From Nabble at slickdeals.endjunk.com  Tue Feb  1 00:01:35 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 31 Jan 2011 13:01:35 -0800 (PST)
Subject: [Freeswitch-users] Problem with mod_shout on FS git
In-Reply-To: <AANLkTikvNTa0OueJHE5qLwnknXDJim100DJJqttt1txF@mail.gmail.com>
References: <1296491863148-5977722.post@n2.nabble.com>
	<AANLkTikvNTa0OueJHE5qLwnknXDJim100DJJqttt1txF@mail.gmail.com>
Message-ID: <1296507695419-5978666.post@n2.nabble.com>



Anthony Minessale wrote:
> Maybe they put in a fake error to stop people from borrowing their audio
> engine.
Interesting and honestly I never thought of that. Anyway, I went ahead to
our  http://wiki.freeswitch.org/wiki/Mod_shout FS wiki mod_shout , followed
one of the examples as shown below, and ended up with the same error
messages. I can't check if the scfire-dll-aa02.stream.aol.com:80/stream/1074
source is broken. If anyone out here has a live stream that works with
mod_shout, please kindly give it a shout so that I can give it a try.

<extension name="shout">
  <condition field="destination_number" expression="^750$">
    <action application="answer"/>
    <action application="playback"
data="shout://scfire-dll-aa02.stream.aol.com:80/stream/1074"/>
  </condition>
</extension>

-----
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/Problem-with-mod-shout-on-FS-git-tp5977722p5978666.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From chris at cloudtel.com  Tue Feb  1 00:20:07 2011
From: chris at cloudtel.com (Chris Burns)
Date: Mon, 31 Jan 2011 16:20:07 -0500
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTimVrhpfX+0D2mFPwGHK1EDsALWjFGp8hh7=ecqt@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
	<AANLkTi=in3CLd-n0mT3SRKYA0kxLMGVWk9KeBx3_jhY6@mail.gmail.com>
	<AANLkTin3Bu+8EEr_v_GaRGNPzOjfcnfzsV2Av2cgnWgO@mail.gmail.com>
	<AANLkTi=OC_jx0+nzK405VZURLDJPD25R4ZjyJ3RXJ1nk@mail.gmail.com>
	<AANLkTi=Sm7CjoayiE8nHmL0Eehqknj1jdpbx0YBQJQGH@mail.gmail.com>
	<AANLkTinr0RdgBs2UiS3=b3i+k6uCwkMihDy_FbAOFPc_@mail.gmail.com>
	<AANLkTi=+aObW3kVBZq5ox4b4E1uFTSgtXUL72tHLOzPi@mail.gmail.com>
	<AANLkTikVDKS40dVsf8x=+5=qnW59LqJbSv1H=vtROFaM@mail.gmail.com>
	<AANLkTi=gJsVKCJifVaBL_H6ZnPHTD__aQPhPeVfdx-ju@mail.gmail.com>
	<AANLkTimeuCHf_7Y1TErPwrJbdGZ3DoQmeqQbJ1_3YNpT@mail.gmail.com>
	<AANLkTikDxzntMFRpFhfiqJ4pzvjsTQR4w_FfyLfT5Y7z@mail.gmail.com>
	<AANLkTimVrhpfX+0D2mFPwGHK1EDsALWjFGp8hh7=ecqt@mail.gmail.com>
Message-ID: <AANLkTi=oqw7fX-kvwqtxUXEWmzDu4C-dOjPrhpsCiwQc@mail.gmail.com>

Not sure the polycoms do it exactly how he would like. Rebooting phones can
trip up many versions of soundpoint firmware. In our case we had a big
opportunity with a client who was stuck on such a version of soundpoint,
with an integrated web app on their 650s. Patrick, if this code works for
you dont let it stop you from applying pressure on your vendor to make your
phones work the way they should. We all have to put the pressure on our
vendors when they make mistakes ... after all these folks are charging
money, as opposed to FreeSWITCH.

Basically this program will store and repeat - on interval - any unique
PRESENCE_IN events it hears from the 'presence' app or api (mod_dptools). It
will also override events from outside the dialplan with stored events (eg.
from a misbehaving SIP phone). You need to build the php module for ESL to
run the code
http://wiki.freeswitch.org/wiki/Event_Socket_Library#Installation

WARNING: this code is just an example pulled from a larger presence
application. I tested it in my lab with polycoms, but I cant guarantee your
safety, molten lava, bat country, etc. This may not be a very smart way to
solve the problem, but this is what I came up with when backed into a corner
and you are welcome to it ...

http://pastebin.freeswitch.org/15193


On Mon, Jan 31, 2011 at 3:04 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> You are heading the wrong way down the ladder but its worth a try.
> Grandstream is notorious for not following specs =D
>
> I have engaged SNOM to again try to sync up for interop.  Stay tuned.
> I would really prefer to have them working.
>
> Meanwhile try firmware 7.1.35 on your snom as that was before things
> started to break.
>
> Also if you have polycom or linksys they have been known to work
> better with blf.
>
>
>
> On Mon, Jan 31, 2011 at 1:19 PM, Patrick Plattes
> <patrick.plattes at niemann-frey.info> wrote:
> > Hi :-)
> >
> > I think SNOM has a problem with the quality of the firmware. maybe
> > they should glp it, so we can fix it ;-)
> >
> > Tomorrow I will test it with an Grandstream GPX-2000 maybe it works
> > with this device. If it does I will also ask SNOM for a fix.
> >
> > Thanks,
> >  Patrick
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From paul at cupis.co.uk  Tue Feb  1 00:28:36 2011
From: paul at cupis.co.uk (Paul Cupis)
Date: Mon, 31 Jan 2011 21:28:36 +0000
Subject: [Freeswitch-users] Outbound faxes fail
In-Reply-To: <AANLkTim12xoc6M+ktBS_Uf5HFd25Q-H3P+sKdN4dpJR0@mail.gmail.com>
References: <AANLkTin7g6g9CwvvLhq_smi4BZvUEMy9OB-k34-g0qfJ@mail.gmail.com>	<13F4AC41-3E40-43A3-B155-2A6FE638B432@ipeva.fr>	<AANLkTimiYu1d8q5=+-o5zh71Fw9b=90DbjCpR_phPTVG@mail.gmail.com>	<AE397829-B4F5-411B-90FB-06B8E387F098@ipeva.fr>	<AANLkTikk-h7UFpBd-aZdAvMpSFFriPDANW_iTqaU4gRK@mail.gmail.com>
	<AANLkTim12xoc6M+ktBS_Uf5HFd25Q-H3P+sKdN4dpJR0@mail.gmail.com>
Message-ID: <4D472984.5020700@cupis.co.uk>

On 31/01/11 20:55, Jonas Gauffin wrote:
> Here is the reinvite when I've enabled t.38:
> 
> Have I configured something
> incorrectly or why do my gw provider respond with 488?

It looks like your provider does not support T.38. You should ask them
to confirm whether they support it or not, and if they say they do ask
them to look at a failed call.


From jonas.gauffin at gmail.com  Tue Feb  1 00:41:41 2011
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Mon, 31 Jan 2011 22:41:41 +0100
Subject: [Freeswitch-users] Outbound faxes fail
In-Reply-To: <4D472984.5020700@cupis.co.uk>
References: <AANLkTin7g6g9CwvvLhq_smi4BZvUEMy9OB-k34-g0qfJ@mail.gmail.com>
	<13F4AC41-3E40-43A3-B155-2A6FE638B432@ipeva.fr>
	<AANLkTimiYu1d8q5=+-o5zh71Fw9b=90DbjCpR_phPTVG@mail.gmail.com>
	<AE397829-B4F5-411B-90FB-06B8E387F098@ipeva.fr>
	<AANLkTikk-h7UFpBd-aZdAvMpSFFriPDANW_iTqaU4gRK@mail.gmail.com>
	<AANLkTim12xoc6M+ktBS_Uf5HFd25Q-H3P+sKdN4dpJR0@mail.gmail.com>
	<4D472984.5020700@cupis.co.uk>
Message-ID: <AANLkTinnrmw8E7O701f2x-wuE=OLJV5qdizAeQGYtm1W@mail.gmail.com>

Their specification says that they do. I've sent the trace to them.

Thanks

On Mon, Jan 31, 2011 at 10:28 PM, Paul Cupis <paul at cupis.co.uk> wrote:

> On 31/01/11 20:55, Jonas Gauffin wrote:
> > Here is the reinvite when I've enabled t.38:
> >
> > Have I configured something
> > incorrectly or why do my gw provider respond with 488?
>
> It looks like your provider does not support T.38. You should ask them
> to confirm whether they support it or not, and if they say they do ask
> them to look at a failed call.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From marcin321 at hotmail.com  Tue Feb  1 01:07:31 2011
From: marcin321 at hotmail.com (Marcin Wojtowicz)
Date: Mon, 31 Jan 2011 17:07:31 -0500
Subject: [Freeswitch-users] Outbound only calls don't connect when
 bypass_media is true.
In-Reply-To: <AANLkTikSs4sn-+x5G9pRpszogW7Hwng_cLJop0zrAueu@mail.gmail.com>
References: <COL118-W52E5226FC3337085AB6AFAE5E00@phx.gbl>,
	<COL118-W38F26048D441DAA2777942E5E30@phx.gbl>,
	<AANLkTinbiU4zNc-ZM+u+iST7E2aZ3zisgh5Lu2FTuBLt@mail.gmail.com>,
	<COL118-W196844C55E71088CAD1FC1E5E20@phx.gbl>,
	<AANLkTi=t0ixs6L3SgUMQzEPtgvFowbFYXCVOtmvzLTEW@mail.gmail.com>,
	<COL118-W34303A74089852D47F0E63E5E20@phx.gbl>,
	<E826C94A3C6BE44198FAF4D66B5AB4D80FEC3E5B24@VA3DIAXVS351.RED001.local>,
	<COL118-W33A8A74086A6BBE2E6B1B7E5E20@phx.gbl>,
	<AANLkTikSs4sn-+x5G9pRpszogW7Hwng_cLJop0zrAueu@mail.gmail.com>
Message-ID: <COL118-W5890CA4168B4879C9AE75CE5E20@phx.gbl>


I tried building with visual C++ express 2010, but there were numerous fatal errors and it didn't finish.

> Date: Mon, 31 Jan 2011 14:00:10 -0600
> From: anthony.minessale at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Outbound only calls don't connect when bypass_media is true.
> 
>  a=fmtp:98 mode=30 is missing in the 200 ok from the phone.
> Also did you mention the revision you are on. I had indicated that the
> very latest code may have more tolerant ilbc codec code in it.
> http://latest.freeswitch.org
> 
> 
> On Mon, Jan 31, 2011 at 12:12 PM, Marcin Wojtowicz
> <marcin321 at hotmail.com> wrote:
> > SDP looks ok to me, but there is one warning about ptime in iLBC below. I
> > don't see how a wrong codec can be selected because I narrowed down my
> > external profile inbound/outbound to PCMU only and my internal is iLBC at 30i
> > only.
> >
> >
> > freeswitch at kuffel> recv 1206 bytes from udp/[74.63.41.218]:5060 at
> > 17:59:32.187500:
> >    ------------------------------------------------------------------------
> >    INVITE sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms SIP/2.0
> >    Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK30d7505b;rport
> >    From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64
> >    To: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
> >    Contact: <sip:MYPHONE#@74.63.41.218>
> >    Call-ID: 1298959c0099d50d177b6bae689fc028 at 74.63.41.218
> >    CSeq: 102 INVITE
> >    User-Agent: VoIPMS/SERAST
> >    Max-Forwards: 70
> >    Remote-Party-ID: "MYPHONE#"
> > <sip:MYPHONE#@74.63.41.218>;privacy=off;screen=no
> >    Date: Mon, 31 Jan 2011 17:59:17 GMT
> >    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> >    Supported: replaces
> >    Content-Type: application/sdp
> >    Content-Length: 515
> >
> >    v=0
> >    o=root 2831 2831 IN IP4 74.63.41.218
> >    s=session
> >    c=IN IP4 74.63.41.218
> >    t=0 0
> >    m=audio 16884 RTP/AVP 0 4 3 8 112 5 10 7 18 111 101
> >    a=rtpmap:0 PCMU/8000
> >    a=rtpmap:4 G723/8000
> >    a=fmtp:4 annexa=no
> >    a=rtpmap:3 GSM/8000
> >    a=rtpmap:8 PCMA/8000
> >    a=rtpmap:112 AAL2-G726-32/8000
> >    a=rtpmap:5 DVI4/8000
> >    a=rtpmap:10 L16/8000
> >    a=rtpmap:7 LPC/8000
> >    a=rtpmap:18 G729/8000
> >    a=fmtp:18 annexb=no
> >    a=rtpmap:111 G726-32/8000
> >    a=rtpmap:101 telephone-event/8000
> >    a=fmtp:101 0-16
> >    a=silenceSupp:off - - - -
> >    a=ptime:20
> >    a=sendrecv
> >    ------------------------------------------------------------------------
> > send 396 bytes to udp/[74.63.41.218]:5060 at 17:59:32.187500:
> >    ------------------------------------------------------------------------
> >    SIP/2.0 100 Trying
> >    Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK30d7505b;rport=5060
> >    From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64
> >    To: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
> >    Call-ID: 1298959c0099d50d177b6bae689fc028 at 74.63.41.218
> >    CSeq: 102 INVITE
> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> > -0600
> >    Content-Length: 0
> >
> >    ------------------------------------------------------------------------
> > 2011-01-31 12:59:32.187500 [NOTICE] switch_channel.c:808 New Channel
> > sofia/external/MYPHONE#@74.63.41.218 [f35f408a-f863-4784-a308-8b4fb3284b70]
> > 2011-01-31 12:59:32.187500 [INFO] mod_dialplan_xml.c:331 Processing MYPHONE#
> > <MYPHONE#>->121628 in context public
> > 2011-01-31 12:59:32.203125 [NOTICE] switch_ivr.c:1606 Transfer
> > sofia/external/MYPHONE#@74.63.41.218 to XML[1001 at default]
> > 2011-01-31 12:59:32.203125 [INFO] mod_dialplan_xml.c:331 Processing MYPHONE#
> > <MYPHONE#>->1001 in context default
> > 2011-01-31 12:59:32.234375 [NOTICE] switch_channel.c:808 New Channel
> > sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060
> > [7230b9e8-37a7-4fc6-9b52-25740a6f7ca4]
> >
> >
> > 2011-01-31 12:59:32.265625 [WARNING] sofia_glue.c:213 Codec iLBC payload 98
> > added to sdp wanting ptime 30 but it's already 20 (PCMU:0:20), disabling
> > ptime.
> >
> >
> > send 1315 bytes to tcp/[32.140.14.196]:46743 at 17:59:32.265625:
> >    ------------------------------------------------------------------------
> >    INVITE sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP SIP/2.0
> >    Via: SIP/2.0/TCP 69.125.20.15;rport;branch=z9hG4bK797eFQ6rgQKmQ
> >    Route: <sip:M9jdt73ig0oOJSbt6Uyy at 32.140.14.196:46743>;transport=TCP
> >    Max-Forwards: 68
> >    From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa
> >    To: <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP>
> >    Call-ID: b281a1fa-a806-122e-f799-c1188a708e17
> >    CSeq: 7912322 INVITE
> >    Contact: <sip:mod_sofia at 69.125.20.15:5060;transport=tcp>
> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> > -0600
> >    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: 234
> >    X-FS-Support: update_display
> >    Remote-Party-ID: "MYPHONE#"
> > <sip:MYPHONE#@192.168.1.100>;party=calling;screen=no;privacy=off
> >
> >    v=0
> >    o=FreeSWITCH 1296474878 1296474879 IN IP4 69.125.20.15
> >    s=FreeSWITCH
> >    c=IN IP4 69.125.20.15
> >    t=0 0
> >    m=audio 21894 RTP/AVP 0 98 101 13
> >    a=rtpmap:98 iLBC/8000
> >    a=fmtp:98 mode=30
> >    a=rtpmap:101 telephone-event/8000
> >    a=fmtp:101 0-16
> >    ------------------------------------------------------------------------
> > recv 318 bytes from tcp/[32.140.14.196]:46743 at 17:59:37.093750:
> >    ------------------------------------------------------------------------
> >    SIP/2.0 100 Trying
> >    Via: SIP/2.0/TCP
> > 69.125.20.15;branch=z9hG4bK797eFQ6rgQKmQ;rport=5060;received=69.125.20.15
> >    To: <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155>
> >    From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa
> >    Call-ID: b281a1fa-a806-122e-f799-c1188a708e17
> >    CSeq: 7912322 INVITE
> >    Content-Length: 0
> >
> >    ------------------------------------------------------------------------
> > recv 476 bytes from tcp/[32.140.14.196]:46743 at 17:59:42.296875:
> >    ------------------------------------------------------------------------
> >    SIP/2.0 180 Ringing
> >    Via: SIP/2.0/TCP
> > 69.125.20.15;branch=z9hG4bK797eFQ6rgQKmQ;rport=5060;received=69.125.20.15
> >    Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP>
> >    From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa
> >    To:
> > <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155>;tag=p4rl1jbfvmnbvfs1d5rktoj2
> >    Supported: 100rel
> >    Call-ID: b281a1fa-a806-122e-f799-c1188a708e17
> >    CSeq: 7912322 INVITE
> >    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE
> >    Content-Length: 0
> >
> >    ------------------------------------------------------------------------
> > 2011-01-31 12:59:42.296875 [INFO] sofia.c:729
> > sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060 Update Callee ID
> > to "Outbound Call" <M9jdt73ig0oOJSbt6Uyy>
> > 2011-01-31 12:59:42.296875 [NOTICE] sofia.c:4724 Ring-Ready
> > sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060!
> > 2011-01-31 12:59:42.312500 [INFO] switch_ivr_originate.c:1101 Sending early
> > media
> > 2011-01-31 12:59:42.343750 [NOTICE] mod_sofia.c:2252 Pre-Answer
> > sofia/external/MYPHONE#@74.63.41.218!
> > send 1079 bytes to udp/[74.63.41.218]:5060 at 17:59:42.343750:
> >    ------------------------------------------------------------------------
> >    SIP/2.0 183 Session Progress
> >    Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK30d7505b;rport=5060
> >    From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64
> >    To:
> > <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>;tag=eK0X80BS0091S
> >    Call-ID: 1298959c0099d50d177b6bae689fc028 at 74.63.41.218
> >    CSeq: 102 INVITE
> >    Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp>
> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> > -0600
> >    Accept: application/sdp
> >    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: 247
> >    Remote-Party-ID: "121628"
> > <sip:121628 at 192.168.1.100>;party=calling;privacy=off;screen=no
> >
> >    v=0
> >    o=FreeSWITCH 1296476876 1296476877 IN IP4 69.125.20.15
> >    s=FreeSWITCH
> >    c=IN IP4 69.125.20.15
> >    t=0 0
> >    m=audio 19906 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
> >    ------------------------------------------------------------------------
> > recv 772 bytes from tcp/[32.140.14.196]:46743 at 17:59:43.812500:
> >    ------------------------------------------------------------------------
> >    SIP/2.0 200 OK
> >    Via: SIP/2.0/TCP
> > 69.125.20.15;branch=z9hG4bK797eFQ6rgQKmQ;rport=5060;received=69.125.20.15
> >    To:
> > <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155>;tag=p4rl1jbfvmnbvfs1d5rktoj2
> >    Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP>
> >    From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa
> >    Supported: timer
> >    Call-ID: b281a1fa-a806-122e-f799-c1188a708e17
> >    CSeq: 7912322 INVITE
> >    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE
> >    Content-Type: application/sdp
> >    Content-Length: 269
> >
> >    v=0
> >    o=M9jdt73ig0oOJSbt6Uyy 63464734759229750 63464734759229750 IN IP4
> > 10.208.245.155
> >    s=-
> >    c=IN IP4 10.208.245.155
> >    t=0 0
> >    m=audio 49152 RTP/AVP 98 101
> >    a=sendrecv
> >    a=rtpmap:98 iLBC/8000
> >    a=ptime:30
> >    a=maxptime:180
> >    a=rtpmap:101 telephone-event/8000
> >    a=fmtp:101 0-15
> >    ------------------------------------------------------------------------
> > send 464 bytes to tcp/[32.140.14.196]:46743 at 17:59:43.828125:
> >    ------------------------------------------------------------------------
> >    ACK sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP SIP/2.0
> >    Via: SIP/2.0/TCP 69.125.20.15;rport;branch=z9hG4bK8j17gjQvD096j
> >    Max-Forwards: 70
> >    From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa
> >    To:
> > <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP>;tag=p4rl1jbfvmnbvfs1d5rktoj2
> >    Call-ID: b281a1fa-a806-122e-f799-c1188a708e17
> >    CSeq: 7912322 ACK
> >    Contact: <sip:mod_sofia at 69.125.20.15:5060;transport=tcp>
> >    Content-Length: 0
> >
> >    ------------------------------------------------------------------------
> > 2011-01-31 12:59:43.828125 [NOTICE] sofia.c:5230 Channel
> > [sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060] has been
> > answered
> > send 1061 bytes to udp/[74.63.41.218]:5060 at 17:59:43.843750:
> >    ------------------------------------------------------------------------
> >    SIP/2.0 200 OK
> >    Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK30d7505b;rport=5060
> >    From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64
> >    To:
> > <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>;tag=eK0X80BS0091S
> >    Call-ID: 1298959c0099d50d177b6bae689fc028 at 74.63.41.218
> >    CSeq: 102 INVITE
> >    Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp>
> > 2011-01-31 12:59:43.843750 [NOTICE] switch_ivr_originate.c:3328 Channel
> > [sofia/external/MYPHONE#@74.63.41.218] has been answered
> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> > -0600
> >    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: 247
> >    Remote-Party-ID: "Outbound Call"
> > <sip:M9jdt73ig0oOJSbt6Uyy at 192.168.1.100>;party=calling;privacy=off;screen=no
> >
> >    v=0
> >    o=FreeSWITCH 1296476876 1296476877 IN IP4 69.125.20.15
> >    s=FreeSWITCH
> >    c=IN IP4 69.125.20.15
> >    t=0 0
> >    m=audio 19906 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
> >    ------------------------------------------------------------------------
> > recv 533 bytes from udp/[74.63.41.218]:5060 at 17:59:43.859375:
> >    ------------------------------------------------------------------------
> >    ACK sip:gw+voip.ms at 69.125.20.15:5080;transport=udp SIP/2.0
> >    Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK0cbbc6ef;rport
> >    From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64
> >    To:
> > <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>;tag=eK0X80BS0091S
> >    Contact: <sip:MYPHONE#@74.63.41.218>
> >    Call-ID: 1298959c0099d50d177b6bae689fc028 at 74.63.41.218
> >    CSeq: 102 ACK
> >    User-Agent: VoIPMS/SERAST
> >    Max-Forwards: 70
> >    Remote-Party-ID: "MYPHONE#"
> > <sip:MYPHONE#@74.63.41.218>;privacy=off;screen=no
> >    Content-Length: 0
> >
> > ________________________________
> > From: robert.hadley at teotech.com
> > To: freeswitch-users at lists.freeswitch.org
> > Date: Mon, 31 Jan 2011 09:33:48 -0800
> > Subject: Re: [Freeswitch-users] Outbound only calls don't connect when
> > bypass_media is true.
> >
> >
> >
> > Check the codecs in the SDP or try manual hardcoding the codecs presented
> > for both legs, we had a squeal problem going to a softphone that turned out
> > to be the BV32 codec was being selected instead of SPEEX16.
> >
> >
> >
> > Robert
> >
> >
> >
> > From: Marcin Wojtowicz [mailto:marcin321 at hotmail.com]
> > Sent: Monday, January 31, 2011 9:17 AM
> > To: freeswitch
> > Subject: Re: [Freeswitch-users] Outbound only calls don't connect when
> > bypass_media is true.
> >
> >
> >
> > Yes, I had it set up to iLBC at 30i. It's not my cell phone (configured to
> > ilbc, ptime=30 and mode=30), because when I call my freeswitch voicemail
> > number, the sound is fine. I suspect it is something on the voip.ms <->
> > freeswitch leg because I created a sample ringback (8khz, mono, 16bit) wave
> > file and directed my dialplan to it, but when I call from my home number to
> > my cell, instead of hearing the ringer, I get choppy squeal.
> >
> > ?
> >
> >> Date: Mon, 31 Jan 2011 10:40:10 -0600
> >> From: anthony.minessale at gmail.com
> >> To: freeswitch-users at lists.freeswitch.org
> >> Subject: Re: [Freeswitch-users] Outbound only calls don't connect when
> >> bypass_media is true.
> >>
> >> Many things have problems doing iLBC right.
> >> I recommend you define it in your configs as iLBC at 30i or it will try
> >> using the 20ms version which is not compatible with many other
> >> platforms. Also make sure you are on the latest version of FS since
> >> we have tweaked iLBC behavior to compensate for problems like this.
> >>
> >>
> >>
> >> On Sun, Jan 30, 2011 at 10:40 PM, Marcin Wojtowicz
> >> <marcin321 at hotmail.com> wrote:
> >> > OK, so I gave up on bypass media, but now I have another problem. This
> >> > time
> >> > I set up freeswitch to communicate with voip.ms using PCMU codec
> >> > (configured
> >> > in my external profile), and use iLBC on my phone (codec configured in
> >> > my
> >> > internal profile, where the phone registers). When I call my mobile it
> >> > rings, but when I pick up all I hear is a high pitched squeal. Am I
> >> > missing
> >> > something here?
> >> >
> >> >> Date: Sun, 30 Jan 2011 16:34:09 -0600
> >> >> From: anthony.minessale at gmail.com
> >> >> To: freeswitch-users at lists.freeswitch.org
> >> >> Subject: Re: [Freeswitch-users] Outbound only calls don't connect when
> >> >> bypass_media is true.
> >> >>
> >> >> Just do not use bypass media.
> >> >> That is all you can do in that situation.
> >> >>
> >> >>
> >> >> On Sun, Jan 30, 2011 at 3:44 PM, Marcin Wojtowicz
> >> >> <marcin321 at hotmail.com>
> >> >> wrote:
> >> >> > I just want to add that I enabled STUN on my cell so now the SDP
> >> >> > message
> >> >> > in
> >> >> > the INVITE to voip.ms contains the public IP of my phone, but it
> >> >> > still
> >> >> > doesn't work.
> >> >> >
> >> >> > ________________________________
> >> >> > From: marcin321 at hotmail.com
> >> >> > To: freeswitch-users at lists.freeswitch.org
> >> >> > Date: Fri, 28 Jan 2011 19:54:19 -0500
> >> >> > Subject: [Freeswitch-users] Outbound only calls don't connect when
> >> >> > bypass_media is true.
> >> >> >
> >> >> > Hello,
> >> >> >
> >> >> > I'm a new user of freeswitch, so please bear with me. I have the
> >> >> > following setup:
> >> >> > voip.ms <- SIP over UDP -> my desktop running freeswitch <- SIP over
> >> >> > TCP
> >> >> > ->
> >> >> > my nokia cellphone on AT&T wireless. This setup is intended to
> >> >> > conserve
> >> >> > the
> >> >> > battery usage.
> >> >> > I've managed to make everything work well when I'm calling in over
> >> >> > any
> >> >> > phone
> >> >> > to my cell phone, and freeswitch is enabled to work in bypass_media =
> >> >> > true,
> >> >> > even though by cell is behind NAT on at&t's network. Things break
> >> >> > when I
> >> >> > pick up my cell and try to call my home phone (or any phone for that
> >> >> > matter). This is the relevant snippet from my dialplan:
> >> >> > <extension name="outbound">
> >> >> >   <condition field="destination_number"
> >> >> > expression="^1?([2-9]\d{2}[2-9]\d{6})$">
> >> >> >     <!--<action application="set" data="bypass_media=true"/>-->
> >> >> >     <action application="bridge" data="sofia/gateway/voip.ms/1$1"/>
> >> >> >   </condition>
> >> >> > </extension>
> >> >> >
> >> >> > Like shown above, my call will go to my home phone. When I uncomment
> >> >> > the
> >> >> > bypass_media tag, my call will not connect. Here are the siptraces
> >> >> > I replaced my real home phone number in the with "MYPHONE".
> >> >> >
> >> >> > recv 940 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.406250:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> >> >> >    Via: SIP/2.0/TCP
> >> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport
> >> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >> >    To: <sip:MYPHONE at 192.168.1.100>
> >> >> >    Contact:
> >> >> > <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>
> >> >> >    Supported: 100rel,timer
> >> >> >    CSeq: 5244503 INVITE
> >> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >> >    Allow:
> >> >> > UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE
> >> >> >    User-Agent: S60 RM-624 v 20.2.042 (en)
> >> >> >    Expires: 120
> >> >> >    Privacy: None
> >> >> >    Session-Expires: 1800
> >> >> >    Max-Forwards: 70
> >> >> >    Content-Type: application/sdp
> >> >> >    Accept-Language: en
> >> >> >    Content-Length: 292
> >> >> >
> >> >> >    v=0
> >> >> >    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> >> >> >    s=-
> >> >> >    c=IN IP4 10.153.174.6
> >> >> >    t=0 0
> >> >> >    m=audio 49152 RTP/AVP 18 97 98
> >> >> >    a=sendrecv
> >> >> >    a=rtpmap:18 G729/8000
> >> >> >    a=ptime:20
> >> >> >    a=maxptime:40
> >> >> >    a=fmtp:18 annexb=no
> >> >> >    a=rtpmap:97 iLBC/8000
> >> >> >    a=rtpmap:98 telephone-event/8000
> >> >> >    a=fmtp:98 0-15
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    SIP/2.0 100 Trying
> >> >> >    Via: SIP/2.0/TCP
> >> >> >
> >> >> >
> >> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180
> >> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >> >    To: <sip:MYPHONE at 192.168.1.100>
> >> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >> >    CSeq: 5244503 INVITE
> >> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> >> > 18-04-05
> >> >> > -0600
> >> >> >    Content-Length: 0
> >> >> >
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > send 875 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    SIP/2.0 407 Proxy Authentication Required
> >> >> >    Via: SIP/2.0/TCP
> >> >> >
> >> >> >
> >> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180
> >> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj2011-01-28
> >> >> > 16:15:58.406250 [WARNING] sofia_reg.c:1241 SIP auth challenge
> >> >> > (INVITE)
> >> >> > on
> >> >> > sofia profile 'internal' for [MYPHONE at 192.168.1.100] from ip
> >> >> > 32.136.78.180
> >> >> >
> >> >> >    To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m
> >> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >> >    CSeq: 5244503 INVITE
> >> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> >> > 18-04-05
> >> >> > -0600
> >> >> >    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
> >> >> >    Proxy-Authenticate: Digest realm="192.168.1.100",
> >> >> > nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28", algorithm=MD5,
> >> >> > qop="auth"
> >> >> >    Content-Length: 0
> >> >> >
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > recv 368 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.765625:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> >> >> >    Via: SIP/2.0/TCP
> >> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport
> >> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >> >    To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m
> >> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >> >    CSeq: 5244503 ACK
> >> >> >    Supported: sec-agree
> >> >> >    Max-Forwards: 70
> >> >> >    Content-Length: 0
> >> >> >
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > recv 1222 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.406250:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> >> >> >    Via: SIP/2.0/TCP
> >> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport
> >> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >> >    To: <sip:MYPHONE at 192.168.1.100>
> >> >> >    Contact:
> >> >> > <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>
> >> >> >    Supported: 100rel,timer
> >> >> >    CSeq: 5244504 INVITE
> >> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >> >    Allow:
> >> >> > UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE
> >> >> >    User-Agent: S60 RM-624 v 20.2.042 (en)
> >> >> >    Expires: 120
> >> >> >    Privacy: None
> >> >> >    Session-Expires: 1800
> >> >> >    Max-Forwards: 70
> >> >> >    Proxy-Authorization: Digest
> >> >> >
> >> >> >
> >> >> > qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"
> >> >> >    Content-Type: application/sdp
> >> >> >    Accept-Language: en
> >> >> >    Content-Length: 292
> >> >> >
> >> >> >    v=0
> >> >> >    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> >> >> >    s=-
> >> >> >    c=IN IP4 10.153.174.6
> >> >> >    t=0 0
> >> >> >    m=audio 49152 RTP/AVP 18 97 98
> >> >> >    a=sendrecv
> >> >> >    a=rtpmap:18 G729/8000
> >> >> >    a=ptime:20
> >> >> >    a=maxptime:40
> >> >> >    a=fmtp:18 annexb=no
> >> >> >    a=rtpmap:97 iLBC/8000
> >> >> >    a=rtpmap:98 telephone-event/8000
> >> >> >    a=fmtp:98 0-15
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.406250:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    SIP/2.0 100 Trying
> >> >> >    Via: SIP/2.0/TCP
> >> >> >
> >> >> >
> >> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180
> >> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >> >    To: <sip:MYPHONE at 192.168.1.100>
> >> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >> >    CSeq: 5244504 INVITE
> >> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> >> > 18-04-05
> >> >> > -0600
> >> >> >    Content-Length: 0
> >> >> >
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > 2011-01-28 16:15:59.406250 [NOTICE] switch_channel.c:808 New Channel
> >> >> > sofia/internal/1001 at 192.168.1.100
> >> >> > [e5841001-04bd-4e16-9519-64ff2c7a8c2f]
> >> >> > 2011-01-28 16:15:59.421875 [INFO] mod_dialplan_xml.c:331 Processing
> >> >> > 1001
> >> >> > <1001>->MYPHONE in context default
> >> >> > 2011-01-28 16:15:59.437500 [NOTICE] switch_channel.c:808 New Channel
> >> >> > sofia/external/1MYPHONE [60940227-9ae0-4c0c-abdb-f3988852fae0]
> >> >> > send 1141 bytes to udp/[74.63.41.218]:5060 at 21:15:59.453125:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0
> >> >> >    Via: SIP/2.0/UDP
> >> >> > 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS
> >> >> >    Max-Forwards: 69
> >> >> >    From: "Extension 1001"
> >> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >> >    To: <sip:1MYPHONE at newyork.voip.ms>
> >> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >> >    CSeq: 7788615 INVITE
> >> >> >    Contact:
> >> >> > <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
> >> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> >> > 18-04-05
> >> >> > -0600
> >> >> >    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: 280
> >> >> >    X-FS-Support: update_display
> >> >> >    Remote-Party-ID: "Extension 1001"
> >> >> > <sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off
> >> >> >
> >> >> >    v=0
> >> >> >    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> >> >> >    s=-
> >> >> >    c=IN IP4 10.153.174.6
> >> >> >    t=0 0
> >> >> >    m=audio 49152 RTP/AVP 18 97 98
> >> >> >    a=rtpmap:18 G729/8000
> >> >> >    a=fmtp:18 annexb=no
> >> >> >    a=rtpmap:97 iLBC/8000
> >> >> >    a=rtpmap:98 telephone-event/8000
> >> >> >    a=fmtp:98 0-15
> >> >> >    a=ptime:20
> >> >> >    a=maxptime:40
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > recv 570 bytes from udp/[74.63.41.218]:5060 at 21:15:59.468750:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    SIP/2.0 407 Proxy Authentication Required
> >> >> >    Via: SIP/2.0/UDP
> >> >> >
> >> >> >
> >> >> > 69.125.20.15:5080;branch=z9hG4bKNp4ZFeKSD43tS;received=69.125.20.15;rport=5080
> >> >> >    From: "Extension 1001"
> >> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >> >    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843
> >> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >> >    CSeq: 7788615 INVITE
> >> >> >    User-Agent: VoIPMS/SERAST
> >> >> >    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> >> >> >    Supported: replaces
> >> >> >    Proxy-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms",
> >> >> > nonce="2d534dd6"
> >> >> >    Content-Length: 0
> >> >> >
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0
> >> >> >    Via: SIP/2.0/UDP
> >> >> > 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS
> >> >> >    Max-Forwards: 69
> >> >> >    From: "Extension 1001"
> >> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >> >    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843
> >> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >> >    CSeq: 7788615 ACK
> >> >> >    Content-Length: 0
> >> >> >
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > send 1344 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0
> >> >> >    Via: SIP/2.0/UDP
> >> >> > 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN
> >> >> >    Max-Forwards: 69
> >> >> >    From: "Extension 1001"
> >> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >> >    To: <sip:1MYPHONE at newyork.voip.ms>
> >> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >> >    CSeq: 7788616 INVITE
> >> >> >    Contact:
> >> >> > <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
> >> >> >    Expires: 300
> >> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> >> > 18-04-05
> >> >> > -0600
> >> >> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> >> > REGISTER, REFER, NOTIFY
> >> >> >    Supported: timer, precondition, path, replaces
> >> >> >    Allow-Events: talk, hold, refer
> >> >> >    Proxy-Authorization: Digest username="121628",
> >> >> > realm="newyork.voip.ms",
> >> >> > nonce="2d534dd6", algorithm=MD5, uri="sip:1MYPHONE at newyork.voip.ms",
> >> >> > response="16f3301efae13df926da7550f709d28a"
> >> >> >    Content-Type: application/sdp
> >> >> >    Content-Disposition: session
> >> >> >    Content-Length: 280
> >> >> >    X-FS-Support: update_display
> >> >> >    Remote-Party-ID: "Extension 1001"
> >> >> > <sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off
> >> >> >
> >> >> >    v=0
> >> >> >    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> >> >> >    s=-
> >> >> >    c=IN IP4 10.153.174.6
> >> >> >    t=0 0
> >> >> >    m=audio 49152 RTP/AVP 18 97 98
> >> >> >    a=rtpmap:18 G729/8000
> >> >> >    a=fmtp:18 annexb=no
> >> >> >    a=rtpmap:97 iLBC/8000
> >> >> >    a=rtpmap:98 telephone-event/8000
> >> >> >    a=fmtp:98 0-15
> >> >> >    a=ptime:20
> >> >> >    a=maxptime:40
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > recv 488 bytes from udp/[74.63.41.218]:5060 at 21:15:59.484375:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    SIP/2.0 100 Trying
> >> >> >    Via: SIP/2.0/UDP
> >> >> >
> >> >> >
> >> >> > 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080
> >> >> >    From: "Extension 1001"
> >> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >> >    To: <sip:1MYPHONE at newyork.voip.ms>
> >> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >> >    CSeq: 7788616 INVITE
> >> >> >    User-Agent: VoIPMS/SERAST
> >> >> >    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> >> >> >    Supported: replaces
> >> >> >    Contact: <sip:1MYPHONE at 74.63.41.218>
> >> >> >    Content-Length: 0
> >> >> >
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > recv 516 bytes from udp/[74.63.41.218]:5060 at 21:15:59.562500:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    SIP/2.0 503 Service Unavailable
> >> >> >    Via: SIP/2.0/UDP
> >> >> >
> >> >> >
> >> >> > 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080
> >> >> >    From: "Extension 1001"
> >> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >> >    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9
> >> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >> >    CSeq: 7788616 INVITE
> >> >> >    User-Agent: VoIPMS/SERAST
> >> >> >    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> >> >> >    Supported: replaces
> >> >> >    Contact: <sip:1MYPHONE at 74.63.41.218>
> >> >> >    Content-Length: 0
> >> >> >
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.562500:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0
> >> >> >    Via: SIP/2.0/UDP
> >> >> > 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN
> >> >> >    Max-Forwards: 69
> >> >> >    From: "Extension 1001"
> >> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >> >    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9
> >> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >> >    CSeq: 7788616 ACK
> >> >> >    Content-Length: 0
> >> >> >
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > 2011-01-28 16:15:59.562500 [INFO] mod_dptools.c:2612 Originate
> >> >> > Failed.
> >> >> > Cause: NO_ANSWER
> >> >> > 2011-01-28 16:15:59.562500 [NOTICE] sofia.c:5286 Hangup
> >> >> > sofia/external/1MYPHONE [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
> >> >> > 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:189
> >> >> > sofia/internal/1001 at 192.168.1.100 has executed the last dialplan
> >> >> > instruction, hanging up.
> >> >> > 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:191
> >> >> > Hangup
> >> >> > sofia/internal/1001 at 192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING]
> >> >> > 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1306
> >> >> > Session 2
> >> >> > (sofia/external/1MYPHONE) Ended
> >> >> > 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1308 Close
> >> >> > Channel
> >> >> > sofia/external/1MYPHONE [CS_DESTROY]
> >> >> > send 887 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.593750:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    SIP/2.0 503 Service Unavailable
> >> >> >    Via: SIP/2.0/TCP
> >> >> >
> >> >> >
> >> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180
> >> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >> >    To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg
> >> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >> >    CSeq: 5244504 INVITE
> >> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> >> > 18-04-05
> >> >> > -0600
> >> >> >    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
> >> >> >    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
> >> >> > 2011-01-28 16:15:59.593750 [NOTICE] switch_core_session.c:1306
> >> >> > Session 1
> >> >> > (sofia/internal/1001 at 192.168.1.100) Ended
> >> >> >    Content-Length: 02011-01-28 16:15:59.593750 [NOTICE]
> >> >> > switch_core_session.c:1308 Close Channel
> >> >> > sofia/internal/1001 at 192.168.1.100
> >> >> > [CS_DESTROY]
> >> >> >
> >> >> >    Remote-Party-ID: "MYPHONE"
> >> >> > <sip:MYPHONE at 192.168.1.100>;party=calling;privacy=off;screen=no
> >> >> >
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > recv 650 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.953125:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> >> >> >    Via: SIP/2.0/TCP
> >> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport
> >> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >> >    To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg
> >> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >> >    CSeq: 5244504 ACK
> >> >> >    Supported: sec-agree
> >> >> >    Max-Forwards: 70
> >> >> >    Proxy-Authorization: Digest
> >> >> >
> >> >> >
> >> >> > qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"
> >> >> >    Content-Length: 0
> >> >> >
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >
> >> >> > 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
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at 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
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 Feb  1 01:15:35 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 31 Jan 2011 16:15:35 -0600
Subject: [Freeswitch-users] freeswitch.com is returned to us!
Message-ID: <AANLkTi=BciYcFugPySe3QU7pK-=avKSpkOUiq119UCYP@mail.gmail.com>

http://www.freeswitch.com/

Thank you everyone who helped with this!



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Tue Feb  1 01:16:49 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 31 Jan 2011 16:16:49 -0600
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTi=oqw7fX-kvwqtxUXEWmzDu4C-dOjPrhpsCiwQc@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
	<AANLkTi=in3CLd-n0mT3SRKYA0kxLMGVWk9KeBx3_jhY6@mail.gmail.com>
	<AANLkTin3Bu+8EEr_v_GaRGNPzOjfcnfzsV2Av2cgnWgO@mail.gmail.com>
	<AANLkTi=OC_jx0+nzK405VZURLDJPD25R4ZjyJ3RXJ1nk@mail.gmail.com>
	<AANLkTi=Sm7CjoayiE8nHmL0Eehqknj1jdpbx0YBQJQGH@mail.gmail.com>
	<AANLkTinr0RdgBs2UiS3=b3i+k6uCwkMihDy_FbAOFPc_@mail.gmail.com>
	<AANLkTi=+aObW3kVBZq5ox4b4E1uFTSgtXUL72tHLOzPi@mail.gmail.com>
	<AANLkTikVDKS40dVsf8x=+5=qnW59LqJbSv1H=vtROFaM@mail.gmail.com>
	<AANLkTi=gJsVKCJifVaBL_H6ZnPHTD__aQPhPeVfdx-ju@mail.gmail.com>
	<AANLkTimeuCHf_7Y1TErPwrJbdGZ3DoQmeqQbJ1_3YNpT@mail.gmail.com>
	<AANLkTikDxzntMFRpFhfiqJ4pzvjsTQR4w_FfyLfT5Y7z@mail.gmail.com>
	<AANLkTimVrhpfX+0D2mFPwGHK1EDsALWjFGp8hh7=ecqt@mail.gmail.com>
	<AANLkTi=oqw7fX-kvwqtxUXEWmzDu4C-dOjPrhpsCiwQc@mail.gmail.com>
Message-ID: <AANLkTikWUfpFpQjKQ_EiGQuUGqa33VwU2K5vYVRvFGFO@mail.gmail.com>

bottom line is, if you use the presence_id how it was designed, it
will report any presence you want throughout the call.
If you reboot the phone, it will persist.  It's just a matter of doing
it all properly.


On Mon, Jan 31, 2011 at 3:20 PM, Chris Burns <chris at cloudtel.com> wrote:
> Not sure the polycoms do it exactly how he would like. Rebooting phones can
> trip up many versions of soundpoint firmware. In our case we had a big
> opportunity with a client who was stuck on such a version of soundpoint,
> with an integrated web app on their 650s. Patrick, if this code works for
> you dont let it stop you from applying pressure on your vendor to make your
> phones work the way they should. We all have to put the pressure on our
> vendors when they make mistakes ... after all these folks are charging
> money, as opposed to FreeSWITCH.
>
> Basically this program will store and repeat - on interval - any unique
> PRESENCE_IN events it hears from the 'presence' app or api (mod_dptools). It
> will also override events from outside the dialplan with stored events (eg.
> from a misbehaving SIP phone). You need to build the php module for ESL to
> run the code
> http://wiki.freeswitch.org/wiki/Event_Socket_Library#Installation
>
> WARNING: this code is just an example pulled from a larger presence
> application. I tested it in my lab with polycoms, but I cant guarantee your
> safety, molten lava, bat country, etc. This may not be a very smart way to
> solve the problem, but this is what I came up with when backed into a corner
> and you are welcome to it ...
>
> http://pastebin.freeswitch.org/15193
>
>
> On Mon, Jan 31, 2011 at 3:04 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> You are heading the wrong way down the ladder but its worth a try.
>> Grandstream is notorious for not following specs =D
>>
>> I have engaged SNOM to again try to sync up for interop. ?Stay tuned.
>> I would really prefer to have them working.
>>
>> Meanwhile try firmware 7.1.35 on your snom as that was before things
>> started to break.
>>
>> Also if you have polycom or linksys they have been known to work
>> better with blf.
>>
>>
>>
>> On Mon, Jan 31, 2011 at 1:19 PM, Patrick Plattes
>> <patrick.plattes at niemann-frey.info> wrote:
>> > Hi :-)
>> >
>> > I think SNOM has a problem with the quality of the firmware. maybe
>> > they should glp it, so we can fix it ;-)
>> >
>> > Tomorrow I will test it with an Grandstream GPX-2000 maybe it works
>> > with this device. If it does I will also ask SNOM for a fix.
>> >
>> > Thanks,
>> > ?Patrick
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at 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 infos at madovsky.org  Tue Feb  1 01:19:30 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 31 Jan 2011 17:19:30 -0500
Subject: [Freeswitch-users] freeswitch.com is returned to us!
References: <AANLkTi=BciYcFugPySe3QU7pK-=avKSpkOUiq119UCYP@mail.gmail.com>
Message-ID: <5FB9599FB0CA4FA496A87C65DAFB9D53@e1705>

yeah nice ! :D

----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com>
To: "Freeswitch-users" <freeswitch-users at lists.freeswitch.org>; 
<freeswitch-dev at lists.freeswitch.org>
Sent: Monday, January 31, 2011 5:15 PM
Subject: [Freeswitch-users] freeswitch.com is returned to us!


> http://www.freeswitch.com/
>
> Thank you everyone who helped with this!
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Tue Feb  1 01:22:23 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 31 Jan 2011 17:22:23 -0500
Subject: [Freeswitch-users] sip_contact_host
Message-ID: <C81EC95BE43D42DCB6FB1727EEF4AAB5@e1705>

I set my internal profile with a domain
but sip_contact_host stays an IP.
how to change it ?

Thanks
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From brian at freeswitch.org  Tue Feb  1 01:37:24 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 31 Jan 2011 16:37:24 -0600
Subject: [Freeswitch-users] sip_contact_host
In-Reply-To: <C81EC95BE43D42DCB6FB1727EEF4AAB5@e1705>
References: <C81EC95BE43D42DCB6FB1727EEF4AAB5@e1705>
Message-ID: <D786BA11-AEBB-4F4A-8E05-E32B9CE910F1@freeswitch.org>

set sip_invite_domain

/b

On Jan 31, 2011, at 4:22 PM, Madovsky wrote:

> I set my internal profile with a domain
> but sip_contact_host stays an IP.
> how to change it ?
>  
> Thanks
> __________

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From chris at cloudtel.com  Tue Feb  1 01:42:30 2011
From: chris at cloudtel.com (Chris Burns)
Date: Mon, 31 Jan 2011 17:42:30 -0500
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTikWUfpFpQjKQ_EiGQuUGqa33VwU2K5vYVRvFGFO@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
	<AANLkTi=in3CLd-n0mT3SRKYA0kxLMGVWk9KeBx3_jhY6@mail.gmail.com>
	<AANLkTin3Bu+8EEr_v_GaRGNPzOjfcnfzsV2Av2cgnWgO@mail.gmail.com>
	<AANLkTi=OC_jx0+nzK405VZURLDJPD25R4ZjyJ3RXJ1nk@mail.gmail.com>
	<AANLkTi=Sm7CjoayiE8nHmL0Eehqknj1jdpbx0YBQJQGH@mail.gmail.com>
	<AANLkTinr0RdgBs2UiS3=b3i+k6uCwkMihDy_FbAOFPc_@mail.gmail.com>
	<AANLkTi=+aObW3kVBZq5ox4b4E1uFTSgtXUL72tHLOzPi@mail.gmail.com>
	<AANLkTikVDKS40dVsf8x=+5=qnW59LqJbSv1H=vtROFaM@mail.gmail.com>
	<AANLkTi=gJsVKCJifVaBL_H6ZnPHTD__aQPhPeVfdx-ju@mail.gmail.com>
	<AANLkTimeuCHf_7Y1TErPwrJbdGZ3DoQmeqQbJ1_3YNpT@mail.gmail.com>
	<AANLkTikDxzntMFRpFhfiqJ4pzvjsTQR4w_FfyLfT5Y7z@mail.gmail.com>
	<AANLkTimVrhpfX+0D2mFPwGHK1EDsALWjFGp8hh7=ecqt@mail.gmail.com>
	<AANLkTi=oqw7fX-kvwqtxUXEWmzDu4C-dOjPrhpsCiwQc@mail.gmail.com>
	<AANLkTikWUfpFpQjKQ_EiGQuUGqa33VwU2K5vYVRvFGFO@mail.gmail.com>
Message-ID: <AANLkTi=z3yysHW7Y7YaB92H6CDe0GFStMdPjHyUBmSvG@mail.gmail.com>

For calls, presence_id works perfectly and that is what I use. In the case
of my client, they were using BLFs to show additional information like which
desks have spanish or french speaking agents (because the agents switch
around desks daily). The call is just to an extension to add/remove the
phone from a fifo queue, which then has to trigger an auxilary BLF on/off.
The call to the extension will be over and the phone's own BLF will unlight,
but the auxilary BLF should remain lit ... showing them as a proud member of
the spanish speaking fifo queue to the managers who rely on blinking red
lights to do so. The end result of my ESL app is that nothing can change my
auxilary BLFs except my dialplan, and any phones that reboot or get plugged
in new get lit up properly.

Was there a way I could have used presence_id and skipped using mod_dptools?
I didnt see how to do it.


On Mon, Jan 31, 2011 at 5:16 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> bottom line is, if you use the presence_id how it was designed, it
> will report any presence you want throughout the call.
> If you reboot the phone, it will persist.  It's just a matter of doing
> it all properly.
>
>
> On Mon, Jan 31, 2011 at 3:20 PM, Chris Burns <chris at cloudtel.com> wrote:
> > Not sure the polycoms do it exactly how he would like. Rebooting phones
> can
> > trip up many versions of soundpoint firmware. In our case we had a big
> > opportunity with a client who was stuck on such a version of soundpoint,
> > with an integrated web app on their 650s. Patrick, if this code works for
> > you dont let it stop you from applying pressure on your vendor to make
> your
> > phones work the way they should. We all have to put the pressure on our
> > vendors when they make mistakes ... after all these folks are charging
> > money, as opposed to FreeSWITCH.
> >
> > Basically this program will store and repeat - on interval - any unique
> > PRESENCE_IN events it hears from the 'presence' app or api (mod_dptools).
> It
> > will also override events from outside the dialplan with stored events
> (eg.
> > from a misbehaving SIP phone). You need to build the php module for ESL
> to
> > run the code
> > http://wiki.freeswitch.org/wiki/Event_Socket_Library#Installation
> >
> > WARNING: this code is just an example pulled from a larger presence
> > application. I tested it in my lab with polycoms, but I cant guarantee
> your
> > safety, molten lava, bat country, etc. This may not be a very smart way
> to
> > solve the problem, but this is what I came up with when backed into a
> corner
> > and you are welcome to it ...
> >
> > http://pastebin.freeswitch.org/15193
> >
> >
> > On Mon, Jan 31, 2011 at 3:04 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> You are heading the wrong way down the ladder but its worth a try.
> >> Grandstream is notorious for not following specs =D
> >>
> >> I have engaged SNOM to again try to sync up for interop.  Stay tuned.
> >> I would really prefer to have them working.
> >>
> >> Meanwhile try firmware 7.1.35 on your snom as that was before things
> >> started to break.
> >>
> >> Also if you have polycom or linksys they have been known to work
> >> better with blf.
> >>
> >>
> >>
> >> On Mon, Jan 31, 2011 at 1:19 PM, Patrick Plattes
> >> <patrick.plattes at niemann-frey.info> wrote:
> >> > Hi :-)
> >> >
> >> > I think SNOM has a problem with the quality of the firmware. maybe
> >> > they should glp it, so we can fix it ;-)
> >> >
> >> > Tomorrow I will test it with an Grandstream GPX-2000 maybe it works
> >> > with this device. If it does I will also ask SNOM for a fix.
> >> >
> >> > Thanks,
> >> >  Patrick
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at 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 <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  1 01:46:05 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 31 Jan 2011 16:46:05 -0600
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTi=z3yysHW7Y7YaB92H6CDe0GFStMdPjHyUBmSvG@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
	<AANLkTi=in3CLd-n0mT3SRKYA0kxLMGVWk9KeBx3_jhY6@mail.gmail.com>
	<AANLkTin3Bu+8EEr_v_GaRGNPzOjfcnfzsV2Av2cgnWgO@mail.gmail.com>
	<AANLkTi=OC_jx0+nzK405VZURLDJPD25R4ZjyJ3RXJ1nk@mail.gmail.com>
	<AANLkTi=Sm7CjoayiE8nHmL0Eehqknj1jdpbx0YBQJQGH@mail.gmail.com>
	<AANLkTinr0RdgBs2UiS3=b3i+k6uCwkMihDy_FbAOFPc_@mail.gmail.com>
	<AANLkTi=+aObW3kVBZq5ox4b4E1uFTSgtXUL72tHLOzPi@mail.gmail.com>
	<AANLkTikVDKS40dVsf8x=+5=qnW59LqJbSv1H=vtROFaM@mail.gmail.com>
	<AANLkTi=gJsVKCJifVaBL_H6ZnPHTD__aQPhPeVfdx-ju@mail.gmail.com>
	<AANLkTimeuCHf_7Y1TErPwrJbdGZ3DoQmeqQbJ1_3YNpT@mail.gmail.com>
	<AANLkTikDxzntMFRpFhfiqJ4pzvjsTQR4w_FfyLfT5Y7z@mail.gmail.com>
	<AANLkTimVrhpfX+0D2mFPwGHK1EDsALWjFGp8hh7=ecqt@mail.gmail.com>
	<AANLkTi=oqw7fX-kvwqtxUXEWmzDu4C-dOjPrhpsCiwQc@mail.gmail.com>
	<AANLkTikWUfpFpQjKQ_EiGQuUGqa33VwU2K5vYVRvFGFO@mail.gmail.com>
	<AANLkTi=z3yysHW7Y7YaB92H6CDe0GFStMdPjHyUBmSvG@mail.gmail.com>
Message-ID: <AANLkTinDmkxEeO2px7NsOSO5DkTiq+=1xTY7togH2-hJ@mail.gmail.com>

So sounds like the guy who started the thread should be able to get
what he needs if we can find out why it's not working on snom then.


On Mon, Jan 31, 2011 at 4:42 PM, Chris Burns <chris at cloudtel.com> wrote:
> For calls, presence_id works perfectly and that is what I use. In the case
> of my client, they were using BLFs to show additional information like which
> desks have spanish or french speaking agents (because the agents switch
> around desks daily). The call is just to an extension to add/remove the
> phone from a fifo queue, which then has to trigger an auxilary BLF on/off.
> The call to the extension will be over and the phone's own BLF will unlight,
> but the auxilary BLF should remain lit ... showing them as a proud member of
> the spanish speaking fifo queue to the managers who rely on blinking red
> lights to do so. The end result of my ESL app is that nothing can change my
> auxilary BLFs except my dialplan, and any phones that reboot or get plugged
> in new get lit up properly.
>
> Was there a way I could have used presence_id and skipped using mod_dptools?
> I didnt see how to do it.
>
>
> On Mon, Jan 31, 2011 at 5:16 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> bottom line is, if you use the presence_id how it was designed, it
>> will report any presence you want throughout the call.
>> If you reboot the phone, it will persist. ?It's just a matter of doing
>> it all properly.
>>
>>
>> On Mon, Jan 31, 2011 at 3:20 PM, Chris Burns <chris at cloudtel.com> wrote:
>> > Not sure the polycoms do it exactly how he would like. Rebooting phones
>> > can
>> > trip up many versions of soundpoint firmware. In our case we had a big
>> > opportunity with a client who was stuck on such a version of soundpoint,
>> > with an integrated web app on their 650s. Patrick, if this code works
>> > for
>> > you dont let it stop you from applying pressure on your vendor to make
>> > your
>> > phones work the way they should. We all have to put the pressure on our
>> > vendors when they make mistakes ... after all these folks are charging
>> > money, as opposed to FreeSWITCH.
>> >
>> > Basically this program will store and repeat - on interval - any unique
>> > PRESENCE_IN events it hears from the 'presence' app or api
>> > (mod_dptools). It
>> > will also override events from outside the dialplan with stored events
>> > (eg.
>> > from a misbehaving SIP phone). You need to build the php module for ESL
>> > to
>> > run the code
>> > http://wiki.freeswitch.org/wiki/Event_Socket_Library#Installation
>> >
>> > WARNING: this code is just an example pulled from a larger presence
>> > application. I tested it in my lab with polycoms, but I cant guarantee
>> > your
>> > safety, molten lava, bat country, etc. This may not be a very smart way
>> > to
>> > solve the problem, but this is what I came up with when backed into a
>> > corner
>> > and you are welcome to it ...
>> >
>> > http://pastebin.freeswitch.org/15193
>> >
>> >
>> > On Mon, Jan 31, 2011 at 3:04 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> You are heading the wrong way down the ladder but its worth a try.
>> >> Grandstream is notorious for not following specs =D
>> >>
>> >> I have engaged SNOM to again try to sync up for interop. ?Stay tuned.
>> >> I would really prefer to have them working.
>> >>
>> >> Meanwhile try firmware 7.1.35 on your snom as that was before things
>> >> started to break.
>> >>
>> >> Also if you have polycom or linksys they have been known to work
>> >> better with blf.
>> >>
>> >>
>> >>
>> >> On Mon, Jan 31, 2011 at 1:19 PM, Patrick Plattes
>> >> <patrick.plattes at niemann-frey.info> wrote:
>> >> > Hi :-)
>> >> >
>> >> > I think SNOM has a problem with the quality of the firmware. maybe
>> >> > they should glp it, so we can fix it ;-)
>> >> >
>> >> > Tomorrow I will test it with an Grandstream GPX-2000 maybe it works
>> >> > with this device. If it does I will also ask SNOM for a fix.
>> >> >
>> >> > Thanks,
>> >> > ?Patrick
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at 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 msc at freeswitch.org  Tue Feb  1 01:56:24 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Jan 2011 14:56:24 -0800
Subject: [Freeswitch-users] CDR Fields
In-Reply-To: <AANLkTim6xqDOGJ+s7B+kLPz2V+UBYG=Em6zKiS_UOHKn@mail.gmail.com>
References: <AANLkTik8anSQqoXFOb6jyX0Rx6f5BzTrDvoii2J_nPA=@mail.gmail.com>
	<AANLkTim6xqDOGJ+s7B+kLPz2V+UBYG=Em6zKiS_UOHKn@mail.gmail.com>
Message-ID: <AANLkTinaGc47UXsye_jnuE3Dg7cHHdHOs+yht53OY3+Y@mail.gmail.com>

On Mon, Jan 31, 2011 at 7:06 AM, Steven Ayre <steveayre at gmail.com> wrote:

> http://wiki.freeswitch.org/wiki/Channel_Variables
> XML CDR also contains extra information such as callflow.
>
> Can't really give you an answer on the quality information...
>
> -Steve
>
>
> http://wiki.freeswitch.org/wiki/Mod_xml_cdr#Channel_Variables

There is a brief description of each piece of data.
-MC
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From infos at madovsky.org  Tue Feb  1 02:08:06 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 31 Jan 2011 18:08:06 -0500
Subject: [Freeswitch-users] sip_contact_host
References: <C81EC95BE43D42DCB6FB1727EEF4AAB5@e1705>
	<D786BA11-AEBB-4F4A-8E05-E32B9CE910F1@freeswitch.org>
Message-ID: <3EC7C77C99CE4C058247FEA98A8BB9B0@e1705>

I just tried

sip_invite_domain=${sip_from_host}
in the bridge and in set application in dialplan
but Bleg still receives sip address with IP.
I tried with x-lite 3, 4 and Linphone

thanks
  ----- Original Message ----- 
  From: Brian West 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 31, 2011 5:37 PM
  Subject: Re: [Freeswitch-users] sip_contact_host


  set sip_invite_domain


  /b


  On Jan 31, 2011, at 4:22 PM, Madovsky wrote:


    I set my internal profile with a domain
    but sip_contact_host stays an IP.
    how to change 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
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From patrick.plattes at niemann-frey.info  Tue Feb  1 02:25:25 2011
From: patrick.plattes at niemann-frey.info (Patrick Plattes)
Date: Tue, 1 Feb 2011 00:25:25 +0100
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTinDmkxEeO2px7NsOSO5DkTiq+=1xTY7togH2-hJ@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
	<AANLkTi=in3CLd-n0mT3SRKYA0kxLMGVWk9KeBx3_jhY6@mail.gmail.com>
	<AANLkTin3Bu+8EEr_v_GaRGNPzOjfcnfzsV2Av2cgnWgO@mail.gmail.com>
	<AANLkTi=OC_jx0+nzK405VZURLDJPD25R4ZjyJ3RXJ1nk@mail.gmail.com>
	<AANLkTi=Sm7CjoayiE8nHmL0Eehqknj1jdpbx0YBQJQGH@mail.gmail.com>
	<AANLkTinr0RdgBs2UiS3=b3i+k6uCwkMihDy_FbAOFPc_@mail.gmail.com>
	<AANLkTi=+aObW3kVBZq5ox4b4E1uFTSgtXUL72tHLOzPi@mail.gmail.com>
	<AANLkTikVDKS40dVsf8x=+5=qnW59LqJbSv1H=vtROFaM@mail.gmail.com>
	<AANLkTi=gJsVKCJifVaBL_H6ZnPHTD__aQPhPeVfdx-ju@mail.gmail.com>
	<AANLkTimeuCHf_7Y1TErPwrJbdGZ3DoQmeqQbJ1_3YNpT@mail.gmail.com>
	<AANLkTikDxzntMFRpFhfiqJ4pzvjsTQR4w_FfyLfT5Y7z@mail.gmail.com>
	<AANLkTimVrhpfX+0D2mFPwGHK1EDsALWjFGp8hh7=ecqt@mail.gmail.com>
	<AANLkTi=oqw7fX-kvwqtxUXEWmzDu4C-dOjPrhpsCiwQc@mail.gmail.com>
	<AANLkTikWUfpFpQjKQ_EiGQuUGqa33VwU2K5vYVRvFGFO@mail.gmail.com>
	<AANLkTi=z3yysHW7Y7YaB92H6CDe0GFStMdPjHyUBmSvG@mail.gmail.com>
	<AANLkTinDmkxEeO2px7NsOSO5DkTiq+=1xTY7togH2-hJ@mail.gmail.com>
Message-ID: <AANLkTinFYHCsPUSmtZ_un=yuJ68TSgh8imxvPwz9iy1B@mail.gmail.com>

hi :-)

i will try it with other phones. if i'll get it wo work i'm going back
to the snom phones and try to get a fix (maybe fw 7.1.35 works for
me).

 it seems like chris had nearly the same problem and i also don't know
if presence_id can  help here, but on the other hand ... you are the
main developer :-)

i'll try it and i'll tell you if it works or not. and i will write an
article inn the wiki :-)

thanks,
 the guy who started the thread ;-)



2011/1/31 Anthony Minessale <anthony.minessale at gmail.com>:
> So sounds like the guy who started the thread should be able to get
> what he needs if we can find out why it's not working on snom then.
>
>
> On Mon, Jan 31, 2011 at 4:42 PM, Chris Burns <chris at cloudtel.com> wrote:
>> For calls, presence_id works perfectly and that is what I use. In the case
>> of my client, they were using BLFs to show additional information like which
>> desks have spanish or french speaking agents (because the agents switch
>> around desks daily). The call is just to an extension to add/remove the
>> phone from a fifo queue, which then has to trigger an auxilary BLF on/off.
>> The call to the extension will be over and the phone's own BLF will unlight,
>> but the auxilary BLF should remain lit ... showing them as a proud member of
>> the spanish speaking fifo queue to the managers who rely on blinking red
>> lights to do so. The end result of my ESL app is that nothing can change my
>> auxilary BLFs except my dialplan, and any phones that reboot or get plugged
>> in new get lit up properly.
>>
>> Was there a way I could have used presence_id and skipped using mod_dptools?
>> I didnt see how to do it.
>>
>>
>> On Mon, Jan 31, 2011 at 5:16 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>>
>>> bottom line is, if you use the presence_id how it was designed, it
>>> will report any presence you want throughout the call.
>>> If you reboot the phone, it will persist. ?It's just a matter of doing
>>> it all properly.
>>>
>>>
>>> On Mon, Jan 31, 2011 at 3:20 PM, Chris Burns <chris at cloudtel.com> wrote:
>>> > Not sure the polycoms do it exactly how he would like. Rebooting phones
>>> > can
>>> > trip up many versions of soundpoint firmware. In our case we had a big
>>> > opportunity with a client who was stuck on such a version of soundpoint,
>>> > with an integrated web app on their 650s. Patrick, if this code works
>>> > for
>>> > you dont let it stop you from applying pressure on your vendor to make
>>> > your
>>> > phones work the way they should. We all have to put the pressure on our
>>> > vendors when they make mistakes ... after all these folks are charging
>>> > money, as opposed to FreeSWITCH.
>>> >
>>> > Basically this program will store and repeat - on interval - any unique
>>> > PRESENCE_IN events it hears from the 'presence' app or api
>>> > (mod_dptools). It
>>> > will also override events from outside the dialplan with stored events
>>> > (eg.
>>> > from a misbehaving SIP phone). You need to build the php module for ESL
>>> > to
>>> > run the code
>>> > http://wiki.freeswitch.org/wiki/Event_Socket_Library#Installation
>>> >
>>> > WARNING: this code is just an example pulled from a larger presence
>>> > application. I tested it in my lab with polycoms, but I cant guarantee
>>> > your
>>> > safety, molten lava, bat country, etc. This may not be a very smart way
>>> > to
>>> > solve the problem, but this is what I came up with when backed into a
>>> > corner
>>> > and you are welcome to it ...
>>> >
>>> > http://pastebin.freeswitch.org/15193
>>> >
>>> >
>>> > On Mon, Jan 31, 2011 at 3:04 PM, Anthony Minessale
>>> > <anthony.minessale at gmail.com> wrote:
>>> >>
>>> >> You are heading the wrong way down the ladder but its worth a try.
>>> >> Grandstream is notorious for not following specs =D
>>> >>
>>> >> I have engaged SNOM to again try to sync up for interop. ?Stay tuned.
>>> >> I would really prefer to have them working.
>>> >>
>>> >> Meanwhile try firmware 7.1.35 on your snom as that was before things
>>> >> started to break.
>>> >>
>>> >> Also if you have polycom or linksys they have been known to work
>>> >> better with blf.
>>> >>
>>> >>
>>> >>
>>> >> On Mon, Jan 31, 2011 at 1:19 PM, Patrick Plattes
>>> >> <patrick.plattes at niemann-frey.info> wrote:
>>> >> > Hi :-)
>>> >> >
>>> >> > I think SNOM has a problem with the quality of the firmware. maybe
>>> >> > they should glp it, so we can fix it ;-)
>>> >> >
>>> >> > Tomorrow I will test it with an Grandstream GPX-2000 maybe it works
>>> >> > with this device. If it does I will also ask SNOM for a fix.
>>> >> >
>>> >> > Thanks,
>>> >> > ?Patrick
>>> >> >
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at 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
>



-- 
Patrick Plattes
IT - Projektleiter

Niemann + Frey GmbH
Adolf-Dembach-Str. 24
47829 Krefeld

Tel.: +49/2151 - 5554-263
Fax : +49/2151 - 5554-123
patrick.plattes at niemann-frey.info

Gesch?ftsf?hrer: Gerd Frey
Sitz und Registergericht: Krefeld HRB 10851

www.niemann-frey.de


From anthony.minessale at gmail.com  Tue Feb  1 02:31:06 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 31 Jan 2011 17:31:06 -0600
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTinFYHCsPUSmtZ_un=yuJ68TSgh8imxvPwz9iy1B@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
	<AANLkTi=in3CLd-n0mT3SRKYA0kxLMGVWk9KeBx3_jhY6@mail.gmail.com>
	<AANLkTin3Bu+8EEr_v_GaRGNPzOjfcnfzsV2Av2cgnWgO@mail.gmail.com>
	<AANLkTi=OC_jx0+nzK405VZURLDJPD25R4ZjyJ3RXJ1nk@mail.gmail.com>
	<AANLkTi=Sm7CjoayiE8nHmL0Eehqknj1jdpbx0YBQJQGH@mail.gmail.com>
	<AANLkTinr0RdgBs2UiS3=b3i+k6uCwkMihDy_FbAOFPc_@mail.gmail.com>
	<AANLkTi=+aObW3kVBZq5ox4b4E1uFTSgtXUL72tHLOzPi@mail.gmail.com>
	<AANLkTikVDKS40dVsf8x=+5=qnW59LqJbSv1H=vtROFaM@mail.gmail.com>
	<AANLkTi=gJsVKCJifVaBL_H6ZnPHTD__aQPhPeVfdx-ju@mail.gmail.com>
	<AANLkTimeuCHf_7Y1TErPwrJbdGZ3DoQmeqQbJ1_3YNpT@mail.gmail.com>
	<AANLkTikDxzntMFRpFhfiqJ4pzvjsTQR4w_FfyLfT5Y7z@mail.gmail.com>
	<AANLkTimVrhpfX+0D2mFPwGHK1EDsALWjFGp8hh7=ecqt@mail.gmail.com>
	<AANLkTi=oqw7fX-kvwqtxUXEWmzDu4C-dOjPrhpsCiwQc@mail.gmail.com>
	<AANLkTikWUfpFpQjKQ_EiGQuUGqa33VwU2K5vYVRvFGFO@mail.gmail.com>
	<AANLkTi=z3yysHW7Y7YaB92H6CDe0GFStMdPjHyUBmSvG@mail.gmail.com>
	<AANLkTinDmkxEeO2px7NsOSO5DkTiq+=1xTY7togH2-hJ@mail.gmail.com>
	<AANLkTinFYHCsPUSmtZ_un=yuJ68TSgh8imxvPwz9iy1B@mail.gmail.com>
Message-ID: <AANLkTinZ2P_b3-mtOX0J3HtsjfbE823cQ_j-0oVNd2-Q@mail.gmail.com>

feel free to stop by IRC irc.freenode.net as well we are there all day
every day.


On Mon, Jan 31, 2011 at 5:25 PM, Patrick Plattes
<patrick.plattes at niemann-frey.info> wrote:
> hi :-)
>
> i will try it with other phones. if i'll get it wo work i'm going back
> to the snom phones and try to get a fix (maybe fw 7.1.35 works for
> me).
>
> ?it seems like chris had nearly the same problem and i also don't know
> if presence_id can ?help here, but on the other hand ... you are the
> main developer :-)
>
> i'll try it and i'll tell you if it works or not. and i will write an
> article inn the wiki :-)
>
> thanks,
> ?the guy who started the thread ;-)
>
>
>
> 2011/1/31 Anthony Minessale <anthony.minessale at gmail.com>:
>> So sounds like the guy who started the thread should be able to get
>> what he needs if we can find out why it's not working on snom then.
>>
>>
>> On Mon, Jan 31, 2011 at 4:42 PM, Chris Burns <chris at cloudtel.com> wrote:
>>> For calls, presence_id works perfectly and that is what I use. In the case
>>> of my client, they were using BLFs to show additional information like which
>>> desks have spanish or french speaking agents (because the agents switch
>>> around desks daily). The call is just to an extension to add/remove the
>>> phone from a fifo queue, which then has to trigger an auxilary BLF on/off.
>>> The call to the extension will be over and the phone's own BLF will unlight,
>>> but the auxilary BLF should remain lit ... showing them as a proud member of
>>> the spanish speaking fifo queue to the managers who rely on blinking red
>>> lights to do so. The end result of my ESL app is that nothing can change my
>>> auxilary BLFs except my dialplan, and any phones that reboot or get plugged
>>> in new get lit up properly.
>>>
>>> Was there a way I could have used presence_id and skipped using mod_dptools?
>>> I didnt see how to do it.
>>>
>>>
>>> On Mon, Jan 31, 2011 at 5:16 PM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>>>
>>>> bottom line is, if you use the presence_id how it was designed, it
>>>> will report any presence you want throughout the call.
>>>> If you reboot the phone, it will persist. ?It's just a matter of doing
>>>> it all properly.
>>>>
>>>>
>>>> On Mon, Jan 31, 2011 at 3:20 PM, Chris Burns <chris at cloudtel.com> wrote:
>>>> > Not sure the polycoms do it exactly how he would like. Rebooting phones
>>>> > can
>>>> > trip up many versions of soundpoint firmware. In our case we had a big
>>>> > opportunity with a client who was stuck on such a version of soundpoint,
>>>> > with an integrated web app on their 650s. Patrick, if this code works
>>>> > for
>>>> > you dont let it stop you from applying pressure on your vendor to make
>>>> > your
>>>> > phones work the way they should. We all have to put the pressure on our
>>>> > vendors when they make mistakes ... after all these folks are charging
>>>> > money, as opposed to FreeSWITCH.
>>>> >
>>>> > Basically this program will store and repeat - on interval - any unique
>>>> > PRESENCE_IN events it hears from the 'presence' app or api
>>>> > (mod_dptools). It
>>>> > will also override events from outside the dialplan with stored events
>>>> > (eg.
>>>> > from a misbehaving SIP phone). You need to build the php module for ESL
>>>> > to
>>>> > run the code
>>>> > http://wiki.freeswitch.org/wiki/Event_Socket_Library#Installation
>>>> >
>>>> > WARNING: this code is just an example pulled from a larger presence
>>>> > application. I tested it in my lab with polycoms, but I cant guarantee
>>>> > your
>>>> > safety, molten lava, bat country, etc. This may not be a very smart way
>>>> > to
>>>> > solve the problem, but this is what I came up with when backed into a
>>>> > corner
>>>> > and you are welcome to it ...
>>>> >
>>>> > http://pastebin.freeswitch.org/15193
>>>> >
>>>> >
>>>> > On Mon, Jan 31, 2011 at 3:04 PM, Anthony Minessale
>>>> > <anthony.minessale at gmail.com> wrote:
>>>> >>
>>>> >> You are heading the wrong way down the ladder but its worth a try.
>>>> >> Grandstream is notorious for not following specs =D
>>>> >>
>>>> >> I have engaged SNOM to again try to sync up for interop. ?Stay tuned.
>>>> >> I would really prefer to have them working.
>>>> >>
>>>> >> Meanwhile try firmware 7.1.35 on your snom as that was before things
>>>> >> started to break.
>>>> >>
>>>> >> Also if you have polycom or linksys they have been known to work
>>>> >> better with blf.
>>>> >>
>>>> >>
>>>> >>
>>>> >> On Mon, Jan 31, 2011 at 1:19 PM, Patrick Plattes
>>>> >> <patrick.plattes at niemann-frey.info> wrote:
>>>> >> > Hi :-)
>>>> >> >
>>>> >> > I think SNOM has a problem with the quality of the firmware. maybe
>>>> >> > they should glp it, so we can fix it ;-)
>>>> >> >
>>>> >> > Tomorrow I will test it with an Grandstream GPX-2000 maybe it works
>>>> >> > with this device. If it does I will also ask SNOM for a fix.
>>>> >> >
>>>> >> > Thanks,
>>>> >> > ?Patrick
>>>> >> >
>>>> >> > _______________________________________________
>>>> >> > FreeSWITCH-users mailing list
>>>> >> > FreeSWITCH-users at 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
>>
>
>
>
> --
> Patrick Plattes
> IT - Projektleiter
>
> Niemann + Frey GmbH
> Adolf-Dembach-Str. 24
> 47829 Krefeld
>
> Tel.: +49/2151 - 5554-263
> Fax : +49/2151 - 5554-123
> patrick.plattes at niemann-frey.info
>
> Gesch?ftsf?hrer: Gerd Frey
> Sitz und Registergericht: Krefeld HRB 10851
>
> www.niemann-frey.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 marcin321 at hotmail.com  Tue Feb  1 02:43:19 2011
From: marcin321 at hotmail.com (Marcin Wojtowicz)
Date: Mon, 31 Jan 2011 18:43:19 -0500
Subject: [Freeswitch-users] Outbound only calls don't connect when
 bypass_media is true.
In-Reply-To: <AANLkTikSs4sn-+x5G9pRpszogW7Hwng_cLJop0zrAueu@mail.gmail.com>
References: <COL118-W52E5226FC3337085AB6AFAE5E00@phx.gbl>,
	<COL118-W38F26048D441DAA2777942E5E30@phx.gbl>,
	<AANLkTinbiU4zNc-ZM+u+iST7E2aZ3zisgh5Lu2FTuBLt@mail.gmail.com>,
	<COL118-W196844C55E71088CAD1FC1E5E20@phx.gbl>,
	<AANLkTi=t0ixs6L3SgUMQzEPtgvFowbFYXCVOtmvzLTEW@mail.gmail.com>,
	<COL118-W34303A74089852D47F0E63E5E20@phx.gbl>,
	<E826C94A3C6BE44198FAF4D66B5AB4D80FEC3E5B24@VA3DIAXVS351.RED001.local>,
	<COL118-W33A8A74086A6BBE2E6B1B7E5E20@phx.gbl>,
	<AANLkTikSs4sn-+x5G9pRpszogW7Hwng_cLJop0zrAueu@mail.gmail.com>
Message-ID: <COL118-W45E72C891B1AB696AFDA8FE5E20@phx.gbl>


Here are the build errors, in order of appearance:
------ Build started: Project: libpcre, Configuration: Release Win32 ------
  pcre_chartables.c
c1 : fatal error C1083: Cannot open source file: 'pcre_chartables.c': No such file or directory
(other files here compile ok)

------ Build started: Project: FreeSwitchCoreLib, Configuration: Release Win32 ------
  Generating switch_version.h
  (stuff here goes ok)
  Generating Code...
LINK : fatal error LNK1181: cannot open input file 'E:\downloads\freeswitch-1.0.7\libs\win32\pcre\Win32\Release\libpcre.lib'

------ Build started: Project: mod_spidermonkey, Configuration: Release Win32 ------
  mod_spidermonkey.c
LINK : fatal error LNK1181: cannot open input file 'E:\downloads\freeswitch-1.0.7\Win32\Release\FreeSwitchCore.lib'

FreeSwitchCore link error repeats many times for other projects, too many to list. It doesn't build.

------ Build started: Project: mod_managed, Configuration: Release_CLR Win32 ------
  freeswitch_managed.cpp
freeswitch_managed.cpp : fatal error C1192: #using failed on 'E:\downloads\freeswitch-1.0.7\Win32\Debug\mod\FreeSWITCH.Managed.dll'
          'The system cannot find the path specified.'
  freeswitch_wrap.2010.cxx
freeswitch_wrap.2010.cxx : fatal error C1192: #using failed on 'E:\downloads\freeswitch-1.0.7\Win32\Debug\mod\FreeSWITCH.Managed.dll'
          'The system cannot find the path specified.'
  mod_managed.cpp
mod_managed.cpp : fatal error C1192: #using failed on 'E:\downloads\freeswitch-1.0.7\Win32\Debug\mod\FreeSWITCH.Managed.dll'
          'The system cannot find the path specified.'
  Generating Code...


The code I downloaded was from latest.freeswitch.org, the tar.bz2 zip file. From the dependencies it looks like FreeSwitchCore depends on libpcre, so all this cascades from the missing pcre_chartables.c file. Can you send it to me?




> Date: Mon, 31 Jan 2011 14:00:10 -0600
> From: anthony.minessale at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Outbound only calls don't connect when bypass_media is true.
> 
>  a=fmtp:98 mode=30 is missing in the 200 ok from the phone.
> Also did you mention the revision you are on. I had indicated that the
> very latest code may have more tolerant ilbc codec code in it.
> http://latest.freeswitch.org
> 
> 
> On Mon, Jan 31, 2011 at 12:12 PM, Marcin Wojtowicz
> <marcin321 at hotmail.com> wrote:
> > SDP looks ok to me, but there is one warning about ptime in iLBC below. I
> > don't see how a wrong codec can be selected because I narrowed down my
> > external profile inbound/outbound to PCMU only and my internal is iLBC at 30i
> > only.
> >
> >
> > freeswitch at kuffel> recv 1206 bytes from udp/[74.63.41.218]:5060 at
> > 17:59:32.187500:
> >    ------------------------------------------------------------------------
> >    INVITE sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms SIP/2.0
> >    Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK30d7505b;rport
> >    From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64
> >    To: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
> >    Contact: <sip:MYPHONE#@74.63.41.218>
> >    Call-ID: 1298959c0099d50d177b6bae689fc028 at 74.63.41.218
> >    CSeq: 102 INVITE
> >    User-Agent: VoIPMS/SERAST
> >    Max-Forwards: 70
> >    Remote-Party-ID: "MYPHONE#"
> > <sip:MYPHONE#@74.63.41.218>;privacy=off;screen=no
> >    Date: Mon, 31 Jan 2011 17:59:17 GMT
> >    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> >    Supported: replaces
> >    Content-Type: application/sdp
> >    Content-Length: 515
> >
> >    v=0
> >    o=root 2831 2831 IN IP4 74.63.41.218
> >    s=session
> >    c=IN IP4 74.63.41.218
> >    t=0 0
> >    m=audio 16884 RTP/AVP 0 4 3 8 112 5 10 7 18 111 101
> >    a=rtpmap:0 PCMU/8000
> >    a=rtpmap:4 G723/8000
> >    a=fmtp:4 annexa=no
> >    a=rtpmap:3 GSM/8000
> >    a=rtpmap:8 PCMA/8000
> >    a=rtpmap:112 AAL2-G726-32/8000
> >    a=rtpmap:5 DVI4/8000
> >    a=rtpmap:10 L16/8000
> >    a=rtpmap:7 LPC/8000
> >    a=rtpmap:18 G729/8000
> >    a=fmtp:18 annexb=no
> >    a=rtpmap:111 G726-32/8000
> >    a=rtpmap:101 telephone-event/8000
> >    a=fmtp:101 0-16
> >    a=silenceSupp:off - - - -
> >    a=ptime:20
> >    a=sendrecv
> >    ------------------------------------------------------------------------
> > send 396 bytes to udp/[74.63.41.218]:5060 at 17:59:32.187500:
> >    ------------------------------------------------------------------------
> >    SIP/2.0 100 Trying
> >    Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK30d7505b;rport=5060
> >    From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64
> >    To: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
> >    Call-ID: 1298959c0099d50d177b6bae689fc028 at 74.63.41.218
> >    CSeq: 102 INVITE
> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> > -0600
> >    Content-Length: 0
> >
> >    ------------------------------------------------------------------------
> > 2011-01-31 12:59:32.187500 [NOTICE] switch_channel.c:808 New Channel
> > sofia/external/MYPHONE#@74.63.41.218 [f35f408a-f863-4784-a308-8b4fb3284b70]
> > 2011-01-31 12:59:32.187500 [INFO] mod_dialplan_xml.c:331 Processing MYPHONE#
> > <MYPHONE#>->121628 in context public
> > 2011-01-31 12:59:32.203125 [NOTICE] switch_ivr.c:1606 Transfer
> > sofia/external/MYPHONE#@74.63.41.218 to XML[1001 at default]
> > 2011-01-31 12:59:32.203125 [INFO] mod_dialplan_xml.c:331 Processing MYPHONE#
> > <MYPHONE#>->1001 in context default
> > 2011-01-31 12:59:32.234375 [NOTICE] switch_channel.c:808 New Channel
> > sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060
> > [7230b9e8-37a7-4fc6-9b52-25740a6f7ca4]
> >
> >
> > 2011-01-31 12:59:32.265625 [WARNING] sofia_glue.c:213 Codec iLBC payload 98
> > added to sdp wanting ptime 30 but it's already 20 (PCMU:0:20), disabling
> > ptime.
> >
> >
> > send 1315 bytes to tcp/[32.140.14.196]:46743 at 17:59:32.265625:
> >    ------------------------------------------------------------------------
> >    INVITE sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP SIP/2.0
> >    Via: SIP/2.0/TCP 69.125.20.15;rport;branch=z9hG4bK797eFQ6rgQKmQ
> >    Route: <sip:M9jdt73ig0oOJSbt6Uyy at 32.140.14.196:46743>;transport=TCP
> >    Max-Forwards: 68
> >    From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa
> >    To: <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP>
> >    Call-ID: b281a1fa-a806-122e-f799-c1188a708e17
> >    CSeq: 7912322 INVITE
> >    Contact: <sip:mod_sofia at 69.125.20.15:5060;transport=tcp>
> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> > -0600
> >    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: 234
> >    X-FS-Support: update_display
> >    Remote-Party-ID: "MYPHONE#"
> > <sip:MYPHONE#@192.168.1.100>;party=calling;screen=no;privacy=off
> >
> >    v=0
> >    o=FreeSWITCH 1296474878 1296474879 IN IP4 69.125.20.15
> >    s=FreeSWITCH
> >    c=IN IP4 69.125.20.15
> >    t=0 0
> >    m=audio 21894 RTP/AVP 0 98 101 13
> >    a=rtpmap:98 iLBC/8000
> >    a=fmtp:98 mode=30
> >    a=rtpmap:101 telephone-event/8000
> >    a=fmtp:101 0-16
> >    ------------------------------------------------------------------------
> > recv 318 bytes from tcp/[32.140.14.196]:46743 at 17:59:37.093750:
> >    ------------------------------------------------------------------------
> >    SIP/2.0 100 Trying
> >    Via: SIP/2.0/TCP
> > 69.125.20.15;branch=z9hG4bK797eFQ6rgQKmQ;rport=5060;received=69.125.20.15
> >    To: <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155>
> >    From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa
> >    Call-ID: b281a1fa-a806-122e-f799-c1188a708e17
> >    CSeq: 7912322 INVITE
> >    Content-Length: 0
> >
> >    ------------------------------------------------------------------------
> > recv 476 bytes from tcp/[32.140.14.196]:46743 at 17:59:42.296875:
> >    ------------------------------------------------------------------------
> >    SIP/2.0 180 Ringing
> >    Via: SIP/2.0/TCP
> > 69.125.20.15;branch=z9hG4bK797eFQ6rgQKmQ;rport=5060;received=69.125.20.15
> >    Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP>
> >    From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa
> >    To:
> > <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155>;tag=p4rl1jbfvmnbvfs1d5rktoj2
> >    Supported: 100rel
> >    Call-ID: b281a1fa-a806-122e-f799-c1188a708e17
> >    CSeq: 7912322 INVITE
> >    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE
> >    Content-Length: 0
> >
> >    ------------------------------------------------------------------------
> > 2011-01-31 12:59:42.296875 [INFO] sofia.c:729
> > sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060 Update Callee ID
> > to "Outbound Call" <M9jdt73ig0oOJSbt6Uyy>
> > 2011-01-31 12:59:42.296875 [NOTICE] sofia.c:4724 Ring-Ready
> > sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060!
> > 2011-01-31 12:59:42.312500 [INFO] switch_ivr_originate.c:1101 Sending early
> > media
> > 2011-01-31 12:59:42.343750 [NOTICE] mod_sofia.c:2252 Pre-Answer
> > sofia/external/MYPHONE#@74.63.41.218!
> > send 1079 bytes to udp/[74.63.41.218]:5060 at 17:59:42.343750:
> >    ------------------------------------------------------------------------
> >    SIP/2.0 183 Session Progress
> >    Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK30d7505b;rport=5060
> >    From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64
> >    To:
> > <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>;tag=eK0X80BS0091S
> >    Call-ID: 1298959c0099d50d177b6bae689fc028 at 74.63.41.218
> >    CSeq: 102 INVITE
> >    Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp>
> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> > -0600
> >    Accept: application/sdp
> >    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: 247
> >    Remote-Party-ID: "121628"
> > <sip:121628 at 192.168.1.100>;party=calling;privacy=off;screen=no
> >
> >    v=0
> >    o=FreeSWITCH 1296476876 1296476877 IN IP4 69.125.20.15
> >    s=FreeSWITCH
> >    c=IN IP4 69.125.20.15
> >    t=0 0
> >    m=audio 19906 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
> >    ------------------------------------------------------------------------
> > recv 772 bytes from tcp/[32.140.14.196]:46743 at 17:59:43.812500:
> >    ------------------------------------------------------------------------
> >    SIP/2.0 200 OK
> >    Via: SIP/2.0/TCP
> > 69.125.20.15;branch=z9hG4bK797eFQ6rgQKmQ;rport=5060;received=69.125.20.15
> >    To:
> > <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155>;tag=p4rl1jbfvmnbvfs1d5rktoj2
> >    Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP>
> >    From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa
> >    Supported: timer
> >    Call-ID: b281a1fa-a806-122e-f799-c1188a708e17
> >    CSeq: 7912322 INVITE
> >    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE
> >    Content-Type: application/sdp
> >    Content-Length: 269
> >
> >    v=0
> >    o=M9jdt73ig0oOJSbt6Uyy 63464734759229750 63464734759229750 IN IP4
> > 10.208.245.155
> >    s=-
> >    c=IN IP4 10.208.245.155
> >    t=0 0
> >    m=audio 49152 RTP/AVP 98 101
> >    a=sendrecv
> >    a=rtpmap:98 iLBC/8000
> >    a=ptime:30
> >    a=maxptime:180
> >    a=rtpmap:101 telephone-event/8000
> >    a=fmtp:101 0-15
> >    ------------------------------------------------------------------------
> > send 464 bytes to tcp/[32.140.14.196]:46743 at 17:59:43.828125:
> >    ------------------------------------------------------------------------
> >    ACK sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP SIP/2.0
> >    Via: SIP/2.0/TCP 69.125.20.15;rport;branch=z9hG4bK8j17gjQvD096j
> >    Max-Forwards: 70
> >    From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa
> >    To:
> > <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP>;tag=p4rl1jbfvmnbvfs1d5rktoj2
> >    Call-ID: b281a1fa-a806-122e-f799-c1188a708e17
> >    CSeq: 7912322 ACK
> >    Contact: <sip:mod_sofia at 69.125.20.15:5060;transport=tcp>
> >    Content-Length: 0
> >
> >    ------------------------------------------------------------------------
> > 2011-01-31 12:59:43.828125 [NOTICE] sofia.c:5230 Channel
> > [sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060] has been
> > answered
> > send 1061 bytes to udp/[74.63.41.218]:5060 at 17:59:43.843750:
> >    ------------------------------------------------------------------------
> >    SIP/2.0 200 OK
> >    Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK30d7505b;rport=5060
> >    From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64
> >    To:
> > <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>;tag=eK0X80BS0091S
> >    Call-ID: 1298959c0099d50d177b6bae689fc028 at 74.63.41.218
> >    CSeq: 102 INVITE
> >    Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp>
> > 2011-01-31 12:59:43.843750 [NOTICE] switch_ivr_originate.c:3328 Channel
> > [sofia/external/MYPHONE#@74.63.41.218] has been answered
> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> > -0600
> >    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: 247
> >    Remote-Party-ID: "Outbound Call"
> > <sip:M9jdt73ig0oOJSbt6Uyy at 192.168.1.100>;party=calling;privacy=off;screen=no
> >
> >    v=0
> >    o=FreeSWITCH 1296476876 1296476877 IN IP4 69.125.20.15
> >    s=FreeSWITCH
> >    c=IN IP4 69.125.20.15
> >    t=0 0
> >    m=audio 19906 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
> >    ------------------------------------------------------------------------
> > recv 533 bytes from udp/[74.63.41.218]:5060 at 17:59:43.859375:
> >    ------------------------------------------------------------------------
> >    ACK sip:gw+voip.ms at 69.125.20.15:5080;transport=udp SIP/2.0
> >    Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK0cbbc6ef;rport
> >    From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64
> >    To:
> > <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>;tag=eK0X80BS0091S
> >    Contact: <sip:MYPHONE#@74.63.41.218>
> >    Call-ID: 1298959c0099d50d177b6bae689fc028 at 74.63.41.218
> >    CSeq: 102 ACK
> >    User-Agent: VoIPMS/SERAST
> >    Max-Forwards: 70
> >    Remote-Party-ID: "MYPHONE#"
> > <sip:MYPHONE#@74.63.41.218>;privacy=off;screen=no
> >    Content-Length: 0
> >
> > ________________________________
> > From: robert.hadley at teotech.com
> > To: freeswitch-users at lists.freeswitch.org
> > Date: Mon, 31 Jan 2011 09:33:48 -0800
> > Subject: Re: [Freeswitch-users] Outbound only calls don't connect when
> > bypass_media is true.
> >
> >
> >
> > Check the codecs in the SDP or try manual hardcoding the codecs presented
> > for both legs, we had a squeal problem going to a softphone that turned out
> > to be the BV32 codec was being selected instead of SPEEX16.
> >
> >
> >
> > Robert
> >
> >
> >
> > From: Marcin Wojtowicz [mailto:marcin321 at hotmail.com]
> > Sent: Monday, January 31, 2011 9:17 AM
> > To: freeswitch
> > Subject: Re: [Freeswitch-users] Outbound only calls don't connect when
> > bypass_media is true.
> >
> >
> >
> > Yes, I had it set up to iLBC at 30i. It's not my cell phone (configured to
> > ilbc, ptime=30 and mode=30), because when I call my freeswitch voicemail
> > number, the sound is fine. I suspect it is something on the voip.ms <->
> > freeswitch leg because I created a sample ringback (8khz, mono, 16bit) wave
> > file and directed my dialplan to it, but when I call from my home number to
> > my cell, instead of hearing the ringer, I get choppy squeal.
> >
> > ?
> >
> >> Date: Mon, 31 Jan 2011 10:40:10 -0600
> >> From: anthony.minessale at gmail.com
> >> To: freeswitch-users at lists.freeswitch.org
> >> Subject: Re: [Freeswitch-users] Outbound only calls don't connect when
> >> bypass_media is true.
> >>
> >> Many things have problems doing iLBC right.
> >> I recommend you define it in your configs as iLBC at 30i or it will try
> >> using the 20ms version which is not compatible with many other
> >> platforms. Also make sure you are on the latest version of FS since
> >> we have tweaked iLBC behavior to compensate for problems like this.
> >>
> >>
> >>
> >> On Sun, Jan 30, 2011 at 10:40 PM, Marcin Wojtowicz
> >> <marcin321 at hotmail.com> wrote:
> >> > OK, so I gave up on bypass media, but now I have another problem. This
> >> > time
> >> > I set up freeswitch to communicate with voip.ms using PCMU codec
> >> > (configured
> >> > in my external profile), and use iLBC on my phone (codec configured in
> >> > my
> >> > internal profile, where the phone registers). When I call my mobile it
> >> > rings, but when I pick up all I hear is a high pitched squeal. Am I
> >> > missing
> >> > something here?
> >> >
> >> >> Date: Sun, 30 Jan 2011 16:34:09 -0600
> >> >> From: anthony.minessale at gmail.com
> >> >> To: freeswitch-users at lists.freeswitch.org
> >> >> Subject: Re: [Freeswitch-users] Outbound only calls don't connect when
> >> >> bypass_media is true.
> >> >>
> >> >> Just do not use bypass media.
> >> >> That is all you can do in that situation.
> >> >>
> >> >>
> >> >> On Sun, Jan 30, 2011 at 3:44 PM, Marcin Wojtowicz
> >> >> <marcin321 at hotmail.com>
> >> >> wrote:
> >> >> > I just want to add that I enabled STUN on my cell so now the SDP
> >> >> > message
> >> >> > in
> >> >> > the INVITE to voip.ms contains the public IP of my phone, but it
> >> >> > still
> >> >> > doesn't work.
> >> >> >
> >> >> > ________________________________
> >> >> > From: marcin321 at hotmail.com
> >> >> > To: freeswitch-users at lists.freeswitch.org
> >> >> > Date: Fri, 28 Jan 2011 19:54:19 -0500
> >> >> > Subject: [Freeswitch-users] Outbound only calls don't connect when
> >> >> > bypass_media is true.
> >> >> >
> >> >> > Hello,
> >> >> >
> >> >> > I'm a new user of freeswitch, so please bear with me. I have the
> >> >> > following setup:
> >> >> > voip.ms <- SIP over UDP -> my desktop running freeswitch <- SIP over
> >> >> > TCP
> >> >> > ->
> >> >> > my nokia cellphone on AT&T wireless. This setup is intended to
> >> >> > conserve
> >> >> > the
> >> >> > battery usage.
> >> >> > I've managed to make everything work well when I'm calling in over
> >> >> > any
> >> >> > phone
> >> >> > to my cell phone, and freeswitch is enabled to work in bypass_media =
> >> >> > true,
> >> >> > even though by cell is behind NAT on at&t's network. Things break
> >> >> > when I
> >> >> > pick up my cell and try to call my home phone (or any phone for that
> >> >> > matter). This is the relevant snippet from my dialplan:
> >> >> > <extension name="outbound">
> >> >> >   <condition field="destination_number"
> >> >> > expression="^1?([2-9]\d{2}[2-9]\d{6})$">
> >> >> >     <!--<action application="set" data="bypass_media=true"/>-->
> >> >> >     <action application="bridge" data="sofia/gateway/voip.ms/1$1"/>
> >> >> >   </condition>
> >> >> > </extension>
> >> >> >
> >> >> > Like shown above, my call will go to my home phone. When I uncomment
> >> >> > the
> >> >> > bypass_media tag, my call will not connect. Here are the siptraces
> >> >> > I replaced my real home phone number in the with "MYPHONE".
> >> >> >
> >> >> > recv 940 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.406250:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> >> >> >    Via: SIP/2.0/TCP
> >> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport
> >> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >> >    To: <sip:MYPHONE at 192.168.1.100>
> >> >> >    Contact:
> >> >> > <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>
> >> >> >    Supported: 100rel,timer
> >> >> >    CSeq: 5244503 INVITE
> >> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >> >    Allow:
> >> >> > UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE
> >> >> >    User-Agent: S60 RM-624 v 20.2.042 (en)
> >> >> >    Expires: 120
> >> >> >    Privacy: None
> >> >> >    Session-Expires: 1800
> >> >> >    Max-Forwards: 70
> >> >> >    Content-Type: application/sdp
> >> >> >    Accept-Language: en
> >> >> >    Content-Length: 292
> >> >> >
> >> >> >    v=0
> >> >> >    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> >> >> >    s=-
> >> >> >    c=IN IP4 10.153.174.6
> >> >> >    t=0 0
> >> >> >    m=audio 49152 RTP/AVP 18 97 98
> >> >> >    a=sendrecv
> >> >> >    a=rtpmap:18 G729/8000
> >> >> >    a=ptime:20
> >> >> >    a=maxptime:40
> >> >> >    a=fmtp:18 annexb=no
> >> >> >    a=rtpmap:97 iLBC/8000
> >> >> >    a=rtpmap:98 telephone-event/8000
> >> >> >    a=fmtp:98 0-15
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    SIP/2.0 100 Trying
> >> >> >    Via: SIP/2.0/TCP
> >> >> >
> >> >> >
> >> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180
> >> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >> >    To: <sip:MYPHONE at 192.168.1.100>
> >> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >> >    CSeq: 5244503 INVITE
> >> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> >> > 18-04-05
> >> >> > -0600
> >> >> >    Content-Length: 0
> >> >> >
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > send 875 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    SIP/2.0 407 Proxy Authentication Required
> >> >> >    Via: SIP/2.0/TCP
> >> >> >
> >> >> >
> >> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180
> >> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj2011-01-28
> >> >> > 16:15:58.406250 [WARNING] sofia_reg.c:1241 SIP auth challenge
> >> >> > (INVITE)
> >> >> > on
> >> >> > sofia profile 'internal' for [MYPHONE at 192.168.1.100] from ip
> >> >> > 32.136.78.180
> >> >> >
> >> >> >    To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m
> >> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >> >    CSeq: 5244503 INVITE
> >> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> >> > 18-04-05
> >> >> > -0600
> >> >> >    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
> >> >> >    Proxy-Authenticate: Digest realm="192.168.1.100",
> >> >> > nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28", algorithm=MD5,
> >> >> > qop="auth"
> >> >> >    Content-Length: 0
> >> >> >
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > recv 368 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.765625:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> >> >> >    Via: SIP/2.0/TCP
> >> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport
> >> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >> >    To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m
> >> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >> >    CSeq: 5244503 ACK
> >> >> >    Supported: sec-agree
> >> >> >    Max-Forwards: 70
> >> >> >    Content-Length: 0
> >> >> >
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > recv 1222 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.406250:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> >> >> >    Via: SIP/2.0/TCP
> >> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport
> >> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >> >    To: <sip:MYPHONE at 192.168.1.100>
> >> >> >    Contact:
> >> >> > <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>
> >> >> >    Supported: 100rel,timer
> >> >> >    CSeq: 5244504 INVITE
> >> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >> >    Allow:
> >> >> > UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE
> >> >> >    User-Agent: S60 RM-624 v 20.2.042 (en)
> >> >> >    Expires: 120
> >> >> >    Privacy: None
> >> >> >    Session-Expires: 1800
> >> >> >    Max-Forwards: 70
> >> >> >    Proxy-Authorization: Digest
> >> >> >
> >> >> >
> >> >> > qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"
> >> >> >    Content-Type: application/sdp
> >> >> >    Accept-Language: en
> >> >> >    Content-Length: 292
> >> >> >
> >> >> >    v=0
> >> >> >    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> >> >> >    s=-
> >> >> >    c=IN IP4 10.153.174.6
> >> >> >    t=0 0
> >> >> >    m=audio 49152 RTP/AVP 18 97 98
> >> >> >    a=sendrecv
> >> >> >    a=rtpmap:18 G729/8000
> >> >> >    a=ptime:20
> >> >> >    a=maxptime:40
> >> >> >    a=fmtp:18 annexb=no
> >> >> >    a=rtpmap:97 iLBC/8000
> >> >> >    a=rtpmap:98 telephone-event/8000
> >> >> >    a=fmtp:98 0-15
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.406250:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    SIP/2.0 100 Trying
> >> >> >    Via: SIP/2.0/TCP
> >> >> >
> >> >> >
> >> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180
> >> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >> >    To: <sip:MYPHONE at 192.168.1.100>
> >> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >> >    CSeq: 5244504 INVITE
> >> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> >> > 18-04-05
> >> >> > -0600
> >> >> >    Content-Length: 0
> >> >> >
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > 2011-01-28 16:15:59.406250 [NOTICE] switch_channel.c:808 New Channel
> >> >> > sofia/internal/1001 at 192.168.1.100
> >> >> > [e5841001-04bd-4e16-9519-64ff2c7a8c2f]
> >> >> > 2011-01-28 16:15:59.421875 [INFO] mod_dialplan_xml.c:331 Processing
> >> >> > 1001
> >> >> > <1001>->MYPHONE in context default
> >> >> > 2011-01-28 16:15:59.437500 [NOTICE] switch_channel.c:808 New Channel
> >> >> > sofia/external/1MYPHONE [60940227-9ae0-4c0c-abdb-f3988852fae0]
> >> >> > send 1141 bytes to udp/[74.63.41.218]:5060 at 21:15:59.453125:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0
> >> >> >    Via: SIP/2.0/UDP
> >> >> > 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS
> >> >> >    Max-Forwards: 69
> >> >> >    From: "Extension 1001"
> >> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >> >    To: <sip:1MYPHONE at newyork.voip.ms>
> >> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >> >    CSeq: 7788615 INVITE
> >> >> >    Contact:
> >> >> > <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
> >> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> >> > 18-04-05
> >> >> > -0600
> >> >> >    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: 280
> >> >> >    X-FS-Support: update_display
> >> >> >    Remote-Party-ID: "Extension 1001"
> >> >> > <sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off
> >> >> >
> >> >> >    v=0
> >> >> >    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> >> >> >    s=-
> >> >> >    c=IN IP4 10.153.174.6
> >> >> >    t=0 0
> >> >> >    m=audio 49152 RTP/AVP 18 97 98
> >> >> >    a=rtpmap:18 G729/8000
> >> >> >    a=fmtp:18 annexb=no
> >> >> >    a=rtpmap:97 iLBC/8000
> >> >> >    a=rtpmap:98 telephone-event/8000
> >> >> >    a=fmtp:98 0-15
> >> >> >    a=ptime:20
> >> >> >    a=maxptime:40
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > recv 570 bytes from udp/[74.63.41.218]:5060 at 21:15:59.468750:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    SIP/2.0 407 Proxy Authentication Required
> >> >> >    Via: SIP/2.0/UDP
> >> >> >
> >> >> >
> >> >> > 69.125.20.15:5080;branch=z9hG4bKNp4ZFeKSD43tS;received=69.125.20.15;rport=5080
> >> >> >    From: "Extension 1001"
> >> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >> >    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843
> >> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >> >    CSeq: 7788615 INVITE
> >> >> >    User-Agent: VoIPMS/SERAST
> >> >> >    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> >> >> >    Supported: replaces
> >> >> >    Proxy-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms",
> >> >> > nonce="2d534dd6"
> >> >> >    Content-Length: 0
> >> >> >
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0
> >> >> >    Via: SIP/2.0/UDP
> >> >> > 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS
> >> >> >    Max-Forwards: 69
> >> >> >    From: "Extension 1001"
> >> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >> >    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843
> >> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >> >    CSeq: 7788615 ACK
> >> >> >    Content-Length: 0
> >> >> >
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > send 1344 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0
> >> >> >    Via: SIP/2.0/UDP
> >> >> > 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN
> >> >> >    Max-Forwards: 69
> >> >> >    From: "Extension 1001"
> >> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >> >    To: <sip:1MYPHONE at newyork.voip.ms>
> >> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >> >    CSeq: 7788616 INVITE
> >> >> >    Contact:
> >> >> > <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
> >> >> >    Expires: 300
> >> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> >> > 18-04-05
> >> >> > -0600
> >> >> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> >> > REGISTER, REFER, NOTIFY
> >> >> >    Supported: timer, precondition, path, replaces
> >> >> >    Allow-Events: talk, hold, refer
> >> >> >    Proxy-Authorization: Digest username="121628",
> >> >> > realm="newyork.voip.ms",
> >> >> > nonce="2d534dd6", algorithm=MD5, uri="sip:1MYPHONE at newyork.voip.ms",
> >> >> > response="16f3301efae13df926da7550f709d28a"
> >> >> >    Content-Type: application/sdp
> >> >> >    Content-Disposition: session
> >> >> >    Content-Length: 280
> >> >> >    X-FS-Support: update_display
> >> >> >    Remote-Party-ID: "Extension 1001"
> >> >> > <sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off
> >> >> >
> >> >> >    v=0
> >> >> >    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> >> >> >    s=-
> >> >> >    c=IN IP4 10.153.174.6
> >> >> >    t=0 0
> >> >> >    m=audio 49152 RTP/AVP 18 97 98
> >> >> >    a=rtpmap:18 G729/8000
> >> >> >    a=fmtp:18 annexb=no
> >> >> >    a=rtpmap:97 iLBC/8000
> >> >> >    a=rtpmap:98 telephone-event/8000
> >> >> >    a=fmtp:98 0-15
> >> >> >    a=ptime:20
> >> >> >    a=maxptime:40
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > recv 488 bytes from udp/[74.63.41.218]:5060 at 21:15:59.484375:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    SIP/2.0 100 Trying
> >> >> >    Via: SIP/2.0/UDP
> >> >> >
> >> >> >
> >> >> > 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080
> >> >> >    From: "Extension 1001"
> >> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >> >    To: <sip:1MYPHONE at newyork.voip.ms>
> >> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >> >    CSeq: 7788616 INVITE
> >> >> >    User-Agent: VoIPMS/SERAST
> >> >> >    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> >> >> >    Supported: replaces
> >> >> >    Contact: <sip:1MYPHONE at 74.63.41.218>
> >> >> >    Content-Length: 0
> >> >> >
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > recv 516 bytes from udp/[74.63.41.218]:5060 at 21:15:59.562500:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    SIP/2.0 503 Service Unavailable
> >> >> >    Via: SIP/2.0/UDP
> >> >> >
> >> >> >
> >> >> > 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080
> >> >> >    From: "Extension 1001"
> >> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >> >    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9
> >> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >> >    CSeq: 7788616 INVITE
> >> >> >    User-Agent: VoIPMS/SERAST
> >> >> >    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> >> >> >    Supported: replaces
> >> >> >    Contact: <sip:1MYPHONE at 74.63.41.218>
> >> >> >    Content-Length: 0
> >> >> >
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.562500:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0
> >> >> >    Via: SIP/2.0/UDP
> >> >> > 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN
> >> >> >    Max-Forwards: 69
> >> >> >    From: "Extension 1001"
> >> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >> >    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9
> >> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >> >    CSeq: 7788616 ACK
> >> >> >    Content-Length: 0
> >> >> >
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > 2011-01-28 16:15:59.562500 [INFO] mod_dptools.c:2612 Originate
> >> >> > Failed.
> >> >> > Cause: NO_ANSWER
> >> >> > 2011-01-28 16:15:59.562500 [NOTICE] sofia.c:5286 Hangup
> >> >> > sofia/external/1MYPHONE [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
> >> >> > 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:189
> >> >> > sofia/internal/1001 at 192.168.1.100 has executed the last dialplan
> >> >> > instruction, hanging up.
> >> >> > 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:191
> >> >> > Hangup
> >> >> > sofia/internal/1001 at 192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING]
> >> >> > 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1306
> >> >> > Session 2
> >> >> > (sofia/external/1MYPHONE) Ended
> >> >> > 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1308 Close
> >> >> > Channel
> >> >> > sofia/external/1MYPHONE [CS_DESTROY]
> >> >> > send 887 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.593750:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    SIP/2.0 503 Service Unavailable
> >> >> >    Via: SIP/2.0/TCP
> >> >> >
> >> >> >
> >> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180
> >> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >> >    To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg
> >> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >> >    CSeq: 5244504 INVITE
> >> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> >> > 18-04-05
> >> >> > -0600
> >> >> >    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
> >> >> >    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
> >> >> > 2011-01-28 16:15:59.593750 [NOTICE] switch_core_session.c:1306
> >> >> > Session 1
> >> >> > (sofia/internal/1001 at 192.168.1.100) Ended
> >> >> >    Content-Length: 02011-01-28 16:15:59.593750 [NOTICE]
> >> >> > switch_core_session.c:1308 Close Channel
> >> >> > sofia/internal/1001 at 192.168.1.100
> >> >> > [CS_DESTROY]
> >> >> >
> >> >> >    Remote-Party-ID: "MYPHONE"
> >> >> > <sip:MYPHONE at 192.168.1.100>;party=calling;privacy=off;screen=no
> >> >> >
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> > recv 650 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.953125:
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >    ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> >> >> >    Via: SIP/2.0/TCP
> >> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport
> >> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >> >    To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg
> >> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >> >    CSeq: 5244504 ACK
> >> >> >    Supported: sec-agree
> >> >> >    Max-Forwards: 70
> >> >> >    Proxy-Authorization: Digest
> >> >> >
> >> >> >
> >> >> > qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"
> >> >> >    Content-Length: 0
> >> >> >
> >> >> >
> >> >> >
> >> >> > ------------------------------------------------------------------------
> >> >> >
> >> >> > 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
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at 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
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 Feb  1 02:54:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Jan 2011 15:54:38 -0800
Subject: [Freeswitch-users] Developer Dinner Coming Up - Help Feed The
	FreeSWITCH Team!
Message-ID: <AANLkTimZjT2SQ=Go10JwLrXJEmjQt+iNZQrUKVHPBgj=@mail.gmail.com>

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

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

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

-Michael Collins
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From msc at freeswitch.org  Tue Feb  1 02:59:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Jan 2011 15:59:38 -0800
Subject: [Freeswitch-users] Outbound only calls don't connect when
 bypass_media is true.
In-Reply-To: <COL118-W196844C55E71088CAD1FC1E5E20@phx.gbl>
References: <COL118-W52E5226FC3337085AB6AFAE5E00@phx.gbl>
	<COL118-W38F26048D441DAA2777942E5E30@phx.gbl>
	<AANLkTinbiU4zNc-ZM+u+iST7E2aZ3zisgh5Lu2FTuBLt@mail.gmail.com>
	<COL118-W196844C55E71088CAD1FC1E5E20@phx.gbl>
Message-ID: <AANLkTikLgZ_-1QVQWBqxKZMF7q=Ep5oDHLdo0o-J=XXs@mail.gmail.com>

I know of another user who had issues with voip.ms. In his case the issue
was that voip.ms was sending a burst of GSM before switching to PCMU. He
eliminated the issue by disabling all codecs except PCMU on the FS <-->
voip.ms configs. You may want to double-check the voip.ms configs and remove
everything except PCMU.

-MC

On Sun, Jan 30, 2011 at 8:40 PM, Marcin Wojtowicz <marcin321 at hotmail.com>wrote:

>  OK, so I gave up on bypass media, but now I have another problem. This
> time I set up freeswitch to communicate with voip.ms using PCMU codec
> (configured in my external profile), and use iLBC on my phone (codec
> configured in my internal profile, where the phone registers). When I call
> my mobile it rings, but when I pick up all I hear is a high pitched squeal.
> Am I missing something here?
>
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From infos at madovsky.org  Tue Feb  1 03:06:57 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 31 Jan 2011 19:06:57 -0500
Subject: [Freeswitch-users] sip_contact_host
References: <C81EC95BE43D42DCB6FB1727EEF4AAB5@e1705>
	<D786BA11-AEBB-4F4A-8E05-E32B9CE910F1@freeswitch.org>
Message-ID: <58540CAE49B84099831D3348670FD069@e1705>

sorry Brian,
apparently there were 2 instances of FS already running
I don't know why so after killall -9 and restart
FS sip_invite_domain works fine.

Thank you 
  ----- Original Message ----- 
  From: Brian West 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 31, 2011 5:37 PM
  Subject: Re: [Freeswitch-users] sip_contact_host


  set sip_invite_domain


  /b


  On Jan 31, 2011, at 4:22 PM, Madovsky wrote:


    I set my internal profile with a domain
    but sip_contact_host stays an IP.
    how to change 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
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From curriegrad2004 at gmail.com  Tue Feb  1 03:09:43 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 31 Jan 2011 16:09:43 -0800
Subject: [Freeswitch-users] freeswitch.com is returned to us!
In-Reply-To: <5FB9599FB0CA4FA496A87C65DAFB9D53@e1705>
References: <AANLkTi=BciYcFugPySe3QU7pK-=avKSpkOUiq119UCYP@mail.gmail.com>
	<5FB9599FB0CA4FA496A87C65DAFB9D53@e1705>
Message-ID: <AANLkTik1QwQs7vo0uaiWzfphh93BCNiSU_kPwoQ9nkjd@mail.gmail.com>

Anthony, are you inclined to share what happened to freeswitch.com originally?

On Mon, Jan 31, 2011 at 2:19 PM, Madovsky <infos at madovsky.org> wrote:
> yeah nice ! :D
>
> ----- Original Message -----
> From: "Anthony Minessale" <anthony.minessale at gmail.com>
> To: "Freeswitch-users" <freeswitch-users at lists.freeswitch.org>;
> <freeswitch-dev at lists.freeswitch.org>
> Sent: Monday, January 31, 2011 5:15 PM
> Subject: [Freeswitch-users] freeswitch.com is returned to us!
>
>
>> http://www.freeswitch.com/
>>
>> Thank you everyone who helped with this!
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb  1 03:12:47 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Jan 2011 16:12:47 -0800
Subject: [Freeswitch-users] How to play Background music during
	conference?
In-Reply-To: <1296457063838-5976296.post@n2.nabble.com>
References: <1296457063838-5976296.post@n2.nabble.com>
Message-ID: <AANLkTimLfDrD6K=Q25Bm9HQRTvqzzOuyEFYGek3hYCrT@mail.gmail.com>

Have you checked the console debug log to see if there are any issues when
the perpetual sound is invoked? Or even to confirm that the perpetual sound
is invoked? You may wish to pastebin your relevant configs so that others
can double-check them for correctness.

-MC

On Sun, Jan 30, 2011 at 10:57 PM, kapil.rastogi
<kapil.rastogi at telemune.net>wrote:

>
> Dear Friends,
>
> I am using javascript as a scripting language in freeSWITCH.
>
> I want to play the background music during conference without mute any
> member in the conference room.
>
> I tried "perpetual-sound" but it is not working. Can anyone help me about
> this? If yes, then please also send me the example code to play background
> music.
>
> -----
> 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/How-to-play-Background-music-during-conference-tp5976296p5976296.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 Feb  1 03:15:23 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Jan 2011 16:15:23 -0800
Subject: [Freeswitch-users] How to play wav files from other path?
In-Reply-To: <1296457348066-5976304.post@n2.nabble.com>
References: <1296457348066-5976304.post@n2.nabble.com>
Message-ID: <AANLkTinN=UK=YSmus9xfY1upMn5ou+AxhmU8u8iV-cUL@mail.gmail.com>

Specify an absolute path instead of a relative path. For example
data="/tmp/file.wav" looks in /tmp whereas
data="ivr/ivr-welcome_to_freeswitch.wav" will look in
/usr/local/freeswitch/sounds/en/us/call/ivr/<rate>/

-MC

On Sun, Jan 30, 2011 at 11:02 PM, kapil.rastogi
<kapil.rastogi at telemune.net>wrote:

>
> Hi,
>
> I want to play wav files from the different path. As now it is playing the
> wav files from:
> "/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/"
> path.
>
> Please tell me how to play wav files from other path using javascript
> application.
>
> Thanks in advance....
>
> -----
> 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/How-to-play-wav-files-from-other-path-tp5976304p5976304.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 anthony.minessale at gmail.com  Tue Feb  1 03:50:36 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 31 Jan 2011 18:50:36 -0600
Subject: [Freeswitch-users] freeswitch.com is returned to us!
In-Reply-To: <AANLkTik1QwQs7vo0uaiWzfphh93BCNiSU_kPwoQ9nkjd@mail.gmail.com>
References: <AANLkTi=BciYcFugPySe3QU7pK-=avKSpkOUiq119UCYP@mail.gmail.com>
	<5FB9599FB0CA4FA496A87C65DAFB9D53@e1705>
	<AANLkTik1QwQs7vo0uaiWzfphh93BCNiSU_kPwoQ9nkjd@mail.gmail.com>
Message-ID: <AANLkTimVnELkhvJF860CKc-PzkYZijUp67RR-fh_27-t@mail.gmail.com>

Some domain squatter registered it and we did not originally realize it.


On Mon, Jan 31, 2011 at 6:09 PM, curriegrad2004
<curriegrad2004 at gmail.com> wrote:
> Anthony, are you inclined to share what happened to freeswitch.com originally?
>
> On Mon, Jan 31, 2011 at 2:19 PM, Madovsky <infos at madovsky.org> wrote:
>> yeah nice ! :D
>>
>> ----- Original Message -----
>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>> To: "Freeswitch-users" <freeswitch-users at lists.freeswitch.org>;
>> <freeswitch-dev at lists.freeswitch.org>
>> Sent: Monday, January 31, 2011 5:15 PM
>> Subject: [Freeswitch-users] freeswitch.com is returned to us!
>>
>>
>>> http://www.freeswitch.com/
>>>
>>> Thank you everyone who helped with this!
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Tue Feb  1 04:44:44 2011
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 1 Feb 2011 09:44:44 +0800
Subject: [Freeswitch-users] freeswitch.com is returned to us!
In-Reply-To: <AANLkTimVnELkhvJF860CKc-PzkYZijUp67RR-fh_27-t@mail.gmail.com>
References: <AANLkTi=BciYcFugPySe3QU7pK-=avKSpkOUiq119UCYP@mail.gmail.com>
	<5FB9599FB0CA4FA496A87C65DAFB9D53@e1705>
	<AANLkTik1QwQs7vo0uaiWzfphh93BCNiSU_kPwoQ9nkjd@mail.gmail.com>
	<AANLkTimVnELkhvJF860CKc-PzkYZijUp67RR-fh_27-t@mail.gmail.com>
Message-ID: <AANLkTinqW0ndoNGH337Wbn2P+KF9rhid83m6KehoSjmT@mail.gmail.com>

Nice !

On Tue, Feb 1, 2011 at 8:50 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Some domain squatter registered it and we did not originally realize it.
>
>
> On Mon, Jan 31, 2011 at 6:09 PM, curriegrad2004
> <curriegrad2004 at gmail.com> wrote:
>> Anthony, are you inclined to share what happened to freeswitch.com originally?
>>
>> On Mon, Jan 31, 2011 at 2:19 PM, Madovsky <infos at madovsky.org> wrote:
>>> yeah nice ! :D
>>>
>>> ----- Original Message -----
>>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>>> To: "Freeswitch-users" <freeswitch-users at lists.freeswitch.org>;
>>> <freeswitch-dev at lists.freeswitch.org>
>>> Sent: Monday, January 31, 2011 5:15 PM
>>> Subject: [Freeswitch-users] freeswitch.com is returned to us!
>>>
>>>
>>>> http://www.freeswitch.com/
>>>>
>>>> Thank you everyone who helped with this!
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 Nabble at slickdeals.endjunk.com  Tue Feb  1 05:10:16 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 31 Jan 2011 18:10:16 -0800 (PST)
Subject: [Freeswitch-users] Developer Dinner Coming Up - Help Feed The
	FreeSWITCH Team!
In-Reply-To: <AANLkTimZjT2SQ=Go10JwLrXJEmjQt+iNZQrUKVHPBgj=@mail.gmail.com>
References: <AANLkTimZjT2SQ=Go10JwLrXJEmjQt+iNZQrUKVHPBgj=@mail.gmail.com>
Message-ID: <1296526216699-5979632.post@n2.nabble.com>


How about donate food?

-----
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/Developer-Dinner-Coming-Up-Help-Feed-The-FreeSWITCH-Team-tp5979347p5979632.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Tue Feb  1 05:22:06 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Jan 2011 18:22:06 -0800
Subject: [Freeswitch-users] Developer Dinner Coming Up - Help Feed The
 FreeSWITCH Team!
In-Reply-To: <1296526216699-5979632.post@n2.nabble.com>
References: <AANLkTimZjT2SQ=Go10JwLrXJEmjQt+iNZQrUKVHPBgj=@mail.gmail.com>
	<1296526216699-5979632.post@n2.nabble.com>
Message-ID: <AANLkTimmnYQdGKP5USuzVCb+w7Y1BBg31W_xnkJTji+c@mail.gmail.com>

I suppose if you can work out the logistics then yes, actually donating food
would be acceptable. If you have a way to do that then let me know.
-MC

On Mon, Jan 31, 2011 at 6:10 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
> How about donate food?
>
> -----
> 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/Developer-Dinner-Coming-Up-Help-Feed-The-FreeSWITCH-Team-tp5979347p5979632.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 Nabble at slickdeals.endjunk.com  Tue Feb  1 05:55:11 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 31 Jan 2011 18:55:11 -0800 (PST)
Subject: [Freeswitch-users] Developer Dinner Coming Up - Help Feed The
	FreeSWITCH Team!
In-Reply-To: <AANLkTimmnYQdGKP5USuzVCb+w7Y1BBg31W_xnkJTji+c@mail.gmail.com>
References: <AANLkTimZjT2SQ=Go10JwLrXJEmjQt+iNZQrUKVHPBgj=@mail.gmail.com>
	<1296526216699-5979632.post@n2.nabble.com>
	<AANLkTimmnYQdGKP5USuzVCb+w7Y1BBg31W_xnkJTji+c@mail.gmail.com>
Message-ID: <1296528911018-5979706.post@n2.nabble.com>


No and I don't know any way to donate food. The reason I asked was who knows
someone who prefers to donate food than money. Perhaps, some people would
also like to donate and/or bring in some can food to participate in the
even. Who knows.

-----
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/Developer-Dinner-Coming-Up-Help-Feed-The-FreeSWITCH-Team-tp5979347p5979706.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Tue Feb  1 05:58:27 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 31 Jan 2011 20:58:27 -0600
Subject: [Freeswitch-users] Developer Dinner Coming Up - Help Feed The
 FreeSWITCH Team!
In-Reply-To: <1296528911018-5979706.post@n2.nabble.com>
References: <AANLkTimZjT2SQ=Go10JwLrXJEmjQt+iNZQrUKVHPBgj=@mail.gmail.com>
	<1296526216699-5979632.post@n2.nabble.com>
	<AANLkTimmnYQdGKP5USuzVCb+w7Y1BBg31W_xnkJTji+c@mail.gmail.com>
	<1296528911018-5979706.post@n2.nabble.com>
Message-ID: <AANLkTimSkXwfpS_nstzTon4_re_fHA7SNv9Cp001+8na@mail.gmail.com>

Due to our limited time together we will probably be eating from a
restaurant to save time.
If someone wants to donate food, I recommend to donate it to someone
locally who needs food on our behalf.




On Mon, Jan 31, 2011 at 8:55 PM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
> No and I don't know any way to donate food. The reason I asked was who knows
> someone who prefers to donate food than money. Perhaps, some people would
> also like to donate and/or bring in some can food to participate in the
> even. Who knows.
>
> -----
> 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/Developer-Dinner-Coming-Up-Help-Feed-The-FreeSWITCH-Team-tp5979347p5979706.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  Tue Feb  1 06:03:57 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 31 Jan 2011 22:03:57 -0500
Subject: [Freeswitch-users] Developer Dinner Coming Up - Help Feed
	TheFreeSWITCH Team!
References: <AANLkTimZjT2SQ=Go10JwLrXJEmjQt+iNZQrUKVHPBgj=@mail.gmail.com>
	<1296526216699-5979632.post@n2.nabble.com>
Message-ID: <BF9E9F3DB6C3408991502E1957F24674@e1705>

I can make delicious pizzas, lasagnes and sushi...

----- Original Message ----- 
From: "mazilo" <Nabble at slickdeals.endjunk.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Monday, January 31, 2011 9:10 PM
Subject: Re: [Freeswitch-users] Developer Dinner Coming Up - Help Feed 
TheFreeSWITCH Team!


>
> How about donate food?
>
> -----
> 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/Developer-Dinner-Coming-Up-Help-Feed-The-FreeSWITCH-Team-tp5979347p5979632.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 joaocarlosleme at gmail.com  Tue Feb  1 06:37:20 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Mon, 31 Jan 2011 19:37:20 -0800
Subject: [Freeswitch-users] Remote LogIn to Freeswitch?
In-Reply-To: <AANLkTinQrp9+Ntp+yvVrwSBERx2JHK9kR94QN=RFVLBi@mail.gmail.com>
References: <10183.49543.qm@web114709.mail.gq1.yahoo.com>
	<A6C135FB-AD49-4479-9A51-78A1A5341F43@gmail.com>
	<AANLkTinQrp9+Ntp+yvVrwSBERx2JHK9kR94QN=RFVLBi@mail.gmail.com>
Message-ID: <AANLkTi=giLnDhX346oYp2jzrcyYOVQk6+O-1i2jj-uKH@mail.gmail.com>

While registered I issued the command "sofia status profile internal" and
notice that my registration information was mentioning "Port: 6395", and
under the firewall rules "http://wiki.freeswitch.org/wiki/Firewall" there is
no rule for this port or range. Do you think it may have something to do
with my "no sound" problem? What is this "Port" used for? How is it assigned
(what is the range)? Later I did the same from home (remote) and it was
using port 5177 (not mentioned on Firewall rules neither).
Thanks,
John

On Sat, Jan 29, 2011 at 2:19 PM, Avi Marcus <avi at avimarcus.net> wrote:

> 5080 for "external" profile means for creating outbound connections to a
> gateway, or for allowing incoming un-authed calls.
> This question about "remote" login was referring to a login from outside
> the local network, which should be going to the internal profile.
>
> -Avi
>
>
> On Sun, Jan 30, 2011 at 12:15 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> a) according to Wiki, you should set port 5080 for remote connections
>>
>>
>> That's for the default config and entirely depends on what you have
>> configured. There is absolutely no reason it *must* be 5080.
>>
>> Steve on iPhone
>>
>> On 28 Jan 2011, at 23:15, Kenan BEKTAS <bektas at yahoo.com> wrote:
>>
>> a) according to Wiki, you should set port 5080 for remote connections.
>> b) if no sound for some reason, then, it is most likely a RTP issue. Make
>> sure RTP port ranges is open on your router.
>>
>> ---
>> Kenan
>> www.dbstreams.ca
>>
>>
>>
>>
>> --- On *Fri, 1/28/11, Joao Leme <joaocarlosleme at gmail.com>* wrote:
>>
>>
>> From: Joao Leme <joaocarlosleme at gmail.com>
>> Subject: Re: [Freeswitch-users] Remote LogIn to Freeswitch?
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Date: Friday, January 28, 2011, 5:31 PM
>>
>> FINALLY I got it to work, I can now remotely log in on the internal
>> profile (port 5060) after placing the computer running Freeswitch on DMZ
>> (changed on Firewall), although that's not the ideal solution, so I was
>> wondering why it wasn't working before. I had the firewall set up according
>> to the wiki on "http://wiki.freeswitch.org/wiki/Firewall" and other than
>> that I changed the "domain" on vars.xml to my external ip of the router, it
>> created an ALIAS so that I can use as the "Domain" on X-lite to log on. So I
>> had the internal profile mod_sofia at 192.168.X.XX:5060 and the Alias
>> 76.XXX.XX.XX, that was before and I was able to LogIn, make calls but NO
>> SOUND. NOW once I forwarded all the port through DMZ, this particular router
>> 2701HG-B Gateway (Att) required me to change my (freeSwitch running
>> computer) static ipv4 (192.XXX.X.XX) to DHCP and then assigned me the
>> external IP (76.XXX.XX.XX) as my a Ipv4. Now the internal profile (sofia
>> status) is mod_sofia at 76.XXX.XX.XX:5060 and I'm not using the ALIAS.
>>
>> SO...I was wondering...it wasn't working before because 1) A Firewall
>> configuration missing? 2) A NAT/Firewall problem or 3) Freeswitch internal
>> profile running on local machine ipv4 192..... and I logging on using the
>> router external IP 76.XX and the Alias didn't do the job?????
>>
>> Sorry there is some nonsense question, I'm a beginner and any help is
>> appreciated.
>> Thanks,
>> John
>>
>> On Tue, Jan 11, 2011 at 1:37 AM, Steven Ayre <steveayre at gmail.com<http://mc/compose?to=steveayre at gmail.com>
>> > wrote:
>>
>> If you're able to dial in but you're getting no sound, it's probably
>> NAT stopping the audio get through.
>>
>> There's quite a bit of information on NAT on the Wiki that might be of
>> use.
>>
>> http://wiki.freeswitch.org/wiki/NAT
>>
>> -Steve
>>
>>
>> On 11 January 2011 00:39, Joao Leme <joaocarlosleme at gmail.com<http://mc/compose?to=joaocarlosleme at gmail.com>>
>> wrote:
>> > Hi There,
>> > What do I have to do to be able to LogIn to Freeswitch from Home (server
>> is
>> > located at office) starting from the basic/original configuration?
>> > I'm using X-Lite. I've been able to LogIn replacing the internal IP by
>> the
>> > external IP from the Office but the sound is not working so I wanted to
>> know
>> > what are the configuration changes that have to be done to allow it. Do
>> I
>> > have to create a different profile? I want be able to do the same just
>> as if
>> > I was at the Office.
>> > Thanks,
>> > John
>> > _______________________________________________
>> > 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
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Tue Feb  1 07:20:44 2011
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 1 Feb 2011 12:20:44 +0800
Subject: [Freeswitch-users] Developer Dinner Coming Up - Help Feed
 TheFreeSWITCH Team!
In-Reply-To: <BF9E9F3DB6C3408991502E1957F24674@e1705>
References: <AANLkTimZjT2SQ=Go10JwLrXJEmjQt+iNZQrUKVHPBgj=@mail.gmail.com>
	<1296526216699-5979632.post@n2.nabble.com>
	<BF9E9F3DB6C3408991502E1957F24674@e1705>
Message-ID: <AANLkTimXcAQ1ZZee3ZDwQHWrTC+FABqf3DQrghstUeWd@mail.gmail.com>

$20 confirmation No. 18U27974AX8119748 . Enjoy!

It's almost 3 years since I'v joined the community. I started using FS
since then and I like the way FS works and so does the community.
People helped me a lot from the basics to deeper VoIP knowledge.
Anthony, Mike and other members also used their owen time helped me to
solve my problems with login into our box.

I didn't make a lot of money from FS directly, but FS allows us to do
funny things and that's why I like it. By the spirit of thanks, I also
reported some bugs and submitted some patches. Also there's $10 in
freeswitch.com (Didn't get thanked I think I didn't left proper
messages on donating). Also, I know that money is not the only way to
help. I also created http://www.freeswitch.org.cn which is 1yr old and
maintains a freeswitch-cn google groups. But due to the policy of
China Internet, only a few people can access google groups and
consequently we didn't get strong enough. The website is deployed on
Heroku which is on EC2 and we have difficulties to access it in the
last month. Anyway, those won't stop me march ahead.

Sorry a bit off topic, just want to say thanks hope not too long.

Thanks again,
Seven.

On Tue, Feb 1, 2011 at 11:03 AM, Madovsky <infos at madovsky.org> wrote:
> I can make delicious pizzas, lasagnes and sushi...
>
> ----- Original Message -----
> From: "mazilo" <Nabble at slickdeals.endjunk.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, January 31, 2011 9:10 PM
> Subject: Re: [Freeswitch-users] Developer Dinner Coming Up - Help Feed
> TheFreeSWITCH Team!
>
>
>>
>> How about donate food?
>>
>> -----
>> 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/Developer-Dinner-Coming-Up-Help-Feed-The-FreeSWITCH-Team-tp5979347p5979632.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
>



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


From kapil.rastogi at telemune.net  Tue Feb  1 10:35:35 2011
From: kapil.rastogi at telemune.net (kapil.rastogi)
Date: Mon, 31 Jan 2011 23:35:35 -0800 (PST)
Subject: [Freeswitch-users] How to play wav files from other path?
In-Reply-To: <AANLkTinN=UK=YSmus9xfY1upMn5ou+AxhmU8u8iV-cUL@mail.gmail.com>
References: <1296457348066-5976304.post@n2.nabble.com>
	<AANLkTinN=UK=YSmus9xfY1upMn5ou+AxhmU8u8iV-cUL@mail.gmail.com>
Message-ID: <1296545735239-5980059.post@n2.nabble.com>


Thanks for ur valuable reply. Now i am able to play wav files from other path
also.

-----
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/How-to-play-wav-files-from-other-path-tp5976304p5980059.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kapil.rastogi at telemune.net  Tue Feb  1 11:04:28 2011
From: kapil.rastogi at telemune.net (kapil.rastogi)
Date: Tue, 1 Feb 2011 00:04:28 -0800 (PST)
Subject: [Freeswitch-users] How to play Background music during
	conference?
In-Reply-To: <AANLkTimLfDrD6K=Q25Bm9HQRTvqzzOuyEFYGek3hYCrT@mail.gmail.com>
References: <1296457063838-5976296.post@n2.nabble.com>
	<AANLkTimLfDrD6K=Q25Bm9HQRTvqzzOuyEFYGek3hYCrT@mail.gmail.com>
Message-ID: <1296547468040-5980114.post@n2.nabble.com>


Hi,

Thanks for your response.
When i uncomment the perpetual-sound line in conference.conf.xml file, then
it is playing the file specified with that configuration in default
conference. But I want to play the file using my javascript application. I
am using the following statement in my javascript file to play file in
conference:

session.execute("conference", "conf-room--1000 at default perpetual-sound
/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/BGFILE.wav");


In logs, it executed the command but i am unable to listen this file.

Can you tell me what I have to do to play this file from my javascript
application?

Thanks in advance.

-----
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/How-to-play-Background-music-during-conference-tp5976296p5980114.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Stefan.Weigel at allianz-warranty.com  Tue Feb  1 11:18:30 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Tue, 1 Feb 2011 09:18:30 +0100
Subject: [Freeswitch-users] mod_callcenter and effective_caller_id_name
In-Reply-To: <5003D7D3E06F514E8C682F18D223265C04717DC7F0@AZWSMS03.azwarranty.int>
References: <5003D7D3E06F514E8C682F18D223265C04717DC7F0@AZWSMS03.azwarranty.int>
Message-ID: <5003D7D3E06F514E8C682F18D223265C04717DC7F1@AZWSMS03.azwarranty.int>

Hi,

to solve this issue you have to export the caller_id_name:

<action inline="true" application="export" data="caller_id_name=Hotline_abc"/>


Best regards

Stefan

Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Weigel, Stefan
Gesendet: Montag, 31. Januar 2011 14:38
An: FreeSWITCH Users Help
Betreff: [Freeswitch-users] mod_callcenter and effective_caller_id_name

 Hi all,

when modifying 'effective_caller_id_name' it's working for user extensions. When routing to application mod_callcenter the change doesn't take effect. Is it a bug or am I missing something ?


Thanks and best regards

Stefan
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From erik.dekkers at wvds.nl  Tue Feb  1 12:28:07 2011
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Tue, 1 Feb 2011 10:28:07 +0100
Subject: [Freeswitch-users] minimum UDP packet size
In-Reply-To: <65064C9F3B344E4A9CAE4A9B63C318D3@e1705>
References: <65064C9F3B344E4A9CAE4A9B63C318D3@e1705>
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE0787301836@FS1.wvds.local>

You probably mean a MAXimum size. If you set a minimum size it would always fragment.
As far as I know it itsn't possible to set a maximum size of a UDP packet. If the UDP packets are getting too large you really should trim down SIP headers / Remove codecs from the SDP.

There are some articles on the internet concerning MTU and SIP.

Erik

Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Madovsky
Verzonden: maandag 31 januari 2011 18:29
Aan: freeswitch-users at lists.freeswitch.org
Onderwerp: [Freeswitch-users] minimum UDP packet size

is it useful to set minimum UDP  pakcet to the
MTU rate like 1500 and avoid any fragmentation ?

Thanks
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From anton.vazir at gmail.com  Tue Feb  1 13:07:54 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Tue, 1 Feb 2011 15:07:54 +0500
Subject: [Freeswitch-users] open g729
In-Reply-To: <AANLkTi=icT=Zw9xrrEcOyyozy5JFm7sXQjxTDxN3psZH@mail.gmail.com>
References: <1296343422.2615.4.camel@gustavo-laptop>
	<AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com>
	<1296477877.3589.24.camel@gustavo-laptop>
	<AANLkTinq9AmN=ZqjGLNUxucw2nJXBLpNHKLTA5jLWfgj@mail.gmail.com>
	<9D7F812E93DC47A99F6A30638FB8092F@e1705>
	<AANLkTikkLnf4BHYgL1k8th=VWk44vpH5gRYSAFW5YfW1@mail.gmail.com>
	<A64FADA67C6D4C1C8B041EEFBB8C1DBB@e1705>
	<AANLkTi=1Fr2e-sz-n5Ds=q6KVntRzfzgrAQOgc0rCYgF@mail.gmail.com>
	<F4AB52F0A27D46418900E2841CB5524B@e1705>
	<AANLkTi=icT=Zw9xrrEcOyyozy5JFm7sXQjxTDxN3psZH@mail.gmail.com>
Message-ID: <AANLkTinXuGfqOBdBcLSdBMaoMJk3JbWg18sfkKyyVofd@mail.gmail.com>

The question that bugs me, that if some one have 1000 licenses for
Asterisk g729 (10K$!) , and wants to switch over from asterisk to a
FreeSWITCH, would one have to buy that 100 liceses again, or there is
a kind of procedure to get the module for existing licenses? Most
likely no way :) But it's worth to mention a problem, which might so
much keeping one on the certain platform. And that is the case when
one may consider, as owner of the licenses to go with the IPP (or
other open implementation) without any law breakage.


From yehavi.bourvine at gmail.com  Tue Feb  1 13:34:42 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 1 Feb 2011 12:34:42 +0200
Subject: [Freeswitch-users] Question about DTMF duration
Message-ID: <AANLkTimvgJ7mOHuDCMvba_=PiAQ1h9Q0rVVC0D7ORJ28@mail.gmail.com>

Hello,

  While trying to debug DTMF problems with some gateways I came across the
following issue: The phone generates the DTMF of a certain period, but I
would like the PBX to send a longer period to the gateway. Thus, I change
switch.conf.xml to have *min-dtmf-duration=*80000  and at the outgoing
profile toward the gateway I put *dtmf-duration=*80000; I tried large values
in order to see the difference. However, wireshark stil shows much smaller
duration (around 1,000).

Do I understand something incorrectly?

                                 Thanks, __Yehavi:
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From lakindia89 at gmail.com  Tue Feb  1 14:01:14 2011
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Tue, 1 Feb 2011 16:31:14 +0530
Subject: [Freeswitch-users] Blind Transfer Not working
In-Reply-To: <AANLkTim7-A=NRjwND3tyzJeDk4boiuCrar5xv-7SB428@mail.gmail.com>
References: <AANLkTim7-A=NRjwND3tyzJeDk4boiuCrar5xv-7SB428@mail.gmail.com>
Message-ID: <AANLkTi=u9uOt-METcshfs2GNr+q7KjZyWCPeOdzh_q_p@mail.gmail.com>

Does any one have any clues on this?

On Mon, Jan 31, 2011 at 6:18 PM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Hi all,
>
> here is my dialplan
>         <context name="default">
>                 <extension name="outbound_soc">
>                         <condition field="destination_number"
> expression="^.*$">
>                                 <action application="log" data="ERR
> ${domain_name}"/>
>                                 <action application="set"
> data="continue_on_fail=true"/>
>                                 <action application="set"
> data="bypass_media=false"/>
>                                 <action application="ring_ready"/>
>                                 <action application="set"
> data="ignore_early_media=true"/>
>                                 <action application="set"
> data="exec_after_bridge_app=park"/>
>                                 <action application="export"
> data="RECORD_STEREO=true"/>
>                                 <action application="bridge"
> data="user/1000"/>
>                         </condition>
>                 </extension>
>         </context>
>
> I made a call from 9952248266 to 39114601. As expected it called to the
> 1000 extension and I answered the call. I was using twinkle as a softphone.
> Now from twinkle I initiated "Transfer". I've given "9976975781" as a
> number and I choose "Blind Transfer".
> But the call didn't get proceeded.
>
> Here is the log with sofia trace enabled.
> http://pastebin.freeswitch.org/15189
>
> Once REFER is received by freeswitch, It replied with "Accept" for that
> refer, but after that the call was not proceeding!!
>
>
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From u2nsam at gmail.com  Tue Feb  1 14:05:57 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 1 Feb 2011 16:35:57 +0530
Subject: [Freeswitch-users] port reg request pbm
Message-ID: <AANLkTinHymqs1CA5fOgFHkd8PW51xGjcpX+jzo8w_g5A@mail.gmail.com>

hello,

The device is sending request to my server ip x.y.z.v from  port 35666

a.b.c.e:35666 -> x.y.z.v :5060
Register

but FS is not sending back to the same port on the device

x.y.z.v:5060 -> a.b.c.e : 5060
401 unauthorized

how this could be solved

Regards
Sam
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From david.ponzone at ipeva.fr  Tue Feb  1 14:10:25 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 1 Feb 2011 12:10:25 +0100
Subject: [Freeswitch-users] port reg request pbm
In-Reply-To: <AANLkTinHymqs1CA5fOgFHkd8PW51xGjcpX+jzo8w_g5A@mail.gmail.com>
References: <AANLkTinHymqs1CA5fOgFHkd8PW51xGjcpX+jzo8w_g5A@mail.gmail.com>
Message-ID: <D71B7E62-D1FE-4C41-9B3B-724507847AA0@ipeva.fr>

This probably means your device is behind NAT, and that it does not send the rport field where it should.
So, either you enable rport on the device, or you use this:

http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#NDLB-force-rport

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 01/02/2011 ? 12:05, Sam a ?crit :

> hello,
> 
> The device is sending request to my server ip x.y.z.v from  port 35666
> 
> a.b.c.e:35666 -> x.y.z.v :5060
> Register
> 
> but FS is not sending back to the same port on the device 
> 
> x.y.z.v:5060 -> a.b.c.e : 5060
> 401 unauthorized
> 
> how this could be solved 
> 
> 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 bernhard.suttner at winet.ch  Tue Feb  1 14:11:12 2011
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Tue, 1 Feb 2011 12:11:12 +0100
Subject: [Freeswitch-users] port reg request pbm
In-Reply-To: <AANLkTinHymqs1CA5fOgFHkd8PW51xGjcpX+jzo8w_g5A@mail.gmail.com>
References: <AANLkTinHymqs1CA5fOgFHkd8PW51xGjcpX+jzo8w_g5A@mail.gmail.com>
Message-ID: <efc1e70e-c4b5-4571-bb34-affd29da49b0@winet.ch>

Sounds like NAT issue. Try around with the NAT possibilities of FreeSWITCH.
 
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Sam
Gesendet: Dienstag, 1. Februar 2011 12:06
An: FreeSWITCH Users Help
Betreff: [Freeswitch-users] port reg request pbm
 
hello,

The device is sending request to my server ip x.y.z.v from  port 35666

a.b.c.e:35666 -> x.y.z.v :5060
Register

but FS is not sending back to the same port on the device 

x.y.z.v:5060 -> a.b.c.e : 5060
401 unauthorized

how this could be solved 

Regards
Sam

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From brent at overthewire.com.au  Tue Feb  1 14:14:46 2011
From: brent at overthewire.com.au (Brent Paddon)
Date: Tue, 1 Feb 2011 21:14:46 +1000
Subject: [Freeswitch-users] open g729
In-Reply-To: <AANLkTinXuGfqOBdBcLSdBMaoMJk3JbWg18sfkKyyVofd@mail.gmail.com>
References: <1296343422.2615.4.camel@gustavo-laptop>
	<AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com>
	<1296477877.3589.24.camel@gustavo-laptop>
	<AANLkTinq9AmN=ZqjGLNUxucw2nJXBLpNHKLTA5jLWfgj@mail.gmail.com>
	<9D7F812E93DC47A99F6A30638FB8092F@e1705>
	<AANLkTikkLnf4BHYgL1k8th=VWk44vpH5gRYSAFW5YfW1@mail.gmail.com>
	<A64FADA67C6D4C1C8B041EEFBB8C1DBB@e1705>
	<AANLkTi=1Fr2e-sz-n5Ds=q6KVntRzfzgrAQOgc0rCYgF@mail.gmail.com>
	<F4AB52F0A27D46418900E2841CB5524B@e1705>
	<AANLkTi=icT=Zw9xrrEcOyyozy5JFm7sXQjxTDxN3psZH@mail.gmail.com>
	<AANLkTinXuGfqOBdBcLSdBMaoMJk3JbWg18sfkKyyVofd@mail.gmail.com>
Message-ID: <AANLkTinYuVNVA=j65hjPiPZX67gbJexO-SrQymQYMqv_@mail.gmail.com>

So buy a hardware solution, then you can move it around.  Plus it (I think)
works out cheaper than the software versions.

I also can't believe that any user of this (or some other) open source
switch can't afford a few measly dollars for their transcoding requirements.

Brent

On Tue, Feb 1, 2011 at 8:07 PM, Anton VG <anton.vazir at gmail.com> wrote:

> The question that bugs me, that if some one have 1000 licenses for
> Asterisk g729 (10K$!) , and wants to switch over from asterisk to a
> FreeSWITCH, would one have to buy that 100 liceses again, or there is
> a kind of procedure to get the module for existing licenses? Most
> likely no way :) But it's worth to mention a problem, which might so
> much keeping one on the certain platform. And that is the case when
> one may consider, as owner of the licenses to go with the IPP (or
> other open implementation) without any law breakage.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From brent at overthewire.com.au  Tue Feb  1 14:22:08 2011
From: brent at overthewire.com.au (Brent Paddon)
Date: Tue, 1 Feb 2011 21:22:08 +1000
Subject: [Freeswitch-users] Hacker Attack?
In-Reply-To: <AANLkTimNOHVYE=uaM8djk10mtqweZfaQLg9GUz4=osBA@mail.gmail.com>
References: <09EC9978-ED26-445F-9B0C-34D6D55DDA41@visionutveckling.se>
	<AANLkTimNOHVYE=uaM8djk10mtqweZfaQLg9GUz4=osBA@mail.gmail.com>
Message-ID: <AANLkTimb+bNiK3u8t=xoSEnBz8=YTKJvLDgM4jJC22HX@mail.gmail.com>

If you are not sure how to secure a box like this down - I sure hope for
your sake that your telephony provider has some good anti-fraud measures in
place or you have deep pockets and don't mind sending great wads of cash off
to your provider(s).

You might want to spend a good amount of time coming up to speed with best
practice security for boxes which do SIP and are connected to the public
Internet.

Brent

On Mon, Jan 31, 2011 at 1:54 AM, Joao Leme <joaocarlosleme at gmail.com> wrote:

> I figured. Same for Fail2Ban I guess. Any suggestions for Windows?
>
> Also I was wondering why it never happened on my 1.0.4 (14460) version
> (precompiled version)? I had it running for a month 24hrs and had never seen
> this before. And after starting the Git Head (below) from Yesterday it
> happened in seconds all 3 times I restarted (restarted the computer to be
> sure). Maybe something wrong with the current version? To be safe I went
> back to my stable 1.0.4 version and haven't had any problems.
>
> 49a5effcdf2cea9e0ddcf146cf3fe85d1872e654
> mod_callcenter: Add error response for queue load and queue reload
> (FS-2988)
> Marc Olivier Chouinard
>  2011-01-29 00:09:06
>
>
> On Sun, Jan 30, 2011 at 2:10 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
>> iptables is a Linux command.
>>
>> /Peter
>>
>>
>> ----- Reply message -----
>> Fr?n: "Joao Leme" <joaocarlosleme at gmail.com>
>> Datum: s?n, jan 30, 2011 13:56
>> Rubrik: [SPAM] - Re: [Freeswitch-users] Hacker Attack?
>> Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>
>> I tried "iptables -I INPUT -s [212.224.71.236] -j DROP" and got " Unknown
>> command: iptables...". Do I must install fail2ban to issue iptables command?
>> I'm on windows 7.
>> Thanks
>>
>> On Sat, Jan 29, 2011 at 4:26 PM, curriegrad2004 <curriegrad2004 at gmail.com
>> <mailto:curriegrad2004 at gmail.com>> wrote:
>> iptables -I INPUT -s [hackerip] -j DROP
>>
>> A better solution is searching the wiki for fail2ban with FreeSwitch.
>>
>> On Sat, Jan 29, 2011 at 4:20 PM, Joao Leme <joaocarlosleme at gmail.com
>> <mailto:joaocarlosleme at gmail.com>> wrote:
>> > How do I do that?
>> > Thanks!
>> > On Sat, Jan 29, 2011 at 4:12 PM, curriegrad2004 <
>> curriegrad2004 at gmail.com<mailto:curriegrad2004 at gmail.com>>
>> > wrote:
>> >>
>> >> Try using iptables and block all incoming traffic from this specific
>> host?
>> >>
>> >> On Sat, Jan 29, 2011 at 3:39 PM, Joao Leme <joaocarlosleme at gmail.com
>> <mailto:joaocarlosleme at gmail.com>>
>> >> wrote:
>> >> > I just downloaded and compiled the latest Git and a little after
>> >> > starting
>> >> > freeswitch I'm getting non stop the following:
>> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> >> > profile
>> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> >> > profile
>> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> >> > profile
>> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> >> > profile
>> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> >> > profile
>> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> >> > profile
>> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> >> > profile
>> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> >> > profile
>> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> >> > profile
>> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> >> > profile
>> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> >> > profile
>> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> >> > profile
>> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> >> > it's non-stop and doesn't let me do nothing else. After the first
>> time I
>> >> > went on to vars and changed the 1234 password....restarted and same
>> >> > thing
>> >> > happened, I also try denying the ip on acl.conf (not sure if has
>> >> > something
>> >> > to do with it but gave it a try):
>> >> >
>> >> > <configuration name="acl.conf" description="Network Lists">
>> >> >         <network-lists>
>> >> >           <list name="test2" default="allow">
>> >> >             <node type="deny" host="212.224.71.236"
>> >> > mask="255.255.255.0"/>
>> >> >           </list>
>> >> >         </network-lists>
>> >> >       </configuration>
>> >> >
>> >> > Restarted the computer but nothing, he (thomas I guess) was back on
>> my
>> >> > console.
>> >> >
>> >> > Any ideas??? p.s. My computer is on DMZ (I know DMZ is not ideal but
>> is
>> >> > the
>> >> > only way I got to be able to connect to the internal profile from out
>> of
>> >> > the
>> >> > office etc).
>> >> > _______________________________________________
>> >> > 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
>>
>> !DSPAM:4d450b3232767678720833!
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From gmaruzz at gmail.com  Tue Feb  1 14:52:57 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 1 Feb 2011 12:52:57 +0100
Subject: [Freeswitch-users] freeswitch.com is returned to us!
In-Reply-To: <AANLkTinqW0ndoNGH337Wbn2P+KF9rhid83m6KehoSjmT@mail.gmail.com>
References: <AANLkTi=BciYcFugPySe3QU7pK-=avKSpkOUiq119UCYP@mail.gmail.com>
	<5FB9599FB0CA4FA496A87C65DAFB9D53@e1705>
	<AANLkTik1QwQs7vo0uaiWzfphh93BCNiSU_kPwoQ9nkjd@mail.gmail.com>
	<AANLkTimVnELkhvJF860CKc-PzkYZijUp67RR-fh_27-t@mail.gmail.com>
	<AANLkTinqW0ndoNGH337Wbn2P+KF9rhid83m6KehoSjmT@mail.gmail.com>
Message-ID: <AANLkTikfp-3JzKNjB1TO-mir8JR-c81X7LSx_gYx-cBy@mail.gmail.com>

Nice!

On Tue, Feb 1, 2011 at 2:44 AM, Seven Du <dujinfang at gmail.com> wrote:
> Nice !
>
> On Tue, Feb 1, 2011 at 8:50 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> Some domain squatter registered it and we did not originally realize it.
>>
>>
>> On Mon, Jan 31, 2011 at 6:09 PM, curriegrad2004
>> <curriegrad2004 at gmail.com> wrote:
>>> Anthony, are you inclined to share what happened to freeswitch.com originally?
>>>
>>> On Mon, Jan 31, 2011 at 2:19 PM, Madovsky <infos at madovsky.org> wrote:
>>>> yeah nice ! :D
>>>>
>>>> ----- Original Message -----
>>>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>>>> To: "Freeswitch-users" <freeswitch-users at lists.freeswitch.org>;
>>>> <freeswitch-dev at lists.freeswitch.org>
>>>> Sent: Monday, January 31, 2011 5:15 PM
>>>> Subject: [Freeswitch-users] freeswitch.com is returned to us!
>>>>
>>>>
>>>>> http://www.freeswitch.com/
>>>>>
>>>>> Thank you everyone who helped with this!
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org
>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From null at invalid.name  Tue Feb  1 14:53:48 2011
From: null at invalid.name (Dan Lane)
Date: Tue, 1 Feb 2011 11:53:48 +0000
Subject: [Freeswitch-users] Choppy VM when using loopback (on Debian
	lenny)
In-Reply-To: <4D466CAE.7030402@estation.dk>
References: <4D42B8EE.7030208@estation.dk>
	<7C6F3FC7-445F-4157-BD59-D76A705E4CE7@carmickle.com>
	<9894FEC6-E451-4993-A85A-2CD910E916B1@ipeva.fr>
	<AANLkTim+aBjRiAvxDUF1O1vut_CJpqgct2OqCNVxi0q4@mail.gmail.com>
	<AANLkTiktckovGv4Ksnn_56EzpvTTSnfG+iD1vPX1_4RM@mail.gmail.com>
	<4D466CAE.7030402@estation.dk>
Message-ID: <AANLkTimhOKmy_yupGtA2HBY_8wXtQKZnDYJ9Ye76bhzq@mail.gmail.com>

FWIW we've been also been having audio issues with loopback recently
on EC2 (with a 1000Hz kernel).

We worked around it in the short term by reverting mod_loopback to
git-4c5426f during the build process.

For anyone else who wants to try this just run "git checkout 4c5426f"
in src/mod/endpoints/mod_loopback then build as usual. This is NOT a
long term solution though.

On Mon, Jan 31, 2011 at 8:02 AM, ?yvind Albrigtsen <oa at estation.dk> wrote:
> Thanks for all your feedback. I'll keep on trying and inform you what
> worked for me.
>
>
> Regards
> Oyvind
>
> On 2011-01-29 21:48, Anthony Minessale wrote:
>> Everyone should try latest GIT before pondering any further because I
>> added a patch like 2 days ago to adress this issue.
>>
>>
>> On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank at telonium.com> ?wrote:
>>
>>> Yeah. I, too, don't see the correlation between the NTP and hw timer..
>>> I am not familiar with the timer_test command and what it's measuring, but
>>> of the 50 tests it ran, min is 19089 and max is 20713.
>>> Frank
>>>
>>>
>>> On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone at ipeva.fr>
>>> wrote:
>>>
>>>> Frank,
>>>> I fail to see the relationship between the hw timer and NTP.
>>>> Can you please elaborate ?
>>>> David Ponzone ?Direction Technique
>>>> email: david.ponzone at ipeva.fr
>>>> tel: ? ? ?01 74 03 18 97
>>>> gsm: ? 06 66 98 76 34
>>>> Service Client IPeva
>>>> tel: ? ? ?0811 46 26 26
>>>> www.ipeva.fr ?- ? www.ipeva-studio.com
>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>
>>>>
>>>>
>>>> Le 28/01/2011 ? 19:49, Frank Carmickle a ?crit :
>>>>
>>>> Hi
>>>>
>>>> On Jan 28, 2011, at 7:39 AM, ?yvind Albrigtsen wrote:
>>>>
>>>> Hi,
>>>>
>>>> I'm using latest git-version of Freeswitch, and when I go to voicemail
>>>>
>>>> when calling a number the sound playback is choppy and it skips some of
>>>>
>>>> the digits in the number I called.
>>>>
>>>> What kind of results do you get from timer_test at the fs_cli? ?Are you
>>>> running on hardware or are you virtualized? ?What is your clock source set
>>>> to and what are your available clock source options? ?See
>>>> /sys/devices/system/clocksource/clocksource0/available_clocksource and
>>>> /sys/devices/system/clocksource/clocksource0/current_clocksource. ?I am
>>>> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to
>>>> hang at 19998/19999 which works very well for me. ?When I was having problem
>>>> it was reporting numbers all over the map from 17400 to 22600 with lots of
>>>> randomness in between. ?I have my clocksource set to jiffies and xen
>>>> independent wallclock set to 1. ?Of course at that point you need to have
>>>> ntp running against a bunch of servers to drive your clock nice and steady.
>>>> ? I know my set up is probably a lot different than yours but I thought I'd
>>>> toss it out there to show that some of the harshest conditions can be dealt
>>>> with and don't give up trying. ?If you are running on hardware with a cpu
>>>> that doesn't have constant_tsc then you might have some problems. ?Just play
>>>> with the different timer options until you find the one that works.
>>>>
>>>> HTH
>>>> --FC
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>>
>>>
>>>
>>
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb  1 14:55:32 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 1 Feb 2011 03:55:32 -0800 (PST)
Subject: [Freeswitch-users] open g729
In-Reply-To: <AANLkTi=icT=Zw9xrrEcOyyozy5JFm7sXQjxTDxN3psZH@mail.gmail.com>
References: <1296343422.2615.4.camel@gustavo-laptop>
	<AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com>
	<1296477877.3589.24.camel@gustavo-laptop>
	<AANLkTinq9AmN=ZqjGLNUxucw2nJXBLpNHKLTA5jLWfgj@mail.gmail.com>
	<9D7F812E93DC47A99F6A30638FB8092F@e1705>
	<AANLkTikkLnf4BHYgL1k8th=VWk44vpH5gRYSAFW5YfW1@mail.gmail.com>
	<A64FADA67C6D4C1C8B041EEFBB8C1DBB@e1705>
	<AANLkTi=1Fr2e-sz-n5Ds=q6KVntRzfzgrAQOgc0rCYgF@mail.gmail.com>
	<F4AB52F0A27D46418900E2841CB5524B@e1705>
	<AANLkTi=icT=Zw9xrrEcOyyozy5JFm7sXQjxTDxN3psZH@mail.gmail.com>
Message-ID: <1296561332402-5980599.post@n2.nabble.com>



Anthony Minessale wrote:
> There is a new codec being developed that is open
> and free and may some day eliminate G729.
Is this an FS in-house CoDec? Do you have its specs and/or link for us to
digest?

-----
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/open-g729-tp5973641p5980599.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Tue Feb  1 14:59:40 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 1 Feb 2011 03:59:40 -0800 (PST)
Subject: [Freeswitch-users] open g729
In-Reply-To: <AANLkTinXuGfqOBdBcLSdBMaoMJk3JbWg18sfkKyyVofd@mail.gmail.com>
References: <AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com>
	<1296477877.3589.24.camel@gustavo-laptop>
	<AANLkTinq9AmN=ZqjGLNUxucw2nJXBLpNHKLTA5jLWfgj@mail.gmail.com>
	<9D7F812E93DC47A99F6A30638FB8092F@e1705>
	<AANLkTikkLnf4BHYgL1k8th=VWk44vpH5gRYSAFW5YfW1@mail.gmail.com>
	<A64FADA67C6D4C1C8B041EEFBB8C1DBB@e1705>
	<AANLkTi=1Fr2e-sz-n5Ds=q6KVntRzfzgrAQOgc0rCYgF@mail.gmail.com>
	<F4AB52F0A27D46418900E2841CB5524B@e1705>
	<AANLkTi=icT=Zw9xrrEcOyyozy5JFm7sXQjxTDxN3psZH@mail.gmail.com>
	<AANLkTinXuGfqOBdBcLSdBMaoMJk3JbWg18sfkKyyVofd@mail.gmail.com>
Message-ID: <1296561580017-5980611.post@n2.nabble.com>



Anton VG wrote:
> 
> The question that bugs me, that if some one have 1000 licenses for
> Asterisk g729 (10K$!) , and wants to switch over from asterisk to a
> FreeSWITCH, would one have to buy that 100 liceses again, or there is
> a kind of procedure to get the module for existing licenses? Most
> likely no way :)
I would concur with you on this.  Otherwise, owners of ATA device (with
included G729 CoDec) no longer have to pay for additional G729 CoDec
licenses for their FS and/or Asterisk. 

-----
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/open-g729-tp5973641p5980611.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveu at coppice.org  Tue Feb  1 15:17:53 2011
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 01 Feb 2011 20:17:53 +0800
Subject: [Freeswitch-users] open g729
In-Reply-To: <AANLkTinXuGfqOBdBcLSdBMaoMJk3JbWg18sfkKyyVofd@mail.gmail.com>
References: <1296343422.2615.4.camel@gustavo-laptop>	<AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com>	<1296477877.3589.24.camel@gustavo-laptop>	<AANLkTinq9AmN=ZqjGLNUxucw2nJXBLpNHKLTA5jLWfgj@mail.gmail.com>	<9D7F812E93DC47A99F6A30638FB8092F@e1705>	<AANLkTikkLnf4BHYgL1k8th=VWk44vpH5gRYSAFW5YfW1@mail.gmail.com>	<A64FADA67C6D4C1C8B041EEFBB8C1DBB@e1705>	<AANLkTi=1Fr2e-sz-n5Ds=q6KVntRzfzgrAQOgc0rCYgF@mail.gmail.com>	<F4AB52F0A27D46418900E2841CB5524B@e1705>	<AANLkTi=icT=Zw9xrrEcOyyozy5JFm7sXQjxTDxN3psZH@mail.gmail.com>
	<AANLkTinXuGfqOBdBcLSdBMaoMJk3JbWg18sfkKyyVofd@mail.gmail.com>
Message-ID: <4D47F9F1.3050401@coppice.org>

On 02/01/2011 06:07 PM, Anton VG wrote:
> The question that bugs me, that if some one have 1000 licenses for
> Asterisk g729 (10K$!) , and wants to switch over from asterisk to a
> FreeSWITCH, would one have to buy that 100 liceses again, or there is
> a kind of procedure to get the module for existing licenses? Most
> likely no way :) But it's worth to mention a problem, which might so
> much keeping one on the certain platform. And that is the case when
> one may consider, as owner of the licenses to go with the IPP (or
> other open implementation) without any law breakage.
>
Those software licence fees do not bestow the right to run any other 
G.729 software. You might feel you are on moral high ground using IPP 
after paying for asterisk licences, but this has no legal standing.

At one time you could get low volume patent licences from the G.729 
patent pool people, to run any code you like. This is no longer offered, 
and it was probably never of much value. I don't think you could get a 
similar low volume arrangement for the patents you need to licence which 
are outside the pool.

Steve



From Nabble at slickdeals.endjunk.com  Tue Feb  1 15:21:51 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 1 Feb 2011 04:21:51 -0800 (PST)
Subject: [Freeswitch-users] FIXED: execute_on_answer=send_dtmf 1
In-Reply-To: <1296183851075-5968605.post@n2.nabble.com>
References: <1296183851075-5968605.post@n2.nabble.com>
Message-ID: <1296562911205-5980707.post@n2.nabble.com>


I just changed the subject line by adding the word FIXED mainly because I got
this problem immediately resolved in other forum that only has a handful of
FS newbie users. Yeah, no need to press 1 no more for an incoming call to my
GV line. 

-----
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/FIXED-execute-on-answer-send-dtmf-1-tp5968605p5980707.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From avi at avimarcus.net  Tue Feb  1 15:31:07 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 1 Feb 2011 14:31:07 +0200
Subject: [Freeswitch-users] FIXED: execute_on_answer=send_dtmf 1
In-Reply-To: <1296562911205-5980707.post@n2.nabble.com>
References: <1296183851075-5968605.post@n2.nabble.com>
	<1296562911205-5980707.post@n2.nabble.com>
Message-ID: <AANLkTi=szvN+kRR4ffrmchMD1cP+7zzs_a0XDBjBuVNC@mail.gmail.com>

So how about telling us what you did to fix it? Share not just the problems,
but the fixes too :)
-Avi

On Tue, Feb 1, 2011 at 2:21 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
> I just changed the subject line by adding the word FIXED mainly because I
> got
> this problem immediately resolved in other forum that only has a handful of
> FS newbie users. Yeah, no need to press 1 no more for an incoming call to
> my
> GV line.
>
> -----
> 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/FIXED-execute-on-answer-send-dtmf-1-tp5968605p5980707.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  Tue Feb  1 16:42:39 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 1 Feb 2011 13:42:39 +0000
Subject: [Freeswitch-users] open g729
In-Reply-To: <AANLkTinXuGfqOBdBcLSdBMaoMJk3JbWg18sfkKyyVofd@mail.gmail.com>
References: <1296343422.2615.4.camel@gustavo-laptop>
	<AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com>
	<1296477877.3589.24.camel@gustavo-laptop>
	<AANLkTinq9AmN=ZqjGLNUxucw2nJXBLpNHKLTA5jLWfgj@mail.gmail.com>
	<9D7F812E93DC47A99F6A30638FB8092F@e1705>
	<AANLkTikkLnf4BHYgL1k8th=VWk44vpH5gRYSAFW5YfW1@mail.gmail.com>
	<A64FADA67C6D4C1C8B041EEFBB8C1DBB@e1705>
	<AANLkTi=1Fr2e-sz-n5Ds=q6KVntRzfzgrAQOgc0rCYgF@mail.gmail.com>
	<F4AB52F0A27D46418900E2841CB5524B@e1705>
	<AANLkTi=icT=Zw9xrrEcOyyozy5JFm7sXQjxTDxN3psZH@mail.gmail.com>
	<AANLkTinXuGfqOBdBcLSdBMaoMJk3JbWg18sfkKyyVofd@mail.gmail.com>
Message-ID: <AANLkTik6My-K6pEW6d75yy9AgoM8wkFiCtz2vA5wuG+i@mail.gmail.com>

No, because they're licensed from different sources.
It's no different to if I buy a G729 license for one softphone, I can't then
use it in another without buying another license.

A sangoma card pays for the license in the hardware costs and the drivers
are free though, so you can switch platform that way while keeping the
license.

"And that is the case when one may consider, as owner of the licenses to go
with the IPP (or other open implementation) without any law breakage."
They won't have a license for the IPP codec, only for the Asterisk one. So
it wouldn't be licensed. Even if you want to justify it to yourself that
way.

-Steve

On 1 February 2011 10:07, Anton VG <anton.vazir at gmail.com> wrote:

> The question that bugs me, that if some one have 1000 licenses for
> Asterisk g729 (10K$!) , and wants to switch over from asterisk to a
> FreeSWITCH, would one have to buy that 100 liceses again, or there is
> a kind of procedure to get the module for existing licenses? Most
> likely no way :) But it's worth to mention a problem, which might so
> much keeping one on the certain platform. And that is the case when
> one may consider, as owner of the licenses to go with the IPP (or
> other open implementation) without any law breakage.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Tue Feb  1 17:34:28 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Tue, 1 Feb 2011 17:34:28 +0300
Subject: [Freeswitch-users] How to prevent source channel hangup after
 uuid_bridge completion (outbound ESL)?
Message-ID: <AANLkTim7PoaVVCopbk6-wT1E04=pabKw9FjZaHCbLH5=@mail.gmail.com>

Hi

I'm trying to originate call from outbound ESL and then bridge it to
original channel where ESL is launched on.
I use api originate with &park(), then uuid_bridge. Everything works
as expected, but when second leg hangs up, first leg gets hangup too.
How can I prevent first leg from hanging up and continue to process it
in ESL?

Thanks in advance


-- 
Best regards,

Dmitry Sytchev,
IT Engineer


From cmrienzo at gmail.com  Tue Feb  1 18:11:25 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Tue, 1 Feb 2011 10:11:25 -0500
Subject: [Freeswitch-users] How to prevent source channel hangup after
 uuid_bridge completion (outbound ESL)?
In-Reply-To: <AANLkTim7PoaVVCopbk6-wT1E04=pabKw9FjZaHCbLH5=@mail.gmail.com>
References: <AANLkTim7PoaVVCopbk6-wT1E04=pabKw9FjZaHCbLH5=@mail.gmail.com>
Message-ID: <AANLkTik-rV6wwBjm3DsEDGVszEizKhgbQfYzN_TKvBhA@mail.gmail.com>

Set the hangup_after_bridge channel variable to false.


On Tue, Feb 1, 2011 at 9:34 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:

> Hi
>
> I'm trying to originate call from outbound ESL and then bridge it to
> original channel where ESL is launched on.
> I use api originate with &park(), then uuid_bridge. Everything works
> as expected, but when second leg hangs up, first leg gets hangup too.
> How can I prevent first leg from hanging up and continue to process it
> in ESL?
>
> 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
>
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From kbdfck at gmail.com  Tue Feb  1 18:17:18 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Tue, 1 Feb 2011 18:17:18 +0300
Subject: [Freeswitch-users] How to prevent source channel hangup after
 uuid_bridge completion (outbound ESL)?
In-Reply-To: <AANLkTim7PoaVVCopbk6-wT1E04=pabKw9FjZaHCbLH5=@mail.gmail.com>
References: <AANLkTim7PoaVVCopbk6-wT1E04=pabKw9FjZaHCbLH5=@mail.gmail.com>
Message-ID: <AANLkTinYt9Ma-bPu357Y+STdx4oAGNL=BSM=nL66yfPe@mail.gmail.com>

I tried to process CHANNEL_HANGUP of second channel to issue uuid_park
on source channel and this worked. But as far as I understand I should
react quickly on this event to prevent source channel hangup? Is there
a better way to ensure that source channel should not be destroyed?

2011/2/1 Dmitry Sytchev <kbdfck at gmail.com>:
> Hi
>
> I'm trying to originate call from outbound ESL and then bridge it to
> original channel where ESL is launched on.
> I use api originate with &park(), then uuid_bridge. Everything works
> as expected, but when second leg hangs up, first leg gets hangup too.
> How can I prevent first leg from hanging up and continue to process it
> in ESL?
>
> Thanks in advance
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>



-- 
Best regards,

Dmitry Sytchev,
IT Engineer


From u2nsam at gmail.com  Tue Feb  1 18:20:39 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 1 Feb 2011 20:50:39 +0530
Subject: [Freeswitch-users] port reg request pbm
In-Reply-To: <efc1e70e-c4b5-4571-bb34-affd29da49b0@winet.ch>
References: <AANLkTinHymqs1CA5fOgFHkd8PW51xGjcpX+jzo8w_g5A@mail.gmail.com>
	<efc1e70e-c4b5-4571-bb34-affd29da49b0@winet.ch>
Message-ID: <AANLkTingMkt9YeUOHrCOFDMk+0Z4AWzQUg07zEY=odd2@mail.gmail.com>

Its hapening only for polycom phones, also NDLB-connectile-dusfunction
dosent works.
Cisco , xlite & audiocodes works fine.

Regards
Sam



On Tue, Feb 1, 2011 at 4:41 PM, Bernhard Suttner
<bernhard.suttner at winet.ch>wrote:

> Sounds like NAT issue. Try around with the NAT possibilities of
> FreeSWITCH.
>
>
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Sam
> *Gesendet:* Dienstag, 1. Februar 2011 12:06
> *An:* FreeSWITCH Users Help
> *Betreff:* [Freeswitch-users] port reg request pbm
>
>
>
> hello,
>
> The device is sending request to my server ip x.y.z.v from  port 35666
>
> a.b.c.e:35666 -> x.y.z.v :5060
> Register
>
> but FS is not sending back to the same port on the device
>
> x.y.z.v:5060 -> a.b.c.e : 5060
> 401 unauthorized
>
> how this could be solved
>
> 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 infos at madovsky.org  Tue Feb  1 18:27:17 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 1 Feb 2011 10:27:17 -0500
Subject: [Freeswitch-users] minimum UDP packet size
References: <65064C9F3B344E4A9CAE4A9B63C318D3@e1705>
	<CDF5E00590C9BA4AB96BE7CDD28C86AE0787301836@FS1.wvds.local>
Message-ID: <C702A689FD9B41949808772070585A50@e1705>

No my question is minimum udp.
usually the max udp packet is around 65000kB,
David was talking about MTU which is different (Max Transmit Unit
which is usually 1500 over the net, but ins some internal network (as my cluster)
I use jumbo frames (9000) with 1GB network cards.
by default linux kernel UDP min packet is set to 4096.
I made some tests, from 1024 to 65000 and the server behavior looks like more
cool when packet is big, but I'm almost sure that it creates also more latency in case
of audio data. I'm finally happy to use exactly the same rate as MTU, performance
seems to be better in multitask.
  ----- Original Message ----- 
  From: Erik Dekkers 
  To: 'FreeSWITCH Users Help' 
  Sent: Tuesday, February 01, 2011 4:28 AM
  Subject: Re: [Freeswitch-users] minimum UDP packet size


  You probably mean a MAXimum size. If you set a minimum size it would always fragment.

  As far as I know it itsn't possible to set a maximum size of a UDP packet. If the UDP packets are getting too large you really should trim down SIP headers / Remove codecs from the SDP.

   

  There are some articles on the internet concerning MTU and SIP.

   

  Erik

   

  Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Madovsky
  Verzonden: maandag 31 januari 2011 18:29
  Aan: freeswitch-users at lists.freeswitch.org
  Onderwerp: [Freeswitch-users] minimum UDP packet size

   

  is it useful to set minimum UDP  pakcet to the

  MTU rate like 1500 and avoid any fragmentation ?

   

  Thanks



------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From brian at freeswitch.org  Tue Feb  1 18:35:36 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 1 Feb 2011 09:35:36 -0600
Subject: [Freeswitch-users] minimum UDP packet size
In-Reply-To: <C702A689FD9B41949808772070585A50@e1705>
References: <65064C9F3B344E4A9CAE4A9B63C318D3@e1705>
	<CDF5E00590C9BA4AB96BE7CDD28C86AE0787301836@FS1.wvds.local>
	<C702A689FD9B41949808772070585A50@e1705>
Message-ID: <076EE673-23EF-40C0-97B3-8D803E3FFF93@freeswitch.org>

While performance will be better with larger packets you're failing to realize that 1440 byte RTP packet would be 180ms of audio.  We only support up to 120ms per packet.  If you start to exceed the MTU most consumer routers will not pass any packets on UDP that exceed the MTU so your testing gains you nothing.

/b

On Feb 1, 2011, at 9:27 AM, Madovsky wrote:

> No my question is minimum udp.
> usually the max udp packet is around 65000kB,
> David was talking about MTU which is different (Max Transmit Unit
> which is usually 1500 over the net, but ins some internal network (as my cluster)
> I use jumbo frames (9000) with 1GB network cards.
> by default linux kernel UDP min packet is set to 4096.
> I made some tests, from 1024 to 65000 and the server behavior looks like more
> cool when packet is big, but I'm almost sure that it creates also more latency in case
> of audio data. I'm finally happy to use exactly the same rate as MTU, performance
> seems to be better in multitask.

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From david.ponzone at ipeva.fr  Tue Feb  1 18:44:12 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 1 Feb 2011 16:44:12 +0100
Subject: [Freeswitch-users] minimum UDP packet size
In-Reply-To: <C702A689FD9B41949808772070585A50@e1705>
References: <65064C9F3B344E4A9CAE4A9B63C318D3@e1705>
	<CDF5E00590C9BA4AB96BE7CDD28C86AE0787301836@FS1.wvds.local>
	<C702A689FD9B41949808772070585A50@e1705>
Message-ID: <364234C2-7E4F-4EFC-8ABD-CD9848BF4C77@ipeva.fr>

65MB ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 01/02/2011 ? 16:27, Madovsky a ?crit :

> No my question is minimum udp.
> usually the max udp packet is around 65000kB,
> David was talking about MTU which is different (Max Transmit Unit
> which is usually 1500 over the net, but ins some internal network (as my cluster)
> I use jumbo frames (9000) with 1GB network cards.
> by default linux kernel UDP min packet is set to 4096.
> I made some tests, from 1024 to 65000 and the server behavior looks like more
> cool when packet is big, but I'm almost sure that it creates also more latency in case
> of audio data. I'm finally happy to use exactly the same rate as MTU, performance
> seems to be better in multitask.
> ----- Original Message -----
> From: Erik Dekkers
> To: 'FreeSWITCH Users Help'
> Sent: Tuesday, February 01, 2011 4:28 AM
> Subject: Re: [Freeswitch-users] minimum UDP packet size
> 
> You probably mean a MAXimum size. If you set a minimum size it would always fragment.
> As far as I know it itsn?t possible to set a maximum size of a UDP packet. If the UDP packets are getting too large you really should trim down SIP headers / Remove codecs from the SDP.
>  
> There are some articles on the internet concerning MTU and SIP.
>  
> Erik
>  
> Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org]Namens Madovsky
> Verzonden: maandag 31 januari 2011 18:29
> Aan: freeswitch-users at lists.freeswitch.org
> Onderwerp: [Freeswitch-users] minimum UDP packet size
>  
> is it useful to set minimum UDP  pakcet to the
> MTU rate like 1500 and avoid any fragmentation ?
>  
> 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 anthony.minessale at gmail.com  Tue Feb  1 18:58:23 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 1 Feb 2011 09:58:23 -0600
Subject: [Freeswitch-users] Choppy VM when using loopback (on Debian
	lenny)
In-Reply-To: <AANLkTimhOKmy_yupGtA2HBY_8wXtQKZnDYJ9Ye76bhzq@mail.gmail.com>
References: <4D42B8EE.7030208@estation.dk>
	<7C6F3FC7-445F-4157-BD59-D76A705E4CE7@carmickle.com>
	<9894FEC6-E451-4993-A85A-2CD910E916B1@ipeva.fr>
	<AANLkTim+aBjRiAvxDUF1O1vut_CJpqgct2OqCNVxi0q4@mail.gmail.com>
	<AANLkTiktckovGv4Ksnn_56EzpvTTSnfG+iD1vPX1_4RM@mail.gmail.com>
	<4D466CAE.7030402@estation.dk>
	<AANLkTimhOKmy_yupGtA2HBY_8wXtQKZnDYJ9Ye76bhzq@mail.gmail.com>
Message-ID: <AANLkTiktQU7kOG_22nG=_oEHSrBad667jcTzBs3n2nQv@mail.gmail.com>

Are you saying you have better results on that version than you do on
the latest?
What conditions do you have that cause you trouble, what is the
endpoint on the other side.

If the last commit to mod_loopback intended to improve audio quality
actually makes it worse I need to investigate it.


On Tue, Feb 1, 2011 at 5:53 AM, Dan Lane <null at invalid.name> wrote:
> FWIW we've been also been having audio issues with loopback recently
> on EC2 (with a 1000Hz kernel).
>
> We worked around it in the short term by reverting mod_loopback to
> git-4c5426f during the build process.
>
> For anyone else who wants to try this just run "git checkout 4c5426f"
> in src/mod/endpoints/mod_loopback then build as usual. This is NOT a
> long term solution though.
>
> On Mon, Jan 31, 2011 at 8:02 AM, ?yvind Albrigtsen <oa at estation.dk> wrote:
>> Thanks for all your feedback. I'll keep on trying and inform you what
>> worked for me.
>>
>>
>> Regards
>> Oyvind
>>
>> On 2011-01-29 21:48, Anthony Minessale wrote:
>>> Everyone should try latest GIT before pondering any further because I
>>> added a patch like 2 days ago to adress this issue.
>>>
>>>
>>> On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank at telonium.com> ?wrote:
>>>
>>>> Yeah. I, too, don't see the correlation between the NTP and hw timer..
>>>> I am not familiar with the timer_test command and what it's measuring, but
>>>> of the 50 tests it ran, min is 19089 and max is 20713.
>>>> Frank
>>>>
>>>>
>>>> On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone at ipeva.fr>
>>>> wrote:
>>>>
>>>>> Frank,
>>>>> I fail to see the relationship between the hw timer and NTP.
>>>>> Can you please elaborate ?
>>>>> David Ponzone ?Direction Technique
>>>>> email: david.ponzone at ipeva.fr
>>>>> tel: ? ? ?01 74 03 18 97
>>>>> gsm: ? 06 66 98 76 34
>>>>> Service Client IPeva
>>>>> tel: ? ? ?0811 46 26 26
>>>>> www.ipeva.fr ?- ? www.ipeva-studio.com
>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>
>>>>>
>>>>>
>>>>> Le 28/01/2011 ? 19:49, Frank Carmickle a ?crit :
>>>>>
>>>>> Hi
>>>>>
>>>>> On Jan 28, 2011, at 7:39 AM, ?yvind Albrigtsen wrote:
>>>>>
>>>>> Hi,
>>>>>
>>>>> I'm using latest git-version of Freeswitch, and when I go to voicemail
>>>>>
>>>>> when calling a number the sound playback is choppy and it skips some of
>>>>>
>>>>> the digits in the number I called.
>>>>>
>>>>> What kind of results do you get from timer_test at the fs_cli? ?Are you
>>>>> running on hardware or are you virtualized? ?What is your clock source set
>>>>> to and what are your available clock source options? ?See
>>>>> /sys/devices/system/clocksource/clocksource0/available_clocksource and
>>>>> /sys/devices/system/clocksource/clocksource0/current_clocksource. ?I am
>>>>> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to
>>>>> hang at 19998/19999 which works very well for me. ?When I was having problem
>>>>> it was reporting numbers all over the map from 17400 to 22600 with lots of
>>>>> randomness in between. ?I have my clocksource set to jiffies and xen
>>>>> independent wallclock set to 1. ?Of course at that point you need to have
>>>>> ntp running against a bunch of servers to drive your clock nice and steady.
>>>>> ? I know my set up is probably a lot different than yours but I thought I'd
>>>>> toss it out there to show that some of the harshest conditions can be dealt
>>>>> with and don't give up trying. ?If you are running on hardware with a cpu
>>>>> that doesn't have constant_tsc then you might have some problems. ?Just play
>>>>> with the different timer options until you find the one that works.
>>>>>
>>>>> HTH
>>>>> --FC
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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
>>>>
>>>>
>>>>
>>>
>>>
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Tue Feb  1 19:12:48 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 1 Feb 2011 11:12:48 -0500
Subject: [Freeswitch-users] minimum UDP packet size
References: <65064C9F3B344E4A9CAE4A9B63C318D3@e1705><CDF5E00590C9BA4AB96BE7CDD28C86AE0787301836@FS1.wvds.local><C702A689FD9B41949808772070585A50@e1705>
	<364234C2-7E4F-4EFC-8ABD-CD9848BF4C77@ipeva.fr>
Message-ID: <57EB680D59A74110AAE9F670BEB9D2D5@e1705>

:) no 65000 bytes
  ----- Original Message ----- 
  From: David Ponzone 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, February 01, 2011 10:44 AM
  Subject: Re: [Freeswitch-users] minimum UDP packet size


  65MB ?


  David Ponzone  Direction Technique
  email: david.ponzone at ipeva.fr
  tel:      01 74 03 18 97
  gsm:   06 66 98 76 34


  Service Client IPeva
  tel:      0811 46 26 26
  www.ipeva.fr  -   www.ipeva-studio.com


  Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.







  Le 01/02/2011 ? 16:27, Madovsky a ?crit :


    No my question is minimum udp.
    usually the max udp packet is around 65000kB,
    David was talking about MTU which is different (Max Transmit Unit
    which is usually 1500 over the net, but ins some internal network (as my cluster)
    I use jumbo frames (9000) with 1GB network cards.
    by default linux kernel UDP min packet is set to 4096.
    I made some tests, from 1024 to 65000 and the server behavior looks like more
    cool when packet is big, but I'm almost sure that it creates also more latency in case
    of audio data. I'm finally happy to use exactly the same rate as MTU, performance
    seems to be better in multitask.
      ----- Original Message -----
      From: Erik Dekkers
      To: 'FreeSWITCH Users Help'
      Sent: Tuesday, February 01, 2011 4:28 AM
      Subject: Re: [Freeswitch-users] minimum UDP packet size


      You probably mean a MAXimum size. If you set a minimum size it would always fragment.
      As far as I know it itsn?t possible to set a maximum size of a UDP packet. If the UDP packets are getting too large you really should trim down SIP headers / Remove codecs from the SDP.

      There are some articles on the internet concerning MTU and SIP.

      Erik

      Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org]Namens Madovsky
      Verzonden: maandag 31 januari 2011 18:29
      Aan: freeswitch-users at lists.freeswitch.org
      Onderwerp: [Freeswitch-users] minimum UDP packet size

      is it useful to set minimum UDP  pakcet to the
      MTU rate like 1500 and avoid any fragmentation ?

      Thanks



--------------------------------------------------------------------------



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      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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      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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------------------------------------------------------------------------------


  _______________________________________________
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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Tue Feb  1 19:15:50 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 1 Feb 2011 11:15:50 -0500
Subject: [Freeswitch-users] minimum UDP packet size
References: <65064C9F3B344E4A9CAE4A9B63C318D3@e1705><CDF5E00590C9BA4AB96BE7CDD28C86AE0787301836@FS1.wvds.local><C702A689FD9B41949808772070585A50@e1705>
	<076EE673-23EF-40C0-97B3-8D803E3FFF93@freeswitch.org>
Message-ID: <D8D9A721B15545ECAEB2CA794B901E7E@e1705>

but UDP packet is independent of MTU.
even if you put your UDP packet higher than MTU it will be fragment as MTU size...

  ----- Original Message ----- 
  From: Brian West 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, February 01, 2011 10:35 AM
  Subject: Re: [Freeswitch-users] minimum UDP packet size


  While performance will be better with larger packets you're failing to realize that 1440 byte RTP packet would be 180ms of audio.  We only support up to 120ms per packet.  If you start to exceed the MTU most consumer routers will not pass any packets on UDP that exceed the MTU so your testing gains you nothing.


  /b


  On Feb 1, 2011, at 9:27 AM, Madovsky wrote:


    No my question is minimum udp.
    usually the max udp packet is around 65000kB,
    David was talking about MTU which is different (Max Transmit Unit
    which is usually 1500 over the net, but ins some internal network (as my cluster)
    I use jumbo frames (9000) with 1GB network cards.
    by default linux kernel UDP min packet is set to 4096.
    I made some tests, from 1024 to 65000 and the server behavior looks like more
    cool when packet is big, but I'm almost sure that it creates also more latency in case
    of audio data. I'm finally happy to use exactly the same rate as MTU, performance
    seems to be better in multitask.




------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing 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 jeff at jefflenk.com  Tue Feb  1 19:19:17 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 1 Feb 2011 08:19:17 -0800 (PST)
Subject: [Freeswitch-users] Choppy VM when using loopback (on Debian
	lenny)
In-Reply-To: <AANLkTiktQU7kOG_22nG=_oEHSrBad667jcTzBs3n2nQv@mail.gmail.com>
References: <4D42B8EE.7030208@estation.dk>
	<7C6F3FC7-445F-4157-BD59-D76A705E4CE7@carmickle.com>
	<9894FEC6-E451-4993-A85A-2CD910E916B1@ipeva.fr>
	<AANLkTim+aBjRiAvxDUF1O1vut_CJpqgct2OqCNVxi0q4@mail.gmail.com>
	<AANLkTiktckovGv4Ksnn_56EzpvTTSnfG+iD1vPX1_4RM@mail.gmail.com>
	<4D466CAE.7030402@estation.dk>
	<AANLkTimhOKmy_yupGtA2HBY_8wXtQKZnDYJ9Ye76bhzq@mail.gmail.com>
	<AANLkTiktQU7kOG_22nG=_oEHSrBad667jcTzBs3n2nQv@mail.gmail.com>
Message-ID: <1296577157282-5981619.post@n2.nabble.com>


Yes under windows the most recent commit causes a severe deterioration in the
audio. I had thought it may only be a problem with windows but I didnt
really know. I have added some debug to
http://jira.freeswitch.org/browse/FS-3011 with some timing information under
windows that illustrates the problem. I can provide more information if
needed.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Choppy-VM-when-using-loopback-on-Debian-lenny-tp5969724p5981619.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Tue Feb  1 19:20:12 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 1 Feb 2011 10:20:12 -0600
Subject: [Freeswitch-users] Developer Dinner Coming Up - Help Feed
 TheFreeSWITCH Team!
In-Reply-To: <AANLkTimXcAQ1ZZee3ZDwQHWrTC+FABqf3DQrghstUeWd@mail.gmail.com>
References: <AANLkTimZjT2SQ=Go10JwLrXJEmjQt+iNZQrUKVHPBgj=@mail.gmail.com>
	<1296526216699-5979632.post@n2.nabble.com>
	<BF9E9F3DB6C3408991502E1957F24674@e1705>
	<AANLkTimXcAQ1ZZee3ZDwQHWrTC+FABqf3DQrghstUeWd@mail.gmail.com>
Message-ID: <AANLkTikOud1ER0AD-nvKRRiM-no=9wUcAWn8w5Gqge5v@mail.gmail.com>

You are thanked now on freeswitch.com and for all your help Seven.
If I missed anyone else, it wasn't intentional I am going from the
UDRP hint in the paypal message.
Just let me know.

On Mon, Jan 31, 2011 at 10:20 PM, Seven Du <dujinfang at gmail.com> wrote:
> $20 confirmation No. 18U27974AX8119748 . Enjoy!
>
> It's almost 3 years since I'v joined the community. I started using FS
> since then and I like the way FS works and so does the community.
> People helped me a lot from the basics to deeper VoIP knowledge.
> Anthony, Mike and other members also used their owen time helped me to
> solve my problems with login into our box.
>
> I didn't make a lot of money from FS directly, but FS allows us to do
> funny things and that's why I like it. By the spirit of thanks, I also
> reported some bugs and submitted some patches. Also there's $10 in
> freeswitch.com (Didn't get thanked I think I didn't left proper
> messages on donating). Also, I know that money is not the only way to
> help. I also created http://www.freeswitch.org.cn which is 1yr old and
> maintains a freeswitch-cn google groups. But due to the policy of
> China Internet, only a few people can access google groups and
> consequently we didn't get strong enough. The website is deployed on
> Heroku which is on EC2 and we have difficulties to access it in the
> last month. Anyway, those won't stop me march ahead.
>
> Sorry a bit off topic, just want to say thanks hope not too long.
>
> Thanks again,
> Seven.
>
> On Tue, Feb 1, 2011 at 11:03 AM, Madovsky <infos at madovsky.org> wrote:
>> I can make delicious pizzas, lasagnes and sushi...
>>
>> ----- Original Message -----
>> From: "mazilo" <Nabble at slickdeals.endjunk.com>
>> To: <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, January 31, 2011 9:10 PM
>> Subject: Re: [Freeswitch-users] Developer Dinner Coming Up - Help Feed
>> TheFreeSWITCH Team!
>>
>>
>>>
>>> How about donate food?
>>>
>>> -----
>>> 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/Developer-Dinner-Coming-Up-Help-Feed-The-FreeSWITCH-Team-tp5979347p5979632.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
>>
>
>
>
> --
> 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
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 Feb  1 19:20:51 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Tue, 1 Feb 2011 19:20:51 +0300
Subject: [Freeswitch-users] How to prevent source channel hangup after
 uuid_bridge completion (outbound ESL)?
In-Reply-To: <AANLkTik-rV6wwBjm3DsEDGVszEizKhgbQfYzN_TKvBhA@mail.gmail.com>
References: <AANLkTim7PoaVVCopbk6-wT1E04=pabKw9FjZaHCbLH5=@mail.gmail.com>
	<AANLkTik-rV6wwBjm3DsEDGVszEizKhgbQfYzN_TKvBhA@mail.gmail.com>
Message-ID: <AANLkTimhBqbp5upa-me21AsWNYACr14FZQkStRK=sYXW@mail.gmail.com>

Thanks, will try this. For some reason I forgot about this, although
using it in dialplan :)))

2011/2/1 Christopher Rienzo <cmrienzo at gmail.com>:
> Set the hangup_after_bridge channel variable to false.
>
>
> On Tue, Feb 1, 2011 at 9:34 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>
>> Hi
>>
>> I'm trying to originate call from outbound ESL and then bridge it to
>> original channel where ESL is launched on.
>> I use api originate with &park(), then uuid_bridge. Everything works
>> as expected, but when second leg hangs up, first leg gets hangup too.
>> How can I prevent first leg from hanging up and continue to process it
>> in ESL?
>>
>> 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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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


From anton.vazir at gmail.com  Tue Feb  1 19:26:40 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Tue, 1 Feb 2011 21:26:40 +0500
Subject: [Freeswitch-users] open g729
In-Reply-To: <AANLkTik6My-K6pEW6d75yy9AgoM8wkFiCtz2vA5wuG+i@mail.gmail.com>
References: <1296343422.2615.4.camel@gustavo-laptop>
	<AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com>
	<1296477877.3589.24.camel@gustavo-laptop>
	<AANLkTinq9AmN=ZqjGLNUxucw2nJXBLpNHKLTA5jLWfgj@mail.gmail.com>
	<9D7F812E93DC47A99F6A30638FB8092F@e1705>
	<AANLkTikkLnf4BHYgL1k8th=VWk44vpH5gRYSAFW5YfW1@mail.gmail.com>
	<A64FADA67C6D4C1C8B041EEFBB8C1DBB@e1705>
	<AANLkTi=1Fr2e-sz-n5Ds=q6KVntRzfzgrAQOgc0rCYgF@mail.gmail.com>
	<F4AB52F0A27D46418900E2841CB5524B@e1705>
	<AANLkTi=icT=Zw9xrrEcOyyozy5JFm7sXQjxTDxN3psZH@mail.gmail.com>
	<AANLkTinXuGfqOBdBcLSdBMaoMJk3JbWg18sfkKyyVofd@mail.gmail.com>
	<AANLkTik6My-K6pEW6d75yy9AgoM8wkFiCtz2vA5wuG+i@mail.gmail.com>
Message-ID: <AANLkTimSNZpM1CXkDRx9R7_uuOSoxDxDBZF8OCgqtzH8@mail.gmail.com>

Considering the above, noone can say "Purchase a LICENSE", but only a
"Licensed SOFTWARE"


From shoney at noblesys.com  Tue Feb  1 19:38:37 2011
From: shoney at noblesys.com (Saji Honey)
Date: Tue, 1 Feb 2011 11:38:37 -0500 (EST)
Subject: [Freeswitch-users] (no subject)
Message-ID: <c8dd01cbc22e$4c24dec0$e46e9c40$@com>


CONFIDENTIAL NOTICE : If you have received this email in error, 
please immediately notify the sender by email at the address 
shown above. This email may contain confidential or legally 
privileged information that is intended only for the use of the 
individual or entity named in this email. If you are not the 
intended recipient, you are hereby notified that any 
disclosure, copying, distribution or reliance upon the contents 
of this email is strictly prohibited. Please delete from your 
files if you are not the intended recipient. Thank you for your 
compliance.


From mstockton at harqen.com  Tue Feb  1 19:10:55 2011
From: mstockton at harqen.com (Matt Stockton)
Date: Tue, 1 Feb 2011 10:10:55 -0600
Subject: [Freeswitch-users] Problem with missing / duplicated DTMF
Message-ID: <AANLkTin+-NCnROr5ZE9GYoao_qVP_-2K7mWqhvJ+dkd3@mail.gmail.com>

I have having trouble with both missing and duplicated DTMF in Freeswitch.

Here are the steps of how I am using it:

1. Leg A - I am calling out from my Freeswitch instance (through iCall), and
I am calling an iCall number that is also connected to the same Freeswitch
instance.

2. Leg B - The above call is routed through iCall and then answered by the
same Freeswitch instance.

3. On Leg B, I play a file and attempt to get DTMF in a lua script. Here is
the code of interest:

callPin = session:playAndGetDigits(1, 10, 4, 30000, "#","/tmp/cw_17.wav",
"", "\\d+");


4. On Leg A, I send DTMF information in a lua script. Here is the code of
interest. I initiate a delay between each digit:

local newPin = "";

for i = 1, string.len(pin) do

  newPin = newPin .. string.sub(pin, i, i) .. "W";

end

session:execute("send_dtmf", newPin .. "#@200");


** Note that there is a session:sleep on Leg A before I send the DTMF to
make sure i don't send it too early **


The problem is that the recognized DTMF on Leg B is wrong about 30% of the
time. For example, if Leg A enters: 22063083, Leg B will get the DTMF digits
222063083. This is an example of duplication, but I have also experienced
missing DTMF codes (and an occasional wrong code completely)


I have messed with a bunch of DTMF settings in hopes of fixing this issue,
but I cannot seem to find something that is reliable 100% of the time.


_____________________________

Here are the DTMF settings I have looked at / messed with. I've tried
various values for the dtmf-duration in the config (and in the send_dtmf
command above)


    <!-- This is in external.xml -->

    <param name="dtmf-duration" value="500"/>

    <param name="dtmf-type" value="rfc2833"/>

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

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

    <!-- This is in vars.xml -->
    <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU"/>
    <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU"/>

________________________

I have run fs_cli with event logging and the DTMF events that Freeswitch
gets do correlate to the wrong value (e.g. the duplication / missing digits
is noticable in the Freeswitch events as well).

Also, I am not running any dtmf-related applications on the session before I
give control to the lua scripts (e.g. not running start_dtmf)

Has anyone experienced this type of issue? Or know what I can do to resolve
it? My next step was going to be trying this against another provider
besides iCall, but I figured I would see if anyone has encountered a similar
problem before. Any help is appreciated.

Thanks
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From eric at bmcrministries.org  Tue Feb  1 19:17:24 2011
From: eric at bmcrministries.org (Eric Michel)
Date: Tue, 1 Feb 2011 09:17:24 -0700
Subject: [Freeswitch-users] Route incoming analog calls.
Message-ID: <AANLkTikmXVZqVAJZ-Av21ohRUnGbmz0Fs9z=kZCK+kp3@mail.gmail.com>

I've got a Sangoma A200 with six FXO ports and two FXS.  I can call out just
fine, but I'm having trouble figuring out how to route incoming calls.
 We've got five POTS lines and all I've been able to do is route all
incoming calls to a specific extension.  How do I go about routing
individual POTS lines to different groups/extensions?

<extension name="analog test2" continue="true">
    <condition field="source" expression="mod_freetdm">
     <action application="set" data="domain_name=$${domain}"/>
       <action application="set" data="transfer_ringback=$${us-ring}"/>
       <action application="answer"/>
       <action application="transfer" data="1000 XML default"/>
    </condition>
 </extension>

That is my current conf/public/inbound dial plan.  Shouldn't I be able to
add a second condition that tests what FXO port is receiving the call and
allow me to route it accordingly?  If yes, could I get an example?  The
answer may be really simple, I've just been unable to find it.

Thanks,
Eric
-------------- next part --------------
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From anthony.minessale at gmail.com  Tue Feb  1 20:28:56 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 1 Feb 2011 11:28:56 -0600
Subject: [Freeswitch-users] Problem with missing / duplicated DTMF
In-Reply-To: <AANLkTin+-NCnROr5ZE9GYoao_qVP_-2K7mWqhvJ+dkd3@mail.gmail.com>
References: <AANLkTin+-NCnROr5ZE9GYoao_qVP_-2K7mWqhvJ+dkd3@mail.gmail.com>
Message-ID: <AANLkTinNkBim5hPgEX0q95j39G_y6ZMYAQXmkWPjrfQp@mail.gmail.com>

You did not include what version of FS you are using.


On Tue, Feb 1, 2011 at 10:10 AM, Matt Stockton <mstockton at harqen.com> wrote:
> I have having trouble with both missing and duplicated DTMF in Freeswitch.
> Here are the steps of how I am using it:
> 1. Leg A - I am calling out from my Freeswitch instance (through iCall), and
> I am calling an iCall number that is also connected to the same Freeswitch
> instance.
> 2. Leg B - The above call is routed through iCall and then answered by the
> same Freeswitch instance.
> 3. On Leg B, I play a file and attempt to get DTMF in a lua script. Here is
> the code of interest:
>
> callPin = session:playAndGetDigits(1, 10, 4, 30000, "#","/tmp/cw_17.wav",
> "", "\\d+");
>
> 4. On Leg A, I send DTMF information in a lua script. Here is the code of
> interest. I initiate a delay between each digit:
>
> local newPin = "";
>
> for i = 1, string.len(pin) do
>
> ??newPin = newPin .. string.sub(pin, i, i) .. "W";
>
> end
>
> session:execute("send_dtmf", newPin .. "#@200");
>
> ** Note that there is a session:sleep on Leg A before I send the DTMF to
> make sure i don't send it too early **
>
> The problem is that the recognized DTMF on Leg B is wrong about 30% of the
> time. For example, if Leg A enters:?22063083, Leg B will get the DTMF digits
> 222063083. This is an example of duplication, but I have also experienced
> missing DTMF codes (and an occasional wrong code completely)
>
> I have messed with a bunch of DTMF settings in hopes of fixing this issue,
> but I cannot seem to find something that is reliable 100% of the time.
>
> _____________________________
>
> Here are the DTMF settings I have looked at / messed with. I've tried
> various values for the dtmf-duration in the config (and in the send_dtmf
> command above)
>
> ?? ?<!-- This is in external.xml -->
>
> ?? ?<param name="dtmf-duration" value="500"/>
>
> ?? ?<param name="dtmf-type" value="rfc2833"/>
>
> ?? ?<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
>
> ?? ?<param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
>
> ?? ?<param name="inbound-codec-negotiation" value="greedy"/>
> ?? ?<!-- This is in vars.xml -->
> ?? ?<X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU"/>
> ?? ?<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU"/>
> ________________________
> I have run fs_cli with event logging and the DTMF events that Freeswitch
> gets do correlate to the wrong value (e.g. the duplication / missing digits
> is noticable in the Freeswitch events as well).
> Also, I am not running any dtmf-related applications on the session before I
> give control to the lua scripts (e.g. not running start_dtmf)
> Has anyone experienced this type of issue? Or know what I can do to resolve
> it? My next step was going to be trying this against another provider
> besides iCall, but I figured I would see if anyone has encountered a similar
> problem before. Any help is appreciated.
> 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 curriegrad2004 at gmail.com  Tue Feb  1 20:43:22 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 1 Feb 2011 09:43:22 -0800
Subject: [Freeswitch-users] open g729
In-Reply-To: <AANLkTimSNZpM1CXkDRx9R7_uuOSoxDxDBZF8OCgqtzH8@mail.gmail.com>
References: <1296343422.2615.4.camel@gustavo-laptop>
	<AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com>
	<1296477877.3589.24.camel@gustavo-laptop>
	<AANLkTinq9AmN=ZqjGLNUxucw2nJXBLpNHKLTA5jLWfgj@mail.gmail.com>
	<9D7F812E93DC47A99F6A30638FB8092F@e1705>
	<AANLkTikkLnf4BHYgL1k8th=VWk44vpH5gRYSAFW5YfW1@mail.gmail.com>
	<A64FADA67C6D4C1C8B041EEFBB8C1DBB@e1705>
	<AANLkTi=1Fr2e-sz-n5Ds=q6KVntRzfzgrAQOgc0rCYgF@mail.gmail.com>
	<F4AB52F0A27D46418900E2841CB5524B@e1705>
	<AANLkTi=icT=Zw9xrrEcOyyozy5JFm7sXQjxTDxN3psZH@mail.gmail.com>
	<AANLkTinXuGfqOBdBcLSdBMaoMJk3JbWg18sfkKyyVofd@mail.gmail.com>
	<AANLkTik6My-K6pEW6d75yy9AgoM8wkFiCtz2vA5wuG+i@mail.gmail.com>
	<AANLkTimSNZpM1CXkDRx9R7_uuOSoxDxDBZF8OCgqtzH8@mail.gmail.com>
Message-ID: <AANLkTinLARHTFP3EX1ow03NS-uyn4+FZTdbyym8ms4cF@mail.gmail.com>

Simple solution to this g.729 patent non-sense: Speex.

'nuff said.

On Tue, Feb 1, 2011 at 8:26 AM, Anton VG <anton.vazir at gmail.com> wrote:
> Considering the above, noone can say "Purchase a LICENSE", but only a
> "Licensed SOFTWARE"
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 fdelawarde at wirelessmundi.com  Tue Feb  1 20:49:03 2011
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Tue, 01 Feb 2011 18:49:03 +0100
Subject: [Freeswitch-users] execute_on_ring executing on answer
In-Reply-To: <1296306637.8986.185.camel@luna.tc.commsmundi.com>
References: <1296215265.8986.147.camel@luna.tc.commsmundi.com>
	<21B4FB4AAC2E428E938F99986CA09165@e1705>
	<1296235095.8986.167.camel@luna.tc.commsmundi.com>
	<0E96944DB29F4CEB9ACA5994D19AB0D6@e1705>
	<AANLkTikHB-QnDyT0WPsEML8yv4AACnYnHrE37gpRsLS2@mail.gmail.com>
	<1296306637.8986.185.camel@luna.tc.commsmundi.com>
Message-ID: <1296582543.5245.87.camel@luna.tc.commsmundi.com>

Since apparently noone reproduces this, it must be a configuration error
from my part.

Any hints of where I could start looking to resolve this issue?

Thanks,
Fran?ois.


On Sat, 2011-01-29 at 14:10 +0100, Fran?ois Delawarde wrote:
> Nice to know, but in that case the destination actually rings (180).
> 
> See commented log:
> http://pastebin.freeswitch.org/15168
> 
> Fran?ois.
> 
> On Fri, 2011-01-28 at 12:34 -0600, Anthony Minessale wrote:
> > if it never rings, answer will still trigger it.
> > 
> > 
> > On Fri, Jan 28, 2011 at 11:24 AM, Madovsky <infos at madovsky.org> wrote:
> > > ah ok, maybe a wiki update would be useful.
> > >
> > >
> > >
> > > ----- Original Message -----
> > > From: "Fran?ois Delawarde" <fdelawarde at wirelessmundi.com>
> > > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > > Sent: Friday, January 28, 2011 12:18 PM
> > > Subject: Re: [Freeswitch-users] execute_on_ring executing on answer
> > >
> > >
> > >> It's some cool feature made by Anthony that allows me to specify the
> > >> separator.
> > >>
> > >> in ^^:PCMA:G722
> > >> ^^: means the separator is now : instead of ,
> > >>
> > >> Useful in the [] or {} case because the coma is already used to separate
> > >> variables.
> > >>
> > >> Fran?ois.
> > >>
> > >> On Fri, 2011-01-28 at 12:12 -0500, Madovsky wrote:
> > >>> what means the ^^ in your codec string ?
> > >>>
> > >>> ----- Original Message -----
> > >>> From: "Fran?ois Delawarde" <fdelawarde at wirelessmundi.com>
> > >>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > >>> Sent: Friday, January 28, 2011 6:47 AM
> > >>> Subject: [Freeswitch-users] execute_on_ring executing on answer
> > >>>
> > >>>
> > >>> > Hi,
> > >>> >
> > >>> > Doing some testing with this morning's git (Fri Jan 28) I just found
> > >>> > out
> > >>> > that the execute_on_ring application runs when the destination answers
> > >>> > instead of when it rings.
> > >>> >
> > >>> > So far, I can't seem to find out the reason. Could it be some
> > >>> > configuration issue?
> > >>> >
> > >>> >
> > >>> > Here a call log showing the phenomenon with a simple bridge:
> > >>> >
> > >>> > <action application="bridge"
> > >>> > data="[execute_on_ring=info,absolute_codec_string=^^:PCMA:G722]sofia/192.168.10.1/sip:2103 at 192.168.10.22:5060"/>
> > >>> >
> > >>> > http://pastebin.freeswitch.org/15168
> > >>> >
> > >>> >
> > >>> > Thanks,
> > >>> > Fran?ois.
> > >>> >
> > >>> >
> > >>> > _______________________________________________
> > >>> > FreeSWITCH-users mailing list
> > >>> > FreeSWITCH-users at lists.freeswitch.org
> > >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >>> > http://www.freeswitch.org
> > >>> >
> > >>>
> > >>>
> > >>> _______________________________________________
> > >>> FreeSWITCH-users mailing list
> > >>> FreeSWITCH-users at lists.freeswitch.org
> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >>> http://www.freeswitch.org
> > >>
> > >>
> > >> _______________________________________________
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> http://www.freeswitch.org
> > >>
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > 
> > 
> > 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb  1 20:50:16 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 1 Feb 2011 09:50:16 -0800
Subject: [Freeswitch-users] Route incoming analog calls.
In-Reply-To: <AANLkTikmXVZqVAJZ-Av21ohRUnGbmz0Fs9z=kZCK+kp3@mail.gmail.com>
References: <AANLkTikmXVZqVAJZ-Av21ohRUnGbmz0Fs9z=kZCK+kp3@mail.gmail.com>
Message-ID: <AANLkTimYZiH6kc49TZZvN6UTofEygu_3XgZaDMQ7YsYr@mail.gmail.com>

You'll have to read the documentation on the Sangoma card itself. I'm
pretty sure there is a variable where the card's driver sets and
passes it on to Freeswitch's switching component.

On Tue, Feb 1, 2011 at 8:17 AM, Eric Michel <eric at bmcrministries.org> wrote:
> I've got a Sangoma A200 with six FXO ports and two FXS. ?I can call out just
> fine, but I'm having trouble figuring out how to route incoming calls.
> ?We've got five POTS lines and all I've been able to do is route all
> incoming calls to a specific?extension. ?How do I go about routing
> individual POTS lines to different groups/extensions?
> <extension name="analog test2" continue="true">
> ?? ?<condition field="source" expression="mod_freetdm">
> ?? ? <action application="set" data="domain_name=$${domain}"/>
> ?? ? ? <action application="set" data="transfer_ringback=$${us-ring}"/>
> ?? ? ? <action application="answer"/>
> ?? ? ? <action application="transfer" data="1000 XML default"/>
> ?? ?</condition>
> ?</extension>
> That is my current conf/public/inbound dial plan. ?Shouldn't I be able to
> add a second condition that tests what FXO port is receiving the call and
> allow me to route it accordingly? ?If yes, could I get an example? ?The
> answer may be really simple, I've just been unable to find it.
> 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
>
>


From curriegrad2004 at gmail.com  Tue Feb  1 20:55:59 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 1 Feb 2011 09:55:59 -0800
Subject: [Freeswitch-users] freeswitch.com is returned to us!
In-Reply-To: <AANLkTimVnELkhvJF860CKc-PzkYZijUp67RR-fh_27-t@mail.gmail.com>
References: <AANLkTi=BciYcFugPySe3QU7pK-=avKSpkOUiq119UCYP@mail.gmail.com>
	<5FB9599FB0CA4FA496A87C65DAFB9D53@e1705>
	<AANLkTik1QwQs7vo0uaiWzfphh93BCNiSU_kPwoQ9nkjd@mail.gmail.com>
	<AANLkTimVnELkhvJF860CKc-PzkYZijUp67RR-fh_27-t@mail.gmail.com>
Message-ID: <AANLkTimOLxbxei+naPxUq3SB78C+fD8H20xV2OfVV4He@mail.gmail.com>

FreeSwitch is a registered trademark as I can see on the front page,
so was that the case that you got the domain back by using the
trademark infringement reason?

I hope you or the project didn't pay the squatter x amount of cash
just to get the domain back...

On Mon, Jan 31, 2011 at 4:50 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Some domain squatter registered it and we did not originally realize it.
>
>
> On Mon, Jan 31, 2011 at 6:09 PM, curriegrad2004
> <curriegrad2004 at gmail.com> wrote:
>> Anthony, are you inclined to share what happened to freeswitch.com originally?
>>
>> On Mon, Jan 31, 2011 at 2:19 PM, Madovsky <infos at madovsky.org> wrote:
>>> yeah nice ! :D
>>>
>>> ----- Original Message -----
>>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>>> To: "Freeswitch-users" <freeswitch-users at lists.freeswitch.org>;
>>> <freeswitch-dev at lists.freeswitch.org>
>>> Sent: Monday, January 31, 2011 5:15 PM
>>> Subject: [Freeswitch-users] freeswitch.com is returned to us!
>>>
>>>
>>>> http://www.freeswitch.com/
>>>>
>>>> Thank you everyone who helped with this!
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 Feb  1 20:59:07 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 1 Feb 2011 11:59:07 -0600
Subject: [Freeswitch-users] freeswitch.com is returned to us!
In-Reply-To: <AANLkTimOLxbxei+naPxUq3SB78C+fD8H20xV2OfVV4He@mail.gmail.com>
References: <AANLkTi=BciYcFugPySe3QU7pK-=avKSpkOUiq119UCYP@mail.gmail.com>
	<5FB9599FB0CA4FA496A87C65DAFB9D53@e1705>
	<AANLkTik1QwQs7vo0uaiWzfphh93BCNiSU_kPwoQ9nkjd@mail.gmail.com>
	<AANLkTimVnELkhvJF860CKc-PzkYZijUp67RR-fh_27-t@mail.gmail.com>
	<AANLkTimOLxbxei+naPxUq3SB78C+fD8H20xV2OfVV4He@mail.gmail.com>
Message-ID: <AANLkTi=aqkQ5WSWp+fp-4Cg-fZUOU9dfvzN+P=z4DF+9@mail.gmail.com>

Yes I had to file a UDRP request and we prevailed.


On Tue, Feb 1, 2011 at 11:55 AM, curriegrad2004
<curriegrad2004 at gmail.com> wrote:
> FreeSwitch is a registered trademark as I can see on the front page,
> so was that the case that you got the domain back by using the
> trademark infringement reason?
>
> I hope you or the project didn't pay the squatter x amount of cash
> just to get the domain back...
>
> On Mon, Jan 31, 2011 at 4:50 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> Some domain squatter registered it and we did not originally realize it.
>>
>>
>> On Mon, Jan 31, 2011 at 6:09 PM, curriegrad2004
>> <curriegrad2004 at gmail.com> wrote:
>>> Anthony, are you inclined to share what happened to freeswitch.com originally?
>>>
>>> On Mon, Jan 31, 2011 at 2:19 PM, Madovsky <infos at madovsky.org> wrote:
>>>> yeah nice ! :D
>>>>
>>>> ----- Original Message -----
>>>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>>>> To: "Freeswitch-users" <freeswitch-users at lists.freeswitch.org>;
>>>> <freeswitch-dev at lists.freeswitch.org>
>>>> Sent: Monday, January 31, 2011 5:15 PM
>>>> Subject: [Freeswitch-users] freeswitch.com is returned to us!
>>>>
>>>>
>>>>> http://www.freeswitch.com/
>>>>>
>>>>> Thank you everyone who helped with this!
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org
>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 mstockton at harqen.com  Tue Feb  1 21:00:51 2011
From: mstockton at harqen.com (Matt Stockton)
Date: Tue, 1 Feb 2011 12:00:51 -0600
Subject: [Freeswitch-users] Problem with missing / duplicated DTMF
In-Reply-To: <AANLkTin+-NCnROr5ZE9GYoao_qVP_-2K7mWqhvJ+dkd3@mail.gmail.com>
References: <AANLkTin+-NCnROr5ZE9GYoao_qVP_-2K7mWqhvJ+dkd3@mail.gmail.com>
Message-ID: <AANLkTi=zyr6+n1+Qk9AWqvpUCNLuWPnf1CiXKYQk8v=Q@mail.gmail.com>

Sorry about not including the version. The version of freeswitch I am using
is.
FreeSWITCH Version 1.0.head (git-256a82d 2011-01-31 10-12-28 -0600)

 I just updated to the latest yesterday to re-test it.

On Tue, Feb 1, 2011 at 10:10 AM, Matt Stockton <mstockton at harqen.com> wrote:

> I have having trouble with both missing and duplicated DTMF in Freeswitch.
>
> Here are the steps of how I am using it:
>
> 1. Leg A - I am calling out from my Freeswitch instance (through iCall),
> and I am calling an iCall number that is also connected to the same
> Freeswitch instance.
>
> 2. Leg B - The above call is routed through iCall and then answered by the
> same Freeswitch instance.
>
> 3. On Leg B, I play a file and attempt to get DTMF in a lua script. Here is
> the code of interest:
>
> callPin = session:playAndGetDigits(1, 10, 4, 30000, "#","/tmp/cw_17.wav",
> "", "\\d+");
>
>
> 4. On Leg A, I send DTMF information in a lua script. Here is the code of
> interest. I initiate a delay between each digit:
>
> local newPin = "";
>
> for i = 1, string.len(pin) do
>
>   newPin = newPin .. string.sub(pin, i, i) .. "W";
>
> end
>
> session:execute("send_dtmf", newPin .. "#@200");
>
>
> ** Note that there is a session:sleep on Leg A before I send the DTMF to
> make sure i don't send it too early **
>
>
> The problem is that the recognized DTMF on Leg B is wrong about 30% of the
> time. For example, if Leg A enters: 22063083, Leg B will get the DTMF digits
> 222063083. This is an example of duplication, but I have also experienced
> missing DTMF codes (and an occasional wrong code completely)
>
>
> I have messed with a bunch of DTMF settings in hopes of fixing this issue,
> but I cannot seem to find something that is reliable 100% of the time.
>
>
> _____________________________
>
> Here are the DTMF settings I have looked at / messed with. I've tried
> various values for the dtmf-duration in the config (and in the send_dtmf
> command above)
>
>
>     <!-- This is in external.xml -->
>
>      <param name="dtmf-duration" value="500"/>
>
>     <param name="dtmf-type" value="rfc2833"/>
>
>     <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
>
>     <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
>     <param name="inbound-codec-negotiation" value="greedy"/>
>
>     <!-- This is in vars.xml -->
>     <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU"/>
>     <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU"/>
>
> ________________________
>
> I have run fs_cli with event logging and the DTMF events that Freeswitch
> gets do correlate to the wrong value (e.g. the duplication / missing digits
> is noticable in the Freeswitch events as well).
>
> Also, I am not running any dtmf-related applications on the session before
> I give control to the lua scripts (e.g. not running start_dtmf)
>
> Has anyone experienced this type of issue? Or know what I can do to resolve
> it? My next step was going to be trying this against another provider
> besides iCall, but I figured I would see if anyone has encountered a similar
> problem before. Any help is appreciated.
>
> Thanks
>
>
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From anthony.minessale at gmail.com  Tue Feb  1 21:22:45 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 1 Feb 2011 12:22:45 -0600
Subject: [Freeswitch-users] Problem with missing / duplicated DTMF
In-Reply-To: <AANLkTi=zyr6+n1+Qk9AWqvpUCNLuWPnf1CiXKYQk8v=Q@mail.gmail.com>
References: <AANLkTin+-NCnROr5ZE9GYoao_qVP_-2K7mWqhvJ+dkd3@mail.gmail.com>
	<AANLkTi=zyr6+n1+Qk9AWqvpUCNLuWPnf1CiXKYQk8v=Q@mail.gmail.com>
Message-ID: <AANLkTinODuGMpAs4mY_29o_FavBw_3t1oVdksjWoAfb9@mail.gmail.com>

I dont think there any current dtmf issues open.
It sounds like maybe you are going across the pstn and encountering
some problems with transition from
2833 to inband and back again or from hair-pinning the call.



On Tue, Feb 1, 2011 at 12:00 PM, Matt Stockton <mstockton at harqen.com> wrote:
> Sorry about not including the version. The version of freeswitch I am using
> is.
> FreeSWITCH Version 1.0.head (git-256a82d 2011-01-31 10-12-28 -0600)
> ?I just updated to the latest yesterday to re-test it.
> On Tue, Feb 1, 2011 at 10:10 AM, Matt Stockton <mstockton at harqen.com> wrote:
>>
>> I have having trouble with both missing and duplicated DTMF in
>> Freeswitch.
>> Here are the steps of how I am using it:
>> 1. Leg A - I am calling out from my Freeswitch instance (through iCall),
>> and I am calling an iCall number that is also connected to the same
>> Freeswitch instance.
>> 2. Leg B - The above call is routed through iCall and then answered by the
>> same Freeswitch instance.
>> 3. On Leg B, I play a file and attempt to get DTMF in a lua script. Here
>> is the code of interest:
>>
>> callPin = session:playAndGetDigits(1, 10, 4, 30000, "#","/tmp/cw_17.wav",
>> "", "\\d+");
>>
>> 4. On Leg A, I send DTMF information in a lua script. Here is the code of
>> interest. I initiate a delay between each digit:
>>
>> local newPin = "";
>>
>> for i = 1, string.len(pin) do
>>
>> ??newPin = newPin .. string.sub(pin, i, i) .. "W";
>>
>> end
>>
>> session:execute("send_dtmf", newPin .. "#@200");
>>
>> ** Note that there is a session:sleep on Leg A before I send the DTMF to
>> make sure i don't send it too early **
>>
>> The problem is that the recognized DTMF on Leg B is wrong about 30% of the
>> time. For example, if Leg A enters:?22063083, Leg B will get the DTMF digits
>> 222063083. This is an example of duplication, but I have also experienced
>> missing DTMF codes (and an occasional wrong code completely)
>>
>> I have messed with a bunch of DTMF settings in hopes of fixing this issue,
>> but I cannot seem to find something that is reliable 100% of the time.
>>
>> _____________________________
>>
>> Here are the DTMF settings I have looked at / messed with. I've tried
>> various values for the dtmf-duration in the config (and in the send_dtmf
>> command above)
>>
>> ?? ?<!-- This is in external.xml -->
>>
>> ?? ?<param name="dtmf-duration" value="500"/>
>>
>> ?? ?<param name="dtmf-type" value="rfc2833"/>
>>
>> ?? ?<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
>>
>> ?? ?<param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
>>
>> ?? ?<param name="inbound-codec-negotiation" value="greedy"/>
>> ?? ?<!-- This is in vars.xml -->
>> ?? ?<X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU"/>
>> ?? ?<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU"/>
>> ________________________
>> I have run fs_cli with event logging and the DTMF events that Freeswitch
>> gets do correlate to the wrong value (e.g. the duplication / missing digits
>> is noticable in the Freeswitch events as well).
>> Also, I am not running any dtmf-related applications on the session before
>> I give control to the lua scripts (e.g. not running start_dtmf)
>> Has anyone experienced this type of issue? Or know what I can do to
>> resolve it? My next step was going to be trying this against another
>> provider besides iCall, but I figured I would see if anyone has encountered
>> a similar problem before. Any help is appreciated.
>> 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 avi at avimarcus.net  Tue Feb  1 21:19:45 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 1 Feb 2011 20:19:45 +0200
Subject: [Freeswitch-users] open g729
In-Reply-To: <AANLkTinLARHTFP3EX1ow03NS-uyn4+FZTdbyym8ms4cF@mail.gmail.com>
References: <1296343422.2615.4.camel@gustavo-laptop>
	<AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com>
	<1296477877.3589.24.camel@gustavo-laptop>
	<AANLkTinq9AmN=ZqjGLNUxucw2nJXBLpNHKLTA5jLWfgj@mail.gmail.com>
	<9D7F812E93DC47A99F6A30638FB8092F@e1705>
	<AANLkTikkLnf4BHYgL1k8th=VWk44vpH5gRYSAFW5YfW1@mail.gmail.com>
	<A64FADA67C6D4C1C8B041EEFBB8C1DBB@e1705>
	<AANLkTi=1Fr2e-sz-n5Ds=q6KVntRzfzgrAQOgc0rCYgF@mail.gmail.com>
	<F4AB52F0A27D46418900E2841CB5524B@e1705>
	<AANLkTi=icT=Zw9xrrEcOyyozy5JFm7sXQjxTDxN3psZH@mail.gmail.com>
	<AANLkTinXuGfqOBdBcLSdBMaoMJk3JbWg18sfkKyyVofd@mail.gmail.com>
	<AANLkTik6My-K6pEW6d75yy9AgoM8wkFiCtz2vA5wuG+i@mail.gmail.com>
	<AANLkTimSNZpM1CXkDRx9R7_uuOSoxDxDBZF8OCgqtzH8@mail.gmail.com>
	<AANLkTinLARHTFP3EX1ow03NS-uyn4+FZTdbyym8ms4cF@mail.gmail.com>
Message-ID: <AANLkTik7CBsimkxum2R08m03MiUwNjn3LY=exNOjUHrO@mail.gmail.com>

Except that many hardware phones don't have speex, so the only low-bandwidth
solution is g729...

On Tue, Feb 1, 2011 at 7:43 PM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:

> Simple solution to this g.729 patent non-sense: Speex.
>
> 'nuff said.
>
> On Tue, Feb 1, 2011 at 8:26 AM, Anton VG <anton.vazir at gmail.com> wrote:
> > Considering the above, noone can say "Purchase a LICENSE", but only a
> > "Licensed SOFTWARE"
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  1 21:45:04 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 1 Feb 2011 13:45:04 -0500
Subject: [Freeswitch-users] open g729
In-Reply-To: <AANLkTik7CBsimkxum2R08m03MiUwNjn3LY=exNOjUHrO@mail.gmail.com>
References: <1296343422.2615.4.camel@gustavo-laptop>
	<AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com>
	<1296477877.3589.24.camel@gustavo-laptop>
	<AANLkTinq9AmN=ZqjGLNUxucw2nJXBLpNHKLTA5jLWfgj@mail.gmail.com>
	<9D7F812E93DC47A99F6A30638FB8092F@e1705>
	<AANLkTikkLnf4BHYgL1k8th=VWk44vpH5gRYSAFW5YfW1@mail.gmail.com>
	<A64FADA67C6D4C1C8B041EEFBB8C1DBB@e1705>
	<AANLkTi=1Fr2e-sz-n5Ds=q6KVntRzfzgrAQOgc0rCYgF@mail.gmail.com>
	<F4AB52F0A27D46418900E2841CB5524B@e1705>
	<AANLkTi=icT=Zw9xrrEcOyyozy5JFm7sXQjxTDxN3psZH@mail.gmail.com>
	<AANLkTinXuGfqOBdBcLSdBMaoMJk3JbWg18sfkKyyVofd@mail.gmail.com>
	<AANLkTik6My-K6pEW6d75yy9AgoM8wkFiCtz2vA5wuG+i@mail.gmail.com>
	<AANLkTimSNZpM1CXkDRx9R7_uuOSoxDxDBZF8OCgqtzH8@mail.gmail.com>
	<AANLkTinLARHTFP3EX1ow03NS-uyn4+FZTdbyym8ms4cF@mail.gmail.com>
	<AANLkTik7CBsimkxum2R08m03MiUwNjn3LY=exNOjUHrO@mail.gmail.com>
Message-ID: <AANLkTikE3viP6KN+28cRzx=39cLQJ+OJZU64xi27UBB9@mail.gmail.com>

Phones, carriers, and just about any commercial gear (reference
hardware, SBCs, DSPs, etc) you'll ever see doesn't support Speex.
Guess how you'll make it work?  Transcoding from G729 to Speex.  Now
you're using licenses, proxying RTP, and wasting CPU while ruining
voice quality transcoding from one lossy codec to another.  It's just
about the worst possible situation you could be in.

On Tue, Feb 1, 2011 at 1:19 PM, Avi Marcus <avi at avimarcus.net> wrote:
> Except that many hardware phones don't have speex, so the only low-bandwidth
> solution is g729...
> On Tue, Feb 1, 2011 at 7:43 PM, curriegrad2004 <curriegrad2004 at gmail.com>
> wrote:
>>
>> Simple solution to this g.729 patent non-sense: Speex.
>>
>> 'nuff said.
>>
>> On Tue, Feb 1, 2011 at 8:26 AM, Anton VG <anton.vazir at gmail.com> wrote:
>> > Considering the above, noone can say "Purchase a LICENSE", but only a
>> > "Licensed SOFTWARE"
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 krice at freeswitch.org  Tue Feb  1 21:55:16 2011
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 01 Feb 2011 12:55:16 -0600
Subject: [Freeswitch-users] open g729
In-Reply-To: <AANLkTikE3viP6KN+28cRzx=39cLQJ+OJZU64xi27UBB9@mail.gmail.com>
Message-ID: <C96DB334.46A6%krice@freeswitch.org>

Why people insist on speex I am not sure... Sure its nice on a closed
network but quickly looses its luster when communicating with the real
world... Not to mention the CPU usages and such...

If you want to use something like speex check out iLBC and theres things
like the Sangoma D100/D500 that you can use to offload the transcoding from
the CPU (disclaimer I sell the sangoma hardware)

K


On 2/1/11 12:45 PM, "Kristian Kielhofner" <kris at kriskinc.com> wrote:

> Phones, carriers, and just about any commercial gear (reference
> hardware, SBCs, DSPs, etc) you'll ever see doesn't support Speex.
> Guess how you'll make it work?  Transcoding from G729 to Speex.  Now
> you're using licenses, proxying RTP, and wasting CPU while ruining
> voice quality transcoding from one lossy codec to another.  It's just
> about the worst possible situation you could be in.
> 
> On Tue, Feb 1, 2011 at 1:19 PM, Avi Marcus <avi at avimarcus.net> wrote:
>> Except that many hardware phones don't have speex, so the only low-bandwidth
>> solution is g729...
>> On Tue, Feb 1, 2011 at 7:43 PM, curriegrad2004 <curriegrad2004 at gmail.com>
>> wrote:
>>> 
>>> Simple solution to this g.729 patent non-sense: Speex.
>>> 
>>> 'nuff said.
>>> 
>>> On Tue, Feb 1, 2011 at 8:26 AM, Anton VG <anton.vazir at gmail.com> wrote:
>>>> Considering the above, noone can say "Purchase a LICENSE", but only a
>>>> "Licensed SOFTWARE"
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 Feb  1 22:51:05 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 1 Feb 2011 13:51:05 -0600
Subject: [Freeswitch-users] execute_on_ring executing on answer
In-Reply-To: <1296582543.5245.87.camel@luna.tc.commsmundi.com>
References: <1296215265.8986.147.camel@luna.tc.commsmundi.com>
	<21B4FB4AAC2E428E938F99986CA09165@e1705>
	<1296235095.8986.167.camel@luna.tc.commsmundi.com>
	<0E96944DB29F4CEB9ACA5994D19AB0D6@e1705>
	<AANLkTikHB-QnDyT0WPsEML8yv4AACnYnHrE37gpRsLS2@mail.gmail.com>
	<1296306637.8986.185.camel@luna.tc.commsmundi.com>
	<1296582543.5245.87.camel@luna.tc.commsmundi.com>
Message-ID: <AANLkTim1Hq0vrOLGpQt4DcNKUY970HzFF1KHF8ewmwxz@mail.gmail.com>

I think I see why.
The app is queued at the right time but not executed until media is active.

I have changed the code so now when you supply ::<arg> at the end of
the app name it will be async and when you don't it will be executed
immediately.

On Tue, Feb 1, 2011 at 11:49 AM, Fran?ois Delawarde
<fdelawarde at wirelessmundi.com> wrote:
> Since apparently noone reproduces this, it must be a configuration error
> from my part.
>
> Any hints of where I could start looking to resolve this issue?
>
> Thanks,
> Fran?ois.
>
>
> On Sat, 2011-01-29 at 14:10 +0100, Fran?ois Delawarde wrote:
>> Nice to know, but in that case the destination actually rings (180).
>>
>> See commented log:
>> http://pastebin.freeswitch.org/15168
>>
>> Fran?ois.
>>
>> On Fri, 2011-01-28 at 12:34 -0600, Anthony Minessale wrote:
>> > if it never rings, answer will still trigger it.
>> >
>> >
>> > On Fri, Jan 28, 2011 at 11:24 AM, Madovsky <infos at madovsky.org> wrote:
>> > > ah ok, maybe a wiki update would be useful.
>> > >
>> > >
>> > >
>> > > ----- Original Message -----
>> > > From: "Fran?ois Delawarde" <fdelawarde at wirelessmundi.com>
>> > > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> > > Sent: Friday, January 28, 2011 12:18 PM
>> > > Subject: Re: [Freeswitch-users] execute_on_ring executing on answer
>> > >
>> > >
>> > >> It's some cool feature made by Anthony that allows me to specify the
>> > >> separator.
>> > >>
>> > >> in ^^:PCMA:G722
>> > >> ^^: means the separator is now : instead of ,
>> > >>
>> > >> Useful in the [] or {} case because the coma is already used to separate
>> > >> variables.
>> > >>
>> > >> Fran?ois.
>> > >>
>> > >> On Fri, 2011-01-28 at 12:12 -0500, Madovsky wrote:
>> > >>> what means the ^^ in your codec string ?
>> > >>>
>> > >>> ----- Original Message -----
>> > >>> From: "Fran?ois Delawarde" <fdelawarde at wirelessmundi.com>
>> > >>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> > >>> Sent: Friday, January 28, 2011 6:47 AM
>> > >>> Subject: [Freeswitch-users] execute_on_ring executing on answer
>> > >>>
>> > >>>
>> > >>> > Hi,
>> > >>> >
>> > >>> > Doing some testing with this morning's git (Fri Jan 28) I just found
>> > >>> > out
>> > >>> > that the execute_on_ring application runs when the destination answers
>> > >>> > instead of when it rings.
>> > >>> >
>> > >>> > So far, I can't seem to find out the reason. Could it be some
>> > >>> > configuration issue?
>> > >>> >
>> > >>> >
>> > >>> > Here a call log showing the phenomenon with a simple bridge:
>> > >>> >
>> > >>> > <action application="bridge"
>> > >>> > data="[execute_on_ring=info,absolute_codec_string=^^:PCMA:G722]sofia/192.168.10.1/sip:2103 at 192.168.10.22:5060"/>
>> > >>> >
>> > >>> > http://pastebin.freeswitch.org/15168
>> > >>> >
>> > >>> >
>> > >>> > Thanks,
>> > >>> > Fran?ois.
>> > >>> >
>> > >>> >
>> > >>> > _______________________________________________
>> > >>> > FreeSWITCH-users mailing list
>> > >>> > FreeSWITCH-users at lists.freeswitch.org
>> > >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > >>> > http://www.freeswitch.org
>> > >>> >
>> > >>>
>> > >>>
>> > >>> _______________________________________________
>> > >>> FreeSWITCH-users mailing list
>> > >>> FreeSWITCH-users at lists.freeswitch.org
>> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > >>> http://www.freeswitch.org
>> > >>
>> > >>
>> > >> _______________________________________________
>> > >> FreeSWITCH-users mailing list
>> > >> FreeSWITCH-users at lists.freeswitch.org
>> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > >> http://www.freeswitch.org
>> > >>
>> > >
>> > >
>> > > _______________________________________________
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> > >
>> >
>> >
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb  1 22:57:43 2011
From: brad at tritelcomm.com (Brad Mina)
Date: Tue, 1 Feb 2011 11:57:43 -0800
Subject: [Freeswitch-users] Modify Registration String to ITSP SipStation
Message-ID: <AANLkTi=nxkFBxtnbWC2M-rJPQTK3awdVTSSDz9Mesymr@mail.gmail.com>

I'm having trouble registering a SipStation account. They require
username:password at trunk1.freepbx.com<username%3Apassword at trunk1.freepbx.com>to
be sent to the registrar. I've done a bit of searching but was unable
to
find any specific details on how to do this. My config is as follows:

<include>
    <gateway name="SipStation">
      <param name="username" value="myUsername"/>
      <param name="password" value="somePassword"/>
      <param name="realm" value="trunk1.freepbx.com"/>
      <param name="from-user" value="TriTelComm"/>
      <param name="proxy" value="trunk1.freepbx.com"/>
      <param name="expire-seconds" value="800"/>
      <param name="register" value="true"/>
      <param name="retry-seconds" value="60"/>
      <param name="context" value="public"/>
    </gateway>
</include>
-------------- next part --------------
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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110201/ea00da46/attachment-0001.html 

From anthony.minessale at gmail.com  Tue Feb  1 23:01:16 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 1 Feb 2011 14:01:16 -0600
Subject: [Freeswitch-users] Modify Registration String to ITSP SipStation
In-Reply-To: <AANLkTi=nxkFBxtnbWC2M-rJPQTK3awdVTSSDz9Mesymr@mail.gmail.com>
References: <AANLkTi=nxkFBxtnbWC2M-rJPQTK3awdVTSSDz9Mesymr@mail.gmail.com>
Message-ID: <AANLkTi=wD5doBH0V48UKyQcK8Atg_=QN2V6Xw0RuTxRZ@mail.gmail.com>

They want you to send it in clear text as part of your username?
I guess you would put the entire user:pass in the username field and
leave the password blank?


On Tue, Feb 1, 2011 at 1:57 PM, Brad Mina <brad at tritelcomm.com> wrote:
> I'm having trouble registering a SipStation account. They require
> username:password at trunk1.freepbx.com to be sent to the registrar. I've done
> a bit of searching but was unable to find any specific details on how to do
> this. My config is as follows:
>
> <include>
> ??? <gateway name="SipStation">
> ????? <param name="username" value="myUsername"/>
> ????? <param name="password" value="somePassword"/>
> ????? <param name="realm" value="trunk1.freepbx.com"/>
> ????? <param name="from-user" value="TriTelComm"/>
> ????? <param name="proxy" value="trunk1.freepbx.com"/>
> ????? <param name="expire-seconds" value="800"/>
> ????? <param name="register" value="true"/>
> ????? <param name="retry-seconds" value="60"/>
> ????? <param name="context" value="public"/>
> ??? </gateway>
> </include>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb  1 23:08:51 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 1 Feb 2011 12:08:51 -0800
Subject: [Freeswitch-users] execute_on_ring executing on answer
In-Reply-To: <AANLkTim1Hq0vrOLGpQt4DcNKUY970HzFF1KHF8ewmwxz@mail.gmail.com>
References: <1296215265.8986.147.camel@luna.tc.commsmundi.com>
	<21B4FB4AAC2E428E938F99986CA09165@e1705>
	<1296235095.8986.167.camel@luna.tc.commsmundi.com>
	<0E96944DB29F4CEB9ACA5994D19AB0D6@e1705>
	<AANLkTikHB-QnDyT0WPsEML8yv4AACnYnHrE37gpRsLS2@mail.gmail.com>
	<1296306637.8986.185.camel@luna.tc.commsmundi.com>
	<1296582543.5245.87.camel@luna.tc.commsmundi.com>
	<AANLkTim1Hq0vrOLGpQt4DcNKUY970HzFF1KHF8ewmwxz@mail.gmail.com>
Message-ID: <AANLkTimC-j3aqgnWbDP7JVf2F2BtGoB5cLnvf-qq=XqC@mail.gmail.com>

I believe I have a new item to document on the wiki. :)
Francois, let us know if the latest git works for you.
Thanks,
MC

On Tue, Feb 1, 2011 at 11:51 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I think I see why.
> The app is queued at the right time but not executed until media is active.
>
> I have changed the code so now when you supply ::<arg> at the end of
> the app name it will be async and when you don't it will be executed
> immediately.
>
> On Tue, Feb 1, 2011 at 11:49 AM, Fran?ois Delawarde
> <fdelawarde at wirelessmundi.com> wrote:
> > Since apparently noone reproduces this, it must be a configuration error
> > from my part.
> >
> > Any hints of where I could start looking to resolve this issue?
> >
> > Thanks,
> > Fran?ois.
> >
> >
> > On Sat, 2011-01-29 at 14:10 +0100, Fran?ois Delawarde wrote:
> >> Nice to know, but in that case the destination actually rings (180).
> >>
> >> See commented log:
> >> http://pastebin.freeswitch.org/15168
> >>
> >> Fran?ois.
> >>
> >> On Fri, 2011-01-28 at 12:34 -0600, Anthony Minessale wrote:
> >> > if it never rings, answer will still trigger it.
> >> >
> >> >
> >> > On Fri, Jan 28, 2011 at 11:24 AM, Madovsky <infos at madovsky.org>
> wrote:
> >> > > ah ok, maybe a wiki update would be useful.
> >> > >
> >> > >
> >> > >
> >> > > ----- Original Message -----
> >> > > From: "Fran?ois Delawarde" <fdelawarde at wirelessmundi.com>
> >> > > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> >> > > Sent: Friday, January 28, 2011 12:18 PM
> >> > > Subject: Re: [Freeswitch-users] execute_on_ring executing on answer
> >> > >
> >> > >
> >> > >> It's some cool feature made by Anthony that allows me to specify
> the
> >> > >> separator.
> >> > >>
> >> > >> in ^^:PCMA:G722
> >> > >> ^^: means the separator is now : instead of ,
> >> > >>
> >> > >> Useful in the [] or {} case because the coma is already used to
> separate
> >> > >> variables.
> >> > >>
> >> > >> Fran?ois.
> >> > >>
> >> > >> On Fri, 2011-01-28 at 12:12 -0500, Madovsky wrote:
> >> > >>> what means the ^^ in your codec string ?
> >> > >>>
> >> > >>> ----- Original Message -----
> >> > >>> From: "Fran?ois Delawarde" <fdelawarde at wirelessmundi.com>
> >> > >>> To: "FreeSWITCH Users Help" <
> freeswitch-users at lists.freeswitch.org>
> >> > >>> Sent: Friday, January 28, 2011 6:47 AM
> >> > >>> Subject: [Freeswitch-users] execute_on_ring executing on answer
> >> > >>>
> >> > >>>
> >> > >>> > Hi,
> >> > >>> >
> >> > >>> > Doing some testing with this morning's git (Fri Jan 28) I just
> found
> >> > >>> > out
> >> > >>> > that the execute_on_ring application runs when the destination
> answers
> >> > >>> > instead of when it rings.
> >> > >>> >
> >> > >>> > So far, I can't seem to find out the reason. Could it be some
> >> > >>> > configuration issue?
> >> > >>> >
> >> > >>> >
> >> > >>> > Here a call log showing the phenomenon with a simple bridge:
> >> > >>> >
> >> > >>> > <action application="bridge"
> >> > >>> >
> data="[execute_on_ring=info,absolute_codec_string=^^:PCMA:G722]sofia/
> 192.168.10.1/sip:2103 at 192.168.10.22:5060"/>
> >> > >>> >
> >> > >>> > http://pastebin.freeswitch.org/15168
> >> > >>> >
> >> > >>> >
> >> > >>> > Thanks,
> >> > >>> > Fran?ois.
> >> > >>> >
> >> > >>> >
> >> > >>> > _______________________________________________
> >> > >>> > FreeSWITCH-users mailing list
> >> > >>> > FreeSWITCH-users at lists.freeswitch.org
> >> > >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > >>> > http://www.freeswitch.org
> >> > >>> >
> >> > >>>
> >> > >>>
> >> > >>> _______________________________________________
> >> > >>> FreeSWITCH-users mailing list
> >> > >>> FreeSWITCH-users at lists.freeswitch.org
> >> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > >>> http://www.freeswitch.org
> >> > >>
> >> > >>
> >> > >> _______________________________________________
> >> > >> FreeSWITCH-users mailing list
> >> > >> FreeSWITCH-users at lists.freeswitch.org
> >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > >> http://www.freeswitch.org
> >> > >>
> >> > >
> >> > >
> >> > > _______________________________________________
> >> > > FreeSWITCH-users mailing list
> >> > > FreeSWITCH-users at lists.freeswitch.org
> >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > > http://www.freeswitch.org
> >> > >
> >> >
> >> >
> >> >
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  1 23:11:00 2011
From: null at invalid.name (Dan Lane)
Date: Tue, 1 Feb 2011 20:11:00 +0000
Subject: [Freeswitch-users] Choppy VM when using loopback (on Debian
	lenny)
In-Reply-To: <AANLkTiktQU7kOG_22nG=_oEHSrBad667jcTzBs3n2nQv@mail.gmail.com>
References: <4D42B8EE.7030208@estation.dk>
	<7C6F3FC7-445F-4157-BD59-D76A705E4CE7@carmickle.com>
	<9894FEC6-E451-4993-A85A-2CD910E916B1@ipeva.fr>
	<AANLkTim+aBjRiAvxDUF1O1vut_CJpqgct2OqCNVxi0q4@mail.gmail.com>
	<AANLkTiktckovGv4Ksnn_56EzpvTTSnfG+iD1vPX1_4RM@mail.gmail.com>
	<4D466CAE.7030402@estation.dk>
	<AANLkTimhOKmy_yupGtA2HBY_8wXtQKZnDYJ9Ye76bhzq@mail.gmail.com>
	<AANLkTiktQU7kOG_22nG=_oEHSrBad667jcTzBs3n2nQv@mail.gmail.com>
Message-ID: <AANLkTiko=qTZbXCL09D-+W-FmibMnBjOeoR30YYzHe0i@mail.gmail.com>

Yes, using 1f1541b our calls that use mod_loopback into mod_conference
are unusable. The audio is choppy and delay increases as time passes.

I thought it was an issue relating to the default 100Hz kernel on EC2
so I spent some time yesterday putting together a 1000Hz kernel but it
didn't make any difference. In the meantime I've compiled mod_loopback
from 4c5426f and loaded it with my 1f1541b build which eliminates the
issue.

I haven't added it to Jira yet as I want to spend some time debugging
it (and I also owe you some debug info for FS-2934) but the problem is
definitely there.

On Tue, Feb 1, 2011 at 3:58 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Are you saying you have better results on that version than you do on
> the latest?
> What conditions do you have that cause you trouble, what is the
> endpoint on the other side.
>
> If the last commit to mod_loopback intended to improve audio quality
> actually makes it worse I need to investigate it.
>
>
> On Tue, Feb 1, 2011 at 5:53 AM, Dan Lane <null at invalid.name> wrote:
>> FWIW we've been also been having audio issues with loopback recently
>> on EC2 (with a 1000Hz kernel).
>>
>> We worked around it in the short term by reverting mod_loopback to
>> git-4c5426f during the build process.
>>
>> For anyone else who wants to try this just run "git checkout 4c5426f"
>> in src/mod/endpoints/mod_loopback then build as usual. This is NOT a
>> long term solution though.
>>
>> On Mon, Jan 31, 2011 at 8:02 AM, ?yvind Albrigtsen <oa at estation.dk> wrote:
>>> Thanks for all your feedback. I'll keep on trying and inform you what
>>> worked for me.
>>>
>>>
>>> Regards
>>> Oyvind
>>>
>>> On 2011-01-29 21:48, Anthony Minessale wrote:
>>>> Everyone should try latest GIT before pondering any further because I
>>>> added a patch like 2 days ago to adress this issue.
>>>>
>>>>
>>>> On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank at telonium.com> ?wrote:
>>>>
>>>>> Yeah. I, too, don't see the correlation between the NTP and hw timer..
>>>>> I am not familiar with the timer_test command and what it's measuring, but
>>>>> of the 50 tests it ran, min is 19089 and max is 20713.
>>>>> Frank
>>>>>
>>>>>
>>>>> On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone at ipeva.fr>
>>>>> wrote:
>>>>>
>>>>>> Frank,
>>>>>> I fail to see the relationship between the hw timer and NTP.
>>>>>> Can you please elaborate ?
>>>>>> David Ponzone ?Direction Technique
>>>>>> email: david.ponzone at ipeva.fr
>>>>>> tel: ? ? ?01 74 03 18 97
>>>>>> gsm: ? 06 66 98 76 34
>>>>>> Service Client IPeva
>>>>>> tel: ? ? ?0811 46 26 26
>>>>>> www.ipeva.fr ?- ? www.ipeva-studio.com
>>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>>>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>>
>>>>>>
>>>>>>
>>>>>> Le 28/01/2011 ? 19:49, Frank Carmickle a ?crit :
>>>>>>
>>>>>> Hi
>>>>>>
>>>>>> On Jan 28, 2011, at 7:39 AM, ?yvind Albrigtsen wrote:
>>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> I'm using latest git-version of Freeswitch, and when I go to voicemail
>>>>>>
>>>>>> when calling a number the sound playback is choppy and it skips some of
>>>>>>
>>>>>> the digits in the number I called.
>>>>>>
>>>>>> What kind of results do you get from timer_test at the fs_cli? ?Are you
>>>>>> running on hardware or are you virtualized? ?What is your clock source set
>>>>>> to and what are your available clock source options? ?See
>>>>>> /sys/devices/system/clocksource/clocksource0/available_clocksource and
>>>>>> /sys/devices/system/clocksource/clocksource0/current_clocksource. ?I am
>>>>>> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to
>>>>>> hang at 19998/19999 which works very well for me. ?When I was having problem
>>>>>> it was reporting numbers all over the map from 17400 to 22600 with lots of
>>>>>> randomness in between. ?I have my clocksource set to jiffies and xen
>>>>>> independent wallclock set to 1. ?Of course at that point you need to have
>>>>>> ntp running against a bunch of servers to drive your clock nice and steady.
>>>>>> ? I know my set up is probably a lot different than yours but I thought I'd
>>>>>> toss it out there to show that some of the harshest conditions can be dealt
>>>>>> with and don't give up trying. ?If you are running on hardware with a cpu
>>>>>> that doesn't have constant_tsc then you might have some problems. ?Just play
>>>>>> with the different timer options until you find the one that works.
>>>>>>
>>>>>> HTH
>>>>>> --FC
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at 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
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 christian.loeschenkohl at xpirio.com  Tue Feb  1 23:18:24 2011
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Tue, 01 Feb 2011 21:18:24 +0100
Subject: [Freeswitch-users] freeswitch.com is returned to us!
In-Reply-To: <AANLkTi=BciYcFugPySe3QU7pK-=avKSpkOUiq119UCYP@mail.gmail.com>
References: <AANLkTi=BciYcFugPySe3QU7pK-=avKSpkOUiq119UCYP@mail.gmail.com>
Message-ID: <4D486A90.7090308@xpirio.com>

hello

i am very happy that this is resolved and the domain got back home.

maybe you can also name us as donators (i'm sure our ceo will like it) :-)
xpirio.com, http://www.xpirio.com

br

On 2011-01-31 23:15, Anthony Minessale wrote:

> http://www.freeswitch.com/
>
> Thank you everyone who helped with this!
>
>
>


-- 
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 freeswitch at servercorps.com  Tue Feb  1 23:30:56 2011
From: freeswitch at servercorps.com (Addison Martin)
Date: Tue, 1 Feb 2011 14:30:56 -0600
Subject: [Freeswitch-users] Load mod_xml_rpc via cli?
Message-ID: <AANLkTin2yGAp03APRs0rdbP-d7QAXpDoQVqgzE5MxfhX@mail.gmail.com>

I seem to be able to load mod_xml_rpc at runtime, but netstat -a isn't
showing port 8080 as listening. Is it possible to do this at runtime, or is
a restart of the switch required?  I'm buidling a click2dial greasemonkey
script, and the xml_rpc api seems the way to do it without a bunch of client
side requirements.


Regards,

Nik
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From tim.compnetwork at gmail.com  Tue Feb  1 22:11:42 2011
From: tim.compnetwork at gmail.com (Tim King)
Date: Tue, 1 Feb 2011 14:11:42 -0500
Subject: [Freeswitch-users] H323 Setup
Message-ID: <AANLkTi=NZM66Mm2KVTdfS=z0rmghQEy7h=hqfa3+s-ST@mail.gmail.com>

>
> I am following http://wiki.freeswitch.org/wiki/Mod_h323 and trying to get
> H.323 setup. I installed using buildopal.sh script in the build directory.I
> am not finding h323.conf.xml anywhere except the sample in
> /usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/h323.conf.xml. Do
> I need to create this file and if so where should it be?
>

Thanks

Tim
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From anthony.minessale at gmail.com  Tue Feb  1 23:47:19 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 1 Feb 2011 14:47:19 -0600
Subject: [Freeswitch-users] Choppy VM when using loopback (on Debian
	lenny)
In-Reply-To: <AANLkTiko=qTZbXCL09D-+W-FmibMnBjOeoR30YYzHe0i@mail.gmail.com>
References: <4D42B8EE.7030208@estation.dk>
	<7C6F3FC7-445F-4157-BD59-D76A705E4CE7@carmickle.com>
	<9894FEC6-E451-4993-A85A-2CD910E916B1@ipeva.fr>
	<AANLkTim+aBjRiAvxDUF1O1vut_CJpqgct2OqCNVxi0q4@mail.gmail.com>
	<AANLkTiktckovGv4Ksnn_56EzpvTTSnfG+iD1vPX1_4RM@mail.gmail.com>
	<4D466CAE.7030402@estation.dk>
	<AANLkTimhOKmy_yupGtA2HBY_8wXtQKZnDYJ9Ye76bhzq@mail.gmail.com>
	<AANLkTiktQU7kOG_22nG=_oEHSrBad667jcTzBs3n2nQv@mail.gmail.com>
	<AANLkTiko=qTZbXCL09D-+W-FmibMnBjOeoR30YYzHe0i@mail.gmail.com>
Message-ID: <AANLkTi=t8AiZG6axtfqmMe9ZHTPNowoAb=7eNjQ-RPoG@mail.gmail.com>

Try the latest GIT, I reverted the last patch and tried to solve the
problem differently.


On Tue, Feb 1, 2011 at 2:11 PM, Dan Lane <null at invalid.name> wrote:
> Yes, using 1f1541b our calls that use mod_loopback into mod_conference
> are unusable. The audio is choppy and delay increases as time passes.
>
> I thought it was an issue relating to the default 100Hz kernel on EC2
> so I spent some time yesterday putting together a 1000Hz kernel but it
> didn't make any difference. In the meantime I've compiled mod_loopback
> from 4c5426f and loaded it with my 1f1541b build which eliminates the
> issue.
>
> I haven't added it to Jira yet as I want to spend some time debugging
> it (and I also owe you some debug info for FS-2934) but the problem is
> definitely there.
>
> On Tue, Feb 1, 2011 at 3:58 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> Are you saying you have better results on that version than you do on
>> the latest?
>> What conditions do you have that cause you trouble, what is the
>> endpoint on the other side.
>>
>> If the last commit to mod_loopback intended to improve audio quality
>> actually makes it worse I need to investigate it.
>>
>>
>> On Tue, Feb 1, 2011 at 5:53 AM, Dan Lane <null at invalid.name> wrote:
>>> FWIW we've been also been having audio issues with loopback recently
>>> on EC2 (with a 1000Hz kernel).
>>>
>>> We worked around it in the short term by reverting mod_loopback to
>>> git-4c5426f during the build process.
>>>
>>> For anyone else who wants to try this just run "git checkout 4c5426f"
>>> in src/mod/endpoints/mod_loopback then build as usual. This is NOT a
>>> long term solution though.
>>>
>>> On Mon, Jan 31, 2011 at 8:02 AM, ?yvind Albrigtsen <oa at estation.dk> wrote:
>>>> Thanks for all your feedback. I'll keep on trying and inform you what
>>>> worked for me.
>>>>
>>>>
>>>> Regards
>>>> Oyvind
>>>>
>>>> On 2011-01-29 21:48, Anthony Minessale wrote:
>>>>> Everyone should try latest GIT before pondering any further because I
>>>>> added a patch like 2 days ago to adress this issue.
>>>>>
>>>>>
>>>>> On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank at telonium.com> ?wrote:
>>>>>
>>>>>> Yeah. I, too, don't see the correlation between the NTP and hw timer..
>>>>>> I am not familiar with the timer_test command and what it's measuring, but
>>>>>> of the 50 tests it ran, min is 19089 and max is 20713.
>>>>>> Frank
>>>>>>
>>>>>>
>>>>>> On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone at ipeva.fr>
>>>>>> wrote:
>>>>>>
>>>>>>> Frank,
>>>>>>> I fail to see the relationship between the hw timer and NTP.
>>>>>>> Can you please elaborate ?
>>>>>>> David Ponzone ?Direction Technique
>>>>>>> email: david.ponzone at ipeva.fr
>>>>>>> tel: ? ? ?01 74 03 18 97
>>>>>>> gsm: ? 06 66 98 76 34
>>>>>>> Service Client IPeva
>>>>>>> tel: ? ? ?0811 46 26 26
>>>>>>> www.ipeva.fr ?- ? www.ipeva-studio.com
>>>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>>>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>>>>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Le 28/01/2011 ? 19:49, Frank Carmickle a ?crit :
>>>>>>>
>>>>>>> Hi
>>>>>>>
>>>>>>> On Jan 28, 2011, at 7:39 AM, ?yvind Albrigtsen wrote:
>>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>> I'm using latest git-version of Freeswitch, and when I go to voicemail
>>>>>>>
>>>>>>> when calling a number the sound playback is choppy and it skips some of
>>>>>>>
>>>>>>> the digits in the number I called.
>>>>>>>
>>>>>>> What kind of results do you get from timer_test at the fs_cli? ?Are you
>>>>>>> running on hardware or are you virtualized? ?What is your clock source set
>>>>>>> to and what are your available clock source options? ?See
>>>>>>> /sys/devices/system/clocksource/clocksource0/available_clocksource and
>>>>>>> /sys/devices/system/clocksource/clocksource0/current_clocksource. ?I am
>>>>>>> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to
>>>>>>> hang at 19998/19999 which works very well for me. ?When I was having problem
>>>>>>> it was reporting numbers all over the map from 17400 to 22600 with lots of
>>>>>>> randomness in between. ?I have my clocksource set to jiffies and xen
>>>>>>> independent wallclock set to 1. ?Of course at that point you need to have
>>>>>>> ntp running against a bunch of servers to drive your clock nice and steady.
>>>>>>> ? I know my set up is probably a lot different than yours but I thought I'd
>>>>>>> toss it out there to show that some of the harshest conditions can be dealt
>>>>>>> with and don't give up trying. ?If you are running on hardware with a cpu
>>>>>>> that doesn't have constant_tsc then you might have some problems. ?Just play
>>>>>>> with the different timer options until you find the one that works.
>>>>>>>
>>>>>>> HTH
>>>>>>> --FC
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at 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
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 steveayre at gmail.com  Wed Feb  2 00:34:14 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 1 Feb 2011 21:34:14 +0000
Subject: [Freeswitch-users] H323 Setup
In-Reply-To: <AANLkTi=NZM66Mm2KVTdfS=z0rmghQEy7h=hqfa3+s-ST@mail.gmail.com>
References: <AANLkTi=NZM66Mm2KVTdfS=z0rmghQEy7h=hqfa3+s-ST@mail.gmail.com>
Message-ID: <AANLkTikvQKt5seDo9W1fX6_+SQ6abG0X9a7=KeskQv8j@mail.gmail.com>

You do not use buildopal.sh - that installs opal (for mod_opal) not h323plus
(for mod_h323). The wiki

The h323.conf.xml you found is the sample config file... adjust it to suit
you and place it in your conf/autoload_configs directory created when you
install.

-Steve


On 1 February 2011 19:11, Tim King <tim.compnetwork at gmail.com> wrote:

> I am following http://wiki.freeswitch.org/wiki/Mod_h323 and trying to get
>> H.323 setup. I installed using buildopal.sh script in the build directory.I
>> am not finding h323.conf.xml anywhere except the sample in
>> /usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/h323.conf.xml. Do
>> I need to create this file and if so where should it be?
>>
>
> Thanks
>
> 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
>
>
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From steveayre at gmail.com  Wed Feb  2 00:35:39 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 1 Feb 2011 21:35:39 +0000
Subject: [Freeswitch-users] H323 Setup
In-Reply-To: <AANLkTikvQKt5seDo9W1fX6_+SQ6abG0X9a7=KeskQv8j@mail.gmail.com>
References: <AANLkTi=NZM66Mm2KVTdfS=z0rmghQEy7h=hqfa3+s-ST@mail.gmail.com>
	<AANLkTikvQKt5seDo9W1fX6_+SQ6abG0X9a7=KeskQv8j@mail.gmail.com>
Message-ID: <AANLkTi=K+d74sMyc2H2N6h-eTT--qt52PK7T9V8nLD8u@mail.gmail.com>

Clicked send early.

You do not use buildopal.sh - that installs opal (for mod_opal) not h323plus
(for mod_h323). The wiki is wrong where it says that - the library headers
are not compatible, mod_h323 can't compile against opal.
The h323.conf.xml you found is the sample config file... adjust it to suit
you and place it in your conf/autoload_configs directory created when you
install.

-Steve



On 1 February 2011 21:34, Steven Ayre <steveayre at gmail.com> wrote:

> You do not use buildopal.sh - that installs opal (for mod_opal) not
> h323plus (for mod_h323). The wiki
>
> The h323.conf.xml you found is the sample config file... adjust it to suit
> you and place it in your conf/autoload_configs directory created when you
> install.
>
> -Steve
>
>
> On 1 February 2011 19:11, Tim King <tim.compnetwork at gmail.com> wrote:
>
>> I am following http://wiki.freeswitch.org/wiki/Mod_h323 and trying to get
>>> H.323 setup. I installed using buildopal.sh script in the build directory.I
>>> am not finding h323.conf.xml anywhere except the sample in
>>> /usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/h323.conf.xml. Do
>>> I need to create this file and if so where should it be?
>>>
>>
>> Thanks
>>
>> 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
>>
>>
>
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From tculjaga at gmail.com  Wed Feb  2 00:42:38 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 1 Feb 2011 22:42:38 +0100
Subject: [Freeswitch-users] H323 Setup
In-Reply-To: <AANLkTikvQKt5seDo9W1fX6_+SQ6abG0X9a7=KeskQv8j@mail.gmail.com>
References: <AANLkTi=NZM66Mm2KVTdfS=z0rmghQEy7h=hqfa3+s-ST@mail.gmail.com>
	<AANLkTikvQKt5seDo9W1fX6_+SQ6abG0X9a7=KeskQv8j@mail.gmail.com>
Message-ID: <AANLkTin2J2yXL+JQGAxe2b88+Kjyy_KPZc0u8CirfhqO@mail.gmail.com>

look for a sample in: freeswitch/src/mod/endpoints/mod_h323/h323.conf.xml
after you update to latest git.

copy it into /usr/local/freeswitch/conf/autoload_config/ directory and start
FS.




On Tue, Feb 1, 2011 at 10:34 PM, Steven Ayre <steveayre at gmail.com> wrote:

> You do not use buildopal.sh - that installs opal (for mod_opal) not
> h323plus (for mod_h323). The wiki
>
> The h323.conf.xml you found is the sample config file... adjust it to suit
> you and place it in your conf/autoload_configs directory created when you
> install.
>
> -Steve
>
>
> On 1 February 2011 19:11, Tim King <tim.compnetwork at gmail.com> wrote:
>
>> I am following http://wiki.freeswitch.org/wiki/Mod_h323 and trying to get
>>> H.323 setup. I installed using buildopal.sh script in the build directory.I
>>> am not finding h323.conf.xml anywhere except the sample in
>>> /usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/h323.conf.xml. Do
>>> I need to create this file and if so where should it be?
>>>
>>
>> Thanks
>>
>> 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 brad at tritelcomm.com  Wed Feb  2 01:51:20 2011
From: brad at tritelcomm.com (Brad Mina)
Date: Tue, 1 Feb 2011 14:51:20 -0800
Subject: [Freeswitch-users] Modify Registration String to ITSP SipStation
In-Reply-To: <AANLkTi=wD5doBH0V48UKyQcK8Atg_=QN2V6Xw0RuTxRZ@mail.gmail.com>
References: <AANLkTi=nxkFBxtnbWC2M-rJPQTK3awdVTSSDz9Mesymr@mail.gmail.com>
	<AANLkTi=wD5doBH0V48UKyQcK8Atg_=QN2V6Xw0RuTxRZ@mail.gmail.com>
Message-ID: <AANLkTinSLk5PzcEVVMDOvSNXp39gv2xsHGrFAM2ZsgHD@mail.gmail.com>

I tried putting username:password in the username field, however I'm still
getting SIP 403 messages back from the registrar.

For reference, from the site (FreePBX and SipStation are closely tied):

The last bit we need to configure is the registration string so that the
> FreePBX.com servers know where to send your inbound calls. This is simply
> derived from you *SIP Username* and *SIP Password*. You should be
> registering to *trunk1.freepbx.com* as currently registrations to *
> trunk2.freepbx.com* will not function:
> http://www.freepbx.org/files/images/freepbx_trunk_config_4-0002.jpg
>



> On Tue, Feb 1, 2011 at 12:01 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> They want you to send it in clear text as part of your username?
> I guess you would put the entire user:pass in the username field and
> leave the password blank?
>
>
> On Tue, Feb 1, 2011 at 1:57 PM, Brad Mina <brad at tritelcomm.com> wrote:
> > I'm having trouble registering a SipStation account. They require
> > username:password at trunk1.freepbx.com<username%3Apassword at trunk1.freepbx.com>to be sent to the registrar. I've done
> > a bit of searching but was unable to find any specific details on how to
> do
> > this. My config is as follows:
> >
> > <include>
> >     <gateway name="SipStation">
> >       <param name="username" value="myUsername"/>
> >       <param name="password" value="somePassword"/>
> >       <param name="realm" value="trunk1.freepbx.com"/>
> >       <param name="from-user" value="TriTelComm"/>
> >       <param name="proxy" value="trunk1.freepbx.com"/>
> >       <param name="expire-seconds" value="800"/>
> >       <param name="register" value="true"/>
> >       <param name="retry-seconds" value="60"/>
> >       <param name="context" value="public"/>
> >     </gateway>
> > </include>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Feb  2 03:38:48 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Wed, 2 Feb 2011 07:38:48 +0700
Subject: [Freeswitch-users] dingalingissue
In-Reply-To: <1296444370502-5976081.post@n2.nabble.com>
References: <AANLkTikbhmD4sB=tOtkYL4oy4Hw7HYf_DAYyYC7Wp7Hk@mail.gmail.com>
	<AANLkTinCDE+1655D+c4T9NwSzLBFot-FDREZ677OJzDt@mail.gmail.com>
	<1296444370502-5976081.post@n2.nabble.com>
Message-ID: <AANLkTinOBBZ-CaQg4TX2BAS-HwF+0BQPpf3CUfstGSkz@mail.gmail.com>

i use FreeSWITCH Version 1.0.head (git-43dd776 2011-02-01 16-36-02 -0600)
and this issue still happen.


On Mon, Jan 31, 2011 at 10:26 AM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
>
> Chris Chen-4 wrote:
> >
> > I have the same issue as you since the official tarball release of 1.0.7,
> > the mod_dingaling.c establish the jingle session, however it loops
> between
> > line 2941 and 3275 until it times out without any RTP traffic.
> > 2011-01-30 20:52:26.973945 [DEBUG] mod_dingaling.c:2941 using Existing
> > session for 3028119339
> > 2011-01-30 20:52:26.973945 [DEBUG] mod_dingaling.c:3275 Already picked an
> > IP
> > [99.xxx.xxx.xxx]
> > I tested with google talk to SIP, or gmail call out using mod_dingaling,
> > both have the same behavior.
> >
> > It was working st least up to Jan 11 2011. I had this issue since
> official
> > release of tarball 1.0.7 and every latest GIT version I tested up to
> > today.
> >
> > Best regards,
> >
> > Chris Chen
> I can confirm what you said above. However, I have updated/recompiled my
> local FS git repository yesterday and now it looks like mod_dingaling is
> back working again. Mine is from a FreeSWITCH Version 1.0.head (git-49a5eff
> 2011-01-29 03-09-06 -0500) on a Seagate DockStar. The only problem my FS is
> facing is no audio in both ways if the caller places a call to my GV DID#
> from the Google Chat inside GMail.
>
> -----
> 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/dingalingissue-tp5975836p5976081.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 mstockton at harqen.com  Wed Feb  2 03:29:12 2011
From: mstockton at harqen.com (Matt Stockton)
Date: Tue, 1 Feb 2011 18:29:12 -0600
Subject: [Freeswitch-users] Problem with missing / duplicated DTMF
In-Reply-To: <AANLkTinODuGMpAs4mY_29o_FavBw_3t1oVdksjWoAfb9@mail.gmail.com>
References: <AANLkTin+-NCnROr5ZE9GYoao_qVP_-2K7mWqhvJ+dkd3@mail.gmail.com>
	<AANLkTi=zyr6+n1+Qk9AWqvpUCNLuWPnf1CiXKYQk8v=Q@mail.gmail.com>
	<AANLkTinODuGMpAs4mY_29o_FavBw_3t1oVdksjWoAfb9@mail.gmail.com>
Message-ID: <AANLkTimJyanCodALy4dD+iSt7N02sHYyDWXNW8P5YtGN@mail.gmail.com>

Thanks Anthony.

We just did some additional testing against a different SIP provider, and we
were not able to re-produce the issue with that SIP provider. It was working
correctly 100% of the time.

We are going to open a ticket with iCall to try to understand the issue.

On Tue, Feb 1, 2011 at 12:22 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I dont think there any current dtmf issues open.
> It sounds like maybe you are going across the pstn and encountering
> some problems with transition from
> 2833 to inband and back again or from hair-pinning the call.
>
>
>
> On Tue, Feb 1, 2011 at 12:00 PM, Matt Stockton <mstockton at harqen.com>
> wrote:
> > Sorry about not including the version. The version of freeswitch I am
> using
> > is.
> > FreeSWITCH Version 1.0.head (git-256a82d 2011-01-31 10-12-28 -0600)
> >  I just updated to the latest yesterday to re-test it.
> > On Tue, Feb 1, 2011 at 10:10 AM, Matt Stockton <mstockton at harqen.com>
> wrote:
> >>
> >> I have having trouble with both missing and duplicated DTMF in
> >> Freeswitch.
> >> Here are the steps of how I am using it:
> >> 1. Leg A - I am calling out from my Freeswitch instance (through iCall),
> >> and I am calling an iCall number that is also connected to the same
> >> Freeswitch instance.
> >> 2. Leg B - The above call is routed through iCall and then answered by
> the
> >> same Freeswitch instance.
> >> 3. On Leg B, I play a file and attempt to get DTMF in a lua script. Here
> >> is the code of interest:
> >>
> >> callPin = session:playAndGetDigits(1, 10, 4, 30000,
> "#","/tmp/cw_17.wav",
> >> "", "\\d+");
> >>
> >> 4. On Leg A, I send DTMF information in a lua script. Here is the code
> of
> >> interest. I initiate a delay between each digit:
> >>
> >> local newPin = "";
> >>
> >> for i = 1, string.len(pin) do
> >>
> >>   newPin = newPin .. string.sub(pin, i, i) .. "W";
> >>
> >> end
> >>
> >> session:execute("send_dtmf", newPin .. "#@200");
> >>
> >> ** Note that there is a session:sleep on Leg A before I send the DTMF to
> >> make sure i don't send it too early **
> >>
> >> The problem is that the recognized DTMF on Leg B is wrong about 30% of
> the
> >> time. For example, if Leg A enters: 22063083, Leg B will get the DTMF
> digits
> >> 222063083. This is an example of duplication, but I have also
> experienced
> >> missing DTMF codes (and an occasional wrong code completely)
> >>
> >> I have messed with a bunch of DTMF settings in hopes of fixing this
> issue,
> >> but I cannot seem to find something that is reliable 100% of the time.
> >>
> >> _____________________________
> >>
> >> Here are the DTMF settings I have looked at / messed with. I've tried
> >> various values for the dtmf-duration in the config (and in the send_dtmf
> >> command above)
> >>
> >>     <!-- This is in external.xml -->
> >>
> >>     <param name="dtmf-duration" value="500"/>
> >>
> >>     <param name="dtmf-type" value="rfc2833"/>
> >>
> >>     <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
> >>
> >>     <param name="outbound-codec-prefs"
> value="$${outbound_codec_prefs}"/>
> >>
> >>     <param name="inbound-codec-negotiation" value="greedy"/>
> >>     <!-- This is in vars.xml -->
> >>     <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU"/>
> >>     <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU"/>
> >> ________________________
> >> I have run fs_cli with event logging and the DTMF events that Freeswitch
> >> gets do correlate to the wrong value (e.g. the duplication / missing
> digits
> >> is noticable in the Freeswitch events as well).
> >> Also, I am not running any dtmf-related applications on the session
> before
> >> I give control to the lua scripts (e.g. not running start_dtmf)
> >> Has anyone experienced this type of issue? Or know what I can do to
> >> resolve it? My next step was going to be trying this against another
> >> provider besides iCall, but I figured I would see if anyone has
> encountered
> >> a similar problem before. Any help is appreciated.
> >> 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  2 05:27:29 2011
From: djbinter at gmail.com (DJB International)
Date: Tue, 1 Feb 2011 18:27:29 -0800
Subject: [Freeswitch-users] Polycom Issues
Message-ID: <AANLkTinTiKQX_H=oVFn1KcL9GfqvVxq=AE6+CM7jQPwb@mail.gmail.com>

Has anyone ever seen Polycom 650 send REGISTER without Authorization even
though FS responds with 401?

The weird thing was that it only happened to this particular Polycom phone
only.

Please see below:

   ------------------------------------------------------------------------
recv 559 bytes from udp/[67.232.144.163]:10058 at 02:24:22.100024:
   ------------------------------------------------------------------------
   REGISTER sip:199.87.44.19:5060 SIP/2.0
   Via: SIP/2.0/UDP 67.232.144.163:10003;branch=z9hG4bK85aaf1325070992B
   From: "Mark" <sip:7025143416 at 199.87.44.19 <sip%3A7025143416 at 199.87.44.19>
>;tag=EED068F-AC7EFF94
   To: <sip:7025143416 at 199.87.44.19 <sip%3A7025143416 at 199.87.44.19>>
   CSeq: 1 REGISTER
   Call-ID: 37fb35c0-279ee281-fd8694be at 67.232.144.163
   Contact: <sip:7025143416 at 67.232.144.163:10003>;methods="INVITE, ACK, BYE,
CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
   User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.4.0244
   Accept-Language: en
   Max-Forwards: 70
   Expires: 3600
   Content-Length: 0

   ------------------------------------------------------------------------
2011-02-01 18:24:28.858417 [WARNING] sofia_reg.c:1247 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [7025143416 at 199.87.44.19] from ip
67.232.144.163
send 676 bytes to udp/[67.232.144.163]:10058 at 02:24:22.174807:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP 67.232.144.163:10003
;branch=z9hG4bK85aaf1325070992B;rport=10058
   From: "Mark" <sip:7025143416 at 199.87.44.19 <sip%3A7025143416 at 199.87.44.19>
>;tag=EED068F-AC7EFF94
   To: <sip:7025143416 at 199.87.44.19 <sip%3A7025143416 at 199.87.44.19>
>;tag=5Byyp046mycjS
   Call-ID: 37fb35c0-279ee281-fd8694be at 67.232.144.163
   CSeq: 1 REGISTER
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9ffca05 2011-01-26 17-24-25
-0500
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   WWW-Authenticate: Digest realm="199.87.44.19",
nonce="4b7a72c9-47ad-4d84-b1a1-fe0f68194efd", algorithm=MD5, qop="auth"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 559 bytes from udp/[67.232.144.163]:10058 at 02:24:22.628250:
   ------------------------------------------------------------------------
   REGISTER sip:199.87.44.19:5060 SIP/2.0
   Via: SIP/2.0/UDP 67.232.144.163:10003;branch=z9hG4bK85aaf1325070992B
   From: "Mark" <sip:7025143416 at 199.87.44.19 <sip%3A7025143416 at 199.87.44.19>
>;tag=EED068F-AC7EFF94
   To: <sip:7025143416 at 199.87.44.19 <sip%3A7025143416 at 199.87.44.19>>
   CSeq: 1 REGISTER
   Call-ID: 37fb35c0-279ee281-fd8694be at 67.232.144.163
   Contact: <sip:7025143416 at 67.232.144.163:10003>;methods="INVITE, ACK, BYE,
CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
   User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.4.0244
   Accept-Language: en
   Max-Forwards: 70
   Expires: 3600
   Content-Length: 0

   ------------------------------------------------------------------------
send 676 bytes to udp/[67.232.144.163]:10058 at 02:24:22.628449:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP 67.232.144.163:10003
;branch=z9hG4bK85aaf1325070992B;rport=10058
   From: "Mark" <sip:7025143416 at 199.87.44.19 <sip%3A7025143416 at 199.87.44.19>
>;tag=EED068F-AC7EFF94
   To: <sip:7025143416 at 199.87.44.19 <sip%3A7025143416 at 199.87.44.19>
>;tag=5Byyp046mycjS
   Call-ID: 37fb35c0-279ee281-fd8694be at 67.232.144.163
   CSeq: 1 REGISTER
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9ffca05 2011-01-26 17-24-25
-0500
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   WWW-Authenticate: Digest realm="199.87.44.19",
nonce="4b7a72c9-47ad-4d84-b1a1-fe0f68194efd", algorithm=MD5, qop="auth"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 559 bytes from udp/[67.232.144.163]:10058 at 02:24:23.628660:
   ------------------------------------------------------------------------
   REGISTER sip:199.87.44.19:5060 SIP/2.0
   Via: SIP/2.0/UDP 67.232.144.163:10003;branch=z9hG4bK85aaf1325070992B
   From: "Mark" <sip:7025143416 at 199.87.44.19 <sip%3A7025143416 at 199.87.44.19>
>;tag=EED068F-AC7EFF94
   To: <sip:7025143416 at 199.87.44.19 <sip%3A7025143416 at 199.87.44.19>>
   CSeq: 1 REGISTER
   Call-ID: 37fb35c0-279ee281-fd8694be at 67.232.144.163
   Contact: <sip:7025143416 at 67.232.144.163:10003>;methods="INVITE, ACK, BYE,
CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
   User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.4.0244
   Accept-Language: en
   Max-Forwards: 70
   Expires: 3600
   Content-Length: 0

   ------------------------------------------------------------------------
send 676 bytes to udp/[67.232.144.163]:10058 at 02:24:23.628895:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP 67.232.144.163:10003
;branch=z9hG4bK85aaf1325070992B;rport=10058
   From: "Mark" <sip:7025143416 at 199.87.44.19 <sip%3A7025143416 at 199.87.44.19>
>;tag=EED068F-AC7EFF94
   To: <sip:7025143416 at 199.87.44.19 <sip%3A7025143416 at 199.87.44.19>
>;tag=5Byyp046mycjS
   Call-ID: 37fb35c0-279ee281-fd8694be at 67.232.144.163
   CSeq: 1 REGISTER
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9ffca05 2011-01-26 17-24-25
-0500
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   WWW-Authenticate: Digest realm="199.87.44.19",
nonce="4b7a72c9-47ad-4d84-b1a1-fe0f68194efd", algorithm=MD5, qop="auth"
   Content-Length: 0


Thank you.
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From sos at sokhapkin.dyndns.org  Wed Feb  2 05:35:41 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 1 Feb 2011 21:35:41 -0500
Subject: [Freeswitch-users] Polycom Issues
In-Reply-To: <AANLkTinTiKQX_H=oVFn1KcL9GfqvVxq=AE6+CM7jQPwb@mail.gmail.com>
References: <AANLkTinTiKQX_H=oVFn1KcL9GfqvVxq=AE6+CM7jQPwb@mail.gmail.com>
Message-ID: <201102012135.41920.sos@sokhapkin.dyndns.org>

The phone doesn't get FS responses back, note the same cseq and call id in 
REGISTER. NAT problem?

On Tuesday 01 February 2011, DJB International wrote:
> Has anyone ever seen Polycom 650 send REGISTER without Authorization even
> though FS responds with 401?
> 
> The weird thing was that it only happened to this particular Polycom phone
> only.
> 
> Please see below:
> 
>    ------------------------------------------------------------------------
> recv 559 bytes from udp/[67.232.144.163]:10058 at 02:24:22.100024:
>    ------------------------------------------------------------------------
>    REGISTER sip:199.87.44.19:5060 SIP/2.0
>    Via: SIP/2.0/UDP 67.232.144.163:10003;branch=z9hG4bK85aaf1325070992B
>    From: "Mark" <sip:7025143416 at 199.87.44.19
> <sip%3A7025143416 at 199.87.44.19>
> 
> >;tag=EED068F-AC7EFF94
> 
>    To: <sip:7025143416 at 199.87.44.19 <sip%3A7025143416 at 199.87.44.19>>
>    CSeq: 1 REGISTER
>    Call-ID: 37fb35c0-279ee281-fd8694be at 67.232.144.163
>    Contact: <sip:7025143416 at 67.232.144.163:10003>;methods="INVITE, ACK,
> BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE,
> REFER" User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.4.0244
>    Accept-Language: en
>    Max-Forwards: 70
>    Expires: 3600
>    Content-Length: 0
> 
>    ------------------------------------------------------------------------
> 2011-02-01 18:24:28.858417 [WARNING] sofia_reg.c:1247 SIP auth challenge
> (REGISTER) on sofia profile 'internal' for [7025143416 at 199.87.44.19] from
> ip 67.232.144.163
> send 676 bytes to udp/[67.232.144.163]:10058 at 02:24:22.174807:
>    ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/UDP 67.232.144.163:10003
> ;branch=z9hG4bK85aaf1325070992B;rport=10058
>    From: "Mark" <sip:7025143416 at 199.87.44.19
> <sip%3A7025143416 at 199.87.44.19>
> 
> >;tag=EED068F-AC7EFF94
> 
>    To: <sip:7025143416 at 199.87.44.19 <sip%3A7025143416 at 199.87.44.19>
> 
> >;tag=5Byyp046mycjS
> 
>    Call-ID: 37fb35c0-279ee281-fd8694be at 67.232.144.163
>    CSeq: 1 REGISTER
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9ffca05 2011-01-26
> 17-24-25 -0500
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, precondition, path, replaces
>    WWW-Authenticate: Digest realm="199.87.44.19",
> nonce="4b7a72c9-47ad-4d84-b1a1-fe0f68194efd", algorithm=MD5, qop="auth"
>    Content-Length: 0
> 
>    ------------------------------------------------------------------------
> recv 559 bytes from udp/[67.232.144.163]:10058 at 02:24:22.628250:
>    ------------------------------------------------------------------------
>    REGISTER sip:199.87.44.19:5060 SIP/2.0
>    Via: SIP/2.0/UDP 67.232.144.163:10003;branch=z9hG4bK85aaf1325070992B
>    From: "Mark" <sip:7025143416 at 199.87.44.19
> <sip%3A7025143416 at 199.87.44.19>
> 
> >;tag=EED068F-AC7EFF94
> 
>    To: <sip:7025143416 at 199.87.44.19 <sip%3A7025143416 at 199.87.44.19>>
>    CSeq: 1 REGISTER
>    Call-ID: 37fb35c0-279ee281-fd8694be at 67.232.144.163
>    Contact: <sip:7025143416 at 67.232.144.163:10003>;methods="INVITE, ACK,
> BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE,
> REFER" User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.4.0244
>    Accept-Language: en
>    Max-Forwards: 70
>    Expires: 3600
>    Content-Length: 0
> 
>    ------------------------------------------------------------------------
> send 676 bytes to udp/[67.232.144.163]:10058 at 02:24:22.628449:
>    ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/UDP 67.232.144.163:10003
> ;branch=z9hG4bK85aaf1325070992B;rport=10058
>    From: "Mark" <sip:7025143416 at 199.87.44.19
> <sip%3A7025143416 at 199.87.44.19>
> 
> >;tag=EED068F-AC7EFF94
> 
>    To: <sip:7025143416 at 199.87.44.19 <sip%3A7025143416 at 199.87.44.19>
> 
> >;tag=5Byyp046mycjS
> 
>    Call-ID: 37fb35c0-279ee281-fd8694be at 67.232.144.163
>    CSeq: 1 REGISTER
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9ffca05 2011-01-26
> 17-24-25 -0500
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, precondition, path, replaces
>    WWW-Authenticate: Digest realm="199.87.44.19",
> nonce="4b7a72c9-47ad-4d84-b1a1-fe0f68194efd", algorithm=MD5, qop="auth"
>    Content-Length: 0
> 
>    ------------------------------------------------------------------------
> recv 559 bytes from udp/[67.232.144.163]:10058 at 02:24:23.628660:
>    ------------------------------------------------------------------------
>    REGISTER sip:199.87.44.19:5060 SIP/2.0
>    Via: SIP/2.0/UDP 67.232.144.163:10003;branch=z9hG4bK85aaf1325070992B
>    From: "Mark" <sip:7025143416 at 199.87.44.19
> <sip%3A7025143416 at 199.87.44.19>
> 
> >;tag=EED068F-AC7EFF94
> 
>    To: <sip:7025143416 at 199.87.44.19 <sip%3A7025143416 at 199.87.44.19>>
>    CSeq: 1 REGISTER
>    Call-ID: 37fb35c0-279ee281-fd8694be at 67.232.144.163
>    Contact: <sip:7025143416 at 67.232.144.163:10003>;methods="INVITE, ACK,
> BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE,
> REFER" User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.4.0244
>    Accept-Language: en
>    Max-Forwards: 70
>    Expires: 3600
>    Content-Length: 0
> 
>    ------------------------------------------------------------------------
> send 676 bytes to udp/[67.232.144.163]:10058 at 02:24:23.628895:
>    ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/UDP 67.232.144.163:10003
> ;branch=z9hG4bK85aaf1325070992B;rport=10058
>    From: "Mark" <sip:7025143416 at 199.87.44.19
> <sip%3A7025143416 at 199.87.44.19>
> 
> >;tag=EED068F-AC7EFF94
> 
>    To: <sip:7025143416 at 199.87.44.19 <sip%3A7025143416 at 199.87.44.19>
> 
> >;tag=5Byyp046mycjS
> 
>    Call-ID: 37fb35c0-279ee281-fd8694be at 67.232.144.163
>    CSeq: 1 REGISTER
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9ffca05 2011-01-26
> 17-24-25 -0500
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, precondition, path, replaces
>    WWW-Authenticate: Digest realm="199.87.44.19",
> nonce="4b7a72c9-47ad-4d84-b1a1-fe0f68194efd", algorithm=MD5, qop="auth"
>    Content-Length: 0
> 
> 
> Thank you.



From david.ponzone at ipeva.fr  Wed Feb  2 05:39:22 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 2 Feb 2011 03:39:22 +0100
Subject: [Freeswitch-users] Polycom Issues
In-Reply-To: <AANLkTinTiKQX_H=oVFn1KcL9GfqvVxq=AE6+CM7jQPwb@mail.gmail.com>
References: <AANLkTinTiKQX_H=oVFn1KcL9GfqvVxq=AE6+CM7jQPwb@mail.gmail.com>
Message-ID: <86E6544D-9446-45F6-8FF7-71A24C8294C3@ipeva.fr>

Well, I don't use Polycom, but I've seen such behaviour in the past with other phones, and most of the time, the cause is that the phone is not receiving the 401 Challenge from FreeSWITCH, so it sends back another REGISTER.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 02/02/2011 ? 03:27, DJB International a ?crit :

> Has anyone ever seen Polycom 650 send REGISTER without Authorization even though FS responds with 401?
> 
> The weird thing was that it only happened to this particular Polycom phone only.
> 
> Please see below:
> 
>    ------------------------------------------------------------------------
> recv 559 bytes from udp/[67.232.144.163]:10058 at 02:24:22.100024:
>    ------------------------------------------------------------------------
>    REGISTER sip:199.87.44.19:5060 SIP/2.0
>    Via: SIP/2.0/UDP 67.232.144.163:10003;branch=z9hG4bK85aaf1325070992B
>    From: "Mark" <sip:7025143416 at 199.87.44.19>;tag=EED068F-AC7EFF94
>    To: <sip:7025143416 at 199.87.44.19>
>    CSeq: 1 REGISTER
>    Call-ID: 37fb35c0-279ee281-fd8694be at 67.232.144.163
>    Contact: <sip:7025143416 at 67.232.144.163:10003>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
>    User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.4.0244
>    Accept-Language: en
>    Max-Forwards: 70
>    Expires: 3600
>    Content-Length: 0
>    
>    ------------------------------------------------------------------------
> 2011-02-01 18:24:28.858417 [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile 'internal' for [7025143416 at 199.87.44.19] from ip 67.232.144.163
> send 676 bytes to udp/[67.232.144.163]:10058 at 02:24:22.174807:
>    ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/UDP 67.232.144.163:10003;branch=z9hG4bK85aaf1325070992B;rport=10058
>    From: "Mark" <sip:7025143416 at 199.87.44.19>;tag=EED068F-AC7EFF94
>    To: <sip:7025143416 at 199.87.44.19>;tag=5Byyp046mycjS
>    Call-ID: 37fb35c0-279ee281-fd8694be at 67.232.144.163
>    CSeq: 1 REGISTER
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9ffca05 2011-01-26 17-24-25 -0500
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, precondition, path, replaces
>    WWW-Authenticate: Digest realm="199.87.44.19", nonce="4b7a72c9-47ad-4d84-b1a1-fe0f68194efd", algorithm=MD5, qop="auth"
>    Content-Length: 0
>    
>    ------------------------------------------------------------------------
> recv 559 bytes from udp/[67.232.144.163]:10058 at 02:24:22.628250:
>    ------------------------------------------------------------------------
>    REGISTER sip:199.87.44.19:5060 SIP/2.0
>    Via: SIP/2.0/UDP 67.232.144.163:10003;branch=z9hG4bK85aaf1325070992B
>    From: "Mark" <sip:7025143416 at 199.87.44.19>;tag=EED068F-AC7EFF94
>    To: <sip:7025143416 at 199.87.44.19>
>    CSeq: 1 REGISTER
>    Call-ID: 37fb35c0-279ee281-fd8694be at 67.232.144.163
>    Contact: <sip:7025143416 at 67.232.144.163:10003>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
>    User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.4.0244
>    Accept-Language: en
>    Max-Forwards: 70
>    Expires: 3600
>    Content-Length: 0
>    
>    ------------------------------------------------------------------------
> send 676 bytes to udp/[67.232.144.163]:10058 at 02:24:22.628449:
>    ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/UDP 67.232.144.163:10003;branch=z9hG4bK85aaf1325070992B;rport=10058
>    From: "Mark" <sip:7025143416 at 199.87.44.19>;tag=EED068F-AC7EFF94
>    To: <sip:7025143416 at 199.87.44.19>;tag=5Byyp046mycjS
>    Call-ID: 37fb35c0-279ee281-fd8694be at 67.232.144.163
>    CSeq: 1 REGISTER
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9ffca05 2011-01-26 17-24-25 -0500
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, precondition, path, replaces
>    WWW-Authenticate: Digest realm="199.87.44.19", nonce="4b7a72c9-47ad-4d84-b1a1-fe0f68194efd", algorithm=MD5, qop="auth"
>    Content-Length: 0
>    
>    ------------------------------------------------------------------------
> recv 559 bytes from udp/[67.232.144.163]:10058 at 02:24:23.628660:
>    ------------------------------------------------------------------------
>    REGISTER sip:199.87.44.19:5060 SIP/2.0
>    Via: SIP/2.0/UDP 67.232.144.163:10003;branch=z9hG4bK85aaf1325070992B
>    From: "Mark" <sip:7025143416 at 199.87.44.19>;tag=EED068F-AC7EFF94
>    To: <sip:7025143416 at 199.87.44.19>
>    CSeq: 1 REGISTER
>    Call-ID: 37fb35c0-279ee281-fd8694be at 67.232.144.163
>    Contact: <sip:7025143416 at 67.232.144.163:10003>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
>    User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.4.0244
>    Accept-Language: en
>    Max-Forwards: 70
>    Expires: 3600
>    Content-Length: 0
>    
>    ------------------------------------------------------------------------
> send 676 bytes to udp/[67.232.144.163]:10058 at 02:24:23.628895:
>    ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/UDP 67.232.144.163:10003;branch=z9hG4bK85aaf1325070992B;rport=10058
>    From: "Mark" <sip:7025143416 at 199.87.44.19>;tag=EED068F-AC7EFF94
>    To: <sip:7025143416 at 199.87.44.19>;tag=5Byyp046mycjS
>    Call-ID: 37fb35c0-279ee281-fd8694be at 67.232.144.163
>    CSeq: 1 REGISTER
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9ffca05 2011-01-26 17-24-25 -0500
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, precondition, path, replaces
>    WWW-Authenticate: Digest realm="199.87.44.19", nonce="4b7a72c9-47ad-4d84-b1a1-fe0f68194efd", algorithm=MD5, qop="auth"
>    Content-Length: 0
> 
> 
> Thank you.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Feb  2 06:00:35 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 1 Feb 2011 19:00:35 -0800
Subject: [Freeswitch-users] open g729
In-Reply-To: <C96DB334.46A6%krice@freeswitch.org>
References: <AANLkTikE3viP6KN+28cRzx=39cLQJ+OJZU64xi27UBB9@mail.gmail.com>
	<C96DB334.46A6%krice@freeswitch.org>
Message-ID: <AANLkTinVX-_UYKhCaBi_1Lh037SQgKZCaS2uM1UgDM+C@mail.gmail.com>

If hardware doesn't support it, make one that supports Speex right out
of the box. I'm pretty sure that somebody will do it.

On Tue, Feb 1, 2011 at 10:55 AM, Ken Rice <krice at freeswitch.org> wrote:
> Why people insist on speex I am not sure... Sure its nice on a closed
> network but quickly looses its luster when communicating with the real
> world... Not to mention the CPU usages and such...
>
> If you want to use something like speex check out iLBC and theres things
> like the Sangoma D100/D500 that you can use to offload the transcoding from
> the CPU (disclaimer I sell the sangoma hardware)
>
> K
>
>
> On 2/1/11 12:45 PM, "Kristian Kielhofner" <kris at kriskinc.com> wrote:
>
>> Phones, carriers, and just about any commercial gear (reference
>> hardware, SBCs, DSPs, etc) you'll ever see doesn't support Speex.
>> Guess how you'll make it work? ?Transcoding from G729 to Speex. ?Now
>> you're using licenses, proxying RTP, and wasting CPU while ruining
>> voice quality transcoding from one lossy codec to another. ?It's just
>> about the worst possible situation you could be in.
>>
>> On Tue, Feb 1, 2011 at 1:19 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>> Except that many hardware phones don't have speex, so the only low-bandwidth
>>> solution is g729...
>>> On Tue, Feb 1, 2011 at 7:43 PM, curriegrad2004 <curriegrad2004 at gmail.com>
>>> wrote:
>>>>
>>>> Simple solution to this g.729 patent non-sense: Speex.
>>>>
>>>> 'nuff said.
>>>>
>>>> On Tue, Feb 1, 2011 at 8:26 AM, Anton VG <anton.vazir at gmail.com> wrote:
>>>>> Considering the above, noone can say "Purchase a LICENSE", but only a
>>>>> "Licensed SOFTWARE"
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 marcin321 at gmail.com  Wed Feb  2 06:44:35 2011
From: marcin321 at gmail.com (Marcin Wojtowicz)
Date: Tue, 1 Feb 2011 22:44:35 -0500
Subject: [Freeswitch-users] Outbound only calls don't connect when
 bypass_media is true.
Message-ID: <AANLkTikBPf3EwhaD7r1atVLDdfe5dteNe36ZW2F0yVGg@mail.gmail.com>

I've found a solution file that creates the pcre_chartables.c source and
compiled freeswitch from the latest git. The problem still remains, and it
looks like it was brought up by somebody else here:
http://lists.freeswitch.org/pipermail/freeswitch-users/2010-October/063891.html




Here are the build errors, in order of appearance:
------ Build started: Project: libpcre, Configuration: Release Win32 ------
  pcre_chartables.c
c1 : fatal error C1083: Cannot open source file: 'pcre_chartables.c': No
such file or directory
(other files here compile ok)

------ Build started: Project: FreeSwitchCoreLib, Configuration: Release
Win32 ------
  Generating switch_version.h
  (stuff here goes ok)
  Generating Code...
LINK : fatal error LNK1181: cannot open input file
'E:\downloads\freeswitch-1.0.7\libs\win32\pcre\Win32\Release\libpcre.lib'

------ Build started: Project: mod_spidermonkey, Configuration: Release
Win32 ------
  mod_spidermonkey.c
LINK : fatal error LNK1181: cannot open input file
'E:\downloads\freeswitch-1.0.7\Win32\Release\FreeSwitchCore.lib'

FreeSwitchCore link error repeats many times for other projects, too many to
list. It doesn't build.

------ Build started: Project: mod_managed, Configuration: Release_CLR Win32
------
  freeswitch_managed.cpp
freeswitch_managed.cpp : fatal error C1192: #using failed on
'E:\downloads\freeswitch-1.0.7\Win32\Debug\mod\FreeSWITCH.Managed.dll'
          'The system cannot find the path specified.'
  freeswitch_wrap.2010.cxx
freeswitch_wrap.2010.cxx : fatal error C1192: #using failed on
'E:\downloads\freeswitch-1.0.7\Win32\Debug\mod\FreeSWITCH.Managed.dll'
          'The system cannot find the path specified.'
  mod_managed.cpp
mod_managed.cpp : fatal error C1192: #using failed on
'E:\downloads\freeswitch-1.0.7\Win32\Debug\mod\FreeSWITCH.Managed.dll'
          'The system cannot find the path specified.'
  Generating Code...


The code I downloaded was from latest.freeswitch.org, the tar.bz2 zip file.
>From the dependencies it looks like FreeSwitchCore depends on libpcre, so
all this cascades from the missing pcre_chartables.c file. Can you send it
to me?
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From infos at madovsky.org  Wed Feb  2 07:20:54 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 1 Feb 2011 23:20:54 -0500
Subject: [Freeswitch-users] voicemail.tpl and sip address
Message-ID: <3A52CE419598478FAE4842B037A61AFC@e1705>

which var can I use in voicemail.tpl to write
correctly a sip address ?
actually vocemail_caller_id_number shows only th username and not the domain

Thanks
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From ayhkor at gmail.com  Wed Feb  2 07:25:09 2011
From: ayhkor at gmail.com (deniro)
Date: Tue, 1 Feb 2011 23:25:09 -0500
Subject: [Freeswitch-users] mod_perl mod_lua
Message-ID: <AANLkTikxbN47Nc7cG8y0HMDVqC0HHJ56aLQZHJYS2WTR@mail.gmail.com>

Hi
I have compiled version of freeswitch 1.6.x on ubuntu 10.04
I want to install extra freeswitch modules like mod_perl and mod_lua
Is there any way to install these modules without re-compileling freeswitch.
I looked at freeswitch site and it is talking about  editing modules.conf
file and compiling from source
and I dont even have modules.conf file as I searched.

thx
deniro--
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From chat2jesse at gmail.com  Wed Feb  2 08:00:12 2011
From: chat2jesse at gmail.com (jesse)
Date: Tue, 1 Feb 2011 21:00:12 -0800
Subject: [Freeswitch-users] keeps receiving 407
In-Reply-To: <AANLkTinWqR0PwrhjScdaBzXb4UoH5CmfzzVxwFrQWk5B@mail.gmail.com>
References: <AANLkTinWqR0PwrhjScdaBzXb4UoH5CmfzzVxwFrQWk5B@mail.gmail.com>
Message-ID: <AANLkTikEJsmuY6uo733BXSuGi-Y75d1qk_7VPN0WOkEo@mail.gmail.com>

bump!!!  my question was posted 2 days ago. It still has not appeared in my
inbox.

-jesse

On Sun, Jan 30, 2011 at 10:00 PM, jesse <chat2jesse at gmail.com> wrote:

> I set up a sipp uac client. here is the call flow messages:
> http://freeswitch.pastebin.com/Sx5Z4egM
> why does FS send 407 after 200 OK?
>
> thanks!
>
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From infos at madovsky.org  Wed Feb  2 08:19:19 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 2 Feb 2011 00:19:19 -0500
Subject: [Freeswitch-users] RTP jitter basic tutorial
Message-ID: <E96400A830E44858B8EE304B4CC9C451@e1705>

interesting

http://toncar.cz/Tutorials/VoIP/VoIP_Basics_Jitter.html
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From mitch.capper at gmail.com  Wed Feb  2 09:39:38 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Tue, 1 Feb 2011 22:39:38 -0800
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client for
	Windows in .NET
Message-ID: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>

In prep for tomorrows conference I am happy to release FSClient.  I would
like to thank DRK and especially jlenk for their continued support and help
on this project.  jlenk created the installer, fixed code, and certainly
helped bring it along quite a bit faster!
FSClient is a windows SIP client:
Most features of any standard sip client
multiple calls at once
transfer, holding, speakerphone, DND
multiple SIP accounts
advanced headset support (caller id, buttons, etc) for jabra and plantronics
out of the box (with plugin support for easily adding others)
basic contact book support (with a sample XML contact book plugin provided)
All codecs (minus commercial g729 support) that freeswitch supports

Give it a shot and let us know about any issues you run into, overall it was
decently tested for many months but some of the rapid changes as of late
means there may be some larger bugs that crept in.  I will also be on the
conference call tomorrow and can answer questions there.

And so to download the binaries:
 http://files.freeswitch.org/windows/installer/x86/FSClient.zip
There will be a readme in the install dir on usage.

The source code is available on the contrib git repo mitchcapper/FSClient.

As for building from source you will need to do a few things to trunk
currently.  In the source folder you need to move mod_portaudio.c to the
freeswitch\src\mod\endpoints\mod_portaudio folder (
http://jira.freeswitch.org/browse/FS-3006 are the changes pending trunk).
In addition move the portaudio.2010.vxcproj over the version in
freeswitch\libs\portaudio\build\msvc to enable direct X support.  Make sure
you have the WPF Toolkit also installed.  Build trunk and then for FSClient
set the ENV var FREESWITCH_SRC_LOCATION to the src location and it will also
auto-copy the needed files into the build folder.


History:
Back in May I wanted a SIP client that supported the Jabra headsets, after
not finding one that would work as I liked I looked to open source and
everyone pointed me towards FSComm.  FSComm is great, and has a huge amount
of time poured into it, but the C++/QT base for it was not for me.  .NET was
the natural choice due to its rapid development time and stability so I went
there.

After a few weeks work I had a client that I used for over 6 months and was
fairly stable with relatively minor changes.   A few weeks ago on the
conference call someone mentioned wanting a windows sip client option and I
figured I would offer up my code base.  Well getting private code ready for
public use was a bit more complex than planned but its here now and
certainly in the past several weeks has had a lot of development.

~Mitch
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From daniel.neubert at solomo.de  Wed Feb  2 11:06:00 2011
From: daniel.neubert at solomo.de (Daniel Neubert)
Date: Wed, 2 Feb 2011 09:06:00 +0100
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
 for	Windows in .NET
In-Reply-To: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
Message-ID: <4D491068.7090308@solomo.de>

Sounds great - but unfortunately installation fails with this error message:



Test environment was Windows Vista Business SP2 X64 installed as virtual machine.


Best regards / Mit freundlichen Gr??en,
Daniel Neubert



On 02.02.2011 07:39, Mitch Capper wrote:
> In prep for tomorrows conference I am happy to release FSClient.  I would like to thank DRK and especially jlenk for their continued support and help on this project.  jlenk created the installer, fixed code, and certainly helped bring it along 
> quite a bit faster!
> FSClient is a windows SIP client:
> Most features of any standard sip client
> multiple calls at once
> transfer, holding, speakerphone, DND
> multiple SIP accounts
> advanced headset support (caller id, buttons, etc) for jabra and plantronics out of the box (with plugin support for easily adding others)
> basic contact book support (with a sample XML contact book plugin provided)
> All codecs (minus commercial g729 support) that freeswitch supports
>
> Give it a shot and let us know about any issues you run into, overall it was decently tested for many months but some of the rapid changes as of late means there may be some larger bugs that crept in.  I will also be on the conference call tomorrow 
> and can answer questions there.
>
> And so to download the binaries:
> http://files.freeswitch.org/windows/installer/x86/FSClient.zip
> There will be a readme in the install dir on usage.
>
> The source code is available on the contrib git repo mitchcapper/FSClient.
>
> As for building from source you will need to do a few things to trunk currently.  In the source folder you need to move mod_portaudio.c to the freeswitch\src\mod\endpoints\mod_portaudio folder (http://jira.freeswitch.org/browse/FS-3006 are the 
> changes pending trunk).  In addition move the portaudio.2010.vxcproj over the version in freeswitch\libs\portaudio\build\msvc to enable direct X support.  Make sure you have the WPF Toolkit also installed.  Build trunk and then for FSClient set the 
> ENV var FREESWITCH_SRC_LOCATION to the src location and it will also auto-copy the needed files into the build folder.
>
>
> History:
> Back in May I wanted a SIP client that supported the Jabra headsets, after not finding one that would work as I liked I looked to open source and everyone pointed me towards FSComm.  FSComm is great, and has a huge amount of time poured into it, 
> but the C++/QT base for it was not for me.  .NET was the natural choice due to its rapid development time and stability so I went there.
>
> After a few weeks work I had a client that I used for over 6 months and was fairly stable with relatively minor changes.   A few weeks ago on the conference call someone mentioned wanting a windows sip client option and I figured I would offer up 
> my code base.  Well getting private code ready for public use was a bit more complex than planned but its here now and certainly in the past several weeks has had a lot of development.
>
> ~Mitch
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From gmaruzz at gmail.com  Wed Feb  2 11:14:22 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 2 Feb 2011 09:14:22 +0100
Subject: [Freeswitch-users] mod_perl mod_lua
In-Reply-To: <AANLkTikxbN47Nc7cG8y0HMDVqC0HHJ56aLQZHJYS2WTR@mail.gmail.com>
References: <AANLkTikxbN47Nc7cG8y0HMDVqC0HHJ56aLQZHJYS2WTR@mail.gmail.com>
Message-ID: <AANLkTinDSQkOJrAikHrTEm6YTfyMcv2yLfxGPucjDwVq@mail.gmail.com>

On Wed, Feb 2, 2011 at 5:25 AM, deniro <ayhkor at gmail.com> wrote:
> Hi
> I have compiled version of freeswitch 1.6.x on ubuntu 10.04
> I want to install extra freeswitch modules like mod_perl and mod_lua
> Is there any way to install these modules without re-compileling freeswitch.
> I looked at freeswitch site and it is talking about? editing modules.conf
> file and compiling from source
> and I dont even have modules.conf file as I searched.

Go to your original sources directory (where you gave the command
"make install"), edit the file modules.conf.xml and give the command
"make install".

-giovanni



>
> thx
> 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
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From erik.dekkers at wvds.nl  Wed Feb  2 11:43:22 2011
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Wed, 2 Feb 2011 09:43:22 +0100
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
 for	Windows in .NET
In-Reply-To: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE0787301840@FS1.wvds.local>

I'm also having the error with security settings. Im on Windows 7 Pro X64. The PC is part of a domain.

Kind regards

Erik

Btw, if I need to debug, please contact me on irc . Nick wvds-nl

Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Mitch Capper
Verzonden: woensdag 2 februari 2011 7:40
Aan: FreeSWITCH Users Help
Onderwerp: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client for Windows in .NET

In prep for tomorrows conference I am happy to release FSClient.  I would like to thank DRK and especially jlenk for their continued support and help on this project.  jlenk created the installer, fixed code, and certainly helped bring it along quite a bit faster!
FSClient is a windows SIP client:
Most features of any standard sip client
multiple calls at once
transfer, holding, speakerphone, DND
multiple SIP accounts
advanced headset support (caller id, buttons, etc) for jabra and plantronics out of the box (with plugin support for easily adding others)
basic contact book support (with a sample XML contact book plugin provided)
All codecs (minus commercial g729 support) that freeswitch supports

Give it a shot and let us know about any issues you run into, overall it was decently tested for many months but some of the rapid changes as of late means there may be some larger bugs that crept in.  I will also be on the conference call tomorrow and can answer questions there.

And so to download the binaries:
 http://files.freeswitch.org/windows/installer/x86/FSClient.zip
There will be a readme in the install dir on usage.

The source code is available on the contrib git repo mitchcapper/FSClient.

As for building from source you will need to do a few things to trunk currently.  In the source folder you need to move mod_portaudio.c to the freeswitch\src\mod\endpoints\mod_portaudio folder (http://jira.freeswitch.org/browse/FS-3006 are the changes pending trunk).  In addition move the portaudio.2010.vxcproj over the version in freeswitch\libs\portaudio\build\msvc to enable direct X support.  Make sure you have the WPF Toolkit also installed.  Build trunk and then for FSClient set the ENV var FREESWITCH_SRC_LOCATION to the src location and it will also auto-copy the needed files into the build folder.


History:
Back in May I wanted a SIP client that supported the Jabra headsets, after not finding one that would work as I liked I looked to open source and everyone pointed me towards FSComm.  FSComm is great, and has a huge amount of time poured into it, but the C++/QT base for it was not for me.  .NET was the natural choice due to its rapid development time and stability so I went there.

After a few weeks work I had a client that I used for over 6 months and was fairly stable with relatively minor changes.   A few weeks ago on the conference call someone mentioned wanting a windows sip client option and I figured I would offer up my code base.  Well getting private code ready for public use was a bit more complex than planned but its here now and certainly in the past several weeks has had a lot of development.

~Mitch
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From oa at estation.dk  Wed Feb  2 11:34:52 2011
From: oa at estation.dk (=?ISO-8859-1?Q?=D8yvind_Albrigtsen?=)
Date: Wed, 02 Feb 2011 09:34:52 +0100
Subject: [Freeswitch-users] Choppy VM when using loopback (on
	Debian	lenny)
In-Reply-To: <AANLkTi=t8AiZG6axtfqmMe9ZHTPNowoAb=7eNjQ-RPoG@mail.gmail.com>
References: <4D42B8EE.7030208@estation.dk>	<7C6F3FC7-445F-4157-BD59-D76A705E4CE7@carmickle.com>	<9894FEC6-E451-4993-A85A-2CD910E916B1@ipeva.fr>	<AANLkTim+aBjRiAvxDUF1O1vut_CJpqgct2OqCNVxi0q4@mail.gmail.com>	<AANLkTiktckovGv4Ksnn_56EzpvTTSnfG+iD1vPX1_4RM@mail.gmail.com>	<4D466CAE.7030402@estation.dk>	<AANLkTimhOKmy_yupGtA2HBY_8wXtQKZnDYJ9Ye76bhzq@mail.gmail.com>	<AANLkTiktQU7kOG_22nG=_oEHSrBad667jcTzBs3n2nQv@mail.gmail.com>	<AANLkTiko=qTZbXCL09D-+W-FmibMnBjOeoR30YYzHe0i@mail.gmail.com>
	<AANLkTi=t8AiZG6axtfqmMe9ZHTPNowoAb=7eNjQ-RPoG@mail.gmail.com>
Message-ID: <4D49172C.5060801@estation.dk>

I've now tested the latest GIT, and it now works fine with the Debian 
lenny default kernel. I didnt get around to recompiling it for 1000 HZ 
though, but I tried changing clocksource which didnt help at all.

This is on a Dell server, and I'm not running it on a virtual server.

FYI I also tried latest GIT yesterday, with the same results as stated 
in my first mail.


