hold_music}"/>
>>> <action application="set" data="hangup_after_bridge=true"/>
>>> <action application="att_xfer"
>>> data="{originate_timeout=5}loopback/${attxfer_digits}/${c
>>> ontext}/${dialplan}"/>
>>> <!--<action application="info" data="ALERT"/>-->
>>> </condition>
>>> </extension>
>>>
>>>> add {origination_timeout=N} before the dial string you pass into it.
>>>>
>>>>
>>>> On Sat, Mar 12, 2011 at 5:40 AM, Boris Kovalenko<boris at tagnet.ru>    wrote:
>>>>> Hello!
>>>>>
>>>>>       Is there a possibility to timeout att_xfer application? For example
>>>>> ->    A calls B, B does att_xfer to C, but C doesn't answer for a 10
>>>>> seconds so B is bridged again to A (of course if B is still off hook).
>>>>>
>>>>> --
>>>>> ? ?????????,
>>>>>     ????? ?????????
>>>>>     ??? "??????"
>>>>>     (3435) 494991
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>
>>> --
>>> ? ?????????,
>>>    ????? ?????????
>>>    ??? "??????"
>>>    (3435) 494991
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> 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
>>
>
>


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From steveu at coppice.org  Sat Mar 12 17:28:16 2011
From: steveu at coppice.org (Steve Underwood)
Date: Sat, 12 Mar 2011 22:28:16 +0800
Subject: [Freeswitch-users] G.711.0
Message-ID: <4D7B8300.6030902@coppice.org>

Hi,

Has anyone seen G.711.0 in real world use? The spec was published quite 
a while ago, but as far as I can tell there is no RFC defining the SDP 
and RTP details needed to deploy it, and nobody advertises that they 
support it in their products.

Steve


From dftoro at yahoo.com  Sat Mar 12 17:35:03 2011
From: dftoro at yahoo.com (Diego Toro)
Date: Sat, 12 Mar 2011 06:35:03 -0800 (PST)
Subject: [Freeswitch-users] FreeSwitch for Windows
In-Reply-To: <4D7A9A32.9090508@gmail.com>
Message-ID: <134839.19366.qm@web33506.mail.mud.yahoo.com>

I don't think like Mr. Colling, I have FS on Windows in production since two year using sangoma cards (FXO, FXS) and works fine.? Windows is a good choice.

Diego Toro

http://voipensando.blogspot.com/

--- On Fri, 3/11/11, Fernando Berretta <fernando.berretta at gmail.com> wrote:

From: Fernando Berretta <fernando.berretta at gmail.com>
Subject: Re: [Freeswitch-users] FreeSwitch for Windows
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Friday, March 11, 2011, 4:54 PM



  

    
    
  Michael,

    

    Thanks for your valuable recommendation

    

    Best Regards,

    Fernando

    

    On 3/11/2011 6:41 PM, Michael Collins wrote:
    If you plan on using TDM then I'd recommend you stay
      away from Windows. Sangoma and Digium/clone cards are all geared
      toward a Linux environment. Stay away from the exotic OSes and use
      something simple like CentOS 5.5 or Debian Lenny.?
      
        

      
      -MC

        

        On Wed, Mar 9, 2011 at 1:38 PM,
          Fernando Berretta <fernando.berretta at gmail.com>
          wrote:

          
            Hi,

            

            I'm newbie in FreeSwitch and would like to know if
            freeswitch could be

            used for production under Windows platforms ? telephony
            boards like

            Digium / OpenVox, Sangoma, etc are going to work ok under
            Windows ? is

            there any problem / limitation ? E1 MFCR2 Trunk works ?

            

            Thanks for your time,

            Best Regards,

            Fernando

            

            _______________________________________________

            FreeSWITCH-users mailing list

            FreeSWITCH-users at lists.freeswitch.org

            http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

            UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

            http://www.freeswitch.org

          
        
        

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

    
  

-----Inline Attachment Follows-----

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From boris at tagnet.ru  Sat Mar 12 17:52:33 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 12 Mar 2011 19:52:33 +0500
Subject: [Freeswitch-users] How to timeout att_xfer?
In-Reply-To: <4D7B805F.1020700@tagnet.ru>
References: <4D7B5B98.3060608@tagnet.ru>	<AANLkTimehsD42iPNByQ1zq0YCRwE44ct+TgDMvG9mSeA@mail.gmail.com>	<4D7B7DA8.7080506@tagnet.ru>	<AANLkTi=wX=Jt=PuokT=ZP9xYY9vLi4zvX4Sq9V5XQyFc@mail.gmail.com>	<AANLkTinkJA6eEvJ3V9Vr7V9BhMTMqeyw=RLem8wKDjUv@mail.gmail.com>
	<4D7B805F.1020700@tagnet.ru>
Message-ID: <4D7B88B1.8060503@tagnet.ru>

Hello!

     bridge_answer_timeout is exact variable I looked for. Thank You, 
Anthony!
> Hello!
>
>       Yes, it works now! But there is another problem ->  while A is
> hearing MOH, the B is hearing nothing... Is there a way for B to hear
> MOH or progress tone?
>> frick,
>>
>> yes {originate_timeout=N}
>>
>> its early still here ....
>>
>> It will work, but you need to use ignore_early_media=true with it
>> otherwise you also need to set bridge_answer_timeout on the inbound
>> leg to the same value.
>>
>> {ignore_early_media=true,originate_timeout=N}
>>
>>
>> On Sat, Mar 12, 2011 at 8:07 AM, Anthony Minessale
>> <anthony.minessale at gmail.com>   wrote:
>>> orignation_timeout inside {} not by set app
>>>
>>>
>>> On Sat, Mar 12, 2011 at 8:05 AM, Boris Kovalenko<boris at tagnet.ru>   wrote:
>>>> Hello!
>>>>
>>>>       origination_timeout or originate_timeout? Indeed, have tried both
>>>> without success. Freeswitch FreeSWITCH Version 1.0.head (git-1c95ad9
>>>> 2011-01-20 22-43-50 -0300)
>>>> My extension:
>>>> <extension name="attented_xfer">
>>>> <condition field="destination_number" expression="^attented_xfer$">
>>>> <action application="read" data="3 3
>>>> $${sound_prefix}/ivr/8000/ivr-enter_ext.wav attxfer
>>>> _digits 10000 #"/>
>>>> <action application="set" data="continue_on_fail=true"/>
>>>> <action application="set" data="origination_cancel_key=#"/>
>>>> <action application="set" data="transfer_ringback=$${hold_music}"/>
>>>> <action application="set" data="hangup_after_bridge=true"/>
>>>> <action application="att_xfer"
>>>> data="{originate_timeout=5}loopback/${attxfer_digits}/${c
>>>> ontext}/${dialplan}"/>
>>>> <!--<action application="info" data="ALERT"/>-->
>>>> </condition>
>>>> </extension>
>>>>
>>>>> add {origination_timeout=N} before the dial string you pass into it.
>>>>>
>>>>>
>>>>> On Sat, Mar 12, 2011 at 5:40 AM, Boris Kovalenko<boris at tagnet.ru>     wrote:
>>>>>> Hello!
>>>>>>
>>>>>>        Is there a possibility to timeout att_xfer application? For example
>>>>>> ->     A calls B, B does att_xfer to C, but C doesn't answer for a 10
>>>>>> seconds so B is bridged again to A (of course if B is still off hook).
>>>>>>
>>>>>> --
>>>>>> ? ?????????,
>>>>>>      ????? ?????????
>>>>>>      ??? "??????"
>>>>>>      (3435) 494991
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>> --
>>>> ? ?????????,
>>>>     ????? ?????????
>>>>     ??? "??????"
>>>>     (3435) 494991
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> 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
>>>
>>
>


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From grsingh750 at gmail.com  Sat Mar 12 19:24:53 2011
From: grsingh750 at gmail.com (guru singh)
Date: Sat, 12 Mar 2011 21:54:53 +0530
Subject: [Freeswitch-users] inbound dtmf issue after upgrade
In-Reply-To: <4D7B2C99.2020401@chatsubo.net>
References: <4D7AA5E1.1050402@chatsubo.net>
	<AANLkTikTjx52euT=P7O-eXC+PCigyQGWLCHxjkQyOyL6@mail.gmail.com>
	<4D7B2C99.2020401@chatsubo.net>
Message-ID: <AANLkTikC7oSwP4_nrP7rADjcQcaiihJDDeErdrMaWEUc@mail.gmail.com>

Sorry, I've no idea about openzap, freeTDM was around when I got here.  :)

guru


On Sat, Mar 12, 2011 at 1:49 PM, Kevin Raison <raison at chatsubo.net> wrote:
> Thanks for the pointer, guru; ?and while I will definitely move toward
> that in the future, I am hoping to remedy my current situation without
> introducing too much risk to my customer, and unfortunately changing out
> oz for freetdm is a little risky in the eyes of the customer. ?So, are
> there any other possibilities that don't involve switching out oz? ?I
> know that wanpipe is setup to handle dtmf properly:
> ?TDMV_HW_DTMF ? ?= YES
>
> Is there something specific in the freeswitch config that needs enabling
> (other than using the "start_dtmf" option)?
>
> Thanks,
> Kevin
>
>
> On 03/11/2011 11:23 PM, guru singh wrote:
>> Hi Kevin,
>>
>> You should look at mod_freetdm. It replaces openzap.
>> http://wiki.freeswitch.org/wiki/FreeTDM
>>
>> guru
>>
>> On Sat, Mar 12, 2011 at 4:14 AM, Kevin Raison <raison at chatsubo.net> wrote:
>>> I upgraded a freeswitch setup from 1.0.4 to 1.0.6 yesterday, and also
>>> upgraded my wanpipe drivers from 3.4.9 to 3.5.10. ?I am using openzap
>>> with a sangoma card. ?Before the upgrade, the following stanza would
>>> correctly send a caller to a mod_event_socket-based application that I
>>> have running on another machine and that caller's dtmf would flow
>>> through to the app:
>>>
>>> <extension name="Kevin_test">
>>> ? ? ? ?<condition field="destination_number" expression="^XXXXXXXXXX$">
>>> ? ? ? ? ? ? ? ?<action application="set" data="ringback=${us-ring}"/>
>>> ? ? ? ? ? ? ? ?<action application="set"
>>> data="transfer_ringback=$${hold_music}"/>
>>> ? ? ? ? ? ? ? ?<action application="bridge"
>>> data="sofia/internal/5000 at prod.XXXXXXX.com"/>
>>> ? ? ? ?</condition>
>>> </extension>
>>>
>>> After the upgrade, everything but inbound dtmf works as it should, but
>>> the only way that I can get inbound dtmf to work is by adding
>>> ? ? ? ?<action application="start_dtmf"/>
>>> to the stanza. ?This is not ideal and I would like to go back to not
>>> using in-band dtmf if possible.
>>>
>>> In my log files, I see absolutely no dtmf being registered when a caller
>>> calls in. ?However, if someone calls out from a desk phone and then
>>> transfers the person that they called to
>>> sofia/internal/5000 at prod.XXXXXXX.com, dtmf works just fine.
>>>
>>> Any ideas?
>>>
>>> Thanks!
>>> Kevin
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From lists at telefaks.de  Sat Mar 12 20:22:48 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Sat, 12 Mar 2011 18:22:48 +0100
Subject: [Freeswitch-users] Where to download zrtp/zfone - server offline?
Message-ID: <4D7BABE8.9090803@telefaks.de>

I tried to install zrtp and to download Zfone, but I get
http://www.zfone.com/server_problems.html
"Sorry, but our download server is offline." (since 29-Jan-2011)

Does anybody know of another download source?

-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de




From mario_fs at mgtech.com  Sat Mar 12 20:37:22 2011
From: mario_fs at mgtech.com (Mario G)
Date: Sat, 12 Mar 2011 09:37:22 -0800
Subject: [Freeswitch-users] Is external_rtp_ip=auto still valid?
Message-ID: <E8474EE0-52E5-41B5-99DC-EFAB728C3A4F@mgtech.com>

The wiki states auto is a valid option under Sofia configuration but it is not mentioned in the vars.xml. Is this still valid? I need and way to make this work without uPNP. Thanks,
Mario G

From katerin.borin at gmail.com  Sat Mar 12 23:39:21 2011
From: katerin.borin at gmail.com (Borin)
Date: Sat, 12 Mar 2011 20:39:21 +0000
Subject: [Freeswitch-users] trunk without authentication
Message-ID: <AANLkTi=78LzPnvN36=H-Pfx=UXwEEdokCHE0pFr=GQAA@mail.gmail.com>

Hello FS users,
Is it possible in FS to allow all calls from a particular IP address without
authentication? like in asterisk with insecure port, invite.
I tried to add a new carrier in
/usr/local/freeswitch/conf/sip_profiles/external/new-carrier.xml
<include>
  <gateway name="new-carrier">
    <param name="username" value="anything"/>
    <param name="password" value="anything"/>
    <param name="realm" value="carrier.ip.address"/>
    <param name="register" value="false"/>
    <param name="caller-id-in-from" value="true"/>
  </gateway>
</include>

But after restart it stil sends 407 Proxy Authentication Required to the
carrier. Is there any way to trust the carrier IP?
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From vetali100 at gmail.com  Sun Mar 13 00:43:32 2011
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sat, 12 Mar 2011 23:43:32 +0200
Subject: [Freeswitch-users] trunk without authentication
In-Reply-To: <AANLkTi=78LzPnvN36=H-Pfx=UXwEEdokCHE0pFr=GQAA@mail.gmail.com>
References: <AANLkTi=78LzPnvN36=H-Pfx=UXwEEdokCHE0pFr=GQAA@mail.gmail.com>
Message-ID: <AANLkTin+60VZvTt=mFT4sGTiiW9BN4zmiAOM5Kxt4Vgm@mail.gmail.com>

Pls. try to add:
<param name="auth-calls" value="false"/>

http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#auth-calls

This should disable authentication check.

And then implement ACL for this specific IP, for details pls. see
http://wiki.freeswitch.org/wiki/Acl

<http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#auth-calls>
Regards,
Vitalie


2011/3/12 Borin <katerin.borin at gmail.com>

> Hello FS users,
> Is it possible in FS to allow all calls from a particular IP address
> without authentication? like in asterisk with insecure port, invite.
> I tried to add a new carrier in
> /usr/local/freeswitch/conf/sip_profiles/external/new-carrier.xml
> <include>
>   <gateway name="new-carrier">
>     <param name="username" value="anything"/>
>     <param name="password" value="anything"/>
>     <param name="realm" value="carrier.ip.address"/>
>     <param name="register" value="false"/>
>     <param name="caller-id-in-from" value="true"/>
>   </gateway>
> </include>
>
> But after restart it stil sends 407 Proxy Authentication Required to the
> carrier. Is there any way to trust the carrier IP?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From iam at onnet.su  Sun Mar 13 01:21:20 2011
From: iam at onnet.su (iam at onnet.su)
Date: Sun, 13 Mar 2011 01:21:20 +0300 (MSK)
Subject: [Freeswitch-users] Disable Goodbye playback for a particular user
In-Reply-To: <AANLkTin+60VZvTt=mFT4sGTiiW9BN4zmiAOM5Kxt4Vgm@mail.gmail.com>
References: <AANLkTi=78LzPnvN36=H-Pfx=UXwEEdokCHE0pFr=GQAA@mail.gmail.com>
	<AANLkTin+60VZvTt=mFT4sGTiiW9BN4zmiAOM5Kxt4Vgm@mail.gmail.com>
Message-ID: <62364.88.201.183.18.1299968480.squirrel@mail.onnet.su>

Hello FS users,

I have some customers trying to utilize automatic redial feature with
their pots lines. But when the analog phone listens for a "Goodbye"
message instead of busy tone it can't determine that the line is busy and
automatic redial fails.

So the question is whether there is a way to disable or skip playback of a
"GoodBye" message for a particular user or change "Goodbye" message for
"User Busy".


Thank you in advance.


From frank at rosengart.de  Sun Mar 13 01:38:30 2011
From: frank at rosengart.de (Frank Rosengart)
Date: Sat, 12 Mar 2011 23:38:30 +0100
Subject: [Freeswitch-users] PA and ALSA
Message-ID: <4D7BF5E6.30201@rosengart.de>

Hi,

I' trying to use FS as a softphone, because it seems to be the only SIP
UA, with up-to-date and full CELT support (thanks for that, great work!)

But I'm really struggling with my audio config. I'm using Ubuntu
10.10, x86 with Gnome -> Pulseaudio.

1. mod_alsa simply segfaults immediately after receiving a call
'application="bridge" data="alsa"' or  'alsa call 123' from the console

2. I am running FS on two different computers. One, which is upgraded
from earlier times, portaudio works ok (not perfect, but I think that
are Alsa related issues). Another computer, with a fresh 10.10 install,
mod_portaudio can not find any audio devices.
mod_portaudio.so links libasound, so I assume that Alsa is included.
aplay -L lists available devices:

pulse
    Playback/recording through the PulseAudio sound server
front:CARD=NVidia,DEV=0
    HDA NVidia, ALC889A Analog
    Front speakers
...

Any ideas what is missing here?

3. Has the portaudio lib, which included with FS, the PA 'dmix patch'
applied? http://www.portaudio.com/trac/ticket/31

4. Has anyone started to build mod_pulseaudio ?
I know, there are latency issues, but we are talking about VoIP with
codec with frame sizes of 10ms and more.. Portaudio is really not the
first choice for sound APIs on Linux anymore...

5. Same config, using CELT on Windows, I get noise glitches every second
when calling an FS server, answering with the echo app. They are very
regular, so it's maybe a sample rate issue somewhere in the path...


Thanks for any hints.


Frank





From steveayre at gmail.com  Sun Mar 13 01:48:20 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 12 Mar 2011 22:48:20 +0000
Subject: [Freeswitch-users] trunk without authentication
In-Reply-To: <AANLkTi=78LzPnvN36=H-Pfx=UXwEEdokCHE0pFr=GQAA@mail.gmail.com>
References: <AANLkTi=78LzPnvN36=H-Pfx=UXwEEdokCHE0pFr=GQAA@mail.gmail.com>
Message-ID: <AANLkTikCwuLZpDXWH_kPG4Ute=yU7WcZn9=WXQyyj9re@mail.gmail.com>

That's for an outgoing gateway.

Look at the cidr parameter of the user directory - it lets you authenticate
a user based on IP rather than password.

http://wiki.freeswitch.org/wiki/Acl#Users

-Steve



On 12 March 2011 20:39, Borin <katerin.borin at gmail.com> wrote:

> Hello FS users,
> Is it possible in FS to allow all calls from a particular IP address
> without authentication? like in asterisk with insecure port, invite.
> I tried to add a new carrier in
> /usr/local/freeswitch/conf/sip_profiles/external/new-carrier.xml
> <include>
>   <gateway name="new-carrier">
>     <param name="username" value="anything"/>
>     <param name="password" value="anything"/>
>     <param name="realm" value="carrier.ip.address"/>
>     <param name="register" value="false"/>
>     <param name="caller-id-in-from" value="true"/>
>   </gateway>
> </include>
>
> But after restart it stil sends 407 Proxy Authentication Required to the
> carrier. Is there any way to trust the carrier IP?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jeff at jefflenk.com  Sun Mar 13 02:34:33 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sat, 12 Mar 2011 15:34:33 -0800 (PST)
Subject: [Freeswitch-users] PA and ALSA
In-Reply-To: <4D7BF5E6.30201@rosengart.de>
References: <4D7BF5E6.30201@rosengart.de>
Message-ID: <1299972873900-6165240.post@n2.nabble.com>

Same config under windows with celt - what are you using? if fs are you using
git head?

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/PA-and-ALSA-tp6165168p6165240.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From boris at tagnet.ru  Sun Mar 13 08:24:01 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sun, 13 Mar 2011 10:24:01 +0500
Subject: [Freeswitch-users] SIP profile and alias ip
Message-ID: <4D7C54F1.5000406@tagnet.ru>

Hello!

     Reading documentation about sip profiles found a <param 
name="alias">. So I may use same profile with two IP's. Is there any 
disadvantages to use alias? If endpoint will be connected to alias IP 
what IP FS will use for outgoing RTP (alias ip or rtp-ip)? Also, if 
gateway (for outbound calls) is reachable (by host route table) via 
alias IP will it be used for RTP instead of primary rtp-ip?

-- 
Regards,
Boris



From boris at tagnet.ru  Sun Mar 13 11:52:15 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sun, 13 Mar 2011 13:52:15 +0500
Subject: [Freeswitch-users] Question about proxy_media
Message-ID: <4D7C85BF.9030804@tagnet.ru>

Hello!

     I have two sip profiles. One is for transit traffic with 
proxy_media enabled by default. Second is a SoftPBX for my clients with 
a features like a IVR, conference, attended call transfer and so on. So 
the question is - what happens when A leg is from transit profile and B 
leg is to SoftPBX profile. What mode of proxy_media will be set when 
call is connected? May my user on B leg use DTMF for call transfer for 
example? What mode will be used for FAX?

-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From grsingh750 at gmail.com  Sun Mar 13 13:25:51 2011
From: grsingh750 at gmail.com (guru singh)
Date: Sun, 13 Mar 2011 15:55:51 +0530
Subject: [Freeswitch-users] mod_callcenter not bridging call to agent
	(callback)
Message-ID: <AANLkTimsbnVMXxyJSYDE=bWYLehba4PDhkmYZ+Z8bG+b@mail.gmail.com>

Hi,

I am trying to get mod_callcenter working.
When the callcenter extension is dialed, the caller keeps listening to
moh until it times out.
The call is not being sent to agents.
call debug log and other info here http://pastebin.freeswitch.org/15665
What am i missing?
Let me know if anything else is required.

thanks
guru


From katerin.borin at gmail.com  Sun Mar 13 17:41:01 2011
From: katerin.borin at gmail.com (Borin)
Date: Sun, 13 Mar 2011 14:41:01 +0000
Subject: [Freeswitch-users] trunk without authentication
In-Reply-To: <AANLkTikCwuLZpDXWH_kPG4Ute=yU7WcZn9=WXQyyj9re@mail.gmail.com>
References: <AANLkTi=78LzPnvN36=H-Pfx=UXwEEdokCHE0pFr=GQAA@mail.gmail.com>
	<AANLkTikCwuLZpDXWH_kPG4Ute=yU7WcZn9=WXQyyj9re@mail.gmail.com>
Message-ID: <AANLkTi=3WuCV+1mk3hhTRFJXns0Yuv7vjvK5H6grHBUr@mail.gmail.com>

Hi guys
thanks a lot

I added
  <list name="carrier" default="deny">
    <node type="allow" cidr="carrier.ip/32"/>
    </list>
to the acl.conf.xml

then    <param name="apply-inbound-acl" value="carrier"/>  in
sip_profiles/external/carrier.xml
<include>
  <gateway name="carrier">
    <param name="username" value="anything"/>
    <param name="password" value="anything"/>
    <param name="auth-calls" value="false"/>
    <param name="apply-inbound-acl" value="carrier"/>
    <param name="register" value="false"/>
</gateway>
</include>

Works now

On Sat, Mar 12, 2011 at 10:48 PM, Steven Ayre <steveayre at gmail.com> wrote:

> That's for an outgoing gateway.
>
> Look at the cidr parameter of the user directory - it lets you authenticate
> a user based on IP rather than password.
>
> http://wiki.freeswitch.org/wiki/Acl#Users
>
> -Steve
>
>
>
> On 12 March 2011 20:39, Borin <katerin.borin at gmail.com> wrote:
>
>> Hello FS users,
>> Is it possible in FS to allow all calls from a particular IP address
>> without authentication? like in asterisk with insecure port, invite.
>> I tried to add a new carrier in
>> /usr/local/freeswitch/conf/sip_profiles/external/new-carrier.xml
>> <include>
>>   <gateway name="new-carrier">
>>     <param name="username" value="anything"/>
>>     <param name="password" value="anything"/>
>>     <param name="realm" value="carrier.ip.address"/>
>>     <param name="register" value="false"/>
>>     <param name="caller-id-in-from" value="true"/>
>>   </gateway>
>> </include>
>>
>> But after restart it stil sends 407 Proxy Authentication Required to the
>> carrier. Is there any way to trust the carrier IP?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anita.hall at simmortel.com  Sun Mar 13 18:18:19 2011
From: anita.hall at simmortel.com (Anita Hall)
Date: Sun, 13 Mar 2011 08:18:19 -0700
Subject: [Freeswitch-users] libss7 + asterisk woes!
Message-ID: <AANLkTikMbfhfGczFYVOCTwG7jnLe5hhyOHiMVv=t+sa_@mail.gmail.com>

Hi

I am using Asterisk on Sangoma cards :(
I know I should have used Digium but now the customer has these cards and I
have to make them work with SS7 :(

(And FreeSWITCH still does not support SS7 , is that right ?)

Unable to receive incoming calls on chan_ss7, I have decided to give another
shot to libss7

Could someone please share their configuration of dahdi/system.conf,
chan_dahdi.conf, wanpipeX.conf with me for multiple signalling links ?

I am especially confusing with the numbering schemes of CICs and signalling
channels. I have signalling links on the first 2 spans, on 16th channel
each.

So , how do I number after the 2nd channel onwards.
Is it 32-64
or 31-63
or what ?
:)

My current config is

debian:/usr/src# cat /etc/dahdi/system.conf
#autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
#autogenrated on 2011-03-13
#Dahdi Channels Configurations
#For detailed Dahdi options, view /etc/dahdi/system.conf.bak
loadzone=us
defaultzone=us

#Sangoma A108 port 1 [slot:4 bus:5 span:1] <wanpipe1>
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
mtp2=16

#Sangoma A108 port 2 [slot:4 bus:5 span:2] <wanpipe2>
span=2,2,0,ccs,hdb3,crc4
bchan=33-46,48-62
mtp2=47

#Sangoma A108 port 3 [slot:4 bus:5 span:3] <wanpipe3>
span=3,0,0,ccs,hdb3,crc4
bchan=63-93

#Sangoma A108 port 4 [slot:4 bus:5 span:4] <wanpipe4>
span=4,0,0,ccs,hdb3,crc4
bchan=94-124

#Sangoma A108 port 5 [slot:4 bus:5 span:5] <wanpipe5>
span=5,0,0,ccs,hdb3,crc4
bchan=125-155

#Sangoma A108 port 6 [slot:4 bus:5 span:6] <wanpipe6>
span=6,0,0,ccs,hdb3,crc4
bchan=156-186

#Sangoma A108 port 7 [slot:4 bus:5 span:7] <wanpipe7>
span=7,0,0,ccs,hdb3,crc4
bchan=187-217

#Sangoma A108 port 8 [slot:4 bus:5 span:8] <wanpipe8>
span=8,0,0,ccs,hdb3,crc4
bchan=218-248

#Sangoma A108 port 1 [slot:4 bus:7 span:9] <wanpipe9>
span=9,0,0,ccs,hdb3,crc4
bchan=249-279

#Sangoma A108 port 2 [slot:4 bus:7 span:10] <wanpipe10>
span=10,0,0,ccs,hdb3,crc4
bchan=280-310

#Sangoma A108 port 3 [slot:4 bus:7 span:11] <wanpipe11>
span=11,0,0,ccs,hdb3,crc4
bchan=311-341

#Sangoma A108 port 4 [slot:4 bus:7 span:12] <wanpipe12>
span=12,0,0,ccs,hdb3,crc4
bchan=342-372

#Sangoma A108 port 5 [slot:4 bus:7 span:13] <wanpipe13>
span=13,0,0,ccs,hdb3,crc4
bchan=373-403

#Sangoma A108 port 6 [slot:4 bus:7 span:14] <wanpipe14>
span=14,0,0,ccs,hdb3,crc4
bchan=404-434

#Sangoma A108 port 7 [slot:4 bus:7 span:15] <wanpipe15>
span=15,0,0,ccs,hdb3,crc4
bchan=435-465

#Sangoma A108 port 8 [slot:4 bus:7 span:16] <wanpipe16>
span=16,0,0,ccs,hdb3,crc4
bchan=466-496

debian:/usr/src# cat /etc/asterisk/chan_dahdi.conf
;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
;autogenrated on 2011-03-13
;Dahdi Channels Configurations
;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak

[trunkgroups]

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=no
echocancelwhenbridged=no
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no

;Sangoma A108 port 1 [slot:4 bus:5 span:1] <wanpipe1>
switchtype=national
context=from-pstn
group=0
echocancel=no
signalling=ss7
ss7type=itu                       ;using the ITU variant
ss7_called_nai=dynamic           ;NAI for outgoing calls
ss7_calling_nai=dynamic          ;NAI for incoming calls
ss7_internationalprefix=00 ;international prefix value for incoming calls
ss7_nationalprefix=0            ;national prefix value for incoming calls
ss7_subscriberprefix=            ;subscriber prefix value for incoming
calls
ss7_unknownprefix=              ;unknown prefix value for incoming calls
ss7_explictacm=yes               ;ACM is send as soon as call enters the
dial plan...may not accepted yet though
linkset=1                         ;arbitrary name for this set of channels
pointcode=13323                      ;the point code for this system...aka
SPC
adjpointcode=12650                  ;the point code for the system that we
are signaling to... aka APC
defaultdpc=12650                     ;the point code for the system that the
CICs will be negotiated with...aka DPC
networkindicator=national_spare  ;NI value for MTP3
cicbeginswith=1                  ;the starting value of the CICs
channel=2-31                     ;the channels that are CICs
sigchan=1                         ;the signaling channel

;Sangoma A108 port 2 [slot:4 bus:5 span:2] <wanpipe2>
switchtype=national
context=from-pstn
group=0
echocancel=no
signalling=ss7
ss7type=itu                       ;using the ITU variant
ss7_called_nai=dynamic           ;NAI for outgoing calls
ss7_calling_nai=dynamic          ;NAI for incoming calls
ss7_internationalprefix=00 ;international prefix value for incoming calls
ss7_nationalprefix=0            ;national prefix value for incoming calls
ss7_subscriberprefix=            ;subscriber prefix value for incoming
calls
ss7_unknownprefix=              ;unknown prefix value for incoming calls
ss7_explictacm=yes               ;ACM is send as soon as call enters the
dial plan...may not accepted yet though
linkset=1                         ;arbitrary name for this set of channels
pointcode=13323                      ;the point code for this system...aka
SPC
adjpointcode=12650                  ;the point code for the system that we
are signaling to... aka APC
defaultdpc=12650                     ;the point code for the system that the
CICs will be negotiated with...aka DPC
networkindicator=national_spare  ;NI value for MTP3
cicbeginswith=1                  ;the starting value of the CICs
channel=2-31                     ;the channels that are CICs
sigchan=1                         ;the signaling channel

;Sangoma A108 port 3 [slot:4 bus:5 span:3] <wanpipe3>
switchtype=national
context=from-pstn
group=0
echocancel=no
signalling=ss7
channel =>63-77,79-93

;Sangoma A108 port 4 [slot:4 bus:5 span:4] <wanpipe4>
switchtype=national
context=from-pstn
group=0
echocancel=no
signalling=ss7
channel =>94-108,110-124

;Sangoma A108 port 5 [slot:4 bus:5 span:5] <wanpipe5>
switchtype=national
context=from-pstn
group=0
echocancel=no
signalling=ss7
channel =>125-139,141-155

;Sangoma A108 port 6 [slot:4 bus:5 span:6] <wanpipe6>
switchtype=national
context=from-pstn
group=0
echocancel=no
signalling=ss7
channel =>156-170,172-186

;Sangoma A108 port 7 [slot:4 bus:5 span:7] <wanpipe7>
switchtype=national
context=from-pstn
group=0
echocancel=no
signalling=ss7
channel =>187-201,203-217

;Sangoma A108 port 8 [slot:4 bus:5 span:8] <wanpipe8>
switchtype=national
context=from-pstn
group=0
echocancel=no
signalling=ss7
channel =>218-232,234-248

;Sangoma A108 port 1 [slot:4 bus:7 span:9] <wanpipe9>
switchtype=national
context=from-pstn
group=0
echocancel=no
signalling=ss7
channel =>249-263,265-279

;Sangoma A108 port 2 [slot:4 bus:7 span:10] <wanpipe10>
switchtype=national
context=from-pstn
group=0
echocancel=no
signalling=ss7
channel =>280-294,296-310

;Sangoma A108 port 3 [slot:4 bus:7 span:11] <wanpipe11>
switchtype=national
context=from-pstn
group=0
echocancel=no
signalling=ss7
channel =>311-325,327-341

;Sangoma A108 port 4 [slot:4 bus:7 span:12] <wanpipe12>
switchtype=national
context=from-pstn
group=0
echocancel=no
signalling=ss7
channel =>342-356,358-372

;Sangoma A108 port 5 [slot:4 bus:7 span:13] <wanpipe13>
switchtype=national
context=from-pstn
group=0
echocancel=no
signalling=ss7
channel =>373-387,389-403

;Sangoma A108 port 6 [slot:4 bus:7 span:14] <wanpipe14>
switchtype=national
context=from-pstn
group=0
echocancel=no
signalling=ss7
channel =>404-418,420-434

;Sangoma A108 port 7 [slot:4 bus:7 span:15] <wanpipe15>
switchtype=national
context=from-pstn
group=0
echocancel=no
signalling=ss7
channel =>435-449,451-465

;Sangoma A108 port 8 [slot:4 bus:7 span:16] <wanpipe16>
switchtype=national
context=from-pstn
group=0
echocancel=no
signalling=ss7
channel =>466-480,482-496


debian:/usr/src# cat /etc/wanpipe/wanpipe1.conf
#================================================
# WANPIPE1 Configuration File
#================================================
#
# Date: Wed Dec  6 20:29:03 UTC 2006
#
# Note: This file was generated automatically
#       by /usr/local/sbin/setup-sangoma program.
#
#       If you want to edit this file, it is
#       recommended that you use wancfg program
#       to do so.
#================================================
# Sangoma Technologies Inc.
#================================================

[devices]
wanpipe1 = WAN_AFT_TE1, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment

[wanpipe1]
CARD_TYPE  = AFT
S514CPU  = A
CommPort  = PRI
AUTO_PCISLOT  = NO
PCISLOT  = 4
PCIBUS   = 5
FE_MEDIA = E1
FE_LCODE = HDB3
FE_FRAME = CRC4
FE_LINE = 1
TE_CLOCK  = NORMAL
TE_REF_CLOCK    = 0
TE_SIG_MODE     = CCS
TE_HIGHIMPEDANCE = NO
TE_RX_SLEVEL    = 430
LBO  = 120OH
FE_TXTRISTATE = NO
MTU  = 1500
UDPPORT = 9000
TTL = 255
IGNORE_FRONT_END = NO
TDMV_SPAN = 1
TDMV_DCHAN = 0
TE_AIS_MAINTENANCE = NO         #NO: defualt  YES: Start port in AIS Blue
Alarm and keep line down
                                #wanpipemon -i w1g1 -c Ttx_ais_off to
disable AIS maintenance mode
#wanpipemon -i w1g1 -c Ttx_ais_on to enable AIS maintenance mode
TDMV_HW_DTMF = NO # YES: receive dtmf events from hardware
TDMV_HW_FAX_DETECT = NO # YES: receive fax 1100hz events from hardware
HWEC_OPERATION_MODE     = OCT_NORMAL    # OCT_NORMAL: echo cancelation
enabled with nlp (default)
# OCT_SPEECH: improves software tone detection by disabling NLP (echo
possible)
# OCT_NO_ECHO:disables echo cancelation but allows VQE/tone functions.
HWEC_DTMF_REMOVAL       = NO    # NO: default  YES: remove dtmf out of
incoming media (must have hwdtmf enabled)
HWEC_NOISE_REDUCTION    = NO    # NO: default  YES: reduces noise on the
line - could break fax
HWEC_ACUSTIC_ECHO       = NO    # NO: default  YES: enables acustic echo
cancelation
HWEC_NLP_DISABLE        = NO    # NO: default  YES: guarantees software tone
detection (possible echo)
HWEC_TX_AUTO_GAIN       = 0     # 0: disable   -40-0: default tx audio level
to be maintained (-20 default)
HWEC_RX_AUTO_GAIN       = 0     # 0: disable   -40-0: default tx audio level
to be maintained (-20 default)
HWEC_TX_GAIN            = 0 # 0: disable   -24-24: db values to be applied
to tx signal
HWEC_RX_GAIN            = 0 # 0: disable   -24-24: db values to be applied
to tx signal

[w1g1]
ACTIVE_CH = ALL
TDMV_HWEC = NO
MTU  = 8
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From julf at julf.com  Sun Mar 13 19:06:01 2011
From: julf at julf.com (Johan Helsingius)
Date: Sun, 13 Mar 2011 17:06:01 +0100
Subject: [Freeswitch-users] Another beginner question: garbled voicemail
In-Reply-To: <AANLkTi=_XwM09bHXuEctmW9Zzts2+tSvWnp-D3Z4_AZX@mail.gmail.com>
References: <4D7A0B5D.6030803@julf.com>	<549CFEF87AEDE841A38E9D15EAB4C04C58B2F6BF79@cooper>	<4D7A17D2.5000307@julf.com>
	<4D7A2721.6020003@julf.com>	<AANLkTik2C95ooHuC2_ux=LfXQXR3ZGC3i6LNQde1rqeO@mail.gmail.com>	<4D7A4CA3.9030701@julf.com>
	<AANLkTi=_XwM09bHXuEctmW9Zzts2+tSvWnp-D3Z4_AZX@mail.gmail.com>
Message-ID: <4D7CEB69.6050800@julf.com>

> try pcap of the call so you can look at the packets and the trace and
> even extract the audio to see what is going on.

Of course I have not had the garbled audio since - not sure what
caused it, but whatever it was, it seems to have gone - knock on
wood!

	Julf


From julf at julf.com  Sun Mar 13 19:09:41 2011
From: julf at julf.com (Johan Helsingius)
Date: Sun, 13 Mar 2011 17:09:41 +0100
Subject: [Freeswitch-users] unknown event 8: 101 NAT detected
Message-ID: <4D7CEC45.3040306@julf.com>

I have a lot of problems getting my SIP provider connections to
work. I am using 3 separate SIP providers (for different
destinations/services), and sometimes things work and sometimes
they don't. Registrations work OK, but often there is no
audio, or the call establishment fails. Makes me suspect a
network/NAT issue.

Something that I do worry about is the log messages saying
"sofia.c:995 nua_i_outbound: unknown event 8: 101 NAT
 detected".

	Julf


From boris at tagnet.ru  Sun Mar 13 19:11:04 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sun, 13 Mar 2011 21:11:04 +0500
Subject: [Freeswitch-users] question about faxes
Message-ID: <4D7CEC98.5090000@tagnet.ru>

Hello!

     Here is a small topology description:

     FaxA -- ClientA -- (internet) -- myFS -- ClientB -- FaxB

ClientA - somewhere in the world, with unknow equipment
myFS - FreeSWITCH Version 1.0.head (git-1c95ad9 2011-01-20 22-43-50 
-0300) managed by me. As I need to process DTMF from clients, 
proxy_media=false
ClientB - any of my clients with some type of equipment Voip<->FXS 
conversion
FaxA/B - Faxes

May someone tell me please what configuration should I do to permit my 
clients send and receive faxes? The main problem is that I really don't 
know when and on what number the user want to use fax machine. I have 
read mod_spandsp wiki but it isn't so clear. Is this enough to add this 
applications before any bridge?:

     <action application="set" data="fax_enable_t38=true"/>  <!-- Enable t.38 for this call -->
     <action application="set" data="fax_enable_t38_request=true"/>  <!-- Enable t38_gateway to send a t.38 reinvite when a fax tone is detected. If using t38_gateway peer then you need to export this variable instead of set -->
     <action application="set" data="execute_on_answer=t38_gateway self"/>  <!--Execute t38_gateway on answer. self or peer. self: send a reinvite back to the a-leg. peer reinvite forward to the b-leg -->


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From julf at julf.com  Sun Mar 13 19:24:53 2011
From: julf at julf.com (Johan Helsingius)
Date: Sun, 13 Mar 2011 17:24:53 +0100
Subject: [Freeswitch-users] unknown event 8: 101 NAT detected
In-Reply-To: <4D7CEC45.3040306@julf.com>
References: <4D7CEC45.3040306@julf.com>
Message-ID: <4D7CEFD5.7060905@julf.com>

> Something that I do worry about is the log messages saying
> "sofia.c:995 nua_i_outbound: unknown event 8: 101 NAT
>  detected".

I also get these fairly often, and quite often associated
with a registration failing - but the retry succeeds:

nua_i_outbound: unknown event 8: 102 NAT binding changed

	Julf


From acrow at integrafin.co.uk  Sun Mar 13 19:28:51 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Sun, 13 Mar 2011 16:28:51 +0000
Subject: [Freeswitch-users] Dialplan tone_detect problem
Message-ID: <4D7CF0C3.3050104@integrafin.co.uk>

All,

I am trying to implement fax detection on FreeSwitch within a dialplan 
that routes inbound calls from a BRI (Sangoma), to an Asterisk (Trixbox) 
machine handling the extensions. I am only doing this until I can get 
British English sounds for FS ;-).

The inbound route without any fax detection works fine, ie the caller 
hears ringing until the Trixbox extension is either answered or hits 
voicemail:

<extension name="60xto35x" >
<condition field="source" expression="freetdm"/>
<condition field="context" expression="public"/>
<condition field="destination_number" expression="60([0-5])">
<action application="bridge" data="sofia/external/35$1 at 192.168.24.15"/>
</condition>
</extension>


The closest I have got to adding the fax detection is the following:

<extension name="60xto35x_copy" >
<condition field="source" expression="freetdm"/>
<condition field="context" expression="public"/>
<condition field="destination_number" expression="60([0-5])">
<action application="tone_detect" data="fax 1100 r +5000 transfer '620 
XML default'"/>
<action application="answer"/>
<action application="playback" data="tone_stream://$${uk-ring};loops=3"/>
<action application="bridge" data="sofia/external/35$1 at 192.168.24.15"/>
</condition>
</extension>

I had to add the playback so the caller would hear ringing during the 
tone_detect timeout.

However, as soon as the tone_stream finishes, and the call is bridged to 
the Trixbox, the caller hears silence instead of ringing. Is there any 
way to pass the ringing from the extension back to the caller coming in 
on the BRI circuit, or otherwise fake it until the remote extension is 
answered?

Thanks

Alex

-- 
This message is intended only for the addressee and may contain
confidential information.  Unless you are that person, you may not
disclose its contents or use it in any way and are requested to delete
the message along with any attachments and notify us immediately.

"Transact" is operated by Integrated Financial Arrangements plc
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Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)



From vetali100 at gmail.com  Sun Mar 13 20:20:15 2011
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 13 Mar 2011 20:20:15 +0300
Subject: [Freeswitch-users] trunk without authentication
In-Reply-To: <AANLkTi=3WuCV+1mk3hhTRFJXns0Yuv7vjvK5H6grHBUr@mail.gmail.com>
References: <AANLkTi=78LzPnvN36=H-Pfx=UXwEEdokCHE0pFr=GQAA@mail.gmail.com>
	<AANLkTikCwuLZpDXWH_kPG4Ute=yU7WcZn9=WXQyyj9re@mail.gmail.com>
	<AANLkTi=3WuCV+1mk3hhTRFJXns0Yuv7vjvK5H6grHBUr@mail.gmail.com>
Message-ID: <AANLkTim2pWDzXVxXpP95PJy2Q46r+mLYw3c0=bM6c=za@mail.gmail.com>

Steven, according to Borin's message she is talking exactly about outgoing
gateway (sip_profiles/external/carrier.xml).

And not about user's directory.

Outgoing gateway can also work as incoming gateway, if you send your calls
to it.

Just want to make things clear.

Vitalie


2011/3/13 Borin <katerin.borin at gmail.com>

> Hi guys
> thanks a lot
>
> I added
>   <list name="carrier" default="deny">
>     <node type="allow" cidr="carrier.ip/32"/>
>     </list>
> to the acl.conf.xml
>
> then    <param name="apply-inbound-acl" value="carrier"/>  in
> sip_profiles/external/carrier.xml
> <include>
>   <gateway name="carrier">
>
>     <param name="username" value="anything"/>
>     <param name="password" value="anything"/>
>     <param name="auth-calls" value="false"/>
>     <param name="apply-inbound-acl" value="carrier"/>
>
>     <param name="register" value="false"/>
> </gateway>
> </include>
>
> Works now
>
>
> On Sat, Mar 12, 2011 at 10:48 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> That's for an outgoing gateway.
>>
>> Look at the cidr parameter of the user directory - it lets you
>> authenticate a user based on IP rather than password.
>>
>> http://wiki.freeswitch.org/wiki/Acl#Users
>>
>> -Steve
>>
>>
>>
>> On 12 March 2011 20:39, Borin <katerin.borin at gmail.com> wrote:
>>
>>> Hello FS users,
>>> Is it possible in FS to allow all calls from a particular IP address
>>> without authentication? like in asterisk with insecure port, invite.
>>> I tried to add a new carrier in
>>> /usr/local/freeswitch/conf/sip_profiles/external/new-carrier.xml
>>> <include>
>>>   <gateway name="new-carrier">
>>>     <param name="username" value="anything"/>
>>>     <param name="password" value="anything"/>
>>>     <param name="realm" value="carrier.ip.address"/>
>>>     <param name="register" value="false"/>
>>>     <param name="caller-id-in-from" value="true"/>
>>>   </gateway>
>>> </include>
>>>
>>> But after restart it stil sends 407 Proxy Authentication Required to the
>>> carrier. Is there any way to trust the carrier IP?
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From katerin.borin at gmail.com  Sun Mar 13 20:35:53 2011
From: katerin.borin at gmail.com (Borin)
Date: Sun, 13 Mar 2011 17:35:53 +0000
Subject: [Freeswitch-users] trunk without authentication
In-Reply-To: <AANLkTim2pWDzXVxXpP95PJy2Q46r+mLYw3c0=bM6c=za@mail.gmail.com>
References: <AANLkTi=78LzPnvN36=H-Pfx=UXwEEdokCHE0pFr=GQAA@mail.gmail.com>
	<AANLkTikCwuLZpDXWH_kPG4Ute=yU7WcZn9=WXQyyj9re@mail.gmail.com>
	<AANLkTi=3WuCV+1mk3hhTRFJXns0Yuv7vjvK5H6grHBUr@mail.gmail.com>
	<AANLkTim2pWDzXVxXpP95PJy2Q46r+mLYw3c0=bM6c=za@mail.gmail.com>
Message-ID: <AANLkTin_VWtAMSnhHAXAg7fh0q4mqxQqXp+vO2uYnjRt@mail.gmail.com>

sorry guys, I just started to play with freeswitch  and  probably I was not
clear with my schema
I use the following scenario: asterisk (carrier in my config) ---> --- (udp
port 5080) freeswitch (here conference application)
I needed that freeswicth did not requie INVITE authentication for the calls
coming from asterisk. So the calls are external incoming for freeswitch.
Until I added this config with acl it did not funtion.


On Sun, Mar 13, 2011 at 5:20 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:

> Steven, according to Borin's message she is talking exactly about outgoing
> gateway (sip_profiles/external/carrier.xml).
>
> And not about user's directory.
>
> Outgoing gateway can also work as incoming gateway, if you send your calls
> to it.
>
> Just want to make things clear.
>
> Vitalie
>
>
> 2011/3/13 Borin <katerin.borin at gmail.com>
>
>> Hi guys
>> thanks a lot
>>
>> I added
>>   <list name="carrier" default="deny">
>>     <node type="allow" cidr="carrier.ip/32"/>
>>     </list>
>> to the acl.conf.xml
>>
>> then    <param name="apply-inbound-acl" value="carrier"/>  in
>> sip_profiles/external/carrier.xml
>> <include>
>>   <gateway name="carrier">
>>
>>     <param name="username" value="anything"/>
>>     <param name="password" value="anything"/>
>>     <param name="auth-calls" value="false"/>
>>     <param name="apply-inbound-acl" value="carrier"/>
>>
>>     <param name="register" value="false"/>
>> </gateway>
>> </include>
>>
>> Works now
>>
>>
>> On Sat, Mar 12, 2011 at 10:48 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>
>>> That's for an outgoing gateway.
>>>
>>> Look at the cidr parameter of the user directory - it lets you
>>> authenticate a user based on IP rather than password.
>>>
>>> http://wiki.freeswitch.org/wiki/Acl#Users
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 12 March 2011 20:39, Borin <katerin.borin at gmail.com> wrote:
>>>
>>>> Hello FS users,
>>>> Is it possible in FS to allow all calls from a particular IP address
>>>> without authentication? like in asterisk with insecure port, invite.
>>>> I tried to add a new carrier in
>>>> /usr/local/freeswitch/conf/sip_profiles/external/new-carrier.xml
>>>> <include>
>>>>   <gateway name="new-carrier">
>>>>     <param name="username" value="anything"/>
>>>>     <param name="password" value="anything"/>
>>>>     <param name="realm" value="carrier.ip.address"/>
>>>>     <param name="register" value="false"/>
>>>>     <param name="caller-id-in-from" value="true"/>
>>>>   </gateway>
>>>> </include>
>>>>
>>>> But after restart it stil sends 407 Proxy Authentication Required to the
>>>> carrier. Is there any way to trust the carrier IP?
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From julf at julf.com  Sun Mar 13 21:18:23 2011
From: julf at julf.com (Johan Helsingius)
Date: Sun, 13 Mar 2011 19:18:23 +0100
Subject: [Freeswitch-users] multiple gateways and NAT
Message-ID: <4D7D0A6F.2020508@julf.com>

I am sure this question comes up all the time...

I want to use several different SIP gateway providers (for
different regions and services). Freeswitch seems to deal
with NAT just great with one external gateway, but multiple
gateways seems to be a rather thornier issue.

So, do I need totally separate profiles for each gateway?
And do they have to use different ports? And does that
mean that I need to get my firewall to port map each
of them separately (I would assume they all try to use
the same 5060 port to reach my system in case of an
incoming call)?

	Julf


From infos at madovsky.org  Sun Mar 13 21:26:34 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 13 Mar 2011 14:26:34 -0400
Subject: [Freeswitch-users] multiple gateways and NAT
References: <4D7D0A6F.2020508@julf.com>
Message-ID: <EE2DD131C4C543B3A0FDF5BBE7196CE9@e1705>

> So, do I need totally separate profiles for each gateway?
yes
> the same 5060 port to reach my system in case of an
> incoming call)?

yes

----- Original Message ----- 
From: "Johan Helsingius" <julf at julf.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Sunday, March 13, 2011 2:18 PM
Subject: [Freeswitch-users] multiple gateways and NAT


>I am sure this question comes up all the time...
> 
> I want to use several different SIP gateway providers (for
> different regions and services). Freeswitch seems to deal
> with NAT just great with one external gateway, but multiple
> gateways seems to be a rather thornier issue.
> 
> So, do I need totally separate profiles for each gateway?
> And do they have to use different ports? And does that
> mean that I need to get my firewall to port map each
> of them separately (I would assume they all try to use
> the same 5060 port to reach my system in case of an
> incoming call)?
> 
> Julf
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From mario_fs at mgtech.com  Sun Mar 13 21:39:42 2011
From: mario_fs at mgtech.com (Mario G)
Date: Sun, 13 Mar 2011 11:39:42 -0700
Subject: [Freeswitch-users] multiple gateways and NAT
In-Reply-To: <EE2DD131C4C543B3A0FDF5BBE7196CE9@e1705>
References: <4D7D0A6F.2020508@julf.com>
	<EE2DD131C4C543B3A0FDF5BBE7196CE9@e1705>
Message-ID: <878669AC-6527-4819-9CEB-98B74040E4C8@mgtech.com>

I have the same problem (3 ITSPs)  but use different ports for each, 5068, 5069, 5080. I had trouble using the same port for them but can't remember what the problem was. Mu big issue is I have 2 DSL lines,1 static and 1 dynamic, and no uPNP. I've never gotten FreeSwitch to handle two external DSLs so I have a new router to try out SIP ALG which I know some people hate but seems to be very good at handling the NAT and dual WANs, Freeswitch now uses both in case one goes down.

On Mar 13, 2011, at 11:26 AM, Madovsky wrote:

>> So, do I need totally separate profiles for each gateway?
> yes
>> the same 5060 port to reach my system in case of an
>> incoming call)?
> 
> yes
> 
> ----- Original Message ----- 
> From: "Johan Helsingius" <julf at julf.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Sunday, March 13, 2011 2:18 PM
> Subject: [Freeswitch-users] multiple gateways and NAT
> 
> 
>> I am sure this question comes up all the time...
>> 
>> I want to use several different SIP gateway providers (for
>> different regions and services). Freeswitch seems to deal
>> with NAT just great with one external gateway, but multiple
>> gateways seems to be a rather thornier issue.
>> 
>> So, do I need totally separate profiles for each gateway?
>> And do they have to use different ports? And does that
>> mean that I need to get my firewall to port map each
>> of them separately (I would assume they all try to use
>> the same 5060 port to reach my system in case of an
>> incoming call)?
>> 
>> Julf
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From covici at ccs.covici.com  Sun Mar 13 21:58:23 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sun, 13 Mar 2011 14:58:23 -0400
Subject: [Freeswitch-users] multiple gateways and NAT
In-Reply-To: <878669AC-6527-4819-9CEB-98B74040E4C8@mgtech.com>
References: <4D7D0A6F.2020508@julf.com>
	<EE2DD131C4C543B3A0FDF5BBE7196CE9@e1705>
	<878669AC-6527-4819-9CEB-98B74040E4C8@mgtech.com>
Message-ID: <23078.1300042703@ccs.covici.com>

I only have one external network, but I have several gateways all using
port 5060 with no problems.  My fs is on the public network and this
makes it easier..

Mario G <mario_fs at mgtech.com> wrote:

> I have the same problem (3 ITSPs)  but use different ports for each, 5068, 5069, 5080. I had trouble using the same port for them but can't remember what the problem was. Mu big issue is I have 2 DSL lines,1 static and 1 dynamic, and no uPNP. I've never gotten FreeSwitch to handle two external DSLs so I have a new router to try out SIP ALG which I know some people hate but seems to be very good at handling the NAT and dual WANs, Freeswitch now uses both in case one goes down.
> 
> On Mar 13, 2011, at 11:26 AM, Madovsky wrote:
> 
> >> So, do I need totally separate profiles for each gateway?
> > yes
> >> the same 5060 port to reach my system in case of an
> >> incoming call)?
> > 
> > yes
> > 
> > ----- Original Message ----- 
> > From: "Johan Helsingius" <julf at julf.com>
> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > Sent: Sunday, March 13, 2011 2:18 PM
> > Subject: [Freeswitch-users] multiple gateways and NAT
> > 
> > 
> >> I am sure this question comes up all the time...
> >> 
> >> I want to use several different SIP gateway providers (for
> >> different regions and services). Freeswitch seems to deal
> >> with NAT just great with one external gateway, but multiple
> >> gateways seems to be a rather thornier issue.
> >> 
> >> So, do I need totally separate profiles for each gateway?
> >> And do they have to use different ports? And does that
> >> mean that I need to get my firewall to port map each
> >> of them separately (I would assume they all try to use
> >> the same 5060 port to reach my system in case of an
> >> incoming call)?
> >> 
> >> Julf
> >> 
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From frank at rosengart.de  Sun Mar 13 21:39:46 2011
From: frank at rosengart.de (Frank Rosengart)
Date: Sun, 13 Mar 2011 19:39:46 +0100
Subject: [Freeswitch-users] PA and ALSA
In-Reply-To: <1299972873900-6165240.post@n2.nabble.com>
References: <4D7BF5E6.30201@rosengart.de>
	<1299972873900-6165240.post@n2.nabble.com>
Message-ID: <4D7D0F72.4050607@rosengart.de>

On 03/13/2011 12:34 AM, Jeff Lenk wrote:
> Same config under windows with celt - what are you using? if fs are you using
> git head?

Ok, did some more debugging. Seems like the "codec-ms" setting is
causing trouble. Once the 'pa looptest' works fine, the echo test has no
regular glitches. Only irregular dropouts (every 1-2 seconds). I will
now try with a second computer and with a LAN connection, to make sure
that are no lost packets while passing the internet.

(I'm using
http://files.freeswitch.org/windows/installer/x86/freeswitch.msi, and
fs/git-trunk/celt-0.10 on all systems)

But anyway, using windows is only my fallback option. Would be nice to
get it running on Linux.

Interestingly, 'Audacity', which is based on portaudio, too, finds all
my Alsa devices, where FS refuses to load mod_portaudio.

Anyone out there using mod_alsa? Or mod_portaudio with a newer Linux distri?


Frank


From infos at madovsky.org  Sun Mar 13 22:19:15 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 13 Mar 2011 15:19:15 -0400
Subject: [Freeswitch-users] invalid format after speex encoding
Message-ID: <EF57AA455929443E8859478AE0E8FDFA@e1705>

I tried to convert vm 16000hz wav sound files
to spx but FS can't read them: "invalid format"
is there any specail encoding settings to make them work ?

Thanks
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From julf at julf.com  Sun Mar 13 22:19:33 2011
From: julf at julf.com (Johan Helsingius)
Date: Sun, 13 Mar 2011 20:19:33 +0100
Subject: [Freeswitch-users] multiple gateways and NAT
In-Reply-To: <EE2DD131C4C543B3A0FDF5BBE7196CE9@e1705>
References: <4D7D0A6F.2020508@julf.com>
	<EE2DD131C4C543B3A0FDF5BBE7196CE9@e1705>
Message-ID: <4D7D18C5.2040404@julf.com>

>> So, do I need totally separate profiles for each gateway?
> yes

OK. And they have to be on different ports?

>> the same 5060 port to reach my system in case of an
>> incoming call)?
> 
> yes

So I have to selectively translate the incoming port
5060 (in the firewall) to a different port for each
gateway on the FS host?

	Julf


From infos at madovsky.org  Sun Mar 13 22:25:49 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 13 Mar 2011 15:25:49 -0400
Subject: [Freeswitch-users] multiple gateways and NAT
References: <4D7D0A6F.2020508@julf.com><EE2DD131C4C543B3A0FDF5BBE7196CE9@e1705>
	<4D7D18C5.2040404@julf.com>
Message-ID: <D81653003224456281F83BFA528134C4@e1705>


----- Original Message ----- 
From: "Johan Helsingius" <julf at julf.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Sunday, March 13, 2011 3:19 PM
Subject: Re: [Freeswitch-users] multiple gateways and NAT


>>> So, do I need totally separate profiles for each gateway?
>> yes
> 
> OK. And they have to be on different ports?
not necessary

> 
>>> the same 5060 port to reach my system in case of an
>>> incoming call)?
>> 
>> yes
> 
> So I have to selectively translate the incoming port
> 5060 (in the firewall) to a different port for each
> gateway on the FS host?
no


From julf at julf.com  Sun Mar 13 22:26:02 2011
From: julf at julf.com (Johan Helsingius)
Date: Sun, 13 Mar 2011 20:26:02 +0100
Subject: [Freeswitch-users] multiple gateways and NAT
In-Reply-To: <878669AC-6527-4819-9CEB-98B74040E4C8@mgtech.com>
References: <4D7D0A6F.2020508@julf.com>	<EE2DD131C4C543B3A0FDF5BBE7196CE9@e1705>
	<878669AC-6527-4819-9CEB-98B74040E4C8@mgtech.com>
Message-ID: <4D7D1A4A.4030505@julf.com>

Mario,

> Mu big issue is I have 2 DSL lines,1 static and 1 dynamic, and no uPNP.

Ouch, yes, I will be facing that issue too going forward... Somehow
I imagine a lot of people have run into these issues.

	Julf



From julf at julf.com  Sun Mar 13 22:27:37 2011
From: julf at julf.com (Johan Helsingius)
Date: Sun, 13 Mar 2011 20:27:37 +0100
Subject: [Freeswitch-users] multiple gateways and NAT
In-Reply-To: <23078.1300042703@ccs.covici.com>
References: <4D7D0A6F.2020508@julf.com>	<EE2DD131C4C543B3A0FDF5BBE7196CE9@e1705>	<878669AC-6527-4819-9CEB-98B74040E4C8@mgtech.com>
	<23078.1300042703@ccs.covici.com>
Message-ID: <4D7D1AA9.6000702@julf.com>

> My fs is on the public network and this makes it easier..

Indeed, that does make it much, much easier. But unfortunately,
with IPv4 space running out pretty quickly, that's a luxury
most of us can't afford... :(

	Julf


From steveayre at gmail.com  Sun Mar 13 22:40:44 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 13 Mar 2011 19:40:44 +0000
Subject: [Freeswitch-users] multiple gateways and NAT
In-Reply-To: <4D7D1AA9.6000702@julf.com>
References: <4D7D0A6F.2020508@julf.com>
	<EE2DD131C4C543B3A0FDF5BBE7196CE9@e1705>
	<878669AC-6527-4819-9CEB-98B74040E4C8@mgtech.com>
	<23078.1300042703@ccs.covici.com> <4D7D1AA9.6000702@julf.com>
Message-ID: <AANLkTimKAB-qNRs-hbr4+mYg=0E085aJr=Fw7QWp1Ri4@mail.gmail.com>

You can use a single IP on multiple profiles by using different ports.

-Steve


On 13 March 2011 19:27, Johan Helsingius <julf at julf.com> wrote:

> > My fs is on the public network and this makes it easier..
>
> Indeed, that does make it much, much easier. But unfortunately,
> with IPv4 space running out pretty quickly, that's a luxury
> most of us can't afford... :(
>
>        Julf
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From julf at julf.com  Sun Mar 13 22:48:49 2011
From: julf at julf.com (Johan Helsingius)
Date: Sun, 13 Mar 2011 20:48:49 +0100
Subject: [Freeswitch-users] multiple gateways and NAT
In-Reply-To: <D81653003224456281F83BFA528134C4@e1705>
References: <4D7D0A6F.2020508@julf.com><EE2DD131C4C543B3A0FDF5BBE7196CE9@e1705>	<4D7D18C5.2040404@julf.com>
	<D81653003224456281F83BFA528134C4@e1705>
Message-ID: <4D7D1FA1.5050600@julf.com>

>>>> So, do I need totally separate profiles for each gateway?
>>> yes
>>
>> OK. And they have to be on different ports?
> not necessary

OK, but... Why are the separate profiles needed then?
If they can use the same port, why can't they use the
same profile?

I am also more than a bit confused about the port 5060
vs. 5080 thing. So the external profile listens on 5080,
but won't the gateway try to use 5060 when it has an incoming
call for me? Or will the registration take care of that?

	Julf


From julf at julf.com  Sun Mar 13 23:01:32 2011
From: julf at julf.com (Johan Helsingius)
Date: Sun, 13 Mar 2011 21:01:32 +0100
Subject: [Freeswitch-users] multiple gateways and NAT
In-Reply-To: <AANLkTimKAB-qNRs-hbr4+mYg=0E085aJr=Fw7QWp1Ri4@mail.gmail.com>
References: <4D7D0A6F.2020508@julf.com>	<EE2DD131C4C543B3A0FDF5BBE7196CE9@e1705>	<878669AC-6527-4819-9CEB-98B74040E4C8@mgtech.com>	<23078.1300042703@ccs.covici.com>
	<4D7D1AA9.6000702@julf.com>
	<AANLkTimKAB-qNRs-hbr4+mYg=0E085aJr=Fw7QWp1Ri4@mail.gmail.com>
Message-ID: <4D7D229C.2000103@julf.com>

> You can use a single IP on multiple profiles by using different ports.

Sure, but most of us only get one IP in total - so unless you want
to run FS on the firewall, there will be NAT between FS and the Internet.

	Julf


From infos at madovsky.org  Sun Mar 13 23:01:37 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 13 Mar 2011 16:01:37 -0400
Subject: [Freeswitch-users] multiple gateways and NAT
References: <4D7D0A6F.2020508@julf.com><EE2DD131C4C543B3A0FDF5BBE7196CE9@e1705>	<4D7D18C5.2040404@julf.com><D81653003224456281F83BFA528134C4@e1705>
	<4D7D1FA1.5050600@julf.com>
Message-ID: <913499CF71FC4CA788510BA78E391891@e1705>

FS default for registered users is port 5060 and external gateway 5080.
but you can invert these port to follow a kind of standard SIP port for 
external gateway which is 5060.
I use on file per each external gateway, more easy to manage...


----- Original Message ----- 
From: "Johan Helsingius" <julf at julf.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Sunday, March 13, 2011 3:48 PM
Subject: Re: [Freeswitch-users] multiple gateways and NAT


>>>>> So, do I need totally separate profiles for each gateway?
>>>> yes
>>>
>>> OK. And they have to be on different ports?
>> not necessary
>
> OK, but... Why are the separate profiles needed then?
> If they can use the same port, why can't they use the
> same profile?
>
> I am also more than a bit confused about the port 5060
> vs. 5080 thing. So the external profile listens on 5080,
> but won't the gateway try to use 5060 when it has an incoming
> call for me? Or will the registration take care of that?
>
> Julf
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From frank at rosengart.de  Mon Mar 14 00:14:41 2011
From: frank at rosengart.de (Frank Rosengart)
Date: Sun, 13 Mar 2011 22:14:41 +0100
Subject: [Freeswitch-users] PA and ALSA
In-Reply-To: <4D7BF5E6.30201@rosengart.de>
References: <4D7BF5E6.30201@rosengart.de>
Message-ID: <4D7D33C1.1060009@rosengart.de>

On 03/12/2011 11:38 PM, Frank Rosengart wrote:

> 2. mod_portaudio can not find any audio devices.
> 3. Has the portaudio lib, which included with FS, the PA 'dmix patch'
> 4. Has anyone started to build mod_pulseaudio ?

Got it! I have replaced the libs/portaudio with the daily v19-snapshot
from http://www.portaudio.com/download.html and recompiled everything.

Now I can even use Pulseaudio! pa looptest runs fine.



Frank


From sos at sokhapkin.dyndns.org  Mon Mar 14 00:24:03 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sun, 13 Mar 2011 17:24:03 -0400
Subject: [Freeswitch-users] PA and ALSA
In-Reply-To: <4D7D33C1.1060009@rosengart.de>
References: <4D7BF5E6.30201@rosengart.de> <4D7D33C1.1060009@rosengart.de>
Message-ID: <201103131724.03970.sos@sokhapkin.dyndns.org>

FS developers' idea to include open source third party libraries to FS source 
distribution is a very bad idea... 

On Sunday 13 March 2011 17:14:41 Frank Rosengart wrote:
> On 03/12/2011 11:38 PM, Frank Rosengart wrote:
> > 2. mod_portaudio can not find any audio devices.
> > 3. Has the portaudio lib, which included with FS, the PA 'dmix patch'
> > 4. Has anyone started to build mod_pulseaudio ?
> 
> Got it! I have replaced the libs/portaudio with the daily v19-snapshot
> from http://www.portaudio.com/download.html and recompiled everything.
> 
> Now I can even use Pulseaudio! pa looptest runs fine.
> 
> 
> 
> Frank
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From lloyd.aloysius at gmail.com  Mon Mar 14 01:20:24 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 13 Mar 2011 18:20:24 -0400
Subject: [Freeswitch-users] Multi tenant and BLF
Message-ID: <AANLkTimwxsT9B7n1+8zzbRhLe2F4mKX3uJh3cb29S9CN@mail.gmail.com>

Hi All,

I could not make the BLF working in multi tenant environment. Any help is
appreciated. Discussion with BKW, he suggest to use multiple profiles. But
if I want to use this box for x number of tenants then I need to setup x no
of profiles. I like to use one profile for all my tenants.

Multi tenant settings works for months without any issue.

Here is my lab setup

*compa.mydomain.com*

Ext 201 - Cisco 504G + SPA500S Expansion Module - BLF 202,203
Ext 202 -SPA942
Ext 203 -SPA942

*BLF : Not working reliably .*

*compb.mydomain.com*

Ext 201 - Aastra 57i  and Topkeys Configured for BLF 202,203
Ext 202 - Aastra 6731i
Ext 203 - Aastra 6731

*BLF: working for couple of hours then completely stop working. A reboot of
the phone required*


*internal profile - The following lines comments*

<!--<param name="force-register-domain" value="$${domain}"/>-->
<!--<param name="force-subscription-domain" value="$${domain}"/>-->
<!--<param name="force-register-db-domain" value="$${domain}"/>-->

---

Any help is appreciated how to make it work with single profile


Thanks
Lloyd
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From anthony.minessale at gmail.com  Mon Mar 14 01:59:52 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 13 Mar 2011 17:59:52 -0500
Subject: [Freeswitch-users] PA and ALSA
In-Reply-To: <201103131724.03970.sos@sokhapkin.dyndns.org>
References: <4D7BF5E6.30201@rosengart.de> <4D7D33C1.1060009@rosengart.de>
	<201103131724.03970.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTikW9ObPMf1P+ruXuWqVdw0VR3f3NfFBTB89rO0y@mail.gmail.com>

That's what you think which you are welcome to do.
Maybe when you build your own cross platform application I can visit your
mailing list and disparage YOUR policies too....
On Mar 13, 2011 4:24 PM, "Sergey Okhapkin" <sos at sokhapkin.dyndns.org> wrote:
> FS developers' idea to include open source third party libraries to FS
source
> distribution is a very bad idea...
>
> On Sunday 13 March 2011 17:14:41 Frank Rosengart wrote:
>> On 03/12/2011 11:38 PM, Frank Rosengart wrote:
>> > 2. mod_portaudio can not find any audio devices.
>> > 3. Has the portaudio lib, which included with FS, the PA 'dmix patch'
>> > 4. Has anyone started to build mod_pulseaudio ?
>>
>> Got it! I have replaced the libs/portaudio with the daily v19-snapshot
>> from http://www.portaudio.com/download.html and recompiled everything.
>>
>> Now I can even use Pulseaudio! pa looptest runs fine.
>>
>>
>>
>> Frank
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From anthony.minessale at gmail.com  Mon Mar 14 02:02:01 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 13 Mar 2011 18:02:01 -0500
Subject: [Freeswitch-users] libss7 + asterisk woes!
In-Reply-To: <AANLkTikMbfhfGczFYVOCTwG7jnLe5hhyOHiMVv=t+sa_@mail.gmail.com>
References: <AANLkTikMbfhfGczFYVOCTwG7jnLe5hhyOHiMVv=t+sa_@mail.gmail.com>
Message-ID: <AANLkTinEoOeNZk5pPqU31DabM0FtWK9q+1KMVM_JzMyd@mail.gmail.com>

Check with sangoma they have ss7 for FreeSWITCH.
On Mar 13, 2011 10:19 AM, "Anita Hall" <anita.hall at simmortel.com> wrote:
> Hi
>
> I am using Asterisk on Sangoma cards :(
> I know I should have used Digium but now the customer has these cards and
I
> have to make them work with SS7 :(
>
> (And FreeSWITCH still does not support SS7 , is that right ?)
>
> Unable to receive incoming calls on chan_ss7, I have decided to give
another
> shot to libss7
>
> Could someone please share their configuration of dahdi/system.conf,
> chan_dahdi.conf, wanpipeX.conf with me for multiple signalling links ?
>
> I am especially confusing with the numbering schemes of CICs and
signalling
> channels. I have signalling links on the first 2 spans, on 16th channel
> each.
>
> So , how do I number after the 2nd channel onwards.
> Is it 32-64
> or 31-63
> or what ?
> :)
>
> My current config is
>
> debian:/usr/src# cat /etc/dahdi/system.conf
> #autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
> #autogenrated on 2011-03-13
> #Dahdi Channels Configurations
> #For detailed Dahdi options, view /etc/dahdi/system.conf.bak
> loadzone=us
> defaultzone=us
>
> #Sangoma A108 port 1 [slot:4 bus:5 span:1] <wanpipe1>
> span=1,1,0,ccs,hdb3,crc4
> bchan=1-15,17-31
> mtp2=16
>
> #Sangoma A108 port 2 [slot:4 bus:5 span:2] <wanpipe2>
> span=2,2,0,ccs,hdb3,crc4
> bchan=33-46,48-62
> mtp2=47
>
> #Sangoma A108 port 3 [slot:4 bus:5 span:3] <wanpipe3>
> span=3,0,0,ccs,hdb3,crc4
> bchan=63-93
>
> #Sangoma A108 port 4 [slot:4 bus:5 span:4] <wanpipe4>
> span=4,0,0,ccs,hdb3,crc4
> bchan=94-124
>
> #Sangoma A108 port 5 [slot:4 bus:5 span:5] <wanpipe5>
> span=5,0,0,ccs,hdb3,crc4
> bchan=125-155
>
> #Sangoma A108 port 6 [slot:4 bus:5 span:6] <wanpipe6>
> span=6,0,0,ccs,hdb3,crc4
> bchan=156-186
>
> #Sangoma A108 port 7 [slot:4 bus:5 span:7] <wanpipe7>
> span=7,0,0,ccs,hdb3,crc4
> bchan=187-217
>
> #Sangoma A108 port 8 [slot:4 bus:5 span:8] <wanpipe8>
> span=8,0,0,ccs,hdb3,crc4
> bchan=218-248
>
> #Sangoma A108 port 1 [slot:4 bus:7 span:9] <wanpipe9>
> span=9,0,0,ccs,hdb3,crc4
> bchan=249-279
>
> #Sangoma A108 port 2 [slot:4 bus:7 span:10] <wanpipe10>
> span=10,0,0,ccs,hdb3,crc4
> bchan=280-310
>
> #Sangoma A108 port 3 [slot:4 bus:7 span:11] <wanpipe11>
> span=11,0,0,ccs,hdb3,crc4
> bchan=311-341
>
> #Sangoma A108 port 4 [slot:4 bus:7 span:12] <wanpipe12>
> span=12,0,0,ccs,hdb3,crc4
> bchan=342-372
>
> #Sangoma A108 port 5 [slot:4 bus:7 span:13] <wanpipe13>
> span=13,0,0,ccs,hdb3,crc4
> bchan=373-403
>
> #Sangoma A108 port 6 [slot:4 bus:7 span:14] <wanpipe14>
> span=14,0,0,ccs,hdb3,crc4
> bchan=404-434
>
> #Sangoma A108 port 7 [slot:4 bus:7 span:15] <wanpipe15>
> span=15,0,0,ccs,hdb3,crc4
> bchan=435-465
>
> #Sangoma A108 port 8 [slot:4 bus:7 span:16] <wanpipe16>
> span=16,0,0,ccs,hdb3,crc4
> bchan=466-496
>
> debian:/usr/src# cat /etc/asterisk/chan_dahdi.conf
> ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
> ;autogenrated on 2011-03-13
> ;Dahdi Channels Configurations
> ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak
>
> [trunkgroups]
>
> [channels]
> context=default
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=no
> echocancelwhenbridged=no
> relaxdtmf=yes
> rxgain=0.0
> txgain=0.0
> group=1
> callgroup=1
> pickupgroup=1
> immediate=no
>
> ;Sangoma A108 port 1 [slot:4 bus:5 span:1] <wanpipe1>
> switchtype=national
> context=from-pstn
> group=0
> echocancel=no
> signalling=ss7
> ss7type=itu ;using the ITU variant
> ss7_called_nai=dynamic ;NAI for outgoing calls
> ss7_calling_nai=dynamic ;NAI for incoming calls
> ss7_internationalprefix=00 ;international prefix value for incoming calls
> ss7_nationalprefix=0 ;national prefix value for incoming calls
> ss7_subscriberprefix= ;subscriber prefix value for incoming
> calls
> ss7_unknownprefix= ;unknown prefix value for incoming calls
> ss7_explictacm=yes ;ACM is send as soon as call enters the
> dial plan...may not accepted yet though
> linkset=1 ;arbitrary name for this set of channels
> pointcode=13323 ;the point code for this system...aka
> SPC
> adjpointcode=12650 ;the point code for the system that we
> are signaling to... aka APC
> defaultdpc=12650 ;the point code for the system that the
> CICs will be negotiated with...aka DPC
> networkindicator=national_spare ;NI value for MTP3
> cicbeginswith=1 ;the starting value of the CICs
> channel=2-31 ;the channels that are CICs
> sigchan=1 ;the signaling channel
>
> ;Sangoma A108 port 2 [slot:4 bus:5 span:2] <wanpipe2>
> switchtype=national
> context=from-pstn
> group=0
> echocancel=no
> signalling=ss7
> ss7type=itu ;using the ITU variant
> ss7_called_nai=dynamic ;NAI for outgoing calls
> ss7_calling_nai=dynamic ;NAI for incoming calls
> ss7_internationalprefix=00 ;international prefix value for incoming calls
> ss7_nationalprefix=0 ;national prefix value for incoming calls
> ss7_subscriberprefix= ;subscriber prefix value for incoming
> calls
> ss7_unknownprefix= ;unknown prefix value for incoming calls
> ss7_explictacm=yes ;ACM is send as soon as call enters the
> dial plan...may not accepted yet though
> linkset=1 ;arbitrary name for this set of channels
> pointcode=13323 ;the point code for this system...aka
> SPC
> adjpointcode=12650 ;the point code for the system that we
> are signaling to... aka APC
> defaultdpc=12650 ;the point code for the system that the
> CICs will be negotiated with...aka DPC
> networkindicator=national_spare ;NI value for MTP3
> cicbeginswith=1 ;the starting value of the CICs
> channel=2-31 ;the channels that are CICs
> sigchan=1 ;the signaling channel
>
> ;Sangoma A108 port 3 [slot:4 bus:5 span:3] <wanpipe3>
> switchtype=national
> context=from-pstn
> group=0
> echocancel=no
> signalling=ss7
> channel =>63-77,79-93
>
> ;Sangoma A108 port 4 [slot:4 bus:5 span:4] <wanpipe4>
> switchtype=national
> context=from-pstn
> group=0
> echocancel=no
> signalling=ss7
> channel =>94-108,110-124
>
> ;Sangoma A108 port 5 [slot:4 bus:5 span:5] <wanpipe5>
> switchtype=national
> context=from-pstn
> group=0
> echocancel=no
> signalling=ss7
> channel =>125-139,141-155
>
> ;Sangoma A108 port 6 [slot:4 bus:5 span:6] <wanpipe6>
> switchtype=national
> context=from-pstn
> group=0
> echocancel=no
> signalling=ss7
> channel =>156-170,172-186
>
> ;Sangoma A108 port 7 [slot:4 bus:5 span:7] <wanpipe7>
> switchtype=national
> context=from-pstn
> group=0
> echocancel=no
> signalling=ss7
> channel =>187-201,203-217
>
> ;Sangoma A108 port 8 [slot:4 bus:5 span:8] <wanpipe8>
> switchtype=national
> context=from-pstn
> group=0
> echocancel=no
> signalling=ss7
> channel =>218-232,234-248
>
> ;Sangoma A108 port 1 [slot:4 bus:7 span:9] <wanpipe9>
> switchtype=national
> context=from-pstn
> group=0
> echocancel=no
> signalling=ss7
> channel =>249-263,265-279
>
> ;Sangoma A108 port 2 [slot:4 bus:7 span:10] <wanpipe10>
> switchtype=national
> context=from-pstn
> group=0
> echocancel=no
> signalling=ss7
> channel =>280-294,296-310
>
> ;Sangoma A108 port 3 [slot:4 bus:7 span:11] <wanpipe11>
> switchtype=national
> context=from-pstn
> group=0
> echocancel=no
> signalling=ss7
> channel =>311-325,327-341
>
> ;Sangoma A108 port 4 [slot:4 bus:7 span:12] <wanpipe12>
> switchtype=national
> context=from-pstn
> group=0
> echocancel=no
> signalling=ss7
> channel =>342-356,358-372
>
> ;Sangoma A108 port 5 [slot:4 bus:7 span:13] <wanpipe13>
> switchtype=national
> context=from-pstn
> group=0
> echocancel=no
> signalling=ss7
> channel =>373-387,389-403
>
> ;Sangoma A108 port 6 [slot:4 bus:7 span:14] <wanpipe14>
> switchtype=national
> context=from-pstn
> group=0
> echocancel=no
> signalling=ss7
> channel =>404-418,420-434
>
> ;Sangoma A108 port 7 [slot:4 bus:7 span:15] <wanpipe15>
> switchtype=national
> context=from-pstn
> group=0
> echocancel=no
> signalling=ss7
> channel =>435-449,451-465
>
> ;Sangoma A108 port 8 [slot:4 bus:7 span:16] <wanpipe16>
> switchtype=national
> context=from-pstn
> group=0
> echocancel=no
> signalling=ss7
> channel =>466-480,482-496
>
>
> debian:/usr/src# cat /etc/wanpipe/wanpipe1.conf
> #================================================
> # WANPIPE1 Configuration File
> #================================================
> #
> # Date: Wed Dec 6 20:29:03 UTC 2006
> #
> # Note: This file was generated automatically
> # by /usr/local/sbin/setup-sangoma program.
> #
> # If you want to edit this file, it is
> # recommended that you use wancfg program
> # to do so.
> #================================================
> # Sangoma Technologies Inc.
> #================================================
>
> [devices]
> wanpipe1 = WAN_AFT_TE1, Comment
>
> [interfaces]
> w1g1 = wanpipe1, , TDM_VOICE, Comment
>
> [wanpipe1]
> CARD_TYPE = AFT
> S514CPU = A
> CommPort = PRI
> AUTO_PCISLOT = NO
> PCISLOT = 4
> PCIBUS = 5
> FE_MEDIA = E1
> FE_LCODE = HDB3
> FE_FRAME = CRC4
> FE_LINE = 1
> TE_CLOCK = NORMAL
> TE_REF_CLOCK = 0
> TE_SIG_MODE = CCS
> TE_HIGHIMPEDANCE = NO
> TE_RX_SLEVEL = 430
> LBO = 120OH
> FE_TXTRISTATE = NO
> MTU = 1500
> UDPPORT = 9000
> TTL = 255
> IGNORE_FRONT_END = NO
> TDMV_SPAN = 1
> TDMV_DCHAN = 0
> TE_AIS_MAINTENANCE = NO #NO: defualt YES: Start port in AIS Blue
> Alarm and keep line down
> #wanpipemon -i w1g1 -c Ttx_ais_off to
> disable AIS maintenance mode
> #wanpipemon -i w1g1 -c Ttx_ais_on to enable AIS maintenance mode
> TDMV_HW_DTMF = NO # YES: receive dtmf events from hardware
> TDMV_HW_FAX_DETECT = NO # YES: receive fax 1100hz events from hardware
> HWEC_OPERATION_MODE = OCT_NORMAL # OCT_NORMAL: echo cancelation
> enabled with nlp (default)
> # OCT_SPEECH: improves software tone detection by disabling NLP (echo
> possible)
> # OCT_NO_ECHO:disables echo cancelation but allows VQE/tone functions.
> HWEC_DTMF_REMOVAL = NO # NO: default YES: remove dtmf out of
> incoming media (must have hwdtmf enabled)
> HWEC_NOISE_REDUCTION = NO # NO: default YES: reduces noise on the
> line - could break fax
> HWEC_ACUSTIC_ECHO = NO # NO: default YES: enables acustic echo
> cancelation
> HWEC_NLP_DISABLE = NO # NO: default YES: guarantees software tone
> detection (possible echo)
> HWEC_TX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio level
> to be maintained (-20 default)
> HWEC_RX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio level
> to be maintained (-20 default)
> HWEC_TX_GAIN = 0 # 0: disable -24-24: db values to be applied
> to tx signal
> HWEC_RX_GAIN = 0 # 0: disable -24-24: db values to be applied
> to tx signal
>
> [w1g1]
> ACTIVE_CH = ALL
> TDMV_HWEC = NO
> MTU = 8
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From jeff at jefflenk.com  Mon Mar 14 02:05:17 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sun, 13 Mar 2011 16:05:17 -0700 (PDT)
Subject: [Freeswitch-users] PA and ALSA
In-Reply-To: <201103131724.03970.sos@sokhapkin.dyndns.org>
References: <4D7BF5E6.30201@rosengart.de> <4D7D33C1.1060009@rosengart.de>
	<201103131724.03970.sos@sokhapkin.dyndns.org>
Message-ID: <1300057517408-6167368.post@n2.nabble.com>

I'm sure there were very good reasons for these decisions when they were
made. So rather than criticizing those decisions why not help by finding the
problems and submitting patches to fix them.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/PA-and-ALSA-tp6165168p6167368.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From sos at sokhapkin.dyndns.org  Mon Mar 14 02:21:17 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sun, 13 Mar 2011 19:21:17 -0400
Subject: [Freeswitch-users] PA and ALSA
In-Reply-To: <1300057517408-6167368.post@n2.nabble.com>
References: <4D7BF5E6.30201@rosengart.de>
	<201103131724.03970.sos@sokhapkin.dyndns.org>
	<1300057517408-6167368.post@n2.nabble.com>
Message-ID: <201103131921.18213.sos@sokhapkin.dyndns.org>

The attempt to isolate core FS code from third party libraries has been done 
by gentoo linux team already (inclusion of third party open source code is 
strictly against gentoo philosophy), but the attempt has been abandoned 
because of lack of support from upstream.

On Sunday 13 March 2011 19:05:17 Jeff Lenk wrote:
> I'm sure there were very good reasons for these decisions when they were
> made. So rather than criticizing those decisions why not help by finding
> the problems and submitting patches to fix them.
> 
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/PA-and-ALSA-tp6165168p616736
> 8.html Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From frank at rosengart.de  Mon Mar 14 02:34:22 2011
From: frank at rosengart.de (Frank Rosengart)
Date: Mon, 14 Mar 2011 00:34:22 +0100
Subject: [Freeswitch-users] PA and ALSA
In-Reply-To: <1300057517408-6167368.post@n2.nabble.com>
References: <4D7BF5E6.30201@rosengart.de>
	<4D7D33C1.1060009@rosengart.de>	<201103131724.03970.sos@sokhapkin.dyndns.org>
	<1300057517408-6167368.post@n2.nabble.com>
Message-ID: <4D7D547E.1020905@rosengart.de>

On 03/14/2011 12:05 AM, Jeff Lenk wrote:
> I'm sure there were very good reasons for these decisions when they were
> made. 

The last release of portaudio dates back to 2007. If FS includes such a
vintage version of PA, a hint to linux users on the FS-wiki might be
appropiate...

Including external libraries is always a trade-off.

Is mod_alsa still maintained?


Frank


From sos at sokhapkin.dyndns.org  Mon Mar 14 03:03:01 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sun, 13 Mar 2011 20:03:01 -0400
Subject: [Freeswitch-users] PA and ALSA
In-Reply-To: <4D7D547E.1020905@rosengart.de>
References: <4D7BF5E6.30201@rosengart.de>
	<1300057517408-6167368.post@n2.nabble.com>
	<4D7D547E.1020905@rosengart.de>
Message-ID: <201103132003.01442.sos@sokhapkin.dyndns.org>

http://bugs.gentoo.org/150527

On Sunday 13 March 2011, Frank Rosengart wrote:
> On 03/14/2011 12:05 AM, Jeff Lenk wrote:
> > I'm sure there were very good reasons for these decisions when they were
> > made.
> 
> The last release of portaudio dates back to 2007. If FS includes such a
> vintage version of PA, a hint to linux users on the FS-wiki might be
> appropiate...
> 
> Including external libraries is always a trade-off.
> 
> Is mod_alsa still maintained?
> 
> 
> Frank
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Mon Mar 14 03:08:20 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 13 Mar 2011 19:08:20 -0500
Subject: [Freeswitch-users] PA and ALSA
In-Reply-To: <201103131921.18213.sos@sokhapkin.dyndns.org>
References: <4D7BF5E6.30201@rosengart.de>
	<201103131724.03970.sos@sokhapkin.dyndns.org>
	<1300057517408-6167368.post@n2.nabble.com>
	<201103131921.18213.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTikYUsbU6=gd=SrRd_a=F=j40eV=D0YW4SLCNuQU@mail.gmail.com>

2 things to keep in mind.

1 Try to wrap your head around making a stable application that runs on
windows, unix including mac and some bsd, and unix-like distros such as
those mentioned.  Not all of our dependancies are stable on all of those
platforms or even easily available in some cases.

2 This is a community project, if some versions of libraries we are using
are older maybe they have been working fine or the modules are not used
enough to warrant attention.

The normal process would be:

The guy reports a problem and suggests its fixed in a newer version.  Now
try this code on every supported platform and make sure it works across the
board, if not, well, we can't blindly add it.  All of this can be done
without the ungrateful cracks and side stories about gentoo and we can
happily update the code.

I don't go to gentoo forum and complain that custom patching the kernel and
libc annoys me or about how I am not interested in being forced into their
religion to have my software in their distro even though it does not install
any third party libs, FS only static links in bits of code from in tree
depends.

Furthermore, we constantly invite people to assist with platform and
packaging and nobody offers anything but complaints.

An example of why we don't just blindly use system libs:
The latest sqlite causes 25% loss in performance and guarenteed segfaults
and occasional deadlock under load.
This is a place where people complain about 1% cpu usage.

Alsa mod is specifically for the nokia N800 N810 devices.
Stick with pa or help us write something new from scratch and don't forget
linux is not alone in the computing world.

On Mar 13, 2011 6:22 PM, "Sergey Okhapkin" <sos at sokhapkin.dyndns.org> wrote:
>
> The attempt to isolate core FS code from third party libraries has been
done
> by gentoo linux team already (inclusion of third party open source code is
> strictly against gentoo philosophy), but the attempt has been abandoned
> because of lack of support from upstream.
>
> On Sunday 13 March 2011 19:05:17 Jeff Lenk wrote:
> > I'm sure there were very good reasons for these decisions when they were
> > made. So rather than criticizing those decisions why not help by finding
> > the problems and submitting patches to fix them.
> >
> > --
> > View this message in context:
> >
http://freeswitch-users.2379917.n2.nabble.com/PA-and-ALSA-tp6165168p616736
> > 8.html Sent from the freeswitch-users mailing list archive at
Nabble.com.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From sos at sokhapkin.dyndns.org  Mon Mar 14 03:32:19 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sun, 13 Mar 2011 20:32:19 -0400
Subject: [Freeswitch-users] PA and ALSA
In-Reply-To: <AANLkTikYUsbU6=gd=SrRd_a=F=j40eV=D0YW4SLCNuQU@mail.gmail.com>
References: <4D7BF5E6.30201@rosengart.de>
	<201103131921.18213.sos@sokhapkin.dyndns.org>
	<AANLkTikYUsbU6=gd=SrRd_a=F=j40eV=D0YW4SLCNuQU@mail.gmail.com>
Message-ID: <201103132032.19044.sos@sokhapkin.dyndns.org>

The power of gentoo linux is that ebuild can specify a range of required 
versions for a third party library. Maybe something can be implemented in FS 
build script too? Just check a version of the installed third party library in 
configure script.

On Sunday 13 March 2011, Anthony Minessale wrote:
> 2 things to keep in mind.
> 
> 1 Try to wrap your head around making a stable application that runs on
> windows, unix including mac and some bsd, and unix-like distros such as
> those mentioned.  Not all of our dependancies are stable on all of those
> platforms or even easily available in some cases.
> 
> 2 This is a community project, if some versions of libraries we are using
> are older maybe they have been working fine or the modules are not used
> enough to warrant attention.
> 
> The normal process would be:
> 
> The guy reports a problem and suggests its fixed in a newer version.  Now
> try this code on every supported platform and make sure it works across the
> board, if not, well, we can't blindly add it.  All of this can be done
> without the ungrateful cracks and side stories about gentoo and we can
> happily update the code.
> 
> I don't go to gentoo forum and complain that custom patching the kernel and
> libc annoys me or about how I am not interested in being forced into their
> religion to have my software in their distro even though it does not
> install any third party libs, FS only static links in bits of code from in
> tree depends.
> 
> Furthermore, we constantly invite people to assist with platform and
> packaging and nobody offers anything but complaints.
> 
> An example of why we don't just blindly use system libs:
> The latest sqlite causes 25% loss in performance and guarenteed segfaults
> and occasional deadlock under load.
> This is a place where people complain about 1% cpu usage.
> 
> Alsa mod is specifically for the nokia N800 N810 devices.
> Stick with pa or help us write something new from scratch and don't forget
> linux is not alone in the computing world.
> 
> On Mar 13, 2011 6:22 PM, "Sergey Okhapkin" <sos at sokhapkin.dyndns.org> wrote:
> > The attempt to isolate core FS code from third party libraries has been
> 
> done
> 
> > by gentoo linux team already (inclusion of third party open source code
> > is strictly against gentoo philosophy), but the attempt has been
> > abandoned because of lack of support from upstream.
> > 
> > On Sunday 13 March 2011 19:05:17 Jeff Lenk wrote:
> > > I'm sure there were very good reasons for these decisions when they
> > > were made. So rather than criticizing those decisions why not help by
> > > finding the problems and submitting patches to fix them.
> > > 
> > > --
> 
> > > View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/PA-and-ALSA-tp6165168p616736
> 
> > > 8.html Sent from the freeswitch-users mailing list archive at
> 
> Nabble.com.
> 
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> > > s http://www.freeswitch.org
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org



From mario_fs at mgtech.com  Mon Mar 14 04:07:11 2011
From: mario_fs at mgtech.com (Mario G)
Date: Sun, 13 Mar 2011 18:07:11 -0700
Subject: [Freeswitch-users] multiple gateways and NAT
In-Reply-To: <913499CF71FC4CA788510BA78E391891@e1705>
References: <4D7D0A6F.2020508@julf.com><EE2DD131C4C543B3A0FDF5BBE7196CE9@e1705>	<4D7D18C5.2040404@julf.com><D81653003224456281F83BFA528134C4@e1705>
	<4D7D1FA1.5050600@julf.com>
	<913499CF71FC4CA788510BA78E391891@e1705>
Message-ID: <C7F29C81-A205-426F-B296-47FA849EF4C3@mgtech.com>

I remembered why I needed to use different ports in one case: Two of the three accounts are at the same ITSP, in this case since the IP address is the same on the two accounts you need to use different ports.

On Mar 13, 2011, at 1:01 PM, Madovsky wrote:

> FS default for registered users is port 5060 and external gateway 5080.
> but you can invert these port to follow a kind of standard SIP port for 
> external gateway which is 5060.
> I use on file per each external gateway, more easy to manage...
> 
> 
> ----- Original Message ----- 
> From: "Johan Helsingius" <julf at julf.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Sunday, March 13, 2011 3:48 PM
> Subject: Re: [Freeswitch-users] multiple gateways and NAT
> 
> 
>>>>>> So, do I need totally separate profiles for each gateway?
>>>>> yes
>>>> 
>>>> OK. And they have to be on different ports?
>>> not necessary
>> 
>> OK, but... Why are the separate profiles needed then?
>> If they can use the same port, why can't they use the
>> same profile?
>> 
>> I am also more than a bit confused about the port 5060
>> vs. 5080 thing. So the external profile listens on 5080,
>> but won't the gateway try to use 5060 when it has an incoming
>> call for me? Or will the registration take care of that?
>> 
>> Julf
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From erkan at speedingtrade.com  Mon Mar 14 11:19:35 2011
From: erkan at speedingtrade.com (=?iso-8859-9?B?RXJrYW4g3G5s/A==?=)
Date: Mon, 14 Mar 2011 10:19:35 +0200
Subject: [Freeswitch-users] FreeSwitch for Windows
References: <E04386CF-006E-4F87-A2B0-FA6C810A34B7@mgtech.com><022B01F7-0577-457B-94C8-D6F47B14E1F0@mgtech.com>
	<4D77F344.3050207@gmail.com>
Message-ID: <81C2CEF80046FB4F863A60D4347DD33A0C56FB@server1.st.local>

Dear Fernando,

?'m using FS on 4 different Server's since over 1 year successfully.
My last server is a 64 bit platform and all things that i use worked fine for me.

The only thing that i see is some file or folder path are not correctly and formated for Unix/Linux systems.
This one must be changed manual for Windows path name.

For example    sounds/xxx/.... and so on the file or path can not be found on Windows.
You see that in the console of FS. But i changed it manual to sounds\xxx\....   and all things work fine.
Also some modules that i need worked for me.

But with boards or other hardware i can not tell you something, because i don't use it.

Kind regards
Erkan

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Fernando Berretta
Sent: Wednesday, March 09, 2011 11:38 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] FreeSwitch for Windows

Hi,

I'm newbie in FreeSwitch and would like to know if freeswitch could be used for production under Windows platforms ? telephony boards like Digium / OpenVox, Sangoma, etc are going to work ok under Windows ? is there any problem / limitation ? E1 MFCR2 Trunk works ?

Thanks for your time,
Best Regards,
Fernando

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


From peter.olsson at visionutveckling.se  Mon Mar 14 11:34:21 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 14 Mar 2011 09:34:21 +0100
Subject: [Freeswitch-users] FreeSwitch for Windows
In-Reply-To: <81C2CEF80046FB4F863A60D4347DD33A0C56FB@server1.st.local>
References: <E04386CF-006E-4F87-A2B0-FA6C810A34B7@mgtech.com><022B01F7-0577-457B-94C8-D6F47B14E1F0@mgtech.com>
	<4D77F344.3050207@gmail.com>
	<81C2CEF80046FB4F863A60D4347DD33A0C56FB@server1.st.local>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2F6C275@cooper>

File paths in Windows should work with either / or \ without any problems, I've never seen any issues with this, and I've been running FS on Windows for at least a year now.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Erkan ?nl?
Skickat: den 14 mars 2011 09:20
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] FreeSwitch for Windows

Dear Fernando,

?'m using FS on 4 different Server's since over 1 year successfully.
My last server is a 64 bit platform and all things that i use worked fine for me.

The only thing that i see is some file or folder path are not correctly and formated for Unix/Linux systems.
This one must be changed manual for Windows path name.

For example    sounds/xxx/.... and so on the file or path can not be found on Windows.
You see that in the console of FS. But i changed it manual to sounds\xxx\....   and all things work fine.
Also some modules that i need worked for me.

But with boards or other hardware i can not tell you something, because i don't use it.

Kind regards
Erkan

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Fernando Berretta
Sent: Wednesday, March 09, 2011 11:38 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] FreeSwitch for Windows

Hi,

I'm newbie in FreeSwitch and would like to know if freeswitch could be used for production under Windows platforms ? telephony boards like Digium / OpenVox, Sangoma, etc are going to work ok under Windows ? is there any problem / limitation ? E1 MFCR2 Trunk works ?

Thanks for your time,
Best Regards,
Fernando

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

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

!DSPAM:4d7dd1c232761971858288!



From julf at julf.com  Mon Mar 14 12:03:16 2011
From: julf at julf.com (Johan Helsingius)
Date: Mon, 14 Mar 2011 10:03:16 +0100
Subject: [Freeswitch-users] multiple gateways and NAT
In-Reply-To: <C7F29C81-A205-426F-B296-47FA849EF4C3@mgtech.com>
References: <4D7D0A6F.2020508@julf.com><EE2DD131C4C543B3A0FDF5BBE7196CE9@e1705>	<4D7D18C5.2040404@julf.com><D81653003224456281F83BFA528134C4@e1705>	<4D7D1FA1.5050600@julf.com>	<913499CF71FC4CA788510BA78E391891@e1705>
	<C7F29C81-A205-426F-B296-47FA849EF4C3@mgtech.com>
Message-ID: <4D7DD9D4.3000002@julf.com>

> I remembered why I needed to use different ports in one case: Two
> of the three accounts are at the same ITSP, in this case since the
> IP address is the same on the two accounts you need to use different ports.

Ah, I have sort of the same case - 4 accounts with one ITSP providing
4 different incoming DID's (replacing an old ISDN connection), and
2 separate single ITSP accounts for international calls.

But I am still very confused about the need for different ports.
This is not a complaint, just an observation, but I do think the
multiple port thing is one of the more confusing aspects of FS,
and something other SIP switches seem to do without.

On outgoing connections, we register with the ITSP gateway (by the
way, does the SIP registration register both an ip address and
a port number, or just the ip address?), and on incoming connections
they either have a destination number/DID that routes them to the
right place in the dialplan, or they register with us (using
username/password) in the case of SIP clients. Security can be
handled with ACL's. So I can't get my head around to why separate
ports are needed - maybe someone can point out what I am missing?

	Julf



From hkalyoncu at gmail.com  Mon Mar 14 12:08:08 2011
From: hkalyoncu at gmail.com (hkalyoncu)
Date: Mon, 14 Mar 2011 02:08:08 -0700 (PDT)
Subject: [Freeswitch-users] strange errors when using mod_python
In-Reply-To: <AANLkTinLJLU3+fc=L3EwAAJe_U-kpTi7HvB40wUu78zD@mail.gmail.com>
References: <1299848563170-6161361.post@n2.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2F6C044@cooper>
	<1299850368472-6161441.post@n2.nabble.com>
	<AANLkTinLJLU3+fc=L3EwAAJe_U-kpTi7HvB40wUu78zD@mail.gmail.com>
Message-ID: <1300093688368-6168292.post@n2.nabble.com>

yes i dumped the generated xml output to a file.
here is a sample generated xml:

        <?xml version="1.0" encoding="UTF-8" standalone="no"?>
        
         
          
           
            
                
                
                
            
           
          
         
        

the xml output is as expected



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/strange-errors-when-using-mod-python-tp6161361p6168292.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From hkalyoncu at gmail.com  Mon Mar 14 12:11:51 2011
From: hkalyoncu at gmail.com (hkalyoncu)
Date: Mon, 14 Mar 2011 02:11:51 -0700 (PDT)
Subject: [Freeswitch-users] strange errors when using mod_python
In-Reply-To: <1300093688368-6168292.post@n2.nabble.com>
References: <1299848563170-6161361.post@n2.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2F6C044@cooper>
	<1299850368472-6161441.post@n2.nabble.com>
	<AANLkTinLJLU3+fc=L3EwAAJe_U-kpTi7HvB40wUu78zD@mail.gmail.com>
	<1300093688368-6168292.post@n2.nabble.com>
Message-ID: <1300093911185-6168304.post@n2.nabble.com>

xml is not displayed correctly after posting 

here it is again without tags:

        xml version="1.0" encoding="UTF-8" standalone="no"
        document type="freeswitch/xml"
         section name="dialplan" description="RE Dial Plan For FreeSWITCH"
          context name="context_3"
           extension name="generated"
            condition
                action application="answer"
                action application="bridge"
data="sofia/sipinterface_5/xxxxxxxxxx at 10.10.1.5:5060"
                action application="hangup"
            condition
           extension
          context
         section
        document
 

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/strange-errors-when-using-mod-python-tp6161361p6168304.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From pkelly at gmail.com  Mon Mar 14 12:23:14 2011
From: pkelly at gmail.com (Pete Kelly)
Date: Mon, 14 Mar 2011 09:23:14 +0000
Subject: [Freeswitch-users] exec app on originate
In-Reply-To: <AANLkTinbgUe0M_MCFZKA96zfErht3pmpw-UhC+sX3fwF@mail.gmail.com>
References: <748093B885AD48CE86278841FBEA1159@e1705>
	<AANLkTikmsGve+DMf9RzyyB3wBH0dgpT_1gkG4hNuq7du@mail.gmail.com>
	<AANLkTinoN7tdVDA8e+XLcjFGV-Z1KhP1a3+gLSGNaGOn@mail.gmail.com>
	<AANLkTinbgUe0M_MCFZKA96zfErht3pmpw-UhC+sX3fwF@mail.gmail.com>
Message-ID: <AANLkTinKBEaO08BzciysDsKiZa6F5vcPKAQ1T1Z_DUYF@mail.gmail.com>

Thank you

as it happens I am using bgapi originate via an XMLRPC call anyway... I'll
 look into the BACKGROUND_JOB event and mod_xml_cdr

On 11 March 2011 21:38, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Mar 11, 2011 at 1:04 AM, Pete Kelly <pkelly at gmail.com> wrote:
>
>>
>>
>> On 25 February 2011 01:04, Anthony Minessale <anthony.minessale at gmail.com
>> > wrote:
>>
>>> early media
>>> if you want after answer do
>>>
>>> originate {ignore_early_media=true}sofia/gateway/blabla/9999999999
>>> &app(hahaha)
>>>
>>
>> Is there any way to get the app launched even earlier? I need to be able
>> to act upon a 4XX/5XX class response to the INVITE within the app.
>>
>
> The originate API does not work that way. The &app(foo) won't be executed
> unless the originate is successful. If you get a 4xx/5xx then the originate
> will fail and &app(foo) won't ever get executed. There is no notion of
> "execute_on_fail" since a failed call means there is no channel/session to
> send through the dialplan or through an application.
>
> Depending on exactly what you're doing you probably need to be using "bgapi
> originate ..." and then checking the BACKGROUND_JOB event to see what
> happened. Alternatively you could be listening/watching for CDRs and then
> acting upon them. Alternatively using mod_xml_cdr is an easy way to receive
> notification of failed calls. The channel variable
> "proto_specific_hangup_cause" will have the "SIP:4xx" value, and you can
> react to it accordingly.
>
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From pkelly at gmail.com  Mon Mar 14 12:30:28 2011
From: pkelly at gmail.com (Pete Kelly)
Date: Mon, 14 Mar 2011 09:30:28 +0000
Subject: [Freeswitch-users] Memory leak
Message-ID: <AANLkTim3UsX4z_URe1Ya40uP4AnVG3s3NiYkew=ZY8+O@mail.gmail.com>

Hi

We are using Freeswitch (from git sources) in a production environment to
handle an IVR system using a simple set of dialplans and lua scripts.
OpenSIPS 1.6 and mysql are also installed and running as part of the same
IVR application.

However over time (approx 1 week) the memory usage of Freeswitch increases,
eventually creeping into swap space forcing us to restart Freeswitch. The
IVR handles approx 40 concurrent calls constantly, and we have instances of
the IVR running on a 32bit Etch and 64bit Lenny boxes - both exhibit the
same issue.

We have also tried the Sangoma freeswitch branch in case that contained any
fixes, however we see the same issues.

Can anybody advise if this is a known issue with Freeswitch? Is it a memory
leak or is it possible to limit the amount of memory Freeswitch will use,
forcing it to garbage collect when it reaches the limit?

Any advice on where to look/debug next would be appreciated.

Thanks

Pete
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From 29prime at gmail.com  Mon Mar 14 11:25:43 2011
From: 29prime at gmail.com (Prime)
Date: Mon, 14 Mar 2011 10:25:43 +0200
Subject: [Freeswitch-users] probem with configuring DIDs
Message-ID: <AANLkTi=aK=K-25NL87WYT4m-GQJK_DT=rW6PjabKzAT0@mail.gmail.com>

Hey!

This most likely is a newb mistake but I'm stuck at the moment.

Recently I had to add another provider to FS to handle incoming calls.
And now I have a problem with calling my new provider's DID number on
my FS.
I looked at FS logs.
Usually FS log says something like this " [INFO]
mod_dialplan_xml.c:418 Processing caller_number->My_DID in context
public" and then routes DID number to local extension,
but now I see something like this " [INFO] mod_dialplan_xml.c:418
Processing caller_number->caller_number in context public ".
Why new numbers does not match the regex?

The number are configured exactly as old ones.

Thanks in advance

Best regards


From gavin.henry at gmail.com  Mon Mar 14 12:39:33 2011
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 14 Mar 2011 09:39:33 +0000
Subject: [Freeswitch-users] FreeSWITCH OCF Resource Agent to run "sofia
	recover"
Message-ID: <AANLkTim09Rd=o4Webt3YVpY83Essna=JXa3k6BFEGedN@mail.gmail.com>

Hi all,

I saw this in Jan but gather the person is at the same point as us:

http://lists.freeswitch.org/pipermail/freeswitch-users/2011-January/066729.html

I'm putting together two test VirtualBox VMs which have pacemaker and
DRBD running. The actual plan is FreeSWITCH, PostgreSQL, Apache and
FusionPBX on the one box to start with in testing. FusionPBX
(http://www.fusionpbx.com) will be running in full Multi-tenant mode
as we've just sponsored the FusionPBX project to get the last bits
done.

Apache and Pg have resource agents already and FS will have its data
as per the FS HA wiki docs pointing to a Pg database.

The last piece is how to add in FreeSWITCH and run "sofia recover"
"correctly". I'd rather not have a "hacked" together OCF resource
agent for FS but a proper one that is in the FS git repo.

I'll happily write this all up on the existing wiki pages, but for the
multi-tenant parts you'll need to speak to the FusionPBX guys for help
on that as you'll need commercial support from them for the setup. FYI
we'll also be using LINBIT packages
(http://www.linbit.com/en/products-services/linbit-cluster-stack-support/)
for our new production platform for the stacked DRBD setup for
off-site disaster recovery:

http://www.drbd.org/users-guide/s-pacemaker-stacked-resources.html
http://www.drbd.org/users-guide/s-three-way-repl.html

Thanks,

Gavin.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.surevoip.co.uk


From benkokakao at gmail.com  Mon Mar 14 13:59:23 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Mon, 14 Mar 2011 11:59:23 +0100
Subject: [Freeswitch-users] Alcatel 4018 - "481 Call Does Not Exist" on
	re-REGISTER
Message-ID: <AANLkTinjt-rh_ieSMwGcWsJbL4RwfRz4SCTFmUht+Z9L@mail.gmail.com>

Hi!

I've setup an Alcatel 4018 in SIP-Mode to register with FreeSWITCH.
The initial registration after startup works fine - the phone sends a
REGISTER -> FreeSWITCH answers with 401 -> UA sends REGISTER including
authorization-data -> 200 OK

But after a while the phone's registration expires, as it sends
subsequent REGISTERs _with_ authorization-data right away, which is
answered by FreeSWITCH with "481 Call Does Not Exist".

I'm not exactly sure if the behaviour of the intitial REGISTER is
required(Send authorization-data after 401 has been received) or if
the proxy should understand a REGISTER including authorization data.
If the latter is true, is there a parameter in FreeSWITCH to allow
this behaviour?

I've attached traces of both communications, the initial successful
registration is "register_alcatel_success.dump", the failed
registration is the trace in "register_alcatel_fail.dump"

Best regards
Christian
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From benkokakao at gmail.com  Mon Mar 14 14:06:09 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Mon, 14 Mar 2011 12:06:09 +0100
Subject: [Freeswitch-users] Alcatel 4018 - "481 Call Does Not Exist" on
	re-REGISTER
In-Reply-To: <AANLkTinjt-rh_ieSMwGcWsJbL4RwfRz4SCTFmUht+Z9L@mail.gmail.com>
References: <AANLkTinjt-rh_ieSMwGcWsJbL4RwfRz4SCTFmUht+Z9L@mail.gmail.com>
Message-ID: <AANLkTi=gi15SR8TRsFuw-YuM0SPPY8LypkQPtpchJJp6@mail.gmail.com>

On 14 March 2011 11:59, Christian Benke <benkokakao at gmail.com> wrote:
> I'm not exactly sure if the behaviour of the intitial REGISTER is
> required(Send authorization-data after 401 has been received) or if
> the proxy should understand a REGISTER including authorization data.
> If the latter is true, is there a parameter in FreeSWITCH to allow
> this behaviour?

Sorry, i left out some information. The traces are from two different
sessions, i restarted the phone in between, so that's why the nonce
has changed. I'll have to verify if the phone is actually reusing the
initial nonce on the re-REGISTER in the same session.


From sascha.daniels at amooma.de  Mon Mar 14 14:55:00 2011
From: sascha.daniels at amooma.de (Sascha Daniels)
Date: Mon, 14 Mar 2011 12:55:00 +0100
Subject: [Freeswitch-users] Compile error on Suse 11.3
In-Reply-To: <4D762F7B.4060108@amooma.de>
References: <4D762F7B.4060108@amooma.de>
Message-ID: <4D7E0214.1090105@amooma.de>

Hi.

Some more information.

Problem exists only with http://files.freeswitch.org/freeswitch-1.0.6.tar.gz

No problems with git source

Greets

                Sascha

Am 08.03.2011 14:30, schrieb Sascha Daniels:
> Hi together.
>
> I did build FS on different systems without any problems.
>
> On Suse 11.3 (32bit) I get the following error:
>
> Compiling src/switch_core_sqldb.c ...
> cc1: warnings being treated as errors
> src/switch_core_sqldb.c: In function '_switch_cache_db_get_db_handle':
> src/switch_core_sqldb.c:314:47: error: comparison between
> 'switch_odbc_status_t' and 'enum <anonymous>'
> src/switch_core_sqldb.c: In function 'switch_cache_db_execute_sql_real':
> src/switch_core_sqldb.c:389:90: error: comparison between
> 'switch_status_t' and 'enum <anonymous>'
> make[1]: *** [libfreeswitch_la-switch_core_sqldb.lo] Fehler 1
> make: *** [all] Fehler 2
>
> Fehler = Error
>   

-- 
AMOOMA GmbH - Bachstr. 124 - 56566 Neuwied  -->  http://www.amooma.de
Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister Montabaur B14998

B?cher: http://das-asterisk-buch.de - http://ruby-auf-schienen.de



From sos at sokhapkin.dyndns.org  Mon Mar 14 14:50:42 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 14 Mar 2011 07:50:42 -0400
Subject: [Freeswitch-users] Memory leak
In-Reply-To: <AANLkTim3UsX4z_URe1Ya40uP4AnVG3s3NiYkew=ZY8+O@mail.gmail.com>
References: <AANLkTim3UsX4z_URe1Ya40uP4AnVG3s3NiYkew=ZY8+O@mail.gmail.com>
Message-ID: <201103140750.42072.sos@sokhapkin.dyndns.org>

Do you use luasql with mysql driver? That could be the cause of the leak.

On Monday 14 March 2011, Pete Kelly wrote:
> Hi
> 
> We are using Freeswitch (from git sources) in a production environment to
> handle an IVR system using a simple set of dialplans and lua scripts.
> OpenSIPS 1.6 and mysql are also installed and running as part of the same
> IVR application.
> 
> However over time (approx 1 week) the memory usage of Freeswitch increases,
> eventually creeping into swap space forcing us to restart Freeswitch. The
> IVR handles approx 40 concurrent calls constantly, and we have instances of
> the IVR running on a 32bit Etch and 64bit Lenny boxes - both exhibit the
> same issue.
> 
> We have also tried the Sangoma freeswitch branch in case that contained any
> fixes, however we see the same issues.
> 
> Can anybody advise if this is a known issue with Freeswitch? Is it a memory
> leak or is it possible to limit the amount of memory Freeswitch will use,
> forcing it to garbage collect when it reaches the limit?
> 
> Any advice on where to look/debug next would be appreciated.
> 
> Thanks
> 
> Pete



From acrow at integrafin.co.uk  Mon Mar 14 15:04:22 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Mon, 14 Mar 2011 12:04:22 +0000
Subject: [Freeswitch-users] Fax detection - ringing not heard by caller
In-Reply-To: <4D7CF0C3.3050104@integrafin.co.uk>
References: <4D7CF0C3.3050104@integrafin.co.uk>
Message-ID: <4D7E0446.90105@integrafin.co.uk>

Is anyone able to assist with this?

I've also tried it with an extension on the FS box instead of the 
Asterisk box - the caller does not hear ringing, just silence, even 
though the extension is ringing, and the call can be completed.

I've fiddled around for ages and I still can't get it to work. The 
Trixbox machine does it perfectly and I need to make the FS box behave 
the same, but it seems once "answer" has been executed I can only use 
"playback" to send ringing to the caller, but I can only do this while 
the tone detect is running. If it set playback to loop then the plan 
never proceeds to the bridge/transfer to the extension.

Thanks

Alex



> All,
>
> I am trying to implement fax detection on FreeSwitch within a dialplan
> that routes inbound calls from a BRI (Sangoma), to an Asterisk (Trixbox)
> machine handling the extensions. I am only doing this until I can get
> British English sounds for FS ;-).
>
> The inbound route without any fax detection works fine, ie the caller
> hears ringing until the Trixbox extension is either answered or hits
> voicemail:
>
> <extension name="60xto35x">
> <condition field="source" expression="freetdm"/>
> <condition field="context" expression="public"/>
> <condition field="destination_number" expression="60([0-5])">
> <action application="bridge" data="sofia/external/35$1 at 192.168.24.15"/>
> </condition>
> </extension>
>
>
> The closest I have got to adding the fax detection is the following:
>
> <extension name="60xto35x_copy">
> <condition field="source" expression="freetdm"/>
> <condition field="context" expression="public"/>
> <condition field="destination_number" expression="60([0-5])">
> <action application="tone_detect" data="fax 1100 r +5000 transfer '620
> XML default'"/>
> <action application="answer"/>
> <action application="playback" data="tone_stream://$${uk-ring};loops=3"/>
> <action application="bridge" data="sofia/external/35$1 at 192.168.24.15"/>
> </condition>
> </extension>
>
> I had to add the playback so the caller would hear ringing during the
> tone_detect timeout.
>
> However, as soon as the tone_stream finishes, and the call is bridged to
> the Trixbox, the caller hears silence instead of ringing. Is there any
> way to pass the ringing from the extension back to the caller coming in
> on the BRI circuit, or otherwise fake it until the remote extension is
> answered?
>
> Thanks
>
> Alex
>


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From peter.olsson at visionutveckling.se  Mon Mar 14 15:09:03 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 14 Mar 2011 13:09:03 +0100
Subject: [Freeswitch-users] Compile error on Suse 11.3
In-Reply-To: <4D7E0214.1090105@amooma.de>
References: <4D762F7B.4060108@amooma.de> <4D7E0214.1090105@amooma.de>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2F6C3F1@cooper>

1.0.6 is pretty old now, so it's not recommended anymore..

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Sascha Daniels
Skickat: den 14 mars 2011 12:55
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Compile error on Suse 11.3

Hi.

Some more information.

Problem exists only with http://files.freeswitch.org/freeswitch-1.0.6.tar.gz

No problems with git source

Greets

                Sascha

Am 08.03.2011 14:30, schrieb Sascha Daniels:
> Hi together.
>
> I did build FS on different systems without any problems.
>
> On Suse 11.3 (32bit) I get the following error:
>
> Compiling src/switch_core_sqldb.c ...
> cc1: warnings being treated as errors
> src/switch_core_sqldb.c: In function '_switch_cache_db_get_db_handle':
> src/switch_core_sqldb.c:314:47: error: comparison between
> 'switch_odbc_status_t' and 'enum <anonymous>'
> src/switch_core_sqldb.c: In function 'switch_cache_db_execute_sql_real':
> src/switch_core_sqldb.c:389:90: error: comparison between
> 'switch_status_t' and 'enum <anonymous>'
> make[1]: *** [libfreeswitch_la-switch_core_sqldb.lo] Fehler 1
> make: *** [all] Fehler 2
>
> Fehler = Error
>   

-- 
AMOOMA GmbH - Bachstr. 124 - 56566 Neuwied  -->  http://www.amooma.de
Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister Montabaur B14998

B?cher: http://das-asterisk-buch.de - http://ruby-auf-schienen.de


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

!DSPAM:4d7e01f332761879716305!



From fernando.berretta at gmail.com  Mon Mar 14 15:10:31 2011
From: fernando.berretta at gmail.com (Fernando Berretta)
Date: Mon, 14 Mar 2011 09:10:31 -0300
Subject: [Freeswitch-users] FreeSwitch for Windows
In-Reply-To: <134839.19366.qm@web33506.mail.mud.yahoo.com>
References: <134839.19366.qm@web33506.mail.mud.yahoo.com>
Message-ID: <4D7E05B7.9040403@gmail.com>

Diego,

Thanks for your answer

It is possible to use also boards like Digium, OpenVox or any other 
similar one ? What about E1 cards ? have you tried one of them over 
Windows ? MFC R2 ?

Best Regards,
Fernando

On 3/12/2011 11:35 AM, Diego Toro wrote:
> I don't think like Mr. Colling, I have FS on Windows in production 
> since two year using sangoma cards (FXO, FXS) and works fine.  Windows 
> is a good choice.
>
> Diego Toro
> http://voipensando.blogspot.com/
>
> --- On *Fri, 3/11/11, Fernando Berretta 
> /<fernando.berretta at gmail.com>/* wrote:
>
>
>     From: Fernando Berretta <fernando.berretta at gmail.com>
>     Subject: Re: [Freeswitch-users] FreeSwitch for Windows
>     To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>     Date: Friday, March 11, 2011, 4:54 PM
>
>     Michael,
>
>     Thanks for your valuable recommendation
>
>     Best Regards,
>     Fernando
>
>     On 3/11/2011 6:41 PM, Michael Collins wrote:
>>     If you plan on using TDM then I'd recommend you stay away from
>>     Windows. Sangoma and Digium/clone cards are all geared toward a
>>     Linux environment. Stay away from the exotic OSes and use
>>     something simple like CentOS 5.5 or Debian Lenny.
>>
>>     -MC
>>
>>     On Wed, Mar 9, 2011 at 1:38 PM, Fernando Berretta
>>     <fernando.berretta at gmail.com
>>     </mc/compose?to=fernando.berretta at gmail.com>> wrote:
>>
>>         Hi,
>>
>>         I'm newbie in FreeSwitch and would like to know if freeswitch
>>         could be
>>         used for production under Windows platforms ? telephony
>>         boards like
>>         Digium / OpenVox, Sangoma, etc are going to work ok under
>>         Windows ? is
>>         there any problem / limitation ? E1 MFCR2 Trunk works ?
>>
>>         Thanks for your time,
>>         Best Regards,
>>         Fernando
>>
>>         _______________________________________________
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         </mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  </mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>     -----Inline Attachment Follows-----
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     </mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-------------- next part --------------
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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110314/e9c99bc6/attachment-0001.html 

From fernando.berretta at gmail.com  Mon Mar 14 15:16:18 2011
From: fernando.berretta at gmail.com (Fernando Berretta)
Date: Mon, 14 Mar 2011 09:16:18 -0300
Subject: [Freeswitch-users] FreeSwitch for Windows
In-Reply-To: <81C2CEF80046FB4F863A60D4347DD33A0C56FB@server1.st.local>
References: <E04386CF-006E-4F87-A2B0-FA6C810A34B7@mgtech.com><022B01F7-0577-457B-94C8-D6F47B14E1F0@mgtech.com>	<4D77F344.3050207@gmail.com>
	<81C2CEF80046FB4F863A60D4347DD33A0C56FB@server1.st.local>
Message-ID: <4D7E0712.6030405@gmail.com>

Erkan,

Thanks for your advice, I'll take care of sounds folder path

Best Regards,
Fernando

On 3/14/2011 5:19 AM, Erkan ?nl? wrote:
> Dear Fernando,
>
> ?'m using FS on 4 different Server's since over 1 year successfully.
> My last server is a 64 bit platform and all things that i use worked fine for me.
>
> The only thing that i see is some file or folder path are not correctly and formated for Unix/Linux systems.
> This one must be changed manual for Windows path name.
>
> For example    sounds/xxx/.... and so on the file or path can not be found on Windows.
> You see that in the console of FS. But i changed it manual to sounds\xxx\....   and all things work fine.
> Also some modules that i need worked for me.
>
> But with boards or other hardware i can not tell you something, because i don't use it.
>
> Kind regards
> Erkan
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Fernando Berretta
> Sent: Wednesday, March 09, 2011 11:38 PM
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] FreeSwitch for Windows
>
> Hi,
>
> I'm newbie in FreeSwitch and would like to know if freeswitch could be used for production under Windows platforms ? telephony boards like Digium / OpenVox, Sangoma, etc are going to work ok under Windows ? is there any problem / limitation ? E1 MFCR2 Trunk works ?
>
> Thanks for your time,
> Best Regards,
> Fernando
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From benkokakao at gmail.com  Mon Mar 14 15:20:38 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Mon, 14 Mar 2011 13:20:38 +0100
Subject: [Freeswitch-users] Alcatel 4018 - "481 Call Does Not Exist" on
	re-REGISTER
In-Reply-To: <AANLkTi=gi15SR8TRsFuw-YuM0SPPY8LypkQPtpchJJp6@mail.gmail.com>
References: <AANLkTinjt-rh_ieSMwGcWsJbL4RwfRz4SCTFmUht+Z9L@mail.gmail.com>
	<AANLkTi=gi15SR8TRsFuw-YuM0SPPY8LypkQPtpchJJp6@mail.gmail.com>
Message-ID: <AANLkTimGzBf_skFLrg99CFGjJs9WDxUpYi4rSgqquDi6@mail.gmail.com>

> I'll have to verify if the phone is actually reusing the
> initial nonce on the re-REGISTER in the same session.

See the attached full trace(I've removed some superfluous and
identical OPTIONS and REGISTERs to make the file smaller), the phone
is in fact reusing the first nonce forever.
Is this a behaviour i could cover with FreeSWITCH?

Unfortunately there's no parameter in the phone's config i could
change(At least no documented parameter).

Best regards,
Christian
-------------- next part --------------
interface: eth1 (192.168.0.0/255.255.255.0)
filter: (ip) and ( host 192.168.0.10 and port 5060 )
#
U 192.168.0.10:5060 -> 192.168.0.1:5060
REGISTER sip:192.168.0.1:5060 SIP/2.0.
From: Jake Doe <sip:400 at 192.168.0.1>;tag=6306aa1707057ea.
To: Jake Doe <sip:400 at 192.168.0.1>.
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK539e84e32.
Call-ID: 0aefb7ccada0398f214fb6633ab58b34 at 192.168.0.10.
CSeq: 1329443678 REGISTER.
Max-Forwards: 70.
Content-Length: 0.
Contact: Jake Doe <sip:400 at 192.168.0.10:5060>;expires=3600.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, MESSAGE, INFO.
.

#
U 192.168.0.1:5060 -> 192.168.0.10:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK539e84e32.
From: Jake Doe <sip:400 at 192.168.0.1>;tag=6306aa1707057ea.
To: Jake Doe <sip:400 at 192.168.0.1>;tag=yypHNKarvU9rQ.
Call-ID: 0aefb7ccada0398f214fb6633ab58b34 at 192.168.0.10.
CSeq: 1329443678 REGISTER.
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-c18835d 2011-02-25 01-53-05 +0100.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
WWW-Authenticate: Digest realm="192.168.0.1", nonce="15a72d85-3c11-43b9-9016-fad60803eab3", algorithm=MD5, qop="auth".
Content-Length: 0.
.

#
U 192.168.0.10:5060 -> 192.168.0.1:5060
REGISTER sip:192.168.0.1:5060 SIP/2.0.
From: Jake Doe <sip:400 at 192.168.0.1>;tag=6306aa1707057ea.
To: Jake Doe <sip:400 at 192.168.0.1>.
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK0d96190c0.
Call-ID: 0aefb7ccada0398f214fb6633ab58b34 at 192.168.0.10.
CSeq: 1329443679 REGISTER.
Max-Forwards: 70.
Content-Length: 0.
Contact: Jake Doe <sip:400 at 192.168.0.10:5060>;expires=3600.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, MESSAGE, INFO.
Authorization:Digest response="818b7844616f93615544d3abd7474e29",nc=00000001,username="400",realm="192.168.0.1",nonce="15a72d85-3c11-43b9-9016-fad60803eab3",algorithm=MD5,qop=auth,cnonce="e3374561f9e3906e60b2dc5f41bee141",uri="sip:192.168.0.1:5060".
.

#
U 192.168.0.1:5060 -> 192.168.0.10:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK0d96190c0.
From: Jake Doe <sip:400 at 192.168.0.1>;tag=6306aa1707057ea.
To: Jake Doe <sip:400 at 192.168.0.1>;tag=Z7FaQeUUS4ZBK.
Call-ID: 0aefb7ccada0398f214fb6633ab58b34 at 192.168.0.10.
CSeq: 1329443679 REGISTER.
Contact: Jake Doe <sip:400 at 192.168.0.10:5060>;expires=3600.
Date: Mon, 14 Mar 2011 12:18:36 GMT.
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-c18835d 2011-02-25 01-53-05 +0100.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Content-Length: 0.
.

#
U 192.168.0.1:5060 -> 192.168.0.10:5060
NOTIFY sip:400 at 192.168.0.10:5060;fs_nat=yes;fs_path=sip:400%40192.168.0.10:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.1;rport;branch=z9hG4bKZU1D2yBN8N1gF.
Route: <sip:400 at 192.168.0.10:5060>.
Max-Forwards: 70.
From: <sip:400 at 192.168.0.1>;tag=0g92r9BZpDpye.
To: <sip:400 at 192.168.0.1>.
Call-ID: 078b76ca-c8d8-122e-c887-00105adc0460.
CSeq: 9716494 NOTIFY.
Contact: <sip:mod_sofia at 192.168.0.1:5060>.
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-c18835d 2011-02-25 01-53-05 +0100.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Event: message-summary.
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
Subscription-State: terminated;reason=timeout.
Content-Type: application/simple-message-summary.
Content-Length: 62.
.
Messages-Waiting: no.
Message-Account: sip:400 at 192.168.0.1.
.

#
U 192.168.0.10:5060 -> 192.168.0.1:5060
SIP/2.0 200 OK.
From: <sip:400 at 192.168.0.1>;tag=0g92r9BZpDpye.
To: <sip:400 at 192.168.0.1>;tag=d52a47311a4b5d5.
Via: SIP/2.0/UDP 192.168.0.1;received=192.168.0.1;branch=z9hG4bKZU1D2yBN8N1gF;rport=5060;rport.
Call-ID: 078b76ca-c8d8-122e-c887-00105adc0460.
CSeq: 9716494 NOTIFY.
Content-Length: 0.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, MESSAGE, INFO.
Supported: replaces.
Server: M5T SIP-UA SAFE/v3.6.9.13.
.

#
U 192.168.0.10:5060 -> 192.168.0.1:5060
SUBSCRIBE sip:400 at 192.168.0.1 SIP/2.0.
From: Jake Doe <sip:400 at 192.168.0.1>;tag=86fa6319ab01706.
To: sip:400 at 192.168.0.1.
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bKe62118e50.
Call-ID: 3b922ea031a98fec52c48f3c158d11a2 at 192.168.0.10.
CSeq: 1565832931 SUBSCRIBE.
Max-Forwards: 70.
Content-Length: 0.
Route: <sip:192.168.0.1:5060;lr>.
Supported: timer.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, MESSAGE, INFO.
Expires: 3600.
Event: message-summary.
Contact: Jake Doe <sip:400 at 192.168.0.10:5060>.
Supported: replaces.
.

#
U 192.168.0.1:5060 -> 192.168.0.10:5060
SIP/2.0 202 Accepted.
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bKe62118e50.
From: Jake Doe <sip:400 at 192.168.0.1>;tag=86fa6319ab01706.
To: <sip:400 at 192.168.0.1>;tag=1S2Ut4v2KpcHa.
Call-ID: 3b922ea031a98fec52c48f3c158d11a2 at 192.168.0.10.
CSeq: 1565832931 SUBSCRIBE.
Contact: <sip:400 at 192.168.0.1:5060>.
Expires: 3600.
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-c18835d 2011-02-25 01-53-05 +0100.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
Subscription-State: active;expires=3600.
Content-Length: 0.
.

#
U 192.168.0.1:5060 -> 192.168.0.10:5060
NOTIFY sip:400 at 192.168.0.10:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.1;rport;branch=z9hG4bK04t63Svr5yQ3a.
Max-Forwards: 70.
From: <sip:400 at 192.168.0.1>;tag=1S2Ut4v2KpcHa.
To: Jake Doe <sip:400 at 192.168.0.1>;tag=86fa6319ab01706.
Call-ID: 3b922ea031a98fec52c48f3c158d11a2 at 192.168.0.10.
CSeq: 9716494 NOTIFY.
Contact: <sip:400 at 192.168.0.1:5060>.
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-c18835d 2011-02-25 01-53-05 +0100.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Event: message-summary.
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
Subscription-State: active;expires=3600.
Content-Type: application/simple-message-summary.
Content-Length: 62.
.
Messages-Waiting: no.
Message-Account: sip:400 at 192.168.0.1.
.

#
U 192.168.0.10:5060 -> 192.168.0.1:5060
SIP/2.0 200 OK.
From: <sip:400 at 192.168.0.1>;tag=1S2Ut4v2KpcHa.
To: Jake Doe <sip:400 at 192.168.0.1>;tag=86fa6319ab01706.
Via: SIP/2.0/UDP 192.168.0.1;received=192.168.0.1;branch=z9hG4bK04t63Svr5yQ3a;rport=5060;rport.
Call-ID: 3b922ea031a98fec52c48f3c158d11a2 at 192.168.0.10.
CSeq: 9716494 NOTIFY.
Content-Length: 0.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, MESSAGE, INFO.
Supported: replaces.
Server: M5T SIP-UA SAFE/v3.6.9.13.
.

#
U 192.168.0.10:5060 -> 192.168.0.1:5060
OPTIONS sip:192.168.0.1:5060 SIP/2.0.
From: Jake Doe <sip:400 at 192.168.0.1>;tag=6306aa1707057ea.
To: Jake Doe <sip:400 at 192.168.0.1>.
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK6d342e095.
Call-ID: 0aefb7ccada0398f214fb6633ab58b34 at 192.168.0.10.
CSeq: 1329443680 OPTIONS.
Max-Forwards: 70.
Content-Length: 0.
Accept: Application/sdp.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, MESSAGE, INFO.
Contact: Jake Doe <sip:400 at 192.168.0.10:5060>.
Authorization:Digest response="bc8dd6c344448e75060249710090cc6c",nc=00000002,username="400",realm="192.168.0.1",nonce="15a72d85-3c11-43b9-9016-fad60803eab3",algorithm=MD5,qop=auth,cnonce="e3374561f9e3906e60b2dc5f41bee141",uri="sip:192.168.0.1:5060".
.

#
U 192.168.0.1:5060 -> 192.168.0.10:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK6d342e095.
From: Jake Doe <sip:400 at 192.168.0.1>;tag=6306aa1707057ea.
To: Jake Doe <sip:400 at 192.168.0.1>;tag=22UmvZD6gZ23N.
Call-ID: 0aefb7ccada0398f214fb6633ab58b34 at 192.168.0.10.
CSeq: 1329443680 OPTIONS.
Contact: <sip:192.168.0.1>.
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-c18835d 2011-02-25 01-53-05 +0100.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
Content-Length: 0.
.

#
U 192.168.0.10:5060 -> 192.168.0.1:5060
OPTIONS sip:192.168.0.1 SIP/2.0.
From: Jake Doe <sip:400 at 192.168.0.1>;tag=6306aa1707057ea.
To: Jake Doe <sip:400 at 192.168.0.1>;tag=22UmvZD6gZ23N.
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bKf800a73ab.
Call-ID: 0aefb7ccada0398f214fb6633ab58b34 at 192.168.0.10.
CSeq: 1329443681 OPTIONS.
Max-Forwards: 70.
Content-Length: 0.
Accept: Application/sdp.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, MESSAGE, INFO.
Contact: Jake Doe <sip:400 at 192.168.0.10:5060>.
Authorization:Digest response="d67a226a40df8dc1c975194259d49e15",nc=00000003,username="400",realm="192.168.0.1",nonce="15a72d85-3c11-43b9-9016-fad60803eab3",algorithm=MD5,qop=auth,cnonce="e3374561f9e3906e60b2dc5f41bee141",uri="sip:192.168.0.1".
.

#
U 192.168.0.1:5060 -> 192.168.0.10:5060
SIP/2.0 481 Call Does Not Exist.
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bKf800a73ab.
From: Jake Doe <sip:400 at 192.168.0.1>;tag=6306aa1707057ea.
To: Jake Doe <sip:400 at 192.168.0.1>;tag=22UmvZD6gZ23N.
Call-ID: 0aefb7ccada0398f214fb6633ab58b34 at 192.168.0.10.
CSeq: 1329443681 OPTIONS.
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-c18835d 2011-02-25 01-53-05 +0100.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Content-Length: 0.
.

#
U 192.168.0.1:5060 -> 192.168.0.10:5060
NOTIFY sip:400 at 192.168.0.10:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.1;rport;branch=z9hG4bK1DmZ5mDv27Dpp.
Max-Forwards: 70.
From: <sip:400 at 192.168.0.1>;tag=yv4y3FmpN1Qcj.
To: Jake Doe <sip:400 at 192.168.0.1>;tag=52a569d0614b8e6.
Call-ID: 77871c9a2c7603027048b3b606932911 at 192.168.0.10.
CSeq: 9713101 NOTIFY.
Contact: <sip:400 at 192.168.0.1:5060>.
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-c18835d 2011-02-25 01-53-05 +0100.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Event: message-summary.
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
Subscription-State: terminated;reason=timeout.
Content-Type: application/simple-message-summary.
Content-Length: 62.
.
Messages-Waiting: no.
Message-Account: sip:400 at 192.168.0.1.
.

#
U 192.168.0.10:5060 -> 192.168.0.1:5060
SIP/2.0 200 OK.
From: <sip:400 at 192.168.0.1>;tag=yv4y3FmpN1Qcj.
To: Jake Doe <sip:400 at 192.168.0.1>;tag=52a569d0614b8e6.
Via: SIP/2.0/UDP 192.168.0.1;received=192.168.0.1;branch=z9hG4bK1DmZ5mDv27Dpp;rport=5060;rport.
Call-ID: 77871c9a2c7603027048b3b606932911 at 192.168.0.10.
CSeq: 9713101 NOTIFY.
Content-Length: 0.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, MESSAGE, INFO.
Supported: replaces.
Server: M5T SIP-UA SAFE/v3.6.9.13.
.

#
U 192.168.0.10:5060 -> 192.168.0.1:5060
OPTIONS sip:192.168.0.1 SIP/2.0.
From: Jake Doe <sip:400 at 192.168.0.1>;tag=6306aa1707057ea.
To: Jake Doe <sip:400 at 192.168.0.1>;tag=22UmvZD6gZ23N.
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK72332f87f.
Call-ID: 0aefb7ccada0398f214fb6633ab58b34 at 192.168.0.10.
CSeq: 1329443695 OPTIONS.
Max-Forwards: 70.
Content-Length: 0.
Accept: Application/sdp.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, MESSAGE, INFO.
Contact: Jake Doe <sip:400 at 192.168.0.10:5060>.
Authorization:Digest response="0f87ba7361f4e8a5b1bc01fb623fc5d4",nc=00000011,username="400",realm="192.168.0.1",nonce="15a72d85-3c11-43b9-9016-fad60803eab3",algorithm=MD5,qop=auth,cnonce="e3374561f9e3906e60b2dc5f41bee141",uri="sip:192.168.0.1".
.

#
U 192.168.0.1:5060 -> 192.168.0.10:5060
SIP/2.0 481 Call Does Not Exist.
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK72332f87f.
From: Jake Doe <sip:400 at 192.168.0.1>;tag=6306aa1707057ea.
To: Jake Doe <sip:400 at 192.168.0.1>;tag=22UmvZD6gZ23N.
Call-ID: 0aefb7ccada0398f214fb6633ab58b34 at 192.168.0.10.
CSeq: 1329443695 OPTIONS.
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-c18835d 2011-02-25 01-53-05 +0100.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Content-Length: 0.
.

#
U 192.168.0.10:5060 -> 192.168.0.1:5060
REGISTER sip:192.168.0.1 SIP/2.0.
From: Jake Doe <sip:400 at 192.168.0.1>;tag=6306aa1707057ea.
To: Jake Doe <sip:400 at 192.168.0.1>;tag=22UmvZD6gZ23N.
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK75e801328.
Call-ID: 0aefb7ccada0398f214fb6633ab58b34 at 192.168.0.10.
CSeq: 1329443707 REGISTER.
Max-Forwards: 70.
Content-Length: 0.
Contact: Jake Doe <sip:400 at 192.168.0.10:5060>;expires=3600.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, MESSAGE, INFO.
Authorization:Digest response="72a25afb32797b5c3b894eb9a45f8ced",nc=0000001d,username="400",realm="192.168.0.1",nonce="15a72d85-3c11-43b9-9016-fad60803eab3",algorithm=MD5,qop=auth,cnonce="e3374561f9e3906e60b2dc5f41bee141",uri="sip:192.168.0.1".
.

#
U 192.168.0.1:5060 -> 192.168.0.10:5060
SIP/2.0 481 Call Does Not Exist.
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK75e801328.
From: Jake Doe <sip:400 at 192.168.0.1>;tag=6306aa1707057ea.
To: Jake Doe <sip:400 at 192.168.0.1>;tag=22UmvZD6gZ23N.
Call-ID: 0aefb7ccada0398f214fb6633ab58b34 at 192.168.0.10.
CSeq: 1329443707 REGISTER.
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-c18835d 2011-02-25 01-53-05 +0100.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Content-Length: 0.
.

From peter.olsson at visionutveckling.se  Mon Mar 14 15:22:11 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 14 Mar 2011 13:22:11 +0100
Subject: [Freeswitch-users] FreeSwitch for Windows
In-Reply-To: <4D7E05B7.9040403@gmail.com>
References: <134839.19366.qm@web33506.mail.mud.yahoo.com>
	<4D7E05B7.9040403@gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2F6C40A@cooper>

Sangoma's PRI (E1) cards are supported under Windows. However, I've had some problems with the sangoma_isdn module for some time, so I havn't used it in about 6 months (it was only in a lab anyway).

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Fernando Berretta
Skickat: den 14 mars 2011 13:11
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] FreeSwitch for Windows

Diego,

Thanks for your answer

It is possible to use also boards like Digium, OpenVox or any other similar one ? What about E1 cards ? have you tried one of them over Windows ? MFC R2 ?

Best Regards,
Fernando

On 3/12/2011 11:35 AM, Diego Toro wrote:
I don't think like Mr. Colling, I have FS on Windows in production since two year using sangoma cards (FXO, FXS) and works fine.  Windows is a good choice.

Diego Toro
http://voipensando.blogspot.com/

--- On Fri, 3/11/11, Fernando Berretta <fernando.berretta at gmail.com><mailto:fernando.berretta at gmail.com> wrote:


From: Fernando Berretta <fernando.berretta at gmail.com><mailto:fernando.berretta at gmail.com>
Subject: Re: [Freeswitch-users] FreeSwitch for Windows
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org><mailto:freeswitch-users at lists.freeswitch.org>
Date: Friday, March 11, 2011, 4:54 PM
Michael,

Thanks for your valuable recommendation

Best Regards,
Fernando

On 3/11/2011 6:41 PM, Michael Collins wrote:
If you plan on using TDM then I'd recommend you stay away from Windows. Sangoma and Digium/clone cards are all geared toward a Linux environment. Stay away from the exotic OSes and use something simple like CentOS 5.5 or Debian Lenny.

-MC
On Wed, Mar 9, 2011 at 1:38 PM, Fernando Berretta <fernando.berretta at gmail.com</mc/compose?to=fernando.berretta at gmail.com>> wrote:
Hi,

I'm newbie in FreeSwitch and would like to know if freeswitch could be
used for production under Windows platforms ? telephony boards like
Digium / OpenVox, Sangoma, etc are going to work ok under Windows ? is
there any problem / limitation ? E1 MFCR2 Trunk works ?

Thanks for your time,
Best Regards,
Fernando

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org</mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org</mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

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From Nabble at slickdeals.endjunk.com  Mon Mar 14 15:30:17 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 14 Mar 2011 05:30:17 -0700 (PDT)
Subject: [Freeswitch-users] Freeswitch / Zywall USG100 problems
In-Reply-To: <7560DFC1-76DD-466C-9D50-33C3A521B6D5@mgtech.com>
References: <E04386CF-006E-4F87-A2B0-FA6C810A34B7@mgtech.com>
	<022B01F7-0577-457B-94C8-D6F47B14E1F0@mgtech.com>
	<7560DFC1-76DD-466C-9D50-33C3A521B6D5@mgtech.com>
Message-ID: <1300105817087-6168856.post@n2.nabble.com>


Mario G wrote:
> Was this thread hijacked?
Mario, I just can't believe the mods here let your thread get hijacked! 

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-Zywall-USG100-problems-tp6155297p6168856.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Mon Mar 14 15:38:42 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 14 Mar 2011 05:38:42 -0700 (PDT)
Subject: [Freeswitch-users] strange errors when using mod_python
In-Reply-To: <1300093911185-6168304.post@n2.nabble.com>
References: <1299848563170-6161361.post@n2.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2F6C044@cooper>
	<1299850368472-6161441.post@n2.nabble.com>
	<AANLkTinLJLU3+fc=L3EwAAJe_U-kpTi7HvB40wUu78zD@mail.gmail.com>
	<1300093688368-6168292.post@n2.nabble.com>
	<1300093911185-6168304.post@n2.nabble.com>
Message-ID: <1300106322794-6168878.post@n2.nabble.com>


hkalyoncu wrote:
> 
extension name="generated"

I don't know if the same generated extension name is an issue.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/strange-errors-when-using-mod-python-tp6161361p6168878.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From pkelly at gmail.com  Mon Mar 14 15:46:39 2011
From: pkelly at gmail.com (Pete Kelly)
Date: Mon, 14 Mar 2011 12:46:39 +0000
Subject: [Freeswitch-users] Memory leak
In-Reply-To: <201103140750.42072.sos@sokhapkin.dyndns.org>
References: <AANLkTim3UsX4z_URe1Ya40uP4AnVG3s3NiYkew=ZY8+O@mail.gmail.com>
	<201103140750.42072.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTimSgCLdC6ZOJBNP6-q3HQ44Ra1eTS7iq5mARwzf@mail.gmail.com>

Yes we do - what prompts you to suggest that could be the source of the
leak?

On 14 March 2011 11:50, Sergey Okhapkin <sos at sokhapkin.dyndns.org> wrote:

> Do you use luasql with mysql driver? That could be the cause of the leak.
>
> On Monday 14 March 2011, Pete Kelly wrote:
> > Hi
> >
> > We are using Freeswitch (from git sources) in a production environment to
> > handle an IVR system using a simple set of dialplans and lua scripts.
> > OpenSIPS 1.6 and mysql are also installed and running as part of the same
> > IVR application.
> >
> > However over time (approx 1 week) the memory usage of Freeswitch
> increases,
> > eventually creeping into swap space forcing us to restart Freeswitch. The
> > IVR handles approx 40 concurrent calls constantly, and we have instances
> of
> > the IVR running on a 32bit Etch and 64bit Lenny boxes - both exhibit the
> > same issue.
> >
> > We have also tried the Sangoma freeswitch branch in case that contained
> any
> > fixes, however we see the same issues.
> >
> > Can anybody advise if this is a known issue with Freeswitch? Is it a
> memory
> > leak or is it possible to limit the amount of memory Freeswitch will use,
> > forcing it to garbage collect when it reaches the limit?
> >
> > Any advice on where to look/debug next would be appreciated.
> >
> > Thanks
> >
> > Pete
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sos at sokhapkin.dyndns.org  Mon Mar 14 15:52:56 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 14 Mar 2011 08:52:56 -0400
Subject: [Freeswitch-users] Memory leak
In-Reply-To: <AANLkTimSgCLdC6ZOJBNP6-q3HQ44Ra1eTS7iq5mARwzf@mail.gmail.com>
References: <AANLkTim3UsX4z_URe1Ya40uP4AnVG3s3NiYkew=ZY8+O@mail.gmail.com>
	<201103140750.42072.sos@sokhapkin.dyndns.org>
	<AANLkTimSgCLdC6ZOJBNP6-q3HQ44Ra1eTS7iq5mARwzf@mail.gmail.com>
Message-ID: <201103140852.56746.sos@sokhapkin.dyndns.org>

Use odbc driver instead of mysql driver to avoid the leak. 

On Monday 14 March 2011, Pete Kelly wrote:
> Yes we do - what prompts you to suggest that could be the source of the
> leak?
> 
> On 14 March 2011 11:50, Sergey Okhapkin <sos at sokhapkin.dyndns.org> wrote:
> > Do you use luasql with mysql driver? That could be the cause of the leak.
> > 
> > On Monday 14 March 2011, Pete Kelly wrote:
> > > Hi
> > > 
> > > We are using Freeswitch (from git sources) in a production environment
> > > to handle an IVR system using a simple set of dialplans and lua
> > > scripts. OpenSIPS 1.6 and mysql are also installed and running as part
> > > of the same IVR application.
> > > 
> > > However over time (approx 1 week) the memory usage of Freeswitch
> > 
> > increases,
> > 
> > > eventually creeping into swap space forcing us to restart Freeswitch.
> > > The IVR handles approx 40 concurrent calls constantly, and we have
> > > instances
> > 
> > of
> > 
> > > the IVR running on a 32bit Etch and 64bit Lenny boxes - both exhibit
> > > the same issue.
> > > 
> > > We have also tried the Sangoma freeswitch branch in case that contained
> > 
> > any
> > 
> > > fixes, however we see the same issues.
> > > 
> > > Can anybody advise if this is a known issue with Freeswitch? Is it a
> > 
> > memory
> > 
> > > leak or is it possible to limit the amount of memory Freeswitch will
> > > use, forcing it to garbage collect when it reaches the limit?
> > > 
> > > Any advice on where to look/debug next would be appreciated.
> > > 
> > > Thanks
> > > 
> > > Pete
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org



From steveayre at gmail.com  Mon Mar 14 16:16:46 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 14 Mar 2011 13:16:46 +0000
Subject: [Freeswitch-users] Memory leak
In-Reply-To: <AANLkTim3UsX4z_URe1Ya40uP4AnVG3s3NiYkew=ZY8+O@mail.gmail.com>
References: <AANLkTim3UsX4z_URe1Ya40uP4AnVG3s3NiYkew=ZY8+O@mail.gmail.com>
Message-ID: <AANLkTincGerDeJtkyqzYXFmtTzjHe-h9RMZJsM5HosCS@mail.gmail.com>

If you use mod_lcr, we were getting a memory leak in that a few weeks ago
which is fixed in the latest git.

-Steve


On 14 March 2011 09:30, Pete Kelly <pkelly at gmail.com> wrote:

> Hi
>
> We are using Freeswitch (from git sources) in a production environment to
> handle an IVR system using a simple set of dialplans and lua scripts.
> OpenSIPS 1.6 and mysql are also installed and running as part of the same
> IVR application.
>
> However over time (approx 1 week) the memory usage of Freeswitch increases,
> eventually creeping into swap space forcing us to restart Freeswitch. The
> IVR handles approx 40 concurrent calls constantly, and we have instances of
> the IVR running on a 32bit Etch and 64bit Lenny boxes - both exhibit the
> same issue.
>
> We have also tried the Sangoma freeswitch branch in case that contained any
> fixes, however we see the same issues.
>
> Can anybody advise if this is a known issue with Freeswitch? Is it a memory
> leak or is it possible to limit the amount of memory Freeswitch will use,
> forcing it to garbage collect when it reaches the limit?
>
> Any advice on where to look/debug next would be appreciated.
>
> Thanks
>
> Pete
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From michelhabib at gmail.com  Mon Mar 14 16:53:25 2011
From: michelhabib at gmail.com (Michel Habib)
Date: Mon, 14 Mar 2011 15:53:25 +0200
Subject: [Freeswitch-users] ASR from Freeswitch to MS Speech Server
 [using MRCP Connector] - Audio Problem
Message-ID: <AANLkTin-_XPH4=K4XF=j0zm4ORXrw4NFVp4Lf=rk7DCw@mail.gmail.com>

Sorry for the delay and thanks Chris,
Please find attached logs: http://www.mediafire.com/?8thg9ey7kd26882
sip client ip 192.168.1.202
freeswitch ip 192.168.1.35
mrcp connector/speech server ip 192.168.1.32
i have attached 3 logs - freeswitch log file, wireshark on freeswitch
server, wireshark on mrcp connector/speech server.
my script simply plays back audio from wave [basic freeswitch function],
then plays back audio from TTS [which cannot be heard]

Thank you,
Michel.

---------- Forwarded message ----------
> From: Christopher Rienzo <cmrienzo at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Wed, 9 Mar 2011 22:25:34 -0500
> Subject: Re: [Freeswitch-users] ASR from Freeswitch to MS Speech Server
> [using MRCP Connector] - Audio Problem
> I still would like to see the logs for your call.
>
>
> On Tue, Mar 8, 2011 at 7:18 AM, Michel Habib <michelhabib at gmail.com>wrote:
>
>> Yes, I get the Audio from FS in regular calls - I already disabled all
>> possible firewalls - all 3 machines [softphone, freeswitch, Speech Server
>> (and mrcp connector) ] are on a switch.
>> 192.168.5.107 is the freeswitch server
>> 192.168.5.110 is the MRCP-Connector/Speech Server 2007 server
>> I made too many iterations on the configuration below:
>>
>> <include>
>>   <profile name="mrcp-connector" version="2">
>>     <param name="client-ip" value="192.168.5.107"/>
>>     <param name="client-port" value="5090"/>
>>     <param name="server-ip" value="192.168.5.110"/>
>>     <param name="server-port" value="5070"/>
>>     <!--param name="force-destination" value="1"/-->
>>     <param name="sip-transport" value="udp"/>
>>     <!--param name="ua-name" value="FreeSWITCH"/-->
>>     <!--param name="sdp-origin" value="FreeSWITCH"/-->
>>     <param name="rtp-ext-ip" value="192.168.5.107"/>
>>     <param name="rtp-ip" value="192.168.5.107"/>
>>     <param name="rtp-port-min" value="4000"/>
>>     <param name="rtp-port-max" value="5000"/>
>>     <!-- enable/disable rtcp support -->
>>     <param name="rtcp" value="1"/>
>>     <!-- rtcp bye policies (rtcp must be enabled first)
>>              0 - disable rtcp bye
>>              1 - send rtcp bye at the end of session
>>              2 - send rtcp bye also at the end of each talkspurt (input)
>>     -->
>>     <param name="rtcp-bye" value="2"/>
>>     <!-- rtcp transmission interval in msec (set 0 to disable) -->
>>     <param name="rtcp-tx-interval" value="5000"/>
>>     <!-- period (timeout) to check for new rtcp messages in msec (set 0 to
>> disable) -->
>>     <param name="rtcp-rx-resolution" value="1000"/>
>>
>>     <!--param name="playout-delay" value="50"/-->
>>     <!--param name="max-playout-delay" value="200"/-->
>>     <!--param name="ptime" value="20"/-->
>>     <param name="codecs" value="PCMU PCMA L16/96/8000"/>
>>
>>     <!-- Add any default MRCP params for SPEAK requests here -->
>>     <synthparams>
>>     </synthparams>
>>
>>     <!-- Add any default MRCP params for RECOGNIZE requests here -->
>>     <recogparams>
>>       <!--param name="start-input-timers" value="false"/-->
>>     </recogparams>
>>   </profile>
>> </include>
>>
>>
>>
>> ---------- Forwarded message ----------
>>
>>> From: Christopher Rienzo <cmrienzo at gmail.com>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Date: Mon, 7 Mar 2011 09:32:51 -0500
>>> Subject: Re: [Freeswitch-users] ASR from Freeswitch to MS Speech Server
>>> [using MRCP Connector] - Audio Problem
>>> Do you get audio between FS and your SIP client when not using ASR/TTS?
>>>
>>> Show me the MRCP profile configuration and your FreeSWITCH logs during
>>> the call.
>>>
>>>
>>>
>>> On Mon, Mar 7, 2011 at 4:55 AM, Michel Habib <michelhabib at gmail.com>wrote:
>>>
>>>> Hello All,
>>>> I have MS OCS Speech Server 2007 [working correctly, as i can make SIP
>>>> calls and use its ASR and TTS Services successfully]
>>>> I am also using MRCP Connector from AumTech - which allows me to use ASR
>>>> and TTS Services through an MRCP Client .
>>>> Now, i am using Freeswitch mod unimrcp to use ASR and TTS.
>>>>
>>>> for TTS, I can successfully make the call, the Audio RTP of the TTS
>>>> voice is transferred succesfully from Speech Server [through MRCP Connector]
>>>> back to the Freeswitch Server.
>>>> However, Freeswitch is not sending back the Audio RTP to the SIP client.
>>>>
>>>> for ASR, I can successfully define the grammar and start recognition,
>>>> but the audio RTP sent to speech server [through MRCP Connector] is silent
>>>> [empty].
>>>>
>>>> I am suspecting something is wrong with the RTP Configuration - can you
>>>> help me?
>>>>
>>>> Let me now if you need any specific logs/scripts/configuration?
>>>>
>>>> Thank you,
>>>> Michel.
>>>>
>>>
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From dftoro at yahoo.com  Mon Mar 14 17:08:43 2011
From: dftoro at yahoo.com (Diego Toro)
Date: Mon, 14 Mar 2011 07:08:43 -0700 (PDT)
Subject: [Freeswitch-users] FreeSwitch for Windows
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B2F6C40A@cooper>
Message-ID: <614202.23687.qm@web33503.mail.mud.yahoo.com>

Hi Fernando, I have not used other cards, I have used sangoma cards with excelent support from sangoma's staff.

Diego Toro

http://voipensando.blogspot.com/

--- On Mon, 3/14/11, Peter Olsson <peter.olsson at visionutveckling.se> wrote:

From: Peter Olsson <peter.olsson at visionutveckling.se>
Subject: Re: [Freeswitch-users] FreeSwitch for Windows
To: "'FreeSWITCH Users Help'" <freeswitch-users at lists.freeswitch.org>
Date: Monday, March 14, 2011, 7:22 AM

Sangoma?s PRI (E1) cards are supported under Windows. However, I?ve had some problems with the sangoma_isdn module for some time, so I havn?t used it in about 6 months (it was only in a lab anyway). ?/Peter ?Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Fernando Berretta
Skickat: den 14 mars 2011 13:11
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] FreeSwitch for Windows ?Diego,

Thanks for your answer

It is possible to use also boards like Digium, OpenVox or any other similar one ? What about E1 cards ? have you tried one of them over Windows ? MFC R2 ?

Best Regards,
Fernando

On 3/12/2011 11:35 AM, Diego Toro wrote: I don't think like Mr. Colling, I have FS on Windows in production since two year using sangoma cards (FXO, FXS) and works fine.? Windows is a good choice.

Diego Toro
http://voipensando.blogspot.com/

--- On Fri, 3/11/11, Fernando Berretta <fernando.berretta at gmail.com> wrote:


From: Fernando Berretta <fernando.berretta at gmail.com>
Subject: Re: [Freeswitch-users] FreeSwitch for Windows
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Friday, March 11, 2011, 4:54 PMMichael,

Thanks for your valuable recommendation

Best Regards,
Fernando

On 3/11/2011 6:41 PM, Michael Collins wrote: If you plan on using TDM then I'd recommend you stay away from Windows. Sangoma and Digium/clone cards are all geared toward a Linux environment. Stay away from the exotic OSes and use something simple like CentOS 5.5 or Debian Lenny.?  ?-MCOn Wed, Mar 9, 2011 at 1:38 PM, Fernando Berretta <fernando.berretta at gmail.com> wrote:Hi,

I'm newbie in FreeSwitch and would like to know if freeswitch could be
used for production under Windows platforms ? telephony boards like
Digium / OpenVox, Sangoma, etc are going to work ok under Windows ? is
there any problem / limitation ? E1 MFCR2 Trunk works ?

Thanks for your time,
Best Regards,
Fernando

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From dftoro at yahoo.com  Mon Mar 14 17:08:54 2011
From: dftoro at yahoo.com (Diego Toro)
Date: Mon, 14 Mar 2011 07:08:54 -0700 (PDT)
Subject: [Freeswitch-users] FreeSwitch for Windows
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B2F6C40A@cooper>
Message-ID: <799432.84909.qm@web33503.mail.mud.yahoo.com>

Hi Fernando, I have not used other cards, I have used sangoma cards with excelent support from sangoma's staff.

Diego Toro

http://voipensando.blogspot.com/

--- On Mon, 3/14/11, Peter Olsson <peter.olsson at visionutveckling.se> wrote:

From: Peter Olsson <peter.olsson at visionutveckling.se>
Subject: Re: [Freeswitch-users] FreeSwitch for Windows
To: "'FreeSWITCH Users Help'" <freeswitch-users at lists.freeswitch.org>
Date: Monday, March 14, 2011, 7:22 AM

Sangoma?s PRI (E1) cards are supported under Windows. However, I?ve had some problems with the sangoma_isdn module for some time, so I havn?t used it in about 6 months (it was only in a lab anyway). ?/Peter ?Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Fernando Berretta
Skickat: den 14 mars 2011 13:11
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] FreeSwitch for Windows ?Diego,

Thanks for your answer

It is possible to use also boards like Digium, OpenVox or any other similar one ? What about E1 cards ? have you tried one of them over Windows ? MFC R2 ?

Best Regards,
Fernando

On 3/12/2011 11:35 AM, Diego Toro wrote: I don't think like Mr. Colling, I have FS on Windows in production since two year using sangoma cards (FXO, FXS) and works fine.? Windows is a good choice.

Diego Toro
http://voipensando.blogspot.com/

--- On Fri, 3/11/11, Fernando Berretta <fernando.berretta at gmail.com> wrote:


From: Fernando Berretta <fernando.berretta at gmail.com>
Subject: Re: [Freeswitch-users] FreeSwitch for Windows
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Friday, March 11, 2011, 4:54 PMMichael,

Thanks for your valuable recommendation

Best Regards,
Fernando

On 3/11/2011 6:41 PM, Michael Collins wrote: If you plan on using TDM then I'd recommend you stay away from Windows. Sangoma and Digium/clone cards are all geared toward a Linux environment. Stay away from the exotic OSes and use something simple like CentOS 5.5 or Debian Lenny.?  ?-MCOn Wed, Mar 9, 2011 at 1:38 PM, Fernando Berretta <fernando.berretta at gmail.com> wrote:Hi,

I'm newbie in FreeSwitch and would like to know if freeswitch could be
used for production under Windows platforms ? telephony boards like
Digium / OpenVox, Sangoma, etc are going to work ok under Windows ? is
there any problem / limitation ? E1 MFCR2 Trunk works ?

Thanks for your time,
Best Regards,
Fernando

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From pkelly at gmail.com  Mon Mar 14 17:28:39 2011
From: pkelly at gmail.com (Pete Kelly)
Date: Mon, 14 Mar 2011 14:28:39 +0000
Subject: [Freeswitch-users] Memory leak
In-Reply-To: <AANLkTincGerDeJtkyqzYXFmtTzjHe-h9RMZJsM5HosCS@mail.gmail.com>
References: <AANLkTim3UsX4z_URe1Ya40uP4AnVG3s3NiYkew=ZY8+O@mail.gmail.com>
	<AANLkTincGerDeJtkyqzYXFmtTzjHe-h9RMZJsM5HosCS@mail.gmail.com>
Message-ID: <AANLkTimLNa73o5Y_h6KUpTptNyX6Q-6bpBOQBjLtsnS6@mail.gmail.com>

On 14 March 2011 13:16, Steven Ayre <steveayre at gmail.com> wrote:

> If you use mod_lcr, we were getting a memory leak in that a few weeks ago
> which is fixed in the latest git.


No we are not. I am going to try strengthening up the code which closes the
DB connections. If that fails then I will compile and try the ODBC driver.


>
> -Steve
>
>
> On 14 March 2011 09:30, Pete Kelly <pkelly at gmail.com> wrote:
>
>> Hi
>>
>> We are using Freeswitch (from git sources) in a production environment to
>> handle an IVR system using a simple set of dialplans and lua scripts.
>> OpenSIPS 1.6 and mysql are also installed and running as part of the same
>> IVR application.
>>
>> However over time (approx 1 week) the memory usage of Freeswitch
>> increases, eventually creeping into swap space forcing us to restart
>> Freeswitch. The IVR handles approx 40 concurrent calls constantly, and we
>> have instances of the IVR running on a 32bit Etch and 64bit Lenny boxes -
>> both exhibit the same issue.
>>
>> We have also tried the Sangoma freeswitch branch in case that contained
>> any fixes, however we see the same issues.
>>
>> Can anybody advise if this is a known issue with Freeswitch? Is it a
>> memory leak or is it possible to limit the amount of memory Freeswitch will
>> use, forcing it to garbage collect when it reaches the limit?
>>
>> Any advice on where to look/debug next would be appreciated.
>>
>> Thanks
>>
>> Pete
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From t.mahe at telemaque.fr  Mon Mar 14 17:39:55 2011
From: t.mahe at telemaque.fr (=?ISO-8859-1?Q?Tristan_Mah=E9?=)
Date: Mon, 14 Mar 2011 15:39:55 +0100
Subject: [Freeswitch-users] Memory leak
In-Reply-To: <AANLkTimLNa73o5Y_h6KUpTptNyX6Q-6bpBOQBjLtsnS6@mail.gmail.com>
References: <AANLkTim3UsX4z_URe1Ya40uP4AnVG3s3NiYkew=ZY8+O@mail.gmail.com>	<AANLkTincGerDeJtkyqzYXFmtTzjHe-h9RMZJsM5HosCS@mail.gmail.com>
	<AANLkTimLNa73o5Y_h6KUpTptNyX6Q-6bpBOQBjLtsnS6@mail.gmail.com>
Message-ID: <4D7E28BB.8020408@telemaque.fr>

Hi,

I've never seen memleaks on luasql using the following code:

Feel free to use it and modify/improve ;)

----------------------------------------------- mysql.lua 
-----------------------------------------------

require 'luasql.mysql'

DB = {}
DB.__index = DB

function DB.create()
         local db = {}
         setmetatable(db,DB)
         db.env = assert( luasql.mysql() )
         db.con = assert( db.env:connect('DB','USER','PASSWORD','HOST') )
         return db
end

function DB:query(query)
         local q,r = self.con:execute(query)
         if q == nil then
                 return nil
         else
                 return q
         end
end

function DB:fetch(cur)
         if (cur ~= nil) then
                 return cur:fetch({},"a");
         else
                 return nil;
         end
end

function DB:close()
         self.con:close()
         self.env:close()
end

----------------------------------------------- /mysql.lua 
-----------------------------------------------

Exemple usage ( in a global do ... end block to ensure no data is global 
and everything is destroyed ):

local db = DB.create();
local r = db:query("SELECT * FROM test");
local f = db:fetch(r);
db:close();


Le 14/03/2011 15:28, Pete Kelly a ?crit :
>
>
> On 14 March 2011 13:16, Steven Ayre <steveayre at gmail.com 
> <mailto:steveayre at gmail.com>> wrote:
>
>     If you use mod_lcr, we were getting a memory leak in that a few
>     weeks ago which is fixed in the latest git.
>
>
> No we are not. I am going to try strengthening up the code which 
> closes the DB connections. If that fails then I will compile and try 
> the ODBC driver.
>
>
>     -Steve
>
>
>     On 14 March 2011 09:30, Pete Kelly <pkelly at gmail.com
>     <mailto:pkelly at gmail.com>> wrote:
>
>         Hi
>
>         We are using Freeswitch (from git sources) in a production
>         environment to handle an IVR system using a simple set of
>         dialplans and lua scripts. OpenSIPS 1.6 and mysql are also
>         installed and running as part of the same IVR application.
>
>         However over time (approx 1 week) the memory usage of
>         Freeswitch increases, eventually creeping into swap space
>         forcing us to restart Freeswitch. The IVR handles approx 40
>         concurrent calls constantly, and we have instances of the IVR
>         running on a 32bit Etch and 64bit Lenny boxes - both exhibit
>         the same issue.
>
>         We have also tried the Sangoma freeswitch branch in case that
>         contained any fixes, however we see the same issues.
>
>         Can anybody advise if this is a known issue with Freeswitch?
>         Is it a memory leak or is it possible to limit the amount of
>         memory Freeswitch will use, forcing it to garbage collect when
>         it reaches the limit?
>
>         Any advice on where to look/debug next would be appreciated.
>
>         Thanks
>
>         Pete
>
>
>
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From fernando.berretta at gmail.com  Mon Mar 14 18:03:26 2011
From: fernando.berretta at gmail.com (Fernando Berretta)
Date: Mon, 14 Mar 2011 12:03:26 -0300
Subject: [Freeswitch-users] FreeSwitch for Windows
In-Reply-To: <799432.84909.qm@web33503.mail.mud.yahoo.com>
References: <799432.84909.qm@web33503.mail.mud.yahoo.com>
Message-ID: <4D7E2E3E.7020304@gmail.com>

Diego,

Thanks for your information, I'm gonna contact sangoma support to get 
more information.

Best Regards,
Fernando

On 3/14/2011 11:08 AM, Diego Toro wrote:
> Hi Fernando, I have not used other cards, I have used sangoma cards 
> with excelent support from sangoma's staff.
>
> Diego Toro
> http://voipensando.blogspot.com/
>
> --- On *Mon, 3/14/11, Peter Olsson 
> /<peter.olsson at visionutveckling.se>/* wrote:
>
>
>     From: Peter Olsson <peter.olsson at visionutveckling.se>
>     Subject: Re: [Freeswitch-users] FreeSwitch for Windows
>     To: "'FreeSWITCH Users Help'" <freeswitch-users at lists.freeswitch.org>
>     Date: Monday, March 14, 2011, 7:22 AM
>
>     Sangoma?s PRI (E1) cards are supported under Windows. However,
>     I?ve had some problems with the sangoma_isdn module for some time,
>     so I havn?t used it in about 6 months (it was only in a lab anyway).
>
>     /Peter
>
>     *Fr?n:*freeswitch-users-bounces at lists.freeswitch.org
>     [mailto:freeswitch-users-bounces at lists.freeswitch.org] *F?r
>     *Fernando Berretta
>     *Skickat:* den 14 mars 2011 13:11
>     *Till:* FreeSWITCH Users Help
>     *?mne:* Re: [Freeswitch-users] FreeSwitch for Windows
>
>     Diego,
>
>     Thanks for your answer
>
>     It is possible to use also boards like Digium, OpenVox or any
>     other similar one ? What about E1 cards ? have you tried one of
>     them over Windows ? MFC R2 ?
>
>     Best Regards,
>     Fernando
>
>     On 3/12/2011 11:35 AM, Diego Toro wrote:
>
>     I don't think like Mr. Colling, I have FS on Windows in production
>     since two year using sangoma cards (FXO, FXS) and works fine. 
>     Windows is a good choice.
>
>     Diego Toro
>     http://voipensando.blogspot.com/
>
>     --- On *Fri, 3/11/11, Fernando Berretta
>     /<fernando.berretta at gmail.com>
>     </mc/compose?to=fernando.berretta at gmail.com>/* wrote:
>
>
>     From: Fernando Berretta <fernando.berretta at gmail.com>
>     </mc/compose?to=fernando.berretta at gmail.com>
>     Subject: Re: [Freeswitch-users] FreeSwitch for Windows
>     To: "FreeSWITCH Users Help"
>     <freeswitch-users at lists.freeswitch.org>
>     </mc/compose?to=freeswitch-users at lists.freeswitch.org>
>     Date: Friday, March 11, 2011, 4:54 PM
>
>     Michael,
>
>     Thanks for your valuable recommendation
>
>     Best Regards,
>     Fernando
>
>     On 3/11/2011 6:41 PM, Michael Collins wrote:
>
>     If you plan on using TDM then I'd recommend you stay away from
>     Windows. Sangoma and Digium/clone cards are all geared toward a
>     Linux environment. Stay away from the exotic OSes and use
>     something simple like CentOS 5.5 or Debian Lenny.
>
>     -MC
>
>     On Wed, Mar 9, 2011 at 1:38 PM, Fernando Berretta
>     <fernando.berretta at gmail.com> wrote:
>
>     Hi,
>
>     I'm newbie in FreeSwitch and would like to know if freeswitch could be
>     used for production under Windows platforms ? telephony boards like
>     Digium / OpenVox, Sangoma, etc are going to work ok under Windows ? is
>     there any problem / limitation ? E1 MFCR2 Trunk works ?
>
>     Thanks for your time,
>     Best Regards,
>     Fernando
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>         
>
>     _______________________________________________
>
>     FreeSWITCH-users mailing list
>
>     FreeSWITCH-users at lists.freeswitch.org
>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>     http://www.freeswitch.org
>
>
>     -----Inline Attachment Follows-----
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>         
>
>         
>
>     _______________________________________________
>
>     FreeSWITCH-users mailing list
>
>     FreeSWITCH-users at lists.freeswitch.org  </mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>     http://www.freeswitch.org
>
>     !DSPAM:4d7e069b32762783310086!
>
>
>     -----Inline Attachment Follows-----
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     </mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From dyatsin at sangoma.com  Mon Mar 14 18:13:33 2011
From: dyatsin at sangoma.com (David Yat Sin)
Date: Mon, 14 Mar 2011 11:13:33 -0400
Subject: [Freeswitch-users] FreeSwitch for Windows
In-Reply-To: <799432.84909.qm@web33503.mail.mud.yahoo.com>
References: <799432.84909.qm@web33503.mail.mud.yahoo.com>
Message-ID: <4D7E309D.7040307@sangoma.com>

Hi Fernando,
You should have no problems to use FS + Sangoma cards on Windows. The 
latest libsng_isdn for Windows can be downloaded from here:

ftp://ftp.sangoma.com/WINDOWS/libsng_isdn/libsng_isdn-7.1.0-win32.exe

Instructions on how to install FS can be found here:
http://wiki.freeswitch.org/wiki/Installation_for_Windows

Once you have compiled Freeswitch and installed libsng_isdn, you can to 
go to the libs/freetdm directory and open the Visual Studio project file 
there, compile FreeTDM core and ftmod_sangoma_isdn module.

Regards,
David

*David Yat Sin, BEng*
*/Senior Software Engineer/*
Sangoma Technologies
100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada

	



t.   +1 800 388 2475 x119
t.   +1 905 474 1990 x119
f.   +1 905 474 9223

	

Description: SANGOMA 
<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>

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| Twitter 
<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>


On 3/14/2011 10:08 AM, Diego Toro wrote:
> Hi Fernando, I have not used other cards, I have used sangoma cards 
> with excelent support from sangoma's staff.
>
> Diego Toro
> http://voipensando.blogspot.com/
>
> --- On *Mon, 3/14/11, Peter Olsson 
> /<peter.olsson at visionutveckling.se>/* wrote:
>
>
>     From: Peter Olsson <peter.olsson at visionutveckling.se>
>     Subject: Re: [Freeswitch-users] FreeSwitch for Windows
>     To: "'FreeSWITCH Users Help'" <freeswitch-users at lists.freeswitch.org>
>     Date: Monday, March 14, 2011, 7:22 AM
>
>     Sangoma?s PRI (E1) cards are supported under Windows. However,
>     I?ve had some problems with the sangoma_isdn module for some time,
>     so I havn?t used it in about 6 months (it was only in a lab anyway).
>
>     /Peter
>
>     *Fr?n:*freeswitch-users-bounces at lists.freeswitch.org
>     [mailto:freeswitch-users-bounces at lists.freeswitch.org] *F?r
>     *Fernando Berretta
>     *Skickat:* den 14 mars 2011 13:11
>     *Till:* FreeSWITCH Users Help
>     *?mne:* Re: [Freeswitch-users] FreeSwitch for Windows
>
>     Diego,
>
>     Thanks for your answer
>
>     It is possible to use also boards like Digium, OpenVox or any
>     other similar one ? What about E1 cards ? have you tried one of
>     them over Windows ? MFC R2 ?
>
>     Best Regards,
>     Fernando
>
>     On 3/12/2011 11:35 AM, Diego Toro wrote:
>
>     I don't think like Mr. Colling, I have FS on Windows in production
>     since two year using sangoma cards (FXO, FXS) and works fine. 
>     Windows is a good choice.
>
>     Diego Toro
>     http://voipensando.blogspot.com/
>
>     --- On *Fri, 3/11/11, Fernando Berretta
>     /<fernando.berretta at gmail.com>
>     </mc/compose?to=fernando.berretta at gmail.com>/* wrote:
>
>
>     From: Fernando Berretta <fernando.berretta at gmail.com>
>     </mc/compose?to=fernando.berretta at gmail.com>
>     Subject: Re: [Freeswitch-users] FreeSwitch for Windows
>     To: "FreeSWITCH Users Help"
>     <freeswitch-users at lists.freeswitch.org>
>     </mc/compose?to=freeswitch-users at lists.freeswitch.org>
>     Date: Friday, March 11, 2011, 4:54 PM
>
>     Michael,
>
>     Thanks for your valuable recommendation
>
>     Best Regards,
>     Fernando
>
>     On 3/11/2011 6:41 PM, Michael Collins wrote:
>
>     If you plan on using TDM then I'd recommend you stay away from
>     Windows. Sangoma and Digium/clone cards are all geared toward a
>     Linux environment. Stay away from the exotic OSes and use
>     something simple like CentOS 5.5 or Debian Lenny.
>
>     -MC
>
>     On Wed, Mar 9, 2011 at 1:38 PM, Fernando Berretta
>     <fernando.berretta at gmail.com> wrote:
>
>     Hi,
>
>     I'm newbie in FreeSwitch and would like to know if freeswitch could be
>     used for production under Windows platforms ? telephony boards like
>     Digium / OpenVox, Sangoma, etc are going to work ok under Windows ? is
>     there any problem / limitation ? E1 MFCR2 Trunk works ?
>
>     Thanks for your time,
>     Best Regards,
>     Fernando
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>         
>
>     _______________________________________________
>
>     FreeSWITCH-users mailing list
>
>     FreeSWITCH-users at lists.freeswitch.org
>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>     http://www.freeswitch.org
>
>
>     -----Inline Attachment Follows-----
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>         
>
>         
>
>     _______________________________________________
>
>     FreeSWITCH-users mailing list
>
>     FreeSWITCH-users at lists.freeswitch.org  </mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>     http://www.freeswitch.org
>
>     !DSPAM:4d7e069b32762783310086!
>
>
>     -----Inline Attachment Follows-----
>
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>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     </mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From acrow at integrafin.co.uk  Mon Mar 14 18:15:20 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Mon, 14 Mar 2011 15:15:20 +0000
Subject: [Freeswitch-users] Fax detection - ringing not heard by caller
In-Reply-To: <4D7E0446.90105@integrafin.co.uk>
References: <4D7CF0C3.3050104@integrafin.co.uk>
	<4D7E0446.90105@integrafin.co.uk>
Message-ID: <4D7E3108.7000707@integrafin.co.uk>

On 14/03/11 12:04, Alex Crow wrote:
>
> I've fiddled around for ages and I still can't get it to work. The
> Trixbox machine does it perfectly and I need to make the FS box behave
> the same, but it seems once "answer" has been executed I can only use
> "playback" to send ringing to the caller, but I can only do this while
> the tone detect is running. If it set playback to loop then the plan
> never proceeds to the bridge/transfer to the extension.
>
>

Figured it out myself - I needed to "set transfer_ringback=$${uk-ring}" 
after "answer".

Alex

-- 
This message is intended only for the addressee and may contain
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From steveayre at gmail.com  Mon Mar 14 18:27:20 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 14 Mar 2011 15:27:20 +0000
Subject: [Freeswitch-users] Memory leak
In-Reply-To: <AANLkTimLNa73o5Y_h6KUpTptNyX6Q-6bpBOQBjLtsnS6@mail.gmail.com>
References: <AANLkTim3UsX4z_URe1Ya40uP4AnVG3s3NiYkew=ZY8+O@mail.gmail.com>
	<AANLkTincGerDeJtkyqzYXFmtTzjHe-h9RMZJsM5HosCS@mail.gmail.com>
	<AANLkTimLNa73o5Y_h6KUpTptNyX6Q-6bpBOQBjLtsnS6@mail.gmail.com>
Message-ID: <AANLkTinxMmO8UCR6wjROjB=DoD_yUoe2EuBNV=rNb3AC@mail.gmail.com>

Try using the freeswitch.Dbh interface:
http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh

It gives you a ODBC interface through freeswitch's odbc support, plus you
get the benefit of the connection caching.

-Steve



On 14 March 2011 14:28, Pete Kelly <pkelly at gmail.com> wrote:

>
>
> On 14 March 2011 13:16, Steven Ayre <steveayre at gmail.com> wrote:
>
>> If you use mod_lcr, we were getting a memory leak in that a few weeks ago
>> which is fixed in the latest git.
>
>
> No we are not. I am going to try strengthening up the code which closes the
> DB connections. If that fails then I will compile and try the ODBC driver.
>
>
>>
>> -Steve
>>
>>
>> On 14 March 2011 09:30, Pete Kelly <pkelly at gmail.com> wrote:
>>
>>> Hi
>>>
>>> We are using Freeswitch (from git sources) in a production environment to
>>> handle an IVR system using a simple set of dialplans and lua scripts.
>>> OpenSIPS 1.6 and mysql are also installed and running as part of the same
>>> IVR application.
>>>
>>> However over time (approx 1 week) the memory usage of Freeswitch
>>> increases, eventually creeping into swap space forcing us to restart
>>> Freeswitch. The IVR handles approx 40 concurrent calls constantly, and we
>>> have instances of the IVR running on a 32bit Etch and 64bit Lenny boxes -
>>> both exhibit the same issue.
>>>
>>> We have also tried the Sangoma freeswitch branch in case that contained
>>> any fixes, however we see the same issues.
>>>
>>> Can anybody advise if this is a known issue with Freeswitch? Is it a
>>> memory leak or is it possible to limit the amount of memory Freeswitch will
>>> use, forcing it to garbage collect when it reaches the limit?
>>>
>>> Any advice on where to look/debug next would be appreciated.
>>>
>>> Thanks
>>>
>>> Pete
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mario_fs at mgtech.com  Mon Mar 14 19:18:23 2011
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 14 Mar 2011 09:18:23 -0700
Subject: [Freeswitch-users] Memory leak
In-Reply-To: <AANLkTinxMmO8UCR6wjROjB=DoD_yUoe2EuBNV=rNb3AC@mail.gmail.com>
References: <AANLkTim3UsX4z_URe1Ya40uP4AnVG3s3NiYkew=ZY8+O@mail.gmail.com>
	<AANLkTincGerDeJtkyqzYXFmtTzjHe-h9RMZJsM5HosCS@mail.gmail.com>
	<AANLkTimLNa73o5Y_h6KUpTptNyX6Q-6bpBOQBjLtsnS6@mail.gmail.com>
	<AANLkTinxMmO8UCR6wjROjB=DoD_yUoe2EuBNV=rNb3AC@mail.gmail.com>
Message-ID: <188EF393-51D5-4740-9572-573F8DA04157@mgtech.com>

This may not be related but I have had a memory leak since last year that requires a restart every night. About 2M an hour on my OS X system.

On Mar 14, 2011, at 8:27 AM, Steven Ayre wrote:

> Try using the freeswitch.Dbh interface:
> http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh
> 
> It gives you a ODBC interface through freeswitch's odbc support, plus you get the benefit of the connection caching.
> 
> -Steve
> 
> 
> 
> On 14 March 2011 14:28, Pete Kelly <pkelly at gmail.com> wrote:
> 
> 
> On 14 March 2011 13:16, Steven Ayre <steveayre at gmail.com> wrote:
> If you use mod_lcr, we were getting a memory leak in that a few weeks ago which is fixed in the latest git.
> 
> No we are not. I am going to try strengthening up the code which closes the DB connections. If that fails then I will compile and try the ODBC driver.
>  
> 
> -Steve
> 
> 
> On 14 March 2011 09:30, Pete Kelly <pkelly at gmail.com> wrote:
> Hi
> 
> We are using Freeswitch (from git sources) in a production environment to handle an IVR system using a simple set of dialplans and lua scripts. OpenSIPS 1.6 and mysql are also installed and running as part of the same IVR application.
> 
> However over time (approx 1 week) the memory usage of Freeswitch increases, eventually creeping into swap space forcing us to restart Freeswitch. The IVR handles approx 40 concurrent calls constantly, and we have instances of the IVR running on a 32bit Etch and 64bit Lenny boxes - both exhibit the same issue.
> 
> We have also tried the Sangoma freeswitch branch in case that contained any fixes, however we see the same issues.
> 
> Can anybody advise if this is a known issue with Freeswitch? Is it a memory leak or is it possible to limit the amount of memory Freeswitch will use, forcing it to garbage collect when it reaches the limit? 
> 
> Any advice on where to look/debug next would be appreciated.
> 
> Thanks
> 
> Pete
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Mon Mar 14 19:25:04 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 14 Mar 2011 11:25:04 -0500
Subject: [Freeswitch-users] Memory leak
In-Reply-To: <AANLkTinxMmO8UCR6wjROjB=DoD_yUoe2EuBNV=rNb3AC@mail.gmail.com>
References: <AANLkTim3UsX4z_URe1Ya40uP4AnVG3s3NiYkew=ZY8+O@mail.gmail.com>
	<AANLkTincGerDeJtkyqzYXFmtTzjHe-h9RMZJsM5HosCS@mail.gmail.com>
	<AANLkTimLNa73o5Y_h6KUpTptNyX6Q-6bpBOQBjLtsnS6@mail.gmail.com>
	<AANLkTinxMmO8UCR6wjROjB=DoD_yUoe2EuBNV=rNb3AC@mail.gmail.com>
Message-ID: <AANLkTin9XpLcop+QnPFE70GSnEEHx5_yxWFhDrEJDBvq@mail.gmail.com>

also make sure to use the _r version of any libs you use such as
myodbc or mysqlclient

look in /etc/odbcinst.ini and chances are its not the _r one

# Driver from the MyODBC package
# Setup from the unixODBC package
[MySQL]
Description     = ODBC for MySQL
Driver          = /usr/lib64/libmyodbc3.so
Setup           = /usr/lib64/libodbcmyS.so
FileUsage       = 1


/usr/lib64/libmyodbc3.so needs to be /usr/lib64/libmyodbc3_r.so




On Mon, Mar 14, 2011 at 10:27 AM, Steven Ayre <steveayre at gmail.com> wrote:
> Try using the freeswitch.Dbh interface:
> http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh
> It gives you a ODBC interface through freeswitch's odbc support, plus you
> get the benefit of the connection caching.
> -Steve
>
>
> On 14 March 2011 14:28, Pete Kelly <pkelly at gmail.com> wrote:
>>
>>
>> On 14 March 2011 13:16, Steven Ayre <steveayre at gmail.com> wrote:
>>>
>>> If you use mod_lcr, we were getting a memory leak in that a few weeks ago
>>> which is fixed in the latest git.
>>
>> No we are not. I am going to try strengthening up the code which closes
>> the DB connections. If that fails then I will compile and try the ODBC
>> driver.
>>
>>>
>>> -Steve
>>>
>>> On 14 March 2011 09:30, Pete Kelly <pkelly at gmail.com> wrote:
>>>>
>>>> Hi
>>>> We are using Freeswitch (from git sources) in a production environment
>>>> to handle an IVR system using a simple set of dialplans and lua scripts.
>>>> OpenSIPS 1.6 and mysql are also installed and running as part of the same
>>>> IVR application.
>>>> However over time (approx 1 week) the memory usage of Freeswitch
>>>> increases, eventually creeping into swap space forcing us to restart
>>>> Freeswitch. The IVR handles approx 40 concurrent calls constantly, and we
>>>> have instances of the IVR running on a 32bit Etch and 64bit Lenny boxes -
>>>> both exhibit the same issue.
>>>> We have also tried the Sangoma freeswitch branch in case that contained
>>>> any fixes, however we see the same issues.
>>>> Can anybody advise if this is a known issue with Freeswitch? Is it a
>>>> memory leak or is it possible to limit the amount of memory Freeswitch will
>>>> use, forcing it to garbage collect when it reaches the limit?
>>>> Any advice on where to look/debug next would be appreciated.
>>>> Thanks
>>>> Pete
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 Mar 14 19:30:56 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 14 Mar 2011 16:30:56 +0000
Subject: [Freeswitch-users] Memory leak
In-Reply-To: <188EF393-51D5-4740-9572-573F8DA04157@mgtech.com>
References: <AANLkTim3UsX4z_URe1Ya40uP4AnVG3s3NiYkew=ZY8+O@mail.gmail.com>
	<AANLkTincGerDeJtkyqzYXFmtTzjHe-h9RMZJsM5HosCS@mail.gmail.com>
	<AANLkTimLNa73o5Y_h6KUpTptNyX6Q-6bpBOQBjLtsnS6@mail.gmail.com>
	<AANLkTinxMmO8UCR6wjROjB=DoD_yUoe2EuBNV=rNb3AC@mail.gmail.com>
	<188EF393-51D5-4740-9572-573F8DA04157@mgtech.com>
Message-ID: <AANLkTikH5gQqvBeLVW8i3GBVwwqVW8vz6kS2ny=x2pNy@mail.gmail.com>

Have you tried upgrading to the latest Git?

-Steve


On 14 March 2011 16:18, Mario G <mario_fs at mgtech.com> wrote:

> This may not be related but I have had a memory leak since last year that
> requires a restart every night. About 2M an hour on my OS X system.
>
> On Mar 14, 2011, at 8:27 AM, Steven Ayre wrote:
>
> Try using the freeswitch.Dbh interface:
> http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh
>
> It gives you a ODBC interface through freeswitch's odbc support, plus you
> get the benefit of the connection caching.
>
> -Steve
>
>
>
> On 14 March 2011 14:28, Pete Kelly <pkelly at gmail.com> wrote:
>
>>
>>
>> On 14 March 2011 13:16, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> If you use mod_lcr, we were getting a memory leak in that a few weeks ago
>>> which is fixed in the latest git.
>>
>>
>> No we are not. I am going to try strengthening up the code which closes
>> the DB connections. If that fails then I will compile and try the ODBC
>> driver.
>>
>>
>>>
>>> -Steve
>>>
>>>
>>> On 14 March 2011 09:30, Pete Kelly <pkelly at gmail.com> wrote:
>>>
>>>> Hi
>>>>
>>>> We are using Freeswitch (from git sources) in a production environment
>>>> to handle an IVR system using a simple set of dialplans and lua scripts.
>>>> OpenSIPS 1.6 and mysql are also installed and running as part of the same
>>>> IVR application.
>>>>
>>>> However over time (approx 1 week) the memory usage of Freeswitch
>>>> increases, eventually creeping into swap space forcing us to restart
>>>> Freeswitch. The IVR handles approx 40 concurrent calls constantly, and we
>>>> have instances of the IVR running on a 32bit Etch and 64bit Lenny boxes -
>>>> both exhibit the same issue.
>>>>
>>>> We have also tried the Sangoma freeswitch branch in case that contained
>>>> any fixes, however we see the same issues.
>>>>
>>>> Can anybody advise if this is a known issue with Freeswitch? Is it a
>>>> memory leak or is it possible to limit the amount of memory Freeswitch will
>>>> use, forcing it to garbage collect when it reaches the limit?
>>>>
>>>> Any advice on where to look/debug next would be appreciated.
>>>>
>>>> Thanks
>>>>
>>>> Pete
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mario_fs at mgtech.com  Mon Mar 14 19:38:05 2011
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 14 Mar 2011 09:38:05 -0700
Subject: [Freeswitch-users] Memory leak
In-Reply-To: <AANLkTikH5gQqvBeLVW8i3GBVwwqVW8vz6kS2ny=x2pNy@mail.gmail.com>
References: <AANLkTim3UsX4z_URe1Ya40uP4AnVG3s3NiYkew=ZY8+O@mail.gmail.com>
	<AANLkTincGerDeJtkyqzYXFmtTzjHe-h9RMZJsM5HosCS@mail.gmail.com>
	<AANLkTimLNa73o5Y_h6KUpTptNyX6Q-6bpBOQBjLtsnS6@mail.gmail.com>
	<AANLkTinxMmO8UCR6wjROjB=DoD_yUoe2EuBNV=rNb3AC@mail.gmail.com>
	<188EF393-51D5-4740-9572-573F8DA04157@mgtech.com>
	<AANLkTikH5gQqvBeLVW8i3GBVwwqVW8vz6kS2ny=x2pNy@mail.gmail.com>
Message-ID: <9F22E106-FBD9-47ED-A9F8-894C9A2ACDC6@mgtech.com>

Been updating gits since last year and even today. This is a known open problem in Jira. It is related to internal registrations, the more phone/faster frequency the faster the leak. It is low priority right now since I am having other major FS problems due to a router change. Thanks for asking!
Mario G

On Mar 14, 2011, at 9:30 AM, Steven Ayre wrote:

> Have you tried upgrading to the latest Git?
> 
> -Steve
> 
> 
> On 14 March 2011 16:18, Mario G <mario_fs at mgtech.com> wrote:
> This may not be related but I have had a memory leak since last year that requires a restart every night. About 2M an hour on my OS X system.
> 
> On Mar 14, 2011, at 8:27 AM, Steven Ayre wrote:
> 
>> Try using the freeswitch.Dbh interface:
>> http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh
>> 
>> It gives you a ODBC interface through freeswitch's odbc support, plus you get the benefit of the connection caching.
>> 
>> -Steve
>> 
>> 
>> 
>> On 14 March 2011 14:28, Pete Kelly <pkelly at gmail.com> wrote:
>> 
>> 
>> On 14 March 2011 13:16, Steven Ayre <steveayre at gmail.com> wrote:
>> If you use mod_lcr, we were getting a memory leak in that a few weeks ago which is fixed in the latest git.
>> 
>> No we are not. I am going to try strengthening up the code which closes the DB connections. If that fails then I will compile and try the ODBC driver.
>>  
>> 
>> -Steve
>> 
>> 
>> On 14 March 2011 09:30, Pete Kelly <pkelly at gmail.com> wrote:
>> Hi
>> 
>> We are using Freeswitch (from git sources) in a production environment to handle an IVR system using a simple set of dialplans and lua scripts. OpenSIPS 1.6 and mysql are also installed and running as part of the same IVR application.
>> 
>> However over time (approx 1 week) the memory usage of Freeswitch increases, eventually creeping into swap space forcing us to restart Freeswitch. The IVR handles approx 40 concurrent calls constantly, and we have instances of the IVR running on a 32bit Etch and 64bit Lenny boxes - both exhibit the same issue.
>> 
>> We have also tried the Sangoma freeswitch branch in case that contained any fixes, however we see the same issues.
>> 
>> Can anybody advise if this is a known issue with Freeswitch? Is it a memory leak or is it possible to limit the amount of memory Freeswitch will use, forcing it to garbage collect when it reaches the limit? 
>> 
>> Any advice on where to look/debug next would be appreciated.
>> 
>> Thanks
>> 
>> Pete
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From moises.silva at gmail.com  Mon Mar 14 19:39:29 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 14 Mar 2011 12:39:29 -0400
Subject: [Freeswitch-users] FreeSwitch for Windows
In-Reply-To: <4D7E05B7.9040403@gmail.com>
References: <134839.19366.qm@web33506.mail.mud.yahoo.com>
	<4D7E05B7.9040403@gmail.com>
Message-ID: <AANLkTi=QOQfvNbRd3QA8N407jU9hVDF9M03V7LXZG14A@mail.gmail.com>

MFC-R2 via the openr2 stack works very well in Windows with Sangoma cards.

There is no formal release yet though (should be ready in about a month) but
the code is not changing any more, the only reason to not have a release is
that we want a Windows installer for the openr2 stack. You need to build the
stack yourself using CMake for Windows and Visual Studio in the meantime.

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 Mon, Mar 14, 2011 at 8:10 AM, Fernando Berretta <
fernando.berretta at gmail.com> wrote:

>  Diego,
>
> Thanks for your answer
>
> It is possible to use also boards like Digium, OpenVox or any other similar
> one ? What about E1 cards ? have you tried one of them over Windows ? MFC R2
> ?
>
> Best Regards,
> Fernando
>
> On 3/12/2011 11:35 AM, Diego Toro wrote:
>
>   I don't think like Mr. Colling, I have FS on Windows in production since
> two year using sangoma cards (FXO, FXS) and works fine.  Windows is a good
> choice.
>
> Diego Toro
> http://voipensando.blogspot.com/
>
> --- On *Fri, 3/11/11, Fernando Berretta <fernando.berretta at gmail.com><fernando.berretta at gmail.com>
> * wrote:
>
>
> From: Fernando Berretta <fernando.berretta at gmail.com><fernando.berretta at gmail.com>
> Subject: Re: [Freeswitch-users] FreeSwitch for Windows
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org><freeswitch-users at lists.freeswitch.org>
> Date: Friday, March 11, 2011, 4:54 PM
>
>  Michael,
>
> Thanks for your valuable recommendation
>
> Best Regards,
> Fernando
>
> On 3/11/2011 6:41 PM, Michael Collins wrote:
>
> If you plan on using TDM then I'd recommend you stay away from Windows.
> Sangoma and Digium/clone cards are all geared toward a Linux environment.
> Stay away from the exotic OSes and use something simple like CentOS 5.5 or
> Debian Lenny.
>
>  -MC
>
>
> On Wed, Mar 9, 2011 at 1:38 PM, Fernando Berretta <
> fernando.berretta at gmail.com<http://mc/compose?to=fernando.berretta at gmail.com>
> > wrote:
>
> Hi,
>
> I'm newbie in FreeSwitch and would like to know if freeswitch could be
> used for production under Windows platforms ? telephony boards like
> Digium / OpenVox, Sangoma, etc are going to work ok under Windows ? is
> there any problem / limitation ? E1 MFCR2 Trunk works ?
>
> Thanks for your time,
> Best Regards,
> Fernando
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.org <http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> -----Inline Attachment Follows-----
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From anthony.minessale at gmail.com  Mon Mar 14 19:56:36 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 14 Mar 2011 11:56:36 -0500
Subject: [Freeswitch-users] Memory leak
In-Reply-To: <9F22E106-FBD9-47ED-A9F8-894C9A2ACDC6@mgtech.com>
References: <AANLkTim3UsX4z_URe1Ya40uP4AnVG3s3NiYkew=ZY8+O@mail.gmail.com>
	<AANLkTincGerDeJtkyqzYXFmtTzjHe-h9RMZJsM5HosCS@mail.gmail.com>
	<AANLkTimLNa73o5Y_h6KUpTptNyX6Q-6bpBOQBjLtsnS6@mail.gmail.com>
	<AANLkTinxMmO8UCR6wjROjB=DoD_yUoe2EuBNV=rNb3AC@mail.gmail.com>
	<188EF393-51D5-4740-9572-573F8DA04157@mgtech.com>
	<AANLkTikH5gQqvBeLVW8i3GBVwwqVW8vz6kS2ny=x2pNy@mail.gmail.com>
	<9F22E106-FBD9-47ED-A9F8-894C9A2ACDC6@mgtech.com>
Message-ID: <AANLkTi=jHhuxW8zbFp94Xz9bb-hJAna29ofPwwerkXKZ@mail.gmail.com>

its also Mac specific and reported to sofia devs.


On Mon, Mar 14, 2011 at 11:38 AM, Mario G <mario_fs at mgtech.com> wrote:
> Been updating gits since last year and even today. This is a known open
> problem in Jira. It is related to internal registrations, the more
> phone/faster frequency the faster the leak. It is low priority right now
> since I am having other major FS problems due to a router change. Thanks for
> asking!
> Mario G
>
> On Mar 14, 2011, at 9:30 AM, Steven Ayre wrote:
>
> Have you tried upgrading to the latest Git?
> -Steve
>
> On 14 March 2011 16:18, Mario G <mario_fs at mgtech.com> wrote:
>>
>> This may not be related but I have had a memory leak since last year that
>> requires a restart every night. About 2M an hour on my OS X system.
>> On Mar 14, 2011, at 8:27 AM, Steven Ayre wrote:
>>
>> Try using the freeswitch.Dbh interface:
>> http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh
>> It gives you a ODBC interface through freeswitch's odbc support, plus you
>> get the benefit of the connection caching.
>> -Steve
>>
>>
>> On 14 March 2011 14:28, Pete Kelly <pkelly at gmail.com> wrote:
>>>
>>>
>>> On 14 March 2011 13:16, Steven Ayre <steveayre at gmail.com> wrote:
>>>>
>>>> If you use mod_lcr, we were getting a memory leak in that a few weeks
>>>> ago which is fixed in the latest git.
>>>
>>> No we are not. I am going to try strengthening up the code which closes
>>> the DB connections. If that fails then I will compile and try the ODBC
>>> driver.
>>>
>>>>
>>>> -Steve
>>>>
>>>> On 14 March 2011 09:30, Pete Kelly <pkelly at gmail.com> wrote:
>>>>>
>>>>> Hi
>>>>> We are using Freeswitch (from git sources) in a production environment
>>>>> to handle an IVR system using a simple set of dialplans and lua scripts.
>>>>> OpenSIPS 1.6 and mysql are also installed and running as part of the same
>>>>> IVR application.
>>>>> However over time (approx 1 week) the memory usage of Freeswitch
>>>>> increases, eventually creeping into swap space forcing us to restart
>>>>> Freeswitch. The IVR handles approx 40 concurrent calls constantly, and we
>>>>> have instances of the IVR running on a 32bit Etch and 64bit Lenny boxes -
>>>>> both exhibit the same issue.
>>>>> We have also tried the Sangoma freeswitch branch in case that contained
>>>>> any fixes, however we see the same issues.
>>>>> Can anybody advise if this is a known issue with Freeswitch? Is it a
>>>>> memory leak or is it possible to limit the amount of memory Freeswitch will
>>>>> use, forcing it to garbage collect when it reaches the limit?
>>>>> Any advice on where to look/debug next would be appreciated.
>>>>> Thanks
>>>>> Pete
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
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>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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MSN:anthony_minessale at hotmail.com
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From peter.olsson at visionutveckling.se  Mon Mar 14 19:57:21 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 14 Mar 2011 17:57:21 +0100
Subject: [Freeswitch-users] Memory leak
In-Reply-To: <9F22E106-FBD9-47ED-A9F8-894C9A2ACDC6@mgtech.com>
References: <AANLkTim3UsX4z_URe1Ya40uP4AnVG3s3NiYkew=ZY8+O@mail.gmail.com>
	<AANLkTincGerDeJtkyqzYXFmtTzjHe-h9RMZJsM5HosCS@mail.gmail.com>
	<AANLkTimLNa73o5Y_h6KUpTptNyX6Q-6bpBOQBjLtsnS6@mail.gmail.com>
	<AANLkTinxMmO8UCR6wjROjB=DoD_yUoe2EuBNV=rNb3AC@mail.gmail.com>
	<188EF393-51D5-4740-9572-573F8DA04157@mgtech.com>
	<AANLkTikH5gQqvBeLVW8i3GBVwwqVW8vz6kS2ny=x2pNy@mail.gmail.com>,
	<9F22E106-FBD9-47ED-A9F8-894C9A2ACDC6@mgtech.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2C4951F@cooper>

Which Jira issue is that - I couldn't find it when searching Jira?

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Mario G [mario_fs at mgtech.com]
Skickat: den 14 mars 2011 17:38
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Memory leak

Been updating gits since last year and even today. This is a known open problem in Jira. It is related to internal registrations, the more phone/faster frequency the faster the leak. It is low priority right now since I am having other major FS problems due to a router change. Thanks for asking!
Mario G

On Mar 14, 2011, at 9:30 AM, Steven Ayre wrote:

Have you tried upgrading to the latest Git?

-Steve


On 14 March 2011 16:18, Mario G <mario_fs at mgtech.com<mailto:mario_fs at mgtech.com>> wrote:
This may not be related but I have had a memory leak since last year that requires a restart every night. About 2M an hour on my OS X system.

On Mar 14, 2011, at 8:27 AM, Steven Ayre wrote:

Try using the freeswitch.Dbh interface:
http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh

It gives you a ODBC interface through freeswitch's odbc support, plus you get the benefit of the connection caching.

-Steve



On 14 March 2011 14:28, Pete Kelly <pkelly at gmail.com<mailto:pkelly at gmail.com>> wrote:


On 14 March 2011 13:16, Steven Ayre <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
If you use mod_lcr, we were getting a memory leak in that a few weeks ago which is fixed in the latest git.

No we are not. I am going to try strengthening up the code which closes the DB connections. If that fails then I will compile and try the ODBC driver.


-Steve


On 14 March 2011 09:30, Pete Kelly <pkelly at gmail.com<mailto:pkelly at gmail.com>> wrote:
Hi

We are using Freeswitch (from git sources) in a production environment to handle an IVR system using a simple set of dialplans and lua scripts. OpenSIPS 1.6 and mysql are also installed and running as part of the same IVR application.

However over time (approx 1 week) the memory usage of Freeswitch increases, eventually creeping into swap space forcing us to restart Freeswitch. The IVR handles approx 40 concurrent calls constantly, and we have instances of the IVR running on a 32bit Etch and 64bit Lenny boxes - both exhibit the same issue.

We have also tried the Sangoma freeswitch branch in case that contained any fixes, however we see the same issues.

Can anybody advise if this is a known issue with Freeswitch? Is it a memory leak or is it possible to limit the amount of memory Freeswitch will use, forcing it to garbage collect when it reaches the limit?

Any advice on where to look/debug next would be appreciated.

Thanks

Pete



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!DSPAM:4d7e458932761594220015!


From anthony.minessale at gmail.com  Mon Mar 14 20:02:30 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 14 Mar 2011 12:02:30 -0500
Subject: [Freeswitch-users] PA and ALSA
In-Reply-To: <201103132032.19044.sos@sokhapkin.dyndns.org>
References: <4D7BF5E6.30201@rosengart.de>
	<201103131921.18213.sos@sokhapkin.dyndns.org>
	<AANLkTikYUsbU6=gd=SrRd_a=F=j40eV=D0YW4SLCNuQU@mail.gmail.com>
	<201103132032.19044.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTim81nXTRSF3SZUME2fzW+zDBqdrVOD7M1R4R-Y4@mail.gmail.com>

Anything is possible if we get patches to make it configurable and our
preferences are the default.


On Sun, Mar 13, 2011 at 7:32 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org> wrote:
> The power of gentoo linux is that ebuild can specify a range of required
> versions for a third party library. Maybe something can be implemented in FS
> build script too? Just check a version of the installed third party library in
> configure script.
>
> On Sunday 13 March 2011, Anthony Minessale wrote:
>> 2 things to keep in mind.
>>
>> 1 Try to wrap your head around making a stable application that runs on
>> windows, unix including mac and some bsd, and unix-like distros such as
>> those mentioned. ?Not all of our dependancies are stable on all of those
>> platforms or even easily available in some cases.
>>
>> 2 This is a community project, if some versions of libraries we are using
>> are older maybe they have been working fine or the modules are not used
>> enough to warrant attention.
>>
>> The normal process would be:
>>
>> The guy reports a problem and suggests its fixed in a newer version. ?Now
>> try this code on every supported platform and make sure it works across the
>> board, if not, well, we can't blindly add it. ?All of this can be done
>> without the ungrateful cracks and side stories about gentoo and we can
>> happily update the code.
>>
>> I don't go to gentoo forum and complain that custom patching the kernel and
>> libc annoys me or about how I am not interested in being forced into their
>> religion to have my software in their distro even though it does not
>> install any third party libs, FS only static links in bits of code from in
>> tree depends.
>>
>> Furthermore, we constantly invite people to assist with platform and
>> packaging and nobody offers anything but complaints.
>>
>> An example of why we don't just blindly use system libs:
>> The latest sqlite causes 25% loss in performance and guarenteed segfaults
>> and occasional deadlock under load.
>> This is a place where people complain about 1% cpu usage.
>>
>> Alsa mod is specifically for the nokia N800 N810 devices.
>> Stick with pa or help us write something new from scratch and don't forget
>> linux is not alone in the computing world.
>>
>> On Mar 13, 2011 6:22 PM, "Sergey Okhapkin" <sos at sokhapkin.dyndns.org> wrote:
>> > The attempt to isolate core FS code from third party libraries has been
>>
>> done
>>
>> > by gentoo linux team already (inclusion of third party open source code
>> > is strictly against gentoo philosophy), but the attempt has been
>> > abandoned because of lack of support from upstream.
>> >
>> > On Sunday 13 March 2011 19:05:17 Jeff Lenk wrote:
>> > > I'm sure there were very good reasons for these decisions when they
>> > > were made. So rather than criticizing those decisions why not help by
>> > > finding the problems and submitting patches to fix them.
>> > >
>> > > --
>>
>> > > View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/PA-and-ALSA-tp6165168p616736
>>
>> > > 8.html Sent from the freeswitch-users mailing list archive at
>>
>> Nabble.com.
>>
>> > > _______________________________________________
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
>> > > s http://www.freeswitch.org
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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
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pstn:+19193869900


From mario_fs at mgtech.com  Mon Mar 14 20:07:36 2011
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 14 Mar 2011 10:07:36 -0700
Subject: [Freeswitch-users] multiple gateways and NAT
In-Reply-To: <4D7DD9D4.3000002@julf.com>
References: <4D7D0A6F.2020508@julf.com><EE2DD131C4C543B3A0FDF5BBE7196CE9@e1705>	<4D7D18C5.2040404@julf.com><D81653003224456281F83BFA528134C4@e1705>	<4D7D1FA1.5050600@julf.com>	<913499CF71FC4CA788510BA78E391891@e1705>
	<C7F29C81-A205-426F-B296-47FA849EF4C3@mgtech.com>
	<4D7DD9D4.3000002@julf.com>
Message-ID: <37078331-559B-4A55-A5DC-9E8A26E23B0E@mgtech.com>

I'll tell you what I know/observed: Based on web searches you need address and port. Here is an example of what happened to me during testing:

I have multiple user IDs on my internal phone, so each phone registers multiple times, one for each extension. When I configured the phone (SPA962) to use the same port (5060) for all extensions the phones had problems such as not ringing when they should. When I gave each extension its own port and problems solved.

As for ITSPs, I remember that I could not get the ITSP that had 2 accounts working right until each account had its own port. My guess is that the connection between the ITSP is registered with the user ID, but then that ID is no longer used, just the IP and port. Since FS had the same ip addr and port for both accounts there is no way to know what data belonged to what account. Making separate ports fixed the problem.

I also am very confused about FS ports. In vars.xml there is only one port that can be setup for internal and external. However, I have separate external profiles each with its own port that is required to make things work. But.... I only have 1 internal profile and yet all extensions work even though they are setup with different ports for each extension. You would think FS would require different internal profiles for each ID as it does for external. Go figure.... Hope this helps a little.

As for using port in registrations I am not sure. The reason is while testing dual WAN I had a call going on one WAN then pulled the plug. The other took over and FS kept the conversation going. So in that case even the IP address changed! But... I am having other major FS problems trying to get a working backup so had to abandon the FS dual wan setup and stick to 1 wan connection for FS.

Mario G


On Mar 14, 2011, at 2:03 AM, Johan Helsingius wrote:

>> I remembered why I needed to use different ports in one case: Two
>> of the three accounts are at the same ITSP, in this case since the
>> IP address is the same on the two accounts you need to use different ports.
> 
> Ah, I have sort of the same case - 4 accounts with one ITSP providing
> 4 different incoming DID's (replacing an old ISDN connection), and
> 2 separate single ITSP accounts for international calls.
> 
> But I am still very confused about the need for different ports.
> This is not a complaint, just an observation, but I do think the
> multiple port thing is one of the more confusing aspects of FS,
> and something other SIP switches seem to do without.
> 
> On outgoing connections, we register with the ITSP gateway (by the
> way, does the SIP registration register both an ip address and
> a port number, or just the ip address?), and on incoming connections
> they either have a destination number/DID that routes them to the
> right place in the dialplan, or they register with us (using
> username/password) in the case of SIP clients. Security can be
> handled with ACL's. So I can't get my head around to why separate
> ports are needed - maybe someone can point out what I am missing?
> 
> 	Julf
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mctch at yahoo.com  Mon Mar 14 20:16:54 2011
From: mctch at yahoo.com (Mark Crane)
Date: Mon, 14 Mar 2011 10:16:54 -0700 (PDT)
Subject: [Freeswitch-users] mod_callcenter features
In-Reply-To: <201012310219.10813.justlikeef@gmail.com>
Message-ID: <901828.47060.qm@web121508.mail.ne1.yahoo.com>

> 1. In mod_callcenter, there is no command like "callcenter_config queue add"

This would be a great feature to have ability to add and delete new call center queues.

Mark J Crane


--- On Fri, 12/31/10, Rob Hutton <justlikeef at gmail.com> wrote:

From: Rob Hutton <justlikeef at gmail.com>
Subject: Re: [Freeswitch-users] mod_callcenter features
To: "Rajkumar K" <rajkumar.kmry at gmail.com>
Cc: freeswitch-users at lists.freeswitch.org
Date: Friday, December 31, 2010, 12:19 AM

1. Try something like:

reloadxml
callcenter_config queue load|reload queue_name

2. I don't believe so.? Hit Moc on the IRC channel.? I am trying, also...


-- 
Thanks,
Rob
On Friday 31 December 2010 00:32:46 Rajkumar K wrote:
> Hi,
> 
> 1. In mod_callcenter, there is no command like "callcenter_config queue add"
> for adding queues dynamically. We need to configure it in
> callcenter.conf.xml file and run "callcenter_config queue reload <queue
> name>". So is there any other ways to add queues.
> 
> 2. The mod_callcenter automatically bridges the call once it identifies the
> available agent. But the requirement is to identify the available agent and
> leave the bridge control to our dial plan. There we need to other operations
> (like announcing caller name to agent) instead of? bridging the call.
> 
> regards
> rajkumar k
> 
> On Thu, Dec 30, 2010 at 6:12 PM, Rob Hutton <justlikeef at gmail.com> wrote:
> 
> > Some of the API commands are on in the Wiki under the heading API:
> >
> > http://wiki.freeswitch.org/wiki/Mod_callcenter
> >
> > Everything starts with callcenter_config, so you can feel your way around
> > the command line and everything is fairly self explanatory.
> >
> > What do you mean control the call flow?? Can you give an example of what
> > you are trying to accomplish?
> >
> > --
> > Thanks,
> > Rob
> > On Monday 27 December 2010 04:42:29 rajkumar wrote:
> > >
> > > Hi,
> > >
> > >???I am developing an application with mod_callcenter. I need to know the
> > > following about mod_callcenter.
> > >
> > >???* Is it possible to add/update/delete the queue configurations
> > dynamically
> > > without using static xml configurations.
> > >???* How can I control the call flow (for playback message and recording)
> > > before and after bridging the call.
> > >
> > > Thanks in advance
> > >
> > > regards
> > > rajkumar k
> > >
> >
> 

_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From peter.olsson at visionutveckling.se  Mon Mar 14 20:35:30 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 14 Mar 2011 18:35:30 +0100
Subject: [Freeswitch-users] Memory leak
In-Reply-To: <AANLkTi=jHhuxW8zbFp94Xz9bb-hJAna29ofPwwerkXKZ@mail.gmail.com>
References: <AANLkTim3UsX4z_URe1Ya40uP4AnVG3s3NiYkew=ZY8+O@mail.gmail.com>
	<AANLkTincGerDeJtkyqzYXFmtTzjHe-h9RMZJsM5HosCS@mail.gmail.com>
	<AANLkTimLNa73o5Y_h6KUpTptNyX6Q-6bpBOQBjLtsnS6@mail.gmail.com>
	<AANLkTinxMmO8UCR6wjROjB=DoD_yUoe2EuBNV=rNb3AC@mail.gmail.com>
	<188EF393-51D5-4740-9572-573F8DA04157@mgtech.com>
	<AANLkTikH5gQqvBeLVW8i3GBVwwqVW8vz6kS2ny=x2pNy@mail.gmail.com>
	<9F22E106-FBD9-47ED-A9F8-894C9A2ACDC6@mgtech.com>,
	<AANLkTi=jHhuxW8zbFp94Xz9bb-hJAna29ofPwwerkXKZ@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2C49521@cooper>

By the way - I just noticed that the Sofia devs released version 1.12.11 about a week ago, but I guess you already knew that :)
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Anthony Minessale [anthony.minessale at gmail.com]
Skickat: den 14 mars 2011 17:56
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Memory leak

its also Mac specific and reported to sofia devs.