Regards
Oyvind


On 2011-02-01 21:47, Anthony Minessale wrote:
> Try the latest GIT, I reverted the last patch and tried to solve the
> problem differently.
>
>
> On Tue, Feb 1, 2011 at 2:11 PM, Dan Lane<null at invalid.name>  wrote:
>    
>> Yes, using 1f1541b our calls that use mod_loopback into mod_conference
>> are unusable. The audio is choppy and delay increases as time passes.
>>
>> I thought it was an issue relating to the default 100Hz kernel on EC2
>> so I spent some time yesterday putting together a 1000Hz kernel but it
>> didn't make any difference. In the meantime I've compiled mod_loopback
>> from 4c5426f and loaded it with my 1f1541b build which eliminates the
>> issue.
>>
>> I haven't added it to Jira yet as I want to spend some time debugging
>> it (and I also owe you some debug info for FS-2934) but the problem is
>> definitely there.
>>
>> On Tue, Feb 1, 2011 at 3:58 PM, Anthony Minessale
>> <anthony.minessale at gmail.com>  wrote:
>>      
>>> Are you saying you have better results on that version than you do on
>>> the latest?
>>> What conditions do you have that cause you trouble, what is the
>>> endpoint on the other side.
>>>
>>> If the last commit to mod_loopback intended to improve audio quality
>>> actually makes it worse I need to investigate it.
>>>
>>>
>>> On Tue, Feb 1, 2011 at 5:53 AM, Dan Lane<null at invalid.name>  wrote:
>>>        
>>>> FWIW we've been also been having audio issues with loopback recently
>>>> on EC2 (with a 1000Hz kernel).
>>>>
>>>> We worked around it in the short term by reverting mod_loopback to
>>>> git-4c5426f during the build process.
>>>>
>>>> For anyone else who wants to try this just run "git checkout 4c5426f"
>>>> in src/mod/endpoints/mod_loopback then build as usual. This is NOT a
>>>> long term solution though.
>>>>
>>>> On Mon, Jan 31, 2011 at 8:02 AM, ?yvind Albrigtsen<oa at estation.dk>  wrote:
>>>>          
>>>>> Thanks for all your feedback. I'll keep on trying and inform you what
>>>>> worked for me.
>>>>>
>>>>>
>>>>> Regards
>>>>> Oyvind
>>>>>
>>>>> On 2011-01-29 21:48, Anthony Minessale wrote:
>>>>>            
>>>>>> Everyone should try latest GIT before pondering any further because I
>>>>>> added a patch like 2 days ago to adress this issue.
>>>>>>
>>>>>>
>>>>>> On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank at telonium.com>    wrote:
>>>>>>
>>>>>>              
>>>>>>> Yeah. I, too, don't see the correlation between the NTP and hw timer..
>>>>>>> I am not familiar with the timer_test command and what it's measuring, but
>>>>>>> of the 50 tests it ran, min is 19089 and max is 20713.
>>>>>>> Frank
>>>>>>>
>>>>>>>
>>>>>>> On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone at ipeva.fr>
>>>>>>> wrote:
>>>>>>>
>>>>>>>                
>>>>>>>> Frank,
>>>>>>>> I fail to see the relationship between the hw timer and NTP.
>>>>>>>> Can you please elaborate ?
>>>>>>>> David Ponzone  Direction Technique
>>>>>>>> email: david.ponzone at ipeva.fr
>>>>>>>> tel:      01 74 03 18 97
>>>>>>>> gsm:   06 66 98 76 34
>>>>>>>> Service Client IPeva
>>>>>>>> tel:      0811 46 26 26
>>>>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>>>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>>>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>>>>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>>>>>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Le 28/01/2011 ? 19:49, Frank Carmickle a ?crit :
>>>>>>>>
>>>>>>>> Hi
>>>>>>>>
>>>>>>>> On Jan 28, 2011, at 7:39 AM, ?yvind Albrigtsen wrote:
>>>>>>>>
>>>>>>>> Hi,
>>>>>>>>
>>>>>>>> I'm using latest git-version of Freeswitch, and when I go to voicemail
>>>>>>>>
>>>>>>>> when calling a number the sound playback is choppy and it skips some of
>>>>>>>>
>>>>>>>> the digits in the number I called.
>>>>>>>>
>>>>>>>> What kind of results do you get from timer_test at the fs_cli?  Are you
>>>>>>>> running on hardware or are you virtualized?  What is your clock source set
>>>>>>>> to and what are your available clock source options?  See
>>>>>>>> /sys/devices/system/clocksource/clocksource0/available_clocksource and
>>>>>>>> /sys/devices/system/clocksource/clocksource0/current_clocksource.  I am
>>>>>>>> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to
>>>>>>>> hang at 19998/19999 which works very well for me.  When I was having problem
>>>>>>>> it was reporting numbers all over the map from 17400 to 22600 with lots of
>>>>>>>> randomness in between.  I have my clocksource set to jiffies and xen
>>>>>>>> independent wallclock set to 1.  Of course at that point you need to have
>>>>>>>> ntp running against a bunch of servers to drive your clock nice and steady.
>>>>>>>>    I know my set up is probably a lot different than yours but I thought I'd
>>>>>>>> toss it out there to show that some of the harshest conditions can be dealt
>>>>>>>> with and don't give up trying.  If you are running on hardware with a cpu
>>>>>>>> that doesn't have constant_tsc then you might have some problems.  Just play
>>>>>>>> with the different timer options until you find the one that works.
>>>>>>>>
>>>>>>>> HTH
>>>>>>>> --FC
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at 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
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>                
>>>>>>
>>>>>>
>>>>>>              
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>            
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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 covici at ccs.covici.com  Wed Feb  2 12:24:18 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 02 Feb 2011 04:24:18 -0500
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
	for Windows in .NET
In-Reply-To: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
Message-ID: <16100.1296638658@ccs.covici.com>

Unfortunately, two problems happened when trying to install:  the
installer did not try to elevate itself and so I had to rerun it as
administrator -- and when I finally did install the thing, all I got was
a window with the title and maybe a graphic or two and nothing else --
no menus or buttons or anything.

Mitch Capper <mitch.capper at gmail.com> wrote:

> In prep for tomorrows conference I am happy to release FSClient.  I would
> like to thank DRK and especially jlenk for their continued support and help
> on this project.  jlenk created the installer, fixed code, and certainly
> helped bring it along quite a bit faster!
> FSClient is a windows SIP client:
> Most features of any standard sip client
> multiple calls at once
> transfer, holding, speakerphone, DND
> multiple SIP accounts
> advanced headset support (caller id, buttons, etc) for jabra and plantronics
> out of the box (with plugin support for easily adding others)
> basic contact book support (with a sample XML contact book plugin provided)
> All codecs (minus commercial g729 support) that freeswitch supports
> 
> Give it a shot and let us know about any issues you run into, overall it was
> decently tested for many months but some of the rapid changes as of late
> means there may be some larger bugs that crept in.  I will also be on the
> conference call tomorrow and can answer questions there.
> 
> And so to download the binaries:
>  http://files.freeswitch.org/windows/installer/x86/FSClient.zip
> There will be a readme in the install dir on usage.
> 
> The source code is available on the contrib git repo mitchcapper/FSClient.
> 
> As for building from source you will need to do a few things to trunk
> currently.  In the source folder you need to move mod_portaudio.c to the
> freeswitch\src\mod\endpoints\mod_portaudio folder (
> http://jira.freeswitch.org/browse/FS-3006 are the changes pending trunk).
> In addition move the portaudio.2010.vxcproj over the version in
> freeswitch\libs\portaudio\build\msvc to enable direct X support.  Make sure
> you have the WPF Toolkit also installed.  Build trunk and then for FSClient
> set the ENV var FREESWITCH_SRC_LOCATION to the src location and it will also
> auto-copy the needed files into the build folder.
> 
> 
> History:
> Back in May I wanted a SIP client that supported the Jabra headsets, after
> not finding one that would work as I liked I looked to open source and
> everyone pointed me towards FSComm.  FSComm is great, and has a huge amount
> of time poured into it, but the C++/QT base for it was not for me.  .NET was
> the natural choice due to its rapid development time and stability so I went
> there.
> 
> After a few weeks work I had a client that I used for over 6 months and was
> fairly stable with relatively minor changes.   A few weeks ago on the
> conference call someone mentioned wanting a windows sip client option and I
> figured I would offer up my code base.  Well getting private code ready for
> public use was a bit more complex than planned but its here now and
> certainly in the past several weeks has had a lot of development.
> 
> ~Mitch
> 
> ----------------------------------------------------
> 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  Wed Feb  2 12:52:33 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 2 Feb 2011 09:52:33 +0000
Subject: [Freeswitch-users] open g729
In-Reply-To: <AANLkTinVX-_UYKhCaBi_1Lh037SQgKZCaS2uM1UgDM+C@mail.gmail.com>
References: <AANLkTikE3viP6KN+28cRzx=39cLQJ+OJZU64xi27UBB9@mail.gmail.com>
	<C96DB334.46A6%krice@freeswitch.org>
	<AANLkTinVX-_UYKhCaBi_1Lh037SQgKZCaS2uM1UgDM+C@mail.gmail.com>
Message-ID: <AANLkTikr2aCnW_CvHFv-Dv_wytyuC+ipUU5mHmrv=0uJ@mail.gmail.com>

But you still need someone to be able to talk to. Ken's point is it works
great when you're only calling your other Speex phones, but as soon as you
want to talk to someone else you'll find they don't have Speex. You then
have to use another codec... a common standard one that's been around for so
long everyone supports it... like G729.

On 2 February 2011 03:00, curriegrad2004 <curriegrad2004 at gmail.com> wrote:

> If hardware doesn't support it, make one that supports Speex right out
> of the box. I'm pretty sure that somebody will do it.
>
> On Tue, Feb 1, 2011 at 10:55 AM, Ken Rice <krice at freeswitch.org> wrote:
> > Why people insist on speex I am not sure... Sure its nice on a closed
> > network but quickly looses its luster when communicating with the real
> > world... Not to mention the CPU usages and such...
> >
> > If you want to use something like speex check out iLBC and theres things
> > like the Sangoma D100/D500 that you can use to offload the transcoding
> from
> > the CPU (disclaimer I sell the sangoma hardware)
> >
> > K
> >
> >
> > On 2/1/11 12:45 PM, "Kristian Kielhofner" <kris at kriskinc.com> wrote:
> >
> >> Phones, carriers, and just about any commercial gear (reference
> >> hardware, SBCs, DSPs, etc) you'll ever see doesn't support Speex.
> >> Guess how you'll make it work?  Transcoding from G729 to Speex.  Now
> >> you're using licenses, proxying RTP, and wasting CPU while ruining
> >> voice quality transcoding from one lossy codec to another.  It's just
> >> about the worst possible situation you could be in.
> >>
> >> On Tue, Feb 1, 2011 at 1:19 PM, Avi Marcus <avi at avimarcus.net> wrote:
> >>> Except that many hardware phones don't have speex, so the only
> low-bandwidth
> >>> solution is g729...
> >>> On Tue, Feb 1, 2011 at 7:43 PM, curriegrad2004 <
> curriegrad2004 at gmail.com>
> >>> wrote:
> >>>>
> >>>> Simple solution to this g.729 patent non-sense: Speex.
> >>>>
> >>>> 'nuff said.
> >>>>
> >>>> On Tue, Feb 1, 2011 at 8:26 AM, Anton VG <anton.vazir at gmail.com>
> wrote:
> >>>>> Considering the above, noone can say "Purchase a LICENSE", but only a
> >>>>> "Licensed SOFTWARE"
> >>>>>
> >>>>> _______________________________________________
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>>
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >>
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  2 12:55:28 2011
From: u2nsam at gmail.com (Sam)
Date: Wed, 2 Feb 2011 15:25:28 +0530
Subject: [Freeswitch-users] context
Message-ID: <AANLkTika5CBO5eVWuRimKY+Bth2YAFC0Tm8_JmFBnXwC@mail.gmail.com>

I have defined a user as :

<user id="2099">
      <params>
        <param name="password" value="1234"/>
        <param name="dial-string" value="{presence_id=${dialed_user}@
${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
        <param name="mwi-account" value="2099 at voicemail"/>
        <param name="vm-password" value="1234"/>
        <param name="vm-email-all-messages" value="true"/>
        <!--param name="vm-attach-file" value="true"/-->
        <!--param name="vm-mailto"  value=""/-->
      </params>
      <variables>
        <variable name="user_context" value="inter"/>
        <variable name="force_transfer_context" value="inter"/>
        <variable name="accountcode" value="2099"/>
        <variable name="internal_caller_id_name" value="2099"/>
        <variable name="internal_caller_id_number" value="2099"/>
        <variable name="callgroup" value="1"/>
        <variable name="sip-force-contact"
value="NDLB-connectile-dysfunction"/>
      </variables>


and in the dialplan i have

<context name="inter">

<extension name="2099">
    <condition field="destination_number" expression="^(2099)">
      <action application="set" data="call_timeout=30"/>
      <action application="set" data="dialed_extension=$1"/>
            <action application="export" data="dialed_extension=$1"/>
            <action application="set"
data="transfer_ringback=$${hold_music}"/>
            <action application="set" data="hangup_after_bridge=true"/>
            <action application="set" data="continue_on_fail=true"/>
            <action application="hash"
data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
            <action application="hash"
data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
            <action application="hash"
data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
            <action application="hash"
data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
            <action application="set"
data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)}"/>
            <action application="hash"
data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
            <action application="bridge"
data="{sip_invite_domain=$${domain}}user/${dialed_extension}@
${domain_name}"/>
    </condition>
  </extension>

</context>


Now extension 2075 is also made on above lines in the same context for user
and dialplan, here the call when initiated from 2075 to 2099 searches for
context public instead of context inter

mod_dialplan_xml.c:331 Processing 2075 <2075>->2099 in context public

any reason why ?

Regards
Sam
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From devel at thom.fr.eu.org  Wed Feb  2 13:21:10 2011
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Wed, 02 Feb 2011 11:21:10 +0100
Subject: [Freeswitch-users] Route incoming analog calls.
In-Reply-To: <AANLkTimYZiH6kc49TZZvN6UTofEygu_3XgZaDMQ7YsYr@mail.gmail.com>
References: <AANLkTikmXVZqVAJZ-Av21ohRUnGbmz0Fs9z=kZCK+kp3@mail.gmail.com>
	<AANLkTimYZiH6kc49TZZvN6UTofEygu_3XgZaDMQ7YsYr@mail.gmail.com>
Message-ID: <85bf91bfa6c3dcf91dfbe3aa6c122541@thom.fr.eu.org>

 Check this http://wiki.freeswitch.org/wiki/FreeTDM#Dial_Plan

 On Tue, 1 Feb 2011 09:50:16 -0800, curriegrad2004 
 <curriegrad2004 at gmail.com> wrote:
> You'll have to read the documentation on the Sangoma card itself. I'm
> pretty sure there is a variable where the card's driver sets and
> passes it on to Freeswitch's switching component.
>
> On Tue, Feb 1, 2011 at 8:17 AM, Eric Michel <eric at bmcrministries.org> 
> wrote:
>> I've got a Sangoma A200 with six FXO ports and two FXS. ?I can call 
>> out just
>> fine, but I'm having trouble figuring out how to route incoming 
>> calls.
>> ?We've got five POTS lines and all I've been able to do is route all
>> incoming calls to a specific?extension. ?How do I go about routing
>> individual POTS lines to different groups/extensions?
>> <extension name="analog test2" continue="true">
>> ?? ?<condition field="source" expression="mod_freetdm">
>> ?? ? <action application="set" data="domain_name=$${domain}"/>
>> ?? ? ? <action application="set" 
>> data="transfer_ringback=$${us-ring}"/>
>> ?? ? ? <action application="answer"/>
>> ?? ? ? <action application="transfer" data="1000 XML default"/>
>> ?? ?</condition>
>> ?</extension>
>> That is my current conf/public/inbound dial plan. ?Shouldn't I be 
>> able to
>> add a second condition that tests what FXO port is receiving the 
>> call and
>> allow me to route it accordingly? ?If yes, could I get an example? 
>> ?The
>> answer may be really simple, I've just been unable to find it.
>> 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
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb  2 13:58:53 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 2 Feb 2011 10:58:53 +0000
Subject: [Freeswitch-users] Route incoming analog calls.
In-Reply-To: <AANLkTikmXVZqVAJZ-Av21ohRUnGbmz0Fs9z=kZCK+kp3@mail.gmail.com>
References: <AANLkTikmXVZqVAJZ-Av21ohRUnGbmz0Fs9z=kZCK+kp3@mail.gmail.com>
Message-ID: <AANLkTimAvYwXmjV9zmWuU2juVpbzYvBn8w9SPLyE78=-@mail.gmail.com>

Try this:

<extension name="analog test2" continue="true">
    <condition field="source" expression="mod_freetdm">
        <action application="info"/>
    </condition>
 </extension>

That'll dump all the fields and variables set. Are there any that indicate
the FXO port? If not, perhaps you need to configure something - check the
card's documentation.

If you find there is a variable containing the port, you can check more than
one condition. For example:
<extension name="analog test2" continue="true">
    <condition field="source" expression="mod_freetdm"/>
    <condition field="${sangoma_fxo_port_variable_name}" expression="^1$">
     <action application="set" data="domain_name=$${domain}"/>
       <action application="set" data="transfer_ringback=$${us-ring}"/>
       <action application="answer"/>
       <action application="transfer" data="1000 XML default"/>
    </condition>
 </extension>

-Steve


On 1 February 2011 16:17, Eric Michel <eric at bmcrministries.org> wrote:

> I've got a Sangoma A200 with six FXO ports and two FXS.  I can call out
> just fine, but I'm having trouble figuring out how to route incoming calls.
>  We've got five POTS lines and all I've been able to do is route all
> incoming calls to a specific extension.  How do I go about routing
> individual POTS lines to different groups/extensions?
>
> <extension name="analog test2" continue="true">
>     <condition field="source" expression="mod_freetdm">
>      <action application="set" data="domain_name=$${domain}"/>
>        <action application="set" data="transfer_ringback=$${us-ring}"/>
>        <action application="answer"/>
>        <action application="transfer" data="1000 XML default"/>
>     </condition>
>  </extension>
>
> That is my current conf/public/inbound dial plan.  Shouldn't I be able to
> add a second condition that tests what FXO port is receiving the call and
> allow me to route it accordingly?  If yes, could I get an example?  The
> answer may be really simple, I've just been unable to find it.
>
> 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 steveayre at gmail.com  Wed Feb  2 14:04:13 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 2 Feb 2011 11:04:13 +0000
Subject: [Freeswitch-users] Route incoming analog calls.
In-Reply-To: <AANLkTimAvYwXmjV9zmWuU2juVpbzYvBn8w9SPLyE78=-@mail.gmail.com>
References: <AANLkTikmXVZqVAJZ-Av21ohRUnGbmz0Fs9z=kZCK+kp3@mail.gmail.com>
	<AANLkTimAvYwXmjV9zmWuU2juVpbzYvBn8w9SPLyE78=-@mail.gmail.com>
Message-ID: <AANLkTinfvSguG+4SD_UKy1qE+iokLQPeStcGXi+9x575@mail.gmail.com>

You can also handle this slightly better by setting the context parameter
for the FXO port in freetdm.conf.xml, then you can have a context that just
handles incoming fxo calls from mod_freetdm, so you no longer have to check
the source.

http://wiki.freeswitch.org/wiki/FreeTDM#FreeSWITCH_FreeTDM_configuration

It looks like the following variables are set:

   - freetdm_span_name
   - freetdm_span_number
   - freetdm_chan_number

Do any of those indicate the FXO/FXS?

-Steve


On 2 February 2011 10:58, Steven Ayre <steveayre at gmail.com> wrote:

> Try this:
>
> <extension name="analog test2" continue="true">
>     <condition field="source" expression="mod_freetdm">
>         <action application="info"/>
>     </condition>
>  </extension>
>
> That'll dump all the fields and variables set. Are there any that indicate
> the FXO port? If not, perhaps you need to configure something - check the
> card's documentation.
>
> If you find there is a variable containing the port, you can check more
> than one condition. For example:
>  <extension name="analog test2" continue="true">
>     <condition field="source" expression="mod_freetdm"/>
>     <condition field="${sangoma_fxo_port_variable_name}" expression="^1$">
>       <action application="set" data="domain_name=$${domain}"/>
>        <action application="set" data="transfer_ringback=$${us-ring}"/>
>        <action application="answer"/>
>        <action application="transfer" data="1000 XML default"/>
>     </condition>
>  </extension>
>

> -Steve
>
>
> On 1 February 2011 16:17, Eric Michel <eric at bmcrministries.org> wrote:
>
>> I've got a Sangoma A200 with six FXO ports and two FXS.  I can call out
>> just fine, but I'm having trouble figuring out how to route incoming calls.
>>  We've got five POTS lines and all I've been able to do is route all
>> incoming calls to a specific extension.  How do I go about routing
>> individual POTS lines to different groups/extensions?
>>
>> <extension name="analog test2" continue="true">
>>     <condition field="source" expression="mod_freetdm">
>>      <action application="set" data="domain_name=$${domain}"/>
>>        <action application="set" data="transfer_ringback=$${us-ring}"/>
>>        <action application="answer"/>
>>        <action application="transfer" data="1000 XML default"/>
>>     </condition>
>>  </extension>
>>
>> That is my current conf/public/inbound dial plan.  Shouldn't I be able to
>> add a second condition that tests what FXO port is receiving the call and
>> allow me to route it accordingly?  If yes, could I get an example?  The
>> answer may be really simple, I've just been unable to find it.
>>
>> 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 kbdfck at gmail.com  Wed Feb  2 14:43:39 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Wed, 2 Feb 2011 14:43:39 +0300
Subject: [Freeswitch-users] How to prevent source channel hangup after
 uuid_bridge completion (outbound ESL)?
In-Reply-To: <AANLkTimhBqbp5upa-me21AsWNYACr14FZQkStRK=sYXW@mail.gmail.com>
References: <AANLkTim7PoaVVCopbk6-wT1E04=pabKw9FjZaHCbLH5=@mail.gmail.com>
	<AANLkTik-rV6wwBjm3DsEDGVszEizKhgbQfYzN_TKvBhA@mail.gmail.com>
	<AANLkTimhBqbp5upa-me21AsWNYACr14FZQkStRK=sYXW@mail.gmail.com>
Message-ID: <AANLkTi=yhk7E30GTQ7ykeJotcXX5KOTWr6aSOj8_wCyY@mail.gmail.com>

hangup_after_bridge=false doesn't prevent source channel to be hangup.
socket application we use to create outbound ESL connection is
finished on originated channel hangup, and since there is no more apps
to run in dialplan, channel hangs up.

But I need to stay in ESL script after channel I originated hangs up.
How to do this?


2011/2/1 Dmitry Sytchev <kbdfck at gmail.com>:
> Thanks, will try this. For some reason I forgot about this, although
> using it in dialplan :)))
>
> 2011/2/1 Christopher Rienzo <cmrienzo at gmail.com>:
>> Set the hangup_after_bridge channel variable to false.
>>
>>
>> On Tue, Feb 1, 2011 at 9:34 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>>
>>> Hi
>>>
>>> I'm trying to originate call from outbound ESL and then bridge it to
>>> original channel where ESL is launched on.
>>> I use api originate with &park(), then uuid_bridge. Everything works
>>> as expected, but when second leg hangs up, first leg gets hangup too.
>>> How can I prevent first leg from hanging up and continue to process it
>>> in ESL?
>>>
>>> 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
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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


From covici at ccs.covici.com  Wed Feb  2 15:08:15 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 02 Feb 2011 07:08:15 -0500
Subject: [Freeswitch-users] Route incoming analog calls.
In-Reply-To: <AANLkTimAvYwXmjV9zmWuU2juVpbzYvBn8w9SPLyE78=-@mail.gmail.com>
References: <AANLkTikmXVZqVAJZ-Av21ohRUnGbmz0Fs9z=kZCK+kp3@mail.gmail.com>
	<AANLkTimAvYwXmjV9zmWuU2juVpbzYvBn8w9SPLyE78=-@mail.gmail.com>
Message-ID: <17853.1296648495@ccs.covici.com>

I used channel_name like this <condition field="${Channel_Name}"
expression="^FreeTDM/1:1/$">  and that has been working -- freetdm is
not working well, but that is another story.

Steven Ayre <steveayre at gmail.com> wrote:

> Try this:
> 
> <extension name="analog test2" continue="true">
>     <condition field="source" expression="mod_freetdm">
>         <action application="info"/>
>     </condition>
>  </extension>
> 
> That'll dump all the fields and variables set. Are there any that indicate
> the FXO port? If not, perhaps you need to configure something - check the
> card's documentation.
> 
> If you find there is a variable containing the port, you can check more than
> one condition. For example:
> <extension name="analog test2" continue="true">
>     <condition field="source" expression="mod_freetdm"/>
>     <condition field="${sangoma_fxo_port_variable_name}" expression="^1$">
>      <action application="set" data="domain_name=$${domain}"/>
>        <action application="set" data="transfer_ringback=$${us-ring}"/>
>        <action application="answer"/>
>        <action application="transfer" data="1000 XML default"/>
>     </condition>
>  </extension>
> 
> -Steve
> 
> 
> On 1 February 2011 16:17, Eric Michel <eric at bmcrministries.org> wrote:
> 
> > I've got a Sangoma A200 with six FXO ports and two FXS.  I can call out
> > just fine, but I'm having trouble figuring out how to route incoming calls.
> >  We've got five POTS lines and all I've been able to do is route all
> > incoming calls to a specific extension.  How do I go about routing
> > individual POTS lines to different groups/extensions?
> >
> > <extension name="analog test2" continue="true">
> >     <condition field="source" expression="mod_freetdm">
> >      <action application="set" data="domain_name=$${domain}"/>
> >        <action application="set" data="transfer_ringback=$${us-ring}"/>
> >        <action application="answer"/>
> >        <action application="transfer" data="1000 XML default"/>
> >     </condition>
> >  </extension>
> >
> > That is my current conf/public/inbound dial plan.  Shouldn't I be able to
> > add a second condition that tests what FXO port is receiving the call and
> > allow me to route it accordingly?  If yes, could I get an example?  The
> > answer may be really simple, I've just been unable to find it.
> >
> > 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
> >
> >
> 
> ----------------------------------------------------
> 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  Wed Feb  2 15:11:01 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 2 Feb 2011 12:11:01 +0000
Subject: [Freeswitch-users] Route incoming analog calls.
In-Reply-To: <17853.1296648495@ccs.covici.com>
References: <AANLkTikmXVZqVAJZ-Av21ohRUnGbmz0Fs9z=kZCK+kp3@mail.gmail.com>
	<AANLkTimAvYwXmjV9zmWuU2juVpbzYvBn8w9SPLyE78=-@mail.gmail.com>
	<17853.1296648495@ccs.covici.com>
Message-ID: <AANLkTi=O_fEFD+vAQ5oHyyxQB9FiBPKTnWSEOLpBuNA5@mail.gmail.com>

Great. :)

I think that 1:1 is equivalent to ${freetdm_span_number}:${
freetdm_chan_number}

-Steve


On 2 February 2011 12:08, <covici at ccs.covici.com> wrote:

> I used channel_name like this <condition field="${Channel_Name}"
> expression="^FreeTDM/1:1/$">  and that has been working -- freetdm is
> not working well, but that is another story.
>
> Steven Ayre <steveayre at gmail.com> wrote:
>
> > Try this:
> >
> > <extension name="analog test2" continue="true">
> >     <condition field="source" expression="mod_freetdm">
> >         <action application="info"/>
> >     </condition>
> >  </extension>
> >
> > That'll dump all the fields and variables set. Are there any that
> indicate
> > the FXO port? If not, perhaps you need to configure something - check the
> > card's documentation.
> >
> > If you find there is a variable containing the port, you can check more
> than
> > one condition. For example:
> > <extension name="analog test2" continue="true">
> >     <condition field="source" expression="mod_freetdm"/>
> >     <condition field="${sangoma_fxo_port_variable_name}"
> expression="^1$">
> >      <action application="set" data="domain_name=$${domain}"/>
> >        <action application="set" data="transfer_ringback=$${us-ring}"/>
> >        <action application="answer"/>
> >        <action application="transfer" data="1000 XML default"/>
> >     </condition>
> >  </extension>
> >
> > -Steve
> >
> >
> > On 1 February 2011 16:17, Eric Michel <eric at bmcrministries.org> wrote:
> >
> > > I've got a Sangoma A200 with six FXO ports and two FXS.  I can call out
> > > just fine, but I'm having trouble figuring out how to route incoming
> calls.
> > >  We've got five POTS lines and all I've been able to do is route all
> > > incoming calls to a specific extension.  How do I go about routing
> > > individual POTS lines to different groups/extensions?
> > >
> > > <extension name="analog test2" continue="true">
> > >     <condition field="source" expression="mod_freetdm">
> > >      <action application="set" data="domain_name=$${domain}"/>
> > >        <action application="set" data="transfer_ringback=$${us-ring}"/>
> > >        <action application="answer"/>
> > >        <action application="transfer" data="1000 XML default"/>
> > >     </condition>
> > >  </extension>
> > >
> > > That is my current conf/public/inbound dial plan.  Shouldn't I be able
> to
> > > add a second condition that tests what FXO port is receiving the call
> and
> > > allow me to route it accordingly?  If yes, could I get an example?  The
> > > answer may be really simple, I've just been unable to find it.
> > >
> > > 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
> > >
> > >
> >
> > ----------------------------------------------------
> > 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
>
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From covici at ccs.covici.com  Wed Feb  2 15:16:59 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 02 Feb 2011 07:16:59 -0500
Subject: [Freeswitch-users] Route incoming analog calls.
In-Reply-To: <AANLkTi=O_fEFD+vAQ5oHyyxQB9FiBPKTnWSEOLpBuNA5@mail.gmail.com>
References: <AANLkTikmXVZqVAJZ-Av21ohRUnGbmz0Fs9z=kZCK+kp3@mail.gmail.com>
	<AANLkTimAvYwXmjV9zmWuU2juVpbzYvBn8w9SPLyE78=-@mail.gmail.com>
	<17853.1296648495@ccs.covici.com>
	<AANLkTi=O_fEFD+vAQ5oHyyxQB9FiBPKTnWSEOLpBuNA5@mail.gmail.com>
Message-ID: <18053.1296649019@ccs.covici.com>

Very possibly - at the time I had never heard of that variable.

Steven Ayre <steveayre at gmail.com> wrote:

> Great. :)
> 
> I think that 1:1 is equivalent to ${freetdm_span_number}:${
> freetdm_chan_number}
> 
> -Steve
> 
> 
> On 2 February 2011 12:08, <covici at ccs.covici.com> wrote:
> 
> > I used channel_name like this <condition field="${Channel_Name}"
> > expression="^FreeTDM/1:1/$">  and that has been working -- freetdm is
> > not working well, but that is another story.
> >
> > Steven Ayre <steveayre at gmail.com> wrote:
> >
> > > Try this:
> > >
> > > <extension name="analog test2" continue="true">
> > >     <condition field="source" expression="mod_freetdm">
> > >         <action application="info"/>
> > >     </condition>
> > >  </extension>
> > >
> > > That'll dump all the fields and variables set. Are there any that
> > indicate
> > > the FXO port? If not, perhaps you need to configure something - check the
> > > card's documentation.
> > >
> > > If you find there is a variable containing the port, you can check more
> > than
> > > one condition. For example:
> > > <extension name="analog test2" continue="true">
> > >     <condition field="source" expression="mod_freetdm"/>
> > >     <condition field="${sangoma_fxo_port_variable_name}"
> > expression="^1$">
> > >      <action application="set" data="domain_name=$${domain}"/>
> > >        <action application="set" data="transfer_ringback=$${us-ring}"/>
> > >        <action application="answer"/>
> > >        <action application="transfer" data="1000 XML default"/>
> > >     </condition>
> > >  </extension>
> > >
> > > -Steve
> > >
> > >
> > > On 1 February 2011 16:17, Eric Michel <eric at bmcrministries.org> wrote:
> > >
> > > > I've got a Sangoma A200 with six FXO ports and two FXS.  I can call out
> > > > just fine, but I'm having trouble figuring out how to route incoming
> > calls.
> > > >  We've got five POTS lines and all I've been able to do is route all
> > > > incoming calls to a specific extension.  How do I go about routing
> > > > individual POTS lines to different groups/extensions?
> > > >
> > > > <extension name="analog test2" continue="true">
> > > >     <condition field="source" expression="mod_freetdm">
> > > >      <action application="set" data="domain_name=$${domain}"/>
> > > >        <action application="set" data="transfer_ringback=$${us-ring}"/>
> > > >        <action application="answer"/>
> > > >        <action application="transfer" data="1000 XML default"/>
> > > >     </condition>
> > > >  </extension>
> > > >
> > > > That is my current conf/public/inbound dial plan.  Shouldn't I be able
> > to
> > > > add a second condition that tests what FXO port is receiving the call
> > and
> > > > allow me to route it accordingly?  If yes, could I get an example?  The
> > > > answer may be really simple, I've just been unable to find it.
> > > >
> > > > 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
> > > >
> > > >
> > >
> > > ----------------------------------------------------
> > > 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
> >
> 
> ----------------------------------------------------
> 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 jmesquita at freeswitch.org  Wed Feb  2 16:48:39 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 2 Feb 2011 08:48:39 -0500
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
 for Windows in .NET
In-Reply-To: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
Message-ID: <AANLkTimghBGjRNXA4GAVXvGFU0KQX5cEqQMREO5U=DvW@mail.gmail.com>

Mitch, you beat me to it...

I think it might be time for me to get back to FSComm and drop being picky
about the AEC thing? Sad but true, I think that the only way for me to draw
attention of someone who knows about this stuff is actually doing the rest
like you did. Great initiative. I truly congratulate you from pulling
something I couldn't so far.

Regards,
Jo?o Mesquita



On Wed, Feb 2, 2011 at 1:39 AM, Mitch Capper <mitch.capper at gmail.com> wrote:

> In prep for tomorrows conference I am happy to release FSClient.  I would
> like to thank DRK and especially jlenk for their continued support and help
> on this project.  jlenk created the installer, fixed code, and certainly
> helped bring it along quite a bit faster!
> FSClient is a windows SIP client:
> Most features of any standard sip client
> multiple calls at once
> transfer, holding, speakerphone, DND
> multiple SIP accounts
> advanced headset support (caller id, buttons, etc) for jabra and
> plantronics out of the box (with plugin support for easily adding others)
> basic contact book support (with a sample XML contact book plugin provided)
> All codecs (minus commercial g729 support) that freeswitch supports
>
> Give it a shot and let us know about any issues you run into, overall it
> was decently tested for many months but some of the rapid changes as of late
> means there may be some larger bugs that crept in.  I will also be on the
> conference call tomorrow and can answer questions there.
>
> And so to download the binaries:
>  http://files.freeswitch.org/windows/installer/x86/FSClient.zip
> There will be a readme in the install dir on usage.
>
> The source code is available on the contrib git repo mitchcapper/FSClient.
>
>
> As for building from source you will need to do a few things to trunk
> currently.  In the source folder you need to move mod_portaudio.c to the
> freeswitch\src\mod\endpoints\mod_portaudio folder (
> http://jira.freeswitch.org/browse/FS-3006 are the changes pending trunk).
> In addition move the portaudio.2010.vxcproj over the version in
> freeswitch\libs\portaudio\build\msvc to enable direct X support.  Make sure
> you have the WPF Toolkit also installed.  Build trunk and then for FSClient
> set the ENV var FREESWITCH_SRC_LOCATION to the src location and it will also
> auto-copy the needed files into the build folder.
>
>
> History:
> Back in May I wanted a SIP client that supported the Jabra headsets, after
> not finding one that would work as I liked I looked to open source and
> everyone pointed me towards FSComm.  FSComm is great, and has a huge amount
> of time poured into it, but the C++/QT base for it was not for me.  .NET was
> the natural choice due to its rapid development time and stability so I went
> there.
>
> After a few weeks work I had a client that I used for over 6 months and was
> fairly stable with relatively minor changes.   A few weeks ago on the
> conference call someone mentioned wanting a windows sip client option and I
> figured I would offer up my code base.  Well getting private code ready for
> public use was a bit more complex than planned but its here now and
> certainly in the past several weeks has had a lot of development.
>
> ~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 null at invalid.name  Wed Feb  2 17:44:09 2011
From: null at invalid.name (Dan Lane)
Date: Wed, 2 Feb 2011 14:44:09 +0000
Subject: [Freeswitch-users] Choppy VM when using loopback (on Debian
	lenny)
In-Reply-To: <AANLkTi=t8AiZG6axtfqmMe9ZHTPNowoAb=7eNjQ-RPoG@mail.gmail.com>
References: <4D42B8EE.7030208@estation.dk>
	<7C6F3FC7-445F-4157-BD59-D76A705E4CE7@carmickle.com>
	<9894FEC6-E451-4993-A85A-2CD910E916B1@ipeva.fr>
	<AANLkTim+aBjRiAvxDUF1O1vut_CJpqgct2OqCNVxi0q4@mail.gmail.com>
	<AANLkTiktckovGv4Ksnn_56EzpvTTSnfG+iD1vPX1_4RM@mail.gmail.com>
	<4D466CAE.7030402@estation.dk>
	<AANLkTimhOKmy_yupGtA2HBY_8wXtQKZnDYJ9Ye76bhzq@mail.gmail.com>
	<AANLkTiktQU7kOG_22nG=_oEHSrBad667jcTzBs3n2nQv@mail.gmail.com>
	<AANLkTiko=qTZbXCL09D-+W-FmibMnBjOeoR30YYzHe0i@mail.gmail.com>
	<AANLkTi=t8AiZG6axtfqmMe9ZHTPNowoAb=7eNjQ-RPoG@mail.gmail.com>
Message-ID: <AANLkTiniBAWw+JMe=64PtAER9B9U7rbbkJ9X27fOfYBn@mail.gmail.com>

This seems to have resolved the issue for us :)

Thanks.

On Tue, Feb 1, 2011 at 8:47 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Try the latest GIT, I reverted the last patch and tried to solve the
> problem differently.
>
>
> On Tue, Feb 1, 2011 at 2:11 PM, Dan Lane <null at invalid.name> wrote:
>> Yes, using 1f1541b our calls that use mod_loopback into mod_conference
>> are unusable. The audio is choppy and delay increases as time passes.
>>
>> I thought it was an issue relating to the default 100Hz kernel on EC2
>> so I spent some time yesterday putting together a 1000Hz kernel but it
>> didn't make any difference. In the meantime I've compiled mod_loopback
>> from 4c5426f and loaded it with my 1f1541b build which eliminates the
>> issue.
>>
>> I haven't added it to Jira yet as I want to spend some time debugging
>> it (and I also owe you some debug info for FS-2934) but the problem is
>> definitely there.
>>
>> On Tue, Feb 1, 2011 at 3:58 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>> Are you saying you have better results on that version than you do on
>>> the latest?
>>> What conditions do you have that cause you trouble, what is the
>>> endpoint on the other side.
>>>
>>> If the last commit to mod_loopback intended to improve audio quality
>>> actually makes it worse I need to investigate it.
>>>
>>>
>>> On Tue, Feb 1, 2011 at 5:53 AM, Dan Lane <null at invalid.name> wrote:
>>>> FWIW we've been also been having audio issues with loopback recently
>>>> on EC2 (with a 1000Hz kernel).
>>>>
>>>> We worked around it in the short term by reverting mod_loopback to
>>>> git-4c5426f during the build process.
>>>>
>>>> For anyone else who wants to try this just run "git checkout 4c5426f"
>>>> in src/mod/endpoints/mod_loopback then build as usual. This is NOT a
>>>> long term solution though.
>>>>
>>>> On Mon, Jan 31, 2011 at 8:02 AM, ?yvind Albrigtsen <oa at estation.dk> wrote:
>>>>> Thanks for all your feedback. I'll keep on trying and inform you what
>>>>> worked for me.
>>>>>
>>>>>
>>>>> Regards
>>>>> Oyvind
>>>>>
>>>>> On 2011-01-29 21:48, Anthony Minessale wrote:
>>>>>> Everyone should try latest GIT before pondering any further because I
>>>>>> added a patch like 2 days ago to adress this issue.
>>>>>>
>>>>>>
>>>>>> On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank at telonium.com> ?wrote:
>>>>>>
>>>>>>> Yeah. I, too, don't see the correlation between the NTP and hw timer..
>>>>>>> I am not familiar with the timer_test command and what it's measuring, but
>>>>>>> of the 50 tests it ran, min is 19089 and max is 20713.
>>>>>>> Frank
>>>>>>>
>>>>>>>
>>>>>>> On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone at ipeva.fr>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> Frank,
>>>>>>>> I fail to see the relationship between the hw timer and NTP.
>>>>>>>> Can you please elaborate ?
>>>>>>>> David Ponzone ?Direction Technique
>>>>>>>> email: david.ponzone at ipeva.fr
>>>>>>>> tel: ? ? ?01 74 03 18 97
>>>>>>>> gsm: ? 06 66 98 76 34
>>>>>>>> Service Client IPeva
>>>>>>>> tel: ? ? ?0811 46 26 26
>>>>>>>> www.ipeva.fr ?- ? www.ipeva-studio.com
>>>>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>>>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>>>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>>>>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>>>>>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Le 28/01/2011 ? 19:49, Frank Carmickle a ?crit :
>>>>>>>>
>>>>>>>> Hi
>>>>>>>>
>>>>>>>> On Jan 28, 2011, at 7:39 AM, ?yvind Albrigtsen wrote:
>>>>>>>>
>>>>>>>> Hi,
>>>>>>>>
>>>>>>>> I'm using latest git-version of Freeswitch, and when I go to voicemail
>>>>>>>>
>>>>>>>> when calling a number the sound playback is choppy and it skips some of
>>>>>>>>
>>>>>>>> the digits in the number I called.
>>>>>>>>
>>>>>>>> What kind of results do you get from timer_test at the fs_cli? ?Are you
>>>>>>>> running on hardware or are you virtualized? ?What is your clock source set
>>>>>>>> to and what are your available clock source options? ?See
>>>>>>>> /sys/devices/system/clocksource/clocksource0/available_clocksource and
>>>>>>>> /sys/devices/system/clocksource/clocksource0/current_clocksource. ?I am
>>>>>>>> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to
>>>>>>>> hang at 19998/19999 which works very well for me. ?When I was having problem
>>>>>>>> it was reporting numbers all over the map from 17400 to 22600 with lots of
>>>>>>>> randomness in between. ?I have my clocksource set to jiffies and xen
>>>>>>>> independent wallclock set to 1. ?Of course at that point you need to have
>>>>>>>> ntp running against a bunch of servers to drive your clock nice and steady.
>>>>>>>> ? I know my set up is probably a lot different than yours but I thought I'd
>>>>>>>> toss it out there to show that some of the harshest conditions can be dealt
>>>>>>>> with and don't give up trying. ?If you are running on hardware with a cpu
>>>>>>>> that doesn't have constant_tsc then you might have some problems. ?Just play
>>>>>>>> with the different timer options until you find the one that works.
>>>>>>>>
>>>>>>>> HTH
>>>>>>>> --FC
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at 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
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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 jeff at jefflenk.com  Wed Feb  2 17:58:21 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 2 Feb 2011 06:58:21 -0800 (PST)
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
	for	Windows in .NET
In-Reply-To: <4D491068.7090308@solomo.de>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
	<4D491068.7090308@solomo.de>
Message-ID: <1296658701797-5985068.post@n2.nabble.com>


I am investigating the cause of this error. If anybody has windows machines
with other locales (non US English) available please report back here with
the results. I have only tested the installer with an English locale at this
time. If someone could confirm that thats the problem I would be grateful
and will then work on a fix.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FSClient-Release-A-FreeSWITCH-SIP-Client-for-Windows-in-NET-tp5983787p5985068.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jeff at jefflenk.com  Wed Feb  2 18:01:19 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 2 Feb 2011 07:01:19 -0800 (PST)
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
	for Windows in .NET
In-Reply-To: <16100.1296638658@ccs.covici.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
	<16100.1296638658@ccs.covici.com>
Message-ID: <1296658879889-5985076.post@n2.nabble.com>


John,

What OS? Did you get prompted for downloading the .Net framework when you
ran setup?
There is much to test here any help diagnosing these problems is great.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FSClient-Release-A-FreeSWITCH-SIP-Client-for-Windows-in-NET-tp5983787p5985076.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From infos at madovsky.org  Wed Feb  2 18:02:35 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 2 Feb 2011 10:02:35 -0500
Subject: [Freeswitch-users] How to prevent source channel hangup after
	uuid_bridge completion (outbound ESL)?
References: <AANLkTim7PoaVVCopbk6-wT1E04=pabKw9FjZaHCbLH5=@mail.gmail.com><AANLkTik-rV6wwBjm3DsEDGVszEizKhgbQfYzN_TKvBhA@mail.gmail.com><AANLkTimhBqbp5upa-me21AsWNYACr14FZQkStRK=sYXW@mail.gmail.com>
	<AANLkTi=yhk7E30GTQ7ykeJotcXX5KOTWr6aSOj8_wCyY@mail.gmail.com>
Message-ID: <88ADC8B9514648949D47C518A0900BC4@e1705>

session_in_hangup_hook ?
----- Original Message ----- 
From: "Dmitry Sytchev" <kbdfck at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, February 02, 2011 6:43 AM
Subject: Re: [Freeswitch-users] How to prevent source channel hangup after 
uuid_bridge completion (outbound ESL)?


> hangup_after_bridge=false doesn't prevent source channel to be hangup.
> socket application we use to create outbound ESL connection is
> finished on originated channel hangup, and since there is no more apps
> to run in dialplan, channel hangs up.
>
> But I need to stay in ESL script after channel I originated hangs up.
> How to do this?
>
>
> 2011/2/1 Dmitry Sytchev <kbdfck at gmail.com>:
>> Thanks, will try this. For some reason I forgot about this, although
>> using it in dialplan :)))
>>
>> 2011/2/1 Christopher Rienzo <cmrienzo at gmail.com>:
>>> Set the hangup_after_bridge channel variable to false.
>>>
>>>
>>> On Tue, Feb 1, 2011 at 9:34 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>>>
>>>> Hi
>>>>
>>>> I'm trying to originate call from outbound ESL and then bridge it to
>>>> original channel where ESL is launched on.
>>>> I use api originate with &park(), then uuid_bridge. Everything works
>>>> as expected, but when second leg hangs up, first leg gets hangup too.
>>>> How can I prevent first leg from hanging up and continue to process it
>>>> in ESL?
>>>>
>>>> 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
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 helmut.kuper at ewetel.de  Wed Feb  2 18:11:51 2011
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 02 Feb 2011 16:11:51 +0100
Subject: [Freeswitch-users] intercom in lua dialplan
In-Reply-To: <4D42DC25.4090407@ewetel.de>
References: <4D3EF036.80701@ewetel.de>	<AANLkTik8jrN_hzom=VEQZfx39YqMgZ15OE8kzZWqJ8iO@mail.gmail.com>	<4D3FD1D9.2@ewetel.de>
	<4D42DC25.4090407@ewetel.de>
Message-ID: <4D497437.20105@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Michael,

did you find some time to look at the log file?


Am 28.01.2011 16:09, schrieb Helmut Kuper:
> Hi Michael,
> 
> any news to this?

regards
helmut
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.10 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/

iEYEARECAAYFAk1JdDcACgkQ4tZeNddg3dwtdgCfYdtrUFRSKFMf7JyujrZwbWJp
zpEAn3Sw7bxZ9LiUMsH3Gv+udNaWRtTm
=Ei5z
-----END PGP SIGNATURE-----


From covici at ccs.covici.com  Wed Feb  2 18:15:05 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 02 Feb 2011 10:15:05 -0500
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
	for Windows in .NET
In-Reply-To: <1296658879889-5985076.post@n2.nabble.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
	<16100.1296638658@ccs.covici.com>
	<1296658879889-5985076.post@n2.nabble.com>
Message-ID: <6549.1296659705@ccs.covici.com>

I am pretty sure I have the .net framework already.  I am using
WindowEyes as the screenreader, so there may be some interaction -- let
me see what .net framework I have -- it says Microsoft net framework 4
client profile.

Jeff Lenk <jeff at jefflenk.com> wrote:

> 
> John,
> 
> What OS? Did you get prompted for downloading the .Net framework when you
> ran setup?
> There is much to test here any help diagnosing these problems is great.
> -- 
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FSClient-Release-A-FreeSWITCH-SIP-Client-for-Windows-in-NET-tp5983787p5985076.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

-- 
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 jeff at jefflenk.com  Wed Feb  2 18:21:16 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 2 Feb 2011 07:21:16 -0800 (PST)
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
	for Windows in .NET
In-Reply-To: <6549.1296659705@ccs.covici.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
	<16100.1296638658@ccs.covici.com>
	<1296658879889-5985076.post@n2.nabble.com>
	<6549.1296659705@ccs.covici.com>
Message-ID: <1296660076398-5985148.post@n2.nabble.com>


Thanks John,

Yes thats the correct version. This is a WPF application so maybe thats the
issue we are having with the screen reader. I will do some investigation to
whether that is the cause.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FSClient-Release-A-FreeSWITCH-SIP-Client-for-Windows-in-NET-tp5983787p5985148.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From ayhkor at gmail.com  Wed Feb  2 18:22:29 2011
From: ayhkor at gmail.com (deniro)
Date: Wed, 2 Feb 2011 10:22:29 -0500
Subject: [Freeswitch-users] mod_perl mod_lua
In-Reply-To: <AANLkTinDSQkOJrAikHrTEm6YTfyMcv2yLfxGPucjDwVq@mail.gmail.com>
References: <AANLkTikxbN47Nc7cG8y0HMDVqC0HHJ56aLQZHJYS2WTR@mail.gmail.com>
	<AANLkTinDSQkOJrAikHrTEm6YTfyMcv2yLfxGPucjDwVq@mail.gmail.com>
Message-ID: <AANLkTi=0=+05cxkP8ER9WKmzgAS4HzymLrTHcrMVho6P@mail.gmail.com>

I think that is to enable loading the module(not the install)
thx



On Wed, Feb 2, 2011 at 3:14 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

>  On Wed, Feb 2, 2011 at 5:25 AM, deniro <ayhkor at gmail.com> wrote:
> > Hi
> > I have compiled version of freeswitch 1.6.x on ubuntu 10.04
> > I want to install extra freeswitch modules like mod_perl and mod_lua
> > Is there any way to install these modules without re-compileling
> freeswitch.
> > I looked at freeswitch site and it is talking about  editing modules.conf
> > file and compiling from source
> > and I dont even have modules.conf file as I searched.
>
> Go to your original sources directory (where you gave the command
> "make install"), edit the file modules.conf.xml and give the command
> "make install".
>
> -giovanni
>
>
>
> >
> > thx
> > 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
> >
> >
>
>
>
> --
> 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 kbdfck at gmail.com  Wed Feb  2 18:29:21 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Wed, 2 Feb 2011 18:29:21 +0300
Subject: [Freeswitch-users] How to prevent source channel hangup after
 uuid_bridge completion (outbound ESL)?
In-Reply-To: <88ADC8B9514648949D47C518A0900BC4@e1705>
References: <AANLkTim7PoaVVCopbk6-wT1E04=pabKw9FjZaHCbLH5=@mail.gmail.com>
	<AANLkTik-rV6wwBjm3DsEDGVszEizKhgbQfYzN_TKvBhA@mail.gmail.com>
	<AANLkTimhBqbp5upa-me21AsWNYACr14FZQkStRK=sYXW@mail.gmail.com>
	<AANLkTi=yhk7E30GTQ7ykeJotcXX5KOTWr6aSOj8_wCyY@mail.gmail.com>
	<88ADC8B9514648949D47C518A0900BC4@e1705>
Message-ID: <AANLkTimmBWk13avJaa6cSw=tDV2xKJYexPj0Bmjc0XUb@mail.gmail.com>

It doesn't help, because even if I set api_hangup_hook, after initial
channel was bridged with uuid_bridge in ESL script I can't get it
parked, for example, with uuid_park since api_hangup_hook seems to be
called after channel is destroyed.

I'm doing uuid_bridge from ESL script, and I need to stay in script
when bridge destroys.

Here is my sequence:

A calls in FS
FS lauches socket app to create outbound ESL connection
ESL script forks and listens for DTMF events
When user presses 1, I launch api "originate sofia/external/something &park()"
When user presses 2, I do "uuid_bridge source_uuid originated_uuid"
After originated channel hangs up, socket app finishes and tries to
continue on dialplan.
I need to prevent this behaviour and let ESL stay alive to decide what
to do next with source channel.



2011/2/2 Madovsky <infos at madovsky.org>:
> session_in_hangup_hook ?
> ----- Original Message -----
> From: "Dmitry Sytchev" <kbdfck at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, February 02, 2011 6:43 AM
> Subject: Re: [Freeswitch-users] How to prevent source channel hangup after
> uuid_bridge completion (outbound ESL)?
>
>
>> hangup_after_bridge=false doesn't prevent source channel to be hangup.
>> socket application we use to create outbound ESL connection is
>> finished on originated channel hangup, and since there is no more apps
>> to run in dialplan, channel hangs up.
>>
>> But I need to stay in ESL script after channel I originated hangs up.
>> How to do this?
>>
>>
>> 2011/2/1 Dmitry Sytchev <kbdfck at gmail.com>:
>>> Thanks, will try this. For some reason I forgot about this, although
>>> using it in dialplan :)))
>>>
>>> 2011/2/1 Christopher Rienzo <cmrienzo at gmail.com>:
>>>> Set the hangup_after_bridge channel variable to false.
>>>>
>>>>
>>>> On Tue, Feb 1, 2011 at 9:34 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>>>>
>>>>> Hi
>>>>>
>>>>> I'm trying to originate call from outbound ESL and then bridge it to
>>>>> original channel where ESL is launched on.
>>>>> I use api originate with &park(), then uuid_bridge. Everything works
>>>>> as expected, but when second leg hangs up, first leg gets hangup too.
>>>>> How can I prevent first leg from hanging up and continue to process it
>>>>> in ESL?
>>>>>
>>>>> 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
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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


From steveayre at gmail.com  Wed Feb  2 18:30:19 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 2 Feb 2011 15:30:19 +0000
Subject: [Freeswitch-users] mod_perl mod_lua
In-Reply-To: <AANLkTi=0=+05cxkP8ER9WKmzgAS4HzymLrTHcrMVho6P@mail.gmail.com>
References: <AANLkTikxbN47Nc7cG8y0HMDVqC0HHJ56aLQZHJYS2WTR@mail.gmail.com>
	<AANLkTinDSQkOJrAikHrTEm6YTfyMcv2yLfxGPucjDwVq@mail.gmail.com>
	<AANLkTi=0=+05cxkP8ER9WKmzgAS4HzymLrTHcrMVho6P@mail.gmail.com>
Message-ID: <AANLkTinjH2xhjkEXkpuBb-+HFJnG-Hr==hKY18b=cwt5@mail.gmail.com>

He meant it without the .xml suffix...

modules.conf in the git checkout controls which modules are built
modules.conf.xml in conf/autoload_configs controls which modules are loaded

-Steve


On 2 February 2011 15:22, deniro <ayhkor at gmail.com> wrote:

> I think that is to enable loading the module(not the install)
> thx
>
>
>
> On Wed, Feb 2, 2011 at 3:14 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:
>
>>  On Wed, Feb 2, 2011 at 5:25 AM, deniro <ayhkor at gmail.com> wrote:
>> > Hi
>> > I have compiled version of freeswitch 1.6.x on ubuntu 10.04
>> > I want to install extra freeswitch modules like mod_perl and mod_lua
>> > Is there any way to install these modules without re-compileling
>> freeswitch.
>> > I looked at freeswitch site and it is talking about  editing
>> modules.conf
>> > file and compiling from source
>> > and I dont even have modules.conf file as I searched.
>>
>> Go to your original sources directory (where you gave the command
>> "make install"), edit the file modules.conf.xml and give the command
>> "make install".
>>
>> -giovanni
>>
>>
>>
>> >
>> > thx
>> > 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
>> >
>> >
>>
>>
>>
>> --
>> 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
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mitch.capper at gmail.com  Wed Feb  2 18:30:31 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 2 Feb 2011 07:30:31 -0800
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
 for Windows in .NET
In-Reply-To: <6549.1296659705@ccs.covici.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
	<16100.1296638658@ccs.covici.com>
	<1296658879889-5985076.post@n2.nabble.com>
	<6549.1296659705@ccs.covici.com>
Message-ID: <AANLkTinCVZqYRxd0ZcQLj_kj42fYHDqUsEocEqW3oaph@mail.gmail.com>

Hi Guys, I am sorry for the issues with the installer we rushed to get one
together to make it easier on everyone for the conference today and did not
have a chance to have too many others test it.   First per the "Elevation
Question" not the installer will not prompt for elevation until after you
select install (basically setup.exe collects the install options then runs
the MSI and should prompt at that time for admin rights).   In the mean time
I have put together a ZIP that you should be able to exact to a directory
for now and run, you can find it at: http://fluky.org/FSClientPortable.zip

Technically the installer should make sure you have .net 4.0 installed so if
you use the zip make sure you already have it installed otherwise it is at:
http://www.microsoft.com/downloads/en/details.aspx?FamilyID=0a391abd-25c1-4fc0-919f-b21f31ab88b7

Also please note, when the app starts up it does take 5-15 seconds to load
while freeswitch is spinning up before you can do anything.

~Mitch
-------------- next part --------------
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From jeff at jefflenk.com  Wed Feb  2 18:32:47 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 2 Feb 2011 07:32:47 -0800 (PST)
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
	for	Windows in .NET
In-Reply-To: <1296658701797-5985068.post@n2.nabble.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
	<4D491068.7090308@solomo.de>
	<1296658701797-5985068.post@n2.nabble.com>
Message-ID: <1296660767248-5985201.post@n2.nabble.com>


I have uploaded a version of the installer that should fix the "Authenticated
User" security problem. Please test and let me know.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FSClient-Release-A-FreeSWITCH-SIP-Client-for-Windows-in-NET-tp5983787p5985201.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From gmaruzz at gmail.com  Wed Feb  2 18:32:46 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 2 Feb 2011 16:32:46 +0100
Subject: [Freeswitch-users] mod_perl mod_lua
In-Reply-To: <AANLkTi=0=+05cxkP8ER9WKmzgAS4HzymLrTHcrMVho6P@mail.gmail.com>
References: <AANLkTikxbN47Nc7cG8y0HMDVqC0HHJ56aLQZHJYS2WTR@mail.gmail.com>
	<AANLkTinDSQkOJrAikHrTEm6YTfyMcv2yLfxGPucjDwVq@mail.gmail.com>
	<AANLkTi=0=+05cxkP8ER9WKmzgAS4HzymLrTHcrMVho6P@mail.gmail.com>
Message-ID: <AANLkTi=bNWH97qZdDmuEvzm6ETaZDRVv1R3Rt7hLTg+L@mail.gmail.com>

On Wed, Feb 2, 2011 at 4:22 PM, deniro <ayhkor at gmail.com> wrote:
> I?think that is to enable loading the module(not the install)
> thx

So, you write to the mailing list for advice, and you don't put
confidence in answers you got back? ;)

>
>
> On Wed, Feb 2, 2011 at 3:14 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>>
>> On Wed, Feb 2, 2011 at 5:25 AM, deniro <ayhkor at gmail.com> wrote:
>> > Hi
>> > I have compiled version of freeswitch 1.6.x on ubuntu 10.04
>> > I want to install extra freeswitch modules like mod_perl and mod_lua
>> > Is there any way to install these modules without re-compileling
>> > freeswitch.
>> > I looked at freeswitch site and it is talking about? editing
>> > modules.conf
>> > file and compiling from source
>> > and I dont even have modules.conf file as I searched.
>>
>> Go to your original sources directory (where you gave the command
>> "make install"), edit the file modules.conf.xml and give the command
>> "make install".
>>
>> -giovanni
>>
>>
>>
>> >
>> > thx
>> > 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
>> >
>> >
>>
>>
>>
>> --
>> 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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 infos at madovsky.org  Wed Feb  2 18:34:53 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 2 Feb 2011 10:34:53 -0500
Subject: [Freeswitch-users] How to prevent source channel hangup after
	uuid_bridge completion (outbound ESL)?
References: <AANLkTim7PoaVVCopbk6-wT1E04=pabKw9FjZaHCbLH5=@mail.gmail.com><AANLkTik-rV6wwBjm3DsEDGVszEizKhgbQfYzN_TKvBhA@mail.gmail.com><AANLkTimhBqbp5upa-me21AsWNYACr14FZQkStRK=sYXW@mail.gmail.com><AANLkTi=yhk7E30GTQ7ykeJotcXX5KOTWr6aSOj8_wCyY@mail.gmail.com><88ADC8B9514648949D47C518A0900BC4@e1705>
	<AANLkTimmBWk13avJaa6cSw=tDV2xKJYexPj0Bmjc0XUb@mail.gmail.com>
Message-ID: <AAEB0F4B1C8E4236AE9B208DB8365F3D@e1705>

not sure you can continue the dialplan after hangup from esl
because I'm looking the same need and didnt' find the trick.
did you try in sync full mode ?

----- Original Message ----- 
From: "Dmitry Sytchev" <kbdfck at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, February 02, 2011 10:29 AM
Subject: Re: [Freeswitch-users] How to prevent source channel hangup after 
uuid_bridge completion (outbound ESL)?


> It doesn't help, because even if I set api_hangup_hook, after initial
> channel was bridged with uuid_bridge in ESL script I can't get it
> parked, for example, with uuid_park since api_hangup_hook seems to be
> called after channel is destroyed.
>
> I'm doing uuid_bridge from ESL script, and I need to stay in script
> when bridge destroys.
>
> Here is my sequence:
>
> A calls in FS
> FS lauches socket app to create outbound ESL connection
> ESL script forks and listens for DTMF events
> When user presses 1, I launch api "originate sofia/external/something 
> &park()"
> When user presses 2, I do "uuid_bridge source_uuid originated_uuid"
> After originated channel hangs up, socket app finishes and tries to
> continue on dialplan.
> I need to prevent this behaviour and let ESL stay alive to decide what
> to do next with source channel.
>
>
>
> 2011/2/2 Madovsky <infos at madovsky.org>:
>> session_in_hangup_hook ?
>> ----- Original Message -----
>> From: "Dmitry Sytchev" <kbdfck at gmail.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Wednesday, February 02, 2011 6:43 AM
>> Subject: Re: [Freeswitch-users] How to prevent source channel hangup 
>> after
>> uuid_bridge completion (outbound ESL)?
>>
>>
>>> hangup_after_bridge=false doesn't prevent source channel to be hangup.
>>> socket application we use to create outbound ESL connection is
>>> finished on originated channel hangup, and since there is no more apps
>>> to run in dialplan, channel hangs up.
>>>
>>> But I need to stay in ESL script after channel I originated hangs up.
>>> How to do this?
>>>
>>>
>>> 2011/2/1 Dmitry Sytchev <kbdfck at gmail.com>:
>>>> Thanks, will try this. For some reason I forgot about this, although
>>>> using it in dialplan :)))
>>>>
>>>> 2011/2/1 Christopher Rienzo <cmrienzo at gmail.com>:
>>>>> Set the hangup_after_bridge channel variable to false.
>>>>>
>>>>>
>>>>> On Tue, Feb 1, 2011 at 9:34 AM, Dmitry Sytchev <kbdfck at gmail.com> 
>>>>> wrote:
>>>>>>
>>>>>> Hi
>>>>>>
>>>>>> I'm trying to originate call from outbound ESL and then bridge it to
>>>>>> original channel where ESL is launched on.
>>>>>> I use api originate with &park(), then uuid_bridge. Everything works
>>>>>> as expected, but when second leg hangs up, first leg gets hangup too.
>>>>>> How can I prevent first leg from hanging up and continue to process 
>>>>>> it
>>>>>> in ESL?
>>>>>>
>>>>>> 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
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 



From covici at ccs.covici.com  Wed Feb  2 18:57:22 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 02 Feb 2011 10:57:22 -0500
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
	for Windows in .NET
In-Reply-To: <1296660076398-5985148.post@n2.nabble.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
	<16100.1296638658@ccs.covici.com>
	<1296658879889-5985076.post@n2.nabble.com>
	<6549.1296659705@ccs.covici.com>
	<1296660076398-5985148.post@n2.nabble.com>
Message-ID: <7068.1296662242@ccs.covici.com>

I am able to use 2011 version of msn messenger, so I should be good to
go if things are properly written.

Jeff Lenk <jeff at jefflenk.com> wrote:

> 
> Thanks John,
> 
> Yes thats the correct version. This is a WPF application so maybe thats the
> issue we are having with the screen reader. I will do some investigation to
> whether that is the cause.
> -- 
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FSClient-Release-A-FreeSWITCH-SIP-Client-for-Windows-in-NET-tp5983787p5985148.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

-- 
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  Wed Feb  2 19:00:50 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 02 Feb 2011 11:00:50 -0500
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
	for Windows in .NET
In-Reply-To: <AANLkTinCVZqYRxd0ZcQLj_kj42fYHDqUsEocEqW3oaph@mail.gmail.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
	<16100.1296638658@ccs.covici.com>
	<1296658879889-5985076.post@n2.nabble.com>
	<6549.1296659705@ccs.covici.com>
	<AANLkTinCVZqYRxd0ZcQLj_kj42fYHDqUsEocEqW3oaph@mail.gmail.com>
Message-ID: <7179.1296662450@ccs.covici.com>

The current installer does not ask for admin rights -- it just dies with
a message about some policy preventing the install, but it does install
if you run as administrator.

Mitch Capper <mitch.capper at gmail.com> wrote:

> Hi Guys, I am sorry for the issues with the installer we rushed to get one
> together to make it easier on everyone for the conference today and did not
> have a chance to have too many others test it.   First per the "Elevation
> Question" not the installer will not prompt for elevation until after you
> select install (basically setup.exe collects the install options then runs
> the MSI and should prompt at that time for admin rights).   In the mean time
> I have put together a ZIP that you should be able to exact to a directory
> for now and run, you can find it at: http://fluky.org/FSClientPortable.zip
> 
> Technically the installer should make sure you have .net 4.0 installed so if
> you use the zip make sure you already have it installed otherwise it is at:
> http://www.microsoft.com/downloads/en/details.aspx?FamilyID=0a391abd-25c1-4fc0-919f-b21f31ab88b7
> 
> Also please note, when the app starts up it does take 5-15 seconds to load
> while freeswitch is spinning up before you can do anything.
> 
> ~Mitch
> 
> ----------------------------------------------------
> 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 kbdfck at gmail.com  Wed Feb  2 19:01:57 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Wed, 2 Feb 2011 19:01:57 +0300
Subject: [Freeswitch-users] How to prevent source channel hangup after
 uuid_bridge completion (outbound ESL)?
In-Reply-To: <AAEB0F4B1C8E4236AE9B208DB8365F3D@e1705>
References: <AANLkTim7PoaVVCopbk6-wT1E04=pabKw9FjZaHCbLH5=@mail.gmail.com>
	<AANLkTik-rV6wwBjm3DsEDGVszEizKhgbQfYzN_TKvBhA@mail.gmail.com>
	<AANLkTimhBqbp5upa-me21AsWNYACr14FZQkStRK=sYXW@mail.gmail.com>
	<AANLkTi=yhk7E30GTQ7ykeJotcXX5KOTWr6aSOj8_wCyY@mail.gmail.com>
	<88ADC8B9514648949D47C518A0900BC4@e1705>
	<AANLkTimmBWk13avJaa6cSw=tDV2xKJYexPj0Bmjc0XUb@mail.gmail.com>
	<AAEB0F4B1C8E4236AE9B208DB8365F3D@e1705>
Message-ID: <AANLkTi=c-UNVkCPc7ie1PpJzQU-r_ABxwmoC3TWMRhar@mail.gmail.com>

I'm trying in async full

I found single solution at the moment - catch CHANNEL_HANGUP from
originated channel and park source channel in event handler, this
works and keeps source channel from hanging up.

But what will happen with source channel I don't get event in time or
delay its processing?

2011/2/2 Madovsky <infos at madovsky.org>:
> not sure you can continue the dialplan after hangup from esl
> because I'm looking the same need and didnt' find the trick.
> did you try in sync full mode ?
>
> ----- Original Message -----
> From: "Dmitry Sytchev" <kbdfck at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, February 02, 2011 10:29 AM
> Subject: Re: [Freeswitch-users] How to prevent source channel hangup after
> uuid_bridge completion (outbound ESL)?
>
>
>> It doesn't help, because even if I set api_hangup_hook, after initial
>> channel was bridged with uuid_bridge in ESL script I can't get it
>> parked, for example, with uuid_park since api_hangup_hook seems to be
>> called after channel is destroyed.
>>
>> I'm doing uuid_bridge from ESL script, and I need to stay in script
>> when bridge destroys.
>>
>> Here is my sequence:
>>
>> A calls in FS
>> FS lauches socket app to create outbound ESL connection
>> ESL script forks and listens for DTMF events
>> When user presses 1, I launch api "originate sofia/external/something
>> &park()"
>> When user presses 2, I do "uuid_bridge source_uuid originated_uuid"
>> After originated channel hangs up, socket app finishes and tries to
>> continue on dialplan.
>> I need to prevent this behaviour and let ESL stay alive to decide what
>> to do next with source channel.
>>
>>
>>
>> 2011/2/2 Madovsky <infos at madovsky.org>:
>>> session_in_hangup_hook ?
>>> ----- Original Message -----
>>> From: "Dmitry Sytchev" <kbdfck at gmail.com>
>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>> Sent: Wednesday, February 02, 2011 6:43 AM
>>> Subject: Re: [Freeswitch-users] How to prevent source channel hangup
>>> after
>>> uuid_bridge completion (outbound ESL)?
>>>
>>>
>>>> hangup_after_bridge=false doesn't prevent source channel to be hangup.
>>>> socket application we use to create outbound ESL connection is
>>>> finished on originated channel hangup, and since there is no more apps
>>>> to run in dialplan, channel hangs up.
>>>>
>>>> But I need to stay in ESL script after channel I originated hangs up.
>>>> How to do this?
>>>>
>>>>
>>>> 2011/2/1 Dmitry Sytchev <kbdfck at gmail.com>:
>>>>> Thanks, will try this. For some reason I forgot about this, although
>>>>> using it in dialplan :)))
>>>>>
>>>>> 2011/2/1 Christopher Rienzo <cmrienzo at gmail.com>:
>>>>>> Set the hangup_after_bridge channel variable to false.
>>>>>>
>>>>>>
>>>>>> On Tue, Feb 1, 2011 at 9:34 AM, Dmitry Sytchev <kbdfck at gmail.com>
>>>>>> wrote:
>>>>>>>
>>>>>>> Hi
>>>>>>>
>>>>>>> I'm trying to originate call from outbound ESL and then bridge it to
>>>>>>> original channel where ESL is launched on.
>>>>>>> I use api originate with &park(), then uuid_bridge. Everything works
>>>>>>> as expected, but when second leg hangs up, first leg gets hangup too.
>>>>>>> How can I prevent first leg from hanging up and continue to process
>>>>>>> it
>>>>>>> in ESL?
>>>>>>>
>>>>>>> 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
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at 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
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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


From jeff at jefflenk.com  Wed Feb  2 19:12:41 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 2 Feb 2011 08:12:41 -0800 (PST)
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
	for Windows in .NET
In-Reply-To: <7179.1296662450@ccs.covici.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
	<16100.1296638658@ccs.covici.com>
	<1296658879889-5985076.post@n2.nabble.com>
	<6549.1296659705@ccs.covici.com>
	<AANLkTinCVZqYRxd0ZcQLj_kj42fYHDqUsEocEqW3oaph@mail.gmail.com>
	<7179.1296662450@ccs.covici.com>
Message-ID: <1296663161529-5985392.post@n2.nabble.com>


John,

What is your OS? I'm guessing its XP.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FSClient-Release-A-FreeSWITCH-SIP-Client-for-Windows-in-NET-tp5983787p5985392.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From infos at madovsky.org  Wed Feb  2 19:15:16 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 2 Feb 2011 11:15:16 -0500
Subject: [Freeswitch-users] execute_on_answer
Message-ID: <37FAA1D493F546F886444D59414F7992@e1705>

possible to use
execute_on_answer=<api> value; <api> value

???

thanks
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From infos at madovsky.org  Wed Feb  2 19:17:27 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 2 Feb 2011 11:17:27 -0500
Subject: [Freeswitch-users] How to prevent source channel hangup after
	uuid_bridge completion (outbound ESL)?
References: <AANLkTim7PoaVVCopbk6-wT1E04=pabKw9FjZaHCbLH5=@mail.gmail.com><AANLkTik-rV6wwBjm3DsEDGVszEizKhgbQfYzN_TKvBhA@mail.gmail.com><AANLkTimhBqbp5upa-me21AsWNYACr14FZQkStRK=sYXW@mail.gmail.com><AANLkTi=yhk7E30GTQ7ykeJotcXX5KOTWr6aSOj8_wCyY@mail.gmail.com><88ADC8B9514648949D47C518A0900BC4@e1705><AANLkTimmBWk13avJaa6cSw=tDV2xKJYexPj0Bmjc0XUb@mail.gmail.com><AAEB0F4B1C8E4236AE9B208DB8365F3D@e1705>
	<AANLkTi=c-UNVkCPc7ie1PpJzQU-r_ABxwmoC3TWMRhar@mail.gmail.com>
Message-ID: <A5F5120D483B499B8F05F330EC09D463@e1705>

> I'm trying in async full
try full only

>
> I found single solution at the moment - catch CHANNEL_HANGUP from
> originated channel and park source channel in event handler, this
> works and keeps source channel from hanging up.

> But what will happen with source channel I don't get event in time or
> delay its processing?
not a stable solution I think


> 2011/2/2 Madovsky <infos at madovsky.org>:
>> not sure you can continue the dialplan after hangup from esl
>> because I'm looking the same need and didnt' find the trick.
>> did you try in sync full mode ?
>>
>> ----- Original Message -----
>> From: "Dmitry Sytchev" <kbdfck at gmail.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Wednesday, February 02, 2011 10:29 AM
>> Subject: Re: [Freeswitch-users] How to prevent source channel hangup 
>> after
>> uuid_bridge completion (outbound ESL)?
>>
>>
>>> It doesn't help, because even if I set api_hangup_hook, after initial
>>> channel was bridged with uuid_bridge in ESL script I can't get it
>>> parked, for example, with uuid_park since api_hangup_hook seems to be
>>> called after channel is destroyed.
>>>
>>> I'm doing uuid_bridge from ESL script, and I need to stay in script
>>> when bridge destroys.
>>>
>>> Here is my sequence:
>>>
>>> A calls in FS
>>> FS lauches socket app to create outbound ESL connection
>>> ESL script forks and listens for DTMF events
>>> When user presses 1, I launch api "originate sofia/external/something
>>> &park()"
>>> When user presses 2, I do "uuid_bridge source_uuid originated_uuid"
>>> After originated channel hangs up, socket app finishes and tries to
>>> continue on dialplan.
>>> I need to prevent this behaviour and let ESL stay alive to decide what
>>> to do next with source channel.
>>>
>>>
>>>
>>> 2011/2/2 Madovsky <infos at madovsky.org>:
>>>> session_in_hangup_hook ?
>>>> ----- Original Message -----
>>>> From: "Dmitry Sytchev" <kbdfck at gmail.com>
>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Wednesday, February 02, 2011 6:43 AM
>>>> Subject: Re: [Freeswitch-users] How to prevent source channel hangup
>>>> after
>>>> uuid_bridge completion (outbound ESL)?
>>>>
>>>>
>>>>> hangup_after_bridge=false doesn't prevent source channel to be hangup.
>>>>> socket application we use to create outbound ESL connection is
>>>>> finished on originated channel hangup, and since there is no more apps
>>>>> to run in dialplan, channel hangs up.
>>>>>
>>>>> But I need to stay in ESL script after channel I originated hangs up.
>>>>> How to do this?
>>>>>
>>>>>
>>>>> 2011/2/1 Dmitry Sytchev <kbdfck at gmail.com>:
>>>>>> Thanks, will try this. For some reason I forgot about this, although
>>>>>> using it in dialplan :)))
>>>>>>
>>>>>> 2011/2/1 Christopher Rienzo <cmrienzo at gmail.com>:
>>>>>>> Set the hangup_after_bridge channel variable to false.
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Feb 1, 2011 at 9:34 AM, Dmitry Sytchev <kbdfck at gmail.com>
>>>>>>> wrote:
>>>>>>>>
>>>>>>>> Hi
>>>>>>>>
>>>>>>>> I'm trying to originate call from outbound ESL and then bridge it 
>>>>>>>> to
>>>>>>>> original channel where ESL is launched on.
>>>>>>>> I use api originate with &park(), then uuid_bridge. Everything 
>>>>>>>> works
>>>>>>>> as expected, but when second leg hangs up, first leg gets hangup 
>>>>>>>> too.
>>>>>>>> How can I prevent first leg from hanging up and continue to process
>>>>>>>> it
>>>>>>>> in ESL?
>>>>>>>>
>>>>>>>> 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
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at 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
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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 



From ayhkor at gmail.com  Wed Feb  2 19:26:10 2011
From: ayhkor at gmail.com (deniro)
Date: Wed, 2 Feb 2011 11:26:10 -0500
Subject: [Freeswitch-users] mod_perl mod_lua
In-Reply-To: <AANLkTi=bNWH97qZdDmuEvzm6ETaZDRVv1R3Rt7hLTg+L@mail.gmail.com>
References: <AANLkTikxbN47Nc7cG8y0HMDVqC0HHJ56aLQZHJYS2WTR@mail.gmail.com>
	<AANLkTinDSQkOJrAikHrTEm6YTfyMcv2yLfxGPucjDwVq@mail.gmail.com>
	<AANLkTi=0=+05cxkP8ER9WKmzgAS4HzymLrTHcrMVho6P@mail.gmail.com>
	<AANLkTi=bNWH97qZdDmuEvzm6ETaZDRVv1R3Rt7hLTg+L@mail.gmail.com>
Message-ID: <AANLkTinJBQ_EskC4Fv72FauNgbr4r4fF_gnOjiD6NDSe@mail.gmail.com>

thanks for your advice and  I appreciate you taking time to reply.
You also note that I first check freeswitch site and I put my questions if I
cant find a solutions myself.
I already checked for options with modules.conf and modules.conf.xml even
before posting.

If you look at my first posting, I stated that I have a compiled freeswitch
and I dont even see modules.conf file (I searched).
so to reiterate, The freeswitch comes installed and compiled already with a
product, so I dont even have the freeswitch source.

All I am loooking for is, if there is any way, to install new modules
without re-installig from source and recompile from the scratch with
existing freeswitch install.
I highly doubt that this is possible but I am checking out with the gurus
here.
I dont wanna break already running freeswitch with custom dialplans and
other custom configurations.

thx again
deniro--







On Wed, Feb 2, 2011 at 10:32 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> On Wed, Feb 2, 2011 at 4:22 PM, deniro <ayhkor at gmail.com> wrote:
> > I think that is to enable loading the module(not the install)
> > thx
>
> So, you write to the mailing list for advice, and you don't put
> confidence in answers you got back? ;)
>
> >
> >
> > On Wed, Feb 2, 2011 at 3:14 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> > wrote:
> >>
> >> On Wed, Feb 2, 2011 at 5:25 AM, deniro <ayhkor at gmail.com> wrote:
> >> > Hi
> >> > I have compiled version of freeswitch 1.6.x on ubuntu 10.04
> >> > I want to install extra freeswitch modules like mod_perl and mod_lua
> >> > Is there any way to install these modules without re-compileling
> >> > freeswitch.
> >> > I looked at freeswitch site and it is talking about  editing
> >> > modules.conf
> >> > file and compiling from source
> >> > and I dont even have modules.conf file as I searched.
> >>
> >> Go to your original sources directory (where you gave the command
> >> "make install"), edit the file modules.conf.xml and give the command
> >> "make install".
> >>
> >> -giovanni
> >>
> >>
> >>
> >> >
> >> > thx
> >> > 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
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> 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
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kapil.rastogi at telemune.net  Wed Feb  2 19:28:22 2011
From: kapil.rastogi at telemune.net (kapil.rastogi)
Date: Wed, 2 Feb 2011 08:28:22 -0800 (PST)
Subject: [Freeswitch-users] How to record a conference using javascript
	application
Message-ID: <1296664102854-5985457.post@n2.nabble.com>


Hi,

I want to record the chatting b/w all members during the conference. I am
using the following statement in my javascript application:

 session.execute("conference", "conf-room--"+room_id+"@default record
/usr/local/freeswitch/abc.wav");

But i am unable to record the conference room chatting. But when i record
the same from fs_cli command line, it is working well.

Please tell me how to record the conference chatting.

Regards,

-----
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/How-to-record-a-conference-using-javascript-application-tp5985457p5985457.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From fdelawarde at wirelessmundi.com  Wed Feb  2 19:33:05 2011
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Wed, 02 Feb 2011 17:33:05 +0100
Subject: [Freeswitch-users] execute_on_ring executing on answer
In-Reply-To: <AANLkTimC-j3aqgnWbDP7JVf2F2BtGoB5cLnvf-qq=XqC@mail.gmail.com>
References: <1296215265.8986.147.camel@luna.tc.commsmundi.com>
	<21B4FB4AAC2E428E938F99986CA09165@e1705>
	<1296235095.8986.167.camel@luna.tc.commsmundi.com>
	<0E96944DB29F4CEB9ACA5994D19AB0D6@e1705>
	<AANLkTikHB-QnDyT0WPsEML8yv4AACnYnHrE37gpRsLS2@mail.gmail.com>
	<1296306637.8986.185.camel@luna.tc.commsmundi.com>
	<1296582543.5245.87.camel@luna.tc.commsmundi.com>
	<AANLkTim1Hq0vrOLGpQt4DcNKUY970HzFF1KHF8ewmwxz@mail.gmail.com>
	<AANLkTimC-j3aqgnWbDP7JVf2F2BtGoB5cLnvf-qq=XqC@mail.gmail.com>
Message-ID: <1296664385.29302.33.camel@luna.tc.commsmundi.com>

Thanks. It does not work, but just a bit of tweaking is necessary.

For what I could read, when using the new "::", app and arg are not
parsed from the variable before calling the
switch_core_session_execute_application_async.

Fran?ois.



On Tue, 2011-02-01 at 12:08 -0800, Michael Collins wrote:
> I believe I have a new item to document on the wiki. :) 
> Francois, let us know if the latest git works for you.
> Thanks,
> MC
> 
> On Tue, Feb 1, 2011 at 11:51 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>         I think I see why.
>         The app is queued at the right time but not executed until
>         media is active.
>         
>         I have changed the code so now when you supply ::<arg> at the
>         end of
>         the app name it will be async and when you don't it will be
>         executed
>         immediately.
>         
>         
>         On Tue, Feb 1, 2011 at 11:49 AM, Fran?ois Delawarde
>         <fdelawarde at wirelessmundi.com> wrote:
>         > Since apparently noone reproduces this, it must be a
>         configuration error
>         > from my part.
>         >
>         > Any hints of where I could start looking to resolve this
>         issue?
>         >
>         > Thanks,
>         > Fran?ois.
>         >
>         >
>         > On Sat, 2011-01-29 at 14:10 +0100, Fran?ois Delawarde wrote:
>         >> Nice to know, but in that case the destination actually
>         rings (180).
>         >>
>         >> See commented log:
>         >> http://pastebin.freeswitch.org/15168
>         >>
>         >> Fran?ois.
>         >>
>         >> On Fri, 2011-01-28 at 12:34 -0600, Anthony Minessale wrote:
>         >> > if it never rings, answer will still trigger it.
>         >> >
>         >> >
>         >> > On Fri, Jan 28, 2011 at 11:24 AM, Madovsky
>         <infos at madovsky.org> wrote:
>         >> > > ah ok, maybe a wiki update would be useful.
>         >> > >
>         >> > >
>         >> > >
>         >> > > ----- Original Message -----
>         >> > > From: "Fran?ois Delawarde"
>         <fdelawarde at wirelessmundi.com>
>         >> > > To: "FreeSWITCH Users Help"
>         <freeswitch-users at lists.freeswitch.org>
>         >> > > Sent: Friday, January 28, 2011 12:18 PM
>         >> > > Subject: Re: [Freeswitch-users] execute_on_ring
>         executing on answer
>         >> > >
>         >> > >
>         >> > >> It's some cool feature made by Anthony that allows me
>         to specify the
>         >> > >> separator.
>         >> > >>
>         >> > >> in ^^:PCMA:G722
>         >> > >> ^^: means the separator is now : instead of ,
>         >> > >>
>         >> > >> Useful in the [] or {} case because the coma is
>         already used to separate
>         >> > >> variables.
>         >> > >>
>         >> > >> Fran?ois.
>         >> > >>
>         >> > >> On Fri, 2011-01-28 at 12:12 -0500, Madovsky wrote:
>         >> > >>> what means the ^^ in your codec string ?
>         >> > >>>
>         >> > >>> ----- Original Message -----
>         >> > >>> From: "Fran?ois Delawarde"
>         <fdelawarde at wirelessmundi.com>
>         >> > >>> To: "FreeSWITCH Users Help"
>         <freeswitch-users at lists.freeswitch.org>
>         >> > >>> Sent: Friday, January 28, 2011 6:47 AM
>         >> > >>> Subject: [Freeswitch-users] execute_on_ring executing
>         on answer
>         >> > >>>
>         >> > >>>
>         >> > >>> > Hi,
>         >> > >>> >
>         >> > >>> > Doing some testing with this morning's git (Fri Jan
>         28) I just found
>         >> > >>> > out
>         >> > >>> > that the execute_on_ring application runs when the
>         destination answers
>         >> > >>> > instead of when it rings.
>         >> > >>> >
>         >> > >>> > So far, I can't seem to find out the reason. Could
>         it be some
>         >> > >>> > configuration issue?
>         >> > >>> >
>         >> > >>> >
>         >> > >>> > Here a call log showing the phenomenon with a
>         simple bridge:
>         >> > >>> >
>         >> > >>> > <action application="bridge"
>         >> > >>> >
>         data="[execute_on_ring=info,absolute_codec_string=^^:PCMA:G722]sofia/192.168.10.1/sip:2103 at 192.168.10.22:5060"/>
>         >> > >>> >
>         >> > >>> > http://pastebin.freeswitch.org/15168
>         >> > >>> >
>         >> > >>> >
>         >> > >>> > Thanks,
>         >> > >>> > Fran?ois.
>         >> > >>> >
>         >> > >>> >
>         >> > >>> > _______________________________________________
>         >> > >>> > FreeSWITCH-users mailing list
>         >> > >>> > FreeSWITCH-users at lists.freeswitch.org
>         >> > >>> >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >> > >>> >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         >> > >>> > http://www.freeswitch.org
>         >> > >>> >
>         >> > >>>
>         >> > >>>
>         >> > >>> _______________________________________________
>         >> > >>> FreeSWITCH-users mailing list
>         >> > >>> FreeSWITCH-users at lists.freeswitch.org
>         >> > >>>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >> > >>>
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         >> > >>> http://www.freeswitch.org
>         >> > >>
>         >> > >>
>         >> > >> _______________________________________________
>         >> > >> FreeSWITCH-users mailing list
>         >> > >> FreeSWITCH-users at lists.freeswitch.org
>         >> > >>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >> > >>
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         >> > >> http://www.freeswitch.org
>         >> > >>
>         >> > >
>         >> > >
>         >> > > _______________________________________________
>         >> > > FreeSWITCH-users mailing list
>         >> > > FreeSWITCH-users at lists.freeswitch.org
>         >> > >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >> > >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         >> > > http://www.freeswitch.org
>         >> > >
>         >> >
>         >> >
>         >> >
>         >>
>         >>
>         >> _______________________________________________
>         >> FreeSWITCH-users mailing list
>         >> FreeSWITCH-users at lists.freeswitch.org
>         >>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >>
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         >> http://www.freeswitch.org
>         >
>         >
>         > _______________________________________________
>         > FreeSWITCH-users mailing list
>         > FreeSWITCH-users at 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 covici at ccs.covici.com  Wed Feb  2 19:34:06 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 02 Feb 2011 11:34:06 -0500
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
	for Windows in .NET
In-Reply-To: <1296663161529-5985392.post@n2.nabble.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
	<16100.1296638658@ccs.covici.com>
	<1296658879889-5985076.post@n2.nabble.com>
	<6549.1296659705@ccs.covici.com>
	<AANLkTinCVZqYRxd0ZcQLj_kj42fYHDqUsEocEqW3oaph@mail.gmail.com>
	<7179.1296662450@ccs.covici.com>
	<1296663161529-5985392.post@n2.nabble.com>
Message-ID: <8132.1296664446@ccs.covici.com>

Windows 7 32-bit.

Jeff Lenk <jeff at jefflenk.com> wrote:

> 
> John,
> 
> What is your OS? I'm guessing its XP.
> -- 
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FSClient-Release-A-FreeSWITCH-SIP-Client-for-Windows-in-NET-tp5983787p5985392.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

-- 
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 paul at cupis.co.uk  Wed Feb  2 19:41:54 2011
From: paul at cupis.co.uk (Paul Cupis)
Date: Wed, 02 Feb 2011 16:41:54 +0000
Subject: [Freeswitch-users] mod_perl mod_lua
In-Reply-To: <AANLkTinJBQ_EskC4Fv72FauNgbr4r4fF_gnOjiD6NDSe@mail.gmail.com>
References: <AANLkTikxbN47Nc7cG8y0HMDVqC0HHJ56aLQZHJYS2WTR@mail.gmail.com>	<AANLkTinDSQkOJrAikHrTEm6YTfyMcv2yLfxGPucjDwVq@mail.gmail.com>	<AANLkTi=0=+05cxkP8ER9WKmzgAS4HzymLrTHcrMVho6P@mail.gmail.com>	<AANLkTi=bNWH97qZdDmuEvzm6ETaZDRVv1R3Rt7hLTg+L@mail.gmail.com>
	<AANLkTinJBQ_EskC4Fv72FauNgbr4r4fF_gnOjiD6NDSe@mail.gmail.com>
Message-ID: <4D498952.9050808@cupis.co.uk>

On 02/02/11 16:26, deniro wrote:
> All I am loooking for is, if there is any way, to install new modules
> without re-installig from source and recompile from the scratch with
> existing freeswitch install.

Yes, you can do this.

Go into the source directory and there is a modules.conf which specifies
which modules will be compiled/installed. Edit this to uncomment the
modules you want. lua is enabled by default.

~/src/freeswitch$ grep -E lua\|perl modules.conf
languages/mod_lua
#languages/mod_perl
~/src/freeswitch$

Then you can run the following to compile install a single module and
add to your existing install:

make mod_lua-install
make mod_perl-install

Regards,



From krice at freeswitch.org  Wed Feb  2 19:43:00 2011
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 02 Feb 2011 10:43:00 -0600
Subject: [Freeswitch-users] open g729
In-Reply-To: <AANLkTikr2aCnW_CvHFv-Dv_wytyuC+ipUU5mHmrv=0uJ@mail.gmail.com>
Message-ID: <C96EE5B4.480D%krice@freeswitch.org>

That?s exactly the whole point... And then even ilBC isnt that widely
deployed but you can get more hardware that speaks iLBC then you can
speex... Face it... In reality you?re more widely find hardware that speaks
AMR then speex or iLBC, because that what the cellular networks went with
(note: I?m not saying that these are better then speex in performance I am
strictly talking ease of interop with outside equipment and VoIP vendors)

Everyone keeps forgetting that what is best doesn?t always win.. See BetaMax
vs VHS...

K


On 2/2/11 3:52 AM, "Steven Ayre" <steveayre at gmail.com> wrote:

> But you still need someone to be able to talk to. Ken's point is it works
> great when you're only calling your other Speex phones, but as soon as you
> want to talk to someone else you'll find they don't have Speex. You then have
> to use another codec... a common standard one that's been around for so long
> everyone supports it... like G729.
> 
> On 2 February 2011 03:00, curriegrad2004 <curriegrad2004 at gmail.com> wrote:
>> If hardware doesn't support it, make one that supports Speex right out
>> of the box. I'm pretty sure that somebody will do it.
>> 
>> On Tue, Feb 1, 2011 at 10:55 AM, Ken Rice <krice at freeswitch.org> wrote:
>>> > Why people insist on speex I am not sure... Sure its nice on a closed
>>> > network but quickly looses its luster when communicating with the real
>>> > world... Not to mention the CPU usages and such...
>>> >
>>> > If you want to use something like speex check out iLBC and theres things
>>> > like the Sangoma D100/D500 that you can use to offload the transcoding
>>> from
>>> > the CPU (disclaimer I sell the sangoma hardware)
>>> >
>>> > K
>>> >
>>> >
>>> > On 2/1/11 12:45 PM, "Kristian Kielhofner" <kris at kriskinc.com> wrote:
>>> >
>>>> >> Phones, carriers, and just about any commercial gear (reference
>>>> >> hardware, SBCs, DSPs, etc) you'll ever see doesn't support Speex.
>>>> >> Guess how you'll make it work? ?Transcoding from G729 to Speex. ?Now
>>>> >> you're using licenses, proxying RTP, and wasting CPU while ruining
>>>> >> voice quality transcoding from one lossy codec to another. ?It's just
>>>> >> about the worst possible situation you could be in.
>>>> >>
>>>> >> On Tue, Feb 1, 2011 at 1:19 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>>>> >>> Except that many hardware phones don't have speex, so the only
>>>>> low-bandwidth
>>>>> >>> solution is g729...
>>>>> >>> On Tue, Feb 1, 2011 at 7:43 PM, curriegrad2004
>>>>> <curriegrad2004 at gmail.com>
>>>>> >>> wrote:
>>>>>> >>>>
>>>>>> >>>> Simple solution to this g.729 patent non-sense: Speex.
>>>>>> >>>>
>>>>>> >>>> 'nuff said.
>>>>>> >>>>
>>>>>> >>>> On Tue, Feb 1, 2011 at 8:26 AM, Anton VG <anton.vazir at gmail.com>
>>>>>> wrote:
>>>>>>> >>>>> Considering the above, noone can say "Purchase a LICENSE", but
>>>>>>> only a
>>>>>>> >>>>> "Licensed SOFTWARE"
>>>>>>> >>>>>
>>>>>>> >>>>> _______________________________________________
>>>>>>> >>>>> FreeSWITCH-users mailing list
>>>>>>> >>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> >>>>> http://www.freeswitch.org
>>>>>>> >>>>>
>>>>>> >>>>
>>>>>> >>>> _______________________________________________
>>>>>> >>>> FreeSWITCH-users mailing list
>>>>>> >>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> >>>> http://www.freeswitch.org
>>>>> >>>
>>>>> >>>
>>>>> >>> _______________________________________________
>>>>> >>> FreeSWITCH-users mailing list
>>>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >>> 
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >>> http://www.freeswitch.org
>>>>> >>>
>>>>> >>>
>>>> >>
>>>> >>
>>> >
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  2 19:46:33 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 2 Feb 2011 16:46:33 +0000
Subject: [Freeswitch-users] mod_perl mod_lua
In-Reply-To: <AANLkTinJBQ_EskC4Fv72FauNgbr4r4fF_gnOjiD6NDSe@mail.gmail.com>
References: <AANLkTikxbN47Nc7cG8y0HMDVqC0HHJ56aLQZHJYS2WTR@mail.gmail.com>
	<AANLkTinDSQkOJrAikHrTEm6YTfyMcv2yLfxGPucjDwVq@mail.gmail.com>
	<AANLkTi=0=+05cxkP8ER9WKmzgAS4HzymLrTHcrMVho6P@mail.gmail.com>
	<AANLkTi=bNWH97qZdDmuEvzm6ETaZDRVv1R3Rt7hLTg+L@mail.gmail.com>
	<AANLkTinJBQ_EskC4Fv72FauNgbr4r4fF_gnOjiD6NDSe@mail.gmail.com>
Message-ID: <AANLkTik0QYcDP-YZ6RSxdipXxYJQ7LH32G45P6zJm8+E@mail.gmail.com>

Check first whether there are freeswitch-lua or freeswitch-perl packages.

The official debian/ubuntu build system does build both mod_lua and
mod_perl, but it places them in separate packages from the rest of
FreeSWITCH.

If you don't you'll need to compile them yourself...

You will need the FreeSWITCH source to compile the modules, since they are
in that source. Some of FreeSWITCH (the core) must also be compiled since
the modules use functions that are in the core - however you can restrict
the modules that are compiled to just the ones you need using modules.conf.

When you checkout/extract the FreeSWITCH source, the first thing you must do
is run bootstrap.sh. It is this file that generates the modules.conf file,
if you haven't created it yourself.

Unless you plan to upgrade (and you should think about doing so - 1.0.6 is
old now and git head has hundreds  of bugfixes and new features) you should
make sure the one you build the modules on is the exact same version the
Ubuntu version was created from, otherwise you'll find that the interface
between the core and the modules may have changed and you'll either get a
unloadable or unstable module. If you installed via APT then you should be
able to use 'apt-get source freeswitch' to get the source package that'll
give you the version they used.

-Steve


On 2 February 2011 16:26, deniro <ayhkor at gmail.com> wrote:

> thanks for your advice and  I appreciate you taking time to reply.
> You also note that I first check freeswitch site and I put my questions if
> I cant find a solutions myself.
> I already checked for options with modules.conf and modules.conf.xml even
> before posting.
>
> If you look at my first posting, I stated that I have a compiled freeswitch
> and I dont even see modules.conf file (I searched).
> so to reiterate, The freeswitch comes installed and compiled already with a
> product, so I dont even have the freeswitch source.
>
> All I am loooking for is, if there is any way, to install new modules
> without re-installig from source and recompile from the scratch with
> existing freeswitch install.
> I highly doubt that this is possible but I am checking out with the gurus
> here.
> I dont wanna break already running freeswitch with custom dialplans and
> other custom configurations.
>
> thx again
> deniro--
>
>
>
>
>
>
>
> On Wed, Feb 2, 2011 at 10:32 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:
>
>> On Wed, Feb 2, 2011 at 4:22 PM, deniro <ayhkor at gmail.com> wrote:
>> > I think that is to enable loading the module(not the install)
>> > thx
>>
>> So, you write to the mailing list for advice, and you don't put
>> confidence in answers you got back? ;)
>>
>> >
>> >
>> > On Wed, Feb 2, 2011 at 3:14 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>> > wrote:
>> >>
>> >> On Wed, Feb 2, 2011 at 5:25 AM, deniro <ayhkor at gmail.com> wrote:
>> >> > Hi
>> >> > I have compiled version of freeswitch 1.6.x on ubuntu 10.04
>> >> > I want to install extra freeswitch modules like mod_perl and mod_lua
>> >> > Is there any way to install these modules without re-compileling
>> >> > freeswitch.
>> >> > I looked at freeswitch site and it is talking about  editing
>> >> > modules.conf
>> >> > file and compiling from source
>> >> > and I dont even have modules.conf file as I searched.
>> >>
>> >> Go to your original sources directory (where you gave the command
>> >> "make install"), edit the file modules.conf.xml and give the command
>> >> "make install".
>> >>
>> >> -giovanni
>> >>
>> >>
>> >>
>> >> >
>> >> > thx
>> >> > 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
>> >> >
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> 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
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at 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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110202/bee15efc/attachment-0001.html 

From helmut.kuper at ewetel.de  Wed Feb  2 19:57:09 2011
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 02 Feb 2011 17:57:09 +0100
Subject: [Freeswitch-users] Need to change "identity display" in dialog-info
	event
Message-ID: <4D498CE5.60607@ewetel.de>

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Hash: SHA1

Hello

is there a way to get the value of "effective_caller_id_name" in
dialog-info's "remote identity display"?

And further: Is it possible to get that value also accessible for
display updates, so that after picking up a call the display is updated
with the picked up call's effective_caller_id_name?

regards
Helmut
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From anthony.minessale at gmail.com  Wed Feb  2 20:21:02 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 2 Feb 2011 11:21:02 -0600
Subject: [Freeswitch-users] execute_on_ring executing on answer
In-Reply-To: <1296664385.29302.33.camel@luna.tc.commsmundi.com>
References: <1296215265.8986.147.camel@luna.tc.commsmundi.com>
	<21B4FB4AAC2E428E938F99986CA09165@e1705>
	<1296235095.8986.167.camel@luna.tc.commsmundi.com>
	<0E96944DB29F4CEB9ACA5994D19AB0D6@e1705>
	<AANLkTikHB-QnDyT0WPsEML8yv4AACnYnHrE37gpRsLS2@mail.gmail.com>
	<1296306637.8986.185.camel@luna.tc.commsmundi.com>
	<1296582543.5245.87.camel@luna.tc.commsmundi.com>
	<AANLkTim1Hq0vrOLGpQt4DcNKUY970HzFF1KHF8ewmwxz@mail.gmail.com>
	<AANLkTimC-j3aqgnWbDP7JVf2F2BtGoB5cLnvf-qq=XqC@mail.gmail.com>
	<1296664385.29302.33.camel@luna.tc.commsmundi.com>
Message-ID: <AANLkTiniqMYJ_mp9viBh0cK8yP3uH5o0-r4tgvOPZfa_@mail.gmail.com>

app::arg is passed as-is into the function where it is automatically
parsed and queued async

so to equate to unix

"app arg" is like running it in the foreground
"app::arg" is like running in the background.

Either way the args will be parsed right.


On Wed, Feb 2, 2011 at 10:33 AM, Fran?ois Delawarde
<fdelawarde at wirelessmundi.com> wrote:
> Thanks. It does not work, but just a bit of tweaking is necessary.
>
> For what I could read, when using the new "::", app and arg are not
> parsed from the variable before calling the
> switch_core_session_execute_application_async.
>
> Fran?ois.
>
>
>
> On Tue, 2011-02-01 at 12:08 -0800, Michael Collins wrote:
>> I believe I have a new item to document on the wiki. :)
>> Francois, let us know if the latest git works for you.
>> Thanks,
>> MC
>>
>> On Tue, Feb 1, 2011 at 11:51 AM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> ? ? ? ? I think I see why.
>> ? ? ? ? The app is queued at the right time but not executed until
>> ? ? ? ? media is active.
>>
>> ? ? ? ? I have changed the code so now when you supply ::<arg> at the
>> ? ? ? ? end of
>> ? ? ? ? the app name it will be async and when you don't it will be
>> ? ? ? ? executed
>> ? ? ? ? immediately.
>>
>>
>> ? ? ? ? On Tue, Feb 1, 2011 at 11:49 AM, Fran?ois Delawarde
>> ? ? ? ? <fdelawarde at wirelessmundi.com> wrote:
>> ? ? ? ? > Since apparently noone reproduces this, it must be a
>> ? ? ? ? configuration error
>> ? ? ? ? > from my part.
>> ? ? ? ? >
>> ? ? ? ? > Any hints of where I could start looking to resolve this
>> ? ? ? ? issue?
>> ? ? ? ? >
>> ? ? ? ? > Thanks,
>> ? ? ? ? > Fran?ois.
>> ? ? ? ? >
>> ? ? ? ? >
>> ? ? ? ? > On Sat, 2011-01-29 at 14:10 +0100, Fran?ois Delawarde wrote:
>> ? ? ? ? >> Nice to know, but in that case the destination actually
>> ? ? ? ? rings (180).
>> ? ? ? ? >>
>> ? ? ? ? >> See commented log:
>> ? ? ? ? >> http://pastebin.freeswitch.org/15168
>> ? ? ? ? >>
>> ? ? ? ? >> Fran?ois.
>> ? ? ? ? >>
>> ? ? ? ? >> On Fri, 2011-01-28 at 12:34 -0600, Anthony Minessale wrote:
>> ? ? ? ? >> > if it never rings, answer will still trigger it.
>> ? ? ? ? >> >
>> ? ? ? ? >> >
>> ? ? ? ? >> > On Fri, Jan 28, 2011 at 11:24 AM, Madovsky
>> ? ? ? ? <infos at madovsky.org> wrote:
>> ? ? ? ? >> > > ah ok, maybe a wiki update would be useful.
>> ? ? ? ? >> > >
>> ? ? ? ? >> > >
>> ? ? ? ? >> > >
>> ? ? ? ? >> > > ----- Original Message -----
>> ? ? ? ? >> > > From: "Fran?ois Delawarde"
>> ? ? ? ? <fdelawarde at wirelessmundi.com>
>> ? ? ? ? >> > > To: "FreeSWITCH Users Help"
>> ? ? ? ? <freeswitch-users at lists.freeswitch.org>
>> ? ? ? ? >> > > Sent: Friday, January 28, 2011 12:18 PM
>> ? ? ? ? >> > > Subject: Re: [Freeswitch-users] execute_on_ring
>> ? ? ? ? executing on answer
>> ? ? ? ? >> > >
>> ? ? ? ? >> > >
>> ? ? ? ? >> > >> It's some cool feature made by Anthony that allows me
>> ? ? ? ? to specify the
>> ? ? ? ? >> > >> separator.
>> ? ? ? ? >> > >>
>> ? ? ? ? >> > >> in ^^:PCMA:G722
>> ? ? ? ? >> > >> ^^: means the separator is now : instead of ,
>> ? ? ? ? >> > >>
>> ? ? ? ? >> > >> Useful in the [] or {} case because the coma is
>> ? ? ? ? already used to separate
>> ? ? ? ? >> > >> variables.
>> ? ? ? ? >> > >>
>> ? ? ? ? >> > >> Fran?ois.
>> ? ? ? ? >> > >>
>> ? ? ? ? >> > >> On Fri, 2011-01-28 at 12:12 -0500, Madovsky wrote:
>> ? ? ? ? >> > >>> what means the ^^ in your codec string ?
>> ? ? ? ? >> > >>>
>> ? ? ? ? >> > >>> ----- Original Message -----
>> ? ? ? ? >> > >>> From: "Fran?ois Delawarde"
>> ? ? ? ? <fdelawarde at wirelessmundi.com>
>> ? ? ? ? >> > >>> To: "FreeSWITCH Users Help"
>> ? ? ? ? <freeswitch-users at lists.freeswitch.org>
>> ? ? ? ? >> > >>> Sent: Friday, January 28, 2011 6:47 AM
>> ? ? ? ? >> > >>> Subject: [Freeswitch-users] execute_on_ring executing
>> ? ? ? ? on answer
>> ? ? ? ? >> > >>>
>> ? ? ? ? >> > >>>
>> ? ? ? ? >> > >>> > Hi,
>> ? ? ? ? >> > >>> >
>> ? ? ? ? >> > >>> > Doing some testing with this morning's git (Fri Jan
>> ? ? ? ? 28) I just found
>> ? ? ? ? >> > >>> > out
>> ? ? ? ? >> > >>> > that the execute_on_ring application runs when the
>> ? ? ? ? destination answers
>> ? ? ? ? >> > >>> > instead of when it rings.
>> ? ? ? ? >> > >>> >
>> ? ? ? ? >> > >>> > So far, I can't seem to find out the reason. Could
>> ? ? ? ? it be some
>> ? ? ? ? >> > >>> > configuration issue?
>> ? ? ? ? >> > >>> >
>> ? ? ? ? >> > >>> >
>> ? ? ? ? >> > >>> > Here a call log showing the phenomenon with a
>> ? ? ? ? simple bridge:
>> ? ? ? ? >> > >>> >
>> ? ? ? ? >> > >>> > <action application="bridge"
>> ? ? ? ? >> > >>> >
>> ? ? ? ? data="[execute_on_ring=info,absolute_codec_string=^^:PCMA:G722]sofia/192.168.10.1/sip:2103 at 192.168.10.22:5060"/>
>> ? ? ? ? >> > >>> >
>> ? ? ? ? >> > >>> > http://pastebin.freeswitch.org/15168
>> ? ? ? ? >> > >>> >
>> ? ? ? ? >> > >>> >
>> ? ? ? ? >> > >>> > Thanks,
>> ? ? ? ? >> > >>> > Fran?ois.
>> ? ? ? ? >> > >>> >
>> ? ? ? ? >> > >>> >
>> ? ? ? ? >> > >>> > _______________________________________________
>> ? ? ? ? >> > >>> > FreeSWITCH-users mailing list
>> ? ? ? ? >> > >>> > FreeSWITCH-users at lists.freeswitch.org
>> ? ? ? ? >> > >>> >
>> ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> ? ? ? ? >> > >>> >
>> ? ? ? ? UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> ? ? ? ? >> > >>> > http://www.freeswitch.org
>> ? ? ? ? >> > >>> >
>> ? ? ? ? >> > >>>
>> ? ? ? ? >> > >>>
>> ? ? ? ? >> > >>> _______________________________________________
>> ? ? ? ? >> > >>> FreeSWITCH-users mailing list
>> ? ? ? ? >> > >>> FreeSWITCH-users at lists.freeswitch.org
>> ? ? ? ? >> > >>>
>> ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> ? ? ? ? >> > >>>
>> ? ? ? ? UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> ? ? ? ? >> > >>> http://www.freeswitch.org
>> ? ? ? ? >> > >>
>> ? ? ? ? >> > >>
>> ? ? ? ? >> > >> _______________________________________________
>> ? ? ? ? >> > >> FreeSWITCH-users mailing list
>> ? ? ? ? >> > >> FreeSWITCH-users at lists.freeswitch.org
>> ? ? ? ? >> > >>
>> ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> ? ? ? ? >> > >>
>> ? ? ? ? UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> ? ? ? ? >> > >> http://www.freeswitch.org
>> ? ? ? ? >> > >>
>> ? ? ? ? >> > >
>> ? ? ? ? >> > >
>> ? ? ? ? >> > > _______________________________________________
>> ? ? ? ? >> > > FreeSWITCH-users mailing list
>> ? ? ? ? >> > > FreeSWITCH-users at lists.freeswitch.org
>> ? ? ? ? >> > >
>> ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> ? ? ? ? >> > >
>> ? ? ? ? UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> ? ? ? ? >> > > http://www.freeswitch.org
>> ? ? ? ? >> > >
>> ? ? ? ? >> >
>> ? ? ? ? >> >
>> ? ? ? ? >> >
>> ? ? ? ? >>
>> ? ? ? ? >>
>> ? ? ? ? >> _______________________________________________
>> ? ? ? ? >> FreeSWITCH-users mailing list
>> ? ? ? ? >> FreeSWITCH-users at lists.freeswitch.org
>> ? ? ? ? >>
>> ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> ? ? ? ? >>
>> ? ? ? ? UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> ? ? ? ? >> http://www.freeswitch.org
>> ? ? ? ? >
>> ? ? ? ? >
>> ? ? ? ? > _______________________________________________
>> ? ? ? ? > FreeSWITCH-users mailing list
>> ? ? ? ? > FreeSWITCH-users at 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 eric at bmcrministries.org  Wed Feb  2 20:18:36 2011
From: eric at bmcrministries.org (Eric Michel)
Date: Wed, 2 Feb 2011 10:18:36 -0700
Subject: [Freeswitch-users] Route incoming analog calls
Message-ID: <AANLkTikueX18uU-4S9+fgHi7V-8N5PZKMXuL+f_-YegJ@mail.gmail.com>

Steve thank you for the pointers, you got me going in the right direction.

Here is what I came up with (so far it seems to work).

     <extension name="Line 2(6820)" continue="true">
<condition field="source" expression="mod_freetdm"/>
<condition field="${channel_name}" expression="^(FreeTDM/2:2/)$">
<action application="info"/>
     <action application="set" data="domain_name=$${domain}"/>
       <action application="set" data="transfer_ringback=$${us-ring}"/>
       <action application="answer"/>
       <action application="transfer" data="2000"/>
     </condition>
    </extension>

All I have to do is modify the second number in the expression. 2:3 line
three, 2:4 line four etc.

-Eric
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From fdelawarde at wirelessmundi.com  Wed Feb  2 20:48:23 2011
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Wed, 02 Feb 2011 18:48:23 +0100
Subject: [Freeswitch-users] execute_on_ring executing on answer
In-Reply-To: <AANLkTiniqMYJ_mp9viBh0cK8yP3uH5o0-r4tgvOPZfa_@mail.gmail.com>
References: <1296215265.8986.147.camel@luna.tc.commsmundi.com>
	<21B4FB4AAC2E428E938F99986CA09165@e1705>
	<1296235095.8986.167.camel@luna.tc.commsmundi.com>
	<0E96944DB29F4CEB9ACA5994D19AB0D6@e1705>
	<AANLkTikHB-QnDyT0WPsEML8yv4AACnYnHrE37gpRsLS2@mail.gmail.com>
	<1296306637.8986.185.camel@luna.tc.commsmundi.com>
	<1296582543.5245.87.camel@luna.tc.commsmundi.com>
	<AANLkTim1Hq0vrOLGpQt4DcNKUY970HzFF1KHF8ewmwxz@mail.gmail.com>
	<AANLkTimC-j3aqgnWbDP7JVf2F2BtGoB5cLnvf-qq=XqC@mail.gmail.com>
	<1296664385.29302.33.camel@luna.tc.commsmundi.com>
	<AANLkTiniqMYJ_mp9viBh0cK8yP3uH5o0-r4tgvOPZfa_@mail.gmail.com>
Message-ID: <1296668903.29302.45.camel@luna.tc.commsmundi.com>

For the record, I understood backwards and execute_on_ring now works
perfectly when executed in the foreground (no need for the "::").

It is when running async (with "::") that the application requires media
to be active and not the contrary. :-)

Thanks,
Fran?ois.


On Wed, 2011-02-02 at 11:21 -0600, Anthony Minessale wrote:
> app::arg is passed as-is into the function where it is automatically
> parsed and queued async
> 
> so to equate to unix
> 
> "app arg" is like running it in the foreground
> "app::arg" is like running in the background.
> 
> Either way the args will be parsed right.
> 
> 
> On Wed, Feb 2, 2011 at 10:33 AM, Fran?ois Delawarde
> <fdelawarde at wirelessmundi.com> wrote:
> > Thanks. It does not work, but just a bit of tweaking is necessary.
> >
> > For what I could read, when using the new "::", app and arg are not
> > parsed from the variable before calling the
> > switch_core_session_execute_application_async.
> >
> > Fran?ois.
> >
> >
> >
> > On Tue, 2011-02-01 at 12:08 -0800, Michael Collins wrote:
> >> I believe I have a new item to document on the wiki. :)
> >> Francois, let us know if the latest git works for you.
> >> Thanks,
> >> MC
> >>
> >> On Tue, Feb 1, 2011 at 11:51 AM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >>         I think I see why.
> >>         The app is queued at the right time but not executed until
> >>         media is active.
> >>
> >>         I have changed the code so now when you supply ::<arg> at the
> >>         end of
> >>         the app name it will be async and when you don't it will be
> >>         executed
> >>         immediately.
> >>
> >>
> >>         On Tue, Feb 1, 2011 at 11:49 AM, Fran?ois Delawarde
> >>         <fdelawarde at wirelessmundi.com> wrote:
> >>         > Since apparently noone reproduces this, it must be a
> >>         configuration error
> >>         > from my part.
> >>         >
> >>         > Any hints of where I could start looking to resolve this
> >>         issue?
> >>         >
> >>         > Thanks,
> >>         > Fran?ois.
> >>         >
> >>         >
> >>         > On Sat, 2011-01-29 at 14:10 +0100, Fran?ois Delawarde wrote:
> >>         >> Nice to know, but in that case the destination actually
> >>         rings (180).
> >>         >>
> >>         >> See commented log:
> >>         >> http://pastebin.freeswitch.org/15168
> >>         >>
> >>         >> Fran?ois.
> >>         >>
> >>         >> On Fri, 2011-01-28 at 12:34 -0600, Anthony Minessale wrote:
> >>         >> > if it never rings, answer will still trigger it.
> >>         >> >
> >>         >> >
> >>         >> > On Fri, Jan 28, 2011 at 11:24 AM, Madovsky
> >>         <infos at madovsky.org> wrote:
> >>         >> > > ah ok, maybe a wiki update would be useful.
> >>         >> > >
> >>         >> > >
> >>         >> > >
> >>         >> > > ----- Original Message -----
> >>         >> > > From: "Fran?ois Delawarde"
> >>         <fdelawarde at wirelessmundi.com>
> >>         >> > > To: "FreeSWITCH Users Help"
> >>         <freeswitch-users at lists.freeswitch.org>
> >>         >> > > Sent: Friday, January 28, 2011 12:18 PM
> >>         >> > > Subject: Re: [Freeswitch-users] execute_on_ring
> >>         executing on answer
> >>         >> > >
> >>         >> > >
> >>         >> > >> It's some cool feature made by Anthony that allows me
> >>         to specify the
> >>         >> > >> separator.
> >>         >> > >>
> >>         >> > >> in ^^:PCMA:G722
> >>         >> > >> ^^: means the separator is now : instead of ,
> >>         >> > >>
> >>         >> > >> Useful in the [] or {} case because the coma is
> >>         already used to separate
> >>         >> > >> variables.
> >>         >> > >>
> >>         >> > >> Fran?ois.
> >>         >> > >>
> >>         >> > >> On Fri, 2011-01-28 at 12:12 -0500, Madovsky wrote:
> >>         >> > >>> what means the ^^ in your codec string ?
> >>         >> > >>>
> >>         >> > >>> ----- Original Message -----
> >>         >> > >>> From: "Fran?ois Delawarde"
> >>         <fdelawarde at wirelessmundi.com>
> >>         >> > >>> To: "FreeSWITCH Users Help"
> >>         <freeswitch-users at lists.freeswitch.org>
> >>         >> > >>> Sent: Friday, January 28, 2011 6:47 AM
> >>         >> > >>> Subject: [Freeswitch-users] execute_on_ring executing
> >>         on answer
> >>         >> > >>>
> >>         >> > >>>
> >>         >> > >>> > Hi,
> >>         >> > >>> >
> >>         >> > >>> > Doing some testing with this morning's git (Fri Jan
> >>         28) I just found
> >>         >> > >>> > out
> >>         >> > >>> > that the execute_on_ring application runs when the
> >>         destination answers
> >>         >> > >>> > instead of when it rings.
> >>         >> > >>> >
> >>         >> > >>> > So far, I can't seem to find out the reason. Could
> >>         it be some
> >>         >> > >>> > configuration issue?
> >>         >> > >>> >
> >>         >> > >>> >
> >>         >> > >>> > Here a call log showing the phenomenon with a
> >>         simple bridge:
> >>         >> > >>> >
> >>         >> > >>> > <action application="bridge"
> >>         >> > >>> >
> >>         data="[execute_on_ring=info,absolute_codec_string=^^:PCMA:G722]sofia/192.168.10.1/sip:2103 at 192.168.10.22:5060"/>
> >>         >> > >>> >
> >>         >> > >>> > http://pastebin.freeswitch.org/15168
> >>         >> > >>> >
> >>         >> > >>> >
> >>         >> > >>> > Thanks,
> >>         >> > >>> > Fran?ois.
> >>         >> > >>> >
> >>         >> > >>> >
> >>         >> > >>> > _______________________________________________
> >>         >> > >>> > FreeSWITCH-users mailing list
> >>         >> > >>> > FreeSWITCH-users at lists.freeswitch.org
> >>         >> > >>> >
> >>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>         >> > >>> >
> >>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>         >> > >>> > http://www.freeswitch.org
> >>         >> > >>> >
> >>         >> > >>>
> >>         >> > >>>
> >>         >> > >>> _______________________________________________
> >>         >> > >>> FreeSWITCH-users mailing list
> >>         >> > >>> FreeSWITCH-users at lists.freeswitch.org
> >>         >> > >>>
> >>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>         >> > >>>
> >>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>         >> > >>> http://www.freeswitch.org
> >>         >> > >>
> >>         >> > >>
> >>         >> > >> _______________________________________________
> >>         >> > >> FreeSWITCH-users mailing list
> >>         >> > >> FreeSWITCH-users at lists.freeswitch.org
> >>         >> > >>
> >>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>         >> > >>
> >>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>         >> > >> http://www.freeswitch.org
> >>         >> > >>
> >>         >> > >
> >>         >> > >
> >>         >> > > _______________________________________________
> >>         >> > > FreeSWITCH-users mailing list
> >>         >> > > FreeSWITCH-users at lists.freeswitch.org
> >>         >> > >
> >>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>         >> > >
> >>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>         >> > > http://www.freeswitch.org
> >>         >> > >
> >>         >> >
> >>         >> >
> >>         >> >
> >>         >>
> >>         >>
> >>         >> _______________________________________________
> >>         >> FreeSWITCH-users mailing list
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> >>         >>
> >>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> >>         >> http://www.freeswitch.org
> >>         >
> >>         >
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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
> >>
> >>
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> >>
> >>
> >>
> >> _______________________________________________
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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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> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 massimiliano.ravelli at gmail.com  Wed Feb  2 20:43:45 2011
From: massimiliano.ravelli at gmail.com (Massimiliano Ravelli)
Date: Wed, 2 Feb 2011 18:43:45 +0100
Subject: [Freeswitch-users] mod_com_g729 - moving licences to a new server
Message-ID: <AANLkTikykPgaD0tErbMw=VOgfXqnwEwY4SzC_bNqSw-i@mail.gmail.com>

I need to move our current pbx to a new hardware.
How can I disable the g729  licences on the old server and enable them on
the new one ?

The current production server has an old installation of mod_com_g729: can I
upgrade it to a more recent one without upgrading freeswitch ?
Does mod_com_g729 upgrade need to stop running calls ?

Thanks in advance
Massimiliano
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From msc at freeswitch.org  Wed Feb  2 20:53:08 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 2 Feb 2011 09:53:08 -0800
Subject: [Freeswitch-users] context
In-Reply-To: <AANLkTika5CBO5eVWuRimKY+Bth2YAFC0Tm8_JmFBnXwC@mail.gmail.com>
References: <AANLkTika5CBO5eVWuRimKY+Bth2YAFC0Tm8_JmFBnXwC@mail.gmail.com>
Message-ID: <AANLkTim84VqQ79xq6amru_4GzvGP9rof001MJ8t2Zpvd@mail.gmail.com>

Turn on console debug level output (default in fs_cli) and make the test
call. Pastebin the output. Most likely the call is not being authorized
because you are letting it in via an ACL or something like that.

-MC

On Wed, Feb 2, 2011 at 1:55 AM, Sam <u2nsam at gmail.com> wrote:

> I have defined a user as :
>
> <user id="2099">
>       <params>
>         <param name="password" value="1234"/>
>         <param name="dial-string" value="{presence_id=${dialed_user}@
> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>         <param name="mwi-account" value="2099 at voicemail"/>
>         <param name="vm-password" value="1234"/>
>         <param name="vm-email-all-messages" value="true"/>
>         <!--param name="vm-attach-file" value="true"/-->
>         <!--param name="vm-mailto"  value=""/-->
>       </params>
>       <variables>
>         <variable name="user_context" value="inter"/>
>         <variable name="force_transfer_context" value="inter"/>
>         <variable name="accountcode" value="2099"/>
>         <variable name="internal_caller_id_name" value="2099"/>
>         <variable name="internal_caller_id_number" value="2099"/>
>         <variable name="callgroup" value="1"/>
>         <variable name="sip-force-contact"
> value="NDLB-connectile-dysfunction"/>
>       </variables>
>
>
> and in the dialplan i have
>
> <context name="inter">
>
> <extension name="2099">
>     <condition field="destination_number" expression="^(2099)">
>       <action application="set" data="call_timeout=30"/>
>       <action application="set" data="dialed_extension=$1"/>
>             <action application="export" data="dialed_extension=$1"/>
>             <action application="set"
> data="transfer_ringback=$${hold_music}"/>
>             <action application="set" data="hangup_after_bridge=true"/>
>             <action application="set" data="continue_on_fail=true"/>
>             <action application="hash"
> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>             <action application="hash"
> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>             <action application="hash"
> data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
>             <action application="hash"
> data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>             <action application="set"
> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
> var callgroup)}"/>
>             <action application="hash"
> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>             <action application="bridge"
> data="{sip_invite_domain=$${domain}}user/${dialed_extension}@
> ${domain_name}"/>
>     </condition>
>   </extension>
>
> </context>
>
>
> Now extension 2075 is also made on above lines in the same context for user
> and dialplan, here the call when initiated from 2075 to 2099 searches for
> context public instead of context inter
>
> mod_dialplan_xml.c:331 Processing 2075 <2075>->2099 in context public
>
> any reason why ?
>
> 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 msc at freeswitch.org  Wed Feb  2 20:54:44 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 2 Feb 2011 09:54:44 -0800
Subject: [Freeswitch-users] execute_on_answer
In-Reply-To: <37FAA1D493F546F886444D59414F7992@e1705>
References: <37FAA1D493F546F886444D59414F7992@e1705>
Message-ID: <AANLkTimtc5oaGVOhJw+G-sVi5Mo426B+aDVedb33eajH@mail.gmail.com>

No, but you could execute an extension that has a bunch of API calls.
-MC

On Wed, Feb 2, 2011 at 8:15 AM, Madovsky <infos at madovsky.org> wrote:

>  possible to use
> execute_on_answer=<api> value; <api> value
>
> ???
>
> 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 marcdecorny at gmail.com  Wed Feb  2 21:03:50 2011
From: marcdecorny at gmail.com (Marc De Corny)
Date: Wed, 2 Feb 2011 18:03:50 +0000
Subject: [Freeswitch-users] Voice quality monitoring via loopback on sip
	phone
In-Reply-To: <AANLkTikykPgaD0tErbMw=VOgfXqnwEwY4SzC_bNqSw-i@mail.gmail.com>
References: <AANLkTikykPgaD0tErbMw=VOgfXqnwEwY4SzC_bNqSw-i@mail.gmail.com>
Message-ID: <85A42F1D-3F9E-4B26-9F0B-1892EE4B9C25@gmail.com>

Hi all,
I am thinking of a way of testing the quality of a voice call. 

Ideally i would like to create a call from FS with a loopback parameter that tells phone to answer the call automatically and loopback the RTP. This is a common functionality in most phoned (for example cisco)
The sent and received packets could be compared for jitter, latency and packet loss and a result extracted.

Does this already exist? I know there some specific tool that accomplish this bit they are expensive and awkward to use. Starting and retrieving the result from an api would be fantastic.


On 2 Feb 2011, at 17:43, Massimiliano Ravelli <massimiliano.ravelli at gmail.com> wrote:

> I need to move our current pbx to a new hardware.
> How can I disable the g729  licences on the old server and enable them on the new one ?
> 
> The current production server has an old installation of mod_com_g729: can I upgrade it to a more recent one without upgrading freeswitch ?
> Does mod_com_g729 upgrade need to stop running calls ?
> 
> Thanks in advance
> Massimiliano
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 david.ponzone at ipeva.fr  Wed Feb  2 21:11:47 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 2 Feb 2011 19:11:47 +0100
Subject: [Freeswitch-users] Voice quality monitoring via loopback on sip
	phone
In-Reply-To: <85A42F1D-3F9E-4B26-9F0B-1892EE4B9C25@gmail.com>
References: <AANLkTikykPgaD0tErbMw=VOgfXqnwEwY4SzC_bNqSw-i@mail.gmail.com>
	<85A42F1D-3F9E-4B26-9F0B-1892EE4B9C25@gmail.com>
Message-ID: <F6B288C7-BE53-493A-BC8E-5DD085C11695@ipeva.fr>

Marc,

I would recommend you resend your message without replying to another one, so it has its own thread.


David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 02/02/2011 ? 19:03, Marc De Corny a ?crit :

> Hi all,
> I am thinking of a way of testing the quality of a voice call. 
> 
> Ideally i would like to create a call from FS with a loopback parameter that tells phone to answer the call automatically and loopback the RTP. This is a common functionality in most phoned (for example cisco)
> The sent and received packets could be compared for jitter, latency and packet loss and a result extracted.
> 
> Does this already exist? I know there some specific tool that accomplish this bit they are expensive and awkward to use. Starting and retrieving the result from an api would be fantastic.
> 
> 
> On 2 Feb 2011, at 17:43, Massimiliano Ravelli <massimiliano.ravelli at gmail.com> wrote:
> 
>> I need to move our current pbx to a new hardware.
>> How can I disable the g729  licences on the old server and enable them on the new one ?
>> 
>> The current production server has an old installation of mod_com_g729: can I upgrade it to a more recent one without upgrading freeswitch ?
>> Does mod_com_g729 upgrade need to stop running calls ?
>> 
>> Thanks in advance
>> Massimiliano
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Feb  2 21:21:41 2011
From: marcdecorny at gmail.com (Marc De Corny)
Date: Wed, 2 Feb 2011 18:21:41 +0000
Subject: [Freeswitch-users] Voice quality monitoring via loopback on sip
	phone
In-Reply-To: <F6B288C7-BE53-493A-BC8E-5DD085C11695@ipeva.fr>
References: <AANLkTikykPgaD0tErbMw=VOgfXqnwEwY4SzC_bNqSw-i@mail.gmail.com>
	<85A42F1D-3F9E-4B26-9F0B-1892EE4B9C25@gmail.com>
	<F6B288C7-BE53-493A-BC8E-5DD085C11695@ipeva.fr>
Message-ID: <FFED9160-E7BB-4175-9ADB-0DA499740815@gmail.com>

My apologies, i thought i had.
Let me try again
Marc

On 2 Feb 2011, at 18:11, David Ponzone <david.ponzone at ipeva.fr> wrote:

> Marc,
> 
> I would recommend you resend your message without replying to another one, so it has its own thread.
> 
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 02/02/2011 ? 19:03, Marc De Corny a ?crit :
> 
>> Hi all,
>> I am thinking of a way of testing the quality of a voice call. 
>> 
>> Ideally i would like to create a call from FS with a loopback parameter that tells phone to answer the call automatically and loopback the RTP. This is a common functionality in most phoned (for example cisco)
>> The sent and received packets could be compared for jitter, latency and packet loss and a result extracted.
>> 
>> Does this already exist? I know there some specific tool that accomplish this bit they are expensive and awkward to use. Starting and retrieving the result from an api would be fantastic.
>> 
>> 
>> On 2 Feb 2011, at 17:43, Massimiliano Ravelli <massimiliano.ravelli at gmail.com> wrote:
>> 
>>> I need to move our current pbx to a new hardware.
>>> How can I disable the g729  licences on the old server and enable them on the new one ?
>>> 
>>> The current production server has an old installation of mod_com_g729: can I upgrade it to a more recent one without upgrading freeswitch ?
>>> Does mod_com_g729 upgrade need to stop running calls ?
>>> 
>>> Thanks in advance
>>> Massimiliano
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Feb  2 21:25:04 2011
From: marcdecorny at gmail.com (Marc De Corny)
Date: Wed, 2 Feb 2011 18:25:04 +0000
Subject: [Freeswitch-users] Voice quality monitoring via loopback on sip
	endpoint
Message-ID: <66D781FD-7248-4A4A-AD83-04D2CDF1AAEF@gmail.com>

Hi all,
I am thinking of a way of testing the quality of a voice call. 

Ideally i would like to create a call from FS with a loopback parameter that tells phone to answer the call automatically and loopback the RTP. This is a common functionality in most phoned (for example cisco)
The sent and received packets could be compared for jitter, latency and packet loss and a result extracted.

Does this already exist? I know there some specific tool that accomplish this bit they are expensive and awkward to use. Starting and retrieving the result from an api would be fantastic.

Any ideas are welcome.
Thanks
Marc
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From david.ponzone at ipeva.fr  Wed Feb  2 21:32:01 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 2 Feb 2011 19:32:01 +0100
Subject: [Freeswitch-users] Voice quality monitoring via loopback on sip
	endpoint
In-Reply-To: <66D781FD-7248-4A4A-AD83-04D2CDF1AAEF@gmail.com>
References: <66D781FD-7248-4A4A-AD83-04D2CDF1AAEF@gmail.com>
Message-ID: <F9F3A5AC-3F5C-45BE-8FAF-C517A56828B2@ipeva.fr>

A common way to test audio quality is to have a device (sipp for instance) calling your FS and playing a known audio file.
FS would then record it and then, by comparing the recorded and the original one and using some signal processing, you can compute a MOSPESQ figure.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 02/02/2011 ? 19:25, Marc De Corny a ?crit :

> Hi all,
> I am thinking of a way of testing the quality of a voice call. 
> 
> Ideally i would like to create a call from FS with a loopback parameter that tells phone to answer the call automatically and loopback the RTP. This is a common functionality in most phoned (for example cisco)
> The sent and received packets could be compared for jitter, latency and packet loss and a result extracted.
> 
> Does this already exist? I know there some specific tool that accomplish this bit they are expensive and awkward to use. Starting and retrieving the result from an api would be fantastic.
> 
> Any ideas are welcome.
> 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 juanito1982 at gmail.com  Wed Feb  2 22:00:23 2011
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Wed, 2 Feb 2011 20:00:23 +0100
Subject: [Freeswitch-users] Voice quality monitoring via loopback on sip
	endpoint
In-Reply-To: <F9F3A5AC-3F5C-45BE-8FAF-C517A56828B2@ipeva.fr>
References: <66D781FD-7248-4A4A-AD83-04D2CDF1AAEF@gmail.com>
	<F9F3A5AC-3F5C-45BE-8FAF-C517A56828B2@ipeva.fr>
Message-ID: <AANLkTimCyD8Dj_5uMT=YhikU6Nbz9VscwQLZ3s+0SCuu@mail.gmail.com>

Could you give any info about which tools to use?

Regards

2011/2/2 David Ponzone <david.ponzone at ipeva.fr>

> A common way to test audio quality is to have a device (sipp for instance)
> calling your FS and playing a known audio file.
> FS would then record it and then, by comparing the recorded and the
> original one and using some signal processing, you can compute a MOSPESQ
> figure.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 02/02/2011 ? 19:25, Marc De Corny a ?crit :
>
> Hi all,
> I am thinking of a way of testing the quality of a voice call.
>
> Ideally i would like to create a call from FS with a loopback parameter
> that tells phone to answer the call automatically and loopback the RTP. This
> is a common functionality in most phoned (for example cisco)
> The sent and received packets could be compared for jitter, latency and
> packet loss and a result extracted.
>
> Does this already exist? I know there some specific tool that accomplish
> this bit they are expensive and awkward to use. Starting and retrieving the
> result from an api would be fantastic.
>
> Any ideas are welcome.
> 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 curriegrad2004 at gmail.com  Wed Feb  2 22:12:22 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Wed, 2 Feb 2011 11:12:22 -0800
Subject: [Freeswitch-users] mod_com_g729 - moving licences to a new
	server
In-Reply-To: <AANLkTikykPgaD0tErbMw=VOgfXqnwEwY4SzC_bNqSw-i@mail.gmail.com>
References: <AANLkTikykPgaD0tErbMw=VOgfXqnwEwY4SzC_bNqSw-i@mail.gmail.com>
Message-ID: <AANLkTinSROc3ETqi8vbs89swryJa1+TNKa8WWDP4_O6H@mail.gmail.com>

Is the network card associated with the server external? If so, you
can move that card and it won't be necessary to give them a call on
transferring the licences over

On Wed, Feb 2, 2011 at 9:43 AM, Massimiliano Ravelli
<massimiliano.ravelli at gmail.com> wrote:
> I need to move our current pbx to a new hardware.
> How can I disable the g729? licences on the old server and enable them on
> the new one ?
>
> The current production server has an old installation of mod_com_g729: can I
> upgrade it to a more recent one without upgrading freeswitch ?
> Does mod_com_g729 upgrade need to stop running calls ?
>
> Thanks in advance
> Massimiliano
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb  2 22:22:33 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 2 Feb 2011 19:22:33 +0000
Subject: [Freeswitch-users] mod_com_g729 - moving licences to a new
	server
In-Reply-To: <AANLkTinSROc3ETqi8vbs89swryJa1+TNKa8WWDP4_O6H@mail.gmail.com>
References: <AANLkTikykPgaD0tErbMw=VOgfXqnwEwY4SzC_bNqSw-i@mail.gmail.com>
	<AANLkTinSROc3ETqi8vbs89swryJa1+TNKa8WWDP4_O6H@mail.gmail.com>
Message-ID: <AANLkTinPmSAR=OiatFKEP0sRVA6_4=DQ7C5UdQcxsTfM@mail.gmail.com>

curriegrad2004 ,

That'll only work if there are no internal cards either.

Massimiliano,

Licenses are linked to the server via the MAC addresses of all network cards
on the server. If you can move all of those (or at least set the same MAC
addresses) it'll let you use the license on the other server. Do not try
using the license on the old server at the same time.

If that's not possible or you'd rather have it linked to the new server try
emailing consulting at freeswitch.org, they'll be able to reissue the license
(but as part of the patent restrictions can only do this a limited number of
times). They'll probably also see this thread.

You'll want to stop the current server before installing the license on the
new server I believe.

mod_com_g729 can be upgraded, but not while you have G729 calls running.
You'll need to do install the newer module, then do "reload mod_com_g729" at
the FS cli. That'll unload the G729 codec module (which'll block until any
current G729 calls end) then load the new one. Be careful on a production
server - you should test doing so on a testbed first so you don't get any
unexpected problems during the upgrade, otherwise your server may be offline
for much longer than you'd expected.

-Steve


On 2 February 2011 19:12, curriegrad2004 <curriegrad2004 at gmail.com> wrote:

> Is the network card associated with the server external? If so, you
> can move that card and it won't be necessary to give them a call on
> transferring the licences over
>
> On Wed, Feb 2, 2011 at 9:43 AM, Massimiliano Ravelli
> <massimiliano.ravelli at gmail.com> wrote:
> > I need to move our current pbx to a new hardware.
> > How can I disable the g729  licences on the old server and enable them on
> > the new one ?
> >
> > The current production server has an old installation of mod_com_g729:
> can I
> > upgrade it to a more recent one without upgrading freeswitch ?
> > Does mod_com_g729 upgrade need to stop running calls ?
> >
> > Thanks in advance
> > Massimiliano
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  2 22:32:30 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 2 Feb 2011 11:32:30 -0800
Subject: [Freeswitch-users] How to record a conference using javascript
	application
In-Reply-To: <1296664102854-5985457.post@n2.nabble.com>
References: <1296664102854-5985457.post@n2.nabble.com>
Message-ID: <AANLkTikZg=tYFKC61EoiFP7rbrQBv0NswzH41tqDS8sE@mail.gmail.com>

I believe that you need to use the conference API, not the conference
dialplan application. Try this:

apiExecute("conference", "conf-room--"+room_id+"@default record
/usr/local/freeswitch/abc.wav");

let us know if that helps...

-MC

On Wed, Feb 2, 2011 at 8:28 AM, kapil.rastogi <kapil.rastogi at telemune.net>wrote:

>
> Hi,
>
> I want to record the chatting b/w all members during the conference. I am
> using the following statement in my javascript application:
>
>  session.execute("conference", "conf-room--"+room_id+"@default record
> /usr/local/freeswitch/abc.wav");
>
> But i am unable to record the conference room chatting. But when i record
> the same from fs_cli command line, it is working well.
>
> Please tell me how to record the conference chatting.
>
> Regards,
>
> -----
> 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/How-to-record-a-conference-using-javascript-application-tp5985457p5985457.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 infos at madovsky.org  Wed Feb  2 22:54:22 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 2 Feb 2011 14:54:22 -0500
Subject: [Freeswitch-users] last git problem ?
Message-ID: <6038028FB6434CCCA1E1DC4E5784A3B7@e1705>

after every hangup now FS quits unexpeclty,without any error message....
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From infos at madovsky.org  Wed Feb  2 23:00:46 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 2 Feb 2011 15:00:46 -0500
Subject: [Freeswitch-users] last git problem ?
Message-ID: <51F625342DF84AF597429D155464E404@e1705>

maybe thies unusual log lines after hangup before FS quits

2011-02-02 14:58:35.549213 [DEBUG] switch_ivr_bridge.c:500 sofia/internal/9999999999999 at default ending bridge by request from read function
2011-02-02 14:58:35.549213 [DEBUG] switch_ivr_bridge.c:494 sofia/internal/9999999999999 at default ending bridge by request from write function

nothing else 


  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, February 02, 2011 2:54 PM
  Subject: last git problem ?


  after every hangup now FS quits unexpeclty,without any error message....
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From steveayre at gmail.com  Wed Feb  2 23:01:51 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 2 Feb 2011 20:01:51 +0000
Subject: [Freeswitch-users] last git problem ?
In-Reply-To: <6038028FB6434CCCA1E1DC4E5784A3B7@e1705>
References: <6038028FB6434CCCA1E1DC4E5784A3B7@e1705>
Message-ID: <AANLkTimRKhCwF9xHixb5psPUf_zoffYk1s=C75_NVENB@mail.gmail.com>

Does dmesg show that any segfaults have occurred?

On 2 February 2011 19:54, Madovsky <infos at madovsky.org> wrote:

>  after every hangup now FS quits unexpeclty,without any error message....
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Feb  2 23:07:31 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 2 Feb 2011 15:07:31 -0500
Subject: [Freeswitch-users] last git problem ?
References: <6038028FB6434CCCA1E1DC4E5784A3B7@e1705>
	<AANLkTimRKhCwF9xHixb5psPUf_zoffYk1s=C75_NVENB@mail.gmail.com>
Message-ID: <1E685A9AA4E24122AF13CE8E8942D844@e1705>

yes

freeswitch[30854]: segfault at 11 ip 00007f4bd1a09285 sp 00007f4bd0bfb9a0 error 4 in libfreeswitch.so.1.0.0[7f4bd19b3000+1c6000]

  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, February 02, 2011 3:01 PM
  Subject: Re: [Freeswitch-users] last git problem ?


  Does dmesg show that any segfaults have occurred?


  On 2 February 2011 19:54, Madovsky <infos at madovsky.org> wrote:

    after every hangup now FS quits unexpeclty,without any error message....

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://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 sos at sokhapkin.dyndns.org  Wed Feb  2 23:14:43 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 2 Feb 2011 15:14:43 -0500
Subject: [Freeswitch-users] last git problem ?
In-Reply-To: <1E685A9AA4E24122AF13CE8E8942D844@e1705>
References: <6038028FB6434CCCA1E1DC4E5784A3B7@e1705>
	<AANLkTimRKhCwF9xHixb5psPUf_zoffYk1s=C75_NVENB@mail.gmail.com>
	<1E685A9AA4E24122AF13CE8E8942D844@e1705>
Message-ID: <201102021514.43150.sos@sokhapkin.dyndns.org>

A friend of mine has similar segfaults with yesterday snapshot too. I suspect 
the cause is in yesterday's execute_on_* changes, but I have no proof.

On Wednesday 02 February 2011, Madovsky wrote:
> yes
> 
> freeswitch[30854]: segfault at 11 ip 00007f4bd1a09285 sp 00007f4bd0bfb9a0
> error 4 in libfreeswitch.so.1.0.0[7f4bd19b3000+1c6000]
> 
>   ----- Original Message -----
>   From: Steven Ayre
>   To: FreeSWITCH Users Help
>   Sent: Wednesday, February 02, 2011 3:01 PM
>   Subject: Re: [Freeswitch-users] last git problem ?
> 
> 
>   Does dmesg show that any segfaults have occurred?
> 
> 
>   On 2 February 2011 19:54, Madovsky <infos at madovsky.org> wrote:
> 
>     after every hangup now FS quits unexpeclty,without any error
> message....
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>    
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> 
> 
> ---------------------------------------------------------------------------
> ---
> 
> 
>   _______________________________________________
>   FreeSWITCH-users mailing list
>   FreeSWITCH-users at 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 Feb  2 23:16:51 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 2 Feb 2011 15:16:51 -0500
Subject: [Freeswitch-users] last git problem ?
References: <6038028FB6434CCCA1E1DC4E5784A3B7@e1705>
	<AANLkTimRKhCwF9xHixb5psPUf_zoffYk1s=C75_NVENB@mail.gmail.com>
Message-ID: <3AEF2A2C49A8453B8C9AC810639A7666@e1705>

it happens on my whole cluster since I updated 2 hours ago

[43671.448504] freeswitch[14963]: segfault at 11 ip 00007fce18a3f285 sp 00007fce17e9d9a0 error 4 in libfreeswitch.so.1.0.0[7fce189e9000+1c6000]
[46054.077174] freeswitch[17916]: segfault at 11 ip 00007fdc09e28285 sp 00007fdc092869a0 error 4 in libfreeswitch.so.1.0.0[7fdc09dd2000+1c6000]
[47426.166723] freeswitch[11108]: segfault at 11 ip 00007f86c7a67285 sp 00007f86c6ec59a0 error 4 in libfreeswitch.so.1.0.0[7f86c7a11000+1c6000]
[57346.162886] freeswitch[24798]: segfault at 11 ip 00007ff8c4958285 sp 00007ff8c3db69a0 error 4 in libfreeswitch.so.1.0.0 (deleted)[7ff8c4902000+1c6000]

  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, February 02, 2011 3:01 PM
  Subject: Re: [Freeswitch-users] last git problem ?


  Does dmesg show that any segfaults have occurred?


  On 2 February 2011 19:54, Madovsky <infos at madovsky.org> wrote:

    after every hangup now FS quits unexpeclty,without any error message....

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://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 steveayre at gmail.com  Wed Feb  2 23:23:14 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 2 Feb 2011 20:23:14 +0000
Subject: [Freeswitch-users] last git problem ?
In-Reply-To: <3AEF2A2C49A8453B8C9AC810639A7666@e1705>
References: <6038028FB6434CCCA1E1DC4E5784A3B7@e1705>
	<AANLkTimRKhCwF9xHixb5psPUf_zoffYk1s=C75_NVENB@mail.gmail.com>
	<3AEF2A2C49A8453B8C9AC810639A7666@e1705>
Message-ID: <AANLkTimDCMZ5aRp2yfcV=paGUvU6nnPF2-6mCC_A=7wG@mail.gmail.com>

I'd suggest you run freeswitch with the -core option which will create a
coredump. You can then use that to report the bug at
http://jira.freeswitch.org/

http://wiki.freeswitch.org/wiki/Reporting_Bugs
http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Simple_bash_script_to_make_debug_easy

You can checkout an earlier git version to compile and roll out a working
earlier version if you need to get your cluster back online. It's a good
idea when upgrading to test on a single server before rolling it out to the
entire cluster in case there's this sort of problem.

$ git clone git://git.freeswitch.org/freeswitch.git
$ cd freeswitch
$ git checkout GIT_COMMIT_ID
$ ./bootstrap.sh
$ ./configure OPTIONS
$ make
$ make install

-Steve



On 2 February 2011 20:16, Madovsky <infos at madovsky.org> wrote:

>  it happens on my whole cluster since I updated 2 hours ago
>
> [43671.448504] freeswitch[14963]: segfault at 11 ip 00007fce18a3f285 sp
> 00007fce17e9d9a0 error 4 in libfreeswitch.so.1.0.0[7fce189e9000+1c6000]
> [46054.077174] freeswitch[17916]: segfault at 11 ip 00007fdc09e28285 sp
> 00007fdc092869a0 error 4 in libfreeswitch.so.1.0.0[7fdc09dd2000+1c6000]
> [47426.166723] freeswitch[11108]: segfault at 11 ip 00007f86c7a67285 sp
> 00007f86c6ec59a0 error 4 in libfreeswitch.so.1.0.0[7f86c7a11000+1c6000]
> [57346.162886] freeswitch[24798]: segfault at 11 ip 00007ff8c4958285 sp
> 00007ff8c3db69a0 error 4 in libfreeswitch.so.1.0.0
> (deleted)[7ff8c4902000+1c6000]
>
> ----- Original Message -----
> *From:* Steven Ayre <steveayre at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, February 02, 2011 3:01 PM
> *Subject:* Re: [Freeswitch-users] last git problem ?
>
> Does dmesg show that any segfaults have occurred?
>
> On 2 February 2011 19:54, Madovsky <infos at madovsky.org> wrote:
>
>>  after every hangup now FS quits unexpeclty,without any error message....
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.ponzone at ipeva.fr  Wed Feb  2 23:25:55 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 2 Feb 2011 21:25:55 +0100
Subject: [Freeswitch-users] Voice quality monitoring via loopback on sip
	endpoint
In-Reply-To: <AANLkTimCyD8Dj_5uMT=YhikU6Nbz9VscwQLZ3s+0SCuu@mail.gmail.com>
References: <66D781FD-7248-4A4A-AD83-04D2CDF1AAEF@gmail.com>
	<F9F3A5AC-3F5C-45BE-8FAF-C517A56828B2@ipeva.fr>
	<AANLkTimCyD8Dj_5uMT=YhikU6Nbz9VscwQLZ3s+0SCuu@mail.gmail.com>
Message-ID: <7FA52E5C-9692-41BA-8BCF-1E82D11D9A01@ipeva.fr>

Well, not really :)
The signal processing involved is not obvious, mainly because what you want is to compare 2 signals, keeping in mind that the human ear is the target (MOSPESQ is normally computed using a group of real people, each of them scoring the quality of the audio).
I don't know any software doing that for free. I am pretty sure in the commercial world, there should be something available.

Check http://www.phoenixdatacom.com/voip.html

Another solution is to use an external quality insurance service to do that for you. We have one like that in France. You give them an entry point in your network where to inject the audio, an exit point where to record the audio they injected, and they give you your MOSPESQ score.
The main reason to go that way, I think, is that the score they give you may be used as a reference against competitors.

And finally, you may probably build your own benchmark reading this:

http://www.freeswitch.org/node/297

I guess Kristian Kielhofner could tell you more about it.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 02/02/2011 ? 20:00, Juan Antonio Iba?ez Santorum a ?crit :

> Could you give any info about which tools to use?
> 
> Regards
> 
> 2011/2/2 David Ponzone <david.ponzone at ipeva.fr>
> A common way to test audio quality is to have a device (sipp for instance) calling your FS and playing a known audio file.
> FS would then record it and then, by comparing the recorded and the original one and using some signal processing, you can compute a MOSPESQ figure.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 02/02/2011 ? 19:25, Marc De Corny a ?crit :
> 
>> Hi all,
>> I am thinking of a way of testing the quality of a voice call. 
>> 
>> Ideally i would like to create a call from FS with a loopback parameter that tells phone to answer the call automatically and loopback the RTP. This is a common functionality in most phoned (for example cisco)
>> The sent and received packets could be compared for jitter, latency and packet loss and a result extracted.
>> 
>> Does this already exist? I know there some specific tool that accomplish this bit they are expensive and awkward to use. Starting and retrieving the result from an api would be fantastic.
>> 
>> Any ideas are welcome.
>> 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
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Feb  2 23:46:13 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 2 Feb 2011 15:46:13 -0500
Subject: [Freeswitch-users] last git problem ?
References: <6038028FB6434CCCA1E1DC4E5784A3B7@e1705><AANLkTimRKhCwF9xHixb5psPUf_zoffYk1s=C75_NVENB@mail.gmail.com><3AEF2A2C49A8453B8C9AC810639A7666@e1705>
	<AANLkTimDCMZ5aRp2yfcV=paGUvU6nnPF2-6mCC_A=7wG@mail.gmail.com>
Message-ID: <E640C3E5DC4C410EA5980C6DA5A8D4E9@e1705>

ok but where to find prevoius git ID ?

  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, February 02, 2011 3:23 PM
  Subject: Re: [Freeswitch-users] last git problem ?


  I'd suggest you run freeswitch with the -core option which will create a coredump. You can then use that to report the bug at http://jira.freeswitch.org/


  http://wiki.freeswitch.org/wiki/Reporting_Bugs
  http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Simple_bash_script_to_make_debug_easy


  You can checkout an earlier git version to compile and roll out a working earlier version if you need to get your cluster back online. It's a good idea when upgrading to test on a single server before rolling it out to the entire cluster in case there's this sort of problem.


  $ git clone git://git.freeswitch.org/freeswitch.git
  $ cd freeswitch
  $ git checkout GIT_COMMIT_ID
  $ ./bootstrap.sh
  $ ./configure OPTIONS
  $ make
  $ make install


  -Steve





  On 2 February 2011 20:16, Madovsky <infos at madovsky.org> wrote:

    it happens on my whole cluster since I updated 2 hours ago

    [43671.448504] freeswitch[14963]: segfault at 11 ip 00007fce18a3f285 sp 00007fce17e9d9a0 error 4 in libfreeswitch.so.1.0.0[7fce189e9000+1c6000]
    [46054.077174] freeswitch[17916]: segfault at 11 ip 00007fdc09e28285 sp 00007fdc092869a0 error 4 in libfreeswitch.so.1.0.0[7fdc09dd2000+1c6000]
    [47426.166723] freeswitch[11108]: segfault at 11 ip 00007f86c7a67285 sp 00007f86c6ec59a0 error 4 in libfreeswitch.so.1.0.0[7f86c7a11000+1c6000]
    [57346.162886] freeswitch[24798]: segfault at 11 ip 00007ff8c4958285 sp 00007ff8c3db69a0 error 4 in libfreeswitch.so.1.0.0 (deleted)[7ff8c4902000+1c6000]

      ----- Original Message ----- 
      From: Steven Ayre 
      To: FreeSWITCH Users Help 
      Sent: Wednesday, February 02, 2011 3:01 PM
      Subject: Re: [Freeswitch-users] last git problem ?


      Does dmesg show that any segfaults have occurred?


      On 2 February 2011 19:54, Madovsky <infos at madovsky.org> wrote:

        after every hangup now FS quits unexpeclty,without any error message....

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






--------------------------------------------------------------------------


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



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






------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.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 Feb  2 23:54:07 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 2 Feb 2011 12:54:07 -0800 (PST)
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
	for Windows in .NET
In-Reply-To: <8132.1296664446@ccs.covici.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
	<16100.1296638658@ccs.covici.com>
	<1296658879889-5985076.post@n2.nabble.com>
	<6549.1296659705@ccs.covici.com>
	<AANLkTinCVZqYRxd0ZcQLj_kj42fYHDqUsEocEqW3oaph@mail.gmail.com>
	<7179.1296662450@ccs.covici.com>
	<1296663161529-5985392.post@n2.nabble.com>
	<8132.1296664446@ccs.covici.com>
Message-ID: <1296680047815-5986486.post@n2.nabble.com>


John,

Do you have UAC disabled?
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FSClient-Release-A-FreeSWITCH-SIP-Client-for-Windows-in-NET-tp5983787p5986486.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From tuyanozipek at gmail.com  Thu Feb  3 00:07:22 2011
From: tuyanozipek at gmail.com (=?ISO-8859-1?Q?Tuyan_=D6zipek?=)
Date: Wed, 2 Feb 2011 16:07:22 -0500
Subject: [Freeswitch-users] last git problem ?
In-Reply-To: <E640C3E5DC4C410EA5980C6DA5A8D4E9@e1705>
References: <6038028FB6434CCCA1E1DC4E5784A3B7@e1705>
	<AANLkTimRKhCwF9xHixb5psPUf_zoffYk1s=C75_NVENB@mail.gmail.com>
	<3AEF2A2C49A8453B8C9AC810639A7666@e1705>
	<AANLkTimDCMZ5aRp2yfcV=paGUvU6nnPF2-6mCC_A=7wG@mail.gmail.com>
	<E640C3E5DC4C410EA5980C6DA5A8D4E9@e1705>
Message-ID: <AANLkTikezK1-_v_X9QrRKnKCdoC0azDZvttMYWhj_8L1@mail.gmail.com>

git checkout -b somebranch `git rev-list -n 1 --before="2011-02-01
00:00" master`

will checkout the first revision which is before 2011-02-01 00:00 from
the master branch
into another branch with the name "somebranch" and will switch to it.


/tyn

On Wed, Feb 2, 2011 at 3:46 PM, Madovsky <infos at madovsky.org> wrote:
> ok but where to find prevoius git ID ?
>
>
> ----- Original Message -----
> From: Steven Ayre
> To: FreeSWITCH Users Help
> Sent: Wednesday, February 02, 2011 3:23 PM
> Subject: Re: [Freeswitch-users] last git problem ?
> I'd suggest you run freeswitch with the -core option which will create a
> coredump. You can then use that to report the bug at
> http://jira.freeswitch.org/
> http://wiki.freeswitch.org/wiki/Reporting_Bugs
> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Simple_bash_script_to_make_debug_easy
> You can checkout an earlier git version to compile and roll out a working
> earlier version if you need to get your cluster back online. It's a good
> idea when upgrading to test on a single server before rolling it out to the
> entire cluster in case there's this sort of problem.
> $ git clone git://git.freeswitch.org/freeswitch.git
> $ cd freeswitch
> $ git checkout GIT_COMMIT_ID
> $ ./bootstrap.sh
> $ ./configure OPTIONS
> $ make
> $ make install
> -Steve
>
>
> On 2 February 2011 20:16, Madovsky <infos at madovsky.org> wrote:
>>
>> it happens on my whole cluster since I updated 2 hours ago
>>
>> [43671.448504] freeswitch[14963]: segfault at 11 ip 00007fce18a3f285 sp
>> 00007fce17e9d9a0 error 4 in libfreeswitch.so.1.0.0[7fce189e9000+1c6000]
>> [46054.077174] freeswitch[17916]: segfault at 11 ip 00007fdc09e28285 sp
>> 00007fdc092869a0 error 4 in libfreeswitch.so.1.0.0[7fdc09dd2000+1c6000]
>> [47426.166723] freeswitch[11108]: segfault at 11 ip 00007f86c7a67285 sp
>> 00007f86c6ec59a0 error 4 in libfreeswitch.so.1.0.0[7f86c7a11000+1c6000]
>> [57346.162886] freeswitch[24798]: segfault at 11 ip 00007ff8c4958285 sp
>> 00007ff8c3db69a0 error 4 in libfreeswitch.so.1.0.0
>> (deleted)[7ff8c4902000+1c6000]
>>
>> ----- Original Message -----
>> From: Steven Ayre
>> To: FreeSWITCH Users Help
>> Sent: Wednesday, February 02, 2011 3:01 PM
>> Subject: Re: [Freeswitch-users] last git problem ?
>> Does dmesg show that any segfaults have occurred?
>>
>> On 2 February 2011 19:54, Madovsky <infos at madovsky.org> wrote:
>>>
>>> after every hangup now FS quits unexpeclty,without any error message....
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> ________________________________
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> ________________________________
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb  3 00:07:48 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 2 Feb 2011 21:07:48 +0000
Subject: [Freeswitch-users] last git problem ?
In-Reply-To: <E640C3E5DC4C410EA5980C6DA5A8D4E9@e1705>
References: <6038028FB6434CCCA1E1DC4E5784A3B7@e1705>
	<AANLkTimRKhCwF9xHixb5psPUf_zoffYk1s=C75_NVENB@mail.gmail.com>
	<3AEF2A2C49A8453B8C9AC810639A7666@e1705>
	<AANLkTimDCMZ5aRp2yfcV=paGUvU6nnPF2-6mCC_A=7wG@mail.gmail.com>
	<E640C3E5DC4C410EA5980C6DA5A8D4E9@e1705>
Message-ID: <AANLkTimPepfmTp7S6hfirwnwjtEk3kJn-L44GOF2e0Rk@mail.gmail.com>

$ git clone git://git.freeswitch.org/freeswitch.git
$ git log

Look through the recent entries and pick one you feel comfortable using,
then test to see if it's affected.

-Steve


On 2 February 2011 20:46, Madovsky <infos at madovsky.org> wrote:

>  ok but where to find prevoius git ID ?
>
>
> ----- Original Message -----
> *From:* Steven Ayre <steveayre at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, February 02, 2011 3:23 PM
> *Subject:* Re: [Freeswitch-users] last git problem ?
>
> I'd suggest you run freeswitch with the -core option which will create a
> coredump. You can then use that to report the bug at
> http://jira.freeswitch.org/
>
> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>
> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Simple_bash_script_to_make_debug_easy
>
> You can checkout an earlier git version to compile and roll out a working
> earlier version if you need to get your cluster back online. It's a good
> idea when upgrading to test on a single server before rolling it out to the
> entire cluster in case there's this sort of problem.
>
> $ git clone git://git.freeswitch.org/freeswitch.git
> $ cd freeswitch
> $ git checkout GIT_COMMIT_ID
> $ ./bootstrap.sh
> $ ./configure OPTIONS
> $ make
> $ make install
>
> -Steve
>
>
>
> On 2 February 2011 20:16, Madovsky <infos at madovsky.org> wrote:
>
>>  it happens on my whole cluster since I updated 2 hours ago
>>
>> [43671.448504] freeswitch[14963]: segfault at 11 ip 00007fce18a3f285 sp
>> 00007fce17e9d9a0 error 4 in libfreeswitch.so.1.0.0[7fce189e9000+1c6000]
>> [46054.077174] freeswitch[17916]: segfault at 11 ip 00007fdc09e28285 sp
>> 00007fdc092869a0 error 4 in libfreeswitch.so.1.0.0[7fdc09dd2000+1c6000]
>> [47426.166723] freeswitch[11108]: segfault at 11 ip 00007f86c7a67285 sp
>> 00007f86c6ec59a0 error 4 in libfreeswitch.so.1.0.0[7f86c7a11000+1c6000]
>> [57346.162886] freeswitch[24798]: segfault at 11 ip 00007ff8c4958285 sp
>> 00007ff8c3db69a0 error 4 in libfreeswitch.so.1.0.0
>> (deleted)[7ff8c4902000+1c6000]
>>
>>  ----- Original Message -----
>> *From:* Steven Ayre <steveayre at gmail.com>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Sent:* Wednesday, February 02, 2011 3:01 PM
>> *Subject:* Re: [Freeswitch-users] last git problem ?
>>
>>  Does dmesg show that any segfaults have occurred?
>>
>> On 2 February 2011 19:54, Madovsky <infos at madovsky.org> wrote:
>>
>>>  after every hangup now FS quits unexpeclty,without any error
>>> message....
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>  ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  3 00:08:10 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 2 Feb 2011 21:08:10 +0000
Subject: [Freeswitch-users] last git problem ?
In-Reply-To: <AANLkTimPepfmTp7S6hfirwnwjtEk3kJn-L44GOF2e0Rk@mail.gmail.com>
References: <6038028FB6434CCCA1E1DC4E5784A3B7@e1705>
	<AANLkTimRKhCwF9xHixb5psPUf_zoffYk1s=C75_NVENB@mail.gmail.com>
	<3AEF2A2C49A8453B8C9AC810639A7666@e1705>
	<AANLkTimDCMZ5aRp2yfcV=paGUvU6nnPF2-6mCC_A=7wG@mail.gmail.com>
	<E640C3E5DC4C410EA5980C6DA5A8D4E9@e1705>
	<AANLkTimPepfmTp7S6hfirwnwjtEk3kJn-L44GOF2e0Rk@mail.gmail.com>
Message-ID: <AANLkTi=VczA-PFC63CgSSEWPec5sBZ0=4UXVodX-GCWm@mail.gmail.com>

There's also http://fisheye.freeswitch.org/changelog/freeswitch.git

On 2 February 2011 21:07, Steven Ayre <steveayre at gmail.com> wrote:

> $ git clone git://git.freeswitch.org/freeswitch.git
> $ git log
>
> Look through the recent entries and pick one you feel comfortable using,
> then test to see if it's affected.
>
> -Steve
>
>
> On 2 February 2011 20:46, Madovsky <infos at madovsky.org> wrote:
>
>>  ok but where to find prevoius git ID ?
>>
>>
>> ----- Original Message -----
>> *From:* Steven Ayre <steveayre at gmail.com>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Sent:* Wednesday, February 02, 2011 3:23 PM
>> *Subject:* Re: [Freeswitch-users] last git problem ?
>>
>> I'd suggest you run freeswitch with the -core option which will create a
>> coredump. You can then use that to report the bug at
>> http://jira.freeswitch.org/
>>
>> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>
>> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Simple_bash_script_to_make_debug_easy
>>
>> You can checkout an earlier git version to compile and roll out a working
>> earlier version if you need to get your cluster back online. It's a good
>> idea when upgrading to test on a single server before rolling it out to the
>> entire cluster in case there's this sort of problem.
>>
>> $ git clone git://git.freeswitch.org/freeswitch.git
>> $ cd freeswitch
>> $ git checkout GIT_COMMIT_ID
>> $ ./bootstrap.sh
>> $ ./configure OPTIONS
>> $ make
>> $ make install
>>
>> -Steve
>>
>>
>>
>> On 2 February 2011 20:16, Madovsky <infos at madovsky.org> wrote:
>>
>>>  it happens on my whole cluster since I updated 2 hours ago
>>>
>>> [43671.448504] freeswitch[14963]: segfault at 11 ip 00007fce18a3f285 sp
>>> 00007fce17e9d9a0 error 4 in libfreeswitch.so.1.0.0[7fce189e9000+1c6000]
>>> [46054.077174] freeswitch[17916]: segfault at 11 ip 00007fdc09e28285 sp
>>> 00007fdc092869a0 error 4 in libfreeswitch.so.1.0.0[7fdc09dd2000+1c6000]
>>> [47426.166723] freeswitch[11108]: segfault at 11 ip 00007f86c7a67285 sp
>>> 00007f86c6ec59a0 error 4 in libfreeswitch.so.1.0.0[7f86c7a11000+1c6000]
>>> [57346.162886] freeswitch[24798]: segfault at 11 ip 00007ff8c4958285 sp
>>> 00007ff8c3db69a0 error 4 in libfreeswitch.so.1.0.0
>>> (deleted)[7ff8c4902000+1c6000]
>>>
>>>  ----- Original Message -----
>>> *From:* Steven Ayre <steveayre at gmail.com>
>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> *Sent:* Wednesday, February 02, 2011 3:01 PM
>>> *Subject:* Re: [Freeswitch-users] last git problem ?
>>>
>>>  Does dmesg show that any segfaults have occurred?
>>>
>>> On 2 February 2011 19:54, Madovsky <infos at madovsky.org> wrote:
>>>
>>>>  after every hangup now FS quits unexpeclty,without any error
>>>> message....
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>  ------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>  ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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 Feb  3 01:01:10 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 2 Feb 2011 16:01:10 -0600
Subject: [Freeswitch-users] last git problem ?
In-Reply-To: <AANLkTi=VczA-PFC63CgSSEWPec5sBZ0=4UXVodX-GCWm@mail.gmail.com>
References: <6038028FB6434CCCA1E1DC4E5784A3B7@e1705>
	<AANLkTimRKhCwF9xHixb5psPUf_zoffYk1s=C75_NVENB@mail.gmail.com>
	<3AEF2A2C49A8453B8C9AC810639A7666@e1705>
	<AANLkTimDCMZ5aRp2yfcV=paGUvU6nnPF2-6mCC_A=7wG@mail.gmail.com>
	<E640C3E5DC4C410EA5980C6DA5A8D4E9@e1705>
	<AANLkTimPepfmTp7S6hfirwnwjtEk3kJn-L44GOF2e0Rk@mail.gmail.com>
	<AANLkTi=VczA-PFC63CgSSEWPec5sBZ0=4UXVodX-GCWm@mail.gmail.com>
Message-ID: <AANLkTimBd1YRaCiO-2-_njyaDmohdpXVdGa7NgNXmKWH@mail.gmail.com>

Did anyone get a backtrace ?
or make sure its not been fixed since.

It sounds like FS-3023 to me. http://jira.freeswitch.org/browse/FS-3023
fixed in

commit 89c5f3bf8226bf605336b66e7761fd9f753d935a
Author: Brian West <brian at freeswitch.org>
Date:   Wed Feb 2 11:04:39 2011 -0600




On Wed, Feb 2, 2011 at 3:08 PM, Steven Ayre <steveayre at gmail.com> wrote:
> There's also?http://fisheye.freeswitch.org/changelog/freeswitch.git
>
> On 2 February 2011 21:07, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> $ git clone git://git.freeswitch.org/freeswitch.git
>> $ git log
>> Look through the recent entries and pick one you feel comfortable using,
>> then test to see if it's affected.
>> -Steve
>>
>> On 2 February 2011 20:46, Madovsky <infos at madovsky.org> wrote:
>>>
>>> ok but where to find prevoius git ID ?
>>>
>>>
>>> ----- Original Message -----
>>> From: Steven Ayre
>>> To: FreeSWITCH Users Help
>>> Sent: Wednesday, February 02, 2011 3:23 PM
>>> Subject: Re: [Freeswitch-users] last git problem ?
>>> I'd suggest you run freeswitch with the -core option which will create a
>>> coredump. You can then use that to report the bug at
>>> http://jira.freeswitch.org/
>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>>
>>> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Simple_bash_script_to_make_debug_easy
>>> You can checkout an earlier git version to compile and roll out a working
>>> earlier version if you need to get your cluster back online. It's a good
>>> idea when upgrading to test on a single server before rolling it out to the
>>> entire cluster in case there's this sort of problem.
>>> $ git clone git://git.freeswitch.org/freeswitch.git
>>> $ cd freeswitch
>>> $ git checkout GIT_COMMIT_ID
>>> $ ./bootstrap.sh
>>> $ ./configure OPTIONS
>>> $ make
>>> $ make install
>>> -Steve
>>>
>>>
>>> On 2 February 2011 20:16, Madovsky <infos at madovsky.org> wrote:
>>>>
>>>> it happens on my whole cluster since I updated 2 hours ago
>>>>
>>>> [43671.448504] freeswitch[14963]: segfault at 11 ip 00007fce18a3f285 sp
>>>> 00007fce17e9d9a0 error 4 in libfreeswitch.so.1.0.0[7fce189e9000+1c6000]
>>>> [46054.077174] freeswitch[17916]: segfault at 11 ip 00007fdc09e28285 sp
>>>> 00007fdc092869a0 error 4 in libfreeswitch.so.1.0.0[7fdc09dd2000+1c6000]
>>>> [47426.166723] freeswitch[11108]: segfault at 11 ip 00007f86c7a67285 sp
>>>> 00007f86c6ec59a0 error 4 in libfreeswitch.so.1.0.0[7f86c7a11000+1c6000]
>>>> [57346.162886] freeswitch[24798]: segfault at 11 ip 00007ff8c4958285 sp
>>>> 00007ff8c3db69a0 error 4 in libfreeswitch.so.1.0.0
>>>> (deleted)[7ff8c4902000+1c6000]
>>>>
>>>> ----- Original Message -----
>>>> From: Steven Ayre
>>>> To: FreeSWITCH Users Help
>>>> Sent: Wednesday, February 02, 2011 3:01 PM
>>>> Subject: Re: [Freeswitch-users] last git problem ?
>>>> Does dmesg show that any segfaults have occurred?
>>>>
>>>> On 2 February 2011 19:54, Madovsky <infos at madovsky.org> wrote:
>>>>>
>>>>> after every hangup now FS quits unexpeclty,without any error
>>>>> message....
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> ________________________________
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> ________________________________
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 chat2jesse at gmail.com  Thu Feb  3 02:15:53 2011
From: chat2jesse at gmail.com (jesse)
Date: Wed, 2 Feb 2011 15:15:53 -0800
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
 for Windows in .NET
In-Reply-To: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
Message-ID: <AANLkTinu683CEVEtGJqx=mugykXbkzZ67wz1UmB=Drdo@mail.gmail.com>

Can you make it across platform?

- jesse
On Feb 1, 2011 10:42 PM, "Mitch Capper" <mitch.capper at gmail.com> wrote:
> In prep for tomorrows conference I am happy to release FSClient. I would
> like to thank DRK and especially jlenk for their continued support and
help
> on this project. jlenk created the installer, fixed code, and certainly
> helped bring it along quite a bit faster!
> FSClient is a windows SIP client:
> Most features of any standard sip client
> multiple calls at once
> transfer, holding, speakerphone, DND
> multiple SIP accounts
> advanced headset support (caller id, buttons, etc) for jabra and
plantronics
> out of the box (with plugin support for easily adding others)
> basic contact book support (with a sample XML contact book plugin
provided)
> All codecs (minus commercial g729 support) that freeswitch supports
>
> Give it a shot and let us know about any issues you run into, overall it
was
> decently tested for many months but some of the rapid changes as of late
> means there may be some larger bugs that crept in. I will also be on the
> conference call tomorrow and can answer questions there.
>
> And so to download the binaries:
> http://files.freeswitch.org/windows/installer/x86/FSClient.zip
> There will be a readme in the install dir on usage.
>
> The source code is available on the contrib git repo mitchcapper/FSClient.
>
> As for building from source you will need to do a few things to trunk
> currently. In the source folder you need to move mod_portaudio.c to the
> freeswitch\src\mod\endpoints\mod_portaudio folder (
> http://jira.freeswitch.org/browse/FS-3006 are the changes pending trunk).
> In addition move the portaudio.2010.vxcproj over the version in
> freeswitch\libs\portaudio\build\msvc to enable direct X support. Make sure
> you have the WPF Toolkit also installed. Build trunk and then for FSClient
> set the ENV var FREESWITCH_SRC_LOCATION to the src location and it will
also
> auto-copy the needed files into the build folder.
>
>
> History:
> Back in May I wanted a SIP client that supported the Jabra headsets, after
> not finding one that would work as I liked I looked to open source and
> everyone pointed me towards FSComm. FSComm is great, and has a huge amount
> of time poured into it, but the C++/QT base for it was not for me. .NET
was
> the natural choice due to its rapid development time and stability so I
went
> there.
>
> After a few weeks work I had a client that I used for over 6 months and
was
> fairly stable with relatively minor changes. A few weeks ago on the
> conference call someone mentioned wanting a windows sip client option and
I
> figured I would offer up my code base. Well getting private code ready for
> public use was a bit more complex than planned but its here now and
> certainly in the past several weeks has had a lot of development.
>
> ~Mitch
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From chat2jesse at gmail.com  Thu Feb  3 02:19:03 2011
From: chat2jesse at gmail.com (jesse)
Date: Wed, 2 Feb 2011 15:19:03 -0800
Subject: [Freeswitch-users] Voice quality monitoring via loopback on sip
	endpoint
In-Reply-To: <F9F3A5AC-3F5C-45BE-8FAF-C517A56828B2@ipeva.fr>
References: <66D781FD-7248-4A4A-AD83-04D2CDF1AAEF@gmail.com>
	<F9F3A5AC-3F5C-45BE-8FAF-C517A56828B2@ipeva.fr>
Message-ID: <AANLkTik+w6u4g+9AP44x2e_E91Xseo1SuyvZK-n-PmVQ@mail.gmail.com>

You can have SIPP UAC plays Rtp , SIPP UAS Echo back , then compare number
of packets.
On Feb 2, 2011 10:34 AM, "David Ponzone" <david.ponzone at ipeva.fr> wrote:
> A common way to test audio quality is to have a device (sipp for instance)
calling your FS and playing a known audio file.
> FS would then record it and then, by comparing the recorded and the
original one and using some signal processing, you can compute a MOSPESQ
figure.
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr
> tel: 01 74 03 18 97
> gsm: 06 66 98 76 34
>
> Service Client IPeva
> tel: 0811 46 26 26
> www.ipeva.fr - www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
non autoris?e est interdite. Tout message ?lectronique est susceptible
d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
> Le 02/02/2011 ? 19:25, Marc De Corny a ?crit :
>
>> Hi all,
>> I am thinking of a way of testing the quality of a voice call.
>>
>> Ideally i would like to create a call from FS with a loopback parameter
that tells phone to answer the call automatically and loopback the RTP. This
is a common functionality in most phoned (for example cisco)
>> The sent and received packets could be compared for jitter, latency and
packet loss and a result extracted.
>>
>> Does this already exist? I know there some specific tool that accomplish
this bit they are expensive and awkward to use. Starting and retrieving the
result from an api would be fantastic.
>>
>> Any ideas are welcome.
>> 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 mitch.capper at gmail.com  Thu Feb  3 02:59:52 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 2 Feb 2011 15:59:52 -0800
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
 for Windows in .NET
In-Reply-To: <AANLkTinu683CEVEtGJqx=mugykXbkzZ67wz1UmB=Drdo@mail.gmail.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
	<AANLkTinu683CEVEtGJqx=mugykXbkzZ67wz1UmB=Drdo@mail.gmail.com>
Message-ID: <AANLkTinF-5GKk7pWNyqw-ayBCst-vkTQ7tua8bSVvLCa@mail.gmail.com>

Originally I expected it to be cross platform, unfortunately mono (.net on
linux) does not support WPF and has stated they have no intention of doing
so (http://www.mono-project.com/WPF).  Unfortunately WinForms is a bit
painful gui wise and the only other choice was silverlight.  Silverlight
could actually replicate the GUI very closely but the one major downside is
that silverlight would require a separate executable that the freeswitch
core runs in.    Ironically this core could also be cross platform using
mono, giving a complete cross platform solution, but is not the way I chose
to go.     It would not be overly hard to port FSClient to silverlight but
will not be a task I am undertaking.   So for now, until WPF support in mono
or it is ported from WPF to silverlight, FSClient will remain windows only.
Sorry!

~Mitch

On Wed, Feb 2, 2011 at 3:15 PM, jesse <chat2jesse at gmail.com> wrote:

> Can you make it across platform?
>
> - jesse
>
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From edpimentl at gmail.com  Thu Feb  3 03:08:05 2011
From: edpimentl at gmail.com (EdPimentl)
Date: Wed, 2 Feb 2011 19:08:05 -0500
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
 for Windows in .NET
In-Reply-To: <AANLkTinCVZqYRxd0ZcQLj_kj42fYHDqUsEocEqW3oaph@mail.gmail.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
	<16100.1296638658@ccs.covici.com>
	<1296658879889-5985076.post@n2.nabble.com>
	<6549.1296659705@ccs.covici.com>
	<AANLkTinCVZqYRxd0ZcQLj_kj42fYHDqUsEocEqW3oaph@mail.gmail.com>
Message-ID: <AANLkTinNjGKvGXKvTh1dMgZVkwdimkPoUZPJKu7233-T@mail.gmail.com>

Can you provide a screenshot of the client?

Thanks in advance,
-E
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From infos at madovsky.org  Thu Feb  3 03:26:16 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 2 Feb 2011 19:26:16 -0500
Subject: [Freeswitch-users] last git problem ?
References: <6038028FB6434CCCA1E1DC4E5784A3B7@e1705><AANLkTimRKhCwF9xHixb5psPUf_zoffYk1s=C75_NVENB@mail.gmail.com><3AEF2A2C49A8453B8C9AC810639A7666@e1705><AANLkTimDCMZ5aRp2yfcV=paGUvU6nnPF2-6mCC_A=7wG@mail.gmail.com><E640C3E5DC4C410EA5980C6DA5A8D4E9@e1705><AANLkTimPepfmTp7S6hfirwnwjtEk3kJn-L44GOF2e0Rk@mail.gmail.com>
	<AANLkTi=VczA-PFC63CgSSEWPec5sBZ0=4UXVodX-GCWm@mail.gmail.com>
Message-ID: <EE4F68EC3C954393BDB5AE7BBC56CF0D@e1705>

I git pulled 30mn ago
and the problem disappear

thanks
  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, February 02, 2011 4:08 PM
  Subject: Re: [Freeswitch-users] last git problem ?


  There's also http://fisheye.freeswitch.org/changelog/freeswitch.git


  On 2 February 2011 21:07, Steven Ayre <steveayre at gmail.com> wrote:

    $ git clone git://git.freeswitch.org/freeswitch.git
    $ git log


    Look through the recent entries and pick one you feel comfortable using, then test to see if it's affected.


    -Steve



    On 2 February 2011 20:46, Madovsky <infos at madovsky.org> wrote:

      ok but where to find prevoius git ID ?

        ----- Original Message ----- 
        From: Steven Ayre 
        To: FreeSWITCH Users Help 
        Sent: Wednesday, February 02, 2011 3:23 PM
        Subject: Re: [Freeswitch-users] last git problem ?


        I'd suggest you run freeswitch with the -core option which will create a coredump. You can then use that to report the bug at http://jira.freeswitch.org/


        http://wiki.freeswitch.org/wiki/Reporting_Bugs
        http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Simple_bash_script_to_make_debug_easy


        You can checkout an earlier git version to compile and roll out a working earlier version if you need to get your cluster back online. It's a good idea when upgrading to test on a single server before rolling it out to the entire cluster in case there's this sort of problem. 


        $ git clone git://git.freeswitch.org/freeswitch.git
        $ cd freeswitch
        $ git checkout GIT_COMMIT_ID
        $ ./bootstrap.sh
        $ ./configure OPTIONS
        $ make
        $ make install


        -Steve





        On 2 February 2011 20:16, Madovsky <infos at madovsky.org> wrote:

          it happens on my whole cluster since I updated 2 hours ago

          [43671.448504] freeswitch[14963]: segfault at 11 ip 00007fce18a3f285 sp 00007fce17e9d9a0 error 4 in libfreeswitch.so.1.0.0[7fce189e9000+1c6000]
          [46054.077174] freeswitch[17916]: segfault at 11 ip 00007fdc09e28285 sp 00007fdc092869a0 error 4 in libfreeswitch.so.1.0.0[7fdc09dd2000+1c6000]
          [47426.166723] freeswitch[11108]: segfault at 11 ip 00007f86c7a67285 sp 00007f86c6ec59a0 error 4 in libfreeswitch.so.1.0.0[7f86c7a11000+1c6000]
          [57346.162886] freeswitch[24798]: segfault at 11 ip 00007ff8c4958285 sp 00007ff8c3db69a0 error 4 in libfreeswitch.so.1.0.0 (deleted)[7ff8c4902000+1c6000]

            ----- Original Message ----- 
            From: Steven Ayre 
            To: FreeSWITCH Users Help 
            Sent: Wednesday, February 02, 2011 3:01 PM
            Subject: Re: [Freeswitch-users] last git problem ?


            Does dmesg show that any segfaults have occurred?


            On 2 February 2011 19:54, Madovsky <infos at madovsky.org> wrote:

              after every hangup now FS quits unexpeclty,without any error message....

              _______________________________________________
              FreeSWITCH-users mailing list
              FreeSWITCH-users at lists.freeswitch.org
              http://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://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






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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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      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From mthakershi at gmail.com  Thu Feb  3 03:45:33 2011
From: mthakershi at gmail.com (Malay Thakershi)
Date: Wed, 2 Feb 2011 18:45:33 -0600
Subject: [Freeswitch-users] Session sleep method
Message-ID: <AANLkTikNbbBF1_g2VimrpdLzR8Ro7_VQ0oHMviYEinkx@mail.gmail.com>

Hello,

I am using .NET module. There is sleep method that has two arguments.

What is second argument for?

I use it:
mObjMainSession.sleep(200, 0);

What happens if I pass something other than zero as second argument?

I am having inconsistent delays between call paths and that is why I am
exploring this argument.

Thank you for any help.

Malay
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From Nabble at slickdeals.endjunk.com  Thu Feb  3 04:12:42 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 2 Feb 2011 17:12:42 -0800 (PST)
Subject: [Freeswitch-users] last git problem ?
In-Reply-To: <6038028FB6434CCCA1E1DC4E5784A3B7@e1705>
References: <6038028FB6434CCCA1E1DC4E5784A3B7@e1705>
Message-ID: <1296695562099-5987183.post@n2.nabble.com>



Madovsky wrote:
> 
> after every hangup now FS quits unexpeclty,without any error message....
I am glad to know that I am not alone. I have been seeing this problem since
last week. Since my built is for a Seagate DockStar, I never bother to post
asking for help here unless there are some other users here also encounter
the same problem.

I can also confirm that I had done another git pull at around 16:22:43
2011-02-02 and my FS hosted on my Seagate DockStar is now running on
FreeSWITCH Version 1.0.head (git-89c5f3b 2011-02-02 11-04-49 -0600) sans the
crashing problem after the call.

While my FS configuration remains the same, I also notice that the problem
with a one to two missing seconds at the beginning of an in/out-bound call
has disappeared with this upgrade. I had had run into this issue
intermittently since I started to compile FS git for my Seagate DockStar
device back in the end of 12/2010. Besides all these problems, all Seagate
DockStar owners can attest that the mod_dingaling on FS git would crash the
FS at the end of a call, especially if the caller hangs up first. This also
happened to Asterisk 1.8.x on a Seagate DockStar in other ways. Such
problems occur to a Seagate DockStar running on a Linux OS with Debian,
OpenWRT, and/or Ubuntu flavor. Apparently, the problem was due to codes
produced by GCC with the word misalignment on an ARM processor.

-----
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/last-git-problem-tp5986283p5987183.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Thu Feb  3 04:19:24 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 2 Feb 2011 17:19:24 -0800 (PST)
Subject: [Freeswitch-users] mod_com_g729 - moving licences to a new
	server
In-Reply-To: <AANLkTinPmSAR=OiatFKEP0sRVA6_4=DQ7C5UdQcxsTfM@mail.gmail.com>
References: <AANLkTikykPgaD0tErbMw=VOgfXqnwEwY4SzC_bNqSw-i@mail.gmail.com>
	<AANLkTinSROc3ETqi8vbs89swryJa1+TNKa8WWDP4_O6H@mail.gmail.com>
	<AANLkTinPmSAR=OiatFKEP0sRVA6_4=DQ7C5UdQcxsTfM@mail.gmail.com>
Message-ID: <1296695964562-5987200.post@n2.nabble.com>



Steven Ayre wrote:
> Licenses are linked to the server via the MAC addresses of all network
> cards
> on the server. If you can move all of those (or at least set the same MAC
> addresses) it'll let you use the license on the other server.
Steve, R U sure about this? Does this only apply to a G729 license purchased
from FS?

-----
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/mod-com-g729-moving-licences-to-a-new-server-tp5985791p5987200.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From infos at madovsky.org  Thu Feb  3 04:38:56 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 2 Feb 2011 20:38:56 -0500
Subject: [Freeswitch-users] last git problem ?
References: <6038028FB6434CCCA1E1DC4E5784A3B7@e1705><AANLkTimRKhCwF9xHixb5psPUf_zoffYk1s=C75_NVENB@mail.gmail.com><3AEF2A2C49A8453B8C9AC810639A7666@e1705><AANLkTimDCMZ5aRp2yfcV=paGUvU6nnPF2-6mCC_A=7wG@mail.gmail.com><E640C3E5DC4C410EA5980C6DA5A8D4E9@e1705><AANLkTimPepfmTp7S6hfirwnwjtEk3kJn-L44GOF2e0Rk@mail.gmail.com>
	<AANLkTi=VczA-PFC63CgSSEWPec5sBZ0=4UXVodX-GCWm@mail.gmail.com>
Message-ID: <D914928923924B16AFC053B411F5C4FB@e1705>

And the sound seems to be better stable
  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, February 02, 2011 4:08 PM
  Subject: Re: [Freeswitch-users] last git problem ?


  There's also http://fisheye.freeswitch.org/changelog/freeswitch.git


  On 2 February 2011 21:07, Steven Ayre <steveayre at gmail.com> wrote:

    $ git clone git://git.freeswitch.org/freeswitch.git
    $ git log


    Look through the recent entries and pick one you feel comfortable using, then test to see if it's affected.


    -Steve



    On 2 February 2011 20:46, Madovsky <infos at madovsky.org> wrote:

      ok but where to find prevoius git ID ?

        ----- Original Message ----- 
        From: Steven Ayre 
        To: FreeSWITCH Users Help 
        Sent: Wednesday, February 02, 2011 3:23 PM
        Subject: Re: [Freeswitch-users] last git problem ?


        I'd suggest you run freeswitch with the -core option which will create a coredump. You can then use that to report the bug at http://jira.freeswitch.org/


        http://wiki.freeswitch.org/wiki/Reporting_Bugs
        http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Simple_bash_script_to_make_debug_easy


        You can checkout an earlier git version to compile and roll out a working earlier version if you need to get your cluster back online. It's a good idea when upgrading to test on a single server before rolling it out to the entire cluster in case there's this sort of problem. 


        $ git clone git://git.freeswitch.org/freeswitch.git
        $ cd freeswitch
        $ git checkout GIT_COMMIT_ID
        $ ./bootstrap.sh
        $ ./configure OPTIONS
        $ make
        $ make install


        -Steve





        On 2 February 2011 20:16, Madovsky <infos at madovsky.org> wrote:

          it happens on my whole cluster since I updated 2 hours ago

          [43671.448504] freeswitch[14963]: segfault at 11 ip 00007fce18a3f285 sp 00007fce17e9d9a0 error 4 in libfreeswitch.so.1.0.0[7fce189e9000+1c6000]
          [46054.077174] freeswitch[17916]: segfault at 11 ip 00007fdc09e28285 sp 00007fdc092869a0 error 4 in libfreeswitch.so.1.0.0[7fdc09dd2000+1c6000]
          [47426.166723] freeswitch[11108]: segfault at 11 ip 00007f86c7a67285 sp 00007f86c6ec59a0 error 4 in libfreeswitch.so.1.0.0[7f86c7a11000+1c6000]
          [57346.162886] freeswitch[24798]: segfault at 11 ip 00007ff8c4958285 sp 00007ff8c3db69a0 error 4 in libfreeswitch.so.1.0.0 (deleted)[7ff8c4902000+1c6000]

            ----- Original Message ----- 
            From: Steven Ayre 
            To: FreeSWITCH Users Help 
            Sent: Wednesday, February 02, 2011 3:01 PM
            Subject: Re: [Freeswitch-users] last git problem ?


            Does dmesg show that any segfaults have occurred?


            On 2 February 2011 19:54, Madovsky <infos at madovsky.org> wrote:

              after every hangup now FS quits unexpeclty,without any error message....

              _______________________________________________
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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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        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From covici at ccs.covici.com  Thu Feb  3 04:49:31 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 02 Feb 2011 20:49:31 -0500
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
	for Windows in .NET
In-Reply-To: <1296680047815-5986486.post@n2.nabble.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
	<16100.1296638658@ccs.covici.com>
	<1296658879889-5985076.post@n2.nabble.com>
	<6549.1296659705@ccs.covici.com>
	<AANLkTinCVZqYRxd0ZcQLj_kj42fYHDqUsEocEqW3oaph@mail.gmail.com>
	<7179.1296662450@ccs.covici.com>
	<1296663161529-5985392.post@n2.nabble.com>
	<8132.1296664446@ccs.covici.com>
	<1296680047815-5986486.post@n2.nabble.com>
Message-ID: <17438.1296697771@ccs.covici.com>

Nope, uac is there.  Drk gave me a hint as to what is happening -- when
you made the templates for the buttons, list boxes, etc. you didn't
label them souia or msaa would see them, and maybe you labeled them
graphically -- this is why I am not seeing any controls.

Jeff Lenk <jeff at jefflenk.com> wrote:

> 
> John,
> 
> Do you have UAC disabled?
> -- 
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FSClient-Release-A-FreeSWITCH-SIP-Client-for-Windows-in-NET-tp5983787p5986486.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

-- 
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 u2nsam at gmail.com  Thu Feb  3 06:02:24 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 3 Feb 2011 08:32:24 +0530
Subject: [Freeswitch-users] context
In-Reply-To: <AANLkTim84VqQ79xq6amru_4GzvGP9rof001MJ8t2Zpvd@mail.gmail.com>
References: <AANLkTika5CBO5eVWuRimKY+Bth2YAFC0Tm8_JmFBnXwC@mail.gmail.com>
	<AANLkTim84VqQ79xq6amru_4GzvGP9rof001MJ8t2Zpvd@mail.gmail.com>
Message-ID: <AANLkTinutzgJ5GF_tP=gpfVaz03+CQ-b-CCjPh3rFL=_@mail.gmail.com>

So you say that the extensions are not registered with FS and they are
getting allowed by the acl, just like DIDs.
How can i rectify this,any thing that needs to be checked ?

http://pastebin.freeswitch.org/15217

Regards
Sam

On Wed, Feb 2, 2011 at 11:23 PM, Michael Collins <msc at freeswitch.org> wrote:

> Turn on console debug level output (default in fs_cli) and make the test
> call. Pastebin the output. Most likely the call is not being authorized
> because you are letting it in via an ACL or something like that.
>
> -MC
>
> On Wed, Feb 2, 2011 at 1:55 AM, Sam <u2nsam at gmail.com> wrote:
>
>> I have defined a user as :
>>
>> <user id="2099">
>>       <params>
>>         <param name="password" value="1234"/>
>>         <param name="dial-string" value="{presence_id=${dialed_user}@
>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>         <param name="mwi-account" value="2099 at voicemail"/>
>>         <param name="vm-password" value="1234"/>
>>         <param name="vm-email-all-messages" value="true"/>
>>         <!--param name="vm-attach-file" value="true"/-->
>>         <!--param name="vm-mailto"  value=""/-->
>>       </params>
>>       <variables>
>>         <variable name="user_context" value="inter"/>
>>         <variable name="force_transfer_context" value="inter"/>
>>         <variable name="accountcode" value="2099"/>
>>         <variable name="internal_caller_id_name" value="2099"/>
>>         <variable name="internal_caller_id_number" value="2099"/>
>>         <variable name="callgroup" value="1"/>
>>         <variable name="sip-force-contact"
>> value="NDLB-connectile-dysfunction"/>
>>       </variables>
>>
>>
>> and in the dialplan i have
>>
>> <context name="inter">
>>
>> <extension name="2099">
>>     <condition field="destination_number" expression="^(2099)">
>>       <action application="set" data="call_timeout=30"/>
>>       <action application="set" data="dialed_extension=$1"/>
>>             <action application="export" data="dialed_extension=$1"/>
>>             <action application="set"
>> data="transfer_ringback=$${hold_music}"/>
>>             <action application="set" data="hangup_after_bridge=true"/>
>>             <action application="set" data="continue_on_fail=true"/>
>>             <action application="hash"
>> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>>             <action application="hash"
>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>>             <action application="hash"
>> data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
>>             <action application="hash"
>> data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>>             <action application="set"
>> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
>> var callgroup)}"/>
>>             <action application="hash"
>> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>>             <action application="bridge"
>> data="{sip_invite_domain=$${domain}}user/${dialed_extension}@
>> ${domain_name}"/>
>>     </condition>
>>   </extension>
>>
>> </context>
>>
>>
>> Now extension 2075 is also made on above lines in the same context for
>> user and dialplan, here the call when initiated from 2075 to 2099 searches
>> for context public instead of context inter
>>
>> mod_dialplan_xml.c:331 Processing 2075 <2075>->2099 in context public
>>
>> any reason why ?
>>
>> 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 mitch.capper at gmail.com  Thu Feb  3 06:07:58 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 2 Feb 2011 19:07:58 -0800
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
 for Windows in .NET
In-Reply-To: <AANLkTinNjGKvGXKvTh1dMgZVkwdimkPoUZPJKu7233-T@mail.gmail.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
	<16100.1296638658@ccs.covici.com>
	<1296658879889-5985076.post@n2.nabble.com>
	<6549.1296659705@ccs.covici.com>
	<AANLkTinCVZqYRxd0ZcQLj_kj42fYHDqUsEocEqW3oaph@mail.gmail.com>
	<AANLkTinNjGKvGXKvTh1dMgZVkwdimkPoUZPJKu7233-T@mail.gmail.com>
Message-ID: <AANLkTikP4H38vmvvVTCHRJdRa=60f4q4RLL+A3GiznNO@mail.gmail.com>

http://wiki.freeswitch.org/wiki/File:Fsclientscreen.png
~Mitch

On Wed, Feb 2, 2011 at 4:08 PM, EdPimentl <edpimentl at gmail.com> wrote:

> Can you provide a screenshot of the client?
>
> Thanks in advance,
> -E
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  3 06:18:40 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 3 Feb 2011 08:48:40 +0530
Subject: [Freeswitch-users] context
In-Reply-To: <AANLkTim84VqQ79xq6amru_4GzvGP9rof001MJ8t2Zpvd@mail.gmail.com>
References: <AANLkTika5CBO5eVWuRimKY+Bth2YAFC0Tm8_JmFBnXwC@mail.gmail.com>
	<AANLkTim84VqQ79xq6amru_4GzvGP9rof001MJ8t2Zpvd@mail.gmail.com>
Message-ID: <AANLkTinLFzDZa+t250Sc3BctDeHwj7Qm+Kw0D+184H0Q@mail.gmail.com>

I could see to registered extensions:-


sofia status profile internal reg

Registrations:
=================================================================================================
Call-ID:        ea5466005648d71f
User:           2075 at x.x.x.x
Contact:        2075 <sip:2075 at x.x.x.x:9061;fs_nat=yes>
Agent:          eyeBeam release 3007n stamp 17816
Status:         Registered(AUTO-NAT)(unknown) EXP(2011-02-03 08:46:39)
EXPSECS(68)
Host:           MediaServer
IP:            x.x.x.x
Port:           9061
Auth-User:      2075
Auth-Realm:     x.x.x.x
MWI-Account:    2075 at voicemail

Call-ID:        ZDRkZGYyOTg0ODBhN2IwNDcyMGQzMjhhYmJiZWI5ZjM.
User:           2099 at x.x.x.x
Contact:        "2099" <sip:2099 at x.x.x.x
:33741;rinstance=540b77d0000a0bb4;fs_nat=yes>
Agent:          X-Lite 4 release 4.0 stamp 58832
Status:         Registered(AUTO-NAT)(unknown) EXP(2011-02-03 08:46:59)
EXPSECS(88)
Host:           MediaServer
IP:             x.x.x.x
Port:           33741
Auth-User:      2099
Auth-Realm:     x.x.x.x
MWI-Account:    2099 at voicemail

Total items returned: 2
=================================================================================================


On Wed, Feb 2, 2011 at 11:23 PM, Michael Collins <msc at freeswitch.org> wrote:

> Turn on console debug level output (default in fs_cli) and make the test
> call. Pastebin the output. Most likely the call is not being authorized
> because you are letting it in via an ACL or something like that.
>
> -MC
>
> On Wed, Feb 2, 2011 at 1:55 AM, Sam <u2nsam at gmail.com> wrote:
>
>> I have defined a user as :
>>
>> <user id="2099">
>>       <params>
>>         <param name="password" value="1234"/>
>>         <param name="dial-string" value="{presence_id=${dialed_user}@
>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>         <param name="mwi-account" value="2099 at voicemail"/>
>>         <param name="vm-password" value="1234"/>
>>         <param name="vm-email-all-messages" value="true"/>
>>         <!--param name="vm-attach-file" value="true"/-->
>>         <!--param name="vm-mailto"  value=""/-->
>>       </params>
>>       <variables>
>>         <variable name="user_context" value="inter"/>
>>         <variable name="force_transfer_context" value="inter"/>
>>         <variable name="accountcode" value="2099"/>
>>         <variable name="internal_caller_id_name" value="2099"/>
>>         <variable name="internal_caller_id_number" value="2099"/>
>>         <variable name="callgroup" value="1"/>
>>         <variable name="sip-force-contact"
>> value="NDLB-connectile-dysfunction"/>
>>       </variables>
>>
>>
>> and in the dialplan i have
>>
>> <context name="inter">
>>
>> <extension name="2099">
>>     <condition field="destination_number" expression="^(2099)">
>>       <action application="set" data="call_timeout=30"/>
>>       <action application="set" data="dialed_extension=$1"/>
>>             <action application="export" data="dialed_extension=$1"/>
>>             <action application="set"
>> data="transfer_ringback=$${hold_music}"/>
>>             <action application="set" data="hangup_after_bridge=true"/>
>>             <action application="set" data="continue_on_fail=true"/>
>>             <action application="hash"
>> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>>             <action application="hash"
>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>>             <action application="hash"
>> data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
>>             <action application="hash"
>> data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>>             <action application="set"
>> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
>> var callgroup)}"/>
>>             <action application="hash"
>> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>>             <action application="bridge"
>> data="{sip_invite_domain=$${domain}}user/${dialed_extension}@
>> ${domain_name}"/>
>>     </condition>
>>   </extension>
>>
>> </context>
>>
>>
>> Now extension 2075 is also made on above lines in the same context for
>> user and dialplan, here the call when initiated from 2075 to 2099 searches
>> for context public instead of context inter
>>
>> mod_dialplan_xml.c:331 Processing 2075 <2075>->2099 in context public
>>
>> any reason why ?
>>
>> 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 edpimentl at gmail.com  Thu Feb  3 06:37:51 2011
From: edpimentl at gmail.com (EdPimentl)
Date: Wed, 2 Feb 2011 22:37:51 -0500
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
 for Windows in .NET
In-Reply-To: <AANLkTikP4H38vmvvVTCHRJdRa=60f4q4RLL+A3GiznNO@mail.gmail.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
	<16100.1296638658@ccs.covici.com>
	<1296658879889-5985076.post@n2.nabble.com>
	<6549.1296659705@ccs.covici.com>
	<AANLkTinCVZqYRxd0ZcQLj_kj42fYHDqUsEocEqW3oaph@mail.gmail.com>
	<AANLkTinNjGKvGXKvTh1dMgZVkwdimkPoUZPJKu7233-T@mail.gmail.com>
	<AANLkTikP4H38vmvvVTCHRJdRa=60f4q4RLL+A3GiznNO@mail.gmail.com>
Message-ID: <AANLkTi=skc8OGLqkiTULMN-pSL9SNgwK-YrcLq6N-GXp@mail.gmail.com>

Thanks

-E
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From dome at tel.co.th  Thu Feb  3 08:13:21 2011
From: dome at tel.co.th (dome at tel.co.th)
Date: Thu, 3 Feb 2011 12:13:21 +0700
Subject: [Freeswitch-users] Fs runninf on Clound Service (XEN)
Message-ID: <AANLkTimhmgVpdSQa+UwBO8DziQ5sSeFcNsU-mh8HVqpa@mail.gmail.com>

Dear All,
   I'm testing FS on http://cserver.cloudwww.com/  then give me IP : 61.x.x.x
and in eth0

eth0      Link encap:Ethernet  HWaddr 00:16:3E:2C:91:DC
          inet addr:10.6.32.11  Bcast:10.6.33.255  Mask:255.255.254.0
          UP BROADCAST RUNNING MULTICAST  MTU:1500  Metric:1
          RX packets:160267 errors:0 dropped:0 overruns:0 frame:0
          TX packets:93251 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:1000
          RX bytes:189671769 (180.8 MiB)  TX bytes:9669248 (9.2 MiB)

So i can access 61.x.x.x without problem. but i want to know how to
config sofia profile for binding 61.x.x.x
in default config. i can send call to 61.x.x.x but when call out i got
10.6.32.11 and i can't find 61.x.x.x in sip header
someone help me please


BG

Dome C.


From dome at tel.co.th  Thu Feb  3 08:27:03 2011
From: dome at tel.co.th (dome at tel.co.th)
Date: Thu, 3 Feb 2011 12:27:03 +0700
Subject: [Freeswitch-users] Fs runninf on Clound Service (XEN)
In-Reply-To: <AANLkTimhmgVpdSQa+UwBO8DziQ5sSeFcNsU-mh8HVqpa@mail.gmail.com>
References: <AANLkTimhmgVpdSQa+UwBO8DziQ5sSeFcNsU-mh8HVqpa@mail.gmail.com>
Message-ID: <AANLkTikpMRQMdBh9XAGaHU-558LzSik4UCe-W4YHCpRq@mail.gmail.com>

Got it :)
like EC2
http://wiki.freeswitch.org/wiki/Amazon_EC2

Dome C.

On Thu, Feb 3, 2011 at 12:13 PM, dome at tel.co.th <dome at tel.co.th> wrote:
> Dear All,
> ? I'm testing FS on http://cserver.cloudwww.com/ ?then give me IP : 61.x.x.x
> and in eth0
>
> eth0 ? ? ?Link encap:Ethernet ?HWaddr 00:16:3E:2C:91:DC
> ? ? ? ? ?inet addr:10.6.32.11 ?Bcast:10.6.33.255 ?Mask:255.255.254.0
> ? ? ? ? ?UP BROADCAST RUNNING MULTICAST ?MTU:1500 ?Metric:1
> ? ? ? ? ?RX packets:160267 errors:0 dropped:0 overruns:0 frame:0
> ? ? ? ? ?TX packets:93251 errors:0 dropped:0 overruns:0 carrier:0
> ? ? ? ? ?collisions:0 txqueuelen:1000
> ? ? ? ? ?RX bytes:189671769 (180.8 MiB) ?TX bytes:9669248 (9.2 MiB)
>
> So i can access 61.x.x.x without problem. but i want to know how to
> config sofia profile for binding 61.x.x.x
> in default config. i can send call to 61.x.x.x but when call out i got
> 10.6.32.11 and i can't find 61.x.x.x in sip header
> someone help me please
>
>
> BG
>
> Dome C.
>


From joaocarlosleme at gmail.com  Thu Feb  3 09:21:43 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Wed, 2 Feb 2011 22:21:43 -0800
Subject: [Freeswitch-users] Caller ID using Fifo
In-Reply-To: <AANLkTinHypD7aCBQwzZxF=DgppbZ_4j8f9x6W6JM=fe+@mail.gmail.com>
References: <AANLkTik6pdgXgND6kmzZA4qe8vDZQ0_7e47+ciVWPfub@mail.gmail.com>
	<AANLkTikKDO3qru-9=nuPEQcW+iCN7ccizyJAh9sqUkCN@mail.gmail.com>
	<AANLkTi=p7_CsUDWS62+weZv50+PBZ8RLop0Q04y-c4hB@mail.gmail.com>
	<AANLkTi=m_hW9vQV5nbC2jwcos0bKy2rCf7YAr6aA23oz@mail.gmail.com>
	<AANLkTinX7fZOO9edgHs739RxhcBJwuh70E9f07mM0mzL@mail.gmail.com>
	<201101251040421253879@asiainfo-linkage.com>
	<AANLkTinOzCVm4rFqm8MGp-_mgc6Ub3t4BZ+c9NLeG=oQ@mail.gmail.com>
	<AANLkTin0BZUeaqbWNrErfpB7_wCoJaYANhAADM2QisDy@mail.gmail.com>
	<AANLkTi=yhRKUpnyApJF4KGqnfECATSTkdURBeevq=iXX@mail.gmail.com>
	<AANLkTinHypD7aCBQwzZxF=DgppbZ_4j8f9x6W6JM=fe+@mail.gmail.com>
Message-ID: <AANLkTin86Qbk-jeihL9-NPZyURFSLb0P=qFY=6C5p+-N@mail.gmail.com>

Thanks Anthony it works great. The problem was my version, I was using the
precompiled one but got the latest git just for that and it worked. Although
to display the variable caller id you have to set on the dialplan (not on
fifo.conf).

On Thu, Jan 27, 2011 at 6:35 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> not the dialplan,
> the fifo config file:
>
> autoload_configs/fifo.conf.xml
>
>
> configuration name="fifo.conf" description="FIFO Configuration">
>  <settings>
>    <param name="delete-all-outbound-member-on-startup" value="false"/>
>  </settings>
>  <fifos>
>    <fifo name="test_fifo@$${domain}" importance="0" outbound_name="sales">
>      <member timeout="60" simo="1"
>
> lag="20">{origination_caller_id_name=fred,origination_caller_id_number=1234}user/1005@
> $${domain}</member>
>    </fifo>
>  </fifos>
> </configuration>
>
>
>
>
> On Thu, Jan 27, 2011 at 12:55 AM, Marc de Corny <marcdecorny at gmail.com>
> wrote:
> > Hi Anthony,
> >
> > I updated it the other day.
> >
> > This is my version :
> > freeswitch at internal> version
> > FreeSWITCH Version 1.0.head (git-6faa4c9 2010-12-02 17-11-04 -0600)
> >
> > Having said that I am using remote SIP endpoints on another SIP platform,
> > maybe that is why. I will take a closer look at the signalling and see.
> >
> > thanks
> > Marc
> >
> > On Thu, Jan 27, 2011 at 1:50 AM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> If they made no difference you are not on latest GIT HEAD
> >>
> >>
> >> On Wed, Jan 26, 2011 at 11:38 AM, Marc de Corny <marcdecorny at gmail.com>
> >> wrote:
> >> > I can see that a number of us are interested in this.
> >> > I have tried to set that outbound_name and fifo_outbound_name before
> >> > send
> >> > ing the call the queue but they made no difference.
> >> >
> >> > and when I do  fifo list, I cannot see any variables with that name
> >> > available to set. Am I not looking in the right place.
> >> >
> >> > I'm looking into <action application="set"
> >> > data="fifo_orbit_dialplan=XML"/>
> >> > as a potential way out, whereby as the call enters the FIFO I could
> >> > record
> >> > the outbound_name requested and then if I can control the call on the
> >> > way
> >> > out I can set it again. How does this command above allow me to send
> the
> >> > calls out a certain way and treat them. what are the options for that
> >> > fifo_orbit_dialplan
> >> >
> >> > thanks
> >> > Marc
> >> > On Tue, Jan 25, 2011 at 2:40 AM, liuyp2 <liuyp2 at asiainfo-linkage.com>
> >> > wrote:
> >> >>
> >> >> mod_fifo can't transfer sip header  message which defined by
> >> >> myself(sip_h_X-xxx) to b-leg also.
> >> >>
> >> >> Is there any solution in latest version?
> >> >>
> >> >> ________________________________
> >> >> liuyp2
> >> >> 2011-01-25
> >> >> ________________________________
> >> >>
> >> >> ???? Anthony Minessale
> >> >> ????? 2011-01-25 09:28:25
> >> >> ???? FreeSWITCH Users Help
> >> >> ???
> >> >> ??? Re: [Freeswitch-users] Caller ID using Fifo
> >> >>
> >> >>
> >> >> You should all confer to make sure you are all using fs latest git
> >> >> because
> >> >> that is the version I am talking about.  Fifo has some major new
> >> >> features in
> >> >> latest that do not exist in older versions including showing the
> >> >> customers
> >> >> cid when it calls agents.  The dilemma jm describes used to be true
> but
> >> >> is
> >> >> no longer the case with the default ringall strategy on latest git.
> >> >>
> >> >> The customers cid is sent to the agent and if the fifo xml defines
> >> >> outbound_name param that will be included as well.
> >> >>
> >> >> If you want to override it you must do what you quoted in the wiki in
> >> >> the
> >> >> dialstring contained in the member tag of the xml for that membership
> >> >> not in
> >> >> the dialplan.
> >> >>
> >> >> On Jan 14, 2011 10:36 AM, "Marc de Corny" <marcdecorny at gmail.com>
> >> >> wrote:
> >> >> >
> >> >> > Just to follow up on this subject.
> >> >> >
> >> >> > I have done a lot of testing on the fifo trying to get the
> >> >> > caller_id_name changed on the outbound call to the agent and to be
> >> >> > honest I
> >> >> > cannot understand the explanation.
> >> >> >
> >> >> > If mod_fifo does not know which call it will connect until the
> agent
> >> >> > answers, how come it displays the CLI correctly, jsut won;t let me
> >> >> > change
> >> >> > it.
> >> >> >
> >> >> > Still seems strange. I am looking into the Mod_callcentre to check
> if
> >> >> > it
> >> >> > sends caller_id information. but the same logic if valid could
> apply
> >> >> >
> >> >> > Also maybe someone should change the Wiki ( I would but do not have
> >> >> > enough expertise on the subject) because the following is a bit
> >> >> > misleading
> >> >> >
> >> >> >  "Note: If you wish to specify the caller ID presented when a fifo
> >> >> > calls
> >> >> > an agent, set the origination_caller_id_name and
> >> >> > origination_caller_id_num
> >> >> > variables to the values desired. These could be set within the {}
> of
> >> >> > the
> >> >> > dialstring, or they could be set using the set application in the
> >> >> > dialplan
> >> >> > which places the caller into the fifo (before the 'fifo in'
> executed
> >> >> > on the
> >> >> > caller). "
> >> >> > thanks
> >> >> > Marc
> >> >> > On Thu, Jan 13, 2011 at 10:47 PM, Joao Leme
> >> >> > <joaocarlosleme at gmail.com>
> >> >> > wrote:
> >> >> >>
> >> >> >> What about showing the Caller ID after it is answered? Any way to
> do
> >> >> >> that?
> >> >> >>
> >> >> >> 2011/1/12 Jo?o Mesquita <jmesquita at freeswitch.org>
> >> >> >>
> >> >> >>> Jo?o Leme,
> >> >> >>>
> >> >> >>> The caller id is not passed when the phone is ringing because
> >> >> >>> mod_fifo
> >> >> >>> does not know which call is going to be sent to that channel once
> >> >> >>> it is
> >> >> >>> answered until it is really answered. I don't know if
> >> >> >>> mod_callcenter does
> >> >> >>> show anything but you should consider looking at the
> documentation
> >> >> >>> if you
> >> >> >>> really need this feature.
> >> >> >>>
> >> >> >>> Regards,
> >> >> >>> Jo?o Mesquita
> >> >> >>>
> >> >> >>>
> >> >> >>> On Wed, Jan 12, 2011 at 9:15 PM, Joao Leme
> >> >> >>> <joaocarlosleme at gmail.com>
> >> >> >>> wrote:
> >> >> >>>>
> >> >> >>>> Hi there,
> >> >> >>>> I would like to know if there is a way to see the caller ID on
> my
> >> >> >>>> Sip
> >> >> >>>> Client (X-Lite for example) of the caller that I answear from a
> >> >> >>>> Fifo queue?
> >> >> >>>> Thanks,
> >> >> >>>> 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
> >> >> >>>
> >> >> >>
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >>
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at 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 <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  3 09:45:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 2 Feb 2011 22:45:02 -0800
Subject: [Freeswitch-users] context
In-Reply-To: <AANLkTinutzgJ5GF_tP=gpfVaz03+CQ-b-CCjPh3rFL=_@mail.gmail.com>
References: <AANLkTika5CBO5eVWuRimKY+Bth2YAFC0Tm8_JmFBnXwC@mail.gmail.com>
	<AANLkTim84VqQ79xq6amru_4GzvGP9rof001MJ8t2Zpvd@mail.gmail.com>
	<AANLkTinutzgJ5GF_tP=gpfVaz03+CQ-b-CCjPh3rFL=_@mail.gmail.com>
Message-ID: <AANLkTikmnwQjXSX9GaC0Q3DGQYOTpv9-LE29AFW-VA4T@mail.gmail.com>

I think I'm going to write an article on this topic for the new folks since
it seems to be a common point of confusion. Don't worry, you're not alone -
this is a common challenge for new FS admins.

#1 - The first line of your pastebin is the key. You are letting the call in
via an ACL. This is "okay" if it is what you want, but I doubt that it is. I
recommend that you remove this phone's IP address from the "domains" section
of acl.conf.xml. When using the default FS configs, if you let a phone call
through using the "domains" ACL then it automatically goes to the public
context. Why? Because the call is not explicitly associated with a local
user on the system. It's like getting a free pass into Disneyland but you
don't have a little badge that says, "Hi, my name is..."

Some get confused by the "Falling back to digest auth" message. It looks
like an error but it's really just information. If the caller's SIP client
properly authenticates (which is different than registration - see below)
then FS knows exactly who is making the call and that it is a local user so
it goes into the "default" context.

#2 - Make sure that you learn the difference between SIP *registration* and
SIP *authentication*.  Registration is where the SIP client tells FS:
"Here's how you can reach me if you get a call for me." It is for calls TO
the phone, not calls from the phone. On the other hand, call authentication
is for calls FROM the phone to FS. When the phone calls FS, FS first checks
to see if the phone's IP is in the "domains" ACL. If it is, then it just
let's the call in to be handled by the public context. If not, then FS sends
an "auth challenge" - basically saying, "What's the password?" If the SIP
client properly responds then the call is "authenticated" - meaning that FS
knows it is from a specific user and thus it goes to the default context.

I recommend that you get the FS book and look at chapter 4. We talk a lot
about the user directory and it will help you understand how it all works.
Hopefully this explanation will whet your appetite for more. :)

Keep hacking away at it - you'll get it soon enough!

-MC

On Wed, Feb 2, 2011 at 7:02 PM, Sam <u2nsam at gmail.com> wrote:

> So you say that the extensions are not registered with FS and they are
> getting allowed by the acl, just like DIDs.
> How can i rectify this,any thing that needs to be checked ?
>
> http://pastebin.freeswitch.org/15217
>
> Regards
> Sam
>
>
> On Wed, Feb 2, 2011 at 11:23 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Turn on console debug level output (default in fs_cli) and make the test
>> call. Pastebin the output. Most likely the call is not being authorized
>> because you are letting it in via an ACL or something like that.
>>
>> -MC
>>
>> On Wed, Feb 2, 2011 at 1:55 AM, Sam <u2nsam at gmail.com> wrote:
>>
>>> I have defined a user as :
>>>
>>> <user id="2099">
>>>       <params>
>>>         <param name="password" value="1234"/>
>>>         <param name="dial-string" value="{presence_id=${dialed_user}@
>>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>>         <param name="mwi-account" value="2099 at voicemail"/>
>>>         <param name="vm-password" value="1234"/>
>>>         <param name="vm-email-all-messages" value="true"/>
>>>         <!--param name="vm-attach-file" value="true"/-->
>>>         <!--param name="vm-mailto"  value=""/-->
>>>       </params>
>>>       <variables>
>>>         <variable name="user_context" value="inter"/>
>>>         <variable name="force_transfer_context" value="inter"/>
>>>         <variable name="accountcode" value="2099"/>
>>>         <variable name="internal_caller_id_name" value="2099"/>
>>>         <variable name="internal_caller_id_number" value="2099"/>
>>>         <variable name="callgroup" value="1"/>
>>>         <variable name="sip-force-contact"
>>> value="NDLB-connectile-dysfunction"/>
>>>       </variables>
>>>
>>>
>>> and in the dialplan i have
>>>
>>> <context name="inter">
>>>
>>> <extension name="2099">
>>>     <condition field="destination_number" expression="^(2099)">
>>>       <action application="set" data="call_timeout=30"/>
>>>       <action application="set" data="dialed_extension=$1"/>
>>>             <action application="export" data="dialed_extension=$1"/>
>>>             <action application="set"
>>> data="transfer_ringback=$${hold_music}"/>
>>>             <action application="set" data="hangup_after_bridge=true"/>
>>>             <action application="set" data="continue_on_fail=true"/>
>>>             <action application="hash"
>>> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>>>             <action application="hash"
>>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>>>             <action application="hash"
>>> data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
>>>             <action application="hash"
>>> data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>>>             <action application="set"
>>> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
>>> var callgroup)}"/>
>>>             <action application="hash"
>>> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>>>             <action application="bridge"
>>> data="{sip_invite_domain=$${domain}}user/${dialed_extension}@
>>> ${domain_name}"/>
>>>     </condition>
>>>   </extension>
>>>
>>> </context>
>>>
>>>
>>> Now extension 2075 is also made on above lines in the same context for
>>> user and dialplan, here the call when initiated from 2075 to 2099 searches
>>> for context public instead of context inter
>>>
>>> mod_dialplan_xml.c:331 Processing 2075 <2075>->2099 in context public
>>>
>>> any reason why ?
>>>
>>> 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 kapil.rastogi at telemune.net  Thu Feb  3 10:10:53 2011
From: kapil.rastogi at telemune.net (kapil.rastogi)
Date: Wed, 2 Feb 2011 23:10:53 -0800 (PST)
Subject: [Freeswitch-users] How to record a conference using javascript
	application
In-Reply-To: <AANLkTikZg=tYFKC61EoiFP7rbrQBv0NswzH41tqDS8sE@mail.gmail.com>
References: <1296664102854-5985457.post@n2.nabble.com>
	<AANLkTikZg=tYFKC61EoiFP7rbrQBv0NswzH41tqDS8sE@mail.gmail.com>
Message-ID: <AANLkTimqZrM+Pe=uQvFVyfqcyy_4WOOm5Qfio0tSaTzF@mail.gmail.com>


Hi,

I have also tried for the same, but it is also not working.

On Thu, Feb 3, 2011 at 1:06 AM, mercutioviz [via freeswitch-users] <
ml-node+5986202-689963855-310600 at n2.nabble.com<ml-node%2B5986202-689963855-310600 at n2.nabble.com>
> wrote:

> I believe that you need to use the conference API, not the conference
> dialplan application. Try this:
>
> apiExecute("conference", "conf-room--"+room_id+"@default record
>
> /usr/local/freeswitch/abc.wav");
>
> let us know if that helps...
>
> -MC
>
> On Wed, Feb 2, 2011 at 8:28 AM, kapil.rastogi <[hidden email]<http://user/SendEmail.jtp?type=node&node=5986202&i=0>
> > wrote:
>
>>
>> Hi,
>>
>> I want to record the chatting b/w all members during the conference. I am
>> using the following statement in my javascript application:
>>
>>  session.execute("conference", "conf-room--"+room_id+"@default record
>> /usr/local/freeswitch/abc.wav");
>>
>> But i am unable to record the conference room chatting. But when i record
>> the same from fs_cli command line, it is working well.
>>
>> Please tell me how to record the conference chatting.
>>
>> Regards,
>>
>> -----
>> Regards,
>> Kapil Rastogi
>> Telemune Software Solutions P Ltd.
>> [hidden email] <http://user/SendEmail.jtp?type=node&node=5986202&i=1>
>> +919013204760
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/How-to-record-a-conference-using-javascript-application-tp5985457p5985457.html<http://freeswitch-users.2379917.n2.nabble.com/How-to-record-a-conference-using-javascript-application-tp5985457p5985457.html?by-user=t>
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> [hidden email] <http://user/SendEmail.jtp?type=node&node=5986202&i=2>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> [hidden email] <http://user/SendEmail.jtp?type=node&node=5986202&i=3>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ------------------------------
>  If you reply to this email, your message will be added to the discussion
> below:
>
> http://freeswitch-users.2379917.n2.nabble.com/How-to-record-a-conference-using-javascript-application-tp5985457p5986202.html
>  To unsubscribe from How to record a conference using javascript
> application, click here<http://freeswitch-users.2379917.n2.nabble.com/template/NamlServlet.jtp?macro=unsubscribe_by_code&node=5985457&code=a2FwaWwucmFzdG9naUB0ZWxlbXVuZS5uZXR8NTk4NTQ1N3wzNjc1NjMwMjg=>.
>
>



-- 
Regards,
Kapil Rastogi
Telemune Software Solutions P Ltd.
kapil.rastogi at telemune.net
+919013204760


-----
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/How-to-record-a-conference-using-javascript-application-tp5985457p5987796.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
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From steveayre at gmail.com  Thu Feb  3 12:09:50 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 3 Feb 2011 09:09:50 +0000
Subject: [Freeswitch-users] mod_com_g729 - moving licences to a new
	server
In-Reply-To: <1296695964562-5987200.post@n2.nabble.com>
References: <AANLkTikykPgaD0tErbMw=VOgfXqnwEwY4SzC_bNqSw-i@mail.gmail.com>
	<AANLkTinSROc3ETqi8vbs89swryJa1+TNKa8WWDP4_O6H@mail.gmail.com>
	<AANLkTinPmSAR=OiatFKEP0sRVA6_4=DQ7C5UdQcxsTfM@mail.gmail.com>
	<1296695964562-5987200.post@n2.nabble.com>
Message-ID: <AANLkTik0Q70zJD1z2gZ6wp1aOHKcrWvT=B-Anv8hEyzk@mail.gmail.com>

That's what's been indicated by Brian in the past... though it could always
be linked to other hardware details too. It's pretty secure because you
can't run two servers on the same network with the same MAC at the same
time, otherwise packets'll go to the wrong server. Plenty of licensing
systems work this way. There's a lot of network cards that don't even let
you change the MAC any more, this might be partly why.

-Steve

On 3 February 2011 01:19, mazilo <Nabble at slickdeals.endjunk.com> wrote:

>
>
> Steven Ayre wrote:
> > Licenses are linked to the server via the MAC addresses of all network
> > cards
> > on the server. If you can move all of those (or at least set the same MAC
> > addresses) it'll let you use the license on the other server.
> Steve, R U sure about this? Does this only apply to a G729 license
> purchased
> from FS?
>
> -----
> 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/mod-com-g729-moving-licences-to-a-new-server-tp5985791p5987200.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 massimiliano.ravelli at gmail.com  Thu Feb  3 11:42:51 2011
From: massimiliano.ravelli at gmail.com (Massimiliano Ravelli)
Date: Thu, 3 Feb 2011 09:42:51 +0100
Subject: [Freeswitch-users] mod_com_g729 - moving licences to a new
	server
In-Reply-To: <AANLkTinPmSAR=OiatFKEP0sRVA6_4=DQ7C5UdQcxsTfM@mail.gmail.com>
References: <AANLkTikykPgaD0tErbMw=VOgfXqnwEwY4SzC_bNqSw-i@mail.gmail.com>
	<AANLkTinSROc3ETqi8vbs89swryJa1+TNKa8WWDP4_O6H@mail.gmail.com>
	<AANLkTinPmSAR=OiatFKEP0sRVA6_4=DQ7C5UdQcxsTfM@mail.gmail.com>
Message-ID: <AANLkTi=oNazw3U-nZ63Ro2oJh+Jgz_3O4_mrFmzS8VZA@mail.gmail.com>

2011/2/2 Steven Ayre <steveayre at gmail.com>

> Licenses are linked to the server via the MAC addresses of all network
> cards on the server. If you can move all of those (or at least set the same
> MAC addresses) it'll let you use the license on the other server. Do not try
> using the license on the old server at the same time.
>

Thanks Steven,
Unluckily the ethernet interfaces are integrated into the motherboard and
even with with network cards the downtime for the swap will be unacceptable.

If that's not possible or you'd rather have it linked to the new server try
> emailing consulting at freeswitch.org, they'll be able to reissue the license
> (but as part of the patent restrictions can only do this a limited number of
> times). They'll probably also see this thread.
>

We plan to swap the production machine back in the future and we need to
swap them quickly so I think we should buy some more licences.
By the way: does anyone know how much is "limited number of times" ?
Just curious, as our customers will need licences too and I'm wondering if
we can buy licences in advance for them.

You'll want to stop the current server before installing the license on the
> new server I believe.
>

Shutting down or, at least, disconnect from the network the current server
is already planned in any case.

mod_com_g729 can be upgraded, but not while you have G729 calls running.
> You'll need to do install the newer module, then do "reload mod_com_g729" at
> the FS cli. That'll unload the G729 codec module (which'll block until any
> current G729 calls end) then load the new one. Be careful on a production
> server - you should test doing so on a testbed first so you don't get any
> unexpected problems during the upgrade, otherwise your server may be offline
> for much longer than you'd expected.
>

Ok, thanks very much !

Regards,
Massimiliano
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From patrick.plattes at niemann-frey.info  Thu Feb  3 13:03:02 2011
From: patrick.plattes at niemann-frey.info (Patrick Plattes)
Date: Thu, 3 Feb 2011 11:03:02 +0100
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTinZ2P_b3-mtOX0J3HtsjfbE823cQ_j-0oVNd2-Q@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
	<AANLkTi=in3CLd-n0mT3SRKYA0kxLMGVWk9KeBx3_jhY6@mail.gmail.com>
	<AANLkTin3Bu+8EEr_v_GaRGNPzOjfcnfzsV2Av2cgnWgO@mail.gmail.com>
	<AANLkTi=OC_jx0+nzK405VZURLDJPD25R4ZjyJ3RXJ1nk@mail.gmail.com>
	<AANLkTi=Sm7CjoayiE8nHmL0Eehqknj1jdpbx0YBQJQGH@mail.gmail.com>
	<AANLkTinr0RdgBs2UiS3=b3i+k6uCwkMihDy_FbAOFPc_@mail.gmail.com>
	<AANLkTi=+aObW3kVBZq5ox4b4E1uFTSgtXUL72tHLOzPi@mail.gmail.com>
	<AANLkTikVDKS40dVsf8x=+5=qnW59LqJbSv1H=vtROFaM@mail.gmail.com>
	<AANLkTi=gJsVKCJifVaBL_H6ZnPHTD__aQPhPeVfdx-ju@mail.gmail.com>
	<AANLkTimeuCHf_7Y1TErPwrJbdGZ3DoQmeqQbJ1_3YNpT@mail.gmail.com>
	<AANLkTikDxzntMFRpFhfiqJ4pzvjsTQR4w_FfyLfT5Y7z@mail.gmail.com>
	<AANLkTimVrhpfX+0D2mFPwGHK1EDsALWjFGp8hh7=ecqt@mail.gmail.com>
	<AANLkTi=oqw7fX-kvwqtxUXEWmzDu4C-dOjPrhpsCiwQc@mail.gmail.com>
	<AANLkTikWUfpFpQjKQ_EiGQuUGqa33VwU2K5vYVRvFGFO@mail.gmail.com>
	<AANLkTi=z3yysHW7Y7YaB92H6CDe0GFStMdPjHyUBmSvG@mail.gmail.com>
	<AANLkTinDmkxEeO2px7NsOSO5DkTiq+=1xTY7togH2-hJ@mail.gmail.com>
	<AANLkTinFYHCsPUSmtZ_un=yuJ68TSgh8imxvPwz9iy1B@mail.gmail.com>
	<AANLkTinZ2P_b3-mtOX0J3HtsjfbE823cQ_j-0oVNd2-Q@mail.gmail.com>
Message-ID: <AANLkTikRuuPD5eX1tApvoE3ZRKf+EYu7vQWBi4eeOgsz@mail.gmail.com>

Thanks Anthony!

I'm also very often there, but it's usually to erly for you ;-). I
think I will spend the Friday night in front of the Phones.

Bye,
 Patrick (mrparity)


From patrick.plattes at niemann-frey.info  Thu Feb  3 13:15:32 2011
From: patrick.plattes at niemann-frey.info (Patrick Plattes)
Date: Thu, 3 Feb 2011 11:15:32 +0100
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTikRuuPD5eX1tApvoE3ZRKf+EYu7vQWBi4eeOgsz@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
	<AANLkTi=in3CLd-n0mT3SRKYA0kxLMGVWk9KeBx3_jhY6@mail.gmail.com>
	<AANLkTin3Bu+8EEr_v_GaRGNPzOjfcnfzsV2Av2cgnWgO@mail.gmail.com>
	<AANLkTi=OC_jx0+nzK405VZURLDJPD25R4ZjyJ3RXJ1nk@mail.gmail.com>
	<AANLkTi=Sm7CjoayiE8nHmL0Eehqknj1jdpbx0YBQJQGH@mail.gmail.com>
	<AANLkTinr0RdgBs2UiS3=b3i+k6uCwkMihDy_FbAOFPc_@mail.gmail.com>
	<AANLkTi=+aObW3kVBZq5ox4b4E1uFTSgtXUL72tHLOzPi@mail.gmail.com>
	<AANLkTikVDKS40dVsf8x=+5=qnW59LqJbSv1H=vtROFaM@mail.gmail.com>
	<AANLkTi=gJsVKCJifVaBL_H6ZnPHTD__aQPhPeVfdx-ju@mail.gmail.com>
	<AANLkTimeuCHf_7Y1TErPwrJbdGZ3DoQmeqQbJ1_3YNpT@mail.gmail.com>
	<AANLkTikDxzntMFRpFhfiqJ4pzvjsTQR4w_FfyLfT5Y7z@mail.gmail.com>
	<AANLkTimVrhpfX+0D2mFPwGHK1EDsALWjFGp8hh7=ecqt@mail.gmail.com>
	<AANLkTi=oqw7fX-kvwqtxUXEWmzDu4C-dOjPrhpsCiwQc@mail.gmail.com>
	<AANLkTikWUfpFpQjKQ_EiGQuUGqa33VwU2K5vYVRvFGFO@mail.gmail.com>
	<AANLkTi=z3yysHW7Y7YaB92H6CDe0GFStMdPjHyUBmSvG@mail.gmail.com>
	<AANLkTinDmkxEeO2px7NsOSO5DkTiq+=1xTY7togH2-hJ@mail.gmail.com>
	<AANLkTinFYHCsPUSmtZ_un=yuJ68TSgh8imxvPwz9iy1B@mail.gmail.com>
	<AANLkTinZ2P_b3-mtOX0J3HtsjfbE823cQ_j-0oVNd2-Q@mail.gmail.com>
	<AANLkTikRuuPD5eX1tApvoE3ZRKf+EYu7vQWBi4eeOgsz@mail.gmail.com>
Message-ID: <AANLkTi=djziwMa_ip7U4-bmwHrvzWSLPD0rZqQ3+9oef@mail.gmail.com>

There is a difference between the NOTIFY message body in the head and
1.0.6. Shouldn't the dialog-info have state child element?

fs 1.0.6:
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0"
state="full" entity="sip:queue-sales_fr-263 at 192.168.56.189">
  <dialog id="" direction="initiator">
    <state>confirmed</state>
  </dialog>
</dialog-info>


fs head:
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0"
state="partial" entity="sip:queue-sales_fr-263 at 192.168.56.189">
</dialog-info>


From steveayre at gmail.com  Thu Feb  3 13:17:56 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 3 Feb 2011 10:17:56 +0000
Subject: [Freeswitch-users] mod_com_g729 - moving licences to a new
	server
In-Reply-To: <AANLkTi=oNazw3U-nZ63Ro2oJh+Jgz_3O4_mrFmzS8VZA@mail.gmail.com>
References: <AANLkTikykPgaD0tErbMw=VOgfXqnwEwY4SzC_bNqSw-i@mail.gmail.com>
	<AANLkTinSROc3ETqi8vbs89swryJa1+TNKa8WWDP4_O6H@mail.gmail.com>
	<AANLkTinPmSAR=OiatFKEP0sRVA6_4=DQ7C5UdQcxsTfM@mail.gmail.com>
	<AANLkTi=oNazw3U-nZ63Ro2oJh+Jgz_3O4_mrFmzS8VZA@mail.gmail.com>
Message-ID: <AANLkTikpKu-T77QNvAGQevDZSOsncyQAYmG6v9=_1OV6@mail.gmail.com>

>
>  If that's not possible or you'd rather have it linked to the new server
>> try emailing consulting at freeswitch.org, they'll be able to reissue the
>> license (but as part of the patent restrictions can only do this a limited
>> number of times). They'll probably also see this thread.
>>
>
> We plan to swap the production machine back in the future and we need to
> swap them quickly so I think we should buy some more licences.
>
>> By the way: does anyone know how much is "limited number of times" ?


Yea, sounds like more licenses would be good then.

I think limited number of times probably means once, then you'll need to
provide proof that the old server is no longer being used. For comparison
that's the system I believe Digium use. You'd need to check the developers
though.

Just curious, as our customers will need licences too and I'm wondering if
> we can buy licences in advance for them.


That should be possible - when you install your license for the first time
it registers the details of the server you're installing on with
freeswitch.org and generates the license file at that time. You should be
able to buy licenses and they won't be activated until you install them on
your customers' servers.




On 3 February 2011 08:42, Massimiliano Ravelli <
massimiliano.ravelli at gmail.com> wrote:

> 2011/2/2 Steven Ayre <steveayre at gmail.com>
>
> Licenses are linked to the server via the MAC addresses of all network
>> cards on the server. If you can move all of those (or at least set the same
>> MAC addresses) it'll let you use the license on the other server. Do not try
>> using the license on the old server at the same time.
>>
>
> Thanks Steven,
> Unluckily the ethernet interfaces are integrated into the motherboard and
> even with with network cards the downtime for the swap will be unacceptable.
>
> If that's not possible or you'd rather have it linked to the new server try
>> emailing consulting at freeswitch.org, they'll be able to reissue the
>> license (but as part of the patent restrictions can only do this a limited
>> number of times). They'll probably also see this thread.
>>
>
> We plan to swap the production machine back in the future and we need to
> swap them quickly so I think we should buy some more licences.
> By the way: does anyone know how much is "limited number of times" ?
> Just curious, as our customers will need licences too and I'm wondering if
> we can buy licences in advance for them.
>
> You'll want to stop the current server before installing the license on the
>> new server I believe.
>>
>
> Shutting down or, at least, disconnect from the network the current server
> is already planned in any case.
>
> mod_com_g729 can be upgraded, but not while you have G729 calls running.
>> You'll need to do install the newer module, then do "reload mod_com_g729" at
>> the FS cli. That'll unload the G729 codec module (which'll block until any
>> current G729 calls end) then load the new one. Be careful on a production
>> server - you should test doing so on a testbed first so you don't get any
>> unexpected problems during the upgrade, otherwise your server may be offline
>> for much longer than you'd expected.
>>
>
> Ok, thanks very much !
>
> Regards,
> Massimiliano
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  3 13:31:01 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Thu, 3 Feb 2011 13:31:01 +0300
Subject: [Freeswitch-users] Choppy VM when using loopback (on Debian
	lenny)
In-Reply-To: <AANLkTiniBAWw+JMe=64PtAER9B9U7rbbkJ9X27fOfYBn@mail.gmail.com>
References: <4D42B8EE.7030208@estation.dk>
	<7C6F3FC7-445F-4157-BD59-D76A705E4CE7@carmickle.com>
	<9894FEC6-E451-4993-A85A-2CD910E916B1@ipeva.fr>
	<AANLkTim+aBjRiAvxDUF1O1vut_CJpqgct2OqCNVxi0q4@mail.gmail.com>
	<AANLkTiktckovGv4Ksnn_56EzpvTTSnfG+iD1vPX1_4RM@mail.gmail.com>
	<4D466CAE.7030402@estation.dk>
	<AANLkTimhOKmy_yupGtA2HBY_8wXtQKZnDYJ9Ye76bhzq@mail.gmail.com>
	<AANLkTiktQU7kOG_22nG=_oEHSrBad667jcTzBs3n2nQv@mail.gmail.com>
	<AANLkTiko=qTZbXCL09D-+W-FmibMnBjOeoR30YYzHe0i@mail.gmail.com>
	<AANLkTi=t8AiZG6axtfqmMe9ZHTPNowoAb=7eNjQ-RPoG@mail.gmail.com>
	<AANLkTiniBAWw+JMe=64PtAER9B9U7rbbkJ9X27fOfYBn@mail.gmail.com>
Message-ID: <AANLkTinRXmR9iHbWvFRrbpxDyocD3g8hTmZ0s_mD9cyd@mail.gmail.com>

We faced same trouble with FreeSWITCH Version 1.0.head (git-33848eb
2011-02-01 08-56-26 +0100), the problem persists with loopback
channels. Even if two sip endpoints are bridged via loopback, sound is
choppy.

We have bridge_early_media=true on loopback channel bridge command and
loopback_bowout=false, loopback_bowout_on_execute=false

2011/2/2 Dan Lane <null at invalid.name>:
> This seems to have resolved the issue for us :)
>
> Thanks.
>
> On Tue, Feb 1, 2011 at 8:47 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> Try the latest GIT, I reverted the last patch and tried to solve the
>> problem differently.
>>
>>
>> On Tue, Feb 1, 2011 at 2:11 PM, Dan Lane <null at invalid.name> wrote:
>>> Yes, using 1f1541b our calls that use mod_loopback into mod_conference
>>> are unusable. The audio is choppy and delay increases as time passes.
>>>
>>> I thought it was an issue relating to the default 100Hz kernel on EC2
>>> so I spent some time yesterday putting together a 1000Hz kernel but it
>>> didn't make any difference. In the meantime I've compiled mod_loopback
>>> from 4c5426f and loaded it with my 1f1541b build which eliminates the
>>> issue.
>>>
>>> I haven't added it to Jira yet as I want to spend some time debugging
>>> it (and I also owe you some debug info for FS-2934) but the problem is
>>> definitely there.
>>>
>>> On Tue, Feb 1, 2011 at 3:58 PM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>>> Are you saying you have better results on that version than you do on
>>>> the latest?
>>>> What conditions do you have that cause you trouble, what is the
>>>> endpoint on the other side.
>>>>
>>>> If the last commit to mod_loopback intended to improve audio quality
>>>> actually makes it worse I need to investigate it.
>>>>
>>>>
>>>> On Tue, Feb 1, 2011 at 5:53 AM, Dan Lane <null at invalid.name> wrote:
>>>>> FWIW we've been also been having audio issues with loopback recently
>>>>> on EC2 (with a 1000Hz kernel).
>>>>>
>>>>> We worked around it in the short term by reverting mod_loopback to
>>>>> git-4c5426f during the build process.
>>>>>
>>>>> For anyone else who wants to try this just run "git checkout 4c5426f"
>>>>> in src/mod/endpoints/mod_loopback then build as usual. This is NOT a
>>>>> long term solution though.
>>>>>
>>>>> On Mon, Jan 31, 2011 at 8:02 AM, ?yvind Albrigtsen <oa at estation.dk> wrote:
>>>>>> Thanks for all your feedback. I'll keep on trying and inform you what
>>>>>> worked for me.
>>>>>>
>>>>>>
>>>>>> Regards
>>>>>> Oyvind
>>>>>>
>>>>>> On 2011-01-29 21:48, Anthony Minessale wrote:
>>>>>>> Everyone should try latest GIT before pondering any further because I
>>>>>>> added a patch like 2 days ago to adress this issue.
>>>>>>>
>>>>>>>
>>>>>>> On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank at telonium.com> ?wrote:
>>>>>>>
>>>>>>>> Yeah. I, too, don't see the correlation between the NTP and hw timer..
>>>>>>>> I am not familiar with the timer_test command and what it's measuring, but
>>>>>>>> of the 50 tests it ran, min is 19089 and max is 20713.
>>>>>>>> Frank
>>>>>>>>
>>>>>>>>
>>>>>>>> On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone at ipeva.fr>
>>>>>>>> wrote:
>>>>>>>>
>>>>>>>>> Frank,
>>>>>>>>> I fail to see the relationship between the hw timer and NTP.
>>>>>>>>> Can you please elaborate ?
>>>>>>>>> David Ponzone ?Direction Technique
>>>>>>>>> email: david.ponzone at ipeva.fr
>>>>>>>>> tel: ? ? ?01 74 03 18 97
>>>>>>>>> gsm: ? 06 66 98 76 34
>>>>>>>>> Service Client IPeva
>>>>>>>>> tel: ? ? ?0811 46 26 26
>>>>>>>>> www.ipeva.fr ?- ? www.ipeva-studio.com
>>>>>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>>>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>>>>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>>>>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>>>>>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>>>>>>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Le 28/01/2011 ? 19:49, Frank Carmickle a ?crit :
>>>>>>>>>
>>>>>>>>> Hi
>>>>>>>>>
>>>>>>>>> On Jan 28, 2011, at 7:39 AM, ?yvind Albrigtsen wrote:
>>>>>>>>>
>>>>>>>>> Hi,
>>>>>>>>>
>>>>>>>>> I'm using latest git-version of Freeswitch, and when I go to voicemail
>>>>>>>>>
>>>>>>>>> when calling a number the sound playback is choppy and it skips some of
>>>>>>>>>
>>>>>>>>> the digits in the number I called.
>>>>>>>>>
>>>>>>>>> What kind of results do you get from timer_test at the fs_cli? ?Are you
>>>>>>>>> running on hardware or are you virtualized? ?What is your clock source set
>>>>>>>>> to and what are your available clock source options? ?See
>>>>>>>>> /sys/devices/system/clocksource/clocksource0/available_clocksource and
>>>>>>>>> /sys/devices/system/clocksource/clocksource0/current_clocksource. ?I am
>>>>>>>>> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to
>>>>>>>>> hang at 19998/19999 which works very well for me. ?When I was having problem
>>>>>>>>> it was reporting numbers all over the map from 17400 to 22600 with lots of
>>>>>>>>> randomness in between. ?I have my clocksource set to jiffies and xen
>>>>>>>>> independent wallclock set to 1. ?Of course at that point you need to have
>>>>>>>>> ntp running against a bunch of servers to drive your clock nice and steady.
>>>>>>>>> ? I know my set up is probably a lot different than yours but I thought I'd
>>>>>>>>> toss it out there to show that some of the harshest conditions can be dealt
>>>>>>>>> with and don't give up trying. ?If you are running on hardware with a cpu
>>>>>>>>> that doesn't have constant_tsc then you might have some problems. ?Just play
>>>>>>>>> with the different timer options until you find the one that works.
>>>>>>>>>
>>>>>>>>> HTH
>>>>>>>>> --FC
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at 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
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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
>



-- 
Best regards,

Dmitry Sytchev,
IT Engineer


From kbdfck at gmail.com  Thu Feb  3 13:33:30 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Thu, 3 Feb 2011 13:33:30 +0300
Subject: [Freeswitch-users] Is it possible to use att_xfer on channels
 already bridged via loopback?
In-Reply-To: <F54ED3D2-5938-4E17-8C1D-7695342BE074@freeswitch.org>
References: <AANLkTimWAPV52349=Yge2zjnJyP40EdgDJXgyH-QcOCd@mail.gmail.com>
	<AANLkTi=WKADqDk-qqevfAh3vR1mZgZdwEEey_cuFUEz8@mail.gmail.com>
	<AANLkTimmE9oirGUznj6RTZHYkf=pWZLUg-5SHfRqbnFH@mail.gmail.com>
	<F54ED3D2-5938-4E17-8C1D-7695342BE074@freeswitch.org>
Message-ID: <AANLkTinUJVdUNdTJbrDjdvOY5CrrwBnn56sPEeLnv8Ce@mail.gmail.com>

Hi!
Any news on this? Did you succeed reproducing our bug?
Problem seems to be present in latest git too.

Also, if first att_xfer attempt is failed for any reason, when
transferer hangs up on subsequent attempts, no transfer is made, all
channels are just hungup.



2011/1/25 Brian West <brian at freeswitch.org>:
> I couldn't replicate it on mine but i'll try his configs.
> /b
> On Jan 25, 2011, at 10:01 AM, Dmitry Sytchev wrote:
>
> http://pastebin.freeswitch.org/15141?- dialplan features.xml
> http://pastebin.freeswitch.org/15140?- extraction from main dialplan
> to only test att_xfers
> http://pastebin.freeswitch.org/15138?- freeswitch.xml
>
> Do you also need my debug logs or any other info?
> My problem can be reproduced on my hardware with this configuration.
> There is a problem with MOH also - if first att_xfer fails for any
> reason, there will be no MOH for transferee on subsequent att_xfer
> calls, just silence
>
>
> 2011/1/24 Michael Collins <msc at freeswitch.org>:
>
> Dmitry,
>
> I'd like to try this myself on one of my boxes. Would you pastebin the
>
> dialplan you are using?
>
> Thanks,
>
> 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


From infos at madovsky.org  Thu Feb  3 14:05:44 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 3 Feb 2011 06:05:44 -0500
Subject: [Freeswitch-users] conference audio in last git
Message-ID: <210D70D652874D9E93576984B739281D@e1705>

seems that the audio (audio files and voice)
is less stable as before in conference only.
some packet loss or like a noise gate with a too high threeshold
also without CNG there's noise between ivr voice
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From u2nsam at gmail.com  Thu Feb  3 14:07:59 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 3 Feb 2011 16:37:59 +0530
Subject: [Freeswitch-users] context
In-Reply-To: <AANLkTikmnwQjXSX9GaC0Q3DGQYOTpv9-LE29AFW-VA4T@mail.gmail.com>
References: <AANLkTika5CBO5eVWuRimKY+Bth2YAFC0Tm8_JmFBnXwC@mail.gmail.com>
	<AANLkTim84VqQ79xq6amru_4GzvGP9rof001MJ8t2Zpvd@mail.gmail.com>
	<AANLkTinutzgJ5GF_tP=gpfVaz03+CQ-b-CCjPh3rFL=_@mail.gmail.com>
	<AANLkTikmnwQjXSX9GaC0Q3DGQYOTpv9-LE29AFW-VA4T@mail.gmail.com>
Message-ID: <AANLkTi=7jxku6ghSw8_r5ccmM=L8BH4sbb3R75WOLTiC@mail.gmail.com>

Wow,

super explanation of the call flow, if such explanations are there in the FS
book , I will take it.

Thanks again.

Regds
Sam

On Thu, Feb 3, 2011 at 12:15 PM, Michael Collins <msc at freeswitch.org> wrote:

> I think I'm going to write an article on this topic for the new folks since
> it seems to be a common point of confusion. Don't worry, you're not alone -
> this is a common challenge for new FS admins.
>
> #1 - The first line of your pastebin is the key. You are letting the call
> in via an ACL. This is "okay" if it is what you want, but I doubt that it
> is. I recommend that you remove this phone's IP address from the "domains"
> section of acl.conf.xml. When using the default FS configs, if you let a
> phone call through using the "domains" ACL then it automatically goes to the
> public context. Why? Because the call is not explicitly associated with a
> local user on the system. It's like getting a free pass into Disneyland but
> you don't have a little badge that says, "Hi, my name is..."
>
> Some get confused by the "Falling back to digest auth" message. It looks
> like an error but it's really just information. If the caller's SIP client
> properly authenticates (which is different than registration - see below)
> then FS knows exactly who is making the call and that it is a local user so
> it goes into the "default" context.
>
> #2 - Make sure that you learn the difference between SIP *registration* and
> SIP *authentication*.  Registration is where the SIP client tells FS:
> "Here's how you can reach me if you get a call for me." It is for calls TO
> the phone, not calls from the phone. On the other hand, call authentication
> is for calls FROM the phone to FS. When the phone calls FS, FS first checks
> to see if the phone's IP is in the "domains" ACL. If it is, then it just
> let's the call in to be handled by the public context. If not, then FS sends
> an "auth challenge" - basically saying, "What's the password?" If the SIP
> client properly responds then the call is "authenticated" - meaning that FS
> knows it is from a specific user and thus it goes to the default context.
>
> I recommend that you get the FS book and look at chapter 4. We talk a lot
> about the user directory and it will help you understand how it all works.
> Hopefully this explanation will whet your appetite for more. :)
>
> Keep hacking away at it - you'll get it soon enough!
>
> -MC
>
> On Wed, Feb 2, 2011 at 7:02 PM, Sam <u2nsam at gmail.com> wrote:
>
>> So you say that the extensions are not registered with FS and they are
>> getting allowed by the acl, just like DIDs.
>> How can i rectify this,any thing that needs to be checked ?
>>
>> http://pastebin.freeswitch.org/15217
>>
>> Regards
>> Sam
>>
>>
>> On Wed, Feb 2, 2011 at 11:23 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Turn on console debug level output (default in fs_cli) and make the test
>>> call. Pastebin the output. Most likely the call is not being authorized
>>> because you are letting it in via an ACL or something like that.
>>>
>>> -MC
>>>
>>> On Wed, Feb 2, 2011 at 1:55 AM, Sam <u2nsam at gmail.com> wrote:
>>>
>>>> I have defined a user as :
>>>>
>>>> <user id="2099">
>>>>       <params>
>>>>         <param name="password" value="1234"/>
>>>>         <param name="dial-string" value="{presence_id=${dialed_user}@
>>>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>>>         <param name="mwi-account" value="2099 at voicemail"/>
>>>>         <param name="vm-password" value="1234"/>
>>>>         <param name="vm-email-all-messages" value="true"/>
>>>>         <!--param name="vm-attach-file" value="true"/-->
>>>>         <!--param name="vm-mailto"  value=""/-->
>>>>       </params>
>>>>       <variables>
>>>>         <variable name="user_context" value="inter"/>
>>>>         <variable name="force_transfer_context" value="inter"/>
>>>>         <variable name="accountcode" value="2099"/>
>>>>         <variable name="internal_caller_id_name" value="2099"/>
>>>>         <variable name="internal_caller_id_number" value="2099"/>
>>>>         <variable name="callgroup" value="1"/>
>>>>         <variable name="sip-force-contact"
>>>> value="NDLB-connectile-dysfunction"/>
>>>>       </variables>
>>>>
>>>>
>>>> and in the dialplan i have
>>>>
>>>> <context name="inter">
>>>>
>>>> <extension name="2099">
>>>>     <condition field="destination_number" expression="^(2099)">
>>>>       <action application="set" data="call_timeout=30"/>
>>>>       <action application="set" data="dialed_extension=$1"/>
>>>>             <action application="export" data="dialed_extension=$1"/>
>>>>             <action application="set"
>>>> data="transfer_ringback=$${hold_music}"/>
>>>>             <action application="set" data="hangup_after_bridge=true"/>
>>>>             <action application="set" data="continue_on_fail=true"/>
>>>>             <action application="hash"
>>>> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>>>>             <action application="hash"
>>>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>>>>             <action application="hash"
>>>> data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
>>>>             <action application="hash"
>>>> data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>>>>             <action application="set"
>>>> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
>>>> var callgroup)}"/>
>>>>             <action application="hash"
>>>> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>>>>             <action application="bridge"
>>>> data="{sip_invite_domain=$${domain}}user/${dialed_extension}@
>>>> ${domain_name}"/>
>>>>     </condition>
>>>>   </extension>
>>>>
>>>> </context>
>>>>
>>>>
>>>> Now extension 2075 is also made on above lines in the same context for
>>>> user and dialplan, here the call when initiated from 2075 to 2099 searches
>>>> for context public instead of context inter
>>>>
>>>> mod_dialplan_xml.c:331 Processing 2075 <2075>->2099 in context public
>>>>
>>>> any reason why ?
>>>>
>>>> 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 u2nsam at gmail.com  Thu Feb  3 14:24:44 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 3 Feb 2011 16:54:44 +0530
Subject: [Freeswitch-users] context
In-Reply-To: <AANLkTi=7jxku6ghSw8_r5ccmM=L8BH4sbb3R75WOLTiC@mail.gmail.com>
References: <AANLkTika5CBO5eVWuRimKY+Bth2YAFC0Tm8_JmFBnXwC@mail.gmail.com>
	<AANLkTim84VqQ79xq6amru_4GzvGP9rof001MJ8t2Zpvd@mail.gmail.com>
	<AANLkTinutzgJ5GF_tP=gpfVaz03+CQ-b-CCjPh3rFL=_@mail.gmail.com>
	<AANLkTikmnwQjXSX9GaC0Q3DGQYOTpv9-LE29AFW-VA4T@mail.gmail.com>
	<AANLkTi=7jxku6ghSw8_r5ccmM=L8BH4sbb3R75WOLTiC@mail.gmail.com>
Message-ID: <AANLkTims8PQp9H2PXW_EghFZ6_Zz4p4xqk-6nQaKwi1D@mail.gmail.com>

Is FS book 1.0.7 available ?

Regds


On Thu, Feb 3, 2011 at 4:37 PM, Sam <u2nsam at gmail.com> wrote:

> Wow,
>
> super explanation of the call flow, if such explanations are there in the
> FS book , I will take it.
>
> Thanks again.
>
> Regds
> Sam
>
>
> On Thu, Feb 3, 2011 at 12:15 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> I think I'm going to write an article on this topic for the new folks
>> since it seems to be a common point of confusion. Don't worry, you're not
>> alone - this is a common challenge for new FS admins.
>>
>> #1 - The first line of your pastebin is the key. You are letting the call
>> in via an ACL. This is "okay" if it is what you want, but I doubt that it
>> is. I recommend that you remove this phone's IP address from the "domains"
>> section of acl.conf.xml. When using the default FS configs, if you let a
>> phone call through using the "domains" ACL then it automatically goes to the
>> public context. Why? Because the call is not explicitly associated with a
>> local user on the system. It's like getting a free pass into Disneyland but
>> you don't have a little badge that says, "Hi, my name is..."
>>
>> Some get confused by the "Falling back to digest auth" message. It looks
>> like an error but it's really just information. If the caller's SIP client
>> properly authenticates (which is different than registration - see below)
>> then FS knows exactly who is making the call and that it is a local user so
>> it goes into the "default" context.
>>
>> #2 - Make sure that you learn the difference between SIP *registration*
>> and SIP *authentication*.  Registration is where the SIP client tells FS:
>> "Here's how you can reach me if you get a call for me." It is for calls TO
>> the phone, not calls from the phone. On the other hand, call authentication
>> is for calls FROM the phone to FS. When the phone calls FS, FS first checks
>> to see if the phone's IP is in the "domains" ACL. If it is, then it just
>> let's the call in to be handled by the public context. If not, then FS sends
>> an "auth challenge" - basically saying, "What's the password?" If the SIP
>> client properly responds then the call is "authenticated" - meaning that FS
>> knows it is from a specific user and thus it goes to the default context.
>>
>> I recommend that you get the FS book and look at chapter 4. We talk a lot
>> about the user directory and it will help you understand how it all works.
>> Hopefully this explanation will whet your appetite for more. :)
>>
>> Keep hacking away at it - you'll get it soon enough!
>>
>> -MC
>>
>> On Wed, Feb 2, 2011 at 7:02 PM, Sam <u2nsam at gmail.com> wrote:
>>
>>> So you say that the extensions are not registered with FS and they are
>>> getting allowed by the acl, just like DIDs.
>>> How can i rectify this,any thing that needs to be checked ?
>>>
>>> http://pastebin.freeswitch.org/15217
>>>
>>> Regards
>>> Sam
>>>
>>>
>>> On Wed, Feb 2, 2011 at 11:23 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> Turn on console debug level output (default in fs_cli) and make the test
>>>> call. Pastebin the output. Most likely the call is not being authorized
>>>> because you are letting it in via an ACL or something like that.
>>>>
>>>> -MC
>>>>
>>>> On Wed, Feb 2, 2011 at 1:55 AM, Sam <u2nsam at gmail.com> wrote:
>>>>
>>>>> I have defined a user as :
>>>>>
>>>>> <user id="2099">
>>>>>       <params>
>>>>>         <param name="password" value="1234"/>
>>>>>         <param name="dial-string" value="{presence_id=${dialed_user}@
>>>>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>>>>         <param name="mwi-account" value="2099 at voicemail"/>
>>>>>         <param name="vm-password" value="1234"/>
>>>>>         <param name="vm-email-all-messages" value="true"/>
>>>>>         <!--param name="vm-attach-file" value="true"/-->
>>>>>         <!--param name="vm-mailto"  value=""/-->
>>>>>       </params>
>>>>>       <variables>
>>>>>         <variable name="user_context" value="inter"/>
>>>>>         <variable name="force_transfer_context" value="inter"/>
>>>>>         <variable name="accountcode" value="2099"/>
>>>>>         <variable name="internal_caller_id_name" value="2099"/>
>>>>>         <variable name="internal_caller_id_number" value="2099"/>
>>>>>         <variable name="callgroup" value="1"/>
>>>>>         <variable name="sip-force-contact"
>>>>> value="NDLB-connectile-dysfunction"/>
>>>>>       </variables>
>>>>>
>>>>>
>>>>> and in the dialplan i have
>>>>>
>>>>> <context name="inter">
>>>>>
>>>>> <extension name="2099">
>>>>>     <condition field="destination_number" expression="^(2099)">
>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>       <action application="set" data="dialed_extension=$1"/>
>>>>>             <action application="export" data="dialed_extension=$1"/>
>>>>>             <action application="set"
>>>>> data="transfer_ringback=$${hold_music}"/>
>>>>>             <action application="set" data="hangup_after_bridge=true"/>
>>>>>             <action application="set" data="continue_on_fail=true"/>
>>>>>             <action application="hash"
>>>>> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>>>>>             <action application="hash"
>>>>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>>>>>             <action application="hash"
>>>>> data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
>>>>>             <action application="hash"
>>>>> data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>>>>>             <action application="set"
>>>>> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
>>>>> var callgroup)}"/>
>>>>>             <action application="hash"
>>>>> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>>>>>             <action application="bridge"
>>>>> data="{sip_invite_domain=$${domain}}user/${dialed_extension}@
>>>>> ${domain_name}"/>
>>>>>     </condition>
>>>>>   </extension>
>>>>>
>>>>> </context>
>>>>>
>>>>>
>>>>> Now extension 2075 is also made on above lines in the same context for
>>>>> user and dialplan, here the call when initiated from 2075 to 2099 searches
>>>>> for context public instead of context inter
>>>>>
>>>>> mod_dialplan_xml.c:331 Processing 2075 <2075>->2099 in context public
>>>>>
>>>>> any reason why ?
>>>>>
>>>>> 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 steveayre at gmail.com  Thu Feb  3 14:46:07 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 3 Feb 2011 11:46:07 +0000
Subject: [Freeswitch-users] context
In-Reply-To: <AANLkTims8PQp9H2PXW_EghFZ6_Zz4p4xqk-6nQaKwi1D@mail.gmail.com>
References: <AANLkTika5CBO5eVWuRimKY+Bth2YAFC0Tm8_JmFBnXwC@mail.gmail.com>
	<AANLkTim84VqQ79xq6amru_4GzvGP9rof001MJ8t2Zpvd@mail.gmail.com>
	<AANLkTinutzgJ5GF_tP=gpfVaz03+CQ-b-CCjPh3rFL=_@mail.gmail.com>
	<AANLkTikmnwQjXSX9GaC0Q3DGQYOTpv9-LE29AFW-VA4T@mail.gmail.com>
	<AANLkTi=7jxku6ghSw8_r5ccmM=L8BH4sbb3R75WOLTiC@mail.gmail.com>
	<AANLkTims8PQp9H2PXW_EghFZ6_Zz4p4xqk-6nQaKwi1D@mail.gmail.com>
Message-ID: <AANLkTi=MuV-Pi26aV7SphZ6WmmOY+ftGQY_DfCcww-0d@mail.gmail.com>

No, and it never will be... 1.0.7 is the daily snapshot release candidate
for the 1.0.8 release. The book might be updated for 1.0.8 once it comes out
though.

-Steve


On 3 February 2011 11:24, Sam <u2nsam at gmail.com> wrote:

> Is FS book 1.0.7 available ?
>
> Regds
>
>
>
> On Thu, Feb 3, 2011 at 4:37 PM, Sam <u2nsam at gmail.com> wrote:
>
>> Wow,
>>
>> super explanation of the call flow, if such explanations are there in the
>> FS book , I will take it.
>>
>> Thanks again.
>>
>> Regds
>> Sam
>>
>>
>> On Thu, Feb 3, 2011 at 12:15 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> I think I'm going to write an article on this topic for the new folks
>>> since it seems to be a common point of confusion. Don't worry, you're not
>>> alone - this is a common challenge for new FS admins.
>>>
>>> #1 - The first line of your pastebin is the key. You are letting the call
>>> in via an ACL. This is "okay" if it is what you want, but I doubt that it
>>> is. I recommend that you remove this phone's IP address from the "domains"
>>> section of acl.conf.xml. When using the default FS configs, if you let a
>>> phone call through using the "domains" ACL then it automatically goes to the
>>> public context. Why? Because the call is not explicitly associated with a
>>> local user on the system. It's like getting a free pass into Disneyland but
>>> you don't have a little badge that says, "Hi, my name is..."
>>>
>>> Some get confused by the "Falling back to digest auth" message. It looks
>>> like an error but it's really just information. If the caller's SIP client
>>> properly authenticates (which is different than registration - see below)
>>> then FS knows exactly who is making the call and that it is a local user so
>>> it goes into the "default" context.
>>>
>>> #2 - Make sure that you learn the difference between SIP *registration*
>>> and SIP *authentication*.  Registration is where the SIP client tells FS:
>>> "Here's how you can reach me if you get a call for me." It is for calls TO
>>> the phone, not calls from the phone. On the other hand, call authentication
>>> is for calls FROM the phone to FS. When the phone calls FS, FS first checks
>>> to see if the phone's IP is in the "domains" ACL. If it is, then it just
>>> let's the call in to be handled by the public context. If not, then FS sends
>>> an "auth challenge" - basically saying, "What's the password?" If the SIP
>>> client properly responds then the call is "authenticated" - meaning that FS
>>> knows it is from a specific user and thus it goes to the default context.
>>>
>>> I recommend that you get the FS book and look at chapter 4. We talk a lot
>>> about the user directory and it will help you understand how it all works.
>>> Hopefully this explanation will whet your appetite for more. :)
>>>
>>> Keep hacking away at it - you'll get it soon enough!
>>>
>>> -MC
>>>
>>> On Wed, Feb 2, 2011 at 7:02 PM, Sam <u2nsam at gmail.com> wrote:
>>>
>>>> So you say that the extensions are not registered with FS and they are
>>>> getting allowed by the acl, just like DIDs.
>>>> How can i rectify this,any thing that needs to be checked ?
>>>>
>>>> http://pastebin.freeswitch.org/15217
>>>>
>>>> Regards
>>>> Sam
>>>>
>>>>
>>>> On Wed, Feb 2, 2011 at 11:23 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>> Turn on console debug level output (default in fs_cli) and make the
>>>>> test call. Pastebin the output. Most likely the call is not being authorized
>>>>> because you are letting it in via an ACL or something like that.
>>>>>
>>>>> -MC
>>>>>
>>>>> On Wed, Feb 2, 2011 at 1:55 AM, Sam <u2nsam at gmail.com> wrote:
>>>>>
>>>>>> I have defined a user as :
>>>>>>
>>>>>> <user id="2099">
>>>>>>       <params>
>>>>>>         <param name="password" value="1234"/>
>>>>>>         <param name="dial-string" value="{presence_id=${dialed_user}@
>>>>>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>>>>>         <param name="mwi-account" value="2099 at voicemail"/>
>>>>>>         <param name="vm-password" value="1234"/>
>>>>>>         <param name="vm-email-all-messages" value="true"/>
>>>>>>         <!--param name="vm-attach-file" value="true"/-->
>>>>>>         <!--param name="vm-mailto"  value=""/-->
>>>>>>       </params>
>>>>>>       <variables>
>>>>>>         <variable name="user_context" value="inter"/>
>>>>>>         <variable name="force_transfer_context" value="inter"/>
>>>>>>         <variable name="accountcode" value="2099"/>
>>>>>>         <variable name="internal_caller_id_name" value="2099"/>
>>>>>>         <variable name="internal_caller_id_number" value="2099"/>
>>>>>>         <variable name="callgroup" value="1"/>
>>>>>>         <variable name="sip-force-contact"
>>>>>> value="NDLB-connectile-dysfunction"/>
>>>>>>       </variables>
>>>>>>
>>>>>>
>>>>>> and in the dialplan i have
>>>>>>
>>>>>> <context name="inter">
>>>>>>
>>>>>> <extension name="2099">
>>>>>>     <condition field="destination_number" expression="^(2099)">
>>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>>       <action application="set" data="dialed_extension=$1"/>
>>>>>>             <action application="export" data="dialed_extension=$1"/>
>>>>>>             <action application="set"
>>>>>> data="transfer_ringback=$${hold_music}"/>
>>>>>>             <action application="set"
>>>>>> data="hangup_after_bridge=true"/>
>>>>>>             <action application="set" data="continue_on_fail=true"/>
>>>>>>             <action application="hash"
>>>>>> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>>>>>>             <action application="hash"
>>>>>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>>>>>>             <action application="hash"
>>>>>> data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
>>>>>>             <action application="hash"
>>>>>> data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>>>>>>             <action application="set"
>>>>>> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
>>>>>> var callgroup)}"/>
>>>>>>             <action application="hash"
>>>>>> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>>>>>>             <action application="bridge"
>>>>>> data="{sip_invite_domain=$${domain}}user/${dialed_extension}@
>>>>>> ${domain_name}"/>
>>>>>>     </condition>
>>>>>>   </extension>
>>>>>>
>>>>>> </context>
>>>>>>
>>>>>>
>>>>>> Now extension 2075 is also made on above lines in the same context for
>>>>>> user and dialplan, here the call when initiated from 2075 to 2099 searches
>>>>>> for context public instead of context inter
>>>>>>
>>>>>> mod_dialplan_xml.c:331 Processing 2075 <2075>->2099 in context public
>>>>>>
>>>>>> any reason why ?
>>>>>>
>>>>>> 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
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  3 14:46:45 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 3 Feb 2011 11:46:45 +0000
Subject: [Freeswitch-users] context
In-Reply-To: <AANLkTi=MuV-Pi26aV7SphZ6WmmOY+ftGQY_DfCcww-0d@mail.gmail.com>
References: <AANLkTika5CBO5eVWuRimKY+Bth2YAFC0Tm8_JmFBnXwC@mail.gmail.com>
	<AANLkTim84VqQ79xq6amru_4GzvGP9rof001MJ8t2Zpvd@mail.gmail.com>
	<AANLkTinutzgJ5GF_tP=gpfVaz03+CQ-b-CCjPh3rFL=_@mail.gmail.com>
	<AANLkTikmnwQjXSX9GaC0Q3DGQYOTpv9-LE29AFW-VA4T@mail.gmail.com>
	<AANLkTi=7jxku6ghSw8_r5ccmM=L8BH4sbb3R75WOLTiC@mail.gmail.com>
	<AANLkTims8PQp9H2PXW_EghFZ6_Zz4p4xqk-6nQaKwi1D@mail.gmail.com>
	<AANLkTi=MuV-Pi26aV7SphZ6WmmOY+ftGQY_DfCcww-0d@mail.gmail.com>
Message-ID: <AANLkTikVRyfQqJbKXH=aw0MCKnzzyyOORzsENxPnJPXm@mail.gmail.com>

Everything in the 1.0.6 book should work on 1.0.7 though...

It'll only be missing some of the new features, but they're on the Wiki

-Steve


On 3 February 2011 11:46, Steven Ayre <steveayre at gmail.com> wrote:

> No, and it never will be... 1.0.7 is the daily snapshot release candidate
> for the 1.0.8 release. The book might be updated for 1.0.8 once it comes out
> though.
>
> -Steve
>
>
> On 3 February 2011 11:24, Sam <u2nsam at gmail.com> wrote:
>
>> Is FS book 1.0.7 available ?
>>
>> Regds
>>
>>
>>
>> On Thu, Feb 3, 2011 at 4:37 PM, Sam <u2nsam at gmail.com> wrote:
>>
>>> Wow,
>>>
>>> super explanation of the call flow, if such explanations are there in the
>>> FS book , I will take it.
>>>
>>> Thanks again.
>>>
>>> Regds
>>> Sam
>>>
>>>
>>> On Thu, Feb 3, 2011 at 12:15 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> I think I'm going to write an article on this topic for the new folks
>>>> since it seems to be a common point of confusion. Don't worry, you're not
>>>> alone - this is a common challenge for new FS admins.
>>>>
>>>> #1 - The first line of your pastebin is the key. You are letting the
>>>> call in via an ACL. This is "okay" if it is what you want, but I doubt that
>>>> it is. I recommend that you remove this phone's IP address from the
>>>> "domains" section of acl.conf.xml. When using the default FS configs, if you
>>>> let a phone call through using the "domains" ACL then it automatically goes
>>>> to the public context. Why? Because the call is not explicitly associated
>>>> with a local user on the system. It's like getting a free pass into
>>>> Disneyland but you don't have a little badge that says, "Hi, my name is..."
>>>>
>>>> Some get confused by the "Falling back to digest auth" message. It looks
>>>> like an error but it's really just information. If the caller's SIP client
>>>> properly authenticates (which is different than registration - see below)
>>>> then FS knows exactly who is making the call and that it is a local user so
>>>> it goes into the "default" context.
>>>>
>>>> #2 - Make sure that you learn the difference between SIP *registration*
>>>> and SIP *authentication*.  Registration is where the SIP client tells FS:
>>>> "Here's how you can reach me if you get a call for me." It is for calls TO
>>>> the phone, not calls from the phone. On the other hand, call authentication
>>>> is for calls FROM the phone to FS. When the phone calls FS, FS first checks
>>>> to see if the phone's IP is in the "domains" ACL. If it is, then it just
>>>> let's the call in to be handled by the public context. If not, then FS sends
>>>> an "auth challenge" - basically saying, "What's the password?" If the SIP
>>>> client properly responds then the call is "authenticated" - meaning that FS
>>>> knows it is from a specific user and thus it goes to the default context.
>>>>
>>>> I recommend that you get the FS book and look at chapter 4. We talk a
>>>> lot about the user directory and it will help you understand how it all
>>>> works. Hopefully this explanation will whet your appetite for more. :)
>>>>
>>>> Keep hacking away at it - you'll get it soon enough!
>>>>
>>>> -MC
>>>>
>>>> On Wed, Feb 2, 2011 at 7:02 PM, Sam <u2nsam at gmail.com> wrote:
>>>>
>>>>> So you say that the extensions are not registered with FS and they are
>>>>> getting allowed by the acl, just like DIDs.
>>>>> How can i rectify this,any thing that needs to be checked ?
>>>>>
>>>>> http://pastebin.freeswitch.org/15217
>>>>>
>>>>> Regards
>>>>> Sam
>>>>>
>>>>>
>>>>> On Wed, Feb 2, 2011 at 11:23 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>> Turn on console debug level output (default in fs_cli) and make the
>>>>>> test call. Pastebin the output. Most likely the call is not being authorized
>>>>>> because you are letting it in via an ACL or something like that.
>>>>>>
>>>>>> -MC
>>>>>>
>>>>>> On Wed, Feb 2, 2011 at 1:55 AM, Sam <u2nsam at gmail.com> wrote:
>>>>>>
>>>>>>> I have defined a user as :
>>>>>>>
>>>>>>> <user id="2099">
>>>>>>>       <params>
>>>>>>>         <param name="password" value="1234"/>
>>>>>>>         <param name="dial-string" value="{presence_id=${dialed_user}@
>>>>>>> ${dialed_domain}}${sofia_contact(${dialed_user}@
>>>>>>> ${dialed_domain})}"/>
>>>>>>>         <param name="mwi-account" value="2099 at voicemail"/>
>>>>>>>         <param name="vm-password" value="1234"/>
>>>>>>>         <param name="vm-email-all-messages" value="true"/>
>>>>>>>         <!--param name="vm-attach-file" value="true"/-->
>>>>>>>         <!--param name="vm-mailto"  value=""/-->
>>>>>>>       </params>
>>>>>>>       <variables>
>>>>>>>         <variable name="user_context" value="inter"/>
>>>>>>>         <variable name="force_transfer_context" value="inter"/>
>>>>>>>         <variable name="accountcode" value="2099"/>
>>>>>>>         <variable name="internal_caller_id_name" value="2099"/>
>>>>>>>         <variable name="internal_caller_id_number" value="2099"/>
>>>>>>>         <variable name="callgroup" value="1"/>
>>>>>>>         <variable name="sip-force-contact"
>>>>>>> value="NDLB-connectile-dysfunction"/>
>>>>>>>       </variables>
>>>>>>>
>>>>>>>
>>>>>>> and in the dialplan i have
>>>>>>>
>>>>>>> <context name="inter">
>>>>>>>
>>>>>>> <extension name="2099">
>>>>>>>     <condition field="destination_number" expression="^(2099)">
>>>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>>>       <action application="set" data="dialed_extension=$1"/>
>>>>>>>             <action application="export" data="dialed_extension=$1"/>
>>>>>>>             <action application="set"
>>>>>>> data="transfer_ringback=$${hold_music}"/>
>>>>>>>             <action application="set"
>>>>>>> data="hangup_after_bridge=true"/>
>>>>>>>             <action application="set" data="continue_on_fail=true"/>
>>>>>>>             <action application="hash"
>>>>>>> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>>>>>>>             <action application="hash"
>>>>>>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>>>>>>>             <action application="hash"
>>>>>>> data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
>>>>>>>             <action application="hash"
>>>>>>> data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>>>>>>>             <action application="set"
>>>>>>> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
>>>>>>> var callgroup)}"/>
>>>>>>>             <action application="hash"
>>>>>>> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>>>>>>>             <action application="bridge"
>>>>>>> data="{sip_invite_domain=$${domain}}user/${dialed_extension}@
>>>>>>> ${domain_name}"/>
>>>>>>>     </condition>
>>>>>>>   </extension>
>>>>>>>
>>>>>>> </context>
>>>>>>>
>>>>>>>
>>>>>>> Now extension 2075 is also made on above lines in the same context
>>>>>>> for user and dialplan, here the call when initiated from 2075 to 2099
>>>>>>> searches for context public instead of context inter
>>>>>>>
>>>>>>> mod_dialplan_xml.c:331 Processing 2075 <2075>->2099 in context public
>>>>>>>
>>>>>>> any reason why ?
>>>>>>>
>>>>>>> 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
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From christian at yellox.de  Thu Feb  3 15:24:24 2011
From: christian at yellox.de (Christian Hiller)
Date: Thu, 03 Feb 2011 13:24:24 +0100
Subject: [Freeswitch-users] ${domain} not working
In-Reply-To: <mailman.3048.1296733629.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.3048.1296733629.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <4D4A9E78.3090305@yellox.de>

Hello,

from CLI > eval ${domain} is returning -ERR with latest git (FreeSWITCH 
Version 1.0.head (git-f60fdf6 2011-02-02 16-22-43 -0600)). Using this 
within a regexp is crashing FS.

Kind regards

Christian


From massimiliano.ravelli at gmail.com  Thu Feb  3 14:38:09 2011
From: massimiliano.ravelli at gmail.com (Massimiliano Ravelli)
Date: Thu, 3 Feb 2011 12:38:09 +0100
Subject: [Freeswitch-users] mod_com_g729 - moving licences to a new
	server
In-Reply-To: <AANLkTikpKu-T77QNvAGQevDZSOsncyQAYmG6v9=_1OV6@mail.gmail.com>
References: <AANLkTikykPgaD0tErbMw=VOgfXqnwEwY4SzC_bNqSw-i@mail.gmail.com>
	<AANLkTinSROc3ETqi8vbs89swryJa1+TNKa8WWDP4_O6H@mail.gmail.com>
	<AANLkTinPmSAR=OiatFKEP0sRVA6_4=DQ7C5UdQcxsTfM@mail.gmail.com>
	<AANLkTi=oNazw3U-nZ63Ro2oJh+Jgz_3O4_mrFmzS8VZA@mail.gmail.com>
	<AANLkTikpKu-T77QNvAGQevDZSOsncyQAYmG6v9=_1OV6@mail.gmail.com>
Message-ID: <AANLkTikk85LxLYBV5pJ3dK7_MeCkFh9oSmM0BRAe-ewC@mail.gmail.com>

2011/2/3 Steven Ayre <steveayre at gmail.com
> You should be able to buy licenses and they won't be activated until you
install them on your customers' servers.

Oh yes, that's true ! :-)

Thanks very much indeed for help !

Massimiliano
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From Nabble at slickdeals.endjunk.com  Thu Feb  3 17:06:58 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 3 Feb 2011 06:06:58 -0800 (PST)
Subject: [Freeswitch-users] mod_com_g729 - moving licences to a new
	server
In-Reply-To: <AANLkTik0Q70zJD1z2gZ6wp1aOHKcrWvT=B-Anv8hEyzk@mail.gmail.com>
References: <AANLkTikykPgaD0tErbMw=VOgfXqnwEwY4SzC_bNqSw-i@mail.gmail.com>
	<AANLkTinSROc3ETqi8vbs89swryJa1+TNKa8WWDP4_O6H@mail.gmail.com>
	<AANLkTinPmSAR=OiatFKEP0sRVA6_4=DQ7C5UdQcxsTfM@mail.gmail.com>
	<1296695964562-5987200.post@n2.nabble.com>
	<AANLkTik0Q70zJD1z2gZ6wp1aOHKcrWvT=B-Anv8hEyzk@mail.gmail.com>
Message-ID: <1296742018936-5988951.post@n2.nabble.com>



Steven Ayre wrote:
> 
> That's what's been indicated by Brian in the past... though it could
> always
> be linked to other hardware details too.
I have no way to confirm this with other vendors, but it sounded like this
only applies to a G729 license bought from FS.

-----
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/mod-com-g729-moving-licences-to-a-new-server-tp5985791p5988951.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From johns1433 at gmail.com  Thu Feb  3 17:43:58 2011
From: johns1433 at gmail.com (John Smith)
Date: Thu, 3 Feb 2011 15:43:58 +0100
Subject: [Freeswitch-users] Simple_conference.lua
Message-ID: <AANLkTi=LRS-DC8Ft1A2Xev70FL=5MU6KGj0p7W5UO2C3@mail.gmail.com>

 Hi,

I?m trying to have the script
http://wiki.freeswitch.org/wiki/Simple_conference.lua running on my
FreeSwitch server.
I get an error when FS wants to play any phrase given in this example:
*[ERR] switch_ivr_play_say.c:150 Can't find macro conference_welcome*
*
*It seems normal as I didn?t find any reference to such phrases in the
different xml phrase configuration files. There seems also to be no adequate
wav file in sounds/en/us/callie/voicemail/8000.

My question is:
Do these files exist somewhere or does the example assume that we have to
record our own wav files?

Thanks

John
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From steveayre at gmail.com  Thu Feb  3 18:13:36 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 3 Feb 2011 15:13:36 +0000
Subject: [Freeswitch-users] mod_com_g729 - moving licences to a new
	server
In-Reply-To: <1296742018936-5988951.post@n2.nabble.com>
References: <AANLkTikykPgaD0tErbMw=VOgfXqnwEwY4SzC_bNqSw-i@mail.gmail.com>
	<AANLkTinSROc3ETqi8vbs89swryJa1+TNKa8WWDP4_O6H@mail.gmail.com>
	<AANLkTinPmSAR=OiatFKEP0sRVA6_4=DQ7C5UdQcxsTfM@mail.gmail.com>
	<1296695964562-5987200.post@n2.nabble.com>
	<AANLkTik0Q70zJD1z2gZ6wp1aOHKcrWvT=B-Anv8hEyzk@mail.gmail.com>
	<1296742018936-5988951.post@n2.nabble.com>
Message-ID: <AANLkTimfKnnOw8j6w-wpQm4DxmKLwc3ALgds3nKcE7it@mail.gmail.com>

Yes, it is specifically their G729 license implementation. They're required
by the patent holders to use some method to prevent you using the same
licence on other servers.

Other people will use various methods but many will probably go for the
MACs. It's a pretty common method that works on any OS.

It's the same one that Digium use for their G729 codec:
http://downloads.digium.com/pub/telephony/codec_g729/README

  "It is extremely important that you backup all of the files located in the

  /var/lib/asterisk/licenses directory. This directory contains the Host-ID
  specific license files for your system. *****These license files are
tied to the
  MAC address of all the ethernet devices installed in your
system*****. Creating a

  backup of this directory will allow you to restore your G.729 license file
  in case you need to reinstall your operating system. This will help prevent
  the need to contact Digium to request authorization to increment your G.729

  key and from needing to purchase a new G.729 key if you exceed the maximum
  number of G.729 key increments allowed."

(That doesn't mean you can use Digium G729 licenses on FS btw - you can't).

-Steve



On 3 February 2011 14:06, mazilo <Nabble at slickdeals.endjunk.com> wrote:

>
>
> Steven Ayre wrote:
> >
> > That's what's been indicated by Brian in the past... though it could
> > always
> > be linked to other hardware details too.
> I have no way to confirm this with other vendors, but it sounded like this
> only applies to a G729 license bought from FS.
>
> -----
> 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/mod-com-g729-moving-licences-to-a-new-server-tp5985791p5988951.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 Feb  3 18:14:25 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 3 Feb 2011 15:14:25 +0000
Subject: [Freeswitch-users] mod_com_g729 - moving licences to a new
	server
In-Reply-To: <1296742018936-5988951.post@n2.nabble.com>
References: <AANLkTikykPgaD0tErbMw=VOgfXqnwEwY4SzC_bNqSw-i@mail.gmail.com>
	<AANLkTinSROc3ETqi8vbs89swryJa1+TNKa8WWDP4_O6H@mail.gmail.com>
	<AANLkTinPmSAR=OiatFKEP0sRVA6_4=DQ7C5UdQcxsTfM@mail.gmail.com>
	<1296695964562-5987200.post@n2.nabble.com>
	<AANLkTik0Q70zJD1z2gZ6wp1aOHKcrWvT=B-Anv8hEyzk@mail.gmail.com>
	<1296742018936-5988951.post@n2.nabble.com>
Message-ID: <AANLkTi=+PtkgYewKLTCPOFToD=QQXuSggz7KDMT=GBfz@mail.gmail.com>

Oh, and it doesn't apply to Sangoma - the G729 licence is part of the
hardware costs in that case.

-Steve



On 3 February 2011 14:06, mazilo <Nabble at slickdeals.endjunk.com> wrote:

>
>
> Steven Ayre wrote:
> >
> > That's what's been indicated by Brian in the past... though it could
> > always
> > be linked to other hardware details too.
> I have no way to confirm this with other vendors, but it sounded like this
> only applies to a G729 license bought from FS.
>
> -----
> 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/mod-com-g729-moving-licences-to-a-new-server-tp5985791p5988951.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 Nabble at slickdeals.endjunk.com  Thu Feb  3 18:55:57 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 3 Feb 2011 07:55:57 -0800 (PST)
Subject: [Freeswitch-users] mod_com_g729 - moving licences to a new
	server
In-Reply-To: <AANLkTimfKnnOw8j6w-wpQm4DxmKLwc3ALgds3nKcE7it@mail.gmail.com>
References: <AANLkTikykPgaD0tErbMw=VOgfXqnwEwY4SzC_bNqSw-i@mail.gmail.com>
	<AANLkTinSROc3ETqi8vbs89swryJa1+TNKa8WWDP4_O6H@mail.gmail.com>
	<AANLkTinPmSAR=OiatFKEP0sRVA6_4=DQ7C5UdQcxsTfM@mail.gmail.com>
	<1296695964562-5987200.post@n2.nabble.com>
	<AANLkTik0Q70zJD1z2gZ6wp1aOHKcrWvT=B-Anv8hEyzk@mail.gmail.com>
	<1296742018936-5988951.post@n2.nabble.com>
	<AANLkTimfKnnOw8j6w-wpQm4DxmKLwc3ALgds3nKcE7it@mail.gmail.com>
Message-ID: <1296748557358-5989291.post@n2.nabble.com>



Steven Ayre wrote:
> Yes, it is specifically their G729 license implementation.
So, if we bought a G729 license from FS (not from G729 directly and/or other
G720 license resellers), then we can basically use the G729 license on other
device provided we manage to change the MAC Address of the new device to an
old one and remove the same G729 license from the old device. This is cool,
especially if one wants to upgrade to a new system and/or if the old system
crashes.

-----
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/mod-com-g729-moving-licences-to-a-new-server-tp5985791p5989291.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Thu Feb  3 19:21:57 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 3 Feb 2011 10:21:57 -0600
Subject: [Freeswitch-users] ${domain} not working
In-Reply-To: <4D4A9E78.3090305@yellox.de>
References: <mailman.3048.1296733629.24555.freeswitch-users@lists.freeswitch.org>
	<4D4A9E78.3090305@yellox.de>
Message-ID: <AANLkTinsbXcdCFv6FDc+1vG_kfUJb24HOf6vsw913PDQ@mail.gmail.com>

that always would happen.  We don't expand variables.
Try "make current" to rebuild clean.


On Thu, Feb 3, 2011 at 6:24 AM, Christian Hiller <christian at yellox.de> wrote:
> Hello,
>
> from CLI > eval ${domain} is returning -ERR with latest git (FreeSWITCH
> Version 1.0.head (git-f60fdf6 2011-02-02 16-22-43 -0600)). Using this
> within a regexp is crashing FS.
>
> Kind 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
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 rajesh.npnr at yahoo.com  Thu Feb  3 19:39:46 2011
From: rajesh.npnr at yahoo.com (rex.alex)
Date: Thu, 3 Feb 2011 08:39:46 -0800 (PST)
Subject: [Freeswitch-users] Grandstream Freeze on Originate command
Message-ID: <1296751186605-5989433.post@n2.nabble.com>


Hi,

I am using the api originate command to establish an internal extn to extn
call (originate sofia/internal/2000%10.10.22.31 1000), where the 2000
registered in Grandstream GXP280 1.2.1.4 and 1000 registered in X-Lite. 

This command freezes the grandstream when I answer. I have pastebin the sip
trace + freeswitch log in below mentioned url. Request your assistance in
resolving this.


http://pastebin.freeswitch.org/15222

Thanks,
Regards,
Rex
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Grandstream-Freeze-on-Originate-command-tp5989433p5989433.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mitch.capper at gmail.com  Thu Feb  3 20:22:46 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Thu, 3 Feb 2011 09:22:46 -0800
Subject: [Freeswitch-users] mod_com_g729 - moving licences to a new
	server
In-Reply-To: <1296748557358-5989291.post@n2.nabble.com>
References: <AANLkTikykPgaD0tErbMw=VOgfXqnwEwY4SzC_bNqSw-i@mail.gmail.com>
	<AANLkTinSROc3ETqi8vbs89swryJa1+TNKa8WWDP4_O6H@mail.gmail.com>
	<AANLkTinPmSAR=OiatFKEP0sRVA6_4=DQ7C5UdQcxsTfM@mail.gmail.com>
	<1296695964562-5987200.post@n2.nabble.com>
	<AANLkTik0Q70zJD1z2gZ6wp1aOHKcrWvT=B-Anv8hEyzk@mail.gmail.com>
	<1296742018936-5988951.post@n2.nabble.com>
	<AANLkTimfKnnOw8j6w-wpQm4DxmKLwc3ALgds3nKcE7it@mail.gmail.com>
	<1296748557358-5989291.post@n2.nabble.com>
Message-ID: <AANLkTikqb_J3H9Dt1+XZ2xYmGc3zuGn1mQiOTO9SC_9X@mail.gmail.com>

It sounds like however the license uses all network cards on the machine, so
while changing macs may work it will probably only work if there are the
same number of network cards.

~Mitch

On Thu, Feb 3, 2011 at 7:55 AM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
>
> Steven Ayre wrote:
> > Yes, it is specifically their G729 license implementation.
> So, if we bought a G729 license from FS (not from G729 directly and/or
> other
> G720 license resellers), then we can basically use the G729 license on
> other
> device provided we manage to change the MAC Address of the new device to an
> old one and remove the same G729 license from the old device. This is cool,
> especially if one wants to upgrade to a new system and/or if the old system
> crashes.
>
> -----
> 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/mod-com-g729-moving-licences-to-a-new-server-tp5985791p5989291.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 Feb  3 20:23:58 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 3 Feb 2011 17:23:58 +0000
Subject: [Freeswitch-users] ${domain} not working
In-Reply-To: <AANLkTinsbXcdCFv6FDc+1vG_kfUJb24HOf6vsw913PDQ@mail.gmail.com>
References: <mailman.3048.1296733629.24555.freeswitch-users@lists.freeswitch.org>
	<4D4A9E78.3090305@yellox.de>
	<AANLkTinsbXcdCFv6FDc+1vG_kfUJb24HOf6vsw913PDQ@mail.gmail.com>
Message-ID: <AANLkTimpAw0tZhCMErpWE6w19nfey02zdx2s+M8NC4PG@mail.gmail.com>

Anthony,

Does that mean the Wiki is wrong? It gives that as an example.
http://wiki.freeswitch.org/wiki/Mod_commands#eval

-Steve


On 3 February 2011 16:21, Anthony Minessale <anthony.minessale at gmail.com>wrote:

> that always would happen.  We don't expand variables.
> Try "make current" to rebuild clean.
>
>
> On Thu, Feb 3, 2011 at 6:24 AM, Christian Hiller <christian at yellox.de>
> wrote:
> > Hello,
> >
> > from CLI > eval ${domain} is returning -ERR with latest git (FreeSWITCH
> > Version 1.0.head (git-f60fdf6 2011-02-02 16-22-43 -0600)). Using this
> > within a regexp is crashing FS.
> >
> > Kind 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
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  3 20:31:34 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 3 Feb 2011 11:31:34 -0600
Subject: [Freeswitch-users] ${domain} not working
In-Reply-To: <AANLkTimpAw0tZhCMErpWE6w19nfey02zdx2s+M8NC4PG@mail.gmail.com>
References: <mailman.3048.1296733629.24555.freeswitch-users@lists.freeswitch.org>
	<4D4A9E78.3090305@yellox.de>
	<AANLkTinsbXcdCFv6FDc+1vG_kfUJb24HOf6vsw913PDQ@mail.gmail.com>
	<AANLkTimpAw0tZhCMErpWE6w19nfey02zdx2s+M8NC4PG@mail.gmail.com>
Message-ID: <AANLkTimk-PsnWCUuXYeybfu2_rL3qzFtYrAdnfhECQvV@mail.gmail.com>

no I missed the "eval" in his first email.


this is a regression, i fixed it already in latest git.


On Thu, Feb 3, 2011 at 11:23 AM, Steven Ayre <steveayre at gmail.com> wrote:
> Anthony,
>
> Does that mean the Wiki is wrong? It gives that as an example.
> http://wiki.freeswitch.org/wiki/Mod_commands#eval
>
> -Steve
>
>
> On 3 February 2011 16:21, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
>>
>> that always would happen. ?We don't expand variables.
>> Try "make current" to rebuild clean.
>>
>>
>> On Thu, Feb 3, 2011 at 6:24 AM, Christian Hiller <christian at yellox.de>
>> wrote:
>> > Hello,
>> >
>> > from CLI > eval ${domain} is returning -ERR with latest git (FreeSWITCH
>> > Version 1.0.head (git-f60fdf6 2011-02-02 16-22-43 -0600)). Using this
>> > within a regexp is crashing FS.
>> >
>> > Kind 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
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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  Thu Feb  3 21:14:34 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 3 Feb 2011 12:14:34 -0600
Subject: [Freeswitch-users] Choppy VM when using loopback (on Debian
	lenny)
In-Reply-To: <AANLkTinRXmR9iHbWvFRrbpxDyocD3g8hTmZ0s_mD9cyd@mail.gmail.com>
References: <4D42B8EE.7030208@estation.dk>
	<7C6F3FC7-445F-4157-BD59-D76A705E4CE7@carmickle.com>
	<9894FEC6-E451-4993-A85A-2CD910E916B1@ipeva.fr>
	<AANLkTim+aBjRiAvxDUF1O1vut_CJpqgct2OqCNVxi0q4@mail.gmail.com>
	<AANLkTiktckovGv4Ksnn_56EzpvTTSnfG+iD1vPX1_4RM@mail.gmail.com>
	<4D466CAE.7030402@estation.dk>
	<AANLkTimhOKmy_yupGtA2HBY_8wXtQKZnDYJ9Ye76bhzq@mail.gmail.com>
	<AANLkTiktQU7kOG_22nG=_oEHSrBad667jcTzBs3n2nQv@mail.gmail.com>
	<AANLkTiko=qTZbXCL09D-+W-FmibMnBjOeoR30YYzHe0i@mail.gmail.com>
	<AANLkTi=t8AiZG6axtfqmMe9ZHTPNowoAb=7eNjQ-RPoG@mail.gmail.com>
	<AANLkTiniBAWw+JMe=64PtAER9B9U7rbbkJ9X27fOfYBn@mail.gmail.com>
	<AANLkTinRXmR9iHbWvFRrbpxDyocD3g8hTmZ0s_mD9cyd@mail.gmail.com>
Message-ID: <AANLkTikDD-1bFEgt0LUnt9zt4q-PqHFZ8Ut0CrUeRMre@mail.gmail.com>

commit fb66abfab4a74055c38cdc67da83e6e0175a4a0b
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Tue Feb 1 16:22:36 2011 -0600

this is the one you need at a bare minimum.

If you are still having problems the most likely cause is ptime
mismatch between the 2 ends of the call.
you cannot leave loopback in the call path when the legs are not on
the same ptime with very good results.


On Thu, Feb 3, 2011 at 4:31 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> We faced same trouble with FreeSWITCH Version 1.0.head (git-33848eb
> 2011-02-01 08-56-26 +0100), the problem persists with loopback
> channels. Even if two sip endpoints are bridged via loopback, sound is
> choppy.
>
> We have bridge_early_media=true on loopback channel bridge command and
> loopback_bowout=false, loopback_bowout_on_execute=false
>
> 2011/2/2 Dan Lane <null at invalid.name>:
>> This seems to have resolved the issue for us :)
>>
>> Thanks.
>>
>> On Tue, Feb 1, 2011 at 8:47 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>> Try the latest GIT, I reverted the last patch and tried to solve the
>>> problem differently.
>>>
>>>
>>> On Tue, Feb 1, 2011 at 2:11 PM, Dan Lane <null at invalid.name> wrote:
>>>> Yes, using 1f1541b our calls that use mod_loopback into mod_conference
>>>> are unusable. The audio is choppy and delay increases as time passes.
>>>>
>>>> I thought it was an issue relating to the default 100Hz kernel on EC2
>>>> so I spent some time yesterday putting together a 1000Hz kernel but it
>>>> didn't make any difference. In the meantime I've compiled mod_loopback
>>>> from 4c5426f and loaded it with my 1f1541b build which eliminates the
>>>> issue.
>>>>
>>>> I haven't added it to Jira yet as I want to spend some time debugging
>>>> it (and I also owe you some debug info for FS-2934) but the problem is
>>>> definitely there.
>>>>
>>>> On Tue, Feb 1, 2011 at 3:58 PM, Anthony Minessale
>>>> <anthony.minessale at gmail.com> wrote:
>>>>> Are you saying you have better results on that version than you do on
>>>>> the latest?
>>>>> What conditions do you have that cause you trouble, what is the
>>>>> endpoint on the other side.
>>>>>
>>>>> If the last commit to mod_loopback intended to improve audio quality
>>>>> actually makes it worse I need to investigate it.
>>>>>
>>>>>
>>>>> On Tue, Feb 1, 2011 at 5:53 AM, Dan Lane <null at invalid.name> wrote:
>>>>>> FWIW we've been also been having audio issues with loopback recently
>>>>>> on EC2 (with a 1000Hz kernel).
>>>>>>
>>>>>> We worked around it in the short term by reverting mod_loopback to
>>>>>> git-4c5426f during the build process.
>>>>>>
>>>>>> For anyone else who wants to try this just run "git checkout 4c5426f"
>>>>>> in src/mod/endpoints/mod_loopback then build as usual. This is NOT a
>>>>>> long term solution though.
>>>>>>
>>>>>> On Mon, Jan 31, 2011 at 8:02 AM, ?yvind Albrigtsen <oa at estation.dk> wrote:
>>>>>>> Thanks for all your feedback. I'll keep on trying and inform you what
>>>>>>> worked for me.
>>>>>>>
>>>>>>>
>>>>>>> Regards
>>>>>>> Oyvind
>>>>>>>
>>>>>>> On 2011-01-29 21:48, Anthony Minessale wrote:
>>>>>>>> Everyone should try latest GIT before pondering any further because I
>>>>>>>> added a patch like 2 days ago to adress this issue.
>>>>>>>>
>>>>>>>>
>>>>>>>> On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank at telonium.com> ?wrote:
>>>>>>>>
>>>>>>>>> Yeah. I, too, don't see the correlation between the NTP and hw timer..
>>>>>>>>> I am not familiar with the timer_test command and what it's measuring, but
>>>>>>>>> of the 50 tests it ran, min is 19089 and max is 20713.
>>>>>>>>> Frank
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone at ipeva.fr>
>>>>>>>>> wrote:
>>>>>>>>>
>>>>>>>>>> Frank,
>>>>>>>>>> I fail to see the relationship between the hw timer and NTP.
>>>>>>>>>> Can you please elaborate ?
>>>>>>>>>> David Ponzone ?Direction Technique
>>>>>>>>>> email: david.ponzone at ipeva.fr
>>>>>>>>>> tel: ? ? ?01 74 03 18 97
>>>>>>>>>> gsm: ? 06 66 98 76 34
>>>>>>>>>> Service Client IPeva
>>>>>>>>>> tel: ? ? ?0811 46 26 26
>>>>>>>>>> www.ipeva.fr ?- ? www.ipeva-studio.com
>>>>>>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>>>>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>>>>>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>>>>>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>>>>>>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>>>>>>>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> Le 28/01/2011 ? 19:49, Frank Carmickle a ?crit :
>>>>>>>>>>
>>>>>>>>>> Hi
>>>>>>>>>>
>>>>>>>>>> On Jan 28, 2011, at 7:39 AM, ?yvind Albrigtsen wrote:
>>>>>>>>>>
>>>>>>>>>> Hi,
>>>>>>>>>>
>>>>>>>>>> I'm using latest git-version of Freeswitch, and when I go to voicemail
>>>>>>>>>>
>>>>>>>>>> when calling a number the sound playback is choppy and it skips some of
>>>>>>>>>>
>>>>>>>>>> the digits in the number I called.
>>>>>>>>>>
>>>>>>>>>> What kind of results do you get from timer_test at the fs_cli? ?Are you
>>>>>>>>>> running on hardware or are you virtualized? ?What is your clock source set
>>>>>>>>>> to and what are your available clock source options? ?See
>>>>>>>>>> /sys/devices/system/clocksource/clocksource0/available_clocksource and
>>>>>>>>>> /sys/devices/system/clocksource/clocksource0/current_clocksource. ?I am
>>>>>>>>>> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to
>>>>>>>>>> hang at 19998/19999 which works very well for me. ?When I was having problem
>>>>>>>>>> it was reporting numbers all over the map from 17400 to 22600 with lots of
>>>>>>>>>> randomness in between. ?I have my clocksource set to jiffies and xen
>>>>>>>>>> independent wallclock set to 1. ?Of course at that point you need to have
>>>>>>>>>> ntp running against a bunch of servers to drive your clock nice and steady.
>>>>>>>>>> ? I know my set up is probably a lot different than yours but I thought I'd
>>>>>>>>>> toss it out there to show that some of the harshest conditions can be dealt
>>>>>>>>>> with and don't give up trying. ?If you are running on hardware with a cpu
>>>>>>>>>> that doesn't have constant_tsc then you might have some problems. ?Just play
>>>>>>>>>> with the different timer options until you find the one that works.
>>>>>>>>>>
>>>>>>>>>> HTH
>>>>>>>>>> --FC
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at 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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at 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
>>
>
>
>
> --
> 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 kbdfck at gmail.com  Thu Feb  3 21:28:37 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Thu, 3 Feb 2011 21:28:37 +0300
Subject: [Freeswitch-users] Choppy VM when using loopback (on Debian
	lenny)
In-Reply-To: <AANLkTikDD-1bFEgt0LUnt9zt4q-PqHFZ8Ut0CrUeRMre@mail.gmail.com>
References: <4D42B8EE.7030208@estation.dk>
	<7C6F3FC7-445F-4157-BD59-D76A705E4CE7@carmickle.com>
	<9894FEC6-E451-4993-A85A-2CD910E916B1@ipeva.fr>
	<AANLkTim+aBjRiAvxDUF1O1vut_CJpqgct2OqCNVxi0q4@mail.gmail.com>
	<AANLkTiktckovGv4Ksnn_56EzpvTTSnfG+iD1vPX1_4RM@mail.gmail.com>
	<4D466CAE.7030402@estation.dk>
	<AANLkTimhOKmy_yupGtA2HBY_8wXtQKZnDYJ9Ye76bhzq@mail.gmail.com>
	<AANLkTiktQU7kOG_22nG=_oEHSrBad667jcTzBs3n2nQv@mail.gmail.com>
	<AANLkTiko=qTZbXCL09D-+W-FmibMnBjOeoR30YYzHe0i@mail.gmail.com>
	<AANLkTi=t8AiZG6axtfqmMe9ZHTPNowoAb=7eNjQ-RPoG@mail.gmail.com>
	<AANLkTiniBAWw+JMe=64PtAER9B9U7rbbkJ9X27fOfYBn@mail.gmail.com>
	<AANLkTinRXmR9iHbWvFRrbpxDyocD3g8hTmZ0s_mD9cyd@mail.gmail.com>
	<AANLkTikDD-1bFEgt0LUnt9zt4q-PqHFZ8Ut0CrUeRMre@mail.gmail.com>
Message-ID: <AANLkTi=Z1P+0C9BvEnOSdQjnu7Va-SmY3Z5+1KSPRFOv@mail.gmail.com>

Thanks, I will test tomorrow.

I'm wondering if I could use att_xfer with loopbacks after these patches:D

2011/2/3 Anthony Minessale <anthony.minessale at gmail.com>:
> commit fb66abfab4a74055c38cdc67da83e6e0175a4a0b
> Author: Anthony Minessale <anthm at freeswitch.org>
> Date: ? Tue Feb 1 16:22:36 2011 -0600
>
> this is the one you need at a bare minimum.
>
> If you are still having problems the most likely cause is ptime
> mismatch between the 2 ends of the call.
> you cannot leave loopback in the call path when the legs are not on
> the same ptime with very good results.
>
>
> On Thu, Feb 3, 2011 at 4:31 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>> We faced same trouble with FreeSWITCH Version 1.0.head (git-33848eb
>> 2011-02-01 08-56-26 +0100), the problem persists with loopback
>> channels. Even if two sip endpoints are bridged via loopback, sound is
>> choppy.
>>
>> We have bridge_early_media=true on loopback channel bridge command and
>> loopback_bowout=false, loopback_bowout_on_execute=false
>>
>> 2011/2/2 Dan Lane <null at invalid.name>:
>>> This seems to have resolved the issue for us :)
>>>
>>> Thanks.
>>>
>>> On Tue, Feb 1, 2011 at 8:47 PM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>>> Try the latest GIT, I reverted the last patch and tried to solve the
>>>> problem differently.
>>>>
>>>>
>>>> On Tue, Feb 1, 2011 at 2:11 PM, Dan Lane <null at invalid.name> wrote:
>>>>> Yes, using 1f1541b our calls that use mod_loopback into mod_conference
>>>>> are unusable. The audio is choppy and delay increases as time passes.
>>>>>
>>>>> I thought it was an issue relating to the default 100Hz kernel on EC2
>>>>> so I spent some time yesterday putting together a 1000Hz kernel but it
>>>>> didn't make any difference. In the meantime I've compiled mod_loopback
>>>>> from 4c5426f and loaded it with my 1f1541b build which eliminates the
>>>>> issue.
>>>>>
>>>>> I haven't added it to Jira yet as I want to spend some time debugging
>>>>> it (and I also owe you some debug info for FS-2934) but the problem is
>>>>> definitely there.
>>>>>
>>>>> On Tue, Feb 1, 2011 at 3:58 PM, Anthony Minessale
>>>>> <anthony.minessale at gmail.com> wrote:
>>>>>> Are you saying you have better results on that version than you do on
>>>>>> the latest?
>>>>>> What conditions do you have that cause you trouble, what is the
>>>>>> endpoint on the other side.
>>>>>>
>>>>>> If the last commit to mod_loopback intended to improve audio quality
>>>>>> actually makes it worse I need to investigate it.
>>>>>>
>>>>>>
>>>>>> On Tue, Feb 1, 2011 at 5:53 AM, Dan Lane <null at invalid.name> wrote:
>>>>>>> FWIW we've been also been having audio issues with loopback recently
>>>>>>> on EC2 (with a 1000Hz kernel).
>>>>>>>
>>>>>>> We worked around it in the short term by reverting mod_loopback to
>>>>>>> git-4c5426f during the build process.
>>>>>>>
>>>>>>> For anyone else who wants to try this just run "git checkout 4c5426f"
>>>>>>> in src/mod/endpoints/mod_loopback then build as usual. This is NOT a
>>>>>>> long term solution though.
>>>>>>>
>>>>>>> On Mon, Jan 31, 2011 at 8:02 AM, ?yvind Albrigtsen <oa at estation.dk> wrote:
>>>>>>>> Thanks for all your feedback. I'll keep on trying and inform you what
>>>>>>>> worked for me.
>>>>>>>>
>>>>>>>>
>>>>>>>> Regards
>>>>>>>> Oyvind
>>>>>>>>
>>>>>>>> On 2011-01-29 21:48, Anthony Minessale wrote:
>>>>>>>>> Everyone should try latest GIT before pondering any further because I
>>>>>>>>> added a patch like 2 days ago to adress this issue.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank at telonium.com> ?wrote:
>>>>>>>>>
>>>>>>>>>> Yeah. I, too, don't see the correlation between the NTP and hw timer..
>>>>>>>>>> I am not familiar with the timer_test command and what it's measuring, but
>>>>>>>>>> of the 50 tests it ran, min is 19089 and max is 20713.
>>>>>>>>>> Frank
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone at ipeva.fr>
>>>>>>>>>> wrote:
>>>>>>>>>>
>>>>>>>>>>> Frank,
>>>>>>>>>>> I fail to see the relationship between the hw timer and NTP.
>>>>>>>>>>> Can you please elaborate ?
>>>>>>>>>>> David Ponzone ?Direction Technique
>>>>>>>>>>> email: david.ponzone at ipeva.fr
>>>>>>>>>>> tel: ? ? ?01 74 03 18 97
>>>>>>>>>>> gsm: ? 06 66 98 76 34
>>>>>>>>>>> Service Client IPeva
>>>>>>>>>>> tel: ? ? ?0811 46 26 26
>>>>>>>>>>> www.ipeva.fr ?- ? www.ipeva-studio.com
>>>>>>>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>>>>>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>>>>>>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>>>>>>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>>>>>>>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>>>>>>>>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> Le 28/01/2011 ? 19:49, Frank Carmickle a ?crit :
>>>>>>>>>>>
>>>>>>>>>>> Hi
>>>>>>>>>>>
>>>>>>>>>>> On Jan 28, 2011, at 7:39 AM, ?yvind Albrigtsen wrote:
>>>>>>>>>>>
>>>>>>>>>>> Hi,
>>>>>>>>>>>
>>>>>>>>>>> I'm using latest git-version of Freeswitch, and when I go to voicemail
>>>>>>>>>>>
>>>>>>>>>>> when calling a number the sound playback is choppy and it skips some of
>>>>>>>>>>>
>>>>>>>>>>> the digits in the number I called.
>>>>>>>>>>>
>>>>>>>>>>> What kind of results do you get from timer_test at the fs_cli? ?Are you
>>>>>>>>>>> running on hardware or are you virtualized? ?What is your clock source set
>>>>>>>>>>> to and what are your available clock source options? ?See
>>>>>>>>>>> /sys/devices/system/clocksource/clocksource0/available_clocksource and
>>>>>>>>>>> /sys/devices/system/clocksource/clocksource0/current_clocksource. ?I am
>>>>>>>>>>> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to
>>>>>>>>>>> hang at 19998/19999 which works very well for me. ?When I was having problem
>>>>>>>>>>> it was reporting numbers all over the map from 17400 to 22600 with lots of
>>>>>>>>>>> randomness in between. ?I have my clocksource set to jiffies and xen
>>>>>>>>>>> independent wallclock set to 1. ?Of course at that point you need to have
>>>>>>>>>>> ntp running against a bunch of servers to drive your clock nice and steady.
>>>>>>>>>>> ? I know my set up is probably a lot different than yours but I thought I'd
>>>>>>>>>>> toss it out there to show that some of the harshest conditions can be dealt
>>>>>>>>>>> with and don't give up trying. ?If you are running on hardware with a cpu
>>>>>>>>>>> that doesn't have constant_tsc then you might have some problems. ?Just play
>>>>>>>>>>> with the different timer options until you find the one that works.
>>>>>>>>>>>
>>>>>>>>>>> HTH
>>>>>>>>>>> --FC
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at 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
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at 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
>>>
>>
>>
>>
>> --
>> 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


From anthony.minessale at gmail.com  Thu Feb  3 21:32:31 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 3 Feb 2011 12:32:31 -0600
Subject: [Freeswitch-users] Choppy VM when using loopback (on Debian
	lenny)
In-Reply-To: <AANLkTi=Z1P+0C9BvEnOSdQjnu7Va-SmY3Z5+1KSPRFOv@mail.gmail.com>
References: <4D42B8EE.7030208@estation.dk>
	<7C6F3FC7-445F-4157-BD59-D76A705E4CE7@carmickle.com>
	<9894FEC6-E451-4993-A85A-2CD910E916B1@ipeva.fr>
	<AANLkTim+aBjRiAvxDUF1O1vut_CJpqgct2OqCNVxi0q4@mail.gmail.com>
	<AANLkTiktckovGv4Ksnn_56EzpvTTSnfG+iD1vPX1_4RM@mail.gmail.com>
	<4D466CAE.7030402@estation.dk>
	<AANLkTimhOKmy_yupGtA2HBY_8wXtQKZnDYJ9Ye76bhzq@mail.gmail.com>
	<AANLkTiktQU7kOG_22nG=_oEHSrBad667jcTzBs3n2nQv@mail.gmail.com>
	<AANLkTiko=qTZbXCL09D-+W-FmibMnBjOeoR30YYzHe0i@mail.gmail.com>
	<AANLkTi=t8AiZG6axtfqmMe9ZHTPNowoAb=7eNjQ-RPoG@mail.gmail.com>
	<AANLkTiniBAWw+JMe=64PtAER9B9U7rbbkJ9X27fOfYBn@mail.gmail.com>
	<AANLkTinRXmR9iHbWvFRrbpxDyocD3g8hTmZ0s_mD9cyd@mail.gmail.com>
	<AANLkTikDD-1bFEgt0LUnt9zt4q-PqHFZ8Ut0CrUeRMre@mail.gmail.com>
	<AANLkTi=Z1P+0C9BvEnOSdQjnu7Va-SmY3Z5+1KSPRFOv@mail.gmail.com>
Message-ID: <AANLkTikkLevVzpoM3YkY5Kvhu_xEMkwGorS08Gr89Rn0@mail.gmail.com>

maybe, but using loopback without the bowout is very costly.
you can also consider hair-pinning a call back to your own box over
SIP for this type of thing.


On Thu, Feb 3, 2011 at 12:28 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> Thanks, I will test tomorrow.
>
> I'm wondering if I could use att_xfer with loopbacks after these patches:D
>
> 2011/2/3 Anthony Minessale <anthony.minessale at gmail.com>:
>> commit fb66abfab4a74055c38cdc67da83e6e0175a4a0b
>> Author: Anthony Minessale <anthm at freeswitch.org>
>> Date: ? Tue Feb 1 16:22:36 2011 -0600
>>
>> this is the one you need at a bare minimum.
>>
>> If you are still having problems the most likely cause is ptime
>> mismatch between the 2 ends of the call.
>> you cannot leave loopback in the call path when the legs are not on
>> the same ptime with very good results.
>>
>>
>> On Thu, Feb 3, 2011 at 4:31 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>> We faced same trouble with FreeSWITCH Version 1.0.head (git-33848eb
>>> 2011-02-01 08-56-26 +0100), the problem persists with loopback
>>> channels. Even if two sip endpoints are bridged via loopback, sound is
>>> choppy.
>>>
>>> We have bridge_early_media=true on loopback channel bridge command and
>>> loopback_bowout=false, loopback_bowout_on_execute=false
>>>
>>> 2011/2/2 Dan Lane <null at invalid.name>:
>>>> This seems to have resolved the issue for us :)
>>>>
>>>> Thanks.
>>>>
>>>> On Tue, Feb 1, 2011 at 8:47 PM, Anthony Minessale
>>>> <anthony.minessale at gmail.com> wrote:
>>>>> Try the latest GIT, I reverted the last patch and tried to solve the
>>>>> problem differently.
>>>>>
>>>>>
>>>>> On Tue, Feb 1, 2011 at 2:11 PM, Dan Lane <null at invalid.name> wrote:
>>>>>> Yes, using 1f1541b our calls that use mod_loopback into mod_conference
>>>>>> are unusable. The audio is choppy and delay increases as time passes.
>>>>>>
>>>>>> I thought it was an issue relating to the default 100Hz kernel on EC2
>>>>>> so I spent some time yesterday putting together a 1000Hz kernel but it
>>>>>> didn't make any difference. In the meantime I've compiled mod_loopback
>>>>>> from 4c5426f and loaded it with my 1f1541b build which eliminates the
>>>>>> issue.
>>>>>>
>>>>>> I haven't added it to Jira yet as I want to spend some time debugging
>>>>>> it (and I also owe you some debug info for FS-2934) but the problem is
>>>>>> definitely there.
>>>>>>
>>>>>> On Tue, Feb 1, 2011 at 3:58 PM, Anthony Minessale
>>>>>> <anthony.minessale at gmail.com> wrote:
>>>>>>> Are you saying you have better results on that version than you do on
>>>>>>> the latest?
>>>>>>> What conditions do you have that cause you trouble, what is the
>>>>>>> endpoint on the other side.
>>>>>>>
>>>>>>> If the last commit to mod_loopback intended to improve audio quality
>>>>>>> actually makes it worse I need to investigate it.
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Feb 1, 2011 at 5:53 AM, Dan Lane <null at invalid.name> wrote:
>>>>>>>> FWIW we've been also been having audio issues with loopback recently
>>>>>>>> on EC2 (with a 1000Hz kernel).
>>>>>>>>
>>>>>>>> We worked around it in the short term by reverting mod_loopback to
>>>>>>>> git-4c5426f during the build process.
>>>>>>>>
>>>>>>>> For anyone else who wants to try this just run "git checkout 4c5426f"
>>>>>>>> in src/mod/endpoints/mod_loopback then build as usual. This is NOT a
>>>>>>>> long term solution though.
>>>>>>>>
>>>>>>>> On Mon, Jan 31, 2011 at 8:02 AM, ?yvind Albrigtsen <oa at estation.dk> wrote:
>>>>>>>>> Thanks for all your feedback. I'll keep on trying and inform you what
>>>>>>>>> worked for me.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Regards
>>>>>>>>> Oyvind
>>>>>>>>>
>>>>>>>>> On 2011-01-29 21:48, Anthony Minessale wrote:
>>>>>>>>>> Everyone should try latest GIT before pondering any further because I
>>>>>>>>>> added a patch like 2 days ago to adress this issue.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank at telonium.com> ?wrote:
>>>>>>>>>>
>>>>>>>>>>> Yeah. I, too, don't see the correlation between the NTP and hw timer..
>>>>>>>>>>> I am not familiar with the timer_test command and what it's measuring, but
>>>>>>>>>>> of the 50 tests it ran, min is 19089 and max is 20713.
>>>>>>>>>>> Frank
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone at ipeva.fr>
>>>>>>>>>>> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> Frank,
>>>>>>>>>>>> I fail to see the relationship between the hw timer and NTP.
>>>>>>>>>>>> Can you please elaborate ?
>>>>>>>>>>>> David Ponzone ?Direction Technique
>>>>>>>>>>>> email: david.ponzone at ipeva.fr
>>>>>>>>>>>> tel: ? ? ?01 74 03 18 97
>>>>>>>>>>>> gsm: ? 06 66 98 76 34
>>>>>>>>>>>> Service Client IPeva
>>>>>>>>>>>> tel: ? ? ?0811 46 26 26
>>>>>>>>>>>> www.ipeva.fr ?- ? www.ipeva-studio.com
>>>>>>>>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>>>>>>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>>>>>>>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>>>>>>>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>>>>>>>>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>>>>>>>>>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> Le 28/01/2011 ? 19:49, Frank Carmickle a ?crit :
>>>>>>>>>>>>
>>>>>>>>>>>> Hi
>>>>>>>>>>>>
>>>>>>>>>>>> On Jan 28, 2011, at 7:39 AM, ?yvind Albrigtsen wrote:
>>>>>>>>>>>>
>>>>>>>>>>>> Hi,
>>>>>>>>>>>>
>>>>>>>>>>>> I'm using latest git-version of Freeswitch, and when I go to voicemail
>>>>>>>>>>>>
>>>>>>>>>>>> when calling a number the sound playback is choppy and it skips some of
>>>>>>>>>>>>
>>>>>>>>>>>> the digits in the number I called.
>>>>>>>>>>>>
>>>>>>>>>>>> What kind of results do you get from timer_test at the fs_cli? ?Are you
>>>>>>>>>>>> running on hardware or are you virtualized? ?What is your clock source set
>>>>>>>>>>>> to and what are your available clock source options? ?See
>>>>>>>>>>>> /sys/devices/system/clocksource/clocksource0/available_clocksource and
>>>>>>>>>>>> /sys/devices/system/clocksource/clocksource0/current_clocksource. ?I am
>>>>>>>>>>>> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to
>>>>>>>>>>>> hang at 19998/19999 which works very well for me. ?When I was having problem
>>>>>>>>>>>> it was reporting numbers all over the map from 17400 to 22600 with lots of
>>>>>>>>>>>> randomness in between. ?I have my clocksource set to jiffies and xen
>>>>>>>>>>>> independent wallclock set to 1. ?Of course at that point you need to have
>>>>>>>>>>>> ntp running against a bunch of servers to drive your clock nice and steady.
>>>>>>>>>>>> ? I know my set up is probably a lot different than yours but I thought I'd
>>>>>>>>>>>> toss it out there to show that some of the harshest conditions can be dealt
>>>>>>>>>>>> with and don't give up trying. ?If you are running on hardware with a cpu
>>>>>>>>>>>> that doesn't have constant_tsc then you might have some problems. ?Just play
>>>>>>>>>>>> with the different timer options until you find the one that works.
>>>>>>>>>>>>
>>>>>>>>>>>> HTH
>>>>>>>>>>>> --FC
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at 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
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at 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
>>>>
>>>
>>>
>>>
>>> --
>>> 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  Thu Feb  3 22:00:52 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 3 Feb 2011 13:00:52 -0600
Subject: [Freeswitch-users] Choppy VM when using loopback (on Debian
	lenny)
In-Reply-To: <AANLkTikkLevVzpoM3YkY5Kvhu_xEMkwGorS08Gr89Rn0@mail.gmail.com>
References: <4D42B8EE.7030208@estation.dk>
	<7C6F3FC7-445F-4157-BD59-D76A705E4CE7@carmickle.com>
	<9894FEC6-E451-4993-A85A-2CD910E916B1@ipeva.fr>
	<AANLkTim+aBjRiAvxDUF1O1vut_CJpqgct2OqCNVxi0q4@mail.gmail.com>
	<AANLkTiktckovGv4Ksnn_56EzpvTTSnfG+iD1vPX1_4RM@mail.gmail.com>
	<4D466CAE.7030402@estation.dk>
	<AANLkTimhOKmy_yupGtA2HBY_8wXtQKZnDYJ9Ye76bhzq@mail.gmail.com>
	<AANLkTiktQU7kOG_22nG=_oEHSrBad667jcTzBs3n2nQv@mail.gmail.com>
	<AANLkTiko=qTZbXCL09D-+W-FmibMnBjOeoR30YYzHe0i@mail.gmail.com>
	<AANLkTi=t8AiZG6axtfqmMe9ZHTPNowoAb=7eNjQ-RPoG@mail.gmail.com>
	<AANLkTiniBAWw+JMe=64PtAER9B9U7rbbkJ9X27fOfYBn@mail.gmail.com>
	<AANLkTinRXmR9iHbWvFRrbpxDyocD3g8hTmZ0s_mD9cyd@mail.gmail.com>
	<AANLkTikDD-1bFEgt0LUnt9zt4q-PqHFZ8Ut0CrUeRMre@mail.gmail.com>
	<AANLkTi=Z1P+0C9BvEnOSdQjnu7Va-SmY3Z5+1KSPRFOv@mail.gmail.com>
	<AANLkTikkLevVzpoM3YkY5Kvhu_xEMkwGorS08Gr89Rn0@mail.gmail.com>
Message-ID: <AANLkTi=Vr7aiPxxE7VQuuDYwGuZeHR2pO7oH=Q-7Drct@mail.gmail.com>

also consider passing {rtp_timer_name=none} through to disable the rtp
timer in the sip leg when butting up against mod_loopback


On Thu, Feb 3, 2011 at 12:32 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> maybe, but using loopback without the bowout is very costly.
> you can also consider hair-pinning a call back to your own box over
> SIP for this type of thing.
>
>
> On Thu, Feb 3, 2011 at 12:28 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>> Thanks, I will test tomorrow.
>>
>> I'm wondering if I could use att_xfer with loopbacks after these patches:D
>>
>> 2011/2/3 Anthony Minessale <anthony.minessale at gmail.com>:
>>> commit fb66abfab4a74055c38cdc67da83e6e0175a4a0b
>>> Author: Anthony Minessale <anthm at freeswitch.org>
>>> Date: ? Tue Feb 1 16:22:36 2011 -0600
>>>
>>> this is the one you need at a bare minimum.
>>>
>>> If you are still having problems the most likely cause is ptime
>>> mismatch between the 2 ends of the call.
>>> you cannot leave loopback in the call path when the legs are not on
>>> the same ptime with very good results.
>>>
>>>
>>> On Thu, Feb 3, 2011 at 4:31 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>>> We faced same trouble with FreeSWITCH Version 1.0.head (git-33848eb
>>>> 2011-02-01 08-56-26 +0100), the problem persists with loopback
>>>> channels. Even if two sip endpoints are bridged via loopback, sound is
>>>> choppy.
>>>>
>>>> We have bridge_early_media=true on loopback channel bridge command and
>>>> loopback_bowout=false, loopback_bowout_on_execute=false
>>>>
>>>> 2011/2/2 Dan Lane <null at invalid.name>:
>>>>> This seems to have resolved the issue for us :)
>>>>>
>>>>> Thanks.
>>>>>
>>>>> On Tue, Feb 1, 2011 at 8:47 PM, Anthony Minessale
>>>>> <anthony.minessale at gmail.com> wrote:
>>>>>> Try the latest GIT, I reverted the last patch and tried to solve the
>>>>>> problem differently.
>>>>>>
>>>>>>
>>>>>> On Tue, Feb 1, 2011 at 2:11 PM, Dan Lane <null at invalid.name> wrote:
>>>>>>> Yes, using 1f1541b our calls that use mod_loopback into mod_conference
>>>>>>> are unusable. The audio is choppy and delay increases as time passes.
>>>>>>>
>>>>>>> I thought it was an issue relating to the default 100Hz kernel on EC2
>>>>>>> so I spent some time yesterday putting together a 1000Hz kernel but it
>>>>>>> didn't make any difference. In the meantime I've compiled mod_loopback
>>>>>>> from 4c5426f and loaded it with my 1f1541b build which eliminates the
>>>>>>> issue.
>>>>>>>
>>>>>>> I haven't added it to Jira yet as I want to spend some time debugging
>>>>>>> it (and I also owe you some debug info for FS-2934) but the problem is
>>>>>>> definitely there.
>>>>>>>
>>>>>>> On Tue, Feb 1, 2011 at 3:58 PM, Anthony Minessale
>>>>>>> <anthony.minessale at gmail.com> wrote:
>>>>>>>> Are you saying you have better results on that version than you do on
>>>>>>>> the latest?
>>>>>>>> What conditions do you have that cause you trouble, what is the
>>>>>>>> endpoint on the other side.
>>>>>>>>
>>>>>>>> If the last commit to mod_loopback intended to improve audio quality
>>>>>>>> actually makes it worse I need to investigate it.
>>>>>>>>
>>>>>>>>
>>>>>>>> On Tue, Feb 1, 2011 at 5:53 AM, Dan Lane <null at invalid.name> wrote:
>>>>>>>>> FWIW we've been also been having audio issues with loopback recently
>>>>>>>>> on EC2 (with a 1000Hz kernel).
>>>>>>>>>
>>>>>>>>> We worked around it in the short term by reverting mod_loopback to
>>>>>>>>> git-4c5426f during the build process.
>>>>>>>>>
>>>>>>>>> For anyone else who wants to try this just run "git checkout 4c5426f"
>>>>>>>>> in src/mod/endpoints/mod_loopback then build as usual. This is NOT a
>>>>>>>>> long term solution though.
>>>>>>>>>
>>>>>>>>> On Mon, Jan 31, 2011 at 8:02 AM, ?yvind Albrigtsen <oa at estation.dk> wrote:
>>>>>>>>>> Thanks for all your feedback. I'll keep on trying and inform you what
>>>>>>>>>> worked for me.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> Regards
>>>>>>>>>> Oyvind
>>>>>>>>>>
>>>>>>>>>> On 2011-01-29 21:48, Anthony Minessale wrote:
>>>>>>>>>>> Everyone should try latest GIT before pondering any further because I
>>>>>>>>>>> added a patch like 2 days ago to adress this issue.
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank at telonium.com> ?wrote:
>>>>>>>>>>>
>>>>>>>>>>>> Yeah. I, too, don't see the correlation between the NTP and hw timer..
>>>>>>>>>>>> I am not familiar with the timer_test command and what it's measuring, but
>>>>>>>>>>>> of the 50 tests it ran, min is 19089 and max is 20713.
>>>>>>>>>>>> Frank
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone at ipeva.fr>
>>>>>>>>>>>> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> Frank,
>>>>>>>>>>>>> I fail to see the relationship between the hw timer and NTP.
>>>>>>>>>>>>> Can you please elaborate ?
>>>>>>>>>>>>> David Ponzone ?Direction Technique
>>>>>>>>>>>>> email: david.ponzone at ipeva.fr
>>>>>>>>>>>>> tel: ? ? ?01 74 03 18 97
>>>>>>>>>>>>> gsm: ? 06 66 98 76 34
>>>>>>>>>>>>> Service Client IPeva
>>>>>>>>>>>>> tel: ? ? ?0811 46 26 26
>>>>>>>>>>>>> www.ipeva.fr ?- ? www.ipeva-studio.com
>>>>>>>>>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>>>>>>>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>>>>>>>>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>>>>>>>>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>>>>>>>>>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>>>>>>>>>>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> Le 28/01/2011 ? 19:49, Frank Carmickle a ?crit :
>>>>>>>>>>>>>
>>>>>>>>>>>>> Hi
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Jan 28, 2011, at 7:39 AM, ?yvind Albrigtsen wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>> Hi,
>>>>>>>>>>>>>
>>>>>>>>>>>>> I'm using latest git-version of Freeswitch, and when I go to voicemail
>>>>>>>>>>>>>
>>>>>>>>>>>>> when calling a number the sound playback is choppy and it skips some of
>>>>>>>>>>>>>
>>>>>>>>>>>>> the digits in the number I called.
>>>>>>>>>>>>>
>>>>>>>>>>>>> What kind of results do you get from timer_test at the fs_cli? ?Are you
>>>>>>>>>>>>> running on hardware or are you virtualized? ?What is your clock source set
>>>>>>>>>>>>> to and what are your available clock source options? ?See
>>>>>>>>>>>>> /sys/devices/system/clocksource/clocksource0/available_clocksource and
>>>>>>>>>>>>> /sys/devices/system/clocksource/clocksource0/current_clocksource. ?I am
>>>>>>>>>>>>> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to
>>>>>>>>>>>>> hang at 19998/19999 which works very well for me. ?When I was having problem
>>>>>>>>>>>>> it was reporting numbers all over the map from 17400 to 22600 with lots of
>>>>>>>>>>>>> randomness in between. ?I have my clocksource set to jiffies and xen
>>>>>>>>>>>>> independent wallclock set to 1. ?Of course at that point you need to have
>>>>>>>>>>>>> ntp running against a bunch of servers to drive your clock nice and steady.
>>>>>>>>>>>>> ? I know my set up is probably a lot different than yours but I thought I'd
>>>>>>>>>>>>> toss it out there to show that some of the harshest conditions can be dealt
>>>>>>>>>>>>> with and don't give up trying. ?If you are running on hardware with a cpu
>>>>>>>>>>>>> that doesn't have constant_tsc then you might have some problems. ?Just play
>>>>>>>>>>>>> with the different timer options until you find the one that works.
>>>>>>>>>>>>>
>>>>>>>>>>>>> HTH
>>>>>>>>>>>>> --FC
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at 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
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at 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
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> 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 kbdfck at gmail.com  Thu Feb  3 22:06:16 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Thu, 3 Feb 2011 22:06:16 +0300
Subject: [Freeswitch-users] Choppy VM when using loopback (on Debian
	lenny)
In-Reply-To: <AANLkTikkLevVzpoM3YkY5Kvhu_xEMkwGorS08Gr89Rn0@mail.gmail.com>
References: <4D42B8EE.7030208@estation.dk>
	<7C6F3FC7-445F-4157-BD59-D76A705E4CE7@carmickle.com>
	<9894FEC6-E451-4993-A85A-2CD910E916B1@ipeva.fr>
	<AANLkTim+aBjRiAvxDUF1O1vut_CJpqgct2OqCNVxi0q4@mail.gmail.com>
	<AANLkTiktckovGv4Ksnn_56EzpvTTSnfG+iD1vPX1_4RM@mail.gmail.com>
	<4D466CAE.7030402@estation.dk>
	<AANLkTimhOKmy_yupGtA2HBY_8wXtQKZnDYJ9Ye76bhzq@mail.gmail.com>
	<AANLkTiktQU7kOG_22nG=_oEHSrBad667jcTzBs3n2nQv@mail.gmail.com>
	<AANLkTiko=qTZbXCL09D-+W-FmibMnBjOeoR30YYzHe0i@mail.gmail.com>
	<AANLkTi=t8AiZG6axtfqmMe9ZHTPNowoAb=7eNjQ-RPoG@mail.gmail.com>
	<AANLkTiniBAWw+JMe=64PtAER9B9U7rbbkJ9X27fOfYBn@mail.gmail.com>
	<AANLkTinRXmR9iHbWvFRrbpxDyocD3g8hTmZ0s_mD9cyd@mail.gmail.com>
	<AANLkTikDD-1bFEgt0LUnt9zt4q-PqHFZ8Ut0CrUeRMre@mail.gmail.com>
	<AANLkTi=Z1P+0C9BvEnOSdQjnu7Va-SmY3Z5+1KSPRFOv@mail.gmail.com>
	<AANLkTikkLevVzpoM3YkY5Kvhu_xEMkwGorS08Gr89Rn0@mail.gmail.com>
Message-ID: <AANLkTim5Ha1YC2W5a6tygWtMg_LeT7BCMRbr6Z0bXywZ@mail.gmail.com>

Yes, I also had this idea, but I thought it would lead to troubles
with loop detection.
How this can be done exactly? I need separate profiles for this? Or I
just can dial from internal to internal?

And what if I use Kamailio in front of FS? Then need to deal with call
re-routing to kamailio. I thought it may lead to spiral detection, if
I understand right what spiral is.

2011/2/3 Anthony Minessale <anthony.minessale at gmail.com>:
> maybe, but using loopback without the bowout is very costly.
> you can also consider hair-pinning a call back to your own box over
> SIP for this type of thing.
>
>
> On Thu, Feb 3, 2011 at 12:28 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>> Thanks, I will test tomorrow.
>>
>> I'm wondering if I could use att_xfer with loopbacks after these patches:D
>>
>> 2011/2/3 Anthony Minessale <anthony.minessale at gmail.com>:
>>> commit fb66abfab4a74055c38cdc67da83e6e0175a4a0b
>>> Author: Anthony Minessale <anthm at freeswitch.org>
>>> Date: ? Tue Feb 1 16:22:36 2011 -0600
>>>
>>> this is the one you need at a bare minimum.
>>>
>>> If you are still having problems the most likely cause is ptime
>>> mismatch between the 2 ends of the call.
>>> you cannot leave loopback in the call path when the legs are not on
>>> the same ptime with very good results.
>>>
>>>
>>> On Thu, Feb 3, 2011 at 4:31 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>>> We faced same trouble with FreeSWITCH Version 1.0.head (git-33848eb
>>>> 2011-02-01 08-56-26 +0100), the problem persists with loopback
>>>> channels. Even if two sip endpoints are bridged via loopback, sound is
>>>> choppy.
>>>>
>>>> We have bridge_early_media=true on loopback channel bridge command and
>>>> loopback_bowout=false, loopback_bowout_on_execute=false
>>>>
>>>> 2011/2/2 Dan Lane <null at invalid.name>:
>>>>> This seems to have resolved the issue for us :)
>>>>>
>>>>> Thanks.
>>>>>
>>>>> On Tue, Feb 1, 2011 at 8:47 PM, Anthony Minessale
>>>>> <anthony.minessale at gmail.com> wrote:
>>>>>> Try the latest GIT, I reverted the last patch and tried to solve the
>>>>>> problem differently.
>>>>>>
>>>>>>
>>>>>> On Tue, Feb 1, 2011 at 2:11 PM, Dan Lane <null at invalid.name> wrote:
>>>>>>> Yes, using 1f1541b our calls that use mod_loopback into mod_conference
>>>>>>> are unusable. The audio is choppy and delay increases as time passes.
>>>>>>>
>>>>>>> I thought it was an issue relating to the default 100Hz kernel on EC2
>>>>>>> so I spent some time yesterday putting together a 1000Hz kernel but it
>>>>>>> didn't make any difference. In the meantime I've compiled mod_loopback
>>>>>>> from 4c5426f and loaded it with my 1f1541b build which eliminates the
>>>>>>> issue.
>>>>>>>
>>>>>>> I haven't added it to Jira yet as I want to spend some time debugging
>>>>>>> it (and I also owe you some debug info for FS-2934) but the problem is
>>>>>>> definitely there.
>>>>>>>
>>>>>>> On Tue, Feb 1, 2011 at 3:58 PM, Anthony Minessale
>>>>>>> <anthony.minessale at gmail.com> wrote:
>>>>>>>> Are you saying you have better results on that version than you do on
>>>>>>>> the latest?
>>>>>>>> What conditions do you have that cause you trouble, what is the
>>>>>>>> endpoint on the other side.
>>>>>>>>
>>>>>>>> If the last commit to mod_loopback intended to improve audio quality
>>>>>>>> actually makes it worse I need to investigate it.
>>>>>>>>
>>>>>>>>
>>>>>>>> On Tue, Feb 1, 2011 at 5:53 AM, Dan Lane <null at invalid.name> wrote:
>>>>>>>>> FWIW we've been also been having audio issues with loopback recently
>>>>>>>>> on EC2 (with a 1000Hz kernel).
>>>>>>>>>
>>>>>>>>> We worked around it in the short term by reverting mod_loopback to
>>>>>>>>> git-4c5426f during the build process.
>>>>>>>>>
>>>>>>>>> For anyone else who wants to try this just run "git checkout 4c5426f"
>>>>>>>>> in src/mod/endpoints/mod_loopback then build as usual. This is NOT a
>>>>>>>>> long term solution though.
>>>>>>>>>
>>>>>>>>> On Mon, Jan 31, 2011 at 8:02 AM, ?yvind Albrigtsen <oa at estation.dk> wrote:
>>>>>>>>>> Thanks for all your feedback. I'll keep on trying and inform you what
>>>>>>>>>> worked for me.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> Regards
>>>>>>>>>> Oyvind
>>>>>>>>>>
>>>>>>>>>> On 2011-01-29 21:48, Anthony Minessale wrote:
>>>>>>>>>>> Everyone should try latest GIT before pondering any further because I
>>>>>>>>>>> added a patch like 2 days ago to adress this issue.
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank at telonium.com> ?wrote:
>>>>>>>>>>>
>>>>>>>>>>>> Yeah. I, too, don't see the correlation between the NTP and hw timer..
>>>>>>>>>>>> I am not familiar with the timer_test command and what it's measuring, but
>>>>>>>>>>>> of the 50 tests it ran, min is 19089 and max is 20713.
>>>>>>>>>>>> Frank
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone at ipeva.fr>
>>>>>>>>>>>> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> Frank,
>>>>>>>>>>>>> I fail to see the relationship between the hw timer and NTP.
>>>>>>>>>>>>> Can you please elaborate ?
>>>>>>>>>>>>> David Ponzone ?Direction Technique
>>>>>>>>>>>>> email: david.ponzone at ipeva.fr
>>>>>>>>>>>>> tel: ? ? ?01 74 03 18 97
>>>>>>>>>>>>> gsm: ? 06 66 98 76 34
>>>>>>>>>>>>> Service Client IPeva
>>>>>>>>>>>>> tel: ? ? ?0811 46 26 26
>>>>>>>>>>>>> www.ipeva.fr ?- ? www.ipeva-studio.com
>>>>>>>>>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>>>>>>>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>>>>>>>>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>>>>>>>>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>>>>>>>>>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>>>>>>>>>>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> Le 28/01/2011 ? 19:49, Frank Carmickle a ?crit :
>>>>>>>>>>>>>
>>>>>>>>>>>>> Hi
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Jan 28, 2011, at 7:39 AM, ?yvind Albrigtsen wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>> Hi,
>>>>>>>>>>>>>
>>>>>>>>>>>>> I'm using latest git-version of Freeswitch, and when I go to voicemail
>>>>>>>>>>>>>
>>>>>>>>>>>>> when calling a number the sound playback is choppy and it skips some of
>>>>>>>>>>>>>
>>>>>>>>>>>>> the digits in the number I called.
>>>>>>>>>>>>>
>>>>>>>>>>>>> What kind of results do you get from timer_test at the fs_cli? ?Are you
>>>>>>>>>>>>> running on hardware or are you virtualized? ?What is your clock source set
>>>>>>>>>>>>> to and what are your available clock source options? ?See
>>>>>>>>>>>>> /sys/devices/system/clocksource/clocksource0/available_clocksource and
>>>>>>>>>>>>> /sys/devices/system/clocksource/clocksource0/current_clocksource. ?I am
>>>>>>>>>>>>> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to
>>>>>>>>>>>>> hang at 19998/19999 which works very well for me. ?When I was having problem
>>>>>>>>>>>>> it was reporting numbers all over the map from 17400 to 22600 with lots of
>>>>>>>>>>>>> randomness in between. ?I have my clocksource set to jiffies and xen
>>>>>>>>>>>>> independent wallclock set to 1. ?Of course at that point you need to have
>>>>>>>>>>>>> ntp running against a bunch of servers to drive your clock nice and steady.
>>>>>>>>>>>>> ? I know my set up is probably a lot different than yours but I thought I'd
>>>>>>>>>>>>> toss it out there to show that some of the harshest conditions can be dealt
>>>>>>>>>>>>> with and don't give up trying. ?If you are running on hardware with a cpu
>>>>>>>>>>>>> that doesn't have constant_tsc then you might have some problems. ?Just play
>>>>>>>>>>>>> with the different timer options until you find the one that works.
>>>>>>>>>>>>>
>>>>>>>>>>>>> HTH
>>>>>>>>>>>>> --FC
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at 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
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at 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
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> 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


From anthony.minessale at gmail.com  Thu Feb  3 22:14:25 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 3 Feb 2011 13:14:25 -0600
Subject: [Freeswitch-users] Choppy VM when using loopback (on Debian
	lenny)
In-Reply-To: <AANLkTim5Ha1YC2W5a6tygWtMg_LeT7BCMRbr6Z0bXywZ@mail.gmail.com>
References: <4D42B8EE.7030208@estation.dk>
	<7C6F3FC7-445F-4157-BD59-D76A705E4CE7@carmickle.com>
	<9894FEC6-E451-4993-A85A-2CD910E916B1@ipeva.fr>
	<AANLkTim+aBjRiAvxDUF1O1vut_CJpqgct2OqCNVxi0q4@mail.gmail.com>
	<AANLkTiktckovGv4Ksnn_56EzpvTTSnfG+iD1vPX1_4RM@mail.gmail.com>
	<4D466CAE.7030402@estation.dk>
	<AANLkTimhOKmy_yupGtA2HBY_8wXtQKZnDYJ9Ye76bhzq@mail.gmail.com>
	<AANLkTiktQU7kOG_22nG=_oEHSrBad667jcTzBs3n2nQv@mail.gmail.com>
	<AANLkTiko=qTZbXCL09D-+W-FmibMnBjOeoR30YYzHe0i@mail.gmail.com>
	<AANLkTi=t8AiZG6axtfqmMe9ZHTPNowoAb=7eNjQ-RPoG@mail.gmail.com>
	<AANLkTiniBAWw+JMe=64PtAER9B9U7rbbkJ9X27fOfYBn@mail.gmail.com>
	<AANLkTinRXmR9iHbWvFRrbpxDyocD3g8hTmZ0s_mD9cyd@mail.gmail.com>
	<AANLkTikDD-1bFEgt0LUnt9zt4q-PqHFZ8Ut0CrUeRMre@mail.gmail.com>
	<AANLkTi=Z1P+0C9BvEnOSdQjnu7Va-SmY3Z5+1KSPRFOv@mail.gmail.com>
	<AANLkTikkLevVzpoM3YkY5Kvhu_xEMkwGorS08Gr89Rn0@mail.gmail.com>
	<AANLkTim5Ha1YC2W5a6tygWtMg_LeT7BCMRbr6Z0bXywZ@mail.gmail.com>
Message-ID: <AANLkTi=0vJC4n2Zr0n11pTJ+FHkvg2mDtwvYHqMYmM0A@mail.gmail.com>

you can just call sofia/profilename/ext@<local ip> even when its the
same profile.


On Thu, Feb 3, 2011 at 1:06 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> Yes, I also had this idea, but I thought it would lead to troubles
> with loop detection.
> How this can be done exactly? I need separate profiles for this? Or I
> just can dial from internal to internal?
>
> And what if I use Kamailio in front of FS? Then need to deal with call
> re-routing to kamailio. I thought it may lead to spiral detection, if
> I understand right what spiral is.
>
> 2011/2/3 Anthony Minessale <anthony.minessale at gmail.com>:
>> maybe, but using loopback without the bowout is very costly.
>> you can also consider hair-pinning a call back to your own box over
>> SIP for this type of thing.
>>
>>
>> On Thu, Feb 3, 2011 at 12:28 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>> Thanks, I will test tomorrow.
>>>
>>> I'm wondering if I could use att_xfer with loopbacks after these patches:D
>>>
>>> 2011/2/3 Anthony Minessale <anthony.minessale at gmail.com>:
>>>> commit fb66abfab4a74055c38cdc67da83e6e0175a4a0b
>>>> Author: Anthony Minessale <anthm at freeswitch.org>
>>>> Date: ? Tue Feb 1 16:22:36 2011 -0600
>>>>
>>>> this is the one you need at a bare minimum.
>>>>
>>>> If you are still having problems the most likely cause is ptime
>>>> mismatch between the 2 ends of the call.
>>>> you cannot leave loopback in the call path when the legs are not on
>>>> the same ptime with very good results.
>>>>
>>>>
>>>> On Thu, Feb 3, 2011 at 4:31 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>>>> We faced same trouble with FreeSWITCH Version 1.0.head (git-33848eb
>>>>> 2011-02-01 08-56-26 +0100), the problem persists with loopback
>>>>> channels. Even if two sip endpoints are bridged via loopback, sound is
>>>>> choppy.
>>>>>
>>>>> We have bridge_early_media=true on loopback channel bridge command and
>>>>> loopback_bowout=false, loopback_bowout_on_execute=false
>>>>>
>>>>> 2011/2/2 Dan Lane <null at invalid.name>:
>>>>>> This seems to have resolved the issue for us :)
>>>>>>
>>>>>> Thanks.
>>>>>>
>>>>>> On Tue, Feb 1, 2011 at 8:47 PM, Anthony Minessale
>>>>>> <anthony.minessale at gmail.com> wrote:
>>>>>>> Try the latest GIT, I reverted the last patch and tried to solve the
>>>>>>> problem differently.
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Feb 1, 2011 at 2:11 PM, Dan Lane <null at invalid.name> wrote:
>>>>>>>> Yes, using 1f1541b our calls that use mod_loopback into mod_conference
>>>>>>>> are unusable. The audio is choppy and delay increases as time passes.
>>>>>>>>
>>>>>>>> I thought it was an issue relating to the default 100Hz kernel on EC2
>>>>>>>> so I spent some time yesterday putting together a 1000Hz kernel but it
>>>>>>>> didn't make any difference. In the meantime I've compiled mod_loopback
>>>>>>>> from 4c5426f and loaded it with my 1f1541b build which eliminates the
>>>>>>>> issue.
>>>>>>>>
>>>>>>>> I haven't added it to Jira yet as I want to spend some time debugging
>>>>>>>> it (and I also owe you some debug info for FS-2934) but the problem is
>>>>>>>> definitely there.
>>>>>>>>
>>>>>>>> On Tue, Feb 1, 2011 at 3:58 PM, Anthony Minessale
>>>>>>>> <anthony.minessale at gmail.com> wrote:
>>>>>>>>> Are you saying you have better results on that version than you do on
>>>>>>>>> the latest?
>>>>>>>>> What conditions do you have that cause you trouble, what is the
>>>>>>>>> endpoint on the other side.
>>>>>>>>>
>>>>>>>>> If the last commit to mod_loopback intended to improve audio quality
>>>>>>>>> actually makes it worse I need to investigate it.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Tue, Feb 1, 2011 at 5:53 AM, Dan Lane <null at invalid.name> wrote:
>>>>>>>>>> FWIW we've been also been having audio issues with loopback recently
>>>>>>>>>> on EC2 (with a 1000Hz kernel).
>>>>>>>>>>
>>>>>>>>>> We worked around it in the short term by reverting mod_loopback to
>>>>>>>>>> git-4c5426f during the build process.
>>>>>>>>>>
>>>>>>>>>> For anyone else who wants to try this just run "git checkout 4c5426f"
>>>>>>>>>> in src/mod/endpoints/mod_loopback then build as usual. This is NOT a
>>>>>>>>>> long term solution though.
>>>>>>>>>>
>>>>>>>>>> On Mon, Jan 31, 2011 at 8:02 AM, ?yvind Albrigtsen <oa at estation.dk> wrote:
>>>>>>>>>>> Thanks for all your feedback. I'll keep on trying and inform you what
>>>>>>>>>>> worked for me.
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> Regards
>>>>>>>>>>> Oyvind
>>>>>>>>>>>
>>>>>>>>>>> On 2011-01-29 21:48, Anthony Minessale wrote:
>>>>>>>>>>>> Everyone should try latest GIT before pondering any further because I
>>>>>>>>>>>> added a patch like 2 days ago to adress this issue.
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank at telonium.com> ?wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> Yeah. I, too, don't see the correlation between the NTP and hw timer..
>>>>>>>>>>>>> I am not familiar with the timer_test command and what it's measuring, but
>>>>>>>>>>>>> of the 50 tests it ran, min is 19089 and max is 20713.
>>>>>>>>>>>>> Frank
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone at ipeva.fr>
>>>>>>>>>>>>> wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>> Frank,
>>>>>>>>>>>>>> I fail to see the relationship between the hw timer and NTP.
>>>>>>>>>>>>>> Can you please elaborate ?
>>>>>>>>>>>>>> David Ponzone ?Direction Technique
>>>>>>>>>>>>>> email: david.ponzone at ipeva.fr
>>>>>>>>>>>>>> tel: ? ? ?01 74 03 18 97
>>>>>>>>>>>>>> gsm: ? 06 66 98 76 34
>>>>>>>>>>>>>> Service Client IPeva
>>>>>>>>>>>>>> tel: ? ? ?0811 46 26 26
>>>>>>>>>>>>>> www.ipeva.fr ?- ? www.ipeva-studio.com
>>>>>>>>>>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>>>>>>>>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>>>>>>>>>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>>>>>>>>>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>>>>>>>>>>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>>>>>>>>>>>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Le 28/01/2011 ? 19:49, Frank Carmickle a ?crit :
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Hi
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Jan 28, 2011, at 7:39 AM, ?yvind Albrigtsen wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Hi,
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> I'm using latest git-version of Freeswitch, and when I go to voicemail
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> when calling a number the sound playback is choppy and it skips some of
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> the digits in the number I called.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> What kind of results do you get from timer_test at the fs_cli? ?Are you
>>>>>>>>>>>>>> running on hardware or are you virtualized? ?What is your clock source set
>>>>>>>>>>>>>> to and what are your available clock source options? ?See
>>>>>>>>>>>>>> /sys/devices/system/clocksource/clocksource0/available_clocksource and
>>>>>>>>>>>>>> /sys/devices/system/clocksource/clocksource0/current_clocksource. ?I am
>>>>>>>>>>>>>> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to
>>>>>>>>>>>>>> hang at 19998/19999 which works very well for me. ?When I was having problem
>>>>>>>>>>>>>> it was reporting numbers all over the map from 17400 to 22600 with lots of
>>>>>>>>>>>>>> randomness in between. ?I have my clocksource set to jiffies and xen
>>>>>>>>>>>>>> independent wallclock set to 1. ?Of course at that point you need to have
>>>>>>>>>>>>>> ntp running against a bunch of servers to drive your clock nice and steady.
>>>>>>>>>>>>>> ? I know my set up is probably a lot different than yours but I thought I'd
>>>>>>>>>>>>>> toss it out there to show that some of the harshest conditions can be dealt
>>>>>>>>>>>>>> with and don't give up trying. ?If you are running on hardware with a cpu
>>>>>>>>>>>>>> that doesn't have constant_tsc then you might have some problems. ?Just play
>>>>>>>>>>>>>> with the different timer options until you find the one that works.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> HTH
>>>>>>>>>>>>>> --FC
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>> FreeSWITCH-users at 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
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at 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
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> 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 mthakershi at gmail.com  Thu Feb  3 22:37:32 2011
From: mthakershi at gmail.com (Malay Thakershi)
Date: Thu, 3 Feb 2011 13:37:32 -0600
Subject: [Freeswitch-users] Session sleep method
In-Reply-To: <AANLkTikNbbBF1_g2VimrpdLzR8Ro7_VQ0oHMviYEinkx@mail.gmail.com>
References: <AANLkTikNbbBF1_g2VimrpdLzR8Ro7_VQ0oHMviYEinkx@mail.gmail.com>
Message-ID: <AANLkTimCayT0bKPjArwSqDzuKEf_jMEhgFMP4wzgWs0b@mail.gmail.com>

Anyone?

On Wed, Feb 2, 2011 at 6:45 PM, Malay Thakershi <mthakershi at gmail.com>wrote:

> Hello,
>
> I am using .NET module. There is sleep method that has two arguments.
>
> What is second argument for?
>
> I use it:
> mObjMainSession.sleep(200, 0);
>
> What happens if I pass something other than zero as second argument?
>
> I am having inconsistent delays between call paths and that is why I am
> exploring this argument.
>
> Thank you for any help.
>
> Malay
>
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From anthony.minessale at gmail.com  Thu Feb  3 23:41:25 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 3 Feb 2011 14:41:25 -0600
Subject: [Freeswitch-users] Session sleep method
In-Reply-To: <AANLkTimCayT0bKPjArwSqDzuKEf_jMEhgFMP4wzgWs0b@mail.gmail.com>
References: <AANLkTikNbbBF1_g2VimrpdLzR8Ro7_VQ0oHMviYEinkx@mail.gmail.com>
	<AANLkTimCayT0bKPjArwSqDzuKEf_jMEhgFMP4wzgWs0b@mail.gmail.com>
Message-ID: <AANLkTi=PW57Pzb-9jS-8Q9M6USm6rYCEEL7d4dbqjZG2@mail.gmail.com>

I'm tempted to not reply due to the impatience.
the 2nd argument is weather or not to flush the input buffers on the rtp stack.
If this is done, any udp packets that have not been read and may be
queued in the udp buffers would be flushed.


On Thu, Feb 3, 2011 at 1:37 PM, Malay Thakershi <mthakershi at gmail.com> wrote:
> Anyone?
>
> On Wed, Feb 2, 2011 at 6:45 PM, Malay Thakershi <mthakershi at gmail.com>
> wrote:
>>
>> Hello,
>> I am using .NET module. There is sleep method that has two arguments.
>> What is second argument for?
>> I use it:
>> mObjMainSession.sleep(200, 0);
>> What happens if I pass something other than zero as second argument?
>> I am having inconsistent delays between call paths and that is why I am
>> exploring this argument.
>> Thank you for any help.
>> 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


From fs-list at communicatefreely.net  Thu Feb  3 23:48:21 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Thu, 03 Feb 2011 15:48:21 -0500
Subject: [Freeswitch-users] Enterprise bridge not continuing?
Message-ID: <4D4B1495.30608@communicatefreely.net>

Hello,

I'm trying to implement a ring group scenario that supports multiple 
registrations per user.

I have been able to get it to work (mostly) by using the enterprise 
bridge syntax, but when I do, I can't continue on in the dialplan to go 
to voicemail, for example.

If I change the syntax and use the parallell dial comma, voice mail 
works fine, but only the first registration of each user entry will 
ring, and I would really like them all to ring.

 From what I can tell, dialplan execution stops after the originate, 
rather than continuing on to process the next steps.  Is there some sort 
of variable I have to set to tell it to continue when using enterprise 
bridge?

Here's the XML debug output of the dialplan I'm sending from xml_curl

<?xml version="1.0"?>
<document type="freeswitch/xml">
 <section name="dialplan">
  <context name="internal">
   <extension name="Intercom" description="Local Extension" continue="true">
    <condition field="destination_number" expression="5240">
     <action application="pre_answer" data=""/>
     <action application="set" data="continue=true"/>
     <action application="set" data="cdr_destination=Internal"/>
     <action application="set" data="ringback=$${us-ring}"/>
     <action application="set" data="sip_callee_id_name=Lab 0"/>
     <action application="set" data="sip_callee_id_number=5240"/>
     <action application="set" data="progress_timeout=4"/>
     <action application="set" data="call_timeout=12"/>
     <action application="set" data="hangup_after_bridge=true"/>
     <action application="set" data="cdr_amaflags=INTERNAL"/>
     <action application="set" data="originate_continue_on_timeout=true"/>
     <action application="set" data="originate_disposition=NO_ANSWER"/>
     <action application="set" data="ignore_early_media=true"/>
     <action application="log" data="INFO Adding user 5242to the group"/>
     <action application="log" data="INFO Adding user 5244to the group"/>
     <action application="bridge" 
data="[call_timeout=12,alert_info=${domain};info=alert-internal]user/5240 at communicatefreely.net:_:[ignore_early_media=true,call_timeout=12,origination_caller_id_name=Cordles-${caller_id_name},origination_caller_id_number=${caller_id_number},export_vars=accountcode 
subdomain,alert_info=${domain};info=alert-group]user/5242@${domain}:_:[ignore_early_media=true,call_timeout=12,origination_caller_id_name=Cordles-${caller_id_name},origination_caller_id_number=${caller_id_number},export_vars=accountcode 
subdomain,alert_info=${domain};info=alert-group]user/5244@${domain}"/>
    </condition>
   </extension>
   <extension name="Voicemail" description="Leave VM message" 
continue="true">
    <condition field="destination_number" expression="5240">
     <action application="set" data="playback_terminators=#0"/>
     <action application="set" data="cdr_destination=VM Box 5240"/>
     <action application="answer" data=""/>
     <action application="set" data="vm_busy_greet=0"/>
     <action application="set" data="vm_unavail_greet=0"/>
     <action application="set" data="vm_direct__greet=0"/>
     <action application="set" data="vm_ivr_enable=false"/>
     <action application="lua" data="vm_leave.lua 5240 lab new"/>
    </condition>
   </extension>
  </context>
 </section>
</document>



From marcin321 at gmail.com  Thu Feb  3 23:47:35 2011
From: marcin321 at gmail.com (Marcin Wojtowicz)
Date: Thu, 3 Feb 2011 15:47:35 -0500
Subject: [Freeswitch-users] iLBC to PCMU transcoding broken on windows?
Message-ID: <AANLkTimWv26=0XppqzZMV8yf_0QKLVCHMenTf4c27Q8X@mail.gmail.com>

I built freeswitch from the latest git on my windows machine, and it looks
like transcoding from iLBC to PCMU is broken, and I hear only squeal on the
line. I have only PCMU on my external profile and iLBC on my internal
profiles, so no codec mismatch is possible, and all my codec settings are
compatible with the devices that I'm using. I inspected the traffic flow
with wireshark, and there is nothing incorrect there, SIP messages and SDP
is ok, and traffic looks like my device <-> iLBC <-> freeswitch <-> PCMU <->
voip.ms. I've searched around and it looks like my issue is identical to the
discussion here:
http://lists.freeswitch.org/pipermail/freeswitch-users/2010-October/063891.html.
If I run PCMU on both ends, there are no problems and if I call into
freeswitch voicemail on iLBC, there are no problems either.
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From mthakershi at gmail.com  Thu Feb  3 23:56:34 2011
From: mthakershi at gmail.com (Malay Thakershi)
Date: Thu, 3 Feb 2011 14:56:34 -0600
Subject: [Freeswitch-users] Session sleep method
In-Reply-To: <AANLkTi=PW57Pzb-9jS-8Q9M6USm6rYCEEL7d4dbqjZG2@mail.gmail.com>
References: <AANLkTikNbbBF1_g2VimrpdLzR8Ro7_VQ0oHMviYEinkx@mail.gmail.com>
	<AANLkTimCayT0bKPjArwSqDzuKEf_jMEhgFMP4wzgWs0b@mail.gmail.com>
	<AANLkTi=PW57Pzb-9jS-8Q9M6USm6rYCEEL7d4dbqjZG2@mail.gmail.com>
Message-ID: <AANLkTingpN0phQt6f7BZQfn8T1W5iCwN_NJyq9Eok7KS@mail.gmail.com>

I am sorry if you thought I was impatient.

It was 20 hours since my post so I thought I could revive it. Anyway, it is
up to you to reply or not. All I am looking for is to get past the issue and
on the way contribute something that is "knowledge" for FS community.

So, 1 = flush packets and 0 = do not flush?

Malay



On Thu, Feb 3, 2011 at 2:41 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I'm tempted to not reply due to the impatience.
> the 2nd argument is weather or not to flush the input buffers on the rtp
> stack.
> If this is done, any udp packets that have not been read and may be
> queued in the udp buffers would be flushed.
>
>
> On Thu, Feb 3, 2011 at 1:37 PM, Malay Thakershi <mthakershi at gmail.com>
> wrote:
> > Anyone?
> >
> > On Wed, Feb 2, 2011 at 6:45 PM, Malay Thakershi <mthakershi at gmail.com>
> > wrote:
> >>
> >> Hello,
> >> I am using .NET module. There is sleep method that has two arguments.
> >> What is second argument for?
> >> I use it:
> >> mObjMainSession.sleep(200, 0);
> >> What happens if I pass something other than zero as second argument?
> >> I am having inconsistent delays between call paths and that is why I am
> >> exploring this argument.
> >> Thank you for any help.
> >> 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  4 00:17:47 2011
From: ayhkor at gmail.com (deniro)
Date: Thu, 3 Feb 2011 16:17:47 -0500
Subject: [Freeswitch-users] mod_perl mod_lua
In-Reply-To: <AANLkTik0QYcDP-YZ6RSxdipXxYJQ7LH32G45P6zJm8+E@mail.gmail.com>
References: <AANLkTikxbN47Nc7cG8y0HMDVqC0HHJ56aLQZHJYS2WTR@mail.gmail.com>
	<AANLkTinDSQkOJrAikHrTEm6YTfyMcv2yLfxGPucjDwVq@mail.gmail.com>
	<AANLkTi=0=+05cxkP8ER9WKmzgAS4HzymLrTHcrMVho6P@mail.gmail.com>
	<AANLkTi=bNWH97qZdDmuEvzm6ETaZDRVv1R3Rt7hLTg+L@mail.gmail.com>
	<AANLkTinJBQ_EskC4Fv72FauNgbr4r4fF_gnOjiD6NDSe@mail.gmail.com>
	<AANLkTik0QYcDP-YZ6RSxdipXxYJQ7LH32G45P6zJm8+E@mail.gmail.com>
Message-ID: <AANLkTimfGCaJxdgrcp1B-9E6EFUO4p75u7CRHPTdA62F@mail.gmail.com>

Paul and Steve thanks many for your great assistance
have couple of puzzles;

1--
What are these packages for?
 freeswitch-lua  and freeswitch-perl (if I am installing  mod_perl and
mod_lua from source)
I installed them anyway.

2--
I downloaded freeswitch 1.0.6 source with wget
('apt-get source freeswitch'  couldn't find the source)
uncommented lua/perl from modules.conf  and run configure, then
make mod_lua-install
make mod_perl-install

mod_lua installed  without problem
but
make mod_perl-install  had  some errors as below

Creating mod_perl.so...
/usr/bin/ld: cannot find -lperl
collect2: ld returned 1 exit status
gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -DDEBIAN
-fno-strict-aliasing -pipe -fstack-protector -I/usr/local/include
-D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -I/usr/lib/perl/5.10/CORE
-DEMBED_PERL -I/usr/src/freeswitch-1.0.6/src/include
-I/usr/src/freeswitch-1.0.6/src/include
-I/usr/src/freeswitch-1.0.6/libs/libteletone/src -fPIC -Werror
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
-g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE
-shared -o .libs/mod_perl.so -shared -Wl,-x .libs/mod_perl.o -Wl,-E
-fstack-protector freeswitch_perl.o mod_perl_wrap.o perlxsi.o
/usr/src/freeswitch-1.0.6/.libs/libfreeswitch.so
-L/usr/src/freeswitch-1.0.6/libs/apr-util/xml/expat/lib
/usr/src/freeswitch-1.0.6/libs/apr-util/xml/expat/lib/.libs/libexpat.a
/usr/src/freeswitch-1.0.6/libs/apr/.libs/libapr-1.a -lrt
-L/usr/src/freeswitch-1.0.6/libs/srtp -lncurses -L/usr/local/lib
-L/usr/lib/perl/5.10/CORE -lperl -lgdbm -lgdbm_compat -ldb -ldl -lm
-lpthread -lc -lcrypt   -Wl,--rpath -Wl,/opt/freeswitch/lib -Wl,--rpath
-Wl,/opt/freeswitch/mod
make[3]: *** [mod_perl.so] Error 1
make[2]: *** [install] Error 1
make[1]: *** [mod_perl-install] Error 1
make: *** [mod_perl-install] Error 2
what am I missing here? (I have perl installed and freeswitch-perl instaled)

3-- how do I check if any module is installed or not? same for the load,
i.e, any command to check wheether or not it is loaded
if not loaded,  how you load a module (after editing modules.conf.xml what
we do? stop/start freeswitch?)

thx
deniro--




On Wed, Feb 2, 2011 at 11:46 AM, Steven Ayre <steveayre at gmail.com> wrote:

>  Check first whether there are freeswitch-lua or freeswitch-perl packages.
>
> The official debian/ubuntu build system does build both mod_lua and
> mod_perl, but it places them in separate packages from the rest of
> FreeSWITCH.
>
> If you don't you'll need to compile them yourself...
>
> You will need the FreeSWITCH source to compile the modules, since they are
> in that source. Some of FreeSWITCH (the core) must also be compiled since
> the modules use functions that are in the core - however you can restrict
> the modules that are compiled to just the ones you need using modules.conf.
>
> When you checkout/extract the FreeSWITCH source, the first thing you must
> do is run bootstrap.sh. It is this file that generates the modules.conf
> file, if you haven't created it yourself.
>
> Unless you plan to upgrade (and you should think about doing so - 1.0.6 is
> old now and git head has hundreds  of bugfixes and new features) you should
> make sure the one you build the modules on is the exact same version the
> Ubuntu version was created from, otherwise you'll find that the interface
> between the core and the modules may have changed and you'll either get a
> unloadable or unstable module. If you installed via APT then you should be
> able to use 'apt-get source freeswitch' to get the source package that'll
> give you the version they used.
>
> -Steve
>
>
> On 2 February 2011 16:26, deniro <ayhkor at gmail.com> wrote:
>
>> thanks for your advice and  I appreciate you taking time to reply.
>> You also note that I first check freeswitch site and I put my questions if
>> I cant find a solutions myself.
>> I already checked for options with modules.conf and modules.conf.xml even
>> before posting.
>>
>> If you look at my first posting, I stated that I have a compiled
>> freeswitch and I dont even see modules.conf file (I searched).
>> so to reiterate, The freeswitch comes installed and compiled already with
>> a product, so I dont even have the freeswitch source.
>>
>> All I am loooking for is, if there is any way, to install new modules
>> without re-installig from source and recompile from the scratch with
>> existing freeswitch install.
>> I highly doubt that this is possible but I am checking out with the gurus
>> here.
>> I dont wanna break already running freeswitch with custom dialplans and
>> other custom configurations.
>>
>> thx again
>> deniro--
>>
>>
>>
>>
>>
>>
>>
>> On Wed, Feb 2, 2011 at 10:32 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:
>>
>>> On Wed, Feb 2, 2011 at 4:22 PM, deniro <ayhkor at gmail.com> wrote:
>>> > I think that is to enable loading the module(not the install)
>>> > thx
>>>
>>> So, you write to the mailing list for advice, and you don't put
>>> confidence in answers you got back? ;)
>>>
>>> >
>>> >
>>> > On Wed, Feb 2, 2011 at 3:14 AM, Giovanni Maruzzelli <gmaruzz at gmail.com
>>> >
>>> > wrote:
>>> >>
>>> >> On Wed, Feb 2, 2011 at 5:25 AM, deniro <ayhkor at gmail.com> wrote:
>>> >> > Hi
>>> >> > I have compiled version of freeswitch 1.6.x on ubuntu 10.04
>>> >> > I want to install extra freeswitch modules like mod_perl and mod_lua
>>> >> > Is there any way to install these modules without re-compileling
>>> >> > freeswitch.
>>> >> > I looked at freeswitch site and it is talking about  editing
>>> >> > modules.conf
>>> >> > file and compiling from source
>>> >> > and I dont even have modules.conf file as I searched.
>>> >>
>>> >> Go to your original sources directory (where you gave the command
>>> >> "make install"), edit the file modules.conf.xml and give the command
>>> >> "make install".
>>> >>
>>> >> -giovanni
>>> >>
>>> >>
>>> >>
>>> >> >
>>> >> > thx
>>> >> > 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
>>> >> >
>>> >> >
>>> >>
>>> >>
>>> >>
>>> >> --
>>> >> 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
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at 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
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Feb  4 00:37:00 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 3 Feb 2011 15:37:00 -0600
Subject: [Freeswitch-users] ANNOUNCEMENT CudaTel IP PBX (powered by
	FreeSWITCH) Reaches Critical Mass
Message-ID: <AANLkTin7xCfqk1kyyyQMW-VZQo2t9MJ9c++_Zjc=FgER@mail.gmail.com>

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

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

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



Check it out at 


-- 
Anthony Minessale II

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

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

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


From david.ponzone at ipeva.fr  Fri Feb  4 00:54:11 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 3 Feb 2011 22:54:11 +0100
Subject: [Freeswitch-users] Strange behaviour with leg_timeout
Message-ID: <240D7F48-8423-42A5-A357-632F0AB54412@ipeva.fr>

It's my time to mess with XXX_timeout variables in FreeSWITCH.

What I need to do is to route a call to a gateway, and if I don't have an answer (like not even the 100) within X seconds, I try to route the call to a backup number.
I tried it this way:

    <extension name="gw_then_backup">
       <condition field="destination_number" expression="^(XXXXXXXXXX)$">
        <action application="bridge" data="{leg_timeout=5}sofia/external/$1 at a.b.c.d|sofia/gateway/provider.out/YYYYYYYYYY"/>
       </condition>
     </extension>

My understanding was that a.b.c.d would be tried for 5 seconds, and if no 100 is received, gateway provider.out would then be used.
It works...but after the first INVITE to provider.out (which sends back 100/180/183), I can see some more INVITEs sent to a.b.c.d.

I am wondering if I am doing something wrong, or if there is an issue.
I use latest git from 5 minutes ago.

Thanks

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




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From msc at freeswitch.org  Fri Feb  4 00:56:04 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 3 Feb 2011 13:56:04 -0800
Subject: [Freeswitch-users] How to record a conference using javascript
	application
In-Reply-To: <AANLkTimqZrM+Pe=uQvFVyfqcyy_4WOOm5Qfio0tSaTzF@mail.gmail.com>
References: <1296664102854-5985457.post@n2.nabble.com>
	<AANLkTikZg=tYFKC61EoiFP7rbrQBv0NswzH41tqDS8sE@mail.gmail.com>
	<AANLkTimqZrM+Pe=uQvFVyfqcyy_4WOOm5Qfio0tSaTzF@mail.gmail.com>
Message-ID: <AANLkTi=07b2ua+OKtjwjC8382QZ8SC=aisygL8JumQgQ@mail.gmail.com>

On Wed, Feb 2, 2011 at 11:10 PM, kapil.rastogi
<kapil.rastogi at telemune.net>wrote:

> Hi,
>
> I have also tried for the same, but it is also not working.
>
> Could you get some logs and put them on pastebin so we can see what's
happening?
-MC
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From msc at freeswitch.org  Fri Feb  4 01:00:56 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 3 Feb 2011 14:00:56 -0800
Subject: [Freeswitch-users] conference audio in last git
In-Reply-To: <210D70D652874D9E93576984B739281D@e1705>
References: <210D70D652874D9E93576984B739281D@e1705>
Message-ID: <AANLkTikRVSbut7nkDiW9YtXfDnUEM591HSXZ_qV=dSXr@mail.gmail.com>

Keep updating. Tony has been working on a few things and it's possible you
got on a version in between fixes. If that doesn't work then try to identify
the version where things go wonky on you.

-MC

On Thu, Feb 3, 2011 at 3:05 AM, Madovsky <infos at madovsky.org> wrote:

>  seems that the audio (audio files and voice)
> is less stable as before in conference only.
> some packet loss or like a noise gate with a too high threeshold
> also without CNG there's noise between ivr voice
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  4 01:01:05 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 3 Feb 2011 22:01:05 +0000
Subject: [Freeswitch-users] mod_perl mod_lua
In-Reply-To: <AANLkTimfGCaJxdgrcp1B-9E6EFUO4p75u7CRHPTdA62F@mail.gmail.com>
References: <AANLkTikxbN47Nc7cG8y0HMDVqC0HHJ56aLQZHJYS2WTR@mail.gmail.com>
	<AANLkTinDSQkOJrAikHrTEm6YTfyMcv2yLfxGPucjDwVq@mail.gmail.com>
	<AANLkTi=0=+05cxkP8ER9WKmzgAS4HzymLrTHcrMVho6P@mail.gmail.com>
	<AANLkTi=bNWH97qZdDmuEvzm6ETaZDRVv1R3Rt7hLTg+L@mail.gmail.com>
	<AANLkTinJBQ_EskC4Fv72FauNgbr4r4fF_gnOjiD6NDSe@mail.gmail.com>
	<AANLkTik0QYcDP-YZ6RSxdipXxYJQ7LH32G45P6zJm8+E@mail.gmail.com>
	<AANLkTimfGCaJxdgrcp1B-9E6EFUO4p75u7CRHPTdA62F@mail.gmail.com>
Message-ID: <AANLkTiktkYAXB7i_+0xrabYe5UZDWf=X7KBv1yc3KRGB@mail.gmail.com>

"What are these packages for?
 freeswitch-lua  and freeswitch-perl (if I am installing  mod_perl and
mod_lua from source)
I installed them anyway."

They're the packaged versions of mod_lua and mod_perl. If you have those
packages you do not need to compile from source.

"make mod_perl-install  had  some errors as below...
Creating mod_perl.so...
/usr/bin/ld: cannot find -lperl...
what am I missing here? (I have perl installed and freeswitch-perl
instaled)"

perl-dev

"3-- how do I check if any module is installed or not? same for the load,
i.e, any command to check wheether or not it is loaded
if not loaded,  how you load a module (after editing modules.conf.xml what
we do? stop/start freeswitch?)"

Installed: ls /opt/freeswitch/mod (at linux cli)
Loaded: show codecs (at fs cli)


-Steve




On 3 February 2011 21:17, deniro <ayhkor at gmail.com> wrote:

> Paul and Steve thanks many for your great assistance
> have couple of puzzles;
>
> 1--
> What are these packages for?
>  freeswitch-lua  and freeswitch-perl (if I am installing  mod_perl and
> mod_lua from source)
> I installed them anyway.
>
> 2--
> I downloaded freeswitch 1.0.6 source with wget
> ('apt-get source freeswitch'  couldn't find the source)
> uncommented lua/perl from modules.conf  and run configure, then
> make mod_lua-install
> make mod_perl-install
>
> mod_lua installed  without problem
> but
> make mod_perl-install  had  some errors as below
>
> Creating mod_perl.so...
> /usr/bin/ld: cannot find -lperl
> collect2: ld returned 1 exit status
> gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -DDEBIAN
> -fno-strict-aliasing -pipe -fstack-protector -I/usr/local/include
> -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -I/usr/lib/perl/5.10/CORE
> -DEMBED_PERL -I/usr/src/freeswitch-1.0.6/src/include
> -I/usr/src/freeswitch-1.0.6/src/include
> -I/usr/src/freeswitch-1.0.6/libs/libteletone/src -fPIC -Werror
> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
> -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE
> -shared -o .libs/mod_perl.so -shared -Wl,-x .libs/mod_perl.o -Wl,-E
> -fstack-protector freeswitch_perl.o mod_perl_wrap.o perlxsi.o
> /usr/src/freeswitch-1.0.6/.libs/libfreeswitch.so
> -L/usr/src/freeswitch-1.0.6/libs/apr-util/xml/expat/lib
> /usr/src/freeswitch-1.0.6/libs/apr-util/xml/expat/lib/.libs/libexpat.a
> /usr/src/freeswitch-1.0.6/libs/apr/.libs/libapr-1.a -lrt
> -L/usr/src/freeswitch-1.0.6/libs/srtp -lncurses -L/usr/local/lib
> -L/usr/lib/perl/5.10/CORE -lperl -lgdbm -lgdbm_compat -ldb -ldl -lm
> -lpthread -lc -lcrypt   -Wl,--rpath -Wl,/opt/freeswitch/lib -Wl,--rpath
> -Wl,/opt/freeswitch/mod
> make[3]: *** [mod_perl.so] Error 1
> make[2]: *** [install] Error 1
> make[1]: *** [mod_perl-install] Error 1
> make: *** [mod_perl-install] Error 2
> what am I missing here? (I have perl installed and freeswitch-perl
> instaled)
>
>

> 3-- how do I check if any module is installed or not? same for the load,
> i.e, any command to check wheether or not it is loaded
> if not loaded,  how you load a module (after editing modules.conf.xml what
> we do? stop/start freeswitch?)
>
> thx
> deniro--
>
>
>
>
> On Wed, Feb 2, 2011 at 11:46 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>>  Check first whether there are freeswitch-lua or freeswitch-perl
>> packages.
>>
>> The official debian/ubuntu build system does build both mod_lua and
>> mod_perl, but it places them in separate packages from the rest of
>> FreeSWITCH.
>>
>> If you don't you'll need to compile them yourself...
>>
>> You will need the FreeSWITCH source to compile the modules, since they are
>> in that source. Some of FreeSWITCH (the core) must also be compiled since
>> the modules use functions that are in the core - however you can restrict
>> the modules that are compiled to just the ones you need using modules.conf.
>>
>> When you checkout/extract the FreeSWITCH source, the first thing you must
>> do is run bootstrap.sh. It is this file that generates the modules.conf
>> file, if you haven't created it yourself.
>>
>> Unless you plan to upgrade (and you should think about doing so - 1.0.6 is
>> old now and git head has hundreds  of bugfixes and new features) you should
>> make sure the one you build the modules on is the exact same version the
>> Ubuntu version was created from, otherwise you'll find that the interface
>> between the core and the modules may have changed and you'll either get a
>> unloadable or unstable module. If you installed via APT then you should be
>> able to use 'apt-get source freeswitch' to get the source package that'll
>> give you the version they used.
>>
>> -Steve
>>
>>
>> On 2 February 2011 16:26, deniro <ayhkor at gmail.com> wrote:
>>
>>> thanks for your advice and  I appreciate you taking time to reply.
>>> You also note that I first check freeswitch site and I put my questions
>>> if I cant find a solutions myself.
>>> I already checked for options with modules.conf and modules.conf.xml even
>>> before posting.
>>>
>>> If you look at my first posting, I stated that I have a compiled
>>> freeswitch and I dont even see modules.conf file (I searched).
>>> so to reiterate, The freeswitch comes installed and compiled already with
>>> a product, so I dont even have the freeswitch source.
>>>
>>> All I am loooking for is, if there is any way, to install new modules
>>> without re-installig from source and recompile from the scratch with
>>> existing freeswitch install.
>>> I highly doubt that this is possible but I am checking out with the gurus
>>> here.
>>> I dont wanna break already running freeswitch with custom dialplans and
>>> other custom configurations.
>>>
>>> thx again
>>> deniro--
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Feb 2, 2011 at 10:32 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:
>>>
>>>> On Wed, Feb 2, 2011 at 4:22 PM, deniro <ayhkor at gmail.com> wrote:
>>>> > I think that is to enable loading the module(not the install)
>>>> > thx
>>>>
>>>> So, you write to the mailing list for advice, and you don't put
>>>> confidence in answers you got back? ;)
>>>>
>>>> >
>>>> >
>>>> > On Wed, Feb 2, 2011 at 3:14 AM, Giovanni Maruzzelli <
>>>> gmaruzz at gmail.com>
>>>> > wrote:
>>>> >>
>>>> >> On Wed, Feb 2, 2011 at 5:25 AM, deniro <ayhkor at gmail.com> wrote:
>>>> >> > Hi
>>>> >> > I have compiled version of freeswitch 1.6.x on ubuntu 10.04
>>>> >> > I want to install extra freeswitch modules like mod_perl and
>>>> mod_lua
>>>> >> > Is there any way to install these modules without re-compileling
>>>> >> > freeswitch.
>>>> >> > I looked at freeswitch site and it is talking about  editing
>>>> >> > modules.conf
>>>> >> > file and compiling from source
>>>> >> > and I dont even have modules.conf file as I searched.
>>>> >>
>>>> >> Go to your original sources directory (where you gave the command
>>>> >> "make install"), edit the file modules.conf.xml and give the command
>>>> >> "make install".
>>>> >>
>>>> >> -giovanni
>>>> >>
>>>> >>
>>>> >>
>>>> >> >
>>>> >> > thx
>>>> >> > 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
>>>> >> >
>>>> >> >
>>>> >>
>>>> >>
>>>> >>
>>>> >> --
>>>> >> 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
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at 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
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Feb  4 01:02:07 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 3 Feb 2011 16:02:07 -0600
Subject: [Freeswitch-users] Strange behaviour with leg_timeout
In-Reply-To: <240D7F48-8423-42A5-A357-632F0AB54412@ipeva.fr>
References: <240D7F48-8423-42A5-A357-632F0AB54412@ipeva.fr>
Message-ID: <AANLkTikeBgwS_u+wt7c9qLJktZ1iqqSkUGtuQdNkVeW7@mail.gmail.com>

We do not have anything to timeout at a 100 the only thing we can do
is 180 or 183 using [leg_progress_timeout=X]  in front of each leg.



On Thu, Feb 3, 2011 at 3:54 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
> It's my time to mess with XXX_timeout variables in FreeSWITCH.
> What I need to do is to route a call to a gateway, and if I don't have an
> answer (like not even the 100) within X seconds, I try to route the call to
> a backup number.
> I tried it this way:
> ?? ?<extension name="gw_then_backup">
> ?? ? ? <condition field="destination_number" expression="^(XXXXXXXXXX)$">
> ?? ? ? ?<action application="bridge"
> data="{leg_timeout=5}sofia/external/$1 at a.b.c.d|sofia/gateway/provider.out/YYYYYYYYYY"/>
> ?? ? ? </condition>
> ?? ? </extension>
> My understanding was that a.b.c.d would be tried for 5 seconds, and if no
> 100 is received, gateway provider.out would then be used.
> It works...but after the first INVITE to provider.out (which sends back
> 100/180/183), I can see some more INVITEs sent to a.b.c.d.
> I am wondering if I am doing something wrong, or if there is an issue.
> I use latest git from 5 minutes ago.
> Thanks
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb  4 01:03:44 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 3 Feb 2011 14:03:44 -0800
Subject: [Freeswitch-users] Grandstream Freeze on Originate command
In-Reply-To: <1296751186605-5989433.post@n2.nabble.com>
References: <1296751186605-5989433.post@n2.nabble.com>
Message-ID: <AANLkTikPiZgq9WsFeMs2U0iZrT7FcL10dptLCFu5HynP@mail.gmail.com>

Is the GS registered with FreeSWITCH? If so then use "originate user/2000
1000" instead of what you have. If not then I'm unsure what else to try. I
know that we have mixed feelings toward GS phones. I personally don't have
any but I've heard nothing positive about them from any of the FS devs.

-MC

On Thu, Feb 3, 2011 at 8:39 AM, rex.alex <rajesh.npnr at yahoo.com> wrote:

>
> Hi,
>
> I am using the api originate command to establish an internal extn to extn
> call (originate sofia/internal/2000%10.10.22.31 1000), where the 2000
> registered in Grandstream GXP280 1.2.1.4 and 1000 registered in X-Lite.
>
> This command freezes the grandstream when I answer. I have pastebin the sip
> trace + freeswitch log in below mentioned url. Request your assistance in
> resolving this.
>
>
> http://pastebin.freeswitch.org/15222
>
> Thanks,
> Regards,
> Rex
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Grandstream-Freeze-on-Originate-command-tp5989433p5989433.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 anthony.minessale at gmail.com  Fri Feb  4 01:04:03 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 3 Feb 2011 16:04:03 -0600
Subject: [Freeswitch-users] conference audio in last git
In-Reply-To: <AANLkTikRVSbut7nkDiW9YtXfDnUEM591HSXZ_qV=dSXr@mail.gmail.com>
References: <210D70D652874D9E93576984B739281D@e1705>
	<AANLkTikRVSbut7nkDiW9YtXfDnUEM591HSXZ_qV=dSXr@mail.gmail.com>
Message-ID: <AANLkTikd6nM5vb_6AQYayO0rXXiAaa5rh3aGUb+TGAeL@mail.gmail.com>

We have not touched the conference code in like 2 months.
The noise gate "or energy level" code has not changed in eons.


On Thu, Feb 3, 2011 at 4:00 PM, Michael Collins <msc at freeswitch.org> wrote:
> Keep updating. Tony has been working on a few things and it's possible you
> got on a version in between fixes. If that doesn't work then try to identify
> the version where things go wonky on you.
> -MC
>
> On Thu, Feb 3, 2011 at 3:05 AM, Madovsky <infos at madovsky.org> wrote:
>>
>> seems that the audio (audio files and voice)
>> is less stable as before in conference only.
>> some packet loss or like a noise gate with a too high threeshold
>> also without CNG there's noise between ivr voice
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Fri Feb  4 01:06:44 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 3 Feb 2011 14:06:44 -0800 (PST)
Subject: [Freeswitch-users] iLBC to PCMU transcoding broken on windows?
In-Reply-To: <AANLkTimWv26=0XppqzZMV8yf_0QKLVCHMenTf4c27Q8X@mail.gmail.com>
References: <AANLkTimWv26=0XppqzZMV8yf_0QKLVCHMenTf4c27Q8X@mail.gmail.com>
Message-ID: <1296770804857-5990709.post@n2.nabble.com>


I asked on that thread for someone to open a Jira at that time but nobody
did. Please open a Jira about this problem so it doesnt get lost. I quickly
checked the problem and I see the same thing as you. Not sure what is wrong
yet.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/iLBC-to-PCMU-transcoding-broken-on-windows-tp5990434p5990709.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Fri Feb  4 01:14:37 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 3 Feb 2011 14:14:37 -0800
Subject: [Freeswitch-users] Enterprise bridge not continuing?
In-Reply-To: <4D4B1495.30608@communicatefreely.net>
References: <4D4B1495.30608@communicatefreely.net>
Message-ID: <AANLkTinpLVkH5Xhs6QWyzS8gMvreROdO_2Rhj2S_zTjR@mail.gmail.com>

Tim,

I don't see that you have set continue_on_fail in your dp. (
http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail)

Also, I'm curious to know if you deliberately are not ignoring early media
on the first leg, the one with user/5420 at communicatefreely.net. If you don't
ignore early media on that leg then that leg will "win" and get the call as
soon as media comes back. Just curious if you had a specific reason for
doing it this way.

-MC
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From anthony.minessale at gmail.com  Fri Feb  4 01:17:54 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 3 Feb 2011 16:17:54 -0600
Subject: [Freeswitch-users] Session sleep method
In-Reply-To: <AANLkTingpN0phQt6f7BZQfn8T1W5iCwN_NJyq9Eok7KS@mail.gmail.com>
References: <AANLkTikNbbBF1_g2VimrpdLzR8Ro7_VQ0oHMviYEinkx@mail.gmail.com>
	<AANLkTimCayT0bKPjArwSqDzuKEf_jMEhgFMP4wzgWs0b@mail.gmail.com>
	<AANLkTi=PW57Pzb-9jS-8Q9M6USm6rYCEEL7d4dbqjZG2@mail.gmail.com>
	<AANLkTingpN0phQt6f7BZQfn8T1W5iCwN_NJyq9Eok7KS@mail.gmail.com>
Message-ID: <AANLkTi=eN4x+kbD5opzLGorunek8UjJA_1Yn+6C5Chw5@mail.gmail.com>

ok well its against our ML etiquette so I am letting you know.  When
someone has time they will reply.
yes 1 = flush 0 = do not flush.  It's most likely nothing to do with
your problem.

I would examine the timing performance of your setup with the results from

"time_test 1000" and "timer_test" from the freeswitch CLI

Also look at the ptime and packet size in the rtp streams.



On Thu, Feb 3, 2011 at 2:56 PM, Malay Thakershi <mthakershi at gmail.com> wrote:
> I am sorry if you thought I was impatient.
> It was 20 hours since my post so I thought I could revive it. Anyway, it is
> up to you to reply or not. All I am looking for is to get past the issue and
> on the way contribute something that is "knowledge" for FS community.
> So, 1 = flush packets and 0 = do not flush?
> Malay
>
>
> On Thu, Feb 3, 2011 at 2:41 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> I'm tempted to not reply due to the impatience.
>> the 2nd argument is weather or not to flush the input buffers on the rtp
>> stack.
>> If this is done, any udp packets that have not been read and may be
>> queued in the udp buffers would be flushed.
>>
>>
>> On Thu, Feb 3, 2011 at 1:37 PM, Malay Thakershi <mthakershi at gmail.com>
>> wrote:
>> > Anyone?
>> >
>> > On Wed, Feb 2, 2011 at 6:45 PM, Malay Thakershi <mthakershi at gmail.com>
>> > wrote:
>> >>
>> >> Hello,
>> >> I am using .NET module. There is sleep method that has two arguments.
>> >> What is second argument for?
>> >> I use it:
>> >> mObjMainSession.sleep(200, 0);
>> >> What happens if I pass something other than zero as second argument?
>> >> I am having inconsistent delays between call paths and that is why I am
>> >> exploring this argument.
>> >> Thank you for any help.
>> >> 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


From anthony.minessale at gmail.com  Fri Feb  4 01:39:36 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 3 Feb 2011 16:39:36 -0600
Subject: [Freeswitch-users] iLBC to PCMU transcoding broken on windows?
In-Reply-To: <1296770804857-5990709.post@n2.nabble.com>
References: <AANLkTimWv26=0XppqzZMV8yf_0QKLVCHMenTf4c27Q8X@mail.gmail.com>
	<1296770804857-5990709.post@n2.nabble.com>
Message-ID: <AANLkTin5JuPefXt49XiCdT88LhS-W94AEtLdEyycS=Ae@mail.gmail.com>

can you try latest GIT?
I HATE iLBC they do the codec negotiation special to the point that no
matter how well we abstract codec negotiation we still need if (codec
== iLBC) in the code grrr.


On Thu, Feb 3, 2011 at 4:06 PM, Jeff Lenk <jeff at jefflenk.com> wrote:
>
> I asked on that thread for someone to open a Jira at that time but nobody
> did. Please open a Jira about this problem so it doesnt get lost. I quickly
> checked the problem and I see the same thing as you. Not sure what is wrong
> yet.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/iLBC-to-PCMU-transcoding-broken-on-windows-tp5990434p5990709.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 marcin321 at gmail.com  Fri Feb  4 03:19:54 2011
From: marcin321 at gmail.com (Marcin Wojtowicz)
Date: Thu, 3 Feb 2011 19:19:54 -0500
Subject: [Freeswitch-users] iLBC to PCMU transcoding broken on windows?
In-Reply-To: <AANLkTin5JuPefXt49XiCdT88LhS-W94AEtLdEyycS=Ae@mail.gmail.com>
References: <AANLkTimWv26=0XppqzZMV8yf_0QKLVCHMenTf4c27Q8X@mail.gmail.com>
	<1296770804857-5990709.post@n2.nabble.com>
	<AANLkTin5JuPefXt49XiCdT88LhS-W94AEtLdEyycS=Ae@mail.gmail.com>
Message-ID: <AANLkTim3Qaxur6kxJTjnqk1FB=+4o+8iKVK_Wd3iYDQ6@mail.gmail.com>

Nope, no dice - it still squeals.

On Thu, Feb 3, 2011 at 5:39 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> can you try latest GIT?
> I HATE iLBC they do the codec negotiation special to the point that no
> matter how well we abstract codec negotiation we still need if (codec
> == iLBC) in the code grrr.
>
>
> On Thu, Feb 3, 2011 at 4:06 PM, Jeff Lenk <jeff at jefflenk.com> wrote:
> >
> > I asked on that thread for someone to open a Jira at that time but nobody
> > did. Please open a Jira about this problem so it doesnt get lost. I
> quickly
> > checked the problem and I see the same thing as you. Not sure what is
> wrong
> > yet.
> > --
> > View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/iLBC-to-PCMU-transcoding-broken-on-windows-tp5990434p5990709.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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  4 03:25:35 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Thu, 03 Feb 2011 19:25:35 -0500
Subject: [Freeswitch-users] Enterprise bridge not continuing?
In-Reply-To: <AANLkTinpLVkH5Xhs6QWyzS8gMvreROdO_2Rhj2S_zTjR@mail.gmail.com>
References: <4D4B1495.30608@communicatefreely.net>
	<AANLkTinpLVkH5Xhs6QWyzS8gMvreROdO_2Rhj2S_zTjR@mail.gmail.com>
Message-ID: <4D4B477F.3020901@communicatefreely.net>

That did it!

I feel a little dumb.  I tried it in the bridge arguments, but I guess 
it needs to be in the A-leg.

It seems to work with or without the ignore_early_media option.  I don't 
think our endpoints send back early media, but I can move that to the <> 
bridge arguments, just in case.

Thanks!

-Tim

Michael Collins wrote:
> Tim,
>
> I don't see that you have set continue_on_fail in your dp. 
> (http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail)
>
> Also, I'm curious to know if you deliberately are not ignoring early 
> media on the first leg, the one with user/5420 @communicatefreely.net 
> <http://communicatefreely.net>. If you don't ignore early media on 
> that leg then that leg will "win" and get the call as soon as media 
> comes back. Just curious if you had a specific reason for doing it 
> this way.
>
> -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 anthony.minessale at gmail.com  Fri Feb  4 03:26:33 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 3 Feb 2011 18:26:33 -0600
Subject: [Freeswitch-users] iLBC to PCMU transcoding broken on windows?
In-Reply-To: <AANLkTim3Qaxur6kxJTjnqk1FB=+4o+8iKVK_Wd3iYDQ6@mail.gmail.com>
References: <AANLkTimWv26=0XppqzZMV8yf_0QKLVCHMenTf4c27Q8X@mail.gmail.com>
	<1296770804857-5990709.post@n2.nabble.com>
	<AANLkTin5JuPefXt49XiCdT88LhS-W94AEtLdEyycS=Ae@mail.gmail.com>
	<AANLkTim3Qaxur6kxJTjnqk1FB=+4o+8iKVK_Wd3iYDQ6@mail.gmail.com>
Message-ID: <AANLkTim3WNf1ZAB0HUekM8k7YXpyVsSYxAvCzWEN80k=@mail.gmail.com>

I don't see the same thing.

can you enable the sip trace

sofia global siptrace on
console loglevel 7

and post the exchange to http://pastebin.freeswitch.org


On Thu, Feb 3, 2011 at 6:19 PM, Marcin Wojtowicz <marcin321 at gmail.com> wrote:
> Nope, no dice - it still squeals.
>
> On Thu, Feb 3, 2011 at 5:39 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> can you try latest GIT?
>> I HATE iLBC they do the codec negotiation special to the point that no
>> matter how well we abstract codec negotiation we still need if (codec
>> == iLBC) in the code grrr.
>>
>>
>> On Thu, Feb 3, 2011 at 4:06 PM, Jeff Lenk <jeff at jefflenk.com> wrote:
>> >
>> > I asked on that thread for someone to open a Jira at that time but
>> > nobody
>> > did. Please open a Jira about this problem so it doesnt get lost. I
>> > quickly
>> > checked the problem and I see the same thing as you. Not sure what is
>> > wrong
>> > yet.
>> > --
>> > View this message in context:
>> > http://freeswitch-users.2379917.n2.nabble.com/iLBC-to-PCMU-transcoding-broken-on-windows-tp5990434p5990709.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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 marcin321 at gmail.com  Fri Feb  4 03:41:58 2011
From: marcin321 at gmail.com (Marcin Wojtowicz)
Date: Thu, 3 Feb 2011 19:41:58 -0500
Subject: [Freeswitch-users] iLBC to PCMU transcoding broken on windows?
In-Reply-To: <AANLkTim3WNf1ZAB0HUekM8k7YXpyVsSYxAvCzWEN80k=@mail.gmail.com>
References: <AANLkTimWv26=0XppqzZMV8yf_0QKLVCHMenTf4c27Q8X@mail.gmail.com>
	<1296770804857-5990709.post@n2.nabble.com>
	<AANLkTin5JuPefXt49XiCdT88LhS-W94AEtLdEyycS=Ae@mail.gmail.com>
	<AANLkTim3Qaxur6kxJTjnqk1FB=+4o+8iKVK_Wd3iYDQ6@mail.gmail.com>
	<AANLkTim3WNf1ZAB0HUekM8k7YXpyVsSYxAvCzWEN80k=@mail.gmail.com>
Message-ID: <AANLkTikV0hhuS-Qu=cCPN5xOJjgxO9mBb1rJkL2EqvTr@mail.gmail.com>

It's up.

On Thu, Feb 3, 2011 at 7:26 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I don't see the same thing.
>
> can you enable the sip trace
>
> sofia global siptrace on
> console loglevel 7
>
> and post the exchange to http://pastebin.freeswitch.org
>
>
> On Thu, Feb 3, 2011 at 6:19 PM, Marcin Wojtowicz <marcin321 at gmail.com>
> wrote:
> > Nope, no dice - it still squeals.
> >
> > On Thu, Feb 3, 2011 at 5:39 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> can you try latest GIT?
> >> I HATE iLBC they do the codec negotiation special to the point that no
> >> matter how well we abstract codec negotiation we still need if (codec
> >> == iLBC) in the code grrr.
> >>
> >>
> >> On Thu, Feb 3, 2011 at 4:06 PM, Jeff Lenk <jeff at jefflenk.com> wrote:
> >> >
> >> > I asked on that thread for someone to open a Jira at that time but
> >> > nobody
> >> > did. Please open a Jira about this problem so it doesnt get lost. I
> >> > quickly
> >> > checked the problem and I see the same thing as you. Not sure what is
> >> > wrong
> >> > yet.
> >> > --
> >> > View this message in context:
> >> >
> http://freeswitch-users.2379917.n2.nabble.com/iLBC-to-PCMU-transcoding-broken-on-windows-tp5990434p5990709.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 <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Feb  4 04:38:18 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 3 Feb 2011 20:38:18 -0500
Subject: [Freeswitch-users] voice quality in conference
Message-ID: <64DD8193985D4FCC8F2F8F92BAEF6627@e1705>

is the audio of outbound call can be normalized
when it enters in conference ?
it seems that some phones with mic wit low sensitivy level
have the start of sentence cut or hashed,
some "ssss" and low level voice are not heard in conference.
if I make a normal bridge with the same phones the problem not
appears..

Thanks
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From dujinfang at gmail.com  Fri Feb  4 04:47:34 2011
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 4 Feb 2011 09:47:34 +0800
Subject: [Freeswitch-users] iLBC to PCMU transcoding broken on windows?
In-Reply-To: <AANLkTikV0hhuS-Qu=cCPN5xOJjgxO9mBb1rJkL2EqvTr@mail.gmail.com>
References: <AANLkTimWv26=0XppqzZMV8yf_0QKLVCHMenTf4c27Q8X@mail.gmail.com>
	<1296770804857-5990709.post@n2.nabble.com>
	<AANLkTin5JuPefXt49XiCdT88LhS-W94AEtLdEyycS=Ae@mail.gmail.com>
	<AANLkTim3Qaxur6kxJTjnqk1FB=+4o+8iKVK_Wd3iYDQ6@mail.gmail.com>
	<AANLkTim3WNf1ZAB0HUekM8k7YXpyVsSYxAvCzWEN80k=@mail.gmail.com>
	<AANLkTikV0hhuS-Qu=cCPN5xOJjgxO9mBb1rJkL2EqvTr@mail.gmail.com>
Message-ID: <AANLkTinByDFpC5uaVAyRq5v2F_0K7+kJNPo11s8tpzTJ@mail.gmail.com>

I was the guy that was asked to make a jira but since I haven't find a
time to try again on a real windows(I have windows in VM on Mac the
sound input doesn't work) so I haven't report.

Do you mix iLBC with 30ms and PCMU 20ms ? Are you happen to use
portaudio? one thing I wanted to try is to change portaudio to use
30ms or iLBC to 20ms so they match, but as I said I never had a chance
to try.

It would be better to report a jira with as much info you found.

btw I give up iLBC at 30i and move to SILK, since most of our provider
only do 20ms and I think a transcode of 20ms and 30 ms results to 60ms
delay....

7.

On Fri, Feb 4, 2011 at 8:41 AM, Marcin Wojtowicz <marcin321 at gmail.com> wrote:
> It's up.
>
> On Thu, Feb 3, 2011 at 7:26 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> I don't see the same thing.
>>
>> can you enable the sip trace
>>
>> sofia global siptrace on
>> console loglevel 7
>>
>> and post the exchange to http://pastebin.freeswitch.org
>>
>>
>> On Thu, Feb 3, 2011 at 6:19 PM, Marcin Wojtowicz <marcin321 at gmail.com>
>> wrote:
>> > Nope, no dice - it still squeals.
>> >
>> > On Thu, Feb 3, 2011 at 5:39 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> can you try latest GIT?
>> >> I HATE iLBC they do the codec negotiation special to the point that no
>> >> matter how well we abstract codec negotiation we still need if (codec
>> >> == iLBC) in the code grrr.
>> >>
>> >>
>> >> On Thu, Feb 3, 2011 at 4:06 PM, Jeff Lenk <jeff at jefflenk.com> wrote:
>> >> >
>> >> > I asked on that thread for someone to open a Jira at that time but
>> >> > nobody
>> >> > did. Please open a Jira about this problem so it doesnt get lost. I
>> >> > quickly
>> >> > checked the problem and I see the same thing as you. Not sure what is
>> >> > wrong
>> >> > yet.
>> >> > --
>> >> > View this message in context:
>> >> >
>> >> > http://freeswitch-users.2379917.n2.nabble.com/iLBC-to-PCMU-transcoding-broken-on-windows-tp5990434p5990709.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
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at 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
>
>



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


From anthony.minessale at gmail.com  Fri Feb  4 04:48:35 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 3 Feb 2011 19:48:35 -0600
Subject: [Freeswitch-users] voice quality in conference
In-Reply-To: <64DD8193985D4FCC8F2F8F92BAEF6627@e1705>
References: <64DD8193985D4FCC8F2F8F92BAEF6627@e1705>
Message-ID: <AANLkTi=71JyHKDtScN2-UTGago7jfhd2=q_5jnM6Uvgj@mail.gmail.com>

turn off the energy level in conference.conf.xml or press 7 a few
times when you are that guy to turn it down to 0


On Thu, Feb 3, 2011 at 7:38 PM, Madovsky <infos at madovsky.org> wrote:
> is the audio of outbound call can be normalized
> when it enters in conference ?
> it seems that some phones with mic wit low sensitivy level
> have the start of sentence cut or hashed,
> some "ssss" and low level voice are not heard in conference.
> if I make a normal bridge with the same phones the problem not
> appears..
>
> 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 mike.tesliuk at ultra.net.br  Fri Feb  4 00:30:17 2011
From: mike.tesliuk at ultra.net.br (Mike Tesliuk)
Date: Thu, 3 Feb 2011 19:30:17 -0200
Subject: [Freeswitch-users] mod_perl mod_lua
In-Reply-To: <AANLkTimfGCaJxdgrcp1B-9E6EFUO4p75u7CRHPTdA62F@mail.gmail.com>
References: <AANLkTikxbN47Nc7cG8y0HMDVqC0HHJ56aLQZHJYS2WTR@mail.gmail.com>
	<AANLkTinDSQkOJrAikHrTEm6YTfyMcv2yLfxGPucjDwVq@mail.gmail.com>
	<AANLkTi=0=+05cxkP8ER9WKmzgAS4HzymLrTHcrMVho6P@mail.gmail.com>
	<AANLkTi=bNWH97qZdDmuEvzm6ETaZDRVv1R3Rt7hLTg+L@mail.gmail.com>
	<AANLkTinJBQ_EskC4Fv72FauNgbr4r4fF_gnOjiD6NDSe@mail.gmail.com>
	<AANLkTik0QYcDP-YZ6RSxdipXxYJQ7LH32G45P6zJm8+E@mail.gmail.com>
	<AANLkTimfGCaJxdgrcp1B-9E6EFUO4p75u7CRHPTdA62F@mail.gmail.com>
Message-ID: <AANLkTimgkTq4BcBnUD+sd8WZnBCDG=w4iYRDyHbDoZFQ@mail.gmail.com>

for the second question -


You need the perl-dev packages, you say that you use apt-get to
install some things, try a apt-cache search perl | grep -i dev and
install the package.

2011/2/3 deniro <ayhkor at gmail.com>:
> Paul and Steve thanks many for your great assistance
> have couple of puzzles;
>
> 1--
> What are these packages for?
> freeswitch-lua? and freeswitch-perl (if I am installing? mod_perl and
> mod_lua from source)
> I installed them anyway.
>
> 2--
> I downloaded freeswitch 1.0.6 source with wget
> ('apt-get source freeswitch'? couldn't find the source)
> uncommented lua/perl from modules.conf? and run configure, then
> make mod_lua-install
> make mod_perl-install
>
> mod_lua installed? without problem
> but
> make mod_perl-install? had? some errors as below
>
> Creating mod_perl.so...
> /usr/bin/ld: cannot find -lperl
> collect2: ld returned 1 exit status
> gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -DDEBIAN
> -fno-strict-aliasing -pipe -fstack-protector -I/usr/local/include
> -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -I/usr/lib/perl/5.10/CORE
> -DEMBED_PERL -I/usr/src/freeswitch-1.0.6/src/include
> -I/usr/src/freeswitch-1.0.6/src/include
> -I/usr/src/freeswitch-1.0.6/libs/libteletone/src -fPIC -Werror
> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
> -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE
> -shared -o .libs/mod_perl.so -shared -Wl,-x .libs/mod_perl.o -Wl,-E
> -fstack-protector freeswitch_perl.o mod_perl_wrap.o perlxsi.o
> /usr/src/freeswitch-1.0.6/.libs/libfreeswitch.so
> -L/usr/src/freeswitch-1.0.6/libs/apr-util/xml/expat/lib
> /usr/src/freeswitch-1.0.6/libs/apr-util/xml/expat/lib/.libs/libexpat.a
> /usr/src/freeswitch-1.0.6/libs/apr/.libs/libapr-1.a -lrt
> -L/usr/src/freeswitch-1.0.6/libs/srtp -lncurses -L/usr/local/lib
> -L/usr/lib/perl/5.10/CORE -lperl -lgdbm -lgdbm_compat -ldb -ldl -lm
> -lpthread -lc -lcrypt?? -Wl,--rpath -Wl,/opt/freeswitch/lib -Wl,--rpath
> -Wl,/opt/freeswitch/mod
> make[3]: *** [mod_perl.so] Error 1
> make[2]: *** [install] Error 1
> make[1]: *** [mod_perl-install] Error 1
> make: *** [mod_perl-install] Error 2
> what am I missing here? (I have perl installed and freeswitch-perl instaled)
>
> 3-- how do I check if any module is installed or not? same for the load,
> i.e, any command to check wheether or not it is loaded
> if not loaded,? how you load a module (after editing modules.conf.xml what
> we do? stop/start freeswitch?)
>
> thx
> deniro--
>
>
>
> On Wed, Feb 2, 2011 at 11:46 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> Check first whether there are freeswitch-lua or freeswitch-perl packages.
>> The official debian/ubuntu build system does build both mod_lua and
>> mod_perl, but it places them in separate packages from the rest of
>> FreeSWITCH.
>> If you don't you'll need to compile them yourself...
>> You will need the FreeSWITCH source to compile the modules, since they are
>> in that source. Some of FreeSWITCH (the core) must also be compiled since
>> the modules use functions that are in the core - however you can restrict
>> the modules that are compiled to just the ones you need using modules.conf.
>> When you checkout/extract the FreeSWITCH source, the first thing you must
>> do is run bootstrap.sh. It is this file that generates the modules.conf
>> file, if you haven't created it yourself.
>> Unless you plan to upgrade (and you should think about doing so - 1.0.6 is
>> old now and git head has hundreds ?of bugfixes and new features) you should
>> make sure the one you build the modules on is the exact same version the
>> Ubuntu version was created from, otherwise you'll find that the interface
>> between the core and the modules may have changed and you'll either get a
>> unloadable or unstable module. If you installed via APT then you should be
>> able to use 'apt-get source freeswitch' to get the source package that'll
>> give you the version they used.
>>
>> -Steve
>>
>> On 2 February 2011 16:26, deniro <ayhkor at gmail.com> wrote:
>>>
>>> thanks for your advice and ?I appreciate you taking time to reply.
>>> You also note that I first check freeswitch site and I?put my questions
>>> if I cant find a solutions myself.
>>> I already checked for options with modules.conf and modules.conf.xml even
>>> before posting.
>>>
>>> If you look at my first posting, I stated that I have a compiled
>>> freeswitch and I dont even see modules.conf file (I searched).
>>> so to reiterate, The freeswitch comes installed and compiled already with
>>> a product, so I dont even have the freeswitch source.
>>>
>>> All I am loooking for is, if there is any way, to install new modules
>>> without re-installig from source and recompile from the scratch with
>>> existing freeswitch install.
>>> I highly doubt that this is possible but I am checking out with the gurus
>>> here.
>>> I dont wanna break already running freeswitch with custom dialplans and
>>> other custom configurations.
>>>
>>> thx again
>>> deniro--
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Feb 2, 2011 at 10:32 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>>> wrote:
>>>>
>>>> On Wed, Feb 2, 2011 at 4:22 PM, deniro <ayhkor at gmail.com> wrote:
>>>> > I?think that is to enable loading the module(not the install)
>>>> > thx
>>>>
>>>> So, you write to the mailing list for advice, and you don't put
>>>> confidence in answers you got back? ;)
>>>>
>>>> >
>>>> >
>>>> > On Wed, Feb 2, 2011 at 3:14 AM, Giovanni Maruzzelli
>>>> > <gmaruzz at gmail.com>
>>>> > wrote:
>>>> >>
>>>> >> On Wed, Feb 2, 2011 at 5:25 AM, deniro <ayhkor at gmail.com> wrote:
>>>> >> > Hi
>>>> >> > I have compiled version of freeswitch 1.6.x on ubuntu 10.04
>>>> >> > I want to install extra freeswitch modules like mod_perl and
>>>> >> > mod_lua
>>>> >> > Is there any way to install these modules without re-compileling
>>>> >> > freeswitch.
>>>> >> > I looked at freeswitch site and it is talking about? editing
>>>> >> > modules.conf
>>>> >> > file and compiling from source
>>>> >> > and I dont even have modules.conf file as I searched.
>>>> >>
>>>> >> Go to your original sources directory (where you gave the command
>>>> >> "make install"), edit the file modules.conf.xml and give the command
>>>> >> "make install".
>>>> >>
>>>> >> -giovanni
>>>> >>
>>>> >>
>>>> >>
>>>> >> >
>>>> >> > thx
>>>> >> > 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
>>>> >> >
>>>> >> >
>>>> >>
>>>> >>
>>>> >>
>>>> >> --
>>>> >> 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
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at 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
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 anu at familytv.com  Fri Feb  4 03:45:09 2011
From: anu at familytv.com (Anirudha Shimpi)
Date: Thu, 3 Feb 2011 17:45:09 -0700
Subject: [Freeswitch-users] mod_fsk not detecting call waiting caller id
Message-ID: <00ee01cbc404$c62c5ae0$528510a0$@familytv.com>

I have an analog line connected through an SPA3102 forwarded to an extension
on FS. Line has call waiting ID on it (verified). I get the tone when call
waiting is sent. A sample dialplan is as follows
<action application="answer"/>
<action application="sleep" data="1000"/>
<action application="fsk_recv"/>
<action application="set" data="execute_on_fsk=log 'Got FSK
[${fsk_phone_num}] [${fsk_phone_name}]'"/>
<action application="transfer" data="1002 XML default"/>

When call waiting tone is detected I get the following
2011-02-03 17:25:06.089086 [DEBUG] mod_fsk.c:276 sofia/internal/xxxxx at xxxxx
processing execute_on_fsk [log 'Got FSK [] []']
EXECUTE sofia/internal/xxxx at xxxx log('Got FSK [] []')
2011-02-03 17:25:06.089086 [DEBUG] mod_dptools.c:1183 FSK [] []'
2011-02-03 17:25:06.269078 [DEBUG] switch_core_media_bug.c:467 Removing BUG
from sofia/internal/xxxxx at xxxx

If I do a uuid_dump, the fsk_ (except variable_execute_on_fsk) variables are
not present. Any idea what I could be doing wrong here? Never used mod_fsk
before, however, need to just log call waiting name and number. Any help
would be greatly appreciated.



From anu at familytv.com  Fri Feb  4 03:48:38 2011
From: anu at familytv.com (Anirudha Shimpi)
Date: Thu, 3 Feb 2011 17:48:38 -0700
Subject: [Freeswitch-users] FS 1.0.7 crash on originate
Message-ID: <00ef01cbc405$42baf3a0$c830dae0$@familytv.com>

Just downloaded (from git) and installed FS 1.0.7, however, every time I do
originate user/1001 &park, FS crashes. Running Centos 5.5 64 bit.

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From infos at madovsky.org  Fri Feb  4 05:05:17 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 3 Feb 2011 21:05:17 -0500
Subject: [Freeswitch-users] voice quality in conference
References: <64DD8193985D4FCC8F2F8F92BAEF6627@e1705>
	<AANLkTi=71JyHKDtScN2-UTGago7jfhd2=q_5jnM6Uvgj@mail.gmail.com>
Message-ID: <A44865E7EFB24845BC03B6A0F43D54AA@e1705>

I set the energy level to 0 in conference.conf.xml
and it works. Thank you !

----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, February 03, 2011 8:48 PM
Subject: Re: [Freeswitch-users] voice quality in conference


> turn off the energy level in conference.conf.xml or press 7 a few
> times when you are that guy to turn it down to 0
> 
> 
> On Thu, Feb 3, 2011 at 7:38 PM, Madovsky <infos at madovsky.org> wrote:
>> is the audio of outbound call can be normalized
>> when it enters in conference ?
>> it seems that some phones with mic wit low sensitivy level
>> have the start of sentence cut or hashed,
>> some "ssss" and low level voice are not heard in conference.
>> if I make a normal bridge with the same phones the problem not
>> appears..
>>
>> 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


From mitch.capper at gmail.com  Fri Feb  4 05:12:41 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Thu, 3 Feb 2011 18:12:41 -0800
Subject: [Freeswitch-users] Portaudio Improvements
In-Reply-To: <AANLkTim9AmtEGZoK0NyG1yKVsmJ_MSwXAppi-6AcCqnz@mail.gmail.com>
References: <AANLkTim9AmtEGZoK0NyG1yKVsmJ_MSwXAppi-6AcCqnz@mail.gmail.com>
Message-ID: <AANLkTim85+Z2Nyysyz1bza-q6f7gsd0RVoR69tcoBMaw@mail.gmail.com>

I wanted to make note that these changes were committed to trunk today, you
can now use it without patching.

~Mitch

On Fri, Jan 21, 2011 at 9:00 AM, Mitch Capper <mitch.capper at gmail.com>wrote:

> I have submitted a ticket with a patch for portaudio and my improvements.
> This includes the improvements I had discussed previously during my call for
> input on the mailing list and is available at:
> http://jira.freeswitch.org/browse/FS-3006
>
> It was only tested in windows, however most of the changes should not
> effect default behavior (but please test if you can).
>
> To test you will want to enable some things in the portaudio config
> including:
>                     <param name="live-stream-switch" value="true" />
>                     <param name="no-auto-resume-call" value="true" />
>                     <param name="no-ring-during-call" value="true" />
>
> you will then be able to use the new features fully.
>
> Many of these changes were made to add better support for softphone's using
> freeswitch, I will be releasing my embedded freeswitch phone (FSClient) for
> the conference call in a week and a half  it is .net/c# but WPF so sadly
> windows only currently.  jlink and drk are helping so we should end up with
> a nice installer also for it.     If you would like to test it please let me
> know.
>
> ~Mitch
>
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From infos at madovsky.org  Fri Feb  4 05:14:14 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 3 Feb 2011 21:14:14 -0500
Subject: [Freeswitch-users] play_and_get_digits and macros
Message-ID: <81EDE8F1DC844D25ABA22B8942787BCC@e1705>

on speech managment wiki page there's explanation
of how to write XML for a macro, but where to put this XML code ?

I'd like to use a macro in play_and_get_digits, possible ? if yes what's the syntax ?

Thanks in advance
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From anthony.minessale at gmail.com  Fri Feb  4 05:30:51 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 3 Feb 2011 20:30:51 -0600
Subject: [Freeswitch-users] mod_fsk not detecting call waiting caller id
In-Reply-To: <00ee01cbc404$c62c5ae0$528510a0$@familytv.com>
References: <00ee01cbc404$c62c5ae0$528510a0$@familytv.com>
Message-ID: <AANLkTi=FZmzLNhnefdAwkKnL7zURQAu7DFLCqxRe6fq6@mail.gmail.com>

This is probably not what you want to hear but I found a bug which is
that it should not have even called your app because there was no fsk
data collected.

This app is not really meant for the way you are using it but maybe it
will work if you use it with bind_digit_action

<action application="bind_digit_action" data="test,D,exec:fsk_recv">

not promising anything but maybe it will work.

On Thu, Feb 3, 2011 at 6:45 PM, Anirudha Shimpi <anu at familytv.com> wrote:
> I have an analog line connected through an SPA3102 forwarded to an extension
> on FS. Line has call waiting ID on it (verified). I get the tone when call
> waiting is sent. A sample dialplan is as follows
> <action application="answer"/>
> <action application="sleep" data="1000"/>
> <action application="fsk_recv"/>
> <action application="set" data="execute_on_fsk=log 'Got FSK
> [${fsk_phone_num}] [${fsk_phone_name}]'"/>
> <action application="transfer" data="1002 XML default"/>
>
> When call waiting tone is detected I get the following
> 2011-02-03 17:25:06.089086 [DEBUG] mod_fsk.c:276 sofia/internal/xxxxx at xxxxx
> processing execute_on_fsk [log 'Got FSK [] []']
> EXECUTE sofia/internal/xxxx at xxxx log('Got FSK [] []')
> 2011-02-03 17:25:06.089086 [DEBUG] mod_dptools.c:1183 FSK [] []'
> 2011-02-03 17:25:06.269078 [DEBUG] switch_core_media_bug.c:467 Removing BUG
> from sofia/internal/xxxxx at xxxx
>
> If I do a uuid_dump, the fsk_ (except variable_execute_on_fsk) variables are
> not present. Any idea what I could be doing wrong here? Never used mod_fsk
> before, however, need to just log call waiting name and number. Any help
> would be greatly appreciated.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Fri Feb  4 05:55:19 2011
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 4 Feb 2011 10:55:19 +0800
Subject: [Freeswitch-users] play_and_get_digits and macros
In-Reply-To: <81EDE8F1DC844D25ABA22B8942787BCC@e1705>
References: <81EDE8F1DC844D25ABA22B8942787BCC@e1705>
Message-ID: <AANLkTik_ocON_JRuTufK7=kmVmaU7Xr6RU71dcBK5dEo@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Speech_Phrase_Management

On Fri, Feb 4, 2011 at 10:14 AM, Madovsky <infos at madovsky.org> wrote:
> on speech managment wiki page there's explanation
> of how to write XML for a macro, but where to put this XML code ?
>
> I'd like to use a macro in play_and_get_digits, possible ? if yes what's the
> syntax ?
>
> 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
>
>



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


From david.ponzone at ipeva.fr  Fri Feb  4 06:02:13 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 4 Feb 2011 04:02:13 +0100
Subject: [Freeswitch-users] Strange behaviour with leg_timeout
In-Reply-To: <AANLkTikeBgwS_u+wt7c9qLJktZ1iqqSkUGtuQdNkVeW7@mail.gmail.com>
References: <240D7F48-8423-42A5-A357-632F0AB54412@ipeva.fr>
	<AANLkTikeBgwS_u+wt7c9qLJktZ1iqqSkUGtuQdNkVeW7@mail.gmail.com>
Message-ID: <6E555D47-0C71-4338-BE09-B11BC7BCC4D9@ipeva.fr>

Anthony,

I tried:
        <action application="bridge" data="[leg_timeout=5]sofia/external/$1 at a.b.c.d|[leg_timeout=5]sofia/gateway/provider.out/YYYYYYYYYY"/>
and
	<action application="bridge" data="[leg_progress_timeout=5]sofia/external/$1 at a.b.c.d|[leg_progress_timeout=5]sofia/gateway/provider.out/YYYYYYYYYY"/>

The result is the same: a.b.c.d does not reply anything, so after 5 seconds, provider.out is tried, but then, INVITEs are still sent to a.b.c.d

I actually think I understand why.
I think it's because those INVITE retries are happening within Sofia.
FreeSWITCH does not have a reply at all, so leg-timeout or leg_progress_timeout are still triggered, and the dialplan continues, but Sofia is still doing its retries on the first target.
I guess to avoid this, we would need a way to control the number of retries sent by Sofia.
It's probably not a big deal anyway for my situation, where i want to send calls to backup numbers when the trunk to a.b.c.d is really dead.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 03/02/2011 ? 23:02, Anthony Minessale a ?crit :

> We do not have anything to timeout at a 100 the only thing we can do
> is 180 or 183 using [leg_progress_timeout=X]  in front of each leg.
> 
> 
> 
> On Thu, Feb 3, 2011 at 3:54 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
>> It's my time to mess with XXX_timeout variables in FreeSWITCH.
>> What I need to do is to route a call to a gateway, and if I don't have an
>> answer (like not even the 100) within X seconds, I try to route the call to
>> a backup number.
>> I tried it this way:
>>     <extension name="gw_then_backup">
>>        <condition field="destination_number" expression="^(XXXXXXXXXX)$">
>>         <action application="bridge"
>> data="{leg_timeout=5}sofia/external/$1 at a.b.c.d|sofia/gateway/provider.out/YYYYYYYYYY"/>
>>        </condition>
>>      </extension>
>> My understanding was that a.b.c.d would be tried for 5 seconds, and if no
>> 100 is received, gateway provider.out would then be used.
>> It works...but after the first INVITE to provider.out (which sends back
>> 100/180/183), I can see some more INVITEs sent to a.b.c.d.
>> I am wondering if I am doing something wrong, or if there is an issue.
>> I use latest git from 5 minutes ago.
>> Thanks
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> 
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From u2nsam at gmail.com  Fri Feb  4 06:24:11 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 4 Feb 2011 08:54:11 +0530
Subject: [Freeswitch-users] context
In-Reply-To: <AANLkTikmnwQjXSX9GaC0Q3DGQYOTpv9-LE29AFW-VA4T@mail.gmail.com>
References: <AANLkTika5CBO5eVWuRimKY+Bth2YAFC0Tm8_JmFBnXwC@mail.gmail.com>
	<AANLkTim84VqQ79xq6amru_4GzvGP9rof001MJ8t2Zpvd@mail.gmail.com>
	<AANLkTinutzgJ5GF_tP=gpfVaz03+CQ-b-CCjPh3rFL=_@mail.gmail.com>
	<AANLkTikmnwQjXSX9GaC0Q3DGQYOTpv9-LE29AFW-VA4T@mail.gmail.com>
Message-ID: <AANLkTik7xyGdbMvfnZbvOWX-P8XUb+zvchQ4X5stAgFQ@mail.gmail.com>

It all happened with the polycom phone not getting registered on FS due to
nat/pat environment and rest of other make phone getting registered.
So added phone ip to acl and tried NDLB and all and landed up getting the
call flow reaching public context and it got out of my mind what changes i
made.
But thanks for you help MC.

Regards
Sam

On Thu, Feb 3, 2011 at 12:15 PM, Michael Collins <msc at freeswitch.org> wrote:

> I think I'm going to write an article on this topic for the new folks since
> it seems to be a common point of confusion. Don't worry, you're not alone -
> this is a common challenge for new FS admins.
>
> #1 - The first line of your pastebin is the key. You are letting the call
> in via an ACL. This is "okay" if it is what you want, but I doubt that it
> is. I recommend that you remove this phone's IP address from the "domains"
> section of acl.conf.xml. When using the default FS configs, if you let a
> phone call through using the "domains" ACL then it automatically goes to the
> public context. Why? Because the call is not explicitly associated with a
> local user on the system. It's like getting a free pass into Disneyland but
> you don't have a little badge that says, "Hi, my name is..."
>
> Some get confused by the "Falling back to digest auth" message. It looks
> like an error but it's really just information. If the caller's SIP client
> properly authenticates (which is different than registration - see below)
> then FS knows exactly who is making the call and that it is a local user so
> it goes into the "default" context.
>
> #2 - Make sure that you learn the difference between SIP *registration* and
> SIP *authentication*.  Registration is where the SIP client tells FS:
> "Here's how you can reach me if you get a call for me." It is for calls TO
> the phone, not calls from the phone. On the other hand, call authentication
> is for calls FROM the phone to FS. When the phone calls FS, FS first checks
> to see if the phone's IP is in the "domains" ACL. If it is, then it just
> let's the call in to be handled by the public context. If not, then FS sends
> an "auth challenge" - basically saying, "What's the password?" If the SIP
> client properly responds then the call is "authenticated" - meaning that FS
> knows it is from a specific user and thus it goes to the default context.
>
> I recommend that you get the FS book and look at chapter 4. We talk a lot
> about the user directory and it will help you understand how it all works.
> Hopefully this explanation will whet your appetite for more. :)
>
> Keep hacking away at it - you'll get it soon enough!
>
> -MC
>
> On Wed, Feb 2, 2011 at 7:02 PM, Sam <u2nsam at gmail.com> wrote:
>
>> So you say that the extensions are not registered with FS and they are
>> getting allowed by the acl, just like DIDs.
>> How can i rectify this,any thing that needs to be checked ?
>>
>> http://pastebin.freeswitch.org/15217
>>
>> Regards
>> Sam
>>
>>
>> On Wed, Feb 2, 2011 at 11:23 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Turn on console debug level output (default in fs_cli) and make the test
>>> call. Pastebin the output. Most likely the call is not being authorized
>>> because you are letting it in via an ACL or something like that.
>>>
>>> -MC
>>>
>>> On Wed, Feb 2, 2011 at 1:55 AM, Sam <u2nsam at gmail.com> wrote:
>>>
>>>> I have defined a user as :
>>>>
>>>> <user id="2099">
>>>>       <params>
>>>>         <param name="password" value="1234"/>
>>>>         <param name="dial-string" value="{presence_id=${dialed_user}@
>>>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>>>         <param name="mwi-account" value="2099 at voicemail"/>
>>>>         <param name="vm-password" value="1234"/>
>>>>         <param name="vm-email-all-messages" value="true"/>
>>>>         <!--param name="vm-attach-file" value="true"/-->
>>>>         <!--param name="vm-mailto"  value=""/-->
>>>>       </params>
>>>>       <variables>
>>>>         <variable name="user_context" value="inter"/>
>>>>         <variable name="force_transfer_context" value="inter"/>
>>>>         <variable name="accountcode" value="2099"/>
>>>>         <variable name="internal_caller_id_name" value="2099"/>
>>>>         <variable name="internal_caller_id_number" value="2099"/>
>>>>         <variable name="callgroup" value="1"/>
>>>>         <variable name="sip-force-contact"
>>>> value="NDLB-connectile-dysfunction"/>
>>>>       </variables>
>>>>
>>>>
>>>> and in the dialplan i have
>>>>
>>>> <context name="inter">
>>>>
>>>> <extension name="2099">
>>>>     <condition field="destination_number" expression="^(2099)">
>>>>       <action application="set" data="call_timeout=30"/>
>>>>       <action application="set" data="dialed_extension=$1"/>
>>>>             <action application="export" data="dialed_extension=$1"/>
>>>>             <action application="set"
>>>> data="transfer_ringback=$${hold_music}"/>
>>>>             <action application="set" data="hangup_after_bridge=true"/>
>>>>             <action application="set" data="continue_on_fail=true"/>
>>>>             <action application="hash"
>>>> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>>>>             <action application="hash"
>>>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>>>>             <action application="hash"
>>>> data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
>>>>             <action application="hash"
>>>> data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>>>>             <action application="set"
>>>> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
>>>> var callgroup)}"/>
>>>>             <action application="hash"
>>>> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>>>>             <action application="bridge"
>>>> data="{sip_invite_domain=$${domain}}user/${dialed_extension}@
>>>> ${domain_name}"/>
>>>>     </condition>
>>>>   </extension>
>>>>
>>>> </context>
>>>>
>>>>
>>>> Now extension 2075 is also made on above lines in the same context for
>>>> user and dialplan, here the call when initiated from 2075 to 2099 searches
>>>> for context public instead of context inter
>>>>
>>>> mod_dialplan_xml.c:331 Processing 2075 <2075>->2099 in context public
>>>>
>>>> any reason why ?
>>>>
>>>> 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 u2nsam at gmail.com  Fri Feb  4 06:31:22 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 4 Feb 2011 09:01:22 +0530
Subject: [Freeswitch-users] deflect
In-Reply-To: <AANLkTik-vBpwkikio_2HqkE1hiBxCuwKUarOOOT1C+Ak@mail.gmail.com>
References: <AANLkTiksYLOeki2_XDJC5YwMW6hjUxvdpT7-Mvg7vmKZ@mail.gmail.com>
	<AANLkTi=EMn55D1xGBrWuGgVE_dW_5itoJ4WpzByyagTE@mail.gmail.com>
	<AANLkTi=F5Lhsk1tjcG5z5LTUjZb4e-cQdNXQjK3OsVD7@mail.gmail.com>
	<AANLkTinV=xZ_ahs=8DH7n8hADGL4hTOR05hNCWCasA85@mail.gmail.com>
	<AANLkTimfCVsJFCVDv1Az-8ERh8UZRqx1cKG1hY+GfKwd@mail.gmail.com>
	<AANLkTi=OnWrFDS_JbijJnpQ+_EmPm6r9=yM_aYVNjLVA@mail.gmail.com>
	<AANLkTik-vBpwkikio_2HqkE1hiBxCuwKUarOOOT1C+Ak@mail.gmail.com>
Message-ID: <AANLkTikixO6_GwBh==P+P24Hv2Gmj4q7nKgLCPH35fxj@mail.gmail.com>

Hi all,

Is there any way around ?

Regards
Sam



On Mon, Jan 31, 2011 at 9:36 AM, Sam <u2nsam at gmail.com> wrote:

> Hi,
>
> Any method to just blind transfer with media to the proxy ahead so that
> call remains connected to 12127773456 ?
> or does it requires only b2bua in the next transferred hop ?
>
> Regards
> Sam
>
>
>
>
> On Thu, Jan 27, 2011 at 1:10 PM, Sam <u2nsam at gmail.com> wrote:
>
>> Hi Michael,
>>
>> Here is it. http://pastebin.freeswitch.org/15156
>>
>> Regds
>> Sam
>>
>>
>> On Wed, Jan 26, 2011 at 12:23 AM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> I strongly recommend that you capture the debug output and drop it into a
>>> pastebin at pastebin.freeswitch.org. You may also wish to capture the
>>> sip traffic as well. If you are using fs_cli then you already see the debug
>>> level console output. To get the sip traffic inline with the debug output
>>> just do "sofia global siptrace on".
>>>
>>> -MC
>>>
>>>
>>> On Tue, Jan 25, 2011 at 8:24 AM, Sam <u2nsam at gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> Is it possible in this scenario,
>>>>
>>>> I have a call (leg a) to an IVR on FS1 , after the ivr the below
>>>> statement is executed,
>>>> <action application="bridge" data="{api_on_answer='uuid_transfer
>>>> ${uuid} both'}sofia/external/12127773456 at 192.168.2.130<9999 at 192.168.2.130>
>>>> "/>
>>>>
>>>> As the FS1 sends invite to 192.168.2.130 and the call is connected to
>>>> the moviephone IVR,
>>>> but here what happens is the call is getting disconnected from leg a and
>>>> the movie phone ivr 12127773456.
>>>>
>>>>
>>>>
>>>> Regds
>>>>
>>>> Sam
>>>>
>>>>
>>>>
>>>> On Mon, Jan 24, 2011 at 3:12 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>>>
>>>>> You could try uuid_simplify with the api_on_answer hook
>>>>>
>>>>> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_simplify
>>>>> http://wiki.freeswitch.org/wiki/Variable_api_on_answer
>>>>>
>>>>> -Steve
>>>>>
>>>>>
>>>>>
>>>>> On 24 January 2011 09:05, Sam <u2nsam at gmail.com> wrote:
>>>>>
>>>>>>  Hi,
>>>>>>
>>>>>> Is it possible by having b2bua in between , would the leg A be
>>>>>> deflected to the another FS server from first server ?
>>>>>>
>>>>>> Regds
>>>>>> Sam
>>>>>>
>>>>>>
>>>>>> On Wed, Jan 12, 2011 at 11:42 AM, Sam <u2nsam at gmail.com> wrote:
>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>> When call comes on 1 server and plays an application and after
>>>>>>> execution of the
>>>>>>> application the call is bridge to the other server ,but here after
>>>>>>> bridging the call
>>>>>>> should refer/deflect to other server, how this can be done ?
>>>>>>>
>>>>>>> Here just using the deflect variable is not recommended as there is
>>>>>>> proxy in between,
>>>>>>> so once the call is bridge the next step would be deflect the leg
>>>>>>> totally to another server via proxy.
>>>>>>>
>>>>>>> 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 avi at avimarcus.net  Fri Feb  4 08:46:15 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 4 Feb 2011 07:46:15 +0200
Subject: [Freeswitch-users] Auto test FS install - now necessary?
Message-ID: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>

Maybe I'm just being paranoid, but I've been seeing a lot of "I just updated
to latest git and X broke"
And that x is always something else...
I'm not a seasoned coder on this magnitude, but perhaps we can develop a
test suite (at least part of one) to test that a fresh install is
functioning?
A FULL test would probably be a huge undertaking and involve recqual for
testing media, too... Also, each mod would need to have it's own test suite.
Due to the rapid updating of FreeSWITCH this not be what you want to put
your effort into, but...
Thoughts?
On Feb 4, 2011 4:04 AM, "Anirudha Shimpi" <anu at familytv.com> wrote:
> Just downloaded (from git) and installed FS 1.0.7, however, every time I
do
> originate user/1001 &park, FS crashes. Running Centos 5.5 64 bit.
>
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From jeff at jefflenk.com  Fri Feb  4 08:48:30 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 3 Feb 2011 21:48:30 -0800 (PST)
Subject: [Freeswitch-users] iLBC to PCMU transcoding broken on windows?
In-Reply-To: <AANLkTikV0hhuS-Qu=cCPN5xOJjgxO9mBb1rJkL2EqvTr@mail.gmail.com>
References: <AANLkTimWv26=0XppqzZMV8yf_0QKLVCHMenTf4c27Q8X@mail.gmail.com>
	<1296770804857-5990709.post@n2.nabble.com>
	<AANLkTin5JuPefXt49XiCdT88LhS-W94AEtLdEyycS=Ae@mail.gmail.com>
	<AANLkTim3Qaxur6kxJTjnqk1FB=+4o+8iKVK_Wd3iYDQ6@mail.gmail.com>
	<AANLkTim3WNf1ZAB0HUekM8k7YXpyVsSYxAvCzWEN80k=@mail.gmail.com>
	<AANLkTikV0hhuS-Qu=cCPN5xOJjgxO9mBb1rJkL2EqvTr@mail.gmail.com>
Message-ID: <1296798510781-5991622.post@n2.nabble.com>


Please try git head

Commit:2d190b37abe00999a2e76861b8c88f0053e0b78f

* fix iLBC under windows
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/iLBC-to-PCMU-transcoding-broken-on-windows-tp5990434p5991622.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveu at coppice.org  Fri Feb  4 10:49:16 2011
From: steveu at coppice.org (Steve Underwood)
Date: Fri, 04 Feb 2011 15:49:16 +0800
Subject: [Freeswitch-users] iLBC to PCMU transcoding broken on windows?
In-Reply-To: <1296798510781-5991622.post@n2.nabble.com>
References: <AANLkTimWv26=0XppqzZMV8yf_0QKLVCHMenTf4c27Q8X@mail.gmail.com>	<1296770804857-5990709.post@n2.nabble.com>	<AANLkTin5JuPefXt49XiCdT88LhS-W94AEtLdEyycS=Ae@mail.gmail.com>	<AANLkTim3Qaxur6kxJTjnqk1FB=+4o+8iKVK_Wd3iYDQ6@mail.gmail.com>	<AANLkTim3WNf1ZAB0HUekM8k7YXpyVsSYxAvCzWEN80k=@mail.gmail.com>	<AANLkTikV0hhuS-Qu=cCPN5xOJjgxO9mBb1rJkL2EqvTr@mail.gmail.com>
	<1296798510781-5991622.post@n2.nabble.com>
Message-ID: <4D4BAF7C.70200@coppice.org>

On 02/04/2011 01:48 PM, Jeff Lenk wrote:
> Please try git head
>
> Commit:2d190b37abe00999a2e76861b8c88f0053e0b78f
>
> * fix iLBC under windows
Interesting change. The original code is what most things use for 
rint(). It is what spandsp uses as well. If there is a problem with the 
code in ilbc, there are probably a number of things which need sorting out.

Steve



From david.ponzone at ipeva.fr  Fri Feb  4 11:05:28 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 4 Feb 2011 09:05:28 +0100
Subject: [Freeswitch-users] How to use sofia_gateway_data ?
Message-ID: <56C92185-F6A1-4C19-B3C3-F685C61CE038@ipeva.fr>

All,

I am still trying to reroute a call in case a gateway is down, but I decided to do that another way.
I'd like to detect my gateway is down, and I was trying to use sofia_gateway_data function for that purpose.
But I can't figure out the expected args.

I try:
sofia_gateway_data <gatewayname> var Status
or ivar/ovar instead of var

I tried with both a gateway I register to, and an IP-auth-based gateway I ping regularly.

So I am stuck now, and I can't find any info anywhere on how to detect a gateway is flagged as DOWN.

TIA

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




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From steveayre at gmail.com  Fri Feb  4 11:45:59 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 4 Feb 2011 08:45:59 +0000
Subject: [Freeswitch-users] How to use sofia_gateway_data ?
In-Reply-To: <56C92185-F6A1-4C19-B3C3-F685C61CE038@ipeva.fr>
References: <56C92185-F6A1-4C19-B3C3-F685C61CE038@ipeva.fr>
Message-ID: <AANLkTi=wqYgNbOQKvQP0yJWvpw1ZUAy=11cj=E9WSLSH@mail.gmail.com>

API commands:
sofia profile <name> gwlist up
sofia profile <name> gwlist down

List the up/down gateways (names separated by space). You can do a regex on
that result from dialplan. For example this will only execute if the gateway
is up:
<extension>
  <condition field="${sofia profile gwlist up}" expression="\bgw001\b">
    <action application="bridge"
data="sofia/gateway/gw001/${destination_number}"/>
  <condition>
<extension>

\b matches word boundary, so will match both the start of word (both at
start of string and after space) and the end of the word (at the end of a
string or before a space).

-Steve


On 4 February 2011 08:05, David Ponzone <david.ponzone at ipeva.fr> wrote:

> All,
>
> I am still trying to reroute a call in case a gateway is down, but I
> decided to do that another way.
> I'd like to detect my gateway is down, and I was trying to use
> sofia_gateway_data function for that purpose.
> But I can't figure out the expected args.
>
> I try:
> sofia_gateway_data <gatewayname> var Status
> or ivar/ovar instead of var
>
> I tried with both a gateway I register to, and an IP-auth-based gateway I
> ping regularly.
>
> So I am stuck now, and I can't find any info anywhere on how to detect a
> gateway is flagged as DOWN.
>
> TIA
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.ponzone at ipeva.fr  Fri Feb  4 12:02:51 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 4 Feb 2011 10:02:51 +0100
Subject: [Freeswitch-users] How to use sofia_gateway_data ?
In-Reply-To: <AANLkTi=wqYgNbOQKvQP0yJWvpw1ZUAy=11cj=E9WSLSH@mail.gmail.com>
References: <56C92185-F6A1-4C19-B3C3-F685C61CE038@ipeva.fr>
	<AANLkTi=wqYgNbOQKvQP0yJWvpw1ZUAy=11cj=E9WSLSH@mail.gmail.com>
Message-ID: <803C8101-BC70-44AF-9707-74EB4F80A8D0@ipeva.fr>

Thanks Steven, that's actually a different approach, and quite straight-forward. I will use that.
I was actually not aware I could include API commands in ${}.

Nevertheless, so I can update the wiki, anyone knows how to use sofia_gateway_data ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 04/02/2011 ? 09:45, Steven Ayre a ?crit :

> API commands:
> sofia profile <name> gwlist up
> sofia profile <name> gwlist down
> 
> List the up/down gateways (names separated by space). You can do a regex on that result from dialplan. For example this will only execute if the gateway is up:
> <extension>
>   <condition field="${sofia profile gwlist up}" expression="\bgw001\b">
>     <action application="bridge" data="sofia/gateway/gw001/${destination_number}"/>
>   <condition>
> <extension>
> 
> \b matches word boundary, so will match both the start of word (both at start of string and after space) and the end of the word (at the end of a string or before a space).
> 
> -Steve
> 
> 
> On 4 February 2011 08:05, David Ponzone <david.ponzone at ipeva.fr> wrote:
> All,
> 
> I am still trying to reroute a call in case a gateway is down, but I decided to do that another way.
> I'd like to detect my gateway is down, and I was trying to use sofia_gateway_data function for that purpose.
> But I can't figure out the expected args.
> 
> I try:
> sofia_gateway_data <gatewayname> var Status
> or ivar/ovar instead of var
> 
> I tried with both a gateway I register to, and an IP-auth-based gateway I ping regularly.
> 
> So I am stuck now, and I can't find any info anywhere on how to detect a gateway is flagged as DOWN.
> 
> TIA
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  4 12:18:29 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 4 Feb 2011 09:18:29 +0000
Subject: [Freeswitch-users] How to use sofia_gateway_data ?
In-Reply-To: <803C8101-BC70-44AF-9707-74EB4F80A8D0@ipeva.fr>
References: <56C92185-F6A1-4C19-B3C3-F685C61CE038@ipeva.fr>
	<AANLkTi=wqYgNbOQKvQP0yJWvpw1ZUAy=11cj=E9WSLSH@mail.gmail.com>
	<803C8101-BC70-44AF-9707-74EB4F80A8D0@ipeva.fr>
Message-ID: <AANLkTimtqOWQx4mVtdOLuvxGhVLrO5eKST76wiNCg1Cb@mail.gmail.com>

I guess regex on the result of sofia_gateway_data and then testing the
result is not empty would do it.

http://wiki.freeswitch.org/wiki/Mod_commands#regex

-Steve


On 4 February 2011 09:02, David Ponzone <david.ponzone at ipeva.fr> wrote:

> Thanks Steven, that's actually a different approach, and quite
> straight-forward. I will use that.
> I was actually not aware I could include API commands in ${}.
>
> Nevertheless, so I can update the wiki, anyone knows how to use
> sofia_gateway_data ?
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 04/02/2011 ? 09:45, Steven Ayre a ?crit :
>
> API commands:
> sofia profile <name> gwlist up
> sofia profile <name> gwlist down
>
> List the up/down gateways (names separated by space). You can do a regex on
> that result from dialplan. For example this will only execute if the gateway
> is up:
> <extension>
>   <condition field="${sofia profile gwlist up}" expression="\bgw001\b">
>     <action application="bridge"
> data="sofia/gateway/gw001/${destination_number}"/>
>   <condition>
> <extension>
>
> \b matches word boundary, so will match both the start of word (both at
> start of string and after space) and the end of the word (at the end of a
> string or before a space).
>
> -Steve
>
>
> On 4 February 2011 08:05, David Ponzone <david.ponzone at ipeva.fr> wrote:
>
>> All,
>>
>> I am still trying to reroute a call in case a gateway is down, but I
>> decided to do that another way.
>> I'd like to detect my gateway is down, and I was trying to use
>> sofia_gateway_data function for that purpose.
>> But I can't figure out the expected args.
>>
>> I try:
>> sofia_gateway_data <gatewayname> var Status
>> or ivar/ovar instead of var
>>
>> I tried with both a gateway I register to, and an IP-auth-based gateway I
>> ping regularly.
>>
>> So I am stuck now, and I can't find any info anywhere on how to detect a
>> gateway is flagged as DOWN.
>>
>> TIA
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.ponzone at ipeva.fr  Fri Feb  4 12:24:25 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 4 Feb 2011 10:24:25 +0100
Subject: [Freeswitch-users] How to use sofia_gateway_data ?
In-Reply-To: <AANLkTimtqOWQx4mVtdOLuvxGhVLrO5eKST76wiNCg1Cb@mail.gmail.com>
References: <56C92185-F6A1-4C19-B3C3-F685C61CE038@ipeva.fr>
	<AANLkTi=wqYgNbOQKvQP0yJWvpw1ZUAy=11cj=E9WSLSH@mail.gmail.com>
	<803C8101-BC70-44AF-9707-74EB4F80A8D0@ipeva.fr>
	<AANLkTimtqOWQx4mVtdOLuvxGhVLrO5eKST76wiNCg1Cb@mail.gmail.com>
Message-ID: <D62B9128-8E58-4D8C-9753-846C261D32D9@ipeva.fr>

Steven,

sure but I can't even manage to get sofia_gateway_data to output anything else than a syntax error.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 04/02/2011 ? 10:18, Steven Ayre a ?crit :

> I guess regex on the result of sofia_gateway_data and then testing the result is not empty would do it.
> 
> http://wiki.freeswitch.org/wiki/Mod_commands#regex
> 
> -Steve
> 
> 
> On 4 February 2011 09:02, David Ponzone <david.ponzone at ipeva.fr> wrote:
> Thanks Steven, that's actually a different approach, and quite straight-forward. I will use that.
> I was actually not aware I could include API commands in ${}.
> 
> Nevertheless, so I can update the wiki, anyone knows how to use sofia_gateway_data ?
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 04/02/2011 ? 09:45, Steven Ayre a ?crit :
> 
>> API commands:
>> sofia profile <name> gwlist up
>> sofia profile <name> gwlist down
>> 
>> List the up/down gateways (names separated by space). You can do a regex on that result from dialplan. For example this will only execute if the gateway is up:
>> <extension>
>>   <condition field="${sofia profile gwlist up}" expression="\bgw001\b">
>>     <action application="bridge" data="sofia/gateway/gw001/${destination_number}"/>
>>   <condition>
>> <extension>
>> 
>> \b matches word boundary, so will match both the start of word (both at start of string and after space) and the end of the word (at the end of a string or before a space).
>> 
>> -Steve
>> 
>> 
>> On 4 February 2011 08:05, David Ponzone <david.ponzone at ipeva.fr> wrote:
>> All,
>> 
>> I am still trying to reroute a call in case a gateway is down, but I decided to do that another way.
>> I'd like to detect my gateway is down, and I was trying to use sofia_gateway_data function for that purpose.
>> But I can't figure out the expected args.
>> 
>> I try:
>> sofia_gateway_data <gatewayname> var Status
>> or ivar/ovar instead of var
>> 
>> I tried with both a gateway I register to, and an IP-auth-based gateway I ping regularly.
>> 
>> So I am stuck now, and I can't find any info anywhere on how to detect a gateway is flagged as DOWN.
>> 
>> TIA
>> 
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> 
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> 
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> 
>> 
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From rajesh.npnr at yahoo.com  Fri Feb  4 14:21:38 2011
From: rajesh.npnr at yahoo.com (rex.alex)
Date: Fri, 4 Feb 2011 03:21:38 -0800 (PST)
Subject: [Freeswitch-users] Grandstream Freeze on Originate command
In-Reply-To: <AANLkTikPiZgq9WsFeMs2U0iZrT7FcL10dptLCFu5HynP@mail.gmail.com>
References: <1296751186605-5989433.post@n2.nabble.com>
	<AANLkTikPiZgq9WsFeMs2U0iZrT7FcL10dptLCFu5HynP@mail.gmail.com>
Message-ID: <1296818498902-5992216.post@n2.nabble.com>


Hello,

Yes GS is registered with FreeSWITCH only. I will try what you said and post
the status.

Thanks,
Rex
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Grandstream-Freeze-on-Originate-command-tp5989433p5992216.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From rajesh.npnr at yahoo.com  Fri Feb  4 14:21:39 2011
From: rajesh.npnr at yahoo.com (rex.alex)
Date: Fri, 4 Feb 2011 03:21:39 -0800 (PST)
Subject: [Freeswitch-users] Grandstream Freeze on Originate command
In-Reply-To: <AANLkTikPiZgq9WsFeMs2U0iZrT7FcL10dptLCFu5HynP@mail.gmail.com>
References: <1296751186605-5989433.post@n2.nabble.com>
	<AANLkTikPiZgq9WsFeMs2U0iZrT7FcL10dptLCFu5HynP@mail.gmail.com>
Message-ID: <1296818499616-5992217.post@n2.nabble.com>


Hello,

Yes GS is registered with FreeSWITCH only. I will try what you said and will
post the status.

Thanks,
Rex
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Grandstream-Freeze-on-Originate-command-tp5989433p5992217.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Fri Feb  4 14:42:35 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 4 Feb 2011 11:42:35 +0000
Subject: [Freeswitch-users] How to use sofia_gateway_data ?
In-Reply-To: <D62B9128-8E58-4D8C-9753-846C261D32D9@ipeva.fr>
References: <56C92185-F6A1-4C19-B3C3-F685C61CE038@ipeva.fr>
	<AANLkTi=wqYgNbOQKvQP0yJWvpw1ZUAy=11cj=E9WSLSH@mail.gmail.com>
	<803C8101-BC70-44AF-9707-74EB4F80A8D0@ipeva.fr>
	<AANLkTimtqOWQx4mVtdOLuvxGhVLrO5eKST76wiNCg1Cb@mail.gmail.com>
	<D62B9128-8E58-4D8C-9753-846C261D32D9@ipeva.fr>
Message-ID: <AANLkTimJL9uqu=azNVuSts-r9OE_5ZL4GPuai+gA-2gY@mail.gmail.com>

Having checked the source code, it doesn't access any param values you have
set, or any runtime status.

It's only for any variables that have been set within the <gateway> tag.

<gateway>
   <variables>
      <variable name="a" value="1" direction="inbound"/>
      <variable name="b" value="2" direction="outound"/>
   </variables>
</gateway>

I'm not sure those are actually documented on the Wiki at the moment...

-Steve


On 4 February 2011 09:24, David Ponzone <david.ponzone at ipeva.fr> wrote:

> Steven,
>
> sure but I can't even manage to get sofia_gateway_data to output anything
> else than a syntax error.
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 04/02/2011 ? 10:18, Steven Ayre a ?crit :
>
> I guess regex on the result of sofia_gateway_data and then testing the
> result is not empty would do it.
>
> http://wiki.freeswitch.org/wiki/Mod_commands#regex
>
> -Steve
>
>
> On 4 February 2011 09:02, David Ponzone <david.ponzone at ipeva.fr> wrote:
>
>> Thanks Steven, that's actually a different approach, and quite
>> straight-forward. I will use that.
>> I was actually not aware I could include API commands in ${}.
>>
>> Nevertheless, so I can update the wiki, anyone knows how to use
>> sofia_gateway_data ?
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 04/02/2011 ? 09:45, Steven Ayre a ?crit :
>>
>> API commands:
>> sofia profile <name> gwlist up
>> sofia profile <name> gwlist down
>>
>> List the up/down gateways (names separated by space). You can do a regex
>> on that result from dialplan. For example this will only execute if the
>> gateway is up:
>> <extension>
>>   <condition field="${sofia profile gwlist up}" expression="\bgw001\b">
>>     <action application="bridge"
>> data="sofia/gateway/gw001/${destination_number}"/>
>>   <condition>
>> <extension>
>>
>> \b matches word boundary, so will match both the start of word (both at
>> start of string and after space) and the end of the word (at the end of a
>> string or before a space).
>>
>> -Steve
>>
>>
>> On 4 February 2011 08:05, David Ponzone <david.ponzone at ipeva.fr> wrote:
>>
>>> All,
>>>
>>> I am still trying to reroute a call in case a gateway is down, but I
>>> decided to do that another way.
>>> I'd like to detect my gateway is down, and I was trying to use
>>> sofia_gateway_data function for that purpose.
>>> But I can't figure out the expected args.
>>>
>>> I try:
>>> sofia_gateway_data <gatewayname> var Status
>>> or ivar/ovar instead of var
>>>
>>> I tried with both a gateway I register to, and an IP-auth-based gateway I
>>> ping regularly.
>>>
>>> So I am stuck now, and I can't find any info anywhere on how to detect a
>>> gateway is flagged as DOWN.
>>>
>>> TIA
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  4 14:50:54 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 4 Feb 2011 11:50:54 +0000
Subject: [Freeswitch-users] How to use sofia_gateway_data ?
In-Reply-To: <AANLkTimJL9uqu=azNVuSts-r9OE_5ZL4GPuai+gA-2gY@mail.gmail.com>
References: <56C92185-F6A1-4C19-B3C3-F685C61CE038@ipeva.fr>
	<AANLkTi=wqYgNbOQKvQP0yJWvpw1ZUAy=11cj=E9WSLSH@mail.gmail.com>
	<803C8101-BC70-44AF-9707-74EB4F80A8D0@ipeva.fr>
	<AANLkTimtqOWQx4mVtdOLuvxGhVLrO5eKST76wiNCg1Cb@mail.gmail.com>
	<D62B9128-8E58-4D8C-9753-846C261D32D9@ipeva.fr>
	<AANLkTimJL9uqu=azNVuSts-r9OE_5ZL4GPuai+gA-2gY@mail.gmail.com>
Message-ID: <AANLkTinhKxLgCoqtuwPbrF34dhsxRZ=a=HuBhSeu_B5K@mail.gmail.com>

I've updated the Wiki to show those can be configured, and update
sofia_gateway_data to explain its usage better.

-Steve

On 4 February 2011 11:42, Steven Ayre <steveayre at gmail.com> wrote:

> Having checked the source code, it doesn't access any param values you have
> set, or any runtime status.
>
> It's only for any variables that have been set within the <gateway> tag.
>
> <gateway>
>    <variables>
>       <variable name="a" value="1" direction="inbound"/>
>       <variable name="b" value="2" direction="outound"/>
>    </variables>
> </gateway>
>
> I'm not sure those are actually documented on the Wiki at the moment...
>
> -Steve
>
>
>
> On 4 February 2011 09:24, David Ponzone <david.ponzone at ipeva.fr> wrote:
>
>> Steven,
>>
>> sure but I can't even manage to get sofia_gateway_data to output anything
>> else than a syntax error.
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 04/02/2011 ? 10:18, Steven Ayre a ?crit :
>>
>> I guess regex on the result of sofia_gateway_data and then testing the
>> result is not empty would do it.
>>
>> http://wiki.freeswitch.org/wiki/Mod_commands#regex
>>
>> -Steve
>>
>>
>> On 4 February 2011 09:02, David Ponzone <david.ponzone at ipeva.fr> wrote:
>>
>>> Thanks Steven, that's actually a different approach, and quite
>>> straight-forward. I will use that.
>>> I was actually not aware I could include API commands in ${}.
>>>
>>> Nevertheless, so I can update the wiki, anyone knows how to use
>>> sofia_gateway_data ?
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 04/02/2011 ? 09:45, Steven Ayre a ?crit :
>>>
>>> API commands:
>>> sofia profile <name> gwlist up
>>> sofia profile <name> gwlist down
>>>
>>> List the up/down gateways (names separated by space). You can do a regex
>>> on that result from dialplan. For example this will only execute if the
>>> gateway is up:
>>> <extension>
>>>   <condition field="${sofia profile gwlist up}" expression="\bgw001\b">
>>>     <action application="bridge"
>>> data="sofia/gateway/gw001/${destination_number}"/>
>>>   <condition>
>>> <extension>
>>>
>>> \b matches word boundary, so will match both the start of word (both at
>>> start of string and after space) and the end of the word (at the end of a
>>> string or before a space).
>>>
>>> -Steve
>>>
>>>
>>> On 4 February 2011 08:05, David Ponzone <david.ponzone at ipeva.fr> wrote:
>>>
>>>> All,
>>>>
>>>> I am still trying to reroute a call in case a gateway is down, but I
>>>> decided to do that another way.
>>>> I'd like to detect my gateway is down, and I was trying to use
>>>> sofia_gateway_data function for that purpose.
>>>> But I can't figure out the expected args.
>>>>
>>>> I try:
>>>> sofia_gateway_data <gatewayname> var Status
>>>> or ivar/ovar instead of var
>>>>
>>>> I tried with both a gateway I register to, and an IP-auth-based gateway
>>>> I ping regularly.
>>>>
>>>> So I am stuck now, and I can't find any info anywhere on how to detect a
>>>> gateway is flagged as DOWN.
>>>>
>>>> TIA
>>>>
>>>>  David Ponzone  Direction Technique
>>>> email: david.ponzone at ipeva.fr
>>>> tel:      01 74 03 18 97
>>>> gsm:   06 66 98 76 34
>>>>
>>>> Service Client IPeva
>>>> tel:      0811 46 26 26
>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>
>>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>>> l'exp?diteur.*
>>>> *
>>>> *
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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.ponzone at ipeva.fr  Fri Feb  4 15:11:55 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 4 Feb 2011 13:11:55 +0100
Subject: [Freeswitch-users] How to use sofia_gateway_data ?
In-Reply-To: <AANLkTinhKxLgCoqtuwPbrF34dhsxRZ=a=HuBhSeu_B5K@mail.gmail.com>
References: <56C92185-F6A1-4C19-B3C3-F685C61CE038@ipeva.fr>
	<AANLkTi=wqYgNbOQKvQP0yJWvpw1ZUAy=11cj=E9WSLSH@mail.gmail.com>
	<803C8101-BC70-44AF-9707-74EB4F80A8D0@ipeva.fr>
	<AANLkTimtqOWQx4mVtdOLuvxGhVLrO5eKST76wiNCg1Cb@mail.gmail.com>
	<D62B9128-8E58-4D8C-9753-846C261D32D9@ipeva.fr>
	<AANLkTimJL9uqu=azNVuSts-r9OE_5ZL4GPuai+gA-2gY@mail.gmail.com>
	<AANLkTinhKxLgCoqtuwPbrF34dhsxRZ=a=HuBhSeu_B5K@mail.gmail.com>
Message-ID: <08709477-63F1-46DB-B666-2380D2757F52@ipeva.fr>

Steven,

It makes sense now, thanks for the update.
It's definitely not what I need!

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 04/02/2011 ? 12:50, Steven Ayre a ?crit :

> I've updated the Wiki to show those can be configured, and update sofia_gateway_data to explain its usage better.
> 
> -Steve
> 
> On 4 February 2011 11:42, Steven Ayre <steveayre at gmail.com> wrote:
> Having checked the source code, it doesn't access any param values you have set, or any runtime status.
> 
> It's only for any variables that have been set within the <gateway> tag.
> 
> <gateway>
>    <variables>
>       <variable name="a" value="1" direction="inbound"/>
>       <variable name="b" value="2" direction="outound"/>
>    </variables>
> </gateway>
> 
> I'm not sure those are actually documented on the Wiki at the moment...
> 
> -Steve
> 
> 
> 
> On 4 February 2011 09:24, David Ponzone <david.ponzone at ipeva.fr> wrote:
> Steven,
> 
> sure but I can't even manage to get sofia_gateway_data to output anything else than a syntax error.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 04/02/2011 ? 10:18, Steven Ayre a ?crit :
> 
>> I guess regex on the result of sofia_gateway_data and then testing the result is not empty would do it.
>> 
>> http://wiki.freeswitch.org/wiki/Mod_commands#regex
>> 
>> -Steve
>> 
>> 
>> On 4 February 2011 09:02, David Ponzone <david.ponzone at ipeva.fr> wrote:
>> Thanks Steven, that's actually a different approach, and quite straight-forward. I will use that.
>> I was actually not aware I could include API commands in ${}.
>> 
>> Nevertheless, so I can update the wiki, anyone knows how to use sofia_gateway_data ?
>> 
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> 
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> 
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> 
>> 
>> 
>> 
>> Le 04/02/2011 ? 09:45, Steven Ayre a ?crit :
>> 
>>> API commands:
>>> sofia profile <name> gwlist up
>>> sofia profile <name> gwlist down
>>> 
>>> List the up/down gateways (names separated by space). You can do a regex on that result from dialplan. For example this will only execute if the gateway is up:
>>> <extension>
>>>   <condition field="${sofia profile gwlist up}" expression="\bgw001\b">
>>>     <action application="bridge" data="sofia/gateway/gw001/${destination_number}"/>
>>>   <condition>
>>> <extension>
>>> 
>>> \b matches word boundary, so will match both the start of word (both at start of string and after space) and the end of the word (at the end of a string or before a space).
>>> 
>>> -Steve
>>> 
>>> 
>>> On 4 February 2011 08:05, David Ponzone <david.ponzone at ipeva.fr> wrote:
>>> All,
>>> 
>>> I am still trying to reroute a call in case a gateway is down, but I decided to do that another way.
>>> I'd like to detect my gateway is down, and I was trying to use sofia_gateway_data function for that purpose.
>>> But I can't figure out the expected args.
>>> 
>>> I try:
>>> sofia_gateway_data <gatewayname> var Status
>>> or ivar/ovar instead of var
>>> 
>>> I tried with both a gateway I register to, and an IP-auth-based gateway I ping regularly.
>>> 
>>> So I am stuck now, and I can't find any info anywhere on how to detect a gateway is flagged as DOWN.
>>> 
>>> TIA
>>> 
>>> David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>> 
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>> 
>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>> 
>>> 
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From rajesh.npnr at yahoo.com  Fri Feb  4 15:16:04 2011
From: rajesh.npnr at yahoo.com (rex.alex)
Date: Fri, 4 Feb 2011 04:16:04 -0800 (PST)
Subject: [Freeswitch-users] Grandstream Freeze on Originate command
In-Reply-To: <1296818499616-5992217.post@n2.nabble.com>
References: <1296751186605-5989433.post@n2.nabble.com>
	<AANLkTikPiZgq9WsFeMs2U0iZrT7FcL10dptLCFu5HynP@mail.gmail.com>
	<1296818499616-5992217.post@n2.nabble.com>
Message-ID: <1296821764473-5992350.post@n2.nabble.com>


Hello,

Even this command (originate user/2000 1000) freezes the GS. I have updated
the latest firmware and tried but the result is same.

But when I originate in the other way (originate user/1000 2000), it's
working absolutely fine. Please help...

Thanks,
Rex


-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Grandstream-Freeze-on-Originate-command-tp5989433p5992350.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kbdfck at gmail.com  Fri Feb  4 17:05:47 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Fri, 4 Feb 2011 17:05:47 +0300
Subject: [Freeswitch-users] Seems att_xfer doesn't work at all?
Message-ID: <AANLkTi=Nnwd2EE6tkt6dUOdn+sn_nn0y8oFcz=Q4q=i7@mail.gmail.com>

Hi all!

In latest git i'm unable to use att_xfer at all.

Channels don't get bridged on transferor hangup, no moh to transferee,
and transferee is locked in no moh state with CS_RESET after
transferor and transfer target is already hangup

Does anybody have working example of using att_xfer with bind_meta_app?


-- 
Best regards,

Dmitry Sytchev,
IT Engineer


From jeff at jefflenk.com  Fri Feb  4 17:28:32 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 4 Feb 2011 06:28:32 -0800 (PST)
Subject: [Freeswitch-users] iLBC to PCMU transcoding broken on windows?
In-Reply-To: <4D4BAF7C.70200@coppice.org>
References: <AANLkTimWv26=0XppqzZMV8yf_0QKLVCHMenTf4c27Q8X@mail.gmail.com>
	<1296770804857-5990709.post@n2.nabble.com>
	<AANLkTin5JuPefXt49XiCdT88LhS-W94AEtLdEyycS=Ae@mail.gmail.com>
	<AANLkTim3Qaxur6kxJTjnqk1FB=+4o+8iKVK_Wd3iYDQ6@mail.gmail.com>
	<AANLkTim3WNf1ZAB0HUekM8k7YXpyVsSYxAvCzWEN80k=@mail.gmail.com>
	<AANLkTikV0hhuS-Qu=cCPN5xOJjgxO9mBb1rJkL2EqvTr@mail.gmail.com>
	<1296798510781-5991622.post@n2.nabble.com>
	<4D4BAF7C.70200@coppice.org>
Message-ID: <1296829712313-5992740.post@n2.nabble.com>


Hi Steve,

Yes I see that spandsp has a similar definition but as far as I can see it
is not used(unless my search failed) which is why we have no problems.

fastconvert.c 282
    __inline double rint(double dbl)
    {
        _asm 
    	{
            fld dbl
            frndint
        }
    }

near as I can tell this code will not work because the result is never
popped from the FPU thats of course what the fstp does.

Please let me know your thoughts here.

Thanks
Jeff
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/iLBC-to-PCMU-transcoding-broken-on-windows-tp5990434p5992740.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Fri Feb  4 17:58:53 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 4 Feb 2011 08:58:53 -0600
Subject: [Freeswitch-users] Auto test FS install - now necessary?
In-Reply-To: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>
Message-ID: <AANLkTi=Rofy64NwXTt3u1hRRMuAEhimKKux-_zAhTkiS@mail.gmail.com>

I've been doing this a long time and it boils down to:

50% superstition/paranoia.
50% regressions from commits.

Stop and think. Without people updating everyday and telling us when
something is not right, we may never find out.
Engaging in the community and this mailing list and irc is a community
effort and its sometimes a BETA program too.
Just like if this was a cool video game, you get to play it before
everyone else.
It's hard to follow every possible execution path and get into every
nook and cranny of 3 million lines of code.
Even with the best unit testing int the world, we can't do it without
real people testing.
If you stop updating or persuade others to stop, now we are in real trouble.

Also, pay attention to the GIT commit messages:

This particular problem (which is fixed now btw) was committed with a
warning to please test.
There was a bigger problem earlier this week that resulted in some
minority invasive changes and its easy to catch the problems and in
the end the software is now more stable.

commit f60fdf653dd2d7f8d3eaa6a9086e1f68bd993c59
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Wed Feb 2 16:22:43 2011 -0600

    fix possible bad pointer in global vars (please test)



On Thu, Feb 3, 2011 at 11:46 PM, Avi Marcus <avi at avimarcus.net> wrote:
> Maybe I'm just being paranoid, but I've been seeing a lot of "I just updated
> to latest git and X broke"
> And that x is always something else...
> I'm not a seasoned coder on this magnitude, but perhaps we can develop a
> test suite (at least part of one) to test that a fresh install is
> functioning?
> A FULL test would probably be a huge undertaking and involve recqual for
> testing media, too... Also, each mod would need to have it's own test suite.
> Due to the rapid updating of FreeSWITCH this not be what you want to put
> your effort into, but...
> Thoughts?
>
> On Feb 4, 2011 4:04 AM, "Anirudha Shimpi" <anu at familytv.com> wrote:
>> Just downloaded (from git) and installed FS 1.0.7, however, every time I
>> do
>> originate user/1001 &park, FS crashes. Running Centos 5.5 64 bit.
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb  4 18:11:13 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 4 Feb 2011 09:11:13 -0600
Subject: [Freeswitch-users] FS 1.0.7 crash on originate
In-Reply-To: <00ef01cbc405$42baf3a0$c830dae0$@familytv.com>
References: <00ef01cbc405$42baf3a0$c830dae0$@familytv.com>
Message-ID: <AANLkTinKuwGgA5CqQazq+hLBGem9Vbeu1V=ukftVDZJL@mail.gmail.com>

sorry, minor regression.  Update to fix.


On Thu, Feb 3, 2011 at 6:48 PM, Anirudha Shimpi <anu at familytv.com> wrote:
> Just downloaded (from git) and installed FS 1.0.7, however, every time I do
> originate user/1001 &park, FS crashes. Running Centos 5.5 64 bit.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Fri Feb  4 18:13:24 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 4 Feb 2011 10:13:24 -0500
Subject: [Freeswitch-users] Auto test FS install - now necessary?
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>
	<AANLkTi=Rofy64NwXTt3u1hRRMuAEhimKKux-_zAhTkiS@mail.gmail.com>
Message-ID: <1105C189976B4970A1F282410B827A7D@e1705>

FS team needs user as users need FS team...
maybe think to a script (developped by all possible users)
that test every new functionality...

----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, February 04, 2011 9:58 AM
Subject: Re: [Freeswitch-users] Auto test FS install - now necessary?


> I've been doing this a long time and it boils down to:
>
> 50% superstition/paranoia.
> 50% regressions from commits.
>
> Stop and think. Without people updating everyday and telling us when
> something is not right, we may never find out.
> Engaging in the community and this mailing list and irc is a community
> effort and its sometimes a BETA program too.
> Just like if this was a cool video game, you get to play it before
> everyone else.
> It's hard to follow every possible execution path and get into every
> nook and cranny of 3 million lines of code.
> Even with the best unit testing int the world, we can't do it without
> real people testing.
> If you stop updating or persuade others to stop, now we are in real 
> trouble.
>
> Also, pay attention to the GIT commit messages:
>
> This particular problem (which is fixed now btw) was committed with a
> warning to please test.
> There was a bigger problem earlier this week that resulted in some
> minority invasive changes and its easy to catch the problems and in
> the end the software is now more stable.
>
> commit f60fdf653dd2d7f8d3eaa6a9086e1f68bd993c59
> Author: Anthony Minessale <anthm at freeswitch.org>
> Date:   Wed Feb 2 16:22:43 2011 -0600
>
>    fix possible bad pointer in global vars (please test)
>
>
>
> On Thu, Feb 3, 2011 at 11:46 PM, Avi Marcus <avi at avimarcus.net> wrote:
>> Maybe I'm just being paranoid, but I've been seeing a lot of "I just 
>> updated
>> to latest git and X broke"
>> And that x is always something else...
>> I'm not a seasoned coder on this magnitude, but perhaps we can develop a
>> test suite (at least part of one) to test that a fresh install is
>> functioning?
>> A FULL test would probably be a huge undertaking and involve recqual for
>> testing media, too... Also, each mod would need to have it's own test 
>> suite.
>> Due to the rapid updating of FreeSWITCH this not be what you want to put
>> your effort into, but...
>> Thoughts?
>>
>> On Feb 4, 2011 4:04 AM, "Anirudha Shimpi" <anu at familytv.com> wrote:
>>> Just downloaded (from git) and installed FS 1.0.7, however, every time I
>>> do
>>> originate user/1001 &park, FS crashes. Running Centos 5.5 64 bit.
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 marcin321 at gmail.com  Fri Feb  4 18:15:56 2011
From: marcin321 at gmail.com (Marcin Wojtowicz)
Date: Fri, 4 Feb 2011 10:15:56 -0500
Subject: [Freeswitch-users] iLBC to PCMU transcoding broken on windows?
In-Reply-To: <1296798510781-5991622.post@n2.nabble.com>
References: <AANLkTimWv26=0XppqzZMV8yf_0QKLVCHMenTf4c27Q8X@mail.gmail.com>
	<1296770804857-5990709.post@n2.nabble.com>
	<AANLkTin5JuPefXt49XiCdT88LhS-W94AEtLdEyycS=Ae@mail.gmail.com>
	<AANLkTim3Qaxur6kxJTjnqk1FB=+4o+8iKVK_Wd3iYDQ6@mail.gmail.com>
	<AANLkTim3WNf1ZAB0HUekM8k7YXpyVsSYxAvCzWEN80k=@mail.gmail.com>
	<AANLkTikV0hhuS-Qu=cCPN5xOJjgxO9mBb1rJkL2EqvTr@mail.gmail.com>
	<1296798510781-5991622.post@n2.nabble.com>
Message-ID: <AANLkTinUQ4vKu8di069=yQ96rM0kEJOo0DwPeczSEDfs@mail.gmail.com>

It works, however I might have a different problem now. When I'm calling out
from my device to a regular phone, I see that iLBC is the only codec offered
to voip.ms, but I have my external profile set up to offer only PCMU on that
leg. The end result is that I have only silence when I'm calling out, but it
works when I'm calling in.

On Fri, Feb 4, 2011 at 12:48 AM, Jeff Lenk <jeff at jefflenk.com> wrote:

>
> Please try git head
>
> Commit:2d190b37abe00999a2e76861b8c88f0053e0b78f
>
> * fix iLBC under windows
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/iLBC-to-PCMU-transcoding-broken-on-windows-tp5990434p5991622.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  Fri Feb  4 18:45:15 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 4 Feb 2011 15:45:15 +0000
Subject: [Freeswitch-users] Auto test FS install - now necessary?
In-Reply-To: <1105C189976B4970A1F282410B827A7D@e1705>
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>
	<AANLkTi=Rofy64NwXTt3u1hRRMuAEhimKKux-_zAhTkiS@mail.gmail.com>
	<1105C189976B4970A1F282410B827A7D@e1705>
Message-ID: <AANLkTik2YTEuXZPXqyMpuuibcxVQo9d1imLGL8+EzzAT@mail.gmail.com>

http://en.wikipedia.org/wiki/Software_testing#Input_combinations_and_preconditions
"A very fundamental problem with software testing is that testing under all
combinations of inputs and preconditions (initial state) is not feasible,
even with a simple product. This means that the number of defects in a
software product can be very large and defects that occur infrequently are
difficult to find in testing. More significantly, non-functional dimensions
of quality (how it is supposed to be versus what it is supposed to
do)?usability, scalability, performance, compatibility, reliability?can be
highly subjective; something that constitutes sufficient value to one person
may be intolerable to another.

Software testing is a very difficult thing. Testing every possible
combination of actions and specifically covering the edge cases most likely
to cause problems is pretty much impossible - to run through every possible
action for something as large as FS would literally take longer than the age
of the universe. As a result a script to test for all possible errors is
pretty much impossible, especially for new functionality that'll be added
before the script's updated to test them. There will always be some bugs,
users testing and reporting is the best way to detect ones that the
developers have missed. Generally FS is pretty good at avoiding them.

Testing yourself is always a good idea before rolling out to a production
system. That'll show any obvious ones up that can be fixed quickly.

-Steve


On 4 February 2011 15:13, Madovsky <infos at madovsky.org> wrote:

> FS team needs user as users need FS team...
> maybe think to a script (developped by all possible users)
> that test every new functionality...
>
> ----- Original Message -----
> From: "Anthony Minessale" <anthony.minessale at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Friday, February 04, 2011 9:58 AM
> Subject: Re: [Freeswitch-users] Auto test FS install - now necessary?
>
>
> > I've been doing this a long time and it boils down to:
> >
> > 50% superstition/paranoia.
> > 50% regressions from commits.
> >
> > Stop and think. Without people updating everyday and telling us when
> > something is not right, we may never find out.
> > Engaging in the community and this mailing list and irc is a community
> > effort and its sometimes a BETA program too.
> > Just like if this was a cool video game, you get to play it before
> > everyone else.
> > It's hard to follow every possible execution path and get into every
> > nook and cranny of 3 million lines of code.
> > Even with the best unit testing int the world, we can't do it without
> > real people testing.
> > If you stop updating or persuade others to stop, now we are in real
> > trouble.
> >
> > Also, pay attention to the GIT commit messages:
> >
> > This particular problem (which is fixed now btw) was committed with a
> > warning to please test.
> > There was a bigger problem earlier this week that resulted in some
> > minority invasive changes and its easy to catch the problems and in
> > the end the software is now more stable.
> >
> > commit f60fdf653dd2d7f8d3eaa6a9086e1f68bd993c59
> > Author: Anthony Minessale <anthm at freeswitch.org>
> > Date:   Wed Feb 2 16:22:43 2011 -0600
> >
> >    fix possible bad pointer in global vars (please test)
> >
> >
> >
> > On Thu, Feb 3, 2011 at 11:46 PM, Avi Marcus <avi at avimarcus.net> wrote:
> >> Maybe I'm just being paranoid, but I've been seeing a lot of "I just
> >> updated
> >> to latest git and X broke"
> >> And that x is always something else...
> >> I'm not a seasoned coder on this magnitude, but perhaps we can develop a
> >> test suite (at least part of one) to test that a fresh install is
> >> functioning?
> >> A FULL test would probably be a huge undertaking and involve recqual for
> >> testing media, too... Also, each mod would need to have it's own test
> >> suite.
> >> Due to the rapid updating of FreeSWITCH this not be what you want to put
> >> your effort into, but...
> >> Thoughts?
> >>
> >> On Feb 4, 2011 4:04 AM, "Anirudha Shimpi" <anu at familytv.com> wrote:
> >>> Just downloaded (from git) and installed FS 1.0.7, however, every time
> I
> >>> do
> >>> originate user/1001 &park, FS crashes. Running Centos 5.5 64 bit.
> >>>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at 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 <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Feb  4 18:55:10 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 4 Feb 2011 07:55:10 -0800 (PST)
Subject: [Freeswitch-users] iLBC to PCMU transcoding broken on windows?
In-Reply-To: <AANLkTinUQ4vKu8di069=yQ96rM0kEJOo0DwPeczSEDfs@mail.gmail.com>
References: <AANLkTimWv26=0XppqzZMV8yf_0QKLVCHMenTf4c27Q8X@mail.gmail.com>
	<1296770804857-5990709.post@n2.nabble.com>
	<AANLkTin5JuPefXt49XiCdT88LhS-W94AEtLdEyycS=Ae@mail.gmail.com>
	<AANLkTim3Qaxur6kxJTjnqk1FB=+4o+8iKVK_Wd3iYDQ6@mail.gmail.com>
	<AANLkTim3WNf1ZAB0HUekM8k7YXpyVsSYxAvCzWEN80k=@mail.gmail.com>
	<AANLkTikV0hhuS-Qu=cCPN5xOJjgxO9mBb1rJkL2EqvTr@mail.gmail.com>
	<1296798510781-5991622.post@n2.nabble.com>
	<AANLkTinUQ4vKu8di069=yQ96rM0kEJOo0DwPeczSEDfs@mail.gmail.com>
Message-ID: <1296834910731-5992975.post@n2.nabble.com>


At this point I know the codec is working in windows(encode and decode) so
you have a different problem now.
You can repost your siptraces, debug log and configuration to pastebin
(provide a link) and someone might see the problem.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/iLBC-to-PCMU-transcoding-broken-on-windows-tp5990434p5992975.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From fs-list at communicatefreely.net  Fri Feb  4 18:58:23 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 04 Feb 2011 10:58:23 -0500
Subject: [Freeswitch-users] Auto test FS install - now necessary?
In-Reply-To: <AANLkTik2YTEuXZPXqyMpuuibcxVQo9d1imLGL8+EzzAT@mail.gmail.com>
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>	<AANLkTi=Rofy64NwXTt3u1hRRMuAEhimKKux-_zAhTkiS@mail.gmail.com>	<1105C189976B4970A1F282410B827A7D@e1705>
	<AANLkTik2YTEuXZPXqyMpuuibcxVQo9d1imLGL8+EzzAT@mail.gmail.com>
Message-ID: <4D4C221F.40905@communicatefreely.net>


>
> Testing yourself is always a good idea before rolling out to a 
> production system. That'll show any obvious ones up that can be fixed 
> quickly.
Not that this will always catch everything, but is there a "release 
notes" or some other simple document that outlines what parts of 
Freeswitch have recently been changed?  We ended up building a sort of 
beta-test network, with a second server and some customers that are very 
benevolent about this sort of thing, and if I had a simple checklist of 
things that had been worked on, I can make sure that I test those areas 
more closely.

I don't have any experience working with software development beyond my 
own little LUA and PHP scripts, so maybe this is already available and I 
just don't know where to look.

Is there a tutorial somewhere on how to be a good user, from a 
developers perspective?

-Tim


From infos at madovsky.org  Fri Feb  4 19:03:56 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 4 Feb 2011 11:03:56 -0500
Subject: [Freeswitch-users] Auto test FS install - now necessary?
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com><AANLkTi=Rofy64NwXTt3u1hRRMuAEhimKKux-_zAhTkiS@mail.gmail.com><1105C189976B4970A1F282410B827A7D@e1705>
	<AANLkTik2YTEuXZPXqyMpuuibcxVQo9d1imLGL8+EzzAT@mail.gmail.com>
Message-ID: <173D9A49B9E545E4A40E9AB8E6C89826@e1705>

I understand well,
but I think everything is possible, even if it seems huge work
by the right method the huge work can be a reasonable work.
It needs only to reflect deeply of how to do this....
  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Friday, February 04, 2011 10:45 AM
  Subject: Re: [Freeswitch-users] Auto test FS install - now necessary?


  http://en.wikipedia.org/wiki/Software_testing#Input_combinations_and_preconditions
  "A very fundamental problem with software testing is that testing under all combinations of inputs and preconditions (initial state) is not feasible, even with a simple product. This means that the number of defects in a software product can be very large and defects that occur infrequently are difficult to find in testing. More significantly, non-functional dimensions of quality (how it is supposed to be versus what it is supposed to do)?usability, scalability, performance, compatibility, reliability?can be highly subjective; something that constitutes sufficient value to one person may be intolerable to another.

  Software testing is a very difficult thing. Testing every possible combination of actions and specifically covering the edge cases most likely to cause problems is pretty much impossible - to run through every possible action for something as large as FS would literally take longer than the age of the universe. As a result a script to test for all possible errors is pretty much impossible, especially for new functionality that'll be added before the script's updated to test them. There will always be some bugs, users testing and reporting is the best way to detect ones that the developers have missed. Generally FS is pretty good at avoiding them.

  Testing yourself is always a good idea before rolling out to a production system. That'll show any obvious ones up that can be fixed quickly.

  -Steve



  On 4 February 2011 15:13, Madovsky <infos at madovsky.org> wrote:

    FS team needs user as users need FS team...
    maybe think to a script (developped by all possible users)
    that test every new functionality...


    ----- Original Message -----
    From: "Anthony Minessale" <anthony.minessale at gmail.com>
    To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
    Sent: Friday, February 04, 2011 9:58 AM
    Subject: Re: [Freeswitch-users] Auto test FS install - now necessary?


    > I've been doing this a long time and it boils down to:
    >
    > 50% superstition/paranoia.
    > 50% regressions from commits.
    >
    > Stop and think. Without people updating everyday and telling us when
    > something is not right, we may never find out.
    > Engaging in the community and this mailing list and irc is a community
    > effort and its sometimes a BETA program too.
    > Just like if this was a cool video game, you get to play it before
    > everyone else.
    > It's hard to follow every possible execution path and get into every
    > nook and cranny of 3 million lines of code.
    > Even with the best unit testing int the world, we can't do it without
    > real people testing.
    > If you stop updating or persuade others to stop, now we are in real
    > trouble.
    >
    > Also, pay attention to the GIT commit messages:
    >
    > This particular problem (which is fixed now btw) was committed with a
    > warning to please test.
    > There was a bigger problem earlier this week that resulted in some
    > minority invasive changes and its easy to catch the problems and in
    > the end the software is now more stable.
    >
    > commit f60fdf653dd2d7f8d3eaa6a9086e1f68bd993c59
    > Author: Anthony Minessale <anthm at freeswitch.org>
    > Date:   Wed Feb 2 16:22:43 2011 -0600
    >
    >    fix possible bad pointer in global vars (please test)
    >
    >
    >
    > On Thu, Feb 3, 2011 at 11:46 PM, Avi Marcus <avi at avimarcus.net> wrote:
    >> Maybe I'm just being paranoid, but I've been seeing a lot of "I just
    >> updated
    >> to latest git and X broke"
    >> And that x is always something else...
    >> I'm not a seasoned coder on this magnitude, but perhaps we can develop a
    >> test suite (at least part of one) to test that a fresh install is
    >> functioning?
    >> A FULL test would probably be a huge undertaking and involve recqual for
    >> testing media, too... Also, each mod would need to have it's own test
    >> suite.
    >> Due to the rapid updating of FreeSWITCH this not be what you want to put
    >> your effort into, but...
    >> Thoughts?
    >>
    >> On Feb 4, 2011 4:04 AM, "Anirudha Shimpi" <anu at familytv.com> wrote:
    >>> Just downloaded (from git) and installed FS 1.0.7, however, every time I
    >>> do
    >>> originate user/1001 &park, FS crashes. Running Centos 5.5 64 bit.
    >>>
    >>
    >> _______________________________________________
    >> FreeSWITCH-users mailing list
    >> FreeSWITCH-users at 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 steveayre at gmail.com  Fri Feb  4 19:03:59 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 4 Feb 2011 16:03:59 +0000
Subject: [Freeswitch-users] Auto test FS install - now necessary?
In-Reply-To: <4D4C221F.40905@communicatefreely.net>
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>
	<AANLkTi=Rofy64NwXTt3u1hRRMuAEhimKKux-_zAhTkiS@mail.gmail.com>
	<1105C189976B4970A1F282410B827A7D@e1705>
	<AANLkTik2YTEuXZPXqyMpuuibcxVQo9d1imLGL8+EzzAT@mail.gmail.com>
	<4D4C221F.40905@communicatefreely.net>
Message-ID: <AANLkTim9CPVnx-zcY+Svdi_Zs-ZfmV2NV_bsedUatEL2@mail.gmail.com>

Best source of that information is the commit log
http://fisheye.freeswitch.org/changelog/freeswitch.git

There's also a changelog file that's periodically brought up-to-date.

-Steve


On 4 February 2011 15:58, Tim St. Pierre <fs-list at communicatefreely.net>wrote:

>
> >
> > Testing yourself is always a good idea before rolling out to a
> > production system. That'll show any obvious ones up that can be fixed
> > quickly.
> Not that this will always catch everything, but is there a "release
> notes" or some other simple document that outlines what parts of
> Freeswitch have recently been changed?  We ended up building a sort of
> beta-test network, with a second server and some customers that are very
> benevolent about this sort of thing, and if I had a simple checklist of
> things that had been worked on, I can make sure that I test those areas
> more closely.
>
> I don't have any experience working with software development beyond my
> own little LUA and PHP scripts, so maybe this is already available and I
> just don't know where to look.
>
> Is there a tutorial somewhere on how to be a good user, from a
> developers perspective?
>
> -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
>
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From peter.olsson at visionutveckling.se  Fri Feb  4 19:26:06 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 4 Feb 2011 17:26:06 +0100
Subject: [Freeswitch-users] Auto test FS install - now necessary?
Message-ID: <5AC6B8AE-5AF0-47D9-A914-A7572B6607FE@visionutveckling.se>

I agree, I always follow the commit log, and usually I get the big idea about what's going on.

/Peter

----- Reply message -----
Fr?n: "Steven Ayre" <steveayre at gmail.com>
Datum: fre, feb 4, 2011 23:10
Rubrik: [Freeswitch-users] Auto test FS install - now necessary?
Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

Best source of that information is the commit log http://fisheye.freeswitch.org/changelog/freeswitch.git

There's also a changelog file that's periodically brought up-to-date.

-Steve


On 4 February 2011 15:58, Tim St. Pierre <fs-list at communicatefreely.net<mailto:fs-list at communicatefreely.net>> wrote:

>
> Testing yourself is always a good idea before rolling out to a
> production system. That'll show any obvious ones up that can be fixed
> quickly.
Not that this will always catch everything, but is there a "release
notes" or some other simple document that outlines what parts of
Freeswitch have recently been changed?  We ended up building a sort of
beta-test network, with a second server and some customers that are very
benevolent about this sort of thing, and if I had a simple checklist of
things that had been worked on, I can make sure that I test those areas
more closely.

I don't have any experience working with software development beyond my
own little LUA and PHP scripts, so maybe this is already available and I
just don't know where to look.

Is there a tutorial somewhere on how to be a good user, from a
developers perspective?

-Tim

_______________________________________________
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:4d4c248132761067889338!


From wstephen80 at gmail.com  Fri Feb  4 19:29:28 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Fri, 4 Feb 2011 17:29:28 +0100
Subject: [Freeswitch-users] Lua Session creation is blocking?
Message-ID: <AANLkTin21o-gZH6FrPU6BxFR+RqxBkN15=DDGD66iTX-@mail.gmail.com>

I have a lua script that originate a new call creating a session as:

        outbound_session = freeswitch.Session(dialstring, session);

The problem I have is that session creation sometimes is blocking and
sometimes is not blocking.

To me it seems that is related to the RTP so the session creation is blocked
until early media is present or until the called answer the call.

Is this true or I have miss something?

There is a way to do a non blocking session creation?

Stephen
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From chat2jesse at gmail.com  Fri Feb  4 19:34:33 2011
From: chat2jesse at gmail.com (jesse)
Date: Fri, 4 Feb 2011 08:34:33 -0800
Subject: [Freeswitch-users] Auto test FS install - now necessary?
In-Reply-To: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>
Message-ID: <AANLkTikSryxPi1-YZ8Wpt0GUbcXf_i=4DMi2BmrVPWix@mail.gmail.com>

I do not see any piece of unit testing code neither.
Reliability is more important than feature. Asterisk is a good example of
such case.

I would recommend developers take a pause to add unit testing cases and auto
integration testings.  Also enforce any future code must be accompanied by
unit testing code.
On Feb 3, 2011 9:47 PM, "Avi Marcus" <avi at avimarcus.net> wrote:
> Maybe I'm just being paranoid, but I've been seeing a lot of "I just
updated
> to latest git and X broke"
> And that x is always something else...
> I'm not a seasoned coder on this magnitude, but perhaps we can develop a
> test suite (at least part of one) to test that a fresh install is
> functioning?
> A FULL test would probably be a huge undertaking and involve recqual for
> testing media, too... Also, each mod would need to have it's own test
suite.
> Due to the rapid updating of FreeSWITCH this not be what you want to put
> your effort into, but...
> Thoughts?
> On Feb 4, 2011 4:04 AM, "Anirudha Shimpi" <anu at familytv.com> wrote:
>> Just downloaded (from git) and installed FS 1.0.7, however, every time I
> do
>> originate user/1001 &park, FS crashes. Running Centos 5.5 64 bit.
>>
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From david.ponzone at ipeva.fr  Fri Feb  4 19:44:21 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 4 Feb 2011 17:44:21 +0100
Subject: [Freeswitch-users] Lua Session creation is blocking?
In-Reply-To: <AANLkTin21o-gZH6FrPU6BxFR+RqxBkN15=DDGD66iTX-@mail.gmail.com>
References: <AANLkTin21o-gZH6FrPU6BxFR+RqxBkN15=DDGD66iTX-@mail.gmail.com>
Message-ID: <13B363F1-1AB3-464C-94D0-01BFD52DD2EA@ipeva.fr>

Stephen,

Use bgapi.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 04/02/2011 ? 17:29, Stephen Wilde a ?crit :

> I have a lua script that originate a new call creating a session as:
> 
>         outbound_session = freeswitch.Session(dialstring, session);
> 
> The problem I have is that session creation sometimes is blocking and sometimes is not blocking.
> 
> To me it seems that is related to the RTP so the session creation is blocked until early media is present or until the called answer the call.
> 
> Is this true or I have miss something?
> 
> There is a way to do a non blocking session creation? 
> 
> 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 marcin321 at gmail.com  Fri Feb  4 19:45:21 2011
From: marcin321 at gmail.com (Marcin Wojtowicz)
Date: Fri, 4 Feb 2011 11:45:21 -0500
Subject: [Freeswitch-users] Wrong codec used on outbound leg
Message-ID: <AANLkTik6wtM=wDdYv4GFL4MMW2f+QT6jP=UvO8bHV9st@mail.gmail.com>

I have the following setup:
voip.ms < (external profile) > freeswitch < (internal profile)> my devices.
On my internal profile, the only inbound and outbound codec I allow is iLBC
(I tried 20ms and 30ms), and on my external profile the only inbound and
outbound codec I allow is PCMU. All my internal settings are compatible with
the devices that I'm using, and everything works well when I call from
outside in, and freeswitch now correctly transcodes from PCMU to iLBC.
However, when I call from my internal devices to outside, the iLBC codec is
offered on outbound leg B, not PCMU like I have on my external profile.
Looking at wireshark voip call flow, my machine is sending iLBC to voip.msand
voip.ms is sending me PCMU RTP.
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From steveayre at gmail.com  Fri Feb  4 19:48:10 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 4 Feb 2011 16:48:10 +0000
Subject: [Freeswitch-users] Lua Session creation is blocking?
In-Reply-To: <AANLkTin21o-gZH6FrPU6BxFR+RqxBkN15=DDGD66iTX-@mail.gmail.com>
References: <AANLkTin21o-gZH6FrPU6BxFR+RqxBkN15=DDGD66iTX-@mail.gmail.com>
Message-ID: <AANLkTikdGqM2QsZ19_4YpkDRTd+b5_wRK9NrEGkoqgnM@mail.gmail.com>

Originate doesn't return until receives 18x, 200, 3xx 4xx, 5xx, or 6xx. I
suspect this is the issue.

Not sure what the workaround would be though...

-Steve


On 4 February 2011 16:29, Stephen Wilde <wstephen80 at gmail.com> wrote:

> I have a lua script that originate a new call creating a session as:
>
>         outbound_session = freeswitch.Session(dialstring, session);
>
> The problem I have is that session creation sometimes is blocking and
> sometimes is not blocking.
>
> To me it seems that is related to the RTP so the session creation is
> blocked until early media is present or until the called answer the call.
>
> Is this true or I have miss something?
>
> There is a way to do a non blocking session creation?
>
> 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 steveayre at gmail.com  Fri Feb  4 19:50:04 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 4 Feb 2011 16:50:04 +0000
Subject: [Freeswitch-users] Wrong codec used on outbound leg
In-Reply-To: <AANLkTik6wtM=wDdYv4GFL4MMW2f+QT6jP=UvO8bHV9st@mail.gmail.com>
References: <AANLkTik6wtM=wDdYv4GFL4MMW2f+QT6jP=UvO8bHV9st@mail.gmail.com>
Message-ID: <AANLkTik2z824zzEm_4ZU9bMdcRL021oFQ7BYsdNvB-+s@mail.gmail.com>

What is your dialstring?

Are you dialing out through a user/gateway registered on internal, or using
sofia/internal/ prefix?

A debug-level log with siptrace would also be useful.

-Steve



On 4 February 2011 16:45, Marcin Wojtowicz <marcin321 at gmail.com> wrote:

> I have the following setup:
> voip.ms < (external profile) > freeswitch < (internal profile)> my
> devices. On my internal profile, the only inbound and outbound codec I allow
> is iLBC (I tried 20ms and 30ms), and on my external profile the only inbound
> and outbound codec I allow is PCMU. All my internal settings are compatible
> with the devices that I'm using, and everything works well when I call from
> outside in, and freeswitch now correctly transcodes from PCMU to iLBC.
> However, when I call from my internal devices to outside, the iLBC codec is
> offered on outbound leg B, not PCMU like I have on my external profile.
> Looking at wireshark voip call flow, my machine is sending iLBC to voip.msand
> voip.ms is sending me PCMU RTP.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  4 19:59:33 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 4 Feb 2011 16:59:33 +0000
Subject: [Freeswitch-users] Lua Session creation is blocking?
In-Reply-To: <13B363F1-1AB3-464C-94D0-01BFD52DD2EA@ipeva.fr>
References: <AANLkTin21o-gZH6FrPU6BxFR+RqxBkN15=DDGD66iTX-@mail.gmail.com>
	<13B363F1-1AB3-464C-94D0-01BFD52DD2EA@ipeva.fr>
Message-ID: <AANLkTikQUNbZ3Y9vgmBunScnO30yRZ0rgBEt0i4pF-kv@mail.gmail.com>

Don't think he can do that and get a session object for the call.

-Steve



On 4 February 2011 16:44, David Ponzone <david.ponzone at ipeva.fr> wrote:

> Stephen,
>
> Use bgapi.
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 04/02/2011 ? 17:29, Stephen Wilde a ?crit :
>
> I have a lua script that originate a new call creating a session as:
>
>         outbound_session = freeswitch.Session(dialstring, session);
>
> The problem I have is that session creation sometimes is blocking and
> sometimes is not blocking.
>
> To me it seems that is related to the RTP so the session creation is
> blocked until early media is present or until the called answer the call.
>
> Is this true or I have miss something?
>
> There is a way to do a non blocking session creation?
>
> 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 anthony.minessale at gmail.com  Fri Feb  4 20:01:59 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 4 Feb 2011 11:01:59 -0600
Subject: [Freeswitch-users] Auto test FS install - now necessary?
In-Reply-To: <AANLkTikSryxPi1-YZ8Wpt0GUbcXf_i=4DMi2BmrVPWix@mail.gmail.com>
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>
	<AANLkTikSryxPi1-YZ8Wpt0GUbcXf_i=4DMi2BmrVPWix@mail.gmail.com>
Message-ID: <AANLkTinQArNvg6jZky6=8JUVYif9kQge6bkjnbqNncLL@mail.gmail.com>

I guess we should try to be more like all the other open source
projects where every single day of GIT HEAD is perfect.
hmm....

The alternative is you can buy a soft-switch and live with it's faults
and pay annually for an update.
Or even wait for our official releases which are very infrequent and
not well-supported once they are more than a few months old.

Also don't forget we use FreeSWITCH in a real product so you don't
even see the amount of testing and QA we inherit from that product as
well as all the large carriers who use FreeSWITCH and do their best to
try beta version for us.  Thank you all.....

We have plans to stable branch soon.  I guess that means everyone will
flock to that and it will be the end of my beta testers.
We do not only get regressions from features we get them sometimes
from fixes as well, other peoples patches, many factors.

I guess what's really saddening since you bring up asterisk, is that I
spent years using it with perpetual problems in the release or the
daily snapshot and it sort of has a reputation for this kind of thing
and all you hear about is how awesome it is from people.  I don't
appreciate being treated like a witch hunt over one regression
especially when we can fix it in 1 minute.  That double-standard is
ridiculous.

I should just make a FAQ about this and paste in the URL:

We only have so much time to give to the world, we spend like 8-14
hours a day on average working on the code.
Any suggestions or improvements (even the ones with the best
intentions) are only viable if they come with volunteers.

Bottom line is by downloading and building the code you are opting in
to our community which involves occasional speed bumps.



On Fri, Feb 4, 2011 at 10:34 AM, jesse <chat2jesse at gmail.com> wrote:
> I do not see any piece of unit testing code neither.
> Reliability is more important than feature. Asterisk is a good example of
> such case.
>
> I would recommend developers take a pause to add unit testing cases and auto
> integration testings.? Also enforce any future code must be accompanied by
> unit testing code.
>
> On Feb 3, 2011 9:47 PM, "Avi Marcus" <avi at avimarcus.net> wrote:
>> Maybe I'm just being paranoid, but I've been seeing a lot of "I just
>> updated
>> to latest git and X broke"
>> And that x is always something else...
>> I'm not a seasoned coder on this magnitude, but perhaps we can develop a
>> test suite (at least part of one) to test that a fresh install is
>> functioning?
>> A FULL test would probably be a huge undertaking and involve recqual for
>> testing media, too... Also, each mod would need to have it's own test
>> suite.
>> Due to the rapid updating of FreeSWITCH this not be what you want to put
>> your effort into, but...
>> Thoughts?
>> On Feb 4, 2011 4:04 AM, "Anirudha Shimpi" <anu at familytv.com> wrote:
>>> Just downloaded (from git) and installed FS 1.0.7, however, every time I
>> do
>>> originate user/1001 &park, FS crashes. Running Centos 5.5 64 bit.
>>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb  4 20:07:29 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 4 Feb 2011 11:07:29 -0600
Subject: [Freeswitch-users] Wrong codec used on outbound leg
In-Reply-To: <AANLkTik6wtM=wDdYv4GFL4MMW2f+QT6jP=UvO8bHV9st@mail.gmail.com>
References: <AANLkTik6wtM=wDdYv4GFL4MMW2f+QT6jP=UvO8bHV9st@mail.gmail.com>
Message-ID: <AANLkTikbYSCPwsN9DhPj8WB30rPVqe3n66YGtzRpRg4G@mail.gmail.com>

try putting your dial string to your provider like this:

{absolute_codec_string=PCMU}sofia/foo/foo at bar.com


On Fri, Feb 4, 2011 at 10:45 AM, Marcin Wojtowicz <marcin321 at gmail.com> wrote:
> I have the following setup:
> voip.ms < (external profile) > freeswitch < (internal profile)> my devices.
> On my internal profile, the only inbound and outbound codec I allow is iLBC
> (I tried 20ms and 30ms), and on my external profile the only inbound and
> outbound codec I allow is PCMU. All my internal settings are compatible with
> the devices that I'm using, and everything works well when I call from
> outside in, and freeswitch now correctly transcodes from PCMU to iLBC.
> However, when I call from my internal devices to outside, the iLBC codec is
> offered on outbound leg B, not PCMU like I have on my external profile.
> Looking at wireshark voip call flow, my machine is sending iLBC to voip.ms
> and voip.ms is sending me PCMU RTP.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 darren at aleph-com.net  Fri Feb  4 20:09:27 2011
From: darren at aleph-com.net (Darren Wiebe)
Date: Fri, 04 Feb 2011 10:09:27 -0700
Subject: [Freeswitch-users] Auto test FS install - now necessary?
In-Reply-To: <AANLkTinQArNvg6jZky6=8JUVYif9kQge6bkjnbqNncLL@mail.gmail.com>
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>	<AANLkTikSryxPi1-YZ8Wpt0GUbcXf_i=4DMi2BmrVPWix@mail.gmail.com>
	<AANLkTinQArNvg6jZky6=8JUVYif9kQge6bkjnbqNncLL@mail.gmail.com>
Message-ID: <4D4C32C7.9050503@aleph-com.net>

On 04/02/2011 10:01 AM, Anthony Minessale wrote:
> Reliability is more important than feature. Asterisk is a good example of
> >  such case.
I snipped a very small part of this thread but the above line provided 
my comedy relief for the day.  Thanks, I needed it.

-- 
Darren Wiebe
Aleph Communications
--------------------
Phone: 1-877-702-2900
Fax:   1-866-274-4506
Email: darren at aleph-com.net



From marcin321 at gmail.com  Fri Feb  4 20:19:11 2011
From: marcin321 at gmail.com (Marcin Wojtowicz)
Date: Fri, 4 Feb 2011 12:19:11 -0500
Subject: [Freeswitch-users] Wrong codec used on outbound leg
In-Reply-To: <AANLkTikbYSCPwsN9DhPj8WB30rPVqe3n66YGtzRpRg4G@mail.gmail.com>
References: <AANLkTik6wtM=wDdYv4GFL4MMW2f+QT6jP=UvO8bHV9st@mail.gmail.com>
	<AANLkTikbYSCPwsN9DhPj8WB30rPVqe3n66YGtzRpRg4G@mail.gmail.com>
Message-ID: <AANLkTin7CFEwCx2N_8WS1uO9bDRzZPh+dhTYSc5V7bBG@mail.gmail.com>

Thank you, the absolute codec string addition worked. My original dialstring
was <action application="bridge" data="sofia/gateway/voip.ms/1$1"/>, BTW
(where voip.ms is imported by the external profile). I just have one more
unrelated question about voice sample times. If one leg is using 20ms and
the other 30ms, does freeswitch introduce a buffer and if so, how much delay
is it?

On Fri, Feb 4, 2011 at 12:07 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> try putting your dial string to your provider like this:
>
> {absolute_codec_string=PCMU}sofia/foo/foo at bar.com
>
>
> On Fri, Feb 4, 2011 at 10:45 AM, Marcin Wojtowicz <marcin321 at gmail.com>
> wrote:
> > I have the following setup:
> > voip.ms < (external profile) > freeswitch < (internal profile)> my
> devices.
> > On my internal profile, the only inbound and outbound codec I allow is
> iLBC
> > (I tried 20ms and 30ms), and on my external profile the only inbound and
> > outbound codec I allow is PCMU. All my internal settings are compatible
> with
> > the devices that I'm using, and everything works well when I call from
> > outside in, and freeswitch now correctly transcodes from PCMU to iLBC.
> > However, when I call from my internal devices to outside, the iLBC codec
> is
> > offered on outbound leg B, not PCMU like I have on my external profile.
> > Looking at wireshark voip call flow, my machine is sending iLBC to
> voip.ms
> > and voip.ms is sending me PCMU RTP.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  4 20:20:04 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 4 Feb 2011 17:20:04 +0000
Subject: [Freeswitch-users] Auto test FS install - now necessary?
In-Reply-To: <AANLkTinQArNvg6jZky6=8JUVYif9kQge6bkjnbqNncLL@mail.gmail.com>
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>
	<AANLkTikSryxPi1-YZ8Wpt0GUbcXf_i=4DMi2BmrVPWix@mail.gmail.com>
	<AANLkTinQArNvg6jZky6=8JUVYif9kQge6bkjnbqNncLL@mail.gmail.com>
Message-ID: <AANLkTiksK5a8W1BEapip2cmBW-V0MvbBTq1dB+0acsBj@mail.gmail.com>

Just to say I think you all do a great job with FreeSWITCH. :)

-Steve



On 4 February 2011 17:01, Anthony Minessale <anthony.minessale at gmail.com>wrote:

> I guess we should try to be more like all the other open source
> projects where every single day of GIT HEAD is perfect.
> hmm....
>
> The alternative is you can buy a soft-switch and live with it's faults
> and pay annually for an update.
> Or even wait for our official releases which are very infrequent and
> not well-supported once they are more than a few months old.
>
> Also don't forget we use FreeSWITCH in a real product so you don't
> even see the amount of testing and QA we inherit from that product as
> well as all the large carriers who use FreeSWITCH and do their best to
> try beta version for us.  Thank you all.....
>
> We have plans to stable branch soon.  I guess that means everyone will
> flock to that and it will be the end of my beta testers.
> We do not only get regressions from features we get them sometimes
> from fixes as well, other peoples patches, many factors.
>
> I guess what's really saddening since you bring up asterisk, is that I
> spent years using it with perpetual problems in the release or the
> daily snapshot and it sort of has a reputation for this kind of thing
> and all you hear about is how awesome it is from people.  I don't
> appreciate being treated like a witch hunt over one regression
> especially when we can fix it in 1 minute.  That double-standard is
> ridiculous.
>
> I should just make a FAQ about this and paste in the URL:
>
> We only have so much time to give to the world, we spend like 8-14
> hours a day on average working on the code.
> Any suggestions or improvements (even the ones with the best
> intentions) are only viable if they come with volunteers.
>
> Bottom line is by downloading and building the code you are opting in
> to our community which involves occasional speed bumps.
>
>
>
> On Fri, Feb 4, 2011 at 10:34 AM, jesse <chat2jesse at gmail.com> wrote:
> > I do not see any piece of unit testing code neither.
> > Reliability is more important than feature. Asterisk is a good example of
> > such case.
> >
> > I would recommend developers take a pause to add unit testing cases and
> auto
> > integration testings.  Also enforce any future code must be accompanied
> by
> > unit testing code.
> >
> > On Feb 3, 2011 9:47 PM, "Avi Marcus" <avi at avimarcus.net> wrote:
> >> Maybe I'm just being paranoid, but I've been seeing a lot of "I just
> >> updated
> >> to latest git and X broke"
> >> And that x is always something else...
> >> I'm not a seasoned coder on this magnitude, but perhaps we can develop a
> >> test suite (at least part of one) to test that a fresh install is
> >> functioning?
> >> A FULL test would probably be a huge undertaking and involve recqual for
> >> testing media, too... Also, each mod would need to have it's own test
> >> suite.
> >> Due to the rapid updating of FreeSWITCH this not be what you want to put
> >> your effort into, but...
> >> Thoughts?
> >> On Feb 4, 2011 4:04 AM, "Anirudha Shimpi" <anu at familytv.com> wrote:
> >>> Just downloaded (from git) and installed FS 1.0.7, however, every time
> I
> >> do
> >>> originate user/1001 &park, FS crashes. Running Centos 5.5 64 bit.
> >>>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  4 20:20:24 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 4 Feb 2011 17:20:24 +0000
Subject: [Freeswitch-users] Auto test FS install - now necessary?
In-Reply-To: <4D4C32C7.9050503@aleph-com.net>
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>
	<AANLkTikSryxPi1-YZ8Wpt0GUbcXf_i=4DMi2BmrVPWix@mail.gmail.com>
	<AANLkTinQArNvg6jZky6=8JUVYif9kQge6bkjnbqNncLL@mail.gmail.com>
	<4D4C32C7.9050503@aleph-com.net>
Message-ID: <AANLkTinz-XCQHVympCqjzBL5=pfCOXiSPqNqbho7UNmX@mail.gmail.com>

I actually spilt my tea. :)


On 4 February 2011 17:09, Darren Wiebe <darren at aleph-com.net> wrote:

> On 04/02/2011 10:01 AM, Anthony Minessale wrote:
> > Reliability is more important than feature. Asterisk is a good example of
> > >  such case.
> I snipped a very small part of this thread but the above line provided
> my comedy relief for the day.  Thanks, I needed it.
>
> --
> Darren Wiebe
> Aleph Communications
> --------------------
> Phone: 1-877-702-2900
> Fax:   1-866-274-4506
> Email: darren at aleph-com.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
>
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From phone.bytes at gmail.com  Fri Feb  4 20:31:51 2011
From: phone.bytes at gmail.com (Phone)
Date: Fri, 04 Feb 2011 10:31:51 -0700
Subject: [Freeswitch-users] Auto test FS install - now necessary?
In-Reply-To: <AANLkTinQArNvg6jZky6=8JUVYif9kQge6bkjnbqNncLL@mail.gmail.com>
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>	<AANLkTikSryxPi1-YZ8Wpt0GUbcXf_i=4DMi2BmrVPWix@mail.gmail.com>
	<AANLkTinQArNvg6jZky6=8JUVYif9kQge6bkjnbqNncLL@mail.gmail.com>
Message-ID: <4D4C3807.2010600@gmail.com>

I hope that we don't ever return to the old days of spending a year or 
more regression testing  PBX firmware between each release.  That was 
okay, if you did not mind having a product that worked really well but 
did not really do much and was always about a year behind the cutting edge.

I applaud all of the FS Devs, I believe that everyone is doing their 
best to avoid introducing new issues as work is done to improve the code 
base and add new features and functionality.

We are happy to participate in the testing and feedback process.
We hope to be able to make meaningful contributions down the road when 
we are more familiar with things.

I have never seen a major issue that was not corrected with lightning speed.
It seems that even whenever someone reports an issue that is important 
to them, they are likewise addressed very rapidly.

In my opinion, this is just the way that software development goes.  Do 
your best to modify (improve the project) code without causing other 
issues.  Unfortunately, sometimes this happens.  Especially when there 
so many people working to make contributions to grow and improve the 
product for all.  Frankly, I think the project is progressing very well.





On 02/04/2011 10:01 AM, Anthony Minessale wrote:
> I guess we should try to be more like all the other open source
> projects where every single day of GIT HEAD is perfect.
> hmm....
>
> The alternative is you can buy a soft-switch and live with it's faults
> and pay annually for an update.
> Or even wait for our official releases which are very infrequent and
> not well-supported once they are more than a few months old.
>
> Also don't forget we use FreeSWITCH in a real product so you don't
> even see the amount of testing and QA we inherit from that product as
> well as all the large carriers who use FreeSWITCH and do their best to
> try beta version for us.  Thank you all.....
>
> We have plans to stable branch soon.  I guess that means everyone will
> flock to that and it will be the end of my beta testers.
> We do not only get regressions from features we get them sometimes
> from fixes as well, other peoples patches, many factors.
>
> I guess what's really saddening since you bring up asterisk, is that I
> spent years using it with perpetual problems in the release or the
> daily snapshot and it sort of has a reputation for this kind of thing
> and all you hear about is how awesome it is from people.  I don't
> appreciate being treated like a witch hunt over one regression
> especially when we can fix it in 1 minute.  That double-standard is
> ridiculous.
>
> I should just make a FAQ about this and paste in the URL:
>
> We only have so much time to give to the world, we spend like 8-14
> hours a day on average working on the code.
> Any suggestions or improvements (even the ones with the best
> intentions) are only viable if they come with volunteers.
>
> Bottom line is by downloading and building the code you are opting in
> to our community which involves occasional speed bumps.
>
>
>



From Nabble at slickdeals.endjunk.com  Fri Feb  4 20:34:40 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 4 Feb 2011 09:34:40 -0800 (PST)
Subject: [Freeswitch-users] Auto test FS install - now necessary?
In-Reply-To: <AANLkTinz-XCQHVympCqjzBL5=pfCOXiSPqNqbho7UNmX@mail.gmail.com>
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>
	<AANLkTikSryxPi1-YZ8Wpt0GUbcXf_i=4DMi2BmrVPWix@mail.gmail.com>
	<AANLkTinQArNvg6jZky6=8JUVYif9kQge6bkjnbqNncLL@mail.gmail.com>
	<4D4C32C7.9050503@aleph-com.net>
	<AANLkTinz-XCQHVympCqjzBL5=pfCOXiSPqNqbho7UNmX@mail.gmail.com>
Message-ID: <1296840880862-5993296.post@n2.nabble.com>



Steven Ayre wrote:
> 
> I actually spilt my tea. :)
Honestly, I have no idea what your statement means. However, I am using both
Asterisk and FreeSWITCH. The only reason I still keep my Asterisk is because
it is configured with a BlackList and incoming calls with CIDs not listed in
the BlackList will be rejected. When I find a way to do this, I definitely
will decommission my Asterisk.

-----
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/Auto-test-FS-install-now-necessary-tp5991624p5993296.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From james.royer at gmail.com  Fri Feb  4 05:27:02 2011
From: james.royer at gmail.com (james.royer at gmail.com)
Date: Thu, 3 Feb 2011 18:27:02 -0800
Subject: [Freeswitch-users] curl: parse XML response?
Message-ID: <AANLkTik3BoGujLx9QBvJg3SWNByarca822ryxBFzaWoK@mail.gmail.com>

Hi All,

I want to use mod_curl to query a web service and then extract information
from the XML response.

<response>
  <value1>test</value1>
  <value2>blah</value2>
</response>

I then want to use TTS to speak the values to the caller.

How can I do this?

I tried searches for "freeswitch parse xml" and other variations but all
just talk about how FreeSWITCH parses config files.

Thanks,
James
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From erick at teal.net  Fri Feb  4 12:00:05 2011
From: erick at teal.net (Erick Baum)
Date: Fri, 4 Feb 2011 01:00:05 -0800
Subject: [Freeswitch-users] mod_directory multi-tenant
Message-ID: <000901cbc449$ea673480$bf359d80$@net>

Is there a trick to getting the mod_directory application working in a
multi-tenant configuration?  I think I've tried every possible combination
of settings and I cannot for the life of me get it to work.  I can get the
directory to answer but no matter what I put in to the search, it comes back
with no results.  Any help would be appreciated!

 

Erick

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From vedran.zeljeznak at gmail.com  Fri Feb  4 15:55:23 2011
From: vedran.zeljeznak at gmail.com (Vedran Zeljeznak)
Date: Fri, 4 Feb 2011 13:55:23 +0100
Subject: [Freeswitch-users] H.323 implementation on FS
Message-ID: <AANLkTin_cDiA9PWcrhO6qSYwzamnxUAJZcOEZy2HMrkC@mail.gmail.com>

hi everyone,

do any of you have any suggestions which H.323 implementation
(mod_opal or mod_h323) should be used on a production Freeswitch
platform (HEAD version)?

Is it better to forward calls from H.323 trunk through Yate
(configured as H.323 to SIP proxy) before terminating them on
Freeswitch?

---
Vedran Zeljeznak


From vedran.zeljeznak at gmail.com  Fri Feb  4 16:18:14 2011
From: vedran.zeljeznak at gmail.com (Vedran Zeljeznak)
Date: Fri, 4 Feb 2011 14:18:14 +0100
Subject: [Freeswitch-users] H.323 implementation on FS
Message-ID: <AANLkTimbQASu1taVqpWi3VFjq4=tFmBhbxyvUL7LcLtZ@mail.gmail.com>

hi everyone,

do any of you have any suggestions which H.323 implementation
(mod_opal or mod_h323) should be used on a production Freeswitch
platform (HEAD version)?

Is it better to forward calls from H.323 trunk through Yate
(configured as H.323 to SIP proxy) before terminating them on
Freeswitch?

---
Vedran Zeljeznak


From james.royer at gmail.com  Fri Feb  4 20:37:31 2011
From: james.royer at gmail.com (james.royer at gmail.com)
Date: Fri, 4 Feb 2011 09:37:31 -0800
Subject: [Freeswitch-users] mod_curl: how to parse xml response?
Message-ID: <AANLkTinbbRufHM7nPkgtW=HXyos18KdhnRgmitVLS7NN@mail.gmail.com>

Hi All,

(This is my second post - are posts delayed? I never received an
acknowledgment message.)

I want to use mod_curl to query a web service and then extract information
from the XML response.

<response>
  <value1>test</value1>
  <value2>blah</value2>
</response>

I then want to use TTS to speak the values to the caller.

How can I do this?

I tried searches for "freeswitch parse xml" and other variations but all
just talk about how FreeSWITCH parses config files.

I see that people have created libraries for Lua proper but looking at the
doc, there doesn't appear to be any functions to parse XML from within Lua.

Thanks,
James
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From grsingh750 at gmail.com  Fri Feb  4 20:52:05 2011
From: grsingh750 at gmail.com (guru singh)
Date: Fri, 4 Feb 2011 23:22:05 +0530
Subject: [Freeswitch-users] Auto test FS install - now necessary?
In-Reply-To: <AANLkTik2YTEuXZPXqyMpuuibcxVQo9d1imLGL8+EzzAT@mail.gmail.com>
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>
	<AANLkTi=Rofy64NwXTt3u1hRRMuAEhimKKux-_zAhTkiS@mail.gmail.com>
	<1105C189976B4970A1F282410B827A7D@e1705>
	<AANLkTik2YTEuXZPXqyMpuuibcxVQo9d1imLGL8+EzzAT@mail.gmail.com>
Message-ID: <AANLkTin7X195XmHYPcSKvqyWVXVF7dV2R7T9Touco716@mail.gmail.com>

On Fri, Feb 4, 2011 at 9:15 PM, Steven Ayre <steveayre at gmail.com> wrote:


> Testing yourself is always a good idea before rolling out to a production
> system. That'll show any obvious ones up that can be fixed quickly.

On a related note, After thorough testing, if a system is put in
production. Would it be 'best practice' not to update it, if no config
changes or new functionality is ever desired?

> -Steve

guru


From msc at freeswitch.org  Fri Feb  4 21:09:20 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 4 Feb 2011 10:09:20 -0800
Subject: [Freeswitch-users] context
In-Reply-To: <AANLkTik7xyGdbMvfnZbvOWX-P8XUb+zvchQ4X5stAgFQ@mail.gmail.com>
References: <AANLkTika5CBO5eVWuRimKY+Bth2YAFC0Tm8_JmFBnXwC@mail.gmail.com>
	<AANLkTim84VqQ79xq6amru_4GzvGP9rof001MJ8t2Zpvd@mail.gmail.com>
	<AANLkTinutzgJ5GF_tP=gpfVaz03+CQ-b-CCjPh3rFL=_@mail.gmail.com>
	<AANLkTikmnwQjXSX9GaC0Q3DGQYOTpv9-LE29AFW-VA4T@mail.gmail.com>
	<AANLkTik7xyGdbMvfnZbvOWX-P8XUb+zvchQ4X5stAgFQ@mail.gmail.com>
Message-ID: <AANLkTi=DHjTttS2E9jHOnj_uB8KHaE1zE7gZtfZzsmS2@mail.gmail.com>

On Thu, Feb 3, 2011 at 7:24 PM, Sam <u2nsam at gmail.com> wrote:

> It all happened with the polycom phone not getting registered on FS due to
> nat/pat environment and rest of other make phone getting registered.
> So added phone ip to acl and tried NDLB and all and landed up getting the
> call flow reaching public context and it got out of my mind what changes i
> made.
> But thanks for you help MC.
>
> Regards
> Sam
>

Thanks for letting us know what happened. :)
-MC
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From msc at freeswitch.org  Fri Feb  4 21:12:47 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 4 Feb 2011 10:12:47 -0800
Subject: [Freeswitch-users] Grandstream Freeze on Originate command
In-Reply-To: <1296821764473-5992350.post@n2.nabble.com>
References: <1296751186605-5989433.post@n2.nabble.com>
	<AANLkTikPiZgq9WsFeMs2U0iZrT7FcL10dptLCFu5HynP@mail.gmail.com>
	<1296818499616-5992217.post@n2.nabble.com>
	<1296821764473-5992350.post@n2.nabble.com>
Message-ID: <AANLkTim+3c7Ui=bPd6_7ROXEA+mxgX3RRzCf03PWdTmi@mail.gmail.com>

Do you have an example of an invite to the GS, perhaps from Asterisk or
another SIP server, that does not freeze the phone? My guess is that GS is
doing something silly. (They're known for doing silly things and freezing up
when you send them an invite is, unfortunately, not at all surprising given
the reputation they've acquired for themselves.)

Wish I could help more...

-MC
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From darren at aleph-com.net  Fri Feb  4 21:12:15 2011
From: darren at aleph-com.net (Darren Wiebe)
Date: Fri, 04 Feb 2011 11:12:15 -0700
Subject: [Freeswitch-users] Auto test FS install - now necessary?
In-Reply-To: <1296840880862-5993296.post@n2.nabble.com>
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>	<AANLkTikSryxPi1-YZ8Wpt0GUbcXf_i=4DMi2BmrVPWix@mail.gmail.com>	<AANLkTinQArNvg6jZky6=8JUVYif9kQge6bkjnbqNncLL@mail.gmail.com>	<4D4C32C7.9050503@aleph-com.net>	<AANLkTinz-XCQHVympCqjzBL5=pfCOXiSPqNqbho7UNmX@mail.gmail.com>
	<1296840880862-5993296.post@n2.nabble.com>
Message-ID: <4D4C417F.9020004@aleph-com.net>

On 04/02/2011 10:34 AM, mazilo wrote:
>
> Steven Ayre wrote:
>> I actually spilt my tea. :)
> Honestly, I have no idea what your statement means. However, I am using both
> Asterisk and FreeSWITCH. The only reason I still keep my Asterisk is because
> it is configured with a BlackList and incoming calls with CIDs not listed in
> the BlackList will be rejected. When I find a way to do this, I definitely
> will decommission my Asterisk.
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
That should be fairly easy using either mod_xml_curl or lua.

-- 
Darren Wiebe
Aleph Communications
--------------------
Phone: 1-877-702-2900
Fax:   1-866-274-4506
Email: darren at aleph-com.net



From wstephen80 at gmail.com  Fri Feb  4 21:19:26 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Fri, 4 Feb 2011 19:19:26 +0100
Subject: [Freeswitch-users] Lua Session creation is blocking?
In-Reply-To: <AANLkTikdGqM2QsZ19_4YpkDRTd+b5_wRK9NrEGkoqgnM@mail.gmail.com>
References: <AANLkTin21o-gZH6FrPU6BxFR+RqxBkN15=DDGD66iTX-@mail.gmail.com>
	<AANLkTikdGqM2QsZ19_4YpkDRTd+b5_wRK9NrEGkoqgnM@mail.gmail.com>
Message-ID: <AANLkTinaixoUDTyR-5gPZPLSaO_W0eNvC491ApAUHg-O@mail.gmail.com>

Ok, 18x will be perfect!

The problem is that when the inbound call is SIP and the originated is ISDN,
the originate doesn't return when a 180 ringing is received (that in ISDN is
an ALERT without in band info).

The originate returns when a 183 (ISDN = in band info available) or 200
(ISDN = CONNECT) is received.

Stephen


On Fri, Feb 4, 2011 at 5:48 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Originate doesn't return until receives 18x, 200, 3xx 4xx, 5xx, or 6xx. I
> suspect this is the issue.
>
> Not sure what the workaround would be though...
>
> -Steve
>
>
> On 4 February 2011 16:29, Stephen Wilde <wstephen80 at gmail.com> wrote:
>
>> I have a lua script that originate a new call creating a session as:
>>
>>         outbound_session = freeswitch.Session(dialstring, session);
>>
>> The problem I have is that session creation sometimes is blocking and
>> sometimes is not blocking.
>>
>> To me it seems that is related to the RTP so the session creation is
>> blocked until early media is present or until the called answer the call.
>>
>> Is this true or I have miss something?
>>
>> There is a way to do a non blocking session creation?
>>
>> 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 djbinter at gmail.com  Fri Feb  4 21:21:56 2011
From: djbinter at gmail.com (DJB International)
Date: Fri, 4 Feb 2011 10:21:56 -0800
Subject: [Freeswitch-users] Polycom Issues
In-Reply-To: <86E6544D-9446-45F6-8FF7-71A24C8294C3@ipeva.fr>
References: <AANLkTinTiKQX_H=oVFn1KcL9GfqvVxq=AE6+CM7jQPwb@mail.gmail.com>
	<86E6544D-9446-45F6-8FF7-71A24C8294C3@ipeva.fr>
Message-ID: <AANLkTi=kyR3=tpNORYM5aun1qoho0h3-mprf7Vc1qyhR@mail.gmail.com>

I finally found out the problem.  This extension is used with internet
provided from Centurylink (Centurylink 660 ADSL modem).  I had to make a
change in the modem to bridge mode.  Then, it resolved the issue of double
natted.  Now, the phone is working great.  Just in case anyone experience
the same problem.

-djbinter

On Tue, Feb 1, 2011 at 6:39 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Well, I don't use Polycom, but I've seen such behaviour in the past with
> other phones, and most of the time, the cause is that the phone is not
> receiving the 401 Challenge from FreeSWITCH, so it sends back another
> REGISTER.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 02/02/2011 ? 03:27, DJB International a ?crit :
>
> Has anyone ever seen Polycom 650 send REGISTER without Authorization even
> though FS responds with 401?
>
> The weird thing was that it only happened to this particular Polycom phone
> only.
>
> Please see below:
>
>    ------------------------------------------------------------------------
> recv 559 bytes from udp/[67.232.144.163]:10058 at 02:24:22.100024:
>    ------------------------------------------------------------------------
>    REGISTER sip:199.87.44.19:5060 SIP/2.0
>    Via: SIP/2.0/UDP 67.232.144.163:10003;branch=z9hG4bK85aaf1325070992B
>    From: "Mark" <sip:7025143416 at 199.87.44.19<sip%3A7025143416 at 199.87.44.19>
> >;tag=EED068F-AC7EFF94
>    To: <sip:7025143416 at 199.87.44.19 <sip%3A7025143416 at 199.87.44.19>>
>    CSeq: 1 REGISTER
>    Call-ID: 37fb35c0-279ee281-fd8694be at 67.232.144.163
>    Contact: <sip:7025143416 at 67.232.144.163:10003>;methods="INVITE, ACK,
> BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE,
> REFER"
>    User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.4.0244
>    Accept-Language: en
>    Max-Forwards: 70
>    Expires: 3600
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> 2011-02-01 18:24:28.858417 [WARNING] sofia_reg.c:1247 SIP auth challenge
> (REGISTER) on sofia profile 'internal' for [7025143416 at 199.87.44.19] from
> ip 67.232.144.163
> send 676 bytes to udp/[67.232.144.163]:10058 at 02:24:22.174807:
>    ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/UDP 67.232.144.163:10003
> ;branch=z9hG4bK85aaf1325070992B;rport=10058
>    From: "Mark" <sip:7025143416 at 199.87.44.19<sip%3A7025143416 at 199.87.44.19>
> >;tag=EED068F-AC7EFF94
>    To: <sip:7025143416 at 199.87.44.19 <sip%3A7025143416 at 199.87.44.19>
> >;tag=5Byyp046mycjS
>    Call-ID: 37fb35c0-279ee281-fd8694be at 67.232.144.163
>    CSeq: 1 REGISTER
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9ffca05 2011-01-26
> 17-24-25 -0500
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, precondition, path, replaces
>    WWW-Authenticate: Digest realm="199.87.44.19",
> nonce="4b7a72c9-47ad-4d84-b1a1-fe0f68194efd", algorithm=MD5, qop="auth"
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 559 bytes from udp/[67.232.144.163]:10058 at 02:24:22.628250:
>    ------------------------------------------------------------------------
>    REGISTER sip:199.87.44.19:5060 SIP/2.0
>    Via: SIP/2.0/UDP 67.232.144.163:10003;branch=z9hG4bK85aaf1325070992B
>    From: "Mark" <sip:7025143416 at 199.87.44.19<sip%3A7025143416 at 199.87.44.19>
> >;tag=EED068F-AC7EFF94
>    To: <sip:7025143416 at 199.87.44.19 <sip%3A7025143416 at 199.87.44.19>>
>    CSeq: 1 REGISTER
>    Call-ID: 37fb35c0-279ee281-fd8694be at 67.232.144.163
>    Contact: <sip:7025143416 at 67.232.144.163:10003>;methods="INVITE, ACK,
> BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE,
> REFER"
>    User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.4.0244
>    Accept-Language: en
>    Max-Forwards: 70
>    Expires: 3600
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> send 676 bytes to udp/[67.232.144.163]:10058 at 02:24:22.628449:
>    ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/UDP 67.232.144.163:10003
> ;branch=z9hG4bK85aaf1325070992B;rport=10058
>    From: "Mark" <sip:7025143416 at 199.87.44.19<sip%3A7025143416 at 199.87.44.19>
> >;tag=EED068F-AC7EFF94
>    To: <sip:7025143416 at 199.87.44.19 <sip%3A7025143416 at 199.87.44.19>
> >;tag=5Byyp046mycjS
>    Call-ID: 37fb35c0-279ee281-fd8694be at 67.232.144.163
>    CSeq: 1 REGISTER
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9ffca05 2011-01-26
> 17-24-25 -0500
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, precondition, path, replaces
>    WWW-Authenticate: Digest realm="199.87.44.19",
> nonce="4b7a72c9-47ad-4d84-b1a1-fe0f68194efd", algorithm=MD5, qop="auth"
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 559 bytes from udp/[67.232.144.163]:10058 at 02:24:23.628660:
>    ------------------------------------------------------------------------
>    REGISTER sip:199.87.44.19:5060 SIP/2.0
>    Via: SIP/2.0/UDP 67.232.144.163:10003;branch=z9hG4bK85aaf1325070992B
>    From: "Mark" <sip:7025143416 at 199.87.44.19<sip%3A7025143416 at 199.87.44.19>
> >;tag=EED068F-AC7EFF94
>    To: <sip:7025143416 at 199.87.44.19 <sip%3A7025143416 at 199.87.44.19>>
>    CSeq: 1 REGISTER
>    Call-ID: 37fb35c0-279ee281-fd8694be at 67.232.144.163
>    Contact: <sip:7025143416 at 67.232.144.163:10003>;methods="INVITE, ACK,
> BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE,
> REFER"
>    User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.4.0244
>    Accept-Language: en
>    Max-Forwards: 70
>    Expires: 3600
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> send 676 bytes to udp/[67.232.144.163]:10058 at 02:24:23.628895:
>    ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/UDP 67.232.144.163:10003
> ;branch=z9hG4bK85aaf1325070992B;rport=10058
>    From: "Mark" <sip:7025143416 at 199.87.44.19<sip%3A7025143416 at 199.87.44.19>
> >;tag=EED068F-AC7EFF94
>    To: <sip:7025143416 at 199.87.44.19 <sip%3A7025143416 at 199.87.44.19>
> >;tag=5Byyp046mycjS
>    Call-ID: 37fb35c0-279ee281-fd8694be at 67.232.144.163
>    CSeq: 1 REGISTER
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9ffca05 2011-01-26
> 17-24-25 -0500
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, precondition, path, replaces
>    WWW-Authenticate: Digest realm="199.87.44.19",
> nonce="4b7a72c9-47ad-4d84-b1a1-fe0f68194efd", algorithm=MD5, qop="auth"
>    Content-Length: 0
>
>
> Thank you.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  4 21:22:45 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 4 Feb 2011 10:22:45 -0800
Subject: [Freeswitch-users] Auto test FS install - now necessary?
In-Reply-To: <AANLkTin7X195XmHYPcSKvqyWVXVF7dV2R7T9Touco716@mail.gmail.com>
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>
	<AANLkTi=Rofy64NwXTt3u1hRRMuAEhimKKux-_zAhTkiS@mail.gmail.com>
	<1105C189976B4970A1F282410B827A7D@e1705>
	<AANLkTik2YTEuXZPXqyMpuuibcxVQo9d1imLGL8+EzzAT@mail.gmail.com>
	<AANLkTin7X195XmHYPcSKvqyWVXVF7dV2R7T9Touco716@mail.gmail.com>
Message-ID: <AANLkTi=8ahEit6vG_mzvjb_EbFC37Tm_DNK-qbNe8aLB@mail.gmail.com>

On Fri, Feb 4, 2011 at 9:52 AM, guru singh <grsingh750 at gmail.com> wrote:

> On Fri, Feb 4, 2011 at 9:15 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>
> > Testing yourself is always a good idea before rolling out to a production
> > system. That'll show any obvious ones up that can be fixed quickly.
>
> On a related note, After thorough testing, if a system is put in
> production. Would it be 'best practice' not to update it, if no config
> changes or new functionality is ever desired?
>

This depends greatly on your scenario. Personally, I think that anyone who
puts FS into production (i.e. in a place where it helps one make money)
should *ALWAYS* have a backup/sandbox/test system for handling updates. If
you are in a mission-critical scenario then it probably means that once you
get it working you don't touch it unless you absolutely have to. That's
where your test system comes in to play. Ideally you have the exact same
hardware on your test system that you do in production. Then your test
system can double as a warm standby.

Just my $.02, but I think that commodity hardware being so cheap these days
that there's no excuse for a for-profit venture using FS (or any other
software, for that matter) not to have a test system for mission-critical
servers.

-MC
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From david.ponzone at ipeva.fr  Fri Feb  4 21:22:22 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 4 Feb 2011 19:22:22 +0100
Subject: [Freeswitch-users] Auto test FS install - now necessary?
In-Reply-To: <1296840880862-5993296.post@n2.nabble.com>
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>
	<AANLkTikSryxPi1-YZ8Wpt0GUbcXf_i=4DMi2BmrVPWix@mail.gmail.com>
	<AANLkTinQArNvg6jZky6=8JUVYif9kQge6bkjnbqNncLL@mail.gmail.com>
	<4D4C32C7.9050503@aleph-com.net>
	<AANLkTinz-XCQHVympCqjzBL5=pfCOXiSPqNqbho7UNmX@mail.gmail.com>
	<1296840880862-5993296.post@n2.nabble.com>
Message-ID: <7D4B9BFA-023D-4426-93C2-F6BDE2CCBF9C@ipeva.fr>

So it's a whitelist, and that is quite easy to do with FreeSWITCH, with some LUA, and a SQL DB.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 04/02/2011 ? 18:34, mazilo a ?crit :

> 
> 
> Steven Ayre wrote:
>> 
>> I actually spilt my tea. :)
> Honestly, I have no idea what your statement means. However, I am using both
> Asterisk and FreeSWITCH. The only reason I still keep my Asterisk is because
> it is configured with a BlackList and incoming calls with CIDs not listed in
> the BlackList will be rejected. When I find a way to do this, I definitely
> will decommission my Asterisk.
> 
> -----
> 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/Auto-test-FS-install-now-necessary-tp5991624p5993296.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  Fri Feb  4 21:26:24 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 4 Feb 2011 10:26:24 -0800
Subject: [Freeswitch-users] Lua Session creation is blocking?
In-Reply-To: <AANLkTinaixoUDTyR-5gPZPLSaO_W0eNvC491ApAUHg-O@mail.gmail.com>
References: <AANLkTin21o-gZH6FrPU6BxFR+RqxBkN15=DDGD66iTX-@mail.gmail.com>
	<AANLkTikdGqM2QsZ19_4YpkDRTd+b5_wRK9NrEGkoqgnM@mail.gmail.com>
	<AANLkTinaixoUDTyR-5gPZPLSaO_W0eNvC491ApAUHg-O@mail.gmail.com>
Message-ID: <AANLkTimU-=7LJXzRy1xTAf+Vz8B6nexLBORJzi_EO5a9@mail.gmail.com>

On Fri, Feb 4, 2011 at 10:19 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> Ok, 18x will be perfect!
>
> The problem is that when the inbound call is SIP and the originated is
> ISDN, the originate doesn't return when a 180 ringing is received (that in
> ISDN is an ALERT without in band info).
>
> The originate returns when a 183 (ISDN = in band info available) or 200
> (ISDN = CONNECT) is received.
>
> Stephen
>
>
Could you tell us a bit more about what you're trying to accomplish? My
guess is that there's a better way to do it. In most cases it is not
necessary to create a new session inside of a script called from the
dialplan.

What is the application that you are working on? Who calls whom, etc.
-MC
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From msc at freeswitch.org  Fri Feb  4 21:29:58 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 4 Feb 2011 10:29:58 -0800
Subject: [Freeswitch-users] curl: parse XML response?
In-Reply-To: <AANLkTik3BoGujLx9QBvJg3SWNByarca822ryxBFzaWoK@mail.gmail.com>
References: <AANLkTik3BoGujLx9QBvJg3SWNByarca822ryxBFzaWoK@mail.gmail.com>
Message-ID: <AANLkTinXDksFq9W35SaUU1eLVCtb+WQL-uM-bRbGE=hM@mail.gmail.com>

Where are you doing the curl call? Just curious if this is happening inside
the dialplan or a script or what. FYI, I covered this in chapter 7 of the
FreeSWITCH book on how to use mod_curl from a Lua script and parse the
results. If you're just doing it from the dialplan then you probably need a
script, although you can try using the regex API. See
http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan for tips on
how to call an API from within the dialplan.

-MC

On Thu, Feb 3, 2011 at 6:27 PM, james.royer at gmail.com <james.royer at gmail.com
> wrote:

> Hi All,
>
> I want to use mod_curl to query a web service and then extract information
> from the XML response.
>
> <response>
>   <value1>test</value1>
>   <value2>blah</value2>
> </response>
>
> I then want to use TTS to speak the values to the caller.
>
> How can I do this?
>
> I tried searches for "freeswitch parse xml" and other variations but all
> just talk about how FreeSWITCH parses config files.
>
> Thanks,
> James
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  4 21:31:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 4 Feb 2011 10:31:02 -0800
Subject: [Freeswitch-users] Seems att_xfer doesn't work at all?
In-Reply-To: <AANLkTi=Nnwd2EE6tkt6dUOdn+sn_nn0y8oFcz=Q4q=i7@mail.gmail.com>
References: <AANLkTi=Nnwd2EE6tkt6dUOdn+sn_nn0y8oFcz=Q4q=i7@mail.gmail.com>
Message-ID: <AANLkTimksnUMs_CUUytJD5Jak1-UOdgU8ENvMYUwyGR9@mail.gmail.com>

Can you get debug log with SIP trace? put it on pastebin and the gang will
take a look.
-MC

On Fri, Feb 4, 2011 at 6:05 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:

> Hi all!
>
> In latest git i'm unable to use att_xfer at all.
>
> Channels don't get bridged on transferor hangup, no moh to transferee,
> and transferee is locked in no moh state with CS_RESET after
> transferor and transfer target is already hangup
>
> Does anybody have working example of using att_xfer with bind_meta_app?
>
>
> --
> 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 msc at freeswitch.org  Fri Feb  4 21:34:49 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 4 Feb 2011 10:34:49 -0800
Subject: [Freeswitch-users] mod_directory multi-tenant
In-Reply-To: <000901cbc449$ea673480$bf359d80$@net>
References: <000901cbc449$ea673480$bf359d80$@net>
Message-ID: <AANLkTinV80XmYE-PXuxLvnMoCpRS9q0c9G=E_3O=owxh@mail.gmail.com>

Pastebin some samples of what you've been trying. We have a few guys who've
done a lot of mod_xml_curl stuff and at least a few of them are experienced
with multi-tenancy.
-MC

On Fri, Feb 4, 2011 at 1:00 AM, Erick Baum <erick at teal.net> wrote:

> Is there a trick to getting the mod_directory application working in a
> multi-tenant configuration?  I think I?ve tried every possible combination
> of settings and I cannot for the life of me get it to work.  I can get the
> directory to answer but no matter what I put in to the search, it comes back
> with no results.  Any help would be appreciated!
>
>
>
> Erick
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jerre at j-cope.com  Fri Feb  4 21:10:47 2011
From: jerre at j-cope.com (Jerre Cope)
Date: Fri, 04 Feb 2011 12:10:47 -0600
Subject: [Freeswitch-users] Auto test FS install - now necessary?
In-Reply-To: <mailman.3138.1296841769.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.3138.1296841769.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <4D4C4127.5020000@j-cope.com>

I second Mr. Ayre.  Thanks everyone.  Especially the Rocket Scientists.


On 02/04/2011 11:49 AM, 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: Auto test FS install - now necessary? (Steven Ayre)
>     2. Re: Auto test FS install - now necessary? (Steven Ayre)
>     3. Re: Auto test FS install - now necessary? (Phone)
>     4. Re: Auto test FS install - now necessary? (mazilo)
>     5. curl: parse XML response? (james.royer at gmail.com)
>     6. mod_directory multi-tenant (Erick Baum)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  4 21:38:17 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 4 Feb 2011 10:38:17 -0800
Subject: [Freeswitch-users] mod_curl: how to parse xml response?
In-Reply-To: <AANLkTinbbRufHM7nPkgtW=HXyos18KdhnRgmitVLS7NN@mail.gmail.com>
References: <AANLkTinbbRufHM7nPkgtW=HXyos18KdhnRgmitVLS7NN@mail.gmail.com>
Message-ID: <AANLkTi=OZhVB33p3FyxrvO0Oi7gVkVMzvXT4i4W9TjMa@mail.gmail.com>

If you sent your first message before you subscribed to the mailing list
then you were most likely moderated. In that case there can be a delay,
especially if our moderators were asleep when your first email went through.
:)

That being said, I responded to your first email so go check that thread. :)

-MC

On Fri, Feb 4, 2011 at 9:37 AM, james.royer at gmail.com <james.royer at gmail.com
> wrote:

> Hi All,
>
> (This is my second post - are posts delayed? I never received an
> acknowledgment message.)
>
> I want to use mod_curl to query a web service and then extract information
> from the XML response.
>
> <response>
>   <value1>test</value1>
>   <value2>blah</value2>
> </response>
>
> I then want to use TTS to speak the values to the caller.
>
> How can I do this?
>
> I tried searches for "freeswitch parse xml" and other variations but all
> just talk about how FreeSWITCH parses config files.
>
> I see that people have created libraries for Lua proper but looking at the
> doc, there doesn't appear to be any functions to parse XML from within Lua.
>
> Thanks,
> James
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  4 22:18:27 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 4 Feb 2011 19:18:27 +0000
Subject: [Freeswitch-users] Auto test FS install - now necessary?
In-Reply-To: <AANLkTin7X195XmHYPcSKvqyWVXVF7dV2R7T9Touco716@mail.gmail.com>
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>
	<AANLkTi=Rofy64NwXTt3u1hRRMuAEhimKKux-_zAhTkiS@mail.gmail.com>
	<1105C189976B4970A1F282410B827A7D@e1705>
	<AANLkTik2YTEuXZPXqyMpuuibcxVQo9d1imLGL8+EzzAT@mail.gmail.com>
	<AANLkTin7X195XmHYPcSKvqyWVXVF7dV2R7T9Touco716@mail.gmail.com>
Message-ID: <AANLkTimEcqS2wOx9_yPcCX2z85NOLLm+KC5pEMbrq38K@mail.gmail.com>

And be stuck with a version that has known bugs?

If they're minor bugs that don't effect you why not. Occasionally there may
be one that's exploitable though and then you'd want to update. That's when
the stable branch Anthm said is being planned could be useful.

-Steve



On 4 February 2011 17:52, guru singh <grsingh750 at gmail.com> wrote:

> On Fri, Feb 4, 2011 at 9:15 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>
> > Testing yourself is always a good idea before rolling out to a production
> > system. That'll show any obvious ones up that can be fixed quickly.
>
> On a related note, After thorough testing, if a system is put in
> production. Would it be 'best practice' not to update it, if no config
> changes or new functionality is ever desired?
>
> > -Steve
>
> 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
>
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From steveayre at gmail.com  Fri Feb  4 22:20:07 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 4 Feb 2011 19:20:07 +0000
Subject: [Freeswitch-users] H.323 implementation on FS
In-Reply-To: <AANLkTin_cDiA9PWcrhO6qSYwzamnxUAJZcOEZy2HMrkC@mail.gmail.com>
References: <AANLkTin_cDiA9PWcrhO6qSYwzamnxUAJZcOEZy2HMrkC@mail.gmail.com>
Message-ID: <AANLkTi=kvGW5ugK41x9gRS6_hGoYNXmZQpBNDjkOBJSA@mail.gmail.com>

I had problems with mod_opal but mod_h323 works for me.

-Steve



On 4 February 2011 12:55, Vedran Zeljeznak <vedran.zeljeznak at gmail.com>wrote:

> hi everyone,
>
> do any of you have any suggestions which H.323 implementation
> (mod_opal or mod_h323) should be used on a production Freeswitch
> platform (HEAD version)?
>
> Is it better to forward calls from H.323 trunk through Yate
> (configured as H.323 to SIP proxy) before terminating them on
> Freeswitch?
>
> ---
> Vedran Zeljeznak
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  4 22:20:54 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 4 Feb 2011 19:20:54 +0000
Subject: [Freeswitch-users] H.323 implementation on FS
In-Reply-To: <AANLkTi=kvGW5ugK41x9gRS6_hGoYNXmZQpBNDjkOBJSA@mail.gmail.com>
References: <AANLkTin_cDiA9PWcrhO6qSYwzamnxUAJZcOEZy2HMrkC@mail.gmail.com>
	<AANLkTi=kvGW5ugK41x9gRS6_hGoYNXmZQpBNDjkOBJSA@mail.gmail.com>
Message-ID: <AANLkTikwcXZxLZ6wKH-7rubL0eW0W7Q4MwNf5R6a+hhZ@mail.gmail.com>

Although there's almost no H323 traffic - we're almost entirely SIP - so I
can't say how stable it is for a large load.

-Steve


On 4 February 2011 19:20, Steven Ayre <steveayre at gmail.com> wrote:

> I had problems with mod_opal but mod_h323 works for me.
>
> -Steve
>
>
>
>
> On 4 February 2011 12:55, Vedran Zeljeznak <vedran.zeljeznak at gmail.com>wrote:
>
>> hi everyone,
>>
>> do any of you have any suggestions which H.323 implementation
>> (mod_opal or mod_h323) should be used on a production Freeswitch
>> platform (HEAD version)?
>>
>> Is it better to forward calls from H.323 trunk through Yate
>> (configured as H.323 to SIP proxy) before terminating them on
>> Freeswitch?
>>
>> ---
>> Vedran Zeljeznak
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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  Fri Feb  4 22:27:24 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 4 Feb 2011 11:27:24 -0800 (PST)
Subject: [Freeswitch-users] Auto test FS install - now necessary?
In-Reply-To: <7D4B9BFA-023D-4426-93C2-F6BDE2CCBF9C@ipeva.fr>
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>
	<AANLkTikSryxPi1-YZ8Wpt0GUbcXf_i=4DMi2BmrVPWix@mail.gmail.com>
	<AANLkTinQArNvg6jZky6=8JUVYif9kQge6bkjnbqNncLL@mail.gmail.com>
	<4D4C32C7.9050503@aleph-com.net>
	<AANLkTinz-XCQHVympCqjzBL5=pfCOXiSPqNqbho7UNmX@mail.gmail.com>
	<1296840880862-5993296.post@n2.nabble.com>
	<7D4B9BFA-023D-4426-93C2-F6BDE2CCBF9C@ipeva.fr>
Message-ID: <1296847644406-5993736.post@n2.nabble.com>



David Ponzone wrote:
> 
> So it's a whitelist, and that is quite easy to do with FreeSWITCH, with
> some LUA, and a SQL DB.
If anyone out here is willing to provide a howto, especially without an SQL
DB, that will be cool. I did this on Asterisk using the built-in database
command with the astdb file.

-----
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/Auto-test-FS-install-now-necessary-tp5991624p5993736.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From rajesh.npnr at yahoo.com  Fri Feb  4 22:28:10 2011
From: rajesh.npnr at yahoo.com (rex.alex)
Date: Fri, 4 Feb 2011 11:28:10 -0800 (PST)
Subject: [Freeswitch-users] Grandstream Freeze on Originate command
In-Reply-To: <AANLkTim+3c7Ui=bPd6_7ROXEA+mxgX3RRzCf03PWdTmi@mail.gmail.com>
References: <1296751186605-5989433.post@n2.nabble.com>
	<AANLkTikPiZgq9WsFeMs2U0iZrT7FcL10dptLCFu5HynP@mail.gmail.com>
	<1296818499616-5992217.post@n2.nabble.com>
	<1296821764473-5992350.post@n2.nabble.com>
	<AANLkTim+3c7Ui=bPd6_7ROXEA+mxgX3RRzCf03PWdTmi@mail.gmail.com>
Message-ID: <1296847690473-5993740.post@n2.nabble.com>


May be I can pastebin the sip trace of GS with freeswitch 1.0.6 which did
work on originate command without any problem

Thanks,
Rex
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Grandstream-Freeze-on-Originate-command-tp5989433p5993740.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From wstephen80 at gmail.com  Fri Feb  4 22:30:14 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Fri, 4 Feb 2011 20:30:14 +0100
Subject: [Freeswitch-users] Lua Session creation is blocking?
In-Reply-To: <AANLkTimU-=7LJXzRy1xTAf+Vz8B6nexLBORJzi_EO5a9@mail.gmail.com>
References: <AANLkTin21o-gZH6FrPU6BxFR+RqxBkN15=DDGD66iTX-@mail.gmail.com>
	<AANLkTikdGqM2QsZ19_4YpkDRTd+b5_wRK9NrEGkoqgnM@mail.gmail.com>
	<AANLkTinaixoUDTyR-5gPZPLSaO_W0eNvC491ApAUHg-O@mail.gmail.com>
	<AANLkTimU-=7LJXzRy1xTAf+Vz8B6nexLBORJzi_EO5a9@mail.gmail.com>
Message-ID: <AANLkTikPNom1k3f=fbixbtcCGw=WGzC9UZj25mBBnvjj@mail.gmail.com>

I have to port some applications on Freeswitch so I'm doing some preliminary
learning tests on Lua scripting.

The result that I would obtain is to have the full control of both inbound
and originated session and to have under control their state.

The script I'm running is simply: http://pastebin.freeswitch.org/15248

But its behaviour is not as I expect.

Stephen


On Fri, Feb 4, 2011 at 7:26 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Feb 4, 2011 at 10:19 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> Ok, 18x will be perfect!
>>
>> The problem is that when the inbound call is SIP and the originated is
>> ISDN, the originate doesn't return when a 180 ringing is received (that in
>> ISDN is an ALERT without in band info).
>>
>> The originate returns when a 183 (ISDN = in band info available) or 200
>> (ISDN = CONNECT) is received.
>>
>> Stephen
>>
>>
> Could you tell us a bit more about what you're trying to accomplish? My
> guess is that there's a better way to do it. In most cases it is not
> necessary to create a new session inside of a script called from the
> dialplan.
>
> What is the application that you are working on? Who calls whom, etc.
> -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 infos at madovsky.org  Fri Feb  4 22:38:19 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 4 Feb 2011 14:38:19 -0500
Subject: [Freeswitch-users] Auto test FS install - now necessary?
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com><AANLkTikSryxPi1-YZ8Wpt0GUbcXf_i=4DMi2BmrVPWix@mail.gmail.com><AANLkTinQArNvg6jZky6=8JUVYif9kQge6bkjnbqNncLL@mail.gmail.com><4D4C32C7.9050503@aleph-com.net><AANLkTinz-XCQHVympCqjzBL5=pfCOXiSPqNqbho7UNmX@mail.gmail.com><1296840880862-5993296.post@n2.nabble.com><7D4B9BFA-023D-4426-93C2-F6BDE2CCBF9C@ipeva.fr>
	<1296847644406-5993736.post@n2.nabble.com>
Message-ID: <91EEF7A314AE4996A055EF2D4F622F4A@e1705>

why not use a file like hosts.deny
from a perl/php/other script called with system ?

----- Original Message ----- 
From: "mazilo" <Nabble at slickdeals.endjunk.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Friday, February 04, 2011 2:27 PM
Subject: Re: [Freeswitch-users] Auto test FS install - now necessary?


>
>
> David Ponzone wrote:
>>
>> So it's a whitelist, and that is quite easy to do with FreeSWITCH, with
>> some LUA, and a SQL DB.
> If anyone out here is willing to provide a howto, especially without an 
> SQL
> DB, that will be cool. I did this on Asterisk using the built-in database
> command with the astdb file.
>
> -----
> 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/Auto-test-FS-install-now-necessary-tp5991624p5993736.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 grsingh750 at gmail.com  Fri Feb  4 23:04:46 2011
From: grsingh750 at gmail.com (guru singh)
Date: Sat, 5 Feb 2011 01:34:46 +0530
Subject: [Freeswitch-users] Auto test FS install - now necessary?
In-Reply-To: <AANLkTi=8ahEit6vG_mzvjb_EbFC37Tm_DNK-qbNe8aLB@mail.gmail.com>
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>
	<AANLkTi=Rofy64NwXTt3u1hRRMuAEhimKKux-_zAhTkiS@mail.gmail.com>
	<1105C189976B4970A1F282410B827A7D@e1705>
	<AANLkTik2YTEuXZPXqyMpuuibcxVQo9d1imLGL8+EzzAT@mail.gmail.com>
	<AANLkTin7X195XmHYPcSKvqyWVXVF7dV2R7T9Touco716@mail.gmail.com>
	<AANLkTi=8ahEit6vG_mzvjb_EbFC37Tm_DNK-qbNe8aLB@mail.gmail.com>
Message-ID: <AANLkTimx7pANjn1ASqigSS=fswefh=k=2-xCuz0jBWiY@mail.gmail.com>

On Fri, Feb 4, 2011 at 11:52 PM, Michael Collins <msc at freeswitch.org> wrote:

>
> This depends greatly on your scenario. Personally, I think that anyone who
> puts FS into production (i.e. in a place where it helps one make money)
> should *ALWAYS* have a backup/sandbox/test system for handling updates. If
> you are in a mission-critical scenario then it probably means that once you
> get it working you don't touch it unless you absolutely have to. That's
> where your test system comes in to play. Ideally you have the exact same
> hardware on your test system that you do in production. Then your test
> system can double as a warm standby.

Thanks for the advice, and also the awesome work you guys are doing. :)
I, for one am more than happy with the 'latest is the best' philosophy.

> Just my $.02, but I think that commodity hardware being so cheap these days
> that there's no excuse for a for-profit venture using FS (or any other
> software, for that matter) not to have a test system for mission-critical
> servers.

Hmmm, someday!

-guru


From larclap at yahoo.com  Fri Feb  4 23:11:11 2011
From: larclap at yahoo.com (Lars Zeb)
Date: Fri, 4 Feb 2011 12:11:11 -0800
Subject: [Freeswitch-users] Group confusions
Message-ID: <00f501cbc4a7$aaee5640$00cb02c0$@yahoo.com>

I went to two individual extensions and registered them for group 01 by
dialing 8101. I then tested that group by dialing 8201 - both extensions
rang.

I then went to the cli and entered the command:

group delete:01:1013 at 192.168.10.29

where 1013 is one of the two extensions I had joined into group 01 earlier
and the url is the domain and fs address. However, when I dialed 8201 after
entering this cli command, both extensions rang. I had expected that only
one would ring.

I am confused. Are these two different types of "groups"?

Thanks, Lars



From list.subscription at alexrambau.com  Fri Feb  4 23:17:32 2011
From: list.subscription at alexrambau.com (Alex Rambau)
Date: Fri, 4 Feb 2011 13:17:32 -0700
Subject: [Freeswitch-users] Grandstream Freeze on Originate command
In-Reply-To: <AANLkTim+3c7Ui=bPd6_7ROXEA+mxgX3RRzCf03PWdTmi@mail.gmail.com>
References: <1296751186605-5989433.post@n2.nabble.com>	<AANLkTikPiZgq9WsFeMs2U0iZrT7FcL10dptLCFu5HynP@mail.gmail.com>	<1296818499616-5992217.post@n2.nabble.com>	<1296821764473-5992350.post@n2.nabble.com>
	<AANLkTim+3c7Ui=bPd6_7ROXEA+mxgX3RRzCf03PWdTmi@mail.gmail.com>
Message-ID: <01af01cbc4a8$8e4fdee0$aaef9ca0$@subscription@alexrambau.com>

This happened to me until I added {absolute_codec_string='PCMU,PCMA'} on the
dial string.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Friday, February 04, 2011 11:13 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Grandstream Freeze on Originate command

 

Do you have an example of an invite to the GS, perhaps from Asterisk or
another SIP server, that does not freeze the phone? My guess is that GS is
doing something silly. (They're known for doing silly things and freezing up
when you send them an invite is, unfortunately, not at all surprising given
the reputation they've acquired for themselves.)

 

Wish I could help more...

 

-MC

 

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From tculjaga at gmail.com  Fri Feb  4 23:18:04 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 4 Feb 2011 21:18:04 +0100
Subject: [Freeswitch-users] H.323 implementation on FS
In-Reply-To: <AANLkTin_cDiA9PWcrhO6qSYwzamnxUAJZcOEZy2HMrkC@mail.gmail.com>
References: <AANLkTin_cDiA9PWcrhO6qSYwzamnxUAJZcOEZy2HMrkC@mail.gmail.com>
Message-ID: <AANLkTin8TQ6d+16BZ0K22ax4kdd-V6dtKM5fFULWUZ0c@mail.gmail.com>

Vedran,

mod_h323 does work fro you...

mod_opal had issues with SIP - H323 interworking

T.

On Fri, Feb 4, 2011 at 1:55 PM, Vedran Zeljeznak <vedran.zeljeznak at gmail.com
> wrote:

> hi everyone,
>
> do any of you have any suggestions which H.323 implementation
> (mod_opal or mod_h323) should be used on a production Freeswitch
> platform (HEAD version)?
>
> Is it better to forward calls from H.323 trunk through Yate
> (configured as H.323 to SIP proxy) before terminating them on
> Freeswitch?
>
> ---
> Vedran Zeljeznak
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Feb  4 23:23:00 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 4 Feb 2011 12:23:00 -0800 (PST)
Subject: [Freeswitch-users] Auto test FS install - now necessary?
In-Reply-To: <91EEF7A314AE4996A055EF2D4F622F4A@e1705>
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>
	<AANLkTikSryxPi1-YZ8Wpt0GUbcXf_i=4DMi2BmrVPWix@mail.gmail.com>
	<AANLkTinQArNvg6jZky6=8JUVYif9kQge6bkjnbqNncLL@mail.gmail.com>
	<4D4C32C7.9050503@aleph-com.net>
	<AANLkTinz-XCQHVympCqjzBL5=pfCOXiSPqNqbho7UNmX@mail.gmail.com>
	<1296840880862-5993296.post@n2.nabble.com>
	<7D4B9BFA-023D-4426-93C2-F6BDE2CCBF9C@ipeva.fr>
	<1296847644406-5993736.post@n2.nabble.com>
	<91EEF7A314AE4996A055EF2D4F622F4A@e1705>
Message-ID: <1296850980061-5993907.post@n2.nabble.com>



Madovsky wrote:
> why not use a file like hosts.deny
> from a perl/php/other script called with system ?
IIRC, the host.deny only filters incoming calls from specific hosts. For
instance, if you want to block crackers out there who try to exploit your FS
system, perhaps this is not a bad idea, but not if you only want to allow
some specific incoming CIDs to be able to reach your internal extensions.

-----
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/Auto-test-FS-install-now-necessary-tp5991624p5993907.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Fri Feb  4 23:26:35 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 4 Feb 2011 20:26:35 +0000
Subject: [Freeswitch-users] Lua Session creation is blocking?
In-Reply-To: <AANLkTikPNom1k3f=fbixbtcCGw=WGzC9UZj25mBBnvjj@mail.gmail.com>
References: <AANLkTin21o-gZH6FrPU6BxFR+RqxBkN15=DDGD66iTX-@mail.gmail.com>
	<AANLkTikdGqM2QsZ19_4YpkDRTd+b5_wRK9NrEGkoqgnM@mail.gmail.com>
	<AANLkTinaixoUDTyR-5gPZPLSaO_W0eNvC491ApAUHg-O@mail.gmail.com>
	<AANLkTimU-=7LJXzRy1xTAf+Vz8B6nexLBORJzi_EO5a9@mail.gmail.com>
	<AANLkTikPNom1k3f=fbixbtcCGw=WGzC9UZj25mBBnvjj@mail.gmail.com>
Message-ID: <AANLkTin-E1cA9oBq_pFO93QMff-SStNHuXcr2s339d2y@mail.gmail.com>

You could also look at ESL.

-Steve



On 4 February 2011 19:30, Stephen Wilde <wstephen80 at gmail.com> wrote:

> I have to port some applications on Freeswitch so I'm doing some
> preliminary learning tests on Lua scripting.
>
> The result that I would obtain is to have the full control of both inbound
> and originated session and to have under control their state.
>
> The script I'm running is simply: http://pastebin.freeswitch.org/15248
>
> But its behaviour is not as I expect.
>
> Stephen
>
>
> On Fri, Feb 4, 2011 at 7:26 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Fri, Feb 4, 2011 at 10:19 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>
>>> Ok, 18x will be perfect!
>>>
>>> The problem is that when the inbound call is SIP and the originated is
>>> ISDN, the originate doesn't return when a 180 ringing is received (that in
>>> ISDN is an ALERT without in band info).
>>>
>>> The originate returns when a 183 (ISDN = in band info available) or 200
>>> (ISDN = CONNECT) is received.
>>>
>>> Stephen
>>>
>>>
>> Could you tell us a bit more about what you're trying to accomplish? My
>> guess is that there's a better way to do it. In most cases it is not
>> necessary to create a new session inside of a script called from the
>> dialplan.
>>
>> What is the application that you are working on? Who calls whom, etc.
>> -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 chat2jesse at gmail.com  Fri Feb  4 23:54:45 2011
From: chat2jesse at gmail.com (jesse)
Date: Fri, 4 Feb 2011 12:54:45 -0800
Subject: [Freeswitch-users] Auto test FS install - now necessary?
In-Reply-To: <4D4C32C7.9050503@aleph-com.net>
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>
	<AANLkTikSryxPi1-YZ8Wpt0GUbcXf_i=4DMi2BmrVPWix@mail.gmail.com>
	<AANLkTinQArNvg6jZky6=8JUVYif9kQge6bkjnbqNncLL@mail.gmail.com>
	<4D4C32C7.9050503@aleph-com.net>
Message-ID: <AANLkTinjCZQSAiA5tdWtpkVTN=wHTcXVqap_aUOr7HyX@mail.gmail.com>

I think you misunderstand me.  I meant asterisk is a good example of bad
code quality system.
On Feb 4, 2011 9:12 AM, "Darren Wiebe" <darren at aleph-com.net> wrote:
> On 04/02/2011 10:01 AM, Anthony Minessale wrote:
>> Reliability is more important than feature. Asterisk is a good example of
>> > such case.
> I snipped a very small part of this thread but the above line provided
> my comedy relief for the day. Thanks, I needed it.
>
> --
> Darren Wiebe
> Aleph Communications
> --------------------
> Phone: 1-877-702-2900
> Fax: 1-866-274-4506
> Email: darren at aleph-com.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
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From chat2jesse at gmail.com  Sat Feb  5 00:02:33 2011
From: chat2jesse at gmail.com (jesse)
Date: Fri, 4 Feb 2011 13:02:33 -0800
Subject: [Freeswitch-users] Auto test FS install - now necessary?
In-Reply-To: <4D4C3807.2010600@gmail.com>
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>
	<AANLkTikSryxPi1-YZ8Wpt0GUbcXf_i=4DMi2BmrVPWix@mail.gmail.com>
	<AANLkTinQArNvg6jZky6=8JUVYif9kQge6bkjnbqNncLL@mail.gmail.com>
	<4D4C3807.2010600@gmail.com>
Message-ID: <AANLkTik=pAVJ+C+LygavDFw0AnPDFog0BbMqyQ9FrwhM@mail.gmail.com>

In my personal experience, I also switched from asterisk to freeswitch. From
engineering perspective, a good process will befits the project in the long
run, especially as it becomes more complicated and involves more people.
On Feb 4, 2011 9:34 AM, "Phone" <phone.bytes at gmail.com> wrote:
> I hope that we don't ever return to the old days of spending a year or
> more regression testing PBX firmware between each release. That was
> okay, if you did not mind having a product that worked really well but
> did not really do much and was always about a year behind the cutting
edge.
>
> I applaud all of the FS Devs, I believe that everyone is doing their
> best to avoid introducing new issues as work is done to improve the code
> base and add new features and functionality.
>
> We are happy to participate in the testing and feedback process.
> We hope to be able to make meaningful contributions down the road when
> we are more familiar with things.
>
> I have never seen a major issue that was not corrected with lightning
speed.
> It seems that even whenever someone reports an issue that is important
> to them, they are likewise addressed very rapidly.
>
> In my opinion, this is just the way that software development goes. Do
> your best to modify (improve the project) code without causing other
> issues. Unfortunately, sometimes this happens. Especially when there
> so many people working to make contributions to grow and improve the
> product for all. Frankly, I think the project is progressing very well.
>
>
>
>
>
> On 02/04/2011 10:01 AM, Anthony Minessale wrote:
>> I guess we should try to be more like all the other open source
>> projects where every single day of GIT HEAD is perfect.
>> hmm....
>>
>> The alternative is you can buy a soft-switch and live with it's faults
>> and pay annually for an update.
>> Or even wait for our official releases which are very infrequent and
>> not well-supported once they are more than a few months old.
>>
>> Also don't forget we use FreeSWITCH in a real product so you don't
>> even see the amount of testing and QA we inherit from that product as
>> well as all the large carriers who use FreeSWITCH and do their best to
>> try beta version for us. Thank you all.....
>>
>> We have plans to stable branch soon. I guess that means everyone will
>> flock to that and it will be the end of my beta testers.
>> We do not only get regressions from features we get them sometimes
>> from fixes as well, other peoples patches, many factors.
>>
>> I guess what's really saddening since you bring up asterisk, is that I
>> spent years using it with perpetual problems in the release or the
>> daily snapshot and it sort of has a reputation for this kind of thing
>> and all you hear about is how awesome it is from people. I don't
>> appreciate being treated like a witch hunt over one regression
>> especially when we can fix it in 1 minute. That double-standard is
>> ridiculous.
>>
>> I should just make a FAQ about this and paste in the URL:
>>
>> We only have so much time to give to the world, we spend like 8-14
>> hours a day on average working on the code.
>> Any suggestions or improvements (even the ones with the best
>> intentions) are only viable if they come with volunteers.
>>
>> Bottom line is by downloading and building the code you are opting in
>> to our community which involves occasional speed bumps.
>>
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  5 00:57:34 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 4 Feb 2011 15:57:34 -0600
Subject: [Freeswitch-users] Lua Session creation is blocking?
In-Reply-To: <AANLkTikPNom1k3f=fbixbtcCGw=WGzC9UZj25mBBnvjj@mail.gmail.com>
References: <AANLkTin21o-gZH6FrPU6BxFR+RqxBkN15=DDGD66iTX-@mail.gmail.com>
	<AANLkTikdGqM2QsZ19_4YpkDRTd+b5_wRK9NrEGkoqgnM@mail.gmail.com>
	<AANLkTinaixoUDTyR-5gPZPLSaO_W0eNvC491ApAUHg-O@mail.gmail.com>
	<AANLkTimU-=7LJXzRy1xTAf+Vz8B6nexLBORJzi_EO5a9@mail.gmail.com>
	<AANLkTikPNom1k3f=fbixbtcCGw=WGzC9UZj25mBBnvjj@mail.gmail.com>
Message-ID: <AANLkTin=J4TqxY4D5OfriK6iD=1-OfLg32YXyyXLL61d@mail.gmail.com>

The way you expect it to work is not the case.

in embedded scripts when you create a new session you must wait for it
to have media.
The scripts are not asynchronous.  if you want asynchronous you 100% want ESL.






On Fri, Feb 4, 2011 at 1:30 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:
> I have to port some applications on Freeswitch so I'm doing some preliminary
> learning tests on Lua scripting.
> The result that I would obtain is to have the full control of both inbound
> and originated session and to have under control their state.
> The script I'm running is simply:?http://pastebin.freeswitch.org/15248
> But its behaviour is not as I expect.
>
> Stephen
>
> On Fri, Feb 4, 2011 at 7:26 PM, Michael Collins <msc at freeswitch.org> wrote:
>>
>>
>> On Fri, Feb 4, 2011 at 10:19 AM, Stephen Wilde <wstephen80 at gmail.com>
>> wrote:
>>>
>>> Ok, 18x will be perfect!
>>> The problem is that when the inbound call is SIP and the originated is
>>> ISDN, the originate doesn't return when a 180 ringing is received (that in
>>> ISDN is an ALERT without in band info).
>>> The originate returns when a 183 (ISDN = in band info available) or 200
>>> (ISDN = CONNECT) is received.
>>> Stephen
>>>
>>
>> Could you tell us a bit more about what you're trying to accomplish? My
>> guess is that there's a better way to do it. In most cases it is not
>> necessary to create a new session inside of a script called from the
>> dialplan.
>> What is the application that you are working on? Who calls whom, etc.
>> -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
>
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 chat2jesse at gmail.com  Sat Feb  5 01:04:15 2011
From: chat2jesse at gmail.com (jesse)
Date: Fri, 4 Feb 2011 14:04:15 -0800
Subject: [Freeswitch-users] Auto test FS install - now necessary?
In-Reply-To: <AANLkTinz-XCQHVympCqjzBL5=pfCOXiSPqNqbho7UNmX@mail.gmail.com>
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>
	<AANLkTikSryxPi1-YZ8Wpt0GUbcXf_i=4DMi2BmrVPWix@mail.gmail.com>
	<AANLkTinQArNvg6jZky6=8JUVYif9kQge6bkjnbqNncLL@mail.gmail.com>
	<4D4C32C7.9050503@aleph-com.net>
	<AANLkTinz-XCQHVympCqjzBL5=pfCOXiSPqNqbho7UNmX@mail.gmail.com>
Message-ID: <AANLkTik7KvMGZh3yz9c894ShK1-7T-WKyiVcsDcMc1mt@mail.gmail.com>

If you misunderstood my point and ruined your keyboard, your own fault!  My
friend.
On Feb 4, 2011 9:21 AM, "Steven Ayre" <steveayre at gmail.com> wrote:
> I actually spilt my tea. :)
>
>
> On 4 February 2011 17:09, Darren Wiebe <darren at aleph-com.net> wrote:
>
>> On 04/02/2011 10:01 AM, Anthony Minessale wrote:
>> > Reliability is more important than feature. Asterisk is a good example
of
>> > > such case.
>> I snipped a very small part of this thread but the above line provided
>> my comedy relief for the day. Thanks, I needed it.
>>
>> --
>> Darren Wiebe
>> Aleph Communications
>> --------------------
>> Phone: 1-877-702-2900
>> Fax: 1-866-274-4506
>> Email: darren at aleph-com.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
>>
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From fs-list at communicatefreely.net  Sat Feb  5 01:09:08 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 04 Feb 2011 17:09:08 -0500
Subject: [Freeswitch-users] Sent aastra-xml event via esl?
Message-ID: <4D4C7904.2030503@communicatefreely.net>

Hello,

I'm trying to use ESL to send an event to aastra phones, so I can get 
them to execute an XML script and do other fancy things.

To start, I'm just testing using a simple telnet connection to make sure 
it will work.

I can successfully connect, and do things like "api help", etc.

Here's what I have been sending:

sendevent NOTIFY
profile: internal
event-string: aastra-xml
user: 5101 at communicatefreely.net
host: stefan.151front.communicatefreely.net
content-type: application/simple-message-summary

I get this response:

Content-Type: command/reply
Reply-Text: +OK

I don't see anything at all go out to the phone.

Here's the phone I want to get the event:

sofia status profile internal user 5101 at communicatefreely.net

Registrations:
=================================================================================================
Call-ID:        996859d51795564d
User:           5101 at communicatefreely.net
Contact:        "Tim St. Pierre"

<contact obscured>

Agent:          Aastra 6731i/2.5.2.30
Status:         Registered(UDP-NAT)(unknown) EXP(2011-02-04 17:21:56) 
EXPSECS(1167)
Host:           stefan.151front.communicatefreely.net
IP:           
User:      5101
Auth-Realm:     pbx.communicatefreely.net
MWI-Account:    5101 at eccentricartists.communicatefreely.net

Total items returned: 1
=================================================================================================

What is the syntax of the user: and host: headers?  Is the host: the 
switch I want to respond to the event, or the IP address of the phone?  
I need FS to send the event to the phone based on it's registration.

Thanks!

-Tim



From gustavo.espeche at upper-soft.com  Sat Feb  5 01:09:06 2011
From: gustavo.espeche at upper-soft.com (Gustavo Espeche)
Date: Fri, 04 Feb 2011 19:09:06 -0300
Subject: [Freeswitch-users] H.323 implementation on FS
In-Reply-To: <AANLkTikwcXZxLZ6wKH-7rubL0eW0W7Q4MwNf5R6a+hhZ@mail.gmail.com>
References: <AANLkTin_cDiA9PWcrhO6qSYwzamnxUAJZcOEZy2HMrkC@mail.gmail.com>
	<AANLkTi=kvGW5ugK41x9gRS6_hGoYNXmZQpBNDjkOBJSA@mail.gmail.com>
	<AANLkTikwcXZxLZ6wKH-7rubL0eW0W7Q4MwNf5R6a+hhZ@mail.gmail.com>
Message-ID: <1296857346.2599.26.camel@gustavo-laptop>

Use Yate we are try to use mod_opal or mod_h323 in production with-out
success, Yate has lot of issue but is a good inter-worker between h323
and sip.
Freeswitch has a lost of good think but the inter-worker isn't it.
Because of then we use Freeswitch for billing propose but Yate for
inter-worker.
But if you need use FS for inter-worker use mod_h323 is a little more
stable.

Best Regards.

Gustavo Espeche
www.easyipcall.com



On Fri, 2011-02-04 at 19:20 +0000, Steven Ayre wrote:
> Although there's almost no H323 traffic - we're almost entirely SIP -
> so I can't say how stable it is for a large load.
> 
> -Steve
> 
> 
> On 4 February 2011 19:20, Steven Ayre <steveayre at gmail.com> wrote:
>         I had problems with mod_opal but mod_h323 works for me.
>         
>         -Steve
>         
>         
>         
>         
>         
>         On 4 February 2011 12:55, Vedran Zeljeznak
>         <vedran.zeljeznak at gmail.com> wrote:
>                 hi everyone,
>                 
>                 do any of you have any suggestions which H.323
>                 implementation
>                 (mod_opal or mod_h323) should be used on a production
>                 Freeswitch
>                 platform (HEAD version)?
>                 
>                 Is it better to forward calls from H.323 trunk through
>                 Yate
>                 (configured as H.323 to SIP proxy) before terminating
>                 them on
>                 Freeswitch?
>                 
>                 ---
>                 Vedran Zeljeznak
>                 
>                 _______________________________________________
>                 FreeSWITCH-users mailing list
>                 FreeSWITCH-users at 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 fs-list at communicatefreely.net  Sat Feb  5 01:11:27 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 04 Feb 2011 17:11:27 -0500
Subject: [Freeswitch-users] mod_directory multi-tenant
In-Reply-To: <AANLkTinV80XmYE-PXuxLvnMoCpRS9q0c9G=E_3O=owxh@mail.gmail.com>
References: <000901cbc449$ea673480$bf359d80$@net>
	<AANLkTinV80XmYE-PXuxLvnMoCpRS9q0c9G=E_3O=owxh@mail.gmail.com>
Message-ID: <4D4C798F.2080108@communicatefreely.net>

It works, but there is a trick.

Take a look at what it's requesting if you use xml_curl.

You have to put all your users in a group called "default" otherwise it 
won't work.

In our situation, we set the domain specific to each customer, then send 
back a response based on the domain that is attached to the request when 
we generate the XML.

Hope that makes sense.

-Tim


>     Is there a trick to getting the mod_directory application working
>     in a multi-tenant configuration?  I think I?ve tried every
>     possible combination of settings and I cannot for the life of me
>     get it to work.  I can get the directory to answer but no matter
>     what I put in to the search, it comes back with no results.  Any
>     help would be appreciated!
>



From steveayre at gmail.com  Sat Feb  5 02:16:42 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 4 Feb 2011 23:16:42 +0000
Subject: [Freeswitch-users] H.323 implementation on FS
In-Reply-To: <1296857346.2599.26.camel@gustavo-laptop>
References: <AANLkTin_cDiA9PWcrhO6qSYwzamnxUAJZcOEZy2HMrkC@mail.gmail.com>
	<AANLkTi=kvGW5ugK41x9gRS6_hGoYNXmZQpBNDjkOBJSA@mail.gmail.com>
	<AANLkTikwcXZxLZ6wKH-7rubL0eW0W7Q4MwNf5R6a+hhZ@mail.gmail.com>
	<1296857346.2599.26.camel@gustavo-laptop>
Message-ID: <AANLkTi=Yfkwo7uz=Y=VkuGxGguNs2u+Z4tgjcVrHWfvf@mail.gmail.com>

I used to use this too. It does work pretty well, for the most part.

Q931 clearing causes did have issues though, because Yate converts the SIP
response code to an internal table (
http://yate.null.ro/pmwiki/index.php?n=Main.CallEndCauses) and then to a
Q931 clearing cause for H323.

It meant our customers didn't see the real clearing cause from the SIP
Reason header (H323 -> SIP calls), which in our case was a big issue since
we needed to pass back DCC 34 which Yate wouldn't give until I made a
patched version.

-Steve


On 4 February 2011 22:09, Gustavo Espeche <gustavo.espeche at upper-soft.com>wrote:

> Use Yate we are try to use mod_opal or mod_h323 in production with-out
> success, Yate has lot of issue but is a good inter-worker between h323
> and sip.
> Freeswitch has a lost of good think but the inter-worker isn't it.
> Because of then we use Freeswitch for billing propose but Yate for
> inter-worker.
> But if you need use FS for inter-worker use mod_h323 is a little more
> stable.
>
> Best Regards.
>
> Gustavo Espeche
> www.easyipcall.com
>
>
>
> On Fri, 2011-02-04 at 19:20 +0000, Steven Ayre wrote:
> > Although there's almost no H323 traffic - we're almost entirely SIP -
> > so I can't say how stable it is for a large load.
> >
> > -Steve
> >
> >
> > On 4 February 2011 19:20, Steven Ayre <steveayre at gmail.com> wrote:
> >         I had problems with mod_opal but mod_h323 works for me.
> >
> >         -Steve
> >
> >
> >
> >
> >
> >         On 4 February 2011 12:55, Vedran Zeljeznak
> >         <vedran.zeljeznak at gmail.com> wrote:
> >                 hi everyone,
> >
> >                 do any of you have any suggestions which H.323
> >                 implementation
> >                 (mod_opal or mod_h323) should be used on a production
> >                 Freeswitch
> >                 platform (HEAD version)?
> >
> >                 Is it better to forward calls from H.323 trunk through
> >                 Yate
> >                 (configured as H.323 to SIP proxy) before terminating
> >                 them on
> >                 Freeswitch?
> >
> >                 ---
> >                 Vedran Zeljeznak
> >
> >                 _______________________________________________
> >                 FreeSWITCH-users mailing list
> >                 FreeSWITCH-users at lists.freeswitch.org
> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >                 UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >                 http://www.freeswitch.org
> >
> >
> >
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  5 02:18:23 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 4 Feb 2011 23:18:23 +0000
Subject: [Freeswitch-users] Auto test FS install - now necessary?
In-Reply-To: <AANLkTik7KvMGZh3yz9c894ShK1-7T-WKyiVcsDcMc1mt@mail.gmail.com>
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com>
	<AANLkTikSryxPi1-YZ8Wpt0GUbcXf_i=4DMi2BmrVPWix@mail.gmail.com>
	<AANLkTinQArNvg6jZky6=8JUVYif9kQge6bkjnbqNncLL@mail.gmail.com>
	<4D4C32C7.9050503@aleph-com.net>
	<AANLkTinz-XCQHVympCqjzBL5=pfCOXiSPqNqbho7UNmX@mail.gmail.com>
	<AANLkTik7KvMGZh3yz9c894ShK1-7T-WKyiVcsDcMc1mt@mail.gmail.com>
Message-ID: <AANLkTinXm8St+5qx4krtoH_u6F=Bw9ZYw8dDRrq6+mi_@mail.gmail.com>

Missed the laptop thankfully ;)

On 4 February 2011 22:04, jesse <chat2jesse at gmail.com> wrote:

> If you misunderstood my point and ruined your keyboard, your own fault!  My
> friend.
> On Feb 4, 2011 9:21 AM, "Steven Ayre" <steveayre at gmail.com> wrote:
> > I actually spilt my tea. :)
> >
> >
> > On 4 February 2011 17:09, Darren Wiebe <darren at aleph-com.net> wrote:
> >
> >> On 04/02/2011 10:01 AM, Anthony Minessale wrote:
> >> > Reliability is more important than feature. Asterisk is a good example
> of
> >> > > such case.
> >> I snipped a very small part of this thread but the above line provided
> >> my comedy relief for the day. Thanks, I needed it.
> >>
> >> --
> >> Darren Wiebe
> >> Aleph Communications
> >> --------------------
> >> Phone: 1-877-702-2900
> >> Fax: 1-866-274-4506
> >> Email: darren at aleph-com.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
> >>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  5 02:32:51 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Sat, 5 Feb 2011 00:32:51 +0100
Subject: [Freeswitch-users] Lua Session creation is blocking?
In-Reply-To: <AANLkTin=J4TqxY4D5OfriK6iD=1-OfLg32YXyyXLL61d@mail.gmail.com>
References: <AANLkTin21o-gZH6FrPU6BxFR+RqxBkN15=DDGD66iTX-@mail.gmail.com>
	<AANLkTikdGqM2QsZ19_4YpkDRTd+b5_wRK9NrEGkoqgnM@mail.gmail.com>
	<AANLkTinaixoUDTyR-5gPZPLSaO_W0eNvC491ApAUHg-O@mail.gmail.com>
	<AANLkTimU-=7LJXzRy1xTAf+Vz8B6nexLBORJzi_EO5a9@mail.gmail.com>
	<AANLkTikPNom1k3f=fbixbtcCGw=WGzC9UZj25mBBnvjj@mail.gmail.com>
	<AANLkTin=J4TqxY4D5OfriK6iD=1-OfLg32YXyyXLL61d@mail.gmail.com>
Message-ID: <AANLkTimxAEMYegb7dmYZpYYvtP87w478Rz4iOXi74Lcg@mail.gmail.com>

Ok, thank you, my way is ESL.

ESL is less simple then embedded Lua but I understand that is the solution
to my problem.

Stephen


On Fri, Feb 4, 2011 at 10:57 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> The way you expect it to work is not the case.
>
> in embedded scripts when you create a new session you must wait for it
> to have media.
> The scripts are not asynchronous.  if you want asynchronous you 100% want
> ESL.
>
>
>
>
>
>
> On Fri, Feb 4, 2011 at 1:30 PM, Stephen Wilde <wstephen80 at gmail.com>
> wrote:
> > I have to port some applications on Freeswitch so I'm doing some
> preliminary
> > learning tests on Lua scripting.
> > The result that I would obtain is to have the full control of both
> inbound
> > and originated session and to have under control their state.
> > The script I'm running is simply: http://pastebin.freeswitch.org/15248
> > But its behaviour is not as I expect.
> >
> > Stephen
> >
> > On Fri, Feb 4, 2011 at 7:26 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> >>
> >>
> >> On Fri, Feb 4, 2011 at 10:19 AM, Stephen Wilde <wstephen80 at gmail.com>
> >> wrote:
> >>>
> >>> Ok, 18x will be perfect!
> >>> The problem is that when the inbound call is SIP and the originated is
> >>> ISDN, the originate doesn't return when a 180 ringing is received (that
> in
> >>> ISDN is an ALERT without in band info).
> >>> The originate returns when a 183 (ISDN = in band info available) or 200
> >>> (ISDN = CONNECT) is received.
> >>> Stephen
> >>>
> >>
> >> Could you tell us a bit more about what you're trying to accomplish? My
> >> guess is that there's a better way to do it. In most cases it is not
> >> necessary to create a new session inside of a script called from the
> >> dialplan.
> >> What is the application that you are working on? Who calls whom, etc.
> >> -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
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-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
>
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From infos at madovsky.org  Sat Feb  5 02:56:32 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 4 Feb 2011 18:56:32 -0500
Subject: [Freeswitch-users] Auto test FS install - now necessary?
References: <AANLkTikJ2ocU-vSMkzuEC0PC2R-ovvgqaxNb_BYPL0Jf@mail.gmail.com><AANLkTikSryxPi1-YZ8Wpt0GUbcXf_i=4DMi2BmrVPWix@mail.gmail.com><AANLkTinQArNvg6jZky6=8JUVYif9kQge6bkjnbqNncLL@mail.gmail.com><4D4C32C7.9050503@aleph-com.net><AANLkTinz-XCQHVympCqjzBL5=pfCOXiSPqNqbho7UNmX@mail.gmail.com><1296840880862-5993296.post@n2.nabble.com><7D4B9BFA-023D-4426-93C2-F6BDE2CCBF9C@ipeva.fr><1296847644406-5993736.post@n2.nabble.com><91EEF7A314AE4996A055EF2D4F622F4A@e1705>
	<1296850980061-5993907.post@n2.nabble.com>
Message-ID: <E52E0E455C1A49F98B8EF74086ED7A44@e1705>

I meant hosts.deny was an example,
you can use your own file

----- Original Message ----- 
From: "mazilo" <Nabble at slickdeals.endjunk.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Friday, February 04, 2011 3:23 PM
Subject: Re: [Freeswitch-users] Auto test FS install - now necessary?


>
>
> Madovsky wrote:
>> why not use a file like hosts.deny
>> from a perl/php/other script called with system ?
> IIRC, the host.deny only filters incoming calls from specific hosts. For
> instance, if you want to block crackers out there who try to exploit your 
> FS
> system, perhaps this is not a bad idea, but not if you only want to allow
> some specific incoming CIDs to be able to reach your internal extensions.
>
> -----
> 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/Auto-test-FS-install-now-necessary-tp5991624p5993907.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 Nabble at slickdeals.endjunk.com  Sat Feb  5 04:06:06 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 4 Feb 2011 17:06:06 -0800 (PST)
Subject: [Freeswitch-users] Using mod_opal with MSN Messenger
Message-ID: <1296867966381-5994575.post@n2.nabble.com>


Has anyone managed to configure mod_opal to register to MSN Messenger server
to place/receive calls to/from any MSN Messenger users? I took a look at the
conf/autoload_configs/opal.conf.xml file and don't know what to fill the
gk-address, gk-identifier, and gk-interface. Anyone?

-----
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/Using-mod-opal-with-MSN-Messenger-tp5994575p5994575.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From saeedahmad1981 at gmail.com  Sat Feb  5 12:06:24 2011
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Sat, 5 Feb 2011 10:06:24 +0100
Subject: [Freeswitch-users] H.323 implementation on FS
In-Reply-To: <AANLkTi=Yfkwo7uz=Y=VkuGxGguNs2u+Z4tgjcVrHWfvf@mail.gmail.com>
References: <AANLkTin_cDiA9PWcrhO6qSYwzamnxUAJZcOEZy2HMrkC@mail.gmail.com>
	<AANLkTi=kvGW5ugK41x9gRS6_hGoYNXmZQpBNDjkOBJSA@mail.gmail.com>
	<AANLkTikwcXZxLZ6wKH-7rubL0eW0W7Q4MwNf5R6a+hhZ@mail.gmail.com>
	<1296857346.2599.26.camel@gustavo-laptop>
	<AANLkTi=Yfkwo7uz=Y=VkuGxGguNs2u+Z4tgjcVrHWfvf@mail.gmail.com>
Message-ID: <AANLkTinoWodbeP-GsBCVUNLO1z-xYFNoz_KO71gaE2j6@mail.gmail.com>

Hi,

My vote for Yate + FS

http://voip.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy

works without issue.

But i really wish to  have it in FS, so we save an extra server.
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From saeedahmad1981 at gmail.com  Sat Feb  5 13:30:41 2011
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Sat, 5 Feb 2011 11:30:41 +0100
Subject: [Freeswitch-users] time_test on Centos 5.5
In-Reply-To: <AANLkTi=NEPwvbF70y3j0p_jfSgLg=xBxW7Es1k7RgAVm@mail.gmail.com>
References: <AANLkTiksCg70iEdZiN1+CnfyBW87mg7T4jDHjeq7JW5Y@mail.gmail.com>
	<0E4A540B-3E61-4940-8246-6FBF67CF91D8@ipeva.fr>
	<41F744A2-90FF-405F-AF62-5E7B8FB5128F@carmickle.com>
	<AANLkTi=NEPwvbF70y3j0p_jfSgLg=xBxW7Es1k7RgAVm@mail.gmail.com>
Message-ID: <AANLkTi=9Zsu2aMifLC406iabet6Q9=n3YOTa0N+BLPck@mail.gmail.com>

how we set kenrel timer freq to 1000hz? on centos?

On Sat, Jan 29, 2011 at 1:10 AM, Steven Ayre <steveayre at gmail.com> wrote:

> I've been using it on Lenny with no problems for ~2 years, timing works
> fine. It will work. CentOS is the reference platform though.
>
> -Steve
>
>
>
>
> On 28 January 2011 19:05, Frank Carmickle <frank at carmickle.com> wrote:
>
>>
>> On Jan 28, 2011, at 6:35 AM, David Ponzone wrote:
>> Snip...
>>
>> > Stay away from Debian, Centos is the right choice.
>> > You could eventually try to fallback to centos 5.3 or 5.4.
>> >
>> Debian can work if that's what people want to use.  I have it working well
>> on a few lenny machines.
>>
>> --FC
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.ponzone at ipeva.fr  Sat Feb  5 13:39:40 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sat, 5 Feb 2011 11:39:40 +0100
Subject: [Freeswitch-users] time_test on Centos 5.5
In-Reply-To: <AANLkTi=9Zsu2aMifLC406iabet6Q9=n3YOTa0N+BLPck@mail.gmail.com>
References: <AANLkTiksCg70iEdZiN1+CnfyBW87mg7T4jDHjeq7JW5Y@mail.gmail.com>
	<0E4A540B-3E61-4940-8246-6FBF67CF91D8@ipeva.fr>
	<41F744A2-90FF-405F-AF62-5E7B8FB5128F@carmickle.com>
	<AANLkTi=NEPwvbF70y3j0p_jfSgLg=xBxW7Es1k7RgAVm@mail.gmail.com>
	<AANLkTi=9Zsu2aMifLC406iabet6Q9=n3YOTa0N+BLPck@mail.gmail.com>
Message-ID: <82D793A3-B5AB-45CC-B4B0-C12742A60C43@ipeva.fr>

You have to recompile it if it's not already set, but AFAIR, CentOs is already a 1000Hz.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 05/02/2011 ? 11:30, Saeed Ahmed a ?crit :

> how we set kenrel timer freq to 1000hz? on centos?
> 
> On Sat, Jan 29, 2011 at 1:10 AM, Steven Ayre <steveayre at gmail.com> wrote:
> I've been using it on Lenny with no problems for ~2 years, timing works fine. It will work. CentOS is the reference platform though.
> 
> -Steve
> 
> 
> 
> 
> On 28 January 2011 19:05, Frank Carmickle <frank at carmickle.com> wrote:
> 
> On Jan 28, 2011, at 6:35 AM, David Ponzone wrote:
> Snip...
> 
> > Stay away from Debian, Centos is the right choice.
> > You could eventually try to fallback to centos 5.3 or 5.4.
> >
> Debian can work if that's what people want to use.  I have it working well on a few lenny machines.
> 
> --FC
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  5 15:54:36 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 5 Feb 2011 12:54:36 +0000
Subject: [Freeswitch-users] H.323 implementation on FS
In-Reply-To: <AANLkTinoWodbeP-GsBCVUNLO1z-xYFNoz_KO71gaE2j6@mail.gmail.com>
References: <AANLkTin_cDiA9PWcrhO6qSYwzamnxUAJZcOEZy2HMrkC@mail.gmail.com>
	<AANLkTi=kvGW5ugK41x9gRS6_hGoYNXmZQpBNDjkOBJSA@mail.gmail.com>
	<AANLkTikwcXZxLZ6wKH-7rubL0eW0W7Q4MwNf5R6a+hhZ@mail.gmail.com>
	<1296857346.2599.26.camel@gustavo-laptop>
	<AANLkTi=Yfkwo7uz=Y=VkuGxGguNs2u+Z4tgjcVrHWfvf@mail.gmail.com>
	<AANLkTinoWodbeP-GsBCVUNLO1z-xYFNoz_KO71gaE2j6@mail.gmail.com>
Message-ID: <2A51D676-DE68-4C6F-AD38-C6B42E747B49@gmail.com>

I used to run yate on the same server as fs... Worked fine.

Just assign the server 2 ips. Listen on one with fs sip and yate h323, and listen on the 2nd with yate sip.

Steve on iPhone

On 5 Feb 2011, at 09:06, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:

> Hi,
> 
> My vote for Yate + FS
> 
> http://voip.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy
> 
> works without issue.
> 
> But i really wish to  have it in FS, so we save an extra server.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  5 18:27:06 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 5 Feb 2011 09:27:06 -0600
Subject: [Freeswitch-users] time_test on Centos 5.5
In-Reply-To: <82D793A3-B5AB-45CC-B4B0-C12742A60C43@ipeva.fr>
References: <AANLkTiksCg70iEdZiN1+CnfyBW87mg7T4jDHjeq7JW5Y@mail.gmail.com>
	<0E4A540B-3E61-4940-8246-6FBF67CF91D8@ipeva.fr>
	<41F744A2-90FF-405F-AF62-5E7B8FB5128F@carmickle.com>
	<AANLkTi=NEPwvbF70y3j0p_jfSgLg=xBxW7Es1k7RgAVm@mail.gmail.com>
	<AANLkTi=9Zsu2aMifLC406iabet6Q9=n3YOTa0N+BLPck@mail.gmail.com>
	<82D793A3-B5AB-45CC-B4B0-C12742A60C43@ipeva.fr>
Message-ID: <AANLkTikorDnhnYmOv-xiu8GCCC+MG=C5uxJviWBdCjHh@mail.gmail.com>

We have not signed off on 5.5 yet. Only 5.2 to 5.4
On Feb 5, 2011 4:41 AM, "David Ponzone" <david.ponzone at ipeva.fr> wrote:
> You have to recompile it if it's not already set, but AFAIR, CentOs is
already a 1000Hz.
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr
> tel: 01 74 03 18 97
> gsm: 06 66 98 76 34
>
> Service Client IPeva
> tel: 0811 46 26 26
> www.ipeva.fr - www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
non autoris?e est interdite. Tout message ?lectronique est susceptible
d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
> Le 05/02/2011 ? 11:30, Saeed Ahmed a ?crit :
>
>> how we set kenrel timer freq to 1000hz? on centos?
>>
>> On Sat, Jan 29, 2011 at 1:10 AM, Steven Ayre <steveayre at gmail.com> wrote:
>> I've been using it on Lenny with no problems for ~2 years, timing works
fine. It will work. CentOS is the reference platform though.
>>
>> -Steve
>>
>>
>>
>>
>> On 28 January 2011 19:05, Frank Carmickle <frank at carmickle.com> wrote:
>>
>> On Jan 28, 2011, at 6:35 AM, David Ponzone wrote:
>> Snip...
>>
>> > Stay away from Debian, Centos is the right choice.
>> > You could eventually try to fallback to centos 5.3 or 5.4.
>> >
>> Debian can work if that's what people want to use. I have it working well
on a few lenny machines.
>>
>> --FC
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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.ponzone at ipeva.fr  Sat Feb  5 21:06:51 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sat, 5 Feb 2011 19:06:51 +0100
Subject: [Freeswitch-users] time_test on Centos 5.5
In-Reply-To: <AANLkTikorDnhnYmOv-xiu8GCCC+MG=C5uxJviWBdCjHh@mail.gmail.com>
References: <AANLkTiksCg70iEdZiN1+CnfyBW87mg7T4jDHjeq7JW5Y@mail.gmail.com>
	<0E4A540B-3E61-4940-8246-6FBF67CF91D8@ipeva.fr>
	<41F744A2-90FF-405F-AF62-5E7B8FB5128F@carmickle.com>
	<AANLkTi=NEPwvbF70y3j0p_jfSgLg=xBxW7Es1k7RgAVm@mail.gmail.com>
	<AANLkTi=9Zsu2aMifLC406iabet6Q9=n3YOTa0N+BLPck@mail.gmail.com>
	<82D793A3-B5AB-45CC-B4B0-C12742A60C43@ipeva.fr>
	<AANLkTikorDnhnYmOv-xiu8GCCC+MG=C5uxJviWBdCjHh@mail.gmail.com>
Message-ID: <0E4DBB6D-77EA-4770-B069-6C543DE49E99@ipeva.fr>

Anthony,

meaning Centos 5.2 to 5.4 does not default to 1000Hz ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 05/02/2011 ? 16:27, Anthony Minessale a ?crit :

> We have not signed off on 5.5 yet. Only 5.2 to 5.4
> 
> On Feb 5, 2011 4:41 AM, "David Ponzone" <david.ponzone at ipeva.fr> wrote:
> > You have to recompile it if it's not already set, but AFAIR, CentOs is already a 1000Hz.
> > 
> > David Ponzone Direction Technique
> > email: david.ponzone at ipeva.fr
> > tel: 01 74 03 18 97
> > gsm: 06 66 98 76 34
> > 
> > Service Client IPeva
> > tel: 0811 46 26 26
> > www.ipeva.fr - www.ipeva-studio.com
> > 
> > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> > 
> > 
> > 
> > 
> > Le 05/02/2011 ? 11:30, Saeed Ahmed a ?crit :
> > 
> >> how we set kenrel timer freq to 1000hz? on centos?
> >> 
> >> On Sat, Jan 29, 2011 at 1:10 AM, Steven Ayre <steveayre at gmail.com> wrote:
> >> I've been using it on Lenny with no problems for ~2 years, timing works fine. It will work. CentOS is the reference platform though.
> >> 
> >> -Steve
> >> 
> >> 
> >> 
> >> 
> >> On 28 January 2011 19:05, Frank Carmickle <frank at carmickle.com> wrote:
> >> 
> >> On Jan 28, 2011, at 6:35 AM, David Ponzone wrote:
> >> Snip...
> >> 
> >> > Stay away from Debian, Centos is the right choice.
> >> > You could eventually try to fallback to centos 5.3 or 5.4.
> >> >
> >> Debian can work if that's what people want to use. I have it working well on a few lenny machines.
> >> 
> >> --FC
> >> 
> >> 
> >> 
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >> 
> >> 
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >> 
> >> 
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> > 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From saeedahmad1981 at gmail.com  Sun Feb  6 00:53:36 2011
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Sat, 5 Feb 2011 22:53:36 +0100
Subject: [Freeswitch-users] H.323 implementation on FS
In-Reply-To: <2A51D676-DE68-4C6F-AD38-C6B42E747B49@gmail.com>
References: <AANLkTin_cDiA9PWcrhO6qSYwzamnxUAJZcOEZy2HMrkC@mail.gmail.com>
	<AANLkTi=kvGW5ugK41x9gRS6_hGoYNXmZQpBNDjkOBJSA@mail.gmail.com>
	<AANLkTikwcXZxLZ6wKH-7rubL0eW0W7Q4MwNf5R6a+hhZ@mail.gmail.com>
	<1296857346.2599.26.camel@gustavo-laptop>
	<AANLkTi=Yfkwo7uz=Y=VkuGxGguNs2u+Z4tgjcVrHWfvf@mail.gmail.com>
	<AANLkTinoWodbeP-GsBCVUNLO1z-xYFNoz_KO71gaE2j6@mail.gmail.com>
	<2A51D676-DE68-4C6F-AD38-C6B42E747B49@gmail.com>
Message-ID: <AANLkTim2V+rDZvGb1X_WF2ZkCPQ-4sByNPP5YZ3pdg5Q@mail.gmail.com>

Hmm good, i'll try...

On Sat, Feb 5, 2011 at 1:54 PM, Steven Ayre <steveayre at gmail.com> wrote:

> I used to run yate on the same server as fs... Worked fine.
>
> Just assign the server 2 ips. Listen on one with fs sip and yate h323, and
> listen on the 2nd with yate sip.
>
> Steve on iPhone
>
> On 5 Feb 2011, at 09:06, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
>
> Hi,
>
> My vote for Yate + FS
>
> <http://voip.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy>
> http://voip.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy
>
> works without issue.
>
> But i really wish to  have it in FS, so we save an extra server.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  6 01:05:04 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 5 Feb 2011 22:05:04 +0000
Subject: [Freeswitch-users] H.323 implementation on FS
In-Reply-To: <AANLkTim2V+rDZvGb1X_WF2ZkCPQ-4sByNPP5YZ3pdg5Q@mail.gmail.com>
References: <AANLkTin_cDiA9PWcrhO6qSYwzamnxUAJZcOEZy2HMrkC@mail.gmail.com>
	<AANLkTi=kvGW5ugK41x9gRS6_hGoYNXmZQpBNDjkOBJSA@mail.gmail.com>
	<AANLkTikwcXZxLZ6wKH-7rubL0eW0W7Q4MwNf5R6a+hhZ@mail.gmail.com>
	<1296857346.2599.26.camel@gustavo-laptop>
	<AANLkTi=Yfkwo7uz=Y=VkuGxGguNs2u+Z4tgjcVrHWfvf@mail.gmail.com>
	<AANLkTinoWodbeP-GsBCVUNLO1z-xYFNoz_KO71gaE2j6@mail.gmail.com>
	<2A51D676-DE68-4C6F-AD38-C6B42E747B49@gmail.com>
	<AANLkTim2V+rDZvGb1X_WF2ZkCPQ-4sByNPP5YZ3pdg5Q@mail.gmail.com>
Message-ID: <5CD94C35-F5C1-4D70-B879-ABDDC1D5056D@gmail.com>

Different port numbers on 1 ip might work too...

Steve on iPhone

On 5 Feb 2011, at 21:53, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:

> Hmm good, i'll try...
> 
> On Sat, Feb 5, 2011 at 1:54 PM, Steven Ayre <steveayre at gmail.com> wrote:
> I used to run yate on the same server as fs... Worked fine.
> 
> Just assign the server 2 ips. Listen on one with fs sip and yate h323, and listen on the 2nd with yate sip.
> 
> Steve on iPhone
> 
> On 5 Feb 2011, at 09:06, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
> 
>> Hi,
>> 
>> My vote for Yate + FS
>> 
>> http://voip.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy
>> 
>> works without issue.
>> 
>> But i really wish to  have it in FS, so we save an extra server.
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From saeedahmad1981 at gmail.com  Sun Feb  6 01:22:42 2011
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Sat, 5 Feb 2011 23:22:42 +0100
Subject: [Freeswitch-users] H.323 implementation on FS
In-Reply-To: <5CD94C35-F5C1-4D70-B879-ABDDC1D5056D@gmail.com>
References: <AANLkTin_cDiA9PWcrhO6qSYwzamnxUAJZcOEZy2HMrkC@mail.gmail.com>
	<AANLkTi=kvGW5ugK41x9gRS6_hGoYNXmZQpBNDjkOBJSA@mail.gmail.com>
	<AANLkTikwcXZxLZ6wKH-7rubL0eW0W7Q4MwNf5R6a+hhZ@mail.gmail.com>
	<1296857346.2599.26.camel@gustavo-laptop>
	<AANLkTi=Yfkwo7uz=Y=VkuGxGguNs2u+Z4tgjcVrHWfvf@mail.gmail.com>
	<AANLkTinoWodbeP-GsBCVUNLO1z-xYFNoz_KO71gaE2j6@mail.gmail.com>
	<2A51D676-DE68-4C6F-AD38-C6B42E747B49@gmail.com>
	<AANLkTim2V+rDZvGb1X_WF2ZkCPQ-4sByNPP5YZ3pdg5Q@mail.gmail.com>
	<5CD94C35-F5C1-4D70-B879-ABDDC1D5056D@gmail.com>
Message-ID: <AANLkTik=0nVEsV=ZNQcK_CJbBJ0UvgbqJU6QezeJYPT6@mail.gmail.com>

yup thats my idea, because for sip 2 sip i use fs only, for sip -> h323,
source sip port (yate) could run on 5061 and h323 -> sip won't create any
prob. I just wish that both (fs+yate) will work fine in high call pressure.

How many cps and CC were you able to handle?

i expect 1000 CC on xecon 8 core, 8 gig ram.


On Sat, Feb 5, 2011 at 11:05 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Different port numbers on 1 ip might work too...
>
> Steve on iPhone
>
> On 5 Feb 2011, at 21:53, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
>
> Hmm good, i'll try...
>
> On Sat, Feb 5, 2011 at 1:54 PM, Steven Ayre < <steveayre at gmail.com>
> steveayre at gmail.com> wrote:
>
>> I used to run yate on the same server as fs... Worked fine.
>>
>> Just assign the server 2 ips. Listen on one with fs sip and yate h323, and
>> listen on the 2nd with yate sip.
>>
>> Steve on iPhone
>>
>> On 5 Feb 2011, at 09:06, Saeed Ahmed < <saeedahmad1981 at gmail.com>
>> saeedahmad1981 at gmail.com> wrote:
>>
>> Hi,
>>
>> My vote for Yate + FS
>>
>> <http://voip.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy><http://voip.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy>
>> http://voip.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy
>>
>> works without issue.
>>
>> But i really wish to  have it in FS, so we save an extra server.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> <FreeSWITCH-users at lists.freeswitch.org>
>> 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>
>> 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
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sun Feb  6 02:54:54 2011
From: william.suffill at gmail.com (William Suffill)
Date: Sat, 5 Feb 2011 18:54:54 -0500
Subject: [Freeswitch-users] time_test on Centos 5.5
In-Reply-To: <0E4DBB6D-77EA-4770-B069-6C543DE49E99@ipeva.fr>
References: <AANLkTiksCg70iEdZiN1+CnfyBW87mg7T4jDHjeq7JW5Y@mail.gmail.com>
	<0E4A540B-3E61-4940-8246-6FBF67CF91D8@ipeva.fr>
	<41F744A2-90FF-405F-AF62-5E7B8FB5128F@carmickle.com>
	<AANLkTi=NEPwvbF70y3j0p_jfSgLg=xBxW7Es1k7RgAVm@mail.gmail.com>
	<AANLkTi=9Zsu2aMifLC406iabet6Q9=n3YOTa0N+BLPck@mail.gmail.com>
	<82D793A3-B5AB-45CC-B4B0-C12742A60C43@ipeva.fr>
	<AANLkTikorDnhnYmOv-xiu8GCCC+MG=C5uxJviWBdCjHh@mail.gmail.com>
	<0E4DBB6D-77EA-4770-B069-6C543DE49E99@ipeva.fr>
Message-ID: <AANLkTik9-EmuK7VrO2nEaDrT7hvUU+BQHdv96ztxYuH6@mail.gmail.com>

Meaning those are the versions of CentOS they have used first hand & sign
off that it will work correctly without any issue.

There has been some issues with I believe it was libc version in the past to
the point the recommended 5.4 instead of 5.5.

-- W
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From Nabble at slickdeals.endjunk.com  Sun Feb  6 05:14:53 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sat, 5 Feb 2011 18:14:53 -0800 (PST)
Subject: [Freeswitch-users] ${domain} not working
In-Reply-To: <AANLkTimk-PsnWCUuXYeybfu2_rL3qzFtYrAdnfhECQvV@mail.gmail.com>
References: <4D4A9E78.3090305@yellox.de>
	<AANLkTinsbXcdCFv6FDc+1vG_kfUJb24HOf6vsw913PDQ@mail.gmail.com>
	<AANLkTimpAw0tZhCMErpWE6w19nfey02zdx2s+M8NC4PG@mail.gmail.com>
	<AANLkTimk-PsnWCUuXYeybfu2_rL3qzFtYrAdnfhECQvV@mail.gmail.com>
Message-ID: <1296958493751-5996900.post@n2.nabble.com>


Thanks. My FS is behind a NAT/Firewall on a private LAN and I just upgraded
and eval ${domain} now returns the IP Address of my FS host (using default
settings from conf/vars.xml)
freeswitch at internal> version
FreeSWITCH Version 1.0.head (git-2944364 2011-02-04 16-53-38 -0600)

freeswitch at internal> eval ${domain}
192.168.1.123
freeswitch at internal>

-----
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/domain-not-working-tp5988752p5996900.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Sun Feb  6 06:19:33 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sat, 5 Feb 2011 19:19:33 -0800 (PST)
Subject: [Freeswitch-users] ext-rtp-ip value behind NAT
Message-ID: <1296962373969-5996978.post@n2.nabble.com>


With FreeSWITCH Version 1.0.head (git-2944364 2011-02-04 16-53-38 -0600) on
my Seagate DockStar, I configured my jingle profile using the following two
different options:

1. 
2. 

With option #1, no incoming call and caller got intercepted by GV voicemail.
fs_cli dumped this  http://pastebin.com/8sJjUZx1 message  as seen on line#
26 and here is the  http://pastebin.com/gnb8dXmW output  of dl_debug on.
With option #2, the call comes in. When I picked up to answer the call, no
audio and caller still heard the ringing tone until the call got intercepted
by GV voicemail. fs_cli dumped this  http://pastebin.com/fJ1s14tX message 
and the  http://pastebin.com/mHBFECs3 output  of dl_debug on.

If anyone here has any idea how to fix this, I sure would appreciate that.
Thanks.

-----
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/ext-rtp-ip-value-behind-NAT-tp5996978p5996978.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From adminjew at gmail.com  Sun Feb  6 07:36:43 2011
From: adminjew at gmail.com (Yitzchok)
Date: Sat, 5 Feb 2011 23:36:43 -0500
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
 for Windows in .NET
In-Reply-To: <AANLkTi=skc8OGLqkiTULMN-pSL9SNgwK-YrcLq6N-GXp@mail.gmail.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
	<16100.1296638658@ccs.covici.com>
	<1296658879889-5985076.post@n2.nabble.com>
	<6549.1296659705@ccs.covici.com>
	<AANLkTinCVZqYRxd0ZcQLj_kj42fYHDqUsEocEqW3oaph@mail.gmail.com>
	<AANLkTinNjGKvGXKvTh1dMgZVkwdimkPoUZPJKu7233-T@mail.gmail.com>
	<AANLkTikP4H38vmvvVTCHRJdRa=60f4q4RLL+A3GiznNO@mail.gmail.com>
	<AANLkTi=skc8OGLqkiTULMN-pSL9SNgwK-YrcLq6N-GXp@mail.gmail.com>
Message-ID: <AANLkTinLND_RPkBG-5jdCoBKUa+SMw+Pe1OVo8igf0mz@mail.gmail.com>

This is what I have in the plugins.log after I try to run the application
(the main screen shows up for a few seconds but an exception is thrown)


---- plugins.log -----
2/5/2011 11:30:30 PM - Headset Plugin Manager: Plugin JabraProvider had an
error Due to: Could not load file or assembly 'JabraTelephonyAPI,
Version=1.2.0.0, Culture=neutral, PublicKeyToken=8f966dc7fd80e1d9' or one of
its dependencies. The system cannot find the file specified.

2/5/2011 11:30:30 PM - Headset Plugin Manager: Error creating headset plugin
from dll "C:\Program Files (x86)\FSClient\Plugins\JabraHeadset.dll" of:
Could not load file or assembly 'JabraTelephonyAPI, Version=1.2.0.0,
Culture=neutral, PublicKeyToken=8f966dc7fd80e1d9' or one of its
dependencies. The system cannot find the file specified.
2/5/2011 11:30:30 PM - Headset Plugin Manager: Plugin Plantronics Provider
had an error Due to: Could not load file or assembly
'Plantronics.Device.Common, Version=2.1.39537.0, Culture=neutral,
PublicKeyToken=a8048bce41894b6e' or one of its dependencies. The system
cannot find the file specified.

2/5/2011 11:30:49 PM - Headset Plugin Manager: Error terminating a plugin:
Could not load file or assembly 'JabraTelephonyAPI, Version=1.2.0.0,
Culture=neutral, PublicKeyToken=8f966dc7fd80e1d9' or one of its
dependencies. The system cannot find the file specified.

----freeswitch.log----
2011-02-05 23:33:48.934072 [ERR] mod_PortAudio.c:1191 Cannot find an input
device
2011-02-05 23:33:48.934072 [WARNING] mod_PortAudio.c:1199 Switching to
default output device
2011-02-05 23:33:48.934072 [WARNING] mod_PortAudio.c:1208 Invalid ring
device configured using output device
2011-02-05 23:33:48.934072 [CRIT] switch_loadable_module.c:928 Error Loading
module C:\Program Files (x86)\FSClient\mod\mod_portaudio.dll
**Module load routine returned an error**



--
Yitzchok


On Wed, Feb 2, 2011 at 10:37 PM, EdPimentl <edpimentl at gmail.com> wrote:

> Thanks
>
> -E
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  6 14:25:32 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 6 Feb 2011 11:25:32 +0000
Subject: [Freeswitch-users] H.323 implementation on FS
In-Reply-To: <AANLkTik=0nVEsV=ZNQcK_CJbBJ0UvgbqJU6QezeJYPT6@mail.gmail.com>
References: <AANLkTin_cDiA9PWcrhO6qSYwzamnxUAJZcOEZy2HMrkC@mail.gmail.com>
	<AANLkTi=kvGW5ugK41x9gRS6_hGoYNXmZQpBNDjkOBJSA@mail.gmail.com>
	<AANLkTikwcXZxLZ6wKH-7rubL0eW0W7Q4MwNf5R6a+hhZ@mail.gmail.com>
	<1296857346.2599.26.camel@gustavo-laptop>
	<AANLkTi=Yfkwo7uz=Y=VkuGxGguNs2u+Z4tgjcVrHWfvf@mail.gmail.com>
	<AANLkTinoWodbeP-GsBCVUNLO1z-xYFNoz_KO71gaE2j6@mail.gmail.com>
	<2A51D676-DE68-4C6F-AD38-C6B42E747B49@gmail.com>
	<AANLkTim2V+rDZvGb1X_WF2ZkCPQ-4sByNPP5YZ3pdg5Q@mail.gmail.com>
	<5CD94C35-F5C1-4D70-B879-ABDDC1D5056D@gmail.com>
	<AANLkTik=0nVEsV=ZNQcK_CJbBJ0UvgbqJU6QezeJYPT6@mail.gmail.com>
Message-ID: <AANLkTinw5UXGL5AGkbAPxgMW+bUVYu2apiwyPW+4AU0E@mail.gmail.com>

Similar hardware, but tested with no more than about 25CPS (almost all our
traffic had already been moved to use SIP). Sure you'll be able to handle
more though.

-Steve



On 5 February 2011 22:22, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:

> yup thats my idea, because for sip 2 sip i use fs only, for sip -> h323,
> source sip port (yate) could run on 5061 and h323 -> sip won't create any
> prob. I just wish that both (fs+yate) will work fine in high call pressure.
>
> How many cps and CC were you able to handle?
>
> i expect 1000 CC on xecon 8 core, 8 gig ram.
>
>
> On Sat, Feb 5, 2011 at 11:05 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Different port numbers on 1 ip might work too...
>>
>> Steve on iPhone
>>
>> On 5 Feb 2011, at 21:53, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
>>
>> Hmm good, i'll try...
>>
>> On Sat, Feb 5, 2011 at 1:54 PM, Steven Ayre < <steveayre at gmail.com>
>> steveayre at gmail.com> wrote:
>>
>>> I used to run yate on the same server as fs... Worked fine.
>>>
>>> Just assign the server 2 ips. Listen on one with fs sip and yate h323,
>>> and listen on the 2nd with yate sip.
>>>
>>> Steve on iPhone
>>>
>>> On 5 Feb 2011, at 09:06, Saeed Ahmed < <saeedahmad1981 at gmail.com>
>>> saeedahmad1981 at gmail.com> wrote:
>>>
>>> Hi,
>>>
>>> My vote for Yate + FS
>>>
>>> <http://voip.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy><http://voip.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy>
>>> http://voip.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy
>>>
>>> works without issue.
>>>
>>> But i really wish to  have it in FS, so we save an extra server.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> <FreeSWITCH-users at lists.freeswitch.org>
>>> 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>
>>> 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
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mbsip at gazeta.pl  Sun Feb  6 16:58:15 2011
From: mbsip at gazeta.pl (Maciej Bylica)
Date: Sun, 6 Feb 2011 14:58:15 +0100
Subject: [Freeswitch-users] Freeswitch as a b2bua
Message-ID: <AANLkTi=DF=5w=RJAVneskBe5=aALYJcye6H6ue0dMkB4@mail.gmail.com>

Hi,

I am thinking about the way to have b2bua with full rtp proxy functionalities.
The one wayout is to incorporate Freeswitch into the sip call flow.
My scenario is as follows: Operator_1 ---> OpenSIPS ----> Operator_2
Of course Opensips has B2b functionality but it may be pretty tricky
to have this done.
The question is could I place FS just after Opensips box like following:
Operator_1 ---> OpenSIPS ----> Freewsitch ----> Operator_2 to achieve
fully topology hiding (so no SIP/SDP  information of Operator_2 should
pass to Operator_1 and vice versa).
Could I achieve this by using FS?
Is the configuration pretty straightforward?
What more less will be cps for Quad-core E5620 with 8GB of RAM for
aforementioned scenario (here i guess Opensips could be unbeatable).
Is 64bit installation a stable one?

Thanks in advance,
Maciej.


From jeff at jefflenk.com  Sun Feb  6 18:43:43 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sun, 6 Feb 2011 07:43:43 -0800 (PST)
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
	for Windows in .NET
In-Reply-To: <AANLkTinLND_RPkBG-5jdCoBKUa+SMw+Pe1OVo8igf0mz@mail.gmail.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
	<16100.1296638658@ccs.covici.com>
	<1296658879889-5985076.post@n2.nabble.com>
	<6549.1296659705@ccs.covici.com>
	<AANLkTinCVZqYRxd0ZcQLj_kj42fYHDqUsEocEqW3oaph@mail.gmail.com>
	<AANLkTinNjGKvGXKvTh1dMgZVkwdimkPoUZPJKu7233-T@mail.gmail.com>
	<AANLkTikP4H38vmvvVTCHRJdRa=60f4q4RLL+A3GiznNO@mail.gmail.com>
	<AANLkTi=skc8OGLqkiTULMN-pSL9SNgwK-YrcLq6N-GXp@mail.gmail.com>
	<AANLkTinLND_RPkBG-5jdCoBKUa+SMw+Pe1OVo8igf0mz@mail.gmail.com>
Message-ID: <1297007023790-5997781.post@n2.nabble.com>


The plugins are not needed to use.
Investigate why mod_portaudio doesnt load. What OS(and service level) and
audio hardware?
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FSClient-Release-A-FreeSWITCH-SIP-Client-for-Windows-in-NET-tp5983787p5997781.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From potxoka at gmail.com  Sun Feb  6 19:07:07 2011
From: potxoka at gmail.com (Antonio)
Date: Sun, 06 Feb 2011 17:07:07 +0100
Subject: [Freeswitch-users] Freeswitch and mysql
In-Reply-To: <AANLkTinYebKhBXQGdwV02bi_kBPB4Vp7Q3iGCv9YRDD7@mail.gmail.com>
References: <4CA11C69.70302@gmail.com> <4D375040.1000209@gmail.com>
	<AANLkTinYebKhBXQGdwV02bi_kBPB4Vp7Q3iGCv9YRDD7@mail.gmail.com>
Message-ID: <4D4EC72B.7080408@gmail.com>

El 19/01/11 22:16, Avi Marcus escribi?:
> Integrate freeswitch with mysql is vague. That itself could mean 
> something like having freeswitch store it's database in mysql, but 
> that doesn't seem to be what you want.
> You are asking how to use a database (any, not just mysql) to 
> configure the dialplan - users, extensions, conferences, whatever.
> 1) You can use lua to process the calls and have it query your database
> 2) If you just need simple sql queries, check the mod_odbc_query from 
> the git contrib. If you already understand the dialplan basics, then 
> this can easily let you query the database as part of that.
> 3) However, if you need more complicated things, then mod_xml_curl is 
> your friend - it lets you grab dynamicly generated XML files for each 
> call.
> I myself use php to query a mysql database for how much to charge for 
> the call, a custom LCR implementation, etc.
> I posted the basic classes to github a while ago: 
> https://github.com/avimar/FreeSWITCH-mod_xml-with-PHP
> Also, intralanman wrote a very modular, all inclusive xml_curl 
> implementation in php - which if you understand it (I didn't know it 
> existed) should be really helpful. You can find that in the git 
> contrib also in: intralanman/PHP/fs_curl
>
> -Avi Marcus
>
Hi

Thanks for everyone's responses, are really helping me to really know to 
look for and how to document ;-). I had thought one thing and I have a 
question: Could use python for this cause?. I'll look at me while the 
modules that have told me;-). Thank you very much.


Regards

<action application="python" data="extension_user"/>

--------- extension_user.xml ---------
<include>
<user id="1000" mailbox="1000">
<params>
<param name="password" value="1234"/>
<param name="vm-password" value="1000"/>
</params>
<variables>
<variable name="toll_allow" value="domestic,international,local"/>
<variable name="accountcode" value="1000"/>
<variable name="user_context" value="default"/>
<variable name="effective_caller_id_name" value="NAME"/>
<variable name="effective_caller_id_number" value="1000"/>
<variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
<variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
<variable name="callgroup" value="techsupport"/>
</variables>
</user>
</include>
--------- extension_user.xml ---------
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From steveu at coppice.org  Sun Feb  6 19:56:03 2011
From: steveu at coppice.org (Steve Underwood)
Date: Mon, 07 Feb 2011 00:56:03 +0800
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
 for Windows in .NET
In-Reply-To: <AANLkTimghBGjRNXA4GAVXvGFU0KQX5cEqQMREO5U=DvW@mail.gmail.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
	<AANLkTimghBGjRNXA4GAVXvGFU0KQX5cEqQMREO5U=DvW@mail.gmail.com>
Message-ID: <4D4ED2A3.1000005@coppice.org>

On 02/02/2011 09:48 PM, Jo?o Mesquita wrote:
> Mitch, you beat me to it...
>
> I think it might be time for me to get back to FSComm and drop being 
> picky about the AEC thing? Sad but true, I think that the only way for 
> me to draw attention of someone who knows about this stuff is actually 
> doing the rest like you did. Great initiative. I truly congratulate 
> you from pulling something I couldn't so far.
>
I am puzzled why the AEC issue would ever have affected your development 
of FSComm. Although a headset or handset needs EC for really good 
telephone, the lack fo AEC is only a killer for speakerphone usage.

On this topic, a little experimentation suggests the following...

Skype appears to do good AEC on a PC, but when you listen carefully is 
seems more like they avoid proper AEC, and use the kind of adaptive gain 
juggling the DSP Group pushed heavily in the early 90s. The final result 
is very acceptable, though, and avoids the problems to non-matching mic 
and speaker sampling rates that you get with many PC sound cards.

Skype on a Symbian handset and a Mac seems to do proper echo 
cancellation, and gives results a bit better than on a PC. The limited 
range of hardware used in these platforms makes AEC more of a practical 
option.

If anyone has further information about the exact schemes used by Skype, 
I would really like to hear from them. Skype is used on so many 
machines, that whatever strategy they have found workable across a wide 
range of hardware should be a safe bet for us to use.

Steve



From mitch.capper at gmail.com  Sun Feb  6 20:22:46 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Sun, 6 Feb 2011 09:22:46 -0800
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
 for Windows in .NET
In-Reply-To: <AANLkTinLND_RPkBG-5jdCoBKUa+SMw+Pe1OVo8igf0mz@mail.gmail.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
	<16100.1296638658@ccs.covici.com>
	<1296658879889-5985076.post@n2.nabble.com>
	<6549.1296659705@ccs.covici.com>
	<AANLkTinCVZqYRxd0ZcQLj_kj42fYHDqUsEocEqW3oaph@mail.gmail.com>
	<AANLkTinNjGKvGXKvTh1dMgZVkwdimkPoUZPJKu7233-T@mail.gmail.com>
	<AANLkTikP4H38vmvvVTCHRJdRa=60f4q4RLL+A3GiznNO@mail.gmail.com>
	<AANLkTi=skc8OGLqkiTULMN-pSL9SNgwK-YrcLq6N-GXp@mail.gmail.com>
	<AANLkTinLND_RPkBG-5jdCoBKUa+SMw+Pe1OVo8igf0mz@mail.gmail.com>
Message-ID: <AANLkTikUmZ8zhtZb35DYBUkHNazevskN5g-+7MPreCM8@mail.gmail.com>

Also are you using the public installer or did you compile yourself?
Looking at the error log there and why mod_portaudio doesn't load the issue
is the fact it is unable to find any input device on your system.  Do you
have a microphone or other input device that shows up in the windows sound
control panel?

~Mitch

On Sat, Feb 5, 2011 at 8:36 PM, Yitzchok <adminjew at gmail.com> wrote:

>
> ----freeswitch.log----
> 2011-02-05 23:33:48.934072 [ERR] mod_PortAudio.c:1191 Cannot find an input
> device
> 2011-02-05 23:33:48.934072 [WARNING] mod_PortAudio.c:1199 Switching to
> default output device
> 2011-02-05 23:33:48.934072 [WARNING] mod_PortAudio.c:1208 Invalid ring
> device configured using output device
> 2011-02-05 23:33:48.934072 [CRIT] switch_loadable_module.c:928 Error
> Loading module C:\Program Files (x86)\FSClient\mod\mod_portaudio.dll
> **Module load routine returned an error**
>
>
>
> --
> Yitzchok
>
>
>
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From infos at madovsky.org  Sun Feb  6 20:29:06 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 6 Feb 2011 12:29:06 -0500
Subject: [Freeswitch-users] play_and_get_digits question
Message-ID: <D95547C5033145B681FDF3E0059D78CA@e1705>

I have this lines in my dialplan

       <action application="play_and_get_digits" data="4 4 3 5000 # phrase:wait_foo phrase:bad_foo foo ${foo}"/>
       <action application="transfer" data="otherext XML features"/>

if I don't press any DTMF, after 4 loops the dialplan continues,
but I thought that it was automatically hangup in case of no digits bad digits...

Did I forget something ?

Thanks

Franck

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From jmesquita at freeswitch.org  Sun Feb  6 20:42:27 2011
From: jmesquita at freeswitch.org (Jmesquita@freeswitch.org)
Date: Sun, 6 Feb 2011 12:42:27 -0500
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
	for Windows in .NET
In-Reply-To: <4D4ED2A3.1000005@coppice.org>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
	<AANLkTimghBGjRNXA4GAVXvGFU0KQX5cEqQMREO5U=DvW@mail.gmail.com>
	<4D4ED2A3.1000005@coppice.org>
Message-ID: <18BD72CA-1B7C-454E-A8B9-3137398CF05D@freeswitch.org>

Steve, the reason why I was developing FSComm is purely out of passion for development. That passion includes me using the software I produce afterwards. As you might have imagined, I use a Mac. I have complained a lot about all the software that claim to be multi platform and that do a lousy job using the features that to me are basic on a soft phone. That includes using it on a speaker on a Mac which is one of the most widely used features for business guys with Skype. Most probably it relates closely to it's success.

That feature has stopped me from using regular soft phones for a long time now on any my devices and being stuck with hard phones just because i cant manage to be attached to the Mac with a corded headset. Granted, there are bluetooth options that i could have bought and i chose not to.

That being said, I have researched for AEC for quite some time now and even though i am don't consider myself too much of a lame tech guy, implementing the theory I have found is not up to my knowledge and thats is precisely why i reached out to the community.

I don't believe we need an AEC technology that is absolutely insanely great like what Skype does, my goal is to find one technology/implementation that does a good job on a small office room. I couldn't get that with the speexdsp which most certainly points that i probably don't know hoe to use it properly.

Sorry for the long reply, but i thought the subject needed it.

Regards,

JM

Sent from my iPad

On Feb 6, 2011, at 11:56 AM, Steve Underwood <steveu at coppice.org> wrote:

> On 02/02/2011 09:48 PM, Jo?o Mesquita wrote:
>> Mitch, you beat me to it...
>> 
>> I think it might be time for me to get back to FSComm and drop being 
>> picky about the AEC thing? Sad but true, I think that the only way for 
>> me to draw attention of someone who knows about this stuff is actually 
>> doing the rest like you did. Great initiative. I truly congratulate 
>> you from pulling something I couldn't so far.
>> 
> I am puzzled why the AEC issue would ever have affected your development 
> of FSComm. Although a headset or handset needs EC for really good 
> telephone, the lack fo AEC is only a killer for speakerphone usage.
> 
> On this topic, a little experimentation suggests the following...
> 
> Skype appears to do good AEC on a PC, but when you listen carefully is 
> seems more like they avoid proper AEC, and use the kind of adaptive gain 
> juggling the DSP Group pushed heavily in the early 90s. The final result 
> is very acceptable, though, and avoids the problems to non-matching mic 
> and speaker sampling rates that you get with many PC sound cards.
> 
> Skype on a Symbian handset and a Mac seems to do proper echo 
> cancellation, and gives results a bit better than on a PC. The limited 
> range of hardware used in these platforms makes AEC more of a practical 
> option.
> 
> If anyone has further information about the exact schemes used by Skype, 
> I would really like to hear from them. Skype is used on so many 
> machines, that whatever strategy they have found workable across a wide 
> range of hardware should be a safe bet for us to use.
> 
> 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 infos at madovsky.org  Sun Feb  6 20:43:38 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 6 Feb 2011 12:43:38 -0500
Subject: [Freeswitch-users] play_and_get_digits question
Message-ID: <EB3EE31EAE274C0D8E5AE2C073CB28C4@e1705>

Ok I think I misunderstood the concept of var where digits should be put in.
it's with this var I need to ckeck after this application.

self forum thank you ;)
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, February 06, 2011 12:29 PM
  Subject: play_and_get_digits question


  I have this lines in my dialplan

         <action application="play_and_get_digits" data="4 4 3 5000 # phrase:wait_foo phrase:bad_foo foo ${foo}"/>
         <action application="transfer" data="otherext XML features"/>

  if I don't press any DTMF, after 4 loops the dialplan continues,
  but I thought that it was automatically hangup in case of no digits bad digits...

  Did I forget something ?

  Thanks

  Franck

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From infos at madovsky.org  Sun Feb  6 21:01:16 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 6 Feb 2011 13:01:16 -0500
Subject: [Freeswitch-users] play_and_get_digits question
Message-ID: <8A02ED1AC7914B4BAD4EEF5D910D28DE@e1705>

or maybe this channel var 
read_result
is the result of play_and_get_digits ?
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, February 06, 2011 12:43 PM
  Subject: Re: play_and_get_digits question


  Ok I think I misunderstood the concept of var where digits should be put in.
  it's with this var I need to ckeck after this application.

  self forum thank you ;)
    ----- Original Message ----- 
    From: Madovsky 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Sunday, February 06, 2011 12:29 PM
    Subject: play_and_get_digits question


    I have this lines in my dialplan

           <action application="play_and_get_digits" data="4 4 3 5000 # phrase:wait_foo phrase:bad_foo foo ${foo}"/>
           <action application="transfer" data="otherext XML features"/>

    if I don't press any DTMF, after 4 loops the dialplan continues,
    but I thought that it was automatically hangup in case of no digits bad digits...

    Did I forget something ?