On Mon, Mar 14, 2011 at 11:38 AM, Mario G <mario_fs at mgtech.com> wrote:
> Been updating gits since last year and even today. This is a known open
> problem in Jira. It is related to internal registrations, the more
> phone/faster frequency the faster the leak. It is low priority right now
> since I am having other major FS problems due to a router change. Thanks for
> asking!
> Mario G
>
> On Mar 14, 2011, at 9:30 AM, Steven Ayre wrote:
>
> Have you tried upgrading to the latest Git?
> -Steve
>
> On 14 March 2011 16:18, Mario G <mario_fs at mgtech.com> wrote:
>>
>> This may not be related but I have had a memory leak since last year that
>> requires a restart every night. About 2M an hour on my OS X system.
>> On Mar 14, 2011, at 8:27 AM, Steven Ayre wrote:
>>
>> Try using the freeswitch.Dbh interface:
>> http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh
>> It gives you a ODBC interface through freeswitch's odbc support, plus you
>> get the benefit of the connection caching.
>> -Steve
>>
>>
>> On 14 March 2011 14:28, Pete Kelly <pkelly at gmail.com> wrote:
>>>
>>>
>>> On 14 March 2011 13:16, Steven Ayre <steveayre at gmail.com> wrote:
>>>>
>>>> If you use mod_lcr, we were getting a memory leak in that a few weeks
>>>> ago which is fixed in the latest git.
>>>
>>> No we are not. I am going to try strengthening up the code which closes
>>> the DB connections. If that fails then I will compile and try the ODBC
>>> driver.
>>>
>>>>
>>>> -Steve
>>>>
>>>> On 14 March 2011 09:30, Pete Kelly <pkelly at gmail.com> wrote:
>>>>>
>>>>> Hi
>>>>> We are using Freeswitch (from git sources) in a production environment
>>>>> to handle an IVR system using a simple set of dialplans and lua scripts.
>>>>> OpenSIPS 1.6 and mysql are also installed and running as part of the same
>>>>> IVR application.
>>>>> However over time (approx 1 week) the memory usage of Freeswitch
>>>>> increases, eventually creeping into swap space forcing us to restart
>>>>> Freeswitch. The IVR handles approx 40 concurrent calls constantly, and we
>>>>> have instances of the IVR running on a 32bit Etch and 64bit Lenny boxes -
>>>>> both exhibit the same issue.
>>>>> We have also tried the Sangoma freeswitch branch in case that contained
>>>>> any fixes, however we see the same issues.
>>>>> Can anybody advise if this is a known issue with Freeswitch? Is it a
>>>>> memory leak or is it possible to limit the amount of memory Freeswitch will
>>>>> use, forcing it to garbage collect when it reaches the limit?
>>>>> Any advice on where to look/debug next would be appreciated.
>>>>> Thanks
>>>>> Pete
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4d7e4a9b32767617011118!



From anthony.minessale at gmail.com  Mon Mar 14 21:00:49 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 14 Mar 2011 13:00:49 -0500
Subject: [Freeswitch-users] Memory leak
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B2C49521@cooper>
References: <AANLkTim3UsX4z_URe1Ya40uP4AnVG3s3NiYkew=ZY8+O@mail.gmail.com>
	<AANLkTincGerDeJtkyqzYXFmtTzjHe-h9RMZJsM5HosCS@mail.gmail.com>
	<AANLkTimLNa73o5Y_h6KUpTptNyX6Q-6bpBOQBjLtsnS6@mail.gmail.com>
	<AANLkTinxMmO8UCR6wjROjB=DoD_yUoe2EuBNV=rNb3AC@mail.gmail.com>
	<188EF393-51D5-4740-9572-573F8DA04157@mgtech.com>
	<AANLkTikH5gQqvBeLVW8i3GBVwwqVW8vz6kS2ny=x2pNy@mail.gmail.com>
	<9F22E106-FBD9-47ED-A9F8-894C9A2ACDC6@mgtech.com>
	<AANLkTi=jHhuxW8zbFp94Xz9bb-hJAna29ofPwwerkXKZ@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2C49521@cooper>
Message-ID: <AANLkTinuH3qQxbs8LWFrV+70O_DS5zvOiFPEzy3qsdca@mail.gmail.com>

yes,

and the release notes are a who's who of FreeSWITCH contributors and
FS JIRA bug references.

http://sofia-sip.sourceforge.net/relnotes/relnotes-sofia-sip-1.12.11.txt


On Mon, Mar 14, 2011 at 12:35 PM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> By the way - I just noticed that the Sofia devs released version 1.12.11 about a week ago, but I guess you already knew that :)
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Anthony Minessale [anthony.minessale at gmail.com]
> Skickat: den 14 mars 2011 17:56
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] Memory leak
>
> its also Mac specific and reported to sofia devs.
>
>
> On Mon, Mar 14, 2011 at 11:38 AM, Mario G <mario_fs at mgtech.com> wrote:
>> Been updating gits since last year and even today. This is a known open
>> problem in Jira. It is related to internal registrations, the more
>> phone/faster frequency the faster the leak. It is low priority right now
>> since I am having other major FS problems due to a router change. Thanks for
>> asking!
>> Mario G
>>
>> On Mar 14, 2011, at 9:30 AM, Steven Ayre wrote:
>>
>> Have you tried upgrading to the latest Git?
>> -Steve
>>
>> On 14 March 2011 16:18, Mario G <mario_fs at mgtech.com> wrote:
>>>
>>> This may not be related but I have had a memory leak since last year that
>>> requires a restart every night. About 2M an hour on my OS X system.
>>> On Mar 14, 2011, at 8:27 AM, Steven Ayre wrote:
>>>
>>> Try using the freeswitch.Dbh interface:
>>> http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh
>>> It gives you a ODBC interface through freeswitch's odbc support, plus you
>>> get the benefit of the connection caching.
>>> -Steve
>>>
>>>
>>> On 14 March 2011 14:28, Pete Kelly <pkelly at gmail.com> wrote:
>>>>
>>>>
>>>> On 14 March 2011 13:16, Steven Ayre <steveayre at gmail.com> wrote:
>>>>>
>>>>> If you use mod_lcr, we were getting a memory leak in that a few weeks
>>>>> ago which is fixed in the latest git.
>>>>
>>>> No we are not. I am going to try strengthening up the code which closes
>>>> the DB connections. If that fails then I will compile and try the ODBC
>>>> driver.
>>>>
>>>>>
>>>>> -Steve
>>>>>
>>>>> On 14 March 2011 09:30, Pete Kelly <pkelly at gmail.com> wrote:
>>>>>>
>>>>>> Hi
>>>>>> We are using Freeswitch (from git sources) in a production environment
>>>>>> to handle an IVR system using a simple set of dialplans and lua scripts.
>>>>>> OpenSIPS 1.6 and mysql are also installed and running as part of the same
>>>>>> IVR application.
>>>>>> However over time (approx 1 week) the memory usage of Freeswitch
>>>>>> increases, eventually creeping into swap space forcing us to restart
>>>>>> Freeswitch. The IVR handles approx 40 concurrent calls constantly, and we
>>>>>> have instances of the IVR running on a 32bit Etch and 64bit Lenny boxes -
>>>>>> both exhibit the same issue.
>>>>>> We have also tried the Sangoma freeswitch branch in case that contained
>>>>>> any fixes, however we see the same issues.
>>>>>> Can anybody advise if this is a known issue with Freeswitch? Is it a
>>>>>> memory leak or is it possible to limit the amount of memory Freeswitch will
>>>>>> use, forcing it to garbage collect when it reaches the limit?
>>>>>> Any advice on where to look/debug next would be appreciated.
>>>>>> Thanks
>>>>>> Pete
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4d7e4a9b32767617011118!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 mario_fs at mgtech.com  Mon Mar 14 21:08:56 2011
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 14 Mar 2011 11:08:56 -0700
Subject: [Freeswitch-users] How to stop SIP auth challenge (REGISTER) msgs?
Message-ID: <6DB632C2-BC5B-4092-BA67-626ED8729DCD@mgtech.com>

I see this was introduced in a recent git. I know they are not an error but does anyone know if there is a way to stop these? Something I need to change on the phones? These are really a problem while trying to working on problems. The log is filled with these. Thanks.

WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [200 at 100.200.1.7] from ip 100.200.1.12



From fernando.berretta at gmail.com  Mon Mar 14 21:17:05 2011
From: fernando.berretta at gmail.com (Fernando Berretta)
Date: Mon, 14 Mar 2011 15:17:05 -0300
Subject: [Freeswitch-users] FreeSwitch for Windows
In-Reply-To: <AANLkTi=QOQfvNbRd3QA8N407jU9hVDF9M03V7LXZG14A@mail.gmail.com>
References: <134839.19366.qm@web33506.mail.mud.yahoo.com>	<4D7E05B7.9040403@gmail.com>
	<AANLkTi=QOQfvNbRd3QA8N407jU9hVDF9M03V7LXZG14A@mail.gmail.com>
Message-ID: <4D7E5BA1.5050006@gmail.com>

Moises,

Thanks for your answer, it is very interesting, I'm gonna try it ASAP.

Best Regards,
Fernando

On 3/14/2011 1:39 PM, Moises Silva wrote:
> MFC-R2 via the openr2 stack works very well in Windows with Sangoma 
> cards.
>
> There is no formal release yet though (should be ready in about a 
> month) but the code is not changing any more, the only reason to not 
> have a release is that we want a Windows installer for the openr2 
> stack. You need to build the stack yourself using CMake for Windows 
> and Visual Studio in the meantime.
>
> 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 <mailto:moy at sangoma.com>
>
>
> On Mon, Mar 14, 2011 at 8:10 AM, Fernando Berretta 
> <fernando.berretta at gmail.com <mailto:fernando.berretta at gmail.com>> wrote:
>
>     Diego,
>
>     Thanks for your answer
>
>     It is possible to use also boards like Digium, OpenVox or any
>     other similar one ? What about E1 cards ? have you tried one of
>     them over Windows ? MFC R2 ?
>
>     Best Regards,
>     Fernando
>
>     On 3/12/2011 11:35 AM, Diego Toro wrote:
>>     I don't think like Mr. Colling, I have FS on Windows in
>>     production since two year using sangoma cards (FXO, FXS) and
>>     works fine.  Windows is a good choice.
>>
>>     Diego Toro
>>     http://voipensando.blogspot.com/
>>
>>     --- On *Fri, 3/11/11, Fernando Berretta
>>     /<fernando.berretta at gmail.com>
>>     <mailto:fernando.berretta at gmail.com>/* wrote:
>>
>>
>>         From: Fernando Berretta <fernando.berretta at gmail.com>
>>         <mailto:fernando.berretta at gmail.com>
>>         Subject: Re: [Freeswitch-users] FreeSwitch for Windows
>>         To: "FreeSWITCH Users Help"
>>         <freeswitch-users at lists.freeswitch.org>
>>         <mailto:freeswitch-users at lists.freeswitch.org>
>>         Date: Friday, March 11, 2011, 4:54 PM
>>
>>         Michael,
>>
>>         Thanks for your valuable recommendation
>>
>>         Best Regards,
>>         Fernando
>>
>>         On 3/11/2011 6:41 PM, Michael Collins wrote:
>>>         If you plan on using TDM then I'd recommend you stay away
>>>         from Windows. Sangoma and Digium/clone cards are all geared
>>>         toward a Linux environment. Stay away from the exotic OSes
>>>         and use something simple like CentOS 5.5 or Debian Lenny.
>>>
>>>         -MC
>>>
>>>
>>>         On Wed, Mar 9, 2011 at 1:38 PM, Fernando Berretta
>>>         <fernando.berretta at gmail.com
>>>         <http://mc/compose?to=fernando.berretta at gmail.com>> wrote:
>>>
>>>             Hi,
>>>
>>>             I'm newbie in FreeSwitch and would like to know if
>>>             freeswitch could be
>>>             used for production under Windows platforms ? telephony
>>>             boards like
>>>             Digium / OpenVox, Sangoma, etc are going to work ok
>>>             under Windows ? is
>>>             there any problem / limitation ? E1 MFCR2 Trunk works ?
>>>
>>>             Thanks for your time,
>>>             Best Regards,
>>>             Fernando
>>>
>>>             _______________________________________________
>>>             FreeSWITCH-users mailing list
>>>             FreeSWITCH-users at lists.freeswitch.org
>>>             <http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>>>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>             http://www.freeswitch.org
>>>
>>>
>>>
>>>         _______________________________________________
>>>         FreeSWITCH-users mailing list
>>>         FreeSWITCH-users at lists.freeswitch.org  <http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         http://www.freeswitch.org
>>
>>         -----Inline Attachment Follows-----
>>
>>
>>         _______________________________________________
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
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From acrow at integrafin.co.uk  Mon Mar 14 21:28:00 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Mon, 14 Mar 2011 18:28:00 +0000
Subject: [Freeswitch-users] api_hangup_hook documentation correction
	required?
Message-ID: <4D7E5E30.9020404@integrafin.co.uk>

Anthony/all,

I found while getting fax working that it seems that if api_hangup_hook 
is set /before/ answer in the following

<extension name="Fax" >
<condition field="destination_number" expression="^620$">
<action application="answer"/>
<action application="playback" data="silence_stream://2000"/>
<action application="set" data="api_hangup_hook=system /usr/bin/php 
/var/www/fusionpbx/secure/fax_to_email.php email=acrow at integrafin.co.uk 
extension=620 name=\\\${last_fax} messages="/>
<action application="set" 
data="last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}"/>
<action application="rxfax" 
data="/opt/freeswitch/storage/fax/620/inbox/${last_fax}.tif"/>
<action application="hangup"/>
</condition>
</extension>

That rxfax strangely fails - the calling fax machine never gets the tone 
to start sending and T4 is retried until rxfax gives up. If it's placed 
as above it works fine.

Not sure if this is a bug or intended and the docs on the Wiki need 
changing.

Thanks

Alex


-- 
This message is intended only for the addressee and may contain
confidential information.  Unless you are that person, you may not
disclose its contents or use it in any way and are requested to delete
the message along with any attachments and notify us immediately.

"Transact" is operated by Integrated Financial Arrangements plc
Domain House, 5-7 Singer Street, London  EC2A 4BQ
Tel: (020) 7608 4900 Fax: (020) 7608 5300
(Registered office: as above; Registered in England and Wales under number: 3727592)
Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)



From infos at madovsky.org  Mon Mar 14 21:42:40 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 14 Mar 2011 14:42:40 -0400
Subject: [Freeswitch-users] api_hangup_hook documentation
	correctionrequired?
References: <4D7E5E30.9020404@integrafin.co.uk>
Message-ID: <04A85C5FFFEE467E8247C9663221C718@e1705>

usually I think it's better to set this kind of vars before answer....

----- Original Message ----- 
From: "Alex Crow" <acrow at integrafin.co.uk>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, March 14, 2011 2:28 PM
Subject: [Freeswitch-users] api_hangup_hook documentation 
correctionrequired?


> Anthony/all,
>
> I found while getting fax working that it seems that if api_hangup_hook
> is set /before/ answer in the following
>
> <extension name="Fax" >
> <condition field="destination_number" expression="^620$">
> <action application="answer"/>
> <action application="playback" data="silence_stream://2000"/>
> <action application="set" data="api_hangup_hook=system /usr/bin/php
> /var/www/fusionpbx/secure/fax_to_email.php email=acrow at integrafin.co.uk
> extension=620 name=\\\${last_fax} messages="/>
> <action application="set"
> data="last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}"/>
> <action application="rxfax"
> data="/opt/freeswitch/storage/fax/620/inbox/${last_fax}.tif"/>
> <action application="hangup"/>
> </condition>
> </extension>
>
> That rxfax strangely fails - the calling fax machine never gets the tone
> to start sending and T4 is retried until rxfax gives up. If it's placed
> as above it works fine.
>
> Not sure if this is a bug or intended and the docs on the Wiki need
> changing.
>
> Thanks
>
> Alex
>
>
> -- 
> This message is intended only for the addressee and may contain
> confidential information.  Unless you are that person, you may not
> disclose its contents or use it in any way and are requested to delete
> the message along with any attachments and notify us immediately.
>
> "Transact" is operated by Integrated Financial Arrangements plc
> Domain House, 5-7 Singer Street, London  EC2A 4BQ
> Tel: (020) 7608 4900 Fax: (020) 7608 5300
> (Registered office: as above; Registered in England and Wales under 
> number: 3727592)
> Authorised and regulated by the Financial Services Authority (entered on 
> the FSA Register; number: 190856)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From infos at madovsky.org  Mon Mar 14 22:03:17 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 14 Mar 2011 15:03:17 -0400
Subject: [Freeswitch-users] debug down trunk
Message-ID: <60BA130CDBF44BD69A43BA7777AE83BD@e1705>

I set external.xml to debug=1 and sip trace to 1 and level debug to 7
but I can get only this in the log

sofia.c:4028 Ping failed trunk_3 with code 408 - count -1/-1/1, state DOWN

so I don't know why this trunk is down.
I contacted them and they said that my account is working.

Thanks
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From anthony.minessale at gmail.com  Mon Mar 14 22:24:04 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 14 Mar 2011 14:24:04 -0500
Subject: [Freeswitch-users] PA and ALSA
In-Reply-To: <AANLkTim81nXTRSF3SZUME2fzW+zDBqdrVOD7M1R4R-Y4@mail.gmail.com>
References: <4D7BF5E6.30201@rosengart.de>
	<201103131921.18213.sos@sokhapkin.dyndns.org>
	<AANLkTikYUsbU6=gd=SrRd_a=F=j40eV=D0YW4SLCNuQU@mail.gmail.com>
	<201103132032.19044.sos@sokhapkin.dyndns.org>
	<AANLkTim81nXTRSF3SZUME2fzW+zDBqdrVOD7M1R4R-Y4@mail.gmail.com>
Message-ID: <AANLkTi=G7ZAK9w7mM_0S0eKaKRyCgEOW5d-ocYhBtwLu@mail.gmail.com>

oh and a another thing,

Many sound cards can't do 8k, just set it to something higher like 16
32 or 48 and it will downsample it in software to 8k.
I will look at latest v19 snapshot.


On Mon, Mar 14, 2011 at 12:02 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Anything is possible if we get patches to make it configurable and our
> preferences are the default.
>
>
> On Sun, Mar 13, 2011 at 7:32 PM, Sergey Okhapkin
> <sos at sokhapkin.dyndns.org> wrote:
>> The power of gentoo linux is that ebuild can specify a range of required
>> versions for a third party library. Maybe something can be implemented in FS
>> build script too? Just check a version of the installed third party library in
>> configure script.
>>
>> On Sunday 13 March 2011, Anthony Minessale wrote:
>>> 2 things to keep in mind.
>>>
>>> 1 Try to wrap your head around making a stable application that runs on
>>> windows, unix including mac and some bsd, and unix-like distros such as
>>> those mentioned. ?Not all of our dependancies are stable on all of those
>>> platforms or even easily available in some cases.
>>>
>>> 2 This is a community project, if some versions of libraries we are using
>>> are older maybe they have been working fine or the modules are not used
>>> enough to warrant attention.
>>>
>>> The normal process would be:
>>>
>>> The guy reports a problem and suggests its fixed in a newer version. ?Now
>>> try this code on every supported platform and make sure it works across the
>>> board, if not, well, we can't blindly add it. ?All of this can be done
>>> without the ungrateful cracks and side stories about gentoo and we can
>>> happily update the code.
>>>
>>> I don't go to gentoo forum and complain that custom patching the kernel and
>>> libc annoys me or about how I am not interested in being forced into their
>>> religion to have my software in their distro even though it does not
>>> install any third party libs, FS only static links in bits of code from in
>>> tree depends.
>>>
>>> Furthermore, we constantly invite people to assist with platform and
>>> packaging and nobody offers anything but complaints.
>>>
>>> An example of why we don't just blindly use system libs:
>>> The latest sqlite causes 25% loss in performance and guarenteed segfaults
>>> and occasional deadlock under load.
>>> This is a place where people complain about 1% cpu usage.
>>>
>>> Alsa mod is specifically for the nokia N800 N810 devices.
>>> Stick with pa or help us write something new from scratch and don't forget
>>> linux is not alone in the computing world.
>>>
>>> On Mar 13, 2011 6:22 PM, "Sergey Okhapkin" <sos at sokhapkin.dyndns.org> wrote:
>>> > The attempt to isolate core FS code from third party libraries has been
>>>
>>> done
>>>
>>> > by gentoo linux team already (inclusion of third party open source code
>>> > is strictly against gentoo philosophy), but the attempt has been
>>> > abandoned because of lack of support from upstream.
>>> >
>>> > On Sunday 13 March 2011 19:05:17 Jeff Lenk wrote:
>>> > > I'm sure there were very good reasons for these decisions when they
>>> > > were made. So rather than criticizing those decisions why not help by
>>> > > finding the problems and submitting patches to fix them.
>>> > >
>>> > > --
>>>
>>> > > View this message in context:
>>> http://freeswitch-users.2379917.n2.nabble.com/PA-and-ALSA-tp6165168p616736
>>>
>>> > > 8.html Sent from the freeswitch-users mailing list archive at
>>>
>>> Nabble.com.
>>>
>>> > > _______________________________________________
>>> > > FreeSWITCH-users mailing list
>>> > > FreeSWITCH-users at lists.freeswitch.org
>>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
>>> > > s http://www.freeswitch.org
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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 cjbujold at accra.ca  Mon Mar 14 22:26:13 2011
From: cjbujold at accra.ca (Charles Bujold)
Date: Mon, 14 Mar 2011 16:26:13 -0300
Subject: [Freeswitch-users] How to route  ata question
Message-ID: <00f801cbe27d$ae479fb0$0ad6df10$@accra.ca>

 

My question is I have an ATA FXO (Grandstream ht503) that currently sends
calls to an extension, I want the calls to be filtered by the other
Freeswitch settings and I am not certain how to proceed.

 

I originally created a Main IVR (ext 5555) and sent the incoming FXO ATA
calls to the IVR extension.   Now I want to send the calls to some other
extension or inbound route so that the calls are routed through all the
Freeswitch  settings such as holidays and time of day  conditions before
being sent to the IVR or some after hours message.  I'm not certain how I
should setup the FXO ATA to send the calls so Freeswitch processes them
properly?    Please advise.

 

cjb

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From anthony.minessale at gmail.com  Mon Mar 14 22:27:42 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 14 Mar 2011 14:27:42 -0500
Subject: [Freeswitch-users] PA and ALSA
In-Reply-To: <AANLkTi=G7ZAK9w7mM_0S0eKaKRyCgEOW5d-ocYhBtwLu@mail.gmail.com>
References: <4D7BF5E6.30201@rosengart.de>
	<201103131921.18213.sos@sokhapkin.dyndns.org>
	<AANLkTikYUsbU6=gd=SrRd_a=F=j40eV=D0YW4SLCNuQU@mail.gmail.com>
	<201103132032.19044.sos@sokhapkin.dyndns.org>
	<AANLkTim81nXTRSF3SZUME2fzW+zDBqdrVOD7M1R4R-Y4@mail.gmail.com>
	<AANLkTi=G7ZAK9w7mM_0S0eKaKRyCgEOW5d-ocYhBtwLu@mail.gmail.com>
Message-ID: <AANLkTik3Xk7ojMNxF6MGNL5Ae9v9Ls1HueXX2t-nnVNs@mail.gmail.com>

I am pretty sure this is the same one we are already using.


On Mon, Mar 14, 2011 at 2:24 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> oh and a another thing,
>
> Many sound cards can't do 8k, just set it to something higher like 16
> 32 or 48 and it will downsample it in software to 8k.
> I will look at latest v19 snapshot.
>
>
> On Mon, Mar 14, 2011 at 12:02 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> Anything is possible if we get patches to make it configurable and our
>> preferences are the default.
>>
>>
>> On Sun, Mar 13, 2011 at 7:32 PM, Sergey Okhapkin
>> <sos at sokhapkin.dyndns.org> wrote:
>>> The power of gentoo linux is that ebuild can specify a range of required
>>> versions for a third party library. Maybe something can be implemented in FS
>>> build script too? Just check a version of the installed third party library in
>>> configure script.
>>>
>>> On Sunday 13 March 2011, Anthony Minessale wrote:
>>>> 2 things to keep in mind.
>>>>
>>>> 1 Try to wrap your head around making a stable application that runs on
>>>> windows, unix including mac and some bsd, and unix-like distros such as
>>>> those mentioned. ?Not all of our dependancies are stable on all of those
>>>> platforms or even easily available in some cases.
>>>>
>>>> 2 This is a community project, if some versions of libraries we are using
>>>> are older maybe they have been working fine or the modules are not used
>>>> enough to warrant attention.
>>>>
>>>> The normal process would be:
>>>>
>>>> The guy reports a problem and suggests its fixed in a newer version. ?Now
>>>> try this code on every supported platform and make sure it works across the
>>>> board, if not, well, we can't blindly add it. ?All of this can be done
>>>> without the ungrateful cracks and side stories about gentoo and we can
>>>> happily update the code.
>>>>
>>>> I don't go to gentoo forum and complain that custom patching the kernel and
>>>> libc annoys me or about how I am not interested in being forced into their
>>>> religion to have my software in their distro even though it does not
>>>> install any third party libs, FS only static links in bits of code from in
>>>> tree depends.
>>>>
>>>> Furthermore, we constantly invite people to assist with platform and
>>>> packaging and nobody offers anything but complaints.
>>>>
>>>> An example of why we don't just blindly use system libs:
>>>> The latest sqlite causes 25% loss in performance and guarenteed segfaults
>>>> and occasional deadlock under load.
>>>> This is a place where people complain about 1% cpu usage.
>>>>
>>>> Alsa mod is specifically for the nokia N800 N810 devices.
>>>> Stick with pa or help us write something new from scratch and don't forget
>>>> linux is not alone in the computing world.
>>>>
>>>> On Mar 13, 2011 6:22 PM, "Sergey Okhapkin" <sos at sokhapkin.dyndns.org> wrote:
>>>> > The attempt to isolate core FS code from third party libraries has been
>>>>
>>>> done
>>>>
>>>> > by gentoo linux team already (inclusion of third party open source code
>>>> > is strictly against gentoo philosophy), but the attempt has been
>>>> > abandoned because of lack of support from upstream.
>>>> >
>>>> > On Sunday 13 March 2011 19:05:17 Jeff Lenk wrote:
>>>> > > I'm sure there were very good reasons for these decisions when they
>>>> > > were made. So rather than criticizing those decisions why not help by
>>>> > > finding the problems and submitting patches to fix them.
>>>> > >
>>>> > > --
>>>>
>>>> > > View this message in context:
>>>> http://freeswitch-users.2379917.n2.nabble.com/PA-and-ALSA-tp6165168p616736
>>>>
>>>> > > 8.html Sent from the freeswitch-users mailing list archive at
>>>>
>>>> Nabble.com.
>>>>
>>>> > > _______________________________________________
>>>> > > FreeSWITCH-users mailing list
>>>> > > FreeSWITCH-users at lists.freeswitch.org
>>>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
>>>> > > s http://www.freeswitch.org
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> 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
>



-- 
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 acrow at integrafin.co.uk  Mon Mar 14 22:25:06 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Mon, 14 Mar 2011 19:25:06 +0000
Subject: [Freeswitch-users] api_hangup_hook
	documentation	correctionrequired?
In-Reply-To: <04A85C5FFFEE467E8247C9663221C718@e1705>
References: <4D7E5E30.9020404@integrafin.co.uk>
	<04A85C5FFFEE467E8247C9663221C718@e1705>
Message-ID: <4D7E6B92.2090702@integrafin.co.uk>

Forgive me, I just realised it was actually a permission error on a 
folder that I fixed before moving the "set"!

No-one is ever going to answer me on this list again given the last few 
days of my posts!

Thanks very much anyway!

Alex

On 14/03/11 18:42, Madovsky wrote:
> usually I think it's better to set this kind of vars before answer....
>
> ----- Original Message -----
> From: "Alex Crow"<acrow at integrafin.co.uk>
> To: "FreeSWITCH Users Help"<freeswitch-users at lists.freeswitch.org>
> Sent: Monday, March 14, 2011 2:28 PM
> Subject: [Freeswitch-users] api_hangup_hook documentation
> correctionrequired?
>
>
>> Anthony/all,
>>
>> I found while getting fax working that it seems that if api_hangup_hook
>> is set /before/ answer in the following
>>
>> <extension name="Fax">
>> <condition field="destination_number" expression="^620$">
>> <action application="answer"/>
>> <action application="playback" data="silence_stream://2000"/>
>> <action application="set" data="api_hangup_hook=system /usr/bin/php
>> /var/www/fusionpbx/secure/fax_to_email.php email=acrow at integrafin.co.uk
>> extension=620 name=\\\${last_fax} messages="/>
>> <action application="set"
>> data="last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}"/>
>> <action application="rxfax"
>> data="/opt/freeswitch/storage/fax/620/inbox/${last_fax}.tif"/>
>> <action application="hangup"/>
>> </condition>
>> </extension>
>>
>> That rxfax strangely fails - the calling fax machine never gets the tone
>> to start sending and T4 is retried until rxfax gives up. If it's placed
>> as above it works fine.
>>
>> Not sure if this is a bug or intended and the docs on the Wiki need
>> changing.
>>
>> Thanks
>>
>> Alex
>>
>>
>> -- 
>> This message is intended only for the addressee and may contain
>> confidential information.  Unless you are that person, you may not
>> disclose its contents or use it in any way and are requested to delete
>> the message along with any attachments and notify us immediately.
>>
>> "Transact" is operated by Integrated Financial Arrangements plc
>> Domain House, 5-7 Singer Street, London  EC2A 4BQ
>> Tel: (020) 7608 4900 Fax: (020) 7608 5300
>> (Registered office: as above; Registered in England and Wales under
>> number: 3727592)
>> Authorised and regulated by the Financial Services Authority (entered on
>> the FSA Register; number: 190856)
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


-- 
This message is intended only for the addressee and may contain
confidential information.  Unless you are that person, you may not
disclose its contents or use it in any way and are requested to delete
the message along with any attachments and notify us immediately.

"Transact" is operated by Integrated Financial Arrangements plc
Domain House, 5-7 Singer Street, London  EC2A 4BQ
Tel: (020) 7608 4900 Fax: (020) 7608 5300
(Registered office: as above; Registered in England and Wales under number: 3727592)
Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)



From steveayre at gmail.com  Mon Mar 14 23:05:16 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 14 Mar 2011 20:05:16 +0000
Subject: [Freeswitch-users] Socket stops receiving INVITE packets
Message-ID: <AANLkTimyiAFTU_8+xjDnV6FnttbB1rtEdG4YnG9YK23_@mail.gmail.com>

Hi everyone,

I've got a strange problem on my servers which has happened 3 times in the
last month now...

I have FS listening on 2 IP addresses, bound to bond0 and bond0:0. FS
suddently stops processing any calls arriving on one of those IPs.

tcpdump/tshark still shows INVITE packets arriving on my server, but FS
doesn't process them and there are no 100 Trying etc replies.

Initially FreeSWITCH still owns the socket, the only odd thing is Send-Q is
744:
    ss1:~# netstat -anp | head -n2
    Active Internet connections (servers and established)
    Proto Recv-Q Send-Q Local Address           Foreign Address
State       PID/Program name
    ss1:~# netstat -anp  | grep 5060
    tcp        0      0 81.27.101.252:5060      0.0.0.0:*
LISTEN      27587/freeswitch
    tcp        0      0 81.27.101.233:5060      0.0.0.0:*
LISTEN      27587/freeswitch
    udp        0    744 81.27.101.252:5060      0.0.0.0:*
27587/freeswitch
    udp        0      0 81.27.101.233:5060      0.0.0.0:*
27587/freeswitch

I stop freeswitch so I can restart it:
ss1:~# killall -9 freeswitch

Freeswitch is no longer running (no zombies etc):
ss1:~# ps faux | grep freeswitch
root     20723  0.0  0.0   5164   788 pts/3    S+   19:56   0:00          \_
grep --exclude-dir=.svn --color freeswitch

But the port is still open!
ss1:~# netstat -anp | grep :5060
udp        0    744 81.27.101.252:5060      0.0.0.0:*
-

This means I can't restart FreeSWITCH because the port is already in use! I
have to reboot to free the port up.


Has anyone seen this before? Does anyone know what could cause this?


Since it's still in use after FS has stopped, I'm thinking a Linux kernel
problem. I'm running Debian Lenny with kernel 2.6.26-2-amd64. The hardware
is a NEC server with 2x E5405 xeon (8cores total) and 8gb ram. It's happened
on 2 servers, so I'm thinking something in software. I updated kernel a
couple of months back so perhaps that's related, before that everything was
fine for 2 years.


Thanks everyone...

-Steve
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From msc at freeswitch.org  Mon Mar 14 23:10:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 14 Mar 2011 13:10:38 -0700
Subject: [Freeswitch-users] Problem with mapping new DIDs
In-Reply-To: <AANLkTimJY4OYsGx_znMy-pUUA19oqXT_-qxf=adygLx7@mail.gmail.com>
References: <AANLkTimJY4OYsGx_znMy-pUUA19oqXT_-qxf=adygLx7@mail.gmail.com>
Message-ID: <AANLkTinfrSYkmZYWAB9_p5xG46cEy8Ln4ttJ2uJ7nbjp@mail.gmail.com>

Sorry for the late reply...
If you have not already figured this one out then I recommend that you
pastebin the debug log of the call coming in. The debug log shows the
processing of the dialplan and will show each extension's patterns trying to
match against the values specified. That should shed some light on the
subject. Use our pastebin at pastebin.freeswitch.org.

-MC

On Thu, Mar 10, 2011 at 1:25 AM, Prime <29prime at gmail.com> wrote:

> Hey!
>
> This most likely is a newb mistake but I'm stuck at the moment.
>
> Recently I had to add another provider to FS to handle incoming calls.
> And now I have a problem with calling my new provider's DID number on
> my FS.
> I looked at FS logs.
> Usually FS log says something like this " [INFO]
> mod_dialplan_xml.c:418 Processing caller_number->My_DID in context
> public" and then routes DID number to local extension,
> but now I see something like this " [INFO] mod_dialplan_xml.c:418
> Processing caller_number->caller_number in context public ".
> Why new numbers does not match the regex?
>
> The number are configured exactly as old ones.
>
> Thanks in advance
>
> Best regards
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Mar 14 23:24:13 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 14 Mar 2011 13:24:13 -0700
Subject: [Freeswitch-users] [freeswitch] <defunct> when using cmd="exec"
 in freeswitch.xml
In-Reply-To: <4D78A5AF.1000808@ewetel.de>
References: <4D78A5AF.1000808@ewetel.de>
Message-ID: <AANLkTin7EUJpz4=9gxp3sTXykjybH5B4Uer7noD+Bgct@mail.gmail.com>

IIRC, 'fork' creates a child PID that stays around and waits for the parent
PID to die. (Don't quote me on that - wait for an expert to chime in.)

An alternative method would be to have a FS start script that launches your
make_my_vars.sh script and then launches FS with the appropriate cmd line
args. You could then drop the 'exec' cmd and then just keep the include
cmd.

-MC

On Thu, Mar 10, 2011 at 2:19 AM, Mitja Thomas <mitja.thomas1 at ewetel.de>wrote:

>  Hi there,
>
> we tried to set up the FreeSWITCH and other Applications, so that we can
> configure them easier and more centralised.
> Thus we defined some Environment Variables (using CentOS) which hold often
> used Configuration Parameter like MySQL IP or FS Event Socket IP.
> We tried to integrate these Env Variables into the FS conf files by
> executing a shell Skript in freeswitch.xml via cmd="exec" which prepares an
> conf file which we include into freeswitch.xml:
>
>   <X-PRE-PROCESS cmd="exec" data="sh conf/make_my_vars.sh"/>
>   <X-PRE-PROCESS cmd="include" data="my_vars.xml"/>
>
> This works as expected and the pre defined variables in my_vars can be
> accessed from the other config Files, except that when we start our
> FreeSWITCH  a zombie child process is spawned.
>
> # ps -eaf | grep free
> ippbx    22191 22190  4 09:41 pts/1    00:00:01
> /opt/app/voip/ippbx/bin/freeswitch -waste -nonat -hp
> ippbx    22197 22191  0 09:41 pts/1    00:00:00 [freeswitch] <defunct>
>
> What I wanna know is: Is this a FS missbehaviour or do we use this in a
> wrong way?
>
> make_my_vars.sh:
> F="conf/my_vars.xml"
> echo "<!-- Auto generated file for getting *NIX evironment variables -->" >
> $F
> echo "<include>" >> $F
>
> fs_ip=`printenv MY_FS_IP`
> if test -n "$fs_ip"
> then
>         echo '<X-PRE-PROCESS cmd="set" data="my_fs_ip='$fs_ip'"/>' >> $F
> fi
> ...
> echo "</include>" >> $F
>
> my_vars.xml (after FS startup):
> <!-- Auto generated file for getting *NIX evironment variables -->
> <include>
> <X-PRE-PROCESS cmd="set" data="my_fs_ip=***.***.***.***"/>
> ...
> </include>
>
> Regards
> Mitja
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Mar 14 23:32:05 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 14 Mar 2011 13:32:05 -0700
Subject: [Freeswitch-users] multiple gateways and NAT
In-Reply-To: <37078331-559B-4A55-A5DC-9E8A26E23B0E@mgtech.com>
References: <4D7D0A6F.2020508@julf.com>
	<EE2DD131C4C543B3A0FDF5BBE7196CE9@e1705>
	<4D7D18C5.2040404@julf.com>
	<D81653003224456281F83BFA528134C4@e1705>
	<4D7D1FA1.5050600@julf.com>
	<913499CF71FC4CA788510BA78E391891@e1705>
	<C7F29C81-A205-426F-B296-47FA849EF4C3@mgtech.com>
	<4D7DD9D4.3000002@julf.com>
	<37078331-559B-4A55-A5DC-9E8A26E23B0E@mgtech.com>
Message-ID: <AANLkTimfkN1yx9r-3akWbbWiJV+yJOv=5pF4QyZjv4re@mail.gmail.com>

On Mon, Mar 14, 2011 at 10:07 AM, Mario G <mario_fs at mgtech.com> wrote:

> I'll tell you what I know/observed: Based on web searches you need address
> and port. Here is an example of what happened to me during testing:
>
> I have multiple user IDs on my internal phone, so each phone registers
> multiple times, one for each extension. When I configured the phone (SPA962)
> to use the same port (5060) for all extensions the phones had problems such
> as not ringing when they should. When I gave each extension its own port and
> problems solved.
>
> As for ITSPs, I remember that I could not get the ITSP that had 2 accounts
> working right until each account had its own port. My guess is that the
> connection between the ITSP is registered with the user ID, but then that ID
> is no longer used, just the IP and port. Since FS had the same ip addr and
> port for both accounts there is no way to know what data belonged to what
> account. Making separate ports fixed the problem.
>
> I also am very confused about FS ports. In vars.xml there is only one port
> that can be setup for internal and external. However, I have separate
> external profiles each with its own port that is required to make things
> work. But.... I only have 1 internal profile and yet all extensions work
> even though they are setup with different ports for each extension. You
> would think FS would require different internal profiles for each ID as it
> does for external. Go figure.... Hope this helps a little.
>

Keep in mind that the ports specified in vars.xml are for the SIP profiles.
Each SIP profile is a user agent. A user agent can service exactly one
IP:Port. So if your internal profile is set with IP of 10.10.1.1 and port
5060, then that's where internal is listening and responding. So, the
internal profile is listening for traffic coming into 10.10.1.1:5060 and
responds from the same IP:port. The phones in the above scenario have their
own user agents who are doing the same thing with their own IPs and ports.
So you could have line key 1 listening/responding on 10.10.1.2:5060 and line
key 2 listening/responding on 10.10.1.2:5062. They both send their SIP
traffic *to* 10.10.1.1:5060 (FS, internal profile) and FS knows which
IP:port to use for responding to traffic "from" each line key. That's an
imperfect explanation, but I hope it illustrates the point.

-MC

>
> As for using port in registrations I am not sure. The reason is while
> testing dual WAN I had a call going on one WAN then pulled the plug. The
> other took over and FS kept the conversation going. So in that case even the
> IP address changed! But... I am having other major FS problems trying to get
> a working backup so had to abandon the FS dual wan setup and stick to 1 wan
> connection for FS.
>
> Mario G
>
>
> On Mar 14, 2011, at 2:03 AM, Johan Helsingius wrote:
>
> >> I remembered why I needed to use different ports in one case: Two
> >> of the three accounts are at the same ITSP, in this case since the
> >> IP address is the same on the two accounts you need to use different
> ports.
> >
> > Ah, I have sort of the same case - 4 accounts with one ITSP providing
> > 4 different incoming DID's (replacing an old ISDN connection), and
> > 2 separate single ITSP accounts for international calls.
> >
> > But I am still very confused about the need for different ports.
> > This is not a complaint, just an observation, but I do think the
> > multiple port thing is one of the more confusing aspects of FS,
> > and something other SIP switches seem to do without.
> >
> > On outgoing connections, we register with the ITSP gateway (by the
> > way, does the SIP registration register both an ip address and
> > a port number, or just the ip address?), and on incoming connections
> > they either have a destination number/DID that routes them to the
> > right place in the dialplan, or they register with us (using
> > username/password) in the case of SIP clients. Security can be
> > handled with ACL's. So I can't get my head around to why separate
> > ports are needed - maybe someone can point out what I am missing?
> >
> >       Julf
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Tue Mar 15 00:04:56 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 14 Mar 2011 21:04:56 +0000
Subject: [Freeswitch-users] How to stop SIP auth challenge (REGISTER)
	msgs?
In-Reply-To: <6DB632C2-BC5B-4092-BA67-626ED8729DCD@mgtech.com>
References: <6DB632C2-BC5B-4092-BA67-626ED8729DCD@mgtech.com>
Message-ID: <AANLkTinCCj1SrDwBD5YFY6-V5_sJGjEU0QOi5SWgmFTw@mail.gmail.com>

Set the SIP profile param log-auth-failures to false. I *think* it's default
(if it's not set) is also false.

The warning level is to make sure they show up in logs at a high log level
so that you can use them with fail2ban.

-Steve


On 14 March 2011 18:08, Mario G <mario_fs at mgtech.com> wrote:

> I see this was introduced in a recent git. I know they are not an error but
> does anyone know if there is a way to stop these? Something I need to change
> on the phones? These are really a problem while trying to working on
> problems. The log is filled with these. Thanks.
>
> WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile
> 'internal' for [200 at 100.200.1.7] from ip 100.200.1.12
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Tue Mar 15 00:06:46 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 14 Mar 2011 21:06:46 +0000
Subject: [Freeswitch-users] debug down trunk
In-Reply-To: <60BA130CDBF44BD69A43BA7777AE83BD@e1705>
References: <60BA130CDBF44BD69A43BA7777AE83BD@e1705>
Message-ID: <AANLkTi=ETiNk4bArqj8VS8nN9j=+LkcwAc9qygN-zkSk@mail.gmail.com>

408 is the error code - this means "408 Request timed out"

FS tries sending an OPTIONS request to the gateway and waits for a reply to
test whether it is online or not..

You can use the siptrace to see those messages and the reply:
sofia global siptrace on

-Steve


On 14 March 2011 19:03, Madovsky <infos at madovsky.org> wrote:

>  I set external.xml to debug=1 and sip trace to 1 and level debug to 7
> but I can get only this in the log
>
> sofia.c:4028 Ping failed trunk_3 with code 408 - count -1/-1/1, state DOWN
>
> so I don't know why this trunk is down.
> I contacted them and they said that my account is working.
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mario_fs at mgtech.com  Tue Mar 15 00:08:27 2011
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 14 Mar 2011 14:08:27 -0700
Subject: [Freeswitch-users] How to stop SIP auth challenge (REGISTER)
	msgs?
In-Reply-To: <AANLkTinCCj1SrDwBD5YFY6-V5_sJGjEU0QOi5SWgmFTw@mail.gmail.com>
References: <6DB632C2-BC5B-4092-BA67-626ED8729DCD@mgtech.com>
	<AANLkTinCCj1SrDwBD5YFY6-V5_sJGjEU0QOi5SWgmFTw@mail.gmail.com>
Message-ID: <4443D92A-C567-49B0-B8B0-6EF516CA44E2@mgtech.com>

Thanks! Will give it a try and let you know.
Mario G

On Mar 14, 2011, at 2:04 PM, Steven Ayre wrote:

> Set the SIP profile param log-auth-failures to false. I *think* it's default (if it's not set) is also false.
> 
> The warning level is to make sure they show up in logs at a high log level so that you can use them with fail2ban.
> 
> -Steve
> 
> 
> On 14 March 2011 18:08, Mario G <mario_fs at mgtech.com> wrote:
> I see this was introduced in a recent git. I know they are not an error but does anyone know if there is a way to stop these? Something I need to change on the phones? These are really a problem while trying to working on problems. The log is filled with these. Thanks.
> 
> WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [200 at 100.200.1.7] from ip 100.200.1.12
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mario_fs at mgtech.com  Tue Mar 15 00:11:47 2011
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 14 Mar 2011 14:11:47 -0700
Subject: [Freeswitch-users] multiple gateways and NAT
In-Reply-To: <AANLkTimfkN1yx9r-3akWbbWiJV+yJOv=5pF4QyZjv4re@mail.gmail.com>
References: <4D7D0A6F.2020508@julf.com>
	<EE2DD131C4C543B3A0FDF5BBE7196CE9@e1705>
	<4D7D18C5.2040404@julf.com>
	<D81653003224456281F83BFA528134C4@e1705>
	<4D7D1FA1.5050600@julf.com>
	<913499CF71FC4CA788510BA78E391891@e1705>
	<C7F29C81-A205-426F-B296-47FA849EF4C3@mgtech.com>
	<4D7DD9D4.3000002@julf.com>
	<37078331-559B-4A55-A5DC-9E8A26E23B0E@mgtech.com>
	<AANLkTimfkN1yx9r-3akWbbWiJV+yJOv=5pF4QyZjv4re@mail.gmail.com>
Message-ID: <B65D66B3-5A7E-4F00-A3CF-766083713720@mgtech.com>

Thanks! That clears it up for me.
Mario G

On Mar 14, 2011, at 1:32 PM, Michael Collins wrote:

> 
> 
> On Mon, Mar 14, 2011 at 10:07 AM, Mario G <mario_fs at mgtech.com> wrote:
> I'll tell you what I know/observed: Based on web searches you need address and port. Here is an example of what happened to me during testing:
> 
> I have multiple user IDs on my internal phone, so each phone registers multiple times, one for each extension. When I configured the phone (SPA962) to use the same port (5060) for all extensions the phones had problems such as not ringing when they should. When I gave each extension its own port and problems solved.
> 
> As for ITSPs, I remember that I could not get the ITSP that had 2 accounts working right until each account had its own port. My guess is that the connection between the ITSP is registered with the user ID, but then that ID is no longer used, just the IP and port. Since FS had the same ip addr and port for both accounts there is no way to know what data belonged to what account. Making separate ports fixed the problem.
> 
> I also am very confused about FS ports. In vars.xml there is only one port that can be setup for internal and external. However, I have separate external profiles each with its own port that is required to make things work. But.... I only have 1 internal profile and yet all extensions work even though they are setup with different ports for each extension. You would think FS would require different internal profiles for each ID as it does for external. Go figure.... Hope this helps a little.
> 
> Keep in mind that the ports specified in vars.xml are for the SIP profiles. Each SIP profile is a user agent. A user agent can service exactly one IP:Port. So if your internal profile is set with IP of 10.10.1.1 and port 5060, then that's where internal is listening and responding. So, the internal profile is listening for traffic coming into 10.10.1.1:5060 and responds from the same IP:port. The phones in the above scenario have their own user agents who are doing the same thing with their own IPs and ports. So you could have line key 1 listening/responding on 10.10.1.2:5060 and line key 2 listening/responding on 10.10.1.2:5062. They both send their SIP traffic *to* 10.10.1.1:5060 (FS, internal profile) and FS knows which IP:port to use for responding to traffic "from" each line key. That's an imperfect explanation, but I hope it illustrates the point.
> 
> -MC
> 
> As for using port in registrations I am not sure. The reason is while testing dual WAN I had a call going on one WAN then pulled the plug. The other took over and FS kept the conversation going. So in that case even the IP address changed! But... I am having other major FS problems trying to get a working backup so had to abandon the FS dual wan setup and stick to 1 wan connection for FS.
> 
> Mario G
> 
> 
> On Mar 14, 2011, at 2:03 AM, Johan Helsingius wrote:
> 
> >> I remembered why I needed to use different ports in one case: Two
> >> of the three accounts are at the same ITSP, in this case since the
> >> IP address is the same on the two accounts you need to use different ports.
> >
> > Ah, I have sort of the same case - 4 accounts with one ITSP providing
> > 4 different incoming DID's (replacing an old ISDN connection), and
> > 2 separate single ITSP accounts for international calls.
> >
> > But I am still very confused about the need for different ports.
> > This is not a complaint, just an observation, but I do think the
> > multiple port thing is one of the more confusing aspects of FS,
> > and something other SIP switches seem to do without.
> >
> > On outgoing connections, we register with the ITSP gateway (by the
> > way, does the SIP registration register both an ip address and
> > a port number, or just the ip address?), and on incoming connections
> > they either have a destination number/DID that routes them to the
> > right place in the dialplan, or they register with us (using
> > username/password) in the case of SIP clients. Security can be
> > handled with ACL's. So I can't get my head around to why separate
> > ports are needed - maybe someone can point out what I am missing?
> >
> >       Julf
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From msc at freeswitch.org  Tue Mar 15 00:11:51 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 14 Mar 2011 14:11:51 -0700
Subject: [Freeswitch-users] Can you use ext-ip.. autonat: with -nonat ?
In-Reply-To: <2DC8F9F6-253E-47A3-9651-D652A911FB80@mgtech.com>
References: <2DC8F9F6-253E-47A3-9651-D652A911FB80@mgtech.com>
Message-ID: <AANLkTinYut4WNGtz2TsPfrYGNLK5tKfRVLw-i0i2852O@mail.gmail.com>

When you do not use -nonat then you *must* supply the external IP addr. The
only thing I can think of is to look at the SIP traces of the failed calls
and compare them to a successful call. See if you can find out what's
different.

-MC

On Fri, Mar 11, 2011 at 1:39 PM, Mario G <mario_fs at mgtech.com> wrote:

> Changed router now only 1 out of 5 calls come in. I previously had
> "ext-sip/rtp-ip:1.2.3.4" and -nonat on another router but having major
> issues on new one. Wiki does not answer this question:
>
> Can you use "exp-rtp/sip-ip"autonat:1.2.3.4" when freeswitch is started
> with -nonat ? Outbound is fine, inbound fails 80 percent if the time.
>
> I really wanted use ALG but when I did all worked fine except call dropped
> while conversing.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Mar 15 00:14:59 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 14 Mar 2011 14:14:59 -0700
Subject: [Freeswitch-users] Multiple Registrations not working
In-Reply-To: <AANLkTinKQ673wyyK-BwKpBv1JN287ssa17sLPKs=vb3o@mail.gmail.com>
References: <AANLkTinKQ673wyyK-BwKpBv1JN287ssa17sLPKs=vb3o@mail.gmail.com>
Message-ID: <AANLkTi=6Z0pHBPtR+Gyu0PMO2RHdXaXz-Hxf0huUDu++@mail.gmail.com>

For kicks try this syntax:

<action application="bridge"
data="<ignore_early_media=true>user/1015:_:user/1016" />

In any case, when you are sending calls to multiple phones you need to
ignore_early_media, otherwise the first phone to send back a 183 (i.e. the
first to send back media) will "win" and all other outbound call legs will
be canceled.

-MC

On Sat, Mar 12, 2011 at 5:58 AM, Yihui Li <xyangni at gmail.com> wrote:

> Hi,
>
> I am trying to use Multiple Registrations feature in bridge, but found it
> not always working when there are more than 1 bridge targets like below:
>
> <action application="bridge" data="user/1015,user/1016" />
>
> where 1015 is registered on 2 PC.
>
> To simplify my case, I have also tested it with CLI command.
> Test 1,
> originate user/1015 &parks
> both PC are ringing,
>
> Test 2,
> originate user/1015,user/1016 &parks
> only 1 PC is ringing
>
> in internal.xml, I added
> <param name="multiple-registrations" value="true"/>
>
> I am running git-head on ubuntu.
> Can anyone help with this issue? Thanks.
>
> Eric
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Mar 15 00:19:11 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 14 Mar 2011 14:19:11 -0700
Subject: [Freeswitch-users] Where to download zrtp/zfone - server
	offline?
In-Reply-To: <4D7BABE8.9090803@telefaks.de>
References: <4D7BABE8.9090803@telefaks.de>
Message-ID: <AANLkTikWO2KLboYyV0-2-77WurDuXmJ7abQ9=z5eLZ9G@mail.gmail.com>

Interesting. Next time we talk to PRZ we'll see what's up with that.
-MC

On Sat, Mar 12, 2011 at 9:22 AM, Peter Steinbach <lists at telefaks.de> wrote:

> I tried to install zrtp and to download Zfone, but I get
> http://www.zfone.com/server_problems.html
> "Sorry, but our download server is offline." (since 29-Jan-2011)
>
> Does anybody know of another download source?
>
> --
> With kind regards
> Peter Steinbach
>
> Telefaks Services GmbH
> mailto:lists (att) telefaks.de
> Internet: www.telefaks.de
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Mar 15 00:24:52 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 14 Mar 2011 14:24:52 -0700
Subject: [Freeswitch-users] Disable Goodbye playback for a particular
	user
In-Reply-To: <62364.88.201.183.18.1299968480.squirrel@mail.onnet.su>
References: <AANLkTi=78LzPnvN36=H-Pfx=UXwEEdokCHE0pFr=GQAA@mail.gmail.com>
	<AANLkTin+60VZvTt=mFT4sGTiiW9BN4zmiAOM5Kxt4Vgm@mail.gmail.com>
	<62364.88.201.183.18.1299968480.squirrel@mail.onnet.su>
Message-ID: <AANLkTin=_7T-e_SFjm65oF+4s=6dBHq4cqLE8=pTdUvY@mail.gmail.com>

You should be able to do this in the dialplan. Without more information I
can't direct you any further, other than to say you can have the dialplan
play a busy tone instead of playing "goodbye":

<action application="playback" data="tone_stream://%(500,500,480,620)"/>

-MC

On Sat, Mar 12, 2011 at 2:21 PM, <iam at onnet.su> wrote:

> Hello FS users,
>
> I have some customers trying to utilize automatic redial feature with
> their pots lines. But when the analog phone listens for a "Goodbye"
> message instead of busy tone it can't determine that the line is busy and
> automatic redial fails.
>
> So the question is whether there is a way to disable or skip playback of a
> "GoodBye" message for a particular user or change "Goodbye" message for
> "User Busy".
>
>
> Thank you in advance.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Mar 15 00:30:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 14 Mar 2011 14:30:38 -0700
Subject: [Freeswitch-users] SIP profile and alias ip
In-Reply-To: <4D7C54F1.5000406@tagnet.ru>
References: <4D7C54F1.5000406@tagnet.ru>
Message-ID: <AANLkTi=Ct0QgAuCTX_7t-PvA2iBzjvFSEEY58jXE9bhm@mail.gmail.com>

The wiki seems to be misleading on this topic. I will confirm with the FS
gurus what should be documented for the alias parameter. Stand by for an
update.

-MC

On Sat, Mar 12, 2011 at 9:24 PM, Boris Kovalenko <boris at tagnet.ru> wrote:

> Hello!
>
>     Reading documentation about sip profiles found a <param
> name="alias">. So I may use same profile with two IP's. Is there any
> disadvantages to use alias? If endpoint will be connected to alias IP
> what IP FS will use for outgoing RTP (alias ip or rtp-ip)? Also, if
> gateway (for outbound calls) is reachable (by host route table) via
> alias IP will it be used for RTP instead of primary rtp-ip?
>
> --
> Regards,
> Boris
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Mar 15 00:31:30 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 14 Mar 2011 14:31:30 -0700
Subject: [Freeswitch-users] Question about proxy_media
In-Reply-To: <4D7C85BF.9030804@tagnet.ru>
References: <4D7C85BF.9030804@tagnet.ru>
Message-ID: <AANLkTik_voT90HA0XXZCG5hPyDBZ9NphHLdM9yuQ5GyF@mail.gmail.com>

Why are you using proxy media? Just curious what it does for you.
-MC

On Sun, Mar 13, 2011 at 12:52 AM, Boris Kovalenko <boris at tagnet.ru> wrote:

> Hello!
>
>     I have two sip profiles. One is for transit traffic with
> proxy_media enabled by default. Second is a SoftPBX for my clients with
> a features like a IVR, conference, attended call transfer and so on. So
> the question is - what happens when A leg is from transit profile and B
> leg is to SoftPBX profile. What mode of proxy_media will be set when
> call is connected? May my user on B leg use DTMF for call transfer for
> example? What mode will be used for FAX?
>
> --
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   (3435) 494991
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Mar 15 00:36:27 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 14 Mar 2011 14:36:27 -0700
Subject: [Freeswitch-users] probem with configuring DIDs
In-Reply-To: <AANLkTi=aK=K-25NL87WYT4m-GQJK_DT=rW6PjabKzAT0@mail.gmail.com>
References: <AANLkTi=aK=K-25NL87WYT4m-GQJK_DT=rW6PjabKzAT0@mail.gmail.com>
Message-ID: <AANLkTimKLWjzUzNEOFGRngebxpLddyXj6KLu1YfR97zQ@mail.gmail.com>

On Mon, Mar 14, 2011 at 1:25 AM, Prime <29prime at gmail.com> wrote:

> Hey!
>
> This most likely is a newb mistake but I'm stuck at the moment.
>
> Recently I had to add another provider to FS to handle incoming calls.
> And now I have a problem with calling my new provider's DID number on
> my FS.
> I looked at FS logs.
> Usually FS log says something like this " [INFO]
> mod_dialplan_xml.c:418 Processing caller_number->My_DID in context
> public" and then routes DID number to local extension,
> but now I see something like this " [INFO] mod_dialplan_xml.c:418
> Processing caller_number->caller_number in context public ".
> Why new numbers does not match the regex?
>
> Best way to know is to look at the debug level output from fs_cli. If you
need help deciphering it then put the output of a failed incoming call into
a pastebin and reply to this thread with the link. Use
pastebin.freeswitch.org for best results.

-MC
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From gavin.henry at gmail.com  Tue Mar 15 01:09:57 2011
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 14 Mar 2011 22:09:57 +0000
Subject: [Freeswitch-users] IP Failover with Sofia Recover - not
	recovering?!
In-Reply-To: <AANLkTinA9f19XPq6d+TzeXt3iE7PW500yBxPGAS+R3GH@mail.gmail.com>
References: <mailman.1927.1295137414.24555.freeswitch-users@lists.freeswitch.org>
	<AANLkTinA9f19XPq6d+TzeXt3iE7PW500yBxPGAS+R3GH@mail.gmail.com>
Message-ID: <AANLkTinBefDiF3mDxjFQm0Q3pW+Y3095Dz02uutV_BUo@mail.gmail.com>

On 16 January 2011 00:27, Avi Marcus <avi at avimarcus.net> wrote:
> I finally figured out how to get pacemaker to start profiles/trigger a sofia
> recover on moving the floating IP (but an FS crash it still can't handle)
>

What did you use for this part? I sent this today to get something
official done, so if you're
up for sharing we can both save others a lot of time:

lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070329.html

Thanks.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.surevoip.co.uk


From avi at avimarcus.net  Tue Mar 15 01:20:29 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 15 Mar 2011 00:20:29 +0200
Subject: [Freeswitch-users] IP Failover with Sofia Recover - not
	recovering?!
In-Reply-To: <AANLkTinBefDiF3mDxjFQm0Q3pW+Y3095Dz02uutV_BUo@mail.gmail.com>
References: <mailman.1927.1295137414.24555.freeswitch-users@lists.freeswitch.org>
	<AANLkTinA9f19XPq6d+TzeXt3iE7PW500yBxPGAS+R3GH@mail.gmail.com>
	<AANLkTinBefDiF3mDxjFQm0Q3pW+Y3095Dz02uutV_BUo@mail.gmail.com>
Message-ID: <AANLkTi=NH0nV9p_ZmBtmKthf-FtSHmDwEuGKjMZ7bx76@mail.gmail.com>

Sorry.
I posted what I currently have here - it's a slight modification of the one
from ledr in the git contrib.. something about it was broken.
http://wiki.freeswitch.org/wiki/Enterprise_deployment_IP_Failover#HeartBeat.2FPacemaker
It seems to try to restart the current one, sometimes thinks it failed.. I
don't really understand it so much.
I find myself doing "crm resource cleanup $fs" often...

-Avi
<http://wiki.freeswitch.org/wiki/Enterprise_deployment_IP_Failover#HeartBeat.2FPacemaker>

On Tue, Mar 15, 2011 at 12:09 AM, Gavin Henry <gavin.henry at gmail.com> wrote:

> On 16 January 2011 00:27, Avi Marcus <avi at avimarcus.net> wrote:
> > I finally figured out how to get pacemaker to start profiles/trigger a
> sofia
> > recover on moving the floating IP (but an FS crash it still can't handle)
> >
>
> What did you use for this part? I sent this today to get something
> official done, so if you're
> up for sharing we can both save others a lot of time:
>
> lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070329.html
>
> Thanks.
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.surevoip.co.uk
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From davidwaf at gmail.com  Tue Mar 15 01:22:49 2011
From: davidwaf at gmail.com (David Wafula)
Date: Tue, 15 Mar 2011 00:22:49 +0200
Subject: [Freeswitch-users] conference comfort noise: recommended setting
Message-ID: <AANLkTinqgtx6TaTDf6iWDkX2bAXMza1EM5pXyw9btwXF@mail.gmail.com>

Hi all,
I wish to include background  noise  in a conference where users are
calling in at 8000khz. What comfort-noise value would be recommended
for this?

Thanks.
-- 
David Wafula


From gavin.henry at gmail.com  Tue Mar 15 01:23:20 2011
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 14 Mar 2011 22:23:20 +0000
Subject: [Freeswitch-users] IP Failover with Sofia Recover - not
	recovering?!
In-Reply-To: <AANLkTi=NH0nV9p_ZmBtmKthf-FtSHmDwEuGKjMZ7bx76@mail.gmail.com>
References: <mailman.1927.1295137414.24555.freeswitch-users@lists.freeswitch.org>
	<AANLkTinA9f19XPq6d+TzeXt3iE7PW500yBxPGAS+R3GH@mail.gmail.com>
	<AANLkTinBefDiF3mDxjFQm0Q3pW+Y3095Dz02uutV_BUo@mail.gmail.com>
	<AANLkTi=NH0nV9p_ZmBtmKthf-FtSHmDwEuGKjMZ7bx76@mail.gmail.com>
Message-ID: <AANLkTimwuyB-xyzPeV_AiHTkQ1vtdVg0sOhN6DvoC2dj@mail.gmail.com>

On 14 March 2011 22:20, Avi Marcus <avi at avimarcus.net> wrote:
> Sorry.
> I posted what I currently have here - it's a slight modification of the one
> from ledr in the git contrib.. something about it was broken.
> http://wiki.freeswitch.org/wiki/Enterprise_deployment_IP_Failover#HeartBeat.2FPacemaker
> It seems to try to restart the current one, sometimes thinks it failed.. I
> don't really understand it so much.
> I find myself doing "crm resource cleanup $fs" often...

Will take a look. That's why I think we need a official one so this
box is ticked!

Gavin.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.surevoip.co.uk


From msc at freeswitch.org  Tue Mar 15 01:32:27 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 14 Mar 2011 15:32:27 -0700
Subject: [Freeswitch-users] How to route ata question
In-Reply-To: <00f801cbe27d$ae479fb0$0ad6df10$@accra.ca>
References: <00f801cbe27d$ae479fb0$0ad6df10$@accra.ca>
Message-ID: <AANLkTinx-AtTQki0xgTed1UX0B23vUf6DWVZssywgpi_@mail.gmail.com>

Well, every call from your ATA to FS already goes through the dialplan. Can
you describe your configuration a bit? For example, did you set up your ATA
to be a user on FreeSWITCH? Also, how are you currently routing your calls
to 5555/IVR?

-MC

On Mon, Mar 14, 2011 at 12:26 PM, Charles Bujold <cjbujold at accra.ca> wrote:

>
>
> My question is I have an ATA FXO (Grandstream ht503) that currently sends
> calls to an extension, I want the calls to be filtered by the other
> Freeswitch settings and I am not certain how to proceed.
>
>
>
> I originally created a Main IVR (ext 5555) and sent the incoming FXO ATA
> calls to the IVR extension.   Now I want to send the calls to some other
> extension or inbound route so that the calls are routed through all the
> Freeswitch  settings such as holidays and time of day  conditions before
> being sent to the IVR or some after hours message.  I?m not certain how I
> should setup the FXO ATA to send the calls so Freeswitch processes them
> properly?    Please advise.
>
>
>
> cjb
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Mar 15 01:39:27 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 14 Mar 2011 15:39:27 -0700
Subject: [Freeswitch-users] conference comfort noise: recommended setting
In-Reply-To: <AANLkTinqgtx6TaTDf6iWDkX2bAXMza1EM5pXyw9btwXF@mail.gmail.com>
References: <AANLkTinqgtx6TaTDf6iWDkX2bAXMza1EM5pXyw9btwXF@mail.gmail.com>
Message-ID: <AANLkTikcmNeDjm+JxMNNhC6PT20RUmUz_rDeJPkoEjq1@mail.gmail.com>

On Mon, Mar 14, 2011 at 3:22 PM, David Wafula <davidwaf at gmail.com> wrote:

> Hi all,
> I wish to include background  noise  in a conference where users are
> calling in at 8000khz. What comfort-noise value would be recommended
> for this?
>
> Hard to say. I think your best bet would be to experiment and see what
actually sounds good to you and your users.
-MC
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From avi at avimarcus.net  Tue Mar 15 02:00:01 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 15 Mar 2011 01:00:01 +0200
Subject: [Freeswitch-users] IP Failover with Sofia Recover - not
	recovering?!
In-Reply-To: <AANLkTimwuyB-xyzPeV_AiHTkQ1vtdVg0sOhN6DvoC2dj@mail.gmail.com>
References: <mailman.1927.1295137414.24555.freeswitch-users@lists.freeswitch.org>
	<AANLkTinA9f19XPq6d+TzeXt3iE7PW500yBxPGAS+R3GH@mail.gmail.com>
	<AANLkTinBefDiF3mDxjFQm0Q3pW+Y3095Dz02uutV_BUo@mail.gmail.com>
	<AANLkTi=NH0nV9p_ZmBtmKthf-FtSHmDwEuGKjMZ7bx76@mail.gmail.com>
	<AANLkTimwuyB-xyzPeV_AiHTkQ1vtdVg0sOhN6DvoC2dj@mail.gmail.com>
Message-ID: <AANLkTinGMg_C0BJVV1XWcuVvYEj5pMhO7hX1S5UFLKCY@mail.gmail.com>

Sorry! I knew that looked wrong!

I had the original fssofia, but I've since updated it to the one I patched.
Take a look again at that section of the wiki..
I think I also turned off the "stop" part because it was propagating SQL
deletion of registrations to the active master...
And notice the hardcoded profile names. Cleanup is appreciated :)
I should get back my sql sync and maybe we can make this exactly reliable..
-Avi Marcus

On Tue, Mar 15, 2011 at 12:23 AM, Gavin Henry <gavin.henry at gmail.com> wrote:

> On 14 March 2011 22:20, Avi Marcus <avi at avimarcus.net> wrote:
> > Sorry.
> > I posted what I currently have here - it's a slight modification of the
> one
> > from ledr in the git contrib.. something about it was broken.
> >
> http://wiki.freeswitch.org/wiki/Enterprise_deployment_IP_Failover#HeartBeat.2FPacemaker
> > It seems to try to restart the current one, sometimes thinks it failed..
> I
> > don't really understand it so much.
> > I find myself doing "crm resource cleanup $fs" often...
>
> Will take a look. That's why I think we need a official one so this
> box is ticked!
>
> Gavin.
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.surevoip.co.uk
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Tue Mar 15 03:01:30 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 14 Mar 2011 20:01:30 -0400
Subject: [Freeswitch-users] debug down trunk
References: <60BA130CDBF44BD69A43BA7777AE83BD@e1705>
	<AANLkTi=ETiNk4bArqj8VS8nN9j=+LkcwAc9qygN-zkSk@mail.gmail.com>
Message-ID: <BE6B9A96642745228DAA6ED4E06102CC@e1705>

thanks Steve,

here the trace :

------------------------------------------------------------------------
send 643 bytes to udp/[10.10.10.10]:5060 at 23:57:10.312821:
------------------------------------------------------------------------
OPTIONS sip:10.10.10.10;transport=udp SIP/2.0
Via: SIP/2.0/UDP 30.30.30.30;rport;branch=z9hG4bKD38154a4e7rXa
Max-Forwards: 70
From: <sip:freeswitch at 10.10.10.10;transport=udp>;tag=eB6B7NSge90vK
To: <sip:freeswitch at 10.10.10.10;transport=udp>
Call-ID: 974aa6c8-c939-122e-45ab-00e0ed0b00c2
CSeq: 9737404 OPTIONS
Contact: <sip:gw+trunk_3 at 30.30.30.30:5060;tport=tcp;transport=udp;gw=trunk_3>
User-Agent: CiscoSystems-SIP-GW-UA
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, refer
Content-Length: 0

------------------------------------------------------------------------

weird because it looks like exactly the same response as the other trunks but
it states to down....

  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Monday, March 14, 2011 5:06 PM
  Subject: Re: [Freeswitch-users] debug down trunk


  408 is the error code - this means "408 Request timed out"

  FS tries sending an OPTIONS request to the gateway and waits for a reply to test whether it is online or not..

  You can use the siptrace to see those messages and the reply:
  sofia global siptrace on

  -Steve



  On 14 March 2011 19:03, Madovsky <infos at madovsky.org> wrote:

    I set external.xml to debug=1 and sip trace to 1 and level debug to 7
    but I can get only this in the log

    sofia.c:4028 Ping failed trunk_3 with code 408 - count -1/-1/1, state DOWN

    so I don't know why this trunk is down.
    I contacted them and they said that my account is working.

    Thanks

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





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From gcd at i.ph  Tue Mar 15 03:01:44 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Tue, 15 Mar 2011 08:01:44 +0800
Subject: [Freeswitch-users] How to route ata question
In-Reply-To: <00f801cbe27d$ae479fb0$0ad6df10$@accra.ca>
References: <00f801cbe27d$ae479fb0$0ad6df10$@accra.ca>
Message-ID: <AANLkTi=dp2L_69RQt+7gjAOYbWcxzrKp4R6j6EnKxraS@mail.gmail.com>

on the BASIC SETTINGS page, set the Unconditional Call Forward to VoIP
setting to match the destination_number you wish in FS.  at the FXO PORT
settings, set the number of rings before the FXO grabs the incoming call.

i hope this helps.

On Tue, Mar 15, 2011 at 3:26 AM, Charles Bujold <cjbujold at accra.ca> wrote:

>
>
> My question is I have an ATA FXO (Grandstream ht503) that currently sends
> calls to an extension, I want the calls to be filtered by the other
> Freeswitch settings and I am not certain how to proceed.
>
>
>
> I originally created a Main IVR (ext 5555) and sent the incoming FXO ATA
> calls to the IVR extension.   Now I want to send the calls to some other
> extension or inbound route so that the calls are routed through all the
> Freeswitch  settings such as holidays and time of day  conditions before
> being sent to the IVR or some after hours message.  I?m not certain how I
> should setup the FXO ATA to send the calls so Freeswitch processes them
> properly?    Please advise.
>
>
>
> cjb
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Tue Mar 15 03:10:12 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 14 Mar 2011 20:10:12 -0400
Subject: [Freeswitch-users] debug down trunk
References: <60BA130CDBF44BD69A43BA7777AE83BD@e1705><AANLkTi=ETiNk4bArqj8VS8nN9j=+LkcwAc9qygN-zkSk@mail.gmail.com>
	<BE6B9A96642745228DAA6ED4E06102CC@e1705>
Message-ID: <E62D9A8F4352426BAD5540CCAEFD64DC@e1705>

oopss, sorry I sent the send message...
apparently their server doesn't answer..
some 10 sends are made without answer
  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Monday, March 14, 2011 8:01 PM
  Subject: Re: [Freeswitch-users] debug down trunk


  thanks Steve,

  here the trace :

  ------------------------------------------------------------------------
  send 643 bytes to udp/[10.10.10.10]:5060 at 23:57:10.312821:
  ------------------------------------------------------------------------
  OPTIONS sip:10.10.10.10;transport=udp SIP/2.0
  Via: SIP/2.0/UDP 30.30.30.30;rport;branch=z9hG4bKD38154a4e7rXa
  Max-Forwards: 70
  From: <sip:freeswitch at 10.10.10.10;transport=udp>;tag=eB6B7NSge90vK
  To: <sip:freeswitch at 10.10.10.10;transport=udp>
  Call-ID: 974aa6c8-c939-122e-45ab-00e0ed0b00c2
  CSeq: 9737404 OPTIONS
  Contact: <sip:gw+trunk_3 at 30.30.30.30:5060;tport=tcp;transport=udp;gw=trunk_3>
  User-Agent: CiscoSystems-SIP-GW-UA
  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REFER, NOTIFY
  Supported: timer, precondition, path, replaces
  Allow-Events: talk, hold, refer
  Content-Length: 0

  ------------------------------------------------------------------------

  weird because it looks like exactly the same response as the other trunks but
  it states to down....

    ----- Original Message ----- 
    From: Steven Ayre 
    To: FreeSWITCH Users Help 
    Sent: Monday, March 14, 2011 5:06 PM
    Subject: Re: [Freeswitch-users] debug down trunk


    408 is the error code - this means "408 Request timed out"

    FS tries sending an OPTIONS request to the gateway and waits for a reply to test whether it is online or not..

    You can use the siptrace to see those messages and the reply:
    sofia global siptrace on

    -Steve



    On 14 March 2011 19:03, Madovsky <infos at madovsky.org> wrote:

      I set external.xml to debug=1 and sip trace to 1 and level debug to 7
      but I can get only this in the log

      sofia.c:4028 Ping failed trunk_3 with code 408 - count -1/-1/1, state DOWN

      so I don't know why this trunk is down.
      I contacted them and they said that my account is working.

      Thanks

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





----------------------------------------------------------------------------


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From gavin.henry at gmail.com  Tue Mar 15 03:10:58 2011
From: gavin.henry at gmail.com (Gavin Henry)
Date: Tue, 15 Mar 2011 00:10:58 +0000
Subject: [Freeswitch-users] IP Failover with Sofia Recover - not
	recovering?!
In-Reply-To: <AANLkTinGMg_C0BJVV1XWcuVvYEj5pMhO7hX1S5UFLKCY@mail.gmail.com>
References: <mailman.1927.1295137414.24555.freeswitch-users@lists.freeswitch.org>
	<AANLkTinA9f19XPq6d+TzeXt3iE7PW500yBxPGAS+R3GH@mail.gmail.com>
	<AANLkTinBefDiF3mDxjFQm0Q3pW+Y3095Dz02uutV_BUo@mail.gmail.com>
	<AANLkTi=NH0nV9p_ZmBtmKthf-FtSHmDwEuGKjMZ7bx76@mail.gmail.com>
	<AANLkTimwuyB-xyzPeV_AiHTkQ1vtdVg0sOhN6DvoC2dj@mail.gmail.com>
	<AANLkTinGMg_C0BJVV1XWcuVvYEj5pMhO7hX1S5UFLKCY@mail.gmail.com>
Message-ID: <AANLkTinheLesNTWinsmGTa-DkpSShDUzuYo9Dyri7k-S@mail.gmail.com>

On 14 March 2011 23:00, Avi Marcus <avi at avimarcus.net> wrote:
> Sorry! I knew that looked wrong!
> I had the original fssofia, but I've since updated it to the one I patched.
> Take a look again at that section of the wiki..
> I think I also turned off the "stop" part because it was?propagating?SQL
> deletion of registrations to the active master...
> And notice the hardcoded profile names. Cleanup is appreciated :)
> I should get back my sql sync and maybe we can make this exactly reliable..

Thanks, will test this on our VMs and get a final version.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.surevoip.co.uk


From boris at tagnet.ru  Tue Mar 15 08:34:28 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 15 Mar 2011 10:34:28 +0500
Subject: [Freeswitch-users] Question about proxy_media
In-Reply-To: <AANLkTik_voT90HA0XXZCG5hPyDBZ9NphHLdM9yuQ5GyF@mail.gmail.com>
References: <4D7C85BF.9030804@tagnet.ru>
	<AANLkTik_voT90HA0XXZCG5hPyDBZ9NphHLdM9yuQ5GyF@mail.gmail.com>
Message-ID: <4D7EFA64.3050101@tagnet.ru>

Hello!

     Only for T.38 faxing. I can't understand how to do it proper with 
mod_spandsp.
> Why are you using proxy media? Just curious what it does for you.
> -MC
>
> On Sun, Mar 13, 2011 at 12:52 AM, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     Hello!
>
>         I have two sip profiles. One is for transit traffic with
>     proxy_media enabled by default. Second is a SoftPBX for my clients
>     with
>     a features like a IVR, conference, attended call transfer and so
>     on. So
>     the question is - what happens when A leg is from transit profile
>     and B
>     leg is to SoftPBX profile. What mode of proxy_media will be set when
>     call is connected? May my user on B leg use DTMF for call transfer for
>     example? What mode will be used for FAX?
>
>     --
>     ? ?????????,
>       ????? ?????????
>       ??? "??????"
>       (3435) 494991
>
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991


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From davidwaf at gmail.com  Tue Mar 15 10:01:19 2011
From: davidwaf at gmail.com (David Wafula)
Date: Tue, 15 Mar 2011 09:01:19 +0200
Subject: [Freeswitch-users] conference comfort noise: recommended setting
In-Reply-To: <AANLkTikcmNeDjm+JxMNNhC6PT20RUmUz_rDeJPkoEjq1@mail.gmail.com>
References: <AANLkTinqgtx6TaTDf6iWDkX2bAXMza1EM5pXyw9btwXF@mail.gmail.com>
	<AANLkTikcmNeDjm+JxMNNhC6PT20RUmUz_rDeJPkoEjq1@mail.gmail.com>
Message-ID: <AANLkTikf6mqXZM=1PtJhm972DrsqXk78H99CZMmpA-3j@mail.gmail.com>

On Tue, Mar 15, 2011 at 12:39 AM, Michael Collins <msc at freeswitch.org> wrote:

>>
> Hard to say. I think your best bet would be to experiment and see what
> actually sounds good to you and your users.
>

Thank you...
-- 
David Wafula


From mitja.thomas1 at ewetel.de  Tue Mar 15 11:36:47 2011
From: mitja.thomas1 at ewetel.de (Mitja Thomas)
Date: Tue, 15 Mar 2011 09:36:47 +0100
Subject: [Freeswitch-users] [freeswitch] <defunct> when using cmd="exec"
 in freeswitch.xml
In-Reply-To: <AANLkTin7EUJpz4=9gxp3sTXykjybH5B4Uer7noD+Bgct@mail.gmail.com>
References: <4D78A5AF.1000808@ewetel.de>
	<AANLkTin7EUJpz4=9gxp3sTXykjybH5B4Uer7noD+Bgct@mail.gmail.com>
Message-ID: <4D7F251F.2070101@ewetel.de>

Hi Micheal,

yeah I think your workaround will work fine and Ill change it that way. 
But this issue might still be a missbehaviour. Either by fs in which 
case I think it might be an interesting information for the Developers 
OR a wrong usage by me in which case this is an interesting information 
for me :)

Thanks,
Mitja
> IIRC, 'fork' creates a child PID that stays around and waits for the 
> parent PID to die. (Don't quote me on that - wait for an expert to 
> chime in.)
>
> An alternative method would be to have a FS start script that launches 
> your make_my_vars.sh script and then launches FS with the appropriate 
> cmd line args. You could then drop the 'exec' cmd and then just keep 
> the include cmd.
>
> -MC
>
> On Thu, Mar 10, 2011 at 2:19 AM, Mitja Thomas <mitja.thomas1 at ewetel.de 
> <mailto:mitja.thomas1 at ewetel.de>> wrote:
>
>     Hi there,
>
>     we tried to set up the FreeSWITCH and other Applications, so that
>     we can configure them easier and more centralised.
>     Thus we defined some Environment Variables (using CentOS) which
>     hold often used Configuration Parameter like MySQL IP or FS Event
>     Socket IP.
>     We tried to integrate these Env Variables into the FS conf files
>     by executing a shell Skript in freeswitch.xml via cmd="exec" which
>     prepares an conf file which we include into freeswitch.xml:
>
>     <X-PRE-PROCESS cmd="exec" data="sh conf/make_my_vars.sh"/>
>     <X-PRE-PROCESS cmd="include" data="my_vars.xml"/>
>
>     This works as expected and the pre defined variables in my_vars
>     can be accessed from the other config Files, except that when we
>     start our FreeSWITCH  a zombie child process is spawned.
>
>     # ps -eaf | grep free
>     ippbx    22191 22190  4 09:41 pts/1    00:00:01
>     /opt/app/voip/ippbx/bin/freeswitch -waste -nonat -hp
>     ippbx    22197 22191  0 09:41 pts/1    00:00:00 [freeswitch] <defunct>
>
>     What I wanna know is: Is this a FS missbehaviour or do we use this
>     in a wrong way?
>
>     make_my_vars.sh:
>     F="conf/my_vars.xml"
>     echo "<!-- Auto generated file for getting *NIX evironment
>     variables -->" > $F
>     echo "<include>" >> $F
>
>     fs_ip=`printenv MY_FS_IP`
>     if test -n "$fs_ip"
>     then
>             echo '<X-PRE-PROCESS cmd="set" data="my_fs_ip='$fs_ip'"/>'
>     >> $F
>     fi
>     ...
>     echo "</include>" >> $F
>
>     my_vars.xml (after FS startup):
>     <!-- Auto generated file for getting *NIX evironment variables -->
>     <include>
>     <X-PRE-PROCESS cmd="set" data="my_fs_ip=***.***.***.***"/>
>     ...
>     </include>
>
>     Regards
>     Mitja
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>

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From chistyakov at directtel.ru  Tue Mar 15 14:21:14 2011
From: chistyakov at directtel.ru (=?UTF-8?B?0KfQuNGB0YLRj9C60L7QsiDQmNCy0LDQvQ==?=)
Date: Tue, 15 Mar 2011 14:21:14 +0300
Subject: [Freeswitch-users] modify ip and port of rtp server in sdp data
Message-ID: <4D7F4BAA.50107@directtel.ru>

how can i modify ip and port of rtp server in sdp data on both legs 
before bridging for using external rtpproxy ?


From steveayre at gmail.com  Tue Mar 15 14:53:42 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 15 Mar 2011 11:53:42 +0000
Subject: [Freeswitch-users] modify ip and port of rtp server in sdp data
In-Reply-To: <4D7F4BAA.50107@directtel.ru>
References: <4D7F4BAA.50107@directtel.ru>
Message-ID: <AANLkTikPyCTOw5AM1sHDBTsRwR7Xnxt11D=r4Q1PmDU7@mail.gmail.com>

Setting ext-rtp-ip on the sip profile might work.

-Steve


On 15 March 2011 11:21, ???????? ???? <chistyakov at directtel.ru> wrote:

> how can i modify ip and port of rtp server in sdp data on both legs
> before bridging for using external rtpproxy ?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Tue Mar 15 15:01:17 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 15 Mar 2011 12:01:17 +0000
Subject: [Freeswitch-users] modify ip and port of rtp server in sdp data
In-Reply-To: <AANLkTikPyCTOw5AM1sHDBTsRwR7Xnxt11D=r4Q1PmDU7@mail.gmail.com>
References: <4D7F4BAA.50107@directtel.ru>
	<AANLkTikPyCTOw5AM1sHDBTsRwR7Xnxt11D=r4Q1PmDU7@mail.gmail.com>
Message-ID: <AANLkTi=NHZyWOvW0s2pW=eaU5BFSKWcB5z6N9FNoJ2a0@mail.gmail.com>

How do you plan to tell the rtpproxy where to send the media?

-Steve


On 15 March 2011 11:53, Steven Ayre <steveayre at gmail.com> wrote:

> Setting ext-rtp-ip on the sip profile might work.
>
> -Steve
>
>
>
> On 15 March 2011 11:21, ???????? ???? <chistyakov at directtel.ru> wrote:
>
>> how can i modify ip and port of rtp server in sdp data on both legs
>> before bridging for using external rtpproxy ?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From lluis.riera at quarea.com  Tue Mar 15 15:21:42 2011
From: lluis.riera at quarea.com (=?iso-8859-1?Q?Llu=EDs_Riera?=)
Date: Tue, 15 Mar 2011 13:21:42 +0100
Subject: [Freeswitch-users] Multiconference with g729 users only without
	transcoding
Message-ID: <738A792011F8794D8CFAE25C00089679DC1AC9@quarea-correu.quarea.com>

Hi,

 

I would like to know if is there a configuration in freeswitch to allow multiconferencing for g729 users but without transcoding? It seems that all users are converted to L16 and it's not possible to have passthrough mode in conference rooms.

 

Best Regards,

 

Llu?s

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From Nabble at slickdeals.endjunk.com  Tue Mar 15 15:40:31 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 15 Mar 2011 05:40:31 -0700 (PDT)
Subject: [Freeswitch-users] conference comfort noise: recommended setting
In-Reply-To: <AANLkTinqgtx6TaTDf6iWDkX2bAXMza1EM5pXyw9btwXF@mail.gmail.com>
References: <AANLkTinqgtx6TaTDf6iWDkX2bAXMza1EM5pXyw9btwXF@mail.gmail.com>
Message-ID: <1300192831488-6172641.post@n2.nabble.com>


David Wafula wrote:
> I wish to include background  noise  in a conference where users are
> calling in at 8000khz.
You're not kidding me with an 8,000KHz, R U? 

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/conference-comfort-noise-recommended-setting-tp6170949p6172641.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From guillaume.renaud at gmail.com  Tue Mar 15 15:51:43 2011
From: guillaume.renaud at gmail.com (Guillaume Renaud)
Date: Tue, 15 Mar 2011 08:51:43 -0400
Subject: [Freeswitch-users] Multiconference with g729 users only without
	transcoding
In-Reply-To: <738A792011F8794D8CFAE25C00089679DC1AC9@quarea-correu.quarea.com>
References: <738A792011F8794D8CFAE25C00089679DC1AC9@quarea-correu.quarea.com>
Message-ID: <AANLkTikgZtsViW3U5tXgGgbiq3z9mN0jcffJggWbC8b1@mail.gmail.com>

Seems logical to have the voice decoded to mix it in a conference, how could
someone mix G.729 encoded voice is your question I guess?

Good luck!

Guillaume Renaud, ing.


On Tue, Mar 15, 2011 at 8:21 AM, Llu?s Riera <lluis.riera at quarea.com> wrote:

>  Hi,
>
>
>
> I would like to know if is there a configuration in freeswitch to allow
> multiconferencing for g729 users but without transcoding? It seems that all
> users are converted to L16 and it?s not possible to have passthrough mode in
> conference rooms.
>
>
>
> Best Regards,
>
>
>
> Llu?s
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From davidwaf at gmail.com  Tue Mar 15 15:53:09 2011
From: davidwaf at gmail.com (David Wafula)
Date: Tue, 15 Mar 2011 14:53:09 +0200
Subject: [Freeswitch-users] conference comfort noise: recommended setting
In-Reply-To: <1300192831488-6172641.post@n2.nabble.com>
References: <AANLkTinqgtx6TaTDf6iWDkX2bAXMza1EM5pXyw9btwXF@mail.gmail.com>
	<1300192831488-6172641.post@n2.nabble.com>
Message-ID: <AANLkTi=-1HvxOjyjRjqoRdk-Um9MC1uR_Vofejfxdb9z@mail.gmail.com>

On Tue, Mar 15, 2011 at 2:40 PM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
> You're not kidding me with an 8,000KHz, R U?
>

Erm...meant 8Khz  : )    :


-- 
David Wafula


From chistyakov at directtel.ru  Tue Mar 15 15:38:38 2011
From: chistyakov at directtel.ru (=?UTF-8?B?0KfQuNGB0YLRj9C60L7QsiDQmNCy0LDQvQ==?=)
Date: Tue, 15 Mar 2011 15:38:38 +0300
Subject: [Freeswitch-users] modify ip and port of rtp server in sdp data
In-Reply-To: <AANLkTi=NHZyWOvW0s2pW=eaU5BFSKWcB5z6N9FNoJ2a0@mail.gmail.com>
References: <4D7F4BAA.50107@directtel.ru>	<AANLkTikPyCTOw5AM1sHDBTsRwR7Xnxt11D=r4Q1PmDU7@mail.gmail.com>
	<AANLkTi=NHZyWOvW0s2pW=eaU5BFSKWcB5z6N9FNoJ2a0@mail.gmail.com>
Message-ID: <4D7F5DCE.7090700@directtel.ru>

I am writing a dialplan-application (in LUA) for managing of all calls 
in FreeSWITCH, and I am planning to use rtpproxy control protocol for 
managing of RTP streams between subscribers.

http://www.rtpproxy.org/wiki/RTPproxy/Protocol


I think  ext-rtp-ip not using in bypass mode.


15.03.2011 15:01, Steven Ayre ?????:
> How do you plan to tell the rtpproxy where to send the media?
>
> -Steve
>
>
> On 15 March 2011 11:53, Steven Ayre <steveayre at gmail.com 
> <mailto:steveayre at gmail.com>> wrote:
>
>     Setting ext-rtp-ip on the sip profile might work.
>
>     -Steve
>
>
>
>     On 15 March 2011 11:21, ???????? ???? <chistyakov at directtel.ru
>     <mailto:chistyakov at directtel.ru>> wrote:
>
>         how can i modify ip and port of rtp server in sdp data on both
>         legs
>         before bridging for using external rtpproxy ?
>
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From lluis.riera at quarea.com  Tue Mar 15 16:00:45 2011
From: lluis.riera at quarea.com (=?UTF-8?B?TGx1w61zIFJpZXJh?=)
Date: Tue, 15 Mar 2011 14:00:45 +0100
Subject: [Freeswitch-users] Multiconference with g729 users only without
	transcoding
References: <738A792011F8794D8CFAE25C00089679DC1AC9@quarea-correu.quarea.com>
	<AANLkTikgZtsViW3U5tXgGgbiq3z9mN0jcffJggWbC8b1@mail.gmail.com>
Message-ID: <738A792011F8794D8CFAE25C00089679DC1AD3@quarea-correu.quarea.com>

I want to have a pure g729 multiconference in passthrough mode, all participants would be g729 phones and all audios would be in g729 format? that way I will not have to buy a transcoding card or license.

 

De: Guillaume Renaud [mailto:guillaume.renaud at gmail.com] 
Enviado el: martes, 15 de marzo de 2011 13:52
Para: FreeSWITCH Users Help
CC: Llu?s Riera
Asunto: Re: [Freeswitch-users] Multiconference with g729 users only without transcoding

 

Seems logical to have the voice decoded to mix it in a conference, how could someone mix G.729 encoded voice is your question I guess?

 

Good luck!


Guillaume Renaud, ing.



On Tue, Mar 15, 2011 at 8:21 AM, Llu?s Riera <lluis.riera at quarea.com> wrote:

Hi,

 

I would like to know if is there a configuration in freeswitch to allow multiconferencing for g729 users but without transcoding? It seems that all users are converted to L16 and it?s not possible to have passthrough mode in conference rooms.

 

Best Regards,

 

Llu?s


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

 

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From steveu at coppice.org  Tue Mar 15 16:27:51 2011
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 15 Mar 2011 21:27:51 +0800
Subject: [Freeswitch-users] Multiconference with g729 users only without
 transcoding
In-Reply-To: <738A792011F8794D8CFAE25C00089679DC1AD3@quarea-correu.quarea.com>
References: <738A792011F8794D8CFAE25C00089679DC1AC9@quarea-correu.quarea.com>	<AANLkTikgZtsViW3U5tXgGgbiq3z9mN0jcffJggWbC8b1@mail.gmail.com>
	<738A792011F8794D8CFAE25C00089679DC1AD3@quarea-correu.quarea.com>
Message-ID: <4D7F6957.9060803@coppice.org>

On 03/15/2011 09:00 PM, Llu?s Riera wrote:
>
> I want to have a pure g729 multiconference in passthrough mode, all 
> participants would be g729 phones and all audios would be in g729 
> format? that way I will not have to buy a transcoding card or license.
>
>
How do you think multiple G.729 input signals might be combined into a 
common G.729 output signal, without decoding and mixing the linear signals?

Steve



From steveu at coppice.org  Tue Mar 15 16:29:19 2011
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 15 Mar 2011 21:29:19 +0800
Subject: [Freeswitch-users] conference comfort noise: recommended setting
In-Reply-To: <AANLkTinqgtx6TaTDf6iWDkX2bAXMza1EM5pXyw9btwXF@mail.gmail.com>
References: <AANLkTinqgtx6TaTDf6iWDkX2bAXMza1EM5pXyw9btwXF@mail.gmail.com>
Message-ID: <4D7F69AF.7000604@coppice.org>

On 03/15/2011 06:22 AM, David Wafula wrote:
> Hi all,
> I wish to include background  noise  in a conference where users are
> calling in at 8000khz. What comfort-noise value would be recommended
> for this?
>
> Thanks.
What is your concept of comfort noise? Are you looking for Hoth noise (a 
model of typical room background mush)?

Steve



From steveayre at gmail.com  Tue Mar 15 16:28:55 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 15 Mar 2011 13:28:55 +0000
Subject: [Freeswitch-users] modify ip and port of rtp server in sdp data
In-Reply-To: <4D7F5DCE.7090700@directtel.ru>
References: <4D7F4BAA.50107@directtel.ru>
	<AANLkTikPyCTOw5AM1sHDBTsRwR7Xnxt11D=r4Q1PmDU7@mail.gmail.com>
	<AANLkTi=NHZyWOvW0s2pW=eaU5BFSKWcB5z6N9FNoJ2a0@mail.gmail.com>
	<4D7F5DCE.7090700@directtel.ru>
Message-ID: <AANLkTimSUNwACbwgLiysoQi0DzAqiac2czB=JqE0H4nC@mail.gmail.com>

>
> I think  ext-rtp-ip not using in bypass mode.
>

No, it isn't. In bypass mode you will not be able to do what you want - the
SDP from either endpoint will be sent to the other, they'll only see the IP
of the other endpoint.

-Steve



On 15 March 2011 12:38, ???????? ???? <chistyakov at directtel.ru> wrote:

>  I am writing a dialplan-application (in LUA) for managing of all calls in
> FreeSWITCH, and I am planning to use rtpproxy control protocol for managing
> of RTP streams between subscribers.
>
> http://www.rtpproxy.org/wiki/RTPproxy/Protocol
>
>
> I think  ext-rtp-ip not using in bypass mode.
>
>
> 15.03.2011 15:01, Steven Ayre ?????:
>
> How do you plan to tell the rtpproxy where to send the media?
>
> -Steve
>
>
> On 15 March 2011 11:53, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Setting ext-rtp-ip on the sip profile might work.
>>
>> -Steve
>>
>>
>>
>> On 15 March 2011 11:21, ???????? ???? <chistyakov at directtel.ru> wrote:
>>
>>> how can i modify ip and port of rtp server in sdp data on both legs
>>> before bridging for using external rtpproxy ?
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Tue Mar 15 16:31:24 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 15 Mar 2011 13:31:24 +0000
Subject: [Freeswitch-users] Multiconference with g729 users only without
	transcoding
In-Reply-To: <738A792011F8794D8CFAE25C00089679DC1AC9@quarea-correu.quarea.com>
References: <738A792011F8794D8CFAE25C00089679DC1AC9@quarea-correu.quarea.com>
Message-ID: <AANLkTinPKf8WvVK6p_5VM=dPA5K_bO8bLW+9Ar-4+Tk5@mail.gmail.com>

Sorry, but that's impossible.

For a conference you have multiple streams coming into the switch from each
caller. They have to be mixed to combine them into a single stream to send
to each member of the conference. Mixing multiple streams can only be done
on the raw audio data (L16) - it requires transcoding.

-Steve


On 15 March 2011 12:21, Llu?s Riera <lluis.riera at quarea.com> wrote:

>  Hi,
>
>
>
> I would like to know if is there a configuration in freeswitch to allow
> multiconferencing for g729 users but without transcoding? It seems that all
> users are converted to L16 and it?s not possible to have passthrough mode in
> conference rooms.
>
>
>
> Best Regards,
>
>
>
> Llu?s
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mustafa.pk at gmail.com  Tue Mar 15 16:34:52 2011
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Tue, 15 Mar 2011 18:34:52 +0500
Subject: [Freeswitch-users] Multiconference with g729 users only without
	transcoding
In-Reply-To: <4D7F6957.9060803@coppice.org>
References: <738A792011F8794D8CFAE25C00089679DC1AC9@quarea-correu.quarea.com>
	<AANLkTikgZtsViW3U5tXgGgbiq3z9mN0jcffJggWbC8b1@mail.gmail.com>
	<738A792011F8794D8CFAE25C00089679DC1AD3@quarea-correu.quarea.com>
	<4D7F6957.9060803@coppice.org>
Message-ID: <AANLkTikcKAuJVuyXYnf=GQsDkrEZEAEHaXar8w5X660A@mail.gmail.com>

You can not make conferences work in paasthrough mode.
On Mar 15, 2011 6:29 PM, "Steve Underwood" <steveu at coppice.org> wrote:
> On 03/15/2011 09:00 PM, Llu?s Riera wrote:
>>
>> I want to have a pure g729 multiconference in passthrough mode, all
>> participants would be g729 phones and all audios would be in g729
>> format? that way I will not have to buy a transcoding card or license.
>>
>>
> How do you think multiple G.729 input signals might be combined into a
> common G.729 output signal, without decoding and mixing the linear
signals?
>
> Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From davidwaf at gmail.com  Tue Mar 15 16:47:30 2011
From: davidwaf at gmail.com (David Wafula)
Date: Tue, 15 Mar 2011 15:47:30 +0200
Subject: [Freeswitch-users] conference comfort noise: recommended setting
In-Reply-To: <4D7F69AF.7000604@coppice.org>
References: <AANLkTinqgtx6TaTDf6iWDkX2bAXMza1EM5pXyw9btwXF@mail.gmail.com>
	<4D7F69AF.7000604@coppice.org>
Message-ID: <AANLkTikkn4d41rPsSCJoec7FK20ZzafFZG1O8hJLDQHL@mail.gmail.com>

On Tue, Mar 15, 2011 at 3:29 PM, Steve Underwood <steveu at coppice.org> wrote:
> What is your concept of comfort noise? Are you looking for Hoth noise (a
> model of typical room background mush)?

Yesterday we were having this conference (just three people), and
immediately i noticed when none of us say anything, the lines get
quite, too quite, we had to keep asking each other "are you there?".
So i need some sound on the line as long as you're connected. So yes.



-- 
David Wafula


From frank at rosengart.de  Tue Mar 15 16:57:52 2011
From: frank at rosengart.de (Frank Rosengart)
Date: Tue, 15 Mar 2011 14:57:52 +0100
Subject: [Freeswitch-users] conference comfort noise: recommended setting
In-Reply-To: <AANLkTikkn4d41rPsSCJoec7FK20ZzafFZG1O8hJLDQHL@mail.gmail.com>
References: <AANLkTinqgtx6TaTDf6iWDkX2bAXMza1EM5pXyw9btwXF@mail.gmail.com>	<4D7F69AF.7000604@coppice.org>
	<AANLkTikkn4d41rPsSCJoec7FK20ZzafFZG1O8hJLDQHL@mail.gmail.com>
Message-ID: <4D7F7060.9070406@rosengart.de>

On 03/15/2011 02:47 PM, David Wafula wrote:

> Yesterday we were having this conference (just three people), and
> immediately i noticed when none of us say anything, the lines get
> quite, too quite, we had to keep asking each other "are you there?".

You can try to deactivate the VAD on the conference by setting
"energy-level" to 0.

Frank




From cjbujold at accra.ca  Tue Mar 15 16:57:32 2011
From: cjbujold at accra.ca (Charles Bujold)
Date: Tue, 15 Mar 2011 10:57:32 -0300
Subject: [Freeswitch-users] How to route ata question
In-Reply-To: <AANLkTinx-AtTQki0xgTed1UX0B23vUf6DWVZssywgpi_@mail.gmail.com>
References: <00f801cbe27d$ae479fb0$0ad6df10$@accra.ca>
	<AANLkTinx-AtTQki0xgTed1UX0B23vUf6DWVZssywgpi_@mail.gmail.com>
Message-ID: <00f701cbe318$ee4f3640$caeda2c0$@accra.ca>

My configuration is I have 3 lines  1- voip connection used for long
distance calling (in/out),  2-pots used for local calls and toll free
calling.  We have 6 telephones/users.   The main lines are the pots lines
and the VOIP is our 1-800 incoming line .    The pots are connected to the
grandstream ht503 which are directly sent to the extension 5555 per the
"Unconditional Call forward to VOIP" option of the ht503.

 

What I'm trying to do is have the incoming calls filtered for holidays and
non-work hours so I can send the call to and after hour message.   I'm Not
sure how to proceed to do this since the HT503 send all incoming calls to
the main reception IVR 5555.

 

cjb

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: March-14-11 7:32 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How to route ata question

 

Well, every call from your ATA to FS already goes through the dialplan. Can
you describe your configuration a bit? For example, did you set up your ATA
to be a user on FreeSWITCH? Also, how are you currently routing your calls
to 5555/IVR? 

 

-MC

On Mon, Mar 14, 2011 at 12:26 PM, Charles Bujold <cjbujold at accra.ca> wrote:

 

My question is I have an ATA FXO (Grandstream ht503) that currently sends
calls to an extension, I want the calls to be filtered by the other
Freeswitch settings and I am not certain how to proceed.

 

I originally created a Main IVR (ext 5555) and sent the incoming FXO ATA
calls to the IVR extension.   Now I want to send the calls to some other
extension or inbound route so that the calls are routed through all the
Freeswitch  settings such as holidays and time of day  conditions before
being sent to the IVR or some after hours message.  I'm not certain how I
should setup the FXO ATA to send the calls so Freeswitch processes them
properly?    Please advise.

 

cjb


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

 

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From avi at avimarcus.net  Tue Mar 15 17:05:04 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 15 Mar 2011 16:05:04 +0200
Subject: [Freeswitch-users] Multiconference with g729 users only without
	transcoding
In-Reply-To: <AANLkTikcKAuJVuyXYnf=GQsDkrEZEAEHaXar8w5X660A@mail.gmail.com>
References: <738A792011F8794D8CFAE25C00089679DC1AC9@quarea-correu.quarea.com>
	<AANLkTikgZtsViW3U5tXgGgbiq3z9mN0jcffJggWbC8b1@mail.gmail.com>
	<738A792011F8794D8CFAE25C00089679DC1AD3@quarea-correu.quarea.com>
	<4D7F6957.9060803@coppice.org>
	<AANLkTikcKAuJVuyXYnf=GQsDkrEZEAEHaXar8w5X660A@mail.gmail.com>
Message-ID: <AANLkTik9=+uZ+zjdq=2M5yKzFjbrP5ZoDn6ZwUq1VCge@mail.gmail.com>

MAYBE if you would *strictly* have only one speaker at a time, with no
volume adjustments etc, then maybe you can hack something together about a
conference in g729 w/o transcoding. Maybe.

-Avi
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From brian at freeswitch.org  Tue Mar 15 17:06:46 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 15 Mar 2011 09:06:46 -0500
Subject: [Freeswitch-users] Multiconference with g729 users only without
	transcoding
In-Reply-To: <AANLkTik9=+uZ+zjdq=2M5yKzFjbrP5ZoDn6ZwUq1VCge@mail.gmail.com>
References: <738A792011F8794D8CFAE25C00089679DC1AC9@quarea-correu.quarea.com>
	<AANLkTikgZtsViW3U5tXgGgbiq3z9mN0jcffJggWbC8b1@mail.gmail.com>
	<738A792011F8794D8CFAE25C00089679DC1AD3@quarea-correu.quarea.com>
	<4D7F6957.9060803@coppice.org>
	<AANLkTikcKAuJVuyXYnf=GQsDkrEZEAEHaXar8w5X660A@mail.gmail.com>
	<AANLkTik9=+uZ+zjdq=2M5yKzFjbrP5ZoDn6ZwUq1VCge@mail.gmail.com>
Message-ID: <FADE6D4D-091A-4D21-BE65-F90DCEE56CC3@freeswitch.org>

The answer would be NO.

/b

On Mar 15, 2011, at 9:05 AM, Avi Marcus wrote:

> MAYBE if you would *strictly* have only one speaker at a time, with no
> volume adjustments etc, then maybe you can hack something together about a
> conference in g729 w/o transcoding. Maybe.
> 
> -Avi
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From chistyakov at directtel.ru  Tue Mar 15 17:03:05 2011
From: chistyakov at directtel.ru (=?UTF-8?B?0KfQuNGB0YLRj9C60L7QsiDQmNCy0LDQvQ==?=)
Date: Tue, 15 Mar 2011 17:03:05 +0300
Subject: [Freeswitch-users] modify ip and port of rtp server in sdp data
In-Reply-To: <AANLkTimSUNwACbwgLiysoQi0DzAqiac2czB=JqE0H4nC@mail.gmail.com>
References: <4D7F4BAA.50107@directtel.ru>	<AANLkTikPyCTOw5AM1sHDBTsRwR7Xnxt11D=r4Q1PmDU7@mail.gmail.com>	<AANLkTi=NHZyWOvW0s2pW=eaU5BFSKWcB5z6N9FNoJ2a0@mail.gmail.com>	<4D7F5DCE.7090700@directtel.ru>
	<AANLkTimSUNwACbwgLiysoQi0DzAqiac2czB=JqE0H4nC@mail.gmail.com>
Message-ID: <4D7F7199.8050903@directtel.ru>

Yes, I understand it.
I am using bypass mode for disable RTP stack of FreeSWITCH.

I am planning to replace ip and port of RTP server in SDP data before 
bridging manualy (in LUA script) for streaming RTP data via rtpproxy 
instead directly between subcribers.

15.03.2011 16:28, Steven Ayre ?????:
>
>     I think  ext-rtp-ip not using in bypass mode.
>
>
> No, it isn't. In bypass mode you will not be able to do what you want 
> - the SDP from either endpoint will be sent to the other, they'll only 
> see the IP of the other endpoint.
>
> -Steve
>
>
>
> On 15 March 2011 12:38, ???????? ???? <chistyakov at directtel.ru 
> <mailto:chistyakov at directtel.ru>> wrote:
>
>     I am writing a dialplan-application (in LUA) for managing of all
>     calls in FreeSWITCH, and I am planning to use rtpproxy control
>     protocol for managing of RTP streams between subscribers.
>
>     http://www.rtpproxy.org/wiki/RTPproxy/Protocol
>
>
>     I think  ext-rtp-ip not using in bypass mode.
>
>
>     15.03.2011 15:01, Steven Ayre ?????:
>>     How do you plan to tell the rtpproxy where to send the media?
>>
>>     -Steve
>>
>>
>>     On 15 March 2011 11:53, Steven Ayre <steveayre at gmail.com
>>     <mailto:steveayre at gmail.com>> wrote:
>>
>>         Setting ext-rtp-ip on the sip profile might work.
>>
>>         -Steve
>>
>>
>>
>>         On 15 March 2011 11:21, ???????? ????
>>         <chistyakov at directtel.ru <mailto:chistyakov at directtel.ru>> wrote:
>>
>>             how can i modify ip and port of rtp server in sdp data on
>>             both legs
>>             before bridging for using external rtpproxy ?
>>
>>             _______________________________________________
>>             FreeSWITCH-users mailing list
>>             FreeSWITCH-users at lists.freeswitch.org
>>             <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>             http://www.freeswitch.org
>>
>>
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From steveu at coppice.org  Tue Mar 15 17:19:20 2011
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 15 Mar 2011 22:19:20 +0800
Subject: [Freeswitch-users] Multiconference with g729 users only without
 transcoding
In-Reply-To: <AANLkTik9=+uZ+zjdq=2M5yKzFjbrP5ZoDn6ZwUq1VCge@mail.gmail.com>
References: <738A792011F8794D8CFAE25C00089679DC1AC9@quarea-correu.quarea.com>	<AANLkTikgZtsViW3U5tXgGgbiq3z9mN0jcffJggWbC8b1@mail.gmail.com>	<738A792011F8794D8CFAE25C00089679DC1AD3@quarea-correu.quarea.com>	<4D7F6957.9060803@coppice.org>	<AANLkTikcKAuJVuyXYnf=GQsDkrEZEAEHaXar8w5X660A@mail.gmail.com>
	<AANLkTik9=+uZ+zjdq=2M5yKzFjbrP5ZoDn6ZwUq1VCge@mail.gmail.com>
Message-ID: <4D7F7568.2050302@coppice.org>

On 03/15/2011 10:05 PM, Avi Marcus wrote:
> MAYBE if you would *strictly* have only one speaker at a time, with no 
> volume adjustments etc, then maybe you can hack something together 
> about a conference in g729 w/o transcoding. Maybe.
You can extract the energy from the encoded frames, but knowing which 
energy to choose is another matter. Just taking the greatest energy will 
lead to rapid hoping amongst speakers. If you are more relaxed about 
switching speaker you will miss the start of voice spurts. The loudest 
might not even be the most relevant speaker.

Steve



From pkelly at gmail.com  Tue Mar 15 17:30:54 2011
From: pkelly at gmail.com (Pete Kelly)
Date: Tue, 15 Mar 2011 14:30:54 +0000
Subject: [Freeswitch-users] Memory leak
In-Reply-To: <AANLkTin9XpLcop+QnPFE70GSnEEHx5_yxWFhDrEJDBvq@mail.gmail.com>
References: <AANLkTim3UsX4z_URe1Ya40uP4AnVG3s3NiYkew=ZY8+O@mail.gmail.com>
	<AANLkTincGerDeJtkyqzYXFmtTzjHe-h9RMZJsM5HosCS@mail.gmail.com>
	<AANLkTimLNa73o5Y_h6KUpTptNyX6Q-6bpBOQBjLtsnS6@mail.gmail.com>
	<AANLkTinxMmO8UCR6wjROjB=DoD_yUoe2EuBNV=rNb3AC@mail.gmail.com>
	<AANLkTin9XpLcop+QnPFE70GSnEEHx5_yxWFhDrEJDBvq@mail.gmail.com>
Message-ID: <AANLkTinRceRe-VmN0a8=UXpqUvSs=ek2iLdviP+KxW81@mail.gmail.com>

I have gone with Freeswitch's inbuilt odbc implementation, and I have the _r
libs in use for the drivers.

Fingers crossed it helps matters, should know in a few days.

On 14 March 2011 16:25, Anthony Minessale <anthony.minessale at gmail.com>wrote:

> also make sure to use the _r version of any libs you use such as
> myodbc or mysqlclient
>
> look in /etc/odbcinst.ini and chances are its not the _r one
>
> # Driver from the MyODBC package
> # Setup from the unixODBC package
> [MySQL]
> Description     = ODBC for MySQL
> Driver          = /usr/lib64/libmyodbc3.so
> Setup           = /usr/lib64/libodbcmyS.so
> FileUsage       = 1
>
>
> /usr/lib64/libmyodbc3.so needs to be /usr/lib64/libmyodbc3_r.so
>
>
>
>
> On Mon, Mar 14, 2011 at 10:27 AM, Steven Ayre <steveayre at gmail.com> wrote:
> > Try using the freeswitch.Dbh interface:
> > http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh
> > It gives you a ODBC interface through freeswitch's odbc support, plus you
> > get the benefit of the connection caching.
> > -Steve
> >
> >
> > On 14 March 2011 14:28, Pete Kelly <pkelly at gmail.com> wrote:
> >>
> >>
> >> On 14 March 2011 13:16, Steven Ayre <steveayre at gmail.com> wrote:
> >>>
> >>> If you use mod_lcr, we were getting a memory leak in that a few weeks
> ago
> >>> which is fixed in the latest git.
> >>
> >> No we are not. I am going to try strengthening up the code which closes
> >> the DB connections. If that fails then I will compile and try the ODBC
> >> driver.
> >>
> >>>
> >>> -Steve
> >>>
> >>> On 14 March 2011 09:30, Pete Kelly <pkelly at gmail.com> wrote:
> >>>>
> >>>> Hi
> >>>> We are using Freeswitch (from git sources) in a production environment
> >>>> to handle an IVR system using a simple set of dialplans and lua
> scripts.
> >>>> OpenSIPS 1.6 and mysql are also installed and running as part of the
> same
> >>>> IVR application.
> >>>> However over time (approx 1 week) the memory usage of Freeswitch
> >>>> increases, eventually creeping into swap space forcing us to restart
> >>>> Freeswitch. The IVR handles approx 40 concurrent calls constantly, and
> we
> >>>> have instances of the IVR running on a 32bit Etch and 64bit Lenny
> boxes -
> >>>> both exhibit the same issue.
> >>>> We have also tried the Sangoma freeswitch branch in case that
> contained
> >>>> any fixes, however we see the same issues.
> >>>> Can anybody advise if this is a known issue with Freeswitch? Is it a
> >>>> memory leak or is it possible to limit the amount of memory Freeswitch
> will
> >>>> use, forcing it to garbage collect when it reaches the limit?
> >>>> Any advice on where to look/debug next would be appreciated.
> >>>> Thanks
> >>>> Pete
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mario_fs at mgtech.com  Tue Mar 15 17:44:39 2011
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 15 Mar 2011 07:44:39 -0700
Subject: [Freeswitch-users] How to stop SIP auth challenge (REGISTER)
	msgs?
In-Reply-To: <4443D92A-C567-49B0-B8B0-6EF516CA44E2@mgtech.com>
References: <6DB632C2-BC5B-4092-BA67-626ED8729DCD@mgtech.com>
	<AANLkTinCCj1SrDwBD5YFY6-V5_sJGjEU0QOi5SWgmFTw@mail.gmail.com>
	<4443D92A-C567-49B0-B8B0-6EF516CA44E2@mgtech.com>
Message-ID: <C75573D8-6F87-4DFD-BF6C-2C24CCE090B3@mgtech.com>

Setting to false in the internal profile worked, the default was true. Thanks very much! That was real pain during debugging.
Mario G

On Mar 14, 2011, at 2:08 PM, Mario G wrote:

> Thanks! Will give it a try and let you know.
> Mario G
> 
> On Mar 14, 2011, at 2:04 PM, Steven Ayre wrote:
> 
>> Set the SIP profile param log-auth-failures to false. I *think* it's default (if it's not set) is also false.
>> 
>> The warning level is to make sure they show up in logs at a high log level so that you can use them with fail2ban.
>> 
>> -Steve
>> 
>> 
>> On 14 March 2011 18:08, Mario G <mario_fs at mgtech.com> wrote:
>> I see this was introduced in a recent git. I know they are not an error but does anyone know if there is a way to stop these? Something I need to change on the phones? These are really a problem while trying to working on problems. The log is filled with these. Thanks.
>> 
>> WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [200 at 100.200.1.7] from ip 100.200.1.12
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From steveayre at gmail.com  Tue Mar 15 17:44:31 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 15 Mar 2011 14:44:31 +0000
Subject: [Freeswitch-users] Multiconference with g729 users only without
	transcoding
In-Reply-To: <AANLkTik9=+uZ+zjdq=2M5yKzFjbrP5ZoDn6ZwUq1VCge@mail.gmail.com>
References: <738A792011F8794D8CFAE25C00089679DC1AC9@quarea-correu.quarea.com>
	<AANLkTikgZtsViW3U5tXgGgbiq3z9mN0jcffJggWbC8b1@mail.gmail.com>
	<738A792011F8794D8CFAE25C00089679DC1AD3@quarea-correu.quarea.com>
	<4D7F6957.9060803@coppice.org>
	<AANLkTikcKAuJVuyXYnf=GQsDkrEZEAEHaXar8w5X660A@mail.gmail.com>
	<AANLkTik9=+uZ+zjdq=2M5yKzFjbrP5ZoDn6ZwUq1VCge@mail.gmail.com>
Message-ID: <AANLkTimtFwvq4F3F_HK1xRLnURYZcxMu3EgRufKpsNSd@mail.gmail.com>

You mean a "bridge"? ;)

If you plan always to do it like that, a bridge or call queues is better.
There's nothing in mod_conference that'd allow you to do that and it
wouldn't be worth the complexity to implement.

-Steve


On 15 March 2011 14:05, Avi Marcus <avi at avimarcus.net> wrote:

> MAYBE if you would *strictly* have only one speaker at a time, with no
> volume adjustments etc, then maybe you can hack something together about a
> conference in g729 w/o transcoding. Maybe.
>
> -Avi
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Tue Mar 15 17:46:06 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 15 Mar 2011 14:46:06 +0000
Subject: [Freeswitch-users] modify ip and port of rtp server in sdp data
In-Reply-To: <4D7F7199.8050903@directtel.ru>
References: <4D7F4BAA.50107@directtel.ru>
	<AANLkTikPyCTOw5AM1sHDBTsRwR7Xnxt11D=r4Q1PmDU7@mail.gmail.com>
	<AANLkTi=NHZyWOvW0s2pW=eaU5BFSKWcB5z6N9FNoJ2a0@mail.gmail.com>
	<4D7F5DCE.7090700@directtel.ru>
	<AANLkTimSUNwACbwgLiysoQi0DzAqiac2czB=JqE0H4nC@mail.gmail.com>
	<4D7F7199.8050903@directtel.ru>
Message-ID: <AANLkTin-+g9WXBt4XQoM4hpBVQ23cMNr9EY=2h2PE1Xs@mail.gmail.com>

Out of curosity, why do you want to use rtpproxy instead of the FreeSWITCH
RTP stack?

-Steve


On 15 March 2011 14:03, ???????? ???? <chistyakov at directtel.ru> wrote:

>  Yes, I understand it.
> I am using bypass mode for disable RTP stack of FreeSWITCH.
>
> I am planning to replace ip and port of RTP server in SDP data before
> bridging manualy (in LUA script) for streaming RTP data via rtpproxy instead
> directly between subcribers.
>
> 15.03.2011 16:28, Steven Ayre ?????:
>
> I think  ext-rtp-ip not using in bypass mode.
>>
>
> No, it isn't. In bypass mode you will not be able to do what you want - the
> SDP from either endpoint will be sent to the other, they'll only see the IP
> of the other endpoint.
>
> -Steve
>
>
>
> On 15 March 2011 12:38, ???????? ???? <chistyakov at directtel.ru> wrote:
>
>>  I am writing a dialplan-application (in LUA) for managing of all calls in
>> FreeSWITCH, and I am planning to use rtpproxy control protocol for managing
>> of RTP streams between subscribers.
>>
>> http://www.rtpproxy.org/wiki/RTPproxy/Protocol
>>
>>
>> I think  ext-rtp-ip not using in bypass mode.
>>
>>
>> 15.03.2011 15:01, Steven Ayre ?????:
>>
>> How do you plan to tell the rtpproxy where to send the media?
>>
>> -Steve
>>
>>
>> On 15 March 2011 11:53, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> Setting ext-rtp-ip on the sip profile might work.
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 15 March 2011 11:21, ???????? ???? <chistyakov at directtel.ru> wrote:
>>>
>>>> how can i modify ip and port of rtp server in sdp data on both legs
>>>> before bridging for using external rtpproxy ?
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gcd at i.ph  Tue Mar 15 17:54:23 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Tue, 15 Mar 2011 22:54:23 +0800
Subject: [Freeswitch-users] How to route ata question
In-Reply-To: <00f701cbe318$ee4f3640$caeda2c0$@accra.ca>
References: <00f801cbe27d$ae479fb0$0ad6df10$@accra.ca>
	<AANLkTinx-AtTQki0xgTed1UX0B23vUf6DWVZssywgpi_@mail.gmail.com>
	<00f701cbe318$ee4f3640$caeda2c0$@accra.ca>
Message-ID: <AANLkTimw9peVXhDKCczfzVMkKpKW=YJo1Nvg1yVPgLx1@mail.gmail.com>

pls try this dialplan. the "open" variable is defined in default.xml

<extension name="office-hours-incoming-pstn" continue="true">
   <condition field="destination_number" expression="^5555$"/>
   <condition field="${open}" expression="true">
       <action application="answer"/>
       <action application="sleep" data="500"/>
       <action application="ivr" data="office-autoattendant"/>  <!-- prompts
only once. if no keys pressed, it bridges to Extn 1000 -->
       <action application="bridge" data="user/1000@${domain_name}"/>
   </condition>
</extension>

<extension name="off-hours-incoming-pstn" >
   <condition field="destination_number" expression="^5555$">
       <action application="bridge" data="user/1000@${domain_name}"/>
   </condition>
</extension>

hope it works.

-nandy

On Tue, Mar 15, 2011 at 9:57 PM, Charles Bujold <cjbujold at accra.ca> wrote:

> My configuration is I have 3 lines  1- voip connection used for long
> distance calling (in/out),  2-pots used for local calls and toll free
> calling.  We have 6 telephones/users.   The main lines are the pots lines
> and the VOIP is our 1-800 incoming line .    The pots are connected to the
> grandstream ht503 which are directly sent to the extension 5555 per the
> ?Unconditional Call forward to VOIP? option of the ht503.
>
>
>
> What I?m trying to do is have the incoming calls filtered for holidays and
> non-work hours so I can send the call to and after hour message.   I?m Not
> sure how to proceed to do this since the HT503 send all incoming calls to
> the main reception IVR 5555.
>
>
>
> cjb
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* March-14-11 7:32 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] How to route ata question
>
>
>
> Well, every call from your ATA to FS already goes through the dialplan. Can
> you describe your configuration a bit? For example, did you set up your ATA
> to be a user on FreeSWITCH? Also, how are you currently routing your calls
> to 5555/IVR?
>
>
>
> -MC
>
> On Mon, Mar 14, 2011 at 12:26 PM, Charles Bujold <cjbujold at accra.ca>
> wrote:
>
>
>
> My question is I have an ATA FXO (Grandstream ht503) that currently sends
> calls to an extension, I want the calls to be filtered by the other
> Freeswitch settings and I am not certain how to proceed.
>
>
>
> I originally created a Main IVR (ext 5555) and sent the incoming FXO ATA
> calls to the IVR extension.   Now I want to send the calls to some other
> extension or inbound route so that the calls are routed through all the
> Freeswitch  settings such as holidays and time of day  conditions before
> being sent to the IVR or some after hours message.  I?m not certain how I
> should setup the FXO ATA to send the calls so Freeswitch processes them
> properly?    Please advise.
>
>
>
> cjb
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chistyakov at directtel.ru  Tue Mar 15 18:13:49 2011
From: chistyakov at directtel.ru (=?UTF-8?B?0KfQuNGB0YLRj9C60L7QsiDQmNCy0LDQvQ==?=)
Date: Tue, 15 Mar 2011 18:13:49 +0300
Subject: [Freeswitch-users] modify ip and port of rtp server in sdp data
In-Reply-To: <AANLkTin-+g9WXBt4XQoM4hpBVQ23cMNr9EY=2h2PE1Xs@mail.gmail.com>
References: <4D7F4BAA.50107@directtel.ru>	<AANLkTikPyCTOw5AM1sHDBTsRwR7Xnxt11D=r4Q1PmDU7@mail.gmail.com>	<AANLkTi=NHZyWOvW0s2pW=eaU5BFSKWcB5z6N9FNoJ2a0@mail.gmail.com>	<4D7F5DCE.7090700@directtel.ru>	<AANLkTimSUNwACbwgLiysoQi0DzAqiac2czB=JqE0H4nC@mail.gmail.com>	<4D7F7199.8050903@directtel.ru>
	<AANLkTin-+g9WXBt4XQoM4hpBVQ23cMNr9EY=2h2PE1Xs@mail.gmail.com>
Message-ID: <4D7F822D.4020306@directtel.ru>

Maybe, it is not normal case, but I got task.

15.03.2011 17:46, Steven Ayre ?????:
> Out of curosity, why do you want to use rtpproxy instead of the 
> FreeSWITCH RTP stack?
>
> -Steve
>
>
> On 15 March 2011 14:03, ???????? ???? <chistyakov at directtel.ru 
> <mailto:chistyakov at directtel.ru>> wrote:
>
>     Yes, I understand it.
>     I am using bypass mode for disable RTP stack of FreeSWITCH.
>
>     I am planning to replace ip and port of RTP server in SDP data
>     before bridging manualy (in LUA script) for streaming RTP data via
>     rtpproxy instead directly between subcribers.
>
>     15.03.2011 16:28, Steven Ayre ?????:
>>
>>         I think  ext-rtp-ip not using in bypass mode.
>>
>>
>>     No, it isn't. In bypass mode you will not be able to do what you
>>     want - the SDP from either endpoint will be sent to the other,
>>     they'll only see the IP of the other endpoint.
>>
>>     -Steve
>>
>>
>>
>>     On 15 March 2011 12:38, ???????? ???? <chistyakov at directtel.ru
>>     <mailto:chistyakov at directtel.ru>> wrote:
>>
>>         I am writing a dialplan-application (in LUA) for managing of
>>         all calls in FreeSWITCH, and I am planning to use rtpproxy
>>         control protocol for managing of RTP streams between subscribers.
>>
>>         http://www.rtpproxy.org/wiki/RTPproxy/Protocol
>>
>>
>>         I think  ext-rtp-ip not using in bypass mode.
>>
>>
>>         15.03.2011 15:01, Steven Ayre ?????:
>>>         How do you plan to tell the rtpproxy where to send the media?
>>>
>>>         -Steve
>>>
>>>
>>>         On 15 March 2011 11:53, Steven Ayre <steveayre at gmail.com
>>>         <mailto:steveayre at gmail.com>> wrote:
>>>
>>>             Setting ext-rtp-ip on the sip profile might work.
>>>
>>>             -Steve
>>>
>>>
>>>
>>>             On 15 March 2011 11:21, ???????? ????
>>>             <chistyakov at directtel.ru
>>>             <mailto:chistyakov at directtel.ru>> wrote:
>>>
>>>                 how can i modify ip and port of rtp server in sdp
>>>                 data on both legs
>>>                 before bridging for using external rtpproxy ?
>>>
>>>                 _______________________________________________
>>>                 FreeSWITCH-users mailing list
>>>                 FreeSWITCH-users at lists.freeswitch.org
>>>                 <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>                 http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>>         _______________________________________________
>>>         FreeSWITCH-users mailing list
>>>         FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         http://www.freeswitch.org
>>
>>         _______________________________________________
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mario_fs at mgtech.com  Tue Mar 15 18:33:39 2011
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 15 Mar 2011 08:33:39 -0700
Subject: [Freeswitch-users] Possible bug commenting X-PRE-PROCESS
Message-ID: <A9075FBE-3868-4D96-8103-66501E67859B@mgtech.com>

The line below was in an external profile. I commented it thinking it would be ignored as such. However, it was still used and the file loaded. This drive me nuts until I moved the extip.xml file out of the directory and FreeSwitch noted errors on startup. The only way was to put a space as "X- PRE..". I can't image why you would want a comment to be processed as a normal statement. Git is 3/13/11.

 <!--<X-PRE-PROCESS cmd="include" data="extip.xml"/>  -->
   
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From lloyd.aloysius at gmail.com  Tue Mar 15 18:49:04 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 15 Mar 2011 11:49:04 -0400
Subject: [Freeswitch-users] Multi tenant and BLF
In-Reply-To: <AANLkTimwxsT9B7n1+8zzbRhLe2F4mKX3uJh3cb29S9CN@mail.gmail.com>
References: <AANLkTimwxsT9B7n1+8zzbRhLe2F4mKX3uJh3cb29S9CN@mail.gmail.com>
Message-ID: <AANLkTikg62nODXyjyF-rdxzE1VkpgNOQif_==-EjuwhE@mail.gmail.com>

I am request a help again how this [Multi tenant and BLF] can be done or can
we add a bounty to make this works.

Thanks again
Lloyd


On Sun, Mar 13, 2011 at 6:20 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> I could not make the BLF working in multi tenant environment. Any help is
> appreciated. Discussion with BKW, he suggest to use multiple profiles. But
> if I want to use this box for x number of tenants then I need to setup x no
> of profiles. I like to use one profile for all my tenants.
>
> Multi tenant settings works for months without any issue.
>
> Here is my lab setup
>
> *compa.mydomain.com*
>
> Ext 201 - Cisco 504G + SPA500S Expansion Module - BLF 202,203
> Ext 202 -SPA942
> Ext 203 -SPA942
>
> *BLF : Not working reliably .*
>
> *compb.mydomain.com*
>
> Ext 201 - Aastra 57i  and Topkeys Configured for BLF 202,203
> Ext 202 - Aastra 6731i
> Ext 203 - Aastra 6731
>
> *BLF: working for couple of hours then completely stop working. A reboot
> of the phone required*
>
>
> *internal profile - The following lines comments*
>
> <!--<param name="force-register-domain" value="$${domain}"/>-->
> <!--<param name="force-subscription-domain" value="$${domain}"/>-->
> <!--<param name="force-register-db-domain" value="$${domain}"/>-->
>
> ---
>
> Any help is appreciated how to make it work with single profile
>
>
> Thanks
> Lloyd
>
>
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From infos at madovsky.org  Tue Mar 15 18:51:05 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 15 Mar 2011 11:51:05 -0400
Subject: [Freeswitch-users] conference comfort noise: recommended setting
References: <AANLkTinqgtx6TaTDf6iWDkX2bAXMza1EM5pXyw9btwXF@mail.gmail.com><1300192831488-6172641.post@n2.nabble.com>
	<AANLkTi=-1HvxOjyjRjqoRdk-Um9MC1uR_Vofejfxdb9z@mail.gmail.com>
Message-ID: <07941AE2CF2648AEADB06C6F86F9F529@e1705>

he's right. we live now
in the 192khz audio era ;)

----- Original Message ----- 
From: "David Wafula" <davidwaf at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, March 15, 2011 8:53 AM
Subject: Re: [Freeswitch-users] conference comfort noise: recommended 
setting


> On Tue, Mar 15, 2011 at 2:40 PM, mazilo <Nabble at slickdeals.endjunk.com> 
> wrote:
>>
>> You're not kidding me with an 8,000KHz, R U?
>>
>
> Erm...meant 8Khz  : )    :
>
>
> -- 
> David Wafula
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From anthony.minessale at gmail.com  Tue Mar 15 19:10:16 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 15 Mar 2011 11:10:16 -0500
Subject: [Freeswitch-users] Possible bug commenting X-PRE-PROCESS
In-Reply-To: <A9075FBE-3868-4D96-8103-66501E67859B@mgtech.com>
References: <A9075FBE-3868-4D96-8103-66501E67859B@mgtech.com>
Message-ID: <AANLkTikQOJ5EhDc=2FocMTXzOmHq71yKp1+2-3j7MnMC@mail.gmail.com>

you cannot comment those.
They are not real xml they are designed to be ignored by XML and
parsed by the pre-processor and are parsed before the xml parser.
All the xml parser would see on your example is <!-- -->
if you want to comment them, change X- to Z- or "cmd" to "comment"


On Tue, Mar 15, 2011 at 10:33 AM, Mario G <mario_fs at mgtech.com> wrote:
> The line below was in an external profile. I commented it thinking it would
> be ignored as such. However, it was still used and the file loaded. This
> drive me nuts until I moved the extip.xml file out of the directory and
> FreeSwitch noted errors on startup. The only way was to put a space as "X-
> PRE..". I can't image why you would want a comment to be processed as a
> normal statement. Git is 3/13/11.
> ?<!--<X-PRE-PROCESS cmd="include" data="extip.xml"/>? -->
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 brad at tritelcomm.com  Tue Mar 15 19:22:38 2011
From: brad at tritelcomm.com (Brad Mina)
Date: Tue, 15 Mar 2011 09:22:38 -0700
Subject: [Freeswitch-users] debug down trunk
In-Reply-To: <E62D9A8F4352426BAD5540CCAEFD64DC@e1705>
References: <60BA130CDBF44BD69A43BA7777AE83BD@e1705>
	<AANLkTi=ETiNk4bArqj8VS8nN9j=+LkcwAc9qygN-zkSk@mail.gmail.com>
	<BE6B9A96642745228DAA6ED4E06102CC@e1705>
	<E62D9A8F4352426BAD5540CCAEFD64DC@e1705>
Message-ID: <AANLkTimkfAURUWKOOMnBDr-9ze8jYg42+-5pK7S_Q+Mk@mail.gmail.com>

I was having a similar problem with SipStation trunks (never fully got it
resolved before I changed providers).
FS would spit out tons of reg packets but receive no response from the other
end, instead it would timeout then re-register as if nothing was wrong.

I was enlightened to the use of the following line in my provider
configuration (however, please keep in mind my problems continued):

    <param name="extension-in-contact" value="true"/>




On Mon, Mar 14, 2011 at 5:10 PM, Madovsky <infos at madovsky.org> wrote:

>  oopss, sorry I sent the send message...
> apparently their server doesn't answer..
> some 10 sends are made without answer
>
> ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, March 14, 2011 8:01 PM
> *Subject:* Re: [Freeswitch-users] debug down trunk
>
> thanks Steve,
>
> here the trace :
>
>
> ------------------------------------------------------------------------
> send 643 bytes to udp/[10.10.10.10]:5060 at 23:57:10.312821:
> ------------------------------------------------------------------------
> OPTIONS sip:10.10.10.10;transport=udp SIP/2.0
> Via: SIP/2.0/UDP 30.30.30.30;rport;branch=z9hG4bKD38154a4e7rXa
> Max-Forwards: 70
> From: *<*sip:freeswitch@*10*.*10*.*10*.*10*;transport=udp*>*
> ;tag=eB6B7NSge90vK
> To: *<*sip:freeswitch@*10*.*10*.*10*.*10*;transport=udp*>*
> Call-ID: 974aa6c8-c939-122e-45ab-00e0ed0b00c2
> CSeq: 9737404 OPTIONS
> Contact: *<*sip:gw+trunk_3@*30*.*30*.*30*.*30*:*5060*
> ;tport=tcp;transport=udp;gw=trunk_3*>*
> User-Agent: CiscoSystems-SIP-GW-UA
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REFER,
> NOTIFY
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, refer
> Content-Length: 0
>
> ------------------------------------------------------------------------
>
> weird because it looks like exactly the same response as the other trunks
> but
> it states to down....
>
> ----- Original Message -----
> *From:* Steven Ayre <steveayre at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, March 14, 2011 5:06 PM
> *Subject:* Re: [Freeswitch-users] debug down trunk
>
> 408 is the error code - this means "408 Request timed out"
>
> FS tries sending an OPTIONS request to the gateway and waits for a reply to
> test whether it is online or not..
>
> You can use the siptrace to see those messages and the reply:
> sofia global siptrace on
>
> -Steve
>
>
> On 14 March 2011 19:03, Madovsky <infos at madovsky.org> wrote:
>
>>  I set external.xml to debug=1 and sip trace to 1 and level debug to 7
>> but I can get only this in the log
>>
>> sofia.c:4028 Ping failed trunk_3 with code 408 - count -1/-1/1, state DOWN
>>
>> so I don't know why this trunk is down.
>> I contacted them and they said that my account is working.
>>
>> Thanks
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mario_fs at mgtech.com  Tue Mar 15 19:36:16 2011
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 15 Mar 2011 09:36:16 -0700
Subject: [Freeswitch-users] Possible bug commenting X-PRE-PROCESS
In-Reply-To: <AANLkTikQOJ5EhDc=2FocMTXzOmHq71yKp1+2-3j7MnMC@mail.gmail.com>
References: <A9075FBE-3868-4D96-8103-66501E67859B@mgtech.com>
	<AANLkTikQOJ5EhDc=2FocMTXzOmHq71yKp1+2-3j7MnMC@mail.gmail.com>
Message-ID: <1CEEC760-86B4-4EF9-865F-693F48EEE670@mgtech.com>

Wish I knew that, it cost me 3 days.... It turned out to be the reason my phones would drop when placing a call on hold. Still have no idea why it affected the SPA962s. I am noting it in case someone else runs across something like this. Thanks for the explanation!
Mario G

On Mar 15, 2011, at 9:10 AM, Anthony Minessale wrote:

> you cannot comment those.
> They are not real xml they are designed to be ignored by XML and
> parsed by the pre-processor and are parsed before the xml parser.
> All the xml parser would see on your example is <!-- -->
> if you want to comment them, change X- to Z- or "cmd" to "comment"
> 
> 
> On Tue, Mar 15, 2011 at 10:33 AM, Mario G <mario_fs at mgtech.com> wrote:
>> The line below was in an external profile. I commented it thinking it would
>> be ignored as such. However, it was still used and the file loaded. This
>> drive me nuts until I moved the extip.xml file out of the directory and
>> FreeSwitch noted errors on startup. The only way was to put a space as "X-
>> PRE..". I can't image why you would want a comment to be processed as a
>> normal statement. Git is 3/13/11.
>>  <!--<X-PRE-PROCESS cmd="include" data="extip.xml"/>  -->
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Tue Mar 15 19:39:29 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 15 Mar 2011 11:39:29 -0500
Subject: [Freeswitch-users] Multi tenant and BLF
In-Reply-To: <AANLkTimwxsT9B7n1+8zzbRhLe2F4mKX3uJh3cb29S9CN@mail.gmail.com>
References: <AANLkTimwxsT9B7n1+8zzbRhLe2F4mKX3uJh3cb29S9CN@mail.gmail.com>
Message-ID: <AANLkTikv_FRDrKy1Uq3aQN=PCB=4Fd+6i+CG_ZSdf4ST@mail.gmail.com>

You need to alias every domain into your profile as well.  (See the
alias tag in the example profile)
This allows the domains to reverse map to the profile.
You also must 100% use hostnames in all of your sip phones for the
domain field.  You can still set the proxy addr to an ip but you must
have the domain present in the sip headers.


On Sun, Mar 13, 2011 at 5:20 PM, Aloysius Lloyd
<lloyd.aloysius at gmail.com> wrote:
> Hi All,
> I could not make the BLF working in?multi tenant?environment. Any help is
> appreciated. Discussion with BKW, he suggest to use multiple profiles. But
> if I want to use this box for x number of?tenants?then I need to setup x no
> of profiles. I like to use one profile for all my tenants.
> Multi tenant?settings works for months without any issue.
> Here is my lab setup
> compa.mydomain.com
> Ext 201 - Cisco 504G + SPA500S Expansion Module - BLF 202,203
> Ext 202 -SPA942
> Ext 203 -SPA942
> BLF : Not working?reliably?.
> compb.mydomain.com
> Ext 201 - Aastra 57i ?and Topkeys Configured for BLF 202,203
> Ext 202 - Aastra 6731i
> Ext 203 -?Aastra 6731
> BLF: working for couple of hours then?completely?stop working. A reboot of
> the phone required
>
> internal profile -?The following lines comments
> <!--<param name="force-register-domain" value="$${domain}"/>-->
> <!--<param name="force-subscription-domain" value="$${domain}"/>-->
> <!--<param name="force-register-db-domain" value="$${domain}"/>-->
> ---
> Any help is appreciated how to make it work with single profile
>
> Thanks
> Lloyd
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 edpimentl at gmail.com  Tue Mar 15 21:59:00 2011
From: edpimentl at gmail.com (EdPimentl)
Date: Tue, 15 Mar 2011 14:59:00 -0400
Subject: [Freeswitch-users] Uncovering spoken phrases in *encrypted* VoIP
	conversations
Message-ID: <AANLkTi=Lj6VajMWP1c3SOWLS+JsLWLXHKtJ22NSmi27n@mail.gmail.com>

New recent study at *(Johns Hopkins / UNC)* says it can *Uncover spoken
phrases in *encrypted* VoIP conversations
*
*(Johns Hopkins / UNC) ACM paper*

    We evaluate our techniques on a standard speech recognition corpus
    containing over 2,000 phonetically rich phrases spoken by 630
    distinct speakers from across the continental United States. Our
    results indicate that we can identify phrases within encrypted calls
    with an average accuracy of 50%, and with accuracy greater than 90%
    for some phrases. Clearly, such an attack calls into question the
    efficacy of current VoIP encryption standards.
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From kris at kriskinc.com  Tue Mar 15 22:36:46 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 15 Mar 2011 15:36:46 -0400
Subject: [Freeswitch-users] Uncovering spoken phrases in *encrypted*
	VoIP conversations
In-Reply-To: <AANLkTi=Lj6VajMWP1c3SOWLS+JsLWLXHKtJ22NSmi27n@mail.gmail.com>
References: <AANLkTi=Lj6VajMWP1c3SOWLS+JsLWLXHKtJ22NSmi27n@mail.gmail.com>
Message-ID: <AANLkTikqV-B3b3HE=0LpDrAN3-00TZv9+Qhh5UvKOwNH@mail.gmail.com>

More scaremonger tactics from paper authors and journalists...

This is for variable bitrate codecs only.  Weaknesses in encryption
for non-padded VBR codecs have been known for some time.

On Tue, Mar 15, 2011 at 2:59 PM, EdPimentl <edpimentl at gmail.com> wrote:
> New recent study at (Johns Hopkins / UNC) says it can Uncover spoken phrases
> in *encrypted* VoIP conversations
>
> (Johns Hopkins / UNC) ACM paper
>
> ? ? We evaluate our techniques on a standard speech recognition corpus
> ? ? containing over 2,000 phonetically rich phrases spoken by 630
> ? ? distinct speakers from across the continental United States. Our
> ? ? results indicate that we can identify phrases within encrypted calls
> ? ? with an average accuracy of 50%, and with accuracy greater than 90%
> ? ? for some phrases. Clearly, such an attack calls into question the
> ? ? efficacy of current VoIP encryption standards.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner


From msc at freeswitch.org  Tue Mar 15 23:34:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 15 Mar 2011 13:34:36 -0700
Subject: [Freeswitch-users] Multiconference with g729 users only without
	transcoding
In-Reply-To: <738A792011F8794D8CFAE25C00089679DC1AC9@quarea-correu.quarea.com>
References: <738A792011F8794D8CFAE25C00089679DC1AC9@quarea-correu.quarea.com>
Message-ID: <AANLkTi=xUfDMA+9aJCHgQoNGdyp_cziJe-m7bReUg-NB@mail.gmail.com>

On Tue, Mar 15, 2011 at 5:21 AM, Llu?s Riera <lluis.riera at quarea.com> wrote:

>  Hi,
>
>
>
> I would like to know if is there a configuration in freeswitch to allow
> multiconferencing for g729 users but without transcoding? It seems that all
> users are converted to L16 and it?s not possible to have passthrough mode in
> conference rooms.
>
>
>

Yeah, this is a bad idea for all the reasons that the others have stated. It
is SOOO much easier just to shell out $10 per channel and get the g729
licenses.
-MC
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From moises.silva at gmail.com  Tue Mar 15 23:38:36 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 15 Mar 2011 16:38:36 -0400
Subject: [Freeswitch-users] Possible bug commenting X-PRE-PROCESS
In-Reply-To: <1CEEC760-86B4-4EF9-865F-693F48EEE670@mgtech.com>
References: <A9075FBE-3868-4D96-8103-66501E67859B@mgtech.com>
	<AANLkTikQOJ5EhDc=2FocMTXzOmHq71yKp1+2-3j7MnMC@mail.gmail.com>
	<1CEEC760-86B4-4EF9-865F-693F48EEE670@mgtech.com>
Message-ID: <AANLkTim521FfOUAB-+HFsy_TTGTt6kGYJapNKOT-w9mG@mail.gmail.com>

On Tue, Mar 15, 2011 at 12:36 PM, Mario G <mario_fs at mgtech.com> wrote:

> Wish I knew that, it cost me 3 days.... It turned out to be the reason my
> phones would drop when placing a call on hold. Still have no idea why it
> affected the SPA962s. I am noting it in case someone else runs across
> something like this. Thanks for the explanation!
>

Also note that the behavior of pre-processor variables is documented in
vars.xml sample configuration.

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
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From moises.silva at gmail.com  Tue Mar 15 23:40:16 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 15 Mar 2011 16:40:16 -0400
Subject: [Freeswitch-users] Multiconference with g729 users only without
	transcoding
In-Reply-To: <AANLkTi=xUfDMA+9aJCHgQoNGdyp_cziJe-m7bReUg-NB@mail.gmail.com>
References: <738A792011F8794D8CFAE25C00089679DC1AC9@quarea-correu.quarea.com>
	<AANLkTi=xUfDMA+9aJCHgQoNGdyp_cziJe-m7bReUg-NB@mail.gmail.com>
Message-ID: <AANLkTin5t5gkE90icHFYvtaz=3M9tp2GpbwecZz3J3ck@mail.gmail.com>

>
> On Tue, Mar 15, 2011 at 5:21 AM, Llu?s Riera <lluis.riera at quarea.com>wrote:
>
>>  Hi,
>>
>>
>>
>> I would like to know if is there a configuration in freeswitch to allow
>> multiconferencing for g729 users but without transcoding? It seems that all
>> users are converted to L16 and it?s not possible to have passthrough mode in
>> conference rooms.
>>
>>
>>
>
You need to mix the audio from all the parties, not just pass it thru,
therefore, transcoding to linear is needed.

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
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From msc at freeswitch.org  Tue Mar 15 23:42:11 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 15 Mar 2011 13:42:11 -0700
Subject: [Freeswitch-users] Uncovering spoken phrases in *encrypted*
	VoIP conversations
In-Reply-To: <AANLkTikqV-B3b3HE=0LpDrAN3-00TZv9+Qhh5UvKOwNH@mail.gmail.com>
References: <AANLkTi=Lj6VajMWP1c3SOWLS+JsLWLXHKtJ22NSmi27n@mail.gmail.com>
	<AANLkTikqV-B3b3HE=0LpDrAN3-00TZv9+Qhh5UvKOwNH@mail.gmail.com>
Message-ID: <AANLkTikifK_mkSiLEiJ52htGY2xoH3WWfzS7oYPDSFY0@mail.gmail.com>

On Tue, Mar 15, 2011 at 12:36 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> More scaremonger tactics from paper authors and journalists...
>
> This is for variable bitrate codecs only.  Weaknesses in encryption
> for non-padded VBR codecs have been known for some time.
>

Agreed. VBR is, by definition, giving away extra information, and it's been
know for a long time to do this. I'd like to see what they got out of
non-VBR encrypted streams. Still, I'm gonna check this out to see if there's
anything useful underneath the hair-on-fire claims.

-MC
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From lloyd.aloysius at sunteltech.ca  Tue Mar 15 23:42:43 2011
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Tue, 15 Mar 2011 16:42:43 -0400
Subject: [Freeswitch-users] Multi tenant and BLF
In-Reply-To: <AANLkTikv_FRDrKy1Uq3aQN=PCB=4Fd+6i+CG_ZSdf4ST@mail.gmail.com>
References: <AANLkTimwxsT9B7n1+8zzbRhLe2F4mKX3uJh3cb29S9CN@mail.gmail.com>
	<AANLkTikv_FRDrKy1Uq3aQN=PCB=4Fd+6i+CG_ZSdf4ST@mail.gmail.com>
Message-ID: <AANLkTim-KNxbq1Wcg_4oUPrQMqs_7=hs3OVCCJ+NVGQr@mail.gmail.com>

I add the alias in the internal.xml inside the alias tag.

 <alias name="compa.mydomain.com"/>
 <alias name="compb.mydomain.com"/>

Then restart the FreeSWITCH. sofia status shows like below

internal       profile          sip:mod_sofia at A.B.C.D:5060      RUNNING (0)
compa.mydomain.com         alias                                   internal
     ALIASED
compb.mydomain.com         alias                                   internal
     ALIASED


Then I dial out from the phone BLF works for 10 sec then switch off
automatically, but the phone still connected to the outside or any other
extension.

Phone configuration SPA942  and SPA504 G I use compa.mydomain.com   in the
proxy and outbound proxy fields. There is no IP involve in the phone
configuration.


Thanks
Lloyd

On Tue, Mar 15, 2011 at 12:39 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> You need to alias every domain into your profile as well.  (See the
> alias tag in the example profile)
> This allows the domains to reverse map to the profile.
> You also must 100% use hostnames in all of your sip phones for the
> domain field.  You can still set the proxy addr to an ip but you must
> have the domain present in the sip headers.
>
>
> On Sun, Mar 13, 2011 at 5:20 PM, Aloysius Lloyd
> <lloyd.aloysius at gmail.com> wrote:
> > Hi All,
> > I could not make the BLF working in multi tenant environment. Any help is
> > appreciated. Discussion with BKW, he suggest to use multiple profiles.
> But
> > if I want to use this box for x number of tenants then I need to setup x
> no
> > of profiles. I like to use one profile for all my tenants.
> > Multi tenant settings works for months without any issue.
> > Here is my lab setup
> > compa.mydomain.com
> > Ext 201 - Cisco 504G + SPA500S Expansion Module - BLF 202,203
> > Ext 202 -SPA942
> > Ext 203 -SPA942
> > BLF : Not working reliably .
> > compb.mydomain.com
> > Ext 201 - Aastra 57i  and Topkeys Configured for BLF 202,203
> > Ext 202 - Aastra 6731i
> > Ext 203 - Aastra 6731
> > BLF: working for couple of hours then completely stop working. A reboot
> of
> > the phone required
> >
> > internal profile - The following lines comments
> > <!--<param name="force-register-domain" value="$${domain}"/>-->
> > <!--<param name="force-subscription-domain" value="$${domain}"/>-->
> > <!--<param name="force-register-db-domain" value="$${domain}"/>-->
> > ---
> > Any help is appreciated how to make it work with single profile
> >
> > Thanks
> > Lloyd
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Mar 15 23:44:28 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 15 Mar 2011 13:44:28 -0700
Subject: [Freeswitch-users] Possible bug commenting X-PRE-PROCESS
In-Reply-To: <AANLkTim521FfOUAB-+HFsy_TTGTt6kGYJapNKOT-w9mG@mail.gmail.com>
References: <A9075FBE-3868-4D96-8103-66501E67859B@mgtech.com>
	<AANLkTikQOJ5EhDc=2FocMTXzOmHq71yKp1+2-3j7MnMC@mail.gmail.com>
	<1CEEC760-86B4-4EF9-865F-693F48EEE670@mgtech.com>
	<AANLkTim521FfOUAB-+HFsy_TTGTt6kGYJapNKOT-w9mG@mail.gmail.com>
Message-ID: <AANLkTikUiesRC1qFD_q3wRMx0+xB5ZyOxk9BsWmh=99d@mail.gmail.com>

On Tue, Mar 15, 2011 at 1:38 PM, Moises Silva <moises.silva at gmail.com>wrote:

> On Tue, Mar 15, 2011 at 12:36 PM, Mario G <mario_fs at mgtech.com> wrote:
>
>> Wish I knew that, it cost me 3 days.... It turned out to be the reason my
>> phones would drop when placing a call on hold. Still have no idea why it
>> affected the SPA962s. I am noting it in case someone else runs across
>> something like this. Thanks for the explanation!
>>
>
> Also note that the behavior of pre-processor variables is documented in
> vars.xml sample configuration.
>

Line #4, to be exact:
<include>
  <!-- Preprocessor Variables
       These are introduced when configuration strings must be consistent
across modules.
       *NOTICE: YOU CAN NOT COMMENT OUT AN X-PRE-PROCESS line, Remove the
line instead.*


-MC
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From benkokakao at gmail.com  Wed Mar 16 01:09:40 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Wed, 16 Mar 2011 00:09:40 +0200
Subject: [Freeswitch-users] German voice prompts - missing exceptions in
	mod_say_de.c
Message-ID: <AANLkTi=6Q6Q-=S-z9pKT8ofMyCQyB7LRZrrwUNB+KGf_@mail.gmail.com>

Hi!

Are there any other german voice-prompts available? The ones provided
by xpirio(http://freeswitch.xpirio.com/) have some flaws
unfortunately.

There's also a possible problem with the word "Eins"(one), as we need
3 different "flexions" of the word:
- Eine - e.g. in Voicemail for "one new message"(Female form)
- Einer - male form of "one" - i didn't see a use for it yet, but that
doesn't mean this case does not exist
- Eins - e.g. spelling of "1", this is the only form currently working
with type "items" in speech management

Additionally we have male, female and neuter versions of counted
numbers, this applies to all numbers, not just "one".
The english equivalent is "XXth", so we can cover one gender with the
h-XX.wav files. I'm not sure if female and neuter-versions of the
files are actually needed(Xpirio has recorded a neutrum-version of the
h-XX.wav files, but for e.g. say type "DATE/TIME" we would need the
male form)

I'm not a c-developer, so i might be wrong - but skimming through
mod_say_de.c i saw some exceptions, but not for these cases(Sorry in
advance if i'm wrong). Also i would not know how to tell the
"say"-application to use these special forms if they existed.

Is it possible to cover this/manipulate mod_say with "speech phrase
management"? Judging from the available documentation it seems not...

Other complications with german grammar might appear - i've just
started using the german sound-files today in the afternoon and tested
with voicemail mostly - but i would dig deeper and happily assist any
developer who wants to get his hands dirty - in real time via phone
and IM and 24h if needed(CET here).

Sorry again if i missed an existing solution, it's been a long day.

Best regards
Christian


From infos at madovsky.org  Wed Mar 16 01:35:00 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 15 Mar 2011 18:35:00 -0400
Subject: [Freeswitch-users] Possible bug commenting X-PRE-PROCESS
References: <A9075FBE-3868-4D96-8103-66501E67859B@mgtech.com><AANLkTikQOJ5EhDc=2FocMTXzOmHq71yKp1+2-3j7MnMC@mail.gmail.com>
	<1CEEC760-86B4-4EF9-865F-693F48EEE670@mgtech.com>
Message-ID: <925C2C70EF054D6A93A31BA26FCDC2AA@e1705>

i> Wish I knew that, it cost me 3 days....

it's only the start my friend ;)

----- Original Message ----- 
From: "Mario G" <mario_fs at mgtech.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, March 15, 2011 12:36 PM
Subject: Re: [Freeswitch-users] Possible bug commenting X-PRE-PROCESS


> Wish I knew that, it cost me 3 days.... It turned out to be the reason my 
> phones would drop when placing a call on hold. Still have no idea why it 
> affected the SPA962s. I am noting it in case someone else runs across 
> something like this. Thanks for the explanation!
> Mario G
>
> On Mar 15, 2011, at 9:10 AM, Anthony Minessale wrote:
>
>> you cannot comment those.
>> They are not real xml they are designed to be ignored by XML and
>> parsed by the pre-processor and are parsed before the xml parser.
>> All the xml parser would see on your example is <!-- -->
>> if you want to comment them, change X- to Z- or "cmd" to "comment"
>>
>>
>> On Tue, Mar 15, 2011 at 10:33 AM, Mario G <mario_fs at mgtech.com> wrote:
>>> The line below was in an external profile. I commented it thinking it 
>>> would
>>> be ignored as such. However, it was still used and the file loaded. This
>>> drive me nuts until I moved the extip.xml file out of the directory and
>>> FreeSwitch noted errors on startup. The only way was to put a space as 
>>> "X-
>>> PRE..". I can't image why you would want a comment to be processed as a
>>> normal statement. Git is 3/13/11.
>>>  <!--<X-PRE-PROCESS cmd="include" data="extip.xml"/>  -->
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From infos at madovsky.org  Wed Mar 16 01:36:16 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 15 Mar 2011 18:36:16 -0400
Subject: [Freeswitch-users] debug down trunk
References: <60BA130CDBF44BD69A43BA7777AE83BD@e1705><AANLkTi=ETiNk4bArqj8VS8nN9j=+LkcwAc9qygN-zkSk@mail.gmail.com><BE6B9A96642745228DAA6ED4E06102CC@e1705><E62D9A8F4352426BAD5540CCAEFD64DC@e1705>
	<AANLkTimkfAURUWKOOMnBDr-9ze8jYg42+-5pK7S_Q+Mk@mail.gmail.com>
Message-ID: <A7A7FCC4D66E4326BA9AB99A1A1C6D08@e1705>

thanks for your suggestion.
the problem was from their firewall....(grrrr)
  ----- Original Message ----- 
  From: Brad Mina 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, March 15, 2011 12:22 PM
  Subject: Re: [Freeswitch-users] debug down trunk


  I was having a similar problem with SipStation trunks (never fully got it resolved before I changed providers).
  FS would spit out tons of reg packets but receive no response from the other end, instead it would timeout then re-register as if nothing was wrong.


  I was enlightened to the use of the following line in my provider configuration (however, please keep in mind my problems continued):
    <param name="extension-in-contact" value="true"/>






  On Mon, Mar 14, 2011 at 5:10 PM, Madovsky <infos at madovsky.org> wrote:

    oopss, sorry I sent the send message...
    apparently their server doesn't answer..
    some 10 sends are made without answer
      ----- Original Message ----- 
      From: Madovsky 
      To: FreeSWITCH Users Help 
      Sent: Monday, March 14, 2011 8:01 PM
      Subject: Re: [Freeswitch-users] debug down trunk


      thanks Steve,

      here the trace :

      ------------------------------------------------------------------------
      send 643 bytes to udp/[10.10.10.10]:5060 at 23:57:10.312821:
      ------------------------------------------------------------------------
      OPTIONS sip:10.10.10.10;transport=udp SIP/2.0
      Via: SIP/2.0/UDP 30.30.30.30;rport;branch=z9hG4bKD38154a4e7rXa
      Max-Forwards: 70
      From: <sip:freeswitch at 10.10.10.10;transport=udp>;tag=eB6B7NSge90vK
      To: <sip:freeswitch at 10.10.10.10;transport=udp>
      Call-ID: 974aa6c8-c939-122e-45ab-00e0ed0b00c2
      CSeq: 9737404 OPTIONS
      Contact: <sip:gw+trunk_3 at 30.30.30.30:5060;tport=tcp;transport=udp;gw=trunk_3>
      User-Agent: CiscoSystems-SIP-GW-UA
      Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REFER, NOTIFY
      Supported: timer, precondition, path, replaces
      Allow-Events: talk, hold, refer
      Content-Length: 0


      ------------------------------------------------------------------------
      weird because it looks like exactly the same response as the other trunks but
      it states to down....

        ----- Original Message ----- 
        From: Steven Ayre 
        To: FreeSWITCH Users Help 
        Sent: Monday, March 14, 2011 5:06 PM
        Subject: Re: [Freeswitch-users] debug down trunk


        408 is the error code - this means "408 Request timed out"

        FS tries sending an OPTIONS request to the gateway and waits for a reply to test whether it is online or not..

        You can use the siptrace to see those messages and the reply:
        sofia global siptrace on

        -Steve



        On 14 March 2011 19:03, Madovsky <infos at madovsky.org> wrote:

          I set external.xml to debug=1 and sip trace to 1 and level debug to 7
          but I can get only this in the log

          sofia.c:4028 Ping failed trunk_3 with code 408 - count -1/-1/1, state DOWN

          so I don't know why this trunk is down.
          I contacted them and they said that my account is working.

          Thanks

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






------------------------------------------------------------------------


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




--------------------------------------------------------------------------


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



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






------------------------------------------------------------------------------


  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From casteven at gmail.com  Wed Mar 16 05:11:03 2011
From: casteven at gmail.com (Campbell Steven)
Date: Wed, 16 Mar 2011 15:11:03 +1300
Subject: [Freeswitch-users] English NZ or English UK Prompt Set Recording
Message-ID: <AANLkTikhqe1mC91VrJQj6tNSn5fPKh4zV6ANK4bBGEuS@mail.gmail.com>

Hi,

I'm trying to get a group of people together to contribute money
towards recording a standard set of New Zealand English prompts for
FreeSWITCH. I'd like to get a New Zealand set done, but if there isn't
enough interest I'd settle for a UK set and I'd have to take guidance
on companies who could do that since I'm based in NZ.

Anyway, let me know if you are interested in contributing money to a
prompt set (either UK or NZ), that would be given back to the
FreeSWITCH project.

Thanks

Campbell


From msc at freeswitch.org  Wed Mar 16 09:07:28 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 15 Mar 2011 23:07:28 -0700
Subject: [Freeswitch-users] English NZ or English UK Prompt Set Recording
In-Reply-To: <AANLkTikhqe1mC91VrJQj6tNSn5fPKh4zV6ANK4bBGEuS@mail.gmail.com>
References: <AANLkTikhqe1mC91VrJQj6tNSn5fPKh4zV6ANK4bBGEuS@mail.gmail.com>
Message-ID: <AANLkTinha-QesOjn583rCoNTGwLhcoywkKPGsNvi9gZu@mail.gmail.com>

Thanks for checking in! Be sure to check with Alex Crow here on the list as
he was working on getting a set of UK En prompts in a female voice.

-MC

On Tue, Mar 15, 2011 at 7:11 PM, Campbell Steven <casteven at gmail.com> wrote:

> Hi,
>
> I'm trying to get a group of people together to contribute money
> towards recording a standard set of New Zealand English prompts for
> FreeSWITCH. I'd like to get a New Zealand set done, but if there isn't
> enough interest I'd settle for a UK set and I'd have to take guidance
> on companies who could do that since I'm based in NZ.
>
> Anyway, let me know if you are interested in contributing money to a
> prompt set (either UK or NZ), that would be given back to the
> FreeSWITCH project.
>
> Thanks
>
> Campbell
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jaybinks at gmail.com  Wed Mar 16 09:14:51 2011
From: jaybinks at gmail.com (jay binks)
Date: Wed, 16 Mar 2011 16:14:51 +1000
Subject: [Freeswitch-users] English NZ or English UK Prompt Set Recording
In-Reply-To: <AANLkTikhqe1mC91VrJQj6tNSn5fPKh4zV6ANK4bBGEuS@mail.gmail.com>
References: <AANLkTikhqe1mC91VrJQj6tNSn5fPKh4zV6ANK4bBGEuS@mail.gmail.com>
Message-ID: <AANLkTinU0xwJdj25trEOjaL1pQ+0kVrP7nru7TOv0NQR@mail.gmail.com>

how about working together to get an Australian Set ? :)

id be slightly interested in UK, but it would have to be a thin UK accent
...
if you know what I mean :)

Jay



On Wed, Mar 16, 2011 at 12:11 PM, Campbell Steven <casteven at gmail.com>wrote:

> Hi,
>
> I'm trying to get a group of people together to contribute money
> towards recording a standard set of New Zealand English prompts for
> FreeSWITCH. I'd like to get a New Zealand set done, but if there isn't
> enough interest I'd settle for a UK set and I'd have to take guidance
> on companies who could do that since I'm based in NZ.
>
> Anyway, let me know if you are interested in contributing money to a
> prompt set (either UK or NZ), that would be given back to the
> FreeSWITCH project.
>
> Thanks
>
> Campbell
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely

Jay
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From acrow at integrafin.co.uk  Wed Mar 16 10:12:37 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Wed, 16 Mar 2011 07:12:37 +0000
Subject: [Freeswitch-users] English NZ or English UK Prompt Set Recording
In-Reply-To: <AANLkTinU0xwJdj25trEOjaL1pQ+0kVrP7nru7TOv0NQR@mail.gmail.com>
References: <AANLkTikhqe1mC91VrJQj6tNSn5fPKh4zV6ANK4bBGEuS@mail.gmail.com>
	<AANLkTinU0xwJdj25trEOjaL1pQ+0kVrP7nru7TOv0NQR@mail.gmail.com>
Message-ID: <4D8062E5.7020602@integrafin.co.uk>

On 16/03/11 06:14, jay binks wrote:
> how about working together to get an Australian Set ? :)
>
> id be slightly interested in UK, but it would have to be a thin UK 
> accent ...
> if you know what I mean :)
>
> Jay
>
>
Jay, Campbell,

I am looking to get commercial prompts ("Rachel") to match what I have 
on a Trixbox machine, so they would not, I'm afraid, be available for 
contribution back to the project.

Rachel's quite plummy too, so I doubt they'd suit NZ or Oz.

Cheers

Alex


-- 
This message is intended only for the addressee and may contain
confidential information.  Unless you are that person, you may not
disclose its contents or use it in any way and are requested to delete
the message along with any attachments and notify us immediately.

"Transact" is operated by Integrated Financial Arrangements plc
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From casteven at gmail.com  Wed Mar 16 10:30:58 2011
From: casteven at gmail.com (Campbell Steven)
Date: Wed, 16 Mar 2011 20:30:58 +1300
Subject: [Freeswitch-users] English NZ or English UK Prompt Set Recording
In-Reply-To: <AANLkTinU0xwJdj25trEOjaL1pQ+0kVrP7nru7TOv0NQR@mail.gmail.com>
References: <AANLkTikhqe1mC91VrJQj6tNSn5fPKh4zV6ANK4bBGEuS@mail.gmail.com>
	<AANLkTinU0xwJdj25trEOjaL1pQ+0kVrP7nru7TOv0NQR@mail.gmail.com>
Message-ID: <1300260658.3027.26.camel@macmini>

I actually need an AU prompt set as well, I was going to approach the
company we get custom prompts from there who produce an Asterisk set of
prompts to see if I could come up with some kind of arrangement with
them. I'll continue to do that and let you know where I get to, so I
guess this is now also a call for those interested in an AU set as well!

Campbell

On Wed, 2011-03-16 at 16:14 +1000, jay binks wrote:

> how about working together to get an Australian Set ? :)
> 
> 
> 
> id be slightly interested in UK, but it would have to be a thin UK
> accent ...
> if you know what I mean :)
> 
> 
> Jay
> 
> 
> 
> 
> 
> On Wed, Mar 16, 2011 at 12:11 PM, Campbell Steven <casteven at gmail.com>
> wrote:
> 
>         Hi,
>         
>         I'm trying to get a group of people together to contribute
>         money
>         towards recording a standard set of New Zealand English
>         prompts for
>         FreeSWITCH. I'd like to get a New Zealand set done, but if
>         there isn't
>         enough interest I'd settle for a UK set and I'd have to take
>         guidance
>         on companies who could do that since I'm based in NZ.
>         
>         Anyway, let me know if you are interested in contributing
>         money to a
>         prompt set (either UK or NZ), that would be given back to the
>         FreeSWITCH project.
>         
>         Thanks
>         
>         Campbell
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Sincerely
> 
> Jay
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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From jaybinks at gmail.com  Wed Mar 16 10:37:27 2011
From: jaybinks at gmail.com (jay binks)
Date: Wed, 16 Mar 2011 17:37:27 +1000
Subject: [Freeswitch-users] English NZ or English UK Prompt Set Recording
In-Reply-To: <1300260658.3027.26.camel@macmini>
References: <AANLkTikhqe1mC91VrJQj6tNSn5fPKh4zV6ANK4bBGEuS@mail.gmail.com>
	<AANLkTinU0xwJdj25trEOjaL1pQ+0kVrP7nru7TOv0NQR@mail.gmail.com>
	<1300260658.3027.26.camel@macmini>
Message-ID: <AANLkTinf-uLJr9quA0ScdcF8s6JwSx=dG=ykm3gn-4uz@mail.gmail.com>

well we COULD just do Australian, and then get them to mis-pronounce some
things and call it NZ prompt set..

all they'd need to do is say 6 funny and it could be passed off as NZ
prompts..  :)

Jay

On Wed, Mar 16, 2011 at 5:30 PM, Campbell Steven <casteven at gmail.com> wrote:

>  I actually need an AU prompt set as well, I was going to approach the
> company we get custom prompts from there who produce an Asterisk set of
> prompts to see if I could come up with some kind of arrangement with them.
> I'll continue to do that and let you know where I get to, so I guess this is
> now also a call for those interested in an AU set as well!
>
> Campbell
>
>
> On Wed, 2011-03-16 at 16:14 +1000, jay binks wrote:
>
> how about working together to get an Australian Set ? :)
>
>
>
>  id be slightly interested in UK, but it would have to be a thin UK accent
> ...
>
>  if you know what I mean :)
>
>
>
>  Jay
>
>
>
>
>
>  On Wed, Mar 16, 2011 at 12:11 PM, Campbell Steven <casteven at gmail.com>
> wrote:
>
> Hi,
>
> I'm trying to get a group of people together to contribute money
> towards recording a standard set of New Zealand English prompts for
> FreeSWITCH. I'd like to get a New Zealand set done, but if there isn't
> enough interest I'd settle for a UK set and I'd have to take guidance
> on companies who could do that since I'm based in NZ.
>
> Anyway, let me know if you are interested in contributing money to a
> prompt set (either UK or NZ), that would be given back to the
> FreeSWITCH project.
>
> Thanks
>
> Campbell
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Sincerely
>
> Jay
>
>  _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely

Jay
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From christian.loeschenkohl at xpirio.com  Wed Mar 16 10:52:48 2011
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Wed, 16 Mar 2011 08:52:48 +0100
Subject: [Freeswitch-users] German voice prompts - missing exceptions
 in	mod_say_de.c
In-Reply-To: <AANLkTi=6Q6Q-=S-z9pKT8ofMyCQyB7LRZrrwUNB+KGf_@mail.gmail.com>
References: <AANLkTi=6Q6Q-=S-z9pKT8ofMyCQyB7LRZrrwUNB+KGf_@mail.gmail.com>
Message-ID: <4D806C50.7050005@xpirio.com>

hello

we have done the voice prompts.
what is online under http://freeswitch.xpirio.com/ is only a start as we wanted to
do a second recording session to complete the prompts.
also the xml files and maybe the mod_say module has to be adapted.

maybe we can join our forces here, i have not that much time right now but
i can organize the recording session give some input and so on.
we also hope that this prompts are not only useable for austria but also for germany.

br

On 2011-03-15 23:09, Christian Benke wrote:

> Hi!
>
> Are there any other german voice-prompts available? The ones provided
> by xpirio(http://freeswitch.xpirio.com/) have some flaws
> unfortunately.
>
> There's also a possible problem with the word "Eins"(one), as we need
> 3 different "flexions" of the word:
> - Eine - e.g. in Voicemail for "one new message"(Female form)
> - Einer - male form of "one" - i didn't see a use for it yet, but that
> doesn't mean this case does not exist
> - Eins - e.g. spelling of "1", this is the only form currently working
> with type "items" in speech management
>
> Additionally we have male, female and neuter versions of counted
> numbers, this applies to all numbers, not just "one".
> The english equivalent is "XXth", so we can cover one gender with the
> h-XX.wav files. I'm not sure if female and neuter-versions of the
> files are actually needed(Xpirio has recorded a neutrum-version of the
> h-XX.wav files, but for e.g. say type "DATE/TIME" we would need the
> male form)
>
> I'm not a c-developer, so i might be wrong - but skimming through
> mod_say_de.c i saw some exceptions, but not for these cases(Sorry in
> advance if i'm wrong). Also i would not know how to tell the
> "say"-application to use these special forms if they existed.
>
> Is it possible to cover this/manipulate mod_say with "speech phrase
> management"? Judging from the available documentation it seems not...
>
> Other complications with german grammar might appear - i've just
> started using the german sound-files today in the afternoon and tested
> with voicemail mostly - but i would dig deeper and happily assist any
> developer who wants to get his hands dirty - in real time via phone
> and IM and 24h if needed(CET here).
>
> Sorry again if i missed an existing solution, it's been a long day.
>
> Best regards
> Christian
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 5 77 11 - 1000
F  +43 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From kbdfck at gmail.com  Wed Mar 16 11:15:36 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Wed, 16 Mar 2011 11:15:36 +0300
Subject: [Freeswitch-users] Already hung channels in 'show channels'
 output never go away
In-Reply-To: <AANLkTimAJ2ULQZk55B63zBoVnKK+E9nrQeCR8tP5CuFQ@mail.gmail.com>
References: <AANLkTi=jiL-aSmkDusoH_9EYRzPZ1ePXkWAQHXn53Np5@mail.gmail.com>
	<AANLkTinUg76_ncbjCwf=W8T5BPSS3FdstGQkkptv-xme@mail.gmail.com>
	<AANLkTimzvf-ZaiaVKCprbHh6JcFx3dXnbaU54T8KL=O6@mail.gmail.com>
	<AANLkTinTApDm-YL5UjqPnc8vaE-e+xpdAXiwaD+DTZhX@mail.gmail.com>
	<AANLkTik_CE66S0FToNMycX+U_WV3m12T=8D4t88BdcKP@mail.gmail.com>
	<AANLkTikTQpMHe67cSHr3T9Xr7UFVTu=RWXnNthxaBqCV@mail.gmail.com>
	<AANLkTimAJ2ULQZk55B63zBoVnKK+E9nrQeCR8tP5CuFQ@mail.gmail.com>
Message-ID: <AANLkTinXhqZ8A=B8QNJwy7AfboaE2bTttHRjrTemNyEB@mail.gmail.com>

I missed I'm using mod_hash for mod_hash_limit, calling
execute("limit","hash $realm $key")
and
api("limit_usage","hash $realm $key")
from perl ESL. And there are messages on FS shutdown (from hash_limit
cleanup task?)
Maybe using mod_hash_limit affect channel destroy sequence in some way?
hash_dump shows nothing after channel hangup, but channel which was hungup
first stays in channel list forever :(

2011-03-16 11:12:07.454839 [ERR] switch_loadable_module.c:489 Giving up on
'sofia' waiting for existing references.

2011-03-16 11:12:07.944692 [CRIT] sofia.c:1646 Waiting for 2 session(s)
2011-03-16 11:12:08.944692 [CRIT] sofia.c:1646 Waiting for 2 session(s)
2011-03-16 11:12:09.944693 [CRIT] sofia.c:1646 Waiting for 2 session(s)
2011-03-16 11:12:10.944678 [CRIT] sofia.c:1646 Waiting for 2 session(s)



2011/3/12 Anthony Minessale <anthony.minessale at gmail.com>

> this sounds odd,
> are you on the latest GIT build?
>
> try doing a gcore on the box when its like that and run gdb on it and
> do thread apply all bt
>
>
> On Fri, Mar 11, 2011 at 3:38 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> > I use cdr_xml only. If the call fails before entering ESL script, seems
> > there are no stuck records, but they appear after some actions in ESL
> even
> > if there was no bridge attempt.
> >
> > 2011/3/11 Anthony Minessale <anthony.minessale at gmail.com>
> >>
> >> that looks like they are all stuck in the CDR module.
> >> What module are you using?
> >>
> >>
> >>
> >> On Fri, Mar 11, 2011 at 5:58 AM, Dmitry Sytchev <kbdfck at gmail.com>
> wrote:
> >> > I think this is not the case, but anyway, what to do with these hung
> >> > channels?
> >> > Maybe I'm doing something wrong while bridging or processing events?
> >> > Maybe
> >> > unprocessed events can affect channel destroy procedure?
> >> > 2011/3/11 Avi Marcus <avi at avimarcus.net>
> >> >>
> >> >> Regarding ram usage, I'd imagine this is the case:
> >> >> http://www.linuxatemyram.com/
> >> >> -Avi
> >> >>
> >> >> On Fri, Mar 11, 2011 at 12:33 PM, Dmitry Sytchev <kbdfck at gmail.com>
> >> >> wrote:
> >> >>>
> >> >>> BTW, uuid_dump 7ae28028-4bc2-11e0-a92f-452f3d3d66ea returns with
> -ERR
> >> >>> No
> >> >>> such channel.
> >> >>>
> >> >>> 2011/3/11 Dmitry Sytchev <kbdfck at gmail.com>
> >> >>>>
> >> >>>> Hi All
> >> >>>> I'm using Perl ESL outbound script to bridge incoming call to sip
> >> >>>> endpoint. I'm doing execute("bridge","sofia/user/somebody"), then
> >> >>>> processing
> >> >>>> events like CHANNEL_HANGUP and EXECUTE_COMPLETE to determine bridge
> >> >>>> result.
> >> >>>> Everything works fine, but original incoming call channel is never
> >> >>>> removed
> >> >>>> from list:
> >> >>>> After few calls I see original incoming channels in 'show channels'
> >> >>>> output:
> >> >>>> 7ae28028-4bc2-11e0-a92f-452f3d3d66ea,inbound,2011-03-11
> >> >>>>
> >> >>>> 12:32:27,1299835947,sofia/external/xxxx at 85.x.x.199
> ,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,
> xxx.xxx.ru,,,HANGUP,,,,
> >> >>>> 883c9eb6-4bc2-11e0-a935-452f3d3d66ea,inbound,2011-03-11
> >> >>>>
> >> >>>> 12:32:50,1299835970,sofia/external/xxxx at 85.x.x.199
> ,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,
> xxx.xxx.ru,,,HANGUP,,,,
> >> >>>> 9f6a0f92-4bc2-11e0-a93b-452f3d3d66ea,inbound,2011-03-11
> >> >>>>
> >> >>>> 12:33:29,1299836009,sofia/external/xxxx at 85.x.x.199
> ,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,
> xxx.xxx.ru,,,HANGUP,,,
> >> >>>> Also, when I try to stop freeswitch i see these messages on
> console:
> >> >>>> 2011-03-11 15:28:22.517063 [CRIT] sofia.c:1632 Waiting for 13
> >> >>>> session(s)
> >> >>>> 2011-03-11 15:28:22.701034 [CRIT] sofia.c:1632 Waiting for 3
> >> >>>> session(s)
> >> >>>> 2011-03-11 15:28:23.517064 [CRIT] sofia.c:1632 Waiting for 13
> >> >>>> session(s)
> >> >>>> 2011-03-11 15:28:23.701033 [CRIT] sofia.c:1632 Waiting for 3
> >> >>>> session(s)
> >> >>>> 2011-03-11 15:28:24.517062 [CRIT] sofia.c:1632 Waiting for 13
> >> >>>> session(s)
> >> >>>> 2011-03-11 15:28:24.705036 [CRIT] sofia.c:1632 Waiting for 3
> >> >>>> session(s)
> >> >>>> 2011-03-11 15:28:25.517064 [CRIT] sofia.c:1632 Waiting for 13
> >> >>>> session(s)
> >> >>>> 2011-03-11 15:28:25.709031 [CRIT] sofia.c:1632 Waiting for 3
> >> >>>> session(s)
> >> >>>> 2011-03-11 15:28:26.517065 [CRIT] sofia.c:1632 Waiting for 13
> >> >>>> session(s)
> >> >>>>
> >> >>>> Why these channels are not removed from list? I also noticed that
> >> >>>> memory
> >> >>>> consumption by freeswitch process constantly grows call by call.
> >> >>>>
> >> >>>> --
> >> >>>> Best regards,
> >> >>>>
> >> >>>> Dmitry Sytchev,
> >> >>>> IT Engineer
> >> >>>
> >> >>>
> >> >>>
> >> >>> --
> >> >>> Best regards,
> >> >>>
> >> >>> Dmitry Sytchev,
> >> >>> IT Engineer
> >> >>>
> >> >>> _______________________________________________
> >> >>> FreeSWITCH-users mailing list
> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>
> >> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> http://www.freeswitch.org
> >> >>>
> >> >>
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >
> >> >
> >> >
> >> > --
> >> > Best regards,
> >> >
> >> > Dmitry Sytchev,
> >> > IT Engineer
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > 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-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Best regards,
> >
> > Dmitry Sytchev,
> > IT Engineer
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From lluis.riera at quarea.com  Wed Mar 16 11:56:51 2011
From: lluis.riera at quarea.com (=?iso-8859-1?Q?Llu=EDs_Riera?=)
Date: Wed, 16 Mar 2011 09:56:51 +0100
Subject: [Freeswitch-users] Multiconference with g729 users only
	withouttranscoding
References: <738A792011F8794D8CFAE25C00089679DC1AC9@quarea-correu.quarea.com>
	<AANLkTi=xUfDMA+9aJCHgQoNGdyp_cziJe-m7bReUg-NB@mail.gmail.com>
Message-ID: <738A792011F8794D8CFAE25C00089679DC1AE8@quarea-correu.quarea.com>

Thank You for all your answers!!! I got a clear picture now J

 

De: Michael Collins [mailto:msc at freeswitch.org] 
Enviado el: martes, 15 de marzo de 2011 21:35
Para: FreeSWITCH Users Help
Asunto: Re: [Freeswitch-users] Multiconference with g729 users only withouttranscoding

 

 

On Tue, Mar 15, 2011 at 5:21 AM, Llu?s Riera <lluis.riera at quarea.com> wrote:

Hi,

 

I would like to know if is there a configuration in freeswitch to allow multiconferencing for g729 users but without transcoding? It seems that all users are converted to L16 and it's not possible to have passthrough mode in conference rooms.

 

 

Yeah, this is a bad idea for all the reasons that the others have stated. It is SOOO much easier just to shell out $10 per channel and get the g729 licenses.

-MC

 

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From casteven at gmail.com  Wed Mar 16 13:31:29 2011
From: casteven at gmail.com (Campbell Steven)
Date: Wed, 16 Mar 2011 23:31:29 +1300
Subject: [Freeswitch-users] English NZ or English UK Prompt Set Recording
In-Reply-To: <AANLkTinf-uLJr9quA0ScdcF8s6JwSx=dG=ykm3gn-4uz@mail.gmail.com>
References: <AANLkTikhqe1mC91VrJQj6tNSn5fPKh4zV6ANK4bBGEuS@mail.gmail.com>
	<AANLkTinU0xwJdj25trEOjaL1pQ+0kVrP7nru7TOv0NQR@mail.gmail.com>
	<1300260658.3027.26.camel@macmini>
	<AANLkTinf-uLJr9quA0ScdcF8s6JwSx=dG=ykm3gn-4uz@mail.gmail.com>
Message-ID: <1300271489.3027.195.camel@macmini>

I could, and it would be an excellent strategy if I wanted to lose all
my NZ customers :-)

Campbell

On Wed, 2011-03-16 at 17:37 +1000, jay binks wrote:

> well we COULD just do Australian, and then get them to mis-pronounce
> some things and call it NZ prompt set..
> 
> 
> 
> all they'd need to do is say 6 funny and it could be passed off as NZ
> prompts..  :)
> 
> 
> Jay
> 
> 
> On Wed, Mar 16, 2011 at 5:30 PM, Campbell Steven <casteven at gmail.com>
> wrote:
> 
>         I actually need an AU prompt set as well, I was going to
>         approach the company we get custom prompts from there who
>         produce an Asterisk set of prompts to see if I could come up
>         with some kind of arrangement with them. I'll continue to do
>         that and let you know where I get to, so I guess this is now
>         also a call for those interested in an AU set as well!
>         
>         Campbell
>         
>         
>         
>         
>         On Wed, 2011-03-16 at 16:14 +1000, jay binks wrote:
>         
>         > how about working together to get an Australian Set ? :)
>         > 
>         > 
>         > id be slightly interested in UK, but it would have to be a
>         > thin UK accent ...
>         > if you know what I mean :)
>         > 
>         > 
>         > Jay
>         > 
>         > 
>         > 
>         > 
>         > On Wed, Mar 16, 2011 at 12:11 PM, Campbell Steven
>         > <casteven at gmail.com> wrote:
>         > 
>         >         Hi,
>         >         
>         >         I'm trying to get a group of people together to
>         >         contribute money
>         >         towards recording a standard set of New Zealand
>         >         English prompts for
>         >         FreeSWITCH. I'd like to get a New Zealand set done,
>         >         but if there isn't
>         >         enough interest I'd settle for a UK set and I'd have
>         >         to take guidance
>         >         on companies who could do that since I'm based in
>         >         NZ.
>         >         
>         >         Anyway, let me know if you are interested in
>         >         contributing money to a
>         >         prompt set (either UK or NZ), that would be given
>         >         back to the
>         >         FreeSWITCH project.
>         >         
>         >         Thanks
>         >         
>         >         Campbell
>         >         
>         >         _______________________________________________
>         >         FreeSWITCH-users mailing list
>         >         FreeSWITCH-users at lists.freeswitch.org
>         >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         >         http://www.freeswitch.org 
>         > 
>         > 
>         > 
>         > 
>         > -- 
>         > Sincerely
>         > 
>         > Jay
>         > 
>         > 
>         > _______________________________________________
>         > FreeSWITCH-users mailing list
>         > FreeSWITCH-users at lists.freeswitch.org
>         > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > http://www.freeswitch.org
>         
>         
>         
>         
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> 
> 
> 
> -- 
> Sincerely
> 
> Jay
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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From benkokakao at gmail.com  Wed Mar 16 13:54:12 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Wed, 16 Mar 2011 11:54:12 +0100
Subject: [Freeswitch-users] German voice prompts - missing exceptions in
	mod_say_de.c
In-Reply-To: <4D806C50.7050005@xpirio.com>
References: <AANLkTi=6Q6Q-=S-z9pKT8ofMyCQyB7LRZrrwUNB+KGf_@mail.gmail.com>
	<4D806C50.7050005@xpirio.com>
Message-ID: <AANLkTinCuxcqoa-aaq-FyXjy8rqEk28JC0ObRt6R5xVS@mail.gmail.com>

2011/3/16 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>:
> also the xml files and maybe the mod_say module has to be adapted.

Hi Christian!

Thank you for your answer and for publishing the prompts you've
recorded! I'll try to get a hold of a dev today to take a closer look
at the say-module and check what needs to be changed.

What is the status of "german prompts 2 -
de2.xls(http://freeswitch.xpirio.com/de2.xls)" - is this list complete
and verifyed? It looks like some conference texts(custom/*) are
missing though.

If you are interested in recording the missing voice-files in the next
few weeks i'll gladly provide you with missing information/texts!

Regards
Christian


From Nabble at slickdeals.endjunk.com  Wed Mar 16 14:00:27 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 16 Mar 2011 04:00:27 -0700 (PDT)
Subject: [Freeswitch-users] Multiconference with g729 users only without
	transcoding
In-Reply-To: <738A792011F8794D8CFAE25C00089679DC1AC9@quarea-correu.quarea.com>
References: <738A792011F8794D8CFAE25C00089679DC1AC9@quarea-correu.quarea.com>
Message-ID: <1300273227906-6176043.post@n2.nabble.com>


Llu?s Riera wrote:
> I would like to know if is there a configuration in freeswitch to allow
> multiconferencing for g729 users but without transcoding? It seems that
> all users are converted to L16 and it's not possible to have passthrough
> mode in conference rooms.
I am just curious here. Does your FS do the transcode in both directions,
i.e. G729 (caller) -> L16 (FS) -> G729 (Caller)?

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Multiconference-with-g729-users-only-without-transcoding-tp6172647p6176043.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From 29prime at gmail.com  Wed Mar 16 14:05:54 2011
From: 29prime at gmail.com (Prime)
Date: Wed, 16 Mar 2011 13:05:54 +0200
Subject: [Freeswitch-users] probem with configuring DIDs
In-Reply-To: <AANLkTimKLWjzUzNEOFGRngebxpLddyXj6KLu1YfR97zQ@mail.gmail.com>
References: <AANLkTi=aK=K-25NL87WYT4m-GQJK_DT=rW6PjabKzAT0@mail.gmail.com>
	<AANLkTimKLWjzUzNEOFGRngebxpLddyXj6KLu1YfR97zQ@mail.gmail.com>
Message-ID: <AANLkTiku6hoNzM5T30kYZ1PnT+d1kgumoBbow7MC9CEn@mail.gmail.com>

On 14 March 2011 23:36, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Mon, Mar 14, 2011 at 1:25 AM, Prime <29prime at gmail.com> wrote:
>>
>> Hey!
>>
>> This most likely is a newb mistake but I'm stuck at the moment.
>>
>> Recently I had to add another provider to FS to handle incoming calls.
>> And now I have a problem with calling my new provider's DID number on
>> my FS.
>> I looked at FS logs.
>> Usually FS log says something like this " [INFO]
>> mod_dialplan_xml.c:418 Processing caller_number->My_DID in context
>> public" and then routes DID number to local extension,
>> but now I see something like this " [INFO] mod_dialplan_xml.c:418
>> Processing caller_number->caller_number in context public ".
>> Why new numbers does not match the regex?
>>
> Best way to know is to look at the debug level output from fs_cli. If you
> need help deciphering it then put the output of a failed incoming call into
> a pastebin and reply to this thread with the link. Use
> pastebin.freeswitch.org for best results.
> -MC
>

Hi!

This case is resolved.
The problem was with provider's SIP invite packet with missing called number.

Sorry for the noise.


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


From pkelly at gmail.com  Wed Mar 16 16:26:24 2011
From: pkelly at gmail.com (Pete Kelly)
Date: Wed, 16 Mar 2011 13:26:24 +0000
Subject: [Freeswitch-users] Memory leak
In-Reply-To: <AANLkTinRceRe-VmN0a8=UXpqUvSs=ek2iLdviP+KxW81@mail.gmail.com>
References: <AANLkTim3UsX4z_URe1Ya40uP4AnVG3s3NiYkew=ZY8+O@mail.gmail.com>
	<AANLkTincGerDeJtkyqzYXFmtTzjHe-h9RMZJsM5HosCS@mail.gmail.com>
	<AANLkTimLNa73o5Y_h6KUpTptNyX6Q-6bpBOQBjLtsnS6@mail.gmail.com>
	<AANLkTinxMmO8UCR6wjROjB=DoD_yUoe2EuBNV=rNb3AC@mail.gmail.com>
	<AANLkTin9XpLcop+QnPFE70GSnEEHx5_yxWFhDrEJDBvq@mail.gmail.com>
	<AANLkTinRceRe-VmN0a8=UXpqUvSs=ek2iLdviP+KxW81@mail.gmail.com>
Message-ID: <AANLkTik9jns-JLjG7p0XCV3Ao-+GDAsjy5v13-tM6aDz@mail.gmail.com>

Yep that seems to have worked.

To recap I was using the luasql.mysql library, and have moved to using
freeswitch.Dbh using the mysql "_r" thread safe odbc driver with unixodbc.

Thanks everyone for your help.

Pete

On 15 March 2011 14:30, Pete Kelly <pkelly at gmail.com> wrote:

> I have gone with Freeswitch's inbuilt odbc implementation, and I have the
> _r libs in use for the drivers.
>
> Fingers crossed it helps matters, should know in a few days.
>
>
> On 14 March 2011 16:25, Anthony Minessale <anthony.minessale at gmail.com>wrote:
>
>> also make sure to use the _r version of any libs you use such as
>> myodbc or mysqlclient
>>
>> look in /etc/odbcinst.ini and chances are its not the _r one
>>
>> # Driver from the MyODBC package
>> # Setup from the unixODBC package
>> [MySQL]
>> Description     = ODBC for MySQL
>> Driver          = /usr/lib64/libmyodbc3.so
>> Setup           = /usr/lib64/libodbcmyS.so
>> FileUsage       = 1
>>
>>
>> /usr/lib64/libmyodbc3.so needs to be /usr/lib64/libmyodbc3_r.so
>>
>>
>>
>>
>> On Mon, Mar 14, 2011 at 10:27 AM, Steven Ayre <steveayre at gmail.com>
>> wrote:
>> > Try using the freeswitch.Dbh interface:
>> > http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh
>> > It gives you a ODBC interface through freeswitch's odbc support, plus
>> you
>> > get the benefit of the connection caching.
>> > -Steve
>> >
>> >
>> > On 14 March 2011 14:28, Pete Kelly <pkelly at gmail.com> wrote:
>> >>
>> >>
>> >> On 14 March 2011 13:16, Steven Ayre <steveayre at gmail.com> wrote:
>> >>>
>> >>> If you use mod_lcr, we were getting a memory leak in that a few weeks
>> ago
>> >>> which is fixed in the latest git.
>> >>
>> >> No we are not. I am going to try strengthening up the code which closes
>> >> the DB connections. If that fails then I will compile and try the ODBC
>> >> driver.
>> >>
>> >>>
>> >>> -Steve
>> >>>
>> >>> On 14 March 2011 09:30, Pete Kelly <pkelly at gmail.com> wrote:
>> >>>>
>> >>>> Hi
>> >>>> We are using Freeswitch (from git sources) in a production
>> environment
>> >>>> to handle an IVR system using a simple set of dialplans and lua
>> scripts.
>> >>>> OpenSIPS 1.6 and mysql are also installed and running as part of the
>> same
>> >>>> IVR application.
>> >>>> However over time (approx 1 week) the memory usage of Freeswitch
>> >>>> increases, eventually creeping into swap space forcing us to restart
>> >>>> Freeswitch. The IVR handles approx 40 concurrent calls constantly,
>> and we
>> >>>> have instances of the IVR running on a 32bit Etch and 64bit Lenny
>> boxes -
>> >>>> both exhibit the same issue.
>> >>>> We have also tried the Sangoma freeswitch branch in case that
>> contained
>> >>>> any fixes, however we see the same issues.
>> >>>> Can anybody advise if this is a known issue with Freeswitch? Is it a
>> >>>> memory leak or is it possible to limit the amount of memory
>> Freeswitch will
>> >>>> use, forcing it to garbage collect when it reaches the limit?
>> >>>> Any advice on where to look/debug next would be appreciated.
>> >>>> Thanks
>> >>>> Pete
>> >>>>
>> >>>>
>> >>>> _______________________________________________
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>>
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From moises.silva at gmail.com  Wed Mar 16 16:44:59 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 16 Mar 2011 09:44:59 -0400
Subject: [Freeswitch-users] Multiconference with g729 users only without
	transcoding
In-Reply-To: <1300273227906-6176043.post@n2.nabble.com>
References: <738A792011F8794D8CFAE25C00089679DC1AC9@quarea-correu.quarea.com>
	<1300273227906-6176043.post@n2.nabble.com>
Message-ID: <AANLkTikODAs3yxkBi2K=iMZvBfRjJLVTo3hbTdN4R8cw@mail.gmail.com>

On Wed, Mar 16, 2011 at 7:00 AM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
> Llu?s Riera wrote:
> > I would like to know if is there a configuration in freeswitch to allow
> > multiconferencing for g729 users but without transcoding? It seems that
> > all users are converted to L16 and it's not possible to have passthrough
> > mode in conference rooms.
> I am just curious here. Does your FS do the transcode in both directions,
> i.e. G729 (caller) -> L16 (FS) -> G729 (Caller)?
>

Without looking at the code, my guess is that it depends. If the caller is
muted only L16 -> G729 will be done since all incoming audio is discarded.

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
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From benkokakao at gmail.com  Wed Mar 16 17:10:58 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Wed, 16 Mar 2011 15:10:58 +0100
Subject: [Freeswitch-users] Alcatel 4018 - "481 Call Does Not Exist" on
	re-REGISTER
In-Reply-To: <AANLkTimGzBf_skFLrg99CFGjJs9WDxUpYi4rSgqquDi6@mail.gmail.com>
References: <AANLkTinjt-rh_ieSMwGcWsJbL4RwfRz4SCTFmUht+Z9L@mail.gmail.com>
	<AANLkTi=gi15SR8TRsFuw-YuM0SPPY8LypkQPtpchJJp6@mail.gmail.com>
	<AANLkTimGzBf_skFLrg99CFGjJs9WDxUpYi4rSgqquDi6@mail.gmail.com>
Message-ID: <AANLkTik+=S7e7q6Y7x4gNQvL1p+okZwWBFqC-9sCa=TE@mail.gmail.com>

I hope my initial information is not the cause why this thread was not
picked up!

I still wonder if there is some way to configure FS to catch this
unusual behaviour - otherwise i'll have to go back to the usual -
reading the rfc and confirming that this is in fact broken behaviour -
to get rid of the issue.

Cheers,
Christian


From mranga at gmail.com  Wed Mar 16 18:18:27 2011
From: mranga at gmail.com (M. Ranganathan)
Date: Wed, 16 Mar 2011 11:18:27 -0400
Subject: [Freeswitch-users] Alcatel 4018 - "481 Call Does Not Exist" on
	re-REGISTER
In-Reply-To: <AANLkTik+=S7e7q6Y7x4gNQvL1p+okZwWBFqC-9sCa=TE@mail.gmail.com>
References: <AANLkTinjt-rh_ieSMwGcWsJbL4RwfRz4SCTFmUht+Z9L@mail.gmail.com>
	<AANLkTi=gi15SR8TRsFuw-YuM0SPPY8LypkQPtpchJJp6@mail.gmail.com>
	<AANLkTimGzBf_skFLrg99CFGjJs9WDxUpYi4rSgqquDi6@mail.gmail.com>
	<AANLkTik+=S7e7q6Y7x4gNQvL1p+okZwWBFqC-9sCa=TE@mail.gmail.com>
Message-ID: <AANLkTikZWiMGT0CUZskxKLYza3coWwbvykLK7pgaQFSk@mail.gmail.com>

Change the cseq number (increment by 1) when you re-register. See if that
fixes it.

On Wed, Mar 16, 2011 at 10:10 AM, Christian Benke <benkokakao at gmail.com>wrote:

> I hope my initial information is not the cause why this thread was not
> picked up!
>
> I still wonder if there is some way to configure FS to catch this
> unusual behaviour - otherwise i'll have to go back to the usual -
> reading the rfc and confirming that this is in fact broken behaviour -
> to get rid of the issue.
>
> Cheers,
> Christian
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
M. Ranganathan
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From msc at freeswitch.org  Wed Mar 16 18:44:33 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 16 Mar 2011 08:44:33 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <AANLkTinEfU5OTM-jrozrL7ZM8U-t0S7fLs_nDh82SXHA@mail.gmail.com>

Hello all! The agenda page for today is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2011_03_16

It's pretty light, so bring your questions and suggestions.

-Michael
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From benkokakao at gmail.com  Wed Mar 16 19:11:34 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Wed, 16 Mar 2011 17:11:34 +0100
Subject: [Freeswitch-users] Alcatel 4018 - "481 Call Does Not Exist" on
	re-REGISTER
In-Reply-To: <AANLkTikZWiMGT0CUZskxKLYza3coWwbvykLK7pgaQFSk@mail.gmail.com>
References: <AANLkTinjt-rh_ieSMwGcWsJbL4RwfRz4SCTFmUht+Z9L@mail.gmail.com>
	<AANLkTi=gi15SR8TRsFuw-YuM0SPPY8LypkQPtpchJJp6@mail.gmail.com>
	<AANLkTimGzBf_skFLrg99CFGjJs9WDxUpYi4rSgqquDi6@mail.gmail.com>
	<AANLkTik+=S7e7q6Y7x4gNQvL1p+okZwWBFqC-9sCa=TE@mail.gmail.com>
	<AANLkTikZWiMGT0CUZskxKLYza3coWwbvykLK7pgaQFSk@mail.gmail.com>
Message-ID: <AANLkTikPk5tQDmYY+W-D57O4FhiQ=RKTjLUwoPBk9YM6@mail.gmail.com>

On 16 March 2011 16:18, M. Ranganathan <mranga at gmail.com> wrote:
> Change the cseq number (increment by 1) when you re-register. See if that
> fixes it.

The phone is in fact incrementing the CSeq already, so no,
unfortunately this does not help(And i actually would not have any
option to change it if it didn't).

So this is what i found in the RFCs:

http://www.ietf.org/rfc/rfc3261.txt 22.4.8. references RFC 2617(HTTP
Authentication: Basic and Digest Access Authentication)
http://www.ietf.org/rfc/rfc2617.txt 3.2.1 says:

     An implementation might choose not to accept a previously used
     nonce or a previously used digest, in order to protect against a
     replay attack. Or, an implementation might choose to use one-time
     nonces or digests for POST or PUT requests and a time-stamp for GET
     requests.

and

  4.3 Limited Use Nonce Values

   The Digest scheme uses a server-specified nonce to seed the
   generation of the request-digest value (as specified in section
   3.2.2.1 above).  As shown in the example nonce in section 3.2.1, the
   server is free to construct the nonce such that it may only be used
   from a particular client, for a particular resource, for a limited
   period of time or number of uses, or any other restrictions.  Doing
   so strengthens the protection provided against, for example, replay
   attacks (see 4.5).

This makes a lot of sense of course, so i have plenty arguments now to
ditch this phone(Alcatel 4018) as it is not behaving correctly(Unless
i find a newer firmware).

Regards,
Christian


From kbdfck at gmail.com  Wed Mar 16 19:28:47 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Wed, 16 Mar 2011 19:28:47 +0300
Subject: [Freeswitch-users] Unexpected out of order command/reply '+OK event
 listener enabled plain' in perl ESL outbound
Message-ID: <AANLkTikZ=pBk99wbg2+M_-vr6qK7=TC9tDZzzvE9a1+u@mail.gmail.com>

In my outbound async full ESL script after some moment there are
out-of-order message on every "execute" call, and this breaks my event
handling in some way.

Content-Type: command/reply.Reply-Text: +OK event listener enabled plain..
I want to execute some commands syncronously, but setEventLock seems to be
broken by this behavior.
As the workaround I had to write lock_for_execute_complete function, which
waits for execute completion of applications that need to be run sync, but
it effectively destroys async events by processing them while waiting for
CHANNEL_EXECUTE_COMPLETE.

here is a part of socket dump, note second "Content-Type:
command/reply.Reply-Text: +OK event listener enabled plain.." right after
execute. There is no event command in script besides the very beginning, so
why these messages appear?

T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
api uuid_setvar 06d55974-4fe8-11e0-a7c0-e3bb2141c22a
pbx_subscriber_customer_id 1
#
T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
..
##
T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
Content-Type: api/response.Content-Length: 4..
#
T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
+OK.
##
T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
  api sofia_contact s000002 at asdasd.com
#
T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
..
##
T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
Content-Type: api/response.Content-Length: 96..
#
T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
sofia/local/sip:s000002 at 172.19.36.54:5061;fs_nat=yes;fs_path=sip%3As000002%40172.19.36.54%3A5061

##
T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
Event-Name: SOCKET_DATA.Content-Type: api/response.Content-Length:
96..sofia/local/sip:s000002 at 172.19.36.54:5061;fs_nat=
yes;fs_path=sip%3As000002%40172.19.36.54%3A5061
#
T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
sendmsg.call-command: execute.execute-app-name: bridge.execute-app-arg:
{originate_timeout=10,hangup_after_bridge=false,
  ignore_early_media=false}user/s000002 at asdasd.com
#
*T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
Content-Type: command/reply.Reply-Text: +OK event listener enabled plain.. *
#
T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
Content-Type: command/reply.Reply-Text: +OK..

-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From peter.olsson at visionutveckling.se  Wed Mar 16 19:40:17 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 16 Mar 2011 17:40:17 +0100
Subject: [Freeswitch-users] Unexpected out of order command/reply '+OK
 event listener enabled plain' in perl ESL outbound
In-Reply-To: <AANLkTikZ=pBk99wbg2+M_-vr6qK7=TC9tDZzzvE9a1+u@mail.gmail.com>
References: <AANLkTikZ=pBk99wbg2+M_-vr6qK7=TC9tDZzzvE9a1+u@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B303CB87@cooper>

Are you on latest GIT HEAD?

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Dmitry Sytchev
Skickat: den 16 mars 2011 17:29
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] Unexpected out of order command/reply '+OK event listener enabled plain' in perl ESL outbound

In my outbound async full ESL script after some moment there are out-of-order message on every "execute" call, and this breaks my event handling in some way.

Content-Type: command/reply.Reply-Text: +OK event listener enabled plain..
I want to execute some commands syncronously, but setEventLock seems to be broken by this behavior.
As the workaround I had to write lock_for_execute_complete function, which waits for execute completion of applications that need to be run sync, but it effectively destroys async events by processing them while waiting for CHANNEL_EXECUTE_COMPLETE.

here is a part of socket dump, note second "Content-Type: command/reply.Reply-Text: +OK event listener enabled plain.." right after execute. There is no event command in script besides the very beginning, so why these messages appear?

T 127.0.0.1:8006<http://127.0.0.1:8006> -> 127.0.0.1:43282<http://127.0.0.1:43282> [AP]
api uuid_setvar 06d55974-4fe8-11e0-a7c0-e3bb2141c22a pbx_subscriber_customer_id 1
#
T 127.0.0.1:8006<http://127.0.0.1:8006> -> 127.0.0.1:43282<http://127.0.0.1:43282> [AP]
..
##
T 127.0.0.1:43282<http://127.0.0.1:43282> -> 127.0.0.1:8006<http://127.0.0.1:8006> [AP]
Content-Type: api/response.Content-Length: 4..
#
T 127.0.0.1:43282<http://127.0.0.1:43282> -> 127.0.0.1:8006<http://127.0.0.1:8006> [AP]
+OK.
##
T 127.0.0.1:8006<http://127.0.0.1:8006> -> 127.0.0.1:43282<http://127.0.0.1:43282> [AP]
  api sofia_contact s000002 at asdasd.com<mailto:s000002 at asdasd.com>
#
T 127.0.0.1:8006<http://127.0.0.1:8006> -> 127.0.0.1:43282<http://127.0.0.1:43282> [AP]
..
##
T 127.0.0.1:43282<http://127.0.0.1:43282> -> 127.0.0.1:8006<http://127.0.0.1:8006> [AP]
Content-Type: api/response.Content-Length: 96..
#
T 127.0.0.1:43282<http://127.0.0.1:43282> -> 127.0.0.1:8006<http://127.0.0.1:8006> [AP]
sofia/local/sip:s000002 at 172.19.36.54:5061;fs_nat=yes;fs_path=sip%3As000002%40172.19.36.54%3A5061
##
T 127.0.0.1:8006<http://127.0.0.1:8006> -> 127.0.0.1:43282<http://127.0.0.1:43282> [AP]
Event-Name: SOCKET_DATA.Content-Type: api/response.Content-Length: 96..sofia/local/sip:s000002 at 172.19.36.54:5061;fs_nat=
yes;fs_path=sip%3As000002%40172.19.36.54%3A5061
#
T 127.0.0.1:8006<http://127.0.0.1:8006> -> 127.0.0.1:43282<http://127.0.0.1:43282> [AP]
sendmsg.call-command: execute.execute-app-name: bridge.execute-app-arg: {originate_timeout=10,hangup_after_bridge=false,
  ignore_early_media=false}user/s000002 at asdasd.com<mailto:s000002 at asdasd.com>
#
T 127.0.0.1:43282<http://127.0.0.1:43282> -> 127.0.0.1:8006<http://127.0.0.1:8006> [AP]
Content-Type: command/reply.Reply-Text: +OK event listener enabled plain..
#
T 127.0.0.1:43282<http://127.0.0.1:43282> -> 127.0.0.1:8006<http://127.0.0.1:8006> [AP]
Content-Type: command/reply.Reply-Text: +OK..

--
Best regards,

Dmitry Sytchev,
IT Engineer
!DSPAM:4d80e61b32768938086861!
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From kbdfck at gmail.com  Wed Mar 16 19:55:54 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Wed, 16 Mar 2011 19:55:54 +0300
Subject: [Freeswitch-users] Unexpected out of order command/reply '+OK
 event listener enabled plain' in perl ESL outbound
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B303CB87@cooper>
References: <AANLkTikZ=pBk99wbg2+M_-vr6qK7=TC9tDZzzvE9a1+u@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B303CB87@cooper>
Message-ID: <AANLkTim=Sxe8iuQncgn+JJgBzP=u3oQZdvwD2wwZCesC@mail.gmail.com>

FreeSWITCH Version 1.0.head (git-2c009dd 2011-03-15 14-29-04 -0500)

2011/3/16 Peter Olsson <peter.olsson at visionutveckling.se>

> Are you on latest GIT HEAD?
>
>
>
> /Peter
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Dmitry Sytchev
> *Skickat:* den 16 mars 2011 17:29
> *Till:* FreeSWITCH Users Help
> *?mne:* [Freeswitch-users] Unexpected out of order command/reply '+OK
> event listener enabled plain' in perl ESL outbound
>
>
>
> In my outbound async full ESL script after some moment there are
> out-of-order message on every "execute" call, and this breaks my event
> handling in some way.
>
>
>
> Content-Type: command/reply.Reply-Text: +OK event listener enabled plain..
>
>
> I want to execute some commands syncronously, but setEventLock seems to be
> broken by this behavior.
>
> As the workaround I had to write lock_for_execute_complete function, which
> waits for execute completion of applications that need to be run sync, but
> it effectively destroys async events by processing them while waiting for
> CHANNEL_EXECUTE_COMPLETE.
>
>
>
> here is a part of socket dump, note second "Content-Type:
> command/reply.Reply-Text: +OK event listener enabled plain.." right after
> execute. There is no event command in script besides the very beginning, so
> why these messages appear?
>
>
> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
> api uuid_setvar 06d55974-4fe8-11e0-a7c0-e3bb2141c22a
> pbx_subscriber_customer_id 1
> #
> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
> ..
> ##
> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
> Content-Type: api/response.Content-Length: 4..
> #
> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
> +OK.
> ##
> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
>   api sofia_contact s000002 at asdasd.com
> #
> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
> ..
> ##
> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
> Content-Type: api/response.Content-Length: 96..
> #
> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
> sofia/local/sip:s000002 at 172.19.36.54:5061;fs_nat=yes;fs_path=sip%3As000002%40172.19.36.54%3A5061
>
> ##
> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
> Event-Name: SOCKET_DATA.Content-Type: api/response.Content-Length:
> 96..sofia/local/sip:s000002 at 172.19.36.54:5061;fs_nat=
> yes;fs_path=sip%3As000002%40172.19.36.54%3A5061
> #
> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
> sendmsg.call-command: execute.execute-app-name: bridge.execute-app-arg:
> {originate_timeout=10,hangup_after_bridge=false,
>   ignore_early_media=false}user/s000002 at asdasd.com
> #
> *T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]**
> Content-Type: command/reply.Reply-Text: +OK event listener enabled plain..
> *
> #
> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
> Content-Type: command/reply.Reply-Text: +OK..
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
> !DSPAM:4d80e61b32768938086861!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
-------------- next part --------------
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From anthony.minessale at gmail.com  Wed Mar 16 20:52:06 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 16 Mar 2011 12:52:06 -0500
Subject: [Freeswitch-users] Unexpected out of order command/reply '+OK
 event listener enabled plain' in perl ESL outbound
In-Reply-To: <AANLkTim=Sxe8iuQncgn+JJgBzP=u3oQZdvwD2wwZCesC@mail.gmail.com>
References: <AANLkTikZ=pBk99wbg2+M_-vr6qK7=TC9tDZzzvE9a1+u@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B303CB87@cooper>
	<AANLkTim=Sxe8iuQncgn+JJgBzP=u3oQZdvwD2wwZCesC@mail.gmail.com>
Message-ID: <AANLkTi=hwzMiU7Dxa8wD4fWmgvf=8Go4DGn+zizJAVub@mail.gmail.com>

Are you using your own client or libESL which takes care of this for you?


On Wed, Mar 16, 2011 at 11:55 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> FreeSWITCH Version 1.0.head (git-2c009dd 2011-03-15 14-29-04 -0500)
> 2011/3/16 Peter Olsson <peter.olsson at visionutveckling.se>
>>
>> Are you on latest GIT HEAD?
>>
>>
>>
>> /Peter
>>
>>
>>
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Dmitry Sytchev
>> Skickat: den 16 mars 2011 17:29
>> Till: FreeSWITCH Users Help
>> ?mne: [Freeswitch-users] Unexpected out of order command/reply '+OK event
>> listener enabled plain' in perl ESL outbound
>>
>>
>>
>> In my outbound async full ESL script after some moment there are
>> out-of-order message on every "execute" call, and this breaks my event
>> handling in some way.
>>
>>
>>
>> Content-Type: command/reply.Reply-Text: +OK event listener enabled plain..
>>
>>
>> I want to execute some commands syncronously, but setEventLock seems to be
>> broken by this behavior.
>>
>> As the workaround I had to write lock_for_execute_complete function, which
>> waits for execute completion of applications that need to be run sync, but
>> it effectively destroys async events by processing them while waiting for
>> CHANNEL_EXECUTE_COMPLETE.
>>
>>
>>
>> here is a part of socket dump,?note second "Content-Type:
>> command/reply.Reply-Text: +OK event listener enabled plain.." right after
>> execute.?There is no event command in script besides the very beginning, so
>> why these messages appear?
>>
>> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
>> api uuid_setvar 06d55974-4fe8-11e0-a7c0-e3bb2141c22a
>> pbx_subscriber_customer_id 1
>> #
>> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
>> ..
>> ##
>> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
>> Content-Type: api/response.Content-Length: 4..
>> #
>> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
>> +OK.
>> ##
>> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
>> ? api sofia_contact s000002 at asdasd.com
>> #
>> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
>> ..
>> ##
>> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
>> Content-Type: api/response.Content-Length: 96..
>> #
>> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
>>
>> sofia/local/sip:s000002 at 172.19.36.54:5061;fs_nat=yes;fs_path=sip%3As000002%40172.19.36.54%3A5061
>> ##
>> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
>> Event-Name: SOCKET_DATA.Content-Type: api/response.Content-Length:
>> 96..sofia/local/sip:s000002 at 172.19.36.54:5061;fs_nat=
>> yes;fs_path=sip%3As000002%40172.19.36.54%3A5061
>> #
>> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
>> sendmsg.call-command: execute.execute-app-name: bridge.execute-app-arg:
>> {originate_timeout=10,hangup_after_bridge=false,
>> ? ignore_early_media=false}user/s000002 at asdasd.com
>> #
>> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
>> Content-Type: command/reply.Reply-Text: +OK event listener enabled plain..
>> #
>> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
>> Content-Type: command/reply.Reply-Text: +OK..
>>
>> --
>> Best regards,
>>
>> Dmitry Sytchev,
>> IT Engineer
>> !DSPAM:4d80e61b32768938086861!
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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  Wed Mar 16 20:55:15 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 16 Mar 2011 12:55:15 -0500
Subject: [Freeswitch-users] Already hung channels in 'show channels'
 output never go away
In-Reply-To: <AANLkTinXhqZ8A=B8QNJwy7AfboaE2bTttHRjrTemNyEB@mail.gmail.com>
References: <AANLkTi=jiL-aSmkDusoH_9EYRzPZ1ePXkWAQHXn53Np5@mail.gmail.com>
	<AANLkTinUg76_ncbjCwf=W8T5BPSS3FdstGQkkptv-xme@mail.gmail.com>
	<AANLkTimzvf-ZaiaVKCprbHh6JcFx3dXnbaU54T8KL=O6@mail.gmail.com>
	<AANLkTinTApDm-YL5UjqPnc8vaE-e+xpdAXiwaD+DTZhX@mail.gmail.com>
	<AANLkTik_CE66S0FToNMycX+U_WV3m12T=8D4t88BdcKP@mail.gmail.com>
	<AANLkTikTQpMHe67cSHr3T9Xr7UFVTu=RWXnNthxaBqCV@mail.gmail.com>
	<AANLkTimAJ2ULQZk55B63zBoVnKK+E9nrQeCR8tP5CuFQ@mail.gmail.com>
	<AANLkTinXhqZ8A=B8QNJwy7AfboaE2bTttHRjrTemNyEB@mail.gmail.com>
Message-ID: <AANLkTi=Gdc8E8-xHeCNdpH5x-FHESEkjWgWN6H_s8_0m@mail.gmail.com>

yes you probably found a use case where the rwlock is not released
Can you open a JIRA and upload a gcore of the box when it's in the stuck state.