    Thanks

    Franck

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From tayeb.meftah at gmail.com  Sun Feb  6 20:59:42 2011
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sun, 06 Feb 2011 18:59:42 +0100
Subject: [Freeswitch-users] Routing calls acording to the divertion header
Message-ID: <4D4EE18E.9070606@gmail.com>

guys,
i have a divertion header like this:
Diversion:<sip:213983200193 at 12.13.14.15;user=phone>;reason=unknown;counter=1 

and another like:
Diversion:<sip:15148001395 at 12.13.14.15;user=phone>;reason=unknown;counter=1
how can i diferentiate calls acording to the divertion header?
thanks

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



From david.ponzone at ipeva.fr  Sun Feb  6 22:03:30 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sun, 6 Feb 2011 20:03:30 +0100
Subject: [Freeswitch-users] Routing calls acording to the divertion
	header
In-Reply-To: <4D4EE18E.9070606@gmail.com>
References: <4D4EE18E.9070606@gmail.com>
Message-ID: <2987F7D9-1A0A-475C-AA7F-E1BF8CB0D02A@ipeva.fr>

I think we need more details because your question does not make sense to me ATM.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 06/02/2011 ? 18:59, Meftah Tayeb a ?crit :

> guys,
> i have a divertion header like this:
> Diversion:<sip:213983200193 at 12.13.14.15;user=phone>;reason=unknown;counter=1 
> 
> and another like:
> Diversion:<sip:15148001395 at 12.13.14.15;user=phone>;reason=unknown;counter=1
> how can i diferentiate calls acording to the divertion header?
> thanks
> 
> -- 
> Meftah Tayeb
> inum: +883510001288000
> phone: +13477595883
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sun Feb  6 22:18:13 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 6 Feb 2011 14:18:13 -0500
Subject: [Freeswitch-users] play_and_get_digits question
Message-ID: <EBED8C31D06F4A319D978371EE1CF24E@e1705>

sorry I think I really didn't undertand the play_and_get_digits concept.
is anyone can help ?

I just want to use this app to check a pin code, hangup if wrong and continue
dialplan if ok

Thanks
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, February 06, 2011 1:01 PM
  Subject: Re: play_and_get_digits question


  or maybe this channel var 
  read_result
  is the result of play_and_get_digits ?
    ----- Original Message ----- 
    From: Madovsky 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Sunday, February 06, 2011 12:43 PM
    Subject: Re: play_and_get_digits question


    Ok I think I misunderstood the concept of var where digits should be put in.
    it's with this var I need to ckeck after this application.

    self forum thank you ;)
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Sunday, February 06, 2011 12:29 PM
      Subject: play_and_get_digits question


      I have this lines in my dialplan

             <action application="play_and_get_digits" data="4 4 3 5000 # phrase:wait_foo phrase:bad_foo foo ${foo}"/>
             <action application="transfer" data="otherext XML features"/>

      if I don't press any DTMF, after 4 loops the dialplan continues,
      but I thought that it was automatically hangup in case of no digits bad digits...

      Did I forget something ?

      Thanks

      Franck

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From adminjew at gmail.com  Sun Feb  6 22:38:49 2011
From: adminjew at gmail.com (Yitzchok)
Date: Sun, 6 Feb 2011 14:38:49 -0500
Subject: [Freeswitch-users] FSClient Release - A FreeSWITCH SIP Client
 for Windows in .NET
In-Reply-To: <AANLkTikUmZ8zhtZb35DYBUkHNazevskN5g-+7MPreCM8@mail.gmail.com>
References: <AANLkTikb8k+c4XVB5x+s34qg68LnNMpZbpS+KpXRmTMR@mail.gmail.com>
	<16100.1296638658@ccs.covici.com>
	<1296658879889-5985076.post@n2.nabble.com>
	<6549.1296659705@ccs.covici.com>
	<AANLkTinCVZqYRxd0ZcQLj_kj42fYHDqUsEocEqW3oaph@mail.gmail.com>
	<AANLkTinNjGKvGXKvTh1dMgZVkwdimkPoUZPJKu7233-T@mail.gmail.com>
	<AANLkTikP4H38vmvvVTCHRJdRa=60f4q4RLL+A3GiznNO@mail.gmail.com>
	<AANLkTi=skc8OGLqkiTULMN-pSL9SNgwK-YrcLq6N-GXp@mail.gmail.com>
	<AANLkTinLND_RPkBG-5jdCoBKUa+SMw+Pe1OVo8igf0mz@mail.gmail.com>
	<AANLkTikUmZ8zhtZb35DYBUkHNazevskN5g-+7MPreCM8@mail.gmail.com>
Message-ID: <AANLkTinvkhqFc6GMnh7nttV11bm8Gbs4R4ZWmMcwv3AM@mail.gmail.com>

Mitch,

That might be it I tried on another computer and it worked.


Yitzchok


On Sun, Feb 6, 2011 at 12:22 PM, Mitch Capper <mitch.capper at gmail.com>wrote:

> Also are you using the public installer or did you compile yourself?
> Looking at the error log there and why mod_portaudio doesn't load the issue
> is the fact it is unable to find any input device on your system.  Do you
> have a microphone or other input device that shows up in the windows sound
> control panel?
>
> ~Mitch
>
>
> On Sat, Feb 5, 2011 at 8:36 PM, Yitzchok <adminjew at gmail.com> wrote:
>
>>
>> ----freeswitch.log----
>> 2011-02-05 23:33:48.934072 [ERR] mod_PortAudio.c:1191 Cannot find an input
>> device
>> 2011-02-05 23:33:48.934072 [WARNING] mod_PortAudio.c:1199 Switching to
>> default output device
>> 2011-02-05 23:33:48.934072 [WARNING] mod_PortAudio.c:1208 Invalid ring
>> device configured using output device
>> 2011-02-05 23:33:48.934072 [CRIT] switch_loadable_module.c:928 Error
>> Loading module C:\Program Files (x86)\FSClient\mod\mod_portaudio.dll
>> **Module load routine returned an error**
>>
>>
>>
>> --
>> Yitzchok
>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From potxoka at gmail.com  Sun Feb  6 23:08:43 2011
From: potxoka at gmail.com (Antonio)
Date: Sun, 06 Feb 2011 21:08:43 +0100
Subject: [Freeswitch-users] Freeswitch and mysql
In-Reply-To: <AANLkTinYebKhBXQGdwV02bi_kBPB4Vp7Q3iGCv9YRDD7@mail.gmail.com>
References: <4CA11C69.70302@gmail.com> <4D375040.1000209@gmail.com>
	<AANLkTinYebKhBXQGdwV02bi_kBPB4Vp7Q3iGCv9YRDD7@mail.gmail.com>
Message-ID: <4D4EFFCB.4070706@gmail.com>

El 19/01/11 22:16, Avi Marcus escribi?:
> Integrate freeswitch with mysql is vague. That itself could mean 
> something like having freeswitch store it's database in mysql, but 
> that doesn't seem to be what you want.
> You are asking how to use a database (any, not just mysql) to 
> configure the dialplan - users, extensions, conferences, whatever.
> 1) You can use lua to process the calls and have it query your database
> 2) If you just need simple sql queries, check the mod_odbc_query from 
> the git contrib. If you already understand the dialplan basics, then 
> this can easily let you query the database as part of that.
> 3) However, if you need more complicated things, then mod_xml_curl is 
> your friend - it lets you grab dynamicly generated XML files for each 
> call.
> I myself use php to query a mysql database for how much to charge for 
> the call, a custom LCR implementation, etc.
> I posted the basic classes to github a while ago: 
> https://github.com/avimar/FreeSWITCH-mod_xml-with-PHP
> Also, intralanman wrote a very modular, all inclusive xml_curl 
> implementation in php - which if you understand it (I didn't know it 
> existed) should be really helpful. You can find that in the git 
> contrib also in: intralanman/PHP/fs_curl
>
> -Avi Marcus
>
> On Wed, Jan 19, 2011 at 10:57 PM, Antonio <potxoka at gmail.com 
> <mailto:potxoka at gmail.com>> wrote:
>
>     El 28/09/10 0:36, Antonio escribi?:
>>     Hello,
>>
>>     I asked and more I searched, I found nothing, how to integrate
>>     FreeSwitch with mysql. Some time ago I found Asterisk-realtime
>>     and wanted to know if there is something similar in FreeSwitch.
>>     Is to avoid double configurations, for example if there is a
>>     conference room that can be accessed from multiple servers, etc.
>>     I'm new in FreeSwitch and I have much knowledge ;-)
>>
>>     Does anyone have any url or book on how to do? Lua?. Thanks.
>>
>>     Greetings 
>     Hello,
>
>     Thanks, is that having to generate hundreds of xml files with the
>     extensions can be very stressful, so we had planned to use php +
>     mysql to configureextensions. Buy the book FreeSwitch and I have
>     yet to start reading and see if I can find a solution to this
>     problem, also tend to read the wiki, but being new to FreeSwitch
>     not quite understand certain concepts :-(. Thanks for the aid; -).
>
>     Greetings
>
Hello

the need I have is that of access to voicemail, because the 
registrar,location and others, is on another computer. FreeSwitch only 
what I have to use gateway, conferences (I'm looking at the issue with 
the wiki FreeSwitch, Javascript or Python) and voicemail. I can access 
the voicemail without having to configure extensions?. Looking at the 
module XML_CURL I understood something you can configure the voicemail 
so it, is not yet clarified me very well even with FreeSwitch: '(. Thanks.

Greetings
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From potxoka at gmail.com  Sun Feb  6 23:12:15 2011
From: potxoka at gmail.com (Antonio)
Date: Sun, 06 Feb 2011 21:12:15 +0100
Subject: [Freeswitch-users] Freeswitch and mysql
In-Reply-To: <4D4EFFCB.4070706@gmail.com>
References: <4CA11C69.70302@gmail.com> <4D375040.1000209@gmail.com>
	<AANLkTinYebKhBXQGdwV02bi_kBPB4Vp7Q3iGCv9YRDD7@mail.gmail.com>
	<4D4EFFCB.4070706@gmail.com>
Message-ID: <4D4F009F.9010109@gmail.com>

Hello

Looking Mod_Voicemail (http://wiki.freeswitch.org/wiki/Mod_voicemail), I 
see that could be configured with python, javascript or similar:-D. 
Every day I like best FreeSwitch. Thanks

REgards
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From sjmudd at pobox.com  Mon Feb  7 02:22:36 2011
From: sjmudd at pobox.com (Simon J Mudd)
Date: Mon, 7 Feb 2011 00:22:36 +0100
Subject: [Freeswitch-users] Confusing SIP auth failure logging message?
Message-ID: <20110206232236.GA10501@mad06.wl0.org>

I've been looking at trying to configure tighter controls for extensions that register.
Doing so made me trigger this error message (adjusted slightly):

2011-02-07 00:07:51.343303 [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1000 at sip.example.com] from ip 192.168.4.99
2011-02-07 00:07:51.343303 [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1000 at sip.example.com] from ip 192.168.4.99

Looking at the message it is not clear if the SIP authentication has succeeded or failed.
Judging by the code it seems this is meant to represent a SIP auth failure. If so should
the code not be patched as shown?

diff --git a/src/mod/endpoints/mod_sofia/sofia_reg.c b/src/mod/endpoints/mod_sofia/sofia_reg.c
index 631cbdb..e42c777 100644
--- a/src/mod/endpoints/mod_sofia/sofia_reg.c
+++ b/src/mod/endpoints/mod_sofia/sofia_reg.c
@@ -1244,7 +1244,7 @@ uint8_t sofia_reg_handle_register(nua_t *nua, sofia_profile_t *profile, nua_hand
                }
                /* Log line added to support Fail2Ban */
                if (sofia_test_pflag(profile, PFLAG_LOG_AUTH_FAIL)) {
-                       switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "SIP auth challenge (%s) on sofia profile '%s' "
+                       switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "SIP auth failure (%s) on sofia profile '%s' "
                                                          "for [%s@%s] from ip %s\n", (regtype == REG_INVITE) ? "INVITE" : "REGISTER", 
                                                          profile->name, to_user, to_host, network_ip);
                }

Thanks,

Simon


From infos at madovsky.org  Mon Feb  7 04:05:21 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 6 Feb 2011 20:05:21 -0500
Subject: [Freeswitch-users] close all conference from fs_cli
Message-ID: <313DB3017F77477FBC815E78800254C3@e1705>

Is there a way to close/hangup all active conferences 
from CLI ?

Thanks
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From david.ponzone at ipeva.fr  Mon Feb  7 05:22:31 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 7 Feb 2011 03:22:31 +0100
Subject: [Freeswitch-users] Freeswitch as a b2bua
In-Reply-To: <AANLkTi=DF=5w=RJAVneskBe5=aALYJcye6H6ue0dMkB4@mail.gmail.com>
References: <AANLkTi=DF=5w=RJAVneskBe5=aALYJcye6H6ue0dMkB4@mail.gmail.com>
Message-ID: <5691DFD2-55B3-4F8D-BB3D-E1F1F21BEFAE@ipeva.fr>

Yes, FreeSWITCH can do this.
The config will be quite simple, particularly if you dont want it to do anything special, just route calls.
It's always an issue to talk about CPS, but with a such host and given what I heard around, you should be able to reach 200 cps.
The 64bits is stable, it's even the recommended one (on CentOs).

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 06/02/2011 ? 14:58, Maciej Bylica a ?crit :

> Hi,
> 
> I am thinking about the way to have b2bua with full rtp proxy functionalities.
> The one wayout is to incorporate Freeswitch into the sip call flow.
> My scenario is as follows: Operator_1 ---> OpenSIPS ----> Operator_2
> Of course Opensips has B2b functionality but it may be pretty tricky
> to have this done.
> The question is could I place FS just after Opensips box like following:
> Operator_1 ---> OpenSIPS ----> Freewsitch ----> Operator_2 to achieve
> fully topology hiding (so no SIP/SDP  information of Operator_2 should
> pass to Operator_1 and vice versa).
> Could I achieve this by using FS?
> Is the configuration pretty straightforward?
> What more less will be cps for Quad-core E5620 with 8GB of RAM for
> aforementioned scenario (here i guess Opensips could be unbeatable).
> Is 64bit installation a stable one?
> 
> Thanks in advance,
> 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

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From gustavo.espeche at upper-soft.com  Mon Feb  7 06:11:50 2011
From: gustavo.espeche at upper-soft.com (Gustavo Espeche)
Date: Mon, 07 Feb 2011 00:11:50 -0300
Subject: [Freeswitch-users] mod_h323 error
Message-ID: <1297048310.2371.6.camel@gustavo-laptop>

Hello we are having a error with mod_h323

2011-02-06 22:04:06.999851 [ERR] mod_h323.cpp:2150 AUDIO RTP REPORTS
ERROR: [Missing local port]

when this error appear not more calls connect.
someone have this error with mod_h323.
regards.


Gustavo Espeche
www.easyipcall.com




From u2nsam at gmail.com  Mon Feb  7 07:05:06 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 7 Feb 2011 09:35:06 +0530
Subject: [Freeswitch-users] blocking 183 w/o sdp
In-Reply-To: <AANLkTikc1prUQweEsbHt94G6AUtPAkHrzf9KB2Dc3QEE@mail.gmail.com>
References: <AANLkTinH1s+Uy-EPvX9oV-x+aXYkyzUEBSYm+hQ+-+K-@mail.gmail.com>
	<AANLkTikJpEPvcQu2reAKXk1wbW7fV4myvXghBCuu2QHt@mail.gmail.com>
	<AANLkTikuQFC2N3CYzreTvCdk8-28MYGUbrP=T83SfPtm@mail.gmail.com>
	<AANLkTinyV-kAWCOiuOVL5jmnL7f9ck00=i_giHY_XGd_@mail.gmail.com>
	<AANLkTik6dUTrSLY+FYCBWQ2gsNQViCK4GW2-UnQPYmE=@mail.gmail.com>
	<AANLkTikc1prUQweEsbHt94G6AUtPAkHrzf9KB2Dc3QEE@mail.gmail.com>
Message-ID: <AANLkTimp8P+o6Ow4tiv_iXJqZ11sfL24uD7k8Hwd3h7h@mail.gmail.com>

Anthony it worked with FS 1.0.7 but FS crashes every 2 hours.

Regds
Sam

On Mon, Jan 31, 2011 at 10:03 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> If it does not work for you, your version of FreeSWITCH may be too old
> for this particular feature.
> Did you try with the latest release snapshot?
>
>
> On Sun, Jan 30, 2011 at 10:00 PM, Sam <u2nsam at gmail.com> wrote:
> > Hi,
> >
> > After using , <action application="bridge"
> > data="{sip_contact_user=${contact},sip_ignore_183nosdp=true}sofia/sbc/$1@
> ${dialed_domain}"/>
> >
> > the 183 without udp is not blocked/ignored .
> >
> > Below are the traces to visualize:
> > 192.168.2.98 is provider
> > 192.168.2.16 is FS
> >
> >
> > U 192.168.2.98:5060 -> 192.168.2.16:5060
> > SIP/2.0 183 Session Progress.
> > Via: SIP/2.0/UDP 192.168.2.16;rport;branch=z9hG4bKjQBQg7vy0y2SF.
> > To: <sip:599261244747199 at 192.168.2.16>;tag=3505434022-138257.
> > From: "0280910101" <sip:0280910101 at 192.168.2.16>;tag=51SjQQQUX14QF.
> > Call-ID: 6ade3e61-a78e-122e-9698-00137256e1a2.
> > CSeq: 7886492 INVITE.
> > Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
> > SUBSCRIBE, PRACK, UPDATE.
> > Contact: <sip:599261244747199 at 192.168.2.98:5060>.
> > Call-Info:
> > <sip:192.168.2.98>;method="NOTIFY;Event=telephone-event;Duration=1000".
> > Content-Length: 0.
> > .
> >
> >
> > U 192.168.2.16:5060 -> 192.168.2.6:5060
> > SIP/2.0 180 Ringing.
> > Via: SIP/2.0/UDP 192.168.2.6;branch=z9hG4bK3fb.51f5e6e1.0.
> > Via: SIP/2.0/UDP
> > 192.168.2.158:5060
> ;received=192.168.2.158;rport=5060;branch=z9hG4bK-f33ff5a-631d780c-50647c36.
> > Record-Route:
> >
> <sip:192.168.2.6;lr=on;ftag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a>.
> > From: "0280910101"
> > <sip:0280910101 at 192.168.2.158
> >;tag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a.
> > To: <sip:61244747199 at 192.168.2.6>;tag=3F70K1Nm3Frjr.
> > Call-ID: 100f6c88-9e3599cb-13c4-50029-f33ff5a-631c036b-f33ff5a.
> > CSeq: 1 INVITE.
> > Contact: <sip:61244747199 at 192.168.2.16:5060;transport=udp>.
> > User-Agent:  SBC.
> > Accept: application/sdp.
> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER,
> > REFER, NOTIFY.
> > Supported: timer, precondition, path, replaces.
> > Allow-Events: talk, refer.
> > Content-Length: 0.
> > Remote-Party-ID: "599261244747199"
> > <sip:599261244747199 at 192.168.2.16>;party=calling;privacy=off;screen=no.
> > .
> >
> >
> > U 192.168.2.98:5060 -> 192.168.2.16:5060
> > SIP/2.0 180 Ringing.
> > Via: SIP/2.0/UDP 192.168.2.16;rport;branch=z9hG4bKjQBQg7vy0y2SF.
> > To: <sip:599261244747199 at 192.168.2.98>;tag=3505434022-138257.
> > From: "0280910101" <sip:0280910101 at 192.168.2.16>;tag=51SjQQQUX14QF.
> > Call-ID: 6ade3e61-a78e-122e-9698-00137256e1a2.
> > CSeq: 7886492 INVITE.
> > Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
> > SUBSCRIBE, PRACK, UPDATE.
> > Contact: <sip:599261244747199 at 192.168.2.98:5060>.
> > Call-Info:
> > <sip:192.168.2.98>;method="NOTIFY;Event=telephone-event;Duration=1000".
> > Content-Type: application/sdp.
> > Content-Length: 209.
> > .
> > v=0.
> > o=vsnl2 770 13521 IN IP4 192.168.2.98.
> > s=sip call.
> > c=IN IP4 115.113.121.99.
> > t=0 0.
> > m=audio 49034 RTP/AVP 18 101.
> > a=rtpmap:101 telephone-event/8000.
> > a=fmtp:101 0-11.
> > a=ptime:20.
> > a=rtpmap:18 G729/8000/1.
> >
> >
> > U 192.168.2.16:5060 -> 192.168.2.6:5060
> > SIP/2.0 183 Session Progress.
> > Via: SIP/2.0/UDP 192.168.2.6;branch=z9hG4bK3fb.51f5e6e1.0.
> > Via: SIP/2.0/UDP
> > 192.168.2.158:5060
> ;received=192.168.2.158;rport=5060;branch=z9hG4bK-f33ff5a-631d780c-50647c36.
> > Record-Route:
> >
> <sip:192.168.2.6;lr=on;ftag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a>.
> > From: "0280910101"
> > <sip:0280910101 at 192.168.2.158
> >;tag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a.
> > To: <sip:61244747199 at 192.168.2.6>;tag=3F70K1Nm3Frjr.
> > Call-ID: 100f6c88-9e3599cb-13c4-50029-f33ff5a-631c036b-f33ff5a.
> > CSeq: 1 INVITE.
> > Contact: <sip:61244747199 at 192.168.2.16:5060;transport=udp>.
> > User-Agent:  SBC.
> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER,
> > REFER, NOTIFY.
> > Supported: timer, precondition, path, replaces.
> > Allow-Events: talk, refer.
> > Content-Type: application/sdp.
> > Content-Disposition: session.
> > Content-Length: 212.
> > Remote-Party-ID: "599261244747199"
> > <sip:599261244747199 at 192.168.2.16>;party=calling;privacy=off;screen=no.
> > .
> > v=0.
> > o=SBC 1019267468 1019267469 IN IP4 192.168.2.16.
> > s=SBC.
> > c=IN IP4 192.168.2.16.
> > t=0 0.
> > m=audio 16922 RTP/AVP 18 101.
> > a=rtpmap:18 G729/8000/1.
> > a=rtpmap:101 telephone-event/8000.
> > a=fmtp:101 0-11.
> > a=ptime:20.
> >
> >
> >
> >
> > Regds
> > Sam
> >
> >
> >
> >
> >
> > On Sat, Jan 29, 2011 at 3:16 PM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >>
> >> Close. You can only have one set of {} brackets. You can separate
> multiple
> >> variables with a comma.
> >>
> >> <action application="bridge"
> >>
> data="{sip_contact_user=${contact},sip_ignore_183nosdp=true}sofia/sbc/$1@
> ${dialed_domain}"/>
> >>
> >> -Steve
> >>
> >>
> >> On 29 January 2011 04:29, Sam <u2nsam at gmail.com> wrote:
> >>>
> >>> Hi,
> >>>
> >>> So you say i need to put
> >>> <action application="bridge"
> >>>
> data="{sip_contact_user=${contact}}{sip_ignore_183nosdp=true}sofia/sbc/$1@
> ${dialed_domain}"/>
> >>>
> >>> Regds
> >>> Sam
> >>>
> >>>
> >>>
> >>>
> >>> On Fri, Jan 28, 2011 at 10:23 PM, Anthony Minessale
> >>> <anthony.minessale at gmail.com> wrote:
> >>>>
> >>>> you need sip_ignore_183nosdp=true set on the b leg not the a leg.
> >>>> Put it in the dial string in {}
> >>>>
> >>>> {sip_ignore_183nosdp=true}sofia/foo/foo at bar.com
> >>>>
> >>>>
> >>>> On Fri, Jan 28, 2011 at 12:41 AM, Sam <u2nsam at gmail.com> wrote:
> >>>> > Hi,
> >>>> >
> >>>> > how can i ignore 183 without sdp,
> >>>> > what happens is the provider sends 183 without sdp and by applying
> >>>> > "<action
> >>>> > application="set" data="sip_ignore_183nosdp=true"/>"  the FS sends
> 180
> >>>> > to
> >>>> > the leg a.
> >>>> > Here i want to block the 183 with SDP just like router as b2bua and
> >>>> > send
> >>>> > nothing to leg a, and when actual 183 with sdp comes it should send
> .
> >>>> >
> >>>> > Its because, providers are sending false signaling by sending 183
> >>>> > without
> >>>> > sdp,and it hampers while @ production,
> >>>> > Although by cisco sbc i have done this but i want to do it by FS,
> >>>> > Take a scenario, when call is send 183 without sdp for 10 secs and
> >>>> > then
> >>>> > followed by 183 with sdp ( actual signal),
> >>>> > but when some one dials invalid number it rings for 10 secs and then
> >>>> > gives
> >>>> > SIP cause 404, which is bad from the providers.
> >>>> > So this is the reason i want to block it.
> >>>> >
> >>>> > Most of the providers do this, the way out is blocking.
> >>>> >
> >>>> > I have got an advice from Tihomir  to do "execute_on_ring and parse
> >>>> > your 180
> >>>> > / 183 messages in search of SDP,
> >>>> > once you get 183 without SDP do not send it back to leg a and send
> >>>> > signal
> >>>> > only when you got 183 with sdp or 180 "
> >>>> > And this could be valid call flow.
> >>>> >
> >>>> > This happens in many cases where the provider is having nextone as a
> >>>> > sbc and
> >>>> > that too tier 1 !
> >>>> >
> >>>> > 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
> >>>> >
> >>>> >
> >>>>
> >>>>
> >>>>
> >>>> --
> >>>> Anthony Minessale II
> >>>>
> >>>> FreeSWITCH http://www.freeswitch.org/
> >>>> ClueCon http://www.cluecon.com/
> >>>> Twitter: http://twitter.com/FreeSWITCH_wire
> >>>>
> >>>> AIM: anthm
> >>>> MSN:anthony_minessale at hotmail.com
> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 david.ponzone at ipeva.fr  Mon Feb  7 09:55:29 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 7 Feb 2011 07:55:29 +0100
Subject: [Freeswitch-users] blocking 183 w/o sdp
In-Reply-To: <AANLkTimp8P+o6Ow4tiv_iXJqZ11sfL24uD7k8Hwd3h7h@mail.gmail.com>
References: <AANLkTinH1s+Uy-EPvX9oV-x+aXYkyzUEBSYm+hQ+-+K-@mail.gmail.com>
	<AANLkTikJpEPvcQu2reAKXk1wbW7fV4myvXghBCuu2QHt@mail.gmail.com>
	<AANLkTikuQFC2N3CYzreTvCdk8-28MYGUbrP=T83SfPtm@mail.gmail.com>
	<AANLkTinyV-kAWCOiuOVL5jmnL7f9ck00=i_giHY_XGd_@mail.gmail.com>
	<AANLkTik6dUTrSLY+FYCBWQ2gsNQViCK4GW2-UnQPYmE=@mail.gmail.com>
	<AANLkTikc1prUQweEsbHt94G6AUtPAkHrzf9KB2Dc3QEE@mail.gmail.com>
	<AANLkTimp8P+o6Ow4tiv_iXJqZ11sfL24uD7k8Hwd3h7h@mail.gmail.com>
Message-ID: <640FBBA5-E9D1-4BBB-BCD1-B6DA6BB72A02@ipeva.fr>

Try the latest git, and if the crashes are still occuring, provide the required traces to the dev team.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 07/02/2011 ? 05:05, Sam a ?crit :

> 
> Anthony it worked with FS 1.0.7 but FS crashes every 2 hours.
> 
> Regds
> Sam
> 
> On Mon, Jan 31, 2011 at 10:03 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> If it does not work for you, your version of FreeSWITCH may be too old
> for this particular feature.
> Did you try with the latest release snapshot?
> 
> 
> On Sun, Jan 30, 2011 at 10:00 PM, Sam <u2nsam at gmail.com> wrote:
> > Hi,
> >
> > After using , <action application="bridge"
> > data="{sip_contact_user=${contact},sip_ignore_183nosdp=true}sofia/sbc/$1@${dialed_domain}"/>
> >
> > the 183 without udp is not blocked/ignored .
> >
> > Below are the traces to visualize:
> > 192.168.2.98 is provider
> > 192.168.2.16 is FS
> >
> >
> > U 192.168.2.98:5060 -> 192.168.2.16:5060
> > SIP/2.0 183 Session Progress.
> > Via: SIP/2.0/UDP 192.168.2.16;rport;branch=z9hG4bKjQBQg7vy0y2SF.
> > To: <sip:599261244747199 at 192.168.2.16>;tag=3505434022-138257.
> > From: "0280910101" <sip:0280910101 at 192.168.2.16>;tag=51SjQQQUX14QF.
> > Call-ID: 6ade3e61-a78e-122e-9698-00137256e1a2.
> > CSeq: 7886492 INVITE.
> > Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
> > SUBSCRIBE, PRACK, UPDATE.
> > Contact: <sip:599261244747199 at 192.168.2.98:5060>.
> > Call-Info:
> > <sip:192.168.2.98>;method="NOTIFY;Event=telephone-event;Duration=1000".
> > Content-Length: 0.
> > .
> >
> >
> > U 192.168.2.16:5060 -> 192.168.2.6:5060
> > SIP/2.0 180 Ringing.
> > Via: SIP/2.0/UDP 192.168.2.6;branch=z9hG4bK3fb.51f5e6e1.0.
> > Via: SIP/2.0/UDP
> > 192.168.2.158:5060;received=192.168.2.158;rport=5060;branch=z9hG4bK-f33ff5a-631d780c-50647c36.
> > Record-Route:
> > <sip:192.168.2.6;lr=on;ftag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a>.
> > From: "0280910101"
> > <sip:0280910101 at 192.168.2.158>;tag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a.
> > To: <sip:61244747199 at 192.168.2.6>;tag=3F70K1Nm3Frjr.
> > Call-ID: 100f6c88-9e3599cb-13c4-50029-f33ff5a-631c036b-f33ff5a.
> > CSeq: 1 INVITE.
> > Contact: <sip:61244747199 at 192.168.2.16:5060;transport=udp>.
> > User-Agent:  SBC.
> > Accept: application/sdp.
> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> > REFER, NOTIFY.
> > Supported: timer, precondition, path, replaces.
> > Allow-Events: talk, refer.
> > Content-Length: 0.
> > Remote-Party-ID: "599261244747199"
> > <sip:599261244747199 at 192.168.2.16>;party=calling;privacy=off;screen=no.
> > .
> >
> >
> > U 192.168.2.98:5060 -> 192.168.2.16:5060
> > SIP/2.0 180 Ringing.
> > Via: SIP/2.0/UDP 192.168.2.16;rport;branch=z9hG4bKjQBQg7vy0y2SF.
> > To: <sip:599261244747199 at 192.168.2.98>;tag=3505434022-138257.
> > From: "0280910101" <sip:0280910101 at 192.168.2.16>;tag=51SjQQQUX14QF.
> > Call-ID: 6ade3e61-a78e-122e-9698-00137256e1a2.
> > CSeq: 7886492 INVITE.
> > Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
> > SUBSCRIBE, PRACK, UPDATE.
> > Contact: <sip:599261244747199 at 192.168.2.98:5060>.
> > Call-Info:
> > <sip:192.168.2.98>;method="NOTIFY;Event=telephone-event;Duration=1000".
> > Content-Type: application/sdp.
> > Content-Length: 209.
> > .
> > v=0.
> > o=vsnl2 770 13521 IN IP4 192.168.2.98.
> > s=sip call.
> > c=IN IP4 115.113.121.99.
> > t=0 0.
> > m=audio 49034 RTP/AVP 18 101.
> > a=rtpmap:101 telephone-event/8000.
> > a=fmtp:101 0-11.
> > a=ptime:20.
> > a=rtpmap:18 G729/8000/1.
> >
> >
> > U 192.168.2.16:5060 -> 192.168.2.6:5060
> > SIP/2.0 183 Session Progress.
> > Via: SIP/2.0/UDP 192.168.2.6;branch=z9hG4bK3fb.51f5e6e1.0.
> > Via: SIP/2.0/UDP
> > 192.168.2.158:5060;received=192.168.2.158;rport=5060;branch=z9hG4bK-f33ff5a-631d780c-50647c36.
> > Record-Route:
> > <sip:192.168.2.6;lr=on;ftag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a>.
> > From: "0280910101"
> > <sip:0280910101 at 192.168.2.158>;tag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a.
> > To: <sip:61244747199 at 192.168.2.6>;tag=3F70K1Nm3Frjr.
> > Call-ID: 100f6c88-9e3599cb-13c4-50029-f33ff5a-631c036b-f33ff5a.
> > CSeq: 1 INVITE.
> > Contact: <sip:61244747199 at 192.168.2.16:5060;transport=udp>.
> > User-Agent:  SBC.
> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> > REFER, NOTIFY.
> > Supported: timer, precondition, path, replaces.
> > Allow-Events: talk, refer.
> > Content-Type: application/sdp.
> > Content-Disposition: session.
> > Content-Length: 212.
> > Remote-Party-ID: "599261244747199"
> > <sip:599261244747199 at 192.168.2.16>;party=calling;privacy=off;screen=no.
> > .
> > v=0.
> > o=SBC 1019267468 1019267469 IN IP4 192.168.2.16.
> > s=SBC.
> > c=IN IP4 192.168.2.16.
> > t=0 0.
> > m=audio 16922 RTP/AVP 18 101.
> > a=rtpmap:18 G729/8000/1.
> > a=rtpmap:101 telephone-event/8000.
> > a=fmtp:101 0-11.
> > a=ptime:20.
> >
> >
> >
> >
> > Regds
> > Sam
> >
> >
> >
> >
> >
> > On Sat, Jan 29, 2011 at 3:16 PM, Steven Ayre <steveayre at gmail.com> wrote:
> >>
> >> Close. You can only have one set of {} brackets. You can separate multiple
> >> variables with a comma.
> >>
> >> <action application="bridge"
> >> data="{sip_contact_user=${contact},sip_ignore_183nosdp=true}sofia/sbc/$1@${dialed_domain}"/>
> >>
> >> -Steve
> >>
> >>
> >> On 29 January 2011 04:29, Sam <u2nsam at gmail.com> wrote:
> >>>
> >>> Hi,
> >>>
> >>> So you say i need to put
> >>> <action application="bridge"
> >>> data="{sip_contact_user=${contact}}{sip_ignore_183nosdp=true}sofia/sbc/$1@${dialed_domain}"/>
> >>>
> >>> Regds
> >>> Sam
> >>>
> >>>
> >>>
> >>>
> >>> On Fri, Jan 28, 2011 at 10:23 PM, Anthony Minessale
> >>> <anthony.minessale at gmail.com> wrote:
> >>>>
> >>>> you need sip_ignore_183nosdp=true set on the b leg not the a leg.
> >>>> Put it in the dial string in {}
> >>>>
> >>>> {sip_ignore_183nosdp=true}sofia/foo/foo at bar.com
> >>>>
> >>>>
> >>>> On Fri, Jan 28, 2011 at 12:41 AM, Sam <u2nsam at gmail.com> wrote:
> >>>> > Hi,
> >>>> >
> >>>> > how can i ignore 183 without sdp,
> >>>> > what happens is the provider sends 183 without sdp and by applying
> >>>> > "<action
> >>>> > application="set" data="sip_ignore_183nosdp=true"/>"  the FS sends 180
> >>>> > to
> >>>> > the leg a.
> >>>> > Here i want to block the 183 with SDP just like router as b2bua and
> >>>> > send
> >>>> > nothing to leg a, and when actual 183 with sdp comes it should send .
> >>>> >
> >>>> > Its because, providers are sending false signaling by sending 183
> >>>> > without
> >>>> > sdp,and it hampers while @ production,
> >>>> > Although by cisco sbc i have done this but i want to do it by FS,
> >>>> > Take a scenario, when call is send 183 without sdp for 10 secs and
> >>>> > then
> >>>> > followed by 183 with sdp ( actual signal),
> >>>> > but when some one dials invalid number it rings for 10 secs and then
> >>>> > gives
> >>>> > SIP cause 404, which is bad from the providers.
> >>>> > So this is the reason i want to block it.
> >>>> >
> >>>> > Most of the providers do this, the way out is blocking.
> >>>> >
> >>>> > I have got an advice from Tihomir  to do "execute_on_ring and parse
> >>>> > your 180
> >>>> > / 183 messages in search of SDP,
> >>>> > once you get 183 without SDP do not send it back to leg a and send
> >>>> > signal
> >>>> > only when you got 183 with sdp or 180 "
> >>>> > And this could be valid call flow.
> >>>> >
> >>>> > This happens in many cases where the provider is having nextone as a
> >>>> > sbc and
> >>>> > that too tier 1 !
> >>>> >
> >>>> > 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
> >>>> >
> >>>> >
> >>>>
> >>>>
> >>>>
> >>>> --
> >>>> Anthony Minessale II
> >>>>
> >>>> FreeSWITCH http://www.freeswitch.org/
> >>>> ClueCon http://www.cluecon.com/
> >>>> Twitter: http://twitter.com/FreeSWITCH_wire
> >>>>
> >>>> AIM: anthm
> >>>> MSN:anthony_minessale at hotmail.com
> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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
> >
> >
> 
> 
> 
> --
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 sjmudd at pobox.com  Mon Feb  7 10:31:24 2011
From: sjmudd at pobox.com (Simon J Mudd)
Date: Mon, 7 Feb 2011 08:31:24 +0100
Subject: [Freeswitch-users] Confusing SIP auth failure logging message?
In-Reply-To: <20110206232236.GA10501@mad06.wl0.org>
References: <20110206232236.GA10501@mad06.wl0.org>
Message-ID: <20110207073124.GA5255@mad06.wl0.org>

On Mon, Feb 07, 2011 at 12:22:36AM +0100, Simon J Mudd wrote:
> I've been looking at trying to configure tighter controls for extensions that register.

Looking at http://wiki.freeswitch.org/wiki/Acl I see the comment lower down:

sip_profiles

... Should you want to protect your FreeSWITCH installation from being contacted by some IP addresses, you will need to setup some firewall rules. To protect your installation, you can look at QoS.

I'm confused. I understand that a firewall can be configured to drop/allow certain packages but given that FreeSWITCH does have acls it seems unusual to me that you
can do this directly in FreeSWITCH.

That is I have an Asterisk configuration which I am trying to migrate from and can easily configure in sip.conf:

[1000]
username=1000
type=friend
secret=1234567890
context=xxxxxx
host=dynamic
registersip=yes
deny=0.0.0.0/0.0.0.0
permit=88.100.50.0/255.255.255.0  -- this is not a real network range but you get the idea.
nat=yes
call-limit=1
...

This specifies a user for registration who:
(1) must provide a password
(2) can only register from the given network range
(3) is only allowed to make 1 call at a time

Basically I want to mimic this functionality.

I'm assuming that FreeSWITCH acls would be the way to do this. The
examples on the wiki don't seem to suggest this is possible.
Could someone help provide an example of if/how this would be done
in FreeSWITCH?

Thanks,

Simon


From steveayre at gmail.com  Mon Feb  7 10:49:46 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 7 Feb 2011 07:49:46 +0000
Subject: [Freeswitch-users] Confusing SIP auth failure logging message?
In-Reply-To: <20110207073124.GA5255@mad06.wl0.org>
References: <20110206232236.GA10501@mad06.wl0.org>
	<20110207073124.GA5255@mad06.wl0.org>
Message-ID: <88F0F936-6527-4379-88F2-412FAD2071A7@gmail.com>

You can do so in freeswitch but sometimes it's better to do it in the firewall. For example it uses far fewer resources to block someone in the firewall. It's also easier to block scanners such as friendly-scanner in the firewall.

Steve on iPhone

On 7 Feb 2011, at 07:31, Simon J Mudd <sjmudd at pobox.com> wrote:

> On Mon, Feb 07, 2011 at 12:22:36AM +0100, Simon J Mudd wrote:
>> I've been looking at trying to configure tighter controls for extensions that register.
> 
> Looking at http://wiki.freeswitch.org/wiki/Acl I see the comment lower down:
> 
> sip_profiles
> 
> ... Should you want to protect your FreeSWITCH installation from being contacted by some IP addresses, you will need to setup some firewall rules. To protect your installation, you can look at QoS.
> 
> I'm confused. I understand that a firewall can be configured to drop/allow certain packages but given that FreeSWITCH does have acls it seems unusual to me that you
> can do this directly in FreeSWITCH.
> 
> That is I have an Asterisk configuration which I am trying to migrate from and can easily configure in sip.conf:
> 
> [1000]
> username=1000
> type=friend
> secret=1234567890
> context=xxxxxx
> host=dynamic
> registersip=yes
> deny=0.0.0.0/0.0.0.0
> permit=88.100.50.0/255.255.255.0  -- this is not a real network range but you get the idea.
> nat=yes
> call-limit=1
> ...
> 
> This specifies a user for registration who:
> (1) must provide a password
> (2) can only register from the given network range
> (3) is only allowed to make 1 call at a time
> 
> Basically I want to mimic this functionality.
> 
> I'm assuming that FreeSWITCH acls would be the way to do this. The
> examples on the wiki don't seem to suggest this is possible.
> Could someone help provide an example of if/how this would be done
> in FreeSWITCH?
> 
> Thanks,
> 
> Simon
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb  7 10:58:24 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 7 Feb 2011 07:58:24 +0000
Subject: [Freeswitch-users] Confusing SIP auth failure logging message?
In-Reply-To: <20110207073124.GA5255@mad06.wl0.org>
References: <20110206232236.GA10501@mad06.wl0.org>
	<20110207073124.GA5255@mad06.wl0.org>
Message-ID: <5F00FAE4-4B64-46A4-901C-6C1BB55E650B@gmail.com>

For call limit, use http://wiki.freeswitch.org/Limit

To use a password on the user, just set a user

For the acl... Not sure. Cidr on the user will allow them to connect from there without a password. There may be a setting I don't know of, or you might want to call http://wiki.freeswitch.org/wiki/Acl#check_acl from the dialplan.

Steve on iPhone

On 7 Feb 2011, at 07:31, Simon J Mudd <sjmudd at pobox.com> wrote:

> On Mon, Feb 07, 2011 at 12:22:36AM +0100, Simon J Mudd wrote:
>> I've been looking at trying to configure tighter controls for extensions that register.
> 
> Looking at http://wiki.freeswitch.org/wiki/Acl I see the comment lower down:
> 
> sip_profiles
> 
> ... Should you want to protect your FreeSWITCH installation from being contacted by some IP addresses, you will need to setup some firewall rules. To protect your installation, you can look at QoS.
> 
> I'm confused. I understand that a firewall can be configured to drop/allow certain packages but given that FreeSWITCH does have acls it seems unusual to me that you
> can do this directly in FreeSWITCH.
> 
> That is I have an Asterisk configuration which I am trying to migrate from and can easily configure in sip.conf:
> 
> [1000]
> username=1000
> type=friend
> secret=1234567890
> context=xxxxxx
> host=dynamic
> registersip=yes
> deny=0.0.0.0/0.0.0.0
> permit=88.100.50.0/255.255.255.0  -- this is not a real network range but you get the idea.
> nat=yes
> call-limit=1
> ...
> 
> This specifies a user for registration who:
> (1) must provide a password
> (2) can only register from the given network range
> (3) is only allowed to make 1 call at a time
> 
> Basically I want to mimic this functionality.
> 
> I'm assuming that FreeSWITCH acls would be the way to do this. The
> examples on the wiki don't seem to suggest this is possible.
> Could someone help provide an example of if/how this would be done
> in FreeSWITCH?
> 
> Thanks,
> 
> Simon
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 david.ponzone at ipeva.fr  Mon Feb  7 11:04:01 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 7 Feb 2011 09:04:01 +0100
Subject: [Freeswitch-users] Confusing SIP auth failure logging message?
In-Reply-To: <88F0F936-6527-4379-88F2-412FAD2071A7@gmail.com>
References: <20110206232236.GA10501@mad06.wl0.org>
	<20110207073124.GA5255@mad06.wl0.org>
	<88F0F936-6527-4379-88F2-412FAD2071A7@gmail.com>
Message-ID: <550C876D-D224-448C-83AD-E29310B8B672@ipeva.fr>

Simon,

You can also add rate-limiting rules in iptables, to prevent SIP DoS (people sending you lots of INVITE).

On the other side, one reason not to use iptables for everything is to avoid sawing off the branch you are sitting on.

When both ACLs are possible, my personal rule would be:
-if changes are required everyday or by various people who are not iptables experts, use FreeSWITCH ACL
-if changes are done by an expert and/or are not frequent, go for iptables

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 07/02/2011 ? 08:49, Steven Ayre a ?crit :

> You can do so in freeswitch but sometimes it's better to do it in the firewall. For example it uses far fewer resources to block someone in the firewall. It's also easier to block scanners such as friendly-scanner in the firewall.
> 
> Steve on iPhone
> 
> On 7 Feb 2011, at 07:31, Simon J Mudd <sjmudd at pobox.com> wrote:
> 
>> On Mon, Feb 07, 2011 at 12:22:36AM +0100, Simon J Mudd wrote:
>>> I've been looking at trying to configure tighter controls for extensions that register.
>> 
>> Looking at http://wiki.freeswitch.org/wiki/Acl I see the comment lower down:
>> 
>> sip_profiles
>> 
>> ... Should you want to protect your FreeSWITCH installation from being contacted by some IP addresses, you will need to setup some firewall rules. To protect your installation, you can look at QoS.
>> 
>> I'm confused. I understand that a firewall can be configured to drop/allow certain packages but given that FreeSWITCH does have acls it seems unusual to me that you
>> can do this directly in FreeSWITCH.
>> 
>> That is I have an Asterisk configuration which I am trying to migrate from and can easily configure in sip.conf:
>> 
>> [1000]
>> username=1000
>> type=friend
>> secret=1234567890
>> context=xxxxxx
>> host=dynamic
>> registersip=yes
>> deny=0.0.0.0/0.0.0.0
>> permit=88.100.50.0/255.255.255.0  -- this is not a real network range but you get the idea.
>> nat=yes
>> call-limit=1
>> ...
>> 
>> This specifies a user for registration who:
>> (1) must provide a password
>> (2) can only register from the given network range
>> (3) is only allowed to make 1 call at a time
>> 
>> Basically I want to mimic this functionality.
>> 
>> I'm assuming that FreeSWITCH acls would be the way to do this. The
>> examples on the wiki don't seem to suggest this is possible.
>> Could someone help provide an example of if/how this would be done
>> in FreeSWITCH?
>> 
>> Thanks,
>> 
>> Simon
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From victor.chukalovskiy at utoronto.ca  Fri Feb  4 22:42:38 2011
From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy)
Date: Fri, 04 Feb 2011 14:42:38 -0500
Subject: [Freeswitch-users] Error installing mod_com_g729
In-Reply-To: <AANLkTikpKu-T77QNvAGQevDZSOsncyQAYmG6v9=_1OV6@mail.gmail.com>
References: <AANLkTikykPgaD0tErbMw=VOgfXqnwEwY4SzC_bNqSw-i@mail.gmail.com>	<AANLkTinSROc3ETqi8vbs89swryJa1+TNKa8WWDP4_O6H@mail.gmail.com>	<AANLkTinPmSAR=OiatFKEP0sRVA6_4=DQ7C5UdQcxsTfM@mail.gmail.com>	<AANLkTi=oNazw3U-nZ63Ro2oJh+Jgz_3O4_mrFmzS8VZA@mail.gmail.com>
	<AANLkTikpKu-T77QNvAGQevDZSOsncyQAYmG6v9=_1OV6@mail.gmail.com>
Message-ID: <4D4C56AE.10303@utoronto.ca>

Hello,

After purchasing a few licenses and installing the latest 
fsg729-191-installer
I'm getting the following error when trying to load the mod_com_g729:
> 2011-02-04 14:27:17.637046 [CRIT] switch_loadable_module.c:926 Error 
> Loading module /opt/fs/mod/mod_com_g729.so
> **Trying to load an out of date module, please rebuild the module.**

Also noticed that g729 installer ran with a couple errors:
> ./installer: line 62: ldconfig: command not found
> ./installer: line 49: useradd: command not found
  Any help or solution is much appreciated.

-Victor


From rhosyn at purplescarab.com  Sat Feb  5 22:14:43 2011
From: rhosyn at purplescarab.com (Rhosyn)
Date: Sat, 5 Feb 2011 19:14:43 +0000
Subject: [Freeswitch-users] Using mod_opal with MSN Messenger
In-Reply-To: <1296867966381-5994575.post@n2.nabble.com>
References: <1296867966381-5994575.post@n2.nabble.com>
Message-ID: <AANLkTikYZDib3t14odY6B6xb=VGNYx-VpEMVU0pDanRE@mail.gmail.com>

Hi Mazilo,

I think you're barking up the wrong tree.

MSN Messenger is (as I understand it from digging into it a few years ago) a
proprietary text-based protocol (see: e.g.
http://www.hypothetic.org/docs/msn/client/invitations.php  for some "reverse
engineered" details)

whereas H.323 (as supported by Opal) is an ITU Standard and is a binary
protocol.

There's a tiny bit of a historical protocol overlap in that both Messenger
(old, now defunct versions?) and H.323 clients (like MS Netmeeting)  have
used T.120 for app sharing but even then, AFAIK, they have never been
interoperable for any kind of call signalling.

To my knowledge, the only major  "chat" program with a large user base that
uses anything close to kind of open standard is gtalk (which uses XMPP)

Hth,

Rhosyn



On 5 February 2011 01:06, mazilo <Nabble at slickdeals.endjunk.com> wrote:

>
> Has anyone managed to configure mod_opal to register to MSN Messenger
> server
> to place/receive calls to/from any MSN Messenger users? I took a look at
> the
> conf/autoload_configs/opal.conf.xml file and don't know what to fill the
> gk-address, gk-identifier, and gk-interface. Anyone?
>
> -----
> 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/Using-mod-opal-with-MSN-Messenger-tp5994575p5994575.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 victor.chukalovskiy at utoronto.ca  Sun Feb  6 00:30:23 2011
From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy)
Date: Sat, 05 Feb 2011 16:30:23 -0500
Subject: [Freeswitch-users] Error installing mod com g729
Message-ID: <4D4DC16F.50500@utoronto.ca>

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From admin at blindi.net  Sun Feb  6 07:24:04 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Sun, 6 Feb 2011 05:24:04 +0100 (CET)
Subject: [Freeswitch-users] conferenceproblems after execute_application
Message-ID: <Pine.LNX.4.64.1102060503170.16105@suse.blindi.net>

Hi all, i have a working freeswitch git on ubuntu lucid 64bit.
I setup a conference. I add these line in my call-control-section:
<control action="execute_application" digits="3" data="playback 
/fssounds/helpconference.wav"/>
I press 3 the file will be played correctly.
After the playback i go back to the conference.
I talk to  others in the conference.
after the "execute_application" command
i have many delays 3 or 5 or more seconds.
My voice is very clearly. the people mean i.m a voicecomputer-).

I must exit the conference and i start again. my voice is normal.

is execute_application a problem?
can you help me please?
thanks.







---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From rhosyn at purplescarab.com  Sun Feb  6 14:59:41 2011
From: rhosyn at purplescarab.com (Rhosyn)
Date: Sun, 6 Feb 2011 11:59:41 +0000
Subject: [Freeswitch-users] Using mod_opal with MSN Messenger
In-Reply-To: <AANLkTikYZDib3t14odY6B6xb=VGNYx-VpEMVU0pDanRE@mail.gmail.com>
References: <1296867966381-5994575.post@n2.nabble.com>
	<AANLkTikYZDib3t14odY6B6xb=VGNYx-VpEMVU0pDanRE@mail.gmail.com>
Message-ID: <AANLkTikR6qZ5D=sSyJZqn-YAjHzUukbcAmb91coU+Rfc@mail.gmail.com>

Re-sending as for some reason the first "send" didn't seem arrive??

On 5 February 2011 19:14, Rhosyn <rhosyn at purplescarab.com> wrote:

> Hi Mazilo,
>
> I think you're barking up the wrong tree.
>
> MSN Messenger is (as I understand it from digging into it a few years ago)
> a proprietary text-based protocol (see: e.g.
> http://www.hypothetic.org/docs/msn/client/invitations.php  for some
> "reverse engineered" details)
>
> whereas H.323 (as supported by Opal) is an ITU Standard and is a binary
> protocol.
>
> There's a tiny bit of a historical protocol overlap in that both Messenger
> (old, now defunct versions?) and H.323 clients (like MS Netmeeting)  have
> used T.120 for app sharing but even then, AFAIK, they have never been
> interoperable for any kind of call signalling.
>
> To my knowledge, the only major  "chat" program with a large user base that
> uses anything close to kind of open standard is gtalk (which uses XMPP)
>
> Hth,
>
> Rhosyn
>
>
>
> On 5 February 2011 01:06, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
>>
>> Has anyone managed to configure mod_opal to register to MSN Messenger
>> server
>> to place/receive calls to/from any MSN Messenger users? I took a look at
>> the
>> conf/autoload_configs/opal.conf.xml file and don't know what to fill the
>> gk-address, gk-identifier, and gk-interface. Anyone?
>>
>> -----
>> 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/Using-mod-opal-with-MSN-Messenger-tp5994575p5994575.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 Duane.Gilbert at patlive.com  Sun Feb  6 22:06:22 2011
From: Duane.Gilbert at patlive.com (Duane Gilbert)
Date: Sun, 6 Feb 2011 14:06:22 -0500
Subject: [Freeswitch-users] Freeswitch and mysql
In-Reply-To: <4D4EC72B.7080408@gmail.com>
References: <4CA11C69.70302@gmail.com>
	<4D375040.1000209@gmail.com><AANLkTinYebKhBXQGdwV02bi_kBPB4Vp7Q3iGCv9YRDD7@mail.gmail.com>
	<4D4EC72B.7080408@gmail.com>
Message-ID: <BD87D3B1554F614190EB6E54CE95DDB60466EF3F@mail2.patlive.local>

This is interesting....Nice to know that we were doing something right...mod_xml_curl....

 

I did not get a chance to look at the php example...I will have a look when I have a free moment...

 

- Duane

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Antonio
Sent: Sunday, February 06, 2011 11:07 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Freeswitch and mysql

 

El 19/01/11 22:16, Avi Marcus escribi?: 

Integrate freeswitch with mysql is vague. That itself could mean something like having freeswitch store it's database in mysql, but that doesn't seem to be what you want. 

You are asking how to use a database (any, not just mysql) to configure the dialplan - users, extensions, conferences, whatever.

1) You can use lua to process the calls and have it query your database

2) If you just need simple sql queries, check the mod_odbc_query from the git contrib. If you already understand the dialplan basics, then this can easily let you query the database as part of that.

3) However, if you need more complicated things, then mod_xml_curl is your friend - it lets you grab dynamicly generated XML files for each call.

I myself use php to query a mysql database for how much to charge for the call, a custom LCR implementation, etc.

I posted the basic classes to github a while ago:  https://github.com/avimar/FreeSWITCH-mod_xml-with-PHP

Also, intralanman wrote a very modular, all inclusive xml_curl implementation in php - which if you understand it (I didn't know it existed) should be really helpful. You can find that in the git contrib also in:  intralanman/PHP/fs_curl

-Avi Marcus

 

Hi

Thanks for everyone's responses, are really helping me to really know to look for and how to document ;-). I had thought one thing and I have a question: Could use python for this cause?. I'll look at me while the modules that have told me;-). Thank you very much.


Regards

<action application="python" data="extension_user"/>

--------- extension_user.xml ---------
<include>
    <user id="1000" mailbox="1000">
        <params>
            <param name="password" value="1234"/>
            <param name="vm-password" value="1000"/>
        </params>
        <variables>
            <variable name="toll_allow" value="domestic,international,local"/>
            <variable name="accountcode" value="1000"/>
            <variable name="user_context" value="default"/>
            <variable name="effective_caller_id_name" value="NAME"/>
            <variable name="effective_caller_id_number" value="1000"/>
            <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
            <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
            <variable name="callgroup" value="techsupport"/>
        </variables>
    </user>
</include>
--------- extension_user.xml ---------

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From jason at jasonjgw.net  Mon Feb  7 04:19:40 2011
From: jason at jasonjgw.net (Jason White)
Date: Mon, 07 Feb 2011 12:19:40 +1100
Subject: [Freeswitch-users] Opus codec in FreeSWITCH
Message-ID: <87hbcgr68j.fsf@jdc.jasonjgw.net>

I have started testing the Opus 0.9.0 experimental codec in a recent
build of FreeSWITCH - thanks to Anthony Minessale for writing the
module, and thanks to William F. Acker for helping with the testing.

For those who aren't familiar with it, Opus is a combination of CELT and
SILK, which is currently under discussion within the IETF.

After building and loading the module, specify Opus-0.9.0 as your codec.
The versioning of the codec's name is a good idea; this should prevent
mutually incompatible versions from interacting with each other as
development of the codec proceeds.

My initial impressions are favourable; the audio quality is comparable
to CELT at 48 khz, based on my early listening experience.

I did notice some occasional pops in the audio, presumably attributable
to packet loss. SILK apparently has some error correction functionality
built in, but I don't know what its status is in the combined Opus
codec. If it's present, then it didn't smooth out the glitches, at least
in my environment.



From admin at blindi.net  Mon Feb  7 12:32:15 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Mon, 7 Feb 2011 10:32:15 +0100 (CET)
Subject: [Freeswitch-users] dialplanhelp  for callforwarding
In-Reply-To: <8A02ED1AC7914B4BAD4EEF5D910D28DE@e1705>
References: <8A02ED1AC7914B4BAD4EEF5D910D28DE@e1705>
Message-ID: <Pine.LNX.4.64.1102070934130.31747@suse.blindi.net>

Hi all, i like to replace my asterisk to fs. FS is very nice.
I hanging:
my problem is: i don.t find a Gotoif command from fs to replace, my 
asterisk dialplan.

my simply dialplan in asterisk is:

;;Callforwarding to a phonenumber from dialplan
;;from: thomas Hoellriegel blindi at gmx.net

;;maimenu for callforwarding
;;anfang is in english: beginn
[callforwarding-only]
exten => s,1,Answer()
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(callforwarding-anfang1&callforwarding-anfang2&callforwarding-anfang3&callforwarding-anfang4&callforwarding-anfang5&callforwarding-anfang6&callforwarding-anfang7&callforwarding-anfang8&mit-stern-exit)
exten => s,n,WaitExten
exten => *,1,Goto(adminmenu,s,1)

;;drop all user errors
exten => i,1,Goto(s,1)

;;menuselection
exten => 1,1,Goto(callforwarding-db,s,1)


exten => t,1,Playback(vm-goodbye)
exten => t,n,Hangup()

[callforwarding-db]
exten => s,1,Answer()
exten => s,n,Set(TIMEOUT(response)=10)

;;status: 
;;aus: (off)  goto: no number in database
;;fwan: (on)  goto: then number is found
exten => s,n,GotoIf($["${DB(CFtommystatus/0)}" != ""]?fwan:fwaus)
exten => s,n(fwan),Playback(all-callfw-is-on)
exten => s,n,Goto(s,30)

exten => s,n(fwaus),Playback(all-fw-ist-off)
exten => s,n,Goto(s,30)

exten => s,30,Background(callforwarding-main)
exten => s,n,WaitExten
exten => *,1,Goto(callforwarding-only,s,1)

exten => i,1,Goto(s,1)

;;nummer is: shortnumber for callforwarding
exten => 1,1,Goto(nummer1,nr1,1)
;;toggle
;;ein aus in english: on off
exten => 0,1,GotoIf($["${DB(CFtommystatus/0)}" != ""]?ein:aus)
exten => 0,n(ein),DBdeltree(CFtommystatus/0)
exten => 0,n,Playback(all-fw-ist-off)
exten => 0,n,Goto(callforwarding-only,s,1)

exten => 0,n(aus),Set(DB(CFtommystatus/0)=p)
exten => 0,n,Playback(alle-callfw-is-on)
exten => 0,n,Goto(callforwarding-only,s,1)

;;number 1
;;kurzwahl is: shortnumber 
[nummer1]
exten => nr1,1,Playback(kurzwahl)
exten => nr1,n,SayDigits(1)
exten => nr1,n,Background(callforwarding-opt)
exten => nr1,n,WaitExten

;;ja: say the existing number
;;falsch:
;;say no number in database found

exten => 1,1,GotoIf($["${DB(CFtommy1/0)}" != ""]?ja1:falsch1)
exten => 1,n(ja1),SayDigits(${DB(CFtommy1/0)})
exten => 1,n,Wait(0.2)
exten => 1,n,Goto(nr1,1)
exten => 1,n(falsch1),Playback(callforwarding-keine)
exten => 1,n,Wait(0.2)
exten => 1,n,Goto(nr1,1)

;;2 to add a number
exten => 2,1,Goto(nummer1-menu1,s,1)

;;aus: deactivate callforwarding
exten => 3,1,Goto(nummer1aus,s,1)
exten => 4,1,SayDigits(${CALLERID(num):})
exten => 4,n,Set(DB(CFtommy1/0)=${CALLERID(num):})
exten => 4,n,Playback(callforwarding-on)
exten => 4,n,Goto(callforwarding-only,s,1)

exten => *,1,Goto(callforwarding-only,s,1)

exten => i,1,Goto(nr1,1)

;;aus:  deactivate
[nummer1aus]
exten => s,1,DBdeltree(CFtommy1/0)
exten => s,n,Background(callforwarding-off)
exten => s,n,Goto(callforwarding-only,s,1)

;;menu for dtmf digits
[nummer1-menu1]
exten => s,1,Set(NR=)
exten => s,n,Background(bitte-callforwarding-nummer)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,WaitExten
exten => _X,1,Set(NR=${NR}${EXTEN})
exten => _X,n,Goto(s,3)
exten => *,1,Goto(s,1)

exten => #,1,Playback(callforwardings-number-is)
exten => #,n,SayDigits(${NR})

;;check the number
exten => #,n,Goto(nummer1-chk,${NR},1)

exten => t,1,Playback(vm-goodbye)
exten => t,n,Hangup()

;;check the number for no allowed numbers 
[nummer1-chk]
exten => i,1,NoOp(Undefined  Nummer ${INVALID_EXTEN}.)
exten => i,n,Answer()
exten => i,n,Playback(tt-somethingwrong)
exten => i,n,Playback(tt-monkeysintro)
exten => i,n,Hangup()

;;forbidden numbers:
;;neu is:  new
exten => _911.,1,Playback(nummer-falsch)
exten => _911.,n,Goto(callback-only-neu,s,1)

exten => _#,1,Playback(notallow)
exten => _#,n,Goto(nummer1-menu1,s,1)
exten => _X,1,Playback(notallow)
exten => _X,n,Goto(nummer1-menu1,s,1)
exten => _X.,1,Goto(nummer1-confirm,s,1)

;;confirmmenu
[nummer1-confirm]
exten => s,1,Background(ist-correct)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,WaitExten
exten => 1,1,Set(DB(CFtommy1/0)=${NR})
exten => 1,n,Playback(callforwarding-on)
exten => 1,n,Goto(callforwarding-only,s,1)

exten => 2,1,Goto(nummer1-menu1,s,1)

exten => #,1,Playback(errormenu)              ; "Thanks for trying the demo"
exten => #,n,Goto(nummer1-menu1,#,1)
exten => t,1,Goto(nummer1-menu1,#,1)
exten => i,1,Playback(conf-errormenu)          ; "That's not valid, try again"
exten => i,n,Goto(nummer1-menu1,#,1)
;;fw_outcontext
[fw_out]
include=trunk
include=international

[default]
;;forwarding extension in defaultcontext

;;read the db entry
exten => answert1,1,GotoIf($["${DB(CFtommy1/0)}" != ""]?yesfw1:nofw1)
exten => answert1,n(yesfw1),Dial(Local/${DB(CFtommy1/0)}@fw_out,,)
exten => answert1,n(nofw1),NoOp()

can you help me please? thankx.



---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
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From wstephen80 at gmail.com  Mon Feb  7 13:23:34 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Mon, 7 Feb 2011 11:23:34 +0100
Subject: [Freeswitch-users] Lua Session creation is blocking?
In-Reply-To: <AANLkTin=J4TqxY4D5OfriK6iD=1-OfLg32YXyyXLL61d@mail.gmail.com>
References: <AANLkTin21o-gZH6FrPU6BxFR+RqxBkN15=DDGD66iTX-@mail.gmail.com>
	<AANLkTikdGqM2QsZ19_4YpkDRTd+b5_wRK9NrEGkoqgnM@mail.gmail.com>
	<AANLkTinaixoUDTyR-5gPZPLSaO_W0eNvC491ApAUHg-O@mail.gmail.com>
	<AANLkTimU-=7LJXzRy1xTAf+Vz8B6nexLBORJzi_EO5a9@mail.gmail.com>
	<AANLkTikPNom1k3f=fbixbtcCGw=WGzC9UZj25mBBnvjj@mail.gmail.com>
	<AANLkTin=J4TqxY4D5OfriK6iD=1-OfLg32YXyyXLL61d@mail.gmail.com>
Message-ID: <AANLkTimE1AwXw=TFaN-82_v8=Ct1jbgHrcPXhkYYJXLR@mail.gmail.com>

I have started learning ESL (using C library) and I have found this
article: http://blog.godson.in/2010/12/use-of-returnringready-originate.html

So I have added in my dialstring the prefix:
*
*
*    {return_ring_ready=true}*
*
*
*and now the Lua script works as I want: the session creation returns also
when a 180 ringing (with no media) is received!*

Stephen

On Fri, Feb 4, 2011 at 10:57 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> The way you expect it to work is not the case.
>
> in embedded scripts when you create a new session you must wait for it
> to have media.
> The scripts are not asynchronous.  if you want asynchronous you 100% want
> ESL.
>
>
>
>
>
>
> On Fri, Feb 4, 2011 at 1:30 PM, Stephen Wilde <wstephen80 at gmail.com>
> wrote:
> > I have to port some applications on Freeswitch so I'm doing some
> preliminary
> > learning tests on Lua scripting.
> > The result that I would obtain is to have the full control of both
> inbound
> > and originated session and to have under control their state.
> > The script I'm running is simply: http://pastebin.freeswitch.org/15248
> > But its behaviour is not as I expect.
> >
> > Stephen
> >
> > On Fri, Feb 4, 2011 at 7:26 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> >>
> >>
> >> On Fri, Feb 4, 2011 at 10:19 AM, Stephen Wilde <wstephen80 at gmail.com>
> >> wrote:
> >>>
> >>> Ok, 18x will be perfect!
> >>> The problem is that when the inbound call is SIP and the originated is
> >>> ISDN, the originate doesn't return when a 180 ringing is received (that
> in
> >>> ISDN is an ALERT without in band info).
> >>> The originate returns when a 183 (ISDN = in band info available) or 200
> >>> (ISDN = CONNECT) is received.
> >>> Stephen
> >>>
> >>
> >> Could you tell us a bit more about what you're trying to accomplish? My
> >> guess is that there's a better way to do it. In most cases it is not
> >> necessary to create a new session inside of a script called from the
> >> dialplan.
> >> What is the application that you are working on? Who calls whom, etc.
> >> -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
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 steveayre at gmail.com  Mon Feb  7 13:49:09 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 7 Feb 2011 10:49:09 +0000
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <4D4DC16F.50500@utoronto.ca>
References: <4D4DC16F.50500@utoronto.ca>
Message-ID: <AANLkTimcheDx7mSDifAA6pBsWN+1Tr=CJbZN2zXJG3L4@mail.gmail.com>

 > ./installer: line 62: ldconfig: command not found
> ./installer: line 49: useradd: command not found

Those really should exist. Were you logged in as root?

Also, which version of FreeSWITCH are you running? Are you on the very
latest Git?

-Steve



On 5 February 2011 21:30, Victor Chukalovskiy <
victor.chukalovskiy at utoronto.ca> wrote:

>  Hello,
>
> After purchasing a few licenses and installing the latest
> fsg729-191-installer
> I'm getting the following error when trying to load the mod_com_g729:
> > 2011-02-04 14:27:17.637046 [CRIT] switch_loadable_module.c:926 Error
> Loading module /opt/fs/mod/mod_com_g729.so
> > **Trying to load an out of date module, please rebuild the module.**
>
> Also noticed that g729 installer ran with a couple errors:
> > ./installer: line 62: ldconfig: command not found
> > ./installer: line 49: useradd: command not found
>  Any help or solution is much appreciated.
>
> -Victor
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  7 18:09:18 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 7 Feb 2011 07:09:18 -0800 (PST)
Subject: [Freeswitch-users] Using mod_opal with MSN Messenger
In-Reply-To: <AANLkTikYZDib3t14odY6B6xb=VGNYx-VpEMVU0pDanRE@mail.gmail.com>
References: <1296867966381-5994575.post@n2.nabble.com>
	<AANLkTikYZDib3t14odY6B6xb=VGNYx-VpEMVU0pDanRE@mail.gmail.com>
Message-ID: <1297091358589-6000576.post@n2.nabble.com>


Rhosyn,

Thanks for your response. So, it looks like there is no way one can
configure an FS system using either mod_opal nor mod_h323 to place/receive
VoIP calls to/from any MSN/HotMail (Live) users. So, what is the use for
either mod_opal or mod_h323?

-----
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/Using-mod-opal-with-MSN-Messenger-tp5994575p6000576.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From francoisbarrouin at hotmail.fr  Mon Feb  7 17:50:18 2011
From: francoisbarrouin at hotmail.fr (Francois Barrouin)
Date: Mon, 7 Feb 2011 15:50:18 +0100
Subject: [Freeswitch-users] FS ESL in Adobe Flex
In-Reply-To: <BLU143-W3595CCF2A425489552F2F9A9EB0@phx.gbl>
References: <BLU143-W3595CCF2A425489552F2F9A9EB0@phx.gbl>
Message-ID: <BLU143-W13F9C31C28DF6BFBCF736EA9EB0@phx.gbl>






Hello,

I would like to use Adobe Flex to access FS event socket. 
I read in the (excellent!) ?FreeSWITCH 1.0.6?  book that FS ESL is based on swig and unfortunately Adobe Flex is not a language supported by this project.

I wonder if anyone has already developed a Flex ESL for FS? 

Thanks

Fran?ois 		 	   		  
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From victor.chukalovskiy at utoronto.ca  Mon Feb  7 18:12:37 2011
From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy)
Date: Mon, 07 Feb 2011 10:12:37 -0500
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <AANLkTimcheDx7mSDifAA6pBsWN+1Tr=CJbZN2zXJG3L4@mail.gmail.com>
References: <4D4DC16F.50500@utoronto.ca>
	<AANLkTimcheDx7mSDifAA6pBsWN+1Tr=CJbZN2zXJG3L4@mail.gmail.com>
Message-ID: <4D500BE5.4080503@utoronto.ca>

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From steveayre at gmail.com  Mon Feb  7 18:30:23 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 7 Feb 2011 15:30:23 +0000
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <4D500BE5.4080503@utoronto.ca>
References: <4D4DC16F.50500@utoronto.ca>
	<AANLkTimcheDx7mSDifAA6pBsWN+1Tr=CJbZN2zXJG3L4@mail.gmail.com>
	<4D500BE5.4080503@utoronto.ca>
Message-ID: <AANLkTinMuFsmvyfpPr9KW3ZOVeEDGRGbad4bS_2F9z1N@mail.gmail.com>

Try installing again, this time as root. If it still does it, try updating
FS.

-Steve



On 7 February 2011 15:12, Victor Chukalovskiy <
victor.chukalovskiy at utoronto.ca> wrote:

>  Hi Steve,
>
> I was running it as sudo. I guess something is absent from the path.
> The FREESwitch version is not the latest, but two month old GIT.
>
> -Victor
>
>
> On 07/02/11 05:49 AM, Steven Ayre wrote:
>
> > ./installer: line 62: ldconfig: command not found
> > ./installer: line 49: useradd: command not found
>
> Those really should exist. Were you logged in as root?
>
> Also, which version of FreeSWITCH are you running? Are you on the very
> latest Git?
>
> -Steve
>
>
>
> On 5 February 2011 21:30, Victor Chukalovskiy <
> victor.chukalovskiy at utoronto.ca> wrote:
>
>>  Hello,
>>
>> After purchasing a few licenses and installing the latest
>> fsg729-191-installer
>> I'm getting the following error when trying to load the mod_com_g729:
>> > 2011-02-04 14:27:17.637046 [CRIT] switch_loadable_module.c:926 Error
>> Loading module /opt/fs/mod/mod_com_g729.so
>> > **Trying to load an out of date module, please rebuild the module.**
>>
>> Also noticed that g729 installer ran with a couple errors:
>> > ./installer: line 62: ldconfig: command not found
>> > ./installer: line 49: useradd: command not found
>>  Any help or solution is much appreciated.
>>
>> -Victor
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> 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 msc at freeswitch.org  Mon Feb  7 18:36:16 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 7 Feb 2011 09:36:16 -0600
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <4D4DC16F.50500@utoronto.ca>
References: <4D4DC16F.50500@utoronto.ca>
Message-ID: <AANLkTimfzmpeR2N2sAf_QpAX0g3=BSRS0P-bPogh-Zbu@mail.gmail.com>

Any chance you can update to latest git? Your life will be easier. There
have been notable improvements in FS in the past few months.

-MC

On Sat, Feb 5, 2011 at 3:30 PM, Victor Chukalovskiy <
victor.chukalovskiy at utoronto.ca> wrote:

>  Hello,
>
> After purchasing a few licenses and installing the latest
> fsg729-191-installer
> I'm getting the following error when trying to load the mod_com_g729:
> > 2011-02-04 14:27:17.637046 [CRIT] switch_loadable_module.c:926 Error
> Loading module /opt/fs/mod/mod_com_g729.so
> > **Trying to load an out of date module, please rebuild the module.**
>
> Also noticed that g729 installer ran with a couple errors:
> > ./installer: line 62: ldconfig: command not found
> > ./installer: line 49: useradd: command not found
>  Any help or solution is much appreciated.
>
> -Victor
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  7 18:56:02 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 7 Feb 2011 16:56:02 +0100
Subject: [Freeswitch-users] Using mod_opal with MSN Messenger
Message-ID: <4BDB6016-806B-4855-9A42-0A8201517F0B@visionutveckling.se>

The use for these modules are to communicate with other h323 stacks/PBX'es. MSN is not one of them AFAIK.

/Peter

----- Reply message -----
Fr?n: "mazilo" <Nabble at slickdeals.endjunk.com>
Datum: m?n, feb 7, 2011 22:17
Rubrik: [Freeswitch-users] Using mod_opal with MSN Messenger
Till: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>


Rhosyn,

Thanks for your response. So, it looks like there is no way one can
configure an FS system using either mod_opal nor mod_h323 to place/receive
VoIP calls to/from any MSN/HotMail (Live) users. So, what is the use for
either mod_opal or mod_h323?

-----
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/Using-mod-opal-with-MSN-Messenger-tp5994575p6000576.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:4d500cac32762891562358!



From msc at freeswitch.org  Mon Feb  7 19:04:56 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 7 Feb 2011 10:04:56 -0600
Subject: [Freeswitch-users] Group confusions
In-Reply-To: <00f501cbc4a7$aaee5640$00cb02c0$@yahoo.com>
References: <AcvEp2ZUeoD+FSG/QdKLQRe6PdRMAA==>
	<00f501cbc4a7$aaee5640$00cb02c0$@yahoo.com>
Message-ID: <AANLkTinz0NuBbN_Q5gQnmuDkkicchhxWMg4XnH=19CJy@mail.gmail.com>

I may be that the value you are trying to delete isn't "1013 at 192.168.10.29".
To find out, go to fs_cli and do this:

group call:01 at 192.168.10.29

It will spit out all the sofia contacts in group 01. Most likely you have
something like this:
1013 at 192.168.10.29:12345;rinstance=aabbccddeeff0011;transport=udp

BTW, the value put into the group call when you dial 8101 is the same value
that you find by doing this at fs_cli:

sofia_contact 1013 at 192.168.10.29

Go try it and let us now.

-MC

P.S. - the 'group' API and dp apps you are using are the same - they come
from mod_db. They are *totally* different from the group_call API...

On Fri, Feb 4, 2011 at 2:11 PM, Lars Zeb <larclap at yahoo.com> wrote:

> I went to two individual extensions and registered them for group 01 by
> dialing 8101. I then tested that group by dialing 8201 - both extensions
> rang.
>
> I then went to the cli and entered the command:
>
> group delete:01:1013 at 192.168.10.29
>
> where 1013 is one of the two extensions I had joined into group 01 earlier
> and the url is the domain and fs address. However, when I dialed 8201 after
> entering this cli command, both extensions rang. I had expected that only
> one would ring.
>
> I am confused. Are these two different types of "groups"?
>
> Thanks, Lars
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  7 19:08:09 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 7 Feb 2011 10:08:09 -0600
Subject: [Freeswitch-users] ext-rtp-ip value behind NAT
In-Reply-To: <1296962373969-5996978.post@n2.nabble.com>
References: <1296962373969-5996978.post@n2.nabble.com>
Message-ID: <AANLkTi==qUfXH9q2x=is=xF+PCxoxFeM-7pJn3+2ZLg9@mail.gmail.com>

Is it just me or did you not actually list anything in 1. and 2. below?
-MC

On Sat, Feb 5, 2011 at 9:19 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
> With FreeSWITCH Version 1.0.head (git-2944364 2011-02-04 16-53-38 -0600) on
> my Seagate DockStar, I configured my jingle profile using the following two
> different options:
>
> 1.
> 2.
>
> With option #1, no incoming call and caller got intercepted by GV
> voicemail.
> fs_cli dumped this  http://pastebin.com/8sJjUZx1 message  as seen on line#
> 26 and here is the  http://pastebin.com/gnb8dXmW output  of dl_debug on.
> With option #2, the call comes in. When I picked up to answer the call, no
> audio and caller still heard the ringing tone until the call got
> intercepted
> by GV voicemail. fs_cli dumped this  http://pastebin.com/fJ1s14tX message
> and the  http://pastebin.com/mHBFECs3 output  of dl_debug on.
>
> If anyone here has any idea how to fix this, I sure would appreciate that.
> Thanks.
>
> -----
> 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/ext-rtp-ip-value-behind-NAT-tp5996978p5996978.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 krice at freeswitch.org  Mon Feb  7 19:13:14 2011
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 07 Feb 2011 10:13:14 -0600
Subject: [Freeswitch-users] FS ESL in Adobe Flex
In-Reply-To: <BLU143-W13F9C31C28DF6BFBCF736EA9EB0@phx.gbl>
Message-ID: <C975763A.8C39%krice@freeswitch.org>

You should not have a webclient such as flex diectly access ELS...  Use a
some sort of proxy such as Java or AMF-PHP based proxy to buffer between the
2... This will allow greater flexibility plus the use of Flex Object
remoting. There are some examples of this in the swk contrib directory
however they are a bit old
K


On 2/7/11 8:50 AM, "Francois Barrouin" <francoisbarrouin at hotmail.fr> wrote:

> Hello,
> 
> I would like to use Adobe Flex to access FS event socket.
> I read in the (excellent!) ?FreeSWITCH 1.0.6?  book that FS ESL is based on
> swig and unfortunately Adobe Flex is not a language supported by this project.
> 
> I wonder if anyone has already developed a Flex ESL for FS?
> 
> Thanks
> 
> Fran?ois
>        
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From rajesh.npnr at yahoo.com  Mon Feb  7 19:15:31 2011
From: rajesh.npnr at yahoo.com (rex.alex)
Date: Mon, 7 Feb 2011 08:15:31 -0800 (PST)
Subject: [Freeswitch-users] Grandstream Freeze on Originate command
In-Reply-To: <01af01cbc4a8$8e4fdee0$aaef9ca0$@subscription@alexrambau.com>
References: <1296751186605-5989433.post@n2.nabble.com>
	<AANLkTikPiZgq9WsFeMs2U0iZrT7FcL10dptLCFu5HynP@mail.gmail.com>
	<1296818499616-5992217.post@n2.nabble.com>
	<1296821764473-5992350.post@n2.nabble.com>
	<AANLkTim+3c7Ui=bPd6_7ROXEA+mxgX3RRzCf03PWdTmi@mail.gmail.com>
	<01af01cbc4a8$8e4fdee0$aaef9ca0$@subscription@alexrambau.com>
Message-ID: <1297095331401-6000804.post@n2.nabble.com>


Hello,

This solution worked for me as well.

Thank you so much.

Rex
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Grandstream-Freeze-on-Originate-command-tp5989433p6000804.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Mon Feb  7 19:19:37 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 7 Feb 2011 10:19:37 -0600
Subject: [Freeswitch-users] play_and_get_digits question
In-Reply-To: <EBED8C31D06F4A319D978371EE1CF24E@e1705>
References: <EBED8C31D06F4A319D978371EE1CF24E@e1705>
Message-ID: <AANLkTi=AVwJ4c80z9F_06+8EyUHrokLzEtRcPGGpa4jG@mail.gmail.com>

I just checked the source (switch_ivr_play_say.c, function name
'switch_play_and_get_digits') and I do not see any indication that a failure
to collect digits after max tries will automatically disconnect the call.
This means you'll need to handle the failure condition in your dialplan or
call a Lua/Perl/Javascript script to do the logic. In this scenario I favor
a simple Lua script that calls PAGD and checks the value. You can hangup on
the caller if they don't enter a PIN.

-MC

On Sun, Feb 6, 2011 at 1:18 PM, Madovsky <infos at madovsky.org> wrote:

>  sorry I think I really didn't undertand the play_and_get_digits concept.
> is anyone can help ?
>
> I just want to use this app to check a pin code, hangup if wrong and
> continue
> dialplan if ok
>
> Thanks
>
> ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Sunday, February 06, 2011 1:01 PM
> *Subject:* Re: play_and_get_digits question
>
> or maybe this channel var
> read_result
> is the result of play_and_get_digits ?
>
> ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Sunday, February 06, 2011 12:43 PM
> *Subject:* Re: play_and_get_digits question
>
> Ok I think I misunderstood the concept of var where digits should be put
> in.
> it's with this var I need to ckeck after this application.
>
> self forum thank you ;)
>
> ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Sunday, February 06, 2011 12:29 PM
> *Subject:* play_and_get_digits question
>
> I have this lines in my dialplan
>
>        <action application="play_and_get_digits" data="4 4 3 5000 #
> phrase:wait_foo phrase:bad_foo foo ${foo}"/>
>        <action application="transfer" data="otherext XML features"/>
>
> if I don't press any DTMF, after 4 loops the dialplan continues,
> but I thought that it was automatically hangup in case of no digits bad
> digits...
>
> Did I forget something ?
>
> Thanks
>
> Franck
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  7 19:26:07 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 7 Feb 2011 11:26:07 -0500
Subject: [Freeswitch-users] play_and_get_digits question
References: <EBED8C31D06F4A319D978371EE1CF24E@e1705>
	<AANLkTi=AVwJ4c80z9F_06+8EyUHrokLzEtRcPGGpa4jG@mail.gmail.com>
Message-ID: <967A19E77EAD4F21A3C148492E88FCDD@e1705>

Hi Mike,

I resolved it after some experiments.
thanks

  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, February 07, 2011 11:19 AM
  Subject: Re: [Freeswitch-users] play_and_get_digits question


  I just checked the source (switch_ivr_play_say.c, function name 'switch_play_and_get_digits') and I do not see any indication that a failure to collect digits after max tries will automatically disconnect the call. This means you'll need to handle the failure condition in your dialplan or call a Lua/Perl/Javascript script to do the logic. In this scenario I favor a simple Lua script that calls PAGD and checks the value. You can hangup on the caller if they don't enter a PIN.


  -MC


  On Sun, Feb 6, 2011 at 1:18 PM, Madovsky <infos at madovsky.org> wrote:

    sorry I think I really didn't undertand the play_and_get_digits concept.
    is anyone can help ?

    I just want to use this app to check a pin code, hangup if wrong and continue
    dialplan if ok

    Thanks
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Sunday, February 06, 2011 1:01 PM
      Subject: Re: play_and_get_digits question


      or maybe this channel var 
      read_result
      is the result of play_and_get_digits ?
        ----- Original Message ----- 
        From: Madovsky 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Sunday, February 06, 2011 12:43 PM
        Subject: Re: play_and_get_digits question


        Ok I think I misunderstood the concept of var where digits should be put in.
        it's with this var I need to ckeck after this application.

        self forum thank you ;)
          ----- Original Message ----- 
          From: Madovsky 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Sunday, February 06, 2011 12:29 PM
          Subject: play_and_get_digits question


          I have this lines in my dialplan

                 <action application="play_and_get_digits" data="4 4 3 5000 # phrase:wait_foo phrase:bad_foo foo ${foo}"/>
                 <action application="transfer" data="otherext XML features"/>

          if I don't press any DTMF, after 4 loops the dialplan continues,
          but I thought that it was automatically hangup in case of no digits bad digits...

          Did I forget something ?

          Thanks

          Franck



    _______________________________________________
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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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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Mon Feb  7 19:32:45 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 7 Feb 2011 11:32:45 -0500
Subject: [Freeswitch-users] play_and_get_digits question
References: <EBED8C31D06F4A319D978371EE1CF24E@e1705>
	<AANLkTi=AVwJ4c80z9F_06+8EyUHrokLzEtRcPGGpa4jG@mail.gmail.com>
Message-ID: <420CA447F8C140BEB942DE8BDB3D63ED@e1705>

but that's would be cool to see this app in an example dialplan
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, February 07, 2011 11:19 AM
  Subject: Re: [Freeswitch-users] play_and_get_digits question


  I just checked the source (switch_ivr_play_say.c, function name 'switch_play_and_get_digits') and I do not see any indication that a failure to collect digits after max tries will automatically disconnect the call. This means you'll need to handle the failure condition in your dialplan or call a Lua/Perl/Javascript script to do the logic. In this scenario I favor a simple Lua script that calls PAGD and checks the value. You can hangup on the caller if they don't enter a PIN.


  -MC


  On Sun, Feb 6, 2011 at 1:18 PM, Madovsky <infos at madovsky.org> wrote:

    sorry I think I really didn't undertand the play_and_get_digits concept.
    is anyone can help ?

    I just want to use this app to check a pin code, hangup if wrong and continue
    dialplan if ok

    Thanks
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Sunday, February 06, 2011 1:01 PM
      Subject: Re: play_and_get_digits question


      or maybe this channel var 
      read_result
      is the result of play_and_get_digits ?
        ----- Original Message ----- 
        From: Madovsky 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Sunday, February 06, 2011 12:43 PM
        Subject: Re: play_and_get_digits question


        Ok I think I misunderstood the concept of var where digits should be put in.
        it's with this var I need to ckeck after this application.

        self forum thank you ;)
          ----- Original Message ----- 
          From: Madovsky 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Sunday, February 06, 2011 12:29 PM
          Subject: play_and_get_digits question


          I have this lines in my dialplan

                 <action application="play_and_get_digits" data="4 4 3 5000 # phrase:wait_foo phrase:bad_foo foo ${foo}"/>
                 <action application="transfer" data="otherext XML features"/>

          if I don't press any DTMF, after 4 loops the dialplan continues,
          but I thought that it was automatically hangup in case of no digits bad digits...

          Did I forget something ?

          Thanks

          Franck



    _______________________________________________
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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  Mon Feb  7 19:59:36 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 7 Feb 2011 11:59:36 -0500
Subject: [Freeswitch-users] play_and_get_digits question
References: <EBED8C31D06F4A319D978371EE1CF24E@e1705>
	<AANLkTi=AVwJ4c80z9F_06+8EyUHrokLzEtRcPGGpa4jG@mail.gmail.com>
Message-ID: <C2CAC53C1A8B4CB3ADC3B5EF97F56DDD@e1705>

Mike,

In fact I almost resolved it.
there's only a little problem when the user doesn't press any key.

I have this in my dialplan

                        <condition field="${conf_admin_pin}" expression="^(\d+)$">
                                <action inline="true" application="set" data="keys_pressed=${conf_admin_pin}"/>
                                <action application="play_and_get_digits" data="4 4 3 5000 # phrase:wait_admin_pin phrase:bad_admin_pin keys_pressed ${conf_admin_pin}"/>
                                <anti-action application="speak" data="cepstral|david|Please define an admin pin code from your account before to create a conference, thank you."/>
                                <anti-action application="hangup"/>
                        </condition>
                        <condition field="${keys_pressed}" expression="^(${conf_admin_pin})$">
                                <action application="transfer" data="otherext XML features"/>
                                <anti-action application="hangup"/>
                        </condition>

if I don't set keys_pressed before, the next condition with  is always false.
so it seems that keys_pressed var is never inline.

Thanks



  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, February 07, 2011 11:19 AM
  Subject: Re: [Freeswitch-users] play_and_get_digits question


  I just checked the source (switch_ivr_play_say.c, function name 'switch_play_and_get_digits') and I do not see any indication that a failure to collect digits after max tries will automatically disconnect the call. This means you'll need to handle the failure condition in your dialplan or call a Lua/Perl/Javascript script to do the logic. In this scenario I favor a simple Lua script that calls PAGD and checks the value. You can hangup on the caller if they don't enter a PIN.


  -MC


  On Sun, Feb 6, 2011 at 1:18 PM, Madovsky <infos at madovsky.org> wrote:

    sorry I think I really didn't undertand the play_and_get_digits concept.
    is anyone can help ?

    I just want to use this app to check a pin code, hangup if wrong and continue
    dialplan if ok

    Thanks
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Sunday, February 06, 2011 1:01 PM
      Subject: Re: play_and_get_digits question


      or maybe this channel var 
      read_result
      is the result of play_and_get_digits ?
        ----- Original Message ----- 
        From: Madovsky 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Sunday, February 06, 2011 12:43 PM
        Subject: Re: play_and_get_digits question


        Ok I think I misunderstood the concept of var where digits should be put in.
        it's with this var I need to ckeck after this application.

        self forum thank you ;)
          ----- Original Message ----- 
          From: Madovsky 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Sunday, February 06, 2011 12:29 PM
          Subject: play_and_get_digits question


          I have this lines in my dialplan

                 <action application="play_and_get_digits" data="4 4 3 5000 # phrase:wait_foo phrase:bad_foo foo ${foo}"/>
                 <action application="transfer" data="otherext XML features"/>

          if I don't press any DTMF, after 4 loops the dialplan continues,
          but I thought that it was automatically hangup in case of no digits bad digits...

          Did I forget something ?

          Thanks

          Franck



    _______________________________________________
    FreeSWITCH-users mailing 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 sjmudd at pobox.com  Mon Feb  7 20:28:34 2011
From: sjmudd at pobox.com (Simon J Mudd)
Date: 07 Feb 2011 18:28:34 +0100
Subject: [Freeswitch-users] Confusing SIP auth failure logging message?
In-Reply-To: <88F0F936-6527-4379-88F2-412FAD2071A7@gmail.com>
References: <20110206232236.GA10501@mad06.wl0.org>
	<20110207073124.GA5255@mad06.wl0.org>
	<88F0F936-6527-4379-88F2-412FAD2071A7@gmail.com>
Message-ID: <m3lj1reou5.fsf@mad06.wl0.org>

steveayre at gmail.com (Steven Ayre) writes:

> You can do so in freeswitch but sometimes it's better to do it in the firewall.

Perhaps. If it's possible to do it in FreeSWITCH how can it be done? I can't find
the "how" in the wiki documentation which is why I was asking.

> For example it uses far fewer resources to block someone in the
> firewall. It's also easier to block scanners such as
> friendly-scanner in the firewall.

Are there examples of this on the wiki?  That is iptables rules intended for
a system such as FreeSWITCH which log problems. Some things may make
more sense in the firewall (or on the FreeSWITCH server itself) but
I'm still inclined to think that other stuff should be provided by
the application itself, even if not enabled by default if this uses
up "precious resources" on a busy system. On a SOHO type system I'd
personally configure things in one place as that is easier to maintain.

Simon


From rhosyn at purplescarab.com  Mon Feb  7 19:35:31 2011
From: rhosyn at purplescarab.com (Rhosyn)
Date: Mon, 7 Feb 2011 16:35:31 +0000
Subject: [Freeswitch-users] Using mod_opal with MSN Messenger
In-Reply-To: <4BDB6016-806B-4855-9A42-0A8201517F0B@visionutveckling.se>
References: <4BDB6016-806B-4855-9A42-0A8201517F0B@visionutveckling.se>
Message-ID: <AANLkTi=9nAY5ryTiYuCXxyat4yzu54KdGmXGPwz6WJgk@mail.gmail.com>

Yeah, the modules can be used to talk to many other H.323 devices probably
including (I think)  PBXes, and  H.323 voice/video
endpoints/MCUs/gateways/gatekeepers/VCS etc.
 (made by TANDBERG, Cisco, Polycom, Lifesize, RADVISION and many other
manufacturers)  - but not MSN I'm afraid :-(

I wonder if anyone is interested in writing a mod_msn?

Tough job though as (like mod_skype) it's dealing with a closed proprietary
signalling spec (though, from the IM (but not voice) side, the likes of
Pidgin etc. have done a good job of creating interoperable open source
stack).

Dunno if there's any stacks that are license compatible with FreeSwitch
around?



On 7 February 2011 15:56, Peter Olsson <peter.olsson at visionutveckling.se>wrote:

> The use for these modules are to communicate with other h323 stacks/PBX'es.
> MSN is not one of them AFAIK.
>
> /Peter
>
> ----- Reply message -----
> Fr?n: "mazilo" <Nabble at slickdeals.endjunk.com>
> Datum: m?n, feb 7, 2011 22:17
> Rubrik: [Freeswitch-users] Using mod_opal with MSN Messenger
> Till: "freeswitch-users at lists.freeswitch.org" <
> freeswitch-users at lists.freeswitch.org>
>
>
> Rhosyn,
>
> Thanks for your response. So, it looks like there is no way one can
> configure an FS system using either mod_opal nor mod_h323 to place/receive
> VoIP calls to/from any MSN/HotMail (Live) users. So, what is the use for
> either mod_opal or mod_h323?
>
> -----
> 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/Using-mod-opal-with-MSN-Messenger-tp5994575p6000576.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
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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:4d500cac32762891562358!
>
>
> _______________________________________________
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From larry at newportpc.com  Mon Feb  7 20:11:13 2011
From: larry at newportpc.com (Larry Marshall)
Date: Mon, 7 Feb 2011 09:11:13 -0800
Subject: [Freeswitch-users] Group confusions
In-Reply-To: <AANLkTinz0NuBbN_Q5gQnmuDkkicchhxWMg4XnH=19CJy@mail.gmail.com>
References: <AcvEp2ZUeoD+FSG/QdKLQRe6PdRMAA==>	<00f501cbc4a7$aaee5640$00cb02c0$@yahoo.com>
	<AANLkTinz0NuBbN_Q5gQnmuDkkicchhxWMg4XnH=19CJy@mail.gmail.com>
Message-ID: <009f01cbc6ea$06beb700$143c2500$@newportpc.com>

Michael,

 

Thanks for your response. The cli tests are:

 

freeswitch at internal> group call:01 at 192.168.10.29

sofia/internal/sip:1002 at 192.168.10.105

freeswitch at internal> sofia_contact 1013 at 192.168.10.29

sofia/internal/sip:1013 at 192.168.10.107

freeswitch at internal> group insert:01:1013 at 192.168.10.29

+OK

freeswitch at internal> group call:01 at 192.168.10.29

sofia/internal/sip:1002 at 192.168.10.105

 

I assume that after adding extension 1013 to group 01 it should be listed in
the 'group call' function. Is this correct?

 

Maybe I should start from the beginning, especially with your warning about
the difference between the group and group_call APIs. I want to ring a group
of extensions on an inbound call. I need to populate the members of this
group based on time constraints (the current time of day, e.g. extension
1013 should only ring between 8:00AM and 5:00PM Monday-Friday), and ring the
appropriate extensions. I wanted to try to do this within a lua script.

 

Which group API should I look at?

 

Thanks, Lars

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Monday, February 07, 2011 8:05 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Group confusions

 

I may be that the value you are trying to delete isn't "1013 at 192.168.10.29".
To find out, go to fs_cli and do this:

 

group call:01 at 192.168.10.29 <mailto:call%3A01 at 192.168.10.29> 

 

It will spit out all the sofia contacts in group 01. Most likely you have
something like this:

1013 at 192.168.10.29:12345;rinstance=aabbccddeeff0011;transport=udp

 

BTW, the value put into the group call when you dial 8101 is the same value
that you find by doing this at fs_cli:

 

sofia_contact 1013 at 192.168.10.29

 

Go try it and let us now.

 

-MC

 

P.S. - the 'group' API and dp apps you are using are the same - they come
from mod_db. They are *totally* different from the group_call API...

On Fri, Feb 4, 2011 at 2:11 PM, Lars Zeb <larclap at yahoo.com> wrote:

I went to two individual extensions and registered them for group 01 by
dialing 8101. I then tested that group by dialing 8201 - both extensions
rang.

I then went to the cli and entered the command:

group delete:01:1013 at 192.168.10.29 <mailto:delete%3A01%3A1013 at 192.168.10.29>


where 1013 is one of the two extensions I had joined into group 01 earlier
and the url is the domain and fs address. However, when I dialed 8201 after
entering this cli command, both extensions rang. I had expected that only
one would ring.

I am confused. Are these two different types of "groups"?

Thanks, Lars


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From francoisbarrouin at hotmail.fr  Mon Feb  7 20:17:08 2011
From: francoisbarrouin at hotmail.fr (Francois Barrouin)
Date: Mon, 7 Feb 2011 18:17:08 +0100
Subject: [Freeswitch-users] FS ESL in Adobe Flex
In-Reply-To: <C975763A.8C39%krice@freeswitch.org>
References: <BLU143-W13F9C31C28DF6BFBCF736EA9EB0@phx.gbl>,
	<C975763A.8C39%krice@freeswitch.org>
Message-ID: <BLU143-W1221E99DA7D217FCD0D746A9EB0@phx.gbl>


Thanks for your answer, I will go in that direction.
I have already developed some code to have a Flex application communicating with a Ruby On Rails server. There is an ESL for ruby, so it may be the best way for me. I have just to find out how to manage FS real time events with RoR.
Francois
 


Date: Mon, 7 Feb 2011 10:13:14 -0600
From: krice at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS ESL in Adobe Flex


You should not have a webclient such as flex diectly access ELS...  Use a some sort of proxy such as Java or AMF-PHP based proxy to buffer between the 2... This will allow greater flexibility plus the use of Flex Object remoting. There are some examples of this in the swk contrib directory however they are a bit old
K


On 2/7/11 8:50 AM, "Francois Barrouin" <francoisbarrouin at hotmail.fr> wrote:


Hello,

I would like to use Adobe Flex to access FS event socket. 
I read in the (excellent!) ?FreeSWITCH 1.0.6?  book that FS ESL is based on swig and unfortunately Adobe Flex is not a language supported by this project.

I wonder if anyone has already developed a Flex ESL for FS? 

Thanks

Fran?ois
      


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From francoisbarrouin at hotmail.fr  Mon Feb  7 20:20:43 2011
From: francoisbarrouin at hotmail.fr (Francois Barrouin)
Date: Mon, 7 Feb 2011 18:20:43 +0100
Subject: [Freeswitch-users] Simple_conference.lua
Message-ID: <BLU143-W134E7A1B03729150AD7344A9EB0@phx.gbl>




You could consider the following alternatives:
- use TTS to generate your prompts: http://wiki.freeswitch.org/wiki/Categories
- take the prompt files from Asterisk

Surprising that FS doesn??t include these prompts.

Fran?ois

 
 
2011/2/3 John Smith <johns1433 at gmail.com>


Hi,

I?m trying to have the script http://wiki.freeswitch.org/wiki/Simple_conference.lua running on my FreeSwitch server.
I get an error when FS wants to play any phrase given in this example:
[ERR] switch_ivr_play_say.c:150 Can't find macro conference_welcome

It seems normal as I didn?t find any reference to such phrases in the different xml phrase configuration files. There seems also to be no adequate wav file in sounds/en/us/callie/voicemail/8000.

My question is: 
Do these files exist somewhere or does the example assume that we have to record our own wav files?

Thanks

John
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From basit.engg at gmail.com  Mon Feb  7 20:47:57 2011
From: basit.engg at gmail.com (Abdul Basit)
Date: Mon, 7 Feb 2011 22:47:57 +0500
Subject: [Freeswitch-users] time_test on Centos 5.5
In-Reply-To: <AANLkTi=9Zsu2aMifLC406iabet6Q9=n3YOTa0N+BLPck@mail.gmail.com>
References: <AANLkTiksCg70iEdZiN1+CnfyBW87mg7T4jDHjeq7JW5Y@mail.gmail.com>
	<0E4A540B-3E61-4940-8246-6FBF67CF91D8@ipeva.fr>
	<41F744A2-90FF-405F-AF62-5E7B8FB5128F@carmickle.com>
	<AANLkTi=NEPwvbF70y3j0p_jfSgLg=xBxW7Es1k7RgAVm@mail.gmail.com>
	<AANLkTi=9Zsu2aMifLC406iabet6Q9=n3YOTa0N+BLPck@mail.gmail.com>
Message-ID: <AANLkTikUABYauEotH=Qg5hXTji9BxQAP2=+uCGPgPEap@mail.gmail.com>

verify kernel timer frequency

grep CONFIG_HZ /boot/config-*

if state

/boot/config-2.6.18-194.3.1.el5PAE:CONFIG_HZ_1000=y
/boot/config-2.6.18-194.3.1.el5PAE:CONFIG_HZ=1000

that means kernel is compiled with timer freq. 1000Hz


On Sat, Feb 5, 2011 at 3:30 PM, Saeed Ahmed <saeedahmad1981 at gmail.com>wrote:

> how we set kenrel timer freq to 1000hz? on centos?
>
>
> On Sat, Jan 29, 2011 at 1:10 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> I've been using it on Lenny with no problems for ~2 years, timing works
>> fine. It will work. CentOS is the reference platform though.
>>
>> -Steve
>>
>>
>>
>>
>> On 28 January 2011 19:05, Frank Carmickle <frank at carmickle.com> wrote:
>>
>>>
>>> On Jan 28, 2011, at 6:35 AM, David Ponzone wrote:
>>> Snip...
>>>
>>> > Stay away from Debian, Centos is the right choice.
>>> > You could eventually try to fallback to centos 5.3 or 5.4.
>>> >
>>> Debian can work if that's what people want to use.  I have it working
>>> well on a few lenny machines.
>>>
>>> --FC
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing 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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>
>


-- 
Regards,

Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445
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From infos at madovsky.org  Mon Feb  7 21:00:56 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 7 Feb 2011 13:00:56 -0500
Subject: [Freeswitch-users] play_and_get_digits question
References: <EBED8C31D06F4A319D978371EE1CF24E@e1705><AANLkTi=AVwJ4c80z9F_06+8EyUHrokLzEtRcPGGpa4jG@mail.gmail.com>
	<C2CAC53C1A8B4CB3ADC3B5EF97F56DDD@e1705>
Message-ID: <371E5F2A00EE49F2B02A7074AA9A0AF6@e1705>

Ok now I understand more.
it seems that it needs to transfer after play_and_get_digits to
get the keys_pressed values
                <extension name="step1">
                        <condition field="${conf_admin_pin}" expression="^(\d+)$">
                                <action application="play_and_get_digits" data="4 4 3 5000 # phrase:wait_admin_pin phrase:bad_admin_pin keys_pressed ${conf_admin_pin}"/>
                                <action application="transfer" data="step2 XML features"/>
                        </condition>
                </extension>
                <extension name="step2">
                        <condition field="${keys_pressed}" expression="^(${conf_admin_pin})$">
                                <action application="transfer" data="otherext XML features"/>
                                <anti-action application="hangup"/>
                        </condition>
                </extension>


as this it works perfectly


thanks

  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Monday, February 07, 2011 11:59 AM
  Subject: Re: [Freeswitch-users] play_and_get_digits question


  Mike,

  In fact I almost resolved it.
  there's only a little problem when the user doesn't press any key.

  I have this in my dialplan

                          <condition field="${conf_admin_pin}" expression="^(\d+)$">
                                  <action inline="true" application="set" data="keys_pressed=${conf_admin_pin}"/>
                                  <action application="play_and_get_digits" data="4 4 3 5000 # phrase:wait_admin_pin phrase:bad_admin_pin keys_pressed ${conf_admin_pin}"/>
                                  <anti-action application="speak" data="cepstral|david|Please define an admin pin code from your account before to create a conference, thank you."/>
                                  <anti-action application="hangup"/>
                          </condition>
                          <condition field="${keys_pressed}" expression="^(${conf_admin_pin})$">
                                  <action application="transfer" data="otherext XML features"/>
                                  <anti-action application="hangup"/>
                          </condition>

  if I don't set keys_pressed before, the next condition with  is always false.
  so it seems that keys_pressed var is never inline.

  Thanks



    ----- Original Message ----- 
    From: Michael Collins 
    To: FreeSWITCH Users Help 
    Sent: Monday, February 07, 2011 11:19 AM
    Subject: Re: [Freeswitch-users] play_and_get_digits question


    I just checked the source (switch_ivr_play_say.c, function name 'switch_play_and_get_digits') and I do not see any indication that a failure to collect digits after max tries will automatically disconnect the call. This means you'll need to handle the failure condition in your dialplan or call a Lua/Perl/Javascript script to do the logic. In this scenario I favor a simple Lua script that calls PAGD and checks the value. You can hangup on the caller if they don't enter a PIN. 


    -MC


    On Sun, Feb 6, 2011 at 1:18 PM, Madovsky <infos at madovsky.org> wrote:

      sorry I think I really didn't undertand the play_and_get_digits concept.
      is anyone can help ?

      I just want to use this app to check a pin code, hangup if wrong and continue
      dialplan if ok

      Thanks
        ----- Original Message ----- 
        From: Madovsky 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Sunday, February 06, 2011 1:01 PM
        Subject: Re: play_and_get_digits question


        or maybe this channel var 
        read_result
        is the result of play_and_get_digits ?
          ----- Original Message ----- 
          From: Madovsky 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Sunday, February 06, 2011 12:43 PM
          Subject: Re: play_and_get_digits question


          Ok I think I misunderstood the concept of var where digits should be put in.
          it's with this var I need to ckeck after this application.

          self forum thank you ;)
            ----- Original Message ----- 
            From: Madovsky 
            To: freeswitch-users at lists.freeswitch.org 
            Sent: Sunday, February 06, 2011 12:29 PM
            Subject: play_and_get_digits question


            I have this lines in my dialplan

                   <action application="play_and_get_digits" data="4 4 3 5000 # phrase:wait_foo phrase:bad_foo foo ${foo}"/>
                   <action application="transfer" data="otherext XML features"/>

            if I don't press any DTMF, after 4 loops the dialplan continues,
            but I thought that it was automatically hangup in case of no digits bad digits...

            Did I forget something ?

            Thanks

            Franck



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From msc at freeswitch.org  Mon Feb  7 21:04:16 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 7 Feb 2011 12:04:16 -0600
Subject: [Freeswitch-users] close all conference from fs_cli
In-Reply-To: <313DB3017F77477FBC815E78800254C3@e1705>
References: <313DB3017F77477FBC815E78800254C3@e1705>
Message-ID: <AANLkTim+fQSOM2KyhnACfa-p9i6MSPTXwMchJBzdV=C2@mail.gmail.com>

There isn't a single command, but you could write a script that does a show
conf to get a list of conferences and then do a conf xxx kick all on each
one.
-MC

On Sun, Feb 6, 2011 at 7:05 PM, Madovsky <infos at madovsky.org> wrote:

>  Is there a way to close/hangup all active conferences
> from CLI ?
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Mon Feb  7 21:10:47 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 7 Feb 2011 13:10:47 -0500
Subject: [Freeswitch-users] close all conference from fs_cli
References: <313DB3017F77477FBC815E78800254C3@e1705>
	<AANLkTim+fQSOM2KyhnACfa-p9i6MSPTXwMchJBzdV=C2@mail.gmail.com>
Message-ID: <57B06820CB8D4E9DA0A00A46FD7D3D67@e1705>

yes it's what I planned to do.
in fact as socket app doesn't have any ping test or check status property,
in case of socket down the only way for now is to
create a script that ping/telnet the connection status and if down
run the right commands to clean what you do in ESL
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, February 07, 2011 1:04 PM
  Subject: Re: [Freeswitch-users] close all conference from fs_cli


  There isn't a single command, but you could write a script that does a show conf to get a list of conferences and then do a conf xxx kick all on each one.
  -MC


  On Sun, Feb 6, 2011 at 7:05 PM, Madovsky <infos at madovsky.org> wrote:

    Is there a way to close/hangup all active conferences 
    from CLI ?

    Thanks

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From msc at freeswitch.org  Mon Feb  7 21:16:39 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 7 Feb 2011 12:16:39 -0600
Subject: [Freeswitch-users] dialplanhelp for callforwarding
In-Reply-To: <Pine.LNX.4.64.1102070934130.31747@suse.blindi.net>
References: <8A02ED1AC7914B4BAD4EEF5D910D28DE@e1705>
	<Pine.LNX.4.64.1102070934130.31747@suse.blindi.net>
Message-ID: <AANLkTi=QbrjG0fA6y7TxwO6P1YL=VF93zJry=WPE0yNO@mail.gmail.com>

There isn't a "gotoif" construct in the FS dialplan. Any time you need a
fair amount of logic in your dialplan then you should consider using a
dialplan script. You can write them in Lua, Perl, or Javascript. Personally,
the FS devs prefer Lua for dialplan scripting because it is pretty easy. I
recommend getting our book and looking at chapter 7. It's a gentle
introduction to writing simple dialplan scripts in Lua.

Remember this rule about the FreeSWITCH XML dialplan: it can do a lot but it
isn't a scripting language - it's a routing "language". Tony designed it
that way. You *can* do some logic-type constructs but I wouldn't go there
when you're still just learning everything.

The first thing you should do in translating your Asterisk dialplan to
FreeSWITCH is to write out the call flow outline. From there you can define
what aspects can be done in the XML dialplan and what can be done in your
Lua script.

The other thing I'd recommend doing is joining the #freeswitch channel on
irc.freenode.net. You can talk to people in realtime and they can help
answer specific questions.

-MC

On Mon, Feb 7, 2011 at 3:32 AM, Thomas Hoellriegel <admin at blindi.net> wrote:

> Hi all, i like to replace my asterisk to fs. FS is very nice.
> I hanging:
> my problem is: i don.t find a Gotoif command from fs to replace, my
> asterisk dialplan.
>
> my simply dialplan in asterisk is:
>
> ;;Callforwarding to a phonenumber from dialplan
> ;;from: thomas Hoellriegel blindi at gmx.net
>
> ;;maimenu for callforwarding
> ;;anfang is in english: beginn
> [callforwarding-only]
> exten => s,1,Answer()
> exten => s,n,Set(TIMEOUT(response)=10)
> exten =>
> s,n,Background(callforwarding-anfang1&callforwarding-anfang2&callforwarding-anfang3&callforwarding-anfang4&callforwarding-anfang5&callforwarding-anfang6&callforwarding-anfang7&callforwarding-anfang8&mit-stern-exit)
> exten => s,n,WaitExten
> exten => *,1,Goto(adminmenu,s,1)
>
> ;;drop all user errors
> exten => i,1,Goto(s,1)
>
> ;;menuselection
> exten => 1,1,Goto(callforwarding-db,s,1)
>
>
> exten => t,1,Playback(vm-goodbye)
> exten => t,n,Hangup()
>
> [callforwarding-db]
> exten => s,1,Answer()
> exten => s,n,Set(TIMEOUT(response)=10)
>
> ;;status: ;;aus: (off)  goto: no number in database
> ;;fwan: (on)  goto: then number is found
> exten => s,n,GotoIf($["${DB(CFtommystatus/0)}" != ""]?fwan:fwaus)
> exten => s,n(fwan),Playback(all-callfw-is-on)
> exten => s,n,Goto(s,30)
>
> exten => s,n(fwaus),Playback(all-fw-ist-off)
> exten => s,n,Goto(s,30)
>
> exten => s,30,Background(callforwarding-main)
> exten => s,n,WaitExten
> exten => *,1,Goto(callforwarding-only,s,1)
>
> exten => i,1,Goto(s,1)
>
> ;;nummer is: shortnumber for callforwarding
> exten => 1,1,Goto(nummer1,nr1,1)
> ;;toggle
> ;;ein aus in english: on off
> exten => 0,1,GotoIf($["${DB(CFtommystatus/0)}" != ""]?ein:aus)
> exten => 0,n(ein),DBdeltree(CFtommystatus/0)
> exten => 0,n,Playback(all-fw-ist-off)
> exten => 0,n,Goto(callforwarding-only,s,1)
>
> exten => 0,n(aus),Set(DB(CFtommystatus/0)=p)
> exten => 0,n,Playback(alle-callfw-is-on)
> exten => 0,n,Goto(callforwarding-only,s,1)
>
> ;;number 1
> ;;kurzwahl is: shortnumber [nummer1]
> exten => nr1,1,Playback(kurzwahl)
> exten => nr1,n,SayDigits(1)
> exten => nr1,n,Background(callforwarding-opt)
> exten => nr1,n,WaitExten
>
> ;;ja: say the existing number
> ;;falsch:
> ;;say no number in database found
>
> exten => 1,1,GotoIf($["${DB(CFtommy1/0)}" != ""]?ja1:falsch1)
> exten => 1,n(ja1),SayDigits(${DB(CFtommy1/0)})
> exten => 1,n,Wait(0.2)
> exten => 1,n,Goto(nr1,1)
> exten => 1,n(falsch1),Playback(callforwarding-keine)
> exten => 1,n,Wait(0.2)
> exten => 1,n,Goto(nr1,1)
>
> ;;2 to add a number
> exten => 2,1,Goto(nummer1-menu1,s,1)
>
> ;;aus: deactivate callforwarding
> exten => 3,1,Goto(nummer1aus,s,1)
> exten => 4,1,SayDigits(${CALLERID(num):})
> exten => 4,n,Set(DB(CFtommy1/0)=${CALLERID(num):})
> exten => 4,n,Playback(callforwarding-on)
> exten => 4,n,Goto(callforwarding-only,s,1)
>
> exten => *,1,Goto(callforwarding-only,s,1)
>
> exten => i,1,Goto(nr1,1)
>
> ;;aus:  deactivate
> [nummer1aus]
> exten => s,1,DBdeltree(CFtommy1/0)
> exten => s,n,Background(callforwarding-off)
> exten => s,n,Goto(callforwarding-only,s,1)
>
> ;;menu for dtmf digits
> [nummer1-menu1]
> exten => s,1,Set(NR=)
> exten => s,n,Background(bitte-callforwarding-nummer)
> exten => s,n,Set(TIMEOUT(response)=10)
> exten => s,n,WaitExten
> exten => _X,1,Set(NR=${NR}${EXTEN})
> exten => _X,n,Goto(s,3)
> exten => *,1,Goto(s,1)
>
> exten => #,1,Playback(callforwardings-number-is)
> exten => #,n,SayDigits(${NR})
>
> ;;check the number
> exten => #,n,Goto(nummer1-chk,${NR},1)
>
> exten => t,1,Playback(vm-goodbye)
> exten => t,n,Hangup()
>
> ;;check the number for no allowed numbers [nummer1-chk]
> exten => i,1,NoOp(Undefined  Nummer ${INVALID_EXTEN}.)
> exten => i,n,Answer()
> exten => i,n,Playback(tt-somethingwrong)
> exten => i,n,Playback(tt-monkeysintro)
> exten => i,n,Hangup()
>
> ;;forbidden numbers:
> ;;neu is:  new
> exten => _911.,1,Playback(nummer-falsch)
> exten => _911.,n,Goto(callback-only-neu,s,1)
>
> exten => _#,1,Playback(notallow)
> exten => _#,n,Goto(nummer1-menu1,s,1)
> exten => _X,1,Playback(notallow)
> exten => _X,n,Goto(nummer1-menu1,s,1)
> exten => _X.,1,Goto(nummer1-confirm,s,1)
>
> ;;confirmmenu
> [nummer1-confirm]
> exten => s,1,Background(ist-correct)
> exten => s,n,Set(TIMEOUT(response)=10)
> exten => s,n,WaitExten
> exten => 1,1,Set(DB(CFtommy1/0)=${NR})
> exten => 1,n,Playback(callforwarding-on)
> exten => 1,n,Goto(callforwarding-only,s,1)
>
> exten => 2,1,Goto(nummer1-menu1,s,1)
>
> exten => #,1,Playback(errormenu)              ; "Thanks for trying the
> demo"
> exten => #,n,Goto(nummer1-menu1,#,1)
> exten => t,1,Goto(nummer1-menu1,#,1)
> exten => i,1,Playback(conf-errormenu)          ; "That's not valid, try
> again"
> exten => i,n,Goto(nummer1-menu1,#,1)
> ;;fw_outcontext
> [fw_out]
> include=trunk
> include=international
>
> [default]
> ;;forwarding extension in defaultcontext
>
> ;;read the db entry
> exten => answert1,1,GotoIf($["${DB(CFtommy1/0)}" != ""]?yesfw1:nofw1)
> exten => answert1,n(yesfw1),Dial(Local/${DB(CFtommy1/0)}@fw_out,,)
> exten => answert1,n(nofw1),NoOp()
>
> can you help me please? thankx.
>
>
>
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/listinfo/blinde-misc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  7 21:17:52 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 7 Feb 2011 12:17:52 -0600
Subject: [Freeswitch-users] conferenceproblems after execute_application
In-Reply-To: <Pine.LNX.4.64.1102060503170.16105@suse.blindi.net>
References: <Pine.LNX.4.64.1102060503170.16105@suse.blindi.net>
Message-ID: <AANLkTiniiqpikj-cTs6+h_xYE=nSxQTnjzPdTxB=wzGk@mail.gmail.com>

Update to the latest git and retest. Let us know how it goes.
-MC

On Sat, Feb 5, 2011 at 10:24 PM, Thomas Hoellriegel <admin at blindi.net>wrote:

> Hi all, i have a working freeswitch git on ubuntu lucid 64bit.
> I setup a conference. I add these line in my call-control-section:
> <control action="execute_application" digits="3" data="playback
> /fssounds/helpconference.wav"/>
> I press 3 the file will be played correctly.
> After the playback i go back to the conference.
> I talk to  others in the conference.
> after the "execute_application" command
> i have many delays 3 or 5 or more seconds.
> My voice is very clearly. the people mean i.m a voicecomputer-).
>
> I must exit the conference and i start again. my voice is normal.
>
> is execute_application a problem?
> can you help me please?
> thanks.
>
>
>
>
>
>
>
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/listinfo/blinde-misc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  7 21:28:21 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 7 Feb 2011 12:28:21 -0600
Subject: [Freeswitch-users] Confusing SIP auth failure logging message?
In-Reply-To: <20110206232236.GA10501@mad06.wl0.org>
References: <20110206232236.GA10501@mad06.wl0.org>
Message-ID: <AANLkTi=JgkwMbmtEsPZRA5NW9kjfbPJ_Uiqqp8ZB3zsB@mail.gmail.com>

On Sun, Feb 6, 2011 at 5:22 PM, Simon J Mudd <sjmudd at pobox.com> wrote:

> I've been looking at trying to configure tighter controls for extensions
> that register.
> Doing so made me trigger this error message (adjusted slightly):
>
> 2011-02-07 00:07:51.343303 [WARNING] sofia_reg.c:1247 SIP auth challenge
> (REGISTER) on sofia profile 'internal' for [1000 at sip.example.com] from ip
> 192.168.4.99
> 2011-02-07 00:07:51.343303 [WARNING] sofia_reg.c:1247 SIP auth challenge
> (REGISTER) on sofia profile 'internal' for [1000 at sip.example.com] from ip
> 192.168.4.99
>
> Looking at the message it is not clear if the SIP authentication has
> succeeded or failed.
> Judging by the code it seems this is meant to represent a SIP auth failure.
> If so should
> the code not be patched as shown?
>

No. This is just saying that there was a challenge, not that there was a
failure. There is already a failure detection routine. To test it, setup a
SIP client with an incorrect password. You'll see two log lines like this:

2011-02-07 12:23:28.490029 [WARNING] sofia_reg.c:1247 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [1002 at 10.10.16.161] from ip
10.10.16.161
2011-02-07 12:23:29.035950 [WARNING] sofia_reg.c:1247 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [1002 at 10.10.16.161] from ip
10.10.16.161
2011-02-07 12:23:29.240695 [WARNING] sofia_reg.c:1205 SIP auth failure
(REGISTER) on sofia profile 'internal' for [1002 at 10.10.16.161] from ip
10.10.16.161

This allows you to differentiate between the mere fact that an auth
challenge was sent to the SIP client vs. the SIP client failing to auth.
(Someone asked for that differentiation a while back - I don't know who or
why...)

-MC
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From daniel_wells at byu.edu  Mon Feb  7 21:42:40 2011
From: daniel_wells at byu.edu (Daniel Wells)
Date: Mon, 7 Feb 2011 11:42:40 -0700
Subject: [Freeswitch-users] Console Shutdown always crashes
Message-ID: <8C68232BC9314C40BBCDDAA480F7B01AEB20514829@harrow.exch.ad.byu.edu>

Any idea why the console always crashes on shutdown (compiled on Windows).

These are the last few logs to the console:

[CONSOLE] switch_loadable_module.c:1405 mod_siren has no shutdown routine
[NOTICE] switch_loadable_module.c:518 Deleting Codec SPEEX 99 Speex 32000hz 20ms
[NOTICE] switch_loadable_module.c:518 Deleting Codec SPEEX 99 Speex 16000hz 20ms
[NOTICE] switch_loadable_module.c:518 Deleting Codec SPEEX 99 Speex 8000hz 20ms
[CONSOLE] switch_loadable_module.c:1405 mod_speex has no shutdown routine

I thought that it must have been because there was no shutdown routine for mod_speex but I noticed mod_siren doesn't either and it passes it ok.

Sincerely,


Daniel Wells
IT Systems Engineer

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From daniel_wells at byu.edu  Mon Feb  7 21:42:21 2011
From: daniel_wells at byu.edu (Daniel Wells)
Date: Mon, 7 Feb 2011 11:42:21 -0700
Subject: [Freeswitch-users] mod_dinglaing errors and google voice
Message-ID: <8C68232BC9314C40BBCDDAA480F7B01AEB20514828@harrow.exch.ad.byu.edu>

So I finally have what appears to be a working version of mod_dingaling on my windows box.  I say working only because the console starts and I see evidence in the logs that freeswitch is communicating with my google account though mod_dingaling  (I see presence information about some of my contacts).

However I am not sure my dialplan is working (just a guess since this my first experience with freeswitch-or any PBX for that matter).

I have been trying to follow the instructions as found here:
http://wiki.freeswitch.org/wiki/Google_Voice

I have a client connected using the default user 1000.  I created a dingaling profile called gv1000.xml (I am using the second example of the dialplan on the above site and inferred that the client profile needed to renamed).  My profile is identical to the one in the example with the exception of the username, password and "exten" (which is set to 1000).  I put a copy of the dial plan in the "default" directory.  But I am getting an error when trying to make outbound calls.  Does this mean I have done something wrong, or is google blocking this now?

freeswitch at dw-laptop> 2011-02-07 10:31:44.361514 [WARNING] sofia_reg.c:1247 SIP auth challenge (INVITE) on sofia profile 'internal' for [18XXXXXXXXX at 192.168.2.97] from ip 192.168.2.97
2011-02-07 10:31:44.477521 [NOTICE] switch_channel.c:811 New Channel sofia/internal/1000 at 192.168.2.97 [cfda05d4-2f61-4ab0-8ca6-f759cfac53b7]
2011-02-07 10:31:44.689533 [INFO] mod_dialplan_xml.c:331 Processing 1000 <1000>->18XXXXXXXXX in context default
2011-02-07 10:31:44.689533 [NOTICE] mod_sofia.c:2185 Ring-Ready sofia/internal/1000 at 192.168.2.97!
2011-02-07 10:31:44.689533 [NOTICE] mod_dptools.c:697 Ring Ready sofia/internal/1000 at 192.168.2.97!
2011-02-07 10:31:44.689533 [NOTICE] mod_dingaling.c:721 Close Channel N/A [CS_NEW]
2011-02-07 10:31:44.689533 [ERR] switch_ivr_originate.c:2638 Cannot create outgoing channel of type [dingaling] cause: [DESTINATION_OUT_OF_ORDER]
2011-02-07 10:31:44.689533 [INFO] mod_dptools.c:2621 Originate Failed.  Cause: DESTINATION_OUT_OF_ORDER
2011-02-07 10:31:44.689533 [NOTICE] mod_dptools.c:2684 Hangup sofia/internal/1000 at 192.168.2.97 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
2011-02-07 10:31:44.900545 [NOTICE] switch_core_session.c:1306 Session 1 (sofia/internal/1000 at 192.168.2.97) Ended
2011-02-07 10:31:44.900545 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/1000 at 192.168.2.97 [CS_DESTROY]

Sincerely,


Daniel Wells

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From msc at freeswitch.org  Mon Feb  7 21:44:23 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 7 Feb 2011 12:44:23 -0600
Subject: [Freeswitch-users] Group confusions
In-Reply-To: <009f01cbc6ea$06beb700$143c2500$@newportpc.com>
References: <00f501cbc4a7$aaee5640$00cb02c0$@yahoo.com>
	<AANLkTinz0NuBbN_Q5gQnmuDkkicchhxWMg4XnH=19CJy@mail.gmail.com>
	<009f01cbc6ea$06beb700$143c2500$@newportpc.com>
Message-ID: <AANLkTik4qK84MNwSBby9KfjqBy7PMU+mxUzqF6mL1ofD@mail.gmail.com>

How much changing will there be in the members who are rung? Do you need to
be able to change the users dynamically? Or can you hard-code them? I see
two options, both of which use the time of day (TOD) routing:

Set up the day-night routing and during 8am-5pm manually dial the group like
this:
<extension name=tod_incoming">
  <condition wday="1-5" hour="8-17">
    <action application="bridge"
data="<ignore_early_media=true>user/xx1:_:user/xx2:_:user/xx3">
    <anti-action application="bridge"
data="<ignore_early_media=true>user/xx4:_:user/xx5:_:user/xx6">
  </condition>
</extension>

This creates a static list of users to be dialed during 8-5, M-F and a
separate, static list to be dialed outside those hours. That's quick and
easy and if you never need to change the users who are dialed then that's
the best option.

If, though, you need to be able to control who is in each group then
designate something like group "1" is day and group "2" is night. Have the
users dial 8101/8001 to join/leave the "day" group and dial 8102/8002 to
join/leave the "night" group. From there your dialplan is very similar to
the previous example:

<extension name=tod_incoming">
  <condition wday="1-5" hour="8-17">
    <action application="bridge"
data="{ignore_early_media=true}${group(call:01@${domain_name})}">
    <anti-action application="bridge"
data="{leg_timeout=15,ignore_early_media=true}${group(call:02@
${domain_name})}">
  </condition>
</extension>

This example simply calls group "01" during business hours and calls group
"02" during non-business hours. If you trust your users to add/remove
themselves properly then this method is nice.

Have fun tinkering, and let us know if you need help setting up
voicemail/auto attendant fallback if none of your users answers the incoming
call. ;)

-MC

On Mon, Feb 7, 2011 at 11:11 AM, Larry Marshall <larry at newportpc.com> wrote:

> Michael,
>
>
>
> Thanks for your response. The cli tests are:
>
>
>
> freeswitch at internal> group call:01 at 192.168.10.29
>
> sofia/internal/sip:1002 at 192.168.10.105
>
> freeswitch at internal> sofia_contact 1013 at 192.168.10.29
>
> sofia/internal/sip:1013 at 192.168.10.107
>
> freeswitch at internal> group insert:01:1013 at 192.168.10.29
>
> +OK
>
> freeswitch at internal> group call:01 at 192.168.10.29
>
> sofia/internal/sip:1002 at 192.168.10.105
>
>
>
> I assume that after adding extension 1013 to group 01 it should be listed
> in the ?group call? function. Is this correct?
>
>
>
> Maybe I should start from the beginning, especially with your warning about
> the difference between the group and group_call APIs. I want to ring a group
> of extensions on an inbound call. I need to populate the members of this
> group based on time constraints (the current time of day, e.g. extension
> 1013 should only ring between 8:00AM and 5:00PM Monday-Friday), and ring the
> appropriate extensions. I wanted to try to do this within a lua script.
>
>
>
> Which group API should I look at?
>
>
>
> Thanks, Lars
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, February 07, 2011 8:05 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Group confusions
>
>
>
> I may be that the value you are trying to delete isn't "1013 at 192.168.10.29".
> To find out, go to fs_cli and do this:
>
>
>
> group call:01 at 192.168.10.29
>
>
>
> It will spit out all the sofia contacts in group 01. Most likely you have
> something like this:
>
> 1013 at 192.168.10.29:12345;rinstance=aabbccddeeff0011;transport=udp
>
>
>
> BTW, the value put into the group call when you dial 8101 is the same value
> that you find by doing this at fs_cli:
>
>
>
> sofia_contact 1013 at 192.168.10.29
>
>
>
> Go try it and let us now.
>
>
>
> -MC
>
>
>
> P.S. - the 'group' API and dp apps you are using are the same - they come
> from mod_db. They are *totally* different from the group_call API...
>
> On Fri, Feb 4, 2011 at 2:11 PM, Lars Zeb <larclap at yahoo.com> wrote:
>
> I went to two individual extensions and registered them for group 01 by
> dialing 8101. I then tested that group by dialing 8201 - both extensions
> rang.
>
> I then went to the cli and entered the command:
>
> group delete:01:1013 at 192.168.10.29
>
> where 1013 is one of the two extensions I had joined into group 01 earlier
> and the url is the domain and fs address. However, when I dialed 8201 after
> entering this cli command, both extensions rang. I had expected that only
> one would ring.
>
> I am confused. Are these two different types of "groups"?
>
> Thanks, Lars
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  7 21:47:15 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Mon, 7 Feb 2011 10:47:15 -0800
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <AANLkTimfzmpeR2N2sAf_QpAX0g3=BSRS0P-bPogh-Zbu@mail.gmail.com>
References: <4D4DC16F.50500@utoronto.ca>
	<AANLkTimfzmpeR2N2sAf_QpAX0g3=BSRS0P-bPogh-Zbu@mail.gmail.com>
Message-ID: <AANLkTimn62PSK2fVETpC5BpZrPO2n3Z5oux+Un0bpE62@mail.gmail.com>

Your sudo is not setting the enviroment to the root, which means /sbin and
/usr/sbin are not in your path (where those commands most likely are).  add
them to the path manually or properly login as root.

~Mitch

>
> On Sat, Feb 5, 2011 at 3:30 PM, Victor Chukalovskiy <
> victor.chukalovskiy at utoronto.ca> wrote:
>
>>  Hello,
>>
>> After purchasing a few licenses and installing the latest
>> fsg729-191-installer
>> I'm getting the following error when trying to load the mod_com_g729:
>> > 2011-02-04 14:27:17.637046 [CRIT] switch_loadable_module.c:926 Error
>> Loading module /opt/fs/mod/mod_com_g729.so
>> > **Trying to load an out of date module, please rebuild the module.**
>>
>> Also noticed that g729 installer ran with a couple errors:
>> > ./installer: line 62: ldconfig: command not found
>> > ./installer: line 49: useradd: command not found
>>  Any help or solution is much appreciated.
>>
>> -Victor
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  7 21:49:27 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 7 Feb 2011 12:49:27 -0600
Subject: [Freeswitch-users] Simple_conference.lua
In-Reply-To: <BLU143-W134E7A1B03729150AD7344A9EB0@phx.gbl>
References: <BLU143-W134E7A1B03729150AD7344A9EB0@phx.gbl>
Message-ID: <AANLkTi=3POMMMXhEHS5TMOOHwAdqV0B4kcqccSyi9pUv@mail.gmail.com>

On Mon, Feb 7, 2011 at 11:20 AM, Francois Barrouin <
francoisbarrouin at hotmail.fr> wrote:

>
> You could consider the following alternatives:
> - use TTS to generate your prompts:
> http://wiki.freeswitch.org/wiki/Categories
> - take the prompt files from Asterisk
>
> Surprising that FS doesn??t include these prompts.
>
Surprising that you didn't look hard enough. ;)

ls /usr/local/freeswitch/sounds/en/us/callie/conference/8000/
conf-alone.wav conf-has_left.wav conf-pin.wav
conf-bad-pin.wav conf-is-locked.wav conf-unmuted.wav
conf-enter_conf_number.wav conf-is-unlocked.wav conf-welcome.wav
conf-enter_conf_pin.wav conf-kicked.wav conf-you_are_already_muted.wav
conf-goodbye.wav conf-locked.wav conf-you_are_now_bidirectionally_muted.wav
conf-has_joined.wav conf-muted.wav


What we don't have, though, is a set of pre-assembled phrase macros. If you
guys whip some up we will be happy to include them in the default config. :)

-MC
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From anu at familytv.com  Mon Feb  7 21:16:34 2011
From: anu at familytv.com (Anirudha Shimpi)
Date: Mon, 7 Feb 2011 11:16:34 -0700
Subject: [Freeswitch-users]  mod_fsk not detecting call waiting caller id
Message-ID: <011301cbc6f3$27259c80$7570d580$@familytv.com>

Waiting for digit 'D' did not yield any results. Any other ideas? I am very
much interested in getting call waiting caller id to work with Freeswitch
and analog lines. Any suggestions would be very helpful. If there are any
modifications that can be done to mod_fsk to get it to work, or writing a
new module for this purpose, I am ready to help in any fashion. Thanks

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From daniel_wells at byu.edu  Mon Feb  7 22:12:35 2011
From: daniel_wells at byu.edu (Daniel Wells)
Date: Mon, 7 Feb 2011 12:12:35 -0700
Subject: [Freeswitch-users] iksemel.lib not compiling
Message-ID: <8C68232BC9314C40BBCDDAA480F7B01AEB20514865@harrow.exch.ad.byu.edu>

So I have been having issues with the iksemel.lib not compiling.  I found a site that gave some helps that fixed the issue (http://etmob.com/blog/index.php/2009/02/how-to-run-freeswitch-mod_dingaling-on-windows/).

The issue lies with the fact that the iksemel library depends on the pthread library which it doesn't know about.  Adding an include to the directory for the pthread project as an additional include to the iksemel project gets us closer.  The article (or rather the comments) says you then need to add a definition for ssize_t to the header file.  This did indeed solve the issue and all seemed to compile.

However this means one of two things is true: either the wiki article describing how to install google voice is incomplete, or something about my visual studio environment differs from those who wrote the wiki articles.

Any ideas on which?  If it is the wiki, I can go in and add the details.  However if it is my environment, then I would rather add instructions for making my environment match those that wrote the article.

Sincerely,


Daniel Wells
IT Systems Engineer
Office of Information Technology
Brigham Young University
daniel_wells at byu.edu ~ (801) 210-0326

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From jonyoung111 at gmail.com  Mon Feb  7 22:32:07 2011
From: jonyoung111 at gmail.com (Jon Young)
Date: Mon, 7 Feb 2011 12:32:07 -0700
Subject: [Freeswitch-users] Conference / Page usage
Message-ID: <AANLkTinz64-9NhK1-hxMEG6Rzb_SxjDymo6nHrrCq3eS@mail.gmail.com>

I utilized the "Mad Boss" intercom section of the sample dialplan to
set up a paging scenario.  This worked great on some slower PC's (and
slightly older FS builds).  I wa utilizing apx 25 stations on a PIII
650Mhz PC.  It takes a few seconds for the call setup, but otherwise
works and sounds great.

I Recently I attempted to move my paging setup (XML files) from the
PIII to a 2xXeon 3.4GHz server running VM.  My results surprised me.
The sound quality is great, but the call setup changed significantly.
While the PIII took a while to setup, I was able to communicate soon
after the call setup.  On the VM machine, the call setup goes quickly
and then I hear 25 conf. entrance tones on all of the destinations as
the conf. is set up.

Can I eliminate the conference announce tones and have just one
announce tone...or none?

The following is one of the paging groups I defined:

  <extension name="paging01">
    <condition field="destination_number" expression="^1999$">
	<action application="set" data="conference_auto_outcall_caller_id_name=Page1"/>
	<action application="set"
data="conference_auto_outcall_caller_id_number=1999"/>
	<action application="set" data="conference_auto_outcall_timeout=60"/>
	<action application="set" data="conference_auto_outcall_flags=mute"/>
	<action application="set"
data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app
2 a s1 transfer::intercept:${uuid} inline'}"/>
	<action application="set" data="sip_exclude_contact=${network_addr}"/>
	<action application="conference_set_auto_outcall"
data="${group_call(pagezone01)}"/>
	<action application="conference" data="paging1 at default+flags{endconf|deaf}"/>
    </condition>
  </extension>


From steveayre at gmail.com  Mon Feb  7 22:28:01 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 7 Feb 2011 19:28:01 +0000
Subject: [Freeswitch-users] close all conference from fs_cli
In-Reply-To: <57B06820CB8D4E9DA0A00A46FD7D3D67@e1705>
References: <313DB3017F77477FBC815E78800254C3@e1705>
	<AANLkTim+fQSOM2KyhnACfa-p9i6MSPTXwMchJBzdV=C2@mail.gmail.com>
	<57B06820CB8D4E9DA0A00A46FD7D3D67@e1705>
Message-ID: <C2551CDC-10F1-42A7-89E1-2EB3DFFCCACE@gmail.com>

Not sure what you mean? Are you connecting in via esl and wanting to detect when the socket closes? You can listen for the heartbeat event, use tcp keepalives or poll with status api. You tcp stack will already tell you if a tcp graceful close occurs.

Steve on iPhone

On 7 Feb 2011, at 18:10, "Madovsky" <infos at madovsky.org> wrote:

> yes it's what I planned to do.
> in fact as socket app doesn't have any ping test or check status property,
> in case of socket down the only way for now is to
> create a script that ping/telnet the connection status and if down
> run the right commands to clean what you do in ESL
> ----- Original Message -----
> From: Michael Collins
> To: FreeSWITCH Users Help
> Sent: Monday, February 07, 2011 1:04 PM
> Subject: Re: [Freeswitch-users] close all conference from fs_cli
> 
> There isn't a single command, but you could write a script that does a show conf to get a list of conferences and then do a conf xxx kick all on each one.
> -MC
> 
> On Sun, Feb 6, 2011 at 7:05 PM, Madovsky <infos at madovsky.org> wrote:
> Is there a way to close/hangup all active conferences
> from CLI ?
>  
> 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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From daniel_wells at byu.edu  Mon Feb  7 22:33:57 2011
From: daniel_wells at byu.edu (Daniel Wells)
Date: Mon, 7 Feb 2011 12:33:57 -0700
Subject: [Freeswitch-users] Setup Project
Message-ID: <8C68232BC9314C40BBCDDAA480F7B01AEB20514885@harrow.exch.ad.byu.edu>

So when I launch the Visual Studio 2010 Project I get an error telling me that the setup project cannot be opened (specifically the setup.wixproj file).  Without this project it would seem a rather complicated process to get my compiled code to run a separate box (compiling on dev box but want this to run on a separate server).

Any ideas?

Sincerely,


Daniel Wells

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From infos at madovsky.org  Mon Feb  7 22:41:34 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 7 Feb 2011 14:41:34 -0500
Subject: [Freeswitch-users] close all conference from fs_cli
References: <313DB3017F77477FBC815E78800254C3@e1705><AANLkTim+fQSOM2KyhnACfa-p9i6MSPTXwMchJBzdV=C2@mail.gmail.com><57B06820CB8D4E9DA0A00A46FD7D3D67@e1705>
	<C2551CDC-10F1-42A7-89E1-2EB3DFFCCACE@gmail.com>
Message-ID: <0E5F23B30298404B813A7886618D3528@e1705>

> Not sure what you mean? Are you connecting in via esl and wanting to detect when the socket closes?
no, in fact I have my own esl socket server in php so in case of this script fails I needed to clean the processes 
linked to esl server (hangup all conferences for example)
  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Monday, February 07, 2011 2:28 PM
  Subject: Re: [Freeswitch-users] close all conference from fs_cli


  Not sure what you mean? Are you connecting in via esl and wanting to detect when the socket closes? You can listen for the heartbeat event, use tcp keepalives or poll with status api. You tcp stack will already tell you if a tcp graceful close occurs.

  Steve on iPhone

  On 7 Feb 2011, at 18:10, "Madovsky" <infos at madovsky.org> wrote:


    yes it's what I planned to do.
    in fact as socket app doesn't have any ping test or check status property,
    in case of socket down the only way for now is to
    create a script that ping/telnet the connection status and if down
    run the right commands to clean what you do in ESL
      ----- Original Message ----- 
      From: Michael Collins 
      To: FreeSWITCH Users Help 
      Sent: Monday, February 07, 2011 1:04 PM
      Subject: Re: [Freeswitch-users] close all conference from fs_cli


      There isn't a single command, but you could write a script that does a show conf to get a list of conferences and then do a conf xxx kick all on each one. 
      -MC


      On Sun, Feb 6, 2011 at 7:05 PM, Madovsky <infos at madovsky.org> wrote:

        Is there a way to close/hangup all active conferences 
        from CLI ?

        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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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From kris at kriskinc.com  Mon Feb  7 22:49:36 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 7 Feb 2011 13:49:36 -0600
Subject: [Freeswitch-users] Opus codec in FreeSWITCH
In-Reply-To: <87hbcgr68j.fsf@jdc.jasonjgw.net>
References: <87hbcgr68j.fsf@jdc.jasonjgw.net>
Message-ID: <AANLkTikCWpYcDFDzMYU+MemBYVBHvB8TJaWKHNsVH2hN@mail.gmail.com>

For the forward error correction in Silk (and presumably Opus) to work
correctly the codec needs access to the frames around the lost
packets.  The codec also needs to communicate this loss to the encoder
at the remote end.  This is a trade-off between bandwidth usage and
the reconstruction of lost packets.

You'll notice the FreeSWITCH implementation of Silk (last time I
checked) hardcodes a packet loss rate of 10%.  This activates some FEC
functionality even if it's less than ideal.

Ideally FreeSWITCH could use and potentially dynamically activate the
jitter buffer to "peek" lost frames when transcoding to/from FEC
capable codecs (including Silk and Opus).

On Sun, Feb 6, 2011 at 7:19 PM, Jason White <jason at jasonjgw.net> wrote:
> I have started testing the Opus 0.9.0 experimental codec in a recent
> build of FreeSWITCH - thanks to Anthony Minessale for writing the
> module, and thanks to William F. Acker for helping with the testing.
>
> For those who aren't familiar with it, Opus is a combination of CELT and
> SILK, which is currently under discussion within the IETF.
>
> After building and loading the module, specify Opus-0.9.0 as your codec.
> The versioning of the codec's name is a good idea; this should prevent
> mutually incompatible versions from interacting with each other as
> development of the codec proceeds.
>
> My initial impressions are favourable; the audio quality is comparable
> to CELT at 48 khz, based on my early listening experience.
>
> I did notice some occasional pops in the audio, presumably attributable
> to packet loss. SILK apparently has some error correction functionality
> built in, but I don't know what its status is in the combined Opus
> codec. If it's present, then it didn't smooth out the glitches, at least
> in my environment.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 infos at madovsky.org  Mon Feb  7 23:24:26 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 7 Feb 2011 15:24:26 -0500
Subject: [Freeswitch-users] session_hangup_hook
Message-ID: <1FBCBE8A1A4344D7B788D736353AC2B5@e1705>

the caller from outside call a registered user,
I'd like to run a script once the caller hangup.
is session_hangup_hook available for outbound call also ?
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From sjmudd at pobox.com  Mon Feb  7 23:31:47 2011
From: sjmudd at pobox.com (Simon J Mudd)
Date: 07 Feb 2011 21:31:47 +0100
Subject: [Freeswitch-users] Confusing SIP auth failure logging message?
In-Reply-To: <AANLkTi=JgkwMbmtEsPZRA5NW9kjfbPJ_Uiqqp8ZB3zsB@mail.gmail.com>
References: <20110206232236.GA10501@mad06.wl0.org>
	<AANLkTi=JgkwMbmtEsPZRA5NW9kjfbPJ_Uiqqp8ZB3zsB@mail.gmail.com>
Message-ID: <m3hbcfegcs.fsf@mad06.wl0.org>

msc at freeswitch.org (Michael Collins) writes:

...

> No. This is just saying that there was a challenge, not that there was a
> failure. There is already a failure detection routine. To test it, setup a
> SIP client with an incorrect password. You'll see two log lines like this:
> 
> 2011-02-07 12:23:28.490029 [WARNING] sofia_reg.c:1247 SIP auth challenge
> (REGISTER) on sofia profile 'internal' for [1002 at 10.10.16.161] from ip
> 10.10.16.161
> 2011-02-07 12:23:29.035950 [WARNING] sofia_reg.c:1247 SIP auth challenge
> (REGISTER) on sofia profile 'internal' for [1002 at 10.10.16.161] from ip
> 10.10.16.161
> 2011-02-07 12:23:29.240695 [WARNING] sofia_reg.c:1205 SIP auth failure
> (REGISTER) on sofia profile 'internal' for [1002 at 10.10.16.161] from ip
> 10.10.16.161
> 
> This allows you to differentiate between the mere fact that an auth
> challenge was sent to the SIP client vs. the SIP client failing to auth.
> (Someone asked for that differentiation a while back - I don't know who or
> why...)

Thanks for the clarification. I must have missed the other message.
Having both makes sense for example if used with fail2ban.

Simon


From msc at freeswitch.org  Tue Feb  8 01:05:53 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 7 Feb 2011 16:05:53 -0600
Subject: [Freeswitch-users] session_hangup_hook
In-Reply-To: <1FBCBE8A1A4344D7B788D736353AC2B5@e1705>
References: <1FBCBE8A1A4344D7B788D736353AC2B5@e1705>
Message-ID: <AANLkTinZLssNUuouT58tR9uA+_dt+J-LGdz7OCu6wSyZ@mail.gmail.com>

Yes, although I am wondering if you mean something different. You set the
api_hangup_hook in the dialplan prior to bridging the call. Then, when
either side hangups up, the hangup hook is executed.

-MC

On Mon, Feb 7, 2011 at 2:24 PM, Madovsky <infos at madovsky.org> wrote:

>  the caller from outside call a registered user,
> I'd like to run a script once the caller hangup.
> is session_hangup_hook available for outbound call also ?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  8 00:47:20 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 7 Feb 2011 21:47:20 +0000
Subject: [Freeswitch-users] Confusing SIP auth failure logging message?
In-Reply-To: <m3hbcfegcs.fsf@mad06.wl0.org>
References: <20110206232236.GA10501@mad06.wl0.org>
	<AANLkTi=JgkwMbmtEsPZRA5NW9kjfbPJ_Uiqqp8ZB3zsB@mail.gmail.com>
	<m3hbcfegcs.fsf@mad06.wl0.org>
Message-ID: <4750E242-AFFF-4ADE-9AA1-3D1EB1E49546@gmail.com>

I expect it's to handle where an attacker keeps sending unauthenticated packets which generate a 407, but don't do a subsequent authenticated invite so never actually fail. Those can still be detected and blocked this way.

Steve on iPhone

On 7 Feb 2011, at 20:31, Simon J Mudd <sjmudd at pobox.com> wrote:

> msc at freeswitch.org (Michael Collins) writes:
> 
> ...
> 
>> No. This is just saying that there was a challenge, not that there was a
>> failure. There is already a failure detection routine. To test it, setup a
>> SIP client with an incorrect password. You'll see two log lines like this:
>> 
>> 2011-02-07 12:23:28.490029 [WARNING] sofia_reg.c:1247 SIP auth challenge
>> (REGISTER) on sofia profile 'internal' for [1002 at 10.10.16.161] from ip
>> 10.10.16.161
>> 2011-02-07 12:23:29.035950 [WARNING] sofia_reg.c:1247 SIP auth challenge
>> (REGISTER) on sofia profile 'internal' for [1002 at 10.10.16.161] from ip
>> 10.10.16.161
>> 2011-02-07 12:23:29.240695 [WARNING] sofia_reg.c:1205 SIP auth failure
>> (REGISTER) on sofia profile 'internal' for [1002 at 10.10.16.161] from ip
>> 10.10.16.161
>> 
>> This allows you to differentiate between the mere fact that an auth
>> challenge was sent to the SIP client vs. the SIP client failing to auth.
>> (Someone asked for that differentiation a while back - I don't know who or
>> why...)
> 
> Thanks for the clarification. I must have missed the other message.
> Having both makes sense for example if used with fail2ban.
> 
> Simon
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 fraserredmond at gmail.com  Tue Feb  8 01:23:34 2011
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Mon, 7 Feb 2011 17:23:34 -0500
Subject: [Freeswitch-users] hold_music var gets unset/lost by
	cancelled/failed att_xfer
Message-ID: <AANLkTimLod3yHrex4LAK=4gdVcPhzXutKYyw=E8ZMcN9@mail.gmail.com>

I'm trying to do a second att_xfer on a call so that if the first attended
transfer fails (c-leg is busy, or presses do-not-answer, or is an extn that
doesn't exist) then the call could be transferred to someone else.

On the first att_xfer the person on hold hears the hold_music correctly.
Once that transfer is cancelled or fails:
-- On any subsequent att_xfer's the person on hold just hears silence.
-- If they are put on hold they just hear silence.

I tried setting hold_music again for each channel after the first att_xfer,
but that didn't work, so it's probably not actually a problem with
hold_music per se, but some other variable/setting that decides whether to
use hold_music.

I also tried doing a uuid_dump before and after each attempt, but didn't
notice anything too different - unless it's a matter of unsetting one of the
couple of changed/new vars like:
variable_originate_disposition
variable_current_application
variable_playback_seconds

I get the feeling other variables are probably also lost by the first failed
transfer as the second att_xfer has some odd things happen if the third
party does answer. Haven't been able to narrow it down as closely as the
hold_music, but two things I've seen happen are:
-- The party that initiated the transfer gets hung up automatically (after
30 sec)
-- When the party that initiated the transfer hangs up it should connect the
other two parties, but instead it hung up all three

Cheers,
Fraser
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From anthony.minessale at gmail.com  Tue Feb  8 01:30:19 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 7 Feb 2011 16:30:19 -0600
Subject: [Freeswitch-users] hold_music var gets unset/lost by
 cancelled/failed att_xfer
In-Reply-To: <AANLkTimLod3yHrex4LAK=4gdVcPhzXutKYyw=E8ZMcN9@mail.gmail.com>
References: <AANLkTimLod3yHrex4LAK=4gdVcPhzXutKYyw=E8ZMcN9@mail.gmail.com>
Message-ID: <AANLkTi=u3VeHPuS5x1e+yu5QNBw5ZSgHvAWOT+F6A_OR@mail.gmail.com>

you really should report this to jira not to the mailing list.
http://jira.freeswitch.org



On Mon, Feb 7, 2011 at 4:23 PM, Fraser Redmond <fraserredmond at gmail.com> wrote:
> I'm trying to do a second att_xfer on a call so that if the first attended
> transfer fails (c-leg is busy, or presses do-not-answer, or is an extn that
> doesn't exist) then the call could be transferred to someone else.
>
> On the first att_xfer the person on hold hears the hold_music correctly.
> Once that transfer is cancelled or fails:
> -- On any subsequent att_xfer's the person on hold just hears silence.
> -- If they are put on hold they just hear silence.
>
> I tried setting hold_music again for each channel after the first att_xfer,
> but that didn't work, so it's probably not actually a problem with
> hold_music per se, but some other variable/setting that decides whether to
> use hold_music.
>
> I also tried doing a uuid_dump before and after each attempt, but didn't
> notice anything too different - unless it's a matter of unsetting one of the
> couple of changed/new vars like:
> variable_originate_disposition
> variable_current_application
> variable_playback_seconds
>
> I get the feeling other variables are probably also lost by the first failed
> transfer as the second att_xfer has some odd things happen if the third
> party does answer. Haven't been able to narrow it down as closely as the
> hold_music, but two things I've seen happen are:
> -- The party that initiated the transfer gets hung up automatically (after
> 30 sec)
> -- When the party that initiated the transfer hangs up it should connect the
> other two parties, but instead it hung up all three
>
> Cheers,
> Fraser
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Nabble at slickdeals.endjunk.com  Tue Feb  8 02:15:09 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 7 Feb 2011 15:15:09 -0800 (PST)
Subject: [Freeswitch-users] Using mod_opal with MSN Messenger
In-Reply-To: <AANLkTi=9nAY5ryTiYuCXxyat4yzu54KdGmXGPwz6WJgk@mail.gmail.com>
References: <1296867966381-5994575.post@n2.nabble.com>
	<4BDB6016-806B-4855-9A42-0A8201517F0B@visionutveckling.se>
	<AANLkTi=9nAY5ryTiYuCXxyat4yzu54KdGmXGPwz6WJgk@mail.gmail.com>
Message-ID: <1297120509137-6002230.post@n2.nabble.com>



Rhosyn wrote:
> I wonder if anyone is interested in writing a mod_msn?
Looks like someone had spent a great deal of time to work on a mod_msn as
shown 
http://freeswitch-users.2379917.n2.nabble.com/Using-Sofia-SIP-with-ICE-NAT-Traversal-Mechanism-td3699335.html#a3699418
here , but that's it and no more news/updates.

-----
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/Using-mod-opal-with-MSN-Messenger-tp5994575p6002230.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mitch.capper at gmail.com  Tue Feb  8 03:02:53 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Mon, 7 Feb 2011 16:02:53 -0800
Subject: [Freeswitch-users] Setup Project
In-Reply-To: <8C68232BC9314C40BBCDDAA480F7B01AEB20514885@harrow.exch.ad.byu.edu>
References: <8C68232BC9314C40BBCDDAA480F7B01AEB20514885@harrow.exch.ad.byu.edu>
Message-ID: <AANLkTi=nXpHYocghOoV7x47UCvQADzWcjCQJzjsVwrz5@mail.gmail.com>

Hi Daniel,
To build the setup you would need the windows installer toolset:
http://wix.sourceforge.net/  its free.   Note you could also just put the
compiled versions into a zip rather than build an installer for
distribution.

~Mitch

On Mon, Feb 7, 2011 at 11:33 AM, Daniel Wells <daniel_wells at byu.edu> wrote:

> So when I launch the Visual Studio 2010 Project I get an error telling me
> that the setup project cannot be opened (specifically the setup.wixproj
> file).  Without this project it would seem a rather complicated process to
> get my compiled code to run a separate box (compiling on dev box but want
> this to run on a separate server).
>
>
>
> Any ideas?
>
>
>
> Sincerely,
>
>
>
>
>
> *Daniel Wells*
>
>
>
>
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From infos at madovsky.org  Tue Feb  8 03:10:56 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 7 Feb 2011 19:10:56 -0500
Subject: [Freeswitch-users] session_hangup_hook
References: <1FBCBE8A1A4344D7B788D736353AC2B5@e1705>
	<AANLkTinZLssNUuouT58tR9uA+_dt+J-LGdz7OCu6wSyZ@mail.gmail.com>
Message-ID: <3C4DCADEFBD24418AAF6193FBEC540ED@e1705>

ok fine I wanted to be sure
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, February 07, 2011 5:05 PM
  Subject: Re: [Freeswitch-users] session_hangup_hook


  Yes, although I am wondering if you mean something different. You set the api_hangup_hook in the dialplan prior to bridging the call. Then, when either side hangups up, the hangup hook is executed.


  -MC


  On Mon, Feb 7, 2011 at 2:24 PM, Madovsky <infos at madovsky.org> wrote:

    the caller from outside call a registered user,
    I'd like to run a script once the caller hangup.
    is session_hangup_hook available for outbound call also ?

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






------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing 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 jeff at jefflenk.com  Tue Feb  8 03:18:43 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 7 Feb 2011 16:18:43 -0800 (PST)
Subject: [Freeswitch-users] iksemel.lib not compiling
In-Reply-To: <8C68232BC9314C40BBCDDAA480F7B01AEB20514865@harrow.exch.ad.byu.edu>
References: <8C68232BC9314C40BBCDDAA480F7B01AEB20514865@harrow.exch.ad.byu.edu>
Message-ID: <1297124323600-6002386.post@n2.nabble.com>


Daniel,

The Wiki needs to be corrected for the manual addition of GnuTLS under
windows. If you have a successful procedure and are willing to document it
that would be great.

Thanks
Jeff

-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/iksemel-lib-not-compiling-tp6001407p6002386.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From daniel_wells at byu.edu  Tue Feb  8 03:22:25 2011
From: daniel_wells at byu.edu (Daniel Wells)
Date: Mon, 7 Feb 2011 17:22:25 -0700
Subject: [Freeswitch-users] Setup Project
Message-ID: <r9bx1i4iytlp9ix9v4lnmbug.1297124557072@email.android.com>

I tried zipping them up and got an error (some missing dependancy that looked like it was related to Visual Studio).  I didn't have time at moment  to look into it much past that.  I will try again.

Do you know if everything it needed from gnutls was imported or will it need to be installed on any machine I move this to.  I'll look into the windows installer toolset.

Thanks for your help.

- Daniel

Mitch Capper <mitch.capper at gmail.com> wrote:



Hi Daniel,
To build the setup you would need the windows installer toolset: http://wix.sourceforge.net/  its free.   Note you could also just put the compiled versions into a zip rather than build an installer for distribution.

~Mitch

On Mon, Feb 7, 2011 at 11:33 AM, Daniel Wells <daniel_wells at byu.edu<mailto:daniel_wells at byu.edu>> wrote:
So when I launch the Visual Studio 2010 Project I get an error telling me that the setup project cannot be opened (specifically the setup.wixproj file).  Without this project it would seem a rather complicated process to get my compiled code to run a separate box (compiling on dev box but want this to run on a separate server).

Any ideas?

Sincerely,


Daniel Wells


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From daniel_wells at byu.edu  Tue Feb  8 03:25:37 2011
From: daniel_wells at byu.edu (Daniel Wells)
Date: Mon, 7 Feb 2011 17:25:37 -0700
Subject: [Freeswitch-users] iksemel.lib not compiling
Message-ID: <n0te4wpye2qd99op2drqg946.1297124749049@email.android.com>

Ok,

I'll try it a few more times to make sure I don't miss anything.

- Daniel W.

Jeff Lenk <jeff at jefflenk.com> wrote:


Daniel,

The Wiki needs to be corrected for the manual addition of GnuTLS under
windows. If you have a successful procedure and are willing to document it
that would be great.

Thanks
Jeff

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/iksemel-lib-not-compiling-tp6001407p6002386.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 mbsip at gazeta.pl  Tue Feb  8 03:34:55 2011
From: mbsip at gazeta.pl (Maciej Bylica)
Date: Tue, 8 Feb 2011 01:34:55 +0100
Subject: [Freeswitch-users] Freeswitch as a b2bua
In-Reply-To: <5691DFD2-55B3-4F8D-BB3D-E1F1F21BEFAE@ipeva.fr>
References: <AANLkTi=DF=5w=RJAVneskBe5=aALYJcye6H6ue0dMkB4@mail.gmail.com>
	<5691DFD2-55B3-4F8D-BB3D-E1F1F21BEFAE@ipeva.fr>
Message-ID: <AANLkTikgmQrdpaqW_yTTaGnna-=CNhQSkbCSrpEwjN8s@mail.gmail.com>

Hi David,

Thanks for clearing this up.
I am about to take a look on this.

Thx,
Maciej.

2011/2/7 David Ponzone <david.ponzone at ipeva.fr>:
> Yes, FreeSWITCH can do this.
> The config will be quite simple, particularly if you dont want it to do
> anything special, just route calls.
> It's always an issue to talk about CPS, but with a such host and given what
> I heard around, you should be able to reach 200 cps.
> The 64bits is stable, it's even the recommended one (on CentOs).
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 06/02/2011 ? 14:58, Maciej Bylica a ?crit :
>
> Hi,
>
> I am thinking about the way to have b2bua with full rtp proxy
> functionalities.
> The one wayout is to incorporate Freeswitch into the sip call flow.
> My scenario is as follows: Operator_1 ---> OpenSIPS ----> Operator_2
> Of course Opensips has B2b functionality but it may be pretty tricky
> to have this done.
> The question is could I place FS just after Opensips box like following:
> Operator_1 ---> OpenSIPS ----> Freewsitch ----> Operator_2 to achieve
> fully topology hiding (so no SIP/SDP ?information of Operator_2 should
> pass to Operator_1 and vice versa).
> Could I achieve this by using FS?
> Is the configuration pretty straightforward?
> What more less will be cps for Quad-core E5620 with 8GB of RAM for
> aforementioned scenario (here i guess Opensips could be unbeatable).
> Is 64bit installation a stable one?
>
> Thanks in advance,
> 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
>
>


From jeff at jefflenk.com  Tue Feb  8 03:52:01 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 7 Feb 2011 16:52:01 -0800 (PST)
Subject: [Freeswitch-users] Setup Project
In-Reply-To: <r9bx1i4iytlp9ix9v4lnmbug.1297124557072@email.android.com>
References: <8C68232BC9314C40BBCDDAA480F7B01AEB20514885@harrow.exch.ad.byu.edu>
	<r9bx1i4iytlp9ix9v4lnmbug.1297124557072@email.android.com>
Message-ID: <1297126321279-6002461.post@n2.nabble.com>


You have to install the crt libraries if you copied the files between
machines.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Setup-Project-tp6001483p6002461.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From daniel_wells at byu.edu  Tue Feb  8 04:41:24 2011
From: daniel_wells at byu.edu (Daniel Wells)
Date: Mon, 7 Feb 2011 18:41:24 -0700
Subject: [Freeswitch-users] Setup Project
In-Reply-To: <1297126321279-6002461.post@n2.nabble.com>
References: <8C68232BC9314C40BBCDDAA480F7B01AEB20514885@harrow.exch.ad.byu.edu>
	<r9bx1i4iytlp9ix9v4lnmbug.1297124557072@email.android.com>
	<1297126321279-6002461.post@n2.nabble.com>
Message-ID: <8C68232BC9314C40BBCDDAA480F7B01AEB20514A0B@harrow.exch.ad.byu.edu>

That definitely makes sense.  Thanks again.

    - Daniel Wells
 
 
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk
Sent: Monday, February 07, 2011 5:52 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setup Project


You have to install the crt libraries if you copied the files between machines.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Setup-Project-tp6001483p6002461.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_______________________________________________
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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 Nabble at slickdeals.endjunk.com  Tue Feb  8 05:44:54 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 7 Feb 2011 18:44:54 -0800 (PST)
Subject: [Freeswitch-users] Doxygen help
In-Reply-To: <b16156851002220200u3736acbdre1a1b82db0e82651@mail.gmail.com>
References: <4B820145.2090109@cartissolutions.com>
	<b16156851002220200u3736acbdre1a1b82db0e82651@mail.gmail.com>
Message-ID: <1297133094749-6002684.post@n2.nabble.com>


Sorry to bring up this old thread. However, I sure would like to know the
status of mod_msn. It sure would be nice to be able to let folks on MSN
network to call our SIP device through FS.

-----
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/Doxygen-help-tp4610125p6002684.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mitch.capper at gmail.com  Tue Feb  8 08:27:11 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Mon, 7 Feb 2011 21:27:11 -0800
Subject: [Freeswitch-users] Embedded Freeswitch .NET Example
Message-ID: <AANLkTi=+5pPf=2c-qFHfK-RMCF=ErL_B92uX9fLTuP=g@mail.gmail.com>

It took awhile but I have updated the Embedded Freeswitch wiki page (
http://wiki.freeswitch.com/wiki/Embedded_FreeSWITCH) with the notes from the
presentation a few weeks ago, at the bottom is also a link to a c# example
project that demos a softphone is around 100 lines of code that makes use of
most of what is talked about on that page.

~Mitch
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From daniel_wells at byu.edu  Tue Feb  8 09:38:01 2011
From: daniel_wells at byu.edu (Daniel Wells)
Date: Mon, 7 Feb 2011 23:38:01 -0700
Subject: [Freeswitch-users] mod_dinglaing errors and google voice
Message-ID: <8C68232BC9314C40BBCDDAA480F7B01AEB20514A13@harrow.exch.ad.byu.edu>

I sent this earlier but am not sure if it went though (I never got it in my inbox).  I apologize if you have already seen it.
____________________________________________________________________________________________

So I finally have what appears to be a working version of mod_dingaling on my windows box.  I say working only because the console starts and I see evidence in the logs that freeswitch is communicating with my google account though mod_dingaling  (I see presence information about some of my contacts).

However I am not sure my dialplan is working (just a guess since this my first experience with freeswitch-or any PBX for that matter).

I have been trying to follow the instructions as found here:
http://wiki.freeswitch.org/wiki/Google_Voice

I have a client connected using the default user 1000.  I created a dingaling profile called gv1000.xml (I am using the second example of the dialplan on the above site and inferred that the client profile needed to renamed).  My profile is identical to the one in the example with the exception of the username, password and "exten" (which is set to 1000).  I put a copy of the dial plan in the "default" directory.  But I am getting an error when trying to make outbound calls.  Does this mean I have done something wrong, or is google blocking this now?

freeswitch at dw-laptop> 2011-02-07 10:31:44.361514 [WARNING] sofia_reg.c:1247 SIP auth challenge (INVITE) on sofia profile 'internal' for [18XXXXXXXXX at 192.168.2.97] from ip 192.168.2.97
2011-02-07 10:31:44.477521 [NOTICE] switch_channel.c:811 New Channel sofia/internal/1000 at 192.168.2.97 [cfda05d4-2f61-4ab0-8ca6-f759cfac53b7]
2011-02-07 10:31:44.689533 [INFO] mod_dialplan_xml.c:331 Processing 1000 <1000>->18XXXXXXXXX in context default
2011-02-07 10:31:44.689533 [NOTICE] mod_sofia.c:2185 Ring-Ready sofia/internal/1000 at 192.168.2.97!
2011-02-07 10:31:44.689533 [NOTICE] mod_dptools.c:697 Ring Ready sofia/internal/1000 at 192.168.2.97!
2011-02-07 10:31:44.689533 [NOTICE] mod_dingaling.c:721 Close Channel N/A [CS_NEW]
2011-02-07 10:31:44.689533 [ERR] switch_ivr_originate.c:2638 Cannot create outgoing channel of type [dingaling] cause: [DESTINATION_OUT_OF_ORDER]
2011-02-07 10:31:44.689533 [INFO] mod_dptools.c:2621 Originate Failed.  Cause: DESTINATION_OUT_OF_ORDER
2011-02-07 10:31:44.689533 [NOTICE] mod_dptools.c:2684 Hangup sofia/internal/1000 at 192.168.2.97 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
2011-02-07 10:31:44.900545 [NOTICE] switch_core_session.c:1306 Session 1 (sofia/internal/1000 at 192.168.2.97) Ended
2011-02-07 10:31:44.900545 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/1000 at 192.168.2.97 [CS_DESTROY]

Sincerely,


Daniel Wells

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From kbdfck at gmail.com  Tue Feb  8 10:28:01 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Tue, 8 Feb 2011 10:28:01 +0300
Subject: [Freeswitch-users] hold_music var gets unset/lost by
 cancelled/failed att_xfer
In-Reply-To: <AANLkTi=u3VeHPuS5x1e+yu5QNBw5ZSgHvAWOT+F6A_OR@mail.gmail.com>
References: <AANLkTimLod3yHrex4LAK=4gdVcPhzXutKYyw=E8ZMcN9@mail.gmail.com>
	<AANLkTi=u3VeHPuS5x1e+yu5QNBw5ZSgHvAWOT+F6A_OR@mail.gmail.com>
Message-ID: <AANLkTi=EndkiUJvCgCSirxCA23EwWD_++0h3Qcr=mg9g@mail.gmail.com>

I have same issue with MOH and att_xfer on failed transfers, music on
hold played only once
At the same time, transfer_ringback always plays correctly to transferer

2011/2/8 Anthony Minessale <anthony.minessale at gmail.com>:
> you really should report this to jira not to the mailing list.
> http://jira.freeswitch.org
>
>
>
> On Mon, Feb 7, 2011 at 4:23 PM, Fraser Redmond <fraserredmond at gmail.com> wrote:
>> I'm trying to do a second att_xfer on a call so that if the first attended
>> transfer fails (c-leg is busy, or presses do-not-answer, or is an extn that
>> doesn't exist) then the call could be transferred to someone else.
>>
>> On the first att_xfer the person on hold hears the hold_music correctly.
>> Once that transfer is cancelled or fails:
>> -- On any subsequent att_xfer's the person on hold just hears silence.
>> -- If they are put on hold they just hear silence.
>>
>> I tried setting hold_music again for each channel after the first att_xfer,
>> but that didn't work, so it's probably not actually a problem with
>> hold_music per se, but some other variable/setting that decides whether to
>> use hold_music.
>>
>> I also tried doing a uuid_dump before and after each attempt, but didn't
>> notice anything too different - unless it's a matter of unsetting one of the
>> couple of changed/new vars like:
>> variable_originate_disposition
>> variable_current_application
>> variable_playback_seconds
>>
>> I get the feeling other variables are probably also lost by the first failed
>> transfer as the second att_xfer has some odd things happen if the third
>> party does answer. Haven't been able to narrow it down as closely as the
>> hold_music, but two things I've seen happen are:
>> -- The party that initiated the transfer gets hung up automatically (after
>> 30 sec)
>> -- When the party that initiated the transfer hangs up it should connect the
>> other two parties, but instead it hung up all three
>>
>> Cheers,
>> Fraser
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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


From kbdfck at gmail.com  Tue Feb  8 10:35:17 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Tue, 8 Feb 2011 10:35:17 +0300
Subject: [Freeswitch-users] FS ESL in Adobe Flex
In-Reply-To: <BLU143-W1221E99DA7D217FCD0D746A9EB0@phx.gbl>
References: <BLU143-W13F9C31C28DF6BFBCF736EA9EB0@phx.gbl>
	<C975763A.8C39%krice@freeswitch.org>
	<BLU143-W1221E99DA7D217FCD0D746A9EB0@phx.gbl>
Message-ID: <AANLkTinzZJtkUmBE3HcCq0xDayhh2rPVHQqrrK3AamhS@mail.gmail.com>

I think it is possible to use sockets available in Flex to connect to
the ESL socket.
There is the only thing to check - Flex socket security subsystem,
which looks for crossdomain permission string from socket it connects
if it fails to get permission from security server on port 843 or
something like that. Freeswitch won't provide crossdomain.xml from its
ESL socket, so connection will fail with security error if you don't
have inetd service, for example, responding with crossdomain.xml.

2011/2/7 Francois Barrouin <francoisbarrouin at hotmail.fr>:
> Thanks for your answer, I will go in that direction.
> I have already developed some code to?have a Flex application communicating
> with a Ruby On Rails server.?There is an ESL for ruby,?so it may be the best
> way for me. I have just to find out how to manage?FS?real time events?with
> RoR.
> Francois
>
> ________________________________
> Date: Mon, 7 Feb 2011 10:13:14 -0600
> From: krice at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] FS ESL in Adobe Flex
>
> You should not have a webclient such as flex diectly access ELS... ?Use a
> some sort of proxy such as Java or AMF-PHP based proxy to buffer between the
> 2... This will allow greater flexibility plus the use of Flex Object
> remoting. There are some examples of this in the swk contrib directory
> however they are a bit old
> K
>
>
> On 2/7/11 8:50 AM, "Francois Barrouin" <francoisbarrouin at hotmail.fr> wrote:
>
> Hello,
>
> I would like to use Adobe Flex to access FS event socket.
> I read in the (excellent!) ?FreeSWITCH 1.0.6? ?book that FS ESL is based on
> swig and unfortunately Adobe Flex is not a language supported by this
> project.
>
> I wonder if anyone has already developed a Flex ESL for FS?
>
> Thanks
>
> Fran?ois
>
> ________________________________
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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


From dujinfang at gmail.com  Tue Feb  8 10:45:57 2011
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 8 Feb 2011 15:45:57 +0800
Subject: [Freeswitch-users] FS ESL in Adobe Flex
In-Reply-To: <AANLkTinzZJtkUmBE3HcCq0xDayhh2rPVHQqrrK3AamhS@mail.gmail.com>
References: <BLU143-W13F9C31C28DF6BFBCF736EA9EB0@phx.gbl>
	<C975763A.8C39%krice@freeswitch.org>
	<BLU143-W1221E99DA7D217FCD0D746A9EB0@phx.gbl>
	<AANLkTinzZJtkUmBE3HcCq0xDayhh2rPVHQqrrK3AamhS@mail.gmail.com>
Message-ID: <AANLkTin2Aj+H=bMaoen3GDskuERC75-+x_xAT15S6A0M@mail.gmail.com>

Check http://wiki.freeswitch.org/wiki/FsAir, src in freeswitch-contrib

On Tue, Feb 8, 2011 at 3:35 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> I think it is possible to use sockets available in Flex to connect to
> the ESL socket.
> There is the only thing to check - Flex socket security subsystem,
> which looks for crossdomain permission string from socket it connects
> if it fails to get permission from security server on port 843 or
> something like that. Freeswitch won't provide crossdomain.xml from its
> ESL socket, so connection will fail with security error if you don't
> have inetd service, for example, responding with crossdomain.xml.
>
> 2011/2/7 Francois Barrouin <francoisbarrouin at hotmail.fr>:
>> Thanks for your answer, I will go in that direction.
>> I have already developed some code to?have a Flex application communicating
>> with a Ruby On Rails server.?There is an ESL for ruby,?so it may be the best
>> way for me. I have just to find out how to manage?FS?real time events?with
>> RoR.
>> Francois
>>
>> ________________________________
>> Date: Mon, 7 Feb 2011 10:13:14 -0600
>> From: krice at freeswitch.org
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] FS ESL in Adobe Flex
>>
>> You should not have a webclient such as flex diectly access ELS... ?Use a
>> some sort of proxy such as Java or AMF-PHP based proxy to buffer between the
>> 2... This will allow greater flexibility plus the use of Flex Object
>> remoting. There are some examples of this in the swk contrib directory
>> however they are a bit old
>> K
>>
>>
>> On 2/7/11 8:50 AM, "Francois Barrouin" <francoisbarrouin at hotmail.fr> wrote:
>>
>> Hello,
>>
>> I would like to use Adobe Flex to access FS event socket.
>> I read in the (excellent!) ?FreeSWITCH 1.0.6? ?book that FS ESL is based on
>> swig and unfortunately Adobe Flex is not a language supported by this
>> project.
>>
>> I wonder if anyone has already developed a Flex ESL for FS?
>>
>> Thanks
>>
>> Fran?ois
>>
>> ________________________________
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________ FreeSWITCH-users mailing
>> list FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>



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


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

Hi there,


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

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

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


< in swig.cs

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

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

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

Thanks,

?Frank 



      
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From yky1628 at yahoo.com  Tue Feb  8 06:55:02 2011
From: yky1628 at yahoo.com (Frankie Yiu)
Date: Mon, 7 Feb 2011 19:55:02 -0800 (PST)
Subject: [Freeswitch-users] C# problem for calling
	"freeswitch.switch_core_session_read_frame()"
Message-ID: <363776.37607.qm@web30506.mail.mud.yahoo.com>

Hi there,


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

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

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


< in swig.cs

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

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

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

Thanks,

?Frank 


      
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From admin at blindi.net  Tue Feb  8 09:26:16 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Tue, 8 Feb 2011 07:26:16 +0100 (CET)
Subject: [Freeswitch-users] how can enable nat for special users?
In-Reply-To: <AANLkTi=+5pPf=2c-qFHfK-RMCF=ErL_B92uX9fLTuP=g@mail.gmail.com>
References: <AANLkTi=+5pPf=2c-qFHfK-RMCF=ErL_B92uX9fLTuP=g@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1102080713070.11553@suse.blindi.net>

Hi, i have a FS on a public Ip i a datacenter.
My  users having problems to connect my Fs.
the users having dsl /cablemodems and UMTS.
I don.t find a Natparameter for example in asterisk "nat=yes"
to enable for a single user.
can you hlep me plese? thanks



---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From admin at blindi.net  Tue Feb  8 10:25:32 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Tue, 8 Feb 2011 08:25:32 +0100 (CET)
Subject: [Freeswitch-users] play_and_get_digits question
In-Reply-To: <C2CAC53C1A8B4CB3ADC3B5EF97F56DDD@e1705>
References: <EBED8C31D06F4A319D978371EE1CF24E@e1705>
	<AANLkTi=AVwJ4c80z9F_06+8EyUHrokLzEtRcPGGpa4jG@mail.gmail.com>
	<C2CAC53C1A8B4CB3ADC3B5EF97F56DDD@e1705>
Message-ID: <Pine.LNX.4.64.1102080811280.11715@suse.blindi.net>

Hi Madovsky,
I found a Securityproblem.
You enter for example 3 invalid attemps, FS don.t disconnect  the line. 
you become full access to the conference.
This is a securityrisk.
A hacker have manny fun to hack 
a conference pin or a pbx-system "to many failures call and fun-)".

I don.t find a parameter, for example: "max-failures" disconnect or give a 
prompt or transfer to a extension to protect the system.





From admin at blindi.net  Tue Feb  8 10:43:18 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Tue, 8 Feb 2011 08:43:18 +0100 (CET)
Subject: [Freeswitch-users] dialplanhelp for callforwarding
In-Reply-To: <AANLkTi=QbrjG0fA6y7TxwO6P1YL=VF93zJry=WPE0yNO@mail.gmail.com>
References: <8A02ED1AC7914B4BAD4EEF5D910D28DE@e1705>
	<Pine.LNX.4.64.1102070934130.31747@suse.blindi.net>
	<AANLkTi=QbrjG0fA6y7TxwO6P1YL=VF93zJry=WPE0yNO@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1102080835450.11715@suse.blindi.net>

Hi Michael,
thank you for you message.
I have a Handycap, i.m blind from birstday on, and i have barrieres to 
help myself and use the internet.
I working under debian textconsole, javachats, pdf-documens grafics is not 
working 
for me.
i don.t find a chatclient for the textconsole.
grafical clients or flash is not useabbility for screenreaders. i using 
suse-blinux, a screenreader under the textconsole.

Best Regards

---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From patrick.plattes at niemann-frey.info  Tue Feb  8 11:48:55 2011
From: patrick.plattes at niemann-frey.info (Patrick Plattes)
Date: Tue, 8 Feb 2011 09:48:55 +0100
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTi=oqw7fX-kvwqtxUXEWmzDu4C-dOjPrhpsCiwQc@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
	<AANLkTi=in3CLd-n0mT3SRKYA0kxLMGVWk9KeBx3_jhY6@mail.gmail.com>
	<AANLkTin3Bu+8EEr_v_GaRGNPzOjfcnfzsV2Av2cgnWgO@mail.gmail.com>
	<AANLkTi=OC_jx0+nzK405VZURLDJPD25R4ZjyJ3RXJ1nk@mail.gmail.com>
	<AANLkTi=Sm7CjoayiE8nHmL0Eehqknj1jdpbx0YBQJQGH@mail.gmail.com>
	<AANLkTinr0RdgBs2UiS3=b3i+k6uCwkMihDy_FbAOFPc_@mail.gmail.com>
	<AANLkTi=+aObW3kVBZq5ox4b4E1uFTSgtXUL72tHLOzPi@mail.gmail.com>
	<AANLkTikVDKS40dVsf8x=+5=qnW59LqJbSv1H=vtROFaM@mail.gmail.com>
	<AANLkTi=gJsVKCJifVaBL_H6ZnPHTD__aQPhPeVfdx-ju@mail.gmail.com>
	<AANLkTimeuCHf_7Y1TErPwrJbdGZ3DoQmeqQbJ1_3YNpT@mail.gmail.com>
	<AANLkTikDxzntMFRpFhfiqJ4pzvjsTQR4w_FfyLfT5Y7z@mail.gmail.com>
	<AANLkTimVrhpfX+0D2mFPwGHK1EDsALWjFGp8hh7=ecqt@mail.gmail.com>
	<AANLkTi=oqw7fX-kvwqtxUXEWmzDu4C-dOjPrhpsCiwQc@mail.gmail.com>
Message-ID: <AANLkTi=xwAromp8SDeExJEOx+nQNpQkr1hcOVmVHxiYB@mail.gmail.com>

Hi Chris :-),

it looks like a solution for us. I hope I will be able to test it
today. Which FS version are you using for this script? Did you try it
with 1.0.7?

Thanks,
 Patrick


From tayeb.meftah at gmail.com  Tue Feb  8 11:59:43 2011
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 08 Feb 2011 09:59:43 +0100
Subject: [Freeswitch-users] Routing calls acording to the divertion
	header
In-Reply-To: <2987F7D9-1A0A-475C-AA7F-E1BF8CB0D02A@ipeva.fr>
References: <4D4EE18E.9070606@gmail.com>
	<2987F7D9-1A0A-475C-AA7F-E1BF8CB0D02A@ipeva.fr>
Message-ID: <4D5105FF.3000402@gmail.com>

david,
this is a incomming did that insert "divertion:sip:xxxxx at x.com
i don't want to match the "to" destination number, but i want to match 
the divertion in the dialplan
so if the number was diverted to: "15148001395", i route it to client 1
and if is diverted to "213983200193" i route it to client 2.
that's the actual status
thank you
Le 06/02/2011 20:03, David Ponzone a ?crit :
> I think we need more details because your question does not make sense 
> to me ATM.
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 06/02/2011 ? 18:59, Meftah Tayeb a ?crit :
>
>> guys,
>> i have a divertion header like this:
>> Diversion:<sip:213983200193 at 12.13.14.15;user=phone>;reason=unknown;counter=1 
>>
>>
>> and another like:
>> Diversion:<sip:15148001395 at 12.13.14.15;user=phone>;reason=unknown;counter=1
>> how can i diferentiate calls acording to the divertion header?
>> thanks
>>
>> -- 
>> Meftah Tayeb
>> inum: +883510001288000
>> phone: +13477595883
>>
>>
>> _______________________________________________
>> 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
>    


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

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From fdelawarde at wirelessmundi.com  Tue Feb  8 12:11:21 2011
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Tue, 08 Feb 2011 10:11:21 +0100
Subject: [Freeswitch-users] Using mod_opal with MSN Messenger
In-Reply-To: <1297120509137-6002230.post@n2.nabble.com>
References: <1296867966381-5994575.post@n2.nabble.com>
	<4BDB6016-806B-4855-9A42-0A8201517F0B@visionutveckling.se>
	<AANLkTi=9nAY5ryTiYuCXxyat4yzu54KdGmXGPwz6WJgk@mail.gmail.com>
	<1297120509137-6002230.post@n2.nabble.com>
Message-ID: <1297156281.29302.258.camel@luna.tc.commsmundi.com>

On Mon, 2011-02-07 at 15:15 -0800, mazilo wrote:
> Rhosyn wrote:
> > I wonder if anyone is interested in writing a mod_msn?
> Looks like someone had spent a great deal of time to work on a mod_msn as
> shown 
> http://freeswitch-users.2379917.n2.nabble.com/Using-Sofia-SIP-with-ICE-NAT-Traversal-Mechanism-td3699335.html#a3699418
> here , but that's it and no more news/updates.

In case someone is interested in implementing it, apparently this
software supports proxying of the latest MSN protocol MSNP21. Might be
useful?
http://www.imspector.org

Fran?ois.



From david.ponzone at ipeva.fr  Tue Feb  8 12:20:22 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 8 Feb 2011 10:20:22 +0100
Subject: [Freeswitch-users] how can enable nat for special users?
In-Reply-To: <Pine.LNX.4.64.1102080713070.11553@suse.blindi.net>
References: <AANLkTi=+5pPf=2c-qFHfK-RMCF=ErL_B92uX9fLTuP=g@mail.gmail.com>
	<Pine.LNX.4.64.1102080713070.11553@suse.blindi.net>
Message-ID: <E272F29D-2DFA-4EAB-A5DC-A5F93FA5F098@ipeva.fr>

Thomas,

can you try not to hijack an existing thread when you post ?
You should send a new mail, not reply to an existing one.
I know about your disability, so I apologize if that's the reason you made the mistake.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 08/02/2011 ? 07:26, Thomas Hoellriegel a ?crit :

> Hi, i have a FS on a public Ip i a datacenter.
> My  users having problems to connect my Fs.
> the users having dsl /cablemodems and UMTS.
> I don.t find a Natparameter for example in asterisk "nat=yes"
> to enable for a single user.
> can you hlep me plese? thanks
> 
> 
> 
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/listinfo/blinde-misc
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.ponzone at ipeva.fr  Tue Feb  8 12:34:46 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 8 Feb 2011 10:34:46 +0100
Subject: [Freeswitch-users] Routing calls acording to the divertion
	header
In-Reply-To: <4D5105FF.3000402@gmail.com>
References: <4D4EE18E.9070606@gmail.com>
	<2987F7D9-1A0A-475C-AA7F-E1BF8CB0D02A@ipeva.fr>
	<4D5105FF.3000402@gmail.com>
Message-ID: <89216625-9282-4E2F-A529-E1053FFC7160@ipeva.fr>

Meftah,

I am not an expert of the Diversion RFC, but let's try to figure this out.
It's an incoming INVITE, coming from PSTN, to one of your DID ?
If the Diversion header is set there, it's just an indication about the number who redirected the call to your DID.
As far as I understand the RFC at least.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 08/02/2011 ? 09:59, Meftah Tayeb a ?crit :

> david,
> this is a incomming did that insert "divertion:sip:xxxxx at x.com
> i don't want to match the "to" destination number, but i want to match the divertion in the dialplan
> so if the number was diverted to: "15148001395", i route it to client 1
> and if is diverted to "213983200193" i route it to client 2.
> that's the actual status
> thank you
> Le 06/02/2011 20:03, David Ponzone a ?crit :
>> 
>> I think we need more details because your question does not make sense to me ATM.
>> 
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> 
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> 
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> 
>> 
>> 
>> 
>> Le 06/02/2011 ? 18:59, Meftah Tayeb a ?crit :
>> 
>>> guys,
>>> i have a divertion header like this:
>>> Diversion:<sip:213983200193 at 12.13.14.15;user=phone>;reason=unknown;counter=1 
>>> 
>>> and another like:
>>> Diversion:<sip:15148001395 at 12.13.14.15;user=phone>;reason=unknown;counter=1
>>> how can i diferentiate calls acording to the divertion header?
>>> thanks
>>> 
>>> -- 
>>> Meftah Tayeb
>>> inum: +883510001288000
>>> phone: +13477595883
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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

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From erik.dekkers at wvds.nl  Tue Feb  8 12:54:25 2011
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Tue, 8 Feb 2011 10:54:25 +0100
Subject: [Freeswitch-users] dialplanhelp for callforwarding
In-Reply-To: <Pine.LNX.4.64.1102080835450.11715@suse.blindi.net>
References: <8A02ED1AC7914B4BAD4EEF5D910D28DE@e1705>
	<Pine.LNX.4.64.1102070934130.31747@suse.blindi.net>
	<AANLkTi=QbrjG0fA6y7TxwO6P1YL=VF93zJry=WPE0yNO@mail.gmail.com>
	<Pine.LNX.4.64.1102080835450.11715@suse.blindi.net>
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE0787301868@FS1.wvds.local>

Hi Thoms,

If you need help getting on IRC you can contact tayeb.meftah at gmail.com . He's also blind but can use IRC with his screenreader.
Tayeb confirmed it's not a problem and he's glad to help you

Success and we'll see you on IRC.

Kind regards,

Erik

-----Oorspronkelijk bericht-----
Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Thomas Hoellriegel
Verzonden: dinsdag 8 februari 2011 8:43
Aan: FreeSWITCH Users Help
Onderwerp: Re: [Freeswitch-users] dialplanhelp for callforwarding

Hi Michael,
thank you for you message.
I have a Handycap, i.m blind from birstday on, and i have barrieres to help myself and use the internet.
I working under debian textconsole, javachats, pdf-documens grafics is not working for me.
i don.t find a chatclient for the textconsole.
grafical clients or flash is not useabbility for screenreaders. i using suse-blinux, a screenreader under the textconsole.

Best Regards

---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc


From victor.chukalovskiy at utoronto.ca  Tue Feb  8 15:47:27 2011
From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy)
Date: Tue, 08 Feb 2011 07:47:27 -0500
Subject: [Freeswitch-users] how can enable nat for special users?
In-Reply-To: <Pine.LNX.4.64.1102080713070.11553@suse.blindi.net>
References: <AANLkTi=+5pPf=2c-qFHfK-RMCF=ErL_B92uX9fLTuP=g@mail.gmail.com>
	<Pine.LNX.4.64.1102080713070.11553@suse.blindi.net>
Message-ID: <4D513B5F.2090503@utoronto.ca>

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From david.ponzone at ipeva.fr  Tue Feb  8 16:05:56 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 8 Feb 2011 14:05:56 +0100
Subject: [Freeswitch-users] how can enable nat for special users?
In-Reply-To: <4D513B5F.2090503@utoronto.ca>
References: <AANLkTi=+5pPf=2c-qFHfK-RMCF=ErL_B92uX9fLTuP=g@mail.gmail.com>
	<Pine.LNX.4.64.1102080713070.11553@suse.blindi.net>
	<4D513B5F.2090503@utoronto.ca>
Message-ID: <F70C2425-C9CB-4EFE-80E7-2171ACB454F2@ipeva.fr>

I dont really see the point in enabling that per user.
FreeSWITCH will cope with clients behind NAT automagically.
It's possible there are situations where this will lead to issues, but it's rare and generally, it's because the endpoint is faulty (like the router has a SIP ALG enabled or the phone does not send rport).
There is no issue to use the NAT-Traversal function of FreeSWITCH (actually, it's called rtp-auto-adjust) with devices which are not behind NAT.
I do that everyday.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 08/02/2011 ? 13:47, Victor Chukalovskiy a ?crit :

> Hi,
> 
> I was not able to find a per-user parameter, 
> but if you know IP address/range of your user network it is possible to use apply-nat-acl and edit appropriate ACL list.
> 
> Also, did you tried agressive-nat-detecetion?
> 
> -Victor
> 
> 
> On 08/02/11 01:26 AM, Thomas Hoellriegel wrote:
>> 
>> Hi, i have a FS on a public Ip i a datacenter. 
>> My  users having problems to connect my Fs. 
>> the users having dsl /cablemodems and UMTS. 
>> I don.t find a Natparameter for example in asterisk "nat=yes" 
>> to enable for a single user. 
>> can you hlep me plese? thanks 
>> 
>> 
>> 
>> --------------- 
>> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: 
>> http://www.blindi.net/callback 
>> homepage: http://www.blindi.net 
>> blinde-misc mailingliste f?r blinde. anmeldung unter: 
>> http://www.blindi.net/mailman/listinfo/blinde-misc 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  8 18:02:57 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 8 Feb 2011 20:32:57 +0530
Subject: [Freeswitch-users] G729
Message-ID: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>

Hi,

Which type of  G.729 codec does freeswitch uses in passthrough mode ?
Is it g729a or g729b or g729ab .

Regards
Sam
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From infos at madovsky.org  Tue Feb  8 18:11:17 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 8 Feb 2011 10:11:17 -0500
Subject: [Freeswitch-users] play_and_get_digits question
References: <EBED8C31D06F4A319D978371EE1CF24E@e1705><AANLkTi=AVwJ4c80z9F_06+8EyUHrokLzEtRcPGGpa4jG@mail.gmail.com><C2CAC53C1A8B4CB3ADC3B5EF97F56DDD@e1705>
	<Pine.LNX.4.64.1102080811280.11715@suse.blindi.net>
Message-ID: <5DCA5F42FA5A4AAA8D5F34DFDDA51157@e1705>

Hi Thomas,

I resolved this in my last email
please check...
you can limit calls with limit app to avoid
hacks

thanks

----- Original Message ----- 
From: "Thomas Hoellriegel" <admin at blindi.net>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, February 08, 2011 2:25 AM
Subject: Re: [Freeswitch-users] play_and_get_digits question


> Hi Madovsky,
> I found a Securityproblem.
> You enter for example 3 invalid attemps, FS don.t disconnect  the line.
> you become full access to the conference.
> This is a securityrisk.
> A hacker have manny fun to hack
> a conference pin or a pbx-system "to many failures call and fun-)".
>
> I don.t find a parameter, for example: "max-failures" disconnect or give a
> prompt or transfer to a extension to protect the system.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Tue Feb  8 18:12:25 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 8 Feb 2011 10:12:25 -0500
Subject: [Freeswitch-users] how can enable nat for special users?
References: <AANLkTi=+5pPf=2c-qFHfK-RMCF=ErL_B92uX9fLTuP=g@mail.gmail.com>
	<Pine.LNX.4.64.1102080713070.11553@suse.blindi.net>
Message-ID: <480BCD99812647829429B4B4BC2A1E4D@e1705>

your user must you nat traversal.
stun for example

----- Original Message ----- 
From: "Thomas Hoellriegel" <admin at blindi.net>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, February 08, 2011 1:26 AM
Subject: [Freeswitch-users] how can enable nat for special users?


Hi, i have a FS on a public Ip i a datacenter.
My  users having problems to connect my Fs.
the users having dsl /cablemodems and UMTS.
I don.t find a Natparameter for example in asterisk "nat=yes"
to enable for a single user.
can you hlep me plese? thanks



---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc



--------------------------------------------------------------------------------


> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb  8 18:31:46 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 8 Feb 2011 07:31:46 -0800
Subject: [Freeswitch-users] G729
In-Reply-To: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>
References: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>
Message-ID: <AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>

G729 only. No A's or AB.

If you want to pass through other variants of G.729, you can simply
modify mod_g729.c to include additional variants of G729

Which by the way, I've pasted the area of the code you might want to
modify. I've done this on mine and it worked fine as pass through.

00217         SWITCH_ADD_CODEC(codec_interface, "G.729A");
00218         for (count = 12; count > 0; count--) {
00219                 switch_core_codec_add_implementation(pool,
codec_interface,
00220
                        SWITCH_CODEC_TYPE_AUDIO, 18, "G729A", NULL,
8000, 8000, 8000,
00221
                        mpf * count, spf * count, bpf * count, ebpf *
count, 1, count * 10,
00222
                        switch_g729_init, switch_g729_encode,
switch_g729_decode, switch_g729_destroy);
00223         }


On Tue, Feb 8, 2011 at 7:02 AM, Sam <u2nsam at gmail.com> wrote:
> Hi,
>
> Which type of? G.729 codec does freeswitch uses in passthrough mode ?
> Is it g729a or g729b or g729ab .
>
> 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
>
>


From david.ponzone at ipeva.fr  Tue Feb  8 18:36:07 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 8 Feb 2011 16:36:07 +0100
Subject: [Freeswitch-users] G729
In-Reply-To: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>
References: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>
Message-ID: <AC48A711-80CA-4FA6-B82B-2FAB4B9DAD78@ipeva.fr>

Sam,

Passthrough, as its name clearly states, is passthrough.
G729 A/B/AB share the same SDP PT (18).
Also, G729 and G729A are compatible.
And B is just the VAD support.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 08/02/2011 ? 16:02, Sam a ?crit :

> Hi,
> 
> Which type of  G.729 codec does freeswitch uses in passthrough mode ? 
> Is it g729a or g729b or g729ab .
> 
> 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 david.ponzone at ipeva.fr  Tue Feb  8 18:39:04 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 8 Feb 2011 16:39:04 +0100
Subject: [Freeswitch-users] G729
In-Reply-To: <AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>
References: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>
	<AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>
Message-ID: <59C40C2C-351C-4544-8B50-099DACD6A292@ipeva.fr>

Ah so, I just replied him the opposite :)
Are you sure about that ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 08/02/2011 ? 16:31, curriegrad2004 a ?crit :

> G729 only. No A's or AB.
> 
> If you want to pass through other variants of G.729, you can simply
> modify mod_g729.c to include additional variants of G729
> 
> Which by the way, I've pasted the area of the code you might want to
> modify. I've done this on mine and it worked fine as pass through.
> 
> 00217         SWITCH_ADD_CODEC(codec_interface, "G.729A");
> 00218         for (count = 12; count > 0; count--) {
> 00219                 switch_core_codec_add_implementation(pool,
> codec_interface,
> 00220
>                        SWITCH_CODEC_TYPE_AUDIO, 18, "G729A", NULL,
> 8000, 8000, 8000,
> 00221
>                        mpf * count, spf * count, bpf * count, ebpf *
> count, 1, count * 10,
> 00222
>                        switch_g729_init, switch_g729_encode,
> switch_g729_decode, switch_g729_destroy);
> 00223         }
> 
> 
> On Tue, Feb 8, 2011 at 7:02 AM, Sam <u2nsam at gmail.com> wrote:
>> Hi,
>> 
>> Which type of  G.729 codec does freeswitch uses in passthrough mode ?
>> Is it g729a or g729b or g729ab .
>> 
>> 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 david.ponzone at ipeva.fr  Tue Feb  8 18:44:08 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 8 Feb 2011 16:44:08 +0100
Subject: [Freeswitch-users] how can enable nat for special users?
In-Reply-To: <480BCD99812647829429B4B4BC2A1E4D@e1705>
References: <AANLkTi=+5pPf=2c-qFHfK-RMCF=ErL_B92uX9fLTuP=g@mail.gmail.com>
	<Pine.LNX.4.64.1102080713070.11553@suse.blindi.net>
	<480BCD99812647829429B4B4BC2A1E4D@e1705>
Message-ID: <A837D9D7-AA05-4013-9DCE-F9ACBD24CE83@ipeva.fr>

No no no.
In most situations (99%) you don't need STUN.
There are 2 ways to defeat NAT:
-a client way (STUN) where the client endpoint is going to send a corrected INVITE
-the server way (often called Autolearn, NAT autodetection, ...) where the rtp forwarder waits for inbound RTP coming into the port it opened for the call, and then uses that IP:port to send back the opposite stream

FreeSWITCH implements autolearn (rtp-auto-adjust) and it is quite excellent.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 08/02/2011 ? 16:12, Madovsky a ?crit :

> your user must you nat traversal.
> stun for example
> 
> ----- Original Message ----- 
> From: "Thomas Hoellriegel" <admin at blindi.net>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, February 08, 2011 1:26 AM
> Subject: [Freeswitch-users] how can enable nat for special users?
> 
> 
> Hi, i have a FS on a public Ip i a datacenter.
> My  users having problems to connect my Fs.
> the users having dsl /cablemodems and UMTS.
> I don.t find a Natparameter for example in asterisk "nat=yes"
> to enable for a single user.
> can you hlep me plese? thanks
> 
> 
> 
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/listinfo/blinde-misc
> 
> 
> 
> --------------------------------------------------------------------------------
> 
> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  8 19:09:38 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 8 Feb 2011 16:09:38 +0000
Subject: [Freeswitch-users] G729
In-Reply-To: <AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>
References: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>
	<AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>
Message-ID: <AANLkTimKH61TNx_BjOthqMOYJe=t_WkD7qBRL-GKbrFa@mail.gmail.com>

Annex A is just an alternative way of encoding the data, using less
processing power but at poorer quality. They generate compatible data -
decoders both with and without annex a support decode both G729 and G729A
encoded data. The difference is purely in the implementation of the
encoder/decoder. If your device uses G729A in the SDP, it's broken.

Annex B uses the same PT & name. It does work in passthrough mode, as does
plain G729.

-Steve


On 8 February 2011 15:31, curriegrad2004 <curriegrad2004 at gmail.com> wrote:

> G729 only. No A's or AB.
>
> If you want to pass through other variants of G.729, you can simply
> modify mod_g729.c to include additional variants of G729
>
> Which by the way, I've pasted the area of the code you might want to
> modify. I've done this on mine and it worked fine as pass through.
>
> 00217         SWITCH_ADD_CODEC(codec_interface, "G.729A");
> 00218         for (count = 12; count > 0; count--) {
> 00219                 switch_core_codec_add_implementation(pool,
> codec_interface,
> 00220
>                        SWITCH_CODEC_TYPE_AUDIO, 18, "G729A", NULL,
> 8000, 8000, 8000,
> 00221
>                        mpf * count, spf * count, bpf * count, ebpf *
> count, 1, count * 10,
> 00222
>                        switch_g729_init, switch_g729_encode,
> switch_g729_decode, switch_g729_destroy);
> 00223         }
>
>
> On Tue, Feb 8, 2011 at 7:02 AM, Sam <u2nsam at gmail.com> wrote:
> > Hi,
> >
> > Which type of  G.729 codec does freeswitch uses in passthrough mode ?
> > Is it g729a or g729b or g729ab .
> >
> > 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 infos at madovsky.org  Tue Feb  8 19:11:53 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 8 Feb 2011 11:11:53 -0500
Subject: [Freeswitch-users] nibblebill sql requests
Message-ID: <AD71F946CFA942858324FAC60CF95D76@e1705>

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
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From diego.viola at gmail.com  Tue Feb  8 19:29:35 2011
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 8 Feb 2011 13:29:35 -0300
Subject: [Freeswitch-users] We need help with the wiki,
	looking for new administrators
Message-ID: <AANLkTinuBj6BqEj_gRzjcEE_6Y3G++kJYZ5dg4KPLSCw@mail.gmail.com>

Hello,

As some of you might know, I'm one of the administrators of the
FreeSWITCH wiki; for the past months I've been trying to keep the wiki
clean from spammers and related abuses, which means I spent my time
blocking spammers, deleting spam pages, and improving the wiki, etc.

Recently I got a job which occupies most of my time so I won't have
the time that I had before for doing such things, so if someone have
the time for this task please reply on this thread so we can grant you
administrator rights.

This task isn't hard and it's basically as follows: keep an eye at
"Recent changes" here:
http://wiki.freeswitch.org/wiki/Special:RecentChanges -- there is also
a "Recent changes" link in the main page and in every wiki page, then
when you find someone spammed the wiki, block that user with
indefinite time, and then delete his/her spam page or the affected
pages.

Please contact us if you're interested in helping.

Sincerely,
Diego Viola


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

Hello all!

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

Thanks,
MC
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From Nabble at slickdeals.endjunk.com  Tue Feb  8 19:48:55 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 8 Feb 2011 08:48:55 -0800 (PST)
Subject: [Freeswitch-users] git-a93623 version (2011/02/07) causes
	[DESTINATION_OUT_OF_ORDER]
Message-ID: <1297183735886-6004663.post@n2.nabble.com>


This morning, when I did a git pull on my local FS git root directory, I
noticed there is a significant update (including the main switch and/or
ivr). So, I went a head to compile/install/run this git version to find out
the following error message through fs_cli when I placed an outbound call:

FreeSWITCH Version 1.0.head (git-a936236 2011-02-07 16-33-45 -0600)
2011-02-08 11:42:05.752459 [CRIT] mod_sofia.c:3844 Error Creating Session
2011-02-08 11:42:05.752459 [ERR] switch_ivr_originate.c:2638 Cannot create
outgoing channel of type [sofia] cause: [DESTINATION_OUT_OF_ORDER]
The dialplan context works with previous git version. So, do I need to make
changes to my dialplan context?

-----
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/git-a93623-version-2011-02-07-causes-DESTINATION-OUT-OF-ORDER-tp6004663p6004663.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Tue Feb  8 19:51:11 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 8 Feb 2011 10:51:11 -0600
Subject: [Freeswitch-users] git-a93623 version (2011/02/07) causes
	[DESTINATION_OUT_OF_ORDER]
In-Reply-To: <1297183735886-6004663.post@n2.nabble.com>
References: <1297183735886-6004663.post@n2.nabble.com>
Message-ID: <AANLkTimOM9dZYr=zrQB+eJAEQ7fh0ugvqJQueOtOGiZH@mail.gmail.com>

no, either you did fsctl pause or are at your session limit.
Did you use "make current"


On Tue, Feb 8, 2011 at 10:48 AM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
> This morning, when I did a git pull on my local FS git root directory, I
> noticed there is a significant update (including the main switch and/or
> ivr). So, I went a head to compile/install/run this git version to find out
> the following error message through fs_cli when I placed an outbound call:
>
> FreeSWITCH Version 1.0.head (git-a936236 2011-02-07 16-33-45 -0600)
> 2011-02-08 11:42:05.752459 [CRIT] mod_sofia.c:3844 Error Creating Session
> 2011-02-08 11:42:05.752459 [ERR] switch_ivr_originate.c:2638 Cannot create
> outgoing channel of type [sofia] cause: [DESTINATION_OUT_OF_ORDER]
> The dialplan context works with previous git version. So, do I need to make
> changes to my dialplan context?
>
> -----
> 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/git-a93623-version-2011-02-07-causes-DESTINATION-OUT-OF-ORDER-tp6004663p6004663.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 max.bridgewater at gmail.com  Tue Feb  8 19:52:25 2011
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Tue, 8 Feb 2011 11:52:25 -0500
Subject: [Freeswitch-users] UPDATE: Buy the devs dinner
In-Reply-To: <AANLkTimqurw-1TdPyYNXw7C6y3tHy9kufqJf9AhP03yD@mail.gmail.com>
References: <AANLkTimqurw-1TdPyYNXw7C6y3tHy9kufqJf9AhP03yD@mail.gmail.com>
Message-ID: <AANLkTi=njM19AMx=WGRDhzm54KJNqiz__cG5n6noWkVa@mail.gmail.com>

Can somebody share the pitch in details? Paypal would be best. Too
lazy to search for ;)

Sorry.
Max.

On Tue, Feb 8, 2011 at 11:42 AM, Michael Collins <msc at freeswitch.org> wrote:
> Hello all!
> Just an update: all the FreeSWITCH developers are together this week and
> will be going to dinner. It's still not too late for those who wish to pitch
> in and help feed them. Just hit the Paypal link on the main page and drop a
> few dollars!
> Thanks,
> MC
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb  8 19:55:09 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 8 Feb 2011 10:55:09 -0600
Subject: [Freeswitch-users] Conference / Page usage
In-Reply-To: <AANLkTinz64-9NhK1-hxMEG6Rzb_SxjDymo6nHrrCq3eS@mail.gmail.com>
References: <AANLkTinz64-9NhK1-hxMEG6Rzb_SxjDymo6nHrrCq3eS@mail.gmail.com>
Message-ID: <AANLkTi=2dNSCE=pH2r60dYHke7oWbvy6YzsM1fhVjFZL@mail.gmail.com>

the announce sounds are in conference.conf.xml in the profile prefs.


On Mon, Feb 7, 2011 at 1:32 PM, Jon Young <jonyoung111 at gmail.com> wrote:
> I utilized the "Mad Boss" intercom section of the sample dialplan to
> set up a paging scenario. ?This worked great on some slower PC's (and
> slightly older FS builds). ?I wa utilizing apx 25 stations on a PIII
> 650Mhz PC. ?It takes a few seconds for the call setup, but otherwise
> works and sounds great.
>
> I Recently I attempted to move my paging setup (XML files) from the
> PIII to a 2xXeon 3.4GHz server running VM. ?My results surprised me.
> The sound quality is great, but the call setup changed significantly.
> While the PIII took a while to setup, I was able to communicate soon
> after the call setup. ?On the VM machine, the call setup goes quickly
> and then I hear 25 conf. entrance tones on all of the destinations as
> the conf. is set up.
>
> Can I eliminate the conference announce tones and have just one
> announce tone...or none?
>
> The following is one of the paging groups I defined:
>
> ?<extension name="paging01">
> ? ?<condition field="destination_number" expression="^1999$">
> ? ? ? ?<action application="set" data="conference_auto_outcall_caller_id_name=Page1"/>
> ? ? ? ?<action application="set"
> data="conference_auto_outcall_caller_id_number=1999"/>
> ? ? ? ?<action application="set" data="conference_auto_outcall_timeout=60"/>
> ? ? ? ?<action application="set" data="conference_auto_outcall_flags=mute"/>
> ? ? ? ?<action application="set"
> data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app
> 2 a s1 transfer::intercept:${uuid} inline'}"/>
> ? ? ? ?<action application="set" data="sip_exclude_contact=${network_addr}"/>
> ? ? ? ?<action application="conference_set_auto_outcall"
> data="${group_call(pagezone01)}"/>
> ? ? ? ?<action application="conference" data="paging1 at default+flags{endconf|deaf}"/>
> ? ?</condition>
> ?</extension>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Tue Feb  8 19:56:13 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 8 Feb 2011 11:56:13 -0500
Subject: [Freeswitch-users] UPDATE: Buy the devs dinner
References: <AANLkTimqurw-1TdPyYNXw7C6y3tHy9kufqJf9AhP03yD@mail.gmail.com>
Message-ID: <4C9E950404614CD89D6299213F3FD878@e1705>

Sorry Mike, I live too far to be at the dinner...
hope you ll enjoy !
  ----- Original Message ----- 
  From: Michael Collins 
  To: freeswitch-users at lists.freeswitch.org ; freeswitch-dev at lists.freeswitch.org 
  Sent: Tuesday, February 08, 2011 11:42 AM
  Subject: [Freeswitch-users] UPDATE: Buy the devs dinner


  Hello all!


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


  Thanks,
  MC


------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.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 Feb  8 20:00:10 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 8 Feb 2011 12:00:10 -0500
Subject: [Freeswitch-users] Conference / Page usage
References: <AANLkTinz64-9NhK1-hxMEG6Rzb_SxjDymo6nHrrCq3eS@mail.gmail.com>
	<AANLkTi=2dNSCE=pH2r60dYHke7oWbvy6YzsM1fhVjFZL@mail.gmail.com>
Message-ID: <F665F14EBCCA4D9ABC9125D2C6B61B0E@e1705>

about this,
if I use say: in alone-sounds it doesn't work. (other params work)
with files it works

Thanks


----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, February 08, 2011 11:55 AM
Subject: Re: [Freeswitch-users] Conference / Page usage


the announce sounds are in conference.conf.xml in the profile prefs.


On Mon, Feb 7, 2011 at 1:32 PM, Jon Young <jonyoung111 at gmail.com> wrote:
> I utilized the "Mad Boss" intercom section of the sample dialplan to
> set up a paging scenario. This worked great on some slower PC's (and
> slightly older FS builds). I wa utilizing apx 25 stations on a PIII
> 650Mhz PC. It takes a few seconds for the call setup, but otherwise
> works and sounds great.
>
> I Recently I attempted to move my paging setup (XML files) from the
> PIII to a 2xXeon 3.4GHz server running VM. My results surprised me.
> The sound quality is great, but the call setup changed significantly.
> While the PIII took a while to setup, I was able to communicate soon
> after the call setup. On the VM machine, the call setup goes quickly
> and then I hear 25 conf. entrance tones on all of the destinations as
> the conf. is set up.
>
> Can I eliminate the conference announce tones and have just one
> announce tone...or none?
>
> The following is one of the paging groups I defined:
>
> <extension name="paging01">
> <condition field="destination_number" expression="^1999$">
> <action application="set" 
> data="conference_auto_outcall_caller_id_name=Page1"/>
> <action application="set"
> data="conference_auto_outcall_caller_id_number=1999"/>
> <action application="set" data="conference_auto_outcall_timeout=60"/>
> <action application="set" data="conference_auto_outcall_flags=mute"/>
> <action application="set"
> data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app
> 2 a s1 transfer::intercept:${uuid} inline'}"/>
> <action application="set" data="sip_exclude_contact=${network_addr}"/>
> <action application="conference_set_auto_outcall"
> data="${group_call(pagezone01)}"/>
> <action application="conference" 
> data="paging1 at default+flags{endconf|deaf}"/>
> </condition>
> </extension>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 max.bridgewater at gmail.com  Tue Feb  8 20:00:29 2011
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Tue, 8 Feb 2011 12:00:29 -0500
Subject: [Freeswitch-users] UPDATE: Buy the devs dinner
In-Reply-To: <4C9E950404614CD89D6299213F3FD878@e1705>
References: <AANLkTimqurw-1TdPyYNXw7C6y3tHy9kufqJf9AhP03yD@mail.gmail.com>
	<4C9E950404614CD89D6299213F3FD878@e1705>
Message-ID: <AANLkTi=pJ+2FH0ksAqhv+wbUJABHHtt-x3j6SAVQ0zPB@mail.gmail.com>

Done. Sorry for not even being able to read.

On Tue, Feb 8, 2011 at 11:56 AM, Madovsky <infos at madovsky.org> wrote:
> Sorry Mike, I live too far to be at the dinner...
> hope you ll enjoy !
>
> ----- Original Message -----
> From: Michael Collins
> To: freeswitch-users at lists.freeswitch.org ;
> freeswitch-dev at lists.freeswitch.org
> Sent: Tuesday, February 08, 2011 11:42 AM
> Subject: [Freeswitch-users] UPDATE: Buy the devs dinner
> Hello all!
> Just an update: all the FreeSWITCH developers are together this week and
> will be going to dinner. It's still not too late for those who wish to pitch
> in and help feed them. Just hit the Paypal link on the main page and drop a
> few dollars!
> Thanks,
> MC
>
> ________________________________
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 math.parent at gmail.com  Tue Feb  8 20:10:14 2011
From: math.parent at gmail.com (Mathieu Parent)
Date: Tue, 8 Feb 2011 18:10:14 +0100
Subject: [Freeswitch-users] We need help with the wiki,
	looking for new administrators
In-Reply-To: <AANLkTinuBj6BqEj_gRzjcEE_6Y3G++kJYZ5dg4KPLSCw@mail.gmail.com>
References: <AANLkTinuBj6BqEj_gRzjcEE_6Y3G++kJYZ5dg4KPLSCw@mail.gmail.com>
Message-ID: <AANLkTim9XAdCS4ia64doKJAjqCy87CfFjHOhbV+eN5Un@mail.gmail.com>

2011/2/8 Diego Viola <diego.viola at gmail.com>:
> Hello,
>
...
> This task isn't hard and it's basically as follows: keep an eye at
> "Recent changes" here:
> http://wiki.freeswitch.org/wiki/Special:RecentChanges -- there is also
> a "Recent changes" link in the main page and in every wiki page, then
> when you find someone spammed the wiki, block that user with
> indefinite time, and then delete his/her spam page or the affected
> pages.
>
> Please contact us if you're interested in helping.

Thanks for this, I know how painfull it can be to read full changlof.

I won't have time myself (I'm pretty busy). But I want to suggest
adding more automatic spam protection.
there is a list of extensions and methods at
http://www.mediawiki.org/wiki/Manual:Combating_spam.
Captcha is already enabled. You can add a $wgSpamRegex* or better use
SpamBlacklist extension ;
use IP address blacklists and DNS BL.

* A lot of words have nothing to do with FS. Like "nursing","X-ray", ...

Regards

Mathieu


From derrick.albers at gmail.com  Tue Feb  8 20:00:37 2011
From: derrick.albers at gmail.com (Derrick Albers)
Date: Tue, 8 Feb 2011 17:00:37 +0000 (UTC)
Subject: [Freeswitch-users] =?utf-8?q?Error_Loading_mod=5Fjava=2Eso?=
References: <AANLkTikz9gbLdKKx=aGJHxUZgwt4MqnLvVfad86PdyWV@mail.gmail.com>
	<AANLkTimYd+7vrObud9iJ5t81u0pdO9+LtT2GC5Nar2KE@mail.gmail.com>
	<AANLkTim0G_FCdrmxx6CFBH3Uu9URhr6EG_rpF_zmP2e=@mail.gmail.com>
Message-ID: <loom.20110208T175844-472@post.gmane.org>

I am trying to load the mod_java, I get the following error:
[ERR] modjava.c:124 Error loading
/usr/java/jdk1.6.0_23/jre/lib/i386/client/libjvm.so 
2011-02-08 10:52:37.540045 [CRIT] switch_loadable_module.c:882 Error Loading
module /usr/local/freeswitch/mod/mod_java.so

I have verified that the libjvm.so does exist and is in that location, I have
tried everything I can to force this load, please help!

Thanks
-Derrick



From Nabble at slickdeals.endjunk.com  Tue Feb  8 20:13:12 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 8 Feb 2011 09:13:12 -0800 (PST)
Subject: [Freeswitch-users] git-a93623 version (2011/02/07) causes
 [DESTINATION_OUT_OF_ORDER]
In-Reply-To: <AANLkTimOM9dZYr=zrQB+eJAEQ7fh0ugvqJQueOtOGiZH@mail.gmail.com>
References: <1297183735886-6004663.post@n2.nabble.com>
	<AANLkTimOM9dZYr=zrQB+eJAEQ7fh0ugvqJQueOtOGiZH@mail.gmail.com>
Message-ID: <1297185192140-6004757.post@n2.nabble.com>



Anthony Minessale wrote:
> no, either you did fsctl pause or are at your session limit.
How do I know if I use fsctl pause or the FS is at a session limit?


Did you use "make current"
No and I did not. However, I did a git pull on a freshly untarred directory
(from a clean tarred file created on 2011/01/19), executed a ./bootstrap.sh,
and then followed by make to compile the source.

-----
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/git-a93623-version-2011-02-07-causes-DESTINATION-OUT-OF-ORDER-tp6004663p6004757.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveu at coppice.org  Tue Feb  8 20:23:57 2011
From: steveu at coppice.org (Steve Underwood)
Date: Wed, 09 Feb 2011 01:23:57 +0800
Subject: [Freeswitch-users] G729
In-Reply-To: <AANLkTimKH61TNx_BjOthqMOYJe=t_WkD7qBRL-GKbrFa@mail.gmail.com>
References: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>	<AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>
	<AANLkTimKH61TNx_BjOthqMOYJe=t_WkD7qBRL-GKbrFa@mail.gmail.com>
Message-ID: <4D517C2D.1050207@coppice.org>

On 02/09/2011 12:09 AM, Steven Ayre wrote:
> Annex A is just an alternative way of encoding the data, using less 
> processing power but at poorer quality. They generate compatible data 
> - decoders both with and without annex a support decode both G729 and 
> G729A encoded data. The difference is purely in the implementation of 
> the encoder/decoder. If your device uses G729A in the SDP, it's broken.
This is correct, but there are still quite a few things around that have 
broken SDP that says g729a. The last time I looked, Freeswitch had a 
fudge in the source code to tolerate that.
>
> Annex B uses the same PT & name. It does work in passthrough mode, as 
> does plain G729.
There is supplementary info in the SDP to say whether AnnexB is enabled, 
like so:

a=fmtp:18 annexb=yes


> -Steve
>
>
> On 8 February 2011 15:31, curriegrad2004 <curriegrad2004 at gmail.com 
> <mailto:curriegrad2004 at gmail.com>> wrote:
>
>     G729 only. No A's or AB.
>
>     If you want to pass through other variants of G.729, you can simply
>     modify mod_g729.c to include additional variants of G729
>
>     Which by the way, I've pasted the area of the code you might want to
>     modify. I've done this on mine and it worked fine as pass through.
>
>     00217         SWITCH_ADD_CODEC(codec_interface, "G.729A");
>     00218         for (count = 12; count > 0; count--) {
>     00219                 switch_core_codec_add_implementation(pool,
>     codec_interface,
>     00220
>                            SWITCH_CODEC_TYPE_AUDIO, 18, "G729A", NULL,
>     8000, 8000, 8000,
>     00221
>                            mpf * count, spf * count, bpf * count, ebpf *
>     count, 1, count * 10,
>     00222
>                            switch_g729_init, switch_g729_encode,
>     switch_g729_decode, switch_g729_destroy);
>     00223         }
>
>
>     On Tue, Feb 8, 2011 at 7:02 AM, Sam <u2nsam at gmail.com
>     <mailto:u2nsam at gmail.com>> wrote:
>     > Hi,
>     >
>     > Which type of  G.729 codec does freeswitch uses in passthrough
>     mode ?
>     > Is it g729a or g729b or g729ab .
>     >
>     > Regards
>     > Sam
>
Steve



From william.suffill at gmail.com  Tue Feb  8 20:32:47 2011
From: william.suffill at gmail.com (William Suffill)
Date: Tue, 8 Feb 2011 12:32:47 -0500
Subject: [Freeswitch-users] We need help with the wiki,
	looking for new administrators
In-Reply-To: <AANLkTim9XAdCS4ia64doKJAjqCy87CfFjHOhbV+eN5Un@mail.gmail.com>
References: <AANLkTinuBj6BqEj_gRzjcEE_6Y3G++kJYZ5dg4KPLSCw@mail.gmail.com>
	<AANLkTim9XAdCS4ia64doKJAjqCy87CfFjHOhbV+eN5Un@mail.gmail.com>
Message-ID: <AANLkTikez+6bftb3n_sKqDyGd+-A+btPgcw_uZC4Ju-q@mail.gmail.com>

That makes alot of sense. Perhaps I can take a look at it and see what
changes can be made with proper coordination to keep it being such a mundane
manual process.

-- W
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From anthony.minessale at gmail.com  Tue Feb  8 20:35:47 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 8 Feb 2011 11:35:47 -0600
Subject: [Freeswitch-users] hold_music var gets unset/lost by
 cancelled/failed att_xfer
In-Reply-To: <AANLkTi=EndkiUJvCgCSirxCA23EwWD_++0h3Qcr=mg9g@mail.gmail.com>
References: <AANLkTimLod3yHrex4LAK=4gdVcPhzXutKYyw=E8ZMcN9@mail.gmail.com>
	<AANLkTi=u3VeHPuS5x1e+yu5QNBw5ZSgHvAWOT+F6A_OR@mail.gmail.com>
	<AANLkTi=EndkiUJvCgCSirxCA23EwWD_++0h3Qcr=mg9g@mail.gmail.com>
Message-ID: <AANLkTikt97dugbhKPgnB2ws4bOFcxg8yhzSdyxHMab_m@mail.gmail.com>

try latest GIT

On Tue, Feb 8, 2011 at 1:28 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> I have same issue with MOH and att_xfer on failed transfers, music on
> hold played only once
> At the same time, transfer_ringback always plays correctly to transferer
>
> 2011/2/8 Anthony Minessale <anthony.minessale at gmail.com>:
>> you really should report this to jira not to the mailing list.
>> http://jira.freeswitch.org
>>
>>
>>
>> On Mon, Feb 7, 2011 at 4:23 PM, Fraser Redmond <fraserredmond at gmail.com> wrote:
>>> I'm trying to do a second att_xfer on a call so that if the first attended
>>> transfer fails (c-leg is busy, or presses do-not-answer, or is an extn that
>>> doesn't exist) then the call could be transferred to someone else.
>>>
>>> On the first att_xfer the person on hold hears the hold_music correctly.
>>> Once that transfer is cancelled or fails:
>>> -- On any subsequent att_xfer's the person on hold just hears silence.
>>> -- If they are put on hold they just hear silence.
>>>
>>> I tried setting hold_music again for each channel after the first att_xfer,
>>> but that didn't work, so it's probably not actually a problem with
>>> hold_music per se, but some other variable/setting that decides whether to
>>> use hold_music.
>>>
>>> I also tried doing a uuid_dump before and after each attempt, but didn't
>>> notice anything too different - unless it's a matter of unsetting one of the
>>> couple of changed/new vars like:
>>> variable_originate_disposition
>>> variable_current_application
>>> variable_playback_seconds
>>>
>>> I get the feeling other variables are probably also lost by the first failed
>>> transfer as the second att_xfer has some odd things happen if the third
>>> party does answer. Haven't been able to narrow it down as closely as the
>>> hold_music, but two things I've seen happen are:
>>> -- The party that initiated the transfer gets hung up automatically (after
>>> 30 sec)
>>> -- When the party that initiated the transfer hangs up it should connect the
>>> other two parties, but instead it hung up all three
>>>
>>> Cheers,
>>> Fraser
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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  Tue Feb  8 20:52:21 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 8 Feb 2011 11:52:21 -0600
Subject: [Freeswitch-users] git-a93623 version (2011/02/07) causes
	[DESTINATION_OUT_OF_ORDER]
In-Reply-To: <1297185192140-6004757.post@n2.nabble.com>
References: <1297183735886-6004663.post@n2.nabble.com>
	<AANLkTimOM9dZYr=zrQB+eJAEQ7fh0ugvqJQueOtOGiZH@mail.gmail.com>
	<1297185192140-6004757.post@n2.nabble.com>
Message-ID: <AANLkTikzn43AqJaEUaWFpa0Yy70KLz-64Cb-8Jk6Fo8W@mail.gmail.com>

check carefully then, maybe you pressed f11 (it used to be mapped to
fsctl pause)


On Tue, Feb 8, 2011 at 11:13 AM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
>
> Anthony Minessale wrote:
>> no, either you did fsctl pause or are at your session limit.
> How do I know if I use fsctl pause or the FS is at a session limit?
>
>
> Did you use "make current"
> No and I did not. However, I did a git pull on a freshly untarred directory
> (from a clean tarred file created on 2011/01/19), executed a ./bootstrap.sh,
> and then followed by make to compile the source.
>
> -----
> 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/git-a93623-version-2011-02-07-causes-DESTINATION-OUT-OF-ORDER-tp6004663p6004757.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
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FreeSWITCH Developer Conference
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From peder at networkoblivion.com  Tue Feb  8 21:05:16 2011
From: peder at networkoblivion.com (Peder)
Date: Tue, 8 Feb 2011 12:05:16 -0600
Subject: [Freeswitch-users] Overriding 480 with 200 from other leg
Message-ID: <02af01cbc7ba$bd8055e0$388101a0$@com>

I've got a strange issue and I am not 100% sure how to get around it.  

We have a system setup where a user calls in and enters a code.  A perl
script is called that does an http get to find a phone number to dial, it
dials the number, bridges the two and then it records the call.  

Once the code is verified and we have the number, we set two carriers to
call, primary and backup.  This all works great 95%+ of the time.  We do 40k
calls per month and about 50 calls a week run into a scenario that I am not
sure of.   On the ones that don't work, it gets all the way through bridging
the calls with no problem, but when the user hangs up, it doesn't hang up,
it falls to the backup bridge and calls the destination number again our
carrier2, all the while still recording.  We get 3 CDR records, incoming,
call out carrier1 and call out carrier2.

Here is the snippet of the perl script that does the dialing ($bridge and
$backupbridge are the two carriers to call).  If $bridge fails, it goes back
to $backupbridge just fine:

$session->execute("set","continue_on_fail=true");
$session->execute("bridge","sofia/gateway/$bridge/$PhoneNo");
$session->execute("export","nolocal:CARRIER=${backupbridge}");
$session->execute("bridge","sofia/gateway/$backupbridge/$PhoneNo");


Every now and then we see this in the logs "Overriding SIP cause 480 with
200 from the other leg":


2011-01-19 09:24:50.751368 [NOTICE] sofia.c:481 Hangup
sofia/external/1xxxxxx8003 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2011-01-19 09:24:50.753368 [DEBUG] switch_channel.c:2071 Send signal
sofia/external/1xxxxxx8003 [KILL]
2011-01-19 09:24:50.753368 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/1xxxxxx8003 [BREAK]
2011-01-19 09:24:50.761364 [DEBUG] switch_ivr_bridge.c:478
sofia/external/1xxxxxx8003 ending bridge by request from read function
2011-01-19 09:24:50.761364 [DEBUG] switch_ivr_bridge.c:565 BRIDGE THREAD
DONE [sofia/external/1xxxxxx8003]
2011-01-19 09:24:50.761364 [DEBUG] switch_ivr_bridge.c:585 Send signal
sofia/internal/yyyyyy6168 at 72.249.14.242 [BREAK]
2011-01-19 09:24:50.761364 [DEBUG] switch_core_state_machine.c:351
(sofia/external/1xxxxxx8003) State EXCHANGE_MEDIA going to sleep
2011-01-19 09:24:50.761364 [DEBUG] switch_core_state_machine.c:314
(sofia/external/1xxxxxx8003) Running State Change CS_HANGUP
2011-01-19 09:24:50.761364 [DEBUG] switch_core_state_machine.c:499
(sofia/external/1xxxxxx8003) State HANGUP
2011-01-19 09:24:50.761364 [DEBUG] mod_sofia.c:405
sofia/external/1xxxxxx8003 Overriding SIP cause 480 with 200 from the other
leg
2011-01-19 09:24:50.761364 [DEBUG] mod_sofia.c:411 Channel
sofia/external/1xxxxxx8003 hanging up, cause: NORMAL_CLEARING
2011-01-19 09:24:50.761364 [DEBUG] switch_core_state_machine.c:46
sofia/external/1xxxxxx8003 Standard HANGUP, cause: NORMAL_CLEARING
2011-01-19 09:24:50.761364 [DEBUG] switch_core_state_machine.c:499
(sofia/external/1xxxxxx8003) State HANGUP going to sleep
2011-01-19 09:24:50.761364 [DEBUG] switch_core_state_machine.c:333
(sofia/external/1xxxxxx8003) State Change CS_HANGUP -> CS_REPORTING
2011-01-19 09:24:50.761364 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/1xxxxxx8003 [BREAK]
2011-01-19 09:24:50.761364 [DEBUG] switch_core_state_machine.c:314
(sofia/external/1xxxxxx8003) Running State Change CS_REPORTING
2011-01-19 09:24:50.761364 [DEBUG] switch_core_state_machine.c:590
(sofia/external/1xxxxxx8003) State REPORTING
2011-01-19 09:24:50.763475 [DEBUG] switch_core_session.c:638 Send signal
sofia/internal/yyyyyy6168 at 72.249.14.242 [BREAK]
2011-01-19 09:24:50.763475 [DEBUG] switch_ivr_bridge.c:565 BRIDGE THREAD
DONE [sofia/internal/yyyyyy6168 at 72.249.14.242]
2011-01-19 09:24:50.763475 [DEBUG] switch_ivr_bridge.c:585 Send signal
sofia/external/1xxxxxx8003 [BREAK]
EXECUTE sofia/internal/yyyyyy6168 at 72.249.14.242
export(nolocal:CARRIER=carrier2)
2011-01-19 09:24:50.766348 [DEBUG] switch_core_state_machine.c:53
sofia/external/1xxxxxx8003 Standard REPORTING, cause: NORMAL_CLEARING
2011-01-19 09:24:50.766348 [DEBUG] switch_core_state_machine.c:590
(sofia/external/1xxxxxx8003) State REPORTING going to sleep
2011-01-19 09:24:50.766348 [DEBUG] switch_core_state_machine.c:327
(sofia/external/1xxxxxx8003) State Change CS_REPORTING -> CS_DESTROY
2011-01-19 09:24:50.766348 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/1xxxxxx8003 [BREAK]
2011-01-19 09:24:50.766348 [DEBUG] switch_core_session.c:1161 Session 96783
(sofia/external/1xxxxxx8003) Locked, Waiting on external entities
2011-01-19 09:24:50.766348 [NOTICE] switch_core_session.c:1179 Session 96783
(sofia/external/1xxxxxx8003) Ended
2011-01-19 09:24:50.766348 [NOTICE] switch_core_session.c:1181 Close Channel
sofia/external/1xxxxxx8003 [CS_DESTROY]
2011-01-19 09:24:50.766348 [DEBUG] mod_dptools.c:898 EXPORT (REMOTE ONLY)
[CARRIER]=[carrier2]
2011-01-19 09:24:50.766348 [DEBUG] switch_core_state_machine.c:428
(sofia/external/1xxxxxx8003) Running State Change CS_DESTROY
2011-01-19 09:24:50.766348 [DEBUG] switch_core_state_machine.c:439
(sofia/external/1xxxxxx8003) State DESTROY
2011-01-19 09:24:50.766348 [DEBUG] mod_sofia.c:338
sofia/external/1xxxxxx8003 SOFIA DESTROY
EXECUTE sofia/internal/yyyyyy6168 at 72.249.14.242
bridge(sofia/gateway/carrier2/1xxxxxx8003)


I am guessing that carrier1 is sending back a 480, so FS thinks it never
worked and then bridges out carrier2.  Our FS is pretty old, FreeSWITCH
Version 1.0.trunk (17043), but I don't think it is an FS issue, I think it
is just a dialplan issue that I don't know how to resolve.  Any idea how to
stop this from happening?  

Peder






From Nabble at slickdeals.endjunk.com  Tue Feb  8 21:25:19 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 8 Feb 2011 10:25:19 -0800 (PST)
Subject: [Freeswitch-users] git-a93623 version (2011/02/07) causes
 [DESTINATION_OUT_OF_ORDER]
In-Reply-To: <AANLkTikzn43AqJaEUaWFpa0Yy70KLz-64Cb-8Jk6Fo8W@mail.gmail.com>
References: <1297183735886-6004663.post@n2.nabble.com>
	<AANLkTimOM9dZYr=zrQB+eJAEQ7fh0ugvqJQueOtOGiZH@mail.gmail.com>
	<1297185192140-6004757.post@n2.nabble.com>
	<AANLkTikzn43AqJaEUaWFpa0Yy70KLz-64Cb-8Jk6Fo8W@mail.gmail.com>
Message-ID: <1297189519810-6005004.post@n2.nabble.com>



Anthony Minessale wrote:
> 
> check carefully then, maybe you pressed f11 (it used to be mapped to
> fsctl pause)
Just to be on the safe side, I shutdown and re-run freeswitch. When I see my
ATA registers and wait for one minute, I execute fs_cli, place an outbount
call, and the problem persists as excerpted below:
FreeSWITCH Version 1.0.head (git-a936236 2011-02-07 16-33-45 -0600)
2011-02-08 13:20:05.318691 [CRIT] mod_sofia.c:3844 Error Creating Session
2011-02-08 13:20:05.318691 [ERR] switch_ivr_originate.c:2638 Cannot create
outgoing channel of type [sofia] cause: [DESTINATION_OUT_OF_ORDER]
If I reboot back to previous git version, placing an outbound call works.
These two tests use the same configurations and dialplans.

-----
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/git-a93623-version-2011-02-07-causes-DESTINATION-OUT-OF-ORDER-tp6004663p6005004.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Tue Feb  8 21:35:01 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 8 Feb 2011 12:35:01 -0600
Subject: [Freeswitch-users] git-a93623 version (2011/02/07) causes
	[DESTINATION_OUT_OF_ORDER]
In-Reply-To: <1297189519810-6005004.post@n2.nabble.com>
References: <1297183735886-6004663.post@n2.nabble.com>
	<AANLkTimOM9dZYr=zrQB+eJAEQ7fh0ugvqJQueOtOGiZH@mail.gmail.com>
	<1297185192140-6004757.post@n2.nabble.com>
	<AANLkTikzn43AqJaEUaWFpa0Yy70KLz-64Cb-8Jk6Fo8W@mail.gmail.com>
	<1297189519810-6005004.post@n2.nabble.com>
Message-ID: <AANLkTing-OkaLem1HLJPQSCaSORSvLZZk-m4M0cXDqFn@mail.gmail.com>

When you are jumping back and forth to different versions that are not
binary compatible you are going to have problems.

Are you erasing the entire contents of
/usr/local/freeswitch/bin,lib,mod when jumping versions and doing full
clean builds in between.

There is nothing wrong with latest git with any outbound calling.





On Tue, Feb 8, 2011 at 12:25 PM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
>
> Anthony Minessale wrote:
>>
>> check carefully then, maybe you pressed f11 (it used to be mapped to
>> fsctl pause)
> Just to be on the safe side, I shutdown and re-run freeswitch. When I see my
> ATA registers and wait for one minute, I execute fs_cli, place an outbount
> call, and the problem persists as excerpted below:
> FreeSWITCH Version 1.0.head (git-a936236 2011-02-07 16-33-45 -0600)
> 2011-02-08 13:20:05.318691 [CRIT] mod_sofia.c:3844 Error Creating Session
> 2011-02-08 13:20:05.318691 [ERR] switch_ivr_originate.c:2638 Cannot create
> outgoing channel of type [sofia] cause: [DESTINATION_OUT_OF_ORDER]
> If I reboot back to previous git version, placing an outbound call works.
> These two tests use the same configurations and dialplans.
>
> -----
> 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/git-a93623-version-2011-02-07-causes-DESTINATION-OUT-OF-ORDER-tp6004663p6005004.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 msc at freeswitch.org  Tue Feb  8 21:43:39 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 8 Feb 2011 12:43:39 -0600
Subject: [Freeswitch-users] We need help with the wiki,
	looking for new administrators
In-Reply-To: <AANLkTikez+6bftb3n_sKqDyGd+-A+btPgcw_uZC4Ju-q@mail.gmail.com>
References: <AANLkTinuBj6BqEj_gRzjcEE_6Y3G++kJYZ5dg4KPLSCw@mail.gmail.com>
	<AANLkTim9XAdCS4ia64doKJAjqCy87CfFjHOhbV+eN5Un@mail.gmail.com>
	<AANLkTikez+6bftb3n_sKqDyGd+-A+btPgcw_uZC4Ju-q@mail.gmail.com>
Message-ID: <AANLkTina11hFrnPxJw0GEzgW5uwRJVGW9CRDaO=2DmaB@mail.gmail.com>

I appreciate everyone's helps with all this. I'm available to load other
mediawiki modules, etc. if someone has thoughts on what we can do.
-MC

On Tue, Feb 8, 2011 at 11:32 AM, William Suffill
<william.suffill at gmail.com>wrote:

> That makes alot of sense. Perhaps I can take a look at it and see what
> changes can be made with proper coordination to keep it being such a mundane
> manual process.
>
> -- 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 david.ponzone at ipeva.fr  Tue Feb  8 21:55:42 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 8 Feb 2011 19:55:42 +0100
Subject: [Freeswitch-users] nibblebill sql requests
In-Reply-To: <AD71F946CFA942858324FAC60CF95D76@e1705>
References: <AD71F946CFA942858324FAC60CF95D76@e1705>
Message-ID: <49831501-F7B9-4C24-BF78-DE4C8AF38A78@ipeva.fr>

AFAIR, they are not unnecessary.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 08/02/2011 ? 17:11, Madovsky a ?crit :

> 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

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From sos at sokhapkin.dyndns.org  Tue Feb  8 22:05:04 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 8 Feb 2011 14:05:04 -0500
Subject: [Freeswitch-users] nibblebill sql requests
In-Reply-To: <49831501-F7B9-4C24-BF78-DE4C8AF38A78@ipeva.fr>
References: <AD71F946CFA942858324FAC60CF95D76@e1705>
	<49831501-F7B9-4C24-BF78-DE4C8AF38A78@ipeva.fr>
Message-ID: <201102081405.04138.sos@sokhapkin.dyndns.org>

There is one more extra update. See http://jira.freeswitch.org/browse/FS-2890 

Mod_nibblebill updates balance in channel hangup and reporting states, but 
should update in hangup state only.

On Tuesday 08 February 2011, David Ponzone wrote:
> AFAIR, they are not unnecessary.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de
> ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> Le 08/02/2011 ? 17:11, Madovsky a ?crit :
> > 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



From locutis at sect001.net  Tue Feb  8 22:07:14 2011
From: locutis at sect001.net (Locutis of Borg)
Date: Tue, 8 Feb 2011 14:07:14 -0500
Subject: [Freeswitch-users] mod_dingaling & inbound audio
Message-ID: <AANLkTim-5LDhfQhHk_2LNLffvTa1JyBXq9U4B3LSY3Nq@mail.gmail.com>

Followed instructions on wiki.  Enabled mod_dingaling to register over at
GV.  Set external rtp to right address.  Outbound calling terminates, and
outbound audio works, but seems that I can not get return audio.

FS is on DMZ - same problem
Moved to internal (behind NAT, port forwarding) - same problem
Moved back to DMZ and killed iptables and router spi - same problem

In and out calling on SIP to provider works fine in both cases.

So my questions are:
  Can mod_dingaling be used for GV or not?
  Since it uses TLS, NAT shouldn't be an issue?
     and
  Does GV not like us FS users?

Thank you for you help.

"Resistance is Fruitful"
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From avi at avimarcus.net  Tue Feb  8 22:12:20 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 8 Feb 2011 21:12:20 +0200
Subject: [Freeswitch-users] We need help with the wiki,
	looking for new administrators
In-Reply-To: <AANLkTina11hFrnPxJw0GEzgW5uwRJVGW9CRDaO=2DmaB@mail.gmail.com>
References: <AANLkTinuBj6BqEj_gRzjcEE_6Y3G++kJYZ5dg4KPLSCw@mail.gmail.com>
	<AANLkTim9XAdCS4ia64doKJAjqCy87CfFjHOhbV+eN5Un@mail.gmail.com>
	<AANLkTikez+6bftb3n_sKqDyGd+-A+btPgcw_uZC4Ju-q@mail.gmail.com>
	<AANLkTina11hFrnPxJw0GEzgW5uwRJVGW9CRDaO=2DmaB@mail.gmail.com>
Message-ID: <AANLkTikHAw+2ZYhFmz_Ez1AeMK91Enp92FJV_eHfFbzw@mail.gmail.com>

I check for recent changes often to try to keep up with new stuff in FS -
sometimes several times per day, so I'm available to help.

I've been noticing a trend of more mediawiki abuse in other places as well,
perhaps additional preventative measures should be put into place...
E.g. 1) htaccess on the submit page (like pb - linode.com does this),  or 2)
manual confirmation of new accounts via email/irc appearance/phone call, or
3) captchas on new user / submit..

-Avi

On Tue, Feb 8, 2011 at 8:43 PM, Michael Collins <msc at freeswitch.org> wrote:

> I appreciate everyone's helps with all this. I'm available to load other
> mediawiki modules, etc. if someone has thoughts on what we can do.
> -MC
>
> On Tue, Feb 8, 2011 at 11:32 AM, William Suffill <
> william.suffill at gmail.com> wrote:
>
>> That makes alot of sense. Perhaps I can take a look at it and see what
>> changes can be made with proper coordination to keep it being such a mundane
>> manual process.
>>
>> -- 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 jaugenstine at gmail.com  Tue Feb  8 22:15:53 2011
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Tue, 8 Feb 2011 11:15:53 -0800
Subject: [Freeswitch-users] We need help with the wiki,
	looking for new administrators
In-Reply-To: <AANLkTina11hFrnPxJw0GEzgW5uwRJVGW9CRDaO=2DmaB@mail.gmail.com>
References: <AANLkTinuBj6BqEj_gRzjcEE_6Y3G++kJYZ5dg4KPLSCw@mail.gmail.com>
	<AANLkTim9XAdCS4ia64doKJAjqCy87CfFjHOhbV+eN5Un@mail.gmail.com>
	<AANLkTikez+6bftb3n_sKqDyGd+-A+btPgcw_uZC4Ju-q@mail.gmail.com>
	<AANLkTina11hFrnPxJw0GEzgW5uwRJVGW9CRDaO=2DmaB@mail.gmail.com>
Message-ID: <AANLkTikyPeQP90rjhMW3m79rMcvtYK6ZLxfa6ts+e8_U@mail.gmail.com>

Diego,

If you would like to contact me offline, I can probably help out with the
monitoring of the wiki, while the automation is put in place.

Jonathan


On Tue, Feb 8, 2011 at 10:43 AM, Michael Collins <msc at freeswitch.org> wrote:

> I appreciate everyone's helps with all this. I'm available to load other
> mediawiki modules, etc. if someone has thoughts on what we can do.
> -MC
>
> On Tue, Feb 8, 2011 at 11:32 AM, William Suffill <
> william.suffill at gmail.com> wrote:
>
>> That makes alot of sense. Perhaps I can take a look at it and see what
>> changes can be made with proper coordination to keep it being such a mundane
>> manual process.
>>
>> -- 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 avi at avimarcus.net  Tue Feb  8 22:18:34 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 8 Feb 2011 21:18:34 +0200
Subject: [Freeswitch-users] nibblebill sql requests
In-Reply-To: <AD71F946CFA942858324FAC60CF95D76@e1705>
References: <AD71F946CFA942858324FAC60CF95D76@e1705>
Message-ID: <AANLkTimLYmuCBRTrJUmVC62QiYohOOMSUVRJT+Oi-Dus@mail.gmail.com>

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
>
>
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From steveayre at gmail.com  Tue Feb  8 22:22:50 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 8 Feb 2011 19:22:50 +0000
Subject: [Freeswitch-users] G729
In-Reply-To: <4D517C2D.1050207@coppice.org>
References: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>
	<AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>
	<AANLkTimKH61TNx_BjOthqMOYJe=t_WkD7qBRL-GKbrFa@mail.gmail.com>
	<4D517C2D.1050207@coppice.org>
Message-ID: <AANLkTi=FHwxTBU6TeMaJ9QJWQK8sotbAa3g42T=YK61W@mail.gmail.com>

On 8 February 2011 17:23, Steve Underwood <steveu at coppice.org> wrote:

> On 02/09/2011 12:09 AM, Steven Ayre wrote:
> > Annex A is just an alternative way of encoding the data, using less
> > processing power but at poorer quality. They generate compatible data
> > - decoders both with and without annex a support decode both G729 and
> > G729A encoded data. The difference is purely in the implementation of
> > the encoder/decoder. If your device uses G729A in the SDP, it's broken.
> This is correct, but there are still quite a few things around that have
> broken SDP that says g729a. The last time I looked, Freeswitch had a
> fudge in the source code to tolerate that.
>

Cool, didn't realise that. So no modifications should be needed then.


> >
> > Annex B uses the same PT & name. It does work in passthrough mode, as
> > does plain G729.
> There is supplementary info in the SDP to say whether AnnexB is enabled,
> like so:
>
> a=fmtp:18 annexb=yes
>

Which is I believe passed to the other leg. Is that correct?


>
>
> > -Steve
> >
> >
> > On 8 February 2011 15:31, curriegrad2004 <curriegrad2004 at gmail.com
> > <mailto:curriegrad2004 at gmail.com>> wrote:
> >
> >     G729 only. No A's or AB.
> >
> >     If you want to pass through other variants of G.729, you can simply
> >     modify mod_g729.c to include additional variants of G729
> >
> >     Which by the way, I've pasted the area of the code you might want to
> >     modify. I've done this on mine and it worked fine as pass through.
> >
> >     00217         SWITCH_ADD_CODEC(codec_interface, "G.729A");
> >     00218         for (count = 12; count > 0; count--) {
> >     00219                 switch_core_codec_add_implementation(pool,
> >     codec_interface,
> >     00220
> >                            SWITCH_CODEC_TYPE_AUDIO, 18, "G729A", NULL,
> >     8000, 8000, 8000,
> >     00221
> >                            mpf * count, spf * count, bpf * count, ebpf *
> >     count, 1, count * 10,
> >     00222
> >                            switch_g729_init, switch_g729_encode,
> >     switch_g729_decode, switch_g729_destroy);
> >     00223         }
> >
> >
> >     On Tue, Feb 8, 2011 at 7:02 AM, Sam <u2nsam at gmail.com
> >     <mailto:u2nsam at gmail.com>> wrote:
> >     > Hi,
> >     >
> >     > Which type of  G.729 codec does freeswitch uses in passthrough
> >     mode ?
> >     > Is it g729a or g729b or g729ab .
> >     >
> >     > Regards
> >     > Sam
> >
> 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 mitch.capper at gmail.com  Tue Feb  8 23:58:32 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Tue, 8 Feb 2011 12:58:32 -0800
Subject: [Freeswitch-users] Error Loading mod_java.so
In-Reply-To: <loom.20110208T175844-472@post.gmane.org>
References: <AANLkTikz9gbLdKKx=aGJHxUZgwt4MqnLvVfad86PdyWV@mail.gmail.com>
	<AANLkTimYd+7vrObud9iJ5t81u0pdO9+LtT2GC5Nar2KE@mail.gmail.com>
	<AANLkTim0G_FCdrmxx6CFBH3Uu9URhr6EG_rpF_zmP2e=@mail.gmail.com>
	<loom.20110208T175844-472@post.gmane.org>
Message-ID: <AANLkTi=isxxC0QHGag_cxA6zEbWSc9VPBZG_FGgTYyB0@mail.gmail.com>

The error ("Module load routine returned an error") means the module itself
has an error during its startup there are 4 places in the module loading
that it would error out:
switch_core_new_memory_pool
load_config
create_java_vm
exec_user_method

Most likely obviously one of the last 3, load_config will always log
something before returning an error status same for create_java_vm,  add
some additional output to modjava.c should be able to help you track it
down.  Additionally turn your console log level to debug, and try reloading
mod_java to make sure you are not missing any error messages.

~Mitch

On Tue, Feb 8, 2011 at 9:00 AM, Derrick Albers <derrick.albers at gmail.com>wrote:

> I am trying to load the mod_java, I get the following error:
> [ERR] modjava.c:124 Error loading
> /usr/java/jdk1.6.0_23/jre/lib/i386/client/libjvm.so
> 2011-02-08 10:52:37.540045 [CRIT] switch_loadable_module.c:882 Error
> Loading
> module /usr/local/freeswitch/mod/mod_java.so
>
> I have verified that the libjvm.so does exist and is in that location, I
> have
> tried everything I can to force this load, please help!
>
> Thanks
> -Derrick
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anu at familytv.com  Wed Feb  9 00:09:21 2011
From: anu at familytv.com (Anirudha Shimpi)
Date: Tue, 8 Feb 2011 14:09:21 -0700
Subject: [Freeswitch-users]  mod_fsk not detecting call waiting caller id
Message-ID: <00c801cbc7d4$75072db0$5f158910$@familytv.com>

Waiting for digit 'D' did not yield any results. Any other ideas? I am very
much interested in getting call waiting caller id to work with Freeswitch
and analog lines.

 

Call waiting caller id works as follow as per my understanding -

1.       CO Sends 440hz tone to indicate call waiting

2.       Followed by CAS signal (DTMF - 2130/2750)

3.       CPE sends DTMF digit 'D'

4.       CO sends FSK caller id in MDMF format

 

I tried this on my original line by adding the following to the dialplan

<action application="tone_detect" data="cwcid_tone_detect 2750,2130 r 0
transfer '1302 XML default' 1"/>

 

With ext 1302 doing the following

<action application="playback" data="tone_stream://%(100,0,697,1633)"/>
<!-DTMF D-->

<action application="fsk_recv"/>

<action application="set" data="execute_on_fsk=log 'Got FSK
[${fsk_phone_num}] [${fsk_phone_name}]'"/>

 

Had two problems with this scenario

1.       FS would never detect the CAS tone

2.       To get around, I decided to detect the 440 hz tone, and ignore the
CAS tone, added delay of about 100ms to ignore the CAS tone. This resulted
in the call being transferred to 1302 but, still no fsk_recv.

 

Questions -

1.       Why does FS not detect the CAS tone?

2.       In case of scenario 2, why did mod_fsk not detect the FSK?

 

I did try the same case with some old Dialogic D/41 cards that I had, and it
seems to work fine (detect CAS tone, send D, wait for FSK). I do get the
caller id fsk after I transmit the D. There is a permitted possible
variation of up to 14hz on the CAS frequencies, however, I do not know how
to specify that for mod_tone_detect.

 

Any help greatly appreciated.

 

Thanks

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From msc at freeswitch.org  Wed Feb  9 00:15:13 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 8 Feb 2011 15:15:13 -0600
Subject: [Freeswitch-users] mod_dingaling & inbound audio
In-Reply-To: <AANLkTim-5LDhfQhHk_2LNLffvTa1JyBXq9U4B3LSY3Nq@mail.gmail.com>
References: <AANLkTim-5LDhfQhHk_2LNLffvTa1JyBXq9U4B3LSY3Nq@mail.gmail.com>
Message-ID: <AANLkTik19FmdGVEukmC4rS3ytu9qhR=wfSkO-fhaRMJQ@mail.gmail.com>

use pastebin.freeswitch.org and pastebin your configs and a debug output of
the failed call. be sure that you have debug level output (which is default
if you use fs_cli)
-MC

On Tue, Feb 8, 2011 at 1:07 PM, Locutis of Borg <locutis at sect001.net> wrote:

> Followed instructions on wiki.  Enabled mod_dingaling to register over at
> GV.  Set external rtp to right address.  Outbound calling terminates, and
> outbound audio works, but seems that I can not get return audio.
>
> FS is on DMZ - same problem
> Moved to internal (behind NAT, port forwarding) - same problem
> Moved back to DMZ and killed iptables and router spi - same problem
>
> In and out calling on SIP to provider works fine in both cases.
>
> So my questions are:
>   Can mod_dingaling be used for GV or not?
>   Since it uses TLS, NAT shouldn't be an issue?
>      and
>   Does GV not like us FS users?
>
> Thank you for you help.
>
> "Resistance is Fruitful"
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  9 00:19:38 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 8 Feb 2011 15:19:38 -0600
Subject: [Freeswitch-users] mod_fsk not detecting call waiting caller id
In-Reply-To: <00c801cbc7d4$75072db0$5f158910$@familytv.com>
References: <00c801cbc7d4$75072db0$5f158910$@familytv.com>
Message-ID: <AANLkTikev=bPRoabJU6MokWHd6mt8Kj2pyxoxEDjV8=n@mail.gmail.com>

I already explained that this is not what this module is intended for.
It's intended for 1 FS to send another FS a series of FSK data.

If you are trying to get the original CID tone on a new call the ata
is probably already eating it up?


On Tue, Feb 8, 2011 at 3:09 PM, Anirudha Shimpi <anu at familytv.com> wrote:
> Waiting for digit ?D? did not yield any results. Any other ideas? I am very
> much interested in getting call waiting caller id to work with Freeswitch
> and analog lines.
>
>
>
> Call waiting caller id works as follow as per my understanding ?
>
> 1.?????? CO Sends 440hz tone to indicate call waiting
>
> 2.?????? Followed by CAS signal (DTMF ? 2130/2750)
>
> 3.?????? CPE sends DTMF digit ?D?
>
> 4.?????? CO sends FSK caller id in MDMF format
>
>
>
> I tried this on my original line by adding the following to the dialplan
>
> <action application="tone_detect" data="cwcid_tone_detect 2750,2130 r 0
> transfer '1302 XML default' 1"/>
>
>
>
> With ext 1302 doing the following
>
> <action application="playback" data="tone_stream://%(100,0,697,1633)"/>
> <!?DTMF D-->
>
> <action application="fsk_recv"/>
>
> <action application="set" data="execute_on_fsk=log 'Got FSK
> [${fsk_phone_num}] [${fsk_phone_name}]'"/>
>
>
>
> Had two problems with this scenario
>
> 1.?????? FS would never detect the CAS tone
>
> 2.?????? To get around, I decided to detect the 440 hz tone, and ignore the
> CAS tone, added delay of about 100ms to ignore the CAS tone. This resulted
> in the call being transferred to 1302 but, still no fsk_recv.
>
>
>
> Questions ?
>
> 1.?????? Why does FS not detect the CAS tone?
>
> 2.?????? In case of scenario 2, why did mod_fsk not detect the FSK?
>
>
>
> I did try the same case with some old Dialogic D/41 cards that I had, and it
> seems to work fine (detect CAS tone, send D, wait for FSK). I do get the
> caller id fsk after I transmit the D. There is a permitted possible
> variation of up to 14hz on the CAS frequencies, however, I do not know how
> to specify that for mod_tone_detect.
>
>
>
> Any help greatly appreciated.
>
>
>
> 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 admin at blindi.net  Wed Feb  9 00:25:28 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Tue, 8 Feb 2011 22:25:28 +0100 (CET)
Subject: [Freeswitch-users] play_and_get_digits question
In-Reply-To: <5DCA5F42FA5A4AAA8D5F34DFDDA51157@e1705>
References: <EBED8C31D06F4A319D978371EE1CF24E@e1705><AANLkTi=AVwJ4c80z9F_06+8EyUHrokLzEtRcPGGpa4jG@mail.gmail.com><C2CAC53C1A8B4CB3ADC3B5EF97F56DDD@e1705>
	<Pine.LNX.4.64.1102080811280.11715@suse.blindi.net>
	<5DCA5F42FA5A4AAA8D5F34DFDDA51157@e1705>
Message-ID: <Pine.LNX.4.64.1102082215470.17347@suse.blindi.net>

Hi Madovsky
Sorry, but i.m. not a FS guru.
i don.t understand your solution.
your create an dialplan with play and get digits.
the application don.t have a if then else construction. is the maximum of 
failled attemps exited.
You say: you have use the limitapplication.
can you post these lines please?
thank you


---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From admin at blindi.net  Wed Feb  9 00:27:49 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Tue, 8 Feb 2011 22:27:49 +0100 (CET)
Subject: [Freeswitch-users] G729
In-Reply-To: <AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>
References: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>
	<AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1102082226490.17347@suse.blindi.net>

Is g729 a free codec?


---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From anthony.minessale at gmail.com  Wed Feb  9 00:32:09 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 8 Feb 2011 15:32:09 -0600
Subject: [Freeswitch-users] G729
In-Reply-To: <Pine.LNX.4.64.1102082226490.17347@suse.blindi.net>
References: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>
	<AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>
	<Pine.LNX.4.64.1102082226490.17347@suse.blindi.net>
Message-ID: <AANLkTikNAJn1XUSpFJjEyz5PhjrJ5Y0eTrM7fvr_SmFe@mail.gmail.com>

the headaches are free but the codec is patented.


On Tue, Feb 8, 2011 at 3:27 PM, Thomas Hoellriegel <admin at blindi.net> wrote:
> Is g729 a free codec?
>
>
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/listinfo/blinde-misc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 admin at blindi.net  Wed Feb  9 00:39:18 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Tue, 8 Feb 2011 22:39:18 +0100 (CET)
Subject: [Freeswitch-users] using skype on a text only ubuntusystem?
In-Reply-To: <59C40C2C-351C-4544-8B50-099DACD6A292@ipeva.fr>
References: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>
	<AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>
	<59C40C2C-351C-4544-8B50-099DACD6A292@ipeva.fr>
Message-ID: <Pine.LNX.4.64.1102082229480.17347@suse.blindi.net>

Hi, i like to configure skype for FS. i have a ubuntu rootserver in a 
datacenter. the server has no X installed. is a text only system.
Can i use skype with no X? Thank you.

---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From gmaruzz at gmail.com  Wed Feb  9 00:44:20 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 8 Feb 2011 22:44:20 +0100
Subject: [Freeswitch-users] using skype on a text only ubuntusystem?
In-Reply-To: <Pine.LNX.4.64.1102082229480.17347@suse.blindi.net>
References: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>
	<AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>
	<59C40C2C-351C-4544-8B50-099DACD6A292@ipeva.fr>
	<Pine.LNX.4.64.1102082229480.17347@suse.blindi.net>
Message-ID: <AANLkTi=hJVcNEDVUpCdXpEZJjYSBq0z6p2QGBRR0Jc73@mail.gmail.com>

Wiki.freeswitch.org/mod_skypopen

On 2/8/11, Thomas Hoellriegel <admin at blindi.net> wrote:
> Hi, i like to configure skype for FS. i have a ubuntu rootserver in a
> datacenter. the server has no X installed. is a text only system.
> Can i use skype with no X? Thank you.
>
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/listinfo/blinde-misc
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From me at nevian.org  Wed Feb  9 01:00:35 2011
From: me at nevian.org (Serge Yuriev)
Date: Wed, 09 Feb 2011 01:00:35 +0300
Subject: [Freeswitch-users] cdr fields
Message-ID: <428041297202435@web1.yandex.ru>

Hello

I noticed difference in cause codes written in csv CDR and RADIUS. Perhaps I need to change something in template?
Current template is
<template name="lanbilling">
${direction},${sip_network_ip},${user_name},${caller_id_number},${destination_number},${start_stamp},${answer_stamp},${end_stamp},${billsec},${hangup_cause_q850},${hangup_cause},${uuid},${bleg_uuid},${accountcode},${read_codec},${write_codec}
</template>

This writes
inbound,109.173.67.229,nevian,nevian,79645835822,2011-02-09 00:18:52,,2011-02-09 00:20:00,0,16,NORMAL_CLEARING,a4da79c4-5356-43b8-894b-78627fb5e243,,nevian,G729,G729

As you can see cause is 16 but in RADiUS is 27 and it's more accurate

In mod_radius_cdr cause got from switch_channel_get_cause(channel)

Please advice.

-- 
wbr,
Serge


From steveayre at gmail.com  Wed Feb  9 01:05:26 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 8 Feb 2011 22:05:26 +0000
Subject: [Freeswitch-users] G729
In-Reply-To: <Pine.LNX.4.64.1102082226490.17347@suse.blindi.net>
References: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>
	<AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>
	<Pine.LNX.4.64.1102082226490.17347@suse.blindi.net>
Message-ID: <AANLkTinwKW6cYrHRfj+j_e8nogX8TtRSynbXH-5LraxK@mail.gmail.com>

No :(

On 8 February 2011 21:27, Thomas Hoellriegel <admin at blindi.net> wrote:

> Is g729 a free codec?
>
>
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/listinfo/blinde-misc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From admin at blindi.net  Wed Feb  9 01:07:07 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Tue, 8 Feb 2011 23:07:07 +0100 (CET)
Subject: [Freeswitch-users] using skype on a text only ubuntusystem?
In-Reply-To: <AANLkTi=hJVcNEDVUpCdXpEZJjYSBq0z6p2QGBRR0Jc73@mail.gmail.com>
References: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>
	<AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>
	<59C40C2C-351C-4544-8B50-099DACD6A292@ipeva.fr>
	<Pine.LNX.4.64.1102082229480.17347@suse.blindi.net>
	<AANLkTi=hJVcNEDVUpCdXpEZJjYSBq0z6p2QGBRR0Jc73@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1102082259120.17347@suse.blindi.net>

Hi  Giovanni,

> Wiki.freeswitch.org/mod_skypopen
I can.t find a textbaseinstallation.
in the wiki:
    On Linux the Skype client can use a lot of CPU if you don't customize
    your setup. To lower its CPU consumption, you use the Xvfb "fake" X
    server and (more importantly) the snd-dummy ALSA "fake" sound driver.

I  can.t use X!!!
---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From me at nevian.org  Wed Feb  9 01:17:02 2011
From: me at nevian.org (Serge Yuriev)
Date: Wed, 09 Feb 2011 01:17:02 +0300
Subject: [Freeswitch-users] destination number variable
Message-ID: <283881297203422@web100.yandex.ru>

Hello

In INFO app after call i see (shortened for readability)
2011-02-09 00:20:00.824547 [INFO] mod_dptools.c:1202 CHANNEL_DATA:
Caller-Direction: [inbound]
Caller-Username: [nevian]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [Serge S. Yuriev]
Caller-Caller-ID-Number: [nevian]
Caller-Callee-ID-Name: [Outbound Call]
Caller-Callee-ID-Number: [79645835822 at 81.16.114.33]
Caller-Network-Addr: [109.173.67.229]
Caller-ANI: [nevian]
Caller-Destination-Number: [79645835822]
variable_sip_from_display: [Serge S. Yuriev]
variable_sip_full_from: ["Serge S. Yuriev" <sip:nevian at cranz.nevian.org>;tag=624579331]
variable_sip_full_to: [<sip:79645835822 at cranz.nevian.org>]
variable_sip_req_user: [79645835822]
variable_sip_req_uri: [79645835822 at cranz.nevian.org]
variable_sip_req_host: [cranz.nevian.org]
variable_sip_to_user: [79645835822]
variable_sip_to_uri: [79645835822 at cranz.nevian.org]

In mod_radius_cdr profile->destination_number returns it as 79645835822 at 81.16.114.33
Which field should I use to get only number?

Termination via h323 if this matters

-- 
wbr,
Serge


From sjmudd at pobox.com  Wed Feb  9 01:18:27 2011
From: sjmudd at pobox.com (Simon J Mudd)
Date: 08 Feb 2011 23:18:27 +0100
Subject: [Freeswitch-users] Enabling extensions with passwords and limiting
	network access via acls (Was: Confusing SIP auth failure
	logging message?)
In-Reply-To: <20110207073124.GA5255@mad06.wl0.org>
References: <20110206232236.GA10501@mad06.wl0.org>
	<20110207073124.GA5255@mad06.wl0.org>
Message-ID: <m3aai69nm4.fsf_-_@mad06.wl0.org>

I don't think I got a definitive answer to the question I posed before:

sjmudd at pobox.com (Simon J Mudd) writes:

...

> That is I have an Asterisk configuration which I am trying to
> migrate from and can easily configure in sip.conf:
> 
> [1000]
> username=1000
> type=friend
> secret=1234567890
> context=xxxxxx
> host=dynamic
> registersip=yes
> deny=0.0.0.0/0.0.0.0
> permit=88.100.50.0/255.255.255.0  -- this is not a real network range but you get the idea.
> nat=yes
> call-limit=1
> ...
> 
> This specifies a user for registration who:
> (1) must provide a password
> (2) can only register from the given network range
> (3) is only allowed to make 1 call at a time

I see that there are ways to implement (3) though it seems that's more on
a per gateway basis than a per extension basis. That's ok.

What really interests me is implementing (1) _and_ (2) together.  Is this
possible? If not it would certainly be a nice new feature.

Perhaps the default FreeSWITCH configuration should limit access to
the default extensions to be registered only from the networks defined
in localnet.auto.  This reduces exposure to external bad
software.

Even if I can configure this extra limitation myself manually I'd be
happy as this would basically leave me more comfortable with
FreeSWITCH running unattended. I'm not confident to do that now if I
configure any gateways due to the issues I've had before.

Simon


From daniel_wells at byu.edu  Wed Feb  9 01:19:37 2011
From: daniel_wells at byu.edu (Daniel Wells)
Date: Tue, 8 Feb 2011 15:19:37 -0700
Subject: [Freeswitch-users] mod_dingaling & inbound audio
In-Reply-To: <AANLkTim-5LDhfQhHk_2LNLffvTa1JyBXq9U4B3LSY3Nq@mail.gmail.com>
References: <AANLkTim-5LDhfQhHk_2LNLffvTa1JyBXq9U4B3LSY3Nq@mail.gmail.com>
Message-ID: <8C68232BC9314C40BBCDDAA480F7B01AEB3C406615@harrow.exch.ad.byu.edu>

I have been trying to do the same.  After a few hiccups and some help on the IRC channel I was able to get outbound working this morning.  I didn't have to mess with any rtp settings though (which could explain my problem below).

However I am having issues with inbound.  It seems like it is trying but I get a strange error in the log regarding a missing port (http://pastebin.freeswitch.org/15311).

    - Daniel Wells


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Locutis of Borg
Sent: Tuesday, February 08, 2011 12:07 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] mod_dingaling & inbound audio

Followed instructions on wiki.  Enabled mod_dingaling to register over at GV.  Set external rtp to right address.  Outbound calling terminates, and outbound audio works, but seems that I can not get return audio.

FS is on DMZ - same problem
Moved to internal (behind NAT, port forwarding) - same problem
Moved back to DMZ and killed iptables and router spi - same problem

In and out calling on SIP to provider works fine in both cases.

So my questions are:
  Can mod_dingaling be used for GV or not?
  Since it uses TLS, NAT shouldn't be an issue?
     and
  Does GV not like us FS users?

Thank you for you help.

"Resistance is Fruitful"
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From gmaruzz at gmail.com  Wed Feb  9 01:29:53 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 8 Feb 2011 23:29:53 +0100
Subject: [Freeswitch-users] using skype on a text only ubuntusystem?
In-Reply-To: <Pine.LNX.4.64.1102082259120.17347@suse.blindi.net>
References: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>
	<AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>
	<59C40C2C-351C-4544-8B50-099DACD6A292@ipeva.fr>
	<Pine.LNX.4.64.1102082229480.17347@suse.blindi.net>
	<AANLkTi=hJVcNEDVUpCdXpEZJjYSBq0z6p2QGBRR0Jc73@mail.gmail.com>
	<Pine.LNX.4.64.1102082259120.17347@suse.blindi.net>
Message-ID: <AANLkTi=ioaNQ2GgsmFQUVtxFpkknh6mdQNQNELGemsG0@mail.gmail.com>

Why you don't read it ?
Xvfb is a fake X server, and is exactly what you want.
Read all the wiki page.
;)


On 2/8/11, Thomas Hoellriegel <admin at blindi.net> wrote:
> Hi  Giovanni,
>
>> Wiki.freeswitch.org/mod_skypopen
> I can.t find a textbaseinstallation.
> in the wiki:
>     On Linux the Skype client can use a lot of CPU if you don't customize
>     your setup. To lower its CPU consumption, you use the Xvfb "fake" X
>     server and (more importantly) the snd-dummy ALSA "fake" sound driver.
>
> I  can.t use X!!!
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/listinfo/blinde-misc
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From ranjtech at gmail.com  Wed Feb  9 01:30:49 2011
From: ranjtech at gmail.com (RR)
Date: Tue, 8 Feb 2011 17:30:49 -0500
Subject: [Freeswitch-users] Microsoft Speech Server/UCMA Experience
Message-ID: <AANLkTin85QEEhWE1rLCuELVXvqBqhMVt9QGkwcqfJ1iP@mail.gmail.com>

 Hello All,

I was wondering if anyone's tried to use OR currently use the Microsoft
Speech Server or their UCMA 3.x SDK etc. as their ASR/TTS backend/engines
etc. If yes, then what's their experience? Please Note, this does NOT need
to be integrated with FS ala MRCP or some module/plugin etc. I just wanted
to know if someone's used it and and what their experience has been in both,
TTS and ASR.

Thanks
\RR
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From infos at madovsky.org  Wed Feb  9 01:49:27 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 8 Feb 2011 17:49:27 -0500
Subject: [Freeswitch-users] play_and_get_digits question
References: <EBED8C31D06F4A319D978371EE1CF24E@e1705><AANLkTi=AVwJ4c80z9F_06+8EyUHrokLzEtRcPGGpa4jG@mail.gmail.com><C2CAC53C1A8B4CB3ADC3B5EF97F56DDD@e1705><Pine.LNX.4.64.1102080811280.11715@suse.blindi.net><5DCA5F42FA5A4AAA8D5F34DFDDA51157@e1705>
	<Pine.LNX.4.64.1102082215470.17347@suse.blindi.net>
Message-ID: <1C1910E661614BED8E7721AA12F14E61@e1705>

in default.xml config in dialplan
there's all for you already
look at the top of thise file "limit extension"

Regards


----- Original Message ----- 
From: "Thomas Hoellriegel" <admin at blindi.net>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, February 08, 2011 4:25 PM
Subject: Re: [Freeswitch-users] play_and_get_digits question


Hi Madovsky
Sorry, but i.m. not a FS guru.
i don.t understand your solution.
your create an dialplan with play and get digits.
the application don.t have a if then else construction. is the maximum of
failled attemps exited.
You say: you have use the limitapplication.
can you post these lines please?
thank you


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From infos at madovsky.org  Wed Feb  9 01:51:10 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 8 Feb 2011 17:51:10 -0500
Subject: [Freeswitch-users] nibblebill sql requests
References: <AD71F946CFA942858324FAC60CF95D76@e1705>
	<AANLkTimLYmuCBRTrJUmVC62QiYohOOMSUVRJT+Oi-Dus@mail.gmail.com>
Message-ID: <76DAFB8BB68C49B79DD0C1A8DCCDBC03@e1705>

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


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From steveayre at gmail.com  Wed Feb  9 02:03:44 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 8 Feb 2011 23:03:44 +0000
Subject: [Freeswitch-users] Enabling extensions with passwords and
	limiting network access via acls (Was: Confusing SIP auth
	failure logging message?)
In-Reply-To: <m3aai69nm4.fsf_-_@mad06.wl0.org>
References: <20110206232236.GA10501@mad06.wl0.org>
	<20110207073124.GA5255@mad06.wl0.org>
	<m3aai69nm4.fsf_-_@mad06.wl0.org>
Message-ID: <142D32EB-C2E9-4E15-A181-E074614D8EA7@gmail.com>

Either set a variable and use it with check_acl in the dialplan for that user, or use mod_XML_curl and don't return the user entry if the network_addr doesn't match the cidr acl.

Steve on iPhone

On 8 Feb 2011, at 22:18, Simon J Mudd <sjmudd at pobox.com> wrote:

> I don't think I got a definitive answer to the question I posed before:
> 
> sjmudd at pobox.com (Simon J Mudd) writes:
> 
> ...
> 
>> That is I have an Asterisk configuration which I am trying to
>> migrate from and can easily configure in sip.conf:
>> 
>> [1000]
>> username=1000
>> type=friend
>> secret=1234567890
>> context=xxxxxx
>> host=dynamic
>> registersip=yes
>> deny=0.0.0.0/0.0.0.0
>> permit=88.100.50.0/255.255.255.0  -- this is not a real network range but you get the idea.
>> nat=yes
>> call-limit=1
>> ...
>> 
>> This specifies a user for registration who:
>> (1) must provide a password
>> (2) can only register from the given network range
>> (3) is only allowed to make 1 call at a time
> 
> I see that there are ways to implement (3) though it seems that's more on
> a per gateway basis than a per extension basis. That's ok.
> 
> What really interests me is implementing (1) _and_ (2) together.  Is this
> possible? If not it would certainly be a nice new feature.
> 
> Perhaps the default FreeSWITCH configuration should limit access to
> the default extensions to be registered only from the networks defined
> in localnet.auto.  This reduces exposure to external bad
> software.
> 
> Even if I can configure this extra limitation myself manually I'd be
> happy as this would basically leave me more comfortable with
> FreeSWITCH running unattended. I'm not confident to do that now if I
> configure any gateways due to the issues I've had before.
> 
> Simon
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 sjmudd at pobox.com  Wed Feb  9 02:06:43 2011
From: sjmudd at pobox.com (Simon J Mudd)
Date: 09 Feb 2011 00:06:43 +0100
Subject: [Freeswitch-users] Enabling extensions with passwords and
	limiting network access via acls (Was: Confusing SIP auth
	failure logging message?)
In-Reply-To: <m3aai69nm4.fsf_-_@mad06.wl0.org>
References: <20110206232236.GA10501@mad06.wl0.org>
	<20110207073124.GA5255@mad06.wl0.org>
	<m3aai69nm4.fsf_-_@mad06.wl0.org>
Message-ID: <m3fwryt9bw.fsf@mad06.wl0.org>

sjmudd at pobox.com (Simon J Mudd) writes:

> I don't think I got a definitive answer to the question I posed before:
> 
> sjmudd at pobox.com (Simon J Mudd) writes:
> 
> ...
> 
> > That is I have an Asterisk configuration which I am trying to
> > migrate from and can easily configure in sip.conf:
> > 
> > [1000]
> > username=1000
> > type=friend
> > secret=1234567890
> > context=xxxxxx
> > host=dynamic
> > registersip=yes
> > deny=0.0.0.0/0.0.0.0
> > permit=88.100.50.0/255.255.255.0  -- this is not a real network range but you get the idea.
> > nat=yes
> > call-limit=1
> > ...
> > 
> > This specifies a user for registration who:
> > (1) must provide a password
> > (2) can only register from the given network range
> > (3) is only allowed to make 1 call at a time
> 
> I see that there are ways to implement (3) though it seems that's more on
> a per gateway basis than a per extension basis. That's ok.
> 
> What really interests me is implementing (1) _and_ (2) together.  Is this
> possible? If not it would certainly be a nice new feature.

Yes, it is. I finally figured that I need something like:

=== snip 1000.xml ===
<include>
  <user id="1000">
    <params>
      <param name="password" value="$${default_password}"/>
      <param name="vm-password" value="1000"/>
      <!-- limit this extension to be used from network 10.1.2.0/24 -->
      <!-- <param name="auth-acl" value="10.1.2.0/24"/> -->
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="1000"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Extension 1000"/>
      <variable name="effective_caller_id_number" value="1000"/>
      <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>
==== snip ====

So thanks. That's good.

Can this somehow be done at a global level in directory/default.xml in
the <params> section?  Would be nice to comment these options in both
places so they are easier to find and apply if needed.

> Perhaps the default FreeSWITCH configuration should limit access to
> the default extensions to be registered only from the networks defined
> in localnet.auto.  This reduces exposure to external bad
> software.

So an option like <param name="auth-acl" value="localnet.auto"/> would
be a nice option to have as this is safe and "auto-configure" at the
same time. In fact it seems that this also works:

I have 2 phones configured to register one configured with the
ip-based acl, the other with the name-based acl.

2011-02-09 00:01:18.332336 [DEBUG] sofia_reg.c:2370 IP [10.1.2.35] passed ACL check [localnet.auto]
2011-02-09 00:01:18.361773 [DEBUG] sofia_reg.c:2370 IP [10.1.2.35] passed ACL check [10.1.2.0/24]

That's even better. _Please_ add this to the default configuration
files, even if it's commented out, as it makes it easy to see how to
tighten down on the config if that is wanted.

Simon


From steveayre at gmail.com  Wed Feb  9 02:09:02 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 8 Feb 2011 23:09:02 +0000
Subject: [Freeswitch-users] Overriding 480 with 200 from other leg
In-Reply-To: <02af01cbc7ba$bd8055e0$388101a0$@com>
References: <02af01cbc7ba$bd8055e0$388101a0$@com>
Message-ID: <E23366A8-DF65-458A-98D5-866E149E4C29@gmail.com>

What is the hangup cause for the blegs from the CDRs?

Can you get debug logs and a siptrace for the call?

Look at the uuid parameter of mod_logfile which would make the log easy to get using grep.

A siptrace could be collected via tcpdump to a file and tshark to extract from it using Call-ID. You can also use 'sofia global siptrace on', but that'll be hard to extract on a high volume server.

I'm guessing the bleg is returning an error cause in the reason header, or there's a codec problem which the log would probably show.

Steve on iPhone

On 8 Feb 2011, at 18:05, "Peder" <peder at networkoblivion.com> wrote:

> I've got a strange issue and I am not 100% sure how to get around it.  
> 
> We have a system setup where a user calls in and enters a code.  A perl
> script is called that does an http get to find a phone number to dial, it
> dials the number, bridges the two and then it records the call.  
> 
> Once the code is verified and we have the number, we set two carriers to
> call, primary and backup.  This all works great 95%+ of the time.  We do 40k
> calls per month and about 50 calls a week run into a scenario that I am not
> sure of.   On the ones that don't work, it gets all the way through bridging
> the calls with no problem, but when the user hangs up, it doesn't hang up,
> it falls to the backup bridge and calls the destination number again our
> carrier2, all the while still recording.  We get 3 CDR records, incoming,
> call out carrier1 and call out carrier2.
> 
> Here is the snippet of the perl script that does the dialing ($bridge and
> $backupbridge are the two carriers to call).  If $bridge fails, it goes back
> to $backupbridge just fine:
> 
> $session->execute("set","continue_on_fail=true");
> $session->execute("bridge","sofia/gateway/$bridge/$PhoneNo");
> $session->execute("export","nolocal:CARRIER=${backupbridge}");
> $session->execute("bridge","sofia/gateway/$backupbridge/$PhoneNo");
> 
> 
> Every now and then we see this in the logs "Overriding SIP cause 480 with
> 200 from the other leg":
> 
> 
> 2011-01-19 09:24:50.751368 [NOTICE] sofia.c:481 Hangup
> sofia/external/1xxxxxx8003 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> 2011-01-19 09:24:50.753368 [DEBUG] switch_channel.c:2071 Send signal
> sofia/external/1xxxxxx8003 [KILL]
> 2011-01-19 09:24:50.753368 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/external/1xxxxxx8003 [BREAK]
> 2011-01-19 09:24:50.761364 [DEBUG] switch_ivr_bridge.c:478
> sofia/external/1xxxxxx8003 ending bridge by request from read function
> 2011-01-19 09:24:50.761364 [DEBUG] switch_ivr_bridge.c:565 BRIDGE THREAD
> DONE [sofia/external/1xxxxxx8003]
> 2011-01-19 09:24:50.761364 [DEBUG] switch_ivr_bridge.c:585 Send signal
> sofia/internal/yyyyyy6168 at 72.249.14.242 [BREAK]
> 2011-01-19 09:24:50.761364 [DEBUG] switch_core_state_machine.c:351
> (sofia/external/1xxxxxx8003) State EXCHANGE_MEDIA going to sleep
> 2011-01-19 09:24:50.761364 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/1xxxxxx8003) Running State Change CS_HANGUP
> 2011-01-19 09:24:50.761364 [DEBUG] switch_core_state_machine.c:499
> (sofia/external/1xxxxxx8003) State HANGUP
> 2011-01-19 09:24:50.761364 [DEBUG] mod_sofia.c:405
> sofia/external/1xxxxxx8003 Overriding SIP cause 480 with 200 from the other
> leg
> 2011-01-19 09:24:50.761364 [DEBUG] mod_sofia.c:411 Channel
> sofia/external/1xxxxxx8003 hanging up, cause: NORMAL_CLEARING
> 2011-01-19 09:24:50.761364 [DEBUG] switch_core_state_machine.c:46
> sofia/external/1xxxxxx8003 Standard HANGUP, cause: NORMAL_CLEARING
> 2011-01-19 09:24:50.761364 [DEBUG] switch_core_state_machine.c:499
> (sofia/external/1xxxxxx8003) State HANGUP going to sleep
> 2011-01-19 09:24:50.761364 [DEBUG] switch_core_state_machine.c:333
> (sofia/external/1xxxxxx8003) State Change CS_HANGUP -> CS_REPORTING
> 2011-01-19 09:24:50.761364 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/external/1xxxxxx8003 [BREAK]
> 2011-01-19 09:24:50.761364 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/1xxxxxx8003) Running State Change CS_REPORTING
> 2011-01-19 09:24:50.761364 [DEBUG] switch_core_state_machine.c:590
> (sofia/external/1xxxxxx8003) State REPORTING
> 2011-01-19 09:24:50.763475 [DEBUG] switch_core_session.c:638 Send signal
> sofia/internal/yyyyyy6168 at 72.249.14.242 [BREAK]
> 2011-01-19 09:24:50.763475 [DEBUG] switch_ivr_bridge.c:565 BRIDGE THREAD
> DONE [sofia/internal/yyyyyy6168 at 72.249.14.242]
> 2011-01-19 09:24:50.763475 [DEBUG] switch_ivr_bridge.c:585 Send signal
> sofia/external/1xxxxxx8003 [BREAK]
> EXECUTE sofia/internal/yyyyyy6168 at 72.249.14.242
> export(nolocal:CARRIER=carrier2)
> 2011-01-19 09:24:50.766348 [DEBUG] switch_core_state_machine.c:53
> sofia/external/1xxxxxx8003 Standard REPORTING, cause: NORMAL_CLEARING
> 2011-01-19 09:24:50.766348 [DEBUG] switch_core_state_machine.c:590
> (sofia/external/1xxxxxx8003) State REPORTING going to sleep
> 2011-01-19 09:24:50.766348 [DEBUG] switch_core_state_machine.c:327
> (sofia/external/1xxxxxx8003) State Change CS_REPORTING -> CS_DESTROY
> 2011-01-19 09:24:50.766348 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/external/1xxxxxx8003 [BREAK]
> 2011-01-19 09:24:50.766348 [DEBUG] switch_core_session.c:1161 Session 96783
> (sofia/external/1xxxxxx8003) Locked, Waiting on external entities
> 2011-01-19 09:24:50.766348 [NOTICE] switch_core_session.c:1179 Session 96783
> (sofia/external/1xxxxxx8003) Ended
> 2011-01-19 09:24:50.766348 [NOTICE] switch_core_session.c:1181 Close Channel
> sofia/external/1xxxxxx8003 [CS_DESTROY]
> 2011-01-19 09:24:50.766348 [DEBUG] mod_dptools.c:898 EXPORT (REMOTE ONLY)
> [CARRIER]=[carrier2]
> 2011-01-19 09:24:50.766348 [DEBUG] switch_core_state_machine.c:428
> (sofia/external/1xxxxxx8003) Running State Change CS_DESTROY
> 2011-01-19 09:24:50.766348 [DEBUG] switch_core_state_machine.c:439
> (sofia/external/1xxxxxx8003) State DESTROY
> 2011-01-19 09:24:50.766348 [DEBUG] mod_sofia.c:338
> sofia/external/1xxxxxx8003 SOFIA DESTROY
> EXECUTE sofia/internal/yyyyyy6168 at 72.249.14.242
> bridge(sofia/gateway/carrier2/1xxxxxx8003)
> 
> 
> I am guessing that carrier1 is sending back a 480, so FS thinks it never
> worked and then bridges out carrier2.  Our FS is pretty old, FreeSWITCH
> Version 1.0.trunk (17043), but I don't think it is an FS issue, I think it
> is just a dialplan issue that I don't know how to resolve.  Any idea how to
> stop this from happening?  
> 
> Peder
> 
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 bcxml at hotmail.com  Wed Feb  9 02:08:21 2011
From: bcxml at hotmail.com (Brian Campbell)
Date: Tue, 8 Feb 2011 18:08:21 -0500
Subject: [Freeswitch-users] Microsoft Speech Server/UCMA Experience
In-Reply-To: <AANLkTin85QEEhWE1rLCuELVXvqBqhMVt9QGkwcqfJ1iP@mail.gmail.com>
References: <AANLkTin85QEEhWE1rLCuELVXvqBqhMVt9QGkwcqfJ1iP@mail.gmail.com>
Message-ID: <SNT135-w480E7C008C08CC312BAB92B7EA0@phx.gbl>


I have been using FreeSwitch with Speech Server 2007 for about 2 years now.
 
I have done six blog postings about integrating the two. 
 
You can check it out at http://gotspeech.net/blogs/verbalinput/archive/2009/12/23/speech-server-2007-marries-freeswitch-part-1-introduction.aspx
 
 
 
Brian Campbell
 
 
 


From: ranjtech at gmail.com
Date: Tue, 8 Feb 2011 17:30:49 -0500
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Microsoft Speech Server/UCMA Experience



Hello All,
 
I was wondering if anyone's tried to use OR currently use the Microsoft Speech Server or their UCMA 3.x SDK etc. as their ASR/TTS backend/engines etc. If yes, then what's their experience? Please Note, this does NOT need to be integrated with FS ala MRCP or some module/plugin etc. I just wanted to know if someone's used it and and what their experience has been in both, TTS and ASR.
 
Thanks
\RR
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.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 Feb  9 02:30:47 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 9 Feb 2011 01:30:47 +0200
Subject: [Freeswitch-users] nibblebill sql requests
In-Reply-To: <76DAFB8BB68C49B79DD0C1A8DCCDBC03@e1705>
References: <AD71F946CFA942858324FAC60CF95D76@e1705>
	<AANLkTimLYmuCBRTrJUmVC62QiYohOOMSUVRJT+Oi-Dus@mail.gmail.com>
	<76DAFB8BB68C49B79DD0C1A8DCCDBC03@e1705>
Message-ID: <AANLkTik2y8MwfJhhETqg1eQ6=UnukpjV2ezAEDHY9CH9@mail.gmail.com>

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 <avi at avimarcus.net>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *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
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>
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From infos at madovsky.org  Wed Feb  9 02:49:19 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 8 Feb 2011 18:49:19 -0500
Subject: [Freeswitch-users] nibblebill sql requests
References: <AD71F946CFA942858324FAC60CF95D76@e1705><AANLkTimLYmuCBRTrJUmVC62QiYohOOMSUVRJT+Oi-Dus@mail.gmail.com><76DAFB8BB68C49B79DD0C1A8DCCDBC03@e1705>
	<AANLkTik2y8MwfJhhETqg1eQ6=UnukpjV2ezAEDHY9CH9@mail.gmail.com>
Message-ID: <D4B3974A30B342AAB9EB1F0AF8E9EDB3@e1705>

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






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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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    _______________________________________________
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From ranjtech at gmail.com  Wed Feb  9 03:12:26 2011
From: ranjtech at gmail.com (RR)
Date: Tue, 8 Feb 2011 19:12:26 -0500
Subject: [Freeswitch-users] Microsoft Speech Server/UCMA Experience
In-Reply-To: <SNT135-w480E7C008C08CC312BAB92B7EA0@phx.gbl>
References: <AANLkTin85QEEhWE1rLCuELVXvqBqhMVt9QGkwcqfJ1iP@mail.gmail.com>
	<SNT135-w480E7C008C08CC312BAB92B7EA0@phx.gbl>
Message-ID: <AANLkTimb6_g92WExet3yeJFsNTmKjE=1ygwUjMZdOHE+@mail.gmail.com>

On Tue, Feb 8, 2011 at 6:08 PM, Brian Campbell <bcxml at hotmail.com> wrote:

> I have been using FreeSwitch with Speech Server 2007 for about 2 years now.
>
> I have done six blog postings about integrating the two.
>
> You can check it out at
> http://gotspeech.net/blogs/verbalinput/archive/2009/12/23/speech-server-2007-marries-freeswitch-part-1-introduction.aspx
>
>

Hi Brian,

So funny...I was just reading that blog when your email hit my mailbox :)

So is Speech Server 2007 the latest version or is there something later than
that available too?

Thanks
\R
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From brian at freeswitch.org  Wed Feb  9 03:21:16 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 8 Feb 2011 18:21:16 -0600
Subject: [Freeswitch-users] Confusing SIP auth failure logging message?
In-Reply-To: <20110206232236.GA10501@mad06.wl0.org>
References: <20110206232236.GA10501@mad06.wl0.org>
Message-ID: <71AD6D54-FA88-49DC-BEB3-5CA91F307BB1@freeswitch.org>

You say "error message" but really its just a WARNING in huge all caps... anyway I giggle every time someone does this... can you please put your patch on jira.

/b

On Feb 6, 2011, at 5:22 PM, Simon J Mudd wrote:

> I've been looking at trying to configure tighter controls for extensions that register.
> Doing so made me trigger this error message (adjusted slightly):
> 
> 2011-02-07 00:07:51.343303 [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1000 at sip.example.com] from ip 192.168.4.99
> 2011-02-07 00:07:51.343303 [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1000 at sip.example.com] from ip 192.168.4.99
> 
> Looking at the message it is not clear if the SIP authentication has succeeded or failed.
> Judging by the code it seems this is meant to represent a SIP auth failure. If so should
> the code not be patched as shown?
> 
> diff --git a/src/mod/endpoints/mod_sofia/sofia_reg.c b/src/mod/endpoints/mod_sofia/sofia_reg.c
> index 631cbdb..e42c777 100644
> --- a/src/mod/endpoints/mod_sofia/sofia_reg.c
> +++ b/src/mod/endpoints/mod_sofia/sofia_reg.c
> @@ -1244,7 +1244,7 @@ uint8_t sofia_reg_handle_register(nua_t *nua, sofia_profile_t *profile, nua_hand
>                }
>                /* Log line added to support Fail2Ban */
>                if (sofia_test_pflag(profile, PFLAG_LOG_AUTH_FAIL)) {
> -                       switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "SIP auth challenge (%s) on sofia profile '%s' "
> +                       switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "SIP auth failure (%s) on sofia profile '%s' "
>                                                          "for [%s@%s] from ip %s\n", (regtype == REG_INVITE) ? "INVITE" : "REGISTER", 
>                                                          profile->name, to_user, to_host, network_ip);
>                }
> 
> Thanks,
> 
> Simon
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Wed Feb  9 03:27:11 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 8 Feb 2011 18:27:11 -0600
Subject: [Freeswitch-users] Enabling extensions with passwords and
	limiting network access via acls (Was: Confusing SIP auth
	failure logging message?)
In-Reply-To: <m3aai69nm4.fsf_-_@mad06.wl0.org>
References: <20110206232236.GA10501@mad06.wl0.org>
	<20110207073124.GA5255@mad06.wl0.org>
	<m3aai69nm4.fsf_-_@mad06.wl0.org>
Message-ID: <5313ADE6-FDD7-4124-AF56-885A16C15A46@freeswitch.org>

TL;DR

So while you can buy a gun and bullets who's fault is it when you get shot in the foot?  Same thing really.  Our dialplan is rather secure since I designed it and I fully understand how our security model works.   Our default is just an example of how to use FreeSWITCH.  I could do a service and just delete it all and leave it up to you to figure it all out but I feel learning by example is a great way to see how to use the software.

/b

On Feb 8, 2011, at 4:18 PM, Simon J Mudd wrote:

> I don't think I got a definitive answer to the question I posed before:



From peder at networkoblivion.com  Wed Feb  9 04:29:07 2011
From: peder at networkoblivion.com (Peder)
Date: Tue, 8 Feb 2011 19:29:07 -0600
Subject: [Freeswitch-users] Overriding 480 with 200 from other leg
In-Reply-To: <E23366A8-DF65-458A-98D5-866E149E4C29@gmail.com>
References: <02af01cbc7ba$bd8055e0$388101a0$@com>
	<E23366A8-DF65-458A-98D5-866E149E4C29@gmail.com>
Message-ID: <045801cbc7f8$bf10d7c0$3d328740$@com>

>What is the hangup cause for the blegs from the CDRs?

All 3 legs show " NORMAL_CLEARING".


>Can you get debug logs and a siptrace for the call?

I had some debug logs on the first message.  It started where it hangs up
the first b-leg and then shows that it bridges another b-leg because of the
480 message.  And as I stated before, the first b-leg was fine and the call
was up for 5+ minutes.  One user hung up, but the call didn't drop
completely, it continued on in the dial plan and bridged another b-leg.


>I'm guessing the bleg is returning an error cause in the reason header, or
there's a codec problem which the log would probably show.

I don't think that is it as the first 2 legs work fine.  The problem is when
the first b leg hangs up, sometimes (maybe 1 in 20), it continues on and
bridges a second b-leg to the a-leg.  Any codec negotiation should be long
gone.  



From u2nsam at gmail.com  Wed Feb  9 06:56:06 2011
From: u2nsam at gmail.com (Sam)
Date: Wed, 9 Feb 2011 09:26:06 +0530
Subject: [Freeswitch-users] G729
In-Reply-To: <AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>
References: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>
	<AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>
Message-ID: <AANLkTinTY5Z+UGfVTRb_8qM-Jb1WbFDdBjreg2PKGB2W@mail.gmail.com>

Hi,

What changes did you made to accommodate all the variations of G729.

Regards
Sam

On Tue, Feb 8, 2011 at 9:01 PM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:

> G729 only. No A's or AB.
>
> If you want to pass through other variants of G.729, you can simply
> modify mod_g729.c to include additional variants of G729
>
> Which by the way, I've pasted the area of the code you might want to
> modify. I've done this on mine and it worked fine as pass through.
>
> 00217         SWITCH_ADD_CODEC(codec_interface, "G.729A");
> 00218         for (count = 12; count > 0; count--) {
> 00219                 switch_core_codec_add_implementation(pool,
> codec_interface,
> 00220
>                        SWITCH_CODEC_TYPE_AUDIO, 18, "G729A", NULL,
> 8000, 8000, 8000,
> 00221
>                        mpf * count, spf * count, bpf * count, ebpf *
> count, 1, count * 10,
> 00222
>                        switch_g729_init, switch_g729_encode,
> switch_g729_decode, switch_g729_destroy);
> 00223         }
>
>
> On Tue, Feb 8, 2011 at 7:02 AM, Sam <u2nsam at gmail.com> wrote:
> > Hi,
> >
> > Which type of  G.729 codec does freeswitch uses in passthrough mode ?
> > Is it g729a or g729b or g729ab .
> >
> > 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 msc at freeswitch.org  Wed Feb  9 07:00:48 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 8 Feb 2011 22:00:48 -0600
Subject: [Freeswitch-users] Enabling extensions with passwords and
 limiting network access via acls (Was: Confusing SIP auth failure logging
 message?)
In-Reply-To: <m3aai69nm4.fsf_-_@mad06.wl0.org>
References: <20110206232236.GA10501@mad06.wl0.org>
	<20110207073124.GA5255@mad06.wl0.org>
	<m3aai69nm4.fsf_-_@mad06.wl0.org>
Message-ID: <AANLkTimZLYfpKVmS7rACTEG3M42eM5a3s5gghhr6Mju8@mail.gmail.com>

On Tue, Feb 8, 2011 at 4:18 PM, Simon J Mudd <sjmudd at pobox.com> wrote:

> I don't think I got a definitive answer to the question I posed before:
>
> I know you came on to the #freeswitch IRC channel but I don't recall where
you left off in your quest for FreeSWITCH knowledge. I'll provide
information here with links to relevant wiki articles, etc. all inline.
However, let me just put these two points out there for everyone:

#1 - If, when learning FreeSWITCH, you feel overwhelmed because there are so
many different config options, etc. then remember this: FreeSWITCH is
designed for power and flexibility, not simplicity. A carrier-grade,
multi-protocol soft-switch is supposed to be complex.

#2 - The default configuration is just an example of some of the many cool
things you can do with FreeSWITCH. It is not meant to be a turn-key,
out-of-box solution to put into a production environment, even a SOHO
environment. This is a feature, not a bug.


> sjmudd at pobox.com (Simon J Mudd) writes:
>
> ...
>

<snip>

> > This specifies a user for registration who:
> > (1) must provide a password
> > (2) can only register from the given network range
> > (3) is only allowed to make 1 call at a time


> I see that there are ways to implement (3) though it seems that's more on
> a per gateway basis than a per extension basis. That's ok.
>
> What really interests me is implementing (1) _and_ (2) together.  Is this
> possible? If not it would certainly be a nice new feature.
>

This is definitely possible. There are two ways to do it that I can think
of:
#1 - The "proper" way with mod_xml_curl
#2 - The less proper but still quite functional way of using a user's
channel variables and dialplan.*

Either option requires a bit of reading up. The good news is that setting
the user information in the directory magically makes FS ready to do auth
challenges for calls and user registrations.

For more information on mod_xml_curl see:
http://wiki.freeswitch.org/wiki/Mod_xml_curl
Also see Raymond's samples in the freeswitch-contrib repo:
http://fisheye.freeswitch.org/browse/freeswitch-contrib/intralanman/PHP/fs_curl


> Perhaps the default FreeSWITCH configuration should limit access to
> the default extensions to be registered only from the networks defined
> in localnet.auto.  This reduces exposure to external bad
> software.
>
Many people don't want this limitation turned on by default. If we did this
then a lot of people would be asking for the ability to have external phones
be able to access FreeSWITCH by default. Besides, there are other
alternatives, such as using a firewall.


> Even if I can configure this extra limitation myself manually I'd be
> happy as this would basically leave me more comfortable with
> FreeSWITCH running unattended. I'm not confident to do that now if I
> configure any gateways due to the issues I've had before.
>
Understood. It is always best to know your VoIP before you throw it out on
the public Internet.

Your blog had several other issues so I'll number them here and then comment
on them below.
Questions:

1. Default FreeSWITCH logging maybe too verbose
2. XML file breakage
3. Relatedly, comments need to be in "<!-- xxx -->" format
4. The external profile should log auth failures by default
5. FreeSWITCH should have some sort of rate-limiting
6. It should be more obvious how to configure network ACLs for for
extensions, and these should be configured by default
7. For registration, a client can use FreeSWITCH's IP address as well as
domain name
8. "For trunk connections if you have a DID number you expect the VoIP
provider to call you..."
9. Allow rate-limiting to/from a gateway or extension

Answers:

1. This goes back to FreeSWITCH being a massively huge piece of software
with lots of configuration options. Having verbose debugging is very handy.
It is much better for a new person to have too much information at first and
then learn about what logging to turn off. The other alternative is for a
new person to come around with a question about something that happened and
not have any logs to help debug. Log files can be archived or deleted, but
the information that never got logged is worthless.

2. XML "file breakage" is well-documented here:
http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Chasing_down_XML_errors
If you wanted to you could take the aggregated file (freeswitch.xml.fsxml)
and put it in conf/freeswitch.xml as the only config file. Of course, now
you're editing a 4500 line XML file... :)

3. These are just standard XML comments. They take the form:
<!-- comment -->
The XML standard dictates that you have two dashes. (Your example seems to
show single dashes.) The XML standard also states that you may not nest
comments, nor may you include "--" inside of a comment. See
http://www.w3.org/TR/REC-xml/#sec-comments for more info

4. This is a reasonable suggestion. We have a saying: "patches accepted". If
anyone has a moment to implement *AND TEST* this please do so and send us a
patch on Jira.

5. Rate-limiting is absolutely possible in FreeSWITCH, and it is not limited
to SIP traffic. Pretty much anything you can think of can be rate-limited
using mod_limit. More information here:
http://wiki.freeswitch.org/wiki/Mod_limit

6. Not sure how to make it more obvious than this:
http://wiki.freeswitch.org/wiki/Acl#Users

7. We allow both IP and domain name by default to lower the barrier to
entry. The default config is supposed to "just work" in a sandbox so that
new users can learn. One great way to learn is by setting up a system and
then making mods to the default config and see what breaks.

8. This is a fun one... What you ask for is something that would work in an
ideal situation, however SIP is involved so idealism must give way to
realism. Most providers will blow up and die if you send them 401's or 407's
so that's not a logical choice for the default. (Again, nothing preventing
you from modifying the default configs to suit your personal needs.) You can
also use ACLs to limit the source of incoming calls, but then you run the
risk of them reconfiguring and changing IP addresses. As long as you are
prepared for that scenario then you're fine, but most new users probably
could do without us adding that burden to them by default.

9. See #5

I hope we haven't scared you off! I'm sure that you can (and will) overcome
all of your challenges in configuring FreeSWITCH for your needs. The
community is here to assist.

Regards,
Michael S Collins (IRC: mercutioviz)

*BONUS CONTENT:
Here's an example of limiting a user's IP address. Add a line like this to
your user's XML file:
<variable name="allowed_ipaddrs" value="^(192\.168\.1\.|10\.10\.16\.161)"/>
This is a regex that matches an exact address (10.10.16.161) or matches a
partial IP address (192.168.1.x)

Then do something like this in your dialplan:
  <extension name="check user's IP address" continue="true">
    <condition field="network_addr" expression="${allowed_ipaddrs}">
      <action application="log" data="WARNING IP Addr test passed"/>
      <anti-action application="log" data="ERR IP Addr test is an EPIC
FAIL"/>
      <anti-action application="answer"/>
      <anti-action application="sleep" data="1000"/>
      <anti-action application="playback"
data="ivr/ivr-all_your_call_are_belong_to_us.wav"/>
      <anti-action application="hangup"/>
    </condition>
  </extension>
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From youstinayacoub at gmail.com  Wed Feb  9 04:05:30 2011
From: youstinayacoub at gmail.com (sherrypioro)
Date: Wed, 9 Feb 2011 01:05:30 +0000 (UTC)
Subject: [Freeswitch-users] Video on X-lite .. FSV module
Message-ID: <loom.20110209T020200-551@post.gmane.org>

Hi,


I am trying to record or play video on X-lite through Freeswitch. When I do a
video call it says faild to send your video. 
 
I have FSV module installed on freeswitch and running. 

Any Ideas???? 

Thank you!!!



From victor.chukalovskiy at utoronto.ca  Wed Feb  9 06:56:25 2011
From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy)
Date: Tue, 08 Feb 2011 22:56:25 -0500
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <AANLkTimfzmpeR2N2sAf_QpAX0g3=BSRS0P-bPogh-Zbu@mail.gmail.com>
References: <4D4DC16F.50500@utoronto.ca>
	<AANLkTimfzmpeR2N2sAf_QpAX0g3=BSRS0P-bPogh-Zbu@mail.gmail.com>
Message-ID: <4D521069.5000802@utoronto.ca>

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From msc at freeswitch.org  Wed Feb  9 07:10:23 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 8 Feb 2011 22:10:23 -0600
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <4D521069.5000802@utoronto.ca>
References: <4D4DC16F.50500@utoronto.ca>
	<AANLkTimfzmpeR2N2sAf_QpAX0g3=BSRS0P-bPogh-Zbu@mail.gmail.com>
	<4D521069.5000802@utoronto.ca>
Message-ID: <AANLkTi=54t5pYfSuLuEnwhycAoou8xhcLNDjVuKX8vJR@mail.gmail.com>

put the license file in /etc/freeswitch
-MC

On Tue, Feb 8, 2011 at 9:56 PM, Victor Chukalovskiy <
victor.chukalovskiy at utoronto.ca> wrote:

>  Hi Michael and others,
>
> After updating to the latest GIT mod_com_g729 loads successfully.
>
> Next problem appears:
> when typing "g729_info" freeswitch replies "can't contact licence server".
> g729_available gives "False"
>
> How to solve this?
>
> I have my xxxxx.conf license file in the root of FS install directory
> /opt/fs
> Should it be placed elsewhere?
>
> Also, xxxxx.conf was created with previous non-successful install of
> mod_com_g729
> Should I run validator again with the same license key?
>
> Thank you,
> Victor
>
> On 07/02/11 10:36 AM, Michael Collins wrote:
>
> Any chance you can update to latest git? Your life will be easier. There
> have been notable improvements in FS in the past few months.
>
>  -MC
>
> On Sat, Feb 5, 2011 at 3:30 PM, Victor Chukalovskiy <
> victor.chukalovskiy at utoronto.ca> wrote:
>
>>  Hello,
>>
>> After purchasing a few licenses and installing the latest
>> fsg729-191-installer
>> I'm getting the following error when trying to load the mod_com_g729:
>> > 2011-02-04 14:27:17.637046 [CRIT] switch_loadable_module.c:926 Error
>> Loading module /opt/fs/mod/mod_com_g729.so
>> > **Trying to load an out of date module, please rebuild the module.**
>>
>> Also noticed that g729 installer ran with a couple errors:
>> > ./installer: line 62: ldconfig: command not found
>> > ./installer: line 49: useradd: command not found
>>  Any help or solution is much appreciated.
>>
>> -Victor
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> 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 curriegrad2004 at gmail.com  Wed Feb  9 07:36:28 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 8 Feb 2011 20:36:28 -0800
Subject: [Freeswitch-users] G729
In-Reply-To: <AANLkTinTY5Z+UGfVTRb_8qM-Jb1WbFDdBjreg2PKGB2W@mail.gmail.com>
References: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>
	<AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>
	<AANLkTinTY5Z+UGfVTRb_8qM-Jb1WbFDdBjreg2PKGB2W@mail.gmail.com>
Message-ID: <AANLkTimVC094R9P4sSurT2XSq_kiLUEpqvAFWjHwVPpL@mail.gmail.com>

Ah, so any devices that use a SDP of G729A is considered to be broken
in implementation, right?

Most of the cisco-linksys SPA adapters actually default to a SDP of
G729A for some reason.

On Tue, Feb 8, 2011 at 7:56 PM, Sam <u2nsam at gmail.com> wrote:
> Hi,
>
> What changes did you made to accommodate all the variations of G729.
>
> Regards
> Sam
>
> On Tue, Feb 8, 2011 at 9:01 PM, curriegrad2004 <curriegrad2004 at gmail.com>
> wrote:
>>
>> G729 only. No A's or AB.
>>
>> If you want to pass through other variants of G.729, you can simply
>> modify mod_g729.c to include additional variants of G729
>>
>> Which by the way, I've pasted the area of the code you might want to
>> modify. I've done this on mine and it worked fine as pass through.
>>
>> 00217 ? ? ? ? SWITCH_ADD_CODEC(codec_interface, "G.729A");
>> 00218 ? ? ? ? for (count = 12; count > 0; count--) {
>> 00219 ? ? ? ? ? ? ? ? switch_core_codec_add_implementation(pool,
>> codec_interface,
>> 00220
>> ? ? ? ? ? ? ? ? ? ? ? ?SWITCH_CODEC_TYPE_AUDIO, 18, "G729A", NULL,
>> 8000, 8000, 8000,
>> 00221
>> ? ? ? ? ? ? ? ? ? ? ? ?mpf * count, spf * count, bpf * count, ebpf *
>> count, 1, count * 10,
>> 00222
>> ? ? ? ? ? ? ? ? ? ? ? ?switch_g729_init, switch_g729_encode,
>> switch_g729_decode, switch_g729_destroy);
>> 00223 ? ? ? ? }
>>
>>
>> On Tue, Feb 8, 2011 at 7:02 AM, Sam <u2nsam at gmail.com> wrote:
>> > Hi,
>> >
>> > Which type of? G.729 codec does freeswitch uses in passthrough mode ?
>> > Is it g729a or g729b or g729ab .
>> >
>> > 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 mthakershi at gmail.com  Wed Feb  9 07:58:14 2011
From: mthakershi at gmail.com (Malay Thakershi)
Date: Tue, 8 Feb 2011 22:58:14 -0600
Subject: [Freeswitch-users] Can ESL work from remote computer?
Message-ID: <AANLkTi=+c3VdnbD2R5vyaTrvBqvFfoOjRG8StPs06TtZ@mail.gmail.com>

Hello,

I am trying to connect to FS server via ESL from another computer on the
LAN.

-------------
            eslConnection = new ESLconnection("10.25.20.202", "8021",
"ClueCon");

            if (eslConnection.Connected() != ESL_SUCCESS)
            {
                return "Fail";
            }
-------------

But I keep getting not connected.

I checked firewall. I tried addressing FS computer by name/IP.

Do I need to change anything in the FS config?

Please share your views.

Malay
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From xyangni at gmail.com  Wed Feb  9 08:43:31 2011
From: xyangni at gmail.com (xuyan yang)
Date: Wed, 9 Feb 2011 05:43:31 +0000
Subject: [Freeswitch-users] Can ESL work from remote computer?
In-Reply-To: <AANLkTi=+c3VdnbD2R5vyaTrvBqvFfoOjRG8StPs06TtZ@mail.gmail.com>
References: <AANLkTi=+c3VdnbD2R5vyaTrvBqvFfoOjRG8StPs06TtZ@mail.gmail.com>
Message-ID: <AANLkTik40JBV-_o-FC-Lrsp0nfdecMv9nysqYYfMf4ub@mail.gmail.com>

you need to set listen-ip to 0.0.0.0 in event_socket.conf.xml

On Wed, Feb 9, 2011 at 4:58 AM, Malay Thakershi <mthakershi at gmail.com>wrote:

> Hello,
>
> I am trying to connect to FS server via ESL from another computer on the
> LAN.
>
> -------------
>             eslConnection = new ESLconnection("10.25.20.202", "8021",
> "ClueCon");
>
>             if (eslConnection.Connected() != ESL_SUCCESS)
>             {
>                 return "Fail";
>             }
> -------------
>
> But I keep getting not connected.
>
> I checked firewall. I tried addressing FS computer by name/IP.
>
> Do I need to change anything in the FS config?
>
> Please share your views.
>
> 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
>
>
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From infos at madovsky.org  Wed Feb  9 09:06:27 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 9 Feb 2011 01:06:27 -0500
Subject: [Freeswitch-users] nibblebill sql requests
Message-ID: <17E3569462944217BBAE2AA65EFDEE97@e1705>

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






------------------------------------------------------------------------


        _______________________________________________
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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 admin at blindi.net  Wed Feb  9 10:05:18 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Wed, 9 Feb 2011 08:05:18 +0100 (CET)
Subject: [Freeswitch-users] using skype on a text only ubuntusystem?
In-Reply-To: <AANLkTi=ioaNQ2GgsmFQUVtxFpkknh6mdQNQNELGemsG0@mail.gmail.com>
References: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>
	<AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>
	<59C40C2C-351C-4544-8B50-099DACD6A292@ipeva.fr>
	<Pine.LNX.4.64.1102082229480.17347@suse.blindi.net>
	<AANLkTi=hJVcNEDVUpCdXpEZJjYSBq0z6p2QGBRR0Jc73@mail.gmail.com>
	<Pine.LNX.4.64.1102082259120.17347@suse.blindi.net>
	<AANLkTi=ioaNQ2GgsmFQUVtxFpkknh6mdQNQNELGemsG0@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1102090757040.24802@suse.blindi.net>

Hi Giovanni

i have read the wike thankx.
the wiki described i must install the skypeclient. The client need a X 
fake server ok.
i have found in the wike a secoundway. The way is 
vnc. The problem is: a vncclient  is not working   in a text only 
system. No framebuffer is defined.

Do you have skype running on a console like system?

---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From gmaruzz at gmail.com  Wed Feb  9 10:25:22 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 9 Feb 2011 08:25:22 +0100
Subject: [Freeswitch-users] using skype on a text only ubuntusystem?
In-Reply-To: <Pine.LNX.4.64.1102090757040.24802@suse.blindi.net>
References: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>
	<AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>
	<59C40C2C-351C-4544-8B50-099DACD6A292@ipeva.fr>
	<Pine.LNX.4.64.1102082229480.17347@suse.blindi.net>
	<AANLkTi=hJVcNEDVUpCdXpEZJjYSBq0z6p2QGBRR0Jc73@mail.gmail.com>
	<Pine.LNX.4.64.1102082259120.17347@suse.blindi.net>
	<AANLkTi=ioaNQ2GgsmFQUVtxFpkknh6mdQNQNELGemsG0@mail.gmail.com>
	<Pine.LNX.4.64.1102090757040.24802@suse.blindi.net>
Message-ID: <AANLkTimm3GTpER_+MxHYBmBzJW5j-jXSP3r3xSxe0GGU@mail.gmail.com>

Skypeclient needs X. You can't use skypeclient without X.
Mod_skypopen needs skypeclient.
Xvfb is a fake X for console only systems.
Xvnc is another fake X for console only systems.
Mod_skypopen runs and has been installed only on "console only" systems.
Eg: rack sysyems in datacenters far away from you, where you connect via ssh.
Is all very extensively documented on the wiki page.
If something is not clear in that wiki page, please let me know
exactly which issue in which part of the wiki page so I can amend it.

On 2/9/11, Thomas Hoellriegel <admin at blindi.net> wrote:
> Hi Giovanni
>
> i have read the wike thankx.
> the wiki described i must install the skypeclient. The client need a X
> fake server ok.
> i have found in the wike a secoundway. The way is
> vnc. The problem is: a vncclient  is not working   in a text only
> system. No framebuffer is defined.
>
> Do you have skype running on a console like system?
>
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/listinfo/blinde-misc
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From sharad at coraltele.com  Wed Feb  9 10:26:13 2011
From: sharad at coraltele.com (sharad)
Date: Wed, 9 Feb 2011 12:56:13 +0530
Subject: [Freeswitch-users] Dial Plan Query
References: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com><AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com><AANLkTinTY5Z+UGfVTRb_8qM-Jb1WbFDdBjreg2PKGB2W@mail.gmail.com>
	<AANLkTimVC094R9P4sSurT2XSq_kiLUEpqvAFWjHwVPpL@mail.gmail.com>
Message-ID: <7B1C266951034491A4A3F20987486A26@sharad>

Hi All

I defined a simple dialplan shown below to speak out the current date & 
time.

Now on dialing 555 followed by any digit / digits, freeswitch starts 
speaking the date & time.

I want to define the duplicate access code of 555 say 444 means on dialing 
444 followed by digits also should perform the same action.

Will be greatfull, if someone can help.

Thanks & regards
Sharad
<!-- Check Current Date & Time from here -->

    <extension name="speak_date_time">

        <condition field="destination_number" expression="^555(.*)$">

        <action application="answer"/>

        <action application="sleep" data="500"/>

        <action application="playback" data="ivr/today_date_is.wav"/>

        <action application="sleep" data="100"/>

        <action application="phrase" data="saydate"/>

        <action application="sleep" data="500"/>

        <action application="playback" data="ivr/and_time_is.wav"/>

        <action application="sleep" data="200"/>

        <action application="phrase" data="saytime"/>

        </condition>

</extension>





From sharad at coraltele.com  Wed Feb  9 10:56:48 2011
From: sharad at coraltele.com (sharad)
Date: Wed, 9 Feb 2011 13:26:48 +0530
Subject: [Freeswitch-users] Dial Plan Query
References: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com><AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com><AANLkTinTY5Z+UGfVTRb_8qM-Jb1WbFDdBjreg2PKGB2W@mail.gmail.com><AANLkTimVC094R9P4sSurT2XSq_kiLUEpqvAFWjHwVPpL@mail.gmail.com>
	<7B1C266951034491A4A3F20987486A26@sharad>
Message-ID: <0C76A5CD3FA44E0EA3905772779D187A@sharad>

Hi

I just realized the mistake I was doing. issue solved. Sorry for taking your 
time.