On Wed, Mar 16, 2011 at 3:15 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> I missed I'm using mod_hash for mod_hash_limit, calling
> execute("limit","hash $realm $key")
> and
> api("limit_usage","hash $realm $key")
> from perl ESL. And there are messages on FS shutdown (from hash_limit
> cleanup task?)
> Maybe using mod_hash_limit affect channel destroy sequence in some way?
> hash_dump shows nothing after channel hangup, but channel which was hungup
> first stays in channel list forever :(
> 2011-03-16 11:12:07.454839 [ERR] switch_loadable_module.c:489 Giving up on
> 'sofia' waiting for existing references.
>
> 2011-03-16 11:12:07.944692 [CRIT] sofia.c:1646 Waiting for 2 session(s)
> 2011-03-16 11:12:08.944692 [CRIT] sofia.c:1646 Waiting for 2 session(s)
> 2011-03-16 11:12:09.944693 [CRIT] sofia.c:1646 Waiting for 2 session(s)
> 2011-03-16 11:12:10.944678 [CRIT] sofia.c:1646 Waiting for 2 session(s)
>
>
>
> 2011/3/12 Anthony Minessale <anthony.minessale at gmail.com>
>>
>> this sounds odd,
>> are you on the latest GIT build?
>>
>> try doing a gcore on the box when its like that and run gdb on it and
>> do thread apply all bt
>>
>>
>> On Fri, Mar 11, 2011 at 3:38 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>> > I use cdr_xml only. If the call fails before entering ESL script, seems
>> > there are no stuck records, but they appear after some actions in ESL
>> > even
>> > if there was no bridge attempt.
>> >
>> > 2011/3/11 Anthony Minessale <anthony.minessale at gmail.com>
>> >>
>> >> that looks like they are all stuck in the CDR module.
>> >> What module are you using?
>> >>
>> >>
>> >>
>> >> On Fri, Mar 11, 2011 at 5:58 AM, Dmitry Sytchev <kbdfck at gmail.com>
>> >> wrote:
>> >> > I think this is not the case, but anyway, what to do with these hung
>> >> > channels?
>> >> > Maybe I'm doing something wrong while bridging or processing events?
>> >> > Maybe
>> >> > unprocessed events can affect channel destroy procedure?
>> >> > 2011/3/11 Avi Marcus <avi at avimarcus.net>
>> >> >>
>> >> >> Regarding ram usage, I'd imagine this is the case:
>> >> >> http://www.linuxatemyram.com/
>> >> >> -Avi
>> >> >>
>> >> >> On Fri, Mar 11, 2011 at 12:33 PM, Dmitry Sytchev <kbdfck at gmail.com>
>> >> >> wrote:
>> >> >>>
>> >> >>> BTW, uuid_dump?7ae28028-4bc2-11e0-a92f-452f3d3d66ea returns with
>> >> >>> -ERR
>> >> >>> No
>> >> >>> such channel.
>> >> >>>
>> >> >>> 2011/3/11 Dmitry Sytchev <kbdfck at gmail.com>
>> >> >>>>
>> >> >>>> Hi All
>> >> >>>> I'm using Perl ESL outbound script to bridge incoming call to sip
>> >> >>>> endpoint. I'm doing execute("bridge","sofia/user/somebody"), then
>> >> >>>> processing
>> >> >>>> events like CHANNEL_HANGUP and EXECUTE_COMPLETE to determine
>> >> >>>> bridge
>> >> >>>> result.
>> >> >>>> Everything works fine, but original incoming call channel is never
>> >> >>>> removed
>> >> >>>> from list:
>> >> >>>> After few calls I see original incoming channels in 'show
>> >> >>>> channels'
>> >> >>>> output:
>> >> >>>> 7ae28028-4bc2-11e0-a92f-452f3d3d66ea,inbound,2011-03-11
>> >> >>>>
>> >> >>>>
>> >> >>>> 12:32:27,1299835947,sofia/external/xxxx at 85.x.x.199,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,xxx.xxx.ru,,,HANGUP,,,,
>> >> >>>> 883c9eb6-4bc2-11e0-a935-452f3d3d66ea,inbound,2011-03-11
>> >> >>>>
>> >> >>>>
>> >> >>>> 12:32:50,1299835970,sofia/external/xxxx at 85.x.x.199,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,xxx.xxx.ru,,,HANGUP,,,,
>> >> >>>> 9f6a0f92-4bc2-11e0-a93b-452f3d3d66ea,inbound,2011-03-11
>> >> >>>>
>> >> >>>>
>> >> >>>> 12:33:29,1299836009,sofia/external/xxxx at 85.x.x.199,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,xxx.xxx.ru,,,HANGUP,,,
>> >> >>>> Also, when I try to stop freeswitch i see these messages on
>> >> >>>> console:
>> >> >>>> 2011-03-11 15:28:22.517063 [CRIT] sofia.c:1632 Waiting for 13
>> >> >>>> session(s)
>> >> >>>> 2011-03-11 15:28:22.701034 [CRIT] sofia.c:1632 Waiting for 3
>> >> >>>> session(s)
>> >> >>>> 2011-03-11 15:28:23.517064 [CRIT] sofia.c:1632 Waiting for 13
>> >> >>>> session(s)
>> >> >>>> 2011-03-11 15:28:23.701033 [CRIT] sofia.c:1632 Waiting for 3
>> >> >>>> session(s)
>> >> >>>> 2011-03-11 15:28:24.517062 [CRIT] sofia.c:1632 Waiting for 13
>> >> >>>> session(s)
>> >> >>>> 2011-03-11 15:28:24.705036 [CRIT] sofia.c:1632 Waiting for 3
>> >> >>>> session(s)
>> >> >>>> 2011-03-11 15:28:25.517064 [CRIT] sofia.c:1632 Waiting for 13
>> >> >>>> session(s)
>> >> >>>> 2011-03-11 15:28:25.709031 [CRIT] sofia.c:1632 Waiting for 3
>> >> >>>> session(s)
>> >> >>>> 2011-03-11 15:28:26.517065 [CRIT] sofia.c:1632 Waiting for 13
>> >> >>>> session(s)
>> >> >>>>
>> >> >>>> Why these channels are not removed from list? I also noticed that
>> >> >>>> memory
>> >> >>>> consumption by freeswitch process constantly grows call by call.
>> >> >>>>
>> >> >>>> --
>> >> >>>> Best regards,
>> >> >>>>
>> >> >>>> Dmitry Sytchev,
>> >> >>>> IT Engineer
>> >> >>>
>> >> >>>
>> >> >>>
>> >> >>> --
>> >> >>> Best regards,
>> >> >>>
>> >> >>> Dmitry Sytchev,
>> >> >>> IT Engineer
>> >> >>>
>> >> >>> _______________________________________________
>> >> >>> FreeSWITCH-users mailing list
>> >> >>> FreeSWITCH-users at lists.freeswitch.org
>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>
>> >> >>>
>> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> http://www.freeswitch.org
>> >> >>>
>> >> >>
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >>
>> >> >
>> >> >
>> >> >
>> >> > --
>> >> > Best regards,
>> >> >
>> >> > Dmitry Sytchev,
>> >> > IT Engineer
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > 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-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Best regards,
>> >
>> > Dmitry Sytchev,
>> > IT Engineer
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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  Wed Mar 16 20:56:46 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 16 Mar 2011 12:56:46 -0500
Subject: [Freeswitch-users] Already hung channels in 'show channels'
 output never go away
In-Reply-To: <AANLkTi=Gdc8E8-xHeCNdpH5x-FHESEkjWgWN6H_s8_0m@mail.gmail.com>
References: <AANLkTi=jiL-aSmkDusoH_9EYRzPZ1ePXkWAQHXn53Np5@mail.gmail.com>
	<AANLkTinUg76_ncbjCwf=W8T5BPSS3FdstGQkkptv-xme@mail.gmail.com>
	<AANLkTimzvf-ZaiaVKCprbHh6JcFx3dXnbaU54T8KL=O6@mail.gmail.com>
	<AANLkTinTApDm-YL5UjqPnc8vaE-e+xpdAXiwaD+DTZhX@mail.gmail.com>
	<AANLkTik_CE66S0FToNMycX+U_WV3m12T=8D4t88BdcKP@mail.gmail.com>
	<AANLkTikTQpMHe67cSHr3T9Xr7UFVTu=RWXnNthxaBqCV@mail.gmail.com>
	<AANLkTimAJ2ULQZk55B63zBoVnKK+E9nrQeCR8tP5CuFQ@mail.gmail.com>
	<AANLkTinXhqZ8A=B8QNJwy7AfboaE2bTttHRjrTemNyEB@mail.gmail.com>
	<AANLkTi=Gdc8E8-xHeCNdpH5x-FHESEkjWgWN6H_s8_0m@mail.gmail.com>
Message-ID: <AANLkTi=NE9Swjk0KYM_KPCUnUQYuwsqAibXOU10SD6w5@mail.gmail.com>

can you also get some console traces on debug level of the traffic
happening on the box.


On Wed, Mar 16, 2011 at 12:55 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> yes you probably found a use case where the rwlock is not released
> Can you open a JIRA and upload a gcore of the box when it's in the stuck state.
>
>
> On Wed, Mar 16, 2011 at 3:15 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>> I missed I'm using mod_hash for mod_hash_limit, calling
>> execute("limit","hash $realm $key")
>> and
>> api("limit_usage","hash $realm $key")
>> from perl ESL. And there are messages on FS shutdown (from hash_limit
>> cleanup task?)
>> Maybe using mod_hash_limit affect channel destroy sequence in some way?
>> hash_dump shows nothing after channel hangup, but channel which was hungup
>> first stays in channel list forever :(
>> 2011-03-16 11:12:07.454839 [ERR] switch_loadable_module.c:489 Giving up on
>> 'sofia' waiting for existing references.
>>
>> 2011-03-16 11:12:07.944692 [CRIT] sofia.c:1646 Waiting for 2 session(s)
>> 2011-03-16 11:12:08.944692 [CRIT] sofia.c:1646 Waiting for 2 session(s)
>> 2011-03-16 11:12:09.944693 [CRIT] sofia.c:1646 Waiting for 2 session(s)
>> 2011-03-16 11:12:10.944678 [CRIT] sofia.c:1646 Waiting for 2 session(s)
>>
>>
>>
>> 2011/3/12 Anthony Minessale <anthony.minessale at gmail.com>
>>>
>>> this sounds odd,
>>> are you on the latest GIT build?
>>>
>>> try doing a gcore on the box when its like that and run gdb on it and
>>> do thread apply all bt
>>>
>>>
>>> On Fri, Mar 11, 2011 at 3:38 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>> > I use cdr_xml only. If the call fails before entering ESL script, seems
>>> > there are no stuck records, but they appear after some actions in ESL
>>> > even
>>> > if there was no bridge attempt.
>>> >
>>> > 2011/3/11 Anthony Minessale <anthony.minessale at gmail.com>
>>> >>
>>> >> that looks like they are all stuck in the CDR module.
>>> >> What module are you using?
>>> >>
>>> >>
>>> >>
>>> >> On Fri, Mar 11, 2011 at 5:58 AM, Dmitry Sytchev <kbdfck at gmail.com>
>>> >> wrote:
>>> >> > I think this is not the case, but anyway, what to do with these hung
>>> >> > channels?
>>> >> > Maybe I'm doing something wrong while bridging or processing events?
>>> >> > Maybe
>>> >> > unprocessed events can affect channel destroy procedure?
>>> >> > 2011/3/11 Avi Marcus <avi at avimarcus.net>
>>> >> >>
>>> >> >> Regarding ram usage, I'd imagine this is the case:
>>> >> >> http://www.linuxatemyram.com/
>>> >> >> -Avi
>>> >> >>
>>> >> >> On Fri, Mar 11, 2011 at 12:33 PM, Dmitry Sytchev <kbdfck at gmail.com>
>>> >> >> wrote:
>>> >> >>>
>>> >> >>> BTW, uuid_dump?7ae28028-4bc2-11e0-a92f-452f3d3d66ea returns with
>>> >> >>> -ERR
>>> >> >>> No
>>> >> >>> such channel.
>>> >> >>>
>>> >> >>> 2011/3/11 Dmitry Sytchev <kbdfck at gmail.com>
>>> >> >>>>
>>> >> >>>> Hi All
>>> >> >>>> I'm using Perl ESL outbound script to bridge incoming call to sip
>>> >> >>>> endpoint. I'm doing execute("bridge","sofia/user/somebody"), then
>>> >> >>>> processing
>>> >> >>>> events like CHANNEL_HANGUP and EXECUTE_COMPLETE to determine
>>> >> >>>> bridge
>>> >> >>>> result.
>>> >> >>>> Everything works fine, but original incoming call channel is never
>>> >> >>>> removed
>>> >> >>>> from list:
>>> >> >>>> After few calls I see original incoming channels in 'show
>>> >> >>>> channels'
>>> >> >>>> output:
>>> >> >>>> 7ae28028-4bc2-11e0-a92f-452f3d3d66ea,inbound,2011-03-11
>>> >> >>>>
>>> >> >>>>
>>> >> >>>> 12:32:27,1299835947,sofia/external/xxxx at 85.x.x.199,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,xxx.xxx.ru,,,HANGUP,,,,
>>> >> >>>> 883c9eb6-4bc2-11e0-a935-452f3d3d66ea,inbound,2011-03-11
>>> >> >>>>
>>> >> >>>>
>>> >> >>>> 12:32:50,1299835970,sofia/external/xxxx at 85.x.x.199,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,xxx.xxx.ru,,,HANGUP,,,,
>>> >> >>>> 9f6a0f92-4bc2-11e0-a93b-452f3d3d66ea,inbound,2011-03-11
>>> >> >>>>
>>> >> >>>>
>>> >> >>>> 12:33:29,1299836009,sofia/external/xxxx at 85.x.x.199,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,xxx.xxx.ru,,,HANGUP,,,
>>> >> >>>> Also, when I try to stop freeswitch i see these messages on
>>> >> >>>> console:
>>> >> >>>> 2011-03-11 15:28:22.517063 [CRIT] sofia.c:1632 Waiting for 13
>>> >> >>>> session(s)
>>> >> >>>> 2011-03-11 15:28:22.701034 [CRIT] sofia.c:1632 Waiting for 3
>>> >> >>>> session(s)
>>> >> >>>> 2011-03-11 15:28:23.517064 [CRIT] sofia.c:1632 Waiting for 13
>>> >> >>>> session(s)
>>> >> >>>> 2011-03-11 15:28:23.701033 [CRIT] sofia.c:1632 Waiting for 3
>>> >> >>>> session(s)
>>> >> >>>> 2011-03-11 15:28:24.517062 [CRIT] sofia.c:1632 Waiting for 13
>>> >> >>>> session(s)
>>> >> >>>> 2011-03-11 15:28:24.705036 [CRIT] sofia.c:1632 Waiting for 3
>>> >> >>>> session(s)
>>> >> >>>> 2011-03-11 15:28:25.517064 [CRIT] sofia.c:1632 Waiting for 13
>>> >> >>>> session(s)
>>> >> >>>> 2011-03-11 15:28:25.709031 [CRIT] sofia.c:1632 Waiting for 3
>>> >> >>>> session(s)
>>> >> >>>> 2011-03-11 15:28:26.517065 [CRIT] sofia.c:1632 Waiting for 13
>>> >> >>>> session(s)
>>> >> >>>>
>>> >> >>>> Why these channels are not removed from list? I also noticed that
>>> >> >>>> memory
>>> >> >>>> consumption by freeswitch process constantly grows call by call.
>>> >> >>>>
>>> >> >>>> --
>>> >> >>>> Best regards,
>>> >> >>>>
>>> >> >>>> Dmitry Sytchev,
>>> >> >>>> IT Engineer
>>> >> >>>
>>> >> >>>
>>> >> >>>
>>> >> >>> --
>>> >> >>> Best regards,
>>> >> >>>
>>> >> >>> Dmitry Sytchev,
>>> >> >>> IT Engineer
>>> >> >>>
>>> >> >>> _______________________________________________
>>> >> >>> FreeSWITCH-users mailing list
>>> >> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >>>
>>> >> >>>
>>> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >>> http://www.freeswitch.org
>>> >> >>>
>>> >> >>
>>> >> >>
>>> >> >> _______________________________________________
>>> >> >> FreeSWITCH-users mailing list
>>> >> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >>
>>> >> >>
>>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> http://www.freeswitch.org
>>> >> >>
>>> >> >
>>> >> >
>>> >> >
>>> >> > --
>>> >> > Best regards,
>>> >> >
>>> >> > Dmitry Sytchev,
>>> >> > IT Engineer
>>> >> >
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >
>>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> > 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-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> >
>>> > --
>>> > Best regards,
>>> >
>>> > Dmitry Sytchev,
>>> > IT Engineer
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > 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-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Best regards,
>>
>> Dmitry Sytchev,
>> IT Engineer
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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 pbdlists at pinboard.com  Wed Mar 16 22:20:01 2011
From: pbdlists at pinboard.com (pbdlists at pinboard.com)
Date: Wed, 16 Mar 2011 20:20:01 +0100
Subject: [Freeswitch-users] Japanese TTS for Linux - anybody?
Message-ID: <20110316192000.GA14301@pinboard.com>

I've been looking for a TTS system I could use with FreeSWITCH which
runs on Linux and has acceptable quality voices in Japanese, German and
English. Cepstral would fit the bill, except there don't seem to be any
Japanese voices at all. Neospeech would be fine, but for a strictly
personal installation it is financially out of reach (USD 950.-- per
year per voice). And the other engines I found run on Windows.

Does anybody on the list know about a TTS system which would run on
Linux, support Japanese voices and have a price tag of no more than a
one-time cost of $200?

Thanks,

Kurt


From vibha_dear6 at yahoo.co.in  Wed Mar 16 22:33:46 2011
From: vibha_dear6 at yahoo.co.in (vibha dear)
Date: Thu, 17 Mar 2011 01:03:46 +0530 (IST)
Subject: [Freeswitch-users] queuing calls on freeswitch
Message-ID: <581318.87007.qm@web137302.mail.in.yahoo.com>

dear all,
how i can queue calls on freeswitch and how to retrieve from queue?

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From a.afzali2003 at gmail.com  Wed Mar 16 23:00:23 2011
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Wed, 16 Mar 2011 23:30:23 +0330
Subject: [Freeswitch-users] queuing calls on freeswitch
In-Reply-To: <581318.87007.qm@web137302.mail.in.yahoo.com>
References: <581318.87007.qm@web137302.mail.in.yahoo.com>
Message-ID: <AANLkTinuTe2=EMimMSeWM-jy3HSxfswpLU-dhB1Yo0zT@mail.gmail.com>

mod_fifo

On Wed, Mar 16, 2011 at 11:03 PM, vibha dear <vibha_dear6 at yahoo.co.in>wrote:

> dear all,
>
> how i can queue calls on freeswitch and how to retrieve from queue?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From avi at avimarcus.net  Wed Mar 16 23:41:25 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 16 Mar 2011 22:41:25 +0200
Subject: [Freeswitch-users] Japanese TTS for Linux - anybody?
In-Reply-To: <20110316192000.GA14301@pinboard.com>
References: <20110316192000.GA14301@pinboard.com>
Message-ID: <AANLkTikBDdsidPB9sDaQ7AuiARbM0UMbCFkNqEV3uzNR@mail.gmail.com>

Try pulling from google translate via mod_shout? I'd try to cache them...
http://wiki.freeswitch.org/wiki/FS_weekly_2010_11_03#Need_some_FREE_text_to_speech.3F

<http://wiki.freeswitch.org/wiki/FS_weekly_2010_11_03#Need_some_FREE_text_to_speech.3F>
-Avi

On Wed, Mar 16, 2011 at 9:20 PM, <pbdlists at pinboard.com> wrote:

> I've been looking for a TTS system I could use with FreeSWITCH which
> runs on Linux and has acceptable quality voices in Japanese, German and
> English. Cepstral would fit the bill, except there don't seem to be any
> Japanese voices at all. Neospeech would be fine, but for a strictly
> personal installation it is financially out of reach (USD 950.-- per
> year per voice). And the other engines I found run on Windows.
>
> Does anybody on the list know about a TTS system which would run on
> Linux, support Japanese voices and have a price tag of no more than a
> one-time cost of $200?
>
> Thanks,
>
> Kurt
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kbdfck at gmail.com  Thu Mar 17 00:21:52 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Thu, 17 Mar 2011 00:21:52 +0300
Subject: [Freeswitch-users] Unexpected out of order command/reply '+OK
 event listener enabled plain' in perl ESL outbound
In-Reply-To: <AANLkTi=hwzMiU7Dxa8wD4fWmgvf=8Go4DGn+zizJAVub@mail.gmail.com>
References: <AANLkTikZ=pBk99wbg2+M_-vr6qK7=TC9tDZzzvE9a1+u@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B303CB87@cooper>
	<AANLkTim=Sxe8iuQncgn+JJgBzP=u3oQZdvwD2wwZCesC@mail.gmail.com>
	<AANLkTi=hwzMiU7Dxa8wD4fWmgvf=8Go4DGn+zizJAVub@mail.gmail.com>
Message-ID: <AANLkTikZoSvNj5AzrMF_4sHNqk7fLFMv0vjzQo4CxSM0@mail.gmail.com>

I'm using ESL.so and ESL.pm for perl outbound ESL, and Net::Server as server
engine.
I already tried to rebuild and reinstall with `make sure` including perlmod
in esl library dir

2011/3/16 Anthony Minessale <anthony.minessale at gmail.com>

> Are you using your own client or libESL which takes care of this for you?
>
>
> On Wed, Mar 16, 2011 at 11:55 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> > FreeSWITCH Version 1.0.head (git-2c009dd 2011-03-15 14-29-04 -0500)
> > 2011/3/16 Peter Olsson <peter.olsson at visionutveckling.se>
> >>
> >> Are you on latest GIT HEAD?
> >>
> >>
> >>
> >> /Peter
> >>
> >>
> >>
> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Dmitry
> Sytchev
> >> Skickat: den 16 mars 2011 17:29
> >> Till: FreeSWITCH Users Help
> >> ?mne: [Freeswitch-users] Unexpected out of order command/reply '+OK
> event
> >> listener enabled plain' in perl ESL outbound
> >>
> >>
> >>
> >> In my outbound async full ESL script after some moment there are
> >> out-of-order message on every "execute" call, and this breaks my event
> >> handling in some way.
> >>
> >>
> >>
> >> Content-Type: command/reply.Reply-Text: +OK event listener enabled
> plain..
> >>
> >>
> >> I want to execute some commands syncronously, but setEventLock seems to
> be
> >> broken by this behavior.
> >>
> >> As the workaround I had to write lock_for_execute_complete function,
> which
> >> waits for execute completion of applications that need to be run sync,
> but
> >> it effectively destroys async events by processing them while waiting
> for
> >> CHANNEL_EXECUTE_COMPLETE.
> >>
> >>
> >>
> >> here is a part of socket dump, note second "Content-Type:
> >> command/reply.Reply-Text: +OK event listener enabled plain.." right
> after
> >> execute. There is no event command in script besides the very beginning,
> so
> >> why these messages appear?
> >>
> >> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
> >> api uuid_setvar 06d55974-4fe8-11e0-a7c0-e3bb2141c22a
> >> pbx_subscriber_customer_id 1
> >> #
> >> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
> >> ..
> >> ##
> >> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
> >> Content-Type: api/response.Content-Length: 4..
> >> #
> >> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
> >> +OK.
> >> ##
> >> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
> >>   api sofia_contact s000002 at asdasd.com
> >> #
> >> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
> >> ..
> >> ##
> >> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
> >> Content-Type: api/response.Content-Length: 96..
> >> #
> >> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
> >>
> >> sofia/local/sip:s000002 at 172.19.36.54:5061
> ;fs_nat=yes;fs_path=sip%3As000002%40172.19.36.54%3A5061
> >> ##
> >> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
> >> Event-Name: SOCKET_DATA.Content-Type: api/response.Content-Length:
> >> 96..sofia/local/sip:s000002 at 172.19.36.54:5061;fs_nat=
> >> yes;fs_path=sip%3As000002%40172.19.36.54%3A5061
> >> #
> >> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
> >> sendmsg.call-command: execute.execute-app-name: bridge.execute-app-arg:
> >> {originate_timeout=10,hangup_after_bridge=false,
> >>   ignore_early_media=false}user/s000002 at asdasd.com
> >> #
> >> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
> >> Content-Type: command/reply.Reply-Text: +OK event listener enabled
> plain..
> >> #
> >> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
> >> Content-Type: command/reply.Reply-Text: +OK..
> >>
> >> --
> >> Best regards,
> >>
> >> Dmitry Sytchev,
> >> IT Engineer
> >> !DSPAM:4d80e61b32768938086861!
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > Best regards,
> >
> > Dmitry Sytchev,
> > IT Engineer
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From anthony.minessale at gmail.com  Thu Mar 17 00:44:01 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 16 Mar 2011 16:44:01 -0500
Subject: [Freeswitch-users] Unexpected out of order command/reply '+OK
 event listener enabled plain' in perl ESL outbound
In-Reply-To: <AANLkTikZoSvNj5AzrMF_4sHNqk7fLFMv0vjzQo4CxSM0@mail.gmail.com>
References: <AANLkTikZ=pBk99wbg2+M_-vr6qK7=TC9tDZzzvE9a1+u@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B303CB87@cooper>
	<AANLkTim=Sxe8iuQncgn+JJgBzP=u3oQZdvwD2wwZCesC@mail.gmail.com>
	<AANLkTi=hwzMiU7Dxa8wD4fWmgvf=8Go4DGn+zizJAVub@mail.gmail.com>
	<AANLkTikZoSvNj5AzrMF_4sHNqk7fLFMv0vjzQo4CxSM0@mail.gmail.com>
Message-ID: <AANLkTinB0CozQfgv6q0wj0_QJZ1BKfYaDD1iURxm4mcB@mail.gmail.com>

You sure you did not install old ESL.so in your perl path?

btw, its "make current" not "make sure" that is an old one.

I would also turn on the debug level in the script as well.

We have no reports of any problems like this.

Speaking of, why are you doing this here instead of JIRA?


On Wed, Mar 16, 2011 at 4:21 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> I'm using ESL.so and ESL.pm for perl outbound ESL, and Net::Server as server
> engine.
> I already tried to rebuild and reinstall with `make sure` including perlmod
> in esl library dir
>
> 2011/3/16 Anthony Minessale <anthony.minessale at gmail.com>
>>
>> Are you using your own client or libESL which takes care of this for you?
>>
>>
>> On Wed, Mar 16, 2011 at 11:55 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>> > FreeSWITCH Version 1.0.head (git-2c009dd 2011-03-15 14-29-04 -0500)
>> > 2011/3/16 Peter Olsson <peter.olsson at visionutveckling.se>
>> >>
>> >> Are you on latest GIT HEAD?
>> >>
>> >>
>> >>
>> >> /Peter
>> >>
>> >>
>> >>
>> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Dmitry
>> >> Sytchev
>> >> Skickat: den 16 mars 2011 17:29
>> >> Till: FreeSWITCH Users Help
>> >> ?mne: [Freeswitch-users] Unexpected out of order command/reply '+OK
>> >> event
>> >> listener enabled plain' in perl ESL outbound
>> >>
>> >>
>> >>
>> >> In my outbound async full ESL script after some moment there are
>> >> out-of-order message on every "execute" call, and this breaks my event
>> >> handling in some way.
>> >>
>> >>
>> >>
>> >> Content-Type: command/reply.Reply-Text: +OK event listener enabled
>> >> plain..
>> >>
>> >>
>> >> I want to execute some commands syncronously, but setEventLock seems to
>> >> be
>> >> broken by this behavior.
>> >>
>> >> As the workaround I had to write lock_for_execute_complete function,
>> >> which
>> >> waits for execute completion of applications that need to be run sync,
>> >> but
>> >> it effectively destroys async events by processing them while waiting
>> >> for
>> >> CHANNEL_EXECUTE_COMPLETE.
>> >>
>> >>
>> >>
>> >> here is a part of socket dump,?note second "Content-Type:
>> >> command/reply.Reply-Text: +OK event listener enabled plain.." right
>> >> after
>> >> execute.?There is no event command in script besides the very
>> >> beginning, so
>> >> why these messages appear?
>> >>
>> >> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
>> >> api uuid_setvar 06d55974-4fe8-11e0-a7c0-e3bb2141c22a
>> >> pbx_subscriber_customer_id 1
>> >> #
>> >> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
>> >> ..
>> >> ##
>> >> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
>> >> Content-Type: api/response.Content-Length: 4..
>> >> #
>> >> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
>> >> +OK.
>> >> ##
>> >> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
>> >> ? api sofia_contact s000002 at asdasd.com
>> >> #
>> >> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
>> >> ..
>> >> ##
>> >> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
>> >> Content-Type: api/response.Content-Length: 96..
>> >> #
>> >> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
>> >>
>> >>
>> >> sofia/local/sip:s000002 at 172.19.36.54:5061;fs_nat=yes;fs_path=sip%3As000002%40172.19.36.54%3A5061
>> >> ##
>> >> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
>> >> Event-Name: SOCKET_DATA.Content-Type: api/response.Content-Length:
>> >> 96..sofia/local/sip:s000002 at 172.19.36.54:5061;fs_nat=
>> >> yes;fs_path=sip%3As000002%40172.19.36.54%3A5061
>> >> #
>> >> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
>> >> sendmsg.call-command: execute.execute-app-name: bridge.execute-app-arg:
>> >> {originate_timeout=10,hangup_after_bridge=false,
>> >> ? ignore_early_media=false}user/s000002 at asdasd.com
>> >> #
>> >> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
>> >> Content-Type: command/reply.Reply-Text: +OK event listener enabled
>> >> plain..
>> >> #
>> >> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
>> >> Content-Type: command/reply.Reply-Text: +OK..
>> >>
>> >> --
>> >> Best regards,
>> >>
>> >> Dmitry Sytchev,
>> >> IT Engineer
>> >> !DSPAM:4d80e61b32768938086861!
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> > --
>> > Best regards,
>> >
>> > Dmitry Sytchev,
>> > IT Engineer
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 avi at avimarcus.net  Thu Mar 17 00:44:56 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 16 Mar 2011 23:44:56 +0200
Subject: [Freeswitch-users] Want more FS English Sounds?
Message-ID: <AANLkTi=Dsdg7jbAmmauc-Q1Ba6n=V47e=gfSNQp_500c@mail.gmail.com>

mercutioviz says that FS is putting in an order for more sounds... is there
a stock sound (not something custom for yourself..) that you think FS would
benefit by having?
List it here!
http://wiki.freeswitch.org/wiki/Sounds_order

-Avi
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From kbdfck at gmail.com  Thu Mar 17 02:36:02 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Thu, 17 Mar 2011 02:36:02 +0300
Subject: [Freeswitch-users] Unexpected out of order command/reply '+OK
 event listener enabled plain' in perl ESL outbound
In-Reply-To: <AANLkTinB0CozQfgv6q0wj0_QJZ1BKfYaDD1iURxm4mcB@mail.gmail.com>
References: <AANLkTikZ=pBk99wbg2+M_-vr6qK7=TC9tDZzzvE9a1+u@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B303CB87@cooper>
	<AANLkTim=Sxe8iuQncgn+JJgBzP=u3oQZdvwD2wwZCesC@mail.gmail.com>
	<AANLkTi=hwzMiU7Dxa8wD4fWmgvf=8Go4DGn+zizJAVub@mail.gmail.com>
	<AANLkTikZoSvNj5AzrMF_4sHNqk7fLFMv0vjzQo4CxSM0@mail.gmail.com>
	<AANLkTinB0CozQfgv6q0wj0_QJZ1BKfYaDD1iURxm4mcB@mail.gmail.com>
Message-ID: <AANLkTinmErNdN52QV2P3M-jVCnrVTR7u--eZdEVgn=6x@mail.gmail.com>

Ok, I'll try to check library and rebuild with `make current`.
I posted it here as I thought that maybe this is my fault somewhere in
script or event handling logic.
I'll open Jira issue if I can't deal with this error by full recompilation.

BTW, if `make sure` is outdated, maybe it's time to remove reference to it
from post-build info that FS displays after make?

Thanks for your help

2011/3/17 Anthony Minessale <anthony.minessale at gmail.com>

> You sure you did not install old ESL.so in your perl path?
>
> btw, its "make current" not "make sure" that is an old one.
>
> I would also turn on the debug level in the script as well.
>
> We have no reports of any problems like this.
>
> Speaking of, why are you doing this here instead of JIRA?
>
>
> On Wed, Mar 16, 2011 at 4:21 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> > I'm using ESL.so and ESL.pm for perl outbound ESL, and Net::Server as
> server
> > engine.
> > I already tried to rebuild and reinstall with `make sure` including
> perlmod
> > in esl library dir
> >
> > 2011/3/16 Anthony Minessale <anthony.minessale at gmail.com>
> >>
> >> Are you using your own client or libESL which takes care of this for
> you?
> >>
> >>
> >> On Wed, Mar 16, 2011 at 11:55 AM, Dmitry Sytchev <kbdfck at gmail.com>
> wrote:
> >> > FreeSWITCH Version 1.0.head (git-2c009dd 2011-03-15 14-29-04 -0500)
> >> > 2011/3/16 Peter Olsson <peter.olsson at visionutveckling.se>
> >> >>
> >> >> Are you on latest GIT HEAD?
> >> >>
> >> >>
> >> >>
> >> >> /Peter
> >> >>
> >> >>
> >> >>
> >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> >> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Dmitry
> >> >> Sytchev
> >> >> Skickat: den 16 mars 2011 17:29
> >> >> Till: FreeSWITCH Users Help
> >> >> ?mne: [Freeswitch-users] Unexpected out of order command/reply '+OK
> >> >> event
> >> >> listener enabled plain' in perl ESL outbound
> >> >>
> >> >>
> >> >>
> >> >> In my outbound async full ESL script after some moment there are
> >> >> out-of-order message on every "execute" call, and this breaks my
> event
> >> >> handling in some way.
> >> >>
> >> >>
> >> >>
> >> >> Content-Type: command/reply.Reply-Text: +OK event listener enabled
> >> >> plain..
> >> >>
> >> >>
> >> >> I want to execute some commands syncronously, but setEventLock seems
> to
> >> >> be
> >> >> broken by this behavior.
> >> >>
> >> >> As the workaround I had to write lock_for_execute_complete function,
> >> >> which
> >> >> waits for execute completion of applications that need to be run
> sync,
> >> >> but
> >> >> it effectively destroys async events by processing them while waiting
> >> >> for
> >> >> CHANNEL_EXECUTE_COMPLETE.
> >> >>
> >> >>
> >> >>
> >> >> here is a part of socket dump, note second "Content-Type:
> >> >> command/reply.Reply-Text: +OK event listener enabled plain.." right
> >> >> after
> >> >> execute. There is no event command in script besides the very
> >> >> beginning, so
> >> >> why these messages appear?
> >> >>
> >> >> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
> >> >> api uuid_setvar 06d55974-4fe8-11e0-a7c0-e3bb2141c22a
> >> >> pbx_subscriber_customer_id 1
> >> >> #
> >> >> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
> >> >> ..
> >> >> ##
> >> >> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
> >> >> Content-Type: api/response.Content-Length: 4..
> >> >> #
> >> >> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
> >> >> +OK.
> >> >> ##
> >> >> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
> >> >>   api sofia_contact s000002 at asdasd.com
> >> >> #
> >> >> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
> >> >> ..
> >> >> ##
> >> >> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
> >> >> Content-Type: api/response.Content-Length: 96..
> >> >> #
> >> >> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
> >> >>
> >> >>
> >> >> sofia/local/sip:s000002 at 172.19.36.54:5061
> ;fs_nat=yes;fs_path=sip%3As000002%40172.19.36.54%3A5061
> >> >> ##
> >> >> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
> >> >> Event-Name: SOCKET_DATA.Content-Type: api/response.Content-Length:
> >> >> 96..sofia/local/sip:s000002 at 172.19.36.54:5061;fs_nat=
> >> >> yes;fs_path=sip%3As000002%40172.19.36.54%3A5061
> >> >> #
> >> >> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
> >> >> sendmsg.call-command: execute.execute-app-name:
> bridge.execute-app-arg:
> >> >> {originate_timeout=10,hangup_after_bridge=false,
> >> >>   ignore_early_media=false}user/s000002 at asdasd.com
> >> >> #
> >> >> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
> >> >> Content-Type: command/reply.Reply-Text: +OK event listener enabled
> >> >> plain..
> >> >> #
> >> >> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
> >> >> Content-Type: command/reply.Reply-Text: +OK..
> >> >>
> >> >> --
> >> >> Best regards,
> >> >>
> >> >> Dmitry Sytchev,
> >> >> IT Engineer
> >> >> !DSPAM:4d80e61b32768938086861!
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >
> >> >
> >> >
> >> > --
> >> > Best regards,
> >> >
> >> > Dmitry Sytchev,
> >> > IT Engineer
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > 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-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Best regards,
> >
> > Dmitry Sytchev,
> > IT Engineer
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From Nabble at slickdeals.endjunk.com  Thu Mar 17 05:27:11 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 16 Mar 2011 19:27:11 -0700 (PDT)
Subject: [Freeswitch-users] Want more FS English Sounds?
In-Reply-To: <AANLkTi=Dsdg7jbAmmauc-Q1Ba6n=V47e=gfSNQp_500c@mail.gmail.com>
References: <AANLkTi=Dsdg7jbAmmauc-Q1Ba6n=V47e=gfSNQp_500c@mail.gmail.com>
Message-ID: <1300328831817-6179533.post@n2.nabble.com>

How about some sound files to deter telemarketers?

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Want-more-FS-English-Sounds-tp6179015p6179533.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From sharad at coraltele.com  Thu Mar 17 08:28:47 2011
From: sharad at coraltele.com (sharad)
Date: Thu, 17 Mar 2011 10:58:47 +0530
Subject: [Freeswitch-users] Mute / Unmute with Freeswitch Conference
References: <mailman.5468.1300318594.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <D0B056F9F4F84EFB9C684F9813351666@sharad>

Hi friends

As of now when we mute a participant, conference server mutes that 
participant while RTP keeps on coming from that UA. This consumes the 
bandwidth & precessing time also is there. This issue elobrates when no. of 
conference participants are increased.

So I am just wondering whether we can mute the conference participants using 
the SIP SDP `reconly' & whenever we wants to unmute him, FS should again 
send the INVITE SDP with `sendonly'.

Plz let me know if it is feasible.

Thanks in advance.

regards
Sharad 



From yehavi.bourvine at gmail.com  Thu Mar 17 10:18:36 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 17 Mar 2011 09:18:36 +0200
Subject: [Freeswitch-users] Freeswitch database polling
Message-ID: <AANLkTi=7dyWZ=Un5yA9S6kndWdRH8zryOZj5wbKNL+p2@mail.gmail.com>

Hello,

  A question for curiosity: We use ODBC over MySQL to serve the FS core
database. This database is replicated (by MySQL means) to a backup system
which runs MySQL also. There is no cluster between the two, only one-way
MySQL replication.

  The FS on on the other system shows an updated current status of all
extensions registered (sofia status profile XXX). How does it do so? a
timely update/polling of the database? If so, what is the interval, and
doesn't it make much load on MySQL  database when there are a lot of
registered phones?

                                      Thanks! __Yehavi:
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From steveayre at gmail.com  Thu Mar 17 10:34:55 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 17 Mar 2011 07:34:55 +0000
Subject: [Freeswitch-users] Freeswitch database polling
In-Reply-To: <AANLkTi=7dyWZ=Un5yA9S6kndWdRH8zryOZj5wbKNL+p2@mail.gmail.com>
References: <AANLkTi=7dyWZ=Un5yA9S6kndWdRH8zryOZj5wbKNL+p2@mail.gmail.com>
Message-ID: <AANLkTikpP3RDGy2rivrYpdASd+oPC6SEx=O+COAxjpf4@mail.gmail.com>

When you run the 'sofia status profile xxx' command it does a database
lookup. There's no polling, just a lookup when it needs to see who is
registered.

-Steve


On 17 March 2011 07:18, Yehavi Bourvine <yehavi.bourvine at gmail.com> wrote:

> Hello,
>
>   A question for curiosity: We use ODBC over MySQL to serve the FS core
> database. This database is replicated (by MySQL means) to a backup system
> which runs MySQL also. There is no cluster between the two, only one-way
> MySQL replication.
>
>   The FS on on the other system shows an updated current status of all
> extensions registered (sofia status profile XXX). How does it do so? a
> timely update/polling of the database? If so, what is the interval, and
> doesn't it make much load on MySQL  database when there are a lot of
> registered phones?
>
>                                       Thanks! __Yehavi:
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kbdfck at gmail.com  Thu Mar 17 13:03:16 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Thu, 17 Mar 2011 13:03:16 +0300
Subject: [Freeswitch-users] Already hung channels in 'show channels'
 output never go away
In-Reply-To: <AANLkTi=NE9Swjk0KYM_KPCUnUQYuwsqAibXOU10SD6w5@mail.gmail.com>
References: <AANLkTi=jiL-aSmkDusoH_9EYRzPZ1ePXkWAQHXn53Np5@mail.gmail.com>
	<AANLkTinUg76_ncbjCwf=W8T5BPSS3FdstGQkkptv-xme@mail.gmail.com>
	<AANLkTimzvf-ZaiaVKCprbHh6JcFx3dXnbaU54T8KL=O6@mail.gmail.com>
	<AANLkTinTApDm-YL5UjqPnc8vaE-e+xpdAXiwaD+DTZhX@mail.gmail.com>
	<AANLkTik_CE66S0FToNMycX+U_WV3m12T=8D4t88BdcKP@mail.gmail.com>
	<AANLkTikTQpMHe67cSHr3T9Xr7UFVTu=RWXnNthxaBqCV@mail.gmail.com>
	<AANLkTimAJ2ULQZk55B63zBoVnKK+E9nrQeCR8tP5CuFQ@mail.gmail.com>
	<AANLkTinXhqZ8A=B8QNJwy7AfboaE2bTttHRjrTemNyEB@mail.gmail.com>
	<AANLkTi=Gdc8E8-xHeCNdpH5x-FHESEkjWgWN6H_s8_0m@mail.gmail.com>
	<AANLkTi=NE9Swjk0KYM_KPCUnUQYuwsqAibXOU10SD6w5@mail.gmail.com>
Message-ID: <AANLkTikmpKfyDGfiHDNOr_r7XXRJBDqraV7sL+g4PBGz@mail.gmail.com>

http://jira.freeswitch.org/browse/FS-3157

2011/3/16 Anthony Minessale <anthony.minessale at gmail.com>

> can you also get some console traces on debug level of the traffic
> happening on the box.
>
>
> On Wed, Mar 16, 2011 at 12:55 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > yes you probably found a use case where the rwlock is not released
> > Can you open a JIRA and upload a gcore of the box when it's in the stuck
> state.
> >
> >
> > On Wed, Mar 16, 2011 at 3:15 AM, Dmitry Sytchev <kbdfck at gmail.com>
> wrote:
> >> I missed I'm using mod_hash for mod_hash_limit, calling
> >> execute("limit","hash $realm $key")
> >> and
> >> api("limit_usage","hash $realm $key")
> >> from perl ESL. And there are messages on FS shutdown (from hash_limit
> >> cleanup task?)
> >> Maybe using mod_hash_limit affect channel destroy sequence in some way?
> >> hash_dump shows nothing after channel hangup, but channel which was
> hungup
> >> first stays in channel list forever :(
> >> 2011-03-16 11:12:07.454839 [ERR] switch_loadable_module.c:489 Giving up
> on
> >> 'sofia' waiting for existing references.
> >>
> >> 2011-03-16 11:12:07.944692 [CRIT] sofia.c:1646 Waiting for 2 session(s)
> >> 2011-03-16 11:12:08.944692 [CRIT] sofia.c:1646 Waiting for 2 session(s)
> >> 2011-03-16 11:12:09.944693 [CRIT] sofia.c:1646 Waiting for 2 session(s)
> >> 2011-03-16 11:12:10.944678 [CRIT] sofia.c:1646 Waiting for 2 session(s)
> >>
> >>
> >>
> >> 2011/3/12 Anthony Minessale <anthony.minessale at gmail.com>
> >>>
> >>> this sounds odd,
> >>> are you on the latest GIT build?
> >>>
> >>> try doing a gcore on the box when its like that and run gdb on it and
> >>> do thread apply all bt
> >>>
> >>>
> >>> On Fri, Mar 11, 2011 at 3:38 PM, Dmitry Sytchev <kbdfck at gmail.com>
> wrote:
> >>> > I use cdr_xml only. If the call fails before entering ESL script,
> seems
> >>> > there are no stuck records, but they appear after some actions in ESL
> >>> > even
> >>> > if there was no bridge attempt.
> >>> >
> >>> > 2011/3/11 Anthony Minessale <anthony.minessale at gmail.com>
> >>> >>
> >>> >> that looks like they are all stuck in the CDR module.
> >>> >> What module are you using?
> >>> >>
> >>> >>
> >>> >>
> >>> >> On Fri, Mar 11, 2011 at 5:58 AM, Dmitry Sytchev <kbdfck at gmail.com>
> >>> >> wrote:
> >>> >> > I think this is not the case, but anyway, what to do with these
> hung
> >>> >> > channels?
> >>> >> > Maybe I'm doing something wrong while bridging or processing
> events?
> >>> >> > Maybe
> >>> >> > unprocessed events can affect channel destroy procedure?
> >>> >> > 2011/3/11 Avi Marcus <avi at avimarcus.net>
> >>> >> >>
> >>> >> >> Regarding ram usage, I'd imagine this is the case:
> >>> >> >> http://www.linuxatemyram.com/
> >>> >> >> -Avi
> >>> >> >>
> >>> >> >> On Fri, Mar 11, 2011 at 12:33 PM, Dmitry Sytchev <
> kbdfck at gmail.com>
> >>> >> >> wrote:
> >>> >> >>>
> >>> >> >>> BTW, uuid_dump 7ae28028-4bc2-11e0-a92f-452f3d3d66ea returns with
> >>> >> >>> -ERR
> >>> >> >>> No
> >>> >> >>> such channel.
> >>> >> >>>
> >>> >> >>> 2011/3/11 Dmitry Sytchev <kbdfck at gmail.com>
> >>> >> >>>>
> >>> >> >>>> Hi All
> >>> >> >>>> I'm using Perl ESL outbound script to bridge incoming call to
> sip
> >>> >> >>>> endpoint. I'm doing execute("bridge","sofia/user/somebody"),
> then
> >>> >> >>>> processing
> >>> >> >>>> events like CHANNEL_HANGUP and EXECUTE_COMPLETE to determine
> >>> >> >>>> bridge
> >>> >> >>>> result.
> >>> >> >>>> Everything works fine, but original incoming call channel is
> never
> >>> >> >>>> removed
> >>> >> >>>> from list:
> >>> >> >>>> After few calls I see original incoming channels in 'show
> >>> >> >>>> channels'
> >>> >> >>>> output:
> >>> >> >>>> 7ae28028-4bc2-11e0-a92f-452f3d3d66ea,inbound,2011-03-11
> >>> >> >>>>
> >>> >> >>>>
> >>> >> >>>> 12:32:27,1299835947,sofia/external/xxxx at 85.x.x.199
> ,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,
> xxx.xxx.ru,,,HANGUP,,,,
> >>> >> >>>> 883c9eb6-4bc2-11e0-a935-452f3d3d66ea,inbound,2011-03-11
> >>> >> >>>>
> >>> >> >>>>
> >>> >> >>>> 12:32:50,1299835970,sofia/external/xxxx at 85.x.x.199
> ,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,
> xxx.xxx.ru,,,HANGUP,,,,
> >>> >> >>>> 9f6a0f92-4bc2-11e0-a93b-452f3d3d66ea,inbound,2011-03-11
> >>> >> >>>>
> >>> >> >>>>
> >>> >> >>>> 12:33:29,1299836009,sofia/external/xxxx at 85.x.x.199
> ,CS_REPORTING,622xxxx,622xxxx,85.114.x.x,622xxxx,hangup,USER_BUSY,XML,route_in,,,,,,,,
> xxx.xxx.ru,,,HANGUP,,,
> >>> >> >>>> Also, when I try to stop freeswitch i see these messages on
> >>> >> >>>> console:
> >>> >> >>>> 2011-03-11 15:28:22.517063 [CRIT] sofia.c:1632 Waiting for 13
> >>> >> >>>> session(s)
> >>> >> >>>> 2011-03-11 15:28:22.701034 [CRIT] sofia.c:1632 Waiting for 3
> >>> >> >>>> session(s)
> >>> >> >>>> 2011-03-11 15:28:23.517064 [CRIT] sofia.c:1632 Waiting for 13
> >>> >> >>>> session(s)
> >>> >> >>>> 2011-03-11 15:28:23.701033 [CRIT] sofia.c:1632 Waiting for 3
> >>> >> >>>> session(s)
> >>> >> >>>> 2011-03-11 15:28:24.517062 [CRIT] sofia.c:1632 Waiting for 13
> >>> >> >>>> session(s)
> >>> >> >>>> 2011-03-11 15:28:24.705036 [CRIT] sofia.c:1632 Waiting for 3
> >>> >> >>>> session(s)
> >>> >> >>>> 2011-03-11 15:28:25.517064 [CRIT] sofia.c:1632 Waiting for 13
> >>> >> >>>> session(s)
> >>> >> >>>> 2011-03-11 15:28:25.709031 [CRIT] sofia.c:1632 Waiting for 3
> >>> >> >>>> session(s)
> >>> >> >>>> 2011-03-11 15:28:26.517065 [CRIT] sofia.c:1632 Waiting for 13
> >>> >> >>>> session(s)
> >>> >> >>>>
> >>> >> >>>> Why these channels are not removed from list? I also noticed
> that
> >>> >> >>>> memory
> >>> >> >>>> consumption by freeswitch process constantly grows call by
> call.
> >>> >> >>>>
> >>> >> >>>> --
> >>> >> >>>> Best regards,
> >>> >> >>>>
> >>> >> >>>> Dmitry Sytchev,
> >>> >> >>>> IT Engineer
> >>> >> >>>
> >>> >> >>>
> >>> >> >>>
> >>> >> >>> --
> >>> >> >>> Best regards,
> >>> >> >>>
> >>> >> >>> Dmitry Sytchev,
> >>> >> >>> IT Engineer
> >>> >> >>>
> >>> >> >>> _______________________________________________
> >>> >> >>> FreeSWITCH-users mailing list
> >>> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> >>>
> >>> >> >>>
> >>> >> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> >>> http://www.freeswitch.org
> >>> >> >>>
> >>> >> >>
> >>> >> >>
> >>> >> >> _______________________________________________
> >>> >> >> FreeSWITCH-users mailing list
> >>> >> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> >>
> >>> >> >>
> >>> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> >> http://www.freeswitch.org
> >>> >> >>
> >>> >> >
> >>> >> >
> >>> >> >
> >>> >> > --
> >>> >> > Best regards,
> >>> >> >
> >>> >> > Dmitry Sytchev,
> >>> >> > IT Engineer
> >>> >> >
> >>> >> > _______________________________________________
> >>> >> > FreeSWITCH-users mailing list
> >>> >> > FreeSWITCH-users at lists.freeswitch.org
> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> >
> >>> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> > 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-users mailing list
> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>
> >>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> http://www.freeswitch.org
> >>> >
> >>> >
> >>> >
> >>> > --
> >>> > Best regards,
> >>> >
> >>> > Dmitry Sytchev,
> >>> > IT Engineer
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > 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-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >> --
> >> Best regards,
> >>
> >> Dmitry Sytchev,
> >> IT Engineer
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From neilp at cs.stanford.edu  Thu Mar 17 13:21:36 2011
From: neilp at cs.stanford.edu (Neil Patel)
Date: Thu, 17 Mar 2011 15:51:36 +0530
Subject: [Freeswitch-users] can't make mod_shout on latest git
Message-ID: <AANLkTimeCqmS58vrV89TUfSRGP1QW4Krg=5d7YPS+Zhc@mail.gmail.com>

>From latest git, I'm getting this error on make all:

making all mod_shout
Creating mod_shout.la...
quiet_libtool: link: cannot find the library
`/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib//libvorbis.la' or
unhandled argument `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib//
libvorbis.la'
make[5]: *** [mod_shout.la] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_shout-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2


Looks like there is a path specification error/typo (I have libvorbis.la in
/usr/lib/)? Not sure how to fix, but would be great if someone could.

Best,
Neil
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From covici at ccs.covici.com  Thu Mar 17 13:51:36 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 17 Mar 2011 06:51:36 -0400
Subject: [Freeswitch-users] very long delays in conference audio
Message-ID: <15324.1300359096@ccs.covici.com>

Hi.  I have been experiencing very long delays in conference audio --
usually over the internet.  For instance, it can take 1 second or more
between the time you hit 0 and when it says you are muted or unmuted.  I
even had this problem during the freeswitch conference.  What can be
done about this -- any way to tell why this is happening, because it
makes conversation on the calls very difficult.  I had one last night
where they gave up because of this problem.

Any ideas would be appreciated.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From steveayre at gmail.com  Thu Mar 17 14:42:17 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 17 Mar 2011 11:42:17 +0000
Subject: [Freeswitch-users] very long delays in conference audio
In-Reply-To: <15324.1300359096@ccs.covici.com>
References: <15324.1300359096@ccs.covici.com>
Message-ID: <AANLkTikNveCvuZOXoKxKTWC11dk5EfnhJB-446iN+C99@mail.gmail.com>

Do you know what the latency (test RTT with ping) and jitter is like? It
could be that over the internet it takes a while for the data to arrive, or
if you have had high latency/jitter a jitterbuffer could be introducing a 1
second delay to compensate.

-Steve


On 17 March 2011 10:51, <covici at ccs.covici.com> wrote:

> Hi.  I have been experiencing very long delays in conference audio --
> usually over the internet.  For instance, it can take 1 second or more
> between the time you hit 0 and when it says you are muted or unmuted.  I
> even had this problem during the freeswitch conference.  What can be
> done about this -- any way to tell why this is happening, because it
> makes conversation on the calls very difficult.  I had one last night
> where they gave up because of this problem.
>
> Any ideas would be appreciated.
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From awais-nazeer at hotmail.com  Thu Mar 17 14:51:50 2011
From: awais-nazeer at hotmail.com (awais nazir)
Date: Thu, 17 Mar 2011 16:51:50 +0500
Subject: [Freeswitch-users] Multiple profile on same ports
Message-ID: <BAY156-w11045DFF35A3CF365E77188B10@phx.gbl>


Hello

I have intermediate level of experience in freeswitch.

I want to achieve a scenario that multiple sip-profiles can share 1 sip bindport 5060.  And sip profile can be attached to a call in using pattern matching in incoming calls.

--Waisee


 		 	   		  
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From avi at avimarcus.net  Thu Mar 17 14:54:44 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 17 Mar 2011 13:54:44 +0200
Subject: [Freeswitch-users] Multiple profile on same ports
In-Reply-To: <BAY156-w11045DFF35A3CF365E77188B10@phx.gbl>
References: <BAY156-w11045DFF35A3CF365E77188B10@phx.gbl>
Message-ID: <AANLkTik9TAZ2oaGtgOBMDk3Tn7A6L+GZMfA++PPorq=T@mail.gmail.com>

Basically, I don't think that possible. But uh, why would you want to?
If you want users authing to different contexts, then that's just a user
variable. What are you trying to achieve here?

-Avi

On Thu, Mar 17, 2011 at 1:51 PM, awais nazir <awais-nazeer at hotmail.com>wrote:

>  Hello
>
> I have intermediate level of experience in freeswitch.
>
> I want to achieve a scenario that multiple sip-profiles can share 1 sip
> bindport 5060.  And sip profile can be attached to a call in using pattern
> matching in incoming calls.
>
> --Waisee
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kbdfck at gmail.com  Thu Mar 17 15:07:40 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Thu, 17 Mar 2011 15:07:40 +0300
Subject: [Freeswitch-users] Unexpected out of order command/reply '+OK
 event listener enabled plain' in perl ESL outbound
In-Reply-To: <AANLkTinmErNdN52QV2P3M-jVCnrVTR7u--eZdEVgn=6x@mail.gmail.com>
References: <AANLkTikZ=pBk99wbg2+M_-vr6qK7=TC9tDZzzvE9a1+u@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B303CB87@cooper>
	<AANLkTim=Sxe8iuQncgn+JJgBzP=u3oQZdvwD2wwZCesC@mail.gmail.com>
	<AANLkTi=hwzMiU7Dxa8wD4fWmgvf=8Go4DGn+zizJAVub@mail.gmail.com>
	<AANLkTikZoSvNj5AzrMF_4sHNqk7fLFMv0vjzQo4CxSM0@mail.gmail.com>
	<AANLkTinB0CozQfgv6q0wj0_QJZ1BKfYaDD1iURxm4mcB@mail.gmail.com>
	<AANLkTinmErNdN52QV2P3M-jVCnrVTR7u--eZdEVgn=6x@mail.gmail.com>
Message-ID: <AANLkTikon7MWc3vDQ9--SBcCB9ZzDy3gGUZHRn=OO8aF@mail.gmail.com>

I have only current ESL.so in perl path.
Debug shows that message has duplicate headers:

[DEBUG] src/esl.c:1166 esl_send() SEND
sendmsg
call-command: execute
execute-app-name: bridge
execute-app-arg:
{originate_timeout=10,hangup_after_bridge=false,ignore_early_media=false}user/
s000001 at yota.access.obit.ru

[DEBUG] src/esl.c:977 esl_recv_event() RECV HEADER [Content-Type] =
[command/reply]
[DEBUG] src/esl.c:977 esl_recv_event() RECV HEADER [Reply-Text] = [+OK event
listener enabled plain]
[DEBUG] src/esl.c:977 esl_recv_event() RECV HEADER [Content-Type] =
[command/reply]
[DEBUG] src/esl.c:977 esl_recv_event() RECV HEADER [Reply-Text] = [+OK]
[DEBUG] src/esl.c:1138 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: command/reply
Reply-Text: +OK


2011/3/17 Dmitry Sytchev <kbdfck at gmail.com>

> Ok, I'll try to check library and rebuild with `make current`.
> I posted it here as I thought that maybe this is my fault somewhere in
> script or event handling logic.
> I'll open Jira issue if I can't deal with this error by full recompilation.
>
> BTW, if `make sure` is outdated, maybe it's time to remove reference to it
> from post-build info that FS displays after make?
>
> Thanks for your help
>
>
> 2011/3/17 Anthony Minessale <anthony.minessale at gmail.com>
>
>> You sure you did not install old ESL.so in your perl path?
>>
>> btw, its "make current" not "make sure" that is an old one.
>>
>> I would also turn on the debug level in the script as well.
>>
>> We have no reports of any problems like this.
>>
>> Speaking of, why are you doing this here instead of JIRA?
>>
>>
>> On Wed, Mar 16, 2011 at 4:21 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>> > I'm using ESL.so and ESL.pm for perl outbound ESL, and Net::Server as
>> server
>> > engine.
>> > I already tried to rebuild and reinstall with `make sure` including
>> perlmod
>> > in esl library dir
>> >
>> > 2011/3/16 Anthony Minessale <anthony.minessale at gmail.com>
>> >>
>> >> Are you using your own client or libESL which takes care of this for
>> you?
>> >>
>> >>
>> >> On Wed, Mar 16, 2011 at 11:55 AM, Dmitry Sytchev <kbdfck at gmail.com>
>> wrote:
>> >> > FreeSWITCH Version 1.0.head (git-2c009dd 2011-03-15 14-29-04 -0500)
>> >> > 2011/3/16 Peter Olsson <peter.olsson at visionutveckling.se>
>> >> >>
>> >> >> Are you on latest GIT HEAD?
>> >> >>
>> >> >>
>> >> >>
>> >> >> /Peter
>> >> >>
>> >> >>
>> >> >>
>> >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> >> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Dmitry
>> >> >> Sytchev
>> >> >> Skickat: den 16 mars 2011 17:29
>> >> >> Till: FreeSWITCH Users Help
>> >> >> ?mne: [Freeswitch-users] Unexpected out of order command/reply '+OK
>> >> >> event
>> >> >> listener enabled plain' in perl ESL outbound
>> >> >>
>> >> >>
>> >> >>
>> >> >> In my outbound async full ESL script after some moment there are
>> >> >> out-of-order message on every "execute" call, and this breaks my
>> event
>> >> >> handling in some way.
>> >> >>
>> >> >>
>> >> >>
>> >> >> Content-Type: command/reply.Reply-Text: +OK event listener enabled
>> >> >> plain..
>> >> >>
>> >> >>
>> >> >> I want to execute some commands syncronously, but setEventLock seems
>> to
>> >> >> be
>> >> >> broken by this behavior.
>> >> >>
>> >> >> As the workaround I had to write lock_for_execute_complete function,
>> >> >> which
>> >> >> waits for execute completion of applications that need to be run
>> sync,
>> >> >> but
>> >> >> it effectively destroys async events by processing them while
>> waiting
>> >> >> for
>> >> >> CHANNEL_EXECUTE_COMPLETE.
>> >> >>
>> >> >>
>> >> >>
>> >> >> here is a part of socket dump, note second "Content-Type:
>> >> >> command/reply.Reply-Text: +OK event listener enabled plain.." right
>> >> >> after
>> >> >> execute. There is no event command in script besides the very
>> >> >> beginning, so
>> >> >> why these messages appear?
>> >> >>
>> >> >> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
>> >> >> api uuid_setvar 06d55974-4fe8-11e0-a7c0-e3bb2141c22a
>> >> >> pbx_subscriber_customer_id 1
>> >> >> #
>> >> >> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
>> >> >> ..
>> >> >> ##
>> >> >> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
>> >> >> Content-Type: api/response.Content-Length: 4..
>> >> >> #
>> >> >> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
>> >> >> +OK.
>> >> >> ##
>> >> >> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
>> >> >>   api sofia_contact s000002 at asdasd.com
>> >> >> #
>> >> >> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
>> >> >> ..
>> >> >> ##
>> >> >> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
>> >> >> Content-Type: api/response.Content-Length: 96..
>> >> >> #
>> >> >> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
>> >> >>
>> >> >>
>> >> >> sofia/local/sip:s000002 at 172.19.36.54:5061
>> ;fs_nat=yes;fs_path=sip%3As000002%40172.19.36.54%3A5061
>> >> >> ##
>> >> >> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
>> >> >> Event-Name: SOCKET_DATA.Content-Type: api/response.Content-Length:
>> >> >> 96..sofia/local/sip:s000002 at 172.19.36.54:5061;fs_nat=
>> >> >> yes;fs_path=sip%3As000002%40172.19.36.54%3A5061
>> >> >> #
>> >> >> T 127.0.0.1:8006 -> 127.0.0.1:43282 [AP]
>> >> >> sendmsg.call-command: execute.execute-app-name:
>> bridge.execute-app-arg:
>> >> >> {originate_timeout=10,hangup_after_bridge=false,
>> >> >>   ignore_early_media=false}user/s000002 at asdasd.com
>> >> >> #
>> >> >> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
>> >> >> Content-Type: command/reply.Reply-Text: +OK event listener enabled
>> >> >> plain..
>> >> >> #
>> >> >> T 127.0.0.1:43282 -> 127.0.0.1:8006 [AP]
>> >> >> Content-Type: command/reply.Reply-Text: +OK..
>> >> >>
>> >> >> --
>> >> >> Best regards,
>> >> >>
>> >> >> Dmitry Sytchev,
>> >> >> IT Engineer
>> >> >> !DSPAM:4d80e61b32768938086861!
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >>
>> >> >
>> >> >
>> >> >
>> >> > --
>> >> > Best regards,
>> >> >
>> >> > Dmitry Sytchev,
>> >> > IT Engineer
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > 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-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Best regards,
>> >
>> > Dmitry Sytchev,
>> > IT Engineer
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>



-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From peter.olsson at visionutveckling.se  Thu Mar 17 15:32:13 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 17 Mar 2011 13:32:13 +0100
Subject: [Freeswitch-users] Unexpected out of order command/reply '+OK
 event listener enabled plain' in perl ESL outbound
In-Reply-To: <AANLkTikon7MWc3vDQ9--SBcCB9ZzDy3gGUZHRn=OO8aF@mail.gmail.com>
References: <AANLkTikZ=pBk99wbg2+M_-vr6qK7=TC9tDZzzvE9a1+u@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B303CB87@cooper>
	<AANLkTim=Sxe8iuQncgn+JJgBzP=u3oQZdvwD2wwZCesC@mail.gmail.com>
	<AANLkTi=hwzMiU7Dxa8wD4fWmgvf=8Go4DGn+zizJAVub@mail.gmail.com>
	<AANLkTikZoSvNj5AzrMF_4sHNqk7fLFMv0vjzQo4CxSM0@mail.gmail.com>
	<AANLkTinB0CozQfgv6q0wj0_QJZ1BKfYaDD1iURxm4mcB@mail.gmail.com>
	<AANLkTinmErNdN52QV2P3M-jVCnrVTR7u--eZdEVgn=6x@mail.gmail.com>
	<AANLkTikon7MWc3vDQ9--SBcCB9ZzDy3gGUZHRn=OO8aF@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B303CDD1@cooper>

What does the perl script look like? Do you have a sample you can share with us? ?+OK event listener enabled plain? should only be sent as a reply after you have subscribed to events.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Dmitry Sytchev
Skickat: den 17 mars 2011 13:08
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Unexpected out of order command/reply '+OK event listener enabled plain' in perl ESL outbound

I have only current ESL.so in perl path.
Debug shows that message has duplicate headers:

[DEBUG] src/esl.c:1166 esl_send() SEND
sendmsg
call-command: execute
execute-app-name: bridge
execute-app-arg: {originate_timeout=10,hangup_after_bridge=false,ignore_early_media=false}user/s000001 at yota.access.obit.ru<mailto:s000001 at yota.access.obit.ru>

[DEBUG] src/esl.c:977 esl_recv_event() RECV HEADER [Content-Type] = [command/reply]
[DEBUG] src/esl.c:977 esl_recv_event() RECV HEADER [Reply-Text] = [+OK event listener enabled plain]
[DEBUG] src/esl.c:977 esl_recv_event() RECV HEADER [Content-Type] = [command/reply]
[DEBUG] src/esl.c:977 esl_recv_event() RECV HEADER [Reply-Text] = [+OK]
[DEBUG] src/esl.c:1138 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: command/reply
Reply-Text: +OK

2011/3/17 Dmitry Sytchev <kbdfck at gmail.com<mailto:kbdfck at gmail.com>>
Ok, I'll try to check library and rebuild with `make current`.
I posted it here as I thought that maybe this is my fault somewhere in script or event handling logic.
I'll open Jira issue if I can't deal with this error by full recompilation.

BTW, if `make sure` is outdated, maybe it's time to remove reference to it from post-build info that FS displays after make?

Thanks for your help

2011/3/17 Anthony Minessale <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>>
You sure you did not install old ESL.so in your perl path?

btw, its "make current" not "make sure" that is an old one.

I would also turn on the debug level in the script as well.

We have no reports of any problems like this.

Speaking of, why are you doing this here instead of JIRA?


On Wed, Mar 16, 2011 at 4:21 PM, Dmitry Sytchev <kbdfck at gmail.com<mailto:kbdfck at gmail.com>> wrote:
> I'm using ESL.so and ESL.pm for perl outbound ESL, and Net::Server as server
> engine.
> I already tried to rebuild and reinstall with `make sure` including perlmod
> in esl library dir
>
> 2011/3/16 Anthony Minessale <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>>
>>
>> Are you using your own client or libESL which takes care of this for you?
>>
>>
>> On Wed, Mar 16, 2011 at 11:55 AM, Dmitry Sytchev <kbdfck at gmail.com<mailto:kbdfck at gmail.com>> wrote:
>> > FreeSWITCH Version 1.0.head (git-2c009dd 2011-03-15 14-29-04 -0500)
>> > 2011/3/16 Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>>
>> >>
>> >> Are you on latest GIT HEAD?
>> >>
>> >>
>> >>
>> >> /Peter
>> >>
>> >>
>> >>
>> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>
>> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] F?r Dmitry
>> >> Sytchev
>> >> Skickat: den 16 mars 2011 17:29
>> >> Till: FreeSWITCH Users Help
>> >> ?mne: [Freeswitch-users] Unexpected out of order command/reply '+OK
>> >> event
>> >> listener enabled plain' in perl ESL outbound
>> >>
>> >>
>> >>
>> >> In my outbound async full ESL script after some moment there are
>> >> out-of-order message on every "execute" call, and this breaks my event
>> >> handling in some way.
>> >>
>> >>
>> >>
>> >> Content-Type: command/reply.Reply-Text: +OK event listener enabled
>> >> plain..
>> >>
>> >>
>> >> I want to execute some commands syncronously, but setEventLock seems to
>> >> be
>> >> broken by this behavior.
>> >>
>> >> As the workaround I had to write lock_for_execute_complete function,
>> >> which
>> >> waits for execute completion of applications that need to be run sync,
>> >> but
>> >> it effectively destroys async events by processing them while waiting
>> >> for
>> >> CHANNEL_EXECUTE_COMPLETE.
>> >>
>> >>
>> >>
>> >> here is a part of socket dump, note second "Content-Type:
>> >> command/reply.Reply-Text: +OK event listener enabled plain.." right
>> >> after
>> >> execute. There is no event command in script besides the very
>> >> beginning, so
>> >> why these messages appear?
>> >>
>> >> T 127.0.0.1:8006<http://127.0.0.1:8006> -> 127.0.0.1:43282<http://127.0.0.1:43282> [AP]
>> >> api uuid_setvar 06d55974-4fe8-11e0-a7c0-e3bb2141c22a
>> >> pbx_subscriber_customer_id 1
>> >> #
>> >> T 127.0.0.1:8006<http://127.0.0.1:8006> -> 127.0.0.1:43282<http://127.0.0.1:43282> [AP]
>> >> ..
>> >> ##
>> >> T 127.0.0.1:43282<http://127.0.0.1:43282> -> 127.0.0.1:8006<http://127.0.0.1:8006> [AP]
>> >> Content-Type: api/response.Content-Length: 4..
>> >> #
>> >> T 127.0.0.1:43282<http://127.0.0.1:43282> -> 127.0.0.1:8006<http://127.0.0.1:8006> [AP]
>> >> +OK.
>> >> ##
>> >> T 127.0.0.1:8006<http://127.0.0.1:8006> -> 127.0.0.1:43282<http://127.0.0.1:43282> [AP]
>> >>   api sofia_contact s000002 at asdasd.com<mailto:s000002 at asdasd.com>
>> >> #
>> >> T 127.0.0.1:8006<http://127.0.0.1:8006> -> 127.0.0.1:43282<http://127.0.0.1:43282> [AP]
>> >> ..
>> >> ##
>> >> T 127.0.0.1:43282<http://127.0.0.1:43282> -> 127.0.0.1:8006<http://127.0.0.1:8006> [AP]
>> >> Content-Type: api/response.Content-Length: 96..
>> >> #
>> >> T 127.0.0.1:43282<http://127.0.0.1:43282> -> 127.0.0.1:8006<http://127.0.0.1:8006> [AP]
>> >>
>> >>
>> >> sofia/local/sip:s000002 at 172.19.36.54:5061;fs_nat=yes;fs_path=sip%3As000002%40172.19.36.54%3A5061
>> >> ##
>> >> T 127.0.0.1:8006<http://127.0.0.1:8006> -> 127.0.0.1:43282<http://127.0.0.1:43282> [AP]
>> >> Event-Name: SOCKET_DATA.Content-Type: api/response.Content-Length:
>> >> 96..sofia/local/sip:s000002 at 172.19.36.54:5061;fs_nat=
>> >> yes;fs_path=sip%3As000002%40172.19.36.54%3A5061
>> >> #
>> >> T 127.0.0.1:8006<http://127.0.0.1:8006> -> 127.0.0.1:43282<http://127.0.0.1:43282> [AP]
>> >> sendmsg.call-command: execute.execute-app-name: bridge.execute-app-arg:
>> >> {originate_timeout=10,hangup_after_bridge=false,
>> >>   ignore_early_media=false}user/s000002 at asdasd.com<mailto:s000002 at asdasd.com>
>> >> #
>> >> T 127.0.0.1:43282<http://127.0.0.1:43282> -> 127.0.0.1:8006<http://127.0.0.1:8006> [AP]
>> >> Content-Type: command/reply.Reply-Text: +OK event listener enabled
>> >> plain..
>> >> #
>> >> T 127.0.0.1:43282<http://127.0.0.1:43282> -> 127.0.0.1:8006<http://127.0.0.1:8006> [AP]
>> >> Content-Type: command/reply.Reply-Text: +OK..
>> >>
>> >> --
>> >> Best regards,
>> >>
>> >> Dmitry Sytchev,
>> >> IT Engineer
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> > --
>> > Best regards,
>> >
>> > Dmitry Sytchev,
>> > IT Engineer
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
>> googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



--
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


--
Best regards,

Dmitry Sytchev,
IT Engineer



--
Best regards,

Dmitry Sytchev,
IT Engineer
!DSPAM:4d81fb9032761898113797!
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From steveayre at gmail.com  Thu Mar 17 15:46:33 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 17 Mar 2011 12:46:33 +0000
Subject: [Freeswitch-users] Multiple profile on same ports
In-Reply-To: <BAY156-w11045DFF35A3CF365E77188B10@phx.gbl>
References: <BAY156-w11045DFF35A3CF365E77188B10@phx.gbl>
Message-ID: <AANLkTinFLQKu=j2u43ga64yjMg=5CK1bPDCXtymMSPh9@mail.gmail.com>

You can listen on multiple profiles on the same port by using a different IP
for each profile.

Listening more than once on the same IP+port *is* impossible.

-Steve


On 17 March 2011 11:51, awais nazir <awais-nazeer at hotmail.com> wrote:

>  Hello
>
> I have intermediate level of experience in freeswitch.
>
> I want to achieve a scenario that multiple sip-profiles can share 1 sip
> bindport 5060.  And sip profile can be attached to a call in using pattern
> matching in incoming calls.
>
> --Waisee
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From erik.dekkers at wvds.nl  Thu Mar 17 15:51:30 2011
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Thu, 17 Mar 2011 13:51:30 +0100
Subject: [Freeswitch-users] Multiple profile on same ports
In-Reply-To: <BAY156-w11045DFF35A3CF365E77188B10@phx.gbl>
References: <BAY156-w11045DFF35A3CF365E77188B10@phx.gbl>
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE078F616BF2@FS1.wvds.local>

Hi,

This is not possible. Like it isn't possible to run 2 different http servers on the same ip+port.



Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens awais nazir
Verzonden: donderdag 17 maart 2011 12:52
Aan: freeswitch-users at lists.freeswitch.org
Onderwerp: [Freeswitch-users] Multiple profile on same ports

Hello

I have intermediate level of experience in freeswitch.

I want to achieve a scenario that multiple sip-profiles can share 1 sip bindport 5060.  And sip profile can be attached to a call in using pattern matching in incoming calls.

--Waisee

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From awais-nazeer at hotmail.com  Thu Mar 17 15:54:13 2011
From: awais-nazeer at hotmail.com (awais nazir)
Date: Thu, 17 Mar 2011 17:54:13 +0500
Subject: [Freeswitch-users] Multiple profile on same ports
In-Reply-To: <BAY156-w11045DFF35A3CF365E77188B10@phx.gbl>
References: <BAY156-w11045DFF35A3CF365E77188B10@phx.gbl>
Message-ID: <bay156-w3C8ECF70D4014AF5BD2AD88B10@phx.gbl>


Hello

I want to achieve a wholesale app in freeswitch where we are considering each endpoint as sip profile. 

So that each profile can have its own in an out codec preferences, gateways , transcoding , by-pass setting etc.

Any guidance of direction will be appreciated.

--waisee
_______________________________________________________________________________________________________
Basically, I don't think that possible. But uh, why would you want to?If you want users authing to different contexts, then that's just a user variable. What are you trying to achieve here?


-Avi

From: awais-nazeer at hotmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Multiple profile on same ports
Date: Thu, 17 Mar 2011 16:51:50 +0500








Hello

I have intermediate level of experience in freeswitch.

I want to achieve a scenario that multiple sip-profiles can share 1 sip bindport 5060.  And sip profile can be attached to a call in using pattern matching in incoming calls.

--Waisee


 		 	   		  
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From avi at avimarcus.net  Thu Mar 17 15:58:19 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 17 Mar 2011 14:58:19 +0200
Subject: [Freeswitch-users] Multiple profile on same ports
In-Reply-To: <bay156-w3C8ECF70D4014AF5BD2AD88B10@phx.gbl>
References: <BAY156-w11045DFF35A3CF365E77188B10@phx.gbl>
	<bay156-w3C8ECF70D4014AF5BD2AD88B10@phx.gbl>
Message-ID: <AANLkTimt3=c3RNmM6H1sn3hZeACb5NOM59Ah=XzR+TPQ@mail.gmail.com>

All of those can be overridden with channel variables - you don't
*need* a different profile, the profile only determines the default.
-Avi

On Thu, Mar 17, 2011 at 2:54 PM, awais nazir <awais-nazeer at hotmail.com> wrote:
> Hello
>
> I want to achieve a wholesale app in freeswitch where we are considering
> each endpoint as sip profile.
>
> So that each profile can have its own in an out codec preferences, gateways
> , transcoding , by-pass setting etc.
>
> Any guidance of direction will be appreciated.
>
> --waisee
> _______________________________________________________________________________________________________
> Basically, I don't think that possible. But uh, why would you want to?
> If you want users authing to different contexts, then that's just a user
> variable. What are you trying to achieve here?
> -Avi
>
> ________________________________
> From: awais-nazeer at hotmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Multiple profile on same ports
> Date: Thu, 17 Mar 2011 16:51:50 +0500
>
> Hello
>
> I have intermediate level of experience in freeswitch.
>
> I want to achieve a scenario that multiple sip-profiles can share 1 sip
> bindport 5060.? And sip profile can be attached to a call in using pattern
> matching in incoming calls.
>
> --Waisee
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From julf at julf.com  Thu Mar 17 16:13:26 2011
From: julf at julf.com (Johan Helsingius)
Date: Thu, 17 Mar 2011 14:13:26 +0100
Subject: [Freeswitch-users] Multiple profile on same ports
In-Reply-To: <AANLkTimt3=c3RNmM6H1sn3hZeACb5NOM59Ah=XzR+TPQ@mail.gmail.com>
References: <BAY156-w11045DFF35A3CF365E77188B10@phx.gbl>	<bay156-w3C8ECF70D4014AF5BD2AD88B10@phx.gbl>
	<AANLkTimt3=c3RNmM6H1sn3hZeACb5NOM59Ah=XzR+TPQ@mail.gmail.com>
Message-ID: <4D8208F6.2020707@julf.com>

On 03/17/2011 01:58 PM, Avi Marcus wrote:
> All of those can be overridden with channel variables - you don't
> *need* a different profile, the profile only determines the default.

Right, so even different outgoing providers could be
handled by one profile, with different gateways for
different providers, as long as you are OK with
handling the incoming connections in one profile?

	Julf


From Nabble at slickdeals.endjunk.com  Thu Mar 17 16:19:19 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 17 Mar 2011 06:19:19 -0700 (PDT)
Subject: [Freeswitch-users] very long delays in conference audio
In-Reply-To: <15324.1300359096@ccs.covici.com>
References: <15324.1300359096@ccs.covici.com>
Message-ID: <1300367959665-6180837.post@n2.nabble.com>

I am not sure if this will help or not. However, when I ran into a 2+ seconds
delay on a Google Voice call through mod_dingaling, Anthony Minessale
suggested that I set the use-rtp-name to none (< param name="rtp-timer-name"
value="none"/ >) as shown 
http://freeswitch-users.2379917.n2.nabble.com/FIXED-Latency-on-Google-Voice-GTalk-tp5741999p5742493.html
here .

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/very-long-delays-in-conference-audio-tp6180420p6180837.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From freeswitch at peely.com  Thu Mar 17 17:16:33 2011
From: freeswitch at peely.com (peely)
Date: Thu, 17 Mar 2011 07:16:33 -0700 (PDT)
Subject: [Freeswitch-users] Multiple profile on same ports
In-Reply-To: <AANLkTimt3=c3RNmM6H1sn3hZeACb5NOM59Ah=XzR+TPQ@mail.gmail.com>
References: <BAY156-w11045DFF35A3CF365E77188B10@phx.gbl>
	<bay156-w3C8ECF70D4014AF5BD2AD88B10@phx.gbl>
	<AANLkTimt3=c3RNmM6H1sn3hZeACb5NOM59Ah=XzR+TPQ@mail.gmail.com>
Message-ID: <1300371393809-6181047.post@n2.nabble.com>

Including transcoding?

I've tried this and I haven't managed to get disable-transcoding to do
anything in the dialplan using set_profile_var.



Avi Marcus-2 wrote:
> 
> All of those can be overridden with channel variables - you don't
> *need* a different profile, the profile only determines the default.
> -Avi
> 
> 


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Multiple-profile-on-same-ports-tp6180600p6181047.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From avi at avimarcus.net  Thu Mar 17 17:20:56 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 17 Mar 2011 16:20:56 +0200
Subject: [Freeswitch-users] Multiple profile on same ports
In-Reply-To: <1300371393809-6181047.post@n2.nabble.com>
References: <BAY156-w11045DFF35A3CF365E77188B10@phx.gbl>
	<bay156-w3C8ECF70D4014AF5BD2AD88B10@phx.gbl>
	<AANLkTimt3=c3RNmM6H1sn3hZeACb5NOM59Ah=XzR+TPQ@mail.gmail.com>
	<1300371393809-6181047.post@n2.nabble.com>
Message-ID: <AANLkTimq9Yeqf7j+15SBkJQrnCNOiaFWHgp3Qm_V7+2e@mail.gmail.com>

absolute_codec_string overrides disable-transcoding setting anyway, as per
the wiki...
http://wiki.freeswitch.org/wiki/Codec_negotiation#Early_Negotiation_.28default_behavior.29

-Avi

On Thu, Mar 17, 2011 at 4:16 PM, peely <freeswitch at peely.com> wrote:

> Including transcoding?
>
> I've tried this and I haven't managed to get disable-transcoding to do
> anything in the dialplan using set_profile_var.
>
>
>
> Avi Marcus-2 wrote:
> >
> > All of those can be overridden with channel variables - you don't
> > *need* a different profile, the profile only determines the default.
> > -Avi
> >
> >
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Multiple-profile-on-same-ports-tp6180600p6181047.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From covici at ccs.covici.com  Thu Mar 17 18:31:01 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 17 Mar 2011 11:31:01 -0400
Subject: [Freeswitch-users] very long delays in conference audio
In-Reply-To: <AANLkTikNveCvuZOXoKxKTWC11dk5EfnhJB-446iN+C99@mail.gmail.com>
References: <15324.1300359096@ccs.covici.com>
	<AANLkTikNveCvuZOXoKxKTWC11dk5EfnhJB-446iN+C99@mail.gmail.com>
Message-ID: <18743.1300375861@ccs.covici.com>

I am not sure, I will have to check the ip address for rtp and see if
this is so.  Anyone else seeing this kind of thing?

Steven Ayre <steveayre at gmail.com> wrote:

> Do you know what the latency (test RTT with ping) and jitter is like? It
> could be that over the internet it takes a while for the data to arrive, or
> if you have had high latency/jitter a jitterbuffer could be introducing a 1
> second delay to compensate.
> 
> -Steve
> 
> 
> On 17 March 2011 10:51, <covici at ccs.covici.com> wrote:
> 
> > Hi.  I have been experiencing very long delays in conference audio --
> > usually over the internet.  For instance, it can take 1 second or more
> > between the time you hit 0 and when it says you are muted or unmuted.  I
> > even had this problem during the freeswitch conference.  What can be
> > done about this -- any way to tell why this is happening, because it
> > makes conversation on the calls very difficult.  I had one last night
> > where they gave up because of this problem.
> >
> > Any ideas would be appreciated.
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >         John Covici
> >         covici at ccs.covici.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From peter.olsson at visionutveckling.se  Thu Mar 17 18:44:48 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 17 Mar 2011 16:44:48 +0100
Subject: [Freeswitch-users] very long delays in conference audio
In-Reply-To: <18743.1300375861@ccs.covici.com>
References: <15324.1300359096@ccs.covici.com>
	<AANLkTikNveCvuZOXoKxKTWC11dk5EfnhJB-446iN+C99@mail.gmail.com>
	<18743.1300375861@ccs.covici.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B303CF45@cooper>

Nope, working fine over here :)

Are you on real hardware, not virtual?

/Peter

-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r covici at ccs.covici.com
Skickat: den 17 mars 2011 16:31
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] very long delays in conference audio

I am not sure, I will have to check the ip address for rtp and see if
this is so.  Anyone else seeing this kind of thing?

Steven Ayre <steveayre at gmail.com> wrote:

> Do you know what the latency (test RTT with ping) and jitter is like? It
> could be that over the internet it takes a while for the data to arrive, or
> if you have had high latency/jitter a jitterbuffer could be introducing a 1
> second delay to compensate.
> 
> -Steve
> 
> 
> On 17 March 2011 10:51, <covici at ccs.covici.com> wrote:
> 
> > Hi.  I have been experiencing very long delays in conference audio --
> > usually over the internet.  For instance, it can take 1 second or more
> > between the time you hit 0 and when it says you are muted or unmuted.  I
> > even had this problem during the freeswitch conference.  What can be
> > done about this -- any way to tell why this is happening, because it
> > makes conversation on the calls very difficult.  I had one last night
> > where they gave up because of this problem.
> >
> > Any ideas would be appreciated.
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >         John Covici
> >         covici at ccs.covici.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com

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

!DSPAM:4d822a8332761303411046!



From infos at madovsky.org  Thu Mar 17 20:04:10 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 17 Mar 2011 13:04:10 -0400
Subject: [Freeswitch-users] very long delays in conference audio
References: <15324.1300359096@ccs.covici.com><AANLkTikNveCvuZOXoKxKTWC11dk5EfnhJB-446iN+C99@mail.gmail.com>
	<18743.1300375861@ccs.covici.com>
Message-ID: <FFF1678367C04E87BD0DAD015559E33E@e1705>

yes I noticed that on very good internet line (data center to data center)
but more latency with keys

----- Original Message ----- 
From: <covici at ccs.covici.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, March 17, 2011 11:31 AM
Subject: Re: [Freeswitch-users] very long delays in conference audio


>I am not sure, I will have to check the ip address for rtp and see if
> this is so.  Anyone else seeing this kind of thing?
>
> Steven Ayre <steveayre at gmail.com> wrote:
>
>> Do you know what the latency (test RTT with ping) and jitter is like? It
>> could be that over the internet it takes a while for the data to arrive, 
>> or
>> if you have had high latency/jitter a jitterbuffer could be introducing a 
>> 1
>> second delay to compensate.
>>
>> -Steve
>>
>>
>> On 17 March 2011 10:51, <covici at ccs.covici.com> wrote:
>>
>> > Hi.  I have been experiencing very long delays in conference audio --
>> > usually over the internet.  For instance, it can take 1 second or more
>> > between the time you hit 0 and when it says you are muted or unmuted. 
>> > I
>> > even had this problem during the freeswitch conference.  What can be
>> > done about this -- any way to tell why this is happening, because it
>> > makes conversation on the calls very difficult.  I had one last night
>> > where they gave up because of this problem.
>> >
>> > Any ideas would be appreciated.
>> >
>> > --
>> > Your life is like a penny.  You're going to lose it.  The question is:
>> > How do
>> > you spend it?
>> >
>> >         John Covici
>> >         covici at ccs.covici.com
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> ----------------------------------------------------
>> Alternatives:
>>
>> ----------------------------------------------------
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From frank at rosengart.de  Fri Mar 18 01:05:52 2011
From: frank at rosengart.de (Frank Rosengart)
Date: Thu, 17 Mar 2011 23:05:52 +0100
Subject: [Freeswitch-users] segfault with stun-enabled=false
Message-ID: <4D8285C0.9020307@rosengart.de>

Hi,

I'm trying to run local tests with FS and need to switch off all stun
stuff. When stun-enabled=false in my sip profile, FS dies when starting
mod_sofia.

I think it's more user friendly to exit with an error message saying
what is wrong. FS git/trunk on Linux/amd64.

How can I force FS not to use stun at all?

Thanks,

Frank


From anthony.minessale at gmail.com  Fri Mar 18 01:25:55 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 17 Mar 2011 17:25:55 -0500
Subject: [Freeswitch-users] segfault with stun-enabled=false
In-Reply-To: <4D8285C0.9020307@rosengart.de>
References: <4D8285C0.9020307@rosengart.de>
Message-ID: <AANLkTi=bm2Cqq9JgnyS_zeF6k0_vkixXR+EQ5XA08ic-@mail.gmail.com>

you should report crashes to jira http://jira.freeswitch.org

reproduce the problem on freshly built FS git HEAD and attach a backtrace.


On Thu, Mar 17, 2011 at 5:05 PM, Frank Rosengart <frank at rosengart.de> wrote:
> Hi,
>
> I'm trying to run local tests with FS and need to switch off all stun
> stuff. When stun-enabled=false in my sip profile, FS dies when starting
> mod_sofia.
>
> I think it's more user friendly to exit with an error message saying
> what is wrong. FS git/trunk on Linux/amd64.
>
> How can I force FS not to use stun at all?
>
> Thanks,
>
> Frank
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 covici at ccs.covici.com  Fri Mar 18 01:45:05 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 17 Mar 2011 18:45:05 -0400
Subject: [Freeswitch-users] very long delays in conference audio
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B303CF45@cooper>
References: <15324.1300359096@ccs.covici.com>
	<AANLkTikNveCvuZOXoKxKTWC11dk5EfnhJB-446iN+C99@mail.gmail.com>
	<18743.1300375861@ccs.covici.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B303CF45@cooper>
Message-ID: <25584.1300401905@ccs.covici.com>

Yep, real hardware.  I wonder if its my itsp, but since it happened on
the fs conference, I doubt it.

Peter Olsson <peter.olsson at visionutveckling.se> wrote:

> Nope, working fine over here :)
> 
> Are you on real hardware, not virtual?
> 
> /Peter
> 
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r covici at ccs.covici.com
> Skickat: den 17 mars 2011 16:31
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] very long delays in conference audio
> 
> I am not sure, I will have to check the ip address for rtp and see if
> this is so.  Anyone else seeing this kind of thing?
> 
> Steven Ayre <steveayre at gmail.com> wrote:
> 
> > Do you know what the latency (test RTT with ping) and jitter is like? It
> > could be that over the internet it takes a while for the data to arrive, or
> > if you have had high latency/jitter a jitterbuffer could be introducing a 1
> > second delay to compensate.
> > 
> > -Steve
> > 
> > 
> > On 17 March 2011 10:51, <covici at ccs.covici.com> wrote:
> > 
> > > Hi.  I have been experiencing very long delays in conference audio --
> > > usually over the internet.  For instance, it can take 1 second or more
> > > between the time you hit 0 and when it says you are muted or unmuted.  I
> > > even had this problem during the freeswitch conference.  What can be
> > > done about this -- any way to tell why this is happening, because it
> > > makes conversation on the calls very difficult.  I had one last night
> > > where they gave up because of this problem.
> > >
> > > Any ideas would be appreciated.
> > >
> > > --
> > > Your life is like a penny.  You're going to lose it.  The question is:
> > > How do
> > > you spend it?
> > >
> > >         John Covici
> > >         covici at ccs.covici.com
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > 
> > ----------------------------------------------------
> > Alternatives:
> > 
> > ----------------------------------------------------
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> !DSPAM:4d822a8332761303411046!
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From covici at ccs.covici.com  Fri Mar 18 01:56:30 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 17 Mar 2011 18:56:30 -0400
Subject: [Freeswitch-users] very long delays in conference audio
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B303CF45@cooper>
References: <15324.1300359096@ccs.covici.com>
	<AANLkTikNveCvuZOXoKxKTWC11dk5EfnhJB-446iN+C99@mail.gmail.com>
	<18743.1300375861@ccs.covici.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B303CF45@cooper>
Message-ID: <25832.1300402590@ccs.covici.com>

Also, I notice -- I was the only one in the conference and it was
playing music on hold, and after a few minutes --maybe 3 minutes -- the
time delay between when I pressed 0 and when I heard the recording went
from 1 second to 3 seconds.

Peter Olsson <peter.olsson at visionutveckling.se> wrote:

> Nope, working fine over here :)
> 
> Are you on real hardware, not virtual?
> 
> /Peter
> 
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r covici at ccs.covici.com
> Skickat: den 17 mars 2011 16:31
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] very long delays in conference audio
> 
> I am not sure, I will have to check the ip address for rtp and see if
> this is so.  Anyone else seeing this kind of thing?
> 
> Steven Ayre <steveayre at gmail.com> wrote:
> 
> > Do you know what the latency (test RTT with ping) and jitter is like? It
> > could be that over the internet it takes a while for the data to arrive, or
> > if you have had high latency/jitter a jitterbuffer could be introducing a 1
> > second delay to compensate.
> > 
> > -Steve
> > 
> > 
> > On 17 March 2011 10:51, <covici at ccs.covici.com> wrote:
> > 
> > > Hi.  I have been experiencing very long delays in conference audio --
> > > usually over the internet.  For instance, it can take 1 second or more
> > > between the time you hit 0 and when it says you are muted or unmuted.  I
> > > even had this problem during the freeswitch conference.  What can be
> > > done about this -- any way to tell why this is happening, because it
> > > makes conversation on the calls very difficult.  I had one last night
> > > where they gave up because of this problem.
> > >
> > > Any ideas would be appreciated.
> > >
> > > --
> > > Your life is like a penny.  You're going to lose it.  The question is:
> > > How do
> > > you spend it?
> > >
> > >         John Covici
> > >         covici at ccs.covici.com
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > 
> > ----------------------------------------------------
> > Alternatives:
> > 
> > ----------------------------------------------------
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> !DSPAM:4d822a8332761303411046!
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From george.niculae79 at gmail.com  Fri Mar 18 02:00:32 2011
From: george.niculae79 at gmail.com (George Niculae)
Date: Fri, 18 Mar 2011 01:00:32 +0200
Subject: [Freeswitch-users] very long delays in conference audio
In-Reply-To: <25584.1300401905@ccs.covici.com>
References: <15324.1300359096@ccs.covici.com>
	<AANLkTikNveCvuZOXoKxKTWC11dk5EfnhJB-446iN+C99@mail.gmail.com>
	<18743.1300375861@ccs.covici.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B303CF45@cooper>
	<25584.1300401905@ccs.covici.com>
Message-ID: <AANLkTikTv6wEN8FG2DK=YQ9UmC39DPq3nBF1fO9FR4qC@mail.gmail.com>

2011/3/18 <covici at ccs.covici.com>

> Yep, real hardware.  I wonder if its my itsp, but since it happened on
> the fs conference, I doubt it.
>
>
I experienced audio delay in fs conferences some time ago when bridging conf
and using mod_loopback. Is this also your case?

George
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From covici at ccs.covici.com  Fri Mar 18 02:49:45 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 17 Mar 2011 19:49:45 -0400
Subject: [Freeswitch-users] very long delays in conference audio
In-Reply-To: <AANLkTikTv6wEN8FG2DK=YQ9UmC39DPq3nBF1fO9FR4qC@mail.gmail.com>
References: <15324.1300359096@ccs.covici.com>
	<AANLkTikNveCvuZOXoKxKTWC11dk5EfnhJB-446iN+C99@mail.gmail.com>
	<18743.1300375861@ccs.covici.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B303CF45@cooper>
	<25584.1300401905@ccs.covici.com>
	<AANLkTikTv6wEN8FG2DK=YQ9UmC39DPq3nBF1fO9FR4qC@mail.gmail.com>
Message-ID: <26926.1300405785@ccs.covici.com>

no loopback used here for that.

George Niculae <george.niculae79 at gmail.com> wrote:

> 2011/3/18 <covici at ccs.covici.com>
> 
> > Yep, real hardware.  I wonder if its my itsp, but since it happened on
> > the fs conference, I doubt it.
> >
> >
> I experienced audio delay in fs conferences some time ago when bridging conf
> and using mod_loopback. Is this also your case?
> 
> George
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From elijah at crankenstein.com  Fri Mar 18 02:53:53 2011
From: elijah at crankenstein.com (elijah)
Date: Thu, 17 Mar 2011 16:53:53 -0700
Subject: [Freeswitch-users] some outbound calls failing
Message-ID: <AANLkTinmvz055Qb7Ep2PMZ-sS6GJmTVVpC2hdapyJXb0@mail.gmail.com>

I have put together a dialer (of sorts) that is failing a fraction of the
time. First, I instantiate a call leg from FreeSwitch to an agent (an
employee in my facility with a softclient registered to FreeSwitch). Upon
answering this call I have FreeSwitch bridge this leg to an outbound leg
destined to a customer (via our carrier). The agent listens as this external
number is dialed then is prepared to speak with that person. So all is well
except a fraction of these calls fail at the step where FreeSwitch calls the
agent. There seems to be a 'hangup' event received from the client at the
moment this leg is answered. This happens maybe 1/3 of the time.

here's my steps

1. call a Lua script via ESL

2. Lua script generates a call leg to agent:
                local session = freeswitch.Session("user/" .. agentExt .. "@
stuff.com")
                session:sleep(1000)
                if session:ready() then
                        session:execute("pre_answer")
                        session:execute("ring_ready")
                        session:execute("sleep", "1000")
                        session:setVariable("outbound_callee_number",
outboundCalleeNumber)
                        session:setVariable("outbound_caller_number",
outboundCallerNumber)
                        session:setVariable("outbound_caller_name",
outboundCallerName)
                        session:transfer("sales_outbound", "XML", "telifi")

******** fails here ~30% of the time - phone rings on the agent's desk,
she answers and the call immediately ends ********

3. this leg is transferred to an XML dialplan that handles bridging to a
customer's phone number:
    <extension name="sales_outbound">
      <condition field="destination_number" expression="^sales_outbound$">
        <action application="log" data="INFO connected to agent, now dialing
out..."/>
        <action application="sleep" data="1000"/>
        <action application="playback"
data="ivr/8000/ivr-call_being_transferred.wav"/>-->
        <action application="set" data="continue_on_fail=true"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="ringback=${us-ring}"/>
        <action application="set" data="call_timeout=30"/>
        <action application="set"
data="effective_caller_id_number=${outbound_caller_number}"/>
        <action application="set"
data="effective_caller_id_name=${outbound_caller_name}"/>
        <action application="sleep" data="1000"/>
        <action application="bridge"
data="sofia/gateway/mycarrier/${outbound_callee_number}"/>
      </condition>
    </extension>

I appreciate any insight or suggestions you could provide.
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From anthony.minessale at gmail.com  Fri Mar 18 04:13:57 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 17 Mar 2011 20:13:57 -0500
Subject: [Freeswitch-users] very long delays in conference audio
In-Reply-To: <26926.1300405785@ccs.covici.com>
References: <15324.1300359096@ccs.covici.com>
	<AANLkTikNveCvuZOXoKxKTWC11dk5EfnhJB-446iN+C99@mail.gmail.com>
	<18743.1300375861@ccs.covici.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B303CF45@cooper>
	<25584.1300401905@ccs.covici.com>
	<AANLkTikTv6wEN8FG2DK=YQ9UmC39DPq3nBF1fO9FR4qC@mail.gmail.com>
	<26926.1300405785@ccs.covici.com>
Message-ID: <AANLkTim0P8Jkv9Pjpc46vsW=u2jwmLWG6tQZr3zCCjP0@mail.gmail.com>

What revision of GIT are you on?  Is this s new development for you?


On Thu, Mar 17, 2011 at 6:49 PM,  <covici at ccs.covici.com> wrote:
> no loopback used here for that.
>
> George Niculae <george.niculae79 at gmail.com> wrote:
>
>> 2011/3/18 <covici at ccs.covici.com>
>>
>> > Yep, real hardware. ?I wonder if its my itsp, but since it happened on
>> > the fs conference, I doubt it.
>> >
>> >
>> I experienced audio delay in fs conferences some time ago when bridging conf
>> and using mod_loopback. Is this also your case?
>>
>> George
>>
>> ----------------------------------------------------
>> Alternatives:
>>
>> ----------------------------------------------------
>
> --
> Your life is like a penny. ?You're going to lose it. ?The question is:
> How do
> you spend it?
>
> ? ? ? ? John Covici
> ? ? ? ? covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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  Fri Mar 18 04:22:56 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 17 Mar 2011 20:22:56 -0500
Subject: [Freeswitch-users] some outbound calls failing
In-Reply-To: <AANLkTinmvz055Qb7Ep2PMZ-sS6GJmTVVpC2hdapyJXb0@mail.gmail.com>
References: <AANLkTinmvz055Qb7Ep2PMZ-sS6GJmTVVpC2hdapyJXb0@mail.gmail.com>
Message-ID: <AANLkTi=6VGzHjt1bSDdy8gWuqUSoZ47A5pR=SGH6Zgu-@mail.gmail.com>

can you see if you have that same problem with the latest build from git
you can update an existing checkout with "make current" assuming you
have git tools etc

if you still can get the problem capture the debug logs of it happening

console loglevel debug


On Thu, Mar 17, 2011 at 6:53 PM, elijah <elijah at crankenstein.com> wrote:
> I have put together a dialer (of sorts) that is failing a fraction of the
> time. First, I instantiate a call leg from FreeSwitch to an agent (an
> employee in my facility with a softclient registered to FreeSwitch). Upon
> answering this call I have FreeSwitch bridge this leg to an outbound leg
> destined to a customer (via our carrier). The agent listens as this external
> number is dialed then is prepared to speak with that person. So all is well
> except a fraction of these calls fail at the step where FreeSwitch calls the
> agent. There seems to be a 'hangup' event received from the client at the
> moment this leg is answered. This happens maybe 1/3 of the time.
> here's my steps
> 1. call a Lua script via ESL
> 2. Lua script generates a call leg to agent:
> ?? ? ? ? ? ? ? ?local session = freeswitch.Session("user/" .. agentExt ..
> "@stuff.com")
> ?? ? ? ? ? ? ? ?session:sleep(1000)
> ?? ? ? ? ? ? ? ?if session:ready() then
> ?? ? ? ? ? ? ? ? ? ? ? ?session:execute("pre_answer")
> ?? ? ? ? ? ? ? ? ? ? ? ?session:execute("ring_ready")
> ?? ? ? ? ? ? ? ? ? ? ? ?session:execute("sleep", "1000")
> ?? ? ? ? ? ? ? ? ? ? ? ?session:setVariable("outbound_callee_number",
> outboundCalleeNumber)
> ?? ? ? ? ? ? ? ? ? ? ? ?session:setVariable("outbound_caller_number",
> outboundCallerNumber)
> ?? ? ? ? ? ? ? ? ? ? ? ?session:setVariable("outbound_caller_name",
> outboundCallerName)
> ?? ? ? ? ? ? ? ? ? ? ? ?session:transfer("sales_outbound", "XML", "telifi")
> ******** fails here ~30% of the time - phone rings on the agent's desk,
> she?answers?and the call immediately ends ********
> 3. this leg is transferred to an XML dialplan that handles bridging to a
> customer's phone number:
> ?? ?<extension name="sales_outbound">
> ?? ? ?<condition field="destination_number" expression="^sales_outbound$">
> ?? ? ? ?<action application="log" data="INFO connected to agent, now dialing
> out..."/>
> ?? ? ? ?<action application="sleep" data="1000"/>
> ?? ? ? ?<action application="playback"
> data="ivr/8000/ivr-call_being_transferred.wav"/>-->
> ?? ? ? ?<action application="set" data="continue_on_fail=true"/>
> ?? ? ? ?<action application="set" data="hangup_after_bridge=true"/>
> ?? ? ? ?<action application="set" data="ringback=${us-ring}"/>
> ?? ? ? ?<action application="set" data="call_timeout=30"/>
> ?? ? ? ?<action application="set"
> data="effective_caller_id_number=${outbound_caller_number}"/>
> ?? ? ? ?<action application="set"
> data="effective_caller_id_name=${outbound_caller_name}"/>
> ?? ? ? ?<action application="sleep" data="1000"/>
> ?? ? ? ?<action application="bridge"
> data="sofia/gateway/mycarrier/${outbound_callee_number}"/>
> ?? ? ?</condition>
> ?? ?</extension>
> I appreciate any insight or suggestions you could provide.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 covici at ccs.covici.com  Fri Mar 18 04:32:17 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 17 Mar 2011 21:32:17 -0400
Subject: [Freeswitch-users] very long delays in conference audio
In-Reply-To: <AANLkTim0P8Jkv9Pjpc46vsW=u2jwmLWG6tQZr3zCCjP0@mail.gmail.com>
References: <15324.1300359096@ccs.covici.com>
	<AANLkTikNveCvuZOXoKxKTWC11dk5EfnhJB-446iN+C99@mail.gmail.com>
	<18743.1300375861@ccs.covici.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B303CF45@cooper>
	<25584.1300401905@ccs.covici.com>
	<AANLkTikTv6wEN8FG2DK=YQ9UmC39DPq3nBF1fO9FR4qC@mail.gmail.com>
	<26926.1300405785@ccs.covici.com>
	<AANLkTim0P8Jkv9Pjpc46vsW=u2jwmLWG6tQZr3zCCjP0@mail.gmail.com>
Message-ID: <29592.1300411937@ccs.covici.com>

It seems to be somewhat new, I am on FreeSWITCH Version 1.0.head
(git-5640464 2011-03-08 12-40-58 -0600) -- I can certainly try to update
and see if that helps -- I don't like to update weekly, unless its
necessary.  I think I was on something from 2/11 before this one.


Anthony Minessale <anthony.minessale at gmail.com> wrote:

> What revision of GIT are you on?  Is this s new development for you?
> 
> 
> On Thu, Mar 17, 2011 at 6:49 PM,  <covici at ccs.covici.com> wrote:
> > no loopback used here for that.
> >
> > George Niculae <george.niculae79 at gmail.com> wrote:
> >
> >> 2011/3/18 <covici at ccs.covici.com>
> >>
> >> > Yep, real hardware. ?I wonder if its my itsp, but since it happened on
> >> > the fs conference, I doubt it.
> >> >
> >> >
> >> I experienced audio delay in fs conferences some time ago when bridging conf
> >> and using mod_loopback. Is this also your case?
> >>
> >> George
> >>
> >> ----------------------------------------------------
> >> Alternatives:
> >>
> >> ----------------------------------------------------
> >
> > --
> > Your life is like a penny. ?You're going to lose it. ?The question is:
> > How do
> > you spend it?
> >
> > ? ? ? ? John Covici
> > ? ? ? ? covici at ccs.covici.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From anthony.minessale at gmail.com  Fri Mar 18 04:37:11 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 17 Mar 2011 20:37:11 -0500
Subject: [Freeswitch-users] very long delays in conference audio
In-Reply-To: <29592.1300411937@ccs.covici.com>
References: <15324.1300359096@ccs.covici.com>
	<AANLkTikNveCvuZOXoKxKTWC11dk5EfnhJB-446iN+C99@mail.gmail.com>
	<18743.1300375861@ccs.covici.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B303CF45@cooper>
	<25584.1300401905@ccs.covici.com>
	<AANLkTikTv6wEN8FG2DK=YQ9UmC39DPq3nBF1fO9FR4qC@mail.gmail.com>
	<26926.1300405785@ccs.covici.com>
	<AANLkTim0P8Jkv9Pjpc46vsW=u2jwmLWG6tQZr3zCCjP0@mail.gmail.com>
	<29592.1300411937@ccs.covici.com>
Message-ID: <AANLkTinSN8rJ2ujm8B8HO5RjMDpyF41cB9mZcg_DjeNt@mail.gmail.com>

yes there was a regression on march 2nd that was fixed about the 9th
that could explain your symptoms but I would update and find out for
sure.


On Thu, Mar 17, 2011 at 8:32 PM,  <covici at ccs.covici.com> wrote:
> It seems to be somewhat new, I am on FreeSWITCH Version 1.0.head
> (git-5640464 2011-03-08 12-40-58 -0600) -- I can certainly try to update
> and see if that helps -- I don't like to update weekly, unless its
> necessary. ?I think I was on something from 2/11 before this one.
>
>
> Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
>> What revision of GIT are you on? ?Is this s new development for you?
>>
>>
>> On Thu, Mar 17, 2011 at 6:49 PM, ?<covici at ccs.covici.com> wrote:
>> > no loopback used here for that.
>> >
>> > George Niculae <george.niculae79 at gmail.com> wrote:
>> >
>> >> 2011/3/18 <covici at ccs.covici.com>
>> >>
>> >> > Yep, real hardware. ?I wonder if its my itsp, but since it happened on
>> >> > the fs conference, I doubt it.
>> >> >
>> >> >
>> >> I experienced audio delay in fs conferences some time ago when bridging conf
>> >> and using mod_loopback. Is this also your case?
>> >>
>> >> George
>> >>
>> >> ----------------------------------------------------
>> >> Alternatives:
>> >>
>> >> ----------------------------------------------------
>> >
>> > --
>> > Your life is like a penny. ?You're going to lose it. ?The question is:
>> > How do
>> > you spend it?
>> >
>> > ? ? ? ? John Covici
>> > ? ? ? ? covici at ccs.covici.com
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> --
> Your life is like a penny. ?You're going to lose it. ?The question is:
> How do
> you spend it?
>
> ? ? ? ? John Covici
> ? ? ? ? covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 avi at avimarcus.net  Fri Mar 18 06:23:26 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 18 Mar 2011 05:23:26 +0200
Subject: [Freeswitch-users] DNS SRV Failover - How about incoming?
Message-ID: <AANLkTim06KmPqqMNYgXnyCVc7cVwjCc60=D=ZF686neS@mail.gmail.com>

I think I understand how to use DNS SRV for failover for outgoing calls, if
the phone supports it, it will try the next lower priority entry upon.. no
response.
So that works fine for outgoing - it will auth for the call.

What about incoming to the phone, via FS? If devices are behind NAT then is
there any way to successfully get the calls back to them? Would it make
sense to have the registration mirrored across the servers that may be
failed over to?

Is there more information about this on the wiki somewhere?

Thanks,
Avi Marcus
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From djbinter at gmail.com  Fri Mar 18 07:15:10 2011
From: djbinter at gmail.com (DJB International)
Date: Thu, 17 Mar 2011 21:15:10 -0700
Subject: [Freeswitch-users] mod_conference question
Message-ID: <AANLkTimtL0A3SzM3ahPgNHQTfmvoxLKsfsbYt2U999LF@mail.gmail.com>

I wonder whether there is a way to get Conference-Unique-ID to show up when
the channel is hang up in CS_REPORTING state.

I am trying to find a way to separate and group the cdrs for each conference
session.


Thank you,
-djbinter
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From covici at ccs.covici.com  Fri Mar 18 09:01:03 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 18 Mar 2011 02:01:03 -0400
Subject: [Freeswitch-users] very long delays in conference audio
In-Reply-To: <AANLkTinSN8rJ2ujm8B8HO5RjMDpyF41cB9mZcg_DjeNt@mail.gmail.com>
References: <15324.1300359096@ccs.covici.com>
	<AANLkTikNveCvuZOXoKxKTWC11dk5EfnhJB-446iN+C99@mail.gmail.com>
	<18743.1300375861@ccs.covici.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B303CF45@cooper>
	<25584.1300401905@ccs.covici.com>
	<AANLkTikTv6wEN8FG2DK=YQ9UmC39DPq3nBF1fO9FR4qC@mail.gmail.com>
	<26926.1300405785@ccs.covici.com>
	<AANLkTim0P8Jkv9Pjpc46vsW=u2jwmLWG6tQZr3zCCjP0@mail.gmail.com>
	<29592.1300411937@ccs.covici.com>
	<AANLkTinSN8rJ2ujm8B8HO5RjMDpyF41cB9mZcg_DjeNt@mail.gmail.com>
Message-ID: <1935.1300428063@ccs.covici.com>

I updated to latest git and still the delay.  
Anthony Minessale <anthony.minessale at gmail.com> wrote:

> yes there was a regression on march 2nd that was fixed about the 9th
> that could explain your symptoms but I would update and find out for
> sure.
> 
> 
> On Thu, Mar 17, 2011 at 8:32 PM,  <covici at ccs.covici.com> wrote:
> > It seems to be somewhat new, I am on FreeSWITCH Version 1.0.head
> > (git-5640464 2011-03-08 12-40-58 -0600) -- I can certainly try to update
> > and see if that helps -- I don't like to update weekly, unless its
> > necessary. ?I think I was on something from 2/11 before this one.
> >
> >
> > Anthony Minessale <anthony.minessale at gmail.com> wrote:
> >
> >> What revision of GIT are you on? ?Is this s new development for you?
> >>
> >>
> >> On Thu, Mar 17, 2011 at 6:49 PM, ?<covici at ccs.covici.com> wrote:
> >> > no loopback used here for that.
> >> >
> >> > George Niculae <george.niculae79 at gmail.com> wrote:
> >> >
> >> >> 2011/3/18 <covici at ccs.covici.com>
> >> >>
> >> >> > Yep, real hardware. ?I wonder if its my itsp, but since it happened on
> >> >> > the fs conference, I doubt it.
> >> >> >
> >> >> >
> >> >> I experienced audio delay in fs conferences some time ago when bridging conf
> >> >> and using mod_loopback. Is this also your case?
> >> >>
> >> >> George
> >> >>
> >> >> ----------------------------------------------------
> >> >> Alternatives:
> >> >>
> >> >> ----------------------------------------------------
> >> >
> >> > --
> >> > Your life is like a penny. ?You're going to lose it. ?The question is:
> >> > How do
> >> > you spend it?
> >> >
> >> > ? ? ? ? John Covici
> >> > ? ? ? ? covici at ccs.covici.com
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > 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-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > --
> > Your life is like a penny. ?You're going to lose it. ?The question is:
> > How do
> > you spend it?
> >
> > ? ? ? ? John Covici
> > ? ? ? ? covici at ccs.covici.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From peter.olsson at visionutveckling.se  Fri Mar 18 09:58:33 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 18 Mar 2011 07:58:33 +0100
Subject: [Freeswitch-users] very long delays in conference audio
Message-ID: <862F4BE5-67D0-4521-A941-0F5E59078AEE@visionutveckling.se>

Is the delay just in DTMF, or do you know for sure that audio is delayed the same way as well?

Do you have jitter buffer enabled - it will delay the audio some.

Also, is it always the same delay, or does it build up over time?

/Peter

----- Reply message -----
Fr?n: "covici at ccs.covici.com" <covici at ccs.covici.com>
Datum: fre, mar 18, 2011 07:08
Rubrik: [Freeswitch-users] very long delays in conference audio
Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

I updated to latest git and still the delay.
Anthony Minessale <anthony.minessale at gmail.com> wrote:

> yes there was a regression on march 2nd that was fixed about the 9th
> that could explain your symptoms but I would update and find out for
> sure.
>
>
> On Thu, Mar 17, 2011 at 8:32 PM,  <covici at ccs.covici.com> wrote:
> > It seems to be somewhat new, I am on FreeSWITCH Version 1.0.head
> > (git-5640464 2011-03-08 12-40-58 -0600) -- I can certainly try to update
> > and see if that helps -- I don't like to update weekly, unless its
> > necessary.  I think I was on something from 2/11 before this one.
> >
> >
> > Anthony Minessale <anthony.minessale at gmail.com> wrote:
> >
> >> What revision of GIT are you on?  Is this s new development for you?
> >>
> >>
> >> On Thu, Mar 17, 2011 at 6:49 PM,  <covici at ccs.covici.com> wrote:
> >> > no loopback used here for that.
> >> >
> >> > George Niculae <george.niculae79 at gmail.com> wrote:
> >> >
> >> >> 2011/3/18 <covici at ccs.covici.com>
> >> >>
> >> >> > Yep, real hardware.  I wonder if its my itsp, but since it happened on
> >> >> > the fs conference, I doubt it.
> >> >> >
> >> >> >
> >> >> I experienced audio delay in fs conferences some time ago when bridging conf
> >> >> and using mod_loopback. Is this also your case?
> >> >>
> >> >> George
> >> >>
> >> >> ----------------------------------------------------
> >> >> Alternatives:
> >> >>
> >> >> ----------------------------------------------------
> >> >
> >> > --
> >> > Your life is like a penny.  You're going to lose it.  The question is:
> >> > How do
> >> > you spend it?
> >> >
> >> >         John Covici
> >> >         covici at ccs.covici.com
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > 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-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >         John Covici
> >         covici at ccs.covici.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

--
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com

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

!DSPAM:4d82f64d32765360412282!



From covici at ccs.covici.com  Fri Mar 18 10:12:16 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 18 Mar 2011 03:12:16 -0400
Subject: [Freeswitch-users] very long delays in conference audio
In-Reply-To: <862F4BE5-67D0-4521-A941-0F5E59078AEE@visionutveckling.se>
References: <862F4BE5-67D0-4521-A941-0F5E59078AEE@visionutveckling.se>
Message-ID: <3141.1300432336@ccs.covici.com>

The delay seems to build up over time.  I put <action
application="export" data="sip_jitter_buffer_during_bridge=false"/> in
my default dialplan in the global extension after it sets the date -- I
hope that turned off jitter buffer.  I am pretty sure the audio is
delayed because a conference experienced the audio delay, not just dtmf.

Peter Olsson <peter.olsson at visionutveckling.se> wrote:

> Is the delay just in DTMF, or do you know for sure that audio is delayed the same way as well?
> 
> Do you have jitter buffer enabled - it will delay the audio some.
> 
> Also, is it always the same delay, or does it build up over time?
> 
> /Peter
> 
> ----- Reply message -----
> Fr?n: "covici at ccs.covici.com" <covici at ccs.covici.com>
> Datum: fre, mar 18, 2011 07:08
> Rubrik: [Freeswitch-users] very long delays in conference audio
> Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> 
> I updated to latest git and still the delay.
> Anthony Minessale <anthony.minessale at gmail.com> wrote:
> 
> > yes there was a regression on march 2nd that was fixed about the 9th
> > that could explain your symptoms but I would update and find out for
> > sure.
> >
> >
> > On Thu, Mar 17, 2011 at 8:32 PM,  <covici at ccs.covici.com> wrote:
> > > It seems to be somewhat new, I am on FreeSWITCH Version 1.0.head
> > > (git-5640464 2011-03-08 12-40-58 -0600) -- I can certainly try to update
> > > and see if that helps -- I don't like to update weekly, unless its
> > > necessary.  I think I was on something from 2/11 before this one.
> > >
> > >
> > > Anthony Minessale <anthony.minessale at gmail.com> wrote:
> > >
> > >> What revision of GIT are you on?  Is this s new development for you?
> > >>
> > >>
> > >> On Thu, Mar 17, 2011 at 6:49 PM,  <covici at ccs.covici.com> wrote:
> > >> > no loopback used here for that.
> > >> >
> > >> > George Niculae <george.niculae79 at gmail.com> wrote:
> > >> >
> > >> >> 2011/3/18 <covici at ccs.covici.com>
> > >> >>
> > >> >> > Yep, real hardware.  I wonder if its my itsp, but since it happened on
> > >> >> > the fs conference, I doubt it.
> > >> >> >
> > >> >> >
> > >> >> I experienced audio delay in fs conferences some time ago when bridging conf
> > >> >> and using mod_loopback. Is this also your case?
> > >> >>
> > >> >> George
> > >> >>
> > >> >> ----------------------------------------------------
> > >> >> Alternatives:
> > >> >>
> > >> >> ----------------------------------------------------
> > >> >
> > >> > --
> > >> > Your life is like a penny.  You're going to lose it.  The question is:
> > >> > How do
> > >> > you spend it?
> > >> >
> > >> >         John Covici
> > >> >         covici at ccs.covici.com
> > >> >
> > >> > _______________________________________________
> > >> > FreeSWITCH-users mailing list
> > >> > FreeSWITCH-users at lists.freeswitch.org
> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> > 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-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> http://www.freeswitch.org
> > >
> > > --
> > > Your life is like a penny.  You're going to lose it.  The question is:
> > > How do
> > > you spend it?
> > >
> > >         John Covici
> > >         covici at ccs.covici.com
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > 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-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> !DSPAM:4d82f64d32765360412282!
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From thisjoy0528 at gmail.com  Fri Mar 18 10:21:56 2011
From: thisjoy0528 at gmail.com (joy this)
Date: Fri, 18 Mar 2011 15:21:56 +0800
Subject: [Freeswitch-users]  Questions of packages loss
Message-ID: <AANLkTi=3xi2pTMd58RzD+A6VnCJLxb8Jai8__e9s+kCu@mail.gmail.com>

    I have some tests for the problem that the voice keeps off and on during
the call. I use Wireshark to check the package flow on the caller, callee,
and FS sides. The network structure:
caller(PCMA)---(Wi-Fi)--->WAN--->FS--->callee(PCMA). FS and the callee are
under the same router. The result is: The caller sends 2688 packages to FS,
and FS receives 2687 packages. Then FS sends only 2380 packages to the
callee, and the callee receives 2380 packages. There is about 11.4% package
loss in FS, and the callee hears incomplete voice.

    I have some questions:

    1. I guess FS drops packages from the caller because these packages
arrived too late. Is it reasonable? If not, what is the reason?

    2. I want to check if FS drops packages because of the late arrival.
Could you tell me the packages flow in the source code?

    3. I set the jitterbuffer_msec from 10 to 10000, but it seems not
effective. Wouldn?t the jitterbuffer_msec affect the result?


Sincerely yours,

 thisjoy.
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From steveayre at gmail.com  Fri Mar 18 11:27:30 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 18 Mar 2011 08:27:30 +0000
Subject: [Freeswitch-users] Questions of packages loss
In-Reply-To: <AANLkTi=3xi2pTMd58RzD+A6VnCJLxb8Jai8__e9s+kCu@mail.gmail.com>
References: <AANLkTi=3xi2pTMd58RzD+A6VnCJLxb8Jai8__e9s+kCu@mail.gmail.com>
Message-ID: <AANLkTi=OB6w8bMWgp7UJWCDd_RzX+iX2tzg-=Zva=NHm@mail.gmail.com>

Some RTP packets will be dropped because they either arrive before the
signalling starts the media (on 18x or 200), or they arrive after the
hangup.

How long was the call you made? 2380 suggests under a minute? Try a much
longer call and see whether you're still seeing the same amount of loss.

-Steve


On 18 March 2011 07:21, joy this <thisjoy0528 at gmail.com> wrote:

>     I have some tests for the problem that the voice keeps off and on
> during the call. I use Wireshark to check the package flow on the caller,
> callee, and FS sides. The network structure:
> caller(PCMA)---(Wi-Fi)--->WAN--->FS--->callee(PCMA). FS and the callee are
> under the same router. The result is: The caller sends 2688 packages to FS,
> and FS receives 2687 packages. Then FS sends only 2380 packages to the
> callee, and the callee receives 2380 packages. There is about 11.4% package
> loss in FS, and the callee hears incomplete voice.
>
>     I have some questions:
>
>     1. I guess FS drops packages from the caller because these packages
> arrived too late. Is it reasonable? If not, what is the reason?
>
>     2. I want to check if FS drops packages because of the late arrival.
> Could you tell me the packages flow in the source code?
>
>     3. I set the jitterbuffer_msec from 10 to 10000, but it seems not
> effective. Wouldn?t the jitterbuffer_msec affect the result?
>
>
> Sincerely yours,
>
>  thisjoy.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From thisjoy0528 at gmail.com  Fri Mar 18 12:50:13 2011
From: thisjoy0528 at gmail.com (joy this)
Date: Fri, 18 Mar 2011 17:50:13 +0800
Subject: [Freeswitch-users] Questions of packages loss
In-Reply-To: <AANLkTi=OB6w8bMWgp7UJWCDd_RzX+iX2tzg-=Zva=NHm@mail.gmail.com>
References: <AANLkTi=3xi2pTMd58RzD+A6VnCJLxb8Jai8__e9s+kCu@mail.gmail.com>
	<AANLkTi=OB6w8bMWgp7UJWCDd_RzX+iX2tzg-=Zva=NHm@mail.gmail.com>
Message-ID: <AANLkTimsn=-Acu0nw8yOsA+siyoM7q=K9Ac=v62yp7qh@mail.gmail.com>

    Thank you for suggestions.
    I test it again, and the callee answers as soon as possible.
    The result is:
        1. caller---(24028)--->(24038)FS---(21679)--->(21679)callee. The
call duration is 8 minutes. The drop rate is 9.8%.
        2. caller---(27042)--->(27054)FS---(24670)--->(24670)callee. The
call duration is 9 minutes. The drop rate is 8.8%.
    The drop rate reduces actually. But I still hear the voice which keeps
off and on. I will try another test in different network structure to check
if it is cause by the network traffic.
Sincerely yours,
thisjoy.

On Fri, Mar 18, 2011 at 4:27 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Some RTP packets will be dropped because they either arrive before the
> signalling starts the media (on 18x or 200), or they arrive after the
> hangup.
>
> How long was the call you made? 2380 suggests under a minute? Try a much
> longer call and see whether you're still seeing the same amount of loss.
>
> -Steve
>
>
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From pkelly at gmail.com  Fri Mar 18 18:57:26 2011
From: pkelly at gmail.com (Pete Kelly)
Date: Fri, 18 Mar 2011 15:57:26 +0000
Subject: [Freeswitch-users] Lua playAndCollectDigits timeout problem.
	Possible bug
Message-ID: <AANLkTin22=Y6pBYSYX885C10HsGV_jXHaZP8w2nMebEJ@mail.gmail.com>

Hi, I am trying to invoke a timeout using the session:playAndCollectDigits
lua function, this is the syntax I am using:

call_number = session:playAndGetDigits(1, 20, 1, 10000, "#*",
"some-file.wav", "", "[0-9]{1,}");

If the user doesn't hit * or # I need the function to return after 10
seconds.

However this timeout seems to only kick in if I start entering DTMF *after*
the initial prompt has played. If I start entering DTMF during the prompt,
the function never returns due to timeout.

Has anyone else seen/experienced this ?

Pete
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From manavid at gmail.com  Fri Mar 18 17:34:48 2011
From: manavid at gmail.com (Moe Navid)
Date: Fri, 18 Mar 2011 07:34:48 -0700
Subject: [Freeswitch-users] Voicemail with no predefined profile
Message-ID: <AANLkTinc5TpC5b=9-br4xMonQE7UK=qWxkqYSt6OE-v+@mail.gmail.com>

Hi all,

I'm in the process of migrating my services from Asterisk to FreeSWITCH.

Is there a way to create voicemail boxes dynamically on the fly
without predefining user profiles?

Right now I'm using OpenSIPS as my registrar and load balancer for my
Asterisk boxes. I use asterisk's real time in conjunction with
res_config_mysql to I define my voicemail boxes in my mysql table and
Asterisk reads it's info from there.

Can I do something similar with FreeSWITCH?

The only way I can find to do it in FreeSWITCH is using mod_xml_curl.

Thank you

Moe
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From jeff at jefflenk.com  Fri Mar 18 19:33:09 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 18 Mar 2011 09:33:09 -0700 (PDT)
Subject: [Freeswitch-users] Lua playAndCollectDigits timeout problem.
	Possible bug
In-Reply-To: <AANLkTin22=Y6pBYSYX885C10HsGV_jXHaZP8w2nMebEJ@mail.gmail.com>
References: <AANLkTin22=Y6pBYSYX885C10HsGV_jXHaZP8w2nMebEJ@mail.gmail.com>
Message-ID: <1300465989227-6185228.post@n2.nabble.com>

What version are you running?

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Lua-playAndCollectDigits-timeout-problem-Possible-bug-tp6185101p6185228.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Fri Mar 18 19:48:43 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 18 Mar 2011 11:48:43 -0500
Subject: [Freeswitch-users] very long delays in conference audio
In-Reply-To: <3141.1300432336@ccs.covici.com>
References: <862F4BE5-67D0-4521-A941-0F5E59078AEE@visionutveckling.se>
	<3141.1300432336@ccs.covici.com>
Message-ID: <AANLkTik5KwBC1QZ4kg40ZrKzg5Q4vX-UbLKc0=-6b=87@mail.gmail.com>

there already is no jitter buffer unless you configured it to be on.

If you actually updated and you still have a problem, your box has
some broken timing issue.
All you can do at this point is export rtp_timer_name=none in your
dialplan before you answer or globally in vars.xml

Chances are you missed something on your update because it would make
more sense to me you didn't actually update.
If there was any real problem we would be seeing it widespread.



On Fri, Mar 18, 2011 at 2:12 AM,  <covici at ccs.covici.com> wrote:
> The delay seems to build up over time. ?I put <action
> application="export" data="sip_jitter_buffer_during_bridge=false"/> in
> my default dialplan in the global extension after it sets the date -- I
> hope that turned off jitter buffer. ?I am pretty sure the audio is
> delayed because a conference experienced the audio delay, not just dtmf.
>
> Peter Olsson <peter.olsson at visionutveckling.se> wrote:
>
>> Is the delay just in DTMF, or do you know for sure that audio is delayed the same way as well?
>>
>> Do you have jitter buffer enabled - it will delay the audio some.
>>
>> Also, is it always the same delay, or does it build up over time?
>>
>> /Peter
>>
>> ----- Reply message -----
>> Fr?n: "covici at ccs.covici.com" <covici at ccs.covici.com>
>> Datum: fre, mar 18, 2011 07:08
>> Rubrik: [Freeswitch-users] very long delays in conference audio
>> Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>
>> I updated to latest git and still the delay.
>> Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>
>> > yes there was a regression on march 2nd that was fixed about the 9th
>> > that could explain your symptoms but I would update and find out for
>> > sure.
>> >
>> >
>> > On Thu, Mar 17, 2011 at 8:32 PM, ?<covici at ccs.covici.com> wrote:
>> > > It seems to be somewhat new, I am on FreeSWITCH Version 1.0.head
>> > > (git-5640464 2011-03-08 12-40-58 -0600) -- I can certainly try to update
>> > > and see if that helps -- I don't like to update weekly, unless its
>> > > necessary. ?I think I was on something from 2/11 before this one.
>> > >
>> > >
>> > > Anthony Minessale <anthony.minessale at gmail.com> wrote:
>> > >
>> > >> What revision of GIT are you on? ?Is this s new development for you?
>> > >>
>> > >>
>> > >> On Thu, Mar 17, 2011 at 6:49 PM, ?<covici at ccs.covici.com> wrote:
>> > >> > no loopback used here for that.
>> > >> >
>> > >> > George Niculae <george.niculae79 at gmail.com> wrote:
>> > >> >
>> > >> >> 2011/3/18 <covici at ccs.covici.com>
>> > >> >>
>> > >> >> > Yep, real hardware. ?I wonder if its my itsp, but since it happened on
>> > >> >> > the fs conference, I doubt it.
>> > >> >> >
>> > >> >> >
>> > >> >> I experienced audio delay in fs conferences some time ago when bridging conf
>> > >> >> and using mod_loopback. Is this also your case?
>> > >> >>
>> > >> >> George
>> > >> >>
>> > >> >> ----------------------------------------------------
>> > >> >> Alternatives:
>> > >> >>
>> > >> >> ----------------------------------------------------
>> > >> >
>> > >> > --
>> > >> > Your life is like a penny. ?You're going to lose it. ?The question is:
>> > >> > How do
>> > >> > you spend it?
>> > >> >
>> > >> > ? ? ? ? John Covici
>> > >> > ? ? ? ? covici at ccs.covici.com
>> > >> >
>> > >> > _______________________________________________
>> > >> > FreeSWITCH-users mailing list
>> > >> > FreeSWITCH-users at lists.freeswitch.org
>> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > >> > 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-users mailing list
>> > >> FreeSWITCH-users at lists.freeswitch.org
>> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > >> http://www.freeswitch.org
>> > >
>> > > --
>> > > Your life is like a penny. ?You're going to lose it. ?The question is:
>> > > How do
>> > > you spend it?
>> > >
>> > > ? ? ? ? John Covici
>> > > ? ? ? ? covici at ccs.covici.com
>> > >
>> > > _______________________________________________
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > 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-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> --
>> Your life is like a penny. ?You're going to lose it. ?The question is:
>> How do
>> you spend it?
>>
>> ? ? ? ? ?John Covici
>> ? ? ? ? ?covici at ccs.covici.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> !DSPAM:4d82f64d32765360412282!
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> --
> Your life is like a penny. ?You're going to lose it. ?The question is:
> How do
> you spend it?
>
> ? ? ? ? John Covici
> ? ? ? ? covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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  Fri Mar 18 19:52:25 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 18 Mar 2011 11:52:25 -0500
Subject: [Freeswitch-users] Lua playAndCollectDigits timeout problem.
 Possible bug
In-Reply-To: <AANLkTin22=Y6pBYSYX885C10HsGV_jXHaZP8w2nMebEJ@mail.gmail.com>
References: <AANLkTin22=Y6pBYSYX885C10HsGV_jXHaZP8w2nMebEJ@mail.gmail.com>
Message-ID: <AANLkTinzvnvJ=5oCaNZSP59kniMtPthJPHSGU7=vikrV@mail.gmail.com>

there is a new final param of digit timeout once you type something.
In older versions it was set to 0 by default meaning forever but on
latest its equiv to the global timeout.

add another ,10 to your arg list


On Fri, Mar 18, 2011 at 10:57 AM, Pete Kelly <pkelly at gmail.com> wrote:
> Hi, I am trying to invoke a timeout using the session:playAndCollectDigits
> lua function, this is the syntax I am using:
> call_number = session:playAndGetDigits(1, 20, 1, 10000, "#*",
> "some-file.wav", "",?"[0-9]{1,}");
> If the user doesn't hit * or # I need the function to return after 10
> seconds.
> However this timeout seems to only kick in if I start entering DTMF *after*
> the initial prompt has played. If I start entering DTMF during the prompt,
> the function never returns due to timeout.
> Has anyone else seen/experienced this ?
> Pete
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 Mar 18 19:53:06 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 18 Mar 2011 09:53:06 -0700
Subject: [Freeswitch-users] Voicemail with no predefined profile
In-Reply-To: <AANLkTinc5TpC5b=9-br4xMonQE7UK=qWxkqYSt6OE-v+@mail.gmail.com>
References: <AANLkTinc5TpC5b=9-br4xMonQE7UK=qWxkqYSt6OE-v+@mail.gmail.com>
Message-ID: <AANLkTimPCHthYSrbhBLh9a7c00Gt0Z+bQ7uFA8ML9Q8G@mail.gmail.com>

On Fri, Mar 18, 2011 at 7:34 AM, Moe Navid <manavid at gmail.com> wrote:

> Hi all,
>
> I'm in the process of migrating my services from Asterisk to FreeSWITCH.
>
> Is there a way to create voicemail boxes dynamically on the fly
> without predefining user profiles?
>
> Right now I'm using OpenSIPS as my registrar and load balancer for my
> Asterisk boxes. I use asterisk's real time in conjunction with
> res_config_mysql to I define my voicemail boxes in my mysql table and
> Asterisk reads it's info from there.
>
> Can I do something similar with FreeSWITCH?
>
> The only way I can find to do it in FreeSWITCH is using mod_xml_curl.
>

In your case, mod_xml_curl is the way to go. You will need to build a little
web server to handle the requests from mod_xml_curl and return the data from
your mysql table in properly formatted XML. I recommend that you look at the
freeswitch-contrib, specifically in intralanman/PHP/fs_curl/ to see an
example of how you can return user directory information. Look at the README
file in there for more information on how to get things set up. As long as
you return properly formatted XML for your dialplan and/or directory then
you should be good to go and FS will do the rest.

-MC
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From manavid at gmail.com  Fri Mar 18 20:14:35 2011
From: manavid at gmail.com (Moe Navid)
Date: Fri, 18 Mar 2011 10:14:35 -0700
Subject: [Freeswitch-users] Voicemail with no predefined profile
In-Reply-To: <AANLkTimPCHthYSrbhBLh9a7c00Gt0Z+bQ7uFA8ML9Q8G@mail.gmail.com>
References: <AANLkTinc5TpC5b=9-br4xMonQE7UK=qWxkqYSt6OE-v+@mail.gmail.com>
	<AANLkTimPCHthYSrbhBLh9a7c00Gt0Z+bQ7uFA8ML9Q8G@mail.gmail.com>
Message-ID: <AANLkTi=b5JUTA+EMwcO0Y0tywxXViQPqrS5ggMhqn+MT@mail.gmail.com>

Thanks Michael,

I guess that's what I have to do. I'm very comfortable with generating
settings on my web server, we are using Rails and it's very easy to add
stuff. This is how currently I'm feeding my settings to my Asterisk cluster.
Asterisk has #exec option for config files I have a custom script
that retrieves the settings from my web server and the server distinguishes
my asterisk boxes based on their IP address. I see that mod_xml_curl is very
sophisticated and very flexible.

One question: does FS has any limits or performance bottlenecks in regards
to xml profiles we define?

Thank you

Moe

On Fri, Mar 18, 2011 at 9:53 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Mar 18, 2011 at 7:34 AM, Moe Navid <manavid at gmail.com> wrote:
>
>> Hi all,
>>
>> I'm in the process of migrating my services from Asterisk to FreeSWITCH.
>>
>> Is there a way to create voicemail boxes dynamically on the fly
>> without predefining user profiles?
>>
>> Right now I'm using OpenSIPS as my registrar and load balancer for my
>> Asterisk boxes. I use asterisk's real time in conjunction with
>> res_config_mysql to I define my voicemail boxes in my mysql table and
>> Asterisk reads it's info from there.
>>
>> Can I do something similar with FreeSWITCH?
>>
>> The only way I can find to do it in FreeSWITCH is using mod_xml_curl.
>>
>
> In your case, mod_xml_curl is the way to go. You will need to build a
> little web server to handle the requests from mod_xml_curl and return the
> data from your mysql table in properly formatted XML. I recommend that you
> look at the freeswitch-contrib, specifically in intralanman/PHP/fs_curl/ to
> see an example of how you can return user directory information. Look at the
> README file in there for more information on how to get things set up. As
> long as you return properly formatted XML for your dialplan and/or directory
> then you should be good to go and FS will do the rest.
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gustavo.espeche at easyipcall.com  Fri Mar 18 20:29:18 2011
From: gustavo.espeche at easyipcall.com (Gustavo Espeche)
Date: Fri, 18 Mar 2011 14:29:18 -0300
Subject: [Freeswitch-users] Diversion Header
Message-ID: <1300469358.2342.6.camel@gustavo-laptop>

Hi all we are having a problem with a customer that send the call with
header Diversion, and our Freeswitch reject the call because don't
understand the header.
Some one know how can fix this problem?
Best Regards.

Gustavo Espeche
www.easyipcall.com






From elijah at crankenstein.com  Fri Mar 18 21:53:13 2011
From: elijah at crankenstein.com (elijah)
Date: Fri, 18 Mar 2011 11:53:13 -0700
Subject: [Freeswitch-users] some outbound calls failing
In-Reply-To: <AANLkTi=6VGzHjt1bSDdy8gWuqUSoZ47A5pR=SGH6Zgu-@mail.gmail.com>
References: <AANLkTinmvz055Qb7Ep2PMZ-sS6GJmTVVpC2hdapyJXb0@mail.gmail.com>
	<AANLkTi=6VGzHjt1bSDdy8gWuqUSoZ47A5pR=SGH6Zgu-@mail.gmail.com>
Message-ID: <AANLkTi=F+WdffeJh2-n2JJOSgrHfAwq3tcr48JDEH1vx@mail.gmail.com>

I updated then tried a few dozen times to reproduce this condition and could
not. So I'll watch for a while and see how it goes. I updated last week but
maybe you changed something in the interim? In any event thanks for the help
and I will update this thread should I catch this event in failure.

On Thu, Mar 17, 2011 at 6:22 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> can you see if you have that same problem with the latest build from git
> you can update an existing checkout with "make current" assuming you
> have git tools etc
>
> if you still can get the problem capture the debug logs of it happening
>
> console loglevel debug
>
>
> On Thu, Mar 17, 2011 at 6:53 PM, elijah <elijah at crankenstein.com> wrote:
> > I have put together a dialer (of sorts) that is failing a fraction of the
> > time. First, I instantiate a call leg from FreeSwitch to an agent (an
> > employee in my facility with a softclient registered to FreeSwitch). Upon
> > answering this call I have FreeSwitch bridge this leg to an outbound leg
> > destined to a customer (via our carrier). The agent listens as this
> external
> > number is dialed then is prepared to speak with that person. So all is
> well
> > except a fraction of these calls fail at the step where FreeSwitch calls
> the
> > agent. There seems to be a 'hangup' event received from the client at the
> > moment this leg is answered. This happens maybe 1/3 of the time.
> > here's my steps
> > 1. call a Lua script via ESL
> > 2. Lua script generates a call leg to agent:
> >                 local session = freeswitch.Session("user/" .. agentExt ..
> > "@stuff.com")
> >                 session:sleep(1000)
> >                 if session:ready() then
> >                         session:execute("pre_answer")
> >                         session:execute("ring_ready")
> >                         session:execute("sleep", "1000")
> >                         session:setVariable("outbound_callee_number",
> > outboundCalleeNumber)
> >                         session:setVariable("outbound_caller_number",
> > outboundCallerNumber)
> >                         session:setVariable("outbound_caller_name",
> > outboundCallerName)
> >                         session:transfer("sales_outbound", "XML",
> "telifi")
> > ******** fails here ~30% of the time - phone rings on the agent's desk,
> > she answers and the call immediately ends ********
> > 3. this leg is transferred to an XML dialplan that handles bridging to a
> > customer's phone number:
> >     <extension name="sales_outbound">
> >       <condition field="destination_number"
> expression="^sales_outbound$">
> >         <action application="log" data="INFO connected to agent, now
> dialing
> > out..."/>
> >         <action application="sleep" data="1000"/>
> >         <action application="playback"
> > data="ivr/8000/ivr-call_being_transferred.wav"/>-->
> >         <action application="set" data="continue_on_fail=true"/>
> >         <action application="set" data="hangup_after_bridge=true"/>
> >         <action application="set" data="ringback=${us-ring}"/>
> >         <action application="set" data="call_timeout=30"/>
> >         <action application="set"
> > data="effective_caller_id_number=${outbound_caller_number}"/>
> >         <action application="set"
> > data="effective_caller_id_name=${outbound_caller_name}"/>
> >         <action application="sleep" data="1000"/>
> >         <action application="bridge"
> > data="sofia/gateway/mycarrier/${outbound_callee_number}"/>
> >       </condition>
> >     </extension>
> > I appreciate any insight or suggestions you could provide.
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Fri Mar 18 22:25:23 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 18 Mar 2011 12:25:23 -0700
Subject: [Freeswitch-users] Voicemail with no predefined profile
In-Reply-To: <AANLkTi=b5JUTA+EMwcO0Y0tywxXViQPqrS5ggMhqn+MT@mail.gmail.com>
References: <AANLkTinc5TpC5b=9-br4xMonQE7UK=qWxkqYSt6OE-v+@mail.gmail.com>
	<AANLkTimPCHthYSrbhBLh9a7c00Gt0Z+bQ7uFA8ML9Q8G@mail.gmail.com>
	<AANLkTi=b5JUTA+EMwcO0Y0tywxXViQPqrS5ggMhqn+MT@mail.gmail.com>
Message-ID: <AANLkTi=C0L0=DYA2ED7E_ZTRr32RCfLL1+C4MMvBsh-=@mail.gmail.com>

On Fri, Mar 18, 2011 at 10:14 AM, Moe Navid <manavid at gmail.com> wrote:

> Thanks Michael,
>
> I guess that's what I have to do. I'm very comfortable with generating
> settings on my web server, we are using Rails and it's very easy to add
> stuff. This is how currently I'm feeding my settings to my Asterisk cluster.
> Asterisk has #exec option for config files I have a custom script
> that retrieves the settings from my web server and the server distinguishes
> my asterisk boxes based on their IP address. I see that mod_xml_curl is very
> sophisticated and very flexible.
>
> One question: does FS has any limits or performance bottlenecks in regards
> to xml profiles we define?
>
Not in any sane configuration. We've got people doing dozens of calls per
second (or more) and things are going quite well.

-MC
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From manavid at gmail.com  Fri Mar 18 23:55:42 2011
From: manavid at gmail.com (Moe Navid)
Date: Fri, 18 Mar 2011 13:55:42 -0700
Subject: [Freeswitch-users] Voicemail with no predefined profile
In-Reply-To: <AANLkTi=C0L0=DYA2ED7E_ZTRr32RCfLL1+C4MMvBsh-=@mail.gmail.com>
References: <AANLkTinc5TpC5b=9-br4xMonQE7UK=qWxkqYSt6OE-v+@mail.gmail.com>
	<AANLkTimPCHthYSrbhBLh9a7c00Gt0Z+bQ7uFA8ML9Q8G@mail.gmail.com>
	<AANLkTi=b5JUTA+EMwcO0Y0tywxXViQPqrS5ggMhqn+MT@mail.gmail.com>
	<AANLkTi=C0L0=DYA2ED7E_ZTRr32RCfLL1+C4MMvBsh-=@mail.gmail.com>
Message-ID: <AANLkTi=RXv44_cjKkLXQ2p3O3Q-tHUzOVECebvFyqa_U@mail.gmail.com>

Thanks Michael.

Moe

On Fri, Mar 18, 2011 at 12:25 PM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Fri, Mar 18, 2011 at 10:14 AM, Moe Navid <manavid at gmail.com> wrote:
>
>> Thanks Michael,
>>
>> I guess that's what I have to do. I'm very comfortable with generating
>> settings on my web server, we are using Rails and it's very easy to add
>> stuff. This is how currently I'm feeding my settings to my Asterisk cluster.
>> Asterisk has #exec option for config files I have a custom script
>> that retrieves the settings from my web server and the server distinguishes
>> my asterisk boxes based on their IP address. I see that mod_xml_curl is very
>> sophisticated and very flexible.
>>
>> One question: does FS has any limits or performance bottlenecks in regards
>> to xml profiles we define?
>>
> Not in any sane configuration. We've got people doing dozens of calls per
> second (or more) and things are going quite well.
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bwibowo at gmail.com  Sat Mar 19 01:16:09 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Sat, 19 Mar 2011 05:16:09 +0700
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <AANLkTimoAGNBvi4mDjU7_DsoTsyanNVVjSDfwOWUr1eJ@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTim0vpXhg16p6LTwbYa7ng5Ki-5-aw=MpfZOT3pY@mail.gmail.com>
	<AANLkTimoAGNBvi4mDjU7_DsoTsyanNVVjSDfwOWUr1eJ@mail.gmail.com>
Message-ID: <AANLkTikizxLZzMyBH3FRrUNCNhpdAFkprXdRMUrRKEi9@mail.gmail.com>

any update for this matter?


thx

budi

On Sat, Mar 5, 2011 at 12:45 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello,
>
> No, i haven't uploaded it yet (i still don't know how); as it is still not
> finished. I might be doing so in a week or so.
>
> Cheers
>
> david
>
>
> On Fri, Mar 4, 2011 at 12:45 AM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> Hi - I didn't notice this in my latest git contrib pull today. Did you get
>> the access worked out?
>> -Avi
>>
>> On Sun, Feb 27, 2011 at 10:06 PM, Saeed Ahmed <saeedahmad1981 at gmail.com>wrote:
>>
>>> press sent too quick..
>>>
>>> what did you use for routing? curl? esl?
>>>
>>> did you use nibble bill for prepaid app?
>>>
>>>
>>> On Sun, Feb 27, 2011 at 9:05 PM, Saeed Ahmed <saeedahmad1981 at gmail.com>wrote:
>>>
>>>> Great!
>>>>
>>>> want to see it soon.
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Sat Mar 19 01:56:13 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 18 Mar 2011 15:56:13 -0700
Subject: [Freeswitch-users] New FreeSWITCH Module: mod_ladspa
Message-ID: <AANLkTimPo4U2bzf+K4ucRq3xZHHDBnjFouhXDd_t+Fzk@mail.gmail.com>

Hello all!

The FreeSWITCH developers have added a cool new module: mod_ladspa. Check
out the story:

http://www.freeswitch.org/node/313

Have fun!
-Michael
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From dmitry.bely at gmail.com  Sat Mar 19 14:27:41 2011
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Sat, 19 Mar 2011 14:27:41 +0300
Subject: [Freeswitch-users] Outbound caller id for a specific gateway?
Message-ID: <AANLkTin=M3MJKB5QA86iL+hkhkZncTQ8eiKQJM2t6ap5@mail.gmail.com>

My VoIP provider requires a specific caller ID set for an outbound
call otherwise the call is rejected. Currently I set it just before
bridge

      <action application="set" data="effective_caller_id_name=..."/>
      <action application="bridge" data="..."/>

But it's tedious as there is a number of bridge commands in the
dialplan and I still have to explicitly specify the caller id for
"originate" command in the FreeSWITCH console. Is it possible to force
an outbound caller id on a gateway basis?

- Dmitry Bely


From boris at tagnet.ru  Sat Mar 19 20:58:15 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 19 Mar 2011 22:58:15 +0500
Subject: [Freeswitch-users] More then one number for registered user
Message-ID: <4D84EEB7.3030600@tagnet.ru>

Hello!

     Is this possible to have one registered user and more then one 
associated number with it (without tricks with dialplan)? For example:
<user id="XXX" number-alias="YYY,ZZZ.....">

-- 
Regards,
Boris




From dunchan at freemail.hu  Sun Mar 20 10:57:57 2011
From: dunchan at freemail.hu (dunchan)
Date: Sun, 20 Mar 2011 08:57:57 +0100
Subject: [Freeswitch-users] pass variable from dialplan to bridge
Message-ID: <4D85B385.4060407@freemail.hu>

Hi!

I have a simple question, how can i pass a varible from dialplan to the 
bridge?
I want to fill the 'from-user' field depends on sip user settings.

I've tryed the following in dialplan:
...
some conditions
...
<action application="set" data="from-user=${outbound_caller_id_number}"/>
<action application="bridge" data="sofia/gateway/upc/$1"/>

in a gateway xml i tryed:
<gateway name="upc">
         <param name="username" value="..."/>
         <param name="password" value="..."/>
         <param name="proxy" value="..."/>
         <param name="register" value="false"/>
         <param name="from-user" value="${from-user}"/>
     </gateway>

Above config ha no effect :(
any suggestions?

thx,
Viktor





From nick.rosier at gmail.com  Sat Mar 19 20:03:18 2011
From: nick.rosier at gmail.com (Nick Rosier)
Date: Sat, 19 Mar 2011 18:03:18 +0100
Subject: [Freeswitch-users] Outbound dialing to SPA3102 behind NAT
Message-ID: <AANLkTik_kYxvxfWxxb5cMjohDEHCj7qroomKbeQyOXK_@mail.gmail.com>

Hi,

I've currently got an Asterisk server running at home but want to
switch to FS on my external server (located in a DC).

Configuration:
- 2 SPA3102 connected to 2 PSTN lines and 2 phones (on Line1)
- Couple of voip phones

The Asterisk server is running on my local network as are all the
other devices; everything is connected to a DSL-router with NAT.
The SPA3102 are configured as Trunks in Asterisk with the following
configuration:
allow=ulaw
canreinvite=no
context=from-trunk
disallow=all
dtmfmode=rfc2833
host=dynamic
incominglimit=1
nat=never
port=5061
qualify=yes
secret=trunk1
username=trunk1
type=friend

The SPA3102 are configured to register their PSTN-line to the Asterisk
server. Outbound routes are configured to use 1 of the 2 Trunks.

I want to do the same thing with my FS running on a server in the DC.
I've setup all the devices and can make internal calls. Inbound calls
on the PSTN lines are also forwarded to the correct voip-devices. I
just cannot figure out how to configure outbound dialing to the
PSTN-lines. The Wiki
(http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo) shows I
should configure them as following:

<include>
  <extension name="To PSTN">
    <condition field="destination_number" expression="(.*)">
      <action application="bridge"
data="sofia/internal/${destination_number}@[IP address of
SPA3102]:5061" />
    </condition>
  </extension>
</include>

The problem here is that the SPA3102 are behind my router so the only
possibility would be to setup some port forwarding to on the router
with different ports for the 2 devices. I would rather not do that so
I was wondering if there is an easier/better way to do that.

Any ideas?

Rgds,
N.


From philippe at ppmt.org  Sat Mar 19 23:27:29 2011
From: philippe at ppmt.org (Philippe Le Toquin)
Date: Sat, 19 Mar 2011 16:27:29 -0400
Subject: [Freeswitch-users] what to do after "make current"
Message-ID: <4D8511B1.4030902@ppmt.org>

Hello,

This might sound like a stupid question but I can't find the answer so...

TO update Freeswitch I do

 git pull

 and then

 make current

After that I generally restart freeswitch. But do I really need to do
it? or will it somehow manage to "hot swap" with
the new binary?

If I need to restart. Do I have to do it right after the update or can
it wait until it is night time?

Thanks
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From covici at ccs.covici.com  Sun Mar 20 12:38:18 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sun, 20 Mar 2011 05:38:18 -0400
Subject: [Freeswitch-users] Outbound dialing to SPA3102 behind NAT
In-Reply-To: <AANLkTik_kYxvxfWxxb5cMjohDEHCj7qroomKbeQyOXK_@mail.gmail.com>
References: <AANLkTik_kYxvxfWxxb5cMjohDEHCj7qroomKbeQyOXK_@mail.gmail.com>
Message-ID: <9538.1300613898@ccs.covici.com>

Only way I can think of is have the spa3102 register as an fs gateway
using user name and password and bridge to the gateway -- and make sure
natis enabled on the 3102, so fs can detect its external ip address.

Nick Rosier <nick.rosier at gmail.com> wrote:

> Hi,
> 
> I've currently got an Asterisk server running at home but want to
> switch to FS on my external server (located in a DC).
> 
> Configuration:
> - 2 SPA3102 connected to 2 PSTN lines and 2 phones (on Line1)
> - Couple of voip phones
> 
> The Asterisk server is running on my local network as are all the
> other devices; everything is connected to a DSL-router with NAT.
> The SPA3102 are configured as Trunks in Asterisk with the following
> configuration:
> allow=ulaw
> canreinvite=no
> context=from-trunk
> disallow=all
> dtmfmode=rfc2833
> host=dynamic
> incominglimit=1
> nat=never
> port=5061
> qualify=yes
> secret=trunk1
> username=trunk1
> type=friend
> 
> The SPA3102 are configured to register their PSTN-line to the Asterisk
> server. Outbound routes are configured to use 1 of the 2 Trunks.
> 
> I want to do the same thing with my FS running on a server in the DC.
> I've setup all the devices and can make internal calls. Inbound calls
> on the PSTN lines are also forwarded to the correct voip-devices. I
> just cannot figure out how to configure outbound dialing to the
> PSTN-lines. The Wiki
> (http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo) shows I
> should configure them as following:
> 
> <include>
>   <extension name="To PSTN">
>     <condition field="destination_number" expression="(.*)">
>       <action application="bridge"
> data="sofia/internal/${destination_number}@[IP address of
> SPA3102]:5061" />
>     </condition>
>   </extension>
> </include>
> 
> The problem here is that the SPA3102 are behind my router so the only
> possibility would be to setup some port forwarding to on the router
> with different ports for the 2 devices. I would rather not do that so
> I was wondering if there is an easier/better way to do that.
> 
> Any ideas?
> 
> Rgds,
> N.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From egbert at redhosting.nl  Sun Mar 20 14:10:25 2011
From: egbert at redhosting.nl ([Redhosting] Egbert Groot)
Date: Sun, 20 Mar 2011 12:10:25 +0100
Subject: [Freeswitch-users] Understanding break="never" in condition-tag
Message-ID: <4D85E0A1.507@redhosting.nl>

Hi All,

I'm struggling understanding the 'break' attribute in condition tags.
Am I correct saying the following condition tag is useless? It doesn't
influence the routing/application stack at all.
For this example the check is on 'wday', but it could be any other
field/expression, my question is about the 'break="never"' part:

<condition wday="1" break="never" />

( though it could be used for clarity or debugging, since it does get
evaluated)


regards,
Egbert.




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From dmitry.bely at gmail.com  Sun Mar 20 15:30:57 2011
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Sun, 20 Mar 2011 15:30:57 +0300
Subject: [Freeswitch-users] Understanding break="never" in condition-tag
In-Reply-To: <4D85E0A1.507@redhosting.nl>
References: <4D85E0A1.507@redhosting.nl>
Message-ID: <AANLkTimNYfHOrZrr7Q70Zea8Nsw-xEmQr2z0g-=PK+hv@mail.gmail.com>

On Sun, Mar 20, 2011 at 2:10 PM, [Redhosting] Egbert Groot
<egbert at redhosting.nl> wrote:
> Hi All,
>
> I'm struggling understanding the 'break' attribute in condition tags.
> Am I correct saying the following condition tag is useless? It doesn't
> influence the routing/application stack at all.
> For this example the check is on 'wday', but it could be any other
> field/expression, my question is about the 'break="never"' part:
>
> <condition wday="1" break="never" />
>
> ( though it could be used for clarity or debugging, since it does get
> evaluated)

It makes sense if you use the form

<condition wday="1" break="never">
  (some actions)
</condition>

- Dmitry Bely


From avi at avimarcus.net  Sun Mar 20 15:54:16 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 20 Mar 2011 14:54:16 +0200
Subject: [Freeswitch-users] Understanding break="never" in condition-tag
In-Reply-To: <AANLkTimNYfHOrZrr7Q70Zea8Nsw-xEmQr2z0g-=PK+hv@mail.gmail.com>
References: <4D85E0A1.507@redhosting.nl>
	<AANLkTimNYfHOrZrr7Q70Zea8Nsw-xEmQr2z0g-=PK+hv@mail.gmail.com>
Message-ID: <AANLkTi=1Ufx3wpkiAkkff34GyTK+7-zg7_CYzEM4zcPg@mail.gmail.com>

Using "on true" or "on false" would stop evaluating the entire extension.
This is really best for *nested* extensions as you'll see the two examples
on the wiki are for when you want to keep evaluating the extension in it's
entirety.

-Avi

On Sun, Mar 20, 2011 at 2:30 PM, Dmitry Bely <dmitry.bely at gmail.com> wrote:

> On Sun, Mar 20, 2011 at 2:10 PM, [Redhosting] Egbert Groot
> <egbert at redhosting.nl> wrote:
> > Hi All,
> >
> > I'm struggling understanding the 'break' attribute in condition tags.
> > Am I correct saying the following condition tag is useless? It doesn't
> > influence the routing/application stack at all.
> > For this example the check is on 'wday', but it could be any other
> > field/expression, my question is about the 'break="never"' part:
> >
> > <condition wday="1" break="never" />
> >
> > ( though it could be used for clarity or debugging, since it does get
> > evaluated)
>
> It makes sense if you use the form
>
> <condition wday="1" break="never">
>   (some actions)
> </condition>
>
> - Dmitry Bely
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kbdfck at gmail.com  Sun Mar 20 16:54:48 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Sun, 20 Mar 2011 16:54:48 +0300
Subject: [Freeswitch-users] Outbound dialing to SPA3102 behind NAT
In-Reply-To: <9538.1300613898@ccs.covici.com>
References: <AANLkTik_kYxvxfWxxb5cMjohDEHCj7qroomKbeQyOXK_@mail.gmail.com>
	<9538.1300613898@ccs.covici.com>
Message-ID: <AANLkTi=XO0FTfTyJSu8KVWJmdEjsK2JC6zHENf9Xq84E@mail.gmail.com>

I think there is no need to enable nat on endpoint device, since freeswitch
is able to detect correct address from SIP/RTP messages if request is
received from networks in nat acl.

2011/3/20 <covici at ccs.covici.com>

> Only way I can think of is have the spa3102 register as an fs gateway
> using user name and password and bridge to the gateway -- and make sure
> natis enabled on the 3102, so fs can detect its external ip address.
>
> Nick Rosier <nick.rosier at gmail.com> wrote:
>
> > Hi,
> >
> > I've currently got an Asterisk server running at home but want to
> > switch to FS on my external server (located in a DC).
> >
> > Configuration:
> > - 2 SPA3102 connected to 2 PSTN lines and 2 phones (on Line1)
> > - Couple of voip phones
> >
> > The Asterisk server is running on my local network as are all the
> > other devices; everything is connected to a DSL-router with NAT.
> > The SPA3102 are configured as Trunks in Asterisk with the following
> > configuration:
> > allow=ulaw
> > canreinvite=no
> > context=from-trunk
> > disallow=all
> > dtmfmode=rfc2833
> > host=dynamic
> > incominglimit=1
> > nat=never
> > port=5061
> > qualify=yes
> > secret=trunk1
> > username=trunk1
> > type=friend
> >
> > The SPA3102 are configured to register their PSTN-line to the Asterisk
> > server. Outbound routes are configured to use 1 of the 2 Trunks.
> >
> > I want to do the same thing with my FS running on a server in the DC.
> > I've setup all the devices and can make internal calls. Inbound calls
> > on the PSTN lines are also forwarded to the correct voip-devices. I
> > just cannot figure out how to configure outbound dialing to the
> > PSTN-lines. The Wiki
> > (http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo) shows I
> > should configure them as following:
> >
> > <include>
> >   <extension name="To PSTN">
> >     <condition field="destination_number" expression="(.*)">
> >       <action application="bridge"
> > data="sofia/internal/${destination_number}@[IP address of
> > SPA3102]:5061" />
> >     </condition>
> >   </extension>
> > </include>
> >
> > The problem here is that the SPA3102 are behind my router so the only
> > possibility would be to setup some port forwarding to on the router
> > with different ports for the 2 devices. I would rather not do that so
> > I was wondering if there is an easier/better way to do that.
> >
> > Any ideas?
> >
> > Rgds,
> > N.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From egbert at redhosting.nl  Sun Mar 20 15:45:47 2011
From: egbert at redhosting.nl ([Redhosting] Egbert Groot)
Date: Sun, 20 Mar 2011 13:45:47 +0100
Subject: [Freeswitch-users] Understanding break="never" in condition-tag
In-Reply-To: <AANLkTimNYfHOrZrr7Q70Zea8Nsw-xEmQr2z0g-=PK+hv@mail.gmail.com>
References: <4D85E0A1.507@redhosting.nl>
	<AANLkTimNYfHOrZrr7Q70Zea8Nsw-xEmQr2z0g-=PK+hv@mail.gmail.com>
Message-ID: <4D85F6FB.8080801@redhosting.nl>

Dmitry Bely schreef:
> On Sun, Mar 20, 2011 at 2:10 PM, [Redhosting] Egbert Groot
> <egbert at redhosting.nl> wrote:
>   
>> Hi All,
>>
>> I'm struggling understanding the 'break' attribute in condition tags.
>> Am I correct saying the following condition tag is useless? It doesn't
>> influence the routing/application stack at all.
>> For this example the check is on 'wday', but it could be any other
>> field/expression, my question is about the 'break="never"' part:
>>
>> <condition wday="1" break="never" />
>>
>> ( though it could be used for clarity or debugging, since it does get
>> evaluated)
>>     
>
> It makes sense if you use the form
>
> <condition wday="1" break="never">
>   (some actions)
> </condition>
>
>   
Indeed, with adding an action, it is/can be usefull.
I was reading in wiki: 
http://wiki.freeswitch.org/wiki/Dialplan_XML#Nesting_Conditions_-OR_.7C.7C
I think the example is not the most clear explanation of the workings of 
'break'. It does use 'break-never' without any action. Even more, am I 
correct stating the provided example doesn't do what one might want to 
accomplish? It seems to me the week days are of no use at all?
I'd like to improve the wiki, problem is I'm not sure I understand the 
working enhough.

regards,
Egbert.















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From philippe at ppmt.org  Sun Mar 20 22:18:24 2011
From: philippe at ppmt.org (Philippe Le Toquin)
Date: Sun, 20 Mar 2011 15:18:24 -0400
Subject: [Freeswitch-users] what to do after "make current"
In-Reply-To: <4d864aaa.08c2640a.2829.507c@mx.google.com>
References: <4d864aaa.08c2640a.2829.507c@mx.google.com>
Message-ID: <4D865300.4030900@ppmt.org>

thanks a lot for the clarification

I actually have a init.d script to restart but I will keep your method
as well

/Philippe

On 11-03-20 02:43 PM, msc at freeswitch.org wrote:
> You can restart freeswitch at your leisure. The new binary won't "hot
> swap" by itself. If you want to do it from the Linux cmd line or shell
> script you can do something like this:
>
> fs_cli -x "fsctl shutdown elegant"
> sleep 20
> freeswitch -nc -nonat
>
> Of course, use a sleep value appropriate for the speed of your server
> and use the FS cmd line args for your config.
>
> -MC
>
> Sent from my HTC on the Now Network from Sprint!
>
> ----- Reply message -----
> From: "Philippe Le Toquin" <philippe at ppmt.org>
> Date: Sat, Mar 19, 2011 1:27 pm
> Subject: [Freeswitch-users] what to do after "make current"
> To: <freeswitch-users at lists.freeswitch.org>
>
> Hello,
>
> This might sound like a stupid question but I can't find the answer so...
>
> TO update Freeswitch I do
>
>  git pull
>
>  and then
>
>  make current
>
> After that I generally restart freeswitch. But do I really need to do
> it? or will it somehow manage to "hot swap" with
> the new binary?
>
> If I need to restart. Do I have to do it right after the update or can
> it wait until it is night time?
>
> Thanks
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From avi at avimarcus.net  Sun Mar 20 22:45:11 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 20 Mar 2011 21:45:11 +0200
Subject: [Freeswitch-users] Understanding break="never" in condition-tag
In-Reply-To: <4D85F6FB.8080801@redhosting.nl>
References: <4D85E0A1.507@redhosting.nl>
	<AANLkTimNYfHOrZrr7Q70Zea8Nsw-xEmQr2z0g-=PK+hv@mail.gmail.com>
	<4D85F6FB.8080801@redhosting.nl>
Message-ID: <AANLkTimXv0PrtnjRPLfi5WFGhpyNmgc4H-Unvr7Bc7Xj@mail.gmail.com>

No, I'm pretty sure that example works as intended.. I created it.
break="never" is NOT "ignore this condition" but rather "keep processing the
extension even if this doesn't match".
So in the example you linked to, if both wday's don't match, it goes to the
"catch-all" at the end.  And if it does match the wday, it still has to
match the rest of the nested conditions to perform the action.
-Avi

On Sun, Mar 20, 2011 at 2:45 PM, [Redhosting] Egbert Groot <
egbert at redhosting.nl> wrote:

> On Sun, Mar 20, 2011 at 2:10 PM, [Redhosting] Egbert Groot<egbert at redhosting.nl> <egbert at redhosting.nl> wrote:
>
>
>    Indeed, with adding an action, it is/can be usefull.
> I was reading in wiki:
> http://wiki.freeswitch.org/wiki/Dialplan_XML#Nesting_Conditions_-OR_.7C.7C
> I think the example is not the most clear explanation of the workings of
> 'break'. It does use 'break-never' without any action. Even more, am I
> correct stating the provided example doesn't do what one might want to
> accomplish? It seems to me the week days are of no use at all?
> I'd like to improve the wiki, problem is I'm not sure I understand the
> working enhough.
>
> regards,
> Egbert.
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From grsingh750 at gmail.com  Sun Mar 20 23:56:13 2011
From: grsingh750 at gmail.com (guru singh)
Date: Mon, 21 Mar 2011 02:26:13 +0530
Subject: [Freeswitch-users] pass variable from dialplan to bridge
In-Reply-To: <4D85B385.4060407@freemail.hu>
References: <4D85B385.4060407@freemail.hu>
Message-ID: <AANLkTinmM2L2YpJUpNKnoe9ib4AtPkOOdfuJji8XT26g@mail.gmail.com>

Hi,

Try export instead of set.
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_export


On Sun, Mar 20, 2011 at 1:27 PM, dunchan <dunchan at freemail.hu> wrote:
> Hi!
>
> I have a simple question, how can i pass a varible from dialplan to the
> bridge?
> I want to fill the 'from-user' field depends on sip user settings.
>
> I've tryed the following in dialplan:
> ...
> some conditions
> ...
> <action application="set" data="from-user=${outbound_caller_id_number}"/>
> <action application="bridge" data="sofia/gateway/upc/$1"/>
>
> in a gateway xml i tryed:
> <gateway name="upc">
> ? ? ? ? <param name="username" value="..."/>
> ? ? ? ? <param name="password" value="..."/>
> ? ? ? ? <param name="proxy" value="..."/>
> ? ? ? ? <param name="register" value="false"/>
> ? ? ? ? <param name="from-user" value="${from-user}"/>
> ? ? </gateway>
>
> Above config ha no effect :(
> any suggestions?
>
> thx,
> Viktor
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From nick.rosier at gmail.com  Mon Mar 21 00:28:17 2011
From: nick.rosier at gmail.com (Nick Rosier)
Date: Sun, 20 Mar 2011 22:28:17 +0100
Subject: [Freeswitch-users] Outbound dialing to SPA3102 behind NAT
In-Reply-To: <9538.1300613898@ccs.covici.com>
References: <AANLkTik_kYxvxfWxxb5cMjohDEHCj7qroomKbeQyOXK_@mail.gmail.com>
	<9538.1300613898@ccs.covici.com>
Message-ID: <AANLkTi=sf9iXbNPLFOSJK6Xy_Tn3=KR74OCSK8=qpESY@mail.gmail.com>

I've using Bluebox to configure the FS. I've defined a gateway as following:

    <gateways>
      <gateway name="trunk1">
        <param name="realm" value=""/>
        <param name="username" value="trunk1"/>
        <param name="password" value="trunk1"/>
        <param name="register" value="false"/>
      </gateway>
    </gateways>

sofia status show the SPA3102:

Call-ID:        abb17a79-760cd2dc at 192.168.1.153
User:           trunk1 at pbx.domain
Contact:        Trunk1
<sip:trunk1@<internal-ip>:5078;transport=tcp;fs_nat=yes;fs_path=sip%3Atrunk1%40<external-ip>%3A5078%3Btransport%3Dtcp>
Agent:          Linksys/SPA3102-5.1.10(GW)
Status:         Registered(TCP-NAT)(unknown) EXP(2011-03-20 23:04:09)
EXPSECS(2795)
Host:           pbx.domain
IP:             <external-ip>
Port:           5078
Auth-User:      unknown
Auth-Realm:     pbx.domain
MWI-Account:    trunk1 at pbx.domain

I can see the dialplan matching but nothing happens; from the log it
looks like the dialing fails instantly:

2011-03-20 22:14:47.713322 [DEBUG] sofia.c:4581 Channel
sofia/sipinterface_1/ entering state [calling][0]
2011-03-20 22:14:47.713322 [DEBUG] sofia.c:4581 Channel
sofia/sipinterface_1/ entering state [terminated][503]

Any ideas what else I need to do?

Rgds,
N.

On 20 March 2011 10:38,  <covici at ccs.covici.com> wrote:
> Only way I can think of is have the spa3102 register as an fs gateway
> using user name and password and bridge to the gateway -- and make sure
> natis enabled on the 3102, so fs can detect its external ip address.
>
> Nick Rosier <nick.rosier at gmail.com> wrote:
>
>> Hi,
>>
>> I've currently got an Asterisk server running at home but want to
>> switch to FS on my external server (located in a DC).
>>
>> Configuration:
>> - 2 SPA3102 connected to 2 PSTN lines and 2 phones (on Line1)
>> - Couple of voip phones
>>
>> The Asterisk server is running on my local network as are all the
>> other devices; everything is connected to a DSL-router with NAT.
>> The SPA3102 are configured as Trunks in Asterisk with the following
>> configuration:
>> allow=ulaw
>> canreinvite=no
>> context=from-trunk
>> disallow=all
>> dtmfmode=rfc2833
>> host=dynamic
>> incominglimit=1
>> nat=never
>> port=5061
>> qualify=yes
>> secret=trunk1
>> username=trunk1
>> type=friend
>>
>> The SPA3102 are configured to register their PSTN-line to the Asterisk
>> server. Outbound routes are configured to use 1 of the 2 Trunks.
>>
>> I want to do the same thing with my FS running on a server in the DC.
>> I've setup all the devices and can make internal calls. Inbound calls
>> on the PSTN lines are also forwarded to the correct voip-devices. I
>> just cannot figure out how to configure outbound dialing to the
>> PSTN-lines. The Wiki
>> (http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo) shows I
>> should configure them as following:
>>
>> <include>
>> ? <extension name="To PSTN">
>> ? ? <condition field="destination_number" expression="(.*)">
>> ? ? ? <action application="bridge"
>> data="sofia/internal/${destination_number}@[IP address of
>> SPA3102]:5061" />
>> ? ? </condition>
>> ? </extension>
>> </include>
>>
>> The problem here is that the SPA3102 are behind my router so the only
>> possibility would be to setup some port forwarding to on the router
>> with different ports for the 2 devices. I would rather not do that so
>> I was wondering if there is an easier/better way to do that.
>>
>> Any ideas?
>>
>> Rgds,
>> N.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> --
> Your life is like a penny. ?You're going to lose it. ?The question is:
> How do
> you spend it?
>
> ? ? ? ? John Covici
> ? ? ? ? covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From avi at avimarcus.net  Mon Mar 21 00:42:43 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 20 Mar 2011 23:42:43 +0200
Subject: [Freeswitch-users] pass variable from dialplan to bridge
In-Reply-To: <AANLkTinmM2L2YpJUpNKnoe9ib4AtPkOOdfuJji8XT26g@mail.gmail.com>
References: <4D85B385.4060407@freemail.hu>
	<AANLkTinmM2L2YpJUpNKnoe9ib4AtPkOOdfuJji8XT26g@mail.gmail.com>
Message-ID: <AANLkTi=HHse+=PVwfPjikqQw-YOkigLGO1C9Jsnx488z@mail.gmail.com>

That doesn't sound right.
I'm not sure exactly what you are trying to do, but I'm guessing you want
the caller ID of this user in the FROM?
So simply set  <param name="caller-id-in-from" value="true"/>
in the gateway, and then "set" "effective_caller_id=
${outbound_caller_id_number}"

-Avi

On Sun, Mar 20, 2011 at 10:56 PM, guru singh <grsingh750 at gmail.com> wrote:

> Hi,
>
> Try export instead of set.
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_export
>
>
> On Sun, Mar 20, 2011 at 1:27 PM, dunchan <dunchan at freemail.hu> wrote:
> > Hi!
> >
> > I have a simple question, how can i pass a varible from dialplan to the
> > bridge?
> > I want to fill the 'from-user' field depends on sip user settings.
> >
> > I've tryed the following in dialplan:
> > ...
> > some conditions
> > ...
> > <action application="set" data="from-user=${outbound_caller_id_number}"/>
> > <action application="bridge" data="sofia/gateway/upc/$1"/>
> >
> > in a gateway xml i tryed:
> > <gateway name="upc">
> >         <param name="username" value="..."/>
> >         <param name="password" value="..."/>
> >         <param name="proxy" value="..."/>
> >         <param name="register" value="false"/>
> >         <param name="from-user" value="${from-user}"/>
> >     </gateway>
> >
> > Above config ha no effect :(
> > any suggestions?
> >
> > thx,
> > Viktor
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Nabble at slickdeals.endjunk.com  Mon Mar 21 03:21:24 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 20 Mar 2011 17:21:24 -0700 (PDT)
Subject: [Freeswitch-users] New FreeSWITCH Module: mod_ladspa
In-Reply-To: <AANLkTimPo4U2bzf+K4ucRq3xZHHDBnjFouhXDd_t+Fzk@mail.gmail.com>
References: <AANLkTimPo4U2bzf+K4ucRq3xZHHDBnjFouhXDd_t+Fzk@mail.gmail.com>
Message-ID: <1300666884615-6190839.post@n2.nabble.com>

Cool. Another reason to use FreeSWITCH.

Just curious, do we have to download and install all plugins mentioned in
the wiki of  http://wiki.freeswitch.org/wiki/Mod_ladspa mod ladspa  just in
order to sound like Cher?

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/New-FreeSWITCH-Module-mod-ladspa-tp6186438p6190839.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From ayhkor at gmail.com  Mon Mar 21 06:44:58 2011
From: ayhkor at gmail.com (deniro)
Date: Sun, 20 Mar 2011 23:44:58 -0400
Subject: [Freeswitch-users] conference dialplan with perl scripting
Message-ID: <AANLkTik_Jqag_APcQv+R13tESbVye_QrMVyMgTmnfyqL@mail.gmail.com>

Hi All,

I'd like to run perl script in dialplan by passing PIN to it
<action application="perl" data="/path/to/your/script.pl $PIN" />

this script is assumed to compare the PIN with the values in database and if
there is a match , let go to the conference room
(with the perl script I am able to compare database values and determine if
there is match for the PIN by manually running it, and I'd like to run this
through dialplan )

so, In the dial plan
1--Where exactly I call this perl script after reading PIN with
"play_and_get_digits"
2-- what should I return from the perl script if there is match or there is
no match for the PIN and pass it to dialplan
3-- if there is a match, how should I proceed to conference
4-- if there is no match how should I terminate(exit) from dial plan

Any guidance, links, documents well appreciated
thans
deniro--
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From chenzhanping at gmail.com  Mon Mar 21 07:38:55 2011
From: chenzhanping at gmail.com (=?GB2312?B?1bzGvbPC?=)
Date: Mon, 21 Mar 2011 12:38:55 +0800
Subject: [Freeswitch-users] Question about mod_dingaling.
In-Reply-To: <AANLkTim-5r58xUao_pVwGAKGBU0cy3Dn-jJ6DXkL6yb3@mail.gmail.com>
References: <AANLkTikd9baOGf0bmL6m3EaiPcGy6Fgfm9=jEYBg+GB2@mail.gmail.com>
	<AANLkTimXmNou2T4tLPMACscdHiHVqUzf3U__WeGVaEJb@mail.gmail.com>
	<AANLkTi=hu9iWu3SQbF5Wj3odkY0ut9HGn0mabsBZJMbQ@mail.gmail.com>
	<AANLkTinanoYB3J-0rmYtM7uw5m5HJC829YmZbxaHWzVh@mail.gmail.com>
	<AANLkTikQnNUSn596FZsm4ZHKi4t_UQMS1JoVfPki+on0@mail.gmail.com>
	<AANLkTikpgKwNmbnd1F_XoUFqu93tcPi3B=vJkE_jRhz5@mail.gmail.com>
	<AANLkTin6E8nKYdx_zn5gLNzoV-oaQpkpmtFDCEDDroGi@mail.gmail.com>
	<AANLkTi=Y-czU6pfGAgrVTvCS=oaM6_VdvSDqFkhyf965@mail.gmail.com>
	<AANLkTikPa22Vh33x+xODysCQo4Q45MFd9X+=xgN8iPis@mail.gmail.com>
	<AANLkTi=Bf89Pae2Qt1Cg1gFhX-xjFHuGz0kv5Dyt9=FK@mail.gmail.com>
	<AANLkTi=BOA1fTfUberjRLzLu-Uts1xvp7Xr4-CzE1xrD@mail.gmail.com>
	<AANLkTik6rKyn6HH8x_9Q=1-O1K-Ax6dxEptwueD8Up-w@mail.gmail.com>
	<45842189-1297744129-cardhu_decombobulator_blackberry.rim.net-1469059183-@b25.c2.bise3.blackberry>
	<AANLkTim-5r58xUao_pVwGAKGBU0cy3Dn-jJ6DXkL6yb3@mail.gmail.com>
Message-ID: <AANLkTimgDNmzd8K853eEUEMbRQgChhgjomrqMurSyVRc@mail.gmail.com>

Now , I install new FS from GIT, the mod_dingaling is bad to work.

I dial use x-lite login my FS server, but cann't connected called.

this is FS log:
http://pastebin.freeswitch.org/15763

What is wrong with this, Thanks.
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From dmitry.bely at gmail.com  Mon Mar 21 09:59:28 2011
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Mon, 21 Mar 2011 09:59:28 +0300
Subject: [Freeswitch-users] Understanding break="never" in condition-tag
In-Reply-To: <AANLkTimXv0PrtnjRPLfi5WFGhpyNmgc4H-Unvr7Bc7Xj@mail.gmail.com>
References: <4D85E0A1.507@redhosting.nl>
	<AANLkTimNYfHOrZrr7Q70Zea8Nsw-xEmQr2z0g-=PK+hv@mail.gmail.com>
	<4D85F6FB.8080801@redhosting.nl>
	<AANLkTimXv0PrtnjRPLfi5WFGhpyNmgc4H-Unvr7Bc7Xj@mail.gmail.com>
Message-ID: <AANLkTimi5pAW-kJif8TVYXKugjbaK18J6hiBtJ712_q4@mail.gmail.com>

On Sun, Mar 20, 2011 at 10:45 PM, Avi Marcus <avi at avimarcus.net> wrote:
> No, I'm pretty sure that example works as intended.. I created it

Really? Correct me if I'm wrong but for wday=1 and hour=9 an action in

	<condition wday="6" break="never"/>
	<condition hour="8-12" break="on-true">
		<action application="transfer" data="1105 XML default"/>
	</condition>

will be executed and further processing will be stopped. Actually the
first condition is a sort of (misleading) comment here.

- Dmitry Bely


From avi at avimarcus.net  Mon Mar 21 11:13:11 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 21 Mar 2011 10:13:11 +0200
Subject: [Freeswitch-users] Understanding break="never" in condition-tag
In-Reply-To: <AANLkTimi5pAW-kJif8TVYXKugjbaK18J6hiBtJ712_q4@mail.gmail.com>
References: <4D85E0A1.507@redhosting.nl>
	<AANLkTimNYfHOrZrr7Q70Zea8Nsw-xEmQr2z0g-=PK+hv@mail.gmail.com>
	<4D85F6FB.8080801@redhosting.nl>
	<AANLkTimXv0PrtnjRPLfi5WFGhpyNmgc4H-Unvr7Bc7Xj@mail.gmail.com>
	<AANLkTimi5pAW-kJif8TVYXKugjbaK18J6hiBtJ712_q4@mail.gmail.com>
Message-ID: <AANLkTikxwL9EHQ984242msZ4kG0wD9pDe8o6AYyOuGcf@mail.gmail.com>

We have two items going on - extensions, and condition-groups within the
extension.
For wday=1, if it was set to on-false - the default - it would end
evaluating this extension and move on to the next.
However, since it says "break=never" it moves to evaluate the *next*
*condition-group* rather than executing or skipping the extension entirely.
-Avi

On Mon, Mar 21, 2011 at 8:59 AM, Dmitry Bely <dmitry.bely at gmail.com> wrote:

> On Sun, Mar 20, 2011 at 10:45 PM, Avi Marcus <avi at avimarcus.net> wrote:
> > No, I'm pretty sure that example works as intended.. I created it
>
> Really? Correct me if I'm wrong but for wday=1 and hour=9 an action in
>
>        <condition wday="6" break="never"/>
>        <condition hour="8-12" break="on-true">
>                <action application="transfer" data="1105 XML default"/>
>        </condition>
>
> will be executed and further processing will be stopped. Actually the
> first condition is a sort of (misleading) comment here.
>
> - Dmitry Bely
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From avi at avimarcus.net  Mon Mar 21 11:18:10 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 21 Mar 2011 10:18:10 +0200
Subject: [Freeswitch-users] Understanding break="never" in condition-tag
In-Reply-To: <AANLkTikxwL9EHQ984242msZ4kG0wD9pDe8o6AYyOuGcf@mail.gmail.com>
References: <4D85E0A1.507@redhosting.nl>
	<AANLkTimNYfHOrZrr7Q70Zea8Nsw-xEmQr2z0g-=PK+hv@mail.gmail.com>
	<4D85F6FB.8080801@redhosting.nl>
	<AANLkTimXv0PrtnjRPLfi5WFGhpyNmgc4H-Unvr7Bc7Xj@mail.gmail.com>
	<AANLkTimi5pAW-kJif8TVYXKugjbaK18J6hiBtJ712_q4@mail.gmail.com>
	<AANLkTikxwL9EHQ984242msZ4kG0wD9pDe8o6AYyOuGcf@mail.gmail.com>
Message-ID: <AANLkTikc0a4ktzy3tNZHNbpNibSAChY5H3avumNSgaOT@mail.gmail.com>

NOTE: break "on-true, on-false, never" don't change the *condition*, merely
the *hunting*. You need ALL the conditions to match for freeswitch to do the
action, and you can't change that!
The break is how to handle failure - "Does 1 failure to match
a condition mean should move on to the next condition extension (on-false)
or keep evaluating this extension? (never)"
-Avi

On Mon, Mar 21, 2011 at 10:13 AM, Avi Marcus <avi at avimarcus.net> wrote:

> We have two items going on - extensions, and condition-groups within the
> extension.
> For wday=1, if it was set to on-false - the default - it would end
> evaluating this extension and move on to the next.
> However, since it says "break=never" it moves to evaluate the *next*
> *condition-group* rather than executing or skipping the extension entirely.
> -Avi
>
>
> On Mon, Mar 21, 2011 at 8:59 AM, Dmitry Bely <dmitry.bely at gmail.com>wrote:
>
>> On Sun, Mar 20, 2011 at 10:45 PM, Avi Marcus <avi at avimarcus.net> wrote:
>> > No, I'm pretty sure that example works as intended.. I created it
>>
>> Really? Correct me if I'm wrong but for wday=1 and hour=9 an action in
>>
>>        <condition wday="6" break="never"/>
>>        <condition hour="8-12" break="on-true">
>>                <action application="transfer" data="1105 XML default"/>
>>        </condition>
>>
>> will be executed and further processing will be stopped. Actually the
>> first condition is a sort of (misleading) comment here.
>>
>> - Dmitry Bely
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From dmitry.bely at gmail.com  Mon Mar 21 12:40:34 2011
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Mon, 21 Mar 2011 12:40:34 +0300
Subject: [Freeswitch-users] Understanding break="never" in condition-tag
In-Reply-To: <AANLkTikc0a4ktzy3tNZHNbpNibSAChY5H3avumNSgaOT@mail.gmail.com>
References: <4D85E0A1.507@redhosting.nl>
	<AANLkTimNYfHOrZrr7Q70Zea8Nsw-xEmQr2z0g-=PK+hv@mail.gmail.com>
	<4D85F6FB.8080801@redhosting.nl>
	<AANLkTimXv0PrtnjRPLfi5WFGhpyNmgc4H-Unvr7Bc7Xj@mail.gmail.com>
	<AANLkTimi5pAW-kJif8TVYXKugjbaK18J6hiBtJ712_q4@mail.gmail.com>
	<AANLkTikxwL9EHQ984242msZ4kG0wD9pDe8o6AYyOuGcf@mail.gmail.com>
	<AANLkTikc0a4ktzy3tNZHNbpNibSAChY5H3avumNSgaOT@mail.gmail.com>
Message-ID: <AANLkTikbejGE2ku-X8VYjc=HctHCuSDDozUdusaKDFr4@mail.gmail.com>

On Mon, Mar 21, 2011 at 11:18 AM, Avi Marcus <avi at avimarcus.net> wrote:
> NOTE: break "on-true, on-false, never" don't change the *condition*, merely
> the *hunting*.

If wday=1 and  hour=9, will transfer to 1105 be executed in your
example? If you think it won't, why?

> You need ALL the conditions to match for freeswitch to do the action, and you can't
> change that!

I don't think so. If some condition evaluates to true, its nested
actions are always executed no matter what preceded it. But of course
the condition should be evaluated first, that's where various break
values come into play.

> The break is how to handle failure - "Does 1 failure to match
> a?condition?mean should move on to the next condition extension (on-false)
> or keep evaluating this extension? (never)"
> -Avi
>
> On Mon, Mar 21, 2011 at 10:13 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>> We have two items going on - extensions, and condition-groups within the
>> extension.
>> For wday=1, if it was set to on-false - the default - it would end
>> evaluating this extension and move on to the next.
>> However, since it says "break=never" it moves to evaluate the *next*
>> *condition-group* rather than executing or skipping the extension entirely.
>> -Avi
>>
>> On Mon, Mar 21, 2011 at 8:59 AM, Dmitry Bely <dmitry.bely at gmail.com>
>> wrote:
>>>
>>> On Sun, Mar 20, 2011 at 10:45 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>> > No, I'm pretty sure that example works as intended.. I created it
>>>
>>> Really? Correct me if I'm wrong but for wday=1 and hour=9 an action in
>>>
>>> ? ? ? ?<condition wday="6" break="never"/>
>>> ? ? ? ?<condition hour="8-12" break="on-true">
>>> ? ? ? ? ? ? ? ?<action application="transfer" data="1105 XML default"/>
>>> ? ? ? ?</condition>
>>>
>>> will be executed and further processing will be stopped. Actually the
>>> first condition is a sort of (misleading) comment here.

- Dmitry Bely


From avi at avimarcus.net  Mon Mar 21 13:01:42 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 21 Mar 2011 12:01:42 +0200
Subject: [Freeswitch-users] Understanding break="never" in condition-tag
In-Reply-To: <AANLkTikbejGE2ku-X8VYjc=HctHCuSDDozUdusaKDFr4@mail.gmail.com>
References: <4D85E0A1.507@redhosting.nl>
	<AANLkTimNYfHOrZrr7Q70Zea8Nsw-xEmQr2z0g-=PK+hv@mail.gmail.com>
	<4D85F6FB.8080801@redhosting.nl>
	<AANLkTimXv0PrtnjRPLfi5WFGhpyNmgc4H-Unvr7Bc7Xj@mail.gmail.com>
	<AANLkTimi5pAW-kJif8TVYXKugjbaK18J6hiBtJ712_q4@mail.gmail.com>
	<AANLkTikxwL9EHQ984242msZ4kG0wD9pDe8o6AYyOuGcf@mail.gmail.com>
	<AANLkTikc0a4ktzy3tNZHNbpNibSAChY5H3avumNSgaOT@mail.gmail.com>
	<AANLkTikbejGE2ku-X8VYjc=HctHCuSDDozUdusaKDFr4@mail.gmail.com>
Message-ID: <AANLkTinty+efpatHL7eu79H5SuAjYdfVxA6o_O_A3rC+@mail.gmail.com>

<extension name="YLE">
	<condition field="destination_number" expression="^\*1105$"
break="on-false"/> <--if this is false, FreeSWITCH skips to the next
extension.
	<condition wday="6" break="never"/> <--The default "on-false" would
make FreeSWITCH move to the next extension, rather than evaluating the
next condition set.
	<condition hour="8-12" break="on-true"> <--Don't evaluate the next
condition set if this is true.
		<action application="transfer" data="1105 XML default"/>
	</condition>



	<condition wday="1-5" break="never"/>
	<condition hour="8-21" break="on-true">
		<action application="transfer" data="1105 XML default"/>
	</condition>



	<condition> <-- this is a catch all. If they didn't dial *1105, they
aren't here anymore.
		<action application="voicemail" data="default ${domain} 1105"/>
	</condition>

   </extension>


On Mon, Mar 21, 2011 at 11:40 AM, Dmitry Bely <dmitry.bely at gmail.com> wrote:

> On Mon, Mar 21, 2011 at 11:18 AM, Avi Marcus <avi at avimarcus.net> wrote:
> > NOTE: break "on-true, on-false, never" don't change the *condition*,
> merely
> > the *hunting*.
>
> If wday=1 and  hour=9, will transfer to 1105 be executed in your
> example? If you think it won't, why?
>
It will go to 1105 - from the second condition-group. But the first
condition-group fails. I'm pretty sure I've seen it in the logs work as I'm
describing..




>
> > You need ALL the conditions to match for freeswitch to do the action, and
> you can't
> > change that!
>
> I don't think so. If some condition evaluates to true, its nested
> actions are always executed no matter what preceded it. But of course
> the condition should be evaluated first, that's where various break
> values come into play.
>
No way - the dialplan is always run on a "match all conditions" to do the
actions! How else would multiple conditions EVER work?

This is a slight edge-case though - the on-true, on-fail, never.. they only
matter when you want MULTIPLE condition-groups within ONE extension. If you
specify your settings as multiple serial extensions, then this stuff doesn't
really matter...

-Avi


> > The break is how to handle failure - "Does 1 failure to match
> > a condition mean should move on to the next condition extension
> (on-false)
> > or keep evaluating this extension? (never)"
> > -Avi
> >
> > On Mon, Mar 21, 2011 at 10:13 AM, Avi Marcus <avi at avimarcus.net> wrote:
> >>
> >> We have two items going on - extensions, and condition-groups within the
> >> extension.
> >> For wday=1, if it was set to on-false - the default - it would end
> >> evaluating this extension and move on to the next.
> >> However, since it says "break=never" it moves to evaluate the *next*
> >> *condition-group* rather than executing or skipping the extension
> entirely.
> >> -Avi
> >>
> >> On Mon, Mar 21, 2011 at 8:59 AM, Dmitry Bely <dmitry.bely at gmail.com>
> >> wrote:
> >>>
> >>> On Sun, Mar 20, 2011 at 10:45 PM, Avi Marcus <avi at avimarcus.net>
> wrote:
> >>> > No, I'm pretty sure that example works as intended.. I created it
> >>>
> >>> Really? Correct me if I'm wrong but for wday=1 and hour=9 an action in
> >>>
> >>>        <condition wday="6" break="never"/>
> >>>        <condition hour="8-12" break="on-true">
> >>>                <action application="transfer" data="1105 XML default"/>
> >>>        </condition>
> >>>
> >>> will be executed and further processing will be stopped. Actually the
> >>> first condition is a sort of (misleading) comment here.
>
> - Dmitry Bely
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From dmitry.bely at gmail.com  Mon Mar 21 13:44:54 2011
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Mon, 21 Mar 2011 13:44:54 +0300
Subject: [Freeswitch-users] Understanding break="never" in condition-tag
In-Reply-To: <AANLkTinty+efpatHL7eu79H5SuAjYdfVxA6o_O_A3rC+@mail.gmail.com>
References: <4D85E0A1.507@redhosting.nl>
	<AANLkTimNYfHOrZrr7Q70Zea8Nsw-xEmQr2z0g-=PK+hv@mail.gmail.com>
	<4D85F6FB.8080801@redhosting.nl>
	<AANLkTimXv0PrtnjRPLfi5WFGhpyNmgc4H-Unvr7Bc7Xj@mail.gmail.com>
	<AANLkTimi5pAW-kJif8TVYXKugjbaK18J6hiBtJ712_q4@mail.gmail.com>
	<AANLkTikxwL9EHQ984242msZ4kG0wD9pDe8o6AYyOuGcf@mail.gmail.com>
	<AANLkTikc0a4ktzy3tNZHNbpNibSAChY5H3avumNSgaOT@mail.gmail.com>
	<AANLkTikbejGE2ku-X8VYjc=HctHCuSDDozUdusaKDFr4@mail.gmail.com>
	<AANLkTinty+efpatHL7eu79H5SuAjYdfVxA6o_O_A3rC+@mail.gmail.com>
Message-ID: <AANLkTi=4Sqb-m+t-o81OYJwK=UTD17-fkcHtG7oe0LDN@mail.gmail.com>

On Mon, Mar 21, 2011 at 1:01 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> > You need ALL the conditions to match for freeswitch to do the action,
>> > and you can't
>> > change that!
>>
>> I don't think so. If some condition evaluates to true, its nested
>> actions are always executed no matter what preceded it. But of course
>> the condition should be evaluated first, that's where various break
>> values come into play.
>
> No way - the dialplan is always run on a "match all conditions" to do the
> actions! How else would multiple conditions EVER work?

Completely wrong. And I don't see relevant logs in your message. Here
is mine. A dialplan fragment:

    <extension name="condition-test">
        <condition wday="6" break="never"/>
        <condition hour="9-18" break="on-true">
            <action application="info"/>
        </condition>
    </extension>

log:

Dialplan: sofia/internal/1000 at 192.168.121.66 parsing
[default->condition-test] continue=false
Dialplan: sofia/internal/1000 at 192.168.121.66 Date/Time Match (FAIL)
[condition-test] break=never
Dialplan: sofia/internal/1000 at 192.168.121.66 Date/Time Match (PASS)
[condition-test] break=on-true
Dialplan: sofia/internal/1000 at 192.168.121.66 Action info()

Guess what? The first condition fails but the action is still
scheduled for execution (and indeed executed later)

- Dmitry Bely


From avi at avimarcus.net  Mon Mar 21 15:08:27 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 21 Mar 2011 14:08:27 +0200
Subject: [Freeswitch-users] Understanding break="never" in condition-tag
In-Reply-To: <AANLkTi=4Sqb-m+t-o81OYJwK=UTD17-fkcHtG7oe0LDN@mail.gmail.com>
References: <4D85E0A1.507@redhosting.nl>
	<AANLkTimNYfHOrZrr7Q70Zea8Nsw-xEmQr2z0g-=PK+hv@mail.gmail.com>
	<4D85F6FB.8080801@redhosting.nl>
	<AANLkTimXv0PrtnjRPLfi5WFGhpyNmgc4H-Unvr7Bc7Xj@mail.gmail.com>
	<AANLkTimi5pAW-kJif8TVYXKugjbaK18J6hiBtJ712_q4@mail.gmail.com>
	<AANLkTikxwL9EHQ984242msZ4kG0wD9pDe8o6AYyOuGcf@mail.gmail.com>
	<AANLkTikc0a4ktzy3tNZHNbpNibSAChY5H3avumNSgaOT@mail.gmail.com>
	<AANLkTikbejGE2ku-X8VYjc=HctHCuSDDozUdusaKDFr4@mail.gmail.com>
	<AANLkTinty+efpatHL7eu79H5SuAjYdfVxA6o_O_A3rC+@mail.gmail.com>
	<AANLkTi=4Sqb-m+t-o81OYJwK=UTD17-fkcHtG7oe0LDN@mail.gmail.com>
Message-ID: <AANLkTinJFMMMBVrLPsAeuKOApECywnCBkTgkMbnjTFzW@mail.gmail.com>

Hmm, looking at your log and mine from the past, you appear to be correct.
It looks like break=never means to ignore the condition. So then.. I'm left
with the original question - what's the point?
And another - how can you do groups of conditions, like I intended to show
in the example? I tried setting it to break on-false, or break on-true, but
I can never get to the second condition set. Am I supposed to create it as
separate extensions?

On Mon, Mar 21, 2011 at 12:44 PM, Dmitry Bely <dmitry.bely at gmail.com> wrote:

> On Mon, Mar 21, 2011 at 1:01 PM, Avi Marcus <avi at avimarcus.net> wrote:
> >
> >> > You need ALL the conditions to match for freeswitch to do the action,
> >> > and you can't
> >> > change that!
> >>
> >> I don't think so. If some condition evaluates to true, its nested
> >> actions are always executed no matter what preceded it. But of course
> >> the condition should be evaluated first, that's where various break
> >> values come into play.
> >
> > No way - the dialplan is always run on a "match all conditions" to do the
> > actions! How else would multiple conditions EVER work?
>
> Completely wrong. And I don't see relevant logs in your message. Here
> is mine. A dialplan fragment:
>
>    <extension name="condition-test">
>         <condition wday="6" break="never"/>
>         <condition hour="9-18" break="on-true">
>            <action application="info"/>
>        </condition>
>    </extension>
>
> log:
>
> Dialplan: sofia/internal/1000 at 192.168.121.66 parsing
> [default->condition-test] continue=false
> Dialplan: sofia/internal/1000 at 192.168.121.66 Date/Time Match (FAIL)
> [condition-test] break=never
> Dialplan: sofia/internal/1000 at 192.168.121.66 Date/Time Match (PASS)
> [condition-test] break=on-true
> Dialplan: sofia/internal/1000 at 192.168.121.66 Action info()
>
> Guess what? The first condition fails but the action is still
> scheduled for execution (and indeed executed later)
>
> - Dmitry Bely
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dmitry.bely at gmail.com  Mon Mar 21 15:53:19 2011
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Mon, 21 Mar 2011 15:53:19 +0300
Subject: [Freeswitch-users] Understanding break="never" in condition-tag
In-Reply-To: <AANLkTinJFMMMBVrLPsAeuKOApECywnCBkTgkMbnjTFzW@mail.gmail.com>
References: <4D85E0A1.507@redhosting.nl>
	<AANLkTimNYfHOrZrr7Q70Zea8Nsw-xEmQr2z0g-=PK+hv@mail.gmail.com>
	<4D85F6FB.8080801@redhosting.nl>
	<AANLkTimXv0PrtnjRPLfi5WFGhpyNmgc4H-Unvr7Bc7Xj@mail.gmail.com>
	<AANLkTimi5pAW-kJif8TVYXKugjbaK18J6hiBtJ712_q4@mail.gmail.com>
	<AANLkTikxwL9EHQ984242msZ4kG0wD9pDe8o6AYyOuGcf@mail.gmail.com>
	<AANLkTikc0a4ktzy3tNZHNbpNibSAChY5H3avumNSgaOT@mail.gmail.com>
	<AANLkTikbejGE2ku-X8VYjc=HctHCuSDDozUdusaKDFr4@mail.gmail.com>
	<AANLkTinty+efpatHL7eu79H5SuAjYdfVxA6o_O_A3rC+@mail.gmail.com>
	<AANLkTi=4Sqb-m+t-o81OYJwK=UTD17-fkcHtG7oe0LDN@mail.gmail.com>
	<AANLkTinJFMMMBVrLPsAeuKOApECywnCBkTgkMbnjTFzW@mail.gmail.com>
Message-ID: <AANLkTikUn_Mdux6XQWx5ouQwEBc7j8uAEPricD==CM4T@mail.gmail.com>

On Mon, Mar 21, 2011 at 3:08 PM, Avi Marcus <avi at avimarcus.net> wrote:
> Hmm, looking at your log and mine from the past, you appear to be correct.
> It looks like break=never means to ignore the condition. So then.. I'm left
> with the original question - what's the point?

I can only replicate my answer to the original poster:

It makes sense if you use the form

<condition wday="1" break="never">
 (some actions)
</condition>

> And another - how can you do groups of conditions, like I intended to show
> in the example? I tried setting it to break on-false, or break on-true, but
> I can never get to the second condition set. Am I supposed to create it as
> separate extensions?

Probably so.

> On Mon, Mar 21, 2011 at 12:44 PM, Dmitry Bely <dmitry.bely at gmail.com> wrote:
>>
>> On Mon, Mar 21, 2011 at 1:01 PM, Avi Marcus <avi at avimarcus.net> wrote:
>> >
>> >> > You need ALL the conditions to match for freeswitch to do the action,
>> >> > and you can't
>> >> > change that!
>> >>
>> >> I don't think so. If some condition evaluates to true, its nested
>> >> actions are always executed no matter what preceded it. But of course
>> >> the condition should be evaluated first, that's where various break
>> >> values come into play.
>> >
>> > No way - the dialplan is always run on a "match all conditions" to do
>> > the
>> > actions! How else would multiple conditions EVER work?
>>
>> Completely wrong. And I don't see relevant logs in your message. Here
>> is mine. A dialplan fragment:
>>
>> ? ?<extension name="condition-test">
>> ? ? ? ?<condition wday="6" break="never"/>
>> ? ? ? ?<condition hour="9-18" break="on-true">
>> ? ? ? ? ? ?<action application="info"/>
>> ? ? ? ?</condition>
>> ? ?</extension>
>>
>> log:
>>
>> Dialplan: sofia/internal/1000 at 192.168.121.66 parsing
>> [default->condition-test] continue=false
>> Dialplan: sofia/internal/1000 at 192.168.121.66 Date/Time Match (FAIL)
>> [condition-test] break=never
>> Dialplan: sofia/internal/1000 at 192.168.121.66 Date/Time Match (PASS)
>> [condition-test] break=on-true
>> Dialplan: sofia/internal/1000 at 192.168.121.66 Action info()
>>
>> Guess what? The first condition fails but the action is still
>> scheduled for execution (and indeed executed later)

- Dmitry Bely


From chris at fowler.cc  Mon Mar 21 19:51:39 2011
From: chris at fowler.cc (Chris Fowler)
Date: Mon, 21 Mar 2011 12:51:39 -0400
Subject: [Freeswitch-users] Recommendations for multi-channel provider of
	Singapore DID's
Message-ID: <7454A296C7EDE34EA57199FAA401E2F121ADDA9A94@VMBX113.ihostexchange.net>

Hi,



We use InPhonex for International DID's.  Unfortunately in Singapore they are unable to provide DID's with more than 2 Channels.



Can anyone recommend a provider that can provide DID's with more than two channels in Singapore (we need 4 - 20 Channel DID's)?



Thanks, Chris.

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From infos at madovsky.org  Mon Mar 21 20:20:11 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 21 Mar 2011 13:20:11 -0400
Subject: [Freeswitch-users] Recommendations for multi-channel provider
	ofSingapore DID's
References: <7454A296C7EDE34EA57199FAA401E2F121ADDA9A94@VMBX113.ihostexchange.net>
Message-ID: <062C15F35FC84A2997EA12C08FF63E38@e1705>

didx dot net
  ----- Original Message ----- 
  From: Chris Fowler 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 21, 2011 12:51 PM
  Subject: [Freeswitch-users] Recommendations for multi-channel provider ofSingapore DID's


  Hi,

   

  We use InPhonex for International DID's.  Unfortunately in Singapore they are unable to provide DID's with more than 2 Channels.

   

  Can anyone recommend a provider that can provide DID's with more than two channels in Singapore (we need 4 - 20 Channel DID's)?

   

  Thanks, Chris.

   



------------------------------------------------------------------------------


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From infos at madovsky.org  Mon Mar 21 20:31:46 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 21 Mar 2011 13:31:46 -0400
Subject: [Freeswitch-users] Understanding break="never" in condition-tag
References: <4D85E0A1.507@redhosting.nl><AANLkTimNYfHOrZrr7Q70Zea8Nsw-xEmQr2z0g-=PK+hv@mail.gmail.com><4D85F6FB.8080801@redhosting.nl><AANLkTimXv0PrtnjRPLfi5WFGhpyNmgc4H-Unvr7Bc7Xj@mail.gmail.com><AANLkTimi5pAW-kJif8TVYXKugjbaK18J6hiBtJ712_q4@mail.gmail.com><AANLkTikxwL9EHQ984242msZ4kG0wD9pDe8o6AYyOuGcf@mail.gmail.com><AANLkTikc0a4ktzy3tNZHNbpNibSAChY5H3avumNSgaOT@mail.gmail.com><AANLkTikbejGE2ku-X8VYjc=HctHCuSDDozUdusaKDFr4@mail.gmail.com><AANLkTinty+efpatHL7eu79H5SuAjYdfVxA6o_O_A3rC+@mail.gmail.com>
	<AANLkTi=4Sqb-m+t-o81OYJwK=UTD17-fkcHtG7oe0LDN@mail.gmail.com>
Message-ID: <FBC25CDDE4E04DC8BD50B52DC9E3590C@e1705>

condition can't be inclusive,
only stacked

----- Original Message ----- 
From: "Dmitry Bely" <dmitry.bely at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, March 21, 2011 6:44 AM
Subject: Re: [Freeswitch-users] Understanding break="never" in condition-tag


> On Mon, Mar 21, 2011 at 1:01 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> > You need ALL the conditions to match for freeswitch to do the action,
>>> > and you can't
>>> > change that!
>>>
>>> I don't think so. If some condition evaluates to true, its nested
>>> actions are always executed no matter what preceded it. But of course
>>> the condition should be evaluated first, that's where various break
>>> values come into play.
>>
>> No way - the dialplan is always run on a "match all conditions" to do the
>> actions! How else would multiple conditions EVER work?
>
> Completely wrong. And I don't see relevant logs in your message. Here
> is mine. A dialplan fragment:
>
>    <extension name="condition-test">
>        <condition wday="6" break="never"/>
>        <condition hour="9-18" break="on-true">
>            <action application="info"/>
>        </condition>
>    </extension>
>
> log:
>
> Dialplan: sofia/internal/1000 at 192.168.121.66 parsing
> [default->condition-test] continue=false
> Dialplan: sofia/internal/1000 at 192.168.121.66 Date/Time Match (FAIL)
> [condition-test] break=never
> Dialplan: sofia/internal/1000 at 192.168.121.66 Date/Time Match (PASS)
> [condition-test] break=on-true
> Dialplan: sofia/internal/1000 at 192.168.121.66 Action info()
>
> Guess what? The first condition fails but the action is still
> scheduled for execution (and indeed executed later)
>
> - Dmitry Bely
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From Nabble at slickdeals.endjunk.com  Mon Mar 21 20:32:37 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 21 Mar 2011 10:32:37 -0700 (PDT)
Subject: [Freeswitch-users] Recommendations for multi-channel provider
	of Singapore DID's
In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F121ADDA9A94@VMBX113.ihostexchange.net>
References: <7454A296C7EDE34EA57199FAA401E2F121ADDA9A94@VMBX113.ihostexchange.net>
Message-ID: <1300728757304-6193169.post@n2.nabble.com>

I used to use  http://www.pfingo.com/home/index.jsp PFingo  many years ago
(perhaps in 2007) when it was just started and gave out free accounts with
free SG DID# for 1 year with free I/O PSTN calls. IIRC, the call quality was
good. That's all I could say about it.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Recommendations-for-multi-channel-provider-of-Singapore-DID-s-tp6193027p6193169.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From randy.andrade at gmail.com  Mon Mar 21 20:53:01 2011
From: randy.andrade at gmail.com (Randy Andrade)
Date: Mon, 21 Mar 2011 13:53:01 -0400
Subject: [Freeswitch-users] I need to register all of my DIDs with a single
	SIP trunk..
Message-ID: <AANLkTikS8rKHBoXxARwG6=qBUqvyNtxBj6d-qpt3-Sg_@mail.gmail.com>

Greetings,

I'm hoping someone can help me out here. I'm relatively new to FS, but not
to general telephony and I have a small amount of SIP experience.

My FS installation is, so far, a fairly straightforward PBX setup with a
single SIP trunk with 21 DIDs to the ITSP that I work at. I have everything
working as far as inbound and outbound calls, except that in order for me to
use a DID as the outbound caller id, that DID needs to be explicitly
registered with the ITSP, and not just the pilot number for the SIP trunk.
When I try to create a second gateway *.xml file to register a second DID, I
get an error "Registration Failed with status Operation has no matching
challenge [904]" on the FS console. As I mentioned, I work at this ITSP and
I program similar SIP trunks into Adtran TA9xx boxes for SIP over T1
deliveries. On the Adtrans, I'm able to define a single SIP trunk, and have
multiple register statements within that trunk, one for each DID. I'm
wondering if there's a way to do something similar in FS.

I'm running a very recent (I update from the GIT tree about twice a week)
version of FS on a Debian Linux OS, and, I have several local extensions
working and able to call in and out; I just want to be able to outpulse
various DIDs as my caller-id without getting calls rejected. Here's a copy
of my gateway .xml file, with relevant information changed:

<include>
  <gateway name="gatewayname">
  <!--/// account username *required* ///-->
  <param name="username" value="1234567890"/>
  <!--/// auth realm: *optional* same as gateway name, if blank ///-->
  <!--<param name="realm" value="asterlink.com"/>-->
  <!--/// username to use in from: *optional* same as  username, if blank
///-->
  <!--<param name="from-user" value="cluecon"/>-->
  <!--/// domain to use in from: *optional* same as  realm, if blank ///-->
  <param name="from-domain" value="gatewayname"/>
  <!--/// account password *required* ///-->
  <param name="password" value="XXXXXXXX"/>
  <!--/// extension for inbound calls: *optional* same as username, if blank
///-->
  <!--<param name="extension" value="cluecon"/>-->
  <!--/// proxy host: *optional* same as realm, if blank ///-->
  <param name="proxy" value="proxyname"/>
  <!--/// send register to this proxy: *optional* same as proxy, if blank
///-->
  <!--<param name="register-proxy" value="mysbc.com"/>-->
  <!--/// expire in seconds: *optional* 3600, if blank ///-->
  <!--<param name="expire-seconds" value="60"/>-->
  <!--/// do not register ///-->
  <param name="register" value="true"/>
  <!-- which transport to use for register -->
  <!--<param name="register-transport" value="tcp"/>-->
  <!--How many seconds before a retry when a failure or timeout occurs -->
  <!--<param name="retry-seconds" value="30"/>-->
  <!--Use the callerid of an inbound call in the from field on outbound
calls via this gateway -->
  <param name="caller-id-in-from" value="true"/>
  <!--extra sip params to send in the contact-->
  <!--<param name="contact-params" value="tport=tcp"/>-->
  <!--send an options ping every x seconds, failure will unregister and/or
mark it down-->
  <param name="ping" value="30"/>
  <param name="extension-in-contact" value="true"/>
  </gateway>
</include>


Again, I don't know if I would do it here (I would assume so) but I'm
wondering if there's some way (maybe multiple "extension" statements? I
dunno..) to register multiple DIDs in the same gateway / SIP trunk.

Thanks!
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From egbert at redhosting.nl  Mon Mar 21 17:41:03 2011
From: egbert at redhosting.nl ([Redhosting] Egbert Groot)
Date: Mon, 21 Mar 2011 15:41:03 +0100
Subject: [Freeswitch-users] Understanding break="never" in condition-tag
In-Reply-To: <AANLkTikUn_Mdux6XQWx5ouQwEBc7j8uAEPricD==CM4T@mail.gmail.com>
References: <4D85E0A1.507@redhosting.nl>	<AANLkTimNYfHOrZrr7Q70Zea8Nsw-xEmQr2z0g-=PK+hv@mail.gmail.com>	<4D85F6FB.8080801@redhosting.nl>	<AANLkTimXv0PrtnjRPLfi5WFGhpyNmgc4H-Unvr7Bc7Xj@mail.gmail.com>	<AANLkTimi5pAW-kJif8TVYXKugjbaK18J6hiBtJ712_q4@mail.gmail.com>	<AANLkTikxwL9EHQ984242msZ4kG0wD9pDe8o6AYyOuGcf@mail.gmail.com>	<AANLkTikc0a4ktzy3tNZHNbpNibSAChY5H3avumNSgaOT@mail.gmail.com>	<AANLkTikbejGE2ku-X8VYjc=HctHCuSDDozUdusaKDFr4@mail.gmail.com>	<AANLkTinty+efpatHL7eu79H5SuAjYdfVxA6o_O_A3rC+@mail.gmail.com>	<AANLkTi=4Sqb-m+t-o81OYJwK=UTD17-fkcHtG7oe0LDN@mail.gmail.com>	<AANLkTinJFMMMBVrLPsAeuKOApECywnCBkTgkMbnjTFzW@mail.gmail.com>
	<AANLkTikUn_Mdux6XQWx5ouQwEBc7j8uAEPricD==CM4T@mail.gmail.com>
Message-ID: <4D87637F.3090309@redhosting.nl>

Dmitry Bely schreef:
> On Mon, Mar 21, 2011 at 3:08 PM, Avi Marcus <avi at avimarcus.net> wrote:
>   
>> Hmm, looking at your log and mine from the past, you appear to be correct.
>> It looks like break=never means to ignore the condition. So then.. I'm left
>> with the original question - what's the point?
>>     
>
> I can only replicate my answer to the original poster:
>
> It makes sense if you use the form
>
> <condition wday="1" break="never">
>  (some actions)
> </condition>
>
>   
>> And another - how can you do groups of conditions, like I intended to show
>> in the example? I tried setting it to break on-false, or break on-true, but
>> I can never get to the second condition set. Am I supposed to create it as
>> separate extensions?
>>     
>
> Probably so.
>
>   
>> On Mon, Mar 21, 2011 at 12:44 PM, Dmitry Bely <dmitry.bely at gmail.com> wrote:
>>     
>>> On Mon, Mar 21, 2011 at 1:01 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>>       
>>>>>> You need ALL the conditions to match for freeswitch to do the action,
>>>>>> and you can't
>>>>>> change that!
>>>>>>             
>>>>> I don't think so. If some condition evaluates to true, its nested
>>>>> actions are always executed no matter what preceded it. But of course
>>>>> the condition should be evaluated first, that's where various break
>>>>> values come into play.
>>>>>           
>>>> No way - the dialplan is always run on a "match all conditions" to do
>>>> the
>>>> actions! How else would multiple conditions EVER work?
>>>>         
>>> Completely wrong. And I don't see relevant logs in your message. Here
>>> is mine. A dialplan fragment:
>>>
>>>    <extension name="condition-test">
>>>        <condition wday="6" break="never"/>
>>>        <condition hour="9-18" break="on-true">
>>>            <action application="info"/>
>>>        </condition>
>>>    </extension>
>>>
>>> log:
>>>
>>> Dialplan: sofia/internal/1000 at 192.168.121.66 parsing
>>> [default->condition-test] continue=false
>>> Dialplan: sofia/internal/1000 at 192.168.121.66 Date/Time Match (FAIL)
>>> [condition-test] break=never
>>> Dialplan: sofia/internal/1000 at 192.168.121.66 Date/Time Match (PASS)
>>> [condition-test] break=on-true
>>> Dialplan: sofia/internal/1000 at 192.168.121.66 Action info()
>>>
>>> Guess what? The first condition fails but the action is still
>>> scheduled for execution (and indeed executed later)
>>>       
>
>   
I can't/didn't contribute much to this discussion, but I do follow it. 
My idea and (little) testing experience supports the explanation of Dimitry.
Either way, it 'proofs' there is confusion about the working of the 
'break' in condition tags.
If I can find the time, I will try to do some tests / build an example 
myself. Thank you both for the answers and insights so far.

regards,
Egbert.







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From mcampbellsmith at gmail.com  Mon Mar 21 13:30:45 2011
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Mon, 21 Mar 2011 21:30:45 +1100
Subject: [Freeswitch-users] No ringback/audio suddenly
Message-ID: <AANLkTikX6J1QnBDxxdRKdL9MDDEseoGZVp9AkS-62M-L@mail.gmail.com>

Hi!

I have an issue with no ringback or audio that has just started in the last
few weeks.  No changes in FS, which has been rock solid for a long time.  I
think my voip provider Pennytel has upgraded or changed their servers.

Anyway, the difference in signalling is as follows:

This does not work:
INVITE sip:number at sip.pennytel.com SIP/2.0
SIP/2.0 100 Trying
SIP/2.0 401 Unauthorized
ACK sip:number at sip.pennytel.com SIP/2.0
INVITE sip:number at sip.pennytel.com SIP/2.0
SIP/2.0 100 Trying
SIP/2.0 183 Session Progress
SIP/2.0 180 Ringing
SIP/2.0 200 OK
ACK sip:202.85.243.105:5061 SIP/2.0
BYE sip:202.85.243.105:5061 SIP/2.0
SIP/2.0 200 OK

This works
INVITE sip:number at sip.pennytel.com SIP/2.0
SIP/2.0 100 Trying
SIP/2.0 401 Unauthorized
ACK sip:number at sip.pennytel.com SIP/2.0
INVITE sip:number at sip.pennytel.com SIP/2.0
SIP/2.0 100 Trying
SIP/2.0 183 Session Progress
SIP/2.0 200 OK
ACK sip:202.85.243.105:5061 SIP/2.0

And I see after the Ringing signal in the case that does not work (there is
no ringing signal in the case that does work):
2011-03-21 20:28:25.677621 [DEBUG] sofia.c:4641 Channel
sofia/external/0393011079 skipping state [proceeding][180] received after a
RINGING

Can anyone point me to a reason for the difference in signalling and what I
can do to get audio 100% of the time?