Thanks

----- Original Message ----- 
From: "sharad" <sharad at coraltele.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, February 09, 2011 12:56 PM
Subject: [Freeswitch-users] Dial Plan Query


> Hi All
>
> I defined a simple dialplan shown below to speak out the current date & 
> time.
>
> Now on dialing 555 followed by any digit / digits, freeswitch starts 
> speaking the date & time.
>
> I want to define the duplicate access code of 555 say 444 means on dialing 
> 444 followed by digits also should perform the same action.
>
> Will be greatfull, if someone can help.
>
> Thanks & regards
> Sharad
> <!-- Check Current Date & Time from here -->
>
>    <extension name="speak_date_time">
>
>        <condition field="destination_number" expression="^555(.*)$">
>
>        <action application="answer"/>
>
>        <action application="sleep" data="500"/>
>
>        <action application="playback" data="ivr/today_date_is.wav"/>
>
>        <action application="sleep" data="100"/>
>
>        <action application="phrase" data="saydate"/>
>
>        <action application="sleep" data="500"/>
>
>        <action application="playback" data="ivr/and_time_is.wav"/>
>
>        <action application="sleep" data="200"/>
>
>        <action application="phrase" data="saytime"/>
>
>        </condition>
>
> </extension>
>
>
>
>
>
> 



From steveayre at gmail.com  Wed Feb  9 11:03:14 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 9 Feb 2011 08:03:14 +0000
Subject: [Freeswitch-users] Overriding 480 with 200 from other leg
In-Reply-To: <045801cbc7f8$bf10d7c0$3d328740$@com>
References: <02af01cbc7ba$bd8055e0$388101a0$@com>
	<E23366A8-DF65-458A-98D5-866E149E4C29@gmail.com>
	<045801cbc7f8$bf10d7c0$3d328740$@com>
Message-ID: <00056758-D1F2-47CB-B670-08E9F001F0AC@gmail.com>

Quite right about codecs. Ignore that.

More debug logging is never a bad thing. It looks like the hangup has already started with what you posted and you're only showing us one leg. Siptrace on both legs really would be good to see what starts the hangup.

Steve on iPhone

On 9 Feb 2011, at 01:29, "Peder" <peder at networkoblivion.com> wrote:

>> What is the hangup cause for the blegs from the CDRs?
> 
> All 3 legs show " NORMAL_CLEARING".
> 
> 
>> Can you get debug logs and a siptrace for the call?
> 
> I had some debug logs on the first message.  It started where it hangs up
> the first b-leg and then shows that it bridges another b-leg because of the
> 480 message.  And as I stated before, the first b-leg was fine and the call
> was up for 5+ minutes.  One user hung up, but the call didn't drop
> completely, it continued on in the dial plan and bridged another b-leg.
> 
> 
>> I'm guessing the bleg is returning an error cause in the reason header, or
> there's a codec problem which the log would probably show.
> 
> I don't think that is it as the first 2 legs work fine.  The problem is when
> the first b leg hangs up, sometimes (maybe 1 in 20), it continues on and
> bridges a second b-leg to the a-leg.  Any codec negotiation should be long
> gone.  
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Wed Feb  9 11:05:34 2011
From: u2nsam at gmail.com (Sam)
Date: Wed, 9 Feb 2011 13:35:34 +0530
Subject: [Freeswitch-users] RTP keep alive
Message-ID: <AANLkTik-o+kpsLyTTtjL09Zg4M9a23aPygFj1d-V=cTb@mail.gmail.com>

Hello,

We have a situation wherein we need to keep alive RTP ,is there any
parameter to do that,
because when someone is on long conversation and not talking fro brief
duration the call
disconnects .

Regards
Sam
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From u2nsam at gmail.com  Wed Feb  9 11:49:02 2011
From: u2nsam at gmail.com (Sam)
Date: Wed, 9 Feb 2011 14:19:02 +0530
Subject: [Freeswitch-users] RTP keep alive
In-Reply-To: <AANLkTik-o+kpsLyTTtjL09Zg4M9a23aPygFj1d-V=cTb@mail.gmail.com>
References: <AANLkTik-o+kpsLyTTtjL09Zg4M9a23aPygFj1d-V=cTb@mail.gmail.com>
Message-ID: <AANLkTi=aUeWAo+54O3U_LAK1kEr-U3wTNty+4AS5HG42@mail.gmail.com>

I have this in the settings,

    <param name="rtp-timeout-sec" value="3000"/>
    <param name="rtp-hold-timeout-sec" value="18000"/>

Any thing i need more regarding that ?

Regards
Sam

On Wed, Feb 9, 2011 at 1:35 PM, Sam <u2nsam at gmail.com> wrote:

> Hello,
>
> We have a situation wherein we need to keep alive RTP ,is there any
> parameter to do that,
> because when someone is on long conversation and not talking fro brief
> duration the call
> disconnects .
>
> Regards
> Sam
>
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From peter.olsson at visionutveckling.se  Wed Feb  9 11:48:42 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 9 Feb 2011 09:48:42 +0100
Subject: [Freeswitch-users] Jira status emails seems to be down since some
	time?
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58ADB7276A@cooper>

I think this has been reported before, but it seems like Jira status emails are not being sent anymore - and I suspect this might cause the core devs to miss out some of the updates :)

Just wanted to let you know..

/Peter


From victor.chukalovskiy at utoronto.ca  Wed Feb  9 07:41:59 2011
From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy)
Date: Tue, 08 Feb 2011 23:41:59 -0500
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <AANLkTi=54t5pYfSuLuEnwhycAoou8xhcLNDjVuKX8vJR@mail.gmail.com>
References: <4D4DC16F.50500@utoronto.ca>	<AANLkTimfzmpeR2N2sAf_QpAX0g3=BSRS0P-bPogh-Zbu@mail.gmail.com>	<4D521069.5000802@utoronto.ca>
	<AANLkTi=54t5pYfSuLuEnwhycAoou8xhcLNDjVuKX8vJR@mail.gmail.com>
Message-ID: <4D521B17.2020408@utoronto.ca>

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From david.ponzone at ipeva.fr  Wed Feb  9 11:54:14 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 9 Feb 2011 09:54:14 +0100
Subject: [Freeswitch-users] G729
In-Reply-To: <AANLkTimVC094R9P4sSurT2XSq_kiLUEpqvAFWjHwVPpL@mail.gmail.com>
References: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>
	<AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>
	<AANLkTinTY5Z+UGfVTRb_8qM-Jb1WbFDdBjreg2PKGB2W@mail.gmail.com>
	<AANLkTimVC094R9P4sSurT2XSq_kiLUEpqvAFWjHwVPpL@mail.gmail.com>
Message-ID: <A6DC62B7-B6A9-4B05-A642-1281870DEDF4@ipeva.fr>

I remember that in the Linksys SPA-crap, you can actually change the string they will use in the SDP.
A such feature means a lot :)

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 09/02/2011 ? 05:36, curriegrad2004 a ?crit :

> Ah, so any devices that use a SDP of G729A is considered to be broken
> in implementation, right?
> 
> Most of the cisco-linksys SPA adapters actually default to a SDP of
> G729A for some reason.
> 
> On Tue, Feb 8, 2011 at 7:56 PM, Sam <u2nsam at gmail.com> wrote:
>> Hi,
>> 
>> What changes did you made to accommodate all the variations of G729.
>> 
>> Regards
>> Sam
>> 
>> On Tue, Feb 8, 2011 at 9:01 PM, curriegrad2004 <curriegrad2004 at gmail.com>
>> wrote:
>>> 
>>> G729 only. No A's or AB.
>>> 
>>> If you want to pass through other variants of G.729, you can simply
>>> modify mod_g729.c to include additional variants of G729
>>> 
>>> Which by the way, I've pasted the area of the code you might want to
>>> modify. I've done this on mine and it worked fine as pass through.
>>> 
>>> 00217         SWITCH_ADD_CODEC(codec_interface, "G.729A");
>>> 00218         for (count = 12; count > 0; count--) {
>>> 00219                 switch_core_codec_add_implementation(pool,
>>> codec_interface,
>>> 00220
>>>                        SWITCH_CODEC_TYPE_AUDIO, 18, "G729A", NULL,
>>> 8000, 8000, 8000,
>>> 00221
>>>                        mpf * count, spf * count, bpf * count, ebpf *
>>> count, 1, count * 10,
>>> 00222
>>>                        switch_g729_init, switch_g729_encode,
>>> switch_g729_decode, switch_g729_destroy);
>>> 00223         }
>>> 
>>> 
>>> On Tue, Feb 8, 2011 at 7:02 AM, Sam <u2nsam at gmail.com> wrote:
>>>> Hi,
>>>> 
>>>> Which type of  G.729 codec does freeswitch uses in passthrough mode ?
>>>> Is it g729a or g729b or g729ab .
>>>> 
>>>> 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 david.ponzone at ipeva.fr  Wed Feb  9 11:55:58 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 9 Feb 2011 09:55:58 +0100
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <4D521B17.2020408@utoronto.ca>
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>
Message-ID: <ED3B5594-990A-4ACC-8206-856650E56F38@ipeva.fr>

There is a file coming with the module which explains the installation quite perfectly.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 09/02/2011 ? 05:41, Victor Chukalovskiy a ?crit :

> This didn't help.
> I reloaded mod_com_g729, tried g729_info and the same "can't contact licence server" error
> -Victor
> 
> On 08/02/11 11:10 PM, Michael Collins wrote:
>> 
>> put the license file in /etc/freeswitch
>> -MC
>> 
>> On Tue, Feb 8, 2011 at 9:56 PM, Victor Chukalovskiy <victor.chukalovskiy at utoronto.ca> wrote:
>> Hi Michael and others,
>> 
>> After updating to the latest GIT mod_com_g729 loads successfully.
>> 
>> Next problem appears: 
>> when typing "g729_info" freeswitch replies "can't contact licence server".
>> g729_available gives "False"
>> 
>> How to solve this?
>> 
>> I have my xxxxx.conf license file in the root of FS install directory /opt/fs
>> Should it be placed elsewhere?
>> 
>> Also, xxxxx.conf was created with previous non-successful install of mod_com_g729
>> Should I run validator again with the same license key?
>> 
>> Thank you,
>> Victor
>> 
>> On 07/02/11 10:36 AM, Michael Collins wrote:
>>> 
>>> Any chance you can update to latest git? Your life will be easier. There have been notable improvements in FS in the past few months. 
>>> 
>>> -MC
>>> 
>>> On Sat, Feb 5, 2011 at 3:30 PM, Victor Chukalovskiy <victor.chukalovskiy at utoronto.ca> wrote:
>>> Hello,
>>> 
>>> After purchasing a few licenses and installing the latest fsg729-191-installer
>>> I'm getting the following error when trying to load the mod_com_g729:
>>> > 2011-02-04 14:27:17.637046 [CRIT] switch_loadable_module.c:926 Error Loading module /opt/fs/mod/mod_com_g729.so
>>> > **Trying to load an out of date module, please rebuild the module.**
>>> 
>>> Also noticed that g729 installer ran with a couple errors:
>>> > ./installer: line 62: ldconfig: command not found
>>> > ./installer: line 49: useradd: command not found
>>>  Any help or solution is much appreciated.
>>> 
>>> -Victor 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.ponzone at ipeva.fr  Wed Feb  9 12:01:43 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 9 Feb 2011 10:01:43 +0100
Subject: [Freeswitch-users] RTP keep alive
In-Reply-To: <AANLkTik-o+kpsLyTTtjL09Zg4M9a23aPygFj1d-V=cTb@mail.gmail.com>
References: <AANLkTik-o+kpsLyTTtjL09Zg4M9a23aPygFj1d-V=cTb@mail.gmail.com>
Message-ID: <37B9FCDE-F129-4763-93E5-77DBB5B137FA@ipeva.fr>

What do you mean by brief ?
Some seconds ?
If you have that behaviour, it's probably that there is an equipement in the path which has a very low RTP-session-timeout.
And that is not FreeSWITCH.

Also, if your RTP stops when you don't talk, it means you have VAD enabled.
But even wit the best VAD in the world, there should be some packets.
It would be probably best if you can trace a such call, and also if you may tell us if FreeSWITCH only is involved, or if there is a provider equipement somewhere in the picture.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 09/02/2011 ? 09:05, Sam a ?crit :

> Hello,
> 
> We have a situation wherein we need to keep alive RTP ,is there any parameter to do that,
> because when someone is on long conversation and not talking fro brief duration the call
> disconnects .
> 
> 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 elihayunfs at gmail.com  Wed Feb  9 12:04:19 2011
From: elihayunfs at gmail.com (Eli Hayun)
Date: Wed, 9 Feb 2011 11:04:19 +0200
Subject: [Freeswitch-users] setting/getting variable in callcenter
Message-ID: <AANLkTi=AK6OpN_sUjJ=J+SJtPhoNMZJ438dgg2uJfobO@mail.gmail.com>

Hi
I am using mod_callcenter and have a question

1) How do I get the value from a callcenter variable from a php or lua
script?
suppose that I want to change the MOH of a queue: I see that the variable
"cc_moh_override" is the one that I have to change, but how do I do that? I
dont have the session (i am in a php /lua script, that run outside FS and
calling to it with ESL api)

2) I am using ESL call to get the list of agents but I want to do more then
just see the list of agents.
I see that when the phone is ringing in an agent phone the status changed to
"Receiving" but without the uid or the caller phone number. How do I get
this information (I prefer to get this without using an event)

Thanks

-- 
Eli Hayun
Hebrew University Jerusalem
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From steveayre at gmail.com  Wed Feb  9 12:25:00 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 9 Feb 2011 09:25:00 +0000
Subject: [Freeswitch-users] G729
In-Reply-To: <AANLkTimVC094R9P4sSurT2XSq_kiLUEpqvAFWjHwVPpL@mail.gmail.com>
References: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>
	<AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>
	<AANLkTinTY5Z+UGfVTRb_8qM-Jb1WbFDdBjreg2PKGB2W@mail.gmail.com>
	<AANLkTimVC094R9P4sSurT2XSq_kiLUEpqvAFWjHwVPpL@mail.gmail.com>
Message-ID: <AANLkTinMd84JqL-s8qcuWCB_LxEbOXSaVMzA1Cs5s5h7@mail.gmail.com>

Yes. They'll only be able to talk to each other nicely.

As Steve Underwood said though, FS apparently has a fudge in it that'll
allow it to work in this case since there's so many of the wild.

-Steve


On 9 February 2011 04:36, curriegrad2004 <curriegrad2004 at gmail.com> wrote:

> Ah, so any devices that use a SDP of G729A is considered to be broken
> in implementation, right?
>
> Most of the cisco-linksys SPA adapters actually default to a SDP of
> G729A for some reason.
>
> On Tue, Feb 8, 2011 at 7:56 PM, Sam <u2nsam at gmail.com> wrote:
> > Hi,
> >
> > What changes did you made to accommodate all the variations of G729.
> >
> > Regards
> > Sam
> >
> > On Tue, Feb 8, 2011 at 9:01 PM, curriegrad2004 <curriegrad2004 at gmail.com
> >
> > wrote:
> >>
> >> G729 only. No A's or AB.
> >>
> >> If you want to pass through other variants of G.729, you can simply
> >> modify mod_g729.c to include additional variants of G729
> >>
> >> Which by the way, I've pasted the area of the code you might want to
> >> modify. I've done this on mine and it worked fine as pass through.
> >>
> >> 00217         SWITCH_ADD_CODEC(codec_interface, "G.729A");
> >> 00218         for (count = 12; count > 0; count--) {
> >> 00219                 switch_core_codec_add_implementation(pool,
> >> codec_interface,
> >> 00220
> >>                        SWITCH_CODEC_TYPE_AUDIO, 18, "G729A", NULL,
> >> 8000, 8000, 8000,
> >> 00221
> >>                        mpf * count, spf * count, bpf * count, ebpf *
> >> count, 1, count * 10,
> >> 00222
> >>                        switch_g729_init, switch_g729_encode,
> >> switch_g729_decode, switch_g729_destroy);
> >> 00223         }
> >>
> >>
> >> On Tue, Feb 8, 2011 at 7:02 AM, Sam <u2nsam at gmail.com> wrote:
> >> > Hi,
> >> >
> >> > Which type of  G.729 codec does freeswitch uses in passthrough mode ?
> >> > Is it g729a or g729b or g729ab .
> >> >
> >> > 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 steveayre at gmail.com  Wed Feb  9 12:28:28 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 9 Feb 2011 09:28:28 +0000
Subject: [Freeswitch-users] Can ESL work from remote computer?
In-Reply-To: <AANLkTi=+c3VdnbD2R5vyaTrvBqvFfoOjRG8StPs06TtZ@mail.gmail.com>
References: <AANLkTi=+c3VdnbD2R5vyaTrvBqvFfoOjRG8StPs06TtZ@mail.gmail.com>
Message-ID: <AANLkTimzfOos3ofz4nkyv_P9Sc7Twfy-UuHMvg+N7iEN@mail.gmail.com>

As xuyan said, you need to change the listen-ip to 0.0.0.0.

The default is to only listen on 127.0.0.1, which is good for security since
the default password ClueCon is well known.

When you change the listen-ip you really should change the default password
too, otherwise anyone will not only get control of FS but also get full
access to your server as the FS user (there's a system api command).

-Steve



On 9 February 2011 04:58, Malay Thakershi <mthakershi at gmail.com> wrote:

> Hello,
>
> I am trying to connect to FS server via ESL from another computer on the
> LAN.
>
> -------------
>             eslConnection = new ESLconnection("10.25.20.202", "8021",
> "ClueCon");
>
>             if (eslConnection.Connected() != ESL_SUCCESS)
>             {
>                 return "Fail";
>             }
> -------------
>
> But I keep getting not connected.
>
> I checked firewall. I tried addressing FS computer by name/IP.
>
> Do I need to change anything in the FS config?
>
> Please share your views.
>
> 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
>
>
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From steveayre at gmail.com  Wed Feb  9 12:30:27 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 9 Feb 2011 09:30:27 +0000
Subject: [Freeswitch-users] Video on X-lite .. FSV module
In-Reply-To: <loom.20110209T020200-551@post.gmane.org>
References: <loom.20110209T020200-551@post.gmane.org>
Message-ID: <AANLkTi=urew0--m1NXKuDpw74aDdCh6FERdPQ2ZyvwT-@mail.gmail.com>

Can you enable debug logging and post it?

-Steve


On 9 February 2011 01:05, sherrypioro <youstinayacoub at gmail.com> wrote:

> Hi,
>
>
> I am trying to record or play video on X-lite through Freeswitch. When I do
> a
> video call it says faild to send your video.
>
> I have FSV module installed on freeswitch and running.
>
> Any Ideas????
>
> Thank you!!!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mattdfong at gmail.com  Wed Feb  9 12:55:55 2011
From: mattdfong at gmail.com (Matthew Fong)
Date: Wed, 9 Feb 2011 01:55:55 -0800
Subject: [Freeswitch-users] Using an Empty FROM field for SIP Messages
Message-ID: <AANLkTikEyL=NpmdqM+Vx=rxaM2+zEyxuJr_oGdFDWLwU@mail.gmail.com>

I am trying to setup a customer who wanted to utilize the airespring
national did presence, where in they rewrite the Caller ID of outbound
calls. In order to do this tho, they say I need to send a blank caller id.
FreeSWITCH will send 0000000000 if the caller id is not specified.
Airespring wants something formatted like

INVITE sip:jungar at alpha-org.com SIP/2.0

Via: SIP/2.0/UDP zetamachine.beta-org.com:5060;branch=as82je8ei4kr

Max-Forwards: 70

To:  2556112121 <sip:jungar at alpha-org.com>

From: <sip:jsmith at beta-org.com>;tag=13456

Call-ID: @zetamachine.beta-org.com


with the From: field looking like any of the following

From: <sip:jsmith at beta-org.com>;tag=13456
From: <  @208.76.54.59>;tag=13456
From:  @208.76.54.59;tag=13456

The closest I got was to make FreeSWITCH to the following, but airespring
won't budge. Can anyone tell me how to make the From string as airespring
wants for their national DID presence service. Thanks.
From: "" <sip:0000000000 at 208.76.54.59>;tag=UyNBXjyUUay8m
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From david.ponzone at ipeva.fr  Wed Feb  9 13:06:10 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 9 Feb 2011 11:06:10 +0100
Subject: [Freeswitch-users] Using an Empty FROM field for SIP Messages
In-Reply-To: <AANLkTikEyL=NpmdqM+Vx=rxaM2+zEyxuJr_oGdFDWLwU@mail.gmail.com>
References: <AANLkTikEyL=NpmdqM+Vx=rxaM2+zEyxuJr_oGdFDWLwU@mail.gmail.com>
Message-ID: <D810E0FC-0474-45E8-9C9A-C8D9D438032C@ipeva.fr>

Try setting effective_caller_id_number to " "

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 09/02/2011 ? 10:55, Matthew Fong a ?crit :

> I am trying to setup a customer who wanted to utilize the airespring national did presence, where in they rewrite the Caller ID of outbound calls. In order to do this tho, they say I need to send a blank caller id. FreeSWITCH will send 0000000000 if the caller id is not specified. Airespring wants something formatted like
> 
> INVITE sip:jungar at alpha-org.com SIP/2.0
> Via: SIP/2.0/UDP zetamachine.beta-org.com:5060;branch=as82je8ei4kr
> Max-Forwards: 70
> To:  2556112121 <sip:jungar at alpha-org.com>
> From: <sip:jsmith at beta-org.com>;tag=13456
> Call-ID: @zetamachine.beta-org.com
> 
> 
> with the From: field looking like any of the following
> 
> From: <sip:jsmith at beta-org.com>;tag=13456
> From: <  @208.76.54.59>;tag=13456
> From:  @208.76.54.59;tag=13456
> 
> The closest I got was to make FreeSWITCH to the following, but airespring won't budge. Can anyone tell me how to make the From string as airespring wants for their national DID presence service. Thanks.
> From: "" <sip:0000000000 at 208.76.54.59>;tag=UyNBXjyUUay8m
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Feb  9 13:15:19 2011
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Wed, 9 Feb 2011 13:15:19 +0300
Subject: [Freeswitch-users] Skype conference call
Message-ID: <AANLkTimXtgR67ZyXXv+YhjtA3dxEsMVEUe5XZUZ6mDD5@mail.gmail.com>

Can I participate in a Skype conference call via FreeSWITCH/Skypopen?
Is there any howto on that?

- Dmitry Bely


From gmaruzz at gmail.com  Wed Feb  9 13:20:39 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 9 Feb 2011 11:20:39 +0100
Subject: [Freeswitch-users] Skype conference call
In-Reply-To: <AANLkTimXtgR67ZyXXv+YhjtA3dxEsMVEUe5XZUZ6mDD5@mail.gmail.com>
References: <AANLkTimXtgR67ZyXXv+YhjtA3dxEsMVEUe5XZUZ6mDD5@mail.gmail.com>
Message-ID: <AANLkTimCFJ4YszYVrWa_7dHq7QTYRtvj4RsrhiZPzLGu@mail.gmail.com>

I'm working on mod_skypopen.
Can you describe more on the use case?
what you would like to do exactly?
I'll take this one as a feature request, and I'm willing to evaluate
its feasability, just need more info on what exactly is the expected
feature.

-giovanni

On Wed, Feb 9, 2011 at 11:15 AM, Dmitry Bely <dmitry.bely at gmail.com> wrote:
> Can I participate in a Skype conference call via FreeSWITCH/Skypopen?
> Is there any howto on that?
>
> - 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
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From dmitry.bely at gmail.com  Wed Feb  9 13:33:50 2011
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Wed, 9 Feb 2011 13:33:50 +0300
Subject: [Freeswitch-users] Skype conference call
In-Reply-To: <AANLkTimCFJ4YszYVrWa_7dHq7QTYRtvj4RsrhiZPzLGu@mail.gmail.com>
References: <AANLkTimXtgR67ZyXXv+YhjtA3dxEsMVEUe5XZUZ6mDD5@mail.gmail.com>
	<AANLkTimCFJ4YszYVrWa_7dHq7QTYRtvj4RsrhiZPzLGu@mail.gmail.com>
Message-ID: <AANLkTin5NvvJOGSqc3EujOmNNeA_cvNW80_L2fhnvZCy@mail.gmail.com>

On Wed, Feb 9, 2011 at 1:20 PM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> I'm working on mod_skypopen.

Yes, I know that. Thanks a lot for your work!

> Can you describe more on the use case?
> what you would like to do exactly?

Well, several Skype users organize a group call as described in

https://support.skype.com/en/faq/FA2831/How-do-I-start-a-conference-call

I would like to participate via FreeSWITCH/Skypopen. Not possible yet?

> I'll take this one as a feature request, and I'm willing to evaluate
> its feasability, just need more info on what exactly is the expected
> feature.

Thanks in advance,

- Dmitry Bely


From gmaruzz at gmail.com  Wed Feb  9 13:37:56 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 9 Feb 2011 11:37:56 +0100
Subject: [Freeswitch-users] Skype conference call
In-Reply-To: <AANLkTin5NvvJOGSqc3EujOmNNeA_cvNW80_L2fhnvZCy@mail.gmail.com>
References: <AANLkTimXtgR67ZyXXv+YhjtA3dxEsMVEUe5XZUZ6mDD5@mail.gmail.com>
	<AANLkTimCFJ4YszYVrWa_7dHq7QTYRtvj4RsrhiZPzLGu@mail.gmail.com>
	<AANLkTin5NvvJOGSqc3EujOmNNeA_cvNW80_L2fhnvZCy@mail.gmail.com>
Message-ID: <AANLkTi=cjQbnpaj8c=1mbQOCCBqWuK=+HUWEWONh3NbC@mail.gmail.com>

how do you participate to a conference from a normal skype client on
your desktop?
there are any special commands to use from that desktop client?
or you just call a skypename? Please give me all the info you deem useful


On Wed, Feb 9, 2011 at 11:33 AM, Dmitry Bely <dmitry.bely at gmail.com> wrote:
> On Wed, Feb 9, 2011 at 1:20 PM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
>> I'm working on mod_skypopen.
>
> Yes, I know that. Thanks a lot for your work!
>
>> Can you describe more on the use case?
>> what you would like to do exactly?
>
> Well, several Skype users organize a group call as described in
>
> https://support.skype.com/en/faq/FA2831/How-do-I-start-a-conference-call
>
> I would like to participate via FreeSWITCH/Skypopen. Not possible yet?
>
>> I'll take this one as a feature request, and I'm willing to evaluate
>> its feasability, just need more info on what exactly is the expected
>> feature.
>
> Thanks in advance,
>
> - 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
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From dmitry.bely at gmail.com  Wed Feb  9 14:02:15 2011
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Wed, 9 Feb 2011 14:02:15 +0300
Subject: [Freeswitch-users] Skype conference call
In-Reply-To: <AANLkTi=cjQbnpaj8c=1mbQOCCBqWuK=+HUWEWONh3NbC@mail.gmail.com>
References: <AANLkTimXtgR67ZyXXv+YhjtA3dxEsMVEUe5XZUZ6mDD5@mail.gmail.com>
	<AANLkTimCFJ4YszYVrWa_7dHq7QTYRtvj4RsrhiZPzLGu@mail.gmail.com>
	<AANLkTin5NvvJOGSqc3EujOmNNeA_cvNW80_L2fhnvZCy@mail.gmail.com>
	<AANLkTi=cjQbnpaj8c=1mbQOCCBqWuK=+HUWEWONh3NbC@mail.gmail.com>
Message-ID: <AANLkTi=Jad3+n+NAKR2W96DCNnCWJEJSF_a9KAP1i03B@mail.gmail.com>

On Wed, Feb 9, 2011 at 1:37 PM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> how do you participate to a conference from a normal skype client on
> your desktop?
> there are any special commands to use from that desktop client?

Please see https://support.skype.com/en/faq/FA2831/How-do-I-start-a-conference-call
I could not describe it better.

> or you just call a skypename? Please give me all the info you deem useful

Just realized that it can be done without additional efforts:

====
To add other people to a call you are already on:
   1. In the conversation window, click the Add People button.
   2. Select the people you wish to add to the call (Note: Ctrl-click
to select multiple people)
   3. Click the Add to Call button. Newly added people will be called.
====

So they just can add me from their desktop client and it results in an
incoming Skype call. I will try that scenario.

I still cannot create a group myself but I can live without it.

- Dmitry Bely


From gmaruzz at gmail.com  Wed Feb  9 14:10:52 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 9 Feb 2011 12:10:52 +0100
Subject: [Freeswitch-users] Skype conference call
In-Reply-To: <AANLkTi=Jad3+n+NAKR2W96DCNnCWJEJSF_a9KAP1i03B@mail.gmail.com>
References: <AANLkTimXtgR67ZyXXv+YhjtA3dxEsMVEUe5XZUZ6mDD5@mail.gmail.com>
	<AANLkTimCFJ4YszYVrWa_7dHq7QTYRtvj4RsrhiZPzLGu@mail.gmail.com>
	<AANLkTin5NvvJOGSqc3EujOmNNeA_cvNW80_L2fhnvZCy@mail.gmail.com>
	<AANLkTi=cjQbnpaj8c=1mbQOCCBqWuK=+HUWEWONh3NbC@mail.gmail.com>
	<AANLkTi=Jad3+n+NAKR2W96DCNnCWJEJSF_a9KAP1i03B@mail.gmail.com>
Message-ID: <AANLkTin_x==imFSY+r_dgQQsMRMoQQYjTNoTWp=QsuaR@mail.gmail.com>

so, you have to be called from a conference, right? (that was my
believe, that I create a conference from my desktop, then I add you,
you receive a call via mod_skypopen and you answer that call. This is
working right now).

To create conferences from mod_skypopen is not possible at the moment.

If this (conference creation from mod_skypopen) is a feature that is
seen as really useful, I can look into that.

-giovanni

On Wed, Feb 9, 2011 at 12:02 PM, Dmitry Bely <dmitry.bely at gmail.com> wrote:
> On Wed, Feb 9, 2011 at 1:37 PM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
>> how do you participate to a conference from a normal skype client on
>> your desktop?
>> there are any special commands to use from that desktop client?
>
> Please see https://support.skype.com/en/faq/FA2831/How-do-I-start-a-conference-call
> I could not describe it better.
>
>> or you just call a skypename? Please give me all the info you deem useful
>
> Just realized that it can be done without additional efforts:
>
> ====
> To add other people to a call you are already on:
> ? 1. In the conversation window, click the Add People button.
> ? 2. Select the people you wish to add to the call (Note: Ctrl-click
> to select multiple people)
> ? 3. Click the Add to Call button. Newly added people will be called.
> ====
>
> So they just can add me from their desktop client and it results in an
> incoming Skype call. I will try that scenario.
>
> I still cannot create a group myself but I can live without it.
>
> - 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
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From admin at blindi.net  Wed Feb  9 14:22:08 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Wed, 9 Feb 2011 12:22:08 +0100 (CET)
Subject: [Freeswitch-users] using skype on a text only ubuntusystem?
In-Reply-To: <AANLkTimm3GTpER_+MxHYBmBzJW5j-jXSP3r3xSxe0GGU@mail.gmail.com>
References: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>
	<AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>
	<59C40C2C-351C-4544-8B50-099DACD6A292@ipeva.fr>
	<Pine.LNX.4.64.1102082229480.17347@suse.blindi.net>
	<AANLkTi=hJVcNEDVUpCdXpEZJjYSBq0z6p2QGBRR0Jc73@mail.gmail.com>
	<Pine.LNX.4.64.1102082259120.17347@suse.blindi.net>
	<AANLkTi=ioaNQ2GgsmFQUVtxFpkknh6mdQNQNELGemsG0@mail.gmail.com>
	<Pine.LNX.4.64.1102090757040.24802@suse.blindi.net>
	<AANLkTimm3GTpER_+MxHYBmBzJW5j-jXSP3r3xSxe0GGU@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1102091213380.27831@suse.blindi.net>

Hi Giovanni,
thankx for you replay.
I using ubuntu lucid. The alsa-driver make a problem.
Alsa is compiled correcly from sources.
when i execute the following command:
root at freeswitch:/# modprobe  snd_dummy
WARNING: Error inserting snd_page_alloc 
(/lib/modules/2.6.32-27-generic-pae/kern
el/sound/acore/snd-page-alloc.ko): Unknown symbol in module, or unknown 
paramete
r (see dmesg)
WARNING: Error inserting snd_timer 
(/lib/modules/2.6.32-27-generic-pae/kernel/so
und/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter 
(see dme
sg)
WARNING: Error inserting snd_pcm 
(/lib/modules/2.6.32-27-generic-pae/kernel/soun
d/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see 
dmesg)
FATAL: Error inserting snd_dummy 
(/lib/modules/2.6.32-27-generic-pae/kernel/soun
d/drivers/snd-dummy.ko): Unknown symbol in module, or unknown parameter 
(see dme
sg)

i make: depmod -a
i reboot the system, the problem is the same.
can you looking please?
tahkx.


---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From yky1628 at yahoo.com  Wed Feb  9 13:38:50 2011
From: yky1628 at yahoo.com (Frankie Yiu)
Date: Wed, 9 Feb 2011 02:38:50 -0800 (PST)
Subject: [Freeswitch-users] How to read RTP package?
Message-ID: <807770.34642.qm@web30502.mail.mud.yahoo.com>

Hi there,

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

If I want to read the?RTP package in C++ or C#,?what function?should?I call to 
get it???Currently I have?can make a call?to a?phone and play an audio.

Thanks?in advance.
?Frankie 


 
____________________________________________________________________________________
Expecting? Get great news right away with email Auto-Check. 
Try the Yahoo! Mail Beta.
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From gmaruzz at gmail.com  Wed Feb  9 14:35:40 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 9 Feb 2011 12:35:40 +0100
Subject: [Freeswitch-users] using skype on a text only ubuntusystem?
In-Reply-To: <Pine.LNX.4.64.1102091213380.27831@suse.blindi.net>
References: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>
	<AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>
	<59C40C2C-351C-4544-8B50-099DACD6A292@ipeva.fr>
	<Pine.LNX.4.64.1102082229480.17347@suse.blindi.net>
	<AANLkTi=hJVcNEDVUpCdXpEZJjYSBq0z6p2QGBRR0Jc73@mail.gmail.com>
	<Pine.LNX.4.64.1102082259120.17347@suse.blindi.net>
	<AANLkTi=ioaNQ2GgsmFQUVtxFpkknh6mdQNQNELGemsG0@mail.gmail.com>
	<Pine.LNX.4.64.1102090757040.24802@suse.blindi.net>
	<AANLkTimm3GTpER_+MxHYBmBzJW5j-jXSP3r3xSxe0GGU@mail.gmail.com>
	<Pine.LNX.4.64.1102091213380.27831@suse.blindi.net>
Message-ID: <AANLkTiku2yazZoR9UkL3ZWzf0uGr2xwj+Uok-D-12K+y@mail.gmail.com>

please do not use a thread for another topic.

It will be less useful to other readers.

I'll answer opening a new topic

-giovanni

On Wed, Feb 9, 2011 at 12:22 PM, Thomas Hoellriegel <admin at blindi.net> wrote:
> Hi Giovanni,
> thankx for you replay.
> I using ubuntu lucid. The alsa-driver make a problem.
> Alsa is compiled correcly from sources.
> when i execute the following command:
> root at freeswitch:/# modprobe ?snd_dummy
> WARNING: Error inserting snd_page_alloc
> (/lib/modules/2.6.32-27-generic-pae/kern
> el/sound/acore/snd-page-alloc.ko): Unknown symbol in module, or unknown
> paramete
> r (see dmesg)
> WARNING: Error inserting snd_timer
> (/lib/modules/2.6.32-27-generic-pae/kernel/so
> und/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see
> dme
> sg)
> WARNING: Error inserting snd_pcm
> (/lib/modules/2.6.32-27-generic-pae/kernel/soun
> d/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see
> dmesg)
> FATAL: Error inserting snd_dummy
> (/lib/modules/2.6.32-27-generic-pae/kernel/soun
> d/drivers/snd-dummy.ko): Unknown symbol in module, or unknown parameter (see
> dme
> sg)
>
> i make: depmod -a
> i reboot the system, the problem is the same.
> can you looking please?
> tahkx.
>
>
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/listinfo/blinde-misc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 dujinfang at gmail.com  Wed Feb  9 14:37:12 2011
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 9 Feb 2011 19:37:12 +0800
Subject: [Freeswitch-users] git HEAD 503 Maximum Calls In Progress
Message-ID: <AANLkTimpGj4GLug8jrK-7-V3Mja9W2tTiwtJsBSu-TZo@mail.gmail.com>

Hi,

It's a fresh install with wget  freeswitch.org/eg/Makefile && make,
and I'm sure I successfully called 9196 and everything was ok.

But a few minutes later I started getting

 503 Maximum Calls In Progress

also originate user/1003 or loopback/9196 shows errors

see
http://pastebin.freeswitch.org/15333

It's on a ubuntu 8.04 32bit, the only thing I did between non-work and
work was I configured odbc with postgresql to work with isql. I
haven't touch the FS config yet. I don't think they are related.


here is also a nua debug

http://pastebin.freeswitch.org/15335

Any hint on this? Thanks.

btw, I also noticed there might be some network problems between me
and the server where sometime sip packets not shown ( sofia debug and
ngrep) on server. Anyway, since loopback cannot create sessions so
something must be wrong in FS side, but what's that?

Thanks.

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


From gmaruzz at gmail.com  Wed Feb  9 14:43:34 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 9 Feb 2011 12:43:34 +0100
Subject: [Freeswitch-users] skypopen: problem loading the custom alsa driver
Message-ID: <AANLkTin=oJpzotF4x4+eYKKrazUaKxoKOdx9yAWA7ewD@mail.gmail.com>

Thomas wrote:
=================
Hi Giovanni,
thankx for you replay.
I using ubuntu lucid. The alsa-driver make a problem.
Alsa is compiled correcly from sources.
when i execute the following command:
root at freeswitch:/# modprobe  snd_dummy
WARNING: Error inserting snd_page_alloc (/lib/modules/2.6.32-27-generic-pae/kern
el/sound/acore/snd-page-alloc.ko): Unknown symbol in module, or unknown paramete
r (see dmesg)
WARNING: Error inserting snd_timer (/lib/modules/2.6.32-27-generic-pae/kernel/so
und/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dme
sg)
WARNING: Error inserting snd_pcm (/lib/modules/2.6.32-27-generic-pae/kernel/soun
d/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg)
FATAL: Error inserting snd_dummy (/lib/modules/2.6.32-27-generic-pae/kernel/soun
d/drivers/snd-dummy.ko): Unknown symbol in module, or unknown parameter (see dme
sg)

i make: depmod -a
i reboot the system, the problem is the same.
can you looking please?
tahkx.
=====================

My answer:

have you followed strictly the guidelines in the wiki page for
compiling and loading the custom snd-dummy? (alsa and drivers are
tricky beasts)

can you paste here the results in dmesg (/var/log/messages) that are
related to that?

anyway, seems that you have one of this problems:
a) maybe you have not done "make" then (after make) "make install"
b) maybe you still have the old sound directory in your /var/lib/modules tree
c) maybe you have compiled and installed the alsa mods for a previous
kernel version

can you report exactly what you have done (and answer the previous questions) ?

-giovanni
-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From steveayre at gmail.com  Wed Feb  9 14:52:00 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 9 Feb 2011 11:52:00 +0000
Subject: [Freeswitch-users] git HEAD 503 Maximum Calls In Progress
In-Reply-To: <AANLkTimpGj4GLug8jrK-7-V3Mja9W2tTiwtJsBSu-TZo@mail.gmail.com>
References: <AANLkTimpGj4GLug8jrK-7-V3Mja9W2tTiwtJsBSu-TZo@mail.gmail.com>
Message-ID: <AANLkTi=2t_gPtzyqHYaNuZz0VUNfOwDkWNmJHJx1YdeQ@mail.gmail.com>

What does show channels show?


On 9 February 2011 11:37, Seven Du <dujinfang at gmail.com> wrote:

> Hi,
>
> It's a fresh install with wget  freeswitch.org/eg/Makefile && make,
> and I'm sure I successfully called 9196 and everything was ok.
>
> But a few minutes later I started getting
>
>  503 Maximum Calls In Progress
>
> also originate user/1003 or loopback/9196 shows errors
>
> see
> http://pastebin.freeswitch.org/15333
>
> It's on a ubuntu 8.04 32bit, the only thing I did between non-work and
> work was I configured odbc with postgresql to work with isql. I
> haven't touch the FS config yet. I don't think they are related.
>
>
> here is also a nua debug
>
> http://pastebin.freeswitch.org/15335
>
> Any hint on this? Thanks.
>
> btw, I also noticed there might be some network problems between me
> and the server where sometime sip packets not shown ( sofia debug and
> ngrep) on server. Anyway, since loopback cannot create sessions so
> something must be wrong in FS side, but what's that?
>
> Thanks.
>
> --
> 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
>
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From david.ponzone at ipeva.fr  Wed Feb  9 15:23:56 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 9 Feb 2011 13:23:56 +0100
Subject: [Freeswitch-users] How to read RTP package?
In-Reply-To: <807770.34642.qm@web30502.mail.mud.yahoo.com>
References: <807770.34642.qm@web30502.mail.mud.yahoo.com>
Message-ID: <E6C5B092-52B1-40F2-B327-ADEE95C90906@ipeva.fr>

Frankie,

Explain your project, that will help people to help you.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 09/02/2011 ? 11:38, Frankie Yiu a ?crit :

> Hi there,
>  
> I am new to FreeSwitch, and I have a question to ask.
>  
> If I want to read the RTP package in C++ or C#, what function should I call to get it?  Currently I have can make a call to a phone and play an audio.
> 
> Thanks in advance.
>  
> Frankie
> 
> Finding fabulous fares is fun.
> Let Yahoo! FareChase search your favorite travel sites to find flight and hotel bargains._______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From admin at blindi.net  Wed Feb  9 15:45:24 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Wed, 9 Feb 2011 13:45:24 +0100 (CET)
Subject: [Freeswitch-users] skypopen: problem loading the custom alsa
 driver
In-Reply-To: <AANLkTin=oJpzotF4x4+eYKKrazUaKxoKOdx9yAWA7ewD@mail.gmail.com>
References: <AANLkTin=oJpzotF4x4+eYKKrazUaKxoKOdx9yAWA7ewD@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1102091322460.28387@suse.blindi.net>

> can you paste here the results in dmesg (/var/log/messages) that are
> related to that?
Yes, mdesg:

[   69.929738] snd: Unknown symbol unregister_sound_special
[   69.930392] snd: Unknown symbol register_sound_special_device
[   69.932477] snd: Unknown symbol sound_class

in /var/log/messages

Feb  9 13:20:51 freeswitch kernel: [   69.929738] snd: Unknown symbol
unregister_sound_special
Feb  9 13:20:51 freeswitch kernel: [   69.930392] snd: Unknown symbol
register_sound_special_device
Feb  9 13:20:51 freeswitch kernel: [   69.932477] snd: Unknown symbol
sound_class

> anyway, seems that you have one of this problems:
> a) maybe you have not done "make" then (after make) "make install"

make install has no errors.

> b) maybe you still have the old sound directory in your /var/lib/modules tree

root at freeswitch:/usr/src/alsa-driver-1.0.20# ls  /var/lib/module*
ls: Zugriff auf /var/lib/module* : No such file or directory

> c) maybe you have compiled and installed the alsa mods for a previous
> kernel version

I ehter these commands:
apt-get  update
apt-get  -y upgrade
dpkg -l |grep alsa
"no alsa found"
  apt-get -y install git-core subversion build-essential \
   autoconf automake libtool libncurses5 libncurses5-dev
apt-get -y install xvfb libx11-dev xfs xfonts-100dpi xfonts-75dpi \
xfonts-scalable
apt-get  -y install unixodbc-dev
apt-get  -y install libtiff4-dev
apt-get  -y install libogg-dev libvorbis-dev
cd /usr/src
wget ftp://ftp.alsa-project.org/pub/driver/alsa-driver-1.0.20.tar.bz2
tar -xjf alsa-driver-1.0.20.tar.bz2
cd alsa-driver-1.0.20
rm -r /lib/modules/`uname -r`/kernel/sound/
./configure --with-redhat=no \
--with-cards=dummy,usb-audio,hda-intel,hrtimer,rtctimer \
--with-card-options=all
make && make install
cp /usr/src/freeswitch.git/src/mod/endpoints/mod_skypopen/alsa/dummy.c \
drivers/dummy.c
make && make install

Alsa is compiled and install corretly.
I can.t load the dummydriver module.

---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From victor.chukalovskiy at utoronto.ca  Wed Feb  9 15:40:38 2011
From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy)
Date: Wed, 09 Feb 2011 07:40:38 -0500
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <ED3B5594-990A-4ACC-8206-856650E56F38@ipeva.fr>
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>
Message-ID: <4D528B46.7010709@utoronto.ca>

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From david.ponzone at ipeva.fr  Wed Feb  9 16:01:26 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 9 Feb 2011 14:01:26 +0100
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <4D528B46.7010709@utoronto.ca>
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>
Message-ID: <9CF0F421-1F14-4718-824F-33A6E1BF85A7@ipeva.fr>

The license server is normally launched automatically.
If you can, restart FreeSWITCH.
It's normally not required.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 09/02/2011 ? 13:40, Victor Chukalovskiy a ?crit :

> Hi David,
> 
> I'm aware of the file and was using it for install.
> 
> There is no single word there what "licence server" is 
> and how to deal with "can't contact licence server".
> 
> Thank you,
> Victor
> 
> On 09/02/11 03:55 AM, David Ponzone wrote:
>> 
>> There is a file coming with the module which explains the installation quite perfectly.
>> 
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> 
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> 
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> 
>> 
>> 
>> 
>> Le 09/02/2011 ? 05:41, Victor Chukalovskiy a ?crit :
>> 
>>> This didn't help.
>>> I reloaded mod_com_g729, tried g729_info and the same "can't contact licence server" error
>>> -Victor
>>> 
>>> On 08/02/11 11:10 PM, Michael Collins wrote:
>>>> 
>>>> put the license file in /etc/freeswitch
>>>> -MC
>>>> 
>>>> On Tue, Feb 8, 2011 at 9:56 PM, Victor Chukalovskiy <victor.chukalovskiy at utoronto.ca> wrote:
>>>> Hi Michael and others,
>>>> 
>>>> After updating to the latest GIT mod_com_g729 loads successfully.
>>>> 
>>>> Next problem appears: 
>>>> when typing "g729_info" freeswitch replies "can't contact licence server".
>>>> g729_available gives "False"
>>>> 
>>>> How to solve this?
>>>> 
>>>> I have my xxxxx.conf license file in the root of FS install directory /opt/fs
>>>> Should it be placed elsewhere?
>>>> 
>>>> Also, xxxxx.conf was created with previous non-successful install of mod_com_g729
>>>> Should I run validator again with the same license key?
>>>> 
>>>> Thank you,
>>>> Victor
>>>> 
>>>> On 07/02/11 10:36 AM, Michael Collins wrote:
>>>>> 
>>>>> Any chance you can update to latest git? Your life will be easier. There have been notable improvements in FS in the past few months. 
>>>>> 
>>>>> -MC
>>>>> 
>>>>> On Sat, Feb 5, 2011 at 3:30 PM, Victor Chukalovskiy <victor.chukalovskiy at utoronto.ca> wrote:
>>>>> Hello,
>>>>> 
>>>>> After purchasing a few licenses and installing the latest fsg729-191-installer
>>>>> I'm getting the following error when trying to load the mod_com_g729:
>>>>> > 2011-02-04 14:27:17.637046 [CRIT] switch_loadable_module.c:926 Error Loading module /opt/fs/mod/mod_com_g729.so
>>>>> > **Trying to load an out of date module, please rebuild the module.**
>>>>> 
>>>>> Also noticed that g729 installer ran with a couple errors:
>>>>> > ./installer: line 62: ldconfig: command not found
>>>>> > ./installer: line 49: useradd: command not found
>>>>>  Any help or solution is much appreciated.
>>>>> 
>>>>> -Victor 
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From thomas at chaschperli.ch  Wed Feb  9 16:03:31 2011
From: thomas at chaschperli.ch (Thomas Mueller)
Date: Wed, 09 Feb 2011 14:03:31 +0100
Subject: [Freeswitch-users] skypopen: problem loading the custom alsa
 driver
In-Reply-To: <AANLkTin=oJpzotF4x4+eYKKrazUaKxoKOdx9yAWA7ewD@mail.gmail.com>
References: <AANLkTin=oJpzotF4x4+eYKKrazUaKxoKOdx9yAWA7ewD@mail.gmail.com>
Message-ID: <4D5290A3.8030208@chaschperli.ch>


> root at freeswitch:/# modprobe  snd_dummy
> WARNING: Error inserting snd_page_alloc (/lib/modules/2.6.32-27-generic-pae/kern
> el/sound/acore/snd-page-alloc.ko): Unknown symbol in module, or unknown paramete
> r (see dmesg)
> WARNING: Error inserting snd_timer (/lib/modules/2.6.32-27-generic-pae/kernel/so
> und/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dme
> sg)
> WARNING: Error inserting snd_pcm (/lib/modules/2.6.32-27-generic-pae/kernel/soun
> d/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg)
> FATAL: Error inserting snd_dummy (/lib/modules/2.6.32-27-generic-pae/kernel/soun
> d/drivers/snd-dummy.ko): Unknown symbol in module, or unknown parameter (see dme
> sg)

for me this error sounds like the module was not built against the 
correct kernel sources.

- Thomas


From steveu at coppice.org  Wed Feb  9 16:23:37 2011
From: steveu at coppice.org (Steve Underwood)
Date: Wed, 09 Feb 2011 21:23:37 +0800
Subject: [Freeswitch-users] G729
In-Reply-To: <AANLkTimVC094R9P4sSurT2XSq_kiLUEpqvAFWjHwVPpL@mail.gmail.com>
References: <AANLkTimEvZiuonWuWevNk2DWknG2uOvJR=AcNdujr3Wj@mail.gmail.com>	<AANLkTim62wJrp1cByF45Lmk4wzJSANBCfeE26Sm+HoZY@mail.gmail.com>	<AANLkTinTY5Z+UGfVTRb_8qM-Jb1WbFDdBjreg2PKGB2W@mail.gmail.com>
	<AANLkTimVC094R9P4sSurT2XSq_kiLUEpqvAFWjHwVPpL@mail.gmail.com>
Message-ID: <4D529559.5030001@coppice.org>

On 02/09/2011 12:36 PM, curriegrad2004 wrote:
> Ah, so any devices that use a SDP of G729A is considered to be broken
> in implementation, right?
>
> Most of the cisco-linksys SPA adapters actually default to a SDP of
> G729A for some reason.
Depends how you view the term broken. They don't comply with the 
standards, that's for sure. A lot of older Cisco kit is far more broken, 
as they have the G.729 bits packed in a different order from the rest of 
humanity.

Most recent versions of Cisco/Linksys software have corrected these 
issues. They allow you to use the wrong packing or the SDP tag "g729a" 
for backwards compatibility with their own kit, and standards compliant 
SDP and/or RTP for compatibility with other kit.

Steve



From victor.chukalovskiy at utoronto.ca  Wed Feb  9 16:25:20 2011
From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy)
Date: Wed, 09 Feb 2011 08:25:20 -0500
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <9CF0F421-1F14-4718-824F-33A6E1BF85A7@ipeva.fr>
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>
Message-ID: <4D5295C0.3080807@utoronto.ca>

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From gmaruzz at gmail.com  Wed Feb  9 16:52:20 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 9 Feb 2011 14:52:20 +0100
Subject: [Freeswitch-users] skypopen: problem loading the custom alsa
	driver
In-Reply-To: <Pine.LNX.4.64.1102091322460.28387@suse.blindi.net>
References: <AANLkTin=oJpzotF4x4+eYKKrazUaKxoKOdx9yAWA7ewD@mail.gmail.com>
	<Pine.LNX.4.64.1102091322460.28387@suse.blindi.net>
Message-ID: <AANLkTi=p5U4Q7tGaz4=pErzhi8esy+gZpJ7U_NL4SESY@mail.gmail.com>

Are you using a normal ubuntu installation? I mean, the normal stock
kernel that Ubuntu gives you?

Do you have done:
apt-get install linux-headers-$(uname -r)
before to do the alsa things?

If not, please delete all the alsa sources directory, decompress the
package again, and start from beginning.



On Wed, Feb 9, 2011 at 1:45 PM, Thomas Hoellriegel <admin at blindi.net> wrote:
>> can you paste here the results in dmesg (/var/log/messages) that are
>> related to that?
>
> Yes, mdesg:
>
> [ ? 69.929738] snd: Unknown symbol unregister_sound_special
> [ ? 69.930392] snd: Unknown symbol register_sound_special_device
> [ ? 69.932477] snd: Unknown symbol sound_class
>
> in /var/log/messages
>
> Feb ?9 13:20:51 freeswitch kernel: [ ? 69.929738] snd: Unknown symbol
> unregister_sound_special
> Feb ?9 13:20:51 freeswitch kernel: [ ? 69.930392] snd: Unknown symbol
> register_sound_special_device
> Feb ?9 13:20:51 freeswitch kernel: [ ? 69.932477] snd: Unknown symbol
> sound_class
>
>> anyway, seems that you have one of this problems:
>> a) maybe you have not done "make" then (after make) "make install"
>
> make install has no errors.
>
>> b) maybe you still have the old sound directory in your /var/lib/modules
>> tree
>
> root at freeswitch:/usr/src/alsa-driver-1.0.20# ls ?/var/lib/module*
> ls: Zugriff auf /var/lib/module* : No such file or directory
>
>> c) maybe you have compiled and installed the alsa mods for a previous
>> kernel version
>
> I ehter these commands:
> apt-get ?update
> apt-get ?-y upgrade
> dpkg -l |grep alsa
> "no alsa found"
> ?apt-get -y install git-core subversion build-essential \
> ?autoconf automake libtool libncurses5 libncurses5-dev
> apt-get -y install xvfb libx11-dev xfs xfonts-100dpi xfonts-75dpi \
> xfonts-scalable
> apt-get ?-y install unixodbc-dev
> apt-get ?-y install libtiff4-dev
> apt-get ?-y install libogg-dev libvorbis-dev
> cd /usr/src
> wget ftp://ftp.alsa-project.org/pub/driver/alsa-driver-1.0.20.tar.bz2
> tar -xjf alsa-driver-1.0.20.tar.bz2
> cd alsa-driver-1.0.20
> rm -r /lib/modules/`uname -r`/kernel/sound/
> ./configure --with-redhat=no \
> --with-cards=dummy,usb-audio,hda-intel,hrtimer,rtctimer \
> --with-card-options=all
> make && make install
> cp /usr/src/freeswitch.git/src/mod/endpoints/mod_skypopen/alsa/dummy.c \
> drivers/dummy.c
> make && make install
>
> Alsa is compiled and install corretly.
> I can.t load the dummydriver module.
>
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/listinfo/blinde-misc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 steveayre at gmail.com  Wed Feb  9 17:01:03 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 9 Feb 2011 14:01:03 +0000
Subject: [Freeswitch-users] How to read RTP package?
In-Reply-To: <807770.34642.qm@web30502.mail.mud.yahoo.com>
References: <807770.34642.qm@web30502.mail.mud.yahoo.com>
Message-ID: <AANLkTin+nNWmTb+wGz9_SoaiDrZYnc_JK_iTVcGOxQBu@mail.gmail.com>

Frankie, I have already responded to your 1st message.

Don't send to the list more than once... someone will reply to your post
even if they don't immediately. If you're new to the list, your 1st post was
probably moderated so didn't show up straight away.

-Steve


On 9 February 2011 10:38, Frankie Yiu <yky1628 at yahoo.com> wrote:

>  Hi there,
>
> I am new to FreeSwitch, and I have a question to ask.
>
> If I want to read the RTP package in C++ or C#, what function should I call
> to get it?  Currently I have can make a call to a phone and play an audio.
>
> Thanks in advance.
>
> Frankie
>
> ------------------------------
> Finding fabulous fares is fun.
> Let Yahoo! FareChase search your favorite travel sites<http://farechase.yahoo.com/promo-generic-14795097;_ylc=X3oDMTFtNW45amVpBF9TAzk3NDA3NTg5BF9zAzI3MTk0ODEEcG9zAzEEc2VjA21haWx0YWdsaW5lBHNsawNxMS0wNw-->to find flight and hotel bargains.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  9 17:02:23 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 9 Feb 2011 14:02:23 +0000
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <4D5295C0.3080807@utoronto.ca>
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>
Message-ID: <AANLkTik_82ZycrpjBmiJSQCDUgScnoJE0WCiH5XwW3rk@mail.gmail.com>

What user do you run FS as?

It's anothe program in the same bin folder as freeswitch, and should run as
the same user as freeswitch itself. There's no logging I'm aware of.

-Steve


On 9 February 2011 13:25, Victor Chukalovskiy <
victor.chukalovskiy at utoronto.ca> wrote:

>  Thanks David,
>
> Freeswitch was restarted.
> Still "can't contact licence server"
>
> Is this license server a process / module running on the same machine?
> If so, does it log anything anywhere? Where is it started from?
>
> -Victor
>
>
> On 09/02/11 08:01 AM, David Ponzone wrote:
>
> The license server is normally launched automatically.
> If you can, restart FreeSWITCH.
> It's normally not required.
>
>   David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
>  Service Client IPeva
>  tel:      0811 46 26 26
>  www.ipeva.fr  -   www.ipeva-studio.com
>
>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
>  Le 09/02/2011 ? 13:40, Victor Chukalovskiy a ?crit :
>
>  Hi David,
>
> I'm aware of the file and was using it for install.
>
> There is no single word there what "licence server" is
> and how to deal with "can't contact licence server".
>
> Thank you,
> Victor
>
> On 09/02/11 03:55 AM, David Ponzone wrote:
>
> There is a file coming with the module which explains the installation
> quite perfectly.
>
>   David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
>  Service Client IPeva
>  tel:      0811 46 26 26
>  www.ipeva.fr  -   www.ipeva-studio.com
>
>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
>  Le 09/02/2011 ? 05:41, Victor Chukalovskiy a ?crit :
>
>  This didn't help.
> I reloaded mod_com_g729, tried g729_info and the same "can't contact
> licence server" error
> -Victor
>
> On 08/02/11 11:10 PM, Michael Collins wrote:
>
> put the license file in /etc/freeswitch
> -MC
>
> On Tue, Feb 8, 2011 at 9:56 PM, Victor Chukalovskiy <
> victor.chukalovskiy at utoronto.ca> wrote:
>
>>  Hi Michael and others,
>>
>> After updating to the latest GIT mod_com_g729 loads successfully.
>>
>> Next problem appears:
>> when typing "g729_info" freeswitch replies "can't contact licence server".
>> g729_available gives "False"
>>
>> How to solve this?
>>
>> I have my xxxxx.conf license file in the root of FS install directory
>> /opt/fs
>> Should it be placed elsewhere?
>>
>> Also, xxxxx.conf was created with previous non-successful install of
>> mod_com_g729
>> Should I run validator again with the same license key?
>>
>> Thank you,
>> Victor
>>
>> On 07/02/11 10:36 AM, Michael Collins wrote:
>>
>> Any chance you can update to latest git? Your life will be easier. There
>> have been notable improvements in FS in the past few months.
>>
>>  -MC
>>
>> On Sat, Feb 5, 2011 at 3:30 PM, Victor Chukalovskiy <
>> victor.chukalovskiy at utoronto.ca> wrote:
>>
>>>  Hello,
>>>
>>> After purchasing a few licenses and installing the latest
>>> fsg729-191-installer
>>> I'm getting the following error when trying to load the mod_com_g729:
>>> > 2011-02-04 14:27:17.637046 [CRIT] switch_loadable_module.c:926 Error
>>> Loading module /opt/fs/mod/mod_com_g729.so
>>> > **Trying to load an out of date module, please rebuild the module.**
>>>
>>> Also noticed that g729 installer ran with a couple errors:
>>> > ./installer: line 62: ldconfig: command not found
>>> > ./installer: line 49: useradd: command not found
>>>  Any help or solution is much appreciated.
>>>
>>> -Victor
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> 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
>
>
>
> _______________________________________________
> 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 admin at blindi.net  Wed Feb  9 17:13:10 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Wed, 9 Feb 2011 15:13:10 +0100 (CET)
Subject: [Freeswitch-users] skypopen: problem loading the custom alsa
 driver
In-Reply-To: <AANLkTi=p5U4Q7tGaz4=pErzhi8esy+gZpJ7U_NL4SESY@mail.gmail.com>
References: <AANLkTin=oJpzotF4x4+eYKKrazUaKxoKOdx9yAWA7ewD@mail.gmail.com>
	<Pine.LNX.4.64.1102091322460.28387@suse.blindi.net>
	<AANLkTi=p5U4Q7tGaz4=pErzhi8esy+gZpJ7U_NL4SESY@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1102091458490.29294@suse.blindi.net>

Am 09.02.11 um 14:52 schrieb Giovanni Maruzzelli:

> Are you using a normal ubuntu installation? I mean, the normal stock
> kernel that Ubuntu gives you?
I have installed  a minimalsystem from debootstrap.
ssh is optional installed.

Yes the standradkernel from ubuntu is installed.
root at freeswitch:~# uname -a
Linux freeswitch 2.6.32-27-generic-pae #49-Ubuntu SMP Thu Dec 2 00:07:52 
UTC 201
0 i686 GNU/Linux

> Do you have done:
> apt-get install linux-headers-$(uname -r)
yes:
the output:
root at freeswitch:~# dpkg -l  |grep -i linux-headers
ii  linux-headers-2.6.32-27             2.6.32-27.49                Header 
files
  related to Linux kernel version
ii  linux-headers-2.6.32-27-generic-pae 2.6.32-27.49                Linux 
kernel
  headers for version 2.6.32 
---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From victor.chukalovskiy at utoronto.ca  Wed Feb  9 17:12:08 2011
From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy)
Date: Wed, 09 Feb 2011 09:12:08 -0500
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <AANLkTik_82ZycrpjBmiJSQCDUgScnoJE0WCiH5XwW3rk@mail.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>
Message-ID: <4D52A0B8.6090005@utoronto.ca>

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From Nabble at slickdeals.endjunk.com  Wed Feb  9 17:22:43 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 9 Feb 2011 06:22:43 -0800 (PST)
Subject: [Freeswitch-users] ext-rtp-ip value behind NAT
In-Reply-To: <AANLkTi==qUfXH9q2x=is=xF+PCxoxFeM-7pJn3+2ZLg9@mail.gmail.com>
References: <1296962373969-5996978.post@n2.nabble.com>
	<AANLkTi==qUfXH9q2x=is=xF+PCxoxFeM-7pJn3+2ZLg9@mail.gmail.com>
Message-ID: <1297261363014-6007811.post@n2.nabble.com>



mazilo wrote:
> 1. 
> 2. 
The above two options show ONLY '<'param name="ext-rtp-ip"
value="auto-nat"/'>' and '<'param name="ext-rtp-ip"
value="$${external_rtp_ip}"/'>'? The apostrophes are needed to enclose the
'<' and '>'.

-----
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/ext-rtp-ip-value-behind-NAT-tp5996978p6007811.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Wed Feb  9 17:24:41 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 9 Feb 2011 14:24:41 +0000
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <4D52A0B8.6090005@utoronto.ca>
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>
Message-ID: <AANLkTimPj_27abwq4RNvpkfczkUXi5hf7-ro5v=yEc6E@mail.gmail.com>

I see you're in a nonstandard directory.

As far as I know the license server only likes being in
/usr/local/freeswitch/bin or /opt/freeswitch/bin.

Because it's neither of those it looks like the installer hasn't installed
the license server.

-Steve


On 9 February 2011 14:12, Victor Chukalovskiy <
victor.chukalovskiy at utoronto.ca> wrote:

>  Hi Steve,
>
> I'm running as user "fs":
> /opt/fs/bin/freeswitch -nf -nc -u fs -g fs -core -nonat
>
> I have following binaries in the directory:
> freeswitch  fs_cli  fs_encode  fs_ivrd  fsxs  gentls_cert  tone2wav
> validator
> Is any of them the required binary? Where is it launched from?
>
> Thank you,
> Victor
>
>
> On 09/02/11 09:02 AM, Steven Ayre wrote:
>
> What user do you run FS as?
>
> It's anothe program in the same bin folder as freeswitch, and should run as
> the same user as freeswitch itself. There's no logging I'm aware of.
>
> -Steve
>
>
>  On 9 February 2011 13:25, Victor Chukalovskiy <
> victor.chukalovskiy at utoronto.ca> wrote:
>
>>  Thanks David,
>>
>> Freeswitch was restarted.
>> Still "can't contact licence server"
>>
>> Is this license server a process / module running on the same machine?
>> If so, does it log anything anywhere? Where is it started from?
>>
>> -Victor
>>
>>
>> On 09/02/11 08:01 AM, David Ponzone wrote:
>>
>> The license server is normally launched automatically.
>> If you can, restart FreeSWITCH.
>> It's normally not required.
>>
>>   David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>>  Service Client IPeva
>>  tel:      0811 46 26 26
>>  www.ipeva.fr  -   www.ipeva-studio.com
>>
>>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>>  Le 09/02/2011 ? 13:40, Victor Chukalovskiy a ?crit :
>>
>>  Hi David,
>>
>> I'm aware of the file and was using it for install.
>>
>> There is no single word there what "licence server" is
>> and how to deal with "can't contact licence server".
>>
>> Thank you,
>> Victor
>>
>> On 09/02/11 03:55 AM, David Ponzone wrote:
>>
>> There is a file coming with the module which explains the installation
>> quite perfectly.
>>
>>   David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>>  Service Client IPeva
>>  tel:      0811 46 26 26
>>  www.ipeva.fr  -   www.ipeva-studio.com
>>
>>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>>  Le 09/02/2011 ? 05:41, Victor Chukalovskiy a ?crit :
>>
>>  This didn't help.
>> I reloaded mod_com_g729, tried g729_info and the same "can't contact
>> licence server" error
>> -Victor
>>
>> On 08/02/11 11:10 PM, Michael Collins wrote:
>>
>> put the license file in /etc/freeswitch
>> -MC
>>
>> On Tue, Feb 8, 2011 at 9:56 PM, Victor Chukalovskiy <
>> victor.chukalovskiy at utoronto.ca> wrote:
>>
>>>  Hi Michael and others,
>>>
>>> After updating to the latest GIT mod_com_g729 loads successfully.
>>>
>>> Next problem appears:
>>> when typing "g729_info" freeswitch replies "can't contact licence
>>> server".
>>> g729_available gives "False"
>>>
>>> How to solve this?
>>>
>>> I have my xxxxx.conf license file in the root of FS install directory
>>> /opt/fs
>>> Should it be placed elsewhere?
>>>
>>> Also, xxxxx.conf was created with previous non-successful install of
>>> mod_com_g729
>>> Should I run validator again with the same license key?
>>>
>>> Thank you,
>>> Victor
>>>
>>> On 07/02/11 10:36 AM, Michael Collins wrote:
>>>
>>> Any chance you can update to latest git? Your life will be easier. There
>>> have been notable improvements in FS in the past few months.
>>>
>>>  -MC
>>>
>>> On Sat, Feb 5, 2011 at 3:30 PM, Victor Chukalovskiy <
>>> victor.chukalovskiy at utoronto.ca> wrote:
>>>
>>>>  Hello,
>>>>
>>>> After purchasing a few licenses and installing the latest
>>>> fsg729-191-installer
>>>> I'm getting the following error when trying to load the mod_com_g729:
>>>> > 2011-02-04 14:27:17.637046 [CRIT] switch_loadable_module.c:926 Error
>>>> Loading module /opt/fs/mod/mod_com_g729.so
>>>> > **Trying to load an out of date module, please rebuild the module.**
>>>>
>>>> Also noticed that g729 installer ran with a couple errors:
>>>> > ./installer: line 62: ldconfig: command not found
>>>> > ./installer: line 49: useradd: command not found
>>>>  Any help or solution is much appreciated.
>>>>
>>>> -Victor
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> 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
>>
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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  Wed Feb  9 17:25:39 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 9 Feb 2011 14:25:39 +0000
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <AANLkTimPj_27abwq4RNvpkfczkUXi5hf7-ro5v=yEc6E@mail.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>
Message-ID: <AANLkTinW0QMGwuTHOzKXh2BepbOJWZsL=PL5yeM6dXfT@mail.gmail.com>

The license server is distributed with mod_com_g729 not freeswitch itself by
the way - the installer i mean is fsg729-191-installer.

-Steve

On 9 February 2011 14:24, Steven Ayre <steveayre at gmail.com> wrote:

> I see you're in a nonstandard directory.
>
> As far as I know the license server only likes being in
> /usr/local/freeswitch/bin or /opt/freeswitch/bin.
>
> Because it's neither of those it looks like the installer hasn't installed
> the license server.
>
> -Steve
>
>
>
> On 9 February 2011 14:12, Victor Chukalovskiy <
> victor.chukalovskiy at utoronto.ca> wrote:
>
>>  Hi Steve,
>>
>> I'm running as user "fs":
>> /opt/fs/bin/freeswitch -nf -nc -u fs -g fs -core -nonat
>>
>> I have following binaries in the directory:
>> freeswitch  fs_cli  fs_encode  fs_ivrd  fsxs  gentls_cert  tone2wav
>> validator
>> Is any of them the required binary? Where is it launched from?
>>
>> Thank you,
>> Victor
>>
>>
>> On 09/02/11 09:02 AM, Steven Ayre wrote:
>>
>> What user do you run FS as?
>>
>> It's anothe program in the same bin folder as freeswitch, and should run
>> as the same user as freeswitch itself. There's no logging I'm aware of.
>>
>> -Steve
>>
>>
>>  On 9 February 2011 13:25, Victor Chukalovskiy <
>> victor.chukalovskiy at utoronto.ca> wrote:
>>
>>>  Thanks David,
>>>
>>> Freeswitch was restarted.
>>> Still "can't contact licence server"
>>>
>>> Is this license server a process / module running on the same machine?
>>> If so, does it log anything anywhere? Where is it started from?
>>>
>>> -Victor
>>>
>>>
>>> On 09/02/11 08:01 AM, David Ponzone wrote:
>>>
>>> The license server is normally launched automatically.
>>> If you can, restart FreeSWITCH.
>>> It's normally not required.
>>>
>>>   David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>>  Service Client IPeva
>>>  tel:      0811 46 26 26
>>>  www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>>  Le 09/02/2011 ? 13:40, Victor Chukalovskiy a ?crit :
>>>
>>>  Hi David,
>>>
>>> I'm aware of the file and was using it for install.
>>>
>>> There is no single word there what "licence server" is
>>> and how to deal with "can't contact licence server".
>>>
>>> Thank you,
>>> Victor
>>>
>>> On 09/02/11 03:55 AM, David Ponzone wrote:
>>>
>>> There is a file coming with the module which explains the installation
>>> quite perfectly.
>>>
>>>   David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>>  Service Client IPeva
>>>  tel:      0811 46 26 26
>>>  www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>>  Le 09/02/2011 ? 05:41, Victor Chukalovskiy a ?crit :
>>>
>>>  This didn't help.
>>> I reloaded mod_com_g729, tried g729_info and the same "can't contact
>>> licence server" error
>>> -Victor
>>>
>>> On 08/02/11 11:10 PM, Michael Collins wrote:
>>>
>>> put the license file in /etc/freeswitch
>>> -MC
>>>
>>> On Tue, Feb 8, 2011 at 9:56 PM, Victor Chukalovskiy <
>>> victor.chukalovskiy at utoronto.ca> wrote:
>>>
>>>>  Hi Michael and others,
>>>>
>>>> After updating to the latest GIT mod_com_g729 loads successfully.
>>>>
>>>> Next problem appears:
>>>> when typing "g729_info" freeswitch replies "can't contact licence
>>>> server".
>>>> g729_available gives "False"
>>>>
>>>> How to solve this?
>>>>
>>>> I have my xxxxx.conf license file in the root of FS install directory
>>>> /opt/fs
>>>> Should it be placed elsewhere?
>>>>
>>>> Also, xxxxx.conf was created with previous non-successful install of
>>>> mod_com_g729
>>>> Should I run validator again with the same license key?
>>>>
>>>> Thank you,
>>>> Victor
>>>>
>>>> On 07/02/11 10:36 AM, Michael Collins wrote:
>>>>
>>>> Any chance you can update to latest git? Your life will be easier. There
>>>> have been notable improvements in FS in the past few months.
>>>>
>>>>  -MC
>>>>
>>>> On Sat, Feb 5, 2011 at 3:30 PM, Victor Chukalovskiy <
>>>> victor.chukalovskiy at utoronto.ca> wrote:
>>>>
>>>>>  Hello,
>>>>>
>>>>> After purchasing a few licenses and installing the latest
>>>>> fsg729-191-installer
>>>>> I'm getting the following error when trying to load the mod_com_g729:
>>>>> > 2011-02-04 14:27:17.637046 [CRIT] switch_loadable_module.c:926 Error
>>>>> Loading module /opt/fs/mod/mod_com_g729.so
>>>>> > **Trying to load an out of date module, please rebuild the module.**
>>>>>
>>>>> Also noticed that g729 installer ran with a couple errors:
>>>>> > ./installer: line 62: ldconfig: command not found
>>>>> > ./installer: line 49: useradd: command not found
>>>>>  Any help or solution is much appreciated.
>>>>>
>>>>> -Victor
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> 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
>>>
>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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 gmaruzz at gmail.com  Wed Feb  9 17:26:25 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 9 Feb 2011 15:26:25 +0100
Subject: [Freeswitch-users] skypopen: problem loading the custom alsa
	driver
In-Reply-To: <Pine.LNX.4.64.1102091458490.29294@suse.blindi.net>
References: <AANLkTin=oJpzotF4x4+eYKKrazUaKxoKOdx9yAWA7ewD@mail.gmail.com>
	<Pine.LNX.4.64.1102091322460.28387@suse.blindi.net>
	<AANLkTi=p5U4Q7tGaz4=pErzhi8esy+gZpJ7U_NL4SESY@mail.gmail.com>
	<Pine.LNX.4.64.1102091458490.29294@suse.blindi.net>
Message-ID: <AANLkTikDzWp=k42132oyTnM3tzWQ-3jTkuoou0g9EDc8@mail.gmail.com>

please install a normal standard version of Ubuntu server (eg: from
downloaded cd).

probably with debootstrap you lack something, or in some way it's different.

btw, for eg, the kernels installed for "virtual" flavor of kvm in
ubuntu, they lack sound support (it's just not compiled. Sound support
is an option when you compile the kernel. It has nothing to do with
ALSA, but ALSA need it to be loaded).

maybe if you repeat your procedure, starting from a bootstrap, make
sure you install a kernel with sound support, and that have alsa
drivers. Check that you can modprobe snd-dummy before trying to
compile and install the new alsa version.

If the kernel you install has no alsa drivers installed by default,
probably it was compiled without sound support. So, you can for sure
compile and install alsa drivers, but they will not be loaded by that
kernel, because the kernel itself is compiled without sound support.

-giovanni

On Wed, Feb 9, 2011 at 3:13 PM, Thomas Hoellriegel <admin at blindi.net> wrote:
> Am 09.02.11 um 14:52 schrieb Giovanni Maruzzelli:
>
>> Are you using a normal ubuntu installation? I mean, the normal stock
>> kernel that Ubuntu gives you?
>
> I have installed ?a minimalsystem from debootstrap.
> ssh is optional installed.
>
> Yes the standradkernel from ubuntu is installed.
> root at freeswitch:~# uname -a
> Linux freeswitch 2.6.32-27-generic-pae #49-Ubuntu SMP Thu Dec 2 00:07:52 UTC
> 201
> 0 i686 GNU/Linux
>
>> Do you have done:
>> apt-get install linux-headers-$(uname -r)
>
> yes:
> the output:
> root at freeswitch:~# dpkg -l ?|grep -i linux-headers
> ii ?linux-headers-2.6.32-27 ? ? ? ? ? ? 2.6.32-27.49 ? ? ? ? ? ? ? ?Header
> files
> ?related to Linux kernel version
> ii ?linux-headers-2.6.32-27-generic-pae 2.6.32-27.49 ? ? ? ? ? ? ? ?Linux
> kernel
> ?headers for version 2.6.32 ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/listinfo/blinde-misc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 infos at madovsky.org  Wed Feb  9 17:29:54 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 9 Feb 2011 09:29:54 -0500
Subject: [Freeswitch-users] RTP keep alive
References: <AANLkTik-o+kpsLyTTtjL09Zg4M9a23aPygFj1d-V=cTb@mail.gmail.com>
	<AANLkTi=aUeWAo+54O3U_LAK1kEr-U3wTNty+4AS5HG42@mail.gmail.com>
Message-ID: <8F741D5883B04D91802BD322CEAE0079@e1705>

some trunk check generate media timeout
if no voice is transited after a while...
  ----- Original Message ----- 
  From: Sam 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, February 09, 2011 3:49 AM
  Subject: Re: [Freeswitch-users] RTP keep alive


  I have this in the settings,

      <param name="rtp-timeout-sec" value="3000"/>
      <param name="rtp-hold-timeout-sec" value="18000"/>

  Any thing i need more regarding that ?

  Regards
  Sam


  On Wed, Feb 9, 2011 at 1:35 PM, Sam <u2nsam at gmail.com> wrote:

    Hello,

    We have a situation wherein we need to keep alive RTP ,is there any parameter to do that,
    because when someone is on long conversation and not talking fro brief duration the call
    disconnects .

    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 Nabble at slickdeals.endjunk.com  Wed Feb  9 17:30:41 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 9 Feb 2011 06:30:41 -0800 (PST)
Subject: [Freeswitch-users] git-a93623 version (2011/02/07) causes
 [DESTINATION_OUT_OF_ORDER]
In-Reply-To: <AANLkTing-OkaLem1HLJPQSCaSORSvLZZk-m4M0cXDqFn@mail.gmail.com>
References: <1297183735886-6004663.post@n2.nabble.com>
	<AANLkTimOM9dZYr=zrQB+eJAEQ7fh0ugvqJQueOtOGiZH@mail.gmail.com>
	<1297185192140-6004757.post@n2.nabble.com>
	<AANLkTikzn43AqJaEUaWFpa0Yy70KLz-64Cb-8Jk6Fo8W@mail.gmail.com>
	<1297189519810-6005004.post@n2.nabble.com>
	<AANLkTing-OkaLem1HLJPQSCaSORSvLZZk-m4M0cXDqFn@mail.gmail.com>
Message-ID: <1297261841390-6007834.post@n2.nabble.com>



Anthony Minessale wrote:
> When you are jumping back and forth to different versions that are not
> binary compatible you are going to have problems.
That's clearly understood. BTW, I kept banging my FS with repeated outbound
calls on the same SIP trunk. Sometimes the calls went through while other
times the output from fs_cli showed [DESTINATION_OUT_OF_ORDER].

Anyway, I have two FS systems, i.e. one for production use and one for
testing use in my private LAN. When I upgraded the test unit with the latest
from a git build, I copied all the files in conf/ directory from the
production unit to the test unit. Perhaps, the files in conf/ directory from
the productions unit (FreeSWITCH Version 1.0.head (git-cf253c3 2011-01-11
15-31-40 -0600)) are not compatible with the ones from the latest git
version.

-----
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/git-a93623-version-2011-02-07-causes-DESTINATION-OUT-OF-ORDER-tp6004663p6007834.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Wed Feb  9 17:33:31 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 9 Feb 2011 14:33:31 +0000
Subject: [Freeswitch-users] RTP keep alive
In-Reply-To: <AANLkTi=aUeWAo+54O3U_LAK1kEr-U3wTNty+4AS5HG42@mail.gmail.com>
References: <AANLkTik-o+kpsLyTTtjL09Zg4M9a23aPygFj1d-V=cTb@mail.gmail.com>
	<AANLkTi=aUeWAo+54O3U_LAK1kEr-U3wTNty+4AS5HG42@mail.gmail.com>
Message-ID: <AANLkTi=0Uo-7qqWfB9uDFownX7TuVzsez3aVnUGMORH4@mail.gmail.com>

That'll do it.

VAD can mean no RTP is transmitted legitimately though, so be careful.
Especially when on hold when the phone might stop sending anything (I see
you've already increased that time).

-Steve


On 9 February 2011 08:49, Sam <u2nsam at gmail.com> wrote:

> I have this in the settings,
>
>     <param name="rtp-timeout-sec" value="3000"/>
>     <param name="rtp-hold-timeout-sec" value="18000"/>
>
> Any thing i need more regarding that ?
>
> Regards
> Sam
>
>
> On Wed, Feb 9, 2011 at 1:35 PM, Sam <u2nsam at gmail.com> wrote:
>
>> Hello,
>>
>> We have a situation wherein we need to keep alive RTP ,is there any
>> parameter to do that,
>> because when someone is on long conversation and not talking fro brief
>> duration the call
>> disconnects .
>>
>> 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 admin at blindi.net  Wed Feb  9 17:48:42 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Wed, 9 Feb 2011 15:48:42 +0100 (CET)
Subject: [Freeswitch-users] skypopen: problem loading the custom alsa
 driver
In-Reply-To: <AANLkTikDzWp=k42132oyTnM3tzWQ-3jTkuoou0g9EDc8@mail.gmail.com>
References: <AANLkTin=oJpzotF4x4+eYKKrazUaKxoKOdx9yAWA7ewD@mail.gmail.com>
	<Pine.LNX.4.64.1102091322460.28387@suse.blindi.net>
	<AANLkTi=p5U4Q7tGaz4=pErzhi8esy+gZpJ7U_NL4SESY@mail.gmail.com>
	<Pine.LNX.4.64.1102091458490.29294@suse.blindi.net>
	<AANLkTikDzWp=k42132oyTnM3tzWQ-3jTkuoou0g9EDc8@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1102091536150.29294@suse.blindi.net>

Am 09.02.11 um 15:26 schrieb Giovanni Maruzzelli:

> please install a normal standard version of Ubuntu server (eg: from
> downloaded cd).

I can.t install from a cd. The server is in a datacenter. I don.t have 
physical acess.
A Recoversystem is available to install a ubuntu plain-system.
then the original ubuntu is installed, the alsadriver works,
root at freeswitch:~# lsmod |grep snd_dummy
snd_dummy               9290  0
snd_pcm                70918  2 snd_dummy,snd_pcm_oss
snd                    54180  9 
snd_dummy,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_
seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device

when i delete the driver and install the compiledversion i become these 
errors.
must i compile the alsasource on a newer kernel?


---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From Nabble at slickdeals.endjunk.com  Wed Feb  9 18:00:54 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 9 Feb 2011 07:00:54 -0800 (PST)
Subject: [Freeswitch-users] How to dial a remote_skypename from a telephone
	keypad?
Message-ID: <1297263654900-6007946.post@n2.nabble.com>


If one uses mod_skypopen + skype client on an FS system with and IP Phone
and/or a telephone attached to an ATA device as an extension to the FS
sytem, how can one place a call to a Skype user whose account isn't in
numeric characters?

-----
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/How-to-dial-a-remote-skypename-from-a-telephone-keypad-tp6007946p6007946.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From david.ponzone at ipeva.fr  Wed Feb  9 18:02:33 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 9 Feb 2011 16:02:33 +0100
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <AANLkTimPj_27abwq4RNvpkfczkUXi5hf7-ro5v=yEc6E@mail.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>
Message-ID: <1AFB9015-306C-492A-AA09-5AD6905FD485@ipeva.fr>

Steven,

I just checked and my license server was installed in /usr/sbin.
So it should not be an issue.
Or perhaps the installer looks for freeswtich in standard paths, and it didn't find it, it doesn't install the license server.
But the module was installed correctly, wasn't it, Victor ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 09/02/2011 ? 15:24, Steven Ayre a ?crit :

> I see you're in a nonstandard directory.
> 
> As far as I know the license server only likes being in /usr/local/freeswitch/bin or /opt/freeswitch/bin.
> 
> Because it's neither of those it looks like the installer hasn't installed the license server.
> 
> -Steve
> 
> 
> On 9 February 2011 14:12, Victor Chukalovskiy <victor.chukalovskiy at utoronto.ca> wrote:
> Hi Steve,
> 
> I'm running as user "fs":
> /opt/fs/bin/freeswitch -nf -nc -u fs -g fs -core -nonat
> 
> I have following binaries in the directory:
> freeswitch  fs_cli  fs_encode  fs_ivrd  fsxs  gentls_cert  tone2wav  validator
> Is any of them the required binary? Where is it launched from?
> 
> Thank you,
> Victor
> 
> 
> On 09/02/11 09:02 AM, Steven Ayre wrote:
>> 
>> What user do you run FS as?
>> 
>> It's anothe program in the same bin folder as freeswitch, and should run as the same user as freeswitch itself. There's no logging I'm aware of.
>> 
>> -Steve
>> 
>> 
>> On 9 February 2011 13:25, Victor Chukalovskiy <victor.chukalovskiy at utoronto.ca> wrote:
>> Thanks David,
>> 
>> Freeswitch was restarted.
>> Still "can't contact licence server"
>> 
>> Is this license server a process / module running on the same machine?
>> If so, does it log anything anywhere? Where is it started from?
>> 
>> -Victor
>> 
>> 
>> On 09/02/11 08:01 AM, David Ponzone wrote:
>>> 
>>> The license server is normally launched automatically.
>>> If you can, restart FreeSWITCH.
>>> It's normally not required.
>>> 
>>> David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>> 
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>> 
>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>> 
>>> 
>>> 
>>> 
>>> Le 09/02/2011 ? 13:40, Victor Chukalovskiy a ?crit :
>>> 
>>>> Hi David,
>>>> 
>>>> I'm aware of the file and was using it for install.
>>>> 
>>>> There is no single word there what "licence server" is 
>>>> and how to deal with "can't contact licence server".
>>>> 
>>>> Thank you,
>>>> Victor
>>>> 
>>>> On 09/02/11 03:55 AM, David Ponzone wrote:
>>>>> 
>>>>> There is a file coming with the module which explains the installation quite perfectly.
>>>>> 
>>>>> David Ponzone  Direction Technique
>>>>> email: david.ponzone at ipeva.fr
>>>>> tel:      01 74 03 18 97
>>>>> gsm:   06 66 98 76 34
>>>>> 
>>>>> Service Client IPeva
>>>>> tel:      0811 46 26 26
>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>> 
>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Le 09/02/2011 ? 05:41, Victor Chukalovskiy a ?crit :
>>>>> 
>>>>>> This didn't help.
>>>>>> I reloaded mod_com_g729, tried g729_info and the same "can't contact licence server" error
>>>>>> -Victor
>>>>>> 
>>>>>> On 08/02/11 11:10 PM, Michael Collins wrote:
>>>>>>> 
>>>>>>> put the license file in /etc/freeswitch
>>>>>>> -MC
>>>>>>> 
>>>>>>> On Tue, Feb 8, 2011 at 9:56 PM, Victor Chukalovskiy <victor.chukalovskiy at utoronto.ca> wrote:
>>>>>>> Hi Michael and others,
>>>>>>> 
>>>>>>> After updating to the latest GIT mod_com_g729 loads successfully.
>>>>>>> 
>>>>>>> Next problem appears: 
>>>>>>> when typing "g729_info" freeswitch replies "can't contact licence server".
>>>>>>> g729_available gives "False"
>>>>>>> 
>>>>>>> How to solve this?
>>>>>>> 
>>>>>>> I have my xxxxx.conf license file in the root of FS install directory /opt/fs
>>>>>>> Should it be placed elsewhere?
>>>>>>> 
>>>>>>> Also, xxxxx.conf was created with previous non-successful install of mod_com_g729
>>>>>>> Should I run validator again with the same license key?
>>>>>>> 
>>>>>>> Thank you,
>>>>>>> Victor
>>>>>>> 
>>>>>>> On 07/02/11 10:36 AM, Michael Collins wrote:
>>>>>>>> 
>>>>>>>> Any chance you can update to latest git? Your life will be easier. There have been notable improvements in FS in the past few months. 
>>>>>>>> 
>>>>>>>> -MC
>>>>>>>> 
>>>>>>>> On Sat, Feb 5, 2011 at 3:30 PM, Victor Chukalovskiy <victor.chukalovskiy at utoronto.ca> wrote:
>>>>>>>> Hello,
>>>>>>>> 
>>>>>>>> After purchasing a few licenses and installing the latest fsg729-191-installer
>>>>>>>> I'm getting the following error when trying to load the mod_com_g729:
>>>>>>>> > 2011-02-04 14:27:17.637046 [CRIT] switch_loadable_module.c:926 Error Loading module /opt/fs/mod/mod_com_g729.so
>>>>>>>> > **Trying to load an out of date module, please rebuild the module.**
>>>>>>>> 
>>>>>>>> Also noticed that g729 installer ran with a couple errors:
>>>>>>>> > ./installer: line 62: ldconfig: command not found
>>>>>>>> > ./installer: line 49: useradd: command not found
>>>>>>>>  Any help or solution is much appreciated.
>>>>>>>> 
>>>>>>>> -Victor 
>>>>>>>> 
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 gmaruzz at gmail.com  Wed Feb  9 18:03:37 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 9 Feb 2011 16:03:37 +0100
Subject: [Freeswitch-users] skypopen: problem loading the custom alsa
	driver
In-Reply-To: <Pine.LNX.4.64.1102091536150.29294@suse.blindi.net>
References: <AANLkTin=oJpzotF4x4+eYKKrazUaKxoKOdx9yAWA7ewD@mail.gmail.com>
	<Pine.LNX.4.64.1102091322460.28387@suse.blindi.net>
	<AANLkTi=p5U4Q7tGaz4=pErzhi8esy+gZpJ7U_NL4SESY@mail.gmail.com>
	<Pine.LNX.4.64.1102091458490.29294@suse.blindi.net>
	<AANLkTikDzWp=k42132oyTnM3tzWQ-3jTkuoou0g9EDc8@mail.gmail.com>
	<Pine.LNX.4.64.1102091536150.29294@suse.blindi.net>
Message-ID: <AANLkTiniayXKmqZZo7HDPYwtR3b4X0vSAHPqTeOpcpEL@mail.gmail.com>

maybe I have not understood your last mail.

My hint would be:

1) install a plain ubuntu (with the recovery procedure)
2) follow strictly, from the beginning to end, step by step the wiki page

particularly, after installing a  plain ubuntu, delete (if is still
there) the directory with the alsa sources, re-download or
re-decompress the package, and do the custom alsa driver procedure
from beginning.

If still does not works, maybe the system that got installed with the
recovery procedure is not a plain ubuntu system.

But I hope it will works.

-giovanni

On Wed, Feb 9, 2011 at 3:48 PM, Thomas Hoellriegel <admin at blindi.net> wrote:
> Am 09.02.11 um 15:26 schrieb Giovanni Maruzzelli:
>
>> please install a normal standard version of Ubuntu server (eg: from
>> downloaded cd).
>
> I can.t install from a cd. The server is in a datacenter. I don.t have
> physical acess.
> A Recoversystem is available to install a ubuntu plain-system.
> then the original ubuntu is installed, the alsadriver works,
> root at freeswitch:~# lsmod |grep snd_dummy
> snd_dummy ? ? ? ? ? ? ? 9290 ?0
> snd_pcm ? ? ? ? ? ? ? ?70918 ?2 snd_dummy,snd_pcm_oss
> snd ? ? ? ? ? ? ? ? ? ?54180 ?9
> snd_dummy,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_
> seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device
>
> when i delete the driver and install the compiledversion i become these
> errors.
> must i compile the alsasource on a newer kernel?
>
>
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/listinfo/blinde-misc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 gmaruzz at gmail.com  Wed Feb  9 18:06:54 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 9 Feb 2011 16:06:54 +0100
Subject: [Freeswitch-users] How to dial a remote_skypename from a
 telephone keypad?
In-Reply-To: <1297263654900-6007946.post@n2.nabble.com>
References: <1297263654900-6007946.post@n2.nabble.com>
Message-ID: <AANLkTikfXRGxqbrassDNhOnaOWkaje3DXQptRAVRzb9d@mail.gmail.com>

you put it in dialplan, or some other hack.

eg: extension "123456" will call skypeuser "giovanni334"

no other way with an ATA

On Wed, Feb 9, 2011 at 4:00 PM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
> If one uses mod_skypopen + skype client on an FS system with and IP Phone
> and/or a telephone attached to an ATA device as an extension to the FS
> sytem, how can one place a call to a Skype user whose account isn't in
> numeric characters?
>
> -----
> 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/How-to-dial-a-remote-skypename-from-a-telephone-keypad-tp6007946p6007946.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 gmaruzz at gmail.com  Wed Feb  9 18:16:01 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 9 Feb 2011 16:16:01 +0100
Subject: [Freeswitch-users] How to dial a remote_skypename from a
 telephone keypad?
In-Reply-To: <AANLkTikfXRGxqbrassDNhOnaOWkaje3DXQptRAVRzb9d@mail.gmail.com>
References: <1297263654900-6007946.post@n2.nabble.com>
	<AANLkTikfXRGxqbrassDNhOnaOWkaje3DXQptRAVRzb9d@mail.gmail.com>
Message-ID: <AANLkTikJNH3-vk=dMkOjAKsAxgr7KaiGD-U-9t2uKjxE@mail.gmail.com>

with a SIP phone that allows you to enter alphanumeric destinations,
you can use something similar to the "skype_uri" described in:
http://wiki.freeswitch.org/wiki/Skypopen_Skype_Endpoint_and_Trunk#Dialplan.2C_and_how_to_use_Skypopen

On Wed, Feb 9, 2011 at 4:06 PM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> you put it in dialplan, or some other hack.
>
> eg: extension "123456" will call skypeuser "giovanni334"
>
> no other way with an ATA
>
> On Wed, Feb 9, 2011 at 4:00 PM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>>
>> If one uses mod_skypopen + skype client on an FS system with and IP Phone
>> and/or a telephone attached to an ATA device as an extension to the FS
>> sytem, how can one place a call to a Skype user whose account isn't in
>> numeric characters?
>>
>> -----
>> 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/How-to-dial-a-remote-skypename-from-a-telephone-keypad-tp6007946p6007946.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
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From dujinfang at gmail.com  Wed Feb  9 18:21:33 2011
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 9 Feb 2011 23:21:33 +0800
Subject: [Freeswitch-users] Video on X-lite .. FSV module
In-Reply-To: <AANLkTi=urew0--m1NXKuDpw74aDdCh6FERdPQ2ZyvwT-@mail.gmail.com>
References: <loom.20110209T020200-551@post.gmane.org>
	<AANLkTi=urew0--m1NXKuDpw74aDdCh6FERdPQ2ZyvwT-@mail.gmail.com>
Message-ID: <AANLkTin=MWHEZjJTkwVAsmn8o7b72UJSRSP5o90dC7C1@mail.gmail.com>

you have to enable video codecs on your profile. check vars.xml

On Wed, Feb 9, 2011 at 5:30 PM, Steven Ayre <steveayre at gmail.com> wrote:
> Can you enable debug logging and post it?
>
> -Steve
>
>
> On 9 February 2011 01:05, sherrypioro <youstinayacoub at gmail.com> wrote:
>>
>> Hi,
>>
>>
>> I am trying to record or play video on X-lite through Freeswitch. When I
>> do a
>> video call it says faild to send your video.
>>
>> I have FSV module installed on freeswitch and running.
>>
>> Any Ideas????
>>
>> Thank you!!!
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 anthony.minessale at gmail.com  Wed Feb  9 18:45:20 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 9 Feb 2011 09:45:20 -0600
Subject: [Freeswitch-users] git HEAD 503 Maximum Calls In Progress
In-Reply-To: <AANLkTi=2t_gPtzyqHYaNuZz0VUNfOwDkWNmJHJx1YdeQ@mail.gmail.com>
References: <AANLkTimpGj4GLug8jrK-7-V3Mja9W2tTiwtJsBSu-TZo@mail.gmail.com>
	<AANLkTi=2t_gPtzyqHYaNuZz0VUNfOwDkWNmJHJx1YdeQ@mail.gmail.com>
Message-ID: <AANLkTik-CcvnBC2+inQ1DVoek=kjcMVBMTUm=PtWY01i@mail.gmail.com>

edit conf/autoload_configs/switch.conf.xml

comment out

<param name="min-idle-cpu" value="25"/>

It seems to not work well on 32 bit.


On Wed, Feb 9, 2011 at 5:52 AM, Steven Ayre <steveayre at gmail.com> wrote:
> What does show channels show?
>
>
> On 9 February 2011 11:37, Seven Du <dujinfang at gmail.com> wrote:
>>
>> Hi,
>>
>> It's a fresh install with wget ?freeswitch.org/eg/Makefile && make,
>> and I'm sure I successfully called 9196 and everything was ok.
>>
>> But a few minutes later I started getting
>>
>> ?503 Maximum Calls In Progress
>>
>> also originate user/1003 or loopback/9196 shows errors
>>
>> see
>> http://pastebin.freeswitch.org/15333
>>
>> It's on a ubuntu 8.04 32bit, the only thing I did between non-work and
>> work was I configured odbc with postgresql to work with isql. I
>> haven't touch the FS config yet. I don't think they are related.
>>
>>
>> here is also a nua debug
>>
>> http://pastebin.freeswitch.org/15335
>>
>> Any hint on this? Thanks.
>>
>> btw, I also noticed there might be some network problems between me
>> and the server where sometime sip packets not shown ( sofia debug and
>> ngrep) on server. Anyway, since loopback cannot create sessions so
>> something must be wrong in FS side, but what's that?
>>
>> Thanks.
>>
>> --
>> 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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 victor.chukalovskiy at utoronto.ca  Wed Feb  9 18:46:04 2011
From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy)
Date: Wed, 09 Feb 2011 10:46:04 -0500
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <1AFB9015-306C-492A-AA09-5AD6905FD485@ipeva.fr>
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>
Message-ID: <4D52B6BC.30306@utoronto.ca>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110209/75c9d073/attachment-0001.html 

From philippe.sultan at gmail.com  Wed Feb  9 18:46:29 2011
From: philippe.sultan at gmail.com (Philippe Sultan)
Date: Wed, 9 Feb 2011 16:46:29 +0100
Subject: [Freeswitch-users] Aastra phone registration lost
Message-ID: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>

Hello to the FreeSWITCH users,

I'm experiencing problems registering an Aastra 6731i phone to
FreeSWITCH. The registration process itself works fine, but the phone
never re-registers to FS after the expiration time specified in the
expires paramater set in the Contact SIP Header Field. And therefore
becomes unreachable from FS after some time.

The phone seems to be buggy because it simply ignores the value given
in the expires parameter. Testing with Asterisk, the phone remains
accessible, because Asterisk adds an Expires HF in addition to the
expires param in the Contact HF.

Did anyone observed what's described here? I've tested several Aastra
firmwares with the same result. I have also tried to set the
registration expiration time on the phone itself without success.

I believe I can cope with this situation by adding an Expires HF from
the (SIP Express Router) proxy that stays between the phone and FS,
but I'm sure there's a simpler way to get a proper registation
process.

I'm using FreeSWITCH 1.0.6 (that comes along with the Blue.box GUI).

Kind regards,

Philippe


From anthony.minessale at gmail.com  Wed Feb  9 18:46:36 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 9 Feb 2011 09:46:36 -0600
Subject: [Freeswitch-users] git-a93623 version (2011/02/07) causes
	[DESTINATION_OUT_OF_ORDER]
In-Reply-To: <1297261841390-6007834.post@n2.nabble.com>
References: <1297183735886-6004663.post@n2.nabble.com>
	<AANLkTimOM9dZYr=zrQB+eJAEQ7fh0ugvqJQueOtOGiZH@mail.gmail.com>
	<1297185192140-6004757.post@n2.nabble.com>
	<AANLkTikzn43AqJaEUaWFpa0Yy70KLz-64Cb-8Jk6Fo8W@mail.gmail.com>
	<1297189519810-6005004.post@n2.nabble.com>
	<AANLkTing-OkaLem1HLJPQSCaSORSvLZZk-m4M0cXDqFn@mail.gmail.com>
	<1297261841390-6007834.post@n2.nabble.com>
Message-ID: <AANLkTi=kTEKr2n7MZy4wuvz5OpsGkD6zF=sNau4Dzu81@mail.gmail.com>

edit conf/autoload_configs/switch.conf.xml

comment out

<param name="min-idle-cpu" value="25"/>

It seems to not work well on 32 bit.


On Wed, Feb 9, 2011 at 8:30 AM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
>
> Anthony Minessale wrote:
>> When you are jumping back and forth to different versions that are not
>> binary compatible you are going to have problems.
> That's clearly understood. BTW, I kept banging my FS with repeated outbound
> calls on the same SIP trunk. Sometimes the calls went through while other
> times the output from fs_cli showed [DESTINATION_OUT_OF_ORDER].
>
> Anyway, I have two FS systems, i.e. one for production use and one for
> testing use in my private LAN. When I upgraded the test unit with the latest
> from a git build, I copied all the files in conf/ directory from the
> production unit to the test unit. Perhaps, the files in conf/ directory from
> the productions unit (FreeSWITCH Version 1.0.head (git-cf253c3 2011-01-11
> 15-31-40 -0600)) are not compatible with the ones from the latest git
> version.
>
> -----
> 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/git-a93623-version-2011-02-07-causes-DESTINATION-OUT-OF-ORDER-tp6004663p6007834.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 david.ponzone at ipeva.fr  Wed Feb  9 18:51:50 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 9 Feb 2011 16:51:50 +0100
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <4D52B6BC.30306@utoronto.ca>
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>
Message-ID: <867B4DFD-1F6C-41FE-B88F-677E1C45E5F3@ipeva.fr>

Normally, it is started when you reload the module.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 09/02/2011 ? 16:46, Victor Chukalovskiy a ?crit :

> Yes, the module is installed correctly. It loads Ok.
> 
> I re-ran sudo ./fsg729-191-installer. 
> license server is indeed installed in /usr/sbin. Would be better if installer puts it into FS install path - just where it belongs.
> 
> It was root:root owned, so I chown'ed it to the correct user fs:fs (this is how I run freeswitch)
> 
> Should it help? I'm not able to restart FS until tonight.
> Is there any other way to (re)-start this licensing server?
> 
> 
> -Victor
> 
> On 09/02/11 10:02 AM, David Ponzone wrote:
>> 
>> Steven,
>> 
>> I just checked and my license server was installed in /usr/sbin.
>> So it should not be an issue.
>> Or perhaps the installer looks for freeswtich in standard paths, and it didn't find it, it doesn't install the license server.
>> But the module was installed correctly, wasn't it, Victor ?
>> 
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> 
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> 
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> 
>> 
>> 
>> 
>> Le 09/02/2011 ? 15:24, Steven Ayre a ?crit :
>> 
>>> I see you're in a nonstandard directory.
>>> 
>>> As far as I know the license server only likes being in /usr/local/freeswitch/bin or /opt/freeswitch/bin.
>>> 
>>> Because it's neither of those it looks like the installer hasn't installed the license server.
>>> 
>>> -Steve
>>> 
>>> 
>>> On 9 February 2011 14:12, Victor Chukalovskiy <victor.chukalovskiy at utoronto.ca> wrote:
>>> Hi Steve,
>>> 
>>> I'm running as user "fs":
>>> /opt/fs/bin/freeswitch -nf -nc -u fs -g fs -core -nonat
>>> 
>>> I have following binaries in the directory:
>>> freeswitch  fs_cli  fs_encode  fs_ivrd  fsxs  gentls_cert  tone2wav  validator
>>> Is any of them the required binary? Where is it launched from?
>>> 
>>> Thank you,
>>> Victor
>>> 
>>> 
>>> On 09/02/11 09:02 AM, Steven Ayre wrote:
>>>> 
>>>> What user do you run FS as?
>>>> 
>>>> It's anothe program in the same bin folder as freeswitch, and should run as the same user as freeswitch itself. There's no logging I'm aware of.
>>>> 
>>>> -Steve
>>>> 
>>>> 
>>>> On 9 February 2011 13:25, Victor Chukalovskiy <victor.chukalovskiy at utoronto.ca> wrote:
>>>> Thanks David,
>>>> 
>>>> Freeswitch was restarted.
>>>> Still "can't contact licence server"
>>>> 
>>>> Is this license server a process / module running on the same machine?
>>>> If so, does it log anything anywhere? Where is it started from?
>>>> 
>>>> -Victor
>>>> 
>>>> 
>>>> On 09/02/11 08:01 AM, David Ponzone wrote:
>>>>> 
>>>>> The license server is normally launched automatically.
>>>>> If you can, restart FreeSWITCH.
>>>>> It's normally not required.
>>>>> 
>>>>> David Ponzone  Direction Technique
>>>>> email: david.ponzone at ipeva.fr
>>>>> tel:      01 74 03 18 97
>>>>> gsm:   06 66 98 76 34
>>>>> 
>>>>> Service Client IPeva
>>>>> tel:      0811 46 26 26
>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>> 
>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Le 09/02/2011 ? 13:40, Victor Chukalovskiy a ?crit :
>>>>> 
>>>>>> Hi David,
>>>>>> 
>>>>>> I'm aware of the file and was using it for install.
>>>>>> 
>>>>>> There is no single word there what "licence server" is 
>>>>>> and how to deal with "can't contact licence server".
>>>>>> 
>>>>>> Thank you,
>>>>>> Victor
>>>>>> 
>>>>>> On 09/02/11 03:55 AM, David Ponzone wrote:
>>>>>>> 
>>>>>>> There is a file coming with the module which explains the installation quite perfectly.
>>>>>>> 
>>>>>>> David Ponzone  Direction Technique
>>>>>>> email: david.ponzone at ipeva.fr
>>>>>>> tel:      01 74 03 18 97
>>>>>>> gsm:   06 66 98 76 34
>>>>>>> 
>>>>>>> Service Client IPeva
>>>>>>> tel:      0811 46 26 26
>>>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>>>> 
>>>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> Le 09/02/2011 ? 05:41, Victor Chukalovskiy a ?crit :
>>>>>>> 
>>>>>>>> This didn't help.
>>>>>>>> I reloaded mod_com_g729, tried g729_info and the same "can't contact licence server" error
>>>>>>>> -Victor
>>>>>>>> 
>>>>>>>> On 08/02/11 11:10 PM, Michael Collins wrote:
>>>>>>>>> 
>>>>>>>>> put the license file in /etc/freeswitch
>>>>>>>>> -MC
>>>>>>>>> 
>>>>>>>>> On Tue, Feb 8, 2011 at 9:56 PM, Victor Chukalovskiy <victor.chukalovskiy at utoronto.ca> wrote:
>>>>>>>>> Hi Michael and others,
>>>>>>>>> 
>>>>>>>>> After updating to the latest GIT mod_com_g729 loads successfully.
>>>>>>>>> 
>>>>>>>>> Next problem appears: 
>>>>>>>>> when typing "g729_info" freeswitch replies "can't contact licence server".
>>>>>>>>> g729_available gives "False"
>>>>>>>>> 
>>>>>>>>> How to solve this?
>>>>>>>>> 
>>>>>>>>> I have my xxxxx.conf license file in the root of FS install directory /opt/fs
>>>>>>>>> Should it be placed elsewhere?
>>>>>>>>> 
>>>>>>>>> Also, xxxxx.conf was created with previous non-successful install of mod_com_g729
>>>>>>>>> Should I run validator again with the same license key?
>>>>>>>>> 
>>>>>>>>> Thank you,
>>>>>>>>> Victor
>>>>>>>>> 
>>>>>>>>> On 07/02/11 10:36 AM, Michael Collins wrote:
>>>>>>>>>> 
>>>>>>>>>> Any chance you can update to latest git? Your life will be easier. There have been notable improvements in FS in the past few months. 
>>>>>>>>>> 
>>>>>>>>>> -MC
>>>>>>>>>> 
>>>>>>>>>> On Sat, Feb 5, 2011 at 3:30 PM, Victor Chukalovskiy <victor.chukalovskiy at utoronto.ca> wrote:
>>>>>>>>>> Hello,
>>>>>>>>>> 
>>>>>>>>>> After purchasing a few licenses and installing the latest fsg729-191-installer
>>>>>>>>>> I'm getting the following error when trying to load the mod_com_g729:
>>>>>>>>>> > 2011-02-04 14:27:17.637046 [CRIT] switch_loadable_module.c:926 Error Loading module /opt/fs/mod/mod_com_g729.so
>>>>>>>>>> > **Trying to load an out of date module, please rebuild the module.**
>>>>>>>>>> 
>>>>>>>>>> Also noticed that g729 installer ran with a couple errors:
>>>>>>>>>> > ./installer: line 62: ldconfig: command not found
>>>>>>>>>> > ./installer: line 49: useradd: command not found
>>>>>>>>>>  Any help or solution is much appreciated.
>>>>>>>>>> 
>>>>>>>>>> -Victor 
>>>>>>>>>> 
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>> 
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Feb  9 18:57:59 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Wed, 09 Feb 2011 10:57:59 -0500
Subject: [Freeswitch-users] Aastra phone registration lost
In-Reply-To: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>
Message-ID: <4D52B987.7050002@communicatefreely.net>

Hi Philippe,

I have been testing just about every model of phone that Aastra makes 
(or has made), on our Freeswitch system, and they all work fine, even 
behind NAT.  It's probably just a setting that isn't right, as you can 
adjust a lot of this stuff in the phone config file.

Can you post your aastra.cfg?  I'll take a look and see if there is 
anything obvious.

What is your topology?  Are the phone behind NAT, or is this a LAN 
environment?

-Tim

Philippe Sultan wrote:
> Hello to the FreeSWITCH users,
>
> I'm experiencing problems registering an Aastra 6731i phone to
> FreeSWITCH. The registration process itself works fine, but the phone
> never re-registers to FS after the expiration time specified in the
> expires paramater set in the Contact SIP Header Field. And therefore
> becomes unreachable from FS after some time.
>
> The phone seems to be buggy because it simply ignores the value given
> in the expires parameter. Testing with Asterisk, the phone remains
> accessible, because Asterisk adds an Expires HF in addition to the
> expires param in the Contact HF.
>
> Did anyone observed what's described here? I've tested several Aastra
> firmwares with the same result. I have also tried to set the
> registration expiration time on the phone itself without success.
>
> I believe I can cope with this situation by adding an Expires HF from
> the (SIP Express Router) proxy that stays between the phone and FS,
> but I'm sure there's a simpler way to get a proper registation
> process.
>
> I'm using FreeSWITCH 1.0.6 (that comes along with the Blue.box GUI).
>
> Kind regards,
>
> Philippe
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb  9 18:57:44 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 9 Feb 2011 15:57:44 +0000
Subject: [Freeswitch-users] git HEAD 503 Maximum Calls In Progress
In-Reply-To: <AANLkTimpGj4GLug8jrK-7-V3Mja9W2tTiwtJsBSu-TZo@mail.gmail.com>
References: <AANLkTimpGj4GLug8jrK-7-V3Mja9W2tTiwtJsBSu-TZo@mail.gmail.com>
Message-ID: <AANLkTi=n72vrWJDKB4G2yNkwm+8hdgnwAuCKBPP7xac2@mail.gmail.com>

Upgrade to 64bit if you can


On 9 February 2011 11:37, Seven Du <dujinfang at gmail.com> wrote:

> Hi,
>
> It's a fresh install with wget  freeswitch.org/eg/Makefile && make,
> and I'm sure I successfully called 9196 and everything was ok.
>
> But a few minutes later I started getting
>
>  503 Maximum Calls In Progress
>
> also originate user/1003 or loopback/9196 shows errors
>
> see
> http://pastebin.freeswitch.org/15333
>
> It's on a ubuntu 8.04 32bit, the only thing I did between non-work and
> work was I configured odbc with postgresql to work with isql. I
> haven't touch the FS config yet. I don't think they are related.
>
>
> here is also a nua debug
>
> http://pastebin.freeswitch.org/15335
>
> Any hint on this? Thanks.
>
> btw, I also noticed there might be some network problems between me
> and the server where sometime sip packets not shown ( sofia debug and
> ngrep) on server. Anyway, since loopback cannot create sessions so
> something must be wrong in FS side, but what's that?
>
> Thanks.
>
> --
> 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
>
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From victor.chukalovskiy at utoronto.ca  Wed Feb  9 18:59:28 2011
From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy)
Date: Wed, 09 Feb 2011 10:59:28 -0500
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <867B4DFD-1F6C-41FE-B88F-677E1C45E5F3@ipeva.fr>
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>
	<867B4DFD-1F6C-41FE-B88F-677E1C45E5F3@ipeva.fr>
Message-ID: <4D52B9E0.7070903@utoronto.ca>

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From steveayre at gmail.com  Wed Feb  9 18:59:14 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 9 Feb 2011 15:59:14 +0000
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <4D52B6BC.30306@utoronto.ca>
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>
Message-ID: <AANLkTinBUe4w3Ui0QRJPOiarDAqO+tkWS1d4X-FmT3ij@mail.gmail.com>

>
> Is there any other way to (re)-start this licensing server?
>

1) I think it runs when you load mod_com_g729
2) Run it manually from command line might work

-Steve



On 9 February 2011 15:46, Victor Chukalovskiy <
victor.chukalovskiy at utoronto.ca> wrote:

>  Yes, the module is installed correctly. It loads Ok.
>
> I re-ran sudo ./fsg729-191-installer.
> license server is indeed installed in /usr/sbin. Would be better if
> installer puts it into FS install path - just where it belongs.
>
> It was root:root owned, so I *chown*'ed it to the correct user fs:fs (this
> is how I run freeswitch)
>
> Should it help? I'm not able to restart FS until tonight.
> Is there any other way to (re)-start this licensing server?
>
>
> -Victor
>
>
> On 09/02/11 10:02 AM, David Ponzone wrote:
>
> Steven,
>
>  I just checked and my license server was installed in /usr/sbin.
> So it should not be an issue.
> Or perhaps the installer looks for freeswtich in standard paths, and it
> didn't find it, it doesn't install the license server.
> But the module was installed correctly, wasn't it, Victor ?
>
>   David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
>  Service Client IPeva
>  tel:      0811 46 26 26
>  www.ipeva.fr  -   www.ipeva-studio.com
>
>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
>  Le 09/02/2011 ? 15:24, Steven Ayre a ?crit :
>
> I see you're in a nonstandard directory.
>
> As far as I know the license server only likes being in
> /usr/local/freeswitch/bin or /opt/freeswitch/bin.
>
> Because it's neither of those it looks like the installer hasn't installed
> the license server.
>
> -Steve
>
>
> On 9 February 2011 14:12, Victor Chukalovskiy <
> victor.chukalovskiy at utoronto.ca> wrote:
>
>>  Hi Steve,
>>
>> I'm running as user "fs":
>> /opt/fs/bin/freeswitch -nf -nc -u fs -g fs -core -nonat
>>
>> I have following binaries in the directory:
>> freeswitch  fs_cli  fs_encode  fs_ivrd  fsxs  gentls_cert  tone2wav
>> validator
>> Is any of them the required binary? Where is it launched from?
>>
>> Thank you,
>>  Victor
>>
>>
>> On 09/02/11 09:02 AM, Steven Ayre wrote:
>>
>> What user do you run FS as?
>>
>> It's anothe program in the same bin folder as freeswitch, and should run
>> as the same user as freeswitch itself. There's no logging I'm aware of.
>>
>> -Steve
>>
>>
>>  On 9 February 2011 13:25, Victor Chukalovskiy <
>> victor.chukalovskiy at utoronto.ca> wrote:
>>
>>>  Thanks David,
>>>
>>> Freeswitch was restarted.
>>> Still "can't contact licence server"
>>>
>>> Is this license server a process / module running on the same machine?
>>> If so, does it log anything anywhere? Where is it started from?
>>>
>>> -Victor
>>>
>>>
>>> On 09/02/11 08:01 AM, David Ponzone wrote:
>>>
>>> The license server is normally launched automatically.
>>> If you can, restart FreeSWITCH.
>>> It's normally not required.
>>>
>>>   David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>>  Service Client IPeva
>>>  tel:      0811 46 26 26
>>>  www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>>  Le 09/02/2011 ? 13:40, Victor Chukalovskiy a ?crit :
>>>
>>>  Hi David,
>>>
>>> I'm aware of the file and was using it for install.
>>>
>>> There is no single word there what "licence server" is
>>> and how to deal with "can't contact licence server".
>>>
>>> Thank you,
>>> Victor
>>>
>>> On 09/02/11 03:55 AM, David Ponzone wrote:
>>>
>>> There is a file coming with the module which explains the installation
>>> quite perfectly.
>>>
>>>   David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>>  Service Client IPeva
>>>  tel:      0811 46 26 26
>>>  www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>>  Le 09/02/2011 ? 05:41, Victor Chukalovskiy a ?crit :
>>>
>>>  This didn't help.
>>> I reloaded mod_com_g729, tried g729_info and the same "can't contact
>>> licence server" error
>>> -Victor
>>>
>>> On 08/02/11 11:10 PM, Michael Collins wrote:
>>>
>>> put the license file in /etc/freeswitch
>>> -MC
>>>
>>> On Tue, Feb 8, 2011 at 9:56 PM, Victor Chukalovskiy <
>>> victor.chukalovskiy at utoronto.ca> wrote:
>>>
>>>>  Hi Michael and others,
>>>>
>>>> After updating to the latest GIT mod_com_g729 loads successfully.
>>>>
>>>> Next problem appears:
>>>> when typing "g729_info" freeswitch replies "can't contact licence
>>>> server".
>>>> g729_available gives "False"
>>>>
>>>> How to solve this?
>>>>
>>>> I have my xxxxx.conf license file in the root of FS install directory
>>>> /opt/fs
>>>> Should it be placed elsewhere?
>>>>
>>>> Also, xxxxx.conf was created with previous non-successful install of
>>>> mod_com_g729
>>>> Should I run validator again with the same license key?
>>>>
>>>> Thank you,
>>>> Victor
>>>>
>>>> On 07/02/11 10:36 AM, Michael Collins wrote:
>>>>
>>>> Any chance you can update to latest git? Your life will be easier. There
>>>> have been notable improvements in FS in the past few months.
>>>>
>>>>  -MC
>>>>
>>>> On Sat, Feb 5, 2011 at 3:30 PM, Victor Chukalovskiy <
>>>> victor.chukalovskiy at utoronto.ca> wrote:
>>>>
>>>>>  Hello,
>>>>>
>>>>> After purchasing a few licenses and installing the latest
>>>>> fsg729-191-installer
>>>>> I'm getting the following error when trying to load the mod_com_g729:
>>>>> > 2011-02-04 14:27:17.637046 [CRIT] switch_loadable_module.c:926 Error
>>>>> Loading module /opt/fs/mod/mod_com_g729.so
>>>>> > **Trying to load an out of date module, please rebuild the module.**
>>>>>
>>>>> Also noticed that g729 installer ran with a couple errors:
>>>>> > ./installer: line 62: ldconfig: command not found
>>>>> > ./installer: line 49: useradd: command not found
>>>>>  Any help or solution is much appreciated.
>>>>>
>>>>> -Victor
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> 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
>>>
>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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 list
> FreeSWITCH-users at lists.freeswitch.org
> 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 philippe.sultan at gmail.com  Wed Feb  9 19:10:44 2011
From: philippe.sultan at gmail.com (Philippe Sultan)
Date: Wed, 9 Feb 2011 17:10:44 +0100
Subject: [Freeswitch-users] Aastra phone registration lost
In-Reply-To: <4D52B987.7050002@communicatefreely.net>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>
	<4D52B987.7050002@communicatefreely.net>
Message-ID: <AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>

Tim,

Thanks a lot for your answer. No NAT involved here. However, the phone
is configured with an outbound proxy and the registrar/proxy
parameters are only used as realms (that is, they don't match with any
DNS A nor SRV record).

Here is my aastra.cfg file :
[root at xcom ~]# cat /tftpboot/aastra.cfg
upgrade file name: 6731i-2.6.0.st
upgrade uri: tftp://128.93.136.254:69
firmware md5: d74b608df20a8e785a7361701d15551e
boot count: 1
time server disabled: 0
time server1: 192.93.2.20
time server1: 128.93.1.9
time zone name: FR-Paris
time zone code: CET
date format: 5
dst config: 3
time format: 1
dhcp: 0
ip: 128.93.136.253
default gateway: 128.93.136.100
dns1: 128.93.1.9
dns2: 128.93.1.23
contact rcs: 0
sip transport protocol: 2
sip outbound support: 1
sip registration retry timer: 10
sip registration renewal timer: 15
sip registration timeout retry timer: 10
sip registration period: 60
sip rport: 1
sip auth name: 492
sip password: XXXXX
sip user name: 492
sip screen name: Philippe
sip screen name 2: 0178097204
sip proxy ip: vm.test.inria.fr
sip proxy port: 5060
sip registrar ip: vm.test.inria.fr
sip registrar port: 5060
sip outbound proxy: 128.93.136.252
sip outbound proxy port: 5060
sip backup proxy ip:
sip backup registrar ip: vm.test.inria.fr
sip backup registrar port: 5060
sip silence suppression: 0
sip transaction timer: 4000
sip rtp port: 10000
sip customized codec: payload=8;ptime=20;silsupp=off
sip line1 registration period: 60
sip line2 registration period: 60
log module sip: 10
lldp: 0
sip gruu: 0
sip instance id: 0

[root at xcom ~]#

Thanks again help,

Philippe

On Wed, Feb 9, 2011 at 4:57 PM, Tim St. Pierre
<fs-list at communicatefreely.net> wrote:
> Hi Philippe,
>
> I have been testing just about every model of phone that Aastra makes
> (or has made), on our Freeswitch system, and they all work fine, even
> behind NAT. ?It's probably just a setting that isn't right, as you can
> adjust a lot of this stuff in the phone config file.
>
> Can you post your aastra.cfg? ?I'll take a look and see if there is
> anything obvious.
>
> What is your topology? ?Are the phone behind NAT, or is this a LAN
> environment?
>
> -Tim
>
> Philippe Sultan wrote:
>> Hello to the FreeSWITCH users,
>>
>> I'm experiencing problems registering an Aastra 6731i phone to
>> FreeSWITCH. The registration process itself works fine, but the phone
>> never re-registers to FS after the expiration time specified in the
>> expires paramater set in the Contact SIP Header Field. And therefore
>> becomes unreachable from FS after some time.
>>
>> The phone seems to be buggy because it simply ignores the value given
>> in the expires parameter. Testing with Asterisk, the phone remains
>> accessible, because Asterisk adds an Expires HF in addition to the
>> expires param in the Contact HF.
>>
>> Did anyone observed what's described here? I've tested several Aastra
>> firmwares with the same result. I have also tried to set the
>> registration expiration time on the phone itself without success.
>>
>> I believe I can cope with this situation by adding an Expires HF from
>> the (SIP Express Router) proxy that stays between the phone and FS,
>> but I'm sure there's a simpler way to get a proper registation
>> process.
>>
>> I'm using FreeSWITCH 1.0.6 (that comes along with the Blue.box GUI).
>>
>> Kind regards,
>>
>> Philippe
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Philippe Sultan


From peder at networkoblivion.com  Wed Feb  9 19:16:25 2011
From: peder at networkoblivion.com (Peder)
Date: Wed, 9 Feb 2011 10:16:25 -0600
Subject: [Freeswitch-users] Re-order or Invalid
Message-ID: <056601cbc874$b3341e90$199c5bb0$@com>

What is the correct way to send a re-order or invalid message/tone to a
caller?  We have a couple hundred DIDs and quite a few of them are not used.
If someone calls into one of those, we just want to send a re-order or busy
tone.  My preference would be to NOT answer the call as I would prefer not
to be charged for the call if I can avoid it.

Peder




From anthony.minessale at gmail.com  Wed Feb  9 19:17:22 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 9 Feb 2011 10:17:22 -0600
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <AANLkTinBUe4w3Ui0QRJPOiarDAqO+tkWS1d4X-FmT3ij@mail.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>
Message-ID: <AANLkTi==uogdP7q2Nsqh7rcoaMy6tvy1at8kcNSqS=1q@mail.gmail.com>

try killing it.
you are probably still running the old one from the first time you tried it.


On Wed, Feb 9, 2011 at 9:59 AM, Steven Ayre <steveayre at gmail.com> wrote:
>> Is there any other way to (re)-start this licensing server?
>
> 1) I think it runs when you load mod_com_g729
> 2) Run it manually from command line might work
>
> -Steve
>
>
>
> On 9 February 2011 15:46, Victor Chukalovskiy
> <victor.chukalovskiy at utoronto.ca> wrote:
>>
>> Yes, the module is installed correctly. It loads Ok.
>>
>> I re-ran sudo ./fsg729-191-installer.
>> license server is indeed installed in /usr/sbin. Would be better if
>> installer puts it into FS install path - just where it belongs.
>>
>> It was root:root owned, so I chown'ed it to the correct user fs:fs (this
>> is how I run freeswitch)
>>
>> Should it help? I'm not able to restart FS until tonight.
>> Is there any other way to (re)-start this licensing server?
>>
>>
>> -Victor
>>
>> On 09/02/11 10:02 AM, David Ponzone wrote:
>>
>> Steven,
>> I just checked and my license server was installed in /usr/sbin.
>> So it should not be an issue.
>> Or perhaps the installer looks for freeswtich in standard paths, and it
>> didn't find it, it doesn't install the license server.
>> But the module was installed correctly, wasn't it, Victor ?
>> David Ponzone ?Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel: ? ? ?01 74 03 18 97
>> gsm: ? 06 66 98 76 34
>> Service Client?IPeva
>> tel: ? ? ?0811 46 26 26
>> www.ipeva.fr? -? ?www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>> Le 09/02/2011 ? 15:24, Steven Ayre a ?crit :
>>
>> I see you're in a nonstandard directory.
>>
>> As far as I know the license server only likes being in
>> /usr/local/freeswitch/bin or /opt/freeswitch/bin.
>>
>> Because it's neither of those it looks like the installer hasn't installed
>> the license server.
>>
>> -Steve
>>
>>
>> On 9 February 2011 14:12, Victor Chukalovskiy
>> <victor.chukalovskiy at utoronto.ca> wrote:
>>>
>>> Hi Steve,
>>>
>>> I'm running as user "fs":
>>> /opt/fs/bin/freeswitch -nf -nc -u fs -g fs -core -nonat
>>>
>>> I have following binaries in the directory:
>>> freeswitch? fs_cli? fs_encode? fs_ivrd? fsxs? gentls_cert? tone2wav
>>> validator
>>> Is any of them the required binary? Where is it launched from?
>>>
>>> Thank you,
>>> Victor
>>>
>>> On 09/02/11 09:02 AM, Steven Ayre wrote:
>>>
>>> What user do you run FS as?
>>>
>>> It's anothe program in the same bin folder as freeswitch, and should run
>>> as the same user as freeswitch itself. There's no logging I'm aware of.
>>>
>>> -Steve
>>>
>>>
>>> On 9 February 2011 13:25, Victor Chukalovskiy
>>> <victor.chukalovskiy at utoronto.ca> wrote:
>>>>
>>>> Thanks David,
>>>>
>>>> Freeswitch was restarted.
>>>> Still "can't contact licence server"
>>>>
>>>> Is this license server a process / module running on the same machine?
>>>> If so, does it log anything anywhere? Where is it started from?
>>>>
>>>> -Victor
>>>>
>>>> On 09/02/11 08:01 AM, David Ponzone wrote:
>>>>
>>>> The license server is normally launched automatically.
>>>> If you can, restart FreeSWITCH.
>>>> It's normally not required.
>>>> David Ponzone ?Direction Technique
>>>> email: david.ponzone at ipeva.fr
>>>> tel: ? ? ?01 74 03 18 97
>>>> gsm: ? 06 66 98 76 34
>>>> Service Client?IPeva
>>>> tel: ? ? ?0811 46 26 26
>>>> www.ipeva.fr? -? ?www.ipeva-studio.com
>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>
>>>>
>>>>
>>>> Le 09/02/2011 ? 13:40, Victor Chukalovskiy a ?crit :
>>>>
>>>> Hi David,
>>>>
>>>> I'm aware of the file and was using it for install.
>>>>
>>>> There is no single word there what "licence server" is
>>>> and how to deal with "can't contact licence server".
>>>>
>>>> Thank you,
>>>> Victor
>>>>
>>>> On 09/02/11 03:55 AM, David Ponzone wrote:
>>>>
>>>> There is a file coming with the module which explains the installation
>>>> quite perfectly.
>>>> David Ponzone ?Direction Technique
>>>> email: david.ponzone at ipeva.fr
>>>> tel: ? ? ?01 74 03 18 97
>>>> gsm: ? 06 66 98 76 34
>>>> Service Client?IPeva
>>>> tel: ? ? ?0811 46 26 26
>>>> www.ipeva.fr? -? ?www.ipeva-studio.com
>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>
>>>>
>>>>
>>>> Le 09/02/2011 ? 05:41, Victor Chukalovskiy a ?crit :
>>>>
>>>> This didn't help.
>>>> I reloaded mod_com_g729, tried g729_info and the same "can't contact
>>>> licence server" error
>>>> -Victor
>>>>
>>>> On 08/02/11 11:10 PM, Michael Collins wrote:
>>>>
>>>> put the license file in /etc/freeswitch
>>>> -MC
>>>>
>>>> On Tue, Feb 8, 2011 at 9:56 PM, Victor Chukalovskiy
>>>> <victor.chukalovskiy at utoronto.ca> wrote:
>>>>>
>>>>> Hi Michael and others,
>>>>>
>>>>> After updating to the latest GIT mod_com_g729 loads successfully.
>>>>>
>>>>> Next problem appears:
>>>>> when typing "g729_info" freeswitch replies "can't contact licence
>>>>> server".
>>>>> g729_available gives "False"
>>>>>
>>>>> How to solve this?
>>>>>
>>>>> I have my xxxxx.conf license file in the root of FS install directory
>>>>> /opt/fs
>>>>> Should it be placed elsewhere?
>>>>>
>>>>> Also, xxxxx.conf was created with previous non-successful install of
>>>>> mod_com_g729
>>>>> Should I run validator again with the same license key?
>>>>>
>>>>> Thank you,
>>>>> Victor
>>>>>
>>>>> On 07/02/11 10:36 AM, Michael Collins wrote:
>>>>>
>>>>> Any chance you can update to latest git? Your life will be easier.
>>>>> There have been notable improvements in FS in the past few months.
>>>>> -MC
>>>>> On Sat, Feb 5, 2011 at 3:30 PM, Victor Chukalovskiy
>>>>> <victor.chukalovskiy at utoronto.ca> wrote:
>>>>>>
>>>>>> Hello,
>>>>>>
>>>>>> After purchasing a few licenses and installing the latest
>>>>>> fsg729-191-installer
>>>>>> I'm getting the following error when trying to load the mod_com_g729:
>>>>>> > 2011-02-04 14:27:17.637046 [CRIT] switch_loadable_module.c:926 Error
>>>>>> > Loading module /opt/fs/mod/mod_com_g729.so
>>>>>> > **Trying to load an out of date module, please rebuild the module.**
>>>>>>
>>>>>> Also noticed that g729 installer ran with a couple errors:
>>>>>> > ./installer: line 62: ldconfig: command not found
>>>>>> > ./installer: line 49: useradd: command not found
>>>>>> ?Any help or solution is much appreciated.
>>>>>>
>>>>>> -Victor
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 david.ponzone at ipeva.fr  Wed Feb  9 19:18:02 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 9 Feb 2011 17:18:02 +0100
Subject: [Freeswitch-users] Re-order or Invalid
In-Reply-To: <056601cbc874$b3341e90$199c5bb0$@com>
References: <056601cbc874$b3341e90$199c5bb0$@com>
Message-ID: <9FB7DDC0-885B-47E8-AD27-47F15B736743@ipeva.fr>

I think you are supposed to send back a SIP 404.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 09/02/2011 ? 17:16, Peder a ?crit :

> What is the correct way to send a re-order or invalid message/tone to a
> caller?  We have a couple hundred DIDs and quite a few of them are not used.
> If someone calls into one of those, we just want to send a re-order or busy
> tone.  My preference would be to NOT answer the call as I would prefer not
> to be charged for the call if I can avoid it.
> 
> Peder
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb  9 19:19:06 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 9 Feb 2011 10:19:06 -0600
Subject: [Freeswitch-users] Re-order or Invalid
In-Reply-To: <056601cbc874$b3341e90$199c5bb0$@com>
References: <056601cbc874$b3341e90$199c5bb0$@com>
Message-ID: <AANLkTin0KMy_Z3mV8FG9mDHRUKF_FsrfkLL0_C+SxTos@mail.gmail.com>

<action application="playback"
data="tone_stream://%(330,15,950);%(330,15,1400);%(330,1000,1800);"/>
<action application="playback" data="invalid.wav"/>
<action application="hangup"/>


On Wed, Feb 9, 2011 at 10:16 AM, Peder <peder at networkoblivion.com> wrote:
> What is the correct way to send a re-order or invalid message/tone to a
> caller? ?We have a couple hundred DIDs and quite a few of them are not used.
> If someone calls into one of those, we just want to send a re-order or busy
> tone. ?My preference would be to NOT answer the call as I would prefer not
> to be charged for the call if I can avoid it.
>
> Peder
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 peder at networkoblivion.com  Wed Feb  9 19:34:40 2011
From: peder at networkoblivion.com (Peder)
Date: Wed, 9 Feb 2011 10:34:40 -0600
Subject: [Freeswitch-users] Re-order or Invalid
In-Reply-To: <9FB7DDC0-885B-47E8-AD27-47F15B736743@ipeva.fr>
References: <056601cbc874$b3341e90$199c5bb0$@com>
	<9FB7DDC0-885B-47E8-AD27-47F15B736743@ipeva.fr>
Message-ID: <057d01cbc877$400abca0$c02035e0$@com>

OK, how do you do that?

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Ponzone
Sent: Wednesday, February 09, 2011 10:18 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Re-order or Invalid

 

I think you are supposed to send back a SIP 404.

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/> 

 

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.

 





 

Le 09/02/2011 ? 17:16, Peder a ?crit :





What is the correct way to send a re-order or invalid message/tone to a
caller?  We have a couple hundred DIDs and quite a few of them are not used.
If someone calls into one of those, we just want to send a re-order or busy
tone.  My preference would be to NOT answer the call as I would prefer not
to be charged for the call if I can avoid it.

Peder



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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.ponzone at ipeva.fr  Wed Feb  9 19:47:51 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 9 Feb 2011 17:47:51 +0100
Subject: [Freeswitch-users] Re-order or Invalid
In-Reply-To: <057d01cbc877$400abca0$c02035e0$@com>
References: <056601cbc874$b3341e90$199c5bb0$@com>
	<9FB7DDC0-885B-47E8-AD27-47F15B736743@ipeva.fr>
	<057d01cbc877$400abca0$c02035e0$@com>
Message-ID: <875BD817-4009-49A7-B511-4AAE4B1345A4@ipeva.fr>

Add in your dialplan for the concerned number:

        <action application="respond" data="404"/>
        <action application="hangup" data="UNALLOCATED_NUMBER"/>

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 09/02/2011 ? 17:34, Peder a ?crit :

> OK, how do you do that?
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Ponzone
> Sent: Wednesday, February 09, 2011 10:18 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Re-order or Invalid
>  
> I think you are supposed to send back a SIP 404.
>  
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>  
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>  
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>  
> 
> 
>  
> Le 09/02/2011 ? 17:16, Peder a ?crit :
> 
> 
> What is the correct way to send a re-order or invalid message/tone to a
> caller?  We have a couple hundred DIDs and quite a few of them are not used.
> If someone calls into one of those, we just want to send a re-order or busy
> tone.  My preference would be to NOT answer the call as I would prefer not
> to be charged for the call if I can avoid it.
> 
> Peder
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Feb  9 19:56:25 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Wed, 09 Feb 2011 11:56:25 -0500
Subject: [Freeswitch-users] Best way to implement real-time ivr from mysql
Message-ID: <4D52C739.8080502@communicatefreely.net>

Hello list,

I'm trying to implement a flexible, but easy to administer IVR system 
for our multi-tenant PBX.

My goal is to be able to build a flexible auto-attendent scheme for 
incoming calls, based on data in a mysql database.

The functionality I need is:
    -Different options based on time of day
    -Playback of 1-2 audio files while listening for digits (Thank you 
for calling abc company)
    -Single digit options set some variables, then transfer to a defined 
extension (press 1 for sales)
    -Multi-digit options are checked against a pattern, then a transfer 
is executed (enter the extension now ...)

Not very complicated, I know, but making it scale is tricky.

I have tried LUA, but LuaSQL has issues.  Even using ODBC, I still get 
memory leaks and random errors with file handles etc.

For the most part, I haven't had any issues with xml_curl getting config 
from another server that generates it with PHP.  I'm exploring this, but 
it looks like I'll have to implement the following logic:
    -Call comes in, and a dialplan is returned that plays the greetings, 
sets some variables, and does a play_and_get_digits, followed by a 
transfer back to the dialplan, so we can figure out what to do based on 
the digits.
    -The callback to the dialplan evaluates the dtmf presented, and 
decides what action to take, returning that in another dialplan piece.

I think this could work. but it means that a dialplan lookup has to be 
done each time someone makes a selection.

Is there a better way, or does that make the most sense?

Thanks!

-Tim


From u2nsam at gmail.com  Wed Feb  9 20:01:30 2011
From: u2nsam at gmail.com (Sam)
Date: Wed, 9 Feb 2011 22:31:30 +0530
Subject: [Freeswitch-users] RTP keep alive
In-Reply-To: <AANLkTi=0Uo-7qqWfB9uDFownX7TuVzsez3aVnUGMORH4@mail.gmail.com>
References: <AANLkTik-o+kpsLyTTtjL09Zg4M9a23aPygFj1d-V=cTb@mail.gmail.com>
	<AANLkTi=aUeWAo+54O3U_LAK1kEr-U3wTNty+4AS5HG42@mail.gmail.com>
	<AANLkTi=0Uo-7qqWfB9uDFownX7TuVzsez3aVnUGMORH4@mail.gmail.com>
Message-ID: <AANLkTinSJ4XPnJvzFfALA8tKwFxnRiq0P8JFk+Uz1kV6@mail.gmail.com>

There is no vad , is there something like rtp keep live after the increase
of rtp timout .
Yes the RTP timeout helps.

Regds
Sam

On Wed, Feb 9, 2011 at 8:03 PM, Steven Ayre <steveayre at gmail.com> wrote:

> That'll do it.
>
> VAD can mean no RTP is transmitted legitimately though, so be careful.
> Especially when on hold when the phone might stop sending anything (I see
> you've already increased that time).
>
> -Steve
>
>
> On 9 February 2011 08:49, Sam <u2nsam at gmail.com> wrote:
>
>> I have this in the settings,
>>
>>     <param name="rtp-timeout-sec" value="3000"/>
>>     <param name="rtp-hold-timeout-sec" value="18000"/>
>>
>> Any thing i need more regarding that ?
>>
>> Regards
>> Sam
>>
>>
>> On Wed, Feb 9, 2011 at 1:35 PM, Sam <u2nsam at gmail.com> wrote:
>>
>>> Hello,
>>>
>>> We have a situation wherein we need to keep alive RTP ,is there any
>>> parameter to do that,
>>> because when someone is on long conversation and not talking fro brief
>>> duration the call
>>> disconnects .
>>>
>>> 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 msc at freeswitch.org  Wed Feb  9 20:02:28 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 9 Feb 2011 11:02:28 -0600
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <AANLkTikkXQaP5Y5p75ou0HE39hNk4areMsAhhyLEtnY1@mail.gmail.com>

Hey gang,

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

-Michael
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From steveayre at gmail.com  Wed Feb  9 20:04:01 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 9 Feb 2011 17:04:01 +0000
Subject: [Freeswitch-users] RTP keep alive
In-Reply-To: <AANLkTinSJ4XPnJvzFfALA8tKwFxnRiq0P8JFk+Uz1kV6@mail.gmail.com>
References: <AANLkTik-o+kpsLyTTtjL09Zg4M9a23aPygFj1d-V=cTb@mail.gmail.com>
	<AANLkTi=aUeWAo+54O3U_LAK1kEr-U3wTNty+4AS5HG42@mail.gmail.com>
	<AANLkTi=0Uo-7qqWfB9uDFownX7TuVzsez3aVnUGMORH4@mail.gmail.com>
	<AANLkTinSJ4XPnJvzFfALA8tKwFxnRiq0P8JFk+Uz1kV6@mail.gmail.com>
Message-ID: <AANLkTinhLp=RkQVuKM9ksXWx8=VUOMBESQ=5qtU758c+@mail.gmail.com>

In media, there's only the possibility of detecting absence of it. Not sure
if RTCP would contain something, but FS doesn't support support it and
neither do all endpoints.

The alternative (probably preferred) method is session invites, which checks
the call is still up through the SIP signalling.

-Steve



On 9 February 2011 17:01, Sam <u2nsam at gmail.com> wrote:

> There is no vad , is there something like rtp keep live after the increase
> of rtp timout .
> Yes the RTP timeout helps.
>
> Regds
> Sam
>
>
> On Wed, Feb 9, 2011 at 8:03 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> That'll do it.
>>
>> VAD can mean no RTP is transmitted legitimately though, so be careful.
>> Especially when on hold when the phone might stop sending anything (I see
>> you've already increased that time).
>>
>> -Steve
>>
>>
>> On 9 February 2011 08:49, Sam <u2nsam at gmail.com> wrote:
>>
>>>  I have this in the settings,
>>>
>>>     <param name="rtp-timeout-sec" value="3000"/>
>>>     <param name="rtp-hold-timeout-sec" value="18000"/>
>>>
>>> Any thing i need more regarding that ?
>>>
>>> Regards
>>> Sam
>>>
>>>
>>> On Wed, Feb 9, 2011 at 1:35 PM, Sam <u2nsam at gmail.com> wrote:
>>>
>>>> Hello,
>>>>
>>>> We have a situation wherein we need to keep alive RTP ,is there any
>>>> parameter to do that,
>>>> because when someone is on long conversation and not talking fro brief
>>>> duration the call
>>>> disconnects .
>>>>
>>>> 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 victor.chukalovskiy at utoronto.ca  Wed Feb  9 20:05:21 2011
From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy)
Date: Wed, 09 Feb 2011 12:05:21 -0500
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <AANLkTi==uogdP7q2Nsqh7rcoaMy6tvy1at8kcNSqS=1q@mail.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>
Message-ID: <4D52C951.5080600@utoronto.ca>

Hi Folks,

Thank you everyone. Solved by lunching server separately.

-Victor

On 09/02/11 11:17 AM, Anthony Minessale wrote:
> try killing it.
> you are probably still running the old one from the first time you tried it.
>
>
> On Wed, Feb 9, 2011 at 9:59 AM, Steven Ayre<steveayre at gmail.com>  wrote:
>>> Is there any other way to (re)-start this licensing server?
>> 1) I think it runs when you load mod_com_g729
>> 2) Run it manually from command line might work
>>
>> -Steve
>>
>>
>>
>> On 9 February 2011 15:46, Victor Chukalovskiy
>> <victor.chukalovskiy at utoronto.ca>  wrote:
>>> Yes, the module is installed correctly. It loads Ok.
>>>
>>> I re-ran sudo ./fsg729-191-installer.
>>> license server is indeed installed in /usr/sbin. Would be better if
>>> installer puts it into FS install path - just where it belongs.
>>>
>>> It was root:root owned, so I chown'ed it to the correct user fs:fs (this
>>> is how I run freeswitch)
>>>
>>> Should it help? I'm not able to restart FS until tonight.
>>> Is there any other way to (re)-start this licensing server?
>>>
>>>
>>> -Victor
>>>
>>> On 09/02/11 10:02 AM, David Ponzone wrote:
>>>
>>> Steven,
>>> I just checked and my license server was installed in /usr/sbin.
>>> So it should not be an issue.
>>> Or perhaps the installer looks for freeswtich in standard paths, and it
>>> didn't find it, it doesn't install the license server.
>>> But the module was installed correctly, wasn't it, Victor ?
>>> David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>
>>>
>>>
>>> Le 09/02/2011 ? 15:24, Steven Ayre a ?crit :
>>>
>>> I see you're in a nonstandard directory.
>>>
>>> As far as I know the license server only likes being in
>>> /usr/local/freeswitch/bin or /opt/freeswitch/bin.
>>>
>>> Because it's neither of those it looks like the installer hasn't installed
>>> the license server.
>>>
>>> -Steve
>>>
>>>
>>> On 9 February 2011 14:12, Victor Chukalovskiy
>>> <victor.chukalovskiy at utoronto.ca>  wrote:
>>>> Hi Steve,
>>>>
>>>> I'm running as user "fs":
>>>> /opt/fs/bin/freeswitch -nf -nc -u fs -g fs -core -nonat
>>>>
>>>> I have following binaries in the directory:
>>>> freeswitch  fs_cli  fs_encode  fs_ivrd  fsxs  gentls_cert  tone2wav
>>>> validator
>>>> Is any of them the required binary? Where is it launched from?
>>>>
>>>> Thank you,
>>>> Victor
>>>>
>>>> On 09/02/11 09:02 AM, Steven Ayre wrote:
>>>>
>>>> What user do you run FS as?
>>>>
>>>> It's anothe program in the same bin folder as freeswitch, and should run
>>>> as the same user as freeswitch itself. There's no logging I'm aware of.
>>>>
>>>> -Steve
>>>>
>>>>
>>>> On 9 February 2011 13:25, Victor Chukalovskiy
>>>> <victor.chukalovskiy at utoronto.ca>  wrote:
>>>>> Thanks David,
>>>>>
>>>>> Freeswitch was restarted.
>>>>> Still "can't contact licence server"
>>>>>
>>>>> Is this license server a process / module running on the same machine?
>>>>> If so, does it log anything anywhere? Where is it started from?
>>>>>
>>>>> -Victor
>>>>>
>>>>> On 09/02/11 08:01 AM, David Ponzone wrote:
>>>>>
>>>>> The license server is normally launched automatically.
>>>>> If you can, restart FreeSWITCH.
>>>>> It's normally not required.
>>>>> David Ponzone  Direction Technique
>>>>> email: david.ponzone at ipeva.fr
>>>>> tel:      01 74 03 18 97
>>>>> gsm:   06 66 98 76 34
>>>>> Service Client IPeva
>>>>> tel:      0811 46 26 26
>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>
>>>>>
>>>>>
>>>>> Le 09/02/2011 ? 13:40, Victor Chukalovskiy a ?crit :
>>>>>
>>>>> Hi David,
>>>>>
>>>>> I'm aware of the file and was using it for install.
>>>>>
>>>>> There is no single word there what "licence server" is
>>>>> and how to deal with "can't contact licence server".
>>>>>
>>>>> Thank you,
>>>>> Victor
>>>>>
>>>>> On 09/02/11 03:55 AM, David Ponzone wrote:
>>>>>
>>>>> There is a file coming with the module which explains the installation
>>>>> quite perfectly.
>>>>> David Ponzone  Direction Technique
>>>>> email: david.ponzone at ipeva.fr
>>>>> tel:      01 74 03 18 97
>>>>> gsm:   06 66 98 76 34
>>>>> Service Client IPeva
>>>>> tel:      0811 46 26 26
>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>
>>>>>
>>>>>
>>>>> Le 09/02/2011 ? 05:41, Victor Chukalovskiy a ?crit :
>>>>>
>>>>> This didn't help.
>>>>> I reloaded mod_com_g729, tried g729_info and the same "can't contact
>>>>> licence server" error
>>>>> -Victor
>>>>>
>>>>> On 08/02/11 11:10 PM, Michael Collins wrote:
>>>>>
>>>>> put the license file in /etc/freeswitch
>>>>> -MC
>>>>>
>>>>> On Tue, Feb 8, 2011 at 9:56 PM, Victor Chukalovskiy
>>>>> <victor.chukalovskiy at utoronto.ca>  wrote:
>>>>>> Hi Michael and others,
>>>>>>
>>>>>> After updating to the latest GIT mod_com_g729 loads successfully.
>>>>>>
>>>>>> Next problem appears:
>>>>>> when typing "g729_info" freeswitch replies "can't contact licence
>>>>>> server".
>>>>>> g729_available gives "False"
>>>>>>
>>>>>> How to solve this?
>>>>>>
>>>>>> I have my xxxxx.conf license file in the root of FS install directory
>>>>>> /opt/fs
>>>>>> Should it be placed elsewhere?
>>>>>>
>>>>>> Also, xxxxx.conf was created with previous non-successful install of
>>>>>> mod_com_g729
>>>>>> Should I run validator again with the same license key?
>>>>>>
>>>>>> Thank you,
>>>>>> Victor
>>>>>>
>>>>>> On 07/02/11 10:36 AM, Michael Collins wrote:
>>>>>>
>>>>>> Any chance you can update to latest git? Your life will be easier.
>>>>>> There have been notable improvements in FS in the past few months.
>>>>>> -MC
>>>>>> On Sat, Feb 5, 2011 at 3:30 PM, Victor Chukalovskiy
>>>>>> <victor.chukalovskiy at utoronto.ca>  wrote:
>>>>>>> Hello,
>>>>>>>
>>>>>>> After purchasing a few licenses and installing the latest
>>>>>>> fsg729-191-installer
>>>>>>> I'm getting the following error when trying to load the mod_com_g729:
>>>>>>>> 2011-02-04 14:27:17.637046 [CRIT] switch_loadable_module.c:926 Error
>>>>>>>> Loading module /opt/fs/mod/mod_com_g729.so
>>>>>>>> **Trying to load an out of date module, please rebuild the module.**
>>>>>>> Also noticed that g729 installer ran with a couple errors:
>>>>>>>> ./installer: line 62: ldconfig: command not found
>>>>>>>> ./installer: line 49: useradd: command not found
>>>>>>>   Any help or solution is much appreciated.
>>>>>>>
>>>>>>> -Victor
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 david.ponzone at ipeva.fr  Wed Feb  9 20:05:22 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 9 Feb 2011 18:05:22 +0100
Subject: [Freeswitch-users] Best way to implement real-time ivr from
	mysql
In-Reply-To: <4D52C739.8080502@communicatefreely.net>
References: <4D52C739.8080502@communicatefreely.net>
Message-ID: <BE42DE63-CA66-40C8-A0BF-17AE303C77B1@ipeva.fr>

Tim,

which ODBC is that ? the one integrated in FreeSWITCH that you can call from LUA with freeswitch.Dbh() ?
I would say that solving those issues would be a better way to achieve this, for you and for everyone around.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 09/02/2011 ? 17:56, Tim St. Pierre a ?crit :

> Hello list,
> 
> I'm trying to implement a flexible, but easy to administer IVR system 
> for our multi-tenant PBX.
> 
> My goal is to be able to build a flexible auto-attendent scheme for 
> incoming calls, based on data in a mysql database.
> 
> The functionality I need is:
>    -Different options based on time of day
>    -Playback of 1-2 audio files while listening for digits (Thank you 
> for calling abc company)
>    -Single digit options set some variables, then transfer to a defined 
> extension (press 1 for sales)
>    -Multi-digit options are checked against a pattern, then a transfer 
> is executed (enter the extension now ...)
> 
> Not very complicated, I know, but making it scale is tricky.
> 
> I have tried LUA, but LuaSQL has issues.  Even using ODBC, I still get 
> memory leaks and random errors with file handles etc.
> 
> For the most part, I haven't had any issues with xml_curl getting config 
> from another server that generates it with PHP.  I'm exploring this, but 
> it looks like I'll have to implement the following logic:
>    -Call comes in, and a dialplan is returned that plays the greetings, 
> sets some variables, and does a play_and_get_digits, followed by a 
> transfer back to the dialplan, so we can figure out what to do based on 
> the digits.
>    -The callback to the dialplan evaluates the dtmf presented, and 
> decides what action to take, returning that in another dialplan piece.
> 
> I think this could work. but it means that a dialplan lookup has to be 
> done each time someone makes a selection.
> 
> Is there a better way, or does that make the most sense?
> 
> Thanks!
> 
> -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

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From peder at networkoblivion.com  Wed Feb  9 20:08:13 2011
From: peder at networkoblivion.com (Peder)
Date: Wed, 9 Feb 2011 11:08:13 -0600
Subject: [Freeswitch-users] Re-order or Invalid
In-Reply-To: <875BD817-4009-49A7-B511-4AAE4B1345A4@ipeva.fr>
References: <056601cbc874$b3341e90$199c5bb0$@com>	<9FB7DDC0-885B-47E8-AD27-47F15B736743@ipeva.fr>	<057d01cbc877$400abca0$c02035e0$@com>
	<875BD817-4009-49A7-B511-4AAE4B1345A4@ipeva.fr>
Message-ID: <059901cbc87b$ef6afd50$ce40f7f0$@com>

Apparently icall doesn?t accept a 404 back as they just keep sending me the call over and over again?.  It appears I have to answer and play tones and a message myself.  For anyone who cares, here is what I have.

 

 

  <extension name="ALLOTHER">

    <condition field="destination_number" expression="^(\d{10})$">

        <action application="answer"/>

        <action application="playback" data="tone_stream://%(330,15,950);%(330,15,1400);%(330,1000,1800);"/>

        <action application="playback" data="misc/8000/invalid_extension.wav"/>

        <action application="hangup"/>

    </condition>

  </extension>

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Ponzone
Sent: Wednesday, February 09, 2011 10:48 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Re-order or Invalid

 

Add in your dialplan for the concerned number:

 

        <action application="respond" data="404"/>

        <action application="hangup" data="UNALLOCATED_NUMBER"/>

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/> 

 

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.

 

 

 

Le 09/02/2011 ? 17:34, Peder a ?crit :





OK, how do you do that?

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Ponzone
Sent: Wednesday, February 09, 2011 10:18 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Re-order or Invalid

 

I think you are supposed to send back a SIP 404.

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/> 

 

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.

 






 

Le 09/02/2011 ? 17:16, Peder a ?crit :






What is the correct way to send a re-order or invalid message/tone to a
caller?  We have a couple hundred DIDs and quite a few of them are not used.
If someone calls into one of those, we just want to send a re-order or busy
tone.  My preference would be to NOT answer the call as I would prefer not
to be charged for the call if I can avoid it.

Peder



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

 

_______________________________________________
FreeSWITCH-users mailing 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  Wed Feb  9 20:09:13 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 9 Feb 2011 11:09:13 -0600
Subject: [Freeswitch-users] Re-order or Invalid
In-Reply-To: <059901cbc87b$ef6afd50$ce40f7f0$@com>
References: <056601cbc874$b3341e90$199c5bb0$@com>
	<9FB7DDC0-885B-47E8-AD27-47F15B736743@ipeva.fr>
	<057d01cbc877$400abca0$c02035e0$@com>
	<875BD817-4009-49A7-B511-4AAE4B1345A4@ipeva.fr>
	<059901cbc87b$ef6afd50$ce40f7f0$@com>
Message-ID: <AANLkTim=xNi1v0-gn_BWwYNezcV6ur4=jdTG4m=awP7p@mail.gmail.com>

you should be able to pre_answer instead? and not actually get billed.


On Wed, Feb 9, 2011 at 11:08 AM, Peder <peder at networkoblivion.com> wrote:
> Apparently icall doesn?t accept a 404 back as they just keep sending me the
> call over and over again?.? It appears I have to answer and play tones and a
> message myself.? For anyone who cares, here is what I have.
>
>
>
>
>
> ? <extension name="ALLOTHER">
>
> ??? <condition field="destination_number" expression="^(\d{10})$">
>
> ??????? <action application="answer"/>
>
> ??????? <action application="playback"
> data="tone_stream://%(330,15,950);%(330,15,1400);%(330,1000,1800);"/>
>
> ??????? <action application="playback"
> data="misc/8000/invalid_extension.wav"/>
>
> ??????? <action application="hangup"/>
>
> ??? </condition>
>
> ? </extension>
>
>
>
>
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David
> Ponzone
> Sent: Wednesday, February 09, 2011 10:48 AM
>
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Re-order or Invalid
>
>
>
> Add in your dialplan for the concerned number:
>
>
>
> ?? ? ? ?<action application="respond" data="404"/>
>
> ?? ? ? ?<action application="hangup" data="UNALLOCATED_NUMBER"/>
>
>
>
> David Ponzone ?Direction Technique
>
> email: david.ponzone at ipeva.fr
>
> tel: ? ? ?01 74 03 18 97
>
> gsm: ? 06 66 98 76 34
>
>
>
> Service Client?IPeva
>
> tel: ? ? ?0811 46 26 26
>
> www.ipeva.fr? -? ?www.ipeva-studio.com
>
>
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
>
>
>
> Le 09/02/2011 ? 17:34, Peder a ?crit :
>
> OK, how do you do that?
>
>
>
> From:?freeswitch-users-bounces at lists.freeswitch.org?[mailto:freeswitch-users-bounces at lists.freeswitch.org]?On
> Behalf Of?David Ponzone
> Sent:?Wednesday, February 09, 2011 10:18 AM
> To:?FreeSWITCH Users Help
> Subject:?Re: [Freeswitch-users] Re-order or Invalid
>
>
>
> I think you are supposed to send back a SIP 404.
>
>
>
> David Ponzone ?Direction Technique
>
> email:?david.ponzone at ipeva.fr
>
> tel: ? ? ?01 74 03 18 97
>
> gsm: ? 06 66 98 76 34
>
>
>
> Service Client?IPeva
>
> tel: ? ? ?0811 46 26 26
>
> www.ipeva.fr? -? ?www.ipeva-studio.com
>
>
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
>
>
> Le 09/02/2011 ? 17:16, Peder a ?crit :
>
>
> What is the correct way to send a re-order or invalid message/tone to a
> caller? ?We have a couple hundred DIDs and quite a few of them are not used.
> If someone calls into one of those, we just want to send a re-order or busy
> tone. ?My preference would be to NOT answer the call as I would prefer not
> to be charged for the call if I can avoid it.
>
> Peder
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 david.ponzone at ipeva.fr  Wed Feb  9 20:10:38 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 9 Feb 2011 18:10:38 +0100
Subject: [Freeswitch-users] RTP keep alive
In-Reply-To: <AANLkTinSJ4XPnJvzFfALA8tKwFxnRiq0P8JFk+Uz1kV6@mail.gmail.com>
References: <AANLkTik-o+kpsLyTTtjL09Zg4M9a23aPygFj1d-V=cTb@mail.gmail.com>
	<AANLkTi=aUeWAo+54O3U_LAK1kEr-U3wTNty+4AS5HG42@mail.gmail.com>
	<AANLkTi=0Uo-7qqWfB9uDFownX7TuVzsez3aVnUGMORH4@mail.gmail.com>
	<AANLkTinSJ4XPnJvzFfALA8tKwFxnRiq0P8JFk+Uz1kV6@mail.gmail.com>
Message-ID: <25A5DCAF-6F3D-4689-8FA2-B282BA6B3235@ipeva.fr>

Sam,

No there is no such thing as RTP keep-alive, AFAIK.

But again, if you don't use VAD, there should be RTP all the time (except when on hold).
So the first thing you should ask yourself is why is this RTP stream missing.
Also you did not provide any information about the setup and the possible involvement of another equipement (provider ?).
Be aware that if G729 is negotiated, it may use VAD by default if it was not disabled explicitly (that's the RFC).

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 09/02/2011 ? 18:01, Sam a ?crit :

> There is no vad , is there something like rtp keep live after the increase of rtp timout .
> Yes the RTP timeout helps.
> 
> Regds
> Sam
> 
> On Wed, Feb 9, 2011 at 8:03 PM, Steven Ayre <steveayre at gmail.com> wrote:
> That'll do it.
> 
> VAD can mean no RTP is transmitted legitimately though, so be careful. Especially when on hold when the phone might stop sending anything (I see you've already increased that time).
> 
> -Steve
> 
> 
> On 9 February 2011 08:49, Sam <u2nsam at gmail.com> wrote:
> I have this in the settings,
> 
>     <param name="rtp-timeout-sec" value="3000"/>
>     <param name="rtp-hold-timeout-sec" value="18000"/>
> 
> Any thing i need more regarding that ?
> 
> Regards
> Sam
> 
> 
> On Wed, Feb 9, 2011 at 1:35 PM, Sam <u2nsam at gmail.com> wrote:
> Hello,
> 
> We have a situation wherein we need to keep alive RTP ,is there any parameter to do that,
> because when someone is on long conversation and not talking fro brief duration the call
> disconnects .
> 
> 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 david.ponzone at ipeva.fr  Wed Feb  9 20:11:03 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 9 Feb 2011 18:11:03 +0100
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <4D52C951.5080600@utoronto.ca>
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>
Message-ID: <1399CAC8-B278-45F0-96BD-D88EEF2CF77E@ipeva.fr>

Victor,

you should not eat your server, that can be bad for health.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 09/02/2011 ? 18:05, Victor Chukalovskiy a ?crit :

> Hi Folks,
> 
> Thank you everyone. Solved by lunching server separately.
> 
> -Victor
> 
> On 09/02/11 11:17 AM, Anthony Minessale wrote:
>> try killing it.
>> you are probably still running the old one from the first time you tried it.
>> 
>> 
>> On Wed, Feb 9, 2011 at 9:59 AM, Steven Ayre<steveayre at gmail.com>  wrote:
>>>> Is there any other way to (re)-start this licensing server?
>>> 1) I think it runs when you load mod_com_g729
>>> 2) Run it manually from command line might work
>>> 
>>> -Steve
>>> 
>>> 
>>> 
>>> On 9 February 2011 15:46, Victor Chukalovskiy
>>> <victor.chukalovskiy at utoronto.ca>  wrote:
>>>> Yes, the module is installed correctly. It loads Ok.
>>>> 
>>>> I re-ran sudo ./fsg729-191-installer.
>>>> license server is indeed installed in /usr/sbin. Would be better if
>>>> installer puts it into FS install path - just where it belongs.
>>>> 
>>>> It was root:root owned, so I chown'ed it to the correct user fs:fs (this
>>>> is how I run freeswitch)
>>>> 
>>>> Should it help? I'm not able to restart FS until tonight.
>>>> Is there any other way to (re)-start this licensing server?
>>>> 
>>>> 
>>>> -Victor
>>>> 
>>>> On 09/02/11 10:02 AM, David Ponzone wrote:
>>>> 
>>>> Steven,
>>>> I just checked and my license server was installed in /usr/sbin.
>>>> So it should not be an issue.
>>>> Or perhaps the installer looks for freeswtich in standard paths, and it
>>>> didn't find it, it doesn't install the license server.
>>>> But the module was installed correctly, wasn't it, Victor ?
>>>> David Ponzone  Direction Technique
>>>> email: david.ponzone at ipeva.fr
>>>> tel:      01 74 03 18 97
>>>> gsm:   06 66 98 76 34
>>>> Service Client IPeva
>>>> tel:      0811 46 26 26
>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>> 
>>>> 
>>>> 
>>>> Le 09/02/2011 ? 15:24, Steven Ayre a ?crit :
>>>> 
>>>> I see you're in a nonstandard directory.
>>>> 
>>>> As far as I know the license server only likes being in
>>>> /usr/local/freeswitch/bin or /opt/freeswitch/bin.
>>>> 
>>>> Because it's neither of those it looks like the installer hasn't installed
>>>> the license server.
>>>> 
>>>> -Steve
>>>> 
>>>> 
>>>> On 9 February 2011 14:12, Victor Chukalovskiy
>>>> <victor.chukalovskiy at utoronto.ca>  wrote:
>>>>> Hi Steve,
>>>>> 
>>>>> I'm running as user "fs":
>>>>> /opt/fs/bin/freeswitch -nf -nc -u fs -g fs -core -nonat
>>>>> 
>>>>> I have following binaries in the directory:
>>>>> freeswitch  fs_cli  fs_encode  fs_ivrd  fsxs  gentls_cert  tone2wav
>>>>> validator
>>>>> Is any of them the required binary? Where is it launched from?
>>>>> 
>>>>> Thank you,
>>>>> Victor
>>>>> 
>>>>> On 09/02/11 09:02 AM, Steven Ayre wrote:
>>>>> 
>>>>> What user do you run FS as?
>>>>> 
>>>>> It's anothe program in the same bin folder as freeswitch, and should run
>>>>> as the same user as freeswitch itself. There's no logging I'm aware of.
>>>>> 
>>>>> -Steve
>>>>> 
>>>>> 
>>>>> On 9 February 2011 13:25, Victor Chukalovskiy
>>>>> <victor.chukalovskiy at utoronto.ca>  wrote:
>>>>>> Thanks David,
>>>>>> 
>>>>>> Freeswitch was restarted.
>>>>>> Still "can't contact licence server"
>>>>>> 
>>>>>> Is this license server a process / module running on the same machine?
>>>>>> If so, does it log anything anywhere? Where is it started from?
>>>>>> 
>>>>>> -Victor
>>>>>> 
>>>>>> On 09/02/11 08:01 AM, David Ponzone wrote:
>>>>>> 
>>>>>> The license server is normally launched automatically.
>>>>>> If you can, restart FreeSWITCH.
>>>>>> It's normally not required.
>>>>>> David Ponzone  Direction Technique
>>>>>> email: david.ponzone at ipeva.fr
>>>>>> tel:      01 74 03 18 97
>>>>>> gsm:   06 66 98 76 34
>>>>>> Service Client IPeva
>>>>>> tel:      0811 46 26 26
>>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>>>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Le 09/02/2011 ? 13:40, Victor Chukalovskiy a ?crit :
>>>>>> 
>>>>>> Hi David,
>>>>>> 
>>>>>> I'm aware of the file and was using it for install.
>>>>>> 
>>>>>> There is no single word there what "licence server" is
>>>>>> and how to deal with "can't contact licence server".
>>>>>> 
>>>>>> Thank you,
>>>>>> Victor
>>>>>> 
>>>>>> On 09/02/11 03:55 AM, David Ponzone wrote:
>>>>>> 
>>>>>> There is a file coming with the module which explains the installation
>>>>>> quite perfectly.
>>>>>> David Ponzone  Direction Technique
>>>>>> email: david.ponzone at ipeva.fr
>>>>>> tel:      01 74 03 18 97
>>>>>> gsm:   06 66 98 76 34
>>>>>> Service Client IPeva
>>>>>> tel:      0811 46 26 26
>>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>>>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Le 09/02/2011 ? 05:41, Victor Chukalovskiy a ?crit :
>>>>>> 
>>>>>> This didn't help.
>>>>>> I reloaded mod_com_g729, tried g729_info and the same "can't contact
>>>>>> licence server" error
>>>>>> -Victor
>>>>>> 
>>>>>> On 08/02/11 11:10 PM, Michael Collins wrote:
>>>>>> 
>>>>>> put the license file in /etc/freeswitch
>>>>>> -MC
>>>>>> 
>>>>>> On Tue, Feb 8, 2011 at 9:56 PM, Victor Chukalovskiy
>>>>>> <victor.chukalovskiy at utoronto.ca>  wrote:
>>>>>>> Hi Michael and others,
>>>>>>> 
>>>>>>> After updating to the latest GIT mod_com_g729 loads successfully.
>>>>>>> 
>>>>>>> Next problem appears:
>>>>>>> when typing "g729_info" freeswitch replies "can't contact licence
>>>>>>> server".
>>>>>>> g729_available gives "False"
>>>>>>> 
>>>>>>> How to solve this?
>>>>>>> 
>>>>>>> I have my xxxxx.conf license file in the root of FS install directory
>>>>>>> /opt/fs
>>>>>>> Should it be placed elsewhere?
>>>>>>> 
>>>>>>> Also, xxxxx.conf was created with previous non-successful install of
>>>>>>> mod_com_g729
>>>>>>> Should I run validator again with the same license key?
>>>>>>> 
>>>>>>> Thank you,
>>>>>>> Victor
>>>>>>> 
>>>>>>> On 07/02/11 10:36 AM, Michael Collins wrote:
>>>>>>> 
>>>>>>> Any chance you can update to latest git? Your life will be easier.
>>>>>>> There have been notable improvements in FS in the past few months.
>>>>>>> -MC
>>>>>>> On Sat, Feb 5, 2011 at 3:30 PM, Victor Chukalovskiy
>>>>>>> <victor.chukalovskiy at utoronto.ca>  wrote:
>>>>>>>> Hello,
>>>>>>>> 
>>>>>>>> After purchasing a few licenses and installing the latest
>>>>>>>> fsg729-191-installer
>>>>>>>> I'm getting the following error when trying to load the mod_com_g729:
>>>>>>>>> 2011-02-04 14:27:17.637046 [CRIT] switch_loadable_module.c:926 Error
>>>>>>>>> Loading module /opt/fs/mod/mod_com_g729.so
>>>>>>>>> **Trying to load an out of date module, please rebuild the module.**
>>>>>>>> Also noticed that g729 installer ran with a couple errors:
>>>>>>>>> ./installer: line 62: ldconfig: command not found
>>>>>>>>> ./installer: line 49: useradd: command not found
>>>>>>>>  Any help or solution is much appreciated.
>>>>>>>> 
>>>>>>>> -Victor
>>>>>>>> 
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> 
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>> 
>> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.ponzone at ipeva.fr  Wed Feb  9 20:13:06 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 9 Feb 2011 18:13:06 +0100
Subject: [Freeswitch-users] Re-order or Invalid
In-Reply-To: <059901cbc87b$ef6afd50$ce40f7f0$@com>
References: <056601cbc874$b3341e90$199c5bb0$@com>	<9FB7DDC0-885B-47E8-AD27-47F15B736743@ipeva.fr>	<057d01cbc877$400abca0$c02035e0$@com>
	<875BD817-4009-49A7-B511-4AAE4B1345A4@ipeva.fr>
	<059901cbc87b$ef6afd50$ce40f7f0$@com>
Message-ID: <6CFBED51-2A0B-4F3A-B705-2C37D2AC0412@ipeva.fr>

They suck.
if they charge you per minute for your inbound DIDs, I would say that's not very fair to force you to answer the call.
In some countries, that would even be illegal.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 09/02/2011 ? 18:08, Peder a ?crit :

> Apparently icall doesn?t accept a 404 back as they just keep sending me the call over and over again?.  It appears I have to answer and play tones and a message myself.  For anyone who cares, here is what I have.
>  
>  
>   <extension name="ALLOTHER">
>     <condition field="destination_number" expression="^(\d{10})$">
>         <action application="answer"/>
>         <action application="playback" data="tone_stream://%(330,15,950);%(330,15,1400);%(330,1000,1800);"/>
>         <action application="playback" data="misc/8000/invalid_extension.wav"/>
>         <action application="hangup"/>
>     </condition>
>   </extension>
>  
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Ponzone
> Sent: Wednesday, February 09, 2011 10:48 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Re-order or Invalid
>  
> Add in your dialplan for the concerned number:
>  
>         <action application="respond" data="404"/>
>         <action application="hangup" data="UNALLOCATED_NUMBER"/>
>  
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>  
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>  
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>  
>  
> 
>  
> Le 09/02/2011 ? 17:34, Peder a ?crit :
> 
> 
> OK, how do you do that?
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Ponzone
> Sent: Wednesday, February 09, 2011 10:18 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Re-order or Invalid
>  
> I think you are supposed to send back a SIP 404.
>  
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>  
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>  
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>  
> 
> 
> 
>  
> Le 09/02/2011 ? 17:16, Peder a ?crit :
> 
> 
> 
> What is the correct way to send a re-order or invalid message/tone to a
> caller?  We have a couple hundred DIDs and quite a few of them are not used.
> If someone calls into one of those, we just want to send a re-order or busy
> tone.  My preference would be to NOT answer the call as I would prefer not
> to be charged for the call if I can avoid it.
> 
> Peder
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From peder at networkoblivion.com  Wed Feb  9 20:17:29 2011
From: peder at networkoblivion.com (Peder)
Date: Wed, 9 Feb 2011 11:17:29 -0600
Subject: [Freeswitch-users] Re-order or Invalid
In-Reply-To: <AANLkTim=xNi1v0-gn_BWwYNezcV6ur4=jdTG4m=awP7p@mail.gmail.com>
References: <056601cbc874$b3341e90$199c5bb0$@com>	<9FB7DDC0-885B-47E8-AD27-47F15B736743@ipeva.fr>	<057d01cbc877$400abca0$c02035e0$@com>	<875BD817-4009-49A7-B511-4AAE4B1345A4@ipeva.fr>	<059901cbc87b$ef6afd50$ce40f7f0$@com>
	<AANLkTim=xNi1v0-gn_BWwYNezcV6ur4=jdTG4m=awP7p@mail.gmail.com>
Message-ID: <05c101cbc87d$3ad42540$b07c6fc0$@com>

How do you end the call if you pre-answer? When I had below, it answered,
played the tone and message and hung up.  If I change answer to pre-answer,
it just keeps playing it over and over again. Or is that the point, it is up
to the caller to hangup?


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Wednesday, February 09, 2011 11:09 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Re-order or Invalid

you should be able to pre_answer instead? and not actually get billed.


On Wed, Feb 9, 2011 at 11:08 AM, Peder <peder at networkoblivion.com> wrote:
> Apparently icall doesn?t accept a 404 back as they just keep sending me
the
> call over and over again
.? It appears I have to answer and play tones and
a
> message myself.? For anyone who cares, here is what I have.
>
>
>
>
>
> ? <extension name="ALLOTHER">
>
> ??? <condition field="destination_number" expression="^(\d{10})$">
>
> ??????? <action application="answer"/>
>
> ??????? <action application="playback"
> data="tone_stream://%(330,15,950);%(330,15,1400);%(330,1000,1800);"/>
>
> ??????? <action application="playback"
> data="misc/8000/invalid_extension.wav"/>
>
> ??????? <action application="hangup"/>
>
> ??? </condition>
>
> ? </extension>
>
>
>
>
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David
> Ponzone
> Sent: Wednesday, February 09, 2011 10:48 AM
>
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Re-order or Invalid
>
>
>
> Add in your dialplan for the concerned number:
>
>
>
> ?? ? ? ?<action application="respond" data="404"/>
>
> ?? ? ? ?<action application="hangup" data="UNALLOCATED_NUMBER"/>
>
>
>
> David Ponzone ?Direction Technique
>
> email: david.ponzone at ipeva.fr
>
> tel: ? ? ?01 74 03 18 97
>
> gsm: ? 06 66 98 76 34
>
>
>
> Service Client?IPeva
>
> tel: ? ? ?0811 46 26 26
>
> www.ipeva.fr? -? ?www.ipeva-studio.com
>
>
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message
s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
>
>
>
> Le 09/02/2011 ? 17:34, Peder a ?crit :
>
> OK, how do you do that?
>
>
>
>
From:?freeswitch-users-bounces at lists.freeswitch.org?[mailto:freeswitch-users
-bounces at lists.freeswitch.org]?On
> Behalf Of?David Ponzone
> Sent:?Wednesday, February 09, 2011 10:18 AM
> To:?FreeSWITCH Users Help
> Subject:?Re: [Freeswitch-users] Re-order or Invalid
>
>
>
> I think you are supposed to send back a SIP 404.
>
>
>
> David Ponzone ?Direction Technique
>
> email:?david.ponzone at ipeva.fr
>
> tel: ? ? ?01 74 03 18 97
>
> gsm: ? 06 66 98 76 34
>
>
>
> Service Client?IPeva
>
> tel: ? ? ?0811 46 26 26
>
> www.ipeva.fr? -? ?www.ipeva-studio.com
>
>
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message
s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
>
>
> Le 09/02/2011 ? 17:16, Peder a ?crit :
>
>
> What is the correct way to send a re-order or invalid message/tone to a
> caller? ?We have a couple hundred DIDs and quite a few of them are not
used.
> If someone calls into one of those, we just want to send a re-order or
busy
> tone. ?My preference would be to NOT answer the call as I would prefer not
> to be charged for the call if I can avoid it.
>
> Peder
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 avi at avimarcus.net  Wed Feb  9 20:19:57 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 9 Feb 2011 19:19:57 +0200
Subject: [Freeswitch-users] Best way to implement real-time ivr from
	mysql
In-Reply-To: <BE42DE63-CA66-40C8-A0BF-17AE303C77B1@ipeva.fr>
References: <4D52C739.8080502@communicatefreely.net>
	<BE42DE63-CA66-40C8-A0BF-17AE303C77B1@ipeva.fr>
Message-ID: <AANLkTikxJ1uNR+2sM-V_z+0YAR_nbX8pyNc2=AK7E9YC@mail.gmail.com>

I haven't heard of any lua sql issues.. I'd keep looking into that.
Otherwise, I'm pretty happy with the flexibility and performance afforded to
me by xml_curl (running nginx + php5-fpm. run mysql tuner and increase
cache, also...).
I have a php class with a bit of abstraction for XML and dialplan functions
(but nothing specific for IVR). You can find it at:
https://github.com/avimar/FreeSWITCH-mod_xml-with-PHP

-Avi Marcus
<https://github.com/avimar/FreeSWITCH-mod_xml-with-PHP>

On Wed, Feb 9, 2011 at 7:05 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Tim,
>
> which ODBC is that ? the one integrated in FreeSWITCH that you can call
> from LUA with freeswitch.Dbh() ?
> I would say that solving those issues would be a better way to achieve
> this, for you and for everyone around.
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 09/02/2011 ? 17:56, Tim St. Pierre a ?crit :
>
> Hello list,
>
> I'm trying to implement a flexible, but easy to administer IVR system
> for our multi-tenant PBX.
>
> My goal is to be able to build a flexible auto-attendent scheme for
> incoming calls, based on data in a mysql database.
>
> The functionality I need is:
>    -Different options based on time of day
>    -Playback of 1-2 audio files while listening for digits (Thank you
> for calling abc company)
>    -Single digit options set some variables, then transfer to a defined
> extension (press 1 for sales)
>    -Multi-digit options are checked against a pattern, then a transfer
> is executed (enter the extension now ...)
>
> Not very complicated, I know, but making it scale is tricky.
>
> I have tried LUA, but LuaSQL has issues.  Even using ODBC, I still get
> memory leaks and random errors with file handles etc.
>
> For the most part, I haven't had any issues with xml_curl getting config
> from another server that generates it with PHP.  I'm exploring this, but
> it looks like I'll have to implement the following logic:
>    -Call comes in, and a dialplan is returned that plays the greetings,
> sets some variables, and does a play_and_get_digits, followed by a
> transfer back to the dialplan, so we can figure out what to do based on
> the digits.
>    -The callback to the dialplan evaluates the dtmf presented, and
> decides what action to take, returning that in another dialplan piece.
>
> I think this could work. but it means that a dialplan lookup has to be
> done each time someone makes a selection.
>
> Is there a better way, or does that make the most sense?
>
> Thanks!
>
> -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 Nabble at slickdeals.endjunk.com  Wed Feb  9 20:22:13 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 9 Feb 2011 09:22:13 -0800 (PST)
Subject: [Freeswitch-users] How to dial a remote_skypename from a
	telephone keypad?
In-Reply-To: <AANLkTikJNH3-vk=dMkOjAKsAxgr7KaiGD-U-9t2uKjxE@mail.gmail.com>
References: <1297263654900-6007946.post@n2.nabble.com>
	<AANLkTikfXRGxqbrassDNhOnaOWkaje3DXQptRAVRzb9d@mail.gmail.com>
	<AANLkTikJNH3-vk=dMkOjAKsAxgr7KaiGD-U-9t2uKjxE@mail.gmail.com>
Message-ID: <1297272133009-6008667.post@n2.nabble.com>



Giovanni Maruzzelli-2 wrote:
> with a SIP phone that allows you to enter alphanumeric destinations,
> you can use something similar to the "skype_uri" described in:
> http://wiki.freeswitch.org/wiki/Skypopen_Skype_Endpoint_and_Trunk#Dialplan.2C_and_how_to_use_Skypopen
If I understood you correctly with the Skype URI as shown below, I should be
able to pick up a telephone handset to dial a number, i.e. 754257,
4711,3019, etc., and the call will be able to reach a Skype user called joe,
marry, ana, etc. respectively (assuming they are also online). Cool.
<!-- dial via SKYPE uri using interface1 interface to go out -->
<extension name="skype_uri">
  <condition field="destination_number" expression="^skype/(.*)$">
    <action application="bridge" data="skypopen/interface1/$1"/>
  </condition>
</extension>

-----
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/How-to-dial-a-remote-skypename-from-a-telephone-keypad-tp6007946p6008667.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Wed Feb  9 20:23:27 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 9 Feb 2011 11:23:27 -0600
Subject: [Freeswitch-users] Re-order or Invalid
In-Reply-To: <05c101cbc87d$3ad42540$b07c6fc0$@com>
References: <056601cbc874$b3341e90$199c5bb0$@com>
	<9FB7DDC0-885B-47E8-AD27-47F15B736743@ipeva.fr>
	<057d01cbc877$400abca0$c02035e0$@com>
	<875BD817-4009-49A7-B511-4AAE4B1345A4@ipeva.fr>
	<059901cbc87b$ef6afd50$ce40f7f0$@com>
	<AANLkTim=xNi1v0-gn_BWwYNezcV6ur4=jdTG4m=awP7p@mail.gmail.com>
	<05c101cbc87d$3ad42540$b07c6fc0$@com>
Message-ID: <AANLkTi=g1n0TLjA1UJDYKZB4H-ur3Z8+0L7Ai=xBWRRi@mail.gmail.com>

maybe they are mis-configured or maybe you need to hangup with a
specific cause code that will make them stop like user_busy or
call_rejected or normal_circuit_congestion or no_route_destination


<action application="hangup" data="user_busy"/>



On Wed, Feb 9, 2011 at 11:17 AM, Peder <peder at networkoblivion.com> wrote:
> How do you end the call if you pre-answer? When I had below, it answered,
> played the tone and message and hung up. ?If I change answer to pre-answer,
> it just keeps playing it over and over again. Or is that the point, it is up
> to the caller to hangup?
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
> Minessale
> Sent: Wednesday, February 09, 2011 11:09 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Re-order or Invalid
>
> you should be able to pre_answer instead? and not actually get billed.
>
>
> On Wed, Feb 9, 2011 at 11:08 AM, Peder <peder at networkoblivion.com> wrote:
>> Apparently icall doesn?t accept a 404 back as they just keep sending me
> the
>> call over and over again?.? It appears I have to answer and play tones and
> a
>> message myself.? For anyone who cares, here is what I have.
>>
>>
>>
>>
>>
>> ? <extension name="ALLOTHER">
>>
>> ??? <condition field="destination_number" expression="^(\d{10})$">
>>
>> ??????? <action application="answer"/>
>>
>> ??????? <action application="playback"
>> data="tone_stream://%(330,15,950);%(330,15,1400);%(330,1000,1800);"/>
>>
>> ??????? <action application="playback"
>> data="misc/8000/invalid_extension.wav"/>
>>
>> ??????? <action application="hangup"/>
>>
>> ??? </condition>
>>
>> ? </extension>
>>
>>
>>
>>
>>
>>
>>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David
>> Ponzone
>> Sent: Wednesday, February 09, 2011 10:48 AM
>>
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Re-order or Invalid
>>
>>
>>
>> Add in your dialplan for the concerned number:
>>
>>
>>
>> ?? ? ? ?<action application="respond" data="404"/>
>>
>> ?? ? ? ?<action application="hangup" data="UNALLOCATED_NUMBER"/>
>>
>>
>>
>> David Ponzone ?Direction Technique
>>
>> email: david.ponzone at ipeva.fr
>>
>> tel: ? ? ?01 74 03 18 97
>>
>> gsm: ? 06 66 98 76 34
>>
>>
>>
>> Service Client?IPeva
>>
>> tel: ? ? ?0811 46 26 26
>>
>> www.ipeva.fr? -? ?www.ipeva-studio.com
>>
>>
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message
> s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>>
>>
>>
>>
>> Le 09/02/2011 ? 17:34, Peder a ?crit :
>>
>> OK, how do you do that?
>>
>>
>>
>>
> From:?freeswitch-users-bounces at lists.freeswitch.org?[mailto:freeswitch-users
> -bounces at lists.freeswitch.org]?On
>> Behalf Of?David Ponzone
>> Sent:?Wednesday, February 09, 2011 10:18 AM
>> To:?FreeSWITCH Users Help
>> Subject:?Re: [Freeswitch-users] Re-order or Invalid
>>
>>
>>
>> I think you are supposed to send back a SIP 404.
>>
>>
>>
>> David Ponzone ?Direction Technique
>>
>> email:?david.ponzone at ipeva.fr
>>
>> tel: ? ? ?01 74 03 18 97
>>
>> gsm: ? 06 66 98 76 34
>>
>>
>>
>> Service Client?IPeva
>>
>> tel: ? ? ?0811 46 26 26
>>
>> www.ipeva.fr? -? ?www.ipeva-studio.com
>>
>>
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message
> s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>>
>>
>>
>> Le 09/02/2011 ? 17:16, Peder a ?crit :
>>
>>
>> What is the correct way to send a re-order or invalid message/tone to a
>> caller? ?We have a couple hundred DIDs and quite a few of them are not
> used.
>> If someone calls into one of those, we just want to send a re-order or
> busy
>> tone. ?My preference would be to NOT answer the call as I would prefer not
>> to be charged for the call if I can avoid it.
>>
>> Peder
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 gmaruzz at gmail.com  Wed Feb  9 20:34:04 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 9 Feb 2011 18:34:04 +0100
Subject: [Freeswitch-users] How to dial a remote_skypename from a
 telephone keypad?
In-Reply-To: <1297272133009-6008667.post@n2.nabble.com>
References: <1297263654900-6007946.post@n2.nabble.com>
	<AANLkTikfXRGxqbrassDNhOnaOWkaje3DXQptRAVRzb9d@mail.gmail.com>
	<AANLkTikJNH3-vk=dMkOjAKsAxgr7KaiGD-U-9t2uKjxE@mail.gmail.com>
	<1297272133009-6008667.post@n2.nabble.com>
Message-ID: <AANLkTikdMDk_NkKVyH6Qd4DBq5izujG77eYPOi1ZCbms@mail.gmail.com>

if I have understood you, I must give you a bad news:

On Wed, Feb 9, 2011 at 6:22 PM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
> If I understood you correctly with the Skype URI as shown below, I should be
> able to pick up a telephone handset to dial a number, i.e. 754257,
> 4711,3019, etc., and the call will be able to reach a Skype user called joe,
> marry, ana, etc. respectively (assuming they are also online). Cool.

not at all

from a SIP phone (softphone or hardphone) in this example you need to
be able to input "skype/giovanni334" to reach the skypeuser
giovanni334

you can do that or because you enter that via the alphanumeric keyboad
(eg, from XLite on your computer keyboard), or because you entered
that input in the phonebook of the phone (or you got a SIP phone with
an alphanumeric keyboard)

> <!-- dial via SKYPE uri using interface1 interface to go out -->
> <extension name="skype_uri">
> ?<condition field="destination_number" expression="^skype/(.*)$">
> ? ?<action application="bridge" data="skypopen/interface1/$1"/>
> ?</condition>
> </extension>
>

The plain old numeric keyboard can only input numbers.
In case a number is mapped in the dialplan, or with some other method,
to a skypeuser, then that call will go to the mapped skypeuser


-giovanni



> -----
> 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/How-to-dial-a-remote-skypename-from-a-telephone-keypad-tp6007946p6008667.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 anthony.minessale at gmail.com  Wed Feb  9 21:09:12 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 9 Feb 2011 12:09:12 -0600
Subject: [Freeswitch-users] RTP keep alive
In-Reply-To: <25A5DCAF-6F3D-4689-8FA2-B282BA6B3235@ipeva.fr>
References: <AANLkTik-o+kpsLyTTtjL09Zg4M9a23aPygFj1d-V=cTb@mail.gmail.com>
	<AANLkTi=aUeWAo+54O3U_LAK1kEr-U3wTNty+4AS5HG42@mail.gmail.com>
	<AANLkTi=0Uo-7qqWfB9uDFownX7TuVzsez3aVnUGMORH4@mail.gmail.com>
	<AANLkTinSJ4XPnJvzFfALA8tKwFxnRiq0P8JFk+Uz1kV6@mail.gmail.com>
	<25A5DCAF-6F3D-4689-8FA2-B282BA6B3235@ipeva.fr>
Message-ID: <AANLkTim4tP9AYSpwUsTJJc4Agi7tu2R17nD4FmEqAVFn@mail.gmail.com>

I think you are talking about CN (payload 13)
This is a rfc standard to send a packet every once in a while to prove
it still works.
if you negotiate CN in the sdp you would have it.


On Wed, Feb 9, 2011 at 11:10 AM, David Ponzone <david.ponzone at ipeva.fr> wrote:
> Sam,
> No there is no such thing as RTP keep-alive, AFAIK.
> But again, if you don't use VAD, there should be RTP all the time (except
> when on hold).
> So the first thing you should ask yourself is why is this RTP stream
> missing.
> Also you did not provide any information about the setup and the possible
> involvement of another equipement (provider ?).
> Be aware that if G729 is negotiated, it may use VAD by default if it was not
> disabled explicitly (that's the RFC).
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 09/02/2011 ? 18:01, Sam a ?crit :
>
> There is no vad , is there something like rtp keep live after the increase
> of rtp timout .
> Yes the RTP timeout helps.
>
> Regds
> Sam
>
> On Wed, Feb 9, 2011 at 8:03 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> That'll do it.
>>
>> VAD can mean no RTP is transmitted legitimately though, so be careful.
>> Especially when on hold when the phone might stop sending anything (I see
>> you've already increased that time).
>>
>> -Steve
>>
>>
>> On 9 February 2011 08:49, Sam <u2nsam at gmail.com> wrote:
>>>
>>> I have this in the settings,
>>>
>>> ??? <param name="rtp-timeout-sec" value="3000"/>
>>> ??? <param name="rtp-hold-timeout-sec" value="18000"/>
>>>
>>> Any thing i need more regarding that ?
>>>
>>> Regards
>>> Sam
>>>
>>> On Wed, Feb 9, 2011 at 1:35 PM, Sam <u2nsam at gmail.com> wrote:
>>>>
>>>> Hello,
>>>>
>>>> We have a situation wherein we need to keep alive RTP ,is there any
>>>> parameter to do that,
>>>> because when someone is on long conversation and not talking fro brief
>>>> duration the call
>>>> disconnects .
>>>>
>>>> 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
>
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 david.ponzone at ipeva.fr  Wed Feb  9 21:19:42 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 9 Feb 2011 19:19:42 +0100
Subject: [Freeswitch-users] RTP keep alive
In-Reply-To: <AANLkTim4tP9AYSpwUsTJJc4Agi7tu2R17nD4FmEqAVFn@mail.gmail.com>
References: <AANLkTik-o+kpsLyTTtjL09Zg4M9a23aPygFj1d-V=cTb@mail.gmail.com>
	<AANLkTi=aUeWAo+54O3U_LAK1kEr-U3wTNty+4AS5HG42@mail.gmail.com>
	<AANLkTi=0Uo-7qqWfB9uDFownX7TuVzsez3aVnUGMORH4@mail.gmail.com>
	<AANLkTinSJ4XPnJvzFfALA8tKwFxnRiq0P8JFk+Uz1kV6@mail.gmail.com>
	<25A5DCAF-6F3D-4689-8FA2-B282BA6B3235@ipeva.fr>
	<AANLkTim4tP9AYSpwUsTJJc4Agi7tu2R17nD4FmEqAVFn@mail.gmail.com>
Message-ID: <0B9C66DB-2B4F-4494-BB64-A8414CF1C74A@ipeva.fr>

Anthony,

I actually thought about telling that to Sam, as it is the behaviour of Siemens DECT IP phones for instance.
In the end, I think it would help to see a tshark trace of a such call.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 09/02/2011 ? 19:09, Anthony Minessale a ?crit :

> I think you are talking about CN (payload 13)
> This is a rfc standard to send a packet every once in a while to prove
> it still works.
> if you negotiate CN in the sdp you would have it.
> 
> 
> On Wed, Feb 9, 2011 at 11:10 AM, David Ponzone <david.ponzone at ipeva.fr> wrote:
>> Sam,
>> No there is no such thing as RTP keep-alive, AFAIK.
>> But again, if you don't use VAD, there should be RTP all the time (except
>> when on hold).
>> So the first thing you should ask yourself is why is this RTP stream
>> missing.
>> Also you did not provide any information about the setup and the possible
>> involvement of another equipement (provider ?).
>> Be aware that if G729 is negotiated, it may use VAD by default if it was not
>> disabled explicitly (that's the RFC).
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> 
>> 
>> 
>> Le 09/02/2011 ? 18:01, Sam a ?crit :
>> 
>> There is no vad , is there something like rtp keep live after the increase
>> of rtp timout .
>> Yes the RTP timeout helps.
>> 
>> Regds
>> Sam
>> 
>> On Wed, Feb 9, 2011 at 8:03 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>> 
>>> That'll do it.
>>> 
>>> VAD can mean no RTP is transmitted legitimately though, so be careful.
>>> Especially when on hold when the phone might stop sending anything (I see
>>> you've already increased that time).
>>> 
>>> -Steve
>>> 
>>> 
>>> On 9 February 2011 08:49, Sam <u2nsam at gmail.com> wrote:
>>>> 
>>>> I have this in the settings,
>>>> 
>>>>     <param name="rtp-timeout-sec" value="3000"/>
>>>>     <param name="rtp-hold-timeout-sec" value="18000"/>
>>>> 
>>>> Any thing i need more regarding that ?
>>>> 
>>>> Regards
>>>> Sam
>>>> 
>>>> On Wed, Feb 9, 2011 at 1:35 PM, Sam <u2nsam at gmail.com> wrote:
>>>>> 
>>>>> Hello,
>>>>> 
>>>>> We have a situation wherein we need to keep alive RTP ,is there any
>>>>> parameter to do that,
>>>>> because when someone is on long conversation and not talking fro brief
>>>>> duration the call
>>>>> disconnects .
>>>>> 
>>>>> 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
>> 
>> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 Nabble at slickdeals.endjunk.com  Wed Feb  9 21:39:32 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 9 Feb 2011 10:39:32 -0800 (PST)
Subject: [Freeswitch-users] How to dial a remote_skypename from a
	telephone keypad?
In-Reply-To: <AANLkTikdMDk_NkKVyH6Qd4DBq5izujG77eYPOi1ZCbms@mail.gmail.com>
References: <1297263654900-6007946.post@n2.nabble.com>
	<AANLkTikfXRGxqbrassDNhOnaOWkaje3DXQptRAVRzb9d@mail.gmail.com>
	<AANLkTikJNH3-vk=dMkOjAKsAxgr7KaiGD-U-9t2uKjxE@mail.gmail.com>
	<1297272133009-6008667.post@n2.nabble.com>
	<AANLkTikdMDk_NkKVyH6Qd4DBq5izujG77eYPOi1ZCbms@mail.gmail.com>
Message-ID: <1297276772026-6009014.post@n2.nabble.com>



Giovanni Maruzzelli-2 wrote:
> 
> if I have understood you, I must give you a bad news:
Now, you seemed to have understood my question. Looks like one needs to
create a phonebook entry to contains all some numerical numbers mapped to
skype user names. Perhaps, someone who is more knowledgeable can update the
FS wiki to include this phonebook feature.

-----
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/How-to-dial-a-remote-skypename-from-a-telephone-keypad-tp6007946p6009014.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From gmaruzz at gmail.com  Wed Feb  9 21:52:28 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 9 Feb 2011 19:52:28 +0100
Subject: [Freeswitch-users] How to dial a remote_skypename from a
 telephone keypad?
In-Reply-To: <1297276772026-6009014.post@n2.nabble.com>
References: <1297263654900-6007946.post@n2.nabble.com>
	<AANLkTikfXRGxqbrassDNhOnaOWkaje3DXQptRAVRzb9d@mail.gmail.com>
	<AANLkTikJNH3-vk=dMkOjAKsAxgr7KaiGD-U-9t2uKjxE@mail.gmail.com>
	<1297272133009-6008667.post@n2.nabble.com>
	<AANLkTikdMDk_NkKVyH6Qd4DBq5izujG77eYPOi1ZCbms@mail.gmail.com>
	<1297276772026-6009014.post@n2.nabble.com>
Message-ID: <AANLkTikDg6yJVHh7M=q0cNkacwhXpJBV5kJSThTaJGTZ@mail.gmail.com>

English is not my first language, so I try to be more clear now:

if you want to use the "skype_uri" (that is just a dialplan hack), you
must find a way to pass the skypeusername to the dialplan. One such
way is through the phonebook of the phone.

The other way is to map extensions to an action that bridges SIP to
the desired outbound skypopen call to desired skypename. This is the
more direct and popular way.

I'm thinking to add some sort of way to call skypenames from the
numeric keyboard (like the directory feature of asterisk) but is not
there yet.

-giovanni


On Wed, Feb 9, 2011 at 7:39 PM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
>
> Giovanni Maruzzelli-2 wrote:
>>
>> if I have understood you, I must give you a bad news:
> Now, you seemed to have understood my question. Looks like one needs to
> create a phonebook entry to contains all some numerical numbers mapped to
> skype user names. Perhaps, someone who is more knowledgeable can update the
> FS wiki to include this phonebook feature.
>
> -----
> 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/How-to-dial-a-remote-skypename-from-a-telephone-keypad-tp6007946p6009014.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 dmitry.bely at gmail.com  Wed Feb  9 23:22:29 2011
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Wed, 9 Feb 2011 23:22:29 +0300
Subject: [Freeswitch-users] How to dial a remote_skypename from a
 telephone keypad?
In-Reply-To: <AANLkTikDg6yJVHh7M=q0cNkacwhXpJBV5kJSThTaJGTZ@mail.gmail.com>
References: <1297263654900-6007946.post@n2.nabble.com>
	<AANLkTikfXRGxqbrassDNhOnaOWkaje3DXQptRAVRzb9d@mail.gmail.com>
	<AANLkTikJNH3-vk=dMkOjAKsAxgr7KaiGD-U-9t2uKjxE@mail.gmail.com>
	<1297272133009-6008667.post@n2.nabble.com>
	<AANLkTikdMDk_NkKVyH6Qd4DBq5izujG77eYPOi1ZCbms@mail.gmail.com>
	<1297276772026-6009014.post@n2.nabble.com>
	<AANLkTikDg6yJVHh7M=q0cNkacwhXpJBV5kJSThTaJGTZ@mail.gmail.com>
Message-ID: <AANLkTik0wPtXNOcd95wX2TH8B0+BaY7mz3aeBZaWa3hi@mail.gmail.com>

I was thinking of the following scheme. As we all know there is
correspondence between digits and letters on the phone's numeric
keypad:

1 -
2 - abc
3 - def
4 - ghi
5 - jkl
6 - mno
7 - pqrs
8 - tuv
9 - wxyz
0 -

What if we would encode letters the following way:

a - 2
b - 22
c - 222
d - 3
e - 33
f - 333

And so on. I.e. a letter is encoded with a digit written on the
appropriate button repeated so many times as is the letter's serial
number on the button. It's very similar to what is used in mobile
phones to type alphabetic info. Remaining important symbols can also
be encoded with same method by digit 1 (as also done in mobile
phones):

1 - .,-

but actually they are rarely needed.

# button can be used to start encoding sequence and ## is to end it.
So e.g. giovanni334 is be encoded as

#4#444#666#888#2#66#66#444##334

or even

#4#444666888266#66444##334 (if digit changes we know that another
character is started. Thus # prefix can be safely omitted)

Then all we would need is a very simple decoder on the
freeswitch/skypopen side. Just an idea...

- Dmitry Bely


From cliff at develix.com  Wed Feb  9 23:25:18 2011
From: cliff at develix.com (Cliff Wells)
Date: Wed, 09 Feb 2011 12:25:18 -0800
Subject: [Freeswitch-users] Gateway with no registration
Message-ID: <1297283118.3511.60.camel@portable-evil>

I'd like to setup a gateway with no registration (other end allows by
IP).

<include>
  <gateway name="outbound">
    <param name="username" value="none"/>
    <param name="password" value="none"/>
    <param name="proxy" value="x.x.x.x"/>
    <param name="register-transport" value="udp"/>
    <param name="retry-seconds" value="30"/>
    <param name="caller-id-in-from" value="true"/>
  </gateway>
</include>

As I understand it, the username and password shouldn't matter since
registration is challenge/auth.

What happens is I get a 403 when FreeSWITCH attempts to register (before
any calls are placed), so FreeSWITCH has the gateway marked as
FAIL_WAIT.   Is this proof the other end is requiring registration?  Or
is FS expecting a challenge, getting none and marking the gateway down?

Most of the solutions I've seen for this type of setup involve using a
dialplan rather than a gateway, but my dialplans are served via
mod_xml_curl and expect a functional gateway named "outbound" on the FS
box.   I'd rather not clutter up the xml dialplans with a bunch of
special cases, so a named gateway is preferred.

The problem certainly may lie on the other end (I don't have control of
that system), but I'm not sure how to verify that.

Regards,
Cliff



From mitch.capper at gmail.com  Wed Feb  9 23:32:26 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 9 Feb 2011 12:32:26 -0800
Subject: [Freeswitch-users] Gateway with no registration
In-Reply-To: <1297283118.3511.60.camel@portable-evil>
References: <1297283118.3511.60.camel@portable-evil>
Message-ID: <AANLkTinjFbqZRPwLcV1c8pe4xZz7YHSxbHBgApt=Aspw@mail.gmail.com>

try register=false
~Mitch

On Wed, Feb 9, 2011 at 12:25 PM, Cliff Wells <cliff at develix.com> wrote:

> I'd like to setup a gateway with no registration (other end allows by
> IP).
>
> <include>
>  <gateway name="outbound">
>    <param name="username" value="none"/>
>    <param name="password" value="none"/>
>    <param name="proxy" value="x.x.x.x"/>
>    <param name="register-transport" value="udp"/>
>    <param name="retry-seconds" value="30"/>
>    <param name="caller-id-in-from" value="true"/>
>  </gateway>
> </include>
>
> As I understand it, the username and password shouldn't matter since
> registration is challenge/auth.
>
> What happens is I get a 403 when FreeSWITCH attempts to register (before
> any calls are placed), so FreeSWITCH has the gateway marked as
> FAIL_WAIT.   Is this proof the other end is requiring registration?  Or
> is FS expecting a challenge, getting none and marking the gateway down?
>
> Most of the solutions I've seen for this type of setup involve using a
> dialplan rather than a gateway, but my dialplans are served via
> mod_xml_curl and expect a functional gateway named "outbound" on the FS
> box.   I'd rather not clutter up the xml dialplans with a bunch of
> special cases, so a named gateway is preferred.
>
> The problem certainly may lie on the other end (I don't have control of
> that system), but I'm not sure how to verify that.
>
> 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 brian at freeswitch.org  Wed Feb  9 23:34:07 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 9 Feb 2011 14:34:07 -0600
Subject: [Freeswitch-users] Gateway with no registration
In-Reply-To: <1297283118.3511.60.camel@portable-evil>
References: <1297283118.3511.60.camel@portable-evil>
Message-ID: <281D641E-15A0-47CF-B8C0-9E88C78BC8CF@freeswitch.org>

Then make sure register is set to false on the gateawy.

/b

On Feb 9, 2011, at 2:25 PM, Cliff Wells wrote:

> I'd like to setup a gateway with no registration (other end allows by
> IP).
> 
> <include>
>  <gateway name="outbound">
>    <param name="username" value="none"/>
>    <param name="password" value="none"/>
>    <param name="proxy" value="x.x.x.x"/>
>    <param name="register-transport" value="udp"/>
>    <param name="retry-seconds" value="30"/>
>    <param name="caller-id-in-from" value="true"/>
>  </gateway>
> </include>
> 
> As I understand it, the username and password shouldn't matter since
> registration is challenge/auth.
> 
> What happens is I get a 403 when FreeSWITCH attempts to register (before
> any calls are placed), so FreeSWITCH has the gateway marked as
> FAIL_WAIT.   Is this proof the other end is requiring registration?  Or
> is FS expecting a challenge, getting none and marking the gateway down?
> 
> Most of the solutions I've seen for this type of setup involve using a
> dialplan rather than a gateway, but my dialplans are served via
> mod_xml_curl and expect a functional gateway named "outbound" on the FS
> box.   I'd rather not clutter up the xml dialplans with a bunch of
> special cases, so a named gateway is preferred.
> 
> The problem certainly may lie on the other end (I don't have control of
> that system), but I'm not sure how to verify that.
> 
> 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



From cliff at develix.com  Wed Feb  9 23:38:43 2011
From: cliff at develix.com (Cliff Wells)
Date: Wed, 09 Feb 2011 12:38:43 -0800
Subject: [Freeswitch-users] Gateway with no registration
In-Reply-To: <AANLkTinjFbqZRPwLcV1c8pe4xZz7YHSxbHBgApt=Aspw@mail.gmail.com>
References: <1297283118.3511.60.camel@portable-evil>
	<AANLkTinjFbqZRPwLcV1c8pe4xZz7YHSxbHBgApt=Aspw@mail.gmail.com>
Message-ID: <1297283923.3511.62.camel@portable-evil>

That was it.  Thanks!

Cliff

On Wed, 2011-02-09 at 12:32 -0800, Mitch Capper wrote:
> try register=false
> ~Mitch
> 
> On Wed, Feb 9, 2011 at 12:25 PM, Cliff Wells <cliff at develix.com>
> wrote:
>         I'd like to setup a gateway with no registration (other end
>         allows by
>         IP).
>         
>         <include>
>          <gateway name="outbound">
>            <param name="username" value="none"/>
>            <param name="password" value="none"/>
>            <param name="proxy" value="x.x.x.x"/>
>            <param name="register-transport" value="udp"/>
>            <param name="retry-seconds" value="30"/>
>            <param name="caller-id-in-from" value="true"/>
>          </gateway>
>         </include>
>         
>         As I understand it, the username and password shouldn't matter
>         since
>         registration is challenge/auth.
>         
>         What happens is I get a 403 when FreeSWITCH attempts to
>         register (before
>         any calls are placed), so FreeSWITCH has the gateway marked as
>         FAIL_WAIT.   Is this proof the other end is requiring
>         registration?  Or
>         is FS expecting a challenge, getting none and marking the
>         gateway down?
>         
>         Most of the solutions I've seen for this type of setup involve
>         using a
>         dialplan rather than a gateway, but my dialplans are served
>         via
>         mod_xml_curl and expect a functional gateway named "outbound"
>         on the FS
>         box.   I'd rather not clutter up the xml dialplans with a
>         bunch of
>         special cases, so a named gateway is preferred.
>         
>         The problem certainly may lie on the other end (I don't have
>         control of
>         that system), but I'm not sure how to verify that.
>         
>         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
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 sjmudd at pobox.com  Thu Feb 10 02:33:01 2011
From: sjmudd at pobox.com (Simon J Mudd)
Date: 10 Feb 2011 00:33:01 +0100
Subject: [Freeswitch-users] Enabling extensions with passwords and
	limiting network access via acls (Was: Confusing SIP auth
	failure logging message?)
In-Reply-To: <5313ADE6-FDD7-4124-AF56-885A16C15A46@freeswitch.org>
References: <20110206232236.GA10501@mad06.wl0.org>
	<20110207073124.GA5255@mad06.wl0.org>
	<m3aai69nm4.fsf_-_@mad06.wl0.org>
	<5313ADE6-FDD7-4124-AF56-885A16C15A46@freeswitch.org>
Message-ID: <m3vd0sdbrm.fsf@mad06.wl0.org>

Hi Brian,

brian at freeswitch.org (Brian West) writes:

> So while you can buy a gun and bullets who's fault is it when you
> get shot in the foot?  Same thing really.

I'm not sure I like examples with guns as there are often very
opposing points of view on topics like that. However, it is true that
with all software you can configure it incorrectly and cause yourself
problems. Ideally that's something you want to avoid if possible.

> Our dialplan is rather secure since I designed it and I fully
> understand how our security model works.

So you understand how to _avoid_ mistakes and what _not_ to do. The
issue I've been having is that for me it's not so obivous when I'm 
making those obvious mistakes and doing things that are not sensible.
I have tried to read the documentation. I believe that if it can happen
to me others will have similar difficulties. Thus the learning curve
even to setup something "simple" is exremely steep.

> Our default is just an example of how to use FreeSWITCH.

Of course. The software is flexible so giving people a starting point
is helpful. Providing pointers (if that's possible about the good
things to do and the things to avoid) aids us even more.

> I could do a service and just delete it all and leave
> it up to you to figure it all out but I feel learning by example is
> a great way to see how to use the software.

Yes, and I for one appreciate the example configuration.

Having said that the page
http://wiki.freeswitch.org/wiki/SIP_Provider_Examples seems to imply
that if I add external gateways they should be added under conf/sip_profiles/external/provider.xml

external.xml says:

<profile name="external">
  <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> 
  <!-- This profile is only for outbound registrations to providers -->
  <gateways>
    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
  </gateways>
...

which seems to confirm this idea, yet

internal.xml says:

<profile name="internal">
...
  <!-- Outbound Registrations -->
  <gateways>
    <X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
  </gateways>
...

It's not clear to me where the providers should go. I've added them
under external/ but I think that's wrong. Understand why I'm confused?

Also I see from: http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Configuring_FreeSWITCH
comments about:

Some common extensions for testing

1000, 1001, ..., 1019 - Generic SIP extensions 

No mention here that these extensions are "reachable from
outside". While there's nothing specifically bad about them being
reachable from outside if that's not the behaviour you want and you
think they are internal extensions only you may have a surprise.

I'm reluctant to go editing the wiki because I don't fully understand
things but perhaps a "(also reachable from outside)" comment would be
useful and perhaps a similar comment in the configuration files.

The files directory/default/10XX.xml "seem" to be internal, and
there's no indication that this might not be the case. Yes, I've
_finally_ figure out that this because of the configuration in
dialplan/public.xml. It's really hard to put this all together unless
you understand the meaning and usage of all the configuration files.

I can:
* suggest patches to the config files to add comments if that would
help.
* edit the wiki if I'm _sure_ that the changes I make are correct

but I'm rather reluctant to jump in when I'm not the one who really
understands and others do much better. I'm silly trying to point
out _I_ I'm having trouble, why I'm having trouble and hoping
that this will aid in the end to improving the documentation so 
that more people can use this software.

Regards,

Simon 





From sjmudd at pobox.com  Thu Feb 10 03:21:13 2011
From: sjmudd at pobox.com (Simon J Mudd)
Date: 10 Feb 2011 01:21:13 +0100
Subject: [Freeswitch-users] Enabling extensions with passwords and
	limiting network access via acls (Was: Confusing SIP auth
	failure logging message?)
In-Reply-To: <AANLkTimZLYfpKVmS7rACTEG3M42eM5a3s5gghhr6Mju8@mail.gmail.com>
References: <20110206232236.GA10501@mad06.wl0.org>
	<20110207073124.GA5255@mad06.wl0.org>
	<m3aai69nm4.fsf_-_@mad06.wl0.org>
	<AANLkTimZLYfpKVmS7rACTEG3M42eM5a3s5gghhr6Mju8@mail.gmail.com>
Message-ID: <m3r5bgd9ja.fsf@mad06.wl0.org>

msc at freeswitch.org (Michael Collins) writes:

> I know you came on to the #freeswitch IRC channel but I don't recall where
> you left off in your quest for FreeSWITCH knowledge.

I'm still on that quest. It's just taking some time. I also appreciate the
time several of you are taking in answering what seem to be "silly" questions.

> I'll provide information here with links to relevant wiki articles,
> etc. all inline.  However, let me just put these two points out
> there for everyone:
> 
> #1 - If, when learning FreeSWITCH, you feel overwhelmed because there are so
> many different config options, etc. then remember this: FreeSWITCH is
> designed for power and flexibility, not simplicity. A carrier-grade,
> multi-protocol soft-switch is supposed to be complex.

Understood. Interestingly I don't see the word "security" mentioned.
Perhaps that's not a primary concern as most "professional FreeSWITCH
installations" are aided by a lot of extra "protection", be that 
quite limited access to external gateways or a lot of time configuring
firewalls before setting up other parts of the system. In that environment
I understand security to be if not less of a concern, probably a well
understood foe.

> #2 - The default configuration is just an example of some of the many cool
> things you can do with FreeSWITCH. It is not meant to be a turn-key,
> out-of-box solution to put into a production environment, even a SOHO
> environment. This is a feature, not a bug.

Configuring _any_ system to work properly for a specific use case
requires tuning it, adjusting it and perhaps setting it up in a
different way to the default configuration. My current day job
involves managing a large number of database servers. A default
configuration is pretty useless and the "deployment" of a new server
requires quite a lot of work, and tools so that process can be
automated. Many people would not immediately think it necessary to do
many of the things I do, or configure things the way I do, but for my
environment that's needed.  The software I use however is configured
by default, like lots of internet software, to basically not allow
"outside access" unless you enable that. Enabling it may be very easy
but initially at least the "tighter configuration" means that you
don't open yourself to problems without realising. The people on this
list are likely not to have that problem, but others trying the
software for the first time are. If you want to encourage new users
for small and large installations helping them to avoid shooting
themselves in the foot is always appreciated.

Perhaps just a different point of view. Different use cases (I'm
looking at this from a hobbyist perspective, you probably earn your
living building systems on this software?)

...

> > > This specifies a user for registration who:
> > > (1) must provide a password
> > > (2) can only register from the given network range
> > > (3) is only allowed to make 1 call at a time
> 
> > I see that there are ways to implement (3) though it seems that's more on
> > a per gateway basis than a per extension basis. That's ok.
> >
> > What really interests me is implementing (1) _and_ (2) together.  Is this
> > possible? If not it would certainly be a nice new feature.
> 
> This is definitely possible. There are two ways to do it that I can think
> of:
> #1 - The "proper" way with mod_xml_curl
> #2 - The less proper but still quite functional way of using a user's
> channel variables and dialplan.*

(1) and (2) have worked with static xml files as I described earlier. I'm
sure there are better ways, but first I need to be comfortable with basic stuff.

...

> For more information on mod_xml_curl see:
> http://wiki.freeswitch.org/wiki/Mod_xml_curl
> Also see Raymond's samples in the freeswitch-contrib repo:
> http://fisheye.freeswitch.org/browse/freeswitch-contrib/intralanman/PHP/fs_curl

mod_curl sounds interesting. I'll take a look. _Needing_ to use something
like this for a "simple" system seems like overkill, even if it provides
more flexibility. Something no doubt for me to look at later. The db backend
is the easy bit: I'm a DBA... And writing code in php or perl doesn't really
worry me.

> > Perhaps the default FreeSWITCH configuration should limit access to
> > the default extensions to be registered only from the networks defined
> > in localnet.auto.  This reduces exposure to external bad
> > software.
>
> Many people don't want this limitation turned on by default. If we did this
> then a lot of people would be asking for the ability to have external phones
> be able to access FreeSWITCH by default. Besides, there are other
> alternatives, such as using a firewall.

I've just found out to my detriment that a firewall is worth little if
you poke any outbound holes out to a provider.

If that's such an issue then make a switch so it works both ways.
It's true that switches in vars.xml mean that the users (like me) can be
dumb and do the wrong thing, but I guess that also calls for some sort
of URL reference to explain the "why", and the different options for
those that need to figure out what's going on.

...

> Your blog had several other issues so I'll number them here and then comment
> on them below.
> Questions:
> 
> 1. Default FreeSWITCH logging maybe too verbose
> 2. XML file breakage
> 3. Relatedly, comments need to be in "<!-- xxx -->" format
> 4. The external profile should log auth failures by default
> 5. FreeSWITCH should have some sort of rate-limiting
> 6. It should be more obvious how to configure network ACLs for for
> extensions, and these should be configured by default
> 7. For registration, a client can use FreeSWITCH's IP address as well as
> domain name
> 8. "For trunk connections if you have a DID number you expect the VoIP
> provider to call you..."
> 9. Allow rate-limiting to/from a gateway or extension
> 
> Answers:
> 
> 1. This goes back to FreeSWITCH being a massively huge piece of software
> with lots of configuration options. Having verbose debugging is very handy.
> It is much better for a new person to have too much information at first and
> then learn about what logging to turn off. The other alternative is for a
> new person to come around with a question about something that happened and
> not have any logs to help debug. Log files can be archived or deleted, but
> the information that never got logged is worthless.
> 
> 2. XML "file breakage" is well-documented here:
> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Chasing_down_XML_errors
> If you wanted to you could take the aggregated file (freeswitch.xml.fsxml)
> and put it in conf/freeswitch.xml as the only config file. Of course, now
> you're editing a 4500 line XML file... :)

Eventually I found this out. I think it took me a couple of months looking 
around and swearing in frustration that I couldn't figure out what the error
was in line 4321.  Again maybe this needs to be pointed out somewhere more prominently,
or if possible the freeswitch logging mentions the fsxml filename itself.

> 3. These are just standard XML comments. They take the form:
> <!-- comment -->
> The XML standard dictates that you have two dashes. (Your example seems to
> show single dashes.) The XML standard also states that you may not nest
> comments, nor may you include "--" inside of a comment. See
> http://www.w3.org/TR/REC-xml/#sec-comments for more info

Then maybe double -- where causing the issue. Thanks for reference.

> 4. This is a reasonable suggestion. We have a saying: "patches accepted". If
> anyone has a moment to implement *AND TEST* this please do so and send us a
> patch on Jira.
> 
> 5. Rate-limiting is absolutely possible in FreeSWITCH, and it is not limited
> to SIP traffic. Pretty much anything you can think of can be rate-limited
> using mod_limit. More information here:
> http://wiki.freeswitch.org/wiki/Mod_limit

Again more for me to read. Thanks again for the pointer.

> 6. Not sure how to make it more obvious than this:
> http://wiki.freeswitch.org/wiki/Acl#Users

If I understand it that's wrong. Using the "apply-inbound-acl" is to disable
the _need_ for passwords, not add the extra acl in addition to the configured
passwords.

my directory/default/1000.xml now says:

<include>
  <user id="1000">
    <params>
      <param name="password" value="$${default_password}"/>
      <param name="vm-password" value="1000"/>
      <param name="auth-acl" value="localnet.auto"/>
...

And that does what I want. That would be nice in the example config
even if you disable it by default (comment out).

> 7. We allow both IP and domain name by default to lower the barrier to
> entry. The default config is supposed to "just work" in a sandbox so that
> new users can learn. One great way to learn is by setting up a system and
> then making mods to the default config and see what breaks.

Or what lets someone break into your system. Yes, I'm insisting on something
because when I get my phone bill I'm not expecting to be happy. So yes
a sandbox is good, and breaking and tweaking configs is needed. I'm just
[over?] paranoid now. So I "added a gateway" and someone got in through
an extension and started using it... (silly mistake of mine, but easy thing
to happen again)
 
> 8. This is a fun one... What you ask for is something that would work in an
> ideal situation, however SIP is involved so idealism must give way to
> realism. Most providers will blow up and die if you send them 401's or 407's
> so that's not a logical choice for the default. (Again, nothing preventing
> you from modifying the default configs to suit your personal needs.) You can
> also use ACLs to limit the source of incoming calls, but then you run the
> risk of them reconfiguring and changing IP addresses. As long as you are
> prepared for that scenario then you're fine, but most new users probably
> could do without us adding that burden to them by default.

I think you misunderstood me.  When I configure a gateway for outgoing
calls then the process seems reasonably clear. What's less clear (and
this was mentioned in today's conference call) is that for an incoming
call to the number that the SIP provider has determined with me that
the call SHOULD only come from them (be that their proxy host, or
maybe a slightly wider network they own). I may be wrong but I don't
see how that is [optionally] enforced with the current default
configuration. That is even if an attacker knows my sip account, or
DID number and password providers password details, if they attempt to
call in from a different host or network I want that call to fail. So
I'd like a way to specify how to tighten incoming calls to that number
to that tighter host/address range. I think a pointer was given to me
on IRC during the conference so I need to check my irc logs, but
it was not obvious to me before.

> 9. See #5

More reading :-)

> I hope we haven't scared you off! I'm sure that you can (and will) overcome
> all of your challenges in configuring FreeSWITCH for your needs. The
> community is here to assist.

Frankly I can only say thanks. I have not meant to be "hostile" but
perhaps my comments may seem that way to you. The irc, and email
exchanges really are helping and I'm making progress.

...

> *BONUS CONTENT:
> Here's an example of limiting a user's IP address. Add a line like this to
> your user's XML file:
> <variable name="allowed_ipaddrs" value="^(192\.168\.1\.|10\.10\.16\.161)"/>
> This is a regex that matches an exact address (10.10.16.161) or matches a
> partial IP address (192.168.1.x)
> 
> Then do something like this in your dialplan:
>   <extension name="check user's IP address" continue="true">
>     <condition field="network_addr" expression="${allowed_ipaddrs}">
>       <action application="log" data="WARNING IP Addr test passed"/>
>       <anti-action application="log" data="ERR IP Addr test is an EPIC
> FAIL"/>
>       <anti-action application="answer"/>
>       <anti-action application="sleep" data="1000"/>
>       <anti-action application="playback"
> data="ivr/ivr-all_your_call_are_belong_to_us.wav"/>
>       <anti-action application="hangup"/>
>     </condition>
>   </extension>

I guess this is similar to the implementation of the solution I found above.
Thanks for showing there's another way to do that.

Simon 

p.s. I know that mail list messages and replies should be kept
short. Sorry this is completely breaking that convention.


From infos at madovsky.org  Thu Feb 10 03:42:30 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 9 Feb 2011 19:42:30 -0500
Subject: [Freeswitch-users] ringback settings
Message-ID: <84A1A2CB83334458BC983DCCEEC2381C@e1705>

I don't find the right settings
to get ringback from a mobile phone and landline
when I call to a DID<
I put instant_ringback=true and/or application ring_ready
but no success

thanks
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From curriegrad2004 at gmail.com  Thu Feb 10 04:13:35 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Wed, 9 Feb 2011 17:13:35 -0800
Subject: [Freeswitch-users] ringback settings
In-Reply-To: <84A1A2CB83334458BC983DCCEEC2381C@e1705>
References: <84A1A2CB83334458BC983DCCEEC2381C@e1705>
Message-ID: <AANLkTi=44B0MDipZD35GDuB9c3P3kbCaCqe8Tmg2zoEP@mail.gmail.com>

Maybe your provider is blocking the ringbacks set by you?

On Wed, Feb 9, 2011 at 4:42 PM, Madovsky <infos at madovsky.org> wrote:
> I don't find the right settings
> to get ringback from a mobile phone and landline
> when I call to a DID<
> I put instant_ringback=true and/or application ring_ready
> but no success
>
> 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  Thu Feb 10 04:32:16 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 9 Feb 2011 20:32:16 -0500
Subject: [Freeswitch-users] ringback settings
References: <84A1A2CB83334458BC983DCCEEC2381C@e1705>
	<AANLkTi=44B0MDipZD35GDuB9c3P3kbCaCqe8Tmg2zoEP@mail.gmail.com>
Message-ID: <885A0865F43C4AA191D6F96C7EA42A8E@e1705>

So what the solution ?


----- Original Message ----- 
From: "curriegrad2004" <curriegrad2004 at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, February 09, 2011 8:13 PM
Subject: Re: [Freeswitch-users] ringback settings


> Maybe your provider is blocking the ringbacks set by you?
> 
> On Wed, Feb 9, 2011 at 4:42 PM, Madovsky <infos at madovsky.org> wrote:
>> I don't find the right settings
>> to get ringback from a mobile phone and landline
>> when I call to a DID<
>> I put instant_ringback=true and/or application ring_ready
>> but no success
>>
>> 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 dujinfang at gmail.com  Thu Feb 10 04:32:22 2011
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 10 Feb 2011 09:32:22 +0800
Subject: [Freeswitch-users] git HEAD 503 Maximum Calls In Progress
In-Reply-To: <AANLkTi=n72vrWJDKB4G2yNkwm+8hdgnwAuCKBPP7xac2@mail.gmail.com>
References: <AANLkTimpGj4GLug8jrK-7-V3Mja9W2tTiwtJsBSu-TZo@mail.gmail.com>
	<AANLkTi=n72vrWJDKB4G2yNkwm+8hdgnwAuCKBPP7xac2@mail.gmail.com>
Message-ID: <AANLkTimnQ+68QR+v-1-rUqyD-YkXA8RKh+Na=8Fvmv6v@mail.gmail.com>

Thank you Anthony, my bad. The server was running mysql and after I
installed postgresql cpu load > 1.0 and 0% idle. I did run
/etc/init.d/mysql stop but mysql-safe still left running which I
didn't noticed, after I killed mysql all problems gone.

I will left the min-idle-cpu untouched until I run into problems. Thanks again.

On Wed, Feb 9, 2011 at 11:57 PM, Steven Ayre <steveayre at gmail.com> wrote:
> Upgrade to 64bit if you can
>

I know 64bit is best but I was trying to help a friend to setup a
simple IVR with lua under 10 channels, I think it's good to use
existing hardwares. Actually the server has a pentium4 CPU with 1G
mem, it must be 10+ year old.

Thanks Steven.

>
> On 9 February 2011 11:37, Seven Du <dujinfang at gmail.com> wrote:
>>
>> Hi,
>>
>> It's a fresh install with wget ?freeswitch.org/eg/Makefile && make,
>> and I'm sure I successfully called 9196 and everything was ok.
>>
>> But a few minutes later I started getting
>>
>> ?503 Maximum Calls In Progress
>>
>> also originate user/1003 or loopback/9196 shows errors
>>
>> see
>> http://pastebin.freeswitch.org/15333
>>
>> It's on a ubuntu 8.04 32bit, the only thing I did between non-work and
>> work was I configured odbc with postgresql to work with isql. I
>> haven't touch the FS config yet. I don't think they are related.
>>
>>
>> here is also a nua debug
>>
>> http://pastebin.freeswitch.org/15335
>>
>> Any hint on this? Thanks.
>>
>> btw, I also noticed there might be some network problems between me
>> and the server where sometime sip packets not shown ( sofia debug and
>> ngrep) on server. Anyway, since loopback cannot create sessions so
>> something must be wrong in FS side, but what's that?
>>
>> Thanks.
>>
>> --
>> 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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 curriegrad2004 at gmail.com  Thu Feb 10 04:37:18 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Wed, 9 Feb 2011 17:37:18 -0800
Subject: [Freeswitch-users] ringback settings
In-Reply-To: <885A0865F43C4AA191D6F96C7EA42A8E@e1705>
References: <84A1A2CB83334458BC983DCCEEC2381C@e1705>
	<AANLkTi=44B0MDipZD35GDuB9c3P3kbCaCqe8Tmg2zoEP@mail.gmail.com>
	<885A0865F43C4AA191D6F96C7EA42A8E@e1705>
Message-ID: <AANLkTi=xaX5ebztQ+qaJpLRX33++mX8ba2--=Nfzk+69@mail.gmail.com>

What I'd do is make it 'answer' the call first then send it to
whatever extension you want it to. Disadvantage to this is the person
on the calling end will have to pay for the call as soon as the switch
'answers' the call.

Just out of curiosity, who's your DID provider that's doing this?

On Wed, Feb 9, 2011 at 5:32 PM, Madovsky <infos at madovsky.org> wrote:
> So what the solution ?
>
>
> ----- Original Message -----
> From: "curriegrad2004" <curriegrad2004 at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, February 09, 2011 8:13 PM
> Subject: Re: [Freeswitch-users] ringback settings
>
>
>> Maybe your provider is blocking the ringbacks set by you?
>>
>> On Wed, Feb 9, 2011 at 4:42 PM, Madovsky <infos at madovsky.org> wrote:
>>> I don't find the right settings
>>> to get ringback from a mobile phone and landline
>>> when I call to a DID<
>>> I put instant_ringback=true and/or application ring_ready
>>> but no success
>>>
>>> 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  Thu Feb 10 04:47:23 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 9 Feb 2011 20:47:23 -0500
Subject: [Freeswitch-users] ringback settings
Message-ID: <0512E8A712BE4ABB893F818E3E67DCBD@e1705>

 found the problem,

before bridge I put

                        <action application="set_audio_level" data="read 
1"/>
                        <action application="set_audio_level" data="write 
1"/>
FS doesn't like it. I guess it needs the bridge done to run them

Any ?

THanks

----- Original Message ----- 
From: "Madovsky" <infos at madovsky.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, February 09, 2011 8:32 PM
Subject: Re: [Freeswitch-users] ringback settings


> So what the solution ?
>
>
> ----- Original Message ----- 
> From: "curriegrad2004" <curriegrad2004 at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, February 09, 2011 8:13 PM
> Subject: Re: [Freeswitch-users] ringback settings
>
>
>> Maybe your provider is blocking the ringbacks set by you?
>>
>> On Wed, Feb 9, 2011 at 4:42 PM, Madovsky <infos at madovsky.org> wrote:
>>> I don't find the right settings
>>> to get ringback from a mobile phone and landline
>>> when I call to a DID<
>>> I put instant_ringback=true and/or application ring_ready
>>> but no success
>>>
>>> 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 fs-list at communicatefreely.net  Thu Feb 10 05:09:21 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Wed, 09 Feb 2011 21:09:21 -0500
Subject: [Freeswitch-users] Best way to implement real-time ivr
	from	mysql
In-Reply-To: <BE42DE63-CA66-40C8-A0BF-17AE303C77B1@ipeva.fr>
References: <4D52C739.8080502@communicatefreely.net>
	<BE42DE63-CA66-40C8-A0BF-17AE303C77B1@ipeva.fr>
Message-ID: <4D5348D1.4010709@communicatefreely.net>

Actually, no.

I was using the lua odbc library that I installed from a package.  I 
didn't know about Dbh()  That might solve some problems, as I don't 
really have any issues with Freeswitch using ODBC for core and things 
like voicemail.  It's only when I want to interact with it from a script 
that it's a problem.

xml_curl is working very nicely for routing, directory, and config, but 
this may be a better solution for things like IVR and other more 
complicated features.

Thanks!

-Tim

David Ponzone wrote:
> Tim,
>
> which ODBC is that ? the one integrated in FreeSWITCH that you can 
> call from LUA with freeswitch.Dbh() ?
> I would say that solving those issues would be a better way to achieve 
> this, for you and for everyone around.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 09/02/2011 ? 17:56, Tim St. Pierre a ?crit :
>
>> Hello list,
>>
>> I'm trying to implement a flexible, but easy to administer IVR system
>> for our multi-tenant PBX.
>>
>> My goal is to be able to build a flexible auto-attendent scheme for
>> incoming calls, based on data in a mysql database.
>>
>> The functionality I need is:
>>    -Different options based on time of day
>>    -Playback of 1-2 audio files while listening for digits (Thank you
>> for calling abc company)
>>    -Single digit options set some variables, then transfer to a defined
>> extension (press 1 for sales)
>>    -Multi-digit options are checked against a pattern, then a transfer
>> is executed (enter the extension now ...)
>>
>> Not very complicated, I know, but making it scale is tricky.
>>
>> I have tried LUA, but LuaSQL has issues.  Even using ODBC, I still get
>> memory leaks and random errors with file handles etc.
>>
>> For the most part, I haven't had any issues with xml_curl getting config
>> from another server that generates it with PHP.  I'm exploring this, but
>> it looks like I'll have to implement the following logic:
>>    -Call comes in, and a dialplan is returned that plays the greetings,
>> sets some variables, and does a play_and_get_digits, followed by a
>> transfer back to the dialplan, so we can figure out what to do based on
>> the digits.
>>    -The callback to the dialplan evaluates the dtmf presented, and
>> decides what action to take, returning that in another dialplan piece.
>>
>> I think this could work. but it means that a dialplan lookup has to be
>> done each time someone makes a selection.
>>
>> Is there a better way, or does that make the most sense?
>>
>> Thanks!
>>
>> -Tim
>>
>> _______________________________________________
>> 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 fs-list at communicatefreely.net  Thu Feb 10 05:18:37 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Wed, 09 Feb 2011 21:18:37 -0500
Subject: [Freeswitch-users] Aastra phone registration lost
In-Reply-To: <AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>	<4D52B987.7050002@communicatefreely.net>
	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>
Message-ID: <4D534AFD.3020401@communicatefreely.net>

I'm wondering about these values:

> sip registration retry timer: 10
> sip registration renewal timer: 15
> sip registration timeout retry timer: 10
> sip registration period: 60
> sip rport: 1
>   
They seem a bit short, especially on a LAN.  Do you really need your 
phones to re-register every 45 seconds?

Try setting sip registration period to 600  That's 10 minutes.
sip registration renewal timer can be a bit longer - 30 seconds maybe, 
or 60 if you are at all worried about reliability.

The sip registration renewal timer sets how long BEFORE expiry that the 
phone will re-register.  I wonder if that value is too short, and 
something is rejecting it as invalid.

Here's a section from my default config.  It seems to work fairly well.

If that doesn't work, we probably need to check the sofia profile.

Let me know how you make out.

-Tim

sip dtmf method: 0
sip rtp port: 10000
sip mode: 0
sip silence suppression: 0
sip digit time out: 3
sip update callerid: 0
sip send mac: 1
sip rport: 0
sip blf subscription period: 1800
sip registration period: 3600
sip registration renewal timer: 1200
sip T2 timer: 2000
sip transaction timer: 12000
sip use basic codecs: 0
sip customized codec: 
payload=115;ptime=20;silsupp=off,payload=97;ptime=20;silsupp=off,payload=9;ptime=20;silsupp=off,payload=0;ptime=20;silsupp=of$



From marcin321 at gmail.com  Thu Feb 10 05:41:11 2011
From: marcin321 at gmail.com (Marcin Wojtowicz)
Date: Wed, 9 Feb 2011 21:41:11 -0500
Subject: [Freeswitch-users] MOH not working in G729 passthru with native
	sound files
Message-ID: <AANLkTi=41CbE_tU8LsW52rZCaD0Z+Y8NbE7WXh+bxTn1@mail.gmail.com>

I'm trying to enable MOH when both legs of the call are using G729 (FS is in
passthru). I converted an edited sample wave file to G729 and put in the
appropriate folder, and FS loads it correctly because this is the message
that keeps popping up in the console:
2011-02-09 21:10:25.609375 [INFO] mod_native_file.c:94 Opening File
[C:\freeswitch\sounds/music/8000\sample.G729] 8000hz

I establish a call, and everything is fine, but when I press hold on my
handset I see an error message that says that G729 is only useable in
passthru (here is the debug message):

2011-02-09 21:14:01.828125 [DEBUG] switch_channel.c:1376 (sofia/internal/
sip:M9jdt73ig0oOJSbt6Uyy at 192.168.1.50:5060) Callstate Change ACTIVE -> HELD
2011-02-09 21:14:01.828125 [DEBUG] switch_core_session.c:954 Send signal
sofia/external/MYHOME#@74.63.41.218 [BREAK]
2011-02-09 21:14:01.859375 [DEBUG] switch_core_session.c:709 Send signal
sofia/external/MYHOME#@74.63.41.218 [BREAK]
2011-02-09 21:14:02.000000 [DEBUG] switch_ivr.c:563 sofia/external/MYHOME#@
74.63.41.218 Command Execute playback(local_stream://moh/8000)
EXECUTE sofia/external/MYHOME#@74.63.41.218playback(local_stream://moh/8000)
2011-02-09 21:14:02.000000 [DEBUG] mod_local_stream.c:421 Opening Stream
[moh/8000] 8000hz
2011-02-09 21:14:02.000000 [DEBUG] switch_ivr_play_say.c:1244 Codec
Activated L16 at 8000hz 1 channels 20ms
2011-02-09 21:14:02.015625 [ERR] mod_g729.c:102 This codec is only usable in
passthrough mode!
2011-02-09 21:14:02.015625 [ERR] switch_core_io.c:1042 Codec G.729 encoder
error!
2011-02-09 21:14:02.015625 [DEBUG] switch_ivr_play_say.c:1581 done playing
file

I don't understand why that would be, since my music file is in G729 so I'm
not asking freeswitch to convert, only stream. My custom ringback (before a
call is established) works just fine using a similar method, so could anyone
explain me why what I want to do is not permitted?
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From brian at freeswitch.org  Thu Feb 10 06:05:32 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 9 Feb 2011 21:05:32 -0600
Subject: [Freeswitch-users] MOH not working in G729 passthru with native
	sound files
In-Reply-To: <AANLkTi=41CbE_tU8LsW52rZCaD0Z+Y8NbE7WXh+bxTn1@mail.gmail.com>
References: <AANLkTi=41CbE_tU8LsW52rZCaD0Z+Y8NbE7WXh+bxTn1@mail.gmail.com>
Message-ID: <E817B540-29EA-4906-BCF7-6274A03D29D8@freeswitch.org>

You would need to set the moh to the file directly and not use local_stream.  I'm pretty sure that local_stream might not support native files.

/b

On Feb 9, 2011, at 8:41 PM, Marcin Wojtowicz wrote:

> I'm trying to enable MOH when both legs of the call are using G729 (FS is in passthru). I converted an edited sample wave file to G729 and put in the appropriate folder, and FS loads it correctly because this is the message that keeps popping up in the console:
> 2011-02-09 21:10:25.609375 [INFO] mod_native_file.c:94 Opening File [C:\freeswitch\sounds/music/8000\sample.G729] 8000hz
> 
> I establish a call, and everything is fine, but when I press hold on my handset I see an error message that says that G729 is only useable in passthru (here is the debug message):
> 
> 2011-02-09 21:14:01.828125 [DEBUG] switch_channel.c:1376 (sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy at 192.168.1.50:5060) Callstate Change ACTIVE -> HELD
> 2011-02-09 21:14:01.828125 [DEBUG] switch_core_session.c:954 Send signal sofia/external/MYHOME#@74.63.41.218 [BREAK]
> 2011-02-09 21:14:01.859375 [DEBUG] switch_core_session.c:709 Send signal sofia/external/MYHOME#@74.63.41.218 [BREAK]
> 2011-02-09 21:14:02.000000 [DEBUG] switch_ivr.c:563 sofia/external/MYHOME#@74.63.41.218 Command Execute playback(local_stream://moh/8000)
> EXECUTE sofia/external/MYHOME#@74.63.41.218 playback(local_stream://moh/8000)
> 2011-02-09 21:14:02.000000 [DEBUG] mod_local_stream.c:421 Opening Stream [moh/8000] 8000hz
> 2011-02-09 21:14:02.000000 [DEBUG] switch_ivr_play_say.c:1244 Codec Activated L16 at 8000hz 1 channels 20ms
> 2011-02-09 21:14:02.015625 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode!
> 2011-02-09 21:14:02.015625 [ERR] switch_core_io.c:1042 Codec G.729 encoder error!
> 2011-02-09 21:14:02.015625 [DEBUG] switch_ivr_play_say.c:1581 done playing file
> 
> I don't understand why that would be, since my music file is in G729 so I'm not asking freeswitch to convert, only stream. My custom ringback (before a call is established) works just fine using a similar method, so could anyone explain me why what I want to do is not permitted?
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 10 06:24:32 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 9 Feb 2011 19:24:32 -0800 (PST)
Subject: [Freeswitch-users] git-a93623 version (2011/02/07) causes
 [DESTINATION_OUT_OF_ORDER]
In-Reply-To: <AANLkTi=kTEKr2n7MZy4wuvz5OpsGkD6zF=sNau4Dzu81@mail.gmail.com>
References: <1297183735886-6004663.post@n2.nabble.com>
	<AANLkTimOM9dZYr=zrQB+eJAEQ7fh0ugvqJQueOtOGiZH@mail.gmail.com>
	<1297185192140-6004757.post@n2.nabble.com>
	<AANLkTikzn43AqJaEUaWFpa0Yy70KLz-64Cb-8Jk6Fo8W@mail.gmail.com>
	<1297189519810-6005004.post@n2.nabble.com>
	<AANLkTing-OkaLem1HLJPQSCaSORSvLZZk-m4M0cXDqFn@mail.gmail.com>
	<1297261841390-6007834.post@n2.nabble.com>
	<AANLkTi=kTEKr2n7MZy4wuvz5OpsGkD6zF=sNau4Dzu81@mail.gmail.com>
Message-ID: <1297308272688-6010429.post@n2.nabble.com>



Anthony Minessale wrote:
> 
> edit conf/autoload_configs/switch.conf.xml
> 
> comment out
> 
> 
> 
> It seems to not work well on 32 bit.
I just did another git pull, recompiled, and tested the code. I made sure
the above requirement is met. Unfortunately, outbound calls still behaved as
before, i.e. sometimes went through and sometimes dead with
[DESTINATION_OUT_OF_ORDER]. To make the matter worst, this
[DESTINATION_OUT_OF_ORDER] thing also seems to affect incoming calls on my
GV trunk served by mod_dingaling. The following is an excerpt from fs_cli
during an incoming call to my GV trunk and the phone did not ring.

FreeSWITCH Version 1.0.head (git-6f9da9a 2011-02-09 13-27-04 -0500)
2011-02-09 22:02:38.190253 [DEBUG] mod_dingaling.c:1008 Send Candidate
192.168.1.123:29448 [d9sZPhj5h6N1RQlA]
2011-02-09 22:02:38.190253 [DEBUG] mod_dingaling.c:3073 Creating an identity
for SIP2041544632 at 10.142.218.17 Google Voice <+14141234567> 1003
2011-02-09 22:02:38.193847 [NOTICE] switch_channel.c:811 New Channel
dingaling/1003 [222a6b26-3927-40ed-91c6-ad07f82e04c3]
2011-02-09 22:02:38.196099 [DEBUG] mod_dingaling.c:3101 Creating a session
for SIP2041544632 at 10.142.218.17
2011-02-09 22:02:38.199360 [NOTICE] switch_channel.c:809 Rename Channel
dingaling/1003->DingaLing/new [222a6b26-3927-40ed-91c6-ad07f82e04c3]
2011-02-09 22:02:38.202711 [DEBUG] mod_dingaling.c:3105 (DingaLing/new)
State Change CS_NEW -> CS_INIT
2011-02-09 22:02:38.205320 [DEBUG] switch_core_session.c:1116 Send signal
DingaLing/new [BREAK]
2011-02-09 22:02:38.206707 [DEBUG] mod_dingaling.c:1348 DingaLing/new
CHANNEL KILL
2011-02-09 22:02:38.208188 [DEBUG] mod_dingaling.c:3207 2 payloads
2011-02-09 22:02:38.208188 [DEBUG] mod_dingaling.c:3209 Available Payload
PCMU 0
2011-02-09 22:02:38.208188 [DEBUG] mod_dingaling.c:3217 compare PCMU 0/8000
to PCMU 0/8000
2011-02-09 22:02:38.208188 [DEBUG] mod_dingaling.c:3228 Choosing Payload
index 0 PCMU 0
2011-02-09 22:02:38.208188 [DEBUG] mod_dingaling.c:1083 Send Describe
[PCMU at 8000]
2011-02-09 22:02:38.209352 [DEBUG] switch_core_state_machine.c:320
(DingaLing/new) Running State Change CS_INIT
2011-02-09 22:02:38.210441 [DEBUG] switch_core_state_machine.c:356
(DingaLing/new) State INIT
2011-02-09 22:02:38.210441 [NOTICE] mod_dingaling.c:1110 Ring-Ready
DingaLing/new!
2011-02-09 22:02:38.454349 [DEBUG] mod_dingaling.c:2941 using Existing
session for SIP2041544632 at 10.142.218.17
2011-02-09 22:02:38.454349 [DEBUG] mod_dingaling.c:3279 3 candidates
2011-02-09 22:02:38.458007 [DEBUG] mod_dingaling.c:3299 candidate
74.125.45.126:19295 PASS ACL wan.auto
2011-02-09 22:02:38.460625 [DEBUG] mod_dingaling.c:3351 Acceptable Candidate
74.125.45.126:19295
2011-02-09 22:02:38.463024 [DEBUG] mod_dingaling.c:865 Set Read Codec to
PCMU at 8000
2011-02-09 22:02:38.465544 [DEBUG] mod_dingaling.c:880 Set Write Codec to
PCMU at 8000
2011-02-09 22:02:38.470235 [DEBUG] mod_dingaling.c:892 SETUP RTP
192.168.1.123:0 -> 74.125.45.126:19295
2011-02-09 22:02:38.470235 [DEBUG] mod_dingaling.c:914 RTP ERROR Missing
local port
2011-02-09 22:02:38.473532 [DEBUG] switch_channel.c:2538 (DingaLing/new)
Callstate Change DOWN -> HANGUP
2011-02-09 22:02:38.476901 [NOTICE] mod_dingaling.c:915 Hangup DingaLing/new
[CS_INIT] [DESTINATION_OUT_OF_ORDER]
2011-02-09 22:02:38.480237 [DEBUG] switch_channel.c:2554 Send signal
DingaLing/new [KILL]
2011-02-09 22:02:38.480237 [DEBUG] mod_dingaling.c:1348 DingaLing/new
CHANNEL KILL
2011-02-09 22:02:38.482580 [DEBUG] switch_core_session.c:1116 Send signal
DingaLing/new [BREAK]
2011-02-09 22:02:38.485926 [DEBUG] mod_dingaling.c:1348 DingaLing/new
CHANNEL KILL
2011-02-09 22:02:38.488213 [DEBUG] mod_dingaling.c:710 Terminate called from
line 1177 state=CS_HANGUP
2011-02-09 22:02:38.490382 [DEBUG] switch_core_state_machine.c:356
(DingaLing/new) State INIT going to sleep
2011-02-09 22:02:38.492582 [DEBUG] switch_core_state_machine.c:320
(DingaLing/new) Running State Change CS_HANGUP
2011-02-09 22:02:38.499246 [DEBUG] switch_core_state_machine.c:557
(DingaLing/new) State HANGUP
2011-02-09 22:02:38.499246 [DEBUG] mod_dingaling.c:1317 DingaLing/new
CHANNEL HANGUP
2011-02-09 22:02:38.499246 [DEBUG] switch_core_state_machine.c:46
DingaLing/new Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER
2011-02-09 22:02:38.499246 [DEBUG] switch_core_state_machine.c:557
(DingaLing/new) State HANGUP going to sleep
2011-02-09 22:02:38.499246 [DEBUG] switch_core_state_machine.c:351
(DingaLing/new) State Change CS_HANGUP -> CS_REPORTING
2011-02-09 22:02:38.500342 [DEBUG] switch_core_session.c:1116 Send signal
DingaLing/new [BREAK]
2011-02-09 22:02:38.500342 [DEBUG] mod_dingaling.c:1348 DingaLing/new
CHANNEL KILL
2011-02-09 22:02:38.500342 [DEBUG] switch_core_state_machine.c:320
(DingaLing/new) Running State Change CS_REPORTING
2011-02-09 22:02:38.501440 [DEBUG] switch_core_state_machine.c:617
(DingaLing/new) State REPORTING
2011-02-09 22:02:38.762679 [DEBUG] switch_core_state_machine.c:53
DingaLing/new Standard REPORTING, cause: DESTINATION_OUT_OF_ORDER
2011-02-09 22:02:38.762679 [DEBUG] switch_core_state_machine.c:617
(DingaLing/new) State REPORTING going to sleep
2011-02-09 22:02:38.762679 [DEBUG] switch_core_state_machine.c:345
(DingaLing/new) State Change CS_REPORTING -> CS_DESTROY
2011-02-09 22:02:38.762679 [DEBUG] switch_core_session.c:1116 Send signal
DingaLing/new [BREAK]
2011-02-09 22:02:38.766170 [DEBUG] mod_dingaling.c:1348 DingaLing/new
CHANNEL KILL
2011-02-09 22:02:38.769238 [DEBUG] switch_core_session.c:1288 Session 1
(DingaLing/new) Locked, Waiting on external entities
2011-02-09 22:02:38.771555 [NOTICE] switch_core_session.c:1306 Session 1
(DingaLing/new) Ended
2011-02-09 22:02:38.773948 [NOTICE] switch_core_session.c:1308 Close Channel
DingaLing/new [CS_DESTROY]
2011-02-09 22:02:38.778327 [DEBUG] switch_core_state_machine.c:449
(DingaLing/new) Callstate Change HANGUP -> DOWN
2011-02-09 22:02:38.782464 [DEBUG] switch_core_state_machine.c:452
(DingaLing/new) Running State Change CS_DESTROY
2011-02-09 22:02:38.784153 [DEBUG] switch_core_state_machine.c:462
(DingaLing/new) State DESTROY
2011-02-09 22:02:38.785378 [DEBUG] switch_core_state_machine.c:60
DingaLing/new Standard DESTROY
2011-02-09 22:02:38.788679 [DEBUG] switch_core_state_machine.c:462
(DingaLing/new) State DESTROY going to sleep
2011-02-09 22:02:52.591019 [DEBUG] mod_dingaling.c:2951 Session is already
dead
The above happened with the following dingaling profile:
<profile type="client">
  
  
  
  
  
  
  
  
  
  
  
  
  
  
  
  
  
  
  
</profile>
If I used '<'param name="ext-rtp-ip" value="$${external_rtp_ip}"/'>' and
'<'param name="rtp-ip" value="$${bind_server_ip}"/'>', the phone will ring
but is a dead air when the telephone handset is picked up and the caller
keeps hearing the ringback tones.

-----
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/git-a93623-version-2011-02-07-causes-DESTINATION-OUT-OF-ORDER-tp6004663p6010429.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Thu Feb 10 06:29:18 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 9 Feb 2011 19:29:18 -0800 (PST)
Subject: [Freeswitch-users] How to dial a remote_skypename from a
	telephone keypad?
In-Reply-To: <AANLkTik0wPtXNOcd95wX2TH8B0+BaY7mz3aeBZaWa3hi@mail.gmail.com>
References: <1297263654900-6007946.post@n2.nabble.com>
	<AANLkTikfXRGxqbrassDNhOnaOWkaje3DXQptRAVRzb9d@mail.gmail.com>
	<AANLkTikJNH3-vk=dMkOjAKsAxgr7KaiGD-U-9t2uKjxE@mail.gmail.com>
	<1297272133009-6008667.post@n2.nabble.com>
	<AANLkTikdMDk_NkKVyH6Qd4DBq5izujG77eYPOi1ZCbms@mail.gmail.com>
	<1297276772026-6009014.post@n2.nabble.com>
	<AANLkTikDg6yJVHh7M=q0cNkacwhXpJBV5kJSThTaJGTZ@mail.gmail.com>
	<AANLkTik0wPtXNOcd95wX2TH8B0+BaY7mz3aeBZaWa3hi@mail.gmail.com>
Message-ID: <1297308558454-6010434.post@n2.nabble.com>



Dmitry Bely wrote:
> # button can be used to start encoding sequence and ## is to end it.
> So e.g. giovanni334 is be encoded as
> 
> #4#444#666#888#2#66#66#444##334
> 
> or even
> 
> #4#444666888266#66444##334 (if digit changes we know that another
> character is started. Thus # prefix can be safely omitted)
> 
> Then all we would need is a very simple decoder on the
> freeswitch/skypopen side. Just an idea...
What will happen if a Skype account is named Skype333 or Skype444?

-----
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/How-to-dial-a-remote-skypename-from-a-telephone-keypad-tp6007946p6010434.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From gabe at gundy.org  Thu Feb 10 09:23:12 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Wed, 9 Feb 2011 23:23:12 -0700
Subject: [Freeswitch-users] Yet another DTMF question
In-Reply-To: <AANLkTi=Trve2-zDb60FEahz2TLD_=BAvhzisf7gSOoFo@mail.gmail.com>
References: <AANLkTik030ecFw2bvKe74pOs+NzfqHknX8keDO3mQ=cH@mail.gmail.com>
	<AANLkTikqQodjsMT_VB1xeQdtY-2DEgL68ndEnHTBTvkb@mail.gmail.com>
	<AANLkTiksK3T24TUDO-7aq_tbkFCkj_4BRNpLRt4=GtBj@mail.gmail.com>
	<62C5671A-8657-4F99-8184-272EEBEC8E77@gmail.com>
	<AANLkTimFMwJ1EkP_+1uCyiKUCKocgh-mSYVEd39navS7@mail.gmail.com>
	<AANLkTikaerx0b+A7EhFLXxuBKBBY+RBog8Lpgyd1ZZmR@mail.gmail.com>
	<AANLkTinL1NxOjMJjShOwXEo5CJERLbY4N7Z7re=QRp8z@mail.gmail.com>
	<AANLkTi=Trve2-zDb60FEahz2TLD_=BAvhzisf7gSOoFo@mail.gmail.com>
Message-ID: <AANLkTi=d_Bzmt+5s+RaGDH6KckBszmzJjsTq2G2JvRZP@mail.gmail.com>

On Wed, Jan 19, 2011 at 10:32 AM, Steven Ayre <steveayre at gmail.com> wrote:
> Try uploading to the latest Git, a post from Brian just reminded me that
> there was a day where some changes to the RTP stack broke RFC2833 support,
> perhaps you have one of the bad versions.

Anyone know what day this might have been?  I WILL update, but it
would be good to know if my issues could also be related.

Thanks,
Gabe


From admin at blindi.net  Thu Feb 10 09:29:45 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Thu, 10 Feb 2011 07:29:45 +0100 (CET)
Subject: [Freeswitch-users] skypopen: problem loading the custom alsa
 driver
In-Reply-To: <AANLkTiniayXKmqZZo7HDPYwtR3b4X0vSAHPqTeOpcpEL@mail.gmail.com>
References: <AANLkTin=oJpzotF4x4+eYKKrazUaKxoKOdx9yAWA7ewD@mail.gmail.com>
	<Pine.LNX.4.64.1102091322460.28387@suse.blindi.net>
	<AANLkTi=p5U4Q7tGaz4=pErzhi8esy+gZpJ7U_NL4SESY@mail.gmail.com>
	<Pine.LNX.4.64.1102091458490.29294@suse.blindi.net>
	<AANLkTikDzWp=k42132oyTnM3tzWQ-3jTkuoou0g9EDc8@mail.gmail.com>
	<Pine.LNX.4.64.1102091536150.29294@suse.blindi.net>
	<AANLkTiniayXKmqZZo7HDPYwtR3b4X0vSAHPqTeOpcpEL@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1102100720120.5671@suse.blindi.net>

Am 09.02.11 um 16:03 schrieb Giovanni Maruzzelli:

> maybe I have not understood your last mail.

I have installed a Debian squeeze, but the problem is the same.
then i install alsa from debian original package "apt-get install -y alsa 
alsa-utils" snd_dummy is loaded successfuly.
then i compile alsa from source, the modules are installed, but the module 
can.t be loadet.
Is the alsaversion-source  incompatible to the actual 2.6.32 kernels?
Is 1.0.20 to old?


---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From gabe at gundy.org  Thu Feb 10 09:35:09 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Wed, 9 Feb 2011 23:35:09 -0700
Subject: [Freeswitch-users] Polycom and registering with domains
	(user@domain.tld@domain.tld)
In-Reply-To: <AANLkTikieRwaDVgNhPRd1BSV1gHFkYyzdiUoASVLhbij@mail.gmail.com>
References: <AANLkTikb2Mzw9oHagSHE=oe8ABSwOWOxwRE6CdNSO8WS@mail.gmail.com>
	<AANLkTikieRwaDVgNhPRd1BSV1gHFkYyzdiUoASVLhbij@mail.gmail.com>
Message-ID: <AANLkTikQs4oWyKoc3yTPPj6AMkobUXyYR6n_nPmhTmMP@mail.gmail.com>

On Sat, Jan 29, 2011 at 2:35 PM, Aloysius Lloyd
<lloyd.aloysius at gmail.com> wrote:
> I had the similar issues with Polycom phones. Stay away from the web
> configuration.
> All you need to setup the FTP or TFTP server and use the configuration
> files.

The Polycom phones are harder to get set up, but when you get things
configured correctly, it's pretty sweet.  Next up, my NATing issues
with Polycom. :)

Thanks for your input,
Gabe


From gmaruzz at gmail.com  Thu Feb 10 09:46:35 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 10 Feb 2011 07:46:35 +0100
Subject: [Freeswitch-users] skypopen: problem loading the custom alsa
	driver
In-Reply-To: <Pine.LNX.4.64.1102100720120.5671@suse.blindi.net>
References: <AANLkTin=oJpzotF4x4+eYKKrazUaKxoKOdx9yAWA7ewD@mail.gmail.com>
	<Pine.LNX.4.64.1102091322460.28387@suse.blindi.net>
	<AANLkTi=p5U4Q7tGaz4=pErzhi8esy+gZpJ7U_NL4SESY@mail.gmail.com>
	<Pine.LNX.4.64.1102091458490.29294@suse.blindi.net>
	<AANLkTikDzWp=k42132oyTnM3tzWQ-3jTkuoou0g9EDc8@mail.gmail.com>
	<Pine.LNX.4.64.1102091536150.29294@suse.blindi.net>
	<AANLkTiniayXKmqZZo7HDPYwtR3b4X0vSAHPqTeOpcpEL@mail.gmail.com>
	<Pine.LNX.4.64.1102100720120.5671@suse.blindi.net>
Message-ID: <AANLkTik2ggpE0TYduoBLriF3gfg5EEe2Gs7HoZPp+r43@mail.gmail.com>

if you prefer you can do try same procedure with alsa 1.0.23 but was
never reported an incompatibility of 1.0.20 with newer kernels.

Let us know how it goes

-giovanni

On Thu, Feb 10, 2011 at 7:29 AM, Thomas Hoellriegel <admin at blindi.net> wrote:
> Am 09.02.11 um 16:03 schrieb Giovanni Maruzzelli:
>
>> maybe I have not understood your last mail.
>
> I have installed a Debian squeeze, but the problem is the same.
> then i install alsa from debian original package "apt-get install -y alsa
> alsa-utils" snd_dummy is loaded successfuly.
> then i compile alsa from source, the modules are installed, but the module
> can.t be loadet.
> Is the alsaversion-source ?incompatible to the actual 2.6.32 kernels?
> Is 1.0.20 to old?
>
>
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/listinfo/blinde-misc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 david.ponzone at ipeva.fr  Thu Feb 10 10:00:11 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 10 Feb 2011 08:00:11 +0100
Subject: [Freeswitch-users] Best way to implement real-time ivr
	from	mysql
In-Reply-To: <4D5348D1.4010709@communicatefreely.net>
References: <4D52C739.8080502@communicatefreely.net>
	<BE42DE63-CA66-40C8-A0BF-17AE303C77B1@ipeva.fr>
	<4D5348D1.4010709@communicatefreely.net>
Message-ID: <2DA00912-4082-4173-BEFF-5DE90C0371C5@ipeva.fr>

I am quite happy with FS odbc access, but I don't use it extensively.
The only issue I have with it is the syntax to access the data, which I find less intuitive than with luaSQL.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 10/02/2011 ? 03:09, Tim St. Pierre a ?crit :

> Actually, no.
> 
> I was using the lua odbc library that I installed from a package.  I 
> didn't know about Dbh()  That might solve some problems, as I don't 
> really have any issues with Freeswitch using ODBC for core and things 
> like voicemail.  It's only when I want to interact with it from a script 
> that it's a problem.
> 
> xml_curl is working very nicely for routing, directory, and config, but 
> this may be a better solution for things like IVR and other more 
> complicated features.
> 
> Thanks!
> 
> -Tim
> 
> David Ponzone wrote:
>> Tim,
>> 
>> which ODBC is that ? the one integrated in FreeSWITCH that you can 
>> call from LUA with freeswitch.Dbh() ?
>> I would say that solving those issues would be a better way to achieve 
>> this, for you and for everyone around.
>> 
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> 
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com 
>> <BLOCKED::http://www.ipeva-studio.com/>
>> 
>> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
>> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
>> diffusion non autoris?e est interdite. Tout message ?lectronique est 
>> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
>> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
>> n'?tes pas destinataire de ce message, merci de le d?truire 
>> imm?diatement et d'avertir l'exp?diteur./
>> /
>> /
>> 
>> 
>> 
>> Le 09/02/2011 ? 17:56, Tim St. Pierre a ?crit :
>> 
>>> Hello list,
>>> 
>>> I'm trying to implement a flexible, but easy to administer IVR system
>>> for our multi-tenant PBX.
>>> 
>>> My goal is to be able to build a flexible auto-attendent scheme for
>>> incoming calls, based on data in a mysql database.
>>> 
>>> The functionality I need is:
>>>   -Different options based on time of day
>>>   -Playback of 1-2 audio files while listening for digits (Thank you
>>> for calling abc company)
>>>   -Single digit options set some variables, then transfer to a defined
>>> extension (press 1 for sales)
>>>   -Multi-digit options are checked against a pattern, then a transfer
>>> is executed (enter the extension now ...)
>>> 
>>> Not very complicated, I know, but making it scale is tricky.
>>> 
>>> I have tried LUA, but LuaSQL has issues.  Even using ODBC, I still get
>>> memory leaks and random errors with file handles etc.
>>> 
>>> For the most part, I haven't had any issues with xml_curl getting config
>>> from another server that generates it with PHP.  I'm exploring this, but
>>> it looks like I'll have to implement the following logic:
>>>   -Call comes in, and a dialplan is returned that plays the greetings,
>>> sets some variables, and does a play_and_get_digits, followed by a
>>> transfer back to the dialplan, so we can figure out what to do based on
>>> the digits.
>>>   -The callback to the dialplan evaluates the dtmf presented, and
>>> decides what action to take, returning that in another dialplan piece.
>>> 
>>> I think this could work. but it means that a dialplan lookup has to be
>>> done each time someone makes a selection.
>>> 
>>> Is there a better way, or does that make the most sense?
>>> 
>>> Thanks!
>>> 
>>> -Tim
>>> 
>>> _______________________________________________
>>> 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
> http://www.freeswitch.org

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From kbdfck at gmail.com  Thu Feb 10 10:03:57 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Thu, 10 Feb 2011 10:03:57 +0300
Subject: [Freeswitch-users] hold_music var gets unset/lost by
 cancelled/failed att_xfer
In-Reply-To: <AANLkTikt97dugbhKPgnB2ws4bOFcxg8yhzSdyxHMab_m@mail.gmail.com>
References: <AANLkTimLod3yHrex4LAK=4gdVcPhzXutKYyw=E8ZMcN9@mail.gmail.com>
	<AANLkTi=u3VeHPuS5x1e+yu5QNBw5ZSgHvAWOT+F6A_OR@mail.gmail.com>
	<AANLkTi=EndkiUJvCgCSirxCA23EwWD_++0h3Qcr=mg9g@mail.gmail.com>
	<AANLkTikt97dugbhKPgnB2ws4bOFcxg8yhzSdyxHMab_m@mail.gmail.com>
Message-ID: <AANLkTikEx9ZwNADRQ8k2JbTwgoLPFmHUPpPL3RLnZZU8@mail.gmail.com>

Thanks! It works now! BTW, att_xfer with
loopback_bowout/bowout_on_execute set to false seems to be working
too. Is there a way to make loopback channel leave the path? Without
bowout turned off att_xfer doesn't work...

2011/2/8 Anthony Minessale <anthony.minessale at gmail.com>:
> try latest GIT
>
> On Tue, Feb 8, 2011 at 1:28 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>> I have same issue with MOH and att_xfer on failed transfers, music on
>> hold played only once
>> At the same time, transfer_ringback always plays correctly to transferer
>>
>> 2011/2/8 Anthony Minessale <anthony.minessale at gmail.com>:
>>> you really should report this to jira not to the mailing list.
>>> http://jira.freeswitch.org
>>>
>>>
>>>
>>> On Mon, Feb 7, 2011 at 4:23 PM, Fraser Redmond <fraserredmond at gmail.com> wrote:
>>>> I'm trying to do a second att_xfer on a call so that if the first attended
>>>> transfer fails (c-leg is busy, or presses do-not-answer, or is an extn that
>>>> doesn't exist) then the call could be transferred to someone else.
>>>>
>>>> On the first att_xfer the person on hold hears the hold_music correctly.
>>>> Once that transfer is cancelled or fails:
>>>> -- On any subsequent att_xfer's the person on hold just hears silence.
>>>> -- If they are put on hold they just hear silence.
>>>>
>>>> I tried setting hold_music again for each channel after the first att_xfer,
>>>> but that didn't work, so it's probably not actually a problem with
>>>> hold_music per se, but some other variable/setting that decides whether to
>>>> use hold_music.
>>>>
>>>> I also tried doing a uuid_dump before and after each attempt, but didn't
>>>> notice anything too different - unless it's a matter of unsetting one of the
>>>> couple of changed/new vars like:
>>>> variable_originate_disposition
>>>> variable_current_application
>>>> variable_playback_seconds
>>>>
>>>> I get the feeling other variables are probably also lost by the first failed
>>>> transfer as the second att_xfer has some odd things happen if the third
>>>> party does answer. Haven't been able to narrow it down as closely as the
>>>> hold_music, but two things I've seen happen are:
>>>> -- The party that initiated the transfer gets hung up automatically (after
>>>> 30 sec)
>>>> -- When the party that initiated the transfer hangs up it should connect the
>>>> other two parties, but instead it hung up all three
>>>>
>>>> Cheers,
>>>> Fraser
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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


From steveayre at gmail.com  Thu Feb 10 11:46:24 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 10 Feb 2011 08:46:24 +0000
Subject: [Freeswitch-users] git HEAD 503 Maximum Calls In Progress
In-Reply-To: <AANLkTimnQ+68QR+v-1-rUqyD-YkXA8RKh+Na=8Fvmv6v@mail.gmail.com>
References: <AANLkTimpGj4GLug8jrK-7-V3Mja9W2tTiwtJsBSu-TZo@mail.gmail.com>
	<AANLkTi=n72vrWJDKB4G2yNkwm+8hdgnwAuCKBPP7xac2@mail.gmail.com>
	<AANLkTimnQ+68QR+v-1-rUqyD-YkXA8RKh+Na=8Fvmv6v@mail.gmail.com>
Message-ID: <AANLkTi=n4bcpPXzyh1sRYnYRbRRnhDETEB3Dij7c0E1U@mail.gmail.com>

>
> I know 64bit is best but I was trying to help a friend to setup a
> simple IVR with lua under 10 channels, I think it's good to use
> existing hardwares. Actually the server has a pentium4 CPU with 1G
> mem, it must be 10+ year old.
>

Fair enough. Some models since Prescott do actually support it by the way,
although most don't

-Steve




On 10 February 2011 01:32, Seven Du <dujinfang at gmail.com> wrote:

> Thank you Anthony, my bad. The server was running mysql and after I
> installed postgresql cpu load > 1.0 and 0% idle. I did run
> /etc/init.d/mysql stop but mysql-safe still left running which I
> didn't noticed, after I killed mysql all problems gone.
>
> I will left the min-idle-cpu untouched until I run into problems. Thanks
> again.
>
> On Wed, Feb 9, 2011 at 11:57 PM, Steven Ayre <steveayre at gmail.com> wrote:
> > Upgrade to 64bit if you can
> >
>
> I know 64bit is best but I was trying to help a friend to setup a
> simple IVR with lua under 10 channels, I think it's good to use
> existing hardwares. Actually the server has a pentium4 CPU with 1G
> mem, it must be 10+ year old.
>
> Thanks Steven.
>
> >
> > On 9 February 2011 11:37, Seven Du <dujinfang at gmail.com> wrote:
> >>
> >> Hi,
> >>
> >> It's a fresh install with wget  freeswitch.org/eg/Makefile && make,
> >> and I'm sure I successfully called 9196 and everything was ok.
> >>
> >> But a few minutes later I started getting
> >>
> >>  503 Maximum Calls In Progress
> >>
> >> also originate user/1003 or loopback/9196 shows errors
> >>
> >> see
> >> http://pastebin.freeswitch.org/15333
> >>
> >> It's on a ubuntu 8.04 32bit, the only thing I did between non-work and
> >> work was I configured odbc with postgresql to work with isql. I
> >> haven't touch the FS config yet. I don't think they are related.
> >>
> >>
> >> here is also a nua debug
> >>
> >> http://pastebin.freeswitch.org/15335
> >>
> >> Any hint on this? Thanks.
> >>
> >> btw, I also noticed there might be some network problems between me
> >> and the server where sometime sip packets not shown ( sofia debug and
> >> ngrep) on server. Anyway, since loopback cannot create sessions so
> >> something must be wrong in FS side, but what's that?
> >>
> >> Thanks.
> >>
> >> --
> >> 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
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
>
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From locutis at sect001.net  Wed Feb  9 22:33:02 2011
From: locutis at sect001.net (Locutis of Borg)
Date: Wed, 9 Feb 2011 14:33:02 -0500
Subject: [Freeswitch-users] mod_dingaling & inbound audio
In-Reply-To: <AANLkTik19FmdGVEukmC4rS3ytu9qhR=wfSkO-fhaRMJQ@mail.gmail.com>
References: <AANLkTim-5LDhfQhHk_2LNLffvTa1JyBXq9U4B3LSY3Nq@mail.gmail.com>
	<AANLkTik19FmdGVEukmC4rS3ytu9qhR=wfSkO-fhaRMJQ@mail.gmail.com>
Message-ID: <AANLkTikoJcrERbEHbdkLM_4ocnCUKkYDoKE-miYbvTxj@mail.gmail.com>

As instructed, I have posted the jingle config and a fs_cli debug log from
starting at the dialplan down through hangup.  Thank you for your help on
this.

Pastebin log and config under the name Locutis  :)

-Borg



On Tue, Feb 8, 2011 at 4:15 PM, Michael Collins <msc at freeswitch.org> wrote:

> use pastebin.freeswitch.org and pastebin your configs and a debug output
> of the failed call. be sure that you have debug level output (which is
> default if you use fs_cli)
> -MC
>
> On Tue, Feb 8, 2011 at 1:07 PM, Locutis of Borg <locutis at sect001.net>wrote:
>
>> Followed instructions on wiki.  Enabled mod_dingaling to register over at
>> GV.  Set external rtp to right address.  Outbound calling terminates, and
>> outbound audio works, but seems that I can not get return audio.
>>
>> FS is on DMZ - same problem
>> Moved to internal (behind NAT, port forwarding) - same problem
>> Moved back to DMZ and killed iptables and router spi - same problem
>>
>> In and out calling on SIP to provider works fine in both cases.
>>
>> So my questions are:
>>   Can mod_dingaling be used for GV or not?
>>   Since it uses TLS, NAT shouldn't be an issue?
>>      and
>>   Does GV not like us FS users?
>>
>> Thank you for you help.
>>
>> "Resistance is Fruitful"
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From garmt.noname at gmail.com  Thu Feb 10 00:44:55 2011
From: garmt.noname at gmail.com (garmt.noname at gmail.com)
Date: Wed, 9 Feb 2011 22:44:55 +0100
Subject: [Freeswitch-users] Best way to implement real-time ivr from
	mysql
In-Reply-To: <AANLkTikxJ1uNR+2sM-V_z+0YAR_nbX8pyNc2=AK7E9YC@mail.gmail.com>
References: <4D52C739.8080502@communicatefreely.net>
	<BE42DE63-CA66-40C8-A0BF-17AE303C77B1@ipeva.fr>
	<AANLkTikxJ1uNR+2sM-V_z+0YAR_nbX8pyNc2=AK7E9YC@mail.gmail.com>
Message-ID: <AANLkTikZ8uaLbGUomEFuiSo6xy0PXPm0ABt8VdvDZv2T@mail.gmail.com>

if you use lua, use Freeswitch.dbh, you may want to have a look at
http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh
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From steveayre at gmail.com  Thu Feb 10 11:53:33 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 10 Feb 2011 08:53:33 +0000
Subject: [Freeswitch-users] Yet another DTMF question
In-Reply-To: <AANLkTi=d_Bzmt+5s+RaGDH6KckBszmzJjsTq2G2JvRZP@mail.gmail.com>
References: <AANLkTik030ecFw2bvKe74pOs+NzfqHknX8keDO3mQ=cH@mail.gmail.com>
	<AANLkTikqQodjsMT_VB1xeQdtY-2DEgL68ndEnHTBTvkb@mail.gmail.com>
	<AANLkTiksK3T24TUDO-7aq_tbkFCkj_4BRNpLRt4=GtBj@mail.gmail.com>
	<62C5671A-8657-4F99-8184-272EEBEC8E77@gmail.com>
	<AANLkTimFMwJ1EkP_+1uCyiKUCKocgh-mSYVEd39navS7@mail.gmail.com>
	<AANLkTikaerx0b+A7EhFLXxuBKBBY+RBog8Lpgyd1ZZmR@mail.gmail.com>
	<AANLkTinL1NxOjMJjShOwXEo5CJERLbY4N7Z7re=QRp8z@mail.gmail.com>
	<AANLkTi=Trve2-zDb60FEahz2TLD_=BAvhzisf7gSOoFo@mail.gmail.com>
	<AANLkTi=d_Bzmt+5s+RaGDH6KckBszmzJjsTq2G2JvRZP@mail.gmail.com>
Message-ID: <AANLkTi=5TEUNnycLKyY_5hh7eZQTAGB34xC6axYjnddO@mail.gmail.com>

Introduced on 12th Jan (commit fe1711fd)
Fixed on 13th Jan (commit b2359797)

Affected commits:
  fe1711fd
  9c7b507d
  a6db66ef
  d9c56345
  8458adeb
  2e074727
  c6bdb303

-Steve


On 10 February 2011 06:23, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Wed, Jan 19, 2011 at 10:32 AM, Steven Ayre <steveayre at gmail.com> wrote:
> > Try uploading to the latest Git, a post from Brian just reminded me that
> > there was a day where some changes to the RTP stack broke RFC2833
> support,
> > perhaps you have one of the bad versions.
>
> Anyone know what day this might have been?  I WILL update, but it
> would be good to know if my issues could also be related.
>
> Thanks,
> Gabe
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Feb 10 12:07:35 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 10 Feb 2011 11:07:35 +0200
Subject: [Freeswitch-users] Language files - French, German, Spanish,
	etc.
In-Reply-To: <AANLkTim4Wc+hV_aCufsa1VkiBheCEWJGC16OhzRQ7fi8@mail.gmail.com>
References: <AANLkTim4Wc+hV_aCufsa1VkiBheCEWJGC16OhzRQ7fi8@mail.gmail.com>
Message-ID: <AANLkTintUvJKvicf7n8-P3fCywqocY8CNVeqVJPZJc+w@mail.gmail.com>

Hello Michael,

  We have Hebrew support: mod_say_he and professional recodrings. How can we
donate them to the public and add them to FreeSwitch source tree?

                                 Thanks! __Yehavi:

 2010/10/29 Michael Collins <msc at freeswitch.org>

> Hello all!
>
> I have heard about several of our intrepid community members creating sound
> sets in various languages. If you have created a sound set in any language
> other than US English please contact me off list. I want to help get this
> sound prompts organized so that the global community can use them.
>
> Thanks!
> -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 u2nsam at gmail.com  Thu Feb 10 12:11:36 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 10 Feb 2011 14:41:36 +0530
Subject: [Freeswitch-users] contact header
Message-ID: <AANLkTimeEwuuJBPnr+VrYs9kA7DN3pqQYDN1ytjW4=Rc@mail.gmail.com>

Hi,

Is there a way to change the contact header to Contact: <sip:abcd at xxxx:5060
from Contact: <sip:gw+foo at xxxx:5060>

Regards
Sam
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From u2nsam at gmail.com  Thu Feb 10 12:23:59 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 10 Feb 2011 14:53:59 +0530
Subject: [Freeswitch-users] Polycom and registering with domains
	(user@domain.tld@domain.tld)
In-Reply-To: <AANLkTikQs4oWyKoc3yTPPj6AMkobUXyYR6n_nPmhTmMP@mail.gmail.com>
References: <AANLkTikb2Mzw9oHagSHE=oe8ABSwOWOxwRE6CdNSO8WS@mail.gmail.com>
	<AANLkTikieRwaDVgNhPRd1BSV1gHFkYyzdiUoASVLhbij@mail.gmail.com>
	<AANLkTikQs4oWyKoc3yTPPj6AMkobUXyYR6n_nPmhTmMP@mail.gmail.com>
Message-ID: <AANLkTiki6vnicN967hiAAxuBsjVJ2h1WvQ_w9h2eFhcA@mail.gmail.com>

I too have natting issues with polycom when FS is on public IP.

Regards

On Thu, Feb 10, 2011 at 12:05 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Sat, Jan 29, 2011 at 2:35 PM, Aloysius Lloyd
> <lloyd.aloysius at gmail.com> wrote:
> > I had the similar issues with Polycom phones. Stay away from the web
> > configuration.
> > All you need to setup the FTP or TFTP server and use the configuration
> > files.
>
> The Polycom phones are harder to get set up, but when you get things
> configured correctly, it's pretty sweet.  Next up, my NATing issues
> with Polycom. :)
>
> Thanks for your input,
> Gabe
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From godson.g at gmail.com  Thu Feb 10 12:33:07 2011
From: godson.g at gmail.com (Godson Gera)
Date: Thu, 10 Feb 2011 15:03:07 +0530
Subject: [Freeswitch-users] C# problem for calling
	"freeswitch.switch_core_session_read_frame()"
In-Reply-To: <363776.37607.qm@web30506.mail.mud.yahoo.com>
References: <363776.37607.qm@web30506.mail.mud.yahoo.com>
Message-ID: <AANLkTinkr04nGE5J5hnbgH3VuchxtaVW3-8CKbesxAW5@mail.gmail.com>

You can use media bug API to read and analyze audio (I think RTP details are
abstracted away by media bug).


On Tue, Feb 8, 2011 at 9:25 AM, Frankie Yiu <yky1628 at yahoo.com> wrote:

>    Hi there,
>
> I would like to dial a phone number, and read the RTP package back when
> connected so that we can analyze the data; (to determine when we should
> play an audio at the right time--human or answer machine.)
>
> We found a code for IVR test (
> http://docs.freeswitch.org/switch__ivr_8c-source.html) Function name:
> switch_ivr_sound_test
>
> We would like to do the same but with C# code, but we encountered a problem
> when calling the function "freeswitch.switch_core_session_read_frame(  ) "
>
> < in swig.cs
>
> switch_core_session_read_frame(SWIGTYPE_p_switch_core_session session,
> SWIGTYPE_p_p_switch_frame frame, uint flags, int stream_id)   >
>
> where the second parameter--frame is a pointer to pointer of switch_frame
> type and in C# code, it is having a difficulty passing an object to the C++
> side and keep the pointer place holder before going deeper into the C code
> (switch_core_io.c)
>
> 1) So is there any way I can call this function in C#?
> 2) Is there another function or routine that you can suggest me to for
> reading RTP package?
>
> Thanks,
>
>
> Frank
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Thanks & Regards,
Godson Gera
FreeSWITCH Asterisk Consultant <http://godson.in/>
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From dmitry.bely at gmail.com  Thu Feb 10 12:36:05 2011
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Thu, 10 Feb 2011 12:36:05 +0300
Subject: [Freeswitch-users] How to dial a remote_skypename from a
 telephone keypad?
In-Reply-To: <1297308558454-6010434.post@n2.nabble.com>
References: <1297263654900-6007946.post@n2.nabble.com>
	<AANLkTikfXRGxqbrassDNhOnaOWkaje3DXQptRAVRzb9d@mail.gmail.com>
	<AANLkTikJNH3-vk=dMkOjAKsAxgr7KaiGD-U-9t2uKjxE@mail.gmail.com>
	<1297272133009-6008667.post@n2.nabble.com>
	<AANLkTikdMDk_NkKVyH6Qd4DBq5izujG77eYPOi1ZCbms@mail.gmail.com>
	<1297276772026-6009014.post@n2.nabble.com>
	<AANLkTikDg6yJVHh7M=q0cNkacwhXpJBV5kJSThTaJGTZ@mail.gmail.com>
	<AANLkTik0wPtXNOcd95wX2TH8B0+BaY7mz3aeBZaWa3hi@mail.gmail.com>
	<1297308558454-6010434.post@n2.nabble.com>
Message-ID: <AANLkTi=bDePMqM+Jf3RzQAHX3COd=hgO_A5iFWk_mA67@mail.gmail.com>

On Thu, Feb 10, 2011 at 6:29 AM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
>
> Dmitry Bely wrote:
>> # button can be used to start encoding sequence and ## is to end it.
>> So e.g. giovanni334 is be encoded as
>>
>> #4#444#666#888#2#66#66#444##334
>>
>> or even
>>
>> #4#444666888266#66444##334 (if digit changes we know that another
>> character is started. Thus # prefix can be safely omitted)
>>
>> Then all we would need is a very simple decoder on the
>> freeswitch/skypopen side. Just an idea...
> What will happen if a Skype account is named Skype333 or Skype444?

Nothing special. Skype names are case-insensitive so one would encode
skype333  string. The result would be:

#777755444733##333

- Dmitry Bely


From dujinfang at gmail.com  Thu Feb 10 12:38:13 2011
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 10 Feb 2011 17:38:13 +0800
Subject: [Freeswitch-users] git HEAD 503 Maximum Calls In Progress
In-Reply-To: <AANLkTi=n4bcpPXzyh1sRYnYRbRRnhDETEB3Dij7c0E1U@mail.gmail.com>
References: <AANLkTimpGj4GLug8jrK-7-V3Mja9W2tTiwtJsBSu-TZo@mail.gmail.com>
	<AANLkTi=n72vrWJDKB4G2yNkwm+8hdgnwAuCKBPP7xac2@mail.gmail.com>
	<AANLkTimnQ+68QR+v-1-rUqyD-YkXA8RKh+Na=8Fvmv6v@mail.gmail.com>
	<AANLkTi=n4bcpPXzyh1sRYnYRbRRnhDETEB3Dij7c0E1U@mail.gmail.com>
Message-ID: <AANLkTimsuYubrJvOP8_oqhsBBsCBSY7a6joye7A27+c-@mail.gmail.com>

Thanks, I got clflush size: 64, doesn't that means it support 64?

But It's in a remote datacenter and I'm not bother to reinstall OS.

cat /proc/cpuinfo
processor       : 0
vendor_id       : GenuineIntel
cpu family      : 15
model           : 2
model name      : Intel(R) Pentium(R) 4 CPU 2.00GHz
stepping        : 7
cpu MHz         : 1996.644
cache size      : 512 KB
fdiv_bug        : no
hlt_bug         : no
f00f_bug        : no
coma_bug        : no
fpu             : yes
fpu_exception   : yes
cpuid level     : 2
wp              : yes
flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe up
pebs bts cid
bogomips        : 3993.28
clflush size    : 64

On Thu, Feb 10, 2011 at 4:46 PM, Steven Ayre <steveayre at gmail.com> wrote:
>> I know 64bit is best but I was trying to help a friend to setup a
>> simple IVR with lua under 10 channels, I think it's good to use
>> existing hardwares. Actually the server has a pentium4 CPU with 1G
>> mem, it must be 10+ year old.
>
> Fair enough. Some models since Prescott do actually support it by the way,
> although most don't
>
> -Steve
>
>
>
>
> On 10 February 2011 01:32, Seven Du <dujinfang at gmail.com> wrote:
>>
>> Thank you Anthony, my bad. The server was running mysql and after I
>> installed postgresql cpu load > 1.0 and 0% idle. I did run
>> /etc/init.d/mysql stop but mysql-safe still left running which I
>> didn't noticed, after I killed mysql all problems gone.
>>
>> I will left the min-idle-cpu untouched until I run into problems. Thanks
>> again.
>>
>> On Wed, Feb 9, 2011 at 11:57 PM, Steven Ayre <steveayre at gmail.com> wrote:
>> > Upgrade to 64bit if you can
>> >
>>
>> I know 64bit is best but I was trying to help a friend to setup a
>> simple IVR with lua under 10 channels, I think it's good to use
>> existing hardwares. Actually the server has a pentium4 CPU with 1G
>> mem, it must be 10+ year old.
>>
>> Thanks Steven.
>>
>> >
>> > On 9 February 2011 11:37, Seven Du <dujinfang at gmail.com> wrote:
>> >>
>> >> Hi,
>> >>
>> >> It's a fresh install with wget ?freeswitch.org/eg/Makefile && make,
>> >> and I'm sure I successfully called 9196 and everything was ok.
>> >>
>> >> But a few minutes later I started getting
>> >>
>> >> ?503 Maximum Calls In Progress
>> >>
>> >> also originate user/1003 or loopback/9196 shows errors
>> >>
>> >> see
>> >> http://pastebin.freeswitch.org/15333
>> >>
>> >> It's on a ubuntu 8.04 32bit, the only thing I did between non-work and
>> >> work was I configured odbc with postgresql to work with isql. I
>> >> haven't touch the FS config yet. I don't think they are related.
>> >>
>> >>
>> >> here is also a nua debug
>> >>
>> >> http://pastebin.freeswitch.org/15335
>> >>
>> >> Any hint on this? Thanks.
>> >>
>> >> btw, I also noticed there might be some network problems between me
>> >> and the server where sometime sip packets not shown ( sofia debug and
>> >> ngrep) on server. Anyway, since loopback cannot create sessions so
>> >> something must be wrong in FS side, but what's that?
>> >>
>> >> Thanks.
>> >>
>> >> --
>> >> 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
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at 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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 yky1628 at yahoo.com  Thu Feb 10 12:34:31 2011
From: yky1628 at yahoo.com (Frankie Yiu)
Date: Thu, 10 Feb 2011 01:34:31 -0800 (PST)
Subject: [Freeswitch-users] How to read RTP package?
Message-ID: <263862.57046.qm@web30508.mail.mud.yahoo.com>

Here is the description of the project:

We want to build a system that would play a recorded message?when a human or an 
answer machine picks up the phone(eventually for multiple of phone numbers).? 
For a human, we?want to?play the message after the human speaks.? For answer 
machine, we want to wait for the beep.? To do this, we think that we have to?get 
the RTP packets and determine when is the right timing to play the message.

Currently, we have code in C# to play a recorded message when someone picks up 
the phone.??Someone suggests me to?try?Media bugs but it?is?not working with the 
wrapper function for C# (swig code, because of the?pointer to pointer type, 
unless I am wrong).? So as?our test, we call a wrapper function?from C# with a 
"session" information, and inside the C++ code, we?initialize a media bug with 
a?call back function for READ.??I got?event for?Media Bugs INIT and when Media 
Bugs?CLOSE message but never got any READ event.

Here are my questions:
1) How can I get the READ event from Media bugs event?? It seems to me that when 
I call a routine to initialize(ADD) the Media bugs, I will get INIT and CLOSE 
event back right away event though we have not hung up the phone yet, and 
therefore we never got any READ events.

2) If I am calling a number from C# code, how can I pass the session information 
to C++ code? (as I mention earlier I have modified function inside 
freeswitch_wrap.cxx to have call back in C++ code, but this is not the proper 
way)

3) Is there a way to?have packet send back to C# (I think those type 
"SWIGTYPE_p_p_..." is not working?properly for C#, am I correct?)
?
Thanks in advance.? Any help would be great!!
Frankie 


      
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From david.ponzone at ipeva.fr  Thu Feb 10 12:50:42 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 10 Feb 2011 10:50:42 +0100
Subject: [Freeswitch-users] contact header
In-Reply-To: <AANLkTimeEwuuJBPnr+VrYs9kA7DN3pqQYDN1ytjW4=Rc@mail.gmail.com>
References: <AANLkTimeEwuuJBPnr+VrYs9kA7DN3pqQYDN1ytjW4=Rc@mail.gmail.com>
Message-ID: <DBFEF45A-B945-43A0-99B3-E5A2ACCEB96F@ipeva.fr>

You can use:
<param name="extension-in-contact" value="true"/>

Depends what is abcd for you.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 10/02/2011 ? 10:11, Sam a ?crit :

> Hi,
> 
> Is there a way to change the contact header to Contact: <sip:abcd at xxxx:5060 from Contact: <sip:gw+foo at xxxx:5060>
> 
> 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 philippe.sultan at gmail.com  Thu Feb 10 13:39:03 2011
From: philippe.sultan at gmail.com (Philippe Sultan)
Date: Thu, 10 Feb 2011 11:39:03 +0100
Subject: [Freeswitch-users] Aastra phone registration lost
In-Reply-To: <4D534AFD.3020401@communicatefreely.net>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>
	<4D52B987.7050002@communicatefreely.net>
	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>
	<4D534AFD.3020401@communicatefreely.net>
Message-ID: <AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>

Tim,

That helped a lot, thanks! More comments inline.

On Thu, Feb 10, 2011 at 3:18 AM, Tim St. Pierre
<fs-list at communicatefreely.net> wrote:
> I'm wondering about these values:
>
>> sip registration retry timer: 10
>> sip registration renewal timer: 15
>> sip registration timeout retry timer: 10
>> sip registration period: 60
>> sip rport: 1
>>
> They seem a bit short, especially on a LAN. ?Do you really need your
> phones to re-register every 45 seconds?

Indeed, I don't. But need a mechanism to keep NAT sessions alive in
case phones are being deployed behind NAT boxes. It seems like
transporting SIP over TCP from the phone, plus using TCP keepalives
will solve that issue though. I also have Thomson ST2030 phones, and
they send SIP OPTIONS packets every 50 seconds for that purpose (all
handled by the SER proxy).

> Try setting sip registration period to 600 ?That's 10 minutes.
> sip registration renewal timer can be a bit longer - 30 seconds maybe,
> or 60 if you are at all worried about reliability.

Here is what I put in the aastra.cfg file from your suggestions :

sip registration period: 3600
sip registration renewal timer: 3565

And that does the trick, that is, the phone registers every 45
seconds. I'll try to update those values to have a larger registration
period, rely on TCP keepalives in order to maintain NAT sessions, and
have all that NAT handling in SER. Here is the topology :

Aastra ---(sip/tcp)--- SER ---(sip/udp)--- FS

ST2030 ---(sip/udp)--- SER ---(sip/udp)--- FS

Thanks again!

Philippe


From david.ponzone at ipeva.fr  Thu Feb 10 14:23:34 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 10 Feb 2011 12:23:34 +0100
Subject: [Freeswitch-users] Aastra phone registration lost
In-Reply-To: <AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>
	<4D52B987.7050002@communicatefreely.net>
	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>
	<4D534AFD.3020401@communicatefreely.net>
	<AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>
Message-ID: <60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>

Philippe,

can you confirm there is no NAT Keepalive feature in the Aastra ?
That's quite weird.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 10/02/2011 ? 11:39, Philippe Sultan a ?crit :

> Tim,
> 
> That helped a lot, thanks! More comments inline.
> 
> On Thu, Feb 10, 2011 at 3:18 AM, Tim St. Pierre
> <fs-list at communicatefreely.net> wrote:
>> I'm wondering about these values:
>> 
>>> sip registration retry timer: 10
>>> sip registration renewal timer: 15
>>> sip registration timeout retry timer: 10
>>> sip registration period: 60
>>> sip rport: 1
>>> 
>> They seem a bit short, especially on a LAN.  Do you really need your
>> phones to re-register every 45 seconds?
> 
> Indeed, I don't. But need a mechanism to keep NAT sessions alive in
> case phones are being deployed behind NAT boxes. It seems like
> transporting SIP over TCP from the phone, plus using TCP keepalives
> will solve that issue though. I also have Thomson ST2030 phones, and
> they send SIP OPTIONS packets every 50 seconds for that purpose (all
> handled by the SER proxy).
> 
>> Try setting sip registration period to 600  That's 10 minutes.
>> sip registration renewal timer can be a bit longer - 30 seconds maybe,
>> or 60 if you are at all worried about reliability.
> 
> Here is what I put in the aastra.cfg file from your suggestions :
> 
> sip registration period: 3600
> sip registration renewal timer: 3565
> 
> And that does the trick, that is, the phone registers every 45
> seconds. I'll try to update those values to have a larger registration
> period, rely on TCP keepalives in order to maintain NAT sessions, and
> have all that NAT handling in SER. Here is the topology :
> 
> Aastra ---(sip/tcp)--- SER ---(sip/udp)--- FS
> 
> ST2030 ---(sip/udp)--- SER ---(sip/udp)--- FS
> 
> Thanks again!
> 
> Philippe
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 10 14:54:38 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 10 Feb 2011 03:54:38 -0800 (PST)
Subject: [Freeswitch-users] How to dial a remote_skypename from a
	telephone keypad?
In-Reply-To: <AANLkTi=bDePMqM+Jf3RzQAHX3COd=hgO_A5iFWk_mA67@mail.gmail.com>
References: <1297263654900-6007946.post@n2.nabble.com>
	<AANLkTikfXRGxqbrassDNhOnaOWkaje3DXQptRAVRzb9d@mail.gmail.com>
	<AANLkTikJNH3-vk=dMkOjAKsAxgr7KaiGD-U-9t2uKjxE@mail.gmail.com>
	<1297272133009-6008667.post@n2.nabble.com>
	<AANLkTikdMDk_NkKVyH6Qd4DBq5izujG77eYPOi1ZCbms@mail.gmail.com>
	<1297276772026-6009014.post@n2.nabble.com>
	<AANLkTikDg6yJVHh7M=q0cNkacwhXpJBV5kJSThTaJGTZ@mail.gmail.com>
	<AANLkTik0wPtXNOcd95wX2TH8B0+BaY7mz3aeBZaWa3hi@mail.gmail.com>
	<1297308558454-6010434.post@n2.nabble.com>
	<AANLkTi=bDePMqM+Jf3RzQAHX3COd=hgO_A5iFWk_mA67@mail.gmail.com>
Message-ID: <1297338878062-6011352.post@n2.nabble.com>



Dmitry Bely wrote:
> 
> On Thu, Feb 10, 2011 at 6:29 AM, mazilo <Nabble at slickdeals.endjunk.com>
> wrote:
>>
>>
>> Dmitry Bely wrote:
>>> # button can be used to start encoding sequence and ## is to end it.
>>> So e.g. giovanni334 is be encoded as
>>>
>>> #4#444#666#888#2#66#66#444##334
>>>
>>> or even
>>>
>>> #4#444666888266#66444##334 (if digit changes we know that another
>>> character is started. Thus # prefix can be safely omitted)
>>>
>>> Then all we would need is a very simple decoder on the
>>> freeswitch/skypopen side. Just an idea...
>> What will happen if a Skype account is named Skype333 or Skype444?
> 
> Nothing special. Skype names are case-insensitive so one would encode
> skype333  string. The result would be:
> 
> #777755444733##333
How about giovanni444 or giovanni6666. How will the encoder differentiate
between the letter 'o' with numeral '444' and the letter 'n' with numeral
'66'? Will they be encoded into giovannio or giovanninn, respectively?


-----
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/How-to-dial-a-remote-skypename-from-a-telephone-keypad-tp6007946p6011352.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From lloyd.aloysius at gmail.com  Thu Feb 10 17:25:04 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 10 Feb 2011 09:25:04 -0500
Subject: [Freeswitch-users] Aastra phone registration lost
In-Reply-To: <60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>
	<4D52B987.7050002@communicatefreely.net>
	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>
	<4D534AFD.3020401@communicatefreely.net>
	<AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>
	<60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>
Message-ID: <AANLkTimux4MBM1K_7itE4GnZQHjy_vR2Y6sr_z+u3gkR@mail.gmail.com>

Tim,

Can you share your success stories FreeSWITCH and Aastra.

Aastra Phones Behind the NAT?

In my case Aastra phones registration not a problem.

But calls drooped every 60 sec ... in the same environment Linksys and
Polycom works perfectly.

How stable the Aastra phones with FreeSWITCH system.

TIA

Lloyd





On Thu, Feb 10, 2011 at 6:23 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Philippe,
>
> can you confirm there is no NAT Keepalive feature in the Aastra ?
> That's quite weird.
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 10/02/2011 ? 11:39, Philippe Sultan a ?crit :
>
> Tim,
>
> That helped a lot, thanks! More comments inline.
>
> On Thu, Feb 10, 2011 at 3:18 AM, Tim St. Pierre
> <fs-list at communicatefreely.net> wrote:
>
> I'm wondering about these values:
>
>
> sip registration retry timer: 10
>
> sip registration renewal timer: 15
>
> sip registration timeout retry timer: 10
>
> sip registration period: 60
>
> sip rport: 1
>
>
> They seem a bit short, especially on a LAN.  Do you really need your
>
> phones to re-register every 45 seconds?
>
>
> Indeed, I don't. But need a mechanism to keep NAT sessions alive in
> case phones are being deployed behind NAT boxes. It seems like
> transporting SIP over TCP from the phone, plus using TCP keepalives
> will solve that issue though. I also have Thomson ST2030 phones, and
> they send SIP OPTIONS packets every 50 seconds for that purpose (all
> handled by the SER proxy).
>
> Try setting sip registration period to 600  That's 10 minutes.
>
> sip registration renewal timer can be a bit longer - 30 seconds maybe,
>
> or 60 if you are at all worried about reliability.
>
>
> Here is what I put in the aastra.cfg file from your suggestions :
>
> sip registration period: 3600
> sip registration renewal timer: 3565
>
> And that does the trick, that is, the phone registers every 45
> seconds. I'll try to update those values to have a larger registration
> period, rely on TCP keepalives in order to maintain NAT sessions, and
> have all that NAT handling in SER. Here is the topology :
>
> Aastra ---(sip/tcp)--- SER ---(sip/udp)--- FS
>
> ST2030 ---(sip/udp)--- SER ---(sip/udp)--- FS
>
> Thanks again!
>
> Philippe
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Feb 10 16:25:52 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Thu, 10 Feb 2011 18:55:52 +0530
Subject: [Freeswitch-users] IBM Server restarts If I start sangoma wanrouter
Message-ID: <AANLkTim_e3UrCj_cdzvozA9OCo+WeyVz=KAZZFMRdzYm@mail.gmail.com>

Hi ,

I am using IBM System X3400 M3 Server with Redhat Enterprise Edition 5.6 and
.
A101E Sangoma PRI Card.



When I start wanrouter , My server getting restart.
Could onyone faced same kind of issue?
Please find attached log files.

I use, wanrouter version 3.5.18

Anyone plz help me to resolve this issue?


-- 

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 dmitry.bely at gmail.com  Thu Feb 10 17:57:00 2011
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Thu, 10 Feb 2011 17:57:00 +0300
Subject: [Freeswitch-users] How to dial a remote_skypename from a
 telephone keypad?
In-Reply-To: <1297338878062-6011352.post@n2.nabble.com>
References: <1297263654900-6007946.post@n2.nabble.com>
	<AANLkTikfXRGxqbrassDNhOnaOWkaje3DXQptRAVRzb9d@mail.gmail.com>
	<AANLkTikJNH3-vk=dMkOjAKsAxgr7KaiGD-U-9t2uKjxE@mail.gmail.com>
	<1297272133009-6008667.post@n2.nabble.com>
	<AANLkTikdMDk_NkKVyH6Qd4DBq5izujG77eYPOi1ZCbms@mail.gmail.com>
	<1297276772026-6009014.post@n2.nabble.com>
	<AANLkTikDg6yJVHh7M=q0cNkacwhXpJBV5kJSThTaJGTZ@mail.gmail.com>
	<AANLkTik0wPtXNOcd95wX2TH8B0+BaY7mz3aeBZaWa3hi@mail.gmail.com>
	<1297308558454-6010434.post@n2.nabble.com>
	<AANLkTi=bDePMqM+Jf3RzQAHX3COd=hgO_A5iFWk_mA67@mail.gmail.com>
	<1297338878062-6011352.post@n2.nabble.com>
Message-ID: <AANLkTi=VFhb9AfHfS0hurOzjOgLHf1tvA6sMztkCmHfL@mail.gmail.com>

On Thu, Feb 10, 2011 at 2:54 PM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
> Dmitry Bely wrote:
>>
>> On Thu, Feb 10, 2011 at 6:29 AM, mazilo <Nabble at slickdeals.endjunk.com>
>> wrote:
>>>
>>>
>>> Dmitry Bely wrote:
>>>> # button can be used to start encoding sequence and ## is to end it.
>>>> So e.g. giovanni334 is be encoded as
>>>>
>>>> #4#444#666#888#2#66#66#444##334
>>>>
>>>> or even
>>>>
>>>> #4#444666888266#66444##334 (if digit changes we know that another
>>>> character is started. Thus # prefix can be safely omitted)
>>>>
>>>> Then all we would need is a very simple decoder on the
>>>> freeswitch/skypopen side. Just an idea...
>>> What will happen if a Skype account is named Skype333 or Skype444?
>>
>> Nothing special. Skype names are case-insensitive so one would encode
>> skype333 ?string. The result would be:
>>
>> #777755444733##333
> How about giovanni444 or giovanni6666.

Just the same way:

#777755444733##444
#777755444733##6666

> How will the encoder differentiate
> between the letter 'o' with numeral '444' and the letter 'n' with numeral
> '66'? Will they be encoded into giovannio or giovanninn, respectively?

Looks like you didn't fully read my proposal. Or my English is not
good enough. One again: an encoded sequence is started by # and ended
by ##. Inside it a group of identical digits is interpreted as a
letter (optionally prefixed by # as delimiter). Dial string may
contain several encoded parts interlaced with ordinary digits:

digits#letters##digits#letters##digits

So there is no problem to differ "n" from 66 - they appear in different contexts

- Dmitry Bely


From erik.dekkers at wvds.nl  Thu Feb 10 18:02:31 2011
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Thu, 10 Feb 2011 16:02:31 +0100
Subject: [Freeswitch-users] Aastra phone registration lost
In-Reply-To: <AANLkTimux4MBM1K_7itE4GnZQHjy_vR2Y6sr_z+u3gkR@mail.gmail.com>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>
	<4D52B987.7050002@communicatefreely.net>
	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>
	<4D534AFD.3020401@communicatefreely.net>
	<AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>
	<60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>
	<AANLkTimux4MBM1K_7itE4GnZQHjy_vR2Y6sr_z+u3gkR@mail.gmail.com>
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE0787301896@FS1.wvds.local>

Im using freeswitch with dozen aastra phones (6757i and 6739i) for about a year. They work perfectly, all the time.
If you're behind NAT, it would also be possible that the NAT device is the source of your calls dropping.

Erik

Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Aloysius Lloyd
Verzonden: donderdag 10 februari 2011 15:25
Aan: FreeSWITCH Users Help
Onderwerp: Re: [Freeswitch-users] Aastra phone registration lost

Tim,

Can you share your success stories FreeSWITCH and Aastra.

Aastra Phones Behind the NAT?

In my case Aastra phones registration not a problem.

But calls drooped every 60 sec ... in the same environment Linksys and Polycom works perfectly.

How stable the Aastra phones with FreeSWITCH system.

TIA

Lloyd




On Thu, Feb 10, 2011 at 6:23 AM, David Ponzone <david.ponzone at ipeva.fr<mailto:david.ponzone at ipeva.fr>> wrote:
Philippe,

can you confirm there is no NAT Keepalive feature in the Aastra ?
That's quite weird.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr<mailto:david.ponzone at ipeva.fr>
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<http://www.ipeva.fr>  -   www.ipeva-studio.com<http://www.ipeva-studio.com>

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.



Le 10/02/2011 ? 11:39, Philippe Sultan a ?crit :


Tim,

That helped a lot, thanks! More comments inline.

On Thu, Feb 10, 2011 at 3:18 AM, Tim St. Pierre
<fs-list at communicatefreely.net<mailto:fs-list at communicatefreely.net>> wrote:

I'm wondering about these values:

sip registration retry timer: 10
sip registration renewal timer: 15
sip registration timeout retry timer: 10
sip registration period: 60
sip rport: 1

They seem a bit short, especially on a LAN.  Do you really need your
phones to re-register every 45 seconds?

Indeed, I don't. But need a mechanism to keep NAT sessions alive in
case phones are being deployed behind NAT boxes. It seems like
transporting SIP over TCP from the phone, plus using TCP keepalives
will solve that issue though. I also have Thomson ST2030 phones, and
they send SIP OPTIONS packets every 50 seconds for that purpose (all
handled by the SER proxy).


Try setting sip registration period to 600  That's 10 minutes.
sip registration renewal timer can be a bit longer - 30 seconds maybe,
or 60 if you are at all worried about reliability.

Here is what I put in the aastra.cfg file from your suggestions :

sip registration period: 3600
sip registration renewal timer: 3565

And that does the trick, that is, the phone registers every 45
seconds. I'll try to update those values to have a larger registration
period, rely on TCP keepalives in order to maintain NAT sessions, and
have all that NAT handling in SER. Here is the topology :

Aastra ---(sip/tcp)--- SER ---(sip/udp)--- FS

ST2030 ---(sip/udp)--- SER ---(sip/udp)--- FS

Thanks again!

Philippe

_______________________________________________
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 Nabble at slickdeals.endjunk.com  Thu Feb 10 18:35:42 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 10 Feb 2011 07:35:42 -0800 (PST)
Subject: [Freeswitch-users] How to dial a remote_skypename from a
	telephone keypad?
In-Reply-To: <AANLkTi=VFhb9AfHfS0hurOzjOgLHf1tvA6sMztkCmHfL@mail.gmail.com>
References: <AANLkTikJNH3-vk=dMkOjAKsAxgr7KaiGD-U-9t2uKjxE@mail.gmail.com>
	<1297272133009-6008667.post@n2.nabble.com>
	<AANLkTikdMDk_NkKVyH6Qd4DBq5izujG77eYPOi1ZCbms@mail.gmail.com>
	<1297276772026-6009014.post@n2.nabble.com>
	<AANLkTikDg6yJVHh7M=q0cNkacwhXpJBV5kJSThTaJGTZ@mail.gmail.com>
	<AANLkTik0wPtXNOcd95wX2TH8B0+BaY7mz3aeBZaWa3hi@mail.gmail.com>
	<1297308558454-6010434.post@n2.nabble.com>
	<AANLkTi=bDePMqM+Jf3RzQAHX3COd=hgO_A5iFWk_mA67@mail.gmail.com>
	<1297338878062-6011352.post@n2.nabble.com>
	<AANLkTi=VFhb9AfHfS0hurOzjOgLHf1tvA6sMztkCmHfL@mail.gmail.com>
Message-ID: <1297352142520-6012135.post@n2.nabble.com>



Dmitry Bely wrote:
> So there is no problem to differ "n" from 66 - they appear in different
> contexts
So, if a skype name is n66n66n66, can you please show the equivalent numeral
key representation a user has to dial?

-----
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/How-to-dial-a-remote-skypename-from-a-telephone-keypad-tp6007946p6012135.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From admin at blindi.net  Thu Feb 10 18:38:19 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Thu, 10 Feb 2011 16:38:19 +0100 (CET)
Subject: [Freeswitch-users] lua question db
In-Reply-To: <AANLkTimux4MBM1K_7itE4GnZQHjy_vR2Y6sr_z+u3gkR@mail.gmail.com>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>
	<4D52B987.7050002@communicatefreely.net>
	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>
	<4D534AFD.3020401@communicatefreely.net>
	<AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>
	<60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>
	<AANLkTimux4MBM1K_7itE4GnZQHjy_vR2Y6sr_z+u3gkR@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1102101600450.18304@suse.blindi.net>

Hi all, i like to write a luascript for callforwaring.
The number store in the internal database "db".
I find a dbh example. i understand: dbh is mysql or postres-databases.
how can i get or  insert entrys in  the database "db"?
tahk you


---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From dmitry.bely at gmail.com  Thu Feb 10 19:19:20 2011
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Thu, 10 Feb 2011 19:19:20 +0300
Subject: [Freeswitch-users] How to dial a remote_skypename from a
 telephone keypad?
In-Reply-To: <1297352142520-6012135.post@n2.nabble.com>
References: <AANLkTikJNH3-vk=dMkOjAKsAxgr7KaiGD-U-9t2uKjxE@mail.gmail.com>
	<1297272133009-6008667.post@n2.nabble.com>
	<AANLkTikdMDk_NkKVyH6Qd4DBq5izujG77eYPOi1ZCbms@mail.gmail.com>
	<1297276772026-6009014.post@n2.nabble.com>
	<AANLkTikDg6yJVHh7M=q0cNkacwhXpJBV5kJSThTaJGTZ@mail.gmail.com>
	<AANLkTik0wPtXNOcd95wX2TH8B0+BaY7mz3aeBZaWa3hi@mail.gmail.com>
	<1297308558454-6010434.post@n2.nabble.com>
	<AANLkTi=bDePMqM+Jf3RzQAHX3COd=hgO_A5iFWk_mA67@mail.gmail.com>
	<1297338878062-6011352.post@n2.nabble.com>
	<AANLkTi=VFhb9AfHfS0hurOzjOgLHf1tvA6sMztkCmHfL@mail.gmail.com>
	<1297352142520-6012135.post@n2.nabble.com>
Message-ID: <AANLkTindVXHbCnA6mNQaq4mFX7K5KSZruxFHAPGhSb6a@mail.gmail.com>

On Thu, Feb 10, 2011 at 6:35 PM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
>
> Dmitry Bely wrote:
>> So there is no problem to differ "n" from 66 - they appear in different
>> contexts
> So, if a skype name is n66n66n66, can you please show the equivalent numeral
> key representation a user has to dial?

#66## 66 #66## 66 #66## 66

I have added spaces for better readability.

- Dmitry Bely


From msc at freeswitch.org  Thu Feb 10 19:24:17 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 10 Feb 2011 10:24:17 -0600
Subject: [Freeswitch-users] destination number variable
In-Reply-To: <283881297203422@web100.yandex.ru>
References: <283881297203422@web100.yandex.ru>
Message-ID: <AANLkTinNzpwDiiZHX8F_vCm2yf_mPMiQ+Y4DrC0w75wp@mail.gmail.com>

Are you calling yourself in this example?

variable_sip_req_user: [79645835822]
variable_sip_to_user: [79645835822]

I'm just curious. In any case, in this example it looks like the sip_to_user
contains the number you are looking for. Another option would be to capture
the phone number part in the regex and add a custom chan var like this:
<action application="set" data="dialed_number=$1"/>

Then you could access "dialed_number" as the name of the chan var.

-MC

On Tue, Feb 8, 2011 at 4:17 PM, Serge Yuriev <me at nevian.org> wrote:

> Hello
>
> In INFO app after call i see (shortened for readability)
> 2011-02-09 00:20:00.824547 [INFO] mod_dptools.c:1202 CHANNEL_DATA:
> Caller-Direction: [inbound]
> Caller-Username: [nevian]
> Caller-Dialplan: [XML]
> Caller-Caller-ID-Name: [Serge S. Yuriev]
> Caller-Caller-ID-Number: [nevian]
> Caller-Callee-ID-Name: [Outbound Call]
> Caller-Callee-ID-Number: [79645835822 at 81.16.114.33]
> Caller-Network-Addr: [109.173.67.229]
> Caller-ANI: [nevian]
> Caller-Destination-Number: [79645835822]
> variable_sip_from_display: [Serge S. Yuriev]
> variable_sip_full_from: ["Serge S. Yuriev" <sip:nevian at cranz.nevian.org
> >;tag=624579331]
> variable_sip_full_to: [<sip:79645835822 at cranz.nevian.org>]
> variable_sip_req_user: [79645835822]
> variable_sip_req_uri: [79645835822 at cranz.nevian.org]
> variable_sip_req_host: [cranz.nevian.org]
> variable_sip_to_user: [79645835822]
> variable_sip_to_uri: [79645835822 at cranz.nevian.org]
>
> In mod_radius_cdr profile->destination_number returns it as
> 79645835822 at 81.16.114.33
> Which field should I use to get only number?
>
> Termination via h323 if this matters
>
> --
> 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 michal.bielicki at seventhsignal.de  Thu Feb 10 19:32:13 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Thu, 10 Feb 2011 17:32:13 +0100
Subject: [Freeswitch-users] lua question db
In-Reply-To: <Pine.LNX.4.64.1102101600450.18304@suse.blindi.net>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>
	<4D52B987.7050002@communicatefreely.net>
	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>
	<4D534AFD.3020401@communicatefreely.net>
	<AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>
	<60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>
	<AANLkTimux4MBM1K_7itE4GnZQHjy_vR2Y6sr_z+u3gkR@mail.gmail.com>
	<Pine.LNX.4.64.1102101600450.18304@suse.blindi.net>
Message-ID: <F44DD6D8-F1DB-4794-94CB-0189BA8016A1@seventhsignal.de>

We do not use db
if you are not using odbc for the core than it uses sqlite.

Am 10.02.2011 um 16:38 schrieb Thomas Hoellriegel:

> Hi all, i like to write a luascript for callforwaring.
> The number store in the internal database "db".
> I find a dbh example. i understand: dbh is mysql or postres-databases.
> how can i get or  insert entrys in  the database "db"?
> tahk you
> 
> 
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/listinfo/blinde-misc
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 philippe.sultan at gmail.com  Thu Feb 10 19:33:08 2011
From: philippe.sultan at gmail.com (Philippe Sultan)
Date: Thu, 10 Feb 2011 17:33:08 +0100
Subject: [Freeswitch-users] Aastra phone registration lost
In-Reply-To: <60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>
	<4D52B987.7050002@communicatefreely.net>
	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>
	<4D534AFD.3020401@communicatefreely.net>
	<AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>
	<60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>
Message-ID: <AANLkTinZdo=TwX-acZjm7_J3FKDqw88bnEDJV=OWU5eD@mail.gmail.com>

Well, I was looking for something like what's implemented in the
Thomson ST2030 series, that is, have the phone initiate SIP OPTIONS,
and process them with SER.

I was not able to find such a mechanism, but digging in the docs, it
appears that I can just use SIP over TCP and keep the connection
maintained using TCP keepalive. That wil be just fine I guess.

On the other hand, you can of course have FS send SIP OPTIONS packets
to the phones, but I'd like to keep the NAT handling on the front end
SIP proxy as much as possible.

Philippe

On Thu, Feb 10, 2011 at 12:23 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
> Philippe,
> can you confirm there is no NAT Keepalive feature in the Aastra ?
> That's quite weird.
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 10/02/2011 ? 11:39, Philippe Sultan a ?crit :
>
> Tim,
>
> That helped a lot, thanks! More comments inline.
>
> On Thu, Feb 10, 2011 at 3:18 AM, Tim St. Pierre
> <fs-list at communicatefreely.net> wrote:
>
> I'm wondering about these values:
>
> sip registration retry timer: 10
>
> sip registration renewal timer: 15
>
> sip registration timeout retry timer: 10
>
> sip registration period: 60
>
> sip rport: 1
>
> They seem a bit short, especially on a LAN. ?Do you really need your
>
> phones to re-register every 45 seconds?
>
> Indeed, I don't. But need a mechanism to keep NAT sessions alive in
> case phones are being deployed behind NAT boxes. It seems like
> transporting SIP over TCP from the phone, plus using TCP keepalives
> will solve that issue though. I also have Thomson ST2030 phones, and
> they send SIP OPTIONS packets every 50 seconds for that purpose (all
> handled by the SER proxy).
>
> Try setting sip registration period to 600 ?That's 10 minutes.
>
> sip registration renewal timer can be a bit longer - 30 seconds maybe,
>
> or 60 if you are at all worried about reliability.
>
> Here is what I put in the aastra.cfg file from your suggestions :
>
> sip registration period: 3600
> sip registration renewal timer: 3565
>
> And that does the trick, that is, the phone registers every 45
> seconds. I'll try to update those values to have a larger registration
> period, rely on TCP keepalives in order to maintain NAT sessions, and
> have all that NAT handling in SER. Here is the topology :
>
> Aastra ---(sip/tcp)--- SER ---(sip/udp)--- FS
>
> ST2030 ---(sip/udp)--- SER ---(sip/udp)--- FS
>
> Thanks again!
>
> Philippe
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Philippe Sultan


From david.ponzone at ipeva.fr  Thu Feb 10 19:36:37 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 10 Feb 2011 17:36:37 +0100
Subject: [Freeswitch-users] lua question db
In-Reply-To: <Pine.LNX.4.64.1102101600450.18304@suse.blindi.net>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>
	<4D52B987.7050002@communicatefreely.net>
	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>
	<4D534AFD.3020401@communicatefreely.net>
	<AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>
	<60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>
	<AANLkTimux4MBM1K_7itE4GnZQHjy_vR2Y6sr_z+u3gkR@mail.gmail.com>
	<Pine.LNX.4.64.1102101600450.18304@suse.blindi.net>
Message-ID: <2243A0E7-C355-4E88-AEC1-2A918747C6BD@ipeva.fr>

You can't use the internal db for your application.
You must use an external DB (Posgres, MySQL, or anyting accessible through ODBC).