THanks
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From marcdecorny at gmail.com  Mon Mar 21 20:59:16 2011
From: marcdecorny at gmail.com (Marc de Corny)
Date: Mon, 21 Mar 2011 17:59:16 +0000
Subject: [Freeswitch-users] Mod_fifo outbound strategy enterprise
Message-ID: <AANLkTiku=gKQAVSy2of8kSZoF+b+zXM5u_yHS45ojUd5@mail.gmail.com>

Hi all,

I am getting the hang of the fifo now and it is proving very stable which is
what we all need.

Got one question however regarding mod_fifo and the outbound strategies.

>From looking at the outbput of the fifo list commands I can see that that
parameter is always set to "ringall". which is fine, but I can also see many
stats on that number of calls and the last call that the members have taken.
Is there also a way of sending the calls to the longest idle or something of
that nature? Is that done by setting the outbound_strategy to "enterprise"
or is there another way to achieve that.

Is there also a command to change that value without editing the XML as I am
trying to make everything dynamic.

any input is very much appreciated.

thanks
Marc
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From yehavi.bourvine at gmail.com  Mon Mar 21 21:46:20 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Mon, 21 Mar 2011 20:46:20 +0200
Subject: [Freeswitch-users] Does Freeswitch complies with SIP Connect
	technical recommendation?
Message-ID: <AANLkTinF=5BX-MagFPfcisa-ionNs2dj7u7EWDLKg_9k@mail.gmail.com>

Hello,

  We published an RFP for our SIP connection to the PSTN. One of the
suppliers asked whether Freeswitch complies with the SIP CONNECT
recomendation. I have to answer them soon, and don't have enough time to
read and evaluate the entire document at present.

Does Freeswitch fullfills all the recomendations there?

                        Thanks, __Yehavi:
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From elijah at crankenstein.com  Mon Mar 21 21:50:44 2011
From: elijah at crankenstein.com (elijah)
Date: Mon, 21 Mar 2011 11:50:44 -0700
Subject: [Freeswitch-users] unable to insert channel variable in new call
	leg w/in Lua script
Message-ID: <AANLkTimQjpqrbuUxgT3g6mOTuBJSR1=eD4uXsJ+-sMLr@mail.gmail.com>

In a Lua script I instantiate a new call leg and (attempt to) insert a new
channel variable:

   local session = freeswitch.Session("user/1001 at stuff.com")
   if session:hangupCause() == "SUCCESS" then
      session:execute("export", "my_variable=" .. myVariable)
      session:transfer("ext_in_mycontext", "XML", "mycontext")

However, I am not able to retrieve my custom channel variable later. This
call leg is established correctly but there is no point in this call leg's
life where I can seem to find my variable. Is there another method by which
I should try to insert a channel variable in a newly created call leg?
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From frank at carmickle.com  Mon Mar 21 22:27:39 2011
From: frank at carmickle.com (Frank Carmickle)
Date: Mon, 21 Mar 2011 15:27:39 -0400
Subject: [Freeswitch-users] New FreeSWITCH Module: mod_ladspa
In-Reply-To: <AANLkTimPo4U2bzf+K4ucRq3xZHHDBnjFouhXDd_t+Fzk@mail.gmail.com>
References: <AANLkTimPo4U2bzf+K4ucRq3xZHHDBnjFouhXDd_t+Fzk@mail.gmail.com>
Message-ID: <BA819A5F-3832-454D-BE67-E1540D420C40@carmickle.com>

Hey Michael

This is cool.  Has there been any thought to making freeswitch a jack client.  That IMHO would be very very useful.  That would allow sip, or any other protocol that fs supports, to be a way to get audio in to professional sound production software like ardour, audacity or ecasound.

Thanks
--FC

On Mar 18, 2011, at 6:56 PM, Michael Collins wrote:

> Hello all!
> 
> The FreeSWITCH developers have added a cool new module: mod_ladspa. Check out the story:
> 
> http://www.freeswitch.org/node/313
> 
> Have fun!
> -Michael
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From wstephen80 at gmail.com  Mon Mar 21 22:33:28 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Mon, 21 Mar 2011 20:33:28 +0100
Subject: [Freeswitch-users] Delayed start of tone_detect
Message-ID: <AANLkTims-MfPkq_tr-JQd_ih=gPd1i8gXSFSX2tF_m8y@mail.gmail.com>

I start the tone_detect on answer using "execute_on_answer" as:

<action application="export" data="nolocal:execute_on_answer=tone_detect
MYTONE 680 ro +5000 set MYTONE=1" />

but I have the necessity to start the tone_detect for example 500ms after
answer: there is a way to do that in dialplan?
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From msc at freeswitch.org  Mon Mar 21 22:39:18 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 21 Mar 2011 12:39:18 -0700
Subject: [Freeswitch-users] New FreeSWITCH Module: mod_ladspa
In-Reply-To: <BA819A5F-3832-454D-BE67-E1540D420C40@carmickle.com>
References: <AANLkTimPo4U2bzf+K4ucRq3xZHHDBnjFouhXDd_t+Fzk@mail.gmail.com>
	<BA819A5F-3832-454D-BE67-E1540D420C40@carmickle.com>
Message-ID: <AANLkTinuRTBgrJWmALSxcdwZba2+z1=xez00ReF2h-1R@mail.gmail.com>

Not that I know of, but "patches welcome"!! :)
-MC

On Mon, Mar 21, 2011 at 12:27 PM, Frank Carmickle <frank at carmickle.com>wrote:

> Hey Michael
>
> This is cool.  Has there been any thought to making freeswitch a jack
> client.  That IMHO would be very very useful.  That would allow sip, or any
> other protocol that fs supports, to be a way to get audio in to professional
> sound production software like ardour, audacity or ecasound.
>
> Thanks
> --FC
>
> On Mar 18, 2011, at 6:56 PM, Michael Collins wrote:
>
> Hello all!
>
> The FreeSWITCH developers have added a cool new module: mod_ladspa. Check
> out the story:
>
> http://www.freeswitch.org/node/313
>
> Have fun!
> -Michael
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Mar 21 23:10:26 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 21 Mar 2011 13:10:26 -0700
Subject: [Freeswitch-users] Understanding break="never" in condition-tag
In-Reply-To: <4D87637F.3090309@redhosting.nl>
References: <4D85E0A1.507@redhosting.nl>
	<AANLkTimNYfHOrZrr7Q70Zea8Nsw-xEmQr2z0g-=PK+hv@mail.gmail.com>
	<4D85F6FB.8080801@redhosting.nl>
	<AANLkTimXv0PrtnjRPLfi5WFGhpyNmgc4H-Unvr7Bc7Xj@mail.gmail.com>
	<AANLkTimi5pAW-kJif8TVYXKugjbaK18J6hiBtJ712_q4@mail.gmail.com>
	<AANLkTikxwL9EHQ984242msZ4kG0wD9pDe8o6AYyOuGcf@mail.gmail.com>
	<AANLkTikc0a4ktzy3tNZHNbpNibSAChY5H3avumNSgaOT@mail.gmail.com>
	<AANLkTikbejGE2ku-X8VYjc=HctHCuSDDozUdusaKDFr4@mail.gmail.com>
	<AANLkTinty+efpatHL7eu79H5SuAjYdfVxA6o_O_A3rC+@mail.gmail.com>
	<AANLkTi=4Sqb-m+t-o81OYJwK=UTD17-fkcHtG7oe0LDN@mail.gmail.com>
	<AANLkTinJFMMMBVrLPsAeuKOApECywnCBkTgkMbnjTFzW@mail.gmail.com>
	<AANLkTikUn_Mdux6XQWx5ouQwEBc7j8uAEPricD==CM4T@mail.gmail.com>
	<4D87637F.3090309@redhosting.nl>
Message-ID: <AANLkTinR9CXqeOXwMu37TnDggb4ffuAd0yGOkAJDr7Mo@mail.gmail.com>

>  I can't/didn't contribute much to this discussion, but I do follow it. My
> idea and (little) testing experience supports the explanation of Dimitry.
> Either way, it 'proofs' there is confusion about the working of the 'break'
> in condition tags.
> If I can find the time, I will try to do some tests / build an example
> myself. Thank you both for the answers and insights so far.
>
>
Okay, for those of you who have the bridge book, please go to page 167-168.
Darren did a really good job of explaining this. For those of you who don't
have the book, SHAME! :)

The break flag determines what happens on processing conditions within a
single extension. Let's say that chan var ${freeswitch} has the value
"rocks" and ${foo} has the value "bar".

<extension name="break_flag_is_not_confusing">
  <condition "${freeswitch}" expression="^lame$">
    <!-- this does not get executed -->
  </condition>
  <!-- the following condition is not even evaluated -->
  <condition "${foo}" expression="bar">
  ...
  </condition>
</extension>

Look at the above snipped. The first condition evaluates to false, which
means all processing for this particular extension stops. The parse "breaks"
out of this extension and moves on to the next one. Now look at this
extension with the the break-flag set to "never"...

<extension name="break_flag_is_not_confusing">
  <condition "${freeswitch}" expression="^lame$" break="never">
    <!-- this does not get executed -->
  </condition>
  <!-- the following condition *is* evaluated -->
  <condition "${foo}" expression="bar">
  ...
  </condition>
</extension>

See the difference? With the break flag set to "never" then it does not
matter whether the condition is evaluated as true or false - the extension's
next condition gets evaluated. Why do you need the break="never"? Simple:
when no "break" is specified there is an implied break="on-false" for the
condition. In other words, by default, if one condition inside of an
extension evaluates to false then all processing for that extension stops.
This has the effect of allowing you to "stack" conditions to create a
logical AND.

So, in summary, the break flag will control how the parser behaves when
evaluating conditions. Normally when a single condition fails, the whole
extension is skipped, however when you do break="never" on a condition then
it does not matter if that particular condition is true or false - extension
processing will continue.

Hope this helps. If it doesn't then read chapters 5 and 8 of the book.
Chapter 5 (me) is a gentle intro into the dialplan. Chapter 8 (Darren) talks
more extensively about the advanced concepts of dialplan processing.

-MC
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From dunchan at freemail.hu  Mon Mar 21 23:09:08 2011
From: dunchan at freemail.hu (dunchan)
Date: Mon, 21 Mar 2011 21:09:08 +0100
Subject: [Freeswitch-users] pass variable from dialplan to bridge
In-Reply-To: <AANLkTi=HHse+=PVwfPjikqQw-YOkigLGO1C9Jsnx488z@mail.gmail.com>
References: <4D85B385.4060407@freemail.hu>	<AANLkTinmM2L2YpJUpNKnoe9ib4AtPkOOdfuJji8XT26g@mail.gmail.com>
	<AANLkTi=HHse+=PVwfPjikqQw-YOkigLGO1C9Jsnx488z@mail.gmail.com>
Message-ID: <4D87B064.9000702@freemail.hu>

Yes, the soulution was the:
  <param name="caller-id-in-from" value="true"/>
parameter!
Thank you for your help!

Viktor
> That doesn't sound right.
> I'm not sure exactly what you are trying to do, but I'm guessing you
> want the caller ID of this user in the FROM?
> So simply set <param name="caller-id-in-from" value="true"/>
> in the gateway, and then "set"
> "effective_caller_id=${outbound_caller_id_number}"
>
> -Avi
>
> On Sun, Mar 20, 2011 at 10:56 PM, guru singh <grsingh750 at gmail.com
> <mailto:grsingh750 at gmail.com>> wrote:
>
>     Hi,
>
>     Try export instead of set.
>     http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_export
>
>
>     On Sun, Mar 20, 2011 at 1:27 PM, dunchan <dunchan at freemail.hu
>     <mailto:dunchan at freemail.hu>> wrote:
>      > Hi!
>      >
>      > I have a simple question, how can i pass a varible from dialplan
>     to the
>      > bridge?
>      > I want to fill the 'from-user' field depends on sip user settings.
>      >
>      > I've tryed the following in dialplan:
>      > ...
>      > some conditions
>      > ...
>      > <action application="set"
>     data="from-user=${outbound_caller_id_number}"/>
>      > <action application="bridge" data="sofia/gateway/upc/$1"/>
>      >
>      > in a gateway xml i tryed:
>      > <gateway name="upc">
>      > <param name="username" value="..."/>
>      > <param name="password" value="..."/>
>      > <param name="proxy" value="..."/>
>      > <param name="register" value="false"/>
>      > <param name="from-user" value="${from-user}"/>
>      > </gateway>
>      >
>      > Above config ha no effect :(
>      > any suggestions?
>      >
>      > thx,
>      > Viktor
>      >
>      >
>      >
>      >
>      > _______________________________________________
>      > FreeSWITCH-users mailing list
>      > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>      > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>      >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>      > http://www.freeswitch.org
>      >
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From curriegrad2004 at gmail.com  Mon Mar 21 23:23:26 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 21 Mar 2011 13:23:26 -0700
Subject: [Freeswitch-users] Does Freeswitch complies with SIP Connect
 technical recommendation?
In-Reply-To: <AANLkTinF=5BX-MagFPfcisa-ionNs2dj7u7EWDLKg_9k@mail.gmail.com>
References: <AANLkTinF=5BX-MagFPfcisa-ionNs2dj7u7EWDLKg_9k@mail.gmail.com>
Message-ID: <AANLkTi=w40JoH=4hEe=b7M0CbODcSYLiiUG6JKq=wTPG@mail.gmail.com>

Most likely no

On Mon, Mar 21, 2011 at 11:46 AM, Yehavi Bourvine
<yehavi.bourvine at gmail.com> wrote:
> Hello,
>
> ? We published an RFP for our SIP connection to the PSTN. One of the
> suppliers asked whether Freeswitch complies with the?SIP?CONNECT
> recomendation. I have to answer them soon, and don't have enough time to
> read and evaluate the entire document at present.
>
> Does Freeswitch fullfills all the recomendations?there?
>
> ??????????????????????? Thanks, __Yehavi:
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From anthony.minessale at gmail.com  Mon Mar 21 23:44:50 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 21 Mar 2011 15:44:50 -0500
Subject: [Freeswitch-users] Understanding break="never" in condition-tag
In-Reply-To: <AANLkTinR9CXqeOXwMu37TnDggb4ffuAd0yGOkAJDr7Mo@mail.gmail.com>
References: <4D85E0A1.507@redhosting.nl>
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	<AANLkTinR9CXqeOXwMu37TnDggb4ffuAd0yGOkAJDr7Mo@mail.gmail.com>
Message-ID: <AANLkTik5YX2PtMU1vfaHbM9kGto_0h+x4zOOHHmsJaNv@mail.gmail.com>

The simple way to put it is that in the example in this email that is
from the wiki, using an empty condition with break=never is a noOP.

The typical usage of break=never would be if you wanted to execute
some extra options when a condition was true and skip it but keep
going when its not true.


<extension>
  <!-- because break=never even when the destionation does not begin
with 1 we just don't execute these actions and keep going -->
  <condition field="destination_number" expression="^1(\d+)$" break="never">
    <action application="set" data="begins_with_one=true"/>
  </condition>

  <condition field="destination_number" expression="^(\d+)$">
    <action application="bridge" data="sofia/internal/$1 at foobar.com"/>
  </condition>

</extension>

So stacking conditions with no actions in them and putting break=never
on them is the same as if they do not exist.




On Mon, Mar 21, 2011 at 3:10 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>>
>> I can't/didn't contribute much to this discussion, but I do follow it. My
>> idea and (little) testing experience supports the explanation of Dimitry.
>> Either way, it 'proofs' there is confusion about the working of the
>> 'break' in condition tags.
>> If I can find the time, I will try to do some tests / build an example
>> myself. Thank you both for the answers and insights so far.
>>
>
> Okay, for those of you who have the bridge book, please go to page 167-168.
> Darren did a really good job of explaining this. For those of you who don't
> have the book, SHAME! :)
> The break flag determines what happens on processing conditions within a
> single extension. Let's say that chan var ${freeswitch} has the value
> "rocks" and ${foo} has the value "bar".
> <extension name="break_flag_is_not_confusing">
> ??<condition "${freeswitch}" expression="^lame$">
> ?? ?<!-- this does not get executed -->
> ??</condition>
> ??<!-- the following condition is not even evaluated -->
> ??<condition "${foo}" expression="bar">
> ??...
> ??</condition>
> </extension>
> Look at the above snipped. The first condition evaluates to false, which
> means all processing for this particular extension stops. The parse "breaks"
> out of this extension and moves on to the next one. Now look at this
> extension with the the break-flag set to "never"...
> <extension name="break_flag_is_not_confusing">
> ??<condition "${freeswitch}" expression="^lame$" break="never">
> ?? ?<!-- this does not get executed -->
> ??</condition>
> ??<!-- the following condition *is* evaluated -->
> ??<condition "${foo}" expression="bar">
> ??...
> ??</condition>
> </extension>
> See the difference? With the break flag set to "never" then it does not
> matter whether the condition is evaluated as true or false - the extension's
> next condition gets evaluated. Why do you need the break="never"? Simple:
> when no "break" is specified there is an implied break="on-false" for the
> condition. In other words, by default, if one condition inside of an
> extension evaluates to false then all processing for that extension stops.
> This has the effect of allowing you to "stack" conditions to create a
> logical AND.
> So, in summary, the break flag will control how the parser behaves when
> evaluating conditions. Normally when a single condition fails, the whole
> extension is skipped, however when you do break="never" on a condition then
> it does not matter if that particular condition is true or false - extension
> processing will continue.
> Hope this helps. If it doesn't then read chapters 5 and 8 of the book.
> Chapter 5 (me) is a gentle intro into the dialplan. Chapter 8 (Darren) talks
> more extensively about the advanced concepts of dialplan processing.
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 joshua at seadragons.us  Tue Mar 22 00:13:34 2011
From: joshua at seadragons.us (Joshua Shaffner)
Date: Mon, 21 Mar 2011 17:13:34 -0400
Subject: [Freeswitch-users] mod_fsv,
	record and playback options (William Kendi ...)
Message-ID: <AANLkTimnhjmsumdTbnOXBJQ2H-dxd+DosCp6r5p1NQVb@mail.gmail.com>

William, I can not find jira ticket. Got a link? I would like to play
with your modification.

Anthm, any reason in particular why one should go with mod_mp4? Can it
play back video to sip video caller?

On Mon, Feb 21, 2011 at 5:21 PM,
<freeswitch-users-request at lists.freeswitch.org> wrote:
> Send FreeSWITCH-users mailing list submissions to
> ? ? ? ?freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> ? ? ? ?http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
> ? ? ? ?freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
> ? ? ? ?freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> Today's Topics:
>
> ? 1. Re: Using 16 KHz sounds (Michael Collins)
> ? 2. Re: FS with Microsoft Forefront TMG (curriegrad2004)
> ? 3. Re: FS with Microsoft Forefront TMG (Malay Thakershi)
> ? 4. Re: mod_fsv, record and playback options (William Kendi ...)
>
>
> ---------- Forwarded message ----------
> From:?Michael Collins <msc at freeswitch.org>
> To:?FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date:?Mon, 21 Feb 2011 12:35:54 -0800
> Subject:?Re: [Freeswitch-users] Using 16 KHz sounds
> It depends on why there is choppy audio. My guess is that going to 16k won't help. You should update to latest git and re-test, preferably on a system that is not in production. See if you can narrow down the conditions under which the audio is not good. Does it happen when the system is under load? Does it happen on every call, or only on certain calls? Things like that.
> -MC
>
> On Sun, Feb 20, 2011 at 12:51 PM, Malay Thakershi <mthakershi at gmail.com> wrote:
>>
>> Hello,
>> I use Cepstral in my mod_managed FS application. I mainly use Session.Speak and PlayAndGetDigits in my code to play WAV / audio text.
>> When I started using FS and got a stable program running, I used Cepstral Allison 8 KHz voice. But frequently I get choppy type of sound. Earlier it was acceptable but now some callers seem to have difficulty understanding the call audio.
>> Would it help if I get 16 KHz sounds / Cepstral license? What are changes I would need to make?
>> Thank you for any help.
>> Malay Thakershi
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> ---------- Forwarded message ----------
> From:?curriegrad2004 <curriegrad2004 at gmail.com>
> To:?FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date:?Mon, 21 Feb 2011 13:28:35 -0800
> Subject:?Re: [Freeswitch-users] FS with Microsoft Forefront TMG
> Publish ports 5060 and the range of rtp ports you specified in the
> switch.conf.xml file. That's how I did it in ISA 2006
>
> On Mon, Feb 21, 2011 at 12:22 PM, Malay Thakershi <mthakershi at gmail.com> wrote:
>> Hello,
>> Anyone has tried using FS successfully behind Microsoft Forefront Threat
>> Management Gateway?
>> Thank you for any direction.
>> Malay
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
>
> ---------- Forwarded message ----------
> From:?Malay Thakershi <mthakershi at gmail.com>
> To:?FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date:?Mon, 21 Feb 2011 16:03:15 -0600
> Subject:?Re: [Freeswitch-users] FS with Microsoft Forefront TMG
> Hello,
> ?? ?<!--RTP port range -->
> ?? ?<!--<param name="rtp-start-port" value="16384"/>-->
> ?? ?<!--<param name="rtp-end-port" value="32768"/>-->
> ?? ?<param name="rtp-enable-zrtp" value="true"/>
> I never had to change anything in this file. So I think my FS configuration working on default settings.
> So therefore, you mean to say if I just follow normal procedure, it should work, right?
> Malay
>
>
> On Mon, Feb 21, 2011 at 3:28 PM, curriegrad2004 <curriegrad2004 at gmail.com> wrote:
>>
>> Publish ports 5060 and the range of rtp ports you specified in the
>> switch.conf.xml file. That's how I did it in ISA 2006
>>
>> On Mon, Feb 21, 2011 at 12:22 PM, Malay Thakershi <mthakershi at gmail.com> wrote:
>> > Hello,
>> > Anyone has tried using FS successfully behind Microsoft Forefront Threat
>> > Management Gateway?
>> > Thank you for any direction.
>> > Malay
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> ---------- Forwarded message ----------
> From:?"William Kendi ..." <william.nishio at voicetechnology.com.br>
> To:?FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date:?Mon, 21 Feb 2011 19:21:26 -0300
> Subject:?Re: [Freeswitch-users] mod_fsv, record and playback options
> I did it!
>
> After some tears and sweats, I finally managed to create a working FSV demuxer module for the FFMPEG project!
>
> With this module, files in the FSV format now can be converted to any other format through the FFMPEG project!
>
> To install:
>
> 1). Put the "fsvdec.c" file in the "libavformat" directory.
> 2). Insert the line "REGISTER_DEMUXER? (FSV, fsv);" in the file "allformats.c" also in the "libavformat" directory.
> 3). Insert the line "OBJS-$(CONFIG_FSV_DEMUXER)?????????????? += fsvdec.o" in the file "Makefile" also in the "libavformat" directory.
> 4). Build the FFMPEG project using "make install".
>
> Now the FSV format seems to be more usable and I am trying to figure how to use the FreeSWITCH JIRA.
>
> 2011/1/21 Anthony Minessale <anthony.minessale at gmail.com>
>>
>> Sure, send it to Jira and we'll get it in.
>> Though, I'm surprised you would not want to use the mod_mp4 now that
>> it exists =D ?the FSV was sort if a hack I made up on a whim.
>>
>>
>>
>> On Fri, Jan 21, 2011 at 5:17 PM, William Suffill
>> <william.suffill at gmail.com> wrote:
>> > Best to add the patches/details into Jira [http://jira.freeswitch.org] so it
>> > can be tracked and reviewed for being added to the source tree.
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From joshua at seadragons.us  Tue Mar 22 00:16:57 2011
From: joshua at seadragons.us (Joshua Shaffner)
Date: Mon, 21 Mar 2011 17:16:57 -0400
Subject: [Freeswitch-users] help
Message-ID: <AANLkTikG1pCU+0CAcKD0rzm=LXKtu75rW-SB-ziz98xp@mail.gmail.com>

On Mon, Mar 21, 2011 at 4:10 PM,
<freeswitch-users-request at lists.freeswitch.org> wrote:
> Send FreeSWITCH-users mailing list submissions to
> ? ? ? ?freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> ? ? ? ?http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
> ? ? ? ?freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
> ? ? ? ?freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> Today's Topics:
>
> ? 1. Mod_fifo outbound strategy enterprise (Marc de Corny)
> ? 2. Does Freeswitch complies with SIP Connect technical
> ? ? ?recommendation? (Yehavi Bourvine)
> ? 3. unable to insert channel variable in new call ? ? leg w/in Lua
> ? ? ?script (elijah)
> ? 4. Re: New FreeSWITCH Module: mod_ladspa (Frank Carmickle)
> ? 5. Delayed start of tone_detect (Stephen Wilde)
> ? 6. Re: New FreeSWITCH Module: mod_ladspa (Michael Collins)
> ? 7. Re: Understanding break="never" in condition-tag (Michael Collins)
>
>
> ---------- Forwarded message ----------
> From:?Marc de Corny <marcdecorny at gmail.com>
> To:?FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date:?Mon, 21 Mar 2011 17:59:16 +0000
> Subject:?[Freeswitch-users] Mod_fifo outbound strategy enterprise
> Hi all,
> I am getting the hang of the fifo now and it is proving very stable which is what we all need.
> Got one question however regarding mod_fifo and the outbound strategies.
> From looking at the outbput of the fifo list commands I can see that that parameter is always set to "ringall". which is fine, but I can also see many stats on that number of calls and the last call that the members have taken. Is there also a way of sending the calls to the longest idle or something of that nature? Is that done by setting the outbound_strategy to "enterprise" or is there another way to achieve that.
> Is there also a command to change that value without editing the XML as I am trying to make everything dynamic.
> any input is very much appreciated.
> thanks
> Marc
>
> ---------- Forwarded message ----------
> From:?Yehavi Bourvine <yehavi.bourvine at gmail.com>
> To:?FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date:?Mon, 21 Mar 2011 20:46:20 +0200
> Subject:?[Freeswitch-users] Does Freeswitch complies with SIP Connect technical recommendation?
> Hello,
>
> ? We published an RFP for our SIP connection to the PSTN. One of the suppliers asked whether Freeswitch complies with the?SIP?CONNECT recomendation. I have to answer them soon, and don't have enough time to read and evaluate the entire document at present.
>
> Does Freeswitch fullfills all the recomendations?there?
>
> ??????????????????????? Thanks, __Yehavi:
>
> ---------- Forwarded message ----------
> From:?elijah <elijah at crankenstein.com>
> To:?FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date:?Mon, 21 Mar 2011 11:50:44 -0700
> Subject:?[Freeswitch-users] unable to insert channel variable in new call leg w/in Lua script
> In a Lua script I instantiate a new call leg and (attempt to) insert a new channel variable:
> ?? local session = freeswitch.Session("user/1001 at stuff.com")
> ?? if session:hangupCause() == "SUCCESS" then
> ?? ? ?session:execute("export", "my_variable=" .. myVariable)
> ?? ? ?session:transfer("ext_in_mycontext", "XML", "mycontext")
> However, I am not able to retrieve my custom channel variable later. This call leg is established correctly but there is no point in this call leg's life where I can seem to find my variable. Is there another method by which I should try to insert a channel variable in a newly created call leg?
>
>
>
> ---------- Forwarded message ----------
> From:?Frank Carmickle <frank at carmickle.com>
> To:?FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date:?Mon, 21 Mar 2011 15:27:39 -0400
> Subject:?Re: [Freeswitch-users] New FreeSWITCH Module: mod_ladspa
> Hey Michael
> This is cool. ?Has there been any thought to making freeswitch a jack client. ?That IMHO would be very very useful. ?That would allow sip, or any other protocol that fs supports, to be a way to get audio in to professional sound production software like ardour, audacity or ecasound.
> Thanks
> --FC
> On Mar 18, 2011, at 6:56 PM, Michael Collins wrote:
>
> Hello all!
> The FreeSWITCH developers have added a cool new module: mod_ladspa. Check out the story:
> http://www.freeswitch.org/node/313
> Have fun!
> -Michael
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ---------- Forwarded message ----------
> From:?Stephen Wilde <wstephen80 at gmail.com>
> To:?FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date:?Mon, 21 Mar 2011 20:33:28 +0100
> Subject:?[Freeswitch-users] Delayed start of tone_detect
> I start the tone_detect on answer using "execute_on_answer" as:
> <action application="export" data="nolocal:execute_on_answer=tone_detect MYTONE 680 ro +5000 set MYTONE=1" />
> but I have the necessity to start the tone_detect for example 500ms after answer: there is a way to do that in dialplan?
>
> ---------- Forwarded message ----------
> From:?Michael Collins <msc at freeswitch.org>
> To:?FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date:?Mon, 21 Mar 2011 12:39:18 -0700
> Subject:?Re: [Freeswitch-users] New FreeSWITCH Module: mod_ladspa
> Not that I know of, but "patches welcome"!! :)
> -MC
>
> On Mon, Mar 21, 2011 at 12:27 PM, Frank Carmickle <frank at carmickle.com> wrote:
>>
>> Hey Michael
>> This is cool. ?Has there been any thought to making freeswitch a jack client. ?That IMHO would be very very useful. ?That would allow sip, or any other protocol that fs supports, to be a way to get audio in to professional sound production software like ardour, audacity or ecasound.
>> Thanks
>> --FC
>> On Mar 18, 2011, at 6:56 PM, Michael Collins wrote:
>>
>> Hello all!
>> The FreeSWITCH developers have added a cool new module: mod_ladspa. Check out the story:
>> http://www.freeswitch.org/node/313
>> Have fun!
>> -Michael
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> ---------- Forwarded message ----------
> From:?Michael Collins <msc at freeswitch.org>
> To:?FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date:?Mon, 21 Mar 2011 13:10:26 -0700
> Subject:?Re: [Freeswitch-users] Understanding break="never" in condition-tag
>
>>
>> I can't/didn't contribute much to this discussion, but I do follow it. My idea and (little) testing experience supports the explanation of Dimitry.
>> Either way, it 'proofs' there is confusion about the working of the 'break' in condition tags.
>> If I can find the time, I will try to do some tests / build an example myself. Thank you both for the answers and insights so far.
>>
>
> Okay, for those of you who have the bridge book, please go to page 167-168. Darren did a really good job of explaining this. For those of you who don't have the book, SHAME! :)
> The break flag determines what happens on processing conditions within a single extension. Let's say that chan var ${freeswitch} has the value "rocks" and ${foo} has the value "bar".
> <extension name="break_flag_is_not_confusing">
> ??<condition "${freeswitch}" expression="^lame$">
> ?? ?<!-- this does not get executed -->
> ??</condition>
> ??<!-- the following condition is not even evaluated -->
> ??<condition "${foo}" expression="bar">
> ??...
> ??</condition>
> </extension>
> Look at the above snipped. The first condition evaluates to false, which means all processing for this particular extension stops. The parse "breaks" out of this extension and moves on to the next one. Now look at this extension with the the break-flag set to "never"...
> <extension name="break_flag_is_not_confusing">
> ??<condition "${freeswitch}" expression="^lame$" break="never">
> ?? ?<!-- this does not get executed -->
> ??</condition>
> ??<!-- the following condition *is* evaluated -->
> ??<condition "${foo}" expression="bar">
> ??...
> ??</condition>
> </extension>
> See the difference? With the break flag set to "never" then it does not matter whether the condition is evaluated as true or false - the extension's next condition gets evaluated. Why do you need the break="never"? Simple: when no "break" is specified there is an implied break="on-false" for the condition. In other words, by default, if one condition inside of an extension evaluates to false then all processing for that extension stops. This has the effect of allowing you to "stack" conditions to create a logical AND.
> So, in summary, the break flag will control how the parser behaves when evaluating conditions. Normally when a single condition fails, the whole extension is skipped, however when you do break="never" on a condition then it does not matter if that particular condition is true or false - extension processing will continue.
> Hope this helps. If it doesn't then read chapters 5 and 8 of the book. Chapter 5 (me) is a gentle intro into the dialplan. Chapter 8 (Darren) talks more extensively about the advanced concepts of dialplan processing.
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Tue Mar 22 00:27:48 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 21 Mar 2011 14:27:48 -0700
Subject: [Freeswitch-users] help
In-Reply-To: <AANLkTikG1pCU+0CAcKD0rzm=LXKtu75rW-SB-ziz98xp@mail.gmail.com>
References: <AANLkTikG1pCU+0CAcKD0rzm=LXKtu75rW-SB-ziz98xp@mail.gmail.com>
Message-ID: <AANLkTi==tHm8yFi5JtJ=q7zECZhtMDfLS035xV2LebpD@mail.gmail.com>

Before everyone starts flaming Joshua please know that this was an
accidental send, so go easy on him. It's is first time in our mailing list.

Welcome, jShaf!
-MC


On Mon, Mar 21, 2011 at 2:16 PM, Joshua Shaffner <joshua at seadragons.us>wrote:

>
<snip>
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From infos at madovsky.org  Tue Mar 22 00:28:29 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 21 Mar 2011 17:28:29 -0400
Subject: [Freeswitch-users] New FreeSWITCH Module: mod_ladspa
References: <AANLkTimPo4U2bzf+K4ucRq3xZHHDBnjFouhXDd_t+Fzk@mail.gmail.com><BA819A5F-3832-454D-BE67-E1540D420C40@carmickle.com>
	<AANLkTinuRTBgrJWmALSxcdwZba2+z1=xez00ReF2h-1R@mail.gmail.com>
Message-ID: <B5CCC1D8F17C49388C412C36648BCE51@e1705>

mhmm, audacity is not a pro software...
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, March 21, 2011 3:39 PM
  Subject: Re: [Freeswitch-users] New FreeSWITCH Module: mod_ladspa


  Not that I know of, but "patches welcome"!! :)
  -MC


  On Mon, Mar 21, 2011 at 12:27 PM, Frank Carmickle <frank at carmickle.com> wrote:

    Hey Michael


    This is cool.  Has there been any thought to making freeswitch a jack client.  That IMHO would be very very useful.  That would allow sip, or any other protocol that fs supports, to be a way to get audio in to professional sound production software like ardour, audacity or ecasound.


    Thanks
    --FC


    On Mar 18, 2011, at 6:56 PM, Michael Collins wrote:


      Hello all!


      The FreeSWITCH developers have added a cool new module: mod_ladspa. Check out the story:


      http://www.freeswitch.org/node/313


      Have fun!
      -Michael
      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org




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






------------------------------------------------------------------------------


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From elijah at crankenstein.com  Tue Mar 22 00:59:05 2011
From: elijah at crankenstein.com (elijah)
Date: Mon, 21 Mar 2011 14:59:05 -0700
Subject: [Freeswitch-users] unable to insert channel variable in new
	call leg w/in Lua script
In-Reply-To: <AANLkTimQjpqrbuUxgT3g6mOTuBJSR1=eD4uXsJ+-sMLr@mail.gmail.com>
References: <AANLkTimQjpqrbuUxgT3g6mOTuBJSR1=eD4uXsJ+-sMLr@mail.gmail.com>
Message-ID: <AANLkTi=vri-K=y2_wLkPAB5zfrgZBrow-T5R_vKABM-X@mail.gmail.com>

Actually, I think the behavior I'm seeing is that I'm not able to retrieve
these variables on the A-leg, but they are present on the B-leg. How should
I insert custom channel variables into the A-leg of a call originated from
Lua, so that they appear in the CALL_HANGUP_COMPLETE event on this first
leg?


On Mon, Mar 21, 2011 at 11:50 AM, elijah <elijah at crankenstein.com> wrote:

> In a Lua script I instantiate a new call leg and (attempt to) insert a new
> channel variable:
>
>    local session = freeswitch.Session("user/1001 at stuff.com")
>    if session:hangupCause() == "SUCCESS" then
>       session:execute("export", "my_variable=" .. myVariable)
>       session:transfer("ext_in_mycontext", "XML", "mycontext")
>
> However, I am not able to retrieve my custom channel variable later. This
> call leg is established correctly but there is no point in this call leg's
> life where I can seem to find my variable. Is there another method by which
> I should try to insert a channel variable in a newly created call leg?
>
>
>
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From grsingh750 at gmail.com  Tue Mar 22 01:15:30 2011
From: grsingh750 at gmail.com (guru singh)
Date: Tue, 22 Mar 2011 03:45:30 +0530
Subject: [Freeswitch-users] Mod_fifo outbound strategy enterprise
In-Reply-To: <AANLkTiku=gKQAVSy2of8kSZoF+b+zXM5u_yHS45ojUd5@mail.gmail.com>
References: <AANLkTiku=gKQAVSy2of8kSZoF+b+zXM5u_yHS45ojUd5@mail.gmail.com>
Message-ID: <AANLkTi=5gfBuLvm41p82nz8M5-cRrPYZtNTv_w_r-GMp@mail.gmail.com>

Hi Marc,

For intelligent routing of calls such as longest-idle and various
other strategies, you should have a look at mod_callcenter

Regards,
guru

On Mon, Mar 21, 2011 at 11:29 PM, Marc de Corny <marcdecorny at gmail.com> wrote:
> Hi all,
> I am getting the hang of the fifo now and it is proving very stable which is
> what we all need.
> Got one question however regarding mod_fifo and the outbound strategies.
> From looking at the outbput of the fifo list commands I can see that that
> parameter is always set to "ringall". which is fine, but I can also see many
> stats on that number of calls and the last call that the members have taken.
> Is there also a way of sending the calls to the longest idle or something of
> that nature? Is that done by setting the outbound_strategy to "enterprise"
> or is there another way to achieve that.
> Is there also a command to change that value without editing the XML as I am
> trying to make everything dynamic.
> any input is very much appreciated.
> thanks
> Marc
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From frank at rosengart.de  Tue Mar 22 01:26:05 2011
From: frank at rosengart.de (Frank Rosengart)
Date: Mon, 21 Mar 2011 23:26:05 +0100
Subject: [Freeswitch-users] audio dropouts with celt
Message-ID: <4D87D07D.9000801@rosengart.de>

Hi,

I'm connecting two FS'es with local audio (PA) each, via SIP using the
CELT codec. I experience occasional audio dropouts, maybe one in five
seconds. But not regulary. Sometimes it's fine for a minute. The
recorded waveform shows there is just silence for some milliseconds.
Sometimes the audio is garbled for a portion of a second.

This happens even when the computers are on the same LAN, but it's
getting worse when using public internet. All systems are running FS
from GIT, with CELT 0.10.0.

I have done the following debugging:

- recording the session: recorded on both sides, local audio is ok,
remote side has dropouts. pa loopback has no issues -> no
soundcard/driver issue

- running wireshark, using the RTP analyzer: on PCMA,G722 there is no
packet loss, no notable jitter (<20ms). The RTP analyzer does not show
any jitter info on CELT payload packets. I'm not sure if it can see lost
packets. -> I'm quite sure there is no packet loss.


How timing critical is CELT? Any ideas how I can go on with debugging?
Or is this better posted to the CELT developers? Is anyone running
FS+CELT with dropout-free audio?


Thanks!

Frank


From anthony.minessale at gmail.com  Tue Mar 22 01:26:24 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 21 Mar 2011 17:26:24 -0500
Subject: [Freeswitch-users] Mod_fifo outbound strategy enterprise
In-Reply-To: <AANLkTiku=gKQAVSy2of8kSZoF+b+zXM5u_yHS45ojUd5@mail.gmail.com>
References: <AANLkTiku=gKQAVSy2of8kSZoF+b+zXM5u_yHS45ojUd5@mail.gmail.com>
Message-ID: <AANLkTim4XgQXUYRqMZstsEMd6oimbfU6iHG9Y0MRSkvU@mail.gmail.com>

The ringall strategy is based on giving every queue a fair chance
based on its configured importance.

The following happens nonstop:

*) Loop all the active call queues containing waiting callers not
already receiving a call sorted on queue priority:
       *) Select agents from a db (omitting any who are on a call or
who are in wrap-up) sorted by
            least amount of consecutive missed calls then least amount
of answered calls.
*) Place an outbound call to this agent with the caller data of the
waiting customer

This distributes the opportunities to have your call answered across
all queues and chooses the most-likely-to-answer-the-phone agent.


The shortcoming of this strategy is simply the fact that you must
pre-allocate agents for each caller to ensure that you can supply the
caller's caller id to the agent when you call them.

The enterprise one is accelerated by figuring out how many callers are
waiting and calling that many agents at once with no caller id info
and inserting them into the queue to service the next customer in
line. Since there is no need to pre-match the caller and agent the
order of what agent and caller are paired is moot.  This was the
original behavior but the importance of caller id seems to prevail so
we made the other method the default.

It would be possible to code in other strategies but to maintain the
simplicity of fifo I did not really bother with any more.








On Mon, Mar 21, 2011 at 12:59 PM, Marc de Corny <marcdecorny at gmail.com> wrote:
> Hi all,
> I am getting the hang of the fifo now and it is proving very stable which is
> what we all need.
> Got one question however regarding mod_fifo and the outbound strategies.
> From looking at the outbput of the fifo list commands I can see that that
> parameter is always set to "ringall". which is fine, but I can also see many
> stats on that number of calls and the last call that the members have taken.
> Is there also a way of sending the calls to the longest idle or something of
> that nature? Is that done by setting the outbound_strategy to "enterprise"
> or is there another way to achieve that.
> Is there also a command to change that value without editing the XML as I am
> trying to make everything dynamic.
> any input is very much appreciated.
> thanks
> Marc
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 elijah at crankenstein.com  Tue Mar 22 02:13:43 2011
From: elijah at crankenstein.com (elijah)
Date: Mon, 21 Mar 2011 16:13:43 -0700
Subject: [Freeswitch-users] unable to insert channel variable in new
	call leg w/in Lua script
In-Reply-To: <AANLkTi=vri-K=y2_wLkPAB5zfrgZBrow-T5R_vKABM-X@mail.gmail.com>
References: <AANLkTimQjpqrbuUxgT3g6mOTuBJSR1=eD4uXsJ+-sMLr@mail.gmail.com>
	<AANLkTi=vri-K=y2_wLkPAB5zfrgZBrow-T5R_vKABM-X@mail.gmail.com>
Message-ID: <AANLkTimF2k7K7NiZaJCzNze=t2ycShW-QWdQrJZa=9Rc@mail.gmail.com>

Sorry if I wasted anyone's time. I've answered my own question. To set the
variable on the A-leg of a call originated by a Lua script I put the
variable here:

   local session = freeswitch.Session("{my_variable='" .. myVariable ..
"'}user/1001 at stuff.com")
   if session:hangupCause() == "SUCCESS" then
      session:transfer("ext_in_mycontext", "XML", "mycontext")


On Mon, Mar 21, 2011 at 2:59 PM, elijah <elijah at crankenstein.com> wrote:

> Actually, I think the behavior I'm seeing is that I'm not able to retrieve
> these variables on the A-leg, but they are present on the B-leg. How should
> I insert custom channel variables into the A-leg of a call originated from
> Lua, so that they appear in the CALL_HANGUP_COMPLETE event on this first
> leg?
>
>
> On Mon, Mar 21, 2011 at 11:50 AM, elijah <elijah at crankenstein.com> wrote:
>
>> In a Lua script I instantiate a new call leg and (attempt to) insert a new
>> channel variable:
>>
>>    local session = freeswitch.Session("user/1001 at stuff.com")
>>    if session:hangupCause() == "SUCCESS" then
>>       session:execute("export", "my_variable=" .. myVariable)
>>       session:transfer("ext_in_mycontext", "XML", "mycontext")
>>
>> However, I am not able to retrieve my custom channel variable later. This
>> call leg is established correctly but there is no point in this call leg's
>> life where I can seem to find my variable. Is there another method by which
>> I should try to insert a channel variable in a newly created call leg?
>>
>>
>>
>
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From anthony.minessale at gmail.com  Tue Mar 22 02:29:18 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 21 Mar 2011 18:29:18 -0500
Subject: [Freeswitch-users] mod_fsv,
 record and playback options (William Kendi ...)
In-Reply-To: <AANLkTimnhjmsumdTbnOXBJQ2H-dxd+DosCp6r5p1NQVb@mail.gmail.com>
References: <AANLkTimnhjmsumdTbnOXBJQ2H-dxd+DosCp6r5p1NQVb@mail.gmail.com>
Message-ID: <AANLkTimxk8Ur4FMubHX+zNrqT1=jVBrwiybW+XNKYNK6@mail.gmail.com>

I really don't know how mod_mp4 works I have no had a chance to try it.
mod_fsv is pretty much a ad-hoc encoding I made up one random day so
it's not really a standard or anything.
It was just a way to save the rtp stream and raw audio to a single file.


On Mon, Mar 21, 2011 at 4:13 PM, Joshua Shaffner <joshua at seadragons.us> wrote:
> William, I can not find jira ticket. Got a link? I would like to play
> with your modification.
>
> Anthm, any reason in particular why one should go with mod_mp4? Can it
> play back video to sip video caller?
>
> On Mon, Feb 21, 2011 at 5:21 PM,
> <freeswitch-users-request at lists.freeswitch.org> wrote:
>> Send FreeSWITCH-users mailing list submissions to
>> ? ? ? ?freeswitch-users at lists.freeswitch.org
>>
>> To subscribe or unsubscribe via the World Wide Web, visit
>> ? ? ? ?http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> or, via email, send a message with subject or body 'help' to
>> ? ? ? ?freeswitch-users-request at lists.freeswitch.org
>>
>> You can reach the person managing the list at
>> ? ? ? ?freeswitch-users-owner at lists.freeswitch.org
>>
>> When replying, please edit your Subject line so it is more specific
>> than "Re: Contents of FreeSWITCH-users digest..."
>>
>> Today's Topics:
>>
>> ? 1. Re: Using 16 KHz sounds (Michael Collins)
>> ? 2. Re: FS with Microsoft Forefront TMG (curriegrad2004)
>> ? 3. Re: FS with Microsoft Forefront TMG (Malay Thakershi)
>> ? 4. Re: mod_fsv, record and playback options (William Kendi ...)
>>
>>
>> ---------- Forwarded message ----------
>> From:?Michael Collins <msc at freeswitch.org>
>> To:?FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Date:?Mon, 21 Feb 2011 12:35:54 -0800
>> Subject:?Re: [Freeswitch-users] Using 16 KHz sounds
>> It depends on why there is choppy audio. My guess is that going to 16k won't help. You should update to latest git and re-test, preferably on a system that is not in production. See if you can narrow down the conditions under which the audio is not good. Does it happen when the system is under load? Does it happen on every call, or only on certain calls? Things like that.
>> -MC
>>
>> On Sun, Feb 20, 2011 at 12:51 PM, Malay Thakershi <mthakershi at gmail.com> wrote:
>>>
>>> Hello,
>>> I use Cepstral in my mod_managed FS application. I mainly use Session.Speak and PlayAndGetDigits in my code to play WAV / audio text.
>>> When I started using FS and got a stable program running, I used Cepstral Allison 8 KHz voice. But frequently I get choppy type of sound. Earlier it was acceptable but now some callers seem to have difficulty understanding the call audio.
>>> Would it help if I get 16 KHz sounds / Cepstral license? What are changes I would need to make?
>>> Thank you for any help.
>>> Malay Thakershi
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From:?curriegrad2004 <curriegrad2004 at gmail.com>
>> To:?FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Date:?Mon, 21 Feb 2011 13:28:35 -0800
>> Subject:?Re: [Freeswitch-users] FS with Microsoft Forefront TMG
>> Publish ports 5060 and the range of rtp ports you specified in the
>> switch.conf.xml file. That's how I did it in ISA 2006
>>
>> On Mon, Feb 21, 2011 at 12:22 PM, Malay Thakershi <mthakershi at gmail.com> wrote:
>>> Hello,
>>> Anyone has tried using FS successfully behind Microsoft Forefront Threat
>>> Management Gateway?
>>> Thank you for any direction.
>>> Malay
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From:?Malay Thakershi <mthakershi at gmail.com>
>> To:?FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Date:?Mon, 21 Feb 2011 16:03:15 -0600
>> Subject:?Re: [Freeswitch-users] FS with Microsoft Forefront TMG
>> Hello,
>> ?? ?<!--RTP port range -->
>> ?? ?<!--<param name="rtp-start-port" value="16384"/>-->
>> ?? ?<!--<param name="rtp-end-port" value="32768"/>-->
>> ?? ?<param name="rtp-enable-zrtp" value="true"/>
>> I never had to change anything in this file. So I think my FS configuration working on default settings.
>> So therefore, you mean to say if I just follow normal procedure, it should work, right?
>> Malay
>>
>>
>> On Mon, Feb 21, 2011 at 3:28 PM, curriegrad2004 <curriegrad2004 at gmail.com> wrote:
>>>
>>> Publish ports 5060 and the range of rtp ports you specified in the
>>> switch.conf.xml file. That's how I did it in ISA 2006
>>>
>>> On Mon, Feb 21, 2011 at 12:22 PM, Malay Thakershi <mthakershi at gmail.com> wrote:
>>> > Hello,
>>> > Anyone has tried using FS successfully behind Microsoft Forefront Threat
>>> > Management Gateway?
>>> > Thank you for any direction.
>>> > Malay
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> ---------- Forwarded message ----------
>> From:?"William Kendi ..." <william.nishio at voicetechnology.com.br>
>> To:?FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Date:?Mon, 21 Feb 2011 19:21:26 -0300
>> Subject:?Re: [Freeswitch-users] mod_fsv, record and playback options
>> I did it!
>>
>> After some tears and sweats, I finally managed to create a working FSV demuxer module for the FFMPEG project!
>>
>> With this module, files in the FSV format now can be converted to any other format through the FFMPEG project!
>>
>> To install:
>>
>> 1). Put the "fsvdec.c" file in the "libavformat" directory.
>> 2). Insert the line "REGISTER_DEMUXER? (FSV, fsv);" in the file "allformats.c" also in the "libavformat" directory.
>> 3). Insert the line "OBJS-$(CONFIG_FSV_DEMUXER)?????????????? += fsvdec.o" in the file "Makefile" also in the "libavformat" directory.
>> 4). Build the FFMPEG project using "make install".
>>
>> Now the FSV format seems to be more usable and I am trying to figure how to use the FreeSWITCH JIRA.
>>
>> 2011/1/21 Anthony Minessale <anthony.minessale at gmail.com>
>>>
>>> Sure, send it to Jira and we'll get it in.
>>> Though, I'm surprised you would not want to use the mod_mp4 now that
>>> it exists =D ?the FSV was sort if a hack I made up on a whim.
>>>
>>>
>>>
>>> On Fri, Jan 21, 2011 at 5:17 PM, William Suffill
>>> <william.suffill at gmail.com> wrote:
>>> > Best to add the patches/details into Jira [http://jira.freeswitch.org] so it
>>> > can be tracked and reviewed for being added to the source tree.
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > 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-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 fs-list at communicatefreely.net  Tue Mar 22 02:30:40 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Mon, 21 Mar 2011 19:30:40 -0400
Subject: [Freeswitch-users] PA and ALSA
In-Reply-To: <1300057517408-6167368.post@n2.nabble.com>
References: <4D7BF5E6.30201@rosengart.de>
	<4D7D33C1.1060009@rosengart.de>	<201103131724.03970.sos@sokhapkin.dyndns.org>
	<1300057517408-6167368.post@n2.nabble.com>
Message-ID: <4D87DFA0.1040805@communicatefreely.net>

I for one am rather grateful that they DID include them.

Not everything makes it into the FreeBSD ports tree (I use FreeBSD, not 
Linux), but I haven't had any problem compiling the required libraries 
that were included with FreeSWITCH.  IT has allowed me to get things 
like SpanDSP working without too much trouble at all.  I tried compiling 
the SpanDSP source, and it wasn't the right version, or there was some 
other issue.  The FS devs include just what you need to make it work, 
without messing around somewhere else.

-Tim

Jeff Lenk wrote:
> I'm sure there were very good reasons for these decisions when they were
> made. So rather than criticizing those decisions why not help by finding the
> problems and submitting patches to fix them.
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/PA-and-ALSA-tp6165168p6167368.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From ejay.greeves at yahoo.com  Tue Mar 22 02:10:51 2011
From: ejay.greeves at yahoo.com (Ejay Greeves)
Date: Mon, 21 Mar 2011 23:10:51 +0000 (GMT)
Subject: [Freeswitch-users] send custom tcp message from esl
Message-ID: <921841.76258.qm@web132304.mail.ird.yahoo.com>

I am using esl in Ruby I want to send tcp messages to a server.? When I tried to make my tcpsocket class it did not work it must be messing with the socket setup of esl. Is there a way without the need to manage the sockets seperatley to use els methods to send out custom tcp message. All I need is for a simple string to be fired to tcp port. How can I do this from inside esl.



      
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From chenzhanping at gmail.com  Tue Mar 22 06:11:37 2011
From: chenzhanping at gmail.com (=?GB2312?B?1bzGvbPC?=)
Date: Tue, 22 Mar 2011 11:11:37 +0800
Subject: [Freeswitch-users] Question about mod_dingaling.
In-Reply-To: <AANLkTikd9baOGf0bmL6m3EaiPcGy6Fgfm9=jEYBg+GB2@mail.gmail.com>
References: <AANLkTikd9baOGf0bmL6m3EaiPcGy6Fgfm9=jEYBg+GB2@mail.gmail.com>
Message-ID: <AANLkTinpUymFWTdoU1Xx0oNFNhyRAKx04SmoWy5avVC9@mail.gmail.com>

Very sorry, the problem has been resolved. Is a problem with my account.
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From msc at freeswitch.org  Tue Mar 22 07:07:25 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 21 Mar 2011 21:07:25 -0700
Subject: [Freeswitch-users] Does Freeswitch complies with SIP Connect
 technical recommendation?
In-Reply-To: <AANLkTinF=5BX-MagFPfcisa-ionNs2dj7u7EWDLKg_9k@mail.gmail.com>
References: <AANLkTinF=5BX-MagFPfcisa-ionNs2dj7u7EWDLKg_9k@mail.gmail.com>
Message-ID: <AANLkTimWQh6aH8FOd3rT249pM0TJBSoDV12psf9EAWZN@mail.gmail.com>

No, we have not paid them thousands of dollars for them to say that we are
"SIP CONNECT Compliant." That being said, I read through some of the 1.1
specs. I didn't see anything that was egregious, but I also did not look at
all 45 pages. This much I can tell you: to be "SIP CONNECT Compliant" you
would need to program your dialplan for some specifics. Here's an example:

Section 13 of the spec says:

*13 Emergency Services*

The SIP-PBX MUST have a dial plan that recognizes emergency calls.

This will differ from one country to the next and is a function of the
administrator's dialplan configuration. There are no inherent limitations in
FreeSWITCH to prevent you from being up to spec on this. However, out of the
box with a default installation you won't have this already configured.

My guess is that we could write tests that demonstrate that FreeSWITCH
*could* be SIP CONNECT compliant, however I doubt that would be of much
value because you have to cough up a minimum of $2500 to have their seal of
approval. Beyond that look at their membership list:

http://www.sipforum.org/component/option,com_fullmember/Itemid,195/

It's a who's who of retarded SIP providers and PBXes. They've got Sonus on
their list for cying out loud! That alone should invalidate the entire SIP
CONNECT venture IMHO.

So, I would answer your supplier this way: As far as I know there aren't any
limitations in FreeSWITCH that would prevent it from complying with the SIP
CONNECT recommendations.

Oh, and GLWT! :)

-MC

On Mon, Mar 21, 2011 at 11:46 AM, Yehavi Bourvine <yehavi.bourvine at gmail.com
> wrote:

> Hello,
>
>   We published an RFP for our SIP connection to the PSTN. One of the
> suppliers asked whether Freeswitch complies with the SIP CONNECT
> recomendation. I have to answer them soon, and don't have enough time to
> read and evaluate the entire document at present.
>
> Does Freeswitch fullfills all the recomendations there?
>
>                         Thanks, __Yehavi:
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveu at coppice.org  Tue Mar 22 07:25:58 2011
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 22 Mar 2011 12:25:58 +0800
Subject: [Freeswitch-users] Does Freeswitch complies with SIP Connect
 technical recommendation?
In-Reply-To: <AANLkTimWQh6aH8FOd3rT249pM0TJBSoDV12psf9EAWZN@mail.gmail.com>
References: <AANLkTinF=5BX-MagFPfcisa-ionNs2dj7u7EWDLKg_9k@mail.gmail.com>
	<AANLkTimWQh6aH8FOd3rT249pM0TJBSoDV12psf9EAWZN@mail.gmail.com>
Message-ID: <4D8824D6.1080107@coppice.org>

I'm not clear how you can get a PBX SIP CONNECT approved. A lot of the 
document comes down to how you configure and use the system. Obviously, 
a product could present a roadblock that prevents SIP CONNECT compliance 
in a working setup, but I doubt that many products would do that.

Characterising the SIP forum as a who's who of retarded SIP providers 
and PBXes is a little unfair. Practically everyone in the VoIP business 
is in that list. Capability certainly doesn't look like a prerequisite, 
though. :-\

Steve


On 03/22/2011 12:07 PM, Michael Collins wrote:
> No, we have not paid them thousands of dollars for them to say that we 
> are "SIP CONNECT Compliant." That being said, I read through some of 
> the 1.1 specs. I didn't see anything that was egregious, but I also 
> did not look at all 45 pages. This much I can tell you: to be "SIP 
> CONNECT Compliant" you would need to program your dialplan for some 
> specifics. Here's an example:
>
> Section 13 of the spec says:
>
> *13 Emergency Services*
>
> The SIP-PBX MUST have a dial plan that recognizes emergency calls.
>
>
> This will differ from one country to the next and is a function of the 
> administrator's dialplan configuration. There are no inherent 
> limitations in FreeSWITCH to prevent you from being up to spec on 
> this. However, out of the box with a default installation you won't 
> have this already configured.
>
> My guess is that we could write tests that demonstrate that FreeSWITCH 
> *could* be SIP CONNECT compliant, however I doubt that would be of 
> much value because you have to cough up a minimum of $2500 to have 
> their seal of approval. Beyond that look at their membership list:
>
> http://www.sipforum.org/component/option,com_fullmember/Itemid,195/
>
> It's a who's who of retarded SIP providers and PBXes. They've got 
> Sonus on their list for cying out loud! That alone should invalidate 
> the entire SIP CONNECT venture IMHO.
>
> So, I would answer your supplier this way: As far as I know there 
> aren't any limitations in FreeSWITCH that would prevent it from 
> complying with the SIP CONNECT recommendations.
>
> Oh, and GLWT! :)
>
> -MC
>
> On Mon, Mar 21, 2011 at 11:46 AM, Yehavi Bourvine 
> <yehavi.bourvine at gmail.com <mailto:yehavi.bourvine at gmail.com>> wrote:
>
>     Hello,
>       We published an RFP for our SIP connection to the PSTN. One of
>     the suppliers asked whether Freeswitch complies with
>     the SIP CONNECT recomendation. I have to answer them soon, and
>     don't have enough time to read and evaluate the entire document at
>     present.
>     Does Freeswitch fullfills all the recomendations there?
>                             Thanks, __Yehavi:
>



From msc at freeswitch.org  Tue Mar 22 07:39:12 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 21 Mar 2011 21:39:12 -0700
Subject: [Freeswitch-users] Does Freeswitch complies with SIP Connect
 technical recommendation?
In-Reply-To: <4D8824D6.1080107@coppice.org>
References: <AANLkTinF=5BX-MagFPfcisa-ionNs2dj7u7EWDLKg_9k@mail.gmail.com>
	<AANLkTimWQh6aH8FOd3rT249pM0TJBSoDV12psf9EAWZN@mail.gmail.com>
	<4D8824D6.1080107@coppice.org>
Message-ID: <AANLkTi=UxjGo=++otrd7vrKL_xJdMx6A4N-cQ-Yx9_Kt@mail.gmail.com>

On Mon, Mar 21, 2011 at 9:25 PM, Steve Underwood <steveu at coppice.org> wrote:

> I'm not clear how you can get a PBX SIP CONNECT approved. A lot of the
> document comes down to how you configure and use the system. Obviously,
> a product could present a roadblock that prevents SIP CONNECT compliance
> in a working setup, but I doubt that many products would do that.
>
> Characterising the SIP forum as a who's who of retarded SIP providers
> and PBXes is a little unfair. Practically everyone in the VoIP business
> is in that list. Capability certainly doesn't look like a prerequisite,
> though. :-\
>

Hehe, true enough. Perhaps I was a little harsh. Still, you are quite right
about the compliance test being incredibly subjective and capability not
being a prerequisite. Sonus and ShoreTel are not exactly known for their SIP
interop features.

-MC
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From msc at freeswitch.org  Tue Mar 22 07:51:46 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 21 Mar 2011 21:51:46 -0700
Subject: [Freeswitch-users] No ringback/audio suddenly
In-Reply-To: <AANLkTikX6J1QnBDxxdRKdL9MDDEseoGZVp9AkS-62M-L@mail.gmail.com>
References: <AANLkTikX6J1QnBDxxdRKdL9MDDEseoGZVp9AkS-62M-L@mail.gmail.com>
Message-ID: <AANLkTikpwt+-s9wc2qEhxcKr1DJz+uXhkGv=nz2TUf+C@mail.gmail.com>

These are on outbound calls through pennytel, correct? Find out if the 183
that they send has media or not. Most likely they're doing something silly
like sending a 183 w/media but not actually sending media. A pcap analyzed
in wireshark would help you see exactly what they are doing. Once you know
that then you can probably figure out what the next step should be.

-MC

On Mon, Mar 21, 2011 at 3:30 AM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Hi!
>
> I have an issue with no ringback or audio that has just started in the last
> few weeks.  No changes in FS, which has been rock solid for a long time.  I
> think my voip provider Pennytel has upgraded or changed their servers.
>
> Anyway, the difference in signalling is as follows:
>
> This does not work:
> INVITE sip:number at sip.pennytel.com SIP/2.0
> SIP/2.0 100 Trying
> SIP/2.0 401 Unauthorized
> ACK sip:number at sip.pennytel.com SIP/2.0
> INVITE sip:number at sip.pennytel.com SIP/2.0
> SIP/2.0 100 Trying
> SIP/2.0 183 Session Progress
> SIP/2.0 180 Ringing
> SIP/2.0 200 OK
> ACK sip:202.85.243.105:5061 SIP/2.0
> BYE sip:202.85.243.105:5061 SIP/2.0
> SIP/2.0 200 OK
>
> This works
> INVITE sip:number at sip.pennytel.com SIP/2.0
> SIP/2.0 100 Trying
> SIP/2.0 401 Unauthorized
> ACK sip:number at sip.pennytel.com SIP/2.0
> INVITE sip:number at sip.pennytel.com SIP/2.0
> SIP/2.0 100 Trying
> SIP/2.0 183 Session Progress
> SIP/2.0 200 OK
> ACK sip:202.85.243.105:5061 SIP/2.0
>
> And I see after the Ringing signal in the case that does not work (there is
> no ringing signal in the case that does work):
> 2011-03-21 20:28:25.677621 [DEBUG] sofia.c:4641 Channel
> sofia/external/0393011079 skipping state [proceeding][180] received after a
> RINGING
>
> Can anyone point me to a reason for the difference in signalling and what I
> can do to get audio 100% of the time?
>
> THanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Mar 22 07:53:10 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 21 Mar 2011 21:53:10 -0700
Subject: [Freeswitch-users] can't make mod_shout on latest git
In-Reply-To: <AANLkTimeCqmS58vrV89TUfSRGP1QW4Krg=5d7YPS+Zhc@mail.gmail.com>
References: <AANLkTimeCqmS58vrV89TUfSRGP1QW4Krg=5d7YPS+Zhc@mail.gmail.com>
Message-ID: <AANLkTi=ZgarsQdTum9ZCZ+HfBPinaaFK=hu-Fn_bJ08V@mail.gmail.com>

Neil,

Try doing a make current and see what happens.

-MC

On Thu, Mar 17, 2011 at 3:21 AM, Neil Patel <neilp at cs.stanford.edu> wrote:

> >From latest git, I'm getting this error on make all:
>
> making all mod_shout
> Creating mod_shout.la...
> quiet_libtool: link: cannot find the library
> `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib//libvorbis.la' or
> unhandled argument `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib//
> libvorbis.la'
> make[5]: *** [mod_shout.la] Error 1
> make[4]: *** [all] Error 1
> make[3]: *** [mod_shout-all] Error 1
> make[2]: *** [all-recursive] Error 1
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
>
>
> Looks like there is a path specification error/typo (I have libvorbis.lain /usr/lib/)? Not sure how to fix, but would be great if someone could.
>
> Best,
> Neil
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Mar 22 07:55:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 21 Mar 2011 21:55:02 -0700
Subject: [Freeswitch-users] Diversion Header
In-Reply-To: <1300469358.2342.6.camel@gustavo-laptop>
References: <1300469358.2342.6.camel@gustavo-laptop>
Message-ID: <AANLkTikEf7wQEmD+0jnf==o+Tk_wqg=56P9vtOCrP=zX@mail.gmail.com>

Could you pastebin a console debug with siptrace? Those details will help us
to see what's going on.

-MC

On Fri, Mar 18, 2011 at 10:29 AM, Gustavo Espeche <
gustavo.espeche at easyipcall.com> wrote:

> Hi all we are having a problem with a customer that send the call with
> header Diversion, and our Freeswitch reject the call because don't
> understand the header.
> Some one know how can fix this problem?
> Best Regards.
>
> Gustavo Espeche
> www.easyipcall.com
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Mar 22 08:02:27 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 21 Mar 2011 22:02:27 -0700
Subject: [Freeswitch-users] Outbound caller id for a specific gateway?
In-Reply-To: <AANLkTin=M3MJKB5QA86iL+hkhkZncTQ8eiKQJM2t6ap5@mail.gmail.com>
References: <AANLkTin=M3MJKB5QA86iL+hkhkZncTQ8eiKQJM2t6ap5@mail.gmail.com>
Message-ID: <AANLkTimqg7TRBULWR2__kxu32K4QuHQi1ChgsAZGiW+G@mail.gmail.com>

How about set the caller id in vars.xml:

<X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/>

Set it to the most common value and then you only have to do something in
bridges and originates that need a CID different from what you set in
vars.xml...

-MC

On Sat, Mar 19, 2011 at 4:27 AM, Dmitry Bely <dmitry.bely at gmail.com> wrote:

> My VoIP provider requires a specific caller ID set for an outbound
> call otherwise the call is rejected. Currently I set it just before
> bridge
>
>      <action application="set" data="effective_caller_id_name=..."/>
>      <action application="bridge" data="..."/>
>
> But it's tedious as there is a number of bridge commands in the
> dialplan and I still have to explicitly specify the caller id for
> "originate" command in the FreeSWITCH console. Is it possible to force
> an outbound caller id on a gateway basis?
>
> - Dmitry Bely
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Mar 22 08:02:57 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 21 Mar 2011 22:02:57 -0700
Subject: [Freeswitch-users] More then one number for registered user
In-Reply-To: <4D84EEB7.3030600@tagnet.ru>
References: <4D84EEB7.3030600@tagnet.ru>
Message-ID: <AANLkTik+fa5gRUupYcGUz2RkOok=0ht4ufRjqS2kOWe1@mail.gmail.com>

Why?

On Sat, Mar 19, 2011 at 10:58 AM, Boris Kovalenko <boris at tagnet.ru> wrote:

> Hello!
>
>     Is this possible to have one registered user and more then one
> associated number with it (without tricks with dialplan)? For example:
> <user id="XXX" number-alias="YYY,ZZZ.....">
>
> --
> Regards,
> Boris
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Mar 22 08:19:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 21 Mar 2011 22:19:02 -0700
Subject: [Freeswitch-users] I need to register all of my DIDs with a
 single SIP trunk..
In-Reply-To: <AANLkTikS8rKHBoXxARwG6=qBUqvyNtxBj6d-qpt3-Sg_@mail.gmail.com>
References: <AANLkTikS8rKHBoXxARwG6=qBUqvyNtxBj6d-qpt3-Sg_@mail.gmail.com>
Message-ID: <AANLkTimGB3V-K56XOfByKfLLUqBrc2-zrZPDwUC3YZ7+@mail.gmail.com>

Are you sure that an explicit registration is required? Or are they merely
sending an auth challenge when you make an outbound call? In any case, you
can create a gateway for each DID. You can put them all in one file or you
can have them each in their own file. You might want to name each gateway
after its corresponding DID number, or name it "gw###" where ### is the DID
number. This will make it easier to do your dialplan. For example, if you
know that you want to send CID "1234567890" then you know that the
corresponding gateway is "gw1234567890".

So, try making a few unique gateways and then do some testing. Let us know
how it goes. If you  run into any trouble then capture some debug output and
drop it into pastebin.freeswitch.org and we'll help you figure it out.

-MC

On Mon, Mar 21, 2011 at 10:53 AM, Randy Andrade <randy.andrade at gmail.com>wrote:

> Greetings,
>
> I'm hoping someone can help me out here. I'm relatively new to FS, but not
> to general telephony and I have a small amount of SIP experience.
>
> My FS installation is, so far, a fairly straightforward PBX setup with a
> single SIP trunk with 21 DIDs to the ITSP that I work at. I have everything
> working as far as inbound and outbound calls, except that in order for me to
> use a DID as the outbound caller id, that DID needs to be explicitly
> registered with the ITSP, and not just the pilot number for the SIP trunk.
> When I try to create a second gateway *.xml file to register a second DID, I
> get an error "Registration Failed with status Operation has no matching
> challenge [904]" on the FS console. As I mentioned, I work at this ITSP and
> I program similar SIP trunks into Adtran TA9xx boxes for SIP over T1
> deliveries. On the Adtrans, I'm able to define a single SIP trunk, and have
> multiple register statements within that trunk, one for each DID. I'm
> wondering if there's a way to do something similar in FS.


> I'm running a very recent (I update from the GIT tree about twice a week)
> version of FS on a Debian Linux OS, and, I have several local extensions
> working and able to call in and out; I just want to be able to outpulse
> various DIDs as my caller-id without getting calls rejected. Here's a copy
> of my gateway .xml file, with relevant information changed:
>
> <include>
>   <gateway name="gatewayname">
>   <!--/// account username *required* ///-->
>   <param name="username" value="1234567890"/>
>   <!--/// auth realm: *optional* same as gateway name, if blank ///-->
>   <!--<param name="realm" value="asterlink.com"/>-->
>   <!--/// username to use in from: *optional* same as  username, if blank
> ///-->
>   <!--<param name="from-user" value="cluecon"/>-->
>   <!--/// domain to use in from: *optional* same as  realm, if blank ///-->
>   <param name="from-domain" value="gatewayname"/>
>   <!--/// account password *required* ///-->
>   <param name="password" value="XXXXXXXX"/>
>   <!--/// extension for inbound calls: *optional* same as username, if
> blank ///-->
>   <!--<param name="extension" value="cluecon"/>-->
>   <!--/// proxy host: *optional* same as realm, if blank ///-->
>   <param name="proxy" value="proxyname"/>
>   <!--/// send register to this proxy: *optional* same as proxy, if blank
> ///-->
>   <!--<param name="register-proxy" value="mysbc.com"/>-->
>   <!--/// expire in seconds: *optional* 3600, if blank ///-->
>   <!--<param name="expire-seconds" value="60"/>-->
>   <!--/// do not register ///-->
>   <param name="register" value="true"/>
>   <!-- which transport to use for register -->
>   <!--<param name="register-transport" value="tcp"/>-->
>   <!--How many seconds before a retry when a failure or timeout occurs -->
>   <!--<param name="retry-seconds" value="30"/>-->
>   <!--Use the callerid of an inbound call in the from field on outbound
> calls via this gateway -->
>   <param name="caller-id-in-from" value="true"/>
>   <!--extra sip params to send in the contact-->
>   <!--<param name="contact-params" value="tport=tcp"/>-->
>   <!--send an options ping every x seconds, failure will unregister and/or
> mark it down-->
>   <param name="ping" value="30"/>
>   <param name="extension-in-contact" value="true"/>
>   </gateway>
> </include>
>
>
> Again, I don't know if I would do it here (I would assume so) but I'm
> wondering if there's some way (maybe multiple "extension" statements? I
> dunno..) to register multiple DIDs in the same gateway / SIP trunk.
>
> Thanks!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Mar 22 08:26:37 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 21 Mar 2011 22:26:37 -0700
Subject: [Freeswitch-users] send custom tcp message from esl
In-Reply-To: <921841.76258.qm@web132304.mail.ird.yahoo.com>
References: <921841.76258.qm@web132304.mail.ird.yahoo.com>
Message-ID: <AANLkTim0f1WN-utJWtTWmozFJZtjj9fvEit2ouaLYbD=@mail.gmail.com>

Are you trying to send a simple string to the event socket port on
FreeSWITCH? If so, what message are you sending? I'm not aware of any event
socket messages that ESL cannot handle.

-MC

On Mon, Mar 21, 2011 at 4:10 PM, Ejay Greeves <ejay.greeves at yahoo.com>wrote:

> I am using esl in Ruby I want to send tcp messages to a server.  When I
> tried to make my tcpsocket class it did not work it must be messing with the
> socket setup of esl. Is there a way without the need to manage the sockets
> seperatley to use els methods to send out custom tcp message. All I need is
> for a simple string to be fired to tcp port. How can I do this from inside
> esl.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From boris at tagnet.ru  Tue Mar 22 08:32:53 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 22 Mar 2011 10:32:53 +0500
Subject: [Freeswitch-users] More then one number for registered user
In-Reply-To: <AANLkTik+fa5gRUupYcGUz2RkOok=0ht4ufRjqS2kOWe1@mail.gmail.com>
References: <4D84EEB7.3030600@tagnet.ru>
	<AANLkTik+fa5gRUupYcGUz2RkOok=0ht4ufRjqS2kOWe1@mail.gmail.com>
Message-ID: <4D883485.20707@tagnet.ru>

Hello!

     Gateway, that has no ability to properly register all associated 
numbers. Rare condition and there is workaround with personal profile on 
so on, but simple way is always better.
> Why?
>
> On Sat, Mar 19, 2011 at 10:58 AM, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     Hello!
>
>         Is this possible to have one registered user and more then one
>     associated number with it (without tricks with dialplan)? For example:
>     <user id="XXX" number-alias="YYY,ZZZ.....">
>
>     --
>     Regards,
>     Boris
>
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991


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From msc at freeswitch.org  Tue Mar 22 08:33:24 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 21 Mar 2011 22:33:24 -0700
Subject: [Freeswitch-users] Delayed start of tone_detect
In-Reply-To: <AANLkTims-MfPkq_tr-JQd_ih=gPd1i8gXSFSX2tF_m8y@mail.gmail.com>
References: <AANLkTims-MfPkq_tr-JQd_ih=gPd1i8gXSFSX2tF_m8y@mail.gmail.com>
Message-ID: <AANLkTi=s1iwAm9qB3=y97U8CmLs+T=pL9ZzAToE2N5Ed@mail.gmail.com>

Add this to your dialplan...

<extension name="start tone detect">
  <condition field="destination_number" data="custom_start_tone_detect">
    <action application="sleep" data="500"/>
    <action application="tone_detect" data="MYTONE 680 ro +5000 set
MYTONE=1"/>
  </condition>
</extension>

...then change your export...
<action application="export"
data="nolocal:execute_on_answer=execute_extension
custom_start_tone_detect"/>

Let us know if that does the trick.

-MC

On Mon, Mar 21, 2011 at 12:33 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:

> I start the tone_detect on answer using "execute_on_answer" as:
>
> <action application="export" data="nolocal:execute_on_answer=tone_detect
> MYTONE 680 ro +5000 set MYTONE=1" />
>
> but I have the necessity to start the tone_detect for example 500ms after
> answer: there is a way to do that in dialplan?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ayhkor at gmail.com  Tue Mar 22 08:46:02 2011
From: ayhkor at gmail.com (deniro)
Date: Tue, 22 Mar 2011 01:46:02 -0400
Subject: [Freeswitch-users] playAndGetDigits not working?
Message-ID: <AANLkTimpyZHpXOf9ZHaKCcXdZH=mo-2y2rWJLWji=Dda@mail.gmail.com>

I am unable to get the pin number and pass it to $digits ($digits comes
blank)
appreciate any feedback
thx
deniro--


my($digits)=$session->playAndGetDigits(1,5,1,8000,"#","conference/conf-pin.wav","ivr/ivr-that_was_an_invalid_entry.wav","","^7\d{4}\$");
or
my($digits)=$session->playAndGetDigits(1,5,1,8000,"#","conference/conf-pin.wav","ivr/ivr-that_was_an_invalid_entry.wav","",
"\d <file://d/>+");
freeswitch::console_log("info", "*Got dtmf: .. $digits ..*\n");

freeswitch console
2011-03-22 01:29:26.208664 [INFO] switch_cpp.cpp:1177* Got dtmf: ..  ..
*2011-03-22 01:29:26.208664 [ERR] mod_perl.c:72 [require
'/opt/freeswitch/scripts/fs1.pl';]
/opt/freeswitch/scripts/fs1.pl did not return a true value at (eval 2) line
1.
 each case getting "that was invalid PIN"
entering pin starting with 7 maximum 5 digits in first case
or any digits upto maximum 5 digits in second case
tried also putting "#" at the end when dialing

none worked
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From dmitry.bely at gmail.com  Tue Mar 22 10:49:53 2011
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Tue, 22 Mar 2011 10:49:53 +0300
Subject: [Freeswitch-users] Outbound caller id for a specific gateway?
In-Reply-To: <AANLkTimqg7TRBULWR2__kxu32K4QuHQi1ChgsAZGiW+G@mail.gmail.com>
References: <AANLkTin=M3MJKB5QA86iL+hkhkZncTQ8eiKQJM2t6ap5@mail.gmail.com>
	<AANLkTimqg7TRBULWR2__kxu32K4QuHQi1ChgsAZGiW+G@mail.gmail.com>
Message-ID: <AANLkTimnorNHZa_3KqeqgecKjgHo4PLQWsm8+xH_FJMQ@mail.gmail.com>

On Tue, Mar 22, 2011 at 8:02 AM, Michael Collins <msc at freeswitch.org> wrote:
> How about set the caller id in vars.xml:
> <X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/>
> Set it to the most common value and then you only have to do something in
> bridges and originates that need a CID different from what you set in
> vars.xml...

I have a number of gateways that require different CIDs. So there is
no "most common" value. I have managed to make a proper dialplan but
originate is still tedious... I just wonder if there is more clean way
(something like caller-id-in-from gateway parameter)

> -MC
>
> On Sat, Mar 19, 2011 at 4:27 AM, Dmitry Bely <dmitry.bely at gmail.com> wrote:
>>
>> My VoIP provider requires a specific caller ID set for an outbound
>> call otherwise the call is rejected. Currently I set it just before
>> bridge
>>
>> ? ? ?<action application="set" data="effective_caller_id_name=..."/>
>> ? ? ?<action application="bridge" data="..."/>
>>
>> But it's tedious as there is a number of bridge commands in the
>> dialplan and I still have to explicitly specify the caller id for
>> "originate" command in the FreeSWITCH console. Is it possible to force
>> an outbound caller id on a gateway basis?

- Dmitry Bely


From wstephen80 at gmail.com  Tue Mar 22 10:54:13 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 22 Mar 2011 08:54:13 +0100
Subject: [Freeswitch-users] Delayed start of tone_detect
In-Reply-To: <AANLkTi=s1iwAm9qB3=y97U8CmLs+T=pL9ZzAToE2N5Ed@mail.gmail.com>
References: <AANLkTims-MfPkq_tr-JQd_ih=gPd1i8gXSFSX2tF_m8y@mail.gmail.com>
	<AANLkTi=s1iwAm9qB3=y97U8CmLs+T=pL9ZzAToE2N5Ed@mail.gmail.com>
Message-ID: <AANLkTimm7+5GxsRfJNRcRcRAqmtCUBwv8UWrn4Qk2_6r@mail.gmail.com>

Thank you Michael, with this trick it works fine!

On Tue, Mar 22, 2011 at 6:33 AM, Michael Collins <msc at freeswitch.org> wrote:

> Add this to your dialplan...
>
> <extension name="start tone detect">
>   <condition field="destination_number" data="custom_start_tone_detect">
>     <action application="sleep" data="500"/>
>     <action application="tone_detect" data="MYTONE 680 ro +5000 set
> MYTONE=1"/>
>   </condition>
> </extension>
>
> ...then change your export...
> <action application="export"
> data="nolocal:execute_on_answer=execute_extension
> custom_start_tone_detect"/>
>
> Let us know if that does the trick.
>
> -MC
>
> On Mon, Mar 21, 2011 at 12:33 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> I start the tone_detect on answer using "execute_on_answer" as:
>>
>> <action application="export" data="nolocal:execute_on_answer=tone_detect
>> MYTONE 680 ro +5000 set MYTONE=1" />
>>
>> but I have the necessity to start the tone_detect for example 500ms after
>> answer: there is a way to do that in dialplan?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From yehavi.bourvine at gmail.com  Tue Mar 22 11:23:45 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 22 Mar 2011 10:23:45 +0200
Subject: [Freeswitch-users] Does Freeswitch complies with SIP Connect
 technical recommendation?
In-Reply-To: <AANLkTi=UxjGo=++otrd7vrKL_xJdMx6A4N-cQ-Yx9_Kt@mail.gmail.com>
References: <AANLkTinF=5BX-MagFPfcisa-ionNs2dj7u7EWDLKg_9k@mail.gmail.com>
	<AANLkTimWQh6aH8FOd3rT249pM0TJBSoDV12psf9EAWZN@mail.gmail.com>
	<4D8824D6.1080107@coppice.org>
	<AANLkTi=UxjGo=++otrd7vrKL_xJdMx6A4N-cQ-Yx9_Kt@mail.gmail.com>
Message-ID: <AANLkTimPpF5hVF698WdiTEJWybxr2rehUq+zFmFuopsP@mail.gmail.com>

I skimmed through the document and it seems that FreeSwitch can be tailored
to comply with most of the requirements by the end user. There are 3 issues
which I am not sure about:

Section 12.1 states that call transfer should be done with INVITE/Re-INVITE
and not by REFER. From the SIP traces I've done I see that FS uses REFER to
transfer a call.
Section 14.1: Does FS accepts INVITE with no SDP inside?
Section 14.3: Does FS supports RFC-4733? This section allows also RFC-2833
for those who do not support 4733.

                                      Thanks, __Yehavi:
 2011/3/22 Michael Collins <msc at freeswitch.org>

>
>
>  On Mon, Mar 21, 2011 at 9:25 PM, Steve Underwood <steveu at coppice.org>wrote:
>
>> I'm not clear how you can get a PBX SIP CONNECT approved. A lot of the
>> document comes down to how you configure and use the system. Obviously,
>> a product could present a roadblock that prevents SIP CONNECT compliance
>> in a working setup, but I doubt that many products would do that.
>>
>> Characterising the SIP forum as a who's who of retarded SIP providers
>> and PBXes is a little unfair. Practically everyone in the VoIP business
>> is in that list. Capability certainly doesn't look like a prerequisite,
>> though. :-\
>>
>
> Hehe, true enough. Perhaps I was a little harsh. Still, you are quite right
> about the compliance test being incredibly subjective and capability not
> being a prerequisite. Sonus and ShoreTel are not exactly known for their SIP
> interop features.
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From wstephen80 at gmail.com  Tue Mar 22 11:29:42 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 22 Mar 2011 09:29:42 +0100
Subject: [Freeswitch-users] Delayed start of tone_detect
In-Reply-To: <AANLkTimm7+5GxsRfJNRcRcRAqmtCUBwv8UWrn4Qk2_6r@mail.gmail.com>
References: <AANLkTims-MfPkq_tr-JQd_ih=gPd1i8gXSFSX2tF_m8y@mail.gmail.com>
	<AANLkTi=s1iwAm9qB3=y97U8CmLs+T=pL9ZzAToE2N5Ed@mail.gmail.com>
	<AANLkTimm7+5GxsRfJNRcRcRAqmtCUBwv8UWrn4Qk2_6r@mail.gmail.com>
Message-ID: <AANLkTikn1V3ypmbZW4NQ6qSgRy_7ENdOWT5eWrCHs-C0@mail.gmail.com>

I have tested this trick with 1 call and it worked fine but when tested on a
production environment doesn't work.

The problem seems to be the "sleep" called by "execute_extension" executed
on answer.

If I activate this extension, I see in freeswitch log:

2011-03-22 09:18:10.191304 [CRIT] switch_core_session.c:1646 Throttle Error!
1326
2011-03-22 09:18:10.193260 [CRIT] switch_core_session.c:1646 Throttle Error!
1326
2011-03-22 09:18:10.194796 [CRIT] switch_core_session.c:1646 Throttle Error!
1323
2011-03-22 09:18:10.194796 [CRIT] switch_core_session.c:1646 Throttle Error!
1323
2011-03-22 09:18:10.199238 [CRIT] switch_core_session.c:1646 Throttle Error!
1317
2011-03-22 09:18:10.204815 [CRIT] switch_core_session.c:1646 Throttle Error!
1314
2011-03-22 09:18:10.204815 [CRIT] mod_sofia.c:3896 Error Creating Session
2011-03-22 09:18:10.214916 [CRIT] switch_core_session.c:1646 Throttle Error!
1302
2011-03-22 09:18:10.214916 [CRIT] mod_sofia.c:3896 Error Creating Session
2011-03-22 09:18:10.377962 [CRIT] switch_time.c:788 Over Session Rate of
200!
2011-03-22 09:18:33.938094 [CRIT] switch_core_session.c:1646 Throttle Error!
1266
2011-03-22 09:18:33.938094 [CRIT] switch_core_session.c:1646 Throttle Error!
1266
2011-03-22 09:18:33.939077 [CRIT] switch_core_session.c:1646 Throttle Error!
1266
2011-03-22 09:18:33.939077 [CRIT] switch_core_session.c:1646 Throttle Error!
1266
2011-03-22 09:18:33.944077 [CRIT] switch_core_session.c:1646 Throttle Error!
1264
2011-03-22 09:18:33.944077 [CRIT] mod_sofia.c:3896 Error Creating Session

Stephen

On Tue, Mar 22, 2011 at 8:54 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> Thank you Michael, with this trick it works fine!
>
>
> On Tue, Mar 22, 2011 at 6:33 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Add this to your dialplan...
>>
>> <extension name="start tone detect">
>>    <condition field="destination_number" data="custom_start_tone_detect">
>>     <action application="sleep" data="500"/>
>>     <action application="tone_detect" data="MYTONE 680 ro +5000 set
>> MYTONE=1"/>
>>   </condition>
>> </extension>
>>
>> ...then change your export...
>> <action application="export"
>> data="nolocal:execute_on_answer=execute_extension
>> custom_start_tone_detect"/>
>>
>> Let us know if that does the trick.
>>
>> -MC
>>
>> On Mon, Mar 21, 2011 at 12:33 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>
>>> I start the tone_detect on answer using "execute_on_answer" as:
>>>
>>> <action application="export" data="nolocal:execute_on_answer=tone_detect
>>> MYTONE 680 ro +5000 set MYTONE=1" />
>>>
>>> but I have the necessity to start the tone_detect for example 500ms after
>>> answer: there is a way to do that in dialplan?
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From benkokakao at gmail.com  Tue Mar 22 13:09:40 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Tue, 22 Mar 2011 11:09:40 +0100
Subject: [Freeswitch-users] New FreeSWITCH Module: mod_ladspa
In-Reply-To: <1300666884615-6190839.post@n2.nabble.com>
References: <AANLkTimPo4U2bzf+K4ucRq3xZHHDBnjFouhXDd_t+Fzk@mail.gmail.com>
	<1300666884615-6190839.post@n2.nabble.com>
Message-ID: <AANLkTikRa9LxKEfZzUYkPdfZObr4W17YxGQSZNobdNqj@mail.gmail.com>

On 21 March 2011 01:21, mazilo <Nabble at slickdeals.endjunk.com> wrote:
> Just curious, do we have to download and install all plugins mentioned in
> the wiki of ?http://wiki.freeswitch.org/wiki/Mod_ladspa mod ladspa ?just in
> order to sound like Cher?

Sounds like an opportunity for a prank on April Fools' Day :-)

I have problems getting some plugins to load though:

http://pastebin.freeswitch.org/15780 (See line 1068 and 1072)

/usr/lib/ladspa/tap_chorusflanger.so and
/usr/lib/ladspa/phasers_1217.so exist, the env-path in mod_ladspa.c
fits("LADSPA_PATH=/usr/lib/ladspa/:/usr/local/lib/ladspa") - i've
moved all libs from /usr/lib64/ladspa to /usr/lib/ladspa and deleted
the directory according to the setup-instructions.

The example-dialplan is unchanged, besides using a local conference-room.

I'm on Centos5.5, all packages mentioned in the wiki are installed.

Regards
Christian


From dmitry.bely at gmail.com  Tue Mar 22 13:11:49 2011
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Tue, 22 Mar 2011 13:11:49 +0300
Subject: [Freeswitch-users] inbound-late-negotiation and attended transfer
Message-ID: <AANLkTimDT=699aCgz-h9yk_+FyXb_tf9oDa=KMMQRZ04@mail.gmail.com>

I have G722 and PCMU enabled for internal extensions and PCMU only for
an external gateway. Inbound-late-negotiation parameter is set so then
an incoming call arrives PCMU is used without transcoding. But it goes
worse when transfer is involved:

call 1: gw <-- PCMU--> FreeSwitch <-- PCMU --> operator
call 2: operator <-- G722 --> FreeSwitch <-- G722--> extension
transfer (refer): gw <-- PCMU--> FreeSwitch <-- G722--> extension

Is it possible to renegotiate a codec during transfer to get rid of transcoding?

- Dmitry Bely


From kris at kriskinc.com  Tue Mar 22 17:31:07 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 22 Mar 2011 10:31:07 -0400
Subject: [Freeswitch-users] Does Freeswitch complies with SIP Connect
 technical recommendation?
In-Reply-To: <AANLkTimPpF5hVF698WdiTEJWybxr2rehUq+zFmFuopsP@mail.gmail.com>
References: <AANLkTinF=5BX-MagFPfcisa-ionNs2dj7u7EWDLKg_9k@mail.gmail.com>
	<AANLkTimWQh6aH8FOd3rT249pM0TJBSoDV12psf9EAWZN@mail.gmail.com>
	<4D8824D6.1080107@coppice.org>
	<AANLkTi=UxjGo=++otrd7vrKL_xJdMx6A4N-cQ-Yx9_Kt@mail.gmail.com>
	<AANLkTimPpF5hVF698WdiTEJWybxr2rehUq+zFmFuopsP@mail.gmail.com>
Message-ID: <AANLkTinKmtYjKQj7p4Bv2CaknLa_jtz81WDOnHnDVf6K@mail.gmail.com>

FreeSWITCH certainly supports most of the features/standards required
by SIPConnect 1.1 (codec support, TLS, Caller ID, etc).  It's a matter
of configuration.  It wouldn't be that difficult for someone to create
a SIPConnect 1.1 compliant SIP profile and dialplan.

12.1 - I'm not sure but doesn't sound that difficult.
14.1 - Yes (3PCC)
14.3 - Yes.  RFC 4733 is essentially the same as RFC 2833.  The
difference is that RFC 4733 requires SRTP to be in use (supported by
FreeSWITCH).

Paying to get certified, however, is ridiculous.

I have a philosophical problem with vendor and (to a lesser extent)
third party certifications.  It's a great way to still extort money
out of people when using a standard protocol.  Avaya, for example,
requires thousands of dollars a year to maintain an Avaya DevConnect
membership so you can do SIP interop with their equipment.  Hmmm, I
wonder what revenue this "certification" is supplanting for them?
Perhaps revenue they used to generate from proprietary protocols,
handsets, and extensions?

On Tue, Mar 22, 2011 at 4:23 AM, Yehavi Bourvine
<yehavi.bourvine at gmail.com> wrote:
> I skimmed through the document and it seems that FreeSwitch can be tailored
> to comply with most of the requirements by the end user. There are 3 issues
> which I am not sure about:
>
> Section 12.1 states that call transfer should be done with INVITE/Re-INVITE
> and not by REFER. From the SIP traces I've done I see that FS uses REFER to
> transfer a call.
> Section 14.1: Does FS accepts INVITE with no SDP inside?
> Section 14.3: Does FS supports RFC-4733? This section allows also RFC-2833
> for those who do not support 4733.
>
> ????????????????????????????????????? Thanks, __Yehavi:
> 2011/3/22 Michael Collins <msc at freeswitch.org>
>>
>> On Mon, Mar 21, 2011 at 9:25 PM, Steve Underwood <steveu at coppice.org>
>> wrote:
>>>
>>> I'm not clear how you can get a PBX SIP CONNECT approved. A lot of the
>>> document comes down to how you configure and use the system. Obviously,
>>> a product could present a roadblock that prevents SIP CONNECT compliance
>>> in a working setup, but I doubt that many products would do that.
>>>
>>> Characterising the SIP forum as a who's who of retarded SIP providers
>>> and PBXes is a little unfair. Practically everyone in the VoIP business
>>> is in that list. Capability certainly doesn't look like a prerequisite,
>>> though. :-\
>>
>> Hehe, true enough. Perhaps I was a little harsh. Still, you are quite
>> right about the compliance test being incredibly subjective and capability
>> not being a prerequisite. Sonus and ShoreTel are not exactly known for their
>> SIP interop features.
>> -MC
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner


From pkelly at gmail.com  Tue Mar 22 17:31:07 2011
From: pkelly at gmail.com (Pete Kelly)
Date: Tue, 22 Mar 2011 14:31:07 +0000
Subject: [Freeswitch-users] Lua playAndCollectDigits timeout problem.
 Possible bug
In-Reply-To: <AANLkTinzvnvJ=5oCaNZSP59kniMtPthJPHSGU7=vikrV@mail.gmail.com>
References: <AANLkTin22=Y6pBYSYX885C10HsGV_jXHaZP8w2nMebEJ@mail.gmail.com>
	<AANLkTinzvnvJ=5oCaNZSP59kniMtPthJPHSGU7=vikrV@mail.gmail.com>
Message-ID: <AANLkTikme467HCXSkZ_uLwRYb3F=W4xg1umrR8BgtjXe@mail.gmail.com>

On 18 March 2011 16:52, Anthony Minessale <anthony.minessale at gmail.com>wrote:

> there is a new final param of digit timeout once you type something.
> In older versions it was set to 0 by default meaning forever but on
> latest its equiv to the global timeout.
>
> add another ,10 to your arg list
>

That worked, thank you again for your help. To clarify are all timeouts in
this function in seconds? I was under the impression (no idea why) that they
would be milliseconds


>
>
> On Fri, Mar 18, 2011 at 10:57 AM, Pete Kelly <pkelly at gmail.com> wrote:
> > Hi, I am trying to invoke a timeout using the
> session:playAndCollectDigits
> > lua function, this is the syntax I am using:
> > call_number = session:playAndGetDigits(1, 20, 1, 10000, "#*",
> > "some-file.wav", "", "[0-9]{1,}");
> > If the user doesn't hit * or # I need the function to return after 10
> > seconds.
> > However this timeout seems to only kick in if I start entering DTMF
> *after*
> > the initial prompt has played. If I start entering DTMF during the
> prompt,
> > the function never returns due to timeout.
> > Has anyone else seen/experienced this ?
> > Pete
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From pkelly at gmail.com  Tue Mar 22 17:31:39 2011
From: pkelly at gmail.com (Pete Kelly)
Date: Tue, 22 Mar 2011 14:31:39 +0000
Subject: [Freeswitch-users] Lua playAndCollectDigits timeout problem.
 Possible bug
In-Reply-To: <AANLkTinzvnvJ=5oCaNZSP59kniMtPthJPHSGU7=vikrV@mail.gmail.com>
References: <AANLkTin22=Y6pBYSYX885C10HsGV_jXHaZP8w2nMebEJ@mail.gmail.com>
	<AANLkTinzvnvJ=5oCaNZSP59kniMtPthJPHSGU7=vikrV@mail.gmail.com>
Message-ID: <AANLkTikAXCxa=AerSTMPNb3FD-e2-QD_rUscxd3J-Xtn@mail.gmail.com>

On 18 March 2011 16:52, Anthony Minessale <anthony.minessale at gmail.com>wrote:

> there is a new final param of digit timeout once you type something.
> In older versions it was set to 0 by default meaning forever but on
> latest its equiv to the global timeout.
>
> add another ,10 to your arg list
>

That worked, thank you again for your help. To clarify are all timeouts in
this function in seconds? I was under the impression (no idea why) that they
would be milliseconds


>
>
> On Fri, Mar 18, 2011 at 10:57 AM, Pete Kelly <pkelly at gmail.com> wrote:
> > Hi, I am trying to invoke a timeout using the
> session:playAndCollectDigits
> > lua function, this is the syntax I am using:
> > call_number = session:playAndGetDigits(1, 20, 1, 10000, "#*",
> > "some-file.wav", "", "[0-9]{1,}");
> > If the user doesn't hit * or # I need the function to return after 10
> > seconds.
> > However this timeout seems to only kick in if I start entering DTMF
> *after*
> > the initial prompt has played. If I start entering DTMF during the
> prompt,
> > the function never returns due to timeout.
> > Has anyone else seen/experienced this ?
> > Pete
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kris at kriskinc.com  Tue Mar 22 17:34:15 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 22 Mar 2011 10:34:15 -0400
Subject: [Freeswitch-users] [Freeswitch-dev] New FreeSWITCH Module:
	mod_ladspa
In-Reply-To: <AANLkTimPo4U2bzf+K4ucRq3xZHHDBnjFouhXDd_t+Fzk@mail.gmail.com>
References: <AANLkTimPo4U2bzf+K4ucRq3xZHHDBnjFouhXDd_t+Fzk@mail.gmail.com>
Message-ID: <AANLkTi=RXmBGW0p_PYxCr=t_Ua6VawiUrZYtu-qgHTTc@mail.gmail.com>

Cool.  Would it be possible to do recqual-like stuff (in real time)
with this module?

On Fri, Mar 18, 2011 at 6:56 PM, Michael Collins <msc at freeswitch.org> wrote:
> Hello all!
> The FreeSWITCH developers have added a cool new module: mod_ladspa. Check
> out the story:
> http://www.freeswitch.org/node/313
> Have fun!
> -Michael
> _______________________________________________
> FreeSWITCH-dev 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
>
>



-- 
Kristian Kielhofner


From infos at madovsky.org  Tue Mar 22 18:32:16 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 22 Mar 2011 11:32:16 -0400
Subject: [Freeswitch-users] [Freeswitch-dev] New FreeSWITCH
	Module:mod_ladspa
References: <AANLkTimPo4U2bzf+K4ucRq3xZHHDBnjFouhXDd_t+Fzk@mail.gmail.com>
	<AANLkTi=RXmBGW0p_PYxCr=t_Ua6VawiUrZYtu-qgHTTc@mail.gmail.com>
Message-ID: <5D32FEEE0569497DB2E3AC1EFD5037A0@e1705>

cool to call with voice confidentiality ;)


----- Original Message ----- 
From: "Kristian Kielhofner" <kris at kriskinc.com>
To: <freeswitch-dev at lists.freeswitch.org>
Cc: <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, March 22, 2011 10:34 AM
Subject: Re: [Freeswitch-users] [Freeswitch-dev] New FreeSWITCH 
Module:mod_ladspa


> Cool.  Would it be possible to do recqual-like stuff (in real time)
> with this module?
>
> On Fri, Mar 18, 2011 at 6:56 PM, Michael Collins <msc at freeswitch.org> 
> wrote:
>> Hello all!
>> The FreeSWITCH developers have added a cool new module: mod_ladspa. Check
>> out the story:
>> http://www.freeswitch.org/node/313
>> Have fun!
>> -Michael
>> _______________________________________________
>> FreeSWITCH-dev 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
>>
>>
>
>
>
> -- 
> Kristian Kielhofner
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From manavid at gmail.com  Tue Mar 22 18:55:40 2011
From: manavid at gmail.com (Moe Navid)
Date: Tue, 22 Mar 2011 08:55:40 -0700
Subject: [Freeswitch-users] Voicemail on GlusterFS
Message-ID: <AANLkTimkv66jjaBTHtT+nDUgo3PsuxEndEi-Z9=Lb0u_@mail.gmail.com>

Hi All,

Has anyone tried using voicemail on GlusterFS?

GlusteFS looks very promising for building distributed file systems. Pandora
Radio recently started using glusterfs
http://www.gluster.com/2011/01/05/gluster-to-help-manage-rapid-data-growth-for-pandora/

I managed to get the recording right for voicemails but retrieving the WAV
files fails. FreeSWITCH gives the following error:

2011-03-22 08:54:06.325001 [ERR] mod_sndfile.c:194 Error Opening File
[/home/moe/freeswitch/storage/voicemail/default/
192.168.1.10/1001/msg_02a108cc-549c-11e0-a9f6-9f1a6f4da0f7.wav] [File
contains data in an unknown format.]

Any thoughts?

Thanks

Moe
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From avi at avimarcus.net  Tue Mar 22 19:13:20 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 22 Mar 2011 18:13:20 +0200
Subject: [Freeswitch-users] Voicemail on GlusterFS
In-Reply-To: <AANLkTimkv66jjaBTHtT+nDUgo3PsuxEndEi-Z9=Lb0u_@mail.gmail.com>
References: <AANLkTimkv66jjaBTHtT+nDUgo3PsuxEndEi-Z9=Lb0u_@mail.gmail.com>
Message-ID: <AANLkTinh4Gbt7PhHCR6Hdhkq3mV_4JxtoOc1tDunX3BR@mail.gmail.com>

This really sounds more like a Gluster question than anything directly
related to FreeSWITCH. I'd try doing basic copies to the file system, and
comparing the hashes (md5?), to make sure it properly reads/writes...
-Avi

p.s. a relevant question for the group, if you want to ask it would still be
only indirectly related to FS: "what file systems do you use in production
to help you manage scaling storage and/or failover and/or hardware fault
tolerance?"

On Tue, Mar 22, 2011 at 5:55 PM, Moe Navid <manavid at gmail.com> wrote:

> Hi All,
>
> Has anyone tried using voicemail on GlusterFS?
>
> GlusteFS looks very promising for building distributed file systems.
> Pandora Radio recently started using glusterfs
> http://www.gluster.com/2011/01/05/gluster-to-help-manage-rapid-data-growth-for-pandora/
>
> I managed to get the recording right for voicemails but retrieving the WAV
> files fails. FreeSWITCH gives the following error:
>
> 2011-03-22 08:54:06.325001 [ERR] mod_sndfile.c:194 Error Opening File
> [/home/moe/freeswitch/storage/voicemail/default/
> 192.168.1.10/1001/msg_02a108cc-549c-11e0-a9f6-9f1a6f4da0f7.wav] [File
> contains data in an unknown format.]
>
> Any thoughts?
>
> Thanks
>
> Moe
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From joshua at seadragons.us  Tue Mar 22 20:13:42 2011
From: joshua at seadragons.us (Joshua Shaffner)
Date: Tue, 22 Mar 2011 13:13:42 -0400
Subject: [Freeswitch-users] inbound-late-negotiation and attended
	transfer
In-Reply-To: <AANLkTimDT=699aCgz-h9yk_+FyXb_tf9oDa=KMMQRZ04@mail.gmail.com>
References: <AANLkTimDT=699aCgz-h9yk_+FyXb_tf9oDa=KMMQRZ04@mail.gmail.com>
Message-ID: <AANLkTi=AP+7ULbOkUBaYn=nuPT7CgHNty=hOUxuvMPsa@mail.gmail.com>

You probably saw this page, http://wiki.freeswitch.org/wiki/Codec_negotiation

I am still learning here but how about experimenting with transcoding
disabled, or setting absolute_codec_string (or codec_string) when
bridging a call?

Let us know how it goes.

HTH,
Joshua

On Tue, Mar 22, 2011 at 6:11 AM, Dmitry Bely <dmitry.bely at gmail.com> wrote:
> I have G722 and PCMU enabled for internal extensions and PCMU only for
> an external gateway. Inbound-late-negotiation parameter is set so then
> an incoming call arrives PCMU is used without transcoding. But it goes
> worse when transfer is involved:
>
> call 1: gw <-- PCMU--> FreeSwitch <-- PCMU --> operator
> call 2: operator <-- G722 --> FreeSwitch <-- G722--> extension
> transfer (refer): gw <-- PCMU--> FreeSwitch <-- G722--> extension
>
> Is it possible to renegotiate a codec during transfer to get rid of transcoding?
>
> - Dmitry Bely
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From msc at freeswitch.org  Tue Mar 22 20:18:37 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Mar 2011 10:18:37 -0700
Subject: [Freeswitch-users] inbound-late-negotiation and attended
	transfer
In-Reply-To: <AANLkTimDT=699aCgz-h9yk_+FyXb_tf9oDa=KMMQRZ04@mail.gmail.com>
References: <AANLkTimDT=699aCgz-h9yk_+FyXb_tf9oDa=KMMQRZ04@mail.gmail.com>
Message-ID: <AANLkTikzP4urgvo6CfvOBFkXGAHrSLn_VjXs0+bDvpsT@mail.gmail.com>

How are you performing the attended transfer? What kind of phone is this?
-MC

On Tue, Mar 22, 2011 at 3:11 AM, Dmitry Bely <dmitry.bely at gmail.com> wrote:

> I have G722 and PCMU enabled for internal extensions and PCMU only for
> an external gateway. Inbound-late-negotiation parameter is set so then
> an incoming call arrives PCMU is used without transcoding. But it goes
> worse when transfer is involved:
>
> call 1: gw <-- PCMU--> FreeSwitch <-- PCMU --> operator
> call 2: operator <-- G722 --> FreeSwitch <-- G722--> extension
> transfer (refer): gw <-- PCMU--> FreeSwitch <-- G722--> extension
>
> Is it possible to renegotiate a codec during transfer to get rid of
> transcoding?
>
> - Dmitry Bely
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Mar 22 20:20:29 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Mar 2011 10:20:29 -0700
Subject: [Freeswitch-users] playAndGetDigits not working?
In-Reply-To: <AANLkTimpyZHpXOf9ZHaKCcXdZH=mo-2y2rWJLWji=Dda@mail.gmail.com>
References: <AANLkTimpyZHpXOf9ZHaKCcXdZH=mo-2y2rWJLWji=Dda@mail.gmail.com>
Message-ID: <AANLkTim253GfYSJaGXAX86u2_YdGzNGwxK6RGb8Z+kUh@mail.gmail.com>

You may need to do \\d+ instead of \d+ in your expression.
-MC

On Mon, Mar 21, 2011 at 10:46 PM, deniro <ayhkor at gmail.com> wrote:

>
> I am unable to get the pin number and pass it to $digits ($digits comes
> blank)
> appreciate any feedback
> thx
> deniro--
>
>
>
> my($digits)=$session->playAndGetDigits(1,5,1,8000,"#","conference/conf-pin.wav","ivr/ivr-that_was_an_invalid_entry.wav","","^7\d{4}\$");
> or
> my($digits)=$session->playAndGetDigits(1,5,1,8000,"#","conference/conf-pin.wav","ivr/ivr-that_was_an_invalid_entry.wav","",
> "\d+");
> freeswitch::console_log("info", "*Got dtmf: .. $digits ..*\n");
>
> freeswitch console
> 2011-03-22 01:29:26.208664 [INFO] switch_cpp.cpp:1177* Got dtmf: ..  ..
> *2011-03-22 01:29:26.208664 [ERR] mod_perl.c:72 [require
> '/opt/freeswitch/scripts/fs1.pl';]
> /opt/freeswitch/scripts/fs1.pl did not return a true value at (eval 2)
> line 1.
>  each case getting "that was invalid PIN"
> entering pin starting with 7 maximum 5 digits in first case
> or any digits upto maximum 5 digits in second case
> tried also putting "#" at the end when dialing
>
> none worked
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Mar 22 20:26:04 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Mar 2011 10:26:04 -0700
Subject: [Freeswitch-users] Delayed start of tone_detect
In-Reply-To: <AANLkTikn1V3ypmbZW4NQ6qSgRy_7ENdOWT5eWrCHs-C0@mail.gmail.com>
References: <AANLkTims-MfPkq_tr-JQd_ih=gPd1i8gXSFSX2tF_m8y@mail.gmail.com>
	<AANLkTi=s1iwAm9qB3=y97U8CmLs+T=pL9ZzAToE2N5Ed@mail.gmail.com>
	<AANLkTimm7+5GxsRfJNRcRcRAqmtCUBwv8UWrn4Qk2_6r@mail.gmail.com>
	<AANLkTikn1V3ypmbZW4NQ6qSgRy_7ENdOWT5eWrCHs-C0@mail.gmail.com>
Message-ID: <AANLkTin8R6roSmw+3ejr8ud_VUMYEkbQ1Hz-xtmmYh54@mail.gmail.com>

On Tue, Mar 22, 2011 at 1:29 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> I have tested this trick with 1 call and it worked fine but when tested on
> a production environment doesn't work.
>
> The problem seems to be the "sleep" called by "execute_extension" executed
> on answer.
>
> If I activate this extension, I see in freeswitch log:
>
> 2011-03-22 09:18:10.191304 [CRIT] switch_core_session.c:1646 Throttle
> Error! 1326
> 2011-03-22 09:18:10.193260 [CRIT] switch_core_session.c:1646 Throttle
> Error! 1326
> 2011-03-22 09:18:10.194796 [CRIT] switch_core_session.c:1646 Throttle
> Error! 1323
> 2011-03-22 09:18:10.194796 [CRIT] switch_core_session.c:1646 Throttle
> Error! 1323
> 2011-03-22 09:18:10.199238 [CRIT] switch_core_session.c:1646 Throttle
> Error! 1317
> 2011-03-22 09:18:10.204815 [CRIT] switch_core_session.c:1646 Throttle
> Error! 1314
> 2011-03-22 09:18:10.204815 [CRIT] mod_sofia.c:3896 Error Creating Session
> 2011-03-22 09:18:10.214916 [CRIT] switch_core_session.c:1646 Throttle
> Error! 1302
> 2011-03-22 09:18:10.214916 [CRIT] mod_sofia.c:3896 Error Creating Session
> *2011-03-22 09:18:10.377962 [CRIT] switch_time.c:788 Over Session Rate of
> 200!*
>
Notice the above line!

You are overwhelming your system with too many calls in such a short period
of time. Slow things down a bit on your call generation. Also, you can
probably experiment with the sleep time. 500ms might be too long. Try
setting it to something shorter, like 100ms or so. That might help speed
things up a bit. Shaving 400ms off the call may not seem like a lot but if
you're doing hundreds of calls per second then that 400ms will add up
quickly.

-MC



> 2011-03-22 09:18:33.938094 [CRIT] switch_core_session.c:1646 Throttle
> Error! 1266
> 2011-03-22 09:18:33.938094 [CRIT] switch_core_session.c:1646 Throttle
> Error! 1266
> 2011-03-22 09:18:33.939077 [CRIT] switch_core_session.c:1646 Throttle
> Error! 1266
> 2011-03-22 09:18:33.939077 [CRIT] switch_core_session.c:1646 Throttle
> Error! 1266
> 2011-03-22 09:18:33.944077 [CRIT] switch_core_session.c:1646 Throttle
> Error! 1264
> 2011-03-22 09:18:33.944077 [CRIT] mod_sofia.c:3896 Error Creating Session
>
> Stephen
>
> On Tue, Mar 22, 2011 at 8:54 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> Thank you Michael, with this trick it works fine!
>>
>>
>> On Tue, Mar 22, 2011 at 6:33 AM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Add this to your dialplan...
>>>
>>> <extension name="start tone detect">
>>>    <condition field="destination_number"
>>> data="custom_start_tone_detect">
>>>     <action application="sleep" data="500"/>
>>>     <action application="tone_detect" data="MYTONE 680 ro +5000 set
>>> MYTONE=1"/>
>>>   </condition>
>>> </extension>
>>>
>>> ...then change your export...
>>> <action application="export"
>>> data="nolocal:execute_on_answer=execute_extension
>>> custom_start_tone_detect"/>
>>>
>>> Let us know if that does the trick.
>>>
>>> -MC
>>>
>>> On Mon, Mar 21, 2011 at 12:33 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>>
>>>> I start the tone_detect on answer using "execute_on_answer" as:
>>>>
>>>> <action application="export" data="nolocal:execute_on_answer=tone_detect
>>>> MYTONE 680 ro +5000 set MYTONE=1" />
>>>>
>>>> but I have the necessity to start the tone_detect for example 500ms
>>>> after answer: there is a way to do that in dialplan?
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From null at invalid.name  Tue Mar 22 20:42:00 2011
From: null at invalid.name (Dan Lane)
Date: Tue, 22 Mar 2011 17:42:00 +0000
Subject: [Freeswitch-users] Voicemail on GlusterFS
In-Reply-To: <AANLkTimkv66jjaBTHtT+nDUgo3PsuxEndEi-Z9=Lb0u_@mail.gmail.com>
References: <AANLkTimkv66jjaBTHtT+nDUgo3PsuxEndEi-Z9=Lb0u_@mail.gmail.com>
Message-ID: <AANLkTi=44afGJ1fni5P+q=0ct=irmVDwOfESUAbdWuo9@mail.gmail.com>

On Tue, Mar 22, 2011 at 3:55 PM, Moe Navid <manavid at gmail.com> wrote:
> Hi All,
> Has anyone tried using voicemail on GlusterFS?
> GlusteFS looks very promising for building distributed file systems. Pandora
> Radio recently started using
> glusterfs?http://www.gluster.com/2011/01/05/gluster-to-help-manage-rapid-data-growth-for-pandora/
> I managed to get the recording right for voicemails but?retrieving?the WAV
> files fails. FreeSWITCH gives the following error:
> 2011-03-22 08:54:06.325001 [ERR] mod_sndfile.c:194 Error Opening File
> [/home/moe/freeswitch/storage/voicemail/default/192.168.1.10/1001/msg_02a108cc-549c-11e0-a9f6-9f1a6f4da0f7.wav]
> [File contains data in an unknown format.]
> Any thoughts?
> Thanks
> Moe

We've been using GlusterFS for storing voicemail in a production
environment for over a year now but we built a custom voicemail
solution rather than using the built-in one.

We didn't experience the issue you're referring to so I'm afraid I
can't offer any assistance.


From kris at kriskinc.com  Tue Mar 22 20:47:08 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 22 Mar 2011 13:47:08 -0400
Subject: [Freeswitch-users] Voicemail on GlusterFS
In-Reply-To: <AANLkTi=44afGJ1fni5P+q=0ct=irmVDwOfESUAbdWuo9@mail.gmail.com>
References: <AANLkTimkv66jjaBTHtT+nDUgo3PsuxEndEi-Z9=Lb0u_@mail.gmail.com>
	<AANLkTi=44afGJ1fni5P+q=0ct=irmVDwOfESUAbdWuo9@mail.gmail.com>
Message-ID: <AANLkTi=bWO48xovSm_am037RWAqnCjGSFOTYTdwiVe4J@mail.gmail.com>

Can I ask how your voicemail solution works?

Have you looked at the Jester framework at all?

http://wiki.freeswitch.org/wiki/Jester

On Tue, Mar 22, 2011 at 1:42 PM, Dan Lane <null at invalid.name> wrote:
>
> We've been using GlusterFS for storing voicemail in a production
> environment for over a year now but we built a custom voicemail
> solution rather than using the built-in one.
>
> We didn't experience the issue you're referring to so I'm afraid I
> can't offer any assistance.

-- 
Kristian Kielhofner


From wstephen80 at gmail.com  Tue Mar 22 20:59:37 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 22 Mar 2011 18:59:37 +0100
Subject: [Freeswitch-users] Delayed start of tone_detect
In-Reply-To: <AANLkTin8R6roSmw+3ejr8ud_VUMYEkbQ1Hz-xtmmYh54@mail.gmail.com>
References: <AANLkTims-MfPkq_tr-JQd_ih=gPd1i8gXSFSX2tF_m8y@mail.gmail.com>
	<AANLkTi=s1iwAm9qB3=y97U8CmLs+T=pL9ZzAToE2N5Ed@mail.gmail.com>
	<AANLkTimm7+5GxsRfJNRcRcRAqmtCUBwv8UWrn4Qk2_6r@mail.gmail.com>
	<AANLkTikn1V3ypmbZW4NQ6qSgRy_7ENdOWT5eWrCHs-C0@mail.gmail.com>
	<AANLkTin8R6roSmw+3ejr8ud_VUMYEkbQ1Hz-xtmmYh54@mail.gmail.com>
Message-ID: <AANLkTiny4+NvH9WFWC51VHDCYM6pB7_QrWn8_CyontFf@mail.gmail.com>

I cannot slow down the call rate, this is a real traffic and is not
generate but the session rate it's not a real problem because my session
rate is never more then 100cps with a medium value of 50cps so, when I
activate this sleep, Freeswitch calculate a wrong CPS value. It seems that
during sleep, FS freezes, as the sleep is performed synchronously.

Another effect is:

2011-03-22 09:17:57.551069 [ERR] mod_freetdm.c:2177 Session for channel
29:15 not found [UUID: N/A]
2011-03-22 09:18:00.181462 [ERR] mod_freetdm.c:2177 Session for channel
29:18 not found [UUID: N/A]
2011-03-22 09:18:00.301001 [ERR] mod_freetdm.c:2177 Session for channel
25:25 not found [UUID: N/A]
2011-03-22 09:18:03.931076 [ERR] mod_freetdm.c:2177 Session for channel 10:3
not found [UUID: N/A]
2011-03-22 09:18:05.331547 [ERR] mod_freetdm.c:2177 Session for channel
10:10 not found [UUID: N/A]
2011-03-22 09:18:06.768937 [ERR] mod_freetdm.c:2177 Session for channel
24:14 not found [UUID: N/A]
2011-03-22 09:18:07.248782 [ERR] mod_freetdm.c:2177 Session for channel
24:26 not found [UUID: N/A]
2011-03-22 09:18:07.797798 [ERR] mod_freetdm.c:2177 Session for channel
24:15 not found [UUID: N/A]

and many calls are dropped (I never seen these errors before).
Removing the sleep from dialplan, all revert to normal.

Stephen

On Tue, Mar 22, 2011 at 6:26 PM, Michael Collins <msc at freeswitch.org> wrote:

>
> On Tue, Mar 22, 2011 at 1:29 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> I have tested this trick with 1 call and it worked fine but when tested on
>> a production environment doesn't work.
>>
>> The problem seems to be the "sleep" called by "execute_extension" executed
>> on answer.
>>
>> If I activate this extension, I see in freeswitch log:
>>
>> 2011-03-22 09:18:10.191304 [CRIT] switch_core_session.c:1646 Throttle
>> Error! 1326
>> 2011-03-22 09:18:10.193260 [CRIT] switch_core_session.c:1646 Throttle
>> Error! 1326
>> 2011-03-22 09:18:10.194796 [CRIT] switch_core_session.c:1646 Throttle
>> Error! 1323
>> 2011-03-22 09:18:10.194796 [CRIT] switch_core_session.c:1646 Throttle
>> Error! 1323
>> 2011-03-22 09:18:10.199238 [CRIT] switch_core_session.c:1646 Throttle
>> Error! 1317
>> 2011-03-22 09:18:10.204815 [CRIT] switch_core_session.c:1646 Throttle
>> Error! 1314
>> 2011-03-22 09:18:10.204815 [CRIT] mod_sofia.c:3896 Error Creating Session
>> 2011-03-22 09:18:10.214916 [CRIT] switch_core_session.c:1646 Throttle
>> Error! 1302
>> 2011-03-22 09:18:10.214916 [CRIT] mod_sofia.c:3896 Error Creating Session
>> *2011-03-22 09:18:10.377962 [CRIT] switch_time.c:788 Over Session Rate of
>> 200!*
>>
> Notice the above line!
>
> You are overwhelming your system with too many calls in such a short period
> of time. Slow things down a bit on your call generation. Also, you can
> probably experiment with the sleep time. 500ms might be too long. Try
> setting it to something shorter, like 100ms or so. That might help speed
> things up a bit. Shaving 400ms off the call may not seem like a lot but if
> you're doing hundreds of calls per second then that 400ms will add up
> quickly.
>
> -MC
>
>
>
>> 2011-03-22 09:18:33.938094 [CRIT] switch_core_session.c:1646 Throttle
>> Error! 1266
>> 2011-03-22 09:18:33.938094 [CRIT] switch_core_session.c:1646 Throttle
>> Error! 1266
>> 2011-03-22 09:18:33.939077 [CRIT] switch_core_session.c:1646 Throttle
>> Error! 1266
>> 2011-03-22 09:18:33.939077 [CRIT] switch_core_session.c:1646 Throttle
>> Error! 1266
>> 2011-03-22 09:18:33.944077 [CRIT] switch_core_session.c:1646 Throttle
>> Error! 1264
>> 2011-03-22 09:18:33.944077 [CRIT] mod_sofia.c:3896 Error Creating Session
>>
>> Stephen
>>
>> On Tue, Mar 22, 2011 at 8:54 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>
>>> Thank you Michael, with this trick it works fine!
>>>
>>>
>>> On Tue, Mar 22, 2011 at 6:33 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> Add this to your dialplan...
>>>>
>>>> <extension name="start tone detect">
>>>>    <condition field="destination_number"
>>>> data="custom_start_tone_detect">
>>>>     <action application="sleep" data="500"/>
>>>>     <action application="tone_detect" data="MYTONE 680 ro +5000 set
>>>> MYTONE=1"/>
>>>>   </condition>
>>>> </extension>
>>>>
>>>> ...then change your export...
>>>> <action application="export"
>>>> data="nolocal:execute_on_answer=execute_extension
>>>> custom_start_tone_detect"/>
>>>>
>>>> Let us know if that does the trick.
>>>>
>>>> -MC
>>>>
>>>> On Mon, Mar 21, 2011 at 12:33 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>>>
>>>>> I start the tone_detect on answer using "execute_on_answer" as:
>>>>>
>>>>> <action application="export"
>>>>> data="nolocal:execute_on_answer=tone_detect MYTONE 680 ro +5000 set
>>>>> MYTONE=1" />
>>>>>
>>>>> but I have the necessity to start the tone_detect for example 500ms
>>>>> after answer: there is a way to do that in dialplan?
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dmitry.bely at gmail.com  Tue Mar 22 21:09:11 2011
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Tue, 22 Mar 2011 21:09:11 +0300
Subject: [Freeswitch-users] inbound-late-negotiation and attended
	transfer
In-Reply-To: <AANLkTikzP4urgvo6CfvOBFkXGAHrSLn_VjXs0+bDvpsT@mail.gmail.com>
References: <AANLkTimDT=699aCgz-h9yk_+FyXb_tf9oDa=KMMQRZ04@mail.gmail.com>
	<AANLkTikzP4urgvo6CfvOBFkXGAHrSLn_VjXs0+bDvpsT@mail.gmail.com>
Message-ID: <AANLkTimVm30-n_p7SiJxsJ9pso692ogNqiaSWp900nUg@mail.gmail.com>

On Tue, Mar 22, 2011 at 8:18 PM, Michael Collins <msc at freeswitch.org> wrote:
> How are you performing the attended transfer? What kind of phone is this?
> -MC

Yes, the attended transfer using phone's capabilities as explained
here: http://www.youtube.com/watch?v=VpEVpr-4y-U
The phone is Grandstream GXP-2000.

Here is an example. There was two active calls:

gw <-> 1000 (PCMU)
1000 <-> 1004 (G722)

Now operator transfers the fist call to 1004. The log:

2011-03-18 17:08:27.221372 [DEBUG] sofia.c:4659 Channel
sofia/internal/1000 at 192.168.121.66 entering
state [received][100]
2011-03-18 17:08:27.222378 [DEBUG] sofia.c:4670 Remote SDP:
v=0
o=1000 8001 8002 IN IP4 192.168.121.153
s=SIP Call
c=IN IP4 192.168.121.153
t=0 0
m=audio 5030 RTP/AVP 9 0 8 18 4 99 3 2
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=20
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=sendonly
a=ptime:20

2011-03-18 17:08:27.222378 [DEBUG] switch_channel.c:1377
(sofia/internal/1000 at 192.168.121.66) Callstate Change ACTIVE -> HELD
2011-03-18 17:08:27.222378 [DEBUG] switch_core_session.c:954 Send
signal sofia/internal/sip:1004 at 192.168.121.136:5060 [BREAK]
2011-03-18 17:08:27.224386 [DEBUG] switch_core_session.c:709 Send
signal sofia/internal/sip:1004 at 192.168.121.136:5060 [BREAK]
2011-03-18 17:08:27.364243 [DEBUG] switch_ivr.c:563
sofia/internal/sip:1004 at 192.168.121.136:5060 Command Execute
playback(local_stream://moh)
EXECUTE sofia/internal/sip:1004 at 192.168.121.136:5060
playback(local_stream://moh)
2011-03-18 17:08:27.364243 [WARNING] mod_local_stream.c:393 Unknown
source moh, trying 'default'
2011-03-18 17:08:27.364243 [DEBUG] mod_local_stream.c:421 Opening
Stream [default] 8000hz
2011-03-18 17:08:27.364243 [DEBUG] switch_core_file.c:176 File moh
sample rate 8000 doesn't match requested rate 16000
2011-03-18 17:08:27.364243 [DEBUG] switch_ivr_play_say.c:1244 Codec
Activated L16 at 16000hz 1 channels 20ms
2011-03-18 17:08:27.472662 [DEBUG] sofia_glue.c:4474 Audio Codec
Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000]
2011-03-18 17:08:27.472662 [DEBUG] sofia_glue.c:2690 Already using G722
2011-03-18 17:08:27.472662 [DEBUG] sofia_glue.c:2972 Audio params are
unchanged for sofia/internal/1000 at 192.168.121.66.
2011-03-18 17:08:27.472662 [DEBUG] sofia.c:5070 Processing updated SDP
2011-03-18 17:08:27.473667 [DEBUG] sofia.c:4659 Channel
sofia/internal/1000 at 192.168.121.66 entering state [completed][200]
2011-03-18 17:08:27.592069 [DEBUG] sofia.c:4659 Channel
sofia/internal/1000 at 192.168.121.66 entering state [ready][200]
2011-03-18 17:08:27.769589 [DEBUG] sofia.c:5520 Process REFER to
[1004 at 192.168.121.66]
2011-03-18 17:08:27.769589 [DEBUG] sofia.c:5539 Replaces:
[5cd372e1e81966ca at 192.168.121.153;from-tag=d55f7fbec6efbde7;to-tag=2N49SS5XNjUeB]
...
(no more codec negotiation)

> On Tue, Mar 22, 2011 at 3:11 AM, Dmitry Bely <dmitry.bely at gmail.com> wrote:
>>
>> I have G722 and PCMU enabled for internal extensions and PCMU only for
>> an external gateway. Inbound-late-negotiation parameter is set so then
>> an incoming call arrives PCMU is used without transcoding. But it goes
>> worse when transfer is involved:
>>
>> call 1: gw <-- PCMU--> FreeSwitch <-- PCMU --> operator
>> call 2: operator <-- G722 --> FreeSwitch <-- G722--> extension
>> transfer (refer): gw <-- PCMU--> FreeSwitch <-- G722--> extension
>>
>> Is it possible to renegotiate a codec during transfer to get rid of
>> transcoding?

- Dmitry Bely


From manavid at gmail.com  Tue Mar 22 21:21:27 2011
From: manavid at gmail.com (Moe Navid)
Date: Tue, 22 Mar 2011 11:21:27 -0700
Subject: [Freeswitch-users] Voicemail on GlusterFS
In-Reply-To: <AANLkTi=bWO48xovSm_am037RWAqnCjGSFOTYTdwiVe4J@mail.gmail.com>
References: <AANLkTimkv66jjaBTHtT+nDUgo3PsuxEndEi-Z9=Lb0u_@mail.gmail.com>
	<AANLkTi=44afGJ1fni5P+q=0ct=irmVDwOfESUAbdWuo9@mail.gmail.com>
	<AANLkTi=bWO48xovSm_am037RWAqnCjGSFOTYTdwiVe4J@mail.gmail.com>
Message-ID: <AANLkTi=hpYhE2qdfuzuxNn=b5atbb7o-DjLSfNcy8Q6e@mail.gmail.com>

Hi Dan,

Thanks for the reply. I'm also working on a custom voicemail.

Is your vm implemented in C/C++ and uses GlusterFS' native library or you
are just mounting with fuse?

On Tue, Mar 22, 2011 at 10:47 AM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> Can I ask how your voicemail solution works?
>
> Have you looked at the Jester framework at all?
>
> http://wiki.freeswitch.org/wiki/Jester
>
> On Tue, Mar 22, 2011 at 1:42 PM, Dan Lane <null at invalid.name> wrote:
> >
> > We've been using GlusterFS for storing voicemail in a production
> > environment for over a year now but we built a custom voicemail
> > solution rather than using the built-in one.
> >
> > We didn't experience the issue you're referring to so I'm afraid I
> > can't offer any assistance.
>
> --
> Kristian Kielhofner
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Mar 22 21:46:20 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Mar 2011 11:46:20 -0700
Subject: [Freeswitch-users] Delayed start of tone_detect
In-Reply-To: <AANLkTiny4+NvH9WFWC51VHDCYM6pB7_QrWn8_CyontFf@mail.gmail.com>
References: <AANLkTims-MfPkq_tr-JQd_ih=gPd1i8gXSFSX2tF_m8y@mail.gmail.com>
	<AANLkTi=s1iwAm9qB3=y97U8CmLs+T=pL9ZzAToE2N5Ed@mail.gmail.com>
	<AANLkTimm7+5GxsRfJNRcRcRAqmtCUBwv8UWrn4Qk2_6r@mail.gmail.com>
	<AANLkTikn1V3ypmbZW4NQ6qSgRy_7ENdOWT5eWrCHs-C0@mail.gmail.com>
	<AANLkTin8R6roSmw+3ejr8ud_VUMYEkbQ1Hz-xtmmYh54@mail.gmail.com>
	<AANLkTiny4+NvH9WFWC51VHDCYM6pB7_QrWn8_CyontFf@mail.gmail.com>
Message-ID: <AANLkTimE=Ovn+HR8dnAZ4NmPydSiepS-Rb7rzkTshP7=@mail.gmail.com>

set the sleep value to something really low, like 20ms or something. Find
the lowest value that still lets the tone_detect work.
-MC

On Tue, Mar 22, 2011 at 10:59 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:

> I cannot slow down the call rate, this is a real traffic and is not
> generate but the session rate it's not a real problem because my session
> rate is never more then 100cps with a medium value of 50cps so, when I
> activate this sleep, Freeswitch calculate a wrong CPS value. It seems that
> during sleep, FS freezes, as the sleep is performed synchronously.
>
> Another effect is:
>
> 2011-03-22 09:17:57.551069 [ERR] mod_freetdm.c:2177 Session for channel
> 29:15 not found [UUID: N/A]
> 2011-03-22 09:18:00.181462 [ERR] mod_freetdm.c:2177 Session for channel
> 29:18 not found [UUID: N/A]
> 2011-03-22 09:18:00.301001 [ERR] mod_freetdm.c:2177 Session for channel
> 25:25 not found [UUID: N/A]
> 2011-03-22 09:18:03.931076 [ERR] mod_freetdm.c:2177 Session for channel
> 10:3 not found [UUID: N/A]
> 2011-03-22 09:18:05.331547 [ERR] mod_freetdm.c:2177 Session for channel
> 10:10 not found [UUID: N/A]
> 2011-03-22 09:18:06.768937 [ERR] mod_freetdm.c:2177 Session for channel
> 24:14 not found [UUID: N/A]
> 2011-03-22 09:18:07.248782 [ERR] mod_freetdm.c:2177 Session for channel
> 24:26 not found [UUID: N/A]
> 2011-03-22 09:18:07.797798 [ERR] mod_freetdm.c:2177 Session for channel
> 24:15 not found [UUID: N/A]
>
> and many calls are dropped (I never seen these errors before).
> Removing the sleep from dialplan, all revert to normal.
>
> Stephen
>
> On Tue, Mar 22, 2011 at 6:26 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>> On Tue, Mar 22, 2011 at 1:29 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>
>>> I have tested this trick with 1 call and it worked fine but when tested
>>> on a production environment doesn't work.
>>>
>>> The problem seems to be the "sleep" called by "execute_extension"
>>> executed on answer.
>>>
>>> If I activate this extension, I see in freeswitch log:
>>>
>>> 2011-03-22 09:18:10.191304 [CRIT] switch_core_session.c:1646 Throttle
>>> Error! 1326
>>> 2011-03-22 09:18:10.193260 [CRIT] switch_core_session.c:1646 Throttle
>>> Error! 1326
>>> 2011-03-22 09:18:10.194796 [CRIT] switch_core_session.c:1646 Throttle
>>> Error! 1323
>>> 2011-03-22 09:18:10.194796 [CRIT] switch_core_session.c:1646 Throttle
>>> Error! 1323
>>> 2011-03-22 09:18:10.199238 [CRIT] switch_core_session.c:1646 Throttle
>>> Error! 1317
>>> 2011-03-22 09:18:10.204815 [CRIT] switch_core_session.c:1646 Throttle
>>> Error! 1314
>>> 2011-03-22 09:18:10.204815 [CRIT] mod_sofia.c:3896 Error Creating Session
>>> 2011-03-22 09:18:10.214916 [CRIT] switch_core_session.c:1646 Throttle
>>> Error! 1302
>>> 2011-03-22 09:18:10.214916 [CRIT] mod_sofia.c:3896 Error Creating Session
>>> *2011-03-22 09:18:10.377962 [CRIT] switch_time.c:788 Over Session Rate
>>> of 200!*
>>>
>> Notice the above line!
>>
>> You are overwhelming your system with too many calls in such a short
>> period of time. Slow things down a bit on your call generation. Also, you
>> can probably experiment with the sleep time. 500ms might be too long. Try
>> setting it to something shorter, like 100ms or so. That might help speed
>> things up a bit. Shaving 400ms off the call may not seem like a lot but if
>> you're doing hundreds of calls per second then that 400ms will add up
>> quickly.
>>
>> -MC
>>
>>
>>
>>> 2011-03-22 09:18:33.938094 [CRIT] switch_core_session.c:1646 Throttle
>>> Error! 1266
>>> 2011-03-22 09:18:33.938094 [CRIT] switch_core_session.c:1646 Throttle
>>> Error! 1266
>>> 2011-03-22 09:18:33.939077 [CRIT] switch_core_session.c:1646 Throttle
>>> Error! 1266
>>> 2011-03-22 09:18:33.939077 [CRIT] switch_core_session.c:1646 Throttle
>>> Error! 1266
>>> 2011-03-22 09:18:33.944077 [CRIT] switch_core_session.c:1646 Throttle
>>> Error! 1264
>>> 2011-03-22 09:18:33.944077 [CRIT] mod_sofia.c:3896 Error Creating Session
>>>
>>> Stephen
>>>
>>> On Tue, Mar 22, 2011 at 8:54 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>>
>>>> Thank you Michael, with this trick it works fine!
>>>>
>>>>
>>>> On Tue, Mar 22, 2011 at 6:33 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>> Add this to your dialplan...
>>>>>
>>>>> <extension name="start tone detect">
>>>>>    <condition field="destination_number"
>>>>> data="custom_start_tone_detect">
>>>>>     <action application="sleep" data="500"/>
>>>>>     <action application="tone_detect" data="MYTONE 680 ro +5000 set
>>>>> MYTONE=1"/>
>>>>>   </condition>
>>>>> </extension>
>>>>>
>>>>> ...then change your export...
>>>>> <action application="export"
>>>>> data="nolocal:execute_on_answer=execute_extension
>>>>> custom_start_tone_detect"/>
>>>>>
>>>>> Let us know if that does the trick.
>>>>>
>>>>> -MC
>>>>>
>>>>> On Mon, Mar 21, 2011 at 12:33 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>>>>
>>>>>> I start the tone_detect on answer using "execute_on_answer" as:
>>>>>>
>>>>>> <action application="export"
>>>>>> data="nolocal:execute_on_answer=tone_detect MYTONE 680 ro +5000 set
>>>>>> MYTONE=1" />
>>>>>>
>>>>>> but I have the necessity to start the tone_detect for example 500ms
>>>>>> after answer: there is a way to do that in dialplan?
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ibc at aliax.net  Tue Mar 22 14:45:41 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Tue, 22 Mar 2011 12:45:41 +0100
Subject: [Freeswitch-users] Event-List: double Content-Lenght?
Message-ID: <AANLkTimDH-bvRh+R+w0TUaj+5WHTR7MZHymtX==vQ9j9@mail.gmail.com>

Hi, I'm trying to understand the following example in the doc:

  http://wiki.freeswitch.org/wiki/Event_List

Upon job completion, server response
----------------------
Content-Length: 625
Content-Type: text/event-plain

Job-UUID: 7f4db78a-17d7-11dd-b7a0-db4edd065621
Job-Command: originate
Job-Command-Arg: sofia/default/1005%20'%26park'
Event-Name: BACKGROUND_JOB
Core-UUID: 42bdf272-16e6-11dd-b7a0-db4edd065621
FreeSWITCH-Hostname: ser
FreeSWITCH-IPv4: 192.168.1.104
FreeSWITCH-IPv6: 127.0.0.1
Event-Date-Local: 2008-05-02%2007%3A37%3A03
Event-Date-GMT: Thu,%2001%20May%202008%2023%3A37%3A03%20GMT
Event-Date-timestamp: 1209685023894968
Event-Calling-File: mod_event_socket.c
Event-Calling-Function: api_exec
Event-Calling-Line-Number: 609
Content-Length: 41

+OK 7f4de4bc-17d7-11dd-b7a0-db4edd065621
----------------------


What is the purpose of the first "Content-Length: 625"? Why not just
send all the headers with no Content-Lenght? (just leave the second
Content-Length:41 which is the length of the job uuid).

So what is the purpose of the first Content-Length? A parser doesn't
need it at all as it can read headers until it finds "Content-Length"
(so it reads N bytes more) or it finds a double \n\n (so event info is
terminated).

Thanks for any clarification.


-- 
I?aki Baz Castillo
<ibc at aliax.net>


From m_g_saeed at hotmail.co.uk  Tue Mar 22 15:09:56 2011
From: m_g_saeed at hotmail.co.uk (Ghazanfar Saeed)
Date: Tue, 22 Mar 2011 12:09:56 +0000
Subject: [Freeswitch-users] use of mobile phone with gsmopen question
In-Reply-To: <SNT111-W427F28F80543FB54AB6AF4EEB40@phx.gbl>
References: <SNT111-W427F28F80543FB54AB6AF4EEB40@phx.gbl>
Message-ID: <SNT111-W562AD83EE8A8BBDE68B8B5EEB40@phx.gbl>










Hi,

My system is Dell XPS L501 and I have installed Ubuntu 10.10 Desktop version on it. The bit I am trying to achieve is to make call to gsm mobile from ubuntu using my gsm mobile (HTC HD2 with T-Mobile-UK SIM card) attached to my Ubuntu 10.10 system via USB. I have followed the below process: 

I have installed FreeSwitch and verified its functionality with Twinkle (able to access the predefined basic IVR). 
command: sudo /usr/local/freeswitch/bin/freeswitch -waste

I have then built/installed the gsmopen as described on http://wiki.freeswitch.org/wiki/GSMopen and loaded module. 

freeswitch at saeed-XPS-L501X> load mod_gsmopen
2011-03-22 07:41:47.623806 [WARNING] mod_gsmopen.cpp:1861 rev exported[(nil)|37     ][WARNINGA  1861 ][interface1][-1, 0, 0] STARTING interface_id=1
open error : No such file or directory
2011-03-22 07:41:48.622951 [ERR] mod_gsmopen.cpp:1892 rev exported[(nil)|37     ][ERRORA  1892 ][interface1][-1, 0, 0] gsmopen_serial_init failed
2011-03-22 07:41:48.622951 [ERR] mod_gsmopen.cpp:1893 rev exported[(nil)|37     ][ERRORA  1893 ][interface1][-1, 0, 0] STARTING interface_id=1 FAILED
2011-03-22 07:41:48.622951 [ERR] mod_gsmopen.cpp:3136 rev exported[(nil)|37     ][ERRORA  3136 ][interface1][-1, 0, 0] ALARM on interface interface1: 
2011-03-22 07:41:48.622951 [CONSOLE] switch_loadable_module.c:900 Successfully Loaded [mod_gsmopen]
2011-03-22 07:41:48.622951 [NOTICE] switch_loadable_module.c:144 Adding Endpoint 'gsmopen'
2011-03-22 07:41:48.622951 [NOTICE] switch_loadable_module.c:272 Adding API Function 'gsm'
2011-03-22 07:41:48.622951 [NOTICE] switch_loadable_module.c:272 Adding API Function 'gsmopen'
2011-03-22 07:41:48.624943 [NOTICE] switch_loadable_module.c:272 Adding API Function 'gsmopen_boost_audio'
2011-03-22 07:41:48.624943 [NOTICE] switch_loadable_module.c:272 Adding API Function 'gsmopen_dump'
2011-03-22 07:41:48.624943 [NOTICE] switch_loadable_module.c:272 Adding API Function 'gsmopen_sendsms'
2011-03-22 07:41:48.624943 [NOTICE] switch_loadable_module.c:377 Adding Chat interface 'SMS'

The interface1 is the default interface (from gspopen.conf.xml)

    <interface id="1" name="interface1">
    <param name="hold-music" value="$${moh_uri}"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="default"/>
        <param name="destination" value="5000"/>
        <param name="alsacname" value="plughw:1"/>
        <param name="alsapname" value="plughw:1"/>
        <param name="controldevice_name" value="/dev/ttyACM0"/>
    </interface>


I am new user to this system (freeswitch+gsmopen) so please can someone put me in the right direction. These errors are probably due to the fact that there is no /dev/ttyACM0 (as I couldn't list this device). Now as a workaround I was thinking if I can find where my HTC HD2 phone got connected e.g. ttyUSB0 etc then I can create a soft link and can proceed but the trouble is that I can't find the which device it is connected to though i can see the phone via lsusb (it is on Bus 003 and device 005)

Bus 003 Device 005: ID 0bb4:0b40 High Tech Computer Corp. 
Bus 003 Device 004: ID 04f3:0212 Elan Microelectronics Corp. Laser Mouse
Bus 003 Device 003: ID 0603:00f2 Novatek Microelectronics Corp. 

Further information is I was able to use internet out of the box when phone is connected (and I opted for the option internet sharing on my phone).

Any recommendation how should I proceed further on this will be very helpful. Please let me know if I need to provide any further information. 

Many thanks.

Kind regards,


 






 		 	   		  
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From ejay.greeves at yahoo.com  Tue Mar 22 16:48:57 2011
From: ejay.greeves at yahoo.com (Ejay Greeves)
Date: Tue, 22 Mar 2011 13:48:57 +0000 (GMT)
Subject: [Freeswitch-users] send custom tcp message from esl
In-Reply-To: <AANLkTim0f1WN-utJWtTWmozFJZtjj9fvEit2ouaLYbD=@mail.gmail.com>
Message-ID: <133921.61849.qm@web132309.mail.ird.yahoo.com>

This is for a collected call being processed in an event socket ruby script. ?I would want to connect out from Freeswitch to an external tcp port. Basic what I want is to sent the output string of a ruby puts command through tcp to a ecternal server. Can I do this with the esl object?

--- On Tue, 22/3/11, Michael Collins <msc at freeswitch.org> wrote:

From: Michael Collins <msc at freeswitch.org>
Subject: Re: [Freeswitch-users] send custom tcp message from esl
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Tuesday, 22 March, 2011, 5:26

Are you trying to send a simple string to the event socket port on FreeSWITCH? If so, what message are you sending? I'm not aware of any event socket messages that ESL cannot handle.
-MC


On Mon, Mar 21, 2011 at 4:10 PM, Ejay Greeves <ejay.greeves at yahoo.com> wrote:

I am using esl in Ruby I want to send tcp messages to a server.? When I tried to make my tcpsocket class it did not work it must be messing with the socket setup of esl. Is there a way without the need to manage the sockets seperatley to use els methods to send out custom tcp message. All I need is for a simple string to be fired to tcp port. How can I do this from inside esl.






      
_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org





-----Inline Attachment Follows-----

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From chistyakov at directtel.ru  Tue Mar 22 16:59:20 2011
From: chistyakov at directtel.ru (=?UTF-8?B?0KfQuNGB0YLRj9C60L7QsiDQmNCy0LDQvQ==?=)
Date: Tue, 22 Mar 2011 16:59:20 +0300
Subject: [Freeswitch-users] How can I send an event via SIP from one
	FreeSWITCH to an other FreeSWITCH?
Message-ID: <4D88AB38.3040001@directtel.ru>

How can I send an event via SIP from one FreeSWITCH to an other FreeSWITCH?


From marketing at cluecon.com  Tue Mar 22 21:43:54 2011
From: marketing at cluecon.com (Michael Collins)
Date: Tue, 22 Mar 2011 11:43:54 -0700
Subject: [Freeswitch-users] ClueCon 2011 Registration Open, Call For Speakers
Message-ID: <AANLkTikfaAd+D_4AORmyVCtXowFc8b6nNOFW8hTN9EBe@mail.gmail.com>

Greetings fellow open source VoIP users and developers!

We are happy to announce that ClueCon 2011 registration is now open. The
event will be held August 9-11, 2011 at the Sofitel Hotel in downtown
Chicago. Additionally, we are now accepting proposals for speaking topics.
If you or your organization would like to speak at ClueCon this year please
contact us with your ideas. Visit us at http://www.cluecon.com for
additional details on registration, speaking, or sponsorship opportunities.

Thank you to all the members of the open source telephony projects who have
supported ClueCon over the years. We hope to see you all again this coming
August!

Michael S Collins
ClueCon Team Member
http://www.cluecon.com
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From msc at freeswitch.org  Tue Mar 22 22:03:21 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Mar 2011 12:03:21 -0700
Subject: [Freeswitch-users] send custom tcp message from esl
In-Reply-To: <133921.61849.qm@web132309.mail.ird.yahoo.com>
References: <AANLkTim0f1WN-utJWtTWmozFJZtjj9fvEit2ouaLYbD=@mail.gmail.com>
	<133921.61849.qm@web132309.mail.ird.yahoo.com>
Message-ID: <AANLkTi==VTD=arWZptZaaRJ=oRc4hTx9oSJRspfRp8kB@mail.gmail.com>

On Tue, Mar 22, 2011 at 6:48 AM, Ejay Greeves <ejay.greeves at yahoo.com>wrote:

> This is for a collected call being processed in an event socket ruby
> script.  I would want to connect out from Freeswitch to an external tcp
> port. Basic what I want is to sent the output string of a ruby puts command
> through tcp to a ecternal server. Can I do this with the esl object?
>

Not that I'm aware of. I'm pretty sure you'd need to handle the socket
connect in whatever Rubyish way you have available.

-MC
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From msc at freeswitch.org  Tue Mar 22 22:07:06 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Mar 2011 12:07:06 -0700
Subject: [Freeswitch-users] How can I send an event via SIP from one
 FreeSWITCH to an other FreeSWITCH?
In-Reply-To: <4D88AB38.3040001@directtel.ru>
References: <4D88AB38.3040001@directtel.ru>
Message-ID: <AANLkTinOUoDRa8bqxJj5tQ7N6Tiod0cSRBtziFUnmH3Q@mail.gmail.com>

On Tue, Mar 22, 2011 at 6:59 AM, ???????? ???? <chistyakov at directtel.ru>wrote:

> How can I send an event via SIP from one FreeSWITCH to an other FreeSWITCH?
>
>
I suppose it depends on what you mean by "event." If you mean event socket
stuff then you probably need to read up on this:
http://wiki.freeswitch.org/wiki/Mod_event_multicast

-MC
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From wstephen80 at gmail.com  Tue Mar 22 22:53:30 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 22 Mar 2011 20:53:30 +0100
Subject: [Freeswitch-users] Delayed start of tone_detect
In-Reply-To: <AANLkTimE=Ovn+HR8dnAZ4NmPydSiepS-Rb7rzkTshP7=@mail.gmail.com>
References: <AANLkTims-MfPkq_tr-JQd_ih=gPd1i8gXSFSX2tF_m8y@mail.gmail.com>
	<AANLkTi=s1iwAm9qB3=y97U8CmLs+T=pL9ZzAToE2N5Ed@mail.gmail.com>
	<AANLkTimm7+5GxsRfJNRcRcRAqmtCUBwv8UWrn4Qk2_6r@mail.gmail.com>
	<AANLkTikn1V3ypmbZW4NQ6qSgRy_7ENdOWT5eWrCHs-C0@mail.gmail.com>
	<AANLkTin8R6roSmw+3ejr8ud_VUMYEkbQ1Hz-xtmmYh54@mail.gmail.com>
	<AANLkTiny4+NvH9WFWC51VHDCYM6pB7_QrWn8_CyontFf@mail.gmail.com>
	<AANLkTimE=Ovn+HR8dnAZ4NmPydSiepS-Rb7rzkTshP7=@mail.gmail.com>
Message-ID: <AANLkTinh4xNnpHK-f_eE9QzCe4Mo8QwuWfec42wDw-jN@mail.gmail.com>

Ok, but this is not the solution for my initial problem: to delay the
tone_detect of 500mS after answer.
But I don't see any other way to do that (except modifying the
"switch_ivr_tone_detect_session" to ignore the first n samples).

Stephen

On Tue, Mar 22, 2011 at 7:46 PM, Michael Collins <msc at freeswitch.org> wrote:

> set the sleep value to something really low, like 20ms or something. Find
> the lowest value that still lets the tone_detect work.
> -MC
>
>
> On Tue, Mar 22, 2011 at 10:59 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> I cannot slow down the call rate, this is a real traffic and is not
>> generate but the session rate it's not a real problem because my session
>> rate is never more then 100cps with a medium value of 50cps so, when I
>> activate this sleep, Freeswitch calculate a wrong CPS value. It seems that
>> during sleep, FS freezes, as the sleep is performed synchronously.
>>
>> Another effect is:
>>
>> 2011-03-22 09:17:57.551069 [ERR] mod_freetdm.c:2177 Session for channel
>> 29:15 not found [UUID: N/A]
>> 2011-03-22 09:18:00.181462 [ERR] mod_freetdm.c:2177 Session for channel
>> 29:18 not found [UUID: N/A]
>> 2011-03-22 09:18:00.301001 [ERR] mod_freetdm.c:2177 Session for channel
>> 25:25 not found [UUID: N/A]
>> 2011-03-22 09:18:03.931076 [ERR] mod_freetdm.c:2177 Session for channel
>> 10:3 not found [UUID: N/A]
>> 2011-03-22 09:18:05.331547 [ERR] mod_freetdm.c:2177 Session for channel
>> 10:10 not found [UUID: N/A]
>> 2011-03-22 09:18:06.768937 [ERR] mod_freetdm.c:2177 Session for channel
>> 24:14 not found [UUID: N/A]
>> 2011-03-22 09:18:07.248782 [ERR] mod_freetdm.c:2177 Session for channel
>> 24:26 not found [UUID: N/A]
>> 2011-03-22 09:18:07.797798 [ERR] mod_freetdm.c:2177 Session for channel
>> 24:15 not found [UUID: N/A]
>>
>> and many calls are dropped (I never seen these errors before).
>> Removing the sleep from dialplan, all revert to normal.
>>
>> Stephen
>>
>> On Tue, Mar 22, 2011 at 6:26 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>>
>>> On Tue, Mar 22, 2011 at 1:29 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>>
>>>> I have tested this trick with 1 call and it worked fine but when tested
>>>> on a production environment doesn't work.
>>>>
>>>> The problem seems to be the "sleep" called by "execute_extension"
>>>> executed on answer.
>>>>
>>>> If I activate this extension, I see in freeswitch log:
>>>>
>>>> 2011-03-22 09:18:10.191304 [CRIT] switch_core_session.c:1646 Throttle
>>>> Error! 1326
>>>> 2011-03-22 09:18:10.193260 [CRIT] switch_core_session.c:1646 Throttle
>>>> Error! 1326
>>>> 2011-03-22 09:18:10.194796 [CRIT] switch_core_session.c:1646 Throttle
>>>> Error! 1323
>>>> 2011-03-22 09:18:10.194796 [CRIT] switch_core_session.c:1646 Throttle
>>>> Error! 1323
>>>> 2011-03-22 09:18:10.199238 [CRIT] switch_core_session.c:1646 Throttle
>>>> Error! 1317
>>>> 2011-03-22 09:18:10.204815 [CRIT] switch_core_session.c:1646 Throttle
>>>> Error! 1314
>>>> 2011-03-22 09:18:10.204815 [CRIT] mod_sofia.c:3896 Error Creating
>>>> Session
>>>> 2011-03-22 09:18:10.214916 [CRIT] switch_core_session.c:1646 Throttle
>>>> Error! 1302
>>>> 2011-03-22 09:18:10.214916 [CRIT] mod_sofia.c:3896 Error Creating
>>>> Session
>>>> *2011-03-22 09:18:10.377962 [CRIT] switch_time.c:788 Over Session Rate
>>>> of 200!*
>>>>
>>> Notice the above line!
>>>
>>> You are overwhelming your system with too many calls in such a short
>>> period of time. Slow things down a bit on your call generation. Also, you
>>> can probably experiment with the sleep time. 500ms might be too long. Try
>>> setting it to something shorter, like 100ms or so. That might help speed
>>> things up a bit. Shaving 400ms off the call may not seem like a lot but if
>>> you're doing hundreds of calls per second then that 400ms will add up
>>> quickly.
>>>
>>> -MC
>>>
>>>
>>>
>>>> 2011-03-22 09:18:33.938094 [CRIT] switch_core_session.c:1646 Throttle
>>>> Error! 1266
>>>> 2011-03-22 09:18:33.938094 [CRIT] switch_core_session.c:1646 Throttle
>>>> Error! 1266
>>>> 2011-03-22 09:18:33.939077 [CRIT] switch_core_session.c:1646 Throttle
>>>> Error! 1266
>>>> 2011-03-22 09:18:33.939077 [CRIT] switch_core_session.c:1646 Throttle
>>>> Error! 1266
>>>> 2011-03-22 09:18:33.944077 [CRIT] switch_core_session.c:1646 Throttle
>>>> Error! 1264
>>>> 2011-03-22 09:18:33.944077 [CRIT] mod_sofia.c:3896 Error Creating
>>>> Session
>>>>
>>>> Stephen
>>>>
>>>> On Tue, Mar 22, 2011 at 8:54 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>>>
>>>>> Thank you Michael, with this trick it works fine!
>>>>>
>>>>>
>>>>> On Tue, Mar 22, 2011 at 6:33 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>> Add this to your dialplan...
>>>>>>
>>>>>> <extension name="start tone detect">
>>>>>>    <condition field="destination_number"
>>>>>> data="custom_start_tone_detect">
>>>>>>     <action application="sleep" data="500"/>
>>>>>>     <action application="tone_detect" data="MYTONE 680 ro +5000 set
>>>>>> MYTONE=1"/>
>>>>>>   </condition>
>>>>>> </extension>
>>>>>>
>>>>>> ...then change your export...
>>>>>> <action application="export"
>>>>>> data="nolocal:execute_on_answer=execute_extension
>>>>>> custom_start_tone_detect"/>
>>>>>>
>>>>>> Let us know if that does the trick.
>>>>>>
>>>>>> -MC
>>>>>>
>>>>>> On Mon, Mar 21, 2011 at 12:33 PM, Stephen Wilde <wstephen80 at gmail.com
>>>>>> > wrote:
>>>>>>
>>>>>>> I start the tone_detect on answer using "execute_on_answer" as:
>>>>>>>
>>>>>>> <action application="export"
>>>>>>> data="nolocal:execute_on_answer=tone_detect MYTONE 680 ro +5000 set
>>>>>>> MYTONE=1" />
>>>>>>>
>>>>>>> but I have the necessity to start the tone_detect for example 500ms
>>>>>>> after answer: there is a way to do that in dialplan?
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From frankie.k.yiu at gmail.com  Tue Mar 22 23:24:08 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Tue, 22 Mar 2011 13:24:08 -0700
Subject: [Freeswitch-users] Gawk.exe crashes when building on Windows 7
	64bits w/ VS 2008
Message-ID: <AANLkTik9kcYZy84LpUNKNytu6PgA0YYPQNxgo8dSJs4m@mail.gmail.com>

Hi there,

I am wondering if anyone else encounters the same issue.  I am running
Windows 7 64bits and when I build FreeSwitch in Visual Studio 2008 (SP1) for
WIN32, it crashes/compilation error when opening a Gawk program which causes
files can not be automatically generated.  I even installed a newer version
3.1.7 and it still crashes.

Please let me know if you have resolved this issue.
Thanks,
Frankie
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From jeff at jefflenk.com  Tue Mar 22 23:31:21 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 22 Mar 2011 13:31:21 -0700 (PDT)
Subject: [Freeswitch-users] Gawk.exe crashes when building on Windows 7
	64bits w/ VS 2008
In-Reply-To: <AANLkTik9kcYZy84LpUNKNytu6PgA0YYPQNxgo8dSJs4m@mail.gmail.com>
References: <AANLkTik9kcYZy84LpUNKNytu6PgA0YYPQNxgo8dSJs4m@mail.gmail.com>
Message-ID: <1300825881905-6197848.post@n2.nabble.com>

This is a problem with your git autocrlf. You must have that turned off.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Gawk-exe-crashes-when-building-on-Windows-7-64bits-w-VS-2008-tp6197846p6197848.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From devel at omninet.eu  Tue Mar 22 22:42:22 2011
From: devel at omninet.eu (Anestis Mavro)
Date: Tue, 22 Mar 2011 21:42:22 +0200
Subject: [Freeswitch-users] How can I set the crypto suite on b-leg for SRTP?
Message-ID: <E266FB1A10564EF0A60EC751762BDEA0@omni1.local>

Hi,

 

I have configured TLS and SRTP on my FS and it seems to be working.

I have several phones and gateways and I have been testing SRTP.

 

My only problem is that FS sends in B-leg only one crypto suite:
AES_CM_128_HMAC_SHA1_32

Is there a way to enable also AES_CM_128_HMAC_SHA1_80 on the b-leg?

 

The calling phone sends all three crypto suites in the sdp, but FS only one
to the called phone.

 

Thank you

Anestis

 

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From anthony.minessale at gmail.com  Wed Mar 23 00:14:53 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 22 Mar 2011 16:14:53 -0500
Subject: [Freeswitch-users] Event-List: double Content-Lenght?
In-Reply-To: <AANLkTimDH-bvRh+R+w0TUaj+5WHTR7MZHymtX==vQ9j9@mail.gmail.com>
References: <AANLkTimDH-bvRh+R+w0TUaj+5WHTR7MZHymtX==vQ9j9@mail.gmail.com>
Message-ID: <AANLkTikr6OybRginxfsmfkjheqF9xCN9V-YOLY=cvf9M@mail.gmail.com>

this is an encapsulated event inside a container event.

if you were framing this on the wire you would read the headers
(everything up to the gap)
the content-length of 625 tells you to read exactly that many bytes
and content type text/event-plain

now when you read 625 bytes you can continue to read on the wire for a
new event.
Meanwhile you url decode that payload and you have another event who
also has a payload of 41 bytes.
Now you can read these 41 bytes to get the payload which is the exact
size of the message contained in the event.

None of this matters to you if you use the ESL lib supplied with FS to
read events into a scripting language.



On Tue, Mar 22, 2011 at 6:45 AM, I?aki Baz Castillo <ibc at aliax.net> wrote:
> Hi, I'm trying to understand the following example in the doc:
>
> ?http://wiki.freeswitch.org/wiki/Event_List
>
> Upon job completion, server response
> ----------------------
> Content-Length: 625
> Content-Type: text/event-plain
>
> Job-UUID: 7f4db78a-17d7-11dd-b7a0-db4edd065621
> Job-Command: originate
> Job-Command-Arg: sofia/default/1005%20'%26park'
> Event-Name: BACKGROUND_JOB
> Core-UUID: 42bdf272-16e6-11dd-b7a0-db4edd065621
> FreeSWITCH-Hostname: ser
> FreeSWITCH-IPv4: 192.168.1.104
> FreeSWITCH-IPv6: 127.0.0.1
> Event-Date-Local: 2008-05-02%2007%3A37%3A03
> Event-Date-GMT: Thu,%2001%20May%202008%2023%3A37%3A03%20GMT
> Event-Date-timestamp: 1209685023894968
> Event-Calling-File: mod_event_socket.c
> Event-Calling-Function: api_exec
> Event-Calling-Line-Number: 609
> Content-Length: 41
>
> +OK 7f4de4bc-17d7-11dd-b7a0-db4edd065621
> ----------------------
>
>
> What is the purpose of the first "Content-Length: 625"? Why not just
> send all the headers with no Content-Lenght? (just leave the second
> Content-Length:41 which is the length of the job uuid).
>
> So what is the purpose of the first Content-Length? A parser doesn't
> need it at all as it can read headers until it finds "Content-Length"
> (so it reads N bytes more) or it finds a double \n\n (so event info is
> terminated).
>
> Thanks for any clarification.
>
>
> --
> I?aki Baz Castillo
> <ibc at aliax.net>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 ibc at aliax.net  Wed Mar 23 01:13:24 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Tue, 22 Mar 2011 23:13:24 +0100
Subject: [Freeswitch-users] Event-List: double Content-Lenght?
In-Reply-To: <AANLkTikr6OybRginxfsmfkjheqF9xCN9V-YOLY=cvf9M@mail.gmail.com>
References: <AANLkTimDH-bvRh+R+w0TUaj+5WHTR7MZHymtX==vQ9j9@mail.gmail.com>
	<AANLkTikr6OybRginxfsmfkjheqF9xCN9V-YOLY=cvf9M@mail.gmail.com>
Message-ID: <AANLkTinMx_z0vmV1ZZQbt0-7P67Jd73gmVDC7ASykD8w@mail.gmail.com>

2011/3/22 Anthony Minessale <anthony.minessale at gmail.com>:
> this is an encapsulated event inside a container event.
>
> if you were framing this on the wire you would read the headers
> (everything up to the gap)
> the content-length of 625 tells you to read exactly that many bytes
> and content type text/event-plain
>
> now when you read 625 bytes you can continue to read on the wire for a
> new event.
> Meanwhile you url decode that payload and you have another event who
> also has a payload of 41 bytes.
> Now you can read these 41 bytes to get the payload which is the exact
> size of the message contained in the event.

Ok, understood. I still think it's a bit strange but I understand.


> None of this matters to you if you use the ESL lib supplied with FS to
> read events into a scripting language.

For now I just want to fully understand the protocol. I will use some
existing erver library (Ruby library) for Outbound Listener, or maybe
create my own one.

Thanks a lot for your explanation.

-- 
I?aki Baz Castillo
<ibc at aliax.net>


From msc at freeswitch.org  Wed Mar 23 01:36:01 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Mar 2011 15:36:01 -0700
Subject: [Freeswitch-users] Delayed start of tone_detect
In-Reply-To: <AANLkTinh4xNnpHK-f_eE9QzCe4Mo8QwuWfec42wDw-jN@mail.gmail.com>
References: <AANLkTims-MfPkq_tr-JQd_ih=gPd1i8gXSFSX2tF_m8y@mail.gmail.com>
	<AANLkTi=s1iwAm9qB3=y97U8CmLs+T=pL9ZzAToE2N5Ed@mail.gmail.com>
	<AANLkTimm7+5GxsRfJNRcRcRAqmtCUBwv8UWrn4Qk2_6r@mail.gmail.com>
	<AANLkTikn1V3ypmbZW4NQ6qSgRy_7ENdOWT5eWrCHs-C0@mail.gmail.com>
	<AANLkTin8R6roSmw+3ejr8ud_VUMYEkbQ1Hz-xtmmYh54@mail.gmail.com>
	<AANLkTiny4+NvH9WFWC51VHDCYM6pB7_QrWn8_CyontFf@mail.gmail.com>
	<AANLkTimE=Ovn+HR8dnAZ4NmPydSiepS-Rb7rzkTshP7=@mail.gmail.com>
	<AANLkTinh4xNnpHK-f_eE9QzCe4Mo8QwuWfec42wDw-jN@mail.gmail.com>
Message-ID: <AANLkTimEAUJnSutb8oLU_=6vkubQ6TtWEop0XS+PUk+u@mail.gmail.com>

How about instead of sleeping you do something like this:
<action application="playback" data="silence_stream://500"/>

-MC

On Tue, Mar 22, 2011 at 12:53 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:

> Ok, but this is not the solution for my initial problem: to delay the
> tone_detect of 500mS after answer.
> But I don't see any other way to do that (except modifying the
> "switch_ivr_tone_detect_session" to ignore the first n samples).
>
> Stephen
>
>
> On Tue, Mar 22, 2011 at 7:46 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> set the sleep value to something really low, like 20ms or something. Find
>> the lowest value that still lets the tone_detect work.
>> -MC
>>
>>
>> On Tue, Mar 22, 2011 at 10:59 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>
>>> I cannot slow down the call rate, this is a real traffic and is not
>>> generate but the session rate it's not a real problem because my session
>>> rate is never more then 100cps with a medium value of 50cps so, when I
>>> activate this sleep, Freeswitch calculate a wrong CPS value. It seems that
>>> during sleep, FS freezes, as the sleep is performed synchronously.
>>>
>>> Another effect is:
>>>
>>> 2011-03-22 09:17:57.551069 [ERR] mod_freetdm.c:2177 Session for channel
>>> 29:15 not found [UUID: N/A]
>>> 2011-03-22 09:18:00.181462 [ERR] mod_freetdm.c:2177 Session for channel
>>> 29:18 not found [UUID: N/A]
>>> 2011-03-22 09:18:00.301001 [ERR] mod_freetdm.c:2177 Session for channel
>>> 25:25 not found [UUID: N/A]
>>> 2011-03-22 09:18:03.931076 [ERR] mod_freetdm.c:2177 Session for channel
>>> 10:3 not found [UUID: N/A]
>>> 2011-03-22 09:18:05.331547 [ERR] mod_freetdm.c:2177 Session for channel
>>> 10:10 not found [UUID: N/A]
>>> 2011-03-22 09:18:06.768937 [ERR] mod_freetdm.c:2177 Session for channel
>>> 24:14 not found [UUID: N/A]
>>> 2011-03-22 09:18:07.248782 [ERR] mod_freetdm.c:2177 Session for channel
>>> 24:26 not found [UUID: N/A]
>>> 2011-03-22 09:18:07.797798 [ERR] mod_freetdm.c:2177 Session for channel
>>> 24:15 not found [UUID: N/A]
>>>
>>> and many calls are dropped (I never seen these errors before).
>>> Removing the sleep from dialplan, all revert to normal.
>>>
>>> Stephen
>>>
>>> On Tue, Mar 22, 2011 at 6:26 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>>
>>>> On Tue, Mar 22, 2011 at 1:29 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>>>
>>>>> I have tested this trick with 1 call and it worked fine but when tested
>>>>> on a production environment doesn't work.
>>>>>
>>>>> The problem seems to be the "sleep" called by "execute_extension"
>>>>> executed on answer.
>>>>>
>>>>> If I activate this extension, I see in freeswitch log:
>>>>>
>>>>> 2011-03-22 09:18:10.191304 [CRIT] switch_core_session.c:1646 Throttle
>>>>> Error! 1326
>>>>> 2011-03-22 09:18:10.193260 [CRIT] switch_core_session.c:1646 Throttle
>>>>> Error! 1326
>>>>> 2011-03-22 09:18:10.194796 [CRIT] switch_core_session.c:1646 Throttle
>>>>> Error! 1323
>>>>> 2011-03-22 09:18:10.194796 [CRIT] switch_core_session.c:1646 Throttle
>>>>> Error! 1323
>>>>> 2011-03-22 09:18:10.199238 [CRIT] switch_core_session.c:1646 Throttle
>>>>> Error! 1317
>>>>> 2011-03-22 09:18:10.204815 [CRIT] switch_core_session.c:1646 Throttle
>>>>> Error! 1314
>>>>> 2011-03-22 09:18:10.204815 [CRIT] mod_sofia.c:3896 Error Creating
>>>>> Session
>>>>> 2011-03-22 09:18:10.214916 [CRIT] switch_core_session.c:1646 Throttle
>>>>> Error! 1302
>>>>> 2011-03-22 09:18:10.214916 [CRIT] mod_sofia.c:3896 Error Creating
>>>>> Session
>>>>> *2011-03-22 09:18:10.377962 [CRIT] switch_time.c:788 Over Session Rate
>>>>> of 200!*
>>>>>
>>>> Notice the above line!
>>>>
>>>> You are overwhelming your system with too many calls in such a short
>>>> period of time. Slow things down a bit on your call generation. Also, you
>>>> can probably experiment with the sleep time. 500ms might be too long. Try
>>>> setting it to something shorter, like 100ms or so. That might help speed
>>>> things up a bit. Shaving 400ms off the call may not seem like a lot but if
>>>> you're doing hundreds of calls per second then that 400ms will add up
>>>> quickly.
>>>>
>>>> -MC
>>>>
>>>>
>>>>
>>>>> 2011-03-22 09:18:33.938094 [CRIT] switch_core_session.c:1646 Throttle
>>>>> Error! 1266
>>>>> 2011-03-22 09:18:33.938094 [CRIT] switch_core_session.c:1646 Throttle
>>>>> Error! 1266
>>>>> 2011-03-22 09:18:33.939077 [CRIT] switch_core_session.c:1646 Throttle
>>>>> Error! 1266
>>>>> 2011-03-22 09:18:33.939077 [CRIT] switch_core_session.c:1646 Throttle
>>>>> Error! 1266
>>>>> 2011-03-22 09:18:33.944077 [CRIT] switch_core_session.c:1646 Throttle
>>>>> Error! 1264
>>>>> 2011-03-22 09:18:33.944077 [CRIT] mod_sofia.c:3896 Error Creating
>>>>> Session
>>>>>
>>>>> Stephen
>>>>>
>>>>> On Tue, Mar 22, 2011 at 8:54 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>>>>
>>>>>> Thank you Michael, with this trick it works fine!
>>>>>>
>>>>>>
>>>>>> On Tue, Mar 22, 2011 at 6:33 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>>
>>>>>>> Add this to your dialplan...
>>>>>>>
>>>>>>> <extension name="start tone detect">
>>>>>>>    <condition field="destination_number"
>>>>>>> data="custom_start_tone_detect">
>>>>>>>     <action application="sleep" data="500"/>
>>>>>>>     <action application="tone_detect" data="MYTONE 680 ro +5000 set
>>>>>>> MYTONE=1"/>
>>>>>>>   </condition>
>>>>>>> </extension>
>>>>>>>
>>>>>>> ...then change your export...
>>>>>>> <action application="export"
>>>>>>> data="nolocal:execute_on_answer=execute_extension
>>>>>>> custom_start_tone_detect"/>
>>>>>>>
>>>>>>> Let us know if that does the trick.
>>>>>>>
>>>>>>> -MC
>>>>>>>
>>>>>>> On Mon, Mar 21, 2011 at 12:33 PM, Stephen Wilde <
>>>>>>> wstephen80 at gmail.com> wrote:
>>>>>>>
>>>>>>>> I start the tone_detect on answer using "execute_on_answer" as:
>>>>>>>>
>>>>>>>> <action application="export"
>>>>>>>> data="nolocal:execute_on_answer=tone_detect MYTONE 680 ro +5000 set
>>>>>>>> MYTONE=1" />
>>>>>>>>
>>>>>>>> but I have the necessity to start the tone_detect for example 500ms
>>>>>>>> after answer: there is a way to do that in dialplan?
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From wstephen80 at gmail.com  Wed Mar 23 01:56:04 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 22 Mar 2011 23:56:04 +0100
Subject: [Freeswitch-users] Delayed start of tone_detect
In-Reply-To: <AANLkTimEAUJnSutb8oLU_=6vkubQ6TtWEop0XS+PUk+u@mail.gmail.com>
References: <AANLkTims-MfPkq_tr-JQd_ih=gPd1i8gXSFSX2tF_m8y@mail.gmail.com>
	<AANLkTi=s1iwAm9qB3=y97U8CmLs+T=pL9ZzAToE2N5Ed@mail.gmail.com>
	<AANLkTimm7+5GxsRfJNRcRcRAqmtCUBwv8UWrn4Qk2_6r@mail.gmail.com>
	<AANLkTikn1V3ypmbZW4NQ6qSgRy_7ENdOWT5eWrCHs-C0@mail.gmail.com>
	<AANLkTin8R6roSmw+3ejr8ud_VUMYEkbQ1Hz-xtmmYh54@mail.gmail.com>
	<AANLkTiny4+NvH9WFWC51VHDCYM6pB7_QrWn8_CyontFf@mail.gmail.com>
	<AANLkTimE=Ovn+HR8dnAZ4NmPydSiepS-Rb7rzkTshP7=@mail.gmail.com>
	<AANLkTinh4xNnpHK-f_eE9QzCe4Mo8QwuWfec42wDw-jN@mail.gmail.com>
	<AANLkTimEAUJnSutb8oLU_=6vkubQ6TtWEop0XS+PUk+u@mail.gmail.com>
Message-ID: <AANLkTi=cjupw5BJhTKL65JNmWWYmhQHqYX+guuq=Cq1L@mail.gmail.com>

The channel where I'm doing tone_detect is an outbound channel that is
bridged to the inbound channel. The playback can works?

Stephen

On Tue, Mar 22, 2011 at 11:36 PM, Michael Collins <msc at freeswitch.org>wrote:

> How about instead of sleeping you do something like this:
> <action application="playback" data="silence_stream://500"/>
>
> -MC
>
>
> On Tue, Mar 22, 2011 at 12:53 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> Ok, but this is not the solution for my initial problem: to delay the
>> tone_detect of 500mS after answer.
>> But I don't see any other way to do that (except modifying the
>> "switch_ivr_tone_detect_session" to ignore the first n samples).
>>
>> Stephen
>>
>>
>> On Tue, Mar 22, 2011 at 7:46 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> set the sleep value to something really low, like 20ms or something. Find
>>> the lowest value that still lets the tone_detect work.
>>> -MC
>>>
>>>
>>> On Tue, Mar 22, 2011 at 10:59 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>>
>>>> I cannot slow down the call rate, this is a real traffic and is not
>>>> generate but the session rate it's not a real problem because my session
>>>> rate is never more then 100cps with a medium value of 50cps so, when I
>>>> activate this sleep, Freeswitch calculate a wrong CPS value. It seems that
>>>> during sleep, FS freezes, as the sleep is performed synchronously.
>>>>
>>>> Another effect is:
>>>>
>>>> 2011-03-22 09:17:57.551069 [ERR] mod_freetdm.c:2177 Session for channel
>>>> 29:15 not found [UUID: N/A]
>>>> 2011-03-22 09:18:00.181462 [ERR] mod_freetdm.c:2177 Session for channel
>>>> 29:18 not found [UUID: N/A]
>>>> 2011-03-22 09:18:00.301001 [ERR] mod_freetdm.c:2177 Session for channel
>>>> 25:25 not found [UUID: N/A]
>>>> 2011-03-22 09:18:03.931076 [ERR] mod_freetdm.c:2177 Session for channel
>>>> 10:3 not found [UUID: N/A]
>>>> 2011-03-22 09:18:05.331547 [ERR] mod_freetdm.c:2177 Session for channel
>>>> 10:10 not found [UUID: N/A]
>>>> 2011-03-22 09:18:06.768937 [ERR] mod_freetdm.c:2177 Session for channel
>>>> 24:14 not found [UUID: N/A]
>>>> 2011-03-22 09:18:07.248782 [ERR] mod_freetdm.c:2177 Session for channel
>>>> 24:26 not found [UUID: N/A]
>>>> 2011-03-22 09:18:07.797798 [ERR] mod_freetdm.c:2177 Session for channel
>>>> 24:15 not found [UUID: N/A]
>>>>
>>>> and many calls are dropped (I never seen these errors before).
>>>> Removing the sleep from dialplan, all revert to normal.
>>>>
>>>> Stephen
>>>>
>>>> On Tue, Mar 22, 2011 at 6:26 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>>
>>>>> On Tue, Mar 22, 2011 at 1:29 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>>>>
>>>>>> I have tested this trick with 1 call and it worked fine but when
>>>>>> tested on a production environment doesn't work.
>>>>>>
>>>>>> The problem seems to be the "sleep" called by "execute_extension"
>>>>>> executed on answer.
>>>>>>
>>>>>> If I activate this extension, I see in freeswitch log:
>>>>>>
>>>>>> 2011-03-22 09:18:10.191304 [CRIT] switch_core_session.c:1646 Throttle
>>>>>> Error! 1326
>>>>>> 2011-03-22 09:18:10.193260 [CRIT] switch_core_session.c:1646 Throttle
>>>>>> Error! 1326
>>>>>> 2011-03-22 09:18:10.194796 [CRIT] switch_core_session.c:1646 Throttle
>>>>>> Error! 1323
>>>>>> 2011-03-22 09:18:10.194796 [CRIT] switch_core_session.c:1646 Throttle
>>>>>> Error! 1323
>>>>>> 2011-03-22 09:18:10.199238 [CRIT] switch_core_session.c:1646 Throttle
>>>>>> Error! 1317
>>>>>> 2011-03-22 09:18:10.204815 [CRIT] switch_core_session.c:1646 Throttle
>>>>>> Error! 1314
>>>>>> 2011-03-22 09:18:10.204815 [CRIT] mod_sofia.c:3896 Error Creating
>>>>>> Session
>>>>>> 2011-03-22 09:18:10.214916 [CRIT] switch_core_session.c:1646 Throttle
>>>>>> Error! 1302
>>>>>> 2011-03-22 09:18:10.214916 [CRIT] mod_sofia.c:3896 Error Creating
>>>>>> Session
>>>>>> *2011-03-22 09:18:10.377962 [CRIT] switch_time.c:788 Over Session
>>>>>> Rate of 200!*
>>>>>>
>>>>> Notice the above line!
>>>>>
>>>>> You are overwhelming your system with too many calls in such a short
>>>>> period of time. Slow things down a bit on your call generation. Also, you
>>>>> can probably experiment with the sleep time. 500ms might be too long. Try
>>>>> setting it to something shorter, like 100ms or so. That might help speed
>>>>> things up a bit. Shaving 400ms off the call may not seem like a lot but if
>>>>> you're doing hundreds of calls per second then that 400ms will add up
>>>>> quickly.
>>>>>
>>>>> -MC
>>>>>
>>>>>
>>>>>
>>>>>> 2011-03-22 09:18:33.938094 [CRIT] switch_core_session.c:1646 Throttle
>>>>>> Error! 1266
>>>>>> 2011-03-22 09:18:33.938094 [CRIT] switch_core_session.c:1646 Throttle
>>>>>> Error! 1266
>>>>>> 2011-03-22 09:18:33.939077 [CRIT] switch_core_session.c:1646 Throttle
>>>>>> Error! 1266
>>>>>> 2011-03-22 09:18:33.939077 [CRIT] switch_core_session.c:1646 Throttle
>>>>>> Error! 1266
>>>>>> 2011-03-22 09:18:33.944077 [CRIT] switch_core_session.c:1646 Throttle
>>>>>> Error! 1264
>>>>>> 2011-03-22 09:18:33.944077 [CRIT] mod_sofia.c:3896 Error Creating
>>>>>> Session
>>>>>>
>>>>>> Stephen
>>>>>>
>>>>>> On Tue, Mar 22, 2011 at 8:54 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>>>>>
>>>>>>> Thank you Michael, with this trick it works fine!
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Mar 22, 2011 at 6:33 AM, Michael Collins <msc at freeswitch.org
>>>>>>> > wrote:
>>>>>>>
>>>>>>>> Add this to your dialplan...
>>>>>>>>
>>>>>>>> <extension name="start tone detect">
>>>>>>>>    <condition field="destination_number"
>>>>>>>> data="custom_start_tone_detect">
>>>>>>>>     <action application="sleep" data="500"/>
>>>>>>>>     <action application="tone_detect" data="MYTONE 680 ro +5000 set
>>>>>>>> MYTONE=1"/>
>>>>>>>>   </condition>
>>>>>>>> </extension>
>>>>>>>>
>>>>>>>> ...then change your export...
>>>>>>>> <action application="export"
>>>>>>>> data="nolocal:execute_on_answer=execute_extension
>>>>>>>> custom_start_tone_detect"/>
>>>>>>>>
>>>>>>>> Let us know if that does the trick.
>>>>>>>>
>>>>>>>> -MC
>>>>>>>>
>>>>>>>> On Mon, Mar 21, 2011 at 12:33 PM, Stephen Wilde <
>>>>>>>> wstephen80 at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> I start the tone_detect on answer using "execute_on_answer" as:
>>>>>>>>>
>>>>>>>>> <action application="export"
>>>>>>>>> data="nolocal:execute_on_answer=tone_detect MYTONE 680 ro +5000 set
>>>>>>>>> MYTONE=1" />
>>>>>>>>>
>>>>>>>>> but I have the necessity to start the tone_detect for example 500ms
>>>>>>>>> after answer: there is a way to do that in dialplan?
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ayhkor at gmail.com  Wed Mar 23 02:07:02 2011
From: ayhkor at gmail.com (deniro)
Date: Tue, 22 Mar 2011 19:07:02 -0400
Subject: [Freeswitch-users] playAndGetDigits not working?
In-Reply-To: <AANLkTim253GfYSJaGXAX86u2_YdGzNGwxK6RGb8Z+kUh@mail.gmail.com>
References: <AANLkTimpyZHpXOf9ZHaKCcXdZH=mo-2y2rWJLWji=Dda@mail.gmail.com>
	<AANLkTim253GfYSJaGXAX86u2_YdGzNGwxK6RGb8Z+kUh@mail.gmail.com>
Message-ID: <AANLkTimmdVo6OcrwiZB7AnH8Lw=QseRLFowvVU4A-B_p@mail.gmail.com>

still same. not working
what else  can it be?
thx



On Tue, Mar 22, 2011 at 1:20 PM, Michael Collins <msc at freeswitch.org> wrote:

> You may need to do \\d+ instead of \d+ in your expression.
> -MC
>
>   On Mon, Mar 21, 2011 at 10:46 PM, deniro <ayhkor at gmail.com> wrote:
>
>>
>> I am unable to get the pin number and pass it to $digits ($digits comes
>> blank)
>> appreciate any feedback
>> thx
>> deniro--
>>
>>
>>
>> my($digits)=$session->playAndGetDigits(1,5,1,8000,"#","conference/conf-pin.wav","ivr/ivr-that_was_an_invalid_entry.wav","","^7\d{4}\$");
>> or
>> my($digits)=$session->playAndGetDigits(1,5,1,8000,"#","conference/conf-pin.wav","ivr/ivr-that_was_an_invalid_entry.wav","",
>> "\d+");
>> freeswitch::console_log("info", "*Got dtmf: .. $digits ..*\n");
>>
>> freeswitch console
>> 2011-03-22 01:29:26.208664 [INFO] switch_cpp.cpp:1177* Got dtmf: ..  ..
>> *2011-03-22 01:29:26.208664 [ERR] mod_perl.c:72 [require
>> '/opt/freeswitch/scripts/fs1.pl';]
>> /opt/freeswitch/scripts/fs1.pl did not return a true value at (eval 2)
>> line 1.
>>  each case getting "that was invalid PIN"
>> entering pin starting with 7 maximum 5 digits in first case
>> or any digits upto maximum 5 digits in second case
>> tried also putting "#" at the end when dialing
>>
>> none worked
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Mar 23 02:09:54 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Mar 2011 16:09:54 -0700
Subject: [Freeswitch-users] inbound-late-negotiation and attended
	transfer
In-Reply-To: <AANLkTimVm30-n_p7SiJxsJ9pso692ogNqiaSWp900nUg@mail.gmail.com>
References: <AANLkTimDT=699aCgz-h9yk_+FyXb_tf9oDa=KMMQRZ04@mail.gmail.com>
	<AANLkTikzP4urgvo6CfvOBFkXGAHrSLn_VjXs0+bDvpsT@mail.gmail.com>
	<AANLkTimVm30-n_p7SiJxsJ9pso692ogNqiaSWp900nUg@mail.gmail.com>
Message-ID: <AANLkTik6GUXYODx6wScD77YcVR4MoF34oto==SndE_BH@mail.gmail.com>

On Tue, Mar 22, 2011 at 11:09 AM, Dmitry Bely <dmitry.bely at gmail.com> wrote:

> On Tue, Mar 22, 2011 at 8:18 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> > How are you performing the attended transfer? What kind of phone is this?
> > -MC
>
> Yes, the attended transfer using phone's capabilities as explained
> here: http://www.youtube.com/watch?v=VpEVpr-4y-U
> The phone is Grandstream GXP-2000.
>
> Here is an example. There was two active calls:
>
> gw <-> 1000 (PCMU)
> 1000 <-> 1004 (G722)
>

Is the gateway in the external profile? If so, try setting the
disable-transcoding param on that profile. Or you could try FreeSWITCH's
built in att-xfer using the *4 key combo.

-MC


> Now operator transfers the fist call to 1004. The log:
>
> 2011-03-18 17:08:27.221372 [DEBUG] sofia.c:4659 Channel
> sofia/internal/1000 at 192.168.121.66 entering
> state [received][100]
> 2011-03-18 17:08:27.222378 [DEBUG] sofia.c:4670 Remote SDP:
> v=0
> o=1000 8001 8002 IN IP4 192.168.121.153
> s=SIP Call
> c=IN IP4 192.168.121.153
> t=0 0
> m=audio 5030 RTP/AVP 9 0 8 18 4 99 3 2
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:99 iLBC/8000
> a=fmtp:99 mode=20
> a=rtpmap:3 GSM/8000
> a=rtpmap:2 G726-32/8000
> a=sendonly
> a=ptime:20
>
> 2011-03-18 17:08:27.222378 [DEBUG] switch_channel.c:1377
> (sofia/internal/1000 at 192.168.121.66) Callstate Change ACTIVE -> HELD
> 2011-03-18 17:08:27.222378 [DEBUG] switch_core_session.c:954 Send
> signal sofia/internal/sip:1004 at 192.168.121.136:5060 [BREAK]
> 2011-03-18 17:08:27.224386 [DEBUG] switch_core_session.c:709 Send
> signal sofia/internal/sip:1004 at 192.168.121.136:5060 [BREAK]
> 2011-03-18 17:08:27.364243 [DEBUG] switch_ivr.c:563
> sofia/internal/sip:1004 at 192.168.121.136:5060 Command Execute
> playback(local_stream://moh)
> EXECUTE sofia/internal/sip:1004 at 192.168.121.136:5060
> playback(local_stream://moh)
> 2011-03-18 17:08:27.364243 [WARNING] mod_local_stream.c:393 Unknown
> source moh, trying 'default'
> 2011-03-18 17:08:27.364243 [DEBUG] mod_local_stream.c:421 Opening
> Stream [default] 8000hz
> 2011-03-18 17:08:27.364243 [DEBUG] switch_core_file.c:176 File moh
> sample rate 8000 doesn't match requested rate 16000
> 2011-03-18 17:08:27.364243 [DEBUG] switch_ivr_play_say.c:1244 Codec
> Activated L16 at 16000hz 1 channels 20ms
> 2011-03-18 17:08:27.472662 [DEBUG] sofia_glue.c:4474 Audio Codec
> Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000]
> 2011-03-18 17:08:27.472662 [DEBUG] sofia_glue.c:2690 Already using G722
> 2011-03-18 17:08:27.472662 [DEBUG] sofia_glue.c:2972 Audio params are
> unchanged for sofia/internal/1000 at 192.168.121.66.
> 2011-03-18 17:08:27.472662 [DEBUG] sofia.c:5070 Processing updated SDP
> 2011-03-18 17:08:27.473667 [DEBUG] sofia.c:4659 Channel
> sofia/internal/1000 at 192.168.121.66 entering state [completed][200]
> 2011-03-18 17:08:27.592069 [DEBUG] sofia.c:4659 Channel
> sofia/internal/1000 at 192.168.121.66 entering state [ready][200]
> 2011-03-18 17:08:27.769589 [DEBUG] sofia.c:5520 Process REFER to
> [1004 at 192.168.121.66]
> 2011-03-18 17:08:27.769589 [DEBUG] sofia.c:5539 Replaces:
> [5cd372e1e81966ca at 192.168.121.153
> ;from-tag=d55f7fbec6efbde7;to-tag=2N49SS5XNjUeB]
> ...
> (no more codec negotiation)
>
> > On Tue, Mar 22, 2011 at 3:11 AM, Dmitry Bely <dmitry.bely at gmail.com>
> wrote:
> >>
> >> I have G722 and PCMU enabled for internal extensions and PCMU only for
> >> an external gateway. Inbound-late-negotiation parameter is set so then
> >> an incoming call arrives PCMU is used without transcoding. But it goes
> >> worse when transfer is involved:
> >>
> >> call 1: gw <-- PCMU--> FreeSwitch <-- PCMU --> operator
> >> call 2: operator <-- G722 --> FreeSwitch <-- G722--> extension
> >> transfer (refer): gw <-- PCMU--> FreeSwitch <-- G722--> extension
> >>
> >> Is it possible to renegotiate a codec during transfer to get rid of
> >> transcoding?
>
> - Dmitry Bely
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From wstephen80 at gmail.com  Wed Mar 23 02:10:36 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Wed, 23 Mar 2011 00:10:36 +0100
Subject: [Freeswitch-users] Delayed start of tone_detect
In-Reply-To: <AANLkTi=cjupw5BJhTKL65JNmWWYmhQHqYX+guuq=Cq1L@mail.gmail.com>
References: <AANLkTims-MfPkq_tr-JQd_ih=gPd1i8gXSFSX2tF_m8y@mail.gmail.com>
	<AANLkTi=s1iwAm9qB3=y97U8CmLs+T=pL9ZzAToE2N5Ed@mail.gmail.com>
	<AANLkTimm7+5GxsRfJNRcRcRAqmtCUBwv8UWrn4Qk2_6r@mail.gmail.com>
	<AANLkTikn1V3ypmbZW4NQ6qSgRy_7ENdOWT5eWrCHs-C0@mail.gmail.com>
	<AANLkTin8R6roSmw+3ejr8ud_VUMYEkbQ1Hz-xtmmYh54@mail.gmail.com>
	<AANLkTiny4+NvH9WFWC51VHDCYM6pB7_QrWn8_CyontFf@mail.gmail.com>
	<AANLkTimE=Ovn+HR8dnAZ4NmPydSiepS-Rb7rzkTshP7=@mail.gmail.com>
	<AANLkTinh4xNnpHK-f_eE9QzCe4Mo8QwuWfec42wDw-jN@mail.gmail.com>
	<AANLkTimEAUJnSutb8oLU_=6vkubQ6TtWEop0XS+PUk+u@mail.gmail.com>
	<AANLkTi=cjupw5BJhTKL65JNmWWYmhQHqYX+guuq=Cq1L@mail.gmail.com>
Message-ID: <AANLkTi=cfX-+k53XbOVo06DSb2GSmPSZBo85_CVGWw1h@mail.gmail.com>

I'll try also with this changes in Freeswitch:

--- a/src/switch_ivr_async.c
+++ b/src/switch_ivr_async.c
@@ -2681,6 +2681,12 @@ SWITCH_DECLARE(switch_status_t)
switch_ivr_tone_detect_session(switch_core_sessi
                }
        }

+    if ((var = switch_channel_get_variable(channel, "tone_detect_delay")))
{
+        int tmp = atoi(var);
+        if (tmp > 0) {
+            cont->list[cont->index].sleep = tmp;
+        }
+    }

        if (zstr(flags)) {
                bflags = SMBF_READ_REPLACE;


using the "tone_detect_delay" channel variable to set the initial delay of
tone detect.

Stephen



On Tue, Mar 22, 2011 at 11:56 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:

> The channel where I'm doing tone_detect is an outbound channel that is
> bridged to the inbound channel. The playback can works?
>
> Stephen
>
>
> On Tue, Mar 22, 2011 at 11:36 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> How about instead of sleeping you do something like this:
>> <action application="playback" data="silence_stream://500"/>
>>
>> -MC
>>
>>
>> On Tue, Mar 22, 2011 at 12:53 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>
>>> Ok, but this is not the solution for my initial problem: to delay the
>>> tone_detect of 500mS after answer.
>>> But I don't see any other way to do that (except modifying the
>>> "switch_ivr_tone_detect_session" to ignore the first n samples).
>>>
>>> Stephen
>>>
>>>
>>> On Tue, Mar 22, 2011 at 7:46 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> set the sleep value to something really low, like 20ms or something.
>>>> Find the lowest value that still lets the tone_detect work.
>>>> -MC
>>>>
>>>>
>>>> On Tue, Mar 22, 2011 at 10:59 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>>>
>>>>> I cannot slow down the call rate, this is a real traffic and is not
>>>>> generate but the session rate it's not a real problem because my session
>>>>> rate is never more then 100cps with a medium value of 50cps so, when I
>>>>> activate this sleep, Freeswitch calculate a wrong CPS value. It seems that
>>>>> during sleep, FS freezes, as the sleep is performed synchronously.
>>>>>
>>>>> Another effect is:
>>>>>
>>>>> 2011-03-22 09:17:57.551069 [ERR] mod_freetdm.c:2177 Session for channel
>>>>> 29:15 not found [UUID: N/A]
>>>>> 2011-03-22 09:18:00.181462 [ERR] mod_freetdm.c:2177 Session for channel
>>>>> 29:18 not found [UUID: N/A]
>>>>> 2011-03-22 09:18:00.301001 [ERR] mod_freetdm.c:2177 Session for channel
>>>>> 25:25 not found [UUID: N/A]
>>>>> 2011-03-22 09:18:03.931076 [ERR] mod_freetdm.c:2177 Session for channel
>>>>> 10:3 not found [UUID: N/A]
>>>>> 2011-03-22 09:18:05.331547 [ERR] mod_freetdm.c:2177 Session for channel
>>>>> 10:10 not found [UUID: N/A]
>>>>> 2011-03-22 09:18:06.768937 [ERR] mod_freetdm.c:2177 Session for channel
>>>>> 24:14 not found [UUID: N/A]
>>>>> 2011-03-22 09:18:07.248782 [ERR] mod_freetdm.c:2177 Session for channel
>>>>> 24:26 not found [UUID: N/A]
>>>>> 2011-03-22 09:18:07.797798 [ERR] mod_freetdm.c:2177 Session for channel
>>>>> 24:15 not found [UUID: N/A]
>>>>>
>>>>> and many calls are dropped (I never seen these errors before).
>>>>> Removing the sleep from dialplan, all revert to normal.
>>>>>
>>>>> Stephen
>>>>>
>>>>> On Tue, Mar 22, 2011 at 6:26 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>>
>>>>>> On Tue, Mar 22, 2011 at 1:29 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>>>>>
>>>>>>> I have tested this trick with 1 call and it worked fine but when
>>>>>>> tested on a production environment doesn't work.
>>>>>>>
>>>>>>> The problem seems to be the "sleep" called by "execute_extension"
>>>>>>> executed on answer.
>>>>>>>
>>>>>>> If I activate this extension, I see in freeswitch log:
>>>>>>>
>>>>>>> 2011-03-22 09:18:10.191304 [CRIT] switch_core_session.c:1646 Throttle
>>>>>>> Error! 1326
>>>>>>> 2011-03-22 09:18:10.193260 [CRIT] switch_core_session.c:1646 Throttle
>>>>>>> Error! 1326
>>>>>>> 2011-03-22 09:18:10.194796 [CRIT] switch_core_session.c:1646 Throttle
>>>>>>> Error! 1323
>>>>>>> 2011-03-22 09:18:10.194796 [CRIT] switch_core_session.c:1646 Throttle
>>>>>>> Error! 1323
>>>>>>> 2011-03-22 09:18:10.199238 [CRIT] switch_core_session.c:1646 Throttle
>>>>>>> Error! 1317
>>>>>>> 2011-03-22 09:18:10.204815 [CRIT] switch_core_session.c:1646 Throttle
>>>>>>> Error! 1314
>>>>>>> 2011-03-22 09:18:10.204815 [CRIT] mod_sofia.c:3896 Error Creating
>>>>>>> Session
>>>>>>> 2011-03-22 09:18:10.214916 [CRIT] switch_core_session.c:1646 Throttle
>>>>>>> Error! 1302
>>>>>>> 2011-03-22 09:18:10.214916 [CRIT] mod_sofia.c:3896 Error Creating
>>>>>>> Session
>>>>>>> *2011-03-22 09:18:10.377962 [CRIT] switch_time.c:788 Over Session
>>>>>>> Rate of 200!*
>>>>>>>
>>>>>> Notice the above line!
>>>>>>
>>>>>> You are overwhelming your system with too many calls in such a short
>>>>>> period of time. Slow things down a bit on your call generation. Also, you
>>>>>> can probably experiment with the sleep time. 500ms might be too long. Try
>>>>>> setting it to something shorter, like 100ms or so. That might help speed
>>>>>> things up a bit. Shaving 400ms off the call may not seem like a lot but if
>>>>>> you're doing hundreds of calls per second then that 400ms will add up
>>>>>> quickly.
>>>>>>
>>>>>> -MC
>>>>>>
>>>>>>
>>>>>>
>>>>>>> 2011-03-22 09:18:33.938094 [CRIT] switch_core_session.c:1646 Throttle
>>>>>>> Error! 1266
>>>>>>> 2011-03-22 09:18:33.938094 [CRIT] switch_core_session.c:1646 Throttle
>>>>>>> Error! 1266
>>>>>>> 2011-03-22 09:18:33.939077 [CRIT] switch_core_session.c:1646 Throttle
>>>>>>> Error! 1266
>>>>>>> 2011-03-22 09:18:33.939077 [CRIT] switch_core_session.c:1646 Throttle
>>>>>>> Error! 1266
>>>>>>> 2011-03-22 09:18:33.944077 [CRIT] switch_core_session.c:1646 Throttle
>>>>>>> Error! 1264
>>>>>>> 2011-03-22 09:18:33.944077 [CRIT] mod_sofia.c:3896 Error Creating
>>>>>>> Session
>>>>>>>
>>>>>>> Stephen
>>>>>>>
>>>>>>> On Tue, Mar 22, 2011 at 8:54 AM, Stephen Wilde <wstephen80 at gmail.com
>>>>>>> > wrote:
>>>>>>>
>>>>>>>> Thank you Michael, with this trick it works fine!
>>>>>>>>
>>>>>>>>
>>>>>>>> On Tue, Mar 22, 2011 at 6:33 AM, Michael Collins <
>>>>>>>> msc at freeswitch.org> wrote:
>>>>>>>>
>>>>>>>>> Add this to your dialplan...
>>>>>>>>>
>>>>>>>>> <extension name="start tone detect">
>>>>>>>>>    <condition field="destination_number"
>>>>>>>>> data="custom_start_tone_detect">
>>>>>>>>>     <action application="sleep" data="500"/>
>>>>>>>>>     <action application="tone_detect" data="MYTONE 680 ro +5000 set
>>>>>>>>> MYTONE=1"/>
>>>>>>>>>   </condition>
>>>>>>>>> </extension>
>>>>>>>>>
>>>>>>>>> ...then change your export...
>>>>>>>>> <action application="export"
>>>>>>>>> data="nolocal:execute_on_answer=execute_extension
>>>>>>>>> custom_start_tone_detect"/>
>>>>>>>>>
>>>>>>>>> Let us know if that does the trick.
>>>>>>>>>
>>>>>>>>> -MC
>>>>>>>>>
>>>>>>>>> On Mon, Mar 21, 2011 at 12:33 PM, Stephen Wilde <
>>>>>>>>> wstephen80 at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> I start the tone_detect on answer using "execute_on_answer" as:
>>>>>>>>>>
>>>>>>>>>> <action application="export"
>>>>>>>>>> data="nolocal:execute_on_answer=tone_detect MYTONE 680 ro +5000 set
>>>>>>>>>> MYTONE=1" />
>>>>>>>>>>
>>>>>>>>>> but I have the necessity to start the tone_detect for example
>>>>>>>>>> 500ms after answer: there is a way to do that in dialplan?
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From msc at freeswitch.org  Wed Mar 23 02:13:17 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Mar 2011 16:13:17 -0700
Subject: [Freeswitch-users] Delayed start of tone_detect
In-Reply-To: <AANLkTi=cjupw5BJhTKL65JNmWWYmhQHqYX+guuq=Cq1L@mail.gmail.com>
References: <AANLkTims-MfPkq_tr-JQd_ih=gPd1i8gXSFSX2tF_m8y@mail.gmail.com>
	<AANLkTi=s1iwAm9qB3=y97U8CmLs+T=pL9ZzAToE2N5Ed@mail.gmail.com>
	<AANLkTimm7+5GxsRfJNRcRcRAqmtCUBwv8UWrn4Qk2_6r@mail.gmail.com>
	<AANLkTikn1V3ypmbZW4NQ6qSgRy_7ENdOWT5eWrCHs-C0@mail.gmail.com>
	<AANLkTin8R6roSmw+3ejr8ud_VUMYEkbQ1Hz-xtmmYh54@mail.gmail.com>
	<AANLkTiny4+NvH9WFWC51VHDCYM6pB7_QrWn8_CyontFf@mail.gmail.com>
	<AANLkTimE=Ovn+HR8dnAZ4NmPydSiepS-Rb7rzkTshP7=@mail.gmail.com>
	<AANLkTinh4xNnpHK-f_eE9QzCe4Mo8QwuWfec42wDw-jN@mail.gmail.com>
	<AANLkTimEAUJnSutb8oLU_=6vkubQ6TtWEop0XS+PUk+u@mail.gmail.com>
	<AANLkTi=cjupw5BJhTKL65JNmWWYmhQHqYX+guuq=Cq1L@mail.gmail.com>
Message-ID: <AANLkTi=1-HEchno8-kEgy4yhkwGTW0eVvD7fxpRK+En+@mail.gmail.com>

On Tue, Mar 22, 2011 at 3:56 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> The channel where I'm doing tone_detect is an outbound channel that is
> bridged to the inbound channel. The playback can works?


Sure. You're just saying to send silence on the line. No biggie. You have to
have media already anyway because tone_detect doesn't work without it, so no
worries.
-MC
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From msc at freeswitch.org  Wed Mar 23 02:20:05 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Mar 2011 16:20:05 -0700
Subject: [Freeswitch-users] playAndGetDigits not working?
In-Reply-To: <AANLkTimmdVo6OcrwiZB7AnH8Lw=QseRLFowvVU4A-B_p@mail.gmail.com>
References: <AANLkTimpyZHpXOf9ZHaKCcXdZH=mo-2y2rWJLWji=Dda@mail.gmail.com>
	<AANLkTim253GfYSJaGXAX86u2_YdGzNGwxK6RGb8Z+kUh@mail.gmail.com>
	<AANLkTimmdVo6OcrwiZB7AnH8Lw=QseRLFowvVU4A-B_p@mail.gmail.com>
Message-ID: <AANLkTi=9fciWjmcvTTXXSYA8XikvhkHe2Bx4-+=A7GCQ@mail.gmail.com>

On Tue, Mar 22, 2011 at 4:07 PM, deniro <ayhkor at gmail.com> wrote:

> still same. not working
> what else  can it be?
> thx
>
> Pastebin your whole script, please and link here.
http://pastebin.freeswitch.org

-MC
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From anthony.minessale at gmail.com  Wed Mar 23 02:31:34 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 22 Mar 2011 18:31:34 -0500
Subject: [Freeswitch-users] Delayed start of tone_detect
In-Reply-To: <AANLkTi=1-HEchno8-kEgy4yhkwGTW0eVvD7fxpRK+En+@mail.gmail.com>
References: <AANLkTims-MfPkq_tr-JQd_ih=gPd1i8gXSFSX2tF_m8y@mail.gmail.com>
	<AANLkTi=s1iwAm9qB3=y97U8CmLs+T=pL9ZzAToE2N5Ed@mail.gmail.com>
	<AANLkTimm7+5GxsRfJNRcRcRAqmtCUBwv8UWrn4Qk2_6r@mail.gmail.com>
	<AANLkTikn1V3ypmbZW4NQ6qSgRy_7ENdOWT5eWrCHs-C0@mail.gmail.com>
	<AANLkTin8R6roSmw+3ejr8ud_VUMYEkbQ1Hz-xtmmYh54@mail.gmail.com>
	<AANLkTiny4+NvH9WFWC51VHDCYM6pB7_QrWn8_CyontFf@mail.gmail.com>
	<AANLkTimE=Ovn+HR8dnAZ4NmPydSiepS-Rb7rzkTshP7=@mail.gmail.com>
	<AANLkTinh4xNnpHK-f_eE9QzCe4Mo8QwuWfec42wDw-jN@mail.gmail.com>
	<AANLkTimEAUJnSutb8oLU_=6vkubQ6TtWEop0XS+PUk+u@mail.gmail.com>
	<AANLkTi=cjupw5BJhTKL65JNmWWYmhQHqYX+guuq=Cq1L@mail.gmail.com>
	<AANLkTi=1-HEchno8-kEgy4yhkwGTW0eVvD7fxpRK+En+@mail.gmail.com>
Message-ID: <AANLkTimLhQtkzBnuujjuL3nN8W4BHJk96RU4Ld5sB649@mail.gmail.com>

or set media_bug_answer_req=true globally in vars.xml or per channel
in export set or {}



On Tue, Mar 22, 2011 at 6:13 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Tue, Mar 22, 2011 at 3:56 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:
>>
>> The channel where I'm doing tone_detect is an outbound channel that is
>> bridged to the inbound channel. The playback can works?
>
> Sure. You're just saying to send silence on the line. No biggie. You have to
> have media already anyway because tone_detect doesn't work without it, so no
> worries.
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 frankie.k.yiu at gmail.com  Wed Mar 23 02:40:30 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Tue, 22 Mar 2011 16:40:30 -0700
Subject: [Freeswitch-users] Gawk.exe crashes when building on Windows 7
	64bits w/ VS 2008
Message-ID: <AANLkTikfjRpu125byJxfXAPi3vhgZ6D7wMCU9t_FUzgW@mail.gmail.com>

Jeff,

I used TortoiseGit x64 bit client to download freeSwitch with AutoCrlf
option unchecked.  I also tried using GIT to download with AutoCrlf=false
but still no luck.

I have no problem compiling it on another machine (Windows Server 2003 with
32bit), it seems like this could be the 64bits windows or windows 7 issue.
Any idea how to get pass this?

Thanks,
Frankie


> ---------- Forwarded message ----------
> From: Frankie Yiu <frankie.k.yiu at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Tue, 22 Mar 2011 13:24:08 -0700
> Subject: [Freeswitch-users] Gawk.exe crashes when building on Windows 7
> 64bits w/ VS 2008
> Hi there,
>
> I am wondering if anyone else encounters the same issue.  I am running
> Windows 7 64bits and when I build FreeSwitch in Visual Studio 2008 (SP1) for
> WIN32, it crashes/compilation error when opening a Gawk program which causes
> files can not be automatically generated.  I even installed a newer version
> 3.1.7 and it still crashes.
>
> Please let me know if you have resolved this issue.
> Thanks,
> Frankie
>
>
> ---------- Forwarded message ----------
> From: Jeff Lenk <jeff at jefflenk.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Tue, 22 Mar 2011 13:31:21 -0700 (PDT)
> Subject: Re: [Freeswitch-users] Gawk.exe crashes when building on Windows 7
> 64bits w/ VS 2008
> This is a problem with your git autocrlf. You must have that turned off.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Gawk-exe-crashes-when-building-on-Windows-7-64bits-w-VS-2008-tp6197846p6197848.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
>
>
>
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From wstephen80 at gmail.com  Wed Mar 23 03:07:18 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Wed, 23 Mar 2011 01:07:18 +0100
Subject: [Freeswitch-users] Delayed start of tone_detect
In-Reply-To: <AANLkTimLhQtkzBnuujjuL3nN8W4BHJk96RU4Ld5sB649@mail.gmail.com>
References: <AANLkTims-MfPkq_tr-JQd_ih=gPd1i8gXSFSX2tF_m8y@mail.gmail.com>
	<AANLkTi=s1iwAm9qB3=y97U8CmLs+T=pL9ZzAToE2N5Ed@mail.gmail.com>
	<AANLkTimm7+5GxsRfJNRcRcRAqmtCUBwv8UWrn4Qk2_6r@mail.gmail.com>
	<AANLkTikn1V3ypmbZW4NQ6qSgRy_7ENdOWT5eWrCHs-C0@mail.gmail.com>
	<AANLkTin8R6roSmw+3ejr8ud_VUMYEkbQ1Hz-xtmmYh54@mail.gmail.com>
	<AANLkTiny4+NvH9WFWC51VHDCYM6pB7_QrWn8_CyontFf@mail.gmail.com>
	<AANLkTimE=Ovn+HR8dnAZ4NmPydSiepS-Rb7rzkTshP7=@mail.gmail.com>
	<AANLkTinh4xNnpHK-f_eE9QzCe4Mo8QwuWfec42wDw-jN@mail.gmail.com>
	<AANLkTimEAUJnSutb8oLU_=6vkubQ6TtWEop0XS+PUk+u@mail.gmail.com>
	<AANLkTi=cjupw5BJhTKL65JNmWWYmhQHqYX+guuq=Cq1L@mail.gmail.com>
	<AANLkTi=1-HEchno8-kEgy4yhkwGTW0eVvD7fxpRK+En+@mail.gmail.com>
	<AANLkTimLhQtkzBnuujjuL3nN8W4BHJk96RU4Ld5sB649@mail.gmail.com>
Message-ID: <AANLkTimn9V3PDEUXJFdNvWcjw2bTJPTJpn7gROk12rd=@mail.gmail.com>

My tone_detect already starts after answer but I want to start it 500ms
after the answer because I want to ignore the first 500ms of media (where
the tone can be present but I'm not interested in).


On Wed, Mar 23, 2011 at 12:31 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> or set media_bug_answer_req=true globally in vars.xml or per channel
> in export set or {}
>
>
>
> On Tue, Mar 22, 2011 at 6:13 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> >
> >
> > On Tue, Mar 22, 2011 at 3:56 PM, Stephen Wilde <wstephen80 at gmail.com>
> wrote:
> >>
> >> The channel where I'm doing tone_detect is an outbound channel that is
> >> bridged to the inbound channel. The playback can works?
> >
> > Sure. You're just saying to send silence on the line. No biggie. You have
> to
> > have media already anyway because tone_detect doesn't work without it, so
> no
> > worries.
> > -MC
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jeff at jefflenk.com  Wed Mar 23 05:55:18 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 22 Mar 2011 19:55:18 -0700 (PDT)
Subject: [Freeswitch-users] Gawk.exe crashes when building on Windows 7
	64bits w/ VS 2008
In-Reply-To: <AANLkTikfjRpu125byJxfXAPi3vhgZ6D7wMCU9t_FUzgW@mail.gmail.com>
References: <AANLkTik9kcYZy84LpUNKNytu6PgA0YYPQNxgo8dSJs4m@mail.gmail.com>
	<AANLkTikfjRpu125byJxfXAPi3vhgZ6D7wMCU9t_FUzgW@mail.gmail.com>
Message-ID: <1300848918347-6198771.post@n2.nabble.com>

Hi Frankie,

Not sure what the problem is. I build all the time on Windows 7 x64 with
vs2008 and vs2010 so I know there is no problem with the solution.

Any differences between your working machine and the new one besides what
you already mentioned?

Are you only having trouble with Gawk?

-Jeff

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Gawk-exe-crashes-when-building-on-Windows-7-64bits-w-VS-2008-tp6197846p6198771.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From ayhkor at gmail.com  Wed Mar 23 06:54:28 2011
From: ayhkor at gmail.com (deniro)
Date: Tue, 22 Mar 2011 23:54:28 -0400
Subject: [Freeswitch-users] playAndGetDigits not working?
In-Reply-To: <AANLkTi=9fciWjmcvTTXXSYA8XikvhkHe2Bx4-+=A7GCQ@mail.gmail.com>
References: <AANLkTimpyZHpXOf9ZHaKCcXdZH=mo-2y2rWJLWji=Dda@mail.gmail.com>
	<AANLkTim253GfYSJaGXAX86u2_YdGzNGwxK6RGb8Z+kUh@mail.gmail.com>
	<AANLkTimmdVo6OcrwiZB7AnH8Lw=QseRLFowvVU4A-B_p@mail.gmail.com>
	<AANLkTi=9fciWjmcvTTXXSYA8XikvhkHe2Bx4-+=A7GCQ@mail.gmail.com>
Message-ID: <AANLkTinMSyNKjnuOt29FqMV07SVHBV1X2J6fR4SiH0dB@mail.gmail.com>

ok, I got that working
$session->anwer() fixed the problem

I am now able to join the conference  but in freeswitch console I am
receiving following when I joined the conf
(it is constantly flowing on the screeen)
What is this and how to fix it?
thx


2011-03-22 23:51:21.740402 [WARNING] mod_local_stream.c:393 Unknown source
moh, trying 'default'
2011-03-22 23:51:21.740402 [ERR] mod_local_stream.c:402 Unknown source
default
2011-03-22 23:51:21.760297 [WARNING] mod_local_stream.c:393 Unknown source
moh, trying 'default'
2011-03-22 23:51:21.760297 [ERR] mod_local_stream.c:402 Unknown source
default
2011-03-22 23:51:21.779849 [WARNING] mod_local_stream.c:393 Unknown source
moh, trying 'default'
2011-03-22 23:51:21.779849 [ERR] mod_local_stream.c:402 Unknown source
default




On Tue, Mar 22, 2011 at 7:20 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
>  On Tue, Mar 22, 2011 at 4:07 PM, deniro <ayhkor at gmail.com> wrote:
>
>> still same. not working
>> what else  can it be?
>> thx
>>
>> Pastebin your whole script, please and link here.
> http://pastebin.freeswitch.org
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ayhkor at gmail.com  Wed Mar 23 08:05:06 2011
From: ayhkor at gmail.com (deniro)
Date: Wed, 23 Mar 2011 01:05:06 -0400
Subject: [Freeswitch-users] playAndGetDigits not working?
In-Reply-To: <AANLkTinMSyNKjnuOt29FqMV07SVHBV1X2J6fR4SiH0dB@mail.gmail.com>
References: <AANLkTimpyZHpXOf9ZHaKCcXdZH=mo-2y2rWJLWji=Dda@mail.gmail.com>
	<AANLkTim253GfYSJaGXAX86u2_YdGzNGwxK6RGb8Z+kUh@mail.gmail.com>
	<AANLkTimmdVo6OcrwiZB7AnH8Lw=QseRLFowvVU4A-B_p@mail.gmail.com>
	<AANLkTi=9fciWjmcvTTXXSYA8XikvhkHe2Bx4-+=A7GCQ@mail.gmail.com>
	<AANLkTinMSyNKjnuOt29FqMV07SVHBV1X2J6fR4SiH0dB@mail.gmail.com>
Message-ID: <AANLkTinOz39n8Fm+sQR7W86rn_gveYb1CR3tUiNMtbwB@mail.gmail.com>

ok, got this one fixed too
installed moh files
thx



On Tue, Mar 22, 2011 at 11:54 PM, deniro <ayhkor at gmail.com> wrote:

>
>
> ok, I got that working
> $session->anwer() fixed the problem
>
> I am now able to join the conference  but in freeswitch console I am
> receiving following when I joined the conf
> (it is constantly flowing on the screeen)
> What is this and how to fix it?
> thx
>
>
> 2011-03-22 23:51:21.740402 [WARNING] mod_local_stream.c:393 Unknown source
> moh, trying 'default'
> 2011-03-22 23:51:21.740402 [ERR] mod_local_stream.c:402 Unknown source
> default
> 2011-03-22 23:51:21.760297 [WARNING] mod_local_stream.c:393 Unknown source
> moh, trying 'default'
> 2011-03-22 23:51:21.760297 [ERR] mod_local_stream.c:402 Unknown source
> default
> 2011-03-22 23:51:21.779849 [WARNING] mod_local_stream.c:393 Unknown source
> moh, trying 'default'
> 2011-03-22 23:51:21.779849 [ERR] mod_local_stream.c:402 Unknown source
> default
>
>
>
>
>  On Tue, Mar 22, 2011 at 7:20 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>>  On Tue, Mar 22, 2011 at 4:07 PM, deniro <ayhkor at gmail.com> wrote:
>>
>>> still same. not working
>>> what else  can it be?
>>> thx
>>>
>>> Pastebin your whole script, please and link here.
>> http://pastebin.freeswitch.org
>>
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From msc at freeswitch.org  Wed Mar 23 08:17:24 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Mar 2011 22:17:24 -0700
Subject: [Freeswitch-users] playAndGetDigits not working?
In-Reply-To: <AANLkTinMSyNKjnuOt29FqMV07SVHBV1X2J6fR4SiH0dB@mail.gmail.com>
References: <AANLkTimpyZHpXOf9ZHaKCcXdZH=mo-2y2rWJLWji=Dda@mail.gmail.com>
	<AANLkTim253GfYSJaGXAX86u2_YdGzNGwxK6RGb8Z+kUh@mail.gmail.com>
	<AANLkTimmdVo6OcrwiZB7AnH8Lw=QseRLFowvVU4A-B_p@mail.gmail.com>
	<AANLkTi=9fciWjmcvTTXXSYA8XikvhkHe2Bx4-+=A7GCQ@mail.gmail.com>
	<AANLkTinMSyNKjnuOt29FqMV07SVHBV1X2J6fR4SiH0dB@mail.gmail.com>
Message-ID: <AANLkTi=WrLkG+W4sYTFSBtENsQ2aiLBJMrdL-hN6PsGy@mail.gmail.com>

Confirm that you have all of the moh files installed for all sampling rates:
"make cd-moh-install"

-MC

On Tue, Mar 22, 2011 at 8:54 PM, deniro <ayhkor at gmail.com> wrote:

>
>
> ok, I got that working
> $session->anwer() fixed the problem
>
> I am now able to join the conference  but in freeswitch console I am
> receiving following when I joined the conf
> (it is constantly flowing on the screeen)
> What is this and how to fix it?
> thx
>
>
> 2011-03-22 23:51:21.740402 [WARNING] mod_local_stream.c:393 Unknown source
> moh, trying 'default'
> 2011-03-22 23:51:21.740402 [ERR] mod_local_stream.c:402 Unknown source
> default
> 2011-03-22 23:51:21.760297 [WARNING] mod_local_stream.c:393 Unknown source
> moh, trying 'default'
> 2011-03-22 23:51:21.760297 [ERR] mod_local_stream.c:402 Unknown source
> default
> 2011-03-22 23:51:21.779849 [WARNING] mod_local_stream.c:393 Unknown source
> moh, trying 'default'
> 2011-03-22 23:51:21.779849 [ERR] mod_local_stream.c:402 Unknown source
> default
>
>
>
>
> On Tue, Mar 22, 2011 at 7:20 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>>  On Tue, Mar 22, 2011 at 4:07 PM, deniro <ayhkor at gmail.com> wrote:
>>
>>> still same. not working
>>> what else  can it be?
>>> thx
>>>
>>> Pastebin your whole script, please and link here.
>> http://pastebin.freeswitch.org
>>
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kris at livecall.com  Wed Mar 23 10:26:17 2011
From: kris at livecall.com (Kris)
Date: Wed, 23 Mar 2011 00:26:17 -0700
Subject: [Freeswitch-users] Voicemail on GlusterFS
References: <AANLkTimkv66jjaBTHtT+nDUgo3PsuxEndEi-Z9=Lb0u_@mail.gmail.com>
	<AANLkTi=44afGJ1fni5P+q=0ct=irmVDwOfESUAbdWuo9@mail.gmail.com>
Message-ID: <63BF7F7A0106425EA39F9632F5D5A292@stor1>

I have noticed that FS has a bug in playback and record APIs where it will 
fail if there are only 2 slashes in front of the server.
This is what I have to do to get it to work- add a third slash.  Something 
to try...hope it helps

string filename = VoiceMessage.ConferenceRecordingFilename(ForumExt, 
ConferenceUUID);

if (filename.StartsWith("\\\\"))//server path

filename = "\\" + filename;//fs expects 3 slashes for some reason..while 
dealing with them only 1 is left

string msg=fsApi.Execute("conference", ConferenceName + " record " + 
filename);



Kris

----- Original Message ----- 
From: "Dan Lane" <null at invalid.name>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, March 22, 2011 10:42 AM
Subject: Re: [Freeswitch-users] Voicemail on GlusterFS


On Tue, Mar 22, 2011 at 3:55 PM, Moe Navid <manavid at gmail.com> wrote:
> Hi All,
> Has anyone tried using voicemail on GlusterFS?
> GlusteFS looks very promising for building distributed file systems. 
> Pandora
> Radio recently started using
> glusterfs 
> http://www.gluster.com/2011/01/05/gluster-to-help-manage-rapid-data-growth-for-pandora/
> I managed to get the recording right for voicemails but retrieving the WAV
> files fails. FreeSWITCH gives the following error:
> 2011-03-22 08:54:06.325001 [ERR] mod_sndfile.c:194 Error Opening File
> [/home/moe/freeswitch/storage/voicemail/default/192.168.1.10/1001/msg_02a108cc-549c-11e0-a9f6-9f1a6f4da0f7.wav]
> [File contains data in an unknown format.]
> Any thoughts?
> Thanks
> Moe

We've been using GlusterFS for storing voicemail in a production
environment for over a year now but we built a custom voicemail
solution rather than using the built-in one.