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 10/02/2011 ? 16:38, Thomas Hoellriegel a ?crit :

> Hi all, i like to write a luascript for callforwaring.
> The number store in the internal database "db".
> I find a dbh example. i understand: dbh is mysql or postres-databases.
> how can i get or  insert entrys in  the database "db"?
> tahk you
> 
> 
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/listinfo/blinde-misc
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 10 19:36:46 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 10 Feb 2011 10:36:46 -0600
Subject: [Freeswitch-users] Video on X-lite .. FSV module
In-Reply-To: <loom.20110209T020200-551@post.gmane.org>
References: <loom.20110209T020200-551@post.gmane.org>
Message-ID: <AANLkTi=vREsYW86fMGQ6dRt_o+R_VQeh5v-e8K-AU=Jo@mail.gmail.com>

>
> I am trying to record or play video on X-lite through Freeswitch. When I do
> a
> video call it says faild to send your video.
>
> Like Seven mentioned you need to go into conf/vars.xml. Find the lines that
have "codec prefs" and add your codec - usually "H263" or "H264". Then
reloadxml and try again. I've personally done this with x-lite on a mac and
it worked as soon as I added the codec. For reference, here's my exact line:

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h
,G722,PCMU,H263"/>

That's an example - be sure to use the codecs that *you* need in your setup.

-MC
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From david.ponzone at ipeva.fr  Thu Feb 10 19:41:20 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 10 Feb 2011 17:41:20 +0100
Subject: [Freeswitch-users] Aastra phone registration lost
In-Reply-To: <AANLkTinZdo=TwX-acZjm7_J3FKDqw88bnEDJV=OWU5eD@mail.gmail.com>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>
	<4D52B987.7050002@communicatefreely.net>
	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>
	<4D534AFD.3020401@communicatefreely.net>
	<AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>
	<60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>
	<AANLkTinZdo=TwX-acZjm7_J3FKDqw88bnEDJV=OWU5eD@mail.gmail.com>
Message-ID: <54568FE4-7A63-4034-9A39-68988708860C@ipeva.fr>

I get you.
The good thing with TCP is that most routers have a TCP NAT KeepAlive timer around 1 hour.
The onlly issue I see with a 3600s registration timer is if the registrar looses the registration for any reason.
It seems some phones do not always try to re-register before the timeout if they loose their registration.
Have you tested the Aastra with this ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 10/02/2011 ? 17:33, Philippe Sultan a ?crit :

> Well, I was looking for something like what's implemented in the
> Thomson ST2030 series, that is, have the phone initiate SIP OPTIONS,
> and process them with SER.
> 
> I was not able to find such a mechanism, but digging in the docs, it
> appears that I can just use SIP over TCP and keep the connection
> maintained using TCP keepalive. That wil be just fine I guess.
> 
> On the other hand, you can of course have FS send SIP OPTIONS packets
> to the phones, but I'd like to keep the NAT handling on the front end
> SIP proxy as much as possible.
> 
> Philippe
> 
> On Thu, Feb 10, 2011 at 12:23 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
>> Philippe,
>> can you confirm there is no NAT Keepalive feature in the Aastra ?
>> That's quite weird.
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> 
>> 
>> 
>> Le 10/02/2011 ? 11:39, Philippe Sultan a ?crit :
>> 
>> Tim,
>> 
>> That helped a lot, thanks! More comments inline.
>> 
>> On Thu, Feb 10, 2011 at 3:18 AM, Tim St. Pierre
>> <fs-list at communicatefreely.net> wrote:
>> 
>> I'm wondering about these values:
>> 
>> sip registration retry timer: 10
>> 
>> sip registration renewal timer: 15
>> 
>> sip registration timeout retry timer: 10
>> 
>> sip registration period: 60
>> 
>> sip rport: 1
>> 
>> They seem a bit short, especially on a LAN.  Do you really need your
>> 
>> phones to re-register every 45 seconds?
>> 
>> Indeed, I don't. But need a mechanism to keep NAT sessions alive in
>> case phones are being deployed behind NAT boxes. It seems like
>> transporting SIP over TCP from the phone, plus using TCP keepalives
>> will solve that issue though. I also have Thomson ST2030 phones, and
>> they send SIP OPTIONS packets every 50 seconds for that purpose (all
>> handled by the SER proxy).
>> 
>> Try setting sip registration period to 600  That's 10 minutes.
>> 
>> sip registration renewal timer can be a bit longer - 30 seconds maybe,
>> 
>> or 60 if you are at all worried about reliability.
>> 
>> Here is what I put in the aastra.cfg file from your suggestions :
>> 
>> sip registration period: 3600
>> sip registration renewal timer: 3565
>> 
>> And that does the trick, that is, the phone registers every 45
>> seconds. I'll try to update those values to have a larger registration
>> period, rely on TCP keepalives in order to maintain NAT sessions, and
>> have all that NAT handling in SER. Here is the topology :
>> 
>> Aastra ---(sip/tcp)--- SER ---(sip/udp)--- FS
>> 
>> ST2030 ---(sip/udp)--- SER ---(sip/udp)--- FS
>> 
>> Thanks again!
>> 
>> Philippe
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> 
> 
> -- 
> Philippe Sultan
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 10 19:49:55 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 10 Feb 2011 10:49:55 -0600
Subject: [Freeswitch-users] Language files - French, German, Spanish,
	etc.
In-Reply-To: <AANLkTintUvJKvicf7n8-P3fCywqocY8CNVeqVJPZJc+w@mail.gmail.com>
References: <AANLkTim4Wc+hV_aCufsa1VkiBheCEWJGC16OhzRQ7fi8@mail.gmail.com>
	<AANLkTintUvJKvicf7n8-P3fCywqocY8CNVeqVJPZJc+w@mail.gmail.com>
Message-ID: <AANLkTimyi3BGKrKoH1TzuSS4dKWv4B=SqjuxVJYvCRiS@mail.gmail.com>

Yehavi,

This is wonderful! Thank you so much! Please email me off list and we'll
handle the particulars.

-MC

On Thu, Feb 10, 2011 at 3:07 AM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> Hello Michael,
>
>   We have Hebrew support: mod_say_he and professional recodrings. How can
> we donate them to the public and add them to FreeSwitch source tree?
>
>                                  Thanks! __Yehavi:
>
>  2010/10/29 Michael Collins <msc at freeswitch.org>
>
>> Hello all!
>>
>>
>> I have heard about several of our intrepid community members creating
>> sound sets in various languages. If you have created a sound set in any
>> language other than US English please contact me off list. I want to help
>> get this sound prompts organized so that the global community can use them.
>>
>> Thanks!
>> -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 wstephen80 at gmail.com  Thu Feb 10 19:53:12 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Thu, 10 Feb 2011 17:53:12 +0100
Subject: [Freeswitch-users] Problem with Outbound ESL in C++
Message-ID: <AANLkTimxR0oJECk8z8R7Yg0_rydBEODSrtK+-a_uY_Ac@mail.gmail.com>

I'm trying to write an ESL Outbound application in C++ but I have problem to
receive events.

My application is: http://pastebin.freeswitch.org/15349

The problem is related to the sending of "event plain ALL" command that
receives an error as reply.

The same problem if I run the Freeswitch "testserver" application of ESL
library: the "event plain" command receives always a reply "-ERR command not
found" and also the "linger" command.

What I'm missing?

Here the log of "testserver" application


[INFO] testserver.c:19 mycallback() Connected! 4
[DEBUG] src/esl.c:1141 esl_send() SEND
filter unique-id 9e159038-d599-4e18-afe9-a8f4eeffd76b


[DEBUG] src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type] =
[command/reply]
[DEBUG] src/esl.c:989 esl_recv_event() RECV HEADER [Reply-Text] = [+OK
filter added. [unique-id]=[9e159038-d599-4e18-afe9-a8f4eeffd76b]]
[DEBUG] src/esl.c:1141 esl_send() SEND
event plain SESSION_HEARTBEAT CHANNEL_ANSWER CHANNEL_ORIGINATE
CHANNEL_PROGRESS CHANNEL_HANGUP CHANNEL_BRIDGE CHANNEL_UNBRIDGE
CHANNEL_OUTGOING CHANNEL_EXECUTE CHANNEL_EXECUTE_COMPLETE DTMF CUSTOM
conference::maintenance


[DEBUG] src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type] =
[command/reply]
[DEBUG] src/esl.c:989 esl_recv_event() RECV HEADER [Reply-Text] = [-ERR
command not found]
[DEBUG] src/esl.c:1141 esl_send() SEND
linger
[DEBUG] src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type] =
[command/reply]
[DEBUG] src/esl.c:989 esl_recv_event() RECV HEADER [Reply-Text] = [-ERR
command not found]
[DEBUG] src/esl.c:1141 esl_send() SEND
sendmsg
call-command: execute
execute-app-name: answer


[DEBUG] src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type] =
[command/reply]
[DEBUG] src/esl.c:989 esl_recv_event() RECV HEADER [Reply-Text] = [+OK]
[DEBUG] src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type] =
[text/disconnect-notice]
[DEBUG] src/esl.c:989 esl_recv_event() RECV HEADER [Controlled-Session-UUID]
= [9e159038-d599-4e18-afe9-a8f4eeffd76b]
[DEBUG] src/esl.c:989 esl_recv_event() RECV HEADER [Content-Disposition] =
[disconnect]
[DEBUG] src/esl.c:989 esl_recv_event() RECV HEADER [Content-Length] = [67]
[INFO] testserver.c:49 mycallback() Disconnected! 4
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From infos at madovsky.org  Thu Feb 10 19:57:11 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 10 Feb 2011 11:57:11 -0500
Subject: [Freeswitch-users] switch.conf.xml port range
Message-ID: <A23A63F76A4642D6BDF80A8B0D3A6B6C@e1705>

in port range settings
is it mean that RTP sockets are created on localhost IP ?
if yes, how can I change the IP to 127.0.0.6 for example ?

thanks
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From massimiliano.ravelli at gmail.com  Thu Feb 10 19:42:34 2011
From: massimiliano.ravelli at gmail.com (Massimiliano Ravelli)
Date: Thu, 10 Feb 2011 17:42:34 +0100
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <4D52C951.5080600@utoronto.ca>
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>
Message-ID: <AANLkTi=nA76tJcdXN2Ta3cyzm5KnQhBadNJE_swg1Xk0@mail.gmail.com>

2011/2/9 Victor Chukalovskiy <victor.chukalovskiy at utoronto.ca>

> Hi Folks,
>
> Thank you everyone. Solved by lunching server separately.
>

I have the same problem:
freeswitch at internal> g729_info
Can't contact licence server.

Quite recent freeswitch version (3 days old - git
2b4f163826132cf55698008a22c065374320796a)
installed in /opt/freeswitch and started with
"/opt/freeswitch/bin/freeswitch -nc"
No problem nor warning using fsg729-191-installer.
The licence server starts automatically upon loading of mod_com_g729.

I registered succesfully some licences.

I even tried to restart freeswitch, I killed licence server and launched
manually...
The licence server starts but I still get "Can't contact licence server."

I notice only a weird thing trying to autocomplete g729_ on fs_cli:
[      g729_available]    [          g729_count]    [
g729_info]    [           g729_used]
[      g729_avaliable]

Any hints ?

Thanks very much in advance

Regards
Massimiliano
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From msc at freeswitch.org  Thu Feb 10 20:01:27 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 10 Feb 2011 11:01:27 -0600
Subject: [Freeswitch-users] lua question db
In-Reply-To: <2243A0E7-C355-4E88-AEC1-2A918747C6BD@ipeva.fr>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>
	<4D52B987.7050002@communicatefreely.net>
	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>
	<4D534AFD.3020401@communicatefreely.net>
	<AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>
	<60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>
	<AANLkTimux4MBM1K_7itE4GnZQHjy_vR2Y6sr_z+u3gkR@mail.gmail.com>
	<Pine.LNX.4.64.1102101600450.18304@suse.blindi.net>
	<2243A0E7-C355-4E88-AEC1-2A918747C6BD@ipeva.fr>
Message-ID: <AANLkTim+B7uN+FKiPJ5B=9tMYnoaGDkm3bCMastkWzhC@mail.gmail.com>

On Thu, Feb 10, 2011 at 10:36 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> You can't use the internal db for your application.
> You must use an external DB (Posgres, MySQL, or anyting accessible through
> ODBC).
>

Technically you *could* use the db (or hash) API to store the call-forward
information in much the same way that the Local_Extension and the
call_return & call-pickup extensions work. I'll leave it as an exercise for
the OP to review these extensions in conf/dialplan/default.xml for hints on
how to proceed:

Local_Extension
global-intercept
group-intercept
intercept-ext
redial
call_return

-MC
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From msc at freeswitch.org  Thu Feb 10 20:03:00 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 10 Feb 2011 11:03:00 -0600
Subject: [Freeswitch-users] switch.conf.xml port range
In-Reply-To: <A23A63F76A4642D6BDF80A8B0D3A6B6C@e1705>
References: <A23A63F76A4642D6BDF80A8B0D3A6B6C@e1705>
Message-ID: <AANLkTi=LnywqgoLkkiO3Yz9x4ThTbRE3jtuX0EAQyZtz@mail.gmail.com>

Do you have a SIP profile bound to an interface with IP address of
127.0.0.6? Does that actually do something useful?
-MC

On Thu, Feb 10, 2011 at 10:57 AM, Madovsky <infos at madovsky.org> wrote:

>  in port range settings
> is it mean that RTP sockets are created on localhost IP ?
> if yes, how can I change the IP to 127.0.0.6 for 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
>
>
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From infos at madovsky.org  Thu Feb 10 20:08:31 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 10 Feb 2011 12:08:31 -0500
Subject: [Freeswitch-users] switch.conf.xml port range
References: <A23A63F76A4642D6BDF80A8B0D3A6B6C@e1705>
	<AANLkTi=LnywqgoLkkiO3Yz9x4ThTbRE3jtuX0EAQyZtz@mail.gmail.com>
Message-ID: <9162BE27CDE9481F88FC4B730195A046@e1705>

yes I created an alias lo:5 espcially reserved for FS
I d only like to reorganize sockets range from other services (bind etc....)

  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Thursday, February 10, 2011 12:03 PM
  Subject: Re: [Freeswitch-users] switch.conf.xml port range


  Do you have a SIP profile bound to an interface with IP address of 127.0.0.6? Does that actually do something useful?
  -MC


  On Thu, Feb 10, 2011 at 10:57 AM, Madovsky <infos at madovsky.org> wrote:

    in port range settings
    is it mean that RTP sockets are created on localhost IP ?
    if yes, how can I change the IP to 127.0.0.6 for 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






------------------------------------------------------------------------------


  _______________________________________________
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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  Thu Feb 10 20:11:47 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 10 Feb 2011 09:11:47 -0800 (PST)
Subject: [Freeswitch-users] How to dial a remote_skypename from a
	telephone keypad?
In-Reply-To: <AANLkTindVXHbCnA6mNQaq4mFX7K5KSZruxFHAPGhSb6a@mail.gmail.com>
References: <AANLkTikdMDk_NkKVyH6Qd4DBq5izujG77eYPOi1ZCbms@mail.gmail.com>
	<1297276772026-6009014.post@n2.nabble.com>
	<AANLkTikDg6yJVHh7M=q0cNkacwhXpJBV5kJSThTaJGTZ@mail.gmail.com>
	<AANLkTik0wPtXNOcd95wX2TH8B0+BaY7mz3aeBZaWa3hi@mail.gmail.com>
	<1297308558454-6010434.post@n2.nabble.com>
	<AANLkTi=bDePMqM+Jf3RzQAHX3COd=hgO_A5iFWk_mA67@mail.gmail.com>
	<1297338878062-6011352.post@n2.nabble.com>
	<AANLkTi=VFhb9AfHfS0hurOzjOgLHf1tvA6sMztkCmHfL@mail.gmail.com>
	<1297352142520-6012135.post@n2.nabble.com>
	<AANLkTindVXHbCnA6mNQaq4mFX7K5KSZruxFHAPGhSb6a@mail.gmail.com>
Message-ID: <1297357907611-6012521.post@n2.nabble.com>



Dmitry Bely wrote:
> 
> On Thu, Feb 10, 2011 at 6:35 PM, mazilo <Nabble at slickdeals.endjunk.com>
> wrote:
>>
>>
>> Dmitry Bely wrote:
>>> So there is no problem to differ "n" from 66 - they appear in different
>>> contexts
>> So, if a skype name is n66n66n66, can you please show the equivalent
>> numeral
>> key representation a user has to dial?
> 
> #66## 66 #66## 66 #66## 66
> 
> I have added spaces for better readability.
So, ## means to decode into alpha characters. Cool.

-----
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/How-to-dial-a-remote-skypename-from-a-telephone-keypad-tp6007946p6012521.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brian at freeswitch.org  Thu Feb 10 20:46:37 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 10 Feb 2011 11:46:37 -0600
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <AANLkTi=nA76tJcdXN2Ta3cyzm5KnQhBadNJE_swg1Xk0@mail.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>
Message-ID: <53C394DD-308B-438C-8D10-93724A453664@freeswitch.org>

The license server MUST be started as root it will then change to user freeswitch but to setup everything it can't start as anyone but root.

/b

On Feb 10, 2011, at 10:42 AM, Massimiliano Ravelli wrote:

> 2011/2/9 Victor Chukalovskiy <victor.chukalovskiy at utoronto.ca>
> Hi Folks,
> 
> Thank you everyone. Solved by lunching server separately.
> 
> I have the same problem:
> freeswitch at internal> g729_info 
> Can't contact licence server.
> 
> Quite recent freeswitch version (3 days old - git 2b4f163826132cf55698008a22c065374320796a) 
> installed in /opt/freeswitch and started with "/opt/freeswitch/bin/freeswitch -nc"
> No problem nor warning using fsg729-191-installer.
> The licence server starts automatically upon loading of mod_com_g729.
> 
> I registered succesfully some licences.
> 
> I even tried to restart freeswitch, I killed licence server and launched manually...
> The licence server starts but I still get "Can't contact licence server." 
> 
> I notice only a weird thing trying to autocomplete g729_ on fs_cli:
> [      g729_available]    [          g729_count]    [           g729_info]    [           g729_used]    
> [      g729_avaliable]    
> 
> Any hints ?
> 
> Thanks very much in advance
> 
> Regards
> Massimiliano

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From br at bsdpad.com  Thu Feb 10 20:31:24 2011
From: br at bsdpad.com (Ruslan Bukin)
Date: Thu, 10 Feb 2011 20:31:24 +0300
Subject: [Freeswitch-users] invalid XML / mod_xml_cdr
Message-ID: <20110210173124.GA22931@bsdjail.com>

Mod_xml_cdr sometimes send invalid XML, smth like:

<read_codec>G729</read_codec>
<1>8000</1>
<write_codec>G729</write_codec>
<write_rate>8000</write_rate>

or

<signal_bridge>true</signal_bridge>
<01>e6c45b44-1535-e011-a46e-001e8c44b22f</01>
<endpoint_disposition>ANSWER</endpoint_disposition>

any recommendations?

I am using freeswitch latest git on freebsd8

thanks

-Ruslan


From massimiliano.ravelli at gmail.com  Thu Feb 10 20:56:48 2011
From: massimiliano.ravelli at gmail.com (Massimiliano Ravelli)
Date: Thu, 10 Feb 2011 18:56:48 +0100
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <53C394DD-308B-438C-8D10-93724A453664@freeswitch.org>
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>
Message-ID: <AANLkTin4S6czPQvTOar-TBrSpRrZeQzKrOxapKk7rcgP@mail.gmail.com>

2011/2/10 Brian West <brian at freeswitch.org>

> The license server MUST be started as root it will then change to user
> freeswitch but to setup everything it can't start as anyone but root.
>

In my previous installation (an older version) it was started automatically
by freeswitch and worked like a charm.
In this case, when I tried to launch manually I did use root.

Massimiliano
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From jerry.richards at teotech.com  Thu Feb 10 21:15:26 2011
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 10 Feb 2011 10:15:26 -0800
Subject: [Freeswitch-users] Quiet packets on non-bypass media calls
Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F750CDE@VA3DIAXVS351.RED001.local>

Hello All,

All of our Freeswitch installations transmit occasional quiet packets in calls when the media is not bypassed.  One case that is easy to analyze is two phones on a meet-me conference.   Packets containing a continuous tone can be seen being transmitted from a phone to Freeswitch with Wireshark.  The media stream to the other phone from Freeswitch has occasional 20 millisecond packets with near zero amplitude.  This is also seen on PRI calls and in voicemail recordings.  It looks like Freeswitch is ignoring a packet that it received.  The jitter, sequence number, and timestamps of all packets are good.  This occurrence rate is similar on switches with only two phones and on more heavily loaded switches.  The rate can be anywhere between no occurrences in an hour to 20 in a minute.

The drop outs are easy to detect when a continuous tone is being sent by one phone.  The drop outs are more difficult to detect in normal conversation.  The servers have fast quad core processors and 4 to 6 gigabytes of RAM.  The latest builds of Freeswitch and 32 or 64 bit versions of Centos 5.3 are used.

Does anyone know what might cause this?

Thanks,
Jerry
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From ryoung at strongholdwax.com  Thu Feb 10 21:16:18 2011
From: ryoung at strongholdwax.com (Roger Young)
Date: Thu, 10 Feb 2011 13:16:18 -0500
Subject: [Freeswitch-users] can't email voicemail messages
Message-ID: <4D542B72.9010806@strongholdwax.com>

Outbound mail is rejected because the "from" address is 
"1000 at localcomputername".  Voicemail messages are sent using the 
sendmail command.  The mailer is actually postfix.  I have set 
vm_mailfrom and email_from both to "workingmail at realdomain.com" both in 
conf/directory/default/1000.xml and 
conf/autoload_configs/voicemail.conf.xml.  I have changed the 
voicemail.tpl so it has the literal, correct sender address.  I have 
tried to change the "mailer-app-args" in 
conf/autoload_confings/switch.conf.xml to include a " -f 
workingmail at realdomain.com ".  All of this is ignored, and 
"1000 at computername" appears as the from address in the rejection 
notices.  Can anybody point me to the correct way to set the "from" 
address so it is actually applied to outbound emails?  The sendmail 
command works fine from the command line.

This is my first post.  Let me know if I am not following the rules.

Thanks,
Roger Young



From dmitry.bely at gmail.com  Thu Feb 10 21:34:54 2011
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Thu, 10 Feb 2011 21:34:54 +0300
Subject: [Freeswitch-users] How to dial a remote_skypename from a
 telephone keypad?
In-Reply-To: <1297357907611-6012521.post@n2.nabble.com>
References: <AANLkTikdMDk_NkKVyH6Qd4DBq5izujG77eYPOi1ZCbms@mail.gmail.com>
	<1297276772026-6009014.post@n2.nabble.com>
	<AANLkTikDg6yJVHh7M=q0cNkacwhXpJBV5kJSThTaJGTZ@mail.gmail.com>
	<AANLkTik0wPtXNOcd95wX2TH8B0+BaY7mz3aeBZaWa3hi@mail.gmail.com>
	<1297308558454-6010434.post@n2.nabble.com>
	<AANLkTi=bDePMqM+Jf3RzQAHX3COd=hgO_A5iFWk_mA67@mail.gmail.com>
	<1297338878062-6011352.post@n2.nabble.com>
	<AANLkTi=VFhb9AfHfS0hurOzjOgLHf1tvA6sMztkCmHfL@mail.gmail.com>
	<1297352142520-6012135.post@n2.nabble.com>
	<AANLkTindVXHbCnA6mNQaq4mFX7K5KSZruxFHAPGhSb6a@mail.gmail.com>
	<1297357907611-6012521.post@n2.nabble.com>
Message-ID: <AANLkTi=_4gzUOZiYuC7dkuJYKzZHZeiv26XqnKQYT3Dq@mail.gmail.com>

On Thu, Feb 10, 2011 at 8:11 PM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
>
> Dmitry Bely wrote:
>>
>> On Thu, Feb 10, 2011 at 6:35 PM, mazilo <Nabble at slickdeals.endjunk.com>
>> wrote:
>>>
>>>
>>> Dmitry Bely wrote:
>>>> So there is no problem to differ "n" from 66 - they appear in different
>>>> contexts
>>> So, if a skype name is n66n66n66, can you please show the equivalent
>>> numeral
>>> key representation a user has to dial?
>>
>> #66## 66 #66## 66 #66## 66
>>
>> I have added spaces for better readability.
> So, ## means to decode into alpha characters. Cool.

Almost so.

# - start to encode digits into chars
## - stop to encode. Following digits will be taken as-is (default state)

- Dmitry Bely


From anthony.minessale at gmail.com  Thu Feb 10 21:38:35 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 10 Feb 2011 12:38:35 -0600
Subject: [Freeswitch-users] Quiet packets on non-bypass media calls
In-Reply-To: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F750CDE@VA3DIAXVS351.RED001.local>
References: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F750CDE@VA3DIAXVS351.RED001.local>
Message-ID: <AANLkTikf2shNKW57hLn8Nv8-ZpiW16-6MOCY80E32k0R@mail.gmail.com>

The packets are probably late and being dropped and replaced with silence.
You can try 2 things:

1) run the app cng_plc before bridge and in execute_on_originate on b legs.
2) enable the jitter buffer at a small value by setting jitterbuffer_msec=60



On Thu, Feb 10, 2011 at 12:15 PM, Jerry Richards
<jerry.richards at teotech.com> wrote:
> Hello All,
>
>
>
> All of our Freeswitch installations transmit occasional quiet packets in
> calls when the media is not bypassed.? One case that is easy to analyze is
> two phones on a meet-me conference.?? Packets containing a continuous tone
> can be seen being transmitted from a phone to Freeswitch with Wireshark.
> The media stream to the other phone from Freeswitch has occasional 20
> millisecond packets with near zero amplitude.? This is also seen on PRI
> calls and in voicemail recordings.? It looks like Freeswitch is ignoring a
> packet that it received.? The jitter, sequence number, and timestamps of all
> packets are good.? This occurrence rate is similar on switches with only two
> phones and on more heavily loaded switches.? The rate can be anywhere
> between no occurrences in an hour to 20 in a minute.
>
>
>
> The drop outs are easy to detect when a continuous tone is being sent by one
> phone.? The drop outs are more difficult to detect in normal conversation.
> The servers have fast quad core processors and 4 to 6 gigabytes of RAM.? The
> latest builds of Freeswitch and 32 or 64 bit versions of Centos 5.3 are
> used.
>
>
>
> Does anyone know what might cause this?
>
>
>
> Thanks,
>
> Jerry
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 tlainvestments.com  Thu Feb 10 21:46:07 2011
From: freeswitch at tlainvestments.com (Troy Anderson)
Date: Thu, 10 Feb 2011 11:46:07 -0700
Subject: [Freeswitch-users] general protection
Message-ID: <EEB02C60-B1E2-47DB-B184-8EDBCFD2C170@tlainvestments.com>

Hi there,

Is there any way to determine what might be the cause of the following message log?  I have this system in a production environment, which makes it difficult to update.  It is running a checkout of FS from Nov 18.  It runs pretty reliably, but once in a while it faults:

kernel: freeswitch[9478] general protection rip:2aaab0f16203 rsp:409f5d08 error:0

Thanks,
Troy

From anthony.minessale at gmail.com  Thu Feb 10 21:49:42 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 10 Feb 2011 12:49:42 -0600
Subject: [Freeswitch-users] hold_music var gets unset/lost by
 cancelled/failed att_xfer
In-Reply-To: <AANLkTikEx9ZwNADRQ8k2JbTwgoLPFmHUPpPL3RLnZZU8@mail.gmail.com>
References: <AANLkTimLod3yHrex4LAK=4gdVcPhzXutKYyw=E8ZMcN9@mail.gmail.com>
	<AANLkTi=u3VeHPuS5x1e+yu5QNBw5ZSgHvAWOT+F6A_OR@mail.gmail.com>
	<AANLkTi=EndkiUJvCgCSirxCA23EwWD_++0h3Qcr=mg9g@mail.gmail.com>
	<AANLkTikt97dugbhKPgnB2ws4bOFcxg8yhzSdyxHMab_m@mail.gmail.com>
	<AANLkTikEx9ZwNADRQ8k2JbTwgoLPFmHUPpPL3RLnZZU8@mail.gmail.com>
Message-ID: <AANLkTinS3wxA8izr9tXX7jhYJPHq_X_coO7UNoUkjs-P@mail.gmail.com>

I pushed a patch that will probably delay the bowout until after the
att_xfer is over
Give it a try.

commit 3546654615f88058fb6769fe79e07162602fa4af
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Thu Feb 10 12:37:14 2011 -0600

    don't bow out on att_xfer bridge


On Thu, Feb 10, 2011 at 1:03 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> Thanks! It works now! BTW, att_xfer with
> loopback_bowout/bowout_on_execute set to false seems to be working
> too. Is there a way to make loopback channel leave the path? Without
> bowout turned off att_xfer doesn't work...
>
> 2011/2/8 Anthony Minessale <anthony.minessale at gmail.com>:
>> try latest GIT
>>
>> On Tue, Feb 8, 2011 at 1:28 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>> I have same issue with MOH and att_xfer on failed transfers, music on
>>> hold played only once
>>> At the same time, transfer_ringback always plays correctly to transferer
>>>
>>> 2011/2/8 Anthony Minessale <anthony.minessale at gmail.com>:
>>>> you really should report this to jira not to the mailing list.
>>>> http://jira.freeswitch.org
>>>>
>>>>
>>>>
>>>> On Mon, Feb 7, 2011 at 4:23 PM, Fraser Redmond <fraserredmond at gmail.com> wrote:
>>>>> I'm trying to do a second att_xfer on a call so that if the first attended
>>>>> transfer fails (c-leg is busy, or presses do-not-answer, or is an extn that
>>>>> doesn't exist) then the call could be transferred to someone else.
>>>>>
>>>>> On the first att_xfer the person on hold hears the hold_music correctly.
>>>>> Once that transfer is cancelled or fails:
>>>>> -- On any subsequent att_xfer's the person on hold just hears silence.
>>>>> -- If they are put on hold they just hear silence.
>>>>>
>>>>> I tried setting hold_music again for each channel after the first att_xfer,
>>>>> but that didn't work, so it's probably not actually a problem with
>>>>> hold_music per se, but some other variable/setting that decides whether to
>>>>> use hold_music.
>>>>>
>>>>> I also tried doing a uuid_dump before and after each attempt, but didn't
>>>>> notice anything too different - unless it's a matter of unsetting one of the
>>>>> couple of changed/new vars like:
>>>>> variable_originate_disposition
>>>>> variable_current_application
>>>>> variable_playback_seconds
>>>>>
>>>>> I get the feeling other variables are probably also lost by the first failed
>>>>> transfer as the second att_xfer has some odd things happen if the third
>>>>> party does answer. Haven't been able to narrow it down as closely as the
>>>>> hold_music, but two things I've seen happen are:
>>>>> -- The party that initiated the transfer gets hung up automatically (after
>>>>> 30 sec)
>>>>> -- When the party that initiated the transfer hangs up it should connect the
>>>>> other two parties, but instead it hung up all three
>>>>>
>>>>> Cheers,
>>>>> Fraser
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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  Thu Feb 10 21:56:25 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 10 Feb 2011 12:56:25 -0600
Subject: [Freeswitch-users] general protection
In-Reply-To: <EEB02C60-B1E2-47DB-B184-8EDBCFD2C170@tlainvestments.com>
References: <EEB02C60-B1E2-47DB-B184-8EDBCFD2C170@tlainvestments.com>
Message-ID: <AANLkTik_cxh2zmHk0WzC-4ngEue+egqRVe3rO5zqTS-O@mail.gmail.com>

you need to find the core file and debug it.
the problem is once we examine it the answer will probably to update
to fix it and on the same token we need to be working with the latest
code to fix it in the event that it still exits.
So you may want to prevent all of that and see if updating fixes it first.



On Thu, Feb 10, 2011 at 12:46 PM, Troy Anderson
<freeswitch at tlainvestments.com> wrote:
> Hi there,
>
> Is there any way to determine what might be the cause of the following message log? ?I have this system in a production environment, which makes it difficult to update. ?It is running a checkout of FS from Nov 18. ?It runs pretty reliably, but once in a while it faults:
>
> kernel: freeswitch[9478] general protection rip:2aaab0f16203 rsp:409f5d08 error:0
>
> Thanks,
> Troy
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 kris at livecall.com  Thu Feb 10 22:01:58 2011
From: kris at livecall.com (Kris)
Date: Thu, 10 Feb 2011 11:01:58 -0800
Subject: [Freeswitch-users] How to dial a remote_skypename from
	atelephone keypad?
References: <AANLkTikJNH3-vk=dMkOjAKsAxgr7KaiGD-U-9t2uKjxE@mail.gmail.com><1297272133009-6008667.post@n2.nabble.com><AANLkTikdMDk_NkKVyH6Qd4DBq5izujG77eYPOi1ZCbms@mail.gmail.com><1297276772026-6009014.post@n2.nabble.com><AANLkTikDg6yJVHh7M=q0cNkacwhXpJBV5kJSThTaJGTZ@mail.gmail.com><AANLkTik0wPtXNOcd95wX2TH8B0+BaY7mz3aeBZaWa3hi@mail.gmail.com><1297308558454-6010434.post@n2.nabble.com><AANLkTi=bDePMqM+Jf3RzQAHX3COd=hgO_A5iFWk_mA67@mail.gmail.com><1297338878062-6011352.post@n2.nabble.com><AANLkTi=VFhb9AfHfS0hurOzjOgLHf1tvA6sMztkCmHfL@mail.gmail.com><1297352142520-6012135.post@n2.nabble.com>
	<AANLkTindVXHbCnA6mNQaq4mFX7K5KSZruxFHAPGhSb6a@mail.gmail.com>
Message-ID: <CAAD618E7377469CBE49604E7EF483FD@stor1>

I think it would be best to read  back to the caller what he is entering. I 
will probably use something like this to enter postal codes.

to take this example:    n66n66n66

"On you dialpad, please enter a digit or the caracter on that button. For 
special characters like coma or space, press 1":(1 can have "1 ,!_.")
Start:
Input: 6
Speak:"Six", "Press again to pick another character or # to save"
Input:6
Speak::"M", "Press again to pick another character or # to save"
Input:6
Speak::"N", "Press again to pick another character or # to save"
(After "O", it starts with "6" again)
Input:#
Save:
Speak"Enter the next digit or character or # for the menu"
Goto Start:
Of course on the next character, the caller presses 6, hears "six" and 
presses #. goes to Save:
Or #
Menu:
"To hear your completed entry, press 1, To continue entering press 2, To 
delete and start over press 3, If you it's correct, press 9 to exit"


Kris

----- Original Message ----- 
From: "Dmitry Bely" <dmitry.bely at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, February 10, 2011 8:19 AM
Subject: Re: [Freeswitch-users] How to dial a remote_skypename from 
atelephone keypad?


> On Thu, Feb 10, 2011 at 6:35 PM, mazilo <Nabble at slickdeals.endjunk.com> 
> wrote:
>>
>>
>> Dmitry Bely wrote:
>>> So there is no problem to differ "n" from 66 - they appear in different
>>> contexts
>> So, if a skype name is n66n66n66, can you please show the equivalent 
>> numeral
>> key representation a user has to dial?
>
> #66## 66 #66## 66 #66## 66
>
> I have added spaces for better readability.
>
> - Dmitry Bely
>
>
> 



From david.villasmil.work at gmail.com  Thu Feb 10 23:15:52 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 10 Feb 2011 21:15:52 +0100
Subject: [Freeswitch-users] mod_distributor
Message-ID: <AANLkTinnH0Z_=tV6Yz1VSzZX=N1x4wBYYp7wQ69wTmYX@mail.gmail.com>

Hello Guys,

The was I use bridgeis like the following from a lua script:

session:execute("bridge","{sip_auth_username=1234,sip_auth_password=1234}sofia/external/5551234566 at 1.2.3.4:5060")

This is because i don't want tothe xml files, i want to do it programmatically.



Lets say i have 3 different gateways i want to use in a round-robin basis.

Normally, i would need to DEFINE 3 gateways in the xml profiles, then
define mod_distributor by creating a "list" and then use it like:

session:execute("bridge","sofia/external/${destination_number}@${distributor(distributor_list)}")

but i would like to define the "distributor_list" when creating the
bridge, is this possible?


Thanks all


David


From nazim.aghabayov at gmail.com  Thu Feb 10 23:16:57 2011
From: nazim.aghabayov at gmail.com (Nazim Aghabayov)
Date: Fri, 11 Feb 2011 00:16:57 +0400
Subject: [Freeswitch-users] IBM Server restarts If I start sangoma
	wanrouter
In-Reply-To: <AANLkTim_e3UrCj_cdzvozA9OCo+WeyVz=KAZZFMRdzYm@mail.gmail.com>
References: <AANLkTim_e3UrCj_cdzvozA9OCo+WeyVz=KAZZFMRdzYm@mail.gmail.com>
Message-ID: <4D5447B9.2030109@gmail.com>

Hi Venkatesan,

It seems that your server runs a XEN kernel with SELinux enabled, that may cause a problem.
1) You may try to disable SELinux completely, reboot the server and see what happens.
2) If it still reboots, try a regular kernel (no xen no pae)
3) If problem persists, try enabling kernel debugger or contact Sangoma support.

b.t.w Sangoma support engineers may help you much more faster than the FreeSWITCH-users participants )

Regards,
Nazim


On 02/10/2011 05:25 PM, ovvenkat wrote:
> Hi ,
>
> I am using IBM System X3400 M3 Server with Redhat Enterprise Edition 5.6 and
> .
> A101E Sangoma PRI Card.
>
>
>
> When I start wanrouter , My server getting restart.
> Could onyone faced same kind of issue?
> Please find attached log files.
>
> I use, wanrouter version 3.5.18
>
> Anyone plz help me to resolve 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



From michal.bielicki at seventhsignal.de  Thu Feb 10 23:19:35 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Thu, 10 Feb 2011 21:19:35 +0100
Subject: [Freeswitch-users] Language files - French, German, Spanish,
	etc.
In-Reply-To: <AANLkTimyi3BGKrKoH1TzuSS4dKWv4B=SqjuxVJYvCRiS@mail.gmail.com>
References: <AANLkTim4Wc+hV_aCufsa1VkiBheCEWJGC16OhzRQ7fi8@mail.gmail.com>
	<AANLkTintUvJKvicf7n8-P3fCywqocY8CNVeqVJPZJc+w@mail.gmail.com>
	<AANLkTimyi3BGKrKoH1TzuSS4dKWv4B=SqjuxVJYvCRiS@mail.gmail.com>
Message-ID: <169322DE-BD65-4752-8258-9512E7AB096B@seventhsignal.de>

We are getting quotes for german sounds. will take a couple of weeks but we are getting there

Am 10.02.2011 um 17:49 schrieb Michael Collins:

> Yehavi,
> 
> This is wonderful! Thank you so much! Please email me off list and we'll handle the particulars.
> 
> -MC
> 
> On Thu, Feb 10, 2011 at 3:07 AM, Yehavi Bourvine <yehavi.bourvine at gmail.com> wrote:
> Hello Michael,
>  
>   We have Hebrew support: mod_say_he and professional recodrings. How can we donate them to the public and add them to FreeSwitch source tree?
>  
>                                  Thanks! __Yehavi:
> 
> 2010/10/29 Michael Collins <msc at freeswitch.org>
> Hello all!
> 
> 
> I have heard about several of our intrepid community members creating sound sets in various languages. If you have created a sound set in any language other than US English please contact me off list. I want to help get this sound prompts organized so that the global community can use them.
> 
> Thanks!
> -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
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 jerry.richards at teotech.com  Thu Feb 10 23:27:10 2011
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 10 Feb 2011 12:27:10 -0800
Subject: [Freeswitch-users] Quiet packets on non-bypass media calls
In-Reply-To: <AANLkTikf2shNKW57hLn8Nv8-ZpiW16-6MOCY80E32k0R@mail.gmail.com>
References: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F750CDE@VA3DIAXVS351.RED001.local>
	<AANLkTikf2shNKW57hLn8Nv8-ZpiW16-6MOCY80E32k0R@mail.gmail.com>
Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F750DC1@VA3DIAXVS351.RED001.local>

Is cng_plc and execute_on_originate documented in the Wiki or somewhere?

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Thursday, February 10, 2011 10:39 AM
To: FreeSWITCH Users Help
Cc: Bob Greaby
Subject: Re: [Freeswitch-users] Quiet packets on non-bypass media calls

The packets are probably late and being dropped and replaced with silence.
You can try 2 things:

1) run the app cng_plc before bridge and in execute_on_originate on b legs.
2) enable the jitter buffer at a small value by setting jitterbuffer_msec=60



On Thu, Feb 10, 2011 at 12:15 PM, Jerry Richards <jerry.richards at teotech.com> wrote:
> Hello All,
>
>
>
> All of our Freeswitch installations transmit occasional quiet packets 
> in calls when the media is not bypassed.? One case that is easy to 
> analyze is two phones on a meet-me conference.?? Packets containing a 
> continuous tone can be seen being transmitted from a phone to Freeswitch with Wireshark.
> The media stream to the other phone from Freeswitch has occasional 20 
> millisecond packets with near zero amplitude.? This is also seen on 
> PRI calls and in voicemail recordings.? It looks like Freeswitch is 
> ignoring a packet that it received.? The jitter, sequence number, and 
> timestamps of all packets are good.? This occurrence rate is similar 
> on switches with only two phones and on more heavily loaded switches.? 
> The rate can be anywhere between no occurrences in an hour to 20 in a minute.
>
>
>
> The drop outs are easy to detect when a continuous tone is being sent 
> by one phone.? The drop outs are more difficult to detect in normal conversation.
> The servers have fast quad core processors and 4 to 6 gigabytes of 
> RAM.? The latest builds of Freeswitch and 32 or 64 bit versions of 
> Centos 5.3 are used.
>
>
>
> Does anyone know what might cause this?
>
>
>
> Thanks,
>
> Jerry
>
> _______________________________________________
> FreeSWITCH-users 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
>
>



--
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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  Thu Feb 10 23:52:32 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 10 Feb 2011 14:52:32 -0600
Subject: [Freeswitch-users] Quiet packets on non-bypass media calls
In-Reply-To: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F750DC1@VA3DIAXVS351.RED001.local>
References: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F750CDE@VA3DIAXVS351.RED001.local>
	<AANLkTikf2shNKW57hLn8Nv8-ZpiW16-6MOCY80E32k0R@mail.gmail.com>
	<2BF7FB90DF25EA4485949F3AF2B9D6960B3F750DC1@VA3DIAXVS351.RED001.local>
Message-ID: <AANLkTinoiMCPp_Z=F4zfs2xWSNuD2ATw7dRqNTDmSNd2@mail.gmail.com>

No they are beta features they are not docd yet. cng plc is just an app that
says to perform plc on any lost packets and execute on originate is like
execute on answer etc but only for outbound calls during originate.
On Feb 10, 2011 2:28 PM, "Jerry Richards" <jerry.richards at teotech.com>
wrote:
> Is cng_plc and execute_on_originate documented in the Wiki or somewhere?
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
> Sent: Thursday, February 10, 2011 10:39 AM
> To: FreeSWITCH Users Help
> Cc: Bob Greaby
> Subject: Re: [Freeswitch-users] Quiet packets on non-bypass media calls
>
> The packets are probably late and being dropped and replaced with silence.
> You can try 2 things:
>
> 1) run the app cng_plc before bridge and in execute_on_originate on b
legs.
> 2) enable the jitter buffer at a small value by setting
jitterbuffer_msec=60
>
>
>
> On Thu, Feb 10, 2011 at 12:15 PM, Jerry Richards <
jerry.richards at teotech.com> wrote:
>> Hello All,
>>
>>
>>
>> All of our Freeswitch installations transmit occasional quiet packets
>> in calls when the media is not bypassed.  One case that is easy to
>> analyze is two phones on a meet-me conference.   Packets containing a
>> continuous tone can be seen being transmitted from a phone to Freeswitch
with Wireshark.
>> The media stream to the other phone from Freeswitch has occasional 20
>> millisecond packets with near zero amplitude.  This is also seen on
>> PRI calls and in voicemail recordings.  It looks like Freeswitch is
>> ignoring a packet that it received.  The jitter, sequence number, and
>> timestamps of all packets are good.  This occurrence rate is similar
>> on switches with only two phones and on more heavily loaded switches.
>> The rate can be anywhere between no occurrences in an hour to 20 in a
minute.
>>
>>
>>
>> The drop outs are easy to detect when a continuous tone is being sent
>> by one phone.  The drop outs are more difficult to detect in normal
conversation.
>> The servers have fast quad core processors and 4 to 6 gigabytes of
>> RAM.  The latest builds of Freeswitch and 32 or 64 bit versions of
>> Centos 5.3 are used.
>>
>>
>>
>> Does anyone know what might cause this?
>>
>>
>>
>> Thanks,
>>
>> Jerry
>>
>> _______________________________________________
>> FreeSWITCH-users 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
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 steveayre at gmail.com  Fri Feb 11 00:07:30 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 10 Feb 2011 21:07:30 +0000
Subject: [Freeswitch-users] general protection
In-Reply-To: <EEB02C60-B1E2-47DB-B184-8EDBCFD2C170@tlainvestments.com>
References: <EEB02C60-B1E2-47DB-B184-8EDBCFD2C170@tlainvestments.com>
Message-ID: <AANLkTimPxs+rvit8y3BXsxrNdMn1A=kTNxsvKvtviHyG@mail.gmail.com>

It's a segmentation fault. You get different messages depending on which
area of memory it accidentally tried to access.

Run freeswitch with the -core option. If it dies it will coredump, saving
all of its memory and state to disk. That lets you find out what it was
trying to do when it crashed, and report it on http://jira.freeswitch.org/.

See:
http://wiki.freeswitch.org/wiki/Reporting_Bugs
http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Simple_bash_script_to_make_debug_easy

-Steve



On 10 February 2011 18:46, Troy Anderson <freeswitch at tlainvestments.com>wrote:

> Hi there,
>
> Is there any way to determine what might be the cause of the following
> message log?  I have this system in a production environment, which makes it
> difficult to update.  It is running a checkout of FS from Nov 18.  It runs
> pretty reliably, but once in a while it faults:
>
> kernel: freeswitch[9478] general protection rip:2aaab0f16203 rsp:409f5d08
> error:0
>
> Thanks,
> Troy
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 11 00:09:12 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 10 Feb 2011 21:09:12 +0000
Subject: [Freeswitch-users] mod_distributor
In-Reply-To: <AANLkTinnH0Z_=tV6Yz1VSzZX=N1x4wBYYp7wQ69wTmYX@mail.gmail.com>
References: <AANLkTinnH0Z_=tV6Yz1VSzZX=N1x4wBYYp7wQ69wTmYX@mail.gmail.com>
Message-ID: <AANLkTinuWiAPnYQLD=aZxJa59qHN7gJb3OL1uh-tLycX@mail.gmail.com>

No.

The reason it must be defined before the bridge is it keeps state
information inbetween the bridges so it knows how many times the current
node has been returned, before moving onto returning the next node. That's
how the weighting's implemented.

-Steve



On 10 February 2011 20:15, David Villasmil
<david.villasmil.work at gmail.com>wrote:

> Hello Guys,
>
> The was I use bridgeis like the following from a lua script:
>
>
> session:execute("bridge","{sip_auth_username=1234,sip_auth_password=1234}sofia/external/
> 5551234566 at 1.2.3.4:5060")
>
> This is because i don't want tothe xml files, i want to do it
> programmatically.
>
>
>
> Lets say i have 3 different gateways i want to use in a round-robin basis.
>
> Normally, i would need to DEFINE 3 gateways in the xml profiles, then
> define mod_distributor by creating a "list" and then use it like:
>
> session:execute("bridge","sofia/external/${destination_number}@
> ${distributor(distributor_list)}")
>
> but i would like to define the "distributor_list" when creating the
> bridge, is this possible?
>
>
> 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 steveayre at gmail.com  Fri Feb 11 00:15:18 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 10 Feb 2011 21:15:18 +0000
Subject: [Freeswitch-users] invalid XML / mod_xml_cdr
In-Reply-To: <20110210173124.GA22931@bsdjail.com>
References: <20110210173124.GA22931@bsdjail.com>
Message-ID: <AANLkTinzwo5oK1uM3Wwq-_KB=aO5qpxAf8pWNsT=na4J@mail.gmail.com>

Define "latest" git, which commit is it?

I don't see anything wrong in the XML you pasted, or is that the entirely of
what mod_xml_cdr is sending?

-Steve



On 10 February 2011 17:31, Ruslan Bukin <br at bsdpad.com> wrote:

> Mod_xml_cdr sometimes send invalid XML, smth like:
>
> <read_codec>G729</read_codec>
> <1>8000</1>
> <write_codec>G729</write_codec>
> <write_rate>8000</write_rate>
>
> or
>
> <signal_bridge>true</signal_bridge>
> <01>e6c45b44-1535-e011-a46e-001e8c44b22f</01>
> <endpoint_disposition>ANSWER</endpoint_disposition>
>
> any recommendations?
>
> I am using freeswitch latest git on freebsd8
>
> thanks
>
> -Ruslan
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 11 00:16:09 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 10 Feb 2011 21:16:09 +0000
Subject: [Freeswitch-users] invalid XML / mod_xml_cdr
In-Reply-To: <20110210173124.GA22931@bsdjail.com>
References: <20110210173124.GA22931@bsdjail.com>
Message-ID: <AANLkTi=2jcD8xFkK=49aNnm-Cu9NGHYGuS7NU1upMkXb@mail.gmail.com>

Oh I see now... the <1> and <01> tags.

Again, which commit are you on?

-Steve


On 10 February 2011 17:31, Ruslan Bukin <br at bsdpad.com> wrote:

> Mod_xml_cdr sometimes send invalid XML, smth like:
>
> <read_codec>G729</read_codec>
> <1>8000</1>
> <write_codec>G729</write_codec>
> <write_rate>8000</write_rate>
>
> or
>
> <signal_bridge>true</signal_bridge>
> <01>e6c45b44-1535-e011-a46e-001e8c44b22f</01>
> <endpoint_disposition>ANSWER</endpoint_disposition>
>
> any recommendations?
>
> I am using freeswitch latest git on freebsd8
>
> thanks
>
> -Ruslan
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 11 00:18:35 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 10 Feb 2011 21:18:35 +0000
Subject: [Freeswitch-users] switch.conf.xml port range
In-Reply-To: <9162BE27CDE9481F88FC4B730195A046@e1705>
References: <A23A63F76A4642D6BDF80A8B0D3A6B6C@e1705>
	<AANLkTi=LnywqgoLkkiO3Yz9x4ThTbRE3jtuX0EAQyZtz@mail.gmail.com>
	<9162BE27CDE9481F88FC4B730195A046@e1705>
Message-ID: <AANLkTimBLOnx1bVqP8qfa2tn2B49og8pXzO4QGeJAAuS@mail.gmail.com>

Any address in the 127.0.0.0/8 netblock maps to 127.0.0.1

If you listen on 127.0.0.6 you'll still be able to connect to that port on
127.0.0.1 - they're equivalent.

-Steve



On 10 February 2011 17:08, Madovsky <infos at madovsky.org> wrote:

>  yes I created an alias lo:5 espcially reserved for FS
> I d only like to reorganize sockets range from other services (bind
> etc....)
>
>
> ----- Original Message -----
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Thursday, February 10, 2011 12:03 PM
> *Subject:* Re: [Freeswitch-users] switch.conf.xml port range
>
> Do you have a SIP profile bound to an interface with IP address of
> 127.0.0.6? Does that actually do something useful?
> -MC
>
> On Thu, Feb 10, 2011 at 10:57 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  in port range settings
>> is it mean that RTP sockets are created on localhost IP ?
>> if yes, how can I change the IP to 127.0.0.6 for 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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From chat2jesse at gmail.com  Fri Feb 11 01:07:50 2011
From: chat2jesse at gmail.com (jesse)
Date: Thu, 10 Feb 2011 14:07:50 -0800
Subject: [Freeswitch-users] FS cluster support of contact center.
Message-ID: <AANLkTinFf6PSQ6+b=6JVHKE5H=+KVk5SH7w_hKBSGLGb@mail.gmail.com>

FreeSWITCH does not share the state of FIFO across multiple instances,
neither does mod_callcenter.
do we have any config example about how to implement ACD in a FS cluster?

thanks!
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From msc at freeswitch.org  Fri Feb 11 01:48:43 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 10 Feb 2011 16:48:43 -0600
Subject: [Freeswitch-users] OSTAG FB Page
Message-ID: <AANLkTinv0am7kVzjJ6VW9d+hTqOE1aem4Q4QBGB_A91W@mail.gmail.com>

Hello all!

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

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

This is our official OSTAG Facebook page. :)

-Michael
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From jerry.richards at teotech.com  Fri Feb 11 02:05:05 2011
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 10 Feb 2011 15:05:05 -0800
Subject: [Freeswitch-users] Can XML CDR Timestamps Be UTC?
Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F750EB4@VA3DIAXVS351.RED001.local>

Hello All,

Is there a way to configure Freeswitch to indicate UTC (instead of local time) in the XML CDR files?

Thanks,
Jerry
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From br at bsdpad.com  Thu Feb 10 23:31:49 2011
From: br at bsdpad.com (Ruslan Bukin)
Date: Thu, 10 Feb 2011 23:31:49 +0300
Subject: [Freeswitch-users] invalid XML / mod_xml_cdr
In-Reply-To: <4d5448f0.8c8ee50a.2876.12a5@mx.google.com>
References: <4d5448f0.8c8ee50a.2876.12a5@mx.google.com>
Message-ID: <20110210203149.GA38373@bsdjail.com>

server cant parse and returns 500,
then module writes the same incorrect data into a file

-Ruslan

On Thu, Feb 10, 2011 at 02:22:18PM -0600, msc at freeswitch.org wrote:
> Is this going straight into a file or is it posting to a web server?
> -MC
> 
> Sent from my HTC on the Now Network from Sprint!
> 
> ----- Reply message -----
> From: "Ruslan Bukin" <br at bsdpad.com>
> Date: Thu, Feb 10, 2011 11:31 am
> Subject: [Freeswitch-users] invalid XML / mod_xml_cdr
> To: <freeswitch-users at lists.freeswitch.org>
> 
> Mod_xml_cdr sometimes send invalid XML, smth like:
> 
> <read_codec>G729</read_codec>
> <1>8000</1>
> <write_codec>G729</write_codec>
> <write_rate>8000</write_rate>
> 
> or
> 
> <signal_bridge>true</signal_bridge>
> <01>e6c45b44-1535-e011-a46e-001e8c44b22f</01>
> <endpoint_disposition>ANSWER</endpoint_disposition>
> 
> any recommendations?
> 
> I am using freeswitch latest git on freebsd8
> 
> thanks
> 
> -Ruslan
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 br at bsdpad.com  Fri Feb 11 00:23:31 2011
From: br at bsdpad.com (Ruslan Bukin)
Date: Fri, 11 Feb 2011 00:23:31 +0300
Subject: [Freeswitch-users] invalid XML / mod_xml_cdr
In-Reply-To: <AANLkTi=2jcD8xFkK=49aNnm-Cu9NGHYGuS7NU1upMkXb@mail.gmail.com>
References: <20110210173124.GA22931@bsdjail.com>
	<AANLkTi=2jcD8xFkK=49aNnm-Cu9NGHYGuS7NU1upMkXb@mail.gmail.com>
Message-ID: <20110210212331.GA45115@bsdjail.com>

commit 6f9da9a070517ce30e9edec6a7d2536db0646c33

-Ruslan

On Thu, Feb 10, 2011 at 09:16:09PM +0000, Steven Ayre wrote:
> Oh I see now... the <1> and <01> tags.
> 
> Again, which commit are you on?
> 
> -Steve
> 
> 
> On 10 February 2011 17:31, Ruslan Bukin <br at bsdpad.com> wrote:
> 
> > Mod_xml_cdr sometimes send invalid XML, smth like:
> >
> > <read_codec>G729</read_codec>
> > <1>8000</1>
> > <write_codec>G729</write_codec>
> > <write_rate>8000</write_rate>
> >
> > or
> >
> > <signal_bridge>true</signal_bridge>
> > <01>e6c45b44-1535-e011-a46e-001e8c44b22f</01>
> > <endpoint_disposition>ANSWER</endpoint_disposition>
> >
> > any recommendations?
> >
> > I am using freeswitch latest git on freebsd8
> >
> > thanks
> >
> > -Ruslan
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >

> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 garmt.noname at gmail.com  Fri Feb 11 00:02:23 2011
From: garmt.noname at gmail.com (Grmt)
Date: Thu, 10 Feb 2011 22:02:23 +0100
Subject: [Freeswitch-users] lua question db
In-Reply-To: <AANLkTim+B7uN+FKiPJ5B=9tMYnoaGDkm3bCMastkWzhC@mail.gmail.com>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>
	<4D52B987.7050002@communicatefreely.net>
	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>
	<4D534AFD.3020401@communicatefreely.net>
	<AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>
	<60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>
	<AANLkTimux4MBM1K_7itE4GnZQHjy_vR2Y6sr_z+u3gkR@mail.gmail.com>
	<Pine.LNX.4.64.1102101600450.18304@suse.blindi.net>
	<2243A0E7-C355-4E88-AEC1-2A918747C6BD@ipeva.fr>
	<AANLkTim+B7uN+FKiPJ5B=9tMYnoaGDkm3bCMastkWzhC@mail.gmail.com>
Message-ID: <4d545261.ca7a0e0a.1d40.02c2@mx.google.com>

 

Hmm, actually you can use the internal sqlite engine of freeswitch. 

Ledr and I recently added that to freeswitch.dbh. You can create new tables
and use them immediately.

 

If you use 'core:tablename'  as the dsn, freeswitch will automatically
create a new tablename.db sqlite file if it did not exist yet.

 

For simple queries and when using lua , there is no need to install  mysql,
postgres or whatever odbc database, and there is also no need to use LuaSQL,
and yet you still have persistent storage (unless you use in memory
filesystem).

 

Garmt

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, 10 February, 2011 18:01
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] lua question db

 

 

On Thu, Feb 10, 2011 at 10:36 AM, David Ponzone <david.ponzone at ipeva.fr>
wrote:

You can't use the internal db for your application.

You must use an external DB (Posgres, MySQL, or anyting accessible through
ODBC).

 

Technically you *could* use the db (or hash) API to store the call-forward
information in much the same way that the Local_Extension and the
call_return & call-pickup extensions work. I'll leave it as an exercise for
the OP to review these extensions in conf/dialplan/default.xml for hints on
how to proceed:

 

Local_Extension

global-intercept

group-intercept

intercept-ext

redial

call_return

 

-MC

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From kahn at vestec.com  Thu Feb 10 22:43:05 2011
From: kahn at vestec.com (Kashif Kahn)
Date: Thu, 10 Feb 2011 14:43:05 -0500
Subject: [Freeswitch-users] Vestec Speech Engine: Windows Release
Message-ID: <4D543FC9.6050303@vestec.com>

Hello Everyone,

Vestec recently launched a Windows version of its latest generation 
speech recognition engine (ASR 2.1) for use with Freeswitch and would 
appreciate if people could try it. We already support all major Linux 
distributions.

Our ASR engine is designed for "command-and-control" type IVR 
applications and offers the best deal around for enabling sophisticated 
speech recognition with Freeswitch. It can be used to speech enable a 
wide variety of DTMF menus or business processes via keywords-based 
speech interaction.

A starter kit - comprising a specially priced perpetual license for a 
full-function, full-feature standard ASR engine - is available for $25. 
Acoustic models cost an additional $9.99 per language. Please visit 
Vestec webstore at: http://www.vestec.com/products

Regards,
-Kashif

-- 

Kashif Kahn
VP Business Development
Vestec Inc
Waterloo, ON Canada
phone: +1 519 885-7615



From kahn at vestec.com  Thu Feb 10 23:14:27 2011
From: kahn at vestec.com (Kashif Kahn)
Date: Thu, 10 Feb 2011 15:14:27 -0500
Subject: [Freeswitch-users] Vestec Speech Engine: Large Vocabulary
Message-ID: <4D544723.8070202@vestec.com>

Hello Everyone,

Vestec recently launched a large vocabulary speech engine for 
"command-and-control" type IVR applications. The new engine - called 
Tier-2 - supports a vocabulary size of 2,500 keywords per recognition 
and complements our existing Tier-1 engine that has a 500 keywords 
vocabulary size. Tier-2 retails for $199 per port (ie. channel) while 
Tier-1 is priced at $99 per port. Acoustic models needs to be licensed 
separately at $9.99 per language.

Our ASR is designed for keywords-based interaction and offers the best 
deal around for speech enabling a wide variety of IVR applications, 
including DTMF menus and multi-step business processes. The speech 
engine is standards based in terms of grammar writing format and 
platform integration protocols. We support SRGS grammar (ABNF & XML) as 
well as MRCP integration (v1 & v2). In addition, we offer a highly 
scalable architecture that is capable of supporting thousands of ports 
in an efficient manner.

A starter kit - comprising a specially priced perpetual license for a 
full-function, full-feature standard ASR engine - is available for $25. 
Please visit Vestec webstore at: http://www.vestec.com/products

Regards,
-Kashif

-- 

Kashif Kahn
VP Business Development
Vestec Inc
Waterloo, ON Canada
phone: +1 519 885-7615



From infos at madovsky.org  Fri Feb 11 06:45:57 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 10 Feb 2011 22:45:57 -0500
Subject: [Freeswitch-users] double instance at relaodxml
Message-ID: <74344B58584B400B9D0AFC74311657FF@e1705>

sometimes when I run from fs_cli "reloadxml"
I notice 2 instances of FS so the last instance
log mod_event socket error , could not listen to socket....

thanks
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From infos at madovsky.org  Fri Feb 11 08:16:03 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 11 Feb 2011 00:16:03 -0500
Subject: [Freeswitch-users] expr()
Message-ID: <D3159CBF58B74DA5AA749382D5D10520@e1705>

Is the greater lower sign available in expre() api ?

I tried 

<condition field="${cond(expr(${myvar} > 0.01) ? YES : NO)}" expression="^YES$">


but retuns always yes....

thanks
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From infos at madovsky.org  Fri Feb 11 08:25:45 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 11 Feb 2011 00:25:45 -0500
Subject: [Freeswitch-users] expr()
References: <D3159CBF58B74DA5AA749382D5D10520@e1705>
Message-ID: <BEA380B5533244C0865C1E4DC2EA1BB3@e1705>

found the right wiki
thanks to forget my question
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, February 11, 2011 12:16 AM
  Subject: [Freeswitch-users] expr()


  Is the greater lower sign available in expre() api ?

  I tried 

  <condition field="${cond(expr(${myvar} > 0.01) ? YES : NO)}" expression="^YES$">


  but retuns always yes....

  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
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From chenzhanping at gmail.com  Fri Feb 11 08:29:11 2011
From: chenzhanping at gmail.com (=?GB2312?B?1bzGvbPC?=)
Date: Fri, 11 Feb 2011 13:29:11 +0800
Subject: [Freeswitch-users] Question about mod_dingaling.
Message-ID: <AANLkTikd9baOGf0bmL6m3EaiPcGy6Fgfm9=jEYBg+GB2@mail.gmail.com>

Good everyone,I use Freeswitch 1.0.6 with mod_dingaling.Now i call out from
mod_dingaling use google voice,this cann't call succeed.

This is cli log:

freeswitch at internal> 2011-02-11 13:27:39.558796 [DEBUG] sofia.c:5847 IP
121.28.15.8 Rejected by acl "domains". Falling back to Digest auth.
2011-02-11 13:27:39.834785 [DEBUG] sofia.c:5847 IP 121.28.15.8 Rejected by
acl "domains". Falling back to Digest auth.
2011-02-11 13:27:39.838761 [NOTICE] switch_channel.c:669 New Channel
sofia/internal/1000 at 184.105.153.247 [1df65ab4-0d7e-48e5-8c75-547ab51d73d7]
2011-02-11 13:27:39.838761 [DEBUG] switch_core_state_machine.c:314 (
sofia/internal/1000 at 184.105.153.247) Running State Change CS_NEW
2011-02-11 13:27:39.838761 [DEBUG] switch_core_state_machine.c:320 (
sofia/internal/1000 at 184.105.153.247) State NEW
2011-02-11 13:27:39.850754 [DEBUG] sofia.c:4153 Channel
sofia/internal/1000 at 184.105.153.247 entering state [received][100]
2011-02-11 13:27:39.850754 [DEBUG] sofia.c:4164 Remote SDP:
v=0
o=- 7 2 IN IP4 61.235.150.11
s=CounterPath eyeBeam 1.5
c=IN IP4 61.235.150.11
t=0 0
m=audio 13410 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 3 : xpehuJM9 ZEXWjImz 192.168.52.1 13410
a=alt:2 2 : 9AzLyh+Z Xgci2e96 192.168.153.1 13410
a=alt:3 1 : U0bonQcp 1CQ61b/S 61.235.150.11 13410
a=x-rtp-session-id:2BC4463425494741B4A9DA4DFD436DDF
2011-02-11 13:27:39.850754 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
[PCMU:0:8000:20]/[G7221:115:32000:20]
2011-02-11 13:27:39.850754 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
[PCMU:0:8000:20]/[G7221:107:16000:20]
2011-02-11 13:27:39.850754 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
[PCMU:0:8000:20]/[G722:9:8000:20]
2011-02-11 13:27:39.850754 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
[PCMU:0:8000:20]/[PCMU:0:8000:20]
2011-02-11 13:27:39.850754 [DEBUG] sofia_glue.c:2354 Set Codec
sofia/internal/1000 at 184.105.153.247 PCMU/8000 20 ms 160 samples
2011-02-11 13:27:39.850754 [DEBUG] sofia_glue.c:3524 Set 2833 dtmf send/recv
payload to 101
2011-02-11 13:27:39.850754 [DEBUG] sofia.c:4310 (
sofia/internal/1000 at 184.105.153.247) State Change CS_NEW -> CS_INIT
2011-02-11 13:27:39.850754 [DEBUG] switch_core_session.c:1021 Send signal
sofia/internal/1000 at 184.105.153.247 [BREAK]
2011-02-11 13:27:39.854744 [DEBUG] switch_core_state_machine.c:314 (
sofia/internal/1000 at 184.105.153.247) Running State Change CS_INIT
2011-02-11 13:27:39.854744 [DEBUG] switch_core_state_machine.c:338 (
sofia/internal/1000 at 184.105.153.247) State INIT
2011-02-11 13:27:39.854744 [DEBUG] mod_sofia.c:83
sofia/internal/1000 at 184.105.153.247 SOFIA INIT
2011-02-11 13:27:39.854744 [DEBUG] mod_sofia.c:117 (
sofia/internal/1000 at 184.105.153.247) State Change CS_INIT -> CS_ROUTING
2011-02-11 13:27:39.854744 [DEBUG] switch_core_session.c:1021 Send signal
sofia/internal/1000 at 184.105.153.247 [BREAK]
2011-02-11 13:27:39.854744 [DEBUG] switch_core_state_machine.c:338 (
sofia/internal/1000 at 184.105.153.247) State INIT going to sleep
2011-02-11 13:27:39.854744 [DEBUG] switch_core_state_machine.c:314 (
sofia/internal/1000 at 184.105.153.247) Running State Change CS_ROUTING
2011-02-11 13:27:39.854744 [DEBUG] switch_core_state_machine.c:341 (
sofia/internal/1000 at 184.105.153.247) State ROUTING
2011-02-11 13:27:39.854744 [DEBUG] mod_sofia.c:140
sofia/internal/1000 at 184.105.153.247 SOFIA ROUTING
2011-02-11 13:27:39.854744 [DEBUG] switch_core_state_machine.c:77
sofia/internal/1000 at 184.105.153.247 Standard ROUTING
2011-02-11 13:27:39.854744 [INFO] mod_dialplan_xml.c:418 Processing
1000->5018009993355 in context default
Dialplan: sofia/internal/1000 at 184.105.153.247 parsing [default->gvoice_out]
continue=false
Dialplan: sofia/internal/1000 at 184.105.153.247 Regex (PASS) [gvoice_out]
destination_number(5018009993355) =~ /^50(1\d{10})$/ break=on-false
Dialplan: sofia/internal/1000 at 184.105.153.247 Action
set(hangup_after_bridge=true)
Dialplan: sofia/internal/1000 at 184.105.153.247 Action bridge(
dingaling/gv1/+18009993355 at voice.google.com)
2011-02-11 13:27:39.854744 [DEBUG] switch_core_state_machine.c:119 (
sofia/internal/1000 at 184.105.153.247) State Change CS_ROUTING -> CS_EXECUTE
2011-02-11 13:27:39.854744 [DEBUG] switch_core_session.c:1021 Send signal
sofia/internal/1000 at 184.105.153.247 [BREAK]
2011-02-11 13:27:39.854744 [DEBUG] switch_core_state_machine.c:341 (
sofia/internal/1000 at 184.105.153.247) State ROUTING going to sleep
2011-02-11 13:27:39.854744 [DEBUG] switch_core_state_machine.c:314 (
sofia/internal/1000 at 184.105.153.247) Running State Change CS_EXECUTE
2011-02-11 13:27:39.854744 [DEBUG] switch_core_state_machine.c:348 (
sofia/internal/1000 at 184.105.153.247) State EXECUTE
2011-02-11 13:27:39.854744 [DEBUG] mod_sofia.c:226
sofia/internal/1000 at 184.105.153.247 SOFIA EXECUTE
2011-02-11 13:27:39.854744 [DEBUG] switch_core_state_machine.c:157
sofia/internal/1000 at 184.105.153.247 Standard EXECUTE
EXECUTE sofia/internal/1000 at 184.105.153.247 set(hangup_after_bridge=true)
2011-02-11 13:27:39.854744 [DEBUG] mod_dptools.c:816
sofia/internal/1000 at 184.105.153.247 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/1000 at 184.105.153.247 bridge(
dingaling/gv1/+18009993355 at voice.google.com)
2011-02-11 13:27:39.894782 [NOTICE] libdingaling.c:1307 SecSEND:
-------------------------------------------------------------------------------
<presence type="subscribe" to="+18009993355 at voice.google.com" from="
husodesar at gmail.com/gtalkAC94A6BE">
  <status>Call Me!</status>
</presence>
2011-02-11 13:27:39.894782 [NOTICE] libdingaling.c:1307 SecSEND:
-------------------------------------------------------------------------------
<presence xmlns="jabber:client" type="probe" to="
+18009993355 at voice.google.com" from="
husodesar at gmail.com/gtalkAC94A6BE"></presence>
2011-02-11 13:27:40.050810 [INFO] libdingaling.c:1305 SecRECV:
-------------------------------------------------------------------------------
<presence type="error" to="husodesar at gmail.com/gtalkAC94A6BE" from="
+18009993355 at voice.google.com">
  <error code="501" type="cancel">
    <feature-not-implemented
xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"></feature-not-implemented>
  </error>
</presence>
2011-02-11 13:27:44.595029 [NOTICE] libdingaling.c:1307 SecSEND:
-------------------------------------------------------------------------------
<presence type="subscribe" to="+18009993355 at voice.google.com" from="
husodesar at gmail.com/gtalkAC94A6BE">
  <status>Call Me!</status>
</presence>
2011-02-11 13:27:44.595029 [NOTICE] libdingaling.c:1307 SecSEND:
-------------------------------------------------------------------------------
<presence xmlns="jabber:client" type="probe" to="
+18009993355 at voice.google.com" from="
husodesar at gmail.com/gtalkAC94A6BE"></presence>
2011-02-11 13:27:44.751042 [INFO] libdingaling.c:1305 SecRECV:
-------------------------------------------------------------------------------
<presence type="error" to="husodesar at gmail.com/gtalkAC94A6BE" from="
+18009993355 at voice.google.com">
  <error code="501" type="cancel">
    <feature-not-implemented
xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"></feature-not-implemented>
  </error>
</presence>
2011-02-11 13:27:49.595349 [NOTICE] libdingaling.c:1307 SecSEND:
-------------------------------------------------------------------------------
<presence type="subscribe" to="+18009993355 at voice.google.com" from="
husodesar at gmail.com/gtalkAC94A6BE">
  <status>Call Me!</status>
</presence>
2011-02-11 13:27:49.595349 [NOTICE] libdingaling.c:1307 SecSEND:
-------------------------------------------------------------------------------
<presence xmlns="jabber:client" type="probe" to="
+18009993355 at voice.google.com" from="
husodesar at gmail.com/gtalkAC94A6BE"></presence>
2011-02-11 13:27:49.751408 [INFO] libdingaling.c:1305 SecRECV:
-------------------------------------------------------------------------------
<presence type="error" to="husodesar at gmail.com/gtalkAC94A6BE" from="
+18009993355 at voice.google.com">
  <error code="501" type="cancel">
    <feature-not-implemented
xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"></feature-not-implemented>
  </error>
</presence>
2011-02-11 13:27:54.595667 [NOTICE] libdingaling.c:1307 SecSEND:
-------------------------------------------------------------------------------
<presence type="subscribe" to="+18009993355 at voice.google.com" from="
husodesar at gmail.com/gtalkAC94A6BE">
  <status>Call Me!</status>
</presence>
2011-02-11 13:27:54.595667 [NOTICE] libdingaling.c:1307 SecSEND:
-------------------------------------------------------------------------------
<presence xmlns="jabber:client" type="probe" to="
+18009993355 at voice.google.com" from="
husodesar at gmail.com/gtalkAC94A6BE"></presence>
2011-02-11 13:27:54.751666 [INFO] libdingaling.c:1305 SecRECV:
-------------------------------------------------------------------------------
<presence type="error" to="husodesar at gmail.com/gtalkAC94A6BE" from="
+18009993355 at voice.google.com">
  <error code="501" type="cancel">
    <feature-not-implemented
xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"></feature-not-implemented>
  </error>
</presence>
2011-02-11 13:27:56.495776 [DEBUG] mod_dingaling.c:1715 Unknown Recipient!
2011-02-11 13:27:56.495776 [DEBUG] mod_dingaling.c:704 Terminate called from
line 1716 state=CS_NEW
2011-02-11 13:27:56.495776 [NOTICE] mod_dingaling.c:715 Close Channel N/A
[CS_NEW]
2011-02-11 13:27:56.495776 [DEBUG] switch_core_state_machine.c:428 ()
Running State Change CS_DESTROY
2011-02-11 13:27:56.495776 [DEBUG] switch_core_state_machine.c:439 (N/A)
State DESTROY
2011-02-11 13:27:56.495776 [DEBUG] switch_core_state_machine.c:439 (N/A)
State DESTROY going to sleep
2011-02-11 13:27:56.495776 [ERR] switch_ivr_originate.c:2430 Cannot create
outgoing channel of type [dingaling] cause: [NO_USER_RESPONSE]
2011-02-11 13:27:56.495776 [DEBUG] switch_ivr_originate.c:3228 Originate
Resulted in Error Cause: 18 [NO_USER_RESPONSE]
2011-02-11 13:27:56.495776 [INFO] mod_dptools.c:2355 Originate Failed.
Cause: NO_USER_RESPONSE
2011-02-11 13:27:56.495776 [NOTICE] switch_core_state_machine.c:185
sofia/internal/1000 at 184.105.153.247 has executed the last dialplan
instruction, hanging up.
2011-02-11 13:27:56.495776 [NOTICE] switch_core_state_machine.c:187 Hangup
sofia/internal/1000 at 184.105.153.247 [CS_EXECUTE] [NORMAL_CLEARING]
2011-02-11 13:27:56.495776 [DEBUG] switch_channel.c:2102 Send signal
sofia/internal/1000 at 184.105.153.247 [KILL]
2011-02-11 13:27:56.495776 [DEBUG] switch_core_session.c:1021 Send signal
sofia/internal/1000 at 184.105.153.247 [BREAK]
2011-02-11 13:27:56.495776 [DEBUG] switch_core_state_machine.c:348 (
sofia/internal/1000 at 184.105.153.247) State EXECUTE going to sleep
2011-02-11 13:27:56.495776 [DEBUG] switch_core_state_machine.c:314 (
sofia/internal/1000 at 184.105.153.247) Running State Change CS_HANGUP
2011-02-11 13:27:56.495776 [DEBUG] switch_core_state_machine.c:499 (
sofia/internal/1000 at 184.105.153.247) State HANGUP
2011-02-11 13:27:56.495776 [DEBUG] mod_sofia.c:414 Channel
sofia/internal/1000 at 184.105.153.247 hanging up, cause: NORMAL_CLEARING
2011-02-11 13:27:56.511787 [DEBUG] mod_sofia.c:476 Responding to INVITE
with: 480
2011-02-11 13:27:56.511787 [DEBUG] switch_core_state_machine.c:46
sofia/internal/1000 at 184.105.153.247 Standard HANGUP, cause: NORMAL_CLEARING
2011-02-11 13:27:56.511787 [DEBUG] switch_core_state_machine.c:499 (
sofia/internal/1000 at 184.105.153.247) State HANGUP going to sleep
2011-02-11 13:27:56.511787 [DEBUG] switch_core_state_machine.c:333 (
sofia/internal/1000 at 184.105.153.247) State Change CS_HANGUP -> CS_REPORTING
2011-02-11 13:27:56.511787 [DEBUG] switch_core_session.c:1021 Send signal
sofia/internal/1000 at 184.105.153.247 [BREAK]
2011-02-11 13:27:56.511787 [DEBUG] switch_core_state_machine.c:314 (
sofia/internal/1000 at 184.105.153.247) Running State Change CS_REPORTING
2011-02-11 13:27:56.511787 [DEBUG] switch_core_state_machine.c:590 (
sofia/internal/1000 at 184.105.153.247) State REPORTING
2011-02-11 13:27:56.511787 [DEBUG] switch_core_state_machine.c:53
sofia/internal/1000 at 184.105.153.247 Standard REPORTING, cause:
NORMAL_CLEARING
2011-02-11 13:27:56.511787 [DEBUG] switch_core_state_machine.c:590 (
sofia/internal/1000 at 184.105.153.247) State REPORTING going to sleep
2011-02-11 13:27:56.511787 [DEBUG] switch_core_state_machine.c:327 (
sofia/internal/1000 at 184.105.153.247) State Change CS_REPORTING -> CS_DESTROY
2011-02-11 13:27:56.511787 [DEBUG] switch_core_session.c:1021 Send signal
sofia/internal/1000 at 184.105.153.247 [BREAK]
2011-02-11 13:27:56.511787 [DEBUG] switch_core_session.c:1164 Session 12 (
sofia/internal/1000 at 184.105.153.247) Locked, Waiting on external entities
2011-02-11 13:27:56.511787 [NOTICE] switch_core_session.c:1182 Session 12 (
sofia/internal/1000 at 184.105.153.247) Ended
2011-02-11 13:27:56.511787 [NOTICE] switch_core_session.c:1184 Close Channel
sofia/internal/1000 at 184.105.153.247 [CS_DESTROY]
2011-02-11 13:27:56.511787 [DEBUG] switch_core_state_machine.c:428 (
sofia/internal/1000 at 184.105.153.247) Running State Change CS_DESTROY
2011-02-11 13:27:56.511787 [DEBUG] switch_core_state_machine.c:439 (
sofia/internal/1000 at 184.105.153.247) State DESTROY
2011-02-11 13:27:56.511787 [DEBUG] mod_sofia.c:341
sofia/internal/1000 at 184.105.153.247 SOFIA DESTROY
2011-02-11 13:27:56.511787 [DEBUG] switch_core_state_machine.c:60
sofia/internal/1000 at 184.105.153.247 Standard DESTROY
2011-02-11 13:27:56.511787 [DEBUG] switch_core_state_machine.c:439 (
sofia/internal/1000 at 184.105.153.247) State DESTROY going to sleep

What's wrong with this?

Please help me, Thanks.
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From anthony.minessale at gmail.com  Fri Feb 11 09:11:39 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 11 Feb 2011 00:11:39 -0600
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: <AANLkTimXmNou2T4tLPMACscdHiHVqUzf3U__WeGVaEJb@mail.gmail.com>

try latest.freeswitch.org or git HEAD

On Thu, Feb 10, 2011 at 11:29 PM, ??? <chenzhanping at gmail.com> wrote:
> Good everyone,I use Freeswitch 1.0.6 with mod_dingaling.Now i call out from
> mod_dingaling use google voice,this cann't call succeed.
>
> This is cli log:
>
> freeswitch at internal> 2011-02-11 13:27:39.558796 [DEBUG] sofia.c:5847 IP
> 121.28.15.8 Rejected by acl "domains". Falling back to Digest auth.
> 2011-02-11 13:27:39.834785 [DEBUG] sofia.c:5847 IP 121.28.15.8 Rejected by
> acl "domains". Falling back to Digest auth.
> 2011-02-11 13:27:39.838761 [NOTICE] switch_channel.c:669 New Channel
> sofia/internal/1000 at 184.105.153.247 [1df65ab4-0d7e-48e5-8c75-547ab51d73d7]
> 2011-02-11 13:27:39.838761 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/1000 at 184.105.153.247) Running State Change CS_NEW
> 2011-02-11 13:27:39.838761 [DEBUG] switch_core_state_machine.c:320
> (sofia/internal/1000 at 184.105.153.247) State NEW
> 2011-02-11 13:27:39.850754 [DEBUG] sofia.c:4153 Channel
> sofia/internal/1000 at 184.105.153.247 entering state [received][100]
> 2011-02-11 13:27:39.850754 [DEBUG] sofia.c:4164 Remote SDP:
> v=0
> o=- 7 2 IN IP4 61.235.150.11
> s=CounterPath eyeBeam 1.5
> c=IN IP4 61.235.150.11
> t=0 0
> m=audio 13410 RTP/AVP 0 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=alt:1 3 : xpehuJM9 ZEXWjImz 192.168.52.1 13410
> a=alt:2 2 : 9AzLyh+Z Xgci2e96 192.168.153.1 13410
> a=alt:3 1 : U0bonQcp 1CQ61b/S 61.235.150.11 13410
> a=x-rtp-session-id:2BC4463425494741B4A9DA4DFD436DDF
> 2011-02-11 13:27:39.850754 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
> [PCMU:0:8000:20]/[G7221:115:32000:20]
> 2011-02-11 13:27:39.850754 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
> [PCMU:0:8000:20]/[G7221:107:16000:20]
> 2011-02-11 13:27:39.850754 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
> [PCMU:0:8000:20]/[G722:9:8000:20]
> 2011-02-11 13:27:39.850754 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
> [PCMU:0:8000:20]/[PCMU:0:8000:20]
> 2011-02-11 13:27:39.850754 [DEBUG] sofia_glue.c:2354 Set Codec
> sofia/internal/1000 at 184.105.153.247 PCMU/8000 20 ms 160 samples
> 2011-02-11 13:27:39.850754 [DEBUG] sofia_glue.c:3524 Set 2833 dtmf send/recv
> payload to 101
> 2011-02-11 13:27:39.850754 [DEBUG] sofia.c:4310
> (sofia/internal/1000 at 184.105.153.247) State Change CS_NEW -> CS_INIT
> 2011-02-11 13:27:39.850754 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1000 at 184.105.153.247 [BREAK]
> 2011-02-11 13:27:39.854744 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/1000 at 184.105.153.247) Running State Change CS_INIT
> 2011-02-11 13:27:39.854744 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/1000 at 184.105.153.247) State INIT
> 2011-02-11 13:27:39.854744 [DEBUG] mod_sofia.c:83
> sofia/internal/1000 at 184.105.153.247 SOFIA INIT
> 2011-02-11 13:27:39.854744 [DEBUG] mod_sofia.c:117
> (sofia/internal/1000 at 184.105.153.247) State Change CS_INIT -> CS_ROUTING
> 2011-02-11 13:27:39.854744 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1000 at 184.105.153.247 [BREAK]
> 2011-02-11 13:27:39.854744 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/1000 at 184.105.153.247) State INIT going to sleep
> 2011-02-11 13:27:39.854744 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/1000 at 184.105.153.247) Running State Change CS_ROUTING
> 2011-02-11 13:27:39.854744 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/1000 at 184.105.153.247) State ROUTING
> 2011-02-11 13:27:39.854744 [DEBUG] mod_sofia.c:140
> sofia/internal/1000 at 184.105.153.247 SOFIA ROUTING
> 2011-02-11 13:27:39.854744 [DEBUG] switch_core_state_machine.c:77
> sofia/internal/1000 at 184.105.153.247 Standard ROUTING
> 2011-02-11 13:27:39.854744 [INFO] mod_dialplan_xml.c:418 Processing
> 1000->5018009993355 in context default
> Dialplan: sofia/internal/1000 at 184.105.153.247 parsing [default->gvoice_out]
> continue=false
> Dialplan: sofia/internal/1000 at 184.105.153.247 Regex (PASS) [gvoice_out]
> destination_number(5018009993355) =~ /^50(1\d{10})$/ break=on-false
> Dialplan: sofia/internal/1000 at 184.105.153.247 Action
> set(hangup_after_bridge=true)
> Dialplan: sofia/internal/1000 at 184.105.153.247 Action
> bridge(dingaling/gv1/+18009993355 at voice.google.com)
> 2011-02-11 13:27:39.854744 [DEBUG] switch_core_state_machine.c:119
> (sofia/internal/1000 at 184.105.153.247) State Change CS_ROUTING -> CS_EXECUTE
> 2011-02-11 13:27:39.854744 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1000 at 184.105.153.247 [BREAK]
> 2011-02-11 13:27:39.854744 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/1000 at 184.105.153.247) State ROUTING going to sleep
> 2011-02-11 13:27:39.854744 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/1000 at 184.105.153.247) Running State Change CS_EXECUTE
> 2011-02-11 13:27:39.854744 [DEBUG] switch_core_state_machine.c:348
> (sofia/internal/1000 at 184.105.153.247) State EXECUTE
> 2011-02-11 13:27:39.854744 [DEBUG] mod_sofia.c:226
> sofia/internal/1000 at 184.105.153.247 SOFIA EXECUTE
> 2011-02-11 13:27:39.854744 [DEBUG] switch_core_state_machine.c:157
> sofia/internal/1000 at 184.105.153.247 Standard EXECUTE
> EXECUTE sofia/internal/1000 at 184.105.153.247 set(hangup_after_bridge=true)
> 2011-02-11 13:27:39.854744 [DEBUG] mod_dptools.c:816
> sofia/internal/1000 at 184.105.153.247 SET [hangup_after_bridge]=[true]
> EXECUTE sofia/internal/1000 at 184.105.153.247
> bridge(dingaling/gv1/+18009993355 at voice.google.com)
> 2011-02-11 13:27:39.894782 [NOTICE] libdingaling.c:1307 SecSEND:
> -------------------------------------------------------------------------------
> <presence type="subscribe" to="+18009993355 at voice.google.com"
> from="husodesar at gmail.com/gtalkAC94A6BE">
> ? <status>Call Me!</status>
> </presence>
> 2011-02-11 13:27:39.894782 [NOTICE] libdingaling.c:1307 SecSEND:
> -------------------------------------------------------------------------------
> <presence xmlns="jabber:client" type="probe"
> to="+18009993355 at voice.google.com"
> from="husodesar at gmail.com/gtalkAC94A6BE"></presence>
> 2011-02-11 13:27:40.050810 [INFO] libdingaling.c:1305 SecRECV:
> -------------------------------------------------------------------------------
> <presence type="error" to="husodesar at gmail.com/gtalkAC94A6BE"
> from="+18009993355 at voice.google.com">
> ? <error code="501" type="cancel">
> ??? <feature-not-implemented
> xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"></feature-not-implemented>
> ? </error>
> </presence>
> 2011-02-11 13:27:44.595029 [NOTICE] libdingaling.c:1307 SecSEND:
> -------------------------------------------------------------------------------
> <presence type="subscribe" to="+18009993355 at voice.google.com"
> from="husodesar at gmail.com/gtalkAC94A6BE">
> ? <status>Call Me!</status>
> </presence>
> 2011-02-11 13:27:44.595029 [NOTICE] libdingaling.c:1307 SecSEND:
> -------------------------------------------------------------------------------
> <presence xmlns="jabber:client" type="probe"
> to="+18009993355 at voice.google.com"
> from="husodesar at gmail.com/gtalkAC94A6BE"></presence>
> 2011-02-11 13:27:44.751042 [INFO] libdingaling.c:1305 SecRECV:
> -------------------------------------------------------------------------------
> <presence type="error" to="husodesar at gmail.com/gtalkAC94A6BE"
> from="+18009993355 at voice.google.com">
> ? <error code="501" type="cancel">
> ??? <feature-not-implemented
> xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"></feature-not-implemented>
> ? </error>
> </presence>
> 2011-02-11 13:27:49.595349 [NOTICE] libdingaling.c:1307 SecSEND:
> -------------------------------------------------------------------------------
> <presence type="subscribe" to="+18009993355 at voice.google.com"
> from="husodesar at gmail.com/gtalkAC94A6BE">
> ? <status>Call Me!</status>
> </presence>
> 2011-02-11 13:27:49.595349 [NOTICE] libdingaling.c:1307 SecSEND:
> -------------------------------------------------------------------------------
> <presence xmlns="jabber:client" type="probe"
> to="+18009993355 at voice.google.com"
> from="husodesar at gmail.com/gtalkAC94A6BE"></presence>
> 2011-02-11 13:27:49.751408 [INFO] libdingaling.c:1305 SecRECV:
> -------------------------------------------------------------------------------
> <presence type="error" to="husodesar at gmail.com/gtalkAC94A6BE"
> from="+18009993355 at voice.google.com">
> ? <error code="501" type="cancel">
> ??? <feature-not-implemented
> xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"></feature-not-implemented>
> ? </error>
> </presence>
> 2011-02-11 13:27:54.595667 [NOTICE] libdingaling.c:1307 SecSEND:
> -------------------------------------------------------------------------------
> <presence type="subscribe" to="+18009993355 at voice.google.com"
> from="husodesar at gmail.com/gtalkAC94A6BE">
> ? <status>Call Me!</status>
> </presence>
> 2011-02-11 13:27:54.595667 [NOTICE] libdingaling.c:1307 SecSEND:
> -------------------------------------------------------------------------------
> <presence xmlns="jabber:client" type="probe"
> to="+18009993355 at voice.google.com"
> from="husodesar at gmail.com/gtalkAC94A6BE"></presence>
> 2011-02-11 13:27:54.751666 [INFO] libdingaling.c:1305 SecRECV:
> -------------------------------------------------------------------------------
> <presence type="error" to="husodesar at gmail.com/gtalkAC94A6BE"
> from="+18009993355 at voice.google.com">
> ? <error code="501" type="cancel">
> ??? <feature-not-implemented
> xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"></feature-not-implemented>
> ? </error>
> </presence>
> 2011-02-11 13:27:56.495776 [DEBUG] mod_dingaling.c:1715 Unknown Recipient!
> 2011-02-11 13:27:56.495776 [DEBUG] mod_dingaling.c:704 Terminate called from
> line 1716 state=CS_NEW
> 2011-02-11 13:27:56.495776 [NOTICE] mod_dingaling.c:715 Close Channel N/A
> [CS_NEW]
> 2011-02-11 13:27:56.495776 [DEBUG] switch_core_state_machine.c:428 ()
> Running State Change CS_DESTROY
> 2011-02-11 13:27:56.495776 [DEBUG] switch_core_state_machine.c:439 (N/A)
> State DESTROY
> 2011-02-11 13:27:56.495776 [DEBUG] switch_core_state_machine.c:439 (N/A)
> State DESTROY going to sleep
> 2011-02-11 13:27:56.495776 [ERR] switch_ivr_originate.c:2430 Cannot create
> outgoing channel of type [dingaling] cause: [NO_USER_RESPONSE]
> 2011-02-11 13:27:56.495776 [DEBUG] switch_ivr_originate.c:3228 Originate
> Resulted in Error Cause: 18 [NO_USER_RESPONSE]
> 2011-02-11 13:27:56.495776 [INFO] mod_dptools.c:2355 Originate Failed.
> Cause: NO_USER_RESPONSE
> 2011-02-11 13:27:56.495776 [NOTICE] switch_core_state_machine.c:185
> sofia/internal/1000 at 184.105.153.247 has executed the last dialplan
> instruction, hanging up.
> 2011-02-11 13:27:56.495776 [NOTICE] switch_core_state_machine.c:187 Hangup
> sofia/internal/1000 at 184.105.153.247 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-02-11 13:27:56.495776 [DEBUG] switch_channel.c:2102 Send signal
> sofia/internal/1000 at 184.105.153.247 [KILL]
> 2011-02-11 13:27:56.495776 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1000 at 184.105.153.247 [BREAK]
> 2011-02-11 13:27:56.495776 [DEBUG] switch_core_state_machine.c:348
> (sofia/internal/1000 at 184.105.153.247) State EXECUTE going to sleep
> 2011-02-11 13:27:56.495776 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/1000 at 184.105.153.247) Running State Change CS_HANGUP
> 2011-02-11 13:27:56.495776 [DEBUG] switch_core_state_machine.c:499
> (sofia/internal/1000 at 184.105.153.247) State HANGUP
> 2011-02-11 13:27:56.495776 [DEBUG] mod_sofia.c:414 Channel
> sofia/internal/1000 at 184.105.153.247 hanging up, cause: NORMAL_CLEARING
> 2011-02-11 13:27:56.511787 [DEBUG] mod_sofia.c:476 Responding to INVITE
> with: 480
> 2011-02-11 13:27:56.511787 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/1000 at 184.105.153.247 Standard HANGUP, cause: NORMAL_CLEARING
> 2011-02-11 13:27:56.511787 [DEBUG] switch_core_state_machine.c:499
> (sofia/internal/1000 at 184.105.153.247) State HANGUP going to sleep
> 2011-02-11 13:27:56.511787 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/1000 at 184.105.153.247) State Change CS_HANGUP -> CS_REPORTING
> 2011-02-11 13:27:56.511787 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1000 at 184.105.153.247 [BREAK]
> 2011-02-11 13:27:56.511787 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/1000 at 184.105.153.247) Running State Change CS_REPORTING
> 2011-02-11 13:27:56.511787 [DEBUG] switch_core_state_machine.c:590
> (sofia/internal/1000 at 184.105.153.247) State REPORTING
> 2011-02-11 13:27:56.511787 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/1000 at 184.105.153.247 Standard REPORTING, cause:
> NORMAL_CLEARING
> 2011-02-11 13:27:56.511787 [DEBUG] switch_core_state_machine.c:590
> (sofia/internal/1000 at 184.105.153.247) State REPORTING going to sleep
> 2011-02-11 13:27:56.511787 [DEBUG] switch_core_state_machine.c:327
> (sofia/internal/1000 at 184.105.153.247) State Change CS_REPORTING ->
> CS_DESTROY
> 2011-02-11 13:27:56.511787 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1000 at 184.105.153.247 [BREAK]
> 2011-02-11 13:27:56.511787 [DEBUG] switch_core_session.c:1164 Session 12
> (sofia/internal/1000 at 184.105.153.247) Locked, Waiting on external entities
> 2011-02-11 13:27:56.511787 [NOTICE] switch_core_session.c:1182 Session 12
> (sofia/internal/1000 at 184.105.153.247) Ended
> 2011-02-11 13:27:56.511787 [NOTICE] switch_core_session.c:1184 Close Channel
> sofia/internal/1000 at 184.105.153.247 [CS_DESTROY]
> 2011-02-11 13:27:56.511787 [DEBUG] switch_core_state_machine.c:428
> (sofia/internal/1000 at 184.105.153.247) Running State Change CS_DESTROY
> 2011-02-11 13:27:56.511787 [DEBUG] switch_core_state_machine.c:439
> (sofia/internal/1000 at 184.105.153.247) State DESTROY
> 2011-02-11 13:27:56.511787 [DEBUG] mod_sofia.c:341
> sofia/internal/1000 at 184.105.153.247 SOFIA DESTROY
> 2011-02-11 13:27:56.511787 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/1000 at 184.105.153.247 Standard DESTROY
> 2011-02-11 13:27:56.511787 [DEBUG] switch_core_state_machine.c:439
> (sofia/internal/1000 at 184.105.153.247) State DESTROY going to sleep
>
> What's wrong with this?
>
> Please help me, 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 peter.olsson at visionutveckling.se  Fri Feb 11 09:28:30 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 11 Feb 2011 07:28:30 +0100
Subject: [Freeswitch-users] FreeTDM + libsng_isdn in Windows - can't get it
 to work (causes FS crasch)
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58ADB72787@cooper>

I'm trying to get FreeTDM and libsng_isdn working in a Windows environment, but with no success. This is what I've been trying to do;

First of all - I'm using lastest git HEAD (as of right now), latest Sangoma driver (6.0.38.0), latest libsng_isdn (7.1.0), I build everything in Visual Studio 2008.

At first I got lots of side-by-side errors (VC runtime errors) when loading mod_freetdm. I found that the reason for this was ftmod_sangoma_isdn.dll was compiled with _DEBUG, even in release mode (which is what I'm using), I changed this to NDEBUG instead, and that got rid of the error (I guess this should be updated in the project file though).

After this I got this error (even though the libsng_isdn was installed on the computer), the reason for this seems to be it can't find libsng_isdn.dll;
2011-02-11 07:02:32.591375 [ERR] ftdm_io.c:4989 Error loading C:\freeswitch\mod\ftmod_sangoma_isdn.dll [dll open error [126l]

Then I simply copied libsng_isdn.dll into the freeswitch folder, and after this FS crasches immediately when trying to load mod_freetdm.

So I guess by now it finds all files and dependencies, but instead it causes a crasch. Did I do something wrong, or is there something in the code that needs to be updated to get things working in Windows? This is just on a lab system, so I'm able to do whatever tests you want me to.

Thanks,

Peter Olsson


From gabe at gundy.org  Fri Feb 11 11:19:47 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 11 Feb 2011 01:19:47 -0700
Subject: [Freeswitch-users] Yet another DTMF question
In-Reply-To: <AANLkTi=5TEUNnycLKyY_5hh7eZQTAGB34xC6axYjnddO@mail.gmail.com>
References: <AANLkTik030ecFw2bvKe74pOs+NzfqHknX8keDO3mQ=cH@mail.gmail.com>
	<AANLkTikqQodjsMT_VB1xeQdtY-2DEgL68ndEnHTBTvkb@mail.gmail.com>
	<AANLkTiksK3T24TUDO-7aq_tbkFCkj_4BRNpLRt4=GtBj@mail.gmail.com>
	<62C5671A-8657-4F99-8184-272EEBEC8E77@gmail.com>
	<AANLkTimFMwJ1EkP_+1uCyiKUCKocgh-mSYVEd39navS7@mail.gmail.com>
	<AANLkTikaerx0b+A7EhFLXxuBKBBY+RBog8Lpgyd1ZZmR@mail.gmail.com>
	<AANLkTinL1NxOjMJjShOwXEo5CJERLbY4N7Z7re=QRp8z@mail.gmail.com>
	<AANLkTi=Trve2-zDb60FEahz2TLD_=BAvhzisf7gSOoFo@mail.gmail.com>
	<AANLkTi=d_Bzmt+5s+RaGDH6KckBszmzJjsTq2G2JvRZP@mail.gmail.com>
	<AANLkTi=5TEUNnycLKyY_5hh7eZQTAGB34xC6axYjnddO@mail.gmail.com>
Message-ID: <AANLkTim4a13Ww3ReOdSHwFvwZ71NB+a3aCMdoJ2A2LU6@mail.gmail.com>

Thanks for the detailed reply.  I guess I'll have to look elsewhere
for the source of my problems :(

Gabe

On Thu, Feb 10, 2011 at 1:53 AM, Steven Ayre <steveayre at gmail.com> wrote:
> Introduced on 12th Jan (commit fe1711fd)
> Fixed on 13th Jan (commit b2359797)
>
> Affected commits:
> ? fe1711fd
> ? 9c7b507d
> ? a6db66ef
> ? d9c56345
> ? 8458adeb
> ? 2e074727
> ? c6bdb303
>
> -Steve
>
>
> On 10 February 2011 06:23, Gabriel Gunderson <gabe at gundy.org> wrote:
>>
>> On Wed, Jan 19, 2011 at 10:32 AM, Steven Ayre <steveayre at gmail.com> wrote:
>> > Try uploading to the latest Git, a post from Brian just reminded me that
>> > there was a day where some changes to the RTP stack broke RFC2833
>> > support,
>> > perhaps you have one of the bad versions.
>>
>> Anyone know what day this might have been? ?I WILL update, but it
>> would be good to know if my issues could also be related.
>>
>> Thanks,
>> Gabe


From steveayre at gmail.com  Fri Feb 11 12:06:02 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 11 Feb 2011 09:06:02 +0000
Subject: [Freeswitch-users] double instance at relaodxml
In-Reply-To: <74344B58584B400B9D0AFC74311657FF@e1705>
References: <74344B58584B400B9D0AFC74311657FF@e1705>
Message-ID: <AANLkTinmfX1+YGpcYW4FzX+Tsj4SRXSy7bmTdeWboQW4@mail.gmail.com>

reloadxml can't start a new FS process. It only reloads the config files for
the FS process you run it in.

The 2nd copy must be being started by something else.

-Steve



On 11 February 2011 03:45, Madovsky <infos at madovsky.org> wrote:

>  sometimes when I run from fs_cli "reloadxml"
> I notice 2 instances of FS so the last instance
> log mod_event socket error , could not listen to socket....
>
> 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  Fri Feb 11 12:07:40 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 11 Feb 2011 09:07:40 +0000
Subject: [Freeswitch-users] Can XML CDR Timestamps Be UTC?
In-Reply-To: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F750EB4@VA3DIAXVS351.RED001.local>
References: <AcvJdvK0FsPB92LXSuymyzLbDIcGSg==>
	<2BF7FB90DF25EA4485949F3AF2B9D6960B3F750EB4@VA3DIAXVS351.RED001.local>
Message-ID: <AANLkTimy4fw7_+n+Z4ej4BGbwU7F9qdaaasE9MBezQPr@mail.gmail.com>

Use the uepoch times and convert them into the representation you want -
they're all in UTC.

-Steve


On 10 February 2011 23:05, Jerry Richards <jerry.richards at teotech.com>wrote:

> Hello All,
>
>
>
> Is there a way to configure Freeswitch to indicate UTC (instead of local
> time) in the XML CDR files?
>
>
>
> Thanks,
>
> Jerry
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 11 13:43:25 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Fri, 11 Feb 2011 13:43:25 +0300
Subject: [Freeswitch-users] hold_music var gets unset/lost by
 cancelled/failed att_xfer
In-Reply-To: <AANLkTinS3wxA8izr9tXX7jhYJPHq_X_coO7UNoUkjs-P@mail.gmail.com>
References: <AANLkTimLod3yHrex4LAK=4gdVcPhzXutKYyw=E8ZMcN9@mail.gmail.com>
	<AANLkTi=u3VeHPuS5x1e+yu5QNBw5ZSgHvAWOT+F6A_OR@mail.gmail.com>
	<AANLkTi=EndkiUJvCgCSirxCA23EwWD_++0h3Qcr=mg9g@mail.gmail.com>
	<AANLkTikt97dugbhKPgnB2ws4bOFcxg8yhzSdyxHMab_m@mail.gmail.com>
	<AANLkTikEx9ZwNADRQ8k2JbTwgoLPFmHUPpPL3RLnZZU8@mail.gmail.com>
	<AANLkTinS3wxA8izr9tXX7jhYJPHq_X_coO7UNoUkjs-P@mail.gmail.com>
Message-ID: <AANLkTi=Hv7kkYeD8TQeCZtfvmzqaCx+xQ=yYC16kpNfa@mail.gmail.com>

I see your patch in git log, is it enough to do 'make current', or I
should apply it in some special way? After make current issues are
still present...

What should i set loopback_bowout/bowout_on_execute variables to? I
tried with false, this works as in previous versions, and loopback
stays in path during the call. While loopback channels are present in
path, there are issues with sound quality, there is robot voice effect
for short periods of time. You mentioned setting rtp timer to none,
where this should be done? In loopback channel execute_extension while
doing att_xfer?

When bowout variables set to true  att_xfer doesn't work just as I
described in my previous posts, and there is also issue with MOH -
after transfer target answers to transferor, MOH is stoped for
transferee. Maybe this can help somehow.




2011/2/10 Anthony Minessale <anthony.minessale at gmail.com>:
> I pushed a patch that will probably delay the bowout until after the
> att_xfer is over
> Give it a try.
>
> commit 3546654615f88058fb6769fe79e07162602fa4af
> Author: Anthony Minessale <anthm at freeswitch.org>
> Date: ? Thu Feb 10 12:37:14 2011 -0600
>
> ? ?don't bow out on att_xfer bridge
>
>
> On Thu, Feb 10, 2011 at 1:03 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>> Thanks! It works now! BTW, att_xfer with
>> loopback_bowout/bowout_on_execute set to false seems to be working
>> too. Is there a way to make loopback channel leave the path? Without
>> bowout turned off att_xfer doesn't work...
>>
>> 2011/2/8 Anthony Minessale <anthony.minessale at gmail.com>:
>>> try latest GIT
>>>
>>> On Tue, Feb 8, 2011 at 1:28 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>>> I have same issue with MOH and att_xfer on failed transfers, music on
>>>> hold played only once
>>>> At the same time, transfer_ringback always plays correctly to transferer
>>>>
>>>> 2011/2/8 Anthony Minessale <anthony.minessale at gmail.com>:
>>>>> you really should report this to jira not to the mailing list.
>>>>> http://jira.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> On Mon, Feb 7, 2011 at 4:23 PM, Fraser Redmond <fraserredmond at gmail.com> wrote:
>>>>>> I'm trying to do a second att_xfer on a call so that if the first attended
>>>>>> transfer fails (c-leg is busy, or presses do-not-answer, or is an extn that
>>>>>> doesn't exist) then the call could be transferred to someone else.
>>>>>>
>>>>>> On the first att_xfer the person on hold hears the hold_music correctly.
>>>>>> Once that transfer is cancelled or fails:
>>>>>> -- On any subsequent att_xfer's the person on hold just hears silence.
>>>>>> -- If they are put on hold they just hear silence.
>>>>>>
>>>>>> I tried setting hold_music again for each channel after the first att_xfer,
>>>>>> but that didn't work, so it's probably not actually a problem with
>>>>>> hold_music per se, but some other variable/setting that decides whether to
>>>>>> use hold_music.
>>>>>>
>>>>>> I also tried doing a uuid_dump before and after each attempt, but didn't
>>>>>> notice anything too different - unless it's a matter of unsetting one of the
>>>>>> couple of changed/new vars like:
>>>>>> variable_originate_disposition
>>>>>> variable_current_application
>>>>>> variable_playback_seconds
>>>>>>
>>>>>> I get the feeling other variables are probably also lost by the first failed
>>>>>> transfer as the second att_xfer has some odd things happen if the third
>>>>>> party does answer. Haven't been able to narrow it down as closely as the
>>>>>> hold_music, but two things I've seen happen are:
>>>>>> -- The party that initiated the transfer gets hung up automatically (after
>>>>>> 30 sec)
>>>>>> -- When the party that initiated the transfer hangs up it should connect the
>>>>>> other two parties, but instead it hung up all three
>>>>>>
>>>>>> Cheers,
>>>>>> Fraser
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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


From mattdfong at gmail.com  Fri Feb 11 14:06:09 2011
From: mattdfong at gmail.com (Matthew Fong)
Date: Fri, 11 Feb 2011 03:06:09 -0800
Subject: [Freeswitch-users] Using an Empty FROM field for SIP Messages
In-Reply-To: <D810E0FC-0474-45E8-9C9A-C8D9D438032C@ipeva.fr>
References: <AANLkTikEyL=NpmdqM+Vx=rxaM2+zEyxuJr_oGdFDWLwU@mail.gmail.com>
	<D810E0FC-0474-45E8-9C9A-C8D9D438032C@ipeva.fr>
Message-ID: <AANLkTikbncOB_=Q4hEc2gg8A+bS59CN0=RXjUdAdT0az@mail.gmail.com>

effective_caller_id_number doesn't seem to work. I was able to get

From: "" <sip:@0.0.0.0>;tag=j0yN36H8tZgDe

using sip_from_uri='<sip:@X.X.X.X>' but there are still the proceeding ""

Does anyone else have any suggestions?

Thanks
--matt
http://www.hellohunter.com
Hosted Dialer
On Wed, Feb 9, 2011 at 2:06 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Try setting effective_caller_id_number to " "
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 09/02/2011 ? 10:55, Matthew Fong a ?crit :
>
> I am trying to setup a customer who wanted to utilize the airespring
> national did presence, where in they rewrite the Caller ID of outbound
> calls. In order to do this tho, they say I need to send a blank caller id.
> FreeSWITCH will send 0000000000 if the caller id is not specified.
> Airespring wants something formatted like
>
> INVITE sip:jungar at alpha-org.com SIP/2.0
>
> Via: SIP/2.0/UDP zetamachine.beta-org.com:5060;branch=as82je8ei4kr
>
> Max-Forwards: 70
>
> To:  2556112121 <sip:jungar at alpha-org.com>
>
> From: <sip:jsmith at beta-org.com>;tag=13456
>
> Call-ID: @zetamachine.beta-org.com
>
>
> with the From: field looking like any of the following
>
> From: <sip:jsmith at beta-org.com>;tag=13456
> From: <  @208.76.54.59>;tag=13456
> From:  @208.76.54.59;tag=13456
>
> The closest I got was to make FreeSWITCH to the following, but airespring
> won't budge. Can anyone tell me how to make the From string as airespring
> wants for their national DID presence service. Thanks.
> From: "" <sip:0000000000 at 208.76.54.59>;tag=UyNBXjyUUay8m
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From br at bsdpad.com  Fri Feb 11 14:24:53 2011
From: br at bsdpad.com (Ruslan Bukin)
Date: Fri, 11 Feb 2011 14:24:53 +0300
Subject: [Freeswitch-users] dp_tools/set- not worked SOMETIMES
Message-ID: <20110211112453.GA700@bsdjail.com>

Application set is not set parameters SOMETIMES

for example, before bridge I set my static parameters:

<action application="set" data="my_own_parameter=test"/>
<action application="set" data="nibble_rate=1"/>
...

sometimes (in about ~1-5% times) one of them is not set up.

I tried to set like this before bridge:

 <extension name="set_var" continue="true">
   <condition field="${my_own_parameter}" expression=".+" break="on-true"/>
   <condition field="destination_number" expression="(777)" break="on-true">
     <action application="set" data="my_own_parameter=test"/>
     <action application="transfer" data="$1 XML public"/>
   </condition>
 </extension>

but no result: after bridge no parameter to read (SOMETIMES)


any recommendations?

-Ruslan



From david.ponzone at ipeva.fr  Fri Feb 11 14:37:02 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 11 Feb 2011 12:37:02 +0100
Subject: [Freeswitch-users] Using an Empty FROM field for SIP Messages
In-Reply-To: <AANLkTikbncOB_=Q4hEc2gg8A+bS59CN0=RXjUdAdT0az@mail.gmail.com>
References: <AANLkTikEyL=NpmdqM+Vx=rxaM2+zEyxuJr_oGdFDWLwU@mail.gmail.com>
	<D810E0FC-0474-45E8-9C9A-C8D9D438032C@ipeva.fr>
	<AANLkTikbncOB_=Q4hEc2gg8A+bS59CN0=RXjUdAdT0az@mail.gmail.com>
Message-ID: <6CE2C3C3-0EAF-43F5-B835-5A954F194ED3@ipeva.fr>

Matthew,

I think I forgot to tell you something.
If you are sending the call to an outbound gateway, you need to add this in the gateway parameter:
 <param name="caller-id-in-from" value="true"/>

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 11/02/2011 ? 12:06, Matthew Fong a ?crit :

> effective_caller_id_number doesn't seem to work. I was able to get
> 
> From: "" <sip:@0.0.0.0>;tag=j0yN36H8tZgDe
> 
> using sip_from_uri='<sip:@X.X.X.X>' but there are still the proceeding ""
> 
> Does anyone else have any suggestions?
> 
> Thanks
> --matt
> http://www.hellohunter.com
> Hosted Dialer
> On Wed, Feb 9, 2011 at 2:06 AM, David Ponzone <david.ponzone at ipeva.fr> wrote:
> Try setting effective_caller_id_number to " "
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 09/02/2011 ? 10:55, Matthew Fong a ?crit :
> 
>> I am trying to setup a customer who wanted to utilize the airespring national did presence, where in they rewrite the Caller ID of outbound calls. In order to do this tho, they say I need to send a blank caller id. FreeSWITCH will send 0000000000 if the caller id is not specified. Airespring wants something formatted like
>> 
>> INVITE sip:jungar at alpha-org.com SIP/2.0
>> Via: SIP/2.0/UDP zetamachine.beta-org.com:5060;branch=as82je8ei4kr
>> Max-Forwards: 70
>> To:  2556112121 <sip:jungar at alpha-org.com>
>> From: <sip:jsmith at beta-org.com>;tag=13456
>> Call-ID: @zetamachine.beta-org.com
>> 
>> 
>> with the From: field looking like any of the following
>> 
>> From: <sip:jsmith at beta-org.com>;tag=13456
>> From: <  @208.76.54.59>;tag=13456
>> From:  @208.76.54.59;tag=13456
>> 
>> The closest I got was to make FreeSWITCH to the following, but airespring won't budge. Can anyone tell me how to make the From string as airespring wants for their national DID presence service. Thanks.
>> From: "" <sip:0000000000 at 208.76.54.59>;tag=UyNBXjyUUay8m
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From br at bsdpad.com  Fri Feb 11 14:40:04 2011
From: br at bsdpad.com (Ruslan Bukin)
Date: Fri, 11 Feb 2011 14:40:04 +0300
Subject: [Freeswitch-users] dp_tools/set- not worked SOMETIMES
In-Reply-To: <20110211112453.GA700@bsdjail.com>
References: <20110211112453.GA700@bsdjail.com>
Message-ID: <20110211114004.GA1491@bsdjail.com>

correction: it seems that parameters always exists after bridge,
but sometimes one of them not exists in CDR result 
(I have checked with mod_xml_cdr and mod_json_cdr)

-Ruslan

On Fri, Feb 11, 2011 at 02:24:53PM +0300, Ruslan Bukin wrote:
> Application set is not set parameters SOMETIMES
> 
> for example, before bridge I set my static parameters:
> 
> <action application="set" data="my_own_parameter=test"/>
> <action application="set" data="nibble_rate=1"/>
> ...
> 
> sometimes (in about ~1-5% times) one of them is not set up.
> 
> I tried to set like this before bridge:
> 
>  <extension name="set_var" continue="true">
>    <condition field="${my_own_parameter}" expression=".+" break="on-true"/>
>    <condition field="destination_number" expression="(777)" break="on-true">
>      <action application="set" data="my_own_parameter=test"/>
>      <action application="transfer" data="$1 XML public"/>
>    </condition>
>  </extension>
> 
> but no result: after bridge no parameter to read (SOMETIMES)
> 
> 
> any recommendations?
> 
> -Ruslan
> 


From chenzhanping at gmail.com  Fri Feb 11 14:43:35 2011
From: chenzhanping at gmail.com (=?GB2312?B?1bzGvbPC?=)
Date: Fri, 11 Feb 2011 19:43:35 +0800
Subject: [Freeswitch-users] Question about mod_dingaling.
In-Reply-To: <AANLkTimXmNou2T4tLPMACscdHiHVqUzf3U__WeGVaEJb@mail.gmail.com>
References: <AANLkTikd9baOGf0bmL6m3EaiPcGy6Fgfm9=jEYBg+GB2@mail.gmail.com>
	<AANLkTimXmNou2T4tLPMACscdHiHVqUzf3U__WeGVaEJb@mail.gmail.com>
Message-ID: <AANLkTi=hu9iWu3SQbF5Wj3odkY0ut9HGn0mabsBZJMbQ@mail.gmail.com>

I try to install *freeswitch 1.0.7* download from latest.freeswitch.org,but can
not call out from mod_dingaling use google voice always.

this is a part of cli log:


2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:865 Set Read Codec to
PCMU at 8000
2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:880 Set Write Codec to
PCMU at 8000
2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:892 SETUP RTP
184.105.153.247:0 -> 74.125.127.126:19295
2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:914* RTP ERROR Missing
local port
*2011-02-11 19:37:21.546355 [DEBUG] switch_channel.c:2538 (
dingaling/gv1/+18009993355 at voice.google.com) Callstate Change DOWN -> HANGUP
2011-02-11 19:37:21.546355 [NOTICE] mod_dingaling.c:915 Hangup
dingaling/gv1/+18009993355 at voice.google.com [CS_INIT] [*
DESTINATION_OUT_OF_ORDER*]
2011-02-11 19:37:21.546355 [DEBUG] switch_channel.c:2554 Send signal
dingaling/gv1/+18009993355 at voice.google.com [KILL]
2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:1348
dingaling/gv1/+18009993355 at voice.google.com CHANNEL KILL
2011-02-11 19:37:21.546355 [DEBUG] switch_core_session.c:1116 Send signal
dingaling/gv1/+18009993355 at voice.google.com [BREAK]
2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:1348
dingaling/gv1/+18009993355 at voice.google.com CHANNEL KILL
2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:710 Terminate called from
line 1177 state=CS_HANGUP
2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:356 (
dingaling/gv1/+18009993355 at voice.google.com) State INIT going to sleep
2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:320 (
dingaling/gv1/+18009993355 at voice.google.com) Running State Change CS_HANGUP
2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:557 (
dingaling/gv1/+18009993355 at voice.google.com) State HANGUP
2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:1317
dingaling/gv1/+18009993355 at voice.google.com CHANNEL HANGUP
2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:46
dingaling/gv1/+18009993355 at voice.google.com *Standard HANGUP, cause:
DESTINATION_OUT_OF_ORDER
*2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:557 (
dingaling/gv1/+18009993355 at voice.google.com) State HANGUP going to sleep
2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:351 (
dingaling/gv1/+18009993355 at voice.google.com) State Change CS_HANGUP ->
CS_REPORTING
2011-02-11 19:37:21.546355 [DEBUG] switch_core_session.c:1116 Send signal
dingaling/gv1/+18009993355 at voice.google.com [BREAK]
2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:1348
dingaling/gv1/+18009993355 at voice.google.com CHANNEL KILL
2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:320 (
dingaling/gv1/+18009993355 at voice.google.com) Running State Change
CS_REPORTING
2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:617 (
dingaling/gv1/+18009993355 at voice.google.com) State REPORTING
2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:53
dingaling/gv1/+18009993355 at voice.google.com Standard REPORTING, cause:
DESTINATION_OUT_OF_ORDER
2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:617 (
dingaling/gv1/+18009993355 at voice.google.com) State REPORTING going to sleep
2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:345 (
dingaling/gv1/+18009993355 at voice.google.com) State Change CS_REPORTING ->
CS_DESTROY
2011-02-11 19:37:21.546355 [DEBUG] switch_core_session.c:1116 Send signal
dingaling/gv1/+18009993355 at voice.google.com [BREAK]
2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:1348
dingaling/gv1/+18009993355 at voice.google.com CHANNEL KILL
2011-02-11 19:37:21.546355 [DEBUG] switch_core_session.c:1288 Session 12 (
dingaling/gv1/+18009993355 at voice.google.com) Locked, Waiting on external
entities
2011-02-11 19:37:21.550059 [DEBUG] switch_ivr_originate.c:3502 Originate
Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2011-02-11 19:37:21.550059 [INFO] mod_dptools.c:2623 Originate Failed.
Cause: DESTINATION_OUT_OF_ORDER
2011-02-11 19:37:21.550059 [DEBUG] switch_channel.c:2538 (
sofia/internal/1000 at 184.105.153.247) Callstate Change RINGING -> HANGUP
2011-02-11 19:37:21.550059 [NOTICE] mod_dptools.c:2686 Hangup
sofia/internal/1000 at 184.105.153.247 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
2011-02-11 19:37:21.550059 [DEBUG] switch_channel.c:2554 Send signal
sofia/internal/1000 at 184.105.153.247 [KILL]
2011-02-11 19:37:21.550059 [DEBUG] switch_core_session.c:1116 Send signal
sofia/internal/1000 at 184.105.153.247 [BREAK]

*all log is here:*


freeswitch at yijiaren.net.cn> 2011-02-11 19:37:17.481827 [DEBUG] sofia.c:6427
IP 121.28.15.8 Rejected by acl "domains". Falling back to Digest auth.
2011-02-11 19:37:17.485846 [WARNING] sofia_reg.c:1247 SIP auth challenge
(INVITE) on sofia profile 'internal' for [5018009993355 at 184.105.153.247]
from ip 121.28.15.8
2011-02-11 19:37:17.785818 [DEBUG] sofia.c:6427 IP 121.28.15.8 Rejected by
acl "domains". Falling back to Digest auth.
2011-02-11 19:37:17.789834 [NOTICE] switch_channel.c:811 New Channel
sofia/internal/1000 at 184.105.153.247 [4716f8ad-de2d-4de6-9848-834add30ff1f]
2011-02-11 19:37:17.789834 [DEBUG] switch_core_state_machine.c:320 (
sofia/internal/1000 at 184.105.153.247) Running State Change CS_NEW
2011-02-11 19:37:17.789834 [DEBUG] switch_core_state_machine.c:338 (
sofia/internal/1000 at 184.105.153.247) State NEW
2011-02-11 19:37:17.830152 [DEBUG] sofia.c:4659 Channel
sofia/internal/1000 at 184.105.153.247 entering state [received][100]
2011-02-11 19:37:17.830152 [DEBUG] sofia.c:4670 Remote SDP:
v=0
o=- 8 2 IN IP4 61.235.150.11
s=CounterPath eyeBeam 1.5
c=IN IP4 61.235.150.11
t=0 0
m=audio 6816 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 3 : H+ZKvDCR 5iYo6mbN 192.168.52.1 6816
a=alt:2 2 : yZueEnyN QCAU4mGC 192.168.153.1 6816
a=alt:3 1 : g4e9fZKB 6/CgnDNR 61.235.150.11 6816
a=x-rtp-session-id:41F847609F534152A0EDFEBE5D1D93FC
2011-02-11 19:37:17.830152 [DEBUG] sofia_glue.c:4474 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000]
2011-02-11 19:37:17.830152 [DEBUG] sofia_glue.c:4474 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000]
2011-02-11 19:37:17.830152 [DEBUG] sofia_glue.c:4474 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-02-11 19:37:17.830152 [DEBUG] sofia_glue.c:2757 Set Codec
sofia/internal/1000 at 184.105.153.247 PCMU/8000 20 ms 160 samples 64000 bits
2011-02-11 19:37:17.830152 [DEBUG] sofia_glue.c:4578 Set 2833 dtmf send/recv
payload to 101
2011-02-11 19:37:17.830152 [DEBUG] sofia.c:4837 (
sofia/internal/1000 at 184.105.153.247) State Change CS_NEW -> CS_INIT
2011-02-11 19:37:17.830152 [DEBUG] switch_core_session.c:1116 Send signal
sofia/internal/1000 at 184.105.153.247 [BREAK]
2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:320 (
sofia/internal/1000 at 184.105.153.247) Running State Change CS_INIT
2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:356 (
sofia/internal/1000 at 184.105.153.247) State INIT
2011-02-11 19:37:17.833837 [DEBUG] mod_sofia.c:84
sofia/internal/1000 at 184.105.153.247 SOFIA INIT
2011-02-11 19:37:17.833837 [DEBUG] mod_sofia.c:124 (
sofia/internal/1000 at 184.105.153.247) State Change CS_INIT -> CS_ROUTING
2011-02-11 19:37:17.833837 [DEBUG] switch_core_session.c:1116 Send signal
sofia/internal/1000 at 184.105.153.247 [BREAK]
2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:356 (
sofia/internal/1000 at 184.105.153.247) State INIT going to sleep
2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:320 (
sofia/internal/1000 at 184.105.153.247) Running State Change CS_ROUTING
2011-02-11 19:37:17.833837 [DEBUG] switch_channel.c:1660 (
sofia/internal/1000 at 184.105.153.247) Callstate Change DOWN -> RINGING
2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:359 (
sofia/internal/1000 at 184.105.153.247) State ROUTING
2011-02-11 19:37:17.833837 [DEBUG] mod_sofia.c:147
sofia/internal/1000 at 184.105.153.247 SOFIA ROUTING
2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:77
sofia/internal/1000 at 184.105.153.247 Standard ROUTING
2011-02-11 19:37:17.833837 [INFO] mod_dialplan_xml.c:331 Processing 1000
<1000>->5018009993355 in context default
Dialplan: sofia/internal/1000 at 184.105.153.247 parsing [default->gvoice_out]
continue=false
Dialplan: sofia/internal/1000 at 184.105.153.247 Regex (PASS) [gvoice_out]
destination_number(5018009993355) =~ /^50(1\d{10})$/ break=on-false
Dialplan: sofia/internal/1000 at 184.105.153.247 Action
set(hangup_after_bridge=true)
Dialplan: sofia/internal/1000 at 184.105.153.247 Action bridge(
dingaling/gv1/+18009993355 at voice.google.com)
2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:119 (
sofia/internal/1000 at 184.105.153.247) State Change CS_ROUTING -> CS_EXECUTE
2011-02-11 19:37:17.833837 [DEBUG] switch_core_session.c:1116 Send signal
sofia/internal/1000 at 184.105.153.247 [BREAK]
2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:359 (
sofia/internal/1000 at 184.105.153.247) State ROUTING going to sleep
2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:320 (
sofia/internal/1000 at 184.105.153.247) Running State Change CS_EXECUTE
2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:366 (
sofia/internal/1000 at 184.105.153.247) State EXECUTE
2011-02-11 19:37:17.833837 [DEBUG] mod_sofia.c:240
sofia/internal/1000 at 184.105.153.247 SOFIA EXECUTE
2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:157
sofia/internal/1000 at 184.105.153.247 Standard EXECUTE
EXECUTE sofia/internal/1000 at 184.105.153.247 set(hangup_after_bridge=true)
2011-02-11 19:37:17.833837 [DEBUG] mod_dptools.c:1059
sofia/internal/1000 at 184.105.153.247 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/1000 at 184.105.153.247 bridge(
dingaling/gv1/+18009993355 at voice.google.com)
2011-02-11 19:37:17.833837 [NOTICE] switch_channel.c:811 New Channel
dingaling/gv1/+18009993355 at voice.google.com[c024cca7-4b61-475f-a04a-023e808cb41f]
2011-02-11 19:37:17.833837 [DEBUG] libdingaling.c:355 Created Session
1162869130
2011-02-11 19:37:17.833837 [DEBUG] mod_dingaling.c:1824 (
dingaling/gv1/+18009993355 at voice.google.com) State Change CS_NEW -> CS_INIT
2011-02-11 19:37:17.833837 [DEBUG] switch_core_session.c:1116 Send signal
dingaling/gv1/+18009993355 at voice.google.com [BREAK]
2011-02-11 19:37:17.833837 [DEBUG] mod_dingaling.c:1348
dingaling/gv1/+18009993355 at voice.google.com CHANNEL KILL
2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:320 (
dingaling/gv1/+18009993355 at voice.google.com) Running State Change CS_INIT
2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:356 (
dingaling/gv1/+18009993355 at voice.google.com) State INIT
2011-02-11 19:37:17.833837 [NOTICE] mod_dingaling.c:1110 Ring-Ready
dingaling/gv1/+18009993355 at voice.google.com!
2011-02-11 19:37:17.833837 [DEBUG] mod_dingaling.c:1063 Don't have my codec
yet here's one
2011-02-11 19:37:17.833837 [DEBUG] mod_dingaling.c:1083 Send Describe
[PCMU at 8000]
2011-02-11 19:37:17.937903 [DEBUG] libdingaling.c:1450 Processing 1 packets
in retry queue
2011-02-11 19:37:17.937903 [DEBUG] libdingaling.c:1467 Sending packet 315 (2
left)
2011-02-11 19:37:17.937903 [NOTICE] libdingaling.c:1368 SecSEND:
-------------------------------------------------------------------------------
<iq xmlns="jabber:client" from="husodesar at gmail.com/gtalk94609708" to="
+18009993355 at voice.google.com" type="set" id="315">
  <ses:session xmlns:ses="http://www.google.com/session" type="initiate"
id="1162869130" initiator="husodesar at gmail.com/gtalk94609708">
    <pho:description xmlns:pho="http://www.google.com/session/phone"
xml:lang="en">
      <pho:payload-type xmlns:pho="http://www.google.com/session/phone"
id="0" name="PCMU" clockrate="8000" bitrate="64000"></pho:payload-type>
    </pho:description>
    <transport xmlns="http://www.google.com/transport/p2p"></transport>
  </ses:session>
</iq>
2011-02-11 19:37:18.037842 [DEBUG] libdingaling.c:1450 Processing 1 packets
in retry queue
2011-02-11 19:37:18.045839 [INFO] libdingaling.c:1366 SecRECV:
-------------------------------------------------------------------------------
<iq from="+18009993355 at voice.google.com" to="
husodesar at gmail.com/gtalk94609708" type="error" id="315">
  <ses:session type="initiate" id="1162869130" initiator="
husodesar at gmail.com/gtalk94609708" xmlns:ses="http://www.google.com/session
">
    <pho:description xml:lang="en" xmlns:pho="
http://www.google.com/session/phone">
      <pho:payload-type id="0" name="PCMU" clockrate="8000"
bitrate="64000"></pho:payload-type>
    </pho:description>
    <transport xmlns="http://www.google.com/transport/p2p"></transport>
  </ses:session>
  <error code="302" type="modify">
    <redirect xmlns="urn:ietf:params:xml:ns:xmpp-stanzas">
xmpp:+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1
</redirect>
    <ses:redirect xmlns:ses="
http://www.google.com/session">xmpp:+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1</ses:redirect<http://www.google.com/session%22%3Exmpp:+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1%3C/ses:redirect>
>
  </error>
</iq>
2011-02-11 19:37:18.045839 [DEBUG] libdingaling.c:968 Cancel packet 315
2011-02-11 19:37:18.045839 [DEBUG] libdingaling.c:383 Message for Session
1162869130
2011-02-11 19:37:18.045839 [DEBUG] mod_dingaling.c:2941 using Existing
session for 1162869130
2011-02-11 19:37:18.045839 [DEBUG] mod_dingaling.c:1083 Send Describe
[PCMU at 8000]
2011-02-11 19:37:18.137840 [NOTICE] libdingaling.c:1368 SecSEND:
-------------------------------------------------------------------------------
<iq type="result" to="+18009993355 at voice.google.com" from="
xmpp:+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1"
id="315">
  <query></query>
</iq>
2011-02-11 19:37:18.137840 [DEBUG] libdingaling.c:1450 Processing 2 packets
in retry queue
2011-02-11 19:37:18.137840 [DEBUG] libdingaling.c:1475 Discarding packet 315
2011-02-11 19:37:18.137840 [DEBUG] libdingaling.c:1467 Sending packet 316 (2
left)
2011-02-11 19:37:18.137840 [NOTICE] libdingaling.c:1368 SecSEND:
-------------------------------------------------------------------------------
<iq xmlns="jabber:client" from="husodesar at gmail.com/gtalk94609708" to="
+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1" type="set"
id="316">
  <ses:session xmlns:ses="http://www.google.com/session" type="initiate"
id="1162869130" initiator="husodesar at gmail.com/gtalk94609708">
    <pho:description xmlns:pho="http://www.google.com/session/phone"
xml:lang="en">
      <pho:payload-type xmlns:pho="http://www.google.com/session/phone"
id="0" name="PCMU" clockrate="8000" bitrate="64000"></pho:payload-type>
    </pho:description>
    <transport xmlns="http://www.google.com/transport/p2p"></transport>
  </ses:session>
</iq>
2011-02-11 19:37:18.237846 [DEBUG] libdingaling.c:1450 Processing 1 packets
in retry queue
2011-02-11 19:37:18.297853 [INFO] libdingaling.c:1366 SecRECV:
-------------------------------------------------------------------------------
<iq from="+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1"
to="husodesar at gmail.com/gtalk94609708"
id="jingle:10.229.92.226-10513507:1:0290B654" type="set">
  <ses:session type="transport-accept" id="1162869130" initiator="
husodesar at gmail.com/gtalk94609708" xmlns:ses="http://www.google.com/session
">
    <transport xmlns="http://www.google.com/transport/p2p"></transport>
  </ses:session>
</iq>
2011-02-11 19:37:18.297853 [DEBUG] libdingaling.c:383 Message for Session
1162869130
2011-02-11 19:37:18.297853 [DEBUG] mod_dingaling.c:2941 using Existing
session for 1162869130
2011-02-11 19:37:18.297853 [DEBUG] mod_dingaling.c:1008 Send Candidate
184.105.153.247:24320 [JwkGkXBkIuntRST4]
2011-02-11 19:37:18.297853 [INFO] libdingaling.c:1366 SecRECV:
-------------------------------------------------------------------------------
<iq to="husodesar at gmail.com/gtalk94609708" from="
+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1" id="316"
type="result"></iq>
2011-02-11 19:37:18.297853 [DEBUG] libdingaling.c:968 Cancel packet 316
2011-02-11 19:37:18.337857 [NOTICE] libdingaling.c:1368 SecSEND:
-------------------------------------------------------------------------------
<iq type="result" to="
+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1" from="
husodesar at gmail.com/gtalk94609708"
id="jingle:10.229.92.226-10513507:1:0290B654">
  <query></query>
</iq>
2011-02-11 19:37:18.337857 [DEBUG] libdingaling.c:1450 Processing 2 packets
in retry queue
2011-02-11 19:37:18.337857 [DEBUG] libdingaling.c:1475 Discarding packet 316
2011-02-11 19:37:18.337857 [DEBUG] libdingaling.c:1467 Sending packet 317 (2
left)
2011-02-11 19:37:18.337857 [NOTICE] libdingaling.c:1368 SecSEND:
-------------------------------------------------------------------------------
<iq xmlns="jabber:client" from="husodesar at gmail.com/gtalk94609708" to="
+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1" type="set"
id="317">
  <ses:session xmlns:ses="http://www.google.com/session"
type="transport-info" id="1162869130" initiator="
husodesar at gmail.com/gtalk94609708">
    <transport xmlns="http://www.google.com/transport/p2p">
      <candidate name="rtp" address="184.105.153.247" port="24320"
username="JwkGkXBkIuntRST4" password="JwkGkXBkIuntRST4" preference="1.0"
protocol="udp" type="local" network="0" generation="0"></candidate>
    </transport>
  </ses:session>
</iq>
2011-02-11 19:37:18.437876 [DEBUG] libdingaling.c:1450 Processing 1 packets
in retry queue
2011-02-11 19:37:18.481871 [INFO] libdingaling.c:1366 SecRECV:
-------------------------------------------------------------------------------
<iq to="husodesar at gmail.com/gtalk94609708" from="
+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1" id="317"
type="result"></iq>
2011-02-11 19:37:18.481871 [DEBUG] libdingaling.c:968 Cancel packet 317
2011-02-11 19:37:18.537869 [DEBUG] libdingaling.c:1450 Processing 1 packets
in retry queue
2011-02-11 19:37:18.537869 [DEBUG] libdingaling.c:1475 Discarding packet 317
2011-02-11 19:37:18.585873 [INFO] libdingaling.c:1366 SecRECV:
-------------------------------------------------------------------------------
<iq from="+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1"
to="husodesar at gmail.com/gtalk94609708"
id="jingle:10.229.92.226-10513507:1:0290B656" type="set">
  <ses:session type="transport-info" id="1162869130" initiator="
husodesar at gmail.com/gtalk94609708" xmlns:ses="http://www.google.com/session
">
    <transport xmlns="http://www.google.com/transport/p2p">
      <candidate name="rtp" address="74.125.127.126" port="19295"
username="wCd9MeksClqQziAR" preference="1.0" protocol="udp"
network="mediaproxy" generation="0"></candidate>
      <candidate name="rtp" address="74.125.127.126" port="19294"
username="wCd9MeksClqQziAR" preference="0.6" protocol="tcp"
network="mediaproxy" generation="0"></candidate>
      <candidate name="rtp" address="74.125.127.126" port="443"
username="wCd9MeksClqQziAR" preference="0.5" protocol="ssltcp"
network="mediaproxy" generation="0"></candidate>
    </transport>
  </ses:session>
</iq>
2011-02-11 19:37:18.585873 [DEBUG] libdingaling.c:383 Message for Session
1162869130
2011-02-11 19:37:18.585873 [DEBUG] libdingaling.c:528 New Candidate 1
name=rtp
type=stun
protocol=udp
username=wCd9MeksClqQziAR
password=(null)
address=74.125.127.126
port=19295
pref=1.00
2011-02-11 19:37:18.585873 [DEBUG] libdingaling.c:528 New Candidate 2
name=rtp
type=stun
protocol=tcp
username=wCd9MeksClqQziAR
password=(null)
address=74.125.127.126
port=19294
pref=0.60
2011-02-11 19:37:18.585873 [DEBUG] libdingaling.c:528 New Candidate 3
name=rtp
type=stun
protocol=ssltcp
username=wCd9MeksClqQziAR
password=(null)
address=74.125.127.126
port=443
pref=0.50
2011-02-11 19:37:18.585873 [DEBUG] mod_dingaling.c:2941 using Existing
session for 1162869130
2011-02-11 19:37:18.585873 [DEBUG] mod_dingaling.c:3279 3 candidates
2011-02-11 19:37:18.585873 [DEBUG] mod_dingaling.c:3315 candidates
74.125.127.126:19295
2011-02-11 19:37:18.585873 [DEBUG] mod_dingaling.c:3315 candidates
74.125.127.126:19294
2011-02-11 19:37:18.585873 [DEBUG] mod_dingaling.c:3315 candidates
74.125.127.126:443
2011-02-11 19:37:18.585873 [DEBUG] mod_dingaling.c:3351 Acceptable Candidate
74.125.127.126:19295
2011-02-11 19:37:18.637916 [NOTICE] libdingaling.c:1368 SecSEND:
-------------------------------------------------------------------------------
<iq type="result" to="
+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1" from="
husodesar at gmail.com/gtalk94609708"
id="jingle:10.229.92.226-10513507:1:0290B656">
  <query></query>
</iq>
2011-02-11 19:37:21.542056 [INFO] libdingaling.c:1366 SecRECV:
-------------------------------------------------------------------------------
<iq from="+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1"
to="husodesar at gmail.com/gtalk94609708"
id="jingle:10.229.92.226-10513507:1:0290B659" type="set">
  <ses:session type="accept" id="1162869130" initiator="
husodesar at gmail.com/gtalk94609708" xmlns:ses="http://www.google.com/session
">
    <pho:description xmlns:pho="http://www.google.com/session/phone">
      <pho:payload-type id="0" name="PCMU"
clockrate="8000"></pho:payload-type>
      <pho:payload-type id="101" name="telephone-event"></pho:payload-type>
    </pho:description>
  </ses:session>
</iq>
2011-02-11 19:37:21.542056 [DEBUG] libdingaling.c:383 Message for Session
1162869130
2011-02-11 19:37:21.542056 [DEBUG] libdingaling.c:440 Add Payload [PCMU]
id='0'
2011-02-11 19:37:21.542056 [DEBUG] libdingaling.c:440 Add Payload
[telephone-event] id='101'
2011-02-11 19:37:21.542056 [DEBUG] mod_dingaling.c:2941 using Existing
session for 1162869130
2011-02-11 19:37:21.542056 [DEBUG] mod_dingaling.c:3193 Already decided on a
codec
2011-02-11 19:37:21.542056 [NOTICE] libdingaling.c:1368 SecSEND:
-------------------------------------------------------------------------------
<iq type="result" to="
+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1" from="
husodesar at gmail.com/gtalk94609708"
id="jingle:10.229.92.226-10513507:1:0290B659">
  <query></query>
</iq>
2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:865 Set Read Codec to
PCMU at 8000
2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:880 Set Write Codec to
PCMU at 8000
2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:892 SETUP RTP
184.105.153.247:0 -> 74.125.127.126:19295
2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:914 RTP ERROR Missing
local port
2011-02-11 19:37:21.546355 [DEBUG] switch_channel.c:2538 (
dingaling/gv1/+18009993355 at voice.google.com) Callstate Change DOWN -> HANGUP
2011-02-11 19:37:21.546355 [NOTICE] mod_dingaling.c:915 Hangup
dingaling/gv1/+18009993355 at voice.google.com [CS_INIT]
[DESTINATION_OUT_OF_ORDER]
2011-02-11 19:37:21.546355 [DEBUG] switch_channel.c:2554 Send signal
dingaling/gv1/+18009993355 at voice.google.com [KILL]
2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:1348
dingaling/gv1/+18009993355 at voice.google.com CHANNEL KILL
2011-02-11 19:37:21.546355 [DEBUG] switch_core_session.c:1116 Send signal
dingaling/gv1/+18009993355 at voice.google.com [BREAK]
2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:1348
dingaling/gv1/+18009993355 at voice.google.com CHANNEL KILL
2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:710 Terminate called from
line 1177 state=CS_HANGUP
2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:356 (
dingaling/gv1/+18009993355 at voice.google.com) State INIT going to sleep
2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:320 (
dingaling/gv1/+18009993355 at voice.google.com) Running State Change CS_HANGUP
2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:557 (
dingaling/gv1/+18009993355 at voice.google.com) State HANGUP
2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:1317
dingaling/gv1/+18009993355 at voice.google.com CHANNEL HANGUP
2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:46
dingaling/gv1/+18009993355 at voice.google.com Standard HANGUP, cause:
DESTINATION_OUT_OF_ORDER
2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:557 (
dingaling/gv1/+18009993355 at voice.google.com) State HANGUP going to sleep
2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:351 (
dingaling/gv1/+18009993355 at voice.google.com) State Change CS_HANGUP ->
CS_REPORTING
2011-02-11 19:37:21.546355 [DEBUG] switch_core_session.c:1116 Send signal
dingaling/gv1/+18009993355 at voice.google.com [BREAK]
2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:1348
dingaling/gv1/+18009993355 at voice.google.com CHANNEL KILL
2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:320 (
dingaling/gv1/+18009993355 at voice.google.com) Running State Change
CS_REPORTING
2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:617 (
dingaling/gv1/+18009993355 at voice.google.com) State REPORTING
2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:53
dingaling/gv1/+18009993355 at voice.google.com Standard REPORTING, cause:
DESTINATION_OUT_OF_ORDER
2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:617 (
dingaling/gv1/+18009993355 at voice.google.com) State REPORTING going to sleep
2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:345 (
dingaling/gv1/+18009993355 at voice.google.com) State Change CS_REPORTING ->
CS_DESTROY
2011-02-11 19:37:21.546355 [DEBUG] switch_core_session.c:1116 Send signal
dingaling/gv1/+18009993355 at voice.google.com [BREAK]
2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:1348
dingaling/gv1/+18009993355 at voice.google.com CHANNEL KILL
2011-02-11 19:37:21.546355 [DEBUG] switch_core_session.c:1288 Session 12 (
dingaling/gv1/+18009993355 at voice.google.com) Locked, Waiting on external
entities
2011-02-11 19:37:21.550059 [DEBUG] switch_ivr_originate.c:3502 Originate
Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2011-02-11 19:37:21.550059 [INFO] mod_dptools.c:2623 Originate Failed.
Cause: DESTINATION_OUT_OF_ORDER
2011-02-11 19:37:21.550059 [DEBUG] switch_channel.c:2538 (
sofia/internal/1000 at 184.105.153.247) Callstate Change RINGING -> HANGUP
2011-02-11 19:37:21.550059 [NOTICE] mod_dptools.c:2686 Hangup
sofia/internal/1000 at 184.105.153.247 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
2011-02-11 19:37:21.550059 [DEBUG] switch_channel.c:2554 Send signal
sofia/internal/1000 at 184.105.153.247 [KILL]
2011-02-11 19:37:21.550059 [DEBUG] switch_core_session.c:1116 Send signal
sofia/internal/1000 at 184.105.153.247 [BREAK]
2011-02-11 19:37:21.550059 [DEBUG] switch_core_session.c:2060
sofia/internal/1000 at 184.105.153.247 skip receive message
[APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2011-02-11 19:37:21.550059 [DEBUG] switch_core_state_machine.c:366 (
sofia/internal/1000 at 184.105.153.247) State EXECUTE going to sleep
2011-02-11 19:37:21.550059 [DEBUG] switch_core_state_machine.c:320 (
sofia/internal/1000 at 184.105.153.247) Running State Change CS_HANGUP
2011-02-11 19:37:21.550059 [DEBUG] switch_core_state_machine.c:557 (
sofia/internal/1000 at 184.105.153.247) State HANGUP
2011-02-11 19:37:21.550059 [DEBUG] mod_sofia.c:457 Channel
sofia/internal/1000 at 184.105.153.247 hanging up, cause:
DESTINATION_OUT_OF_ORDER
2011-02-11 19:37:21.550059 [NOTICE] switch_core_session.c:1306 Session 12 (
dingaling/gv1/+18009993355 at voice.google.com) Ended
2011-02-11 19:37:21.550059 [NOTICE] switch_core_session.c:1308 Close Channel
dingaling/gv1/+18009993355 at voice.google.com [CS_DESTROY]
2011-02-11 19:37:21.550059 [DEBUG] switch_core_state_machine.c:449 (
dingaling/gv1/+18009993355 at voice.google.com) Callstate Change HANGUP -> DOWN
2011-02-11 19:37:21.550059 [DEBUG] switch_core_state_machine.c:452 (
dingaling/gv1/+18009993355 at voice.google.com) Running State Change CS_DESTROY
2011-02-11 19:37:21.550059 [DEBUG] switch_core_state_machine.c:462 (
dingaling/gv1/+18009993355 at voice.google.com) State DESTROY
2011-02-11 19:37:21.550059 [DEBUG] libdingaling.c:299 Destroyed Session
1162869130
2011-02-11 19:37:21.550059 [DEBUG] switch_core_state_machine.c:60
dingaling/gv1/+18009993355 at voice.google.com Standard DESTROY
2011-02-11 19:37:21.550059 [DEBUG] switch_core_state_machine.c:462 (
dingaling/gv1/+18009993355 at voice.google.com) State DESTROY going to sleep
2011-02-11 19:37:21.562076 [DEBUG] mod_sofia.c:519 Responding to INVITE
with: 502
2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:46
sofia/internal/1000 at 184.105.153.247 Standard HANGUP, cause:
DESTINATION_OUT_OF_ORDER
2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:557 (
sofia/internal/1000 at 184.105.153.247) State HANGUP going to sleep
2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:351 (
sofia/internal/1000 at 184.105.153.247) State Change CS_HANGUP -> CS_REPORTING
2011-02-11 19:37:21.562076 [DEBUG] switch_core_session.c:1116 Send signal
sofia/internal/1000 at 184.105.153.247 [BREAK]
2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:320 (
sofia/internal/1000 at 184.105.153.247) Running State Change CS_REPORTING
2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:617 (
sofia/internal/1000 at 184.105.153.247) State REPORTING
2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:53
sofia/internal/1000 at 184.105.153.247 Standard REPORTING, cause:
DESTINATION_OUT_OF_ORDER
2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:617 (
sofia/internal/1000 at 184.105.153.247) State REPORTING going to sleep
2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:345 (
sofia/internal/1000 at 184.105.153.247) State Change CS_REPORTING -> CS_DESTROY
2011-02-11 19:37:21.562076 [DEBUG] switch_core_session.c:1116 Send signal
sofia/internal/1000 at 184.105.153.247 [BREAK]
2011-02-11 19:37:21.562076 [DEBUG] switch_core_session.c:1288 Session 11 (
sofia/internal/1000 at 184.105.153.247) Locked, Waiting on external entities
2011-02-11 19:37:21.562076 [NOTICE] switch_core_session.c:1306 Session 11 (
sofia/internal/1000 at 184.105.153.247) Ended
2011-02-11 19:37:21.562076 [NOTICE] switch_core_session.c:1308 Close Channel
sofia/internal/1000 at 184.105.153.247 [CS_DESTROY]
2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:449 (
sofia/internal/1000 at 184.105.153.247) Callstate Change HANGUP -> DOWN
2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:452 (
sofia/internal/1000 at 184.105.153.247) Running State Change CS_DESTROY
2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:462 (
sofia/internal/1000 at 184.105.153.247) State DESTROY
2011-02-11 19:37:21.562076 [DEBUG] mod_sofia.c:362
sofia/internal/1000 at 184.105.153.247 SOFIA DESTROY
2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:60
sofia/internal/1000 at 184.105.153.247 Standard DESTROY
2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:462 (
sofia/internal/1000 at 184.105.153.247) State DESTROY going to sleep
2011-02-11 19:37:48.963766 [INFO] libdingaling.c:1366 SecRECV:
-------------------------------------------------------------------------------
<iq from="+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1"
to="husodesar at gmail.com/gtalk94609708"
id="jingle:10.229.92.226-10513507:1:0290B66D" type="set">
  <ses:session type="terminate" id="1162869130" initiator="
husodesar at gmail.com/gtalk94609708" xmlns:ses="http://www.google.com/session
">
    <pho:internal-server-error xmlns:pho="
http://www.google.com/session/phone">Mediaterminated</pho:internal-server-error>
  </ses:session>
</iq>
2011-02-11 19:37:48.963766 [DEBUG] libdingaling.c:355 Created Session
1162869130
2011-02-11 19:37:48.963766 [DEBUG] libdingaling.c:383 Message for Session
1162869130
2011-02-11 19:37:48.963766 [DEBUG] mod_dingaling.c:2951 Session is already
dead
2011-02-11 19:37:49.055819 [NOTICE] libdingaling.c:1368 SecSEND:
-------------------------------------------------------------------------------
<iq type="result" to="
+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1" from="
husodesar at gmail.com/gtalk94609708"
id="jingle:10.229.92.226-10513507:1:0290B66D">
  <query></query>
</iq>
-------------- next part --------------
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From anthony.minessale at gmail.com  Fri Feb 11 14:56:01 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 11 Feb 2011 05:56:01 -0600
Subject: [Freeswitch-users] hold_music var gets unset/lost by
 cancelled/failed att_xfer
In-Reply-To: <AANLkTi=Hv7kkYeD8TQeCZtfvmzqaCx+xQ=yYC16kpNfa@mail.gmail.com>
References: <AANLkTimLod3yHrex4LAK=4gdVcPhzXutKYyw=E8ZMcN9@mail.gmail.com>
	<AANLkTi=u3VeHPuS5x1e+yu5QNBw5ZSgHvAWOT+F6A_OR@mail.gmail.com>
	<AANLkTi=EndkiUJvCgCSirxCA23EwWD_++0h3Qcr=mg9g@mail.gmail.com>
	<AANLkTikt97dugbhKPgnB2ws4bOFcxg8yhzSdyxHMab_m@mail.gmail.com>
	<AANLkTikEx9ZwNADRQ8k2JbTwgoLPFmHUPpPL3RLnZZU8@mail.gmail.com>
	<AANLkTinS3wxA8izr9tXX7jhYJPHq_X_coO7UNoUkjs-P@mail.gmail.com>
	<AANLkTi=Hv7kkYeD8TQeCZtfvmzqaCx+xQ=yYC16kpNfa@mail.gmail.com>
Message-ID: <AANLkTikKt8TC-g3Q5LpXEZh62dGY6+f=HRO00n4G4PU3@mail.gmail.com>

I reverted the patch, I guess it doesn't work based on your feedback.
We will leave well enough alone when you said in your last email
everything was working.

There is a time when maybe you are taking things too far trying to
solve things a silly way and maybe you should consider nicer
equipment.


On Fri, Feb 11, 2011 at 4:43 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> I see your patch in git log, is it enough to do 'make current', or I
> should apply it in some special way? After make current issues are
> still present...
>
> What should i set loopback_bowout/bowout_on_execute variables to? I
> tried with false, this works as in previous versions, and loopback
> stays in path during the call. While loopback channels are present in
> path, there are issues with sound quality, there is robot voice effect
> for short periods of time. You mentioned setting rtp timer to none,
> where this should be done? In loopback channel execute_extension while
> doing att_xfer?
>
> When bowout variables set to true ?att_xfer doesn't work just as I
> described in my previous posts, and there is also issue with MOH -
> after transfer target answers to transferor, MOH is stoped for
> transferee. Maybe this can help somehow.
>
>
>
>
> 2011/2/10 Anthony Minessale <anthony.minessale at gmail.com>:
>> I pushed a patch that will probably delay the bowout until after the
>> att_xfer is over
>> Give it a try.
>>
>> commit 3546654615f88058fb6769fe79e07162602fa4af
>> Author: Anthony Minessale <anthm at freeswitch.org>
>> Date: ? Thu Feb 10 12:37:14 2011 -0600
>>
>> ? ?don't bow out on att_xfer bridge
>>
>>
>> On Thu, Feb 10, 2011 at 1:03 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>> Thanks! It works now! BTW, att_xfer with
>>> loopback_bowout/bowout_on_execute set to false seems to be working
>>> too. Is there a way to make loopback channel leave the path? Without
>>> bowout turned off att_xfer doesn't work...
>>>
>>> 2011/2/8 Anthony Minessale <anthony.minessale at gmail.com>:
>>>> try latest GIT
>>>>
>>>> On Tue, Feb 8, 2011 at 1:28 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>>>> I have same issue with MOH and att_xfer on failed transfers, music on
>>>>> hold played only once
>>>>> At the same time, transfer_ringback always plays correctly to transferer
>>>>>
>>>>> 2011/2/8 Anthony Minessale <anthony.minessale at gmail.com>:
>>>>>> you really should report this to jira not to the mailing list.
>>>>>> http://jira.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Mon, Feb 7, 2011 at 4:23 PM, Fraser Redmond <fraserredmond at gmail.com> wrote:
>>>>>>> I'm trying to do a second att_xfer on a call so that if the first attended
>>>>>>> transfer fails (c-leg is busy, or presses do-not-answer, or is an extn that
>>>>>>> doesn't exist) then the call could be transferred to someone else.
>>>>>>>
>>>>>>> On the first att_xfer the person on hold hears the hold_music correctly.
>>>>>>> Once that transfer is cancelled or fails:
>>>>>>> -- On any subsequent att_xfer's the person on hold just hears silence.
>>>>>>> -- If they are put on hold they just hear silence.
>>>>>>>
>>>>>>> I tried setting hold_music again for each channel after the first att_xfer,
>>>>>>> but that didn't work, so it's probably not actually a problem with
>>>>>>> hold_music per se, but some other variable/setting that decides whether to
>>>>>>> use hold_music.
>>>>>>>
>>>>>>> I also tried doing a uuid_dump before and after each attempt, but didn't
>>>>>>> notice anything too different - unless it's a matter of unsetting one of the
>>>>>>> couple of changed/new vars like:
>>>>>>> variable_originate_disposition
>>>>>>> variable_current_application
>>>>>>> variable_playback_seconds
>>>>>>>
>>>>>>> I get the feeling other variables are probably also lost by the first failed
>>>>>>> transfer as the second att_xfer has some odd things happen if the third
>>>>>>> party does answer. Haven't been able to narrow it down as closely as the
>>>>>>> hold_music, but two things I've seen happen are:
>>>>>>> -- The party that initiated the transfer gets hung up automatically (after
>>>>>>> 30 sec)
>>>>>>> -- When the party that initiated the transfer hangs up it should connect the
>>>>>>> other two parties, but instead it hung up all three
>>>>>>>
>>>>>>> Cheers,
>>>>>>> Fraser
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at 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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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  Fri Feb 11 15:00:16 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 11 Feb 2011 06:00:16 -0600
Subject: [Freeswitch-users] dp_tools/set- not worked SOMETIMES
In-Reply-To: <20110211114004.GA1491@bsdjail.com>
References: <20110211112453.GA700@bsdjail.com>
	<20110211114004.GA1491@bsdjail.com>
Message-ID: <AANLkTim=+R9snJzYiCdUNZiixxOcOy-6C2j=SjpzsBSN@mail.gmail.com>

This is your second strange issue. Maybe you have some problem with
your platform.
Are you using unaltered GIT HEAD?  Have you compared results on other
platforms like Linux 64 bit or Windows?


On Fri, Feb 11, 2011 at 5:40 AM, Ruslan Bukin <br at bsdpad.com> wrote:
> correction: it seems that parameters always exists after bridge,
> but sometimes one of them not exists in CDR result
> (I have checked with mod_xml_cdr and mod_json_cdr)
>
> -Ruslan
>
> On Fri, Feb 11, 2011 at 02:24:53PM +0300, Ruslan Bukin wrote:
>> Application set is not set parameters SOMETIMES
>>
>> for example, before bridge I set my static parameters:
>>
>> <action application="set" data="my_own_parameter=test"/>
>> <action application="set" data="nibble_rate=1"/>
>> ...
>>
>> sometimes (in about ~1-5% times) one of them is not set up.
>>
>> I tried to set like this before bridge:
>>
>> ?<extension name="set_var" continue="true">
>> ? ?<condition field="${my_own_parameter}" expression=".+" break="on-true"/>
>> ? ?<condition field="destination_number" expression="(777)" break="on-true">
>> ? ? ?<action application="set" data="my_own_parameter=test"/>
>> ? ? ?<action application="transfer" data="$1 XML public"/>
>> ? ?</condition>
>> ?</extension>
>>
>> but no result: after bridge no parameter to read (SOMETIMES)
>>
>>
>> any recommendations?
>>
>> -Ruslan
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb 11 15:02:11 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 11 Feb 2011 06:02:11 -0600
Subject: [Freeswitch-users] Question about mod_dingaling.
In-Reply-To: <AANLkTi=hu9iWu3SQbF5Wj3odkY0ut9HGn0mabsBZJMbQ@mail.gmail.com>
References: <AANLkTikd9baOGf0bmL6m3EaiPcGy6Fgfm9=jEYBg+GB2@mail.gmail.com>
	<AANLkTimXmNou2T4tLPMACscdHiHVqUzf3U__WeGVaEJb@mail.gmail.com>
	<AANLkTi=hu9iWu3SQbF5Wj3odkY0ut9HGn0mabsBZJMbQ@mail.gmail.com>
Message-ID: <AANLkTinanoYB3J-0rmYtM7uw5m5HJC829YmZbxaHWzVh@mail.gmail.com>

dont post logs on the mailing list. Put them on http://pastebin.freeswitch.org

Also did you use the wiki to learn how to do this?

Start with fresh default configuration and follow:
http://wiki.freeswitch.org/wiki/Google_Voice



2011/2/11 ??? <chenzhanping at gmail.com>:
> I try to install freeswitch 1.0.7 download from latest.freeswitch.org,but
> can not call out from mod_dingaling use google voice always.
>
> this is a part of cli log:
>
>
> 2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:865 Set Read Codec to
> PCMU at 8000
> 2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:880 Set Write Codec to
> PCMU at 8000
> 2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:892 SETUP RTP
> 184.105.153.247:0 -> 74.125.127.126:19295
> 2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:914 RTP ERROR Missing
> local port
> 2011-02-11 19:37:21.546355 [DEBUG] switch_channel.c:2538
> (dingaling/gv1/+18009993355 at voice.google.com) Callstate Change DOWN ->
> HANGUP
> 2011-02-11 19:37:21.546355 [NOTICE] mod_dingaling.c:915 Hangup
> dingaling/gv1/+18009993355 at voice.google.com [CS_INIT]
> [DESTINATION_OUT_OF_ORDER]
> 2011-02-11 19:37:21.546355 [DEBUG] switch_channel.c:2554 Send signal
> dingaling/gv1/+18009993355 at voice.google.com [KILL]
> 2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:1348
> dingaling/gv1/+18009993355 at voice.google.com CHANNEL KILL
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_session.c:1116 Send signal
> dingaling/gv1/+18009993355 at voice.google.com [BREAK]
> 2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:1348
> dingaling/gv1/+18009993355 at voice.google.com CHANNEL KILL
> 2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:710 Terminate called from
> line 1177 state=CS_HANGUP
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:356
> (dingaling/gv1/+18009993355 at voice.google.com) State INIT going to sleep
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:320
> (dingaling/gv1/+18009993355 at voice.google.com) Running State Change CS_HANGUP
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:557
> (dingaling/gv1/+18009993355 at voice.google.com) State HANGUP
> 2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:1317
> dingaling/gv1/+18009993355 at voice.google.com CHANNEL HANGUP
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:46
> dingaling/gv1/+18009993355 at voice.google.com Standard HANGUP, cause:
> DESTINATION_OUT_OF_ORDER
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:557
> (dingaling/gv1/+18009993355 at voice.google.com) State HANGUP going to sleep
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:351
> (dingaling/gv1/+18009993355 at voice.google.com) State Change CS_HANGUP ->
> CS_REPORTING
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_session.c:1116 Send signal
> dingaling/gv1/+18009993355 at voice.google.com [BREAK]
> 2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:1348
> dingaling/gv1/+18009993355 at voice.google.com CHANNEL KILL
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:320
> (dingaling/gv1/+18009993355 at voice.google.com) Running State Change
> CS_REPORTING
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:617
> (dingaling/gv1/+18009993355 at voice.google.com) State REPORTING
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:53
> dingaling/gv1/+18009993355 at voice.google.com Standard REPORTING, cause:
> DESTINATION_OUT_OF_ORDER
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:617
> (dingaling/gv1/+18009993355 at voice.google.com) State REPORTING going to sleep
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:345
> (dingaling/gv1/+18009993355 at voice.google.com) State Change CS_REPORTING ->
> CS_DESTROY
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_session.c:1116 Send signal
> dingaling/gv1/+18009993355 at voice.google.com [BREAK]
> 2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:1348
> dingaling/gv1/+18009993355 at voice.google.com CHANNEL KILL
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_session.c:1288 Session 12
> (dingaling/gv1/+18009993355 at voice.google.com) Locked, Waiting on external
> entities
> 2011-02-11 19:37:21.550059 [DEBUG] switch_ivr_originate.c:3502 Originate
> Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
> 2011-02-11 19:37:21.550059 [INFO] mod_dptools.c:2623 Originate Failed.
> Cause: DESTINATION_OUT_OF_ORDER
> 2011-02-11 19:37:21.550059 [DEBUG] switch_channel.c:2538
> (sofia/internal/1000 at 184.105.153.247) Callstate Change RINGING -> HANGUP
> 2011-02-11 19:37:21.550059 [NOTICE] mod_dptools.c:2686 Hangup
> sofia/internal/1000 at 184.105.153.247 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
> 2011-02-11 19:37:21.550059 [DEBUG] switch_channel.c:2554 Send signal
> sofia/internal/1000 at 184.105.153.247 [KILL]
> 2011-02-11 19:37:21.550059 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/internal/1000 at 184.105.153.247 [BREAK]
>
> all log is here:
>
>
> freeswitch at yijiaren.net.cn> 2011-02-11 19:37:17.481827 [DEBUG] sofia.c:6427
> IP 121.28.15.8 Rejected by acl "domains". Falling back to Digest auth.
> 2011-02-11 19:37:17.485846 [WARNING] sofia_reg.c:1247 SIP auth challenge
> (INVITE) on sofia profile 'internal' for [5018009993355 at 184.105.153.247]
> from ip 121.28.15.8
> 2011-02-11 19:37:17.785818 [DEBUG] sofia.c:6427 IP 121.28.15.8 Rejected by
> acl "domains". Falling back to Digest auth.
> 2011-02-11 19:37:17.789834 [NOTICE] switch_channel.c:811 New Channel
> sofia/internal/1000 at 184.105.153.247 [4716f8ad-de2d-4de6-9848-834add30ff1f]
> 2011-02-11 19:37:17.789834 [DEBUG] switch_core_state_machine.c:320
> (sofia/internal/1000 at 184.105.153.247) Running State Change CS_NEW
> 2011-02-11 19:37:17.789834 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/1000 at 184.105.153.247) State NEW
> 2011-02-11 19:37:17.830152 [DEBUG] sofia.c:4659 Channel
> sofia/internal/1000 at 184.105.153.247 entering state [received][100]
> 2011-02-11 19:37:17.830152 [DEBUG] sofia.c:4670 Remote SDP:
> v=0
> o=- 8 2 IN IP4 61.235.150.11
> s=CounterPath eyeBeam 1.5
> c=IN IP4 61.235.150.11
> t=0 0
> m=audio 6816 RTP/AVP 0 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=alt:1 3 : H+ZKvDCR 5iYo6mbN 192.168.52.1 6816
> a=alt:2 2 : yZueEnyN QCAU4mGC 192.168.153.1 6816
> a=alt:3 1 : g4e9fZKB 6/CgnDNR 61.235.150.11 6816
> a=x-rtp-session-id:41F847609F534152A0EDFEBE5D1D93FC
> 2011-02-11 19:37:17.830152 [DEBUG] sofia_glue.c:4474 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000]
> 2011-02-11 19:37:17.830152 [DEBUG] sofia_glue.c:4474 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000]
> 2011-02-11 19:37:17.830152 [DEBUG] sofia_glue.c:4474 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
> 2011-02-11 19:37:17.830152 [DEBUG] sofia_glue.c:2757 Set Codec
> sofia/internal/1000 at 184.105.153.247 PCMU/8000 20 ms 160 samples 64000 bits
> 2011-02-11 19:37:17.830152 [DEBUG] sofia_glue.c:4578 Set 2833 dtmf send/recv
> payload to 101
> 2011-02-11 19:37:17.830152 [DEBUG] sofia.c:4837
> (sofia/internal/1000 at 184.105.153.247) State Change CS_NEW -> CS_INIT
> 2011-02-11 19:37:17.830152 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/internal/1000 at 184.105.153.247 [BREAK]
> 2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:320
> (sofia/internal/1000 at 184.105.153.247) Running State Change CS_INIT
> 2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:356
> (sofia/internal/1000 at 184.105.153.247) State INIT
> 2011-02-11 19:37:17.833837 [DEBUG] mod_sofia.c:84
> sofia/internal/1000 at 184.105.153.247 SOFIA INIT
> 2011-02-11 19:37:17.833837 [DEBUG] mod_sofia.c:124
> (sofia/internal/1000 at 184.105.153.247) State Change CS_INIT -> CS_ROUTING
> 2011-02-11 19:37:17.833837 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/internal/1000 at 184.105.153.247 [BREAK]
> 2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:356
> (sofia/internal/1000 at 184.105.153.247) State INIT going to sleep
> 2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:320
> (sofia/internal/1000 at 184.105.153.247) Running State Change CS_ROUTING
> 2011-02-11 19:37:17.833837 [DEBUG] switch_channel.c:1660
> (sofia/internal/1000 at 184.105.153.247) Callstate Change DOWN -> RINGING
> 2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:359
> (sofia/internal/1000 at 184.105.153.247) State ROUTING
> 2011-02-11 19:37:17.833837 [DEBUG] mod_sofia.c:147
> sofia/internal/1000 at 184.105.153.247 SOFIA ROUTING
> 2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:77
> sofia/internal/1000 at 184.105.153.247 Standard ROUTING
> 2011-02-11 19:37:17.833837 [INFO] mod_dialplan_xml.c:331 Processing 1000
> <1000>->5018009993355 in context default
> Dialplan: sofia/internal/1000 at 184.105.153.247 parsing [default->gvoice_out]
> continue=false
> Dialplan: sofia/internal/1000 at 184.105.153.247 Regex (PASS) [gvoice_out]
> destination_number(5018009993355) =~ /^50(1\d{10})$/ break=on-false
> Dialplan: sofia/internal/1000 at 184.105.153.247 Action
> set(hangup_after_bridge=true)
> Dialplan: sofia/internal/1000 at 184.105.153.247 Action
> bridge(dingaling/gv1/+18009993355 at voice.google.com)
> 2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:119
> (sofia/internal/1000 at 184.105.153.247) State Change CS_ROUTING -> CS_EXECUTE
> 2011-02-11 19:37:17.833837 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/internal/1000 at 184.105.153.247 [BREAK]
> 2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:359
> (sofia/internal/1000 at 184.105.153.247) State ROUTING going to sleep
> 2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:320
> (sofia/internal/1000 at 184.105.153.247) Running State Change CS_EXECUTE
> 2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:366
> (sofia/internal/1000 at 184.105.153.247) State EXECUTE
> 2011-02-11 19:37:17.833837 [DEBUG] mod_sofia.c:240
> sofia/internal/1000 at 184.105.153.247 SOFIA EXECUTE
> 2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:157
> sofia/internal/1000 at 184.105.153.247 Standard EXECUTE
> EXECUTE sofia/internal/1000 at 184.105.153.247 set(hangup_after_bridge=true)
> 2011-02-11 19:37:17.833837 [DEBUG] mod_dptools.c:1059
> sofia/internal/1000 at 184.105.153.247 SET [hangup_after_bridge]=[true]
> EXECUTE sofia/internal/1000 at 184.105.153.247
> bridge(dingaling/gv1/+18009993355 at voice.google.com)
> 2011-02-11 19:37:17.833837 [NOTICE] switch_channel.c:811 New Channel
> dingaling/gv1/+18009993355 at voice.google.com
> [c024cca7-4b61-475f-a04a-023e808cb41f]
> 2011-02-11 19:37:17.833837 [DEBUG] libdingaling.c:355 Created Session
> 1162869130
> 2011-02-11 19:37:17.833837 [DEBUG] mod_dingaling.c:1824
> (dingaling/gv1/+18009993355 at voice.google.com) State Change CS_NEW -> CS_INIT
> 2011-02-11 19:37:17.833837 [DEBUG] switch_core_session.c:1116 Send signal
> dingaling/gv1/+18009993355 at voice.google.com [BREAK]
> 2011-02-11 19:37:17.833837 [DEBUG] mod_dingaling.c:1348
> dingaling/gv1/+18009993355 at voice.google.com CHANNEL KILL
> 2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:320
> (dingaling/gv1/+18009993355 at voice.google.com) Running State Change CS_INIT
> 2011-02-11 19:37:17.833837 [DEBUG] switch_core_state_machine.c:356
> (dingaling/gv1/+18009993355 at voice.google.com) State INIT
> 2011-02-11 19:37:17.833837 [NOTICE] mod_dingaling.c:1110 Ring-Ready
> dingaling/gv1/+18009993355 at voice.google.com!
> 2011-02-11 19:37:17.833837 [DEBUG] mod_dingaling.c:1063 Don't have my codec
> yet here's one
> 2011-02-11 19:37:17.833837 [DEBUG] mod_dingaling.c:1083 Send Describe
> [PCMU at 8000]
> 2011-02-11 19:37:17.937903 [DEBUG] libdingaling.c:1450 Processing 1 packets
> in retry queue
> 2011-02-11 19:37:17.937903 [DEBUG] libdingaling.c:1467 Sending packet 315 (2
> left)
> 2011-02-11 19:37:17.937903 [NOTICE] libdingaling.c:1368 SecSEND:
> -------------------------------------------------------------------------------
> <iq xmlns="jabber:client" from="husodesar at gmail.com/gtalk94609708"
> to="+18009993355 at voice.google.com" type="set" id="315">
>   <ses:session xmlns:ses="http://www.google.com/session" type="initiate"
> id="1162869130" initiator="husodesar at gmail.com/gtalk94609708">
>     <pho:description xmlns:pho="http://www.google.com/session/phone"
> xml:lang="en">
>       <pho:payload-type xmlns:pho="http://www.google.com/session/phone"
> id="0" name="PCMU" clockrate="8000" bitrate="64000"></pho:payload-type>
>     </pho:description>
>     <transport xmlns="http://www.google.com/transport/p2p"></transport>
>   </ses:session>
> </iq>
> 2011-02-11 19:37:18.037842 [DEBUG] libdingaling.c:1450 Processing 1 packets
> in retry queue
> 2011-02-11 19:37:18.045839 [INFO] libdingaling.c:1366 SecRECV:
> -------------------------------------------------------------------------------
> <iq from="+18009993355 at voice.google.com"
> to="husodesar at gmail.com/gtalk94609708" type="error" id="315">
>   <ses:session type="initiate" id="1162869130"
> initiator="husodesar at gmail.com/gtalk94609708"
> xmlns:ses="http://www.google.com/session">
>     <pho:description xml:lang="en"
> xmlns:pho="http://www.google.com/session/phone">
>       <pho:payload-type id="0" name="PCMU" clockrate="8000"
> bitrate="64000"></pho:payload-type>
>     </pho:description>
>     <transport xmlns="http://www.google.com/transport/p2p"></transport>
>   </ses:session>
>   <error code="302" type="modify">
>     <redirect
> xmlns="urn:ietf:params:xml:ns:xmpp-stanzas">xmpp:+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1</redirect>
>     <ses:redirect
> xmlns:ses="http://www.google.com/session">xmpp:+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1</ses:redirect>
>   </error>
> </iq>
> 2011-02-11 19:37:18.045839 [DEBUG] libdingaling.c:968 Cancel packet 315
> 2011-02-11 19:37:18.045839 [DEBUG] libdingaling.c:383 Message for Session
> 1162869130
> 2011-02-11 19:37:18.045839 [DEBUG] mod_dingaling.c:2941 using Existing
> session for 1162869130
> 2011-02-11 19:37:18.045839 [DEBUG] mod_dingaling.c:1083 Send Describe
> [PCMU at 8000]
> 2011-02-11 19:37:18.137840 [NOTICE] libdingaling.c:1368 SecSEND:
> -------------------------------------------------------------------------------
> <iq type="result" to="+18009993355 at voice.google.com"
> from="xmpp:+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1"
> id="315">
>   <query></query>
> </iq>
> 2011-02-11 19:37:18.137840 [DEBUG] libdingaling.c:1450 Processing 2 packets
> in retry queue
> 2011-02-11 19:37:18.137840 [DEBUG] libdingaling.c:1475 Discarding packet 315
> 2011-02-11 19:37:18.137840 [DEBUG] libdingaling.c:1467 Sending packet 316 (2
> left)
> 2011-02-11 19:37:18.137840 [NOTICE] libdingaling.c:1368 SecSEND:
> -------------------------------------------------------------------------------
> <iq xmlns="jabber:client" from="husodesar at gmail.com/gtalk94609708"
> to="+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1"
> type="set" id="316">
>   <ses:session xmlns:ses="http://www.google.com/session" type="initiate"
> id="1162869130" initiator="husodesar at gmail.com/gtalk94609708">
>     <pho:description xmlns:pho="http://www.google.com/session/phone"
> xml:lang="en">
>       <pho:payload-type xmlns:pho="http://www.google.com/session/phone"
> id="0" name="PCMU" clockrate="8000" bitrate="64000"></pho:payload-type>
>     </pho:description>
>     <transport xmlns="http://www.google.com/transport/p2p"></transport>
>   </ses:session>
> </iq>
> 2011-02-11 19:37:18.237846 [DEBUG] libdingaling.c:1450 Processing 1 packets
> in retry queue
> 2011-02-11 19:37:18.297853 [INFO] libdingaling.c:1366 SecRECV:
> -------------------------------------------------------------------------------
> <iq from="+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1"
> to="husodesar at gmail.com/gtalk94609708"
> id="jingle:10.229.92.226-10513507:1:0290B654" type="set">
>   <ses:session type="transport-accept" id="1162869130"
> initiator="husodesar at gmail.com/gtalk94609708"
> xmlns:ses="http://www.google.com/session">
>     <transport xmlns="http://www.google.com/transport/p2p"></transport>
>   </ses:session>
> </iq>
> 2011-02-11 19:37:18.297853 [DEBUG] libdingaling.c:383 Message for Session
> 1162869130
> 2011-02-11 19:37:18.297853 [DEBUG] mod_dingaling.c:2941 using Existing
> session for 1162869130
> 2011-02-11 19:37:18.297853 [DEBUG] mod_dingaling.c:1008 Send Candidate
> 184.105.153.247:24320 [JwkGkXBkIuntRST4]
> 2011-02-11 19:37:18.297853 [INFO] libdingaling.c:1366 SecRECV:
> -------------------------------------------------------------------------------
> <iq to="husodesar at gmail.com/gtalk94609708"
> from="+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1"
> id="316" type="result"></iq>
> 2011-02-11 19:37:18.297853 [DEBUG] libdingaling.c:968 Cancel packet 316
> 2011-02-11 19:37:18.337857 [NOTICE] libdingaling.c:1368 SecSEND:
> -------------------------------------------------------------------------------
> <iq type="result"
> to="+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1"
> from="husodesar at gmail.com/gtalk94609708"
> id="jingle:10.229.92.226-10513507:1:0290B654">
>   <query></query>
> </iq>
> 2011-02-11 19:37:18.337857 [DEBUG] libdingaling.c:1450 Processing 2 packets
> in retry queue
> 2011-02-11 19:37:18.337857 [DEBUG] libdingaling.c:1475 Discarding packet 316
> 2011-02-11 19:37:18.337857 [DEBUG] libdingaling.c:1467 Sending packet 317 (2
> left)
> 2011-02-11 19:37:18.337857 [NOTICE] libdingaling.c:1368 SecSEND:
> -------------------------------------------------------------------------------
> <iq xmlns="jabber:client" from="husodesar at gmail.com/gtalk94609708"
> to="+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1"
> type="set" id="317">
>   <ses:session xmlns:ses="http://www.google.com/session"
> type="transport-info" id="1162869130"
> initiator="husodesar at gmail.com/gtalk94609708">
>     <transport xmlns="http://www.google.com/transport/p2p">
>       <candidate name="rtp" address="184.105.153.247" port="24320"
> username="JwkGkXBkIuntRST4" password="JwkGkXBkIuntRST4" preference="1.0"
> protocol="udp" type="local" network="0" generation="0"></candidate>
>     </transport>
>   </ses:session>
> </iq>
> 2011-02-11 19:37:18.437876 [DEBUG] libdingaling.c:1450 Processing 1 packets
> in retry queue
> 2011-02-11 19:37:18.481871 [INFO] libdingaling.c:1366 SecRECV:
> -------------------------------------------------------------------------------
> <iq to="husodesar at gmail.com/gtalk94609708"
> from="+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1"
> id="317" type="result"></iq>
> 2011-02-11 19:37:18.481871 [DEBUG] libdingaling.c:968 Cancel packet 317
> 2011-02-11 19:37:18.537869 [DEBUG] libdingaling.c:1450 Processing 1 packets
> in retry queue
> 2011-02-11 19:37:18.537869 [DEBUG] libdingaling.c:1475 Discarding packet 317
> 2011-02-11 19:37:18.585873 [INFO] libdingaling.c:1366 SecRECV:
> -------------------------------------------------------------------------------
> <iq from="+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1"
> to="husodesar at gmail.com/gtalk94609708"
> id="jingle:10.229.92.226-10513507:1:0290B656" type="set">
>   <ses:session type="transport-info" id="1162869130"
> initiator="husodesar at gmail.com/gtalk94609708"
> xmlns:ses="http://www.google.com/session">
>     <transport xmlns="http://www.google.com/transport/p2p">
>       <candidate name="rtp" address="74.125.127.126" port="19295"
> username="wCd9MeksClqQziAR" preference="1.0" protocol="udp"
> network="mediaproxy" generation="0"></candidate>
>       <candidate name="rtp" address="74.125.127.126" port="19294"
> username="wCd9MeksClqQziAR" preference="0.6" protocol="tcp"
> network="mediaproxy" generation="0"></candidate>
>       <candidate name="rtp" address="74.125.127.126" port="443"
> username="wCd9MeksClqQziAR" preference="0.5" protocol="ssltcp"
> network="mediaproxy" generation="0"></candidate>
>     </transport>
>   </ses:session>
> </iq>
> 2011-02-11 19:37:18.585873 [DEBUG] libdingaling.c:383 Message for Session
> 1162869130
> 2011-02-11 19:37:18.585873 [DEBUG] libdingaling.c:528 New Candidate 1
> name=rtp
> type=stun
> protocol=udp
> username=wCd9MeksClqQziAR
> password=(null)
> address=74.125.127.126
> port=19295
> pref=1.00
> 2011-02-11 19:37:18.585873 [DEBUG] libdingaling.c:528 New Candidate 2
> name=rtp
> type=stun
> protocol=tcp
> username=wCd9MeksClqQziAR
> password=(null)
> address=74.125.127.126
> port=19294
> pref=0.60
> 2011-02-11 19:37:18.585873 [DEBUG] libdingaling.c:528 New Candidate 3
> name=rtp
> type=stun
> protocol=ssltcp
> username=wCd9MeksClqQziAR
> password=(null)
> address=74.125.127.126
> port=443
> pref=0.50
> 2011-02-11 19:37:18.585873 [DEBUG] mod_dingaling.c:2941 using Existing
> session for 1162869130
> 2011-02-11 19:37:18.585873 [DEBUG] mod_dingaling.c:3279 3 candidates
> 2011-02-11 19:37:18.585873 [DEBUG] mod_dingaling.c:3315 candidates
> 74.125.127.126:19295
> 2011-02-11 19:37:18.585873 [DEBUG] mod_dingaling.c:3315 candidates
> 74.125.127.126:19294
> 2011-02-11 19:37:18.585873 [DEBUG] mod_dingaling.c:3315 candidates
> 74.125.127.126:443
> 2011-02-11 19:37:18.585873 [DEBUG] mod_dingaling.c:3351 Acceptable Candidate
> 74.125.127.126:19295
> 2011-02-11 19:37:18.637916 [NOTICE] libdingaling.c:1368 SecSEND:
> -------------------------------------------------------------------------------
> <iq type="result"
> to="+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1"
> from="husodesar at gmail.com/gtalk94609708"
> id="jingle:10.229.92.226-10513507:1:0290B656">
>   <query></query>
> </iq>
> 2011-02-11 19:37:21.542056 [INFO] libdingaling.c:1366 SecRECV:
> -------------------------------------------------------------------------------
> <iq from="+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1"
> to="husodesar at gmail.com/gtalk94609708"
> id="jingle:10.229.92.226-10513507:1:0290B659" type="set">
>   <ses:session type="accept" id="1162869130"
> initiator="husodesar at gmail.com/gtalk94609708"
> xmlns:ses="http://www.google.com/session">
>     <pho:description xmlns:pho="http://www.google.com/session/phone">
>       <pho:payload-type id="0" name="PCMU"
> clockrate="8000"></pho:payload-type>
>       <pho:payload-type id="101" name="telephone-event"></pho:payload-type>
>     </pho:description>
>   </ses:session>
> </iq>
> 2011-02-11 19:37:21.542056 [DEBUG] libdingaling.c:383 Message for Session
> 1162869130
> 2011-02-11 19:37:21.542056 [DEBUG] libdingaling.c:440 Add Payload [PCMU]
> id='0'
> 2011-02-11 19:37:21.542056 [DEBUG] libdingaling.c:440 Add Payload
> [telephone-event] id='101'
> 2011-02-11 19:37:21.542056 [DEBUG] mod_dingaling.c:2941 using Existing
> session for 1162869130
> 2011-02-11 19:37:21.542056 [DEBUG] mod_dingaling.c:3193 Already decided on a
> codec
> 2011-02-11 19:37:21.542056 [NOTICE] libdingaling.c:1368 SecSEND:
> -------------------------------------------------------------------------------
> <iq type="result"
> to="+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1"
> from="husodesar at gmail.com/gtalk94609708"
> id="jingle:10.229.92.226-10513507:1:0290B659">
>   <query></query>
> </iq>
> 2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:865 Set Read Codec to
> PCMU at 8000
> 2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:880 Set Write Codec to
> PCMU at 8000
> 2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:892 SETUP RTP
> 184.105.153.247:0 -> 74.125.127.126:19295
> 2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:914 RTP ERROR Missing
> local port
> 2011-02-11 19:37:21.546355 [DEBUG] switch_channel.c:2538
> (dingaling/gv1/+18009993355 at voice.google.com) Callstate Change DOWN ->
> HANGUP
> 2011-02-11 19:37:21.546355 [NOTICE] mod_dingaling.c:915 Hangup
> dingaling/gv1/+18009993355 at voice.google.com [CS_INIT]
> [DESTINATION_OUT_OF_ORDER]
> 2011-02-11 19:37:21.546355 [DEBUG] switch_channel.c:2554 Send signal
> dingaling/gv1/+18009993355 at voice.google.com [KILL]
> 2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:1348
> dingaling/gv1/+18009993355 at voice.google.com CHANNEL KILL
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_session.c:1116 Send signal
> dingaling/gv1/+18009993355 at voice.google.com [BREAK]
> 2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:1348
> dingaling/gv1/+18009993355 at voice.google.com CHANNEL KILL
> 2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:710 Terminate called from
> line 1177 state=CS_HANGUP
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:356
> (dingaling/gv1/+18009993355 at voice.google.com) State INIT going to sleep
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:320
> (dingaling/gv1/+18009993355 at voice.google.com) Running State Change CS_HANGUP
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:557
> (dingaling/gv1/+18009993355 at voice.google.com) State HANGUP
> 2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:1317
> dingaling/gv1/+18009993355 at voice.google.com CHANNEL HANGUP
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:46
> dingaling/gv1/+18009993355 at voice.google.com Standard HANGUP, cause:
> DESTINATION_OUT_OF_ORDER
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:557
> (dingaling/gv1/+18009993355 at voice.google.com) State HANGUP going to sleep
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:351
> (dingaling/gv1/+18009993355 at voice.google.com) State Change CS_HANGUP ->
> CS_REPORTING
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_session.c:1116 Send signal
> dingaling/gv1/+18009993355 at voice.google.com [BREAK]
> 2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:1348
> dingaling/gv1/+18009993355 at voice.google.com CHANNEL KILL
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:320
> (dingaling/gv1/+18009993355 at voice.google.com) Running State Change
> CS_REPORTING
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:617
> (dingaling/gv1/+18009993355 at voice.google.com) State REPORTING
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:53
> dingaling/gv1/+18009993355 at voice.google.com Standard REPORTING, cause:
> DESTINATION_OUT_OF_ORDER
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:617
> (dingaling/gv1/+18009993355 at voice.google.com) State REPORTING going to sleep
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_state_machine.c:345
> (dingaling/gv1/+18009993355 at voice.google.com) State Change CS_REPORTING ->
> CS_DESTROY
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_session.c:1116 Send signal
> dingaling/gv1/+18009993355 at voice.google.com [BREAK]
> 2011-02-11 19:37:21.546355 [DEBUG] mod_dingaling.c:1348
> dingaling/gv1/+18009993355 at voice.google.com CHANNEL KILL
> 2011-02-11 19:37:21.546355 [DEBUG] switch_core_session.c:1288 Session 12
> (dingaling/gv1/+18009993355 at voice.google.com) Locked, Waiting on external
> entities
> 2011-02-11 19:37:21.550059 [DEBUG] switch_ivr_originate.c:3502 Originate
> Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
> 2011-02-11 19:37:21.550059 [INFO] mod_dptools.c:2623 Originate Failed.
> Cause: DESTINATION_OUT_OF_ORDER
> 2011-02-11 19:37:21.550059 [DEBUG] switch_channel.c:2538
> (sofia/internal/1000 at 184.105.153.247) Callstate Change RINGING -> HANGUP
> 2011-02-11 19:37:21.550059 [NOTICE] mod_dptools.c:2686 Hangup
> sofia/internal/1000 at 184.105.153.247 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
> 2011-02-11 19:37:21.550059 [DEBUG] switch_channel.c:2554 Send signal
> sofia/internal/1000 at 184.105.153.247 [KILL]
> 2011-02-11 19:37:21.550059 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/internal/1000 at 184.105.153.247 [BREAK]
> 2011-02-11 19:37:21.550059 [DEBUG] switch_core_session.c:2060
> sofia/internal/1000 at 184.105.153.247 skip receive message
> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 2011-02-11 19:37:21.550059 [DEBUG] switch_core_state_machine.c:366
> (sofia/internal/1000 at 184.105.153.247) State EXECUTE going to sleep
> 2011-02-11 19:37:21.550059 [DEBUG] switch_core_state_machine.c:320
> (sofia/internal/1000 at 184.105.153.247) Running State Change CS_HANGUP
> 2011-02-11 19:37:21.550059 [DEBUG] switch_core_state_machine.c:557
> (sofia/internal/1000 at 184.105.153.247) State HANGUP
> 2011-02-11 19:37:21.550059 [DEBUG] mod_sofia.c:457 Channel
> sofia/internal/1000 at 184.105.153.247 hanging up, cause:
> DESTINATION_OUT_OF_ORDER
> 2011-02-11 19:37:21.550059 [NOTICE] switch_core_session.c:1306 Session 12
> (dingaling/gv1/+18009993355 at voice.google.com) Ended
> 2011-02-11 19:37:21.550059 [NOTICE] switch_core_session.c:1308 Close Channel
> dingaling/gv1/+18009993355 at voice.google.com [CS_DESTROY]
> 2011-02-11 19:37:21.550059 [DEBUG] switch_core_state_machine.c:449
> (dingaling/gv1/+18009993355 at voice.google.com) Callstate Change HANGUP ->
> DOWN
> 2011-02-11 19:37:21.550059 [DEBUG] switch_core_state_machine.c:452
> (dingaling/gv1/+18009993355 at voice.google.com) Running State Change
> CS_DESTROY
> 2011-02-11 19:37:21.550059 [DEBUG] switch_core_state_machine.c:462
> (dingaling/gv1/+18009993355 at voice.google.com) State DESTROY
> 2011-02-11 19:37:21.550059 [DEBUG] libdingaling.c:299 Destroyed Session
> 1162869130
> 2011-02-11 19:37:21.550059 [DEBUG] switch_core_state_machine.c:60
> dingaling/gv1/+18009993355 at voice.google.com Standard DESTROY
> 2011-02-11 19:37:21.550059 [DEBUG] switch_core_state_machine.c:462
> (dingaling/gv1/+18009993355 at voice.google.com) State DESTROY going to sleep
> 2011-02-11 19:37:21.562076 [DEBUG] mod_sofia.c:519 Responding to INVITE
> with: 502
> 2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/1000 at 184.105.153.247 Standard HANGUP, cause:
> DESTINATION_OUT_OF_ORDER
> 2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:557
> (sofia/internal/1000 at 184.105.153.247) State HANGUP going to sleep
> 2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:351
> (sofia/internal/1000 at 184.105.153.247) State Change CS_HANGUP -> CS_REPORTING
> 2011-02-11 19:37:21.562076 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/internal/1000 at 184.105.153.247 [BREAK]
> 2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:320
> (sofia/internal/1000 at 184.105.153.247) Running State Change CS_REPORTING
> 2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:617
> (sofia/internal/1000 at 184.105.153.247) State REPORTING
> 2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/1000 at 184.105.153.247 Standard REPORTING, cause:
> DESTINATION_OUT_OF_ORDER
> 2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:617
> (sofia/internal/1000 at 184.105.153.247) State REPORTING going to sleep
> 2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:345
> (sofia/internal/1000 at 184.105.153.247) State Change CS_REPORTING ->
> CS_DESTROY
> 2011-02-11 19:37:21.562076 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/internal/1000 at 184.105.153.247 [BREAK]
> 2011-02-11 19:37:21.562076 [DEBUG] switch_core_session.c:1288 Session 11
> (sofia/internal/1000 at 184.105.153.247) Locked, Waiting on external entities
> 2011-02-11 19:37:21.562076 [NOTICE] switch_core_session.c:1306 Session 11
> (sofia/internal/1000 at 184.105.153.247) Ended
> 2011-02-11 19:37:21.562076 [NOTICE] switch_core_session.c:1308 Close Channel
> sofia/internal/1000 at 184.105.153.247 [CS_DESTROY]
> 2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:449
> (sofia/internal/1000 at 184.105.153.247) Callstate Change HANGUP -> DOWN
> 2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:452
> (sofia/internal/1000 at 184.105.153.247) Running State Change CS_DESTROY
> 2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:462
> (sofia/internal/1000 at 184.105.153.247) State DESTROY
> 2011-02-11 19:37:21.562076 [DEBUG] mod_sofia.c:362
> sofia/internal/1000 at 184.105.153.247 SOFIA DESTROY
> 2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/1000 at 184.105.153.247 Standard DESTROY
> 2011-02-11 19:37:21.562076 [DEBUG] switch_core_state_machine.c:462
> (sofia/internal/1000 at 184.105.153.247) State DESTROY going to sleep
> 2011-02-11 19:37:48.963766 [INFO] libdingaling.c:1366 SecRECV:
> -------------------------------------------------------------------------------
> <iq from="+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1"
> to="husodesar at gmail.com/gtalk94609708"
> id="jingle:10.229.92.226-10513507:1:0290B66D" type="set">
>   <ses:session type="terminate" id="1162869130"
> initiator="husodesar at gmail.com/gtalk94609708"
> xmlns:ses="http://www.google.com/session">
>     <pho:internal-server-error
> xmlns:pho="http://www.google.com/session/phone">Media
> terminated</pho:internal-server-error>
>   </ses:session>
> </iq>
> 2011-02-11 19:37:48.963766 [DEBUG] libdingaling.c:355 Created Session
> 1162869130
> 2011-02-11 19:37:48.963766 [DEBUG] libdingaling.c:383 Message for Session
> 1162869130
> 2011-02-11 19:37:48.963766 [DEBUG] mod_dingaling.c:2951 Session is already
> dead
> 2011-02-11 19:37:49.055819 [NOTICE] libdingaling.c:1368 SecSEND:
> -------------------------------------------------------------------------------
> <iq type="result"
> to="+18009993355 at voice.google.com/srvres-MTAuMjI5LjkyLjIyNjo5ODc1"
> from="husodesar at gmail.com/gtalk94609708"
> id="jingle:10.229.92.226-10513507:1:0290B66D">
>   <query></query>
> </iq>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb 11 15:04:16 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 11 Feb 2011 06:04:16 -0600
Subject: [Freeswitch-users] Using an Empty FROM field for SIP Messages
In-Reply-To: <6CE2C3C3-0EAF-43F5-B835-5A954F194ED3@ipeva.fr>
References: <AANLkTikEyL=NpmdqM+Vx=rxaM2+zEyxuJr_oGdFDWLwU@mail.gmail.com>
	<D810E0FC-0474-45E8-9C9A-C8D9D438032C@ipeva.fr>
	<AANLkTikbncOB_=Q4hEc2gg8A+bS59CN0=RXjUdAdT0az@mail.gmail.com>
	<6CE2C3C3-0EAF-43F5-B835-5A954F194ED3@ipeva.fr>
Message-ID: <AANLkTin7tStk2f=dJv-P8kRyU+upvmUfmW2Y1DRSx8QG@mail.gmail.com>

set it to _undef_ for blank in the sip packet.
one of many hacks for stupid behaviors in commercial products.



On Fri, Feb 11, 2011 at 5:37 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Matthew,
>
> I think I forgot to tell you something.
> If you are sending the call to an outbound gateway, you need to add this in
> the gateway parameter:
>  <param name="caller-id-in-from" value="true"/>
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 11/02/2011 ? 12:06, Matthew Fong a ?crit :
>
> effective_caller_id_number doesn't seem to work. I was able to get
>
> From: "" <sip:@0.0.0.0>;tag=j0yN36H8tZgDe
>
> using sip_from_uri='<sip:@X.X.X.X>' but there are still the proceeding ""
>
> Does anyone else have any suggestions?
>
> Thanks
> --matt
> http://www.hellohunter.com
> Hosted Dialer
> On Wed, Feb 9, 2011 at 2:06 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>
>> Try setting effective_caller_id_number to " "
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 09/02/2011 ? 10:55, Matthew Fong a ?crit :
>>
>> I am trying to setup a customer who wanted to utilize the airespring
>> national did presence, where in they rewrite the Caller ID of outbound
>> calls. In order to do this tho, they say I need to send a blank caller id.
>> FreeSWITCH will send 0000000000 if the caller id is not specified.
>> Airespring wants something formatted like
>>
>> INVITE sip:jungar at alpha-org.com SIP/2.0
>>
>> Via: SIP/2.0/UDP zetamachine.beta-org.com:5060;branch=as82je8ei4kr
>>
>> Max-Forwards: 70
>>
>> To:  2556112121 <sip:jungar at alpha-org.com>
>>
>> From: <sip:jsmith at beta-org.com>;tag=13456
>>
>> Call-ID: @zetamachine.beta-org.com
>>
>>
>> with the From: field looking like any of the following
>>
>> From: <sip:jsmith at beta-org.com>;tag=13456
>> From: <  @208.76.54.59>;tag=13456
>> From:  @208.76.54.59;tag=13456
>>
>> The closest I got was to make FreeSWITCH to the following, but airespring
>> won't budge. Can anyone tell me how to make the From string as airespring
>> wants for their national DID presence service. Thanks.
>> From: "" <sip:0000000000 at 208.76.54.59>;tag=UyNBXjyUUay8m
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 br at bsdpad.com  Fri Feb 11 16:02:09 2011
From: br at bsdpad.com (Ruslan Bukin)
Date: Fri, 11 Feb 2011 16:02:09 +0300
Subject: [Freeswitch-users] dp_tools/set- not worked SOMETIMES
In-Reply-To: <AANLkTim=+R9snJzYiCdUNZiixxOcOy-6C2j=SjpzsBSN@mail.gmail.com>
References: <20110211112453.GA700@bsdjail.com>
	<20110211114004.GA1491@bsdjail.com>
	<AANLkTim=+R9snJzYiCdUNZiixxOcOy-6C2j=SjpzsBSN@mail.gmail.com>
Message-ID: <20110211130209.GA4821@bsdjail.com>

I'm using not changed git head and freebsd amd64 only
(checked on two different fbsd servers -problem exists in both)

-Ruslan

On Fri, Feb 11, 2011 at 06:00:16AM -0600, Anthony Minessale wrote:
> This is your second strange issue. Maybe you have some problem with
> your platform.
> Are you using unaltered GIT HEAD?  Have you compared results on other
> platforms like Linux 64 bit or Windows?
> 
> 
> On Fri, Feb 11, 2011 at 5:40 AM, Ruslan Bukin <br at bsdpad.com> wrote:
> > correction: it seems that parameters always exists after bridge,
> > but sometimes one of them not exists in CDR result
> > (I have checked with mod_xml_cdr and mod_json_cdr)
> >
> > -Ruslan
> >
> > On Fri, Feb 11, 2011 at 02:24:53PM +0300, Ruslan Bukin wrote:
> >> Application set is not set parameters SOMETIMES
> >>
> >> for example, before bridge I set my static parameters:
> >>
> >> <action application="set" data="my_own_parameter=test"/>
> >> <action application="set" data="nibble_rate=1"/>
> >> ...
> >>
> >> sometimes (in about ~1-5% times) one of them is not set up.
> >>
> >> I tried to set like this before bridge:
> >>
> >> ?<extension name="set_var" continue="true">
> >> ? ?<condition field="${my_own_parameter}" expression=".+" break="on-true"/>
> >> ? ?<condition field="destination_number" expression="(777)" break="on-true">
> >> ? ? ?<action application="set" data="my_own_parameter=test"/>
> >> ? ? ?<action application="transfer" data="$1 XML public"/>
> >> ? ?</condition>
> >> ?</extension>
> >>
> >> but no result: after bridge no parameter to read (SOMETIMES)
> >>
> >>
> >> any recommendations?
> >>
> >> -Ruslan
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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  Fri Feb 11 16:11:36 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 11 Feb 2011 07:11:36 -0600
Subject: [Freeswitch-users] dp_tools/set- not worked SOMETIMES
In-Reply-To: <20110211130209.GA4821@bsdjail.com>
References: <20110211112453.GA700@bsdjail.com>
	<20110211114004.GA1491@bsdjail.com>
	<AANLkTim=+R9snJzYiCdUNZiixxOcOy-6C2j=SjpzsBSN@mail.gmail.com>
	<20110211130209.GA4821@bsdjail.com>
Message-ID: <AANLkTi=uc-bprji8C82T=xukDZ3PEP3yHj=h-ro488wU@mail.gmail.com>

Is it reproducable enough to provide a recipe so someone can test it on
linux.  Its possibly only a fbsd issue since I have not heard this ever
before.  Is it the latest freebsd since they stopped using their own thread
lib?
On Feb 11, 2011 7:03 AM, "Ruslan Bukin" <br at bsdpad.com> wrote:
>
> I'm using not changed git head and freebsd amd64 only
> (checked on two different fbsd servers -problem exists in both)
>
> -Ruslan
>
> On Fri, Feb 11, 2011 at 06:00:16AM -0600, Anthony Minessale wrote:
> > This is your second strange issue. Maybe you have some problem with
> > your platform.
> > Are you using unaltered GIT HEAD?  Have you compared results on other
> > platforms like Linux 64 bit or Windows?
> >
> >
> > On Fri, Feb 11, 2011 at 5:40 AM, Ruslan Bukin <br at bsdpad.com> wrote:
> > > correction: it seems that parameters always exists after bridge,
> > > but sometimes one of them not exists in CDR result
> > > (I have checked with mod_xml_cdr and mod_json_cdr)
> > >
> > > -Ruslan
> > >
> > > On Fri, Feb 11, 2011 at 02:24:53PM +0300, Ruslan Bukin wrote:
> > >> Application set is not set parameters SOMETIMES
> > >>
> > >> for example, before bridge I set my static parameters:
> > >>
> > >> <action application="set" data="my_own_parameter=test"/>
> > >> <action application="set" data="nibble_rate=1"/>
> > >> ...
> > >>
> > >> sometimes (in about ~1-5% times) one of them is not set up.
> > >>
> > >> I tried to set like this before bridge:
> > >>
> > >>  <extension name="set_var" continue="true">
> > >>    <condition field="${my_own_parameter}" expression=".+"
break="on-true"/>
> > >>    <condition field="destination_number" expression="(777)"
break="on-true">
> > >>      <action application="set" data="my_own_parameter=test"/>
> > >>      <action application="transfer" data="$1 XML public"/>
> > >>    </condition>
> > >>  </extension>
> > >>
> > >> but no result: after bridge no parameter to read (SOMETIMES)
> > >>
> > >>
> > >> any recommendations?
> > >>
> > >> -Ruslan
> > >>
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at 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 kbdfck at gmail.com  Fri Feb 11 16:22:33 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Fri, 11 Feb 2011 16:22:33 +0300
Subject: [Freeswitch-users] hold_music var gets unset/lost by
 cancelled/failed att_xfer
In-Reply-To: <AANLkTikKt8TC-g3Q5LpXEZh62dGY6+f=HRO00n4G4PU3@mail.gmail.com>
References: <AANLkTimLod3yHrex4LAK=4gdVcPhzXutKYyw=E8ZMcN9@mail.gmail.com>
	<AANLkTi=u3VeHPuS5x1e+yu5QNBw5ZSgHvAWOT+F6A_OR@mail.gmail.com>
	<AANLkTi=EndkiUJvCgCSirxCA23EwWD_++0h3Qcr=mg9g@mail.gmail.com>
	<AANLkTikt97dugbhKPgnB2ws4bOFcxg8yhzSdyxHMab_m@mail.gmail.com>
	<AANLkTikEx9ZwNADRQ8k2JbTwgoLPFmHUPpPL3RLnZZU8@mail.gmail.com>
	<AANLkTinS3wxA8izr9tXX7jhYJPHq_X_coO7UNoUkjs-P@mail.gmail.com>
	<AANLkTi=Hv7kkYeD8TQeCZtfvmzqaCx+xQ=yYC16kpNfa@mail.gmail.com>
	<AANLkTikKt8TC-g3Q5LpXEZh62dGY6+f=HRO00n4G4PU3@mail.gmail.com>
Message-ID: <AANLkTimjWc_Tn2Du0w_Zgy2VihsN8Av40M1UAQV7NXtT@mail.gmail.com>

Maybe you are right, but then tell me the right way :) att_xfer uses
loopback by itself, and it doesn't work without tricks with not well
documented loopback_bowout/bowout_on_execute variables... This
behavior is not documented anywhere, but I can't propose wiki updates
until I get it working or just give up with att_xfer in FS at all :)

All I need is just working in-call att_xfer with Sofia channels :( SIP
transfers are cool, but  I need in-call DTMF triggered transfers :((

Is there a developer documentation about FS basic work principles?
I'll try to dive in the code :))

2011/2/11 Anthony Minessale <anthony.minessale at gmail.com>:
> I reverted the patch, I guess it doesn't work based on your feedback.
> We will leave well enough alone when you said in your last email
> everything was working.
>
> There is a time when maybe you are taking things too far trying to
> solve things a silly way and maybe you should consider nicer
> equipment.
>
>
> On Fri, Feb 11, 2011 at 4:43 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>> I see your patch in git log, is it enough to do 'make current', or I
>> should apply it in some special way? After make current issues are
>> still present...
>>
>> What should i set loopback_bowout/bowout_on_execute variables to? I
>> tried with false, this works as in previous versions, and loopback
>> stays in path during the call. While loopback channels are present in
>> path, there are issues with sound quality, there is robot voice effect
>> for short periods of time. You mentioned setting rtp timer to none,
>> where this should be done? In loopback channel execute_extension while
>> doing att_xfer?
>>
>> When bowout variables set to true ?att_xfer doesn't work just as I
>> described in my previous posts, and there is also issue with MOH -
>> after transfer target answers to transferor, MOH is stoped for
>> transferee. Maybe this can help somehow.
>>
>>
>>
>>
>> 2011/2/10 Anthony Minessale <anthony.minessale at gmail.com>:
>>> I pushed a patch that will probably delay the bowout until after the
>>> att_xfer is over
>>> Give it a try.
>>>
>>> commit 3546654615f88058fb6769fe79e07162602fa4af
>>> Author: Anthony Minessale <anthm at freeswitch.org>
>>> Date: ? Thu Feb 10 12:37:14 2011 -0600
>>>
>>> ? ?don't bow out on att_xfer bridge
>>>
>>>
>>> On Thu, Feb 10, 2011 at 1:03 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>>> Thanks! It works now! BTW, att_xfer with
>>>> loopback_bowout/bowout_on_execute set to false seems to be working
>>>> too. Is there a way to make loopback channel leave the path? Without
>>>> bowout turned off att_xfer doesn't work...
>>>>
>>>> 2011/2/8 Anthony Minessale <anthony.minessale at gmail.com>:
>>>>> try latest GIT
>>>>>
>>>>> On Tue, Feb 8, 2011 at 1:28 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>>>>> I have same issue with MOH and att_xfer on failed transfers, music on
>>>>>> hold played only once
>>>>>> At the same time, transfer_ringback always plays correctly to transferer
>>>>>>
>>>>>> 2011/2/8 Anthony Minessale <anthony.minessale at gmail.com>:
>>>>>>> you really should report this to jira not to the mailing list.
>>>>>>> http://jira.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Mon, Feb 7, 2011 at 4:23 PM, Fraser Redmond <fraserredmond at gmail.com> wrote:
>>>>>>>> I'm trying to do a second att_xfer on a call so that if the first attended
>>>>>>>> transfer fails (c-leg is busy, or presses do-not-answer, or is an extn that
>>>>>>>> doesn't exist) then the call could be transferred to someone else.
>>>>>>>>
>>>>>>>> On the first att_xfer the person on hold hears the hold_music correctly.
>>>>>>>> Once that transfer is cancelled or fails:
>>>>>>>> -- On any subsequent att_xfer's the person on hold just hears silence.
>>>>>>>> -- If they are put on hold they just hear silence.
>>>>>>>>
>>>>>>>> I tried setting hold_music again for each channel after the first att_xfer,
>>>>>>>> but that didn't work, so it's probably not actually a problem with
>>>>>>>> hold_music per se, but some other variable/setting that decides whether to
>>>>>>>> use hold_music.
>>>>>>>>
>>>>>>>> I also tried doing a uuid_dump before and after each attempt, but didn't
>>>>>>>> notice anything too different - unless it's a matter of unsetting one of the
>>>>>>>> couple of changed/new vars like:
>>>>>>>> variable_originate_disposition
>>>>>>>> variable_current_application
>>>>>>>> variable_playback_seconds
>>>>>>>>
>>>>>>>> I get the feeling other variables are probably also lost by the first failed
>>>>>>>> transfer as the second att_xfer has some odd things happen if the third
>>>>>>>> party does answer. Haven't been able to narrow it down as closely as the
>>>>>>>> hold_music, but two things I've seen happen are:
>>>>>>>> -- The party that initiated the transfer gets hung up automatically (after
>>>>>>>> 30 sec)
>>>>>>>> -- When the party that initiated the transfer hangs up it should connect the
>>>>>>>> other two parties, but instead it hung up all three
>>>>>>>>
>>>>>>>> Cheers,
>>>>>>>> Fraser
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at 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
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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


From gchen00 at insightbb.com  Fri Feb 11 17:51:30 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Fri, 11 Feb 2011 09:51:30 -0500
Subject: [Freeswitch-users] 503 Maximum Calls In Progress
Message-ID: <fc78b8a07f3e.4d5506a2@insightbb.com>

FreeSWITCH Version 1.0.7 (hacked-20110119T213949Z)

I have a freeswitch installed for testing. It just up for about an hour and there are two Cisco 7960 phones registered. After made several test calls between these two Cisco phones, FS stop working and give me these message on fs_cli:
2011-02-11 17:39:58.691008 [CRIT] switch_core_session.c:1629 The system cannot create any sessions at this time.

In the SIP header it shows:?503 Maximum Calls In Progress

What should I do to fix this? Here is the message by press F2, F3 and F4:
F2:
freeswitch at internal> status
UP 0 years, 0 days, 1 hour, 12 minutes, 55 seconds, 590 milliseconds, 20 microseconds
39 session(s) since startup
0 session(s) 0/30
1000 session(s) max
min idle cpu 0.00/100.00
F3:
freeswitch at internal> show channels
0 total.

F4:
freeswitch at internal> show calls
0 total.

Gary?


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From peter.olsson at visionutveckling.se  Fri Feb 11 18:11:42 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 11 Feb 2011 16:11:42 +0100
Subject: [Freeswitch-users] 503 Maximum Calls In Progress
In-Reply-To: <fc78b8a07f3e.4d5506a2@insightbb.com>
References: <fc78b8a07f3e.4d5506a2@insightbb.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58ADD56FBD@cooper>

Update to latest git. I think Tony did an update for this, it was in one of the config files.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Gary Chen
Skickat: den 11 februari 2011 15:52
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] 503 Maximum Calls In Progress

FreeSWITCH Version 1.0.7 (hacked-20110119T213949Z)

I have a freeswitch installed for testing. It just up for about an hour and there are two Cisco 7960 phones registered. After made several test calls between these two Cisco phones, FS stop working and give me these message on fs_cli:
2011-02-11 17:39:58.691008 [CRIT] switch_core_session.c:1629 The system cannot create any sessions at this time.

In the SIP header it shows: 503 Maximum Calls In Progress

What should I do to fix this? Here is the message by press F2, F3 and F4:
F2:
freeswitch at internal> status
UP 0 years, 0 days, 1 hour, 12 minutes, 55 seconds, 590 milliseconds, 20 microseconds
39 session(s) since startup
0 session(s) 0/30
1000 session(s) max
min idle cpu 0.00/100.00
F3:
freeswitch at internal> show channels
0 total.

F4:
freeswitch at internal> show calls
0 total.

Gary

!DSPAM:4d554ec432767061315535!
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From u2nsam at gmail.com  Fri Feb 11 18:16:42 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 11 Feb 2011 20:46:42 +0530
Subject: [Freeswitch-users] no voice for call in same lan network
Message-ID: <AANLkTing0KFwW9LqmkaNFqDeEF70O0MVSLtoFjVGTJKA@mail.gmail.com>

Here ,

i have bunch extension in the same lan network and when they talk with each
other there is no voice ,
i have another bunch in other lan network the too do not get voice when
talking with each other.
Here the public ip for 2 bunches are different and when an extension from
bunch A calls to extension
of bunch B then they have voice.
FS is on public IP   :(

>From the trace i could see that when making an internal call in the same LAN
the RTP flows to private IP
and when making and external call from bunch A to bunch B the RTP is send to
public IP where the voice is heard,
also while making and outbound call the RTP is send to public IP of the LAN
and there is voice.

Is this a known scenario to anybody ?

Regards
Sam
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From gchen00 at insightbb.com  Fri Feb 11 18:28:11 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Fri, 11 Feb 2011 10:28:11 -0500
Subject: [Freeswitch-users] 503 Maximum Calls In Progress
Message-ID: <fc808c5b1180.4d550f3b@insightbb.com>

?My company blocked Git access.

Can you tell me which config file need to be changed to fix the problem?

Gary

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson
Sent: Friday, February 11, 2011 10:12 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] 503 Maximum Calls In Progress

?

Update to latest git. I think Tony did an update for this, it was in one of the config files.

?

/Peter

?

?

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Gary Chen
Skickat: den 11 februari 2011 15:52
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] 503 Maximum Calls In Progress

?

FreeSWITCH Version 1.0.7 (hacked-20110119T213949Z)

?

I have a freeswitch installed for testing. It just up for about an hour and there are two Cisco 7960 phones registered. After made several test calls between these two Cisco phones, FS stop working and give me these message on fs_cli:

2011-02-11 17:39:58.691008 [CRIT] switch_core_session.c:1629 The system cannot create any sessions at this time.

?

In the SIP header it shows:?503 Maximum Calls In Progress

?

What should I do to fix this? Here is the message by press F2, F3 and F4:

F2:

freeswitch at internal> status

UP 0 years, 0 days, 1 hour, 12 minutes, 55 seconds, 590 milliseconds, 20 microseconds

39 session(s) since startup

0 session(s) 0/30

1000 session(s) max

min idle cpu 0.00/100.00

F3:

freeswitch at internal> show channels

0 total.

?

F4:

freeswitch at internal> show calls

0 total.

?

Gary?

?

!DSPAM:4d554ec432767061315535!?
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From peter.olsson at visionutveckling.se  Fri Feb 11 18:39:26 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 11 Feb 2011 16:39:26 +0100
Subject: [Freeswitch-users] 503 Maximum Calls In Progress
In-Reply-To: <fc808c5b1180.4d550f3b@insightbb.com>
References: <fc808c5b1180.4d550f3b@insightbb.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58ADD56FDF@cooper>

Comment from anthm;


edit conf/autoload_configs/switch.conf.xml



comment out



<param name="min-idle-cpu" value="25"/>



It seems to not work well on 32 bit.


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Gary Chen
Skickat: den 11 februari 2011 16:28
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] 503 Maximum Calls In Progress

 My company blocked Git access.
Can you tell me which config file need to be changed to fix the problem?
Gary
________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson
Sent: Friday, February 11, 2011 10:12 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] 503 Maximum Calls In Progress

Update to latest git. I think Tony did an update for this, it was in one of the config files.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Gary Chen
Skickat: den 11 februari 2011 15:52
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] 503 Maximum Calls In Progress

FreeSWITCH Version 1.0.7 (hacked-20110119T213949Z)

I have a freeswitch installed for testing. It just up for about an hour and there are two Cisco 7960 phones registered. After made several test calls between these two Cisco phones, FS stop working and give me these message on fs_cli:
2011-02-11 17:39:58.691008 [CRIT] switch_core_session.c:1629 The system cannot create any sessions at this time.

In the SIP header it shows: 503 Maximum Calls In Progress

What should I do to fix this? Here is the message by press F2, F3 and F4:
F2:
freeswitch at internal> status
UP 0 years, 0 days, 1 hour, 12 minutes, 55 seconds, 590 milliseconds, 20 microseconds
39 session(s) since startup
0 session(s) 0/30
1000 session(s) max
min idle cpu 0.00/100.00
F3:
freeswitch at internal> show channels
0 total.

F4:
freeswitch at internal> show calls
0 total.

Gary


!DSPAM:4d55571e32761412459665!
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From gchen00 at insightbb.com  Fri Feb 11 19:01:30 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Fri, 11 Feb 2011 11:01:30 -0500
Subject: [Freeswitch-users] 503 Maximum Calls In Progress
Message-ID: <fc5eb12a398e.4d55170a@insightbb.com>

?My switch.conf.xml does not have this param.

Gary

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson
Sent: Friday, February 11, 2011 10:39 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] 503 Maximum Calls In Progress

?

Comment from anthm;

?

edit conf/autoload_configs/switch.conf.xml

?

comment out

?

<param name="min-idle-cpu" value="25"/>

?

It seems to not work well on 32 bit.

?
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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110211/97f7c860/attachment.html 

From david.ponzone at ipeva.fr  Fri Feb 11 19:11:03 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 11 Feb 2011 17:11:03 +0100
Subject: [Freeswitch-users] 503 Maximum Calls In Progress
In-Reply-To: <fc5eb12a398e.4d55170a@insightbb.com>
References: <fc5eb12a398e.4d55170a@insightbb.com>
Message-ID: <6B4D073D-32AE-4E1D-B919-1169F29059C7@ipeva.fr>

So add it.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 11/02/2011 ? 17:01, Gary Chen a ?crit :

>  My switch.conf.xml does not have this param.
> 
> Gary
> 
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson
> Sent: Friday, February 11, 2011 10:39 AM
> To: 'FreeSWITCH Users Help'
> Subject: Re: [Freeswitch-users] 503 Maximum Calls In Progress
> 
>  
> 
> Comment from anthm;
> 
>  
> 
> edit conf/autoload_configs/switch.conf.xml
> 
>  
> 
> comment out
> 
>  
> 
> <param name="min-idle-cpu" value="25"/>
> 
>  
> 
> It seems to not work well on 32 bit.
> 
>  
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mattdfong at gmail.com  Fri Feb 11 19:11:10 2011
From: mattdfong at gmail.com (Matthew Fong)
Date: Fri, 11 Feb 2011 08:11:10 -0800
Subject: [Freeswitch-users] Using an Empty FROM field for SIP Messages
In-Reply-To: <AANLkTin7tStk2f=dJv-P8kRyU+upvmUfmW2Y1DRSx8QG@mail.gmail.com>
References: <AANLkTikEyL=NpmdqM+Vx=rxaM2+zEyxuJr_oGdFDWLwU@mail.gmail.com>
	<D810E0FC-0474-45E8-9C9A-C8D9D438032C@ipeva.fr>
	<AANLkTikbncOB_=Q4hEc2gg8A+bS59CN0=RXjUdAdT0az@mail.gmail.com>
	<6CE2C3C3-0EAF-43F5-B835-5A954F194ED3@ipeva.fr>
	<AANLkTin7tStk2f=dJv-P8kRyU+upvmUfmW2Y1DRSx8QG@mail.gmail.com>
Message-ID: <AANLkTiniQkwgfqZNHfaa_imuEGFDo3sEYbsKmkTLb5Yb@mail.gmail.com>

inside my conf/sip_profiles/external/airespring.xml file I set

<param name="caller-id-in-from" value="_undef_"/>

and restarted but

   ------------------------------------------------------------------------
   INVITE sip:14159927717 at 64.211.41.115 SIP/2.0
   Via: SIP/2.0/UDP 67.43.59.58:5080;rport;branch=z9hG4bK92ypSpD2Q6D6H
   Max-Forwards: 70
   From: "" <sip:@67.43.59.58>;tag=9tyvjr55tmyXe

is still the output of the INVITE

Airespring needs (without the "" after From:, but before <sip:...)

   ------------------------------------------------------------------------
   INVITE sip:14159927717 at 64.211.41.115 SIP/2.0
   Via: SIP/2.0/UDP 67.43.59.58:5080;rport;branch=z9hG4bK92ypSpD2Q6D6H
   Max-Forwards: 70
   From: <sip:@67.43.59.58>;tag=9tyvjr55tmyXe

I also tried setting sip_from_uri, effective_caller_id_number to _undef_ but
did not see a different result. Am I setting _undef_ in the correct places?
Thanks.

I am using the latest nightly snapshop.

--matt

On Fri, Feb 11, 2011 at 4:04 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> set it to _undef_ for blank in the sip packet.
> one of many hacks for stupid behaviors in commercial products.
>
>
>
> On Fri, Feb 11, 2011 at 5:37 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>
>> Matthew,
>>
>> I think I forgot to tell you something.
>> If you are sending the call to an outbound gateway, you need to add this
>> in the gateway parameter:
>>  <param name="caller-id-in-from" value="true"/>
>>
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 11/02/2011 ? 12:06, Matthew Fong a ?crit :
>>
>> effective_caller_id_number doesn't seem to work. I was able to get
>>
>> From: "" <sip:@0.0.0.0>;tag=j0yN36H8tZgDe
>>
>> using sip_from_uri='<sip:@X.X.X.X>' but there are still the proceeding ""
>>
>> Does anyone else have any suggestions?
>>
>> Thanks
>> --matt
>> http://www.hellohunter.com
>> Hosted Dialer
>> On Wed, Feb 9, 2011 at 2:06 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>>
>>> Try setting effective_caller_id_number to " "
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 09/02/2011 ? 10:55, Matthew Fong a ?crit :
>>>
>>> I am trying to setup a customer who wanted to utilize the airespring
>>> national did presence, where in they rewrite the Caller ID of outbound
>>> calls. In order to do this tho, they say I need to send a blank caller id.
>>> FreeSWITCH will send 0000000000 if the caller id is not specified.
>>> Airespring wants something formatted like
>>>
>>> INVITE sip:jungar at alpha-org.com SIP/2.0
>>>
>>> Via: SIP/2.0/UDP zetamachine.beta-org.com:5060;branch=as82je8ei4kr
>>>
>>> Max-Forwards: 70
>>>
>>> To:  2556112121 <sip:jungar at alpha-org.com>
>>>
>>> From: <sip:jsmith at beta-org.com>;tag=13456
>>>
>>> Call-ID: @zetamachine.beta-org.com
>>>
>>>
>>> with the From: field looking like any of the following
>>>
>>> From: <sip:jsmith at beta-org.com>;tag=13456
>>> From: <  @208.76.54.59>;tag=13456
>>> From:  @208.76.54.59;tag=13456
>>>
>>> The closest I got was to make FreeSWITCH to the following, but airespring
>>> won't budge. Can anyone tell me how to make the From string as airespring
>>> wants for their national DID presence service. Thanks.
>>> From: "" <sip:0000000000 at 208.76.54.59>;tag=UyNBXjyUUay8m
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 david.ponzone at ipeva.fr  Fri Feb 11 19:18:28 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 11 Feb 2011 17:18:28 +0100
Subject: [Freeswitch-users] Using an Empty FROM field for SIP Messages
In-Reply-To: <AANLkTiniQkwgfqZNHfaa_imuEGFDo3sEYbsKmkTLb5Yb@mail.gmail.com>
References: <AANLkTikEyL=NpmdqM+Vx=rxaM2+zEyxuJr_oGdFDWLwU@mail.gmail.com>
	<D810E0FC-0474-45E8-9C9A-C8D9D438032C@ipeva.fr>
	<AANLkTikbncOB_=Q4hEc2gg8A+bS59CN0=RXjUdAdT0az@mail.gmail.com>
	<6CE2C3C3-0EAF-43F5-B835-5A954F194ED3@ipeva.fr>
	<AANLkTin7tStk2f=dJv-P8kRyU+upvmUfmW2Y1DRSx8QG@mail.gmail.com>
	<AANLkTiniQkwgfqZNHfaa_imuEGFDo3sEYbsKmkTLb5Yb@mail.gmail.com>
Message-ID: <4430AE8C-0C39-43F4-A085-42A7AACFE0CC@ipeva.fr>

Set effective_caller_id_name to _undef_.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 11/02/2011 ? 17:11, Matthew Fong a ?crit :

> inside my conf/sip_profiles/external/airespring.xml file I set
> 
> <param name="caller-id-in-from" value="_undef_"/>
> 
> and restarted but 
> 
>    ------------------------------------------------------------------------
>    INVITE sip:14159927717 at 64.211.41.115 SIP/2.0
>    Via: SIP/2.0/UDP 67.43.59.58:5080;rport;branch=z9hG4bK92ypSpD2Q6D6H
>    Max-Forwards: 70
>    From: "" <sip:@67.43.59.58>;tag=9tyvjr55tmyXe
> 
> is still the output of the INVITE
> 
> Airespring needs (without the "" after From:, but before <sip:...)
> 
>    ------------------------------------------------------------------------
>    INVITE sip:14159927717 at 64.211.41.115 SIP/2.0
>    Via: SIP/2.0/UDP 67.43.59.58:5080;rport;branch=z9hG4bK92ypSpD2Q6D6H
>    Max-Forwards: 70
>    From: <sip:@67.43.59.58>;tag=9tyvjr55tmyXe
> 
> I also tried setting sip_from_uri, effective_caller_id_number to _undef_ but did not see a different result. Am I setting _undef_ in the correct places? Thanks.
> 
> I am using the latest nightly snapshop.
> 
> --matt
> 
> On Fri, Feb 11, 2011 at 4:04 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> set it to _undef_ for blank in the sip packet.
> one of many hacks for stupid behaviors in commercial products.
> 
> 
> 
> On Fri, Feb 11, 2011 at 5:37 AM, David Ponzone <david.ponzone at ipeva.fr> wrote:
> Matthew,
> 
> I think I forgot to tell you something.
> If you are sending the call to an outbound gateway, you need to add this in the gateway parameter:
>  <param name="caller-id-in-from" value="true"/>
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 11/02/2011 ? 12:06, Matthew Fong a ?crit :
> 
>> effective_caller_id_number doesn't seem to work. I was able to get
>> 
>> From: "" <sip:@0.0.0.0>;tag=j0yN36H8tZgDe
>> 
>> using sip_from_uri='<sip:@X.X.X.X>' but there are still the proceeding ""
>> 
>> Does anyone else have any suggestions?
>> 
>> Thanks
>> --matt
>> http://www.hellohunter.com
>> Hosted Dialer
>> On Wed, Feb 9, 2011 at 2:06 AM, David Ponzone <david.ponzone at ipeva.fr> wrote:
>> Try setting effective_caller_id_number to " "
>> 
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> 
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> 
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> 
>> 
>> 
>> 
>> Le 09/02/2011 ? 10:55, Matthew Fong a ?crit :
>> 
>>> I am trying to setup a customer who wanted to utilize the airespring national did presence, where in they rewrite the Caller ID of outbound calls. In order to do this tho, they say I need to send a blank caller id. FreeSWITCH will send 0000000000 if the caller id is not specified. Airespring wants something formatted like
>>> 
>>> INVITE sip:jungar at alpha-org.com SIP/2.0
>>> Via: SIP/2.0/UDP zetamachine.beta-org.com:5060;branch=as82je8ei4kr
>>> Max-Forwards: 70
>>> To:  2556112121 <sip:jungar at alpha-org.com>
>>> From: <sip:jsmith at beta-org.com>;tag=13456
>>> Call-ID: @zetamachine.beta-org.com
>>> 
>>> 
>>> with the From: field looking like any of the following
>>> 
>>> From: <sip:jsmith at beta-org.com>;tag=13456
>>> From: <  @208.76.54.59>;tag=13456
>>> From:  @208.76.54.59;tag=13456
>>> 
>>> The closest I got was to make FreeSWITCH to the following, but airespring won't budge. Can anyone tell me how to make the From string as airespring wants for their national DID presence service. Thanks.
>>> From: "" <sip:0000000000 at 208.76.54.59>;tag=UyNBXjyUUay8m
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mattdfong at gmail.com  Fri Feb 11 19:30:04 2011
From: mattdfong at gmail.com (Matthew Fong)
Date: Fri, 11 Feb 2011 08:30:04 -0800
Subject: [Freeswitch-users] Using an Empty FROM field for SIP Messages
In-Reply-To: <4430AE8C-0C39-43F4-A085-42A7AACFE0CC@ipeva.fr>
References: <AANLkTikEyL=NpmdqM+Vx=rxaM2+zEyxuJr_oGdFDWLwU@mail.gmail.com>
	<D810E0FC-0474-45E8-9C9A-C8D9D438032C@ipeva.fr>
	<AANLkTikbncOB_=Q4hEc2gg8A+bS59CN0=RXjUdAdT0az@mail.gmail.com>
	<6CE2C3C3-0EAF-43F5-B835-5A954F194ED3@ipeva.fr>
	<AANLkTin7tStk2f=dJv-P8kRyU+upvmUfmW2Y1DRSx8QG@mail.gmail.com>
	<AANLkTiniQkwgfqZNHfaa_imuEGFDo3sEYbsKmkTLb5Yb@mail.gmail.com>
	<4430AE8C-0C39-43F4-A085-42A7AACFE0CC@ipeva.fr>
Message-ID: <AANLkTinEwBKfDCZ-4Tu8aQV13HYJtXtR69UtYdm19Bfv@mail.gmail.com>

No luck :(

freeswitch at ubuntu> originate
{ignore_early_media=true,effective_caller_id_name=_undef_}sofia/gateway/
airwest.com/1415992XXXX 5000

   ------------------------------------------------------------------------
   INVITE sip:14159927717 at 64.211.41.115 SIP/2.0
   Via: SIP/2.0/UDP 67.43.59.58:5080;rport;branch=z9hG4bK774H41greZpXc
   Max-Forwards: 70
   From: "" <sip:unused at 64.211.41.115;transport=udp>;tag=7tjDN85Qyt48r

On Fri, Feb 11, 2011 at 8:18 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Set effective_caller_id_name to _undef_.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 11/02/2011 ? 17:11, Matthew Fong a ?crit :
>
> inside my conf/sip_profiles/external/airespring.xml file I set
>
> <param name="caller-id-in-from" value="_undef_"/>
>
> and restarted but
>
>    ------------------------------------------------------------------------
>    INVITE sip:14159927717 at 64.211.41.115 SIP/2.0
>    Via: SIP/2.0/UDP 67.43.59.58:5080;rport;branch=z9hG4bK92ypSpD2Q6D6H
>    Max-Forwards: 70
>    From: "" <sip:@67.43.59.58>;tag=9tyvjr55tmyXe
>
> is still the output of the INVITE
>
> Airespring needs (without the "" after From:, but before <sip:...)
>
>    ------------------------------------------------------------------------
>    INVITE sip:14159927717 at 64.211.41.115 SIP/2.0
>    Via: SIP/2.0/UDP 67.43.59.58:5080;rport;branch=z9hG4bK92ypSpD2Q6D6H
>    Max-Forwards: 70
>    From: <sip:@67.43.59.58>;tag=9tyvjr55tmyXe
>
> I also tried setting sip_from_uri, effective_caller_id_number to _undef_
> but did not see a different result. Am I setting _undef_ in the correct
> places? Thanks.
>
> I am using the latest nightly snapshop.
>
> --matt
>
> On Fri, Feb 11, 2011 at 4:04 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> set it to _undef_ for blank in the sip packet.
>> one of many hacks for stupid behaviors in commercial products.
>>
>>
>>
>> On Fri, Feb 11, 2011 at 5:37 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>>
>>> Matthew,
>>>
>>> I think I forgot to tell you something.
>>> If you are sending the call to an outbound gateway, you need to add this
>>> in the gateway parameter:
>>>  <param name="caller-id-in-from" value="true"/>
>>>
>>> David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 11/02/2011 ? 12:06, Matthew Fong a ?crit :
>>>
>>> effective_caller_id_number doesn't seem to work. I was able to get
>>>
>>> From: "" <sip:@0.0.0.0>;tag=j0yN36H8tZgDe
>>>
>>> using sip_from_uri='<sip:@X.X.X.X>' but there are still the proceeding
>>> ""
>>>
>>> Does anyone else have any suggestions?
>>>
>>> Thanks
>>> --matt
>>> http://www.hellohunter.com
>>> Hosted Dialer
>>> On Wed, Feb 9, 2011 at 2:06 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>>>
>>>> Try setting effective_caller_id_number to " "
>>>>
>>>>  David Ponzone  Direction Technique
>>>> email: david.ponzone at ipeva.fr
>>>> tel:      01 74 03 18 97
>>>> gsm:   06 66 98 76 34
>>>>
>>>> Service Client IPeva
>>>> tel:      0811 46 26 26
>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>
>>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>>> l'exp?diteur.*
>>>> *
>>>> *
>>>>
>>>>
>>>>
>>>> Le 09/02/2011 ? 10:55, Matthew Fong a ?crit :
>>>>
>>>> I am trying to setup a customer who wanted to utilize the airespring
>>>> national did presence, where in they rewrite the Caller ID of outbound
>>>> calls. In order to do this tho, they say I need to send a blank caller id.
>>>> FreeSWITCH will send 0000000000 if the caller id is not specified.
>>>> Airespring wants something formatted like
>>>>
>>>> INVITE sip:jungar at alpha-org.com SIP/2.0
>>>>
>>>> Via: SIP/2.0/UDP zetamachine.beta-org.com:5060;branch=as82je8ei4kr
>>>>
>>>> Max-Forwards: 70
>>>>
>>>> To:  2556112121 <sip:jungar at alpha-org.com>
>>>>
>>>> From: <sip:jsmith at beta-org.com>;tag=13456
>>>>
>>>> Call-ID: @zetamachine.beta-org.com
>>>>
>>>>
>>>> with the From: field looking like any of the following
>>>>
>>>> From: <sip:jsmith at beta-org.com>;tag=13456
>>>> From: <  @208.76.54.59>;tag=13456
>>>> From:  @208.76.54.59;tag=13456
>>>>
>>>> The closest I got was to make FreeSWITCH to the following, but
>>>> airespring won't budge. Can anyone tell me how to make the From string as
>>>> airespring wants for their national DID presence service. Thanks.
>>>> From: "" <sip:0000000000 at 208.76.54.59>;tag=UyNBXjyUUay8m
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 msc at freeswitch.org  Fri Feb 11 19:34:40 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 11 Feb 2011 10:34:40 -0600
Subject: [Freeswitch-users] no voice for call in same lan network
In-Reply-To: <AANLkTing0KFwW9LqmkaNFqDeEF70O0MVSLtoFjVGTJKA@mail.gmail.com>
References: <AANLkTing0KFwW9LqmkaNFqDeEF70O0MVSLtoFjVGTJKA@mail.gmail.com>
Message-ID: <AANLkTikt6BH4q4K=M5H_X3Di7bMtVvtsuaYm=3Fw-aRO@mail.gmail.com>

Get a console log w/ sip trace of a call without audio and put it in a
pastebin. Hopefully there will be some clues. Also, be sure to document your
network layout - what IPs and subnets belong to what, etc.

-MC

On Fri, Feb 11, 2011 at 9:16 AM, Sam <u2nsam at gmail.com> wrote:

> Here ,
>
> i have bunch extension in the same lan network and when they talk with each
> other there is no voice ,
> i have another bunch in other lan network the too do not get voice when
> talking with each other.
> Here the public ip for 2 bunches are different and when an extension from
> bunch A calls to extension
> of bunch B then they have voice.
> FS is on public IP   :(
>
> From the trace i could see that when making an internal call in the same
> LAN the RTP flows to private IP
> and when making and external call from bunch A to bunch B the RTP is send
> to public IP where the voice is heard,
> also while making and outbound call the RTP is send to public IP of the LAN
> and there is voice.
>
> Is this a known scenario to anybody ?
>
> 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 msc at freeswitch.org  Fri Feb 11 19:52:50 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 11 Feb 2011 10:52:50 -0600
Subject: [Freeswitch-users] Using an Empty FROM field for SIP Messages
In-Reply-To: <AANLkTinEwBKfDCZ-4Tu8aQV13HYJtXtR69UtYdm19Bfv@mail.gmail.com>
References: <AANLkTikEyL=NpmdqM+Vx=rxaM2+zEyxuJr_oGdFDWLwU@mail.gmail.com>
	<D810E0FC-0474-45E8-9C9A-C8D9D438032C@ipeva.fr>
	<AANLkTikbncOB_=Q4hEc2gg8A+bS59CN0=RXjUdAdT0az@mail.gmail.com>
	<6CE2C3C3-0EAF-43F5-B835-5A954F194ED3@ipeva.fr>
	<AANLkTin7tStk2f=dJv-P8kRyU+upvmUfmW2Y1DRSx8QG@mail.gmail.com>
	<AANLkTiniQkwgfqZNHfaa_imuEGFDo3sEYbsKmkTLb5Yb@mail.gmail.com>
	<4430AE8C-0C39-43F4-A085-42A7AACFE0CC@ipeva.fr>
	<AANLkTinEwBKfDCZ-4Tu8aQV13HYJtXtR69UtYdm19Bfv@mail.gmail.com>
Message-ID: <AANLkTinwkA5hm5e=ETGW_NxL3b+aHO5t_XKhrF+1JBS1@mail.gmail.com>

>
>
>>>>> with the From: field looking like any of the following
>>>>>
>>>>> From: <sip:jsmith at beta-org.com>;tag=13456
>>>>> From: <  @208.76.54.59>;tag=13456
>>>>> From:  @208.76.54.59;tag=13456
>>>>>
>>>>> Let me get this straight - they are barfing on empty double quotes?
They don't actually consider that an empty caller ID? Can you have them give
you written justification that one SHOULD consider "" to mean anything other
than an empty string in this context? Any IETF references would be most
appreciated.

-MC
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From megahohol at gmail.com  Fri Feb 11 19:55:27 2011
From: megahohol at gmail.com (Grygoriy Dobrovolskyy)
Date: Fri, 11 Feb 2011 17:55:27 +0100
Subject: [Freeswitch-users] Vestec Speech Engine: Large Vocabulary
In-Reply-To: <4D544723.8070202@vestec.com>
References: <4D544723.8070202@vestec.com>
Message-ID: <AANLkTint0ic-93i8nov5=StRb1DoBfc0V-hTGfMYLb0_@mail.gmail.com>

2011/2/10 Kashif Kahn <kahn at vestec.com>:
> Hello Everyone,
>
> Vestec recently launched a large vocabulary speech engine for
> "command-and-control" type IVR applications. The new engine - called
> Tier-2 - supports a vocabulary size of 2,500 keywords per recognition
> and complements our existing Tier-1 engine that has a 500 keywords
> vocabulary size. Tier-2 retails for $199 per port (ie. channel) while
> Tier-1 is priced at $99 per port. Acoustic models needs to be licensed
> separately at $9.99 per language.
>
> Our ASR is designed for keywords-based interaction and offers the best
> deal around for speech enabling a wide variety of IVR applications,
> including DTMF menus and multi-step business processes. The speech
> engine is standards based in terms of grammar writing format and
> platform integration protocols. We support SRGS grammar (ABNF & XML) as
> well as MRCP integration (v1 & v2). In addition, we offer a highly
> scalable architecture that is capable of supporting thousands of ports
> in an efficient manner.
>
> A starter kit - comprising a specially priced perpetual license for a
> full-function, full-feature standard ASR engine - is available for $25.
> Please visit Vestec webstore at: http://www.vestec.com/products
>
> Regards,
> -Kashif
>
> --
>
> Kashif Kahn
> VP Business Development
> Vestec Inc
> Waterloo, ON Canada
> phone: +1 519 885-7615
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

Hello, there is no info on your site about the language support.
Do you support French ?


From reply at matthewfong.com  Fri Feb 11 19:58:09 2011
From: reply at matthewfong.com (Matthew Fong)
Date: Fri, 11 Feb 2011 08:58:09 -0800
Subject: [Freeswitch-users] Using an Empty FROM field for SIP Messages
In-Reply-To: <AANLkTinwkA5hm5e=ETGW_NxL3b+aHO5t_XKhrF+1JBS1@mail.gmail.com>
References: <AANLkTikEyL=NpmdqM+Vx=rxaM2+zEyxuJr_oGdFDWLwU@mail.gmail.com>
	<D810E0FC-0474-45E8-9C9A-C8D9D438032C@ipeva.fr>
	<AANLkTikbncOB_=Q4hEc2gg8A+bS59CN0=RXjUdAdT0az@mail.gmail.com>
	<6CE2C3C3-0EAF-43F5-B835-5A954F194ED3@ipeva.fr>
	<AANLkTin7tStk2f=dJv-P8kRyU+upvmUfmW2Y1DRSx8QG@mail.gmail.com>
	<AANLkTiniQkwgfqZNHfaa_imuEGFDo3sEYbsKmkTLb5Yb@mail.gmail.com>
	<4430AE8C-0C39-43F4-A085-42A7AACFE0CC@ipeva.fr>
	<AANLkTinEwBKfDCZ-4Tu8aQV13HYJtXtR69UtYdm19Bfv@mail.gmail.com>
	<AANLkTinwkA5hm5e=ETGW_NxL3b+aHO5t_XKhrF+1JBS1@mail.gmail.com>
Message-ID: <AANLkTimkWi5bUDf_ts4FiymnK-Z-aMJFkk6_pBdNubVc@mail.gmail.com>

Yes, they do not consider "" an empty caller id string. Here is a reply to
the email

Matt,



Please change the  following  FROM : ""
<sip:jsmith at 208.76.54.59>;tag=pjUp5yDSHtDXQ
  to :   FROM : <  @208.76.54.59> ; tag?..  or    FROM:  @208.76.54.59;
tag?..



Dumitru


of course there is no IETF justification, it's much easier for them to
hangup on me. :( I know it's a problem with airespring and not freeswitch, I
just thought maybe there would be a hack. thanks anyway.

--matt

On Fri, Feb 11, 2011 at 8:52 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>>>>>> with the From: field looking like any of the following
>>>>>>
>>>>>> From: <sip:jsmith at beta-org.com>;tag=13456
>>>>>> From: <  @208.76.54.59>;tag=13456
>>>>>> From:  @208.76.54.59;tag=13456
>>>>>>
>>>>>> Let me get this straight - they are barfing on empty double quotes?
> They don't actually consider that an empty caller ID? Can you have them give
> you written justification that one SHOULD consider "" to mean anything other
> than an empty string in this context? Any IETF references would be most
> appreciated.
>
> -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 mattdfong at gmail.com  Fri Feb 11 19:58:19 2011
From: mattdfong at gmail.com (Matthew Fong)
Date: Fri, 11 Feb 2011 08:58:19 -0800
Subject: [Freeswitch-users] Using an Empty FROM field for SIP Messages
In-Reply-To: <AANLkTinwkA5hm5e=ETGW_NxL3b+aHO5t_XKhrF+1JBS1@mail.gmail.com>
References: <AANLkTikEyL=NpmdqM+Vx=rxaM2+zEyxuJr_oGdFDWLwU@mail.gmail.com>
	<D810E0FC-0474-45E8-9C9A-C8D9D438032C@ipeva.fr>
	<AANLkTikbncOB_=Q4hEc2gg8A+bS59CN0=RXjUdAdT0az@mail.gmail.com>
	<6CE2C3C3-0EAF-43F5-B835-5A954F194ED3@ipeva.fr>
	<AANLkTin7tStk2f=dJv-P8kRyU+upvmUfmW2Y1DRSx8QG@mail.gmail.com>
	<AANLkTiniQkwgfqZNHfaa_imuEGFDo3sEYbsKmkTLb5Yb@mail.gmail.com>
	<4430AE8C-0C39-43F4-A085-42A7AACFE0CC@ipeva.fr>
	<AANLkTinEwBKfDCZ-4Tu8aQV13HYJtXtR69UtYdm19Bfv@mail.gmail.com>
	<AANLkTinwkA5hm5e=ETGW_NxL3b+aHO5t_XKhrF+1JBS1@mail.gmail.com>
Message-ID: <AANLkTimFFG4+J077jzxpoVs1RQz-KMQY6UnzFD2aPFua@mail.gmail.com>

Yes, they do not consider "" an empty caller id string. Here is a reply to
the email

Matt,



Please change the  following  FROM : ""
<sip:jsmith at 208.76.54.59>;tag=pjUp5yDSHtDXQ
  to :   FROM : <  @208.76.54.59> ; tag?..  or    FROM:  @208.76.54.59;
tag?..



Dumitru


of course there is no IETF justification, it's much easier for them to
hangup on me. :( I know it's a problem with airespring and not freeswitch, I
just thought maybe there would be a hack. thanks anyway.

--matt

On Fri, Feb 11, 2011 at 8:52 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>>>>>> with the From: field looking like any of the following
>>>>>>
>>>>>> From: <sip:jsmith at beta-org.com>;tag=13456
>>>>>> From: <  @208.76.54.59>;tag=13456
>>>>>> From:  @208.76.54.59;tag=13456
>>>>>>
>>>>>> Let me get this straight - they are barfing on empty double quotes?
> They don't actually consider that an empty caller ID? Can you have them give
> you written justification that one SHOULD consider "" to mean anything other
> than an empty string in this context? Any IETF references would be most
> appreciated.
>
> -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  Fri Feb 11 20:00:50 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 11 Feb 2011 11:00:50 -0600
Subject: [Freeswitch-users] Using an Empty FROM field for SIP Messages
In-Reply-To: <AANLkTimFFG4+J077jzxpoVs1RQz-KMQY6UnzFD2aPFua@mail.gmail.com>
References: <AANLkTikEyL=NpmdqM+Vx=rxaM2+zEyxuJr_oGdFDWLwU@mail.gmail.com>
	<D810E0FC-0474-45E8-9C9A-C8D9D438032C@ipeva.fr>
	<AANLkTikbncOB_=Q4hEc2gg8A+bS59CN0=RXjUdAdT0az@mail.gmail.com>
	<6CE2C3C3-0EAF-43F5-B835-5A954F194ED3@ipeva.fr>
	<AANLkTin7tStk2f=dJv-P8kRyU+upvmUfmW2Y1DRSx8QG@mail.gmail.com>
	<AANLkTiniQkwgfqZNHfaa_imuEGFDo3sEYbsKmkTLb5Yb@mail.gmail.com>
	<4430AE8C-0C39-43F4-A085-42A7AACFE0CC@ipeva.fr>
	<AANLkTinEwBKfDCZ-4Tu8aQV13HYJtXtR69UtYdm19Bfv@mail.gmail.com>
	<AANLkTinwkA5hm5e=ETGW_NxL3b+aHO5t_XKhrF+1JBS1@mail.gmail.com>
	<AANLkTimFFG4+J077jzxpoVs1RQz-KMQY6UnzFD2aPFua@mail.gmail.com>
Message-ID: <AANLkTi=5V=3bAHqmfQhNaLzZ5=jeWh6-3Ze3-MJ=SsuW@mail.gmail.com>

Ugh. You may need to HIYDS on this one...
-MC

On Fri, Feb 11, 2011 at 10:58 AM, Matthew Fong <mattdfong at gmail.com> wrote:

> Yes, they do not consider "" an empty caller id string. Here is a reply to
> the email
>
> Matt,
>
>
>
> Please change the  following  FROM : "" <sip:jsmith at 208.76.54.59>;tag=pjUp5yDSHtDXQ
>   to :   FROM : <  @208.76.54.59> ; tag?..  or    FROM:  @208.76.54.59;
> tag?..
>
>
>
> Dumitru
>
>
> of course there is no IETF justification, it's much easier for them to
> hangup on me. :( I know it's a problem with airespring and not freeswitch, I
> just thought maybe there would be a hack. thanks anyway.
>
> --matt
>
> On Fri, Feb 11, 2011 at 8:52 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>>>>>> with the From: field looking like any of the following
>>>>>>>
>>>>>>> From: <sip:jsmith at beta-org.com>;tag=13456
>>>>>>> From: <  @208.76.54.59>;tag=13456
>>>>>>> From:  @208.76.54.59;tag=13456
>>>>>>>
>>>>>>> Let me get this straight - they are barfing on empty double quotes?
>> They don't actually consider that an empty caller ID? Can you have them give
>> you written justification that one SHOULD consider "" to mean anything other
>> than an empty string in this context? Any IETF references would be most
>> appreciated.
>>
>> -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 david.ponzone at ipeva.fr  Fri Feb 11 20:04:32 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 11 Feb 2011 18:04:32 +0100
Subject: [Freeswitch-users] Using an Empty FROM field for SIP Messages
In-Reply-To: <AANLkTimFFG4+J077jzxpoVs1RQz-KMQY6UnzFD2aPFua@mail.gmail.com>
References: <AANLkTikEyL=NpmdqM+Vx=rxaM2+zEyxuJr_oGdFDWLwU@mail.gmail.com>
	<D810E0FC-0474-45E8-9C9A-C8D9D438032C@ipeva.fr>
	<AANLkTikbncOB_=Q4hEc2gg8A+bS59CN0=RXjUdAdT0az@mail.gmail.com>
	<6CE2C3C3-0EAF-43F5-B835-5A954F194ED3@ipeva.fr>
	<AANLkTin7tStk2f=dJv-P8kRyU+upvmUfmW2Y1DRSx8QG@mail.gmail.com>
	<AANLkTiniQkwgfqZNHfaa_imuEGFDo3sEYbsKmkTLb5Yb@mail.gmail.com>
	<4430AE8C-0C39-43F4-A085-42A7AACFE0CC@ipeva.fr>
	<AANLkTinEwBKfDCZ-4Tu8aQV13HYJtXtR69UtYdm19Bfv@mail.gmail.com>
	<AANLkTinwkA5hm5e=ETGW_NxL3b+aHO5t_XKhrF+1JBS1@mail.gmail.com>
	<AANLkTimFFG4+J077jzxpoVs1RQz-KMQY6UnzFD2aPFua@mail.gmail.com>
Message-ID: <30CBB919-BBD7-41DD-8AFD-F97525AC748A@ipeva.fr>

Really, dump them.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 11/02/2011 ? 17:58, Matthew Fong a ?crit :

> Yes, they do not consider "" an empty caller id string. Here is a reply to the email
> 
> Matt,
> 
>  
> Please change the  following  FROM : "" <sip:jsmith at 208.76.54.59>;tag=pjUp5yDSHtDXQ    to :   FROM : <  @208.76.54.59> ; tag?..  or    FROM:  @208.76.54.59; tag?..
> 
>  
> Dumitru
> 
> 
> 
> of course there is no IETF justification, it's much easier for them to hangup on me. :( I know it's a problem with airespring and not freeswitch, I just thought maybe there would be a hack. thanks anyway.
> 
> --matt
> 
> On Fri, Feb 11, 2011 at 8:52 AM, Michael Collins <msc at freeswitch.org> wrote:
>>>> 
>>>> with the From: field looking like any of the following
>>>> 
>>>> From: <sip:jsmith at beta-org.com>;tag=13456
>>>> From: <  @208.76.54.59>;tag=13456
>>>> From:  @208.76.54.59;tag=13456
> Let me get this straight - they are barfing on empty double quotes? They don't actually consider that an empty caller ID? Can you have them give you written justification that one SHOULD consider "" to mean anything other than an empty string in this context? Any IETF references would be most appreciated.
> 
> -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 steveayre at gmail.com  Fri Feb 11 20:12:44 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 11 Feb 2011 17:12:44 +0000
Subject: [Freeswitch-users] 503 Maximum Calls In Progress
In-Reply-To: <6B4D073D-32AE-4E1D-B919-1169F29059C7@ipeva.fr>
References: <fc5eb12a398e.4d55170a@insightbb.com>
	<6B4D073D-32AE-4E1D-B919-1169F29059C7@ipeva.fr>
Message-ID: <AANLkTinuJ8aiqMyo38=VZDiAvSb0f9jwHCfk7D23vd22@mail.gmail.com>

Uhhh... which is the opposite of commenting it out?

Gary, if you don't have that param in your file then it's not this.

You say FS had been up an hour. Had it handled any successful calls in that
time?

Was FS shutting down at the time?

-Steve



2011/2/11 David Ponzone <david.ponzone at ipeva.fr>

> So add it.
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 11/02/2011 ? 17:01, Gary Chen a ?crit :
>
>  My switch.conf.xml does not have this param.
>
> Gary
>  ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Peter Olsson
> *Sent:* Friday, February 11, 2011 10:39 AM
> *To:* 'FreeSWITCH Users Help'
> *Subject:* Re: [Freeswitch-users] 503 Maximum Calls In Progress
>
>
>
> Comment from anthm;
>
>
>
> edit conf/autoload_configs/switch.conf.xml
>
>
>
> comment out
>
>
>
> <param name="min-idle-cpu" value="25"/>
>
>
>
> It seems to not work well on 32 bit.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.ponzone at ipeva.fr  Fri Feb 11 20:18:04 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 11 Feb 2011 18:18:04 +0100
Subject: [Freeswitch-users] 503 Maximum Calls In Progress
In-Reply-To: <AANLkTinuJ8aiqMyo38=VZDiAvSb0f9jwHCfk7D23vd22@mail.gmail.com>
References: <fc5eb12a398e.4d55170a@insightbb.com>
	<6B4D073D-32AE-4E1D-B919-1169F29059C7@ipeva.fr>
	<AANLkTinuJ8aiqMyo38=VZDiAvSb0f9jwHCfk7D23vd22@mail.gmail.com>
Message-ID: <F61402A2-A7F6-4E62-B79C-A21F28EDCB17@ipeva.fr>

Sorry, replied too quickly :)

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 11/02/2011 ? 18:12, Steven Ayre a ?crit :

> Uhhh... which is the opposite of commenting it out?
> 
> Gary, if you don't have that param in your file then it's not this.
> 
> You say FS had been up an hour. Had it handled any successful calls in that time?
> 
> Was FS shutting down at the time?
> 
> -Steve
> 
> 
> 
> 2011/2/11 David Ponzone <david.ponzone at ipeva.fr>
> So add it.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 11/02/2011 ? 17:01, Gary Chen a ?crit :
> 
>>  My switch.conf.xml does not have this param.
>> 
>> Gary
>> 
>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson
>> Sent: Friday, February 11, 2011 10:39 AM
>> To: 'FreeSWITCH Users Help'
>> Subject: Re: [Freeswitch-users] 503 Maximum Calls In Progress
>> 
>>  
>> Comment from anthm;
>> 
>>  
>> edit conf/autoload_configs/switch.conf.xml
>> 
>>  
>> comment out
>> 
>>  
>> <param name="min-idle-cpu" value="25"/>
>> 
>>  
>> It seems to not work well on 32 bit.
>> 
>>  
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Feb 11 20:26:50 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Fri, 11 Feb 2011 12:26:50 -0500
Subject: [Freeswitch-users] 503 Maximum Calls In Progress
Message-ID: <fc75fc6b47c5.4d552b0a@insightbb.com>

Yes. I did some test calls between two Cisco sip phones and they all work fine and then it just stop working. At that point, FS is still up and I can see the system information by press F keys. It just won?t accept any calls. I looked at log file and it just has that one line:

2011-02-11 17:39:58.691008 [CRIT] switch_core_session.c:1629 The system cannot create any sessions at this time.

?

The register message is still working at that time.

?

Gary

?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: Friday, February 11, 2011 12:13 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] 503 Maximum Calls In Progress

?

Uhhh... which is the opposite of commenting it out?

?

Gary, if you don't have that param in your file then it's not this.

?

You say FS had been up an hour. Had it handled any successful calls in that time?

?

Was FS shutting down at the time?

?

-Steve

?

?

?

2011/2/11 David Ponzone <david.ponzone at ipeva.fr>

So add it.

?

David Ponzone ?Direction Technique

email: david.ponzone at ipeva.fr

tel: ? ? ?01 74 03 18 97

gsm: ? 06 66 98 76 34

?

Service Client?IPeva

tel: ? ? ?0811 46 26 26

www.ipeva.fr? -? ?www.ipeva-studio.com

?

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.

?




?

Le 11/02/2011 ? 17:01, Gary Chen a ?crit :




?My switch.conf.xml does not have this param.

Gary

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson
Sent: Friday, February 11, 2011 10:39 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] 503 Maximum Calls In Progress

?

Comment from anthm;

?

edit conf/autoload_configs/switch.conf.xml

?

comment out

?

<param name="min-idle-cpu" value="25"/>

?

It seems to not work well on 32 bit.

?

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

?


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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 Feb 11 20:53:40 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 11 Feb 2011 11:53:40 -0600
Subject: [Freeswitch-users] 503 Maximum Calls In Progress
In-Reply-To: <fc75fc6b47c5.4d552b0a@insightbb.com>
References: <fc75fc6b47c5.4d552b0a@insightbb.com>
Message-ID: <AANLkTin5TwBxbDKpXt29FzFo-B2BUir1587emzV_HjFp@mail.gmail.com>

If you stop freeswitch and restart it does it still say that or are you able
to make some calls for a while before it throws that error?
-MC

On Fri, Feb 11, 2011 at 11:26 AM, Gary Chen <gchen00 at insightbb.com> wrote:

> Yes. I did some test calls between two Cisco sip phones and they all work
> fine and then it just stop working. At that point, FS is still up and I can
> see the system information by press F keys. It just won?t accept any calls.
> I looked at log file and it just has that one line:
>
> 2011-02-11 17:39:58.691008 [CRIT] switch_core_session.c:1629 The system
> cannot create any sessions at this time.
>
>
>
> The register message is still working at that time.
>
>
>
> Gary
>
>
>  ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* Friday, February 11, 2011 12:13 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] 503 Maximum Calls In Progress
>
>
>
> Uhhh... which is the opposite of commenting it out?
>
>
>
> Gary, if you don't have that param in your file then it's not this.
>
>
>
> You say FS had been up an hour. Had it handled any successful calls in that
> time?
>
>
>
> Was FS shutting down at the time?
>
>
>
> -Steve
>
>
>
>
>
>
>
> 2011/2/11 David Ponzone <david.ponzone at ipeva.fr>
>
> So add it.
>
>
>
> David Ponzone  Direction Technique
>
> email: david.ponzone at ipeva.fr
>
> tel:      01 74 03 18 97
>
> gsm:   06 66 98 76 34
>
>
>
> Service Client IPeva
>
> tel:      0811 46 26 26
>
> *www.ipeva.fr*  -   *www.ipeva-studio.com*
>
>
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de
> ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.*
>
> * *
>
>
>
>
>
> Le 11/02/2011 ? 17:01, Gary Chen a ?crit :
>
>
>
>   My switch.conf.xml does not have this param.
>
> Gary
>  ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Peter Olsson
> *Sent:* Friday, February 11, 2011 10:39 AM
> *To:* 'FreeSWITCH Users Help'
> *Subject:* Re: [Freeswitch-users] 503 Maximum Calls In Progress
>
>
>
> Comment from anthm;
>
>
>
> edit conf/autoload_configs/switch.conf.xml
>
>
>
> comment out
>
>
>
> <param name="min-idle-cpu" value="25"/>
>
>
>
> It seems to not work well on 32 bit.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 11 20:54:22 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 11 Feb 2011 12:54:22 -0500
Subject: [Freeswitch-users] Aastra phone registration lost
In-Reply-To: <AANLkTimux4MBM1K_7itE4GnZQHjy_vR2Y6sr_z+u3gkR@mail.gmail.com>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>	<4D52B987.7050002@communicatefreely.net>	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>	<4D534AFD.3020401@communicatefreely.net>	<AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>	<60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>
	<AANLkTimux4MBM1K_7itE4GnZQHjy_vR2Y6sr_z+u3gkR@mail.gmail.com>
Message-ID: <4D5577CE.1000603@communicatefreely.net>

Sure,

I normally administer about 300 Aastra phones, with every model they 
make represented.

I have 22 connected to our Freeswitch "beta" system, which will 
eventually become production.

All the endpoints are behind NAT without exception.  There are a number 
of legacy 9133i and 480i phones on the network that don't have the newer 
NAT traversal features available, but this doesn't seem to be a 
problem.  I have some of the nat traversal options turned on in the 
sofia profile though, so fs will send media back to the originating 
address and port.

They have been quite reliable, and the sound quality has been excellent, 
with the newer phones using g722 at 16KHz.

There are a few advanced features that I haven't had a chance to play 
with yet, but here's what I have working:

Regular calls, in and out.
Intercom calls (auto-answer to speaker phone)
Automatic update of destination name and number (updates when checking 
voice mail, and when calling an extension).  Only on newer phones
Blind and attended transfer
Music on hold
SIP using udp or tcp (haven't tried TLS yet)
Fewer issues with DTMF than with asterisk, using rfc2833 dtmf (no issues 
as of yet).
BLF lamps work correctly, flashing when the phone rings, lit steady when 
they are on the phone.
Distinctive ringing works.
I haven't tried SLA yet, but Aastra recently released a firmware update 
that fixes a missing header, reported to have broken correct SLA 
operation.  I'm hoping to test that in the next week or two.

The phones provision very nicely - we auto generate config using PHP 
scripts that generate a config file on the fly from the user database.  
These are very easy phones to deploy in large installations, or to the 
outside world (not readily accessible).  They have just added some new 
features that allow for remote diagnostics of the phones as well.

There is a great deal of XML programmability in the phones too, which 
I'm starting to use for call control and other useful things (updating 
forwarding rules in the database, or conference and recording control 
using ESL).

Hope that helps!

-Tim

Aloysius Lloyd wrote:
> Tim,
>
> Can you share your success stories FreeSWITCH and Aastra.
>
> Aastra Phones Behind the NAT?
>
> In my case Aastra phones registration not a problem. 
>
> But calls drooped every 60 sec ... in the same environment Linksys and 
> Polycom works perfectly.
>
> How stable the Aastra phones with FreeSWITCH system.
>
> TIA
>
> Lloyd



From gchen00 at insightbb.com  Fri Feb 11 21:31:10 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Fri, 11 Feb 2011 13:31:10 -0500
Subject: [Freeswitch-users] 503 Maximum Calls In Progress
Message-ID: <fc75ff0036de.4d553a1e@insightbb.com>

After restart, everything back to normal. I can make call again.

I will do more tests. I will let you know if it happen again.

?

Gary

?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Friday, February 11, 2011 12:54 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] 503 Maximum Calls In Progress

?

If you stop freeswitch and restart it does it still say that or are you able to make some calls for a while before it throws that error?

-MC

On Fri, Feb 11, 2011 at 11:26 AM, Gary Chen <gchen00 at insightbb.com> wrote:

Yes. I did some test calls between two Cisco sip phones and they all work fine and then it just stop working. At that point, FS is still up and I can see the system information by press F keys. It just won?t accept any calls. I looked at log file and it just has that one line:

2011-02-11 17:39:58.691008 [CRIT] switch_core_session.c:1629 The system cannot create any sessions at this time.

?

The register message is still working at that time.

?

Gary
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From yky1628 at yahoo.com  Fri Feb 11 23:11:03 2011
From: yky1628 at yahoo.com (Frankie Yiu)
Date: Fri, 11 Feb 2011 12:11:03 -0800 (PST)
Subject: [Freeswitch-users] how to set up a DTMF callback function
Message-ID: <93057.68597.qm@web30502.mail.mud.yahoo.com>

If I want to get a DTMF?callBack?to a function when a caller presses a key, what 
function should I use to set it up?

Thanks,?
?Frankie 


 
____________________________________________________________________________________
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From yky1628 at yahoo.com  Fri Feb 11 23:13:44 2011
From: yky1628 at yahoo.com (Frankie Yiu)
Date: Fri, 11 Feb 2011 12:13:44 -0800 (PST)
Subject: [Freeswitch-users] How to set up a DTMF callback in C or C#?
Message-ID: <812941.11103.qm@web30507.mail.mud.yahoo.com>

Hi there,

If I want to set up a DTMF?callBack?to a function (in either C or C#) when a 
caller presses a key, what function should I use to set it up?

Thanks,?
?Frankie 


      
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From anthony.minessale at gmail.com  Fri Feb 11 23:42:42 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 11 Feb 2011 14:42:42 -0600
Subject: [Freeswitch-users] Using an Empty FROM field for SIP Messages
In-Reply-To: <AANLkTinEwBKfDCZ-4Tu8aQV13HYJtXtR69UtYdm19Bfv@mail.gmail.com>
References: <AANLkTikEyL=NpmdqM+Vx=rxaM2+zEyxuJr_oGdFDWLwU@mail.gmail.com>
	<D810E0FC-0474-45E8-9C9A-C8D9D438032C@ipeva.fr>
	<AANLkTikbncOB_=Q4hEc2gg8A+bS59CN0=RXjUdAdT0az@mail.gmail.com>
	<6CE2C3C3-0EAF-43F5-B835-5A954F194ED3@ipeva.fr>
	<AANLkTin7tStk2f=dJv-P8kRyU+upvmUfmW2Y1DRSx8QG@mail.gmail.com>
	<AANLkTiniQkwgfqZNHfaa_imuEGFDo3sEYbsKmkTLb5Yb@mail.gmail.com>
	<4430AE8C-0C39-43F4-A085-42A7AACFE0CC@ipeva.fr>
	<AANLkTinEwBKfDCZ-4Tu8aQV13HYJtXtR69UtYdm19Bfv@mail.gmail.com>
Message-ID: <AANLkTikj5p5o2XHJBbO94cXvN+qFyJB20UbEhhk1dL2=@mail.gmail.com>

B.S.

It's right here in the code:

if (!from_display && !strcasecmp(tech_pvt->caller_profile->caller_id_name,
"_undef_")) {
            from_str = switch_core_session_sprintf(session, "<%s>",
use_from_str);
        } else {
            from_str = switch_core_session_sprintf(session, "\"%s\" <%s>",
from_display ? from_display :

tech_pvt->caller_profile->caller_id_name, use_from_str);
        }


do you see any "" in the From: on this sip trace?


freeswitch at deathstar.freeswitch.org> originate
{origination_caller_id_name=_undef_}sofia/internal/
1235 at conference.freeswitch.org 9999
2011-02-11 14:29:54.249877 [NOTICE] switch_channel.c:811 New Channel
sofia/internal/1235 at conference.freeswitch.org[f779eb8b-7b2c-480e-b262-824bda48548b]
2011-02-11 14:29:54.249877 [WARNING] switch_core_port_allocator.c:78
Rounding odd end port 65535 to 65534
send 1229 bytes to udp/[74.112.133.77]:5060 at 20:29:54.399158:
   ------------------------------------------------------------------------
   INVITE sip:1235 at conference.freeswitch.org SIP/2.0
   Via: SIP/2.0/UDP 8.19.97.170;rport;branch=z9hG4bKBc6y0DN2mvgXc
   Max-Forwards: 70
   From: <sip:0000000000 at 8.19.97.170>;tag=Bc78t3mKKHDHp
   To: <sip:1235 at conference.freeswitch.org>
   Call-ID: 86929dca-b0c0-122e-2091-00219b8b1186
   CSeq: 8392033 INVITE
   Contact: <sip:mod_sofia at 8.19.97.170:5060>
   User-Agent: The Guy In IRC IS WRONG
   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
   Session-Expires: 300;refresher=uac
   Min-SE: 120
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 305
   X-FS-Support: update_display
   Remote-Party-ID: <sip:0000000000 at 8.19.97.170
>;party=calling;screen=yes;privacy=off



On Fri, Feb 11, 2011 at 10:30 AM, Matthew Fong <mattdfong at gmail.com> wrote:

> No luck :(
>
> freeswitch at ubuntu> originate
> {ignore_early_media=true,effective_caller_id_name=_undef_}sofia/gateway/
> airwest.com/1415992XXXX 5000
>
>
>    ------------------------------------------------------------------------
>    INVITE sip:14159927717 at 64.211.41.115 SIP/2.0
>    Via: SIP/2.0/UDP 67.43.59.58:5080;rport;branch=z9hG4bK774H41greZpXc
>    Max-Forwards: 70
>    From: "" <sip:unused at 64.211.41.115;transport=udp>;tag=7tjDN85Qyt48r
>
>
> On Fri, Feb 11, 2011 at 8:18 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>
>> Set effective_caller_id_name to _undef_.
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 11/02/2011 ? 17:11, Matthew Fong a ?crit :
>>
>> inside my conf/sip_profiles/external/airespring.xml file I set
>>
>> <param name="caller-id-in-from" value="_undef_"/>
>>
>> and restarted but
>>
>>
>> ------------------------------------------------------------------------
>>    INVITE sip:14159927717 at 64.211.41.115 SIP/2.0
>>    Via: SIP/2.0/UDP 67.43.59.58:5080;rport;branch=z9hG4bK92ypSpD2Q6D6H
>>    Max-Forwards: 70
>>    From: "" <sip:@67.43.59.58>;tag=9tyvjr55tmyXe
>>
>> is still the output of the INVITE
>>
>> Airespring needs (without the "" after From:, but before <sip:...)
>>
>>
>> ------------------------------------------------------------------------
>>    INVITE sip:14159927717 at 64.211.41.115 SIP/2.0
>>    Via: SIP/2.0/UDP 67.43.59.58:5080;rport;branch=z9hG4bK92ypSpD2Q6D6H
>>    Max-Forwards: 70
>>    From: <sip:@67.43.59.58>;tag=9tyvjr55tmyXe
>>
>> I also tried setting sip_from_uri, effective_caller_id_number to _undef_
>> but did not see a different result. Am I setting _undef_ in the correct
>> places? Thanks.
>>
>> I am using the latest nightly snapshop.
>>
>> --matt
>>
>> On Fri, Feb 11, 2011 at 4:04 AM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> set it to _undef_ for blank in the sip packet.
>>> one of many hacks for stupid behaviors in commercial products.
>>>
>>>
>>>
>>> On Fri, Feb 11, 2011 at 5:37 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>>>
>>>> Matthew,
>>>>
>>>> I think I forgot to tell you something.
>>>> If you are sending the call to an outbound gateway, you need to add this
>>>> in the gateway parameter:
>>>>  <param name="caller-id-in-from" value="true"/>
>>>>
>>>> David Ponzone  Direction Technique
>>>> email: david.ponzone at ipeva.fr
>>>> tel:      01 74 03 18 97
>>>> gsm:   06 66 98 76 34
>>>>
>>>> Service Client IPeva
>>>> tel:      0811 46 26 26
>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>
>>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>>> l'exp?diteur.*
>>>> *
>>>> *
>>>>
>>>>
>>>>
>>>> Le 11/02/2011 ? 12:06, Matthew Fong a ?crit :
>>>>
>>>> effective_caller_id_number doesn't seem to work. I was able to get
>>>>
>>>> From: "" <sip:@0.0.0.0>;tag=j0yN36H8tZgDe
>>>>
>>>> using sip_from_uri='<sip:@X.X.X.X>' but there are still the proceeding
>>>> ""
>>>>
>>>> Does anyone else have any suggestions?
>>>>
>>>> Thanks
>>>> --matt
>>>> http://www.hellohunter.com
>>>> Hosted Dialer
>>>> On Wed, Feb 9, 2011 at 2:06 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>>>>
>>>>> Try setting effective_caller_id_number to " "
>>>>>
>>>>>  David Ponzone  Direction Technique
>>>>> email: david.ponzone at ipeva.fr
>>>>> tel:      01 74 03 18 97
>>>>> gsm:   06 66 98 76 34
>>>>>
>>>>> Service Client IPeva
>>>>> tel:      0811 46 26 26
>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>>
>>>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>>>> l'exp?diteur.*
>>>>> *
>>>>> *
>>>>>
>>>>>
>>>>>
>>>>> Le 09/02/2011 ? 10:55, Matthew Fong a ?crit :
>>>>>
>>>>> I am trying to setup a customer who wanted to utilize the airespring
>>>>> national did presence, where in they rewrite the Caller ID of outbound
>>>>> calls. In order to do this tho, they say I need to send a blank caller id.
>>>>> FreeSWITCH will send 0000000000 if the caller id is not specified.
>>>>> Airespring wants something formatted like
>>>>>
>>>>> INVITE sip:jungar at alpha-org.com SIP/2.0
>>>>>
>>>>> Via: SIP/2.0/UDP zetamachine.beta-org.com:5060;branch=as82je8ei4kr
>>>>>
>>>>> Max-Forwards: 70
>>>>>
>>>>> To:  2556112121 <sip:jungar at alpha-org.com>
>>>>>
>>>>> From: <sip:jsmith at beta-org.com>;tag=13456
>>>>>
>>>>> Call-ID: @zetamachine.beta-org.com
>>>>>
>>>>>
>>>>> with the From: field looking like any of the following
>>>>>
>>>>> From: <sip:jsmith at beta-org.com>;tag=13456
>>>>> From: <  @208.76.54.59>;tag=13456
>>>>> From:  @208.76.54.59;tag=13456
>>>>>
>>>>> The closest I got was to make FreeSWITCH to the following, but
>>>>> airespring won't budge. Can anyone tell me how to make the From string as
>>>>> airespring wants for their national DID presence service. Thanks.
>>>>> From: "" <sip:0000000000 at 208.76.54.59>;tag=UyNBXjyUUay8m
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 anthony.minessale at gmail.com  Fri Feb 11 23:44:45 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 11 Feb 2011 14:44:45 -0600
Subject: [Freeswitch-users] 503 Maximum Calls In Progress
In-Reply-To: <fc75ff0036de.4d553a1e@insightbb.com>
References: <fc75ff0036de.4d553a1e@insightbb.com>
Message-ID: <AANLkTi=kz_kchMuEXMHeoYw=_z083bmT2NxXUqqrnFte@mail.gmail.com>

I bet you pressed F11 which issued fsctl pause in older config sets.


On Fri, Feb 11, 2011 at 12:31 PM, Gary Chen <gchen00 at insightbb.com> wrote:
> After restart, everything back to normal. I can make call again.
>
> I will do more tests. I will let you know if it happen again.
>
>
>
> Gary
>
>
>
> ________________________________
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
> Collins
> Sent: Friday, February 11, 2011 12:54 PM
>
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] 503 Maximum Calls In Progress
>
>
>
> If you stop freeswitch and restart it does it still say that or are you able
> to make some calls for a while before it throws that error?
>
> -MC
>
> On Fri, Feb 11, 2011 at 11:26 AM, Gary Chen <gchen00 at insightbb.com> wrote:
>
> Yes. I did some test calls between two Cisco sip phones and they all work
> fine and then it just stop working. At that point, FS is still up and I can
> see the system information by press F keys. It just won?t accept any calls.
> I looked at log file and it just has that one line:
>
> 2011-02-11 17:39:58.691008 [CRIT] switch_core_session.c:1629 The system
> cannot create any sessions at this time.
>
>
>
> The register message is still working at that time.
>
>
>
> Gary
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb 12 00:06:43 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 11 Feb 2011 15:06:43 -0600
Subject: [Freeswitch-users] hold_music var gets unset/lost by
 cancelled/failed att_xfer
In-Reply-To: <AANLkTimjWc_Tn2Du0w_Zgy2VihsN8Av40M1UAQV7NXtT@mail.gmail.com>
References: <AANLkTimLod3yHrex4LAK=4gdVcPhzXutKYyw=E8ZMcN9@mail.gmail.com>
	<AANLkTi=u3VeHPuS5x1e+yu5QNBw5ZSgHvAWOT+F6A_OR@mail.gmail.com>
	<AANLkTi=EndkiUJvCgCSirxCA23EwWD_++0h3Qcr=mg9g@mail.gmail.com>
	<AANLkTikt97dugbhKPgnB2ws4bOFcxg8yhzSdyxHMab_m@mail.gmail.com>
	<AANLkTikEx9ZwNADRQ8k2JbTwgoLPFmHUPpPL3RLnZZU8@mail.gmail.com>
	<AANLkTinS3wxA8izr9tXX7jhYJPHq_X_coO7UNoUkjs-P@mail.gmail.com>
	<AANLkTi=Hv7kkYeD8TQeCZtfvmzqaCx+xQ=yYC16kpNfa@mail.gmail.com>
	<AANLkTikKt8TC-g3Q5LpXEZh62dGY6+f=HRO00n4G4PU3@mail.gmail.com>
	<AANLkTimjWc_Tn2Du0w_Zgy2VihsN8Av40M1UAQV7NXtT@mail.gmail.com>
Message-ID: <AANLkTimFbvsfk9bTMORWXeWEAa-68WG5nJ=KFpF94gk-@mail.gmail.com>

bowout_on_execute is only useful to 1 leg calls you never have to set it.
loopback_bowout is true by default it tries to cut loopback out by
doing uuid_bridge which is certainly why it breaks for you.  The last
patch which is now reverted, prevented bowout when the CF_INNER_BRIDGE
flag was set which is the case during att_xfer.

You could save yourself some time and call your own box back over sip instead.



On Fri, Feb 11, 2011 at 7:22 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> Maybe you are right, but then tell me the right way :) att_xfer uses
> loopback by itself, and it doesn't work without tricks with not well
> documented loopback_bowout/bowout_on_execute variables... This
> behavior is not documented anywhere, but I can't propose wiki updates
> until I get it working or just give up with att_xfer in FS at all :)
>
> All I need is just working in-call att_xfer with Sofia channels :( SIP
> transfers are cool, but ?I need in-call DTMF triggered transfers :((
>
> Is there a developer documentation about FS basic work principles?
> I'll try to dive in the code :))
>
> 2011/2/11 Anthony Minessale <anthony.minessale at gmail.com>:
>> I reverted the patch, I guess it doesn't work based on your feedback.
>> We will leave well enough alone when you said in your last email
>> everything was working.
>>
>> There is a time when maybe you are taking things too far trying to
>> solve things a silly way and maybe you should consider nicer
>> equipment.
>>
>>
>> On Fri, Feb 11, 2011 at 4:43 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>> I see your patch in git log, is it enough to do 'make current', or I
>>> should apply it in some special way? After make current issues are
>>> still present...
>>>
>>> What should i set loopback_bowout/bowout_on_execute variables to? I
>>> tried with false, this works as in previous versions, and loopback
>>> stays in path during the call. While loopback channels are present in
>>> path, there are issues with sound quality, there is robot voice effect
>>> for short periods of time. You mentioned setting rtp timer to none,
>>> where this should be done? In loopback channel execute_extension while
>>> doing att_xfer?
>>>
>>> When bowout variables set to true ?att_xfer doesn't work just as I
>>> described in my previous posts, and there is also issue with MOH -
>>> after transfer target answers to transferor, MOH is stoped for
>>> transferee. Maybe this can help somehow.
>>>
>>>
>>>
>>>
>>> 2011/2/10 Anthony Minessale <anthony.minessale at gmail.com>:
>>>> I pushed a patch that will probably delay the bowout until after the
>>>> att_xfer is over
>>>> Give it a try.
>>>>
>>>> commit 3546654615f88058fb6769fe79e07162602fa4af
>>>> Author: Anthony Minessale <anthm at freeswitch.org>
>>>> Date: ? Thu Feb 10 12:37:14 2011 -0600
>>>>
>>>> ? ?don't bow out on att_xfer bridge
>>>>
>>>>
>>>> On Thu, Feb 10, 2011 at 1:03 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>>>> Thanks! It works now! BTW, att_xfer with
>>>>> loopback_bowout/bowout_on_execute set to false seems to be working
>>>>> too. Is there a way to make loopback channel leave the path? Without
>>>>> bowout turned off att_xfer doesn't work...
>>>>>
>>>>> 2011/2/8 Anthony Minessale <anthony.minessale at gmail.com>:
>>>>>> try latest GIT
>>>>>>
>>>>>> On Tue, Feb 8, 2011 at 1:28 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>>>>>> I have same issue with MOH and att_xfer on failed transfers, music on
>>>>>>> hold played only once
>>>>>>> At the same time, transfer_ringback always plays correctly to transferer
>>>>>>>
>>>>>>> 2011/2/8 Anthony Minessale <anthony.minessale at gmail.com>:
>>>>>>>> you really should report this to jira not to the mailing list.
>>>>>>>> http://jira.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Mon, Feb 7, 2011 at 4:23 PM, Fraser Redmond <fraserredmond at gmail.com> wrote:
>>>>>>>>> I'm trying to do a second att_xfer on a call so that if the first attended
>>>>>>>>> transfer fails (c-leg is busy, or presses do-not-answer, or is an extn that
>>>>>>>>> doesn't exist) then the call could be transferred to someone else.
>>>>>>>>>
>>>>>>>>> On the first att_xfer the person on hold hears the hold_music correctly.
>>>>>>>>> Once that transfer is cancelled or fails:
>>>>>>>>> -- On any subsequent att_xfer's the person on hold just hears silence.
>>>>>>>>> -- If they are put on hold they just hear silence.
>>>>>>>>>
>>>>>>>>> I tried setting hold_music again for each channel after the first att_xfer,
>>>>>>>>> but that didn't work, so it's probably not actually a problem with
>>>>>>>>> hold_music per se, but some other variable/setting that decides whether to
>>>>>>>>> use hold_music.
>>>>>>>>>
>>>>>>>>> I also tried doing a uuid_dump before and after each attempt, but didn't
>>>>>>>>> notice anything too different - unless it's a matter of unsetting one of the
>>>>>>>>> couple of changed/new vars like:
>>>>>>>>> variable_originate_disposition
>>>>>>>>> variable_current_application
>>>>>>>>> variable_playback_seconds
>>>>>>>>>
>>>>>>>>> I get the feeling other variables are probably also lost by the first failed
>>>>>>>>> transfer as the second att_xfer has some odd things happen if the third
>>>>>>>>> party does answer. Haven't been able to narrow it down as closely as the
>>>>>>>>> hold_music, but two things I've seen happen are:
>>>>>>>>> -- The party that initiated the transfer gets hung up automatically (after
>>>>>>>>> 30 sec)
>>>>>>>>> -- When the party that initiated the transfer hangs up it should connect the
>>>>>>>>> other two parties, but instead it hung up all three
>>>>>>>>>
>>>>>>>>> Cheers,
>>>>>>>>> Fraser
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at 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
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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 frank at telonium.com  Sat Feb 12 01:26:16 2011
From: frank at telonium.com (Frank Park)
Date: Fri, 11 Feb 2011 17:26:16 -0500
Subject: [Freeswitch-users] Intermittent curl execution after rxfax
Message-ID: <AANLkTi=qqaP0bAhdVMW309fm2vfr41BP13c81epOieW0@mail.gmail.com>

Hello,

I am trying to troubleshoot this dialplan. I have a dial that receives the
fax and executes a curl that invokes a script on a remote site (emails the
fax, and does few more things).
rxfax works fine and saves the tiff file accordingly, but every so often
(about 50% of the time), curl call doesn't get executed. It doesn't even
show up on the fs_cli log of it being executed, not to mention the apache
side on the other end.


Dialplan looks as follows actually values are obfuscated:
        <condition field="destination_number" expression="^(99999)$">
          <action application="answer"/>
          <action application="playback" data="silence_stream://2000"/>
          <action application="rxfax"
data="/usr/local/freeswitch/storage/fax/${uuid}.tiff"/>
          <action application="curl" data="http://abc.test post
uuid=${uuid}"/>
          <action application="hangup"/>
        </condition>

Thanks,
Frank

-- 

----=======================----
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 kris at livecall.com  Sat Feb 12 02:23:37 2011
From: kris at livecall.com (Kris)
Date: Fri, 11 Feb 2011 15:23:37 -0800
Subject: [Freeswitch-users] How to set up a DTMF callback in C or C#?
References: <812941.11103.qm@web30507.mail.mud.yahoo.com>
Message-ID: <3F83903C359140E3BF786504D7507457@stor1>


        protected string DtmfReceived(Char Digit, TimeSpan 
DurationMilliseconds) //t is the duration of the 
tone -TimeSpan.FromMilliseconds(dtmf.duration));
        {
            SaveDigit(Digit.ToString());
            BaseLog.WriteLine(BaseLogLevel.Info, CallID, sv_uuid, 
CurrentAction, Termination.Reason, "DtmfReceived {0} Duration:{1}", 
Digit.ToString(), DurationMilliseconds.ToString());
            //
            if (Termination.TerminatorDigits.IndexOf(Digit) >= 0 || 
Termination.TerminatorDigits.IndexOf('@') >= 0) //@ means any digit
                return "break";  //returning anything breaks play
            if (Termination.MaximumDigits > 0) //break the play
                return "break";  //returning anything but 
SWITCH_STATUS_SUCCESS breaks play
            return String.Empty;// play continues
        }
        void SetDTMFFunction(bool set)
        {
            if (Session != null && Session.Ready())
            {
                if (set)
                    Session.DtmfReceivedFunction = DtmfReceived;// 
public Func<Char, TimeSpan, string>
                else
                    Session.DtmfReceivedFunction = null;
            }
        }

----- Original Message ----- 
From: "Frankie Yiu" <yky1628 at yahoo.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Friday, February 11, 2011 12:13 PM
Subject: [Freeswitch-users] How to set up a DTMF callback in C or C#?


Hi there,

If I want to set up a DTMF callBack to a function (in either C or C#) when a
caller presses a key, what function should I use to set it up?

Thanks,
Frankie





From chenzhanping at gmail.com  Sat Feb 12 03:57:01 2011
From: chenzhanping at gmail.com (=?GB2312?B?1bzGvbPC?=)
Date: Sat, 12 Feb 2011 08:57:01 +0800
Subject: [Freeswitch-users] Question about mod_dingaling.
In-Reply-To: <AANLkTinanoYB3J-0rmYtM7uw5m5HJC829YmZbxaHWzVh@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>
Message-ID: <AANLkTikQnNUSn596FZsm4ZHKi4t_UQMS1JoVfPki+on0@mail.gmail.com>

I use the configuration in http://wiki.freeswitch.org/wiki/Google_Voice ,but
still can not call out from mod_dingaling use google voice.

this is log:

http://pastebin.freeswitch.org/15363

Is there a problem here:  Thanks.

   1.  2011-02-12 08:50:07.335702 [DEBUG] mod_dingaling.c:892 SETUP RTP
   184.105.153.247:0 -> 74.125.127.126:19295
   2.  2011-02-12 08:50:07.335702 [DEBUG] mod_dingaling.c:914 RTP ERROR
   Missing local port
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From chenzhanping at gmail.com  Sat Feb 12 03:58:01 2011
From: chenzhanping at gmail.com (=?GB2312?B?1bzGvbPC?=)
Date: Sat, 12 Feb 2011 08:58:01 +0800
Subject: [Freeswitch-users] Question about mod_dingaling.
In-Reply-To: <AANLkTikQnNUSn596FZsm4ZHKi4t_UQMS1JoVfPki+on0@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>
Message-ID: <AANLkTikpgKwNmbnd1F_XoUFqu93tcPi3B=vJkE_jRhz5@mail.gmail.com>

client.xml:

<include>
  <!-- Client Profile (Original mode) -->
  <!-- to use this profile take the x- away from the open and close tags so
its <profile> and </profile> -->
  <profile type="client">
    <param name="name" value="gv1"/>
    <param name="login" value="XXXXX at gmail.com/talk"/>
    <param name="password" value="XXXXXX"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="message" value="Jingle all the way"/>
    <param name="rtp-ip" value="$${bind_server_ip}"/>
    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="auto-login" value="true"/>
    <param name="auto-reply" value="Press *Call* to join my conference"/>
    <!-- SASL "plain" or "md5" -->
    <param name="sasl" value="plain"/>
    <!-- if the server where the jabber is hosted is not the same as the one
in the jid -->
    <param name="server" value="talk.google.com"/>-->
    <!-- Enable TLS or not -->
    <param name="tls" value="true"/>
    <!-- disable to trade async for more calls -->
    <param name="use-rtp-timer" value="true"/>
    <!-- default extension (if one cannot be determined) -->
    <param name="exten" value="1000"/>
    <!-- VAD choose one -->
    <!-- <param name="vad" value="in"/> -->
    <!-- <param name="vad" value="out"/> -->
    <param name="vad" value="both"/>
    <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
    <!--<param name="candidate-acl" value="wan.auto"/>-->
    <param name="local-network-acl" value="localnet.auto"/>
  </profile>
</include>


? 2011?2?12? ??8:57???? <chenzhanping at gmail.com>???

> I use the configuration in http://wiki.freeswitch.org/wiki/Google_Voice ,but
> still can not call out from mod_dingaling use google voice.
>
> this is log:
>
> http://pastebin.freeswitch.org/15363
>
> Is there a problem here:  Thanks.
>
>    1.  2011-02-12 08:50:07.335702 [DEBUG] mod_dingaling.c:892 SETUP RTP
>    184.105.153.247:0 -> 74.125.127.126:19295
>    2.  2011-02-12 08:50:07.335702 [DEBUG] mod_dingaling.c:914 RTP ERROR
>    Missing local port
>
>
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From david.villasmil.work at gmail.com  Sat Feb 12 04:37:22 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 12 Feb 2011 02:37:22 +0100
Subject: [Freeswitch-users] mod_distributor and dial strings
Message-ID: <AANLkTins0BSsBxu3uBL-yV5VCU3puwJFgyzgBAZBtoEw@mail.gmail.com>

hello all

I'm playing with mod_distributor... it's nice.. but i'm having this problem:

suppose i have 2 gateways i want to use in round-trip fashion.. that
works fine... but suppose i must send to gw1 with a prefix and to gw2
with a different prefix... is this possible?

thanks

David


From infos at madovsky.org  Sat Feb 12 04:46:19 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 11 Feb 2011 20:46:19 -0500
Subject: [Freeswitch-users] mod_distributor and dial strings
References: <AANLkTins0BSsBxu3uBL-yV5VCU3puwJFgyzgBAZBtoEw@mail.gmail.com>
Message-ID: <29452B0C55CF4C6C92E0AB760DAFE074@e1705>

it remembers me my FS start.
had the same question one year ago...
you have to use a script or play with your dialplan to use different prefix,
you can also use mod_easyroute and mod_lcr.

----- Original Message ----- 
From: "David Villasmil" <david.villasmil.work at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, February 11, 2011 8:37 PM
Subject: [Freeswitch-users] mod_distributor and dial strings


> hello all
>
> I'm playing with mod_distributor... it's nice.. but i'm having this 
> problem:
>
> suppose i have 2 gateways i want to use in round-trip fashion.. that
> works fine... but suppose i must send to gw1 with a prefix and to gw2
> with a different prefix... is this possible?
>
> 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 



From david.villasmil.work at gmail.com  Sat Feb 12 05:10:20 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 12 Feb 2011 03:10:20 +0100
Subject: [Freeswitch-users] mod_distributor and dial strings
In-Reply-To: <29452B0C55CF4C6C92E0AB760DAFE074@e1705>
References: <AANLkTins0BSsBxu3uBL-yV5VCU3puwJFgyzgBAZBtoEw@mail.gmail.com>
	<29452B0C55CF4C6C92E0AB760DAFE074@e1705>
Message-ID: <AANLkTimw02-WSQqENkaTzC7B7iwxwi5sU_aGbP58si3G@mail.gmail.com>

Hello and thanks for answering,

easyroute looks good... thing now is: how about failover and
distribution based on weight? :(
I'm actually looking for something like mod_distributor/mod_easyroute
combined... is there such a module?

Thanks again

David

On Sat, Feb 12, 2011 at 2:46 AM, Madovsky <infos at madovsky.org> wrote:
> it remembers me my FS start.
> had the same question one year ago...
> you have to use a script or play with your dialplan to use different prefix,
> you can also use mod_easyroute and mod_lcr.
>
> ----- Original Message -----
> From: "David Villasmil" <david.villasmil.work at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Friday, February 11, 2011 8:37 PM
> Subject: [Freeswitch-users] mod_distributor and dial strings
>
>
>> hello all
>>
>> I'm playing with mod_distributor... it's nice.. but i'm having this
>> problem:
>>
>> suppose i have 2 gateways i want to use in round-trip fashion.. that
>> works fine... but suppose i must send to gw1 with a prefix and to gw2
>> with a different prefix... is this possible?
>>
>> 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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Sat Feb 12 05:29:40 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 11 Feb 2011 20:29:40 -0600
Subject: [Freeswitch-users] mod_distributor and dial strings
In-Reply-To: <AANLkTins0BSsBxu3uBL-yV5VCU3puwJFgyzgBAZBtoEw@mail.gmail.com>
References: <AANLkTins0BSsBxu3uBL-yV5VCU3puwJFgyzgBAZBtoEw@mail.gmail.com>
Message-ID: <AANLkTikhTC0WRqyuR6HraBsaCZYadOLAvKZ=KyMSCfNz@mail.gmail.com>

think about it.
it returns any string you want
mix that with variable expansion on the xml DP.


On Fri, Feb 11, 2011 at 7:37 PM, David Villasmil
<david.villasmil.work at gmail.com> wrote:
> hello all
>
> I'm playing with mod_distributor... it's nice.. but i'm having this problem:
>
> suppose i have 2 gateways i want to use in round-trip fashion.. that
> works fine... but suppose i must send to gw1 with a prefix and to gw2
> with a different prefix... is this possible?
>
> 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
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 Feb 12 05:46:14 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 11 Feb 2011 20:46:14 -0600
Subject: [Freeswitch-users] Question about mod_dingaling.
In-Reply-To: <AANLkTikpgKwNmbnd1F_XoUFqu93tcPi3B=vJkE_jRhz5@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>
Message-ID: <AANLkTin6E8nKYdx_zn5gLNzoV-oaQpkpmtFDCEDDroGi@mail.gmail.com>

change $${bind_server_ip} to whatever the box's eth0 ip is.


2011/2/11 ??? <chenzhanping at gmail.com>:
> client.xml:
>
> <include>
>   <!-- Client Profile (Original mode) -->
>   <!-- to use this profile take the x- away from the open and close tags so
> its <profile> and </profile> -->
>   <profile type="client">
>     <param name="name" value="gv1"/>
>     <param name="login" value="XXXXX at gmail.com/talk"/>
>     <param name="password" value="XXXXXX"/>
>     <param name="dialplan" value="XML"/>
>     <param name="context" value="public"/>
>     <param name="message" value="Jingle all the way"/>
>     <param name="rtp-ip" value="$${bind_server_ip}"/>
>     <param name="ext-rtp-ip" value="auto-nat"/>
>     <param name="auto-login" value="true"/>
>     <param name="auto-reply" value="Press *Call* to join my conference"/>
>     <!-- SASL "plain" or "md5" -->
>     <param name="sasl" value="plain"/>
>     <!-- if the server where the jabber is hosted is not the same as the one
> in the jid -->
>     <param name="server" value="talk.google.com"/>-->
>     <!-- Enable TLS or not -->
>     <param name="tls" value="true"/>
>     <!-- disable to trade async for more calls -->
>     <param name="use-rtp-timer" value="true"/>
>     <!-- default extension (if one cannot be determined) -->
>     <param name="exten" value="1000"/>
>     <!-- VAD choose one -->
>     <!-- <param name="vad" value="in"/> -->
>     <!-- <param name="vad" value="out"/> -->
>     <param name="vad" value="both"/>
>     <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
>     <!--<param name="candidate-acl" value="wan.auto"/>-->
>     <param name="local-network-acl" value="localnet.auto"/>
>   </profile>
> </include>
>
>
> ? 2011?2?12? ??8:57???? <chenzhanping at gmail.com>???
>>
>> I use the configuration in
>> http://wiki.freeswitch.org/wiki/Google_Voice ,but still can not call out
>> from mod_dingaling use google voice.
>>
>> this is log:
>>
>> http://pastebin.freeswitch.org/15363
>>
>> Is there a problem here:  Thanks.
>>
>> 2011-02-12 08:50:07.335702 [DEBUG] mod_dingaling.c:892 SETUP RTP
>> 184.105.153.247:0 -> 74.125.127.126:19295
>> 2011-02-12 08:50:07.335702 [DEBUG] mod_dingaling.c:914 RTP ERROR Missing
>> local port
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 david.villasmil.work at gmail.com  Sat Feb 12 05:49:10 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 12 Feb 2011 03:49:10 +0100
Subject: [Freeswitch-users] mod_distributor and dial strings
In-Reply-To: <AANLkTikhTC0WRqyuR6HraBsaCZYadOLAvKZ=KyMSCfNz@mail.gmail.com>
References: <AANLkTins0BSsBxu3uBL-yV5VCU3puwJFgyzgBAZBtoEw@mail.gmail.com>
	<AANLkTikhTC0WRqyuR6HraBsaCZYadOLAvKZ=KyMSCfNz@mail.gmail.com>
Message-ID: <AANLkTi=LQr2jYKyEz3Ayh7a0Q2+gcLLSThT1VOVJ1Wqw@mail.gmail.com>

Anthony,

Yes, i just found out in the irc... nice trick, i will try it now.

thanks

David

On Sat, Feb 12, 2011 at 3:29 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> think about it.
> it returns any string you want
> mix that with variable expansion on the xml DP.
>
>
> On Fri, Feb 11, 2011 at 7:37 PM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
>> hello all
>>
>> I'm playing with mod_distributor... it's nice.. but i'm having this problem:
>>
>> suppose i have 2 gateways i want to use in round-trip fashion.. that
>> works fine... but suppose i must send to gw1 with a prefix and to gw2
>> with a different prefix... is this possible?
>>
>> 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
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 bwibowo at gmail.com  Sat Feb 12 10:27:03 2011
From: bwibowo at gmail.com (Budi wibowo)
Date: Sat, 12 Feb 2011 07:27:03 +0000
Subject: [Freeswitch-users] mod_dingaling & inbound audio
In-Reply-To: <AANLkTikoJcrERbEHbdkLM_4ocnCUKkYDoKE-miYbvTxj@mail.gmail.com>
References: <AANLkTim-5LDhfQhHk_2LNLffvTa1JyBXq9U4B3LSY3Nq@mail.gmail.com><AANLkTik19FmdGVEukmC4rS3ytu9qhR=wfSkO-fhaRMJQ@mail.gmail.com><AANLkTikoJcrERbEHbdkLM_4ocnCUKkYDoKE-miYbvTxj@mail.gmail.com>
Message-ID: <1977194672-1297495611-cardhu_decombobulator_blackberry.rim.net-1261789055-@b25.c2.bise3.blackberry>

On previous git version I have working mod dingaling, but until today my dingaling is never working. 
If anybody has freeswitch source with dingaling works, please share the url. I will download 


Thx
Budi
-----Original Message-----
From: Locutis of Borg <locutis at sect001.net>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Wed, 9 Feb 2011 14:33:02 
To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] mod_dingaling & inbound audio

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at 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 kris at livecall.com  Sat Feb 12 11:32:54 2011
From: kris at livecall.com (Kris)
Date: Sat, 12 Feb 2011 00:32:54 -0800
Subject: [Freeswitch-users] How to set up a DTMF callback in C or C#?
References: <582070.8303.qm@web30505.mail.mud.yahoo.com>
Message-ID: <A876649131D642C992E4A9485AEE962A@stor1>


 I'll try to put it on the list again.

I don't use any of those functions, but there is a way to do it somehow with 
delegates(the C# pointers). I use the API like this. I don't know much..3 
months ago I didn't know anything about C# or FreeSwitch. Just have to read 
and try things since there aren't many C# examples out there.

           FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api();

                                                        string filename;

                                                        filename = 
@"E:\LiveMatch\Sounds\en\us\female\conference\conf-help_menu.vox";
                                                        fsApi.Execute("uuid_displace", 
ev_all.GetHeader("Caller-Unique-ID") + " start " + filename + " 10 mux");

----- Original Message ----- 
From: "Frankie Yiu" <yky1628 at yahoo.com>
To: <kris at livecall.com>
Sent: Friday, February 11, 2011 7:03 PM
Subject: Re: [Freeswitch-users] How to set up a DTMF callback in C or C#?


Thanks for your reply earlier, it works.
For some reason, I can not reply the thread so I send you this
message privately.

Since you might know about using C# w/ FreeSwitch, I might ask you this:

1) If I want to use the Media bugs, how can I do this in C#? I see a
method that I want to use in swig.cs: "switch_core_media_bug_add", but what 
is
the parameters for the call back function?


>From the definition in switch_types.h:
switch_bool_t (*switch_media_bug_callback_t) (switch_media_bug_t *, void *,
switch_abc_type_t);

but what is it in C# since it does not have pointers?

2) I see that some functions have type like this "SWIGTYPE_p_p_ ..." (for
example: SWIGTYPE_p_p_switch) , do these type really work (expecially for 
those
pointer to pointer type in C/C++)? I tried to call:
"switch_core_session_read_frame(SWIGTYPE_p_switch_core_session session,
SWIGTYPE_p_p_switch_frame frame, uint flags, int stream_id) "

but seem like when I pass in a variable for frame, when the program gets to
freeswitch_wrap.cxx side, it lost its frame's memory address/place holder. 
Am I
doing something wrong?

Thanks in advance.

Frankie



____________________________________________________________________________________
Bored stiff? Loosen up...
Download and play hundreds of games for free on Yahoo! Games.
http://games.yahoo.com/games/front 



From steveayre at gmail.com  Sat Feb 12 12:24:05 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 12 Feb 2011 09:24:05 +0000
Subject: [Freeswitch-users] Intermittent curl execution after rxfax
In-Reply-To: <AANLkTi=qqaP0bAhdVMW309fm2vfr41BP13c81epOieW0@mail.gmail.com>
References: <AANLkTi=qqaP0bAhdVMW309fm2vfr41BP13c81epOieW0@mail.gmail.com>
Message-ID: <4961923E-BF1C-4202-9441-A2FF2B7E5E57@gmail.com>

Perhaps the aleg hangs up after rxfax but before curl is executed? If the aleg hangs up the remaining extension actions won't run.

Looks like you're only using it to indicate a fax has been received? Try using ESL or CDRs. Mod_xml_cdr can post a CDR to a server and will have various variables set that indicate the fax has been received.

Steve on iPhone

On 11 Feb 2011, at 22:26, Frank Park <frank at telonium.com> wrote:

> Hello,
> 
> I am trying to troubleshoot this dialplan. I have a dial that receives the fax and executes a curl that invokes a script on a remote site (emails the fax, and does few more things).
> rxfax works fine and saves the tiff file accordingly, but every so often (about 50% of the time), curl call doesn't get executed. It doesn't even show up on the fs_cli log of it being executed, not to mention the apache side on the other end.
> 
> 
> Dialplan looks as follows actually values are obfuscated:
>         <condition field="destination_number" expression="^(99999)$">
>           <action application="answer"/>
>           <action application="playback" data="silence_stream://2000"/>
>           <action application="rxfax" data="/usr/local/freeswitch/storage/fax/${uuid}.tiff"/>
>           <action application="curl" data="http://abc.test post uuid=${uuid}"/>
>           <action application="hangup"/>
>         </condition>
> 
> Thanks,
> Frank
> 
> -- 
> 
> ----=======================----
> 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
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From ovvenkatesan at gmail.com  Sat Feb 12 13:51:02 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Sat, 12 Feb 2011 16:21:02 +0530
Subject: [Freeswitch-users] call getting hangup after git update
In-Reply-To: <AANLkTimptf=rLg4SFV4bzseUf9zQwqo_Ee-yh7U4_UKB@mail.gmail.com>
References: <AANLkTi=z5KOP0sHwdYg+=tiF38tDS_X6QY1nk7ehuuLQ@mail.gmail.com>
	<2B571B49-1886-445D-8473-C34CC9039301@freeswitch.org>
	<AANLkTikqr10zdWni96-RUWyRrpzus+J1God_yh7JiH9A@mail.gmail.com>
	<AANLkTimptf=rLg4SFV4bzseUf9zQwqo_Ee-yh7U4_UKB@mail.gmail.com>
Message-ID: <AANLkTim8i7B_hztSmTXxrzPw25ef-Aycnb=ZcqKFbObc@mail.gmail.com>

HI.

It works now, thank you very much.

Regards,
Venkat.


On Sat, Jan 8, 2011 at 9:12 AM, Sam <u2nsam at gmail.com> wrote:

> Hello,
>
> Set NPI/TON to unknown and it will work !
>
> Regards
> Sam
>
>
>
>
>
> On Fri, Jan 7, 2011 at 10:55 PM, ovvenkat <ovvenkatesan at gmail.com> wrote:
>
>> Hi Brian,
>>
>> On Fri, Jan 7, 2011 at 8:44 PM, Brian West <brian at freeswitch.org> wrote:
>>
>>> You're getting no route back from your freetdm circuit.
>>>
>>>
>>
>> When I am calling to mobile number, it works fine.
>> Its giving the problem only with fixed line numbers.
>> Can you please tell , why its so?
>>
>>
>>
>>
>>>  /b
>>>
>>> On Jan 7, 2011, at 8:39 AM, ovvenkat wrote:
>>>
>>>
>>> Hi,
>>>
>>> Today, I have updated freeSwitch to latest git.
>>> After that, When I am trying to do outbound call,
>>> Call is getting hangup. When I check the fs_cli
>>> logs, its showing that,
>>>
>>> *mod_dptools.c:2610 Originate Failed.  Cause: NO_ROUTE_DESTINATION*
>>>
>>> Here is the logs for the same
>>> http://pastebin.freeswitch.org/14951
>>>
>>> I have added STD code , before the number,
>>> Still no luck. I am getting error like
>>>
>>> *Originate Failed.  Cause: NORMAL_UNSPECIFIED*
>>>
>>>
>>> Here is the log after adding STD code
>>>
>>> http://pastebin.freeswitch.org/14952
>>>
>>>
>>> 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.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From emss.mail at gmail.com  Sat Feb 12 15:07:52 2011
From: emss.mail at gmail.com (Eric Masson)
Date: Sat, 12 Feb 2011 13:07:52 +0100
Subject: [Freeswitch-users] can't email voicemail messages
In-Reply-To: <4D542B72.9010806@strongholdwax.com>
References: <4D542B72.9010806@strongholdwax.com>
Message-ID: <ij5t6r$7l4$1@dough.gmane.org>

Le 10/02/2011 19:16, Roger Young a ?crit :

Hello,

> Outbound mail is rejected because the "from" address is
> "1000 at localcomputername".

I suppose you have not changed the default domain on your FS 
installation, right ?

In this case, Postfix rejects the mail because of the enveloppe sender 
(right part of the enveloppe from is the ip address of the box).

You have 2 solutions :
- change the domain in vars.xml to something else than $${local_ip_v4}
- change postfix configuration not to enforce strict RFC821 checks.

Regards

?ric Masson



From chenzhanping at gmail.com  Sat Feb 12 16:17:11 2011
From: chenzhanping at gmail.com (=?GB2312?B?1bzGvbPC?=)
Date: Sat, 12 Feb 2011 21:17:11 +0800
Subject: [Freeswitch-users] Question about mod_dingaling.
In-Reply-To: <AANLkTin6E8nKYdx_zn5gLNzoV-oaQpkpmtFDCEDDroGi@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>
Message-ID: <AANLkTi=Y-czU6pfGAgrVTvCS=oaM6_VdvSDqFkhyf965@mail.gmail.com>

I change $${bind_server_ip} to my ip,but still not to call out from google
voice.

this is new log:

http://pastebin.freeswitch.org/15366
Please help me,  Thanks a lot.
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From chenzhanping at gmail.com  Sat Feb 12 16:18:35 2011
From: chenzhanping at gmail.com (=?GB2312?B?1bzGvbPC?=)
Date: Sat, 12 Feb 2011 21:18:35 +0800
Subject: [Freeswitch-users] Question about mod_dingaling.
In-Reply-To: <AANLkTi=Y-czU6pfGAgrVTvCS=oaM6_VdvSDqFkhyf965@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>
Message-ID: <AANLkTinH-OHWPc7jHEUzBr3RxtG29Ntd+3XV8xxXAT+-@mail.gmail.com>

By the way,i use the XEN vps to install freeswitch.

2011/2/12 ??? <chenzhanping at gmail.com>

> I change $${bind_server_ip} to my ip,but still not to call out from google
> voice.
>
> this is new log:
>
> http://pastebin.freeswitch.org/15366
> Please help me,  Thanks a lot.
>
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From anthony.minessale at gmail.com  Sat Feb 12 17:45:20 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 12 Feb 2011 08:45:20 -0600
Subject: [Freeswitch-users] Question about mod_dingaling.
In-Reply-To: <AANLkTi=Y-czU6pfGAgrVTvCS=oaM6_VdvSDqFkhyf965@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>
Message-ID: <AANLkTikPa22Vh33x+xODysCQo4Q45MFd9X+=xgN8iPis@mail.gmail.com>

run freeswitch with -nonat and comment out the auto-nat line in your profile.
your upnp router is returning port 0


On Sat, Feb 12, 2011 at 7:17 AM, ??? <chenzhanping at gmail.com> wrote:
> I change $${bind_server_ip} to my ip,but still not to call out from google
> voice.
>
> this is new log:
>
> http://pastebin.freeswitch.org/15366
> Please help me,? Thanks a lot.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 david.villasmil.work at gmail.com  Sat Feb 12 18:04:42 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 12 Feb 2011 16:04:42 +0100
Subject: [Freeswitch-users] mod_distributor, lua and loop
Message-ID: <AANLkTinDD-n_J7B3VjvJcB=0iHsr6nnggUNyr0Nk8O7a@mail.gmail.com>

Hello All,

from a lua script, i'm executing the following:


session:execute("bridge","sofia/gateway/${distributor(" .. route_name
.. ")}".. out_number .."")

but the call doesn't falls to the next gw, i'm trying to add the
loop="3" to the bridge but it doesn't work.

I've trued like this:


session:execute("set","loop=\"3\"")
session:execute("bridge","sofia/gateway/${distributor(" .. route_name
.. ")}".. out_number .."")

and


session:execute("bridge","sofia/gateway/${distributor(" .. route_name
.. ")}".. out_number .." loop=\"3\"")

and

session:execute("bridge","{loop=\"3\"}sofia/gateway/${distributor(" ..
route_name .. ")}".. out_number .."")


Still nothing...

any thoughts?

Thanks

David


From david.villasmil.work at gmail.com  Sat Feb 12 18:06:56 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 12 Feb 2011 16:06:56 +0100
Subject: [Freeswitch-users] mod_distributor, lua and loop
In-Reply-To: <AANLkTinDD-n_J7B3VjvJcB=0iHsr6nnggUNyr0Nk8O7a@mail.gmail.com>
References: <AANLkTinDD-n_J7B3VjvJcB=0iHsr6nnggUNyr0Nk8O7a@mail.gmail.com>
Message-ID: <AANLkTik3m+zr3BLugtVG1+J9a7jK57fJHcWuvxygse76@mail.gmail.com>

Oh I forgot,

I also have:

session:execute("set","continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407")

thanks

David

On Sat, Feb 12, 2011 at 4:04 PM, David Villasmil
<david.villasmil.work at gmail.com> wrote:
> Hello All,
>
> from a lua script, i'm executing the following:
>
>
> session:execute("bridge","sofia/gateway/${distributor(" .. route_name
> .. ")}".. out_number .."")
>
> but the call doesn't falls to the next gw, i'm trying to add the
> loop="3" to the bridge but it doesn't work.
>
> I've trued like this:
>
>
> session:execute("set","loop=\"3\"")
> session:execute("bridge","sofia/gateway/${distributor(" .. route_name
> .. ")}".. out_number .."")
>
> and
>
>
> session:execute("bridge","sofia/gateway/${distributor(" .. route_name
> .. ")}".. out_number .." loop=\"3\"")
>
> and
>
> session:execute("bridge","{loop=\"3\"}sofia/gateway/${distributor(" ..
> route_name .. ")}".. out_number .."")
>
>
> Still nothing...
>
> any thoughts?
>
> Thanks
>
> David
>


From jeff at jefflenk.com  Sat Feb 12 20:06:32 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sat, 12 Feb 2011 09:06:32 -0800 (PST)
Subject: [Freeswitch-users] How to set up a DTMF callback in C or C#?
In-Reply-To: <812941.11103.qm@web30507.mail.mud.yahoo.com>
References: <812941.11103.qm@web30507.mail.mud.yahoo.com>
Message-ID: <1297530392276-6019100.post@n2.nabble.com>


The code sample with the source also has a complete example for this.

src\mod\languages\mod_managed\managed\demo.csx

When this type of file is copied to managed in your runtime location fs will
dynamically compile it - watch the cli.

Look at the dialplan and example section on the mod_managed Wiki
http://wiki.freeswitch.org/wiki/Mod_managed

replace <action application="managed" data="AppDemo"/>

-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-set-up-a-DTMF-callback-in-C-or-C-tp6017058p6019100.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Sat Feb 12 20:26:54 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 12 Feb 2011 17:26:54 +0000
Subject: [Freeswitch-users] mod_distributor, lua and loop
In-Reply-To: <AANLkTik3m+zr3BLugtVG1+J9a7jK57fJHcWuvxygse76@mail.gmail.com>
References: <AANLkTinDD-n_J7B3VjvJcB=0iHsr6nnggUNyr0Nk8O7a@mail.gmail.com>
	<AANLkTik3m+zr3BLugtVG1+J9a7jK57fJHcWuvxygse76@mail.gmail.com>
Message-ID: <1D6E8634-BF1E-42C4-91DC-B776B87CD894@gmail.com>

I believe loop is specific to XML dialplan tags.

Try implementing a lua loop, checking the hangup cause after the bridge and looping if it looks bad.