We didn't experience the issue you're referring to so I'm afraid I
can't offer any assistance.





From kbdfck at gmail.com  Wed Mar 23 10:51:22 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Wed, 23 Mar 2011 10:51:22 +0300
Subject: [Freeswitch-users] Event-lock,
	sync/async on outbound ESL and execute() tagging question
Message-ID: <AANLkTingnGhvshmDK3TxKSkTpFeno76KTrFULRhbSCiP@mail.gmail.com>

Hi All

I'm trying to understand setEventLock and setAsyncExecute parameters on
outbound ESL socket. My aim is to have ability to choose execution method of
freeswitch dialplan command via execute(). I'm using async full, but I want
to use sync mode for some applications for which I don't want to receive
events, for example. Is there a way to run application in sync mode and just
get return value, and discard its events?

I want this because If later I will run some app in async mode, I'll get
previous events like 'CHANNEL_EXECUTE_COMPLETE' and so on. Another approach
is to have ability to mark calls to execute or API with some tag to
distinguish between events from previous execute() or api() calls, like with
BGAPI job id. What can be done to achieve this?

Or maybe I should just process all events by myself if I use async?

And finally, could anybody explain the Event-Lock / AsyncExecute difference?
Seems I can't really understand the wiki notes about that.

-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From frankie.k.yiu at gmail.com  Wed Mar 23 13:02:23 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Wed, 23 Mar 2011 03:02:23 -0700
Subject: [Freeswitch-users] Gawk.exe crashes when building on Windows 7
 64bits w/ VS 2008
Message-ID: <AANLkTimpmOAZweiffzr6D3uxd1GSS-GO9wfvPSr1qUyQ@mail.gmail.com>

>
> Hi Jeff,
>

Because Gawk is not running properly and therefore I have more than 100
errors complaining missing files.
I don't know what other differences could there be besides the Windows 7 x64
and the Windows 2003 server x86.

Thanks,
Frankie

>
>
> ---------- Forwarded message ----------
> From: Frankie Yiu <frankie.k.yiu at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Tue, 22 Mar 2011 16:40:30 -0700
> Subject: Re: [Freeswitch-users] Gawk.exe crashes when building on Windows 7
> 64bits w/ VS 2008
> Jeff,
>
> I used TortoiseGit x64 bit client to download freeSwitch with AutoCrlf
> option unchecked.  I also tried using GIT to download with AutoCrlf=false
> but still no luck.
>
> I have no problem compiling it on another machine (Windows Server 2003 with
> 32bit), it seems like this could be the 64bits windows or windows 7 issue.
> Any idea how to get pass this?
>
> Thanks,
> Frankie
>
>
>> ---------- Forwarded message ----------
>> From: Frankie Yiu <frankie.k.yiu at gmail.com>
>> To: freeswitch-users at lists.freeswitch.org
>> Date: Tue, 22 Mar 2011 13:24:08 -0700
>> Subject: [Freeswitch-users] Gawk.exe crashes when building on Windows 7
>> 64bits w/ VS 2008
>> Hi there,
>>
>> I am wondering if anyone else encounters the same issue.  I am running
>> Windows 7 64bits and when I build FreeSwitch in Visual Studio 2008 (SP1) for
>> WIN32, it crashes/compilation error when opening a Gawk program which causes
>> files can not be automatically generated.  I even installed a newer version
>> 3.1.7 and it still crashes.
>>
>> Please let me know if you have resolved this issue.
>> Thanks,
>> Frankie
>>
>>
>> ---------- Forwarded message ----------
>> From: Jeff Lenk <jeff at jefflenk.com>
>> To: freeswitch-users at lists.freeswitch.org
>> Date: Tue, 22 Mar 2011 13:31:21 -0700 (PDT)
>> Subject: Re: [Freeswitch-users] Gawk.exe crashes when building on Windows
>> 7 64bits w/ VS 2008
>> This is a problem with your git autocrlf. You must have that turned off.
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Gawk-exe-crashes-when-building-on-Windows-7-64bits-w-VS-2008-tp6197846p6197848.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>>
>>
>>
>>
>
> ---------- Forwarded message ----------
> From: Jeff Lenk <jeff at jefflenk.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Tue, 22 Mar 2011 19:55:18 -0700 (PDT)
> Subject: Re: [Freeswitch-users] Gawk.exe crashes when building on Windows 7
> 64bits w/ VS 2008
> Hi Frankie,
>
> Not sure what the problem is. I build all the time on Windows 7 x64 with
> vs2008 and vs2010 so I know there is no problem with the solution.
>
> Any differences between your working machine and the new one besides what
> you already mentioned?
>
> Are you only having trouble with Gawk?
>
> -Jeff
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Gawk-exe-crashes-when-building-on-Windows-7-64bits-w-VS-2008-tp6197846p6198771.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
>
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From all.eforums at gmail.com  Wed Mar 23 13:44:21 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Wed, 23 Mar 2011 06:44:21 -0400
Subject: [Freeswitch-users] Bug ID: FS-2868 Status?
Message-ID: <AANLkTikTZbhXT-99gwFK6fA=opMQyajR4A5smzVbM6bc@mail.gmail.com>

Hello FreeSwitchers!

Am attempting to compile the latest (off of Git) code for FS on Solaris 11
Express on an 64-Bit x86 machine. However, I'm running into the error that
Bruce McAllister has reported on 23rd Nov 2010. The bug is still sitting
unresolved. So I'm assuming there's a backdoor trick to fixing this OR that
some other version of FS code doesn't have this issue. Could someone please
let me on the secret on which build compiles on x86 Solaris v10+ and/or what
the backdoor trick to bypass this is? Should I not compile the hash module
where this bug seems to be originating from?

Your help is appreciated in advance,
Cheers,
\AEG
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From harishjprabhu at gmail.com  Wed Mar 23 09:54:34 2011
From: harishjprabhu at gmail.com (Harish Prabhu)
Date: Wed, 23 Mar 2011 12:24:34 +0530
Subject: [Freeswitch-users] Cross-compiling Freeswitch
Message-ID: <AANLkTine3CtFNGPbkZPhCYNqXAT-je65Qf7Yd5_EQibr@mail.gmail.com>

Hi All,

I need to install Freeswitch on an embedded system.
I use an Ubuntu machine for compilation.
The target binaries/shared libs have to be generated in a /tmp folder, but
in such a way that it would run at /usr/local/freeswitch on the target.
Is it possible to do this without any major changes to Freeswitch build
environment ?

I tried using --prefix=/tmp/abc and --exec-prefix=/usr/local/freeswitch as
options to configure but this does not work.
If I use /tmp/abc for both prefix and exec-prefix, I am forced to run
freeswitch from the /tmp/abc folder on the embedded system.

Regards,
Harish
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From egbert at redhosting.nl  Wed Mar 23 15:03:27 2011
From: egbert at redhosting.nl ([Redhosting] Egbert Groot)
Date: Wed, 23 Mar 2011 13:03:27 +0100
Subject: [Freeswitch-users] Understanding break="never" in condition-tag
In-Reply-To: <AANLkTinR9CXqeOXwMu37TnDggb4ffuAd0yGOkAJDr7Mo@mail.gmail.com>
References: <4D85E0A1.507@redhosting.nl>	<AANLkTimNYfHOrZrr7Q70Zea8Nsw-xEmQr2z0g-=PK+hv@mail.gmail.com>	<4D85F6FB.8080801@redhosting.nl>	<AANLkTimXv0PrtnjRPLfi5WFGhpyNmgc4H-Unvr7Bc7Xj@mail.gmail.com>	<AANLkTimi5pAW-kJif8TVYXKugjbaK18J6hiBtJ712_q4@mail.gmail.com>	<AANLkTikxwL9EHQ984242msZ4kG0wD9pDe8o6AYyOuGcf@mail.gmail.com>	<AANLkTikc0a4ktzy3tNZHNbpNibSAChY5H3avumNSgaOT@mail.gmail.com>	<AANLkTikbejGE2ku-X8VYjc=HctHCuSDDozUdusaKDFr4@mail.gmail.com>	<AANLkTinty+efpatHL7eu79H5SuAjYdfVxA6o_O_A3rC+@mail.gmail.com>	<AANLkTi=4Sqb-m+t-o81OYJwK=UTD17-fkcHtG7oe0LDN@mail.gmail.com>	<AANLkTinJFMMMBVrLPsAeuKOApECywnCBkTgkMbnjTFzW@mail.gmail.com>	<AANLkTikUn_Mdux6XQWx5ouQwEBc7j8uAEPricD==CM4T@mail.gmail.com>	<4D87637F.3090309@redhosting.nl>
	<AANLkTinR9CXqeOXwMu37TnDggb4ffuAd0yGOkAJDr7Mo@mail.gmail.com>
Message-ID: <4D89E18F.70204@redhosting.nl>

Michael Collins schreef:
>  
>
>     I can't/didn't contribute much to this discussion, but I do follow
>     it. My idea and (little) testing experience supports the
>     explanation of Dimitry.
>     Either way, it 'proofs' there is confusion about the working of
>     the 'break' in condition tags.
>     If I can find the time, I will try to do some tests / build an
>     example myself. Thank you both for the answers and insights so far.
>
>
> Okay, for those of you who have the bridge book, please go to page 
> 167-168. Darren did a really good job of explaining this. For those of 
> you who don't have the book, SHAME! :)
>

Thanks for the explanation, and I just ordered the book :)

regards,
Egbert.

> The break flag determines what happens on processing conditions within 
> a single extension. Let's say that chan var ${freeswitch} has the 
> value "rocks" and ${foo} has the value "bar". 
>
> <extension name="break_flag_is_not_confusing">
>   <condition "${freeswitch}" expression="^lame$">
>     <!-- this does not get executed -->
>   </condition>
>   <!-- the following condition is not even evaluated -->
>   <condition "${foo}" expression="bar">
>   ...
>   </condition>
> </extension> 
>
> Look at the above snipped. The first condition evaluates to false, 
> which means all processing for this particular extension stops. The 
> parse "breaks" out of this extension and moves on to the next one. Now 
> look at this extension with the the break-flag set to "never"...
>
> <extension name="break_flag_is_not_confusing">
>   <condition "${freeswitch}" expression="^lame$" break="never">
>     <!-- this does not get executed -->
>   </condition>
>   <!-- the following condition *is* evaluated -->
>   <condition "${foo}" expression="bar">
>   ...
>   </condition>
> </extension> 
>
> See the difference? With the break flag set to "never" then it does 
> not matter whether the condition is evaluated as true or false - the 
> extension's next condition gets evaluated. Why do you need the 
> break="never"? Simple: when no "break" is specified there is an 
> implied break="on-false" for the condition. In other words, by 
> default, if one condition inside of an extension evaluates to false 
> then all processing for that extension stops. This has the effect of 
> allowing you to "stack" conditions to create a logical AND. 
>
> So, in summary, the break flag will control how the parser behaves 
> when evaluating conditions. Normally when a single condition fails, 
> the whole extension is skipped, however when you do break="never" on a 
> condition then it does not matter if that particular condition is true 
> or false - extension processing will continue.
>
> Hope this helps. If it doesn't then read chapters 5 and 8 of the book. 
> Chapter 5 (me) is a gentle intro into the dialplan. Chapter 8 (Darren) 
> talks more extensively about the advanced concepts of dialplan 
> processing. 
>
> -MC
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From michal.bielicki at seventhsignal.de  Wed Mar 23 16:15:58 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Wed, 23 Mar 2011 14:15:58 +0100
Subject: [Freeswitch-users] Bug ID: FS-2868 Status?
In-Reply-To: <AANLkTikTZbhXT-99gwFK6fA=opMQyajR4A5smzVbM6bc@mail.gmail.com>
References: <AANLkTikTZbhXT-99gwFK6fA=opMQyajR4A5smzVbM6bc@mail.gmail.com>
Message-ID: <616EFD97-246A-46EE-9D7C-AE2F2DD2BF7C@seventhsignal.de>

The problem is a bug in solaris perl development package that I haven't been doing anything about since I do not use perl for roughly a decade.
Its not the hash module, its the perl module.

Am 23.03.2011 um 11:44 schrieb A E [Gmail]:

> Hello FreeSwitchers!
> 
> Am attempting to compile the latest (off of Git) code for FS on Solaris 11 Express on an 64-Bit x86 machine. However, I'm running into the error that Bruce McAllister has reported on 23rd Nov 2010. The bug is still sitting unresolved. So I'm assuming there's a backdoor trick to fixing this OR that some other version of FS code doesn't have this issue. Could someone please let me on the secret on which build compiles on x86 Solaris v10+ and/or what the backdoor trick to bypass this is? Should I not compile the hash module where this bug seems to be originating from?
> 
> Your help is appreciated in advance,
> Cheers,
> \AEG
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de

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From chistyakov at directtel.ru  Wed Mar 23 16:21:38 2011
From: chistyakov at directtel.ru (=?UTF-8?B?0KfQuNGB0YLRj9C60L7QsiDQmNCy0LDQvQ==?=)
Date: Wed, 23 Mar 2011 16:21:38 +0300
Subject: [Freeswitch-users] Can I create an EventConsumer only for specified
	uuid?
Message-ID: <4D89F3E2.40308@directtel.ru>

Can I create an EventConsumer only for specified uuid?


From peter.olsson at visionutveckling.se  Wed Mar 23 16:32:59 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 23 Mar 2011 14:32:59 +0100
Subject: [Freeswitch-users] Can I create an EventConsumer only for
 specified	uuid?
In-Reply-To: <4D89F3E2.40308@directtel.ru>
References: <4D89F3E2.40308@directtel.ru>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B30F0C28@cooper>

If you add a filter that should work just fine.

Mvh
Peter Olsson

-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r ???????? ????
Skickat: den 23 mars 2011 14:22
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] Can I create an EventConsumer only for specified uuid?

Can I create an EventConsumer only for specified uuid?

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

!DSPAM:4d89f56632761192711445!



From gmaruzz at gmail.com  Wed Mar 23 16:43:40 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 23 Mar 2011 14:43:40 +0100
Subject: [Freeswitch-users] use of mobile phone with gsmopen question
In-Reply-To: <SNT111-W562AD83EE8A8BBDE68B8B5EEB40@phx.gbl>
References: <SNT111-W427F28F80543FB54AB6AF4EEB40@phx.gbl>
	<SNT111-W562AD83EE8A8BBDE68B8B5EEB40@phx.gbl>
Message-ID: <AANLkTinFZsqxZwc9iknCXv=oFVWXeG3_P25JgJVNY-z4@mail.gmail.com>

heh,

first: you have to be sure it is recognized by your system, eg: that
Linux recognize it and can use it as a modem.
Browse google for your phone on linux.

second: when and if your phone can be used by linux as a modem, linux
will automatically create a /dev/ttyACM? or a /dev/ttyUSB?

third: at that point, you just put that device in the config file of gsmopen

fourth: you will be able to send and receive sms, hopefully, at this point

fifth: to be able to make and receive voice calls, you'll need to
create an audio cable that goes from the headset jack in the cellphone
to the soundcard mic/speaker

If you want to use a device that is ready made, tested, and do not
needs any additional cables (it conyains in itself the soundcard, and
is connected to the pc with just one normal usb cable for both modem
and voice), check out www.mobigater.com

-giovanni

On Tue, Mar 22, 2011 at 1:09 PM, Ghazanfar Saeed
<m_g_saeed at hotmail.co.uk> wrote:
>
> Hi,
>
> My system is Dell XPS L501 and I have installed Ubuntu 10.10 Desktop version
> on it. The bit I am trying to achieve is to make call to gsm mobile from
> ubuntu using my gsm mobile (HTC HD2 with T-Mobile-UK SIM card) attached to
> my Ubuntu 10.10 system via USB. I have followed the below process:
>
> I have installed FreeSwitch and verified its functionality with Twinkle
> (able to access the predefined basic IVR).
> command: sudo /usr/local/freeswitch/bin/freeswitch -waste
>
> I have then built/installed the gsmopen as described on
> http://wiki.freeswitch.org/wiki/GSMopen and loaded module.
>
> freeswitch at saeed-XPS-L501X> load mod_gsmopen
> 2011-03-22 07:41:47.623806 [WARNING] mod_gsmopen.cpp:1861 rev
> exported[(nil)|37???? ][WARNINGA? 1861 ][interface1][-1, 0, 0] STARTING
> interface_id=1
> open error : No such file or directory
> 2011-03-22 07:41:48.622951 [ERR] mod_gsmopen.cpp:1892 rev
> exported[(nil)|37???? ][ERRORA? 1892 ][interface1][-1, 0, 0]
> gsmopen_serial_init failed
> 2011-03-22 07:41:48.622951 [ERR] mod_gsmopen.cpp:1893 rev
> exported[(nil)|37???? ][ERRORA? 1893 ][interface1][-1, 0, 0] STARTING
> interface_id=1 FAILED
> 2011-03-22 07:41:48.622951 [ERR] mod_gsmopen.cpp:3136 rev
> exported[(nil)|37???? ][ERRORA? 3136 ][interface1][-1, 0, 0] ALARM on
> interface interface1:
> 2011-03-22 07:41:48.622951 [CONSOLE] switch_loadable_module.c:900
> Successfully Loaded [mod_gsmopen]
> 2011-03-22 07:41:48.622951 [NOTICE] switch_loadable_module.c:144 Adding
> Endpoint 'gsmopen'
> 2011-03-22 07:41:48.622951 [NOTICE] switch_loadable_module.c:272 Adding API
> Function 'gsm'
> 2011-03-22 07:41:48.622951 [NOTICE] switch_loadable_module.c:272 Adding API
> Function 'gsmopen'
> 2011-03-22 07:41:48.624943 [NOTICE] switch_loadable_module.c:272 Adding API
> Function 'gsmopen_boost_audio'
> 2011-03-22 07:41:48.624943 [NOTICE] switch_loadable_module.c:272 Adding API
> Function 'gsmopen_dump'
> 2011-03-22 07:41:48.624943 [NOTICE] switch_loadable_module.c:272 Adding API
> Function 'gsmopen_sendsms'
> 2011-03-22 07:41:48.624943 [NOTICE] switch_loadable_module.c:377 Adding Chat
> interface 'SMS'
>
> The interface1 is the default interface (from gspopen.conf.xml)
>
> ??? <interface id="1" name="interface1">
> ??? <param name="hold-music" value="$${moh_uri}"/>
> ??? <param name="dialplan" value="XML"/>
> ??? <param name="context" value="default"/>
> ??? ??? <param name="destination" value="5000"/>
> ??? ??? <param name="alsacname" value="plughw:1"/>
> ??? ??? <param name="alsapname" value="plughw:1"/>
> ??? ??? <param name="controldevice_name" value="/dev/ttyACM0"/>
> ??? </interface>
>
>
> I am new user to this system (freeswitch+gsmopen) so please can someone put
> me in the right direction. These errors are probably due to the fact that
> there is no /dev/ttyACM0 (as I couldn't list this device). Now as a
> workaround I was thinking if I can find where my HTC HD2 phone got connected
> e.g. ttyUSB0 etc then I can create a soft link and can proceed but the
> trouble is that I can't find the which device it is connected to though i
> can see the phone via lsusb (it is on Bus 003 and device 005)
>
> Bus 003 Device 005: ID 0bb4:0b40 High Tech Computer Corp.
> Bus 003 Device 004: ID 04f3:0212 Elan Microelectronics Corp. Laser Mouse
> Bus 003 Device 003: ID 0603:00f2 Novatek Microelectronics Corp.
>
> Further information is I was able to use internet out of the box when phone
> is connected (and I opted for the option internet sharing on my phone).
>
> Any recommendation how should I proceed further on this will be very
> helpful. Please let me know if I need to provide any further information.
>
> Many thanks.
>
> Kind regards,
>
>
>
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From lakindia89 at gmail.com  Wed Mar 23 16:48:24 2011
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Wed, 23 Mar 2011 19:18:24 +0530
Subject: [Freeswitch-users] Event-lock,
 sync/async on outbound ESL and execute() tagging question
In-Reply-To: <AANLkTingnGhvshmDK3TxKSkTpFeno76KTrFULRhbSCiP@mail.gmail.com>
References: <AANLkTingnGhvshmDK3TxKSkTpFeno76KTrFULRhbSCiP@mail.gmail.com>
Message-ID: <AANLkTim9ftv7Q40i2ZhkfH6U8Qk+W5-NjziNt+B6anXW@mail.gmail.com>

Hi,
Eventlock is used to say "Whether to queue the commands or not".

Assume I'm giving playback and bridge command via outbound socket to
freeswitch.

If eventlock is "true" then, first "playback will be executed" and then only
the bridge will be executed.

If eventlock is "false", then order of execution is not guaranteed.

I'm also using Outbound event socket. I'll run it in async mode. After each
execute() statement, if I need the output for that event, I will wait for
the event and then only I'll proceed. If I don't need event, ( playback ), I
just do the other operations.

This works well for me.

On Wed, Mar 23, 2011 at 1:21 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:

> Hi All
>
> I'm trying to understand setEventLock and setAsyncExecute parameters on
> outbound ESL socket. My aim is to have ability to choose execution method of
> freeswitch dialplan command via execute(). I'm using async full, but I want
> to use sync mode for some applications for which I don't want to receive
> events, for example. Is there a way to run application in sync mode and just
> get return value, and discard its events?
>
> I want this because If later I will run some app in async mode, I'll get
> previous events like 'CHANNEL_EXECUTE_COMPLETE' and so on. Another approach
> is to have ability to mark calls to execute or API with some tag to
> distinguish between events from previous execute() or api() calls, like with
> BGAPI job id. What can be done to achieve this?
>
> Or maybe I should just process all events by myself if I use async?
>
> And finally, could anybody explain the Event-Lock / AsyncExecute
> difference? Seems I can't really understand the wiki notes about that.
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From michal.bielicki at seventhsignal.de  Wed Mar 23 16:53:14 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Wed, 23 Mar 2011 14:53:14 +0100
Subject: [Freeswitch-users] Bug ID: FS-2868 Status?
In-Reply-To: <AANLkTikTZbhXT-99gwFK6fA=opMQyajR4A5smzVbM6bc@mail.gmail.com>
References: <AANLkTikTZbhXT-99gwFK6fA=opMQyajR4A5smzVbM6bc@mail.gmail.com>
Message-ID: <639D8939-0972-4496-93DF-AB1968F3401E@seventhsignal.de>

You can see the results of my builds here: http://www.freeswitch.de:8080/ for both openindiana and solaris express 1

Am 23.03.2011 um 11:44 schrieb A E [Gmail]:

> Hello FreeSwitchers!
> 
> Am attempting to compile the latest (off of Git) code for FS on Solaris 11 Express on an 64-Bit x86 machine. However, I'm running into the error that Bruce McAllister has reported on 23rd Nov 2010. The bug is still sitting unresolved. So I'm assuming there's a backdoor trick to fixing this OR that some other version of FS code doesn't have this issue. Could someone please let me on the secret on which build compiles on x86 Solaris v10+ and/or what the backdoor trick to bypass this is? Should I not compile the hash module where this bug seems to be originating from?
> 
> Your help is appreciated in advance,
> Cheers,
> \AEG
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de

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From chistyakov at directtel.ru  Wed Mar 23 16:52:33 2011
From: chistyakov at directtel.ru (=?UTF-8?B?0KfQuNGB0YLRj9C60L7QsiDQmNCy0LDQvQ==?=)
Date: Wed, 23 Mar 2011 16:52:33 +0300
Subject: [Freeswitch-users] Can I create an EventConsumer only for
 specified uuid?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B30F0C28@cooper>
References: <4D89F3E2.40308@directtel.ru>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B30F0C28@cooper>
Message-ID: <4D89FB21.4080109@directtel.ru>

How can I add it?

23.03.2011 16:32, Peter Olsson ?????:
> If you add a filter that should work just fine.
>
> Mvh
> Peter Olsson
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r ???????? ????
> Skickat: den 23 mars 2011 14:22
> Till: FreeSWITCH Users Help
> ?mne: [Freeswitch-users] Can I create an EventConsumer only for specified uuid?
>
> Can I create an EventConsumer only for specified uuid?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4d89f56632761192711445!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From fieldpeak at gmail.com  Wed Mar 23 17:26:18 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Wed, 23 Mar 2011 22:26:18 +0800
Subject: [Freeswitch-users] How to disable the 407 challenge for the call
	from registerred users
Message-ID: <AANLkTi=gUTuTgaptYAa2AQOP4o8J6M0kscPJZ8-qpvxE@mail.gmail.com>

Is there any one can help how to configure the FreeSwitch to disable the 407
challenge for the call from registerred users?

P.S. the register user maybe use dynamic IP address or IP segment, so it
could not directly use *apply*-*inbound*-*acl* in profile.

Thanks.
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From jeff at jefflenk.com  Wed Mar 23 17:40:02 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 23 Mar 2011 07:40:02 -0700 (PDT)
Subject: [Freeswitch-users] Gawk.exe crashes when building on Windows 7
	64bits w/ VS 2008
In-Reply-To: <AANLkTimpmOAZweiffzr6D3uxd1GSS-GO9wfvPSr1qUyQ@mail.gmail.com>
References: <AANLkTik9kcYZy84LpUNKNytu6PgA0YYPQNxgo8dSJs4m@mail.gmail.com>
	<AANLkTimpmOAZweiffzr6D3uxd1GSS-GO9wfvPSr1qUyQ@mail.gmail.com>
Message-ID: <1300891202380-6200499.post@n2.nabble.com>

I just did a fresh clone and built with vs2008(x64 debug build) on windows 7
x64 without any problems.
Maybe if you post a Pastebin of your build log something will help.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Gawk-exe-crashes-when-building-on-Windows-7-64bits-w-VS-2008-tp6197846p6200499.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From fieldpeak at gmail.com  Wed Mar 23 17:38:37 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Wed, 23 Mar 2011 22:38:37 +0800
Subject: [Freeswitch-users] sofia_reg_internal.db grow bigger and bigger
	along with time
Message-ID: <AANLkTi=FVvghm6H=BiV8jT+mYqE9SVtfNa0zst+BqFvF@mail.gmail.com>

i found sofia_reg_internal.db grow bigger and bigger after FreeSwitch for
some time.
the change comes from the sip-authentication table inside this db, it always
create new record for nonce for the new challenge ('407 Proxy Authentication
Required') for the call from registered users.
any one can help how to avoid the db grow, i'm afaid the hard disk will be
full along with time...

Thanks!
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From peter.olsson at visionutveckling.se  Wed Mar 23 18:00:00 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 23 Mar 2011 16:00:00 +0100
Subject: [Freeswitch-users] Gawk.exe crashes when building on Windows
	7	64bits w/ VS 2008
In-Reply-To: <1300891202380-6200499.post@n2.nabble.com>
References: <AANLkTik9kcYZy84LpUNKNytu6PgA0YYPQNxgo8dSJs4m@mail.gmail.com>
	<AANLkTimpmOAZweiffzr6D3uxd1GSS-GO9wfvPSr1qUyQ@mail.gmail.com>
	<1300891202380-6200499.post@n2.nabble.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B30F0C8E@cooper>

Also, check UAC related stuff - it wouldn't be the first time it caused problems :)

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Jeff Lenk
Skickat: den 23 mars 2011 15:40
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Gawk.exe crashes when building on Windows 7 64bits w/ VS 2008

I just did a fresh clone and built with vs2008(x64 debug build) on windows 7
x64 without any problems.
Maybe if you post a Pastebin of your build log something will help.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Gawk-exe-crashes-when-building-on-Windows-7-64bits-w-VS-2008-tp6197846p6200499.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

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

!DSPAM:4d8a074832761734911477!



From msc at freeswitch.org  Wed Mar 23 19:27:54 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 23 Mar 2011 09:27:54 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <AANLkTin7QFGA3aOCaBFeWKXY_tp5MWZ9eTnEC9=N8Xqp@mail.gmail.com>

Hey all,

Here's today's agenda:
http://wiki.freeswitch.org/wiki/FS_weekly_2011_03_23

We have a few items to discuss today, including the highlights from
February's git commits. Also, we are going to have a discussion on some
troubleshooting and data collection techniques.

Talk to you soon!

-MC
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From gustavo.espeche at easyipcall.com  Wed Mar 23 16:39:01 2011
From: gustavo.espeche at easyipcall.com (Gustavo Espeche)
Date: Wed, 23 Mar 2011 10:39:01 -0300
Subject: [Freeswitch-users] Diversion Header
Message-ID: <1300887541.2279.0.camel@gustavo-laptop>

i'm attaching a tcpdump capture.
Thanks.

Gustavo Espeche
http://www.easyipcall.com



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From all.eforums at gmail.com  Wed Mar 23 17:38:15 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Wed, 23 Mar 2011 10:38:15 -0400
Subject: [Freeswitch-users] Bug ID: FS-2868 Status?
In-Reply-To: <639D8939-0972-4496-93DF-AB1968F3401E@seventhsignal.de>
References: <AANLkTikTZbhXT-99gwFK6fA=opMQyajR4A5smzVbM6bc@mail.gmail.com>
	<639D8939-0972-4496-93DF-AB1968F3401E@seventhsignal.de>
Message-ID: <AANLkTiknJZXQbndprQcc2ybdiESzWonee201iTyFvDrP@mail.gmail.com>

On Wed, Mar 23, 2011 at 9:53 AM, Michal Bielicki <
michal.bielicki at seventhsignal.de> wrote:

> You can see the results of my builds here: http://www.freeswitch.de:8080/ for
> both openindiana and solaris express 1
>
>
> Thanks Michael. I noticed that you aren't making mod_hash which might be
how you're getting through the entire build? I know you said it's not
mod-hash but the PERL module but I guess it's getting encountered/used in
the compilation of mod_hash?

BTW, I am also getting these WARNINGs for all sorts of "op", <<, = etc.

*warning: integer overflow detected: op "<<" (E_INTEGER_OVERFLOW_DETECTED)*

But I noticed that you don't have them. Did you ever get them? and if you
did then how did you solve them?

Thanks
\AEG
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From peter.olsson at visionutveckling.se  Wed Mar 23 19:39:07 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 23 Mar 2011 17:39:07 +0100
Subject: [Freeswitch-users] Can I create an EventConsumer only for
 specified uuid?
In-Reply-To: <4D89FB21.4080109@directtel.ru>
References: <4D89F3E2.40308@directtel.ru>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B30F0C28@cooper>
	<4D89FB21.4080109@directtel.ru>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B30F0CF5@cooper>

Send a command like this on the event socket.

Also, here you have some more info: http://wiki.freeswitch.org/wiki/Mod_event_socket

filter Unique-ID YOUR_UUID

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r ???????? ????
Skickat: den 23 mars 2011 14:53
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Can I create an EventConsumer only for specified uuid?

How can I add it?

23.03.2011 16:32, Peter Olsson ?????:
> If you add a filter that should work just fine.
>
> Mvh
> Peter Olsson
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r ???????? ????
> Skickat: den 23 mars 2011 14:22
> Till: FreeSWITCH Users Help
> ?mne: [Freeswitch-users] Can I create an EventConsumer only for specified uuid?
>
> Can I create an EventConsumer only for specified uuid?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> 
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

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

!DSPAM:4d89fbf532761360168620!


From msc at freeswitch.org  Wed Mar 23 19:41:07 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 23 Mar 2011 09:41:07 -0700
Subject: [Freeswitch-users] Understanding break="never" in condition-tag
In-Reply-To: <4D89E18F.70204@redhosting.nl>
References: <4D85E0A1.507@redhosting.nl>
	<AANLkTimNYfHOrZrr7Q70Zea8Nsw-xEmQr2z0g-=PK+hv@mail.gmail.com>
	<4D85F6FB.8080801@redhosting.nl>
	<AANLkTimXv0PrtnjRPLfi5WFGhpyNmgc4H-Unvr7Bc7Xj@mail.gmail.com>
	<AANLkTimi5pAW-kJif8TVYXKugjbaK18J6hiBtJ712_q4@mail.gmail.com>
	<AANLkTikxwL9EHQ984242msZ4kG0wD9pDe8o6AYyOuGcf@mail.gmail.com>
	<AANLkTikc0a4ktzy3tNZHNbpNibSAChY5H3avumNSgaOT@mail.gmail.com>
	<AANLkTikbejGE2ku-X8VYjc=HctHCuSDDozUdusaKDFr4@mail.gmail.com>
	<AANLkTinty+efpatHL7eu79H5SuAjYdfVxA6o_O_A3rC+@mail.gmail.com>
	<AANLkTi=4Sqb-m+t-o81OYJwK=UTD17-fkcHtG7oe0LDN@mail.gmail.com>
	<AANLkTinJFMMMBVrLPsAeuKOApECywnCBkTgkMbnjTFzW@mail.gmail.com>
	<AANLkTikUn_Mdux6XQWx5ouQwEBc7j8uAEPricD==CM4T@mail.gmail.com>
	<4D87637F.3090309@redhosting.nl>
	<AANLkTinR9CXqeOXwMu37TnDggb4ffuAd0yGOkAJDr7Mo@mail.gmail.com>
	<4D89E18F.70204@redhosting.nl>
Message-ID: <AANLkTinQA3tUCtndMs6+0g6nVFw8CpCNsKTx+bN6s1WS@mail.gmail.com>

On Wed, Mar 23, 2011 at 5:03 AM, [Redhosting] Egbert Groot <
egbert at redhosting.nl> wrote:

> Michael Collins schreef:
>
>>
>>    I can't/didn't contribute much to this discussion, but I do follow
>>    it. My idea and (little) testing experience supports the
>>    explanation of Dimitry.
>>    Either way, it 'proofs' there is confusion about the working of
>>    the 'break' in condition tags.
>>    If I can find the time, I will try to do some tests / build an
>>    example myself. Thank you both for the answers and insights so far.
>>
>>
>> Okay, for those of you who have the bridge book, please go to page
>> 167-168. Darren did a really good job of explaining this. For those of you
>> who don't have the book, SHAME! :)
>>
>>
> Thanks for the explanation, and I just ordered the book :)
>

Glad to hear it! Also, it turns out that Darren and I both discuss this in
the book. I wrote chapter 5, and I talk about this on page 89. When Packt
did the index on the word "break" they got confused because Darren and I
used the expressions "break attribute", "break flag", and "break parameter"
all referring to the same thing.

I have my copy of the book and I write notes on every page where I find
something wrong or otherwise needs to be changed. Hopefully if/when we do a
2nd edition of this book we will be more consistent so that the index is
more useful.

-MC
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From msc at freeswitch.org  Wed Mar 23 19:51:19 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 23 Mar 2011 09:51:19 -0700
Subject: [Freeswitch-users] Event-lock,
 sync/async on outbound ESL and execute() tagging question
In-Reply-To: <AANLkTim9ftv7Q40i2ZhkfH6U8Qk+W5-NjziNt+B6anXW@mail.gmail.com>
References: <AANLkTingnGhvshmDK3TxKSkTpFeno76KTrFULRhbSCiP@mail.gmail.com>
	<AANLkTim9ftv7Q40i2ZhkfH6U8Qk+W5-NjziNt+B6anXW@mail.gmail.com>
Message-ID: <AANLkTi=uNARtTnWAjuH42129o-LN64J+n26=Reqzzrqy@mail.gmail.com>

On Wed, Mar 23, 2011 at 6:48 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Hi,
> Eventlock is used to say "Whether to queue the commands or not".
>
> Assume I'm giving playback and bridge command via outbound socket to
> freeswitch.
>
> If eventlock is "true" then, first "playback will be executed" and then
> only the bridge will be executed.
>
> If eventlock is "false", then order of execution is not guaranteed.
>
> I'm also using Outbound event socket. I'll run it in async mode. After each
> execute() statement, if I need the output for that event, I will wait for
> the event and then only I'll proceed. If I don't need event, ( playback ), I
> just do the other operations.
>
> This works well for me.
>
> Be sure that you are using the terms "inbound" and "outbound" event socket
correctly. It's inbound or outbound from the perspective of FreeSWITCH, not
from the perspective of your script. An easy rule of thumb is this: if you
use the 'socket' application in your dialplan then you are using "outbound
event socket" because FS has to make an outbound connection to your script.

http://wiki.freeswitch.org/wiki/Event_socket

-MC
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From ibc at aliax.net  Wed Mar 23 20:09:22 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Wed, 23 Mar 2011 18:09:22 +0100
Subject: [Freeswitch-users] How to disable the 407 challenge for the
 call from registerred users
In-Reply-To: <AANLkTi=gUTuTgaptYAa2AQOP4o8J6M0kscPJZ8-qpvxE@mail.gmail.com>
References: <AANLkTi=gUTuTgaptYAa2AQOP4o8J6M0kscPJZ8-qpvxE@mail.gmail.com>
Message-ID: <AANLkTi=E+WDPw6X-pq3R_fnF5xSwgrLkiQxyAUc+D5-a@mail.gmail.com>

2011/3/23 fieldpeak <fieldpeak at gmail.com>:
> Is there any one can help how to configure the FreeSwitch to disable the 407
> challenge for the call from registerred users?

That makes no sense at all. SIP is a "disconnected" protocol. Being
"registered" just means that the server does know where the user is so
can route calls to him. But if such user wants to make a call through
the server it must authenticate for every call (except in case the
server is configured to trust the user's IP:port).

In MySQL the client opens a TCP connection with the MySQL server,
sends user:password and then it can send any number of queries on top
of that connection.

SIP is not the same. In SIP each request from the client (a REGISTER,
an INVITE, etc) is an independent communication so it must be
authenticated by the server.

-- 
I?aki Baz Castillo
<ibc at aliax.net>


From richocet2 at hotmail.com  Wed Mar 23 20:16:05 2011
From: richocet2 at hotmail.com (Dave Bracken)
Date: Wed, 23 Mar 2011 12:16:05 -0500
Subject: [Freeswitch-users] Stopping uuid_record error
Message-ID: <SNT144-w107EBF7DF2D9C9EA0FB742F0B70@phx.gbl>


I have figured out how to start recording using uuid_record, but when i try to stop the record, i get: -ERR Cannot stop record session! 
here is my code(with a sudo uuid in place here) 
api uuid_record 36dfdf636dfd6f36df start test.wav 

and to stop the record: 
api uuid_record 36dfdf636dfd6f36df stop test.wav 

the start one works fine, but the stop one does not. What am i missing here? 

Thanks in advance, 
Dave  		 	   		  
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From frankie.k.yiu at gmail.com  Wed Mar 23 20:24:24 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Wed, 23 Mar 2011 10:24:24 -0700
Subject: [Freeswitch-users] Gawk.exe crashes when building on Windows 7
	64bits w/ VS
Message-ID: <AANLkTimpiqGG7fn98Up9gH65C7vgnmgM5K4Fusk7o-_V@mail.gmail.com>

My UAC setting is off.  Here is part of the build log when running Gawk.exe.
Thanks,
Frankie


33>------ Build started: Project: libsofia_sip_ua_static, Configuration:
Debug Win32 ------
32>jsscript.c
33>Performing Pre-Build Event...
33>Downloading: http://files.freeswitch.org/downloads/win32/gawk.exe
32>jsstr.c
32>jsutil.c
33>gawk: ../libsofia-sip-ua/msg/msg_parser.awk:39: fatal error: internal
error
33>This application has requested the Runtime to terminate it in an unusual
way.
33>Please contact the application's support team for more information.
32>jsxdrapi.c
32>jsxml.c
32>k_cos.c
32>Generating Code...
32>Compiling...
32>k_rem_pio2.c
32>k_sin.c
32>k_standard.c
32>k_tan.c
32>prmjtime.c
32>s_asinh.c
32>s_atan.c
32>s_cbrt.c
32>s_ceil.c
32>s_copysign.c
32>s_cos.c
32>s_erf.c
32>s_expm1.c
32>s_fabs.c
32>s_finite.c
32>s_floor.c
32>s_frexp.c
32>s_ilogb.c
32>s_isnan.c
32>s_ldexp.c
32>Generating Code...
32>Compiling...
32>s_lib_version.c
32>s_log1p.c
32>s_logb.c
32>s_matherr.c
32>s_modf.c
32>s_nextafter.c
32>s_rint.c
32>s_scalbn.c
32>s_signgam.c
32>s_significand.c
32>s_sin.c
32>s_tan.c
32>s_tanh.c
32>w_acos.c
32>w_acosh.c
32>w_asin.c
32>w_atan2.c
32>w_atanh.c
32>w_cosh.c
32>w_exp.c
32>Generating Code...
32>Compiling...
32>w_fmod.c
32>w_gamma.c
32>w_gamma_r.c
32>w_hypot.c
32>w_j0.c
32>w_j1.c
32>w_jn.c
32>w_lgamma.c
32>w_lgamma_r.c
32>w_log.c
32>w_log10.c
32>w_pow.c
32>w_remainder.c
32>w_scalb.c
32>w_sinh.c
32>w_sqrt.c
32>ntinrval.c
32>ntio.c
32>ntmisc.c
32>ntsec.c
32>Generating Code...
32>Compiling...
32>ntthread.c
32>pratom.c
32>prcthr.c
32>prdir.c
32>prerror.c
32>prfdcach.c
33>*** FAILED ***
33>NOTE:
33>NOTE: Remember to install pthreadVC2.dll to your path, too!
33>NOTE:
33>Compiling...
32>prfile.c
33>inet_pton.c
32>prinit.c
33>smoothsort.c
33>string0.c
33>su.c
32>prinrval.c
33>su_addrinfo.c
33>su_alloc.c
32>prio.c
33>su_alloc_lock.c
33>su_base_port.c
32>priometh.c
33>su_bm.c
33>su_default_log.c
33>su_errno.c
32>prlayer.c
33>su_global_log.c
33>su_localinfo.c
32>prlog.c
32>prmem.c
33>su_log.c
33>su_md5.c
33>su_os_nw.c
32>prmmap.c
33>su_port.c
33>su_pthread_port.c
32>prmwait.c
33>su_root.c
32>prolock.c
33>su_socket_port.c
32>prosdep.c
33>Generating Code...
32>prprf.c
33>Compiling...
33>su_sprintf.c
33>su_strdup.c
33>su_string.c
33>..\..\sofia-sip\libsofia-sip-ua\su\su_string.c : error C2220: warning
treated as error - no 'object' file generated
33>..\..\sofia-sip\libsofia-sip-ua\su\su_string.c : warning C4819: The file
contains a character that cannot be represented in the current code page
(950). Save the file in Unicode format to prevent data loss
32>prseg.c
33>su_strlst.c
33>su_tag.c
33>su_tag_io.c
33>su_taglist.c
32>Generating Code...
33>su_time.c
32>Compiling...
32>prtime.c
32>prtpd.c
33>su_time0.c
33>su_timer.c
32>prucpu.c
33>su_uniqueid.c
32>prucv.c
33>su_vector.c
33>su_wait.c
33>su_win32_port.c
32>prulock.c
33>base64.c
33>rc4.c
32>prustack.c
33>token64.c
33>url.c
33>url_tag.c
32>pruthr.c
33>url_tag_ref.c
33>c1 : fatal error C1083: Cannot open source file:
'..\..\sofia-sip\libsofia-sip-ua\url\url_tag_ref.c': No such file or
directory
33>Generating Code...
33>Compiling...
33>features.c
32>w32poll.c
33>bnf.c
33>msg.c
32>win32_errors.c
33>c:\work\FreeSwitch3\freeswitch\libs\sofia-sip\libsofia-sip-ua\msg\sofia-sip/msg_header.h(312)
: fatal error C1083: Cannot open include file: 'sofia-sip/msg_protos.h': No
such file or directory
33>msg_auth.c
33>c:\work\FreeSwitch3\freeswitch\libs\sofia-sip\libsofia-sip-ua\msg\sofia-sip/msg_header.h(312)
: fatal error C1083: Cannot open include file: 'sofia-sip/msg_protos.h': No
such file or directory
33>msg_basic.c
32>Generating Code...
33>c:\work\FreeSwitch3\freeswitch\libs\sofia-sip\libsofia-sip-ua\msg\sofia-sip/msg_header.h(312)
: fatal error C1083: Cannot open include file: 'sofia-sip/msg_protos.h': No
such file or directory
33>msg_date.c
32>Linking...
33>msg_generic.c
33>c:\work\FreeSwitch3\freeswitch\libs\sofia-sip\libsofia-sip-ua\msg\sofia-sip/msg_header.h(312)
: fatal error C1083: Cannot open include file: 'sofia-sip/msg_protos.h': No
such file or directory
33>msg_header_copy.c
32>   Creating library Debug/js32.lib and object Debug/js32.exp
33>c:\work\FreeSwitch3\freeswitch\libs\sofia-sip\libsofia-sip-ua\msg\sofia-sip/msg_header.h(312)
: fatal error C1083: Cannot open include file: 'sofia-sip/msg_protos.h': No
such file or directory
33>msg_header_make.c
33>c:\work\FreeSwitch3\freeswitch\libs\sofia-sip\libsofia-sip-ua\msg\sofia-sip/msg_header.h(312)
: fatal error C1083: Cannot open include file: 'sofia-sip/msg_protos.h': No
such file or directory
33>msg_mclass.c
32>Embedding manifest...
33>c:\work\FreeSwitch3\freeswitch\libs\sofia-sip\libsofia-sip-ua\msg\sofia-sip/msg_header.h(312)
: fatal error C1083: Cannot open include file: 'sofia-sip/msg_protos.h': No
such file or directory
33>msg_mime.c
33>c:\work\FreeSwitch3\freeswitch\libs\sofia-sip\libsofia-sip-ua\msg\sofia-sip/msg_header.h(312)
: fatal error C1083: Cannot open include file: 'sofia-sip/msg_protos.h': No
such file or directory
33>msg_mime_table.c
33>c1 : fatal error C1083: Cannot open source file:
'..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime_table.c': No such file or
directory
33>msg_parser.c
32>Build log was saved at "
file://c:\work\FreeSwitch3\freeswitch\libs\win32\js\Debug\BuildLog.htm"
32>js - 0 error(s), 0 warning(s)



---------- Forwarded message ----------
From: Jeff Lenk <jeff at jefflenk.com>
To: freeswitch-users at lists.freeswitch.org
Date: Wed, 23 Mar 2011 07:40:02 -0700 (PDT)
Subject: Re: [Freeswitch-users] Gawk.exe crashes when building on Windows 7
64bits w/ VS 2008
I just did a fresh clone and built with vs2008(x64 debug build) on windows 7
x64 without any problems.
Maybe if you post a Pastebin of your build log something will help.

--
View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/Gawk-exe-crashes-when-building-on-Windows-7-64bits-w-VS-2008-tp6197846p6200499.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



---------- Forwarded message ----------
From: Peter Olsson <peter.olsson at visionutveckling.se>
To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
Date: Wed, 23 Mar 2011 16:00:00 +0100
Subject: Re: [Freeswitch-users] Gawk.exe crashes when building on Windows 7
64bits w/ VS 2008
Also, check UAC related stuff - it wouldn't be the first time it caused
problems :)

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] F?r Jeff Lenk
Skickat: den 23 mars 2011 15:40
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Gawk.exe crashes when building on Windows 7
64bits w/ VS 2008

I just did a fresh clone and built with vs2008(x64 debug build) on windows 7
x64 without any problems.
Maybe if you post a Pastebin of your build log something will help.

--
View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/Gawk-exe-crashes-when-building-on-Windows-7-64bits-w-VS-2008-tp6197846p6200499.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
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From jeff at jefflenk.com  Wed Mar 23 21:11:45 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 23 Mar 2011 11:11:45 -0700 (PDT)
Subject: [Freeswitch-users] Gawk.exe crashes when building on Windows 7
	64bits w/ VS
In-Reply-To: <AANLkTimpiqGG7fn98Up9gH65C7vgnmgM5K4Fusk7o-_V@mail.gmail.com>
References: <AANLkTimpiqGG7fn98Up9gH65C7vgnmgM5K4Fusk7o-_V@mail.gmail.com>
Message-ID: <1300903905597-6201322.post@n2.nabble.com>

I wonder if there is some kind of codepage issue - can you try with a
different one?

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Re-Gawk-exe-crashes-when-building-on-Windows-7-64bits-w-VS-tp6201122p6201322.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From null at invalid.name  Wed Mar 23 21:59:55 2011
From: null at invalid.name (Dan Lane)
Date: Wed, 23 Mar 2011 18:59:55 +0000
Subject: [Freeswitch-users] Voicemail on GlusterFS
In-Reply-To: <AANLkTi=hpYhE2qdfuzuxNn=b5atbb7o-DjLSfNcy8Q6e@mail.gmail.com>
References: <AANLkTimkv66jjaBTHtT+nDUgo3PsuxEndEi-Z9=Lb0u_@mail.gmail.com>
	<AANLkTi=44afGJ1fni5P+q=0ct=irmVDwOfESUAbdWuo9@mail.gmail.com>
	<AANLkTi=bWO48xovSm_am037RWAqnCjGSFOTYTdwiVe4J@mail.gmail.com>
	<AANLkTi=hpYhE2qdfuzuxNn=b5atbb7o-DjLSfNcy8Q6e@mail.gmail.com>
Message-ID: <AANLkTinDYc2RKXMrj=ycs0zKa36gW2PZpiDgSLh-ajs1@mail.gmail.com>

On Tue, Mar 22, 2011 at 6:21 PM, Moe Navid <manavid at gmail.com> wrote:
> Hi Dan,
> Thanks for the reply. I'm also working on a custom voicemail.
> Is your vm implemented in C/C++ and uses GlusterFS' native library or you
> are just mounting with fuse?

It's written in LUA and we're using FUSE to mount the GlusterFS


From anthony.minessale at gmail.com  Wed Mar 23 22:34:02 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 23 Mar 2011 14:34:02 -0500
Subject: [Freeswitch-users] sofia_reg_internal.db grow bigger and bigger
 along with time
In-Reply-To: <AANLkTi=FVvghm6H=BiV8jT+mYqE9SVtfNa0zst+BqFvF@mail.gmail.com>
References: <AANLkTi=FVvghm6H=BiV8jT+mYqE9SVtfNa0zst+BqFvF@mail.gmail.com>
Message-ID: <AANLkTikzwCDxGnCbMCNu5rGPyF9n03F_KwxvvR+SMpDa@mail.gmail.com>

they expire so they will be deleted when the expire time happens.
If you have a lot you were probably attacked by a sip worm.


On Wed, Mar 23, 2011 at 9:38 AM, fieldpeak <fieldpeak at gmail.com> wrote:
> i found sofia_reg_internal.db grow bigger and bigger after FreeSwitch for
> some time.
> the change comes from the sip-authentication table inside this db, it always
> create new record for nonce for the new challenge ('407 Proxy Authentication
> Required') for the call from registered users.
> any one can help how to avoid the db grow, i'm afaid the hard disk will be
> full along with time...
>
> Thanks!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 kbdfck at gmail.com  Wed Mar 23 23:41:24 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Wed, 23 Mar 2011 23:41:24 +0300
Subject: [Freeswitch-users] Event-lock,
 sync/async on outbound ESL and execute() tagging question
In-Reply-To: <AANLkTi=uNARtTnWAjuH42129o-LN64J+n26=Reqzzrqy@mail.gmail.com>
References: <AANLkTingnGhvshmDK3TxKSkTpFeno76KTrFULRhbSCiP@mail.gmail.com>
	<AANLkTim9ftv7Q40i2ZhkfH6U8Qk+W5-NjziNt+B6anXW@mail.gmail.com>
	<AANLkTi=uNARtTnWAjuH42129o-LN64J+n26=Reqzzrqy@mail.gmail.com>
Message-ID: <AANLkTin6oFsWUX7gm-gA26RLzi-fuDz7ie9FtMUR0SU5@mail.gmail.com>

I understand this. I use outbound async full ESL mode, launching socket app
from dialplan, because in sync mode I have some problems with bridge app and
I need to catch parent channel hangup and server disconnection.

I written wrapper around ESL.pm to implement executeSync which waits for
CHANNEL_EXECUTE_COMPLETE while in async mode, works for now.
But of course it is not the best way, because of possible error when
'execute' is used before my 'executeSync' by mistake and without proper
event handling so that CHANNEL_EXECUTE_COMPLETE from previous app is catched
by my wrapper and it returns prematurely. This is why I want to have some
tag I can mark execute call to filter its events later... Maybe there is
better way?

2011/3/23 Michael Collins <msc at freeswitch.org>

>
>
> On Wed, Mar 23, 2011 at 6:48 AM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> Hi,
>> Eventlock is used to say "Whether to queue the commands or not".
>>
>> Assume I'm giving playback and bridge command via outbound socket to
>> freeswitch.
>>
>> If eventlock is "true" then, first "playback will be executed" and then
>> only the bridge will be executed.
>>
>> If eventlock is "false", then order of execution is not guaranteed.
>>
>> I'm also using Outbound event socket. I'll run it in async mode. After
>> each execute() statement, if I need the output for that event, I will wait
>> for the event and then only I'll proceed. If I don't need event, ( playback
>> ), I just do the other operations.
>>
>> This works well for me.
>>
>> Be sure that you are using the terms "inbound" and "outbound" event socket
> correctly. It's inbound or outbound from the perspective of FreeSWITCH, not
> from the perspective of your script. An easy rule of thumb is this: if you
> use the 'socket' application in your dialplan then you are using "outbound
> event socket" because FS has to make an outbound connection to your script.
>
> http://wiki.freeswitch.org/wiki/Event_socket
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From brian at freeswitch.org  Thu Mar 24 00:35:43 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 23 Mar 2011 16:35:43 -0500
Subject: [Freeswitch-users] How can I set the crypto suite on b-leg for
	SRTP?
In-Reply-To: <E266FB1A10564EF0A60EC751762BDEA0@omni1.local>
References: <E266FB1A10564EF0A60EC751762BDEA0@omni1.local>
Message-ID: <24CFB054-E52F-4839-83AF-A6C839502824@freeswitch.org>

Then you set the suite in {sip_secure_media=AES_CM_128_HMAC_SHA1_80}sofia/blah/blah on the originate/bridge.

/b

On Mar 22, 2011, at 2:42 PM, Anestis Mavro wrote:

> Hi,
> 
> 
> 
> I have configured TLS and SRTP on my FS and it seems to be working.
> 
> I have several phones and gateways and I have been testing SRTP.
> 
> 
> 
> My only problem is that FS sends in B-leg only one crypto suite:
> AES_CM_128_HMAC_SHA1_32
> 
> Is there a way to enable also AES_CM_128_HMAC_SHA1_80 on the b-leg?
> 
> 
> 
> The calling phone sends all three crypto suites in the sdp, but FS only one
> to the called phone.
> 
> 
> 
> Thank you
> 
> Anestis
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Thu Mar 24 01:12:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 23 Mar 2011 15:12:38 -0700
Subject: [Freeswitch-users] Diversion Header
In-Reply-To: <1300887541.2279.0.camel@gustavo-laptop>
References: <1300887541.2279.0.camel@gustavo-laptop>
Message-ID: <AANLkTin8HWEho8_gzfei3XPuLzN=Xp6ecoKSHvoYVUqy@mail.gmail.com>

On Wed, Mar 23, 2011 at 6:39 AM, Gustavo Espeche <
gustavo.espeche at easyipcall.com> wrote:

> i'm attaching a tcpdump capture.
> Thanks.
>
Did you actually look at the capture? :D
You have a ton of these:
SIP/2.0 400 Bad To Header

So, this has nothing to do with a Diversion Header, it has to do with the
fact that you have a '#' in your To Header, which is a no-no. Whoever is
sending this invite needs to URL-encode it or remove the # sign. The
offending header is this:

To: 548#254722577500 <sip:548#254722577500 at 38.108.107.228>

Brian West is wondering if this is from a Nextone device? Just curious.
Anyway, have them fix the To header and you should be okay. (Note: the
Diversion header also needs to have the # URL endcoded.)

-MC

P.S. - If they get their knickers in a twist over the URL encoding then
throw the RFC at them:
http://www.ietf.org/rfc/rfc1738.txt

Specifically, page 2 under "Unsafe:" you have this sentence:

The character "#" is unsafe and should always be encoded because it is
used in World Wide Web and in other systems to delimit a URL from a
fragment/anchor identifier that might follow it.
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From vibha_dear6 at yahoo.co.in  Wed Mar 23 21:08:32 2011
From: vibha_dear6 at yahoo.co.in (vibha dear)
Date: Wed, 23 Mar 2011 23:38:32 +0530 (IST)
Subject: [Freeswitch-users] errors while building freeswitch
Message-ID: <246320.467.qm@web137316.mail.in.yahoo.com>

while building .tar freeswitch (freeswitch.2008.express.sln) on ms visual studio 2008 i'm getting following errors. how to resolve the problem? is there any files missing in the setup?

Error	1	fatal error C1083: Cannot open source file: '..\..\pthreads-w32-2-7-0-release\pthread.c': No such file or directory	c1	pthreadError	2	fatal error C1027: Inconsistent values for /Ym between creation and use of precompiled header	c1	FreeSwitchCoreLibError	50	fatal error C1083: Cannot open source file: '..\..\sphinxbase-0.4.99\src\libsphinxbase\util\blas_lite.c': No such file or directory	c1	sphinxbase


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From curriegrad2004 at gmail.com  Thu Mar 24 01:21:49 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Wed, 23 Mar 2011 15:21:49 -0700
Subject: [Freeswitch-users] errors while building freeswitch
In-Reply-To: <246320.467.qm@web137316.mail.in.yahoo.com>
References: <246320.467.qm@web137316.mail.in.yahoo.com>
Message-ID: <AANLkTi=ah5HDb5ZiTsOknZc3qzWpDP=cJtzBYY5Zd0rf@mail.gmail.com>

pull the latest git and try again

On Wed, Mar 23, 2011 at 11:08 AM, vibha dear <vibha_dear6 at yahoo.co.in>wrote:

> while building .tar freeswitch (freeswitch.2008.express.sln) on ms visual
> studio 2008 i'm getting following errors. how to resolve the problem? is
> there any files missing in the setup?
>
>
> Error 1 fatal error C1083: Cannot open source file:
> '..\..\pthreads-w32-2-7-0-release\pthread.c': No such file or directory c1
> pthread
> Error 2 fatal error C1027: Inconsistent values for /Ym between creation
> and use of precompiled header c1 FreeSwitchCoreLib
> Error 50 fatal error C1083: Cannot open source file:
> '..\..\sphinxbase-0.4.99\src\libsphinxbase\util\blas_lite.c': No such file
> or directory c1 sphinxbase
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at microcomaustralia.com.au  Thu Mar 24 02:49:52 2011
From: brian at microcomaustralia.com.au (Brian May)
Date: Thu, 24 Mar 2011 10:49:52 +1100
Subject: [Freeswitch-users] answering calls weirdness
Message-ID: <AANLkTin5mP7QzDe+CAvaPjsbxhq934P-49LARGQJwR8Y@mail.gmail.com>

When freeswitch answers the call, everything seems fine at first, but
then it redirects the call to the magic number of 6201 which fails
because I haven't defined this number anywhere. For an incoming SIP
call the call is hang up after the first ring. For an incoming call
from FreeDTM, callerid doesn't work, and the phone keeps ringing
randomly until well after the caller has hang up. As it a result it is
not possible to answer the incoming call.

At first I thought maybe a problem with my analog telephone line, I
think I have ruled this out now that I see similar logs for SIP calls.

What is weird is that I haven't made any changes to the setup, and
everything was working fine until yesterday.

Here is an excerpt from my logs for an incoming SIP call. Where is
this magic number of 6201 coming from? I did a grep of all my config
files and came up with nothing.

Last night I rebooted the box, it worked for one call, then failed for
subsequent calls.


Any ideas??


Thanks



2011-03-24 10:31:22.061336 [DEBUG] mod_dptools.c:854
sofia/internal/1004 at microcomaustralia.com.au SET
[dialed_extension]=[2000]
EXECUTE sofia/internal/1004 at microcomaustralia.com.au
export(dialed_extension=2000)
2011-03-24 10:31:22.080434 [DEBUG] mod_dptools.c:938 EXPORT
[dialed_extension]=[2000]
EXECUTE sofia/internal/1004 at microcomaustralia.com.au bind_meta_app(1 b
s execute_extension::dx XML features)
2011-03-24 10:31:22.095423 [INFO] switch_ivr_async.c:2464 Bound B-Leg:
*1 execute_extension::dx XML features
EXECUTE sofia/internal/1004 at microcomaustralia.com.au bind_meta_app(2 b
s record_session::/opt/freeswitch/recordings/1004.2011-03-24-10-31-22.wav)
2011-03-24 10:31:22.116351 [INFO] switch_ivr_async.c:2464 Bound B-Leg:
*2 record_session::/opt/freeswitch/recordings/1004.2011-03-24-10-31-22.wav
EXECUTE sofia/internal/1004 at microcomaustralia.com.au bind_meta_app(3 b
s execute_extension::cf XML features)
2011-03-24 10:31:22.130335 [INFO] switch_ivr_async.c:2464 Bound B-Leg:
*3 execute_extension::cf XML features
EXECUTE sofia/internal/1004 at microcomaustralia.com.au set(ringback=)
2011-03-24 10:31:22.141503 [DEBUG] mod_dptools.c:854
sofia/internal/1004 at microcomaustralia.com.au SET [ringback]=[UNDEF]
EXECUTE sofia/internal/1004 at microcomaustralia.com.au
set(transfer_ringback=local_stream://moh)
2011-03-24 10:31:22.164352 [DEBUG] mod_dptools.c:854
sofia/internal/1004 at microcomaustralia.com.au SET
[transfer_ringback]=[local_stream://moh]
EXECUTE sofia/internal/1004 at microcomaustralia.com.au set(call_timeout=30)
2011-03-24 10:31:22.179416 [DEBUG] mod_dptools.c:854
sofia/internal/1004 at microcomaustralia.com.au SET [call_timeout]=[30]
EXECUTE sofia/internal/1004 at microcomaustralia.com.au
set(hangup_after_bridge=true)
2011-03-24 10:31:22.192366 [DEBUG] mod_dptools.c:854
sofia/internal/1004 at microcomaustralia.com.au SET
[hangup_after_bridge]=[true]
EXECUTE sofia/internal/1004 at microcomaustralia.com.au set(continue_on_fail=true)
2011-03-24 10:31:22.206368 [DEBUG] mod_dptools.c:854
sofia/internal/1004 at microcomaustralia.com.au SET
[continue_on_fail]=[true]
EXECUTE sofia/internal/1004 at microcomaustralia.com.au
bridge([presence_id=1000 at microcomaustralia.com.au]error/user_not_registered,FreeTDM/1/1,[presence_id=1001 at microcomaustralia.com.au]error/user_not_registered,FreeTDM/2/1,[presence_id=1002 at microcomaustralia.com.au]sofia/internal/sip:1002 at 192.168.2.13:5060,[presence_id=1003 at microcomaustralia.com.au]sofia/internal/sip:1003 at 192.168.2.13:5061,[presence_id=1004 at microcomaustralia.com.au]sofia/internal/sip:1004 at 192.168.2.16)

Everything fine up to (and beyond) this point.


2011-03-24 10:31:22.374349 [DEBUG] switch_ivr_originate.c:2599 local
variable string 0 = [presence_id=1000 at microcomaustralia.com.au]
2011-03-24 10:31:22.374349 [ERR] switch_ivr_originate.c:2632 Cannot
create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
2011-03-24 10:31:22.380323 [INFO] ftmod_zt.c:636 Setting echo cancel
to 64 taps for 1:1
2011-03-24 10:31:22.381586 [DEBUG] mod_freetdm.c:377 Set codec PCMU 20ms
2011-03-24 10:31:22.393312 [DEBUG] mod_freetdm.c:1334 Connect outbound
channel FreeTDM/1:1/
2011-03-24 10:31:22.393312 [NOTICE] switch_channel.c:779 New Channel
FreeTDM/1:1/ [d1b1124a-e8cf-485a-8c1e-dafbcaf0d09b]
2011-03-24 10:31:22.399504 [DEBUG] mod_freetdm.c:1348 (FreeTDM/1:1/)
State Change CS_NEW -> CS_INIT
2011-03-24 10:31:22.399504 [DEBUG] switch_core_session.c:1047 Send
signal FreeTDM/1:1/ [BREAK]
2011-03-24 10:31:22.399504 [DEBUG] ftmod_analog.c:361 [s1c1][1:1]
ANALOG CHANNEL thread starting.
2011-03-24 10:31:22.404683 [DEBUG] ftmod_analog.c:88 [s1c1][1:1]
Changed state from DOWN to GENRING
2011-03-24 10:31:23.514411 [INFO] ftmod_zt.c:636 Setting echo cancel
to 64 taps for 1:1
2011-03-24 10:31:23.514411 [DEBUG] ftdm_io.c:2539 [s1c1][1:1] Enabled
software DTMF detector
2011-03-24 10:31:23.514411 [DEBUG] ftmod_analog.c:381 [s1c1][1:1]
Initialized DTMF detection
2011-03-24 10:31:23.514411 [DEBUG] ftmod_analog.c:534 [s1c1][1:1]
Executing state handler on 1:1 for GENRING
2011-03-24 10:31:23.514411 [DEBUG] mod_freetdm.c:1715 got FXS sig [PROGRESS]
2011-03-24 10:31:23.514411 [NOTICE] mod_freetdm.c:1732 Ring-Ready FreeTDM/1:1/!
2011-03-24 10:31:23.525898 [DEBUG] switch_core_state_machine.c:314
(FreeTDM/1:1/) Running State Change CS_INIT
2011-03-24 10:31:23.525898 [DEBUG] switch_core_state_machine.c:338
(FreeTDM/1:1/) State INIT
2011-03-24 10:31:23.525898 [DEBUG] mod_freetdm.c:405 (FreeTDM/1:1/)
State Change CS_INIT -> CS_ROUTING
2011-03-24 10:31:23.525898 [DEBUG] switch_core_session.c:1047 Send
signal FreeTDM/1:1/ [BREAK]
2011-03-24 10:31:23.525898 [DEBUG] switch_core_state_machine.c:338
(FreeTDM/1:1/) State INIT going to sleep
2011-03-24 10:31:23.525898 [DEBUG] switch_core_state_machine.c:314
(FreeTDM/1:1/) Running State Change CS_ROUTING
2011-03-24 10:31:23.525898 [DEBUG] switch_channel.c:1512
(FreeTDM/1:1/) Callstate Change DOWN -> RINGING
2011-03-24 10:31:23.525898 [DEBUG] switch_core_state_machine.c:341
(FreeTDM/1:1/) State ROUTING
2011-03-24 10:31:23.525898 [DEBUG] mod_freetdm.c:428 FreeTDM/1:1/
CHANNEL ROUTING
2011-03-24 10:31:23.525898 [DEBUG] switch_ivr_originate.c:66
(FreeTDM/1:1/) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2011-03-24 10:31:23.525898 [DEBUG] switch_core_session.c:1047 Send
signal FreeTDM/1:1/ [BREAK]
2011-03-24 10:31:23.525898 [DEBUG] switch_core_state_machine.c:341
(FreeTDM/1:1/) State ROUTING going to sleep
2011-03-24 10:31:23.525898 [DEBUG] switch_core_state_machine.c:314
(FreeTDM/1:1/) Running State Change CS_CONSUME_MEDIA
2011-03-24 10:31:23.525898 [DEBUG] switch_core_state_machine.c:360
(FreeTDM/1:1/) State CONSUME_MEDIA
2011-03-24 10:31:23.525898 [DEBUG] switch_core_state_machine.c:360
(FreeTDM/1:1/) State CONSUME_MEDIA going to sleep
2011-03-24 10:31:23.536490 [DEBUG] switch_ivr_originate.c:2599 local
variable string 0 = [presence_id=1001 at microcomaustralia.com.au]
2011-03-24 10:31:23.538398 [ERR] switch_ivr_originate.c:2632 Cannot
create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
2011-03-24 10:31:23.538398 [INFO] ftmod_zt.c:636 Setting echo cancel
to 64 taps for 2:1
2011-03-24 10:31:23.544342 [DEBUG] mod_freetdm.c:377 Set codec PCMU 20ms
2011-03-24 10:31:23.551431 [DEBUG] mod_freetdm.c:1334 Connect outbound
channel FreeTDM/2:1/
2011-03-24 10:31:23.555384 [NOTICE] switch_channel.c:779 New Channel
FreeTDM/2:1/ [86275359-c4be-4663-9561-c5973344c1a7]
2011-03-24 10:31:23.559646 [DEBUG] mod_freetdm.c:1348 (FreeTDM/2:1/)
State Change CS_NEW -> CS_INIT
2011-03-24 10:31:23.561459 [DEBUG] switch_core_session.c:1047 Send
signal FreeTDM/2:1/ [BREAK]
2011-03-24 10:31:23.561459 [DEBUG] ftmod_analog.c:361 [s2c1][1:2]
ANALOG CHANNEL thread starting.
2011-03-24 10:31:23.565467 [DEBUG] ftmod_analog.c:88 [s2c1][1:2]
Changed state from DOWN to GENRING
2011-03-24 10:31:24.665349 [INFO] ftmod_zt.c:636 Setting echo cancel
to 64 taps for 2:1
2011-03-24 10:31:24.665349 [DEBUG] ftdm_io.c:2539 [s2c1][1:2] Enabled
software DTMF detector
2011-03-24 10:31:24.665349 [DEBUG] ftmod_analog.c:381 [s2c1][1:2]
Initialized DTMF detection
2011-03-24 10:31:24.665349 [DEBUG] ftmod_analog.c:534 [s2c1][1:2]
Executing state handler on 2:1 for GENRING
2011-03-24 10:31:24.665349 [DEBUG] mod_freetdm.c:1715 got FXS sig [PROGRESS]
2011-03-24 10:31:24.665349 [NOTICE] mod_freetdm.c:1732 Ring-Ready FreeTDM/2:1/!
2011-03-24 10:31:24.677691 [DEBUG] switch_core_state_machine.c:314
(FreeTDM/2:1/) Running State Change CS_INIT
2011-03-24 10:31:24.677691 [DEBUG] switch_core_state_machine.c:338
(FreeTDM/2:1/) State INIT
2011-03-24 10:31:24.677691 [DEBUG] mod_freetdm.c:405 (FreeTDM/2:1/)
State Change CS_INIT -> CS_ROUTING
2011-03-24 10:31:24.677691 [DEBUG] switch_core_session.c:1047 Send
signal FreeTDM/2:1/ [BREAK]
2011-03-24 10:31:24.677691 [DEBUG] switch_core_state_machine.c:338
(FreeTDM/2:1/) State INIT going to sleep
2011-03-24 10:31:24.677691 [DEBUG] switch_core_state_machine.c:314
(FreeTDM/2:1/) Running State Change CS_ROUTING
2011-03-24 10:31:24.677691 [DEBUG] switch_channel.c:1512
(FreeTDM/2:1/) Callstate Change DOWN -> RINGING
2011-03-24 10:31:24.677691 [DEBUG] switch_core_state_machine.c:341
(FreeTDM/2:1/) State ROUTING
2011-03-24 10:31:24.677691 [DEBUG] mod_freetdm.c:428 FreeTDM/2:1/
CHANNEL ROUTING
2011-03-24 10:31:24.677691 [DEBUG] switch_ivr_originate.c:66
(FreeTDM/2:1/) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2011-03-24 10:31:24.677691 [DEBUG] switch_core_session.c:1047 Send
signal FreeTDM/2:1/ [BREAK]
2011-03-24 10:31:24.677691 [DEBUG] switch_core_state_machine.c:341
(FreeTDM/2:1/) State ROUTING going to sleep
2011-03-24 10:31:24.677691 [DEBUG] switch_core_state_machine.c:314
(FreeTDM/2:1/) Running State Change CS_CONSUME_MEDIA
2011-03-24 10:31:24.677691 [DEBUG] switch_core_state_machine.c:360
(FreeTDM/2:1/) State CONSUME_MEDIA
2011-03-24 10:31:24.677691 [DEBUG] switch_core_state_machine.c:360
(FreeTDM/2:1/) State CONSUME_MEDIA going to sleep
2011-03-24 10:31:24.692818 [DEBUG] switch_ivr_originate.c:2599 local
variable string 0 = [presence_id=1002 at microcomaustralia.com.au]
2011-03-24 10:31:24.697423 [NOTICE] switch_channel.c:779 New Channel
sofia/internal/sip:1002 at 192.168.2.13:5060
[0da6383c-4633-4a03-99ed-839918a74161]
2011-03-24 10:31:24.737484 [DEBUG] mod_sofia.c:3920
(sofia/internal/sip:1002 at 192.168.2.13:5060) State Change CS_NEW ->
CS_INIT
2011-03-24 10:31:24.737484 [DEBUG] switch_core_session.c:1047 Send
signal sofia/internal/sip:1002 at 192.168.2.13:5060 [BREAK]
2011-03-24 10:31:24.752679 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1002 at 192.168.2.13:5060) Running State Change
CS_INIT
2011-03-24 10:31:24.756503 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/sip:1002 at 192.168.2.13:5060) State INIT
2011-03-24 10:31:24.758368 [DEBUG] mod_sofia.c:83
sofia/internal/sip:1002 at 192.168.2.13:5060 SOFIA INIT
2011-03-24 10:31:24.777518 [DEBUG] mod_sofia.c:123
(sofia/internal/sip:1002 at 192.168.2.13:5060) State Change CS_INIT ->
CS_ROUTING
2011-03-24 10:31:24.777518 [DEBUG] switch_core_session.c:1047 Send
signal sofia/internal/sip:1002 at 192.168.2.13:5060 [BREAK]
2011-03-24 10:31:24.777518 [DEBUG] sofia.c:4402 Channel
sofia/internal/sip:1002 at 192.168.2.13:5060 entering state [calling][0]
2011-03-24 10:31:24.781657 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/sip:1002 at 192.168.2.13:5060) State INIT going to sleep
2011-03-24 10:31:24.781657 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1002 at 192.168.2.13:5060) Running State Change
CS_ROUTING
2011-03-24 10:31:24.785366 [DEBUG] switch_channel.c:1512
(sofia/internal/sip:1002 at 192.168.2.13:5060) Callstate Change DOWN ->
RINGING
2011-03-24 10:31:24.799334 [INFO] sofia.c:709
sofia/internal/sip:1002 at 192.168.2.13:5060 Update Callee ID to "1002"
<1002>
2011-03-24 10:31:24.808910 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/sip:1002 at 192.168.2.13:5060) State ROUTING
2011-03-24 10:31:24.808910 [DEBUG] sofia.c:4402 Channel
sofia/internal/sip:1002 at 192.168.2.13:5060 entering state
[proceeding][180]
2011-03-24 10:31:24.812354 [NOTICE] sofia.c:4474 Ring-Ready
sofia/internal/sip:1002 at 192.168.2.13:5060!
2011-03-24 10:31:24.815429 [DEBUG] mod_sofia.c:146
sofia/internal/sip:1002 at 192.168.2.13:5060 SOFIA ROUTING
2011-03-24 10:31:24.815429 [DEBUG] switch_ivr_originate.c:66
(sofia/internal/sip:1002 at 192.168.2.13:5060) State Change CS_ROUTING ->
CS_CONSUME_MEDIA
2011-03-24 10:31:24.815429 [DEBUG] switch_core_session.c:1047 Send
signal sofia/internal/sip:1002 at 192.168.2.13:5060 [BREAK]
2011-03-24 10:31:24.815429 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/sip:1002 at 192.168.2.13:5060) State ROUTING going to
sleep
2011-03-24 10:31:24.815429 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1002 at 192.168.2.13:5060) Running State Change
CS_CONSUME_MEDIA
2011-03-24 10:31:24.817284 [DEBUG] switch_core_state_machine.c:360
(sofia/internal/sip:1002 at 192.168.2.13:5060) State CONSUME_MEDIA
2011-03-24 10:31:24.817284 [DEBUG] switch_core_state_machine.c:360
(sofia/internal/sip:1002 at 192.168.2.13:5060) State CONSUME_MEDIA going
to sleep
2011-03-24 10:31:24.819480 [DEBUG] switch_ivr_originate.c:2599 local
variable string 0 = [presence_id=1003 at microcomaustralia.com.au]
2011-03-24 10:31:24.819480 [NOTICE] switch_channel.c:779 New Channel
sofia/internal/sip:1003 at 192.168.2.13:5061
[644a2402-7658-4831-8599-2439dc82e1f2]
2011-03-24 10:31:24.850900 [DEBUG] mod_sofia.c:3920
(sofia/internal/sip:1003 at 192.168.2.13:5061) State Change CS_NEW ->
CS_INIT
2011-03-24 10:31:24.851903 [DEBUG] switch_core_session.c:1047 Send
signal sofia/internal/sip:1003 at 192.168.2.13:5061 [BREAK]
2011-03-24 10:31:24.865923 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1003 at 192.168.2.13:5061) Running State Change
CS_INIT
2011-03-24 10:31:24.865923 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/sip:1003 at 192.168.2.13:5061) State INIT
2011-03-24 10:31:24.872367 [DEBUG] mod_sofia.c:83
sofia/internal/sip:1003 at 192.168.2.13:5061 SOFIA INIT
2011-03-24 10:31:24.897626 [DEBUG] mod_sofia.c:123
(sofia/internal/sip:1003 at 192.168.2.13:5061) State Change CS_INIT ->
CS_ROUTING
2011-03-24 10:31:24.898904 [DEBUG] switch_core_session.c:1047 Send
signal sofia/internal/sip:1003 at 192.168.2.13:5061 [BREAK]
2011-03-24 10:31:24.898904 [DEBUG] sofia.c:4402 Channel
sofia/internal/sip:1003 at 192.168.2.13:5061 entering state [calling][0]
2011-03-24 10:31:24.899882 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/sip:1003 at 192.168.2.13:5061) State INIT going to sleep
2011-03-24 10:31:24.899882 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1003 at 192.168.2.13:5061) Running State Change
CS_ROUTING
2011-03-24 10:31:24.920400 [DEBUG] sofia.c:4402 Channel
sofia/internal/sip:1003 at 192.168.2.13:5061 entering state [calling][0]
2011-03-24 10:31:24.930315 [DEBUG] sofia.c:6117 IP 59.167.180.194
Rejected by acl "domains". Falling back to Digest auth.
2011-03-24 10:31:24.930315 [NOTICE] switch_channel.c:779 New Channel
sofia/internal/1004 at 59.167.180.194
[294c0493-b397-481a-ac78-09fa727d79b9]
2011-03-24 10:31:24.939488 [DEBUG] switch_channel.c:1512
(sofia/internal/sip:1003 at 192.168.2.13:5061) Callstate Change DOWN ->
RINGING
2011-03-24 10:31:24.947209 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/sip:1003 at 192.168.2.13:5061) State ROUTING
2011-03-24 10:31:24.947209 [DEBUG] mod_sofia.c:146
sofia/internal/sip:1003 at 192.168.2.13:5061 SOFIA ROUTING
2011-03-24 10:31:24.949210 [DEBUG] switch_ivr_originate.c:66
(sofia/internal/sip:1003 at 192.168.2.13:5061) State Change CS_ROUTING ->
CS_CONSUME_MEDIA
2011-03-24 10:31:24.950890 [DEBUG] switch_core_session.c:1047 Send
signal sofia/internal/sip:1003 at 192.168.2.13:5061 [BREAK]
2011-03-24 10:31:24.950890 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/sip:1003 at 192.168.2.13:5061) State ROUTING going to
sleep
2011-03-24 10:31:24.950890 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1003 at 192.168.2.13:5061) Running State Change
CS_CONSUME_MEDIA
2011-03-24 10:31:24.954269 [DEBUG] switch_core_state_machine.c:360
(sofia/internal/sip:1003 at 192.168.2.13:5061) State CONSUME_MEDIA
2011-03-24 10:31:24.954269 [DEBUG] switch_core_state_machine.c:360
(sofia/internal/sip:1003 at 192.168.2.13:5061) State CONSUME_MEDIA going
to sleep
2011-03-24 10:31:24.954269 [DEBUG] switch_ivr_originate.c:2599 local
variable string 0 = [presence_id=1004 at microcomaustralia.com.au]
2011-03-24 10:31:24.960337 [NOTICE] switch_channel.c:779 New Channel
sofia/internal/sip:1004 at 192.168.2.16
[55c3e984-ddf8-45db-a157-65c50e18c8d0]
2011-03-24 10:31:24.970358 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1004 at 59.167.180.194) Running State Change CS_NEW
2011-03-24 10:31:24.970358 [DEBUG] switch_core_state_machine.c:320
(sofia/internal/1004 at 59.167.180.194) State NEW
2011-03-24 10:31:24.994428 [DEBUG] mod_sofia.c:3920
(sofia/internal/sip:1004 at 192.168.2.16) State Change CS_NEW -> CS_INIT
2011-03-24 10:31:24.996392 [DEBUG] switch_core_session.c:1047 Send
signal sofia/internal/sip:1004 at 192.168.2.16 [BREAK]
2011-03-24 10:31:25.023275 [DEBUG] sofia.c:4402 Channel
sofia/internal/1004 at 59.167.180.194 entering state [received][100]
2011-03-24 10:31:25.029365 [DEBUG] sofia.c:4413 Remote SDP:
v=0
o=FreeSWITCH 1300899030 1300899032 IN IP4 59.167.180.194
s=FreeSWITCH
c=IN IP4 59.167.180.194
t=0 0
m=audio 24054 RTP/AVP 8 115 107 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:107 G7221/16000
a=fmtp:107 bitrate=32000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

2011-03-24 10:31:25.035447 [DEBUG] sofia_glue.c:3859 Audio Codec
Compare [PCMA:8:8000:20]/[G7221:115:32000:20]
2011-03-24 10:31:25.035447 [DEBUG] sofia_glue.c:3859 Audio Codec
Compare [PCMA:8:8000:20]/[G7221:107:16000:20]
2011-03-24 10:31:25.035447 [DEBUG] sofia_glue.c:3859 Audio Codec
Compare [PCMA:8:8000:20]/[PCMU:0:8000:20]
2011-03-24 10:31:25.035447 [DEBUG] sofia_glue.c:3859 Audio Codec
Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
2011-03-24 10:31:25.037256 [DEBUG] sofia_glue.c:2454 Set Codec
sofia/internal/1004 at 59.167.180.194 PCMA/8000 20 ms 160 samples
2011-03-24 10:31:25.049229 [DEBUG] sofia_glue.c:3955 Set 2833 dtmf
send/recv payload to 101
2011-03-24 10:31:25.049229 [DEBUG] sofia.c:4573
(sofia/internal/1004 at 59.167.180.194) State Change CS_NEW -> CS_INIT
2011-03-24 10:31:25.049229 [DEBUG] switch_core_session.c:1047 Send
signal sofia/internal/1004 at 59.167.180.194 [BREAK]
2011-03-24 10:31:25.053356 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1004 at 59.167.180.194) Running State Change CS_INIT
2011-03-24 10:31:25.053356 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/1004 at 59.167.180.194) State INIT
2011-03-24 10:31:25.055393 [DEBUG] mod_sofia.c:83
sofia/internal/1004 at 59.167.180.194 SOFIA INIT
2011-03-24 10:31:25.055393 [DEBUG] mod_sofia.c:123
(sofia/internal/1004 at 59.167.180.194) State Change CS_INIT ->
CS_ROUTING
2011-03-24 10:31:25.055393 [DEBUG] switch_core_session.c:1047 Send
signal sofia/internal/1004 at 59.167.180.194 [BREAK]
2011-03-24 10:31:25.055393 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/1004 at 59.167.180.194) State INIT going to sleep
2011-03-24 10:31:25.055393 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1004 at 59.167.180.194) Running State Change CS_ROUTING
2011-03-24 10:31:25.055393 [DEBUG] switch_channel.c:1512
(sofia/internal/1004 at 59.167.180.194) Callstate Change DOWN -> RINGING
2011-03-24 10:31:25.055393 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/1004 at 59.167.180.194) State ROUTING
2011-03-24 10:31:25.055393 [DEBUG] mod_sofia.c:146
sofia/internal/1004 at 59.167.180.194 SOFIA ROUTING
2011-03-24 10:31:25.055393 [DEBUG] switch_core_state_machine.c:77
sofia/internal/1004 at 59.167.180.194 Standard ROUTING
2011-03-24 10:31:25.055393 [INFO] mod_dialplan_xml.c:331 Processing
Brian May <1004>->6201 in context public


Something went wrong here. It starts processing 6201 as per normal.
-- 
Brian May <brian at microcomaustralia.com.au>


From fieldpeak at gmail.com  Thu Mar 24 05:42:48 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Thu, 24 Mar 2011 10:42:48 +0800
Subject: [Freeswitch-users] sofia_reg_internal.db grow bigger and bigger
 along with time
In-Reply-To: <AANLkTikzwCDxGnCbMCNu5rGPyF9n03F_KwxvvR+SMpDa@mail.gmail.com>
References: <AANLkTi=FVvghm6H=BiV8jT+mYqE9SVtfNa0zst+BqFvF@mail.gmail.com>
	<AANLkTikzwCDxGnCbMCNu5rGPyF9n03F_KwxvvR+SMpDa@mail.gmail.com>
Message-ID: <AANLkTi=aHzQQe4b=Su5Fr7ybwvNQ42-JU4gMZY9=Cird@mail.gmail.com>

Hi Anthony,

Thanks for your reply.
i tried to understand how long it is the expire time and where to cnofigure
it.
so i made a test, see the attached screen shot, in the profile internal.xml,
the nonce-ttl = 60, when i make a call from a registerred user, a new record
was added in the 'sip_authentication' table with a very big value
(1300933453, i don't know what the unit is, s or ms?) , after 60 seconds
(actually i wait for more than 15 minutes), even the call finished, the new
record still stay their, it was not deleted. want to know when it will be
deleted...

thanks for your help!

2011/3/24 Anthony Minessale <anthony.minessale at gmail.com>

> they expire so they will be deleted when the expire time happens.
> If you have a lot you were probably attacked by a sip worm.
>
>
> On Wed, Mar 23, 2011 at 9:38 AM, fieldpeak <fieldpeak at gmail.com> wrote:
> > i found sofia_reg_internal.db grow bigger and bigger after FreeSwitch for
> > some time.
> > the change comes from the sip-authentication table inside this db, it
> always
> > create new record for nonce for the new challenge ('407 Proxy
> Authentication
> > Required') for the call from registered users.
> > any one can help how to avoid the db grow, i'm afaid the hard disk will
> be
> > full along with time...
> >
> > Thanks!
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fieldpeak at gmail.com  Thu Mar 24 05:47:06 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Thu, 24 Mar 2011 10:47:06 +0800
Subject: [Freeswitch-users] sofia_reg_internal.db grow bigger and bigger
 along with time
In-Reply-To: <AANLkTi=aHzQQe4b=Su5Fr7ybwvNQ42-JU4gMZY9=Cird@mail.gmail.com>
References: <AANLkTi=FVvghm6H=BiV8jT+mYqE9SVtfNa0zst+BqFvF@mail.gmail.com>
	<AANLkTikzwCDxGnCbMCNu5rGPyF9n03F_KwxvvR+SMpDa@mail.gmail.com>
	<AANLkTi=aHzQQe4b=Su5Fr7ybwvNQ42-JU4gMZY9=Cird@mail.gmail.com>
Message-ID: <AANLkTikKdb7ey7U2x-6QkfUrR-uTcyAT=JR8Bs7tVjHi@mail.gmail.com>

sorry, forgot attach the screenshot file.

2011/3/24 fieldpeak <fieldpeak at gmail.com>

> Hi Anthony,
>
> Thanks for your reply.
> i tried to understand how long it is the expire time and where to cnofigure
> it.
> so i made a test, see the attached screen shot, in the profile
> internal.xml, the nonce-ttl = 60, when i make a call from a registerred
> user, a new record was added in the 'sip_authentication' table with a very
> big value (1300933453, i don't know what the unit is, s or ms?) , after 60
> seconds (actually i wait for more than 15 minutes), even the call finished,
> the new record still stay their, it was not deleted. want to know when it
> will be deleted...
>
> thanks for your help!
>
>
> 2011/3/24 Anthony Minessale <anthony.minessale at gmail.com>
>
>> they expire so they will be deleted when the expire time happens.
>> If you have a lot you were probably attacked by a sip worm.
>>
>>
>> On Wed, Mar 23, 2011 at 9:38 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>> > i found sofia_reg_internal.db grow bigger and bigger after FreeSwitch
>> for
>> > some time.
>> > the change comes from the sip-authentication table inside this db, it
>> always
>> > create new record for nonce for the new challenge ('407 Proxy
>> Authentication
>> > Required') for the call from registered users.
>> > any one can help how to avoid the db grow, i'm afaid the hard disk will
>> be
>> > full along with time...
>> >
>> > Thanks!
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From msc at freeswitch.org  Thu Mar 24 05:55:50 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 23 Mar 2011 19:55:50 -0700
Subject: [Freeswitch-users] sofia_reg_internal.db grow bigger and bigger
 along with time
In-Reply-To: <AANLkTi=aHzQQe4b=Su5Fr7ybwvNQ42-JU4gMZY9=Cird@mail.gmail.com>
References: <AANLkTi=FVvghm6H=BiV8jT+mYqE9SVtfNa0zst+BqFvF@mail.gmail.com>
	<AANLkTikzwCDxGnCbMCNu5rGPyF9n03F_KwxvvR+SMpDa@mail.gmail.com>
	<AANLkTi=aHzQQe4b=Su5Fr7ybwvNQ42-JU4gMZY9=Cird@mail.gmail.com>
Message-ID: <AANLkTinGr6YfSjDx9oAJV-wKNSmUx9Nq_i7KQfXR3RDQ@mail.gmail.com>

1300933453 is a unix epoch time in seconds. You can get the exact time from
the fs_cli by issuing the strepoch command. Anyway, the size of that number
is not the issue. I'll defer to Tony on the database record remaining in the
table for longer than expected...

-MC

On Wed, Mar 23, 2011 at 7:42 PM, fieldpeak <fieldpeak at gmail.com> wrote:

> Hi Anthony,
>
> Thanks for your reply.
> i tried to understand how long it is the expire time and where to cnofigure
> it.
> so i made a test, see the attached screen shot, in the profile
> internal.xml, the nonce-ttl = 60, when i make a call from a registerred
> user, a new record was added in the 'sip_authentication' table with a very
> big value (1300933453, i don't know what the unit is, s or ms?) , after 60
> seconds (actually i wait for more than 15 minutes), even the call finished,
> the new record still stay their, it was not deleted. want to know when it
> will be deleted...
>
> thanks for your help!
>
> 2011/3/24 Anthony Minessale <anthony.minessale at gmail.com>
>
>> they expire so they will be deleted when the expire time happens.
>> If you have a lot you were probably attacked by a sip worm.
>>
>>
>> On Wed, Mar 23, 2011 at 9:38 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>> > i found sofia_reg_internal.db grow bigger and bigger after FreeSwitch
>> for
>> > some time.
>> > the change comes from the sip-authentication table inside this db, it
>> always
>> > create new record for nonce for the new challenge ('407 Proxy
>> Authentication
>> > Required') for the call from registered users.
>> > any one can help how to avoid the db grow, i'm afaid the hard disk will
>> be
>> > full along with time...
>> >
>> > Thanks!
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chistyakov at directtel.ru  Thu Mar 24 10:01:38 2011
From: chistyakov at directtel.ru (=?UTF-8?B?0KfQuNGB0YLRj9C60L7QsiDQmNCy0LDQvQ==?=)
Date: Thu, 24 Mar 2011 10:01:38 +0300
Subject: [Freeswitch-users] Can I create an EventConsumer only for
 specified uuid?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B30F0CF5@cooper>
References: <4D89F3E2.40308@directtel.ru>	<549CFEF87AEDE841A38E9D15EAB4C04C58B30F0C28@cooper>	<4D89FB21.4080109@directtel.ru>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B30F0CF5@cooper>
Message-ID: <4D8AEC52.6010607@directtel.ru>

I don't using mod_event_socket. I am using freeswitch.EventConsumer with 
LUA.
How can I set this filter?

23.03.2011 19:39, Peter Olsson ?????:
> Send a command like this on the event socket.
>
> Also, here you have some more info: http://wiki.freeswitch.org/wiki/Mod_event_socket
>
> filter Unique-ID YOUR_UUID
>
> /Peter
>
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r ???????? ????
> Skickat: den 23 mars 2011 14:53
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] Can I create an EventConsumer only for specified uuid?
>
> How can I add it?
>
> 23.03.2011 16:32, Peter Olsson ?????:
>> If you add a filter that should work just fine.
>>
>> Mvh
>> Peter Olsson
>>
>> -----Ursprungligt meddelande-----
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r ???????? ????
>> Skickat: den 23 mars 2011 14:22
>> Till: FreeSWITCH Users Help
>> ?mne: [Freeswitch-users] Can I create an EventConsumer only for specified uuid?
>>
>> Can I create an EventConsumer only for specified uuid?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4d89fbf532761360168620!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From frankie.k.yiu at gmail.com  Thu Mar 24 10:56:08 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Thu, 24 Mar 2011 00:56:08 -0700
Subject: [Freeswitch-users] Gawk.exe crashes when building on Windows 7
	64bits w/ VS
Message-ID: <AANLkTinFgrm_SsBoBsgmCROfZG2B0jrGkmU5YOVBtPfF@mail.gmail.com>

How exactly do I change that?

Thanks,
Frankie



---------- Forwarded message ----------
From: Jeff Lenk <jeff at jefflenk.com>
To: freeswitch-users at lists.freeswitch.org
Date: Wed, 23 Mar 2011 11:11:45 -0700 (PDT)
Subject: Re: [Freeswitch-users] Gawk.exe crashes when building on Windows 7
64bits w/ VS
I wonder if there is some kind of codepage issue - can you try with a
different one?

--
View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/Re-Gawk-exe-crashes-when-building-on-Windows-7-64bits-w-VS-tp6201122p6201322.html
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From fieldpeak at gmail.com  Thu Mar 24 11:34:47 2011
From: fieldpeak at gmail.com (=?utf-8?B?Q2hhcmxlcw==?=)
Date: Thu, 24 Mar 2011 16:34:47 +0800
Subject: [Freeswitch-users]
	=?utf-8?q?sofia=5Freg=5Finternal=2Edb_grow_big?=
	=?utf-8?q?ger_and_biggeralong_with_time?=
References: <AANLkTi=FVvghm6H=BiV8jT+mYqE9SVtfNa0zst+BqFvF@mail.gmail.com>,
	<AANLkTikzwCDxGnCbMCNu5rGPyF9n03F_KwxvvR+SMpDa@mail.gmail.com>,
	<AANLkTi=aHzQQe4b=Su5Fr7ybwvNQ42-JU4gMZY9=Cird@mail.gmail.com>,
	<AANLkTinGr6YfSjDx9oAJV-wKNSmUx9Nq_i7KQfXR3RDQ@mail.gmail.com>
Message-ID: <4d8b022b.0a13e70a.112f.652a@mx.google.com>

Hi Michael,

unstood it's a unix epoch time now.
how, it always stay in the table even the call finshied which will lead the database get bigger and bigger till disk reach full, especially we set db on RAM disk which is limited by the physical RAM size, can you help advise how to avoid this happen.
thanks!


2011-03-24 



Charles 



???? Michael Collins 
????? 2011-03-24  10:56:52 
???? FreeSWITCH Users Help 
??? 
??? Re: [Freeswitch-users] sofia_reg_internal.db grow bigger and biggeralong with time 
 
1300933453 is a unix epoch time in seconds. You can get the exact time from the fs_cli by issuing the strepoch command. Anyway, the size of that number is not the issue. I'll defer to Tony on the database record remaining in the table for longer than expected...


-MC


On Wed, Mar 23, 2011 at 7:42 PM, fieldpeak <fieldpeak at gmail.com> wrote:

Hi Anthony,

Thanks for your reply.
i tried to understand how long it is the expire time and where to cnofigure it.
so i made a test, see the attached screen shot, in the profile internal.xml, the nonce-ttl = 60, when i make a call from a registerred user, a new record was added in the 'sip_authentication' table with a very big value (1300933453, i don't know what the unit is, s or ms?) , after 60 seconds (actually i wait for more than 15 minutes), even the call finished, the new record still stay their, it was not deleted. want to know when it will be deleted...

thanks for your help!


2011/3/24 Anthony Minessale <anthony.minessale at gmail.com>

they expire so they will be deleted when the expire time happens.
If you have a lot you were probably attacked by a sip worm.



On Wed, Mar 23, 2011 at 9:38 AM, fieldpeak <fieldpeak at gmail.com> wrote:
> i found sofia_reg_internal.db grow bigger and bigger after FreeSwitch for
> some time.
> the change comes from the sip-authentication table inside this db, it always
> create new record for nonce for the new challenge ('407 Proxy Authentication
> Required') for the call from registered users.
> any one can help how to avoid the db grow, i'm afaid the hard disk will be
> full along with time...
>
> Thanks!
>

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



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From Nabble at slickdeals.endjunk.com  Thu Mar 24 13:48:09 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 24 Mar 2011 03:48:09 -0700 (PDT)
Subject: [Freeswitch-users] sofia_reg_internal.db grow bigger and bigger
	along with time
In-Reply-To: <AANLkTi=FVvghm6H=BiV8jT+mYqE9SVtfNa0zst+BqFvF@mail.gmail.com>
References: <AANLkTi=FVvghm6H=BiV8jT+mYqE9SVtfNa0zst+BqFvF@mail.gmail.com>
Message-ID: <1300963689185-6203596.post@n2.nabble.com>

I am just curious what version of FreeSWITCH are you using that gives you
this problem.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/sofia-reg-internal-db-grow-bigger-and-bigger-along-with-time-tp6200516p6203596.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From abeka at greatiam.com  Thu Mar 24 15:05:16 2011
From: abeka at greatiam.com (Samuel)
Date: Thu, 24 Mar 2011 12:05:16 +0000
Subject: [Freeswitch-users] static build of the stable Skype client 2.0.72
In-Reply-To: <mailman.5925.1300955723.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.5925.1300955723.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <4D8B337C.6010600@greatiam.com>

Hello

I am trying out Mod skyopen Skype fromn the wiki but cannot locate the 
recommended skype client. Has anyone got the link  and can the please 
let me have it ?

thanks



From fieldpeak at gmail.com  Thu Mar 24 15:11:47 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Thu, 24 Mar 2011 20:11:47 +0800
Subject: [Freeswitch-users] sofia_reg_internal.db grow bigger and bigger
 along with time
In-Reply-To: <1300963689185-6203596.post@n2.nabble.com>
References: <AANLkTi=FVvghm6H=BiV8jT+mYqE9SVtfNa0zst+BqFvF@mail.gmail.com>
	<1300963689185-6203596.post@n2.nabble.com>
Message-ID: <AANLkTimAjOrO3Qp6r09G+--tEKbbPrvTHjVxkfmoTZ=o@mail.gmail.com>

The fs version is git-head on Noveber,30,2010

? 2011-3-24 ??6:48?"mazilo" <Nabble at slickdeals.endjunk.com>???

I am just curious what version of FreeSWITCH are you using that gives you
this problem.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/sofia-reg-internal-db-grow-bigger-and-bigger-along-with-time-tp6200516p6203596.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at list...
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From chistyakov at directtel.ru  Thu Mar 24 15:18:19 2011
From: chistyakov at directtel.ru (=?UTF-8?B?0KfQuNGB0YLRj9C60L7QsiDQmNCy0LDQvQ==?=)
Date: Thu, 24 Mar 2011 15:18:19 +0300
Subject: [Freeswitch-users] I am trying to use ESL with LUA - undefined
	symbol: _ZN8ESLeventC1EPKcS1_
Message-ID: <4D8B368B.7050909@directtel.ru>

I am trying to use ESL with LUA:

/libs/esl/lua# ./single_command.lua help
/usr/local/bin/lua: error loading module 'ESL' from file 
'/usr/lib/lua/5.1/ESL.so':
     /usr/lib/lua/5.1/ESL.so: undefined symbol: _ZN8ESLeventC1EPKcS1_
stack traceback:
     [C]: ?
     [C]: in function 'require'
     ./single_command.lua:2: in main chunk
     [C]: ?


What's wrong?


From fieldpeak at gmail.com  Thu Mar 24 15:19:59 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Thu, 24 Mar 2011 20:19:59 +0800
Subject: [Freeswitch-users] sofia_reg_internal.db grow bigger and bigger
 along with time
In-Reply-To: <AANLkTimjjtKju704KWokqXYySjx9=hZk5E_H2dAaqtxG@mail.gmail.com>
References: <AANLkTi=FVvghm6H=BiV8jT+mYqE9SVtfNa0zst+BqFvF@mail.gmail.com>
	<1300963689185-6203596.post@n2.nabble.com>
	<AANLkTimAjOrO3Qp6r09G+--tEKbbPrvTHjVxkfmoTZ=o@mail.gmail.com>
	<AANLkTimjjtKju704KWokqXYySjx9=hZk5E_H2dAaqtxG@mail.gmail.com>
Message-ID: <AANLkTi=KZ1TgsUC48ttL2ER4EhfRA2kBbNm5ctZqxA1j@mail.gmail.com>

is it a known bug and fixed already by the newer version?
thanks!

? 2011-3-24 ??6:48?"mazilo" <Nabble at slickdeals.endjunk.com>???


>
> I am just curious what version of FreeSWITCH are you using that gives you
> this problem.
>
> -...


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at list...
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From w8hdkim at gmail.com  Thu Mar 24 16:22:55 2011
From: w8hdkim at gmail.com (Kim Culhan)
Date: Thu, 24 Mar 2011 09:22:55 -0400
Subject: [Freeswitch-users] Bug ID: FS-2868 Status?
Message-ID: <AANLkTinhDPN8VaHUYmSktvcAiV-bsU47j_A1JhMQuMOf@mail.gmail.com>

On Wed, March 23, 2011 9:15 am, Michal Bielicki wrote:
> The problem is a bug in solaris perl development package that I haven't
been doing
> anything about since I do not use perl for roughly a decade.
> Its not the hash module, its the perl module.
>
> Am 23.03.2011 um 11:44 schrieb A E [Gmail]:
>
>> Hello FreeSwitchers!
>>
>> Am attempting to compile the latest (off of Git) code for FS on Solaris
11 Express
>> on an 64-Bit x86 machine. However, I'm running into the error that Bruce
>> McAllister has reported on 23rd Nov 2010. The bug is still sitting
unresolved. So
>> I'm assuming there's a backdoor trick to fixing this OR that some other
version of
>> FS code doesn't have this issue. Could someone please let me on the
secret on
>> which build compiles on x86 Solaris v10+ and/or what the backdoor trick
to bypass
>> this is? Should I not compile the hash module where this bug seems to be
>> originating from?

I'm seeing the same error here with the perl module commented-out in
modules.conf

Please describe the workaround you are using to avoid the problem caused
by the perl module.

thanks
-kim
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From Nabble at slickdeals.endjunk.com  Thu Mar 24 16:33:55 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 24 Mar 2011 06:33:55 -0700 (PDT)
Subject: [Freeswitch-users] Question about mod_dingaling.
In-Reply-To: <AANLkTinpUymFWTdoU1Xx0oNFNhyRAKx04SmoWy5avVC9@mail.gmail.com>
References: <AANLkTikd9baOGf0bmL6m3EaiPcGy6Fgfm9=jEYBg+GB2@mail.gmail.com>
	<AANLkTinpUymFWTdoU1Xx0oNFNhyRAKx04SmoWy5avVC9@mail.gmail.com>
Message-ID: <1300973635054-6204107.post@n2.nabble.com>


??? wrote:
> Very sorry, the problem has been resolved. Is a problem with my account.
Enjoy! I just did a git pull and recompiled. Now, my FS is up with a
FreeSWITCH Version 1.0.head (git-f3c33c7 2011-03-23 14-57-16 -0500) and it
is happy to serve GV calls through dingaling so far. I will now have more
fun to start playing to mod_ladspa.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Question-about-mod-dingaling-tp6014600p6204107.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jeff at jefflenk.com  Thu Mar 24 16:48:30 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 24 Mar 2011 06:48:30 -0700 (PDT)
Subject: [Freeswitch-users] Gawk.exe crashes when building on Windows 7
	64bits w/ VS
In-Reply-To: <AANLkTimpiqGG7fn98Up9gH65C7vgnmgM5K4Fusk7o-_V@mail.gmail.com>
References: <AANLkTimpiqGG7fn98Up9gH65C7vgnmgM5K4Fusk7o-_V@mail.gmail.com>
Message-ID: <1300974510562-6204157.post@n2.nabble.com>

http://freeswitch-users.2379917.n2.nabble.com/file/n6204157/New_Bitmap_Image.bmp 

See attached image.

I recall an issue with this in the past but I cant find the thread. Try
building with the locale set to english once to generate the sofia gawk
files then you should be able to switch back.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Re-Gawk-exe-crashes-when-building-on-Windows-7-64bits-w-VS-tp6201122p6204157.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From ibc at aliax.net  Thu Mar 24 17:29:54 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Thu, 24 Mar 2011 15:29:54 +0100
Subject: [Freeswitch-users] mod_fifo and changing From header (RFC 4916
	"Connected Identity")
Message-ID: <AANLkTimm8ZXha-ViB-Ut-gWG8PRx=btCUw4ynL=NkR0z@mail.gmail.com>

Hi, if I understand correctly, mod_fifo works by calling initialy to
every agents. Later when an inblund call arrives, FS brigdes such call
into an existing call with an agent. This means that the agent's phone
uses the same SIP dialog for every inblund call to the queue. So, the
agent doesn't know the callerid number of each call.

For this, there is a RFC which provides dynamic From header within the
same dialog. Of course this requires the phone to support it, but it
will occur soon (I hope). The spec is RFC 4916 and basically means
that an endpoint of the dialog can send a re-INVITE or UPDATE within
an existing dialog containing a different From header. The negociation
of this feature is handled by common Supported/Require headers.

http://tools.ietf.org/html/rfc4916

Would it be useful to implement such feature in FS?

Thanks a lot.

-- 
I?aki Baz Castillo
<ibc at aliax.net>


From infos at madovsky.org  Thu Mar 24 18:12:38 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 24 Mar 2011 11:12:38 -0400
Subject: [Freeswitch-users] invite in conference
Message-ID: <910B7BF54F224C2AAB4F4E3D1E69026F@e1705>

I'm experimenting hot invite in conference.
so I used the example on wiki

conference testconf dial {originate_timeout=30}sofia/default/1000 at softswitch 1234567890 FreeSWITCH_Conference

while the conference testconf is running. I noticed that:
-  my TTS pronounces the extension number rather than caller_id_name (if someone enters in conference normally the TTS is right)

question:
- how to hangup the channel in case of answer machine or busy for example while the caller is in conference ?

Thanks

Franck
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From dujinfang at gmail.com  Thu Mar 24 18:19:22 2011
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 24 Mar 2011 23:19:22 +0800
Subject: [Freeswitch-users] Core dump on unimrcp
Message-ID: <465C7B35554247F297EA9C356F0619DB@gmail.com>


I'm using a recent version of FS unimrcp, and tried to connect to a mrcp V1 server from a China vendor. Core dump after the log:

http://pastebin.freeswitch.org/15830



I traced the core and found that in mod_unimrcp.c around line 1824

if (schannel->type == SPEECH_CHANNEL_SYNTHESIZER) {
descriptor = mrcp_application_sink_descriptor_get(channel);
} else {
descriptor = mrcp_application_source_descriptor_get(channel);
}


where mrcp_application_sink_descriptor_get(channel) returns NULL.

configurations is fairly "standard". anyone could help take a look is appreciated. Let me know if I need open a jira with update the the latest version and backtraces.

Thanks.
Seven. 


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From peter.olsson at visionutveckling.se  Thu Mar 24 18:31:57 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 24 Mar 2011 16:31:57 +0100
Subject: [Freeswitch-users] Core dump on unimrcp
In-Reply-To: <465C7B35554247F297EA9C356F0619DB@gmail.com>
References: <465C7B35554247F297EA9C356F0619DB@gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B30F1116@cooper>

Please report this to Jira ? we really need to keep track of the bugs being reported.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Seven Du
Skickat: den 24 mars 2011 16:19
Till: freeswitch-users
?mne: [Freeswitch-users] Core dump on unimrcp


I'm using a recent version of FS unimrcp, and tried to connect to a mrcp V1 server from a China vendor. Core dump after the log:

http://pastebin.freeswitch.org/15830



I traced the core and found that in mod_unimrcp.c around line 1824

                                            if (schannel->type == SPEECH_CHANNEL_SYNTHESIZER) {
                                                                 descriptor = mrcp_application_sink_descriptor_get(channel);
                                            } else {
                                                                 descriptor = mrcp_application_source_descriptor_get(channel);
                                            }


where mrcp_application_sink_descriptor_get(channel) returns NULL.

configurations is fairly "standard". anyone could help take a look is appreciated. Let me know if I need open a jira with update the the latest version and backtraces.

Thanks.
Seven.
!DSPAM:4d8b61a932768348013829!
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From manavid at gmail.com  Thu Mar 24 18:49:28 2011
From: manavid at gmail.com (Moe Navid)
Date: Thu, 24 Mar 2011 08:49:28 -0700
Subject: [Freeswitch-users] Event-lock,
 sync/async on outbound ESL and execute() tagging question
In-Reply-To: <AANLkTin6oFsWUX7gm-gA26RLzi-fuDz7ie9FtMUR0SU5@mail.gmail.com>
References: <AANLkTingnGhvshmDK3TxKSkTpFeno76KTrFULRhbSCiP@mail.gmail.com>
	<AANLkTim9ftv7Q40i2ZhkfH6U8Qk+W5-NjziNt+B6anXW@mail.gmail.com>
	<AANLkTi=uNARtTnWAjuH42129o-LN64J+n26=Reqzzrqy@mail.gmail.com>
	<AANLkTin6oFsWUX7gm-gA26RLzi-fuDz7ie9FtMUR0SU5@mail.gmail.com>
Message-ID: <AANLkTikvTcH5mscmC7q06_VHgmVVXw=6SAG7RHc0vPd3@mail.gmail.com>

I use outbound socket for bridging, I also do my accounting from there too,
As soon as I get the CHANNEL_ANSWER event with the direction:outbound (in
content) I defer a new process and create a timer which does 6 seconds
billing increments.

The trick for not receiving events irrelevant to my session was issuing
following at the beginning of my session:

filter unique-id <uuid>\n\n
filter channel-call-uuid <uuid>\n\n

I wrote my server in Ruby, I'm using EventMachine for TCP handling and Ruby
Fibers to write my apps synchronous fashion while talking to FreeSWITCH in
async mode. Whenever I need to simulate the blocking mode and pause the
execution at some point (with Fibers) I just start listening for the event
that has do be received which indicates the job is done, in case of bridge,
I enqueue the CHANNEL_HANGUP_COMPLETE and resume my app as soon as I receive
it.

Moe

On Wed, Mar 23, 2011 at 1:41 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:

> I understand this. I use outbound async full ESL mode, launching socket app
> from dialplan, because in sync mode I have some problems with bridge app and
> I need to catch parent channel hangup and server disconnection.
>
> I written wrapper around ESL.pm to implement executeSync which waits for
> CHANNEL_EXECUTE_COMPLETE while in async mode, works for now.
> But of course it is not the best way, because of possible error when
> 'execute' is used before my 'executeSync' by mistake and without proper
> event handling so that CHANNEL_EXECUTE_COMPLETE from previous app is catched
> by my wrapper and it returns prematurely. This is why I want to have some
> tag I can mark execute call to filter its events later... Maybe there is
> better way?
>
> 2011/3/23 Michael Collins <msc at freeswitch.org>
>
>>
>>
>> On Wed, Mar 23, 2011 at 6:48 AM, lakshmanan ganapathy <
>> lakindia89 at gmail.com> wrote:
>>
>>> Hi,
>>> Eventlock is used to say "Whether to queue the commands or not".
>>>
>>> Assume I'm giving playback and bridge command via outbound socket to
>>> freeswitch.
>>>
>>> If eventlock is "true" then, first "playback will be executed" and then
>>> only the bridge will be executed.
>>>
>>> If eventlock is "false", then order of execution is not guaranteed.
>>>
>>> I'm also using Outbound event socket. I'll run it in async mode. After
>>> each execute() statement, if I need the output for that event, I will wait
>>> for the event and then only I'll proceed. If I don't need event, ( playback
>>> ), I just do the other operations.
>>>
>>> This works well for me.
>>>
>>> Be sure that you are using the terms "inbound" and "outbound" event
>> socket correctly. It's inbound or outbound from the perspective of
>> FreeSWITCH, not from the perspective of your script. An easy rule of thumb
>> is this: if you use the 'socket' application in your dialplan then you are
>> using "outbound event socket" because FS has to make an outbound connection
>> to your script.
>>
>> http://wiki.freeswitch.org/wiki/Event_socket
>>
>> -MC
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lists at telefaks.de  Thu Mar 24 19:11:48 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Thu, 24 Mar 2011 17:11:48 +0100
Subject: [Freeswitch-users] mod_conference and max-members
Message-ID: <4D8B6D44.4000909@telefaks.de>

Hello,

when we open a conference and set the value of "max-members" to 5 then
only 5 members will be able to join the conference. This is fine.
But just in case the conference moderator wants to invite an additional
person: Is it possible to temporarily disable this function and to add
(invite) another member to the conference?

What we have tried so far:

    * 5 members dialed into the conference
    * changed number of members temporarily to 6 (via XML-Curl)
    * Invited another member into the conference (potential 6th member)
    * Freeswitch requests the new conference.conf XML with "max-members"
      == 6
    * but still it shows "Conference is full" in the logs to the
      potential 6th member

Is there any other way to overcome this, maybe by sending some api
command via event socket?

-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de


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From infos at madovsky.org  Thu Mar 24 19:19:55 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 24 Mar 2011 12:19:55 -0400
Subject: [Freeswitch-users] voicemail sounds format for internal and
	external sip profiles
Message-ID: <355D22D1232E4D9A899DA76966719185@e1705>

Is there a way to configure voicemail sound format to internal and external profiles independently ?
I'd like for example set  ".spx" for inbound calls and ".wav" for outbound calls.

thanks
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From infos at madovsky.org  Thu Mar 24 19:31:24 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 24 Mar 2011 12:31:24 -0400
Subject: [Freeswitch-users] member_id of conference
Message-ID: <A4C72EF1FA19481EBD30DB4E58617E75@e1705>

Is conference <conf> list the only way
to get a specific member_id ?

Thanks much
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From infos at madovsky.org  Thu Mar 24 19:41:07 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 24 Mar 2011 12:41:07 -0400
Subject: [Freeswitch-users] invite in conference
Message-ID: <9D4138A0879448C8B3A2B5F6930FB155@e1705>

for those who need caller_id_name effective in conference from an invite,
it has to specify caller_id_name in brackets

conference testconf dial {originate_timeout=30,caller_id_name='bibi baba'}sofia/default/1000 at softswitch 1234567890 FreeSWITCH_Conference
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, March 24, 2011 11:12 AM
  Subject: invite in conference


  I'm experimenting hot invite in conference.
  so I used the example on wiki

  conference testconf dial {originate_timeout=30}sofia/default/1000 at softswitch 1234567890 FreeSWITCH_Conference

  while the conference testconf is running. I noticed that:
  -  my TTS pronounces the extension number rather than caller_id_name (if someone enters in conference normally the TTS is right)

  question:
  - how to hangup the channel in case of answer machine or busy for example while the caller is in conference ?

  Thanks

  Franck
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From jack at livecall.com  Thu Mar 24 06:05:07 2011
From: jack at livecall.com (Jack)
Date: Wed, 23 Mar 2011 20:05:07 -0700
Subject: [Freeswitch-users] Sending Custom Session State
Message-ID: <4D8AB4E3.4030705@livecall.com>

I would like to send a custom session state to my User Agent that I can 
capture and react to with my user agent.
Is there a way to send something like an extension number instead of the 
ACTIVE or INVITE etc..... ?


From yivzhenko at mksat.net  Thu Mar 24 17:44:27 2011
From: yivzhenko at mksat.net (Yuriy Ivzhenko)
Date: Thu, 24 Mar 2011 16:44:27 +0200
Subject: [Freeswitch-users] mod_fifo & CID
Message-ID: <201103241644.27845.yivzhenko@mksat.net>

Hi All,

After upgrading to last Git (git-73ca862) i see changes in CID on
calls to agents, generated by mod_fifo.

If agent answer
caller_profile.caller_id_name = "Outbound Call"
caller_profile.caller_id_number = <agent number> 

If agent not answer 
caller_profile.caller_id_name = caller caller_id_name
caller_profile.caller_id_number = caller caller_id_number (or 
origination_caller_id_name/number of set in agents dialstring)
In old version, so it was always.

Is this correct?
It seems to me it should not depend on, answered the call or not.



From msc at freeswitch.org  Thu Mar 24 20:42:06 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 24 Mar 2011 10:42:06 -0700
Subject: [Freeswitch-users] I am trying to use ESL with LUA - undefined
 symbol: _ZN8ESLeventC1EPKcS1_
In-Reply-To: <4D8B368B.7050909@directtel.ru>
References: <4D8B368B.7050909@directtel.ru>
Message-ID: <AANLkTinLH5u=W0eJSZmDvhWNjpG6Xr45YOE38w+ZA_2M@mail.gmail.com>

>
>
> /usr/local/bin/lua: error loading module 'ESL' from file
>

Did you successfully build both esl and the Ruby mod?
cd libs/esl
make
make rubymod

-MC
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From frankie.k.yiu at gmail.com  Thu Mar 24 20:44:48 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Thu, 24 Mar 2011 10:44:48 -0700
Subject: [Freeswitch-users] Gawk.exe crashes when building on Windows 7
	64bits w/ VS
Message-ID: <AANLkTi=HpEJqm=Sfu7UzixYJpCkOvp8Fd-0xSVqYTHBk@mail.gmail.com>

I still got the compiliation error of Gawk.exe.  Could that be other
setting?

Thanks,
Frankie

---------- Forwarded message ----------
From: Jeff Lenk <jeff at jefflenk.com>
To: freeswitch-users at lists.freeswitch.org
Date: Thu, 24 Mar 2011 06:48:30 -0700 (PDT)
Subject: Re: [Freeswitch-users] Gawk.exe crashes when building on Windows 7
64bits w/ VS
http://freeswitch-users.2379917.n2.nabble.com/file/n6204157/New_Bitmap_Image.bmp

See attached image.

I recall an issue with this in the past but I cant find the thread. Try
building with the locale set to english once to generate the sofia gawk
files then you should be able to switch back.

--
View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/Re-Gawk-exe-crashes-when-building-on-Windows-7-64bits-w-VS-tp6201122p6204157.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
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From msc at freeswitch.org  Thu Mar 24 20:50:33 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 24 Mar 2011 10:50:33 -0700
Subject: [Freeswitch-users] member_id of conference
In-Reply-To: <A4C72EF1FA19481EBD30DB4E58617E75@e1705>
References: <A4C72EF1FA19481EBD30DB4E58617E75@e1705>
Message-ID: <AANLkTinLw86HQM8SSHcrw5j_3K4VP6XB4nX+C53f9+Jh@mail.gmail.com>

On Thu, Mar 24, 2011 at 9:31 AM, Madovsky <infos at madovsky.org> wrote:

>  Is conference <conf> list the only way
> to get a specific member_id ?
>
What other method did you have in mind?
-MC
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From manavid at gmail.com  Thu Mar 24 20:39:15 2011
From: manavid at gmail.com (Moe Navid)
Date: Thu, 24 Mar 2011 10:39:15 -0700
Subject: [Freeswitch-users] Event Lock and Bind Meta App
Message-ID: <AANLkTikc0YueMabrnHaNck7Bu7ydOOyB=21FZ5m+AJMV@mail.gmail.com>

Hi all,

I noticed when bridging the call in event socket and adding event-lock:true,
the lock prevents the bind_meta_app to get executed while on the call it
waits till bridge is done and then executes. Any suggestions or work
arounds?

Thanks

Moe
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From Nabble at slickdeals.endjunk.com  Thu Mar 24 21:03:06 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 24 Mar 2011 11:03:06 -0700 (PDT)
Subject: [Freeswitch-users] I am trying to use ESL with LUA - undefined
 symbol: _ZN8ESLeventC1EPKcS1_
In-Reply-To: <AANLkTinLH5u=W0eJSZmDvhWNjpG6Xr45YOE38w+ZA_2M@mail.gmail.com>
References: <4D8B368B.7050909@directtel.ru>
	<AANLkTinLH5u=W0eJSZmDvhWNjpG6Xr45YOE38w+ZA_2M@mail.gmail.com>
Message-ID: <1300989786193-6205132.post@n2.nabble.com>


mercutioviz wrote:
> 
> &gt;
> &gt;
> &gt; /usr/local/bin/lua: error loading module 'ESL' from file
> &gt;
> 
> Did you successfully build both esl and the Ruby mod?
> cd libs/esl
> make
> make rubymod
Doesn't make all or make current on FS root will take care of that?

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/I-am-trying-to-use-ESL-with-LUA-undefined-symbol-ZN8ESLeventC1EPKcS1-tp6203883p6205132.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kbdfck at gmail.com  Thu Mar 24 21:11:58 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Thu, 24 Mar 2011 21:11:58 +0300
Subject: [Freeswitch-users] Event-lock,
 sync/async on outbound ESL and execute() tagging question
In-Reply-To: <AANLkTikvTcH5mscmC7q06_VHgmVVXw=6SAG7RHc0vPd3@mail.gmail.com>
References: <AANLkTingnGhvshmDK3TxKSkTpFeno76KTrFULRhbSCiP@mail.gmail.com>
	<AANLkTim9ftv7Q40i2ZhkfH6U8Qk+W5-NjziNt+B6anXW@mail.gmail.com>
	<AANLkTi=uNARtTnWAjuH42129o-LN64J+n26=Reqzzrqy@mail.gmail.com>
	<AANLkTin6oFsWUX7gm-gA26RLzi-fuDz7ie9FtMUR0SU5@mail.gmail.com>
	<AANLkTikvTcH5mscmC7q06_VHgmVVXw=6SAG7RHc0vPd3@mail.gmail.com>
Message-ID: <AANLkTinUAo2bUEtmHEz09U3rvxb=c1D-Jv3bZ+gp6GR-@mail.gmail.com>

I don't get any irrelevant channel events, it's my events or my child
channel events.
I want to distinguish events from previously launched execute() commands in
my event loop, like job id in bgapi. These events can be from my own
original channel as well, so uuid filter doesn't help me.


2011/3/24 Moe Navid <manavid at gmail.com>

> I use outbound socket for bridging, I also do my accounting from there too,
> As soon as I get the CHANNEL_ANSWER event with the direction:outbound (in
> content) I defer a new process and create a timer which does 6 seconds
> billing increments.
>
> The trick for not receiving events irrelevant to my session was issuing
> following at the beginning of my session:
>
> filter unique-id <uuid>\n\n
> filter channel-call-uuid <uuid>\n\n
>
> I wrote my server in Ruby, I'm using EventMachine for TCP handling and Ruby
> Fibers to write my apps synchronous fashion while talking to FreeSWITCH in
> async mode. Whenever I need to simulate the blocking mode and pause the
> execution at some point (with Fibers) I just start listening for the event
> that has do be received which indicates the job is done, in case of bridge,
> I enqueue the CHANNEL_HANGUP_COMPLETE and resume my app as soon as I receive
> it.
>
> Moe
>
>
> On Wed, Mar 23, 2011 at 1:41 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>
>> I understand this. I use outbound async full ESL mode, launching socket
>> app from dialplan, because in sync mode I have some problems with bridge app
>> and I need to catch parent channel hangup and server disconnection.
>>
>> I written wrapper around ESL.pm to implement executeSync which waits for
>> CHANNEL_EXECUTE_COMPLETE while in async mode, works for now.
>> But of course it is not the best way, because of possible error when
>> 'execute' is used before my 'executeSync' by mistake and without proper
>> event handling so that CHANNEL_EXECUTE_COMPLETE from previous app is catched
>> by my wrapper and it returns prematurely. This is why I want to have some
>> tag I can mark execute call to filter its events later... Maybe there is
>> better way?
>>
>> 2011/3/23 Michael Collins <msc at freeswitch.org>
>>
>>>
>>>
>>> On Wed, Mar 23, 2011 at 6:48 AM, lakshmanan ganapathy <
>>> lakindia89 at gmail.com> wrote:
>>>
>>>> Hi,
>>>> Eventlock is used to say "Whether to queue the commands or not".
>>>>
>>>> Assume I'm giving playback and bridge command via outbound socket to
>>>> freeswitch.
>>>>
>>>> If eventlock is "true" then, first "playback will be executed" and then
>>>> only the bridge will be executed.
>>>>
>>>> If eventlock is "false", then order of execution is not guaranteed.
>>>>
>>>> I'm also using Outbound event socket. I'll run it in async mode. After
>>>> each execute() statement, if I need the output for that event, I will wait
>>>> for the event and then only I'll proceed. If I don't need event, ( playback
>>>> ), I just do the other operations.
>>>>
>>>> This works well for me.
>>>>
>>>> Be sure that you are using the terms "inbound" and "outbound" event
>>> socket correctly. It's inbound or outbound from the perspective of
>>> FreeSWITCH, not from the perspective of your script. An easy rule of thumb
>>> is this: if you use the 'socket' application in your dialplan then you are
>>> using "outbound event socket" because FS has to make an outbound connection
>>> to your script.
>>>
>>> http://wiki.freeswitch.org/wiki/Event_socket
>>>
>>> -MC
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Best regards,
>>
>> Dmitry Sytchev,
>> IT Engineer
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From msc at freeswitch.org  Thu Mar 24 21:26:09 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 24 Mar 2011 11:26:09 -0700
Subject: [Freeswitch-users] I am trying to use ESL with LUA - undefined
 symbol: _ZN8ESLeventC1EPKcS1_
In-Reply-To: <1300989786193-6205132.post@n2.nabble.com>
References: <4D8B368B.7050909@directtel.ru>
	<AANLkTinLH5u=W0eJSZmDvhWNjpG6Xr45YOE38w+ZA_2M@mail.gmail.com>
	<1300989786193-6205132.post@n2.nabble.com>
Message-ID: <AANLkTi=8zq5Dguug4o5RjdW9WL8c+G4AsrhPnpSE_E04@mail.gmail.com>

On Thu, Mar 24, 2011 at 11:03 AM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
> mercutioviz wrote:
> >
> > &gt;
> > &gt;
> > &gt; /usr/local/bin/lua: error loading module 'ESL' from file
> > &gt;
> >
> > Did you successfully build both esl and the Ruby mod?
> > cd libs/esl
> > make
> > make rubymod
> Doesn't make all or make current on FS root will take care of that?
>
It looks like the make install process takes care of ESL's make but not the
individual module's makes.
-MC
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From manavid at gmail.com  Thu Mar 24 21:27:08 2011
From: manavid at gmail.com (Mohammad Amin)
Date: Thu, 24 Mar 2011 11:27:08 -0700
Subject: [Freeswitch-users] Event-lock,
	sync/async on outbound ESL and execute() tagging question
In-Reply-To: <AANLkTinUAo2bUEtmHEz09U3rvxb=c1D-Jv3bZ+gp6GR-@mail.gmail.com>
References: <AANLkTingnGhvshmDK3TxKSkTpFeno76KTrFULRhbSCiP@mail.gmail.com>
	<AANLkTim9ftv7Q40i2ZhkfH6U8Qk+W5-NjziNt+B6anXW@mail.gmail.com>
	<AANLkTi=uNARtTnWAjuH42129o-LN64J+n26=Reqzzrqy@mail.gmail.com>
	<AANLkTin6oFsWUX7gm-gA26RLzi-fuDz7ie9FtMUR0SU5@mail.gmail.com>
	<AANLkTikvTcH5mscmC7q06_VHgmVVXw=6SAG7RHc0vPd3@mail.gmail.com>
	<AANLkTinUAo2bUEtmHEz09U3rvxb=c1D-Jv3bZ+gp6GR-@mail.gmail.com>
Message-ID: <883D3DA8-9A8F-4E44-A0CD-93833172C48D@gmail.com>

I have hash table which keeps track of api & dp applications I have executed to process the events & responses I get from FS, based on that I control the flow of execution in my application.

For instance for the 'read' application my server returns the result  variable as soon as It receives CHANNEL_EXECUTE_COMPLETE with the 'application: read' in it's content.

The fired events are very self explanatory it has all the info you need to control the execution flow. If you watch for the 'application' in the content of CHANNEL_EXECUTE_COMPLETE and you keep track of what applications you have executed so far you can map them very easily.

Moe

On Mar 24, 2011, at 11:11 AM, Dmitry Sytchev wrote:

> I don't get any irrelevant channel events, it's my events or my child channel events.
> I want to distinguish events from previously launched execute() commands in my event loop, like job id in bgapi. These events can be from my own original channel as well, so uuid filter doesn't help me.
> 
> 
> 2011/3/24 Moe Navid <manavid at gmail.com>
> I use outbound socket for bridging, I also do my accounting from there too, As soon as I get the CHANNEL_ANSWER event with the direction:outbound (in content) I defer a new process and create a timer which does 6 seconds billing increments.
> 
> The trick for not receiving events irrelevant to my session was issuing following at the beginning of my session:
> 
> filter unique-id <uuid>\n\n
> filter channel-call-uuid <uuid>\n\n
> 
> I wrote my server in Ruby, I'm using EventMachine for TCP handling and Ruby Fibers to write my apps synchronous fashion while talking to FreeSWITCH in async mode. Whenever I need to simulate the blocking mode and pause the execution at some point (with Fibers) I just start listening for the event that has do be received which indicates the job is done, in case of bridge, I enqueue the CHANNEL_HANGUP_COMPLETE and resume my app as soon as I receive it. 
> 
> Moe
> 
> 
> On Wed, Mar 23, 2011 at 1:41 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> I understand this. I use outbound async full ESL mode, launching socket app from dialplan, because in sync mode I have some problems with bridge app and I need to catch parent channel hangup and server disconnection.
> 
> I written wrapper around ESL.pm to implement executeSync which waits for CHANNEL_EXECUTE_COMPLETE while in async mode, works for now.
> But of course it is not the best way, because of possible error when 'execute' is used before my 'executeSync' by mistake and without proper event handling so that CHANNEL_EXECUTE_COMPLETE from previous app is catched by my wrapper and it returns prematurely. This is why I want to have some tag I can mark execute call to filter its events later... Maybe there is better way?
> 
> 2011/3/23 Michael Collins <msc at freeswitch.org>
> 
> 
> On Wed, Mar 23, 2011 at 6:48 AM, lakshmanan ganapathy <lakindia89 at gmail.com> wrote:
> Hi,
> Eventlock is used to say "Whether to queue the commands or not".
> 
> Assume I'm giving playback and bridge command via outbound socket to freeswitch.
> 
> If eventlock is "true" then, first "playback will be executed" and then only the bridge will be executed.
> 
> If eventlock is "false", then order of execution is not guaranteed. 
> 
> I'm also using Outbound event socket. I'll run it in async mode. After each execute() statement, if I need the output for that event, I will wait for the event and then only I'll proceed. If I don't need event, ( playback ), I just do the other operations.
> 
> This works well for me.
> 
> Be sure that you are using the terms "inbound" and "outbound" event socket correctly. It's inbound or outbound from the perspective of FreeSWITCH, not from the perspective of your script. An easy rule of thumb is this: if you use the 'socket' application in your dialplan then you are using "outbound event socket" because FS has to make an outbound connection to your script.
> 
> http://wiki.freeswitch.org/wiki/Event_socket
> 
> -MC
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Best regards,
> 
> Dmitry Sytchev,
> IT Engineer
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Best regards,
> 
> Dmitry Sytchev,
> IT Engineer
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jeff at jefflenk.com  Thu Mar 24 21:28:24 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 24 Mar 2011 11:28:24 -0700 (PDT)
Subject: [Freeswitch-users] Gawk.exe crashes when building on Windows 7
	64bits w/ VS
In-Reply-To: <AANLkTi=HpEJqm=Sfu7UzixYJpCkOvp8Fd-0xSVqYTHBk@mail.gmail.com>
References: <AANLkTi=HpEJqm=Sfu7UzixYJpCkOvp8Fd-0xSVqYTHBk@mail.gmail.com>
Message-ID: <1300991304784-6205227.post@n2.nabble.com>

Kinda out of options must be something specific to your system. Are you
running a virus scanner? try disabling that.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Gawk-exe-crashes-when-building-on-Windows-7-64bits-w-VS-tp6205075p6205227.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Thu Mar 24 21:29:15 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 24 Mar 2011 11:29:15 -0700
Subject: [Freeswitch-users] voicemail sounds format for internal and
 external sip profiles
In-Reply-To: <355D22D1232E4D9A899DA76966719185@e1705>
References: <355D22D1232E4D9A899DA76966719185@e1705>
Message-ID: <AANLkTiksS0jXV5L_4BxfWM4Y+b9gs6PHoAMNBaweSGt_@mail.gmail.com>

On Thu, Mar 24, 2011 at 9:19 AM, Madovsky <infos at madovsky.org> wrote:

>  Is there a way to configure voicemail sound format to internal and
> external profiles independently ?
> I'd like for example set  ".spx" for inbound calls and ".wav" for outbound
> calls.
>
I may be missing something here, but how can you send outbound calls to your
own voicemail system?

-MC
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From msc at freeswitch.org  Thu Mar 24 21:37:25 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 24 Mar 2011 11:37:25 -0700
Subject: [Freeswitch-users] mod_fifo & CID
In-Reply-To: <201103241644.27845.yivzhenko@mksat.net>
References: <201103241644.27845.yivzhenko@mksat.net>
Message-ID: <AANLkTim6S7kh=g-kSpT3x5jtNhS5-BHmitdf8o7FyYW9@mail.gmail.com>

How old was your previous FS? The default mod_fifo behavior changed a few
months back.
-MC

On Thu, Mar 24, 2011 at 7:44 AM, Yuriy Ivzhenko <yivzhenko at mksat.net> wrote:

> Hi All,
>
> After upgrading to last Git (git-73ca862) i see changes in CID on
> calls to agents, generated by mod_fifo.
>
> If agent answer
> caller_profile.caller_id_name = "Outbound Call"
> caller_profile.caller_id_number = <agent number>
>
> If agent not answer
> caller_profile.caller_id_name = caller caller_id_name
> caller_profile.caller_id_number = caller caller_id_number (or
> origination_caller_id_name/number of set in agents dialstring)
> In old version, so it was always.
>
> Is this correct?
> It seems to me it should not depend on, answered the call or not.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Mar 24 21:53:19 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 24 Mar 2011 11:53:19 -0700
Subject: [Freeswitch-users] mod_conference and max-members
In-Reply-To: <4D8B6D44.4000909@telefaks.de>
References: <4D8B6D44.4000909@telefaks.de>
Message-ID: <AANLkTinEZooEw9rhcHwYVAyjyJ-+p8xBZsxcCKmt8ouP@mail.gmail.com>

I don't believe you can change the parameters from a conference profile for
an active conference. However, if you are using the event socket and have
some sort of conference manager then you can use the "conference lock" and
"conference unlock" APIs to accomplish the same purpose. When the conference
gets to X number of users your conf manager program can issue the lock
command. The moderator can then lock or unlock as he/she sees fit.

-MC

On Thu, Mar 24, 2011 at 9:11 AM, Peter Steinbach <lists at telefaks.de> wrote:

>  Hello,
>
> when we open a conference and set the value of "max-members" to 5 then only
> 5 members will be able to join the conference. This is fine.
> But just in case the conference moderator wants to invite an additional
> person: Is it possible to temporarily disable this function and to add
> (invite) another member to the conference?
>
> What we have tried so far:
>
>    - 5 members dialed into the conference
>    - changed number of members temporarily to 6 (via XML-Curl)
>    - Invited another member into the conference (potential 6th member)
>     - Freeswitch requests the new conference.conf XML with "max-members"
>    == 6
>     - but still it shows "Conference is full" in the logs to the potential
>    6th member
>
> Is there any other way to overcome this, maybe by sending some api command
> via event socket?
>
> --
> With kind regards
> Peter Steinbach
>
> Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de
> Internet: www.telefaks.de
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Thu Mar 24 21:59:34 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 24 Mar 2011 14:59:34 -0400
Subject: [Freeswitch-users] voicemail sounds format for internal and
	external sip profiles
References: <355D22D1232E4D9A899DA76966719185@e1705>
	<AANLkTiksS0jXV5L_4BxfWM4Y+b9gs6PHoAMNBaweSGt_@mail.gmail.com>
Message-ID: <DAE04DA7346B4433A54E97B6E0CCB971@e1705>

sorry, I meant incoming calls from external profiles ;)
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Thursday, March 24, 2011 2:29 PM
  Subject: Re: [Freeswitch-users] voicemail sounds format for internal and external sip profiles





  On Thu, Mar 24, 2011 at 9:19 AM, Madovsky <infos at madovsky.org> wrote:

    Is there a way to configure voicemail sound format to internal and external profiles independently ?
    I'd like for example set  ".spx" for inbound calls and ".wav" for outbound calls.
  I may be missing something here, but how can you send outbound calls to your own voicemail system?


  -MC


------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From all.eforums at gmail.com  Thu Mar 24 22:03:14 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Thu, 24 Mar 2011 15:03:14 -0400
Subject: [Freeswitch-users] Bug ID: FS-2868 Status?
In-Reply-To: <AANLkTinhDPN8VaHUYmSktvcAiV-bsU47j_A1JhMQuMOf@mail.gmail.com>
References: <AANLkTinhDPN8VaHUYmSktvcAiV-bsU47j_A1JhMQuMOf@mail.gmail.com>
Message-ID: <AANLkTi=jBap-X-rcGfV7apvK7M-e3rNRnMpBfTPu_vZv@mail.gmail.com>

On Thu, Mar 24, 2011 at 9:22 AM, Kim Culhan <w8hdkim at gmail.com> wrote:

>
> On Wed, March 23, 2011 9:15 am, Michal Bielicki wrote:
> > The problem is a bug in solaris perl development package that I haven't
> been doing
> > anything about since I do not use perl for roughly a decade.
> > Its not the hash module, its the perl module.
> >
> > Am 23.03.2011 um 11:44 schrieb A E [Gmail]:
> >
> >> Hello FreeSwitchers!
> >>
> >> Am attempting to compile the latest (off of Git) code for FS on Solaris
> 11 Express
> >> on an 64-Bit x86 machine. However, I'm running into the error that Bruce
> >> McAllister has reported on 23rd Nov 2010. The bug is still sitting
> unresolved. So
> >> I'm assuming there's a backdoor trick to fixing this OR that some other
> version of
> >> FS code doesn't have this issue. Could someone please let me on the
> secret on
> >> which build compiles on x86 Solaris v10+ and/or what the backdoor trick
> to bypass
> >> this is? Should I not compile the hash module where this bug seems to be
> >> originating from?
>
> I'm seeing the same error here with the perl module commented-out in
> modules.conf
>
> Please describe the workaround you are using to avoid the problem caused
> by the perl module.
>
> thanks
> -kim
>
>
Hello Kim,

for now, I've just commented out mod_hash in modules.conf to see if I can
compile the rest of it successfully. But there's a LOT of problems along the
way. So I will come back to this later. I don't want to open a Jira on it as
it's already there. However, if you're not getting this with mod_hash but
something else, then you will have to look carefully what is causing it. I
have noticed other places where the make process is crashing due
to unrecognised options being passed to the Solaris Studio express CC
compiler. So I will probably open a Jira on other issues I have encountered.
I'm still hoping Michal would reply and let us know how he is getting it to
compile every time. For now, I'm just editing the Makefile for each of such
application/module which is encountering this problem so I can keep moving
forward

HTH
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From msc at freeswitch.org  Thu Mar 24 22:03:10 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 24 Mar 2011 12:03:10 -0700
Subject: [Freeswitch-users] voicemail sounds format for internal and
 external sip profiles
In-Reply-To: <DAE04DA7346B4433A54E97B6E0CCB971@e1705>
References: <355D22D1232E4D9A899DA76966719185@e1705>
	<AANLkTiksS0jXV5L_4BxfWM4Y+b9gs6PHoAMNBaweSGt_@mail.gmail.com>
	<DAE04DA7346B4433A54E97B6E0CCB971@e1705>
Message-ID: <AANLkTikd0MtGOhyTmThnXRtaDhn-rJO4r-3XGstDH5rw@mail.gmail.com>

On Thu, Mar 24, 2011 at 11:59 AM, Madovsky <infos at madovsky.org> wrote:

>  sorry, I meant incoming calls from external profiles ;)
>
okay, that makes more sense.

All I can think to try is creating a second voicemail profile in
conf/autoload_configs/voicemail.conf.xml and using that for the external
calls. Give it a try and see what happens...

-MC
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From infos at madovsky.org  Thu Mar 24 22:05:44 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 24 Mar 2011 15:05:44 -0400
Subject: [Freeswitch-users] member_id of conference
References: <A4C72EF1FA19481EBD30DB4E58617E75@e1705>
	<AANLkTinLw86HQM8SSHcrw5j_3K4VP6XB4nX+C53f9+Jh@mail.gmail.com>
Message-ID: <BC3FD7C9CDE244CFB6A3A30EC360E12A@e1705>

for example the conferencer invites 9999999999
and does it from fs_cli. the return string of conference dial command is
the status of call after success or failure (seems that the command is synchronous)
but not uuid information nor id_member in return. So maybe have the member_id in the
return string would be useful in case of the call fails in a not useful answered call like answer machine
or "caller is unreacheable" from telcom.
as this it will avoid to run "conference list", parse the result and find the right member_id.

what folks do you think ?

thanks
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Thursday, March 24, 2011 1:50 PM
  Subject: Re: [Freeswitch-users] member_id of conference





  On Thu, Mar 24, 2011 at 9:31 AM, Madovsky <infos at madovsky.org> wrote:

    Is conference <conf> list the only way
    to get a specific member_id ?
  What other method did you have in mind?
  -MC




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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Thu Mar 24 22:07:16 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 24 Mar 2011 15:07:16 -0400
Subject: [Freeswitch-users] voicemail sounds format for internal and
	external sip profiles
References: <355D22D1232E4D9A899DA76966719185@e1705><AANLkTiksS0jXV5L_4BxfWM4Y+b9gs6PHoAMNBaweSGt_@mail.gmail.com><DAE04DA7346B4433A54E97B6E0CCB971@e1705>
	<AANLkTikd0MtGOhyTmThnXRtaDhn-rJO4r-3XGstDH5rw@mail.gmail.com>
Message-ID: <B6DF963850B647DD9B3186F38929F4CD@e1705>

nice ok will do it tonight
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Thursday, March 24, 2011 3:03 PM
  Subject: Re: [Freeswitch-users] voicemail sounds format for internal and external sip profiles





  On Thu, Mar 24, 2011 at 11:59 AM, Madovsky <infos at madovsky.org> wrote:

    sorry, I meant incoming calls from external profiles ;)
  okay, that makes more sense.


  All I can think to try is creating a second voicemail profile in conf/autoload_configs/voicemail.conf.xml and using that for the external calls. Give it a try and see what happens...


  -MC


------------------------------------------------------------------------------


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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From msc at freeswitch.org  Thu Mar 24 22:32:53 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 24 Mar 2011 12:32:53 -0700
Subject: [Freeswitch-users] member_id of conference
In-Reply-To: <BC3FD7C9CDE244CFB6A3A30EC360E12A@e1705>
References: <A4C72EF1FA19481EBD30DB4E58617E75@e1705>
	<AANLkTinLw86HQM8SSHcrw5j_3K4VP6XB4nX+C53f9+Jh@mail.gmail.com>
	<BC3FD7C9CDE244CFB6A3A30EC360E12A@e1705>
Message-ID: <AANLkTin8NvtCdSiap2AGLPski-ry=ZR71ukMDN_v4HWm@mail.gmail.com>

On Thu, Mar 24, 2011 at 12:05 PM, Madovsky <infos at madovsky.org> wrote:

>  for example the conferencer invites 9999999999
> and does it from fs_cli. the return string of conference dial command is
> the status of call after success or failure (seems that the command is
> synchronous)
> but not uuid information nor id_member in return. So maybe have the
> member_id in the
> return string would be useful in case of the call fails in a not useful
> answered call like answer machine
> or "caller is unreacheable" from telcom.
> as this it will avoid to run "conference list", parse the result and find
> the right member_id.
>
> what folks do you think ?
>
Well, you have the uuid of the call, correct? You can get the member_id with
uuid_getvar:

uuid_getvar <uuid> member_id

If there is no member_id then your call never made it into the conference.
 BTW, we recently added conference_uuid channel variable so you can even see
which exact instance of the conference that the user was/is connected to.

-MC
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From msc at freeswitch.org  Thu Mar 24 22:44:15 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 24 Mar 2011 12:44:15 -0700
Subject: [Freeswitch-users] answering calls weirdness
In-Reply-To: <AANLkTin5mP7QzDe+CAvaPjsbxhq934P-49LARGQJwR8Y@mail.gmail.com>
References: <AANLkTin5mP7QzDe+CAvaPjsbxhq934P-49LARGQJwR8Y@mail.gmail.com>
Message-ID: <AANLkTin8nv3N=XsPa=jepebenxRJ4EvtNSMJ3D37KZKu@mail.gmail.com>

Can you drop this into pastebin? pastebin.freeswitch.org. Also include the
dialplan that is running all this. Use the "freeswitch log" syntax
highlighting because it makes it much easier to read.

-MC

On Wed, Mar 23, 2011 at 4:49 PM, Brian May
<brian at microcomaustralia.com.au>wrote:

> When freeswitch answers the call, everything seems fine at first, but
> then it redirects the call to the magic number of 6201 which fails
> because I haven't defined this number anywhere. For an incoming SIP
> call the call is hang up after the first ring. For an incoming call
> from FreeDTM, callerid doesn't work, and the phone keeps ringing
> randomly until well after the caller has hang up. As it a result it is
> not possible to answer the incoming call.
>
> At first I thought maybe a problem with my analog telephone line, I
> think I have ruled this out now that I see similar logs for SIP calls.
>
> What is weird is that I haven't made any changes to the setup, and
> everything was working fine until yesterday.
>
> Here is an excerpt from my logs for an incoming SIP call. Where is
> this magic number of 6201 coming from? I did a grep of all my config
> files and came up with nothing.
>
> Last night I rebooted the box, it worked for one call, then failed for
> subsequent calls.
>
>
> Any ideas??
>
>
> Thanks
>
>
>
> 2011-03-24 10:31:22.061336 [DEBUG] mod_dptools.c:854
> sofia/internal/1004 at microcomaustralia.com.au SET
> [dialed_extension]=[2000]
> EXECUTE sofia/internal/1004 at microcomaustralia.com.au
> export(dialed_extension=2000)
> 2011-03-24 10:31:22.080434 [DEBUG] mod_dptools.c:938 EXPORT
> [dialed_extension]=[2000]
> EXECUTE sofia/internal/1004 at microcomaustralia.com.au bind_meta_app(1 b
> s execute_extension::dx XML features)
> 2011-03-24 10:31:22.095423 [INFO] switch_ivr_async.c:2464 Bound B-Leg:
> *1 execute_extension::dx XML features
> EXECUTE sofia/internal/1004 at microcomaustralia.com.au bind_meta_app(2 b
> s record_session::/opt/freeswitch/recordings/1004.2011-03-24-10-31-22.wav)
> 2011-03-24 10:31:22.116351 [INFO] switch_ivr_async.c:2464 Bound B-Leg:
> *2 record_session::/opt/freeswitch/recordings/1004.2011-03-24-10-31-22.wav
> EXECUTE sofia/internal/1004 at microcomaustralia.com.au bind_meta_app(3 b
> s execute_extension::cf XML features)
> 2011-03-24 10:31:22.130335 [INFO] switch_ivr_async.c:2464 Bound B-Leg:
> *3 execute_extension::cf XML features
> EXECUTE sofia/internal/1004 at microcomaustralia.com.au set(ringback=)
> 2011-03-24 10:31:22.141503 [DEBUG] mod_dptools.c:854
> sofia/internal/1004 at microcomaustralia.com.au SET [ringback]=[UNDEF]
> EXECUTE sofia/internal/1004 at microcomaustralia.com.au
> set(transfer_ringback=local_stream://moh)
> 2011-03-24 10:31:22.164352 [DEBUG] mod_dptools.c:854
> sofia/internal/1004 at microcomaustralia.com.au SET
> [transfer_ringback]=[local_stream://moh]
> EXECUTE sofia/internal/1004 at microcomaustralia.com.au set(call_timeout=30)
> 2011-03-24 10:31:22.179416 [DEBUG] mod_dptools.c:854
> sofia/internal/1004 at microcomaustralia.com.au SET [call_timeout]=[30]
> EXECUTE sofia/internal/1004 at microcomaustralia.com.au
> set(hangup_after_bridge=true)
> 2011-03-24 10:31:22.192366 [DEBUG] mod_dptools.c:854
> sofia/internal/1004 at microcomaustralia.com.au SET
> [hangup_after_bridge]=[true]
> EXECUTE sofia/internal/1004 at microcomaustralia.com.auset(continue_on_fail=true)
> 2011-03-24 10:31:22.206368 [DEBUG] mod_dptools.c:854
> sofia/internal/1004 at microcomaustralia.com.au SET
> [continue_on_fail]=[true]
> EXECUTE sofia/internal/1004 at microcomaustralia.com.au
> bridge([presence_id=1000 at microcomaustralia.com.au
> ]error/user_not_registered,FreeTDM/1/1,[presence_id=
> 1001 at microcomaustralia.com.au
> ]error/user_not_registered,FreeTDM/2/1,[presence_id=
> 1002 at microcomaustralia.com.au]sofia/internal/sip:1002 at 192.168.2.13:5060
> ,[presence_id=1003 at microcomaustralia.com.au]sofia/internal/
> sip:1003 at 192.168.2.13:5061,[presence_id=1004 at microcomaustralia.com.au
> ]sofia/internal/sip:1004 at 192.168.2.16)
>
> Everything fine up to (and beyond) this point.
>
>
> 2011-03-24 10:31:22.374349 [DEBUG] switch_ivr_originate.c:2599 local
> variable string 0 = [presence_id=1000 at microcomaustralia.com.au]
> 2011-03-24 10:31:22.374349 [ERR] switch_ivr_originate.c:2632 Cannot
> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> 2011-03-24 10:31:22.380323 [INFO] ftmod_zt.c:636 Setting echo cancel
> to 64 taps for 1:1
> 2011-03-24 10:31:22.381586 [DEBUG] mod_freetdm.c:377 Set codec PCMU 20ms
> 2011-03-24 10:31:22.393312 [DEBUG] mod_freetdm.c:1334 Connect outbound
> channel FreeTDM/1:1/
> 2011-03-24 10:31:22.393312 [NOTICE] switch_channel.c:779 New Channel
> FreeTDM/1:1/ [d1b1124a-e8cf-485a-8c1e-dafbcaf0d09b]
> 2011-03-24 10:31:22.399504 [DEBUG] mod_freetdm.c:1348 (FreeTDM/1:1/)
> State Change CS_NEW -> CS_INIT
> 2011-03-24 10:31:22.399504 [DEBUG] switch_core_session.c:1047 Send
> signal FreeTDM/1:1/ [BREAK]
> 2011-03-24 10:31:22.399504 [DEBUG] ftmod_analog.c:361 [s1c1][1:1]
> ANALOG CHANNEL thread starting.
> 2011-03-24 10:31:22.404683 [DEBUG] ftmod_analog.c:88 [s1c1][1:1]
> Changed state from DOWN to GENRING
> 2011-03-24 10:31:23.514411 [INFO] ftmod_zt.c:636 Setting echo cancel
> to 64 taps for 1:1
> 2011-03-24 10:31:23.514411 [DEBUG] ftdm_io.c:2539 [s1c1][1:1] Enabled
> software DTMF detector
> 2011-03-24 10:31:23.514411 [DEBUG] ftmod_analog.c:381 [s1c1][1:1]
> Initialized DTMF detection
> 2011-03-24 10:31:23.514411 [DEBUG] ftmod_analog.c:534 [s1c1][1:1]
> Executing state handler on 1:1 for GENRING
> 2011-03-24 10:31:23.514411 [DEBUG] mod_freetdm.c:1715 got FXS sig
> [PROGRESS]
> 2011-03-24 10:31:23.514411 [NOTICE] mod_freetdm.c:1732 Ring-Ready
> FreeTDM/1:1/!
> 2011-03-24 10:31:23.525898 [DEBUG] switch_core_state_machine.c:314
> (FreeTDM/1:1/) Running State Change CS_INIT
> 2011-03-24 10:31:23.525898 [DEBUG] switch_core_state_machine.c:338
> (FreeTDM/1:1/) State INIT
> 2011-03-24 10:31:23.525898 [DEBUG] mod_freetdm.c:405 (FreeTDM/1:1/)
> State Change CS_INIT -> CS_ROUTING
> 2011-03-24 10:31:23.525898 [DEBUG] switch_core_session.c:1047 Send
> signal FreeTDM/1:1/ [BREAK]
> 2011-03-24 10:31:23.525898 [DEBUG] switch_core_state_machine.c:338
> (FreeTDM/1:1/) State INIT going to sleep
> 2011-03-24 10:31:23.525898 [DEBUG] switch_core_state_machine.c:314
> (FreeTDM/1:1/) Running State Change CS_ROUTING
> 2011-03-24 10:31:23.525898 [DEBUG] switch_channel.c:1512
> (FreeTDM/1:1/) Callstate Change DOWN -> RINGING
> 2011-03-24 10:31:23.525898 [DEBUG] switch_core_state_machine.c:341
> (FreeTDM/1:1/) State ROUTING
> 2011-03-24 10:31:23.525898 [DEBUG] mod_freetdm.c:428 FreeTDM/1:1/
> CHANNEL ROUTING
> 2011-03-24 10:31:23.525898 [DEBUG] switch_ivr_originate.c:66
> (FreeTDM/1:1/) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2011-03-24 10:31:23.525898 [DEBUG] switch_core_session.c:1047 Send
> signal FreeTDM/1:1/ [BREAK]
> 2011-03-24 10:31:23.525898 [DEBUG] switch_core_state_machine.c:341
> (FreeTDM/1:1/) State ROUTING going to sleep
> 2011-03-24 10:31:23.525898 [DEBUG] switch_core_state_machine.c:314
> (FreeTDM/1:1/) Running State Change CS_CONSUME_MEDIA
> 2011-03-24 10:31:23.525898 [DEBUG] switch_core_state_machine.c:360
> (FreeTDM/1:1/) State CONSUME_MEDIA
> 2011-03-24 10:31:23.525898 [DEBUG] switch_core_state_machine.c:360
> (FreeTDM/1:1/) State CONSUME_MEDIA going to sleep
> 2011-03-24 10:31:23.536490 [DEBUG] switch_ivr_originate.c:2599 local
> variable string 0 = [presence_id=1001 at microcomaustralia.com.au]
> 2011-03-24 10:31:23.538398 [ERR] switch_ivr_originate.c:2632 Cannot
> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> 2011-03-24 10:31:23.538398 [INFO] ftmod_zt.c:636 Setting echo cancel
> to 64 taps for 2:1
> 2011-03-24 10:31:23.544342 [DEBUG] mod_freetdm.c:377 Set codec PCMU 20ms
> 2011-03-24 10:31:23.551431 [DEBUG] mod_freetdm.c:1334 Connect outbound
> channel FreeTDM/2:1/
> 2011-03-24 10:31:23.555384 [NOTICE] switch_channel.c:779 New Channel
> FreeTDM/2:1/ [86275359-c4be-4663-9561-c5973344c1a7]
> 2011-03-24 10:31:23.559646 [DEBUG] mod_freetdm.c:1348 (FreeTDM/2:1/)
> State Change CS_NEW -> CS_INIT
> 2011-03-24 10:31:23.561459 [DEBUG] switch_core_session.c:1047 Send
> signal FreeTDM/2:1/ [BREAK]
> 2011-03-24 10:31:23.561459 [DEBUG] ftmod_analog.c:361 [s2c1][1:2]
> ANALOG CHANNEL thread starting.
> 2011-03-24 10:31:23.565467 [DEBUG] ftmod_analog.c:88 [s2c1][1:2]
> Changed state from DOWN to GENRING
> 2011-03-24 10:31:24.665349 [INFO] ftmod_zt.c:636 Setting echo cancel
> to 64 taps for 2:1
> 2011-03-24 10:31:24.665349 [DEBUG] ftdm_io.c:2539 [s2c1][1:2] Enabled
> software DTMF detector
> 2011-03-24 10:31:24.665349 [DEBUG] ftmod_analog.c:381 [s2c1][1:2]
> Initialized DTMF detection
> 2011-03-24 10:31:24.665349 [DEBUG] ftmod_analog.c:534 [s2c1][1:2]
> Executing state handler on 2:1 for GENRING
> 2011-03-24 10:31:24.665349 [DEBUG] mod_freetdm.c:1715 got FXS sig
> [PROGRESS]
> 2011-03-24 10:31:24.665349 [NOTICE] mod_freetdm.c:1732 Ring-Ready
> FreeTDM/2:1/!
> 2011-03-24 10:31:24.677691 [DEBUG] switch_core_state_machine.c:314
> (FreeTDM/2:1/) Running State Change CS_INIT
> 2011-03-24 10:31:24.677691 [DEBUG] switch_core_state_machine.c:338
> (FreeTDM/2:1/) State INIT
> 2011-03-24 10:31:24.677691 [DEBUG] mod_freetdm.c:405 (FreeTDM/2:1/)
> State Change CS_INIT -> CS_ROUTING
> 2011-03-24 10:31:24.677691 [DEBUG] switch_core_session.c:1047 Send
> signal FreeTDM/2:1/ [BREAK]
> 2011-03-24 10:31:24.677691 [DEBUG] switch_core_state_machine.c:338
> (FreeTDM/2:1/) State INIT going to sleep
> 2011-03-24 10:31:24.677691 [DEBUG] switch_core_state_machine.c:314
> (FreeTDM/2:1/) Running State Change CS_ROUTING
> 2011-03-24 10:31:24.677691 [DEBUG] switch_channel.c:1512
> (FreeTDM/2:1/) Callstate Change DOWN -> RINGING
> 2011-03-24 10:31:24.677691 [DEBUG] switch_core_state_machine.c:341
> (FreeTDM/2:1/) State ROUTING
> 2011-03-24 10:31:24.677691 [DEBUG] mod_freetdm.c:428 FreeTDM/2:1/
> CHANNEL ROUTING
> 2011-03-24 10:31:24.677691 [DEBUG] switch_ivr_originate.c:66
> (FreeTDM/2:1/) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2011-03-24 10:31:24.677691 [DEBUG] switch_core_session.c:1047 Send
> signal FreeTDM/2:1/ [BREAK]
> 2011-03-24 10:31:24.677691 [DEBUG] switch_core_state_machine.c:341
> (FreeTDM/2:1/) State ROUTING going to sleep
> 2011-03-24 10:31:24.677691 [DEBUG] switch_core_state_machine.c:314
> (FreeTDM/2:1/) Running State Change CS_CONSUME_MEDIA
> 2011-03-24 10:31:24.677691 [DEBUG] switch_core_state_machine.c:360
> (FreeTDM/2:1/) State CONSUME_MEDIA
> 2011-03-24 10:31:24.677691 [DEBUG] switch_core_state_machine.c:360
> (FreeTDM/2:1/) State CONSUME_MEDIA going to sleep
> 2011-03-24 10:31:24.692818 [DEBUG] switch_ivr_originate.c:2599 local
> variable string 0 = [presence_id=1002 at microcomaustralia.com.au]
> 2011-03-24 10:31:24.697423 [NOTICE] switch_channel.c:779 New Channel
> sofia/internal/sip:1002 at 192.168.2.13:5060
> [0da6383c-4633-4a03-99ed-839918a74161]
> 2011-03-24 10:31:24.737484 [DEBUG] mod_sofia.c:3920
> (sofia/internal/sip:1002 at 192.168.2.13:5060) State Change CS_NEW ->
> CS_INIT
> 2011-03-24 10:31:24.737484 [DEBUG] switch_core_session.c:1047 Send
> signal sofia/internal/sip:1002 at 192.168.2.13:5060 [BREAK]
> 2011-03-24 10:31:24.752679 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:1002 at 192.168.2.13:5060) Running State Change
> CS_INIT
> 2011-03-24 10:31:24.756503 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/sip:1002 at 192.168.2.13:5060) State INIT
> 2011-03-24 10:31:24.758368 [DEBUG] mod_sofia.c:83
> sofia/internal/sip:1002 at 192.168.2.13:5060 SOFIA INIT
> 2011-03-24 10:31:24.777518 [DEBUG] mod_sofia.c:123
> (sofia/internal/sip:1002 at 192.168.2.13:5060) State Change CS_INIT ->
> CS_ROUTING
> 2011-03-24 10:31:24.777518 [DEBUG] switch_core_session.c:1047 Send
> signal sofia/internal/sip:1002 at 192.168.2.13:5060 [BREAK]
> 2011-03-24 10:31:24.777518 [DEBUG] sofia.c:4402 Channel
> sofia/internal/sip:1002 at 192.168.2.13:5060 entering state [calling][0]
> 2011-03-24 10:31:24.781657 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/sip:1002 at 192.168.2.13:5060) State INIT going to sleep
> 2011-03-24 10:31:24.781657 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:1002 at 192.168.2.13:5060) Running State Change
> CS_ROUTING
> 2011-03-24 10:31:24.785366 [DEBUG] switch_channel.c:1512
> (sofia/internal/sip:1002 at 192.168.2.13:5060) Callstate Change DOWN ->
> RINGING
> 2011-03-24 10:31:24.799334 [INFO] sofia.c:709
> sofia/internal/sip:1002 at 192.168.2.13:5060 Update Callee ID to "1002"
> <1002>
> 2011-03-24 10:31:24.808910 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/sip:1002 at 192.168.2.13:5060) State ROUTING
> 2011-03-24 10:31:24.808910 [DEBUG] sofia.c:4402 Channel
> sofia/internal/sip:1002 at 192.168.2.13:5060 entering state
> [proceeding][180]
> 2011-03-24 10:31:24.812354 [NOTICE] sofia.c:4474 Ring-Ready
> sofia/internal/sip:1002 at 192.168.2.13:5060!
> 2011-03-24 10:31:24.815429 [DEBUG] mod_sofia.c:146
> sofia/internal/sip:1002 at 192.168.2.13:5060 SOFIA ROUTING
> 2011-03-24 10:31:24.815429 [DEBUG] switch_ivr_originate.c:66
> (sofia/internal/sip:1002 at 192.168.2.13:5060) State Change CS_ROUTING ->
> CS_CONSUME_MEDIA
> 2011-03-24 10:31:24.815429 [DEBUG] switch_core_session.c:1047 Send
> signal sofia/internal/sip:1002 at 192.168.2.13:5060 [BREAK]
> 2011-03-24 10:31:24.815429 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/sip:1002 at 192.168.2.13:5060) State ROUTING going to
> sleep
> 2011-03-24 10:31:24.815429 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:1002 at 192.168.2.13:5060) Running State Change
> CS_CONSUME_MEDIA
> 2011-03-24 10:31:24.817284 [DEBUG] switch_core_state_machine.c:360
> (sofia/internal/sip:1002 at 192.168.2.13:5060) State CONSUME_MEDIA
> 2011-03-24 10:31:24.817284 [DEBUG] switch_core_state_machine.c:360
> (sofia/internal/sip:1002 at 192.168.2.13:5060) State CONSUME_MEDIA going
> to sleep
> 2011-03-24 10:31:24.819480 [DEBUG] switch_ivr_originate.c:2599 local
> variable string 0 = [presence_id=1003 at microcomaustralia.com.au]
> 2011-03-24 10:31:24.819480 [NOTICE] switch_channel.c:779 New Channel
> sofia/internal/sip:1003 at 192.168.2.13:5061
> [644a2402-7658-4831-8599-2439dc82e1f2]
> 2011-03-24 10:31:24.850900 [DEBUG] mod_sofia.c:3920
> (sofia/internal/sip:1003 at 192.168.2.13:5061) State Change CS_NEW ->
> CS_INIT
> 2011-03-24 10:31:24.851903 [DEBUG] switch_core_session.c:1047 Send
> signal sofia/internal/sip:1003 at 192.168.2.13:5061 [BREAK]
> 2011-03-24 10:31:24.865923 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:1003 at 192.168.2.13:5061) Running State Change
> CS_INIT
> 2011-03-24 10:31:24.865923 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/sip:1003 at 192.168.2.13:5061) State INIT
> 2011-03-24 10:31:24.872367 [DEBUG] mod_sofia.c:83
> sofia/internal/sip:1003 at 192.168.2.13:5061 SOFIA INIT
> 2011-03-24 10:31:24.897626 [DEBUG] mod_sofia.c:123
> (sofia/internal/sip:1003 at 192.168.2.13:5061) State Change CS_INIT ->
> CS_ROUTING
> 2011-03-24 10:31:24.898904 [DEBUG] switch_core_session.c:1047 Send
> signal sofia/internal/sip:1003 at 192.168.2.13:5061 [BREAK]
> 2011-03-24 10:31:24.898904 [DEBUG] sofia.c:4402 Channel
> sofia/internal/sip:1003 at 192.168.2.13:5061 entering state [calling][0]
> 2011-03-24 10:31:24.899882 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/sip:1003 at 192.168.2.13:5061) State INIT going to sleep
> 2011-03-24 10:31:24.899882 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:1003 at 192.168.2.13:5061) Running State Change
> CS_ROUTING
> 2011-03-24 10:31:24.920400 [DEBUG] sofia.c:4402 Channel
> sofia/internal/sip:1003 at 192.168.2.13:5061 entering state [calling][0]
> 2011-03-24 10:31:24.930315 [DEBUG] sofia.c:6117 IP 59.167.180.194
> Rejected by acl "domains". Falling back to Digest auth.
> 2011-03-24 10:31:24.930315 [NOTICE] switch_channel.c:779 New Channel
> sofia/internal/1004 at 59.167.180.194
> [294c0493-b397-481a-ac78-09fa727d79b9]
> 2011-03-24 10:31:24.939488 [DEBUG] switch_channel.c:1512
> (sofia/internal/sip:1003 at 192.168.2.13:5061) Callstate Change DOWN ->
> RINGING
> 2011-03-24 10:31:24.947209 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/sip:1003 at 192.168.2.13:5061) State ROUTING
> 2011-03-24 10:31:24.947209 [DEBUG] mod_sofia.c:146
> sofia/internal/sip:1003 at 192.168.2.13:5061 SOFIA ROUTING
> 2011-03-24 10:31:24.949210 [DEBUG] switch_ivr_originate.c:66
> (sofia/internal/sip:1003 at 192.168.2.13:5061) State Change CS_ROUTING ->
> CS_CONSUME_MEDIA
> 2011-03-24 10:31:24.950890 [DEBUG] switch_core_session.c:1047 Send
> signal sofia/internal/sip:1003 at 192.168.2.13:5061 [BREAK]
> 2011-03-24 10:31:24.950890 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/sip:1003 at 192.168.2.13:5061) State ROUTING going to
> sleep
> 2011-03-24 10:31:24.950890 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:1003 at 192.168.2.13:5061) Running State Change
> CS_CONSUME_MEDIA
> 2011-03-24 10:31:24.954269 [DEBUG] switch_core_state_machine.c:360
> (sofia/internal/sip:1003 at 192.168.2.13:5061) State CONSUME_MEDIA
> 2011-03-24 10:31:24.954269 [DEBUG] switch_core_state_machine.c:360
> (sofia/internal/sip:1003 at 192.168.2.13:5061) State CONSUME_MEDIA going
> to sleep
> 2011-03-24 10:31:24.954269 [DEBUG] switch_ivr_originate.c:2599 local
> variable string 0 = [presence_id=1004 at microcomaustralia.com.au]
> 2011-03-24 10:31:24.960337 [NOTICE] switch_channel.c:779 New Channel
> sofia/internal/sip:1004 at 192.168.2.16
> [55c3e984-ddf8-45db-a157-65c50e18c8d0]
> 2011-03-24 10:31:24.970358 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/1004 at 59.167.180.194) Running State Change CS_NEW
> 2011-03-24 10:31:24.970358 [DEBUG] switch_core_state_machine.c:320
> (sofia/internal/1004 at 59.167.180.194) State NEW
> 2011-03-24 10:31:24.994428 [DEBUG] mod_sofia.c:3920
> (sofia/internal/sip:1004 at 192.168.2.16) State Change CS_NEW -> CS_INIT
> 2011-03-24 10:31:24.996392 [DEBUG] switch_core_session.c:1047 Send
> signal sofia/internal/sip:1004 at 192.168.2.16 [BREAK]
> 2011-03-24 10:31:25.023275 [DEBUG] sofia.c:4402 Channel
> sofia/internal/1004 at 59.167.180.194 entering state [received][100]
> 2011-03-24 10:31:25.029365 [DEBUG] sofia.c:4413 Remote SDP:
> v=0
> o=FreeSWITCH 1300899030 1300899032 IN IP4 59.167.180.194
> s=FreeSWITCH
> c=IN IP4 59.167.180.194
> t=0 0
> m=audio 24054 RTP/AVP 8 115 107 0 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:115 G7221/32000
> a=fmtp:115 bitrate=48000
> a=rtpmap:107 G7221/16000
> a=fmtp:107 bitrate=32000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
>
> 2011-03-24 10:31:25.035447 [DEBUG] sofia_glue.c:3859 Audio Codec
> Compare [PCMA:8:8000:20]/[G7221:115:32000:20]
> 2011-03-24 10:31:25.035447 [DEBUG] sofia_glue.c:3859 Audio Codec
> Compare [PCMA:8:8000:20]/[G7221:107:16000:20]
> 2011-03-24 10:31:25.035447 [DEBUG] sofia_glue.c:3859 Audio Codec
> Compare [PCMA:8:8000:20]/[PCMU:0:8000:20]
> 2011-03-24 10:31:25.035447 [DEBUG] sofia_glue.c:3859 Audio Codec
> Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
> 2011-03-24 10:31:25.037256 [DEBUG] sofia_glue.c:2454 Set Codec
> sofia/internal/1004 at 59.167.180.194 PCMA/8000 20 ms 160 samples
> 2011-03-24 10:31:25.049229 [DEBUG] sofia_glue.c:3955 Set 2833 dtmf
> send/recv payload to 101
> 2011-03-24 10:31:25.049229 [DEBUG] sofia.c:4573
> (sofia/internal/1004 at 59.167.180.194) State Change CS_NEW -> CS_INIT
> 2011-03-24 10:31:25.049229 [DEBUG] switch_core_session.c:1047 Send
> signal sofia/internal/1004 at 59.167.180.194 [BREAK]
> 2011-03-24 10:31:25.053356 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/1004 at 59.167.180.194) Running State Change CS_INIT
> 2011-03-24 10:31:25.053356 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/1004 at 59.167.180.194) State INIT
> 2011-03-24 10:31:25.055393 [DEBUG] mod_sofia.c:83
> sofia/internal/1004 at 59.167.180.194 SOFIA INIT
> 2011-03-24 10:31:25.055393 [DEBUG] mod_sofia.c:123
> (sofia/internal/1004 at 59.167.180.194) State Change CS_INIT ->
> CS_ROUTING
> 2011-03-24 10:31:25.055393 [DEBUG] switch_core_session.c:1047 Send
> signal sofia/internal/1004 at 59.167.180.194 [BREAK]
> 2011-03-24 10:31:25.055393 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/1004 at 59.167.180.194) State INIT going to sleep
> 2011-03-24 10:31:25.055393 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/1004 at 59.167.180.194) Running State Change CS_ROUTING
> 2011-03-24 10:31:25.055393 [DEBUG] switch_channel.c:1512
> (sofia/internal/1004 at 59.167.180.194) Callstate Change DOWN -> RINGING
> 2011-03-24 10:31:25.055393 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/1004 at 59.167.180.194) State ROUTING
> 2011-03-24 10:31:25.055393 [DEBUG] mod_sofia.c:146
> sofia/internal/1004 at 59.167.180.194 SOFIA ROUTING
> 2011-03-24 10:31:25.055393 [DEBUG] switch_core_state_machine.c:77
> sofia/internal/1004 at 59.167.180.194 Standard ROUTING
> 2011-03-24 10:31:25.055393 [INFO] mod_dialplan_xml.c:331 Processing
> Brian May <1004>->6201 in context public
>
>
> Something went wrong here. It starts processing 6201 as per normal.
> --
> Brian May <brian at microcomaustralia.com.au>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Thu Mar 24 22:49:04 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 24 Mar 2011 15:49:04 -0400
Subject: [Freeswitch-users] member_id of conference
References: <A4C72EF1FA19481EBD30DB4E58617E75@e1705><AANLkTinLw86HQM8SSHcrw5j_3K4VP6XB4nX+C53f9+Jh@mail.gmail.com><BC3FD7C9CDE244CFB6A3A30EC360E12A@e1705>
	<AANLkTin8NvtCdSiap2AGLPski-ry=ZR71ukMDN_v4HWm@mail.gmail.com>
Message-ID: <7C97681A65F54EC3B68CA47820AB3F8D@e1705>

> Well, you have the uuid of the call, correct? 
no since this command 
conference testconf dial {originate_timeout=30}sofia/default/1000 at softswitch 1234567890 FreeSWITCH_Conference

doesn't return any uuid

> we recently added conference_uuid channel variable so you can even see which exact instance of the conference that the user was/is connected to.

I'm confused. if I use the conference_uuid channel to get the member_id it will be the same as conference list since
it will return all member_id and not the only one the conferencer just called for ?


  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Thursday, March 24, 2011 3:32 PM
  Subject: Re: [Freeswitch-users] member_id of conference





  On Thu, Mar 24, 2011 at 12:05 PM, Madovsky <infos at madovsky.org> wrote:

    for example the conferencer invites 9999999999
    and does it from fs_cli. the return string of conference dial command is
    the status of call after success or failure (seems that the command is synchronous)
    but not uuid information nor id_member in return. So maybe have the member_id in the
    return string would be useful in case of the call fails in a not useful answered call like answer machine
    or "caller is unreacheable" from telcom.
    as this it will avoid to run "conference list", parse the result and find the right member_id.

    what folks do you think ?
  Well, you have the uuid of the call, correct? You can get the member_id with uuid_getvar:


  uuid_getvar <uuid> member_id


  If there is no member_id then your call never made it into the conference.  BTW, we recently added conference_uuid channel variable so you can even see which exact instance of the conference that the user was/is connected to.


  -MC


------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From lists at telefaks.de  Fri Mar 25 00:49:17 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Thu, 24 Mar 2011 22:49:17 +0100
Subject: [Freeswitch-users] mod_conference and max-members
In-Reply-To: <AANLkTinEZooEw9rhcHwYVAyjyJ-+p8xBZsxcCKmt8ouP@mail.gmail.com>
References: <4D8B6D44.4000909@telefaks.de>
	<AANLkTinEZooEw9rhcHwYVAyjyJ-+p8xBZsxcCKmt8ouP@mail.gmail.com>
Message-ID: <4D8BBC5D.6020309@telefaks.de>

Thanks, I understood that for this case we have to implement the
max-members functionality outside of freeswitch.
We will then do it that way.

Best regards
Peter

Michael Collins schrieb:
> I don't believe you can change the parameters from a conference
> profile for an active conference. However, if you are using the event
> socket and have some sort of conference manager then you can use the
> "conference lock" and "conference unlock" APIs to accomplish the same
> purpose. When the conference gets to X number of users your conf
> manager program can issue the lock command. The moderator can then
> lock or unlock as he/she sees fit.
>
> -MC
>
> On Thu, Mar 24, 2011 at 9:11 AM, Peter Steinbach <lists at telefaks.de
> <mailto:lists at telefaks.de>> wrote:
>
>     Hello,
>
>     when we open a conference and set the value of "max-members" to 5
>     then only 5 members will be able to join the conference. This is fine.
>     But just in case the conference moderator wants to invite an
>     additional person: Is it possible to temporarily disable this
>     function and to add (invite) another member to the conference?
>
>     What we have tried so far:
>
>         * 5 members dialed into the conference
>         * changed number of members temporarily to 6 (via XML-Curl)
>         * Invited another member into the conference (potential 6th
>           member)
>         * Freeswitch requests the new conference.conf XML with
>           "max-members" == 6
>         * but still it shows "Conference is full" in the logs to the
>           potential 6th member
>
>     Is there any other way to overcome this, maybe by sending some api
>     command via event socket?
>
>     -- 
>     With kind regards
>     Peter Steinbach 
>
>     Telefaks Services GmbH
>     mailto:lists (att) telefaks.de <http://telefaks.de>
>     Internet: www.telefaks.de <http://www.telefaks.de>
>
>         
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de




From kbdfck at gmail.com  Fri Mar 25 01:05:06 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Fri, 25 Mar 2011 01:05:06 +0300
Subject: [Freeswitch-users] Event-lock,
 sync/async on outbound ESL and execute() tagging question
In-Reply-To: <883D3DA8-9A8F-4E44-A0CD-93833172C48D@gmail.com>
References: <AANLkTingnGhvshmDK3TxKSkTpFeno76KTrFULRhbSCiP@mail.gmail.com>
	<AANLkTim9ftv7Q40i2ZhkfH6U8Qk+W5-NjziNt+B6anXW@mail.gmail.com>
	<AANLkTi=uNARtTnWAjuH42129o-LN64J+n26=Reqzzrqy@mail.gmail.com>
	<AANLkTin6oFsWUX7gm-gA26RLzi-fuDz7ie9FtMUR0SU5@mail.gmail.com>
	<AANLkTikvTcH5mscmC7q06_VHgmVVXw=6SAG7RHc0vPd3@mail.gmail.com>
	<AANLkTinUAo2bUEtmHEz09U3rvxb=c1D-Jv3bZ+gp6GR-@mail.gmail.com>
	<883D3DA8-9A8F-4E44-A0CD-93833172C48D@gmail.com>
Message-ID: <AANLkTim+gybTWoMt-LXTLpfmLow4eGmA5cFm_+7LYkcs@mail.gmail.com>

Application name can be the same for multiple execute() calls, so we need to
hash by name+params+uuid+something else, but single user- or autogenerated
tag field in request/response will be enough to track application execution
:)

Anyway, thanks to all for your ideas! I have similiar working solution, but
I thought if it can be done in some better way.

2011/3/24 Mohammad Amin <manavid at gmail.com>

> I have hash table which keeps track of api & dp applications I have
> executed to process the events & responses I get from FS, based on that I
> control the flow of execution in my application.
>
> For instance for the 'read' application my server returns the result
>  variable as soon as It receives CHANNEL_EXECUTE_COMPLETE with the
> 'application: read' in it's content.
>
> The fired events are very self explanatory it has all the info you need to
> control the execution flow. If you watch for the 'application' in the
> content of CHANNEL_EXECUTE_COMPLETE and you keep track of what applications
> you have executed so far you can map them very easily.
>
> Moe
>
> On Mar 24, 2011, at 11:11 AM, Dmitry Sytchev wrote:
>
> I don't get any irrelevant channel events, it's my events or my child
> channel events.
> I want to distinguish events from previously launched execute() commands in
> my event loop, like job id in bgapi. These events can be from my own
> original channel as well, so uuid filter doesn't help me.
>
>
> 2011/3/24 Moe Navid <manavid at gmail.com>
>
>> I use outbound socket for bridging, I also do my accounting from there
>> too, As soon as I get the CHANNEL_ANSWER event with the direction:outbound
>> (in content) I defer a new process and create a timer which does 6 seconds
>> billing increments.
>>
>> The trick for not receiving events irrelevant to my session was issuing
>> following at the beginning of my session:
>>
>> filter unique-id <uuid>\n\n
>> filter channel-call-uuid <uuid>\n\n
>>
>> I wrote my server in Ruby, I'm using EventMachine for TCP handling and
>> Ruby Fibers to write my apps synchronous fashion while talking to FreeSWITCH
>> in async mode. Whenever I need to simulate the blocking mode and pause the
>> execution at some point (with Fibers) I just start listening for the event
>> that has do be received which indicates the job is done, in case of bridge,
>> I enqueue the CHANNEL_HANGUP_COMPLETE and resume my app as soon as I receive
>> it.
>>
>> Moe
>>
>>
>> On Wed, Mar 23, 2011 at 1:41 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>
>>> I understand this. I use outbound async full ESL mode, launching socket
>>> app from dialplan, because in sync mode I have some problems with bridge app
>>> and I need to catch parent channel hangup and server disconnection.
>>>
>>> I written wrapper around ESL.pm to implement executeSync which waits for
>>> CHANNEL_EXECUTE_COMPLETE while in async mode, works for now.
>>> But of course it is not the best way, because of possible error when
>>> 'execute' is used before my 'executeSync' by mistake and without proper
>>> event handling so that CHANNEL_EXECUTE_COMPLETE from previous app is catched
>>> by my wrapper and it returns prematurely. This is why I want to have some
>>> tag I can mark execute call to filter its events later... Maybe there is
>>> better way?
>>>
>>> 2011/3/23 Michael Collins <msc at freeswitch.org>
>>>
>>>>
>>>>
>>>> On Wed, Mar 23, 2011 at 6:48 AM, lakshmanan ganapathy <
>>>> lakindia89 at gmail.com> wrote:
>>>>
>>>>> Hi,
>>>>> Eventlock is used to say "Whether to queue the commands or not".
>>>>>
>>>>> Assume I'm giving playback and bridge command via outbound socket to
>>>>> freeswitch.
>>>>>
>>>>> If eventlock is "true" then, first "playback will be executed" and then
>>>>> only the bridge will be executed.
>>>>>
>>>>> If eventlock is "false", then order of execution is not guaranteed.
>>>>>
>>>>> I'm also using Outbound event socket. I'll run it in async mode. After
>>>>> each execute() statement, if I need the output for that event, I will wait
>>>>> for the event and then only I'll proceed. If I don't need event, ( playback
>>>>> ), I just do the other operations.
>>>>>
>>>>> This works well for me.
>>>>>
>>>>> Be sure that you are using the terms "inbound" and "outbound" event
>>>> socket correctly. It's inbound or outbound from the perspective of
>>>> FreeSWITCH, not from the perspective of your script. An easy rule of thumb
>>>> is this: if you use the 'socket' application in your dialplan then you are
>>>> using "outbound event socket" because FS has to make an outbound connection
>>>> to your script.
>>>>
>>>> http://wiki.freeswitch.org/wiki/Event_socket
>>>>
>>>> -MC
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Best regards,
>>>
>>> Dmitry Sytchev,
>>> IT Engineer
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From brian at microcomaustralia.com.au  Fri Mar 25 02:01:34 2011
From: brian at microcomaustralia.com.au (Brian May)
Date: Fri, 25 Mar 2011 10:01:34 +1100
Subject: [Freeswitch-users] answering calls weirdness
In-Reply-To: <AANLkTin8nv3N=XsPa=jepebenxRJ4EvtNSMJ3D37KZKu@mail.gmail.com>
References: <AANLkTin5mP7QzDe+CAvaPjsbxhq934P-49LARGQJwR8Y@mail.gmail.com>
	<AANLkTin8nv3N=XsPa=jepebenxRJ4EvtNSMJ3D37KZKu@mail.gmail.com>
Message-ID: <AANLkTink5QsRAeQc8xRw=-2Jog+_PDRebfwSMFUF0BGw@mail.gmail.com>

On 25 March 2011 06:44, Michael Collins <msc at freeswitch.org> wrote:
> Can you drop this into pastebin? pastebin.freeswitch.org. Also include the
> dialplan that is running all this. Use the "freeswitch log" syntax
> highlighting because it makes it much easier to read.

That seems to require a username and password, without telling me how
to register. How do I get a username and password?
-- 
Brian May <brian at microcomaustralia.com.au>


From brian at microcomaustralia.com.au  Fri Mar 25 02:05:05 2011
From: brian at microcomaustralia.com.au (Brian May)
Date: Fri, 25 Mar 2011 10:05:05 +1100
Subject: [Freeswitch-users] answering calls weirdness
In-Reply-To: <AANLkTink5QsRAeQc8xRw=-2Jog+_PDRebfwSMFUF0BGw@mail.gmail.com>
References: <AANLkTin5mP7QzDe+CAvaPjsbxhq934P-49LARGQJwR8Y@mail.gmail.com>
	<AANLkTin8nv3N=XsPa=jepebenxRJ4EvtNSMJ3D37KZKu@mail.gmail.com>
	<AANLkTink5QsRAeQc8xRw=-2Jog+_PDRebfwSMFUF0BGw@mail.gmail.com>
Message-ID: <AANLkTimXFmm5sqnn5eH0+Zk=8ue0r3D9RtajkUQpvsKe@mail.gmail.com>

On 25 March 2011 10:01, Brian May <brian at microcomaustralia.com.au> wrote:
> That seems to require a username and password, without telling me how
> to register. How do I get a username and password?

Oh, never mind, I worked it out....
-- 
Brian May <brian at microcomaustralia.com.au>


From brian at microcomaustralia.com.au  Fri Mar 25 02:19:07 2011
From: brian at microcomaustralia.com.au (Brian May)
Date: Fri, 25 Mar 2011 10:19:07 +1100
Subject: [Freeswitch-users] answering calls weirdness
In-Reply-To: <AANLkTin8nv3N=XsPa=jepebenxRJ4EvtNSMJ3D37KZKu@mail.gmail.com>
References: <AANLkTin5mP7QzDe+CAvaPjsbxhq934P-49LARGQJwR8Y@mail.gmail.com>
	<AANLkTin8nv3N=XsPa=jepebenxRJ4EvtNSMJ3D37KZKu@mail.gmail.com>
Message-ID: <AANLkTi=GC1u5oZeACdpUeF=ggT3c5AAbDPH5KAgYzTKW@mail.gmail.com>

On 25 March 2011 06:44, Michael Collins <msc at freeswitch.org> wrote:
> Can you drop this into pastebin? pastebin.freeswitch.org. Also include the
> dialplan that is running all this.

freeswitch log:

http://pastebin.freeswitch.org/15837

relevant part of the dialplan:

http://pastebin.freeswitch.org/15838

> Use the "freeswitch log" syntax
> highlighting because it makes it much easier to read.
> -MC

Not really sure about that. It seems to have put broken HTML or
something before the first line??? "pan class="re0">"
-- 
Brian May <brian at microcomaustralia.com.au>


From fieldpeak at gmail.com  Fri Mar 25 05:47:41 2011
From: fieldpeak at gmail.com (Charles)
Date: Fri, 25 Mar 2011 10:47:41 +0800
Subject: [Freeswitch-users] sofia_reg_internal.db grow bigger and
	biggeralong with time
References: <AANLkTi=FVvghm6H=BiV8jT+mYqE9SVtfNa0zst+BqFvF@mail.gmail.com>,
	<1300963689185-6203596.post@n2.nabble.com>,
	<AANLkTimAjOrO3Qp6r09G+--tEKbbPrvTHjVxkfmoTZ=o@mail.gmail.com>,
	<AANLkTimjjtKju704KWokqXYySjx9=hZk5E_H2dAaqtxG@mail.gmail.com>
Message-ID: <4d8c0255.254b640a.36b5.1f98@mx.google.com>

i've tried the latest weekly build (freeswitch.msi 20-Mar-2011 18:30), there is the same issue.

http://files.freeswitch.org/windows/installer/x86/ 


2011-03-25 



Charles 



???? fieldpeak 
????? 2011-03-24  20:19:59 
???? FreeSWITCH Users Help 
??? 
??? Re: [Freeswitch-users] sofia_reg_internal.db grow bigger and biggeralong with time 
 
is it a known bug and fixed already by the newer version?
thanks!
? 2011-3-24 ??6:48?"mazilo" <Nabble at slickdeals.endjunk.com>???

>
> I am just curious what version of FreeSWITCH are you using that gives you
> this problem.
>
> -...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at list...
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From fieldpeak at gmail.com  Fri Mar 25 05:52:29 2011
From: fieldpeak at gmail.com (Charles)
Date: Fri, 25 Mar 2011 10:52:29 +0800
Subject: [Freeswitch-users] sofia_reg_internal.db grow bigger and
	biggeralong with time
References: <AANLkTi=FVvghm6H=BiV8jT+mYqE9SVtfNa0zst+BqFvF@mail.gmail.com>,
	<1300963689185-6203596.post@n2.nabble.com>,
	<AANLkTimAjOrO3Qp6r09G+--tEKbbPrvTHjVxkfmoTZ=o@mail.gmail.com>,
	<AANLkTimjjtKju704KWokqXYySjx9=hZk5E_H2dAaqtxG@mail.gmail.com>,
	<4d8c0255.254b640a.36b5.1f98@mx.google.com>
Message-ID: <4d8c0372.08a9640a.6448.220a@mx.google.com>

version cmd shows below exactly version,

FreeSWITCH Version 1.0.head (git-069f5f7 2011-03-18 16-59-00 -0500)


2011-03-25 



Charles 



???? Charles 
????? 2011-03-25  10:47:49 
???? fieldpeak; FreeSWITCH Users Help; msc; Nabble 
??? 
??? Re: Re: [Freeswitch-users] sofia_reg_internal.db grow bigger and biggeralong with time 
 
i've tried the latest weekly build (freeswitch.msi 20-Mar-2011 18:30), there is the same issue.

http://files.freeswitch.org/windows/installer/x86/ 


2011-03-25 



Charles 



???? fieldpeak 
????? 2011-03-24  20:19:59 
???? FreeSWITCH Users Help 
??? 
??? Re: [Freeswitch-users] sofia_reg_internal.db grow bigger and biggeralong with time 
is it a known bug and fixed already by the newer version?
thanks!
? 2011-3-24 ??6:48?"mazilo" <Nabble at slickdeals.endjunk.com>??? 

>
> I am just curious what version of FreeSWITCH are you using that gives you
> this problem.
>
> -...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at list...
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From mervyn.yeo at gmail.com  Fri Mar 25 04:52:47 2011
From: mervyn.yeo at gmail.com (Mervyn Yeo)
Date: Fri, 25 Mar 2011 09:52:47 +0800
Subject: [Freeswitch-users] FS does not respond to second 401 Unauthorized
 that has new nonce during registration with gateway
Message-ID: <AANLkTikvSjXBcfjcHuiRk5ea3t-33gGG9xvvdRtF6P=v@mail.gmail.com>

Hi Everyone,

I'm trying to register FS with a gateway but it's been unsuccessful so
far with the last message being 401 Unauthorized. I'm quite certain
that the configuration in the gateway xml file is correct. The same
username, password, etc has been used with X-Lite, Zoiper, Yealink,
Cisco phones and it's been able to register. I've compared SIP traces
of FS's registration and all the others and it seems that FS is not
responding to the last 401 message from the other end even when the
it's been challenged with a new nonce. What I found out was that the
second 401 has an additional domain parameter as well. Are all the
other clients being lenient and is not following the RFC or is this a
bug in FS?

SIP trace http://pastebin.freeswitch.org/15831
Configuration http://pastebin.freeswitch.org/15832

Mervyn


From jeff at jefflenk.com  Fri Mar 25 07:20:35 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 24 Mar 2011 21:20:35 -0700 (PDT)
Subject: [Freeswitch-users] sofia_reg_internal.db grow bigger and
	biggeralong with time
In-Reply-To: <4d8c0372.08a9640a.6448.220a@mx.google.com>
References: <AANLkTi=FVvghm6H=BiV8jT+mYqE9SVtfNa0zst+BqFvF@mail.gmail.com>
	<1300963689185-6203596.post@n2.nabble.com>
	<AANLkTimAjOrO3Qp6r09G+--tEKbbPrvTHjVxkfmoTZ=o@mail.gmail.com>
	<4d8c0255.254b640a.36b5.1f98@mx.google.com>
	<4d8c0372.08a9640a.6448.220a@mx.google.com>
Message-ID: <1301026835845-6206535.post@n2.nabble.com>

I will run some tests on the builds coming of that build server to verify
whether there is some build skew in those builds. I do run the code that
comes off that machine regularly so I dont think there is a problem with it.

You should really build the code yourself from git to make sure.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/sofia-reg-internal-db-grow-bigger-and-bigger-along-with-time-tp6200516p6206535.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From fieldpeak at gmail.com  Fri Mar 25 07:29:31 2011
From: fieldpeak at gmail.com (Charles)
Date: Fri, 25 Mar 2011 12:29:31 +0800
Subject: [Freeswitch-users] sofia_reg_internal.db grow bigger
	andbiggeralong with time
References: <AANLkTi=FVvghm6H=BiV8jT+mYqE9SVtfNa0zst+BqFvF@mail.gmail.com>,
	<1300963689185-6203596.post@n2.nabble.com>,
	<AANLkTimAjOrO3Qp6r09G+--tEKbbPrvTHjVxkfmoTZ=o@mail.gmail.com>,
	<4d8c0255.254b640a.36b5.1f98@mx.google.com>,
	<4d8c0372.08a9640a.6448.220a@mx.google.com>,
	<1301026835845-6206535.post@n2.nabble.com>
Message-ID: <4d8c1a2f.14ce640a.3e4c.259d@mx.google.com>

Hi Jeff,

I did really build on my pc (windows 7 and VS 2008), i got the git head at Nov 30, 2010
thanks for your help!


2011-03-25 



Charles 



???? Jeff Lenk 
????? 2011-03-25  12:21:36 
???? freeswitch-users 
??? 
??? Re: [Freeswitch-users] sofia_reg_internal.db grow bigger andbiggeralong with time 
 
I will run some tests on the builds coming of that build server to verify
whether there is some build skew in those builds. I do run the code that
comes off that machine regularly so I dont think there is a problem with it.
You should really build the code yourself from git to make sure.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/sofia-reg-internal-db-grow-bigger-and-bigger-along-with-time-tp6200516p6206535.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From chistyakov at directtel.ru  Fri Mar 25 09:38:08 2011
From: chistyakov at directtel.ru (=?UTF-8?B?0KfQuNGB0YLRj9C60L7QsiDQmNCy0LDQvQ==?=)
Date: Fri, 25 Mar 2011 09:38:08 +0300
Subject: [Freeswitch-users] I am trying to use ESL with LUA - undefined
 symbol: _ZN8ESLeventC1EPKcS1_
In-Reply-To: <AANLkTinLH5u=W0eJSZmDvhWNjpG6Xr45YOE38w+ZA_2M@mail.gmail.com>
References: <4D8B368B.7050909@directtel.ru>
	<AANLkTinLH5u=W0eJSZmDvhWNjpG6Xr45YOE38w+ZA_2M@mail.gmail.com>
Message-ID: <4D8C3850.5030204@directtel.ru>

Ohh... I am really newby )

It's work:
make luamod LOCAL_CFLAGS="-I/usr/include/lua5.1" LOCAL_LDFLAGS="-llua 
-lpthread -I/usr/include/lua5.1"

PS Originally I have tried: make -C lua...

24.03.2011 20:42, Michael Collins ?????:
>
>
>     /usr/local/bin/lua: error loading module 'ESL' from file
>
>
> Did you successfully build both esl and the Ruby mod?
> cd libs/esl
> make
> make rubymod
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From peter.olsson at visionutveckling.se  Fri Mar 25 09:51:48 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 25 Mar 2011 07:51:48 +0100
Subject: [Freeswitch-users] sofia_reg_internal.db grow
	bigger	andbiggeralong with time
Message-ID: <04371062-578D-4D69-94DA-2D8D633FE3DE@visionutveckling.se>

Sounds to me like you're being attacked from outside. I've never seen this behaviour on either Linux or Windows systems.

Just a guess though...

/Peter

----- Reply message -----
Fr?n: "Charles" <fieldpeak at gmail.com>
Datum: fre, mar 25, 2011 05:36
Rubrik: [Freeswitch-users] sofia_reg_internal.db grow bigger andbiggeralong with time
Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>, "freeswitch-users" <freeswitch-users at lists.freeswitch.org>, "jeff" <jeff at jefflenk.com>

Hi Jeff,

I did really build on my pc (windows 7 and VS 2008), i got the git head at Nov 30, 2010
thanks for your help!


2011-03-25
________________________________
Charles
________________________________
???? Jeff Lenk
????? 2011-03-25  12:21:36
???? freeswitch-users
???
??? Re: [Freeswitch-users] sofia_reg_internal.db grow bigger andbiggeralong with time
I will run some tests on the builds coming of that build server to verify
whether there is some build skew in those builds. I do run the code that
comes off that machine regularly so I dont think there is a problem with it.
You should really build the code yourself from git to make sure.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/sofia-reg-internal-db-grow-bigger-and-bigger-along-with-time-tp6200516p6206535.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
!DSPAM:4d8c1b2a32765493338506!

From fieldpeak at gmail.com  Fri Mar 25 10:13:06 2011
From: fieldpeak at gmail.com (=?utf-8?B?Q2hhcmxlcw==?=)
Date: Fri, 25 Mar 2011 15:13:06 +0800
Subject: [Freeswitch-users]
	=?utf-8?q?sofia=5Freg=5Finternal=2Edb_growbigg?=
	=?utf-8?q?er=09andbiggeralong_with_time?=
References: <04371062-578D-4D69-94DA-2D8D633FE3DE@visionutveckling.se>
Message-ID: <4d8c4085.8b3b2b0a.7686.1e19@mx.google.com>

no, this is not the case, i can reproduce it in my lab, only one sip client register to the FS...
wireshark trace shows no any attack.


2011-03-25 



Charles 



???? Peter Olsson 
????? 2011-03-25  14:52:50 
???? freeswitch-users at lists.freeswitch.org 
??? 
??? Re: [Freeswitch-users] sofia_reg_internal.db growbigger andbiggeralong with time 
 
Sounds to me like you're being attacked from outside. I've never seen this behaviour on either Linux or Windows systems.
Just a guess though...
/Peter
----- Reply message -----
Fr?n: "Charles" <fieldpeak at gmail.com>
Datum: fre, mar 25, 2011 05:36
Rubrik: [Freeswitch-users] sofia_reg_internal.db grow bigger andbiggeralong with time
Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>, "freeswitch-users" <freeswitch-users at lists.freeswitch.org>, "jeff" <jeff at jefflenk.com>
Hi Jeff,
I did really build on my pc (windows 7 and VS 2008), i got the git head at Nov 30, 2010
thanks for your help!
2011-03-25
________________________________
Charles
________________________________
???? Jeff Lenk
????? 2011-03-25  12:21:36
???? freeswitch-users
???
??? Re: [Freeswitch-users] sofia_reg_internal.db grow bigger andbiggeralong with time
I will run some tests on the builds coming of that build server to verify
whether there is some build skew in those builds. I do run the code that
comes off that machine regularly so I dont think there is a problem with it.
You should really build the code yourself from git to make sure.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/sofia-reg-internal-db-grow-bigger-and-bigger-along-with-time-tp6200516p6206535.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
!DSPAM:4d8c1b2a32765493338506!
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From yivzhenko at mksat.net  Fri Mar 25 10:56:37 2011
From: yivzhenko at mksat.net (Yuriy Ivzhenko)
Date: Fri, 25 Mar 2011 09:56:37 +0200
Subject: [Freeswitch-users] mod_fifo & CID
In-Reply-To: <AANLkTim6S7kh=g-kSpT3x5jtNhS5-BHmitdf8o7FyYW9@mail.gmail.com>
References: <201103241644.27845.yivzhenko@mksat.net>
	<AANLkTim6S7kh=g-kSpT3x5jtNhS5-BHmitdf8o7FyYW9@mail.gmail.com>
Message-ID: <201103250956.37556.yivzhenko@mksat.net>

My previous version was 12 Oct 2010.


On Thursday 24 March 2011 20:37:25 Michael Collins wrote:
> How old was your previous FS? The default mod_fifo behavior changed a few
> months back.
> -MC
> 
> On Thu, Mar 24, 2011 at 7:44 AM, Yuriy Ivzhenko <yivzhenko at mksat.net> wrote:
> > Hi All,
> >
> > After upgrading to last Git (git-73ca862) i see changes in CID on
> > calls to agents, generated by mod_fifo.
> >
> > If agent answer
> > caller_profile.caller_id_name = "Outbound Call"
> > caller_profile.caller_id_number = <agent number>
> >
> > If agent not answer
> > caller_profile.caller_id_name = caller caller_id_name
> > caller_profile.caller_id_number = caller caller_id_number (or
> > origination_caller_id_name/number of set in agents dialstring)
> > In old version, so it was always.
> >
> > Is this correct?
> > It seems to me it should not depend on, answered the call or not.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 


From fieldpeak at gmail.com  Fri Mar 25 07:23:02 2011
From: fieldpeak at gmail.com (=?utf-8?B?Q2hhcmxlcw==?=)
Date: Fri, 25 Mar 2011 12:23:02 +0800
Subject: [Freeswitch-users]
	=?utf-8?q?sofia=5Freg=5Finternal=2Edb_grow_big?=
	=?utf-8?q?ger_and_biggeralong_with_time?=
References: <AANLkTi=FVvghm6H=BiV8jT+mYqE9SVtfNa0zst+BqFvF@mail.gmail.com>,
	<AANLkTikzwCDxGnCbMCNu5rGPyF9n03F_KwxvvR+SMpDa@mail.gmail.com>,
	<AANLkTi=aHzQQe4b=Su5Fr7ybwvNQ42-JU4gMZY9=Cird@mail.gmail.com>
Message-ID: <4d8c18b0.8235ec0a.539d.1f89@mx.google.com>

Hi Michael, 

It looks that I found the root cause, at the line#801 of sofia_reg.c, when excute the SQL statement ('INSERT') to the table 'sip_authentication', it wrongly insert as screenshot below, profile_name should be 'internal' and hostname should be 'mypc' (name of my computer).
however, i did not find anything wrong regarding the code itselft, so failed to fix it...
could you please help focus here to fix it, thanks.

P.S. i manually insert the correct record into this talbe, after a while (tens of seconds), it automatically deteleted by FS. 

btw, i found there is a bug report like this http://jira.freeswitch.org/browse/FS-498# and it is marked 
Resolved:14/Sep/10 12:32 PM, however, my version is Nov, 30, 2010, there is still exsit.





 line#801 of sofia_reg.c:
sql = switch_mprintf("insert into sip_authentication (nonce,expires,profile_name,hostname, last_nc) "
 "values('%q', %ld, '%q', '%q', 0)", uuid_str,
 switch_epoch_time_now(NULL) + (profile->nonce_ttl ? profile->nonce_ttl : DEFAULT_NONCE_TTL),
 profile->name, mod_sofia_globals.hostname);
switch_assert(sql != NULL);
sofia_glue_actually_execute_sql(profile, sql, profile->ireg_mutex);


2011-03-25 



Charles 



???? Michael Collins 
????? 2011-03-24  10:56:52 
???? FreeSWITCH Users Help 
??? 
??? Re: [Freeswitch-users] sofia_reg_internal.db grow bigger and biggeralong with time 
 
1300933453 is a unix epoch time in seconds. You can get the exact time from the fs_cli by issuing the strepoch command. Anyway, the size of that number is not the issue. I'll defer to Tony on the database record remaining in the table for longer than expected...


-MC


On Wed, Mar 23, 2011 at 7:42 PM, fieldpeak <fieldpeak at gmail.com> wrote:

Hi Anthony,

Thanks for your reply.
i tried to understand how long it is the expire time and where to cnofigure it.
so i made a test, see the attached screen shot, in the profile internal.xml, the nonce-ttl = 60, when i make a call from a registerred user, a new record was added in the 'sip_authentication' table with a very big value (1300933453, i don't know what the unit is, s or ms?) , after 60 seconds (actually i wait for more than 15 minutes), even the call finished, the new record still stay their, it was not deleted. want to know when it will be deleted...

thanks for your help!


2011/3/24 Anthony Minessale <anthony.minessale at gmail.com>

they expire so they will be deleted when the expire time happens.
If you have a lot you were probably attacked by a sip worm.



On Wed, Mar 23, 2011 at 9:38 AM, fieldpeak <fieldpeak at gmail.com> wrote:
> i found sofia_reg_internal.db grow bigger and bigger after FreeSwitch for
> some time.
> the change comes from the sip-authentication table inside this db, it always
> create new record for nonce for the new challenge ('407 Proxy Authentication
> Required') for the call from registered users.
> any one can help how to avoid the db grow, i'm afaid the hard disk will be
> full along with time...
>
> Thanks!
>

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



--
Anthony Minessale II

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

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

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

_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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http://www.freeswitch.org



_______________________________________________
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From jonas.gauffin at gmail.com  Fri Mar 25 12:12:37 2011
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Fri, 25 Mar 2011 10:12:37 +0100
Subject: [Freeswitch-users] sofia_reg_internal.db growbigger
 andbiggeralong with time
In-Reply-To: <4d8c4085.8b3b2b0a.7686.1e19@mx.google.com>
References: <04371062-578D-4D69-94DA-2D8D633FE3DE@visionutveckling.se>
	<4d8c4085.8b3b2b0a.7686.1e19@mx.google.com>
Message-ID: <AANLkTi==wH3un+te_an8ZX8gzi2_Xub3mNbGV7J8We1L@mail.gmail.com>

might be the same bug as I reported here:
http://jira.freeswitch.org/browse/FS-498



On Fri, Mar 25, 2011 at 8:13 AM, Charles <fieldpeak at gmail.com> wrote:

>  no, this is not the case, i can reproduce it in my lab, only one sip
> client register to the FS...
> wireshark trace shows no any attack.
>
>
> 2011-03-25
> ------------------------------
>  Charles
> ------------------------------
> *????* Peter Olsson
> *?????* 2011-03-25  14:52:50
> *????* freeswitch-users at lists.freeswitch.org
>   *???*
> *???* Re: [Freeswitch-users] sofia_reg_internal.db growbigger
> andbiggeralong with time
>
> Sounds to me like you're being attacked from outside. I've never seen this behaviour on either Linux or Windows systems.
>  Just a guess though...
>  /Peter
>  ----- Reply message -----
> Fr?n: "Charles" <fieldpeak at gmail.com>
> Datum: fre, mar 25, 2011 05:36
>
> Rubrik: [Freeswitch-users] sofia_reg_internal.db grow bigger andbiggeralong with time
> Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org
> >, "freeswitch-users" <freeswitch-users at lists.freeswitch.org>, "jeff" <
> jeff at jefflenk.com>
>  Hi Jeff,
>
> I did really build on my pc (windows 7 and VS 2008), i got the git head at Nov 30, 2010
> thanks for your help!
>  2011-03-25
> ________________________________
> Charles
> ________________________________
> ???? Jeff Lenk
> ????? 2011-03-25  12:21:36
> ???? freeswitch-users
> ???
>
> ??? Re: [Freeswitch-users] sofia_reg_internal.db grow bigger andbiggeralong with time
> I will run some tests on the builds coming of that build server to verify
> whether there is some build skew in those builds. I do run the code that
>
> comes off that machine regularly so I dont think there is a problem with it.
> You should really build the code yourself from git to make sure.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/sofia-reg-internal-db-grow-bigger-and-bigger-along-with-time-tp6200516p6206535.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> !DSPAM:4d8c1b2a32765493338506!
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
>
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From fieldpeak at gmail.com  Fri Mar 25 13:53:57 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Fri, 25 Mar 2011 18:53:57 +0800
Subject: [Freeswitch-users] sofia_reg_internal.db growbigger
 andbiggeralong with time
In-Reply-To: <AANLkTimh=bygFkZfgcw2D=PmHxzXudKA-r75-d08oc5r@mail.gmail.com>
References: <04371062-578D-4D69-94DA-2D8D633FE3DE@visionutveckling.se>
	<4d8c4085.8b3b2b0a.7686.1e19@mx.google.com>
	<AANLkTi==wH3un+te_an8ZX8gzi2_Xub3mNbGV7J8We1L@mail.gmail.com>
	<AANLkTikNF0ZjXBF_3iZonX2wkkRu2aOxx8hNsdBgrNt8@mail.gmail.com>
	<AANLkTimh=bygFkZfgcw2D=PmHxzXudKA-r75-d08oc5r@mail.gmail.com>
Message-ID: <AANLkTi=fNk_uv6LyFU93no6mt1mdyzu+2DbgeASOxXac@mail.gmail.com>

Hi Jonas,
i also think so. can you please update was the bug fixed?

? 2011-3-25 ??5:14?"Jonas Gauffin" <jonas.gauffin at gmail.com>???

might be the same bug as I reported here:
http://jira.freeswitch.org/browse/FS-498





On Fri, Mar 25, 2011 at 8:13 AM, Charles <fieldpeak at gmail.com> wrote:
>
> no, this is not the c...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Fri Mar 25 19:10:23 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 25 Mar 2011 11:10:23 -0500
Subject: [Freeswitch-users] sofia_reg_internal.db growbigger
 andbiggeralong with time
In-Reply-To: <AANLkTi=fNk_uv6LyFU93no6mt1mdyzu+2DbgeASOxXac@mail.gmail.com>
References: <04371062-578D-4D69-94DA-2D8D633FE3DE@visionutveckling.se>
	<4d8c4085.8b3b2b0a.7686.1e19@mx.google.com>
	<AANLkTi==wH3un+te_an8ZX8gzi2_Xub3mNbGV7J8We1L@mail.gmail.com>
	<AANLkTikNF0ZjXBF_3iZonX2wkkRu2aOxx8hNsdBgrNt8@mail.gmail.com>
	<AANLkTimh=bygFkZfgcw2D=PmHxzXudKA-r75-d08oc5r@mail.gmail.com>
	<AANLkTi=fNk_uv6LyFU93no6mt1mdyzu+2DbgeASOxXac@mail.gmail.com>
Message-ID: <AANLkTin9Lnnvf8MGqhoM6SkpnT5J3Xe-rUdUKB4qxXUb@mail.gmail.com>

can you try deleting the db files sofia_reg_* db files?  Maybe they
have become corrupt?



2011/3/25 fieldpeak <fieldpeak at gmail.com>:
> Hi Jonas,
> i also think so. can you please update was the bug fixed?
>
> ? 2011-3-25 ??5:14?"Jonas Gauffin" <jonas.gauffin at gmail.com>???
>
> might be the same bug as I reported here:
> http://jira.freeswitch.org/browse/FS-498
>
>
>
> On Fri, Mar 25, 2011 at 8:13 AM, Charles <fieldpeak at gmail.com> wrote:
>>
>> no, this is not the c...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 egable+freeswitch at gmail.com  Fri Mar 25 20:39:00 2011
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Fri, 25 Mar 2011 13:39:00 -0400
Subject: [Freeswitch-users] Voicemail on GlusterFS
In-Reply-To: <AANLkTinDYc2RKXMrj=ycs0zKa36gW2PZpiDgSLh-ajs1@mail.gmail.com>
References: <AANLkTimkv66jjaBTHtT+nDUgo3PsuxEndEi-Z9=Lb0u_@mail.gmail.com>
	<AANLkTi=44afGJ1fni5P+q=0ct=irmVDwOfESUAbdWuo9@mail.gmail.com>
	<AANLkTi=bWO48xovSm_am037RWAqnCjGSFOTYTdwiVe4J@mail.gmail.com>
	<AANLkTi=hpYhE2qdfuzuxNn=b5atbb7o-DjLSfNcy8Q6e@mail.gmail.com>
	<AANLkTinDYc2RKXMrj=ycs0zKa36gW2PZpiDgSLh-ajs1@mail.gmail.com>
Message-ID: <AANLkTik7ExTSGi62uEd4fyE0s+HthySYiT9gzZjxVHx0@mail.gmail.com>

Also check out MooseFS. I was very impressed by how quickly I was able to
set it up and use it. IIRC, it was < 2 hours to set up six nodes with shared
voicemail and sounds files from start to finish.

On Wed, Mar 23, 2011 at 2:59 PM, Dan Lane <null at invalid.name> wrote:

> On Tue, Mar 22, 2011 at 6:21 PM, Moe Navid <manavid at gmail.com> wrote:
> > Hi Dan,
> > Thanks for the reply. I'm also working on a custom voicemail.
> > Is your vm implemented in C/C++ and uses GlusterFS' native library or you
> > are just mounting with fuse?
>
> It's written in LUA and we're using FUSE to mount the GlusterFS
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing from
our children, we're stealing from them--and it's not even considered to be a
crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not
live to eat.) ~Marcus Tullius Cicero
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From frankie.k.yiu at gmail.com  Fri Mar 25 20:50:54 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Fri, 25 Mar 2011 10:50:54 -0700
Subject: [Freeswitch-users] Gawk.exe crashes when building on Windows 7
	64bits w/ VS
Message-ID: <AANLkTi=YLMvG8i=azBq3uBnoX6e6NE5iqscHYbNzsjUj@mail.gmail.com>

Disabled the virus scan and it compiled successfully.
FYI--I have AVG anti-virus.

Thanks,
Frankie


> ---------- Forwarded message ----------
> From: Jeff Lenk <jeff at jefflenk.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Thu, 24 Mar 2011 11:28:24 -0700 (PDT)
> Subject: Re: [Freeswitch-users] Gawk.exe crashes when building on Windows 7
> 64bits w/ VS
> Kinda out of options must be something specific to your system. Are you
> running a virus scanner? try disabling that.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Gawk-exe-crashes-when-building-on-Windows-7-64bits-w-VS-tp6205075p6205227.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
>
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From manavid at gmail.com  Fri Mar 25 20:51:46 2011
From: manavid at gmail.com (Moe Navid)
Date: Fri, 25 Mar 2011 10:51:46 -0700
Subject: [Freeswitch-users] Voicemail on GlusterFS
In-Reply-To: <AANLkTik7ExTSGi62uEd4fyE0s+HthySYiT9gzZjxVHx0@mail.gmail.com>
References: <AANLkTimkv66jjaBTHtT+nDUgo3PsuxEndEi-Z9=Lb0u_@mail.gmail.com>
	<AANLkTi=44afGJ1fni5P+q=0ct=irmVDwOfESUAbdWuo9@mail.gmail.com>
	<AANLkTi=bWO48xovSm_am037RWAqnCjGSFOTYTdwiVe4J@mail.gmail.com>
	<AANLkTi=hpYhE2qdfuzuxNn=b5atbb7o-DjLSfNcy8Q6e@mail.gmail.com>
	<AANLkTinDYc2RKXMrj=ycs0zKa36gW2PZpiDgSLh-ajs1@mail.gmail.com>
	<AANLkTik7ExTSGi62uEd4fyE0s+HthySYiT9gzZjxVHx0@mail.gmail.com>
Message-ID: <AANLkTi=Uo_pzfKe_VmbXjJi4RNY9Jor7EZkNxL9hvrt7@mail.gmail.com>

Thanks Eliot, I'll give it a try.

Moe

On Fri, Mar 25, 2011 at 10:39 AM, Eliot Gable
<egable+freeswitch at gmail.com>wrote:

> Also check out MooseFS. I was very impressed by how quickly I was able to
> set it up and use it. IIRC, it was < 2 hours to set up six nodes with shared
> voicemail and sounds files from start to finish.
>
>
> On Wed, Mar 23, 2011 at 2:59 PM, Dan Lane <null at invalid.name> wrote:
>
>> On Tue, Mar 22, 2011 at 6:21 PM, Moe Navid <manavid at gmail.com> wrote:
>> > Hi Dan,
>> > Thanks for the reply. I'm also working on a custom voicemail.
>> > Is your vm implemented in C/C++ and uses GlusterFS' native library or
>> you
>> > are just mounting with fuse?
>>
>> It's written in LUA and we're using FUSE to mount the GlusterFS
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Eliot Gable
>
> "We do not inherit the Earth from our ancestors: we borrow it from our
> children." ~David Brower
>
> "I decided the words were too conservative for me. We're not borrowing from
> our children, we're stealing from them--and it's not even considered to be a
> crime." ~David Brower
>
> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live;
> not live to eat.) ~Marcus Tullius Cicero
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From null at invalid.name  Fri Mar 25 22:22:18 2011
From: null at invalid.name (Dan Lane)
Date: Fri, 25 Mar 2011 19:22:18 +0000
Subject: [Freeswitch-users] transcoding GSM with a high ptime
Message-ID: <AANLkTimc_1WppCuu90n668=LZDfmVduV0X9v5yaiKDfj@mail.gmail.com>

I need to do a bit more work on this before I have enough information
to submit a Jira but just on the off-chance anyone else has had this
issue:

We have handsets using GSM with 120ms ptime but when this gets
transcoded to something else (for example 20ms PCMU) the resulting
audio is choppy as though there was a timing error. We only experience
this with GSM, 120ms packets to 20ms transcoding between other formats
is fine.

Has anyone else experienced this? I've experienced it on git-e7acd4d
(our production kit) and f3c33c7 (due to become our next production
kit)

I'll continue to test it before submitting a Jira but I wanted to know
if it was just us experiencing this.

Regards,
Dan


From anthony.minessale at gmail.com  Fri Mar 25 22:41:23 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 25 Mar 2011 14:41:23 -0500
Subject: [Freeswitch-users] mod_fifo & CID
In-Reply-To: <201103241644.27845.yivzhenko@mksat.net>
References: <201103241644.27845.yivzhenko@mksat.net>
Message-ID: <AANLkTim=ER2syNeDVWM6C6C84pqvshg7R3bNR4M2xEnf@mail.gmail.com>

when you define the agents in the config add this:

{origination_callee_id_name='agent
name',origination_callee_id_number='agents number'}

Its display update when the call changes direction from outbound to bridge.




On Thu, Mar 24, 2011 at 9:44 AM, Yuriy Ivzhenko <yivzhenko at mksat.net> wrote:
> Hi All,
>
> After upgrading to last Git (git-73ca862) i see changes in CID on
> calls to agents, generated by mod_fifo.
>
> If agent answer
> caller_profile.caller_id_name = "Outbound Call"
> caller_profile.caller_id_number = <agent number>
>
> If agent not answer
> caller_profile.caller_id_name = caller caller_id_name
> caller_profile.caller_id_number = caller caller_id_number (or
> origination_caller_id_name/number of set in agents dialstring)
> In old version, so it was always.
>
> Is this correct?
> It seems to me it should not depend on, answered the call or not.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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  Fri Mar 25 22:42:56 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 25 Mar 2011 14:42:56 -0500
Subject: [Freeswitch-users] Voicemail on GlusterFS
In-Reply-To: <63BF7F7A0106425EA39F9632F5D5A292@stor1>
References: <AANLkTimkv66jjaBTHtT+nDUgo3PsuxEndEi-Z9=Lb0u_@mail.gmail.com>
	<AANLkTi=44afGJ1fni5P+q=0ct=irmVDwOfESUAbdWuo9@mail.gmail.com>
	<63BF7F7A0106425EA39F9632F5D5A292@stor1>
Message-ID: <AANLkTi==qp2vxS5u4dmo1YVG+Kd_yabbCMs3zd+S_yJQ@mail.gmail.com>

not a bug \ is an escape character and you need 2 to imply 1 \\ will
reduce to \ by the parser.


On Wed, Mar 23, 2011 at 2:26 AM, Kris <kris at livecall.com> wrote:
> I have noticed that FS has a bug in playback and record APIs where it will
> fail if there are only 2 slashes in front of the server.
> This is what I have to do to get it to work- add a third slash. ?Something
> to try...hope it helps
>
> string filename = VoiceMessage.ConferenceRecordingFilename(ForumExt,
> ConferenceUUID);
>
> if (filename.StartsWith("\\\\"))//server path
>
> filename = "\\" + filename;//fs expects 3 slashes for some reason..while
> dealing with them only 1 is left
>
> string msg=fsApi.Execute("conference", ConferenceName + " record " +
> filename);
>
>
>
> Kris
>
> ----- Original Message -----
> From: "Dan Lane" <null at invalid.name>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, March 22, 2011 10:42 AM
> Subject: Re: [Freeswitch-users] Voicemail on GlusterFS
>
>
> On Tue, Mar 22, 2011 at 3:55 PM, Moe Navid <manavid at gmail.com> wrote:
>> Hi All,
>> Has anyone tried using voicemail on GlusterFS?
>> GlusteFS looks very promising for building distributed file systems.
>> Pandora
>> Radio recently started using
>> glusterfs
>> http://www.gluster.com/2011/01/05/gluster-to-help-manage-rapid-data-growth-for-pandora/
>> I managed to get the recording right for voicemails but retrieving the WAV
>> files fails. FreeSWITCH gives the following error:
>> 2011-03-22 08:54:06.325001 [ERR] mod_sndfile.c:194 Error Opening File
>> [/home/moe/freeswitch/storage/voicemail/default/192.168.1.10/1001/msg_02a108cc-549c-11e0-a9f6-9f1a6f4da0f7.wav]
>> [File contains data in an unknown format.]
>> Any thoughts?
>> Thanks
>> Moe
>
> We've been using GlusterFS for storing voicemail in a production
> environment for over a year now but we built a custom voicemail
> solution rather than using the built-in one.
>
> We didn't experience the issue you're referring to so I'm afraid I
> can't offer any assistance.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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  Fri Mar 25 23:39:33 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 25 Mar 2011 15:39:33 -0500
Subject: [Freeswitch-users] transcoding GSM with a high ptime
In-Reply-To: <AANLkTimc_1WppCuu90n668=LZDfmVduV0X9v5yaiKDfj@mail.gmail.com>
References: <AANLkTimc_1WppCuu90n668=LZDfmVduV0X9v5yaiKDfj@mail.gmail.com>
Message-ID: <AANLkTik=Yzg90dP=BMDZCYrJnODm0yJpR1qw42fJe1hy@mail.gmail.com>

other formats also with 120 ms ptime?

You do run the risk of losing a significant amount of audio if you
drop even 1 120ms packet.
The other end would be starved for 5 intervals.
you could try rtp_timer_name=none on the 120ms leg


On Fri, Mar 25, 2011 at 2:22 PM, Dan Lane <null at invalid.name> wrote:
> I need to do a bit more work on this before I have enough information
> to submit a Jira but just on the off-chance anyone else has had this
> issue:
>
> We have handsets using GSM with 120ms ptime but when this gets
> transcoded to something else (for example 20ms PCMU) the resulting
> audio is choppy as though there was a timing error. We only experience
> this with GSM, 120ms packets to 20ms transcoding between other formats
> is fine.
>
> Has anyone else experienced this? I've experienced it on git-e7acd4d
> (our production kit) and f3c33c7 (due to become our next production
> kit)
>
> I'll continue to test it before submitting a Jira but I wanted to know
> if it was just us experiencing this.
>
> Regards,
> Dan
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 manavid at gmail.com  Sat Mar 26 00:55:00 2011
From: manavid at gmail.com (Moe Navid)
Date: Fri, 25 Mar 2011 14:55:00 -0700
Subject: [Freeswitch-users] Voicemail on GlusterFS
In-Reply-To: <AANLkTik7ExTSGi62uEd4fyE0s+HthySYiT9gzZjxVHx0@mail.gmail.com>
References: <AANLkTimkv66jjaBTHtT+nDUgo3PsuxEndEi-Z9=Lb0u_@mail.gmail.com>
	<AANLkTi=44afGJ1fni5P+q=0ct=irmVDwOfESUAbdWuo9@mail.gmail.com>
	<AANLkTi=bWO48xovSm_am037RWAqnCjGSFOTYTdwiVe4J@mail.gmail.com>
	<AANLkTi=hpYhE2qdfuzuxNn=b5atbb7o-DjLSfNcy8Q6e@mail.gmail.com>
	<AANLkTinDYc2RKXMrj=ycs0zKa36gW2PZpiDgSLh-ajs1@mail.gmail.com>
	<AANLkTik7ExTSGi62uEd4fyE0s+HthySYiT9gzZjxVHx0@mail.gmail.com>
Message-ID: <AANLkTimFXON+riwnJQd7r1nagtJF-MhrUr4p0nNSU2Zb@mail.gmail.com>

Justed installed MooseFS, it's working like a charm :) Thanks

On Fri, Mar 25, 2011 at 10:39 AM, Eliot Gable
<egable+freeswitch at gmail.com>wrote:

> Also check out MooseFS. I was very impressed by how quickly I was able to
> set it up and use it. IIRC, it was < 2 hours to set up six nodes with shared
> voicemail and sounds files from start to finish.
>
>
> On Wed, Mar 23, 2011 at 2:59 PM, Dan Lane <null at invalid.name> wrote:
>
>> On Tue, Mar 22, 2011 at 6:21 PM, Moe Navid <manavid at gmail.com> wrote:
>> > Hi Dan,
>> > Thanks for the reply. I'm also working on a custom voicemail.
>> > Is your vm implemented in C/C++ and uses GlusterFS' native library or
>> you
>> > are just mounting with fuse?
>>
>> It's written in LUA and we're using FUSE to mount the GlusterFS
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Eliot Gable
>
> "We do not inherit the Earth from our ancestors: we borrow it from our
> children." ~David Brower
>
> "I decided the words were too conservative for me. We're not borrowing from
> our children, we're stealing from them--and it's not even considered to be a
> crime." ~David Brower
>
> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live;
> not live to eat.) ~Marcus Tullius Cicero
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From wstephen80 at gmail.com  Sat Mar 26 01:14:25 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Fri, 25 Mar 2011 23:14:25 +0100
Subject: [Freeswitch-users] Hangup a call with silence
Message-ID: <AANLkTi=T9hKyzb-KrdaKQZCg6YWet_TMLNc-NEvh3MY7@mail.gmail.com>

I have a provider that doesn't send the bye when the b-leb hangup the call
and the call is active until a-leg hangup.
Sometime happens that the a-leg forget to hangup and the call remain active.
There is a way to auto disconnect the call where there is, for example, 30s
of silence?

Stephen
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From frank at rosengart.de  Sat Mar 26 01:37:05 2011
From: frank at rosengart.de (Frank Rosengart)
Date: Fri, 25 Mar 2011 23:37:05 +0100
Subject: [Freeswitch-users] Hangup a call with silence
In-Reply-To: <AANLkTi=T9hKyzb-KrdaKQZCg6YWet_TMLNc-NEvh3MY7@mail.gmail.com>
References: <AANLkTi=T9hKyzb-KrdaKQZCg6YWet_TMLNc-NEvh3MY7@mail.gmail.com>
Message-ID: <4D8D1911.9030303@rosengart.de>

On 03/25/2011 11:14 PM, Stephen Wilde wrote:

> There is a way to auto disconnect the call where there is, for example,
> 30s of silence?

What about <param name="rtp-timeout-sec" value="30"/> in your sofia profile?

Frank


From Nabble at slickdeals.endjunk.com  Sat Mar 26 05:49:18 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 25 Mar 2011 19:49:18 -0700 (PDT)
Subject: [Freeswitch-users] mod_dingaling terminates incoming GV call if
	answer before 3rd ring
Message-ID: <1301107758212-6209770.post@n2.nabble.com>

I just did a git pull, compiled, and now my FS is running on FreeSWITCH
Version 1.0.head (git-b8d93de 2011-03-25 17-05-13 -0500). So far, so good,
except mod_dingaling seems to have encountered some problems to handle an
incoming Google Voice call. If the call is picked up before the 3rd ring, it
gets terminated as shown  http://pastebin.com/1HfHNp0z here  on line #27.
Otherwise, the incoming call is properly handled when it is picked up after
the 3rd ring.

This problem did not happen with the FreeSWITCH Version 1.0.head
(git-f3c33c7 2011-03-23 14-57-16 -0500.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-dingaling-terminates-incoming-GV-call-if-answer-before-3rd-ring-tp6209770p6209770.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From fieldpeak at gmail.com  Sat Mar 26 06:06:28 2011
From: fieldpeak at gmail.com (Charles)
Date: Sat, 26 Mar 2011 11:06:28 +0800
Subject: [Freeswitch-users] sofia_reg_internal.db
	growbiggerandbiggeralong with time
References: <04371062-578D-4D69-94DA-2D8D633FE3DE@visionutveckling.se>,
	<4d8c4085.8b3b2b0a.7686.1e19@mx.google.com>,
	<AANLkTi==wH3un+te_an8ZX8gzi2_Xub3mNbGV7J8We1L@mail.gmail.com>,
	<AANLkTikNF0ZjXBF_3iZonX2wkkRu2aOxx8hNsdBgrNt8@mail.gmail.com>,
	<AANLkTimh=bygFkZfgcw2D=PmHxzXudKA-r75-d08oc5r@mail.gmail.com>,
	<AANLkTi=fNk_uv6LyFU93no6mt1mdyzu+2DbgeASOxXac@mail.gmail.com>
Message-ID: <4d8d5850.0c87970a.7e04.1729@mx.google.com>

Hi Anthony,

I tried to delete the db files sofia_reg_* db files, it is the same.

Actually i've located the bug where the source code, when INSERTing the new record to the table sip_authentication(line#801 of sofia_reg.c), 
It compose a wrong SQL statement which caused insert the wrong values ('NULL' at 'profile' column, and 'internal' at hostname column) to columns, consequetnly, failed to delete when expires time arrive.

Please FOCUS here.

the mail below shows the details, 
thanks!

sender? Charles
sent time? 2011-03-25 12:22:57
receiver? FreeSWITCH Users Help; msc; jeff
cc? jonas.gauffin
subject? Re: Re: [Freeswitch-users] sofia_reg_internal.db grow bigger and biggeralong with time

It looks that I found the root cause, at the line#801 of sofia_reg.c, when excute the SQL statement ('INSERT') to the table 'sip_authentication', it wrongly insert as screenshot below, profile_name should be 'internal' and hostname should be 'mypc' (name of my computer).
however, i did not find anything wrong regarding the code itselft, so failed to fix it...
could you please help focus here to fix it, thanks.

P.S. i manually insert the correct record into this talbe, after a while (tens of seconds), it automatically deteleted by FS. 

btw, i found there is a bug report like this http://jira.freeswitch.org/browse/FS-498# and it is marked 
Resolved:14/Sep/10 12:32 PM, however, my version is Nov, 30, 2010, there is still exsit.





 line#801 of sofia_reg.c:
sql = switch_mprintf("insert into sip_authentication (nonce,expires,profile_name,hostname, last_nc) "
 "values('%q', %ld, '%q', '%q', 0)", uuid_str,
 switch_epoch_time_now(NULL) + (profile->nonce_ttl ? profile->nonce_ttl : DEFAULT_NONCE_TTL),
 profile->name, mod_sofia_globals.hostname);
switch_assert(sql != NULL);
sofia_glue_actually_execute_sql(profile, sql, profile->ireg_mutex);





2011-03-26 



Charles 



???? Anthony Minessale 
????? 2011-03-26  00:12:12 
???? FreeSWITCH Users Help 
??? 
??? Re: [Freeswitch-users] sofia_reg_internal.db growbiggerandbiggeralong with time 
 
can you try deleting the db files sofia_reg_* db files?  Maybe they
have become corrupt?
2011/3/25 fieldpeak <fieldpeak at gmail.com>:
> Hi Jonas,
> i also think so. can you please update was the bug fixed?
>
> ? 2011-3-25 ??5:14?"Jonas Gauffin" <jonas.gauffin at gmail.com>???
>
> might be the same bug as I reported here:
> http://jira.freeswitch.org/browse/FS-498
>
>
>
> On Fri, Mar 25, 2011 at 8:13 AM, Charles <fieldpeak at gmail.com> wrote:
>>
>> no, this is not the c...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From infos at madovsky.org  Sat Mar 26 08:00:43 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 26 Mar 2011 01:00:43 -0400
Subject: [Freeswitch-users] conference unlock and pin code
Message-ID: <BAE1717664864AA6BD72093A63126047@e1705>

if I activate lock and unlock  iteems that if lock unlock activated pincode restriction disappear...

Thanks

Franck
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From brian at microcomaustralia.com.au  Sat Mar 26 10:27:33 2011
From: brian at microcomaustralia.com.au (Brian May)
Date: Sat, 26 Mar 2011 18:27:33 +1100
Subject: [Freeswitch-users] answering calls weirdness
In-Reply-To: <AANLkTi=GC1u5oZeACdpUeF=ggT3c5AAbDPH5KAgYzTKW@mail.gmail.com>
References: <AANLkTin5mP7QzDe+CAvaPjsbxhq934P-49LARGQJwR8Y@mail.gmail.com>
	<AANLkTin8nv3N=XsPa=jepebenxRJ4EvtNSMJ3D37KZKu@mail.gmail.com>
	<AANLkTi=GC1u5oZeACdpUeF=ggT3c5AAbDPH5KAgYzTKW@mail.gmail.com>
Message-ID: <AANLkTinaEPAJ7Gc3bYb0nnguWfXvnC=XL=L+EJMgG180@mail.gmail.com>

I upgraded to the latest git version of freeswitch and reproduced the
same issues.

Latest log file now:

http://pastebin.freeswitch.org/15860

Where does it get 6201 from? Surely freeswitch doesn't normally pull
phone numbers out of thin air???

-- 
Brian May <brian at microcomaustralia.com.au>


From brian at microcomaustralia.com.au  Sat Mar 26 10:50:47 2011
From: brian at microcomaustralia.com.au (Brian May)
Date: Sat, 26 Mar 2011 18:50:47 +1100
Subject: [Freeswitch-users] answering calls weirdness
In-Reply-To: <AANLkTinaEPAJ7Gc3bYb0nnguWfXvnC=XL=L+EJMgG180@mail.gmail.com>
References: <AANLkTin5mP7QzDe+CAvaPjsbxhq934P-49LARGQJwR8Y@mail.gmail.com>
	<AANLkTin8nv3N=XsPa=jepebenxRJ4EvtNSMJ3D37KZKu@mail.gmail.com>
	<AANLkTi=GC1u5oZeACdpUeF=ggT3c5AAbDPH5KAgYzTKW@mail.gmail.com>
	<AANLkTinaEPAJ7Gc3bYb0nnguWfXvnC=XL=L+EJMgG180@mail.gmail.com>
Message-ID: <AANLkTikBtvfpR4ryNXCcUhMdEqubVjw7fZ1SRR_WqHHX@mail.gmail.com>

On 26 March 2011 18:27, Brian May <brian at microcomaustralia.com.au> wrote:
> I upgraded to the latest git version of freeswitch and reproduced the
> same issues.
>
> Latest log file now:
>
> http://pastebin.freeswitch.org/15860
>
> Where does it get 6201 from? Surely freeswitch doesn't normally pull
> phone numbers out of thin air???

Also worth noting that this only occurs when dialing group numbers,
calls to single destinations work fine.


Also callerid seems to be broken, I suspect this broke when I updated
to the latest freeswitch, I removed the patch to fix freetdm, is this
still required?

http://pastebin.freeswitch.org/15861

I would have hoped these changes had already been incorporated into
the upstream repository by now...
-- 
Brian May <brian at microcomaustralia.com.au>


From peter.olsson at visionutveckling.se  Sat Mar 26 11:08:15 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sat, 26 Mar 2011 09:08:15 +0100
Subject: [Freeswitch-users]
	sofia_reg_internal.db	growbiggerandbiggeralong with time
In-Reply-To: <4d8d5850.0c87970a.7e04.1729@mx.google.com>
References: <04371062-578D-4D69-94DA-2D8D633FE3DE@visionutveckling.se>,
	<4d8c4085.8b3b2b0a.7686.1e19@mx.google.com>,
	<AANLkTi==wH3un+te_an8ZX8gzi2_Xub3mNbGV7J8We1L@mail.gmail.com>,
	<AANLkTikNF0ZjXBF_3iZonX2wkkRu2aOxx8hNsdBgrNt8@mail.gmail.com>,
	<AANLkTimh=bygFkZfgcw2D=PmHxzXudKA-r75-d08oc5r@mail.gmail.com>,
	<AANLkTi=fNk_uv6LyFU93no6mt1mdyzu+2DbgeASOxXac@mail.gmail.com>,
	<4d8d5850.0c87970a.7e04.1729@mx.google.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2C495A8@cooper>

If you've found a bug - please submit it to Jira. Also, make sure to attach your sip profile config to it.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Charles [fieldpeak at gmail.com]
Skickat: den 26 mars 2011 04:06
Till: FreeSWITCH Users Help; FreeSWITCH Users Help; anthony.minessale
?mne: Re: [Freeswitch-users] sofia_reg_internal.db      growbiggerandbiggeralong with time

Hi Anthony,

I tried to delete the db files sofia_reg_* db files, it is the same.

Actually i've located the bug where the source code, when INSERTing the new record to the table sip_authentication(line#801 of sofia_reg.c),
It compose a wrong SQL statement which caused insert the wrong values ('NULL' at 'profile' column, and 'internal' at hostname column) to columns, consequetnly, failed to delete when expires time arrive.

Please FOCUS here.

the mail below shows the details,
thanks!

sender? Charles
sent time? 2011-03-25 12:22:57
receiver? FreeSWITCH Users Help; msc; jeff
cc? jonas.gauffin
subject? Re: Re: [Freeswitch-users] sofia_reg_internal.db grow bigger and biggeralong with time

It looks that I found the root cause, at the line#801 of sofia_reg.c, when excute the SQL statement ('INSERT') to the table 'sip_authentication', it wrongly insert as screenshot below, profile_name should be 'internal' and hostname should be 'mypc' (name of my computer).
however, i did not find anything wrong regarding the code itselft, so failed to fix it...
could you please help focus here to fix it, thanks.

P.S. i manually insert the correct record into this talbe, after a while (tens of seconds), it automatically deteleted by FS.

btw, i found there is a bug report like this http://jira.freeswitch.org/browse/FS-498# and it is marked

Resolved:14/Sep/10 12:32 PM, however, my version is Nov, 30, 2010, there is still exsit.


[cid:__0 at Foxmail.net]


 line#801 of sofia_reg.c:
sql = switch_mprintf("insert into sip_authentication (nonce,expires,profile_name,hostname, last_nc) "
 "values('%q', %ld, '%q', '%q', 0)", uuid_str,
 switch_epoch_time_now(NULL) + (profile->nonce_ttl ? profile->nonce_ttl : DEFAULT_NONCE_TTL),
 profile->name, mod_sofia_globals.hostname);
switch_assert(sql != NULL);
sofia_glue_actually_execute_sql(profile, sql, profile->ireg_mutex);





2011-03-26
________________________________
Charles
________________________________
???? Anthony Minessale
????? 2011-03-26  00:12:12
???? FreeSWITCH Users Help
???
??? Re: [Freeswitch-users] sofia_reg_internal.db growbiggerandbiggeralong with time
can you try deleting the db files sofia_reg_* db files?  Maybe they
have become corrupt?
2011/3/25 fieldpeak <fieldpeak at gmail.com>:
> Hi Jonas,
> i also think so. can you please update was the bug fixed?
>
> ? 2011-3-25 ??5:14?"Jonas Gauffin" <jonas.gauffin at gmail.com>???
>
> might be the same bug as I reported here:
> http://jira.freeswitch.org/browse/FS-498
>
>
>
> On Fri, Mar 25, 2011 at 8:13 AM, Charles <fieldpeak at gmail.com> wrote:
>>
>> no, this is not the c...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
!DSPAM:4d8d70e832765221816297!

From fieldpeak at gmail.com  Sat Mar 26 13:33:18 2011
From: fieldpeak at gmail.com (=?utf-8?B?Q2hhcmxlcw==?=)
Date: Sat, 26 Mar 2011 18:33:18 +0800
Subject: [Freeswitch-users]
	=?utf-8?q?sofia=5Freg=5Finternal=2Edb=09growbi?=
	=?utf-8?q?ggerandbiggeralong_with_time?=
References: <04371062-578D-4D69-94DA-2D8D633FE3DE@visionutveckling.se>,
	<4d8c4085.8b3b2b0a.7686.1e19@mx.google.com>,
	<AANLkTi==wH3un+te_an8ZX8gzi2_Xub3mNbGV7J8We1L@mail.gmail.com>,
	<AANLkTikNF0ZjXBF_3iZonX2wkkRu2aOxx8hNsdBgrNt8@mail.gmail.com>,
	<AANLkTimh=bygFkZfgcw2D=PmHxzXudKA-r75-d08oc5r@mail.gmail.com>,
	<AANLkTi=fNk_uv6LyFU93no6mt1mdyzu+2DbgeASOxXac@mail.gmail.com>,
	<4d8d5850.0c87970a.7e04.1729@mx.google.com>,
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2C495A8@cooper>
Message-ID: <4d8dc0f8.6219e70a.4bee.0faa@mx.google.com>

Hi Peter,

Thanks for reminding, i've submit it to Jira.

FreeSWITCH 
FS-3190
Table 'sip_authentication' was not cleared after nonce expired -caused sofia_reg_internal.db grow bigger and bigger along with time


2011-03-26 



Charles 



???? Peter Olsson 
????? 2011-03-26  16:14:02 
???? FreeSWITCH Users Help 
??? 
??? Re: [Freeswitch-users]sofia_reg_internal.db growbiggerandbiggeralong with time 
 
If you've found a bug - please submit it to Jira. Also, make sure to attach your sip profile config to it.
/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Charles [fieldpeak at gmail.com]
Skickat: den 26 mars 2011 04:06
Till: FreeSWITCH Users Help; FreeSWITCH Users Help; anthony.minessale
?mne: Re: [Freeswitch-users] sofia_reg_internal.db      growbiggerandbiggeralong with time
Hi Anthony,
I tried to delete the db files sofia_reg_* db files, it is the same.
Actually i've located the bug where the source code, when INSERTing the new record to the table sip_authentication(line#801 of sofia_reg.c),
It compose a wrong SQL statement which caused insert the wrong values ('NULL' at 'profile' column, and 'internal' at hostname column) to columns, consequetnly, failed to delete when expires time arrive.
Please FOCUS here.
the mail below shows the details,
thanks!
sender? Charles
sent time? 2011-03-25 12:22:57
receiver? FreeSWITCH Users Help; msc; jeff
cc? jonas.gauffin
subject? Re: Re: [Freeswitch-users] sofia_reg_internal.db grow bigger and biggeralong with time
It looks that I found the root cause, at the line#801 of sofia_reg.c, when excute the SQL statement ('INSERT') to the table 'sip_authentication', it wrongly insert as screenshot below, profile_name should be 'internal' and hostname should be 'mypc' (name of my computer).
however, i did not find anything wrong regarding the code itselft, so failed to fix it...
could you please help focus here to fix it, thanks.
P.S. i manually insert the correct record into this talbe, after a while (tens of seconds), it automatically deteleted by FS.
btw, i found there is a bug report like this http://jira.freeswitch.org/browse/FS-498# and it is marked
Resolved:14/Sep/10 12:32 PM, however, my version is Nov, 30, 2010, there is still exsit.
[cid:__0 at Foxmail.net]
 line#801 of sofia_reg.c:
sql = switch_mprintf("insert into sip_authentication (nonce,expires,profile_name,hostname, last_nc) "
 "values('%q', %ld, '%q', '%q', 0)", uuid_str,
 switch_epoch_time_now(NULL) + (profile->nonce_ttl ? profile->nonce_ttl : DEFAULT_NONCE_TTL),
 profile->name, mod_sofia_globals.hostname);
switch_assert(sql != NULL);
sofia_glue_actually_execute_sql(profile, sql, profile->ireg_mutex);
2011-03-26
________________________________
Charles
________________________________
???? Anthony Minessale
????? 2011-03-26  00:12:12
???? FreeSWITCH Users Help
???
??? Re: [Freeswitch-users] sofia_reg_internal.db growbiggerandbiggeralong with time
can you try deleting the db files sofia_reg_* db files?  Maybe they
have become corrupt?
2011/3/25 fieldpeak <fieldpeak at gmail.com>:
> Hi Jonas,
> i also think so. can you please update was the bug fixed?
>
> ? 2011-3-25 ??5:14?"Jonas Gauffin" <jonas.gauffin at gmail.com>???
>
> might be the same bug as I reported here:
> http://jira.freeswitch.org/browse/FS-498
>
>
>
> On Fri, Mar 25, 2011 at 8:13 AM, Charles <fieldpeak at gmail.com> wrote:
>>
>> no, this is not the c...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
!DSPAM:4d8d70e832765221816297!
_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Sat Mar 26 17:59:54 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 26 Mar 2011 10:59:54 -0400
Subject: [Freeswitch-users]
	sofia_reg_internal.db	growbiggerandbiggeralong with time
References: <04371062-578D-4D69-94DA-2D8D633FE3DE@visionutveckling.se>,
	<4d8c4085.8b3b2b0a.7686.1e19@mx.google.com>,
	<AANLkTi==wH3un+te_an8ZX8gzi2_Xub3mNbGV7J8We1L@mail.gmail.com>,
	<AANLkTikNF0ZjXBF_3iZonX2wkkRu2aOxx8hNsdBgrNt8@mail.gmail.com>,
	<AANLkTimh=bygFkZfgcw2D=PmHxzXudKA-r75-d08oc5r@mail.gmail.com>,
	<AANLkTi=fNk_uv6LyFU93no6mt1mdyzu+2DbgeASOxXac@mail.gmail.com>,
	<4d8d5850.0c87970a.7e04.1729@mx.google.com>,
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2C495A8@cooper>
	<4d8dc0f8.6219e70a.4bee.0faa@mx.google.com>
Message-ID: <8FCF27AAD14D468EB268E59A4CABC2EE@e1705>

seems that even the text of your email becomes bigger and bigger :)

  ----- Original Message ----- 
  From: Charles 
  To: FreeSWITCH Users Help ; FreeSWITCH Users Help 
  Sent: Saturday, March 26, 2011 6:33 AM
  Subject: Re: [Freeswitch-users]sofia_reg_internal.db growbiggerandbiggeralong with time


  Hi Peter,

  Thanks for reminding, i've submit it to Jira.

    a.. FreeSWITCH 
    b.. FS-3190
  Table 'sip_authentication' was not cleared after nonce expired -caused sofia_reg_internal.db grow bigger and bigger along with time


  2011-03-26 

------------------------------------------------------------------------------

  Charles 

------------------------------------------------------------------------------

  ???? Peter Olsson 
  ????? 2011-03-26  16:14:02 
  ???? FreeSWITCH Users Help 
  ??? 
  ??? Re: [Freeswitch-users]sofia_reg_internal.db growbiggerandbiggeralong with time 
  If you've found a bug - please submit it to Jira. Also, make sure to attach your sip profile config to it.
  /Peter
  ________________________________________
  Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Charles [fieldpeak at gmail.com]
  Skickat: den 26 mars 2011 04:06
  Till: FreeSWITCH Users Help; FreeSWITCH Users Help; anthony.minessale
  ?mne: Re: [Freeswitch-users] sofia_reg_internal.db      growbiggerandbiggeralong with time
  Hi Anthony,
  I tried to delete the db files sofia_reg_* db files, it is the same.
  Actually i've located the bug where the source code, when INSERTing the new record to the table sip_authentication(line#801 of sofia_reg.c),
  It compose a wrong SQL statement which caused insert the wrong values ('NULL' at 'profile' column, and 'internal' at hostname column) to columns, consequetnly, failed to delete when expires time arrive.
  Please FOCUS here.
  the mail below shows the details,
  thanks!
  sender? Charles
  sent time? 2011-03-25 12:22:57
  receiver? FreeSWITCH Users Help; msc; jeff
  cc? jonas.gauffin
  subject? Re: Re: [Freeswitch-users] sofia_reg_internal.db grow bigger and biggeralong with time
  It looks that I found the root cause, at the line#801 of sofia_reg.c, when excute the SQL statement ('INSERT') to the table 'sip_authentication', it wrongly insert as screenshot below, profile_name should be 'internal' and hostname should be 'mypc' (name of my computer).
  however, i did not find anything wrong regarding the code itselft, so failed to fix it...
  could you please help focus here to fix it, thanks.
  P.S. i manually insert the correct record into this talbe, after a while (tens of seconds), it automatically deteleted by FS.
  btw, i found there is a bug report like this http://jira.freeswitch.org/browse/FS-498# and it is marked
  Resolved:14/Sep/10 12:32 PM, however, my version is Nov, 30, 2010, there is still exsit.
  [cid:__0 at Foxmail.net]
   line#801 of sofia_reg.c:
  sql = switch_mprintf("insert into sip_authentication (nonce,expires,profile_name,hostname, last_nc) "
   "values('%q', %ld, '%q', '%q', 0)", uuid_str,
   switch_epoch_time_now(NULL) + (profile->nonce_ttl ? profile->nonce_ttl : DEFAULT_NONCE_TTL),
   profile->name, mod_sofia_globals.hostname);
  switch_assert(sql != NULL);
  sofia_glue_actually_execute_sql(profile, sql, profile->ireg_mutex);
  2011-03-26
  ________________________________
  Charles
  ________________________________
  ???? Anthony Minessale
  ????? 2011-03-26  00:12:12
  ???? FreeSWITCH Users Help
  ???
  ??? Re: [Freeswitch-users] sofia_reg_internal.db growbiggerandbiggeralong with time
  can you try deleting the db files sofia_reg_* db files?  Maybe they
  have become corrupt?
  2011/3/25 fieldpeak <fieldpeak at gmail.com>:
  > Hi Jonas,
  > i also think so. can you please update was the bug fixed?
  >
  > ? 2011-3-25 ??5:14?"Jonas Gauffin" <jonas.gauffin at gmail.com>???
  >
  > might be the same bug as I reported here:
  > http://jira.freeswitch.org/browse/FS-498
  >
  >
  >
  > On Fri, Mar 25, 2011 at 8:13 AM, Charles <fieldpeak at gmail.com> wrote:
  >>
  >> no, this is not the c...
  >
  > _______________________________________________
  > FreeSWITCH-users mailing list
  > FreeSWITCH-users at lists.freeswitch.org
  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  > http://www.freeswitch.org
  >
  >
  > _______________________________________________
  > FreeSWITCH-users mailing list
  > FreeSWITCH-users at lists.freeswitch.org
  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  > 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-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
  !DSPAM:4d8d70e832765221816297!
  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org


------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From Nabble at slickdeals.endjunk.com  Sat Mar 26 21:43:26 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sat, 26 Mar 2011 11:43:26 -0700 (PDT)
Subject: [Freeswitch-users] mod_dingaling terminates incoming GV call if
 answer before 3rd ring
In-Reply-To: <1301107758212-6209770.post@n2.nabble.com>
References: <1301107758212-6209770.post@n2.nabble.com>
Message-ID: <1301165006938-6211076.post@n2.nabble.com>

I just did another git pull, compiled, and now my FS is running FreeSWITCH
Version 1.0.head (git-9795dd2 2011-03-26 11-07-34 -0500) to happily serve
incoming GV calls. So far, no more problems for mod_dingaling to process
incoming calls even if answered on the 1st ring. Thanks.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-dingaling-terminates-incoming-GV-call-if-answer-before-3rd-ring-tp6209770p6211076.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jcasale at activenetwerx.com  Sat Mar 26 21:48:56 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sat, 26 Mar 2011 18:48:56 +0000
Subject: [Freeswitch-users] Phone opinions
Message-ID: <assp.0066b9f461.92097A6A775D5147B1078E3F15430B922D7C58@prato.activenetwerx.local>

Hey guys,
I need to grab a few phones to be used in a shop environment so spending
a lot of money isn't desired as they will probably get wrecked soon. My only
experience is with Aastra and Snom, the Aastra's I have are expensive and
work well (at least sound quality is good), the Snom's not so.

Given all the <$100.00 Grandstream's and Polycom's, anyone have an opinion
on what works best w/ FS? None of our existing phones have configs handed
out via tftp but its desired that these would so that replacement is easy.

Thanks for any opinions...
jlc



From Nabble at slickdeals.endjunk.com  Sat Mar 26 22:23:16 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sat, 26 Mar 2011 12:23:16 -0700 (PDT)
Subject: [Freeswitch-users] [CRIT] switch_loadable_module.c:928 Error
	loading module ...
Message-ID: <1301167396505-6211171.post@n2.nabble.com>

With FreeSWITCH Version 1.0.head (git-9795dd2 2011-03-26 11-07-34 -0500) and
many versions before, I noticed (but forgot to report) the following error
messages if I enable some of the modules, i.e. mod_erlang_event, mod_lcr,
etc., even though I have successfully cross compiled the modules and
installed. I don't know if this has anything to do with a cross compilation.
2011-03-26 14:32:36.251260 [CRIT] switch_loadable_module.c:928 Error Loading
module /usr/lib/freeswitch/mod_erlang_event.so
**Unknown error**
2011-03-26 14:32:43.288400 [CRIT] switch_loadable_module.c:928 Error Loading
module /usr/lib/freeswitch/mod_lcr.so
**Module load routine returned an error**

I have both modules compiled and installed as seen below. The corrupted
section header size trailing message returned by file utility is harmless,
AFAICT. Otherwise, the /usr/bin/freeswitch and among other programs listed
below, i.e. busybox and dropbear, won't even run.
root at DockStar:/# ls -l /usr/lib/freeswitch/mod_erlang_event.so
/usr/lib/freeswitch/mod_lcr.so
-rwxr-xr-x    1 root     root         58794 Mar 26 14:00
/usr/lib/freeswitch/mod_erlang_event.so
-rwxr-xr-x    1 root     root         39814 Mar 26 14:00
/usr/lib/freeswitch/mod_lcr.so
root at DockStar:/# file /usr/lib/freeswitch/mod_erlang_event.so
/usr/lib/freeswitch/mod_lcr.so /usr/lib/freeswitch/mod_sofia.so /bin/busybox
/usr/sbin/dropbear /usr/bin/freeswitch
/usr/lib/freeswitch/mod_erlang_event.so: ELF 32-bit LSB shared object, ARM,
version 1 (SYSV), dynamically linked, corrupted section header size
/usr/lib/freeswitch/mod_lcr.so:          ELF 32-bit LSB shared object, ARM,
version 1 (SYSV), dynamically linked, corrupted section header size
/usr/lib/freeswitch/mod_sofia.so:        ELF 32-bit LSB shared object, ARM,
version 1 (SYSV), dynamically linked, corrupted section header size
/bin/busybox: ELF 32-bit LSB executable, ARM, version 1 (SYSV), dynamically
linked (uses shared libs), corrupted section header size
/usr/sbin/dropbear: ELF 32-bit LSB executable, ARM, version 1 (SYSV),
dynamically linked (uses shared libs), corrupted section header size
/usr/bin/freeswitch: ELF 32-bit LSB executable, ARM, version 1 (SYSV),
dynamically linked (uses shared libs), corrupted section header size
root at DockStar:/#


-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/CRIT-switch-loadable-module-c-928-Error-loading-module-tp6211171p6211171.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jacredit at gmail.com  Sat Mar 26 22:30:41 2011
From: jacredit at gmail.com (John Hyde)
Date: Sat, 26 Mar 2011 12:30:41 -0700
Subject: [Freeswitch-users] Phone opinions
In-Reply-To: <assp.0066b9f461.92097A6A775D5147B1078E3F15430B922D7C58@prato.activenetwerx.local>
References: <assp.0066b9f461.92097A6A775D5147B1078E3F15430B922D7C58@prato.activenetwerx.local>
Message-ID: <AANLkTim_cAqSYjH+Anb2Oiro9V8fr1+eKkGHgOuXjvis@mail.gmail.com>

Actually - Aastra has the 30i (1 LAN) and 31i which you can find for $100 or
less. And in my opinion - Aastra is the easiest to write tftp configs for,
even after writing Polycoms for 10 years, I stumble, and their web gui is
not much better.

Regards,

John

On Sat, Mar 26, 2011 at 11:48 AM, Joseph L. Casale <
jcasale at activenetwerx.com> wrote:

> Hey guys,
> I need to grab a few phones to be used in a shop environment so spending
> a lot of money isn't desired as they will probably get wrecked soon. My
> only
> experience is with Aastra and Snom, the Aastra's I have are expensive and
> work well (at least sound quality is good), the Snom's not so.
>
> Given all the <$100.00 Grandstream's and Polycom's, anyone have an opinion
> on what works best w/ FS? None of our existing phones have configs handed
> out via tftp but its desired that these would so that replacement is easy.
>
> Thanks for any opinions...
> jlc
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
- j
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From infos at madovsky.org  Sat Mar 26 22:42:02 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 26 Mar 2011 15:42:02 -0400
Subject: [Freeswitch-users] bridge loop attribute
Message-ID: <7AB85CAEC3C84072A8A6454BD294AD01@e1705>

is loop attribute be used (syntax?) in CLI dialstring ?

thanks
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From philippe at ppmt.org  Sat Mar 26 22:59:43 2011
From: philippe at ppmt.org (Philippe Le Toquin)
Date: Sat, 26 Mar 2011 15:59:43 -0400
Subject: [Freeswitch-users] problem with failed called when dialled from
	phone memory
Message-ID: <4D8E45AF.9090009@ppmt.org>

Hello,

I have received a complaint from my wife and as every married person
will know it has priority over anything else!

I have freeswitch compiled on my Guruplug and it works fine except that
sometime outgoing calls fail.
I can see it in the failed outgoing list from sofia

Apparently this happens when the number is dialled from the phone memory
(but sometime it works as well so
not sure there is a pattern)

I have a trace for when it works and one where it doesn't work but I am
not good enough to read and fully understand the trace :(

Is there someone that could look into them and guide me? I can provide
the logs but before that I would like to know if there are DOs  and
DON't for providing logs

a second question would be: is there a limit to how fast you can dial a
number?

Thanks

Philippe
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From jcasale at activenetwerx.com  Sat Mar 26 23:34:14 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sat, 26 Mar 2011 20:34:14 +0000
Subject: [Freeswitch-users] Phone opinions
In-Reply-To: <AANLkTim_cAqSYjH+Anb2Oiro9V8fr1+eKkGHgOuXjvis@mail.gmail.com>
References: <92097A6A775D5147B1078E3F15430B922D7C58@prato.activenetwerx.local>
	<AANLkTim_cAqSYjH+Anb2Oiro9V8fr1+eKkGHgOuXjvis@mail.gmail.com>
Message-ID: <assp.0066776003.92097A6A775D5147B1078E3F15430B922DBE69@prato.activenetwerx.local>

>Actually - Aastra has the 30i (1 LAN) and 31i which you can find for $100 or less. And in my opinion -
>Aastra is the easiest to write tftp configs for, even after writing Polycoms for 10 years, I stumble,
>and their web gui is not much better.

Very much appreciate that info. Aastra it is, those 6730/1i's fit the bill...
Thanks John,
jlc



From infos at madovsky.org  Sun Mar 27 00:17:20 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 26 Mar 2011 17:17:20 -0400
Subject: [Freeswitch-users] invite in conference
Message-ID: <36A45615B7C24998A37DC5C720388B14@e1705>

I noticed that in case of hot invitation in conference
the welcome sound starts while the invited leg is still ringing

Thanks

Franck
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From frank at rosengart.de  Sun Mar 27 03:11:38 2011
From: frank at rosengart.de (Frank Rosengart)
Date: Sun, 27 Mar 2011 00:11:38 +0100
Subject: [Freeswitch-users] Phone opinions
In-Reply-To: <assp.0066b9f461.92097A6A775D5147B1078E3F15430B922D7C58@prato.activenetwerx.local>
References: <assp.0066b9f461.92097A6A775D5147B1078E3F15430B922D7C58@prato.activenetwerx.local>
Message-ID: <4D8E72AA.6030900@rosengart.de>

On 03/26/2011 07:48 PM, Joseph L. Casale wrote:

> My only
> experience is with Aastra and Snom, the Aastra's I have are expensive and
> work well (at least sound quality is good), the Snom's not so.

What's wrong with the audio of Snom phones? What codec are you using?

Frank


From rhuddleston at gmail.com  Sun Mar 27 04:55:21 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Sat, 26 Mar 2011 20:55:21 -0400
Subject: [Freeswitch-users] Mod_xml_cdr BLeg Account Code
Message-ID: <26ef01cbec19$a77a0380$f66e0a80$@com>

Anyone have any suggestions on obtaining / setting the Account Code for BLeg
for Mod_xml_cdr.

 

I applied accountcode but that is only handling for ALeg. I also looked at
effective_caller_id_number - but I need to dynamically (at time of bridging
a-leg to b-leg) define the accountcode?

 

Probably some custom coding I would assume to pull this off.

 

Any thoughts.

 

Thanks

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From steveayre at gmail.com  Sun Mar 27 05:34:10 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 27 Mar 2011 02:34:10 +0100
Subject: [Freeswitch-users] [CRIT] switch_loadable_module.c:928 Error
 loading module ...
In-Reply-To: <1301167396505-6211171.post@n2.nabble.com>
References: <1301167396505-6211171.post@n2.nabble.com>
Message-ID: <AANLkTi=G4bVeP84Boz6gUTK6Up1Tc+ebn-YhJo8nY5Sz@mail.gmail.com>

Are there any other messages before the CRIT lines? More context would be
useful. It could be that it's loading, but failing to load a dependancy or
config file.

-Steve



On 26 March 2011 19:23, mazilo <Nabble at slickdeals.endjunk.com> wrote:

> With FreeSWITCH Version 1.0.head (git-9795dd2 2011-03-26 11-07-34 -0500)
> and
> many versions before, I noticed (but forgot to report) the following error
> messages if I enable some of the modules, i.e. mod_erlang_event, mod_lcr,
> etc., even though I have successfully cross compiled the modules and
> installed. I don't know if this has anything to do with a cross
> compilation.
> 2011-03-26 14:32:36.251260 [CRIT] switch_loadable_module.c:928 Error
> Loading
> module /usr/lib/freeswitch/mod_erlang_event.so
> **Unknown error**
> 2011-03-26 14:32:43.288400 [CRIT] switch_loadable_module.c:928 Error
> Loading
> module /usr/lib/freeswitch/mod_lcr.so
> **Module load routine returned an error**
>
> I have both modules compiled and installed as seen below. The corrupted
> section header size trailing message returned by file utility is harmless,
> AFAICT. Otherwise, the /usr/bin/freeswitch and among other programs listed
> below, i.e. busybox and dropbear, won't even run.
> root at DockStar:/# ls -l /usr/lib/freeswitch/mod_erlang_event.so
> /usr/lib/freeswitch/mod_lcr.so
> -rwxr-xr-x    1 root     root         58794 Mar 26 14:00
> /usr/lib/freeswitch/mod_erlang_event.so
> -rwxr-xr-x    1 root     root         39814 Mar 26 14:00
> /usr/lib/freeswitch/mod_lcr.so
> root at DockStar:/# file /usr/lib/freeswitch/mod_erlang_event.so
> /usr/lib/freeswitch/mod_lcr.so /usr/lib/freeswitch/mod_sofia.so
> /bin/busybox
> /usr/sbin/dropbear /usr/bin/freeswitch
> /usr/lib/freeswitch/mod_erlang_event.so: ELF 32-bit LSB shared object, ARM,
> version 1 (SYSV), dynamically linked, corrupted section header size
> /usr/lib/freeswitch/mod_lcr.so:          ELF 32-bit LSB shared object, ARM,
> version 1 (SYSV), dynamically linked, corrupted section header size
> /usr/lib/freeswitch/mod_sofia.so:        ELF 32-bit LSB shared object, ARM,
> version 1 (SYSV), dynamically linked, corrupted section header size
> /bin/busybox: ELF 32-bit LSB executable, ARM, version 1 (SYSV), dynamically
> linked (uses shared libs), corrupted section header size
> /usr/sbin/dropbear: ELF 32-bit LSB executable, ARM, version 1 (SYSV),
> dynamically linked (uses shared libs), corrupted section header size
> /usr/bin/freeswitch: ELF 32-bit LSB executable, ARM, version 1 (SYSV),
> dynamically linked (uses shared libs), corrupted section header size
> root at DockStar:/#
>
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/CRIT-switch-loadable-module-c-928-Error-loading-module-tp6211171p6211171.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Sun Mar 27 05:36:17 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 27 Mar 2011 02:36:17 +0100
Subject: [Freeswitch-users] Mod_xml_cdr BLeg Account Code
In-Reply-To: <26ef01cbec19$a77a0380$f66e0a80$@com>
References: <26ef01cbec19$a77a0380$f66e0a80$@com>
Message-ID: <AANLkTikYyPE8BUfc6xhmt27P4Zn5CquSeVZcFq3FFHBi@mail.gmail.com>

Use the export app to export the variable from the aleg from the bleg.

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_export

-Steve



On 27 March 2011 00:55, Robert Huddleston <rhuddleston at gmail.com> wrote:

> Anyone have any suggestions on obtaining / setting the Account Code for
> BLeg for Mod_xml_cdr.
>
>
>
> I applied accountcode but that is only handling for ALeg. I also looked at
> effective_caller_id_number ? but I need to dynamically (at time of bridging
> a-leg to b-leg) define the accountcode?
>
>
>
> Probably some custom coding I would assume to pull this off?
>
>
>
> Any thoughts?
>
>
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Sun Mar 27 05:38:04 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 27 Mar 2011 02:38:04 +0100
Subject: [Freeswitch-users] bridge loop attribute
In-Reply-To: <7AB85CAEC3C84072A8A6454BD294AD01@e1705>
References: <7AB85CAEC3C84072A8A6454BD294AD01@e1705>
Message-ID: <AANLkTimkDfbo++7G19e6LGGTCJBnbx9NrHU9YQ=CB-qW@mail.gmail.com>

No, it's part of the XML dialplan.

You can repeat the route in the dialstring by using the | separator though:
sofia/gateway/gw1/12345|sofia/gateway/gw1/12345|sofia/gateway/gw1/12345

-Steve


On 26 March 2011 19:42, Madovsky <infos at madovsky.org> wrote:

>  is loop attribute be used (syntax?) in CLI dialstring ?
>
> thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Sun Mar 27 05:38:35 2011
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 26 Mar 2011 18:38:35 -0700
Subject: [Freeswitch-users] Mod_xml_cdr BLeg Account Code
In-Reply-To: <26ef01cbec19$a77a0380$f66e0a80$@com>
References: <26ef01cbec19$a77a0380$f66e0a80$@com>
Message-ID: <AANLkTi=Kns-bKdQjSuDXhSUxkR7vyMzp0mnycbDyMp7E@mail.gmail.com>

You can export it or set it in the dialstring. Just be sure to use "nolocal"
if you export it:

<action application="export" data="nolocal:accountcode=12345"/>

Or do this:

<action application="bridge" data="{accountcode=12345}sofia/foo/bar at baz"/>

-MC

On Sat, Mar 26, 2011 at 5:55 PM, Robert Huddleston <rhuddleston at gmail.com>wrote:

> Anyone have any suggestions on obtaining / setting the Account Code for
> BLeg for Mod_xml_cdr.
>
>
>
> I applied accountcode but that is only handling for ALeg. I also looked at
> effective_caller_id_number ? but I need to dynamically (at time of bridging
> a-leg to b-leg) define the accountcode?
>
>
>
> Probably some custom coding I would assume to pull this off?
>
>
>
> Any thoughts?
>
>
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rhuddleston at gmail.com  Sun Mar 27 05:41:02 2011
From: rhuddleston at gmail.com (Rob Huddleston)
Date: Sat, 26 Mar 2011 21:41:02 -0400
Subject: [Freeswitch-users] Mod_xml_cdr BLeg Account Code
In-Reply-To: <AANLkTi=Kns-bKdQjSuDXhSUxkR7vyMzp0mnycbDyMp7E@mail.gmail.com>
References: <26ef01cbec19$a77a0380$f66e0a80$@com>
	<AANLkTi=Kns-bKdQjSuDXhSUxkR7vyMzp0mnycbDyMp7E@mail.gmail.com>
Message-ID: <AANLkTikKNqe5g_B7d=D2aB-b8AfS3FWKvREriLyeYZGt@mail.gmail.com>

Duh... completely forgot about the preceding data in the gateway call...

Thanks a bunch

On Sat, Mar 26, 2011 at 9:38 PM, Michael Collins <msc at freeswitch.org> wrote:

> You can export it or set it in the dialstring. Just be sure to use
> "nolocal" if you export it:
>
> <action application="export" data="nolocal:accountcode=12345"/>
>
> Or do this:
>
> <action application="bridge" data="{accountcode=12345}sofia/foo/bar at baz"/>
>
> -MC
>
> On Sat, Mar 26, 2011 at 5:55 PM, Robert Huddleston <rhuddleston at gmail.com>wrote:
>
>> Anyone have any suggestions on obtaining / setting the Account Code for
>> BLeg for Mod_xml_cdr.
>>
>>
>>
>> I applied accountcode but that is only handling for ALeg. I also looked at
>> effective_caller_id_number ? but I need to dynamically (at time of bridging
>> a-leg to b-leg) define the accountcode?
>>
>>
>>
>> Probably some custom coding I would assume to pull this off?
>>
>>
>>
>> Any thoughts?
>>
>>
>>
>> Thanks
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
I could put... strychnine in the guacamole
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From fieldpeak at gmail.com  Sun Mar 27 06:15:33 2011
From: fieldpeak at gmail.com (=?utf-8?B?Q2hhcmxlcw==?=)
Date: Sun, 27 Mar 2011 10:15:33 +0800
Subject: [Freeswitch-users]
	=?utf-8?q?sofia=5Freg=5Finternal=2Edb=09growbi?=
	=?utf-8?q?ggerandbiggeralong_with_time?=
References: <04371062-578D-4D69-94DA-2D8D633FE3DE@visionutveckling.se>,
	<4d8c4085.8b3b2b0a.7686.1e19@mx.google.com>,
	<AANLkTi==wH3un+te_an8ZX8gzi2_Xub3mNbGV7J8We1L@mail.gmail.com>,
	<AANLkTikNF0ZjXBF_3iZonX2wkkRu2aOxx8hNsdBgrNt8@mail.gmail.com>,
	<AANLkTimh=bygFkZfgcw2D=PmHxzXudKA-r75-d08oc5r@mail.gmail.com>,
	<AANLkTi=fNk_uv6LyFU93no6mt1mdyzu+2DbgeASOxXac@mail.gmail.com>,
	<4d8d5850.0c87970a.7e04.1729@mx.google.com>,
	<549CFEF87AEDE841A38E9D15EAB4C04C58B2C495A8@cooper>,
	<4d8dc0f8.6219e70a.4bee.0faa@mx.google.com>,
	<8FCF27AAD14D468EB268E59A4CABC2EE@e1705>
Message-ID: <4d8e9dcb.c126e70a.6f33.7948@mx.google.com>

, hope it will be fixed soon...
using the latest version, same problem, i tried it just today...


2011-03-27 



Charles 



???? Madovsky 
????? 2011-03-26  23:01:13 
???? FreeSWITCH Users Help 
??? 
??? Re: [Freeswitch-users]sofia_reg_internal.db growbiggerandbiggeralong with time 
 
seems that even the text of your email becomes bigger and bigger :)

----- Original Message ----- 
From: Charles 
To: FreeSWITCH Users Help ; FreeSWITCH Users Help 
Sent: Saturday, March 26, 2011 6:33 AM
Subject: Re: [Freeswitch-users]sofia_reg_internal.db growbiggerandbiggeralong with time


Hi Peter,

Thanks for reminding, i've submit it to Jira.

FreeSWITCH 
FS-3190
Table 'sip_authentication' was not cleared after nonce expired -caused sofia_reg_internal.db grow bigger and bigger along with time


2011-03-26 



Charles 



???? Peter Olsson 
????? 2011-03-26  16:14:02 
???? FreeSWITCH Users Help 
??? 
??? Re: [Freeswitch-users]sofia_reg_internal.db growbiggerandbiggeralong with time 
If you've found a bug - please submit it to Jira. Also, make sure to attach your sip profile config to it.
/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Charles [fieldpeak at gmail.com]
Skickat: den 26 mars 2011 04:06
Till: FreeSWITCH Users Help; FreeSWITCH Users Help; anthony.minessale
?mne: Re: [Freeswitch-users] sofia_reg_internal.db      growbiggerandbiggeralong with time
Hi Anthony,
I tried to delete the db files sofia_reg_* db files, it is the same.
Actually i've located the bug where the source code, when INSERTing the new record to the table sip_authentication(line#801 of sofia_reg.c),
It compose a wrong SQL statement which caused insert the wrong values ('NULL' at 'profile' column, and 'internal' at hostname column) to columns, consequetnly, failed to delete when expires time arrive.
Please FOCUS here.
the mail below shows the details,
thanks!
sender? Charles
sent time? 2011-03-25 12:22:57
receiver? FreeSWITCH Users Help; msc; jeff
cc? jonas.gauffin
subject? Re: Re: [Freeswitch-users] sofia_reg_internal.db grow bigger and biggeralong with time
It looks that I found the root cause, at the line#801 of sofia_reg.c, when excute the SQL statement ('INSERT') to the table 'sip_authentication', it wrongly insert as screenshot below, profile_name should be 'internal' and hostname should be 'mypc' (name of my computer).
however, i did not find anything wrong regarding the code itselft, so failed to fix it...
could you please help focus here to fix it, thanks.
P.S. i manually insert the correct record into this talbe, after a while (tens of seconds), it automatically deteleted by FS.
btw, i found there is a bug report like this http://jira.freeswitch.org/browse/FS-498# and it is marked
Resolved:14/Sep/10 12:32 PM, however, my version is Nov, 30, 2010, there is still exsit.
[cid:__0 at Foxmail.net]
 line#801 of sofia_reg.c:
sql = switch_mprintf("insert into sip_authentication (nonce,expires,profile_name,hostname, last_nc) "
 "values('%q', %ld, '%q', '%q', 0)", uuid_str,
 switch_epoch_time_now(NULL) + (profile->nonce_ttl ? profile->nonce_ttl : DEFAULT_NONCE_TTL),
 profile->name, mod_sofia_globals.hostname);
switch_assert(sql != NULL);
sofia_glue_actually_execute_sql(profile, sql, profile->ireg_mutex);
2011-03-26
________________________________
Charles
________________________________
???? Anthony Minessale
????? 2011-03-26  00:12:12
???? FreeSWITCH Users Help
???
??? Re: [Freeswitch-users] sofia_reg_internal.db growbiggerandbiggeralong with time
can you try deleting the db files sofia_reg_* db files?  Maybe they
have become corrupt?
2011/3/25 fieldpeak <fieldpeak at gmail.com>:
> Hi Jonas,
> i also think so. can you please update was the bug fixed?
>
> ? 2011-3-25 ??5:14?"Jonas Gauffin" <jonas.gauffin at gmail.com>???
>
> might be the same bug as I reported here:
> http://jira.freeswitch.org/browse/FS-498
>
>
>
> On Fri, Mar 25, 2011 at 8:13 AM, Charles <fieldpeak at gmail.com> wrote:
>>
>> no, this is not the c...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
!DSPAM:4d8d70e832765221816297!
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From Nabble at slickdeals.endjunk.com  Sun Mar 27 07:19:16 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sat, 26 Mar 2011 20:19:16 -0700 (PDT)
Subject: [Freeswitch-users] [CRIT] switch_loadable_module.c:928 Error
	loading module ...
In-Reply-To: <AANLkTi=G4bVeP84Boz6gUTK6Up1Tc+ebn-YhJo8nY5Sz@mail.gmail.com>
References: <1301167396505-6211171.post@n2.nabble.com>
	<AANLkTi=G4bVeP84Boz6gUTK6Up1Tc+ebn-YhJo8nY5Sz@mail.gmail.com>
Message-ID: <1301195956866-6211719.post@n2.nabble.com>

For the 1st error, here is the leading messages before the error message:
2011-03-26 14:32:36.143317 [NOTICE] switch_loadable_module.c:212 Adding
Dialplan 'enum'
2011-03-26 14:32:36.143756 [NOTICE] switch_loadable_module.c:252 Adding
Application 'enum'
2011-03-26 14:32:36.144259 [NOTICE] switch_loadable_module.c:274 Adding API
Function 'enum'
2011-03-26 14:32:36.144820 [NOTICE] switch_loadable_module.c:274 Adding API
Function 'enum_auto'
2011-03-26 14:32:36.193681 [NOTICE] mod_xml_cdr.c:99 Rotating log file paths
2011-03-26 14:32:36.193799 [NOTICE] mod_xml_cdr.c:126 Setting log file path
to /var/log/freeswitch/xml_cdr
2011-03-26 14:32:36.193927 [NOTICE] mod_xml_cdr.c:164 Setting err log file
path to /var/log/freeswitch/xml_cdr
2011-03-26 14:32:36.205563 [DEBUG] mod_cdr_csv.c:321 Adding default
template.
2011-03-26 14:32:36.205748 [DEBUG] mod_cdr_csv.c:368 Adding template sql.
2011-03-26 14:32:36.205895 [DEBUG] mod_cdr_csv.c:368 Adding template
example.
2011-03-26 14:32:36.206038 [DEBUG] mod_cdr_csv.c:368 Adding template snom.
2011-03-26 14:32:36.206177 [DEBUG] mod_cdr_csv.c:368 Adding template
linksys.
2011-03-26 14:32:36.206316 [DEBUG] mod_cdr_csv.c:368 Adding template
asterisk.
2011-03-26 14:32:36.211749 [DEBUG] mod_cdr_sqlite.c:198 Adding default
template.
2011-03-26 14:32:36.211921 [DEBUG] mod_cdr_sqlite.c:237 Adding template
example.
2011-03-26 14:32:36.213596 [DEBUG] switch_core_sqldb.c:875 SQL ERR [no such
table: cdr]
[SELECT * FROM cdr LIMIT 1]
Auto Generating Table!
2011-03-26 14:32:36.223605 [NOTICE] switch_loadable_module.c:274 Adding API
Function 'multicast_peers'
2011-03-26 14:32:36.232992 [NOTICE] switch_loadable_module.c:252 Adding
Application 'socket'
2011-03-26 14:32:36.233480 [NOTICE] switch_loadable_module.c:274 Adding API
Function 'event_sink'
2011-03-26 14:32:36.251260 [CRIT] switch_loadable_module.c:928 Error Loading
module /usr/lib/freeswitch/mod_erlang_event.so
**Unknown error**
2011-03-26 14:32:36.279999 [NOTICE] switch_loadable_module.c:359 Adding
Directory interface 'ldap'

For the 2nd error message, here is the leading messages:
2011-03-26 14:32:43.288115 [ERR] mod_lcr.c:1906 You must have ODBC support
in FreeSWITCH to use this module
2011-03-26 14:32:43.288278 [ERR] mod_lcr.c:1907         ./configure
--enable-core-odbc-support        
2011-03-26 14:32:43.288400 [CRIT] switch_loadable_module.c:928 Error Loading
module /usr/lib/freeswitch/mod_lcr.so
**Module load routine returned an error**


-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/CRIT-switch-loadable-module-c-928-Error-loading-module-tp6211171p6211719.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From dunchan at freemail.hu  Sun Mar 27 19:12:13 2011
From: dunchan at freemail.hu (dunchan)
Date: Sun, 27 Mar 2011 17:12:13 +0200
Subject: [Freeswitch-users] Hangup problem, missing SIP BYE
Message-ID: <4D8F53CD.7000607@freemail.hu>

Hi!

I have a server machine with Freeswitch, and SIP gateway to make 
outgoung call. The gateway has only IP address authentication.
I have own created UA, and if i run it from the server and the called 
party hangs up, it don't get BYE message.

If my UA runs in different machine (other IP), it works well.

I have no idea about this. :(

I've checked the logs, and compared the two methods, and it seems there 
are no differnces.
I use the newest FS.

Any suggestions?

thanks,
Viktor


From infos at madovsky.org  Sun Mar 27 20:26:28 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 27 Mar 2011 12:26:28 -0400
Subject: [Freeswitch-users] api conference dial
Message-ID: <A59442461B06476FBC0F491AAE91071C@e1705>

I try to do this

conference testconf dial loopback/9999999999 at domain.ltd 5555555555 freeSwitch

to match default dialplan but the result is a loop.
the log shows that the destination number is 9999999999 at domain.ltd-b,
why a "-b" is added ? must I change my default dialplan to match it ?

Thanks
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From ejay.greeves at yahoo.com  Sun Mar 27 20:56:15 2011
From: ejay.greeves at yahoo.com (Ejay Greeves)
Date: Sun, 27 Mar 2011 17:56:15 +0100 (BST)
Subject: [Freeswitch-users] cleanup on hangup
Message-ID: <128796.69350.qm@web132303.mail.ird.yahoo.com>

Is it possible to hook onto events? using esl in ruby. I am getting <defunct> ruby processes. 

I think this is happening because when a call is hung up it causes my script to terminate there and then and the ruby child processes are not being closed/cleaned up properly. While this does not stop new incoming calls because new fork process is spwaned  I need a more gracefull handling when a call ends. I think I could do something if I can collect the hangup event.

Is? event hook/handler possible in esl?



      
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From rhuddleston at gmail.com  Sun Mar 27 21:06:37 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Sun, 27 Mar 2011 13:06:37 -0400
Subject: [Freeswitch-users] Voice Over / Talent / Prompts
Message-ID: <287d01cbeca1$56615420$0323fc60$@com>

Anyone have any good leads on Voice Over / Talents / Prompts?

 

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From infos at madovsky.org  Sun Mar 27 21:07:52 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 27 Mar 2011 13:07:52 -0400
Subject: [Freeswitch-users] api conference dial
Message-ID: <B6F001400A1343B18C27EFB19C84D7B7@e1705>

sorry forget my request.
I needed to set sip_to_uri variables to match my default dialplan correctly

Thanks
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, March 27, 2011 12:26 PM
  Subject: api conference dial


  I try to do this

  conference testconf dial loopback/9999999999 at domain.ltd 5555555555 freeSwitch

  to match default dialplan but the result is a loop.
  the log shows that the destination number is 9999999999 at domain.ltd-b,
  why a "-b" is added ? must I change my default dialplan to match it ?

  Thanks
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From william.suffill at gmail.com  Sun Mar 27 21:13:15 2011
From: william.suffill at gmail.com (William Suffill)
Date: Sun, 27 Mar 2011 13:13:15 -0400
Subject: [Freeswitch-users] Voice Over / Talent / Prompts
In-Reply-To: <287d01cbeca1$56615420$0323fc60$@com>
References: <287d01cbeca1$56615420$0323fc60$@com>
Message-ID: <AANLkTik+RFafO_4FLMF9bHTYocnb5HX8o4N=WBikhS_A@mail.gmail.com>

It would require a bit more info as to the specifics you are looking for to
give you proper suggestions. The Stock American prompts are by GM Voices.

-- W
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From rhuddleston at gmail.com  Sun Mar 27 21:18:31 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Sun, 27 Mar 2011 13:18:31 -0400
Subject: [Freeswitch-users] Voice Over / Talent / Prompts
In-Reply-To: <AANLkTik+RFafO_4FLMF9bHTYocnb5HX8o4N=WBikhS_A@mail.gmail.com>
References: <287d01cbeca1$56615420$0323fc60$@com>
	<AANLkTik+RFafO_4FLMF9bHTYocnb5HX8o4N=WBikhS_A@mail.gmail.com>
Message-ID: <289d01cbeca2$ffa98d30$fefca790$@com>

I need about 10 - 20 prompts for a calling card platform in both English and
Spanish. I have the prompts already laid out - just looking for an
affordable talent to record the prompts.

 

The two recommendations I've gotten were for GM Voices and VocesEnLaRed -
however the later requires fluency in Spanish to be able to navigate the
site - and I'm somewhat challenged in that area.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William
Suffill
Sent: Sunday, March 27, 2011 1:13 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Voice Over / Talent / Prompts

 

It would require a bit more info as to the specifics you are looking for to
give you proper suggestions. The Stock American prompts are by GM Voices.

 

-- W

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From infos at madovsky.org  Sun Mar 27 21:50:22 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 27 Mar 2011 13:50:22 -0400
Subject: [Freeswitch-users] api conference dial
Message-ID: <88AEE5D99EBB43F08CC0D5BFC0BEBCAF@e1705>

So after some invite tests I notced 6/8 seconds of audio
delay between the invited call and conference.
if the invited leg call himself the conference the latency is reasonable
I heard in previous threads that there was maybe a latency problem
if loopback is used in conference ?

  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, March 27, 2011 1:07 PM
  Subject: Re: api conference dial


  sorry forget my request.
  I needed to set sip_to_uri variables to match my default dialplan correctly

  Thanks
    ----- Original Message ----- 
    From: Madovsky 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Sunday, March 27, 2011 12:26 PM
    Subject: api conference dial


    I try to do this

    conference testconf dial loopback/9999999999 at domain.ltd 5555555555 freeSwitch

    to match default dialplan but the result is a loop.
    the log shows that the destination number is 9999999999 at domain.ltd-b,
    why a "-b" is added ? must I change my default dialplan to match it ?

    Thanks
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From avi at avimarcus.net  Sun Mar 27 21:55:53 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 27 Mar 2011 19:55:53 +0200
Subject: [Freeswitch-users] Voice Over / Talent / Prompts
In-Reply-To: <289d01cbeca2$ffa98d30$fefca790$@com>
References: <287d01cbeca1$56615420$0323fc60$@com>
	<AANLkTik+RFafO_4FLMF9bHTYocnb5HX8o4N=WBikhS_A@mail.gmail.com>
	<289d01cbeca2$ffa98d30$fefca790$@com>
Message-ID: <AANLkTinQFPQC1VooHtSJngCDh-F=AyMmrwkM=WoQ_caS@mail.gmail.com>

Just FYI, FreeSWITCH default English sounds already include quite a few that
would probably fit your needs...
-Avi

On Sun, Mar 27, 2011 at 7:18 PM, Robert Huddleston <rhuddleston at gmail.com>wrote:

> I need about 10 ? 20 prompts for a calling card platform in both English
> and Spanish. I have the prompts already laid out ? just looking for an
> affordable talent to record the prompts.
>
>
>
> The two recommendations I?ve gotten were for GM Voices and VocesEnLaRed ?
> however the later requires fluency in Spanish to be able to navigate the
> site ? and I?m somewhat challenged in that area.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *William
> Suffill
> *Sent:* Sunday, March 27, 2011 1:13 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Voice Over / Talent / Prompts
>
>
>
> It would require a bit more info as to the specifics you are looking for to
> give you proper suggestions. The Stock American prompts are by GM Voices.
>
>
>
> -- W
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jcasale at activenetwerx.com  Sun Mar 27 21:59:21 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sun, 27 Mar 2011 17:59:21 +0000
Subject: [Freeswitch-users] Phone opinions
In-Reply-To: <4D8E72AA.6030900@rosengart.de>
References: <92097A6A775D5147B1078E3F15430B922D7C58@prato.activenetwerx.local>
	<4D8E72AA.6030900@rosengart.de>
Message-ID: <assp.0067f73758.92097A6A775D5147B1078E3F15430B922DDA61@prato.activenetwerx.local>

>What's wrong with the audio of Snom phones? What codec are you using?

Well, more so the overall quality of Snom and the competency of their support.
I bought a few of the first M3's which where total sh!t. A quote from the support
guy about firmware upgrades was something like "well, if you fiddle with this
it might just work, if not try this, then that..." I was thinking you have got to be
kidding me?

Based on the really shoddy overall of that phone and their tech support while I
used those handsets, I won't buy their phones ever again.


From rhuddleston at gmail.com  Sun Mar 27 22:56:46 2011
From: rhuddleston at gmail.com (rhuddleston at gmail.com)
Date: Sun, 27 Mar 2011 14:56:46 -0400
Subject: [Freeswitch-users] Voice Over / Talent / Prompts
In-Reply-To: <AANLkTinQFPQC1VooHtSJngCDh-F=AyMmrwkM=WoQ_caS@mail.gmail.com>
References: <287d01cbeca1$56615420$0323fc60$@com>
	<AANLkTik+RFafO_4FLMF9bHTYocnb5HX8o4N=WBikhS_A@mail.gmail.com>
	<289d01cbeca2$ffa98d30$fefca790$@com>
	<AANLkTinQFPQC1VooHtSJngCDh-F=AyMmrwkM=WoQ_caS@mail.gmail.com>
Message-ID: <9F7FC3DA-A886-435A-B44C-5E730D749A6A@gmail.com>

I agree there are numerous - but to keep in same tongue/accent/vocals I need a custom set
Thanks

On Mar 27, 2011, at 1:55 PM, Avi Marcus <avi at avimarcus.net> wrote:

> Just FYI, FreeSWITCH default English sounds already include quite a few that would probably fit your needs...
> -Avi
> 
> On Sun, Mar 27, 2011 at 7:18 PM, Robert Huddleston <rhuddleston at gmail.com> wrote:
> I need about 10 ? 20 prompts for a calling card platform in both English and Spanish. I have the prompts already laid out ? just looking for an affordable talent to record the prompts.
> 
>  
> 
> The two recommendations I?ve gotten were for GM Voices and VocesEnLaRed ? however the later requires fluency in Spanish to be able to navigate the site ? and I?m somewhat challenged in that area.
> 
>  
> 
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William Suffill
> Sent: Sunday, March 27, 2011 1:13 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Voice Over / Talent / Prompts
> 
>  
> 
> It would require a bit more info as to the specifics you are looking for to give you proper suggestions. The Stock American prompts are by GM Voices.
> 
>  
> 
> -- W
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From infos at madovsky.org  Mon Mar 28 00:32:59 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 27 Mar 2011 16:32:59 -0400
Subject: [Freeswitch-users] api conference dial
Message-ID: <26C2B9BF7B2F48549C000624CF08A53E@e1705>

If I understand the conference concept it means that
the invited call is never really inside the conference loop ?
because I can't see any events of the invite from ESL

thanks
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, March 27, 2011 1:50 PM
  Subject: Re: api conference dial


  So after some invite tests I notced 6/8 seconds of audio
  delay between the invited call and conference.
  if the invited leg call himself the conference the latency is reasonable
  I heard in previous threads that there was maybe a latency problem
  if loopback is used in conference ?

    ----- Original Message ----- 
    From: Madovsky 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Sunday, March 27, 2011 1:07 PM
    Subject: Re: api conference dial


    sorry forget my request.
    I needed to set sip_to_uri variables to match my default dialplan correctly

    Thanks
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Sunday, March 27, 2011 12:26 PM
      Subject: api conference dial


      I try to do this

      conference testconf dial loopback/9999999999 at domain.ltd 5555555555 freeSwitch

      to match default dialplan but the result is a loop.
      the log shows that the destination number is 9999999999 at domain.ltd-b,
      why a "-b" is added ? must I change my default dialplan to match it ?

      Thanks
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From steveayre at gmail.com  Mon Mar 28 02:01:51 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 27 Mar 2011 23:01:51 +0100
Subject: [Freeswitch-users] Hangup problem, missing SIP BYE
In-Reply-To: <4D8F53CD.7000607@freemail.hu>
References: <4D8F53CD.7000607@freemail.hu>
Message-ID: <AANLkTinkKO97_466mRnN+AFPO+Ra7dagbasOLxQ04nNH@mail.gmail.com>

Do you have any NAT in place? What's the Contact header of the INVITE
message?

The BYE is a separate request to the INVITE, and so it is sent to the
address in the Contact header from the INVITE. If you're behind NAT then FS
might not be able to reach you at that address (e.g. if it's an internal
IP).

-Steve


On 27 March 2011 16:12, dunchan <dunchan at freemail.hu> wrote:

> Hi!
>
> I have a server machine with Freeswitch, and SIP gateway to make
> outgoung call. The gateway has only IP address authentication.
> I have own created UA, and if i run it from the server and the called
> party hangs up, it don't get BYE message.
>
> If my UA runs in different machine (other IP), it works well.
>
> I have no idea about this. :(
>
> I've checked the logs, and compared the two methods, and it seems there
> are no differnces.
> I use the newest FS.
>
> Any suggestions?
>
> thanks,
> Viktor
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Mon Mar 28 02:04:56 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 27 Mar 2011 23:04:56 +0100
Subject: [Freeswitch-users] [CRIT] switch_loadable_module.c:928 Error
 loading module ...
In-Reply-To: <1301195956866-6211719.post@n2.nabble.com>
References: <1301167396505-6211171.post@n2.nabble.com>
	<AANLkTi=G4bVeP84Boz6gUTK6Up1Tc+ebn-YhJo8nY5Sz@mail.gmail.com>
	<1301195956866-6211719.post@n2.nabble.com>
Message-ID: <AANLkTi=xj+oikTOw3UQ7NgJ2GxuF_w83f4gA50XPaxaF@mail.gmail.com>

I don't see any messages indicating the reason for mod_erlang_event, but
mod_lcr is easy:

2011-03-26 14:32:43.288115 [ERR] mod_lcr.c:1906 You must have ODBC support
in FreeSWITCH to use this module
2011-03-26 14:32:43.288278 [ERR] mod_lcr.c:1907         ./configure
--enable-core-odbc-support

You must compile in ODBC support to use that module, and you haven't
(--enable-core-odbc-support option to configure).

It's possible mod_erlang_event is failing for the same reason but not
logging the reason. Compile in ODBC support and then see if you're still
unable to load that module.

-Steve


On 27 March 2011 04:19, mazilo <Nabble at slickdeals.endjunk.com> wrote:

> For the 1st error, here is the leading messages before the error message:
> 2011-03-26 14:32:36.143317 [NOTICE] switch_loadable_module.c:212 Adding
> Dialplan 'enum'
> 2011-03-26 14:32:36.143756 [NOTICE] switch_loadable_module.c:252 Adding
> Application 'enum'
> 2011-03-26 14:32:36.144259 [NOTICE] switch_loadable_module.c:274 Adding API
> Function 'enum'
> 2011-03-26 14:32:36.144820 [NOTICE] switch_loadable_module.c:274 Adding API
> Function 'enum_auto'
> 2011-03-26 14:32:36.193681 [NOTICE] mod_xml_cdr.c:99 Rotating log file
> paths
> 2011-03-26 14:32:36.193799 [NOTICE] mod_xml_cdr.c:126 Setting log file path
> to /var/log/freeswitch/xml_cdr
> 2011-03-26 14:32:36.193927 [NOTICE] mod_xml_cdr.c:164 Setting err log file
> path to /var/log/freeswitch/xml_cdr
> 2011-03-26 14:32:36.205563 [DEBUG] mod_cdr_csv.c:321 Adding default
> template.
> 2011-03-26 14:32:36.205748 [DEBUG] mod_cdr_csv.c:368 Adding template sql.
> 2011-03-26 14:32:36.205895 [DEBUG] mod_cdr_csv.c:368 Adding template
> example.
> 2011-03-26 14:32:36.206038 [DEBUG] mod_cdr_csv.c:368 Adding template snom.
> 2011-03-26 14:32:36.206177 [DEBUG] mod_cdr_csv.c:368 Adding template
> linksys.
> 2011-03-26 14:32:36.206316 [DEBUG] mod_cdr_csv.c:368 Adding template
> asterisk.
> 2011-03-26 14:32:36.211749 [DEBUG] mod_cdr_sqlite.c:198 Adding default
> template.
> 2011-03-26 14:32:36.211921 [DEBUG] mod_cdr_sqlite.c:237 Adding template
> example.
> 2011-03-26 14:32:36.213596 [DEBUG] switch_core_sqldb.c:875 SQL ERR [no such
> table: cdr]
> [SELECT * FROM cdr LIMIT 1]
> Auto Generating Table!
> 2011-03-26 14:32:36.223605 [NOTICE] switch_loadable_module.c:274 Adding API
> Function 'multicast_peers'
> 2011-03-26 14:32:36.232992 [NOTICE] switch_loadable_module.c:252 Adding
> Application 'socket'
> 2011-03-26 14:32:36.233480 [NOTICE] switch_loadable_module.c:274 Adding API
> Function 'event_sink'
> 2011-03-26 14:32:36.251260 [CRIT] switch_loadable_module.c:928 Error
> Loading
> module /usr/lib/freeswitch/mod_erlang_event.so
> **Unknown error**
> 2011-03-26 14:32:36.279999 [NOTICE] switch_loadable_module.c:359 Adding
> Directory interface 'ldap'
>
> For the 2nd error message, here is the leading messages:
> 2011-03-26 14:32:43.288115 [ERR] mod_lcr.c:1906 You must have ODBC support
> in FreeSWITCH to use this module
> 2011-03-26 14:32:43.288278 [ERR] mod_lcr.c:1907         ./configure
> --enable-core-odbc-support
> 2011-03-26 14:32:43.288400 [CRIT] switch_loadable_module.c:928 Error
> Loading
> module /usr/lib/freeswitch/mod_lcr.so
> **Module load routine returned an error**
>
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/CRIT-switch-loadable-module-c-928-Error-loading-module-tp6211171p6211719.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at microcomaustralia.com.au  Mon Mar 28 07:45:03 2011
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 28 Mar 2011 14:45:03 +1100
Subject: [Freeswitch-users] answering calls weirdness
In-Reply-To: <AANLkTikBtvfpR4ryNXCcUhMdEqubVjw7fZ1SRR_WqHHX@mail.gmail.com>
References: <AANLkTin5mP7QzDe+CAvaPjsbxhq934P-49LARGQJwR8Y@mail.gmail.com>
	<AANLkTin8nv3N=XsPa=jepebenxRJ4EvtNSMJ3D37KZKu@mail.gmail.com>
	<AANLkTi=GC1u5oZeACdpUeF=ggT3c5AAbDPH5KAgYzTKW@mail.gmail.com>
	<AANLkTinaEPAJ7Gc3bYb0nnguWfXvnC=XL=L+EJMgG180@mail.gmail.com>
	<AANLkTikBtvfpR4ryNXCcUhMdEqubVjw7fZ1SRR_WqHHX@mail.gmail.com>
Message-ID: <AANLkTi=qKxEz3HqY4KNW0OXVuTjqq4D3VfnZ+CM6bU81@mail.gmail.com>

On 26 March 2011 18:50, Brian May <brian at microcomaustralia.com.au> wrote:
> On 26 March 2011 18:27, Brian May <brian at microcomaustralia.com.au> wrote:
>> I upgraded to the latest git version of freeswitch and reproduced the
>> same issues.
>>
>> Latest log file now:
>>
>> http://pastebin.freeswitch.org/15860
>>
>> Where does it get 6201 from? Surely freeswitch doesn't normally pull
>> phone numbers out of thin air???
>
> Also worth noting that this only occurs when dialing group numbers,
> calls to single destinations work fine.

Any ideas on how to debug this problem? It is kind of making
freeswitch useless :-(
-- 
Brian May <brian at microcomaustralia.com.au>


From jeff at jefflenk.com  Mon Mar 28 08:07:31 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sun, 27 Mar 2011 21:07:31 -0700 (PDT)
Subject: [Freeswitch-users] answering calls weirdness
In-Reply-To: <AANLkTi=qKxEz3HqY4KNW0OXVuTjqq4D3VfnZ+CM6bU81@mail.gmail.com>
References: <AANLkTin5mP7QzDe+CAvaPjsbxhq934P-49LARGQJwR8Y@mail.gmail.com>
	<AANLkTin8nv3N=XsPa=jepebenxRJ4EvtNSMJ3D37KZKu@mail.gmail.com>
	<AANLkTi=GC1u5oZeACdpUeF=ggT3c5AAbDPH5KAgYzTKW@mail.gmail.com>
	<AANLkTinaEPAJ7Gc3bYb0nnguWfXvnC=XL=L+EJMgG180@mail.gmail.com>
	<AANLkTikBtvfpR4ryNXCcUhMdEqubVjw7fZ1SRR_WqHHX@mail.gmail.com>
	<AANLkTi=qKxEz3HqY4KNW0OXVuTjqq4D3VfnZ+CM6bU81@mail.gmail.com>
Message-ID: <1301285251126-6213679.post@n2.nabble.com>

get a siptrace to go with the debug log you have - sofia global siptrace on


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/answering-calls-weirdness-tp6202405p6213679.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brian at microcomaustralia.com.au  Mon Mar 28 08:28:12 2011
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 28 Mar 2011 15:28:12 +1100
Subject: [Freeswitch-users] answering calls weirdness
In-Reply-To: <1301285251126-6213679.post@n2.nabble.com>
References: <AANLkTin5mP7QzDe+CAvaPjsbxhq934P-49LARGQJwR8Y@mail.gmail.com>
	<AANLkTin8nv3N=XsPa=jepebenxRJ4EvtNSMJ3D37KZKu@mail.gmail.com>
	<AANLkTi=GC1u5oZeACdpUeF=ggT3c5AAbDPH5KAgYzTKW@mail.gmail.com>
	<AANLkTinaEPAJ7Gc3bYb0nnguWfXvnC=XL=L+EJMgG180@mail.gmail.com>
	<AANLkTikBtvfpR4ryNXCcUhMdEqubVjw7fZ1SRR_WqHHX@mail.gmail.com>
	<AANLkTi=qKxEz3HqY4KNW0OXVuTjqq4D3VfnZ+CM6bU81@mail.gmail.com>
	<1301285251126-6213679.post@n2.nabble.com>
Message-ID: <AANLkTi=rVM_1Fkqw=LxA=squVS1qSwTG+60Ty_Qm=YfN@mail.gmail.com>

On 28 March 2011 15:07, Jeff Lenk <jeff at jefflenk.com> wrote:
> get a siptrace to go with the debug log you have - sofia global siptrace on

Never mind, think I have resolved it. At the risk of looking very very
silly, here goes:

First I noticed the problem only occured when I included a certain
phone (ATA box) in the group.

I noticed that ATA box has call forwarding on busy setup, to phone
number ... you guessed it ... 6201.

So my guess is everything normally works fine, but just recently
somebody left that phone off hook (will check when I get home), so all
phone calls were getting redirected to ... 6201.

In other words freeswitch was doing what is was suppose to do.

Now just to fix callerid on the freetdm port since upgrading to the
latest freeswitch.
-- 
Brian May <brian at microcomaustralia.com.au>


From infos at madovsky.org  Mon Mar 28 09:24:32 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 28 Mar 2011 01:24:32 -0400
Subject: [Freeswitch-users] inline syntax in ESL
Message-ID: <353BF1FF43AB4531B925CBC58AFEA2B9@e1705>

I'm looking for to set an inline variable in ESL message format.
any example ?

Thanks
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From msc at freeswitch.org  Mon Mar 28 09:46:27 2011
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 27 Mar 2011 22:46:27 -0700
Subject: [Freeswitch-users] inline syntax in ESL
In-Reply-To: <353BF1FF43AB4531B925CBC58AFEA2B9@e1705>
References: <353BF1FF43AB4531B925CBC58AFEA2B9@e1705>
Message-ID: <AANLkTinM9DOyjytYQGvZZXpEZMENU=A5TuCuUNPSGik4@mail.gmail.com>

Can you expand upon this question a bit? What is the scenario?
-MC

On Sun, Mar 27, 2011 at 10:24 PM, Madovsky <infos at madovsky.org> wrote:
> I'm looking for to set an inline variable in ESL message format.
> any example ?
>
> Thanks
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From infos at madovsky.org  Mon Mar 28 09:57:17 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 28 Mar 2011 01:57:17 -0400
Subject: [Freeswitch-users] inline syntax in ESL
References: <353BF1FF43AB4531B925CBC58AFEA2B9@e1705>
	<AANLkTinM9DOyjytYQGvZZXpEZMENU=A5TuCuUNPSGik4@mail.gmail.com>
Message-ID: <4D41F3843F824974B0598897624E2AC2@e1705>

I send msg from my esl script as the wiki says
SendMsg <uuid>
call-command: execute
execute-app-name: set
execute-app-arg: myvar=yeswhere I put "inline=true" ?----- Original Message ----- 
From: "Michael Collins" <msc at freeswitch.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, March 28, 2011 1:46 AM
Subject: Re: [Freeswitch-users] inline syntax in ESL


> Can you expand upon this question a bit? What is the scenario?
> -MC
> 
> On Sun, Mar 27, 2011 at 10:24 PM, Madovsky <infos at madovsky.org> wrote:
>> I'm looking for to set an inline variable in ESL message format.
>> any example ?
>>
>> Thanks
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From msc at freeswitch.org  Mon Mar 28 10:03:50 2011
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 27 Mar 2011 23:03:50 -0700
Subject: [Freeswitch-users] inline syntax in ESL
In-Reply-To: <4D41F3843F824974B0598897624E2AC2@e1705>
References: <353BF1FF43AB4531B925CBC58AFEA2B9@e1705>
	<AANLkTinM9DOyjytYQGvZZXpEZMENU=A5TuCuUNPSGik4@mail.gmail.com>
	<4D41F3843F824974B0598897624E2AC2@e1705>
Message-ID: <AANLkTinW-TXnF2rKdMjnq--Bu8nDnv__R9b6A3BMDQ6n@mail.gmail.com>

I'm pretty sure that "inline" does not apply here as that is a
dialplan convention. IIRC, as soon as you execute the app then the
channel variable is set. Are you not seeing that happen?

-MC

On Sun, Mar 27, 2011 at 10:57 PM, Madovsky <infos at madovsky.org> wrote:
> I send msg from my esl script as the wiki says
>
> SendMsg <uuid>
> call-command: execute
> execute-app-name: set
> execute-app-arg: myvar=yes
>
> where I put "inline=true" ?
>
> ----- Original Message -----
> From: "Michael Collins" <msc at freeswitch.org>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, March 28, 2011 1:46 AM
> Subject: Re: [Freeswitch-users] inline syntax in ESL
>> Can you expand upon this question a bit? What is the scenario?
>> -MC
>>
>> On Sun, Mar 27, 2011 at 10:24 PM, Madovsky <infos at madovsky.org> wrote:
>>> I'm looking for to set an inline variable in ESL message format.
>>> any example ?
>>>
>>> Thanks
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From infos at madovsky.org  Mon Mar 28 10:10:49 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 28 Mar 2011 02:10:49 -0400
Subject: [Freeswitch-users] inline syntax in ESL
References: <353BF1FF43AB4531B925CBC58AFEA2B9@e1705><AANLkTinM9DOyjytYQGvZZXpEZMENU=A5TuCuUNPSGik4@mail.gmail.com><4D41F3843F824974B0598897624E2AC2@e1705>
	<AANLkTinW-TXnF2rKdMjnq--Bu8nDnv__R9b6A3BMDQ6n@mail.gmail.com>
Message-ID: <F1D28339F14B45C896196F3A2C9557B2@e1705>

ah ok,
> as soon as you execute the app then the
> channel variable is set. Are you not seeing that happen?
no the var is empty, nevermind as you said I can set it in the dialplan,
but the logical will be less accurate.

Thanks

----- Original Message ----- 
From: "Michael Collins" <msc at freeswitch.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, March 28, 2011 2:03 AM
Subject: Re: [Freeswitch-users] inline syntax in ESL


> I'm pretty sure that "inline" does not apply here as that is a
> dialplan convention. IIRC, as soon as you execute the app then the
> channel variable is set. Are you not seeing that happen?
>
> -MC
>
> On Sun, Mar 27, 2011 at 10:57 PM, Madovsky <infos at madovsky.org> wrote:
>> I send msg from my esl script as the wiki says
>>
>> SendMsg <uuid>
>> call-command: execute
>> execute-app-name: set
>> execute-app-arg: myvar=yes
>>
>> where I put "inline=true" ?
>>
>> ----- Original Message -----
>> From: "Michael Collins" <msc at freeswitch.org>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, March 28, 2011 1:46 AM
>> Subject: Re: [Freeswitch-users] inline syntax in ESL
>>> Can you expand upon this question a bit? What is the scenario?
>>> -MC
>>>
>>> On Sun, Mar 27, 2011 at 10:24 PM, Madovsky <infos at madovsky.org> wrote:
>>>> I'm looking for to set an inline variable in ESL message format.
>>>> any example ?
>>>>
>>>> Thanks
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From peter.olsson at visionutveckling.se  Mon Mar 28 10:22:53 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 28 Mar 2011 08:22:53 +0200
Subject: [Freeswitch-users] inline syntax in ESL
Message-ID: <C2E843A6-748C-46F1-98EF-B5CF88B65DAD@visionutveckling.se>

execute/set will queue to the channel, api setvar will set immediately.

/Peter

----- Reply message -----
Fr?n: "Madovsky" <infos at madovsky.org>
Datum: m?n, mar 28, 2011 08:18
Rubrik: [Freeswitch-users] inline syntax in ESL
Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

ah ok,
> as soon as you execute the app then the
> channel variable is set. Are you not seeing that happen?
no the var is empty, nevermind as you said I can set it in the dialplan,
but the logical will be less accurate.

Thanks

----- Original Message -----
From: "Michael Collins" <msc at freeswitch.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, March 28, 2011 2:03 AM
Subject: Re: [Freeswitch-users] inline syntax in ESL


> I'm pretty sure that "inline" does not apply here as that is a
> dialplan convention. IIRC, as soon as you execute the app then the
> channel variable is set. Are you not seeing that happen?
>
> -MC
>
> On Sun, Mar 27, 2011 at 10:57 PM, Madovsky <infos at madovsky.org> wrote:
>> I send msg from my esl script as the wiki says
>>
>> SendMsg <uuid>
>> call-command: execute
>> execute-app-name: set
>> execute-app-arg: myvar=yes
>>
>> where I put "inline=true" ?
>>
>> ----- Original Message -----
>> From: "Michael Collins" <msc at freeswitch.org>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, March 28, 2011 1:46 AM
>> Subject: Re: [Freeswitch-users] inline syntax in ESL
>>> Can you expand upon this question a bit? What is the scenario?
>>> -MC
>>>
>>> On Sun, Mar 27, 2011 at 10:24 PM, Madovsky <infos at madovsky.org> wrote:
>>>> I'm looking for to set an inline variable in ESL message format.
>>>> any example ?
>>>>
>>>> Thanks
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _______________________________________________
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>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From mitch.capper at gmail.com  Mon Mar 28 10:32:21 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Sun, 27 Mar 2011 23:32:21 -0700
Subject: [Freeswitch-users] Voice Over / Talent / Prompts
In-Reply-To: <9F7FC3DA-A886-435A-B44C-5E730D749A6A@gmail.com>
References: <287d01cbeca1$56615420$0323fc60$@com>
	<AANLkTik+RFafO_4FLMF9bHTYocnb5HX8o4N=WBikhS_A@mail.gmail.com>
	<289d01cbeca2$ffa98d30$fefca790$@com>
	<AANLkTinQFPQC1VooHtSJngCDh-F=AyMmrwkM=WoQ_caS@mail.gmail.com>
	<9F7FC3DA-A886-435A-B44C-5E730D749A6A@gmail.com>
Message-ID: <AANLkTikORjFLUGQa5oB+Ckbu=oeR+Ek-Tuxe0AMTzxub@mail.gmail.com>

Flowroute has some affordable options: http://flowroute.com/services/voice/

~Mitch

On Sun, Mar 27, 2011 at 11:56 AM,  <rhuddleston at gmail.com> wrote:
> I agree there are numerous - but to keep in same tongue/accent/vocals I need
> a custom set
> Thanks
>
> On Mar 27, 2011, at 1:55 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
> Just FYI, FreeSWITCH default?English?sounds already include quite a few that
> would probably fit your needs...
> -Avi
>
> On Sun, Mar 27, 2011 at 7:18 PM, Robert Huddleston <rhuddleston at gmail.com>
> wrote:
>>
>> I need about 10 ? 20 prompts for a calling card platform in both English
>> and Spanish. I have the prompts already laid out ? just looking for an
>> affordable talent to record the prompts.
>>
>>
>>
>> The two recommendations I?ve gotten were for GM Voices and VocesEnLaRed ?
>> however the later requires fluency in Spanish to be able to navigate the
>> site ? and I?m somewhat challenged in that area.
>>
>>
>>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William
>> Suffill
>> Sent: Sunday, March 27, 2011 1:13 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Voice Over / Talent / Prompts
>>
>>
>>
>> It would require a bit more info as to the specifics you are looking for
>> to give you proper suggestions. The Stock American prompts are by GM Voices.
>>
>>
>>
>> -- W
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From infos at madovsky.org  Mon Mar 28 10:41:49 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 28 Mar 2011 02:41:49 -0400
Subject: [Freeswitch-users] inline syntax in ESL
References: <C2E843A6-748C-46F1-98EF-B5CF88B65DAD@visionutveckling.se>
Message-ID: <296C54D239C24D2DBFF6F9DC88F9FDAE@e1705>

good to know, thanks Peter.
you mean uuid_setvar for ex ?


----- Original Message ----- 
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Monday, March 28, 2011 2:22 AM
Subject: Re: [Freeswitch-users] inline syntax in ESL


execute/set will queue to the channel, api setvar will set immediately.

/Peter

----- Reply message -----
Fr?n: "Madovsky" <infos at madovsky.org>
Datum: m?n, mar 28, 2011 08:18
Rubrik: [Freeswitch-users] inline syntax in ESL
Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

ah ok,
> as soon as you execute the app then the
> channel variable is set. Are you not seeing that happen?
no the var is empty, nevermind as you said I can set it in the dialplan,
but the logical will be less accurate.

Thanks

----- Original Message -----
From: "Michael Collins" <msc at freeswitch.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, March 28, 2011 2:03 AM
Subject: Re: [Freeswitch-users] inline syntax in ESL


> I'm pretty sure that "inline" does not apply here as that is a
> dialplan convention. IIRC, as soon as you execute the app then the
> channel variable is set. Are you not seeing that happen?
>
> -MC
>
> On Sun, Mar 27, 2011 at 10:57 PM, Madovsky <infos at madovsky.org> wrote:
>> I send msg from my esl script as the wiki says
>>
>> SendMsg <uuid>
>> call-command: execute
>> execute-app-name: set
>> execute-app-arg: myvar=yes
>>
>> where I put "inline=true" ?
>>
>> ----- Original Message -----
>> From: "Michael Collins" <msc at freeswitch.org>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, March 28, 2011 1:46 AM
>> Subject: Re: [Freeswitch-users] inline syntax in ESL
>>> Can you expand upon this question a bit? What is the scenario?
>>> -MC
>>>
>>> On Sun, Mar 27, 2011 at 10:24 PM, Madovsky <infos at madovsky.org> wrote:
>>>> I'm looking for to set an inline variable in ESL message format.
>>>> any example ?
>>>>
>>>> Thanks
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _______________________________________________
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>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From peter.olsson at visionutveckling.se  Mon Mar 28 11:05:21 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 28 Mar 2011 09:05:21 +0200
Subject: [Freeswitch-users] inline syntax in ESL
In-Reply-To: <296C54D239C24D2DBFF6F9DC88F9FDAE@e1705>
References: <C2E843A6-748C-46F1-98EF-B5CF88B65DAD@visionutveckling.se>
	<296C54D239C24D2DBFF6F9DC88F9FDAE@e1705>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B3178F6F@cooper>

Yes, exactly. uuid_setvar will set the variable before it returns an answer back to ESL client. Execute/set will return immediately and queue the application to the channel, so it will be executed next time it checks the queue in the loop (so, usually 20ms delay).

/Peter

-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Madovsky
Skickat: den 28 mars 2011 08:42
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] inline syntax in ESL

good to know, thanks Peter.
you mean uuid_setvar for ex ?


----- Original Message ----- 
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Monday, March 28, 2011 2:22 AM
Subject: Re: [Freeswitch-users] inline syntax in ESL


execute/set will queue to the channel, api setvar will set immediately.

/Peter

----- Reply message -----
Fr?n: "Madovsky" <infos at madovsky.org>
Datum: m?n, mar 28, 2011 08:18
Rubrik: [Freeswitch-users] inline syntax in ESL
Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

ah ok,
> as soon as you execute the app then the
> channel variable is set. Are you not seeing that happen?
no the var is empty, nevermind as you said I can set it in the dialplan,
but the logical will be less accurate.

Thanks

----- Original Message -----
From: "Michael Collins" <msc at freeswitch.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, March 28, 2011 2:03 AM
Subject: Re: [Freeswitch-users] inline syntax in ESL


> I'm pretty sure that "inline" does not apply here as that is a
> dialplan convention. IIRC, as soon as you execute the app then the
> channel variable is set. Are you not seeing that happen?
>
> -MC
>
> On Sun, Mar 27, 2011 at 10:57 PM, Madovsky <infos at madovsky.org> wrote:
>> I send msg from my esl script as the wiki says
>>
>> SendMsg <uuid>
>> call-command: execute
>> execute-app-name: set
>> execute-app-arg: myvar=yes
>>
>> where I put "inline=true" ?
>>
>> ----- Original Message -----
>> From: "Michael Collins" <msc at freeswitch.org>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, March 28, 2011 1:46 AM
>> Subject: Re: [Freeswitch-users] inline syntax in ESL
>>> Can you expand upon this question a bit? What is the scenario?
>>> -MC
>>>
>>> On Sun, Mar 27, 2011 at 10:24 PM, Madovsky <infos at madovsky.org> wrote:
>>>> I'm looking for to set an inline variable in ESL message format.
>>>> any example ?
>>>>
>>>> Thanks
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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From infos at madovsky.org  Mon Mar 28 11:07:57 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 28 Mar 2011 03:07:57 -0400
Subject: [Freeswitch-users] inline syntax in ESL
References: <C2E843A6-748C-46F1-98EF-B5CF88B65DAD@visionutveckling.se>
Message-ID: <FFA093E2172A4FC5930744401614D527@e1705>

solved. thanks Mike and Peter


----- Original Message ----- 
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Monday, March 28, 2011 2:22 AM
Subject: Re: [Freeswitch-users] inline syntax in ESL


execute/set will queue to the channel, api setvar will set immediately.

/Peter

----- Reply message -----
Fr?n: "Madovsky" <infos at madovsky.org>
Datum: m?n, mar 28, 2011 08:18
Rubrik: [Freeswitch-users] inline syntax in ESL
Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

ah ok,
> as soon as you execute the app then the
> channel variable is set. Are you not seeing that happen?
no the var is empty, nevermind as you said I can set it in the dialplan,
but the logical will be less accurate.

Thanks

----- Original Message -----
From: "Michael Collins" <msc at freeswitch.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, March 28, 2011 2:03 AM
Subject: Re: [Freeswitch-users] inline syntax in ESL


> I'm pretty sure that "inline" does not apply here as that is a
> dialplan convention. IIRC, as soon as you execute the app then the
> channel variable is set. Are you not seeing that happen?
>
> -MC
>
> On Sun, Mar 27, 2011 at 10:57 PM, Madovsky <infos at madovsky.org> wrote:
>> I send msg from my esl script as the wiki says
>>
>> SendMsg <uuid>
>> call-command: execute
>> execute-app-name: set
>> execute-app-arg: myvar=yes
>>
>> where I put "inline=true" ?
>>
>> ----- Original Message -----
>> From: "Michael Collins" <msc at freeswitch.org>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, March 28, 2011 1:46 AM
>> Subject: Re: [Freeswitch-users] inline syntax in ESL
>>> Can you expand upon this question a bit? What is the scenario?
>>> -MC
>>>
>>> On Sun, Mar 27, 2011 at 10:24 PM, Madovsky <infos at madovsky.org> wrote:
>>>> I'm looking for to set an inline variable in ESL message format.
>>>> any example ?
>>>>
>>>> Thanks
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From brian at microcomaustralia.com.au  Mon Mar 28 11:25:21 2011
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 28 Mar 2011 18:25:21 +1100
Subject: [Freeswitch-users] answering calls weirdness
In-Reply-To: <AANLkTi=rVM_1Fkqw=LxA=squVS1qSwTG+60Ty_Qm=YfN@mail.gmail.com>
References: <AANLkTin5mP7QzDe+CAvaPjsbxhq934P-49LARGQJwR8Y@mail.gmail.com>
	<AANLkTin8nv3N=XsPa=jepebenxRJ4EvtNSMJ3D37KZKu@mail.gmail.com>
	<AANLkTi=GC1u5oZeACdpUeF=ggT3c5AAbDPH5KAgYzTKW@mail.gmail.com>
	<AANLkTinaEPAJ7Gc3bYb0nnguWfXvnC=XL=L+EJMgG180@mail.gmail.com>
	<AANLkTikBtvfpR4ryNXCcUhMdEqubVjw7fZ1SRR_WqHHX@mail.gmail.com>
	<AANLkTi=qKxEz3HqY4KNW0OXVuTjqq4D3VfnZ+CM6bU81@mail.gmail.com>
	<1301285251126-6213679.post@n2.nabble.com>
	<AANLkTi=rVM_1Fkqw=LxA=squVS1qSwTG+60Ty_Qm=YfN@mail.gmail.com>
Message-ID: <AANLkTinmBxPdskziZp5D1fgXxg1K2TDaHtd0i9b8t8Fj@mail.gmail.com>

On 28 March 2011 15:28, Brian May <brian at microcomaustralia.com.au> wrote:
> So my guess is everything normally works fine, but just recently
> somebody left that phone off hook (will check when I get home), so all
> phone calls were getting redirected to ... 6201.
>
> In other words freeswitch was doing what is was suppose to do.

Then again, after thinking about it, I don't consider this expected
behaviour. If making a bridge to a group, and one of the phones in the
group forwards the call to somewhere else, I would expect this should
only affect the call to that phone, the other phones in the group
should continue ringing.
-- 
Brian May <brian at microcomaustralia.com.au>


From frankie.k.yiu at gmail.com  Mon Mar 28 11:50:01 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Mon, 28 Mar 2011 00:50:01 -0700
Subject: [Freeswitch-users] Question on creating my own event or message
Message-ID: <AANLkTik2fVzON=j7epRvnziL3yCO1zFj0=0xZb6XffSW@mail.gmail.com>

Hi there,

If I want to send my own event or message to notify my other part of
the system when a certain condition is met, how do I implement that in C++
code?
For example, I might want to do data analysis of the RTP packets in C++,
and when I detected the right timing where I want to play an audio, I can
send an event to my other part of the system to play an audio file.

Is there any hint or sample code that I can take a look?

Thanks,

Frankie
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From yivzhenko at mksat.net  Mon Mar 28 11:54:52 2011
From: yivzhenko at mksat.net (Yuriy Ivzhenko)
Date: Mon, 28 Mar 2011 09:54:52 +0200
Subject: [Freeswitch-users] mod_fifo & CID
In-Reply-To: <AANLkTim=ER2syNeDVWM6C6C84pqvshg7R3bNR4M2xEnf@mail.gmail.com>
References: <201103241644.27845.yivzhenko@mksat.net>
	<AANLkTim=ER2syNeDVWM6C6C84pqvshg7R3bNR4M2xEnf@mail.gmail.com>
Message-ID: <201103281054.52190.yivzhenko@mksat.net>

This change only caller_id_name from "Outbound Call" to 'agent name' and only 
when call Answered.

On Friday 25 March 2011 21:41:23 Anthony Minessale wrote:
> when you define the agents in the config add this:
> 
> {origination_callee_id_name='agent
> name',origination_callee_id_number='agents number'}
> 
> Its display update when the call changes direction from outbound to bridge.
> 
> On Thu, Mar 24, 2011 at 9:44 AM, Yuriy Ivzhenko <yivzhenko at mksat.net> wrote:
> > Hi All,
> >
> > After upgrading to last Git (git-73ca862) i see changes in CID on
> > calls to agents, generated by mod_fifo.
> >
> > If agent answer
> > caller_profile.caller_id_name = "Outbound Call"
> > caller_profile.caller_id_number = <agent number>
> >
> > If agent not answer
> > caller_profile.caller_id_name = caller caller_id_name
> > caller_profile.caller_id_number = caller caller_id_number (or
> > origination_caller_id_name/number of set in agents dialstring)
> > In old version, so it was always.
> >
> > Is this correct?
> > It seems to me it should not depend on, answered the call or not.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 


From brian at microcomaustralia.com.au  Mon Mar 28 12:02:44 2011
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 28 Mar 2011 19:02:44 +1100
Subject: [Freeswitch-users] outgoing callerid freetdm TDM400
In-Reply-To: <AANLkTinPq9TwEaL5fzFkgL0g5G3k21nVy61PNBLcr94Z@mail.gmail.com>
References: <e4aa86dec5cce313b5ca697bcab0e620@thom.fr.eu.org>
	<AANLkTinrExBqouNizfj6G_jo5z9956RTwW_Ba1v1c4Y=@mail.gmail.com>
	<d08269f78135c08016658f7faace0b65@thom.fr.eu.org>
	<AANLkTimxnj-2JvKwA8P938yg9xqdvqJgGLM9vYHw57GB@mail.gmail.com>
	<31329b0ef5bc12ecc522e18a9a83c764@thom.fr.eu.org>
	<AANLkTinyC4GfKaVvNCciBAbo=VStrG1suiLO5PUO2jYH@mail.gmail.com>
	<f181ca10a3c711fbe1c3e71767cb58c6@thom.fr.eu.org>
	<AANLkTikP-wPt_FeiRjotb+ouirfuSPPYoVERZe_-qBiu@mail.gmail.com>
	<729917f2a65c3b6c32a40c25bd81a89a@thom.fr.eu.org>
	<AANLkTikci7V+WELZV5yS7K6ac5MBpSi4YUqXrDAxunY6@mail.gmail.com>
	<693eb681021b33ccdf84a48451678505@thom.fr.eu.org>
	<AANLkTinPq9TwEaL5fzFkgL0g5G3k21nVy61PNBLcr94Z@mail.gmail.com>
Message-ID: <AANLkTimvfmz=Vfk39KY84KZP5DbOMy47-vMs8iHH=hGA@mail.gmail.com>

On 9 September 2010 21:03, Brian May <brian at microcomaustralia.com.au> wrote:
> On 9 September 2010 18:54, Fran?ois Legal <devel at thom.fr.eu.org> wrote:
>> I have updated the CID patch against today's git. The patch now contains
>> only the CID part. I created a separate MWI patch that I attached to
>> OPENZAP-101.
>
> Wow! It seems to work. I get caller id again.

Is there any progress in getting these changes incorporated into the
official git repository?

Looks like when I updated to the latest git version I broken callerid :-(.

Unfortunately the patch in OPENZAP-100 no longer applies cleanly either.

http://jira.freeswitch.org/browse/OPENZAP-100
-- 
Brian May <brian at microcomaustralia.com.au>


From richocet2 at hotmail.com  Mon Mar 28 10:04:29 2011
From: richocet2 at hotmail.com (Dave Bracken)
Date: Mon, 28 Mar 2011 06:04:29 +0000
Subject: [Freeswitch-users] freeswitch-users@lists.freeswitch.org
Message-ID: <BLU0-SMTP56CDA3B84F8900C0993AFFF0BA0@phx.gbl>

freeswitch-users at lists.freeswitch.org I always borrowed money knowing I couldnt pay it back I had reached the end of the line this was such a relief http://bit.ly/fJNMDE now im blowing money fast without consequences you have potential for greatness


From all.eforums at gmail.com  Mon Mar 28 14:42:18 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Mon, 28 Mar 2011 06:42:18 -0400
Subject: [Freeswitch-users] Embarrassing Question: Local extension
 registering in the Public context on default configuration?
Message-ID: <AANLkTinDZ0xBMxFssO9NrP8y6opAXHUrMOPoNzhS1U2M@mail.gmail.com>

Hi All,

I've used FS in the past to varying degrees but never as a PBX or a
REGISTRAR and just to test basic config, I built and installed it and didn't
change a damn thing. Had a softphone on the same internal network registered
to it using the default password, but it seems to register itself in the
public context when I'd have thought it'd register in default context so I
can test all those pre-built extensions for an echo test, dumping channel
variables etc. But they don't work as the call originated to say 5000 from
extension 1000 goes into the public context which doesn't have 5000. What is
going on? It's so embarrassing to ask such a simple question :(

freeswitch at internal> sofia status profile internal 1000
=================================================================================================
Name                    internal
Domain Name             N/A
Auto-NAT                false
DBName                  sofia_reg_internal
Pres Hosts              192.168.3.101,192.168.3.101
Dialplan                XML
Context                 public
Challenge Realm         auto_from
RTP-IP                  192.168.3.101
SIP-IP                  192.168.3.101
URL                     sip:mod_sofia at 192.168.3.101:5060
BIND-URL                sip:mod_sofia at 192.168.3.101:5060
HOLD-MUSIC              local_stream://moh
OUTBOUND-PROXY          N/A
CODECS IN               G7221 at 32000h,G7221 at 16000h,G722,BV32,PCMU,PCMA,GSM
CODECS OUT              G7221 at 32000h,G7221 at 16000h,G722,BV32,PCMU,PCMA,GSM
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
NOMEDIA                 false
LATE-NEG                false
PROXY-MEDIA             false
AGGRESSIVENAT           false
STUN-ENABLED            true
STUN-AUTO-DISABLE       false
CALLS-IN                0
FAILED-CALLS-IN         0
CALLS-OUT               0
FAILED-CALLS-OUT        0

Registrations:
=================================================================================================
Call-ID:        ZDVjOThlZmZhN2JhOGMyYjA1MTUxMzliYzAwMzNhYjU.
User:           1000 at 192.168.3.101
Contact:        "FreeSWITCH 1000" <sip:1000 at 192.168.3.121:49950
;rinstance=15dfcfcc0353b7c7;fs_nat=yes;fs_path=sip%3A1000%40192.168.3.121%3A49950%3Brinstance%3D15dfcfcc0353b7c7>
Agent:          Bria Professional release 2.4.3 stamp 50906
Status:         Registered(UDP-NAT)(unknown) EXP(2011-03-26 02:18:52)
EXPSECS(1146)
Host:           Solaris11Ex
IP:             192.168.3.121
Port:           49950
Auth-User:      1000
Auth-Realm:     192.168.3.101
MWI-Account:    1000 at 192.168.3.101
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From abeka at greatiam.com  Mon Mar 28 15:16:39 2011
From: abeka at greatiam.com (Samuel)
Date: Mon, 28 Mar 2011 12:16:39 +0100
Subject: [Freeswitch-users] Download link to Skyopen
Message-ID: <4D906E17.4000609@greatiam.com>

Hello

Could someone give the link to  the *static* build of the *stable* Skype 
client (2.0.72) as specified in the wiki 
<http://wiki.freeswitch.org/wiki/Skypiax#SHORT_BLUEPRINT:_STEPS_NEEDED_TO_USE_SKYPOPEN>

Thanks

Samuel
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From abeka at greatiam.com  Mon Mar 28 15:21:03 2011
From: abeka at greatiam.com (Samuel)
Date: Mon, 28 Mar 2011 12:21:03 +0100
Subject: [Freeswitch-users] Link to skypopen
Message-ID: <4D906F1F.4010502@greatiam.com>

Hello,

Could someone please give me the link to  the static build of the stable 
Skype client (2.0.72) as specified in the wiki 
<http://wiki.freeswitch.org/wiki/Skypiax#SHORT_BLUEPRINT:_STEPS_NEEDED_TO_USE_SKYPOPEN>


Thanks
Samuel
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From ejay.greeves at yahoo.com  Mon Mar 28 15:46:02 2011
From: ejay.greeves at yahoo.com (Ejay Greeves)
Date: Mon, 28 Mar 2011 12:46:02 +0100 (BST)
Subject: [Freeswitch-users] gsmopen
Message-ID: <618419.12110.qm@web132302.mail.ird.yahoo.com>

I have bluetooth service installed and gsmopen compiled.Now can anyone advise to setup bluetooth in gsmopen. Thanks


      
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From boris at tagnet.ru  Mon Mar 28 15:47:45 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Mon, 28 Mar 2011 17:47:45 +0600
Subject: [Freeswitch-users] Need help with T.38
Message-ID: <4D907561.9050508@tagnet.ru>

Hello!

     My network configuration is:

FAX -- Audiocodecs MP-104 (FXO) -- Freeswitch -- External SIP provider 
--- FAX

When my Freeswitch worked in proxy_media mode there was no troubles with 
sending faxes. Sometimes ago I switched it to default mode and now I 
have many troubles with faxes. I have read wiki about mod_spandsp and 
t38 gateway, but in my situation I don't know from which number fax is 
sending and also to what number fax is sending. And there is no example 
in wiki what to do if FS acts only as proxy when sending fax. May 
somebody give me good example what should I configure to get faxes work?

-- 
Regards,
Boris




From randy.andrade at gmail.com  Mon Mar 28 16:28:16 2011
From: randy.andrade at gmail.com (Randy Andrade)
Date: Mon, 28 Mar 2011 08:28:16 -0400
Subject: [Freeswitch-users] Embarrassing Question: Local extension
 registering in the Public context on default configuration?
In-Reply-To: <AANLkTinDZ0xBMxFssO9NrP8y6opAXHUrMOPoNzhS1U2M@mail.gmail.com>
References: <AANLkTinDZ0xBMxFssO9NrP8y6opAXHUrMOPoNzhS1U2M@mail.gmail.com>
Message-ID: <AANLkTimCcZ33aEgEKi6qmNj8K9Uq9RYBoHr-sPXAByzA@mail.gmail.com>

I'm no FS guru, but I believe that would happen based on the port that your
softphone is using to register w/ the FS system. By default (at least in my
experience), the internal / default profile is setup to listen on port
5060/5061 (ssl) and the external / public profile is setup to listen on port
5080/5081 (ssl). Check to make sure the softphone is set to register on port
5060, and it should re-register on the internal / default profile, unless
there's a local firewall issue on the FS machine preventing it..

Randy

On Mon, Mar 28, 2011 at 6:42 AM, A E [Gmail] <all.eforums at gmail.com> wrote:

> Hi All,
>
> I've used FS in the past to varying degrees but never as a PBX or a
> REGISTRAR and just to test basic config, I built and installed it and didn't
> change a damn thing. Had a softphone on the same internal network registered
> to it using the default password, but it seems to register itself in the
> public context when I'd have thought it'd register in default context so I
> can test all those pre-built extensions for an echo test, dumping channel
> variables etc. But they don't work as the call originated to say 5000 from
> extension 1000 goes into the public context which doesn't have 5000. What is
> going on? It's so embarrassing to ask such a simple question :(
>
> freeswitch at internal> sofia status profile internal 1000
>
> =================================================================================================
> Name                    internal
> Domain Name             N/A
> Auto-NAT                false
> DBName                  sofia_reg_internal
> Pres Hosts              192.168.3.101,192.168.3.101
> Dialplan                XML
> Context                 public
> Challenge Realm         auto_from
> RTP-IP                  192.168.3.101
> SIP-IP                  192.168.3.101
> URL                     sip:mod_sofia at 192.168.3.101:5060
> BIND-URL                sip:mod_sofia at 192.168.3.101:5060
> HOLD-MUSIC              local_stream://moh
> OUTBOUND-PROXY          N/A
> CODECS IN               G7221 at 32000h,G7221 at 16000h,G722,BV32,PCMU,PCMA,GSM
> CODECS OUT              G7221 at 32000h,G7221 at 16000h,G722,BV32,PCMU,PCMA,GSM
> TEL-EVENT               101
> DTMF-MODE               rfc2833
> CNG                     13
> SESSION-TO              0
> MAX-DIALOG              0
> NOMEDIA                 false
> LATE-NEG                false
> PROXY-MEDIA             false
> AGGRESSIVENAT           false
> STUN-ENABLED            true
> STUN-AUTO-DISABLE       false
> CALLS-IN                0
> FAILED-CALLS-IN         0
> CALLS-OUT               0
> FAILED-CALLS-OUT        0
>
> Registrations:
>
> =================================================================================================
> Call-ID:        ZDVjOThlZmZhN2JhOGMyYjA1MTUxMzliYzAwMzNhYjU.
> User:           1000 at 192.168.3.101
> Contact:        "FreeSWITCH 1000" <sip:1000 at 192.168.3.121:49950
> ;rinstance=15dfcfcc0353b7c7;fs_nat=yes;fs_path=sip%3A1000%40192.168.3.121%3A49950%3Brinstance%3D15dfcfcc0353b7c7>
> Agent:          Bria Professional release 2.4.3 stamp 50906
> Status:         Registered(UDP-NAT)(unknown) EXP(2011-03-26 02:18:52)
> EXPSECS(1146)
> Host:           Solaris11Ex
> IP:             192.168.3.121
> Port:           49950
> Auth-User:      1000
> Auth-Realm:     192.168.3.101
> MWI-Account:    1000 at 192.168.3.101
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From motosota at gmail.com  Mon Mar 28 16:43:38 2011
From: motosota at gmail.com (Mike)
Date: Mon, 28 Mar 2011 13:43:38 +0100
Subject: [Freeswitch-users] Stopping uuid_record error
In-Reply-To: <SNT144-w107EBF7DF2D9C9EA0FB742F0B70@phx.gbl>
References: <SNT144-w107EBF7DF2D9C9EA0FB742F0B70@phx.gbl>
Message-ID: <BANLkTimdhumY=KMGyWp9oBhZ7VRHmbkYzw@mail.gmail.com>

Dave,

it seems that if you don't specify a complete path, the 'start' option will
use the sound_prefix directory to put the file in - so unless you have
changed it in vars.xml, in your case the recording will
be /usr/local/freeswitch/sounds/en/us/callie/test.wav

It looks like the 'stop' option doesn't use the same assumption - but if you
specify the full path it will work - so

uuid_record 36dfdf636dfd6f36df
stop /usr/local/freeswitch/sounds/en/us/callie/test.wav

Or a better solution would be to specify the directory you want to use
explicitly when starting the recoding.

Mike

On Wed, Mar 23, 2011 at 5:16 PM, Dave Bracken <richocet2 at hotmail.com> wrote:

>  I have figured out how to start recording using uuid_record, but when i
> try to stop the record, i get: -ERR Cannot stop record session!
> here is my code(with a sudo uuid in place here)
> api uuid_record 36dfdf636dfd6f36df start test.wav
>
> and to stop the record:
> api uuid_record 36dfdf636dfd6f36df stop test.wav
>
> the start one works fine, but the stop one does not. What am i missing
> here?
>
> Thanks in advance,
> Dave
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From benkokakao at gmail.com  Mon Mar 28 16:55:29 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Mon, 28 Mar 2011 14:55:29 +0200
Subject: [Freeswitch-users] German voice prompts - missing exceptions in
	mod_say_de.c
In-Reply-To: <AANLkTinCuxcqoa-aaq-FyXjy8rqEk28JC0ObRt6R5xVS@mail.gmail.com>
References: <AANLkTi=6Q6Q-=S-z9pKT8ofMyCQyB7LRZrrwUNB+KGf_@mail.gmail.com>
	<4D806C50.7050005@xpirio.com>
	<AANLkTinCuxcqoa-aaq-FyXjy8rqEk28JC0ObRt6R5xVS@mail.gmail.com>
Message-ID: <AANLkTim_cdXPqsibbaMHO4o=p+TEKH6iuZQDz64GV91U@mail.gmail.com>

Follow-up: I've patched mod_say_de.c and added
conf/lang/de/vm/sounds.xml - you can find the changes here:
http://jira.freeswitch.org/browse/FS-3195 - please test and let me
know if i missed anything for method "pronounce".

I've made use of the (undocumented) "gender"-parameter for mod_say,
see the updated wiki-doc.

What's still missing are some sound-files as specified in
http://freeswitch.xpirio.com/de2.xls, plus "ein.wav"("ein") and
"1_f.wav"("eine")

Also the patch doesn't fix the 12h vs. 24h issue, mod_say_de still
announces am/pm(mod_say_fr.c might be used as a guide to fix this)

Regards
Christian


From gmaruzz at gmail.com  Mon Mar 28 17:06:24 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 28 Mar 2011 15:06:24 +0200
Subject: [Freeswitch-users] gsmopen and skypopen
Message-ID: <AANLkTimjyiE5veGOPDy+6nhydiKexoDT3i+s+2EyFmvs@mail.gmail.com>

Dear FreeSWITCHers,

in the last weeks I've been too much busy, and I was not present in the
community as I would have liked to be.

I'll be much more present starting next week.

I'm planning new features and more easy of installation for mod_skypopen.

In the last month I saw some interest for gsmopen, that I was thinking to
discontinue.

This obviously is gratifying, and is pushing me to continue with gsmopen
too, and make it more useful.

Stay tuned starting next week ;).

ciao,

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From gmaruzz at gmail.com  Mon Mar 28 17:08:27 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 28 Mar 2011 15:08:27 +0200
Subject: [Freeswitch-users] Link to skypopen
In-Reply-To: <4D906F1F.4010502@greatiam.com>
References: <4D906F1F.4010502@greatiam.com>
Message-ID: <AANLkTik1E4s1an94gyV3P++ERk_79r=E-YzUH=Av_mGR@mail.gmail.com>

That static build is not available from the skype official site.

Some nice person maybe will send it to you.

-giovanni

On Mon, Mar 28, 2011 at 1:21 PM, Samuel <abeka at greatiam.com> wrote:

>  Hello,
>
> Could someone please give me the link to  the static build of the stable
> Skype client (2.0.72) as specified in the wiki<http://wiki.freeswitch.org/wiki/Skypiax#SHORT_BLUEPRINT:_STEPS_NEEDED_TO_USE_SKYPOPEN>
>
>
> Thanks
> Samuel
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From gmaruzz at gmail.com  Mon Mar 28 17:07:05 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 28 Mar 2011 15:07:05 +0200
Subject: [Freeswitch-users] gsmopen
In-Reply-To: <618419.12110.qm@web132302.mail.ird.yahoo.com>
References: <618419.12110.qm@web132302.mail.ird.yahoo.com>
Message-ID: <AANLkTi=Bcmsyk-HAKOKAkZN-D_U2dNojEbKpynGjndS+@mail.gmail.com>

no bluetooth for gsmopen at the moment

On Mon, Mar 28, 2011 at 1:46 PM, Ejay Greeves <ejay.greeves at yahoo.com>wrote:

> I have bluetooth service installed and gsmopen compiled.
> Now can anyone advise to setup bluetooth in gsmopen. Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From davidwaf at gmail.com  Mon Mar 28 17:31:45 2011
From: davidwaf at gmail.com (David Wafula)
Date: Mon, 28 Mar 2011 15:31:45 +0200
Subject: [Freeswitch-users] gsmopen and skypopen
In-Reply-To: <AANLkTimjyiE5veGOPDy+6nhydiKexoDT3i+s+2EyFmvs@mail.gmail.com>
References: <AANLkTimjyiE5veGOPDy+6nhydiKexoDT3i+s+2EyFmvs@mail.gmail.com>
Message-ID: <AANLkTinfn-rPgm-JTbgvmUSRgv1uT1+h9b6LyhPGBwNj@mail.gmail.com>

On Mon, Mar 28, 2011 at 3:06 PM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> Dear FreeSWITCHers,
>
> in the last weeks I've been too much busy, and I was not present in the
> community as I would have liked to be.
>
> I'll be much more present starting next week.
>
> I'm planning new features and more easy of installation for mod_skypopen.
>
> In the last month I saw some interest for gsmopen, that I was thinking to
> discontinue.
>
> This obviously is gratifying, and is pushing me to continue with gsmopen
> too, and make it more useful.
>
> Stay tuned starting next week ;).
>
>


+1 from me!


-- 
David Wafula


From jeff at jefflenk.com  Mon Mar 28 18:07:12 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 28 Mar 2011 07:07:12 -0700 (PDT)
Subject: [Freeswitch-users] Stopping uuid_record error
In-Reply-To: <SNT144-w107EBF7DF2D9C9EA0FB742F0B70@phx.gbl>
References: <SNT144-w107EBF7DF2D9C9EA0FB742F0B70@phx.gbl>
Message-ID: <1301321232027-6215119.post@n2.nabble.com>

you can also specify "all" as the file which removes all bugs for that uuid

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Stopping-uuid-record-error-tp6201087p6215119.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From null at invalid.name  Mon Mar 28 18:14:56 2011
From: null at invalid.name (Dan Lane)
Date: Mon, 28 Mar 2011 15:14:56 +0100
Subject: [Freeswitch-users] transcoding GSM with a high ptime
In-Reply-To: <AANLkTik=Yzg90dP=BMDZCYrJnODm0yJpR1qw42fJe1hy@mail.gmail.com>
References: <AANLkTimc_1WppCuu90n668=LZDfmVduV0X9v5yaiKDfj@mail.gmail.com>
	<AANLkTik=Yzg90dP=BMDZCYrJnODm0yJpR1qw42fJe1hy@mail.gmail.com>
Message-ID: <AANLkTin1BpYf+dmFpSo7gXfkm1XUigKE9Acm+HaaBZ1r@mail.gmail.com>

On Fri, Mar 25, 2011 at 8:39 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> other formats also with 120 ms ptime?

Namely PCMA / PCMU. We've noticed that G722 is limited to 60ms and I
don't seem to be able to get iLBC to work above 40ms.

> You do run the risk of losing a significant amount of audio if you
> drop even 1 120ms packet.
> The other end would be starved for 5 intervals.

Yes, but we're dealing with mobile endpoints behind very jittery connections.

> you could try rtp_timer_name=none on the 120ms leg

That fixed it... rtp_timer_name doesn't appear to be documented very
well on the wiki so are there any negative side-effects to having this
set to none? (I'll update the wiki with anything I learn as I'm doing
this).

Thanks :)


From anthony.minessale at gmail.com  Mon Mar 28 19:04:46 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Mar 2011 10:04:46 -0500
Subject: [Freeswitch-users] transcoding GSM with a high ptime
In-Reply-To: <AANLkTin1BpYf+dmFpSo7gXfkm1XUigKE9Acm+HaaBZ1r@mail.gmail.com>
References: <AANLkTimc_1WppCuu90n668=LZDfmVduV0X9v5yaiKDfj@mail.gmail.com>
	<AANLkTik=Yzg90dP=BMDZCYrJnODm0yJpR1qw42fJe1hy@mail.gmail.com>
	<AANLkTin1BpYf+dmFpSo7gXfkm1XUigKE9Acm+HaaBZ1r@mail.gmail.com>
Message-ID: <AANLkTinuP3_yM3h2CeVhKWkdtyfsQ+VcjBVeff0CYX7K@mail.gmail.com>

The only negative side effect is that with no timer, the channel only
times off the inbound stream so if there is no rtp it would not
playback sound.  This is similar to how asterisk behaves by default.
It's not noticeable at all during bridge but could be noticeable while
on hold if you have vad enabled.


On Mon, Mar 28, 2011 at 9:14 AM, Dan Lane <null at invalid.name> wrote:
> On Fri, Mar 25, 2011 at 8:39 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> other formats also with 120 ms ptime?
>
> Namely PCMA / PCMU. We've noticed that G722 is limited to 60ms and I
> don't seem to be able to get iLBC to work above 40ms.
>
>> You do run the risk of losing a significant amount of audio if you
>> drop even 1 120ms packet.
>> The other end would be starved for 5 intervals.
>
> Yes, but we're dealing with mobile endpoints behind very jittery connections.
>
>> you could try rtp_timer_name=none on the 120ms leg
>
> That fixed it... rtp_timer_name doesn't appear to be documented very
> well on the wiki so are there any negative side-effects to having this
> set to none? (I'll update the wiki with anything I learn as I'm doing
> this).
>
> Thanks :)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 vardar.melih at gmail.com  Mon Mar 28 18:10:03 2011
From: vardar.melih at gmail.com (Melih Vardar)
Date: Mon, 28 Mar 2011 16:10:03 +0200
Subject: [Freeswitch-users] Getting RC=200 OK instead of RC=183 Session
	Progress // Reason: Q.850; cause=16; text="NORMAL_CLEARING"
Message-ID: <AANLkTi=32Qrrmmo4_EU_SnSdSMq5w2=3B-ttQFgdWY+y@mail.gmail.com>

Hello,
I am trying to call a softphone via a Java Applet application.
I have a VPN and there is no proxy just Freeswitch between the 2 phones.

Everything works fine with 2 SJPhones. But trying it with my Applet causes
-> Reason: Q.850;cause=16;text="NORMAL_CLEARING".

When I check the network traffic I see that after 100(Trying) comes directly
200(OK). By using SJPhone it is 183(Session Progress) after 100(Trying) and
the connection is esablished for voice communication.

I also attached the SIP outputs and the log from server ( generated with
"sofia loglevel all 9" ).

I have searched a lot of sites but I haven't found the problem.

I appreciate any help,
Thanks,
Melih




+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++




INVITE sip:1000 at xx.yy.zz.1 SIP/2.0

Call-ID: a830cf50124857e3ba148763920bf674 at xx.yy.zz.66

CSeq: 1 INVITE

From: "My Name" <sip:1016 at xx.yy.zz.1>;tag=3799

To: <sip:1000 at xx.yy.zz.1>

Via: SIP/2.0/UDP
xx.yy.zz.66:4829;rport;branch=z9hG4bK89f52f6729c740781eac05ab4a1a99bb

Max-Forwards: 70

Contact: <sip:1016 at xx.yy.zz.66>

User-Agent: JavaSoftPhone v1.0

Supported: replaces,norefersub,timer

Content-Type: application/sdp

Content-Length: 306



v=0

o=- 814489 815858 IN IP4 xx.yy.zz.66

s=JavaSoftPhone

c=IN IP4 xx.yy.zz.66

t=0 0

m=audio 3454 RTP/AVP 0 3 4 5 101

c=IN IP4 xx.yy.zz.66

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:4 G723/8000

a=rtpmap:5 DVI4/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=setup:active

a=sendrecv

SIP/2.0 100 Trying

Via: SIP/2.0/UDP
xx.yy.zz.66:4829;rport=4829;branch=z9hG4bK89f52f6729c740781eac05ab4a1a99bb

From: "My Name" <sip:1016 at xx.yy.zz.1>;tag=3799

To: <sip:1000 at xx.yy.zz.1>

Call-ID: a830cf50124857e3ba148763920bf674 at xx.yy.zz.66

CSeq: 1 INVITE

User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked

Content-Length: 0



SIP/2.0 200 OK

Via: SIP/2.0/UDP
xx.yy.zz.66:4829;rport=4829;branch=z9hG4bK89f52f6729c740781eac05ab4a1a99bb

From: "My Name" <sip:1016 at xx.yy.zz.1>;tag=3799

To: <sip:1000 at xx.yy.zz.1>;tag=HFr8848ZK8yjQ

Call-ID: a830cf50124857e3ba148763920bf674 at xx.yy.zz.66

CSeq: 1 INVITE

Contact: <sip:1000 at xx.yy.zz.1:5060;transport=udp>

User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked

Accept: application/sdp

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE

Require: timer

Supported: timer, precondition, path, replaces

Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer

Session-Expires: 120;refresher=uac

Min-SE: 120

Content-Type: application/sdp

Content-Disposition: session

Content-Length: 239



v=0

o=FreeSWITCH 1301299753 1301299754 IN IP4 xx.yy.zz.1

s=FreeSWITCH

c=IN IP4 xx.yy.zz.1

t=0 0

m=audio 21030 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

ACK sip:1000 at xx.yy.zz.1:5060;transport=udp SIP/2.0

Call-ID: a830cf50124857e3ba148763920bf674 at xx.yy.zz.66

CSeq: 1 ACK

From: "My Name" <sip:1016 at xx.yy.zz.1>;tag=3799

To: <sip:1000 at xx.yy.zz.1>;tag=HFr8848ZK8yjQ

Via: SIP/2.0/UDP
xx.yy.zz.66:4829;rport;branch=6b3a31302e352e352e36363a3438323

Max-Forwards: 70

Content-Length: 0



BYE sip:1016 at xx.yy.zz.66 SIP/2.0

Via: SIP/2.0/UDP xx.yy.zz.1;rport;branch=z9hG4bKUcjg7QjFeXSyB

Max-Forwards: 70

From: <sip:1000 at xx.yy.zz.1>;tag=HFr8848ZK8yjQ

To: "My Name" <sip:1016 at xx.yy.zz.1>;tag=3799

Call-ID: a830cf50124857e3ba148763920bf674 at xx.yy.zz.66

CSeq: 10324328 BYE

Contact: <sip:1000 at xx.yy.zz.1:5060;transport=udp>

User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE

Supported: timer, precondition, path, replaces

*Reason: Q.850;cause=16;text="NORMAL_CLEARING"*

Content-Length: 0



BYE sip:1016 at xx.yy.zz.66 SIP/2.0

Via: SIP/2.0/UDP xx.yy.zz.1;rport;branch=z9hG4bKUcjg7QjFeXSyB

Max-Forwards: 70

From: <sip:1000 at xx.yy.zz.1>;tag=HFr8848ZK8yjQ

To: "My Name" <sip:1016 at xx.yy.zz.1>;tag=3799

Call-ID: a830cf50124857e3ba148763920bf674 at xx.yy.zz.66

CSeq: 10324328 BYE

Contact: <sip:1000 at xx.yy.zz.1:5060;transport=udp>

User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE

Supported: timer, precondition, path, replaces

*Reason: Q.850;cause=16;text="NORMAL_CLEARING"*

Content-Length: 0



BYE sip:1016 at xx.yy.zz.66 SIP/2.0

Via: SIP/2.0/UDP xx.yy.zz.1;rport;branch=z9hG4bKUcjg7QjFeXSyB

Max-Forwards: 70

From: <sip:1000 at xx.yy.zz.1>;tag=HFr8848ZK8yjQ

To: "My Name" <sip:1016 at xx.yy.zz.1>;tag=3799

Call-ID: a830cf50124857e3ba148763920bf674 at xx.yy.zz.66

CSeq: 10324328 BYE

Contact: <sip:1000 at xx.yy.zz.1:5060;transport=udp>

User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE

Supported: timer, precondition, path, replaces

*Reason: Q.850;cause=16;text="NORMAL_CLEARING"*

Content-Length: 0



BYE sip:1016 at xx.yy.zz.66 SIP/2.0

Via: SIP/2.0/UDP xx.yy.zz.1;rport;branch=z9hG4bKUcjg7QjFeXSyB

Max-Forwards: 70

From: <sip:1000 at xx.yy.zz.1>;tag=HFr8848ZK8yjQ

To: "My Name" <sip:1016 at xx.yy.zz.1>;tag=3799

Call-ID: a830cf50124857e3ba148763920bf674 at xx.yy.zz.66

CSeq: 10324328 BYE

Contact: <sip:1000 at xx.yy.zz.1:5060;transport=udp>

User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE

Supported: timer, precondition, path, replaces

*Reason: Q.850;cause=16;text="NORMAL_CLEARING"*

Content-Length: 0



BYE sip:1016 at xx.yy.zz.66 SIP/2.0

Via: SIP/2.0/UDP xx.yy.zz.1;rport;branch=z9hG4bKUcjg7QjFeXSyB

Max-Forwards: 70

From: <sip:1000 at xx.yy.zz.1>;tag=HFr8848ZK8yjQ

To: "My Name" <sip:1016 at xx.yy.zz.1>;tag=3799

Call-ID: a830cf50124857e3ba148763920bf674 at xx.yy.zz.66

CSeq: 10324328 BYE

Contact: <sip:1000 at xx.yy.zz.1:5060;transport=udp>

User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE

Supported: timer, precondition, path, replaces

*Reason: Q.850;cause=16;text="NORMAL_CLEARING"*

Content-Length: 0



SIP/2.0 200 OK

Via: SIP/2.0/UDP xx.yy.zz.1;rport;branch=z9hG4bKUcjg7QjFeXSyB

Max-Forwards: 70

From: <sip:1000 at xx.yy.zz.1>;tag=HFr8848ZK8yjQ

To: "My Name" <sip:1016 at xx.yy.zz.1>;tag=3799

Call-ID: a830cf50124857e3ba148763920bf674 at xx.yy.zz.66

CSeq: 10324328 BYE

Content-Length: 0



++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++



freeswitch at internal> sofia status profile xx.yy.zz.1
=================================================================================================
Name                    xx.yy.zz.1
Domain Name             N/A
Alias Of                internal
DBName                  sofia_reg_internal
Pres Hosts
Dialplan                XML
Context                 public
Challenge Realm         auto_from
RTP-IP                  xx.yy.zz.1
Ext-RTP-IP              xx.yy.zz.1
SIP-IP                  xx.yy.zz.1
Ext-SIP-IP              xx.yy.zz.1
URL                     sip:mod_sofia at xx.yy.zz.1:5060
BIND-URL                sip:mod_sofia at xx.yy.zz.1:5060
HOLD-MUSIC              local_stream://moh
OUTBOUND-PROXY          N/A
CODECS                  G7221 at 32000h,G7221 at 16000h
,G722,PCMU,PCMA,GSM,speex at 16000h@20i
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
NOMEDIA                 false
LATE-NEG                false
PROXY-MEDIA             false
AGGRESSIVENAT           false
STUN-ENABLED            true
STUN-AUTO-DISABLE       false
CALLS-IN                483
FAILED-CALLS-IN         34
CALLS-OUT               1098
FAILED-CALLS-OUT        865

Registrations:
=================================================================================================
=================================================================================================

freeswitch at internal> nta: timer J fired, terminate 401 response
incoming_reclaim_all((nil), (nil), 0x4123cea0)
nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/17 free
nta: timer set next to 2192 ms
nta: timer J fired, terminate 403 response
incoming_reclaim_all((nil), (nil), 0x4123cea0)
nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/16 free
nta: timer not set
tport_wakeup_pri(0x7028b0): events IN
tport_recv_event(0x7028b0)
tport_recv_iovec(0x7028b0) msg 0x11662d0 from (udp/xx.yy.zz.1:5060) has 364
bytes, veclen = 1
tport_deliver(0x7028b0): msg 0x11662d0 (364 bytes) from
udp/xx.yy.zz.66:5060/sip next=(nil)
nta: received REGISTER sip:xx.yy.zz.1:5060 SIP/2.0 (CSeq 1)
nta: canonizing sip:xx.yy.zz.1:5060 with contact
nta: REGISTER (1) going to a default leg
nua: nua_stack_process_request: entering
nua: nh_create: entering
nua: nh_create_handle: entering
nua: nua_stack_set_params: entering
soa_clone(static::0x6efae0, 0x7154c0, 0x71df40) called
soa_set_params(static::0x1148020, ...) called
nua(0x71df40): event i_register 100 Trying
nua: nua_application_event: entering
nua: nua_respond: entering
nua(0x71df40): sent signal r_respond
nua: nua_handle_destroy: entering
nua(0x71df40): sent signal r_destroy
nua: nua_handle_magic: entering
nua: nua_handle_destroy: entering
nua(0x71df40): recv signal r_respond 401 Unauthorized
nua: nua_stack_set_params: entering
soa_set_params(static::0x1148020, ...) called
tport_tsend(0x7028b0) tpn = UDP/xx.yy.zz.66:4829
tport_resolve addrinfo = xx.yy.zz.66:4829
tport_by_addrinfo(0x7028b0): not found by name UDP/xx.yy.zz.66:4829
tport_vsend(0x7028b0): 628 bytes of 628 to udp/xx.yy.zz.66:4829
tport_vsend returned 628
nta: sent 401 Unauthorized for REGISTER (1)
nta: timer set to 32000 ms
nua(0x71df40): recv signal r_destroy
nta_leg_destroy((nil))
soa_destroy(static::0x1148020) called

freeswitch at internal>
freeswitch at internal>
freeswitch at internal>
freeswitch at internal> tport_wakeup_pri(0x7028b0): events IN
tport_recv_event(0x7028b0)
tport_recv_iovec(0x7028b0) msg 0x768b30 from (udp/xx.yy.zz.1:5060) has 568
bytes, veclen = 1
tport_deliver(0x7028b0): msg 0x768b30 (568 bytes) from
udp/xx.yy.zz.66:5060/sip next=(nil)
nta: received REGISTER sip:xx.yy.zz.1:5060 SIP/2.0 (CSeq 2)
nta: canonizing sip:xx.yy.zz.1:5060 with contact
nta: REGISTER (2) going to a default leg
nua: nua_stack_process_request: entering
nua: nh_create: entering
nua: nh_create_handle: entering
nua: nua_stack_set_params: entering
soa_clone(static::0x6efae0, 0x7154c0, 0x1118070) called
soa_set_params(static::0x1148020, ...) called
nua(0x1118070): event i_register 100 Trying
nua: nua_application_event: entering
nua: nua_respond: entering
nua(0x1118070): sent signal r_respond
nua(0x1118070): recv signal r_respond 200 OK
nua: nua_stack_set_params: entering
soa_set_params(static::0x1148020, ...) called
tport_tsend(0x7028b0) tpn = UDP/xx.yy.zz.66:4829
tport_resolve addrinfo = xx.yy.zz.66:4829
tport_by_addrinfo(0x7028b0): not found by name UDP/xx.yy.zz.66:4829
tport_vsend(0x7028b0): 551 bytes of 551 to udp/xx.yy.zz.66:4829
tport_vsend returned 551
nta: sent 200 OK for REGISTER (2)
nua: nua_handle_destroy: entering
nua(0x1118070): sent signal r_destroy
nua: nua_handle_magic: entering
nua: nua_handle_destroy: entering
nua(0x1118070): recv signal r_destroy
nta_leg_destroy((nil))
soa_destroy(static::0x1148020) called

freeswitch at internal>
freeswitch at internal>
freeswitch at internal>
freeswitch at internal>
freeswitch at internal> tport_wakeup_pri(0x7028b0): events IN
tport_recv_event(0x7028b0)
tport_recv_iovec(0x7028b0) msg 0x1108a50 from (udp/xx.yy.zz.1:5060) has 744
bytes, veclen = 1
tport_deliver(0x7028b0): msg 0x1108a50 (744 bytes) from
udp/xx.yy.zz.66:5060/sip next=(nil)
nta: received INVITE sip:1000 at xx.yy.zz.1 SIP/2.0 (CSeq 1)
nta: canonizing sip:1000 at xx.yy.zz.1 with contact
nta: INVITE (1) going to a default leg
nta: timer shortened to 200 ms
nua: nua_stack_process_request: entering
nua: nh_create: entering
nua: nh_create_handle: entering
nua: nua_stack_set_params: entering
soa_clone(static::0x6efae0, 0x7154c0, 0x11170f0) called
soa_set_params(static::0x1148020, ...) called
nta_leg_tcreate(0x110a160)
soa_init_offer_answer(static::0x1148020) called
soa_set_remote_sdp(static::0x1148020, (nil), 0x117ad96, 306) called
nua(0x11170f0): adding session usage
tport_tsend(0x7028b0) tpn = UDP/xx.yy.zz.66:4829
tport_resolve addrinfo = xx.yy.zz.66:4829
tport_by_addrinfo(0x7028b0): not found by name UDP/xx.yy.zz.66:4829
tport_vsend(0x7028b0): 328 bytes of 328 to udp/xx.yy.zz.66:4829
tport_vsend returned 328
nta: sent 100 Trying for INVITE (1)
nua(0x11170f0): event i_invite 100 Trying
nua(0x11170f0): call state changed: init -> received, received offer
soa_get_remote_sdp(static::0x1148020, [0x4123c558], [0x4123c550], [(nil)])
called
nua(0x11170f0): event i_state 100 Trying
nua: nua_application_event: entering
2011-03-28 15:57:20.161392 [DEBUG] sofia.c:4671 IP xx.yy.zz.66 Approved by
acl "domains[]". Access Granted.
2011-03-28 15:57:20.161392 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/1016 at xx.yy.zz.1 [4c7785ba-5943-11e0-8c96-03c3bfaa409b]
nua: nua_handle_bind: entering
2011-03-28 15:57:20.161392 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/1016 at xx.yy.zz.1) Running State Change CS_NEW
2011-03-28 15:57:20.161392 [DEBUG] switch_core_state_machine.c:404
(sofia/internal/1016 at xx.yy.zz.1) State NEW
2011-03-28 15:57:20.161392 [DEBUG] sofia.c:5408 Setting NAT mode based on
nat.auto
nua: nua_handle_magic: entering
nua: nua_application_event: entering
2011-03-28 15:57:20.161392 [DEBUG] sofia.c:3334 Channel
sofia/internal/1016 at xx.yy.zz.1 entering state [received][100]
2011-03-28 15:57:20.161392 [DEBUG] sofia.c:3341 Remote SDP:
v=0
o=- 652619 653988 IN IP4 xx.yy.zz.66
s=JavaSoftPhone
c=IN IP4 xx.yy.zz.66
t=0 0
m=audio 1370 RTP/AVP 0 3 4 5 101
c=IN IP4 xx.yy.zz.66
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=setup:active

2011-03-28 15:57:20.161392 [DEBUG] sofia_glue.c:3136 Audio Codec Compare
[PCMU:0:8000:0]/[G7221:115:32000:20]
2011-03-28 15:57:20.161392 [DEBUG] sofia_glue.c:3136 Audio Codec Compare
[PCMU:0:8000:0]/[G7221:107:16000:20]
2011-03-28 15:57:20.161392 [DEBUG] sofia_glue.c:3136 Audio Codec Compare
[PCMU:0:8000:0]/[G722:9:8000:20]
2011-03-28 15:57:20.161392 [DEBUG] sofia_glue.c:3136 Audio Codec Compare
[PCMU:0:8000:0]/[PCMU:0:8000:20]
2011-03-28 15:57:20.161392 [DEBUG] sofia_glue.c:2094 Set Codec
sofia/internal/1016 at xx.yy.zz.1 PCMU/8000 20 ms 160 samples
2011-03-28 15:57:20.161392 [DEBUG] sofia_glue.c:3096 Set 2833 dtmf payload
to 101
2011-03-28 15:57:20.161392 [DEBUG] sofia.c:3502
(sofia/internal/1016 at xx.yy.zz.1) State Change CS_NEW -> CS_INIT
2011-03-28 15:57:20.161392 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/1016 at xx.yy.zz.1 [BREAK]
nua: nua_handle_magic: entering
2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/1016 at xx.yy.zz.1) Running State Change CS_INIT
2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:422
(sofia/internal/1016 at xx.yy.zz.1) State INIT
2011-03-28 15:57:20.181410 [DEBUG] mod_sofia.c:83
sofia/internal/1016 at xx.yy.zz.1 SOFIA INIT
2011-03-28 15:57:20.181410 [DEBUG] mod_sofia.c:111
(sofia/internal/1016 at xx.yy.zz.1) State Change CS_INIT -> CS_ROUTING
2011-03-28 15:57:20.181410 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/1016 at xx.yy.zz.1 [BREAK]
2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:422
(sofia/internal/1016 at xx.yy.zz.1) State INIT going to sleep
2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/1016 at xx.yy.zz.1) Running State Change CS_ROUTING
2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:425
(sofia/internal/1016 at xx.yy.zz.1) State ROUTING
2011-03-28 15:57:20.181410 [DEBUG] mod_sofia.c:130
sofia/internal/1016 at xx.yy.zz.1 SOFIA ROUTING
2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:78
sofia/internal/1016 at xx.yy.zz.1 Standard ROUTING
2011-03-28 15:57:20.181410 [INFO] mod_dialplan_xml.c:391 Processing My
Name->1000 in context public
Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [public->unloop]
continue=false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [public->outside_call]
continue=true
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Absolute Condition [outside_call]
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action set(outside_call=true)
Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [public->call_debug]
continue=true
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [call_debug]
${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [public->public_extensions]
continue=false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (PASS) [public_extensions]
destination_number(1000) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action transfer(1000 XML default)
2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:114
(sofia/internal/1016 at xx.yy.zz.1) State Change CS_ROUTING -> CS_EXECUTE
2011-03-28 15:57:20.181410 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/1016 at xx.yy.zz.1 [BREAK]
2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:425
(sofia/internal/1016 at xx.yy.zz.1) State ROUTING going to sleep
2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/1016 at xx.yy.zz.1) Running State Change CS_EXECUTE
2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:432
(sofia/internal/1016 at xx.yy.zz.1) State EXECUTE
2011-03-28 15:57:20.181410 [DEBUG] mod_sofia.c:173
sofia/internal/1016 at xx.yy.zz.1 SOFIA EXECUTE
2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:151
sofia/internal/1016 at xx.yy.zz.1 Standard EXECUTE
EXECUTE sofia/internal/1016 at xx.yy.zz.1 set(outside_call=true)
2011-03-28 15:57:20.181410 [DEBUG] mod_dptools.c:748
sofia/internal/1016 at xx.yy.zz.1 SET [outside_call]=[true]
EXECUTE sofia/internal/1016 at xx.yy.zz.1 transfer(1000 XML default)
2011-03-28 15:57:20.181410 [DEBUG] switch_ivr.c:1344
(sofia/internal/1016 at xx.yy.zz.1) State Change CS_EXECUTE -> CS_ROUTING
2011-03-28 15:57:20.181410 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/1016 at xx.yy.zz.1 [BREAK]
2011-03-28 15:57:20.181410 [DEBUG] switch_ivr.c:1348
sofia/internal/1016 at xx.yy.zz.1 receive message [TRANSFER]
2011-03-28 15:57:20.181410 [DEBUG] switch_core_session.c:630 Send signal
sofia/internal/1016 at xx.yy.zz.1 [BREAK]
2011-03-28 15:57:20.181410 [NOTICE] switch_ivr.c:1350 Transfer
sofia/internal/1016 at xx.yy.zz.1 to XML[1000 at default]
2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:432
(sofia/internal/1016 at xx.yy.zz.1) State EXECUTE going to sleep
2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/1016 at xx.yy.zz.1) Running State Change CS_ROUTING
2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:425
(sofia/internal/1016 at xx.yy.zz.1) State ROUTING
2011-03-28 15:57:20.181410 [DEBUG] mod_sofia.c:130
sofia/internal/1016 at xx.yy.zz.1 SOFIA ROUTING
2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:78
sofia/internal/1016 at xx.yy.zz.1 Standard ROUTING
2011-03-28 15:57:20.181410 [INFO] mod_dialplan_xml.c:391 Processing My
Name->1000 in context default
Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->unloop]
continue=false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->tod_example]
continue=true
Dialplan: day of week[2] =~ 2-6 (PASS)
Dialplan: hour[15] =~ 9-18 (PASS)
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Date/Time Match (PASS)
[tod_example] break=on-false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action set(open=true)
Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->global-intercept]
continue=false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [global-intercept]
destination_number(1000) =~ /^886$/ break=on-false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->group-intercept]
continue=false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [group-intercept]
destination_number(1000) =~ /^\*8$/ break=on-false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->intercept-ext]
continue=false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [intercept-ext]
destination_number(1000) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->redial]
continue=false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [redial]
destination_number(1000) =~ /^870$/ break=on-false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->global]
continue=true
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [global]
${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [global]
${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/
break=never
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Absolute Condition [global]
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action
hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action
hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})

Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action
hash(insert/${domain_name}-last_dial/global/${uuid})
Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->snom-demo-2]
continue=false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [snom-demo-2]
destination_number(1000) =~ /^9001$/ break=on-false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->snom-demo-1]
continue=false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [snom-demo-1]
destination_number(1000) =~ /^9000$/ break=on-false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->eavesdrop]
continue=false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [eavesdrop]
destination_number(1000) =~ /^88(.*)$|^\*0(.*)$/ break=on-false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->eavesdrop]
continue=false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [eavesdrop]
destination_number(1000) =~ /^779$/ break=on-false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->call_return]
continue=false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [call_return]
destination_number(1000) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->del-group]
continue=false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [del-group]
destination_number(1000) =~ /^80(\d{2})$/ break=on-false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->add-group]
continue=false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [add-group]
destination_number(1000) =~ /^81(\d{2})$/ break=on-false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->call-group-simo]
continue=false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [call-group-simo]
destination_number(1000) =~ /^82(\d{2})$/ break=on-false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->call-group-order]
continue=false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [call-group-order]
destination_number(1000) =~ /^83(\d{2})$/ break=on-false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing
[default->extension-intercom] continue=false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [extension-intercom]
destination_number(1000) =~ /^8(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->Local_Extension]
continue=false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (PASS) [Local_Extension]
destination_number(1000) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action set(dialed_extension=1000)
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action
export(dialed_extension=1000)
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action bind_meta_app(1 b s
execute_extension::dx XML features)
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action bind_meta_app(2 b s
record_session::/opt/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav)

Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action bind_meta_app(3 b s
execute_extension::cf XML features)
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action set(ringback=${us-ring})
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action
set(transfer_ringback=local_stream://moh)
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action set(call_timeout=30)
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action
set(hangup_after_bridge=true)
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action set(continue_on_fail=true)
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action
hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number})

Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action
hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action
set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)})
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action
hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action
bridge(user/${dialed_extension}@${domain_name})
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action answer()
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action sleep(1000)
Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action voicemail(default
${domain_name} ${dialed_extension})
2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:114
(sofia/internal/1016 at xx.yy.zz.1) State Change CS_ROUTING -> CS_EXECUTE
2011-03-28 15:57:20.181410 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/1016 at xx.yy.zz.1 [BREAK]
2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:425
(sofia/internal/1016 at xx.yy.zz.1) State ROUTING going to sleep
2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/1016 at xx.yy.zz.1) Running State Change CS_EXECUTE
2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:432
(sofia/internal/1016 at xx.yy.zz.1) State EXECUTE
2011-03-28 15:57:20.181410 [DEBUG] mod_sofia.c:173
sofia/internal/1016 at xx.yy.zz.1 SOFIA EXECUTE
2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:151
sofia/internal/1016 at xx.yy.zz.1 Standard EXECUTE
EXECUTE sofia/internal/1016 at xx.yy.zz.1 set(open=true)
2011-03-28 15:57:20.181410 [DEBUG] mod_dptools.c:748
sofia/internal/1016 at xx.yy.zz.1 SET [open]=[true]
EXECUTE sofia/internal/1016 at xx.yy.zz.1hash(insert/xx.yy.zz.1-spymap/1016/4c7785ba-5943-11e0-8c96-03c3bfaa409b)
EXECUTE sofia/internal/1016 at xx.yy.zz.1hash(insert/xx.yy.zz.1-last_dial/1016/1000)
EXECUTE sofia/internal/1016 at xx.yy.zz.1hash(insert/xx.yy.zz.1-last_dial/global/4c7785ba-5943-11e0-8c96-03c3bfaa409b)
EXECUTE sofia/internal/1016 at xx.yy.zz.1 set(dialed_extension=1000)
2011-03-28 15:57:20.181410 [DEBUG] mod_dptools.c:748
sofia/internal/1016 at xx.yy.zz.1 SET [dialed_extension]=[1000]
EXECUTE sofia/internal/1016 at xx.yy.zz.1 export(dialed_extension=1000)
2011-03-28 15:57:20.181410 [DEBUG] mod_dptools.c:886 EXPORT
[dialed_extension]=[1000]
EXECUTE sofia/internal/1016 at xx.yy.zz.1 bind_meta_app(1 b s
execute_extension::dx XML features)
2011-03-28 15:57:20.181410 [INFO] switch_ivr_async.c:1835 Bound B-Leg: 1
execute_extension::dx XML features
EXECUTE sofia/internal/1016 at xx.yy.zz.1 bind_meta_app(2 b s
record_session::/opt/freeswitch/recordings/1016.2011-03-28-15-57-20.wav)
2011-03-28 15:57:20.181410 [INFO] switch_ivr_async.c:1835 Bound B-Leg: 2
record_session::/opt/freeswitch/recordings/1016.2011-03-28-15-57-20.wav
EXECUTE sofia/internal/1016 at xx.yy.zz.1 bind_meta_app(3 b s
execute_extension::cf XML features)
2011-03-28 15:57:20.181410 [INFO] switch_ivr_async.c:1835 Bound B-Leg: 3
execute_extension::cf XML features
EXECUTE sofia/internal/1016 at xx.yy.zz.1set(ringback=%(2000,4000,440.0,480.0))
2011-03-28 15:57:20.181410 [DEBUG] mod_dptools.c:748
sofia/internal/1016 at xx.yy.zz.1 SET [ringback]=[%(2000,4000,440.0,480.0)]
EXECUTE sofia/internal/1016 at xx.yy.zz.1set(transfer_ringback=local_stream://moh)
2011-03-28 15:57:20.181410 [DEBUG] mod_dptools.c:748
sofia/internal/1016 at xx.yy.zz.1 SET [transfer_ringback]=[local_stream://moh]
EXECUTE sofia/internal/1016 at xx.yy.zz.1 set(call_timeout=30)
2011-03-28 15:57:20.191371 [DEBUG] mod_dptools.c:748
sofia/internal/1016 at xx.yy.zz.1 SET [call_timeout]=[30]
EXECUTE sofia/internal/1016 at xx.yy.zz.1 set(hangup_after_bridge=true)
2011-03-28 15:57:20.191371 [DEBUG] mod_dptools.c:748
sofia/internal/1016 at xx.yy.zz.1 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/1016 at xx.yy.zz.1 set(continue_on_fail=true)
2011-03-28 15:57:20.191371 [DEBUG] mod_dptools.c:748
sofia/internal/1016 at xx.yy.zz.1 SET [continue_on_fail]=[true]
EXECUTE sofia/internal/1016 at xx.yy.zz.1hash(insert/xx.yy.zz.1-call_return/1000/1016)
EXECUTE sofia/internal/1016 at xx.yy.zz.1hash(insert/xx.yy.zz.1-last_dial_ext/1000/4c7785ba-5943-11e0-8c96-03c3bfaa409b)
EXECUTE sofia/internal/1016 at xx.yy.zz.1set(called_party_callgroup=techsupport)
2011-03-28 15:57:20.191371 [DEBUG] mod_dptools.c:748
sofia/internal/1016 at xx.yy.zz.1 SET [called_party_callgroup]=[techsupport]
EXECUTE sofia/internal/1016 at xx.yy.zz.1hash(insert/xx.yy.zz.1-last_dial/techsupport/4c7785ba-5943-11e0-8c96-03c3bfaa409b)
EXECUTE sofia/internal/1016 at xx.yy.zz.1 bridge(user/1000 at xx.yy.zz.1)
2011-03-28 15:57:20.191371 [DEBUG] switch_ivr_originate.c:1047 variable
string 0 = [presence_id=1000 at xx.yy.zz.1]
2011-03-28 15:57:20.191371 [ERR] switch_ivr_originate.c:1531 Cannot create
outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
2011-03-28 15:57:20.191371 [DEBUG] switch_ivr_originate.c:2170 Originate
Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2011-03-28 15:57:20.191371 [ERR] switch_ivr_originate.c:1531 Cannot create
outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
2011-03-28 15:57:20.191371 [DEBUG] switch_ivr_originate.c:2170 Originate
Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2011-03-28 15:57:20.191371 [INFO] mod_dptools.c:2114 Originate Failed.
Cause: USER_NOT_REGISTERED
2011-03-28 15:57:20.191371 [DEBUG] mod_dptools.c:2141 Continue on fail
[true]:  Cause: USER_NOT_REGISTERED
EXECUTE sofia/internal/1016 at xx.yy.zz.1 answer()
2011-03-28 15:57:20.191371 [DEBUG] mod_dptools.c:649
sofia/internal/1016 at xx.yy.zz.1 receive message [ANSWER]
2011-03-28 15:57:20.191371 [DEBUG] sofia_glue.c:2328 AUDIO RTP
[sofia/internal/1016 at xx.yy.zz.1] xx.yy.zz.1 port 17768 -> xx.yy.zz.66 port
1370 codec: 0 ms: 20
2011-03-28 15:57:20.191371 [DEBUG] switch_rtp.c:1155 Starting timer [soft]
160 bytes per 20ms
2011-03-28 15:57:20.201410 [DEBUG] mod_sofia.c:536 Local SDP
sofia/internal/1016 at xx.yy.zz.1:
v=0
o=FreeSWITCH 1301302872 1301302873 IN IP4 xx.yy.zz.1
s=FreeSWITCH
c=IN IP4 xx.yy.zz.1
t=0 0
m=audio 17768 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

nua: nua_respond: entering
nua(0x11170f0): sent signal r_respond
2011-03-28 15:57:20.201410 [DEBUG] switch_core_session.c:630 Send signal
sofia/internal/1016 at xx.yy.zz.1 [BREAK]
2011-03-28 15:57:20.201410 [NOTICE] mod_dptools.c:649 Channel
[sofia/internal/1016 at xx.yy.zz.1] has been answered
2011-03-28 15:57:20.201410 [DEBUG] switch_channel.c:182
sofia/internal/1016 at xx.yy.zz.1 receive message [AUDIO_SYNC]
EXECUTE sofia/internal/1016 at xx.yy.zz.1 sleep(1000)
nua(0x11170f0): recv signal r_respond 200 OK
nua: nua_stack_set_params: entering
soa_set_params(static::0x1148020, ...) called
soa_set_user_sdp(static::0x1148020, (nil), 0x7fb13c02316d, -1) called
soa_set_capability_sdp(static::0x1148020, (nil), 0x7fb13c02316d, -1) called
nua: nua_invite_server_respond: entering
soa_generate_answer(static::0x1148020) called
soa_static_offer_answer_action(0x1148020, soa_generate_answer): called
soa_static(0x1148020, soa_generate_answer): generating local description
soa_static(0x1148020, soa_generate_answer): upgrade with remote description
soa_sdp_mode_set(0x4123caa0, 0x116ef80, ""): called
soa_static(0x1148020, soa_generate_answer): storing local description
soa_activate(static::0x1148020, (nil)) called
soa_get_local_sdp(static::0x1148020, [(nil)], [0x4123cc90], [0x4123cc9c])
called
tport_tsend(0x7028b0) tpn = UDP/xx.yy.zz.66:4829
tport_resolve addrinfo = xx.yy.zz.66:4829
tport_by_addrinfo(0x7028b0): not found by name UDP/xx.yy.zz.66:4829
tport_vsend(0x7028b0): 1064 bytes of 1064 to udp/xx.yy.zz.66:4829
tport_vsend returned 1064
nta: sent 200 OK for INVITE (1)
nua(0x11170f0): call state changed: received -> completed, sent answer
soa_get_local_sdp(static::0x1148020, [0x4123cde8], [0x4123cde0], [(nil)])
called
soa_get_params(static::0x1148020, ...) called
nua(0x11170f0): event i_state 200 OK
2011-03-28 15:57:20.201410 [DEBUG] switch_channel.c:182
sofia/internal/1016 at xx.yy.zz.1 receive message [AUDIO_SYNC]
nua: nua_application_event: entering
2011-03-28 15:57:20.201410 [DEBUG] sofia.c:3334 Channel
sofia/internal/1016 at xx.yy.zz.1 entering state [completed][200]
nua: nua_handle_magic: entering
tport_wakeup_pri(0x7028b0): events IN
tport_recv_event(0x7028b0)
tport_recv_iovec(0x7028b0) msg 0x1125e50 from (udp/xx.yy.zz.1:5060) has 327
bytes, veclen = 1
tport_deliver(0x7028b0): msg 0x1125e50 (327 bytes) from
udp/xx.yy.zz.66:5060/sip next=(nil)
nta: received ACK sip:1000 at xx.yy.zz.1:5060;transport=udp SIP/2.0 (CSeq 1)
nta: ACK (1) is going to INVITE (1)
nua: process_ack_or_cancel: entering
soa_clear_remote_sdp(static::0x1148020) called
nua(0x11170f0): event i_ack 200 OK
nua(0x11170f0): call state changed: completed -> ready
nua(0x11170f0): event i_state 200 OK
nua(0x11170f0): event i_active 200 Call active
nua(): refresh session after 88 seconds (in [88..88])
nua: nua_application_event: entering
nua: nua_handle_magic: entering
nua: nua_application_event: entering
2011-03-28 15:57:20.231367 [DEBUG] sofia.c:3334 Channel
sofia/internal/1016 at xx.yy.zz.1 entering state [ready][200]
nua: nua_handle_magic: entering
nua: nua_application_event: entering
nua: nua_handle_magic: entering
nta: timer set next to 4862 ms
EXECUTE sofia/internal/1016 at xx.yy.zz.1 voicemail(default xx.yy.zz.1 1000)
2011-03-28 15:57:21.201425 [DEBUG] mod_voicemail.c:788 [default] rwlock
2011-03-28 15:57:21.201425 [ERR] mod_voicemail.c:2927 Error creating
/opt/freeswitch/storage/voicemail/default/xx.yy.zz.1/1000
2011-03-28 15:57:21.201425 [DEBUG] switch_ivr_play_say.c:118 No language
specified - Using [en]
2011-03-28 15:57:21.201425 [ERR] switch_ivr_play_say.c:148 Can't find
language tag.
2011-03-28 15:57:21.201425 [WARNING] switch_ivr_play_say.c:364 Macro
[voicemail_goodbye] did not match any patterns
2011-03-28 15:57:21.201425 [NOTICE] switch_core_state_machine.c:179 Hangup
sofia/internal/1016 at xx.yy.zz.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-28 15:57:21.201425 [DEBUG] switch_channel.c:1726 Send signal
sofia/internal/1016 at xx.yy.zz.1 [KILL]
2011-03-28 15:57:21.201425 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/1016 at xx.yy.zz.1 [BREAK]
2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:560
(sofia/internal/1016 at xx.yy.zz.1) State HANGUP
2011-03-28 15:57:21.201425 [DEBUG] mod_sofia.c:338 Channel
sofia/internal/1016 at xx.yy.zz.1 hanging up, cause: NORMAL_CLEARING
2011-03-28 15:57:21.201425 [DEBUG] mod_sofia.c:376 Sending BYE to
sofia/internal/1016 at xx.yy.zz.1
nua: nua_bye: entering
nua(0x11170f0): sent signal r_bye
nua(0x11170f0): recv signal r_bye
nua: nua_stack_set_params: entering
soa_set_params(static::0x1148020, ...) called
soa_terminate(static::0x1148020) called
soa_init_offer_answer(static::0x1148020) called
nta: selecting scheme sip
tport_tsend(0x7028b0) tpn = udp/xx.yy.zz.66:4829
tport_resolve addrinfo = xx.yy.zz.66:4829
tport_by_addrinfo(0x7028b0): not found by name udp/xx.yy.zz.66:4829
tport_vsend(0x7028b0): 604 bytes of 604 to udp/xx.yy.zz.66:4829
tport_vsend returned 604
nta: sent BYE (10324256) to udp/xx.yy.zz.66:4829
tport_pend(0x7028b0): pending 0x7fb1440d6100 for udp/xx.yy.zz.1:5060
(already 0)
nta: timer shortened to 500 ms
2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:46
sofia/internal/1016 at xx.yy.zz.1 Standard HANGUP, cause: NORMAL_CLEARING
2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:560
(sofia/internal/1016 at xx.yy.zz.1) State HANGUP going to sleep
2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:432
(sofia/internal/1016 at xx.yy.zz.1) State EXECUTE going to sleep
2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/1016 at xx.yy.zz.1) Running State Change CS_HANGUP
2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:538 handler
already called, skipping state handler.
2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:417
(sofia/internal/1016 at xx.yy.zz.1) State Change CS_HANGUP -> CS_REPORTING
2011-03-28 15:57:21.201425 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/1016 at xx.yy.zz.1 [BREAK]
2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/1016 at xx.yy.zz.1) Running State Change CS_REPORTING
2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:651
(sofia/internal/1016 at xx.yy.zz.1) State REPORTING
2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:53
sofia/internal/1016 at xx.yy.zz.1 Standard REPORTING, cause: NORMAL_CLEARING
2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:651
(sofia/internal/1016 at xx.yy.zz.1) State REPORTING going to sleep
2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:411
(sofia/internal/1016 at xx.yy.zz.1) State Change CS_REPORTING -> CS_DESTROY
2011-03-28 15:57:21.201425 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/1016 at xx.yy.zz.1 [BREAK]
2011-03-28 15:57:21.201425 [DEBUG] switch_core_session.c:1069 Session 1582
(sofia/internal/1016 at xx.yy.zz.1) Locked, Waiting on external entities
2011-03-28 15:57:21.201425 [NOTICE] switch_core_session.c:1087 Session 1582
(sofia/internal/1016 at xx.yy.zz.1) Ended
2011-03-28 15:57:21.201425 [NOTICE] switch_core_session.c:1089 Close Channel
sofia/internal/1016 at xx.yy.zz.1 [CS_DESTROY]
2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:497
(sofia/internal/1016 at xx.yy.zz.1) Running State Change CS_DESTROY
2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:508
(sofia/internal/1016 at xx.yy.zz.1) State DESTROY
2011-03-28 15:57:21.201425 [DEBUG] mod_sofia.c:255
sofia/internal/1016 at xx.yy.zz.1 SOFIA DESTROY
2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:60
sofia/internal/1016 at xx.yy.zz.1 Standard DESTROY
2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:508
(sofia/internal/1016 at xx.yy.zz.1) State DESTROY going to sleep
nta: timer E fired, retransmit BYE (10324256)
tport_release(0x7028b0): 0x7fb1440d6100 by 0x7fb144068530 with (nil)
tport_tsend(0x7028b0) tpn = udp/xx.yy.zz.66:4829
tport_resolve addrinfo = xx.yy.zz.66:4829
tport_by_addrinfo(0x7028b0): not found by name udp/xx.yy.zz.66:4829
tport_vsend(0x7028b0): 604 bytes of 604 to udp/xx.yy.zz.66:4829
tport_vsend returned 604
nta: resent BYE (10324256) to udp/xx.yy.zz.66:4829
tport_pend(0x7028b0): pending 0x7fb1440d6100 for udp/xx.yy.zz.1:5060
(already 0)
nta_outgoing_timer: 1/1 resent, 0/1 tout, 0/0 term, 0/1 free
nta: timer set next to 1000 ms
nta: timer E fired, retransmit BYE (10324256)
tport_release(0x7028b0): 0x7fb1440d6100 by 0x7fb144068530 with (nil)
tport_tsend(0x7028b0) tpn = udp/xx.yy.zz.66:4829
tport_resolve addrinfo = xx.yy.zz.66:4829
tport_by_addrinfo(0x7028b0): not found by name udp/xx.yy.zz.66:4829
tport_vsend(0x7028b0): 604 bytes of 604 to udp/xx.yy.zz.66:4829
tport_vsend returned 604
nta: resent BYE (10324256) to udp/xx.yy.zz.66:4829
tport_pend(0x7028b0): pending 0x7fb1440d6100 for udp/xx.yy.zz.1:5060
(already 0)
nta_outgoing_timer: 1/1 resent, 0/1 tout, 0/0 term, 0/1 free
nta: timer set next to 2000 ms
nta: timer E fired, retransmit BYE (10324256)
tport_release(0x7028b0): 0x7fb1440d6100 by 0x7fb144068530 with (nil)
tport_tsend(0x7028b0) tpn = udp/xx.yy.zz.66:4829
tport_resolve addrinfo = xx.yy.zz.66:4829
tport_by_addrinfo(0x7028b0): not found by name udp/xx.yy.zz.66:4829
tport_vsend(0x7028b0): 604 bytes of 604 to udp/xx.yy.zz.66:4829
tport_vsend returned 604
nta: resent BYE (10324256) to udp/xx.yy.zz.66:4829
tport_pend(0x7028b0): pending 0x7fb1440d6100 for udp/xx.yy.zz.1:5060
(already 0)
nta_outgoing_timer: 1/1 resent, 0/1 tout, 0/0 term, 0/1 free
nta: timer set next to 522 ms
nta: timer I fired, terminate 200 response
incoming_reclaim_all((nil), (nil), 0x4123cea0)
nta_incoming_timer: 0/0 resent, 0/0 tout, 1/3 term, 1/18 free
nta: timer set next to 2738 ms
nta: timer J fired, terminate 401 response
incoming_reclaim_all((nil), (nil), 0x4123cea0)
nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/17 free
nta: timer set next to 730 ms
nta: timer E fired, retransmit BYE (10324256)
tport_release(0x7028b0): 0x7fb1440d6100 by 0x7fb144068530 with (nil)
tport_tsend(0x7028b0) tpn = udp/xx.yy.zz.66:4829
tport_resolve addrinfo = xx.yy.zz.66:4829
tport_by_addrinfo(0x7028b0): not found by name udp/xx.yy.zz.66:4829
tport_vsend(0x7028b0): 604 bytes of 604 to udp/xx.yy.zz.66:4829
tport_vsend returned 604
nta: resent BYE (10324256) to udp/xx.yy.zz.66:4829
tport_pend(0x7028b0): pending 0x7fb1440d6100 for udp/xx.yy.zz.1:5060
(already 0)
nta_outgoing_timer: 1/1 resent, 0/1 tout, 0/0 term, 0/1 free
nta: timer set next to 4000 ms
tport_wakeup_pri(0x7028b0): events IN
tport_recv_event(0x7028b0)
tport_recv_iovec(0x7028b0) msg 0x117e8c0 from (udp/xx.yy.zz.1:5060) has 283
bytes, veclen = 1
tport_deliver(0x7028b0): msg 0x117e8c0 (283 bytes) from
udp/xx.yy.zz.66:5060/sip next=(nil)
nta: received 200 OK for BYE (10324256)
nta: 200 OK is going to a transaction
nta_outgoing: RTT is 8366.26 ms
tport_release(0x7028b0): 0x7fb1440d6100 by 0x7fb144068530 with 0x117e8c0
nua(0x11170f0): event r_bye 200 OK
nua(0x11170f0): call state changed: terminating -> terminated
nua(0x11170f0): event i_state 200 to BYE
nua(0x11170f0): event i_terminated 200 to BYE
nua(0x11170f0): removing session usage
soa_destroy(static::0x1148020) called
nta_leg_destroy(0x110a160)
nua: terminated session 0x11170f0
nua: nua_application_event: entering
nua: nua_handle_magic: entering
nua: nua_application_event: entering
nua: nua_handle_magic: entering
nua: nua_handle_bind: entering
nua: nua_application_event: entering
nua: nua_handle_magic: entering
nua: nua_handle_destroy: entering
nua(0x11170f0): recv signal r_destroy
nta_leg_destroy((nil))
nua(0x11170f0): sent signal r_destroy
nta: timer set next to 1856 ms
nta: timer K fired, terminate BYE (10324256)
outgoing_reclaim_all((nil), (nil), 0x4123ce90)
nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
nta: timer set next to 8799 ms
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From Nabble at slickdeals.endjunk.com  Mon Mar 28 20:34:11 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 28 Mar 2011 09:34:11 -0700 (PDT)
Subject: [Freeswitch-users] Link to skypopen
In-Reply-To: <4D906F1F.4010502@greatiam.com>
References: <4D906F1F.4010502@greatiam.com>
Message-ID: <1301330051052-6215693.post@n2.nabble.com>

Have a look at my 
http://freeswitch-users.2379917.n2.nabble.com/Skype-2-0-72-for-Linux-tp6038235p6038482.html
post .

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Link-to-skypopen-tp6214508p6215693.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Mon Mar 28 20:56:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 28 Mar 2011 09:56:38 -0700
Subject: [Freeswitch-users] transcoding GSM with a high ptime
In-Reply-To: <AANLkTin1BpYf+dmFpSo7gXfkm1XUigKE9Acm+HaaBZ1r@mail.gmail.com>
References: <AANLkTimc_1WppCuu90n668=LZDfmVduV0X9v5yaiKDfj@mail.gmail.com>
	<AANLkTik=Yzg90dP=BMDZCYrJnODm0yJpR1qw42fJe1hy@mail.gmail.com>
	<AANLkTin1BpYf+dmFpSo7gXfkm1XUigKE9Acm+HaaBZ1r@mail.gmail.com>
Message-ID: <AANLkTinEqpdiERoBv2qRh1AmFzGBm_QB2wt4zTGTxyQL@mail.gmail.com>

>> you could try rtp_timer_name=none on the 120ms leg
>
> That fixed it... rtp_timer_name doesn't appear to be documented very
> well on the wiki so are there any negative side-effects to having this
> set to none? (I'll update the wiki with anything I learn as I'm doing
> this).

Dan,

Thanks for helping w/ the wiki. Let me know if you have any questions
about editing on the wiki.
-MC


From kerem.erciyes at gmail.com  Mon Mar 28 20:54:39 2011
From: kerem.erciyes at gmail.com (Kerem Erciyes)
Date: Mon, 28 Mar 2011 19:54:39 +0300
Subject: [Freeswitch-users] Getting RC=200 OK instead of RC=183 Session
 Progress // Reason: Q.850; cause=16; text="NORMAL_CLEARING"
In-Reply-To: <AANLkTi=32Qrrmmo4_EU_SnSdSMq5w2=3B-ttQFgdWY+y@mail.gmail.com>
References: <AANLkTi=32Qrrmmo4_EU_SnSdSMq5w2=3B-ttQFgdWY+y@mail.gmail.com>
Message-ID: <AANLkTik3jqymoufNUTSrbgMhA6y7vbsXRyG31NDY-Roz@mail.gmail.com>

Hi Melih,

Read the output, the error is USER_NOT_REGISTERED. You have
registration and authentication problems.

<snip>
EXECUTE sofia/internal/1016 at xx.yy.zz.1 bridge(user/1000 at xx.yy.zz.1)
2011-03-28 15:57:20.191371 [DEBUG] switch_ivr_originate.c:1047
variable string 0 = [presence_id=1000 at xx.yy.zz.1]
2011-03-28 15:57:20.191371 [ERR] switch_ivr_originate.c:1531 Cannot
create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
2011-03-28 15:57:20.191371 [DEBUG] switch_ivr_originate.c:2170
Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2011-03-28 15:57:20.191371 [ERR] switch_ivr_originate.c:1531 Cannot
create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
2011-03-28 15:57:20.191371 [DEBUG] switch_ivr_originate.c:2170
Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
</snip>

Also try using pastebin for pasting your logs and conf files. See:
http://wiki.freeswitch.org/wiki/Reporting_Bugs#Using_The_Pastebin

--KE

On Mon, Mar 28, 2011 at 5:10 PM, Melih Vardar <vardar.melih at gmail.com> wrote:
> Hello,
> I am trying to call a softphone via a Java Applet application.
> I have a VPN and there is no proxy just Freeswitch between the 2 phones.
>
> Everything works fine with 2 SJPhones. But trying it with my Applet causes
> -> Reason: Q.850;cause=16;text="NORMAL_CLEARING".
>
> When I check the network traffic I see that after 100(Trying) comes directly
> 200(OK). By using SJPhone it is 183(Session Progress) after 100(Trying) and
> the connection is esablished for voice communication.
>
> I also attached the SIP outputs and the log from server ( generated with
> "sofia loglevel all 9" ).
>
> I have searched a lot of sites but I haven't found the problem.
>
> I appreciate any help,
> Thanks,
> Melih
>
>
>
>
> +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
>
>
>
>
> INVITE sip:1000 at xx.yy.zz.1 SIP/2.0
>
> Call-ID: a830cf50124857e3ba148763920bf674 at xx.yy.zz.66
>
> CSeq: 1 INVITE
>
> From: "My Name" <sip:1016 at xx.yy.zz.1>;tag=3799
>
> To: <sip:1000 at xx.yy.zz.1>
>
> Via: SIP/2.0/UDP
> xx.yy.zz.66:4829;rport;branch=z9hG4bK89f52f6729c740781eac05ab4a1a99bb
>
> Max-Forwards: 70
>
> Contact: <sip:1016 at xx.yy.zz.66>
>
> User-Agent: JavaSoftPhone v1.0
>
> Supported: replaces,norefersub,timer
>
> Content-Type: application/sdp
>
> Content-Length: 306
>
>
>
> v=0
>
> o=- 814489 815858 IN IP4 xx.yy.zz.66
>
> s=JavaSoftPhone
>
> c=IN IP4 xx.yy.zz.66
>
> t=0 0
>
> m=audio 3454 RTP/AVP 0 3 4 5 101
>
> c=IN IP4 xx.yy.zz.66
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:3 GSM/8000
>
> a=rtpmap:4 G723/8000
>
> a=rtpmap:5 DVI4/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-16
>
> a=setup:active
>
> a=sendrecv
>
> SIP/2.0 100 Trying
>
> Via: SIP/2.0/UDP
> xx.yy.zz.66:4829;rport=4829;branch=z9hG4bK89f52f6729c740781eac05ab4a1a99bb
>
> From: "My Name" <sip:1016 at xx.yy.zz.1>;tag=3799
>
> To: <sip:1000 at xx.yy.zz.1>
>
> Call-ID: a830cf50124857e3ba148763920bf674 at xx.yy.zz.66
>
> CSeq: 1 INVITE
>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked
>
> Content-Length: 0
>
>
>
> SIP/2.0 200 OK
>
> Via: SIP/2.0/UDP
> xx.yy.zz.66:4829;rport=4829;branch=z9hG4bK89f52f6729c740781eac05ab4a1a99bb
>
> From: "My Name" <sip:1016 at xx.yy.zz.1>;tag=3799
>
> To: <sip:1000 at xx.yy.zz.1>;tag=HFr8848ZK8yjQ
>
> Call-ID: a830cf50124857e3ba148763920bf674 at xx.yy.zz.66
>
> CSeq: 1 INVITE
>
> Contact: <sip:1000 at xx.yy.zz.1:5060;transport=udp>
>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked
>
> Accept: application/sdp
>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
>
> Require: timer
>
> Supported: timer, precondition, path, replaces
>
> Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>
> Session-Expires: 120;refresher=uac
>
> Min-SE: 120
>
> Content-Type: application/sdp
>
> Content-Disposition: session
>
> Content-Length: 239
>
>
>
> v=0
>
> o=FreeSWITCH 1301299753 1301299754 IN IP4 xx.yy.zz.1
>
> s=FreeSWITCH
>
> c=IN IP4 xx.yy.zz.1
>
> t=0 0
>
> m=audio 21030 RTP/AVP 0 101
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-16
>
> a=silenceSupp:off - - - -
>
> a=ptime:20
>
> ACK sip:1000 at xx.yy.zz.1:5060;transport=udp SIP/2.0
>
> Call-ID: a830cf50124857e3ba148763920bf674 at xx.yy.zz.66
>
> CSeq: 1 ACK
>
> From: "My Name" <sip:1016 at xx.yy.zz.1>;tag=3799
>
> To: <sip:1000 at xx.yy.zz.1>;tag=HFr8848ZK8yjQ
>
> Via: SIP/2.0/UDP
> xx.yy.zz.66:4829;rport;branch=6b3a31302e352e352e36363a3438323
>
> Max-Forwards: 70
>
> Content-Length: 0
>
>
>
> BYE sip:1016 at xx.yy.zz.66 SIP/2.0
>
> Via: SIP/2.0/UDP xx.yy.zz.1;rport;branch=z9hG4bKUcjg7QjFeXSyB
>
> Max-Forwards: 70
>
> From: <sip:1000 at xx.yy.zz.1>;tag=HFr8848ZK8yjQ
>
> To: "My Name" <sip:1016 at xx.yy.zz.1>;tag=3799
>
> Call-ID: a830cf50124857e3ba148763920bf674 at xx.yy.zz.66
>
> CSeq: 10324328 BYE
>
> Contact: <sip:1000 at xx.yy.zz.1:5060;transport=udp>
>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked
>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
>
> Supported: timer, precondition, path, replaces
>
> Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>
> Content-Length: 0
>
>
>
> BYE sip:1016 at xx.yy.zz.66 SIP/2.0
>
> Via: SIP/2.0/UDP xx.yy.zz.1;rport;branch=z9hG4bKUcjg7QjFeXSyB
>
> Max-Forwards: 70
>
> From: <sip:1000 at xx.yy.zz.1>;tag=HFr8848ZK8yjQ
>
> To: "My Name" <sip:1016 at xx.yy.zz.1>;tag=3799
>
> Call-ID: a830cf50124857e3ba148763920bf674 at xx.yy.zz.66
>
> CSeq: 10324328 BYE
>
> Contact: <sip:1000 at xx.yy.zz.1:5060;transport=udp>
>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked
>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
>
> Supported: timer, precondition, path, replaces
>
> Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>
> Content-Length: 0
>
>
>
> BYE sip:1016 at xx.yy.zz.66 SIP/2.0
>
> Via: SIP/2.0/UDP xx.yy.zz.1;rport;branch=z9hG4bKUcjg7QjFeXSyB
>
> Max-Forwards: 70
>
> From: <sip:1000 at xx.yy.zz.1>;tag=HFr8848ZK8yjQ
>
> To: "My Name" <sip:1016 at xx.yy.zz.1>;tag=3799
>
> Call-ID: a830cf50124857e3ba148763920bf674 at xx.yy.zz.66
>
> CSeq: 10324328 BYE
>
> Contact: <sip:1000 at xx.yy.zz.1:5060;transport=udp>
>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked
>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
>
> Supported: timer, precondition, path, replaces
>
> Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>
> Content-Length: 0
>
>
>
> BYE sip:1016 at xx.yy.zz.66 SIP/2.0
>
> Via: SIP/2.0/UDP xx.yy.zz.1;rport;branch=z9hG4bKUcjg7QjFeXSyB
>
> Max-Forwards: 70
>
> From: <sip:1000 at xx.yy.zz.1>;tag=HFr8848ZK8yjQ
>
> To: "My Name" <sip:1016 at xx.yy.zz.1>;tag=3799
>
> Call-ID: a830cf50124857e3ba148763920bf674 at xx.yy.zz.66
>
> CSeq: 10324328 BYE
>
> Contact: <sip:1000 at xx.yy.zz.1:5060;transport=udp>
>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked
>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
>
> Supported: timer, precondition, path, replaces
>
> Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>
> Content-Length: 0
>
>
>
> BYE sip:1016 at xx.yy.zz.66 SIP/2.0
>
> Via: SIP/2.0/UDP xx.yy.zz.1;rport;branch=z9hG4bKUcjg7QjFeXSyB
>
> Max-Forwards: 70
>
> From: <sip:1000 at xx.yy.zz.1>;tag=HFr8848ZK8yjQ
>
> To: "My Name" <sip:1016 at xx.yy.zz.1>;tag=3799
>
> Call-ID: a830cf50124857e3ba148763920bf674 at xx.yy.zz.66
>
> CSeq: 10324328 BYE
>
> Contact: <sip:1000 at xx.yy.zz.1:5060;transport=udp>
>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked
>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
>
> Supported: timer, precondition, path, replaces
>
> Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>
> Content-Length: 0
>
>
>
> SIP/2.0 200 OK
>
> Via: SIP/2.0/UDP xx.yy.zz.1;rport;branch=z9hG4bKUcjg7QjFeXSyB
>
> Max-Forwards: 70
>
> From: <sip:1000 at xx.yy.zz.1>;tag=HFr8848ZK8yjQ
>
> To: "My Name" <sip:1016 at xx.yy.zz.1>;tag=3799
>
> Call-ID: a830cf50124857e3ba148763920bf674 at xx.yy.zz.66
>
> CSeq: 10324328 BYE
>
> Content-Length: 0
>
>
>
> ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
>
>
>
> freeswitch at internal> sofia status profile xx.yy.zz.1
> =================================================================================================
> Name??????????????????? xx.yy.zz.1
> Domain Name???????????? N/A
> Alias Of??????????????? internal
> DBName????????????????? sofia_reg_internal
> Pres Hosts
> Dialplan??????????????? XML
> Context???????????????? public
> Challenge Realm???????? auto_from
> RTP-IP????????????????? xx.yy.zz.1
> Ext-RTP-IP????????????? xx.yy.zz.1
> SIP-IP????????????????? xx.yy.zz.1
> Ext-SIP-IP????????????? xx.yy.zz.1
> URL???????????????????? sip:mod_sofia at xx.yy.zz.1:5060
> BIND-URL??????????????? sip:mod_sofia at xx.yy.zz.1:5060
> HOLD-MUSIC????????????? local_stream://moh
> OUTBOUND-PROXY????????? N/A
> CODECS
> G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM,speex at 16000h@20i
> TEL-EVENT?????????????? 101
> DTMF-MODE?????????????? rfc2833
> CNG???????????????????? 13
> SESSION-TO????????????? 0
> MAX-DIALOG????????????? 0
> NOMEDIA???????????????? false
> LATE-NEG??????????????? false
> PROXY-MEDIA???????????? false
> AGGRESSIVENAT?????????? false
> STUN-ENABLED??????????? true
> STUN-AUTO-DISABLE?????? false
> CALLS-IN??????????????? 483
> FAILED-CALLS-IN???????? 34
> CALLS-OUT?????????????? 1098
> FAILED-CALLS-OUT??????? 865
>
> Registrations:
> =================================================================================================
> =================================================================================================
>
> freeswitch at internal> nta: timer J fired, terminate 401 response
> incoming_reclaim_all((nil), (nil), 0x4123cea0)
> nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/17 free
> nta: timer set next to 2192 ms
> nta: timer J fired, terminate 403 response
> incoming_reclaim_all((nil), (nil), 0x4123cea0)
> nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/16 free
> nta: timer not set
> tport_wakeup_pri(0x7028b0): events IN
> tport_recv_event(0x7028b0)
> tport_recv_iovec(0x7028b0) msg 0x11662d0 from (udp/xx.yy.zz.1:5060) has 364
> bytes, veclen = 1
> tport_deliver(0x7028b0): msg 0x11662d0 (364 bytes) from
> udp/xx.yy.zz.66:5060/sip next=(nil)
> nta: received REGISTER sip:xx.yy.zz.1:5060 SIP/2.0 (CSeq 1)
> nta: canonizing sip:xx.yy.zz.1:5060 with contact
> nta: REGISTER (1) going to a default leg
> nua: nua_stack_process_request: entering
> nua: nh_create: entering
> nua: nh_create_handle: entering
> nua: nua_stack_set_params: entering
> soa_clone(static::0x6efae0, 0x7154c0, 0x71df40) called
> soa_set_params(static::0x1148020, ...) called
> nua(0x71df40): event i_register 100 Trying
> nua: nua_application_event: entering
> nua: nua_respond: entering
> nua(0x71df40): sent signal r_respond
> nua: nua_handle_destroy: entering
> nua(0x71df40): sent signal r_destroy
> nua: nua_handle_magic: entering
> nua: nua_handle_destroy: entering
> nua(0x71df40): recv signal r_respond 401 Unauthorized
> nua: nua_stack_set_params: entering
> soa_set_params(static::0x1148020, ...) called
> tport_tsend(0x7028b0) tpn = UDP/xx.yy.zz.66:4829
> tport_resolve addrinfo = xx.yy.zz.66:4829
> tport_by_addrinfo(0x7028b0): not found by name UDP/xx.yy.zz.66:4829
> tport_vsend(0x7028b0): 628 bytes of 628 to udp/xx.yy.zz.66:4829
> tport_vsend returned 628
> nta: sent 401 Unauthorized for REGISTER (1)
> nta: timer set to 32000 ms
> nua(0x71df40): recv signal r_destroy
> nta_leg_destroy((nil))
> soa_destroy(static::0x1148020) called
>
> freeswitch at internal>
> freeswitch at internal>
> freeswitch at internal>
> freeswitch at internal> tport_wakeup_pri(0x7028b0): events IN
> tport_recv_event(0x7028b0)
> tport_recv_iovec(0x7028b0) msg 0x768b30 from (udp/xx.yy.zz.1:5060) has 568
> bytes, veclen = 1
> tport_deliver(0x7028b0): msg 0x768b30 (568 bytes) from
> udp/xx.yy.zz.66:5060/sip next=(nil)
> nta: received REGISTER sip:xx.yy.zz.1:5060 SIP/2.0 (CSeq 2)
> nta: canonizing sip:xx.yy.zz.1:5060 with contact
> nta: REGISTER (2) going to a default leg
> nua: nua_stack_process_request: entering
> nua: nh_create: entering
> nua: nh_create_handle: entering
> nua: nua_stack_set_params: entering
> soa_clone(static::0x6efae0, 0x7154c0, 0x1118070) called
> soa_set_params(static::0x1148020, ...) called
> nua(0x1118070): event i_register 100 Trying
> nua: nua_application_event: entering
> nua: nua_respond: entering
> nua(0x1118070): sent signal r_respond
> nua(0x1118070): recv signal r_respond 200 OK
> nua: nua_stack_set_params: entering
> soa_set_params(static::0x1148020, ...) called
> tport_tsend(0x7028b0) tpn = UDP/xx.yy.zz.66:4829
> tport_resolve addrinfo = xx.yy.zz.66:4829
> tport_by_addrinfo(0x7028b0): not found by name UDP/xx.yy.zz.66:4829
> tport_vsend(0x7028b0): 551 bytes of 551 to udp/xx.yy.zz.66:4829
> tport_vsend returned 551
> nta: sent 200 OK for REGISTER (2)
> nua: nua_handle_destroy: entering
> nua(0x1118070): sent signal r_destroy
> nua: nua_handle_magic: entering
> nua: nua_handle_destroy: entering
> nua(0x1118070): recv signal r_destroy
> nta_leg_destroy((nil))
> soa_destroy(static::0x1148020) called
>
> freeswitch at internal>
> freeswitch at internal>
> freeswitch at internal>
> freeswitch at internal>
> freeswitch at internal> tport_wakeup_pri(0x7028b0): events IN
> tport_recv_event(0x7028b0)
> tport_recv_iovec(0x7028b0) msg 0x1108a50 from (udp/xx.yy.zz.1:5060) has 744
> bytes, veclen = 1
> tport_deliver(0x7028b0): msg 0x1108a50 (744 bytes) from
> udp/xx.yy.zz.66:5060/sip next=(nil)
> nta: received INVITE sip:1000 at xx.yy.zz.1 SIP/2.0 (CSeq 1)
> nta: canonizing sip:1000 at xx.yy.zz.1 with contact
> nta: INVITE (1) going to a default leg
> nta: timer shortened to 200 ms
> nua: nua_stack_process_request: entering
> nua: nh_create: entering
> nua: nh_create_handle: entering
> nua: nua_stack_set_params: entering
> soa_clone(static::0x6efae0, 0x7154c0, 0x11170f0) called
> soa_set_params(static::0x1148020, ...) called
> nta_leg_tcreate(0x110a160)
> soa_init_offer_answer(static::0x1148020) called
> soa_set_remote_sdp(static::0x1148020, (nil), 0x117ad96, 306) called
> nua(0x11170f0): adding session usage
> tport_tsend(0x7028b0) tpn = UDP/xx.yy.zz.66:4829
> tport_resolve addrinfo = xx.yy.zz.66:4829
> tport_by_addrinfo(0x7028b0): not found by name UDP/xx.yy.zz.66:4829
> tport_vsend(0x7028b0): 328 bytes of 328 to udp/xx.yy.zz.66:4829
> tport_vsend returned 328
> nta: sent 100 Trying for INVITE (1)
> nua(0x11170f0): event i_invite 100 Trying
> nua(0x11170f0): call state changed: init -> received, received offer
> soa_get_remote_sdp(static::0x1148020, [0x4123c558], [0x4123c550], [(nil)])
> called
> nua(0x11170f0): event i_state 100 Trying
> nua: nua_application_event: entering
> 2011-03-28 15:57:20.161392 [DEBUG] sofia.c:4671 IP xx.yy.zz.66 Approved by
> acl "domains[]". Access Granted.
> 2011-03-28 15:57:20.161392 [NOTICE] switch_channel.c:613 New Channel
> sofia/internal/1016 at xx.yy.zz.1 [4c7785ba-5943-11e0-8c96-03c3bfaa409b]
> nua: nua_handle_bind: entering
> 2011-03-28 15:57:20.161392 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/1016 at xx.yy.zz.1) Running State Change CS_NEW
> 2011-03-28 15:57:20.161392 [DEBUG] switch_core_state_machine.c:404
> (sofia/internal/1016 at xx.yy.zz.1) State NEW
> 2011-03-28 15:57:20.161392 [DEBUG] sofia.c:5408 Setting NAT mode based on
> nat.auto
> nua: nua_handle_magic: entering
> nua: nua_application_event: entering
> 2011-03-28 15:57:20.161392 [DEBUG] sofia.c:3334 Channel
> sofia/internal/1016 at xx.yy.zz.1 entering state [received][100]
> 2011-03-28 15:57:20.161392 [DEBUG] sofia.c:3341 Remote SDP:
> v=0
> o=- 652619 653988 IN IP4 xx.yy.zz.66
> s=JavaSoftPhone
> c=IN IP4 xx.yy.zz.66
> t=0 0
> m=audio 1370 RTP/AVP 0 3 4 5 101
> c=IN IP4 xx.yy.zz.66
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:5 DVI4/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=setup:active
>
> 2011-03-28 15:57:20.161392 [DEBUG] sofia_glue.c:3136 Audio Codec Compare
> [PCMU:0:8000:0]/[G7221:115:32000:20]
> 2011-03-28 15:57:20.161392 [DEBUG] sofia_glue.c:3136 Audio Codec Compare
> [PCMU:0:8000:0]/[G7221:107:16000:20]
> 2011-03-28 15:57:20.161392 [DEBUG] sofia_glue.c:3136 Audio Codec Compare
> [PCMU:0:8000:0]/[G722:9:8000:20]
> 2011-03-28 15:57:20.161392 [DEBUG] sofia_glue.c:3136 Audio Codec Compare
> [PCMU:0:8000:0]/[PCMU:0:8000:20]
> 2011-03-28 15:57:20.161392 [DEBUG] sofia_glue.c:2094 Set Codec
> sofia/internal/1016 at xx.yy.zz.1 PCMU/8000 20 ms 160 samples
> 2011-03-28 15:57:20.161392 [DEBUG] sofia_glue.c:3096 Set 2833 dtmf payload
> to 101
> 2011-03-28 15:57:20.161392 [DEBUG] sofia.c:3502
> (sofia/internal/1016 at xx.yy.zz.1) State Change CS_NEW -> CS_INIT
> 2011-03-28 15:57:20.161392 [DEBUG] switch_core_session.c:932 Send signal
> sofia/internal/1016 at xx.yy.zz.1 [BREAK]
> nua: nua_handle_magic: entering
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/1016 at xx.yy.zz.1) Running State Change CS_INIT
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:422
> (sofia/internal/1016 at xx.yy.zz.1) State INIT
> 2011-03-28 15:57:20.181410 [DEBUG] mod_sofia.c:83
> sofia/internal/1016 at xx.yy.zz.1 SOFIA INIT
> 2011-03-28 15:57:20.181410 [DEBUG] mod_sofia.c:111
> (sofia/internal/1016 at xx.yy.zz.1) State Change CS_INIT -> CS_ROUTING
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_session.c:932 Send signal
> sofia/internal/1016 at xx.yy.zz.1 [BREAK]
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:422
> (sofia/internal/1016 at xx.yy.zz.1) State INIT going to sleep
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/1016 at xx.yy.zz.1) Running State Change CS_ROUTING
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:425
> (sofia/internal/1016 at xx.yy.zz.1) State ROUTING
> 2011-03-28 15:57:20.181410 [DEBUG] mod_sofia.c:130
> sofia/internal/1016 at xx.yy.zz.1 SOFIA ROUTING
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:78
> sofia/internal/1016 at xx.yy.zz.1 Standard ROUTING
> 2011-03-28 15:57:20.181410 [INFO] mod_dialplan_xml.c:391 Processing My
> Name->1000 in context public
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [public->unloop]
> continue=false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (PASS) [unloop]
> ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [unloop]
> ${sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [public->outside_call]
> continue=true
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Absolute Condition [outside_call]
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action set(outside_call=true)
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [public->call_debug]
> continue=true
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [call_debug]
> ${call_debug}(false) =~ /^true$/ break=never
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [public->public_extensions]
> continue=false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (PASS) [public_extensions]
> destination_number(1000) =~ /^(10[01][0-9])$/ break=on-false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action transfer(1000 XML default)
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:114
> (sofia/internal/1016 at xx.yy.zz.1) State Change CS_ROUTING -> CS_EXECUTE
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_session.c:932 Send signal
> sofia/internal/1016 at xx.yy.zz.1 [BREAK]
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:425
> (sofia/internal/1016 at xx.yy.zz.1) State ROUTING going to sleep
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/1016 at xx.yy.zz.1) Running State Change CS_EXECUTE
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:432
> (sofia/internal/1016 at xx.yy.zz.1) State EXECUTE
> 2011-03-28 15:57:20.181410 [DEBUG] mod_sofia.c:173
> sofia/internal/1016 at xx.yy.zz.1 SOFIA EXECUTE
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:151
> sofia/internal/1016 at xx.yy.zz.1 Standard EXECUTE
> EXECUTE sofia/internal/1016 at xx.yy.zz.1 set(outside_call=true)
> 2011-03-28 15:57:20.181410 [DEBUG] mod_dptools.c:748
> sofia/internal/1016 at xx.yy.zz.1 SET [outside_call]=[true]
> EXECUTE sofia/internal/1016 at xx.yy.zz.1 transfer(1000 XML default)
> 2011-03-28 15:57:20.181410 [DEBUG] switch_ivr.c:1344
> (sofia/internal/1016 at xx.yy.zz.1) State Change CS_EXECUTE -> CS_ROUTING
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_session.c:932 Send signal
> sofia/internal/1016 at xx.yy.zz.1 [BREAK]
> 2011-03-28 15:57:20.181410 [DEBUG] switch_ivr.c:1348
> sofia/internal/1016 at xx.yy.zz.1 receive message [TRANSFER]
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_session.c:630 Send signal
> sofia/internal/1016 at xx.yy.zz.1 [BREAK]
> 2011-03-28 15:57:20.181410 [NOTICE] switch_ivr.c:1350 Transfer
> sofia/internal/1016 at xx.yy.zz.1 to XML[1000 at default]
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:432
> (sofia/internal/1016 at xx.yy.zz.1) State EXECUTE going to sleep
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/1016 at xx.yy.zz.1) Running State Change CS_ROUTING
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:425
> (sofia/internal/1016 at xx.yy.zz.1) State ROUTING
> 2011-03-28 15:57:20.181410 [DEBUG] mod_sofia.c:130
> sofia/internal/1016 at xx.yy.zz.1 SOFIA ROUTING
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:78
> sofia/internal/1016 at xx.yy.zz.1 Standard ROUTING
> 2011-03-28 15:57:20.181410 [INFO] mod_dialplan_xml.c:391 Processing My
> Name->1000 in context default
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->unloop]
> continue=false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (PASS) [unloop]
> ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [unloop]
> ${sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->tod_example]
> continue=true
> Dialplan: day of week[2] =~ 2-6 (PASS)
> Dialplan: hour[15] =~ 9-18 (PASS)
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Date/Time Match (PASS)
> [tod_example] break=on-false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action set(open=true)
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->global-intercept]
> continue=false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [global-intercept]
> destination_number(1000) =~ /^886$/ break=on-false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->group-intercept]
> continue=false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [group-intercept]
> destination_number(1000) =~ /^\*8$/ break=on-false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->intercept-ext]
> continue=false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [intercept-ext]
> destination_number(1000) =~ /^\*\*(\d+)$/ break=on-false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->redial]
> continue=false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [redial]
> destination_number(1000) =~ /^870$/ break=on-false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->global]
> continue=true
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [global]
> ${call_debug}(false) =~ /^true$/ break=never
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [global]
> ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/
> break=never
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Absolute Condition [global]
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action
> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action
> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action
> hash(insert/${domain_name}-last_dial/global/${uuid})
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->snom-demo-2]
> continue=false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [snom-demo-2]
> destination_number(1000) =~ /^9001$/ break=on-false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->snom-demo-1]
> continue=false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [snom-demo-1]
> destination_number(1000) =~ /^9000$/ break=on-false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->eavesdrop]
> continue=false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [eavesdrop]
> destination_number(1000) =~ /^88(.*)$|^\*0(.*)$/ break=on-false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->eavesdrop]
> continue=false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [eavesdrop]
> destination_number(1000) =~ /^779$/ break=on-false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->call_return]
> continue=false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [call_return]
> destination_number(1000) =~ /^\*69$|^869$|^lcr$/ break=on-false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->del-group]
> continue=false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [del-group]
> destination_number(1000) =~ /^80(\d{2})$/ break=on-false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->add-group]
> continue=false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [add-group]
> destination_number(1000) =~ /^81(\d{2})$/ break=on-false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->call-group-simo]
> continue=false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [call-group-simo]
> destination_number(1000) =~ /^82(\d{2})$/ break=on-false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->call-group-order]
> continue=false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [call-group-order]
> destination_number(1000) =~ /^83(\d{2})$/ break=on-false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing
> [default->extension-intercom] continue=false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (FAIL) [extension-intercom]
> destination_number(1000) =~ /^8(10[01][0-9])$/ break=on-false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 parsing [default->Local_Extension]
> continue=false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Regex (PASS) [Local_Extension]
> destination_number(1000) =~ /^(10[01][0-9])$/ break=on-false
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action set(dialed_extension=1000)
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action
> export(dialed_extension=1000)
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action bind_meta_app(1 b s
> execute_extension::dx XML features)
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action bind_meta_app(2 b s
> record_session::/opt/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav)
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action bind_meta_app(3 b s
> execute_extension::cf XML features)
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action set(ringback=${us-ring})
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action
> set(transfer_ringback=local_stream://moh)
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action set(call_timeout=30)
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action
> set(hangup_after_bridge=true)
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action set(continue_on_fail=true)
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action
> hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number})
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action
> hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action
> set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
> var callgroup)})
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action
> hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action
> bridge(user/${dialed_extension}@${domain_name})
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action answer()
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action sleep(1000)
> Dialplan: sofia/internal/1016 at xx.yy.zz.1 Action voicemail(default
> ${domain_name} ${dialed_extension})
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:114
> (sofia/internal/1016 at xx.yy.zz.1) State Change CS_ROUTING -> CS_EXECUTE
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_session.c:932 Send signal
> sofia/internal/1016 at xx.yy.zz.1 [BREAK]
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:425
> (sofia/internal/1016 at xx.yy.zz.1) State ROUTING going to sleep
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/1016 at xx.yy.zz.1) Running State Change CS_EXECUTE
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:432
> (sofia/internal/1016 at xx.yy.zz.1) State EXECUTE
> 2011-03-28 15:57:20.181410 [DEBUG] mod_sofia.c:173
> sofia/internal/1016 at xx.yy.zz.1 SOFIA EXECUTE
> 2011-03-28 15:57:20.181410 [DEBUG] switch_core_state_machine.c:151
> sofia/internal/1016 at xx.yy.zz.1 Standard EXECUTE
> EXECUTE sofia/internal/1016 at xx.yy.zz.1 set(open=true)
> 2011-03-28 15:57:20.181410 [DEBUG] mod_dptools.c:748
> sofia/internal/1016 at xx.yy.zz.1 SET [open]=[true]
> EXECUTE sofia/internal/1016 at xx.yy.zz.1
> hash(insert/xx.yy.zz.1-spymap/1016/4c7785ba-5943-11e0-8c96-03c3bfaa409b)
> EXECUTE sofia/internal/1016 at xx.yy.zz.1
> hash(insert/xx.yy.zz.1-last_dial/1016/1000)
> EXECUTE sofia/internal/1016 at xx.yy.zz.1
> hash(insert/xx.yy.zz.1-last_dial/global/4c7785ba-5943-11e0-8c96-03c3bfaa409b)
> EXECUTE sofia/internal/1016 at xx.yy.zz.1 set(dialed_extension=1000)
> 2011-03-28 15:57:20.181410 [DEBUG] mod_dptools.c:748
> sofia/internal/1016 at xx.yy.zz.1 SET [dialed_extension]=[1000]
> EXECUTE sofia/internal/1016 at xx.yy.zz.1 export(dialed_extension=1000)
> 2011-03-28 15:57:20.181410 [DEBUG] mod_dptools.c:886 EXPORT
> [dialed_extension]=[1000]
> EXECUTE sofia/internal/1016 at xx.yy.zz.1 bind_meta_app(1 b s
> execute_extension::dx XML features)
> 2011-03-28 15:57:20.181410 [INFO] switch_ivr_async.c:1835 Bound B-Leg: 1
> execute_extension::dx XML features
> EXECUTE sofia/internal/1016 at xx.yy.zz.1 bind_meta_app(2 b s
> record_session::/opt/freeswitch/recordings/1016.2011-03-28-15-57-20.wav)
> 2011-03-28 15:57:20.181410 [INFO] switch_ivr_async.c:1835 Bound B-Leg: 2
> record_session::/opt/freeswitch/recordings/1016.2011-03-28-15-57-20.wav
> EXECUTE sofia/internal/1016 at xx.yy.zz.1 bind_meta_app(3 b s
> execute_extension::cf XML features)
> 2011-03-28 15:57:20.181410 [INFO] switch_ivr_async.c:1835 Bound B-Leg: 3
> execute_extension::cf XML features
> EXECUTE sofia/internal/1016 at xx.yy.zz.1
> set(ringback=%(2000,4000,440.0,480.0))
> 2011-03-28 15:57:20.181410 [DEBUG] mod_dptools.c:748
> sofia/internal/1016 at xx.yy.zz.1 SET [ringback]=[%(2000,4000,440.0,480.0)]
> EXECUTE sofia/internal/1016 at xx.yy.zz.1
> set(transfer_ringback=local_stream://moh)
> 2011-03-28 15:57:20.181410 [DEBUG] mod_dptools.c:748
> sofia/internal/1016 at xx.yy.zz.1 SET [transfer_ringback]=[local_stream://moh]
> EXECUTE sofia/internal/1016 at xx.yy.zz.1 set(call_timeout=30)
> 2011-03-28 15:57:20.191371 [DEBUG] mod_dptools.c:748
> sofia/internal/1016 at xx.yy.zz.1 SET [call_timeout]=[30]
> EXECUTE sofia/internal/1016 at xx.yy.zz.1 set(hangup_after_bridge=true)
> 2011-03-28 15:57:20.191371 [DEBUG] mod_dptools.c:748
> sofia/internal/1016 at xx.yy.zz.1 SET [hangup_after_bridge]=[true]
> EXECUTE sofia/internal/1016 at xx.yy.zz.1 set(continue_on_fail=true)
> 2011-03-28 15:57:20.191371 [DEBUG] mod_dptools.c:748
> sofia/internal/1016 at xx.yy.zz.1 SET [continue_on_fail]=[true]
> EXECUTE sofia/internal/1016 at xx.yy.zz.1
> hash(insert/xx.yy.zz.1-call_return/1000/1016)
> EXECUTE sofia/internal/1016 at xx.yy.zz.1
> hash(insert/xx.yy.zz.1-last_dial_ext/1000/4c7785ba-5943-11e0-8c96-03c3bfaa409b)
> EXECUTE sofia/internal/1016 at xx.yy.zz.1
> set(called_party_callgroup=techsupport)
> 2011-03-28 15:57:20.191371 [DEBUG] mod_dptools.c:748
> sofia/internal/1016 at xx.yy.zz.1 SET [called_party_callgroup]=[techsupport]
> EXECUTE sofia/internal/1016 at xx.yy.zz.1
> hash(insert/xx.yy.zz.1-last_dial/techsupport/4c7785ba-5943-11e0-8c96-03c3bfaa409b)
> EXECUTE sofia/internal/1016 at xx.yy.zz.1 bridge(user/1000 at xx.yy.zz.1)
> 2011-03-28 15:57:20.191371 [DEBUG] switch_ivr_originate.c:1047 variable
> string 0 = [presence_id=1000 at xx.yy.zz.1]
> 2011-03-28 15:57:20.191371 [ERR] switch_ivr_originate.c:1531 Cannot create
> outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> 2011-03-28 15:57:20.191371 [DEBUG] switch_ivr_originate.c:2170 Originate
> Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
> 2011-03-28 15:57:20.191371 [ERR] switch_ivr_originate.c:1531 Cannot create
> outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> 2011-03-28 15:57:20.191371 [DEBUG] switch_ivr_originate.c:2170 Originate
> Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
> 2011-03-28 15:57:20.191371 [INFO] mod_dptools.c:2114 Originate Failed.
> Cause: USER_NOT_REGISTERED
> 2011-03-28 15:57:20.191371 [DEBUG] mod_dptools.c:2141 Continue on fail
> [true]:? Cause: USER_NOT_REGISTERED
> EXECUTE sofia/internal/1016 at xx.yy.zz.1 answer()
> 2011-03-28 15:57:20.191371 [DEBUG] mod_dptools.c:649
> sofia/internal/1016 at xx.yy.zz.1 receive message [ANSWER]
> 2011-03-28 15:57:20.191371 [DEBUG] sofia_glue.c:2328 AUDIO RTP
> [sofia/internal/1016 at xx.yy.zz.1] xx.yy.zz.1 port 17768 -> xx.yy.zz.66 port
> 1370 codec: 0 ms: 20
> 2011-03-28 15:57:20.191371 [DEBUG] switch_rtp.c:1155 Starting timer [soft]
> 160 bytes per 20ms
> 2011-03-28 15:57:20.201410 [DEBUG] mod_sofia.c:536 Local SDP
> sofia/internal/1016 at xx.yy.zz.1:
> v=0
> o=FreeSWITCH 1301302872 1301302873 IN IP4 xx.yy.zz.1
> s=FreeSWITCH
> c=IN IP4 xx.yy.zz.1
> t=0 0
> m=audio 17768 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> nua: nua_respond: entering
> nua(0x11170f0): sent signal r_respond
> 2011-03-28 15:57:20.201410 [DEBUG] switch_core_session.c:630 Send signal
> sofia/internal/1016 at xx.yy.zz.1 [BREAK]
> 2011-03-28 15:57:20.201410 [NOTICE] mod_dptools.c:649 Channel
> [sofia/internal/1016 at xx.yy.zz.1] has been answered
> 2011-03-28 15:57:20.201410 [DEBUG] switch_channel.c:182
> sofia/internal/1016 at xx.yy.zz.1 receive message [AUDIO_SYNC]
> EXECUTE sofia/internal/1016 at xx.yy.zz.1 sleep(1000)
> nua(0x11170f0): recv signal r_respond 200 OK
> nua: nua_stack_set_params: entering
> soa_set_params(static::0x1148020, ...) called
> soa_set_user_sdp(static::0x1148020, (nil), 0x7fb13c02316d, -1) called
> soa_set_capability_sdp(static::0x1148020, (nil), 0x7fb13c02316d, -1) called
> nua: nua_invite_server_respond: entering
> soa_generate_answer(static::0x1148020) called
> soa_static_offer_answer_action(0x1148020, soa_generate_answer): called
> soa_static(0x1148020, soa_generate_answer): generating local description
> soa_static(0x1148020, soa_generate_answer): upgrade with remote description
> soa_sdp_mode_set(0x4123caa0, 0x116ef80, ""): called
> soa_static(0x1148020, soa_generate_answer): storing local description
> soa_activate(static::0x1148020, (nil)) called
> soa_get_local_sdp(static::0x1148020, [(nil)], [0x4123cc90], [0x4123cc9c])
> called
> tport_tsend(0x7028b0) tpn = UDP/xx.yy.zz.66:4829
> tport_resolve addrinfo = xx.yy.zz.66:4829
> tport_by_addrinfo(0x7028b0): not found by name UDP/xx.yy.zz.66:4829
> tport_vsend(0x7028b0): 1064 bytes of 1064 to udp/xx.yy.zz.66:4829
> tport_vsend returned 1064
> nta: sent 200 OK for INVITE (1)
> nua(0x11170f0): call state changed: received -> completed, sent answer
> soa_get_local_sdp(static::0x1148020, [0x4123cde8], [0x4123cde0], [(nil)])
> called
> soa_get_params(static::0x1148020, ...) called
> nua(0x11170f0): event i_state 200 OK
> 2011-03-28 15:57:20.201410 [DEBUG] switch_channel.c:182
> sofia/internal/1016 at xx.yy.zz.1 receive message [AUDIO_SYNC]
> nua: nua_application_event: entering
> 2011-03-28 15:57:20.201410 [DEBUG] sofia.c:3334 Channel
> sofia/internal/1016 at xx.yy.zz.1 entering state [completed][200]
> nua: nua_handle_magic: entering
> tport_wakeup_pri(0x7028b0): events IN
> tport_recv_event(0x7028b0)
> tport_recv_iovec(0x7028b0) msg 0x1125e50 from (udp/xx.yy.zz.1:5060) has 327
> bytes, veclen = 1
> tport_deliver(0x7028b0): msg 0x1125e50 (327 bytes) from
> udp/xx.yy.zz.66:5060/sip next=(nil)
> nta: received ACK sip:1000 at xx.yy.zz.1:5060;transport=udp SIP/2.0 (CSeq 1)
> nta: ACK (1) is going to INVITE (1)
> nua: process_ack_or_cancel: entering
> soa_clear_remote_sdp(static::0x1148020) called
> nua(0x11170f0): event i_ack 200 OK
> nua(0x11170f0): call state changed: completed -> ready
> nua(0x11170f0): event i_state 200 OK
> nua(0x11170f0): event i_active 200 Call active
> nua(): refresh session after 88 seconds (in [88..88])
> nua: nua_application_event: entering
> nua: nua_handle_magic: entering
> nua: nua_application_event: entering
> 2011-03-28 15:57:20.231367 [DEBUG] sofia.c:3334 Channel
> sofia/internal/1016 at xx.yy.zz.1 entering state [ready][200]
> nua: nua_handle_magic: entering
> nua: nua_application_event: entering
> nua: nua_handle_magic: entering
> nta: timer set next to 4862 ms
> EXECUTE sofia/internal/1016 at xx.yy.zz.1 voicemail(default xx.yy.zz.1 1000)
> 2011-03-28 15:57:21.201425 [DEBUG] mod_voicemail.c:788 [default] rwlock
> 2011-03-28 15:57:21.201425 [ERR] mod_voicemail.c:2927 Error creating
> /opt/freeswitch/storage/voicemail/default/xx.yy.zz.1/1000
> 2011-03-28 15:57:21.201425 [DEBUG] switch_ivr_play_say.c:118 No language
> specified - Using [en]
> 2011-03-28 15:57:21.201425 [ERR] switch_ivr_play_say.c:148 Can't find
> language tag.
> 2011-03-28 15:57:21.201425 [WARNING] switch_ivr_play_say.c:364 Macro
> [voicemail_goodbye] did not match any patterns
> 2011-03-28 15:57:21.201425 [NOTICE] switch_core_state_machine.c:179 Hangup
> sofia/internal/1016 at xx.yy.zz.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-28 15:57:21.201425 [DEBUG] switch_channel.c:1726 Send signal
> sofia/internal/1016 at xx.yy.zz.1 [KILL]
> 2011-03-28 15:57:21.201425 [DEBUG] switch_core_session.c:932 Send signal
> sofia/internal/1016 at xx.yy.zz.1 [BREAK]
> 2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:560
> (sofia/internal/1016 at xx.yy.zz.1) State HANGUP
> 2011-03-28 15:57:21.201425 [DEBUG] mod_sofia.c:338 Channel
> sofia/internal/1016 at xx.yy.zz.1 hanging up, cause: NORMAL_CLEARING
> 2011-03-28 15:57:21.201425 [DEBUG] mod_sofia.c:376 Sending BYE to
> sofia/internal/1016 at xx.yy.zz.1
> nua: nua_bye: entering
> nua(0x11170f0): sent signal r_bye
> nua(0x11170f0): recv signal r_bye
> nua: nua_stack_set_params: entering
> soa_set_params(static::0x1148020, ...) called
> soa_terminate(static::0x1148020) called
> soa_init_offer_answer(static::0x1148020) called
> nta: selecting scheme sip
> tport_tsend(0x7028b0) tpn = udp/xx.yy.zz.66:4829
> tport_resolve addrinfo = xx.yy.zz.66:4829
> tport_by_addrinfo(0x7028b0): not found by name udp/xx.yy.zz.66:4829
> tport_vsend(0x7028b0): 604 bytes of 604 to udp/xx.yy.zz.66:4829
> tport_vsend returned 604
> nta: sent BYE (10324256) to udp/xx.yy.zz.66:4829
> tport_pend(0x7028b0): pending 0x7fb1440d6100 for udp/xx.yy.zz.1:5060
> (already 0)
> nta: timer shortened to 500 ms
> 2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/1016 at xx.yy.zz.1 Standard HANGUP, cause: NORMAL_CLEARING
> 2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:560
> (sofia/internal/1016 at xx.yy.zz.1) State HANGUP going to sleep
> 2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:432
> (sofia/internal/1016 at xx.yy.zz.1) State EXECUTE going to sleep
> 2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/1016 at xx.yy.zz.1) Running State Change CS_HANGUP
> 2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:538 handler
> already called, skipping state handler.
> 2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:417
> (sofia/internal/1016 at xx.yy.zz.1) State Change CS_HANGUP -> CS_REPORTING
> 2011-03-28 15:57:21.201425 [DEBUG] switch_core_session.c:932 Send signal
> sofia/internal/1016 at xx.yy.zz.1 [BREAK]
> 2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/1016 at xx.yy.zz.1) Running State Change CS_REPORTING
> 2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:651
> (sofia/internal/1016 at xx.yy.zz.1) State REPORTING
> 2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/1016 at xx.yy.zz.1 Standard REPORTING, cause: NORMAL_CLEARING
> 2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:651
> (sofia/internal/1016 at xx.yy.zz.1) State REPORTING going to sleep
> 2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:411
> (sofia/internal/1016 at xx.yy.zz.1) State Change CS_REPORTING -> CS_DESTROY
> 2011-03-28 15:57:21.201425 [DEBUG] switch_core_session.c:932 Send signal
> sofia/internal/1016 at xx.yy.zz.1 [BREAK]
> 2011-03-28 15:57:21.201425 [DEBUG] switch_core_session.c:1069 Session 1582
> (sofia/internal/1016 at xx.yy.zz.1) Locked, Waiting on external entities
> 2011-03-28 15:57:21.201425 [NOTICE] switch_core_session.c:1087 Session 1582
> (sofia/internal/1016 at xx.yy.zz.1) Ended
> 2011-03-28 15:57:21.201425 [NOTICE] switch_core_session.c:1089 Close Channel
> sofia/internal/1016 at xx.yy.zz.1 [CS_DESTROY]
> 2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:497
> (sofia/internal/1016 at xx.yy.zz.1) Running State Change CS_DESTROY
> 2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:508
> (sofia/internal/1016 at xx.yy.zz.1) State DESTROY
> 2011-03-28 15:57:21.201425 [DEBUG] mod_sofia.c:255
> sofia/internal/1016 at xx.yy.zz.1 SOFIA DESTROY
> 2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/1016 at xx.yy.zz.1 Standard DESTROY
> 2011-03-28 15:57:21.201425 [DEBUG] switch_core_state_machine.c:508
> (sofia/internal/1016 at xx.yy.zz.1) State DESTROY going to sleep
> nta: timer E fired, retransmit BYE (10324256)
> tport_release(0x7028b0): 0x7fb1440d6100 by 0x7fb144068530 with (nil)
> tport_tsend(0x7028b0) tpn = udp/xx.yy.zz.66:4829
> tport_resolve addrinfo = xx.yy.zz.66:4829
> tport_by_addrinfo(0x7028b0): not found by name udp/xx.yy.zz.66:4829
> tport_vsend(0x7028b0): 604 bytes of 604 to udp/xx.yy.zz.66:4829
> tport_vsend returned 604
> nta: resent BYE (10324256) to udp/xx.yy.zz.66:4829
> tport_pend(0x7028b0): pending 0x7fb1440d6100 for udp/xx.yy.zz.1:5060
> (already 0)
> nta_outgoing_timer: 1/1 resent, 0/1 tout, 0/0 term, 0/1 free
> nta: timer set next to 1000 ms
> nta: timer E fired, retransmit BYE (10324256)
> tport_release(0x7028b0): 0x7fb1440d6100 by 0x7fb144068530 with (nil)
> tport_tsend(0x7028b0) tpn = udp/xx.yy.zz.66:4829
> tport_resolve addrinfo = xx.yy.zz.66:4829
> tport_by_addrinfo(0x7028b0): not found by name udp/xx.yy.zz.66:4829
> tport_vsend(0x7028b0): 604 bytes of 604 to udp/xx.yy.zz.66:4829
> tport_vsend returned 604
> nta: resent BYE (10324256) to udp/xx.yy.zz.66:4829
> tport_pend(0x7028b0): pending 0x7fb1440d6100 for udp/xx.yy.zz.1:5060
> (already 0)
> nta_outgoing_timer: 1/1 resent, 0/1 tout, 0/0 term, 0/1 free
> nta: timer set next to 2000 ms
> nta: timer E fired, retransmit BYE (10324256)
> tport_release(0x7028b0): 0x7fb1440d6100 by 0x7fb144068530 with (nil)
> tport_tsend(0x7028b0) tpn = udp/xx.yy.zz.66:4829
> tport_resolve addrinfo = xx.yy.zz.66:4829
> tport_by_addrinfo(0x7028b0): not found by name udp/xx.yy.zz.66:4829
> tport_vsend(0x7028b0): 604 bytes of 604 to udp/xx.yy.zz.66:4829
> tport_vsend returned 604
> nta: resent BYE (10324256) to udp/xx.yy.zz.66:4829
> tport_pend(0x7028b0): pending 0x7fb1440d6100 for udp/xx.yy.zz.1:5060
> (already 0)
> nta_outgoing_timer: 1/1 resent, 0/1 tout, 0/0 term, 0/1 free
> nta: timer set next to 522 ms
> nta: timer I fired, terminate 200 response
> incoming_reclaim_all((nil), (nil), 0x4123cea0)
> nta_incoming_timer: 0/0 resent, 0/0 tout, 1/3 term, 1/18 free
> nta: timer set next to 2738 ms
> nta: timer J fired, terminate 401 response
> incoming_reclaim_all((nil), (nil), 0x4123cea0)
> nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/17 free
> nta: timer set next to 730 ms
> nta: timer E fired, retransmit BYE (10324256)
> tport_release(0x7028b0): 0x7fb1440d6100 by 0x7fb144068530 with (nil)
> tport_tsend(0x7028b0) tpn = udp/xx.yy.zz.66:4829
> tport_resolve addrinfo = xx.yy.zz.66:4829
> tport_by_addrinfo(0x7028b0): not found by name udp/xx.yy.zz.66:4829
> tport_vsend(0x7028b0): 604 bytes of 604 to udp/xx.yy.zz.66:4829
> tport_vsend returned 604
> nta: resent BYE (10324256) to udp/xx.yy.zz.66:4829
> tport_pend(0x7028b0): pending 0x7fb1440d6100 for udp/xx.yy.zz.1:5060
> (already 0)
> nta_outgoing_timer: 1/1 resent, 0/1 tout, 0/0 term, 0/1 free
> nta: timer set next to 4000 ms
> tport_wakeup_pri(0x7028b0): events IN
> tport_recv_event(0x7028b0)
> tport_recv_iovec(0x7028b0) msg 0x117e8c0 from (udp/xx.yy.zz.1:5060) has 283
> bytes, veclen = 1
> tport_deliver(0x7028b0): msg 0x117e8c0 (283 bytes) from
> udp/xx.yy.zz.66:5060/sip next=(nil)
> nta: received 200 OK for BYE (10324256)
> nta: 200 OK is going to a transaction
> nta_outgoing: RTT is 8366.26 ms
> tport_release(0x7028b0): 0x7fb1440d6100 by 0x7fb144068530 with 0x117e8c0
> nua(0x11170f0): event r_bye 200 OK
> nua(0x11170f0): call state changed: terminating -> terminated
> nua(0x11170f0): event i_state 200 to BYE
> nua(0x11170f0): event i_terminated 200 to BYE
> nua(0x11170f0): removing session usage
> soa_destroy(static::0x1148020) called
> nta_leg_destroy(0x110a160)
> nua: terminated session 0x11170f0
> nua: nua_application_event: entering
> nua: nua_handle_magic: entering
> nua: nua_application_event: entering
> nua: nua_handle_magic: entering
> nua: nua_handle_bind: entering
> nua: nua_application_event: entering
> nua: nua_handle_magic: entering
> nua: nua_handle_destroy: entering
> nua(0x11170f0): recv signal r_destroy
> nta_leg_destroy((nil))
> nua(0x11170f0): sent signal r_destroy
> nta: timer set next to 1856 ms
> nta: timer K fired, terminate BYE (10324256)
> outgoing_reclaim_all((nil), (nil), 0x4123ce90)
> nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
> nta: timer set next to 8799 ms
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kerem Erciyes - Sistem Danismani
http://keremerciyes.com


From anthony.minessale at gmail.com  Tue Mar 29 00:17:10 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Mar 2011 15:17:10 -0500
Subject: [Freeswitch-users] mod_fifo & CID
In-Reply-To: <201103281054.52190.yivzhenko@mksat.net>
References: <201103241644.27845.yivzhenko@mksat.net>
	<AANLkTim=ER2syNeDVWM6C6C84pqvshg7R3bNR4M2xEnf@mail.gmail.com>
	<201103281054.52190.yivzhenko@mksat.net>
Message-ID: <AANLkTikiJ1ERs1bec9uC9kAVx_a-TXv7JrYfhHe927ge@mail.gmail.com>

Callee ID name becomes caller id name once they answer and re-enter
the dial plan.
This is a side-effect of making an outbound call then transferring it
inward and is the new default behavior.