Steve on iPhone


On 12 Feb 2011, at 15:06, David Villasmil <david.villasmil.work at gmail.com> wrote:

> Oh I forgot,
> 
> I also have:
> 
> session:execute("set","continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407")
> 
> thanks
> 
> David
> 
> On Sat, Feb 12, 2011 at 4:04 PM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
>> Hello All,
>> 
>> from a lua script, i'm executing the following:
>> 
>> 
>> session:execute("bridge","sofia/gateway/${distributor(" .. route_name
>> .. ")}".. out_number .."")
>> 
>> but the call doesn't falls to the next gw, i'm trying to add the
>> loop="3" to the bridge but it doesn't work.
>> 
>> I've trued like this:
>> 
>> 
>> session:execute("set","loop=\"3\"")
>> session:execute("bridge","sofia/gateway/${distributor(" .. route_name
>> .. ")}".. out_number .."")
>> 
>> and
>> 
>> 
>> session:execute("bridge","sofia/gateway/${distributor(" .. route_name
>> .. ")}".. out_number .." loop=\"3\"")
>> 
>> and
>> 
>> session:execute("bridge","{loop=\"3\"}sofia/gateway/${distributor(" ..
>> route_name .. ")}".. out_number .."")
>> 
>> 
>> Still nothing...
>> 
>> any thoughts?
>> 
>> 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


From david.villasmil.work at gmail.com  Sat Feb 12 20:35:01 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 12 Feb 2011 18:35:01 +0100
Subject: [Freeswitch-users] mod_distributor, lua and loop
In-Reply-To: <1D6E8634-BF1E-42C4-91DC-B776B87CD894@gmail.com>
References: <AANLkTinDD-n_J7B3VjvJcB=0iHsr6nnggUNyr0Nk8O7a@mail.gmail.com>
	<AANLkTik3m+zr3BLugtVG1+J9a7jK57fJHcWuvxygse76@mail.gmail.com>
	<1D6E8634-BF1E-42C4-91DC-B776B87CD894@gmail.com>
Message-ID: <AANLkTinvZAZVT-3rdaG=mG6GKeGDRRrm_U0EdFU9B6o2@mail.gmail.com>

Thanks for answering Steve,

That's exactly what i don't want to do, because i wanted to use the
"weight" feature in mod_distributor... but i gues i will end up doing
it :)

Thanks

David

On Sat, Feb 12, 2011 at 6:26 PM, Steven Ayre <steveayre at gmail.com> wrote:
> I believe loop is specific to XML dialplan tags.
>
> Try implementing a lua loop, checking the hangup cause after the bridge and looping if it looks bad.
>
> Steve on iPhone
>
>
> On 12 Feb 2011, at 15:06, David Villasmil <david.villasmil.work at gmail.com> wrote:
>
>> Oh I forgot,
>>
>> I also have:
>>
>> session:execute("set","continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407")
>>
>> thanks
>>
>> David
>>
>> On Sat, Feb 12, 2011 at 4:04 PM, David Villasmil
>> <david.villasmil.work at gmail.com> wrote:
>>> Hello All,
>>>
>>> from a lua script, i'm executing the following:
>>>
>>>
>>> session:execute("bridge","sofia/gateway/${distributor(" .. route_name
>>> .. ")}".. out_number .."")
>>>
>>> but the call doesn't falls to the next gw, i'm trying to add the
>>> loop="3" to the bridge but it doesn't work.
>>>
>>> I've trued like this:
>>>
>>>
>>> session:execute("set","loop=\"3\"")
>>> session:execute("bridge","sofia/gateway/${distributor(" .. route_name
>>> .. ")}".. out_number .."")
>>>
>>> and
>>>
>>>
>>> session:execute("bridge","sofia/gateway/${distributor(" .. route_name
>>> .. ")}".. out_number .." loop=\"3\"")
>>>
>>> and
>>>
>>> session:execute("bridge","{loop=\"3\"}sofia/gateway/${distributor(" ..
>>> route_name .. ")}".. out_number .."")
>>>
>>>
>>> Still nothing...
>>>
>>> any thoughts?
>>>
>>> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Sat Feb 12 21:45:25 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 12 Feb 2011 12:45:25 -0600
Subject: [Freeswitch-users] mod_distributor, lua and loop
In-Reply-To: <AANLkTinvZAZVT-3rdaG=mG6GKeGDRRrm_U0EdFU9B6o2@mail.gmail.com>
References: <AANLkTinDD-n_J7B3VjvJcB=0iHsr6nnggUNyr0Nk8O7a@mail.gmail.com>
	<AANLkTik3m+zr3BLugtVG1+J9a7jK57fJHcWuvxygse76@mail.gmail.com>
	<1D6E8634-BF1E-42C4-91DC-B776B87CD894@gmail.com>
	<AANLkTinvZAZVT-3rdaG=mG6GKeGDRRrm_U0EdFU9B6o2@mail.gmail.com>
Message-ID: <AANLkTimuNPdCt+_tRkV-qEVk2nHBhjvLkNjNuHPXBPtX@mail.gmail.com>

You are misunderstanding.  He means do the line of lua code 3 times in a
row. Or use a for loop with 1 line.
On Feb 12, 2011 11:36 AM, "David Villasmil" <david.villasmil.work at gmail.com>
wrote:
> Thanks for answering Steve,
>
> That's exactly what i don't want to do, because i wanted to use the
> "weight" feature in mod_distributor... but i gues i will end up doing
> it :)
>
> Thanks
>
> David
>
> On Sat, Feb 12, 2011 at 6:26 PM, Steven Ayre <steveayre at gmail.com> wrote:
>> I believe loop is specific to XML dialplan tags.
>>
>> Try implementing a lua loop, checking the hangup cause after the bridge
and looping if it looks bad.
>>
>> Steve on iPhone
>>
>>
>> On 12 Feb 2011, at 15:06, David Villasmil <david.villasmil.work at gmail.com>
wrote:
>>
>>> Oh I forgot,
>>>
>>> I also have:
>>>
>>>
session:execute("set","continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407")
>>>
>>> thanks
>>>
>>> David
>>>
>>> On Sat, Feb 12, 2011 at 4:04 PM, David Villasmil
>>> <david.villasmil.work at gmail.com> wrote:
>>>> Hello All,
>>>>
>>>> from a lua script, i'm executing the following:
>>>>
>>>>
>>>> session:execute("bridge","sofia/gateway/${distributor(" .. route_name
>>>> .. ")}".. out_number .."")
>>>>
>>>> but the call doesn't falls to the next gw, i'm trying to add the
>>>> loop="3" to the bridge but it doesn't work.
>>>>
>>>> I've trued like this:
>>>>
>>>>
>>>> session:execute("set","loop=\"3\"")
>>>> session:execute("bridge","sofia/gateway/${distributor(" .. route_name
>>>> .. ")}".. out_number .."")
>>>>
>>>> and
>>>>
>>>>
>>>> session:execute("bridge","sofia/gateway/${distributor(" .. route_name
>>>> .. ")}".. out_number .." loop=\"3\"")
>>>>
>>>> and
>>>>
>>>> session:execute("bridge","{loop=\"3\"}sofia/gateway/${distributor(" ..
>>>> route_name .. ")}".. out_number .."")
>>>>
>>>>
>>>> Still nothing...
>>>>
>>>> any thoughts?
>>>>
>>>> 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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 12 23:24:32 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 12 Feb 2011 20:24:32 +0000
Subject: [Freeswitch-users] mod_distributor, lua and loop
In-Reply-To: <AANLkTinvZAZVT-3rdaG=mG6GKeGDRRrm_U0EdFU9B6o2@mail.gmail.com>
References: <AANLkTinDD-n_J7B3VjvJcB=0iHsr6nnggUNyr0Nk8O7a@mail.gmail.com>
	<AANLkTik3m+zr3BLugtVG1+J9a7jK57fJHcWuvxygse76@mail.gmail.com>
	<1D6E8634-BF1E-42C4-91DC-B776B87CD894@gmail.com>
	<AANLkTinvZAZVT-3rdaG=mG6GKeGDRRrm_U0EdFU9B6o2@mail.gmail.com>
Message-ID: <AANLkTimM=80_NoAaY18wix+z84c1ecfvFEN2QPWggxB0@mail.gmail.com>

See Anthony's reply... that'll produce a different call to distributor each
time you do the bridge - which'll possibly return a different gateway each
time, depending on your weightings and which other calls are calling
distributor.

-Steve


On 12 February 2011 17:35, David Villasmil
<david.villasmil.work at gmail.com>wrote:

> Thanks for answering Steve,
>
> That's exactly what i don't want to do, because i wanted to use the
> "weight" feature in mod_distributor... but i gues i will end up doing
> it :)
>
> Thanks
>
> David
>
> On Sat, Feb 12, 2011 at 6:26 PM, Steven Ayre <steveayre at gmail.com> wrote:
> > I believe loop is specific to XML dialplan tags.
> >
> > Try implementing a lua loop, checking the hangup cause after the bridge
> and looping if it looks bad.
> >
> > Steve on iPhone
> >
> >
> > On 12 Feb 2011, at 15:06, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
> >
> >> Oh I forgot,
> >>
> >> I also have:
> >>
> >>
> session:execute("set","continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407")
> >>
> >> thanks
> >>
> >> David
> >>
> >> On Sat, Feb 12, 2011 at 4:04 PM, David Villasmil
> >> <david.villasmil.work at gmail.com> wrote:
> >>> Hello All,
> >>>
> >>> from a lua script, i'm executing the following:
> >>>
> >>>
> >>> session:execute("bridge","sofia/gateway/${distributor(" .. route_name
> >>> .. ")}".. out_number .."")
> >>>
> >>> but the call doesn't falls to the next gw, i'm trying to add the
> >>> loop="3" to the bridge but it doesn't work.
> >>>
> >>> I've trued like this:
> >>>
> >>>
> >>> session:execute("set","loop=\"3\"")
> >>> session:execute("bridge","sofia/gateway/${distributor(" .. route_name
> >>> .. ")}".. out_number .."")
> >>>
> >>> and
> >>>
> >>>
> >>> session:execute("bridge","sofia/gateway/${distributor(" .. route_name
> >>> .. ")}".. out_number .." loop=\"3\"")
> >>>
> >>> and
> >>>
> >>> session:execute("bridge","{loop=\"3\"}sofia/gateway/${distributor(" ..
> >>> route_name .. ")}".. out_number .."")
> >>>
> >>>
> >>> Still nothing...
> >>>
> >>> any thoughts?
> >>>
> >>> 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
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 13 01:49:35 2011
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 12 Feb 2011 16:49:35 -0600
Subject: [Freeswitch-users] mod_dingaling & inbound audio
In-Reply-To: <1977194672-1297495611-cardhu_decombobulator_blackberry.rim.net-1261789055-@b25.c2.bise3.blackberry>
References: <AANLkTim-5LDhfQhHk_2LNLffvTa1JyBXq9U4B3LSY3Nq@mail.gmail.com>
	<AANLkTik19FmdGVEukmC4rS3ytu9qhR=wfSkO-fhaRMJQ@mail.gmail.com>
	<AANLkTikoJcrERbEHbdkLM_4ocnCUKkYDoKE-miYbvTxj@mail.gmail.com>
	<1977194672-1297495611-cardhu_decombobulator_blackberry.rim.net-1261789055-@b25.c2.bise3.blackberry>
Message-ID: <AANLkTi=UUVuXaoTyPe_xSJd06DYzXi9anHgjc6nEm9Ok@mail.gmail.com>

On Sat, Feb 12, 2011 at 1:27 AM, Budi wibowo <bwibowo at gmail.com> wrote:

> On previous git version I have working mod dingaling, but until today my
> dingaling is never working.
> If anybody has freeswitch source with dingaling works, please share the
> url. I will download
>
> What does the console log show? Also, turn on dingaling debugging w/ the
"dl_debug on" command.
-MC
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From msc at freeswitch.org  Sun Feb 13 01:50:15 2011
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 12 Feb 2011 16:50:15 -0600
Subject: [Freeswitch-users] Intermittent curl execution after rxfax
In-Reply-To: <4961923E-BF1C-4202-9441-A2FF2B7E5E57@gmail.com>
References: <AANLkTi=qqaP0bAhdVMW309fm2vfr41BP13c81epOieW0@mail.gmail.com>
	<4961923E-BF1C-4202-9441-A2FF2B7E5E57@gmail.com>
Message-ID: <AANLkTimeCzSwCMfA7FoCzQn8a=xGSRuPUUkitN3EA2qa@mail.gmail.com>

How about an api_hangup_hook?

On Sat, Feb 12, 2011 at 3:24 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Perhaps the aleg hangs up after rxfax but before curl is executed? If the
> aleg hangs up the remaining extension actions won't run.
>
> Looks like you're only using it to indicate a fax has been received? Try
> using ESL or CDRs. Mod_xml_cdr can post a CDR to a server and will have
> various variables set that indicate the fax has been received.
>
> Steve on iPhone
>
>
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From bwibowo at gmail.com  Sun Feb 13 03:30:42 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Sun, 13 Feb 2011 07:30:42 +0700
Subject: [Freeswitch-users] mod_dingaling & inbound audio
In-Reply-To: <AANLkTi=UUVuXaoTyPe_xSJd06DYzXi9anHgjc6nEm9Ok@mail.gmail.com>
References: <AANLkTim-5LDhfQhHk_2LNLffvTa1JyBXq9U4B3LSY3Nq@mail.gmail.com>
	<AANLkTik19FmdGVEukmC4rS3ytu9qhR=wfSkO-fhaRMJQ@mail.gmail.com>
	<AANLkTikoJcrERbEHbdkLM_4ocnCUKkYDoKE-miYbvTxj@mail.gmail.com>
	<1977194672-1297495611-cardhu_decombobulator_blackberry.rim.net-1261789055-@b25.c2.bise3.blackberry>
	<AANLkTi=UUVuXaoTyPe_xSJd06DYzXi9anHgjc6nEm9Ok@mail.gmail.com>
Message-ID: <AANLkTinvzvssxGewxP13tPqy3ywdsgvFTTrB3Ofm3Y7h@mail.gmail.com>

hi michael
this is my output screen, i consider on BREAK and CHANNEL KILL here, i use
xlite and
acrobits with different ip provider to test. and both silent with dingaling

thx

budi

2011-02-13 08:26:06.523030 [INFO] libdingaling.c:1366 SecRECV:
-------------------------------------------------------------------------------
<iq to="bwibowo at gmail.com/gtalkE0340285" from="+
14085264000 at voice.google.com/srvres-MTAuMTMuMTE4LjI2Ojk4MDI=" id="306"
type="result"></iq>

2011-02-13 08:26:06.523030 [DEBUG] libdingaling.c:968 Cancel packet 306

2011-02-13 08:26:06.617029 [DEBUG] libdingaling.c:1450 Processing 1 packets
in retry queue

2011-02-13 08:26:06.617029 [DEBUG] libdingaling.c:1475 Discarding packet 306

2011-02-13 08:26:06.721027 [INFO] libdingaling.c:1366 SecRECV:
-------------------------------------------------------------------------------
<iq from="+14085264000 at voice.google.com/srvres-MTAuMTMuMTE4LjI2Ojk4MDI="
to="bwibowo at gmail.com/gtalkE0340285"
id="jingle:10.13.118.26-26031162:1:03888679" type="set">
  <ses:session type="transport-info" id="9537230161" initiator="
bwibowo at gmail.com/gtalkE0340285" xmlns:ses="http://www.google.com/session">
    <transport xmlns="http://www.google.com/transport/p2p">
      <candidate name="rtp" address="74.125.153.126" port="19295"
username="f1ipq5kmX1kt1UlH" preference="1.0" protocol="udp"
network="mediaproxy" generation="0"></candidate>
      <candidate name="rtp" address="74.125.153.126" port="19294"
username="f1ipq5kmX1kt1UlH" preference="0.6" protocol="tcp"
network="mediaproxy" generation="0"></candidate>
      <candidate name="rtp" address="74.125.153.126" port="443"
username="f1ipq5kmX1kt1UlH" preference="0.5" protocol="ssltcp"
network="mediaproxy" generation="0"></candidate>
    </transport>
  </ses:session>
</iq>

2011-02-13 08:26:06.721027 [DEBUG] libdingaling.c:383 Message for Session
9537230161

2011-02-13 08:26:06.721027 [DEBUG] libdingaling.c:528 New Candidate 1
name=rtp
type=stun
protocol=udp
username=f1ipq5kmX1kt1UlH
password=(null)
address=74.125.153.126
port=19295
pref=1.00

2011-02-13 08:26:06.721027 [DEBUG] libdingaling.c:528 New Candidate 2
name=rtp
type=stun
protocol=tcp
username=f1ipq5kmX1kt1UlH
password=(null)
address=74.125.153.126
port=19294
pref=0.60

2011-02-13 08:26:06.721027 [DEBUG] libdingaling.c:528 New Candidate 3
name=rtp
type=stun
protocol=ssltcp
username=f1ipq5kmX1kt1UlH
password=(null)
address=74.125.153.126
port=443
pref=0.50

2011-02-13 08:26:06.721027 [DEBUG] mod_dingaling.c:2941 using Existing
session for 9537230161
2011-02-13 08:26:06.721027 [DEBUG] mod_dingaling.c:3279 3 candidates
2011-02-13 08:26:06.721027 [DEBUG] mod_dingaling.c:3299 candidate
74.125.153.126:19295 PASS ACL wan.auto
2011-02-13 08:26:06.721027 [DEBUG] mod_dingaling.c:3351 Acceptable Candidate
74.125.153.126:19295
2011-02-13 08:26:06.817031 [NOTICE] libdingaling.c:1368 SecSEND:
-------------------------------------------------------------------------------
<iq type="result" to="+
14085264000 at voice.google.com/srvres-MTAuMTMuMTE4LjI2Ojk4MDI=" from="
bwibowo at gmail.com/gtalkE0340285"
id="jingle:10.13.118.26-26031162:1:03888679">
  <query></query>
</iq>

2011-02-13 08:26:08.841032 [INFO] libdingaling.c:1366 SecRECV:
-------------------------------------------------------------------------------
<iq from="+14085264000 at voice.google.com/srvres-MTAuMTMuMTE4LjI2Ojk4MDI="
to="bwibowo at gmail.com/gtalkE0340285"
id="jingle:10.13.118.26-26031162:1:0388867F" type="set">
  <ses:session type="accept" id="9537230161" initiator="
bwibowo at gmail.com/gtalkE0340285" xmlns:ses="http://www.google.com/session">
    <pho:description xmlns:pho="http://www.google.com/session/phone">
      <pho:payload-type id="0" name="PCMU"
clockrate="8000"></pho:payload-type>
      <pho:payload-type id="101" name="telephone-event"></pho:payload-type>
    </pho:description>
  </ses:session>
</iq>

2011-02-13 08:26:08.841957 [DEBUG] libdingaling.c:383 Message for Session
9537230161

2011-02-13 08:26:08.841957 [DEBUG] libdingaling.c:440 Add Payload [PCMU]
id='0'

2011-02-13 08:26:08.841957 [DEBUG] libdingaling.c:440 Add Payload
[telephone-event] id='101'

2011-02-13 08:26:08.841957 [DEBUG] mod_dingaling.c:2941 using Existing
session for 9537230161
2011-02-13 08:26:08.841957 [DEBUG] mod_dingaling.c:3193 Already decided on a
codec
2011-02-13 08:26:08.843032 [DEBUG] mod_dingaling.c:865 Set Read Codec to
PCMU at 8000
2011-02-13 08:26:08.843032 [DEBUG] mod_dingaling.c:880 Set Write Codec to
PCMU at 8000
2011-02-13 08:26:08.843032 [DEBUG] mod_dingaling.c:892 SETUP RTP
202.122.99.99:24240 -> 74.125.153.126:19295
2011-02-13 08:26:08.843032 [DEBUG] switch_rtp.c:1607 Starting timer [soft]
160 bytes per 20ms
2011-02-13 08:26:08.845035 [DEBUG] switch_channel.c:2788 (dingaling/gtalk/+
14085264000 at voice.google.com) Callstate Change DOWN -> ACTIVE
2011-02-13 08:26:08.845035 [DEBUG] switch_channel.c:2800 Send signal
sofia/internal/1000 at 202.122.99.99 [BREAK]
2011-02-13 08:26:08.845035 [NOTICE] mod_dingaling.c:1202 Channel
[dingaling/gtalk/+14085264000 at voice.google.com] has been answered
2011-02-13 08:26:08.845035 [DEBUG] mod_dingaling.c:1205 (dingaling/gtalk/+
14085264000 at voice.google.com) State Change CS_INIT -> CS_ROUTING
2011-02-13 08:26:08.845035 [DEBUG] switch_core_session.c:1116 Send signal
dingaling/gtalk/+14085264000 at voice.google.com [BREAK]
2011-02-13 08:26:08.845035 [DEBUG] mod_dingaling.c:1348 dingaling/gtalk/+
14085264000 at voice.google.com CHANNEL KILL
2011-02-13 08:26:08.845035 [DEBUG] switch_core_state_machine.c:356
(dingaling/gtalk/+14085264000 at voice.google.com) State INIT going to sleep
2011-02-13 08:26:08.845035 [DEBUG] switch_core_state_machine.c:320
(dingaling/gtalk/+14085264000 at voice.google.com) Running State Change
CS_ROUTING
2011-02-13 08:26:08.845035 [DEBUG] switch_channel.c:1660 (dingaling/gtalk/+
14085264000 at voice.google.com) Callstate Change ACTIVE -> RINGING
2011-02-13 08:26:08.845035 [DEBUG] switch_core_state_machine.c:359
(dingaling/gtalk/+14085264000 at voice.google.com) State ROUTING
2011-02-13 08:26:08.845035 [DEBUG] mod_dingaling.c:1219 dingaling/gtalk/+
14085264000 at voice.google.com CHANNEL ROUTING
2011-02-13 08:26:08.845035 [DEBUG] switch_ivr_originate.c:66
(dingaling/gtalk/+14085264000 at voice.google.com) State Change CS_ROUTING ->
CS_CONSUME_MEDIA
2011-02-13 08:26:08.845035 [DEBUG] switch_core_session.c:1116 Send signal
dingaling/gtalk/+14085264000 at voice.google.com [BREAK]
2011-02-13 08:26:08.845035 [DEBUG] mod_dingaling.c:1348 dingaling/gtalk/+
14085264000 at voice.google.com CHANNEL KILL
2011-02-13 08:26:08.845035 [DEBUG] switch_core_state_machine.c:359
(dingaling/gtalk/+14085264000 at voice.google.com) State ROUTING going to sleep
2011-02-13 08:26:08.845035 [DEBUG] switch_core_state_machine.c:320
(dingaling/gtalk/+14085264000 at voice.google.com) Running State Change
CS_CONSUME_MEDIA
2011-02-13 08:26:08.845035 [DEBUG] switch_channel.c:1662 (dingaling/gtalk/+
14085264000 at voice.google.com) Callstate Change RINGING -> ACTIVE
2011-02-13 08:26:08.845035 [DEBUG] switch_core_state_machine.c:378
(dingaling/gtalk/+14085264000 at voice.google.com) State CONSUME_MEDIA
2011-02-13 08:26:08.845035 [DEBUG] switch_core_state_machine.c:378
(dingaling/gtalk/+14085264000 at voice.google.com) State CONSUME_MEDIA going to
sleep
2011-02-13 08:26:08.845035 [NOTICE] mod_sofia.c:2185 Ring-Ready
sofia/internal/1000 at 202.122.99.99!
2011-02-13 08:26:08.845035 [DEBUG] switch_core_session.c:709 Send signal
sofia/internal/1000 at 202.122.99.99 [BREAK]
2011-02-13 08:26:08.845035 [NOTICE] switch_ivr_originate.c:479 Ring Ready
sofia/internal/1000 at 202.122.99.99!
2011-02-13 08:26:08.845035 [DEBUG] sofia.c:4659 Channel sofia/internal/
1000 at 202.122.99.99 entering state [early][180]
2011-02-13 08:26:08.866018 [DEBUG] sofia_glue.c:2987 AUDIO RTP
[sofia/internal/1000 at 202.122.99.99] 202.122.99.99 port 20318 ->
202.43.188.15 port 4516 codec: 3 ms: 20
2011-02-13 08:26:08.866971 [DEBUG] switch_rtp.c:1607 Starting timer [soft]
160 bytes per 20ms
2011-02-13 08:26:08.868036 [DEBUG] sofia_glue.c:3228 Set 2833 dtmf send
payload to 100
2011-02-13 08:26:08.868036 [DEBUG] sofia_glue.c:3233 Set 2833 dtmf receive
payload to 100
2011-02-13 08:26:08.868036 [DEBUG] mod_sofia.c:681 Local SDP sofia/internal/
1000 at 202.122.99.99:
v=0
o=FreeSWITCH 1297536450 1297536451 IN IP4 202.122.99.99
s=FreeSWITCH
c=IN IP4 202.122.99.99
t=0 0
m=audio 20318 RTP/AVP 3 100
a=rtpmap:3 GSM/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2011-02-13 08:26:08.868036 [DEBUG] switch_core_session.c:709 Send signal
sofia/internal/1000 at 202.122.99.99 [BREAK]
2011-02-13 08:26:08.868036 [DEBUG] switch_channel.c:2788 (sofia/internal/
1000 at 202.122.99.99) Callstate Change RINGING -> ACTIVE
2011-02-13 08:26:08.868036 [NOTICE] switch_ivr_originate.c:3363 Channel
[sofia/internal/1000 at 202.122.99.99] has been answered
2011-02-13 08:26:08.868036 [DEBUG] switch_ivr_originate.c:3408 Originate
Resulted in Success: [dingaling/gtalk/+14085264000 at voice.google.com]
2011-02-13 08:26:08.868036 [DEBUG] switch_core_session.c:709 Send signal
dingaling/gtalk/+14085264000 at voice.google.com [BREAK]
2011-02-13 08:26:08.868036 [DEBUG] mod_dingaling.c:1348 dingaling/gtalk/+
14085264000 at voice.google.com CHANNEL KILL
2011-02-13 08:26:08.868036 [DEBUG] switch_core_session.c:709 Send signal
sofia/internal/1000 at 202.122.99.99 [BREAK]
2011-02-13 08:26:08.868036 [DEBUG] switch_ivr_bridge.c:1234
(dingaling/gtalk/+14085264000 at voice.google.com) State Change
CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2011-02-13 08:26:08.868036 [DEBUG] switch_core_session.c:1116 Send signal
dingaling/gtalk/+14085264000 at voice.google.com [BREAK]
2011-02-13 08:26:08.868036 [DEBUG] mod_dingaling.c:1348 dingaling/gtalk/+
14085264000 at voice.google.com CHANNEL KILL
2011-02-13 08:26:08.868036 [DEBUG] sofia.c:4659 Channel sofia/internal/
1000 at 202.122.99.99 entering state [completed][200]
2011-02-13 08:26:08.868036 [DEBUG] switch_core_state_machine.c:320
(dingaling/gtalk/+14085264000 at voice.google.com) Running State Change
CS_EXCHANGE_MEDIA
2011-02-13 08:26:08.868036 [DEBUG] switch_core_state_machine.c:369
(dingaling/gtalk/+14085264000 at voice.google.com) State EXCHANGE_MEDIA
2011-02-13 08:26:08.868036 [DEBUG] mod_dingaling.c:1356 CHANNEL LOOPBACK
2011-02-13 08:26:08.869918 [DEBUG] mod_nibblebill.c:572 Received request via
SESSION_HEARTBEAT!
2011-02-13 08:26:08.869918 [DEBUG] mod_nibblebill.c:433 Attempting to bill
at $0.05 per minute to account 1000
2011-02-13 08:26:08.869918 [INFO] mod_nibblebill.c:485 Beginning new billing
on d5cbf051-f0d5-4d02-b136-574e80d7c676
2011-02-13 08:26:08.869918 [DEBUG] mod_nibblebill.c:491 0 seconds passed
since last bill time of 2011-02-13 08:26:08
2011-02-13 08:26:08.869918 [DEBUG] mod_nibblebill.c:498 Billing $0.000002 to
1000 (Call: d5cbf051-f0d5-4d02-b136-574e80d7c676 / 0.000000 so far)
2011-02-13 08:26:08.869918 [DEBUG] mod_nibblebill.c:321 Doing update query
[UPDATE accounts SET cash=cash-0.000002 WHERE id='1000']
2011-02-13 08:26:08.869918 [DEBUG] mod_nibblebill.c:366 Doing lookup query
[SELECT cash AS nibble_balance FROM accounts WHERE id='1000']
2011-02-13 08:26:08.869918 [DEBUG] mod_nibblebill.c:376 Retrieved current
balance for account 1000 (balance = 96.581215)
2011-02-13 08:26:08.917032 [NOTICE] libdingaling.c:1368 SecSEND:
-------------------------------------------------------------------------------
<iq type="result" to="+
14085264000 at voice.google.com/srvres-MTAuMTMuMTE4LjI2Ojk4MDI=" from="
bwibowo at gmail.com/gtalkE0340285"
id="jingle:10.13.118.26-26031162:1:0388867F">
  <query></query>
</iq>

2011-02-13 08:26:09.221034 [DEBUG] sofia.c:4659 Channel sofia/internal/
1000 at 202.122.99.99 entering state [ready][200]
2011-02-13 08:26:09.228033 [DEBUG] switch_core_session.c:771 Send signal
dingaling/gtalk/+14085264000 at voice.google.com [BREAK]
2011-02-13 08:26:09.228033 [DEBUG] mod_dingaling.c:1348 dingaling/gtalk/+
14085264000 at voice.google.com CHANNEL KILL
2011-02-13 08:26:09.228033 [DEBUG] switch_core_session.c:771 Send signal
sofia/internal/1000 at 202.122.99.99 [BREAK]
2011-02-13 08:26:10.568036 [INFO] switch_rtp.c:2920 Auto Changing port from
202.43.188.15:4516 to 202.43.188.76:20256
2011-02-13 08:26:58.767084 [INFO] libdingaling.c:1366 SecRECV:
-------------------------------------------------------------------------------
<presence from="bwibowo at gmail.com/gmail.16565A66" to="bwibowo at gmail.com">
  <priority>24</priority>
  <caps:c node="http://mail.google.com/xmpp/client/caps" ver="1.1"
ext="pmuc-v1 sms-v1 camera-v1 video-v1 voice-v1" xmlns:caps="
http://jabber.org/protocol/caps"></caps:c>
  <status>Ngantuk</status>
  <x xmlns="vcard-temp:x:update">
    <photo></photo>
  </x>
</presence>

2011-02-13 08:26:58.817032 [NOTICE] libdingaling.c:1368 SecSEND:
-------------------------------------------------------------------------------
<presence type="subscribed" to="bwibowo at gmail.com">
  <status>Ding A Ling....</status>
</presence>

2011-02-13 08:27:08.868031 [DEBUG] mod_nibblebill.c:572 Received request via
SESSION_HEARTBEAT!
2011-02-13 08:27:08.868031 [DEBUG] mod_nibblebill.c:433 Attempting to bill
at $0.05 per minute to account 1000
2011-02-13 08:27:08.868031 [DEBUG] mod_nibblebill.c:491 59 seconds passed
since last bill time of 2011-02-13 08:26:08
2011-02-13 08:27:08.868031 [DEBUG] mod_nibblebill.c:498 Billing $0.049998 to
1000 (Call: d5cbf051-f0d5-4d02-b136-574e80d7c676 / 0.000002 so far)
2011-02-13 08:27:08.868031 [DEBUG] mod_nibblebill.c:321 Doing update query
[UPDATE accounts SET cash=cash-0.049998 WHERE id='1000']
2011-02-13 08:27:08.868979 [DEBUG] mod_nibblebill.c:366 Doing lookup query
[SELECT cash AS nibble_balance FROM accounts WHERE id='1000']
2011-02-13 08:27:08.868979 [DEBUG] mod_nibblebill.c:376 Retrieved current
balance for account 1000 (balance = 96.531219)


On Sun, Feb 13, 2011 at 5:49 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Sat, Feb 12, 2011 at 1:27 AM, Budi wibowo <bwibowo at gmail.com> wrote:
>
>> On previous git version I have working mod dingaling, but until today my
>> dingaling is never working.
>> If anybody has freeswitch source with dingaling works, please share the
>> url. I will download
>>
>> What does the console log show? Also, turn on dingaling debugging w/ the
> "dl_debug on" command.
> -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 chris.chen2004 at gmail.com  Sun Feb 13 04:16:30 2011
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Sat, 12 Feb 2011 20:16:30 -0500
Subject: [Freeswitch-users] mod_dingaling & inbound audio
In-Reply-To: <AANLkTi=UUVuXaoTyPe_xSJd06DYzXi9anHgjc6nEm9Ok@mail.gmail.com>
References: <AANLkTim-5LDhfQhHk_2LNLffvTa1JyBXq9U4B3LSY3Nq@mail.gmail.com>
	<AANLkTik19FmdGVEukmC4rS3ytu9qhR=wfSkO-fhaRMJQ@mail.gmail.com>
	<AANLkTikoJcrERbEHbdkLM_4ocnCUKkYDoKE-miYbvTxj@mail.gmail.com>
	<1977194672-1297495611-cardhu_decombobulator_blackberry.rim.net-1261789055-@b25.c2.bise3.blackberry>
	<AANLkTi=UUVuXaoTyPe_xSJd06DYzXi9anHgjc6nEm9Ok@mail.gmail.com>
Message-ID: <AANLkTimcsPbwT7thhopGCoiX56Z-VFjuhbnksz7ms4Mc@mail.gmail.com>

Hi Michael, this is a known issue with mod_dingaling since mid January
around the time of 1.0.7 tarbar release this year. I tested with different
kinds of setups, the mod_dingaling always loops at mod_dingaling.c line
2941and 3275, which results in no audio (rtp packets) even though the jingle
session seems to have already established.

I asked for core dev's review of mod_dingaling on IRC a couple of times,
looks like everybody is so busy that no one really takes a closer looks at
the mod_dingaling.c yet.

This behavior affects Gtalk to sip calling  as well as gmail outbound call
which is sip to jingle.

Hope FS core dev will have some time to review this, we really appreciate
that FS's gtalk calling can be back running again.

Thanks,

Chris


On Sat, Feb 12, 2011 at 5:49 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Sat, Feb 12, 2011 at 1:27 AM, Budi wibowo <bwibowo at gmail.com> wrote:
>
>> On previous git version I have working mod dingaling, but until today my
>> dingaling is never working.
>> If anybody has freeswitch source with dingaling works, please share the
>> url. I will download
>>
>> What does the console log show? Also, turn on dingaling debugging w/ the
> "dl_debug on" command.
> -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 rmbertjones at comcast.net  Sun Feb 13 06:27:42 2011
From: rmbertjones at comcast.net (Bert Jones)
Date: Sat, 12 Feb 2011 22:27:42 -0500
Subject: [Freeswitch-users] Possible to communicate with FS using
	event_socket without ESL?
Message-ID: <003101cbcb2d$fa0e9e60$ee2bdb20$@net>

Hello,

 

Am I correct in assuming that an app can talk directly to FS using
event_socket without the use of the ESL?  I am attempting to do this, but
having trouble authenticating. I wanted to make sure I was not overlooking
something basis in my approach.

 

I have successfully loaded FS on a windows XP server and can register phones
and make calls.  Further I can talk to the server, authenticate and make
calls using telnet, but when attempting to do the same via an app written in
.net and using a socket, I am unable to authenticate.  

 

Running my app using a tcpSocket, I receive the "Content-Type: auth/request"
upon connection of the socket, but when sending "auth password\n\n" it
appears that the connection times out after about 10 seconds and I receive:
"Content-Type: text/disconnect-notice" & vbLf & "Content-Length: 67".
mod_event_socket is enabled in modules.conf.xml .

 

Is there something obvious that I am missing conceptually, or should this
work?

 

Thanks!

 

 

 

 

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From ashley at midletearth.com  Sun Feb 13 09:25:18 2011
From: ashley at midletearth.com (Ashley B)
Date: Sun, 13 Feb 2011 08:25:18 +0200
Subject: [Freeswitch-users] Passing custom variables for managed code in
	xml_curl
Message-ID: <36B53B15-A1ED-4A0A-809D-EF0A1D030C9E@midletearth.com>

Hi

How do I set my own global vars (X-PRE-PROCESS) using mod_xml_curl that I can use in managed code / script?). 

Eg mod_managed does not always have the values of the variables  required @ runtime, they are always changing. I have to curl them from another server, i know i can set them up as channel variables when calling for dial plan with xml_curl but need to be able to fetch them @ the same time as configuration when calling reloadxml.

Any assistance is appreciated.

Thanks
Ashley




From chenzhanping at gmail.com  Sun Feb 13 11:17:34 2011
From: chenzhanping at gmail.com (=?GB2312?B?1bzGvbPC?=)
Date: Sun, 13 Feb 2011 16:17:34 +0800
Subject: [Freeswitch-users] Question about mod_dingaling.
In-Reply-To: <AANLkTikPa22Vh33x+xODysCQo4Q45MFd9X+=xgN8iPis@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>
Message-ID: <AANLkTi=Bf89Pae2Qt1Cg1gFhX-xjFHuGz0kv5Dyt9=FK@mail.gmail.com>

Thank u very much,

My server is have public ip,and router not have upnp.

I modified client.xml   :

<!-- <param name="ext-rtp-ip" value="auto-nat"/> -->   to   <param
name="ext-rtp-ip" value="My ip"/>

Now my Freeswitch + mod_dingaling is working.

But there is *no voice* when i call out from google voice and connect called
Successed.

I try to called my other google phone use freeswitch + mod_dingaling ,
and into voicemail, the voicemail system can heared my voice,and record
good.

This mean: caller is no voice,but called is have voice good.

there is client.xml:

http://pastebin.freeswitch.org/15371

there is cli log:

http://pastebin.freeswitch.org/15372

freeswitch version:FreeSWITCH Version 1.0.head (git-a2c0da5 2011-02-11
23-10-12 -0600)
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From mayamatakeshi at gmail.com  Sun Feb 13 12:28:06 2011
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Sun, 13 Feb 2011 18:28:06 +0900
Subject: [Freeswitch-users] Possible to communicate with FS using
 event_socket without ESL?
In-Reply-To: <003101cbcb2d$fa0e9e60$ee2bdb20$@net>
References: <003101cbcb2d$fa0e9e60$ee2bdb20$@net>
Message-ID: <AANLkTinhUwUfcdnMTkdtgpV7W+wBYzCC9oULrexsjFEZ@mail.gmail.com>

On Sun, Feb 13, 2011 at 12:27 PM, Bert Jones <rmbertjones at comcast.net>wrote:

> Hello,
>
>
>
> Am I correct in assuming that an app can talk directly to FS using
> event_socket without the use of the ESL?  I am attempting to do this, but
> having trouble authenticating. I wanted to make sure I was not overlooking
> something basis in my approach.
>
>
>
> I have successfully loaded FS on a windows XP server and can register
> phones and make calls.  Further I can talk to the server, authenticate and
> make calls using telnet, but when attempting to do the same via an app
> written in .net and using a socket, I am unable to authenticate.
>
>
>
> Running my app using a tcpSocket, I receive the "Content-Type:
> auth/request" upon connection of the socket, but when sending ?auth
> password\n\n? it appears that the connection times out after about 10
> seconds and I receive:  ?Content-Type: text/disconnect-notice" & vbLf &
> "Content-Length: 67".  mod_event_socket is enabled in modules.conf.xml .
>
>
>
> Is there something obvious that I am missing conceptually, or should this
> work?
>

Maybe the .NET component is set to buffer data before sending it. There
might be some method to flush the data immediately.
If this doesn't solve it. I would get a packet capture.

r,
takeshi
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From bwibowo at gmail.com  Sun Feb 13 13:52:20 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Sun, 13 Feb 2011 17:52:20 +0700
Subject: [Freeswitch-users] mod managed install
Message-ID: <AANLkTinKJgRcROp2RcTw4vtyu59QBmLGOz4o0bqnGnxr@mail.gmail.com>

hi
i have mono installed using yum following
http://stackoverflow.com/questions/3510320/install-mono-on-centos5-5-using-yum
when i install mod_managed i got this
making all mod_managed
Package mono was not found in the pkg-config search path.
Perhaps you should add the directory containing `mono.pc'
to the PKG_CONFIG_PATH environment variable
No package 'mono' found
Package mono was not found in the pkg-config search path.
Perhaps you should add the directory containing `mono.pc'
to the PKG_CONFIG_PATH environment variable
No package 'mono' found
Compiling freeswitch_managed.cpp...
g++ -I/usr/local/src/freeswitch-1.0.7/src/include
-I/usr/local/src/freeswitch-1.0.7/src/include
-I/usr/local/src/freeswitch-1.0.7/libs/libteletone/src -fPIC
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2
-D_GNU_SOURCE -DHAVE_CONFIG_H -c -o freeswitch_managed.o
freeswitch_managed.cpp
In file included from freeswitch_managed.cpp:35:
freeswitch_managed.h:43:18: error: glib.h: No such file or directory
freeswitch_managed.h:44:26: error: mono/jit/jit.h: No such file or directory
freeswitch_managed.h:45:36: error: mono/metadata/assembly.h: No such file or
directory
freeswitch_managed.h:46:39: error: mono/metadata/environment.h: No such file
or directory
freeswitch_managed.h:47:39: error: mono/metadata/mono-config.h: No such file
or directory

any parameter i need to change?


thx

budi wibowo

<http://stackoverflow.com/questions/3510320/install-mono-on-centos5-5-using-yum>
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From br at bsdpad.com  Sun Feb 13 14:43:36 2011
From: br at bsdpad.com (Ruslan Bukin)
Date: Sun, 13 Feb 2011 14:43:36 +0300
Subject: [Freeswitch-users] dp_tools/set- not worked SOMETIMES
In-Reply-To: <AANLkTi=uc-bprji8C82T=xukDZ3PEP3yHj=h-ro488wU@mail.gmail.com>
References: <20110211112453.GA700@bsdjail.com>
	<20110211114004.GA1491@bsdjail.com>
	<AANLkTim=+R9snJzYiCdUNZiixxOcOy-6C2j=SjpzsBSN@mail.gmail.com>
	<20110211130209.GA4821@bsdjail.com>
	<AANLkTi=uc-bprji8C82T=xukDZ3PEP3yHj=h-ro488wU@mail.gmail.com>
Message-ID: <20110213114336.GA12277@bsdjail.com>

It is simplest usage of fs => so the issues is platform specific.
yes, it is latest freebsd 8.1 (thread lib was changed in fbsd7)
3d issue: fs crashes time by time

-Ruslan

On Fri, Feb 11, 2011 at 07:11:36AM -0600, Anthony Minessale wrote:
> Is it reproducable enough to provide a recipe so someone can test it on
> linux.  Its possibly only a fbsd issue since I have not heard this ever
> before.  Is it the latest freebsd since they stopped using their own thread
> lib?
> On Feb 11, 2011 7:03 AM, "Ruslan Bukin" <br at bsdpad.com> wrote:
> >
> > I'm using not changed git head and freebsd amd64 only
> > (checked on two different fbsd servers -problem exists in both)
> >
> > -Ruslan
> >
> > On Fri, Feb 11, 2011 at 06:00:16AM -0600, Anthony Minessale wrote:
> > > This is your second strange issue. Maybe you have some problem with
> > > your platform.
> > > Are you using unaltered GIT HEAD?  Have you compared results on other
> > > platforms like Linux 64 bit or Windows?
> > >
> > >
> > > On Fri, Feb 11, 2011 at 5:40 AM, Ruslan Bukin <br at bsdpad.com> wrote:
> > > > correction: it seems that parameters always exists after bridge,
> > > > but sometimes one of them not exists in CDR result
> > > > (I have checked with mod_xml_cdr and mod_json_cdr)
> > > >
> > > > -Ruslan
> > > >
> > > > On Fri, Feb 11, 2011 at 02:24:53PM +0300, Ruslan Bukin wrote:
> > > >> Application set is not set parameters SOMETIMES
> > > >>
> > > >> for example, before bridge I set my static parameters:
> > > >>
> > > >> <action application="set" data="my_own_parameter=test"/>
> > > >> <action application="set" data="nibble_rate=1"/>
> > > >> ...
> > > >>
> > > >> sometimes (in about ~1-5% times) one of them is not set up.
> > > >>
> > > >> I tried to set like this before bridge:
> > > >>
> > > >>  <extension name="set_var" continue="true">
> > > >>    <condition field="${my_own_parameter}" expression=".+"
> break="on-true"/>
> > > >>    <condition field="destination_number" expression="(777)"
> break="on-true">
> > > >>      <action application="set" data="my_own_parameter=test"/>
> > > >>      <action application="transfer" data="$1 XML public"/>
> > > >>    </condition>
> > > >>  </extension>
> > > >>
> > > >> but no result: after bridge no parameter to read (SOMETIMES)
> > > >>
> > > >>
> > > >> any recommendations?
> > > >>
> > > >> -Ruslan
> > > >>
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at 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



From Nabble at slickdeals.endjunk.com  Sun Feb 13 15:16:12 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 13 Feb 2011 04:16:12 -0800 (PST)
Subject: [Freeswitch-users] Possible to communicate with FS using
	event_socket without ESL?
In-Reply-To: <003101cbcb2d$fa0e9e60$ee2bdb20$@net>
References: <003101cbcb2d$fa0e9e60$ee2bdb20$@net>
Message-ID: <1297599372197-6020712.post@n2.nabble.com>



Bert Jones wrote:
> Further I can talk to the server, authenticate and make
> calls using telnet, 
I am just curious how you do this. Do care to explain how to do this using
telnet?

-----
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/Possible-to-communicate-with-FS-using-event-socket-without-ESL-tp6020307p6020712.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Sun Feb 13 15:27:48 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 13 Feb 2011 12:27:48 +0000
Subject: [Freeswitch-users] Possible to communicate with FS using
 event_socket without ESL?
In-Reply-To: <1297599372197-6020712.post@n2.nabble.com>
References: <003101cbcb2d$fa0e9e60$ee2bdb20$@net>
	<1297599372197-6020712.post@n2.nabble.com>
Message-ID: <AANLkTin8R0e8i29L6rVogEuBB05rcQObPWN4G_HvH_uj@mail.gmail.com>

The ESL is a plain text protocol. You can connect to the ESL port via telnet
and type in the ESL protocol commands manually and it'll connect fine.

-Steve


On 13 February 2011 12:16, mazilo <Nabble at slickdeals.endjunk.com> wrote:

>
>
> Bert Jones wrote:
> > Further I can talk to the server, authenticate and make
> > calls using telnet,
> I am just curious how you do this. Do care to explain how to do this using
> telnet?
>
> -----
> 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/Possible-to-communicate-with-FS-using-event-socket-without-ESL-tp6020307p6020712.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 Feb 13 15:29:01 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 13 Feb 2011 12:29:01 +0000
Subject: [Freeswitch-users] Possible to communicate with FS using
 event_socket without ESL?
In-Reply-To: <003101cbcb2d$fa0e9e60$ee2bdb20$@net>
References: <003101cbcb2d$fa0e9e60$ee2bdb20$@net>
Message-ID: <AANLkTi=5zyKN1Ef+nJspyMrOuw=VE+fV-RcCQ5S148L=@mail.gmail.com>

Without the use of the ESL *protocol* - no. Without the use of the supplied
ESL *libraries* - correct.

You're along the right lines on implementing your own protocol. If you can
connect and authenticate via telnet from the same computer you're trying
your app from then there should be no problem with FS.

The fact you say it takes 10s for FS to respond indicates you're not sending
the auth line. You should get either a success or failure message straight
away in reply to that. So it sounds like the auth isn't getting sent at all.

Try using Wireshark to view what's going over the network to check it's what
you expect.

-Steve






On 13 February 2011 03:27, Bert Jones <rmbertjones at comcast.net> wrote:

> Hello,
>
>
>
> Am I correct in assuming that an app can talk directly to FS using
> event_socket without the use of the ESL?  I am attempting to do this, but
> having trouble authenticating. I wanted to make sure I was not overlooking
> something basis in my approach.
>
>
>
> I have successfully loaded FS on a windows XP server and can register
> phones and make calls.  Further I can talk to the server, authenticate and
> make calls using telnet, but when attempting to do the same via an app
> written in .net and using a socket, I am unable to authenticate.
>
>
>
> Running my app using a tcpSocket, I receive the "Content-Type:
> auth/request" upon connection of the socket, but when sending ?auth
> password\n\n? it appears that the connection times out after about 10
> seconds and I receive:  ?Content-Type: text/disconnect-notice" & vbLf &
> "Content-Length: 67".  mod_event_socket is enabled in modules.conf.xml .
>
>
>
> Is there something obvious that I am missing conceptually, or should this
> work?
>
>
>
> 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 w8hdkim at gmail.com  Sun Feb 13 15:30:52 2011
From: w8hdkim at gmail.com (Kim Culhan)
Date: Sun, 13 Feb 2011 07:30:52 -0500
Subject: [Freeswitch-users] dp_tools/set- not worked SOMETIMES
Message-ID: <AANLkTimwKcuTJHJQp-Q4ER7h8Pd+Qj3vq5rkcRA+L4Uv@mail.gmail.com>

On Sun, February 13, 2011 6:43 am, Ruslan Bukin wrote:
> It is simplest usage of fs => so the issues is platform specific.
> yes, it is latest freebsd 8.1 (thread lib was changed in fbsd7)
> 3d issue: fs crashes time by time
I am running git head on freebsd 8.2-RC2  amd64 and do not have any cases of
fs crashing.

I wanted to mention this because it reminded me of Anthony's comment:

>> > On Fri, Feb 11, 2011 at 06:00:16AM -0600, Anthony Minessale wrote:
>> > > This is your second strange issue. Maybe you have some problem with
>> > > your platform.
I do not have enough  information to duplicate your situation with the loss
of parameters.

If you would like I could try to help if you could supply a test recipe as
Anthony suggested to duplicate the problem.

thanks
-kim

--
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From Nabble at slickdeals.endjunk.com  Sun Feb 13 16:49:20 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 13 Feb 2011 05:49:20 -0800 (PST)
Subject: [Freeswitch-users] mod_dingaling & inbound audio
In-Reply-To: <AANLkTimcsPbwT7thhopGCoiX56Z-VFjuhbnksz7ms4Mc@mail.gmail.com>
References: <AANLkTim-5LDhfQhHk_2LNLffvTa1JyBXq9U4B3LSY3Nq@mail.gmail.com>
	<AANLkTik19FmdGVEukmC4rS3ytu9qhR=wfSkO-fhaRMJQ@mail.gmail.com>
	<AANLkTikoJcrERbEHbdkLM_4ocnCUKkYDoKE-miYbvTxj@mail.gmail.com>
	<1977194672-1297495611-cardhu_decombobulator_blackberry.rim.net-1261789055-@b25.c2.bise3.blackberry>
	<AANLkTi=UUVuXaoTyPe_xSJd06DYzXi9anHgjc6nEm9Ok@mail.gmail.com>
	<AANLkTimcsPbwT7thhopGCoiX56Z-VFjuhbnksz7ms4Mc@mail.gmail.com>
Message-ID: <1297604960568-6020844.post@n2.nabble.com>



Chris Chen-4 wrote:
> Hi Michael, this is a known issue with mod_dingaling since mid January
> around the time of 1.0.7 tarbar release this year. I tested with different
> kinds of setups, the mod_dingaling always loops at mod_dingaling.c line
> 2941and 3275, which results in no audio (rtp packets) even though the
> jingle
> session seems to have already established.
I can concur with you on this issue.


I asked for core dev's review of mod_dingaling on IRC a couple of times,
> looks like everybody is so busy that no one really takes a closer looks at
> the mod_dingaling.c yet.
I could understand, particularly working hard with an empty stomach.


Hope FS core dev will have some time to review this, we really appreciate
> that FS's gtalk calling can be back running again.
Especially, after they have returned from their good meals paid for and/or
donated by some of readers as shown 
http://freeswitch-users.2379917.n2.nabble.com/UPDATE-Buy-the-devs-dinner-tp6004646p6004646.html
here .

-----
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/mod-dingaling-inbound-audio-tp6005151p6020844.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From david.villasmil.work at gmail.com  Sun Feb 13 17:09:01 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sun, 13 Feb 2011 15:09:01 +0100
Subject: [Freeswitch-users] mod_distributor, lua and loop
In-Reply-To: <AANLkTimM=80_NoAaY18wix+z84c1ecfvFEN2QPWggxB0@mail.gmail.com>
References: <AANLkTinDD-n_J7B3VjvJcB=0iHsr6nnggUNyr0Nk8O7a@mail.gmail.com>
	<AANLkTik3m+zr3BLugtVG1+J9a7jK57fJHcWuvxygse76@mail.gmail.com>
	<1D6E8634-BF1E-42C4-91DC-B776B87CD894@gmail.com>
	<AANLkTinvZAZVT-3rdaG=mG6GKeGDRRrm_U0EdFU9B6o2@mail.gmail.com>
	<AANLkTimM=80_NoAaY18wix+z84c1ecfvFEN2QPWggxB0@mail.gmail.com>
Message-ID: <AANLkTinMAvZcijWO5UnaHKiuVemGjoEA7M7ZAePo_x7c@mail.gmail.com>

Hello,

You guys mean something like the following?

while session:answered()==fasle do
     session:execute("set","continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407")
     session:execute("bridge","sofia/gateway/${distributor(" ..
route_name.. ")}".. out_number)
end

mmm... don't know if i llike that. ;)

David

On Sat, Feb 12, 2011 at 9:24 PM, Steven Ayre <steveayre at gmail.com> wrote:
> See Anthony's reply... that'll produce a different call to distributor each
> time you do the bridge - which'll possibly return a different gateway each
> time, depending on your weightings and which other calls are calling
> distributor.
>
> -Steve
>
>
> On 12 February 2011 17:35, David Villasmil <david.villasmil.work at gmail.com>
> wrote:
>>
>> Thanks for answering Steve,
>>
>> That's exactly what i don't want to do, because i wanted to use the
>> "weight" feature in mod_distributor... but i gues i will end up doing
>> it :)
>>
>> Thanks
>>
>> David
>>
>> On Sat, Feb 12, 2011 at 6:26 PM, Steven Ayre <steveayre at gmail.com> wrote:
>> > I believe loop is specific to XML dialplan tags.
>> >
>> > Try implementing a lua loop, checking the hangup cause after the bridge
>> > and looping if it looks bad.
>> >
>> > Steve on iPhone
>> >
>> >
>> > On 12 Feb 2011, at 15:06, David Villasmil
>> > <david.villasmil.work at gmail.com> wrote:
>> >
>> >> Oh I forgot,
>> >>
>> >> I also have:
>> >>
>> >>
>> >> session:execute("set","continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407")
>> >>
>> >> thanks
>> >>
>> >> David
>> >>
>> >> On Sat, Feb 12, 2011 at 4:04 PM, David Villasmil
>> >> <david.villasmil.work at gmail.com> wrote:
>> >>> Hello All,
>> >>>
>> >>> from a lua script, i'm executing the following:
>> >>>
>> >>>
>> >>> session:execute("bridge","sofia/gateway/${distributor(" .. route_name
>> >>> .. ")}".. out_number .."")
>> >>>
>> >>> but the call doesn't falls to the next gw, i'm trying to add the
>> >>> loop="3" to the bridge but it doesn't work.
>> >>>
>> >>> I've trued like this:
>> >>>
>> >>>
>> >>> session:execute("set","loop=\"3\"")
>> >>> session:execute("bridge","sofia/gateway/${distributor(" .. route_name
>> >>> .. ")}".. out_number .."")
>> >>>
>> >>> and
>> >>>
>> >>>
>> >>> session:execute("bridge","sofia/gateway/${distributor(" .. route_name
>> >>> .. ")}".. out_number .." loop=\"3\"")
>> >>>
>> >>> and
>> >>>
>> >>> session:execute("bridge","{loop=\"3\"}sofia/gateway/${distributor(" ..
>> >>> route_name .. ")}".. out_number .."")
>> >>>
>> >>>
>> >>> Still nothing...
>> >>>
>> >>> any thoughts?
>> >>>
>> >>> 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
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 br at bsdpad.com  Mon Feb 14 00:36:16 2011
From: br at bsdpad.com (Ruslan Bukin)
Date: Mon, 14 Feb 2011 00:36:16 +0300
Subject: [Freeswitch-users] dp_tools/set- not worked SOMETIMES
In-Reply-To: <AANLkTimwKcuTJHJQp-Q4ER7h8Pd+Qj3vq5rkcRA+L4Uv@mail.gmail.com>
References: <AANLkTimwKcuTJHJQp-Q4ER7h8Pd+Qj3vq5rkcRA+L4Uv@mail.gmail.com>
Message-ID: <20110213213616.GA44119@bsdjail.com>

On Sun, Feb 13, 2011 at 07:30:52AM -0500, Kim Culhan wrote:
> On Sun, February 13, 2011 6:43 am, Ruslan Bukin wrote:
> > It is simplest usage of fs => so the issues is platform specific.
> > yes, it is latest freebsd 8.1 (thread lib was changed in fbsd7)
> > 3d issue: fs crashes time by time
> I am running git head on freebsd 8.2-RC2  amd64 and do not have any cases of
> fs crashing.

cool, I will update to 8.2rc2 and check again

> 
> I wanted to mention this because it reminded me of Anthony's comment:
> 
> >> > On Fri, Feb 11, 2011 at 06:00:16AM -0600, Anthony Minessale wrote:
> >> > > This is your second strange issue. Maybe you have some problem with
> >> > > your platform.
> I do not have enough  information to duplicate your situation with the loss
> of parameters.
> 
> If you would like I could try to help if you could supply a test recipe as
> Anthony suggested to duplicate the problem.

no recipe.. if you have no very strange behavior of fs then everything OK

> 
> thanks
> -kim
> 
> --

> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 abrods01 at gmail.com  Sun Feb 13 19:28:17 2011
From: abrods01 at gmail.com (A Brodskiy)
Date: Sun, 13 Feb 2011 11:28:17 -0500
Subject: [Freeswitch-users] DTMF issues
Message-ID: <AANLkTinx8U+=uukrAF0p0aSrWTpp02XPb55=YxuYC308@mail.gmail.com>

I am trying to use RFC 2833 <http://tools.ietf.org/html/rfc2833> for DTMF
with Freeswitch and tollfree termination providers like
tollfreetollfree.comand others, however it only seem to work with
1-800 numbers not 866. InBand
DTMF works perfectly, however I want to get 2833 working. Trying that with
SJPhone direct connection to termination provider RFC DTMF works fine. So it
seems that issue is with FreeSwitch, I also tried diffrent phone with FS
X-Lite and Pap2T and both have problems. I collected tcpdump captures and i
dont see any issues on the wire that is DTMF are being send and decodded by
wireshark as they go to provider. I am really lost for solution, Please
Help.

Thank you, Alex.
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From enver.yilmaz at esoft.com.tr  Mon Feb 14 00:01:03 2011
From: enver.yilmaz at esoft.com.tr (=?iso-8859-9?Q?Enver_Y=FDlmaz?=)
Date: Sun, 13 Feb 2011 21:01:03 +0000
Subject: [Freeswitch-users] ODBC Multiple Active Result Set (MARS) Problem
Message-ID: <2A4AA626FF646141AB1B10970E3FB4DB17141B0D@srv.esoft.com.tr>

Hi,

I'm using git head on windows 2008 r2 with sql 2008 r2. When I enable ODBC with SQLNCLI10 driver I experienced following problem ;

2011-02-11 20:45:52.556958 [CRIT] switch_odbc.c:280 The sql server is not responding for DSN FreeSWITCH [STATE: HY000 CODE 0 ERROR: [Microsoft][SQL Server Native Client 10.0]Connection is busy with results for another command
][244]

Freeswitch running and after minutes stop with above error then quit. I try to narrow ODBC config and I found when I enable ODBC on multiple profiles of sofia then this happens. On single profile and core and bunch of other modules is working without error.

As I understand, FS trying to query a resultset without closing previous one on same connection. MS has an option MARS (Multiple Active Result Set) on ODBC. If I enable MARS on ODBC then everything working but I'm not sure is this safe or not. MS Says MARS should not be enabled unless application specifically designed to work that way. http://msdn.microsoft.com/en-us/library/ms345109(v=sql.90).aspx

So what is your opinion? Should I go with mars option? Is this deserve to go to jira?

Thanks,
Enver Y?lmaz

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From philippe.sultan at gmail.com  Mon Feb 14 01:00:40 2011
From: philippe.sultan at gmail.com (Philippe Sultan)
Date: Sun, 13 Feb 2011 23:00:40 +0100
Subject: [Freeswitch-users] Aastra phone registration lost
In-Reply-To: <4D5577CE.1000603@communicatefreely.net>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>
	<4D52B987.7050002@communicatefreely.net>
	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>
	<4D534AFD.3020401@communicatefreely.net>
	<AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>
	<60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>
	<AANLkTimux4MBM1K_7itE4GnZQHjy_vR2Y6sr_z+u3gkR@mail.gmail.com>
	<4D5577CE.1000603@communicatefreely.net>
Message-ID: <AANLkTi=kavk9e7YgNKJ+QQo3X89yhir-Cqj9bL1KnSdj@mail.gmail.com>

Tim,

Based on your input, I kept those settings :
sip registration period: 3600
sip registration renewal timer: 3000

The phone registers to FS every 10 minutes. However, any other
registration period does not work like it should. It's obvious to me
that the phone ignores the expires parameter given in the Contact HF
and actually uses what I believe to be its default registation period
(3600 sec).

I tried to insert an Expires HF to the 200 REGISTER response sent back
from FS, and then the phone re-registers based on the time given in
this Header Field. It therefore seems to me that the phone is buggy,
even though adjusting the two parameters like you suggested it
perfectly solves the registration issue.

Thanks again,

Philippe

On Fri, Feb 11, 2011 at 6:54 PM, Tim St. Pierre
<fs-list at communicatefreely.net> wrote:
> Sure,
>
> I normally administer about 300 Aastra phones, with every model they
> make represented.
>
> I have 22 connected to our Freeswitch "beta" system, which will
> eventually become production.
>
> All the endpoints are behind NAT without exception. ?There are a number
> of legacy 9133i and 480i phones on the network that don't have the newer
> NAT traversal features available, but this doesn't seem to be a
> problem. ?I have some of the nat traversal options turned on in the
> sofia profile though, so fs will send media back to the originating
> address and port.
>
> They have been quite reliable, and the sound quality has been excellent,
> with the newer phones using g722 at 16KHz.
>
> There are a few advanced features that I haven't had a chance to play
> with yet, but here's what I have working:
>
> Regular calls, in and out.
> Intercom calls (auto-answer to speaker phone)
> Automatic update of destination name and number (updates when checking
> voice mail, and when calling an extension). ?Only on newer phones
> Blind and attended transfer
> Music on hold
> SIP using udp or tcp (haven't tried TLS yet)
> Fewer issues with DTMF than with asterisk, using rfc2833 dtmf (no issues
> as of yet).
> BLF lamps work correctly, flashing when the phone rings, lit steady when
> they are on the phone.
> Distinctive ringing works.
> I haven't tried SLA yet, but Aastra recently released a firmware update
> that fixes a missing header, reported to have broken correct SLA
> operation. ?I'm hoping to test that in the next week or two.
>
> The phones provision very nicely - we auto generate config using PHP
> scripts that generate a config file on the fly from the user database.
> These are very easy phones to deploy in large installations, or to the
> outside world (not readily accessible). ?They have just added some new
> features that allow for remote diagnostics of the phones as well.
>
> There is a great deal of XML programmability in the phones too, which
> I'm starting to use for call control and other useful things (updating
> forwarding rules in the database, or conference and recording control
> using ESL).
>
> Hope that helps!
>
> -Tim
>
> Aloysius Lloyd wrote:
>> Tim,
>>
>> Can you share your success stories FreeSWITCH and Aastra.
>>
>> Aastra Phones Behind the NAT?
>>
>> In my case Aastra phones registration not a problem.
>>
>> But calls drooped every 60 sec ... in the same environment Linksys and
>> Polycom works perfectly.
>>
>> How stable the Aastra phones with FreeSWITCH system.
>>
>> TIA
>>
>> 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
>



-- 
Philippe Sultan


From moises.silva at gmail.com  Mon Feb 14 05:31:52 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Sun, 13 Feb 2011 21:31:52 -0500
Subject: [Freeswitch-users] FreeTDM + libsng_isdn in Windows - can't get
 it to work (causes FS crasch)
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58ADB72787@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C58ADB72787@cooper>
Message-ID: <AANLkTimL667GpOsu+1c2prrpkz+qiqk4BeE996xihO=9@mail.gmail.com>

On Fri, Feb 11, 2011 at 1:28 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> So I guess by now it finds all files and dependencies, but instead it
> causes a crasch. Did I do something wrong, or is there something in the code
> that needs to be updated to get things working in Windows? This is just on a
> lab system, so I'm able to do whatever tests you want me to.
>
> Hi Peter,

We've been stress testing FreeTDM BRI support with libsng_isdn (not with
FreeSWITCH though), so I'd bet this is just a mismatch between libsng_isdn
and freetdm somehow.

Is this a 64 bit box?

I'll get you an updated libsng_isdn for Windows this week. As we speak we're
working towards getting automated builds/releases for libsng_isdn in
Windows, so you might be the first in trying them :-)

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 peter.olsson at visionutveckling.se  Mon Feb 14 08:46:32 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 14 Feb 2011 06:46:32 +0100
Subject: [Freeswitch-users] FreeTDM + libsng_isdn in Windows - can't get
 it to work (causes FS crasch)
In-Reply-To: <AANLkTimL667GpOsu+1c2prrpkz+qiqk4BeE996xihO=9@mail.gmail.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C58ADB72787@cooper>,
	<AANLkTimL667GpOsu+1c2prrpkz+qiqk4BeE996xihO=9@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58ADB7278F@cooper>

Thanks for the reply.

This is on a 32-bit box. I look forward testing your new release :)

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Moises Silva [moises.silva at gmail.com]
Skickat: den 14 februari 2011 03:31
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] FreeTDM + libsng_isdn in Windows - can't get it to work (causes FS crasch)

On Fri, Feb 11, 2011 at 1:28 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
So I guess by now it finds all files and dependencies, but instead it causes a crasch. Did I do something wrong, or is there something in the code that needs to be updated to get things working in Windows? This is just on a lab system, so I'm able to do whatever tests you want me to.

Hi Peter,

We've been stress testing FreeTDM BRI support with libsng_isdn (not with FreeSWITCH though), so I'd bet this is just a mismatch between libsng_isdn and freetdm somehow.

Is this a 64 bit box?

I'll get you an updated libsng_isdn for Windows this week. As we speak we're working towards getting automated builds/releases for libsng_isdn in Windows, so you might be the first in trying them :-)

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>
!DSPAM:4d5895b632761278110023!


From rmbertjones at comcast.net  Mon Feb 14 07:35:49 2011
From: rmbertjones at comcast.net (Bert Jones)
Date: Sun, 13 Feb 2011 23:35:49 -0500
Subject: [Freeswitch-users] Possible to communicate with FS using
	event_socket without ESL?
In-Reply-To: <AANLkTin8R0e8i29L6rVogEuBB05rcQObPWN4G_HvH_uj@mail.gmail.com>
References: <003101cbcb2d$fa0e9e60$ee2bdb20$@net>	<1297599372197-6020712.post@n2.nabble.com>
	<AANLkTin8R0e8i29L6rVogEuBB05rcQObPWN4G_HvH_uj@mail.gmail.com>
Message-ID: <007e01cbcc00$a825ffc0$f871ff40$@net>

Yes, thanks for the clarification Steve.  

Bert

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven
Ayre
Sent: Sunday, February 13, 2011 7:28 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Possible to communicate with FS using
event_socket without ESL?

 

The ESL is a plain text protocol. You can connect to the ESL port via telnet
and type in the ESL protocol commands manually and it'll connect fine.

-Steve



On 13 February 2011 12:16, mazilo <Nabble at slickdeals.endjunk.com> wrote:



Bert Jones wrote:
> Further I can talk to the server, authenticate and make
> calls using telnet,

I am just curious how you do this. Do care to explain how to do this using
telnet?

-----
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/Possible-to-communicate-with-F
S-using-event-socket-without-ESL-tp6020307p6020712.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 rmbertjones at comcast.net  Mon Feb 14 07:35:49 2011
From: rmbertjones at comcast.net (Bert Jones)
Date: Sun, 13 Feb 2011 23:35:49 -0500
Subject: [Freeswitch-users] Possible to communicate with FS
	using	event_socket without ESL?
In-Reply-To: <1297599372197-6020712.post@n2.nabble.com>
References: <003101cbcb2d$fa0e9e60$ee2bdb20$@net>
	<1297599372197-6020712.post@n2.nabble.com>
Message-ID: <008301cbcc00$a8452080$f8cf6180$@net>

Sure, here is what worked for me using Putty

First: enable mod_event_socket module in modules.conf.xml
	(It is enabled by default )
	<load module="mod_event_socket"/>

Next: configure in autoload_configs/event_socket.conf.xml as follows:
	<settings>
    		<param name="nat-map" value="false"/>
    		<param name="listen-ip" value="192.168.0.15"/>
   		 <param name="listen-port" value="8021"/>
   		 <param name="password" value="myPassword"/>
   		 <param name="apply-inbound-acl" value="lan"/>
  	</settings>
 
Then: configure Putty client as follows
	Set Putty to talk to server running FS over port 8021 using telnet
protocol, open session,

Finally:
	Open a telnet session and: 
		Send blank line, 
		Send "auth yourPassword", 
		Send blank line.  
		Receive "Ok"

Bert


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of mazilo
Sent: Sunday, February 13, 2011 7:16 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Possible to communicate with FS using
event_socket without ESL?



Bert Jones wrote:
> Further I can talk to the server, authenticate and make
> calls using telnet, 
I am just curious how you do this. Do care to explain how to do this using
telnet?

-----
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/Possible-to-communicate-with-F
S-using-event-socket-without-ESL-tp6020307p6020712.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 rmbertjones at comcast.net  Mon Feb 14 07:35:49 2011
From: rmbertjones at comcast.net (Bert Jones)
Date: Sun, 13 Feb 2011 23:35:49 -0500
Subject: [Freeswitch-users] Possible to communicate with FS using
	event_socket without ESL?
In-Reply-To: <AANLkTinhUwUfcdnMTkdtgpV7W+wBYzCC9oULrexsjFEZ@mail.gmail.com>
References: <003101cbcb2d$fa0e9e60$ee2bdb20$@net>
	<AANLkTinhUwUfcdnMTkdtgpV7W+wBYzCC9oULrexsjFEZ@mail.gmail.com>
Message-ID: <008401cbcc00$a8666420$f9332c60$@net>

Thanks mayamatakeshi.  I added flushes after each send, but still not
working.  Setting up packet capture now.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
mayamatakeshi
Sent: Sunday, February 13, 2011 4:28 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Possible to communicate with FS using
event_socket without ESL?

 

 

On Sun, Feb 13, 2011 at 12:27 PM, Bert Jones <rmbertjones at comcast.net>
wrote:

Hello,

 

Am I correct in assuming that an app can talk directly to FS using
event_socket without the use of the ESL?  I am attempting to do this, but
having trouble authenticating. I wanted to make sure I was not overlooking
something basis in my approach.

 

I have successfully loaded FS on a windows XP server and can register phones
and make calls.  Further I can talk to the server, authenticate and make
calls using telnet, but when attempting to do the same via an app written in
.net and using a socket, I am unable to authenticate.  

 

Running my app using a tcpSocket, I receive the "Content-Type: auth/request"
upon connection of the socket, but when sending "auth password\n\n" it
appears that the connection times out after about 10 seconds and I receive:
"Content-Type: text/disconnect-notice" & vbLf & "Content-Length: 67".
mod_event_socket is enabled in modules.conf.xml .

 

Is there something obvious that I am missing conceptually, or should this
work?


Maybe the .NET component is set to buffer data before sending it. There
might be some method to flush the data immediately.
If this doesn't solve it. I would get a packet capture.

r,
takeshi

 

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From rmbertjones at comcast.net  Mon Feb 14 07:40:32 2011
From: rmbertjones at comcast.net (Bert Jones)
Date: Sun, 13 Feb 2011 23:40:32 -0500
Subject: [Freeswitch-users] Possible to communicate with FS using
	event_socket without ESL?
In-Reply-To: <AANLkTi=5zyKN1Ef+nJspyMrOuw=VE+fV-RcCQ5S148L=@mail.gmail.com>
References: <003101cbcb2d$fa0e9e60$ee2bdb20$@net>
	<AANLkTi=5zyKN1Ef+nJspyMrOuw=VE+fV-RcCQ5S148L=@mail.gmail.com>
Message-ID: <008901cbcc01$517903b0$f46b0b10$@net>

Thanks for the clarification.  I too suspect something in my communications.
Just wanted to make sure I was not missing something. Will move my
communication app to a separate machine from the FS server and run
Wireshark.  Thanks for recommendation.

Bert

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven
Ayre
Sent: Sunday, February 13, 2011 7:29 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Possible to communicate with FS using
event_socket without ESL?

 

Without the use of the ESL *protocol* - no. Without the use of the supplied
ESL *libraries* - correct.

You're along the right lines on implementing your own protocol. If you can
connect and authenticate via telnet from the same computer you're trying
your app from then there should be no problem with FS.

The fact you say it takes 10s for FS to respond indicates you're not sending
the auth line. You should get either a success or failure message straight
away in reply to that. So it sounds like the auth isn't getting sent at all.

Try using Wireshark to view what's going over the network to check it's what
you expect.

-Steve







On 13 February 2011 03:27, Bert Jones <rmbertjones at comcast.net> wrote:

Hello,

 

Am I correct in assuming that an app can talk directly to FS using
event_socket without the use of the ESL?  I am attempting to do this, but
having trouble authenticating. I wanted to make sure I was not overlooking
something basis in my approach.

 

I have successfully loaded FS on a windows XP server and can register phones
and make calls.  Further I can talk to the server, authenticate and make
calls using telnet, but when attempting to do the same via an app written in
.net and using a socket, I am unable to authenticate.  

 

Running my app using a tcpSocket, I receive the "Content-Type: auth/request"
upon connection of the socket, but when sending "auth password\n\n" it
appears that the connection times out after about 10 seconds and I receive:
"Content-Type: text/disconnect-notice" & vbLf & "Content-Length: 67".
mod_event_socket is enabled in modules.conf.xml .

 

Is there something obvious that I am missing conceptually, or should this
work?

 

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 cowboys550 at hotmail.com  Mon Feb 14 06:38:05 2011
From: cowboys550 at hotmail.com (Chris A)
Date: Mon, 14 Feb 2011 05:38:05 +0200
Subject: [Freeswitch-users] Freeswitch design!
Message-ID: <COL115-W3548A7784C17257AEB46E883D00@phx.gbl>


Hi ! 
Can i know the design overview of the "freeswitch" ?
what is inside it ? 
how the media is precessed ?
what is the controlled protocol used?
 
thx u  		 	   		  
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From erik.dekkers at wvds.nl  Mon Feb 14 11:28:05 2011
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Mon, 14 Feb 2011 09:28:05 +0100
Subject: [Freeswitch-users] Freeswitch design!
In-Reply-To: <COL115-W3548A7784C17257AEB46E883D00@phx.gbl>
References: <COL115-W3548A7784C17257AEB46E883D00@phx.gbl>
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE07873018A9@FS1.wvds.local>

DUDE, are you on drugs?

Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Chris A
Verzonden: maandag 14 februari 2011 4:38
Aan: freeswitch-users at lists.freeswitch.org
Onderwerp: [Freeswitch-users] Freeswitch design!

Hi !
Can i know the design overview of the "freeswitch" ?
what is inside it ?
how the media is precessed ?
what is the controlled protocol used?

thx u
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From frankie.k.yiu at gmail.com  Mon Feb 14 12:30:10 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Mon, 14 Feb 2011 01:30:10 -0800
Subject: [Freeswitch-users] How to create a C++ application (project),
	that can code FS C code function?
Message-ID: <AANLkTikfxCrHsC5S-DZVfQejSWiWtF0HRM15MXksX-qK@mail.gmail.com>

Hi there,

I am new to FreeSwitch and I would like to implement a C++ application / mod
that would do some analysis of RTP packets.  I would like to ask what mod I
should use as a template to create my C++ project so that I can access
the core FreeSwitch APIs / functions, since they are in C language?

I know under Endpoints, there are 2 MODs that are using C++ code (mod_h323
and mod_opal), but I don't know if I should implement the same way if I am
creating an "application" instead of "endpoints".

Please kindly let me know if there is any special / required function that I
need to implemented in C++ code.

Thanks in advance.

Frankie
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From jgallartm at gmail.com  Mon Feb 14 13:48:45 2011
From: jgallartm at gmail.com (Javier Gallart)
Date: Mon, 14 Feb 2011 11:48:45 +0100
Subject: [Freeswitch-users] g729 packets skipped in passthrough mode
Message-ID: <AANLkTikebJ1RKxDtJp9RzK6fbGyeR1pJQKE+X+G1iJ88@mail.gmail.com>

Hello list

I've just noticed a problem in our Freeswitch  (version 1.0.head
(git-3fd7b8f 2011-02-13 18-04-17 -0500)) when it stays in the middle of a
g729 negotiated call. I haven't found information to this particular problem
previous posts to the list.

The dialplan is trivial, the only relevant line is a bridge to the
destination ip address:
<action application="bridge" data="sofia/internal/${cld_number}@${destip}"/>
The call goes through but the quality is very bad becuse half of the packets
are not relayed from the FS to the originating ip. From the logs:
----A-LEG----

2011-02-13 23:17:50.375026 [DEBUG] sofia.c:4670 Remote SDP:
c=IN IP4 x.x.x.x
m=audio 14098 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=silenceSupp:off - - - -
2011-02-13 23:17:50.375026 [DEBUG] sofia_glue.c:4474 Audio Codec Compare
[G729:18:8000:20:8000]/[G729:18:8000:20:8000]
2011-02-13 23:17:50.375026 [DEBUG] sofia_glue.c:2757 Set Codec
sofia/internal/34917019888 at 79.170.68.158:5060 G729/8000 20 ms 160 samples
8000 bits

------B-LEG------

2011-02-13 23:17:50.762967 [DEBUG] sofia.c:4670 Remote SDP:
c=IN IP4 y.y.y.y
m=audio 20098 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
2011-02-13 23:17:50.762967 [DEBUG] sofia_glue.c:4474 Audio Codec Compare
[G729:18:8000:20:8000]/[G729:18:8000:20:8000]
2011-02-13 23:17:50.762967 [DEBUG] sofia_glue.c:2757 Set Codec
sofia/internal/59381855597 at 200.25.204.69 G729/8000 20 ms 160 samples 8000
bits

And this is what tshark says (x.x.x.x origin media ip: y.y.y.y dest media
ip: z.z.z.z FS ip):
 2.289121 y.y.y.y -> z.z.z.z RTP PT=ITU-T G.729, SSRC=0x75061F76, Seq=8158,
Time=2931222712, Mark
 2.309166 y.y.y.y -> z.z.z.z RTP PT=ITU-T G.729, SSRC=0x75061F76, Seq=8159,
Time=2931222872, Mark
 2.328814 z.z.z.z -> x.x.x.x RTP PT=ITU-T G.729, SSRC=0x621B8D66, Seq=47609,
Time=15680
 2.329147 y.y.y.y -> z.z.z.z RTP PT=ITU-T G.729, SSRC=0x75061F76, Seq=8160,
Time=2931223032, Mark
 2.349253 y.y.y.y -> z.z.z.z RTP PT=ITU-T G.729, SSRC=0x75061F76, Seq=8161,
Time=2931223192, Mark
 2.369036 z.z.z.z -> x.x.x.x RTP PT=ITU-T G.729, SSRC=0x621B8D66, Seq=47610,
Time=16160
 2.369274 y.y.y.y -> z.z.z.z RTP PT=ITU-T G.729, SSRC=0x75061F76, Seq=8162,
Time=2931223352, Mark
 2.389271 y.y.y.y -> z.z.z.z RTP PT=ITU-T G.729, SSRC=0x75061F76, Seq=8163,
Time=2931223512, Mark

I've tried to explicitly set ptime at 20ms at switch.conf -although it's not
necessary afaik-. Has any one experienced this same issue?

Thanks in advance..

Regards
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From wstephen80 at gmail.com  Mon Feb 14 14:46:28 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Mon, 14 Feb 2011 12:46:28 +0100
Subject: [Freeswitch-users] ESL: how to filter two uuid's?
Message-ID: <AANLkTikjDdsyZFf=zcYcerLz7P_w-1+hvxE_rbZwccTj@mail.gmail.com>

If I have 2 uuid (uuid_inbound and uuid_outbound) in an ESL socket outbound
application, there is a way to filter events related to this two uuid?

If I try to do:

connection.filter("unique-id", uuid_inbound);
connection.filter("unique-id", uuid_outbound);

it seems that the second filter overwrite the first one and my application
receives only uuid_outbound related events.

I have tried also an:

connection.filter("unique-id", uuid_inbound + "|" + uuid_outbound);

but in this way no event are received.

There is a way to filter 2 specific uuid's or I have to receive all events
and implement the filtering in my ESL application?

Stephen
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From chris at cheeky.org  Mon Feb 14 16:04:12 2011
From: chris at cheeky.org (Chris Hemmings)
Date: Mon, 14 Feb 2011 13:04:12 +0000
Subject: [Freeswitch-users] ESL: how to filter two uuid's?
In-Reply-To: <AANLkTikjDdsyZFf=zcYcerLz7P_w-1+hvxE_rbZwccTj@mail.gmail.com>
References: <AANLkTikjDdsyZFf=zcYcerLz7P_w-1+hvxE_rbZwccTj@mail.gmail.com>
Message-ID: <AANLkTikmL03v+0o6eqvbci30sXZSH7GXURFwi-ZuXGu3@mail.gmail.com>

This seemed to work for me in Perl:

$con->sendRecv("event plain ALL");
$con->sendRecv("filter Unique-ID ".$uuid);
$con->sendRecv("filter Unique-ID ".$out_uuid);

Cheers,

Chris

On 14 February 2011 11:46, Stephen Wilde <wstephen80 at gmail.com> wrote:

> If I have 2 uuid (uuid_inbound and uuid_outbound) in an ESL socket outbound
> application, there is a way to filter events related to this two uuid?
>
> If I try to do:
>
> connection.filter("unique-id", uuid_inbound);
> connection.filter("unique-id", uuid_outbound);
>
> it seems that the second filter overwrite the first one and my application
> receives only uuid_outbound related events.
>
> I have tried also an:
>
> connection.filter("unique-id", uuid_inbound + "|" + uuid_outbound);
>
> but in this way no event are received.
>
> There is a way to filter 2 specific uuid's or I have to receive all events
> and implement the filtering in my ESL application?
>
> 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 edpimentl at gmail.com  Mon Feb 14 17:44:40 2011
From: edpimentl at gmail.com (EdPimentl)
Date: Mon, 14 Feb 2011 09:44:40 -0500
Subject: [Freeswitch-users] Freeswitch design!
In-Reply-To: <COL115-W3548A7784C17257AEB46E883D00@phx.gbl>
References: <COL115-W3548A7784C17257AEB46E883D00@phx.gbl>
Message-ID: <AANLkTiki6xQ8cxJhf90sijQCQyi0rnT0jJMGiCNmSQ2v@mail.gmail.com>

https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-telephony-systems/book

http://wiki.freeswitch.org/wiki/Users_Guide_Introduction
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From frank at telonium.com  Mon Feb 14 18:03:14 2011
From: frank at telonium.com (Frank Park)
Date: Mon, 14 Feb 2011 10:03:14 -0500
Subject: [Freeswitch-users] Intermittent curl execution after rxfax
In-Reply-To: <AANLkTimeCzSwCMfA7FoCzQn8a=xGSRuPUUkitN3EA2qa@mail.gmail.com>
References: <AANLkTi=qqaP0bAhdVMW309fm2vfr41BP13c81epOieW0@mail.gmail.com>
	<4961923E-BF1C-4202-9441-A2FF2B7E5E57@gmail.com>
	<AANLkTimeCzSwCMfA7FoCzQn8a=xGSRuPUUkitN3EA2qa@mail.gmail.com>
Message-ID: <AANLkTin9tsFqjKz+40QjG7PRkHTnnw0X9FUgCD_O0KOM@mail.gmail.com>

Didn't think about the other end hanging up so quickly.
I do currently use xml_cdr for CDR, and prefer that I keep that separate.
I will try the api_hangup_hook, that might work better.

Thank you, sirs

On Sat, Feb 12, 2011 at 5:50 PM, Michael Collins <msc at freeswitch.org> wrote:

> How about an api_hangup_hook?
>
>
> On Sat, Feb 12, 2011 at 3:24 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Perhaps the aleg hangs up after rxfax but before curl is executed? If the
>> aleg hangs up the remaining extension actions won't run.
>>
>> Looks like you're only using it to indicate a fax has been received? Try
>> using ESL or CDRs. Mod_xml_cdr can post a CDR to a server and will have
>> various variables set that indicate the fax has been received.
>>
>> Steve on 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
>
>


-- 

----=======================----
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 brian at freeswitch.org  Mon Feb 14 19:28:37 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 14 Feb 2011 10:28:37 -0600
Subject: [Freeswitch-users] g729 packets skipped in passthrough mode
In-Reply-To: <AANLkTikebJ1RKxDtJp9RzK6fbGyeR1pJQKE+X+G1iJ88@mail.gmail.com>
References: <AANLkTikebJ1RKxDtJp9RzK6fbGyeR1pJQKE+X+G1iJ88@mail.gmail.com>
Message-ID: <8704BA31-1913-4034-9DAF-ACD60B1BE8B8@freeswitch.org>

What exactly is the problem?  I see no issue here can you elaborate on what you're seeing?

/b

On Feb 14, 2011, at 4:48 AM, Javier Gallart wrote:

> I've tried to explicitly set ptime at 20ms at switch.conf -although it's not necessary afaik-. Has any one experienced this same issue?
> 
> Thanks in advance..
> 
> Regards



From leonardo.bidinoto at voicetechnology.com.br  Mon Feb 14 16:50:50 2011
From: leonardo.bidinoto at voicetechnology.com.br (Leonardo P. Bidinoto)
Date: Mon, 14 Feb 2011 11:50:50 -0200
Subject: [Freeswitch-users] pthread_mutex_lock error in conference transfer
	function
Message-ID: <AANLkTinY6koyWcVzE8AqgSyxCTMmXMBvwgxdcGb1imHk@mail.gmail.com>

Bug url = http://jira.freeswitch.org/browse/FS-3052

Hello guys,
I'm new here.
Getting this problem that are taking nights of sleep.


http://dl.dropbox.com/u/710501/bug%23002%20Freeswitch%20lock%20null/backtrace.txt
as you can see this link above which i get from core file with gdb, i.m get
problem in the transfer function on mod_conference, and this is making
FreeSwitch to crash.

i'm providing the link to download the core file to analyses:
http://dl.dropbox.com/u/21117321/core.11120.gz

Plz, can someone help me with that?!

Thanks.

-- 
Leonardo Pires Bidinoto
Voice Technology
www.voicetechnology.com.br
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From leonardo.bidinoto at voicetechnology.com.br  Mon Feb 14 17:27:41 2011
From: leonardo.bidinoto at voicetechnology.com.br (Leonardo P. Bidinoto)
Date: Mon, 14 Feb 2011 12:27:41 -0200
Subject: [Freeswitch-users] Problems with mp3 larger than 70MB while playing
 files into the conference in the "perpetual-sound" variable
Message-ID: <AANLkTikWBYU_suW2i1tg5HHhe4P1U18Hqs5E1UZZ=UaQ@mail.gmail.com>

Bug url = http://jira.freeswitch.org/browse/FS-3053

Hello guys, me again. Now with another bug that i found.
I set the "perpetual-sound" variable in the conference.conf.xml to play mp3
files that my customer chooses.
but when a mp3 with 71mb or more is played, it crashed the machine when was
more than one user listening to the mp3.
i tried a file with 99mb and crash too.
but when i use a file with 10mb or less, works fine.

i looked at the core file and see this:
===================================================================================================================================================================
FreeSWITCH Version 1.0.head (git-2401fec 2011-02-08 13-01-42 -0600)
---------------------------------------------------------------------------------
Core was generated by `/usr/local/freeswitch/bin/freeswitch -nc -core
-waste'.
Program terminated with signal 6, Aborted.
#0  0x002f6402 in __kernel_vsyscall ()
(gdb) bt
#0  0x002f6402 in __kernel_vsyscall ()
#1  0x003d9df0 in vfprintf () from /lib/libc.so.6
#2  0x003db701 in vfprintf () from /lib/libc.so.6
#3  0x0041228b in strcoll_l () from /lib/libc.so.6
#4  0x0041a5a5 in ____wcstold_l_internal () from /lib/libc.so.6
#5  0x0041a9e9 in ____wcstold_l_internal () from /lib/libc.so.6
#6  0x0033e675 in parse_new_id3 (fr=0xbc53390, first4bytes=<value optimized
out>) at id3.c:736
#7  0x0035a185 in read_frame (fr=0xbc53390) at parse.c:442
#8  0x0034f692 in get_next_frame (mh=0xbc53390) at libmpg123.c:495
#9  0x00350837 in mpg123_decode (mh=0xbc53390, inmemory=0x0, inmemsize=0,
outmemory=0xa1b21e0 "", outmemsize=16384, done=0x700d01f8) at
libmpg123.c:692
#10 0x003509a3 in mpg123_read (mh=0xbc53390, out=0xa1b21e0 "", size=16384,
done=0x700d01f8) at libmpg123.c:602
#11 0x00305f1b in decode_fd (handle=0xa28b210, data=0xa62b7e0,
len=0x700d0258) at
/usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:310
#12 shout_file_read (handle=0xa28b210, data=0xa62b7e0, len=0x700d0258) at
/usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:853
#13 0x00c89eba in switch_core_file_read (fh=0xa28b210, data=0xa289148,
len=0x700d3350) at src/switch_core_file.c:241
#14 0x001a42fc in conference_thread_run (thread=0x9ff50e0, obj=0x9ff4c30) at
/usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c:1188
#15 0x00d27726 in dummy_worker (opaque=0x9ff50e0) at
threadproc/unix/thread.c:138
#16 0x00504832 in _L_unlock_3870 () from /lib/libpthread.so.0
#17 0x0050483d in _L_unlock_3870 () from /lib/libpthread.so.0
Backtrace stopped: previous frame identical to this frame (corrupt stack?)
(gdb) frame 6
#6  0x0033e675 in parse_new_id3 (fr=0xbc53390, first4bytes=<value optimized
out>) at id3.c:736
736                             free(tagdata);
(gdb) list
731                             {
732                                     if(NOQUIET) error("ID3v2: Duh, not
able to read ID3v2 tag data.");
733                                     ret = ret2;
734                             }
735     tagparse_cleanup:
736                             free(tagdata);
737                     }
738                     else
739                     {
740                             if(NOQUIET) error1("ID3v2: Arrg! Unable to
allocate %lu bytes for interpreting ID3v2 data - trying to skip instead.",
length);
(gdb) print tag data
No symbol "tag" in current context.
(gdb) print tagdata
$1 = <value optimized out>
(gdb) frame 13
#13 0x00c89eba in switch_core_file_read (fh=0xa28b210, data=0xa289148,
len=0x700d3350) at src/switch_core_file.c:241
241                                     if ((status =
fh->file_interface->file_read(fh, fh->pre_buffer_data, &rlen)) !=
SWITCH_STATUS_SUCCESS || !rlen) {
(gdb) list
236
237                     if (!switch_test_flag(fh, SWITCH_FILE_BUFFER_DONE))
{
238                             rlen = asis ? fh->pre_buffer_datalen :
fh->pre_buffer_datalen / 2;
239
240                             if (switch_buffer_inuse(fh->pre_buffer) <
rlen * 2) {
241                                     if ((status =
fh->file_interface->file_read(fh, fh->pre_buffer_data, &rlen)) !=
SWITCH_STATUS_SUCCESS || !rlen) {
242                                             switch_set_flag(fh,
SWITCH_FILE_BUFFER_DONE);
243                                     } else {
244                                             fh->samples_in += rlen;
245                                             if (fh->channels > 1) {
(gdb) print fh
$2 = (switch_file_handle_t *) 0xa28b210
(gdb) print *fh
$3 = {file_interface = 0xb212ed68, flags = 2057, fd = 0x0, samples = 0,
samplerate = 8000, native_rate = 8000, channels = 1 '\001', format = 0,
sections = 0, seekable = 1,
  sample_count = 0, speed = 0, memory_pool = 0xa28b150, prebuf = 0, interval
= 0, private_info = 0xa1b20c8, handler = 0x0, pos = 0, audio_buffer = 0x0,
sp_audio_buffer = 0x0, thresh = 0,
  silence_hits = 0, offset_pos = 0, samples_in = 0, samples_out = 0, vol =
0, resampler = 0x0, buffer = 0x0, dbuf = 0x0, dbuflen = 0, pre_buffer =
0xa624858,
  pre_buffer_data = 0xa62b7e0 "", pre_buffer_datalen = 65536, file =
0x1abf00
"/usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c",
  func = 0x1acedb "conference_play_file", line = 2993, file_path = 0xa28b2d0
"/usr/local/freeswitch/sounds/flex/app186/en/conferencemusics/73036.mp3",
spool_path = 0x0, prefix = 0x0}
(gdb)
===================================================================================================================================================================
Someone have a ideia of whats happenning?

i'm providing the link to download the core file to analyses:
http://dl.dropbox.com/u/21117321/core.26585.gz

Thanks.

-- 
Leonardo Pires Bidinoto
Voice Technology
www.voicetechnology.com.br
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From eric.lamer at phoenixsecure.com  Mon Feb 14 18:45:08 2011
From: eric.lamer at phoenixsecure.com (Eric Lamer)
Date: Mon, 14 Feb 2011 15:45:08 +0000
Subject: [Freeswitch-users] VMWare
Message-ID: <0B5E13F15EF4C0459656824B8D16134B4CC62498@PhoenixSrv01.phoenixsecure.com>

Hi,

 

   I am trying to install the ISO 1.0.3 on a VMWare workstation to test it.
The install work but I don't have any web server installed.  Any idea why?

 

Thanks.

 

 

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From brian at freeswitch.org  Mon Feb 14 19:29:39 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 14 Feb 2011 10:29:39 -0600
Subject: [Freeswitch-users] Freeswitch design!
In-Reply-To: <CDF5E00590C9BA4AB96BE7CDD28C86AE07873018A9@FS1.wvds.local>
References: <COL115-W3548A7784C17257AEB46E883D00@phx.gbl>
	<CDF5E00590C9BA4AB96BE7CDD28C86AE07873018A9@FS1.wvds.local>
Message-ID: <756AD8B6-1403-4B4F-BDF5-87FDCB2C94C0@freeswitch.org>

DUDE, Clearly you're the one on drugs and don't respond like that again please.  Be respectful if possible I know sometimes its hard... but don't drive people away or I'll get the clue-by-4 out and beat you with it.

/b

On Feb 14, 2011, at 2:28 AM, Erik Dekkers wrote:

> DUDE, are you on drugs?
>  

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From brian at freeswitch.org  Mon Feb 14 19:30:51 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 14 Feb 2011 10:30:51 -0600
Subject: [Freeswitch-users] How to create a C++ application (project),
	that can code FS C code function?
In-Reply-To: <AANLkTikfxCrHsC5S-DZVfQejSWiWtF0HRM15MXksX-qK@mail.gmail.com>
References: <AANLkTikfxCrHsC5S-DZVfQejSWiWtF0HRM15MXksX-qK@mail.gmail.com>
Message-ID: <54918FB2-4DFD-47AF-9F39-8335E7187ED4@freeswitch.org>

Every bit of libFreeSWITCH is able to be used in a C++ application see we also have some modules that are in C++ and it all just works fine.  

/b

On Feb 14, 2011, at 3:30 AM, Frankie Yiu wrote:

> Hi there,
>  
> I am new to FreeSwitch and I would like to implement a C++ application / mod that would do some analysis of RTP packets.  I would like to ask what mod I should use as a template to create my C++ project so that I can access the core FreeSwitch APIs / functions, since they are in C language? 
>  
> I know under Endpoints, there are 2 MODs that are using C++ code (mod_h323 and mod_opal), but I don't know if I should implement the same way if I am creating an "application" instead of "endpoints". 
>  
> Please kindly let me know if there is any special / required function that I need to implemented in C++ code.
> 
> Thanks in advance.
>  
> Frankie
> __________



From brian at freeswitch.org  Mon Feb 14 19:33:17 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 14 Feb 2011 10:33:17 -0600
Subject: [Freeswitch-users] Problems with mp3 larger than 70MB while
	playing files into the conference in the "perpetual-sound" variable
In-Reply-To: <AANLkTikWBYU_suW2i1tg5HHhe4P1U18Hqs5E1UZZ=UaQ@mail.gmail.com>
References: <AANLkTikWBYU_suW2i1tg5HHhe4P1U18Hqs5E1UZZ=UaQ@mail.gmail.com>
Message-ID: <8B6F3C08-5497-4C84-9D2C-5182762BB4F7@freeswitch.org>

Wait for it... NEVER run 32bit OS's on a 64bit CPU please.  You'll get way better performance out of things.  Have you tried this?  I don't trust that MP3 lib to make the right decisions on data types when pulling this type of thing off.

/b

On Feb 14, 2011, at 8:27 AM, Leonardo P. Bidinoto wrote:

> Bug url = http://jira.freeswitch.org/browse/FS-3053
> 
> Hello guys, me again. Now with another bug that i found.
> I set the "perpetual-sound" variable in the conference.conf.xml to play mp3 files that my customer chooses.
> but when a mp3 with 71mb or more is played, it crashed the machine when was more than one user listening to the mp3.
> i tried a file with 99mb and crash too.
> but when i use a file with 10mb or less, works fine.

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From brian at freeswitch.org  Mon Feb 14 19:35:07 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 14 Feb 2011 10:35:07 -0600
Subject: [Freeswitch-users] VMWare
In-Reply-To: <0B5E13F15EF4C0459656824B8D16134B4CC62498@PhoenixSrv01.phoenixsecure.com>
References: <0B5E13F15EF4C0459656824B8D16134B4CC62498@PhoenixSrv01.phoenixsecure.com>
Message-ID: <0F5E8440-621D-43A7-8848-8918F8E0FA29@freeswitch.org>

Please refer to this http://wiki.freeswitch.org/wiki/Installation_Guide

I don't recommend using 1.0.3 EVER.

/b

On Feb 14, 2011, at 9:45 AM, Eric Lamer wrote:

> Hi,
>  
>    I am trying to install the ISO 1.0.3 on a VMWare workstation to test it.  The install work but I don?t have any web server installed.  Any idea why?
>  
> Thanks.
>  

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From anthony.minessale at gmail.com  Mon Feb 14 19:38:57 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 14 Feb 2011 10:38:57 -0600
Subject: [Freeswitch-users] g729 packets skipped in passthrough mode
In-Reply-To: <8704BA31-1913-4034-9DAF-ACD60B1BE8B8@freeswitch.org>
References: <AANLkTikebJ1RKxDtJp9RzK6fbGyeR1pJQKE+X+G1iJ88@mail.gmail.com>
	<8704BA31-1913-4034-9DAF-ACD60B1BE8B8@freeswitch.org>
Message-ID: <AANLkTin9SV49V9BqFRoi_6CDTLRUjWf3yS82oetKRXzV@mail.gmail.com>

I bet its 20ms vs 30
set passthru_ptime_mismatch to true either in vars.xml or in your
dialplan both legs through export on the a leg or set in the a leg and
in the {} on b.


On Mon, Feb 14, 2011 at 10:28 AM, Brian West <brian at freeswitch.org> wrote:
> What exactly is the problem? ?I see no issue here can you elaborate on what you're seeing?
>
> /b
>
> On Feb 14, 2011, at 4:48 AM, Javier Gallart wrote:
>
>> I've tried to explicitly set ptime at 20ms at switch.conf -although it's not necessary afaik-. Has any one experienced this same issue?
>>
>> Thanks in advance..
>>
>> 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
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 Feb 14 19:54:43 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 14 Feb 2011 10:54:43 -0600
Subject: [Freeswitch-users] pthread_mutex_lock error in conference
 transfer function
In-Reply-To: <AANLkTinY6koyWcVzE8AqgSyxCTMmXMBvwgxdcGb1imHk@mail.gmail.com>
References: <AANLkTinY6koyWcVzE8AqgSyxCTMmXMBvwgxdcGb1imHk@mail.gmail.com>
Message-ID: <AANLkTimr2ZQYWmriVE-G3cVg162TKmwFVP8XczoC-nwC@mail.gmail.com>

Reporting your problem to jira was the right thing to do.
Reporting it here too telling us to look at it was the wrong thing to
do, we already saw the jira.

You are making several mistakes please see the comments on your bug notes.


On Mon, Feb 14, 2011 at 7:50 AM, Leonardo P. Bidinoto
<leonardo.bidinoto at voicetechnology.com.br> wrote:
> Bug url = http://jira.freeswitch.org/browse/FS-3052
>
> Hello guys,
> I'm new here.
> Getting this problem that are taking nights of sleep.
>
>
> http://dl.dropbox.com/u/710501/bug%23002%20Freeswitch%20lock%20null/backtrace.txt
> as you can see this link above which i get from core file with gdb, i.m get
> problem in the transfer function on mod_conference, and this is making
> FreeSwitch to crash.
>
> i'm providing the link to download the core file to analyses:
> http://dl.dropbox.com/u/21117321/core.11120.gz
>
> Plz, can someone help me with that?!
>
> Thanks.
>
> --
> Leonardo Pires Bidinoto
> Voice Technology
> www.voicetechnology.com.br
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 lloyd.aloysius at sunteltech.ca  Mon Feb 14 19:54:38 2011
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Mon, 14 Feb 2011 11:54:38 -0500
Subject: [Freeswitch-users] Aastra phone registration lost
In-Reply-To: <4D5577CE.1000603@communicatefreely.net>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>
	<4D52B987.7050002@communicatefreely.net>
	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>
	<4D534AFD.3020401@communicatefreely.net>
	<AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>
	<60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>
	<AANLkTimux4MBM1K_7itE4GnZQHjy_vR2Y6sr_z+u3gkR@mail.gmail.com>
	<4D5577CE.1000603@communicatefreely.net>
Message-ID: <AANLkTinJK+e9hZJbk4ur_OQwGV5B7tRLrpPMH=TU5MOo@mail.gmail.com>

Tim,

Thank you for the information.

I have around 275 Aastra 9133i models phones in production .These phones
installed 31/2 years ago. I am trying to migrate to FreeSWITCH could not
make it work reliably.

What are the profile settings turned on for these phones works reliably?


Thanks
Lloyd



On Fri, Feb 11, 2011 at 12:54 PM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:

> Sure,
>
> I normally administer about 300 Aastra phones, with every model they
> make represented.
>
> I have 22 connected to our Freeswitch "beta" system, which will
> eventually become production.
>
> All the endpoints are behind NAT without exception.  There are a number
> of legacy 9133i and 480i phones on the network that don't have the newer
> NAT traversal features available, but this doesn't seem to be a
> problem.  I have some of the nat traversal options turned on in the
> sofia profile though, so fs will send media back to the originating
> address and port.
>
> They have been quite reliable, and the sound quality has been excellent,
> with the newer phones using g722 at 16KHz.
>
> There are a few advanced features that I haven't had a chance to play
> with yet, but here's what I have working:
>
> Regular calls, in and out.
> Intercom calls (auto-answer to speaker phone)
> Automatic update of destination name and number (updates when checking
> voice mail, and when calling an extension).  Only on newer phones
> Blind and attended transfer
> Music on hold
> SIP using udp or tcp (haven't tried TLS yet)
> Fewer issues with DTMF than with asterisk, using rfc2833 dtmf (no issues
> as of yet).
> BLF lamps work correctly, flashing when the phone rings, lit steady when
> they are on the phone.
> Distinctive ringing works.
> I haven't tried SLA yet, but Aastra recently released a firmware update
> that fixes a missing header, reported to have broken correct SLA
> operation.  I'm hoping to test that in the next week or two.
>
> The phones provision very nicely - we auto generate config using PHP
> scripts that generate a config file on the fly from the user database.
> These are very easy phones to deploy in large installations, or to the
> outside world (not readily accessible).  They have just added some new
> features that allow for remote diagnostics of the phones as well.
>
> There is a great deal of XML programmability in the phones too, which
> I'm starting to use for call control and other useful things (updating
> forwarding rules in the database, or conference and recording control
> using ESL).
>
> Hope that helps!
>
> -Tim
>
> Aloysius Lloyd wrote:
> > Tim,
> >
> > Can you share your success stories FreeSWITCH and Aastra.
> >
> > Aastra Phones Behind the NAT?
> >
> > In my case Aastra phones registration not a problem.
> >
> > But calls drooped every 60 sec ... in the same environment Linksys and
> > Polycom works perfectly.
> >
> > How stable the Aastra phones with FreeSWITCH system.
> >
> > TIA
> >
> > 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
>
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From anthony.minessale at gmail.com  Mon Feb 14 20:02:33 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 14 Feb 2011 11:02:33 -0600
Subject: [Freeswitch-users] ESL: how to filter two uuid's?
In-Reply-To: <AANLkTikmL03v+0o6eqvbci30sXZSH7GXURFwi-ZuXGu3@mail.gmail.com>
References: <AANLkTikjDdsyZFf=zcYcerLz7P_w-1+hvxE_rbZwccTj@mail.gmail.com>
	<AANLkTikmL03v+0o6eqvbci30sXZSH7GXURFwi-ZuXGu3@mail.gmail.com>
Message-ID: <AANLkTikRQXPbrL3yZLJ+FPkF_4=CTU5x-QrpY+iehN-_@mail.gmail.com>

try patch in latest GIT HEAD

On Mon, Feb 14, 2011 at 7:04 AM, Chris Hemmings <chris at cheeky.org> wrote:
> This seemed to work for me in Perl:
>
> $con->sendRecv("event plain ALL");
> $con->sendRecv("filter Unique-ID ".$uuid);
> $con->sendRecv("filter Unique-ID ".$out_uuid);
>
> Cheers,
>
> Chris
>
> On 14 February 2011 11:46, Stephen Wilde <wstephen80 at gmail.com> wrote:
>>
>> If I have 2 uuid (uuid_inbound and uuid_outbound) in an ESL socket
>> outbound application, there is a way to filter events related to this two
>> uuid?
>> If I try to do:
>> connection.filter("unique-id", uuid_inbound);
>> connection.filter("unique-id", uuid_outbound);
>> it seems that the second filter overwrite the first one and my application
>> receives only uuid_outbound related events.
>> I have tried also an:
>> connection.filter("unique-id", uuid_inbound + "|" + uuid_outbound);
>> but in this way no event are received.
>> There is a way to filter 2 specific uuid's or I have to receive all events
>> and implement the filtering in my ESL application?
>> 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
>
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 leonardo.bidinoto at voicetechnology.com.br  Mon Feb 14 20:03:01 2011
From: leonardo.bidinoto at voicetechnology.com.br (Leonardo P. Bidinoto)
Date: Mon, 14 Feb 2011 15:03:01 -0200
Subject: [Freeswitch-users] Problems with mp3 larger than 70MB while
 playing files into the conference in the "perpetual-sound" variable
In-Reply-To: <8B6F3C08-5497-4C84-9D2C-5182762BB4F7@freeswitch.org>
References: <AANLkTikWBYU_suW2i1tg5HHhe4P1U18Hqs5E1UZZ=UaQ@mail.gmail.com>
	<8B6F3C08-5497-4C84-9D2C-5182762BB4F7@freeswitch.org>
Message-ID: <AANLkTi=7YOEm5+fm=ASEVFO1EgJFVYZPOXJT=wyUc8RU@mail.gmail.com>

ok, understood ! never use 32bit OS on 64bit CPU. But i have tried with with
CentOS 64-bit and got the same error, not in the same machine, but in a
machine test with the same configuration as the other that we made it to
reproduce the problem here in our lab.
here is the core file:
http://dl.dropbox.com/u/21117321/core.4128.gz

2011/2/14 Brian West <brian at freeswitch.org>

> Wait for it... NEVER run 32bit OS's on a 64bit CPU please.  You'll get way
> better performance out of things.  Have you tried this?  I don't trust that
> MP3 lib to make the right decisions on data types when pulling this type of
> thing off.
>
> /b
>
> On Feb 14, 2011, at 8:27 AM, Leonardo P. Bidinoto wrote:
>
> Bug url = http://jira.freeswitch.org/browse/FS-3053
>
> Hello guys, me again. Now with another bug that i found.
> I set the "perpetual-sound" variable in the conference.conf.xml to play mp3
> files that my customer chooses.
> but when a mp3 with 71mb or more is played, it crashed the machine when was
> more than one user listening to the mp3.
> i tried a file with 99mb and crash too.
> but when i use a file with 10mb or less, works fine.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Leonardo Pires Bidinoto
Voice Technology
www.voicetechnology.com.br
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From anthony.minessale at gmail.com  Mon Feb 14 20:07:22 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 14 Feb 2011 11:07:22 -0600
Subject: [Freeswitch-users] Possible to communicate with FS using
 event_socket without ESL?
In-Reply-To: <008901cbcc01$517903b0$f46b0b10$@net>
References: <003101cbcb2d$fa0e9e60$ee2bdb20$@net>
	<AANLkTi=5zyKN1Ef+nJspyMrOuw=VE+fV-RcCQ5S148L=@mail.gmail.com>
	<008901cbcc01$517903b0$f46b0b10$@net>
Message-ID: <AANLkTikGKFLQ074418Bj_spqkgeAC9UWwngBea4bArYX@mail.gmail.com>

This is why I stopped and took the time to write the ESL library.  If
you want to do it yourself that's fine but were batting 1000 on people
who try to write their own and ask for help or report their own
mistakes as bugs along the way.

I'm not sure why you would want to take the risk to properly implement
the interface we have already spent the time on.


On Sun, Feb 13, 2011 at 10:40 PM, Bert Jones <rmbertjones at comcast.net> wrote:
> Thanks for the clarification.? I too suspect something in my
> communications.? Just wanted to make sure I was not missing something. Will
> move my communication app to a separate machine from the FS server and run
> Wireshark.? Thanks for recommendation.
>
> Bert
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven
> Ayre
> Sent: Sunday, February 13, 2011 7:29 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Possible to communicate with FS using
> event_socket without ESL?
>
>
>
> Without the use of the ESL *protocol* - no. Without the use of the supplied
> ESL *libraries* - correct.
>
> You're along the right lines on implementing your own protocol. If you can
> connect and authenticate via telnet from the same computer you're trying
> your app from then there should be no problem with FS.
>
> The fact you say it takes 10s for FS to respond indicates you're not sending
> the auth line. You should get either a success or failure message straight
> away in reply to that. So it sounds like the auth isn't getting sent at all.
>
> Try using Wireshark to view what's going over the network to check it's what
> you expect.
>
> -Steve
>
>
>
>
>
> On 13 February 2011 03:27, Bert Jones <rmbertjones at comcast.net> wrote:
>
> Hello,
>
>
>
> Am I correct in assuming that an app can talk directly to FS using
> event_socket without the use of the ESL?? I am attempting to do this, but
> having trouble authenticating. I wanted to make sure I was not overlooking
> something basis in my approach.
>
>
>
> I have successfully loaded FS on a windows XP server and can register phones
> and make calls.? Further I can talk to the server, authenticate and make
> calls using telnet, but when attempting to do the same via an app written in
> .net and using a socket, I am unable to authenticate.
>
>
>
> Running my app using a tcpSocket, I receive the "Content-Type: auth/request"
> upon connection of the socket, but when sending ?auth password\n\n? it
> appears that the connection times out after about 10 seconds and I receive:
> ??Content-Type: text/disconnect-notice" & vbLf & "Content-Length: 67".
> ?mod_event_socket is enabled in modules.conf.xml .
>
>
>
> Is there something obvious that I am missing conceptually, or should this
> work?
>
>
>
> 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 steveayre at gmail.com  Mon Feb 14 20:26:46 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 14 Feb 2011 17:26:46 +0000
Subject: [Freeswitch-users] Freeswitch design!
In-Reply-To: <COL115-W3548A7784C17257AEB46E883D00@phx.gbl>
References: <COL115-W3548A7784C17257AEB46E883D00@phx.gbl>
Message-ID: <AANLkTiknA-PkDJd3Hc-CNy3XOpobSd8rYHB7VV-6VT_A@mail.gmail.com>

All the best information is on the Wiki and in the source.

Wiki: http://wiki.freeswitch.org/wiki
Source: git://git.freeswitch.org/freeswitch.git

Media is processed using the RTP protocol using FS's own stack.

By control protocols I assume you mean the signalling? There are several
supported: http://wiki.freeswitch.org/wiki/Specsheet#Protocols
Each protocol is implemented in an endpoint module which wraps around a FOSS
library that implements the protocol, e.g. SIP is implemented in mod_sofia
using the Nokia Sofia-SIP stack.

There's also a management interface implemented using the custom ESL
protocol. That's all documented on the wiki and there's a client using it
and implementations of it in a range of languages in the source so you can
write your own (libs/esl).

The only high-level design overview I know of is this one on the Wiki:
http://wiki.freeswitch.org/wiki/Core_Outline_of_FreeSWITCH

Most of it is written in C (there's a little C++ there too). You can extend
it yourself by writing your own C modules or using the runtime support for
languages such as Lua, .NET and Javascript.

-Steve


On 14 February 2011 03:38, Chris A <cowboys550 at hotmail.com> wrote:

>  Hi !
> Can i know the design overview of the "freeswitch" ?
> what is inside it ?
> how the media is precessed ?
> what is the controlled protocol used?
>
> thx u *
> *
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 14 20:30:26 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 14 Feb 2011 11:30:26 -0600
Subject: [Freeswitch-users] mod_dingaling & inbound audio
In-Reply-To: <1297604960568-6020844.post@n2.nabble.com>
References: <AANLkTim-5LDhfQhHk_2LNLffvTa1JyBXq9U4B3LSY3Nq@mail.gmail.com>
	<AANLkTik19FmdGVEukmC4rS3ytu9qhR=wfSkO-fhaRMJQ@mail.gmail.com>
	<AANLkTikoJcrERbEHbdkLM_4ocnCUKkYDoKE-miYbvTxj@mail.gmail.com>
	<1977194672-1297495611-cardhu_decombobulator_blackberry.rim.net-1261789055-@b25.c2.bise3.blackberry>
	<AANLkTi=UUVuXaoTyPe_xSJd06DYzXi9anHgjc6nEm9Ok@mail.gmail.com>
	<AANLkTimcsPbwT7thhopGCoiX56Z-VFjuhbnksz7ms4Mc@mail.gmail.com>
	<1297604960568-6020844.post@n2.nabble.com>
Message-ID: <AANLkTi=bBCp+JQrivaXDcX21MemWo_Bd-2qy6879J1Gy@mail.gmail.com>

I will test this on one of my linux boxen when i get a chance later today. I
will keep you posted.
-MC


On Sun, Feb 13, 2011 at 7:49 AM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
>
> Chris Chen-4 wrote:
> > Hi Michael, this is a known issue with mod_dingaling since mid January
> > around the time of 1.0.7 tarbar release this year. I tested with
> different
> > kinds of setups, the mod_dingaling always loops at mod_dingaling.c line
> > 2941and 3275, which results in no audio (rtp packets) even though the
> > jingle
> > session seems to have already established.
> I can concur with you on this issue.
>
> |
>
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From msc at freeswitch.org  Mon Feb 14 20:32:44 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 14 Feb 2011 11:32:44 -0600
Subject: [Freeswitch-users] Freeswitch design!
In-Reply-To: <AANLkTiknA-PkDJd3Hc-CNy3XOpobSd8rYHB7VV-6VT_A@mail.gmail.com>
References: <COL115-W3548A7784C17257AEB46E883D00@phx.gbl>
	<AANLkTiknA-PkDJd3Hc-CNy3XOpobSd8rYHB7VV-6VT_A@mail.gmail.com>
Message-ID: <AANLkTimzGfVXPUE-V9C324PNGHHy8KPgGRO6YJmSmdp8@mail.gmail.com>

Also,
Don't forget that chapter 1 and appendix B of the FreeSWITCH book have a lot
of excellent information.
-MC

https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-telephony-systems/book


On Mon, Feb 14, 2011 at 11:26 AM, Steven Ayre <steveayre at gmail.com> wrote:

> All the best information is on the Wiki and in the source.
>
> Wiki: http://wiki.freeswitch.org/wiki
> Source: git://git.freeswitch.org/freeswitch.git
>
> Media is processed using the RTP protocol using FS's own stack.
>
> By control protocols I assume you mean the signalling? There are several
> supported: http://wiki.freeswitch.org/wiki/Specsheet#Protocols
> Each protocol is implemented in an endpoint module which wraps around a
> FOSS library that implements the protocol, e.g. SIP is implemented in
> mod_sofia using the Nokia Sofia-SIP stack.
>
> There's also a management interface implemented using the custom ESL
> protocol. That's all documented on the wiki and there's a client using it
> and implementations of it in a range of languages in the source so you can
> write your own (libs/esl).
>
> The only high-level design overview I know of is this one on the Wiki:
> http://wiki.freeswitch.org/wiki/Core_Outline_of_FreeSWITCH
>
> Most of it is written in C (there's a little C++ there too). You can extend
> it yourself by writing your own C modules or using the runtime support for
> languages such as Lua, .NET and Javascript.
>
> -Steve
>
>
> On 14 February 2011 03:38, Chris A <cowboys550 at hotmail.com> wrote:
>
>>  Hi !
>> Can i know the design overview of the "freeswitch" ?
>> what is inside it ?
>> how the media is precessed ?
>> what is the controlled protocol used?
>>
>> thx u *
>> *
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 14 20:59:50 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 14 Feb 2011 11:59:50 -0600
Subject: [Freeswitch-users] mod_dingaling & inbound audio
In-Reply-To: <AANLkTimcsPbwT7thhopGCoiX56Z-VFjuhbnksz7ms4Mc@mail.gmail.com>
References: <AANLkTim-5LDhfQhHk_2LNLffvTa1JyBXq9U4B3LSY3Nq@mail.gmail.com>
	<AANLkTik19FmdGVEukmC4rS3ytu9qhR=wfSkO-fhaRMJQ@mail.gmail.com>
	<AANLkTikoJcrERbEHbdkLM_4ocnCUKkYDoKE-miYbvTxj@mail.gmail.com>
	<1977194672-1297495611-cardhu_decombobulator_blackberry.rim.net-1261789055-@b25.c2.bise3.blackberry>
	<AANLkTi=UUVuXaoTyPe_xSJd06DYzXi9anHgjc6nEm9Ok@mail.gmail.com>
	<AANLkTimcsPbwT7thhopGCoiX56Z-VFjuhbnksz7ms4Mc@mail.gmail.com>
Message-ID: <AANLkTikqtNE2AZEkPdD=cmf+_GV=Swr9yROTERytP+rC@mail.gmail.com>

Have you considered using JIRA?
Imagine if you get 200 emails and 1200 im messages a day, if you can
keep track of it in your head?


On Sat, Feb 12, 2011 at 7:16 PM, Chris Chen <chris.chen2004 at gmail.com> wrote:
> Hi Michael, this is a known issue with mod_dingaling since mid January
> around the time of 1.0.7 tarbar release this year. I tested with different
> kinds of setups, the mod_dingaling always loops at mod_dingaling.c line
> 2941and 3275, which results in no audio (rtp packets) even though the jingle
> session seems to have already established.
> I asked for core dev's review of mod_dingaling on IRC a couple of times,
> looks like everybody is so busy that no one really takes a closer looks at
> the mod_dingaling.c yet.
> This behavior affects Gtalk to sip calling ?as well as gmail outbound call
> which is sip to jingle.
> Hope FS core dev will have some time to review this, we really appreciate
> that FS's gtalk calling can be back running again.
> Thanks,
> Chris
>
>
> On Sat, Feb 12, 2011 at 5:49 PM, Michael Collins <msc at freeswitch.org> wrote:
>>
>>
>> On Sat, Feb 12, 2011 at 1:27 AM, Budi wibowo <bwibowo at gmail.com> wrote:
>>>
>>> On previous git version I have working mod dingaling, but until today my
>>> dingaling is never working.
>>> If anybody has freeswitch source with dingaling works, please share the
>>> url. I will download
>>>
>> What does the console log show? Also, turn on dingaling debugging w/ the
>> "dl_debug on" command.
>> -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
>
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 cliff at develix.com  Mon Feb 14 21:27:53 2011
From: cliff at develix.com (Cliff Wells)
Date: Mon, 14 Feb 2011 10:27:53 -0800
Subject: [Freeswitch-users] Hacker Attack?
In-Reply-To: <AANLkTimcrJmG=CVTKMy-_4kZYA6hCU-_aORbbdRkO_P5@mail.gmail.com>
References: <AANLkTimcrJmG=CVTKMy-_4kZYA6hCU-_aORbbdRkO_P5@mail.gmail.com>
Message-ID: <1297708073.2842.28.camel@portable-evil>

On Sat, 2011-01-29 at 15:39 -0800, Joao Leme wrote:
> I just downloaded and compiled the latest Git and a little after
> starting freeswitch I'm getting non stop the following:
> 
> 
> [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> profile ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236

> it's non-stop and doesn't let me do nothing else. After the first time
> I went on to vars and changed the 1234 password....restarted and same
> thing happened, I also try denying the ip on acl.conf (not sure if has
> something to do with it but gave it a try):

It seems obvious, but since no one else mentioned it, I will: have you
tried contacting the owner of attacking machine?   Chances are it's a
compromised machine and the owner is unaware of the situation.   I
experienced the same thing about a week ago, did a WHOIS lookup on the
IP, contacted the owner who was quite grateful to be alerted of the
issue, and had it resolved pretty quickly.

As an aside, if the script that's attacking you is the same one that was
attacking me, it won't stop once you setup the iptables rule.  In my
case it continued to send over 1Mbit/s of registration attempts even
when it could no longer connect (as measured by pfSense).

Regards,
Cliff




From testa at voicetechnology.com.br  Mon Feb 14 22:07:17 2011
From: testa at voicetechnology.com.br (Fernando Testa)
Date: Mon, 14 Feb 2011 17:07:17 -0200
Subject: [Freeswitch-users] Centos version
Message-ID: <AANLkTimbRLjOqR-9CyDC5BfSoyjZFwc1ATowi7=Oh_nU@mail.gmail.com>

Hi all,
What CentOS version is recommended? 5.3, 5.4 or 5.5?
Thanks!
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From msc at freeswitch.org  Mon Feb 14 22:47:43 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 14 Feb 2011 13:47:43 -0600
Subject: [Freeswitch-users] Centos version
In-Reply-To: <AANLkTimbRLjOqR-9CyDC5BfSoyjZFwc1ATowi7=Oh_nU@mail.gmail.com>
References: <AANLkTimbRLjOqR-9CyDC5BfSoyjZFwc1ATowi7=Oh_nU@mail.gmail.com>
Message-ID: <AANLkTikDJMGFUFta7Xp3EbHxj_AGWqAm3023eY=UMddh@mail.gmail.com>

At this point we're comfortable with 5.3 and 5.4 of CentOS. Many community
members are reporting success with 5.5 as well, but the devs are more comfy
with 5.3/5.4 right now.
-MC

On Mon, Feb 14, 2011 at 1:07 PM, Fernando Testa <
testa at voicetechnology.com.br> wrote:

> Hi all,
> What CentOS version is recommended? 5.3, 5.4 or 5.5?
> Thanks!
>
>
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From anthony.minessale at gmail.com  Mon Feb 14 23:24:10 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 14 Feb 2011 14:24:10 -0600
Subject: [Freeswitch-users] Question about mod_dingaling.
In-Reply-To: <AANLkTi=Bf89Pae2Qt1Cg1gFhX-xjFHuGz0kv5Dyt9=FK@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>
Message-ID: <AANLkTi=BOA1fTfUberjRLzLu-Uts1xvp7Xr4-CzE1xrD@mail.gmail.com>

update again, I think I found the problem in the code.


2011/2/13 ??? <chenzhanping at gmail.com>:
> Thank u very much,
>
> My server is have public ip,and router not have upnp.
>
> I modified client.xml   :
>
> <!-- <param name="ext-rtp-ip" value="auto-nat"/> -->   to   <param
> name="ext-rtp-ip" value="My ip"/>
>
> Now my Freeswitch + mod_dingaling is working.
>
> But there is no voice when i call out from google voice and connect called
> Successed.
>
> I try to called my other google phone use freeswitch + mod_dingaling ,
> and into voicemail, the voicemail system can heared my voice,and record
> good.
>
> This mean: caller is no voice,but called is have voice good.
>
> there is client.xml:
>
> http://pastebin.freeswitch.org/15371
>
> there is cli log:
>
> http://pastebin.freeswitch.org/15372
>
> freeswitch version:FreeSWITCH Version 1.0.head (git-a2c0da5 2011-02-11
> 23-10-12 -0600)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb 14 23:24:54 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 14 Feb 2011 14:24:54 -0600
Subject: [Freeswitch-users] mod_dingaling & inbound audio
In-Reply-To: <AANLkTikqtNE2AZEkPdD=cmf+_GV=Swr9yROTERytP+rC@mail.gmail.com>
References: <AANLkTim-5LDhfQhHk_2LNLffvTa1JyBXq9U4B3LSY3Nq@mail.gmail.com>
	<AANLkTik19FmdGVEukmC4rS3ytu9qhR=wfSkO-fhaRMJQ@mail.gmail.com>
	<AANLkTikoJcrERbEHbdkLM_4ocnCUKkYDoKE-miYbvTxj@mail.gmail.com>
	<1977194672-1297495611-cardhu_decombobulator_blackberry.rim.net-1261789055-@b25.c2.bise3.blackberry>
	<AANLkTi=UUVuXaoTyPe_xSJd06DYzXi9anHgjc6nEm9Ok@mail.gmail.com>
	<AANLkTimcsPbwT7thhopGCoiX56Z-VFjuhbnksz7ms4Mc@mail.gmail.com>
	<AANLkTikqtNE2AZEkPdD=cmf+_GV=Swr9yROTERytP+rC@mail.gmail.com>
Message-ID: <AANLkTinSHvXi6sj9S4AO_zeqq+so=5P4LytB48q+H-f8@mail.gmail.com>

Oh, well I found it i think, again I would have preferred a JIRA ticket.

it's hard to juggle issues from people over email.


On Mon, Feb 14, 2011 at 11:59 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Have you considered using JIRA?
> Imagine if you get 200 emails and 1200 im messages a day, if you can
> keep track of it in your head?
>
>
> On Sat, Feb 12, 2011 at 7:16 PM, Chris Chen <chris.chen2004 at gmail.com> wrote:
>> Hi Michael, this is a known issue with mod_dingaling since mid January
>> around the time of 1.0.7 tarbar release this year. I tested with different
>> kinds of setups, the mod_dingaling always loops at mod_dingaling.c line
>> 2941and 3275, which results in no audio (rtp packets) even though the jingle
>> session seems to have already established.
>> I asked for core dev's review of mod_dingaling on IRC a couple of times,
>> looks like everybody is so busy that no one really takes a closer looks at
>> the mod_dingaling.c yet.
>> This behavior affects Gtalk to sip calling ?as well as gmail outbound call
>> which is sip to jingle.
>> Hope FS core dev will have some time to review this, we really appreciate
>> that FS's gtalk calling can be back running again.
>> Thanks,
>> Chris
>>
>>
>> On Sat, Feb 12, 2011 at 5:49 PM, Michael Collins <msc at freeswitch.org> wrote:
>>>
>>>
>>> On Sat, Feb 12, 2011 at 1:27 AM, Budi wibowo <bwibowo at gmail.com> wrote:
>>>>
>>>> On previous git version I have working mod dingaling, but until today my
>>>> dingaling is never working.
>>>> If anybody has freeswitch source with dingaling works, please share the
>>>> url. I will download
>>>>
>>> What does the console log show? Also, turn on dingaling debugging w/ the
>>> "dl_debug on" command.
>>> -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
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 joaocarlosleme at gmail.com  Mon Feb 14 23:25:00 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Mon, 14 Feb 2011 12:25:00 -0800
Subject: [Freeswitch-users] Hacker Attack?
In-Reply-To: <1297708073.2842.28.camel@portable-evil>
References: <AANLkTimcrJmG=CVTKMy-_4kZYA6hCU-_aORbbdRkO_P5@mail.gmail.com>
	<1297708073.2842.28.camel@portable-evil>
Message-ID: <AANLkTi=CjKmPvwv8hfVO5JPK=jB_eXyCKxWff6YD5DpL@mail.gmail.com>

Thanks Cliff, but I haven't had this problem for almost 2 weeks now (since I
updated FS (again) to the latest GIT, even though FS version seems to have
nothing to do with it). I wanted to remove DMZ but I can't get around the NO
SOUND problem when connecting from outside the local network (I opened
another thread for that but no solution so far from my end).
Thanks,
John

On Mon, Feb 14, 2011 at 10:27 AM, Cliff Wells <cliff at develix.com> wrote:

> On Sat, 2011-01-29 at 15:39 -0800, Joao Leme wrote:
> > I just downloaded and compiled the latest Git and a little after
> > starting freeswitch I'm getting non stop the following:
> >
> >
> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> > profile ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>
> > it's non-stop and doesn't let me do nothing else. After the first time
> > I went on to vars and changed the 1234 password....restarted and same
> > thing happened, I also try denying the ip on acl.conf (not sure if has
> > something to do with it but gave it a try):
>
> It seems obvious, but since no one else mentioned it, I will: have you
> tried contacting the owner of attacking machine?   Chances are it's a
> compromised machine and the owner is unaware of the situation.   I
> experienced the same thing about a week ago, did a WHOIS lookup on the
> IP, contacted the owner who was quite grateful to be alerted of the
> issue, and had it resolved pretty quickly.
>
> As an aside, if the script that's attacking you is the same one that was
> attacking me, it won't stop once you setup the iptables rule.  In my
> case it continued to send over 1Mbit/s of registration attempts even
> when it could no longer connect (as measured by pfSense).
>
> 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 Nabble at slickdeals.endjunk.com  Mon Feb 14 23:43:37 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 14 Feb 2011 12:43:37 -0800 (PST)
Subject: [Freeswitch-users] mod_dingaling & inbound audio
In-Reply-To: <AANLkTinSHvXi6sj9S4AO_zeqq+so=5P4LytB48q+H-f8@mail.gmail.com>
References: <AANLkTim-5LDhfQhHk_2LNLffvTa1JyBXq9U4B3LSY3Nq@mail.gmail.com>
	<AANLkTik19FmdGVEukmC4rS3ytu9qhR=wfSkO-fhaRMJQ@mail.gmail.com>
	<AANLkTikoJcrERbEHbdkLM_4ocnCUKkYDoKE-miYbvTxj@mail.gmail.com>
	<1977194672-1297495611-cardhu_decombobulator_blackberry.rim.net-1261789055-@b25.c2.bise3.blackberry>
	<AANLkTi=UUVuXaoTyPe_xSJd06DYzXi9anHgjc6nEm9Ok@mail.gmail.com>
	<AANLkTimcsPbwT7thhopGCoiX56Z-VFjuhbnksz7ms4Mc@mail.gmail.com>
	<AANLkTikqtNE2AZEkPdD=cmf+_GV=Swr9yROTERytP+rC@mail.gmail.com>
	<AANLkTinSHvXi6sj9S4AO_zeqq+so=5P4LytB48q+H-f8@mail.gmail.com>
Message-ID: <1297716217329-6025112.post@n2.nabble.com>



Anthony Minessale wrote:
> 
> Oh, well I found it i think, ...
By disabling VAD?

-----
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/mod-dingaling-inbound-audio-tp6005151p6025112.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Tue Feb 15 00:22:56 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 14 Feb 2011 15:22:56 -0600
Subject: [Freeswitch-users] mod_dingaling & inbound audio
In-Reply-To: <1297716217329-6025112.post@n2.nabble.com>
References: <AANLkTim-5LDhfQhHk_2LNLffvTa1JyBXq9U4B3LSY3Nq@mail.gmail.com>
	<AANLkTik19FmdGVEukmC4rS3ytu9qhR=wfSkO-fhaRMJQ@mail.gmail.com>
	<AANLkTikoJcrERbEHbdkLM_4ocnCUKkYDoKE-miYbvTxj@mail.gmail.com>
	<1977194672-1297495611-cardhu_decombobulator_blackberry.rim.net-1261789055-@b25.c2.bise3.blackberry>
	<AANLkTi=UUVuXaoTyPe_xSJd06DYzXi9anHgjc6nEm9Ok@mail.gmail.com>
	<AANLkTimcsPbwT7thhopGCoiX56Z-VFjuhbnksz7ms4Mc@mail.gmail.com>
	<AANLkTikqtNE2AZEkPdD=cmf+_GV=Swr9yROTERytP+rC@mail.gmail.com>
	<AANLkTinSHvXi6sj9S4AO_zeqq+so=5P4LytB48q+H-f8@mail.gmail.com>
	<1297716217329-6025112.post@n2.nabble.com>
Message-ID: <AANLkTikYfG9d=JXqWeG7nYF-miH-FmvpJVgS8X-DEJWF@mail.gmail.com>

no

On Mon, Feb 14, 2011 at 2:43 PM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
>
> Anthony Minessale wrote:
>>
>> Oh, well I found it i think, ...
> By disabling VAD?
>
> -----
> 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/mod-dingaling-inbound-audio-tp6005151p6025112.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 wstephen80 at gmail.com  Tue Feb 15 01:57:13 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Mon, 14 Feb 2011 23:57:13 +0100
Subject: [Freeswitch-users] ESL Outbound: how to bridge media?
Message-ID: <AANLkTikLVLpSGkrRqeD3WVFZZgSaLnQvypCOG39hEhLX@mail.gmail.com>

The problem I have is to propagate the audio coming from an "originated"
outbound session to the inbound session when the outbound is in the PROGRESS
MEDIA phase.

When my application receives the "CHANNEL_PROGRESS_MEDIA" event from
outbound session I can do a "pre_answer" on inbound session but I'm not
capable to do an audio bridge.

I have tried with "uuid_bridge <outbound_uuid> <inbound_uuid>" with no
result probably because this api requires that at least one session must be
answered.

I don't want to answer to the inbound session to propagate the outbound
progressing media but I want to answer to inbound only on outbound answer.

Any way to do that?

Stephen
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From msc at freeswitch.org  Tue Feb 15 02:19:52 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 14 Feb 2011 17:19:52 -0600
Subject: [Freeswitch-users] ESL Outbound: how to bridge media?
In-Reply-To: <AANLkTikLVLpSGkrRqeD3WVFZZgSaLnQvypCOG39hEhLX@mail.gmail.com>
References: <AANLkTikLVLpSGkrRqeD3WVFZZgSaLnQvypCOG39hEhLX@mail.gmail.com>
Message-ID: <AANLkTimwX8Uz4t9gp6Ppzfu+WE+UJa3FO4qHE7XKFOBz@mail.gmail.com>

My head is spinning after reading this email. :)
It sounds like you just need a simple bridge from the incoming leg to the
outgoing leg. Can you pre_answer the A leg then execute a good old-fashioned
bridge to the b-leg?

-MC

On Mon, Feb 14, 2011 at 4:57 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> The problem I have is to propagate the audio coming from an "originated"
> outbound session to the inbound session when the outbound is in the PROGRESS
> MEDIA phase.
>
> When my application receives the "CHANNEL_PROGRESS_MEDIA" event from
> outbound session I can do a "pre_answer" on inbound session but I'm not
> capable to do an audio bridge.
>
> I have tried with "uuid_bridge <outbound_uuid> <inbound_uuid>" with no
> result probably because this api requires that at least one session must be
> answered.
>
> I don't want to answer to the inbound session to propagate the outbound
> progressing media but I want to answer to inbound only on outbound answer.
>
> Any way to do that?
>
> 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  Tue Feb 15 02:44:50 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 15 Feb 2011 00:44:50 +0100
Subject: [Freeswitch-users] ESL Outbound: how to bridge media?
In-Reply-To: <AANLkTimwX8Uz4t9gp6Ppzfu+WE+UJa3FO4qHE7XKFOBz@mail.gmail.com>
References: <AANLkTikLVLpSGkrRqeD3WVFZZgSaLnQvypCOG39hEhLX@mail.gmail.com>
	<AANLkTimwX8Uz4t9gp6Ppzfu+WE+UJa3FO4qHE7XKFOBz@mail.gmail.com>
Message-ID: <AANLkTi=t-a7U9ov3wJ6Dh-PuniSfZbyeQ==BCAvMkUPN@mail.gmail.com>

Sorry but I have missed something.

I know that I can solve this problem directly in dialplan with a bridge but
what I'm doing is an "Freeswitch ESL learning" because I have to port some
application in Freeswitch and I'm learning how to implement some
functionality.

For me it's important to take control of both inbound/outbound in full async
way and I have the necessity to do the complete call control.

I'm not sure but to me it seems that with a normal bridge I lose the control
of two sessions, for example, an outbound answer is propagated by bridge
application as inbound answer.

What I want to do is an audio bridging so my application can take control of
"signaling bridging".

I'm wrong? There are other way to do that?

Stephen

On Tue, Feb 15, 2011 at 12:19 AM, Michael Collins <msc at freeswitch.org>wrote:

> My head is spinning after reading this email. :)
> It sounds like you just need a simple bridge from the incoming leg to the
> outgoing leg. Can you pre_answer the A leg then execute a good old-fashioned
> bridge to the b-leg?
>
> -MC
>
> On Mon, Feb 14, 2011 at 4:57 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> The problem I have is to propagate the audio coming from an "originated"
>> outbound session to the inbound session when the outbound is in the PROGRESS
>> MEDIA phase.
>>
>> When my application receives the "CHANNEL_PROGRESS_MEDIA" event from
>> outbound session I can do a "pre_answer" on inbound session but I'm not
>> capable to do an audio bridge.
>>
>> I have tried with "uuid_bridge <outbound_uuid> <inbound_uuid>" with no
>> result probably because this api requires that at least one session must be
>> answered.
>>
>> I don't want to answer to the inbound session to propagate the outbound
>> progressing media but I want to answer to inbound only on outbound answer.
>>
>> Any way to do that?
>>
>> 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 fs-list at communicatefreely.net  Tue Feb 15 02:53:41 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Mon, 14 Feb 2011 18:53:41 -0500
Subject: [Freeswitch-users] Aastra phone registration lost
In-Reply-To: <AANLkTinJK+e9hZJbk4ur_OQwGV5B7tRLrpPMH=TU5MOo@mail.gmail.com>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>	<4D52B987.7050002@communicatefreely.net>	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>	<4D534AFD.3020401@communicatefreely.net>	<AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>	<60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>	<AANLkTimux4MBM1K_7itE4GnZQHjy_vR2Y6sr_z+u3gkR@mail.gmail.com>	<4D5577CE.1000603@communicatefreely.net>
	<AANLkTinJK+e9hZJbk4ur_OQwGV5B7tRLrpPMH=TU5MOo@mail.gmail.com>
Message-ID: <4D59C085.9040008@communicatefreely.net>

<domains>
    <domain name="all" alias="true" parse="true"/>
    <domain name="pbx.MYDOMAIN.NET" alias="true" parse="false"/>
  </domains>

<settings>
    <param name="user-agent-string" value="Communicate Freely 2.0"/>
    <param name="debug" value="0"/>
    <param name="sip-trace" value="no"/>
    <param name="log-auth-failures" value="true"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="sip-port" value="$${internal_sip_port}"/>
    <param name="dialplan" value="xml"/>
    <param name="context" value="internal"/>
    <param name="dtmf-duration" value="100"/>
    <param name="inbound-codec-prefs" value="$${internal_codec_prefs}"/>
    <param name="outbound-codec-prefs" value="$${internal_codec_prefs}"/>
    <param name="rtp-timer-name" value="soft"/>
    <param name="sip-ip" value="$${public_ip}"/>
    <param name="rtp-ip" value="$${public_ip}"/>
    <param name="hold-music" value="$${moh_prefix}alt"/>
    <param name="dtmf-type" value="rfc2833"/>

    <param name="force-register-domain" value="$${domain}"/>
    <param name="force-subscription-domain" value="pbx.$${domain}"/>
    <param name="force-register-db-domain" value="$${domain}"/>
    <param name="force-subscription-expires" value="600"/>

These are the most important ones I think.

    <param name="NDLB-received-in-nat-reg-contact" value="true"/>
    <param name="sip-force-contact" value=" NDLB-connectile-dysfunction"/>

I'm also using sip-force-expires to 600 at the moment, and ping = 10.  I 
will probably increase those eventually to reduce bandwidth.  I'm still 
in Beta right now, but I'm not having too many issues.

Some of those params are added per-device using the directory, so I can 
tweak them depending on which device registers, and what the NAT status 
is of that device.  I'm really pushing to get IPv6 on the phones, as 
well as on some of the more prominent (but competitive) DSL providers 
here so that we can forego NAT altogether some day.

Hope that's helpful.  I haven't really gone through and figured out 
which variables do what at the moment, but it seems to work as it is.

-Tim
Aloysius Lloyd wrote:
> Tim,
>
> Thank you for the information.
>
> I have around 275 Aastra 9133i models phones in production .These 
> phones installed 31/2 years ago. I am trying to migrate to FreeSWITCH 
> could not make it work reliably.
>
> What are the profile settings turned on for these phones works reliably?
>
>
> Thanks 
> Lloyd
>
>
>
> On Fri, Feb 11, 2011 at 12:54 PM, Tim St. Pierre 
> <fs-list at communicatefreely.net <mailto:fs-list at communicatefreely.net>> 
> wrote:
>
>     Sure,
>
>     I normally administer about 300 Aastra phones, with every model they
>     make represented.
>
>     I have 22 connected to our Freeswitch "beta" system, which will
>     eventually become production.
>
>     All the endpoints are behind NAT without exception.  There are a
>     number
>     of legacy 9133i and 480i phones on the network that don't have the
>     newer
>     NAT traversal features available, but this doesn't seem to be a
>     problem.  I have some of the nat traversal options turned on in the
>     sofia profile though, so fs will send media back to the originating
>     address and port.
>
>     They have been quite reliable, and the sound quality has been
>     excellent,
>     with the newer phones using g722 at 16KHz.
>
>     There are a few advanced features that I haven't had a chance to play
>     with yet, but here's what I have working:
>
>     Regular calls, in and out.
>     Intercom calls (auto-answer to speaker phone)
>     Automatic update of destination name and number (updates when checking
>     voice mail, and when calling an extension).  Only on newer phones
>     Blind and attended transfer
>     Music on hold
>     SIP using udp or tcp (haven't tried TLS yet)
>     Fewer issues with DTMF than with asterisk, using rfc2833 dtmf (no
>     issues
>     as of yet).
>     BLF lamps work correctly, flashing when the phone rings, lit
>     steady when
>     they are on the phone.
>     Distinctive ringing works.
>     I haven't tried SLA yet, but Aastra recently released a firmware
>     update
>     that fixes a missing header, reported to have broken correct SLA
>     operation.  I'm hoping to test that in the next week or two.
>
>     The phones provision very nicely - we auto generate config using PHP
>     scripts that generate a config file on the fly from the user database.
>     These are very easy phones to deploy in large installations, or to the
>     outside world (not readily accessible).  They have just added some new
>     features that allow for remote diagnostics of the phones as well.
>
>     There is a great deal of XML programmability in the phones too, which
>     I'm starting to use for call control and other useful things (updating
>     forwarding rules in the database, or conference and recording control
>     using ESL).
>
>     Hope that helps!
>
>     -Tim
>
>     Aloysius Lloyd wrote:
>     > Tim,
>     >
>     > Can you share your success stories FreeSWITCH and Aastra.
>     >
>     > Aastra Phones Behind the NAT?
>     >
>     > In my case Aastra phones registration not a problem.
>     >
>     > But calls drooped every 60 sec ... in the same environment
>     Linksys and
>     > Polycom works perfectly.
>     >
>     > How stable the Aastra phones with FreeSWITCH system.
>     >
>     > TIA
>     >
>     > Lloyd
>
>
>     _______________________________________________
>     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 anthony.minessale at gmail.com  Tue Feb 15 02:53:17 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 14 Feb 2011 17:53:17 -0600
Subject: [Freeswitch-users] ESL Outbound: how to bridge media?
In-Reply-To: <AANLkTi=t-a7U9ov3wJ6Dh-PuniSfZbyeQ==BCAvMkUPN@mail.gmail.com>
References: <AANLkTikLVLpSGkrRqeD3WVFZZgSaLnQvypCOG39hEhLX@mail.gmail.com>
	<AANLkTimwX8Uz4t9gp6Ppzfu+WE+UJa3FO4qHE7XKFOBz@mail.gmail.com>
	<AANLkTi=t-a7U9ov3wJ6Dh-PuniSfZbyeQ==BCAvMkUPN@mail.gmail.com>
Message-ID: <AANLkTik27F7JjyMRLKryQZq_N+oVDsDzdmiPPa+d2KWN@mail.gmail.com>

you need to make sure originate has ended on the outbound leg before
you use it in a bridge etc.
you also need to supply the inbound leg first in uuid_bridge if that
is something you want to do.

Easier would be to originate the B leg to park inline and tell A leg
to execute intercept on the B leg uuid.



On Mon, Feb 14, 2011 at 5:44 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:
> Sorry but I have missed something.
> I know that I can solve this problem directly in dialplan with a bridge but
> what I'm doing is an "Freeswitch ESL learning"?because I have to port some
> application in Freeswitch and I'm learning how to implement some
> functionality.
> For me it's important to take control of both inbound/outbound in full async
> way and I have the necessity to do the complete call control.
> I'm not sure but to me it seems that with a normal bridge I lose the control
> of two sessions, for example, an outbound answer is propagated by bridge
> application as inbound answer.
> What I want to do is an audio bridging so my application can take control of
> "signaling bridging".
>
> I'm wrong? There are other way to do that?
> Stephen
> On Tue, Feb 15, 2011 at 12:19 AM, Michael Collins <msc at freeswitch.org>
> wrote:
>>
>> My head is spinning after reading this email. :)
>> It sounds like you just need a simple bridge from the incoming leg to the
>> outgoing leg. Can you pre_answer the A leg then execute a good old-fashioned
>> bridge to the b-leg?
>> -MC
>> On Mon, Feb 14, 2011 at 4:57 PM, Stephen Wilde <wstephen80 at gmail.com>
>> wrote:
>>>
>>> The problem I have is to propagate the audio coming from an "originated"
>>> outbound session to the inbound session when the outbound is in the PROGRESS
>>> MEDIA phase.
>>> When my application receives the "CHANNEL_PROGRESS_MEDIA" event from
>>> outbound session I can do a "pre_answer" on inbound session but I'm not
>>> capable to do an audio bridge.
>>> I have tried with "uuid_bridge <outbound_uuid> <inbound_uuid>" with no
>>> result probably because this api requires that at least one session must be
>>> answered.
>>> I don't want to answer to the inbound session to propagate the outbound
>>> progressing media but I want to answer to inbound only on outbound answer.
>>> Any way to do that?
>>> 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
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb 15 03:11:49 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 15 Feb 2011 01:11:49 +0100
Subject: [Freeswitch-users] ESL Outbound: how to bridge media?
In-Reply-To: <AANLkTik27F7JjyMRLKryQZq_N+oVDsDzdmiPPa+d2KWN@mail.gmail.com>
References: <AANLkTikLVLpSGkrRqeD3WVFZZgSaLnQvypCOG39hEhLX@mail.gmail.com>
	<AANLkTimwX8Uz4t9gp6Ppzfu+WE+UJa3FO4qHE7XKFOBz@mail.gmail.com>
	<AANLkTi=t-a7U9ov3wJ6Dh-PuniSfZbyeQ==BCAvMkUPN@mail.gmail.com>
	<AANLkTik27F7JjyMRLKryQZq_N+oVDsDzdmiPPa+d2KWN@mail.gmail.com>
Message-ID: <AANLkTine=jG-dc98pDySashEvo3T5cSPwVXYwp54mhhC@mail.gmail.com>

Ok, now I'm doing bridge when I receive "CHANNEL_ANSWER" or
"CHANNEL_PROGRESS_MEDIA" on outbound leg but you say that I have wait that
"originate has ended": you mean that I have to wait for "BACKGROUND_JOB"
event related to my "bgapi originate ... &park"?

I'm already doing "uuid_bridge <inbound_uuid> <outbound_uuid>".

I'll try also with intercept and inline originate. Thank you!

Stephen


On Tue, Feb 15, 2011 at 12:53 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> you need to make sure originate has ended on the outbound leg before
> you use it in a bridge etc.
> you also need to supply the inbound leg first in uuid_bridge if that
> is something you want to do.
>
> Easier would be to originate the B leg to park inline and tell A leg
> to execute intercept on the B leg uuid.
>
>
>
> On Mon, Feb 14, 2011 at 5:44 PM, Stephen Wilde <wstephen80 at gmail.com>
> wrote:
> > Sorry but I have missed something.
> > I know that I can solve this problem directly in dialplan with a bridge
> but
> > what I'm doing is an "Freeswitch ESL learning" because I have to port
> some
> > application in Freeswitch and I'm learning how to implement some
> > functionality.
> > For me it's important to take control of both inbound/outbound in full
> async
> > way and I have the necessity to do the complete call control.
> > I'm not sure but to me it seems that with a normal bridge I lose the
> control
> > of two sessions, for example, an outbound answer is propagated by bridge
> > application as inbound answer.
> > What I want to do is an audio bridging so my application can take control
> of
> > "signaling bridging".
> >
> > I'm wrong? There are other way to do that?
> > Stephen
> > On Tue, Feb 15, 2011 at 12:19 AM, Michael Collins <msc at freeswitch.org>
> > wrote:
> >>
> >> My head is spinning after reading this email. :)
> >> It sounds like you just need a simple bridge from the incoming leg to
> the
> >> outgoing leg. Can you pre_answer the A leg then execute a good
> old-fashioned
> >> bridge to the b-leg?
> >> -MC
> >> On Mon, Feb 14, 2011 at 4:57 PM, Stephen Wilde <wstephen80 at gmail.com>
> >> wrote:
> >>>
> >>> The problem I have is to propagate the audio coming from an
> "originated"
> >>> outbound session to the inbound session when the outbound is in the
> PROGRESS
> >>> MEDIA phase.
> >>> When my application receives the "CHANNEL_PROGRESS_MEDIA" event from
> >>> outbound session I can do a "pre_answer" on inbound session but I'm not
> >>> capable to do an audio bridge.
> >>> I have tried with "uuid_bridge <outbound_uuid> <inbound_uuid>" with no
> >>> result probably because this api requires that at least one session
> must be
> >>> answered.
> >>> I don't want to answer to the inbound session to propagate the outbound
> >>> progressing media but I want to answer to inbound only on outbound
> answer.
> >>> Any way to do that?
> >>> 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
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 Feb 15 03:14:18 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 14 Feb 2011 18:14:18 -0600
Subject: [Freeswitch-users] ESL Outbound: how to bridge media?
In-Reply-To: <AANLkTine=jG-dc98pDySashEvo3T5cSPwVXYwp54mhhC@mail.gmail.com>
References: <AANLkTikLVLpSGkrRqeD3WVFZZgSaLnQvypCOG39hEhLX@mail.gmail.com>
	<AANLkTimwX8Uz4t9gp6Ppzfu+WE+UJa3FO4qHE7XKFOBz@mail.gmail.com>
	<AANLkTi=t-a7U9ov3wJ6Dh-PuniSfZbyeQ==BCAvMkUPN@mail.gmail.com>
	<AANLkTik27F7JjyMRLKryQZq_N+oVDsDzdmiPPa+d2KWN@mail.gmail.com>
	<AANLkTine=jG-dc98pDySashEvo3T5cSPwVXYwp54mhhC@mail.gmail.com>
Message-ID: <AANLkTi=JAt=RkQ5b=DBajH_YR6TmiqExdjgF1=OL=fJ_@mail.gmail.com>

if you do originate without ignore_early_media=true set it will end
the soonest possible where it's suitable for a bridge.
so that is the best bet to wait for originate to end.


On Mon, Feb 14, 2011 at 6:11 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:
> Ok, now I'm doing bridge when I receive "CHANNEL_ANSWER" or
> "CHANNEL_PROGRESS_MEDIA" on outbound leg but you say that I have wait that
> "originate has ended": you mean that I have to wait for "BACKGROUND_JOB"
> event related to my "bgapi originate ... &park"?
> I'm already doing "uuid_bridge <inbound_uuid> <outbound_uuid>".
> I'll try also with intercept and inline originate.?Thank you!
> Stephen
>
> On Tue, Feb 15, 2011 at 12:53 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> you need to make sure originate has ended on the outbound leg before
>> you use it in a bridge etc.
>> you also need to supply the inbound leg first in uuid_bridge if that
>> is something you want to do.
>>
>> Easier would be to originate the B leg to park inline and tell A leg
>> to execute intercept on the B leg uuid.
>>
>>
>>
>> On Mon, Feb 14, 2011 at 5:44 PM, Stephen Wilde <wstephen80 at gmail.com>
>> wrote:
>> > Sorry but I have missed something.
>> > I know that I can solve this problem directly in dialplan with a bridge
>> > but
>> > what I'm doing is an "Freeswitch ESL learning"?because I have to port
>> > some
>> > application in Freeswitch and I'm learning how to implement some
>> > functionality.
>> > For me it's important to take control of both inbound/outbound in full
>> > async
>> > way and I have the necessity to do the complete call control.
>> > I'm not sure but to me it seems that with a normal bridge I lose the
>> > control
>> > of two sessions, for example, an outbound answer is propagated by bridge
>> > application as inbound answer.
>> > What I want to do is an audio bridging so my application can take
>> > control of
>> > "signaling bridging".
>> >
>> > I'm wrong? There are other way to do that?
>> > Stephen
>> > On Tue, Feb 15, 2011 at 12:19 AM, Michael Collins <msc at freeswitch.org>
>> > wrote:
>> >>
>> >> My head is spinning after reading this email. :)
>> >> It sounds like you just need a simple bridge from the incoming leg to
>> >> the
>> >> outgoing leg. Can you pre_answer the A leg then execute a good
>> >> old-fashioned
>> >> bridge to the b-leg?
>> >> -MC
>> >> On Mon, Feb 14, 2011 at 4:57 PM, Stephen Wilde <wstephen80 at gmail.com>
>> >> wrote:
>> >>>
>> >>> The problem I have is to propagate the audio coming from an
>> >>> "originated"
>> >>> outbound session to the inbound session when the outbound is in the
>> >>> PROGRESS
>> >>> MEDIA phase.
>> >>> When my application receives the "CHANNEL_PROGRESS_MEDIA" event from
>> >>> outbound session I can do a "pre_answer" on inbound session but I'm
>> >>> not
>> >>> capable to do an audio bridge.
>> >>> I have tried with "uuid_bridge <outbound_uuid> <inbound_uuid>" with no
>> >>> result probably because this api requires that at least one session
>> >>> must be
>> >>> answered.
>> >>> I don't want to answer to the inbound session to propagate the
>> >>> outbound
>> >>> progressing media but I want to answer to inbound only on outbound
>> >>> answer.
>> >>> Any way to do that?
>> >>> 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
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at 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 msc at freeswitch.org  Tue Feb 15 03:42:47 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 14 Feb 2011 18:42:47 -0600
Subject: [Freeswitch-users] INFORMATION: FreeSWITCH Training, West Coast
Message-ID: <AANLkTinDFXwz+9Ln3xPvFxQbTfYKDOLn=fU2pE5_1wOD@mail.gmail.com>

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

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

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

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

-Michael
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From frank at telonium.com  Tue Feb 15 04:22:27 2011
From: frank at telonium.com (Frank Park)
Date: Mon, 14 Feb 2011 20:22:27 -0500
Subject: [Freeswitch-users] Simultaneous ring out (external)
Message-ID: <AANLkTin983TbMribnEchrrG-TUMvRuV4hvCaFR-k7sjG@mail.gmail.com>

Hello,

Yet another puzzle I've been trying to resolve.
I have been trying to implement a simultaneous ring group that calls
multiple external phone numbers (ie cellphones). Similar to call group,
first person who answers gets the bridging.
Bridge with commas works well for internal extensions and call groups, but
it seems like it doesn't like multiple external legs.

Should I be using something else besides bridge function?

Thank you,
Frank

-- 

----=======================----
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  Tue Feb 15 06:50:20 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 14 Feb 2011 22:50:20 -0500
Subject: [Freeswitch-users] Aastra phone registration lost
In-Reply-To: <4D59C085.9040008@communicatefreely.net>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>
	<4D52B987.7050002@communicatefreely.net>
	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>
	<4D534AFD.3020401@communicatefreely.net>
	<AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>
	<60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>
	<AANLkTimux4MBM1K_7itE4GnZQHjy_vR2Y6sr_z+u3gkR@mail.gmail.com>
	<4D5577CE.1000603@communicatefreely.net>
	<AANLkTinJK+e9hZJbk4ur_OQwGV5B7tRLrpPMH=TU5MOo@mail.gmail.com>
	<4D59C085.9040008@communicatefreely.net>
Message-ID: <AANLkTikZbWPqf7N1fT-9fF8Fvydi85mJVZ+o8_J=RLjE@mail.gmail.com>

Tim,

Thank you for the settings will give a try.


Thanks
Lloyd

On Mon, Feb 14, 2011 at 6:53 PM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:

> <domains>
>    <domain name="all" alias="true" parse="true"/>
>    <domain name="pbx.MYDOMAIN.NET" alias="true" parse="false"/>
>  </domains>
>
> <settings>
>    <param name="user-agent-string" value="Communicate Freely 2.0"/>
>    <param name="debug" value="0"/>
>    <param name="sip-trace" value="no"/>
>    <param name="log-auth-failures" value="true"/>
>    <param name="rfc2833-pt" value="101"/>
>    <param name="sip-port" value="$${internal_sip_port}"/>
>    <param name="dialplan" value="xml"/>
>    <param name="context" value="internal"/>
>    <param name="dtmf-duration" value="100"/>
>    <param name="inbound-codec-prefs" value="$${internal_codec_prefs}"/>
>    <param name="outbound-codec-prefs" value="$${internal_codec_prefs}"/>
>    <param name="rtp-timer-name" value="soft"/>
>    <param name="sip-ip" value="$${public_ip}"/>
>    <param name="rtp-ip" value="$${public_ip}"/>
>    <param name="hold-music" value="$${moh_prefix}alt"/>
>    <param name="dtmf-type" value="rfc2833"/>
>
>    <param name="force-register-domain" value="$${domain}"/>
>    <param name="force-subscription-domain" value="pbx.$${domain}"/>
>    <param name="force-register-db-domain" value="$${domain}"/>
>    <param name="force-subscription-expires" value="600"/>
>
> These are the most important ones I think.
>
>    <param name="NDLB-received-in-nat-reg-contact" value="true"/>
>    <param name="sip-force-contact" value=" NDLB-connectile-dysfunction"/>
>
> I'm also using sip-force-expires to 600 at the moment, and ping = 10.  I
> will probably increase those eventually to reduce bandwidth.  I'm still
> in Beta right now, but I'm not having too many issues.
>
> Some of those params are added per-device using the directory, so I can
> tweak them depending on which device registers, and what the NAT status
> is of that device.  I'm really pushing to get IPv6 on the phones, as
> well as on some of the more prominent (but competitive) DSL providers
> here so that we can forego NAT altogether some day.
>
> Hope that's helpful.  I haven't really gone through and figured out
> which variables do what at the moment, but it seems to work as it is.
>
> -Tim
> Aloysius Lloyd wrote:
> > Tim,
> >
> > Thank you for the information.
> >
> > I have around 275 Aastra 9133i models phones in production .These
> > phones installed 31/2 years ago. I am trying to migrate to FreeSWITCH
> > could not make it work reliably.
> >
> > What are the profile settings turned on for these phones works reliably?
> >
> >
> > Thanks
> > Lloyd
> >
> >
> >
> > On Fri, Feb 11, 2011 at 12:54 PM, Tim St. Pierre
> > <fs-list at communicatefreely.net <mailto:fs-list at communicatefreely.net>>
> > wrote:
> >
> >     Sure,
> >
> >     I normally administer about 300 Aastra phones, with every model they
> >     make represented.
> >
> >     I have 22 connected to our Freeswitch "beta" system, which will
> >     eventually become production.
> >
> >     All the endpoints are behind NAT without exception.  There are a
> >     number
> >     of legacy 9133i and 480i phones on the network that don't have the
> >     newer
> >     NAT traversal features available, but this doesn't seem to be a
> >     problem.  I have some of the nat traversal options turned on in the
> >     sofia profile though, so fs will send media back to the originating
> >     address and port.
> >
> >     They have been quite reliable, and the sound quality has been
> >     excellent,
> >     with the newer phones using g722 at 16KHz.
> >
> >     There are a few advanced features that I haven't had a chance to play
> >     with yet, but here's what I have working:
> >
> >     Regular calls, in and out.
> >     Intercom calls (auto-answer to speaker phone)
> >     Automatic update of destination name and number (updates when
> checking
> >     voice mail, and when calling an extension).  Only on newer phones
> >     Blind and attended transfer
> >     Music on hold
> >     SIP using udp or tcp (haven't tried TLS yet)
> >     Fewer issues with DTMF than with asterisk, using rfc2833 dtmf (no
> >     issues
> >     as of yet).
> >     BLF lamps work correctly, flashing when the phone rings, lit
> >     steady when
> >     they are on the phone.
> >     Distinctive ringing works.
> >     I haven't tried SLA yet, but Aastra recently released a firmware
> >     update
> >     that fixes a missing header, reported to have broken correct SLA
> >     operation.  I'm hoping to test that in the next week or two.
> >
> >     The phones provision very nicely - we auto generate config using PHP
> >     scripts that generate a config file on the fly from the user
> database.
> >     These are very easy phones to deploy in large installations, or to
> the
> >     outside world (not readily accessible).  They have just added some
> new
> >     features that allow for remote diagnostics of the phones as well.
> >
> >     There is a great deal of XML programmability in the phones too, which
> >     I'm starting to use for call control and other useful things
> (updating
> >     forwarding rules in the database, or conference and recording control
> >     using ESL).
> >
> >     Hope that helps!
> >
> >     -Tim
> >
> >     Aloysius Lloyd wrote:
> >     > Tim,
> >     >
> >     > Can you share your success stories FreeSWITCH and Aastra.
> >     >
> >     > Aastra Phones Behind the NAT?
> >     >
> >     > In my case Aastra phones registration not a problem.
> >     >
> >     > But calls drooped every 60 sec ... in the same environment
> >     Linksys and
> >     > Polycom works perfectly.
> >     >
> >     > How stable the Aastra phones with FreeSWITCH system.
> >     >
> >     > TIA
> >     >
> >     > Lloyd
> >
> >
> >     _______________________________________________
> >     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
> http://www.freeswitch.org
>
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From lloyd.aloysius at gmail.com  Tue Feb 15 07:06:50 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 14 Feb 2011 23:06:50 -0500
Subject: [Freeswitch-users] Getting SIP trace on user level
Message-ID: <AANLkTimhW139WdDKXiQkRm43Ai13f_7Z+N9Z-K2y8Ldi@mail.gmail.com>

Hi All,

Is there any console command available to get SIP trance on user level .
Equivalent to Asterisk "sip set debug peer <peername>"


TIA

Lloyd
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From chris.chen2004 at gmail.com  Tue Feb 15 07:13:44 2011
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Mon, 14 Feb 2011 23:13:44 -0500
Subject: [Freeswitch-users] Question about mod_dingaling.
In-Reply-To: <AANLkTi=BOA1fTfUberjRLzLu-Uts1xvp7Xr4-CzE1xrD@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>
Message-ID: <AANLkTik6rKyn6HH8x_9Q=1-O1K-Ax6dxEptwueD8Up-w@mail.gmail.com>

Thanks Tony, the git 0bce777a4a06701798e426b13b8b7e0153836077 did fix the
issue. I am happy to have Gtalk calling back. It is too hard for me to
figure out the issue actually in switch_rtp.c as all I could see was
mod_dingaling.c looping between two lines without rtp packets until it timed
out. That's why sometimes we can observe the issue but really cannot locate
the problem.

Thanks again.

Chris Chen

2011/2/14 Anthony Minessale <anthony.minessale at gmail.com>

> update again, I think I found the problem in the code.
>
>
> 2011/2/13 ??? <chenzhanping at gmail.com>:
> > Thank u very much,
> >
> > My server is have public ip,and router not have upnp.
> >
> > I modified client.xml   :
> >
> > <!-- <param name="ext-rtp-ip" value="auto-nat"/> -->   to   <param
> > name="ext-rtp-ip" value="My ip"/>
> >
> > Now my Freeswitch + mod_dingaling is working.
> >
> > But there is no voice when i call out from google voice and connect
> called
> > Successed.
> >
> > I try to called my other google phone use freeswitch + mod_dingaling ,
> > and into voicemail, the voicemail system can heared my voice,and record
> > good.
> >
> > This mean: caller is no voice,but called is have voice good.
> >
> > there is client.xml:
> >
> > http://pastebin.freeswitch.org/15371
> >
> > there is cli log:
> >
> > http://pastebin.freeswitch.org/15372
> >
> > freeswitch version:FreeSWITCH Version 1.0.head (git-a2c0da5 2011-02-11
> > 23-10-12 -0600)
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 bwibowo at gmail.com  Tue Feb 15 07:28:46 2011
From: bwibowo at gmail.com (Budi wibowo)
Date: Tue, 15 Feb 2011 04:28:46 +0000
Subject: [Freeswitch-users] Question about mod_dingaling.
In-Reply-To: <AANLkTik6rKyn6HH8x_9Q=1-O1K-Ax6dxEptwueD8Up-w@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>
Message-ID: <45842189-1297744129-cardhu_decombobulator_blackberry.rim.net-1469059183-@b25.c2.bise3.blackberry>

Just update to the current version by issuing make current. And it works again 

Thx a lot 
Budi wibowo
-----Original Message-----
From: Chris Chen <chris.chen2004 at gmail.com>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Mon, 14 Feb 2011 23:13:44 
To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Question about mod_dingaling.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at 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 chenzhanping at gmail.com  Tue Feb 15 07:47:34 2011
From: chenzhanping at gmail.com (=?GB2312?B?1bzGvbPC?=)
Date: Tue, 15 Feb 2011 12:47:34 +0800
Subject: [Freeswitch-users] Question about mod_dingaling.
In-Reply-To: <45842189-1297744129-cardhu_decombobulator_blackberry.rim.net-1469059183-@b25.c2.bise3.blackberry>
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>
Message-ID: <AANLkTim-5r58xUao_pVwGAKGBU0cy3Dn-jJ6DXkL6yb3@mail.gmail.com>

I updated,now, i heard the voice,Thank u very much.

2011/2/15 Budi wibowo <bwibowo at gmail.com>

> Just update to the current version by issuing make current. And it works
> again
>
> Thx a lot
> Budi wibowo
>  -----Original Message-----
> From: Chris Chen <chris.chen2004 at gmail.com>
> Sender: freeswitch-users-bounces at lists.freeswitch.org
> Date: Mon, 14 Feb 2011 23:13:44
> To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Question about mod_dingaling.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jgallartm at gmail.com  Tue Feb 15 10:12:42 2011
From: jgallartm at gmail.com (Javier Gallart)
Date: Tue, 15 Feb 2011 08:12:42 +0100
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 56, Issue 177
In-Reply-To: <mailman.3855.1297702532.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.3855.1297702532.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <AANLkTimf9wTpy36Hogi+kEVVAKWMZ2z+R2jA47x6NEfy@mail.gmail.com>

Anthony, thanks for the tip. I haven't seen any change though. I should have
mentioned that I'm focusing in the rtp stream coming form the callee to the
caller. The problem is that FS relays to the caller only half of the packets
received from the callee. The packets arrive at FS at a rate of 1 packet
every 20 ms, and with a payload of 20 bytes each. Packets from FS to the
caller are sent every 40 ms with a payload of 20 bytes, thus skipping half
of the information.

Thanks

>
> ---------- Forwarded message ----------
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Mon, 14 Feb 2011 10:38:57 -0600
> Subject: Re: [Freeswitch-users] g729 packets skipped in passthrough mode
> I bet its 20ms vs 30
> set passthru_ptime_mismatch to true either in vars.xml or in your
> dialplan both legs through export on the a leg or set in the a leg and
> in the {} on b.
>
>
> On Mon, Feb 14, 2011 at 10:28 AM, Brian West <brian at freeswitch.org> wrote:
> > What exactly is the problem?  I see no issue here can you elaborate on
> what you're seeing?
> >
> > /b
> >
> > On Feb 14, 2011, at 4:48 AM, Javier Gallart wrote:
> >
> >> I've tried to explicitly set ptime at 20ms at switch.conf -although it's
> not necessary afaik-. Has any one experienced this same issue?
> >>
> >> Thanks in advance..
> >>
> >> 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
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 avi at avimarcus.net  Tue Feb 15 10:47:23 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 15 Feb 2011 09:47:23 +0200
Subject: [Freeswitch-users] Simultaneous ring out (external)
In-Reply-To: <AANLkTin983TbMribnEchrrG-TUMvRuV4hvCaFR-k7sjG@mail.gmail.com>
References: <AANLkTin983TbMribnEchrrG-TUMvRuV4hvCaFR-k7sjG@mail.gmail.com>
Message-ID: <AANLkTik-7Rfyn1VX2F6xp7WqgSTeJmkpCuDPPO2q9r87@mail.gmail.com>

No, it should work the same.
Does your sip carrier allow multiple simultaneous calls with the same
credentials?
Is your bridge directly to "external/gateway/sample_gateway/1npaxxxxxxx" ?

And... as always, a trace of what happened, posted to pastebin, could
provide more feedback. Rather than just "it seems like it doesn't like" -
what happened?

-Avi

On Tue, Feb 15, 2011 at 3:22 AM, Frank Park <frank at telonium.com> wrote:

> Hello,
>
> Yet another puzzle I've been trying to resolve.
> I have been trying to implement a simultaneous ring group that calls
> multiple external phone numbers (ie cellphones). Similar to call group,
> first person who answers gets the bridging.
> Bridge with commas works well for internal extensions and call groups, but
> it seems like it doesn't like multiple external legs.
>
> Should I be using something else besides bridge function?
>
> Thank you,
> Frank
>
> --
>
> ----=======================----
> 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
>
>
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From wstephen80 at gmail.com  Tue Feb 15 12:08:57 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 15 Feb 2011 10:08:57 +0100
Subject: [Freeswitch-users] ESL: how to filter two uuid's?
In-Reply-To: <AANLkTikRQXPbrL3yZLJ+FPkF_4=CTU5x-QrpY+iehN-_@mail.gmail.com>
References: <AANLkTikjDdsyZFf=zcYcerLz7P_w-1+hvxE_rbZwccTj@mail.gmail.com>
	<AANLkTikmL03v+0o6eqvbci30sXZSH7GXURFwi-ZuXGu3@mail.gmail.com>
	<AANLkTikRQXPbrL3yZLJ+FPkF_4=CTU5x-QrpY+iehN-_@mail.gmail.com>
Message-ID: <AANLkTin-NwggyeBai51RfzgeZOa1ON+cMb7e+PSVvvsx@mail.gmail.com>

Ok, thank you, with latest git head the filter works fine!

Stephen

On Mon, Feb 14, 2011 at 6:02 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> try patch in latest GIT HEAD
>
> On Mon, Feb 14, 2011 at 7:04 AM, Chris Hemmings <chris at cheeky.org> wrote:
> > This seemed to work for me in Perl:
> >
> > $con->sendRecv("event plain ALL");
> > $con->sendRecv("filter Unique-ID ".$uuid);
> > $con->sendRecv("filter Unique-ID ".$out_uuid);
> >
> > Cheers,
> >
> > Chris
> >
> > On 14 February 2011 11:46, Stephen Wilde <wstephen80 at gmail.com> wrote:
> >>
> >> If I have 2 uuid (uuid_inbound and uuid_outbound) in an ESL socket
> >> outbound application, there is a way to filter events related to this
> two
> >> uuid?
> >> If I try to do:
> >> connection.filter("unique-id", uuid_inbound);
> >> connection.filter("unique-id", uuid_outbound);
> >> it seems that the second filter overwrite the first one and my
> application
> >> receives only uuid_outbound related events.
> >> I have tried also an:
> >> connection.filter("unique-id", uuid_inbound + "|" + uuid_outbound);
> >> but in this way no event are received.
> >> There is a way to filter 2 specific uuid's or I have to receive all
> events
> >> and implement the filtering in my ESL application?
> >> 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
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 wstephen80 at gmail.com  Tue Feb 15 12:48:54 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 15 Feb 2011 10:48:54 +0100
Subject: [Freeswitch-users] ESL Outbound: how to bridge media?
In-Reply-To: <AANLkTi=JAt=RkQ5b=DBajH_YR6TmiqExdjgF1=OL=fJ_@mail.gmail.com>
References: <AANLkTikLVLpSGkrRqeD3WVFZZgSaLnQvypCOG39hEhLX@mail.gmail.com>
	<AANLkTimwX8Uz4t9gp6Ppzfu+WE+UJa3FO4qHE7XKFOBz@mail.gmail.com>
	<AANLkTi=t-a7U9ov3wJ6Dh-PuniSfZbyeQ==BCAvMkUPN@mail.gmail.com>
	<AANLkTik27F7JjyMRLKryQZq_N+oVDsDzdmiPPa+d2KWN@mail.gmail.com>
	<AANLkTine=jG-dc98pDySashEvo3T5cSPwVXYwp54mhhC@mail.gmail.com>
	<AANLkTi=JAt=RkQ5b=DBajH_YR6TmiqExdjgF1=OL=fJ_@mail.gmail.com>
Message-ID: <AANLkTinhkOznPh98kkHp=rCU9Qm0xv_hW3c41rRJr=Kn@mail.gmail.com>

I have tried to wait originate completition but the "uuid_bridge" doesn't
works also in this case.

It seems that works only when at least one leg is answered so it's not
possible to do the "uuid_bridge" during progressing phase also if the
originate has ended (I don't set the ignore_early_media).

My application is this: http://pastebin.freeswitch.org/15387

The application:

1. receive a call with an "inbound_uuid"

2. create a new "outbound_uuid"

3. do a "bgapi originate" using the new "outbound_uuid"

4. when the called phone is ringing, receive a "CHANNEL_PROGRESS_MEDIA"
event on "outbound_uuid" and do a "pre_answer" on "inbound_uuid"

5. receive a "CHANNEL_PROGRESS_MEDIA" on "inbound_uuid"

6. receive a job completition event related to the "originate" so issue an
"uuid_bridge inbound_uuid outbound_uuid"

7. when a job completition event related to "uuid_bridge" is received, no
audio flow from outbound to inbound channel

8. when outbound answer the call, the application answer also the inbound
call but no audio flow in both directions

If I do the uuid_bridge after legB answer, then all is ok (obviously with no
audio during progressing phase).


The log of my application is:

[ERROR] newacme.cpp:46 mycallback() Connected!
[ERROR] newacme.cpp:65 mycallback() Inbound uuid =
8b2c39db-1ad9-489c-b72f-a92d4087bf99
[ERROR] newacme.cpp:68 mycallback() create_uuid:
394167aa-2811-4fcd-95c9-85576bdd9a7a
[ERROR] newacme.cpp:89 mycallback() bgapi originate
[origination_uuid=394167aa-2811-4fcd-95c9-85576bdd9a7a]sofia/external/xxxx at yyyy&park()
[ERROR] newacme.cpp:91 mycallback() Job-ID:
1c654ac2-c22d-418f-9fad-bb6b6b35aeff
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_UUID]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_OUTGOING]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_ORIGINATE]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_STATE]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_CALLSTATE]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_STATE]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_STATE]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CALL_UPDATE]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] - [CODEC]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] - [CODEC]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_CALLSTATE]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_PROGRESS_MEDIA]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_STATE]
[ERROR] newacme.cpp:112 mycallback() Got an event: [BACKGROUND_JOB] -
[1c654ac2-c22d-418f-9fad-bb6b6b35aeff]
[ERROR] newacme.cpp:119 mycallback() bgapi uuid_bridge
8b2c39db-1ad9-489c-b72f-a92d4087bf99 394167aa-2811-4fcd-95c9-85576bdd9a7a
[ERROR] newacme.cpp:123 mycallback() Job-ID:
0d43cffe-fdc6-4a13-97e6-aeee199c45d5
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_EXECUTE]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_PARK]
[ERROR] newacme.cpp:112 mycallback() Got an event: [BACKGROUND_JOB] -
[0d43cffe-fdc6-4a13-97e6-aeee199c45d5]
[ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
[CHANNEL_EXECUTE]
[ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
[CHANNEL_CALLSTATE]
[ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
[CHANNEL_PROGRESS_MEDIA]
[ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
[CHANNEL_EXECUTE_COMPLETE]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CALL_UPDATE]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_CALLSTATE]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_ANSWER]
[ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
[CHANNEL_EXECUTE]
[ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
[CHANNEL_CALLSTATE]
[ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
[CHANNEL_ANSWER]
[ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
[CHANNEL_EXECUTE_COMPLETE]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_CALLSTATE]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_HANGUP]
[ERROR] newacme.cpp:160 mycallback() hangup
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_UNPARK]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_EXECUTE_COMPLETE]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_STATE]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_STATE]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_HANGUP_COMPLETE]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_DESTROY]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_CALLSTATE]
[ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_STATE]
[ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
[CHANNEL_EXECUTE]
[ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
[CHANNEL_CALLSTATE]
[ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
[CHANNEL_HANGUP]
[ERROR] newacme.cpp:171 mycallback() hangup
[ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
[CHANNEL_EXECUTE_COMPLETE]
[ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
[CHANNEL_UNPARK]
[ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
[CHANNEL_EXECUTE_COMPLETE]
[ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
[CHANNEL_STATE]
[ERROR] newacme.cpp:184 mycallback() End.


Stephen


On Tue, Feb 15, 2011 at 1:14 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> if you do originate without ignore_early_media=true set it will end
> the soonest possible where it's suitable for a bridge.
> so that is the best bet to wait for originate to end.
>
>
> On Mon, Feb 14, 2011 at 6:11 PM, Stephen Wilde <wstephen80 at gmail.com>
> wrote:
> > Ok, now I'm doing bridge when I receive "CHANNEL_ANSWER" or
> > "CHANNEL_PROGRESS_MEDIA" on outbound leg but you say that I have wait
> that
> > "originate has ended": you mean that I have to wait for "BACKGROUND_JOB"
> > event related to my "bgapi originate ... &park"?
> > I'm already doing "uuid_bridge <inbound_uuid> <outbound_uuid>".
> > I'll try also with intercept and inline originate. Thank you!
> > Stephen
> >
> > On Tue, Feb 15, 2011 at 12:53 AM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> you need to make sure originate has ended on the outbound leg before
> >> you use it in a bridge etc.
> >> you also need to supply the inbound leg first in uuid_bridge if that
> >> is something you want to do.
> >>
> >> Easier would be to originate the B leg to park inline and tell A leg
> >> to execute intercept on the B leg uuid.
> >>
> >>
> >>
> >> On Mon, Feb 14, 2011 at 5:44 PM, Stephen Wilde <wstephen80 at gmail.com>
> >> wrote:
> >> > Sorry but I have missed something.
> >> > I know that I can solve this problem directly in dialplan with a
> bridge
> >> > but
> >> > what I'm doing is an "Freeswitch ESL learning" because I have to port
> >> > some
> >> > application in Freeswitch and I'm learning how to implement some
> >> > functionality.
> >> > For me it's important to take control of both inbound/outbound in full
> >> > async
> >> > way and I have the necessity to do the complete call control.
> >> > I'm not sure but to me it seems that with a normal bridge I lose the
> >> > control
> >> > of two sessions, for example, an outbound answer is propagated by
> bridge
> >> > application as inbound answer.
> >> > What I want to do is an audio bridging so my application can take
> >> > control of
> >> > "signaling bridging".
> >> >
> >> > I'm wrong? There are other way to do that?
> >> > Stephen
> >> > On Tue, Feb 15, 2011 at 12:19 AM, Michael Collins <msc at freeswitch.org
> >
> >> > wrote:
> >> >>
> >> >> My head is spinning after reading this email. :)
> >> >> It sounds like you just need a simple bridge from the incoming leg to
> >> >> the
> >> >> outgoing leg. Can you pre_answer the A leg then execute a good
> >> >> old-fashioned
> >> >> bridge to the b-leg?
> >> >> -MC
> >> >> On Mon, Feb 14, 2011 at 4:57 PM, Stephen Wilde <wstephen80 at gmail.com
> >
> >> >> wrote:
> >> >>>
> >> >>> The problem I have is to propagate the audio coming from an
> >> >>> "originated"
> >> >>> outbound session to the inbound session when the outbound is in the
> >> >>> PROGRESS
> >> >>> MEDIA phase.
> >> >>> When my application receives the "CHANNEL_PROGRESS_MEDIA" event from
> >> >>> outbound session I can do a "pre_answer" on inbound session but I'm
> >> >>> not
> >> >>> capable to do an audio bridge.
> >> >>> I have tried with "uuid_bridge <outbound_uuid> <inbound_uuid>" with
> no
> >> >>> result probably because this api requires that at least one session
> >> >>> must be
> >> >>> answered.
> >> >>> I don't want to answer to the inbound session to propagate the
> >> >>> outbound
> >> >>> progressing media but I want to answer to inbound only on outbound
> >> >>> answer.
> >> >>> Any way to do that?
> >> >>> 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
> >> >>
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at 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 yehavi.bourvine at gmail.com  Tue Feb 15 13:22:28 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 15 Feb 2011 12:22:28 +0200
Subject: [Freeswitch-users] Lua API broken???
Message-ID: <AANLkTimnX5FoTOP_UhTaFPzDUZ3cOoon3YnRJrxA9zqz@mail.gmail.com>

Hello,

  There is a problem with of our LUA scripts which started with GIT of a
about a week ago. From the dialplan I call a LUA script which does the
following (stripped down just to show the problem):

local api = freeswitch.API();
local isShared = api:execute("user_data", "86111 var SharedLine");
It causes Freeswitch to do segmentation fault, sometimes with an error about
double free call.

If I take the orignal script and remove the api:execute() calls it works
fine.

Is it some known problem/change in the API, or shall I open a JIRA for it?

                            Thanks, __Yehavi:
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From steveayre at gmail.com  Tue Feb 15 13:49:23 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 15 Feb 2011 10:49:23 +0000
Subject: [Freeswitch-users] Lua API broken???
In-Reply-To: <AANLkTimnX5FoTOP_UhTaFPzDUZ3cOoon3YnRJrxA9zqz@mail.gmail.com>
References: <AANLkTimnX5FoTOP_UhTaFPzDUZ3cOoon3YnRJrxA9zqz@mail.gmail.com>
Message-ID: <AANLkTiky0Yoi1qagc_gzb-cqLxdx8hYM1yvxD64ZsQaU@mail.gmail.com>

Try to reproduce it on the latest Git, if you can do so open a Jira.

-Steve



On 15 February 2011 10:22, Yehavi Bourvine <yehavi.bourvine at gmail.com>wrote:

> Hello,
>
>   There is a problem with of our LUA scripts which started with GIT of a
> about a week ago. From the dialplan I call a LUA script which does the
> following (stripped down just to show the problem):
>
> local api = freeswitch.API();
> local isShared = api:execute("user_data", "86111 var SharedLine");
> It causes Freeswitch to do segmentation fault, sometimes with an error
> about double free call.
>
> If I take the orignal script and remove the api:execute() calls it works
> fine.
>
> Is it some known problem/change in the API, or shall I open a JIRA for it?
>
>                             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 mitch.johnson7 at gmail.com  Tue Feb 15 08:08:38 2011
From: mitch.johnson7 at gmail.com (Mitch Johnson)
Date: Tue, 15 Feb 2011 00:08:38 -0500
Subject: [Freeswitch-users] Cisco Call Manager question
Message-ID: <D6FA37B2-E9F1-4E95-AA9B-16AA4892B3EC@gmail.com>

Has anyone connected a Freeswitch PBX to a Cisco CalManager, version 7 and above?

I need to send all digits down to the callmanager that do not reside on the switch.  I assume I have to have a SIP trunk between the two and then include a dial plan reference to send all digits down the trunk.

I've just started working with FreeSwitch, so excuse my ignorance,

Thanks,

Mitch Johnson

From kris at kriskinc.com  Tue Feb 15 16:12:24 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 15 Feb 2011 08:12:24 -0500
Subject: [Freeswitch-users] Cisco Call Manager question
In-Reply-To: <D6FA37B2-E9F1-4E95-AA9B-16AA4892B3EC@gmail.com>
References: <D6FA37B2-E9F1-4E95-AA9B-16AA4892B3EC@gmail.com>
Message-ID: <AANLkTikrgPEHo9sUmv=8fmtNOcmws9vhRmbNFNKDQzDy@mail.gmail.com>

I have done various Call Manager integrations (including 7).

Sending "all digits" down the trunk to CUCM should be fairly simple as
long as the Cisco returns something sane (like a 404) for unknown
extensions.  It should.

On Tue, Feb 15, 2011 at 12:08 AM, Mitch Johnson
<mitch.johnson7 at gmail.com> wrote:
> Has anyone connected a Freeswitch PBX to a Cisco CalManager, version 7 and above?
>
> I need to send all digits down to the callmanager that do not reside on the switch. ?I assume I have to have a SIP trunk between the two and then include a dial plan reference to send all digits down the trunk.
>
> I've just started working with FreeSwitch, so excuse my ignorance,
>
> Thanks,
>
> Mitch Johnson


-- 
Kristian Kielhofner


From kris at kriskinc.com  Tue Feb 15 16:14:15 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 15 Feb 2011 08:14:15 -0500
Subject: [Freeswitch-users] Getting SIP trace on user level
In-Reply-To: <AANLkTimhW139WdDKXiQkRm43Ai13f_7Z+N9Z-K2y8Ldi@mail.gmail.com>
References: <AANLkTimhW139WdDKXiQkRm43Ai13f_7Z+N9Z-K2y8Ldi@mail.gmail.com>
Message-ID: <AANLkTikxBuc960_-sZvXQXdXAC5LzmiNvagj9RUC2GPx@mail.gmail.com>

In FreeSWITCH: not that I know of.

In Linux, yes: http://ngrep.sourceforge.net/

Use ngrep and match on anything, including IP and port combination
from a specific peer (that's all Asterisk does anyway).  It would be
trivial to throw a script together to pull out the sofia contact for a
user and pass the IP:port to ngrep for filtering.

You'll find ngrep to be very handy.

On Mon, Feb 14, 2011 at 11:06 PM, Aloysius Lloyd
<lloyd.aloysius at gmail.com> wrote:
> Hi All,
> Is there any console command?available?to get SIP trance on user level .
> Equivalent to Asterisk "sip set debug peer <peername>"
>
>
> TIA
> 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
>
>



-- 
Kristian Kielhofner


From kris at kriskinc.com  Tue Feb 15 16:17:49 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 15 Feb 2011 08:17:49 -0500
Subject: [Freeswitch-users] Simultaneous ring out (external)
In-Reply-To: <AANLkTin983TbMribnEchrrG-TUMvRuV4hvCaFR-k7sjG@mail.gmail.com>
References: <AANLkTin983TbMribnEchrrG-TUMvRuV4hvCaFR-k7sjG@mail.gmail.com>
Message-ID: <AANLkTi=kkuz7Tn+b4EmzjCGppRANS39-sjGfN_A_PhYF@mail.gmail.com>

Frank,

  This is easy to do with bridge:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Calling_multiple_destinations

  However, one issue you'll likely experience is early audio and
answers.  When calling multiple cell phones, for example, it's very
likely at least one will be out of service, ignored (callee declining
call), etc.  As long as voicemail is active it's a race to see what
answers first - an actual human being or a cell carrier voicemail
system.

  The first thing you can do is pass
http://wiki.freeswitch.org/wiki/Variable_ignore_early_media to bridge.
 That way you'll only have to worry about actual answers - even if it
is a voicemail system.

On Mon, Feb 14, 2011 at 8:22 PM, Frank Park <frank at telonium.com> wrote:
> Hello,
> Yet another puzzle I've been trying to resolve.
> I have been trying to implement a simultaneous ring group that calls
> multiple external phone numbers (ie cellphones). Similar to call group,
> first person who answers gets the bridging.
> Bridge with commas works well for internal extensions and call groups, but
> it seems like it doesn't like multiple external legs.
> Should I be using something else besides bridge function?
> Thank you,
> Frank
>
> --
>
> ----=======================----
> 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
>
>



-- 
Kristian Kielhofner


From steveayre at gmail.com  Tue Feb 15 16:47:28 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 15 Feb 2011 13:47:28 +0000
Subject: [Freeswitch-users] Getting SIP trace on user level
In-Reply-To: <AANLkTikxBuc960_-sZvXQXdXAC5LzmiNvagj9RUC2GPx@mail.gmail.com>
References: <AANLkTimhW139WdDKXiQkRm43Ai13f_7Z+N9Z-K2y8Ldi@mail.gmail.com>
	<AANLkTikxBuc960_-sZvXQXdXAC5LzmiNvagj9RUC2GPx@mail.gmail.com>
Message-ID: <AANLkTi=+yL16rqZs9ig0R4JUfsfFOtmfr1s9jDY=-Gpf@mail.gmail.com>

tshark can also do the same (part of wireshark)

-Steve


On 15 February 2011 13:14, Kristian Kielhofner <kris at kriskinc.com> wrote:

> In FreeSWITCH: not that I know of.
>
> In Linux, yes: http://ngrep.sourceforge.net/
>
> Use ngrep and match on anything, including IP and port combination
> from a specific peer (that's all Asterisk does anyway).  It would be
> trivial to throw a script together to pull out the sofia contact for a
> user and pass the IP:port to ngrep for filtering.
>
> You'll find ngrep to be very handy.
>
> On Mon, Feb 14, 2011 at 11:06 PM, Aloysius Lloyd
> <lloyd.aloysius at gmail.com> wrote:
> > Hi All,
> > Is there any console command available to get SIP trance on user level .
> > Equivalent to Asterisk "sip set debug peer <peername>"
> >
> >
> > TIA
> > 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
> >
> >
>
>
>
> --
> 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 frank at telonium.com  Tue Feb 15 17:54:37 2011
From: frank at telonium.com (Frank Park)
Date: Tue, 15 Feb 2011 09:54:37 -0500
Subject: [Freeswitch-users] Simultaneous ring out (external)
In-Reply-To: <AANLkTi=kkuz7Tn+b4EmzjCGppRANS39-sjGfN_A_PhYF@mail.gmail.com>
References: <AANLkTin983TbMribnEchrrG-TUMvRuV4hvCaFR-k7sjG@mail.gmail.com>
	<AANLkTi=kkuz7Tn+b4EmzjCGppRANS39-sjGfN_A_PhYF@mail.gmail.com>
Message-ID: <AANLkTinprN1_xaBkqp=hupzeVPK4VvnsE3aWVT_dJ8p9@mail.gmail.com>

Let me see if I can generate the the log and post it on a pastebin today.
The ignore_early_media is disabled, last time I checked, but I will also
confirm that it's doing that on this dialplan.

More to come :)

Frank

On Tue, Feb 15, 2011 at 8:17 AM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> Frank,
>
>  This is easy to do with bridge:
>
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Calling_multiple_destinations
>
>  However, one issue you'll likely experience is early audio and
> answers.  When calling multiple cell phones, for example, it's very
> likely at least one will be out of service, ignored (callee declining
> call), etc.  As long as voicemail is active it's a race to see what
> answers first - an actual human being or a cell carrier voicemail
> system.
>
>  The first thing you can do is pass
> http://wiki.freeswitch.org/wiki/Variable_ignore_early_media to bridge.
>  That way you'll only have to worry about actual answers - even if it
> is a voicemail system.
>
> On Mon, Feb 14, 2011 at 8:22 PM, Frank Park <frank at telonium.com> wrote:
> > Hello,
> > Yet another puzzle I've been trying to resolve.
> > I have been trying to implement a simultaneous ring group that calls
> > multiple external phone numbers (ie cellphones). Similar to call group,
> > first person who answers gets the bridging.
> > Bridge with commas works well for internal extensions and call groups,
> but
> > it seems like it doesn't like multiple external legs.
> > Should I be using something else besides bridge function?
> > Thank you,
> > Frank
> >
> > --
> >
> > ----=======================----
> > 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
> >
> >
>
>
>
> --
> 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
>



-- 

----=======================----
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
----=======================----
-------------- next part --------------
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From fs-list at communicatefreely.net  Tue Feb 15 18:29:46 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Tue, 15 Feb 2011 10:29:46 -0500
Subject: [Freeswitch-users] Aastra phone registration lost
In-Reply-To: <AANLkTikZbWPqf7N1fT-9fF8Fvydi85mJVZ+o8_J=RLjE@mail.gmail.com>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>	<4D52B987.7050002@communicatefreely.net>	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>	<4D534AFD.3020401@communicatefreely.net>	<AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>	<60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>	<AANLkTimux4MBM1K_7itE4GnZQHjy_vR2Y6sr_z+u3gkR@mail.gmail.com>	<4D5577CE.1000603@communicatefreely.net>	<AANLkTinJK+e9hZJbk4ur_OQwGV5B7tRLrpPMH=TU5MOo@mail.gmail.com>	<4D59C085.9040008@communicatefreely.net>
	<AANLkTikZbWPqf7N1fT-9fF8Fvydi85mJVZ+o8_J=RLjE@mail.gmail.com>
Message-ID: <4D5A9BEA.7070302@communicatefreely.net>

Good luck!

There are some aastra.cfg sections earlier in this thread.  One thing I 
found about the 9133i and other legacy phones is that they work better 
if you don't subscribe to MWI but instead send unsolicited updates.  
They also have odd default codec settings, so you want to specify the 
codec string with the ptime that you want (I think they default to 30 
instead of 20).

I like the newer phones much better, but we have about 150 of these in 
the field, so we have to make them work.

-Tim

Aloysius Lloyd wrote:
> Tim,
>
> Thank you for the settings will give a try.
>
>
> Thanks
> Lloyd
>
> On Mon, Feb 14, 2011 at 6:53 PM, Tim St. Pierre 
> <fs-list at communicatefreely.net <mailto:fs-list at communicatefreely.net>> 
> wrote:
>
>     <domains>
>        <domain name="all" alias="true" parse="true"/>
>        <domain name="pbx.MYDOMAIN.NET <http://pbx.MYDOMAIN.NET>"
>     alias="true" parse="false"/>
>      </domains>
>
>     <settings>
>        <param name="user-agent-string" value="Communicate Freely 2.0"/>
>        <param name="debug" value="0"/>
>        <param name="sip-trace" value="no"/>
>        <param name="log-auth-failures" value="true"/>
>        <param name="rfc2833-pt" value="101"/>
>        <param name="sip-port" value="$${internal_sip_port}"/>
>        <param name="dialplan" value="xml"/>
>        <param name="context" value="internal"/>
>        <param name="dtmf-duration" value="100"/>
>        <param name="inbound-codec-prefs"
>     value="$${internal_codec_prefs}"/>
>        <param name="outbound-codec-prefs"
>     value="$${internal_codec_prefs}"/>
>        <param name="rtp-timer-name" value="soft"/>
>        <param name="sip-ip" value="$${public_ip}"/>
>        <param name="rtp-ip" value="$${public_ip}"/>
>        <param name="hold-music" value="$${moh_prefix}alt"/>
>        <param name="dtmf-type" value="rfc2833"/>
>
>        <param name="force-register-domain" value="$${domain}"/>
>        <param name="force-subscription-domain" value="pbx.$${domain}"/>
>        <param name="force-register-db-domain" value="$${domain}"/>
>        <param name="force-subscription-expires" value="600"/>
>
>     These are the most important ones I think.
>
>        <param name="NDLB-received-in-nat-reg-contact" value="true"/>
>        <param name="sip-force-contact" value="
>     NDLB-connectile-dysfunction"/>
>
>     I'm also using sip-force-expires to 600 at the moment, and ping =
>     10.  I
>     will probably increase those eventually to reduce bandwidth.  I'm
>     still
>     in Beta right now, but I'm not having too many issues.
>
>     Some of those params are added per-device using the directory, so
>     I can
>     tweak them depending on which device registers, and what the NAT
>     status
>     is of that device.  I'm really pushing to get IPv6 on the phones, as
>     well as on some of the more prominent (but competitive) DSL providers
>     here so that we can forego NAT altogether some day.
>
>     Hope that's helpful.  I haven't really gone through and figured out
>     which variables do what at the moment, but it seems to work as it is.
>
>     -Tim
>     Aloysius Lloyd wrote:
>     > Tim,
>     >
>     > Thank you for the information.
>     >
>     > I have around 275 Aastra 9133i models phones in production .These
>     > phones installed 31/2 years ago. I am trying to migrate to
>     FreeSWITCH
>     > could not make it work reliably.
>     >
>     > What are the profile settings turned on for these phones works
>     reliably?
>     >
>     >
>     > Thanks
>     > Lloyd
>     >
>     >
>     >
>     > On Fri, Feb 11, 2011 at 12:54 PM, Tim St. Pierre
>     > <fs-list at communicatefreely.net
>     <mailto:fs-list at communicatefreely.net>
>     <mailto:fs-list at communicatefreely.net
>     <mailto:fs-list at communicatefreely.net>>>
>     > wrote:
>     >
>     >     Sure,
>     >
>     >     I normally administer about 300 Aastra phones, with every
>     model they
>     >     make represented.
>     >
>     >     I have 22 connected to our Freeswitch "beta" system, which will
>     >     eventually become production.
>     >
>     >     All the endpoints are behind NAT without exception.  There are a
>     >     number
>     >     of legacy 9133i and 480i phones on the network that don't
>     have the
>     >     newer
>     >     NAT traversal features available, but this doesn't seem to be a
>     >     problem.  I have some of the nat traversal options turned on
>     in the
>     >     sofia profile though, so fs will send media back to the
>     originating
>     >     address and port.
>     >
>     >     They have been quite reliable, and the sound quality has been
>     >     excellent,
>     >     with the newer phones using g722 at 16KHz.
>     >
>     >     There are a few advanced features that I haven't had a
>     chance to play
>     >     with yet, but here's what I have working:
>     >
>     >     Regular calls, in and out.
>     >     Intercom calls (auto-answer to speaker phone)
>     >     Automatic update of destination name and number (updates
>     when checking
>     >     voice mail, and when calling an extension).  Only on newer
>     phones
>     >     Blind and attended transfer
>     >     Music on hold
>     >     SIP using udp or tcp (haven't tried TLS yet)
>     >     Fewer issues with DTMF than with asterisk, using rfc2833
>     dtmf (no
>     >     issues
>     >     as of yet).
>     >     BLF lamps work correctly, flashing when the phone rings, lit
>     >     steady when
>     >     they are on the phone.
>     >     Distinctive ringing works.
>     >     I haven't tried SLA yet, but Aastra recently released a firmware
>     >     update
>     >     that fixes a missing header, reported to have broken correct SLA
>     >     operation.  I'm hoping to test that in the next week or two.
>     >
>     >     The phones provision very nicely - we auto generate config
>     using PHP
>     >     scripts that generate a config file on the fly from the user
>     database.
>     >     These are very easy phones to deploy in large installations,
>     or to the
>     >     outside world (not readily accessible).  They have just
>     added some new
>     >     features that allow for remote diagnostics of the phones as
>     well.
>     >
>     >     There is a great deal of XML programmability in the phones
>     too, which
>     >     I'm starting to use for call control and other useful things
>     (updating
>     >     forwarding rules in the database, or conference and
>     recording control
>     >     using ESL).
>     >
>     >     Hope that helps!
>     >
>     >     -Tim
>     >
>     >     Aloysius Lloyd wrote:
>     >     > Tim,
>     >     >
>     >     > Can you share your success stories FreeSWITCH and Aastra.
>     >     >
>     >     > Aastra Phones Behind the NAT?
>     >     >
>     >     > In my case Aastra phones registration not a problem.
>     >     >
>     >     > But calls drooped every 60 sec ... in the same environment
>     >     Linksys and
>     >     > Polycom works perfectly.
>     >     >
>     >     > How stable the Aastra phones with FreeSWITCH system.
>     >     >
>     >     > TIA
>     >     >
>     >     > Lloyd
>     >
>     >
>     >     _______________________________________________
>     >     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 lloyd.aloysius at gmail.com  Tue Feb 15 18:42:48 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 15 Feb 2011 10:42:48 -0500
Subject: [Freeswitch-users] Aastra phone registration lost
In-Reply-To: <4D5A9BEA.7070302@communicatefreely.net>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>
	<4D52B987.7050002@communicatefreely.net>
	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>
	<4D534AFD.3020401@communicatefreely.net>
	<AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>
	<60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>
	<AANLkTimux4MBM1K_7itE4GnZQHjy_vR2Y6sr_z+u3gkR@mail.gmail.com>
	<4D5577CE.1000603@communicatefreely.net>
	<AANLkTinJK+e9hZJbk4ur_OQwGV5B7tRLrpPMH=TU5MOo@mail.gmail.com>
	<4D59C085.9040008@communicatefreely.net>
	<AANLkTikZbWPqf7N1fT-9fF8Fvydi85mJVZ+o8_J=RLjE@mail.gmail.com>
	<4D5A9BEA.7070302@communicatefreely.net>
Message-ID: <AANLkTikuozcozSiFFL8-6+s3iRP4Lfy9DT2xFi9TVBOd@mail.gmail.com>

I have the same issue, around 275 phones in the field. I want the 275 phones
work with FreeSWITCH.


Thanks
Lloyd


On Tue, Feb 15, 2011 at 10:29 AM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:

> Good luck!
>
> There are some aastra.cfg sections earlier in this thread.  One thing I
> found about the 9133i and other legacy phones is that they work better
> if you don't subscribe to MWI but instead send unsolicited updates.
> They also have odd default codec settings, so you want to specify the
> codec string with the ptime that you want (I think they default to 30
> instead of 20).
>
> I like the newer phones much better, but we have about 150 of these in
> the field, so we have to make them work.
>
> -Tim
>
> Aloysius Lloyd wrote:
> > Tim,
> >
> > Thank you for the settings will give a try.
> >
> >
> > Thanks
> > Lloyd
> >
> > On Mon, Feb 14, 2011 at 6:53 PM, Tim St. Pierre
> > <fs-list at communicatefreely.net <mailto:fs-list at communicatefreely.net>>
> > wrote:
> >
> >     <domains>
> >        <domain name="all" alias="true" parse="true"/>
> >        <domain name="pbx.MYDOMAIN.NET <http://pbx.MYDOMAIN.NET>"
> >     alias="true" parse="false"/>
> >      </domains>
> >
> >     <settings>
> >        <param name="user-agent-string" value="Communicate Freely 2.0"/>
> >        <param name="debug" value="0"/>
> >        <param name="sip-trace" value="no"/>
> >        <param name="log-auth-failures" value="true"/>
> >        <param name="rfc2833-pt" value="101"/>
> >        <param name="sip-port" value="$${internal_sip_port}"/>
> >        <param name="dialplan" value="xml"/>
> >        <param name="context" value="internal"/>
> >        <param name="dtmf-duration" value="100"/>
> >        <param name="inbound-codec-prefs"
> >     value="$${internal_codec_prefs}"/>
> >        <param name="outbound-codec-prefs"
> >     value="$${internal_codec_prefs}"/>
> >        <param name="rtp-timer-name" value="soft"/>
> >        <param name="sip-ip" value="$${public_ip}"/>
> >        <param name="rtp-ip" value="$${public_ip}"/>
> >        <param name="hold-music" value="$${moh_prefix}alt"/>
> >        <param name="dtmf-type" value="rfc2833"/>
> >
> >        <param name="force-register-domain" value="$${domain}"/>
> >        <param name="force-subscription-domain" value="pbx.$${domain}"/>
> >        <param name="force-register-db-domain" value="$${domain}"/>
> >        <param name="force-subscription-expires" value="600"/>
> >
> >     These are the most important ones I think.
> >
> >        <param name="NDLB-received-in-nat-reg-contact" value="true"/>
> >        <param name="sip-force-contact" value="
> >     NDLB-connectile-dysfunction"/>
> >
> >     I'm also using sip-force-expires to 600 at the moment, and ping =
> >     10.  I
> >     will probably increase those eventually to reduce bandwidth.  I'm
> >     still
> >     in Beta right now, but I'm not having too many issues.
> >
> >     Some of those params are added per-device using the directory, so
> >     I can
> >     tweak them depending on which device registers, and what the NAT
> >     status
> >     is of that device.  I'm really pushing to get IPv6 on the phones, as
> >     well as on some of the more prominent (but competitive) DSL providers
> >     here so that we can forego NAT altogether some day.
> >
> >     Hope that's helpful.  I haven't really gone through and figured out
> >     which variables do what at the moment, but it seems to work as it is.
> >
> >     -Tim
> >     Aloysius Lloyd wrote:
> >     > Tim,
> >     >
> >     > Thank you for the information.
> >     >
> >     > I have around 275 Aastra 9133i models phones in production .These
> >     > phones installed 31/2 years ago. I am trying to migrate to
> >     FreeSWITCH
> >     > could not make it work reliably.
> >     >
> >     > What are the profile settings turned on for these phones works
> >     reliably?
> >     >
> >     >
> >     > Thanks
> >     > Lloyd
> >     >
> >     >
> >     >
> >     > On Fri, Feb 11, 2011 at 12:54 PM, Tim St. Pierre
> >     > <fs-list at communicatefreely.net
> >     <mailto:fs-list at communicatefreely.net>
> >     <mailto:fs-list at communicatefreely.net
> >     <mailto:fs-list at communicatefreely.net>>>
> >     > wrote:
> >     >
> >     >     Sure,
> >     >
> >     >     I normally administer about 300 Aastra phones, with every
> >     model they
> >     >     make represented.
> >     >
> >     >     I have 22 connected to our Freeswitch "beta" system, which will
> >     >     eventually become production.
> >     >
> >     >     All the endpoints are behind NAT without exception.  There are
> a
> >     >     number
> >     >     of legacy 9133i and 480i phones on the network that don't
> >     have the
> >     >     newer
> >     >     NAT traversal features available, but this doesn't seem to be a
> >     >     problem.  I have some of the nat traversal options turned on
> >     in the
> >     >     sofia profile though, so fs will send media back to the
> >     originating
> >     >     address and port.
> >     >
> >     >     They have been quite reliable, and the sound quality has been
> >     >     excellent,
> >     >     with the newer phones using g722 at 16KHz.
> >     >
> >     >     There are a few advanced features that I haven't had a
> >     chance to play
> >     >     with yet, but here's what I have working:
> >     >
> >     >     Regular calls, in and out.
> >     >     Intercom calls (auto-answer to speaker phone)
> >     >     Automatic update of destination name and number (updates
> >     when checking
> >     >     voice mail, and when calling an extension).  Only on newer
> >     phones
> >     >     Blind and attended transfer
> >     >     Music on hold
> >     >     SIP using udp or tcp (haven't tried TLS yet)
> >     >     Fewer issues with DTMF than with asterisk, using rfc2833
> >     dtmf (no
> >     >     issues
> >     >     as of yet).
> >     >     BLF lamps work correctly, flashing when the phone rings, lit
> >     >     steady when
> >     >     they are on the phone.
> >     >     Distinctive ringing works.
> >     >     I haven't tried SLA yet, but Aastra recently released a
> firmware
> >     >     update
> >     >     that fixes a missing header, reported to have broken correct
> SLA
> >     >     operation.  I'm hoping to test that in the next week or two.
> >     >
> >     >     The phones provision very nicely - we auto generate config
> >     using PHP
> >     >     scripts that generate a config file on the fly from the user
> >     database.
> >     >     These are very easy phones to deploy in large installations,
> >     or to the
> >     >     outside world (not readily accessible).  They have just
> >     added some new
> >     >     features that allow for remote diagnostics of the phones as
> >     well.
> >     >
> >     >     There is a great deal of XML programmability in the phones
> >     too, which
> >     >     I'm starting to use for call control and other useful things
> >     (updating
> >     >     forwarding rules in the database, or conference and
> >     recording control
> >     >     using ESL).
> >     >
> >     >     Hope that helps!
> >     >
> >     >     -Tim
> >     >
> >     >     Aloysius Lloyd wrote:
> >     >     > Tim,
> >     >     >
> >     >     > Can you share your success stories FreeSWITCH and Aastra.
> >     >     >
> >     >     > Aastra Phones Behind the NAT?
> >     >     >
> >     >     > In my case Aastra phones registration not a problem.
> >     >     >
> >     >     > But calls drooped every 60 sec ... in the same environment
> >     >     Linksys and
> >     >     > Polycom works perfectly.
> >     >     >
> >     >     > How stable the Aastra phones with FreeSWITCH system.
> >     >     >
> >     >     > TIA
> >     >     >
> >     >     > Lloyd
> >     >
> >     >
> >     >     _______________________________________________
> >     >     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
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From christian at yellox.de  Tue Feb 15 18:48:15 2011
From: christian at yellox.de (Christian Hiller)
Date: Tue, 15 Feb 2011 16:48:15 +0100
Subject: [Freeswitch-users] javascript hanguphook
Message-ID: <4D5AA03F.8030307@yellox.de>

Hello,

i have a javscript that is called from xml-dialplan
...
function on_hangup(hup_session, how)
{
     console_log("info","got hungup");
     exit();
}

session.execute('bridge','sofia/internal....');
session.setHangupHook(on_hangup);
...

Now i experience, that this function is only called if the session is 
not answered yet.
Once its answered and then got hung up, the function on_hangup is not 
called anymore.
Any ideas ?

Kind regards

Christian Hiller


From anthony.minessale at gmail.com  Tue Feb 15 18:58:06 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 15 Feb 2011 09:58:06 -0600
Subject: [Freeswitch-users] Lua API broken???
In-Reply-To: <AANLkTimnX5FoTOP_UhTaFPzDUZ3cOoon3YnRJrxA9zqz@mail.gmail.com>
References: <AANLkTimnX5FoTOP_UhTaFPzDUZ3cOoon3YnRJrxA9zqz@mail.gmail.com>
Message-ID: <AANLkTimpGDx0sJb-XC_u0aW=fi0+NfggM-sBfTguR48m@mail.gmail.com>

http://jira.freeswitch.org/browse/FS-3056

fixed in HEAD

Yes you should always open a jira when you have a problem like this.



On Tue, Feb 15, 2011 at 4:22 AM, Yehavi Bourvine
<yehavi.bourvine at gmail.com> wrote:
> Hello,
>
> ? There is a?problem with of our LUA scripts which started with GIT of a
> about a week ago. From the dialplan I call a LUA script which does the
> following (stripped down just to show the problem):
>
> local api = freeswitch.API();
> local isShared = api:execute("user_data", "86111 var SharedLine");
> It causes Freeswitch to do segmentation fault, sometimes with an error about
> double free call.
>
> If I take the orignal script and remove the api:execute() calls it works
> fine.
>
> Is it some known problem/change in the API,?or shall I open a JIRA for it?
>
> ??????????????????????????? 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
>
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 brian at freeswitch.org  Tue Feb 15 19:06:43 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 15 Feb 2011 10:06:43 -0600
Subject: [Freeswitch-users] Aastra phone registration lost
In-Reply-To: <AANLkTikuozcozSiFFL8-6+s3iRP4Lfy9DT2xFi9TVBOd@mail.gmail.com>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>
	<4D52B987.7050002@communicatefreely.net>
	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>
	<4D534AFD.3020401@communicatefreely.net>
	<AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>
	<60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>
	<AANLkTimux4MBM1K_7itE4GnZQHjy_vR2Y6sr_z+u3gkR@mail.gmail.com>
	<4D5577CE.1000603@communicatefreely.net>
	<AANLkTinJK+e9hZJbk4ur_OQwGV5B7tRLrpPMH=TU5MOo@mail.gmail.com>
	<4D59C085.9040008@communicatefreely.net>
	<AANLkTikZbWPqf7N1fT-9fF8Fvydi85mJVZ+o8_J=RLjE@mail.gmail.com>
	<4D5A9BEA.7070302@communicatefreely.net>
	<AANLkTikuozcozSiFFL8-6+s3iRP4Lfy9DT2xFi9TVBOd@mail.gmail.com>
Message-ID: <16242449-AADE-4068-8523-C1727F66FDF8@freeswitch.org>

Are you setting your expires to > 300 seconds?

/b

On Feb 15, 2011, at 9:42 AM, Aloysius Lloyd wrote:

> 
> 
> I have the same issue, around 275 phones in the field. I want the 275 phones work with FreeSWITCH.
> 
> 
> Thanks
> Lloyd

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From anthony.minessale at gmail.com  Tue Feb 15 19:09:54 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 15 Feb 2011 10:09:54 -0600
Subject: [Freeswitch-users] ESL Outbound: how to bridge media?
In-Reply-To: <AANLkTinhkOznPh98kkHp=rCU9Qm0xv_hW3c41rRJr=Kn@mail.gmail.com>
References: <AANLkTikLVLpSGkrRqeD3WVFZZgSaLnQvypCOG39hEhLX@mail.gmail.com>
	<AANLkTimwX8Uz4t9gp6Ppzfu+WE+UJa3FO4qHE7XKFOBz@mail.gmail.com>
	<AANLkTi=t-a7U9ov3wJ6Dh-PuniSfZbyeQ==BCAvMkUPN@mail.gmail.com>
	<AANLkTik27F7JjyMRLKryQZq_N+oVDsDzdmiPPa+d2KWN@mail.gmail.com>
	<AANLkTine=jG-dc98pDySashEvo3T5cSPwVXYwp54mhhC@mail.gmail.com>
	<AANLkTi=JAt=RkQ5b=DBajH_YR6TmiqExdjgF1=OL=fJ_@mail.gmail.com>
	<AANLkTinhkOznPh98kkHp=rCU9Qm0xv_hW3c41rRJr=Kn@mail.gmail.com>
Message-ID: <AANLkTikAJoP7_cmhoEDorq4uCTdw+SVQeMNMpcFBkdYn@mail.gmail.com>

Well you can't bridge 2 unanswered channels together.
You need at least one of them to have been pre_answered and that
should be the first uuid in the list.


On Tue, Feb 15, 2011 at 3:48 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:
> I have tried to wait originate completition but the "uuid_bridge" doesn't
> works also in this case.
> It seems that works only when at least one leg is answered so it's not
> possible to do the "uuid_bridge" during progressing phase also if the
> originate has ended (I don't set the ignore_early_media).
> My application is this:?http://pastebin.freeswitch.org/15387
> The application:
> 1. receive a call with an "inbound_uuid"
> 2. create a new "outbound_uuid"
> 3. do a "bgapi originate" using the new "outbound_uuid"
> 4. when the called phone is ringing, receive a "CHANNEL_PROGRESS_MEDIA"
> event on "outbound_uuid" and do a "pre_answer" on "inbound_uuid"
> 5. receive a?"CHANNEL_PROGRESS_MEDIA" on "inbound_uuid"
> 6. receive a job completition event related to the "originate" so issue an
> "uuid_bridge inbound_uuid outbound_uuid"
> 7. when a job completition event related to "uuid_bridge" is received, no
> audio flow from outbound to inbound channel
> 8. when outbound answer the call, the application answer also the inbound
> call but no audio flow in both directions
> If I do the uuid_bridge after legB answer, then all is ok (obviously with no
> audio during progressing phase).
>
> The log of my application is:
> [ERROR] newacme.cpp:46 mycallback() Connected!
> [ERROR] newacme.cpp:65 mycallback() Inbound uuid =
> 8b2c39db-1ad9-489c-b72f-a92d4087bf99
> [ERROR] newacme.cpp:68 mycallback() create_uuid:
> 394167aa-2811-4fcd-95c9-85576bdd9a7a
> [ERROR] newacme.cpp:89 mycallback() bgapi originate
> [origination_uuid=394167aa-2811-4fcd-95c9-85576bdd9a7a]sofia/external/xxxx at yyyy
> &park()
> [ERROR] newacme.cpp:91 mycallback() Job-ID:
> 1c654ac2-c22d-418f-9fad-bb6b6b35aeff
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_UUID]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_OUTGOING]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_ORIGINATE]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_STATE]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_CALLSTATE]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_STATE]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_STATE]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CALL_UPDATE]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] - [CODEC]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] - [CODEC]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_CALLSTATE]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_PROGRESS_MEDIA]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_STATE]
> [ERROR] newacme.cpp:112 mycallback() Got an event: [BACKGROUND_JOB] -
> [1c654ac2-c22d-418f-9fad-bb6b6b35aeff]
> [ERROR] newacme.cpp:119 mycallback() bgapi uuid_bridge
> 8b2c39db-1ad9-489c-b72f-a92d4087bf99 394167aa-2811-4fcd-95c9-85576bdd9a7a
> [ERROR] newacme.cpp:123 mycallback() Job-ID:
> 0d43cffe-fdc6-4a13-97e6-aeee199c45d5
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_EXECUTE]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_PARK]
> [ERROR] newacme.cpp:112 mycallback() Got an event: [BACKGROUND_JOB] -
> [0d43cffe-fdc6-4a13-97e6-aeee199c45d5]
> [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> [CHANNEL_EXECUTE]
> [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> [CHANNEL_CALLSTATE]
> [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> [CHANNEL_PROGRESS_MEDIA]
> [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> [CHANNEL_EXECUTE_COMPLETE]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CALL_UPDATE]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_CALLSTATE]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_ANSWER]
> [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> [CHANNEL_EXECUTE]
> [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> [CHANNEL_CALLSTATE]
> [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> [CHANNEL_ANSWER]
> [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> [CHANNEL_EXECUTE_COMPLETE]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_CALLSTATE]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_HANGUP]
> [ERROR] newacme.cpp:160 mycallback() hangup
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_UNPARK]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_EXECUTE_COMPLETE]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_STATE]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_STATE]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_HANGUP_COMPLETE]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_DESTROY]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_CALLSTATE]
> [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_STATE]
> [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> [CHANNEL_EXECUTE]
> [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> [CHANNEL_CALLSTATE]
> [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> [CHANNEL_HANGUP]
> [ERROR] newacme.cpp:171 mycallback() hangup
> [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> [CHANNEL_EXECUTE_COMPLETE]
> [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> [CHANNEL_UNPARK]
> [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> [CHANNEL_EXECUTE_COMPLETE]
> [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> [CHANNEL_STATE]
> [ERROR] newacme.cpp:184 mycallback() End.
>
> Stephen
>
> On Tue, Feb 15, 2011 at 1:14 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> if you do originate without ignore_early_media=true set it will end
>> the soonest possible where it's suitable for a bridge.
>> so that is the best bet to wait for originate to end.
>>
>>
>> On Mon, Feb 14, 2011 at 6:11 PM, Stephen Wilde <wstephen80 at gmail.com>
>> wrote:
>> > Ok, now I'm doing bridge when I receive "CHANNEL_ANSWER" or
>> > "CHANNEL_PROGRESS_MEDIA" on outbound leg but you say that I have wait
>> > that
>> > "originate has ended": you mean that I have to wait for "BACKGROUND_JOB"
>> > event related to my "bgapi originate ... &park"?
>> > I'm already doing "uuid_bridge <inbound_uuid> <outbound_uuid>".
>> > I'll try also with intercept and inline originate.?Thank you!
>> > Stephen
>> >
>> > On Tue, Feb 15, 2011 at 12:53 AM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> you need to make sure originate has ended on the outbound leg before
>> >> you use it in a bridge etc.
>> >> you also need to supply the inbound leg first in uuid_bridge if that
>> >> is something you want to do.
>> >>
>> >> Easier would be to originate the B leg to park inline and tell A leg
>> >> to execute intercept on the B leg uuid.
>> >>
>> >>
>> >>
>> >> On Mon, Feb 14, 2011 at 5:44 PM, Stephen Wilde <wstephen80 at gmail.com>
>> >> wrote:
>> >> > Sorry but I have missed something.
>> >> > I know that I can solve this problem directly in dialplan with a
>> >> > bridge
>> >> > but
>> >> > what I'm doing is an "Freeswitch ESL learning"?because I have to port
>> >> > some
>> >> > application in Freeswitch and I'm learning how to implement some
>> >> > functionality.
>> >> > For me it's important to take control of both inbound/outbound in
>> >> > full
>> >> > async
>> >> > way and I have the necessity to do the complete call control.
>> >> > I'm not sure but to me it seems that with a normal bridge I lose the
>> >> > control
>> >> > of two sessions, for example, an outbound answer is propagated by
>> >> > bridge
>> >> > application as inbound answer.
>> >> > What I want to do is an audio bridging so my application can take
>> >> > control of
>> >> > "signaling bridging".
>> >> >
>> >> > I'm wrong? There are other way to do that?
>> >> > Stephen
>> >> > On Tue, Feb 15, 2011 at 12:19 AM, Michael Collins
>> >> > <msc at freeswitch.org>
>> >> > wrote:
>> >> >>
>> >> >> My head is spinning after reading this email. :)
>> >> >> It sounds like you just need a simple bridge from the incoming leg
>> >> >> to
>> >> >> the
>> >> >> outgoing leg. Can you pre_answer the A leg then execute a good
>> >> >> old-fashioned
>> >> >> bridge to the b-leg?
>> >> >> -MC
>> >> >> On Mon, Feb 14, 2011 at 4:57 PM, Stephen Wilde
>> >> >> <wstephen80 at gmail.com>
>> >> >> wrote:
>> >> >>>
>> >> >>> The problem I have is to propagate the audio coming from an
>> >> >>> "originated"
>> >> >>> outbound session to the inbound session when the outbound is in the
>> >> >>> PROGRESS
>> >> >>> MEDIA phase.
>> >> >>> When my application receives the "CHANNEL_PROGRESS_MEDIA" event
>> >> >>> from
>> >> >>> outbound session I can do a "pre_answer" on inbound session but I'm
>> >> >>> not
>> >> >>> capable to do an audio bridge.
>> >> >>> I have tried with "uuid_bridge <outbound_uuid> <inbound_uuid>" with
>> >> >>> no
>> >> >>> result probably because this api requires that at least one session
>> >> >>> must be
>> >> >>> answered.
>> >> >>> I don't want to answer to the inbound session to propagate the
>> >> >>> outbound
>> >> >>> progressing media but I want to answer to inbound only on outbound
>> >> >>> answer.
>> >> >>> Any way to do that?
>> >> >>> 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
>> >> >>
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at 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 Feb 15 19:29:21 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 15 Feb 2011 17:29:21 +0100
Subject: [Freeswitch-users] ESL Outbound: how to bridge media?
In-Reply-To: <AANLkTikAJoP7_cmhoEDorq4uCTdw+SVQeMNMpcFBkdYn@mail.gmail.com>
References: <AANLkTikLVLpSGkrRqeD3WVFZZgSaLnQvypCOG39hEhLX@mail.gmail.com>
	<AANLkTimwX8Uz4t9gp6Ppzfu+WE+UJa3FO4qHE7XKFOBz@mail.gmail.com>
	<AANLkTi=t-a7U9ov3wJ6Dh-PuniSfZbyeQ==BCAvMkUPN@mail.gmail.com>
	<AANLkTik27F7JjyMRLKryQZq_N+oVDsDzdmiPPa+d2KWN@mail.gmail.com>
	<AANLkTine=jG-dc98pDySashEvo3T5cSPwVXYwp54mhhC@mail.gmail.com>
	<AANLkTi=JAt=RkQ5b=DBajH_YR6TmiqExdjgF1=OL=fJ_@mail.gmail.com>
	<AANLkTinhkOznPh98kkHp=rCU9Qm0xv_hW3c41rRJr=Kn@mail.gmail.com>
	<AANLkTikAJoP7_cmhoEDorq4uCTdw+SVQeMNMpcFBkdYn@mail.gmail.com>
Message-ID: <AANLkTi=XbutFHsC1w_n3zG+pRW6gtmVC=+YPH0cvK=eA@mail.gmail.com>

Ok, I have tried with "pre_answer" to the inbound channel (that will be the
first uuid in the "uuid_bridge") but with same result: no audio from
outbound (in progressing media state) to inbound (in progressing media state
due to my pre_answer).

Here the sequence of events:

[ERROR] newacme.cpp:193 mycallback() Connected!
[ERROR] newacme.cpp:113 Originate() bgapi originate
[origination_uuid=61e00b94-f4d3-4ac4-bcf5-8fba02a18de6]sofia/external/xxxx at yyyy&park()
[ERROR] newacme.cpp:115 Originate() Job-ID:
eee79d97-dbf8-49d4-9bf9-578ef75f73ea
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_UUID]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_OUTGOING]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_ORIGINATE]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_STATE]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_CALLSTATE]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_STATE]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_STATE]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CALL_UPDATE]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] - [CODEC]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] - [CODEC]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_CALLSTATE]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_PROGRESS_MEDIA]
[ERROR] newacme.cpp:250 mycallback() Executing 'pre_answer' on INBOUND
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_STATE]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_EXECUTE]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_PARK]
[ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
[CHANNEL_EXECUTE]
[ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
[CHANNEL_CALLSTATE]
[ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
[CHANNEL_PROGRESS_MEDIA]
[ERROR] newacme.cpp:127 BridgeTo() bgapi uuid_bridge
902e2cd4-10ee-4938-97cf-c9eb2ac969f8 61e00b94-f4d3-4ac4-bcf5-8fba02a18de6
[ERROR] newacme.cpp:129 BridgeTo() Job-ID:
7a66a5c7-721c-47f4-aff2-373ec20123c8
[ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
[CHANNEL_EXECUTE_COMPLETE]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CALL_UPDATE]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_CALLSTATE]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_ANSWER]
[ERROR] newacme.cpp:255 mycallback() Executing 'answer' on INBOUND
[ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
[CHANNEL_EXECUTE]
[ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
[CHANNEL_CALLSTATE]
[ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
[CHANNEL_ANSWER]
[ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
[CHANNEL_EXECUTE_COMPLETE]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_CALLSTATE]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_HANGUP]
[ERROR] newacme.cpp:260 mycallback() Executing 'hangup' on INBOUND
[ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
[CHANNEL_EXECUTE]
[ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
[CHANNEL_CALLSTATE]
[ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
[CHANNEL_HANGUP]
[ERROR] newacme.cpp:272 mycallback() Executing 'hangup' on OUTBOUND
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_UNPARK]
[ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
[CHANNEL_EXECUTE_COMPLETE]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_EXECUTE_COMPLETE]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_STATE]
[ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
[CHANNEL_UNPARK]
[ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
[CHANNEL_EXECUTE_COMPLETE]
[ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
[CHANNEL_STATE]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_STATE]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_HANGUP_COMPLETE]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_DESTROY]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_CALLSTATE]
[ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
[CHANNEL_STATE]
[ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
[CHANNEL_STATE]
[ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
[CHANNEL_HANGUP_COMPLETE]
[ERROR] newacme.cpp:290 mycallback() End.


On Tue, Feb 15, 2011 at 5:09 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Well you can't bridge 2 unanswered channels together.
> You need at least one of them to have been pre_answered and that
> should be the first uuid in the list.
>
>
> On Tue, Feb 15, 2011 at 3:48 AM, Stephen Wilde <wstephen80 at gmail.com>
> wrote:
> > I have tried to wait originate completition but the "uuid_bridge" doesn't
> > works also in this case.
> > It seems that works only when at least one leg is answered so it's not
> > possible to do the "uuid_bridge" during progressing phase also if the
> > originate has ended (I don't set the ignore_early_media).
> > My application is this: http://pastebin.freeswitch.org/15387
> > The application:
> > 1. receive a call with an "inbound_uuid"
> > 2. create a new "outbound_uuid"
> > 3. do a "bgapi originate" using the new "outbound_uuid"
> > 4. when the called phone is ringing, receive a "CHANNEL_PROGRESS_MEDIA"
> > event on "outbound_uuid" and do a "pre_answer" on "inbound_uuid"
> > 5. receive a "CHANNEL_PROGRESS_MEDIA" on "inbound_uuid"
> > 6. receive a job completition event related to the "originate" so issue
> an
> > "uuid_bridge inbound_uuid outbound_uuid"
> > 7. when a job completition event related to "uuid_bridge" is received, no
> > audio flow from outbound to inbound channel
> > 8. when outbound answer the call, the application answer also the inbound
> > call but no audio flow in both directions
> > If I do the uuid_bridge after legB answer, then all is ok (obviously with
> no
> > audio during progressing phase).
> >
> > The log of my application is:
> > [ERROR] newacme.cpp:46 mycallback() Connected!
> > [ERROR] newacme.cpp:65 mycallback() Inbound uuid =
> > 8b2c39db-1ad9-489c-b72f-a92d4087bf99
> > [ERROR] newacme.cpp:68 mycallback() create_uuid:
> > 394167aa-2811-4fcd-95c9-85576bdd9a7a
> > [ERROR] newacme.cpp:89 mycallback() bgapi originate
> >
> [origination_uuid=394167aa-2811-4fcd-95c9-85576bdd9a7a]sofia/external/xxxx at yyyy
> > &park()
> > [ERROR] newacme.cpp:91 mycallback() Job-ID:
> > 1c654ac2-c22d-418f-9fad-bb6b6b35aeff
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_UUID]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_OUTGOING]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_ORIGINATE]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_STATE]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_CALLSTATE]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_STATE]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_STATE]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CALL_UPDATE]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] - [CODEC]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] - [CODEC]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_CALLSTATE]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_PROGRESS_MEDIA]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_STATE]
> > [ERROR] newacme.cpp:112 mycallback() Got an event: [BACKGROUND_JOB] -
> > [1c654ac2-c22d-418f-9fad-bb6b6b35aeff]
> > [ERROR] newacme.cpp:119 mycallback() bgapi uuid_bridge
> > 8b2c39db-1ad9-489c-b72f-a92d4087bf99 394167aa-2811-4fcd-95c9-85576bdd9a7a
> > [ERROR] newacme.cpp:123 mycallback() Job-ID:
> > 0d43cffe-fdc6-4a13-97e6-aeee199c45d5
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_EXECUTE]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_PARK]
> > [ERROR] newacme.cpp:112 mycallback() Got an event: [BACKGROUND_JOB] -
> > [0d43cffe-fdc6-4a13-97e6-aeee199c45d5]
> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_EXECUTE]
> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_CALLSTATE]
> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_PROGRESS_MEDIA]
> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_EXECUTE_COMPLETE]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CALL_UPDATE]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_CALLSTATE]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_ANSWER]
> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_EXECUTE]
> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_CALLSTATE]
> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_ANSWER]
> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_EXECUTE_COMPLETE]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_CALLSTATE]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_HANGUP]
> > [ERROR] newacme.cpp:160 mycallback() hangup
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_UNPARK]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_EXECUTE_COMPLETE]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_STATE]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_STATE]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_HANGUP_COMPLETE]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_DESTROY]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_CALLSTATE]
> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_STATE]
> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_EXECUTE]
> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_CALLSTATE]
> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_HANGUP]
> > [ERROR] newacme.cpp:171 mycallback() hangup
> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_EXECUTE_COMPLETE]
> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_UNPARK]
> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_EXECUTE_COMPLETE]
> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_STATE]
> > [ERROR] newacme.cpp:184 mycallback() End.
> >
> > Stephen
> >
> > On Tue, Feb 15, 2011 at 1:14 AM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> if you do originate without ignore_early_media=true set it will end
> >> the soonest possible where it's suitable for a bridge.
> >> so that is the best bet to wait for originate to end.
> >>
> >>
> >> On Mon, Feb 14, 2011 at 6:11 PM, Stephen Wilde <wstephen80 at gmail.com>
> >> wrote:
> >> > Ok, now I'm doing bridge when I receive "CHANNEL_ANSWER" or
> >> > "CHANNEL_PROGRESS_MEDIA" on outbound leg but you say that I have wait
> >> > that
> >> > "originate has ended": you mean that I have to wait for
> "BACKGROUND_JOB"
> >> > event related to my "bgapi originate ... &park"?
> >> > I'm already doing "uuid_bridge <inbound_uuid> <outbound_uuid>".
> >> > I'll try also with intercept and inline originate. Thank you!
> >> > Stephen
> >> >
> >> > On Tue, Feb 15, 2011 at 12:53 AM, Anthony Minessale
> >> > <anthony.minessale at gmail.com> wrote:
> >> >>
> >> >> you need to make sure originate has ended on the outbound leg before
> >> >> you use it in a bridge etc.
> >> >> you also need to supply the inbound leg first in uuid_bridge if that
> >> >> is something you want to do.
> >> >>
> >> >> Easier would be to originate the B leg to park inline and tell A leg
> >> >> to execute intercept on the B leg uuid.
> >> >>
> >> >>
> >> >>
> >> >> On Mon, Feb 14, 2011 at 5:44 PM, Stephen Wilde <wstephen80 at gmail.com
> >
> >> >> wrote:
> >> >> > Sorry but I have missed something.
> >> >> > I know that I can solve this problem directly in dialplan with a
> >> >> > bridge
> >> >> > but
> >> >> > what I'm doing is an "Freeswitch ESL learning" because I have to
> port
> >> >> > some
> >> >> > application in Freeswitch and I'm learning how to implement some
> >> >> > functionality.
> >> >> > For me it's important to take control of both inbound/outbound in
> >> >> > full
> >> >> > async
> >> >> > way and I have the necessity to do the complete call control.
> >> >> > I'm not sure but to me it seems that with a normal bridge I lose
> the
> >> >> > control
> >> >> > of two sessions, for example, an outbound answer is propagated by
> >> >> > bridge
> >> >> > application as inbound answer.
> >> >> > What I want to do is an audio bridging so my application can take
> >> >> > control of
> >> >> > "signaling bridging".
> >> >> >
> >> >> > I'm wrong? There are other way to do that?
> >> >> > Stephen
> >> >> > On Tue, Feb 15, 2011 at 12:19 AM, Michael Collins
> >> >> > <msc at freeswitch.org>
> >> >> > wrote:
> >> >> >>
> >> >> >> My head is spinning after reading this email. :)
> >> >> >> It sounds like you just need a simple bridge from the incoming leg
> >> >> >> to
> >> >> >> the
> >> >> >> outgoing leg. Can you pre_answer the A leg then execute a good
> >> >> >> old-fashioned
> >> >> >> bridge to the b-leg?
> >> >> >> -MC
> >> >> >> On Mon, Feb 14, 2011 at 4:57 PM, Stephen Wilde
> >> >> >> <wstephen80 at gmail.com>
> >> >> >> wrote:
> >> >> >>>
> >> >> >>> The problem I have is to propagate the audio coming from an
> >> >> >>> "originated"
> >> >> >>> outbound session to the inbound session when the outbound is in
> the
> >> >> >>> PROGRESS
> >> >> >>> MEDIA phase.
> >> >> >>> When my application receives the "CHANNEL_PROGRESS_MEDIA" event
> >> >> >>> from
> >> >> >>> outbound session I can do a "pre_answer" on inbound session but
> I'm
> >> >> >>> not
> >> >> >>> capable to do an audio bridge.
> >> >> >>> I have tried with "uuid_bridge <outbound_uuid> <inbound_uuid>"
> with
> >> >> >>> no
> >> >> >>> result probably because this api requires that at least one
> session
> >> >> >>> must be
> >> >> >>> answered.
> >> >> >>> I don't want to answer to the inbound session to propagate the
> >> >> >>> outbound
> >> >> >>> progressing media but I want to answer to inbound only on
> outbound
> >> >> >>> answer.
> >> >> >>> Any way to do that?
> >> >> >>> 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
> >> >> >>
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at 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 tim.compnetwork at gmail.com  Tue Feb 15 20:02:26 2011
From: tim.compnetwork at gmail.com (Tim King)
Date: Tue, 15 Feb 2011 12:02:26 -0500
Subject: [Freeswitch-users] Alarm Conditions
Message-ID: <AANLkTimhirHNT=S52=8hAgCtQvT+snzNovMRskpD=Gvf@mail.gmail.com>

I am looking for a way to gather the information in the following list so I
may utilize it to set alarm conditions.  Can someone please let me know a
good method for gathering this information or if perhaps there is a newer
module for reporting alarms that I am not aware of. Thanks - Tim

?         CDR Failed to write/send

?         Call Failure

?         H323 Gatekeeper registration error

?         Number of Active calls

?         Active Calls based on trunk

?         No route for call

?         SIP Error

      Codec Translation error
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From steveayre at gmail.com  Tue Feb 15 20:12:37 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 15 Feb 2011 17:12:37 +0000
Subject: [Freeswitch-users] Alarm Conditions
In-Reply-To: <AANLkTimhirHNT=S52=8hAgCtQvT+snzNovMRskpD=Gvf@mail.gmail.com>
References: <AANLkTimhirHNT=S52=8hAgCtQvT+snzNovMRskpD=Gvf@mail.gmail.com>
Message-ID: <AANLkTi=w9qnsM0WPHBws_R_JaW9O+K7HF3=5kSo60gWG@mail.gmail.com>

- CDR Failed to write/send
Using mod_xml_cdr I assume? If they fail to send to the web server they'll
save to the err-log-dir if it's set. They shouldn't fail to save there - if
they do (e.g. missing path or permissions) there'll probably be something in
the logfile. If there's no space to save the CDR you have bigger problems
since there'll also be nowhere to log to.

- Call Failure
CDRs

- H323 Gatekeeper registration error
Not sure, but it probably puts something in the log files

- Number of Active calls
ESL + show calls (or ODBC if you put the core db in odbc)

- Active Calls based on trunk
Limit + ODBC/ESL

- No route for call
Handled in dialplan, probably via CDRs

- SIP Error
CDRs
- Codec Translation error
Can you clarify what type of errors? G729 logs in the logfile, no matching
codec will generate a 488 unless the call has already been answered in which
case it'll hangup. CDRs can show the 488s.


-Steve


On 15 February 2011 17:02, Tim King <tim.compnetwork at gmail.com> wrote:

> I am looking for a way to gather the information in the following list so I
> may utilize it to set alarm conditions.  Can someone please let me know a
> good method for gathering this information or if perhaps there is a newer
> module for reporting alarms that I am not aware of. Thanks - Tim
>
> ?         CDR Failed to write/send
>
> ?         Call Failure
>
> ?         H323 Gatekeeper registration error
>
> ?         Number of Active calls
>
> ?         Active Calls based on trunk
>
> ?         No route for call
>
> ?         SIP Error
>
>       Codec Translation error
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 15 20:15:08 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 15 Feb 2011 11:15:08 -0600
Subject: [Freeswitch-users] ESL Outbound: how to bridge media?
In-Reply-To: <AANLkTi=XbutFHsC1w_n3zG+pRW6gtmVC=+YPH0cvK=eA@mail.gmail.com>
References: <AANLkTikLVLpSGkrRqeD3WVFZZgSaLnQvypCOG39hEhLX@mail.gmail.com>
	<AANLkTimwX8Uz4t9gp6Ppzfu+WE+UJa3FO4qHE7XKFOBz@mail.gmail.com>
	<AANLkTi=t-a7U9ov3wJ6Dh-PuniSfZbyeQ==BCAvMkUPN@mail.gmail.com>
	<AANLkTik27F7JjyMRLKryQZq_N+oVDsDzdmiPPa+d2KWN@mail.gmail.com>
	<AANLkTine=jG-dc98pDySashEvo3T5cSPwVXYwp54mhhC@mail.gmail.com>
	<AANLkTi=JAt=RkQ5b=DBajH_YR6TmiqExdjgF1=OL=fJ_@mail.gmail.com>
	<AANLkTinhkOznPh98kkHp=rCU9Qm0xv_hW3c41rRJr=Kn@mail.gmail.com>
	<AANLkTikAJoP7_cmhoEDorq4uCTdw+SVQeMNMpcFBkdYn@mail.gmail.com>
	<AANLkTi=XbutFHsC1w_n3zG+pRW6gtmVC=+YPH0cvK=eA@mail.gmail.com>
Message-ID: <AANLkTimMDkdWi_V8TDXrxzn9paLEsthTGaOXMGD2_-PN@mail.gmail.com>

What happens once they answer?

This is not one of those attempts to speed up click to call by calling
them both at once is it?


On Tue, Feb 15, 2011 at 10:29 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:
> Ok, I have tried with "pre_answer" to the inbound channel (that will be the
> first uuid in the "uuid_bridge")?but with same result: no audio from
> outbound (in progressing media state) to inbound (in progressing media state
> due to my pre_answer).
> Here the sequence of events:
> [ERROR] newacme.cpp:193 mycallback() Connected!
> [ERROR] newacme.cpp:113 Originate() bgapi originate
> [origination_uuid=61e00b94-f4d3-4ac4-bcf5-8fba02a18de6]sofia/external/xxxx at yyyy
> &park()
> [ERROR] newacme.cpp:115 Originate() Job-ID:
> eee79d97-dbf8-49d4-9bf9-578ef75f73ea
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_UUID]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_OUTGOING]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_ORIGINATE]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_STATE]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_CALLSTATE]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_STATE]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_STATE]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CALL_UPDATE]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] - [CODEC]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] - [CODEC]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_CALLSTATE]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_PROGRESS_MEDIA]
> [ERROR] newacme.cpp:250 mycallback() Executing 'pre_answer' on INBOUND
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_STATE]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_EXECUTE]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_PARK]
> [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> [CHANNEL_EXECUTE]
> [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> [CHANNEL_CALLSTATE]
> [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> [CHANNEL_PROGRESS_MEDIA]
> [ERROR] newacme.cpp:127 BridgeTo() bgapi uuid_bridge
> 902e2cd4-10ee-4938-97cf-c9eb2ac969f8 61e00b94-f4d3-4ac4-bcf5-8fba02a18de6
> [ERROR] newacme.cpp:129 BridgeTo() Job-ID:
> 7a66a5c7-721c-47f4-aff2-373ec20123c8
> [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> [CHANNEL_EXECUTE_COMPLETE]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CALL_UPDATE]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_CALLSTATE]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_ANSWER]
> [ERROR] newacme.cpp:255 mycallback() Executing 'answer' on INBOUND
> [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> [CHANNEL_EXECUTE]
> [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> [CHANNEL_CALLSTATE]
> [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> [CHANNEL_ANSWER]
> [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> [CHANNEL_EXECUTE_COMPLETE]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_CALLSTATE]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_HANGUP]
> [ERROR] newacme.cpp:260 mycallback() Executing 'hangup' on INBOUND
> [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> [CHANNEL_EXECUTE]
> [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> [CHANNEL_CALLSTATE]
> [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> [CHANNEL_HANGUP]
> [ERROR] newacme.cpp:272 mycallback() Executing 'hangup' on OUTBOUND
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_UNPARK]
> [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> [CHANNEL_EXECUTE_COMPLETE]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_EXECUTE_COMPLETE]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_STATE]
> [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> [CHANNEL_UNPARK]
> [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> [CHANNEL_EXECUTE_COMPLETE]
> [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> [CHANNEL_STATE]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_STATE]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_HANGUP_COMPLETE]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_DESTROY]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_CALLSTATE]
> [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CHANNEL_STATE]
> [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> [CHANNEL_STATE]
> [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> [CHANNEL_HANGUP_COMPLETE]
> [ERROR] newacme.cpp:290 mycallback() End.
>
>
> On Tue, Feb 15, 2011 at 5:09 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> Well you can't bridge 2 unanswered channels together.
>> You need at least one of them to have been pre_answered and that
>> should be the first uuid in the list.
>>
>>
>> On Tue, Feb 15, 2011 at 3:48 AM, Stephen Wilde <wstephen80 at gmail.com>
>> wrote:
>> > I have tried to wait originate completition but the "uuid_bridge"
>> > doesn't
>> > works also in this case.
>> > It seems that works only when at least one leg is answered so it's not
>> > possible to do the "uuid_bridge" during progressing phase also if the
>> > originate has ended (I don't set the ignore_early_media).
>> > My application is this:?http://pastebin.freeswitch.org/15387
>> > The application:
>> > 1. receive a call with an "inbound_uuid"
>> > 2. create a new "outbound_uuid"
>> > 3. do a "bgapi originate" using the new "outbound_uuid"
>> > 4. when the called phone is ringing, receive a "CHANNEL_PROGRESS_MEDIA"
>> > event on "outbound_uuid" and do a "pre_answer" on "inbound_uuid"
>> > 5. receive a?"CHANNEL_PROGRESS_MEDIA" on "inbound_uuid"
>> > 6. receive a job completition event related to the "originate" so issue
>> > an
>> > "uuid_bridge inbound_uuid outbound_uuid"
>> > 7. when a job completition event related to "uuid_bridge" is received,
>> > no
>> > audio flow from outbound to inbound channel
>> > 8. when outbound answer the call, the application answer also the
>> > inbound
>> > call but no audio flow in both directions
>> > If I do the uuid_bridge after legB answer, then all is ok (obviously
>> > with no
>> > audio during progressing phase).
>> >
>> > The log of my application is:
>> > [ERROR] newacme.cpp:46 mycallback() Connected!
>> > [ERROR] newacme.cpp:65 mycallback() Inbound uuid =
>> > 8b2c39db-1ad9-489c-b72f-a92d4087bf99
>> > [ERROR] newacme.cpp:68 mycallback() create_uuid:
>> > 394167aa-2811-4fcd-95c9-85576bdd9a7a
>> > [ERROR] newacme.cpp:89 mycallback() bgapi originate
>> >
>> > [origination_uuid=394167aa-2811-4fcd-95c9-85576bdd9a7a]sofia/external/xxxx at yyyy
>> > &park()
>> > [ERROR] newacme.cpp:91 mycallback() Job-ID:
>> > 1c654ac2-c22d-418f-9fad-bb6b6b35aeff
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_UUID]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_OUTGOING]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_ORIGINATE]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_STATE]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_CALLSTATE]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_STATE]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_STATE]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CALL_UPDATE]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] - [CODEC]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] - [CODEC]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_CALLSTATE]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_PROGRESS_MEDIA]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_STATE]
>> > [ERROR] newacme.cpp:112 mycallback() Got an event: [BACKGROUND_JOB] -
>> > [1c654ac2-c22d-418f-9fad-bb6b6b35aeff]
>> > [ERROR] newacme.cpp:119 mycallback() bgapi uuid_bridge
>> > 8b2c39db-1ad9-489c-b72f-a92d4087bf99
>> > 394167aa-2811-4fcd-95c9-85576bdd9a7a
>> > [ERROR] newacme.cpp:123 mycallback() Job-ID:
>> > 0d43cffe-fdc6-4a13-97e6-aeee199c45d5
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_EXECUTE]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_PARK]
>> > [ERROR] newacme.cpp:112 mycallback() Got an event: [BACKGROUND_JOB] -
>> > [0d43cffe-fdc6-4a13-97e6-aeee199c45d5]
>> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_EXECUTE]
>> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_CALLSTATE]
>> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_PROGRESS_MEDIA]
>> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_EXECUTE_COMPLETE]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CALL_UPDATE]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_CALLSTATE]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_ANSWER]
>> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_EXECUTE]
>> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_CALLSTATE]
>> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_ANSWER]
>> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_EXECUTE_COMPLETE]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_CALLSTATE]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_HANGUP]
>> > [ERROR] newacme.cpp:160 mycallback() hangup
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_UNPARK]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_EXECUTE_COMPLETE]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_STATE]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_STATE]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_HANGUP_COMPLETE]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_DESTROY]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_CALLSTATE]
>> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_STATE]
>> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_EXECUTE]
>> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_CALLSTATE]
>> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_HANGUP]
>> > [ERROR] newacme.cpp:171 mycallback() hangup
>> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_EXECUTE_COMPLETE]
>> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_UNPARK]
>> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_EXECUTE_COMPLETE]
>> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_STATE]
>> > [ERROR] newacme.cpp:184 mycallback() End.
>> >
>> > Stephen
>> >
>> > On Tue, Feb 15, 2011 at 1:14 AM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> if you do originate without ignore_early_media=true set it will end
>> >> the soonest possible where it's suitable for a bridge.
>> >> so that is the best bet to wait for originate to end.
>> >>
>> >>
>> >> On Mon, Feb 14, 2011 at 6:11 PM, Stephen Wilde <wstephen80 at gmail.com>
>> >> wrote:
>> >> > Ok, now I'm doing bridge when I receive "CHANNEL_ANSWER" or
>> >> > "CHANNEL_PROGRESS_MEDIA" on outbound leg but you say that I have wait
>> >> > that
>> >> > "originate has ended": you mean that I have to wait for
>> >> > "BACKGROUND_JOB"
>> >> > event related to my "bgapi originate ... &park"?
>> >> > I'm already doing "uuid_bridge <inbound_uuid> <outbound_uuid>".
>> >> > I'll try also with intercept and inline originate.?Thank you!
>> >> > Stephen
>> >> >
>> >> > On Tue, Feb 15, 2011 at 12:53 AM, Anthony Minessale
>> >> > <anthony.minessale at gmail.com> wrote:
>> >> >>
>> >> >> you need to make sure originate has ended on the outbound leg before
>> >> >> you use it in a bridge etc.
>> >> >> you also need to supply the inbound leg first in uuid_bridge if that
>> >> >> is something you want to do.
>> >> >>
>> >> >> Easier would be to originate the B leg to park inline and tell A leg
>> >> >> to execute intercept on the B leg uuid.
>> >> >>
>> >> >>
>> >> >>
>> >> >> On Mon, Feb 14, 2011 at 5:44 PM, Stephen Wilde
>> >> >> <wstephen80 at gmail.com>
>> >> >> wrote:
>> >> >> > Sorry but I have missed something.
>> >> >> > I know that I can solve this problem directly in dialplan with a
>> >> >> > bridge
>> >> >> > but
>> >> >> > what I'm doing is an "Freeswitch ESL learning"?because I have to
>> >> >> > port
>> >> >> > some
>> >> >> > application in Freeswitch and I'm learning how to implement some
>> >> >> > functionality.
>> >> >> > For me it's important to take control of both inbound/outbound in
>> >> >> > full
>> >> >> > async
>> >> >> > way and I have the necessity to do the complete call control.
>> >> >> > I'm not sure but to me it seems that with a normal bridge I lose
>> >> >> > the
>> >> >> > control
>> >> >> > of two sessions, for example, an outbound answer is propagated by
>> >> >> > bridge
>> >> >> > application as inbound answer.
>> >> >> > What I want to do is an audio bridging so my application can take
>> >> >> > control of
>> >> >> > "signaling bridging".
>> >> >> >
>> >> >> > I'm wrong? There are other way to do that?
>> >> >> > Stephen
>> >> >> > On Tue, Feb 15, 2011 at 12:19 AM, Michael Collins
>> >> >> > <msc at freeswitch.org>
>> >> >> > wrote:
>> >> >> >>
>> >> >> >> My head is spinning after reading this email. :)
>> >> >> >> It sounds like you just need a simple bridge from the incoming
>> >> >> >> leg
>> >> >> >> to
>> >> >> >> the
>> >> >> >> outgoing leg. Can you pre_answer the A leg then execute a good
>> >> >> >> old-fashioned
>> >> >> >> bridge to the b-leg?
>> >> >> >> -MC
>> >> >> >> On Mon, Feb 14, 2011 at 4:57 PM, Stephen Wilde
>> >> >> >> <wstephen80 at gmail.com>
>> >> >> >> wrote:
>> >> >> >>>
>> >> >> >>> The problem I have is to propagate the audio coming from an
>> >> >> >>> "originated"
>> >> >> >>> outbound session to the inbound session when the outbound is in
>> >> >> >>> the
>> >> >> >>> PROGRESS
>> >> >> >>> MEDIA phase.
>> >> >> >>> When my application receives the "CHANNEL_PROGRESS_MEDIA" event
>> >> >> >>> from
>> >> >> >>> outbound session I can do a "pre_answer" on inbound session but
>> >> >> >>> I'm
>> >> >> >>> not
>> >> >> >>> capable to do an audio bridge.
>> >> >> >>> I have tried with "uuid_bridge <outbound_uuid> <inbound_uuid>"
>> >> >> >>> with
>> >> >> >>> no
>> >> >> >>> result probably because this api requires that at least one
>> >> >> >>> session
>> >> >> >>> must be
>> >> >> >>> answered.
>> >> >> >>> I don't want to answer to the inbound session to propagate the
>> >> >> >>> outbound
>> >> >> >>> progressing media but I want to answer to inbound only on
>> >> >> >>> outbound
>> >> >> >>> answer.
>> >> >> >>> Any way to do that?
>> >> >> >>> 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
>> >> >> >>
>> >> >> >
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at 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 jeff at jefflenk.com  Tue Feb 15 20:37:50 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 15 Feb 2011 09:37:50 -0800 (PST)
Subject: [Freeswitch-users] Possible to communicate with FS using
	event_socket without ESL?
In-Reply-To: <003101cbcb2d$fa0e9e60$ee2bdb20$@net>
References: <003101cbcb2d$fa0e9e60$ee2bdb20$@net>
Message-ID: <1297791470877-6028546.post@n2.nabble.com>


If your using .Net use this from the source tree -
libs\esl\managed\managed_esl*

works great and uses the esl lib as its base
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Possible-to-communicate-with-FS-using-event-socket-without-ESL-tp6020307p6028546.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Tue Feb 15 20:56:53 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 15 Feb 2011 11:56:53 -0600
Subject: [Freeswitch-users] Getting SIP trace on user level
In-Reply-To: <AANLkTi=+yL16rqZs9ig0R4JUfsfFOtmfr1s9jDY=-Gpf@mail.gmail.com>
References: <AANLkTimhW139WdDKXiQkRm43Ai13f_7Z+N9Z-K2y8Ldi@mail.gmail.com>
	<AANLkTikxBuc960_-sZvXQXdXAC5LzmiNvagj9RUC2GPx@mail.gmail.com>
	<AANLkTi=+yL16rqZs9ig0R4JUfsfFOtmfr1s9jDY=-Gpf@mail.gmail.com>
Message-ID: <AANLkTi=YNTzEYUiwaW2bZN9YRTr+VURWVgSz4xsw_gdn@mail.gmail.com>

FYI,
If you know the IP address of the phone you can also use tcpdump:

tcpdump -nq -s 0 -w myfile.pcap host 1.2.3.4 and port 5060

-MC

On Tue, Feb 15, 2011 at 7:47 AM, Steven Ayre <steveayre at gmail.com> wrote:

> tshark can also do the same (part of wireshark)
>
> -Steve
>
>
>
> On 15 February 2011 13:14, Kristian Kielhofner <kris at kriskinc.com> wrote:
>
>> In FreeSWITCH: not that I know of.
>>
>> In Linux, yes: http://ngrep.sourceforge.net/
>>
>> Use ngrep and match on anything, including IP and port combination
>> from a specific peer (that's all Asterisk does anyway).  It would be
>> trivial to throw a script together to pull out the sofia contact for a
>> user and pass the IP:port to ngrep for filtering.
>>
>> You'll find ngrep to be very handy.
>>
>> On Mon, Feb 14, 2011 at 11:06 PM, Aloysius Lloyd
>> <lloyd.aloysius at gmail.com> wrote:
>> > Hi All,
>> > Is there any console command available to get SIP trance on user level .
>> > Equivalent to Asterisk "sip set debug peer <peername>"
>> >
>> >
>> > TIA
>> > 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
>> >
>> >
>>
>>
>>
>> --
>> 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
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jack at hotglass.cc  Tue Feb 15 20:55:50 2011
From: jack at hotglass.cc (Jack Loranger)
Date: Tue, 15 Feb 2011 09:55:50 -0800
Subject: [Freeswitch-users] Fail2ban for Windows?
Message-ID: <4D5ABE26.5040304@hotglass.cc>

Is there a Fail2ban type solution for windows?
Thanks,
  Jack
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From wstephen80 at gmail.com  Tue Feb 15 21:08:58 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 15 Feb 2011 19:08:58 +0100
Subject: [Freeswitch-users] ESL Outbound: how to bridge media?
In-Reply-To: <AANLkTimMDkdWi_V8TDXrxzn9paLEsthTGaOXMGD2_-PN@mail.gmail.com>
References: <AANLkTikLVLpSGkrRqeD3WVFZZgSaLnQvypCOG39hEhLX@mail.gmail.com>
	<AANLkTimwX8Uz4t9gp6Ppzfu+WE+UJa3FO4qHE7XKFOBz@mail.gmail.com>
	<AANLkTi=t-a7U9ov3wJ6Dh-PuniSfZbyeQ==BCAvMkUPN@mail.gmail.com>
	<AANLkTik27F7JjyMRLKryQZq_N+oVDsDzdmiPPa+d2KWN@mail.gmail.com>
	<AANLkTine=jG-dc98pDySashEvo3T5cSPwVXYwp54mhhC@mail.gmail.com>
	<AANLkTi=JAt=RkQ5b=DBajH_YR6TmiqExdjgF1=OL=fJ_@mail.gmail.com>
	<AANLkTinhkOznPh98kkHp=rCU9Qm0xv_hW3c41rRJr=Kn@mail.gmail.com>
	<AANLkTikAJoP7_cmhoEDorq4uCTdw+SVQeMNMpcFBkdYn@mail.gmail.com>
	<AANLkTi=XbutFHsC1w_n3zG+pRW6gtmVC=+YPH0cvK=eA@mail.gmail.com>
	<AANLkTimMDkdWi_V8TDXrxzn9paLEsthTGaOXMGD2_-PN@mail.gmail.com>
Message-ID: <AANLkTikghfSv1pu-V8-qKB1C6Oc-JFipv2Wm=eWwJfgr@mail.gmail.com>

LegA is an incoming call, legB is an outgoing call.

The dialplan is:

    <extension name="myhandler">
      <condition field="destination_number" expression="^xxxx$">
        <action application="set" data="dialstring=sofia/external/yyyy at zzzz/>
        <action application="socket" data="127.0.0.1:8084 async full"/>
      </condition>
    </extension>

When the legB answer the call, I receive the "CHANNEL_ANSWER" event on legB
and I send the "answer" to legA (I have already done a uuid_bridge when
originate has ended).
The audio continue to be mute in both direction.

The "uuid_bridge" works only if I call it after received "CHANNEL_ANSWER" in
legB or after I send "answer" to legA.

It's not enough to do a "pre_answer".

Stephen

On Tue, Feb 15, 2011 at 6:15 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> What happens once they answer?
>
> This is not one of those attempts to speed up click to call by calling
> them both at once is it?
>
>
> On Tue, Feb 15, 2011 at 10:29 AM, Stephen Wilde <wstephen80 at gmail.com>
> wrote:
> > Ok, I have tried with "pre_answer" to the inbound channel (that will be
> the
> > first uuid in the "uuid_bridge") but with same result: no audio from
> > outbound (in progressing media state) to inbound (in progressing media
> state
> > due to my pre_answer).
> > Here the sequence of events:
> > [ERROR] newacme.cpp:193 mycallback() Connected!
> > [ERROR] newacme.cpp:113 Originate() bgapi originate
> >
> [origination_uuid=61e00b94-f4d3-4ac4-bcf5-8fba02a18de6]sofia/external/xxxx at yyyy
> > &park()
> > [ERROR] newacme.cpp:115 Originate() Job-ID:
> > eee79d97-dbf8-49d4-9bf9-578ef75f73ea
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_UUID]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_OUTGOING]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_ORIGINATE]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_STATE]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_CALLSTATE]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_STATE]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_STATE]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CALL_UPDATE]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] - [CODEC]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] - [CODEC]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_CALLSTATE]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_PROGRESS_MEDIA]
> > [ERROR] newacme.cpp:250 mycallback() Executing 'pre_answer' on INBOUND
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_STATE]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_EXECUTE]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_PARK]
> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_EXECUTE]
> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_CALLSTATE]
> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_PROGRESS_MEDIA]
> > [ERROR] newacme.cpp:127 BridgeTo() bgapi uuid_bridge
> > 902e2cd4-10ee-4938-97cf-c9eb2ac969f8 61e00b94-f4d3-4ac4-bcf5-8fba02a18de6
> > [ERROR] newacme.cpp:129 BridgeTo() Job-ID:
> > 7a66a5c7-721c-47f4-aff2-373ec20123c8
> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_EXECUTE_COMPLETE]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CALL_UPDATE]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_CALLSTATE]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_ANSWER]
> > [ERROR] newacme.cpp:255 mycallback() Executing 'answer' on INBOUND
> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_EXECUTE]
> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_CALLSTATE]
> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_ANSWER]
> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_EXECUTE_COMPLETE]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_CALLSTATE]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_HANGUP]
> > [ERROR] newacme.cpp:260 mycallback() Executing 'hangup' on INBOUND
> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_EXECUTE]
> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_CALLSTATE]
> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_HANGUP]
> > [ERROR] newacme.cpp:272 mycallback() Executing 'hangup' on OUTBOUND
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_UNPARK]
> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_EXECUTE_COMPLETE]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_EXECUTE_COMPLETE]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_STATE]
> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_UNPARK]
> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_EXECUTE_COMPLETE]
> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_STATE]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_STATE]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_HANGUP_COMPLETE]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_DESTROY]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_CALLSTATE]
> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> > [CHANNEL_STATE]
> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_STATE]
> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> > [CHANNEL_HANGUP_COMPLETE]
> > [ERROR] newacme.cpp:290 mycallback() End.
> >
> >
> > On Tue, Feb 15, 2011 at 5:09 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> Well you can't bridge 2 unanswered channels together.
> >> You need at least one of them to have been pre_answered and that
> >> should be the first uuid in the list.
> >>
> >>
> >> On Tue, Feb 15, 2011 at 3:48 AM, Stephen Wilde <wstephen80 at gmail.com>
> >> wrote:
> >> > I have tried to wait originate completition but the "uuid_bridge"
> >> > doesn't
> >> > works also in this case.
> >> > It seems that works only when at least one leg is answered so it's not
> >> > possible to do the "uuid_bridge" during progressing phase also if the
> >> > originate has ended (I don't set the ignore_early_media).
> >> > My application is this: http://pastebin.freeswitch.org/15387
> >> > The application:
> >> > 1. receive a call with an "inbound_uuid"
> >> > 2. create a new "outbound_uuid"
> >> > 3. do a "bgapi originate" using the new "outbound_uuid"
> >> > 4. when the called phone is ringing, receive a
> "CHANNEL_PROGRESS_MEDIA"
> >> > event on "outbound_uuid" and do a "pre_answer" on "inbound_uuid"
> >> > 5. receive a "CHANNEL_PROGRESS_MEDIA" on "inbound_uuid"
> >> > 6. receive a job completition event related to the "originate" so
> issue
> >> > an
> >> > "uuid_bridge inbound_uuid outbound_uuid"
> >> > 7. when a job completition event related to "uuid_bridge" is received,
> >> > no
> >> > audio flow from outbound to inbound channel
> >> > 8. when outbound answer the call, the application answer also the
> >> > inbound
> >> > call but no audio flow in both directions
> >> > If I do the uuid_bridge after legB answer, then all is ok (obviously
> >> > with no
> >> > audio during progressing phase).
> >> >
> >> > The log of my application is:
> >> > [ERROR] newacme.cpp:46 mycallback() Connected!
> >> > [ERROR] newacme.cpp:65 mycallback() Inbound uuid =
> >> > 8b2c39db-1ad9-489c-b72f-a92d4087bf99
> >> > [ERROR] newacme.cpp:68 mycallback() create_uuid:
> >> > 394167aa-2811-4fcd-95c9-85576bdd9a7a
> >> > [ERROR] newacme.cpp:89 mycallback() bgapi originate
> >> >
> >> >
> [origination_uuid=394167aa-2811-4fcd-95c9-85576bdd9a7a]sofia/external/xxxx at yyyy
> >> > &park()
> >> > [ERROR] newacme.cpp:91 mycallback() Job-ID:
> >> > 1c654ac2-c22d-418f-9fad-bb6b6b35aeff
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_UUID]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_OUTGOING]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_ORIGINATE]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_STATE]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_CALLSTATE]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_STATE]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_STATE]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CALL_UPDATE]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CODEC]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> [CODEC]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_CALLSTATE]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_PROGRESS_MEDIA]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_STATE]
> >> > [ERROR] newacme.cpp:112 mycallback() Got an event: [BACKGROUND_JOB] -
> >> > [1c654ac2-c22d-418f-9fad-bb6b6b35aeff]
> >> > [ERROR] newacme.cpp:119 mycallback() bgapi uuid_bridge
> >> > 8b2c39db-1ad9-489c-b72f-a92d4087bf99
> >> > 394167aa-2811-4fcd-95c9-85576bdd9a7a
> >> > [ERROR] newacme.cpp:123 mycallback() Job-ID:
> >> > 0d43cffe-fdc6-4a13-97e6-aeee199c45d5
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_EXECUTE]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_PARK]
> >> > [ERROR] newacme.cpp:112 mycallback() Got an event: [BACKGROUND_JOB] -
> >> > [0d43cffe-fdc6-4a13-97e6-aeee199c45d5]
> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_EXECUTE]
> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_CALLSTATE]
> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_PROGRESS_MEDIA]
> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CALL_UPDATE]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_CALLSTATE]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_ANSWER]
> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_EXECUTE]
> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_CALLSTATE]
> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_ANSWER]
> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_CALLSTATE]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_HANGUP]
> >> > [ERROR] newacme.cpp:160 mycallback() hangup
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_UNPARK]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_STATE]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_STATE]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_HANGUP_COMPLETE]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_DESTROY]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_CALLSTATE]
> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_STATE]
> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_EXECUTE]
> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_CALLSTATE]
> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_HANGUP]
> >> > [ERROR] newacme.cpp:171 mycallback() hangup
> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_UNPARK]
> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_STATE]
> >> > [ERROR] newacme.cpp:184 mycallback() End.
> >> >
> >> > Stephen
> >> >
> >> > On Tue, Feb 15, 2011 at 1:14 AM, Anthony Minessale
> >> > <anthony.minessale at gmail.com> wrote:
> >> >>
> >> >> if you do originate without ignore_early_media=true set it will end
> >> >> the soonest possible where it's suitable for a bridge.
> >> >> so that is the best bet to wait for originate to end.
> >> >>
> >> >>
> >> >> On Mon, Feb 14, 2011 at 6:11 PM, Stephen Wilde <wstephen80 at gmail.com
> >
> >> >> wrote:
> >> >> > Ok, now I'm doing bridge when I receive "CHANNEL_ANSWER" or
> >> >> > "CHANNEL_PROGRESS_MEDIA" on outbound leg but you say that I have
> wait
> >> >> > that
> >> >> > "originate has ended": you mean that I have to wait for
> >> >> > "BACKGROUND_JOB"
> >> >> > event related to my "bgapi originate ... &park"?
> >> >> > I'm already doing "uuid_bridge <inbound_uuid> <outbound_uuid>".
> >> >> > I'll try also with intercept and inline originate. Thank you!
> >> >> > Stephen
> >> >> >
> >> >> > On Tue, Feb 15, 2011 at 12:53 AM, Anthony Minessale
> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >>
> >> >> >> you need to make sure originate has ended on the outbound leg
> before
> >> >> >> you use it in a bridge etc.
> >> >> >> you also need to supply the inbound leg first in uuid_bridge if
> that
> >> >> >> is something you want to do.
> >> >> >>
> >> >> >> Easier would be to originate the B leg to park inline and tell A
> leg
> >> >> >> to execute intercept on the B leg uuid.
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >> On Mon, Feb 14, 2011 at 5:44 PM, Stephen Wilde
> >> >> >> <wstephen80 at gmail.com>
> >> >> >> wrote:
> >> >> >> > Sorry but I have missed something.
> >> >> >> > I know that I can solve this problem directly in dialplan with a
> >> >> >> > bridge
> >> >> >> > but
> >> >> >> > what I'm doing is an "Freeswitch ESL learning" because I have to
> >> >> >> > port
> >> >> >> > some
> >> >> >> > application in Freeswitch and I'm learning how to implement some
> >> >> >> > functionality.
> >> >> >> > For me it's important to take control of both inbound/outbound
> in
> >> >> >> > full
> >> >> >> > async
> >> >> >> > way and I have the necessity to do the complete call control.
> >> >> >> > I'm not sure but to me it seems that with a normal bridge I lose
> >> >> >> > the
> >> >> >> > control
> >> >> >> > of two sessions, for example, an outbound answer is propagated
> by
> >> >> >> > bridge
> >> >> >> > application as inbound answer.
> >> >> >> > What I want to do is an audio bridging so my application can
> take
> >> >> >> > control of
> >> >> >> > "signaling bridging".
> >> >> >> >
> >> >> >> > I'm wrong? There are other way to do that?
> >> >> >> > Stephen
> >> >> >> > On Tue, Feb 15, 2011 at 12:19 AM, Michael Collins
> >> >> >> > <msc at freeswitch.org>
> >> >> >> > wrote:
> >> >> >> >>
> >> >> >> >> My head is spinning after reading this email. :)
> >> >> >> >> It sounds like you just need a simple bridge from the incoming
> >> >> >> >> leg
> >> >> >> >> to
> >> >> >> >> the
> >> >> >> >> outgoing leg. Can you pre_answer the A leg then execute a good
> >> >> >> >> old-fashioned
> >> >> >> >> bridge to the b-leg?
> >> >> >> >> -MC
> >> >> >> >> On Mon, Feb 14, 2011 at 4:57 PM, Stephen Wilde
> >> >> >> >> <wstephen80 at gmail.com>
> >> >> >> >> wrote:
> >> >> >> >>>
> >> >> >> >>> The problem I have is to propagate the audio coming from an
> >> >> >> >>> "originated"
> >> >> >> >>> outbound session to the inbound session when the outbound is
> in
> >> >> >> >>> the
> >> >> >> >>> PROGRESS
> >> >> >> >>> MEDIA phase.
> >> >> >> >>> When my application receives the "CHANNEL_PROGRESS_MEDIA"
> event
> >> >> >> >>> from
> >> >> >> >>> outbound session I can do a "pre_answer" on inbound session
> but
> >> >> >> >>> I'm
> >> >> >> >>> not
> >> >> >> >>> capable to do an audio bridge.
> >> >> >> >>> I have tried with "uuid_bridge <outbound_uuid> <inbound_uuid>"
> >> >> >> >>> with
> >> >> >> >>> no
> >> >> >> >>> result probably because this api requires that at least one
> >> >> >> >>> session
> >> >> >> >>> must be
> >> >> >> >>> answered.
> >> >> >> >>> I don't want to answer to the inbound session to propagate the
> >> >> >> >>> outbound
> >> >> >> >>> progressing media but I want to answer to inbound only on
> >> >> >> >>> outbound
> >> >> >> >>> answer.
> >> >> >> >>> Any way to do that?
> >> >> >> >>> 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
> >> >> >> >>
> >> >> >> >
> >> >> >> >
> >> >> >> > _______________________________________________
> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> > FreeSWITCH-users at 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 Feb 15 23:53:51 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 15 Feb 2011 14:53:51 -0600
Subject: [Freeswitch-users] ESL Outbound: how to bridge media?
In-Reply-To: <AANLkTikghfSv1pu-V8-qKB1C6Oc-JFipv2Wm=eWwJfgr@mail.gmail.com>
References: <AANLkTikLVLpSGkrRqeD3WVFZZgSaLnQvypCOG39hEhLX@mail.gmail.com>
	<AANLkTimwX8Uz4t9gp6Ppzfu+WE+UJa3FO4qHE7XKFOBz@mail.gmail.com>
	<AANLkTi=t-a7U9ov3wJ6Dh-PuniSfZbyeQ==BCAvMkUPN@mail.gmail.com>
	<AANLkTik27F7JjyMRLKryQZq_N+oVDsDzdmiPPa+d2KWN@mail.gmail.com>
	<AANLkTine=jG-dc98pDySashEvo3T5cSPwVXYwp54mhhC@mail.gmail.com>
	<AANLkTi=JAt=RkQ5b=DBajH_YR6TmiqExdjgF1=OL=fJ_@mail.gmail.com>
	<AANLkTinhkOznPh98kkHp=rCU9Qm0xv_hW3c41rRJr=Kn@mail.gmail.com>
	<AANLkTikAJoP7_cmhoEDorq4uCTdw+SVQeMNMpcFBkdYn@mail.gmail.com>
	<AANLkTi=XbutFHsC1w_n3zG+pRW6gtmVC=+YPH0cvK=eA@mail.gmail.com>
	<AANLkTimMDkdWi_V8TDXrxzn9paLEsthTGaOXMGD2_-PN@mail.gmail.com>
	<AANLkTikghfSv1pu-V8-qKB1C6Oc-JFipv2Wm=eWwJfgr@mail.gmail.com>
Message-ID: <AANLkTi=Sjtsw6KPqDXJmB-F8g-ZcUfz7n62ro7b4jmid@mail.gmail.com>

I meant when you have no audio, and you wait until leg b answers. what
happens then?


If you have leg A inbound and you send it to park.
then you originate outbound to b leg to park
wait for park event on b leg
then you uuid_bridge them

also you can sendmsg with application = intercept app_arg = uuid of B


On Tue, Feb 15, 2011 at 12:08 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:
> LegA is an incoming call, legB is an outgoing call.
> The dialplan is:
> ?? ?<extension name="myhandler">
> ?? ? ?<condition field="destination_number" expression="^xxxx$">
> ?? ? ? ?<action application="set" data="dialstring=sofia/external/yyyy at zzzz
> />
> ?? ? ? ?<action application="socket" data="127.0.0.1:8084 async full"/>
> ?? ? ?</condition>
> ?? ?</extension>
> When the legB answer the call, I receive the "CHANNEL_ANSWER" event on legB
> and I send the "answer" to legA (I have already done a uuid_bridge when
> originate has ended).
> The audio continue to be mute in both direction.
> The "uuid_bridge" works only if I call it after received "CHANNEL_ANSWER" in
> legB or after I send "answer" to legA.
> It's not enough to do a "pre_answer".
> Stephen
>
> On Tue, Feb 15, 2011 at 6:15 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> What happens once they answer?
>>
>> This is not one of those attempts to speed up click to call by calling
>> them both at once is it?
>>
>>
>> On Tue, Feb 15, 2011 at 10:29 AM, Stephen Wilde <wstephen80 at gmail.com>
>> wrote:
>> > Ok, I have tried with "pre_answer" to the inbound channel (that will be
>> > the
>> > first uuid in the "uuid_bridge")?but with same result: no audio from
>> > outbound (in progressing media state) to inbound (in progressing media
>> > state
>> > due to my pre_answer).
>> > Here the sequence of events:
>> > [ERROR] newacme.cpp:193 mycallback() Connected!
>> > [ERROR] newacme.cpp:113 Originate() bgapi originate
>> >
>> > [origination_uuid=61e00b94-f4d3-4ac4-bcf5-8fba02a18de6]sofia/external/xxxx at yyyy
>> > &park()
>> > [ERROR] newacme.cpp:115 Originate() Job-ID:
>> > eee79d97-dbf8-49d4-9bf9-578ef75f73ea
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_UUID]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_OUTGOING]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_ORIGINATE]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_STATE]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_CALLSTATE]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_STATE]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_STATE]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CALL_UPDATE]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] - [CODEC]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] - [CODEC]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_CALLSTATE]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_PROGRESS_MEDIA]
>> > [ERROR] newacme.cpp:250 mycallback() Executing 'pre_answer' on INBOUND
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_STATE]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_EXECUTE]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_PARK]
>> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_EXECUTE]
>> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_CALLSTATE]
>> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_PROGRESS_MEDIA]
>> > [ERROR] newacme.cpp:127 BridgeTo() bgapi uuid_bridge
>> > 902e2cd4-10ee-4938-97cf-c9eb2ac969f8
>> > 61e00b94-f4d3-4ac4-bcf5-8fba02a18de6
>> > [ERROR] newacme.cpp:129 BridgeTo() Job-ID:
>> > 7a66a5c7-721c-47f4-aff2-373ec20123c8
>> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_EXECUTE_COMPLETE]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CALL_UPDATE]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_CALLSTATE]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_ANSWER]
>> > [ERROR] newacme.cpp:255 mycallback() Executing 'answer' on INBOUND
>> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_EXECUTE]
>> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_CALLSTATE]
>> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_ANSWER]
>> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_EXECUTE_COMPLETE]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_CALLSTATE]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_HANGUP]
>> > [ERROR] newacme.cpp:260 mycallback() Executing 'hangup' on INBOUND
>> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_EXECUTE]
>> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_CALLSTATE]
>> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_HANGUP]
>> > [ERROR] newacme.cpp:272 mycallback() Executing 'hangup' on OUTBOUND
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_UNPARK]
>> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_EXECUTE_COMPLETE]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_EXECUTE_COMPLETE]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_STATE]
>> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_UNPARK]
>> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_EXECUTE_COMPLETE]
>> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_STATE]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_STATE]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_HANGUP_COMPLETE]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_DESTROY]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_CALLSTATE]
>> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> > [CHANNEL_STATE]
>> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_STATE]
>> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> > [CHANNEL_HANGUP_COMPLETE]
>> > [ERROR] newacme.cpp:290 mycallback() End.
>> >
>> >
>> > On Tue, Feb 15, 2011 at 5:09 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> Well you can't bridge 2 unanswered channels together.
>> >> You need at least one of them to have been pre_answered and that
>> >> should be the first uuid in the list.
>> >>
>> >>
>> >> On Tue, Feb 15, 2011 at 3:48 AM, Stephen Wilde <wstephen80 at gmail.com>
>> >> wrote:
>> >> > I have tried to wait originate completition but the "uuid_bridge"
>> >> > doesn't
>> >> > works also in this case.
>> >> > It seems that works only when at least one leg is answered so it's
>> >> > not
>> >> > possible to do the "uuid_bridge" during progressing phase also if the
>> >> > originate has ended (I don't set the ignore_early_media).
>> >> > My application is this:?http://pastebin.freeswitch.org/15387
>> >> > The application:
>> >> > 1. receive a call with an "inbound_uuid"
>> >> > 2. create a new "outbound_uuid"
>> >> > 3. do a "bgapi originate" using the new "outbound_uuid"
>> >> > 4. when the called phone is ringing, receive a
>> >> > "CHANNEL_PROGRESS_MEDIA"
>> >> > event on "outbound_uuid" and do a "pre_answer" on "inbound_uuid"
>> >> > 5. receive a?"CHANNEL_PROGRESS_MEDIA" on "inbound_uuid"
>> >> > 6. receive a job completition event related to the "originate" so
>> >> > issue
>> >> > an
>> >> > "uuid_bridge inbound_uuid outbound_uuid"
>> >> > 7. when a job completition event related to "uuid_bridge" is
>> >> > received,
>> >> > no
>> >> > audio flow from outbound to inbound channel
>> >> > 8. when outbound answer the call, the application answer also the
>> >> > inbound
>> >> > call but no audio flow in both directions
>> >> > If I do the uuid_bridge after legB answer, then all is ok (obviously
>> >> > with no
>> >> > audio during progressing phase).
>> >> >
>> >> > The log of my application is:
>> >> > [ERROR] newacme.cpp:46 mycallback() Connected!
>> >> > [ERROR] newacme.cpp:65 mycallback() Inbound uuid =
>> >> > 8b2c39db-1ad9-489c-b72f-a92d4087bf99
>> >> > [ERROR] newacme.cpp:68 mycallback() create_uuid:
>> >> > 394167aa-2811-4fcd-95c9-85576bdd9a7a
>> >> > [ERROR] newacme.cpp:89 mycallback() bgapi originate
>> >> >
>> >> >
>> >> > [origination_uuid=394167aa-2811-4fcd-95c9-85576bdd9a7a]sofia/external/xxxx at yyyy
>> >> > &park()
>> >> > [ERROR] newacme.cpp:91 mycallback() Job-ID:
>> >> > 1c654ac2-c22d-418f-9fad-bb6b6b35aeff
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_UUID]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_OUTGOING]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_ORIGINATE]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_STATE]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_CALLSTATE]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_STATE]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_STATE]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CALL_UPDATE]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CODEC]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CODEC]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_CALLSTATE]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_PROGRESS_MEDIA]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_STATE]
>> >> > [ERROR] newacme.cpp:112 mycallback() Got an event: [BACKGROUND_JOB] -
>> >> > [1c654ac2-c22d-418f-9fad-bb6b6b35aeff]
>> >> > [ERROR] newacme.cpp:119 mycallback() bgapi uuid_bridge
>> >> > 8b2c39db-1ad9-489c-b72f-a92d4087bf99
>> >> > 394167aa-2811-4fcd-95c9-85576bdd9a7a
>> >> > [ERROR] newacme.cpp:123 mycallback() Job-ID:
>> >> > 0d43cffe-fdc6-4a13-97e6-aeee199c45d5
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_EXECUTE]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_PARK]
>> >> > [ERROR] newacme.cpp:112 mycallback() Got an event: [BACKGROUND_JOB] -
>> >> > [0d43cffe-fdc6-4a13-97e6-aeee199c45d5]
>> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_EXECUTE]
>> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_CALLSTATE]
>> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_PROGRESS_MEDIA]
>> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CALL_UPDATE]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_CALLSTATE]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_ANSWER]
>> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_EXECUTE]
>> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_CALLSTATE]
>> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_ANSWER]
>> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_CALLSTATE]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_HANGUP]
>> >> > [ERROR] newacme.cpp:160 mycallback() hangup
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_UNPARK]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_STATE]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_STATE]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_HANGUP_COMPLETE]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_DESTROY]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_CALLSTATE]
>> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_STATE]
>> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_EXECUTE]
>> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_CALLSTATE]
>> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_HANGUP]
>> >> > [ERROR] newacme.cpp:171 mycallback() hangup
>> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_UNPARK]
>> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_STATE]
>> >> > [ERROR] newacme.cpp:184 mycallback() End.
>> >> >
>> >> > Stephen
>> >> >
>> >> > On Tue, Feb 15, 2011 at 1:14 AM, Anthony Minessale
>> >> > <anthony.minessale at gmail.com> wrote:
>> >> >>
>> >> >> if you do originate without ignore_early_media=true set it will end
>> >> >> the soonest possible where it's suitable for a bridge.
>> >> >> so that is the best bet to wait for originate to end.
>> >> >>
>> >> >>
>> >> >> On Mon, Feb 14, 2011 at 6:11 PM, Stephen Wilde
>> >> >> <wstephen80 at gmail.com>
>> >> >> wrote:
>> >> >> > Ok, now I'm doing bridge when I receive "CHANNEL_ANSWER" or
>> >> >> > "CHANNEL_PROGRESS_MEDIA" on outbound leg but you say that I have
>> >> >> > wait
>> >> >> > that
>> >> >> > "originate has ended": you mean that I have to wait for
>> >> >> > "BACKGROUND_JOB"
>> >> >> > event related to my "bgapi originate ... &park"?
>> >> >> > I'm already doing "uuid_bridge <inbound_uuid> <outbound_uuid>".
>> >> >> > I'll try also with intercept and inline originate.?Thank you!
>> >> >> > Stephen
>> >> >> >
>> >> >> > On Tue, Feb 15, 2011 at 12:53 AM, Anthony Minessale
>> >> >> > <anthony.minessale at gmail.com> wrote:
>> >> >> >>
>> >> >> >> you need to make sure originate has ended on the outbound leg
>> >> >> >> before
>> >> >> >> you use it in a bridge etc.
>> >> >> >> you also need to supply the inbound leg first in uuid_bridge if
>> >> >> >> that
>> >> >> >> is something you want to do.
>> >> >> >>
>> >> >> >> Easier would be to originate the B leg to park inline and tell A
>> >> >> >> leg
>> >> >> >> to execute intercept on the B leg uuid.
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> On Mon, Feb 14, 2011 at 5:44 PM, Stephen Wilde
>> >> >> >> <wstephen80 at gmail.com>
>> >> >> >> wrote:
>> >> >> >> > Sorry but I have missed something.
>> >> >> >> > I know that I can solve this problem directly in dialplan with
>> >> >> >> > a
>> >> >> >> > bridge
>> >> >> >> > but
>> >> >> >> > what I'm doing is an "Freeswitch ESL learning"?because I have
>> >> >> >> > to
>> >> >> >> > port
>> >> >> >> > some
>> >> >> >> > application in Freeswitch and I'm learning how to implement
>> >> >> >> > some
>> >> >> >> > functionality.
>> >> >> >> > For me it's important to take control of both inbound/outbound
>> >> >> >> > in
>> >> >> >> > full
>> >> >> >> > async
>> >> >> >> > way and I have the necessity to do the complete call control.
>> >> >> >> > I'm not sure but to me it seems that with a normal bridge I
>> >> >> >> > lose
>> >> >> >> > the
>> >> >> >> > control
>> >> >> >> > of two sessions, for example, an outbound answer is propagated
>> >> >> >> > by
>> >> >> >> > bridge
>> >> >> >> > application as inbound answer.
>> >> >> >> > What I want to do is an audio bridging so my application can
>> >> >> >> > take
>> >> >> >> > control of
>> >> >> >> > "signaling bridging".
>> >> >> >> >
>> >> >> >> > I'm wrong? There are other way to do that?
>> >> >> >> > Stephen
>> >> >> >> > On Tue, Feb 15, 2011 at 12:19 AM, Michael Collins
>> >> >> >> > <msc at freeswitch.org>
>> >> >> >> > wrote:
>> >> >> >> >>
>> >> >> >> >> My head is spinning after reading this email. :)
>> >> >> >> >> It sounds like you just need a simple bridge from the incoming
>> >> >> >> >> leg
>> >> >> >> >> to
>> >> >> >> >> the
>> >> >> >> >> outgoing leg. Can you pre_answer the A leg then execute a good
>> >> >> >> >> old-fashioned
>> >> >> >> >> bridge to the b-leg?
>> >> >> >> >> -MC
>> >> >> >> >> On Mon, Feb 14, 2011 at 4:57 PM, Stephen Wilde
>> >> >> >> >> <wstephen80 at gmail.com>
>> >> >> >> >> wrote:
>> >> >> >> >>>
>> >> >> >> >>> The problem I have is to propagate the audio coming from an
>> >> >> >> >>> "originated"
>> >> >> >> >>> outbound session to the inbound session when the outbound is
>> >> >> >> >>> in
>> >> >> >> >>> the
>> >> >> >> >>> PROGRESS
>> >> >> >> >>> MEDIA phase.
>> >> >> >> >>> When my application receives the "CHANNEL_PROGRESS_MEDIA"
>> >> >> >> >>> event
>> >> >> >> >>> from
>> >> >> >> >>> outbound session I can do a "pre_answer" on inbound session
>> >> >> >> >>> but
>> >> >> >> >>> I'm
>> >> >> >> >>> not
>> >> >> >> >>> capable to do an audio bridge.
>> >> >> >> >>> I have tried with "uuid_bridge <outbound_uuid>
>> >> >> >> >>> <inbound_uuid>"
>> >> >> >> >>> with
>> >> >> >> >>> no
>> >> >> >> >>> result probably because this api requires that at least one
>> >> >> >> >>> session
>> >> >> >> >>> must be
>> >> >> >> >>> answered.
>> >> >> >> >>> I don't want to answer to the inbound session to propagate
>> >> >> >> >>> the
>> >> >> >> >>> outbound
>> >> >> >> >>> progressing media but I want to answer to inbound only on
>> >> >> >> >>> outbound
>> >> >> >> >>> answer.
>> >> >> >> >>> Any way to do that?
>> >> >> >> >>> 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
>> >> >> >> >>
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > _______________________________________________
>> >> >> >> > FreeSWITCH-users mailing list
>> >> >> >> > FreeSWITCH-users at 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/
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>> >> >> >>
>> >> >> >> AIM: anthm
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>> >> >> >>
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>> >> >>
>> >> >> --
>> >> >> Anthony Minessale II
>> >> >>
>> >> >> FreeSWITCH http://www.freeswitch.org/
>> >> >> ClueCon http://www.cluecon.com/
>> >> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >>
>> >> >> AIM: anthm
>> >> >> MSN:anthony_minessale at hotmail.com
>> >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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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>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
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>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> IRC: irc.freenode.net #freeswitch
>> >>
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>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
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>>
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>
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>



-- 
Anthony Minessale II

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

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MSN:anthony_minessale at hotmail.com
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FreeSWITCH Developer Conference
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googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From wstephen80 at gmail.com  Wed Feb 16 00:46:36 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 15 Feb 2011 22:46:36 +0100
Subject: [Freeswitch-users] ESL Outbound: how to bridge media?
In-Reply-To: <AANLkTi=Sjtsw6KPqDXJmB-F8g-ZcUfz7n62ro7b4jmid@mail.gmail.com>
References: <AANLkTikLVLpSGkrRqeD3WVFZZgSaLnQvypCOG39hEhLX@mail.gmail.com>
	<AANLkTimwX8Uz4t9gp6Ppzfu+WE+UJa3FO4qHE7XKFOBz@mail.gmail.com>
	<AANLkTi=t-a7U9ov3wJ6Dh-PuniSfZbyeQ==BCAvMkUPN@mail.gmail.com>
	<AANLkTik27F7JjyMRLKryQZq_N+oVDsDzdmiPPa+d2KWN@mail.gmail.com>
	<AANLkTine=jG-dc98pDySashEvo3T5cSPwVXYwp54mhhC@mail.gmail.com>
	<AANLkTi=JAt=RkQ5b=DBajH_YR6TmiqExdjgF1=OL=fJ_@mail.gmail.com>
	<AANLkTinhkOznPh98kkHp=rCU9Qm0xv_hW3c41rRJr=Kn@mail.gmail.com>
	<AANLkTikAJoP7_cmhoEDorq4uCTdw+SVQeMNMpcFBkdYn@mail.gmail.com>
	<AANLkTi=XbutFHsC1w_n3zG+pRW6gtmVC=+YPH0cvK=eA@mail.gmail.com>
	<AANLkTimMDkdWi_V8TDXrxzn9paLEsthTGaOXMGD2_-PN@mail.gmail.com>
	<AANLkTikghfSv1pu-V8-qKB1C6Oc-JFipv2Wm=eWwJfgr@mail.gmail.com>
	<AANLkTi=Sjtsw6KPqDXJmB-F8g-ZcUfz7n62ro7b4jmid@mail.gmail.com>
Message-ID: <AANLkTim22-SPAnWp6OttQt7AcUVBzRS=z1aiDvsgsD3j@mail.gmail.com>

When legB answers there is also no audio.

If I do the bridge after legB answers the call (instead of when
"origination" has ended), then the audio is ok and the legA can ear the legB
also if legA is in "pre_answer" state.

Now I have found one problem: the "BACKGROUND_JOB" event related to "bgapi
uuid_bridge ..." contains: "-ERR Invalid uuid" but both uuid's are correct
so probably is due to the state of channels.

With the same uuid's, the uuid_bridge after the legB answer works fine.

Stephen


On Tue, Feb 15, 2011 at 9:53 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I meant when you have no audio, and you wait until leg b answers. what
> happens then?
>
>
> If you have leg A inbound and you send it to park.
> then you originate outbound to b leg to park
> wait for park event on b leg
> then you uuid_bridge them
>
> also you can sendmsg with application = intercept app_arg = uuid of B
>
>
> On Tue, Feb 15, 2011 at 12:08 PM, Stephen Wilde <wstephen80 at gmail.com>
> wrote:
> > LegA is an incoming call, legB is an outgoing call.
> > The dialplan is:
> >     <extension name="myhandler">
> >       <condition field="destination_number" expression="^xxxx$">
> >         <action application="set"
> data="dialstring=sofia/external/yyyy at zzzz
> > />
> >         <action application="socket" data="127.0.0.1:8084 async full"/>
> >       </condition>
> >     </extension>
> > When the legB answer the call, I receive the "CHANNEL_ANSWER" event on
> legB
> > and I send the "answer" to legA (I have already done a uuid_bridge when
> > originate has ended).
> > The audio continue to be mute in both direction.
> > The "uuid_bridge" works only if I call it after received "CHANNEL_ANSWER"
> in
> > legB or after I send "answer" to legA.
> > It's not enough to do a "pre_answer".
> > Stephen
> >
> > On Tue, Feb 15, 2011 at 6:15 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> What happens once they answer?
> >>
> >> This is not one of those attempts to speed up click to call by calling
> >> them both at once is it?
> >>
> >>
> >> On Tue, Feb 15, 2011 at 10:29 AM, Stephen Wilde <wstephen80 at gmail.com>
> >> wrote:
> >> > Ok, I have tried with "pre_answer" to the inbound channel (that will
> be
> >> > the
> >> > first uuid in the "uuid_bridge") but with same result: no audio from
> >> > outbound (in progressing media state) to inbound (in progressing media
> >> > state
> >> > due to my pre_answer).
> >> > Here the sequence of events:
> >> > [ERROR] newacme.cpp:193 mycallback() Connected!
> >> > [ERROR] newacme.cpp:113 Originate() bgapi originate
> >> >
> >> >
> [origination_uuid=61e00b94-f4d3-4ac4-bcf5-8fba02a18de6]sofia/external/xxxx at yyyy
> >> > &park()
> >> > [ERROR] newacme.cpp:115 Originate() Job-ID:
> >> > eee79d97-dbf8-49d4-9bf9-578ef75f73ea
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_UUID]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_OUTGOING]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_ORIGINATE]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_STATE]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_CALLSTATE]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_STATE]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_STATE]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CALL_UPDATE]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CODEC]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> [CODEC]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_CALLSTATE]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_PROGRESS_MEDIA]
> >> > [ERROR] newacme.cpp:250 mycallback() Executing 'pre_answer' on INBOUND
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_STATE]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_EXECUTE]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_PARK]
> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_EXECUTE]
> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_CALLSTATE]
> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_PROGRESS_MEDIA]
> >> > [ERROR] newacme.cpp:127 BridgeTo() bgapi uuid_bridge
> >> > 902e2cd4-10ee-4938-97cf-c9eb2ac969f8
> >> > 61e00b94-f4d3-4ac4-bcf5-8fba02a18de6
> >> > [ERROR] newacme.cpp:129 BridgeTo() Job-ID:
> >> > 7a66a5c7-721c-47f4-aff2-373ec20123c8
> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CALL_UPDATE]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_CALLSTATE]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_ANSWER]
> >> > [ERROR] newacme.cpp:255 mycallback() Executing 'answer' on INBOUND
> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_EXECUTE]
> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_CALLSTATE]
> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_ANSWER]
> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_CALLSTATE]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_HANGUP]
> >> > [ERROR] newacme.cpp:260 mycallback() Executing 'hangup' on INBOUND
> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_EXECUTE]
> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_CALLSTATE]
> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_HANGUP]
> >> > [ERROR] newacme.cpp:272 mycallback() Executing 'hangup' on OUTBOUND
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_UNPARK]
> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_STATE]
> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_UNPARK]
> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_STATE]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_STATE]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_HANGUP_COMPLETE]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_DESTROY]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_CALLSTATE]
> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> > [CHANNEL_STATE]
> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_STATE]
> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> > [CHANNEL_HANGUP_COMPLETE]
> >> > [ERROR] newacme.cpp:290 mycallback() End.
> >> >
> >> >
> >> > On Tue, Feb 15, 2011 at 5:09 PM, Anthony Minessale
> >> > <anthony.minessale at gmail.com> wrote:
> >> >>
> >> >> Well you can't bridge 2 unanswered channels together.
> >> >> You need at least one of them to have been pre_answered and that
> >> >> should be the first uuid in the list.
> >> >>
> >> >>
> >> >> On Tue, Feb 15, 2011 at 3:48 AM, Stephen Wilde <wstephen80 at gmail.com
> >
> >> >> wrote:
> >> >> > I have tried to wait originate completition but the "uuid_bridge"
> >> >> > doesn't
> >> >> > works also in this case.
> >> >> > It seems that works only when at least one leg is answered so it's
> >> >> > not
> >> >> > possible to do the "uuid_bridge" during progressing phase also if
> the
> >> >> > originate has ended (I don't set the ignore_early_media).
> >> >> > My application is this: http://pastebin.freeswitch.org/15387
> >> >> > The application:
> >> >> > 1. receive a call with an "inbound_uuid"
> >> >> > 2. create a new "outbound_uuid"
> >> >> > 3. do a "bgapi originate" using the new "outbound_uuid"
> >> >> > 4. when the called phone is ringing, receive a
> >> >> > "CHANNEL_PROGRESS_MEDIA"
> >> >> > event on "outbound_uuid" and do a "pre_answer" on "inbound_uuid"
> >> >> > 5. receive a "CHANNEL_PROGRESS_MEDIA" on "inbound_uuid"
> >> >> > 6. receive a job completition event related to the "originate" so
> >> >> > issue
> >> >> > an
> >> >> > "uuid_bridge inbound_uuid outbound_uuid"
> >> >> > 7. when a job completition event related to "uuid_bridge" is
> >> >> > received,
> >> >> > no
> >> >> > audio flow from outbound to inbound channel
> >> >> > 8. when outbound answer the call, the application answer also the
> >> >> > inbound
> >> >> > call but no audio flow in both directions
> >> >> > If I do the uuid_bridge after legB answer, then all is ok
> (obviously
> >> >> > with no
> >> >> > audio during progressing phase).
> >> >> >
> >> >> > The log of my application is:
> >> >> > [ERROR] newacme.cpp:46 mycallback() Connected!
> >> >> > [ERROR] newacme.cpp:65 mycallback() Inbound uuid =
> >> >> > 8b2c39db-1ad9-489c-b72f-a92d4087bf99
> >> >> > [ERROR] newacme.cpp:68 mycallback() create_uuid:
> >> >> > 394167aa-2811-4fcd-95c9-85576bdd9a7a
> >> >> > [ERROR] newacme.cpp:89 mycallback() bgapi originate
> >> >> >
> >> >> >
> >> >> >
> [origination_uuid=394167aa-2811-4fcd-95c9-85576bdd9a7a]sofia/external/xxxx at yyyy
> >> >> > &park()
> >> >> > [ERROR] newacme.cpp:91 mycallback() Job-ID:
> >> >> > 1c654ac2-c22d-418f-9fad-bb6b6b35aeff
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_UUID]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_OUTGOING]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_ORIGINATE]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_STATE]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_CALLSTATE]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_STATE]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_STATE]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CALL_UPDATE]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CODEC]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CODEC]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_CALLSTATE]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_PROGRESS_MEDIA]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_STATE]
> >> >> > [ERROR] newacme.cpp:112 mycallback() Got an event: [BACKGROUND_JOB]
> -
> >> >> > [1c654ac2-c22d-418f-9fad-bb6b6b35aeff]
> >> >> > [ERROR] newacme.cpp:119 mycallback() bgapi uuid_bridge
> >> >> > 8b2c39db-1ad9-489c-b72f-a92d4087bf99
> >> >> > 394167aa-2811-4fcd-95c9-85576bdd9a7a
> >> >> > [ERROR] newacme.cpp:123 mycallback() Job-ID:
> >> >> > 0d43cffe-fdc6-4a13-97e6-aeee199c45d5
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_EXECUTE]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_PARK]
> >> >> > [ERROR] newacme.cpp:112 mycallback() Got an event: [BACKGROUND_JOB]
> -
> >> >> > [0d43cffe-fdc6-4a13-97e6-aeee199c45d5]
> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_EXECUTE]
> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_CALLSTATE]
> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_PROGRESS_MEDIA]
> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CALL_UPDATE]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_CALLSTATE]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_ANSWER]
> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_EXECUTE]
> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_CALLSTATE]
> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_ANSWER]
> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_CALLSTATE]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_HANGUP]
> >> >> > [ERROR] newacme.cpp:160 mycallback() hangup
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_UNPARK]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_STATE]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_STATE]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_HANGUP_COMPLETE]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_DESTROY]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_CALLSTATE]
> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_STATE]
> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_EXECUTE]
> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_CALLSTATE]
> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_HANGUP]
> >> >> > [ERROR] newacme.cpp:171 mycallback() hangup
> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_UNPARK]
> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_STATE]
> >> >> > [ERROR] newacme.cpp:184 mycallback() End.
> >> >> >
> >> >> > Stephen
> >> >> >
> >> >> > On Tue, Feb 15, 2011 at 1:14 AM, Anthony Minessale
> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >>
> >> >> >> if you do originate without ignore_early_media=true set it will
> end
> >> >> >> the soonest possible where it's suitable for a bridge.
> >> >> >> so that is the best bet to wait for originate to end.
> >> >> >>
> >> >> >>
> >> >> >> On Mon, Feb 14, 2011 at 6:11 PM, Stephen Wilde
> >> >> >> <wstephen80 at gmail.com>
> >> >> >> wrote:
> >> >> >> > Ok, now I'm doing bridge when I receive "CHANNEL_ANSWER" or
> >> >> >> > "CHANNEL_PROGRESS_MEDIA" on outbound leg but you say that I have
> >> >> >> > wait
> >> >> >> > that
> >> >> >> > "originate has ended": you mean that I have to wait for
> >> >> >> > "BACKGROUND_JOB"
> >> >> >> > event related to my "bgapi originate ... &park"?
> >> >> >> > I'm already doing "uuid_bridge <inbound_uuid> <outbound_uuid>".
> >> >> >> > I'll try also with intercept and inline originate. Thank you!
> >> >> >> > Stephen
> >> >> >> >
> >> >> >> > On Tue, Feb 15, 2011 at 12:53 AM, Anthony Minessale
> >> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >> >>
> >> >> >> >> you need to make sure originate has ended on the outbound leg
> >> >> >> >> before
> >> >> >> >> you use it in a bridge etc.
> >> >> >> >> you also need to supply the inbound leg first in uuid_bridge if
> >> >> >> >> that
> >> >> >> >> is something you want to do.
> >> >> >> >>
> >> >> >> >> Easier would be to originate the B leg to park inline and tell
> A
> >> >> >> >> leg
> >> >> >> >> to execute intercept on the B leg uuid.
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> On Mon, Feb 14, 2011 at 5:44 PM, Stephen Wilde
> >> >> >> >> <wstephen80 at gmail.com>
> >> >> >> >> wrote:
> >> >> >> >> > Sorry but I have missed something.
> >> >> >> >> > I know that I can solve this problem directly in dialplan
> with
> >> >> >> >> > a
> >> >> >> >> > bridge
> >> >> >> >> > but
> >> >> >> >> > what I'm doing is an "Freeswitch ESL learning" because I have
> >> >> >> >> > to
> >> >> >> >> > port
> >> >> >> >> > some
> >> >> >> >> > application in Freeswitch and I'm learning how to implement
> >> >> >> >> > some
> >> >> >> >> > functionality.
> >> >> >> >> > For me it's important to take control of both
> inbound/outbound
> >> >> >> >> > in
> >> >> >> >> > full
> >> >> >> >> > async
> >> >> >> >> > way and I have the necessity to do the complete call control.
> >> >> >> >> > I'm not sure but to me it seems that with a normal bridge I
> >> >> >> >> > lose
> >> >> >> >> > the
> >> >> >> >> > control
> >> >> >> >> > of two sessions, for example, an outbound answer is
> propagated
> >> >> >> >> > by
> >> >> >> >> > bridge
> >> >> >> >> > application as inbound answer.
> >> >> >> >> > What I want to do is an audio bridging so my application can
> >> >> >> >> > take
> >> >> >> >> > control of
> >> >> >> >> > "signaling bridging".
> >> >> >> >> >
> >> >> >> >> > I'm wrong? There are other way to do that?
> >> >> >> >> > Stephen
> >> >> >> >> > On Tue, Feb 15, 2011 at 12:19 AM, Michael Collins
> >> >> >> >> > <msc at freeswitch.org>
> >> >> >> >> > wrote:
> >> >> >> >> >>
> >> >> >> >> >> My head is spinning after reading this email. :)
> >> >> >> >> >> It sounds like you just need a simple bridge from the
> incoming
> >> >> >> >> >> leg
> >> >> >> >> >> to
> >> >> >> >> >> the
> >> >> >> >> >> outgoing leg. Can you pre_answer the A leg then execute a
> good
> >> >> >> >> >> old-fashioned
> >> >> >> >> >> bridge to the b-leg?
> >> >> >> >> >> -MC
> >> >> >> >> >> On Mon, Feb 14, 2011 at 4:57 PM, Stephen Wilde
> >> >> >> >> >> <wstephen80 at gmail.com>
> >> >> >> >> >> wrote:
> >> >> >> >> >>>
> >> >> >> >> >>> The problem I have is to propagate the audio coming from an
> >> >> >> >> >>> "originated"
> >> >> >> >> >>> outbound session to the inbound session when the outbound
> is
> >> >> >> >> >>> in
> >> >> >> >> >>> the
> >> >> >> >> >>> PROGRESS
> >> >> >> >> >>> MEDIA phase.
> >> >> >> >> >>> When my application receives the "CHANNEL_PROGRESS_MEDIA"
> >> >> >> >> >>> event
> >> >> >> >> >>> from
> >> >> >> >> >>> outbound session I can do a "pre_answer" on inbound session
> >> >> >> >> >>> but
> >> >> >> >> >>> I'm
> >> >> >> >> >>> not
> >> >> >> >> >>> capable to do an audio bridge.
> >> >> >> >> >>> I have tried with "uuid_bridge <outbound_uuid>
> >> >> >> >> >>> <inbound_uuid>"
> >> >> >> >> >>> with
> >> >> >> >> >>> no
> >> >> >> >> >>> result probably because this api requires that at least one
> >> >> >> >> >>> session
> >> >> >> >> >>> must be
> >> >> >> >> >>> answered.
> >> >> >> >> >>> I don't want to answer to the inbound session to propagate
> >> >> >> >> >>> the
> >> >> >> >> >>> outbound
> >> >> >> >> >>> progressing media but I want to answer to inbound only on
> >> >> >> >> >>> outbound
> >> >> >> >> >>> answer.
> >> >> >> >> >>> Any way to do that?
> >> >> >> >> >>> 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
> >> >> >> >> >>
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > _______________________________________________
> >> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> >> > FreeSWITCH-users at 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 lloyd.aloysius at gmail.com  Wed Feb 16 00:49:52 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 15 Feb 2011 16:49:52 -0500
Subject: [Freeswitch-users] Getting SIP trace on user level
In-Reply-To: <AANLkTi=YNTzEYUiwaW2bZN9YRTr+VURWVgSz4xsw_gdn@mail.gmail.com>
References: <AANLkTimhW139WdDKXiQkRm43Ai13f_7Z+N9Z-K2y8Ldi@mail.gmail.com>
	<AANLkTikxBuc960_-sZvXQXdXAC5LzmiNvagj9RUC2GPx@mail.gmail.com>
	<AANLkTi=+yL16rqZs9ig0R4JUfsfFOtmfr1s9jDY=-Gpf@mail.gmail.com>
	<AANLkTi=YNTzEYUiwaW2bZN9YRTr+VURWVgSz4xsw_gdn@mail.gmail.com>
Message-ID: <AANLkTi=0rDWMNk3aS29QvoRu=GL4QjMuo3A_psK+iD6G@mail.gmail.com>

Thank you for all the replies . I am currently using ngrep and tcpdump. But
it is nice to have in FreeSWITCH CLI this option.

Thanks
Lloyd


On Tue, Feb 15, 2011 at 12:56 PM, Michael Collins <msc at freeswitch.org>wrote:

> FYI,
> If you know the IP address of the phone you can also use tcpdump:
>
> tcpdump -nq -s 0 -w myfile.pcap host 1.2.3.4 and port 5060
>
> -MC
>
>
> On Tue, Feb 15, 2011 at 7:47 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> tshark can also do the same (part of wireshark)
>>
>> -Steve
>>
>>
>>
>> On 15 February 2011 13:14, Kristian Kielhofner <kris at kriskinc.com> wrote:
>>
>>> In FreeSWITCH: not that I know of.
>>>
>>> In Linux, yes: http://ngrep.sourceforge.net/
>>>
>>> Use ngrep and match on anything, including IP and port combination
>>> from a specific peer (that's all Asterisk does anyway).  It would be
>>> trivial to throw a script together to pull out the sofia contact for a
>>> user and pass the IP:port to ngrep for filtering.
>>>
>>> You'll find ngrep to be very handy.
>>>
>>> On Mon, Feb 14, 2011 at 11:06 PM, Aloysius Lloyd
>>> <lloyd.aloysius at gmail.com> wrote:
>>> > Hi All,
>>> > Is there any console command available to get SIP trance on user level
>>> .
>>> > Equivalent to Asterisk "sip set debug peer <peername>"
>>> >
>>> >
>>> > TIA
>>> > 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
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> 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
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 16 01:07:57 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 15 Feb 2011 22:07:57 +0000
Subject: [Freeswitch-users] Getting SIP trace on user level
In-Reply-To: <AANLkTi=0rDWMNk3aS29QvoRu=GL4QjMuo3A_psK+iD6G@mail.gmail.com>
References: <AANLkTimhW139WdDKXiQkRm43Ai13f_7Z+N9Z-K2y8Ldi@mail.gmail.com>
	<AANLkTikxBuc960_-sZvXQXdXAC5LzmiNvagj9RUC2GPx@mail.gmail.com>
	<AANLkTi=+yL16rqZs9ig0R4JUfsfFOtmfr1s9jDY=-Gpf@mail.gmail.com>
	<AANLkTi=YNTzEYUiwaW2bZN9YRTr+VURWVgSz4xsw_gdn@mail.gmail.com>
	<AANLkTi=0rDWMNk3aS29QvoRu=GL4QjMuo3A_psK+iD6G@mail.gmail.com>
Message-ID: <8DCCF5BD-9B1A-46DB-B559-762AF06A81CA@gmail.com>

I believe the problem is that the logging is handled by the Sofia stack on a per-profile basis not itself, and occurs before it is matched to a call/user.

Steve on iPhone

On 15 Feb 2011, at 21:49, Aloysius Lloyd <lloyd.aloysius at gmail.com> wrote:

> Thank you for all the replies . I am currently using ngrep and tcpdump. But it is nice to have in FreeSWITCH CLI this option.
> 
> Thanks
> Lloyd
> 
> 
> On Tue, Feb 15, 2011 at 12:56 PM, Michael Collins <msc at freeswitch.org> wrote:
> FYI,
> If you know the IP address of the phone you can also use tcpdump:
> 
> tcpdump -nq -s 0 -w myfile.pcap host 1.2.3.4 and port 5060
> 
> -MC
> 
> 
> On Tue, Feb 15, 2011 at 7:47 AM, Steven Ayre <steveayre at gmail.com> wrote:
> tshark can also do the same (part of wireshark)
> 
> -Steve
> 
> 
> 
> On 15 February 2011 13:14, Kristian Kielhofner <kris at kriskinc.com> wrote:
> In FreeSWITCH: not that I know of.
> 
> In Linux, yes: http://ngrep.sourceforge.net/
> 
> Use ngrep and match on anything, including IP and port combination
> from a specific peer (that's all Asterisk does anyway).  It would be
> trivial to throw a script together to pull out the sofia contact for a
> user and pass the IP:port to ngrep for filtering.
> 
> You'll find ngrep to be very handy.
> 
> On Mon, Feb 14, 2011 at 11:06 PM, Aloysius Lloyd
> <lloyd.aloysius at gmail.com> wrote:
> > Hi All,
> > Is there any console command available to get SIP trance on user level .
> > Equivalent to Asterisk "sip set debug peer <peername>"
> >
> >
> > TIA
> > 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
> >
> >
> 
> 
> 
> --
> Kristian Kielhofner
> 
> _______________________________________________
> 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  Wed Feb 16 01:38:35 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 15 Feb 2011 16:38:35 -0600
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 56, Issue 177
In-Reply-To: <AANLkTimf9wTpy36Hogi+kEVVAKWMZ2z+R2jA47x6NEfy@mail.gmail.com>
References: <mailman.3855.1297702532.24555.freeswitch-users@lists.freeswitch.org>
	<AANLkTimf9wTpy36Hogi+kEVVAKWMZ2z+R2jA47x6NEfy@mail.gmail.com>
Message-ID: <AANLkTinUC1u_6rKYqVm4EGbziQoPg7A4hNyw9jc32eZ1@mail.gmail.com>

Well that's all I could think of:

passthrough is passthrough FS never modifies anything.
you need to look harder at sip trace, pcaps or other diagnostics for
some misconfiguration.
many people use this daily.