On Mon, Mar 28, 2011 at 2:54 AM, Yuriy Ivzhenko <yivzhenko at mksat.net> wrote:
> This change only caller_id_name from "Outbound Call" to 'agent name' and only
> when call Answered.
>
> On Friday 25 March 2011 21:41:23 Anthony Minessale wrote:
>> when you define the agents in the config add this:
>>
>> {origination_callee_id_name='agent
>> name',origination_callee_id_number='agents number'}
>>
>> Its display update when the call changes direction from outbound to bridge.
>>
>> On Thu, Mar 24, 2011 at 9:44 AM, Yuriy Ivzhenko <yivzhenko at mksat.net> wrote:
>> > Hi All,
>> >
>> > After upgrading to last Git (git-73ca862) i see changes in CID on
>> > calls to agents, generated by mod_fifo.
>> >
>> > If agent answer
>> > caller_profile.caller_id_name = "Outbound Call"
>> > caller_profile.caller_id_number = <agent number>
>> >
>> > If agent not answer
>> > caller_profile.caller_id_name = caller caller_id_name
>> > caller_profile.caller_id_number = caller caller_id_number (or
>> > origination_caller_id_name/number of set in agents dialstring)
>> > In old version, so it was always.
>> >
>> > Is this correct?
>> > It seems to me it should not depend on, answered the call or not.
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 all.eforums at gmail.com  Tue Mar 29 00:33:01 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Mon, 28 Mar 2011 16:33:01 -0400
Subject: [Freeswitch-users] Embarrassing Question: Local extension
 registering in the Public context on default configuration?
In-Reply-To: <AANLkTimCcZ33aEgEKi6qmNj8K9Uq9RYBoHr-sPXAByzA@mail.gmail.com>
References: <AANLkTinDZ0xBMxFssO9NrP8y6opAXHUrMOPoNzhS1U2M@mail.gmail.com>
	<AANLkTimCcZ33aEgEKi6qmNj8K9Uq9RYBoHr-sPXAByzA@mail.gmail.com>
Message-ID: <AANLkTikhKTF3TvXAf_vHPWxn6VHkwF2gVZpYcsP401bT@mail.gmail.com>

On Mon, Mar 28, 2011 at 8:28 AM, Randy Andrade <randy.andrade at gmail.com>wrote:

> I'm no FS guru, but I believe that would happen based on the port that your
> softphone is using to register w/ the FS system. By default (at least in my
> experience), the internal / default profile is setup to listen on port
> 5060/5061 (ssl) and the external / public profile is setup to listen on port
> 5080/5081 (ssl). Check to make sure the softphone is set to register on port
> 5060, and it should re-register on the internal / default profile, unless
> there's a local firewall issue on the FS machine preventing it..
>
> Randy
>
>
> mmmm that isn't it. As is shown in the status of this endpoint it IS
actually registered internally on 5060 and I'm not adding any port to the
end of the domain/server name in Bria client.

Anymore ideas? With the gurus on the list, I'd have thought this would be
answered in 10 mins of it being posted :)
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From msc at freeswitch.org  Tue Mar 29 00:41:55 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 28 Mar 2011 13:41:55 -0700
Subject: [Freeswitch-users] Embarrassing Question: Local extension
 registering in the Public context on default configuration?
In-Reply-To: <AANLkTikhKTF3TvXAf_vHPWxn6VHkwF2gVZpYcsP401bT@mail.gmail.com>
References: <AANLkTinDZ0xBMxFssO9NrP8y6opAXHUrMOPoNzhS1U2M@mail.gmail.com>
	<AANLkTimCcZ33aEgEKi6qmNj8K9Uq9RYBoHr-sPXAByzA@mail.gmail.com>
	<AANLkTikhKTF3TvXAf_vHPWxn6VHkwF2gVZpYcsP401bT@mail.gmail.com>
Message-ID: <AANLkTim3dZh0rit8cEicCRm73maVeOaHDy9snBfnPBa-@mail.gmail.com>

> Anymore ideas? With the gurus on the list, I'd have thought this would be
> answered in 10 mins of it being posted :)

:)

Did you nail down the question about whether you have an ACL set?
Also, did you pastebin a debug log of the call hitting the dialplan?

-MC


From all.eforums at gmail.com  Tue Mar 29 01:01:38 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Mon, 28 Mar 2011 17:01:38 -0400
Subject: [Freeswitch-users] Embarrassing Question: Local extension
 registering in the Public context on default configuration?
In-Reply-To: <AANLkTim3dZh0rit8cEicCRm73maVeOaHDy9snBfnPBa-@mail.gmail.com>
References: <AANLkTinDZ0xBMxFssO9NrP8y6opAXHUrMOPoNzhS1U2M@mail.gmail.com>
	<AANLkTimCcZ33aEgEKi6qmNj8K9Uq9RYBoHr-sPXAByzA@mail.gmail.com>
	<AANLkTikhKTF3TvXAf_vHPWxn6VHkwF2gVZpYcsP401bT@mail.gmail.com>
	<AANLkTim3dZh0rit8cEicCRm73maVeOaHDy9snBfnPBa-@mail.gmail.com>
Message-ID: <AANLkTin2YgsH+xVkP9VT4usbpEMxxx-b17To18KO3Ed+@mail.gmail.com>

On Mon, Mar 28, 2011 at 4:41 PM, Michael Collins <msc at freeswitch.org> wrote:

> > Anymore ideas? With the gurus on the list, I'd have thought this would be
> > answered in 10 mins of it being posted :)
>
> :)
>
> Did you nail down the question about whether you have an ACL set?
> Also, did you pastebin a debug log of the call hitting the dialplan?
>
> -MC
>
>
So, I did muck around a tad with the acl.conf.xml file although I was under
the impression that it was auto-magically generated at install time? It
looks like this.

    <list name="lan" default="allow">
      <node type="deny" cidr="0.0.0.0/0"/>
      <node type="allow" cidr="192.168.3.0/24"/>
    </list>

    <list name="domains" default="deny">
     <node type="allow" domain="$${domain}"/>
      <node type="allow" cidr="192.168.3.0/24"/>
    </list>

and yes, I put the pastebin here: http://pastebin.freeswitch.org/15881

I also keep seeing this on the console in fs_cli:

2011-03-26 03:48:54.397538 [WARNING] sofia_reg.c:1246 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [1000 at 192.168.3.101] from ip
192.168.3.121

Not sure why this is a "warning", and that might be an indication that FS
thinks this UA is "foreign"?

HTH?
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From jeff at jefflenk.com  Tue Mar 29 01:45:02 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 28 Mar 2011 14:45:02 -0700 (PDT)
Subject: [Freeswitch-users] Embarrassing Question: Local extension
 registering in the Public context on default configuration?
In-Reply-To: <AANLkTin2YgsH+xVkP9VT4usbpEMxxx-b17To18KO3Ed+@mail.gmail.com>
References: <AANLkTinDZ0xBMxFssO9NrP8y6opAXHUrMOPoNzhS1U2M@mail.gmail.com>
	<AANLkTimCcZ33aEgEKi6qmNj8K9Uq9RYBoHr-sPXAByzA@mail.gmail.com>
	<AANLkTikhKTF3TvXAf_vHPWxn6VHkwF2gVZpYcsP401bT@mail.gmail.com>
	<AANLkTim3dZh0rit8cEicCRm73maVeOaHDy9snBfnPBa-@mail.gmail.com>
	<AANLkTin2YgsH+xVkP9VT4usbpEMxxx-b17To18KO3Ed+@mail.gmail.com>
Message-ID: <1301348702013-6216937.post@n2.nabble.com>

return your acl.conf.xml to default and let the phones register to
authenticate. You could do it with acls to but if you dont understand that
yet try it simpler first:)

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Embarrassing-Question-Local-extension-registering-in-the-Public-context-on-default-configuration-tp6214422p6216937.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From all.eforums at gmail.com  Tue Mar 29 01:50:56 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Mon, 28 Mar 2011 17:50:56 -0400
Subject: [Freeswitch-users] Embarrassing Question: Local extension
 registering in the Public context on default configuration?
In-Reply-To: <1301348702013-6216937.post@n2.nabble.com>
References: <AANLkTinDZ0xBMxFssO9NrP8y6opAXHUrMOPoNzhS1U2M@mail.gmail.com>
	<AANLkTimCcZ33aEgEKi6qmNj8K9Uq9RYBoHr-sPXAByzA@mail.gmail.com>
	<AANLkTikhKTF3TvXAf_vHPWxn6VHkwF2gVZpYcsP401bT@mail.gmail.com>
	<AANLkTim3dZh0rit8cEicCRm73maVeOaHDy9snBfnPBa-@mail.gmail.com>
	<AANLkTin2YgsH+xVkP9VT4usbpEMxxx-b17To18KO3Ed+@mail.gmail.com>
	<1301348702013-6216937.post@n2.nabble.com>
Message-ID: <AANLkTinKg+KnKqGHQ7W9aBE2O17564_BR5f_fCKGYij-@mail.gmail.com>

On Mon, Mar 28, 2011 at 5:45 PM, Jeff Lenk <jeff at jefflenk.com> wrote:

> return your acl.conf.xml to default and let the phones register to
> authenticate. You could do it with acls to but if you dont understand that
> yet try it simpler first:)
>
>
Hey Jeff,
I only started to mess around with the acl file after I had this problem
with the default config :( So that can't be it.
What else you got? :P
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From msc at freeswitch.org  Tue Mar 29 01:52:55 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 28 Mar 2011 14:52:55 -0700
Subject: [Freeswitch-users] Embarrassing Question: Local extension
 registering in the Public context on default configuration?
In-Reply-To: <AANLkTin2YgsH+xVkP9VT4usbpEMxxx-b17To18KO3Ed+@mail.gmail.com>
References: <AANLkTinDZ0xBMxFssO9NrP8y6opAXHUrMOPoNzhS1U2M@mail.gmail.com>
	<AANLkTimCcZ33aEgEKi6qmNj8K9Uq9RYBoHr-sPXAByzA@mail.gmail.com>
	<AANLkTikhKTF3TvXAf_vHPWxn6VHkwF2gVZpYcsP401bT@mail.gmail.com>
	<AANLkTim3dZh0rit8cEicCRm73maVeOaHDy9snBfnPBa-@mail.gmail.com>
	<AANLkTin2YgsH+xVkP9VT4usbpEMxxx-b17To18KO3Ed+@mail.gmail.com>
Message-ID: <AANLkTimQkBwOQokB1S5tC1q+woRp3cXO7L4qKQq7uHNp@mail.gmail.com>

DING DING DING DING! We have a weener! :P
Look at line 3 of your pastebin:

2011-03-26 02:00:54.058641 [DEBUG] sofia.c:6466 IP 192.168.3.121
Approved by acl "domains[]". Access Granted.

Your "domains" ACL is letting the call in, but is not "authenticating"
it so it's hitting the default context. If you absolutely want IP auth
then you need to go into each directory entry for each user you want
to have IP auth and add a tag like this:

<user id="1000" cidr="192.168.3.10">

The better way to go, IMHO, is simply to use digest authentication.
That way you don't have to worry about what IP address for what user,
etc. Just make sure that you change the SIP password to something
better than "1234" :)

I answered this yesterday asking about the ACL. Granted it wasn't
within 10 minutes but it was a Sunday for me... ;)

-MC

BTW,

On Mon, Mar 28, 2011 at 2:01 PM, A E [Gmail] <all.eforums at gmail.com> wrote:
> On Mon, Mar 28, 2011 at 4:41 PM, Michael Collins <msc at freeswitch.org> wrote:
>>
>> > Anymore ideas? With the gurus on the list, I'd have thought this would
>> > be
>> > answered in 10 mins of it being posted :)
>>
>> :)
>>
>> Did you nail down the question about whether you have an ACL set?
>> Also, did you pastebin a debug log of the call hitting the dialplan?
>>
>> -MC
>>
>
> So, I did muck around a tad with the acl.conf.xml file although I was under
> the impression that it was auto-magically generated at install time? It
> looks like this.
> ?? ?<list name="lan" default="allow">
> ?? ? ?<node type="deny" cidr="0.0.0.0/0"/>
> ?? ? ?<node type="allow" cidr="192.168.3.0/24"/>
> ?? ?</list>
> ?? ?<list name="domains" default="deny">
> ?? ? <node type="allow" domain="$${domain}"/>
> ?? ? ?<node type="allow" cidr="192.168.3.0/24"/>
> ?? ?</list>
> and yes, I put the pastebin here:?http://pastebin.freeswitch.org/15881
> I also keep seeing this on the console in fs_cli:
> 2011-03-26 03:48:54.397538 [WARNING] sofia_reg.c:1246 SIP auth challenge
> (REGISTER) on sofia profile 'internal' for [1000 at 192.168.3.101] from ip
> 192.168.3.121
> Not sure why this is a "warning", and that might be an indication that FS
> thinks this UA is "foreign"?
> HTH?
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Tue Mar 29 01:53:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 28 Mar 2011 14:53:36 -0700
Subject: [Freeswitch-users] Embarrassing Question: Local extension
 registering in the Public context on default configuration?
In-Reply-To: <AANLkTinKg+KnKqGHQ7W9aBE2O17564_BR5f_fCKGYij-@mail.gmail.com>
References: <AANLkTinDZ0xBMxFssO9NrP8y6opAXHUrMOPoNzhS1U2M@mail.gmail.com>
	<AANLkTimCcZ33aEgEKi6qmNj8K9Uq9RYBoHr-sPXAByzA@mail.gmail.com>
	<AANLkTikhKTF3TvXAf_vHPWxn6VHkwF2gVZpYcsP401bT@mail.gmail.com>
	<AANLkTim3dZh0rit8cEicCRm73maVeOaHDy9snBfnPBa-@mail.gmail.com>
	<AANLkTin2YgsH+xVkP9VT4usbpEMxxx-b17To18KO3Ed+@mail.gmail.com>
	<1301348702013-6216937.post@n2.nabble.com>
	<AANLkTinKg+KnKqGHQ7W9aBE2O17564_BR5f_fCKGYij-@mail.gmail.com>
Message-ID: <AANLkTik=M+Z5DJ3JJv-tA8ti_r__VYwc8iqCdxt1h_9=@mail.gmail.com>

On Mon, Mar 28, 2011 at 2:50 PM, A E [Gmail] <all.eforums at gmail.com> wrote:
> On Mon, Mar 28, 2011 at 5:45 PM, Jeff Lenk <jeff at jefflenk.com> wrote:
>>
>> return your acl.conf.xml to default and let the phones register to
>> authenticate. You could do it with acls to but if you dont understand that
>> yet try it simpler first:)
>>
>
> Hey Jeff,
> I only started to mess around with the acl file after I had this problem
> with the default config :( So that can't be it.
> What else you got? :P
Actually, it is that. See my other post.
-MC


From gourav at rentec.com  Tue Mar 29 00:56:53 2011
From: gourav at rentec.com (Gourav Vohra)
Date: Mon, 28 Mar 2011 16:56:53 -0400 (EDT)
Subject: [Freeswitch-users] CALL_REJECT on freeswitch
In-Reply-To: <2081591548.41728.1301345164462.JavaMail.root@zinnia1>
Message-ID: <1366439243.41832.1301345813484.JavaMail.root@zinnia1>




Does the call have to be rejected twice by the end user for calls to be transferred to voicemail on freeswitch?

In my case I have to hit the reject button twice on the phone for calls to be transferred to voicemail.

Thanks for your help.


From msc at freeswitch.org  Tue Mar 29 02:04:04 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 28 Mar 2011 15:04:04 -0700
Subject: [Freeswitch-users] CALL_REJECT on freeswitch
In-Reply-To: <1366439243.41832.1301345813484.JavaMail.root@zinnia1>
References: <2081591548.41728.1301345164462.JavaMail.root@zinnia1>
	<1366439243.41832.1301345813484.JavaMail.root@zinnia1>
Message-ID: <AANLkTin6eOcbaSvJv6+X9=2Rr6NE55VE0A-Yc-hkgOmd@mail.gmail.com>

> Does the call have to be rejected twice by the end user for calls to be transferred to voicemail on freeswitch?
Probably not. Can you get a siptrace of this behavior? Most likely the
phone isn't actually sending the BYE until you hit reject a second
time. What kind of phone is it?
-MC


From all.eforums at gmail.com  Tue Mar 29 02:07:20 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Mon, 28 Mar 2011 18:07:20 -0400
Subject: [Freeswitch-users] Embarrassing Question: Local extension
 registering in the Public context on default configuration?
In-Reply-To: <AANLkTimQkBwOQokB1S5tC1q+woRp3cXO7L4qKQq7uHNp@mail.gmail.com>
References: <AANLkTinDZ0xBMxFssO9NrP8y6opAXHUrMOPoNzhS1U2M@mail.gmail.com>
	<AANLkTimCcZ33aEgEKi6qmNj8K9Uq9RYBoHr-sPXAByzA@mail.gmail.com>
	<AANLkTikhKTF3TvXAf_vHPWxn6VHkwF2gVZpYcsP401bT@mail.gmail.com>
	<AANLkTim3dZh0rit8cEicCRm73maVeOaHDy9snBfnPBa-@mail.gmail.com>
	<AANLkTin2YgsH+xVkP9VT4usbpEMxxx-b17To18KO3Ed+@mail.gmail.com>
	<AANLkTimQkBwOQokB1S5tC1q+woRp3cXO7L4qKQq7uHNp@mail.gmail.com>
Message-ID: <AANLkTinoSx_A-LzOJMm2cJqbghQN4oceLoq4=jScTSW9@mail.gmail.com>

On Mon, Mar 28, 2011 at 5:52 PM, Michael Collins <msc at freeswitch.org> wrote:

> DING DING DING DING! We have a weener! :P
> Look at line 3 of your pastebin:
>
> 2011-03-26 02:00:54.058641 [DEBUG] sofia.c:6466 IP 192.168.3.121
> Approved by acl "domains[]". Access Granted.
>
>
Believe it or not, I SWEAR I was going to highlight that and point it out
that my $${domain} was coming up empty and it didn't have anyway to know who
was local and who wasn't. So then I added the line

<node type="allow" cidr="192.168.3.0/24"/> under <list name="domains"> to
"teach" it! But that didn't work either.


> Your "domains" ACL is letting the call in, but is not "authenticating"
> it so it's hitting the default context. If you absolutely want IP auth
> then you need to go into each directory entry for each user you want
> to have IP auth and add a tag like this:
>
> <user id="1000" cidr="192.168.3.10">
>
> The better way to go, IMHO, is simply to use digest authentication.
> That way you don't have to worry about what IP address for what user,
> etc. Just make sure that you change the SIP password to something
> better than "1234" :)
>
>
Sorry, how do I make it do that? Doesn't it do that by default?


> I answered this yesterday asking about the ACL. Granted it wasn't
> within 10 minutes but it was a Sunday for me... ;)
>
>
Might have missed it :(


> -MC
>
>
So bottom line, how do I fix this? like I kinda knew what the problem was,
just wasn't sure and didn't know how to fix it :( If changing back to the
original/default acl is the answer then why did this problem happen to begin
with when I started out with a the basic default config that "make samples"
spits out...hell I didn't even change the default pwd (don't worry this is
running inside a VM and has no outside access) for registering the UAs. But
when I had this issue, I started to much around.
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From richocet2 at hotmail.com  Tue Mar 29 02:19:14 2011
From: richocet2 at hotmail.com (Dave Bracken)
Date: Mon, 28 Mar 2011 17:19:14 -0500
Subject: [Freeswitch-users] Weird results with "show calls"
Message-ID: <SNT144-w55951A42941A69D9CA77C4F0BA0@phx.gbl>


I am just wondering what your thoughts are on this. When i use Delphi to originate a call to a softphone, and then do a "show calls", it shows zero calls. If o call one softphone to another, then it shows 1 call. Why is it not showing any info for a software originated call? anybody?
Thanks in advance,
Dave 		 	   		  
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From richocet2 at hotmail.com  Tue Mar 29 02:20:03 2011
From: richocet2 at hotmail.com (Dave Bracken)
Date: Mon, 28 Mar 2011 17:20:03 -0500
Subject: [Freeswitch-users] Stopping uuid_record error
In-Reply-To: <1301321232027-6215119.post@n2.nabble.com>
References: <SNT144-w107EBF7DF2D9C9EA0FB742F0B70@phx.gbl>,
	<1301321232027-6215119.post@n2.nabble.com>
Message-ID: <SNT144-w256C238CDAEDD8C0A0ADD4F0BA0@phx.gbl>


Thanks for the reply. But it turned out the file path solution worked for me.
Dave
 
> Date: Mon, 28 Mar 2011 07:07:12 -0700
> From: jeff at jefflenk.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Stopping uuid_record error
> 
> you can also specify "all" as the file which removes all bugs for that uuid
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Stopping-uuid-record-error-tp6201087p6215119.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
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From all.eforums at gmail.com  Tue Mar 29 02:22:40 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Mon, 28 Mar 2011 18:22:40 -0400
Subject: [Freeswitch-users] Embarrassing Question: Local extension
 registering in the Public context on default configuration?
In-Reply-To: <AANLkTinoSx_A-LzOJMm2cJqbghQN4oceLoq4=jScTSW9@mail.gmail.com>
References: <AANLkTinDZ0xBMxFssO9NrP8y6opAXHUrMOPoNzhS1U2M@mail.gmail.com>
	<AANLkTimCcZ33aEgEKi6qmNj8K9Uq9RYBoHr-sPXAByzA@mail.gmail.com>
	<AANLkTikhKTF3TvXAf_vHPWxn6VHkwF2gVZpYcsP401bT@mail.gmail.com>
	<AANLkTim3dZh0rit8cEicCRm73maVeOaHDy9snBfnPBa-@mail.gmail.com>
	<AANLkTin2YgsH+xVkP9VT4usbpEMxxx-b17To18KO3Ed+@mail.gmail.com>
	<AANLkTimQkBwOQokB1S5tC1q+woRp3cXO7L4qKQq7uHNp@mail.gmail.com>
	<AANLkTinoSx_A-LzOJMm2cJqbghQN4oceLoq4=jScTSW9@mail.gmail.com>
Message-ID: <AANLkTinvvsZN+OL5WAjwv+bGVzDEqokx65Lfdkh_SnNB@mail.gmail.com>

On Mon, Mar 28, 2011 at 6:07 PM, A E [Gmail] <all.eforums at gmail.com> wrote:

> On Mon, Mar 28, 2011 at 5:52 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> DING DING DING DING! We have a weener! :P
>> Look at line 3 of your pastebin:
>>
>> 2011-03-26 02:00:54.058641 [DEBUG] sofia.c:6466 IP 192.168.3.121
>> Approved by acl "domains[]". Access Granted.
>>
>>
> Believe it or not, I SWEAR I was going to highlight that and point it out
> that my $${domain} was coming up empty and it didn't have anyway to know who
> was local and who wasn't. So then I added the line
>
> <node type="allow" cidr="192.168.3.0/24"/> under <list name="domains"> to
> "teach" it! But that didn't work either.
>
>
>> Your "domains" ACL is letting the call in, but is not "authenticating"
>> it so it's hitting the default context. If you absolutely want IP auth
>> then you need to go into each directory entry for each user you want
>> to have IP auth and add a tag like this:
>>
>> <user id="1000" cidr="192.168.3.10">
>>
>> The better way to go, IMHO, is simply to use digest authentication.
>> That way you don't have to worry about what IP address for what user,
>> etc. Just make sure that you change the SIP password to something
>> better than "1234" :)
>>
>>
> Sorry, how do I make it do that? Doesn't it do that by default?
>
>
>> I answered this yesterday asking about the ACL. Granted it wasn't
>> within 10 minutes but it was a Sunday for me... ;)
>>
>>
> Might have missed it :(
>
>
>> -MC
>>
>>
> So bottom line, how do I fix this? like I kinda knew what the problem was,
> just wasn't sure and didn't know how to fix it :( If changing back to the
> original/default acl is the answer then why did this problem happen to begin
> with when I started out with a the basic default config that "make samples"
> spits out...hell I didn't even change the default pwd (don't worry this is
> running inside a VM and has no outside access) for registering the UAs. But
> when I had this issue, I started to much around.
>

Bloody hell! I moved the modified acl file and did a "make samples" again
and now it's going through the extensions in default context. Of course I
get a "502" back but, with other bizarre issues going on...coz of all the
hacky stuff I did to make it compile on Solaris 11, primarily coz it uses
the "mod_hash" application that didn't compile so I commented it out but now
it hangs up the call whenever that application is called in the dialplan.
Wonder if anyone will answer my other email which has my Solaris woes!

Thanks Michael and Jeff, I have no idea how, but you were right! I still
don't get why the default config didn't work the first time!
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From msc at freeswitch.org  Tue Mar 29 02:50:21 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 28 Mar 2011 15:50:21 -0700
Subject: [Freeswitch-users] Weird results with "show calls"
In-Reply-To: <SNT144-w55951A42941A69D9CA77C4F0BA0@phx.gbl>
References: <SNT144-w55951A42941A69D9CA77C4F0BA0@phx.gbl>
Message-ID: <AANLkTikdJPitmQHySCGjHuLbDpbeR6DzWVN2XdsYSbZ2@mail.gmail.com>

This is the classic issue of "channels" vs. "calls":
A "channel' is just a single call leg, but a "call" is two channels
bridged together.

So, dialing from one phone to another phone will yield 2 channels and 1 call.

You homework assignment tonight is to make several calls and
experiment with these commands:

show calls
show channels
show distinct_channels

Try calling from phone to phone as well as trying things like calling
voicemail. Let us know if you have any more questions.
-MC

On Mon, Mar 28, 2011 at 3:19 PM, Dave Bracken <richocet2 at hotmail.com> wrote:
> I am just wondering what your thoughts are on this. When i use Delphi to
> originate a call to a softphone, and then do a "show calls", it shows zero
> calls. If o call one softphone to another, then it shows 1 call. Why is it
> not showing any info for a software originated call? anybody?
> Thanks in advance,
> Dave
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From lists at telefaks.de  Tue Mar 29 04:29:01 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Tue, 29 Mar 2011 02:29:01 +0200
Subject: [Freeswitch-users] Dingaling and sasl authentication failed
Message-ID: <4D9127CD.7060300@telefaks.de>

Hello,

I installed mod_dingaling and having problems with the registration

freeswitch at internal> dingaling status
--DingaLing status--
login   |       connected
my.account at googlemail.com/talk   |       UNCONNECTED

2011-03-29 02:02:48.756832 [DEBUG] libdingaling.c:1289 sasl
authentication failed
2011-03-29 02:02:48.756832 [DEBUG] libdingaling.c:1607 io error 2 7
retry in 47 second(s)

Searching the mailing list led me to look at libgnutls26.
But libgnutls26 libgnutls-dev are both installed, and both the are
newest version, and both were installed long before I configured and
compiled mod_dingaling. And TLs for SIP is working sucessfully since a
while.

Anybody has a hint where to look further?

-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de




From all.eforums at gmail.com  Tue Mar 29 05:29:11 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Mon, 28 Mar 2011 21:29:11 -0400
Subject: [Freeswitch-users] Invalid Application hash
In-Reply-To: <AANLkTimyis5WmY0D2L+6zUKdJbirGMM0viAuHDZ-N_Ev@mail.gmail.com>
References: <AANLkTimX=5mruJzK683hdqOrjZ_unwTmVyWo3oNa601+@mail.gmail.com>
	<AANLkTimShuxrmaNA6MQDipFsZqD9pxiJ1UOGOC7LZJH6@mail.gmail.com>
	<AANLkTimyis5WmY0D2L+6zUKdJbirGMM0viAuHDZ-N_Ev@mail.gmail.com>
Message-ID: <AANLkTin57hx=u6s0zHCW=PjROJV3XFE+o7=Bhkh2Dce0@mail.gmail.com>

On Mon, Oct 25, 2010 at 7:07 AM, David Wafula <davidwaf at gmail.com> wrote:

> Thank you, i upgraded ....
> <load module="mod_hash"/>
> did it..
>

You're lucky, as this ain't that easy in Solaris and no one seems to be
telling us if someone managed to successfully compile/build mod_hash in
Solaris 11. I'm still stuck! That and ESL!
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From all.eforums at gmail.com  Tue Mar 29 05:34:15 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Mon, 28 Mar 2011 21:34:15 -0400
Subject: [Freeswitch-users] How to Build mod_hash and ESL on Solaris 11
	Express?
Message-ID: <AANLkTikt=SSKdmLpBwofkjvhNV8_TqBvV3iUzUqxMMS1@mail.gmail.com>

On Wed, Mar 23, 2011 at 10:38 AM, A E [Gmail] <all.eforums at gmail.com> wrote:

> On Wed, Mar 23, 2011 at 9:53 AM, Michal Bielicki <
> michal.bielicki at seventhsignal.de> wrote:
>
>> You can see the results of my builds here: http://www.freeswitch.de:8080/ for
>> both openindiana and solaris express 1
>>
>>
>> Thanks Michael. I noticed that you aren't making mod_hash which might be
> how you're getting through the entire build? I know you said it's not
> mod-hash but the PERL module but I guess it's getting encountered/used in
> the compilation of mod_hash?
>
> BTW, I am also getting these WARNINGs for all sorts of "op", <<, = etc.
>
> *warning: integer overflow detected: op "<<" (E_INTEGER_OVERFLOW_DETECTED)
> *
>
> But I noticed that you don't have them. Did you ever get them? and if you
> did then how did you solve them?
>
> Thanks
> \AEG
>
>
>
Hello Michal,

sorry to be a pain but could you help us with this? I managed to
successfully compile and run FS on Solaris 11 Express 64-Bit i386 platform
but using the default dialplan fails as it needs the hash application when
FS tries to write stuff to it's internal database (I'm assuming). Has anyone
compiled mod_hash and ESL on Solaris? Once the -Wxxx CXX flags are removed
from the Makefile, I get all of this
[http://pastebin.freeswitch.org/15886] when
trying to "gmake mod_hash"

Help! please? :D
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From all.eforums at gmail.com  Tue Mar 29 05:39:13 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Mon, 28 Mar 2011 21:39:13 -0400
Subject: [Freeswitch-users] Build on solaris fails in libs/esl
In-Reply-To: <AANLkTimGtXWdRPAv8OJYWeu8VYtCF8ZfQt0osGVgU7G=@mail.gmail.com>
References: <AANLkTimGtXWdRPAv8OJYWeu8VYtCF8ZfQt0osGVgU7G=@mail.gmail.com>
Message-ID: <AANLkTinab+WgVXUYtWaSccbwBV7G7LO-qjjcE4bYAKXj@mail.gmail.com>

On Fri, Feb 18, 2011 at 3:17 PM, Kim Culhan <w8hdkim at gmail.com> wrote:

> Seeing this on Opensolaris b134 and on latest Solaris 11:
>
> making all mod_hash
> Creating mod_hash_la-mod_hash.lo
> mkdir .libs
> Compiling mod_hash.c ...
> "../../../../src/include/switch_types.h", line 563: warning: integer
> overflow detected: op "<<" (E_INTEGER_OVERFLOW_DETECTED)
> cc: Warning: Option -db passed to ld, if ld is invoked, ignored otherwise
> cc: Warning: Option -ffast-math passed to ld, if ld is invoked, ignored
> otherwise
> cc: -W option with unknown program all
> make[5]: *** [src/esl.o] Error 1
> make[4]: *** [/usr/local/src/freeswitch/freeswitch/libs/esl/libesl.so]
> Error 2
> make[3]: *** [mod_hash-all] Error 1
> make[2]: *** [all-recursive] Error 1
> make[1]: *** [all-recursive] Error 1
> gmake: *** [all] Error 2
>
> In freeswitch/libs/esl/Makefile:
>
> CXXFLAGS=$(BASE_FLAGS) -Wall -Werror -Wno-unused-variable
>
> With mod_hash commented in modules.conf all of fs compiles.
>
> Any help is greatly appreciated.
>
> -kim
>
>
>
Just saw this email. Did you ever succeed in doing this? I know you asked me
another post of what I did, and I had done the same as you, commenting it
out, but FS crashes all over the place without this application :(
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From rupa at rupa.com  Tue Mar 29 07:21:47 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 28 Mar 2011 22:21:47 -0500
Subject: [Freeswitch-users] nibblebill sql requests
In-Reply-To: <17E3569462944217BBAE2AA65EFDEE97@e1705>
References: <17E3569462944217BBAE2AA65EFDEE97@e1705>
Message-ID: <AANLkTinocOdxT3v6W1OTJQwKrdzFhQGsQ47ykk25Rc3U@mail.gmail.com>

The hook at hangup can probably be removed and only have the hook at
reporting.  Jira?

On Wed, Feb 9, 2011 at 12:06 AM, Madovsky <infos at madovsky.org> wrote:
> I retried and it's the same
> in fact nibblebill makes sql request at channel destroy and normal
> clearing....
> I don't know how to resolve it...
>
>
> ----- Original Message -----
> From: Madovsky
> To: FreeSWITCH Users Help
> Sent: Tuesday, February 08, 2011 6:49 PM
> Subject: Re: [Freeswitch-users] nibblebill sql requests
> no 60s
> but maybe it could be I hanged up at the same time
> of the last second of 60 seconds
> but I'd like to be sure... I test again..
>
> ----- Original Message -----
> From: Avi Marcus
> To: FreeSWITCH Users Help
> Sent: Tuesday, February 08, 2011 6:30 PM
> Subject: Re: [Freeswitch-users] nibblebill sql requests
> I don't know why the first is billing so soon - is your heartbeat set to 1
> second?
> But the second is billing again because the call hung up.
> -Avi
>
> On Wed, Feb 9, 2011 at 12:51 AM, Madovsky <infos at madovsky.org> wrote:
>>
>> Avi,
>>
>> look at the time, there are?2 identical?select / update in the same second
>>
>> ----- Original Message -----
>> From: Avi Marcus
>> To: FreeSWITCH Users Help
>> Sent: Tuesday, February 08, 2011 2:18 PM
>> Subject: Re: [Freeswitch-users] nibblebill sql requests
>> I don't see why you think this is duplicate.
>> nibblebill automatically bills every X as per the "heartbeat" variable.
>> Also, since the timer is sometimes slightly off, it has to make up for it
>> with a fraction at the end.
>> Also, the constant "selects" is to enable the lowbal/nobal actions. That
>> LAST select does seem suspect, however.
>> If you only want POST-call billing, then set the heartbeat variable to 0 I
>> think? or just something big? in the conf xml -?<param
>> name="global_heartbeat" value="300">
>> -Avi
>>
>> On Tue, Feb 8, 2011 at 6:11 PM, Madovsky <infos at madovsky.org> wrote:
>>>
>>> after hangup,
>>> nibblebill makes unnecessary duplicated SQL requests :
>>>
>>>
>>> 2011-02-08 11:06:36.894238 [DEBUG] mod_sofia.c:457 Channel
>>> sofia/internal/9999999999999 at domain.com hanging up, cause: NORMAL_CLEARING
>>> 2011-02-08 11:06:37.018489 [DEBUG] mod_nibblebill.c:433 Attempting to
>>> bill at $0.03864 per minute to account 9999999999999
>>> 2011-02-08 11:06:37.018489 [INFO] mod_nibblebill.c:485 Beginning new
>>> billing on 461772290 at 12.34.56.78
>>> 2011-02-08 11:06:37.018489 [DEBUG] mod_nibblebill.c:491 1 seconds passed
>>> since last bill time of 2011-02-08 11:06:35
>>> 2011-02-08 11:06:37.018489 [DEBUG] mod_nibblebill.c:498 Billing $0.000874
>>> to 9999999999999 (Call: 461772290 at 12.34.56.78 / 0.000000 so far)
>>> 2011-02-08 11:06:37.018489 [DEBUG] mod_nibblebill.c:321 Doing update
>>> query
>>> [UPDATE accounts SET cash=cash-0.000874 WHERE id='9999999999999']
>>> 2011-02-08 11:06:37.074740 [DEBUG] mod_nibblebill.c:366 Doing lookup
>>> query
>>> [SELECT cash AS nibble_balance FROM accounts WHERE id='9999999999999']
>>> 2011-02-08 11:06:37.082844 [DEBUG] mod_nibblebill.c:376 Retrieved current
>>> balance for account 9999999999999 (balance = 9.579666)
>>> 2011-02-08 11:06:37.082844 [DEBUG] switch_core_state_machine.c:46
>>> sofia/internal/9999999999999 at domain.com Standard HANGUP, cause:
>>> NORMAL_CLEARING
>>> 2011-02-08 11:06:37.082844 [DEBUG] switch_core_state_machine.c:557
>>> (sofia/internal/9999999999999 at domain.com) State HANGUP going to sleep
>>> 2011-02-08 11:06:37.082844 [DEBUG] switch_core_state_machine.c:351
>>> (sofia/internal/9999999999999 at domain.com) State Change CS_HANGUP ->
>>> CS_REPORTING
>>> 2011-02-08 11:06:37.082844 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/internal/9999999999999 at domain.com [BREAK]
>>> 2011-02-08 11:06:37.082844 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/internal/9999999999999 at domain.com) Running State Change CS_REPORTING
>>> 2011-02-08 11:06:37.082844 [DEBUG] switch_core_state_machine.c:617
>>> (sofia/internal/9999999999999 at domain.com) State REPORTING
>>> 2011-02-08 11:06:37.082844 [DEBUG] mod_nibblebill.c:433 Attempting to
>>> bill at $0.03864 per minute to account 9999999999999
>>> 2011-02-08 11:06:37.082844 [DEBUG] mod_nibblebill.c:491 0 seconds passed
>>> since last bill time of 2011-02-08 11:06:37
>>> 2011-02-08 11:06:37.082844 [DEBUG] mod_nibblebill.c:498 Billing $0.000041
>>> to 9999999999999 (Call: 461772290 at 12.34.56.78 / 0.000874 so far)
>>> 2011-02-08 11:06:37.082844 [DEBUG] mod_nibblebill.c:321 Doing update
>>> query
>>> [UPDATE accounts SET cash=cash-0.000041 WHERE id='9999999999999']
>>> 2011-02-08 11:06:37.133170 [DEBUG] mod_nibblebill.c:366 Doing lookup
>>> query
>>> [SELECT cash AS nibble_balance FROM accounts WHERE id='9999999999999']
>>> 2011-02-08 11:06:37.201557 [DEBUG] mod_nibblebill.c:376 Retrieved current
>>> balance for account 9999999999999 (balance = 9.579625)
>>>
>>> How to avoid duplicate SQL requests with nibblebill ?
>>>
>>> Thanks
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> ________________________________
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> ________________________________
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From anthony.minessale at gmail.com  Tue Mar 29 07:40:36 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Mar 2011 22:40:36 -0500
Subject: [Freeswitch-users] Build on solaris fails in libs/esl
In-Reply-To: <AANLkTinab+WgVXUYtWaSccbwBV7G7LO-qjjcE4bYAKXj@mail.gmail.com>
References: <AANLkTimGtXWdRPAv8OJYWeu8VYtCF8ZfQt0osGVgU7G=@mail.gmail.com>
	<AANLkTinab+WgVXUYtWaSccbwBV7G7LO-qjjcE4bYAKXj@mail.gmail.com>
Message-ID: <AANLkTinAtxnnmbVm-tBvEuOxUQ7=z2JebB9TK+0WdBuF@mail.gmail.com>

try latest git to see if the silly out of bound thing is fixed.


On Mon, Mar 28, 2011 at 8:39 PM, A E [Gmail] <all.eforums at gmail.com> wrote:
> On Fri, Feb 18, 2011 at 3:17 PM, Kim Culhan <w8hdkim at gmail.com> wrote:
>>
>> Seeing this on Opensolaris b134 and on latest Solaris 11:
>>
>> making all mod_hash
>> Creating mod_hash_la-mod_hash.lo
>> mkdir .libs
>> Compiling mod_hash.c ...
>> "../../../../src/include/switch_types.h", line 563: warning: integer
>> overflow detected: op "<<" (E_INTEGER_OVERFLOW_DETECTED)
>> cc: Warning: Option -db passed to ld, if ld is invoked, ignored otherwise
>> cc: Warning: Option -ffast-math passed to ld, if ld is invoked, ignored
>> otherwise
>> cc: -W option with unknown program all
>> make[5]: *** [src/esl.o] Error 1
>> make[4]: *** [/usr/local/src/freeswitch/freeswitch/libs/esl/libesl.so]
>> Error 2
>> make[3]: *** [mod_hash-all] Error 1
>> make[2]: *** [all-recursive] Error 1
>> make[1]: *** [all-recursive] Error 1
>> gmake: *** [all] Error 2
>>
>> In freeswitch/libs/esl/Makefile:
>>
>> CXXFLAGS=$(BASE_FLAGS) -Wall -Werror -Wno-unused-variable
>>
>> With mod_hash commented in modules.conf all of fs compiles.
>>
>> Any help is greatly appreciated.
>>
>> -kim
>>
>
> Just saw this email. Did you ever succeed in doing this? I know you asked me
> another post of what I did, and I had done the same as you, commenting it
> out, but FS crashes all over the place without this application :(
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 all.eforums at gmail.com  Tue Mar 29 09:27:30 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Tue, 29 Mar 2011 01:27:30 -0400
Subject: [Freeswitch-users] Build on solaris fails in libs/esl
In-Reply-To: <AANLkTinAtxnnmbVm-tBvEuOxUQ7=z2JebB9TK+0WdBuF@mail.gmail.com>
References: <AANLkTimGtXWdRPAv8OJYWeu8VYtCF8ZfQt0osGVgU7G=@mail.gmail.com>
	<AANLkTinab+WgVXUYtWaSccbwBV7G7LO-qjjcE4bYAKXj@mail.gmail.com>
	<AANLkTinAtxnnmbVm-tBvEuOxUQ7=z2JebB9TK+0WdBuF@mail.gmail.com>
Message-ID: <AANLkTim77j3aBU0YrR8YGZEp=bDf9ND3-T0NXSznW4ov@mail.gmail.com>

On Mon, Mar 28, 2011 at 11:40 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> try latest git to see if the silly out of bound thing is fixed.
>
>
Hi Anthony,

It seems to have been, yes. I don't see the these "integer overflow"
messages, but now some other ones have been noticed:

e.g.
make: *** [current] Error 2"/home/freeswitch/fs/libs/esl/src/include/esl.h",
line 465: warning: implicit function declaration: strcasecmp
(E_NO_IMPLICIT_DECL_ALLOWED)
"libs/esl/src/esl.c", line 1191: warning: implicit function declaration:
strdup (E_NO_IMPLICIT_DECL_ALLOWED)
"libs/esl/src/esl.c", line 1191: warning: improper pointer/integer
combination: op "=" (E_BAD_PTR_INT_COMBINATION)
"libs/esl/src/esl.c", line 1232: warning: improper pointer/integer
combination: op "=" (E_BAD_PTR_INT_COMBINATION)
"/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
"/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
"/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
"/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
"libs/esl/ivrd.c", line 98: warning: implicit function declaration: signal
(E_NO_IMPLICIT_DECL_ALLOWED)
"/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
"/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)

As well as a load of these:

-- "/home/freeswitch/fs/libs/spandsp/src/spandsp/bit_operations.h", line 50:
warning: "__asm__" is an extension of ANSI C (E_KW_IS_AN_EXTENSION_OF_ANSI)

And lastly,

BTW, Not that you claimed that you fixed it, might as well tell you that,
esl and mod_hash compilation still fails. Pastebin here:
http://pastebin.freeswitch.org/15887
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From boris at tagnet.ru  Tue Mar 29 09:51:08 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 29 Mar 2011 11:51:08 +0600
Subject: [Freeswitch-users] bind_digit_action help
Message-ID: <4D91734C.8010208@tagnet.ru>

Hello!

     Is this possible to get bind_digit_action work on B leg? I got it 
working on A leg and can't on B. Would someone provide a little example?

-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From u2nsam at gmail.com  Tue Mar 29 10:01:55 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 29 Mar 2011 11:31:55 +0530
Subject: [Freeswitch-users] signaling change
Message-ID: <AANLkTimGcMgZFN3mhWJUJ95OhFXiDtY05XiRDg0dHBBc@mail.gmail.com>

Hello,

I am trying to use freeswitch as sbc and when calls passes through FS it
gives signaling change
from 183 to 180 , but I want to pass 183 to the leg b, how should i do it ?



Call is passed from leg a --> FS --> leg b

leg b ( 183 ) --> FS --> leg a (180 )

fscli output :--  http://pastebin.freeswitch.org/15888

line 70 to line 75



Regards
Sam
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From chistyakov at directtel.ru  Tue Mar 29 10:27:28 2011
From: chistyakov at directtel.ru (=?UTF-8?B?0KfQuNGB0YLRj9C60L7QsiDQmNCy0LDQvQ==?=)
Date: Tue, 29 Mar 2011 10:27:28 +0400
Subject: [Freeswitch-users] How to capture an events by setInputCallback ?
Message-ID: <4D917BD0.2000405@directtel.ru>

function my_cb(s, type, obj, arg)

    if (arg) then
       print("type: " .. type .. "\n" .. "arg: " .. arg .. "\n");
    else
       print("type: " .. type .. "\n");
    end

    if (type == "dtmf") then
       print("digit: [" .. obj['digit'] .. "]\nduration: [" .. 
obj['duration'] .. "]\n");
    else
       print(obj:serialize("xml"));

    end
return true
end

blah="w00t";                                                                                                                                                           
$

local session = freeswitch.Session("sofia/internal/Dt002%172.16.0.49");
session:answer();
session:execute("enable_heartbeat", 5);
session:setInputCallback("my_cb", "blah");
session:streamFile("/tmp/1.wav");

This script is work, but only for DTMF.



From frankie.k.yiu at gmail.com  Tue Mar 29 10:55:49 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Mon, 28 Mar 2011 23:55:49 -0700
Subject: [Freeswitch-users] How to create an event or message?
Message-ID: <AANLkTim8a8B8G-ZNuRp7CN06bJPkdopGP3J6OvRoBEhY@mail.gmail.com>

Hi there,

If I want to send an event or message to notify my other part of the system
(in C#) when a certain condition is met, how do I implement the event in C++
code?

For example, I might want to do data analysis of the RTP packets in C++,
and when I detected the right timing where I want to play an audio, I would
create and send an event to notify another part of the system.

Is there any hint or sample code that I can take a look?

Thanks,

Frankie

On Mon, Mar 28, 2011 at 12:50 AM, Frankie Yiu <frankie.k.yiu at gmail.com>wrote:

> Hi there,
>
> If I want to send my own event or message to notify my other part of
> the system when a certain condition is met, how do I implement that in C++
> code?
> For example, I might want to do data analysis of the RTP packets in C++,
> and when I detected the right timing where I want to play an audio, I can
> send an event to my other part of the system to play an audio file.
>
> Is there any hint or sample code that I can take a look?
>
> Thanks,
>
> Frankie
>
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From ovvenkatesan at gmail.com  Tue Mar 29 12:52:41 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Tue, 29 Mar 2011 14:22:41 +0530
Subject: [Freeswitch-users] need help on : kenyan pri line setting
Message-ID: <AANLkTi=8KVWALfs80Y6W6+CsPRx6NLY7vTPdf9VX5xD3@mail.gmail.com>

Hi,

Any one please tell me,
E1 line setting for kenyan E1 line.

Which *E1 Line coding* and *framing* I have to use it for kenyan E1 line.

My current setting works fine in India.

When I am trying to connect to kenyan E1 line,
wanrouter status,  says,

Device name | Protocol | Station | Status        |
wanpipe1    | AFT TE1  | N/A     | Disconnected  |



I know, its not related to freeswitch.
But, I know no anyother place to get
help on this. If anyone done this before
in kenyan network please help me.


Regards,
Venkat.
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From fieldpeak at gmail.com  Tue Mar 29 13:08:04 2011
From: fieldpeak at gmail.com (Charles)
Date: Tue, 29 Mar 2011 17:08:04 +0800
Subject: [Freeswitch-users] FS performance test -Hard disk alway busy when
	registerred 500 users
Message-ID: <4d91a177.4323e70a.2cca.ffffefc8@mx.google.com>

I'm makeing the load test of the FreeSWITCH, I registerred 500 users to FS, and make NO calls, then the LED of Hard disk alway blink meaning it always busy... below is my hardware configuration, would someone can help how to improve the performance? thanks in advance.
i tried to increased the nonce-ttl to 600 from 60 tring to decrease the times of writing db. however it doesn't make help...
attached is the internal.xml for reference.
 
 
Windows server 2008 64bit.
CPU: Intel Core2 Duo L7400 with Intel 310 1.5GHz, 
Memory: 2G,
bit: 64bit,
HD: 160G 
 
P.S. i can think about RAMDISK, however, I?m afraid of it is so stable...  if it can be increase the performance by HD itself, it will be great.
 
Regards,
Charles
 
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From peter.olsson at visionutveckling.se  Tue Mar 29 13:26:24 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 29 Mar 2011 11:26:24 +0200
Subject: [Freeswitch-users] FS performance test -Hard disk alway busy
 when	registerred 500 users
In-Reply-To: <4d91a177.4323e70a.2cca.ffffefc8@mx.google.com>
References: <4d91a177.4323e70a.2cca.ffffefc8@mx.google.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B394B66A@cooper>

The recommendation is to use a RAM disk if you see performance problems. If your disk is "blinking" I wouldn't say that it's that much loaded though - in that case it should have a more or less steady light :) RAM disk shouldn't be a stability problem, and if you reboot your computer, FS will re-create all databases needed.
Also make sure to use latest GIT, there have been some changes how the sqlite db handles are cached internally.
/Peter
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Charles
Skickat: den 29 mars 2011 11:08
Till: freeswitch-users
?mne: [Freeswitch-users] FS performance test -Hard disk alway busy when registerred 500 users

I'm makeing the load test of the FreeSWITCH, I registerred 500 users to FS, and make NO calls, then the LED of Hard disk alway blink meaning it always busy... below is my hardware configuration, would someone can help how to improve the performance? thanks in advance.
i tried to increased the nonce-ttl to 600 from 60 tring to decrease the times of writing db. however it doesn't make help...
attached is the internal.xml for reference.


Windows server 2008 64bit.
CPU: Intel Core2 Duo L7400 with Intel 310 1.5GHz,
Memory: 2G,
bit: 64bit,
HD: 160G

P.S. i can think about RAMDISK, however, I'm afraid of it is so stable...  if it can be increase the performance by HD itself, it will be great.

Regards,
Charles

!DSPAM:4d91a2a832769362989175!
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From avi at avimarcus.net  Tue Mar 29 13:29:45 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 29 Mar 2011 11:29:45 +0200
Subject: [Freeswitch-users] FS performance test -Hard disk alway busy
 when registerred 500 users
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B394B66A@cooper>
References: <4d91a177.4323e70a.2cca.ffffefc8@mx.google.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B394B66A@cooper>
Message-ID: <AANLkTik8MryDa4Hbe9aPBYYjxMDXFzZk-RmDmAJTdL_5@mail.gmail.com>

500 registered users with an expiry of 3600 seconds/1 hour gives you only
one request every 7 seconds. What kind of registration timing do you have on
your phones?
-Avi

On Tue, Mar 29, 2011 at 11:26 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> The recommendation is to use a RAM disk if you see performance problems. If
> your disk is ?blinking? I wouldn?t say that it?s that much loaded though ?
> in that case it should have a more or less steady light :) RAM disk
> shouldn?t be a stability problem, and if you reboot your computer, FS will
> re-create all databases needed.
>
> Also make sure to use latest GIT, there have been some changes how the
> sqlite db handles are cached internally.
>
> /Peter
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Charles
> *Skickat:* den 29 mars 2011 11:08
> *Till:* freeswitch-users
> *?mne:* [Freeswitch-users] FS performance test -Hard disk alway busy when
> registerred 500 users
>
>
>
> I'm makeing the load test of the FreeSWITCH, I registerred 500 users to FS, and
> make NO calls,
> then the LED of Hard disk alway blink meaning it always busy... below is my hardware configuration, would someone can help how to improve the performance? thanks in advance.
>
>
> i tried to increased the nonce-ttl to 600 from 60 tring to decrease the times of writing db. however it doesn't make help...
>
> attached is the internal.xml for reference.
>
>
>
>
>
> Windows server 2008 64bit.
>
> CPU: Intel Core2 Duo L7400 with Intel 310 1.5GHz,
>
> Memory: 2G,
>
> bit: 64bit,
>
> HD: 160G
>
>
>
> P.S. i can think about RAMDISK, however, I?m afraid of it is so stable...
> if it can be increase the performance by HD itself, it will be great.
>
>
>
> Regards,
>
> Charles
>
>
>
> !DSPAM:4d91a2a832769362989175!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From wasim at convergence.pk  Tue Mar 29 13:36:21 2011
From: wasim at convergence.pk (Wasim Baig)
Date: Tue, 29 Mar 2011 14:36:21 +0500
Subject: [Freeswitch-users] need help on : kenyan pri line setting
In-Reply-To: <AANLkTi=8KVWALfs80Y6W6+CsPRx6NLY7vTPdf9VX5xD3@mail.gmail.com>
References: <AANLkTi=8KVWALfs80Y6W6+CsPRx6NLY7vTPdf9VX5xD3@mail.gmail.com>
Message-ID: <AANLkTikoiumv95u=UVy0Y7TTZJqLGMNBy1DmdVRth_M0@mail.gmail.com>

line coding should be hdb3
framing can be either crc4 or ncrc4 depending on what the other side has set
its most probably a cable fault, have then give you a loop from far end
set your clock to master and you should get e1 up, if not, then the issue is
in the cable
and should be resolved first

-wasim

On Tue, Mar 29, 2011 at 13:52, ovvenkat <ovvenkatesan at gmail.com> wrote:

> Hi,
>
> Any one please tell me,
> E1 line setting for kenyan E1 line.
>
> Which *E1 Line coding* and *framing* I have to use it for kenyan E1 line.
>
> My current setting works fine in India.
>
> When I am trying to connect to kenyan E1 line,
> wanrouter status,  says,
>
> Device name | Protocol | Station | Status        |
> wanpipe1    | AFT TE1  | N/A     | Disconnected  |
>
>
>
>  I know, its not related to freeswitch.
> But, I know no anyother place to get
> help on this. If anyone done this before
> in kenyan network please help me.
>
>
> Regards,
> Venkat.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
peace be upon you ...
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From ovvenkatesan at gmail.com  Tue Mar 29 13:42:20 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Tue, 29 Mar 2011 15:12:20 +0530
Subject: [Freeswitch-users] need help on : kenyan pri line setting
In-Reply-To: <AANLkTi=_xPtM8BSUO90jjGgdX9CtHfufgqDwZaHO2g7i@mail.gmail.com>
References: <AANLkTi=8KVWALfs80Y6W6+CsPRx6NLY7vTPdf9VX5xD3@mail.gmail.com>
	<AANLkTi=_xPtM8BSUO90jjGgdX9CtHfufgqDwZaHO2g7i@mail.gmail.com>
Message-ID: <AANLkTimp0c2q05vXr3YmY9_-rJiVrxZW52F35HOUrq40@mail.gmail.com>

Select signalling typeHi Frank.

Thank you very much for your quick reply.

I did not see anything like,

frame type : CSS

Here are my settings  I have selected
while configuring sangoma PRI Card .

1. Select media type

    I have selected  *E1*

2. Select line coding

    I have selected *HDB3*

3. Select framing

    I have selected *CRC4*

4. Select clock

    I have selected *NORMAL*

5. Select Switchtype

   I have selected *EuroISDN/ETSI*

6. Select signalling type

   I have selected *PRI CPE


*Still, When I check the *wanrouter status*
Its says , Disconnected *
*
Thanks in advance,

Regards,
Venkat. *

*On Tue, Mar 29, 2011 at 2:36 PM, Frank Ochere <ochere at gmail.com> wrote:

> Venkat,
>
> Pri Type EuroISDN
> CRC4 Enabled
> Coding HDB3
> Framing Type CCS
> Timing Source set to Incoming
> Bearer Channels 1-15, 17-31
> Data Channel 16
>
> Regards
>
> Frank
>
>
> On Tue, Mar 29, 2011 at 11:52 AM, ovvenkat <ovvenkatesan at gmail.com> wrote:
>
>> Hi,
>>
>> Any one please tell me,
>> E1 line setting for kenyan E1 line.
>>
>> Which *E1 Line coding* and *framing* I have to use it for kenyan E1 line.
>>
>>
>> My current setting works fine in India.
>>
>> When I am trying to connect to kenyan E1 line,
>> wanrouter status,  says,
>>
>> Device name | Protocol | Station | Status        |
>> wanpipe1    | AFT TE1  | N/A     | Disconnected  |
>>
>>
>>
>>  I know, its not related to freeswitch.
>> But, I know no anyother place to get
>> help on this. If anyone done this before
>> in kenyan network please help me.
>>
>>
>> Regards,
>> Venkat.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>


-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
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From fieldpeak at gmail.com  Tue Mar 29 15:03:23 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Tue, 29 Mar 2011 19:03:23 +0800
Subject: [Freeswitch-users] FS performance test -Hard disk alway busy
 when registerred 500 users
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B394B66A@cooper>
References: <4d91a177.4323e70a.2cca.ffffefc8@mx.google.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B394B66A@cooper>
Message-ID: <AANLkTimRHZUdrg+aeBYSyfwuA-r0Ee7ry4WTbLedq4VD@mail.gmail.com>

Hi Peter,

Thanks for your prompt reponse and recommendation.

I ever tried two ramdisk softwares (VSuite and DataRAM free edition),
both work. Can you advise which one is preferred using in the
production? or anyother recomended product running in windows
platform?

RAMDisk is a 'program' after all, i'm afraid whether it can be loaded
prior to freeswitch service?

if using ramdisk, we have to seperatedly install RAMdisk with FS
package in the filed, it maybe involve work comparing one installation
pakage...

is there any windows native support Ramdisk like the one in Linux?

sorry for coming up so many question :)

Regards,
Charles

2011/3/29, Peter Olsson <peter.olsson at visionutveckling.se>:
> The recommendation is to use a RAM disk if you see performance problems. If
> your disk is "blinking" I wouldn't say that it's that much loaded though -
> in that case it should have a more or less steady light :) RAM disk
> shouldn't be a stability problem, and if you reboot your computer, FS will
> re-create all databases needed.
> Also make sure to use latest GIT, there have been some changes how the
> sqlite db handles are cached internally.
> /Peter
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Charles
> Skickat: den 29 mars 2011 11:08
> Till: freeswitch-users
> ?mne: [Freeswitch-users] FS performance test -Hard disk alway busy when
> registerred 500 users
>
> I'm makeing the load test of the FreeSWITCH, I registerred 500 users to FS,
> and make NO calls, then the LED of Hard disk alway blink meaning it always
> busy... below is my hardware configuration, would someone can help how to
> improve the performance? thanks in advance.
> i tried to increased the nonce-ttl to 600 from 60 tring to decrease the
> times of writing db. however it doesn't make help...
> attached is the internal.xml for reference.
>
>
> Windows server 2008 64bit.
> CPU: Intel Core2 Duo L7400 with Intel 310 1.5GHz,
> Memory: 2G,
> bit: 64bit,
> HD: 160G
>
> P.S. i can think about RAMDISK, however, I'm afraid of it is so stable...
> if it can be increase the performance by HD itself, it will be great.
>
> Regards,
> Charles
>
> !DSPAM:4d91a2a832769362989175!
>


From ejay.greeves at yahoo.com  Tue Mar 29 15:09:57 2011
From: ejay.greeves at yahoo.com (Ejay Greeves)
Date: Tue, 29 Mar 2011 12:09:57 +0100 (BST)
Subject: [Freeswitch-users] recvEvent causes zombies process
Message-ID: <25112.37795.qm@web132301.mail.ird.yahoo.com>

I am wanting to code an inbound event program.
In order to see why I am getting zombie processes I stripped out all my code and used the minimum skeletcon code to do a test.
con = ESL::ESLconnection.new('127.0.0.1', '8021', 'ClueCon')con.events("plain", "CHANNEL_HANGUP_COMPLETE")
loop do	event = con.recvEvent		pid = fork do		
	end
	Process.detach(pid)	end

This code does nothing but wait for an incoming event, creates an fork then detach however it still produces a defunct process.
I think con.recvEvent is the cause?


      
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From fieldpeak at gmail.com  Tue Mar 29 15:10:51 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Tue, 29 Mar 2011 19:10:51 +0800
Subject: [Freeswitch-users] FS performance test -Hard disk alway busy
 when registerred 500 users
In-Reply-To: <AANLkTik8MryDa4Hbe9aPBYYjxMDXFzZk-RmDmAJTdL_5@mail.gmail.com>
References: <4d91a177.4323e70a.2cca.ffffefc8@mx.google.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B394B66A@cooper>
	<AANLkTik8MryDa4Hbe9aPBYYjxMDXFzZk-RmDmAJTdL_5@mail.gmail.com>
Message-ID: <AANLkTik10dzX1G2b-a48bkdrGQ64n8G5i66QcPo5-yOg@mail.gmail.com>

Hi Avi,

You're right. the registertation timing is 60 seconds before 3600
seconds, meaning send regisetertion to FS every 3510 seconds, so i
belive regiseration is not a issue...

i think the performance consumption coming from freqentely update
'nonce' value to table 'sip_authentication' in the
sofia_reg_internal.db, however, i already tired set the nonce-ttl to
600 seconds, it doensn't help....

now really don't know to what caused so mass HD I/O operation... but
see the resource monior of the OS, it surely the most one is the
writing sofia_reg_internal.db...

Regards,
Charles

2011/3/29, Avi Marcus <avi at avimarcus.net>:
> 500 registered users with an expiry of 3600 seconds/1 hour gives you only
> one request every 7 seconds. What kind of registration timing do you have on
> your phones?
> -Avi
>
> On Tue, Mar 29, 2011 at 11:26 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
>> The recommendation is to use a RAM disk if you see performance problems.
>> If
>> your disk is ?blinking? I wouldn?t say that it?s that much loaded though ?
>> in that case it should have a more or less steady light :) RAM disk
>> shouldn?t be a stability problem, and if you reboot your computer, FS will
>> re-create all databases needed.
>>
>> Also make sure to use latest GIT, there have been some changes how the
>> sqlite db handles are cached internally.
>>
>> /Peter
>>
>> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Charles
>> *Skickat:* den 29 mars 2011 11:08
>> *Till:* freeswitch-users
>> *?mne:* [Freeswitch-users] FS performance test -Hard disk alway busy when
>> registerred 500 users
>>
>>
>>
>> I'm makeing the load test of the FreeSWITCH, I registerred 500 users to
>> FS, and
>> make NO calls,
>> then the LED of Hard disk alway blink meaning it always busy... below is
>> my hardware configuration, would someone can help how to improve the
>> performance? thanks in advance.
>>
>>
>> i tried to increased the nonce-ttl to 600 from 60 tring to decrease the
>> times of writing db. however it doesn't make help...
>>
>> attached is the internal.xml for reference.
>>
>>
>>
>>
>>
>> Windows server 2008 64bit.
>>
>> CPU: Intel Core2 Duo L7400 with Intel 310 1.5GHz,
>>
>> Memory: 2G,
>>
>> bit: 64bit,
>>
>> HD: 160G
>>
>>
>>
>> P.S. i can think about RAMDISK, however, I?m afraid of it is so stable...
>> if it can be increase the performance by HD itself, it will be great.
>>
>>
>>
>> Regards,
>>
>> Charles
>>
>>
>>
>> !DSPAM:4d91a2a832769362989175!
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>


From ejay.greeves at yahoo.com  Tue Mar 29 15:42:21 2011
From: ejay.greeves at yahoo.com (Ejay Greeves)
Date: Tue, 29 Mar 2011 12:42:21 +0100 (BST)
Subject: [Freeswitch-users] phones for gsmopen
Message-ID: <452327.64364.qm@web132308.mail.ird.yahoo.com>


Is there any current phones that work with gsmopen. What would I need to look for in a phone to find out if the phone will be supported


      
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From gmaruzz at gmail.com  Tue Mar 29 15:48:54 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 29 Mar 2011 13:48:54 +0200
Subject: [Freeswitch-users] phones for gsmopen
In-Reply-To: <452327.64364.qm@web132308.mail.ird.yahoo.com>
References: <452327.64364.qm@web132308.mail.ird.yahoo.com>
Message-ID: <AANLkTik_n1+-WZ4E-ykOzno5aZi-JXnDkmug_XTtqSzY@mail.gmail.com>

1) modem recognized by linux, with AT command support
2) if you want voice operation, you will need to make an audiocable that
goes from the cellphone headset jack to the mic+spk of the soundcard
3) without audiocable you will only have SMS send/receive capabilities, not
voice calls
4) cellphones made by motorola, nokia, ericsson are known to work
5) if you want a ready made "black box", known to work, that contains
cellphone+modem+soundcard and that do not need any special cable (just one
only standard USB cable) you may want to look into www.mobigater.com

-giovanni

On Tue, Mar 29, 2011 at 1:42 PM, Ejay Greeves <ejay.greeves at yahoo.com>wrote:

>
> Is there any current phones that work with gsmopen. What would I need to
> look for in a phone to find out if the phone will be supported
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From peter.olsson at visionutveckling.se  Tue Mar 29 16:16:26 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 29 Mar 2011 14:16:26 +0200
Subject: [Freeswitch-users] FS performance test -Hard disk alway busy
 when registerred 500 users
In-Reply-To: <AANLkTimRHZUdrg+aeBYSyfwuA-r0Ee7ry4WTbLedq4VD@mail.gmail.com>
References: <4d91a177.4323e70a.2cca.ffffefc8@mx.google.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B394B66A@cooper>
	<AANLkTimRHZUdrg+aeBYSyfwuA-r0Ee7ry4WTbLedq4VD@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B394B78D@cooper>

Ahh - I didn't read you were on Windows.. :)

Are you using latest git though - there has been lots of rewrites last month to handle cached sqlite handles better.

/Peter

-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r fieldpeak
Skickat: den 29 mars 2011 13:03
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] FS performance test -Hard disk alway busy when registerred 500 users

Hi Peter,

Thanks for your prompt reponse and recommendation.

I ever tried two ramdisk softwares (VSuite and DataRAM free edition),
both work. Can you advise which one is preferred using in the
production? or anyother recomended product running in windows
platform?

RAMDisk is a 'program' after all, i'm afraid whether it can be loaded
prior to freeswitch service?

if using ramdisk, we have to seperatedly install RAMdisk with FS
package in the filed, it maybe involve work comparing one installation
pakage...

is there any windows native support Ramdisk like the one in Linux?

sorry for coming up so many question :)

Regards,
Charles

2011/3/29, Peter Olsson <peter.olsson at visionutveckling.se>:
> The recommendation is to use a RAM disk if you see performance problems. If
> your disk is "blinking" I wouldn't say that it's that much loaded though -
> in that case it should have a more or less steady light :) RAM disk
> shouldn't be a stability problem, and if you reboot your computer, FS will
> re-create all databases needed.
> Also make sure to use latest GIT, there have been some changes how the
> sqlite db handles are cached internally.
> /Peter
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Charles
> Skickat: den 29 mars 2011 11:08
> Till: freeswitch-users
> ?mne: [Freeswitch-users] FS performance test -Hard disk alway busy when
> registerred 500 users
>
> I'm makeing the load test of the FreeSWITCH, I registerred 500 users to FS,
> and make NO calls, then the LED of Hard disk alway blink meaning it always
> busy... below is my hardware configuration, would someone can help how to
> improve the performance? thanks in advance.
> i tried to increased the nonce-ttl to 600 from 60 tring to decrease the
> times of writing db. however it doesn't make help...
> attached is the internal.xml for reference.
>
>
> Windows server 2008 64bit.
> CPU: Intel Core2 Duo L7400 with Intel 310 1.5GHz,
> Memory: 2G,
> bit: 64bit,
> HD: 160G
>
> P.S. i can think about RAMDISK, however, I'm afraid of it is so stable...
> if it can be increase the performance by HD itself, it will be great.
>
> Regards,
> Charles
>
> 
>

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

!DSPAM:4d91bd9932761987414457!



From fieldpeak at gmail.com  Tue Mar 29 16:25:18 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Tue, 29 Mar 2011 20:25:18 +0800
Subject: [Freeswitch-users] FS performance test -Hard disk alway busy
 when registerred 500 users
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B394B78D@cooper>
References: <4d91a177.4323e70a.2cca.ffffefc8@mx.google.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B394B66A@cooper>
	<AANLkTimRHZUdrg+aeBYSyfwuA-r0Ee7ry4WTbLedq4VD@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B394B78D@cooper>
Message-ID: <AANLkTim8b+rGjCxah1HTD9w30y_p1aYB9EkcNS6EhCd-@mail.gmail.com>

Not yet, i will try the latest version tomorrow.  wish it will be better :)

? 2011-3-29 ??8:17?"Peter Olsson" <peter.olsson at visionutveckling.se>???

Ahh - I didn't read you were on Windows.. :)

Are you using latest git though - there has been lots of rewrites last month
to handle cached sqlite handles better.

/Peter

-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] F?r fieldpeak
Skickat: den 29 mars 2011 13:03
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] FS performance test -Hard disk alway busy when
registerred 500 users


Hi Peter,

Thanks for your prompt reponse and recommendation.

I ever tried two ramdisk softwares (...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists...
!DSPAM:4d91bd9932761987414457!



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lis...
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From ochere at gmail.com  Tue Mar 29 13:06:48 2011
From: ochere at gmail.com (Frank Ochere)
Date: Tue, 29 Mar 2011 12:06:48 +0300
Subject: [Freeswitch-users] need help on : kenyan pri line setting
In-Reply-To: <AANLkTi=8KVWALfs80Y6W6+CsPRx6NLY7vTPdf9VX5xD3@mail.gmail.com>
References: <AANLkTi=8KVWALfs80Y6W6+CsPRx6NLY7vTPdf9VX5xD3@mail.gmail.com>
Message-ID: <AANLkTi=_xPtM8BSUO90jjGgdX9CtHfufgqDwZaHO2g7i@mail.gmail.com>

Venkat,

Pri Type EuroISDN
CRC4 Enabled
Coding HDB3
Framing Type CCS
Timing Source set to Incoming
Bearer Channels 1-15, 17-31
Data Channel 16

Regards

Frank


On Tue, Mar 29, 2011 at 11:52 AM, ovvenkat <ovvenkatesan at gmail.com> wrote:

> Hi,
>
> Any one please tell me,
> E1 line setting for kenyan E1 line.
>
> Which *E1 Line coding* and *framing* I have to use it for kenyan E1 line.
>
> My current setting works fine in India.
>
> When I am trying to connect to kenyan E1 line,
> wanrouter status,  says,
>
> Device name | Protocol | Station | Status        |
> wanpipe1    | AFT TE1  | N/A     | Disconnected  |
>
>
>
>  I know, its not related to freeswitch.
> But, I know no anyother place to get
> help on this. If anyone done this before
> in kenyan network please help me.
>
>
> Regards,
> Venkat.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From duncan.pierce at gmail.com  Tue Mar 29 15:17:17 2011
From: duncan.pierce at gmail.com (Duncan Pierce)
Date: Tue, 29 Mar 2011 12:17:17 +0100
Subject: [Freeswitch-users] How to set PIN after conference created?
Message-ID: <AANLkTimMvsofsbCMUjSRAzNEBLZiba3buEb5j=yRdDn+@mail.gmail.com>

Hi everyone,

I'm new to FreeSWITCH. I've managed to set up many features quite easily but
I'm struggling with mod_conference. I want to allow a user to create a
conference and set its PIN by dialling a special number
81<conference_number>*<pin>:

<extension name="initiate_conference">
<condition field="destination_number" expression="^81(\d*)\*(\d*)$">
<action application="log" data="INFO Creating conference $1 with PIN $2"/>
<action application="answer"/>
<action application="conference" data="$1 at default+$2"/>
</condition>
</extension>

Other users can dial into the same conference using a different access
number, but are challenged for the PIN set by the first user:

<extension name="join_conference">
<condition field="destination_number" expression="^82(\d*)$">
<action application="answer"/>
<action application="conference" data="$1"/>
</condition>
</extension>

There is no <param name="pin" ...> set in <configuration
name="conference.conf"><profiles><profile name="default">.

I was expecting <action application="conference" data="$1 at default+$2"/> to
create the conference and set its PIN, so that a subsequent <action
application="conference" data="$1"/> would be challenged to provide the same
PIN. Instead what happens is the first caller has the required PIN set for
their call, but the second caller is not challenged to provide the PIN at
all.

Looking at the code in mod_conference.c on git checkout v1.0.6 I see there
only 2 ways conference->pin can be set: via <param name="pin" .../> in the
profile or via API 'conference $1 pin $2'. I was expecting there would be a
third way: assigning conference->pin = dpin when creating a new conference.
Although I find this a bit confusing it could be by design.

I tried using the API to set the PIN after the conference has been created
but I lack the knowledge to do this successfully. I've tried:

* <action application="set" data="result=${conference($1 pin $2)}"/> after
<action application="conference" data="$1 at default+$2"/> - no actions execute
after the <action application="conference">
* <action application="set" data="result=${conference($1 pin $2)}"/> before
<action application="conference" data="$1 at default+$2"/> - unsurprisingly, it
reports conference not found
* <action application="export" data="nolocal:api_on_answer=conference $1 pin
$2"/> - appears not to execute but it's difficult to be sure in this case

Thanks in advance for any help you can provide!

Best regards,
Duncan
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From curriegrad2004 at gmail.com  Tue Mar 29 17:42:38 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 29 Mar 2011 06:42:38 -0700
Subject: [Freeswitch-users] FS performance test -Hard disk alway busy
 when registerred 500 users
In-Reply-To: <AANLkTim8b+rGjCxah1HTD9w30y_p1aYB9EkcNS6EhCd-@mail.gmail.com>
References: <4d91a177.4323e70a.2cca.ffffefc8@mx.google.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B394B66A@cooper>
	<AANLkTimRHZUdrg+aeBYSyfwuA-r0Ee7ry4WTbLedq4VD@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B394B78D@cooper>
	<AANLkTim8b+rGjCxah1HTD9w30y_p1aYB9EkcNS6EhCd-@mail.gmail.com>
Message-ID: <AANLkTi=mETV2-jMJiv-_KvzJ4q87OcOCdz1COmE86pVr@mail.gmail.com>

I would also suggest that you not use Windows under this situation.
>From what I experienced, running FreeSwitch under Windows does have a
slight performance penalty. Running it under CentOS 5.x or the newer
RHEL 6.x series does have a significant improvement over running the
softswitch under Windows.

On Tue, Mar 29, 2011 at 5:25 AM, fieldpeak <fieldpeak at gmail.com> wrote:
> Not yet, i will try the latest version tomorrow.? wish it will be better :)
>
> ? 2011-3-29 ??8:17?"Peter Olsson" <peter.olsson at visionutveckling.se>???
>
> Ahh - I didn't read you were on Windows.. :)
>
> Are you using latest git though - there has been lots of rewrites last month
> to handle cached sqlite handles better.
>
> /Peter
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r fieldpeak
> Skickat: den 29 mars 2011 13:03
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] FS performance test -Hard disk alway busy when
> registerred 500 users
>
> Hi Peter,
>
> Thanks for your prompt reponse and recommendation.
>
> I ever tried two ramdisk softwares (...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists...
>
> !DSPAM:4d91bd9932761987414457!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lis...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From ovvenkatesan at gmail.com  Tue Mar 29 18:24:28 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Tue, 29 Mar 2011 17:24:28 +0300
Subject: [Freeswitch-users] track of number of calls
Message-ID: <AANLkTinsrFqZGYgfsMS8Ocw2WKGzy5wpCJkGCt+6CMZ-@mail.gmail.com>

Hi to all,

Can you anyone please tell,
I wanted to check number of calls answered,
before answer the new calls.




-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
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From avi at avimarcus.net  Tue Mar 29 18:42:39 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 29 Mar 2011 16:42:39 +0200
Subject: [Freeswitch-users] track of number of calls
In-Reply-To: <AANLkTinsrFqZGYgfsMS8Ocw2WKGzy5wpCJkGCt+6CMZ-@mail.gmail.com>
References: <AANLkTinsrFqZGYgfsMS8Ocw2WKGzy5wpCJkGCt+6CMZ-@mail.gmail.com>
Message-ID: <AANLkTi=0oVanX-PdZKTXaPfGH3_CcWZmbKKHmebbGK9m@mail.gmail.com>

Do you want to know how many calls have been answered in the past since
freeswitch started up, or how many are currently on - e.g. to limit the
channels in use?
If you want the second, take a look at:
http://wiki.freeswitch.org/wiki/Limit

<http://wiki.freeswitch.org/wiki/Limit>-Avi


On Tue, Mar 29, 2011 at 4:24 PM, ovvenkat <ovvenkatesan at gmail.com> wrote:

> Hi to all,
>
> Can you anyone please tell,
> I wanted to check number of calls answered,
> before answer the new calls.
>
>
>
>
> --
>
> If you have come to help me, you are wasting your time.
> If you have come to because your liberation is bound up in mine, we can
> work together.
>
>
> Regards
> Venkatesan OV.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Tue Mar 29 18:52:47 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 29 Mar 2011 09:52:47 -0500
Subject: [Freeswitch-users] recvEvent causes zombies process
In-Reply-To: <25112.37795.qm@web132301.mail.ird.yahoo.com>
References: <25112.37795.qm@web132301.mail.ird.yahoo.com>
Message-ID: <AANLkTim4a3uh6O1qMBQ=GGc5=HmEwK4P_H7E6ZE7DQtO@mail.gmail.com>

you are calling fork every time you receive an event.
you need to exit the process from the child and reap or ignore it on the
parent.


On Tue, Mar 29, 2011 at 6:09 AM, Ejay Greeves <ejay.greeves at yahoo.com>wrote:

> I am wanting to code an inbound event program.
>
> In order to see why I am getting zombie processes I stripped out all my
> code and used the minimum skeletcon code to do a test.
>
> con = ESL::ESLconnection.new('127.0.0.1', '8021', 'ClueCon')
> con.events("plain", "CHANNEL_HANGUP_COMPLETE")
>
> loop do
> event = con.recvEvent
>  pid = fork do
>
> end
>
> Process.detach(pid)
> end
>
>
> This code does nothing but wait for an incoming event, creates an fork then
> detach however it still produces a defunct process.
>
> I think con.recvEvent is the cause?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From ovvenkatesan at gmail.com  Tue Mar 29 18:55:23 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Tue, 29 Mar 2011 17:55:23 +0300
Subject: [Freeswitch-users] track of number of calls
In-Reply-To: <AANLkTi=0oVanX-PdZKTXaPfGH3_CcWZmbKKHmebbGK9m@mail.gmail.com>
References: <AANLkTinsrFqZGYgfsMS8Ocw2WKGzy5wpCJkGCt+6CMZ-@mail.gmail.com>
	<AANLkTi=0oVanX-PdZKTXaPfGH3_CcWZmbKKHmebbGK9m@mail.gmail.com>
Message-ID: <AANLkTin1-cP7skuAfncUK0-i9jZpbUj6qeaZGqriVF_b@mail.gmail.com>

Hi Avi,

Thanks for your response.

Let me tell you the scenario call connection

one E1 line consist of 30 channels.

users will be calling this number to access to the IVR.

Here, 10 chennels are reserved for premium customers and

remaining channels for normal users.

so, I need to block the normal users ,
if already 20 normal users are currently accessing the IVR.

Regards.
Venkat.


On Tue, Mar 29, 2011 at 5:42 PM, Avi Marcus <avi at avimarcus.net> wrote:

> Do you want to know how many calls have been answered in the past since
> freeswitch started up, or how many are currently on - e.g. to limit the
> channels in use?
> If you want the second, take a look at:
> http://wiki.freeswitch.org/wiki/Limit
>
> <http://wiki.freeswitch.org/wiki/Limit>-Avi
>
>
> On Tue, Mar 29, 2011 at 4:24 PM, ovvenkat <ovvenkatesan at gmail.com> wrote:
>
>> Hi to all,
>>
>> Can you anyone please tell,
>> I wanted to check number of calls answered,
>> before answer the new calls.
>>
>>
>>
>>
>> --
>>
>> If you have come to help me, you are wasting your time.
>> If you have come to because your liberation is bound up in mine, we can
>> work together.
>>
>>
>> Regards
>> Venkatesan OV.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
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From Lars.Bobka at web.de  Tue Mar 29 17:42:56 2011
From: Lars.Bobka at web.de (Lars Bobka)
Date: Tue, 29 Mar 2011 15:42:56 +0200 (CEST)
Subject: [Freeswitch-users] Disable MoH
Message-ID: <1180382136.4187907.1301406176465.JavaMail.fmail@mwmweb036>


 Hi,
 
I want to disable MoH for all profiles.
So I disabled moh in the internam and external.xml:
 
<param name="disable-hold" value="true"/>
 
When I restart and reloadxml from the internal or external profile, I can also see that the MoH is disables:
 
2011-03-29 15:17:56.442811 [DEBUG] sofia.c:2755 disable-hold [true]
 
But when I get call over a gateway and transfer it to the default 1000 and the 1000 makes the MoH Request, MoH is played for the Caller
 
 
########################
2011-03-29 15:20:36.365954 [DEBUG] sofia.c:4153 Channel sofia/internal/sip:1000 at 213.148.128.70:16348 entering state [received][100]
2011-03-29 15:20:36.365954 [DEBUG] sofia.c:4164 Remote SDP:
v=0
o=- 3013619149 4 IN IP4 192.168.4.76
s=SIPPER for PhonerLite
c=IN IP4 192.168.4.76
t=0 0
m=audio 5078 RTP/AVP 9 0 8 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendonly
 
2011-03-29 15:20:36.365954 [DEBUG] switch_core_session.c:859 Send signal sofia/internal/035 at 62.206.162.10 [BREAK]
2011-03-29 15:20:36.371840 [DEBUG] switch_core_session.c:641 Send signal sofia/internal/035 at 62.206.162.10 [BREAK]
2011-03-29 15:20:36.511515 [DEBUG] switch_ivr.c:551 sofia/internal/035 at 62.206.162.10 Command Execute playback(local_stream://moh) EXECUTE sofia/internal/035 at 62.206.162.10 playback(local_stream://moh)
2011-03-29 15:20:36.511515 [DEBUG] mod_local_stream.c:421 Opening Stream [moh/16000] 16000hz ######################
 
But I disabled it?!
Can you please help me?
 
Regards Lars


___________________________________________________________
Schon geh?rt? WEB.DE hat einen genialen Phishing-Filter in die
Toolbar eingebaut! http://produkte.web.de/go/toolbar


From anthony.minessale at gmail.com  Tue Mar 29 19:11:24 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 29 Mar 2011 10:11:24 -0500
Subject: [Freeswitch-users] Disable MoH
In-Reply-To: <1180382136.4187907.1301406176465.JavaMail.fmail@mwmweb036>
References: <1180382136.4187907.1301406176465.JavaMail.fmail@mwmweb036>
Message-ID: <AANLkTinEfYZW7_YKCJiQZdBLEKVaLeUQVJ7LAUFn+L6r@mail.gmail.com>

what you want to do is set the variable hold_music to "slience"
you can set it global in vars.xml or in your dialplan on a per-channel basis.


On Tue, Mar 29, 2011 at 8:42 AM, Lars Bobka <Lars.Bobka at web.de> wrote:
>
> ?Hi,
>
> I want to disable MoH for all profiles.
> So I disabled moh in the internam and external.xml:
>
> <param name="disable-hold" value="true"/>
>
> When I restart and reloadxml from the internal or external profile, I can also see that the MoH is disables:
>
> 2011-03-29 15:17:56.442811 [DEBUG] sofia.c:2755 disable-hold [true]
>
> But when I get call over a gateway and transfer it to the default 1000 and the 1000 makes the MoH Request, MoH is played for the Caller
>
>
> ########################
> 2011-03-29 15:20:36.365954 [DEBUG] sofia.c:4153 Channel sofia/internal/sip:1000 at 213.148.128.70:16348 entering state [received][100]
> 2011-03-29 15:20:36.365954 [DEBUG] sofia.c:4164 Remote SDP:
> v=0
> o=- 3013619149 4 IN IP4 192.168.4.76
> s=SIPPER for PhonerLite
> c=IN IP4 192.168.4.76
> t=0 0
> m=audio 5078 RTP/AVP 9 0 8 101
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=sendonly
>
> 2011-03-29 15:20:36.365954 [DEBUG] switch_core_session.c:859 Send signal sofia/internal/035 at 62.206.162.10 [BREAK]
> 2011-03-29 15:20:36.371840 [DEBUG] switch_core_session.c:641 Send signal sofia/internal/035 at 62.206.162.10 [BREAK]
> 2011-03-29 15:20:36.511515 [DEBUG] switch_ivr.c:551 sofia/internal/035 at 62.206.162.10 Command Execute playback(local_stream://moh) EXECUTE sofia/internal/035 at 62.206.162.10 playback(local_stream://moh)
> 2011-03-29 15:20:36.511515 [DEBUG] mod_local_stream.c:421 Opening Stream [moh/16000] 16000hz ######################
>
> But I disabled it?!
> Can you please help me?
>
> Regards Lars
>
>
> ___________________________________________________________
> Schon geh?rt? WEB.DE hat einen genialen Phishing-Filter in die
> Toolbar eingebaut! http://produkte.web.de/go/toolbar
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 jeff at jefflenk.com  Tue Mar 29 19:23:19 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 29 Mar 2011 08:23:19 -0700 (PDT)
Subject: [Freeswitch-users] FS performance test -Hard disk alway busy
 when registerred 500 users
In-Reply-To: <AANLkTi=mETV2-jMJiv-_KvzJ4q87OcOCdz1COmE86pVr@mail.gmail.com>
References: <4d91a177.4323e70a.2cca.ffffefc8@mx.google.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B394B66A@cooper>
	<AANLkTimRHZUdrg+aeBYSyfwuA-r0Ee7ry4WTbLedq4VD@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B394B78D@cooper>
	<AANLkTim8b+rGjCxah1HTD9w30y_p1aYB9EkcNS6EhCd-@mail.gmail.com>
	<AANLkTi=mETV2-jMJiv-_KvzJ4q87OcOCdz1COmE86pVr@mail.gmail.com>
Message-ID: <1301412199027-6219631.post@n2.nabble.com>

FreeSWITCH builds as a native binary on Windows, there is no additional
overhead versus any other platform. All performance testing is very
subjective and highly dependant on the machine being tested and all of the
other related hardware and infrastructure.


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-performance-test-Hard-disk-alway-busy-when-registerred-500-users-tp6218312p6219631.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Tue Mar 29 19:24:20 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 29 Mar 2011 10:24:20 -0500
Subject: [Freeswitch-users] Build on solaris fails in libs/esl
In-Reply-To: <AANLkTim77j3aBU0YrR8YGZEp=bDf9ND3-T0NXSznW4ov@mail.gmail.com>
References: <AANLkTimGtXWdRPAv8OJYWeu8VYtCF8ZfQt0osGVgU7G=@mail.gmail.com>
	<AANLkTinab+WgVXUYtWaSccbwBV7G7LO-qjjcE4bYAKXj@mail.gmail.com>
	<AANLkTinAtxnnmbVm-tBvEuOxUQ7=z2JebB9TK+0WdBuF@mail.gmail.com>
	<AANLkTim77j3aBU0YrR8YGZEp=bDf9ND3-T0NXSznW4ov@mail.gmail.com>
Message-ID: <AANLkTikJ4bYiWcsU5uLjtV4pfcL_LV1bQbqUNsfAAZu0@mail.gmail.com>

If you send me a private email with login credentials and location of
the source, I can see about correcting it for you.
I don't currently have a box to test this on.


On Tue, Mar 29, 2011 at 12:27 AM, A E [Gmail] <all.eforums at gmail.com> wrote:
> On Mon, Mar 28, 2011 at 11:40 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> try latest git to see if the silly out of bound thing is fixed.
>>
>
> Hi Anthony,
>
> It seems to have been, yes. I don't see the these "integer overflow"
> messages, but now some other ones have been noticed:
>
> e.g.
> make: *** [current] Error 2"/home/freeswitch/fs/libs/esl/src/include/esl.h",
> line 465: warning: implicit function declaration: strcasecmp
> (E_NO_IMPLICIT_DECL_ALLOWED)
> "libs/esl/src/esl.c", line 1191: warning: implicit function declaration:
> strdup (E_NO_IMPLICIT_DECL_ALLOWED)
> "libs/esl/src/esl.c", line 1191: warning: improper pointer/integer
> combination: op "=" (E_BAD_PTR_INT_COMBINATION)
> "libs/esl/src/esl.c", line 1232: warning: improper pointer/integer
> combination: op "=" (E_BAD_PTR_INT_COMBINATION)
> "/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
> implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
> "/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
> implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
> "/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
> implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
> "/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
> implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
> "libs/esl/ivrd.c", line 98: warning: implicit function declaration: signal
> (E_NO_IMPLICIT_DECL_ALLOWED)
> "/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
> implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
> "/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
> implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
>
> As well as a load of these:
>
> -- "/home/freeswitch/fs/libs/spandsp/src/spandsp/bit_operations.h", line 50:
> warning: "__asm__" is an extension of ANSI C (E_KW_IS_AN_EXTENSION_OF_ANSI)
>
> And lastly,
>
> BTW, Not that you claimed that you fixed it, might as well tell you that,
> esl and mod_hash compilation still fails. Pastebin here:
> http://pastebin.freeswitch.org/15887
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Tue Mar 29 19:31:17 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 29 Mar 2011 10:31:17 -0500
Subject: [Freeswitch-users] bind_digit_action help
In-Reply-To: <4D91734C.8010208@tagnet.ru>
References: <4D91734C.8010208@tagnet.ru>
Message-ID: <AANLkTin9NegHMEOrCWyGZXka_mhsNdRsV3ZhegivmQQ9@mail.gmail.com>

Here are several choices:

These 2 vars on the A leg using the set app
<action application="set" data="bridge_pre_execute_bleg_app=bind_digit_action"/>
<action application="set" data="bridge_pre_execute_bleg_data=whatever"/>

This var with export app
<action application="export"
data="nolocal:execute_on_answer=bind_digit_action whatever"/>


These vars with export app
<action application="export"
data="nolocal:bridge_pre_execute_app=bind_digit_action"/>
<action application="export" data="nolocal:bridge_pre_execute_data=whatever"/>



or one of these vars in the dial string {}

{bridge_pre_execute_app=bind_digit_action,bridge_pre_execute_data='whatever'}
{execute_on_answer='bind_digit_action whatever'}




On Tue, Mar 29, 2011 at 12:51 AM, Boris Kovalenko <boris at tagnet.ru> wrote:
> Hello!
>
> ? ? Is this possible to get bind_digit_action work on B leg? I got it
> working on A leg and can't on B. Would someone provide a little example?
>
> --
> ? ?????????,
> ? ????? ?????????
> ? ??? "??????"
> ? (3435) 494991
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 michal.bielicki at seventhsignal.de  Tue Mar 29 19:33:43 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Tue, 29 Mar 2011 17:33:43 +0200
Subject: [Freeswitch-users] Build on solaris fails in libs/esl
In-Reply-To: <AANLkTikJ4bYiWcsU5uLjtV4pfcL_LV1bQbqUNsfAAZu0@mail.gmail.com>
References: <AANLkTimGtXWdRPAv8OJYWeu8VYtCF8ZfQt0osGVgU7G=@mail.gmail.com>
	<AANLkTinab+WgVXUYtWaSccbwBV7G7LO-qjjcE4bYAKXj@mail.gmail.com>
	<AANLkTinAtxnnmbVm-tBvEuOxUQ7=z2JebB9TK+0WdBuF@mail.gmail.com>
	<AANLkTim77j3aBU0YrR8YGZEp=bDf9ND3-T0NXSznW4ov@mail.gmail.com>
	<AANLkTikJ4bYiWcsU5uLjtV4pfcL_LV1bQbqUNsfAAZu0@mail.gmail.com>
Message-ID: <6ADAA4B6-802B-49F6-8C52-6BBCDE176F16@seventhsignal.de>

Its a problem with the build scripts for esl. looking at that right now ..

Am 29.03.2011 um 17:24 schrieb Anthony Minessale:

> If you send me a private email with login credentials and location of
> the source, I can see about correcting it for you.
> I don't currently have a box to test this on.
> 
> 
> On Tue, Mar 29, 2011 at 12:27 AM, A E [Gmail] <all.eforums at gmail.com> wrote:
>> On Mon, Mar 28, 2011 at 11:40 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>> 
>>> try latest git to see if the silly out of bound thing is fixed.
>>> 
>> 
>> Hi Anthony,
>> 
>> It seems to have been, yes. I don't see the these "integer overflow"
>> messages, but now some other ones have been noticed:
>> 
>> e.g.
>> make: *** [current] Error 2"/home/freeswitch/fs/libs/esl/src/include/esl.h",
>> line 465: warning: implicit function declaration: strcasecmp
>> (E_NO_IMPLICIT_DECL_ALLOWED)
>> "libs/esl/src/esl.c", line 1191: warning: implicit function declaration:
>> strdup (E_NO_IMPLICIT_DECL_ALLOWED)
>> "libs/esl/src/esl.c", line 1191: warning: improper pointer/integer
>> combination: op "=" (E_BAD_PTR_INT_COMBINATION)
>> "libs/esl/src/esl.c", line 1232: warning: improper pointer/integer
>> combination: op "=" (E_BAD_PTR_INT_COMBINATION)
>> "/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
>> implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
>> "/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
>> implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
>> "/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
>> implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
>> "/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
>> implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
>> "libs/esl/ivrd.c", line 98: warning: implicit function declaration: signal
>> (E_NO_IMPLICIT_DECL_ALLOWED)
>> "/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
>> implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
>> "/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
>> implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
>> 
>> As well as a load of these:
>> 
>> -- "/home/freeswitch/fs/libs/spandsp/src/spandsp/bit_operations.h", line 50:
>> warning: "__asm__" is an extension of ANSI C (E_KW_IS_AN_EXTENSION_OF_ANSI)
>> 
>> And lastly,
>> 
>> BTW, Not that you claimed that you fixed it, might as well tell you that,
>> esl and mod_hash compilation still fails. Pastebin here:
>> http://pastebin.freeswitch.org/15887
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de



From msc at freeswitch.org  Tue Mar 29 19:37:03 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 29 Mar 2011 08:37:03 -0700
Subject: [Freeswitch-users] How to set PIN after conference created?
In-Reply-To: <AANLkTimMvsofsbCMUjSRAzNEBLZiba3buEb5j=yRdDn+@mail.gmail.com>
References: <AANLkTimMvsofsbCMUjSRAzNEBLZiba3buEb5j=yRdDn+@mail.gmail.com>
Message-ID: <AANLkTimETc09FbB3M2D79ZfrFhAn12ac_p+4WNFEya2i@mail.gmail.com>

This is one of those cases where the devs tried very hard not to force
feed specific functionality onto the user. The bottom line here is
that the devs consider mod_conference's primary function to be that of
getting channels into a conference, muxing the audio, etc. Things like
PINs and security, etc. can all be handled in the dialplan. This is,
indeed, by design. Instead of forcing the security into mod_conference
itself we let the FS implementor decide how he/she wants to perform
security functions. One advantage to doing all the security in the
dialplan is that you have a lot of power and flexibility: you do
security however YOU feel is best.

I believe that if you start viewing PINs and security/privacy from the
perspective that they are functions of the dialplan instead of
functions of mod_conference then you will be able to solve your
problem much easier.

-MC

On Tue, Mar 29, 2011 at 4:17 AM, Duncan Pierce <duncan.pierce at gmail.com> wrote:
> Hi everyone,
> I'm new to FreeSWITCH. I've managed to set up many features quite easily but
> I'm struggling with mod_conference. I want to allow a user to create a
> conference and set its PIN by dialling a special number
> 81<conference_number>*<pin>:
> <extension name="initiate_conference">
> <condition field="destination_number" expression="^81(\d*)\*(\d*)$">
> <action application="log" data="INFO Creating conference $1 with PIN $2"/>
> <action application="answer"/>
> <action application="conference" data="$1 at default+$2"/>
> </condition>
> </extension>
> Other users can dial into the same conference using a different access
> number, but are challenged for the PIN set by the first user:
> <extension name="join_conference">
> <condition field="destination_number" expression="^82(\d*)$">
> <action application="answer"/>
> <action application="conference" data="$1"/>
> </condition>
> </extension>
> There is no <param name="pin" ...> set in <configuration
> name="conference.conf"><profiles><profile name="default">.
> I was expecting <action application="conference" data="$1 at default+$2"/> to
> create the conference and set its PIN, so that a subsequent <action
> application="conference" data="$1"/> would be challenged to provide the same
> PIN. Instead what happens is the first caller has the required PIN set for
> their call, but the second caller is not challenged to provide the PIN at
> all.
> Looking at the code in mod_conference.c on git checkout v1.0.6 I see there
> only 2 ways conference->pin can be set: via <param name="pin" .../> in the
> profile or via API 'conference $1 pin $2'. I was expecting there would be a
> third way: assigning conference->pin = dpin when creating a new conference.
> Although I find this a bit confusing it could be by design.
> I tried using the API to set the PIN after the conference has been created
> but I lack the knowledge to do this successfully. I've tried:
> * <action application="set" data="result=${conference($1 pin $2)}"/> after
> <action application="conference" data="$1 at default+$2"/> - no actions execute
> after the <action application="conference">
> * <action application="set" data="result=${conference($1 pin $2)}"/> before
> <action application="conference" data="$1 at default+$2"/> - unsurprisingly, it
> reports conference not found
> * <action application="export" data="nolocal:api_on_answer=conference $1 pin
> $2"/> - appears not to execute but it's difficult to be sure in this case
> Thanks in advance for any help you can provide!
> Best regards,
> Duncan
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Tue Mar 29 19:44:13 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 29 Mar 2011 08:44:13 -0700
Subject: [Freeswitch-users] bind_digit_action help
In-Reply-To: <AANLkTin9NegHMEOrCWyGZXka_mhsNdRsV3ZhegivmQQ9@mail.gmail.com>
References: <4D91734C.8010208@tagnet.ru>
	<AANLkTin9NegHMEOrCWyGZXka_mhsNdRsV3ZhegivmQQ9@mail.gmail.com>
Message-ID: <AANLkTi=aidD4PFjdj+gQ8F-aE_mgPGM9eALXw8xhEAqb@mail.gmail.com>

FYI,

I added this to the bind_digit_action wiki page:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_digit_action#Setting_on_the_B_leg

If anyone has more specific information and/or examples please add
them to that section.

Thanks,
MC

On Tue, Mar 29, 2011 at 8:31 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Here are several choices:
>
> These 2 vars on the A leg using the set app
> <action application="set" data="bridge_pre_execute_bleg_app=bind_digit_action"/>
> <action application="set" data="bridge_pre_execute_bleg_data=whatever"/>
>
> This var with export app
> <action application="export"
> data="nolocal:execute_on_answer=bind_digit_action whatever"/>
>
>
> These vars with export app
> <action application="export"
> data="nolocal:bridge_pre_execute_app=bind_digit_action"/>
> <action application="export" data="nolocal:bridge_pre_execute_data=whatever"/>
>
>
>
> or one of these vars in the dial string {}
>
> {bridge_pre_execute_app=bind_digit_action,bridge_pre_execute_data='whatever'}
> {execute_on_answer='bind_digit_action whatever'}
>
>
>
>
> On Tue, Mar 29, 2011 at 12:51 AM, Boris Kovalenko <boris at tagnet.ru> wrote:
>> Hello!
>>
>> ? ? Is this possible to get bind_digit_action work on B leg? I got it
>> working on A leg and can't on B. Would someone provide a little example?
>>
>> --
>> ? ?????????,
>> ? ????? ?????????
>> ? ??? "??????"
>> ? (3435) 494991
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From jpatten at co.brazos.tx.us  Tue Mar 29 19:57:53 2011
From: jpatten at co.brazos.tx.us (Josh M. Patten)
Date: Tue, 29 Mar 2011 15:57:53 +0000
Subject: [Freeswitch-users] FreeSWITCH forcing 8000Hz??
Message-ID: <8C8A3D4965236A42BDFF1758727F049A26310C@ITEX1.bc.local>

When attempting to utilize the conference service in FreeSWITCH I am unable to get FreeSWITCH to allow any wideband codecs. From fs_cli I see the SDP coming in from a counterpath softphone with SPEEX/16000 as one of the codecs:


2011-03-29 10:31:31.362549 [DEBUG] sofia.c:4675 Remote SDP:
v=0
o=- 12945886409318361 1 IN IP4 10.200.18.105
s=CounterPath X-Lite 4.0
c=IN IP4 10.200.18.105
t=0 0
a=ice-ufrag:76ed8b
a=ice-pwd:d3ed6f0a9947e9798cf0b0ee04d70777
m=audio 57398 RTP/AVP 100 101
a=rtpmap:100 SPEEX/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=candidate:1 1 UDP 659136 10.200.18.105 57398 typ host
a=candidate:1 2 UDP 659134 10.200.18.105 57399 typ host

But then FreeSWITCH forces the codec down to 8000Hz:

2011-03-29 10:31:31.363546 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [SPEEX:100:16000:20:0]/[SPEEX:99:8000:20:24600]
2011-03-29 10:31:31.363546 [DEBUG] sofia_glue.c:4520 Bah HUMBUG! Sticking with SPEEX at 8000h@20i


>From what I can tell I have allowed 16000Hz sample rates and codecs everywhere they can be set, including vars.xml, conference.conf.xml, my sofia profile, local_stream.conf.xml...

Any help would be appreciated. Thanks!



Josh Patten
Brazos County Network Engineer
979.361.4676

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From max.clark at gmail.com  Tue Mar 29 20:54:35 2011
From: max.clark at gmail.com (Max Clark)
Date: Tue, 29 Mar 2011 09:54:35 -0700
Subject: [Freeswitch-users] PRI Test Equipment
Message-ID: <AANLkTi=txKH5Yj3MHaMrX1uHhnqMpuS=PuLgGsxpR=H8@mail.gmail.com>

I'm looking for an appliance that can be plugged into a PRI connected
to a PBX and either display or read back the digits that are passed to
it. Before I build something to simulate this, is there anything out
there commercially that can be purchased?

Thanks,
Max


From msc at freeswitch.org  Tue Mar 29 21:02:43 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 29 Mar 2011 10:02:43 -0700
Subject: [Freeswitch-users] PRI Test Equipment
In-Reply-To: <AANLkTi=txKH5Yj3MHaMrX1uHhnqMpuS=PuLgGsxpR=H8@mail.gmail.com>
References: <AANLkTi=txKH5Yj3MHaMrX1uHhnqMpuS=PuLgGsxpR=H8@mail.gmail.com>
Message-ID: <AANLkTi=emvxczkxopd4+OYLdY7gJjyd7qgONO8sK-PKT@mail.gmail.com>

depends on what all you want the appliance to do. if you just want to
do basic testing then you can buy or rent a "T-BERD" (google it) or
something similar.

-MC

On Tue, Mar 29, 2011 at 9:54 AM, Max Clark <max.clark at gmail.com> wrote:
> I'm looking for an appliance that can be plugged into a PRI connected
> to a PBX and either display or read back the digits that are passed to
> it. Before I build something to simulate this, is there anything out
> there commercially that can be purchased?
>
> Thanks,
> Max
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From ayhkor at gmail.com  Tue Mar 29 21:39:40 2011
From: ayhkor at gmail.com (deniro)
Date: Tue, 29 Mar 2011 13:39:40 -0400
Subject: [Freeswitch-users] list conference name?
Message-ID: <BANLkTik9mEqR9UL5u+NpS5d8r7iVH5f9XA@mail.gmail.com>

Hi,
I have a web +voice conferencing  and trying below

$list=$api->executeString("conference <conf_name> list");

What determines the conference name?
is it voicebridge number or meeting id or  attendee PIN or moderator PIN

I am unable to determine conference name and list it accordingly (performed
many trials)

is there any way to get the conference name by a command like above?
what is the best way to get the conference name and list it and execute
accordingly

thx
deniro--
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From max.clark at gmail.com  Tue Mar 29 21:47:09 2011
From: max.clark at gmail.com (Max Clark)
Date: Tue, 29 Mar 2011 10:47:09 -0700
Subject: [Freeswitch-users] PRI Test Equipment
In-Reply-To: <AANLkTi=emvxczkxopd4+OYLdY7gJjyd7qgONO8sK-PKT@mail.gmail.com>
References: <AANLkTi=txKH5Yj3MHaMrX1uHhnqMpuS=PuLgGsxpR=H8@mail.gmail.com>
	<AANLkTi=emvxczkxopd4+OYLdY7gJjyd7qgONO8sK-PKT@mail.gmail.com>
Message-ID: <AANLkTinexeb64gsV_ih8fwub1F=K8V+fKEJtEPUeWuwQ@mail.gmail.com>

I want to display/read back the digits passed on the PRI.

On Tue, Mar 29, 2011 at 10:02 AM, Michael Collins <msc at freeswitch.org> wrote:
> depends on what all you want the appliance to do. if you just want to
> do basic testing then you can buy or rent a "T-BERD" (google it) or
> something similar.
>
> -MC
>
> On Tue, Mar 29, 2011 at 9:54 AM, Max Clark <max.clark at gmail.com> wrote:
>> I'm looking for an appliance that can be plugged into a PRI connected
>> to a PBX and either display or read back the digits that are passed to
>> it. Before I build something to simulate this, is there anything out
>> there commercially that can be purchased?
>>
>> Thanks,
>> Max
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From max.clark at gmail.com  Tue Mar 29 21:49:14 2011
From: max.clark at gmail.com (Max Clark)
Date: Tue, 29 Mar 2011 10:49:14 -0700
Subject: [Freeswitch-users] "Free" Conference Calling
Message-ID: <AANLkTikK8zUBnPtBX4n+4S3dYn+_VPiNJieSLzO7Vuhi@mail.gmail.com>

Hello,

I've noticed a consistent pattern for SIP Termination providers not
completing calls to the "Free" Conference lines due to costs. What's
the best way to deal with this? Are there published lists of these
providers numbers that I can use to influence my LCR? Are there SIP
Termination providers that explicitly deal with these lines?

Thanks,
Max


From peter.olsson at visionutveckling.se  Tue Mar 29 21:50:50 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 29 Mar 2011 19:50:50 +0200
Subject: [Freeswitch-users] list conference name?
In-Reply-To: <BANLkTik9mEqR9UL5u+NpS5d8r7iVH5f9XA@mail.gmail.com>
References: <BANLkTik9mEqR9UL5u+NpS5d8r7iVH5f9XA@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B394B8E0@cooper>

Just do a "conference list" and you will get all conferences, with their members. You will also see the name in there.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r deniro
Skickat: den 29 mars 2011 19:40
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] list conference name?

Hi,
I have a web +voice conferencing  and trying below

$list=$api->executeString("conference <conf_name> list");

What determines the conference name?
is it voicebridge number or meeting id or  attendee PIN or moderator PIN

I am unable to determine conference name and list it accordingly (performed many trials)

is there any way to get the conference name by a command like above?
what is the best way to get the conference name and list it and execute accordingly

thx
deniro--


!DSPAM:4d921a7332761434317129!
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From sos at sokhapkin.dyndns.org  Tue Mar 29 22:00:04 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 29 Mar 2011 14:00:04 -0400
Subject: [Freeswitch-users] "Free" Conference Calling
In-Reply-To: <AANLkTikK8zUBnPtBX4n+4S3dYn+_VPiNJieSLzO7Vuhi@mail.gmail.com>
References: <AANLkTikK8zUBnPtBX4n+4S3dYn+_VPiNJieSLzO7Vuhi@mail.gmail.com>
Message-ID: <201103291400.04549.sos@sokhapkin.dyndns.org>

Pay as you go providers do not block those numbers, but providers with a fixed 
monthly cost do.

On Tuesday 29 March 2011 13:49:14 Max Clark wrote:
> Hello,
> 
> I've noticed a consistent pattern for SIP Termination providers not
> completing calls to the "Free" Conference lines due to costs. What's
> the best way to deal with this? Are there published lists of these
> providers numbers that I can use to influence my LCR? Are there SIP
> Termination providers that explicitly deal with these lines?
> 
> Thanks,
> Max
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From anthony.minessale at gmail.com  Tue Mar 29 22:03:12 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 29 Mar 2011 13:03:12 -0500
Subject: [Freeswitch-users] FreeSWITCH forcing 8000Hz??
In-Reply-To: <8C8A3D4965236A42BDFF1758727F049A26310C@ITEX1.bc.local>
References: <8C8A3D4965236A42BDFF1758727F049A26310C@ITEX1.bc.local>
Message-ID: <AANLkTimo9aK1xxN4pPYr6uh-=r5T5Ni+YJKns0nyxzQe@mail.gmail.com>

you seem to have codec negotiation set to scrooge for some reason
which is a special work-around option that should not be generally
used.


On Tue, Mar 29, 2011 at 10:57 AM, Josh M. Patten
<jpatten at co.brazos.tx.us> wrote:
> When attempting to utilize the conference service in FreeSWITCH I am unable
> to get FreeSWITCH to allow any wideband codecs. From fs_cli I see the SDP
> coming in from a counterpath softphone with SPEEX/16000 as one of the
> codecs:
>
>
>
>
>
> 2011-03-29 10:31:31.362549 [DEBUG] sofia.c:4675 Remote SDP:
>
> v=0
>
> o=- 12945886409318361 1 IN IP4 10.200.18.105
>
> s=CounterPath X-Lite 4.0
>
> c=IN IP4 10.200.18.105
>
> t=0 0
>
> a=ice-ufrag:76ed8b
>
> a=ice-pwd:d3ed6f0a9947e9798cf0b0ee04d70777
>
> m=audio 57398 RTP/AVP 100 101
>
> a=rtpmap:100 SPEEX/16000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-15
>
> a=candidate:1 1 UDP 659136 10.200.18.105 57398 typ host
>
> a=candidate:1 2 UDP 659134 10.200.18.105 57399 typ host
>
>
>
> But then FreeSWITCH forces the codec down to 8000Hz:
>
>
>
> 2011-03-29 10:31:31.363546 [DEBUG] sofia_glue.c:4504 Audio Codec Compare
> [SPEEX:100:16000:20:0]/[SPEEX:99:8000:20:24600]
>
> 2011-03-29 10:31:31.363546 [DEBUG] sofia_glue.c:4520 Bah HUMBUG! Sticking
> with SPEEX at 8000h@20i
>
>
>
>
>
> From what I can tell I have allowed 16000Hz sample rates and codecs
> everywhere they can be set, including vars.xml, conference.conf.xml, my
> sofia profile, local_stream.conf.xml?
>
>
>
> Any help would be appreciated. Thanks!
>
>
>
>
>
>
>
> Josh Patten
>
> Brazos County Network Engineer
>
> 979.361.4676
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Tue Mar 29 22:05:22 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 29 Mar 2011 13:05:22 -0500
Subject: [Freeswitch-users] Build on solaris fails in libs/esl
In-Reply-To: <6ADAA4B6-802B-49F6-8C52-6BBCDE176F16@seventhsignal.de>
References: <AANLkTimGtXWdRPAv8OJYWeu8VYtCF8ZfQt0osGVgU7G=@mail.gmail.com>
	<AANLkTinab+WgVXUYtWaSccbwBV7G7LO-qjjcE4bYAKXj@mail.gmail.com>
	<AANLkTinAtxnnmbVm-tBvEuOxUQ7=z2JebB9TK+0WdBuF@mail.gmail.com>
	<AANLkTim77j3aBU0YrR8YGZEp=bDf9ND3-T0NXSznW4ov@mail.gmail.com>
	<AANLkTikJ4bYiWcsU5uLjtV4pfcL_LV1bQbqUNsfAAZu0@mail.gmail.com>
	<6ADAA4B6-802B-49F6-8C52-6BBCDE176F16@seventhsignal.de>
Message-ID: <AANLkTi=YpGBXoFNEw+rrKSBd-cGbDk_8MQuwMD_N=a7y@mail.gmail.com>

It looks to me like the biggest problem is they can't find the proto
for strdup and some other things that normally would have been in
string.h

On Tue, Mar 29, 2011 at 10:33 AM, Michal Bielicki
<michal.bielicki at seventhsignal.de> wrote:
> Its a problem with the build scripts for esl. looking at that right now ..
>
> Am 29.03.2011 um 17:24 schrieb Anthony Minessale:
>
>> If you send me a private email with login credentials and location of
>> the source, I can see about correcting it for you.
>> I don't currently have a box to test this on.
>>
>>
>> On Tue, Mar 29, 2011 at 12:27 AM, A E [Gmail] <all.eforums at gmail.com> wrote:
>>> On Mon, Mar 28, 2011 at 11:40 PM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>>>
>>>> try latest git to see if the silly out of bound thing is fixed.
>>>>
>>>
>>> Hi Anthony,
>>>
>>> It seems to have been, yes. I don't see the these "integer overflow"
>>> messages, but now some other ones have been noticed:
>>>
>>> e.g.
>>> make: *** [current] Error 2"/home/freeswitch/fs/libs/esl/src/include/esl.h",
>>> line 465: warning: implicit function declaration: strcasecmp
>>> (E_NO_IMPLICIT_DECL_ALLOWED)
>>> "libs/esl/src/esl.c", line 1191: warning: implicit function declaration:
>>> strdup (E_NO_IMPLICIT_DECL_ALLOWED)
>>> "libs/esl/src/esl.c", line 1191: warning: improper pointer/integer
>>> combination: op "=" (E_BAD_PTR_INT_COMBINATION)
>>> "libs/esl/src/esl.c", line 1232: warning: improper pointer/integer
>>> combination: op "=" (E_BAD_PTR_INT_COMBINATION)
>>> "/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
>>> implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
>>> "/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
>>> implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
>>> "/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
>>> implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
>>> "/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
>>> implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
>>> "libs/esl/ivrd.c", line 98: warning: implicit function declaration: signal
>>> (E_NO_IMPLICIT_DECL_ALLOWED)
>>> "/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
>>> implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
>>> "/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
>>> implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
>>>
>>> As well as a load of these:
>>>
>>> -- "/home/freeswitch/fs/libs/spandsp/src/spandsp/bit_operations.h", line 50:
>>> warning: "__asm__" is an extension of ANSI C (E_KW_IS_AN_EXTENSION_OF_ANSI)
>>>
>>> And lastly,
>>>
>>> BTW, Not that you claimed that you fixed it, might as well tell you that,
>>> esl and mod_hash compilation still fails. Pastebin here:
>>> http://pastebin.freeswitch.org/15887
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> Michal Bielicki
> Gesch?ftsf?hrer / CEO
>
> Seventh Signal Ltd. & Co. KG
> Weigandufer 45, B?ro 115, D-12059 Berlin
> Voice: +49 30 60988730
>
> Amtsgericht Charlottenburg HRA 44413 B
> Ust.-ID: DE266981999
> Gesch?ftsf?hrer: Michal Bielicki
> Pers?nlich Haftende Gesellschafterin:
> Seventh Signal Ltd, 69 Great Hampton St. Birmingham,
> B18 6EW, GB, Company Nr.: 06889439
> WWW.: http://www.seventhsignal.de
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 avi at avimarcus.net  Tue Mar 29 22:07:07 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 29 Mar 2011 20:07:07 +0200
Subject: [Freeswitch-users] "Free" Conference Calling
In-Reply-To: <AANLkTikK8zUBnPtBX4n+4S3dYn+_VPiNJieSLzO7Vuhi@mail.gmail.com>
References: <AANLkTikK8zUBnPtBX4n+4S3dYn+_VPiNJieSLzO7Vuhi@mail.gmail.com>
Message-ID: <AANLkTini7aKRL2mcV2LYsCPxEcUr+G+eg_93m8rPSyzP@mail.gmail.com>

grnvoip doesn't block them either, but does charge a premium for termination
to those numbers..
-Avi

On Tue, Mar 29, 2011 at 7:49 PM, Max Clark <max.clark at gmail.com> wrote:

> Hello,
>
> I've noticed a consistent pattern for SIP Termination providers not
> completing calls to the "Free" Conference lines due to costs. What's
> the best way to deal with this? Are there published lists of these
> providers numbers that I can use to influence my LCR? Are there SIP
> Termination providers that explicitly deal with these lines?
>
> Thanks,
> Max
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From max.clark at gmail.com  Tue Mar 29 22:07:36 2011
From: max.clark at gmail.com (Max Clark)
Date: Tue, 29 Mar 2011 11:07:36 -0700
Subject: [Freeswitch-users] "Free" Conference Calling
In-Reply-To: <201103291400.04549.sos@sokhapkin.dyndns.org>
References: <AANLkTikK8zUBnPtBX4n+4S3dYn+_VPiNJieSLzO7Vuhi@mail.gmail.com>
	<201103291400.04549.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTi=teqs=zVa8acRCMhm+7=MEAtZj-BiGrL4u_tyu@mail.gmail.com>

Sergey,

I agree that "pay as you go" providers should not block these number.
What I've found however is that they do (or I should clarify by saying
"wholesale" providers). I've contemplated provisioning a single PRI
for this purpose to the ILEC but I'm trying to find a better way.

-Max

On Tue, Mar 29, 2011 at 11:00 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org> wrote:
> Pay as you go providers do not block those numbers, but providers with a fixed
> monthly cost do.
>
> On Tuesday 29 March 2011 13:49:14 Max Clark wrote:
>> Hello,
>>
>> I've noticed a consistent pattern for SIP Termination providers not
>> completing calls to the "Free" Conference lines due to costs. What's
>> the best way to deal with this? Are there published lists of these
>> providers numbers that I can use to influence my LCR? Are there SIP
>> Termination providers that explicitly deal with these lines?
>>
>> Thanks,
>> Max
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From ayhkor at gmail.com  Tue Mar 29 22:12:03 2011
From: ayhkor at gmail.com (deniro)
Date: Tue, 29 Mar 2011 14:12:03 -0400
Subject: [Freeswitch-users] list conference name?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58B394B8E0@cooper>
References: <BANLkTik9mEqR9UL5u+NpS5d8r7iVH5f9XA@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B394B8E0@cooper>
Message-ID: <BANLkTinQB--W-dVFxi4SNaRkBk9ZkUBmOA@mail.gmail.com>

I waanna list a particular conference as there will be many
thx



On Tue, Mar 29, 2011 at 1:50 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

>  Just do a ?conference list? and you will get all conferences, with their
> members. You will also see the name in there.
>
>
>
> /Peter
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *deniro
> *Skickat:* den 29 mars 2011 19:40
> *Till:* FreeSWITCH Users Help
> *?mne:* [Freeswitch-users] list conference name?
>
>
>
> Hi,
>
> I have a web +voice conferencing  and trying below
>
>
>
> $list=$api->executeString("conference <conf_name> list");
>
>
>
> What determines the conference name?
>
> is it voicebridge number or meeting id or  attendee PIN or moderator PIN
>
>
>
> I am unable to determine conference name and list it accordingly (performed
> many trials)
>
>
>
> is there any way to get the conference name by a command like above?
>
> what is the best way to get the conference name and list it and execute
> accordingly
>
>
>
> thx
>
> deniro--
>
>
>
>
>
> !DSPAM:4d921a7332761434317129!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ayhkor at gmail.com  Tue Mar 29 22:16:29 2011
From: ayhkor at gmail.com (deniro)
Date: Tue, 29 Mar 2011 14:16:29 -0400
Subject: [Freeswitch-users] list conference name?
In-Reply-To: <BANLkTinQB--W-dVFxi4SNaRkBk9ZkUBmOA@mail.gmail.com>
References: <BANLkTik9mEqR9UL5u+NpS5d8r7iVH5f9XA@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B394B8E0@cooper>
	<BANLkTinQB--W-dVFxi4SNaRkBk9ZkUBmOA@mail.gmail.com>
Message-ID: <BANLkTikmquCG4nztX1kdSEh-BR2=227MGw@mail.gmail.com>

I mean I wanna list conference that I am or someone is joining to
(during join session list only that conference details that I am joining)
not all conferences
thx




On Tue, Mar 29, 2011 at 2:12 PM, deniro <ayhkor at gmail.com> wrote:

> I waanna list a particular conference as there will be many
> thx
>
>
>
>   On Tue, Mar 29, 2011 at 1:50 PM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
>>    Just do a ?conference list? and you will get all conferences, with
>> their members. You will also see the name in there.
>>
>>
>>
>> /Peter
>>
>>
>>
>> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *F?r *deniro
>> *Skickat:* den 29 mars 2011 19:40
>> *Till:* FreeSWITCH Users Help
>> *?mne:* [Freeswitch-users] list conference name?
>>
>>
>>
>> Hi,
>>
>> I have a web +voice conferencing  and trying below
>>
>>
>>
>> $list=$api->executeString("conference <conf_name> list");
>>
>>
>>
>> What determines the conference name?
>>
>> is it voicebridge number or meeting id or  attendee PIN or moderator PIN
>>
>>
>>
>> I am unable to determine conference name and list it accordingly
>> (performed many trials)
>>
>>
>>
>> is there any way to get the conference name by a command like above?
>>
>> what is the best way to get the conference name and list it and execute
>> accordingly
>>
>>
>>
>> thx
>>
>> deniro--
>>
>>
>>
>>
>>
>> !DSPAM:4d921a7332761434317129!
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From michal.bielicki at seventhsignal.de  Tue Mar 29 22:31:02 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Tue, 29 Mar 2011 20:31:02 +0200
Subject: [Freeswitch-users] Build on solaris fails in libs/esl
In-Reply-To: <AANLkTi=YpGBXoFNEw+rrKSBd-cGbDk_8MQuwMD_N=a7y@mail.gmail.com>
References: <AANLkTimGtXWdRPAv8OJYWeu8VYtCF8ZfQt0osGVgU7G=@mail.gmail.com>	<AANLkTinab+WgVXUYtWaSccbwBV7G7LO-qjjcE4bYAKXj@mail.gmail.com>	<AANLkTinAtxnnmbVm-tBvEuOxUQ7=z2JebB9TK+0WdBuF@mail.gmail.com>	<AANLkTim77j3aBU0YrR8YGZEp=bDf9ND3-T0NXSznW4ov@mail.gmail.com>	<AANLkTikJ4bYiWcsU5uLjtV4pfcL_LV1bQbqUNsfAAZu0@mail.gmail.com>	<6ADAA4B6-802B-49F6-8C52-6BBCDE176F16@seventhsignal.de>
	<AANLkTi=YpGBXoFNEw+rrKSBd-cGbDk_8MQuwMD_N=a7y@mail.gmail.com>
Message-ID: <4D922566.6050204@seventhsignal.de>

Basically all the settings we carefully fix in automake are not in there.

On 29.03.2011 20:05, Anthony Minessale wrote:
> It looks to me like the biggest problem is they can't find the proto
> for strdup and some other things that normally would have been in
> string.h
>
> On Tue, Mar 29, 2011 at 10:33 AM, Michal Bielicki
> <michal.bielicki at seventhsignal.de>  wrote:
>> Its a problem with the build scripts for esl. looking at that right now ..
>>
>> Am 29.03.2011 um 17:24 schrieb Anthony Minessale:
>>
>>> If you send me a private email with login credentials and location of
>>> the source, I can see about correcting it for you.
>>> I don't currently have a box to test this on.
>>>
>>>
>>> On Tue, Mar 29, 2011 at 12:27 AM, A E [Gmail]<all.eforums at gmail.com>  wrote:
>>>> On Mon, Mar 28, 2011 at 11:40 PM, Anthony Minessale
>>>> <anthony.minessale at gmail.com>  wrote:
>>>>> try latest git to see if the silly out of bound thing is fixed.
>>>>>
>>>> Hi Anthony,
>>>>
>>>> It seems to have been, yes. I don't see the these "integer overflow"
>>>> messages, but now some other ones have been noticed:
>>>>
>>>> e.g.
>>>> make: *** [current] Error 2"/home/freeswitch/fs/libs/esl/src/include/esl.h",
>>>> line 465: warning: implicit function declaration: strcasecmp
>>>> (E_NO_IMPLICIT_DECL_ALLOWED)
>>>> "libs/esl/src/esl.c", line 1191: warning: implicit function declaration:
>>>> strdup (E_NO_IMPLICIT_DECL_ALLOWED)
>>>> "libs/esl/src/esl.c", line 1191: warning: improper pointer/integer
>>>> combination: op "=" (E_BAD_PTR_INT_COMBINATION)
>>>> "libs/esl/src/esl.c", line 1232: warning: improper pointer/integer
>>>> combination: op "=" (E_BAD_PTR_INT_COMBINATION)
>>>> "/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
>>>> implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
>>>> "/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
>>>> implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
>>>> "/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
>>>> implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
>>>> "/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
>>>> implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
>>>> "libs/esl/ivrd.c", line 98: warning: implicit function declaration: signal
>>>> (E_NO_IMPLICIT_DECL_ALLOWED)
>>>> "/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
>>>> implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
>>>> "/home/freeswitch/fs/libs/esl/src/include/esl.h", line 465: warning:
>>>> implicit function declaration: strcasecmp (E_NO_IMPLICIT_DECL_ALLOWED)
>>>>
>>>> As well as a load of these:
>>>>
>>>> -- "/home/freeswitch/fs/libs/spandsp/src/spandsp/bit_operations.h", line 50:
>>>> warning: "__asm__" is an extension of ANSI C (E_KW_IS_AN_EXTENSION_OF_ANSI)
>>>>
>>>> And lastly,
>>>>
>>>> BTW, Not that you claimed that you fixed it, might as well tell you that,
>>>> esl and mod_hash compilation still fails. Pastebin here:
>>>> http://pastebin.freeswitch.org/15887
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> 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-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> Michal Bielicki
>> Gesch?ftsf?hrer / CEO
>>
>> Seventh Signal Ltd.&  Co. KG
>> Weigandufer 45, B?ro 115, D-12059 Berlin
>> Voice: +49 30 60988730
>>
>> Amtsgericht Charlottenburg HRA 44413 B
>> Ust.-ID: DE266981999
>> Gesch?ftsf?hrer: Michal Bielicki
>> Pers?nlich Haftende Gesellschafterin:
>> Seventh Signal Ltd, 69 Great Hampton St. Birmingham,
>> B18 6EW, GB, Company Nr.: 06889439
>> WWW.: http://www.seventhsignal.de
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>



From ejay.greeves at yahoo.com  Tue Mar 29 23:49:39 2011
From: ejay.greeves at yahoo.com (Ejay Greeves)
Date: Tue, 29 Mar 2011 20:49:39 +0100 (BST)
Subject: [Freeswitch-users] recvEvent causes zombies process
In-Reply-To: <AANLkTim4a3uh6O1qMBQ=GGc5=HmEwK4P_H7E6ZE7DQtO@mail.gmail.com>
Message-ID: <755992.33294.qm@web132301.mail.ird.yahoo.com>

I have added the second suggestion you made, so far I done it this way because it was easier to get the result. Is it correct how I have done it and if so does it take less resources the other way you suggest because at this stage I don't yet that


loop do	event =
 con.recvEvent?????? Signal.trap('CHLD', 'IGNORE')	pid = fork do

	end
	Process.detach(pid)	end


--- On Tue, 29/3/11, Anthony Minessale <anthony.minessale at gmail.com> wrote:

From: Anthony Minessale <anthony.minessale at gmail.com>
Subject: Re: [Freeswitch-users] recvEvent causes zombies process
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Tuesday, 29 March, 2011, 15:52

you are calling fork every time you receive an event.you need to exit the process from the child and reap or ignore it on the parent.

On Tue, Mar 29, 2011 at 6:09 AM, Ejay Greeves <ejay.greeves at yahoo.com> wrote:


I am wanting to code an inbound event program.

In order to see why I am getting zombie processes I stripped out all my code and used the minimum skeletcon code to do a test.
con = ESL::ESLconnection.new('127.0.0.1', '8021', 'ClueCon')
con.events("plain", "CHANNEL_HANGUP_COMPLETE")
loop do	event =
 con.recvEvent		pid = fork do		

	end
	Process.detach(pid)	end

This code does nothing but wait for an incoming event, creates an fork then detach however it still produces a defunct process.

I think con.recvEvent is the cause?




      
_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

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



-----Inline Attachment Follows-----

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



      
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From william.suffill at gmail.com  Wed Mar 30 00:16:30 2011
From: william.suffill at gmail.com (William Suffill)
Date: Tue, 29 Mar 2011 16:16:30 -0400
Subject: [Freeswitch-users] "Free" Conference Calling
In-Reply-To: <AANLkTi=teqs=zVa8acRCMhm+7=MEAtZj-BiGrL4u_tyu@mail.gmail.com>
References: <AANLkTikK8zUBnPtBX4n+4S3dYn+_VPiNJieSLzO7Vuhi@mail.gmail.com>
	<201103291400.04549.sos@sokhapkin.dyndns.org>
	<AANLkTi=teqs=zVa8acRCMhm+7=MEAtZj-BiGrL4u_tyu@mail.gmail.com>
Message-ID: <AANLkTi=XUMyXz47FBwiOwiet82JM2r_oKX7Civa6x2uW@mail.gmail.com>

I have seen term routes including the conference prefixs since not all
routes take it at this point.  Best to see how much of your customer basis
cares about them and if it's great enough see about getting additional
routes to cover that market. I'm sure someone has gone to the ILECs
regarding getting capacity and reselling it via sip already.
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From gourav at rentec.com  Wed Mar 30 00:43:37 2011
From: gourav at rentec.com (Gourav Vohra)
Date: Tue, 29 Mar 2011 16:43:37 -0400 (EDT)
Subject: [Freeswitch-users] CALL_REJECT on freeswitch
In-Reply-To: <1391996406.51705.1301430239613.JavaMail.root@zinnia1>
Message-ID: <1900960152.51877.1301431417366.JavaMail.root@zinnia1>

I have two freeswitch servers.  First one is installed using the official release and the second is from a snapshot from mar 15.

Call reject on the official release works without any problem.  It doesn't work on the snapshot release.  I am also having some trouble with SLA and barge on the snapshot release whereas the official release works without any problems.

I am assuming the Official release should be used in production and snapshot releases are only for testing.  Please correct me if I am wrong.


Thanks.


From duncan.pierce at gmail.com  Tue Mar 29 20:04:14 2011
From: duncan.pierce at gmail.com (Duncan Pierce)
Date: Tue, 29 Mar 2011 17:04:14 +0100
Subject: [Freeswitch-users] How to set PIN after conference created?
In-Reply-To: <AANLkTimETc09FbB3M2D79ZfrFhAn12ac_p+4WNFEya2i@mail.gmail.com>
References: <AANLkTimMvsofsbCMUjSRAzNEBLZiba3buEb5j=yRdDn+@mail.gmail.com>
	<AANLkTimETc09FbB3M2D79ZfrFhAn12ac_p+4WNFEya2i@mail.gmail.com>
Message-ID: <AANLkTimZe9kqpZTz6GbaXAMtgccHxLvq3PeWxWJT8trf@mail.gmail.com>

Thanks for responding Michael. Do you have any hints for getting started
with implementing PIN security in the dialplan? I can see how a PIN could be
prompted for and gathered but I can't see how to "attach" a PIN to a
conference.

Best regards,
Duncan


On 29 March 2011 16:37, Michael Collins <msc at freeswitch.org> wrote:

> This is one of those cases where the devs tried very hard not to force
> feed specific functionality onto the user. The bottom line here is
> that the devs consider mod_conference's primary function to be that of
> getting channels into a conference, muxing the audio, etc. Things like
> PINs and security, etc. can all be handled in the dialplan. This is,
> indeed, by design. Instead of forcing the security into mod_conference
> itself we let the FS implementor decide how he/she wants to perform
> security functions. One advantage to doing all the security in the
> dialplan is that you have a lot of power and flexibility: you do
> security however YOU feel is best.
>
> I believe that if you start viewing PINs and security/privacy from the
> perspective that they are functions of the dialplan instead of
> functions of mod_conference then you will be able to solve your
> problem much easier.
>
> -MC
>
> On Tue, Mar 29, 2011 at 4:17 AM, Duncan Pierce <duncan.pierce at gmail.com>
> wrote:
> > Hi everyone,
> > I'm new to FreeSWITCH. I've managed to set up many features quite easily
> but
> > I'm struggling with mod_conference. I want to allow a user to create a
> > conference and set its PIN by dialling a special number
> > 81<conference_number>*<pin>:
> > <extension name="initiate_conference">
> > <condition field="destination_number" expression="^81(\d*)\*(\d*)$">
> > <action application="log" data="INFO Creating conference $1 with PIN
> $2"/>
> > <action application="answer"/>
> > <action application="conference" data="$1 at default+$2"/>
> > </condition>
> > </extension>
> > Other users can dial into the same conference using a different access
> > number, but are challenged for the PIN set by the first user:
> > <extension name="join_conference">
> > <condition field="destination_number" expression="^82(\d*)$">
> > <action application="answer"/>
> > <action application="conference" data="$1"/>
> > </condition>
> > </extension>
> > There is no <param name="pin" ...> set in <configuration
> > name="conference.conf"><profiles><profile name="default">.
> > I was expecting <action application="conference" data="$1 at default+$2"/>
> to
> > create the conference and set its PIN, so that a subsequent <action
> > application="conference" data="$1"/> would be challenged to provide the
> same
> > PIN. Instead what happens is the first caller has the required PIN set
> for
> > their call, but the second caller is not challenged to provide the PIN at
> > all.
> > Looking at the code in mod_conference.c on git checkout v1.0.6 I see
> there
> > only 2 ways conference->pin can be set: via <param name="pin" .../> in
> the
> > profile or via API 'conference $1 pin $2'. I was expecting there would be
> a
> > third way: assigning conference->pin = dpin when creating a new
> conference.
> > Although I find this a bit confusing it could be by design.
> > I tried using the API to set the PIN after the conference has been
> created
> > but I lack the knowledge to do this successfully. I've tried:
> > * <action application="set" data="result=${conference($1 pin $2)}"/>
> after
> > <action application="conference" data="$1 at default+$2"/> - no actions
> execute
> > after the <action application="conference">
> > * <action application="set" data="result=${conference($1 pin $2)}"/>
> before
> > <action application="conference" data="$1 at default+$2"/> -
> unsurprisingly, it
> > reports conference not found
> > * <action application="export" data="nolocal:api_on_answer=conference $1
> pin
> > $2"/> - appears not to execute but it's difficult to be sure in this case
> > Thanks in advance for any help you can provide!
> > Best regards,
> > Duncan
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Duncan Pierce
duncanpierce.org
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From msc at freeswitch.org  Wed Mar 30 01:15:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 29 Mar 2011 14:15:02 -0700
Subject: [Freeswitch-users] How to set PIN after conference created?
In-Reply-To: <AANLkTimZe9kqpZTz6GbaXAMtgccHxLvq3PeWxWJT8trf@mail.gmail.com>
References: <AANLkTimMvsofsbCMUjSRAzNEBLZiba3buEb5j=yRdDn+@mail.gmail.com>
	<AANLkTimETc09FbB3M2D79ZfrFhAn12ac_p+4WNFEya2i@mail.gmail.com>
	<AANLkTimZe9kqpZTz6GbaXAMtgccHxLvq3PeWxWJT8trf@mail.gmail.com>
Message-ID: <AANLkTinW3iZG2RgB5_Rq_ta7acYquGGGaE8m8M-g4Dz5@mail.gmail.com>

On Tue, Mar 29, 2011 at 9:04 AM, Duncan Pierce <duncan.pierce at gmail.com> wrote:
> Thanks for responding Michael. Do you have any hints for getting started
> with implementing PIN security in the dialplan? I can see how a PIN could be
> prompted for and gathered but I can't see how to "attach" a PIN to a
> conference.

In this case you'd need to keep a database of PINs and their
corresponding conferences. You could do your lookup with mod_xml_curl
which is the best long-term solution. In the short term you could do
something in mod_lua/mod_perl via dialplan and have the script poll
the database. Heck, you could even hard-code the pin into your
dialplan if you're desperate to give it a test drive. :)

-MC


From gourav at rentec.com  Wed Mar 30 01:19:23 2011
From: gourav at rentec.com (Gourav Vohra)
Date: Tue, 29 Mar 2011 17:19:23 -0400 (EDT)
Subject: [Freeswitch-users] CALL_REJECT on freeswitch
In-Reply-To: <1900960152.51877.1301431417366.JavaMail.root@zinnia1>
Message-ID: <828065258.52145.1301433563225.JavaMail.root@zinnia1>

Please disregard my post.  

The issue is not with freeswitch but my config.  I have to look into it more.



regards,
Gourav.-

----- Original Message -----
From: "Gourav Vohra" <gourav at rentec.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Tuesday, March 29, 2011 4:43:37 PM
Subject: Re: CALL_REJECT on freeswitch

I have two freeswitch servers.  First one is installed using the official release and the second is from a snapshot from mar 15.

Call reject on the official release works without any problem.  It doesn't work on the snapshot release.  I am also having some trouble with SLA and barge on the snapshot release whereas the official release works without any problems.

I am assuming the Official release should be used in production and snapshot releases are only for testing.  Please correct me if I am wrong.


Thanks.


From msc at freeswitch.org  Wed Mar 30 01:24:32 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 29 Mar 2011 14:24:32 -0700
Subject: [Freeswitch-users] list conference name?
In-Reply-To: <BANLkTikmquCG4nztX1kdSEh-BR2=227MGw@mail.gmail.com>
References: <BANLkTik9mEqR9UL5u+NpS5d8r7iVH5f9XA@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B394B8E0@cooper>
	<BANLkTinQB--W-dVFxi4SNaRkBk9ZkUBmOA@mail.gmail.com>
	<BANLkTikmquCG4nztX1kdSEh-BR2=227MGw@mail.gmail.com>
Message-ID: <AANLkTinRYY3+bP1bpxQ5wLJesgHC141LxVNiFHMuDrT9@mail.gmail.com>

the conference name is determined by the dialplan. From default.xml:

   <extension name="cdquality_conferences">
     <condition field="destination_number" expression="^(33\d{2})$">
       <action application="answer"/>
       <action application="conference" data="$1-${domain_name}@cdquality"/>
     </condition>
   </extension>

The argument to the conference app is <conf-name>@<conf-profile>

If you have a default install and a phone connected then do this:
call 3300
at the cli type "conference list"
you'll see a conference named "3300-x.x.x.x" where x.x.x.x is your domain
name, which is usually your ipv4 address.

you can also do "conference list 3300-x.x.x.x" and get just the information
on that conference. For extra fun try "conference xml_list 3300-x.x.x.x"

-MC

On Tue, Mar 29, 2011 at 11:16 AM, deniro <ayhkor at gmail.com> wrote:
> I mean I wanna list conference that I am or someone is joining to
> (during join session list only that conference details that I am joining)
> not all conferences
> thx
>
>
>
> On Tue, Mar 29, 2011 at 2:12 PM, deniro <ayhkor at gmail.com> wrote:
>>
>> I waanna list a particular conference as there will be many
>> thx
>>
>>
>> On Tue, Mar 29, 2011 at 1:50 PM, Peter Olsson
>> <peter.olsson at visionutveckling.se> wrote:
>>>
>>> Just do a ?conference list? and you will get all conferences, with their
>>> members. You will also see the name in there.
>>>
>>>
>>>
>>> /Peter
>>>
>>>
>>>
>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r deniro
>>> Skickat: den 29 mars 2011 19:40
>>> Till: FreeSWITCH Users Help
>>> ?mne: [Freeswitch-users] list conference name?
>>>
>>>
>>>
>>> Hi,
>>>
>>> I have a web +voice conferencing  and trying below
>>>
>>>
>>>
>>> $list=$api->executeString("conference <conf_name> list");
>>>
>>>
>>>
>>> What determines the conference name?
>>>
>>> is it voicebridge number or meeting id or  attendee PIN or moderator PIN
>>>
>>>
>>>
>>> I am unable to determine conference name and list it accordingly
>>> (performed many trials)
>>>
>>>
>>>
>>> is there any way to get the conference name by a command like above?
>>>
>>> what is the best way to get the conference name and list it and execute
>>> accordingly
>>>
>>>
>>>
>>> thx
>>>
>>> deniro--
>>>
>>>
>>>
>>>
>>>
>>> !DSPAM:4d921a7332761434317129!
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Wed Mar 30 01:25:12 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 29 Mar 2011 16:25:12 -0500
Subject: [Freeswitch-users] recvEvent causes zombies process
In-Reply-To: <755992.33294.qm@web132301.mail.ird.yahoo.com>
References: <AANLkTim4a3uh6O1qMBQ=GGc5=HmEwK4P_H7E6ZE7DQtO@mail.gmail.com>
	<755992.33294.qm@web132301.mail.ird.yahoo.com>
Message-ID: <AANLkTinMNMLOLmJ0RptQCGyEQc6JofdqPzTsnFkGJu49@mail.gmail.com>

it's just a necessary thing to do when forking a child process.
so its fine that way if it's working.


On Tue, Mar 29, 2011 at 2:49 PM, Ejay Greeves <ejay.greeves at yahoo.com>wrote:

> I have added the second suggestion you made, so far I done it this way
> because it was easier to get the result. Is it correct how I have done it
> and if so does it take less resources the other way you suggest because at
> this stage I don't yet that
>
>
>
> loop do
> event = con.recvEvent
>        Signal.trap('CHLD', 'IGNORE')
> pid = fork do
>
> end
>
> Process.detach(pid)
> end
>
>
>
> --- On *Tue, 29/3/11, Anthony Minessale <anthony.minessale at gmail.com>*wrote:
>
>
> From: Anthony Minessale <anthony.minessale at gmail.com>
> Subject: Re: [Freeswitch-users] recvEvent causes zombies process
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Tuesday, 29 March, 2011, 15:52
>
>
> you are calling fork every time you receive an event.
> you need to exit the process from the child and reap or ignore it on the
> parent.
>
>
> On Tue, Mar 29, 2011 at 6:09 AM, Ejay Greeves <ejay.greeves at yahoo.com<http://mc/compose?to=ejay.greeves at yahoo.com>
> > wrote:
>
> I am wanting to code an inbound event program.
>
> In order to see why I am getting zombie processes I stripped out all my
> code and used the minimum skeletcon code to do a test.
>
> con = ESL::ESLconnection.new('127.0.0.1', '8021', 'ClueCon')
> con.events("plain", "CHANNEL_HANGUP_COMPLETE")
>
> loop do
> event = con.recvEvent
>  pid = fork do
>
>
> end
>
> Process.detach(pid)
> end
>
>
> This code does nothing but wait for an incoming event, creates an fork then
> detach however it still produces a defunct process.
>
> I think con.recvEvent is the cause?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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<http://mc/compose?to=MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<http://mc/compose?to=PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org<http://mc/compose?to=sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<http://mc/compose?to=googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> -----Inline Attachment Follows-----
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From ayhkor at gmail.com  Wed Mar 30 02:27:20 2011
From: ayhkor at gmail.com (deniro)
Date: Tue, 29 Mar 2011 18:27:20 -0400
Subject: [Freeswitch-users] list conference name?
In-Reply-To: <AANLkTinRYY3+bP1bpxQ5wLJesgHC141LxVNiFHMuDrT9@mail.gmail.com>
References: <BANLkTik9mEqR9UL5u+NpS5d8r7iVH5f9XA@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58B394B8E0@cooper>
	<BANLkTinQB--W-dVFxi4SNaRkBk9ZkUBmOA@mail.gmail.com>
	<BANLkTikmquCG4nztX1kdSEh-BR2=227MGw@mail.gmail.com>
	<AANLkTinRYY3+bP1bpxQ5wLJesgHC141LxVNiFHMuDrT9@mail.gmail.com>
Message-ID: <BANLkTikV-f0P9G1qZiD_eF3WD8UUu+Xz2Q@mail.gmail.com>

thanks Michael
In my case I am using mod_perl
/opt/freeswitch/scripts/xxx.pl from within public.xml to join to conference
and $PIN at profile  in the perl program
so the conference name must be $PIN



On Tue, Mar 29, 2011 at 5:24 PM, Michael Collins <msc at freeswitch.org> wrote:

> the conference name is determined by the dialplan. From default.xml:
>
>    <extension name="cdquality_conferences">
>      <condition field="destination_number" expression="^(33\d{2})$">
>        <action application="answer"/>
>        <action application="conference" data="$1-${domain_name}@cdquality
> "/>
>      </condition>
>    </extension>
>
> The argument to the conference app is <conf-name>@<conf-profile>
>
> If you have a default install and a phone connected then do this:
> call 3300
> at the cli type "conference list"
> you'll see a conference named "3300-x.x.x.x" where x.x.x.x is your domain
> name, which is usually your ipv4 address.
>
> you can also do "conference list 3300-x.x.x.x" and get just the information
> on that conference. For extra fun try "conference xml_list 3300-x.x.x.x"
>
> -MC
>
>
> On Tue, Mar 29, 2011 at 11:16 AM, deniro <ayhkor at gmail.com> wrote:
> > I mean I wanna list conference that I am or someone is joining to
> > (during join session list only that conference details that I am joining)
> > not all conferences
> > thx
> >
> >
> >
> > On Tue, Mar 29, 2011 at 2:12 PM, deniro <ayhkor at gmail.com> wrote:
> >>
> >> I waanna list a particular conference as there will be many
> >> thx
> >>
> >>
> >> On Tue, Mar 29, 2011 at 1:50 PM, Peter Olsson
> >> <peter.olsson at visionutveckling.se> wrote:
> >>>
> >>> Just do a ?conference list? and you will get all conferences, with
> their
> >>> members. You will also see the name in there.
> >>>
> >>>
> >>>
> >>> /Peter
> >>>
> >>>
> >>>
> >>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r deniro
> >>> Skickat: den 29 mars 2011 19:40
> >>> Till: FreeSWITCH Users Help
> >>> ?mne: [Freeswitch-users] list conference name?
> >>>
> >>>
> >>>
> >>> Hi,
> >>>
> >>> I have a web +voice conferencing  and trying below
> >>>
> >>>
> >>>
> >>> $list=$api->executeString("conference <conf_name> list");
> >>>
> >>>
> >>>
> >>> What determines the conference name?
> >>>
> >>> is it voicebridge number or meeting id or  attendee PIN or moderator
> PIN
> >>>
> >>>
> >>>
> >>> I am unable to determine conference name and list it accordingly
> >>> (performed many trials)
> >>>
> >>>
> >>>
> >>> is there any way to get the conference name by a command like above?
> >>>
> >>> what is the best way to get the conference name and list it and execute
> >>> accordingly
> >>>
> >>>
> >>>
> >>> thx
> >>>
> >>> deniro--
> >>>
> >>>
> >>>
> >>>
> >>>
> >>> !DSPAM:4d921a7332761434317129!
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Wed Mar 30 03:11:30 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 30 Mar 2011 00:11:30 +0100
Subject: [Freeswitch-users] CALL_REJECT on freeswitch
In-Reply-To: <828065258.52145.1301433563225.JavaMail.root@zinnia1>
References: <828065258.52145.1301433563225.JavaMail.root@zinnia1>
Message-ID: <2D56929E-E52B-47C5-B6AE-E505C9568527@gmail.com>

In answer to your other question - official=1.0.6?

Git head is probably best. There are many bugfixes and improvements since 1.0.6, and it's almost always stable (you should if course always test when upgrading before sending live traffic). The snapshot is a daily build from git head. I always run off git head and periodically update personally, but there are people who prefer using the latest official one. It's largely personal preference I guess unless you want a feature that's only in git.

Steve on iPhone


On 29 Mar 2011, at 22:19, Gourav Vohra <gourav at rentec.com> wrote:

> Please disregard my post.  
> 
> The issue is not with freeswitch but my config.  I have to look into it more.
> 
> 
> 
> regards,
> Gourav.-
> 
> ----- Original Message -----
> From: "Gourav Vohra" <gourav at rentec.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Tuesday, March 29, 2011 4:43:37 PM
> Subject: Re: CALL_REJECT on freeswitch
> 
> I have two freeswitch servers.  First one is installed using the official release and the second is from a snapshot from mar 15.
> 
> Call reject on the official release works without any problem.  It doesn't work on the snapshot release.  I am also having some trouble with SLA and barge on the snapshot release whereas the official release works without any problems.
> 
> I am assuming the Official release should be used in production and snapshot releases are only for testing.  Please correct me if I am wrong.
> 
> 
> Thanks.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From msc at freeswitch.org  Wed Mar 30 05:49:03 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 29 Mar 2011 18:49:03 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Tomorrow - Author of
	SIPVicious Joining Us!
Message-ID: <AANLkTimpFmsmcCrPRY5ixbw+DDojs+=1u+gya6useNtW@mail.gmail.com>

Hello all!

We have a special guest on our Wednesday conference call: Sandro Gauci,
author of SIPVicious. He will be joining us to discuss SIPVicious and
related issues. Please join us Wednesday, March 30th, at 1PM Eastern.
Dial-in instructions can be found on the meeting agenda page:

http://wiki.freeswitch.org/wiki/FS_weekly_2011_03_30

Thanks,
Michael
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From andrew.keil at askinteractive.net  Wed Mar 30 02:42:13 2011
From: andrew.keil at askinteractive.net (Andrew Keil)
Date: Wed, 30 Mar 2011 09:42:13 +1100
Subject: [Freeswitch-users] Freeswitch latest source rebuild issue under
	Windows
Message-ID: <C42A22554C0CF44DAC0AA1F132B86DE65716763FA2@ASKSERVER.asksbs.local>

To Freeswitch developers....



I previously posted this to Freeswitch-dev and it is waiting Moderator approval.  However Darren Schreiber suggested I post this here.



OK.  Here goes, what I have done so far...



1) Setup a Virtual Machine running 32-bit XP SP3 (with all Critical Updates applied).  That way I can quickly go back in time and try an install again if necessary.

2) Since I noticed that you now support Visual C++ 2010 Express, I downloaded and installed that (without the option for SQL Server Express 2008).

3) I also then installed the Windows SDK (7.1 (the latest version)).  The reason for this is when I download your latest build of Freeswitch 1.0.7 today it complained about 64bit settings throughout various project files when opening Freeswitch.2010.express.sln

4) Then I simply downloaded your latest build from http://latest.freeswitch.org/ and extracted it out to c:\FreeSWITCH (so that there is a sub-directory freeswitch-1.0.7 below it with all the files) - this looks the same layout as the Freeswitch book example, except 1.0.7 version instead.

5) Opened the Freeswitch.2010.express.sln

                5.1) Stated inside the Visual Studio Output Window: "Some of the properties associated with the solution could not be read."

6) Built the solution via F7



See attached for the build errors from Visual Studio.



I guess my questions are the following:



Q1) Should I be running Visual C++ 2010 Express or Visual C++ 2008 Express - it does not matter to me, however I always like to move with the times?

Q2) Was I right in installing the Windows SDK (http://msdn.microsoft.com/en-us/windows/bb980924.aspx) after the installation of Visual C++ 2010 Express?  Perhaps this should be added to the Freeswitch windows installation instructions page: http://wiki.freeswitch.org/wiki/Installation_for_Windows

Q3) Should I expect the http://latest.freeswitch.org/ version of freeswitch to Build first time?  Has this been tested?

Q4) Should I simply use the 1.0.6 version with Visual C++ 2008 Express and follow the Freeswitch book word for word?  This seems a reliable path, but I am a 'C' developer myself and would like to start out with a more up-to-date dev. environment and a version of freeswitch that is closer to the current version.

Q5) Can someone make comments on the Build errors that are inside the attached file.



Obviously I could go through and debug each build error, however I feel it best to have my questions answered above to avoid wasting time.



Please note.  I can happily start again and go back in time to the point prior to the installation of Visual C++ 2010 Express, since I am running a ESXi Virtual Machine.  All I aiming for is a clean setup of FreeSWITCH under XP for testing and developing purposes (this is not to run in a production environment).  I would like to be as up-to-date as possible in order to progress quickly (obviously I will install GIT/TortoiseGIT etc.. to make my dev. environment more integrated once I have a working base version of Freeswitch).



Thanks in advance,



Andrew Keil

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From boris at tagnet.ru  Wed Mar 30 07:25:45 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Wed, 30 Mar 2011 09:25:45 +0600
Subject: [Freeswitch-users] bind_digit_action help
In-Reply-To: <AANLkTi=aidD4PFjdj+gQ8F-aE_mgPGM9eALXw8xhEAqb@mail.gmail.com>
References: <4D91734C.8010208@tagnet.ru>	<AANLkTin9NegHMEOrCWyGZXka_mhsNdRsV3ZhegivmQQ9@mail.gmail.com>
	<AANLkTi=aidD4PFjdj+gQ8F-aE_mgPGM9eALXw8xhEAqb@mail.gmail.com>
Message-ID: <4D92A2B9.6020006@tagnet.ru>

Hello!

     Thank You! It works....
> FYI,
>
> I added this to the bind_digit_action wiki page:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_digit_action#Setting_on_the_B_leg
>
> If anyone has more specific information and/or examples please add
> them to that section.
>
> Thanks,
> MC
>
> On Tue, Mar 29, 2011 at 8:31 AM, Anthony Minessale
> <anthony.minessale at gmail.com>  wrote:
>> Here are several choices:
>>
>> These 2 vars on the A leg using the set app
>> <action application="set" data="bridge_pre_execute_bleg_app=bind_digit_action"/>
>> <action application="set" data="bridge_pre_execute_bleg_data=whatever"/>
>>
>> This var with export app
>> <action application="export"
>> data="nolocal:execute_on_answer=bind_digit_action whatever"/>
>>
>>
>> These vars with export app
>> <action application="export"
>> data="nolocal:bridge_pre_execute_app=bind_digit_action"/>
>> <action application="export" data="nolocal:bridge_pre_execute_data=whatever"/>
>>
>>
>>
>> or one of these vars in the dial string {}
>>
>> {bridge_pre_execute_app=bind_digit_action,bridge_pre_execute_data='whatever'}
>> {execute_on_answer='bind_digit_action whatever'}
>>
>>
>>
>>
>> On Tue, Mar 29, 2011 at 12:51 AM, Boris Kovalenko<boris at tagnet.ru>  wrote:
>>> Hello!
>>>
>>>      Is this possible to get bind_digit_action work on B leg? I got it
>>> working on A leg and can't on B. Would someone provide a little example?
>>>
>>> --
>>> ? ?????????,
>>>    ????? ?????????
>>>    ??? "??????"
>>>    (3435) 494991
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From jeff at jefflenk.com  Wed Mar 30 07:31:15 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 29 Mar 2011 20:31:15 -0700 (PDT)
Subject: [Freeswitch-users] Freeswitch latest source rebuild issue under
	Windows
In-Reply-To: <C42A22554C0CF44DAC0AA1F132B86DE65716763FA2@ASKSERVER.asksbs.local>
References: <C42A22554C0CF44DAC0AA1F132B86DE65716763FA2@ASKSERVER.asksbs.local>
Message-ID: <1301455875395-6221733.post@n2.nabble.com>

VS2010 express is fine.

You should not need to install any additional SDK's  

As far as I know Express does not support x64 projects which is the warnings
you get when you open the solution. You can ignore the warnings.

Your build error is caused by the following line:
c1 : fatal error C1083: Cannot open source file: 'pcre_chartables.c': No
such file or directory
This is a generated file by the build system which must have failed???

Did you try the build again?

I normally use VS2010 Pro or Ultimate but I will check the build with vs2010
Express again.


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-latest-source-rebuild-issue-under-Windows-tp6221661p6221733.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From fieldpeak at gmail.com  Wed Mar 30 07:49:21 2011
From: fieldpeak at gmail.com (=?utf-8?B?Q2hhcmxlcw==?=)
Date: Wed, 30 Mar 2011 11:49:21 +0800
Subject: [Freeswitch-users]
	=?utf-8?q?FS_performance_test_-Hard_disk_alway?=
	=?utf-8?q?_busywhen_registerred_500_users?=
References: <4d91a177.4323e70a.2cca.ffffefc8@mx.google.com>,
	<549CFEF87AEDE841A38E9D15EAB4C04C58B394B66A@cooper>,
	<AANLkTimRHZUdrg+aeBYSyfwuA-r0Ee7ry4WTbLedq4VD@mail.gmail.com>,
	<549CFEF87AEDE841A38E9D15EAB4C04C58B394B78D@cooper>
Message-ID: <4d92a844.254b640a.36b5.ffffb7c6@mx.google.com>

Hi Peter,

Tested with the latest GIT, the performace increase a lot, 1/10 Hardisk I/O that i used previous version. 
the LED of HD blink each 1-3 seconds, fully acceptable.

Thanks for your valuable infomation!!

2011-03-30 



Charles 



???? Peter Olsson 
????? 2011-03-29  20:18:05 
???? 'FreeSWITCH Users Help' 
??? 
??? Re: [Freeswitch-users] FS performance test -Hard disk alway busywhen registerred 500 users 
 
Ahh - I didn't read you were on Windows.. :)
Are you using latest git though - there has been lots of rewrites last month to handle cached sqlite handles better.
/Peter
-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r fieldpeak
Skickat: den 29 mars 2011 13:03
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] FS performance test -Hard disk alway busy when registerred 500 users
Hi Peter,
Thanks for your prompt reponse and recommendation.
I ever tried two ramdisk softwares (VSuite and DataRAM free edition),
both work. Can you advise which one is preferred using in the
production? or anyother recomended product running in windows
platform?
RAMDisk is a 'program' after all, i'm afraid whether it can be loaded
prior to freeswitch service?
if using ramdisk, we have to seperatedly install RAMdisk with FS
package in the filed, it maybe involve work comparing one installation
pakage...
is there any windows native support Ramdisk like the one in Linux?
sorry for coming up so many question :)
Regards,
Charles
2011/3/29, Peter Olsson <peter.olsson at visionutveckling.se>:
> The recommendation is to use a RAM disk if you see performance problems. If
> your disk is "blinking" I wouldn't say that it's that much loaded though -
> in that case it should have a more or less steady light :) RAM disk
> shouldn't be a stability problem, and if you reboot your computer, FS will
> re-create all databases needed.
> Also make sure to use latest GIT, there have been some changes how the
> sqlite db handles are cached internally.
> /Peter
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Charles
> Skickat: den 29 mars 2011 11:08
> Till: freeswitch-users
> ?mne: [Freeswitch-users] FS performance test -Hard disk alway busy when
> registerred 500 users
>
> I'm makeing the load test of the FreeSWITCH, I registerred 500 users to FS,
> and make NO calls, then the LED of Hard disk alway blink meaning it always
> busy... below is my hardware configuration, would someone can help how to
> improve the performance? thanks in advance.
> i tried to increased the nonce-ttl to 600 from 60 tring to decrease the
> times of writing db. however it doesn't make help...
> attached is the internal.xml for reference.
>
>
> Windows server 2008 64bit.
> CPU: Intel Core2 Duo L7400 with Intel 310 1.5GHz,
> Memory: 2G,
> bit: 64bit,
> HD: 160G
>
> P.S. i can think about RAMDISK, however, I'm afraid of it is so stable...
> if it can be increase the performance by HD itself, it will be great.
>
> Regards,
> Charles
>
> 
>
_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From peter.olsson at visionutveckling.se  Wed Mar 30 08:15:11 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 30 Mar 2011 06:15:11 +0200
Subject: [Freeswitch-users] FS performance test -Hard disk alway
 busywhen registerred 500 users
In-Reply-To: <4d92a844.254b640a.36b5.ffffb7c6@mx.google.com>
References: <4d91a177.4323e70a.2cca.ffffefc8@mx.google.com>,
	<549CFEF87AEDE841A38E9D15EAB4C04C58B394B66A@cooper>,
	<AANLkTimRHZUdrg+aeBYSyfwuA-r0Ee7ry4WTbLedq4VD@mail.gmail.com>,
	<549CFEF87AEDE841A38E9D15EAB4C04C58B394B78D@cooper>,
	<4d92a844.254b640a.36b5.ffffb7c6@mx.google.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58BE186BCE@cooper>

No problem - good to know it helped! :)

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Charles [fieldpeak at gmail.com]
Skickat: den 30 mars 2011 05:49
Till: FreeSWITCH Users Help; 'FreeSWITCH Users Help'
?mne: Re: [Freeswitch-users]    FS performance test -Hard disk alway busywhen registerred 500 users

Hi Peter,

Tested with the latest GIT, the performace increase a lot, 1/10 Hardisk I/O that i used previous version.
the LED of HD blink each 1-3 seconds, fully acceptable.

Thanks for your valuable infomation!!

2011-03-30
________________________________
Charles
________________________________
???? Peter Olsson
????? 2011-03-29  20:18:05
???? 'FreeSWITCH Users Help'
???
??? Re: [Freeswitch-users] FS performance test -Hard disk alway busywhen registerred 500 users
Ahh - I didn't read you were on Windows.. :)
Are you using latest git though - there has been lots of rewrites last month to handle cached sqlite handles better.
/Peter
-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r fieldpeak
Skickat: den 29 mars 2011 13:03
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] FS performance test -Hard disk alway busy when registerred 500 users
Hi Peter,
Thanks for your prompt reponse and recommendation.
I ever tried two ramdisk softwares (VSuite and DataRAM free edition),
both work. Can you advise which one is preferred using in the
production? or anyother recomended product running in windows
platform?
RAMDisk is a 'program' after all, i'm afraid whether it can be loaded
prior to freeswitch service?
if using ramdisk, we have to seperatedly install RAMdisk with FS
package in the filed, it maybe involve work comparing one installation
pakage...
is there any windows native support Ramdisk like the one in Linux?
sorry for coming up so many question :)
Regards,
Charles
2011/3/29, Peter Olsson <peter.olsson at visionutveckling.se>:
> The recommendation is to use a RAM disk if you see performance problems. If
> your disk is "blinking" I wouldn't say that it's that much loaded though -
> in that case it should have a more or less steady light :) RAM disk
> shouldn't be a stability problem, and if you reboot your computer, FS will
> re-create all databases needed.
> Also make sure to use latest GIT, there have been some changes how the
> sqlite db handles are cached internally.
> /Peter
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Charles
> Skickat: den 29 mars 2011 11:08
> Till: freeswitch-users
> ?mne: [Freeswitch-users] FS performance test -Hard disk alway busy when
> registerred 500 users
>
> I'm makeing the load test of the FreeSWITCH, I registerred 500 users to FS,
> and make NO calls, then the LED of Hard disk alway blink meaning it always
> busy... below is my hardware configuration, would someone can help how to
> improve the performance? thanks in advance.
> i tried to increased the nonce-ttl to 600 from 60 tring to decrease the
> times of writing db. however it doesn't make help...
> attached is the internal.xml for reference.
>
>
> Windows server 2008 64bit.
> CPU: Intel Core2 Duo L7400 with Intel 310 1.5GHz,
> Memory: 2G,
> bit: 64bit,
> HD: 160G
>
> P.S. i can think about RAMDISK, however, I'm afraid of it is so stable...
> if it can be increase the performance by HD itself, it will be great.
>
> Regards,
> Charles
>
>
>
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
!DSPAM:4d92a96732769097113148!

From andrew.keil at askinteractive.net  Wed Mar 30 08:07:59 2011
From: andrew.keil at askinteractive.net (Andrew Keil)
Date: Wed, 30 Mar 2011 15:07:59 +1100
Subject: [Freeswitch-users] Freeswitch latest source rebuild issue under
 Windows
Message-ID: <C42A22554C0CF44DAC0AA1F132B86DE65716763FA4@ASKSERVER.asksbs.local>

Jeff,

I have tried the build again with no luck - same main fatal error as before.

Followed from http://wiki.freeswitch.org/wiki/Installation_for_Windows

Note that building for Windows within a tree that has previously been built for a different platform will result in numerous errors and build failures. To resolve, delete the following generated files:
 libs/apr/include/apr.h 
libs/js/config.h 
libs/js/src/jsautocfg.h 
libs/js/nsprpub/pr/include/prcpucfg.h 
libs/iksemel/include/config.h 
libs/xmlrpc/xmlrpc_config.h 
libs/libsndfile/src/sfconfig.h; also rename libs/win32/libsndfile/config.h to libs/win32/libsndfile/sfconfig.h  {Note: this line is the only one that I actually did anything - since the rest of the files do not exist}
 libs/sofia-sip/libsofia-sip-ua/su/sofia-sip/su_configure.h 

Then Rebuild the solution.

Tried to rebuild three more times with no joy - same main fatal error as before.  See attached.

Regards,

Andrew Keil

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk
Sent: Wednesday, 30 March 2011 2:31 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Freeswitch latest source rebuild issue under Windows

VS2010 express is fine.

You should not need to install any additional SDK's  

As far as I know Express does not support x64 projects which is the warnings you get when you open the solution. You can ignore the warnings.

Your build error is caused by the following line:
c1 : fatal error C1083: Cannot open source file: 'pcre_chartables.c': No such file or directory This is a generated file by the build system which must have failed???

Did you try the build again?

I normally use VS2010 Pro or Ultimate but I will check the build with vs2010 Express again.


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-latest-source-rebuild-issue-under-Windows-tp6221661p6221733.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

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

__________ Information from ESET NOD32 Antivirus, version of virus signature database 5998 (20110329) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
 
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From curriegrad2004 at gmail.com  Wed Mar 30 08:29:50 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 29 Mar 2011 21:29:50 -0700
Subject: [Freeswitch-users] Freeswitch latest source rebuild issue under
	Windows
In-Reply-To: <C42A22554C0CF44DAC0AA1F132B86DE65716763FA4@ASKSERVER.asksbs.local>
References: <C42A22554C0CF44DAC0AA1F132B86DE65716763FA4@ASKSERVER.asksbs.local>
Message-ID: <AANLkTinsAd8LTzsZQF4DuT4P2wYQhTHsoHCubZMtfnC1@mail.gmail.com>

Did you set the autocrlf option to false under git? If this is unset
under git all kinds of problems will happen with compiling FS under
Windows.

On Tue, Mar 29, 2011 at 9:07 PM, Andrew Keil
<andrew.keil at askinteractive.net> wrote:
> Jeff,
>
> I have tried the build again with no luck - same main fatal error as before.
>
> Followed from http://wiki.freeswitch.org/wiki/Installation_for_Windows
>
> Note that building for Windows within a tree that has previously been built for a different platform will result in numerous errors and build failures. To resolve, delete the following generated files:
> ?libs/apr/include/apr.h
> libs/js/config.h
> libs/js/src/jsautocfg.h
> libs/js/nsprpub/pr/include/prcpucfg.h
> libs/iksemel/include/config.h
> libs/xmlrpc/xmlrpc_config.h
> libs/libsndfile/src/sfconfig.h; also rename libs/win32/libsndfile/config.h to libs/win32/libsndfile/sfconfig.h ?{Note: this line is the only one that I actually did anything - since the rest of the files do not exist}
> ?libs/sofia-sip/libsofia-sip-ua/su/sofia-sip/su_configure.h
>
> Then Rebuild the solution.
>
> Tried to rebuild three more times with no joy - same main fatal error as before. ?See attached.
>
> Regards,
>
> Andrew Keil
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk
> Sent: Wednesday, 30 March 2011 2:31 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Freeswitch latest source rebuild issue under Windows
>
> VS2010 express is fine.
>
> You should not need to install any additional SDK's
>
> As far as I know Express does not support x64 projects which is the warnings you get when you open the solution. You can ignore the warnings.
>
> Your build error is caused by the following line:
> c1 : fatal error C1083: Cannot open source file: 'pcre_chartables.c': No such file or directory This is a generated file by the build system which must have failed???
>
> Did you try the build again?
>
> I normally use VS2010 Pro or Ultimate but I will check the build with vs2010 Express again.
>
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-latest-source-rebuild-issue-under-Windows-tp6221661p6221733.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 5998 (20110329) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From jeff at jefflenk.com  Wed Mar 30 08:39:16 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 29 Mar 2011 21:39:16 -0700 (PDT)
Subject: [Freeswitch-users] Freeswitch latest source rebuild issue under
	Windows
In-Reply-To: <C42A22554C0CF44DAC0AA1F132B86DE65716763FA4@ASKSERVER.asksbs.local>
References: <C42A22554C0CF44DAC0AA1F132B86DE65716763FA2@ASKSERVER.asksbs.local>
	<C42A22554C0CF44DAC0AA1F132B86DE65716763FA4@ASKSERVER.asksbs.local>
Message-ID: <1301459956302-6221815.post@n2.nabble.com>

Please try git head just submitted fixes

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-latest-source-rebuild-issue-under-Windows-tp6221661p6221815.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From peter.olsson at visionutveckling.se  Wed Mar 30 08:39:29 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 30 Mar 2011 06:39:29 +0200
Subject: [Freeswitch-users] Freeswitch latest source rebuild issue under
 Windows
In-Reply-To: <C42A22554C0CF44DAC0AA1F132B86DE65716763FA4@ASKSERVER.asksbs.local>
References: <C42A22554C0CF44DAC0AA1F132B86DE65716763FA4@ASKSERVER.asksbs.local>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58BE186BD0@cooper>

When you tried to build for a different platform, did you use "clean solution" in the old platform before trying to to do it? I've never tried this myself though, so just an idea.

I've never used the Express version much either (not for a couple of years now), but in the other versions it works fine for me, both in VS2008 and VS2010. Also, chances are that this works in latest git, the 1.0.7 release is pretty old by now. You should always use latest git.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Andrew Keil [andrew.keil at askinteractive.net]
Skickat: den 30 mars 2011 06:07
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Freeswitch latest source rebuild issue under Windows

Jeff,

I have tried the build again with no luck - same main fatal error as before.

Followed from http://wiki.freeswitch.org/wiki/Installation_for_Windows

Note that building for Windows within a tree that has previously been built for a different platform will result in numerous errors and build failures. To resolve, delete the following generated files:
 libs/apr/include/apr.h
libs/js/config.h
libs/js/src/jsautocfg.h
libs/js/nsprpub/pr/include/prcpucfg.h
libs/iksemel/include/config.h
libs/xmlrpc/xmlrpc_config.h
libs/libsndfile/src/sfconfig.h; also rename libs/win32/libsndfile/config.h to libs/win32/libsndfile/sfconfig.h  {Note: this line is the only one that I actually did anything - since the rest of the files do not exist}
 libs/sofia-sip/libsofia-sip-ua/su/sofia-sip/su_configure.h

Then Rebuild the solution.

Tried to rebuild three more times with no joy - same main fatal error as before.  See attached.

Regards,

Andrew Keil

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk
Sent: Wednesday, 30 March 2011 2:31 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Freeswitch latest source rebuild issue under Windows

VS2010 express is fine.

You should not need to install any additional SDK's

As far as I know Express does not support x64 projects which is the warnings you get when you open the solution. You can ignore the warnings.

Your build error is caused by the following line:
c1 : fatal error C1083: Cannot open source file: 'pcre_chartables.c': No such file or directory This is a generated file by the build system which must have failed???

Did you try the build again?

I normally use VS2010 Pro or Ultimate but I will check the build with vs2010 Express again.


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-latest-source-rebuild-issue-under-Windows-tp6221661p6221733.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

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


__________ Information from ESET NOD32 Antivirus, version of virus signature database 5998 (20110329) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com



From chistyakov at directtel.ru  Wed Mar 30 09:17:48 2011
From: chistyakov at directtel.ru (=?UTF-8?B?0KfQuNGB0YLRj9C60L7QsiDQmNCy0LDQvQ==?=)
Date: Wed, 30 Mar 2011 09:17:48 +0400
Subject: [Freeswitch-users] Plz Help me! How to capture a session events by
 setInputCallback ?
Message-ID: <4D92BCFC.4050209@directtel.ru>

This script is work, but only for DTMF.


function my_cb(s, type, obj, arg)

     if (arg) then
        print("type: " .. type .. "\n" .. "arg: " .. arg .. "\n");
     else
        print("type: " .. type .. "\n");
     end

     if (type == "dtmf") then
        print("digit: [" .. obj['digit'] .. "]\nduration: [" ..
obj['duration'] .. "]\n");
     else
        print(obj:serialize("xml"));

     end
return true
end

blah="w00t";
$

local session = freeswitch.Session("sofia/internal/Dt002%172.16.0.49");
session:answer();
session:execute("enable_heartbeat", 5);
session:setInputCallback("my_cb", "blah");
session:streamFile("/tmp/1.wav");




From all.eforums at gmail.com  Wed Mar 30 11:09:24 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Wed, 30 Mar 2011 03:09:24 -0400
Subject: [Freeswitch-users] Build on solaris fails in libs/esl
In-Reply-To: <6ADAA4B6-802B-49F6-8C52-6BBCDE176F16@seventhsignal.de>
References: <AANLkTimGtXWdRPAv8OJYWeu8VYtCF8ZfQt0osGVgU7G=@mail.gmail.com>
	<AANLkTinab+WgVXUYtWaSccbwBV7G7LO-qjjcE4bYAKXj@mail.gmail.com>
	<AANLkTinAtxnnmbVm-tBvEuOxUQ7=z2JebB9TK+0WdBuF@mail.gmail.com>
	<AANLkTim77j3aBU0YrR8YGZEp=bDf9ND3-T0NXSznW4ov@mail.gmail.com>
	<AANLkTikJ4bYiWcsU5uLjtV4pfcL_LV1bQbqUNsfAAZu0@mail.gmail.com>
	<6ADAA4B6-802B-49F6-8C52-6BBCDE176F16@seventhsignal.de>
Message-ID: <AANLkTi=ksjc+aeGHfnvbnzGhPUvkyxWCXb-Gn5wvDD0y@mail.gmail.com>

On Tue, Mar 29, 2011 at 11:33 AM, Michal Bielicki <
michal.bielicki at seventhsignal.de> wrote:

> Its a problem with the build scripts for esl. looking at that right now ..
>
>
Hi Michal,

did you find anything. I did see your later post which said that all
settings in automake are missing? Is this just a case with my installation
or is it missing in the source in git? I'm assuming it's only the settings
that affect Solaris? I'm still confused how you get it to build all the time
although like I'd said, I noticed you don't build any of the problem modules
like hash, esl, silk and a few others I get problems with.
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From dujinfang at gmail.com  Wed Mar 30 11:52:13 2011
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 30 Mar 2011 15:52:13 +0800
Subject: [Freeswitch-users] Videos in FS questions
Message-ID: <05A231906A2F4E6381AC28429404A6E5@gmail.com>

Hi,

I finally got a chance to play with videos in FS. I read some discussions of mod_fsv and mod_mp4, and I haven't try mod_mp4 but I think video passthrough is enough for me for now.

Videos in conference.

In my test video is sort of works, it seems that all parties see the video of the current speaker ( who is in control of floor). I didn't find much info about floor only in a email said it's auto selected and we have not much use about that param. So my questions are:

- is it possible to lock the video of one speaker? so it's like a broadcast conference where all seeing and listening to one speaker? I think we can get that by mute all others but it would be good if we can control

- if we can lock one speaker's video, can we manually switch to another speaker's video?

- If we could use sth. like mod_mp4, is it possible to mix videos lively? I guess it would be a performance killer.

- how to record videos in a conference? is it possible to conference xxx dial loopback/9193 to record video ? (I haven't try that).

I'd like to work on a patch to make the above work if not already supported. Also, other ideas about cool things with video in FS are welcome.

Talk to flash

I heard someone started working on a mod_fmtp like things, how's the progress? I'd like to take a look maybe I can help working on that. I now there's already some sip-flash gateways out there but seems that only adobe fms supports rtfmp:

http://askmeflash.com/article/10/comparison-wowza-vs-fms-vs-red5


Thanks,
7.

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

Sent with Sparrow
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From andrew.keil at askinteractive.net  Wed Mar 30 11:41:09 2011
From: andrew.keil at askinteractive.net (Andrew Keil)
Date: Wed, 30 Mar 2011 18:41:09 +1100
Subject: [Freeswitch-users] Freeswitch latest source rebuild issue under
 Windows
Message-ID: <C42A22554C0CF44DAC0AA1F132B86DE65716763FA6@ASKSERVER.asksbs.local>

Jeff,

Just tried the latest (git head).  Note: autocrlf option is set to false (tested via: git config --get-all core.autocrlf)

Almost there.  The original fatal error has gone now.

The only error left is the following:

------ Build started: Project: mod_managed, Configuration: Debug_CLR Win32 ------
  freeswitch_managed.cpp
freeswitch_managed.cpp : fatal error C1192: #using failed on 'C:\FreeSWITCH\Win32\Debug\mod\FreeSWITCH.Managed.dll'
          'The system cannot find the file specified.'
  freeswitch_wrap.2010.cxx
freeswitch_wrap.2010.cxx : fatal error C1192: #using failed on 'C:\FreeSWITCH\Win32\Debug\mod\FreeSWITCH.Managed.dll'
          'The system cannot find the file specified.'
  mod_managed.cpp
mod_managed.cpp : fatal error C1192: #using failed on 'C:\FreeSWITCH\Win32\Debug\mod\FreeSWITCH.Managed.dll'
          'The system cannot find the file specified.'
  Running Code Analysis for C/C++...
  Generating Code...

I tried to rebuild a few times, however the same error above happened.  Any ideas?

See attached for my build output log.

Regards,

Andrew Keil



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk
Sent: Wednesday, 30 March 2011 3:39 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Freeswitch latest source rebuild issue under Windows

Please try git head just submitted fixes

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-latest-source-rebuild-issue-under-Windows-tp6221661p6221815.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

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

__________ Information from ESET NOD32 Antivirus, version of virus signature database 5998 (20110329) __________

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From latysheff at gmail.com  Wed Mar 30 12:05:10 2011
From: latysheff at gmail.com (Vladimir Latyshev)
Date: Wed, 30 Mar 2011 12:05:10 +0400
Subject: [Freeswitch-users] mod_skypopen produces snaps
Message-ID: <AANLkTikS7asjH3akOoXcLdMizDNYySKv7fScCTPM=m81@mail.gmail.com>

Hi everybody!
mod_skypopen produces snaps every 20 sec when incoming skype call directed
to playback application.
Clearly, this happens because of zeroing read and write buffers, as it's
seen in source code. How can this problem be avoided?
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From fieldpeak at gmail.com  Wed Mar 30 13:38:31 2011
From: fieldpeak at gmail.com (Charles)
Date: Wed, 30 Mar 2011 17:38:31 +0800
Subject: [Freeswitch-users] How to check the FS source code version on
	Windows platform
Message-ID: <4d92fa1c.4a2c2b0a.0a54.1bbf@mx.google.com>


Can anyone help advise how to check the exact version of Freeswitch on windows platform? 
I can see the info like screenshot below in the properties of FS folder which git from latest source, it the correct way? 
if yes, the version should call 'Git-Head: d5ef86d7788ef0080ca3be7e2ff39bda989d4b4d ', however, it looks very strange...
thanks.




2011-03-30 



Charles 
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From mitch.capper at gmail.com  Wed Mar 30 13:53:29 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 30 Mar 2011 02:53:29 -0700
Subject: [Freeswitch-users] Freeswitch latest source rebuild issue under
	Windows
In-Reply-To: <C42A22554C0CF44DAC0AA1F132B86DE65716763FA6@ASKSERVER.asksbs.local>
References: <C42A22554C0CF44DAC0AA1F132B86DE65716763FA6@ASKSERVER.asksbs.local>
Message-ID: <AANLkTinLmBcybYuz1EyEFOjHV7bdd3DtcfeLVrP=ToZB@mail.gmail.com>

Hi Andrew,
It is saying it can't find the managed dll.  Visual Studio Express
comes in different versions with there being a different version for
c#.   I am not sure if you can build the c# module from the c++
edition, although if you don't need .net/mod_managed support you may
be able to just not compile/use that module.

~Mitch

On Wed, Mar 30, 2011 at 12:41 AM, Andrew Keil
<andrew.keil at askinteractive.net> wrote:
> Jeff,
>
> Just tried the latest (git head). ?Note: autocrlf option is set to false (tested via: git config --get-all core.autocrlf)
>
> Almost there. ?The original fatal error has gone now.
>
> The only error left is the following:
>
> ------ Build started: Project: mod_managed, Configuration: Debug_CLR Win32 ------
> ?freeswitch_managed.cpp
> freeswitch_managed.cpp : fatal error C1192: #using failed on 'C:\FreeSWITCH\Win32\Debug\mod\FreeSWITCH.Managed.dll'
> ? ? ? ? ?'The system cannot find the file specified.'
> ?freeswitch_wrap.2010.cxx
> freeswitch_wrap.2010.cxx : fatal error C1192: #using failed on 'C:\FreeSWITCH\Win32\Debug\mod\FreeSWITCH.Managed.dll'
> ? ? ? ? ?'The system cannot find the file specified.'
> ?mod_managed.cpp
> mod_managed.cpp : fatal error C1192: #using failed on 'C:\FreeSWITCH\Win32\Debug\mod\FreeSWITCH.Managed.dll'
> ? ? ? ? ?'The system cannot find the file specified.'
> ?Running Code Analysis for C/C++...
> ?Generating Code...
>
> I tried to rebuild a few times, however the same error above happened. ?Any ideas?
>
> See attached for my build output log.
>
> Regards,
>
> Andrew Keil
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk
> Sent: Wednesday, 30 March 2011 3:39 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Freeswitch latest source rebuild issue under Windows
>
> Please try git head just submitted fixes
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-latest-source-rebuild-issue-under-Windows-tp6221661p6221815.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 5998 (20110329) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From andrew.keil at askinteractive.net  Wed Mar 30 14:25:40 2011
From: andrew.keil at askinteractive.net (Andrew Keil)
Date: Wed, 30 Mar 2011 21:25:40 +1100
Subject: [Freeswitch-users] Freeswitch latest source rebuild issue
	under	Windows
In-Reply-To: <AANLkTinLmBcybYuz1EyEFOjHV7bdd3DtcfeLVrP=ToZB@mail.gmail.com>
References: <C42A22554C0CF44DAC0AA1F132B86DE65716763FA6@ASKSERVER.asksbs.local>
	<AANLkTinLmBcybYuz1EyEFOjHV7bdd3DtcfeLVrP=ToZB@mail.gmail.com>
Message-ID: <C42A22554C0CF44DAC0AA1F132B86DE65716763FA9@ASKSERVER.asksbs.local>

Mitch,

Well spotted!  I simply unchecked mod_managed from the build list and now nothing fails!

I usually use Visual Studio Pro and completely forgot about express being split into C++, C# etc....

Perhaps the Windows installation documentation (http://wiki.freeswitch.org/wiki/Installation_for_Windows) on freeswitch.org should be updated to include this one.  Especially since it mentions the Express version and only mentions VC++ Express.

Now time to do some basic testing.

Thanks,

Andrew Keil


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitch Capper
Sent: Wednesday, 30 March 2011 8:53 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Freeswitch latest source rebuild issue under Windows

Hi Andrew,
It is saying it can't find the managed dll.  Visual Studio Express comes in different versions with there being a different version for
c#.   I am not sure if you can build the c# module from the c++
edition, although if you don't need .net/mod_managed support you may be able to just not compile/use that module.

~Mitch

On Wed, Mar 30, 2011 at 12:41 AM, Andrew Keil <andrew.keil at askinteractive.net> wrote:
> Jeff,
>
> Just tried the latest (git head). ?Note: autocrlf option is set to 
> false (tested via: git config --get-all core.autocrlf)
>
> Almost there. ?The original fatal error has gone now.
>
> The only error left is the following:
>
> ------ Build started: Project: mod_managed, Configuration: Debug_CLR 
> Win32 ------
> ?freeswitch_managed.cpp
> freeswitch_managed.cpp : fatal error C1192: #using failed on 'C:\FreeSWITCH\Win32\Debug\mod\FreeSWITCH.Managed.dll'
> ? ? ? ? ?'The system cannot find the file specified.'
> ?freeswitch_wrap.2010.cxx
> freeswitch_wrap.2010.cxx : fatal error C1192: #using failed on 'C:\FreeSWITCH\Win32\Debug\mod\FreeSWITCH.Managed.dll'
> ? ? ? ? ?'The system cannot find the file specified.'
> ?mod_managed.cpp
> mod_managed.cpp : fatal error C1192: #using failed on 'C:\FreeSWITCH\Win32\Debug\mod\FreeSWITCH.Managed.dll'
> ? ? ? ? ?'The system cannot find the file specified.'
> ?Running Code Analysis for C/C++...
> ?Generating Code...
>
> I tried to rebuild a few times, however the same error above happened. ?Any ideas?
>
> See attached for my build output log.
>
> Regards,
>
> Andrew Keil
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of 
> Jeff Lenk
> Sent: Wednesday, 30 March 2011 3:39 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Freeswitch latest source rebuild issue 
> under Windows
>
> Please try git head just submitted fixes
>
> --
> View this message in context: 
> http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-latest-source
> -rebuild-issue-under-Windows-tp6221661p6221815.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus 
> signature database 5998 (20110329) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>
>

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

__________ Information from ESET NOD32 Antivirus, version of virus signature database 5999 (20110330) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
 


From gmaruzz at gmail.com  Wed Mar 30 15:15:28 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 30 Mar 2011 13:15:28 +0200
Subject: [Freeswitch-users] mod_skypopen produces snaps
In-Reply-To: <AANLkTikS7asjH3akOoXcLdMizDNYySKv7fScCTPM=m81@mail.gmail.com>
References: <AANLkTikS7asjH3akOoXcLdMizDNYySKv7fScCTPM=m81@mail.gmail.com>
Message-ID: <AANLkTi=XbGj9mwQirYQg9tVpwWXqd-R-Zp2tKV5M5n3W@mail.gmail.com>

On Wed, Mar 30, 2011 at 10:05 AM, Vladimir Latyshev <latysheff at gmail.com>wrote:

> Hi everybody!
> mod_skypopen produces snaps every 20 sec when incoming skype call directed
> to playback application.
> Clearly, this happens because of zeroing read and write buffers, as it's
> seen in source code. How can this problem be avoided?
>
> Hi Vladimir,

never heard of this problem before, and is like three years that we're
zeroing buffers.

What platform, sound driver, skype client version, code version, operating
system, hardware, etc are you using?

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From dujinfang at gmail.com  Wed Mar 30 15:31:17 2011
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 30 Mar 2011 19:31:17 +0800
Subject: [Freeswitch-users] video problem in conference with H264
Message-ID: <AD0F952609504D9D8E8AC968ADABE0C8@gmail.com>

I tested with default 3000 conference and it just OK. But I have problem on H264.

I tested with one Bria 3.1 on Mac and two XTP8886 hardware phones.

http://www.gvscusa.com/xtp8886.html

Bria 1003
XTP 1011/1012

call from 1003 to 1011 and from 1011 to 1003 both ok with videos.

http://pastebin.freeswitch.org/15910
http://pastebin.freeswitch.org/15911

When 3 phones calling into 3000(conference), Everyone call see Bria(1003), but no one can say 1011 and 1012. Even when I muted 1003. 

http://pastebin.freeswitch.org/15913

As I said there's no problem with similar test with h263.

Can anyone help take a look, thanks.

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

Sent with Sparrow
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From dujinfang at gmail.com  Wed Mar 30 15:42:39 2011
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 30 Mar 2011 19:42:39 +0800
Subject: [Freeswitch-users] Videos in FS questions
In-Reply-To: <05A231906A2F4E6381AC28429404A6E5@gmail.com>
References: <05A231906A2F4E6381AC28429404A6E5@gmail.com>
Message-ID: <3FA4FC94587F44C1B08BD4FC8458EAD3@gmail.com>

add calling in from 2 xtp8886 phones, no video on both phones.

http://pastebin.freeswitch.org/15914


but they calling each other just fine.

Thanks,
7.

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

Sent with Sparrow
On Wednesday, March 30, 2011 at 3:52 PM, Seven Du wrote: 
> Hi,
> 
> I finally got a chance to play with videos in FS. I read some discussions of mod_fsv and mod_mp4, and I haven't try mod_mp4 but I think video passthrough is enough for me for now.
> 
> Videos in conference.
> 
> In my test video is sort of works, it seems that all parties see the video of the current speaker ( who is in control of floor). I didn't find much info about floor only in a email said it's auto selected and we have not much use about that param. So my questions are:
> 
> - is it possible to lock the video of one speaker? so it's like a broadcast conference where all seeing and listening to one speaker? I think we can get that by mute all others but it would be good if we can control
> 
> - if we can lock one speaker's video, can we manually switch to another speaker's video?
> 
> - If we could use sth. like mod_mp4, is it possible to mix videos lively? I guess it would be a performance killer.
> 
> - how to record videos in a conference? is it possible to conference xxx dial loopback/9193 to record video ? (I haven't try that).
> 
> I'd like to work on a patch to make the above work if not already supported. Also, other ideas about cool things with video in FS are welcome.
> 
> Talk to flash
> 
> I heard someone started working on a mod_fmtp like things, how's the progress? I'd like to take a look maybe I can help working on that. I now there's already some sip-flash gateways out there but seems that only adobe fms supports rtfmp:
> 
> http://askmeflash.com/article/10/comparison-wowza-vs-fms-vs-red5
> 
> 
> Thanks,
> 7.
> 
> -- 
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj: http://www.freeswitch.org.cn
> 
> Sent with Sparrow
> 
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From gmaruzz at gmail.com  Wed Mar 30 16:18:28 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 30 Mar 2011 14:18:28 +0200
Subject: [Freeswitch-users] mod_skypopen produces snaps
In-Reply-To: <AANLkTim9JNFH_5stBXmYoW96yewKcrvkZMx0SeF16YHA@mail.gmail.com>
References: <AANLkTikS7asjH3akOoXcLdMizDNYySKv7fScCTPM=m81@mail.gmail.com>
	<AANLkTi=XbGj9mwQirYQg9tVpwWXqd-R-Zp2tKV5M5n3W@mail.gmail.com>
	<AANLkTim9JNFH_5stBXmYoW96yewKcrvkZMx0SeF16YHA@mail.gmail.com>
Message-ID: <AANLkTinWuw8zMgpw3Uei9zRpj4e5XvdaRqrek3ecvNKn@mail.gmail.com>

On Wed, Mar 30, 2011 at 1:53 PM, Vladimir Latyshev <latysheff at gmail.com>wrote:

> Giovanni, thanks for developing this module!
> I use Windows 7, mod_skypopen compiled with VS2008, precompiled win32 FS
> git, Skype 4 and 5, and I guess any (tried many different for other
> reasons).
> wav file parameters may post a bit later..
>
>
I can't check this until next week because I'm away from my office and I
have no way to access a windows box. With Vista and XP I never had this
problem, but maybe it has surfaced recently, I don't know for sure.

In the mean time you can do the following:

comment out the code in skypopen_protocol.c from line 452 to line 566,
included.

This will avoid the zeroing of the buffers every 20 seconds.

-giovanni


> 2011/3/30 Giovanni Maruzzelli <gmaruzz at gmail.com>
>
>>
>> On Wed, Mar 30, 2011 at 10:05 AM, Vladimir Latyshev <latysheff at gmail.com>wrote:
>>
>>> Hi everybody!
>>> mod_skypopen produces snaps every 20 sec when incoming skype call
>>> directed to playback application.
>>> Clearly, this happens because of zeroing read and write buffers, as it's
>>> seen in source code. How can this problem be avoided?
>>>
>>> Hi Vladimir,
>>
>> never heard of this problem before, and is like three years that we're
>> zeroing buffers.
>>
>> What platform, sound driver, skype client version, code version, operating
>> system, hardware, etc are you using?
>>
>> -giovanni
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>>
>


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From david.villasmil.work at gmail.com  Wed Mar 30 16:26:55 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 30 Mar 2011 14:26:55 +0200
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <AANLkTikizxLZzMyBH3FRrUNCNhpdAFkprXdRMUrRKEi9@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTim0vpXhg16p6LTwbYa7ng5Ki-5-aw=MpfZOT3pY@mail.gmail.com>
	<AANLkTimoAGNBvi4mDjU7_DsoTsyanNVVjSDfwOWUr1eJ@mail.gmail.com>
	<AANLkTikizxLZzMyBH3FRrUNCNhpdAFkprXdRMUrRKEi9@mail.gmail.com>
Message-ID: <AANLkTikM=eELSjUAaD2nFX85s3Bx7xXCRrFNSJL7Vt=u@mail.gmail.com>

How you doing guys!

Sorry for my not replying before but i was out of the country.

I will be finishing the app between this and the next week :)

I WILL be needing some help uploading and working wirh git, as i've never
used it before save for downloading.


Thanks

On Fri, Mar 18, 2011 at 11:16 PM, budi wibowo <bwibowo at gmail.com> wrote:

> any update for this matter?
>
>
> thx
>
> budi
>
>
> On Sat, Mar 5, 2011 at 12:45 AM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Hello,
>>
>> No, i haven't uploaded it yet (i still don't know how); as it is still not
>> finished. I might be doing so in a week or so.
>>
>> Cheers
>>
>> david
>>
>>
>> On Fri, Mar 4, 2011 at 12:45 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> Hi - I didn't notice this in my latest git contrib pull today. Did you
>>> get the access worked out?
>>> -Avi
>>>
>>> On Sun, Feb 27, 2011 at 10:06 PM, Saeed Ahmed <saeedahmad1981 at gmail.com>wrote:
>>>
>>>> press sent too quick..
>>>>
>>>> what did you use for routing? curl? esl?
>>>>
>>>> did you use nibble bill for prepaid app?
>>>>
>>>>
>>>> On Sun, Feb 27, 2011 at 9:05 PM, Saeed Ahmed <saeedahmad1981 at gmail.com>wrote:
>>>>
>>>>> Great!
>>>>>
>>>>> want to see it soon.
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gmaruzz at gmail.com  Wed Mar 30 16:27:03 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 30 Mar 2011 14:27:03 +0200
Subject: [Freeswitch-users] mod_skypopen produces snaps
In-Reply-To: <AANLkTinWuw8zMgpw3Uei9zRpj4e5XvdaRqrek3ecvNKn@mail.gmail.com>
References: <AANLkTikS7asjH3akOoXcLdMizDNYySKv7fScCTPM=m81@mail.gmail.com>
	<AANLkTi=XbGj9mwQirYQg9tVpwWXqd-R-Zp2tKV5M5n3W@mail.gmail.com>
	<AANLkTim9JNFH_5stBXmYoW96yewKcrvkZMx0SeF16YHA@mail.gmail.com>
	<AANLkTinWuw8zMgpw3Uei9zRpj4e5XvdaRqrek3ecvNKn@mail.gmail.com>
Message-ID: <AANLkTimC0545qxHGkHZ4BwWODmJFcDjHOTrVy5-zCsgB@mail.gmail.com>

Maybe you right in the mail you sent me privately: this can have something
to do with the new sound subsystem of Windows. Check the soundcard settings,
and the settings of Skype client. Use something like "skype has exclusive
use of the soundcard" or something similar.
Let us know how it goes.
Also, would be nice if you open a Jira issue on this one, so it's easier to
track.
Go to http://jira.freeswitch.org

-giovanni

On Wed, Mar 30, 2011 at 2:18 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> On Wed, Mar 30, 2011 at 1:53 PM, Vladimir Latyshev <latysheff at gmail.com>wrote:
>
>> Giovanni, thanks for developing this module!
>> I use Windows 7, mod_skypopen compiled with VS2008, precompiled win32 FS
>> git, Skype 4 and 5, and I guess any (tried many different for other
>> reasons).
>> wav file parameters may post a bit later..
>>
>>
> I can't check this until next week because I'm away from my office and I
> have no way to access a windows box. With Vista and XP I never had this
> problem, but maybe it has surfaced recently, I don't know for sure.
>
> In the mean time you can do the following:
>
> comment out the code in skypopen_protocol.c from line 452 to line 566,
> included.
>
> This will avoid the zeroing of the buffers every 20 seconds.
>
> -giovanni
>
>
>> 2011/3/30 Giovanni Maruzzelli <gmaruzz at gmail.com>
>>
>>>
>>> On Wed, Mar 30, 2011 at 10:05 AM, Vladimir Latyshev <latysheff at gmail.com
>>> > wrote:
>>>
>>>> Hi everybody!
>>>> mod_skypopen produces snaps every 20 sec when incoming skype call
>>>> directed to playback application.
>>>> Clearly, this happens because of zeroing read and write buffers, as it's
>>>> seen in source code. How can this problem be avoided?
>>>>
>>>> Hi Vladimir,
>>>
>>> never heard of this problem before, and is like three years that we're
>>> zeroing buffers.
>>>
>>> What platform, sound driver, skype client version, code version,
>>> operating system, hardware, etc are you using?
>>>
>>> -giovanni
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>>
>>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From david.villasmil.work at gmail.com  Wed Mar 30 16:21:37 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 30 Mar 2011 14:21:37 +0200
Subject: [Freeswitch-users] XML parser bug
Message-ID: <AANLkTi=2zkLLKyLg9qaKDsF=y_oKqUEo_ErKwvEpgkvM@mail.gmail.com>

Hello,

I noticed the following:

I have my sofia.conf.xml like this:

<configuration name="sofia.conf" description="sofia Endpoint">

  <global_settings>
    <param name="log-level" value="0"/>

    <!-- <param name="auto-restart" value="false"/> -->

    <param name="debug-presence" value="0"/>
  </global_settings>

  <!--
           The rabbit hole goes deep.  This includes all the
           profiles in the sip_profiles directory that is up
           one level from this directory.
       <profiles>
         <X-PRE-PROCESS cmd="include" data="../sip_profiles/*.xml" />
       </profiles>
  -->

</configuration>

when I start FS, latest GIT:
freeswitch -version
FreeSWITCH version: 1.0.head (git-7e52acf 2011-03-28 22-18-47 -0500)

I get the following output:

./freeswitch -waste
WARNING: Wasting up to 8 megs of memory per thread.
2011-03-30 14:02:23.200097 [INFO] switch_event.c:615 Activate Eventing
Engine.
2011-03-30 14:02:23.211052 [DEBUG] switch_event.c:594 Create event dispatch
thread 0
Cannot Initialize [[error near line 1521]: unclosed <!--]

If i then change my sofia to:

<configuration name="sofia.conf" description="sofia Endpoint">

  <global_settings>
    <param name="log-level" value="0"/>

    <!-- <param name="auto-restart" value="false"/> -->

    <param name="debug-presence" value="0"/>
  </global_settings>

  <!--
           The rabbit hole goes deep.  This includes all the
           profiles in the sip_profiles directory that is up
           one level from this directory.
       <profiles>
       </profiles>
  -->

</configuration>


Please note the absence of:          <X-PRE-PROCESS cmd="include"
data="../sip_profiles/*.xml" />


FS Starts normally!

Is this the correct behaviour? Isn't comments supposed NOT to be read?

Thanks all.



David
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From jmesquita at freeswitch.org  Wed Mar 30 16:34:04 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 30 Mar 2011 09:34:04 -0300
Subject: [Freeswitch-users] XML parser bug
In-Reply-To: <AANLkTi=2zkLLKyLg9qaKDsF=y_oKqUEo_ErKwvEpgkvM@mail.gmail.com>
References: <AANLkTi=2zkLLKyLg9qaKDsF=y_oKqUEo_ErKwvEpgkvM@mail.gmail.com>
Message-ID: <AANLkTimVxhaN4f3TU3Ni9vv0FxjRaYz=H1rGY9aZvFje@mail.gmail.com>

This is not a bug and has been discussed several times on this mailing list.
You can't comment X-PRE-PROCESS tags like that. Make a quick google search
and you'll find several discussions about that including an explanation from
Tony on the subject.

Regards,
Jo?o Mesquita



On Wed, Mar 30, 2011 at 9:21 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello,
>
> I noticed the following:
>
> I have my sofia.conf.xml like this:
>
> <configuration name="sofia.conf" description="sofia Endpoint">
>
>   <global_settings>
>     <param name="log-level" value="0"/>
>
>     <!-- <param name="auto-restart" value="false"/> -->
>
>     <param name="debug-presence" value="0"/>
>   </global_settings>
>
>   <!--
>            The rabbit hole goes deep.  This includes all the
>            profiles in the sip_profiles directory that is up
>            one level from this directory.
>        <profiles>
>          <X-PRE-PROCESS cmd="include" data="../sip_profiles/*.xml" />
>        </profiles>
>   -->
>
> </configuration>
>
> when I start FS, latest GIT:
> freeswitch -version
> FreeSWITCH version: 1.0.head (git-7e52acf 2011-03-28 22-18-47 -0500)
>
> I get the following output:
>
> ./freeswitch -waste
> WARNING: Wasting up to 8 megs of memory per thread.
> 2011-03-30 14:02:23.200097 [INFO] switch_event.c:615 Activate Eventing
> Engine.
> 2011-03-30 14:02:23.211052 [DEBUG] switch_event.c:594 Create event dispatch
> thread 0
> Cannot Initialize [[error near line 1521]: unclosed <!--]
>
> If i then change my sofia to:
>
> <configuration name="sofia.conf" description="sofia Endpoint">
>
>   <global_settings>
>     <param name="log-level" value="0"/>
>
>     <!-- <param name="auto-restart" value="false"/> -->
>
>     <param name="debug-presence" value="0"/>
>   </global_settings>
>
>   <!--
>            The rabbit hole goes deep.  This includes all the
>            profiles in the sip_profiles directory that is up
>            one level from this directory.
>        <profiles>
>        </profiles>
>   -->
>
> </configuration>
>
>
> Please note the absence of:          <X-PRE-PROCESS cmd="include"
> data="../sip_profiles/*.xml" />
>
>
> FS Starts normally!
>
> Is this the correct behaviour? Isn't comments supposed NOT to be read?
>
> Thanks all.
>
>
>
> David
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From robin at swip.net  Wed Mar 30 16:36:53 2011
From: robin at swip.net (Robin Vleij)
Date: Wed, 30 Mar 2011 14:36:53 +0200
Subject: [Freeswitch-users] RTP Proxy mode, DTMF conversion problem
In-Reply-To: <AANLkTin4LE4pUBv0cLsFnm=S86Nsx6Lj9jMEvLGk+dX5@mail.gmail.com>
References: <AANLkTimUvhO65MTg7DaMRbbzO1M5bz4oKg5JfnR0_6uC@mail.gmail.com>	<12CA5F58-378A-4231-9E1E-3F89DD59049B@gmail.com>	<AANLkTikABFQ4hwj9+dkUyKGTU02+dL4gY8DYdRH8RNhz@mail.gmail.com>	<AANLkTikO+U5=A9yLepzeA8-LxggWM2SLcHRDfzfr4ieg@mail.gmail.com>	<AANLkTikH6EKOMzYriW=egWA-dFo0-KckRY=Qsxn-FgoL@mail.gmail.com>	<AANLkTik3=WPdnj_WpE-vK8b8SfEp5NXxyXonL7YT+UN=@mail.gmail.com>
	<AANLkTin4LE4pUBv0cLsFnm=S86Nsx6Lj9jMEvLGk+dX5@mail.gmail.com>
Message-ID: <4D9323E5.3030603@swip.net>

On 02/25/2011 12:07 AM, Anthony Minessale wrote:

Hi!

> you need to run the start_dtmf() application on the leg that has inband DTMF
>
> if it's the outbound leg you need to set it in execute_on_answer in
> the bridge line
>
> <action application="bridge"
> data="{execute_on_answer=start_dtmf}sofia/OPERATOR_2/46111223344"/>

I think he's running into the same kind of problem I'm running into. I 
reported it as FS-2819.

UA (2833) -> FS -> (INBAND) UA

On the b-leg that only supports inband, FS keeps pushing telephone-event 
on the SDP. Even though the UA on the b-leg only replies with 0 or 8. I 
tried with the start_dtmf as written in this thread, but still only 
rtp-events are passed onto the B-side UA, which doesn't know what to do 
with that.

/Robin

-- 
Robin Vleij


From david.villasmil.work at gmail.com  Wed Mar 30 16:54:13 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 30 Mar 2011 14:54:13 +0200
Subject: [Freeswitch-users] XML parser bug
In-Reply-To: <AANLkTimVxhaN4f3TU3Ni9vv0FxjRaYz=H1rGY9aZvFje@mail.gmail.com>
References: <AANLkTi=2zkLLKyLg9qaKDsF=y_oKqUEo_ErKwvEpgkvM@mail.gmail.com>
	<AANLkTimVxhaN4f3TU3Ni9vv0FxjRaYz=H1rGY9aZvFje@mail.gmail.com>
Message-ID: <AANLkTinUhFVvbx364NVyNgC0_jApVyi9E+p8jb_59efw@mail.gmail.com>

Hello Joao,

Ok, thanks

David

2011/3/30 Jo?o Mesquita <jmesquita at freeswitch.org>

> This is not a bug and has been discussed several times on this mailing
> list. You can't comment X-PRE-PROCESS tags like that. Make a quick google
> search and you'll find several discussions about that including an
> explanation from Tony on the subject.
>
> Regards,
> Jo?o Mesquita
>
>
>
> On Wed, Mar 30, 2011 at 9:21 AM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Hello,
>>
>> I noticed the following:
>>
>> I have my sofia.conf.xml like this:
>>
>> <configuration name="sofia.conf" description="sofia Endpoint">
>>
>>   <global_settings>
>>     <param name="log-level" value="0"/>
>>
>>     <!-- <param name="auto-restart" value="false"/> -->
>>
>>     <param name="debug-presence" value="0"/>
>>   </global_settings>
>>
>>   <!--
>>            The rabbit hole goes deep.  This includes all the
>>            profiles in the sip_profiles directory that is up
>>            one level from this directory.
>>        <profiles>
>>          <X-PRE-PROCESS cmd="include" data="../sip_profiles/*.xml" />
>>        </profiles>
>>   -->
>>
>> </configuration>
>>
>> when I start FS, latest GIT:
>> freeswitch -version
>> FreeSWITCH version: 1.0.head (git-7e52acf 2011-03-28 22-18-47 -0500)
>>
>> I get the following output:
>>
>> ./freeswitch -waste
>> WARNING: Wasting up to 8 megs of memory per thread.
>> 2011-03-30 14:02:23.200097 [INFO] switch_event.c:615 Activate Eventing
>> Engine.
>> 2011-03-30 14:02:23.211052 [DEBUG] switch_event.c:594 Create event
>> dispatch thread 0
>> Cannot Initialize [[error near line 1521]: unclosed <!--]
>>
>> If i then change my sofia to:
>>
>> <configuration name="sofia.conf" description="sofia Endpoint">
>>
>>   <global_settings>
>>     <param name="log-level" value="0"/>
>>
>>     <!-- <param name="auto-restart" value="false"/> -->
>>
>>     <param name="debug-presence" value="0"/>
>>   </global_settings>
>>
>>   <!--
>>            The rabbit hole goes deep.  This includes all the
>>            profiles in the sip_profiles directory that is up
>>            one level from this directory.
>>        <profiles>
>>        </profiles>
>>   -->
>>
>> </configuration>
>>
>>
>> Please note the absence of:          <X-PRE-PROCESS cmd="include"
>> data="../sip_profiles/*.xml" />
>>
>>
>> FS Starts normally!
>>
>> Is this the correct behaviour? Isn't comments supposed NOT to be read?
>>
>> Thanks all.
>>
>>
>>
>> David
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.villasmil.work at gmail.com  Wed Mar 30 16:56:29 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 30 Mar 2011 14:56:29 +0200
Subject: [Freeswitch-users] Possible bug commenting X-PRE-PROCESS
In-Reply-To: <925C2C70EF054D6A93A31BA26FCDC2AA@e1705>
References: <A9075FBE-3868-4D96-8103-66501E67859B@mgtech.com>
	<AANLkTikQOJ5EhDc=2FocMTXzOmHq71yKp1+2-3j7MnMC@mail.gmail.com>
	<1CEEC760-86B4-4EF9-865F-693F48EEE670@mgtech.com>
	<925C2C70EF054D6A93A31BA26FCDC2AA@e1705>
Message-ID: <AANLkTin9STZ=52aa=NZ8q2_4KAsZamcYT1SrB2Z+p-b6@mail.gmail.com>

Duh! for a day i was going crazy... obviously didn't read well the comments
in vars.xml

Thanks

David

On Tue, Mar 15, 2011 at 11:35 PM, Madovsky <infos at madovsky.org> wrote:

> i> Wish I knew that, it cost me 3 days....
>
> it's only the start my friend ;)
>
> ----- Original Message -----
> From: "Mario G" <mario_fs at mgtech.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, March 15, 2011 12:36 PM
> Subject: Re: [Freeswitch-users] Possible bug commenting X-PRE-PROCESS
>
>
> > Wish I knew that, it cost me 3 days.... It turned out to be the reason my
> > phones would drop when placing a call on hold. Still have no idea why it
> > affected the SPA962s. I am noting it in case someone else runs across
> > something like this. Thanks for the explanation!
> > Mario G
> >
> > On Mar 15, 2011, at 9:10 AM, Anthony Minessale wrote:
> >
> >> you cannot comment those.
> >> They are not real xml they are designed to be ignored by XML and
> >> parsed by the pre-processor and are parsed before the xml parser.
> >> All the xml parser would see on your example is <!-- -->
> >> if you want to comment them, change X- to Z- or "cmd" to "comment"
> >>
> >>
> >> On Tue, Mar 15, 2011 at 10:33 AM, Mario G <mario_fs at mgtech.com> wrote:
> >>> The line below was in an external profile. I commented it thinking it
> >>> would
> >>> be ignored as such. However, it was still used and the file loaded.
> This
> >>> drive me nuts until I moved the extip.xml file out of the directory and
> >>> FreeSwitch noted errors on startup. The only way was to put a space as
> >>> "X-
> >>> PRE..". I can't image why you would want a comment to be processed as a
> >>> normal statement. Git is 3/13/11.
> >>>  <!--<X-PRE-PROCESS cmd="include" data="extip.xml"/>  -->
> >>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> 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-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mitch.capper at gmail.com  Wed Mar 30 18:19:11 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 30 Mar 2011 07:19:11 -0700
Subject: [Freeswitch-users] Freeswitch latest source rebuild issue under
	Windows
In-Reply-To: <C42A22554C0CF44DAC0AA1F132B86DE65716763FA9@ASKSERVER.asksbs.local>
References: <C42A22554C0CF44DAC0AA1F132B86DE65716763FA6@ASKSERVER.asksbs.local>
	<AANLkTinLmBcybYuz1EyEFOjHV7bdd3DtcfeLVrP=ToZB@mail.gmail.com>
	<C42A22554C0CF44DAC0AA1F132B86DE65716763FA9@ASKSERVER.asksbs.local>
Message-ID: <AANLkTimWpJonmpawmd1oTkREM72BgzV8gV_QU3vBSFUQ@mail.gmail.com>

Thats the great part of the wiki anyone can edit it:)

~Mitch

On Wed, Mar 30, 2011 at 3:25 AM, Andrew Keil
<andrew.keil at askinteractive.net> wrote:
> Mitch,
>
> Well spotted! ?I simply unchecked mod_managed from the build list and now nothing fails!
>
> I usually use Visual Studio Pro and completely forgot about express being split into C++, C# etc....
>
> Perhaps the Windows installation documentation (http://wiki.freeswitch.org/wiki/Installation_for_Windows) on freeswitch.org should be updated to include this one. ?Especially since it mentions the Express version and only mentions VC++ Express.
>
> Now time to do some basic testing.
>
> Thanks,
>
> Andrew Keil
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitch Capper
> Sent: Wednesday, 30 March 2011 8:53 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Freeswitch latest source rebuild issue under Windows
>
> Hi Andrew,
> It is saying it can't find the managed dll. ?Visual Studio Express comes in different versions with there being a different version for
> c#. ? I am not sure if you can build the c# module from the c++
> edition, although if you don't need .net/mod_managed support you may be able to just not compile/use that module.
>
> ~Mitch
>
> On Wed, Mar 30, 2011 at 12:41 AM, Andrew Keil <andrew.keil at askinteractive.net> wrote:
>> Jeff,
>>
>> Just tried the latest (git head). ?Note: autocrlf option is set to
>> false (tested via: git config --get-all core.autocrlf)
>>
>> Almost there. ?The original fatal error has gone now.
>>
>> The only error left is the following:
>>
>> ------ Build started: Project: mod_managed, Configuration: Debug_CLR
>> Win32 ------
>> ?freeswitch_managed.cpp
>> freeswitch_managed.cpp : fatal error C1192: #using failed on 'C:\FreeSWITCH\Win32\Debug\mod\FreeSWITCH.Managed.dll'
>> ? ? ? ? ?'The system cannot find the file specified.'
>> ?freeswitch_wrap.2010.cxx
>> freeswitch_wrap.2010.cxx : fatal error C1192: #using failed on 'C:\FreeSWITCH\Win32\Debug\mod\FreeSWITCH.Managed.dll'
>> ? ? ? ? ?'The system cannot find the file specified.'
>> ?mod_managed.cpp
>> mod_managed.cpp : fatal error C1192: #using failed on 'C:\FreeSWITCH\Win32\Debug\mod\FreeSWITCH.Managed.dll'
>> ? ? ? ? ?'The system cannot find the file specified.'
>> ?Running Code Analysis for C/C++...
>> ?Generating Code...
>>
>> I tried to rebuild a few times, however the same error above happened. ?Any ideas?
>>
>> See attached for my build output log.
>>
>> Regards,
>>
>> Andrew Keil
>>
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Jeff Lenk
>> Sent: Wednesday, 30 March 2011 3:39 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Freeswitch latest source rebuild issue
>> under Windows
>>
>> Please try git head just submitted fixes
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-latest-source
>> -rebuild-issue-under-Windows-tp6221661p6221815.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>> http://www.freeswitch.org
>>
>>
>> __________ Information from ESET NOD32 Antivirus, version of virus
>> signature database 5998 (20110329) __________
>>
>> The message was checked by ESET NOD32 Antivirus.
>>
>> http://www.eset.com
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 5999 (20110330) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From steveayre at gmail.com  Wed Mar 30 18:26:26 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 30 Mar 2011 15:26:26 +0100
Subject: [Freeswitch-users] How to check the FS source code version on
 Windows platform
In-Reply-To: <4d92fa1c.4a2c2b0a.0a54.1bbf@mx.google.com>
References: <4d92fa1c.4a2c2b0a.0a54.1bbf@mx.google.com>
Message-ID: <AANLkTikBwf=QXzr1dT2=_5rLXCnB_XYY0Eu1zrvz87e8@mail.gmail.com>

d5ef86d7788ef0080ca3be7e2ff39bda989d4b4d is indeed the version.

Git uses hashes as version number, not a sequential number as with SVN. It
can't do sequential since it's distributed so it avoids having the same
version number at different locations at the same time.

The version when you're at the CLI will be something like:
FreeSWITCH Version 1.0.head (git-b041edc8 2011-03-01 06-35-06 -0600)

Note git-b041edc8 is the version number, it's just the first 8 digits of the
full hash (so it's easier to remember). Yours would be git-d5ef86d7.

The changelog (git log) will show you how recent your version is.

-Steve





On 30 March 2011 10:38, Charles <fieldpeak at gmail.com> wrote:

>
> Can anyone help advise
> how to check the exact version of Freeswitch on windows platform?
> I can see the info like screenshot
> below in the properties of FS folder which git from latest source, it the correct way?
>
> if yes, the version should call 'Git-Head:
> d5ef86d7788ef0080ca3be7e2ff39bda989d4b4d ', however, it looks very
> strange...
> thanks.
>
>
>
> 2011-03-30
> ------------------------------
> Charles
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kbdfck at gmail.com  Wed Mar 30 19:24:05 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Wed, 30 Mar 2011 19:24:05 +0400
Subject: [Freeswitch-users] T.38 reinvite and 488 Inacceptable Here relay
 problem; codec issue with proxy_media=true
Message-ID: <AANLkTinpHst2HgPVkHQ2gASM+2LofYVhXZZwpJO8+34i@mail.gmail.com>

Hi All!

I'm trying to launch t.38 passthough on FS.
I have two sip profiles - local and external, one on customer side looking
to ATAs, and one looking to PSTN gateways.
On both profiles I have enabled t38-passthrough, and this works when T.38 is
enabled on ATA and PSTN gateway.

Problems come when T.38 is turned off on ATA, so PSTN gateway detects fax on
its side and tries to re-invite ATA behind freeswitch with T.38. ATA answers
488 Inacceptable Here, but this answer doesn't get relayed to PSTN gateway,
so while ATA transmits fax signals inband, PSTN gateway still doesn't know
that re-invite is failed and fax machines can't negotiate t.30.

My question is how to allow relay this 488 to another side of bridge?

I thought that maybe proxy_media=true will help, and this is where second
issue comes. I enabled late negotiation and proxy_media, the only codec I
use is PCMA. But when I get incoming call, initially in PCMA and it goes to
outbound ESL app via socket, in the middle of processing I get strange
errors:


EXECUTE sofia/external/1234567 at XXXXX.199 set(continue_on_fail=true)
2011-03-30 18:59:30.172363 [DEBUG] mod_dptools.c:1060
sofia/external/1234567 at XX.XX.XX.199 SET [continue_on_fail]=[true]
EXECUTE sofia/external/1234567 at XXXXX.199 socket(localhost:8006 async full)
*2011-03-30 18:59:30.184605 [DEBUG] switch_ivr.c:766 Codec Activated
L16 at 0hz0 channels 0ms
*
2011-03-30 18:59:30.188640 [DEBUG] switch_core_session.c:954 Send signal
sofia/external/1234567 at XXXXX.199 [BREAK]
2011-03-30 18:59:30.205638 [DEBUG] switch_ivr.c:563 sofia/external/
1234567 at 85.114.2.199 Command Execute limit(hash OutboundExternalLimit 1)
EXECUTE sofia/external/1234567 at XXXXX.199 limit(hash OutboundExternalLimit 1)
2011-03-30 18:59:30.205638 [INFO] switch_limit.c:126 incr called:
OutboundExternalLimit_1 max:-1, interval:0
2011-03-30 18:59:30.205638 [INFO] mod_hash.c:200 Usage for
OutboundExternalLimit_1 is now 1
*2011-03-30 18:59:30.205638 [ERR] switch_core_io.c:724
sofia/external/1234567 at XXXXX.199 has no write codec.*
2011-03-30 18:59:30.205638 [DEBUG] switch_channel.c:2563
(sofia/external/1234567 at XXXXX.199) Callstate Change RINGING -> HANGUP
2011-03-30 18:59:30.205638 [NOTICE] switch_core_io.c:725 Hangup
sofia/external/1234567 at XXXXX.199 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]

I don't answer channel in my script, it doesn't even try to execute bridge,
failing somewhere in the middle.
Is this a bug or misconfiguration of my FS?

I tried latest git FreeSWITCH Version 1.0.head (git-6e78f6f 2011-03-30
11-41-45 +0200) with same result.


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From infos at madovsky.org  Wed Mar 30 20:14:21 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 30 Mar 2011 12:14:21 -0400
Subject: [Freeswitch-users] hupall in cluster
Message-ID: <E8AFD7E4531E4DB5B0465D236BF945BE@e1705>

I'm experimenting how to use hupall from a node to another node in a cluster.
is there a way to do it ?

Thanks
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From lautram.mathieu at gmail.com  Wed Mar 30 20:21:48 2011
From: lautram.mathieu at gmail.com (Mathieu Lautram)
Date: Wed, 30 Mar 2011 18:21:48 +0200
Subject: [Freeswitch-users] stuck channels
Message-ID: <AANLkTimuPUcyEonfie4K=W_0JCmXE6YAvguBScSBQk4Q@mail.gmail.com>

Hi everyone,

I have a problem of stuck channels. Indeed, when I do a show channels in
fs_cli, there is a lot of hangup callstate channels. I even tried to do a
uuid_kill but fs_cli tells me "-ERR No Such Channels"...
So I d'ont know where the problem is. Someone tells me that it could be a
sqlite problem. I hope you could help me because it's been 3 weeks that I'm
on it and I have no solutions...
Here is the log corresponding to my problem:

2011-03-30 15:26:54.866347 [NOTICE] sofia.c:537 Hangup sofia/external/
0142560840 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:26:54.866347 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0142560840 at 192.168.0.1 [KILL]
2011-03-30 15:26:54.866347 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0142560840 at 192.168.0.1 [BREAK]
2011-03-30 15:26:56.268289 [DEBUG] switch_channel.c:2563 (sofia/external/
0134747924 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:26:56.268289 [NOTICE] sofia.c:537 Hangup sofia/external/
0134747924 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:26:56.268289 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0134747924 at 192.168.0.1 [KILL]
2011-03-30 15:26:56.268289 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0134747924 at 192.168.0.1 [BREAK]
2011-03-30 15:27:00.066249 [DEBUG] switch_channel.c:2563 (sofia/external/
0139339445 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:27:00.066249 [NOTICE] sofia.c:537 Hangup sofia/external/
0139339445 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:27:00.066249 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0139339445 at 192.168.0.1 [KILL]
2011-03-30 15:27:00.066249 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0139339445 at 192.168.0.1 [BREAK]
2011-03-30 15:27:00.563121 [DEBUG] switch_channel.c:2563 (sofia/external/
0143708446 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:27:00.563121 [NOTICE] sofia.c:537 Hangup sofia/external/
0143708446 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:27:00.563121 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0143708446 at 192.168.0.1 [KILL]
2011-03-30 15:27:00.563121 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0143708446 at 192.168.0.1 [BREAK]
2011-03-30 15:27:01.376088 [DEBUG] switch_channel.c:2563 (sofia/external/
0148863352 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:27:01.376088 [NOTICE] sofia.c:537 Hangup sofia/external/
0148863352 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:27:01.376088 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0148863352 at 192.168.0.1 [KILL]
2011-03-30 15:27:01.376088 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0148863352 at 192.168.0.1 [BREAK]
2011-03-30 15:27:01.377142 [DEBUG] switch_channel.c:2563 (sofia/external/
0140411621 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:27:01.377142 [NOTICE] sofia.c:537 Hangup sofia/external/
0140411621 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:27:01.377142 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0140411621 at 192.168.0.1 [KILL]
2011-03-30 15:27:01.377142 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0140411621 at 192.168.0.1 [BREAK]
2011-03-30 15:27:04.264971 [DEBUG] switch_channel.c:2563 (sofia/external/
0130510861 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:27:04.264971 [NOTICE] sofia.c:537 Hangup sofia/external/
0130510861 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:27:04.264971 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0130510861 at 192.168.0.1 [KILL]
2011-03-30 15:27:04.264971 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0130510861 at 192.168.0.1 [BREAK]
2011-03-30 15:27:05.165960 [DEBUG] switch_channel.c:2563 (sofia/external/
0134868273 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:27:05.165960 [NOTICE] sofia.c:537 Hangup sofia/external/
0134868273 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:27:05.165960 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0134868273 at 192.168.0.1 [KILL]
2011-03-30 15:27:05.165960 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0134868273 at 192.168.0.1 [BREAK]
2011-03-30 15:27:05.665936 [DEBUG] switch_channel.c:2563 (sofia/external/
0142861137 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:27:05.665936 [NOTICE] sofia.c:537 Hangup sofia/external/
0142861137 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:27:05.665936 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0142861137 at 192.168.0.1 [KILL]
2011-03-30 15:27:05.665936 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0142861137 at 192.168.0.1 [BREAK]
2011-03-30 15:27:05.962900 [DEBUG] switch_channel.c:2563 (sofia/external/
0139880615 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:27:05.962900 [NOTICE] sofia.c:537 Hangup sofia/external/
0139880615 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:27:05.962900 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0139880615 at 192.168.0.1 [KILL]
2011-03-30 15:27:05.962900 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0139880615 at 192.168.0.1 [BREAK]
2011-03-30 15:27:08.964780 [DEBUG] switch_channel.c:2563 (sofia/external/
0140209135 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:27:08.964780 [NOTICE] sofia.c:537 Hangup sofia/external/
0140209135 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:27:08.964780 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0140209135 at 192.168.0.1 [KILL]
2011-03-30 15:27:08.964780 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0140209135 at 192.168.0.1 [BREAK]
2011-03-30 15:27:09.664751 [DEBUG] switch_channel.c:2563 (sofia/external/
0134524074 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:27:09.664751 [NOTICE] sofia.c:537 Hangup sofia/external/
0134524074 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:27:09.664751 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0134524074 at 192.168.0.1 [KILL]
2011-03-30 15:27:09.664751 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0134524074 at 192.168.0.1 [BREAK]
2011-03-30 15:27:09.767745 [DEBUG] switch_channel.c:2563 (sofia/external/
0130326184 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:27:09.767745 [NOTICE] sofia.c:537 Hangup sofia/external/
0130326184 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:27:09.767745 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0130326184 at 192.168.0.1 [KILL]
2011-03-30 15:27:09.767745 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0130326184 at 192.168.0.1 [BREAK]
2011-03-30 15:27:09.864742 [DEBUG] switch_channel.c:2563 (sofia/external/
0139902937 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:27:09.864742 [NOTICE] sofia.c:537 Hangup sofia/external/
0139902937 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:27:09.864742 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0139902937 at 192.168.0.1 [KILL]
2011-03-30 15:27:09.864742 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0139902937 at 192.168.0.1 [BREAK]
2011-03-30 15:27:12.064653 [DEBUG] switch_channel.c:2563 (sofia/external/
0144311010 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:27:12.064653 [NOTICE] sofia.c:537 Hangup sofia/external/
0144311010 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:27:12.064653 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0144311010 at 192.168.0.1 [KILL]
2011-03-30 15:27:12.064653 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0144311010 at 192.168.0.1 [BREAK]
2011-03-30 15:27:12.466637 [DEBUG] switch_channel.c:2563 (sofia/external/
0139199893 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:27:12.466637 [NOTICE] sofia.c:537 Hangup sofia/external/
0139199893 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:27:12.466637 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0139199893 at 192.168.0.1 [KILL]
2011-03-30 15:27:12.466637 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0139199893 at 192.168.0.1 [BREAK]
2011-03-30 15:27:14.062573 [DEBUG] switch_channel.c:2563 (sofia/external/
0142533724 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:27:14.062573 [NOTICE] sofia.c:537 Hangup sofia/external/
0142533724 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:27:14.062573 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0142533724 at 192.168.0.1 [KILL]
2011-03-30 15:27:14.062573 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0142533724 at 192.168.0.1 [BREAK]
2011-03-30 15:27:14.066573 [DEBUG] switch_channel.c:2563 (sofia/external/
0141322567 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:27:14.066573 [NOTICE] sofia.c:537 Hangup sofia/external/
0141322567 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:27:14.066573 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0141322567 at 192.168.0.1 [KILL]
2011-03-30 15:27:14.066573 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0141322567 at 192.168.0.1 [BREAK]
2011-03-30 15:27:16.564476 [DEBUG] switch_channel.c:2563 (sofia/external/
0130344121 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:27:16.564476 [NOTICE] sofia.c:537 Hangup sofia/external/
0130344121 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:27:16.564476 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0130344121 at 192.168.0.1 [KILL]
2011-03-30 15:27:16.564476 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0130344121 at 192.168.0.1 [BREAK]
2011-03-30 15:27:26.464071 [DEBUG] switch_channel.c:2563 (sofia/external/
0139595901 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:27:26.464071 [NOTICE] sofia.c:537 Hangup sofia/external/
0139595901 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:27:26.464071 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0139595901 at 192.168.0.1 [KILL]
2011-03-30 15:27:26.464071 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0139595901 at 192.168.0.1 [BREAK]
2011-03-30 15:27:27.066047 [DEBUG] switch_channel.c:2563 (sofia/external/
0143758609 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:27:27.066047 [NOTICE] sofia.c:537 Hangup sofia/external/
0143758609 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:27:27.066047 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0143758609 at 192.168.0.1 [KILL]
2011-03-30 15:27:27.066047 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0143758609 at 192.168.0.1 [BREAK]
2011-03-30 15:27:29.953931 [DEBUG] switch_channel.c:2563 (sofia/external/
0148831626 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:27:29.953931 [NOTICE] sofia.c:537 Hangup sofia/external/
0148831626 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:27:29.953931 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0148831626 at 192.168.0.1 [KILL]
2011-03-30 15:27:29.953931 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0148831626 at 192.168.0.1 [BREAK]
2011-03-30 15:27:36.965646 [DEBUG] switch_channel.c:2563 (sofia/external/
0140419154 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:27:36.965646 [NOTICE] sofia.c:537 Hangup sofia/external/
0140419154 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:27:36.965646 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0140419154 at 192.168.0.1 [KILL]
2011-03-30 15:27:36.965646 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0140419154 at 192.168.0.1 [BREAK]
2011-03-30 15:27:53.565145 [DEBUG] switch_channel.c:2563 (sofia/external/
0142569040 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:27:53.565145 [NOTICE] sofia.c:537 Hangup sofia/external/
0142569040 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:27:53.565145 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0142569040 at 192.168.0.1 [KILL]
2011-03-30 15:27:53.565145 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0142569040 at 192.168.0.1 [BREAK]
2011-03-30 15:27:59.260817 [DEBUG] switch_channel.c:2563 (sofia/external/
0140261579 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:27:59.260817 [NOTICE] sofia.c:537 Hangup sofia/external/
0140261579 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:27:59.260817 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0140261579 at 192.168.0.1 [KILL]
2011-03-30 15:27:59.260817 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0140261579 at 192.168.0.1 [BREAK]
2011-03-30 15:28:07.062534 [DEBUG] switch_channel.c:2563 (sofia/external/
0134291660 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:28:07.062534 [NOTICE] sofia.c:537 Hangup sofia/external/
0134291660 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:28:07.062534 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0134291660 at 192.168.0.1 [KILL]
2011-03-30 15:28:07.062534 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0134291660 at 192.168.0.1 [BREAK]
2011-03-30 15:28:08.761352 [DEBUG] switch_channel.c:2563 (sofia/external/
0142335551 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:28:08.761352 [NOTICE] sofia.c:537 Hangup sofia/external/
0142335551 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:28:08.761352 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0142335551 at 192.168.0.1 [KILL]
2011-03-30 15:28:08.761352 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0142335551 at 192.168.0.1 [BREAK]
2011-03-30 15:28:26.160649 [DEBUG] switch_channel.c:2563 (sofia/external/
0145932250 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:28:26.160649 [NOTICE] sofia.c:537 Hangup sofia/external/
0145932250 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:28:26.160649 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0145932250 at 192.168.0.1 [KILL]
2011-03-30 15:28:26.160649 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0145932250 at 192.168.0.1 [BREAK]
2011-03-30 15:28:31.958419 [DEBUG] switch_channel.c:2563 (sofia/external/
0146377247 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-03-30 15:28:31.958419 [NOTICE] sofia.c:537 Hangup sofia/external/
0146377247 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2011-03-30 15:28:31.958419 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/0146377247 at 192.168.0.1 [KILL]
2011-03-30 15:28:31.958419 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/0146377247 at 192.168.0.1 [BREAK]

freeswitch at internal>
freeswitch at internal>
freeswitch at internal> show channels
uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid
7c91effe-7af7-47cc-a9c2-5a8d7a13e070,outbound,2011-03-30
15:24:18,1301491458,sofia/external/0146426664 at 192.168.0.1
,CS_EXECUTE,,0000000000,,0146426664,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,0146426664,0146426664,RECV,d2ea58aa-e3a4-496e-a031-ac501910d18e
19b44e66-0cc8-4bf2-b8e4-8587986c4e22,outbound,2011-03-30
15:24:46,1301491486,sofia/external/0140675411 at 192.168.0.1,CS_EXECUTE,,0000000000,,0140675411,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0140675411,RECV,472fb609-8d90-4d7d-bc0f-7353bb01139f
4dca35aa-589b-4ba8-9efa-f7510514385b,outbound,2011-03-30
15:25:02,1301491502,sofia/external/0134291660 at 192.168.0.1,CS_EXECUTE,,0000000000,,0134291660,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0134291660,RECV,f62c2f5d-0b17-4e1e-8f5c-9c495407e95e
309fd56c-9d4b-4904-be94-8803cc757dd3,outbound,2011-03-30
15:25:08,1301491508,sofia/external/0146377247 at 192.168.0.1,CS_EXECUTE,,0000000000,,0146377247,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0146377247,RECV,cdcfd0eb-5efc-4541-9abd-845abfa63fa9
384fdd12-908c-4ad5-a488-a67f25b253ed,outbound,2011-03-30
15:25:20,1301491520,sofia/external/0141955961 at 192.168.0.1,CS_EXECUTE,,0000000000,,0141955961,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0141955961,RECV,f1a4da83-7ff9-478b-9323-0b9a69104ea2
66ba4d1c-a8de-4a5f-8da3-53eea2e353be,outbound,2011-03-30
15:25:22,1301491522,sofia/external/0146770127 at 192.168.0.1,CS_EXECUTE,,0000000000,,0146770127,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0146770127,RECV,50a004d4-bb3e-4baf-afe5-a9eeb5a58cfc
398d2132-7f7a-4ddf-95af-fe8afcb2a253,outbound,2011-03-30
15:25:23,1301491523,sofia/external/0144598175 at 192.168.0.1,CS_EXECUTE,,0000000000,,0144598175,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound
Call,0144598175,RECV,98170d54-5c3d-40be-8d20-30da9a478730
2b90e1ba-16c4-4d48-bc4f-90c753a9244e,outbound,2011-03-30
15:25:23,1301491523,sofia/external/0130310549 at 192.168.0.1,CS_EXECUTE,,0000000000,,0130310549,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0130310549,RECV,5a3cba1e-f559-4116-93d8-3cca45e3084b
13671770-a92a-4f18-98bb-e8488d926cdb,outbound,2011-03-30
15:25:25,1301491525,sofia/external/0148863352 at 192.168.0.1,CS_EXECUTE,,0000000000,,0148863352,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0148863352,RECV,0c01dc58-90ed-408f-a574-1091c35b2c13
6da9d0de-ef0d-4dfc-85bf-d771b3c6f098,outbound,2011-03-30
15:25:25,1301491525,sofia/external/0146682550 at 192.168.0.1,CS_EXECUTE,,0000000000,,0146682550,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound
Call,0146682550,RECV,e8754f05-8870-4b25-8f7a-107ee129b1e2
cb332c48-443b-4326-86b4-ea3d89346fef,outbound,2011-03-30
15:25:25,1301491525,sofia/external/0142378320 at 192.168.0.1,CS_EXECUTE,,0000000000,,0142378320,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0142378320,RECV,0759c1a2-9f3a-4259-9e3a-7f567318cf67
4f960d60-8079-4225-bea2-58f8ccba67f5,outbound,2011-03-30
15:25:26,1301491526,sofia/external/0148521177 at 192.168.0.1,CS_EXECUTE,,0000000000,,0148521177,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound
Call,0148521177,RECV,37b401ba-3077-43b8-bf8a-ee3a836ac843
c03c2320-1ddb-40b0-884b-d4b07ac4d189,outbound,2011-03-30
15:25:27,1301491527,sofia/external/0142569040 at 192.168.0.1,CS_EXECUTE,,0000000000,,0142569040,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0142569040,RECV,1be9a468-b720-4398-b8c6-621c52aad6ac
b4a6ef35-79d3-4520-9574-1c37c2fecec7,outbound,2011-03-30
15:25:28,1301491528,sofia/external/0140411621 at 192.168.0.1
,CS_EXECUTE,,0000000000,,0140411621,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,0140411621,0140411621,RECV,81e14c30-0b77-46e6-96a4-ae8a0678e6c5
67b126e7-2a5f-4b99-9843-ee71e2c08c10,outbound,2011-03-30
15:25:35,1301491535,sofia/external/0139745628 at 192.168.0.1,CS_EXECUTE,,0000000000,,0139745628,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound
Call,0139745628,RECV,1b173b0d-69c6-40e0-8b71-1496106cbead
dbd23a86-64bf-417d-bb77-6b4a927b9efd,outbound,2011-03-30
15:25:35,1301491535,sofia/external/0139595901 at 192.168.0.1,CS_EXECUTE,,0000000000,,0139595901,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0139595901,RECV,395e8762-4f70-48f3-b8b5-b5040a2e5fbe
dfe644c4-d453-4c39-93bd-34f7f3e9b570,outbound,2011-03-30
15:25:36,1301491536,sofia/external/0142335551 at 192.168.0.1,CS_EXECUTE,,0000000000,,0142335551,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0142335551,RECV,51995424-c0f4-4252-b4ad-3b4ae7fcdad6
96a82658-74eb-4508-8f6b-7d88705274cb,outbound,2011-03-30
15:25:36,1301491536,sofia/external/0147662091 at 192.168.0.1,CS_EXECUTE,,0000000000,,0147662091,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound
Call,0147662091,RECV,2a5464d8-c50d-4088-bfaa-0e23ff24db08
9d507251-73f6-4f96-9026-37eeabae525c,outbound,2011-03-30
15:25:38,1301491538,sofia/external/0147050416 at 192.168.0.1,CS_EXECUTE,,0000000000,,0147050416,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound
Call,0147050416,RECV,59603591-4298-41f8-af0a-ad939e0b16f2
0a327bf4-b3a1-43f1-bcfa-e8c8770f8a2b,outbound,2011-03-30
15:25:39,1301491539,sofia/external/0143387777 at 192.168.0.1,CS_EXECUTE,,0000000000,,0143387777,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound
Call,0143387777,RECV,dc9713a8-57a9-465c-bd6e-006aadb67903
c769b907-cea2-41c4-a3a6-1b1699960f27,outbound,2011-03-30
15:25:39,1301491539,sofia/external/0143758609 at 192.168.0.1,CS_EXECUTE,,0000000000,,0143758609,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0143758609,RECV,2c9b1ab3-d1ee-499b-8771-91168acbd091
99cabd94-db28-47cb-8257-c94be0cd279f,outbound,2011-03-30
15:25:40,1301491540,sofia/external/0134162490 at 192.168.0.1,CS_EXECUTE,,0000000000,,0134162490,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound
Call,0134162490,RECV,3ed6414e-e3e7-4176-986d-47f69f6dd2c4
89b66ced-eea9-4f0c-a56c-7b8a96a0416b,outbound,2011-03-30
15:25:42,1301491542,sofia/external/0145932250 at 192.168.0.1,CS_EXECUTE,,0000000000,,0145932250,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0145932250,RECV,f420c519-b869-48fc-8190-4cf0e86c36e3
5a1a30c4-5df7-4618-b52c-ab3e4782734c,outbound,2011-03-30
15:25:44,1301491544,sofia/external/0143318179 at 192.168.0.1,CS_EXECUTE,,0000000000,,0143318179,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound
Call,0143318179,RECV,84e3bb19-a94f-4d26-ba46-5a3347644377
d07e50bd-303f-4181-85f2-27bdbb74b93d,outbound,2011-03-30
15:25:45,1301491545,sofia/external/0143708446 at 192.168.0.1,CS_EXECUTE,,0000000000,,0143708446,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0143708446,RECV,70c6ec80-78fe-4f1f-9e3a-732d544f74f2
dab44fd9-8f4b-42fd-855e-9ab4a5723687,outbound,2011-03-30
15:25:46,1301491546,sofia/external/0134988997 at 192.168.0.1,CS_EXECUTE,,0000000000,,0134988997,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0134988997,RECV,2b4c5913-4131-4387-94d1-30ab4366a49e
0ec7264f-5eaf-4f19-9b40-792ccf62b401,outbound,2011-03-30
15:25:47,1301491547,sofia/external/0141322567 at 192.168.0.1,CS_EXECUTE,,0000000000,,0141322567,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0141322567,RECV,15d192af-716e-44cb-9d00-08a9ebbdbb79
5a984ed3-6dd8-4991-912d-877469896bfa,outbound,2011-03-30
15:25:49,1301491549,sofia/external/0145234010 at 192.168.0.1,CS_EXECUTE,,0000000000,,0145234010,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound
Call,0145234010,RECV,ee680d98-9bec-4830-ae67-9d8a70aa3b08
688a2a96-bb50-4d12-8fe9-ec9423a42bd7,outbound,2011-03-30
15:25:49,1301491549,sofia/external/0134747924 at 192.168.0.1,CS_EXECUTE,,0000000000,,0134747924,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0134747924,RECV,fb9aecb6-85e8-4d3e-8c0c-b8a252425e71
3402d9bd-e33e-4e6f-9bc2-43ab559fa8aa,outbound,2011-03-30
15:25:49,1301491549,sofia/external/0147239297 at 192.168.0.1
,CS_EXECUTE,,0000000000,,0147239297,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,0147239297,0147239297,RECV,0924c5b1-fd35-40de-b47e-526b767cfce1
66101a12-5c62-4d03-9eb3-dd0a7a73a22d,outbound,2011-03-30
15:25:50,1301491550,sofia/external/0142533724 at 192.168.0.1,CS_EXECUTE,,0000000000,,0142533724,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0142533724,RECV,55115eca-c7c3-4657-98a8-fc21f13a10c9
905d5a84-c7e2-4218-881f-39655efd040a,outbound,2011-03-30
15:25:50,1301491550,sofia/external/0140261579 at 192.168.0.1,CS_EXECUTE,,0000000000,,0140261579,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0140261579,RECV,b79dc96f-146d-4a94-9025-ea82e06d30e5
0cc72f20-e347-47ad-aa7a-73cf5e486770,outbound,2011-03-30
15:25:53,1301491553,sofia/external/0141734848 at 192.168.0.1
,CS_EXECUTE,,0000000000,,0141734848,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,0141734848,0141734848,RECV,f3f48827-fe5c-4b92-a83e-8f1e88f5263d
6a7c903d-a1b8-4898-92a9-17e8e21bc477,outbound,2011-03-30
15:25:53,1301491553,sofia/external/0130926822 at 192.168.0.1,CS_EXECUTE,,0000000000,,0130926822,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0130926822,RECV,1a6d58ae-d93b-4a82-a311-e478dcf6c972
cf48337a-bd0e-4739-8aad-dd01534de325,outbound,2011-03-30
15:25:55,1301491555,sofia/external/0139194547 at 192.168.0.1,CS_EXECUTE,,0000000000,,0139194547,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0139194547,RECV,2687f485-f872-44e9-b04b-3ab45ccdbda2
9c4e2841-0d70-49bd-9876-298cdbfd07d8,outbound,2011-03-30
15:26:01,1301491561,sofia/external/0142560840 at 192.168.0.1,CS_EXECUTE,,0000000000,,0142560840,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0142560840,RECV,223d8889-5006-4043-8792-88f0059a726f
097527a5-af50-4544-95f9-41e5d2db4329,outbound,2011-03-30
15:26:01,1301491561,sofia/external/0139339445 at 192.168.0.1,CS_EXECUTE,,0000000000,,0139339445,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0139339445,RECV,ef818d92-7a22-4d00-8e17-8f6428db7255
0c734637-b2fb-4b3d-b4f3-33bc1d6565e5,outbound,2011-03-30
15:26:05,1301491565,sofia/external/0134868273 at 192.168.0.1,CS_EXECUTE,,0000000000,,0134868273,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0134868273,RECV,2364ea0b-d80c-44d5-915c-b4ee0498f50d
65a8d0fc-26e9-4f21-9117-1d833111a428,outbound,2011-03-30
15:26:07,1301491567,sofia/external/0139199893 at 192.168.0.1,CS_EXECUTE,,0000000000,,0139199893,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0139199893,RECV,a14ad031-c059-45a7-a822-4fbfddeb56fd
ea33d5ab-4dc5-45a8-8e5d-6dfbeb65c1b0,outbound,2011-03-30
15:26:08,1301491568,sofia/external/0140209135 at 192.168.0.1,CS_EXECUTE,,0000000000,,0140209135,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0140209135,RECV,0786ebcf-be37-48f0-b00c-6e1b6f22af75
473be842-169d-4ea3-8c86-c7f1d0bc01db,outbound,2011-03-30
15:26:08,1301491568,sofia/external/0142861137 at 192.168.0.1,CS_EXECUTE,,0000000000,,0142861137,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0142861137,RECV,b4ae8669-20f0-423f-951a-a14affbe76e1
f25d32b4-3b8e-42aa-a38a-b3fc7b11bf97,outbound,2011-03-30
15:26:09,1301491569,sofia/external/0134524074 at 192.168.0.1,CS_EXECUTE,,0000000000,,0134524074,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0134524074,RECV,9f82e9cb-5f79-4f76-8e1e-7f8cf8fb39e1
0694de2a-1251-4ba7-a5e3-302403bda1ec,outbound,2011-03-30
15:26:09,1301491569,sofia/external/0130326184 at 192.168.0.1,CS_EXECUTE,,0000000000,,0130326184,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0130326184,RECV,3dd6da1f-a59d-4b72-8b87-0657aee2a7eb
6db28a24-99c7-4050-9c7e-7df94ee05860,outbound,2011-03-30
15:26:10,1301491570,sofia/external/0130510861 at 192.168.0.1,CS_EXECUTE,,0000000000,,0130510861,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0130510861,RECV,f1d0b27f-a7d0-4dd0-a2a1-873525e6ed47
676d1965-4bd0-43a3-85a2-84d1c618098c,outbound,2011-03-30
15:26:12,1301491572,sofia/external/0139880615 at 192.168.0.1,CS_EXECUTE,,0000000000,,0139880615,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0139880615,RECV,69fb0f7e-b279-46a0-a9ff-0a685b2a2d9a
10efef01-29bb-4181-bf34-70390cdbe964,outbound,2011-03-30
15:26:13,1301491573,sofia/external/0139902937 at 192.168.0.1,CS_EXECUTE,,0000000000,,0139902937,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0139902937,RECV,505c212f-7459-4ace-836c-7d35737a5749
9efbd4c6-17f9-4035-b185-84c53f8e67f5,outbound,2011-03-30
15:26:14,1301491574,sofia/external/0130344121 at 192.168.0.1,CS_EXECUTE,,0000000000,,0130344121,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0130344121,RECV,2a79f551-dd7a-454d-af0b-b893a05a3bee
f111d343-9a56-4d18-890f-9482f7743264,outbound,2011-03-30
15:26:16,1301491576,sofia/external/0144311010 at 192.168.0.1,CS_EXECUTE,,0000000000,,0144311010,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0144311010,RECV,eb7e7eed-d4d5-46ad-ac91-6a90b3bab9b6
a2f60b85-204b-4dd7-933d-af04d5807b3c,outbound,2011-03-30
15:26:32,1301491592,sofia/external/0148831626 at 192.168.0.1,CS_EXECUTE,,0000000000,,0148831626,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0148831626,RECV,3bd290c2-2275-428c-9edd-e940c659e730
9a40f095-69e6-4a1b-97ea-d464c6ea2f4a,outbound,2011-03-30
15:26:39,1301491599,sofia/external/0140419154 at 192.168.0.1,CS_EXECUTE,,0000000000,,0140419154,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
Call,0140419154,RECV,d79fa0b6-6e77-4fb8-8d37-bcd1d7f68a8d

50 total.

Thank you =)

-- 
Mathieu LAUTRAM
Application developer

BJT Partners - FRANCE
+33 1 79 75 99 60
+33 6 61 59 07 25
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From infos at madovsky.org  Wed Mar 30 20:27:54 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 30 Mar 2011 12:27:54 -0400
Subject: [Freeswitch-users] invite in conference
Message-ID: <DA9A496894454615B726C8598E9737F4@e1705>

When I invite in conference, I can't see any conference esl event of the new member invited and accepted in conference
is it normal ?

Thanks
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From anthony.minessale at gmail.com  Wed Mar 30 20:48:57 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 30 Mar 2011 11:48:57 -0500
Subject: [Freeswitch-users] stuck channels
In-Reply-To: <AANLkTimuPUcyEonfie4K=W_0JCmXE6YAvguBScSBQk4Q@mail.gmail.com>
References: <AANLkTimuPUcyEonfie4K=W_0JCmXE6YAvguBScSBQk4Q@mail.gmail.com>
Message-ID: <AANLkTimb01fCuTkTt8t4jHxz+-FfDm6xTm8NJCBq3KrM@mail.gmail.com>

Have you tried to reproduce on the latest GIT HEAD?


On Wed, Mar 30, 2011 at 11:21 AM, Mathieu Lautram
<lautram.mathieu at gmail.com> wrote:
> Hi everyone,
> I have a problem of stuck channels. Indeed, when I do a show channels in
> fs_cli, there is a lot of hangup callstate channels. I even tried to do a
> uuid_kill but fs_cli tells me "-ERR No Such Channels"...
> So I d'ont know where the problem is. Someone tells me that it could be a
> sqlite problem. I hope you could help me because it's been 3 weeks that I'm
> on it and I have no solutions...
> Here is the log corresponding to my problem:
> 2011-03-30 15:26:54.866347 [NOTICE] sofia.c:537 Hangup
> sofia/external/0142560840 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:26:54.866347 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0142560840 at 192.168.0.1 [KILL]
> 2011-03-30 15:26:54.866347 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0142560840 at 192.168.0.1 [BREAK]
> 2011-03-30 15:26:56.268289 [DEBUG] switch_channel.c:2563
> (sofia/external/0134747924 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:26:56.268289 [NOTICE] sofia.c:537 Hangup
> sofia/external/0134747924 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:26:56.268289 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0134747924 at 192.168.0.1 [KILL]
> 2011-03-30 15:26:56.268289 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0134747924 at 192.168.0.1 [BREAK]
> 2011-03-30 15:27:00.066249 [DEBUG] switch_channel.c:2563
> (sofia/external/0139339445 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:27:00.066249 [NOTICE] sofia.c:537 Hangup
> sofia/external/0139339445 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:27:00.066249 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0139339445 at 192.168.0.1 [KILL]
> 2011-03-30 15:27:00.066249 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0139339445 at 192.168.0.1 [BREAK]
> 2011-03-30 15:27:00.563121 [DEBUG] switch_channel.c:2563
> (sofia/external/0143708446 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:27:00.563121 [NOTICE] sofia.c:537 Hangup
> sofia/external/0143708446 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:27:00.563121 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0143708446 at 192.168.0.1 [KILL]
> 2011-03-30 15:27:00.563121 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0143708446 at 192.168.0.1 [BREAK]
> 2011-03-30 15:27:01.376088 [DEBUG] switch_channel.c:2563
> (sofia/external/0148863352 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:27:01.376088 [NOTICE] sofia.c:537 Hangup
> sofia/external/0148863352 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:27:01.376088 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0148863352 at 192.168.0.1 [KILL]
> 2011-03-30 15:27:01.376088 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0148863352 at 192.168.0.1 [BREAK]
> 2011-03-30 15:27:01.377142 [DEBUG] switch_channel.c:2563
> (sofia/external/0140411621 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:27:01.377142 [NOTICE] sofia.c:537 Hangup
> sofia/external/0140411621 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:27:01.377142 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0140411621 at 192.168.0.1 [KILL]
> 2011-03-30 15:27:01.377142 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0140411621 at 192.168.0.1 [BREAK]
> 2011-03-30 15:27:04.264971 [DEBUG] switch_channel.c:2563
> (sofia/external/0130510861 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:27:04.264971 [NOTICE] sofia.c:537 Hangup
> sofia/external/0130510861 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:27:04.264971 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0130510861 at 192.168.0.1 [KILL]
> 2011-03-30 15:27:04.264971 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0130510861 at 192.168.0.1 [BREAK]
> 2011-03-30 15:27:05.165960 [DEBUG] switch_channel.c:2563
> (sofia/external/0134868273 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:27:05.165960 [NOTICE] sofia.c:537 Hangup
> sofia/external/0134868273 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:27:05.165960 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0134868273 at 192.168.0.1 [KILL]
> 2011-03-30 15:27:05.165960 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0134868273 at 192.168.0.1 [BREAK]
> 2011-03-30 15:27:05.665936 [DEBUG] switch_channel.c:2563
> (sofia/external/0142861137 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:27:05.665936 [NOTICE] sofia.c:537 Hangup
> sofia/external/0142861137 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:27:05.665936 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0142861137 at 192.168.0.1 [KILL]
> 2011-03-30 15:27:05.665936 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0142861137 at 192.168.0.1 [BREAK]
> 2011-03-30 15:27:05.962900 [DEBUG] switch_channel.c:2563
> (sofia/external/0139880615 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:27:05.962900 [NOTICE] sofia.c:537 Hangup
> sofia/external/0139880615 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:27:05.962900 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0139880615 at 192.168.0.1 [KILL]
> 2011-03-30 15:27:05.962900 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0139880615 at 192.168.0.1 [BREAK]
> 2011-03-30 15:27:08.964780 [DEBUG] switch_channel.c:2563
> (sofia/external/0140209135 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:27:08.964780 [NOTICE] sofia.c:537 Hangup
> sofia/external/0140209135 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:27:08.964780 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0140209135 at 192.168.0.1 [KILL]
> 2011-03-30 15:27:08.964780 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0140209135 at 192.168.0.1 [BREAK]
> 2011-03-30 15:27:09.664751 [DEBUG] switch_channel.c:2563
> (sofia/external/0134524074 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:27:09.664751 [NOTICE] sofia.c:537 Hangup
> sofia/external/0134524074 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:27:09.664751 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0134524074 at 192.168.0.1 [KILL]
> 2011-03-30 15:27:09.664751 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0134524074 at 192.168.0.1 [BREAK]
> 2011-03-30 15:27:09.767745 [DEBUG] switch_channel.c:2563
> (sofia/external/0130326184 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:27:09.767745 [NOTICE] sofia.c:537 Hangup
> sofia/external/0130326184 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:27:09.767745 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0130326184 at 192.168.0.1 [KILL]
> 2011-03-30 15:27:09.767745 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0130326184 at 192.168.0.1 [BREAK]
> 2011-03-30 15:27:09.864742 [DEBUG] switch_channel.c:2563
> (sofia/external/0139902937 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:27:09.864742 [NOTICE] sofia.c:537 Hangup
> sofia/external/0139902937 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:27:09.864742 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0139902937 at 192.168.0.1 [KILL]
> 2011-03-30 15:27:09.864742 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0139902937 at 192.168.0.1 [BREAK]
> 2011-03-30 15:27:12.064653 [DEBUG] switch_channel.c:2563
> (sofia/external/0144311010 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:27:12.064653 [NOTICE] sofia.c:537 Hangup
> sofia/external/0144311010 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:27:12.064653 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0144311010 at 192.168.0.1 [KILL]
> 2011-03-30 15:27:12.064653 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0144311010 at 192.168.0.1 [BREAK]
> 2011-03-30 15:27:12.466637 [DEBUG] switch_channel.c:2563
> (sofia/external/0139199893 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:27:12.466637 [NOTICE] sofia.c:537 Hangup
> sofia/external/0139199893 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:27:12.466637 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0139199893 at 192.168.0.1 [KILL]
> 2011-03-30 15:27:12.466637 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0139199893 at 192.168.0.1 [BREAK]
> 2011-03-30 15:27:14.062573 [DEBUG] switch_channel.c:2563
> (sofia/external/0142533724 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:27:14.062573 [NOTICE] sofia.c:537 Hangup
> sofia/external/0142533724 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:27:14.062573 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0142533724 at 192.168.0.1 [KILL]
> 2011-03-30 15:27:14.062573 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0142533724 at 192.168.0.1 [BREAK]
> 2011-03-30 15:27:14.066573 [DEBUG] switch_channel.c:2563
> (sofia/external/0141322567 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:27:14.066573 [NOTICE] sofia.c:537 Hangup
> sofia/external/0141322567 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:27:14.066573 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0141322567 at 192.168.0.1 [KILL]
> 2011-03-30 15:27:14.066573 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0141322567 at 192.168.0.1 [BREAK]
> 2011-03-30 15:27:16.564476 [DEBUG] switch_channel.c:2563
> (sofia/external/0130344121 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:27:16.564476 [NOTICE] sofia.c:537 Hangup
> sofia/external/0130344121 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:27:16.564476 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0130344121 at 192.168.0.1 [KILL]
> 2011-03-30 15:27:16.564476 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0130344121 at 192.168.0.1 [BREAK]
> 2011-03-30 15:27:26.464071 [DEBUG] switch_channel.c:2563
> (sofia/external/0139595901 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:27:26.464071 [NOTICE] sofia.c:537 Hangup
> sofia/external/0139595901 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:27:26.464071 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0139595901 at 192.168.0.1 [KILL]
> 2011-03-30 15:27:26.464071 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0139595901 at 192.168.0.1 [BREAK]
> 2011-03-30 15:27:27.066047 [DEBUG] switch_channel.c:2563
> (sofia/external/0143758609 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:27:27.066047 [NOTICE] sofia.c:537 Hangup
> sofia/external/0143758609 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:27:27.066047 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0143758609 at 192.168.0.1 [KILL]
> 2011-03-30 15:27:27.066047 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0143758609 at 192.168.0.1 [BREAK]
> 2011-03-30 15:27:29.953931 [DEBUG] switch_channel.c:2563
> (sofia/external/0148831626 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:27:29.953931 [NOTICE] sofia.c:537 Hangup
> sofia/external/0148831626 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:27:29.953931 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0148831626 at 192.168.0.1 [KILL]
> 2011-03-30 15:27:29.953931 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0148831626 at 192.168.0.1 [BREAK]
> 2011-03-30 15:27:36.965646 [DEBUG] switch_channel.c:2563
> (sofia/external/0140419154 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:27:36.965646 [NOTICE] sofia.c:537 Hangup
> sofia/external/0140419154 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:27:36.965646 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0140419154 at 192.168.0.1 [KILL]
> 2011-03-30 15:27:36.965646 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0140419154 at 192.168.0.1 [BREAK]
> 2011-03-30 15:27:53.565145 [DEBUG] switch_channel.c:2563
> (sofia/external/0142569040 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:27:53.565145 [NOTICE] sofia.c:537 Hangup
> sofia/external/0142569040 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:27:53.565145 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0142569040 at 192.168.0.1 [KILL]
> 2011-03-30 15:27:53.565145 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0142569040 at 192.168.0.1 [BREAK]
> 2011-03-30 15:27:59.260817 [DEBUG] switch_channel.c:2563
> (sofia/external/0140261579 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:27:59.260817 [NOTICE] sofia.c:537 Hangup
> sofia/external/0140261579 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:27:59.260817 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0140261579 at 192.168.0.1 [KILL]
> 2011-03-30 15:27:59.260817 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0140261579 at 192.168.0.1 [BREAK]
> 2011-03-30 15:28:07.062534 [DEBUG] switch_channel.c:2563
> (sofia/external/0134291660 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:28:07.062534 [NOTICE] sofia.c:537 Hangup
> sofia/external/0134291660 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:28:07.062534 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0134291660 at 192.168.0.1 [KILL]
> 2011-03-30 15:28:07.062534 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0134291660 at 192.168.0.1 [BREAK]
> 2011-03-30 15:28:08.761352 [DEBUG] switch_channel.c:2563
> (sofia/external/0142335551 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:28:08.761352 [NOTICE] sofia.c:537 Hangup
> sofia/external/0142335551 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:28:08.761352 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0142335551 at 192.168.0.1 [KILL]
> 2011-03-30 15:28:08.761352 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0142335551 at 192.168.0.1 [BREAK]
> 2011-03-30 15:28:26.160649 [DEBUG] switch_channel.c:2563
> (sofia/external/0145932250 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:28:26.160649 [NOTICE] sofia.c:537 Hangup
> sofia/external/0145932250 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:28:26.160649 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0145932250 at 192.168.0.1 [KILL]
> 2011-03-30 15:28:26.160649 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0145932250 at 192.168.0.1 [BREAK]
> 2011-03-30 15:28:31.958419 [DEBUG] switch_channel.c:2563
> (sofia/external/0146377247 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
> 2011-03-30 15:28:31.958419 [NOTICE] sofia.c:537 Hangup
> sofia/external/0146377247 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-03-30 15:28:31.958419 [DEBUG] switch_channel.c:2579 Send signal
> sofia/external/0146377247 at 192.168.0.1 [KILL]
> 2011-03-30 15:28:31.958419 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/0146377247 at 192.168.0.1 [BREAK]
> freeswitch at internal>
> freeswitch at internal>
> freeswitch at internal> show channels
> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid
> 7c91effe-7af7-47cc-a9c2-5a8d7a13e070,outbound,2011-03-30
> 15:24:18,1301491458,sofia/external/0146426664 at 192.168.0.1,CS_EXECUTE,,0000000000,,0146426664,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,0146426664,0146426664,RECV,d2ea58aa-e3a4-496e-a031-ac501910d18e
> 19b44e66-0cc8-4bf2-b8e4-8587986c4e22,outbound,2011-03-30
> 15:24:46,1301491486,sofia/external/0140675411 at 192.168.0.1,CS_EXECUTE,,0000000000,,0140675411,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0140675411,RECV,472fb609-8d90-4d7d-bc0f-7353bb01139f
> 4dca35aa-589b-4ba8-9efa-f7510514385b,outbound,2011-03-30
> 15:25:02,1301491502,sofia/external/0134291660 at 192.168.0.1,CS_EXECUTE,,0000000000,,0134291660,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0134291660,RECV,f62c2f5d-0b17-4e1e-8f5c-9c495407e95e
> 309fd56c-9d4b-4904-be94-8803cc757dd3,outbound,2011-03-30
> 15:25:08,1301491508,sofia/external/0146377247 at 192.168.0.1,CS_EXECUTE,,0000000000,,0146377247,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0146377247,RECV,cdcfd0eb-5efc-4541-9abd-845abfa63fa9
> 384fdd12-908c-4ad5-a488-a67f25b253ed,outbound,2011-03-30
> 15:25:20,1301491520,sofia/external/0141955961 at 192.168.0.1,CS_EXECUTE,,0000000000,,0141955961,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0141955961,RECV,f1a4da83-7ff9-478b-9323-0b9a69104ea2
> 66ba4d1c-a8de-4a5f-8da3-53eea2e353be,outbound,2011-03-30
> 15:25:22,1301491522,sofia/external/0146770127 at 192.168.0.1,CS_EXECUTE,,0000000000,,0146770127,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0146770127,RECV,50a004d4-bb3e-4baf-afe5-a9eeb5a58cfc
> 398d2132-7f7a-4ddf-95af-fe8afcb2a253,outbound,2011-03-30
> 15:25:23,1301491523,sofia/external/0144598175 at 192.168.0.1,CS_EXECUTE,,0000000000,,0144598175,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound
> Call,0144598175,RECV,98170d54-5c3d-40be-8d20-30da9a478730
> 2b90e1ba-16c4-4d48-bc4f-90c753a9244e,outbound,2011-03-30
> 15:25:23,1301491523,sofia/external/0130310549 at 192.168.0.1,CS_EXECUTE,,0000000000,,0130310549,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0130310549,RECV,5a3cba1e-f559-4116-93d8-3cca45e3084b
> 13671770-a92a-4f18-98bb-e8488d926cdb,outbound,2011-03-30
> 15:25:25,1301491525,sofia/external/0148863352 at 192.168.0.1,CS_EXECUTE,,0000000000,,0148863352,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0148863352,RECV,0c01dc58-90ed-408f-a574-1091c35b2c13
> 6da9d0de-ef0d-4dfc-85bf-d771b3c6f098,outbound,2011-03-30
> 15:25:25,1301491525,sofia/external/0146682550 at 192.168.0.1,CS_EXECUTE,,0000000000,,0146682550,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound
> Call,0146682550,RECV,e8754f05-8870-4b25-8f7a-107ee129b1e2
> cb332c48-443b-4326-86b4-ea3d89346fef,outbound,2011-03-30
> 15:25:25,1301491525,sofia/external/0142378320 at 192.168.0.1,CS_EXECUTE,,0000000000,,0142378320,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0142378320,RECV,0759c1a2-9f3a-4259-9e3a-7f567318cf67
> 4f960d60-8079-4225-bea2-58f8ccba67f5,outbound,2011-03-30
> 15:25:26,1301491526,sofia/external/0148521177 at 192.168.0.1,CS_EXECUTE,,0000000000,,0148521177,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound
> Call,0148521177,RECV,37b401ba-3077-43b8-bf8a-ee3a836ac843
> c03c2320-1ddb-40b0-884b-d4b07ac4d189,outbound,2011-03-30
> 15:25:27,1301491527,sofia/external/0142569040 at 192.168.0.1,CS_EXECUTE,,0000000000,,0142569040,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0142569040,RECV,1be9a468-b720-4398-b8c6-621c52aad6ac
> b4a6ef35-79d3-4520-9574-1c37c2fecec7,outbound,2011-03-30
> 15:25:28,1301491528,sofia/external/0140411621 at 192.168.0.1,CS_EXECUTE,,0000000000,,0140411621,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,0140411621,0140411621,RECV,81e14c30-0b77-46e6-96a4-ae8a0678e6c5
> 67b126e7-2a5f-4b99-9843-ee71e2c08c10,outbound,2011-03-30
> 15:25:35,1301491535,sofia/external/0139745628 at 192.168.0.1,CS_EXECUTE,,0000000000,,0139745628,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound
> Call,0139745628,RECV,1b173b0d-69c6-40e0-8b71-1496106cbead
> dbd23a86-64bf-417d-bb77-6b4a927b9efd,outbound,2011-03-30
> 15:25:35,1301491535,sofia/external/0139595901 at 192.168.0.1,CS_EXECUTE,,0000000000,,0139595901,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0139595901,RECV,395e8762-4f70-48f3-b8b5-b5040a2e5fbe
> dfe644c4-d453-4c39-93bd-34f7f3e9b570,outbound,2011-03-30
> 15:25:36,1301491536,sofia/external/0142335551 at 192.168.0.1,CS_EXECUTE,,0000000000,,0142335551,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0142335551,RECV,51995424-c0f4-4252-b4ad-3b4ae7fcdad6
> 96a82658-74eb-4508-8f6b-7d88705274cb,outbound,2011-03-30
> 15:25:36,1301491536,sofia/external/0147662091 at 192.168.0.1,CS_EXECUTE,,0000000000,,0147662091,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound
> Call,0147662091,RECV,2a5464d8-c50d-4088-bfaa-0e23ff24db08
> 9d507251-73f6-4f96-9026-37eeabae525c,outbound,2011-03-30
> 15:25:38,1301491538,sofia/external/0147050416 at 192.168.0.1,CS_EXECUTE,,0000000000,,0147050416,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound
> Call,0147050416,RECV,59603591-4298-41f8-af0a-ad939e0b16f2
> 0a327bf4-b3a1-43f1-bcfa-e8c8770f8a2b,outbound,2011-03-30
> 15:25:39,1301491539,sofia/external/0143387777 at 192.168.0.1,CS_EXECUTE,,0000000000,,0143387777,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound
> Call,0143387777,RECV,dc9713a8-57a9-465c-bd6e-006aadb67903
> c769b907-cea2-41c4-a3a6-1b1699960f27,outbound,2011-03-30
> 15:25:39,1301491539,sofia/external/0143758609 at 192.168.0.1,CS_EXECUTE,,0000000000,,0143758609,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0143758609,RECV,2c9b1ab3-d1ee-499b-8771-91168acbd091
> 99cabd94-db28-47cb-8257-c94be0cd279f,outbound,2011-03-30
> 15:25:40,1301491540,sofia/external/0134162490 at 192.168.0.1,CS_EXECUTE,,0000000000,,0134162490,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound
> Call,0134162490,RECV,3ed6414e-e3e7-4176-986d-47f69f6dd2c4
> 89b66ced-eea9-4f0c-a56c-7b8a96a0416b,outbound,2011-03-30
> 15:25:42,1301491542,sofia/external/0145932250 at 192.168.0.1,CS_EXECUTE,,0000000000,,0145932250,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0145932250,RECV,f420c519-b869-48fc-8190-4cf0e86c36e3
> 5a1a30c4-5df7-4618-b52c-ab3e4782734c,outbound,2011-03-30
> 15:25:44,1301491544,sofia/external/0143318179 at 192.168.0.1,CS_EXECUTE,,0000000000,,0143318179,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound
> Call,0143318179,RECV,84e3bb19-a94f-4d26-ba46-5a3347644377
> d07e50bd-303f-4181-85f2-27bdbb74b93d,outbound,2011-03-30
> 15:25:45,1301491545,sofia/external/0143708446 at 192.168.0.1,CS_EXECUTE,,0000000000,,0143708446,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0143708446,RECV,70c6ec80-78fe-4f1f-9e3a-732d544f74f2
> dab44fd9-8f4b-42fd-855e-9ab4a5723687,outbound,2011-03-30
> 15:25:46,1301491546,sofia/external/0134988997 at 192.168.0.1,CS_EXECUTE,,0000000000,,0134988997,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0134988997,RECV,2b4c5913-4131-4387-94d1-30ab4366a49e
> 0ec7264f-5eaf-4f19-9b40-792ccf62b401,outbound,2011-03-30
> 15:25:47,1301491547,sofia/external/0141322567 at 192.168.0.1,CS_EXECUTE,,0000000000,,0141322567,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0141322567,RECV,15d192af-716e-44cb-9d00-08a9ebbdbb79
> 5a984ed3-6dd8-4991-912d-877469896bfa,outbound,2011-03-30
> 15:25:49,1301491549,sofia/external/0145234010 at 192.168.0.1,CS_EXECUTE,,0000000000,,0145234010,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound
> Call,0145234010,RECV,ee680d98-9bec-4830-ae67-9d8a70aa3b08
> 688a2a96-bb50-4d12-8fe9-ec9423a42bd7,outbound,2011-03-30
> 15:25:49,1301491549,sofia/external/0134747924 at 192.168.0.1,CS_EXECUTE,,0000000000,,0134747924,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0134747924,RECV,fb9aecb6-85e8-4d3e-8c0c-b8a252425e71
> 3402d9bd-e33e-4e6f-9bc2-43ab559fa8aa,outbound,2011-03-30
> 15:25:49,1301491549,sofia/external/0147239297 at 192.168.0.1,CS_EXECUTE,,0000000000,,0147239297,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,0147239297,0147239297,RECV,0924c5b1-fd35-40de-b47e-526b767cfce1
> 66101a12-5c62-4d03-9eb3-dd0a7a73a22d,outbound,2011-03-30
> 15:25:50,1301491550,sofia/external/0142533724 at 192.168.0.1,CS_EXECUTE,,0000000000,,0142533724,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0142533724,RECV,55115eca-c7c3-4657-98a8-fc21f13a10c9
> 905d5a84-c7e2-4218-881f-39655efd040a,outbound,2011-03-30
> 15:25:50,1301491550,sofia/external/0140261579 at 192.168.0.1,CS_EXECUTE,,0000000000,,0140261579,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0140261579,RECV,b79dc96f-146d-4a94-9025-ea82e06d30e5
> 0cc72f20-e347-47ad-aa7a-73cf5e486770,outbound,2011-03-30
> 15:25:53,1301491553,sofia/external/0141734848 at 192.168.0.1,CS_EXECUTE,,0000000000,,0141734848,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,0141734848,0141734848,RECV,f3f48827-fe5c-4b92-a83e-8f1e88f5263d
> 6a7c903d-a1b8-4898-92a9-17e8e21bc477,outbound,2011-03-30
> 15:25:53,1301491553,sofia/external/0130926822 at 192.168.0.1,CS_EXECUTE,,0000000000,,0130926822,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0130926822,RECV,1a6d58ae-d93b-4a82-a311-e478dcf6c972
> cf48337a-bd0e-4739-8aad-dd01534de325,outbound,2011-03-30
> 15:25:55,1301491555,sofia/external/0139194547 at 192.168.0.1,CS_EXECUTE,,0000000000,,0139194547,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0139194547,RECV,2687f485-f872-44e9-b04b-3ab45ccdbda2
> 9c4e2841-0d70-49bd-9876-298cdbfd07d8,outbound,2011-03-30
> 15:26:01,1301491561,sofia/external/0142560840 at 192.168.0.1,CS_EXECUTE,,0000000000,,0142560840,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0142560840,RECV,223d8889-5006-4043-8792-88f0059a726f
> 097527a5-af50-4544-95f9-41e5d2db4329,outbound,2011-03-30
> 15:26:01,1301491561,sofia/external/0139339445 at 192.168.0.1,CS_EXECUTE,,0000000000,,0139339445,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0139339445,RECV,ef818d92-7a22-4d00-8e17-8f6428db7255
> 0c734637-b2fb-4b3d-b4f3-33bc1d6565e5,outbound,2011-03-30
> 15:26:05,1301491565,sofia/external/0134868273 at 192.168.0.1,CS_EXECUTE,,0000000000,,0134868273,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0134868273,RECV,2364ea0b-d80c-44d5-915c-b4ee0498f50d
> 65a8d0fc-26e9-4f21-9117-1d833111a428,outbound,2011-03-30
> 15:26:07,1301491567,sofia/external/0139199893 at 192.168.0.1,CS_EXECUTE,,0000000000,,0139199893,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0139199893,RECV,a14ad031-c059-45a7-a822-4fbfddeb56fd
> ea33d5ab-4dc5-45a8-8e5d-6dfbeb65c1b0,outbound,2011-03-30
> 15:26:08,1301491568,sofia/external/0140209135 at 192.168.0.1,CS_EXECUTE,,0000000000,,0140209135,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0140209135,RECV,0786ebcf-be37-48f0-b00c-6e1b6f22af75
> 473be842-169d-4ea3-8c86-c7f1d0bc01db,outbound,2011-03-30
> 15:26:08,1301491568,sofia/external/0142861137 at 192.168.0.1,CS_EXECUTE,,0000000000,,0142861137,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0142861137,RECV,b4ae8669-20f0-423f-951a-a14affbe76e1
> f25d32b4-3b8e-42aa-a38a-b3fc7b11bf97,outbound,2011-03-30
> 15:26:09,1301491569,sofia/external/0134524074 at 192.168.0.1,CS_EXECUTE,,0000000000,,0134524074,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0134524074,RECV,9f82e9cb-5f79-4f76-8e1e-7f8cf8fb39e1
> 0694de2a-1251-4ba7-a5e3-302403bda1ec,outbound,2011-03-30
> 15:26:09,1301491569,sofia/external/0130326184 at 192.168.0.1,CS_EXECUTE,,0000000000,,0130326184,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0130326184,RECV,3dd6da1f-a59d-4b72-8b87-0657aee2a7eb
> 6db28a24-99c7-4050-9c7e-7df94ee05860,outbound,2011-03-30
> 15:26:10,1301491570,sofia/external/0130510861 at 192.168.0.1,CS_EXECUTE,,0000000000,,0130510861,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0130510861,RECV,f1d0b27f-a7d0-4dd0-a2a1-873525e6ed47
> 676d1965-4bd0-43a3-85a2-84d1c618098c,outbound,2011-03-30
> 15:26:12,1301491572,sofia/external/0139880615 at 192.168.0.1,CS_EXECUTE,,0000000000,,0139880615,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0139880615,RECV,69fb0f7e-b279-46a0-a9ff-0a685b2a2d9a
> 10efef01-29bb-4181-bf34-70390cdbe964,outbound,2011-03-30
> 15:26:13,1301491573,sofia/external/0139902937 at 192.168.0.1,CS_EXECUTE,,0000000000,,0139902937,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0139902937,RECV,505c212f-7459-4ace-836c-7d35737a5749
> 9efbd4c6-17f9-4035-b185-84c53f8e67f5,outbound,2011-03-30
> 15:26:14,1301491574,sofia/external/0130344121 at 192.168.0.1,CS_EXECUTE,,0000000000,,0130344121,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0130344121,RECV,2a79f551-dd7a-454d-af0b-b893a05a3bee
> f111d343-9a56-4d18-890f-9482f7743264,outbound,2011-03-30
> 15:26:16,1301491576,sofia/external/0144311010 at 192.168.0.1,CS_EXECUTE,,0000000000,,0144311010,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0144311010,RECV,eb7e7eed-d4d5-46ad-ac91-6a90b3bab9b6
> a2f60b85-204b-4dd7-933d-af04d5807b3c,outbound,2011-03-30
> 15:26:32,1301491592,sofia/external/0148831626 at 192.168.0.1,CS_EXECUTE,,0000000000,,0148831626,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0148831626,RECV,3bd290c2-2275-428c-9edd-e940c659e730
> 9a40f095-69e6-4a1b-97ea-d464c6ea2f4a,outbound,2011-03-30
> 15:26:39,1301491599,sofia/external/0140419154 at 192.168.0.1,CS_EXECUTE,,0000000000,,0140419154,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound
> Call,0140419154,RECV,d79fa0b6-6e77-4fb8-8d37-bcd1d7f68a8d
> 50 total.
> Thank you =)
> --
> Mathieu LAUTRAM
> Application developer
>
> BJT Partners - FRANCE
> +33 1 79 75 99 60
> +33 6 61 59 07 25
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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  Wed Mar 30 20:56:34 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 30 Mar 2011 11:56:34 -0500
Subject: [Freeswitch-users] T.38 reinvite and 488 Inacceptable Here
 relay problem; codec issue with proxy_media=true
In-Reply-To: <AANLkTinpHst2HgPVkHQ2gASM+2LofYVhXZZwpJO8+34i@mail.gmail.com>
References: <AANLkTinpHst2HgPVkHQ2gASM+2LofYVhXZZwpJO8+34i@mail.gmail.com>
Message-ID: <AANLkTikpZSM+yZm0z8OEs+Jpa_7GBOW6Y7PZ014M2tH2@mail.gmail.com>

Can you try this with the patch I just pushed to GIT.
In the future can you use JIRA for this type of report so we have a
ticket to reference the patch to?


On Wed, Mar 30, 2011 at 10:24 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> Hi All!
>
> I'm trying to launch t.38 passthough on FS.
> I have two sip profiles - local and external, one on customer side looking
> to ATAs, and one looking to PSTN gateways.
> On both profiles I have enabled t38-passthrough, and this works when T.38 is
> enabled on ATA and PSTN gateway.
> Problems come when T.38 is turned off on ATA, so PSTN gateway detects fax on
> its side and tries to re-invite ATA behind freeswitch with T.38. ATA answers
> 488 Inacceptable Here, but this answer doesn't get relayed to PSTN gateway,
> so while ATA transmits fax signals inband, PSTN gateway still doesn't know
> that re-invite is failed and fax machines can't negotiate t.30.
> My question is how to allow relay this 488 to another side of bridge?
> I thought that maybe proxy_media=true will help, and this is where second
> issue comes. I enabled late negotiation and proxy_media, the only codec I
> use is PCMA. But when I get incoming call, initially in PCMA and it goes to
> outbound ESL app via socket, in the middle of processing I get strange
> errors:
>
> EXECUTE sofia/external/1234567 at XXXXX.199 set(continue_on_fail=true)
> 2011-03-30 18:59:30.172363 [DEBUG] mod_dptools.c:1060
> sofia/external/1234567 at XX.XX.XX.199 SET [continue_on_fail]=[true]
> EXECUTE sofia/external/1234567 at XXXXX.199 socket(localhost:8006 async full)
> 2011-03-30 18:59:30.184605 [DEBUG] switch_ivr.c:766 Codec Activated L16 at 0hz
> 0 channels 0ms
> 2011-03-30 18:59:30.188640 [DEBUG] switch_core_session.c:954 Send signal
> sofia/external/1234567 at XXXXX.199 [BREAK]
> 2011-03-30 18:59:30.205638 [DEBUG] switch_ivr.c:563
> sofia/external/1234567 at 85.114.2.199 Command Execute limit(hash
> OutboundExternalLimit 1)
> EXECUTE sofia/external/1234567 at XXXXX.199 limit(hash OutboundExternalLimit 1)
> 2011-03-30 18:59:30.205638 [INFO] switch_limit.c:126 incr called:
> OutboundExternalLimit_1 max:-1, interval:0
> 2011-03-30 18:59:30.205638 [INFO] mod_hash.c:200 Usage for
> OutboundExternalLimit_1 is now 1
> 2011-03-30 18:59:30.205638 [ERR] switch_core_io.c:724
> sofia/external/1234567 at XXXXX.199 has no write codec.
> 2011-03-30 18:59:30.205638 [DEBUG] switch_channel.c:2563
> (sofia/external/1234567 at XXXXX.199) Callstate Change RINGING -> HANGUP
> 2011-03-30 18:59:30.205638 [NOTICE] switch_core_io.c:725 Hangup
> sofia/external/1234567 at XXXXX.199 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
>
> I don't answer channel in my script, it doesn't even try to execute bridge,
> failing somewhere in the middle.
> Is this a bug or misconfiguration of my FS?
> I tried latest git FreeSWITCH Version 1.0.head (git-6e78f6f 2011-03-30
> 11-41-45 +0200) with same result.
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 latysheff at gmail.com  Wed Mar 30 15:53:21 2011
From: latysheff at gmail.com (Vladimir Latyshev)
Date: Wed, 30 Mar 2011 15:53:21 +0400
Subject: [Freeswitch-users] mod_skypopen produces snaps
In-Reply-To: <AANLkTi=XbGj9mwQirYQg9tVpwWXqd-R-Zp2tKV5M5n3W@mail.gmail.com>
References: <AANLkTikS7asjH3akOoXcLdMizDNYySKv7fScCTPM=m81@mail.gmail.com>
	<AANLkTi=XbGj9mwQirYQg9tVpwWXqd-R-Zp2tKV5M5n3W@mail.gmail.com>
Message-ID: <AANLkTim9JNFH_5stBXmYoW96yewKcrvkZMx0SeF16YHA@mail.gmail.com>

Giovanni, thanks for developing this module!
I use Windows 7, mod_skypopen compiled with VS2008, precompiled win32 FS
git, Skype 4 and 5, and I guess any (tried many different for other
reasons).
wav file parameters may post a bit later..

2011/3/30 Giovanni Maruzzelli <gmaruzz at gmail.com>

>
> On Wed, Mar 30, 2011 at 10:05 AM, Vladimir Latyshev <latysheff at gmail.com>wrote:
>
>> Hi everybody!
>> mod_skypopen produces snaps every 20 sec when incoming skype call directed
>> to playback application.
>> Clearly, this happens because of zeroing read and write buffers, as it's
>> seen in source code. How can this problem be avoided?
>>
>> Hi Vladimir,
>
> never heard of this problem before, and is like three years that we're
> zeroing buffers.
>
> What platform, sound driver, skype client version, code version, operating
> system, hardware, etc are you using?
>
> -giovanni
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>
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From jpatten at co.brazos.tx.us  Wed Mar 30 19:53:55 2011
From: jpatten at co.brazos.tx.us (Josh M. Patten)
Date: Wed, 30 Mar 2011 15:53:55 +0000
Subject: [Freeswitch-users] Valet Parking improvements
Message-ID: <8C8A3D4965236A42BDFF1758727F049A263D10@ITEX1.bc.local>

Our development team is in the process of writing some improvements into the valet parking module that we hope to get pushed upstream to the community. Among these improvements are:

*         Ability to set timeout value

o   Ability to send call back to extension that originally parked the call after timeout

o   Ability to send call to static extension (for example, an operator) after timeout

o   Ability to disconnect the call after timeout

*         Ability to set escape button, such as *, #, 0-9

o   Ability to send call back to extension that originally parked the call after caller presses escape button

o   Ability to send call to static extension (for example, an operator) after caller presses escape button

o   Ability to disconnect the call after caller presses escape button

In order to make these changes we are going to have to add a few arguments to the module. My questions are:

How should we get this code submitted for review and testing?
What do we need to do to get this code merged after it is reviewed and tested?

Thanks!


Josh Patten
Brazos County Network Engineer
979.361.4676

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From msc at freeswitch.org  Wed Mar 30 21:09:57 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 30 Mar 2011 10:09:57 -0700
Subject: [Freeswitch-users] Valet Parking improvements
In-Reply-To: <8C8A3D4965236A42BDFF1758727F049A263D10@ITEX1.bc.local>
References: <8C8A3D4965236A42BDFF1758727F049A263D10@ITEX1.bc.local>
Message-ID: <AANLkTi=8qdfCvO0VeTYE2WT=DiDQRnGVpgKBvvUVc0Rk@mail.gmail.com>

Josh,

Many thanks for your willingness to help! The best way to submit patches is
through our Jira tracking system at jira.freeswitch.org. Please submit your
proposed changes there as text attachment(s) of diff files. Choose
"improvement" as the issue type.

If you have any other questions feel free to email us here or join us in
#freeswitch on irc.freenode.net.

-Michael

On Wed, Mar 30, 2011 at 8:53 AM, Josh M. Patten <jpatten at co.brazos.tx.us>wrote:

>  Our development team is in the process of writing some improvements into
> the valet parking module that we hope to get pushed upstream to the
> community. Among these improvements are:
>
> ?         Ability to set timeout value
>
> o   Ability to send call back to extension that originally parked the call
> after timeout
>
> o   Ability to send call to static extension (for example, an operator)
> after timeout
>
> o   Ability to disconnect the call after timeout
>
> ?         Ability to set escape button, such as *, #, 0-9
>
> o   Ability to send call back to extension that originally parked the call
> after caller presses escape button
>
> o   Ability to send call to static extension (for example, an operator)
> after caller presses escape button
>
> o   Ability to disconnect the call after caller presses escape button
>
>
>
> In order to make these changes we are going to have to add a few arguments
> to the module. My questions are:
>
>
>
> How should we get this code submitted for review and testing?
>
> What do we need to do to get this code merged after it is reviewed and
> tested?
>
>
>
> Thanks!
>
>
>
>
>
> Josh Patten
>
> Brazos County Network Engineer
>
> 979.361.4676
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Mar 30 21:18:26 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 30 Mar 2011 10:18:26 -0700
Subject: [Freeswitch-users] hupall in cluster
In-Reply-To: <E8AFD7E4531E4DB5B0465D236BF945BE@e1705>
References: <E8AFD7E4531E4DB5B0465D236BF945BE@e1705>
Message-ID: <AANLkTi=7AjRx_nrAQi8a+PJjaiZ6rpoQVHCZ4+e9wbSP@mail.gmail.com>

Do you have socket access to the node in question?

On Wed, Mar 30, 2011 at 9:14 AM, Madovsky <infos at madovsky.org> wrote:

>  I'm experimenting how to use hupall from a node to another node in a
> cluster.
> is there a way to do it ?
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Mar 30 21:20:21 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 30 Mar 2011 10:20:21 -0700
Subject: [Freeswitch-users] invite in conference
In-Reply-To: <DA9A496894454615B726C8598E9737F4@e1705>
References: <DA9A496894454615B726C8598E9737F4@e1705>
Message-ID: <AANLkTikZxNH0_28LOW5FAF7ohDy4zdgyFV-PbTC9rJcy@mail.gmail.com>

what syntax are you using for the invitation? I would like to try it on my
system and see if i can reproduce.
-MC

On Wed, Mar 30, 2011 at 9:27 AM, Madovsky <infos at madovsky.org> wrote:

>  When I invite in conference, I can't see any conference esl event of the
> new member invited and accepted in conference
> is it normal ?
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Mar 30 21:23:22 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 30 Mar 2011 10:23:22 -0700
Subject: [Freeswitch-users] RTP Proxy mode, DTMF conversion problem
In-Reply-To: <4D9323E5.3030603@swip.net>
References: <AANLkTimUvhO65MTg7DaMRbbzO1M5bz4oKg5JfnR0_6uC@mail.gmail.com>
	<12CA5F58-378A-4231-9E1E-3F89DD59049B@gmail.com>
	<AANLkTikABFQ4hwj9+dkUyKGTU02+dL4gY8DYdRH8RNhz@mail.gmail.com>
	<AANLkTikO+U5=A9yLepzeA8-LxggWM2SLcHRDfzfr4ieg@mail.gmail.com>
	<AANLkTikH6EKOMzYriW=egWA-dFo0-KckRY=Qsxn-FgoL@mail.gmail.com>
	<AANLkTik3=WPdnj_WpE-vK8b8SfEp5NXxyXonL7YT+UN=@mail.gmail.com>
	<AANLkTin4LE4pUBv0cLsFnm=S86Nsx6Lj9jMEvLGk+dX5@mail.gmail.com>
	<4D9323E5.3030603@swip.net>
Message-ID: <AANLkTimmeA3LHw1=w+JP=dXuOCY2DCuWBuCLJ7=zEYoQ@mail.gmail.com>

Tony just added a patch today, so update to latest git and re-test. Add
comments to FS-2819 on Jira.

-MC

On Wed, Mar 30, 2011 at 5:36 AM, Robin Vleij <robin at swip.net> wrote:

> On 02/25/2011 12:07 AM, Anthony Minessale wrote:
>
> Hi!
>
> > you need to run the start_dtmf() application on the leg that has inband
> DTMF
> >
> > if it's the outbound leg you need to set it in execute_on_answer in
> > the bridge line
> >
> > <action application="bridge"
> > data="{execute_on_answer=start_dtmf}sofia/OPERATOR_2/46111223344"/>
>
> I think he's running into the same kind of problem I'm running into. I
> reported it as FS-2819.
>
> UA (2833) -> FS -> (INBAND) UA
>
> On the b-leg that only supports inband, FS keeps pushing telephone-event
> on the SDP. Even though the UA on the b-leg only replies with 0 or 8. I
> tried with the start_dtmf as written in this thread, but still only
> rtp-events are passed onto the B-side UA, which doesn't know what to do
> with that.
>
> /Robin
>
> --
> Robin Vleij
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From andrewkt at aktzero.com  Thu Mar 31 00:39:52 2011
From: andrewkt at aktzero.com (Andrew Thompson)
Date: Wed, 30 Mar 2011 16:39:52 -0400
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <4D5D6F66.9060606@gmail.com>
References: <4D4DC16F.50500@utoronto.ca>	<AANLkTimfzmpeR2N2sAf_QpAX0g3=BSRS0P-bPogh-Zbu@mail.gmail.com>	<4D521069.5000802@utoronto.ca>	<AANLkTi=54t5pYfSuLuEnwhycAoou8xhcLNDjVuKX8vJR@mail.gmail.com>	<4D521B17.2020408@utoronto.ca>	<ED3B5594-990A-4ACC-8206-856650E56F38@ipeva.fr>	<4D528B46.7010709@utoronto.ca>	<9CF0F421-1F14-4718-824F-33A6E1BF85A7@ipeva.fr>	<4D5295C0.3080807@utoronto.ca>	<AANLkTik_82ZycrpjBmiJSQCDUgScnoJE0WCiH5XwW3rk@mail.gmail.com>	<4D52A0B8.6090005@utoronto.ca>	<AANLkTimPj_27abwq4RNvpkfczkUXi5hf7-ro5v=yEc6E@mail.gmail.com>	<1AFB9015-306C-492A-AA09-5AD6905FD485@ipeva.fr>	<4D52B6BC.30306@utoronto.ca>	<AANLkTinBUe4w3Ui0QRJPOiarDAqO+tkWS1d4X-FmT3ij@mail.gmail.com>	<AANLkTi==uogdP7q2Nsqh7rcoaMy6tvy1at8kcNSqS=1q@mail.gmail.com>	<4D52C951.5080600@utoronto.ca>	<AANLkTi=nA76tJcdXN2Ta3cyzm5KnQhBadNJE_swg1Xk0@mail.gmail.com>	<53C394DD-308B-438C-8D10-93724A453664@freeswitch.org>	<AANLkTin4S6czPQvTOar-TBrSpRrZeQzKrOxapKk7rcgP@mail.gmail.com>
	<4D5D6F66.9060606@gmail.com>
Message-ID: <4D939518.10000@aktzero.com>

On 2/17/2011 1:56 PM, Nazim Aghabayov wrote:
> Hello All
>
> I have a same problem loading mod_com_g729.
> I'm on a Debian Lenny x64, latest git.
> FS is installed in /usr/local/freeswitch.
>
> During the load module outputs:
> "[INFO] mod_com_g729.c:243 Failed to get G.729A status".
>
> Every time I try to load the module, freeswitch_licence_server outputs
> on console:
> "Unrecognised resource G.729A/0"
>
> FreeSWITCH and freeswith_licence_server are started as root, I've tried
> to load licensing server separately, but still no luck.
>
> Anybody had this problem before?
> www.freeswitch.org

Did you find a resolution for this?

I've not dug on it much, as I didn't have time, I just disabled it and 
moved on.

-- 
Andrew Thompson
http://aktzero.com/



From egable+freeswitch at gmail.com  Thu Mar 31 01:55:00 2011
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Wed, 30 Mar 2011 17:55:00 -0400
Subject: [Freeswitch-users] XML parser bug
In-Reply-To: <AANLkTinUhFVvbx364NVyNgC0_jApVyi9E+p8jb_59efw@mail.gmail.com>
References: <AANLkTi=2zkLLKyLg9qaKDsF=y_oKqUEo_ErKwvEpgkvM@mail.gmail.com>
	<AANLkTimVxhaN4f3TU3Ni9vv0FxjRaYz=H1rGY9aZvFje@mail.gmail.com>
	<AANLkTinUhFVvbx364NVyNgC0_jApVyi9E+p8jb_59efw@mail.gmail.com>
Message-ID: <AANLkTi=qJn+GL+4m6bXW2VSdMUDsWFcGqnh7WsY6P6it@mail.gmail.com>

On a side note, why are you running with -waste flag? You really should not
be doing that unless you have very good and very specific reasons to do it
and you know what that does and why you want to do it. Perhaps you do, but I
would double check. Personally, I've run FS on several different versions of
Linux without -waste for two years without ever needing it.

On Wed, Mar 30, 2011 at 8:54 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello Joao,
>
> Ok, thanks
>
> David
>
>
> 2011/3/30 Jo?o Mesquita <jmesquita at freeswitch.org>
>
>> This is not a bug and has been discussed several times on this mailing
>> list. You can't comment X-PRE-PROCESS tags like that. Make a quick google
>> search and you'll find several discussions about that including an
>> explanation from Tony on the subject.
>>
>> Regards,
>> Jo?o Mesquita
>>
>>
>>
>> On Wed, Mar 30, 2011 at 9:21 AM, David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> Hello,
>>>
>>> I noticed the following:
>>>
>>> I have my sofia.conf.xml like this:
>>>
>>> <configuration name="sofia.conf" description="sofia Endpoint">
>>>
>>>   <global_settings>
>>>     <param name="log-level" value="0"/>
>>>
>>>     <!-- <param name="auto-restart" value="false"/> -->
>>>
>>>     <param name="debug-presence" value="0"/>
>>>   </global_settings>
>>>
>>>   <!--
>>>            The rabbit hole goes deep.  This includes all the
>>>            profiles in the sip_profiles directory that is up
>>>            one level from this directory.
>>>        <profiles>
>>>          <X-PRE-PROCESS cmd="include" data="../sip_profiles/*.xml" />
>>>        </profiles>
>>>   -->
>>>
>>> </configuration>
>>>
>>> when I start FS, latest GIT:
>>> freeswitch -version
>>> FreeSWITCH version: 1.0.head (git-7e52acf 2011-03-28 22-18-47 -0500)
>>>
>>> I get the following output:
>>>
>>> ./freeswitch -waste
>>> WARNING: Wasting up to 8 megs of memory per thread.
>>> 2011-03-30 14:02:23.200097 [INFO] switch_event.c:615 Activate Eventing
>>> Engine.
>>> 2011-03-30 14:02:23.211052 [DEBUG] switch_event.c:594 Create event
>>> dispatch thread 0
>>> Cannot Initialize [[error near line 1521]: unclosed <!--]
>>>
>>> If i then change my sofia to:
>>>
>>> <configuration name="sofia.conf" description="sofia Endpoint">
>>>
>>>   <global_settings>
>>>     <param name="log-level" value="0"/>
>>>
>>>     <!-- <param name="auto-restart" value="false"/> -->
>>>
>>>     <param name="debug-presence" value="0"/>
>>>   </global_settings>
>>>
>>>   <!--
>>>            The rabbit hole goes deep.  This includes all the
>>>            profiles in the sip_profiles directory that is up
>>>            one level from this directory.
>>>        <profiles>
>>>        </profiles>
>>>   -->
>>>
>>> </configuration>
>>>
>>>
>>> Please note the absence of:          <X-PRE-PROCESS cmd="include"
>>> data="../sip_profiles/*.xml" />
>>>
>>>
>>> FS Starts normally!
>>>
>>> Is this the correct behaviour? Isn't comments supposed NOT to be read?
>>>
>>> Thanks all.
>>>
>>>
>>>
>>> David
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing from
our children, we're stealing from them--and it's not even considered to be a
crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not
live to eat.) ~Marcus Tullius Cicero
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From andrew.keil at askinteractive.net  Thu Mar 31 02:13:59 2011
From: andrew.keil at askinteractive.net (Andrew Keil)
Date: Thu, 31 Mar 2011 09:13:59 +1100
Subject: [Freeswitch-users] Freeswitch latest source rebuild issue
	under	Windows
In-Reply-To: <AANLkTimWpJonmpawmd1oTkREM72BgzV8gV_QU3vBSFUQ@mail.gmail.com>
References: <C42A22554C0CF44DAC0AA1F132B86DE65716763FA6@ASKSERVER.asksbs.local>
	<AANLkTinLmBcybYuz1EyEFOjHV7bdd3DtcfeLVrP=ToZB@mail.gmail.com>
	<C42A22554C0CF44DAC0AA1F132B86DE65716763FA9@ASKSERVER.asksbs.local>
	<AANLkTimWpJonmpawmd1oTkREM72BgzV8gV_QU3vBSFUQ@mail.gmail.com>
Message-ID: <C42A22554C0CF44DAC0AA1F132B86DE65716763FAB@ASKSERVER.asksbs.local>

Mitch,

I have edited the wiki (http://wiki.freeswitch.org/wiki/Installation_for_Windows) to include this information.

Thanks,

Andrew Keil

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitch Capper
Sent: Thursday, 31 March 2011 1:19 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Freeswitch latest source rebuild issue under Windows

Thats the great part of the wiki anyone can edit it:)

~Mitch

On Wed, Mar 30, 2011 at 3:25 AM, Andrew Keil <andrew.keil at askinteractive.net> wrote:
> Mitch,
>
> Well spotted! ?I simply unchecked mod_managed from the build list and now nothing fails!
>
> I usually use Visual Studio Pro and completely forgot about express being split into C++, C# etc....
>
> Perhaps the Windows installation documentation (http://wiki.freeswitch.org/wiki/Installation_for_Windows) on freeswitch.org should be updated to include this one. ?Especially since it mentions the Express version and only mentions VC++ Express.
>
> Now time to do some basic testing.
>
> Thanks,
>
> Andrew Keil
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of 
> Mitch Capper
> Sent: Wednesday, 30 March 2011 8:53 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Freeswitch latest source rebuild issue 
> under Windows
>
> Hi Andrew,
> It is saying it can't find the managed dll. ?Visual Studio Express 
> comes in different versions with there being a different version for 
> c#. ? I am not sure if you can build the c# module from the c++ edition, although if you don't need .net/mod_managed support you may be able to just not compile/use that module.
>
> ~Mitch
>
> On Wed, Mar 30, 2011 at 12:41 AM, Andrew Keil <andrew.keil at askinteractive.net> wrote:
>> Jeff,
>>
>> Just tried the latest (git head). ?Note: autocrlf option is set to 
>> false (tested via: git config --get-all core.autocrlf)
>>
>> Almost there. ?The original fatal error has gone now.
>>
>> The only error left is the following:
>>
>> ------ Build started: Project: mod_managed, Configuration: Debug_CLR
>> Win32 ------
>> ?freeswitch_managed.cpp
>> freeswitch_managed.cpp : fatal error C1192: #using failed on 'C:\FreeSWITCH\Win32\Debug\mod\FreeSWITCH.Managed.dll'
>> ? ? ? ? ?'The system cannot find the file specified.'
>> ?freeswitch_wrap.2010.cxx
>> freeswitch_wrap.2010.cxx : fatal error C1192: #using failed on 'C:\FreeSWITCH\Win32\Debug\mod\FreeSWITCH.Managed.dll'
>> ? ? ? ? ?'The system cannot find the file specified.'
>> ?mod_managed.cpp
>> mod_managed.cpp : fatal error C1192: #using failed on 'C:\FreeSWITCH\Win32\Debug\mod\FreeSWITCH.Managed.dll'
>> ? ? ? ? ?'The system cannot find the file specified.'
>> ?Running Code Analysis for C/C++...
>> ?Generating Code...
>>
>> I tried to rebuild a few times, however the same error above happened. ?Any ideas?
>>
>> See attached for my build output log.
>>
>> Regards,
>>
>> Andrew Keil
>>
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of 
>> Jeff Lenk
>> Sent: Wednesday, 30 March 2011 3:39 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Freeswitch latest source rebuild 
>> issue under Windows
>>
>> Please try git head just submitted fixes
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-latest-sourc
>> e -rebuild-issue-under-Windows-tp6221661p6221815.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>> e
>> rs
>> http://www.freeswitch.org
>>
>>
>> __________ Information from ESET NOD32 Antivirus, version of virus 
>> signature database 5998 (20110329) __________
>>
>> The message was checked by ESET NOD32 Antivirus.
>>
>> http://www.eset.com
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>> e
>> rs
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus 
> signature database 5999 (20110330) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>

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

__________ Information from ESET NOD32 Antivirus, version of virus signature database 6001 (20110330) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
 


From infos at madovsky.org  Thu Mar 31 02:27:35 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 30 Mar 2011 18:27:35 -0400
Subject: [Freeswitch-users] invite in conference
References: <DA9A496894454615B726C8598E9737F4@e1705>
	<AANLkTikZxNH0_28LOW5FAF7ohDy4zdgyFV-PbTC9rJcy@mail.gmail.com>
Message-ID: <3F77E2C4CE1D4DDC84BD50F5B980B500@e1705>

/usr/local/freeswitch/bin/fs_cli -x "conference confText dial\{inconf=true,originate_timeout=20,ignore_early_media=true,instant_ringback=true}user/11111 22222 hiConf"

and

 /usr/local/freeswitch/bin/fs_cli -x "conference confText dial\{inconf=true,originate_timeout=20,ignore_early_media=true,instant_ringback=true}loopback/11111 22222 hiConf"

is this dial event can be in other place that conference::maintenance in ESL ?

thanks

  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, March 30, 2011 1:20 PM
  Subject: Re: [Freeswitch-users] invite in conference


  what syntax are you using for the invitation? I would like to try it on my system and see if i can reproduce.
  -MC


  On Wed, Mar 30, 2011 at 9:27 AM, Madovsky <infos at madovsky.org> wrote:

    When I invite in conference, I can't see any conference esl event of the new member invited and accepted in conference
    is it normal ?

    Thanks

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






------------------------------------------------------------------------------


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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Thu Mar 31 02:31:28 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 30 Mar 2011 18:31:28 -0400
Subject: [Freeswitch-users] hupall in cluster
References: <E8AFD7E4531E4DB5B0465D236BF945BE@e1705>
	<AANLkTi=7AjRx_nrAQi8a+PJjaiZ6rpoQVHCZ4+e9wbSP@mail.gmail.com>
Message-ID: <57724C4888E84B8AA7F25F011949E5C6@e1705>

unfortunatly all sockets are listening on 127.0.0.1.
yes I know what you think, listen on network IP and send message to socket.
but as I have only public IP It means that I need to create iptables rules...
if it's the only solution I will do it.

thanks
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, March 30, 2011 1:18 PM
  Subject: Re: [Freeswitch-users] hupall in cluster


  Do you have socket access to the node in question?


  On Wed, Mar 30, 2011 at 9:14 AM, Madovsky <infos at madovsky.org> wrote:

    I'm experimenting how to use hupall from a node to another node in a cluster.
    is there a way to do it ?

    Thanks

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






------------------------------------------------------------------------------


  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at microcomaustralia.com.au  Thu Mar 31 02:39:45 2011
From: brian at microcomaustralia.com.au (Brian May)
Date: Thu, 31 Mar 2011 09:39:45 +1100
Subject: [Freeswitch-users] DOS attack
Message-ID: <AANLkTinzLYjWNeYZka9yJMVdFy46JFiLeKf87ce=Fh8U@mail.gmail.com>

Hello,

This morning, I got the following message:

[241824.279299] Out of memory: kill process 20570 (freeswitch) score
17388 or a child

Since then I have plenty of memory.

Since then I have noticed that I am receiving almost 400 packets a
second along the lines of:

2011-03-31 06:57:25.541284 [WARNING] sofia_reg.c:1246 SIP auth
challenge (REGISTER) on sofia profile 'internal' for
[224586792 at 59.167.180.194] from ip 95.154.248.17
2011-03-31 06:57:25.543256 [WARNING] sofia_reg.c:1246 SIP auth
challenge (REGISTER) on sofia profile 'internal' for
[3728015026 at 59.167.180.194] from ip 95.154.248.17
2011-03-31 06:57:25.547261 [WARNING] sofia_reg.c:1246 SIP auth
challenge (REGISTER) on sofia profile 'internal' for
[224586792 at 59.167.180.194] from ip 95.154.248.17
2011-03-31 06:57:25.559259 [WARNING] sofia_reg.c:1246 SIP auth
challenge (REGISTER) on sofia profile 'internal' for
[3728015026 at 59.167.180.194] from ip 95.154.248.17
2011-03-31 06:57:25.564311 [WARNING] sofia_reg.c:1246 SIP auth
challenge (REGISTER) on sofia profile 'internal' for
[224586792 at 59.167.180.194] from ip 95.154.248.17
2011-03-31 06:57:25.574287 [WARNING] sofia_reg.c:1246 SIP auth
challenge (REGISTER) on sofia profile 'internal' for
[3728015026 at 59.167.180.194] from ip 95.154.248.17
2011-03-31 06:57:25.578259 [WARNING] sofia_reg.c:1246 SIP auth
challenge (REGISTER) on sofia profile 'internal' for
[3728015026 at 59.167.180.194] from ip 95.154.248.17
2011-03-31 06:57:25.587276 [WARNING] sofia_reg.c:1246 SIP auth
challenge (REGISTER) on sofia profile 'internal' for
[224586792 at 59.167.180.194] from ip 95.154.248.17
2011-03-31 06:57:25.593266 [WARNING] sofia_reg.c:1246 SIP auth
challenge (REGISTER) on sofia profile 'internal' for
[3728015026 at 59.167.180.194] from ip 95.154.248.17
2011-03-31 06:57:25.595256 [WARNING] sofia_reg.c:1246 SIP auth
challenge (REGISTER) on sofia profile 'internal' for
[3728015026 at 59.167.180.194] from ip 95.154.248.17

These packets continue even though I stoped freeswitch:

09:38:30.132408 IP 95.154.248.17.5124 > 59.167.180.194.5060: SIP, length: 362
09:38:30.132915 IP 95.154.248.17.5115 > 59.167.180.194.5060: SIP, length: 366
09:38:30.137077 IP 95.154.248.17.5124 > 59.167.180.194.5060: SIP, length: 362
09:38:30.138790 IP 95.154.248.17.5115 > 59.167.180.194.5060: SIP, length: 364
09:38:30.142020 IP 95.154.248.17.5124 > 59.167.180.194.5060: SIP, length: 361
09:38:30.144696 IP 95.154.248.17.5115 > 59.167.180.194.5060: SIP, length: 366
09:38:30.147442 IP 95.154.248.17.5124 > 59.167.180.194.5060: SIP, length: 362
09:38:30.150147 IP 95.154.248.17.5115 > 59.167.180.194.5060: SIP, length: 366
09:38:30.153407 IP 95.154.248.17.5124 > 59.167.180.194.5060: SIP, length: 362
09:38:30.155827 IP 95.154.248.17.5115 > 59.167.180.194.5060: SIP, length: 367
09:38:30.159236 IP 95.154.248.17.5124 > 59.167.180.194.5060: SIP, length: 363
09:38:30.161730 IP 95.154.248.17.5115 > 59.167.180.194.5060: SIP, length: 366
09:38:30.165435 IP 95.154.248.17.5124 > 59.167.180.194.5060: SIP, length: 363
09:38:30.168153 IP 95.154.248.17.5115 > 59.167.180.194.5060: SIP, length: 366

I don't recognise this IP address - 95.154.248.17.

Could this be related to the out of memory issue? If so, does this
indicate some sort of memory leak inside freeswitch? Or is this normal
expected behaviour when receiving so many connection attempts?

Thanks
-- 
Brian May <brian at microcomaustralia.com.au>


From msc at freeswitch.org  Thu Mar 31 02:46:29 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 30 Mar 2011 15:46:29 -0700
Subject: [Freeswitch-users] DOS attack
In-Reply-To: <AANLkTinzLYjWNeYZka9yJMVdFy46JFiLeKf87ce=Fh8U@mail.gmail.com>
References: <AANLkTinzLYjWNeYZka9yJMVdFy46JFiLeKf87ce=Fh8U@mail.gmail.com>
Message-ID: <AANLkTintAc0t2ffksWoRuMtT2uojcWOgZzzXUDCa5aid@mail.gmail.com>

Sounds like the friend-scanner. Check this out:
http://wiki.freeswitch.org/wiki/FS_weekly_2011_02_23#Featured_Presentation

Of course, you should look into those packets to see what, exactly they are.
Also, if you can block that IP address outright on your firewall that would
be good, too.

-MC

On Wed, Mar 30, 2011 at 3:39 PM, Brian May
<brian at microcomaustralia.com.au>wrote:

> Hello,
>
> This morning, I got the following message:
>
> [241824.279299] Out of memory: kill process 20570 (freeswitch) score
> 17388 or a child
>
> Since then I have plenty of memory.
>
> Since then I have noticed that I am receiving almost 400 packets a
> second along the lines of:
>
> 2011-03-31 06:57:25.541284 [WARNING] sofia_reg.c:1246 SIP auth
> challenge (REGISTER) on sofia profile 'internal' for
> [224586792 at 59.167.180.194] from ip 95.154.248.17
> 2011-03-31 06:57:25.543256 [WARNING] sofia_reg.c:1246 SIP auth
> challenge (REGISTER) on sofia profile 'internal' for
> [3728015026 at 59.167.180.194] from ip 95.154.248.17
> 2011-03-31 06:57:25.547261 [WARNING] sofia_reg.c:1246 SIP auth
> challenge (REGISTER) on sofia profile 'internal' for
> [224586792 at 59.167.180.194] from ip 95.154.248.17
> 2011-03-31 06:57:25.559259 [WARNING] sofia_reg.c:1246 SIP auth
> challenge (REGISTER) on sofia profile 'internal' for
> [3728015026 at 59.167.180.194] from ip 95.154.248.17
> 2011-03-31 06:57:25.564311 [WARNING] sofia_reg.c:1246 SIP auth
> challenge (REGISTER) on sofia profile 'internal' for
> [224586792 at 59.167.180.194] from ip 95.154.248.17
> 2011-03-31 06:57:25.574287 [WARNING] sofia_reg.c:1246 SIP auth
> challenge (REGISTER) on sofia profile 'internal' for
> [3728015026 at 59.167.180.194] from ip 95.154.248.17
> 2011-03-31 06:57:25.578259 [WARNING] sofia_reg.c:1246 SIP auth
> challenge (REGISTER) on sofia profile 'internal' for
> [3728015026 at 59.167.180.194] from ip 95.154.248.17
> 2011-03-31 06:57:25.587276 [WARNING] sofia_reg.c:1246 SIP auth
> challenge (REGISTER) on sofia profile 'internal' for
> [224586792 at 59.167.180.194] from ip 95.154.248.17
> 2011-03-31 06:57:25.593266 [WARNING] sofia_reg.c:1246 SIP auth
> challenge (REGISTER) on sofia profile 'internal' for
> [3728015026 at 59.167.180.194] from ip 95.154.248.17
> 2011-03-31 06:57:25.595256 [WARNING] sofia_reg.c:1246 SIP auth
> challenge (REGISTER) on sofia profile 'internal' for
> [3728015026 at 59.167.180.194] from ip 95.154.248.17
>
> These packets continue even though I stoped freeswitch:
>
> 09:38:30.132408 IP 95.154.248.17.5124 > 59.167.180.194.5060: SIP, length:
> 362
> 09:38:30.132915 IP 95.154.248.17.5115 > 59.167.180.194.5060: SIP, length:
> 366
> 09:38:30.137077 IP 95.154.248.17.5124 > 59.167.180.194.5060: SIP, length:
> 362
> 09:38:30.138790 IP 95.154.248.17.5115 > 59.167.180.194.5060: SIP, length:
> 364
> 09:38:30.142020 IP 95.154.248.17.5124 > 59.167.180.194.5060: SIP, length:
> 361
> 09:38:30.144696 IP 95.154.248.17.5115 > 59.167.180.194.5060: SIP, length:
> 366
> 09:38:30.147442 IP 95.154.248.17.5124 > 59.167.180.194.5060: SIP, length:
> 362
> 09:38:30.150147 IP 95.154.248.17.5115 > 59.167.180.194.5060: SIP, length:
> 366
> 09:38:30.153407 IP 95.154.248.17.5124 > 59.167.180.194.5060: SIP, length:
> 362
> 09:38:30.155827 IP 95.154.248.17.5115 > 59.167.180.194.5060: SIP, length:
> 367
> 09:38:30.159236 IP 95.154.248.17.5124 > 59.167.180.194.5060: SIP, length:
> 363
> 09:38:30.161730 IP 95.154.248.17.5115 > 59.167.180.194.5060: SIP, length:
> 366
> 09:38:30.165435 IP 95.154.248.17.5124 > 59.167.180.194.5060: SIP, length:
> 363
> 09:38:30.168153 IP 95.154.248.17.5115 > 59.167.180.194.5060: SIP, length:
> 366
>
> I don't recognise this IP address - 95.154.248.17.
>
> Could this be related to the out of memory issue? If so, does this
> indicate some sort of memory leak inside freeswitch? Or is this normal
> expected behaviour when receiving so many connection attempts?
>
> Thanks
> --
> Brian May <brian at microcomaustralia.com.au>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jaybinks at gmail.com  Thu Mar 31 02:51:45 2011
From: jaybinks at gmail.com (jay binks)
Date: Thu, 31 Mar 2011 08:51:45 +1000
Subject: [Freeswitch-users] DOS attack
In-Reply-To: <AANLkTintAc0t2ffksWoRuMtT2uojcWOgZzzXUDCa5aid@mail.gmail.com>
References: <AANLkTinzLYjWNeYZka9yJMVdFy46JFiLeKf87ce=Fh8U@mail.gmail.com>
	<AANLkTintAc0t2ffksWoRuMtT2uojcWOgZzzXUDCa5aid@mail.gmail.com>
Message-ID: <AANLkTikbQ6MiRRzfKOnJhmuvA=ttPsB9B36PfVBzm5se@mail.gmail.com>

then setup and run Fail2Ban
http://wiki.freeswitch.org/wiki/Fail2ban

and to help with the register flood you should look at using kristian's
SIP Dos script to put packet per sec limits on registers.

http://etel.wiki.oreilly.com/wiki/index.php/SIP_DoS/DDoS_Mitigation

Jay


On Thu, Mar 31, 2011 at 8:46 AM, Michael Collins <msc at freeswitch.org> wrote:

> Sounds like the friend-scanner. Check this out:
> http://wiki.freeswitch.org/wiki/FS_weekly_2011_02_23#Featured_Presentation
>
> Of course, you should look into those packets to see what, exactly they
> are. Also, if you can block that IP address outright on your firewall that
> would be good, too.
>
> -MC
>
>
> On Wed, Mar 30, 2011 at 3:39 PM, Brian May <brian at microcomaustralia.com.au
> > wrote:
>
>> Hello,
>>
>> This morning, I got the following message:
>>
>> [241824.279299] Out of memory: kill process 20570 (freeswitch) score
>> 17388 or a child
>>
>> Since then I have plenty of memory.
>>
>> Since then I have noticed that I am receiving almost 400 packets a
>> second along the lines of:
>>
>> 2011-03-31 06:57:25.541284 [WARNING] sofia_reg.c:1246 SIP auth
>> challenge (REGISTER) on sofia profile 'internal' for
>> [224586792 at 59.167.180.194] from ip 95.154.248.17
>> 2011-03-31 06:57:25.543256 [WARNING] sofia_reg.c:1246 SIP auth
>> challenge (REGISTER) on sofia profile 'internal' for
>> [3728015026 at 59.167.180.194] from ip 95.154.248.17
>> 2011-03-31 06:57:25.547261 [WARNING] sofia_reg.c:1246 SIP auth
>> challenge (REGISTER) on sofia profile 'internal' for
>> [224586792 at 59.167.180.194] from ip 95.154.248.17
>> 2011-03-31 06:57:25.559259 [WARNING] sofia_reg.c:1246 SIP auth
>> challenge (REGISTER) on sofia profile 'internal' for
>> [3728015026 at 59.167.180.194] from ip 95.154.248.17
>> 2011-03-31 06:57:25.564311 [WARNING] sofia_reg.c:1246 SIP auth
>> challenge (REGISTER) on sofia profile 'internal' for
>> [224586792 at 59.167.180.194] from ip 95.154.248.17
>> 2011-03-31 06:57:25.574287 [WARNING] sofia_reg.c:1246 SIP auth
>> challenge (REGISTER) on sofia profile 'internal' for
>> [3728015026 at 59.167.180.194] from ip 95.154.248.17
>> 2011-03-31 06:57:25.578259 [WARNING] sofia_reg.c:1246 SIP auth
>> challenge (REGISTER) on sofia profile 'internal' for
>> [3728015026 at 59.167.180.194] from ip 95.154.248.17
>> 2011-03-31 06:57:25.587276 [WARNING] sofia_reg.c:1246 SIP auth
>> challenge (REGISTER) on sofia profile 'internal' for
>> [224586792 at 59.167.180.194] from ip 95.154.248.17
>> 2011-03-31 06:57:25.593266 [WARNING] sofia_reg.c:1246 SIP auth
>> challenge (REGISTER) on sofia profile 'internal' for
>> [3728015026 at 59.167.180.194] from ip 95.154.248.17
>> 2011-03-31 06:57:25.595256 [WARNING] sofia_reg.c:1246 SIP auth
>> challenge (REGISTER) on sofia profile 'internal' for
>> [3728015026 at 59.167.180.194] from ip 95.154.248.17
>>
>> These packets continue even though I stoped freeswitch:
>>
>> 09:38:30.132408 IP 95.154.248.17.5124 > 59.167.180.194.5060: SIP, length:
>> 362
>> 09:38:30.132915 IP 95.154.248.17.5115 > 59.167.180.194.5060: SIP, length:
>> 366
>> 09:38:30.137077 IP 95.154.248.17.5124 > 59.167.180.194.5060: SIP, length:
>> 362
>> 09:38:30.138790 IP 95.154.248.17.5115 > 59.167.180.194.5060: SIP, length:
>> 364
>> 09:38:30.142020 IP 95.154.248.17.5124 > 59.167.180.194.5060: SIP, length:
>> 361
>> 09:38:30.144696 IP 95.154.248.17.5115 > 59.167.180.194.5060: SIP, length:
>> 366
>> 09:38:30.147442 IP 95.154.248.17.5124 > 59.167.180.194.5060: SIP, length:
>> 362
>> 09:38:30.150147 IP 95.154.248.17.5115 > 59.167.180.194.5060: SIP, length:
>> 366
>> 09:38:30.153407 IP 95.154.248.17.5124 > 59.167.180.194.5060: SIP, length:
>> 362
>> 09:38:30.155827 IP 95.154.248.17.5115 > 59.167.180.194.5060: SIP, length:
>> 367
>> 09:38:30.159236 IP 95.154.248.17.5124 > 59.167.180.194.5060: SIP, length:
>> 363
>> 09:38:30.161730 IP 95.154.248.17.5115 > 59.167.180.194.5060: SIP, length:
>> 366
>> 09:38:30.165435 IP 95.154.248.17.5124 > 59.167.180.194.5060: SIP, length:
>> 363
>> 09:38:30.168153 IP 95.154.248.17.5115 > 59.167.180.194.5060: SIP, length:
>> 366
>>
>> I don't recognise this IP address - 95.154.248.17.
>>
>> Could this be related to the out of memory issue? If so, does this
>> indicate some sort of memory leak inside freeswitch? Or is this normal
>> expected behaviour when receiving so many connection attempts?
>>
>> Thanks
>> --
>> Brian May <brian at microcomaustralia.com.au>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely

Jay
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From steveu at coppice.org  Thu Mar 31 03:45:58 2011
From: steveu at coppice.org (Steve Underwood)
Date: Thu, 31 Mar 2011 07:45:58 +0800
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <4D939518.10000@aktzero.com>
References: <4D4DC16F.50500@utoronto.ca>	<4D521069.5000802@utoronto.ca>	<AANLkTi=54t5pYfSuLuEnwhycAoou8xhcLNDjVuKX8vJR@mail.gmail.com>	<4D521B17.2020408@utoronto.ca>	<ED3B5594-990A-4ACC-8206-856650E56F38@ipeva.fr>	<4D528B46.7010709@utoronto.ca>	<9CF0F421-1F14-4718-824F-33A6E1BF85A7@ipeva.fr>	<4D5295C0.3080807@utoronto.ca>	<AANLkTik_82ZycrpjBmiJSQCDUgScnoJE0WCiH5XwW3rk@mail.gmail.com>	<4D52A0B8.6090005@utoronto.ca>	<AANLkTimPj_27abwq4RNvpkfczkUXi5hf7-ro5v=yEc6E@mail.gmail.com>	<1AFB9015-306C-492A-AA09-5AD6905FD485@ipeva.fr>	<4D52B6BC.30306@utoronto.ca>	<AANLkTinBUe4w3Ui0QRJPOiarDAqO+tkWS1d4X-FmT3ij@mail.gmail.com>	<AANLkTi==uogdP7q2Nsqh7rcoaMy6tvy1at8kcNSqS=1q@mail.gmail.com>	<4D52C951.5080600@utoronto.ca>	<AANLkTi=nA76tJcdXN2Ta3cyzm5KnQhBadNJE_swg1Xk0@mail.gmail.com>	<53C394DD-308B-438C-8D10-93724A453664@freeswitch.org>	<AANLkTin4S6czPQvTOar-TBrSpRrZeQzKrOxapKk7rcgP@mail.gmail.com>	<4D5D6F66.9060606@gmail.com>
	<4D939518.10000@aktzero.com>
Message-ID: <4D93C0B6.40606@coppice.org>

On 03/31/2011 04:39 AM, Andrew Thompson wrote:
> On 2/17/2011 1:56 PM, Nazim Aghabayov wrote:
>> Hello All
>>
>> I have a same problem loading mod_com_g729.
>> I'm on a Debian Lenny x64, latest git.
>> FS is installed in /usr/local/freeswitch.
>>
>> During the load module outputs:
>> "[INFO] mod_com_g729.c:243 Failed to get G.729A status".
>>
>> Every time I try to load the module, freeswitch_licence_server outputs
>> on console:
>> "Unrecognised resource G.729A/0"
>>
>> FreeSWITCH and freeswith_licence_server are started as root, I've tried
>> to load licensing server separately, but still no luck.
>>
>> Anybody had this problem before?
>> www.freeswitch.org
> Did you find a resolution for this?
>
> I've not dug on it much, as I didn't have time, I just disabled it and
> moved on.
>
"Unrecognised resource G.729A/0" is a message from the licence server, 
indicating that a client has requested a resource (G.729A in this case) 
for which there are no valid licences available.

Steve



From kris at livecall.com  Thu Mar 31 03:57:18 2011
From: kris at livecall.com (Kris)
Date: Wed, 30 Mar 2011 16:57:18 -0700
Subject: [Freeswitch-users] hupall in cluster
References: <E8AFD7E4531E4DB5B0465D236BF945BE@e1705><AANLkTi=7AjRx_nrAQi8a+PJjaiZ6rpoQVHCZ4+e9wbSP@mail.gmail.com>
	<57724C4888E84B8AA7F25F011949E5C6@e1705>
Message-ID: <49E9749B090440D88501762D2723E3AD@stor1>

What about multicast event, capture it on the other boxes and hang up. I 
heard about it ,I haven't actually done it. Curious to see how it works...

"multicast::event": //events from other FS boxes

----- Original Message ----- 
From: "Madovsky" <infos at madovsky.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, March 30, 2011 3:31 PM
Subject: Re: [Freeswitch-users] hupall in cluster


unfortunatly all sockets are listening on 127.0.0.1.
yes I know what you think, listen on network IP and send message to socket.
but as I have only public IP It means that I need to create iptables 
rules...
if it's the only solution I will do it.

thanks
  ----- Original Message ----- 
  From: Michael Collins
  To: FreeSWITCH Users Help
  Sent: Wednesday, March 30, 2011 1:18 PM
  Subject: Re: [Freeswitch-users] hupall in cluster


  Do you have socket access to the node in question?


  On Wed, Mar 30, 2011 at 9:14 AM, Madovsky <infos at madovsky.org> wrote:

    I'm experimenting how to use hupall from a node to another node in a 
cluster.
    is there a way to do it ?

    Thanks

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






------------------------------------------------------------------------------


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



From steveu at coppice.org  Thu Mar 31 04:01:01 2011
From: steveu at coppice.org (Steve Underwood)
Date: Thu, 31 Mar 2011 08:01:01 +0800
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <4D93C0B6.40606@coppice.org>
References: <4D4DC16F.50500@utoronto.ca>	<4D521069.5000802@utoronto.ca>	<AANLkTi=54t5pYfSuLuEnwhycAoou8xhcLNDjVuKX8vJR@mail.gmail.com>	<4D521B17.2020408@utoronto.ca>	<ED3B5594-990A-4ACC-8206-856650E56F38@ipeva.fr>	<4D528B46.7010709@utoronto.ca>	<9CF0F421-1F14-4718-824F-33A6E1BF85A7@ipeva.fr>	<4D5295C0.3080807@utoronto.ca>	<AANLkTik_82ZycrpjBmiJSQCDUgScnoJE0WCiH5XwW3rk@mail.gmail.com>	<4D52A0B8.6090005@utoronto.ca>	<AANLkTimPj_27abwq4RNvpkfczkUXi5hf7-ro5v=yEc6E@mail.gmail.com>	<1AFB9015-306C-492A-AA09-5AD6905FD485@ipeva.fr>	<4D52B6BC.30306@utoronto.ca>	<AANLkTinBUe4w3Ui0QRJPOiarDAqO+tkWS1d4X-FmT3ij@mail.gmail.com>	<AANLkTi==uogdP7q2Nsqh7rcoaMy6tvy1at8kcNSqS=1q@mail.gmail.com>	<4D52C951.5080600@utoronto.ca>	<AANLkTi=nA76tJcdXN2Ta3cyzm5KnQhBadNJE_swg1Xk0@mail.gmail.com>	<53C394DD-308B-438C-8D10-93724A453664@freeswitch.org>	<AANLkTin4S6czPQvTOar-TBrSpRrZeQzKrOxapKk7rcgP@mail.gmail.com>	<4D5D6F66.9060606@gmail.com>	<4D939518.10000@aktzero.com>
	<4D93C0B6.40606@coppice.org>
Message-ID: <4D93C43D.1040507@coppice.org>

On 03/31/2011 07:45 AM, Steve Underwood wrote:
> On 03/31/2011 04:39 AM, Andrew Thompson wrote:
>> On 2/17/2011 1:56 PM, Nazim Aghabayov wrote:
>>> Hello All
>>>
>>> I have a same problem loading mod_com_g729.
>>> I'm on a Debian Lenny x64, latest git.
>>> FS is installed in /usr/local/freeswitch.
>>>
>>> During the load module outputs:
>>> "[INFO] mod_com_g729.c:243 Failed to get G.729A status".
>>>
>>> Every time I try to load the module, freeswitch_licence_server outputs
>>> on console:
>>> "Unrecognised resource G.729A/0"
>>>
>>> FreeSWITCH and freeswith_licence_server are started as root, I've tried
>>> to load licensing server separately, but still no luck.
>>>
>>> Anybody had this problem before?
>>> www.freeswitch.org
>> Did you find a resolution for this?
>>
>> I've not dug on it much, as I didn't have time, I just disabled it and
>> moved on.
>>
> "Unrecognised resource G.729A/0" is a message from the licence server,
> indicating that a client has requested a resource (G.729A in this case)
> for which there are no valid licences available.
Rereading that, I realise the word "available" is a little vague in that 
context. Let me rephrase that as...

"Unrecognised resource G.729A/0" is a message from the licence server, 
indicating that a client has requested a resource (G.729A in this case) 
for which there are no valid licences on the machine.

In other words, your freeswitch box has no valid G.729A licences 
installed. If you think you have valid licences, are you sure they are 
installed in the correct directory?
Regards,
Steve



From infos at madovsky.org  Thu Mar 31 04:07:07 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 30 Mar 2011 20:07:07 -0400
Subject: [Freeswitch-users] hupall in cluster
References: <E8AFD7E4531E4DB5B0465D236BF945BE@e1705><AANLkTi=7AjRx_nrAQi8a+PJjaiZ6rpoQVHCZ4+e9wbSP@mail.gmail.com><57724C4888E84B8AA7F25F011949E5C6@e1705>
	<49E9749B090440D88501762D2723E3AD@stor1>
Message-ID: <7B8239EB848C4AE3AE2EBF04AA6EF06C@e1705>

good suggestion.
do you mean that multicast::event is available with mod_event_socket ?

if I read wiki for mod-event_multicast
--------------
You get access to all of the same events that you would get with 
mod_event_socket and "event plain ALL".
This can be both a blessing and a curse. Please take care to audit the 
events, as they'll be sent everywhere that the
router of your subnet and the adjoining routers are configured to send 
multicast packets. You may wish to explicitly
plan for this and use a VLAN or similar precaution to protect yourself from 
information leaking to places it need not go.
---------------

the problem is if my esl script listens all events from all nodes so it can 
unecessary grow the data incoming


----- Original Message ----- 
From: "Kris" <kris at livecall.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, March 30, 2011 7:57 PM
Subject: Re: [Freeswitch-users] hupall in cluster


> What about multicast event, capture it on the other boxes and hang up. I
> heard about it ,I haven't actually done it. Curious to see how it works...
>
> "multicast::event": //events from other FS boxes
>
> ----- Original Message ----- 
> From: "Madovsky" <infos at madovsky.org>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, March 30, 2011 3:31 PM
> Subject: Re: [Freeswitch-users] hupall in cluster
>
>
> unfortunatly all sockets are listening on 127.0.0.1.
> yes I know what you think, listen on network IP and send message to 
> socket.
> but as I have only public IP It means that I need to create iptables
> rules...
> if it's the only solution I will do it.
>
> thanks
>  ----- Original Message ----- 
>  From: Michael Collins
>  To: FreeSWITCH Users Help
>  Sent: Wednesday, March 30, 2011 1:18 PM
>  Subject: Re: [Freeswitch-users] hupall in cluster
>
>
>  Do you have socket access to the node in question?
>
>
>  On Wed, Mar 30, 2011 at 9:14 AM, Madovsky <infos at madovsky.org> wrote:
>
>    I'm experimenting how to use hupall from a node to another node in a
> cluster.
>    is there a way to do it ?
>
>    Thanks
>
>    _______________________________________________
>    FreeSWITCH-users mailing list
>    FreeSWITCH-users at lists.freeswitch.org
>    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>    http://www.freeswitch.org
>
>
>
>
>
>
> ------------------------------------------------------------------------------
>
>
>  _______________________________________________
>  FreeSWITCH-users mailing list
>  FreeSWITCH-users at lists.freeswitch.org
>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From msc at freeswitch.org  Thu Mar 31 04:14:27 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 30 Mar 2011 17:14:27 -0700
Subject: [Freeswitch-users] hupall in cluster
In-Reply-To: <7B8239EB848C4AE3AE2EBF04AA6EF06C@e1705>
References: <E8AFD7E4531E4DB5B0465D236BF945BE@e1705>
	<AANLkTi=7AjRx_nrAQi8a+PJjaiZ6rpoQVHCZ4+e9wbSP@mail.gmail.com>
	<57724C4888E84B8AA7F25F011949E5C6@e1705>
	<49E9749B090440D88501762D2723E3AD@stor1>
	<7B8239EB848C4AE3AE2EBF04AA6EF06C@e1705>
Message-ID: <AANLkTinG-pxP-V=m7gQCP2FDwK5RFAF5ozbfp34q7ZgG@mail.gmail.com>

On Wed, Mar 30, 2011 at 5:07 PM, Madovsky <infos at madovsky.org> wrote:

> good suggestion.
> do you mean that multicast::event is available with mod_event_socket ?
>
> if I read wiki for mod-event_multicast
> --------------
> You get access to all of the same events that you would get with
> mod_event_socket and "event plain ALL".
> This can be both a blessing and a curse. Please take care to audit the
> events, as they'll be sent everywhere that the
> router of your subnet and the adjoining routers are configured to send
> multicast packets. You may wish to explicitly
> plan for this and use a VLAN or similar precaution to protect yourself from
> information leaking to places it need not go.
> ---------------
>
> the problem is if my esl script listens all events from all nodes so it can
> unecessary grow the data incoming
>

Yep. You can't have it both ways - either you have to connect to each node
to send hupall via the socket or you need to listen to multicast traffic and
pick out the events you wish to act upon. I suppose the third choice would
be to create a "multicast event proxy" that handles the filtering of events
to/from each box.

Any other suggestions out there?

-MC
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From jan.berger at video24.no  Thu Mar 31 04:18:11 2011
From: jan.berger at video24.no (Jan Berger)
Date: Thu, 31 Mar 2011 02:18:11 +0200
Subject: [Freeswitch-users] PRI Test Equipment
In-Reply-To: <AANLkTinexeb64gsV_ih8fwub1F=K8V+fKEJtEPUeWuwQ@mail.gmail.com>
References: <AANLkTi=txKH5Yj3MHaMrX1uHhnqMpuS=PuLgGsxpR=H8@mail.gmail.com>
	<AANLkTi=emvxczkxopd4+OYLdY7gJjyd7qgONO8sK-PKT@mail.gmail.com>
	<AANLkTinexeb64gsV_ih8fwub1F=K8V+fKEJtEPUeWuwQ@mail.gmail.com>
Message-ID: <2E80FBC43F4F464AB3730923CA7CC754@dell9400>

http://www.utelsystems.com/instruments/isdn-pra/index.php

Utel Systems have a software + hardware package, and I believe their pricing
is "reasonable" - you are in USD 5000.- ++ range, but that is worth it if
you ever need a 3rd party tester/analyzer for conformance testing.

I have always wanted to try them, but I question their conformance & support
on standards used outside Norway. Usage of terms like PRA rather than PRI
etc + I know their main customer is the Norwegian PTT.

Other than that their package looks very awesome.

Jan 

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Max
Clark
Sent: 29. mars 2011 19:47
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] PRI Test Equipment

I want to display/read back the digits passed on the PRI.

On Tue, Mar 29, 2011 at 10:02 AM, Michael Collins <msc at freeswitch.org>
wrote:
> depends on what all you want the appliance to do. if you just want to
> do basic testing then you can buy or rent a "T-BERD" (google it) or
> something similar.
>
> -MC
>
> On Tue, Mar 29, 2011 at 9:54 AM, Max Clark <max.clark at gmail.com> wrote:
>> I'm looking for an appliance that can be plugged into a PRI connected
>> to a PBX and either display or read back the digits that are passed to
>> it. Before I build something to simulate this, is there anything out
>> there commercially that can be purchased?
>>
>> Thanks,
>> Max
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

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



From infos at madovsky.org  Thu Mar 31 04:28:23 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 30 Mar 2011 20:28:23 -0400
Subject: [Freeswitch-users] hupall in cluster
References: <E8AFD7E4531E4DB5B0465D236BF945BE@e1705><AANLkTi=7AjRx_nrAQi8a+PJjaiZ6rpoQVHCZ4+e9wbSP@mail.gmail.com><57724C4888E84B8AA7F25F011949E5C6@e1705><49E9749B090440D88501762D2723E3AD@stor1><7B8239EB848C4AE3AE2EBF04AA6EF06C@e1705>
	<AANLkTinG-pxP-V=m7gQCP2FDwK5RFAF5ozbfp34q7ZgG@mail.gmail.com>
Message-ID: <8E6F834D6F4442E592C7AB95040827BE@e1705>

do you think that an optional arguments to some application or api command
like "hupall <local|nodeXX|ALL> normal_clearing var ${var}" would be easy to program ?
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, March 30, 2011 8:14 PM
  Subject: Re: [Freeswitch-users] hupall in cluster





  On Wed, Mar 30, 2011 at 5:07 PM, Madovsky <infos at madovsky.org> wrote:

    good suggestion.
    do you mean that multicast::event is available with mod_event_socket ?

    if I read wiki for mod-event_multicast
    --------------
    You get access to all of the same events that you would get with
    mod_event_socket and "event plain ALL".
    This can be both a blessing and a curse. Please take care to audit the
    events, as they'll be sent everywhere that the
    router of your subnet and the adjoining routers are configured to send
    multicast packets. You may wish to explicitly
    plan for this and use a VLAN or similar precaution to protect yourself from
    information leaking to places it need not go.
    ---------------

    the problem is if my esl script listens all events from all nodes so it can
    unecessary grow the data incoming



  Yep. You can't have it both ways - either you have to connect to each node to send hupall via the socket or you need to listen to multicast traffic and pick out the events you wish to act upon. I suppose the third choice would be to create a "multicast event proxy" that handles the filtering of events to/from each box. 


  Any other suggestions out there?


  -MC




------------------------------------------------------------------------------


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Thu Mar 31 05:13:35 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 30 Mar 2011 21:13:35 -0400
Subject: [Freeswitch-users] set_user effect
Message-ID: <AEEF6390FE2F4AE49BC424D266E8A8B6@e1705>

If I change user with set_user, it seems that all the sip_from_xx are changed
according to the new set_user value.But how can I get the original sip_from_xxx vars ?

Thanks
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From infos at madovsky.org  Thu Mar 31 05:31:27 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 30 Mar 2011 21:31:27 -0400
Subject: [Freeswitch-users] set_user effect
Message-ID: <FC6F422E54AD48529086F465BC08C183@e1705>

sounds weird.
sip_from_user took the new set_user user value but not sip_from_user_stripped.
so now I use sip_from_user_stripped to get the original caller_id_number....
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 30, 2011 9:13 PM
  Subject: set_user effect


  If I change user with set_user, it seems that all the sip_from_xx are changed
  according to the new set_user value.But how can I get the original sip_from_xxx vars ?

  Thanks
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From brian at freeswitch.org  Thu Mar 31 05:39:33 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 30 Mar 2011 20:39:33 -0500
Subject: [Freeswitch-users] set_user effect
In-Reply-To: <FC6F422E54AD48529086F465BC08C183@e1705>
References: <FC6F422E54AD48529086F465BC08C183@e1705>
Message-ID: <F82993E7-29CA-412E-B476-2C0625EF812A@freeswitch.org>

set_user only sets the variables from the user in the directory on the current session.

If you look at the usage you can "<user>@<domain> [prefix]"

So you can set_user bob at example.com user_  

It will prefix the vars from the user with user_  and your assessment of what is going on is wrong.

/b

On Mar 30, 2011, at 8:31 PM, Madovsky wrote:

> sounds weird.
> sip_from_user took the new set_user user value but not sip_from_user_stripped.
> so now I use sip_from_user_stripped to get the original caller_id_number....
>  ----- Original Message ----- 
>  From: Madovsky 
>  To: freeswitch-users at lists.freeswitch.org 
>  Sent: Wednesday, March 30, 2011 9:13 PM
>  Subject: set_user effect

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From brian at freeswitch.org  Thu Mar 31 05:40:40 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 30 Mar 2011 20:40:40 -0500
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <4D93C0B6.40606@coppice.org>
References: <4D4DC16F.50500@utoronto.ca>	<4D521069.5000802@utoronto.ca>	<AANLkTi=54t5pYfSuLuEnwhycAoou8xhcLNDjVuKX8vJR@mail.gmail.com>	<4D521B17.2020408@utoronto.ca>	<ED3B5594-990A-4ACC-8206-856650E56F38@ipeva.fr>	<4D528B46.7010709@utoronto.ca>	<9CF0F421-1F14-4718-824F-33A6E1BF85A7@ipeva.fr>	<4D5295C0.3080807@utoronto.ca>	<AANLkTik_82ZycrpjBmiJSQCDUgScnoJE0WCiH5XwW3rk@mail.gmail.com>	<4D52A0B8.6090005@utoronto.ca>	<AANLkTimPj_27abwq4RNvpkfczkUXi5hf7-ro5v=yEc6E@mail.gmail.com>	<1AFB9015-306C-492A-AA09-5AD6905FD485@ipeva.fr>	<4D52B6BC.30306@utoronto.ca>	<AANLkTinBUe4w3Ui0QRJPOiarDAqO+tkWS1d4X-FmT3ij@mail.gmail.com>	<AANLkTi==uogdP7q2Nsqh7rcoaMy6tvy1at8kcNSqS=1q@mail.gmail.com>	<4D52C951.5080600@utoronto.ca>	<AANLkTi=nA76tJcdXN2Ta3cyzm5KnQhBadNJE_swg1Xk0@mail.gmail.com>	<53C394DD-308B-438C-8D10-93724A453664@freeswitch.org>	<AANLkTin4S6czPQvTOar-TBrSpRrZeQzKrOxapKk7rcgP@mail.gmail.com>	<4D5D6F66.9060606@gmail.com>
	<4D939518.10000@aktzero.com> <4D93C0B6.40606@coppice.org>
Message-ID: <F9992467-127C-45EC-890A-E93873840B16@freeswitch.org>

Everyone can reactivate their license with 194 installer to get over this issue.  I have reset everyones activation accounts.  Mr. Collins will be sending an email to cover this process.

/b

On Mar 30, 2011, at 6:45 PM, Steve Underwood wrote:

> "Unrecognised resource G.729A/0" is a message from the licence server, 
> indicating that a client has requested a resource (G.729A in this case) 
> for which there are no valid licences available.
> 
> Steve



From infos at madovsky.org  Thu Mar 31 05:50:00 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 30 Mar 2011 21:50:00 -0400
Subject: [Freeswitch-users] set_user effect
References: <FC6F422E54AD48529086F465BC08C183@e1705>
	<F82993E7-29CA-412E-B476-2C0625EF812A@freeswitch.org>
Message-ID: <8BC0E9EC8ECA4F25B7619AE4323450C1@e1705>

Ok Brian,

but I did this in my dialplan:

<action inline="true" application="set_user" data="${dialed_extension}@$${domain} dest_"/>

and after this line sip_from_uri, sip_from_user take the set_user values unless sip_from_user_stripped.
I use the latest git (6 hours ago),

thanks
  ----- Original Message ----- 
  From: Brian West 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, March 30, 2011 9:39 PM
  Subject: Re: [Freeswitch-users] set_user effect


  set_user only sets the variables from the user in the directory on the current session.


  If you look at the usage you can "<user>@<domain> [prefix]"


  So you can set_user bob at example.com user_  


  It will prefix the vars from the user with user_  and your assessment of what is going on is wrong.



  /b


  On Mar 30, 2011, at 8:31 PM, Madovsky wrote:


    sounds weird.
    sip_from_user took the new set_user user value but not sip_from_user_stripped.
    so now I use sip_from_user_stripped to get the original caller_id_number....
     ----- Original Message ----- 
     From: Madovsky 
     To: freeswitch-users at lists.freeswitch.org 
     Sent: Wednesday, March 30, 2011 9:13 PM
     Subject: set_user effect





------------------------------------------------------------------------------


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From brian at freeswitch.org  Thu Mar 31 06:00:49 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 30 Mar 2011 21:00:49 -0500
Subject: [Freeswitch-users] set_user effect
In-Reply-To: <8BC0E9EC8ECA4F25B7619AE4323450C1@e1705>
References: <FC6F422E54AD48529086F465BC08C183@e1705>
	<F82993E7-29CA-412E-B476-2C0625EF812A@freeswitch.org>
	<8BC0E9EC8ECA4F25B7619AE4323450C1@e1705>
Message-ID: <F8AB9415-F8C1-421E-AE23-F0DDDFCC0429@freeswitch.org>

I'm pretty sure thats not possible if you go check switch_ivr_set_user in switch_ivr.c you'll see what it does... and setting or any knowledge of any sip_ variables is not there.

/b

On Mar 30, 2011, at 8:50 PM, Madovsky wrote:

> and after this line sip_from_uri, sip_from_user take the set_user values unless sip_from_user_stripped.
> I use the latest git (6 hours ago),



From brian at freeswitch.org  Thu Mar 31 06:01:37 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 30 Mar 2011 21:01:37 -0500
Subject: [Freeswitch-users] XML parser bug
In-Reply-To: <AANLkTinUhFVvbx364NVyNgC0_jApVyi9E+p8jb_59efw@mail.gmail.com>
References: <AANLkTi=2zkLLKyLg9qaKDsF=y_oKqUEo_ErKwvEpgkvM@mail.gmail.com>
	<AANLkTimVxhaN4f3TU3Ni9vv0FxjRaYz=H1rGY9aZvFje@mail.gmail.com>
	<AANLkTinUhFVvbx364NVyNgC0_jApVyi9E+p8jb_59efw@mail.gmail.com>
Message-ID: <C8D814D9-D38F-4DBA-8E7C-FFC352FCFD4F@freeswitch.org>

Just change it to XX-PRE-PROCESS

/b

On Mar 30, 2011, at 7:54 AM, David Villasmil wrote:

> Hello Joao,
> 
> Ok, thanks
> 
> David
> 
> 2011/3/30 Jo?o Mesquita <jmesquita at freeswitch.org>
> 
>> This is not a bug and has been discussed several times on this mailing
>> list. You can't comment X-PRE-PROCESS tags like that. Make a quick google
>> search and you'll find several discussions about that including an
>> explanation from Tony on the subject.
>> 
>> Regards,
>> Jo?o Mesquita
>> 
>> 
>> 
>> On Wed, Mar 30, 2011 at 9:21 AM, David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>> 
>>> Hello,
>>> 
>>> I noticed the following:
>>> 
>>> I have my sofia.conf.xml like this:
>>> 
>>> <configuration name="sofia.conf" description="sofia Endpoint">
>>> 
>>>  <global_settings>
>>>    <param name="log-level" value="0"/>
>>> 
>>>    <!-- <param name="auto-restart" value="false"/> -->
>>> 
>>>    <param name="debug-presence" value="0"/>
>>>  </global_settings>
>>> 
>>>  <!--
>>>           The rabbit hole goes deep.  This includes all the
>>>           profiles in the sip_profiles directory that is up
>>>           one level from this directory.
>>>       <profiles>
>>>         <X-PRE-PROCESS cmd="include" data="../sip_profiles/*.xml" />
>>>       </profiles>
>>>  -->
>>> 
>>> </configuration>
>>> 
>>> when I start FS, latest GIT:
>>> freeswitch -version
>>> FreeSWITCH version: 1.0.head (git-7e52acf 2011-03-28 22-18-47 -0500)
>>> 
>>> I get the following output:
>>> 
>>> ./freeswitch -waste
>>> WARNING: Wasting up to 8 megs of memory per thread.
>>> 2011-03-30 14:02:23.200097 [INFO] switch_event.c:615 Activate Eventing
>>> Engine.
>>> 2011-03-30 14:02:23.211052 [DEBUG] switch_event.c:594 Create event
>>> dispatch thread 0
>>> Cannot Initialize [[error near line 1521]: unclosed <!--]
>>> 
>>> If i then change my sofia to:
>>> 
>>> <configuration name="sofia.conf" description="sofia Endpoint">
>>> 
>>>  <global_settings>
>>>    <param name="log-level" value="0"/>
>>> 
>>>    <!-- <param name="auto-restart" value="false"/> -->
>>> 
>>>    <param name="debug-presence" value="0"/>
>>>  </global_settings>
>>> 
>>>  <!--
>>>           The rabbit hole goes deep.  This includes all the
>>>           profiles in the sip_profiles directory that is up
>>>           one level from this directory.
>>>       <profiles>
>>>       </profiles>
>>>  -->
>>> 
>>> </configuration>
>>> 
>>> 
>>> Please note the absence of:          <X-PRE-PROCESS cmd="include"
>>> data="../sip_profiles/*.xml" />
>>> 
>>> 
>>> FS Starts normally!
>>> 
>>> Is this the correct behaviour? Isn't comments supposed NOT to be read?
>>> 
>>> Thanks all.
>>> 
>>> 
>>> 
>>> David
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From infos at madovsky.org  Thu Mar 31 06:09:09 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 30 Mar 2011 22:09:09 -0400
Subject: [Freeswitch-users] set_user effect
References: <FC6F422E54AD48529086F465BC08C183@e1705><F82993E7-29CA-412E-B476-2C0625EF812A@freeswitch.org><8BC0E9EC8ECA4F25B7619AE4323450C1@e1705>
	<F8AB9415-F8C1-421E-AE23-F0DDDFCC0429@freeswitch.org>
Message-ID: <CD61AE0B5AFA486DB77D3D6C4D735CB4@e1705>

ok thanks, maybe the cause is elsewhere

----- Original Message ----- 
From: "Brian West" <brian at freeswitch.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, March 30, 2011 10:00 PM
Subject: Re: [Freeswitch-users] set_user effect


> I'm pretty sure thats not possible if you go check switch_ivr_set_user in 
> switch_ivr.c you'll see what it does... and setting or any knowledge of 
> any sip_ variables is not there.
>
> /b
>
> On Mar 30, 2011, at 8:50 PM, Madovsky wrote:
>
>> and after this line sip_from_uri, sip_from_user take the set_user values 
>> unless sip_from_user_stripped.
>> I use the latest git (6 hours ago),
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From ayhkor at gmail.com  Thu Mar 31 07:06:25 2011
From: ayhkor at gmail.com (deniro)
Date: Wed, 30 Mar 2011 23:06:25 -0400
Subject: [Freeswitch-users] phone callers muted when joining to conference
Message-ID: <BANLkTikQDd+X_AV-2LddEhXV_EG6rTC6xg@mail.gmail.com>

Hi,
   When join to the conference  through perl program
   $session->execute("conference",conf-name at conf-profile);
somehow phones were getting muted
is there any parameters to pass while joining conference that will prevent
phone callers be muted automatically

thx
deniro--
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From fieldpeak at gmail.com  Thu Mar 31 07:26:33 2011
From: fieldpeak at gmail.com (=?utf-8?B?Q2hhcmxlcw==?=)
Date: Thu, 31 Mar 2011 11:26:33 +0800
Subject: [Freeswitch-users]
	=?utf-8?q?How_to_check_the_FS_source_code_vers?=
	=?utf-8?q?ion_onWindows_platform?=
References: <4d92fa1c.4a2c2b0a.0a54.1bbf@mx.google.com>
Message-ID: <4d93f46f.0e37640a.7a10.3818@mx.google.com>

Hi Steven,

Very good! Appreicated very much for your elabrated explaination!! professional!

howerver, when i run 'version' cmd in the console, it shows below (without e.g. 'b041edc8 2011-03-01 06-35-06 -0600'), it looks we need manually write the exact version inside the code then re-build, right?
if i manally write it inside the code, for my case, it should be git-d5ef86d7 2011-03-30 08-55-28, however, it abenst  '-0600''... where i can get it... it looks a millisecond value 

freeswitch at WIN-9E8E1NBP4RJ> version
FreeSWITCH Version 1.0.head (git-)


2011-03-31 



Charles 



???? Steven Ayre 
????? 2011-03-30  22:28:59 
???? FreeSWITCH Users Help 
??? 
??? Re: [Freeswitch-users] How to check the FS source code version onWindows platform 
 
d5ef86d7788ef0080ca3be7e2ff39bda989d4b4d is indeed the version.

Git uses hashes as version number, not a sequential number as with SVN. It can't do sequential since it's distributed so it avoids having the same version number at different locations at the same time.

The version when you're at the CLI will be something like:
FreeSWITCH Version 1.0.head (git-b041edc8 2011-03-01 06-35-06 -0600)

Note git-b041edc8 is the version number, it's just the first 8 digits of the full hash (so it's easier to remember). Yours would be git-d5ef86d7.

The changelog (git log) will show you how recent your version is.

-Steve






On 30 March 2011 10:38, Charles <fieldpeak at gmail.com> wrote:


Can anyone help advise how to check the exact version of Freeswitch on windows platform? 
I can see the info like screenshot below in the properties of FS folder which git from latest source, it the correct way? 
if yes, the version should call 'Git-Head: d5ef86d7788ef0080ca3be7e2ff39bda989d4b4d ', however, it looks very strange...
thanks.



 
2011-03-30 



Charles 

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From jeff at jefflenk.com  Thu Mar 31 07:49:46 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 30 Mar 2011 20:49:46 -0700 (PDT)
Subject: [Freeswitch-users] How to check the FS source code version
	onWindows platform
In-Reply-To: <4d93f46f.0e37640a.7a10.3818@mx.google.com>
References: <4d92fa1c.4a2c2b0a.0a54.1bbf@mx.google.com>
	<4d93f46f.0e37640a.7a10.3818@mx.google.com>
Message-ID: <1301543386697-6225824.post@n2.nabble.com>

You must have git in your path or else the version string will be blank

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-check-the-FS-source-code-version-on-Windows-platform-tp6222745p6225824.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jeff at jefflenk.com  Thu Mar 31 07:51:27 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 30 Mar 2011 20:51:27 -0700 (PDT)
Subject: [Freeswitch-users] How to check the FS source code version
	onWindows platform
In-Reply-To: <1301543386697-6225824.post@n2.nabble.com>
References: <4d92fa1c.4a2c2b0a.0a54.1bbf@mx.google.com>
	<4d93f46f.0e37640a.7a10.3818@mx.google.com>
	<1301543386697-6225824.post@n2.nabble.com>
Message-ID: <1301543487032-6225826.post@n2.nabble.com>

The windows build automation will automatically place the version string in
the code when git is found in the path.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-check-the-FS-source-code-version-on-Windows-platform-tp6222745p6225826.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From fieldpeak at gmail.com  Thu Mar 31 09:41:38 2011
From: fieldpeak at gmail.com (Charles)
Date: Thu, 31 Mar 2011 13:41:38 +0800
Subject: [Freeswitch-users] How to check the FS source code
	versiononWindows platform
References: <4d92fa1c.4a2c2b0a.0a54.1bbf@mx.google.com>,
	<4d93f46f.0e37640a.7a10.3818@mx.google.com>,
	<1301543386697-6225824.post@n2.nabble.com>
Message-ID: <4d941415.08a9640a.53d7.33ff@mx.google.com>

Hi Jeff,

it works after added the git to the path, thanks  a lot!


2011-03-31 



Charles 



???? Jeff Lenk 
????? 2011-03-31  11:51:50 
???? freeswitch-users 
??? 
??? Re: [Freeswitch-users] How to check the FS source code versiononWindows platform 
 
The windows build automation will automatically place the version string in
the code when git is found in the path.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-check-the-FS-source-code-version-on-Windows-platform-tp6222745p6225826.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From me at nevian.org  Thu Mar 31 12:08:11 2011
From: me at nevian.org (Serge S. Yuriev)
Date: Thu, 31 Mar 2011 12:08:11 +0400
Subject: [Freeswitch-users] Full username in caller_profile->username
Message-ID: <367151301558892@web113.yandex.ru>

Hello,

caller_profile->channel_name shows sofia/internal/user at domain
but caller_profile->username shows only user w/o domain part.
How i can set username to include domain name in caller_profile->username?

-- 
wbr,
Serge


From mayamatakeshi at gmail.com  Thu Mar 31 12:32:13 2011
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 31 Mar 2011 17:32:13 +0900
Subject: [Freeswitch-users] Variable interpolation of bridge b leg
Message-ID: <AANLkTi=ofqj3yY80U7WGVvRhD2ZJiYTB_3r=Gr9zQvoa@mail.gmail.com>

I am setting channel variable execute_on_answer in my call to application
bridge. Like this:

<action application='bridge' data='{execute_on_answer=record_session
$${base_dir}/storage/recorded_sessions/${uuid}_${strftime(%Y%m%d-%H%M%S)}.wav}sofia/external/
DestinationNumber at 192.168.2.2' />

The above works, and the application record_session is executed on the leg
b. However, the uuid it gets is from leg a, and the timestamp is from the
time bridge was executed, which as I understand, is happening because the
variable interpolation is performed at the moment the application bridge is
executed.
So, is there a way to delay variable interpolation to the instant the b leg
app is executed?

br,
takeshi
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From david.villasmil.work at gmail.com  Thu Mar 31 14:36:55 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 31 Mar 2011 12:36:55 +0200
Subject: [Freeswitch-users] XML parser bug
In-Reply-To: <AANLkTi=qJn+GL+4m6bXW2VSdMUDsWFcGqnh7WsY6P6it@mail.gmail.com>
References: <AANLkTi=2zkLLKyLg9qaKDsF=y_oKqUEo_ErKwvEpgkvM@mail.gmail.com>
	<AANLkTimVxhaN4f3TU3Ni9vv0FxjRaYz=H1rGY9aZvFje@mail.gmail.com>
	<AANLkTinUhFVvbx364NVyNgC0_jApVyi9E+p8jb_59efw@mail.gmail.com>
	<AANLkTi=qJn+GL+4m6bXW2VSdMUDsWFcGqnh7WsY6P6it@mail.gmail.com>
Message-ID: <AANLkTi=-yaOswXcV7iSt=r1QtAXPD7Eu8Za652XaZAez@mail.gmail.com>

Hello,

that's just testing :P i just don't like the warnings when testing
I don't run it like that for production.

David

On Wed, Mar 30, 2011 at 11:55 PM, Eliot Gable
<egable+freeswitch at gmail.com>wrote:

> On a side note, why are you running with -waste flag? You really should not
> be doing that unless you have very good and very specific reasons to do it
> and you know what that does and why you want to do it. Perhaps you do, but I
> would double check. Personally, I've run FS on several different versions of
> Linux without -waste for two years without ever needing it.
>
>
> On Wed, Mar 30, 2011 at 8:54 AM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Hello Joao,
>>
>> Ok, thanks
>>
>> David
>>
>>
>> 2011/3/30 Jo?o Mesquita <jmesquita at freeswitch.org>
>>
>>> This is not a bug and has been discussed several times on this mailing
>>> list. You can't comment X-PRE-PROCESS tags like that. Make a quick google
>>> search and you'll find several discussions about that including an
>>> explanation from Tony on the subject.
>>>
>>> Regards,
>>> Jo?o Mesquita
>>>
>>>
>>>
>>> On Wed, Mar 30, 2011 at 9:21 AM, David Villasmil <
>>> david.villasmil.work at gmail.com> wrote:
>>>
>>>> Hello,
>>>>
>>>> I noticed the following:
>>>>
>>>> I have my sofia.conf.xml like this:
>>>>
>>>> <configuration name="sofia.conf" description="sofia Endpoint">
>>>>
>>>>   <global_settings>
>>>>     <param name="log-level" value="0"/>
>>>>
>>>>     <!-- <param name="auto-restart" value="false"/> -->
>>>>
>>>>     <param name="debug-presence" value="0"/>
>>>>   </global_settings>
>>>>
>>>>   <!--
>>>>            The rabbit hole goes deep.  This includes all the
>>>>            profiles in the sip_profiles directory that is up
>>>>            one level from this directory.
>>>>        <profiles>
>>>>          <X-PRE-PROCESS cmd="include" data="../sip_profiles/*.xml" />
>>>>        </profiles>
>>>>   -->
>>>>
>>>> </configuration>
>>>>
>>>> when I start FS, latest GIT:
>>>> freeswitch -version
>>>> FreeSWITCH version: 1.0.head (git-7e52acf 2011-03-28 22-18-47 -0500)
>>>>
>>>> I get the following output:
>>>>
>>>> ./freeswitch -waste
>>>> WARNING: Wasting up to 8 megs of memory per thread.
>>>> 2011-03-30 14:02:23.200097 [INFO] switch_event.c:615 Activate Eventing
>>>> Engine.
>>>> 2011-03-30 14:02:23.211052 [DEBUG] switch_event.c:594 Create event
>>>> dispatch thread 0
>>>> Cannot Initialize [[error near line 1521]: unclosed <!--]
>>>>
>>>> If i then change my sofia to:
>>>>
>>>> <configuration name="sofia.conf" description="sofia Endpoint">
>>>>
>>>>   <global_settings>
>>>>     <param name="log-level" value="0"/>
>>>>
>>>>     <!-- <param name="auto-restart" value="false"/> -->
>>>>
>>>>     <param name="debug-presence" value="0"/>
>>>>   </global_settings>
>>>>
>>>>   <!--
>>>>            The rabbit hole goes deep.  This includes all the
>>>>            profiles in the sip_profiles directory that is up
>>>>            one level from this directory.
>>>>        <profiles>
>>>>        </profiles>
>>>>   -->
>>>>
>>>> </configuration>
>>>>
>>>>
>>>> Please note the absence of:          <X-PRE-PROCESS cmd="include"
>>>> data="../sip_profiles/*.xml" />
>>>>
>>>>
>>>> FS Starts normally!
>>>>
>>>> Is this the correct behaviour? Isn't comments supposed NOT to be read?
>>>>
>>>> Thanks all.
>>>>
>>>>
>>>>
>>>> David
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Eliot Gable
>
> "We do not inherit the Earth from our ancestors: we borrow it from our
> children." ~David Brower
>
> "I decided the words were too conservative for me. We're not borrowing from
> our children, we're stealing from them--and it's not even considered to be a
> crime." ~David Brower
>
> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live;
> not live to eat.) ~Marcus Tullius Cicero
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From robin at swip.net  Thu Mar 31 15:53:02 2011
From: robin at swip.net (Robin Vleij)
Date: Thu, 31 Mar 2011 13:53:02 +0200
Subject: [Freeswitch-users] RTP Proxy mode, DTMF conversion problem
In-Reply-To: <AANLkTimmeA3LHw1=w+JP=dXuOCY2DCuWBuCLJ7=zEYoQ@mail.gmail.com>
References: <AANLkTimUvhO65MTg7DaMRbbzO1M5bz4oKg5JfnR0_6uC@mail.gmail.com>	<12CA5F58-378A-4231-9E1E-3F89DD59049B@gmail.com>	<AANLkTikABFQ4hwj9+dkUyKGTU02+dL4gY8DYdRH8RNhz@mail.gmail.com>	<AANLkTikO+U5=A9yLepzeA8-LxggWM2SLcHRDfzfr4ieg@mail.gmail.com>	<AANLkTikH6EKOMzYriW=egWA-dFo0-KckRY=Qsxn-FgoL@mail.gmail.com>	<AANLkTik3=WPdnj_WpE-vK8b8SfEp5NXxyXonL7YT+UN=@mail.gmail.com>	<AANLkTin4LE4pUBv0cLsFnm=S86Nsx6Lj9jMEvLGk+dX5@mail.gmail.com>	<4D9323E5.3030603@swip.net>
	<AANLkTimmeA3LHw1=w+JP=dXuOCY2DCuWBuCLJ7=zEYoQ@mail.gmail.com>
Message-ID: <4D946B1E.8060704@swip.net>

On 03/30/2011 07:23 PM, Michael Collins wrote:

Hi!

I entered the following comment to the Jira (well, if I get over the 
access denied I get in my client :-). THough it posting it here as well, 
to maybe create a discussion about if this is the right thing to do or not.

My comment to the bug:
"The behaviour changed now.
As expected from the code change if the B-side doesn't support 2833, FS 
doesn't send any RTP-events any longer. But that's not what we want, I 
think. I would expect that FS would switch to inband instead on that 
channel, like fallback to the default for DTMF. I tried by putting in 
start_dtmf_generate (both on the dialplan and "on_answer_execute" on the 
B-leg). Putting in the dialplan gives as a result that we always 
generate inband, instead of just when the B-side doesn't support 
anything else. RTP-Events dissapear then, independant if the B-side 
support 2833.

Basically, I think the patch should not read DTMF-Type=None, but 
DTMF-Type=inband.

In the larger picture, we should have some kind of DTMF-Transcoding. One 
leg can be any of SIP-INFO, 2833 or inband, while the B-Leg can have any 
of these as well, and FS should "transcode"."

/Robin

> Tony just added a patch today, so update to latest git and re-test. Add
> comments to FS-2819 on Jira.
>
> -MC
>
> On Wed, Mar 30, 2011 at 5:36 AM, Robin Vleij <robin at swip.net
> <mailto:robin at swip.net>> wrote:
>
>     On 02/25/2011 12:07 AM, Anthony Minessale wrote:
>
>     Hi!
>
>      > you need to run the start_dtmf() application on the leg that has
>     inband DTMF
>      >
>      > if it's the outbound leg you need to set it in execute_on_answer in
>      > the bridge line
>      >
>      > <action application="bridge"
>      > data="{execute_on_answer=start_dtmf}sofia/OPERATOR_2/46111223344"/>
>
>     I think he's running into the same kind of problem I'm running into. I
>     reported it as FS-2819.
>
>     UA (2833) -> FS -> (INBAND) UA
>
>     On the b-leg that only supports inband, FS keeps pushing telephone-event
>     on the SDP. Even though the UA on the b-leg only replies with 0 or 8. I
>     tried with the start_dtmf as written in this thread, but still only
>     rtp-events are passed onto the B-side UA, which doesn't know what to do
>     with that.
>
>     /Robin
>
>     --
>     Robin Vleij
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Robin Vleij


From lautram.mathieu at gmail.com  Thu Mar 31 16:05:29 2011
From: lautram.mathieu at gmail.com (Mathieu Lautram)
Date: Thu, 31 Mar 2011 14:05:29 +0200
Subject: [Freeswitch-users] stuck channels
Message-ID: <AANLkTimykLH-7i1PivZK8SQHrAFRzuXDMr0QjPPX2Ky3@mail.gmail.com>

 Yes, I have the last one... Someone tells me it was due to an old release,
so updated it. But I still have the problem =(


Have you tried to reproduce on the latest GIT HEAD?


On Wed, Mar 30, 2011 at 11:21 AM, Mathieu Lautram
<lautram.mathieu at gmail.com
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>
wrote:
>* Hi everyone,*>* I have a problem of stuck channels. Indeed, when I do a show channels in*>* fs_cli, there is a lot of hangup callstate channels. I even tried to do a*>* uuid_kill but fs_cli tells me "-ERR No Such Channels"...*>* So I d'ont know where the problem is. Someone tells me that it could be a*>* sqlite problem. I hope you could help me because it's been 3 weeks that I'm*>* on it and I have no solutions...*>* Here is the log corresponding to my problem:*>* 2011-03-30 15:26:54.866347 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0142560840 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:26:54.866347 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0142560840 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:26:54.866347 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0142560840 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:26:56.268289 [DEBUG] switch_channel.c:2563*>* (sofia/external/0134747924 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:26:56.268289 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0134747924 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:26:56.268289 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0134747924 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:26:56.268289 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0134747924 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:27:00.066249 [DEBUG] switch_channel.c:2563*>* (sofia/external/0139339445 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:27:00.066249 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0139339445 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:27:00.066249 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0139339445 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:27:00.066249 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0139339445 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:27:00.563121 [DEBUG] switch_channel.c:2563*>* (sofia/external/0143708446 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:27:00.563121 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0143708446 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:27:00.563121 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0143708446 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:27:00.563121 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0143708446 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:27:01.376088 [DEBUG] switch_channel.c:2563*>* (sofia/external/0148863352 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:27:01.376088 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0148863352 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:27:01.376088 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0148863352 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:27:01.376088 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0148863352 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:27:01.377142 [DEBUG] switch_channel.c:2563*>* (sofia/external/0140411621 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:27:01.377142 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0140411621 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:27:01.377142 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0140411621 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:27:01.377142 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0140411621 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:27:04.264971 [DEBUG] switch_channel.c:2563*>* (sofia/external/0130510861 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:27:04.264971 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0130510861 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:27:04.264971 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0130510861 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:27:04.264971 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0130510861 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:27:05.165960 [DEBUG] switch_channel.c:2563*>* (sofia/external/0134868273 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:27:05.165960 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0134868273 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:27:05.165960 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0134868273 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:27:05.165960 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0134868273 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:27:05.665936 [DEBUG] switch_channel.c:2563*>* (sofia/external/0142861137 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:27:05.665936 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0142861137 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:27:05.665936 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0142861137 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:27:05.665936 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0142861137 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:27:05.962900 [DEBUG] switch_channel.c:2563*>* (sofia/external/0139880615 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:27:05.962900 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0139880615 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:27:05.962900 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0139880615 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:27:05.962900 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0139880615 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:27:08.964780 [DEBUG] switch_channel.c:2563*>* (sofia/external/0140209135 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:27:08.964780 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0140209135 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:27:08.964780 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0140209135 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:27:08.964780 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0140209135 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:27:09.664751 [DEBUG] switch_channel.c:2563*>* (sofia/external/0134524074 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:27:09.664751 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0134524074 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:27:09.664751 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0134524074 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:27:09.664751 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0134524074 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:27:09.767745 [DEBUG] switch_channel.c:2563*>* (sofia/external/0130326184 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:27:09.767745 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0130326184 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:27:09.767745 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0130326184 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:27:09.767745 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0130326184 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:27:09.864742 [DEBUG] switch_channel.c:2563*>* (sofia/external/0139902937 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:27:09.864742 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0139902937 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:27:09.864742 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0139902937 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:27:09.864742 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0139902937 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:27:12.064653 [DEBUG] switch_channel.c:2563*>* (sofia/external/0144311010 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:27:12.064653 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0144311010 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:27:12.064653 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0144311010 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:27:12.064653 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0144311010 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:27:12.466637 [DEBUG] switch_channel.c:2563*>* (sofia/external/0139199893 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:27:12.466637 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0139199893 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:27:12.466637 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0139199893 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:27:12.466637 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0139199893 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:27:14.062573 [DEBUG] switch_channel.c:2563*>* (sofia/external/0142533724 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:27:14.062573 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0142533724 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:27:14.062573 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0142533724 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:27:14.062573 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0142533724 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:27:14.066573 [DEBUG] switch_channel.c:2563*>* (sofia/external/0141322567 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:27:14.066573 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0141322567 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:27:14.066573 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0141322567 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:27:14.066573 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0141322567 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:27:16.564476 [DEBUG] switch_channel.c:2563*>* (sofia/external/0130344121 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:27:16.564476 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0130344121 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:27:16.564476 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0130344121 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:27:16.564476 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0130344121 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:27:26.464071 [DEBUG] switch_channel.c:2563*>* (sofia/external/0139595901 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:27:26.464071 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0139595901 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:27:26.464071 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0139595901 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:27:26.464071 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0139595901 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:27:27.066047 [DEBUG] switch_channel.c:2563*>* (sofia/external/0143758609 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:27:27.066047 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0143758609 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:27:27.066047 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0143758609 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:27:27.066047 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0143758609 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:27:29.953931 [DEBUG] switch_channel.c:2563*>* (sofia/external/0148831626 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:27:29.953931 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0148831626 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:27:29.953931 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0148831626 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:27:29.953931 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0148831626 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:27:36.965646 [DEBUG] switch_channel.c:2563*>* (sofia/external/0140419154 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:27:36.965646 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0140419154 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:27:36.965646 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0140419154 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:27:36.965646 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0140419154 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:27:53.565145 [DEBUG] switch_channel.c:2563*>* (sofia/external/0142569040 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:27:53.565145 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0142569040 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:27:53.565145 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0142569040 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:27:53.565145 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0142569040 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:27:59.260817 [DEBUG] switch_channel.c:2563*>* (sofia/external/0140261579 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:27:59.260817 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0140261579 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:27:59.260817 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0140261579 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:27:59.260817 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0140261579 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:28:07.062534 [DEBUG] switch_channel.c:2563*>* (sofia/external/0134291660 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:28:07.062534 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0134291660 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:28:07.062534 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0134291660 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:28:07.062534 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0134291660 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:28:08.761352 [DEBUG] switch_channel.c:2563*>* (sofia/external/0142335551 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:28:08.761352 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0142335551 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:28:08.761352 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0142335551 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:28:08.761352 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0142335551 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:28:26.160649 [DEBUG] switch_channel.c:2563*>* (sofia/external/0145932250 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:28:26.160649 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0145932250 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:28:26.160649 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0145932250 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:28:26.160649 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0145932250 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* 2011-03-30 15:28:31.958419 [DEBUG] switch_channel.c:2563*>* (sofia/external/0146377247 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>) Callstate Change ACTIVE -> HANGUP*>* 2011-03-30 15:28:31.958419 [NOTICE] sofia.c:537 Hangup*>* sofia/external/0146377247 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [CS_EXECUTE] [NORMAL_CLEARING]*>* 2011-03-30 15:28:31.958419 [DEBUG] switch_channel.c:2579 Send signal*>* sofia/external/0146377247 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [KILL]*>* 2011-03-30 15:28:31.958419 [DEBUG] switch_core_session.c:1116 Send signal*>* sofia/external/0146377247 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [BREAK]*>* freeswitch at internal <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>*>* freeswitch at internal <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>*>* freeswitch at internal <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> show channels*>* uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid*>* 7c91effe-7af7-47cc-a9c2-5a8d7a13e070,outbound,2011-03-30*>* 15:24:18,1301491458,sofia/external/0146426664 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0146426664,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,0146426664,0146426664,RECV,d2ea58aa-e3a4-496e-a031-ac501910d18e*>* 19b44e66-0cc8-4bf2-b8e4-8587986c4e22,outbound,2011-03-30*>* 15:24:46,1301491486,sofia/external/0140675411 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0140675411,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0140675411,RECV,472fb609-8d90-4d7d-bc0f-7353bb01139f*>* 4dca35aa-589b-4ba8-9efa-f7510514385b,outbound,2011-03-30*>* 15:25:02,1301491502,sofia/external/0134291660 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0134291660,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0134291660,RECV,f62c2f5d-0b17-4e1e-8f5c-9c495407e95e*>* 309fd56c-9d4b-4904-be94-8803cc757dd3,outbound,2011-03-30*>* 15:25:08,1301491508,sofia/external/0146377247 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0146377247,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0146377247,RECV,cdcfd0eb-5efc-4541-9abd-845abfa63fa9*>* 384fdd12-908c-4ad5-a488-a67f25b253ed,outbound,2011-03-30*>* 15:25:20,1301491520,sofia/external/0141955961 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0141955961,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0141955961,RECV,f1a4da83-7ff9-478b-9323-0b9a69104ea2*>* 66ba4d1c-a8de-4a5f-8da3-53eea2e353be,outbound,2011-03-30*>* 15:25:22,1301491522,sofia/external/0146770127 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0146770127,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0146770127,RECV,50a004d4-bb3e-4baf-afe5-a9eeb5a58cfc*>* 398d2132-7f7a-4ddf-95af-fe8afcb2a253,outbound,2011-03-30*>* 15:25:23,1301491523,sofia/external/0144598175 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0144598175,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound*>* Call,0144598175,RECV,98170d54-5c3d-40be-8d20-30da9a478730*>* 2b90e1ba-16c4-4d48-bc4f-90c753a9244e,outbound,2011-03-30*>* 15:25:23,1301491523,sofia/external/0130310549 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0130310549,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0130310549,RECV,5a3cba1e-f559-4116-93d8-3cca45e3084b*>* 13671770-a92a-4f18-98bb-e8488d926cdb,outbound,2011-03-30*>* 15:25:25,1301491525,sofia/external/0148863352 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0148863352,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0148863352,RECV,0c01dc58-90ed-408f-a574-1091c35b2c13*>* 6da9d0de-ef0d-4dfc-85bf-d771b3c6f098,outbound,2011-03-30*>* 15:25:25,1301491525,sofia/external/0146682550 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0146682550,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound*>* Call,0146682550,RECV,e8754f05-8870-4b25-8f7a-107ee129b1e2*>* cb332c48-443b-4326-86b4-ea3d89346fef,outbound,2011-03-30*>* 15:25:25,1301491525,sofia/external/0142378320 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0142378320,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0142378320,RECV,0759c1a2-9f3a-4259-9e3a-7f567318cf67*>* 4f960d60-8079-4225-bea2-58f8ccba67f5,outbound,2011-03-30*>* 15:25:26,1301491526,sofia/external/0148521177 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0148521177,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound*>* Call,0148521177,RECV,37b401ba-3077-43b8-bf8a-ee3a836ac843*>* c03c2320-1ddb-40b0-884b-d4b07ac4d189,outbound,2011-03-30*>* 15:25:27,1301491527,sofia/external/0142569040 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0142569040,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0142569040,RECV,1be9a468-b720-4398-b8c6-621c52aad6ac*>* b4a6ef35-79d3-4520-9574-1c37c2fecec7,outbound,2011-03-30*>* 15:25:28,1301491528,sofia/external/0140411621 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0140411621,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,0140411621,0140411621,RECV,81e14c30-0b77-46e6-96a4-ae8a0678e6c5*>* 67b126e7-2a5f-4b99-9843-ee71e2c08c10,outbound,2011-03-30*>* 15:25:35,1301491535,sofia/external/0139745628 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0139745628,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound*>* Call,0139745628,RECV,1b173b0d-69c6-40e0-8b71-1496106cbead*>* dbd23a86-64bf-417d-bb77-6b4a927b9efd,outbound,2011-03-30*>* 15:25:35,1301491535,sofia/external/0139595901 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0139595901,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0139595901,RECV,395e8762-4f70-48f3-b8b5-b5040a2e5fbe*>* dfe644c4-d453-4c39-93bd-34f7f3e9b570,outbound,2011-03-30*>* 15:25:36,1301491536,sofia/external/0142335551 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0142335551,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0142335551,RECV,51995424-c0f4-4252-b4ad-3b4ae7fcdad6*>* 96a82658-74eb-4508-8f6b-7d88705274cb,outbound,2011-03-30*>* 15:25:36,1301491536,sofia/external/0147662091 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0147662091,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound*>* Call,0147662091,RECV,2a5464d8-c50d-4088-bfaa-0e23ff24db08*>* 9d507251-73f6-4f96-9026-37eeabae525c,outbound,2011-03-30*>* 15:25:38,1301491538,sofia/external/0147050416 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0147050416,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound*>* Call,0147050416,RECV,59603591-4298-41f8-af0a-ad939e0b16f2*>* 0a327bf4-b3a1-43f1-bcfa-e8c8770f8a2b,outbound,2011-03-30*>* 15:25:39,1301491539,sofia/external/0143387777 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0143387777,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound*>* Call,0143387777,RECV,dc9713a8-57a9-465c-bd6e-006aadb67903*>* c769b907-cea2-41c4-a3a6-1b1699960f27,outbound,2011-03-30*>* 15:25:39,1301491539,sofia/external/0143758609 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0143758609,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0143758609,RECV,2c9b1ab3-d1ee-499b-8771-91168acbd091*>* 99cabd94-db28-47cb-8257-c94be0cd279f,outbound,2011-03-30*>* 15:25:40,1301491540,sofia/external/0134162490 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0134162490,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound*>* Call,0134162490,RECV,3ed6414e-e3e7-4176-986d-47f69f6dd2c4*>* 89b66ced-eea9-4f0c-a56c-7b8a96a0416b,outbound,2011-03-30*>* 15:25:42,1301491542,sofia/external/0145932250 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0145932250,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0145932250,RECV,f420c519-b869-48fc-8190-4cf0e86c36e3*>* 5a1a30c4-5df7-4618-b52c-ab3e4782734c,outbound,2011-03-30*>* 15:25:44,1301491544,sofia/external/0143318179 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0143318179,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound*>* Call,0143318179,RECV,84e3bb19-a94f-4d26-ba46-5a3347644377*>* d07e50bd-303f-4181-85f2-27bdbb74b93d,outbound,2011-03-30*>* 15:25:45,1301491545,sofia/external/0143708446 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0143708446,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0143708446,RECV,70c6ec80-78fe-4f1f-9e3a-732d544f74f2*>* dab44fd9-8f4b-42fd-855e-9ab4a5723687,outbound,2011-03-30*>* 15:25:46,1301491546,sofia/external/0134988997 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0134988997,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0134988997,RECV,2b4c5913-4131-4387-94d1-30ab4366a49e*>* 0ec7264f-5eaf-4f19-9b40-792ccf62b401,outbound,2011-03-30*>* 15:25:47,1301491547,sofia/external/0141322567 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0141322567,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0141322567,RECV,15d192af-716e-44cb-9d00-08a9ebbdbb79*>* 5a984ed3-6dd8-4991-912d-877469896bfa,outbound,2011-03-30*>* 15:25:49,1301491549,sofia/external/0145234010 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0145234010,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound*>* Call,0145234010,RECV,ee680d98-9bec-4830-ae67-9d8a70aa3b08*>* 688a2a96-bb50-4d12-8fe9-ec9423a42bd7,outbound,2011-03-30*>* 15:25:49,1301491549,sofia/external/0134747924 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0134747924,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0134747924,RECV,fb9aecb6-85e8-4d3e-8c0c-b8a252425e71*>* 3402d9bd-e33e-4e6f-9bc2-43ab559fa8aa,outbound,2011-03-30*>* 15:25:49,1301491549,sofia/external/0147239297 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0147239297,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,0147239297,0147239297,RECV,0924c5b1-fd35-40de-b47e-526b767cfce1*>* 66101a12-5c62-4d03-9eb3-dd0a7a73a22d,outbound,2011-03-30*>* 15:25:50,1301491550,sofia/external/0142533724 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0142533724,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0142533724,RECV,55115eca-c7c3-4657-98a8-fc21f13a10c9*>* 905d5a84-c7e2-4218-881f-39655efd040a,outbound,2011-03-30*>* 15:25:50,1301491550,sofia/external/0140261579 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0140261579,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0140261579,RECV,b79dc96f-146d-4a94-9025-ea82e06d30e5*>* 0cc72f20-e347-47ad-aa7a-73cf5e486770,outbound,2011-03-30*>* 15:25:53,1301491553,sofia/external/0141734848 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0141734848,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,0141734848,0141734848,RECV,f3f48827-fe5c-4b92-a83e-8f1e88f5263d*>* 6a7c903d-a1b8-4898-92a9-17e8e21bc477,outbound,2011-03-30*>* 15:25:53,1301491553,sofia/external/0130926822 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0130926822,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0130926822,RECV,1a6d58ae-d93b-4a82-a311-e478dcf6c972*>* cf48337a-bd0e-4739-8aad-dd01534de325,outbound,2011-03-30*>* 15:25:55,1301491555,sofia/external/0139194547 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0139194547,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0139194547,RECV,2687f485-f872-44e9-b04b-3ab45ccdbda2*>* 9c4e2841-0d70-49bd-9876-298cdbfd07d8,outbound,2011-03-30*>* 15:26:01,1301491561,sofia/external/0142560840 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0142560840,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0142560840,RECV,223d8889-5006-4043-8792-88f0059a726f*>* 097527a5-af50-4544-95f9-41e5d2db4329,outbound,2011-03-30*>* 15:26:01,1301491561,sofia/external/0139339445 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0139339445,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0139339445,RECV,ef818d92-7a22-4d00-8e17-8f6428db7255*>* 0c734637-b2fb-4b3d-b4f3-33bc1d6565e5,outbound,2011-03-30*>* 15:26:05,1301491565,sofia/external/0134868273 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0134868273,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0134868273,RECV,2364ea0b-d80c-44d5-915c-b4ee0498f50d*>* 65a8d0fc-26e9-4f21-9117-1d833111a428,outbound,2011-03-30*>* 15:26:07,1301491567,sofia/external/0139199893 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0139199893,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0139199893,RECV,a14ad031-c059-45a7-a822-4fbfddeb56fd*>* ea33d5ab-4dc5-45a8-8e5d-6dfbeb65c1b0,outbound,2011-03-30*>* 15:26:08,1301491568,sofia/external/0140209135 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0140209135,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0140209135,RECV,0786ebcf-be37-48f0-b00c-6e1b6f22af75*>* 473be842-169d-4ea3-8c86-c7f1d0bc01db,outbound,2011-03-30*>* 15:26:08,1301491568,sofia/external/0142861137 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0142861137,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0142861137,RECV,b4ae8669-20f0-423f-951a-a14affbe76e1*>* f25d32b4-3b8e-42aa-a38a-b3fc7b11bf97,outbound,2011-03-30*>* 15:26:09,1301491569,sofia/external/0134524074 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0134524074,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0134524074,RECV,9f82e9cb-5f79-4f76-8e1e-7f8cf8fb39e1*>* 0694de2a-1251-4ba7-a5e3-302403bda1ec,outbound,2011-03-30*>* 15:26:09,1301491569,sofia/external/0130326184 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0130326184,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0130326184,RECV,3dd6da1f-a59d-4b72-8b87-0657aee2a7eb*>* 6db28a24-99c7-4050-9c7e-7df94ee05860,outbound,2011-03-30*>* 15:26:10,1301491570,sofia/external/0130510861 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0130510861,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0130510861,RECV,f1d0b27f-a7d0-4dd0-a2a1-873525e6ed47*>* 676d1965-4bd0-43a3-85a2-84d1c618098c,outbound,2011-03-30*>* 15:26:12,1301491572,sofia/external/0139880615 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0139880615,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0139880615,RECV,69fb0f7e-b279-46a0-a9ff-0a685b2a2d9a*>* 10efef01-29bb-4181-bf34-70390cdbe964,outbound,2011-03-30*>* 15:26:13,1301491573,sofia/external/0139902937 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0139902937,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0139902937,RECV,505c212f-7459-4ace-836c-7d35737a5749*>* 9efbd4c6-17f9-4035-b185-84c53f8e67f5,outbound,2011-03-30*>* 15:26:14,1301491574,sofia/external/0130344121 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0130344121,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0130344121,RECV,2a79f551-dd7a-454d-af0b-b893a05a3bee*>* f111d343-9a56-4d18-890f-9482f7743264,outbound,2011-03-30*>* 15:26:16,1301491576,sofia/external/0144311010 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0144311010,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0144311010,RECV,eb7e7eed-d4d5-46ad-ac91-6a90b3bab9b6*>* a2f60b85-204b-4dd7-933d-af04d5807b3c,outbound,2011-03-30*>* 15:26:32,1301491592,sofia/external/0148831626 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0148831626,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0148831626,RECV,3bd290c2-2275-428c-9edd-e940c659e730*>* 9a40f095-69e6-4a1b-97ea-d464c6ea2f4a,outbound,2011-03-30*>* 15:26:39,1301491599,sofia/external/0140419154 at 192.168.0.1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>,CS_EXECUTE,,0000000000,,0140419154,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound*>* Call,0140419154,RECV,d79fa0b6-6e77-4fb8-8d37-bcd1d7f68a8d*>* 50 total.*>* Thank you =)*>* --*>* Mathieu LAUTRAM*>* Application developer*>**>* BJT Partners - FRANCE*>* +33 1 79 75 99 60*>* +33 6 61 59 07 25*>* _______________________________________________*>* FreeSWITCH-users mailing list*>* FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>*>* http://lists.freeswitch.org/mailman/listinfo/freeswitch-users*>* UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users*>* 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
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From steveayre at gmail.com  Thu Mar 31 16:35:16 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 31 Mar 2011 13:35:16 +0100
Subject: [Freeswitch-users] XML parser bug
In-Reply-To: <AANLkTi=-yaOswXcV7iSt=r1QtAXPD7Eu8Za652XaZAez@mail.gmail.com>
References: <AANLkTi=2zkLLKyLg9qaKDsF=y_oKqUEo_ErKwvEpgkvM@mail.gmail.com>
	<AANLkTimVxhaN4f3TU3Ni9vv0FxjRaYz=H1rGY9aZvFje@mail.gmail.com>
	<AANLkTinUhFVvbx364NVyNgC0_jApVyi9E+p8jb_59efw@mail.gmail.com>
	<AANLkTi=qJn+GL+4m6bXW2VSdMUDsWFcGqnh7WsY6P6it@mail.gmail.com>
	<AANLkTi=-yaOswXcV7iSt=r1QtAXPD7Eu8Za652XaZAez@mail.gmail.com>
Message-ID: <AANLkTimGoP07q3dzpvfEtdZNXCUo+acqJVB8EUswnj89@mail.gmail.com>

Which warnings?

"WARNING: Wasting up to 8 megs of memory per thread." only appears if you're
giving -waste

"Error: stacksize x is too large" will only appear if you haven't set
"ulimit -s" correctly. If you're on 64bit I don't think it appears at all.

-Steve



On 31 March 2011 11:36, David Villasmil <david.villasmil.work at gmail.com>wrote:

> Hello,
>
> that's just testing :P i just don't like the warnings when testing
> I don't run it like that for production.
>
> David
>
>
> On Wed, Mar 30, 2011 at 11:55 PM, Eliot Gable <egable+freeswitch at gmail.com
> > wrote:
>
>> On a side note, why are you running with -waste flag? You really should
>> not be doing that unless you have very good and very specific reasons to do
>> it and you know what that does and why you want to do it. Perhaps you do,
>> but I would double check. Personally, I've run FS on several different
>> versions of Linux without -waste for two years without ever needing it.
>>
>>
>> On Wed, Mar 30, 2011 at 8:54 AM, David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> Hello Joao,
>>>
>>> Ok, thanks
>>>
>>> David
>>>
>>>
>>> 2011/3/30 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>
>>>> This is not a bug and has been discussed several times on this mailing
>>>> list. You can't comment X-PRE-PROCESS tags like that. Make a quick google
>>>> search and you'll find several discussions about that including an
>>>> explanation from Tony on the subject.
>>>>
>>>> Regards,
>>>> Jo?o Mesquita
>>>>
>>>>
>>>>
>>>> On Wed, Mar 30, 2011 at 9:21 AM, David Villasmil <
>>>> david.villasmil.work at gmail.com> wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> I noticed the following:
>>>>>
>>>>> I have my sofia.conf.xml like this:
>>>>>
>>>>> <configuration name="sofia.conf" description="sofia Endpoint">
>>>>>
>>>>>   <global_settings>
>>>>>     <param name="log-level" value="0"/>
>>>>>
>>>>>     <!-- <param name="auto-restart" value="false"/> -->
>>>>>
>>>>>     <param name="debug-presence" value="0"/>
>>>>>   </global_settings>
>>>>>
>>>>>   <!--
>>>>>            The rabbit hole goes deep.  This includes all the
>>>>>            profiles in the sip_profiles directory that is up
>>>>>            one level from this directory.
>>>>>        <profiles>
>>>>>          <X-PRE-PROCESS cmd="include" data="../sip_profiles/*.xml" />
>>>>>        </profiles>
>>>>>   -->
>>>>>
>>>>> </configuration>
>>>>>
>>>>> when I start FS, latest GIT:
>>>>> freeswitch -version
>>>>> FreeSWITCH version: 1.0.head (git-7e52acf 2011-03-28 22-18-47 -0500)
>>>>>
>>>>> I get the following output:
>>>>>
>>>>> ./freeswitch -waste
>>>>> WARNING: Wasting up to 8 megs of memory per thread.
>>>>> 2011-03-30 14:02:23.200097 [INFO] switch_event.c:615 Activate Eventing
>>>>> Engine.
>>>>> 2011-03-30 14:02:23.211052 [DEBUG] switch_event.c:594 Create event
>>>>> dispatch thread 0
>>>>> Cannot Initialize [[error near line 1521]: unclosed <!--]
>>>>>
>>>>> If i then change my sofia to:
>>>>>
>>>>> <configuration name="sofia.conf" description="sofia Endpoint">
>>>>>
>>>>>   <global_settings>
>>>>>     <param name="log-level" value="0"/>
>>>>>
>>>>>     <!-- <param name="auto-restart" value="false"/> -->
>>>>>
>>>>>     <param name="debug-presence" value="0"/>
>>>>>   </global_settings>
>>>>>
>>>>>   <!--
>>>>>            The rabbit hole goes deep.  This includes all the
>>>>>            profiles in the sip_profiles directory that is up
>>>>>            one level from this directory.
>>>>>        <profiles>
>>>>>        </profiles>
>>>>>   -->
>>>>>
>>>>> </configuration>
>>>>>
>>>>>
>>>>> Please note the absence of:          <X-PRE-PROCESS cmd="include"
>>>>> data="../sip_profiles/*.xml" />
>>>>>
>>>>>
>>>>> FS Starts normally!
>>>>>
>>>>> Is this the correct behaviour? Isn't comments supposed NOT to be read?
>>>>>
>>>>> Thanks all.
>>>>>
>>>>>
>>>>>
>>>>> David
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Eliot Gable
>>
>> "We do not inherit the Earth from our ancestors: we borrow it from our
>> children." ~David Brower
>>
>> "I decided the words were too conservative for me. We're not borrowing
>> from our children, we're stealing from them--and it's not even considered to
>> be a crime." ~David Brower
>>
>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live;
>> not live to eat.) ~Marcus Tullius Cicero
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From me at nevian.org  Thu Mar 31 16:45:39 2011
From: me at nevian.org (Serge S. Yuriev)
Date: Thu, 31 Mar 2011 16:45:39 +0400
Subject: [Freeswitch-users] Why FS rewrites From header?
Message-ID: <538261301575539@web100.yandex.ru>


Hello,

Please see highlighted lines - why on B-leg From trashed? Is my misconfiguration?

Log
http://pastebin.freeswitch.org/15937

Configs
http://pastebin.freeswitch.org/15941
http://pastebin.freeswitch.org/15939
http://pastebin.freeswitch.org/15940


-- 
wbr,
Serge


From mitch.capper at gmail.com  Thu Mar 31 16:56:56 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Thu, 31 Mar 2011 05:56:56 -0700
Subject: [Freeswitch-users] Freeswitch latest source rebuild issue under
	Windows
In-Reply-To: <C42A22554C0CF44DAC0AA1F132B86DE65716763FAB@ASKSERVER.asksbs.local>
References: <C42A22554C0CF44DAC0AA1F132B86DE65716763FA6@ASKSERVER.asksbs.local>
	<AANLkTinLmBcybYuz1EyEFOjHV7bdd3DtcfeLVrP=ToZB@mail.gmail.com>
	<C42A22554C0CF44DAC0AA1F132B86DE65716763FA9@ASKSERVER.asksbs.local>
	<AANLkTimWpJonmpawmd1oTkREM72BgzV8gV_QU3vBSFUQ@mail.gmail.com>
	<C42A22554C0CF44DAC0AA1F132B86DE65716763FAB@ASKSERVER.asksbs.local>
Message-ID: <AANLkTin5cvrmihpdHn474bVVCoDfG_34RqDjNOqn=rNC@mail.gmail.com>

Thanks!

~Mitch

On Wed, Mar 30, 2011 at 3:13 PM, Andrew Keil
<andrew.keil at askinteractive.net> wrote:
> Mitch,
>
> I have edited the wiki (http://wiki.freeswitch.org/wiki/Installation_for_Windows) to include this information.
>
> Thanks,
>
> Andrew Keil
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitch Capper
> Sent: Thursday, 31 March 2011 1:19 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Freeswitch latest source rebuild issue under Windows
>
> Thats the great part of the wiki anyone can edit it:)
>
> ~Mitch
>
> On Wed, Mar 30, 2011 at 3:25 AM, Andrew Keil <andrew.keil at askinteractive.net> wrote:
>> Mitch,
>>
>> Well spotted! ?I simply unchecked mod_managed from the build list and now nothing fails!
>>
>> I usually use Visual Studio Pro and completely forgot about express being split into C++, C# etc....
>>
>> Perhaps the Windows installation documentation (http://wiki.freeswitch.org/wiki/Installation_for_Windows) on freeswitch.org should be updated to include this one. ?Especially since it mentions the Express version and only mentions VC++ Express.
>>
>> Now time to do some basic testing.
>>
>> Thanks,
>>
>> Andrew Keil
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Mitch Capper
>> Sent: Wednesday, 30 March 2011 8:53 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Freeswitch latest source rebuild issue
>> under Windows
>>
>> Hi Andrew,
>> It is saying it can't find the managed dll. ?Visual Studio Express
>> comes in different versions with there being a different version for
>> c#. ? I am not sure if you can build the c# module from the c++ edition, although if you don't need .net/mod_managed support you may be able to just not compile/use that module.
>>
>> ~Mitch
>>
>> On Wed, Mar 30, 2011 at 12:41 AM, Andrew Keil <andrew.keil at askinteractive.net> wrote:
>>> Jeff,
>>>
>>> Just tried the latest (git head). ?Note: autocrlf option is set to
>>> false (tested via: git config --get-all core.autocrlf)
>>>
>>> Almost there. ?The original fatal error has gone now.
>>>
>>> The only error left is the following:
>>>
>>> ------ Build started: Project: mod_managed, Configuration: Debug_CLR
>>> Win32 ------
>>> ?freeswitch_managed.cpp
>>> freeswitch_managed.cpp : fatal error C1192: #using failed on 'C:\FreeSWITCH\Win32\Debug\mod\FreeSWITCH.Managed.dll'
>>> ? ? ? ? ?'The system cannot find the file specified.'
>>> ?freeswitch_wrap.2010.cxx
>>> freeswitch_wrap.2010.cxx : fatal error C1192: #using failed on 'C:\FreeSWITCH\Win32\Debug\mod\FreeSWITCH.Managed.dll'
>>> ? ? ? ? ?'The system cannot find the file specified.'
>>> ?mod_managed.cpp
>>> mod_managed.cpp : fatal error C1192: #using failed on 'C:\FreeSWITCH\Win32\Debug\mod\FreeSWITCH.Managed.dll'
>>> ? ? ? ? ?'The system cannot find the file specified.'
>>> ?Running Code Analysis for C/C++...
>>> ?Generating Code...
>>>
>>> I tried to rebuild a few times, however the same error above happened. ?Any ideas?
>>>
>>> See attached for my build output log.
>>>
>>> Regards,
>>>
>>> Andrew Keil
>>>
>>>
>>>
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>>> Jeff Lenk
>>> Sent: Wednesday, 30 March 2011 3:39 PM
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] Freeswitch latest source rebuild
>>> issue under Windows
>>>
>>> Please try git head just submitted fixes
>>>
>>> --
>>> View this message in context:
>>> http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-latest-sourc
>>> e -rebuild-issue-under-Windows-tp6221661p6221815.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>>> e
>>> rs
>>> http://www.freeswitch.org
>>>
>>>
>>> __________ Information from ESET NOD32 Antivirus, version of virus
>>> signature database 5998 (20110329) __________
>>>
>>> The message was checked by ESET NOD32 Antivirus.
>>>
>>> http://www.eset.com
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>>> e
>>> rs
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>> http://www.freeswitch.org
>>
>>
>> __________ Information from ESET NOD32 Antivirus, version of virus
>> signature database 5999 (20110330) __________
>>
>> The message was checked by ESET NOD32 Antivirus.
>>
>> http://www.eset.com
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 6001 (20110330) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From nagalenoj at gmail.com  Thu Mar 31 17:03:21 2011
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Thu, 31 Mar 2011 18:33:21 +0530
Subject: [Freeswitch-users] Knowing TON of each call
Message-ID: <AANLkTimBkFTEB+thcUXw9So1Cwf2Z2oVbkbTrBSfXh6T@mail.gmail.com>

Dear Friends,
   I'm using ftmod_sangom_isdn in my setup and I would want to know the
TON(Type of Numbering) presented for each call. Is it possible to know the
TON through channel variable in freeswitch?!

   Based on the Type Of Numbering of a call, I would want to modify the
caller_id to a standard format and store in my database.

It's always pleasure working with FS. Thanks.
-- 
Nagalenoj H.
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From peter.olsson at visionutveckling.se  Thu Mar 31 17:14:13 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 31 Mar 2011 15:14:13 +0200
Subject: [Freeswitch-users] Knowing TON of each call
In-Reply-To: <AANLkTimBkFTEB+thcUXw9So1Cwf2Z2oVbkbTrBSfXh6T@mail.gmail.com>
References: <AANLkTimBkFTEB+thcUXw9So1Cwf2Z2oVbkbTrBSfXh6T@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58BE0F4C0B@cooper>

Try the variable sip_h_X-FreeTDM-ANI-TON

Or just run the "info" app on the incoming call to show all variables.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Nagalenoj H.
Skickat: den 31 mars 2011 15:03
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] Knowing TON of each call

Dear Friends,
   I'm using ftmod_sangom_isdn in my setup and I would want to know the TON(Type of Numbering) presented for each call. Is it possible to know the TON through channel variable in freeswitch?!

   Based on the Type Of Numbering of a call, I would want to modify the caller_id to a standard format and store in my database.

It's always pleasure working with FS. Thanks.
--
Nagalenoj H.
!DSPAM:4d947c2432764944577751!
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From jeff at jefflenk.com  Thu Mar 31 18:57:31 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 31 Mar 2011 07:57:31 -0700 (PDT)
Subject: [Freeswitch-users] stuck channels
In-Reply-To: <AANLkTimykLH-7i1PivZK8SQHrAFRzuXDMr0QjPPX2Ky3@mail.gmail.com>
References: <AANLkTimykLH-7i1PivZK8SQHrAFRzuXDMr0QjPPX2Ky3@mail.gmail.com>
Message-ID: <1301583451169-6227463.post@n2.nabble.com>

So you used make current and the problem still occurs?

Is it always related to fax calls?

If the above is correct you should file this a Jira with all relevant
information.

see http://wiki.freeswitch.org/wiki/Reporting_Bugs

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/stuck-channels-tp6226874p6227463.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From dyatsin at sangoma.com  Thu Mar 31 19:33:16 2011
From: dyatsin at sangoma.com (David Yat Sin)
Date: Thu, 31 Mar 2011 11:33:16 -0400
Subject: [Freeswitch-users] Knowing TON of each call
In-Reply-To: <AANLkTimBkFTEB+thcUXw9So1Cwf2Z2oVbkbTrBSfXh6T@mail.gmail.com>
References: <AANLkTimBkFTEB+thcUXw9So1Cwf2Z2oVbkbTrBSfXh6T@mail.gmail.com>
Message-ID: <4D949EBC.4010503@sangoma.com>

Hi Nagalenoj,
You can obtain the TON on incoming calls via the following channel 
variables:

caller_ton: TON from the Calling Party Number
destination_number_ton: TON from the Called Party Number

Regards,
David

*David Yat Sin, BEng*
*/Senior Software Engineer/*
Sangoma Technologies
100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada

	



t.   +1 800 388 2475 x119
t.   +1 905 474 1990 x119
f.   +1 905 474 9223

	

Description: SANGOMA 
<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>

Products 
<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 
| Solutions 
<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 
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| Wiki 
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| Facebook 
<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 
| Twitter 
<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>


On 3/31/2011 9:03 AM, Nagalenoj H. wrote:
> Dear Friends,
>    I'm using ftmod_sangom_isdn in my setup and I would want to know 
> the TON(Type of Numbering) presented for each call. Is it possible to 
> know the TON through channel variable in freeswitch?!
>
>    Based on the Type Of Numbering of a call, I would want to modify 
> the caller_id to a standard format and store in my database.
>
> It's always pleasure working with FS. Thanks.
> -- 
> Nagalenoj H.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From lautram.mathieu at gmail.com  Thu Mar 31 19:36:38 2011
From: lautram.mathieu at gmail.com (Mathieu Lautram)
Date: Thu, 31 Mar 2011 17:36:38 +0200
Subject: [Freeswitch-users] stuck channels
In-Reply-To: <1301583451169-6227463.post@n2.nabble.com>
References: <AANLkTimykLH-7i1PivZK8SQHrAFRzuXDMr0QjPPX2Ky3@mail.gmail.com>
	<1301583451169-6227463.post@n2.nabble.com>
Message-ID: <AANLkTik5ahRedWCETwPhYB8kSqtGckX-vTGA2aXL2ZG1@mail.gmail.com>

Thank you for your answer.
Yes, I use the last Freeswitch (FreeSWITCH Version 1.0.head (git-6e78f6f
2011-03-30 11-41-45 +0200)) and the problem is still there and it's always
related to fax calls.
I'm reporting this bug to Jira, hope it will be fix soon.


2011/3/31 Jeff Lenk <jeff at jefflenk.com>

> So you used make current and the problem still occurs?
>
> Is it always related to fax calls?
>
> If the above is correct you should file this a Jira with all relevant
> information.
>
> see http://wiki.freeswitch.org/wiki/Reporting_Bugs
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/stuck-channels-tp6226874p6227463.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Mathieu LAUTRAM
Application developer

BJT Partners - FRANCE
+33 1 79 75 99 60
+33 6 61 59 07 25
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From max.clark at gmail.com  Thu Mar 31 19:41:10 2011
From: max.clark at gmail.com (Max Clark)
Date: Thu, 31 Mar 2011 08:41:10 -0700
Subject: [Freeswitch-users] PRI Test Equipment
In-Reply-To: <2E80FBC43F4F464AB3730923CA7CC754@dell9400>
References: <AANLkTi=txKH5Yj3MHaMrX1uHhnqMpuS=PuLgGsxpR=H8@mail.gmail.com>
	<AANLkTi=emvxczkxopd4+OYLdY7gJjyd7qgONO8sK-PKT@mail.gmail.com>
	<AANLkTinexeb64gsV_ih8fwub1F=K8V+fKEJtEPUeWuwQ@mail.gmail.com>
	<2E80FBC43F4F464AB3730923CA7CC754@dell9400>
Message-ID: <AANLkTi=t418uxQL37eaD-HCmky=7UOCPsH0NyLbkgybQ@mail.gmail.com>

Thanks Jan I'll check this out.

On Wed, Mar 30, 2011 at 5:18 PM, Jan Berger <jan.berger at video24.no> wrote:
> http://www.utelsystems.com/instruments/isdn-pra/index.php
>
> Utel Systems have a software + hardware package, and I believe their pricing
> is "reasonable" - you are in USD 5000.- ++ range, but that is worth it if
> you ever need a 3rd party tester/analyzer for conformance testing.
>
> I have always wanted to try them, but I question their conformance & support
> on standards used outside Norway. Usage of terms like PRA rather than PRI
> etc + I know their main customer is the Norwegian PTT.
>
> Other than that their package looks very awesome.
>
> Jan
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Max
> Clark
> Sent: 29. mars 2011 19:47
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] PRI Test Equipment
>
> I want to display/read back the digits passed on the PRI.
>
> On Tue, Mar 29, 2011 at 10:02 AM, Michael Collins <msc at freeswitch.org>
> wrote:
>> depends on what all you want the appliance to do. if you just want to
>> do basic testing then you can buy or rent a "T-BERD" (google it) or
>> something similar.
>>
>> -MC
>>
>> On Tue, Mar 29, 2011 at 9:54 AM, Max Clark <max.clark at gmail.com> wrote:
>>> I'm looking for an appliance that can be plugged into a PRI connected
>>> to a PBX and either display or read back the digits that are passed to
>>> it. Before I build something to simulate this, is there anything out
>>> there commercially that can be purchased?
>>>
>>> Thanks,
>>> Max
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From infos at madovsky.org  Thu Mar 31 20:33:05 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 31 Mar 2011 12:33:05 -0400
Subject: [Freeswitch-users] user and public dialplan
Message-ID: <5FE749A45FF3409AADF1D9B6AE0C2EA1@e1705>

example:
- 9999 extension exisits in conf/directory
- no public dialplan that matches 9999

external call is coming to public dialplan.
is FS will consider that 9999 exists or not ?

Thanks
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From infos at madovsky.org  Thu Mar 31 20:48:12 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 31 Mar 2011 12:48:12 -0400
Subject: [Freeswitch-users] user and public dialplan
Message-ID: <E1899A6727B348329FE4734599F4232A@e1705>

forgot to say
it's an external call but from inside a cluster
and the dialstring is like /sofia/external/9999 at domain.ltd

thanks
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, March 31, 2011 12:33 PM
  Subject: user and public dialplan


  example:
  - 9999 extension exisits in conf/directory
  - no public dialplan that matches 9999

  external call is coming to public dialplan.
  is FS will consider that 9999 exists or not ?

  Thanks
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From juan.wajnerman at gmail.com  Thu Mar 31 17:25:40 2011
From: juan.wajnerman at gmail.com (Juan Wajnerman)
Date: Thu, 31 Mar 2011 10:25:40 -0300
Subject: [Freeswitch-users] Gateway with dynamic IP address
Message-ID: <C9C8E1CC-4621-44FF-8D19-D8DB35D41530@gmail.com>

I asked this question yesterday in the IRC but I couldn't get a solution.
I'd like to have a gateway configured in FreeSwitch without specifying the static IP address.
I have this configuration:

<user id="gw">
   <gateways>
     <gateway name="gw">
       <param name="username" value="gw"/>
       <param name="password" value="password"/>
       <param name="register" value="false"/>
       <param name="extension" value="2"/>
     </gateway>
   </gateways>
   <params>
     <param name="password" value="password"/>
   </params>
   <variables>
     <variable name="user_context" value="default" />
   </variables>
 </user>

and the SIP device is registering properly, but I cannot dial with addresses like: "sofia/gateway/gw/123456789".
Note that this works if the gateway name is the IP address or host name, or if I add a "proxy" setting with the IP address.

I have a similar configuration in asterisk, where the sip.conf contains:

[gw]
type=friend
secret=password
context=default
host=dynamic

And once the gateway is registered in asterisk, I can dial with "SIP/gw/123456789".
Is there any way to make a similar configuration in FreeSwitch?

Thanks!
- Juan

From infos at madovsky.org  Thu Mar 31 21:10:58 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 31 Mar 2011 13:10:58 -0400
Subject: [Freeswitch-users] incoming fax calls
Message-ID: <2AEA11608B0642348D4C867C5058F0AC@e1705>

I'm trying to find a way to dectect a fax or call from the same extension

        <extension name="myphone">
                <condition field="${sip_to_uri}" expression="^(9999)@$${domain}$">
                        <action inline="true" application="export" data="dialed_extension=$1"/>
                        <action application="set" data="session_in_hangup_hook=true"/>
                        <action application="set" data="hangup_after_bridge=true"/>
                        <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,CALL_REJECTED,NORMAL_CLEARING,USER_NOT_REGISTERED,NO_ANSWER,NO_USER_RESPONSE,USER_BUSY"/>
                        <action application="set" data="group_confirm_cancel_timeout=1"/>
                        <action application="ring_ready"/>
                        <action application="set" data="execute_on_media=tone_detect fax 1100 r +5000 transfer 'receivefax XML features' 1"/>
                        <action application="export" data="originate_timeout=25"/>
                        <action application="bridge" data="{sip_invite_domain=${sip_from_host},nibble_account=,nibble_rate=,origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number,}}user/${dialed_extension}"/>
                </condition>
         </extension>

is there a way to detect a fax before answer (2 rings for example) and avoid 
phone rings until no answer ?

thanks
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From dome at tel.co.th  Thu Mar 31 21:33:25 2011
From: dome at tel.co.th (dome at tel.co.th)
Date: Fri, 1 Apr 2011 00:33:25 +0700
Subject: [Freeswitch-users] Multitenant and text chat via sip
Message-ID: <AANLkTinPjkoisyk==Nmqe9HbNb2i5boauSf21rVUNkpt@mail.gmail.com>

Hi all,
When i setup multitenant text chat via sip don't work. i got error
invalid profile

2011-04-01 00:32:08.114100 [ERR] sofia_presence.c:133 Chat proto [sip]
from [0838833133 at wellcom.callbuzz.net]
to [0816167580 at wellcom.callbuzz.net]
Xxx
Invalid Profile wellcom.callbuzz.net

wellcom.callbuzz.net is one of domain.

It's bug or miss config somewhere in sip profile ?

BG
Dome C.


From covici at ccs.covici.com  Thu Mar 31 22:58:55 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 31 Mar 2011 14:58:55 -0400
Subject: [Freeswitch-users] user and public dialplan
In-Reply-To: <5FE749A45FF3409AADF1D9B6AE0C2EA1@e1705>
References: <5FE749A45FF3409AADF1D9B6AE0C2EA1@e1705>
Message-ID: <16998.1301597935@ccs.covici.com>

What I do is to put some extension in the public dial plan which finds
out who is calling -- ip address or whatever and take the appropriate
action -- transfer to somewhere or whatever you want. 

Hope this helps.

Madovsky <infos at madovsky.org> wrote:

> example:
> - 9999 extension exisits in conf/directory
> - no public dialplan that matches 9999
> 
> external call is coming to public dialplan.
> is FS will consider that 9999 exists or not ?
> 
> Thanks
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From steveayre at gmail.com  Thu Mar 31 23:58:10 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 31 Mar 2011 20:58:10 +0100
Subject: [Freeswitch-users] Why FS rewrites From header?
In-Reply-To: <538261301575539@web100.yandex.ru>
References: <538261301575539@web100.yandex.ru>
Message-ID: <AANLkTin2X8FvBhPf65rDXqDXMv8O36FwUS_Par9iPR3s@mail.gmail.com>

FreeSWITCH is a B2BUA, not a proxy.

The aleg and bleg are 2 different separate calls, and FS joins the
signalling media on the 2.

The From etc headers have to have the address of FS because that's what's
making the call.

-Steve


On 31 March 2011 13:45, Serge S. Yuriev <me at nevian.org> wrote:

>
> Hello,
>
> Please see highlighted lines - why on B-leg From trashed? Is my
> misconfiguration?
>
> Log
> http://pastebin.freeswitch.org/15937
>
> Configs
> http://pastebin.freeswitch.org/15941
> http://pastebin.freeswitch.org/15939
> http://pastebin.freeswitch.org/15940
>
>
> --
> wbr,
> Serge
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Info at KennedySoftware.ie  Thu Mar 31 22:05:54 2011
From: Info at KennedySoftware.ie (Michael Kennedy)
Date: Thu, 31 Mar 2011 19:05:54 +0100
Subject: [Freeswitch-users] Recommended SIP IP Wifi Handsets?
Message-ID: <4D94C282.1090903@KennedySoftware.ie>

Hello,

Newbie here, and newbie on FS - but been lurking for a few YEARS!...

I'm hoping to roll out FS where some areas in a building are wired, and 
other areas are on WiFi, and to deploy some SIP phones in both areas.

I expected that many phone suppliers would have handsets with EITHER 
RJ45 or WiFi connectivity to the LAN, or even both! I've found only a 
single device, a Cisco SPA525G2! Furthermore, searching the FS site, and 
various VoIP sites, and running general searches, I've found no other 
SIP WiFi phones that look like standard desktop handsets.

I'd appreciate any pointers to WiFi devices that are recommended with 
FS. Preferably "standard-looking" desktop units, and better still, if 
they had wired "sisters" - in appearance and functionality!

Or... maybe best to invest in a few rolls of Cat-6 cable!!

Thank you - and many thanks again to some of the regulars here for 
off-list guidance.
   - Mike