On Tue, Feb 15, 2011 at 1:12 AM, Javier Gallart <jgallartm at gmail.com> wrote:
>
> Anthony, thanks for the tip. I haven't seen any change though. I should have
> mentioned that I'm focusing in the rtp stream coming form the callee to the
> caller. The problem is that FS relays to the caller only half of the packets
> received from the callee. The packets arrive at FS at a rate of 1 packet
> every 20 ms, and with a payload of 20 bytes each. Packets from FS to the
> caller are sent every 40 ms with a payload of 20 bytes, thus skipping half
> of the information.
>
> Thanks
>>
>> ---------- Forwarded message ----------
>> From:?Anthony Minessale <anthony.minessale at gmail.com>
>> To:?FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Date:?Mon, 14 Feb 2011 10:38:57 -0600
>> Subject:?Re: [Freeswitch-users] g729 packets skipped in passthrough mode
>> I bet its 20ms vs 30
>> set passthru_ptime_mismatch to true either in vars.xml or in your
>> dialplan both legs through export on the a leg or set in the a leg and
>> in the {} on b.
>>
>>
>> On Mon, Feb 14, 2011 at 10:28 AM, Brian West <brian at freeswitch.org> wrote:
>> > What exactly is the problem? ?I see no issue here can you elaborate on
>> > what you're seeing?
>> >
>> > /b
>> >
>> > On Feb 14, 2011, at 4:48 AM, Javier Gallart wrote:
>> >
>> >> I've tried to explicitly set ptime at 20ms at switch.conf -although
>> >> it's not necessary afaik-. Has any one experienced this same issue?
>> >>
>> >> Thanks in advance..
>> >>
>> >> 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
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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
>
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 Feb 16 01:32:54 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 15 Feb 2011 16:32:54 -0600
Subject: [Freeswitch-users] ESL Outbound: how to bridge media?
In-Reply-To: <AANLkTim22-SPAnWp6OttQt7AcUVBzRS=z1aiDvsgsD3j@mail.gmail.com>
References: <AANLkTikLVLpSGkrRqeD3WVFZZgSaLnQvypCOG39hEhLX@mail.gmail.com>
	<AANLkTimwX8Uz4t9gp6Ppzfu+WE+UJa3FO4qHE7XKFOBz@mail.gmail.com>
	<AANLkTi=t-a7U9ov3wJ6Dh-PuniSfZbyeQ==BCAvMkUPN@mail.gmail.com>
	<AANLkTik27F7JjyMRLKryQZq_N+oVDsDzdmiPPa+d2KWN@mail.gmail.com>
	<AANLkTine=jG-dc98pDySashEvo3T5cSPwVXYwp54mhhC@mail.gmail.com>
	<AANLkTi=JAt=RkQ5b=DBajH_YR6TmiqExdjgF1=OL=fJ_@mail.gmail.com>
	<AANLkTinhkOznPh98kkHp=rCU9Qm0xv_hW3c41rRJr=Kn@mail.gmail.com>
	<AANLkTikAJoP7_cmhoEDorq4uCTdw+SVQeMNMpcFBkdYn@mail.gmail.com>
	<AANLkTi=XbutFHsC1w_n3zG+pRW6gtmVC=+YPH0cvK=eA@mail.gmail.com>
	<AANLkTimMDkdWi_V8TDXrxzn9paLEsthTGaOXMGD2_-PN@mail.gmail.com>
	<AANLkTikghfSv1pu-V8-qKB1C6Oc-JFipv2Wm=eWwJfgr@mail.gmail.com>
	<AANLkTi=Sjtsw6KPqDXJmB-F8g-ZcUfz7n62ro7b4jmid@mail.gmail.com>
	<AANLkTim22-SPAnWp6OttQt7AcUVBzRS=z1aiDvsgsD3j@mail.gmail.com>
Message-ID: <AANLkTikE8cGwPppjS76ABot0YYgHzMToJGszFpZMNBLu@mail.gmail.com>

Did you read the little chart i put in my last email?

i'll be more specific:

1) leg A who has a_uuid has called inbound and has been put in park app.

from ESL:
generate uuid (we'll  call  it b_uuid for clarity)
bgapi originate {origination_uuid=b_uuid}soifa/internal/bleg at box.com park inline

Wait for park event from b_uuid (do not touch it any sooner)

now either do:

uuid_bridge a_uuid b_uuid

or:

uuid_transfer a_uuid intercept:b_uuid inline

or:

sendmsg to a_uuid telling it to execute intercept app on b_uuid





On Tue, Feb 15, 2011 at 3:46 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:
> When legB answers there is also no audio.
> If I do the bridge after legB answers the call (instead of when
> "origination" has ended), then the audio is ok?and the legA can ear the legB
> also if legA is in "pre_answer" state.
> Now I have found one problem: the "BACKGROUND_JOB" event related to "bgapi
> uuid_bridge ..." contains: "-ERR Invalid uuid" but?both uuid's are correct
> so probably is due to the state of channels.
> With the same uuid's, the uuid_bridge after the legB answer works fine.
> Stephen
>
> On Tue, Feb 15, 2011 at 9:53 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> I meant when you have no audio, and you wait until leg b answers. what
>> happens then?
>>
>>
>> If you have leg A inbound and you send it to park.
>> then you originate outbound to b leg to park
>> wait for park event on b leg
>> then you uuid_bridge them
>>
>> also you can sendmsg with application = intercept app_arg = uuid of B
>>
>>
>> On Tue, Feb 15, 2011 at 12:08 PM, Stephen Wilde <wstephen80 at gmail.com>
>> wrote:
>> > LegA is an incoming call, legB is an outgoing call.
>> > The dialplan is:
>> > ?? ?<extension name="myhandler">
>> > ?? ? ?<condition field="destination_number" expression="^xxxx$">
>> > ?? ? ? ?<action application="set"
>> > data="dialstring=sofia/external/yyyy at zzzz
>> > />
>> > ?? ? ? ?<action application="socket" data="127.0.0.1:8084 async full"/>
>> > ?? ? ?</condition>
>> > ?? ?</extension>
>> > When the legB answer the call, I receive the "CHANNEL_ANSWER" event on
>> > legB
>> > and I send the "answer" to legA (I have already done a uuid_bridge when
>> > originate has ended).
>> > The audio continue to be mute in both direction.
>> > The "uuid_bridge" works only if I call it after received
>> > "CHANNEL_ANSWER" in
>> > legB or after I send "answer" to legA.
>> > It's not enough to do a "pre_answer".
>> > Stephen
>> >
>> > On Tue, Feb 15, 2011 at 6:15 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> What happens once they answer?
>> >>
>> >> This is not one of those attempts to speed up click to call by calling
>> >> them both at once is it?
>> >>
>> >>
>> >> On Tue, Feb 15, 2011 at 10:29 AM, Stephen Wilde <wstephen80 at gmail.com>
>> >> wrote:
>> >> > Ok, I have tried with "pre_answer" to the inbound channel (that will
>> >> > be
>> >> > the
>> >> > first uuid in the "uuid_bridge")?but with same result: no audio from
>> >> > outbound (in progressing media state) to inbound (in progressing
>> >> > media
>> >> > state
>> >> > due to my pre_answer).
>> >> > Here the sequence of events:
>> >> > [ERROR] newacme.cpp:193 mycallback() Connected!
>> >> > [ERROR] newacme.cpp:113 Originate() bgapi originate
>> >> >
>> >> >
>> >> > [origination_uuid=61e00b94-f4d3-4ac4-bcf5-8fba02a18de6]sofia/external/xxxx at yyyy
>> >> > &park()
>> >> > [ERROR] newacme.cpp:115 Originate() Job-ID:
>> >> > eee79d97-dbf8-49d4-9bf9-578ef75f73ea
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_UUID]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_OUTGOING]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_ORIGINATE]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_STATE]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_CALLSTATE]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_STATE]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_STATE]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CALL_UPDATE]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CODEC]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CODEC]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_CALLSTATE]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_PROGRESS_MEDIA]
>> >> > [ERROR] newacme.cpp:250 mycallback() Executing 'pre_answer' on
>> >> > INBOUND
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_STATE]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_EXECUTE]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_PARK]
>> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_EXECUTE]
>> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_CALLSTATE]
>> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_PROGRESS_MEDIA]
>> >> > [ERROR] newacme.cpp:127 BridgeTo() bgapi uuid_bridge
>> >> > 902e2cd4-10ee-4938-97cf-c9eb2ac969f8
>> >> > 61e00b94-f4d3-4ac4-bcf5-8fba02a18de6
>> >> > [ERROR] newacme.cpp:129 BridgeTo() Job-ID:
>> >> > 7a66a5c7-721c-47f4-aff2-373ec20123c8
>> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CALL_UPDATE]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_CALLSTATE]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_ANSWER]
>> >> > [ERROR] newacme.cpp:255 mycallback() Executing 'answer' on INBOUND
>> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_EXECUTE]
>> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_CALLSTATE]
>> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_ANSWER]
>> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_CALLSTATE]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_HANGUP]
>> >> > [ERROR] newacme.cpp:260 mycallback() Executing 'hangup' on INBOUND
>> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_EXECUTE]
>> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_CALLSTATE]
>> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_HANGUP]
>> >> > [ERROR] newacme.cpp:272 mycallback() Executing 'hangup' on OUTBOUND
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_UNPARK]
>> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_STATE]
>> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_UNPARK]
>> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_STATE]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_STATE]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_HANGUP_COMPLETE]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_DESTROY]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_CALLSTATE]
>> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> > [CHANNEL_STATE]
>> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_STATE]
>> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> > [CHANNEL_HANGUP_COMPLETE]
>> >> > [ERROR] newacme.cpp:290 mycallback() End.
>> >> >
>> >> >
>> >> > On Tue, Feb 15, 2011 at 5:09 PM, Anthony Minessale
>> >> > <anthony.minessale at gmail.com> wrote:
>> >> >>
>> >> >> Well you can't bridge 2 unanswered channels together.
>> >> >> You need at least one of them to have been pre_answered and that
>> >> >> should be the first uuid in the list.
>> >> >>
>> >> >>
>> >> >> On Tue, Feb 15, 2011 at 3:48 AM, Stephen Wilde
>> >> >> <wstephen80 at gmail.com>
>> >> >> wrote:
>> >> >> > I have tried to wait originate completition but the "uuid_bridge"
>> >> >> > doesn't
>> >> >> > works also in this case.
>> >> >> > It seems that works only when at least one leg is answered so it's
>> >> >> > not
>> >> >> > possible to do the "uuid_bridge" during progressing phase also if
>> >> >> > the
>> >> >> > originate has ended (I don't set the ignore_early_media).
>> >> >> > My application is this:?http://pastebin.freeswitch.org/15387
>> >> >> > The application:
>> >> >> > 1. receive a call with an "inbound_uuid"
>> >> >> > 2. create a new "outbound_uuid"
>> >> >> > 3. do a "bgapi originate" using the new "outbound_uuid"
>> >> >> > 4. when the called phone is ringing, receive a
>> >> >> > "CHANNEL_PROGRESS_MEDIA"
>> >> >> > event on "outbound_uuid" and do a "pre_answer" on "inbound_uuid"
>> >> >> > 5. receive a?"CHANNEL_PROGRESS_MEDIA" on "inbound_uuid"
>> >> >> > 6. receive a job completition event related to the "originate" so
>> >> >> > issue
>> >> >> > an
>> >> >> > "uuid_bridge inbound_uuid outbound_uuid"
>> >> >> > 7. when a job completition event related to "uuid_bridge" is
>> >> >> > received,
>> >> >> > no
>> >> >> > audio flow from outbound to inbound channel
>> >> >> > 8. when outbound answer the call, the application answer also the
>> >> >> > inbound
>> >> >> > call but no audio flow in both directions
>> >> >> > If I do the uuid_bridge after legB answer, then all is ok
>> >> >> > (obviously
>> >> >> > with no
>> >> >> > audio during progressing phase).
>> >> >> >
>> >> >> > The log of my application is:
>> >> >> > [ERROR] newacme.cpp:46 mycallback() Connected!
>> >> >> > [ERROR] newacme.cpp:65 mycallback() Inbound uuid =
>> >> >> > 8b2c39db-1ad9-489c-b72f-a92d4087bf99
>> >> >> > [ERROR] newacme.cpp:68 mycallback() create_uuid:
>> >> >> > 394167aa-2811-4fcd-95c9-85576bdd9a7a
>> >> >> > [ERROR] newacme.cpp:89 mycallback() bgapi originate
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > [origination_uuid=394167aa-2811-4fcd-95c9-85576bdd9a7a]sofia/external/xxxx at yyyy
>> >> >> > &park()
>> >> >> > [ERROR] newacme.cpp:91 mycallback() Job-ID:
>> >> >> > 1c654ac2-c22d-418f-9fad-bb6b6b35aeff
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_UUID]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_OUTGOING]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_ORIGINATE]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_STATE]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_CALLSTATE]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_STATE]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_STATE]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CALL_UPDATE]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CODEC]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CODEC]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_CALLSTATE]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_PROGRESS_MEDIA]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_STATE]
>> >> >> > [ERROR] newacme.cpp:112 mycallback() Got an event:
>> >> >> > [BACKGROUND_JOB] -
>> >> >> > [1c654ac2-c22d-418f-9fad-bb6b6b35aeff]
>> >> >> > [ERROR] newacme.cpp:119 mycallback() bgapi uuid_bridge
>> >> >> > 8b2c39db-1ad9-489c-b72f-a92d4087bf99
>> >> >> > 394167aa-2811-4fcd-95c9-85576bdd9a7a
>> >> >> > [ERROR] newacme.cpp:123 mycallback() Job-ID:
>> >> >> > 0d43cffe-fdc6-4a13-97e6-aeee199c45d5
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_EXECUTE]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_PARK]
>> >> >> > [ERROR] newacme.cpp:112 mycallback() Got an event:
>> >> >> > [BACKGROUND_JOB] -
>> >> >> > [0d43cffe-fdc6-4a13-97e6-aeee199c45d5]
>> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_EXECUTE]
>> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_CALLSTATE]
>> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_PROGRESS_MEDIA]
>> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CALL_UPDATE]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_CALLSTATE]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_ANSWER]
>> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_EXECUTE]
>> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_CALLSTATE]
>> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_ANSWER]
>> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_CALLSTATE]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_HANGUP]
>> >> >> > [ERROR] newacme.cpp:160 mycallback() hangup
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_UNPARK]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_STATE]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_STATE]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_HANGUP_COMPLETE]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_DESTROY]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_CALLSTATE]
>> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_STATE]
>> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_EXECUTE]
>> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_CALLSTATE]
>> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_HANGUP]
>> >> >> > [ERROR] newacme.cpp:171 mycallback() hangup
>> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_UNPARK]
>> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_STATE]
>> >> >> > [ERROR] newacme.cpp:184 mycallback() End.
>> >> >> >
>> >> >> > Stephen
>> >> >> >
>> >> >> > On Tue, Feb 15, 2011 at 1:14 AM, Anthony Minessale
>> >> >> > <anthony.minessale at gmail.com> wrote:
>> >> >> >>
>> >> >> >> if you do originate without ignore_early_media=true set it will
>> >> >> >> end
>> >> >> >> the soonest possible where it's suitable for a bridge.
>> >> >> >> so that is the best bet to wait for originate to end.
>> >> >> >>
>> >> >> >>
>> >> >> >> On Mon, Feb 14, 2011 at 6:11 PM, Stephen Wilde
>> >> >> >> <wstephen80 at gmail.com>
>> >> >> >> wrote:
>> >> >> >> > Ok, now I'm doing bridge when I receive "CHANNEL_ANSWER" or
>> >> >> >> > "CHANNEL_PROGRESS_MEDIA" on outbound leg but you say that I
>> >> >> >> > have
>> >> >> >> > wait
>> >> >> >> > that
>> >> >> >> > "originate has ended": you mean that I have to wait for
>> >> >> >> > "BACKGROUND_JOB"
>> >> >> >> > event related to my "bgapi originate ... &park"?
>> >> >> >> > I'm already doing "uuid_bridge <inbound_uuid> <outbound_uuid>".
>> >> >> >> > I'll try also with intercept and inline originate.?Thank you!
>> >> >> >> > Stephen
>> >> >> >> >
>> >> >> >> > On Tue, Feb 15, 2011 at 12:53 AM, Anthony Minessale
>> >> >> >> > <anthony.minessale at gmail.com> wrote:
>> >> >> >> >>
>> >> >> >> >> you need to make sure originate has ended on the outbound leg
>> >> >> >> >> before
>> >> >> >> >> you use it in a bridge etc.
>> >> >> >> >> you also need to supply the inbound leg first in uuid_bridge
>> >> >> >> >> if
>> >> >> >> >> that
>> >> >> >> >> is something you want to do.
>> >> >> >> >>
>> >> >> >> >> Easier would be to originate the B leg to park inline and tell
>> >> >> >> >> A
>> >> >> >> >> leg
>> >> >> >> >> to execute intercept on the B leg uuid.
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> On Mon, Feb 14, 2011 at 5:44 PM, Stephen Wilde
>> >> >> >> >> <wstephen80 at gmail.com>
>> >> >> >> >> wrote:
>> >> >> >> >> > Sorry but I have missed something.
>> >> >> >> >> > I know that I can solve this problem directly in dialplan
>> >> >> >> >> > with
>> >> >> >> >> > a
>> >> >> >> >> > bridge
>> >> >> >> >> > but
>> >> >> >> >> > what I'm doing is an "Freeswitch ESL learning"?because I
>> >> >> >> >> > have
>> >> >> >> >> > to
>> >> >> >> >> > port
>> >> >> >> >> > some
>> >> >> >> >> > application in Freeswitch and I'm learning how to implement
>> >> >> >> >> > some
>> >> >> >> >> > functionality.
>> >> >> >> >> > For me it's important to take control of both
>> >> >> >> >> > inbound/outbound
>> >> >> >> >> > in
>> >> >> >> >> > full
>> >> >> >> >> > async
>> >> >> >> >> > way and I have the necessity to do the complete call
>> >> >> >> >> > control.
>> >> >> >> >> > I'm not sure but to me it seems that with a normal bridge I
>> >> >> >> >> > lose
>> >> >> >> >> > the
>> >> >> >> >> > control
>> >> >> >> >> > of two sessions, for example, an outbound answer is
>> >> >> >> >> > propagated
>> >> >> >> >> > by
>> >> >> >> >> > bridge
>> >> >> >> >> > application as inbound answer.
>> >> >> >> >> > What I want to do is an audio bridging so my application can
>> >> >> >> >> > take
>> >> >> >> >> > control of
>> >> >> >> >> > "signaling bridging".
>> >> >> >> >> >
>> >> >> >> >> > I'm wrong? There are other way to do that?
>> >> >> >> >> > Stephen
>> >> >> >> >> > On Tue, Feb 15, 2011 at 12:19 AM, Michael Collins
>> >> >> >> >> > <msc at freeswitch.org>
>> >> >> >> >> > wrote:
>> >> >> >> >> >>
>> >> >> >> >> >> My head is spinning after reading this email. :)
>> >> >> >> >> >> It sounds like you just need a simple bridge from the
>> >> >> >> >> >> incoming
>> >> >> >> >> >> leg
>> >> >> >> >> >> to
>> >> >> >> >> >> the
>> >> >> >> >> >> outgoing leg. Can you pre_answer the A leg then execute a
>> >> >> >> >> >> good
>> >> >> >> >> >> old-fashioned
>> >> >> >> >> >> bridge to the b-leg?
>> >> >> >> >> >> -MC
>> >> >> >> >> >> On Mon, Feb 14, 2011 at 4:57 PM, Stephen Wilde
>> >> >> >> >> >> <wstephen80 at gmail.com>
>> >> >> >> >> >> wrote:
>> >> >> >> >> >>>
>> >> >> >> >> >>> The problem I have is to propagate the audio coming from
>> >> >> >> >> >>> an
>> >> >> >> >> >>> "originated"
>> >> >> >> >> >>> outbound session to the inbound session when the outbound
>> >> >> >> >> >>> is
>> >> >> >> >> >>> in
>> >> >> >> >> >>> the
>> >> >> >> >> >>> PROGRESS
>> >> >> >> >> >>> MEDIA phase.
>> >> >> >> >> >>> When my application receives the "CHANNEL_PROGRESS_MEDIA"
>> >> >> >> >> >>> event
>> >> >> >> >> >>> from
>> >> >> >> >> >>> outbound session I can do a "pre_answer" on inbound
>> >> >> >> >> >>> session
>> >> >> >> >> >>> but
>> >> >> >> >> >>> I'm
>> >> >> >> >> >>> not
>> >> >> >> >> >>> capable to do an audio bridge.
>> >> >> >> >> >>> I have tried with "uuid_bridge <outbound_uuid>
>> >> >> >> >> >>> <inbound_uuid>"
>> >> >> >> >> >>> with
>> >> >> >> >> >>> no
>> >> >> >> >> >>> result probably because this api requires that at least
>> >> >> >> >> >>> one
>> >> >> >> >> >>> session
>> >> >> >> >> >>> must be
>> >> >> >> >> >>> answered.
>> >> >> >> >> >>> I don't want to answer to the inbound session to propagate
>> >> >> >> >> >>> the
>> >> >> >> >> >>> outbound
>> >> >> >> >> >>> progressing media but I want to answer to inbound only on
>> >> >> >> >> >>> outbound
>> >> >> >> >> >>> answer.
>> >> >> >> >> >>> Any way to do that?
>> >> >> >> >> >>> 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
>> >> >> >> >> >>
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> > _______________________________________________
>> >> >> >> >> > FreeSWITCH-users mailing list
>> >> >> >> >> > FreeSWITCH-users at 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 wstephen80 at gmail.com  Wed Feb 16 02:30:00 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Wed, 16 Feb 2011 00:30:00 +0100
Subject: [Freeswitch-users] ESL Outbound: how to bridge media?
In-Reply-To: <AANLkTikE8cGwPppjS76ABot0YYgHzMToJGszFpZMNBLu@mail.gmail.com>
References: <AANLkTikLVLpSGkrRqeD3WVFZZgSaLnQvypCOG39hEhLX@mail.gmail.com>
	<AANLkTimwX8Uz4t9gp6Ppzfu+WE+UJa3FO4qHE7XKFOBz@mail.gmail.com>
	<AANLkTi=t-a7U9ov3wJ6Dh-PuniSfZbyeQ==BCAvMkUPN@mail.gmail.com>
	<AANLkTik27F7JjyMRLKryQZq_N+oVDsDzdmiPPa+d2KWN@mail.gmail.com>
	<AANLkTine=jG-dc98pDySashEvo3T5cSPwVXYwp54mhhC@mail.gmail.com>
	<AANLkTi=JAt=RkQ5b=DBajH_YR6TmiqExdjgF1=OL=fJ_@mail.gmail.com>
	<AANLkTinhkOznPh98kkHp=rCU9Qm0xv_hW3c41rRJr=Kn@mail.gmail.com>
	<AANLkTikAJoP7_cmhoEDorq4uCTdw+SVQeMNMpcFBkdYn@mail.gmail.com>
	<AANLkTi=XbutFHsC1w_n3zG+pRW6gtmVC=+YPH0cvK=eA@mail.gmail.com>
	<AANLkTimMDkdWi_V8TDXrxzn9paLEsthTGaOXMGD2_-PN@mail.gmail.com>
	<AANLkTikghfSv1pu-V8-qKB1C6Oc-JFipv2Wm=eWwJfgr@mail.gmail.com>
	<AANLkTi=Sjtsw6KPqDXJmB-F8g-ZcUfz7n62ro7b4jmid@mail.gmail.com>
	<AANLkTim22-SPAnWp6OttQt7AcUVBzRS=z1aiDvsgsD3j@mail.gmail.com>
	<AANLkTikE8cGwPppjS76ABot0YYgHzMToJGszFpZMNBLu@mail.gmail.com>
Message-ID: <AANLkTikftaVhR1jXO5sPqZTOwh9AAPGR4bOo764hPqw8@mail.gmail.com>

Sorry, I have now followed your indication doing the bridge only after
receiving "CHANNEL_PARK" on legB. The problem is the same with no audio.

I have also tried to do a pre_answer in legA (before originate legB) with no
success.

I see in the fs_cli:

[CRIT] switch_ivr_bridge.c:1412 Neither channel is answered, cannot bridge
them.

The sequence of events I receive before doing the bridge is:

[OUTBOUND] - [CHANNEL_UUID]
[OUTBOUND] - [CHANNEL_OUTGOING]
[OUTBOUND] - [CHANNEL_ORIGINATE]
[OUTBOUND] - [CHANNEL_STATE]
[OUTBOUND] - [CHANNEL_CALLSTATE]
[OUTBOUND] - [CHANNEL_STATE]
[OUTBOUND] - [CHANNEL_STATE]
[INBOUND] - [CHANNEL_EXECUTE]
[INBOUND] - [CHANNEL_CALLSTATE]
[INBOUND] - [CHANNEL_PROGRESS_MEDIA]
[INBOUND] - [CHANNEL_EXECUTE_COMPLETE]
[OUTBOUND] - [CALL_UPDATE]
[OUTBOUND] - [CHANNEL_PROGRESS]
[OUTBOUND] - [CALL_UPDATE]
[OUTBOUND] - [CODEC]
[OUTBOUND] - [CODEC]
[OUTBOUND] - [CHANNEL_CALLSTATE]
[OUTBOUND] - [CHANNEL_PROGRESS_MEDIA]
[OUTBOUND] - [CHANNEL_STATE]
[OUTBOUND] - [CHANNEL_EXECUTE]
[OUTBOUND] - [CHANNEL_PARK]

After this CHANNEL_PARK I do the "uuid_bridge" with no succes. These event
are related to the case of putting legA in pre_answer. Same problem if I
leave it in park.

Stephen


On Tue, Feb 15, 2011 at 11:32 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Did you read the little chart i put in my last email?
>
> i'll be more specific:
>
> 1) leg A who has a_uuid has called inbound and has been put in park app.
>
> from ESL:
> generate uuid (we'll  call  it b_uuid for clarity)
> bgapi originate {origination_uuid=b_uuid}soifa/internal/bleg at box.com park
> inline
>
> Wait for park event from b_uuid (do not touch it any sooner)
>
> now either do:
>
> uuid_bridge a_uuid b_uuid
>
> or:
>
> uuid_transfer a_uuid intercept:b_uuid inline
>
> or:
>
> sendmsg to a_uuid telling it to execute intercept app on b_uuid
>
>
>
>
>
> On Tue, Feb 15, 2011 at 3:46 PM, Stephen Wilde <wstephen80 at gmail.com>
> wrote:
> > When legB answers there is also no audio.
> > If I do the bridge after legB answers the call (instead of when
> > "origination" has ended), then the audio is ok and the legA can ear the
> legB
> > also if legA is in "pre_answer" state.
> > Now I have found one problem: the "BACKGROUND_JOB" event related to
> "bgapi
> > uuid_bridge ..." contains: "-ERR Invalid uuid" but both uuid's are
> correct
> > so probably is due to the state of channels.
> > With the same uuid's, the uuid_bridge after the legB answer works fine.
> > Stephen
> >
> > On Tue, Feb 15, 2011 at 9:53 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> I meant when you have no audio, and you wait until leg b answers. what
> >> happens then?
> >>
> >>
> >> If you have leg A inbound and you send it to park.
> >> then you originate outbound to b leg to park
> >> wait for park event on b leg
> >> then you uuid_bridge them
> >>
> >> also you can sendmsg with application = intercept app_arg = uuid of B
> >>
> >>
> >> On Tue, Feb 15, 2011 at 12:08 PM, Stephen Wilde <wstephen80 at gmail.com>
> >> wrote:
> >> > LegA is an incoming call, legB is an outgoing call.
> >> > The dialplan is:
> >> >     <extension name="myhandler">
> >> >       <condition field="destination_number" expression="^xxxx$">
> >> >         <action application="set"
> >> > data="dialstring=sofia/external/yyyy at zzzz
> >> > />
> >> >         <action application="socket" data="127.0.0.1:8084 async
> full"/>
> >> >       </condition>
> >> >     </extension>
> >> > When the legB answer the call, I receive the "CHANNEL_ANSWER" event on
> >> > legB
> >> > and I send the "answer" to legA (I have already done a uuid_bridge
> when
> >> > originate has ended).
> >> > The audio continue to be mute in both direction.
> >> > The "uuid_bridge" works only if I call it after received
> >> > "CHANNEL_ANSWER" in
> >> > legB or after I send "answer" to legA.
> >> > It's not enough to do a "pre_answer".
> >> > Stephen
> >> >
> >> > On Tue, Feb 15, 2011 at 6:15 PM, Anthony Minessale
> >> > <anthony.minessale at gmail.com> wrote:
> >> >>
> >> >> What happens once they answer?
> >> >>
> >> >> This is not one of those attempts to speed up click to call by
> calling
> >> >> them both at once is it?
> >> >>
> >> >>
> >> >> On Tue, Feb 15, 2011 at 10:29 AM, Stephen Wilde <
> wstephen80 at gmail.com>
> >> >> wrote:
> >> >> > Ok, I have tried with "pre_answer" to the inbound channel (that
> will
> >> >> > be
> >> >> > the
> >> >> > first uuid in the "uuid_bridge") but with same result: no audio
> from
> >> >> > outbound (in progressing media state) to inbound (in progressing
> >> >> > media
> >> >> > state
> >> >> > due to my pre_answer).
> >> >> > Here the sequence of events:
> >> >> > [ERROR] newacme.cpp:193 mycallback() Connected!
> >> >> > [ERROR] newacme.cpp:113 Originate() bgapi originate
> >> >> >
> >> >> >
> >> >> >
> [origination_uuid=61e00b94-f4d3-4ac4-bcf5-8fba02a18de6]sofia/external/xxxx at yyyy
> >> >> > &park()
> >> >> > [ERROR] newacme.cpp:115 Originate() Job-ID:
> >> >> > eee79d97-dbf8-49d4-9bf9-578ef75f73ea
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_UUID]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_OUTGOING]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_ORIGINATE]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_STATE]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_CALLSTATE]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_STATE]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_STATE]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CALL_UPDATE]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CODEC]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CODEC]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_CALLSTATE]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_PROGRESS_MEDIA]
> >> >> > [ERROR] newacme.cpp:250 mycallback() Executing 'pre_answer' on
> >> >> > INBOUND
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_STATE]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_EXECUTE]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_PARK]
> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_EXECUTE]
> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_CALLSTATE]
> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_PROGRESS_MEDIA]
> >> >> > [ERROR] newacme.cpp:127 BridgeTo() bgapi uuid_bridge
> >> >> > 902e2cd4-10ee-4938-97cf-c9eb2ac969f8
> >> >> > 61e00b94-f4d3-4ac4-bcf5-8fba02a18de6
> >> >> > [ERROR] newacme.cpp:129 BridgeTo() Job-ID:
> >> >> > 7a66a5c7-721c-47f4-aff2-373ec20123c8
> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CALL_UPDATE]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_CALLSTATE]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_ANSWER]
> >> >> > [ERROR] newacme.cpp:255 mycallback() Executing 'answer' on INBOUND
> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_EXECUTE]
> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_CALLSTATE]
> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_ANSWER]
> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_CALLSTATE]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_HANGUP]
> >> >> > [ERROR] newacme.cpp:260 mycallback() Executing 'hangup' on INBOUND
> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_EXECUTE]
> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_CALLSTATE]
> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_HANGUP]
> >> >> > [ERROR] newacme.cpp:272 mycallback() Executing 'hangup' on OUTBOUND
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_UNPARK]
> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_STATE]
> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_UNPARK]
> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_STATE]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_STATE]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_HANGUP_COMPLETE]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_DESTROY]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_CALLSTATE]
> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> > [CHANNEL_STATE]
> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_STATE]
> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> > [CHANNEL_HANGUP_COMPLETE]
> >> >> > [ERROR] newacme.cpp:290 mycallback() End.
> >> >> >
> >> >> >
> >> >> > On Tue, Feb 15, 2011 at 5:09 PM, Anthony Minessale
> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >>
> >> >> >> Well you can't bridge 2 unanswered channels together.
> >> >> >> You need at least one of them to have been pre_answered and that
> >> >> >> should be the first uuid in the list.
> >> >> >>
> >> >> >>
> >> >> >> On Tue, Feb 15, 2011 at 3:48 AM, Stephen Wilde
> >> >> >> <wstephen80 at gmail.com>
> >> >> >> wrote:
> >> >> >> > I have tried to wait originate completition but the
> "uuid_bridge"
> >> >> >> > doesn't
> >> >> >> > works also in this case.
> >> >> >> > It seems that works only when at least one leg is answered so
> it's
> >> >> >> > not
> >> >> >> > possible to do the "uuid_bridge" during progressing phase also
> if
> >> >> >> > the
> >> >> >> > originate has ended (I don't set the ignore_early_media).
> >> >> >> > My application is this: http://pastebin.freeswitch.org/15387
> >> >> >> > The application:
> >> >> >> > 1. receive a call with an "inbound_uuid"
> >> >> >> > 2. create a new "outbound_uuid"
> >> >> >> > 3. do a "bgapi originate" using the new "outbound_uuid"
> >> >> >> > 4. when the called phone is ringing, receive a
> >> >> >> > "CHANNEL_PROGRESS_MEDIA"
> >> >> >> > event on "outbound_uuid" and do a "pre_answer" on "inbound_uuid"
> >> >> >> > 5. receive a "CHANNEL_PROGRESS_MEDIA" on "inbound_uuid"
> >> >> >> > 6. receive a job completition event related to the "originate"
> so
> >> >> >> > issue
> >> >> >> > an
> >> >> >> > "uuid_bridge inbound_uuid outbound_uuid"
> >> >> >> > 7. when a job completition event related to "uuid_bridge" is
> >> >> >> > received,
> >> >> >> > no
> >> >> >> > audio flow from outbound to inbound channel
> >> >> >> > 8. when outbound answer the call, the application answer also
> the
> >> >> >> > inbound
> >> >> >> > call but no audio flow in both directions
> >> >> >> > If I do the uuid_bridge after legB answer, then all is ok
> >> >> >> > (obviously
> >> >> >> > with no
> >> >> >> > audio during progressing phase).
> >> >> >> >
> >> >> >> > The log of my application is:
> >> >> >> > [ERROR] newacme.cpp:46 mycallback() Connected!
> >> >> >> > [ERROR] newacme.cpp:65 mycallback() Inbound uuid =
> >> >> >> > 8b2c39db-1ad9-489c-b72f-a92d4087bf99
> >> >> >> > [ERROR] newacme.cpp:68 mycallback() create_uuid:
> >> >> >> > 394167aa-2811-4fcd-95c9-85576bdd9a7a
> >> >> >> > [ERROR] newacme.cpp:89 mycallback() bgapi originate
> >> >> >> >
> >> >> >> >
> >> >> >> >
> >> >> >> >
> [origination_uuid=394167aa-2811-4fcd-95c9-85576bdd9a7a]sofia/external/xxxx at yyyy
> >> >> >> > &park()
> >> >> >> > [ERROR] newacme.cpp:91 mycallback() Job-ID:
> >> >> >> > 1c654ac2-c22d-418f-9fad-bb6b6b35aeff
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_UUID]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_OUTGOING]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_ORIGINATE]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_STATE]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_STATE]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_STATE]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CALL_UPDATE]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CODEC]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CODEC]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_PROGRESS_MEDIA]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_STATE]
> >> >> >> > [ERROR] newacme.cpp:112 mycallback() Got an event:
> >> >> >> > [BACKGROUND_JOB] -
> >> >> >> > [1c654ac2-c22d-418f-9fad-bb6b6b35aeff]
> >> >> >> > [ERROR] newacme.cpp:119 mycallback() bgapi uuid_bridge
> >> >> >> > 8b2c39db-1ad9-489c-b72f-a92d4087bf99
> >> >> >> > 394167aa-2811-4fcd-95c9-85576bdd9a7a
> >> >> >> > [ERROR] newacme.cpp:123 mycallback() Job-ID:
> >> >> >> > 0d43cffe-fdc6-4a13-97e6-aeee199c45d5
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_EXECUTE]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_PARK]
> >> >> >> > [ERROR] newacme.cpp:112 mycallback() Got an event:
> >> >> >> > [BACKGROUND_JOB] -
> >> >> >> > [0d43cffe-fdc6-4a13-97e6-aeee199c45d5]
> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_EXECUTE]
> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_PROGRESS_MEDIA]
> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CALL_UPDATE]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_ANSWER]
> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_EXECUTE]
> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_ANSWER]
> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_HANGUP]
> >> >> >> > [ERROR] newacme.cpp:160 mycallback() hangup
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_UNPARK]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_STATE]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_STATE]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_HANGUP_COMPLETE]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_DESTROY]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_STATE]
> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_EXECUTE]
> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_HANGUP]
> >> >> >> > [ERROR] newacme.cpp:171 mycallback() hangup
> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_UNPARK]
> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_STATE]
> >> >> >> > [ERROR] newacme.cpp:184 mycallback() End.
> >> >> >> >
> >> >> >> > Stephen
> >> >> >> >
> >> >> >> > On Tue, Feb 15, 2011 at 1:14 AM, Anthony Minessale
> >> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >> >>
> >> >> >> >> if you do originate without ignore_early_media=true set it will
> >> >> >> >> end
> >> >> >> >> the soonest possible where it's suitable for a bridge.
> >> >> >> >> so that is the best bet to wait for originate to end.
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> On Mon, Feb 14, 2011 at 6:11 PM, Stephen Wilde
> >> >> >> >> <wstephen80 at gmail.com>
> >> >> >> >> wrote:
> >> >> >> >> > Ok, now I'm doing bridge when I receive "CHANNEL_ANSWER" or
> >> >> >> >> > "CHANNEL_PROGRESS_MEDIA" on outbound leg but you say that I
> >> >> >> >> > have
> >> >> >> >> > wait
> >> >> >> >> > that
> >> >> >> >> > "originate has ended": you mean that I have to wait for
> >> >> >> >> > "BACKGROUND_JOB"
> >> >> >> >> > event related to my "bgapi originate ... &park"?
> >> >> >> >> > I'm already doing "uuid_bridge <inbound_uuid>
> <outbound_uuid>".
> >> >> >> >> > I'll try also with intercept and inline originate. Thank you!
> >> >> >> >> > Stephen
> >> >> >> >> >
> >> >> >> >> > On Tue, Feb 15, 2011 at 12:53 AM, Anthony Minessale
> >> >> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >> >> >>
> >> >> >> >> >> you need to make sure originate has ended on the outbound
> leg
> >> >> >> >> >> before
> >> >> >> >> >> you use it in a bridge etc.
> >> >> >> >> >> you also need to supply the inbound leg first in uuid_bridge
> >> >> >> >> >> if
> >> >> >> >> >> that
> >> >> >> >> >> is something you want to do.
> >> >> >> >> >>
> >> >> >> >> >> Easier would be to originate the B leg to park inline and
> tell
> >> >> >> >> >> A
> >> >> >> >> >> leg
> >> >> >> >> >> to execute intercept on the B leg uuid.
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >> On Mon, Feb 14, 2011 at 5:44 PM, Stephen Wilde
> >> >> >> >> >> <wstephen80 at gmail.com>
> >> >> >> >> >> wrote:
> >> >> >> >> >> > Sorry but I have missed something.
> >> >> >> >> >> > I know that I can solve this problem directly in dialplan
> >> >> >> >> >> > with
> >> >> >> >> >> > a
> >> >> >> >> >> > bridge
> >> >> >> >> >> > but
> >> >> >> >> >> > what I'm doing is an "Freeswitch ESL learning" because I
> >> >> >> >> >> > have
> >> >> >> >> >> > to
> >> >> >> >> >> > port
> >> >> >> >> >> > some
> >> >> >> >> >> > application in Freeswitch and I'm learning how to
> implement
> >> >> >> >> >> > some
> >> >> >> >> >> > functionality.
> >> >> >> >> >> > For me it's important to take control of both
> >> >> >> >> >> > inbound/outbound
> >> >> >> >> >> > in
> >> >> >> >> >> > full
> >> >> >> >> >> > async
> >> >> >> >> >> > way and I have the necessity to do the complete call
> >> >> >> >> >> > control.
> >> >> >> >> >> > I'm not sure but to me it seems that with a normal bridge
> I
> >> >> >> >> >> > lose
> >> >> >> >> >> > the
> >> >> >> >> >> > control
> >> >> >> >> >> > of two sessions, for example, an outbound answer is
> >> >> >> >> >> > propagated
> >> >> >> >> >> > by
> >> >> >> >> >> > bridge
> >> >> >> >> >> > application as inbound answer.
> >> >> >> >> >> > What I want to do is an audio bridging so my application
> can
> >> >> >> >> >> > take
> >> >> >> >> >> > control of
> >> >> >> >> >> > "signaling bridging".
> >> >> >> >> >> >
> >> >> >> >> >> > I'm wrong? There are other way to do that?
> >> >> >> >> >> > Stephen
> >> >> >> >> >> > On Tue, Feb 15, 2011 at 12:19 AM, Michael Collins
> >> >> >> >> >> > <msc at freeswitch.org>
> >> >> >> >> >> > wrote:
> >> >> >> >> >> >>
> >> >> >> >> >> >> My head is spinning after reading this email. :)
> >> >> >> >> >> >> It sounds like you just need a simple bridge from the
> >> >> >> >> >> >> incoming
> >> >> >> >> >> >> leg
> >> >> >> >> >> >> to
> >> >> >> >> >> >> the
> >> >> >> >> >> >> outgoing leg. Can you pre_answer the A leg then execute a
> >> >> >> >> >> >> good
> >> >> >> >> >> >> old-fashioned
> >> >> >> >> >> >> bridge to the b-leg?
> >> >> >> >> >> >> -MC
> >> >> >> >> >> >> On Mon, Feb 14, 2011 at 4:57 PM, Stephen Wilde
> >> >> >> >> >> >> <wstephen80 at gmail.com>
> >> >> >> >> >> >> wrote:
> >> >> >> >> >> >>>
> >> >> >> >> >> >>> The problem I have is to propagate the audio coming from
> >> >> >> >> >> >>> an
> >> >> >> >> >> >>> "originated"
> >> >> >> >> >> >>> outbound session to the inbound session when the
> outbound
> >> >> >> >> >> >>> is
> >> >> >> >> >> >>> in
> >> >> >> >> >> >>> the
> >> >> >> >> >> >>> PROGRESS
> >> >> >> >> >> >>> MEDIA phase.
> >> >> >> >> >> >>> When my application receives the
> "CHANNEL_PROGRESS_MEDIA"
> >> >> >> >> >> >>> event
> >> >> >> >> >> >>> from
> >> >> >> >> >> >>> outbound session I can do a "pre_answer" on inbound
> >> >> >> >> >> >>> session
> >> >> >> >> >> >>> but
> >> >> >> >> >> >>> I'm
> >> >> >> >> >> >>> not
> >> >> >> >> >> >>> capable to do an audio bridge.
> >> >> >> >> >> >>> I have tried with "uuid_bridge <outbound_uuid>
> >> >> >> >> >> >>> <inbound_uuid>"
> >> >> >> >> >> >>> with
> >> >> >> >> >> >>> no
> >> >> >> >> >> >>> result probably because this api requires that at least
> >> >> >> >> >> >>> one
> >> >> >> >> >> >>> session
> >> >> >> >> >> >>> must be
> >> >> >> >> >> >>> answered.
> >> >> >> >> >> >>> I don't want to answer to the inbound session to
> propagate
> >> >> >> >> >> >>> the
> >> >> >> >> >> >>> outbound
> >> >> >> >> >> >>> progressing media but I want to answer to inbound only
> on
> >> >> >> >> >> >>> outbound
> >> >> >> >> >> >>> answer.
> >> >> >> >> >> >>> Any way to do that?
> >> >> >> >> >> >>> 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
> >> >> >> >> >> >>
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> > _______________________________________________
> >> >> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> >> >> > FreeSWITCH-users at 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  Wed Feb 16 02:35:22 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 15 Feb 2011 17:35:22 -0600
Subject: [Freeswitch-users] ESL Outbound: how to bridge media?
In-Reply-To: <AANLkTikftaVhR1jXO5sPqZTOwh9AAPGR4bOo764hPqw8@mail.gmail.com>
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Message-ID: <AANLkTimZ_1ACgw5O+zRN7iUZ2-LicrWfpGMjfdTEFcZp@mail.gmail.com>

That used to be a problem to bridge 2 unanswered channels.
I think it *might* let you now if you have the right conditions but it
may take some code and testing.

can you test this for me?

edit switch_ivr_bridge.c

erase or comment out the entire else block starting at line 1411

Try again.



On Tue, Feb 15, 2011 at 5:30 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:
> Sorry, I have now followed your indication doing the bridge only after
> receiving "CHANNEL_PARK" on legB.?The problem is the same with no audio.
> I have also tried to do a pre_answer in legA (before originate legB) with no
> success.
> I see in the fs_cli:
> [CRIT] switch_ivr_bridge.c:1412 Neither channel is answered, cannot bridge
> them.
> The sequence of events I receive before doing the bridge is:
> [OUTBOUND] - [CHANNEL_UUID]
> [OUTBOUND] - [CHANNEL_OUTGOING]
> [OUTBOUND] - [CHANNEL_ORIGINATE]
> [OUTBOUND] - [CHANNEL_STATE]
> [OUTBOUND] - [CHANNEL_CALLSTATE]
> [OUTBOUND] - [CHANNEL_STATE]
> [OUTBOUND] - [CHANNEL_STATE]
> [INBOUND] - [CHANNEL_EXECUTE]
> [INBOUND] - [CHANNEL_CALLSTATE]
> [INBOUND] - [CHANNEL_PROGRESS_MEDIA]
> [INBOUND] - [CHANNEL_EXECUTE_COMPLETE]
> [OUTBOUND] - [CALL_UPDATE]
> [OUTBOUND] - [CHANNEL_PROGRESS]
> [OUTBOUND] - [CALL_UPDATE]
> [OUTBOUND] - [CODEC]
> [OUTBOUND] - [CODEC]
> [OUTBOUND] - [CHANNEL_CALLSTATE]
> [OUTBOUND] - [CHANNEL_PROGRESS_MEDIA]
> [OUTBOUND] - [CHANNEL_STATE]
> [OUTBOUND] - [CHANNEL_EXECUTE]
> [OUTBOUND] - [CHANNEL_PARK]
> After this CHANNEL_PARK I do the "uuid_bridge" with no succes. These event
> are related to the case of putting legA in pre_answer. Same problem if I
> leave it in park.
> Stephen
>
> On Tue, Feb 15, 2011 at 11:32 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> Did you read the little chart i put in my last email?
>>
>> i'll be more specific:
>>
>> 1) leg A who has a_uuid has called inbound and has been put in park app.
>>
>> from ESL:
>> generate uuid (we'll ?call ?it b_uuid for clarity)
>> bgapi originate {origination_uuid=b_uuid}soifa/internal/bleg at box.com park
>> inline
>>
>> Wait for park event from b_uuid (do not touch it any sooner)
>>
>> now either do:
>>
>> uuid_bridge a_uuid b_uuid
>>
>> or:
>>
>> uuid_transfer a_uuid intercept:b_uuid inline
>>
>> or:
>>
>> sendmsg to a_uuid telling it to execute intercept app on b_uuid
>>
>>
>>
>>
>>
>> On Tue, Feb 15, 2011 at 3:46 PM, Stephen Wilde <wstephen80 at gmail.com>
>> wrote:
>> > When legB answers there is also no audio.
>> > If I do the bridge after legB answers the call (instead of when
>> > "origination" has ended), then the audio is ok?and the legA can ear the
>> > legB
>> > also if legA is in "pre_answer" state.
>> > Now I have found one problem: the "BACKGROUND_JOB" event related to
>> > "bgapi
>> > uuid_bridge ..." contains: "-ERR Invalid uuid" but?both uuid's are
>> > correct
>> > so probably is due to the state of channels.
>> > With the same uuid's, the uuid_bridge after the legB answer works fine.
>> > Stephen
>> >
>> > On Tue, Feb 15, 2011 at 9:53 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> I meant when you have no audio, and you wait until leg b answers. what
>> >> happens then?
>> >>
>> >>
>> >> If you have leg A inbound and you send it to park.
>> >> then you originate outbound to b leg to park
>> >> wait for park event on b leg
>> >> then you uuid_bridge them
>> >>
>> >> also you can sendmsg with application = intercept app_arg = uuid of B
>> >>
>> >>
>> >> On Tue, Feb 15, 2011 at 12:08 PM, Stephen Wilde <wstephen80 at gmail.com>
>> >> wrote:
>> >> > LegA is an incoming call, legB is an outgoing call.
>> >> > The dialplan is:
>> >> > ?? ?<extension name="myhandler">
>> >> > ?? ? ?<condition field="destination_number" expression="^xxxx$">
>> >> > ?? ? ? ?<action application="set"
>> >> > data="dialstring=sofia/external/yyyy at zzzz
>> >> > />
>> >> > ?? ? ? ?<action application="socket" data="127.0.0.1:8084 async
>> >> > full"/>
>> >> > ?? ? ?</condition>
>> >> > ?? ?</extension>
>> >> > When the legB answer the call, I receive the "CHANNEL_ANSWER" event
>> >> > on
>> >> > legB
>> >> > and I send the "answer" to legA (I have already done a uuid_bridge
>> >> > when
>> >> > originate has ended).
>> >> > The audio continue to be mute in both direction.
>> >> > The "uuid_bridge" works only if I call it after received
>> >> > "CHANNEL_ANSWER" in
>> >> > legB or after I send "answer" to legA.
>> >> > It's not enough to do a "pre_answer".
>> >> > Stephen
>> >> >
>> >> > On Tue, Feb 15, 2011 at 6:15 PM, Anthony Minessale
>> >> > <anthony.minessale at gmail.com> wrote:
>> >> >>
>> >> >> What happens once they answer?
>> >> >>
>> >> >> This is not one of those attempts to speed up click to call by
>> >> >> calling
>> >> >> them both at once is it?
>> >> >>
>> >> >>
>> >> >> On Tue, Feb 15, 2011 at 10:29 AM, Stephen Wilde
>> >> >> <wstephen80 at gmail.com>
>> >> >> wrote:
>> >> >> > Ok, I have tried with "pre_answer" to the inbound channel (that
>> >> >> > will
>> >> >> > be
>> >> >> > the
>> >> >> > first uuid in the "uuid_bridge")?but with same result: no audio
>> >> >> > from
>> >> >> > outbound (in progressing media state) to inbound (in progressing
>> >> >> > media
>> >> >> > state
>> >> >> > due to my pre_answer).
>> >> >> > Here the sequence of events:
>> >> >> > [ERROR] newacme.cpp:193 mycallback() Connected!
>> >> >> > [ERROR] newacme.cpp:113 Originate() bgapi originate
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > [origination_uuid=61e00b94-f4d3-4ac4-bcf5-8fba02a18de6]sofia/external/xxxx at yyyy
>> >> >> > &park()
>> >> >> > [ERROR] newacme.cpp:115 Originate() Job-ID:
>> >> >> > eee79d97-dbf8-49d4-9bf9-578ef75f73ea
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_UUID]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_OUTGOING]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_ORIGINATE]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_STATE]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_CALLSTATE]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_STATE]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_STATE]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CALL_UPDATE]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CODEC]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CODEC]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_CALLSTATE]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_PROGRESS_MEDIA]
>> >> >> > [ERROR] newacme.cpp:250 mycallback() Executing 'pre_answer' on
>> >> >> > INBOUND
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_STATE]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_EXECUTE]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_PARK]
>> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_EXECUTE]
>> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_CALLSTATE]
>> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_PROGRESS_MEDIA]
>> >> >> > [ERROR] newacme.cpp:127 BridgeTo() bgapi uuid_bridge
>> >> >> > 902e2cd4-10ee-4938-97cf-c9eb2ac969f8
>> >> >> > 61e00b94-f4d3-4ac4-bcf5-8fba02a18de6
>> >> >> > [ERROR] newacme.cpp:129 BridgeTo() Job-ID:
>> >> >> > 7a66a5c7-721c-47f4-aff2-373ec20123c8
>> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CALL_UPDATE]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_CALLSTATE]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_ANSWER]
>> >> >> > [ERROR] newacme.cpp:255 mycallback() Executing 'answer' on INBOUND
>> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_EXECUTE]
>> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_CALLSTATE]
>> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_ANSWER]
>> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_CALLSTATE]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_HANGUP]
>> >> >> > [ERROR] newacme.cpp:260 mycallback() Executing 'hangup' on INBOUND
>> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_EXECUTE]
>> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_CALLSTATE]
>> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_HANGUP]
>> >> >> > [ERROR] newacme.cpp:272 mycallback() Executing 'hangup' on
>> >> >> > OUTBOUND
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_UNPARK]
>> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_STATE]
>> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_UNPARK]
>> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_STATE]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_STATE]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_HANGUP_COMPLETE]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_DESTROY]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_CALLSTATE]
>> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> > [CHANNEL_STATE]
>> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_STATE]
>> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> > [CHANNEL_HANGUP_COMPLETE]
>> >> >> > [ERROR] newacme.cpp:290 mycallback() End.
>> >> >> >
>> >> >> >
>> >> >> > On Tue, Feb 15, 2011 at 5:09 PM, Anthony Minessale
>> >> >> > <anthony.minessale at gmail.com> wrote:
>> >> >> >>
>> >> >> >> Well you can't bridge 2 unanswered channels together.
>> >> >> >> You need at least one of them to have been pre_answered and that
>> >> >> >> should be the first uuid in the list.
>> >> >> >>
>> >> >> >>
>> >> >> >> On Tue, Feb 15, 2011 at 3:48 AM, Stephen Wilde
>> >> >> >> <wstephen80 at gmail.com>
>> >> >> >> wrote:
>> >> >> >> > I have tried to wait originate completition but the
>> >> >> >> > "uuid_bridge"
>> >> >> >> > doesn't
>> >> >> >> > works also in this case.
>> >> >> >> > It seems that works only when at least one leg is answered so
>> >> >> >> > it's
>> >> >> >> > not
>> >> >> >> > possible to do the "uuid_bridge" during progressing phase also
>> >> >> >> > if
>> >> >> >> > the
>> >> >> >> > originate has ended (I don't set the ignore_early_media).
>> >> >> >> > My application is this:?http://pastebin.freeswitch.org/15387
>> >> >> >> > The application:
>> >> >> >> > 1. receive a call with an "inbound_uuid"
>> >> >> >> > 2. create a new "outbound_uuid"
>> >> >> >> > 3. do a "bgapi originate" using the new "outbound_uuid"
>> >> >> >> > 4. when the called phone is ringing, receive a
>> >> >> >> > "CHANNEL_PROGRESS_MEDIA"
>> >> >> >> > event on "outbound_uuid" and do a "pre_answer" on
>> >> >> >> > "inbound_uuid"
>> >> >> >> > 5. receive a?"CHANNEL_PROGRESS_MEDIA" on "inbound_uuid"
>> >> >> >> > 6. receive a job completition event related to the "originate"
>> >> >> >> > so
>> >> >> >> > issue
>> >> >> >> > an
>> >> >> >> > "uuid_bridge inbound_uuid outbound_uuid"
>> >> >> >> > 7. when a job completition event related to "uuid_bridge" is
>> >> >> >> > received,
>> >> >> >> > no
>> >> >> >> > audio flow from outbound to inbound channel
>> >> >> >> > 8. when outbound answer the call, the application answer also
>> >> >> >> > the
>> >> >> >> > inbound
>> >> >> >> > call but no audio flow in both directions
>> >> >> >> > If I do the uuid_bridge after legB answer, then all is ok
>> >> >> >> > (obviously
>> >> >> >> > with no
>> >> >> >> > audio during progressing phase).
>> >> >> >> >
>> >> >> >> > The log of my application is:
>> >> >> >> > [ERROR] newacme.cpp:46 mycallback() Connected!
>> >> >> >> > [ERROR] newacme.cpp:65 mycallback() Inbound uuid =
>> >> >> >> > 8b2c39db-1ad9-489c-b72f-a92d4087bf99
>> >> >> >> > [ERROR] newacme.cpp:68 mycallback() create_uuid:
>> >> >> >> > 394167aa-2811-4fcd-95c9-85576bdd9a7a
>> >> >> >> > [ERROR] newacme.cpp:89 mycallback() bgapi originate
>> >> >> >> >
>> >> >> >> >
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > [origination_uuid=394167aa-2811-4fcd-95c9-85576bdd9a7a]sofia/external/xxxx at yyyy
>> >> >> >> > &park()
>> >> >> >> > [ERROR] newacme.cpp:91 mycallback() Job-ID:
>> >> >> >> > 1c654ac2-c22d-418f-9fad-bb6b6b35aeff
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_UUID]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_OUTGOING]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_ORIGINATE]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_STATE]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_CALLSTATE]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_STATE]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_STATE]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CALL_UPDATE]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CODEC]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CODEC]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_CALLSTATE]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_PROGRESS_MEDIA]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_STATE]
>> >> >> >> > [ERROR] newacme.cpp:112 mycallback() Got an event:
>> >> >> >> > [BACKGROUND_JOB] -
>> >> >> >> > [1c654ac2-c22d-418f-9fad-bb6b6b35aeff]
>> >> >> >> > [ERROR] newacme.cpp:119 mycallback() bgapi uuid_bridge
>> >> >> >> > 8b2c39db-1ad9-489c-b72f-a92d4087bf99
>> >> >> >> > 394167aa-2811-4fcd-95c9-85576bdd9a7a
>> >> >> >> > [ERROR] newacme.cpp:123 mycallback() Job-ID:
>> >> >> >> > 0d43cffe-fdc6-4a13-97e6-aeee199c45d5
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_EXECUTE]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_PARK]
>> >> >> >> > [ERROR] newacme.cpp:112 mycallback() Got an event:
>> >> >> >> > [BACKGROUND_JOB] -
>> >> >> >> > [0d43cffe-fdc6-4a13-97e6-aeee199c45d5]
>> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_EXECUTE]
>> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_CALLSTATE]
>> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_PROGRESS_MEDIA]
>> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CALL_UPDATE]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_CALLSTATE]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_ANSWER]
>> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_EXECUTE]
>> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_CALLSTATE]
>> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_ANSWER]
>> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_CALLSTATE]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_HANGUP]
>> >> >> >> > [ERROR] newacme.cpp:160 mycallback() hangup
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_UNPARK]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_STATE]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_STATE]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_HANGUP_COMPLETE]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_DESTROY]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_CALLSTATE]
>> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_STATE]
>> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_EXECUTE]
>> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_CALLSTATE]
>> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_HANGUP]
>> >> >> >> > [ERROR] newacme.cpp:171 mycallback() hangup
>> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_UNPARK]
>> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_STATE]
>> >> >> >> > [ERROR] newacme.cpp:184 mycallback() End.
>> >> >> >> >
>> >> >> >> > Stephen
>> >> >> >> >
>> >> >> >> > On Tue, Feb 15, 2011 at 1:14 AM, Anthony Minessale
>> >> >> >> > <anthony.minessale at gmail.com> wrote:
>> >> >> >> >>
>> >> >> >> >> if you do originate without ignore_early_media=true set it
>> >> >> >> >> will
>> >> >> >> >> end
>> >> >> >> >> the soonest possible where it's suitable for a bridge.
>> >> >> >> >> so that is the best bet to wait for originate to end.
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> On Mon, Feb 14, 2011 at 6:11 PM, Stephen Wilde
>> >> >> >> >> <wstephen80 at gmail.com>
>> >> >> >> >> wrote:
>> >> >> >> >> > Ok, now I'm doing bridge when I receive "CHANNEL_ANSWER" or
>> >> >> >> >> > "CHANNEL_PROGRESS_MEDIA" on outbound leg but you say that I
>> >> >> >> >> > have
>> >> >> >> >> > wait
>> >> >> >> >> > that
>> >> >> >> >> > "originate has ended": you mean that I have to wait for
>> >> >> >> >> > "BACKGROUND_JOB"
>> >> >> >> >> > event related to my "bgapi originate ... &park"?
>> >> >> >> >> > I'm already doing "uuid_bridge <inbound_uuid>
>> >> >> >> >> > <outbound_uuid>".
>> >> >> >> >> > I'll try also with intercept and inline originate.?Thank
>> >> >> >> >> > you!
>> >> >> >> >> > Stephen
>> >> >> >> >> >
>> >> >> >> >> > On Tue, Feb 15, 2011 at 12:53 AM, Anthony Minessale
>> >> >> >> >> > <anthony.minessale at gmail.com> wrote:
>> >> >> >> >> >>
>> >> >> >> >> >> you need to make sure originate has ended on the outbound
>> >> >> >> >> >> leg
>> >> >> >> >> >> before
>> >> >> >> >> >> you use it in a bridge etc.
>> >> >> >> >> >> you also need to supply the inbound leg first in
>> >> >> >> >> >> uuid_bridge
>> >> >> >> >> >> if
>> >> >> >> >> >> that
>> >> >> >> >> >> is something you want to do.
>> >> >> >> >> >>
>> >> >> >> >> >> Easier would be to originate the B leg to park inline and
>> >> >> >> >> >> tell
>> >> >> >> >> >> A
>> >> >> >> >> >> leg
>> >> >> >> >> >> to execute intercept on the B leg uuid.
>> >> >> >> >> >>
>> >> >> >> >> >>
>> >> >> >> >> >>
>> >> >> >> >> >> On Mon, Feb 14, 2011 at 5:44 PM, Stephen Wilde
>> >> >> >> >> >> <wstephen80 at gmail.com>
>> >> >> >> >> >> wrote:
>> >> >> >> >> >> > Sorry but I have missed something.
>> >> >> >> >> >> > I know that I can solve this problem directly in dialplan
>> >> >> >> >> >> > with
>> >> >> >> >> >> > a
>> >> >> >> >> >> > bridge
>> >> >> >> >> >> > but
>> >> >> >> >> >> > what I'm doing is an "Freeswitch ESL learning"?because I
>> >> >> >> >> >> > have
>> >> >> >> >> >> > to
>> >> >> >> >> >> > port
>> >> >> >> >> >> > some
>> >> >> >> >> >> > application in Freeswitch and I'm learning how to
>> >> >> >> >> >> > implement
>> >> >> >> >> >> > some
>> >> >> >> >> >> > functionality.
>> >> >> >> >> >> > For me it's important to take control of both
>> >> >> >> >> >> > inbound/outbound
>> >> >> >> >> >> > in
>> >> >> >> >> >> > full
>> >> >> >> >> >> > async
>> >> >> >> >> >> > way and I have the necessity to do the complete call
>> >> >> >> >> >> > control.
>> >> >> >> >> >> > I'm not sure but to me it seems that with a normal bridge
>> >> >> >> >> >> > I
>> >> >> >> >> >> > lose
>> >> >> >> >> >> > the
>> >> >> >> >> >> > control
>> >> >> >> >> >> > of two sessions, for example, an outbound answer is
>> >> >> >> >> >> > propagated
>> >> >> >> >> >> > by
>> >> >> >> >> >> > bridge
>> >> >> >> >> >> > application as inbound answer.
>> >> >> >> >> >> > What I want to do is an audio bridging so my application
>> >> >> >> >> >> > can
>> >> >> >> >> >> > take
>> >> >> >> >> >> > control of
>> >> >> >> >> >> > "signaling bridging".
>> >> >> >> >> >> >
>> >> >> >> >> >> > I'm wrong? There are other way to do that?
>> >> >> >> >> >> > Stephen
>> >> >> >> >> >> > On Tue, Feb 15, 2011 at 12:19 AM, Michael Collins
>> >> >> >> >> >> > <msc at freeswitch.org>
>> >> >> >> >> >> > wrote:
>> >> >> >> >> >> >>
>> >> >> >> >> >> >> My head is spinning after reading this email. :)
>> >> >> >> >> >> >> It sounds like you just need a simple bridge from the
>> >> >> >> >> >> >> incoming
>> >> >> >> >> >> >> leg
>> >> >> >> >> >> >> to
>> >> >> >> >> >> >> the
>> >> >> >> >> >> >> outgoing leg. Can you pre_answer the A leg then execute
>> >> >> >> >> >> >> a
>> >> >> >> >> >> >> good
>> >> >> >> >> >> >> old-fashioned
>> >> >> >> >> >> >> bridge to the b-leg?
>> >> >> >> >> >> >> -MC
>> >> >> >> >> >> >> On Mon, Feb 14, 2011 at 4:57 PM, Stephen Wilde
>> >> >> >> >> >> >> <wstephen80 at gmail.com>
>> >> >> >> >> >> >> wrote:
>> >> >> >> >> >> >>>
>> >> >> >> >> >> >>> The problem I have is to propagate the audio coming
>> >> >> >> >> >> >>> from
>> >> >> >> >> >> >>> an
>> >> >> >> >> >> >>> "originated"
>> >> >> >> >> >> >>> outbound session to the inbound session when the
>> >> >> >> >> >> >>> outbound
>> >> >> >> >> >> >>> is
>> >> >> >> >> >> >>> in
>> >> >> >> >> >> >>> the
>> >> >> >> >> >> >>> PROGRESS
>> >> >> >> >> >> >>> MEDIA phase.
>> >> >> >> >> >> >>> When my application receives the
>> >> >> >> >> >> >>> "CHANNEL_PROGRESS_MEDIA"
>> >> >> >> >> >> >>> event
>> >> >> >> >> >> >>> from
>> >> >> >> >> >> >>> outbound session I can do a "pre_answer" on inbound
>> >> >> >> >> >> >>> session
>> >> >> >> >> >> >>> but
>> >> >> >> >> >> >>> I'm
>> >> >> >> >> >> >>> not
>> >> >> >> >> >> >>> capable to do an audio bridge.
>> >> >> >> >> >> >>> I have tried with "uuid_bridge <outbound_uuid>
>> >> >> >> >> >> >>> <inbound_uuid>"
>> >> >> >> >> >> >>> with
>> >> >> >> >> >> >>> no
>> >> >> >> >> >> >>> result probably because this api requires that at least
>> >> >> >> >> >> >>> one
>> >> >> >> >> >> >>> session
>> >> >> >> >> >> >>> must be
>> >> >> >> >> >> >>> answered.
>> >> >> >> >> >> >>> I don't want to answer to the inbound session to
>> >> >> >> >> >> >>> propagate
>> >> >> >> >> >> >>> the
>> >> >> >> >> >> >>> outbound
>> >> >> >> >> >> >>> progressing media but I want to answer to inbound only
>> >> >> >> >> >> >>> on
>> >> >> >> >> >> >>> outbound
>> >> >> >> >> >> >>> answer.
>> >> >> >> >> >> >>> Any way to do that?
>> >> >> >> >> >> >>> 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
>> >> >> >> >> >> >>
>> >> >> >> >> >> >
>> >> >> >> >> >> >
>> >> >> >> >> >> > _______________________________________________
>> >> >> >> >> >> > FreeSWITCH-users mailing list
>> >> >> >> >> >> > FreeSWITCH-users at 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
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>> >> >> >> >
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>> >> >> >> >
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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
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>> >> >> >> 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
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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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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/
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Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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  Wed Feb 16 02:49:25 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Wed, 16 Feb 2011 00:49:25 +0100
Subject: [Freeswitch-users] ESL Outbound: how to bridge media?
In-Reply-To: <AANLkTimZ_1ACgw5O+zRN7iUZ2-LicrWfpGMjfdTEFcZp@mail.gmail.com>
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Message-ID: <AANLkTimLgZDK8jTdRPqrxN3Fhe8grKCHyB+=9H200vVH@mail.gmail.com>

Whow! With this change I can ear progressing messages! It works now!

Stephen

On Wed, Feb 16, 2011 at 12:35 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> That used to be a problem to bridge 2 unanswered channels.
> I think it *might* let you now if you have the right conditions but it
> may take some code and testing.
>
> can you test this for me?
>
> edit switch_ivr_bridge.c
>
> erase or comment out the entire else block starting at line 1411
>
> Try again.
>
>
>
> On Tue, Feb 15, 2011 at 5:30 PM, Stephen Wilde <wstephen80 at gmail.com>
> wrote:
> > Sorry, I have now followed your indication doing the bridge only after
> > receiving "CHANNEL_PARK" on legB. The problem is the same with no audio.
> > I have also tried to do a pre_answer in legA (before originate legB) with
> no
> > success.
> > I see in the fs_cli:
> > [CRIT] switch_ivr_bridge.c:1412 Neither channel is answered, cannot
> bridge
> > them.
> > The sequence of events I receive before doing the bridge is:
> > [OUTBOUND] - [CHANNEL_UUID]
> > [OUTBOUND] - [CHANNEL_OUTGOING]
> > [OUTBOUND] - [CHANNEL_ORIGINATE]
> > [OUTBOUND] - [CHANNEL_STATE]
> > [OUTBOUND] - [CHANNEL_CALLSTATE]
> > [OUTBOUND] - [CHANNEL_STATE]
> > [OUTBOUND] - [CHANNEL_STATE]
> > [INBOUND] - [CHANNEL_EXECUTE]
> > [INBOUND] - [CHANNEL_CALLSTATE]
> > [INBOUND] - [CHANNEL_PROGRESS_MEDIA]
> > [INBOUND] - [CHANNEL_EXECUTE_COMPLETE]
> > [OUTBOUND] - [CALL_UPDATE]
> > [OUTBOUND] - [CHANNEL_PROGRESS]
> > [OUTBOUND] - [CALL_UPDATE]
> > [OUTBOUND] - [CODEC]
> > [OUTBOUND] - [CODEC]
> > [OUTBOUND] - [CHANNEL_CALLSTATE]
> > [OUTBOUND] - [CHANNEL_PROGRESS_MEDIA]
> > [OUTBOUND] - [CHANNEL_STATE]
> > [OUTBOUND] - [CHANNEL_EXECUTE]
> > [OUTBOUND] - [CHANNEL_PARK]
> > After this CHANNEL_PARK I do the "uuid_bridge" with no succes. These
> event
> > are related to the case of putting legA in pre_answer. Same problem if I
> > leave it in park.
> > Stephen
> >
> > On Tue, Feb 15, 2011 at 11:32 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> Did you read the little chart i put in my last email?
> >>
> >> i'll be more specific:
> >>
> >> 1) leg A who has a_uuid has called inbound and has been put in park app.
> >>
> >> from ESL:
> >> generate uuid (we'll  call  it b_uuid for clarity)
> >> bgapi originate {origination_uuid=b_uuid}soifa/internal/bleg at box.compark
> >> inline
> >>
> >> Wait for park event from b_uuid (do not touch it any sooner)
> >>
> >> now either do:
> >>
> >> uuid_bridge a_uuid b_uuid
> >>
> >> or:
> >>
> >> uuid_transfer a_uuid intercept:b_uuid inline
> >>
> >> or:
> >>
> >> sendmsg to a_uuid telling it to execute intercept app on b_uuid
> >>
> >>
> >>
> >>
> >>
> >> On Tue, Feb 15, 2011 at 3:46 PM, Stephen Wilde <wstephen80 at gmail.com>
> >> wrote:
> >> > When legB answers there is also no audio.
> >> > If I do the bridge after legB answers the call (instead of when
> >> > "origination" has ended), then the audio is ok and the legA can ear
> the
> >> > legB
> >> > also if legA is in "pre_answer" state.
> >> > Now I have found one problem: the "BACKGROUND_JOB" event related to
> >> > "bgapi
> >> > uuid_bridge ..." contains: "-ERR Invalid uuid" but both uuid's are
> >> > correct
> >> > so probably is due to the state of channels.
> >> > With the same uuid's, the uuid_bridge after the legB answer works
> fine.
> >> > Stephen
> >> >
> >> > On Tue, Feb 15, 2011 at 9:53 PM, Anthony Minessale
> >> > <anthony.minessale at gmail.com> wrote:
> >> >>
> >> >> I meant when you have no audio, and you wait until leg b answers.
> what
> >> >> happens then?
> >> >>
> >> >>
> >> >> If you have leg A inbound and you send it to park.
> >> >> then you originate outbound to b leg to park
> >> >> wait for park event on b leg
> >> >> then you uuid_bridge them
> >> >>
> >> >> also you can sendmsg with application = intercept app_arg = uuid of B
> >> >>
> >> >>
> >> >> On Tue, Feb 15, 2011 at 12:08 PM, Stephen Wilde <
> wstephen80 at gmail.com>
> >> >> wrote:
> >> >> > LegA is an incoming call, legB is an outgoing call.
> >> >> > The dialplan is:
> >> >> >     <extension name="myhandler">
> >> >> >       <condition field="destination_number" expression="^xxxx$">
> >> >> >         <action application="set"
> >> >> > data="dialstring=sofia/external/yyyy at zzzz
> >> >> > />
> >> >> >         <action application="socket" data="127.0.0.1:8084 async
> >> >> > full"/>
> >> >> >       </condition>
> >> >> >     </extension>
> >> >> > When the legB answer the call, I receive the "CHANNEL_ANSWER" event
> >> >> > on
> >> >> > legB
> >> >> > and I send the "answer" to legA (I have already done a uuid_bridge
> >> >> > when
> >> >> > originate has ended).
> >> >> > The audio continue to be mute in both direction.
> >> >> > The "uuid_bridge" works only if I call it after received
> >> >> > "CHANNEL_ANSWER" in
> >> >> > legB or after I send "answer" to legA.
> >> >> > It's not enough to do a "pre_answer".
> >> >> > Stephen
> >> >> >
> >> >> > On Tue, Feb 15, 2011 at 6:15 PM, Anthony Minessale
> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >>
> >> >> >> What happens once they answer?
> >> >> >>
> >> >> >> This is not one of those attempts to speed up click to call by
> >> >> >> calling
> >> >> >> them both at once is it?
> >> >> >>
> >> >> >>
> >> >> >> On Tue, Feb 15, 2011 at 10:29 AM, Stephen Wilde
> >> >> >> <wstephen80 at gmail.com>
> >> >> >> wrote:
> >> >> >> > Ok, I have tried with "pre_answer" to the inbound channel (that
> >> >> >> > will
> >> >> >> > be
> >> >> >> > the
> >> >> >> > first uuid in the "uuid_bridge") but with same result: no audio
> >> >> >> > from
> >> >> >> > outbound (in progressing media state) to inbound (in progressing
> >> >> >> > media
> >> >> >> > state
> >> >> >> > due to my pre_answer).
> >> >> >> > Here the sequence of events:
> >> >> >> > [ERROR] newacme.cpp:193 mycallback() Connected!
> >> >> >> > [ERROR] newacme.cpp:113 Originate() bgapi originate
> >> >> >> >
> >> >> >> >
> >> >> >> >
> >> >> >> >
> [origination_uuid=61e00b94-f4d3-4ac4-bcf5-8fba02a18de6]sofia/external/xxxx at yyyy
> >> >> >> > &park()
> >> >> >> > [ERROR] newacme.cpp:115 Originate() Job-ID:
> >> >> >> > eee79d97-dbf8-49d4-9bf9-578ef75f73ea
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_UUID]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_OUTGOING]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_ORIGINATE]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_STATE]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_STATE]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_STATE]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CALL_UPDATE]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CODEC]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CODEC]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_PROGRESS_MEDIA]
> >> >> >> > [ERROR] newacme.cpp:250 mycallback() Executing 'pre_answer' on
> >> >> >> > INBOUND
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_STATE]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_EXECUTE]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_PARK]
> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_EXECUTE]
> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_PROGRESS_MEDIA]
> >> >> >> > [ERROR] newacme.cpp:127 BridgeTo() bgapi uuid_bridge
> >> >> >> > 902e2cd4-10ee-4938-97cf-c9eb2ac969f8
> >> >> >> > 61e00b94-f4d3-4ac4-bcf5-8fba02a18de6
> >> >> >> > [ERROR] newacme.cpp:129 BridgeTo() Job-ID:
> >> >> >> > 7a66a5c7-721c-47f4-aff2-373ec20123c8
> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CALL_UPDATE]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_ANSWER]
> >> >> >> > [ERROR] newacme.cpp:255 mycallback() Executing 'answer' on
> INBOUND
> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_EXECUTE]
> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_ANSWER]
> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_HANGUP]
> >> >> >> > [ERROR] newacme.cpp:260 mycallback() Executing 'hangup' on
> INBOUND
> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_EXECUTE]
> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_HANGUP]
> >> >> >> > [ERROR] newacme.cpp:272 mycallback() Executing 'hangup' on
> >> >> >> > OUTBOUND
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_UNPARK]
> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_STATE]
> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_UNPARK]
> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_STATE]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_STATE]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_HANGUP_COMPLETE]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_DESTROY]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
> >> >> >> > [CHANNEL_STATE]
> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_STATE]
> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
> >> >> >> > [CHANNEL_HANGUP_COMPLETE]
> >> >> >> > [ERROR] newacme.cpp:290 mycallback() End.
> >> >> >> >
> >> >> >> >
> >> >> >> > On Tue, Feb 15, 2011 at 5:09 PM, Anthony Minessale
> >> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >> >>
> >> >> >> >> Well you can't bridge 2 unanswered channels together.
> >> >> >> >> You need at least one of them to have been pre_answered and
> that
> >> >> >> >> should be the first uuid in the list.
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> On Tue, Feb 15, 2011 at 3:48 AM, Stephen Wilde
> >> >> >> >> <wstephen80 at gmail.com>
> >> >> >> >> wrote:
> >> >> >> >> > I have tried to wait originate completition but the
> >> >> >> >> > "uuid_bridge"
> >> >> >> >> > doesn't
> >> >> >> >> > works also in this case.
> >> >> >> >> > It seems that works only when at least one leg is answered so
> >> >> >> >> > it's
> >> >> >> >> > not
> >> >> >> >> > possible to do the "uuid_bridge" during progressing phase
> also
> >> >> >> >> > if
> >> >> >> >> > the
> >> >> >> >> > originate has ended (I don't set the ignore_early_media).
> >> >> >> >> > My application is this: http://pastebin.freeswitch.org/15387
> >> >> >> >> > The application:
> >> >> >> >> > 1. receive a call with an "inbound_uuid"
> >> >> >> >> > 2. create a new "outbound_uuid"
> >> >> >> >> > 3. do a "bgapi originate" using the new "outbound_uuid"
> >> >> >> >> > 4. when the called phone is ringing, receive a
> >> >> >> >> > "CHANNEL_PROGRESS_MEDIA"
> >> >> >> >> > event on "outbound_uuid" and do a "pre_answer" on
> >> >> >> >> > "inbound_uuid"
> >> >> >> >> > 5. receive a "CHANNEL_PROGRESS_MEDIA" on "inbound_uuid"
> >> >> >> >> > 6. receive a job completition event related to the
> "originate"
> >> >> >> >> > so
> >> >> >> >> > issue
> >> >> >> >> > an
> >> >> >> >> > "uuid_bridge inbound_uuid outbound_uuid"
> >> >> >> >> > 7. when a job completition event related to "uuid_bridge" is
> >> >> >> >> > received,
> >> >> >> >> > no
> >> >> >> >> > audio flow from outbound to inbound channel
> >> >> >> >> > 8. when outbound answer the call, the application answer also
> >> >> >> >> > the
> >> >> >> >> > inbound
> >> >> >> >> > call but no audio flow in both directions
> >> >> >> >> > If I do the uuid_bridge after legB answer, then all is ok
> >> >> >> >> > (obviously
> >> >> >> >> > with no
> >> >> >> >> > audio during progressing phase).
> >> >> >> >> >
> >> >> >> >> > The log of my application is:
> >> >> >> >> > [ERROR] newacme.cpp:46 mycallback() Connected!
> >> >> >> >> > [ERROR] newacme.cpp:65 mycallback() Inbound uuid =
> >> >> >> >> > 8b2c39db-1ad9-489c-b72f-a92d4087bf99
> >> >> >> >> > [ERROR] newacme.cpp:68 mycallback() create_uuid:
> >> >> >> >> > 394167aa-2811-4fcd-95c9-85576bdd9a7a
> >> >> >> >> > [ERROR] newacme.cpp:89 mycallback() bgapi originate
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> >
> [origination_uuid=394167aa-2811-4fcd-95c9-85576bdd9a7a]sofia/external/xxxx at yyyy
> >> >> >> >> > &park()
> >> >> >> >> > [ERROR] newacme.cpp:91 mycallback() Job-ID:
> >> >> >> >> > 1c654ac2-c22d-418f-9fad-bb6b6b35aeff
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_UUID]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_OUTGOING]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_ORIGINATE]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CALL_UPDATE]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CODEC]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CODEC]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_PROGRESS_MEDIA]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> > [ERROR] newacme.cpp:112 mycallback() Got an event:
> >> >> >> >> > [BACKGROUND_JOB] -
> >> >> >> >> > [1c654ac2-c22d-418f-9fad-bb6b6b35aeff]
> >> >> >> >> > [ERROR] newacme.cpp:119 mycallback() bgapi uuid_bridge
> >> >> >> >> > 8b2c39db-1ad9-489c-b72f-a92d4087bf99
> >> >> >> >> > 394167aa-2811-4fcd-95c9-85576bdd9a7a
> >> >> >> >> > [ERROR] newacme.cpp:123 mycallback() Job-ID:
> >> >> >> >> > 0d43cffe-fdc6-4a13-97e6-aeee199c45d5
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_EXECUTE]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_PARK]
> >> >> >> >> > [ERROR] newacme.cpp:112 mycallback() Got an event:
> >> >> >> >> > [BACKGROUND_JOB] -
> >> >> >> >> > [0d43cffe-fdc6-4a13-97e6-aeee199c45d5]
> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_EXECUTE]
> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_PROGRESS_MEDIA]
> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CALL_UPDATE]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_ANSWER]
> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_EXECUTE]
> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_ANSWER]
> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_HANGUP]
> >> >> >> >> > [ERROR] newacme.cpp:160 mycallback() hangup
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_UNPARK]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_HANGUP_COMPLETE]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_DESTROY]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_EXECUTE]
> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_HANGUP]
> >> >> >> >> > [ERROR] newacme.cpp:171 mycallback() hangup
> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_UNPARK]
> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> > [ERROR] newacme.cpp:184 mycallback() End.
> >> >> >> >> >
> >> >> >> >> > Stephen
> >> >> >> >> >
> >> >> >> >> > On Tue, Feb 15, 2011 at 1:14 AM, Anthony Minessale
> >> >> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >> >> >>
> >> >> >> >> >> if you do originate without ignore_early_media=true set it
> >> >> >> >> >> will
> >> >> >> >> >> end
> >> >> >> >> >> the soonest possible where it's suitable for a bridge.
> >> >> >> >> >> so that is the best bet to wait for originate to end.
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >> On Mon, Feb 14, 2011 at 6:11 PM, Stephen Wilde
> >> >> >> >> >> <wstephen80 at gmail.com>
> >> >> >> >> >> wrote:
> >> >> >> >> >> > Ok, now I'm doing bridge when I receive "CHANNEL_ANSWER"
> or
> >> >> >> >> >> > "CHANNEL_PROGRESS_MEDIA" on outbound leg but you say that
> I
> >> >> >> >> >> > have
> >> >> >> >> >> > wait
> >> >> >> >> >> > that
> >> >> >> >> >> > "originate has ended": you mean that I have to wait for
> >> >> >> >> >> > "BACKGROUND_JOB"
> >> >> >> >> >> > event related to my "bgapi originate ... &park"?
> >> >> >> >> >> > I'm already doing "uuid_bridge <inbound_uuid>
> >> >> >> >> >> > <outbound_uuid>".
> >> >> >> >> >> > I'll try also with intercept and inline originate. Thank
> >> >> >> >> >> > you!
> >> >> >> >> >> > Stephen
> >> >> >> >> >> >
> >> >> >> >> >> > On Tue, Feb 15, 2011 at 12:53 AM, Anthony Minessale
> >> >> >> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >> >> >> >>
> >> >> >> >> >> >> you need to make sure originate has ended on the outbound
> >> >> >> >> >> >> leg
> >> >> >> >> >> >> before
> >> >> >> >> >> >> you use it in a bridge etc.
> >> >> >> >> >> >> you also need to supply the inbound leg first in
> >> >> >> >> >> >> uuid_bridge
> >> >> >> >> >> >> if
> >> >> >> >> >> >> that
> >> >> >> >> >> >> is something you want to do.
> >> >> >> >> >> >>
> >> >> >> >> >> >> Easier would be to originate the B leg to park inline and
> >> >> >> >> >> >> tell
> >> >> >> >> >> >> A
> >> >> >> >> >> >> leg
> >> >> >> >> >> >> to execute intercept on the B leg uuid.
> >> >> >> >> >> >>
> >> >> >> >> >> >>
> >> >> >> >> >> >>
> >> >> >> >> >> >> On Mon, Feb 14, 2011 at 5:44 PM, Stephen Wilde
> >> >> >> >> >> >> <wstephen80 at gmail.com>
> >> >> >> >> >> >> wrote:
> >> >> >> >> >> >> > Sorry but I have missed something.
> >> >> >> >> >> >> > I know that I can solve this problem directly in
> dialplan
> >> >> >> >> >> >> > with
> >> >> >> >> >> >> > a
> >> >> >> >> >> >> > bridge
> >> >> >> >> >> >> > but
> >> >> >> >> >> >> > what I'm doing is an "Freeswitch ESL learning" because
> I
> >> >> >> >> >> >> > have
> >> >> >> >> >> >> > to
> >> >> >> >> >> >> > port
> >> >> >> >> >> >> > some
> >> >> >> >> >> >> > application in Freeswitch and I'm learning how to
> >> >> >> >> >> >> > implement
> >> >> >> >> >> >> > some
> >> >> >> >> >> >> > functionality.
> >> >> >> >> >> >> > For me it's important to take control of both
> >> >> >> >> >> >> > inbound/outbound
> >> >> >> >> >> >> > in
> >> >> >> >> >> >> > full
> >> >> >> >> >> >> > async
> >> >> >> >> >> >> > way and I have the necessity to do the complete call
> >> >> >> >> >> >> > control.
> >> >> >> >> >> >> > I'm not sure but to me it seems that with a normal
> bridge
> >> >> >> >> >> >> > I
> >> >> >> >> >> >> > lose
> >> >> >> >> >> >> > the
> >> >> >> >> >> >> > control
> >> >> >> >> >> >> > of two sessions, for example, an outbound answer is
> >> >> >> >> >> >> > propagated
> >> >> >> >> >> >> > by
> >> >> >> >> >> >> > bridge
> >> >> >> >> >> >> > application as inbound answer.
> >> >> >> >> >> >> > What I want to do is an audio bridging so my
> application
> >> >> >> >> >> >> > can
> >> >> >> >> >> >> > take
> >> >> >> >> >> >> > control of
> >> >> >> >> >> >> > "signaling bridging".
> >> >> >> >> >> >> >
> >> >> >> >> >> >> > I'm wrong? There are other way to do that?
> >> >> >> >> >> >> > Stephen
> >> >> >> >> >> >> > On Tue, Feb 15, 2011 at 12:19 AM, Michael Collins
> >> >> >> >> >> >> > <msc at freeswitch.org>
> >> >> >> >> >> >> > wrote:
> >> >> >> >> >> >> >>
> >> >> >> >> >> >> >> My head is spinning after reading this email. :)
> >> >> >> >> >> >> >> It sounds like you just need a simple bridge from the
> >> >> >> >> >> >> >> incoming
> >> >> >> >> >> >> >> leg
> >> >> >> >> >> >> >> to
> >> >> >> >> >> >> >> the
> >> >> >> >> >> >> >> outgoing leg. Can you pre_answer the A leg then
> execute
> >> >> >> >> >> >> >> a
> >> >> >> >> >> >> >> good
> >> >> >> >> >> >> >> old-fashioned
> >> >> >> >> >> >> >> bridge to the b-leg?
> >> >> >> >> >> >> >> -MC
> >> >> >> >> >> >> >> On Mon, Feb 14, 2011 at 4:57 PM, Stephen Wilde
> >> >> >> >> >> >> >> <wstephen80 at gmail.com>
> >> >> >> >> >> >> >> wrote:
> >> >> >> >> >> >> >>>
> >> >> >> >> >> >> >>> The problem I have is to propagate the audio coming
> >> >> >> >> >> >> >>> from
> >> >> >> >> >> >> >>> an
> >> >> >> >> >> >> >>> "originated"
> >> >> >> >> >> >> >>> outbound session to the inbound session when the
> >> >> >> >> >> >> >>> outbound
> >> >> >> >> >> >> >>> is
> >> >> >> >> >> >> >>> in
> >> >> >> >> >> >> >>> the
> >> >> >> >> >> >> >>> PROGRESS
> >> >> >> >> >> >> >>> MEDIA phase.
> >> >> >> >> >> >> >>> When my application receives the
> >> >> >> >> >> >> >>> "CHANNEL_PROGRESS_MEDIA"
> >> >> >> >> >> >> >>> event
> >> >> >> >> >> >> >>> from
> >> >> >> >> >> >> >>> outbound session I can do a "pre_answer" on inbound
> >> >> >> >> >> >> >>> session
> >> >> >> >> >> >> >>> but
> >> >> >> >> >> >> >>> I'm
> >> >> >> >> >> >> >>> not
> >> >> >> >> >> >> >>> capable to do an audio bridge.
> >> >> >> >> >> >> >>> I have tried with "uuid_bridge <outbound_uuid>
> >> >> >> >> >> >> >>> <inbound_uuid>"
> >> >> >> >> >> >> >>> with
> >> >> >> >> >> >> >>> no
> >> >> >> >> >> >> >>> result probably because this api requires that at
> least
> >> >> >> >> >> >> >>> one
> >> >> >> >> >> >> >>> session
> >> >> >> >> >> >> >>> must be
> >> >> >> >> >> >> >>> answered.
> >> >> >> >> >> >> >>> I don't want to answer to the inbound session to
> >> >> >> >> >> >> >>> propagate
> >> >> >> >> >> >> >>> the
> >> >> >> >> >> >> >>> outbound
> >> >> >> >> >> >> >>> progressing media but I want to answer to inbound
> only
> >> >> >> >> >> >> >>> on
> >> >> >> >> >> >> >>> outbound
> >> >> >> >> >> >> >>> answer.
> >> >> >> >> >> >> >>> Any way to do that?
> >> >> >> >> >> >> >>> 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
> >> >> >> >> >> >> >>
> >> >> >> >> >> >> >
> >> >> >> >> >> >> >
> >> >> >> >> >> >> > _______________________________________________
> >> >> >> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> >> >> >> > FreeSWITCH-users at 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:
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> >> Anthony Minessale II
> >>
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> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
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> >> IRC: irc.freenode.net #freeswitch
> >>
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> >> sip:888 at conference.freeswitch.org
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From anthony.minessale at gmail.com  Wed Feb 16 02:56:21 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 15 Feb 2011 17:56:21 -0600
Subject: [Freeswitch-users] ESL Outbound: how to bridge media?
In-Reply-To: <AANLkTimLgZDK8jTdRPqrxN3Fhe8grKCHyB+=9H200vVH@mail.gmail.com>
References: <AANLkTikLVLpSGkrRqeD3WVFZZgSaLnQvypCOG39hEhLX@mail.gmail.com>
	<AANLkTimwX8Uz4t9gp6Ppzfu+WE+UJa3FO4qHE7XKFOBz@mail.gmail.com>
	<AANLkTi=t-a7U9ov3wJ6Dh-PuniSfZbyeQ==BCAvMkUPN@mail.gmail.com>
	<AANLkTik27F7JjyMRLKryQZq_N+oVDsDzdmiPPa+d2KWN@mail.gmail.com>
	<AANLkTine=jG-dc98pDySashEvo3T5cSPwVXYwp54mhhC@mail.gmail.com>
	<AANLkTi=JAt=RkQ5b=DBajH_YR6TmiqExdjgF1=OL=fJ_@mail.gmail.com>
	<AANLkTinhkOznPh98kkHp=rCU9Qm0xv_hW3c41rRJr=Kn@mail.gmail.com>
	<AANLkTikAJoP7_cmhoEDorq4uCTdw+SVQeMNMpcFBkdYn@mail.gmail.com>
	<AANLkTi=XbutFHsC1w_n3zG+pRW6gtmVC=+YPH0cvK=eA@mail.gmail.com>
	<AANLkTimMDkdWi_V8TDXrxzn9paLEsthTGaOXMGD2_-PN@mail.gmail.com>
	<AANLkTikghfSv1pu-V8-qKB1C6Oc-JFipv2Wm=eWwJfgr@mail.gmail.com>
	<AANLkTi=Sjtsw6KPqDXJmB-F8g-ZcUfz7n62ro7b4jmid@mail.gmail.com>
	<AANLkTim22-SPAnWp6OttQt7AcUVBzRS=z1aiDvsgsD3j@mail.gmail.com>
	<AANLkTikE8cGwPppjS76ABot0YYgHzMToJGszFpZMNBLu@mail.gmail.com>
	<AANLkTikftaVhR1jXO5sPqZTOwh9AAPGR4bOo764hPqw8@mail.gmail.com>
	<AANLkTimZ_1ACgw5O+zRN7iUZ2-LicrWfpGMjfdTEFcZp@mail.gmail.com>
	<AANLkTimLgZDK8jTdRPqrxN3Fhe8grKCHyB+=9H200vVH@mail.gmail.com>
Message-ID: <AANLkTin9gR2p3GapiRZrTPOtunvYHze5HQruJ7KBjkzQ@mail.gmail.com>

ok if that's the case, then try latest git which should also work.
If not let me know.  I'll be back tomorrow.

commit 4f93ea25ece53df029e6dc6c913ad7c190df22d5
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Tue Feb 15 17:43:13 2011 -0600

    allow uuid bridge on unaswered channels as long as there is media
available on at least one



On Tue, Feb 15, 2011 at 5:49 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:
> Whow! With this change I can ear progressing messages! It works now!
> Stephen
>
> On Wed, Feb 16, 2011 at 12:35 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> That used to be a problem to bridge 2 unanswered channels.
>> I think it *might* let you now if you have the right conditions but it
>> may take some code and testing.
>>
>> can you test this for me?
>>
>> edit switch_ivr_bridge.c
>>
>> erase or comment out the entire else block starting at line 1411
>>
>> Try again.
>>
>>
>>
>> On Tue, Feb 15, 2011 at 5:30 PM, Stephen Wilde <wstephen80 at gmail.com>
>> wrote:
>> > Sorry, I have now followed your indication doing the bridge only after
>> > receiving "CHANNEL_PARK" on legB.?The problem is the same with no audio.
>> > I have also tried to do a pre_answer in legA (before originate legB)
>> > with no
>> > success.
>> > I see in the fs_cli:
>> > [CRIT] switch_ivr_bridge.c:1412 Neither channel is answered, cannot
>> > bridge
>> > them.
>> > The sequence of events I receive before doing the bridge is:
>> > [OUTBOUND] - [CHANNEL_UUID]
>> > [OUTBOUND] - [CHANNEL_OUTGOING]
>> > [OUTBOUND] - [CHANNEL_ORIGINATE]
>> > [OUTBOUND] - [CHANNEL_STATE]
>> > [OUTBOUND] - [CHANNEL_CALLSTATE]
>> > [OUTBOUND] - [CHANNEL_STATE]
>> > [OUTBOUND] - [CHANNEL_STATE]
>> > [INBOUND] - [CHANNEL_EXECUTE]
>> > [INBOUND] - [CHANNEL_CALLSTATE]
>> > [INBOUND] - [CHANNEL_PROGRESS_MEDIA]
>> > [INBOUND] - [CHANNEL_EXECUTE_COMPLETE]
>> > [OUTBOUND] - [CALL_UPDATE]
>> > [OUTBOUND] - [CHANNEL_PROGRESS]
>> > [OUTBOUND] - [CALL_UPDATE]
>> > [OUTBOUND] - [CODEC]
>> > [OUTBOUND] - [CODEC]
>> > [OUTBOUND] - [CHANNEL_CALLSTATE]
>> > [OUTBOUND] - [CHANNEL_PROGRESS_MEDIA]
>> > [OUTBOUND] - [CHANNEL_STATE]
>> > [OUTBOUND] - [CHANNEL_EXECUTE]
>> > [OUTBOUND] - [CHANNEL_PARK]
>> > After this CHANNEL_PARK I do the "uuid_bridge" with no succes. These
>> > event
>> > are related to the case of putting legA in pre_answer. Same problem if I
>> > leave it in park.
>> > Stephen
>> >
>> > On Tue, Feb 15, 2011 at 11:32 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> Did you read the little chart i put in my last email?
>> >>
>> >> i'll be more specific:
>> >>
>> >> 1) leg A who has a_uuid has called inbound and has been put in park
>> >> app.
>> >>
>> >> from ESL:
>> >> generate uuid (we'll ?call ?it b_uuid for clarity)
>> >> bgapi originate {origination_uuid=b_uuid}soifa/internal/bleg at box.com
>> >> park
>> >> inline
>> >>
>> >> Wait for park event from b_uuid (do not touch it any sooner)
>> >>
>> >> now either do:
>> >>
>> >> uuid_bridge a_uuid b_uuid
>> >>
>> >> or:
>> >>
>> >> uuid_transfer a_uuid intercept:b_uuid inline
>> >>
>> >> or:
>> >>
>> >> sendmsg to a_uuid telling it to execute intercept app on b_uuid
>> >>
>> >>
>> >>
>> >>
>> >>
>> >> On Tue, Feb 15, 2011 at 3:46 PM, Stephen Wilde <wstephen80 at gmail.com>
>> >> wrote:
>> >> > When legB answers there is also no audio.
>> >> > If I do the bridge after legB answers the call (instead of when
>> >> > "origination" has ended), then the audio is ok?and the legA can ear
>> >> > the
>> >> > legB
>> >> > also if legA is in "pre_answer" state.
>> >> > Now I have found one problem: the "BACKGROUND_JOB" event related to
>> >> > "bgapi
>> >> > uuid_bridge ..." contains: "-ERR Invalid uuid" but?both uuid's are
>> >> > correct
>> >> > so probably is due to the state of channels.
>> >> > With the same uuid's, the uuid_bridge after the legB answer works
>> >> > fine.
>> >> > Stephen
>> >> >
>> >> > On Tue, Feb 15, 2011 at 9:53 PM, Anthony Minessale
>> >> > <anthony.minessale at gmail.com> wrote:
>> >> >>
>> >> >> I meant when you have no audio, and you wait until leg b answers.
>> >> >> what
>> >> >> happens then?
>> >> >>
>> >> >>
>> >> >> If you have leg A inbound and you send it to park.
>> >> >> then you originate outbound to b leg to park
>> >> >> wait for park event on b leg
>> >> >> then you uuid_bridge them
>> >> >>
>> >> >> also you can sendmsg with application = intercept app_arg = uuid of
>> >> >> B
>> >> >>
>> >> >>
>> >> >> On Tue, Feb 15, 2011 at 12:08 PM, Stephen Wilde
>> >> >> <wstephen80 at gmail.com>
>> >> >> wrote:
>> >> >> > LegA is an incoming call, legB is an outgoing call.
>> >> >> > The dialplan is:
>> >> >> > ?? ?<extension name="myhandler">
>> >> >> > ?? ? ?<condition field="destination_number" expression="^xxxx$">
>> >> >> > ?? ? ? ?<action application="set"
>> >> >> > data="dialstring=sofia/external/yyyy at zzzz
>> >> >> > />
>> >> >> > ?? ? ? ?<action application="socket" data="127.0.0.1:8084 async
>> >> >> > full"/>
>> >> >> > ?? ? ?</condition>
>> >> >> > ?? ?</extension>
>> >> >> > When the legB answer the call, I receive the "CHANNEL_ANSWER"
>> >> >> > event
>> >> >> > on
>> >> >> > legB
>> >> >> > and I send the "answer" to legA (I have already done a uuid_bridge
>> >> >> > when
>> >> >> > originate has ended).
>> >> >> > The audio continue to be mute in both direction.
>> >> >> > The "uuid_bridge" works only if I call it after received
>> >> >> > "CHANNEL_ANSWER" in
>> >> >> > legB or after I send "answer" to legA.
>> >> >> > It's not enough to do a "pre_answer".
>> >> >> > Stephen
>> >> >> >
>> >> >> > On Tue, Feb 15, 2011 at 6:15 PM, Anthony Minessale
>> >> >> > <anthony.minessale at gmail.com> wrote:
>> >> >> >>
>> >> >> >> What happens once they answer?
>> >> >> >>
>> >> >> >> This is not one of those attempts to speed up click to call by
>> >> >> >> calling
>> >> >> >> them both at once is it?
>> >> >> >>
>> >> >> >>
>> >> >> >> On Tue, Feb 15, 2011 at 10:29 AM, Stephen Wilde
>> >> >> >> <wstephen80 at gmail.com>
>> >> >> >> wrote:
>> >> >> >> > Ok, I have tried with "pre_answer" to the inbound channel (that
>> >> >> >> > will
>> >> >> >> > be
>> >> >> >> > the
>> >> >> >> > first uuid in the "uuid_bridge")?but with same result: no audio
>> >> >> >> > from
>> >> >> >> > outbound (in progressing media state) to inbound (in
>> >> >> >> > progressing
>> >> >> >> > media
>> >> >> >> > state
>> >> >> >> > due to my pre_answer).
>> >> >> >> > Here the sequence of events:
>> >> >> >> > [ERROR] newacme.cpp:193 mycallback() Connected!
>> >> >> >> > [ERROR] newacme.cpp:113 Originate() bgapi originate
>> >> >> >> >
>> >> >> >> >
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > [origination_uuid=61e00b94-f4d3-4ac4-bcf5-8fba02a18de6]sofia/external/xxxx at yyyy
>> >> >> >> > &park()
>> >> >> >> > [ERROR] newacme.cpp:115 Originate() Job-ID:
>> >> >> >> > eee79d97-dbf8-49d4-9bf9-578ef75f73ea
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_UUID]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_OUTGOING]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_ORIGINATE]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_STATE]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_CALLSTATE]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_STATE]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_STATE]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CALL_UPDATE]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CODEC]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CODEC]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_CALLSTATE]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_PROGRESS_MEDIA]
>> >> >> >> > [ERROR] newacme.cpp:250 mycallback() Executing 'pre_answer' on
>> >> >> >> > INBOUND
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_STATE]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_EXECUTE]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_PARK]
>> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_EXECUTE]
>> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_CALLSTATE]
>> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_PROGRESS_MEDIA]
>> >> >> >> > [ERROR] newacme.cpp:127 BridgeTo() bgapi uuid_bridge
>> >> >> >> > 902e2cd4-10ee-4938-97cf-c9eb2ac969f8
>> >> >> >> > 61e00b94-f4d3-4ac4-bcf5-8fba02a18de6
>> >> >> >> > [ERROR] newacme.cpp:129 BridgeTo() Job-ID:
>> >> >> >> > 7a66a5c7-721c-47f4-aff2-373ec20123c8
>> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CALL_UPDATE]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_CALLSTATE]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_ANSWER]
>> >> >> >> > [ERROR] newacme.cpp:255 mycallback() Executing 'answer' on
>> >> >> >> > INBOUND
>> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_EXECUTE]
>> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_CALLSTATE]
>> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_ANSWER]
>> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_CALLSTATE]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_HANGUP]
>> >> >> >> > [ERROR] newacme.cpp:260 mycallback() Executing 'hangup' on
>> >> >> >> > INBOUND
>> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_EXECUTE]
>> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_CALLSTATE]
>> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_HANGUP]
>> >> >> >> > [ERROR] newacme.cpp:272 mycallback() Executing 'hangup' on
>> >> >> >> > OUTBOUND
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_UNPARK]
>> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_STATE]
>> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_UNPARK]
>> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_STATE]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_STATE]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_HANGUP_COMPLETE]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_DESTROY]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_CALLSTATE]
>> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND] -
>> >> >> >> > [CHANNEL_STATE]
>> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_STATE]
>> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND] -
>> >> >> >> > [CHANNEL_HANGUP_COMPLETE]
>> >> >> >> > [ERROR] newacme.cpp:290 mycallback() End.
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > On Tue, Feb 15, 2011 at 5:09 PM, Anthony Minessale
>> >> >> >> > <anthony.minessale at gmail.com> wrote:
>> >> >> >> >>
>> >> >> >> >> Well you can't bridge 2 unanswered channels together.
>> >> >> >> >> You need at least one of them to have been pre_answered and
>> >> >> >> >> that
>> >> >> >> >> should be the first uuid in the list.
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> On Tue, Feb 15, 2011 at 3:48 AM, Stephen Wilde
>> >> >> >> >> <wstephen80 at gmail.com>
>> >> >> >> >> wrote:
>> >> >> >> >> > I have tried to wait originate completition but the
>> >> >> >> >> > "uuid_bridge"
>> >> >> >> >> > doesn't
>> >> >> >> >> > works also in this case.
>> >> >> >> >> > It seems that works only when at least one leg is answered
>> >> >> >> >> > so
>> >> >> >> >> > it's
>> >> >> >> >> > not
>> >> >> >> >> > possible to do the "uuid_bridge" during progressing phase
>> >> >> >> >> > also
>> >> >> >> >> > if
>> >> >> >> >> > the
>> >> >> >> >> > originate has ended (I don't set the ignore_early_media).
>> >> >> >> >> > My application is this:?http://pastebin.freeswitch.org/15387
>> >> >> >> >> > The application:
>> >> >> >> >> > 1. receive a call with an "inbound_uuid"
>> >> >> >> >> > 2. create a new "outbound_uuid"
>> >> >> >> >> > 3. do a "bgapi originate" using the new "outbound_uuid"
>> >> >> >> >> > 4. when the called phone is ringing, receive a
>> >> >> >> >> > "CHANNEL_PROGRESS_MEDIA"
>> >> >> >> >> > event on "outbound_uuid" and do a "pre_answer" on
>> >> >> >> >> > "inbound_uuid"
>> >> >> >> >> > 5. receive a?"CHANNEL_PROGRESS_MEDIA" on "inbound_uuid"
>> >> >> >> >> > 6. receive a job completition event related to the
>> >> >> >> >> > "originate"
>> >> >> >> >> > so
>> >> >> >> >> > issue
>> >> >> >> >> > an
>> >> >> >> >> > "uuid_bridge inbound_uuid outbound_uuid"
>> >> >> >> >> > 7. when a job completition event related to "uuid_bridge" is
>> >> >> >> >> > received,
>> >> >> >> >> > no
>> >> >> >> >> > audio flow from outbound to inbound channel
>> >> >> >> >> > 8. when outbound answer the call, the application answer
>> >> >> >> >> > also
>> >> >> >> >> > the
>> >> >> >> >> > inbound
>> >> >> >> >> > call but no audio flow in both directions
>> >> >> >> >> > If I do the uuid_bridge after legB answer, then all is ok
>> >> >> >> >> > (obviously
>> >> >> >> >> > with no
>> >> >> >> >> > audio during progressing phase).
>> >> >> >> >> >
>> >> >> >> >> > The log of my application is:
>> >> >> >> >> > [ERROR] newacme.cpp:46 mycallback() Connected!
>> >> >> >> >> > [ERROR] newacme.cpp:65 mycallback() Inbound uuid =
>> >> >> >> >> > 8b2c39db-1ad9-489c-b72f-a92d4087bf99
>> >> >> >> >> > [ERROR] newacme.cpp:68 mycallback() create_uuid:
>> >> >> >> >> > 394167aa-2811-4fcd-95c9-85576bdd9a7a
>> >> >> >> >> > [ERROR] newacme.cpp:89 mycallback() bgapi originate
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> > [origination_uuid=394167aa-2811-4fcd-95c9-85576bdd9a7a]sofia/external/xxxx at yyyy
>> >> >> >> >> > &park()
>> >> >> >> >> > [ERROR] newacme.cpp:91 mycallback() Job-ID:
>> >> >> >> >> > 1c654ac2-c22d-418f-9fad-bb6b6b35aeff
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CHANNEL_UUID]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CHANNEL_OUTGOING]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CHANNEL_ORIGINATE]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CHANNEL_STATE]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CHANNEL_CALLSTATE]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CHANNEL_STATE]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CHANNEL_STATE]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CALL_UPDATE]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CODEC]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CODEC]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CHANNEL_CALLSTATE]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CHANNEL_PROGRESS_MEDIA]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CHANNEL_STATE]
>> >> >> >> >> > [ERROR] newacme.cpp:112 mycallback() Got an event:
>> >> >> >> >> > [BACKGROUND_JOB] -
>> >> >> >> >> > [1c654ac2-c22d-418f-9fad-bb6b6b35aeff]
>> >> >> >> >> > [ERROR] newacme.cpp:119 mycallback() bgapi uuid_bridge
>> >> >> >> >> > 8b2c39db-1ad9-489c-b72f-a92d4087bf99
>> >> >> >> >> > 394167aa-2811-4fcd-95c9-85576bdd9a7a
>> >> >> >> >> > [ERROR] newacme.cpp:123 mycallback() Job-ID:
>> >> >> >> >> > 0d43cffe-fdc6-4a13-97e6-aeee199c45d5
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CHANNEL_EXECUTE]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CHANNEL_PARK]
>> >> >> >> >> > [ERROR] newacme.cpp:112 mycallback() Got an event:
>> >> >> >> >> > [BACKGROUND_JOB] -
>> >> >> >> >> > [0d43cffe-fdc6-4a13-97e6-aeee199c45d5]
>> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
>> >> >> >> >> > -
>> >> >> >> >> > [CHANNEL_EXECUTE]
>> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
>> >> >> >> >> > -
>> >> >> >> >> > [CHANNEL_CALLSTATE]
>> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
>> >> >> >> >> > -
>> >> >> >> >> > [CHANNEL_PROGRESS_MEDIA]
>> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
>> >> >> >> >> > -
>> >> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CALL_UPDATE]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CHANNEL_CALLSTATE]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CHANNEL_ANSWER]
>> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
>> >> >> >> >> > -
>> >> >> >> >> > [CHANNEL_EXECUTE]
>> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
>> >> >> >> >> > -
>> >> >> >> >> > [CHANNEL_CALLSTATE]
>> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
>> >> >> >> >> > -
>> >> >> >> >> > [CHANNEL_ANSWER]
>> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
>> >> >> >> >> > -
>> >> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CHANNEL_CALLSTATE]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CHANNEL_HANGUP]
>> >> >> >> >> > [ERROR] newacme.cpp:160 mycallback() hangup
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CHANNEL_UNPARK]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CHANNEL_STATE]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CHANNEL_STATE]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CHANNEL_HANGUP_COMPLETE]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CHANNEL_DESTROY]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CHANNEL_CALLSTATE]
>> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
>> >> >> >> >> > [OUTBOUND] -
>> >> >> >> >> > [CHANNEL_STATE]
>> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
>> >> >> >> >> > -
>> >> >> >> >> > [CHANNEL_EXECUTE]
>> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
>> >> >> >> >> > -
>> >> >> >> >> > [CHANNEL_CALLSTATE]
>> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
>> >> >> >> >> > -
>> >> >> >> >> > [CHANNEL_HANGUP]
>> >> >> >> >> > [ERROR] newacme.cpp:171 mycallback() hangup
>> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
>> >> >> >> >> > -
>> >> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
>> >> >> >> >> > -
>> >> >> >> >> > [CHANNEL_UNPARK]
>> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
>> >> >> >> >> > -
>> >> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
>> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event: [INBOUND]
>> >> >> >> >> > -
>> >> >> >> >> > [CHANNEL_STATE]
>> >> >> >> >> > [ERROR] newacme.cpp:184 mycallback() End.
>> >> >> >> >> >
>> >> >> >> >> > Stephen
>> >> >> >> >> >
>> >> >> >> >> > On Tue, Feb 15, 2011 at 1:14 AM, Anthony Minessale
>> >> >> >> >> > <anthony.minessale at gmail.com> wrote:
>> >> >> >> >> >>
>> >> >> >> >> >> if you do originate without ignore_early_media=true set it
>> >> >> >> >> >> will
>> >> >> >> >> >> end
>> >> >> >> >> >> the soonest possible where it's suitable for a bridge.
>> >> >> >> >> >> so that is the best bet to wait for originate to end.
>> >> >> >> >> >>
>> >> >> >> >> >>
>> >> >> >> >> >> On Mon, Feb 14, 2011 at 6:11 PM, Stephen Wilde
>> >> >> >> >> >> <wstephen80 at gmail.com>
>> >> >> >> >> >> wrote:
>> >> >> >> >> >> > Ok, now I'm doing bridge when I receive "CHANNEL_ANSWER"
>> >> >> >> >> >> > or
>> >> >> >> >> >> > "CHANNEL_PROGRESS_MEDIA" on outbound leg but you say that
>> >> >> >> >> >> > I
>> >> >> >> >> >> > have
>> >> >> >> >> >> > wait
>> >> >> >> >> >> > that
>> >> >> >> >> >> > "originate has ended": you mean that I have to wait for
>> >> >> >> >> >> > "BACKGROUND_JOB"
>> >> >> >> >> >> > event related to my "bgapi originate ... &park"?
>> >> >> >> >> >> > I'm already doing "uuid_bridge <inbound_uuid>
>> >> >> >> >> >> > <outbound_uuid>".
>> >> >> >> >> >> > I'll try also with intercept and inline originate.?Thank
>> >> >> >> >> >> > you!
>> >> >> >> >> >> > Stephen
>> >> >> >> >> >> >
>> >> >> >> >> >> > On Tue, Feb 15, 2011 at 12:53 AM, Anthony Minessale
>> >> >> >> >> >> > <anthony.minessale at gmail.com> wrote:
>> >> >> >> >> >> >>
>> >> >> >> >> >> >> you need to make sure originate has ended on the
>> >> >> >> >> >> >> outbound
>> >> >> >> >> >> >> leg
>> >> >> >> >> >> >> before
>> >> >> >> >> >> >> you use it in a bridge etc.
>> >> >> >> >> >> >> you also need to supply the inbound leg first in
>> >> >> >> >> >> >> uuid_bridge
>> >> >> >> >> >> >> if
>> >> >> >> >> >> >> that
>> >> >> >> >> >> >> is something you want to do.
>> >> >> >> >> >> >>
>> >> >> >> >> >> >> Easier would be to originate the B leg to park inline
>> >> >> >> >> >> >> and
>> >> >> >> >> >> >> tell
>> >> >> >> >> >> >> A
>> >> >> >> >> >> >> leg
>> >> >> >> >> >> >> to execute intercept on the B leg uuid.
>> >> >> >> >> >> >>
>> >> >> >> >> >> >>
>> >> >> >> >> >> >>
>> >> >> >> >> >> >> On Mon, Feb 14, 2011 at 5:44 PM, Stephen Wilde
>> >> >> >> >> >> >> <wstephen80 at gmail.com>
>> >> >> >> >> >> >> wrote:
>> >> >> >> >> >> >> > Sorry but I have missed something.
>> >> >> >> >> >> >> > I know that I can solve this problem directly in
>> >> >> >> >> >> >> > dialplan
>> >> >> >> >> >> >> > with
>> >> >> >> >> >> >> > a
>> >> >> >> >> >> >> > bridge
>> >> >> >> >> >> >> > but
>> >> >> >> >> >> >> > what I'm doing is an "Freeswitch ESL learning"?because
>> >> >> >> >> >> >> > I
>> >> >> >> >> >> >> > have
>> >> >> >> >> >> >> > to
>> >> >> >> >> >> >> > port
>> >> >> >> >> >> >> > some
>> >> >> >> >> >> >> > application in Freeswitch and I'm learning how to
>> >> >> >> >> >> >> > implement
>> >> >> >> >> >> >> > some
>> >> >> >> >> >> >> > functionality.
>> >> >> >> >> >> >> > For me it's important to take control of both
>> >> >> >> >> >> >> > inbound/outbound
>> >> >> >> >> >> >> > in
>> >> >> >> >> >> >> > full
>> >> >> >> >> >> >> > async
>> >> >> >> >> >> >> > way and I have the necessity to do the complete call
>> >> >> >> >> >> >> > control.
>> >> >> >> >> >> >> > I'm not sure but to me it seems that with a normal
>> >> >> >> >> >> >> > bridge
>> >> >> >> >> >> >> > I
>> >> >> >> >> >> >> > lose
>> >> >> >> >> >> >> > the
>> >> >> >> >> >> >> > control
>> >> >> >> >> >> >> > of two sessions, for example, an outbound answer is
>> >> >> >> >> >> >> > propagated
>> >> >> >> >> >> >> > by
>> >> >> >> >> >> >> > bridge
>> >> >> >> >> >> >> > application as inbound answer.
>> >> >> >> >> >> >> > What I want to do is an audio bridging so my
>> >> >> >> >> >> >> > application
>> >> >> >> >> >> >> > can
>> >> >> >> >> >> >> > take
>> >> >> >> >> >> >> > control of
>> >> >> >> >> >> >> > "signaling bridging".
>> >> >> >> >> >> >> >
>> >> >> >> >> >> >> > I'm wrong? There are other way to do that?
>> >> >> >> >> >> >> > Stephen
>> >> >> >> >> >> >> > On Tue, Feb 15, 2011 at 12:19 AM, Michael Collins
>> >> >> >> >> >> >> > <msc at freeswitch.org>
>> >> >> >> >> >> >> > wrote:
>> >> >> >> >> >> >> >>
>> >> >> >> >> >> >> >> My head is spinning after reading this email. :)
>> >> >> >> >> >> >> >> It sounds like you just need a simple bridge from the
>> >> >> >> >> >> >> >> incoming
>> >> >> >> >> >> >> >> leg
>> >> >> >> >> >> >> >> to
>> >> >> >> >> >> >> >> the
>> >> >> >> >> >> >> >> outgoing leg. Can you pre_answer the A leg then
>> >> >> >> >> >> >> >> execute
>> >> >> >> >> >> >> >> a
>> >> >> >> >> >> >> >> good
>> >> >> >> >> >> >> >> old-fashioned
>> >> >> >> >> >> >> >> bridge to the b-leg?
>> >> >> >> >> >> >> >> -MC
>> >> >> >> >> >> >> >> On Mon, Feb 14, 2011 at 4:57 PM, Stephen Wilde
>> >> >> >> >> >> >> >> <wstephen80 at gmail.com>
>> >> >> >> >> >> >> >> wrote:
>> >> >> >> >> >> >> >>>
>> >> >> >> >> >> >> >>> The problem I have is to propagate the audio coming
>> >> >> >> >> >> >> >>> from
>> >> >> >> >> >> >> >>> an
>> >> >> >> >> >> >> >>> "originated"
>> >> >> >> >> >> >> >>> outbound session to the inbound session when the
>> >> >> >> >> >> >> >>> outbound
>> >> >> >> >> >> >> >>> is
>> >> >> >> >> >> >> >>> in
>> >> >> >> >> >> >> >>> the
>> >> >> >> >> >> >> >>> PROGRESS
>> >> >> >> >> >> >> >>> MEDIA phase.
>> >> >> >> >> >> >> >>> When my application receives the
>> >> >> >> >> >> >> >>> "CHANNEL_PROGRESS_MEDIA"
>> >> >> >> >> >> >> >>> event
>> >> >> >> >> >> >> >>> from
>> >> >> >> >> >> >> >>> outbound session I can do a "pre_answer" on inbound
>> >> >> >> >> >> >> >>> session
>> >> >> >> >> >> >> >>> but
>> >> >> >> >> >> >> >>> I'm
>> >> >> >> >> >> >> >>> not
>> >> >> >> >> >> >> >>> capable to do an audio bridge.
>> >> >> >> >> >> >> >>> I have tried with "uuid_bridge <outbound_uuid>
>> >> >> >> >> >> >> >>> <inbound_uuid>"
>> >> >> >> >> >> >> >>> with
>> >> >> >> >> >> >> >>> no
>> >> >> >> >> >> >> >>> result probably because this api requires that at
>> >> >> >> >> >> >> >>> least
>> >> >> >> >> >> >> >>> one
>> >> >> >> >> >> >> >>> session
>> >> >> >> >> >> >> >>> must be
>> >> >> >> >> >> >> >>> answered.
>> >> >> >> >> >> >> >>> I don't want to answer to the inbound session to
>> >> >> >> >> >> >> >>> propagate
>> >> >> >> >> >> >> >>> the
>> >> >> >> >> >> >> >>> outbound
>> >> >> >> >> >> >> >>> progressing media but I want to answer to inbound
>> >> >> >> >> >> >> >>> only
>> >> >> >> >> >> >> >>> on
>> >> >> >> >> >> >> >>> outbound
>> >> >> >> >> >> >> >>> answer.
>> >> >> >> >> >> >> >>> Any way to do that?
>> >> >> >> >> >> >> >>> 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
>> >> >> >> >> >> >> >>
>> >> >> >> >> >> >> >
>> >> >> >> >> >> >> >
>> >> >> >> >> >> >> > _______________________________________________
>> >> >> >> >> >> >> > FreeSWITCH-users mailing list
>> >> >> >> >> >> >> > FreeSWITCH-users at 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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Wed Feb 16 03:21:41 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Wed, 16 Feb 2011 01:21:41 +0100
Subject: [Freeswitch-users] ESL Outbound: how to bridge media?
In-Reply-To: <AANLkTin9gR2p3GapiRZrTPOtunvYHze5HQruJ7KBjkzQ@mail.gmail.com>
References: <AANLkTikLVLpSGkrRqeD3WVFZZgSaLnQvypCOG39hEhLX@mail.gmail.com>
	<AANLkTimwX8Uz4t9gp6Ppzfu+WE+UJa3FO4qHE7XKFOBz@mail.gmail.com>
	<AANLkTi=t-a7U9ov3wJ6Dh-PuniSfZbyeQ==BCAvMkUPN@mail.gmail.com>
	<AANLkTik27F7JjyMRLKryQZq_N+oVDsDzdmiPPa+d2KWN@mail.gmail.com>
	<AANLkTine=jG-dc98pDySashEvo3T5cSPwVXYwp54mhhC@mail.gmail.com>
	<AANLkTi=JAt=RkQ5b=DBajH_YR6TmiqExdjgF1=OL=fJ_@mail.gmail.com>
	<AANLkTinhkOznPh98kkHp=rCU9Qm0xv_hW3c41rRJr=Kn@mail.gmail.com>
	<AANLkTikAJoP7_cmhoEDorq4uCTdw+SVQeMNMpcFBkdYn@mail.gmail.com>
	<AANLkTi=XbutFHsC1w_n3zG+pRW6gtmVC=+YPH0cvK=eA@mail.gmail.com>
	<AANLkTimMDkdWi_V8TDXrxzn9paLEsthTGaOXMGD2_-PN@mail.gmail.com>
	<AANLkTikghfSv1pu-V8-qKB1C6Oc-JFipv2Wm=eWwJfgr@mail.gmail.com>
	<AANLkTi=Sjtsw6KPqDXJmB-F8g-ZcUfz7n62ro7b4jmid@mail.gmail.com>
	<AANLkTim22-SPAnWp6OttQt7AcUVBzRS=z1aiDvsgsD3j@mail.gmail.com>
	<AANLkTikE8cGwPppjS76ABot0YYgHzMToJGszFpZMNBLu@mail.gmail.com>
	<AANLkTikftaVhR1jXO5sPqZTOwh9AAPGR4bOo764hPqw8@mail.gmail.com>
	<AANLkTimZ_1ACgw5O+zRN7iUZ2-LicrWfpGMjfdTEFcZp@mail.gmail.com>
	<AANLkTimLgZDK8jTdRPqrxN3Fhe8grKCHyB+=9H200vVH@mail.gmail.com>
	<AANLkTin9gR2p3GapiRZrTPOtunvYHze5HQruJ7KBjkzQ@mail.gmail.com>
Message-ID: <AANLkTikgDEX_3oeGS5-m+rdyrKSfCu2oVZEFLDQak4==@mail.gmail.com>

Thank you Anthony, with latest git it works fine.

Stephen

On Wed, Feb 16, 2011 at 12:56 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> ok if that's the case, then try latest git which should also work.
> If not let me know.  I'll be back tomorrow.
>
> commit 4f93ea25ece53df029e6dc6c913ad7c190df22d5
> Author: Anthony Minessale <anthm at freeswitch.org>
> Date:   Tue Feb 15 17:43:13 2011 -0600
>
>    allow uuid bridge on unaswered channels as long as there is media
> available on at least one
>
>
>
> On Tue, Feb 15, 2011 at 5:49 PM, Stephen Wilde <wstephen80 at gmail.com>
> wrote:
> > Whow! With this change I can ear progressing messages! It works now!
> > Stephen
> >
> > On Wed, Feb 16, 2011 at 12:35 AM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> That used to be a problem to bridge 2 unanswered channels.
> >> I think it *might* let you now if you have the right conditions but it
> >> may take some code and testing.
> >>
> >> can you test this for me?
> >>
> >> edit switch_ivr_bridge.c
> >>
> >> erase or comment out the entire else block starting at line 1411
> >>
> >> Try again.
> >>
> >>
> >>
> >> On Tue, Feb 15, 2011 at 5:30 PM, Stephen Wilde <wstephen80 at gmail.com>
> >> wrote:
> >> > Sorry, I have now followed your indication doing the bridge only after
> >> > receiving "CHANNEL_PARK" on legB. The problem is the same with no
> audio.
> >> > I have also tried to do a pre_answer in legA (before originate legB)
> >> > with no
> >> > success.
> >> > I see in the fs_cli:
> >> > [CRIT] switch_ivr_bridge.c:1412 Neither channel is answered, cannot
> >> > bridge
> >> > them.
> >> > The sequence of events I receive before doing the bridge is:
> >> > [OUTBOUND] - [CHANNEL_UUID]
> >> > [OUTBOUND] - [CHANNEL_OUTGOING]
> >> > [OUTBOUND] - [CHANNEL_ORIGINATE]
> >> > [OUTBOUND] - [CHANNEL_STATE]
> >> > [OUTBOUND] - [CHANNEL_CALLSTATE]
> >> > [OUTBOUND] - [CHANNEL_STATE]
> >> > [OUTBOUND] - [CHANNEL_STATE]
> >> > [INBOUND] - [CHANNEL_EXECUTE]
> >> > [INBOUND] - [CHANNEL_CALLSTATE]
> >> > [INBOUND] - [CHANNEL_PROGRESS_MEDIA]
> >> > [INBOUND] - [CHANNEL_EXECUTE_COMPLETE]
> >> > [OUTBOUND] - [CALL_UPDATE]
> >> > [OUTBOUND] - [CHANNEL_PROGRESS]
> >> > [OUTBOUND] - [CALL_UPDATE]
> >> > [OUTBOUND] - [CODEC]
> >> > [OUTBOUND] - [CODEC]
> >> > [OUTBOUND] - [CHANNEL_CALLSTATE]
> >> > [OUTBOUND] - [CHANNEL_PROGRESS_MEDIA]
> >> > [OUTBOUND] - [CHANNEL_STATE]
> >> > [OUTBOUND] - [CHANNEL_EXECUTE]
> >> > [OUTBOUND] - [CHANNEL_PARK]
> >> > After this CHANNEL_PARK I do the "uuid_bridge" with no succes. These
> >> > event
> >> > are related to the case of putting legA in pre_answer. Same problem if
> I
> >> > leave it in park.
> >> > Stephen
> >> >
> >> > On Tue, Feb 15, 2011 at 11:32 PM, Anthony Minessale
> >> > <anthony.minessale at gmail.com> wrote:
> >> >>
> >> >> Did you read the little chart i put in my last email?
> >> >>
> >> >> i'll be more specific:
> >> >>
> >> >> 1) leg A who has a_uuid has called inbound and has been put in park
> >> >> app.
> >> >>
> >> >> from ESL:
> >> >> generate uuid (we'll  call  it b_uuid for clarity)
> >> >> bgapi originate {origination_uuid=b_uuid}soifa/internal/bleg at box.com
> >> >> park
> >> >> inline
> >> >>
> >> >> Wait for park event from b_uuid (do not touch it any sooner)
> >> >>
> >> >> now either do:
> >> >>
> >> >> uuid_bridge a_uuid b_uuid
> >> >>
> >> >> or:
> >> >>
> >> >> uuid_transfer a_uuid intercept:b_uuid inline
> >> >>
> >> >> or:
> >> >>
> >> >> sendmsg to a_uuid telling it to execute intercept app on b_uuid
> >> >>
> >> >>
> >> >>
> >> >>
> >> >>
> >> >> On Tue, Feb 15, 2011 at 3:46 PM, Stephen Wilde <wstephen80 at gmail.com
> >
> >> >> wrote:
> >> >> > When legB answers there is also no audio.
> >> >> > If I do the bridge after legB answers the call (instead of when
> >> >> > "origination" has ended), then the audio is ok and the legA can ear
> >> >> > the
> >> >> > legB
> >> >> > also if legA is in "pre_answer" state.
> >> >> > Now I have found one problem: the "BACKGROUND_JOB" event related to
> >> >> > "bgapi
> >> >> > uuid_bridge ..." contains: "-ERR Invalid uuid" but both uuid's are
> >> >> > correct
> >> >> > so probably is due to the state of channels.
> >> >> > With the same uuid's, the uuid_bridge after the legB answer works
> >> >> > fine.
> >> >> > Stephen
> >> >> >
> >> >> > On Tue, Feb 15, 2011 at 9:53 PM, Anthony Minessale
> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >>
> >> >> >> I meant when you have no audio, and you wait until leg b answers.
> >> >> >> what
> >> >> >> happens then?
> >> >> >>
> >> >> >>
> >> >> >> If you have leg A inbound and you send it to park.
> >> >> >> then you originate outbound to b leg to park
> >> >> >> wait for park event on b leg
> >> >> >> then you uuid_bridge them
> >> >> >>
> >> >> >> also you can sendmsg with application = intercept app_arg = uuid
> of
> >> >> >> B
> >> >> >>
> >> >> >>
> >> >> >> On Tue, Feb 15, 2011 at 12:08 PM, Stephen Wilde
> >> >> >> <wstephen80 at gmail.com>
> >> >> >> wrote:
> >> >> >> > LegA is an incoming call, legB is an outgoing call.
> >> >> >> > The dialplan is:
> >> >> >> >     <extension name="myhandler">
> >> >> >> >       <condition field="destination_number" expression="^xxxx$">
> >> >> >> >         <action application="set"
> >> >> >> > data="dialstring=sofia/external/yyyy at zzzz
> >> >> >> > />
> >> >> >> >         <action application="socket" data="127.0.0.1:8084 async
> >> >> >> > full"/>
> >> >> >> >       </condition>
> >> >> >> >     </extension>
> >> >> >> > When the legB answer the call, I receive the "CHANNEL_ANSWER"
> >> >> >> > event
> >> >> >> > on
> >> >> >> > legB
> >> >> >> > and I send the "answer" to legA (I have already done a
> uuid_bridge
> >> >> >> > when
> >> >> >> > originate has ended).
> >> >> >> > The audio continue to be mute in both direction.
> >> >> >> > The "uuid_bridge" works only if I call it after received
> >> >> >> > "CHANNEL_ANSWER" in
> >> >> >> > legB or after I send "answer" to legA.
> >> >> >> > It's not enough to do a "pre_answer".
> >> >> >> > Stephen
> >> >> >> >
> >> >> >> > On Tue, Feb 15, 2011 at 6:15 PM, Anthony Minessale
> >> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >> >>
> >> >> >> >> What happens once they answer?
> >> >> >> >>
> >> >> >> >> This is not one of those attempts to speed up click to call by
> >> >> >> >> calling
> >> >> >> >> them both at once is it?
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> On Tue, Feb 15, 2011 at 10:29 AM, Stephen Wilde
> >> >> >> >> <wstephen80 at gmail.com>
> >> >> >> >> wrote:
> >> >> >> >> > Ok, I have tried with "pre_answer" to the inbound channel
> (that
> >> >> >> >> > will
> >> >> >> >> > be
> >> >> >> >> > the
> >> >> >> >> > first uuid in the "uuid_bridge") but with same result: no
> audio
> >> >> >> >> > from
> >> >> >> >> > outbound (in progressing media state) to inbound (in
> >> >> >> >> > progressing
> >> >> >> >> > media
> >> >> >> >> > state
> >> >> >> >> > due to my pre_answer).
> >> >> >> >> > Here the sequence of events:
> >> >> >> >> > [ERROR] newacme.cpp:193 mycallback() Connected!
> >> >> >> >> > [ERROR] newacme.cpp:113 Originate() bgapi originate
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> >
> [origination_uuid=61e00b94-f4d3-4ac4-bcf5-8fba02a18de6]sofia/external/xxxx at yyyy
> >> >> >> >> > &park()
> >> >> >> >> > [ERROR] newacme.cpp:115 Originate() Job-ID:
> >> >> >> >> > eee79d97-dbf8-49d4-9bf9-578ef75f73ea
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_UUID]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_OUTGOING]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_ORIGINATE]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CALL_UPDATE]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CODEC]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CODEC]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_PROGRESS_MEDIA]
> >> >> >> >> > [ERROR] newacme.cpp:250 mycallback() Executing 'pre_answer'
> on
> >> >> >> >> > INBOUND
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_EXECUTE]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_PARK]
> >> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_EXECUTE]
> >> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_PROGRESS_MEDIA]
> >> >> >> >> > [ERROR] newacme.cpp:127 BridgeTo() bgapi uuid_bridge
> >> >> >> >> > 902e2cd4-10ee-4938-97cf-c9eb2ac969f8
> >> >> >> >> > 61e00b94-f4d3-4ac4-bcf5-8fba02a18de6
> >> >> >> >> > [ERROR] newacme.cpp:129 BridgeTo() Job-ID:
> >> >> >> >> > 7a66a5c7-721c-47f4-aff2-373ec20123c8
> >> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CALL_UPDATE]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_ANSWER]
> >> >> >> >> > [ERROR] newacme.cpp:255 mycallback() Executing 'answer' on
> >> >> >> >> > INBOUND
> >> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_EXECUTE]
> >> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_ANSWER]
> >> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_HANGUP]
> >> >> >> >> > [ERROR] newacme.cpp:260 mycallback() Executing 'hangup' on
> >> >> >> >> > INBOUND
> >> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_EXECUTE]
> >> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_HANGUP]
> >> >> >> >> > [ERROR] newacme.cpp:272 mycallback() Executing 'hangup' on
> >> >> >> >> > OUTBOUND
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_UNPARK]
> >> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_UNPARK]
> >> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_HANGUP_COMPLETE]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_DESTROY]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> >> > [ERROR] newacme.cpp:246 mycallback() Got an event: [OUTBOUND]
> -
> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> > [ERROR] newacme.cpp:268 mycallback() Got an event: [INBOUND]
> -
> >> >> >> >> > [CHANNEL_HANGUP_COMPLETE]
> >> >> >> >> > [ERROR] newacme.cpp:290 mycallback() End.
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > On Tue, Feb 15, 2011 at 5:09 PM, Anthony Minessale
> >> >> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >> >> >>
> >> >> >> >> >> Well you can't bridge 2 unanswered channels together.
> >> >> >> >> >> You need at least one of them to have been pre_answered and
> >> >> >> >> >> that
> >> >> >> >> >> should be the first uuid in the list.
> >> >> >> >> >>
> >> >> >> >> >>
> >> >> >> >> >> On Tue, Feb 15, 2011 at 3:48 AM, Stephen Wilde
> >> >> >> >> >> <wstephen80 at gmail.com>
> >> >> >> >> >> wrote:
> >> >> >> >> >> > I have tried to wait originate completition but the
> >> >> >> >> >> > "uuid_bridge"
> >> >> >> >> >> > doesn't
> >> >> >> >> >> > works also in this case.
> >> >> >> >> >> > It seems that works only when at least one leg is answered
> >> >> >> >> >> > so
> >> >> >> >> >> > it's
> >> >> >> >> >> > not
> >> >> >> >> >> > possible to do the "uuid_bridge" during progressing phase
> >> >> >> >> >> > also
> >> >> >> >> >> > if
> >> >> >> >> >> > the
> >> >> >> >> >> > originate has ended (I don't set the ignore_early_media).
> >> >> >> >> >> > My application is this:
> http://pastebin.freeswitch.org/15387
> >> >> >> >> >> > The application:
> >> >> >> >> >> > 1. receive a call with an "inbound_uuid"
> >> >> >> >> >> > 2. create a new "outbound_uuid"
> >> >> >> >> >> > 3. do a "bgapi originate" using the new "outbound_uuid"
> >> >> >> >> >> > 4. when the called phone is ringing, receive a
> >> >> >> >> >> > "CHANNEL_PROGRESS_MEDIA"
> >> >> >> >> >> > event on "outbound_uuid" and do a "pre_answer" on
> >> >> >> >> >> > "inbound_uuid"
> >> >> >> >> >> > 5. receive a "CHANNEL_PROGRESS_MEDIA" on "inbound_uuid"
> >> >> >> >> >> > 6. receive a job completition event related to the
> >> >> >> >> >> > "originate"
> >> >> >> >> >> > so
> >> >> >> >> >> > issue
> >> >> >> >> >> > an
> >> >> >> >> >> > "uuid_bridge inbound_uuid outbound_uuid"
> >> >> >> >> >> > 7. when a job completition event related to "uuid_bridge"
> is
> >> >> >> >> >> > received,
> >> >> >> >> >> > no
> >> >> >> >> >> > audio flow from outbound to inbound channel
> >> >> >> >> >> > 8. when outbound answer the call, the application answer
> >> >> >> >> >> > also
> >> >> >> >> >> > the
> >> >> >> >> >> > inbound
> >> >> >> >> >> > call but no audio flow in both directions
> >> >> >> >> >> > If I do the uuid_bridge after legB answer, then all is ok
> >> >> >> >> >> > (obviously
> >> >> >> >> >> > with no
> >> >> >> >> >> > audio during progressing phase).
> >> >> >> >> >> >
> >> >> >> >> >> > The log of my application is:
> >> >> >> >> >> > [ERROR] newacme.cpp:46 mycallback() Connected!
> >> >> >> >> >> > [ERROR] newacme.cpp:65 mycallback() Inbound uuid =
> >> >> >> >> >> > 8b2c39db-1ad9-489c-b72f-a92d4087bf99
> >> >> >> >> >> > [ERROR] newacme.cpp:68 mycallback() create_uuid:
> >> >> >> >> >> > 394167aa-2811-4fcd-95c9-85576bdd9a7a
> >> >> >> >> >> > [ERROR] newacme.cpp:89 mycallback() bgapi originate
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> >
> >> >> >> >> >> >
> [origination_uuid=394167aa-2811-4fcd-95c9-85576bdd9a7a]sofia/external/xxxx at yyyy
> >> >> >> >> >> > &park()
> >> >> >> >> >> > [ERROR] newacme.cpp:91 mycallback() Job-ID:
> >> >> >> >> >> > 1c654ac2-c22d-418f-9fad-bb6b6b35aeff
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CHANNEL_UUID]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CHANNEL_OUTGOING]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CHANNEL_ORIGINATE]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CALL_UPDATE]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CODEC]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CODEC]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CHANNEL_PROGRESS_MEDIA]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> >> > [ERROR] newacme.cpp:112 mycallback() Got an event:
> >> >> >> >> >> > [BACKGROUND_JOB] -
> >> >> >> >> >> > [1c654ac2-c22d-418f-9fad-bb6b6b35aeff]
> >> >> >> >> >> > [ERROR] newacme.cpp:119 mycallback() bgapi uuid_bridge
> >> >> >> >> >> > 8b2c39db-1ad9-489c-b72f-a92d4087bf99
> >> >> >> >> >> > 394167aa-2811-4fcd-95c9-85576bdd9a7a
> >> >> >> >> >> > [ERROR] newacme.cpp:123 mycallback() Job-ID:
> >> >> >> >> >> > 0d43cffe-fdc6-4a13-97e6-aeee199c45d5
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CHANNEL_EXECUTE]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CHANNEL_PARK]
> >> >> >> >> >> > [ERROR] newacme.cpp:112 mycallback() Got an event:
> >> >> >> >> >> > [BACKGROUND_JOB] -
> >> >> >> >> >> > [0d43cffe-fdc6-4a13-97e6-aeee199c45d5]
> >> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event:
> [INBOUND]
> >> >> >> >> >> > -
> >> >> >> >> >> > [CHANNEL_EXECUTE]
> >> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event:
> [INBOUND]
> >> >> >> >> >> > -
> >> >> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event:
> [INBOUND]
> >> >> >> >> >> > -
> >> >> >> >> >> > [CHANNEL_PROGRESS_MEDIA]
> >> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event:
> [INBOUND]
> >> >> >> >> >> > -
> >> >> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CALL_UPDATE]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CHANNEL_ANSWER]
> >> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event:
> [INBOUND]
> >> >> >> >> >> > -
> >> >> >> >> >> > [CHANNEL_EXECUTE]
> >> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event:
> [INBOUND]
> >> >> >> >> >> > -
> >> >> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event:
> [INBOUND]
> >> >> >> >> >> > -
> >> >> >> >> >> > [CHANNEL_ANSWER]
> >> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event:
> [INBOUND]
> >> >> >> >> >> > -
> >> >> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CHANNEL_HANGUP]
> >> >> >> >> >> > [ERROR] newacme.cpp:160 mycallback() hangup
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CHANNEL_UNPARK]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CHANNEL_HANGUP_COMPLETE]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CHANNEL_DESTROY]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> >> >> > [ERROR] newacme.cpp:138 mycallback() Got an event:
> >> >> >> >> >> > [OUTBOUND] -
> >> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event:
> [INBOUND]
> >> >> >> >> >> > -
> >> >> >> >> >> > [CHANNEL_EXECUTE]
> >> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event:
> [INBOUND]
> >> >> >> >> >> > -
> >> >> >> >> >> > [CHANNEL_CALLSTATE]
> >> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event:
> [INBOUND]
> >> >> >> >> >> > -
> >> >> >> >> >> > [CHANNEL_HANGUP]
> >> >> >> >> >> > [ERROR] newacme.cpp:171 mycallback() hangup
> >> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event:
> [INBOUND]
> >> >> >> >> >> > -
> >> >> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event:
> [INBOUND]
> >> >> >> >> >> > -
> >> >> >> >> >> > [CHANNEL_UNPARK]
> >> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event:
> [INBOUND]
> >> >> >> >> >> > -
> >> >> >> >> >> > [CHANNEL_EXECUTE_COMPLETE]
> >> >> >> >> >> > [ERROR] newacme.cpp:167 mycallback() Got an event:
> [INBOUND]
> >> >> >> >> >> > -
> >> >> >> >> >> > [CHANNEL_STATE]
> >> >> >> >> >> > [ERROR] newacme.cpp:184 mycallback() End.
> >> >> >> >> >> >
> >> >> >> >> >> > Stephen
> >> >> >> >> >> >
> >> >> >> >> >> > On Tue, Feb 15, 2011 at 1:14 AM, Anthony Minessale
> >> >> >> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >> >> >> >>
> >> >> >> >> >> >> if you do originate without ignore_early_media=true set
> it
> >> >> >> >> >> >> will
> >> >> >> >> >> >> end
> >> >> >> >> >> >> the soonest possible where it's suitable for a bridge.
> >> >> >> >> >> >> so that is the best bet to wait for originate to end.
> >> >> >> >> >> >>
> >> >> >> >> >> >>
> >> >> >> >> >> >> On Mon, Feb 14, 2011 at 6:11 PM, Stephen Wilde
> >> >> >> >> >> >> <wstephen80 at gmail.com>
> >> >> >> >> >> >> wrote:
> >> >> >> >> >> >> > Ok, now I'm doing bridge when I receive
> "CHANNEL_ANSWER"
> >> >> >> >> >> >> > or
> >> >> >> >> >> >> > "CHANNEL_PROGRESS_MEDIA" on outbound leg but you say
> that
> >> >> >> >> >> >> > I
> >> >> >> >> >> >> > have
> >> >> >> >> >> >> > wait
> >> >> >> >> >> >> > that
> >> >> >> >> >> >> > "originate has ended": you mean that I have to wait for
> >> >> >> >> >> >> > "BACKGROUND_JOB"
> >> >> >> >> >> >> > event related to my "bgapi originate ... &park"?
> >> >> >> >> >> >> > I'm already doing "uuid_bridge <inbound_uuid>
> >> >> >> >> >> >> > <outbound_uuid>".
> >> >> >> >> >> >> > I'll try also with intercept and inline
> originate. Thank
> >> >> >> >> >> >> > you!
> >> >> >> >> >> >> > Stephen
> >> >> >> >> >> >> >
> >> >> >> >> >> >> > On Tue, Feb 15, 2011 at 12:53 AM, Anthony Minessale
> >> >> >> >> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >> >> >> >> >>
> >> >> >> >> >> >> >> you need to make sure originate has ended on the
> >> >> >> >> >> >> >> outbound
> >> >> >> >> >> >> >> leg
> >> >> >> >> >> >> >> before
> >> >> >> >> >> >> >> you use it in a bridge etc.
> >> >> >> >> >> >> >> you also need to supply the inbound leg first in
> >> >> >> >> >> >> >> uuid_bridge
> >> >> >> >> >> >> >> if
> >> >> >> >> >> >> >> that
> >> >> >> >> >> >> >> is something you want to do.
> >> >> >> >> >> >> >>
> >> >> >> >> >> >> >> Easier would be to originate the B leg to park inline
> >> >> >> >> >> >> >> and
> >> >> >> >> >> >> >> tell
> >> >> >> >> >> >> >> A
> >> >> >> >> >> >> >> leg
> >> >> >> >> >> >> >> to execute intercept on the B leg uuid.
> >> >> >> >> >> >> >>
> >> >> >> >> >> >> >>
> >> >> >> >> >> >> >>
> >> >> >> >> >> >> >> On Mon, Feb 14, 2011 at 5:44 PM, Stephen Wilde
> >> >> >> >> >> >> >> <wstephen80 at gmail.com>
> >> >> >> >> >> >> >> wrote:
> >> >> >> >> >> >> >> > Sorry but I have missed something.
> >> >> >> >> >> >> >> > I know that I can solve this problem directly in
> >> >> >> >> >> >> >> > dialplan
> >> >> >> >> >> >> >> > with
> >> >> >> >> >> >> >> > a
> >> >> >> >> >> >> >> > bridge
> >> >> >> >> >> >> >> > but
> >> >> >> >> >> >> >> > what I'm doing is an "Freeswitch ESL
> learning" because
> >> >> >> >> >> >> >> > I
> >> >> >> >> >> >> >> > have
> >> >> >> >> >> >> >> > to
> >> >> >> >> >> >> >> > port
> >> >> >> >> >> >> >> > some
> >> >> >> >> >> >> >> > application in Freeswitch and I'm learning how to
> >> >> >> >> >> >> >> > implement
> >> >> >> >> >> >> >> > some
> >> >> >> >> >> >> >> > functionality.
> >> >> >> >> >> >> >> > For me it's important to take control of both
> >> >> >> >> >> >> >> > inbound/outbound
> >> >> >> >> >> >> >> > in
> >> >> >> >> >> >> >> > full
> >> >> >> >> >> >> >> > async
> >> >> >> >> >> >> >> > way and I have the necessity to do the complete call
> >> >> >> >> >> >> >> > control.
> >> >> >> >> >> >> >> > I'm not sure but to me it seems that with a normal
> >> >> >> >> >> >> >> > bridge
> >> >> >> >> >> >> >> > I
> >> >> >> >> >> >> >> > lose
> >> >> >> >> >> >> >> > the
> >> >> >> >> >> >> >> > control
> >> >> >> >> >> >> >> > of two sessions, for example, an outbound answer is
> >> >> >> >> >> >> >> > propagated
> >> >> >> >> >> >> >> > by
> >> >> >> >> >> >> >> > bridge
> >> >> >> >> >> >> >> > application as inbound answer.
> >> >> >> >> >> >> >> > What I want to do is an audio bridging so my
> >> >> >> >> >> >> >> > application
> >> >> >> >> >> >> >> > can
> >> >> >> >> >> >> >> > take
> >> >> >> >> >> >> >> > control of
> >> >> >> >> >> >> >> > "signaling bridging".
> >> >> >> >> >> >> >> >
> >> >> >> >> >> >> >> > I'm wrong? There are other way to do that?
> >> >> >> >> >> >> >> > Stephen
> >> >> >> >> >> >> >> > On Tue, Feb 15, 2011 at 12:19 AM, Michael Collins
> >> >> >> >> >> >> >> > <msc at freeswitch.org>
> >> >> >> >> >> >> >> > wrote:
> >> >> >> >> >> >> >> >>
> >> >> >> >> >> >> >> >> My head is spinning after reading this email. :)
> >> >> >> >> >> >> >> >> It sounds like you just need a simple bridge from
> the
> >> >> >> >> >> >> >> >> incoming
> >> >> >> >> >> >> >> >> leg
> >> >> >> >> >> >> >> >> to
> >> >> >> >> >> >> >> >> the
> >> >> >> >> >> >> >> >> outgoing leg. Can you pre_answer the A leg then
> >> >> >> >> >> >> >> >> execute
> >> >> >> >> >> >> >> >> a
> >> >> >> >> >> >> >> >> good
> >> >> >> >> >> >> >> >> old-fashioned
> >> >> >> >> >> >> >> >> bridge to the b-leg?
> >> >> >> >> >> >> >> >> -MC
> >> >> >> >> >> >> >> >> On Mon, Feb 14, 2011 at 4:57 PM, Stephen Wilde
> >> >> >> >> >> >> >> >> <wstephen80 at gmail.com>
> >> >> >> >> >> >> >> >> wrote:
> >> >> >> >> >> >> >> >>>
> >> >> >> >> >> >> >> >>> The problem I have is to propagate the audio
> coming
> >> >> >> >> >> >> >> >>> from
> >> >> >> >> >> >> >> >>> an
> >> >> >> >> >> >> >> >>> "originated"
> >> >> >> >> >> >> >> >>> outbound session to the inbound session when the
> >> >> >> >> >> >> >> >>> outbound
> >> >> >> >> >> >> >> >>> is
> >> >> >> >> >> >> >> >>> in
> >> >> >> >> >> >> >> >>> the
> >> >> >> >> >> >> >> >>> PROGRESS
> >> >> >> >> >> >> >> >>> MEDIA phase.
> >> >> >> >> >> >> >> >>> When my application receives the
> >> >> >> >> >> >> >> >>> "CHANNEL_PROGRESS_MEDIA"
> >> >> >> >> >> >> >> >>> event
> >> >> >> >> >> >> >> >>> from
> >> >> >> >> >> >> >> >>> outbound session I can do a "pre_answer" on
> inbound
> >> >> >> >> >> >> >> >>> session
> >> >> >> >> >> >> >> >>> but
> >> >> >> >> >> >> >> >>> I'm
> >> >> >> >> >> >> >> >>> not
> >> >> >> >> >> >> >> >>> capable to do an audio bridge.
> >> >> >> >> >> >> >> >>> I have tried with "uuid_bridge <outbound_uuid>
> >> >> >> >> >> >> >> >>> <inbound_uuid>"
> >> >> >> >> >> >> >> >>> with
> >> >> >> >> >> >> >> >>> no
> >> >> >> >> >> >> >> >>> result probably because this api requires that at
> >> >> >> >> >> >> >> >>> least
> >> >> >> >> >> >> >> >>> one
> >> >> >> >> >> >> >> >>> session
> >> >> >> >> >> >> >> >>> must be
> >> >> >> >> >> >> >> >>> answered.
> >> >> >> >> >> >> >> >>> I don't want to answer to the inbound session to
> >> >> >> >> >> >> >> >>> propagate
> >> >> >> >> >> >> >> >>> the
> >> >> >> >> >> >> >> >>> outbound
> >> >> >> >> >> >> >> >>> progressing media but I want to answer to inbound
> >> >> >> >> >> >> >> >>> only
> >> >> >> >> >> >> >> >>> on
> >> >> >> >> >> >> >> >>> outbound
> >> >> >> >> >> >> >> >>> answer.
> >> >> >> >> >> >> >> >>> Any way to do that?
> >> >> >> >> >> >> >> >>> 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
> >> >> >> >> >> >> >> >>
> >> >> >> >> >> >> >> >
> >> >> >> >> >> >> >> >
> >> >> >> >> >> >> >> > _______________________________________________
> >> >> >> >> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> >> >> >> >> > FreeSWITCH-users at 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
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 asilva at wirelessmundi.com  Wed Feb 16 03:22:17 2011
From: asilva at wirelessmundi.com (asilva at wirelessmundi.com)
Date: Wed, 16 Feb 2011 01:22:17 +0100
Subject: [Freeswitch-users] =?utf-8?q?Mod=5Fcallcenter_reload_xml_error?=
Message-ID: <40187d1d593bee16127c518fe1d994a3@mail.wirelessmundi.com>

Hi,

Is it possible to reload mod_callcenter to force it to reload / reset the
configuration form the XML and synchronized it to callcenter database. 
I'm using the latest git and i couldn't find any option that allows me to
do that...

My idea is to add / or remove queues or even change some parameters, or
add more agents to the xml and with a single command reload the new
configuration.  
I notice that i have 

callcenter_config queue load | unload   
callcenter_config agent add | del  

is very painful to add or remove one element by one when i remove it form
the xml. 

I tried to reload the mod_callcenter:
# reload mod_callcenter
R: 
+OK module unloaded
+OK Reloading XML
+OK module loaded

[...]
2011-02-16 01:34:38.155841 [DEBUG] mod_callcenter.c:1045 Updated Agent
cc1_queue1_agent1 set contact = [call_timeout=30]loopback/101/local
2011-02-16 01:34:38.156843 [DEBUG] mod_callcenter.c:1045 Updated Agent
cc1_queue1_agent1 set status = Available
2011-02-16 01:34:38.157845 [DEBUG] mod_callcenter.c:1045 Updated Agent
cc1_queue1_agent1 set wrap_up_time = 40
2011-02-16 01:34:38.159849 [DEBUG] mod_callcenter.c:1045 Updated Agent
cc1_queue1_agent1 set type = callback
2011-02-16 01:34:38.161865 [DEBUG] mod_callcenter.c:1166 Updated tier:
Agent cc1_queue1_agent1 in Queue cc1_queue1 set level = 1
2011-02-16 01:34:38.163868 [DEBUG] mod_callcenter.c:1166 Updated tier:
Agent cc1_queue1_agent1 in Queue cc1_queue1 set position = 5


Seams to work, in the messages i see only the agents that are defined in
the XML, but when i do:

# callcenter_config agent list (i still have the previous agents that
where defined)

freeswitch at internal> callcenter_config agent list
name|system|uuid|type|contact|status|state|max_no_answer|wrap_up_time|reject_delay_time|busy_delay_time|last_bridge_start|last_bridge_end|last_offered_call|last_status_change|no_answer_count|calls_answered|talk_time|ready_time
cc1_queue1_agent1|single_box||callback|[call_timeout=30]loopback/101/local|Available|Waiting|0|40|0|0|0|0|0|1297815798|0|0|0|0
cc1_queue1_agent2|single_box||callback|[call_timeout=30]loopback/105/local|Available|Waiting|0|40|0|0|0|0|0|1297815798|0|0|0|0
+OK

Is it a bug in the CLI? 

Can anyone point me how to reload the XML configuration for the callcenter
without doing it manually queue by queue, agent by agent, or tier by tier?

Thanks,

Ant?nio Silva










From chris at cloudtel.com  Wed Feb 16 07:51:55 2011
From: chris at cloudtel.com (Chris Burns)
Date: Tue, 15 Feb 2011 23:51:55 -0500
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTi=xwAromp8SDeExJEOx+nQNpQkr1hcOVmVHxiYB@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
	<AANLkTi=in3CLd-n0mT3SRKYA0kxLMGVWk9KeBx3_jhY6@mail.gmail.com>
	<AANLkTin3Bu+8EEr_v_GaRGNPzOjfcnfzsV2Av2cgnWgO@mail.gmail.com>
	<AANLkTi=OC_jx0+nzK405VZURLDJPD25R4ZjyJ3RXJ1nk@mail.gmail.com>
	<AANLkTi=Sm7CjoayiE8nHmL0Eehqknj1jdpbx0YBQJQGH@mail.gmail.com>
	<AANLkTinr0RdgBs2UiS3=b3i+k6uCwkMihDy_FbAOFPc_@mail.gmail.com>
	<AANLkTi=+aObW3kVBZq5ox4b4E1uFTSgtXUL72tHLOzPi@mail.gmail.com>
	<AANLkTikVDKS40dVsf8x=+5=qnW59LqJbSv1H=vtROFaM@mail.gmail.com>
	<AANLkTi=gJsVKCJifVaBL_H6ZnPHTD__aQPhPeVfdx-ju@mail.gmail.com>
	<AANLkTimeuCHf_7Y1TErPwrJbdGZ3DoQmeqQbJ1_3YNpT@mail.gmail.com>
	<AANLkTikDxzntMFRpFhfiqJ4pzvjsTQR4w_FfyLfT5Y7z@mail.gmail.com>
	<AANLkTimVrhpfX+0D2mFPwGHK1EDsALWjFGp8hh7=ecqt@mail.gmail.com>
	<AANLkTi=oqw7fX-kvwqtxUXEWmzDu4C-dOjPrhpsCiwQc@mail.gmail.com>
	<AANLkTi=xwAromp8SDeExJEOx+nQNpQkr1hcOVmVHxiYB@mail.gmail.com>
Message-ID: <AANLkTi=EdwMoLo9vGVrBcFO+rLfTHAVCAUGf_p2n-Lju@mail.gmail.com>

I don't know the version numbers. I updated to HEAD and deployed just before
you reported the problem., which should be post 1.06. When the phones were
rebooting, BLF statuses set from the dialplan to which they were subscribed
were getting reset... but my script just talks over it to correct the issue.
Presence otherwise works normally.

I'm not holding out for a fix, as everything works just fine in my system
albeit not 100% efficiently. If I had time I would determine which device is
at fault, switch or phone, and help to remedy it at the source (I highly
suspect phone). But I had just enough time to get it working and move on :)

On Tue, Feb 8, 2011 at 3:48 AM, Patrick Plattes <
patrick.plattes at niemann-frey.info> wrote:

> Hi Chris :-),
>
> it looks like a solution for us. I hope I will be able to test it
> today. Which FS version are you using for this script? Did you try it
> with 1.0.7?
>
> Thanks,
>  Patrick
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From babak.freeswitch at gmail.com  Wed Feb 16 10:14:06 2011
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Wed, 16 Feb 2011 10:44:06 +0330
Subject: [Freeswitch-users] connecting freeswitch to a kx-td1232
Message-ID: <AANLkTing2BBWR8QW_6K_KMoONJ2VjGeNh=ab76mSiwQP@mail.gmail.com>

Hi
Is it possible to connect a kx-td1232 to freeswitch for connecting an
office with 8 employees to the current kx-td1232 setup? and if it is
possible what kind of cards do I need and how should I do that?
thanx


From oseslija at gmail.com  Wed Feb 16 10:28:21 2011
From: oseslija at gmail.com (Ognjen Seslija)
Date: Wed, 16 Feb 2011 08:28:21 +0100
Subject: [Freeswitch-users] connecting freeswitch to a kx-td1232
In-Reply-To: <AANLkTing2BBWR8QW_6K_KMoONJ2VjGeNh=ab76mSiwQP@mail.gmail.com>
References: <AANLkTing2BBWR8QW_6K_KMoONJ2VjGeNh=ab76mSiwQP@mail.gmail.com>
Message-ID: <AANLkTikuQE5F43qWGr9O_oW2CpCBgfXVE4=yDeRwtmUk@mail.gmail.com>

I did a lot of Panasonic interconnections with Linksys ATAs connected to CO
ports. I guess it's the only method you can use on this model.


On Wed, Feb 16, 2011 at 8:14 AM, babak yakhchali <babak.freeswitch at gmail.com
> wrote:

> Hi
> Is it possible to connect a kx-td1232 to freeswitch for connecting an
> office with 8 employees to the current kx-td1232 setup? and if it is
> possible what kind of cards do I need and how should I do that?
> thanx
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Feb 16 11:19:44 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 16 Feb 2011 03:19:44 -0500
Subject: [Freeswitch-users] txfax and multiple files
Message-ID: <8D3CBF9281FF476CB88E77DC2FCD7ADD@e1705>

What is the right syntax to send multiple files with
mod_spandsp txfax ?

thanks
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From erik.dekkers at wvds.nl  Wed Feb 16 13:41:35 2011
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Wed, 16 Feb 2011 11:41:35 +0100
Subject: [Freeswitch-users] javascript hanguphook
In-Reply-To: <4D5AA03F.8030307@yellox.de>
References: <4D5AA03F.8030307@yellox.de>
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE07873018D3@FS1.wvds.local>

Hi Christian,

The hangup hook api should be set in the dialplan:
<action application="set" data="api_hangup_hook=jsrun cleanup.js ${uuid}"/>

This will trigger the hangup api once the session is destroyed.

Regards,

Erik

-----Oorspronkelijk bericht-----
Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Christian Hiller
Verzonden: dinsdag 15 februari 2011 16:48
Aan: freeswitch-users at lists.freeswitch.org
Onderwerp: [Freeswitch-users] javascript hanguphook

Hello,

i have a javscript that is called from xml-dialplan ...
function on_hangup(hup_session, how)
{
     console_log("info","got hungup");
     exit();
}

session.execute('bridge','sofia/internal....');
session.setHangupHook(on_hangup);
...

Now i experience, that this function is only called if the session is not answered yet.
Once its answered and then got hung up, the function on_hangup is not called anymore.
Any ideas ?

Kind regards

Christian Hiller

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at 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 k-b-24 at live.com  Wed Feb 16 07:32:48 2011
From: k-b-24 at live.com (Jason b.a)
Date: Wed, 16 Feb 2011 04:32:48 +0000
Subject: [Freeswitch-users] application and media server
Message-ID: <COL111-W50BA6C8284D44C80B5369DACD20@phx.gbl>


Hi  ,
should i use sip server to connect the application server with the freeswitch (media server).
 
client ---- application server --- sip server ---- freeswitch
 
and is MSCML or VXML embedded in the sip message so the application server can control the freeswitch.
 
thank you . 		 	   		  
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From ziga.jakhel at integrum.si  Wed Feb 16 12:54:14 2011
From: ziga.jakhel at integrum.si (=?iso-8859-2?Q?=AEiga_Jakhel?=)
Date: Wed, 16 Feb 2011 09:54:14 +0000
Subject: [Freeswitch-users] Sofia and SIP Service-Route header
Message-ID: <D5C8AC44DB7916499074AB1754372DF0D0F019@HAIL.hq.integrum-it.com>

Hi!

We're having issues with a >funky< ISP - they started advertising a SIP Service-Route header pointing to nowhere (go figure), claiming they need it for their systems to function, and that our UA should not use it, if the initial address is working.
Sofia, however, uses the header dilligently, sending the request to the address specified in the header (in my case going nowhere), despite the functioning base SBC address.

Any ideas? Is my ISP off the rocker, or is Sofia maybe a little overzealous?

Regards,

?iga Jakhel, MSc

Integrum d.o.o.
Smoletova 6, 1230 Dom?ale
Slovenia, EU
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From jgallartm at gmail.com  Wed Feb 16 15:28:35 2011
From: jgallartm at gmail.com (Javier Gallart)
Date: Wed, 16 Feb 2011 13:28:35 +0100
Subject: [Freeswitch-users] g729 packets skipped in passthrough mode
Message-ID: <AANLkTimsYaF+jO_rthDx9Z2Mp-RQt6PdTfE2T+pp1R84@mail.gmail.com>

Hello,

I've looked deeper into this issue and I might have hit a known problem:

RTP_BUG_IGNORE_MARK_BIT = (1 << 2)

/*
  A Huawei SBC has been discovered that sends the mark bit on every single
RTP packet.
  Since this causes the RTP stack to flush it's buffers, it horribly messes
up the timing on the channel.

  This flag will do nothing when an inbound packet contains the mark bit.

 */

The carrier is actually a Huawei SBC, and the problem I reported only
happens when it sends the marker bit  = true in all the rtp packets. I've
tried
<param name="auto-rtp-bugs" data="RTP_BUG_IGNORE_MARK_BIT"/> in the profile
but the problem is still there, is this the right way to override this
nonsense from the Huawei?

Thanks

Javier

>
> ---------- Forwarded message ----------
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 15 Feb 2011 16:38:35 -0600
> Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 56, Issue 177
> Well that's all I could think of:
>
> passthrough is passthrough FS never modifies anything.
> you need to look harder at sip trace, pcaps or other diagnostics for
> some misconfiguration.
> many people use this daily.
>
>
> On Tue, Feb 15, 2011 at 1:12 AM, Javier Gallart <jgallartm at gmail.com>
> wrote:
> >
> > Anthony, thanks for the tip. I haven't seen any change though. I should
> have
> > mentioned that I'm focusing in the rtp stream coming form the callee to
> the
> > caller. The problem is that FS relays to the caller only half of the
> packets
> > received from the callee. The packets arrive at FS at a rate of 1 packet
> > every 20 ms, and with a payload of 20 bytes each. Packets from FS to the
> > caller are sent every 40 ms with a payload of 20 bytes, thus skipping
> half
> > of the information.
> >
> > Thanks
> >>
> >> ---------- Forwarded message ----------
> >> From: Anthony Minessale <anthony.minessale at gmail.com>
> >> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> >> Date: Mon, 14 Feb 2011 10:38:57 -0600
> >> Subject: Re: [Freeswitch-users] g729 packets skipped in passthrough mode
> >> I bet its 20ms vs 30
> >> set passthru_ptime_mismatch to true either in vars.xml or in your
> >> dialplan both legs through export on the a leg or set in the a leg and
> >> in the {} on b.
> >>
> >>
> >> On Mon, Feb 14, 2011 at 10:28 AM, Brian West <brian at freeswitch.org>
> wrote:
> >> > What exactly is the problem?  I see no issue here can you elaborate on
> >> > what you're seeing?
> >> >
> >> > /b
> >> >
> >> > On Feb 14, 2011, at 4:48 AM, Javier Gallart wrote:
> >> >
> >> >> I've tried to explicitly set ptime at 20ms at switch.conf -although
> >> >> it's not necessary afaik-. Has any one experienced this same issue?
> >> >>
> >> >> Thanks in advance..
> >> >>
> >> >> Regards
> >> >
> >> >
>
>
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From Nabble at slickdeals.endjunk.com  Wed Feb 16 17:41:04 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 16 Feb 2011 06:41:04 -0800 (PST)
Subject: [Freeswitch-users] Sofia and SIP Service-Route header
In-Reply-To: <D5C8AC44DB7916499074AB1754372DF0D0F019@HAIL.hq.integrum-it.com>
References: <D5C8AC44DB7916499074AB1754372DF0D0F019@HAIL.hq.integrum-it.com>
Message-ID: <1297867264738-6032006.post@n2.nabble.com>



?iga Jakhel wrote:
> We're having issues with a >funky< ISP - they started advertising a SIP
> Service-Route header pointing to nowhere (go figure), claiming they need
> it for their systems to function, and that our UA should not use it, if
> the initial address is working.
I would definitely stay away from an Internet Service Provider (ISP) that
meddles with a SIP service-route header.

-----
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/Sofia-and-SIP-Service-Route-header-tp6031278p6032006.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From rajesh.npnr at yahoo.com  Wed Feb 16 18:28:05 2011
From: rajesh.npnr at yahoo.com (rex.alex)
Date: Wed, 16 Feb 2011 07:28:05 -0800 (PST)
Subject: [Freeswitch-users] The call dropped prematurely error on txfax
Message-ID: <1297870085994-6032194.post@n2.nabble.com>


Hello,

I am testing the FreeSWITCH mod_spandsp module. I have got a gateway
registered in FreeSWITCH which supports fax and using the following command
to send the fax, but It always ended up with the "result (49) The call
dropped prematurely" error. Also if I remove fax_enable_t38 variables, it's
giving "Reinvite Codec Error!". I have pastebin the entire log in
http://pastebin.freeswitch.org/15394.

originate {absolute_codec_string='PCMU,PCMA',
fax_enable_t38=true,fax_enable_t38_request=true}sofia/gateway/Pulse/1XXXXXXXXXX
&txfax(/var/www/html/faxtest/docs/15Feb2011103600.pdf.tiff)

Please assist.

Thanks,
Rex
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/The-call-dropped-prematurely-error-on-txfax-tp6032194p6032194.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From infos at madovsky.org  Wed Feb 16 18:49:16 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 16 Feb 2011 10:49:16 -0500
Subject: [Freeswitch-users] The call dropped prematurely error on txfax
References: <1297870085994-6032194.post@n2.nabble.com>
Message-ID: <643BF67696204EE089E833DEE6BA2116@e1705>

maybe a media_timeout problem ?

----- Original Message ----- 
From: "rex.alex" <rajesh.npnr at yahoo.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, February 16, 2011 10:28 AM
Subject: [Freeswitch-users] The call dropped prematurely error on txfax


>
> Hello,
>
> I am testing the FreeSWITCH mod_spandsp module. I have got a gateway
> registered in FreeSWITCH which supports fax and using the following 
> command
> to send the fax, but It always ended up with the "result (49) The call
> dropped prematurely" error. Also if I remove fax_enable_t38 variables, 
> it's
> giving "Reinvite Codec Error!". I have pastebin the entire log in
> http://pastebin.freeswitch.org/15394.
>
> originate {absolute_codec_string='PCMU,PCMA',
> fax_enable_t38=true,fax_enable_t38_request=true}sofia/gateway/Pulse/1XXXXXXXXXX
> &txfax(/var/www/html/faxtest/docs/15Feb2011103600.pdf.tiff)
>
> Please assist.
>
> Thanks,
> Rex
> -- 
> View this message in context: 
> http://freeswitch-users.2379917.n2.nabble.com/The-call-dropped-prematurely-error-on-txfax-tp6032194p6032194.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 fs-list at communicatefreely.net  Wed Feb 16 18:53:00 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Wed, 16 Feb 2011 10:53:00 -0500
Subject: [Freeswitch-users] Aastra phone registration lost
In-Reply-To: <16242449-AADE-4068-8523-C1727F66FDF8@freeswitch.org>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>	<4D52B987.7050002@communicatefreely.net>	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>	<4D534AFD.3020401@communicatefreely.net>	<AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>	<60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>	<AANLkTimux4MBM1K_7itE4GnZQHjy_vR2Y6sr_z+u3gkR@mail.gmail.com>	<4D5577CE.1000603@communicatefreely.net>	<AANLkTinJK+e9hZJbk4ur_OQwGV5B7tRLrpPMH=TU5MOo@mail.gmail.com>	<4D59C085.9040008@communicatefreely.net>	<AANLkTikZbWPqf7N1fT-9fF8Fvydi85mJVZ+o8_J=RLjE@mail.gmail.com>	<4D5A9BEA.7070302@communicatefreely.net>	<AANLkTikuozcozSiFFL8-6+s3iRP4Lfy9DT2xFi9TVBOd@mail.gmail.com>
	<16242449-AADE-4068-8523-C1727F66FDF8@freeswitch.org>
Message-ID: <4D5BF2DC.20409@communicatefreely.net>

I have been setting my expires to 600.

Brian West wrote:
> Are you setting your expires to > 300 seconds?
>
> /b
>
> On Feb 15, 2011, at 9:42 AM, Aloysius Lloyd wrote:
>
>>
>>
>> I have the same issue, around 275 phones in the field. I want the 275 
>> phones work with FreeSWITCH.
>>
>>
>> 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
>   



From rajesh.npnr at yahoo.com  Wed Feb 16 19:22:35 2011
From: rajesh.npnr at yahoo.com (rex.alex)
Date: Wed, 16 Feb 2011 08:22:35 -0800 (PST)
Subject: [Freeswitch-users] The call dropped prematurely error on txfax
In-Reply-To: <643BF67696204EE089E833DEE6BA2116@e1705>
References: <1297870085994-6032194.post@n2.nabble.com>
	<643BF67696204EE089E833DEE6BA2116@e1705>
Message-ID: <1297873355400-6032435.post@n2.nabble.com>


Hi,

I have commented the rtp-timeout variables in both sip_profiles (internal &
external) like

 <!--
    
             
 -->

But the result is same. Please assist.

Thanks,
Rex
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/The-call-dropped-prematurely-error-on-txfax-tp6032194p6032435.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Wed Feb 16 19:59:40 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 16 Feb 2011 10:59:40 -0600
Subject: [Freeswitch-users] The call dropped prematurely error on txfax
In-Reply-To: <1297873355400-6032435.post@n2.nabble.com>
References: <1297870085994-6032194.post@n2.nabble.com>
	<643BF67696204EE089E833DEE6BA2116@e1705>
	<1297873355400-6032435.post@n2.nabble.com>
Message-ID: <AANLkTingYEfo=4oaANLS8ZzySqLSaUOOm7ME=HpxxQ-C@mail.gmail.com>

you did not include the revision of FreeSWITCH you have.


On Wed, Feb 16, 2011 at 10:22 AM, rex.alex <rajesh.npnr at yahoo.com> wrote:
>
> Hi,
>
> I have commented the rtp-timeout variables in both sip_profiles (internal &
> external) like
>
> ?<!--
>
>
> ?-->
>
> But the result is same. Please assist.
>
> Thanks,
> Rex
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/The-call-dropped-prematurely-error-on-txfax-tp6032194p6032435.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 msc at freeswitch.org  Wed Feb 16 20:34:40 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 16 Feb 2011 09:34:40 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <AANLkTinWchNsK7HCoe71BC+XjU84AiKJP-JXcRZW3A+B@mail.gmail.com>

Hey folks,

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

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

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

Talk to you soon,
Michael
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From neilp at cs.stanford.edu  Wed Feb 16 21:33:57 2011
From: neilp at cs.stanford.edu (Neil Patel)
Date: Thu, 17 Feb 2011 00:03:57 +0530
Subject: [Freeswitch-users] bridging?
Message-ID: <AANLkTikkWZtatkiLRLdvu9-T2-VdsNRtyAk7ZCMCcF+4@mail.gmail.com>

Hi Folks, newbie question:

I have a simple dialplan that takes calls to an extension and executes a lua
app:

*<extension name="incoming">*
*        <condition field="destination_number" expression="^7777$">*
*            <action application="lua" data="lua/app.lua" />*
*        </condition>*
* </extension>*
*
*

>From the same extension I am initiating calls from FS to endpoints and
executing another lua app. What I want to do is connect (bridge?) those
outbound calls to app.lua. From FS perspective this call should now look
like it was initiated by the caller to 7777.

Is this what call bridging is? What commands/wiki page should I be looking
at to do this?

Thanks in advance,
Neil
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From wagnerspi at gmail.com  Wed Feb 16 21:45:06 2011
From: wagnerspi at gmail.com (Wagner)
Date: Wed, 16 Feb 2011 16:45:06 -0200
Subject: [Freeswitch-users] Benchmarking
Message-ID: <AANLkTinGJLO9EXk3+fLfBUwAUyNc9MzfCLu7dg=dnfHG@mail.gmail.com>

Hello,

I'd like to know how could I benchmark a FreeSwitch server?

Is there any tool to stress it?

Thanks
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From msc at freeswitch.org  Wed Feb 16 22:40:52 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 16 Feb 2011 11:40:52 -0800
Subject: [Freeswitch-users] Benchmarking
In-Reply-To: <AANLkTinGJLO9EXk3+fLfBUwAUyNc9MzfCLu7dg=dnfHG@mail.gmail.com>
References: <AANLkTinGJLO9EXk3+fLfBUwAUyNc9MzfCLu7dg=dnfHG@mail.gmail.com>
Message-ID: <AANLkTin6ryyD9NBH5_SbCPvL76tBvSHiFaoW_upKpd1h@mail.gmail.com>

The best benchmark is to put some real traffic on it on a real network
connected to real VoIP carriers.

You can simulate call traffic with tools like SIPp but those simulations are
limited. It's better to sketch out your needs before hand and then look at
your available options for processing power, RAM, HDD speed, and bandwidth.

What kind of system are you building?

-MC

On Wed, Feb 16, 2011 at 10:45 AM, Wagner <wagnerspi at gmail.com> wrote:

> Hello,
>
> I'd like to know how could I benchmark a FreeSwitch server?
>
> Is there any tool to stress 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 mitch.capper at gmail.com  Wed Feb 16 22:40:41 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 16 Feb 2011 11:40:41 -0800
Subject: [Freeswitch-users] Anyone able to get freeswitch to register a
 gateway over TLS or TCP? I can't get it to not send UDP packets for reg.
Message-ID: <AANLkTinis3fd8Uivvh96UDEmfbwDt3_uASAxuOKgRDNX@mail.gmail.com>

I am trying to get freeswitch to register with another freeswitch server (on
TLS port 5061) however default the config settings I believe should set it,
it always trys to register over UDP (after registration or with registration
set to false however it does send TCP then for outbound calls).   If anyone
has a gateway config that actually registers over TCP or TLS that would be
great.
The client gateway/profile config at:
http://pastebin.freeswitch.org/15383is what i believe should work but
isn't

I have tried setting contact-params on the gateway to tport=tls tport=tcp
tport=tcp;transport=tls and transport=tls  without any luck.  Anyway even
TCP would be helpful as it would allow me to trace better how the sip
registration decides to use TCP or UDP.   I tried digging into libsofia but
its a bit hard to trace and understand with its generic event structure and
tagging (plus the documentation isn't the most rebust on registration
options).

Thanks

~Mitch
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From rajesh.npnr at yahoo.com  Wed Feb 16 22:49:33 2011
From: rajesh.npnr at yahoo.com (rex.alex)
Date: Wed, 16 Feb 2011 11:49:33 -0800 (PST)
Subject: [Freeswitch-users] The call dropped prematurely error on txfax
In-Reply-To: <AANLkTingYEfo=4oaANLS8ZzySqLSaUOOm7ME=HpxxQ-C@mail.gmail.com>
References: <1297870085994-6032194.post@n2.nabble.com>
	<643BF67696204EE089E833DEE6BA2116@e1705>
	<1297873355400-6032435.post@n2.nabble.com>
	<AANLkTingYEfo=4oaANLS8ZzySqLSaUOOm7ME=HpxxQ-C@mail.gmail.com>
Message-ID: <1297885773534-6033268.post@n2.nabble.com>


I am sorry. It's 

FreeSWITCH Version 1.0.head (git-564dc7e 2010-10-21 22-35-24 -0500)

Thanks,
Rex
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/The-call-dropped-prematurely-error-on-txfax-tp6032194p6033268.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Wed Feb 16 22:56:31 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 16 Feb 2011 11:56:31 -0800
Subject: [Freeswitch-users] bridging?
In-Reply-To: <AANLkTikkWZtatkiLRLdvu9-T2-VdsNRtyAk7ZCMCcF+4@mail.gmail.com>
References: <AANLkTikkWZtatkiLRLdvu9-T2-VdsNRtyAk7ZCMCcF+4@mail.gmail.com>
Message-ID: <AANLkTikWBopbR5yk6xAy-y6Ueqm+4wS+=_MNVah9bqD1@mail.gmail.com>

Can you use originate?

originate sofia/gateway/gwname/123456789 7777

that will create the outbound call to the destination number (1233456789)
and then bridge it to 7777.

-MC

On Wed, Feb 16, 2011 at 10:33 AM, Neil Patel <neilp at cs.stanford.edu> wrote:

> Hi Folks, newbie question:
>
> I have a simple dialplan that takes calls to an extension and executes a
> lua app:
>
> *<extension name="incoming">*
> *        <condition field="destination_number" expression="^7777$">*
> *            <action application="lua" data="lua/app.lua" />*
> *        </condition>*
> * </extension>*
> *
> *
>
> From the same extension I am initiating calls from FS to endpoints and
> executing another lua app. What I want to do is connect (bridge?) those
> outbound calls to app.lua. From FS perspective this call should now look
> like it was initiated by the caller to 7777.
>
> Is this what call bridging is? What commands/wiki page should I be looking
> at to do this?
>
> Thanks in advance,
> 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 rajesh.npnr at yahoo.com  Wed Feb 16 23:21:52 2011
From: rajesh.npnr at yahoo.com (rex.alex)
Date: Wed, 16 Feb 2011 12:21:52 -0800 (PST)
Subject: [Freeswitch-users] Core dump error
Message-ID: <1297887712012-6033400.post@n2.nabble.com>


Hi 

I was using the FreeSWITCH 1.0.6 for quite some time in production and was
very stable and recently I had upgraded to latest git to use the mod_spandsp
module. From then on I used to get core dump issue sometimes and hence
yesterday I did a git pull && make current on source. But today, till now
FreeSWITCH has been core dumped for 5 times,  could not find the reason. I
have pastebin the core trace in http://pastebin.freeswitch.org/15402

Version :
FreeSWITCH Version 1.0.head (git-722b2ce 2011-02-15 19-01-45 -0600)

OS : Cent OS 5.5

Please help me to resolve the same.

Thanks,
Rex

-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Core-dump-error-tp6033400p6033400.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Thu Feb 17 00:01:21 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 16 Feb 2011 15:01:21 -0600
Subject: [Freeswitch-users] Core dump error
In-Reply-To: <1297887712012-6033400.post@n2.nabble.com>
References: <1297887712012-6033400.post@n2.nabble.com>
Message-ID: <AANLkTin=diX+LxsuU2ach=g+OStGC34UK+stzeXkxhPG@mail.gmail.com>

your backtrace looks like an issue that is already fixed.
please do the following

1) stop FreeSWITCH
2) git pull and visually verify that the GIT is updating with no errors
3) make current
4) restart FreeSWITCH

if you still get cores after that post them to http://jira.freeswitch.org


On Wed, Feb 16, 2011 at 2:21 PM, rex.alex <rajesh.npnr at yahoo.com> wrote:
>
> Hi
>
> I was using the FreeSWITCH 1.0.6 for quite some time in production and was
> very stable and recently I had upgraded to latest git to use the mod_spandsp
> module. From then on I used to get core dump issue sometimes and hence
> yesterday I did a git pull && make current on source. But today, till now
> FreeSWITCH has been core dumped for 5 times, ?could not find the reason. I
> have pastebin the core trace in http://pastebin.freeswitch.org/15402
>
> Version :
> FreeSWITCH Version 1.0.head (git-722b2ce 2011-02-15 19-01-45 -0600)
>
> OS : Cent OS 5.5
>
> Please help me to resolve the same.
>
> Thanks,
> Rex
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Core-dump-error-tp6033400p6033400.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.capper at gmail.com  Thu Feb 17 00:08:18 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 16 Feb 2011 13:08:18 -0800
Subject: [Freeswitch-users] Anyone able to get freeswitch to register a
 gateway over TLS or TCP? I can't get it to not send UDP packets for reg.
In-Reply-To: <AANLkTinis3fd8Uivvh96UDEmfbwDt3_uASAxuOKgRDNX@mail.gmail.com>
References: <AANLkTinis3fd8Uivvh96UDEmfbwDt3_uASAxuOKgRDNX@mail.gmail.com>
Message-ID: <AANLkTimSNzA6S_7zVXSZcBNfjkTMkEyoicqhqBaaQUyQ@mail.gmail.com>

Nevermind, jlenk figured out the problem.

Thanks
~Mitch
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From Nabble at slickdeals.endjunk.com  Thu Feb 17 00:21:53 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 16 Feb 2011 13:21:53 -0800 (PST)
Subject: [Freeswitch-users] Anyone able to get freeswitch to register a
 gateway over TLS or TCP? I can't get it to not send UDP packets for reg.
In-Reply-To: <AANLkTimSNzA6S_7zVXSZcBNfjkTMkEyoicqhqBaaQUyQ@mail.gmail.com>
References: <AANLkTinis3fd8Uivvh96UDEmfbwDt3_uASAxuOKgRDNX@mail.gmail.com>
	<AANLkTimSNzA6S_7zVXSZcBNfjkTMkEyoicqhqBaaQUyQ@mail.gmail.com>
Message-ID: <1297891313399-6033586.post@n2.nabble.com>



Mitch Capper wrote:
> Nevermind, jlenk figured out the problem.
You should post here the solution for posterities.

-----
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/Anyone-able-to-get-freeswitch-to-register-a-gateway-over-TLS-or-TCP-I-can-t-get-it-to-not-send-UDP-p-tp6033245p6033586.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Thu Feb 17 00:24:20 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 16 Feb 2011 15:24:20 -0600
Subject: [Freeswitch-users] The call dropped prematurely error on txfax
In-Reply-To: <1297885773534-6033268.post@n2.nabble.com>
References: <1297870085994-6032194.post@n2.nabble.com>
	<643BF67696204EE089E833DEE6BA2116@e1705>
	<1297873355400-6032435.post@n2.nabble.com>
	<AANLkTingYEfo=4oaANLS8ZzySqLSaUOOm7ME=HpxxQ-C@mail.gmail.com>
	<1297885773534-6033268.post@n2.nabble.com>
Message-ID: <AANLkTiku+q9B_FeSErvMuXSog6DQ4mMtDmxn_TjtG9Zw@mail.gmail.com>

There were some big changes for t.38 in that time, you should try latest git


On Wed, Feb 16, 2011 at 1:49 PM, rex.alex <rajesh.npnr at yahoo.com> wrote:
>
> I am sorry. It's
>
> FreeSWITCH Version 1.0.head (git-564dc7e 2010-10-21 22-35-24 -0500)
>
> Thanks,
> Rex
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/The-call-dropped-prematurely-error-on-txfax-tp6032194p6033268.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.capper at gmail.com  Thu Feb 17 01:16:50 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 16 Feb 2011 14:16:50 -0800
Subject: [Freeswitch-users] Anyone able to get freeswitch to register a
 gateway over TLS or TCP? I can't get it to not send UDP packets for reg.
In-Reply-To: <1297891313399-6033586.post@n2.nabble.com>
References: <AANLkTinis3fd8Uivvh96UDEmfbwDt3_uASAxuOKgRDNX@mail.gmail.com>
	<AANLkTimSNzA6S_7zVXSZcBNfjkTMkEyoicqhqBaaQUyQ@mail.gmail.com>
	<1297891313399-6033586.post@n2.nabble.com>
Message-ID: <AANLkTik+vzwO+Zbun8zBOX5KhpgD8TOizUQfZn_xb8vL@mail.gmail.com>

There is a bug in freeswitch with alternate transports and external-sip-ip
not being set to an actual ip.  We are going to fix the bug and once its
fixed then freeswitch will behave as expected.

~Mitch

On Wed, Feb 16, 2011 at 1:21 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
>
> Mitch Capper wrote:
> > Nevermind, jlenk figured out the problem.
> You should post here the solution for posterities.
>
> -----
> 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/Anyone-able-to-get-freeswitch-to-register-a-gateway-over-TLS-or-TCP-I-can-t-get-it-to-not-send-UDP-p-tp6033245p6033586.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 freeswitch.org  Thu Feb 17 01:53:51 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 16 Feb 2011 16:53:51 -0600
Subject: [Freeswitch-users] OSTAG Facebook page
Message-ID: <407F60C9-39E3-4C1D-8AF7-C42982F8C130@freeswitch.org>

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

Please join the OSTAG facebook community... it has taken a long time to get non-profit status so please help out and donate if you can... get involved.

Thanks,
Brian



From msc at freeswitch.org  Thu Feb 17 01:56:41 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 16 Feb 2011 14:56:41 -0800
Subject: [Freeswitch-users] OSTAG Facebook page
In-Reply-To: <407F60C9-39E3-4C1D-8AF7-C42982F8C130@freeswitch.org>
References: <407F60C9-39E3-4C1D-8AF7-C42982F8C130@freeswitch.org>
Message-ID: <AANLkTikTDfzh8Usy8qd6kN4xNYTnbC23VYA743iX0xSr@mail.gmail.com>

Also, follow us on twitter:
http://twitter.com/ostag

-MC

On Wed, Feb 16, 2011 at 2:53 PM, Brian West <brian at freeswitch.org> wrote:

>
> http://www.facebook.com/pages/Open-Source-Telephony-Advancement-Group-Inc/188675947819805
>
> Please join the OSTAG facebook community... it has taken a long time to get
> non-profit status so please help out and donate if you can... get involved.
>
> Thanks,
> Brian
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From k-b-24 at live.com  Thu Feb 17 02:51:30 2011
From: k-b-24 at live.com (Jason b.a)
Date: Wed, 16 Feb 2011 23:51:30 +0000
Subject: [Freeswitch-users] IVR application
Message-ID: <COL111-W141D8C49354C749B3AF2F7ACD20@phx.gbl>


Hey guys, 
i need to implement an IVR application, but i am wondering if it is required to use a SIP SERVER between my IVR application and FREESWITCH.
thank you 		 	   		  
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From Nabble at slickdeals.endjunk.com  Thu Feb 17 06:29:29 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 16 Feb 2011 19:29:29 -0800 (PST)
Subject: [Freeswitch-users] Parallel compilation on a Linux machine
Message-ID: <1297913369896-6034541.post@n2.nabble.com>


I tried to use make -j 4 to compile FS git on my AMD64 PhenomII X3 machine.
The compilation doesn't seem to fully utilize all the 3 CPU cores. As a
matter of fact, it takes almost the same amount of time (about 55 minutes)
to compile FS git with only -j 4 switch. So, does anyone here know how to
make compile FS git in parallel to make use of all CPU cores? This way, I
don't have to spend about an hour just to compile FS git.

-----
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/Parallel-compilation-on-a-Linux-machine-tp6034541p6034541.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From lloyd.aloysius at gmail.com  Thu Feb 17 06:34:02 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Wed, 16 Feb 2011 22:34:02 -0500
Subject: [Freeswitch-users] Aastra phone registration lost
In-Reply-To: <4D5BF2DC.20409@communicatefreely.net>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>
	<4D52B987.7050002@communicatefreely.net>
	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>
	<4D534AFD.3020401@communicatefreely.net>
	<AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>
	<60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>
	<AANLkTimux4MBM1K_7itE4GnZQHjy_vR2Y6sr_z+u3gkR@mail.gmail.com>
	<4D5577CE.1000603@communicatefreely.net>
	<AANLkTinJK+e9hZJbk4ur_OQwGV5B7tRLrpPMH=TU5MOo@mail.gmail.com>
	<4D59C085.9040008@communicatefreely.net>
	<AANLkTikZbWPqf7N1fT-9fF8Fvydi85mJVZ+o8_J=RLjE@mail.gmail.com>
	<4D5A9BEA.7070302@communicatefreely.net>
	<AANLkTikuozcozSiFFL8-6+s3iRP4Lfy9DT2xFi9TVBOd@mail.gmail.com>
	<16242449-AADE-4068-8523-C1727F66FDF8@freeswitch.org>
	<4D5BF2DC.20409@communicatefreely.net>
Message-ID: <AANLkTik5pgWLDqg3Q2FDsc3i+NJ9vo77VGjrX7v-2bK6@mail.gmail.com>

Hi All,


I stay away with Aastra phone for a long time  and today I did some tests.
All of my test ... the phones not reliable with FreeSWITCH

I try both 6731i and 57i with the most recent firmware

Here is the configuration files

*aastra.cfg*

dhcp: 1
sip digit timeout: 3
sip dial plan:
"x+#|xx+*|[2-9]XX[2-9]XXXXXX|1[2-9]XX[2-9]XXXXXX|1XXXXXXXXXX|[2-3]XX|67[2-9]XX[2-9]XXXXXX"
sip rport: 1
sip customized
codec:payload=9;ptime=20;silsupp=off,payload=0;ptime=20;silsupp=off,payload=18;ptime=20;silsupp=off
#sip registration period: 120
#sip registration renewal timer: 15
headset tx gain: -3
headset sidetone gain: -3
handset tx gain: -3
handset sidetone gain: -3
handsfree tx gain: 0
handset volume: 5     #RX volumes - user adjustable, so easily changed
speaker volume: 5
ringer volume: 3
web interface enabled: 1
live dialpad: 1
missed calls indicator disabled: 1
suppress dtmf playback: 0
#audio mode: 2 #0 = speaker (default)1 = headset 2 = speaker/headset 3 =
headset/speaker
time server disabled: 0
time server1: pool.ntp.org
#directory
directory 1: internal_list.csv
directory 2: external_list.csv

*mac.cfg*

directed call pickup: 1
directed call pickup prefix: **
#
sip line1 screen name: Ext 203
sip line1 display name: Ext 203
sip line1 auth name: 203
sip line1 user name: 203
sip line1 password: *********
sip line1 vmail: *97
sip line1 mode: 0
#
sip line1 proxy ip: aastra.mydomain.com
#sip line1 proxy port: 5060
sip line1 registrar ip: aastra.mydomain.com
sip line1 registration period: 300
#sip line1 registrar port: 5060

------------

1. FreeSWITCH Registration shows two entires in the internal profile and
when I try to call to the extension two lines on the phones rings .... no
idea why this is happen?

Call-ID:        d8fd4c8795801cd2
User:           203 at aastra.mydomain.com
Contact:        "Ext 203" <sip:203 at 192.168.240.106:5060
;transport=udp;fs_nat=yes;fs_path=sip%3A203%40173.33.178.49%3A1627%3Btransport%3Dudp>
Agent:          Aastra 6731i/2.6.0.2010
Status:         Registered(UDP-NAT)(unknown) EXP(2011-02-16 22:24:36)
EXPSECS(310)
Host:           li176-12
IP:             173.33.178.49
Port:           1627
Auth-User:      203
Auth-Realm:     aastra.mydomain.com
MWI-Account:    203 at aastra.mydomain.com

Call-ID:        d8fd4c8795801cd2
User:           203 at aastra.mydomain.com
Contact:        "Ext 203" <sip:203 at 173.33.178.49:1627;transport=udp>
Agent:          Aastra 6731i/2.6.0.2010
Status:         Registered(UDP)(unknown) EXP(2011-02-16 22:24:36)
EXPSECS(310)
Host:           li176-12
IP:             173.33.178.49
Port:           1627
Auth-User:      203
Auth-Realm:     aastra.mydomain.com
MWI-Account:    203 at aastra.mydomain.com


2. After first registration expires .... FreeSWITCH internal registration
status shows the following entry. When I dial the extension now there is a
long delay ... FreeSWITCH dialing and waiting then goes to voicemail.

Call-ID:        d8fd4c8795801cd2
User:           203 at aastra.mydomain.com
Contact:        "Ext 203" <sip:203 at 173.33.178.49:1627;transport=udp>
Agent:          Aastra 6731i/2.6.0.2010
Status:         Registered(UDP)(unknown) EXP(2011-02-16 22:31:25)
EXPSECS(266)
Host:           li176-12
IP:             173.33.178.49
Port:           1631
Auth-User:      203
Auth-Realm:     aastra.mydomain.com
MWI-Account:    203 at aastra.mydomain.com

3. Phones Goes randomly "No Service" .... I see this problem several times
in the past.

-------------

At the same time in my lab Linksys and Polycom Phones working without any
issues with the default settings.

-------------

my question what is Aastra Doing differently?

Thanks
Lloyd

On Wed, Feb 16, 2011 at 10:53 AM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:

> I have been setting my expires to 600.
>
> Brian West wrote:
> > Are you setting your expires to > 300 seconds?
> >
> > /b
> >
> > On Feb 15, 2011, at 9:42 AM, Aloysius Lloyd wrote:
> >
> >>
> >>
> >> I have the same issue, around 275 phones in the field. I want the 275
> >> phones work with FreeSWITCH.
> >>
> >>
> >> 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
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 17 06:54:53 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Wed, 16 Feb 2011 19:54:53 -0800
Subject: [Freeswitch-users] Parallel compilation on a Linux machine
In-Reply-To: <1297913369896-6034541.post@n2.nabble.com>
References: <1297913369896-6034541.post@n2.nabble.com>
Message-ID: <AANLkTinscs9H1tGtJB4tvLDgK--cxK+CZU8H0ZRgFQOk@mail.gmail.com>

How many modules are you compiling at once anyways? FS git never seem
to take more than approx. 15-20 minutes when it's being compiled on my
AMD Athlon II Dual Core machine. May I ask how much RAM do you have on
your box?

On Wed, Feb 16, 2011 at 7:29 PM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
> I tried to use make -j 4 to compile FS git on my AMD64 PhenomII X3 machine.
> The compilation doesn't seem to fully utilize all the 3 CPU cores. As a
> matter of fact, it takes almost the same amount of time (about 55 minutes)
> to compile FS git with only -j 4 switch. So, does anyone here know how to
> make compile FS git in parallel to make use of all CPU cores? This way, I
> don't have to spend about an hour just to compile FS git.
>
> -----
> 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/Parallel-compilation-on-a-Linux-machine-tp6034541p6034541.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 curriegrad2004 at gmail.com  Thu Feb 17 07:02:29 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Wed, 16 Feb 2011 20:02:29 -0800
Subject: [Freeswitch-users] IVR application
In-Reply-To: <COL111-W141D8C49354C749B3AF2F7ACD20@phx.gbl>
References: <COL111-W141D8C49354C749B3AF2F7ACD20@phx.gbl>
Message-ID: <AANLkTi=0b4iMTbrsjJ7+PcPoyJMZo+gc44+2KYbko2Ld@mail.gmail.com>

FreeSwitch can definitely load your IVR through it's own API when it
needs to interact with modules. How is your IVR application set up
anyways?

On Wed, Feb 16, 2011 at 3:51 PM, Jason b.a <k-b-24 at live.com> wrote:
> Hey guys,
> i need to implement an IVR application, but i am wondering if it is required
> to use a SIP SERVER between my IVR application and FREESWITCH.
> thank you
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 abubacker at bksystems.co.in  Thu Feb 17 08:15:27 2011
From: abubacker at bksystems.co.in (abubacker)
Date: Thu, 17 Feb 2011 10:45:27 +0530
Subject: [Freeswitch-users] non-preemption in play_and_get_digits
Message-ID: <4D5CAEEF.4000700@bksys.co.in>

Is this possible to set the non  preemption mode int the 
play_and_get_digits_application ?
If yes then how ?


-- 
Best regards,
N.Abubacker ,
Associate system engineer ,
bk systems pvt ltd ,
Ph : 9144-43902701

Disclaimer: http://www.bksystems.co.in/email-policy



From sharad at coraltele.com  Thu Feb 17 10:33:03 2011
From: sharad at coraltele.com (sharad)
Date: Thu, 17 Feb 2011 13:03:03 +0530
Subject: [Freeswitch-users] IVR application
References: <COL111-W141D8C49354C749B3AF2F7ACD20@phx.gbl>
Message-ID: <896E9DAD5C82443A8A13EE7DBEC959A2@sharad>


  ----- Original Message ----- 
  From: Jason b.a 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, February 17, 2011 5:21 AM
  Subject: [Freeswitch-users] IVR application


  Hey guys, 
  i need to implement an IVR application, but i am wondering if it is required to use a SIP SERVER between my IVR application and FREESWITCH.
  thank you
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From sharad at coraltele.com  Thu Feb 17 10:35:46 2011
From: sharad at coraltele.com (sharad)
Date: Thu, 17 Feb 2011 13:05:46 +0530
Subject: [Freeswitch-users] IVR application
References: <COL111-W141D8C49354C749B3AF2F7ACD20@phx.gbl>
Message-ID: <9F60E5CFBCF04E968E7657100481A51B@sharad>

It deponds upon the application. If you are expecting the calls from PSTN, SIP server / external sip gateways may be required. If only IP users are expected, no SIP server / gateway should be required.

This is my openion...

Thanks
Sharad, India


 ----- Original Message ----- 
  From: Jason b.a 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, February 17, 2011 5:21 AM
  Subject: [Freeswitch-users] IVR application


  Hey guys, 
  i need to implement an IVR application, but i am wondering if it is required to use a SIP SERVER between my IVR application and FREESWITCH.
  thank you
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From neilp at cs.stanford.edu  Thu Feb 17 11:18:09 2011
From: neilp at cs.stanford.edu (Neil Patel)
Date: Thu, 17 Feb 2011 13:48:09 +0530
Subject: [Freeswitch-users] bridging?
In-Reply-To: <AANLkTikWBopbR5yk6xAy-y6Ueqm+4wS+=_MNVah9bqD1@mail.gmail.com>
References: <AANLkTikkWZtatkiLRLdvu9-T2-VdsNRtyAk7ZCMCcF+4@mail.gmail.com>
	<AANLkTikWBopbR5yk6xAy-y6Ueqm+4wS+=_MNVah9bqD1@mail.gmail.com>
Message-ID: <AANLkTimCeaoCc5pTs+a2vhhyQkDnuFX2dstB1uNy3uX_@mail.gmail.com>

Can I use this from within a live outbound (FS -> 123456789) call initiated
through a lua app? Once this command is executed, what happens to that
original call?

On Thu, Feb 17, 2011 at 1:26 AM, Michael Collins <msc at freeswitch.org> wrote:

> Can you use originate?
>
> originate sofia/gateway/gwname/123456789 7777
>
> that will create the outbound call to the destination number (1233456789)
> and then bridge it to 7777.
>
> -MC
>
> On Wed, Feb 16, 2011 at 10:33 AM, Neil Patel <neilp at cs.stanford.edu>wrote:
>
>> Hi Folks, newbie question:
>>
>> I have a simple dialplan that takes calls to an extension and executes a
>> lua app:
>>
>> *<extension name="incoming">*
>> *        <condition field="destination_number" expression="^7777$">*
>> *            <action application="lua" data="lua/app.lua" />*
>> *        </condition>*
>> * </extension>*
>> *
>> *
>>
>> From the same extension I am initiating calls from FS to endpoints and
>> executing another lua app. What I want to do is connect (bridge?) those
>> outbound calls to app.lua. From FS perspective this call should now look
>> like it was initiated by the caller to 7777.
>>
>> Is this what call bridging is? What commands/wiki page should I be looking
>> at to do this?
>>
>> Thanks in advance,
>> 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
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mayamatakeshi at gmail.com  Thu Feb 17 12:46:08 2011
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 17 Feb 2011 18:46:08 +0900
Subject: [Freeswitch-users] Can we Intercept a sleeping channel?
Message-ID: <AANLkTim6WGevD7dWnGZWExNR6ioFZ4KHmA7BVLwjHFit@mail.gmail.com>

To hold a call to be picked up using intercept, I've tried this with the
intention of having the call sent to voicemail if no one does the pickup:

<action application="sleep" data="20000"/>
<action application="answer"/>
<action application="voicemail" data="default user at domain.com"/>

However, when I use application intercept, sleep is not aborted and we have
no audio until it finishes.
I was expecting intercept would abort the sleep application the same way as
it does for example for application playback.

I believe this can be solved by using application park and sched_transfer.
But I thought I could ask if there would be a way of doing it without
splitting the dialplan entities and keep it simple.

br,
takeshi
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From Nabble at slickdeals.endjunk.com  Thu Feb 17 15:05:36 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 17 Feb 2011 04:05:36 -0800 (PST)
Subject: [Freeswitch-users] Parallel compilation on a Linux machine
In-Reply-To: <AANLkTinscs9H1tGtJB4tvLDgK--cxK+CZU8H0ZRgFQOk@mail.gmail.com>
References: <1297913369896-6034541.post@n2.nabble.com>
	<AANLkTinscs9H1tGtJB4tvLDgK--cxK+CZU8H0ZRgFQOk@mail.gmail.com>
Message-ID: <1297944336320-6035621.post@n2.nabble.com>



curriegrad2004 wrote:
> How many modules are you compiling at once anyways?
Almost all modules.


FS git never seem to take more than approx. 15-20 minutes when it's being
compiled on my AMD Athlon II Dual Core machine. May I ask how much RAM do
you have on your box?
2GB RAM with no X11 windows running.

-----
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/Parallel-compilation-on-a-Linux-machine-tp6034541p6035621.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jgallartm at gmail.com  Thu Feb 17 15:21:11 2011
From: jgallartm at gmail.com (Javier Gallart)
Date: Thu, 17 Feb 2011 13:21:11 +0100
Subject: [Freeswitch-users] g729 packets skipped in passthrough mode
Message-ID: <AANLkTikQQJYBLMkiXjqJvZPtkBu8C4-D2CUvaYcsM2KW@mail.gmail.com>

>
> Hello
>

my mistake... I had written data="RTP_BUG_IGNORE_MARK_BIT" instead of
value="RTP_BUG_IGNORE_MARK_BIT". It's working now. Keep this in mind if you
ever happen to put a Huawei SBC in from of a FS!

Thanks

Javier

>
> ---------- Forwarded message ----------
> From: Javier Gallart <jgallartm at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Wed, 16 Feb 2011 13:28:35 +0100
> Subject: Re: [Freeswitch-users] g729 packets skipped in passthrough mode
> Hello,
>
> I've looked deeper into this issue and I might have hit a known problem:
>
> RTP_BUG_IGNORE_MARK_BIT = (1 << 2)
>
> /*
>   A Huawei SBC has been discovered that sends the mark bit on every single
> RTP packet.
>   Since this causes the RTP stack to flush it's buffers, it horribly messes
> up the timing on the channel.
>
>   This flag will do nothing when an inbound packet contains the mark bit.
>
>  */
>
> The carrier is actually a Huawei SBC, and the problem I reported only
> happens when it sends the marker bit  = true in all the rtp packets. I've
> tried
> <param name="auto-rtp-bugs" data="RTP_BUG_IGNORE_MARK_BIT"/> in the profile
> but the problem is still there, is this the right way to override this
> nonsense from the Huawei?
>
> Thanks
>
> Javier
>
>>
>> ---------- Forwarded message ----------
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Date: Tue, 15 Feb 2011 16:38:35 -0600
>> Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 56, Issue 177
>> Well that's all I could think of:
>>
>> passthrough is passthrough FS never modifies anything.
>> you need to look harder at sip trace, pcaps or other diagnostics for
>> some misconfiguration.
>> many people use this daily.
>>
>>
>> On Tue, Feb 15, 2011 at 1:12 AM, Javier Gallart <jgallartm at gmail.com>
>> wrote:
>> >
>> > Anthony, thanks for the tip. I haven't seen any change though. I should
>> have
>> > mentioned that I'm focusing in the rtp stream coming form the callee to
>> the
>> > caller. The problem is that FS relays to the caller only half of the
>> packets
>> > received from the callee. The packets arrive at FS at a rate of 1 packet
>> > every 20 ms, and with a payload of 20 bytes each. Packets from FS to the
>> > caller are sent every 40 ms with a payload of 20 bytes, thus skipping
>> half
>> > of the information.
>> >
>> > Thanks
>> >>
>> >> ---------- Forwarded message ----------
>> >> From: Anthony Minessale <anthony.minessale at gmail.com>
>> >> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> >> Date: Mon, 14 Feb 2011 10:38:57 -0600
>> >> Subject: Re: [Freeswitch-users] g729 packets skipped in passthrough
>> mode
>> >> I bet its 20ms vs 30
>> >> set passthru_ptime_mismatch to true either in vars.xml or in your
>> >> dialplan both legs through export on the a leg or set in the a leg and
>> >> in the {} on b.
>> >>
>> >>
>> >> On Mon, Feb 14, 2011 at 10:28 AM, Brian West <brian at freeswitch.org>
>> wrote:
>> >> > What exactly is the problem?  I see no issue here can you elaborate
>> on
>> >> > what you're seeing?
>> >> >
>> >> > /b
>> >> >
>> >> > On Feb 14, 2011, at 4:48 AM, Javier Gallart wrote:
>> >> >
>> >> >> I've tried to explicitly set ptime at 20ms at switch.conf -although
>> >> >> it's not necessary afaik-. Has any one experienced this same issue?
>> >> >>
>> >> >> Thanks in advance..
>> >> >>
>> >> >> Regards
>> >> >
>> >> >
>>
>>
>
>
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From vetali100 at gmail.com  Thu Feb 17 15:31:22 2011
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Thu, 17 Feb 2011 14:31:22 +0200
Subject: [Freeswitch-users] g729 packets skipped in passthrough mode
In-Reply-To: <AANLkTikQQJYBLMkiXjqJvZPtkBu8C4-D2CUvaYcsM2KW@mail.gmail.com>
References: <AANLkTikQQJYBLMkiXjqJvZPtkBu8C4-D2CUvaYcsM2KW@mail.gmail.com>
Message-ID: <AANLkTikW4+Xiy9RFY3jTwANe0uC0m=++a-iSRRX5A4fZ@mail.gmail.com>

Just wondering - what is this brilliant idea which made Huawei engineers to
setup their switch behave this way?

Is there any REAL benefit? What did they try to achieve by duplicating each
RTP packet...

(Or maybe it is so strange way to assure the reliability of the traffic? At
least one packet out of 2 will reach the destination) :)




2011/2/17 Javier Gallart <jgallartm at gmail.com>

> Hello
>>
>
> my mistake... I had written data="RTP_BUG_IGNORE_MARK_BIT" instead of
> value="RTP_BUG_IGNORE_MARK_BIT". It's working now. Keep this in mind if you
> ever happen to put a Huawei SBC in from of a FS!
>
> Thanks
>
> Javier
>
>>
>> ---------- Forwarded message ----------
>> From: Javier Gallart <jgallartm at gmail.com>
>> To: freeswitch-users at lists.freeswitch.org
>> Date: Wed, 16 Feb 2011 13:28:35 +0100
>> Subject: Re: [Freeswitch-users] g729 packets skipped in passthrough mode
>> Hello,
>>
>> I've looked deeper into this issue and I might have hit a known problem:
>>
>> RTP_BUG_IGNORE_MARK_BIT = (1 << 2)
>>
>> /*
>>   A Huawei SBC has been discovered that sends the mark bit on every
>> single RTP packet.
>>   Since this causes the RTP stack to flush it's buffers, it horribly
>> messes up the timing on the channel.
>>
>>   This flag will do nothing when an inbound packet contains the mark bit.
>>
>>  */
>>
>> The carrier is actually a Huawei SBC, and the problem I reported only
>> happens when it sends the marker bit  = true in all the rtp packets. I've
>> tried
>> <param name="auto-rtp-bugs" data="RTP_BUG_IGNORE_MARK_BIT"/> in the
>> profile but the problem is still there, is this the right way to override
>> this nonsense from the Huawei?
>>
>> Thanks
>>
>> Javier
>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Anthony Minessale <anthony.minessale at gmail.com>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Date: Tue, 15 Feb 2011 16:38:35 -0600
>>> Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 56, Issue
>>> 177
>>> Well that's all I could think of:
>>>
>>> passthrough is passthrough FS never modifies anything.
>>> you need to look harder at sip trace, pcaps or other diagnostics for
>>> some misconfiguration.
>>> many people use this daily.
>>>
>>>
>>> On Tue, Feb 15, 2011 at 1:12 AM, Javier Gallart <jgallartm at gmail.com>
>>> wrote:
>>> >
>>> > Anthony, thanks for the tip. I haven't seen any change though. I should
>>> have
>>> > mentioned that I'm focusing in the rtp stream coming form the callee to
>>> the
>>> > caller. The problem is that FS relays to the caller only half of the
>>> packets
>>> > received from the callee. The packets arrive at FS at a rate of 1
>>> packet
>>> > every 20 ms, and with a payload of 20 bytes each. Packets from FS to
>>> the
>>> > caller are sent every 40 ms with a payload of 20 bytes, thus skipping
>>> half
>>> > of the information.
>>> >
>>> > Thanks
>>> >>
>>> >> ---------- Forwarded message ----------
>>> >> From: Anthony Minessale <anthony.minessale at gmail.com>
>>> >> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> >> Date: Mon, 14 Feb 2011 10:38:57 -0600
>>> >> Subject: Re: [Freeswitch-users] g729 packets skipped in passthrough
>>> mode
>>> >> I bet its 20ms vs 30
>>> >> set passthru_ptime_mismatch to true either in vars.xml or in your
>>> >> dialplan both legs through export on the a leg or set in the a leg and
>>> >> in the {} on b.
>>> >>
>>> >>
>>> >> On Mon, Feb 14, 2011 at 10:28 AM, Brian West <brian at freeswitch.org>
>>> wrote:
>>> >> > What exactly is the problem?  I see no issue here can you elaborate
>>> on
>>> >> > what you're seeing?
>>> >> >
>>> >> > /b
>>> >> >
>>> >> > On Feb 14, 2011, at 4:48 AM, Javier Gallart wrote:
>>> >> >
>>> >> >> I've tried to explicitly set ptime at 20ms at switch.conf -although
>>> >> >> it's not necessary afaik-. Has any one experienced this same issue?
>>> >> >>
>>> >> >> Thanks in advance..
>>> >> >>
>>> >> >> 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 steveayre at gmail.com  Thu Feb 17 15:43:33 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 17 Feb 2011 12:43:33 +0000
Subject: [Freeswitch-users] g729 packets skipped in passthrough mode
In-Reply-To: <AANLkTikW4+Xiy9RFY3jTwANe0uC0m=++a-iSRRX5A4fZ@mail.gmail.com>
References: <AANLkTikQQJYBLMkiXjqJvZPtkBu8C4-D2CUvaYcsM2KW@mail.gmail.com>
	<AANLkTikW4+Xiy9RFY3jTwANe0uC0m=++a-iSRRX5A4fZ@mail.gmail.com>
Message-ID: <AANLkTinY-=Lnq34YSt66=93oiDaD8XBRMPpb6Ps34JUj@mail.gmail.com>

They're not duplicating any packets, just setting the Marker flag on them
all.

According to the RTP RFC "It is intended to allow significant events such as
frame boundaries to be marked in the packet stream".

You generally see them on the first packet containing voice after a period
of silence. Why Huawei decided to set it on every packet is a mystery.

-Steve




On 17 February 2011 12:31, Vitalii Colosov <vetali100 at gmail.com> wrote:

> Just wondering - what is this brilliant idea which made Huawei engineers to
> setup their switch behave this way?
>
> Is there any REAL benefit? What did they try to achieve by duplicating each
> RTP packet...
>
> (Or maybe it is so strange way to assure the reliability of the traffic? At
> least one packet out of 2 will reach the destination) :)
>
>
>
>
> 2011/2/17 Javier Gallart <jgallartm at gmail.com>
>
>>  Hello
>>>
>>
>> my mistake... I had written data="RTP_BUG_IGNORE_MARK_BIT" instead of
>> value="RTP_BUG_IGNORE_MARK_BIT". It's working now. Keep this in mind if you
>> ever happen to put a Huawei SBC in from of a FS!
>>
>> Thanks
>>
>> Javier
>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Javier Gallart <jgallartm at gmail.com>
>>> To: freeswitch-users at lists.freeswitch.org
>>> Date: Wed, 16 Feb 2011 13:28:35 +0100
>>> Subject: Re: [Freeswitch-users] g729 packets skipped in passthrough mode
>>> Hello,
>>>
>>> I've looked deeper into this issue and I might have hit a known problem:
>>>
>>> RTP_BUG_IGNORE_MARK_BIT = (1 << 2)
>>>
>>> /*
>>>   A Huawei SBC has been discovered that sends the mark bit on every
>>> single RTP packet.
>>>   Since this causes the RTP stack to flush it's buffers, it horribly
>>> messes up the timing on the channel.
>>>
>>>   This flag will do nothing when an inbound packet contains the mark bit.
>>>
>>>  */
>>>
>>> The carrier is actually a Huawei SBC, and the problem I reported only
>>> happens when it sends the marker bit  = true in all the rtp packets. I've
>>> tried
>>> <param name="auto-rtp-bugs" data="RTP_BUG_IGNORE_MARK_BIT"/> in the
>>> profile but the problem is still there, is this the right way to override
>>> this nonsense from the Huawei?
>>>
>>> Thanks
>>>
>>> Javier
>>>
>>>>
>>>> ---------- Forwarded message ----------
>>>> From: Anthony Minessale <anthony.minessale at gmail.com>
>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>> Date: Tue, 15 Feb 2011 16:38:35 -0600
>>>> Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 56, Issue
>>>> 177
>>>> Well that's all I could think of:
>>>>
>>>> passthrough is passthrough FS never modifies anything.
>>>> you need to look harder at sip trace, pcaps or other diagnostics for
>>>> some misconfiguration.
>>>> many people use this daily.
>>>>
>>>>
>>>> On Tue, Feb 15, 2011 at 1:12 AM, Javier Gallart <jgallartm at gmail.com>
>>>> wrote:
>>>> >
>>>> > Anthony, thanks for the tip. I haven't seen any change though. I
>>>> should have
>>>> > mentioned that I'm focusing in the rtp stream coming form the callee
>>>> to the
>>>> > caller. The problem is that FS relays to the caller only half of the
>>>> packets
>>>> > received from the callee. The packets arrive at FS at a rate of 1
>>>> packet
>>>> > every 20 ms, and with a payload of 20 bytes each. Packets from FS to
>>>> the
>>>> > caller are sent every 40 ms with a payload of 20 bytes, thus skipping
>>>> half
>>>> > of the information.
>>>> >
>>>> > Thanks
>>>> >>
>>>> >> ---------- Forwarded message ----------
>>>> >> From: Anthony Minessale <anthony.minessale at gmail.com>
>>>> >> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>> >> Date: Mon, 14 Feb 2011 10:38:57 -0600
>>>> >> Subject: Re: [Freeswitch-users] g729 packets skipped in passthrough
>>>> mode
>>>> >> I bet its 20ms vs 30
>>>> >> set passthru_ptime_mismatch to true either in vars.xml or in your
>>>> >> dialplan both legs through export on the a leg or set in the a leg
>>>> and
>>>> >> in the {} on b.
>>>> >>
>>>> >>
>>>> >> On Mon, Feb 14, 2011 at 10:28 AM, Brian West <brian at freeswitch.org>
>>>> wrote:
>>>> >> > What exactly is the problem?  I see no issue here can you elaborate
>>>> on
>>>> >> > what you're seeing?
>>>> >> >
>>>> >> > /b
>>>> >> >
>>>> >> > On Feb 14, 2011, at 4:48 AM, Javier Gallart wrote:
>>>> >> >
>>>> >> >> I've tried to explicitly set ptime at 20ms at switch.conf
>>>> -although
>>>> >> >> it's not necessary afaik-. Has any one experienced this same
>>>> issue?
>>>> >> >>
>>>> >> >> Thanks in advance..
>>>> >> >>
>>>> >> >> 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
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From helmut.kuper at ewetel.de  Thu Feb 17 16:33:32 2011
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Thu, 17 Feb 2011 14:33:32 +0100
Subject: [Freeswitch-users] SIP SUBSCRIBE for as-feature-event events
Message-ID: <4D5D23AC.7090804@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

Snom is supporting subscriptions to as-feature-event Events.
Unfortunately FS replys with "Bad Event" to this.

Are there plans to support such subscriptions? Maybe the headers and
body can be passed to an application? Or the whole thing could be
forwared to an application server to work on it.

The SIP message looks like this:


Session Initiation Protocol
    Request-Line: SUBSCRIBE sip:2850 at a.a.a.a SIP/2.0
    Message Header
        Via: SIP/2.0/UDP b.b.b.b:5060;branch=z9hG4bK-b3v8fu4shagb;rport
        From: <sip:2850 at a.a.a.a>;tag=dq168o7qj9
        To: "Helmut Kuper" <sip:2850 at a.a.a.a>
        Call-ID: 3c26705696f0-6ntjya2y7b3s
        CSeq: 2 SUBSCRIBE
        Max-Forwards: 70
        Contact: <sip:2850 at b.b.b.b:5060>;reg-id=1
        Event: as-feature-event
        User-Agent: snom370/8.4.27
        Expires: 3600
        Content-Type: application/x-as-feature-event+xml
        Content-Length: 185
    Message Body
        <?xml version="1.0" encoding="ISO-8859-1"?><SetDoNotDisturb
xmlns="http://www.ecma-international.org/standards/ecma-323/csta/ed3"><doNotDisturbOn>true</doNotDisturbOn></SetDoNotDisturb>

Best regards
Helmut
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.10 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/

iEYEARECAAYFAk1dI6wACgkQ4tZeNddg3dx0LQCdGPIHZWb5tUEaP4SOMl7cWJrE
f38AoK1NEd9VeOUJ6Ut+XAyZgS6LdcZz
=JLAn
-----END PGP SIGNATURE-----


From rajesh.npnr at yahoo.com  Thu Feb 17 17:51:00 2011
From: rajesh.npnr at yahoo.com (rex.alex)
Date: Thu, 17 Feb 2011 06:51:00 -0800 (PST)
Subject: [Freeswitch-users] The call dropped prematurely error on txfax
In-Reply-To: <AANLkTiku+q9B_FeSErvMuXSog6DQ4mMtDmxn_TjtG9Zw@mail.gmail.com>
References: <1297870085994-6032194.post@n2.nabble.com>
	<643BF67696204EE089E833DEE6BA2116@e1705>
	<1297873355400-6032435.post@n2.nabble.com>
	<AANLkTingYEfo=4oaANLS8ZzySqLSaUOOm7ME=HpxxQ-C@mail.gmail.com>
	<1297885773534-6033268.post@n2.nabble.com>
	<AANLkTiku+q9B_FeSErvMuXSog6DQ4mMtDmxn_TjtG9Zw@mail.gmail.com>
Message-ID: <1297954260038-6036197.post@n2.nabble.com>


I have upgraded to latest git (FreeSWITCH Version 1.0.head (git-65d5932
2011-02-17 10-09-26 +0100)) but I am getting the same error when executing
the command,

originate
{fax_enable_t38=true,fax_enable_t38_request=true}sofia/gateway/VivaCom/1XXXXXXXXXX
&txfax(/var/www/html/faxtest/docs/15Feb2011103600.pdf.tiff)

Thanks,
Rex
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/The-call-dropped-prematurely-error-on-txfax-tp6032194p6036197.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From rmbertjones at comcast.net  Thu Feb 17 17:36:51 2011
From: rmbertjones at comcast.net (Bert Jones)
Date: Thu, 17 Feb 2011 09:36:51 -0500
Subject: [Freeswitch-users] Possible to communicate with FS
	using	event_socket without ESL?
In-Reply-To: <008901cbcc01$517903b0$f46b0b10$@net>
References: <003101cbcb2d$fa0e9e60$ee2bdb20$@net>	<AANLkTi=5zyKN1Ef+nJspyMrOuw=VE+fV-RcCQ5S148L=@mail.gmail.com>
	<008901cbcc01$517903b0$f46b0b10$@net>
Message-ID: <011d01cbceb0$1f7246e0$5e56d4a0$@net>

Hi Again Steve,

 

This is resolved now.  It appears that I was not doing the initiation and
authentication with the proper number of linefeeds and with the proper
sequence of writes AND reads.  For those interested, here is a working
function written in vb.net to do the communications on the localhost
machine.  The assumption is made that the password has been changed from
ClueCon to myPassword in modules.config.xml and that mod_event_socket is
loaded in modules.conf.xml  using the entry: <load
module="mod_event_socket"/>.

 

Thanks to all for the help. 

Bert

 

 

  Function SendCommands_socket(ByVal id, ByVal Cmnd)

            'Funciton to initiate conneciton with FreeSWITCH event_socket
via a tcpsocket, authenticate and send a command.

            'Note that the writes AND reads are all mandatory, as are the
double linefeeds.

 

            'Dims

            Dim returndata As String = "" 'Holds return data

            Dim sendBytes As [Byte]() ' byte array used to hold outbound
data after conversion from ascii to byte array

            Dim tcpClient As New System.Net.Sockets.TcpClient() ' the tcp
socket client used to communicate with the host

            Dim rcvBytes(tcpClient.ReceiveBufferSize) As Byte 'byte array
used to receive bytes returned from host

 

            Try

                'Initiate conneciton with event_socket via the
socket--------------------

                tcpClient.Connect("localhost", 8021)

                Dim stream As NetworkStream = tcpClient.GetStream() 'open
the socket

                stream.Read(rcvBytes, 0, CInt(tcpClient.ReceiveBufferSize))

                returndata = Encoding.ASCII.GetString(rcvBytes)

                MsgBox(returndata) 'should say "Content-Type: auth/request"

 

                If stream.CanWrite And stream.CanRead Then

                    'Do
authentication.-------------------------------------------------

                    ' convert text to byte array

                    sendBytes = Encoding.ASCII.GetBytes("auth myPassword" &
vbCrLf & vbCrLf) '(The 2 crlf are mandatory )

                    ' write data to stream

                    stream.Write(sendBytes, 0, sendBytes.Length)

                    ' read the NetworkStream into a seperate byte buffer. 

                    stream.Read(rcvBytes, 0,
CInt(tcpClient.ReceiveBufferSize))

                    ' convert binary bytes received into ascii

                    returndata = Encoding.ASCII.GetString(rcvBytes)

                    MsgBox(returndata) 'should say "Content-Type:
command/reply Reply-Text: +Ok accepted"

 

                    'Send the
command.-------------------------------------------------

                    sendBytes = Encoding.ASCII.GetBytes(Cmnd & vbCrLf &
vbCrLf)

                    stream.Write(sendBytes, 0, sendBytes.Length)

                    stream.Read(rcvBytes, 0,
CInt(tcpClient.ReceiveBufferSize))

                    returndata = Encoding.ASCII.GetString(rcvBytes)

                    MsgBox(returndata)

                Else

                    If Not stream.CanRead Then

                        Console.WriteLine("cannot not write data to this
stream")

                        tcpClient.Close()

                    Else

                        If Not stream.CanWrite Then

                            Console.WriteLine("cannot read data from this
stream")

                            tcpClient.Close()

                        End If

                    End If

                End If

 

                ' Disconnect

                tcpClient.Close()

 

                Return returndata

 

            Catch oEX As SocketException

                ' Disconnect

                tcpClient.Close()

                Return oEX.ToString

                Exit Function

            End Try

 

        End Function

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bert
Jones
Sent: Sunday, February 13, 2011 11:41 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] Possible to communicate with FS using
event_socket without ESL?

 

Thanks for the clarification.  I too suspect something in my communications.
Just wanted to make sure I was not missing something. Will move my
communication app to a separate machine from the FS server and run
Wireshark.  Thanks for recommendation.

Bert

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven
Ayre
Sent: Sunday, February 13, 2011 7:29 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Possible to communicate with FS using
event_socket without ESL?

 

Without the use of the ESL *protocol* - no. Without the use of the supplied
ESL *libraries* - correct.

You're along the right lines on implementing your own protocol. If you can
connect and authenticate via telnet from the same computer you're trying
your app from then there should be no problem with FS.

The fact you say it takes 10s for FS to respond indicates you're not sending
the auth line. You should get either a success or failure message straight
away in reply to that. So it sounds like the auth isn't getting sent at all.

Try using Wireshark to view what's going over the network to check it's what
you expect.

-Steve






On 13 February 2011 03:27, Bert Jones <rmbertjones at comcast.net> wrote:

Hello,

 

Am I correct in assuming that an app can talk directly to FS using
event_socket without the use of the ESL?  I am attempting to do this, but
having trouble authenticating. I wanted to make sure I was not overlooking
something basis in my approach.

 

I have successfully loaded FS on a windows XP server and can register phones
and make calls.  Further I can talk to the server, authenticate and make
calls using telnet, but when attempting to do the same via an app written in
.net and using a socket, I am unable to authenticate.  

 

Running my app using a tcpSocket, I receive the "Content-Type: auth/request"
upon connection of the socket, but when sending "auth password\n\n" it
appears that the connection times out after about 10 seconds and I receive:
"Content-Type: text/disconnect-notice" & vbLf & "Content-Length: 67".
mod_event_socket is enabled in modules.conf.xml .

 

Is there something obvious that I am missing conceptually, or should this
work?

 

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 victor.chukalovskiy at utoronto.ca  Thu Feb 17 18:07:10 2011
From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy)
Date: Thu, 17 Feb 2011 10:07:10 -0500
Subject: [Freeswitch-users] Broad-voice configuration trouble.
In-Reply-To: <1297954260038-6036197.post@n2.nabble.com>
References: <1297870085994-6032194.post@n2.nabble.com>	<643BF67696204EE089E833DEE6BA2116@e1705>	<1297873355400-6032435.post@n2.nabble.com>	<AANLkTingYEfo=4oaANLS8ZzySqLSaUOOm7ME=HpxxQ-C@mail.gmail.com>	<1297885773534-6033268.post@n2.nabble.com>	<AANLkTiku+q9B_FeSErvMuXSog6DQ4mMtDmxn_TjtG9Zw@mail.gmail.com>
	<1297954260038-6036197.post@n2.nabble.com>
Message-ID: <4D5D399E.40805@utoronto.ca>

Hello,

I can't get BroadVoice running.
Currently I follow this page:
http://wiki.freeswitch.org/wiki/Provider_Configuration:_Broadvoice
and it registers Ok.

However, outgoing calls fail with Broadvoice sending me "403 Forbidden".
I do set absolute_codec_string to PCMU and I see that PCMU is indeed the 
only codec in my SDP.

Any help is appreciated.

Thank you,
Victor


From davidwaf at gmail.com  Thu Feb 17 18:00:25 2011
From: davidwaf at gmail.com (David Wafula)
Date: Thu, 17 Feb 2011 17:00:25 +0200
Subject: [Freeswitch-users] mod_xml_curl response debug
Message-ID: <AANLkTimt1g5ORXkD_8cY3urQKUS6jq=ksBfkJDw5WU=2@mail.gmail.com>

Am using mod_xml_curl to fetch dialplans from a web app. However, am running
into:

mod_xml_curl.c:301 Error Parsing Result! I have checked my xml responses,
they look fine, but obviously something is wrong somewhere. So how do i get
to see the response freeswitch gets in the logs, to help me narrow down the
problem?


-- 
David Wafula
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From avi at avimarcus.net  Thu Feb 17 18:17:20 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 17 Feb 2011 17:17:20 +0200
Subject: [Freeswitch-users] mod_xml_curl response debug
In-Reply-To: <AANLkTimt1g5ORXkD_8cY3urQKUS6jq=ksBfkJDw5WU=2@mail.gmail.com>
References: <AANLkTimt1g5ORXkD_8cY3urQKUS6jq=ksBfkJDw5WU=2@mail.gmail.com>
Message-ID: <AANLkTinEu1v8DZUoB+m0NMckWqx0NM8JK2N6goUsQhtj@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Mod_xml_curl#Debugging
<http://wiki.freeswitch.org/wiki/Mod_xml_curl#Debugging>Run xml_curl
debug_on
Then, FS saves a file with debug info / the returned XML. I think the CLI
says where the log is.
-Avi


On Thu, Feb 17, 2011 at 5:00 PM, David Wafula <davidwaf at gmail.com> wrote:

> Am using mod_xml_curl to fetch dialplans from a web app. However, am
> running into:
>
> mod_xml_curl.c:301 Error Parsing Result! I have checked my xml responses,
> they look fine, but obviously something is wrong somewhere. So how do i get
> to see the response freeswitch gets in the logs, to help me narrow down the
> problem?
>
>
> --
> 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
>
>
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From jeff at jefflenk.com  Thu Feb 17 18:20:29 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 17 Feb 2011 07:20:29 -0800 (PST)
Subject: [Freeswitch-users] Broad-voice configuration trouble.
In-Reply-To: <4D5D399E.40805@utoronto.ca>
References: <1297870085994-6032194.post@n2.nabble.com>
	<643BF67696204EE089E833DEE6BA2116@e1705>
	<1297873355400-6032435.post@n2.nabble.com>
	<AANLkTingYEfo=4oaANLS8ZzySqLSaUOOm7ME=HpxxQ-C@mail.gmail.com>
	<1297885773534-6033268.post@n2.nabble.com>
	<AANLkTiku+q9B_FeSErvMuXSog6DQ4mMtDmxn_TjtG9Zw@mail.gmail.com>
	<1297954260038-6036197.post@n2.nabble.com>
	<4D5D399E.40805@utoronto.ca>
Message-ID: <1297956029731-6036326.post@n2.nabble.com>


Please dont hijack threads - do not reply to an existing thread and change
the subject line as this makes it very difficult for people watching these
threads to make any sense of it. Thanks.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/The-call-dropped-prematurely-error-on-txfax-tp6032194p6036326.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Thu Feb 17 18:20:43 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 17 Feb 2011 07:20:43 -0800 (PST)
Subject: [Freeswitch-users] Broad-voice configuration trouble.
In-Reply-To: <4D5D399E.40805@utoronto.ca>
References: <1297870085994-6032194.post@n2.nabble.com>
	<643BF67696204EE089E833DEE6BA2116@e1705>
	<1297873355400-6032435.post@n2.nabble.com>
	<AANLkTingYEfo=4oaANLS8ZzySqLSaUOOm7ME=HpxxQ-C@mail.gmail.com>
	<1297885773534-6033268.post@n2.nabble.com>
	<AANLkTiku+q9B_FeSErvMuXSog6DQ4mMtDmxn_TjtG9Zw@mail.gmail.com>
	<1297954260038-6036197.post@n2.nabble.com>
	<4D5D399E.40805@utoronto.ca>
Message-ID: <1297956043436-6036328.post@n2.nabble.com>



Victor Chukalovskiy-3 wrote:
> 
> Hello,
> 
> I can't get BroadVoice running.
> Currently I follow this page:
> http://wiki.freeswitch.org/wiki/Provider_Configuration:_Broadvoice
> and it registers Ok.
> 
> However, outgoing calls fail with Broadvoice sending me "403 Forbidden".
> I do set absolute_codec_string to PCMU and I see that PCMU is indeed the 
> only codec in my SDP.
> 
> Any help is appreciated.
> 
> Thank you,
> Victor
Your are off topic. Please kindly post on a new discussion thread and be
sure to include the log output from fs_cli.

-----
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/The-call-dropped-prematurely-error-on-txfax-tp6032194p6036328.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From davidwaf at gmail.com  Thu Feb 17 18:19:49 2011
From: davidwaf at gmail.com (David Wafula)
Date: Thu, 17 Feb 2011 17:19:49 +0200
Subject: [Freeswitch-users] mod_xml_curl response debug
In-Reply-To: <AANLkTimt1g5ORXkD_8cY3urQKUS6jq=ksBfkJDw5WU=2@mail.gmail.com>
References: <AANLkTimt1g5ORXkD_8cY3urQKUS6jq=ksBfkJDw5WU=2@mail.gmail.com>
Message-ID: <AANLkTimci-N97n+06ByffSRqLVWJsmhi1GaN7rShBGWv@mail.gmail.com>

Never mind,
found xml_curl debug_on in the docs.

regards.

On Thu, Feb 17, 2011 at 5:00 PM, David Wafula <davidwaf at gmail.com> wrote:

> Am using mod_xml_curl to fetch dialplans from a web app. However, am
> running into:
>
> mod_xml_curl.c:301 Error Parsing Result! I have checked my xml responses,
> they look fine, but obviously something is wrong somewhere. So how do i get
> to see the response freeswitch gets in the logs, to help me narrow down the
> problem?
>
>
> --
> David Wafula
>



-- 
David Wafula
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From infos at madovsky.org  Thu Feb 17 18:29:42 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 17 Feb 2011 10:29:42 -0500
Subject: [Freeswitch-users] The call dropped prematurely error on txfax
References: <1297870085994-6032194.post@n2.nabble.com><643BF67696204EE089E833DEE6BA2116@e1705><1297873355400-6032435.post@n2.nabble.com><AANLkTingYEfo=4oaANLS8ZzySqLSaUOOm7ME=HpxxQ-C@mail.gmail.com><1297885773534-6033268.post@n2.nabble.com><AANLkTiku+q9B_FeSErvMuXSog6DQ4mMtDmxn_TjtG9Zw@mail.gmail.com>
	<1297954260038-6036197.post@n2.nabble.com>
Message-ID: <140D740BF92640F7A514BD35BD4BD1AA@e1705>

did you try another fax number ?

----- Original Message ----- 
From: "rex.alex" <rajesh.npnr at yahoo.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, February 17, 2011 9:51 AM
Subject: Re: [Freeswitch-users] The call dropped prematurely error on txfax


>
> I have upgraded to latest git (FreeSWITCH Version 1.0.head (git-65d5932
> 2011-02-17 10-09-26 +0100)) but I am getting the same error when executing
> the command,
>
> originate
> {fax_enable_t38=true,fax_enable_t38_request=true}sofia/gateway/VivaCom/1XXXXXXXXXX
> &txfax(/var/www/html/faxtest/docs/15Feb2011103600.pdf.tiff)
>
> Thanks,
> Rex
> -- 
> View this message in context: 
> http://freeswitch-users.2379917.n2.nabble.com/The-call-dropped-prematurely-error-on-txfax-tp6032194p6036197.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 victor.chukalovskiy at utoronto.ca  Thu Feb 17 18:30:24 2011
From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy)
Date: Thu, 17 Feb 2011 10:30:24 -0500
Subject: [Freeswitch-users] Broad-voice configuration trouble.
Message-ID: <4D5D3F10.2060905@utoronto.ca>

Hello,

I can't get BroadVoice running.
Currently I follow this page:
http://wiki.freeswitch.org/wiki/Provider_Configuration:_Broadvoice
and it registers Ok.

However, outgoing calls fail with Broadvoice sending me "403 Forbidden".
I do set absolute_codec_string to PCMU and I see that PCMU is indeed the
only codec in my SDP.

Any help is appreciated.
Actually, can anyone share an INVITE packet that works with Broad-voice?
I think I'll be able to solve it then.

Thank you,
Victor

> Sorry for hijacking previous thread!



From rajesh.npnr at yahoo.com  Thu Feb 17 19:07:37 2011
From: rajesh.npnr at yahoo.com (rex.alex)
Date: Thu, 17 Feb 2011 08:07:37 -0800 (PST)
Subject: [Freeswitch-users] The call dropped prematurely error on txfax
In-Reply-To: <140D740BF92640F7A514BD35BD4BD1AA@e1705>
References: <1297870085994-6032194.post@n2.nabble.com>
	<643BF67696204EE089E833DEE6BA2116@e1705>
	<1297873355400-6032435.post@n2.nabble.com>
	<AANLkTingYEfo=4oaANLS8ZzySqLSaUOOm7ME=HpxxQ-C@mail.gmail.com>
	<1297885773534-6033268.post@n2.nabble.com>
	<AANLkTiku+q9B_FeSErvMuXSog6DQ4mMtDmxn_TjtG9Zw@mail.gmail.com>
	<1297954260038-6036197.post@n2.nabble.com>
	<140D740BF92640F7A514BD35BD4BD1AA@e1705>
Message-ID: <1297958857213-6036503.post@n2.nabble.com>


Hi,

Yes I tried another number without any channel variables and it's sending
the same document successfully. But if I include
{fax_enable_t38=true,fax_enable_t38_request=true} in originate command, even
the 2nd number fails always with "The call dropped prematurely" error.

But the 1st tried number fails in any combination of channel variables with
error. What could be the reason?

Thanks,
Rex
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/The-call-dropped-prematurely-error-on-txfax-tp6032194p6036503.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brian at freeswitch.org  Thu Feb 17 19:12:17 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 17 Feb 2011 10:12:17 -0600
Subject: [Freeswitch-users] The call dropped prematurely error on txfax
In-Reply-To: <1297958857213-6036503.post@n2.nabble.com>
References: <1297870085994-6032194.post@n2.nabble.com>
	<643BF67696204EE089E833DEE6BA2116@e1705>
	<1297873355400-6032435.post@n2.nabble.com>
	<AANLkTingYEfo=4oaANLS8ZzySqLSaUOOm7ME=HpxxQ-C@mail.gmail.com>
	<1297885773534-6033268.post@n2.nabble.com>
	<AANLkTiku+q9B_FeSErvMuXSog6DQ4mMtDmxn_TjtG9Zw@mail.gmail.com>
	<1297954260038-6036197.post@n2.nabble.com>
	<140D740BF92640F7A514BD35BD4BD1AA@e1705>
	<1297958857213-6036503.post@n2.nabble.com>
Message-ID: <FA5EFAF9-332F-41F2-BC8E-5A35D958C239@freeswitch.org>

without logs of what is going on I can't tell you .

/b

On Feb 17, 2011, at 10:07 AM, rex.alex wrote:

> 
> Hi,
> 
> Yes I tried another number without any channel variables and it's sending
> the same document successfully. But if I include
> {fax_enable_t38=true,fax_enable_t38_request=true} in originate command, even
> the 2nd number fails always with "The call dropped prematurely" error.
> 
> But the 1st tried number fails in any combination of channel variables with
> error. What could be the reason?
> 
> Thanks,
> Rex



From infos at madovsky.org  Thu Feb 17 19:17:59 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 17 Feb 2011 11:17:59 -0500
Subject: [Freeswitch-users] The call dropped prematurely error on txfax
References: <1297870085994-6032194.post@n2.nabble.com><643BF67696204EE089E833DEE6BA2116@e1705><1297873355400-6032435.post@n2.nabble.com><AANLkTingYEfo=4oaANLS8ZzySqLSaUOOm7ME=HpxxQ-C@mail.gmail.com><1297885773534-6033268.post@n2.nabble.com><AANLkTiku+q9B_FeSErvMuXSog6DQ4mMtDmxn_TjtG9Zw@mail.gmail.com><1297954260038-6036197.post@n2.nabble.com><140D740BF92640F7A514BD35BD4BD1AA@e1705>
	<1297958857213-6036503.post@n2.nabble.com>
Message-ID: <C0AB523AE4774FA5837E88F48243F970@e1705>

another stupid question, did you try another trunk also ?

----- Original Message ----- 
From: "rex.alex" <rajesh.npnr at yahoo.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, February 17, 2011 11:07 AM
Subject: Re: [Freeswitch-users] The call dropped prematurely error on txfax


>
> Hi,
>
> Yes I tried another number without any channel variables and it's sending
> the same document successfully. But if I include
> {fax_enable_t38=true,fax_enable_t38_request=true} in originate command, 
> even
> the 2nd number fails always with "The call dropped prematurely" error.
>
> But the 1st tried number fails in any combination of channel variables 
> with
> error. What could be the reason?
>
> Thanks,
> Rex
> -- 
> View this message in context: 
> http://freeswitch-users.2379917.n2.nabble.com/The-call-dropped-prematurely-error-on-txfax-tp6032194p6036503.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 k-b-24 at live.com  Thu Feb 17 19:18:40 2011
From: k-b-24 at live.com (Jason b.a)
Date: Thu, 17 Feb 2011 16:18:40 +0000
Subject: [Freeswitch-users] IVR application
Message-ID: <col111-w4579CBD2496FFFEAEB7B10ACD50@phx.gbl>


yeh i am using xlite , so my design will be 
xlite--IVR application--freeswitch
but what about the signaling or control protocol used between app and Fs.
  		 	   		  
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From msc at freeswitch.org  Thu Feb 17 19:21:53 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 17 Feb 2011 08:21:53 -0800
Subject: [Freeswitch-users] Broad-voice configuration trouble.
In-Reply-To: <4D5D3F10.2060905@utoronto.ca>
References: <4D5D3F10.2060905@utoronto.ca>
Message-ID: <AANLkTimLD2gRoYicAfweBEVt7gf=tPaLE+w-2QRRb9n5@mail.gmail.com>

403 usually means bad password or they don't like the IP addr you're calling
from. Call them up and confirm all your settings and ask them to give you
some further details regarding the 403.
-MC

On Thu, Feb 17, 2011 at 7:30 AM, Victor Chukalovskiy <
victor.chukalovskiy at utoronto.ca> wrote:

> Hello,
>
> I can't get BroadVoice running.
> Currently I follow this page:
> http://wiki.freeswitch.org/wiki/Provider_Configuration:_Broadvoice
> and it registers Ok.
>
> However, outgoing calls fail with Broadvoice sending me "403 Forbidden".
> I do set absolute_codec_string to PCMU and I see that PCMU is indeed the
> only codec in my SDP.
>
> Any help is appreciated.
> Actually, can anyone share an INVITE packet that works with Broad-voice?
> I think I'll be able to solve it then.
>
> Thank you,
> Victor
>
> > Sorry for hijacking previous thread!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From victor.chukalovskiy at utoronto.ca  Thu Feb 17 19:32:28 2011
From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy)
Date: Thu, 17 Feb 2011 11:32:28 -0500
Subject: [Freeswitch-users] Broad-voice configuration trouble.
In-Reply-To: <AANLkTimLD2gRoYicAfweBEVt7gf=tPaLE+w-2QRRb9n5@mail.gmail.com>
References: <4D5D3F10.2060905@utoronto.ca>
	<AANLkTimLD2gRoYicAfweBEVt7gf=tPaLE+w-2QRRb9n5@mail.gmail.com>
Message-ID: <4D5D4D9C.2010308@utoronto.ca>

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From msc at freeswitch.org  Thu Feb 17 19:38:52 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 17 Feb 2011 08:38:52 -0800
Subject: [Freeswitch-users] Broad-voice configuration trouble.
In-Reply-To: <4D5D4D9C.2010308@utoronto.ca>
References: <4D5D3F10.2060905@utoronto.ca>
	<AANLkTimLD2gRoYicAfweBEVt7gf=tPaLE+w-2QRRb9n5@mail.gmail.com>
	<4D5D4D9C.2010308@utoronto.ca>
Message-ID: <AANLkTinrBPPbRp58eOpz5FEtXK0VwYmYTHtWE__X9+Nx@mail.gmail.com>

What about the IP address issue? Are they expecting you to be coming from a
specific IP address? What about your account? Is it locked? Are you able to
receive inbound calls?

There are a lot of questions left to be asked and answered. Also, get a SIP
trace and put it on pastebin. Hopefully someone else who has Broadvoice can
offer some insight. If not then consider a provider who actually supports
FreeSWITCH and isn't so abusive to a paying customer.

-MC

On Thu, Feb 17, 2011 at 8:32 AM, Victor Chukalovskiy <
victor.chukalovskiy at utoronto.ca> wrote:

>  Hi Michael,
>
> I assume password is Ok since it registers fine. It shouldn't register with
> no password.
> They say they don't officially support PBX's and that they can't check
> logs/traces to see why "403" is sent back.
>
> -Victor
>
>
>
> On 17/02/11 11:21 AM, Michael Collins wrote:
>
> 403 usually means bad password or they don't like the IP addr you're
> calling from. Call them up and confirm all your settings and ask them to
> give you some further details regarding the 403.
> -MC
>
>  On Thu, Feb 17, 2011 at 7:30 AM, Victor Chukalovskiy <
> victor.chukalovskiy at utoronto.ca> wrote:
>
>> Hello,
>>
>> I can't get BroadVoice running.
>> Currently I follow this page:
>> http://wiki.freeswitch.org/wiki/Provider_Configuration:_Broadvoice
>> and it registers Ok.
>>
>> However, outgoing calls fail with Broadvoice sending me "403 Forbidden".
>> I do set absolute_codec_string to PCMU and I see that PCMU is indeed the
>> only codec in my SDP.
>>
>> Any help is appreciated.
>> Actually, can anyone share an INVITE packet that works with Broad-voice?
>> I think I'll be able to solve it then.
>>
>> Thank you,
>> Victor
>>
>> > Sorry for hijacking previous thread!
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> 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 msc at freeswitch.org  Thu Feb 17 19:41:07 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 17 Feb 2011 08:41:07 -0800
Subject: [Freeswitch-users] txfax and multiple files
In-Reply-To: <8D3CBF9281FF476CB88E77DC2FCD7ADD@e1705>
References: <8D3CBF9281FF476CB88E77DC2FCD7ADD@e1705>
Message-ID: <AANLkTimh9d3Jb4GnXQzyR_sVQFCbthqT1vEhHzia9P3j@mail.gmail.com>

Did you figure this out? I didn't know that you could queue up multiple
files for transmission, but if you can then I'd like to know so that I can
get it properly documented...

-MC

On Wed, Feb 16, 2011 at 12:19 AM, Madovsky <infos at madovsky.org> wrote:

>  What is the right syntax to send multiple files with
> mod_spandsp txfax ?
>
> 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  Thu Feb 17 19:41:42 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 17 Feb 2011 16:41:42 +0000
Subject: [Freeswitch-users] Broad-voice configuration trouble.
In-Reply-To: <AANLkTimLD2gRoYicAfweBEVt7gf=tPaLE+w-2QRRb9n5@mail.gmail.com>
References: <4D5D3F10.2060905@utoronto.ca>
	<AANLkTimLD2gRoYicAfweBEVt7gf=tPaLE+w-2QRRb9n5@mail.gmail.com>
Message-ID: <AANLkTi=Bb6Jw0gkxwOpCauA-DR2iPCadvLBH7_7GxuXU@mail.gmail.com>

403 can also happen with them if the INVITE's too big. They're not RFC
compliant.

Victor, can you post the siptrace?

-Steve



On 17 February 2011 16:21, Michael Collins <msc at freeswitch.org> wrote:

> 403 usually means bad password or they don't like the IP addr you're
> calling from. Call them up and confirm all your settings and ask them to
> give you some further details regarding the 403.
> -MC
>
>
> On Thu, Feb 17, 2011 at 7:30 AM, Victor Chukalovskiy <
> victor.chukalovskiy at utoronto.ca> wrote:
>
>> Hello,
>>
>> I can't get BroadVoice running.
>> Currently I follow this page:
>> http://wiki.freeswitch.org/wiki/Provider_Configuration:_Broadvoice
>> and it registers Ok.
>>
>> However, outgoing calls fail with Broadvoice sending me "403 Forbidden".
>> I do set absolute_codec_string to PCMU and I see that PCMU is indeed the
>> only codec in my SDP.
>>
>> Any help is appreciated.
>> Actually, can anyone share an INVITE packet that works with Broad-voice?
>> I think I'll be able to solve it then.
>>
>> Thank you,
>> Victor
>>
>> > Sorry for hijacking previous thread!
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 17 19:54:49 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 17 Feb 2011 11:54:49 -0500
Subject: [Freeswitch-users] txfax and multiple files
References: <8D3CBF9281FF476CB88E77DC2FCD7ADD@e1705>
	<AANLkTimh9d3Jb4GnXQzyR_sVQFCbthqT1vEhHzia9P3j@mail.gmail.com>
Message-ID: <3D5EA10B77DF4A40BA5B3A6854EB98DD@e1705>

not yet, will do test later today.
If I remember it's possible by separate every path files by a space or comma
Will confirm soon
in case of it's not possible, can I run tfax app several times while the channel is up ?

Thanks
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Thursday, February 17, 2011 11:41 AM
  Subject: Re: [Freeswitch-users] txfax and multiple files


  Did you figure this out? I didn't know that you could queue up multiple files for transmission, but if you can then I'd like to know so that I can get it properly documented...


  -MC


  On Wed, Feb 16, 2011 at 12:19 AM, Madovsky <infos at madovsky.org> wrote:

    What is the right syntax to send multiple files with
    mod_spandsp txfax ?

    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 rajesh.npnr at yahoo.com  Thu Feb 17 19:57:06 2011
From: rajesh.npnr at yahoo.com (rex.alex)
Date: Thu, 17 Feb 2011 08:57:06 -0800 (PST)
Subject: [Freeswitch-users] The call dropped prematurely error on txfax
In-Reply-To: <C0AB523AE4774FA5837E88F48243F970@e1705>
References: <1297870085994-6032194.post@n2.nabble.com>
	<643BF67696204EE089E833DEE6BA2116@e1705>
	<1297873355400-6032435.post@n2.nabble.com>
	<AANLkTingYEfo=4oaANLS8ZzySqLSaUOOm7ME=HpxxQ-C@mail.gmail.com>
	<1297885773534-6033268.post@n2.nabble.com>
	<AANLkTiku+q9B_FeSErvMuXSog6DQ4mMtDmxn_TjtG9Zw@mail.gmail.com>
	<1297954260038-6036197.post@n2.nabble.com>
	<140D740BF92640F7A514BD35BD4BD1AA@e1705>
	<1297958857213-6036503.post@n2.nabble.com>
	<C0AB523AE4774FA5837E88F48243F970@e1705>
Message-ID: <1297961826262-6036700.post@n2.nabble.com>


Yes I have tried through another trunk also but the result is same. I have
pastebin freeswitch log in http://pastebin.freeswitch.org/15407.

If you require any other log, please let me know and how to take the same
also so that I can pastebin the same.

Thanks,
Rex
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/The-call-dropped-prematurely-error-on-txfax-tp6032194p6036700.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Thu Feb 17 20:05:43 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 17 Feb 2011 11:05:43 -0600
Subject: [Freeswitch-users] The call dropped prematurely error on txfax
In-Reply-To: <1297961826262-6036700.post@n2.nabble.com>
References: <1297870085994-6032194.post@n2.nabble.com>
	<643BF67696204EE089E833DEE6BA2116@e1705>
	<1297873355400-6032435.post@n2.nabble.com>
	<AANLkTingYEfo=4oaANLS8ZzySqLSaUOOm7ME=HpxxQ-C@mail.gmail.com>
	<1297885773534-6033268.post@n2.nabble.com>
	<AANLkTiku+q9B_FeSErvMuXSog6DQ4mMtDmxn_TjtG9Zw@mail.gmail.com>
	<1297954260038-6036197.post@n2.nabble.com>
	<140D740BF92640F7A514BD35BD4BD1AA@e1705>
	<1297958857213-6036503.post@n2.nabble.com>
	<C0AB523AE4774FA5837E88F48243F970@e1705>
	<1297961826262-6036700.post@n2.nabble.com>
Message-ID: <AANLkTikmEvZB5tF6qiT2Jf4Ca6VzBB8bqQ_k=O59=eO=@mail.gmail.com>

you also need to execute
sofia global siptrace on
and do the trace.


On Thu, Feb 17, 2011 at 10:57 AM, rex.alex <rajesh.npnr at yahoo.com> wrote:
>
> Yes I have tried through another trunk also but the result is same. I have
> pastebin freeswitch log in http://pastebin.freeswitch.org/15407.
>
> If you require any other log, please let me know and how to take the same
> also so that I can pastebin the same.
>
> Thanks,
> Rex
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/The-call-dropped-prematurely-error-on-txfax-tp6032194p6036700.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
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From curriegrad2004 at gmail.com  Thu Feb 17 20:06:33 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 17 Feb 2011 09:06:33 -0800
Subject: [Freeswitch-users] Parallel compilation on a Linux machine
In-Reply-To: <1297944336320-6035621.post@n2.nabble.com>
References: <1297913369896-6034541.post@n2.nabble.com>
	<AANLkTinscs9H1tGtJB4tvLDgK--cxK+CZU8H0ZRgFQOk@mail.gmail.com>
	<1297944336320-6035621.post@n2.nabble.com>
Message-ID: <AANLkTikaL=0GbPP-Vy5Y2CqEoJuriaqWkjrEQVyC5gDQ@mail.gmail.com>

Similar configuration here, but all performed under a Fedora 13 x86 VM
running on VirtualBox with the host running on Windows 7. Guest has
2GB of RAM allocated to it and no GUI installed at all.

On Thu, Feb 17, 2011 at 4:05 AM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
>
> curriegrad2004 wrote:
>> How many modules are you compiling at once anyways?
> Almost all modules.
>
>
> FS git never seem to take more than approx. 15-20 minutes when it's being
> compiled on my AMD Athlon II Dual Core machine. May I ask how much RAM do
> you have on your box?
> 2GB RAM with no X11 windows running.
>
> -----
> 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/Parallel-compilation-on-a-Linux-machine-tp6034541p6035621.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 Nabble at slickdeals.endjunk.com  Thu Feb 17 20:45:26 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 17 Feb 2011 09:45:26 -0800 (PST)
Subject: [Freeswitch-users] Parallel compilation on a Linux machine
In-Reply-To: <AANLkTikaL=0GbPP-Vy5Y2CqEoJuriaqWkjrEQVyC5gDQ@mail.gmail.com>
References: <1297913369896-6034541.post@n2.nabble.com>
	<AANLkTinscs9H1tGtJB4tvLDgK--cxK+CZU8H0ZRgFQOk@mail.gmail.com>
	<1297944336320-6035621.post@n2.nabble.com>
	<AANLkTikaL=0GbPP-Vy5Y2CqEoJuriaqWkjrEQVyC5gDQ@mail.gmail.com>
Message-ID: <1297964726895-6036887.post@n2.nabble.com>



curriegrad2004 wrote:
> Similar configuration here, but all performed under a Fedora 13 x86 VM
> running on VirtualBox with the host running on Windows 7. Guest has
> 2GB of RAM allocated to it and no GUI installed at all.
Interesting. You do a compilation through a VM running on a Win7 and it
takes about 20minutes. Does the compilation use both CPU? Mine is a straight
AMD64 PhenomII X3 running on an OpenSuSE v11.3 and it takes almost an hour
long to compile. Perhaps, it has something to do with cross compilation.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Parallel-compilation-on-a-Linux-machine-tp6034541p6036887.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From krice at freeswitch.org  Thu Feb 17 20:55:38 2011
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 17 Feb 2011 11:55:38 -0600
Subject: [Freeswitch-users] Parallel compilation on a Linux machine
In-Reply-To: <1297964726895-6036887.post@n2.nabble.com>
Message-ID: <C982BD3A.6063%krice@freeswitch.org>

Tony might have to chime in here... But does the turbo-build.sh script still
work right?

K


On 2/17/11 11:45 AM, "mazilo" <Nabble at slickdeals.endjunk.com> wrote:

> 
> 
> curriegrad2004 wrote:
>> Similar configuration here, but all performed under a Fedora 13 x86 VM
>> running on VirtualBox with the host running on Windows 7. Guest has
>> 2GB of RAM allocated to it and no GUI installed at all.
> Interesting. You do a compilation through a VM running on a Win7 and it
> takes about 20minutes. Does the compilation use both CPU? Mine is a straight
> AMD64 PhenomII X3 running on an OpenSuSE v11.3 and it takes almost an hour
> long to compile. Perhaps, it has something to do with cross compilation.
> 
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.




From victor.chukalovskiy at utoronto.ca  Thu Feb 17 21:03:42 2011
From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy)
Date: Thu, 17 Feb 2011 13:03:42 -0500
Subject: [Freeswitch-users] Broad-voice configuration trouble.
In-Reply-To: <AANLkTi=Bb6Jw0gkxwOpCauA-DR2iPCadvLBH7_7GxuXU@mail.gmail.com>
References: <4D5D3F10.2060905@utoronto.ca>	<AANLkTimLD2gRoYicAfweBEVt7gf=tPaLE+w-2QRRb9n5@mail.gmail.com>
	<AANLkTi=Bb6Jw0gkxwOpCauA-DR2iPCadvLBH7_7GxuXU@mail.gmail.com>
Message-ID: <4D5D62FE.5020507@utoronto.ca>

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From neilp at cs.stanford.edu  Thu Feb 17 21:15:39 2011
From: neilp at cs.stanford.edu (Neil Patel)
Date: Thu, 17 Feb 2011 23:45:39 +0530
Subject: [Freeswitch-users] premature disconnect on outbound call:
	destroy/unlink session from object
Message-ID: <AANLkTi=BzQsCy10jCiR4inKO0eER-gLt6i9W14oBP0a2@mail.gmail.com>

Hi All,

I am running into an intermittent problem with outbound calling. I originate
calls (using FreeTDM to Sangoma hardware) from a lua script, and every so
often (10-50% of the time) the call disconnects just as the endpoints
answers the call. I am executing these calls identically and to the same
phone number. Here is CLI output for a call that doesn't disconnect:
http://pastebin.freeswitch.org/15409

And here is an example of one that does upon answering on the endpoint:
http://pastebin.freeswitch.org/15408

I noticed that one difference is that in the disconnected call there is the
following output:

2011-02-17 23:11:48.747752 [DEBUG] switch_core_state_machine.c:204
FreeTDM/2:1/09586550654 Standard SOFT_EXECUTE
2011-02-17 23:11:48.747752 [DEBUG] switch_core_state_machine.c:372
(FreeTDM/2:1/09586550654) State SOFT_EXECUTE going to sleep
*2011-02-17 23:11:48.747752 [DEBUG] switch_cpp.cpp:981
FreeTDM/2:1/09586550654 destroy/unlink session from object*
2011-02-17 23:11:48.747752 [DEBUG] switch_channel.c:2535
(FreeTDM/2:1/09586550654) Callstate Change ACTIVE -> HANGUP

I'm wondering what's triggering that destroy/unlink. I've taken this issue
up with Sangoma support, they are saying that there is nothing from the HW
side that would trigger this, though freeTDM may possible be passing back
some variables in a way that causes this and subsequent callstate change.

In case it's useful, pcaps for the normal,connected call and disconnected
fault call are attached.

Thanks in advance,
Neil
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From steveayre at gmail.com  Thu Feb 17 21:26:55 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 17 Feb 2011 18:26:55 +0000
Subject: [Freeswitch-users] Broad-voice configuration trouble.
In-Reply-To: <4D5D62FE.5020507@utoronto.ca>
References: <4D5D3F10.2060905@utoronto.ca>
	<AANLkTimLD2gRoYicAfweBEVt7gf=tPaLE+w-2QRRb9n5@mail.gmail.com>
	<AANLkTi=Bb6Jw0gkxwOpCauA-DR2iPCadvLBH7_7GxuXU@mail.gmail.com>
	<4D5D62FE.5020507@utoronto.ca>
Message-ID: <AANLkTinavPKNAgejSmzuoDUjB_SKLRmt1+A2TirpRG7U@mail.gmail.com>

Does <action application="set" data="verbose_sdp=true"/> before the bridge
help at all?

Some softswitches don't like the rtpmap missing even though it's optional
for static payload numbers. Perhaps broadvoice is one of them.

-Steve


On 17 February 2011 18:03, Victor Chukalovskiy <
victor.chukalovskiy at utoronto.ca> wrote:

>  Done with a paste-bin:
>
> http://pastebin.freeswitch.org/15410
>
> 1.2.3.4 is my PBX public IP address (masked with 1234 for security)
> 2014581234 is my Broadvoice DID (masked with 1234 for security)
>
> Thanks,
> Victor
>
>
> On 17/02/11 11:41 AM, Steven Ayre wrote:
>
> 403 can also happen with them if the INVITE's too big. They're not RFC
> compliant.
>
> Victor, can you post the siptrace?
>
> -Steve
>
>
>
> On 17 February 2011 16:21, Michael Collins <msc at freeswitch.org> wrote:
>
>> 403 usually means bad password or they don't like the IP addr you're
>> calling from. Call them up and confirm all your settings and ask them to
>> give you some further details regarding the 403.
>> -MC
>>
>>
>>  On Thu, Feb 17, 2011 at 7:30 AM, Victor Chukalovskiy <
>> victor.chukalovskiy at utoronto.ca> wrote:
>>
>>> Hello,
>>>
>>> I can't get BroadVoice running.
>>> Currently I follow this page:
>>> http://wiki.freeswitch.org/wiki/Provider_Configuration:_Broadvoice
>>> and it registers Ok.
>>>
>>> However, outgoing calls fail with Broadvoice sending me "403 Forbidden".
>>> I do set absolute_codec_string to PCMU and I see that PCMU is indeed the
>>> only codec in my SDP.
>>>
>>> Any help is appreciated.
>>> Actually, can anyone share an INVITE packet that works with Broad-voice?
>>> I think I'll be able to solve it then.
>>>
>>> Thank you,
>>> Victor
>>>
>>> > Sorry for hijacking previous thread!
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> 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 victor.chukalovskiy at utoronto.ca  Thu Feb 17 21:40:49 2011
From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy)
Date: Thu, 17 Feb 2011 13:40:49 -0500
Subject: [Freeswitch-users] Broad-voice configuration trouble.
In-Reply-To: <AANLkTinavPKNAgejSmzuoDUjB_SKLRmt1+A2TirpRG7U@mail.gmail.com>
References: <4D5D3F10.2060905@utoronto.ca>	<AANLkTimLD2gRoYicAfweBEVt7gf=tPaLE+w-2QRRb9n5@mail.gmail.com>	<AANLkTi=Bb6Jw0gkxwOpCauA-DR2iPCadvLBH7_7GxuXU@mail.gmail.com>	<4D5D62FE.5020507@utoronto.ca>
	<AANLkTinavPKNAgejSmzuoDUjB_SKLRmt1+A2TirpRG7U@mail.gmail.com>
Message-ID: <4D5D6BB1.8030901@utoronto.ca>

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From nazim.aghabayov at gmail.com  Thu Feb 17 21:56:38 2011
From: nazim.aghabayov at gmail.com (Nazim Aghabayov)
Date: Thu, 17 Feb 2011 22:56:38 +0400
Subject: [Freeswitch-users] Error installing mod com g729
In-Reply-To: <AANLkTin4S6czPQvTOar-TBrSpRrZeQzKrOxapKk7rcgP@mail.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>
Message-ID: <4D5D6F66.9060606@gmail.com>

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?


Thanks and Regards,
Nazim


From msc at freeswitch.org  Thu Feb 17 22:12:49 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 17 Feb 2011 11:12:49 -0800
Subject: [Freeswitch-users] IVR application
In-Reply-To: <col111-w4579CBD2496FFFEAEB7B10ACD50@phx.gbl>
References: <col111-w4579CBD2496FFFEAEB7B10ACD50@phx.gbl>
Message-ID: <AANLkTikknsrX1ZqHmnKkBQRfgCAhZ=fwhsMtuT+X92Qe@mail.gmail.com>

FS supports SIP for sure, and if you're feeling crazy you can try h.323. Or
you can build the IVR in FS itself and cut out the middle man.
-MC

On Thu, Feb 17, 2011 at 8:18 AM, Jason b.a <k-b-24 at live.com> wrote:

>  yeh i am using xlite , so my design will be
> xlite--IVR application--freeswitch
> but what about the signaling or control protocol used between app and Fs.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 17 22:15:32 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 17 Feb 2011 11:15:32 -0800
Subject: [Freeswitch-users] non-preemption in play_and_get_digits
In-Reply-To: <4D5CAEEF.4000700@bksys.co.in>
References: <4D5CAEEF.4000700@bksys.co.in>
Message-ID: <AANLkTikf3ve7m2mxqujnUmRMkGk5rE7iWB1Zw9-Luzm-@mail.gmail.com>

Can you rephrase the question? I'm not sure what you need.
-MC

On Wed, Feb 16, 2011 at 9:15 PM, abubacker <abubacker at bksystems.co.in>wrote:

> Is this possible to set the non  preemption mode int the
> play_and_get_digits_application ?
> If yes then how ?
>
>
> --
> Best regards,
> N.Abubacker ,
> Associate system engineer ,
> bk systems pvt ltd ,
> Ph : 9144-43902701
>
> Disclaimer: http://www.bksystems.co.in/email-policy
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 17 22:18:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 17 Feb 2011 11:18:02 -0800
Subject: [Freeswitch-users] application and media server
In-Reply-To: <COL111-W50BA6C8284D44C80B5369DACD20@phx.gbl>
References: <COL111-W50BA6C8284D44C80B5369DACD20@phx.gbl>
Message-ID: <AANLkTin2m8s8=a4pQoZV9891pQaHFFdsQZ7TEVypV8kB@mail.gmail.com>

What, exactly, are you building? And do you truly need 3 different servers?
-MC

On Tue, Feb 15, 2011 at 8:32 PM, Jason b.a <k-b-24 at live.com> wrote:

>  Hi  ,
> should i use sip server to connect the application server with the
> freeswitch (media server).
>
> client ---- application server --- sip server ---- freeswitch
>
> and is MSCML or VXML embedded in the sip message so the application server
> can control the freeswitch.
>
> thank you .
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 17 22:25:14 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 17 Feb 2011 11:25:14 -0800
Subject: [Freeswitch-users] Parallel compilation on a Linux machine
In-Reply-To: <1297964726895-6036887.post@n2.nabble.com>
References: <1297913369896-6034541.post@n2.nabble.com>
	<AANLkTinscs9H1tGtJB4tvLDgK--cxK+CZU8H0ZRgFQOk@mail.gmail.com>
	<1297944336320-6035621.post@n2.nabble.com>
	<AANLkTikaL=0GbPP-Vy5Y2CqEoJuriaqWkjrEQVyC5gDQ@mail.gmail.com>
	<1297964726895-6036887.post@n2.nabble.com>
Message-ID: <AANLkTinVe_oBLNhA+iLbo9fN2pD_HUPDqRwjbHv6YOfY@mail.gmail.com>

It's only configured to use one core, same result too when I used
CentOS 5.5 x64 on the same machine. The host is running on x64 if
anything helps. Maybe is it the fact that you forgot to disable cool
n' quiet or CPU scaling in the BIOS? For me I'm pretty sure I flipped
off CPU Scaling/cool n' quiet on my machine.

On Thu, Feb 17, 2011 at 9:45 AM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
>
> curriegrad2004 wrote:
>> Similar configuration here, but all performed under a Fedora 13 x86 VM
>> running on VirtualBox with the host running on Windows 7. Guest has
>> 2GB of RAM allocated to it and no GUI installed at all.
> Interesting. You do a compilation through a VM running on a Win7 and it
> takes about 20minutes. Does the compilation use both CPU? Mine is a straight
> AMD64 PhenomII X3 running on an OpenSuSE v11.3 and it takes almost an hour
> long to compile. Perhaps, it has something to do with cross compilation.
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Parallel-compilation-on-a-Linux-machine-tp6034541p6036887.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 cjbujold at accra.ca  Thu Feb 17 22:20:04 2011
From: cjbujold at accra.ca (Charles Bujold)
Date: Thu, 17 Feb 2011 15:20:04 -0400
Subject: [Freeswitch-users] Unable to dial IVR menu options
Message-ID: <003501cbced7$ae7c18d0$0b744a70$@accra.ca>

Here is the scenario, I have a telephone connected to a Grandstream HT503
fxs with a pstn connected to the fxo and a SIP connection via the Lan to a
freeswitch PBX. If I dial my voicemail (located on the SIP Freeswitch) I get
the IVR for the voice mail but when I dial my password it does not get the
numbers (password#)  I dial to hear my messages. If I try this from a SIP
phone everything works and I can dial an IVR option.

It looks like if Freeswitch is not able to pickup the dial numbers of the
phone once the call is started.   Any suggestion on how to fix?

Thanks
cjb

 

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From msc at freeswitch.org  Thu Feb 17 22:40:57 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 17 Feb 2011 11:40:57 -0800
Subject: [Freeswitch-users] Unable to dial IVR menu options
In-Reply-To: <003501cbced7$ae7c18d0$0b744a70$@accra.ca>
References: <003501cbced7$ae7c18d0$0b744a70$@accra.ca>
Message-ID: <AANLkTinSa9qwZJ-ED6S2tPD4PDegqDQEjDi0U4dYVdE3@mail.gmail.com>

Make sure that you have the RFC2833 turned on in the unit.
http://wiki.freeswitch.org/wiki/Interop_List#Grandstream_HandyTone_503

-MC

On Thu, Feb 17, 2011 at 11:20 AM, Charles Bujold <cjbujold at accra.ca> wrote:

> Here is the scenario, I have a telephone connected to a Grandstream HT503
> fxs with a pstn connected to the fxo and a SIP connection via the Lan to a
> freeswitch PBX. If I dial my voicemail (located on the SIP Freeswitch) I get
> the IVR for the voice mail but when I dial my password it does not get the
> numbers (password#)  I dial to hear my messages. If I try this from a SIP
> phone everything works and I can dial an IVR option.
>
> It looks like if Freeswitch is not able to pickup the dial numbers of the
> phone once the call is started.   Any suggestion on how to fix?
>
> Thanks
> 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 Nabble at slickdeals.endjunk.com  Thu Feb 17 22:59:21 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 17 Feb 2011 11:59:21 -0800 (PST)
Subject: [Freeswitch-users] Parallel compilation on a Linux machine
In-Reply-To: <AANLkTinVe_oBLNhA+iLbo9fN2pD_HUPDqRwjbHv6YOfY@mail.gmail.com>
References: <1297913369896-6034541.post@n2.nabble.com>
	<AANLkTinscs9H1tGtJB4tvLDgK--cxK+CZU8H0ZRgFQOk@mail.gmail.com>
	<1297944336320-6035621.post@n2.nabble.com>
	<AANLkTikaL=0GbPP-Vy5Y2CqEoJuriaqWkjrEQVyC5gDQ@mail.gmail.com>
	<1297964726895-6036887.post@n2.nabble.com>
	<AANLkTinVe_oBLNhA+iLbo9fN2pD_HUPDqRwjbHv6YOfY@mail.gmail.com>
Message-ID: <1297972761985-6037387.post@n2.nabble.com>



curriegrad2004 wrote:
> 
> It's only configured to use one core, same result too when I used
> CentOS 5.5 x64 on the same machine. The host is running on x64 if
> anything helps. Maybe is it the fact that you forgot to disable cool
> n' quiet or CPU scaling in the BIOS? For me I'm pretty sure I flipped
> off CPU Scaling/cool n' quiet on my machine.
Good point. But, when I cross compile Linux kernel with -j switch, the
compilation only takes a little less than two minutes. Without the -j
switch, the compilation takes more than 10 minutes, IIRC.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Parallel-compilation-on-a-Linux-machine-tp6034541p6037387.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mitch.capper at gmail.com  Thu Feb 17 23:20:25 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Thu, 17 Feb 2011 12:20:25 -0800
Subject: [Freeswitch-users] TLS / SSL improvements patch request for testing
Message-ID: <AANLkTikxmWqLkkm6GfCzGh9uEktkXcaaovP1MaEw_Voy@mail.gmail.com>

Earlier today on trunk there was a fix committed so that if you have
had an issue getting registrations to work over TLS/TCP try again as
it may just work now without any config changes.  In addition there is
a set of patches to expose some additional TLS options.  The patches
are at: http://jira.freeswitch.org/browse/FS-3071  the first patch
(sofia_tls_extra_options.patch) is the main one, the other two are
only needed if you want to use private keys that are passworded, note
the upstream patch needs to be applied in the
freeswitch/libs/sofia-sip.

If you are using freeswitch to connect to a remote server securely it
is highly advisable to be using the options in this patch.  Right now
freeswitch does no certificate validation which means you could easily
be MITM defeating the encryption.

The main config options for sip profiles are:
tls-verify-policy - this is the important one,  set to 'out' to ensure
remote servers we connect to(gateways) have a valid certificate, set
to 'in' to ensure any clients connecting to us have a valid
certificate, and set to 'all' to validate both.
tls-only - set to true and freeswitch will not even bind for
un-encrypted registrations
tls-no-verify-date - set to true to disable checking of the dates in
certificates
tls-verify-depth - max depth for validating a certificate

and optionally with the extra two patches:
tls-passphrase - set the passphrase for the private keys used


NOTE: once you enable certificate validation you will want to ensure
in your tls-cert-dir you have a cafile.pem which contains the entire
certificate chain to validate connections.  Most of the time this is
going to just be the CA cert for the certificate issuer.   Also you
can optionally put an agent.pem in the cert-dir this is the key/cert
that will be used for freeswitch when it connects out (its client
certificate/key).   It is not needed however if the remote side is not
validating you.

If you are running into problems try turning up the debug level by
setting "sofia loglevel tport 9"  it should give you more details
about why things fail (as otherwise you may just see gateway timeout
messages or something similar).


Please test if you can!

Also for those not using TLS /SSL right now and going over the
internet you should try and switch to it if possible.  There is very
low overhead (using sslv23 you still have udp packets for the actual
rtp stream and they are just 167 bytes vs 164 without encryption)  to
get started with it take a look at:
http://wiki.freeswitch.org/wiki/SIP_TLS

Note I would suggest sslv23 over tlsv1,  sslv23 keeps the RTP stream
as UDP which should mean least chance of a difference between no
encryption and with encryption.   The page does not take into account
these new changes yet however.  For a 30 second tutorial on getting
started in linux:

from the freeswitch bin dir on the server:
  ./gentls_cert setup  -cn pbx.yourdomain.com -alt
DNS:pbx.yourdomain.com -org pbx.yourdomain.com
  ./gentls_cert create -cn pbx.yourdomain.com -alt
DNS:pbx.yourdomain.com -org pbx.yourdomain.com

then make sure your freeswitch user owns the conf/ssl folder
generated.  Copy cafile.pem from it to any clients then set the
following options in your profile config:
<param name="tls" value="true"/>
<param name="tls-bind-params" value="transport=tls"/>
<param name="tls-version" value="sslv23"/>
<param name="tls-verify-policy" value="none"/>

and you should be able to have clients register securely( again see
wiki page for more information).

For freeswitch client:
enable the above tls options, put the cafile.pem from the server in
the conf/ssl folder, set:
<param name="tls-verify-policy" value="out"/>
and in the gateway:
<param name="register-transport" value="tls" />
<param name="contact-params" value="tport=tcp"/>
<param name="realm" value="sip.yourserver.com:5061" />  (or whatever
the TLS port you are using is)

~Mitch


From cjbujold at accra.ca  Thu Feb 17 23:47:07 2011
From: cjbujold at accra.ca (Charles Bujold)
Date: Thu, 17 Feb 2011 16:47:07 -0400
Subject: [Freeswitch-users] Unable to dial IVR menu options
In-Reply-To: <AANLkTinSa9qwZJ-ED6S2tPD4PDegqDQEjDi0U4dYVdE3@mail.gmail.com>
References: <003501cbced7$ae7c18d0$0b744a70$@accra.ca>
	<AANLkTinSa9qwZJ-ED6S2tPD4PDegqDQEjDi0U4dYVdE3@mail.gmail.com>
Message-ID: <005d01cbcee3$d7576c80$86064580$@accra.ca>

Already set to RFC2833, any other ideas?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: February-17-11 3:41 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Unable to dial IVR menu options

 

Make sure that you have the RFC2833 turned on in the unit. 

http://wiki.freeswitch.org/wiki/Interop_List#Grandstream_HandyTone_503

 

-MC

On Thu, Feb 17, 2011 at 11:20 AM, Charles Bujold <cjbujold at accra.ca> wrote:

Here is the scenario, I have a telephone connected to a Grandstream HT503
fxs with a pstn connected to the fxo and a SIP connection via the Lan to a
freeswitch PBX. If I dial my voicemail (located on the SIP Freeswitch) I get
the IVR for the voice mail but when I dial my password it does not get the
numbers (password#)  I dial to hear my messages. If I try this from a SIP
phone everything works and I can dial an IVR option.

It looks like if Freeswitch is not able to pickup the dial numbers of the
phone once the call is started.   Any suggestion on how to fix?

Thanks
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  Fri Feb 18 00:58:46 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 17 Feb 2011 16:58:46 -0500
Subject: [Freeswitch-users] txfax and multiple files
Message-ID: <541CCE77D6AC47478665282FC736888A@e1705>

I didn't succeed so I use tiffcp to condensate
all tiff files in one with several pages
  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Thursday, February 17, 2011 11:54 AM
  Subject: Re: [Freeswitch-users] txfax and multiple files


  not yet, will do test later today.
  If I remember it's possible by separate every path files by a space or comma
  Will confirm soon
  in case of it's not possible, can I run tfax app several times while the channel is up ?

  Thanks
    ----- Original Message ----- 
    From: Michael Collins 
    To: FreeSWITCH Users Help 
    Sent: Thursday, February 17, 2011 11:41 AM
    Subject: Re: [Freeswitch-users] txfax and multiple files


    Did you figure this out? I didn't know that you could queue up multiple files for transmission, but if you can then I'd like to know so that I can get it properly documented... 


    -MC


    On Wed, Feb 16, 2011 at 12:19 AM, Madovsky <infos at madovsky.org> wrote:

      What is the right syntax to send multiple files with
      mod_spandsp txfax ?

      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
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From george.niculae79 at gmail.com  Fri Feb 18 01:37:00 2011
From: george.niculae79 at gmail.com (George Niculae)
Date: Fri, 18 Feb 2011 00:37:00 +0200
Subject: [Freeswitch-users] big audio delay in bridged conference
Message-ID: <AANLkTikNn40pM56bYb6okRipL=0tivvUEEy7apSLEbkn@mail.gmail.com>

 Hi All,

I have two conferences set in my dialplan - one is bridged and is using
mod_loopback (as advised in http://jira.freeswitch.org/browse/FS-2741 ) the
other is a normal one. Extensions in dialplan looks like:

 <extension name="3202">
    <condition field="destination_number" expression="^202-conference$">
      <action application="bridge"
data="loopback/app=conference:202-conference at 3202"/>
    </condition>
  </extension>
  <extension name="3203">
    <condition field="destination_number" expression="^203-conference$">
      <action application="conference" data="203-conference at 3203"/>
    </condition>
  </extension>

With the bridged conference I hit weird behaviour: there is a delay in audio
that grows in time and can even touch 9 seconds in about 10 minutes. The
other conference works just fine tested in same conditions as the first one.
I have monitored FS console during both conferences but noticed nothing
special. (running HEAD, commit 49a5effcdf2cea9e0ddcf146cf3fe85d1872e654)

Any pointer greatly appreciated,

Thanks,
George
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From msc at freeswitch.org  Fri Feb 18 02:03:43 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 17 Feb 2011 15:03:43 -0800
Subject: [Freeswitch-users] Unable to dial IVR menu options
In-Reply-To: <005d01cbcee3$d7576c80$86064580$@accra.ca>
References: <003501cbced7$ae7c18d0$0b744a70$@accra.ca>
	<AANLkTinSa9qwZJ-ED6S2tPD4PDegqDQEjDi0U4dYVdE3@mail.gmail.com>
	<005d01cbcee3$d7576c80$86064580$@accra.ca>
Message-ID: <AANLkTi=gBVWEcuOX+bPr8geS0-JiViyHFHx0oYAXxtkA@mail.gmail.com>

Capture the media between the ATA and FS. Analyze w/ Wireshark - is the ATA
*really* sending RFC2833?

Check out this cool video on how to do SIP debugging on your network. You
won't be disappointed.

http://www.viddler.com/explore/cluecon/videos/39/

-MC

On Thu, Feb 17, 2011 at 12:47 PM, Charles Bujold <cjbujold at accra.ca> wrote:

> Already set to RFC2833, any other ideas?
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* February-17-11 3:41 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Unable to dial IVR menu options
>
>
>
> Make sure that you have the RFC2833 turned on in the unit.
>
> http://wiki.freeswitch.org/wiki/Interop_List#Grandstream_HandyTone_503
>
>
>
> -MC
>
> On Thu, Feb 17, 2011 at 11:20 AM, Charles Bujold <cjbujold at accra.ca>
> wrote:
>
> Here is the scenario, I have a telephone connected to a Grandstream HT503
> fxs with a pstn connected to the fxo and a SIP connection via the Lan to a
> freeswitch PBX. If I dial my voicemail (located on the SIP Freeswitch) I get
> the IVR for the voice mail but when I dial my password it does not get the
> numbers (password#)  I dial to hear my messages. If I try this from a SIP
> phone everything works and I can dial an IVR option.
>
> It looks like if Freeswitch is not able to pickup the dial numbers of the
> phone once the call is started.   Any suggestion on how to fix?
>
> Thanks
> 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 k-b-24 at live.com  Fri Feb 18 02:21:31 2011
From: k-b-24 at live.com (Jason b.a)
Date: Thu, 17 Feb 2011 23:21:31 +0000
Subject: [Freeswitch-users] IVR application
Message-ID: <COL111-W10259DC7F442E309D2BC40ACD50@phx.gbl>



no i need to host my ivr on a application server, but how can i control a running FreeSWITCH server from another machine, 
for example how the application will send an info message to freeswitch to play a prompt !   		 	   		  
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From brad at tritelcomm.com  Fri Feb 18 02:30:09 2011
From: brad at tritelcomm.com (Brad Mina)
Date: Thu, 17 Feb 2011 15:30:09 -0800
Subject: [Freeswitch-users] IVR application
In-Reply-To: <COL111-W10259DC7F442E309D2BC40ACD50@phx.gbl>
References: <COL111-W10259DC7F442E309D2BC40ACD50@phx.gbl>
Message-ID: <AANLkTi=o2ihWfoo8fcT9nv-O+SnAGqjG+km2mtNMo+M0@mail.gmail.com>

What you might be looking for is this:
http://wiki.freeswitch.org/wiki/Mod_xmpp_event

On Thu, Feb 17, 2011 at 3:21 PM, Jason b.a <k-b-24 at live.com> wrote:

>  no i need to host my ivr on a application server, but how can i control a
> running FreeSWITCH server from another machine,
> for example how the application will send an info message to freeswitch to
> play a prompt !
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 18 03:27:54 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 17 Feb 2011 16:27:54 -0800
Subject: [Freeswitch-users] IVR application
In-Reply-To: <COL111-W10259DC7F442E309D2BC40ACD50@phx.gbl>
References: <COL111-W10259DC7F442E309D2BC40ACD50@phx.gbl>
Message-ID: <AANLkTi=wkOTYEsWhb98eCcuMdf18yBnLJDgNheCob4G0@mail.gmail.com>

Boy, you picked a not-so-easy one. You'll need to get familiar with the
event socket and ESL (the event socket library). Can you give us an example
of call flow? We might be able to offer more specific suggestions.

-MC

On Thu, Feb 17, 2011 at 3:21 PM, Jason b.a <k-b-24 at live.com> wrote:

>  no i need to host my ivr on a application server, but how can i control a
> running FreeSWITCH server from another machine,
> for example how the application will send an info message to freeswitch to
> play a prompt !
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 18 03:29:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 17 Feb 2011 16:29:38 -0800
Subject: [Freeswitch-users] Skype 2.0.72 for Linux
Message-ID: <AANLkTinYLAuoB9wy_vjbZwbO38PthoFzSia-2bS9yCwf@mail.gmail.com>

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

Any suggestions are appreciated.

-MC
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From k-b-24 at live.com  Fri Feb 18 03:49:14 2011
From: k-b-24 at live.com (Jason b.a)
Date: Fri, 18 Feb 2011 00:49:14 +0000
Subject: [Freeswitch-users] IVR application
Message-ID: <COL111-W41EE8237801F7EE7F81570ACD40@phx.gbl>


Here is a call flow sample for playing a prompt, later on i need to make my IVR application virtual using cloud computing , in this example freeswitch can play the role of media server,
for media processing , and the application server for signal processing and controlling the media server sure, is it right ! or should i use different media server such as SEMS ! 
 
sip Caller                      Application Server                             Media server
 
   -----------invite---------------->
                                             ----------------invite--------------------->
  <---------Trying-----------------
                                             <--------------200ok----------------------
  <-------------200ok-------------
   --------------Ack--------------->
                                             -----------------Ack----------------------->
 
                          Caller in interactiong with media server
 
                                            ---------Info(play prompt X.wav)--------->
                                            <----------------Ok--------------------------- 		 	   		  
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From krice at freeswitch.org  Fri Feb 18 05:33:49 2011
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 17 Feb 2011 20:33:49 -0600
Subject: [Freeswitch-users] IVR application
In-Reply-To: <COL111-W41EE8237801F7EE7F81570ACD40@phx.gbl>
Message-ID: <C98336AD.6098%krice@freeswitch.org>

There are many ways to do what you are asking about with freeswitch

Keep in mind that freeswitch itself can be an IVR media server and
processor... The question really is how do you want to control the IVRs?

FreeSWITCH offers multiple ways to script and interact with a DB such as
LUA, JavaScript (via mod_spidermonkey), and others, or you can use the
event_socket stuff in FreeSWITCH to control calls, yet another method is to
write a C Application that plugs into FreeSWITCH and directs what media is
played to the remote party as well as reacts to input from the remote end.

Then there is MRCP which is another whole can of worms...

So keep in mind here that the flow is more like


Caller -> SIP -> FreeSWITCH -> ${CONTROL_PROTOCOL} -> Application middleware

That control protocol can be many many things


On 2/17/11 6:49 PM, "Jason b.a" <k-b-24 at live.com> wrote:

> Here is a call flow sample for playing a prompt, later on i need to make my
> IVR application virtual using cloud computing , in this example freeswitch can
> play the role of media server,
> for media processing , and the application server for signal processing and
> controlling the media server sure, is it right ! or should i use different
> media server such as SEMS !
>  
> sip Caller                      Application Server
> Media server
>  
>    -----------invite---------------->
>                  
> ----------------invite--------------------->
>   <---------Trying-----------------
>                  
> <--------------200ok----------------------
>   <-------------200ok-------------
>    --------------Ack--------------->
>                  
> -----------------Ack----------------------->
>  
>                           Caller in interactiong with media server
>  
>                                             ---------Info(play prompt
> X.wav)--------->
>                  
> <----------------Ok---------------------------
>        
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Feb 18 05:42:12 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 17 Feb 2011 18:42:12 -0800 (PST)
Subject: [Freeswitch-users] Skype 2.0.72 for Linux
In-Reply-To: <AANLkTinYLAuoB9wy_vjbZwbO38PthoFzSia-2bS9yCwf@mail.gmail.com>
References: <AANLkTinYLAuoB9wy_vjbZwbO38PthoFzSia-2bS9yCwf@mail.gmail.com>
Message-ID: <1297996932418-6038482.post@n2.nabble.com>


You can download it from 
ftp://ftp.isu.edu.tw/pub/Linux/PLD/dists/2.0/test/SRPMS/skype-2.0.0.27-1.src.rpm
here . Then, use this shell scripts 
http://www.mombu.com/gnu_linux/red-hat/t-disrpm-shell-script-8907760.html
disrpm  utility to extract it.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Skype-2-0-72-for-Linux-tp6038235p6038482.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Fri Feb 18 05:48:03 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 17 Feb 2011 18:48:03 -0800 (PST)
Subject: [Freeswitch-users] Call Recording
In-Reply-To: <3367A6EC-EEC2-4930-ABCE-B6CA261932C2@freeswitch.org>
References: <011201cb5f4e$89dbc3c0$9d934b40$@teotech.com>
	<3367A6EC-EEC2-4930-ABCE-B6CA261932C2@freeswitch.org>
Message-ID: <1297997283132-6038514.post@n2.nabble.com>



Brian West wrote:
> 
> Yes it works fine but are you sure the file path is correct?
Cool. Once recorded, what extension number should I dial to listen to the
recording file?

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Call-Recording-tp5581208p6038514.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From fcintrono at gmail.com  Fri Feb 18 04:42:50 2011
From: fcintrono at gmail.com (=?ISO-8859-1?Q?Francisco_Javier_Cintr=F3n_Olgu=EDn?=)
Date: Thu, 17 Feb 2011 19:42:50 -0600
Subject: [Freeswitch-users] =?iso-8859-1?q?Newbie=B4s_question_about_FreeS?=
	=?iso-8859-1?q?witch=2E=2E=2E?=
Message-ID: <AANLkTinf=z4JEp4G2xK_0n+u8v0OhR++GBvPzzbT1Y_-@mail.gmail.com>

Hi, My name is Francisco from M?xico.

Here, in my work we have a very very old panasonic PBX(12 years old). We are
growing and we need to increase our external lines(from 3 to 4) and our
internal lines(from 6 to 10). Besides we need voice mail and voice menu too.


We asked for a quote to our panasonic dealer. The whole thing cost about
4,500 dollars.

My boss just saw a thing called FreeSwitch this morning looking for options
in Google. He asked my to investigate what this thing called FreeSwitch is
and if we could save some money using it instead of the panasonic solution.
So, here I am.

I have some experience as linux sysadmin(we have 1 oracle linux server and 1
linux print server) nevertheless I don?t have any idea where and how to
start this evaluation?


Please
Would you give us a clue where to see If FreeSwitch could work for us?

Thanks for your kind help.
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From edward.dejong at voicecarrier.com  Fri Feb 18 09:17:47 2011
From: edward.dejong at voicecarrier.com (Edward de Jong)
Date: Thu, 17 Feb 2011 22:17:47 -0800
Subject: [Freeswitch-users] which language to implement dial plans in?
Message-ID: <200AE28D-0F21-422F-9171-6D4384A29014@voicecarrier.com>

I am new to freeswitch, and wondering which of the many languages to implement my rather complex dial plan in. I know i can't express my desires in the normal XML language, because I need looping and database access, so which language is the best tested and most reliable in terms of usage with FreeSwitch. My finalists so far are Lua, Python, and good old C. Haven't used python or lua yet, but I figure C will have a big performance advantage.

but maybe not important considering how much greater the CPU load in the switch is devoted to handling audio streams...

edj



From oseslija at gmail.com  Fri Feb 18 10:28:53 2011
From: oseslija at gmail.com (Ognjen Seslija)
Date: Fri, 18 Feb 2011 08:28:53 +0100
Subject: [Freeswitch-users]
	=?iso-8859-1?q?Newbie=B4s_question_about_FreeS?=
	=?iso-8859-1?q?witch=2E=2E=2E?=
In-Reply-To: <AANLkTinf=z4JEp4G2xK_0n+u8v0OhR++GBvPzzbT1Y_-@mail.gmail.com>
References: <AANLkTinf=z4JEp4G2xK_0n+u8v0OhR++GBvPzzbT1Y_-@mail.gmail.com>
Message-ID: <AANLkTik6hFX05paABERb_9mfJgvJq82iQyLwQT-LRKt1@mail.gmail.com>

Join us on irc (irc.freenode.net , #freeswitch) and we'll help you.

In the meantime, check wiki.freeswitch.org.

On Fri, Feb 18, 2011 at 2:42 AM, Francisco Javier Cintr?n Olgu?n <
fcintrono at gmail.com> wrote:

> Hi, My name is Francisco from M?xico.
>
> Here, in my work we have a very very old panasonic PBX(12 years old). We
> are growing and we need to increase our external lines(from 3 to 4) and our
> internal lines(from 6 to 10). Besides we need voice mail and voice menu too.
>
>
> We asked for a quote to our panasonic dealer. The whole thing cost about
> 4,500 dollars.
>
> My boss just saw a thing called FreeSwitch this morning looking for options
> in Google. He asked my to investigate what this thing called FreeSwitch is
> and if we could save some money using it instead of the panasonic solution.
> So, here I am.
>
> I have some experience as linux sysadmin(we have 1 oracle linux server and
> 1 linux print server) nevertheless I don?t have any idea where and how to
> start this evaluation?
>
>
> Please
> Would you give us a clue where to see If FreeSwitch could work for us?
>
> Thanks for your kind help.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From edpimentl at gmail.com  Fri Feb 18 10:31:06 2011
From: edpimentl at gmail.com (EdPimentl)
Date: Fri, 18 Feb 2011 02:31:06 -0500
Subject: [Freeswitch-users]
	=?utf-8?q?Newbie=C2=B4s_question_about_FreeSwi?=
	=?utf-8?b?dGNoLi4u?=
In-Reply-To: <AANLkTinf=z4JEp4G2xK_0n+u8v0OhR++GBvPzzbT1Y_-@mail.gmail.com>
References: <AANLkTinf=z4JEp4G2xK_0n+u8v0OhR++GBvPzzbT1Y_-@mail.gmail.com>
Message-ID: <AANLkTi=hdr-Okh+NMJhXJ4mco2h_1iHELkixOWWjb1U-@mail.gmail.com>

Start here
http://wiki.freeswitch.org/wiki/Main_Page
http://wiki.freeswitch.org/wiki/Specsheet
https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-telephony-systems/book
/ (commercial )

Sincerely,
-E
vCardCloud.com




On Thu, Feb 17, 2011 at 8:42 PM, Francisco Javier Cintr?n Olgu?n <
fcintrono at gmail.com> wrote:

> Hi, My name is Francisco from M?xico.
>
> Here, in my work we have a very very old panasonic PBX(12 years old). We
> are growing and we need to increase our external lines(from 3 to 4) and our
> internal lines(from 6 to 10). Besides we need voice mail and voice menu too.
>
>
> We asked for a quote to our panasonic dealer. The whole thing cost about
> 4,500 dollars.
>
> My boss just saw a thing called FreeSwitch this morning looking for options
> in Google. He asked my to investigate what this thing called FreeSwitch is
> and if we could save some money using it instead of the panasonic solution.
> So, here I am.
>
> I have some experience as linux sysadmin(we have 1 oracle linux server and
> 1 linux print server) nevertheless I don?t have any idea where and how to
> start this evaluation?
>
>
> Please
> Would you give us a clue where to see If FreeSwitch could work for us?
>
> Thanks for your kind help.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Feb 18 11:17:02 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 18 Feb 2011 10:17:02 +0200
Subject: [Freeswitch-users] which language to implement dial plans in?
In-Reply-To: <200AE28D-0F21-422F-9171-6D4384A29014@voicecarrier.com>
References: <200AE28D-0F21-422F-9171-6D4384A29014@voicecarrier.com>
Message-ID: <AANLkTiknPESOVjnYTw-3gQVoXqBTj6nMPWzaUU2veNde@mail.gmail.com>

On Feb 18, 2011 8:51 AM, "Edward de Jong" <edward.dejong at voicecarrier.com>
wrote:
>
> I am new to freeswitch, and wondering which of the many languages to
implement my rather complex dial plan in. I know i can't express my desires
in the normal XML language, because I need looping and database access
There is a mod called mod_odbc_query in the git contrib that lets you make
sql queries from the dialplan. The static xml can do a lot.
I'm not sure what you mean by looping, but you can transfer to a new number,
and it can restart going through thr dialplan.

, so which language is the best tested and most reliable in terms of usage
with FreeSwitch. My finalists so far are Lua, Python, and good old C.
Haven't used python or lua yet, but I figure C will have a big performance
advantage.
>
> but maybe not important considering how much greater the CPU load in the
switch is devoted to handling audio streams...
>
Lua can be run directly, whereas python would be used eith mod_xml_curl,
served from a server (I do tis with php) so if you know lua, use it. If you
like python, use that.
C is probably overkill becaue you would need to compile it for any change
you make. If you have standardized functions that don't already have a mod,
then by all means write a modin C and share it wih the community.
-Avi

> edj
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From kond at nstel.ru  Fri Feb 18 11:57:12 2011
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Fri, 18 Feb 2011 11:57:12 +0300
Subject: [Freeswitch-users] dtmf via info
Message-ID: <mailman.0.1298019471.24552.freeswitch-users@lists.freeswitch.org>

Hi all,

I?d like to send dtmf via info and I?d like to set it in the dial plan.

Do I just need to set dtmf_type channel variable in the dial plan?

I tried to set <action application="set" data="dtmf_type=info"/> in the dialplan, but I do not see Info messages?.
Should it work in this way at all?
Did I miss something?
Thanks in advance,
Nikolay.

-- 
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.

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From steveayre at gmail.com  Fri Feb 18 12:00:59 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 18 Feb 2011 09:00:59 +0000
Subject: [Freeswitch-users] which language to implement dial plans in?
In-Reply-To: <AANLkTiknPESOVjnYTw-3gQVoXqBTj6nMPWzaUU2veNde@mail.gmail.com>
References: <200AE28D-0F21-422F-9171-6D4384A29014@voicecarrier.com>
	<AANLkTiknPESOVjnYTw-3gQVoXqBTj6nMPWzaUU2veNde@mail.gmail.com>
Message-ID: <AANLkTimK2va_j9RjuFVic1reVMb5hxcRTw_Qua4Wpyca@mail.gmail.com>

>
> Lua can be run directly, whereas python would be used eith mod_xml_curl,
> served from a server (I do tis with php) so if you know lua, use it. If you
> like python, use that.


Not true - while you can implement the script that mod_xml_curl using
Python, you can also use it directly within FreeSWITCH just like Lua:
http://wiki.freeswitch.org/wiki/Mod_python

-Steve

On 18 February 2011 08:17, Avi Marcus <avi at avimarcus.net> wrote:

>
> On Feb 18, 2011 8:51 AM, "Edward de Jong" <edward.dejong at voicecarrier.com>
> wrote:
> >
> > I am new to freeswitch, and wondering which of the many languages to
> implement my rather complex dial plan in. I know i can't express my desires
> in the normal XML language, because I need looping and database access
>  There is a mod called mod_odbc_query in the git contrib that lets you make
> sql queries from the dialplan. The static xml can do a lot.
> I'm not sure what you mean by looping, but you can transfer to a new
> number, and it can restart going through thr dialplan.
>
> , so which language is the best tested and most reliable in terms of usage
> with FreeSwitch. My finalists so far are Lua, Python, and good old C.
> Haven't used python or lua yet, but I figure C will have a big performance
> advantage.
> >
> > but maybe not important considering how much greater the CPU load in the
> switch is devoted to handling audio streams...
> >
> Lua can be run directly, whereas python would be used eith mod_xml_curl,
> served from a server (I do tis with php) so if you know lua, use it. If you
> like python, use that.
> C is probably overkill becaue you would need to compile it for any change
> you make. If you have standardized functions that don't already have a mod,
> then by all means write a modin C and share it wih the community.
>  -Avi
>
> > edj
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 18 12:07:13 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 18 Feb 2011 09:07:13 +0000
Subject: [Freeswitch-users] which language to implement dial plans in?
In-Reply-To: <200AE28D-0F21-422F-9171-6D4384A29014@voicecarrier.com>
References: <200AE28D-0F21-422F-9171-6D4384A29014@voicecarrier.com>
Message-ID: <AANLkTim+RdyCfSzeb7HyX5rUkC-UY-mdTjXsEbo18njn@mail.gmail.com>

C is faster, but it's probably not worth it unless you can show that there's
enough processing being done by your script that it'd have an impact. It's
slightly more work to tweak it than just editing a script since you'd need
to recompile and reload the module, which might also be trickier if you're
handling traffic at the time.

Lua (mod_lua) is probably the most commonly used language, but there's also
Python (mod_python) as you've spotted. There's also JavaScript
(mod_spidermonkey) which you haven't mentioned. Which you use is really up
to you though.

As far as database access goes, from Lua check out:
http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh

It gives you a database interface without having to install and configure
luasql, and you get to take advantage of the FS connection pooling which you
wouldn't if you were using the native support. That'll should speed up
database access times (you'll in most cases already be connected to the
database so the call doesn't have to wait while you connect). I don't think
that interface is available in the other languages just yet though.

-Steve


On 18 February 2011 06:17, Edward de Jong <edward.dejong at voicecarrier.com>wrote:

> I am new to freeswitch, and wondering which of the many languages to
> implement my rather complex dial plan in. I know i can't express my desires
> in the normal XML language, because I need looping and database access, so
> which language is the best tested and most reliable in terms of usage with
> FreeSwitch. My finalists so far are Lua, Python, and good old C. Haven't
> used python or lua yet, but I figure C will have a big performance
> advantage.
>
> but maybe not important considering how much greater the CPU load in the
> switch is devoted to handling audio streams...
>
> edj
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tayeb.meftah at gmail.com  Fri Feb 18 12:52:20 2011
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Fri, 18 Feb 2011 10:52:20 +0100
Subject: [Freeswitch-users] dtmf via info
In-Reply-To: <4d5e34bf.04a7960a.413d.61a0SMTPIN_ADDED@mx.google.com>
References: <4d5e34bf.04a7960a.413d.61a0SMTPIN_ADDED@mx.google.com>
Message-ID: <4D5E4154.2050108@gmail.com>

ok,
1. why you need sip info DTMF?
is not standard
2. you could try out <action application="set" data="dtmf-type=info"/>. 
if don't work, set it in the profile.
3. i strongly recomand you to use rfc2833 or inband dtmf.
thank you
Le 18/02/2011 09:57, Nikolay Kondratyev a ?crit :
>
> Hi all,
>
> I?d like to send dtmf via info and I?d like to set it in the dial plan.
>
> Do I just need to set dtmf_type channel variable in the dial plan?
>
> I tried to set<action application="set" data="dtmf_type=info"/>  in the dialplan, but I do not see Info messages?.
> Should it work in this way at all?
> Did I miss something?
> Thanks in advance,
> Nikolay.
>
> -- 
> 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
>    


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

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From avi at avimarcus.net  Fri Feb 18 13:37:41 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 18 Feb 2011 12:37:41 +0200
Subject: [Freeswitch-users] which language to implement dial plans in?
In-Reply-To: <AANLkTim+RdyCfSzeb7HyX5rUkC-UY-mdTjXsEbo18njn@mail.gmail.com>
References: <200AE28D-0F21-422F-9171-6D4384A29014@voicecarrier.com>
	<AANLkTim+RdyCfSzeb7HyX5rUkC-UY-mdTjXsEbo18njn@mail.gmail.com>
Message-ID: <AANLkTi=0fGt8RR+PMu8oMGWqqf4V1YHhX1uS2XCUdC68@mail.gmail.com>

mod_python, cool. I've heard lua has a very low overhead, but what about
mod_python (or PHP ESL)?
I'd imagine the script would have to be left "running" until the call
completed.
mod_python seems pretty well fleshed out...
-Avi

On Fri, Feb 18, 2011 at 11:07 AM, Steven Ayre <steveayre at gmail.com> wrote:

> C is faster, but it's probably not worth it unless you can show that
> there's enough processing being done by your script that it'd have an
> impact. It's slightly more work to tweak it than just editing a script since
> you'd need to recompile and reload the module, which might also be trickier
> if you're handling traffic at the time.
>
> Lua (mod_lua) is probably the most commonly used language, but there's also
> Python (mod_python) as you've spotted. There's also JavaScript
> (mod_spidermonkey) which you haven't mentioned. Which you use is really up
> to you though.
>
> As far as database access goes, from Lua check out:
> http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh
>
> It gives you a database interface without having to install and configure
> luasql, and you get to take advantage of the FS connection pooling which you
> wouldn't if you were using the native support. That'll should speed up
> database access times (you'll in most cases already be connected to the
> database so the call doesn't have to wait while you connect). I don't think
> that interface is available in the other languages just yet though.
>
> -Steve
>
>
> On 18 February 2011 06:17, Edward de Jong <edward.dejong at voicecarrier.com>wrote:
>
>> I am new to freeswitch, and wondering which of the many languages to
>> implement my rather complex dial plan in. I know i can't express my desires
>> in the normal XML language, because I need looping and database access, so
>> which language is the best tested and most reliable in terms of usage with
>> FreeSwitch. My finalists so far are Lua, Python, and good old C. Haven't
>> used python or lua yet, but I figure C will have a big performance
>> advantage.
>>
>> but maybe not important considering how much greater the CPU load in the
>> switch is devoted to handling audio streams...
>>
>> edj
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 18 13:44:47 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 18 Feb 2011 10:44:47 +0000
Subject: [Freeswitch-users] which language to implement dial plans in?
In-Reply-To: <AANLkTi=0fGt8RR+PMu8oMGWqqf4V1YHhX1uS2XCUdC68@mail.gmail.com>
References: <200AE28D-0F21-422F-9171-6D4384A29014@voicecarrier.com>
	<AANLkTim+RdyCfSzeb7HyX5rUkC-UY-mdTjXsEbo18njn@mail.gmail.com>
	<AANLkTi=0fGt8RR+PMu8oMGWqqf4V1YHhX1uS2XCUdC68@mail.gmail.com>
Message-ID: <AANLkTinp3gAZT24pfEdtfnquWC3dbSgR0JNoAVZ-paSB@mail.gmail.com>

In terms of running python scripts, it's running for the same time the
equivalent lua one would be. Don't know how the overhead compares, although
I expect lua is lower.

-Steve



On 18 February 2011 10:37, Avi Marcus <avi at avimarcus.net> wrote:

> mod_python, cool. I've heard lua has a very low overhead, but what about
> mod_python (or PHP ESL)?
> I'd imagine the script would have to be left "running" until the call
> completed.
>  mod_python seems pretty well fleshed out...
> -Avi
>
>
> On Fri, Feb 18, 2011 at 11:07 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> C is faster, but it's probably not worth it unless you can show that
>> there's enough processing being done by your script that it'd have an
>> impact. It's slightly more work to tweak it than just editing a script since
>> you'd need to recompile and reload the module, which might also be trickier
>> if you're handling traffic at the time.
>>
>> Lua (mod_lua) is probably the most commonly used language, but there's
>> also Python (mod_python) as you've spotted. There's also JavaScript
>> (mod_spidermonkey) which you haven't mentioned. Which you use is really up
>> to you though.
>>
>> As far as database access goes, from Lua check out:
>> http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh
>>
>> It gives you a database interface without having to install and configure
>> luasql, and you get to take advantage of the FS connection pooling which you
>> wouldn't if you were using the native support. That'll should speed up
>> database access times (you'll in most cases already be connected to the
>> database so the call doesn't have to wait while you connect). I don't think
>> that interface is available in the other languages just yet though.
>>
>> -Steve
>>
>>
>> On 18 February 2011 06:17, Edward de Jong <edward.dejong at voicecarrier.com
>> > wrote:
>>
>>> I am new to freeswitch, and wondering which of the many languages to
>>> implement my rather complex dial plan in. I know i can't express my desires
>>> in the normal XML language, because I need looping and database access, so
>>> which language is the best tested and most reliable in terms of usage with
>>> FreeSwitch. My finalists so far are Lua, Python, and good old C. Haven't
>>> used python or lua yet, but I figure C will have a big performance
>>> advantage.
>>>
>>> but maybe not important considering how much greater the CPU load in the
>>> switch is devoted to handling audio streams...
>>>
>>> edj
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kond at nstel.ru  Fri Feb 18 14:40:03 2011
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Fri, 18 Feb 2011 14:40:03 +0300
Subject: [Freeswitch-users] dtmf via info
In-Reply-To: <4D5E4154.2050108@gmail.com>
Message-ID: <mailman.1.1298029244.24552.freeswitch-users@lists.freeswitch.org>

I agree, that 2833 is better then sip info for dtmf. 

I need sip info dtmf because of the itsp.

Due to some particular features of my network I need to be able to choose dtmf method on the basis of destination number.

So I can not set dtmf-type in the profile?

I?ve already tried to set channel variable dtmf_type in the corresponding extention in my dialplan, that did not worked?

I tried to set ?dtmf-type? (as you proposed) too, but with the same result ? I don?t see info messages?

My scheme is as following: Avaya IPOFFICE ---- h323 ---- FS ----sip ----- SIPX ----- sip ITSP. 

FS is used ad h.323 ? sip gateway. 

So?

Is it possible to choose dtmf method on the basis of destination number?

Rgds,

Nikolay.

 

 

  _____  

From: Meftah Tayeb [mailto:tayeb.meftah at gmail.com] 
Sent: Friday, February 18, 2011 12:52 PM
To: FreeSWITCH Users Help
Cc: Nikolay Kondratyev
Subject: Re: [Freeswitch-users] dtmf via info

 

ok,
1. why you need sip info DTMF?
is not standard
2. you could try out     <action application="set" data="dtmf-type=info"/>. if don't work, set it in the profile.
3. i strongly recomand you to use rfc2833 or inband dtmf.
thank you
Le 18/02/2011 09:57, Nikolay Kondratyev a ?crit : 

Hi all,

I?d like to send dtmf via info and I?d like to set it in the dial plan.

Do I just need to set dtmf_type channel variable in the dial plan?

I tried to set <action application="set" data="dtmf_type=info"/> in the dialplan, but I do not see Info messages?.
Should it work in this way at all?
Did I miss something?
Thanks in advance,
Nikolay.


-- 
This message has been scanned for viruses and 
dangerous content by  <http://www.mailscanner.info/> MailScanner, and is 
believed to be clean. 

 
 
_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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inum: +883510001288000
phone: +13477595883


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From rajesh.npnr at yahoo.com  Fri Feb 18 14:58:20 2011
From: rajesh.npnr at yahoo.com (rex.alex)
Date: Fri, 18 Feb 2011 03:58:20 -0800 (PST)
Subject: [Freeswitch-users] The call dropped prematurely error on txfax
In-Reply-To: <AANLkTikmEvZB5tF6qiT2Jf4Ca6VzBB8bqQ_k=O59=eO=@mail.gmail.com>
References: <1297873355400-6032435.post@n2.nabble.com>
	<AANLkTingYEfo=4oaANLS8ZzySqLSaUOOm7ME=HpxxQ-C@mail.gmail.com>
	<1297885773534-6033268.post@n2.nabble.com>
	<AANLkTiku+q9B_FeSErvMuXSog6DQ4mMtDmxn_TjtG9Zw@mail.gmail.com>
	<1297954260038-6036197.post@n2.nabble.com>
	<140D740BF92640F7A514BD35BD4BD1AA@e1705>
	<1297958857213-6036503.post@n2.nabble.com>
	<C0AB523AE4774FA5837E88F48243F970@e1705>
	<1297961826262-6036700.post@n2.nabble.com>
	<AANLkTikmEvZB5tF6qiT2Jf4Ca6VzBB8bqQ_k=O59=eO=@mail.gmail.com>
Message-ID: <1298030300075-6039655.post@n2.nabble.com>


Hi,

I have pasted the sip trace log in pastebin url
http://pastebin.freeswitch.org/15417.

Please help me resolve this problem.

Thanks,
Rex
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/The-call-dropped-prematurely-error-on-txfax-tp6032194p6039655.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From michelhabib at gmail.com  Fri Feb 18 13:42:09 2011
From: michelhabib at gmail.com (Michel Habib)
Date: Fri, 18 Feb 2011 12:42:09 +0200
Subject: [Freeswitch-users] ASR from Freeswitch to Loquendo - No Audio
	passed to MRCP Server
Message-ID: <AANLkTiknFyESH2kuj2erYAEp1tKN6eoDRrFGcvQ-drvC@mail.gmail.com>

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

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

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

Thank you,
Michel.
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From vetali100 at gmail.com  Fri Feb 18 15:50:47 2011
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Fri, 18 Feb 2011 14:50:47 +0200
Subject: [Freeswitch-users]  Which CPU to select
Message-ID: <AANLkTi=s-CbsTv6-J6BJKHxTWZWQnP6XcWdHD-nHh6gX@mail.gmail.com>

Hello list,

Could you please suggest, based on your experience, which of the CPU will be
enough to handle 1000 simultaneous calls using G711 codec on all end points?

Is it really needed to buy Xeon with 4 cores? or Core 2 Duo or Quad will be
perfectly enough? Assuming CPU frequency is ~2.5 Ghz

The following page seems like not too  much updated:
http://wiki.freeswitch.org/wiki/Real-world_results


Thank you,
Vitalie
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From cmrienzo at gmail.com  Fri Feb 18 17:16:41 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Fri, 18 Feb 2011 09:16:41 -0500
Subject: [Freeswitch-users] ASR from Freeswitch to Loquendo - No Audio
 passed to MRCP Server
In-Reply-To: <AANLkTiknFyESH2kuj2erYAEp1tKN6eoDRrFGcvQ-drvC@mail.gmail.com>
References: <AANLkTiknFyESH2kuj2erYAEp1tKN6eoDRrFGcvQ-drvC@mail.gmail.com>
Message-ID: <AANLkTimfzq36csrvpTJ9umyX01eD4OsCFs74GfV4aN5v@mail.gmail.com>

Show me the lua script.


On Fri, Feb 18, 2011 at 5:42 AM, Michel Habib <michelhabib at gmail.com> wrote:

> Dears, I am seeking expert advice on setting up Freeswitch to Loquendo MRCP
> Connection.
> I have freeswitch on one computer, configured for MRCP V1 - Loquendo Speech
> Suite 7.
> I have Loquendo set-up on the other computer - with MRCP V1 Profile.
> Using a simple lua script driving my extension, I can successfully:-
> - establish the connection to Loquendo Server.
> - define grammar
> - compile grammar successfully.
> - Begin Recognition
> Then it waits forever until i hangup the call.
>
> Looking at the Wireshark Traces, the RTP packets are transferred
> succesfully from softphone to freeswitch [i could hear my voice on the
> playback]
> But, when playing back the RTP packets between freeswitch and Loquendo,
> there is only deep silence [same thing with audio dump created by Loquendo].
>
> Can you direct me where to troubleshoot this on the Freeswitch side? I will
> be more than happy to provide any logs you need to resolve this.
>
> Thank you,
> Michel.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Feb 18 17:41:00 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 18 Feb 2011 08:41:00 -0600
Subject: [Freeswitch-users] big audio delay in bridged conference
In-Reply-To: <AANLkTikNn40pM56bYb6okRipL=0tivvUEEy7apSLEbkn@mail.gmail.com>
References: <AANLkTikNn40pM56bYb6okRipL=0tivvUEEy7apSLEbkn@mail.gmail.com>
Message-ID: <AANLkTinh-EAyoyjMY+RCVYS1XGoME0uMRsy4UKWh09LR@mail.gmail.com>

try updating again, there are 3 or 4 more commits to mod_loopback
since that one.


On Thu, Feb 17, 2011 at 4:37 PM, George Niculae
<george.niculae79 at gmail.com> wrote:
> ?Hi All,
>
> I have two conferences set in my dialplan - one is bridged and is using
> mod_loopback (as advised in http://jira.freeswitch.org/browse/FS-2741 ) the
> other is a normal one. Extensions in dialplan looks like:
>
> ?<extension name="3202">
> ??? <condition field="destination_number" expression="^202-conference$">
> ????? <action application="bridge"
> data="loopback/app=conference:202-conference at 3202"/>
> ??? </condition>
> ? </extension>
> ? <extension name="3203">
> ??? <condition field="destination_number" expression="^203-conference$">
> ????? <action application="conference" data="203-conference at 3203"/>
> ??? </condition>
> ? </extension>
>
> With the bridged conference I hit weird behaviour: there is a delay in audio
> that grows in time and can even touch 9 seconds in about 10 minutes. The
> other conference works just fine tested in same conditions as the first one.
> I have monitored FS console during both conferences but noticed nothing
> special. (running HEAD, commit 49a5effcdf2cea9e0ddcf146cf3fe85d1872e654)
>
> Any pointer greatly appreciated,
>
> 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


From anthony.minessale at gmail.com  Fri Feb 18 18:02:56 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 18 Feb 2011 09:02:56 -0600
Subject: [Freeswitch-users] Which CPU to select
In-Reply-To: <AANLkTi=s-CbsTv6-J6BJKHxTWZWQnP6XcWdHD-nHh6gX@mail.gmail.com>
References: <AANLkTi=s-CbsTv6-J6BJKHxTWZWQnP6XcWdHD-nHh6gX@mail.gmail.com>
Message-ID: <AANLkTikgWwmN+-hANt0kRwX02y0z5HD7WJfxET4dDGUg@mail.gmail.com>

You want the most recent, well-priced INTEL Xeon with the most cores
on a superb motherboard.
INTEL shows you what to buy with their prices.


On Fri, Feb 18, 2011 at 6:50 AM, Vitalii Colosov <vetali100 at gmail.com> wrote:
> Hello list,
> Could you please suggest, based on your experience, which of the CPU will be
> enough to handle 1000 simultaneous calls using G711 codec on all end points?
> Is it really needed to buy Xeon with 4 cores? or Core 2 Duo or Quad will be
> perfectly enough? Assuming CPU frequency is ~2.5 Ghz
> The following page seems like not too ?much updated:
> http://wiki.freeswitch.org/wiki/Real-world_results
>
> Thank you,
> Vitalie
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb 18 18:36:49 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 18 Feb 2011 09:36:49 -0600
Subject: [Freeswitch-users] The call dropped prematurely error on txfax
In-Reply-To: <1298030300075-6039655.post@n2.nabble.com>
References: <1297873355400-6032435.post@n2.nabble.com>
	<AANLkTingYEfo=4oaANLS8ZzySqLSaUOOm7ME=HpxxQ-C@mail.gmail.com>
	<1297885773534-6033268.post@n2.nabble.com>
	<AANLkTiku+q9B_FeSErvMuXSog6DQ4mMtDmxn_TjtG9Zw@mail.gmail.com>
	<1297954260038-6036197.post@n2.nabble.com>
	<140D740BF92640F7A514BD35BD4BD1AA@e1705>
	<1297958857213-6036503.post@n2.nabble.com>
	<C0AB523AE4774FA5837E88F48243F970@e1705>
	<1297961826262-6036700.post@n2.nabble.com>
	<AANLkTikmEvZB5tF6qiT2Jf4Ca6VzBB8bqQ_k=O59=eO=@mail.gmail.com>
	<1298030300075-6039655.post@n2.nabble.com>
Message-ID: <AANLkTinTX+vJj8qibEVsfyrSMVK=jPHbYksPkD20O=Z_@mail.gmail.com>

They appear to not support you doing the re-invite, they are ignoring
yours then they try to send you one while you are sending them one.
use fax_enable_t38_request=false instead of true when using this
particular destination and allow them to do the re-invite.


On Fri, Feb 18, 2011 at 5:58 AM, rex.alex <rajesh.npnr at yahoo.com> wrote:
>
> Hi,
>
> I have pasted the sip trace log in pastebin url
> http://pastebin.freeswitch.org/15417.
>
> Please help me resolve this problem.
>
> Thanks,
> Rex
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/The-call-dropped-prematurely-error-on-txfax-tp6032194p6039655.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 wstephen80 at gmail.com  Fri Feb 18 19:25:32 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Fri, 18 Feb 2011 17:25:32 +0100
Subject: [Freeswitch-users] ESL socket outbound in C++
Message-ID: <AANLkTi=+LAue0ZEiJ2zGg4Kx_KBBSrcXEyKMunw9m=kX@mail.gmail.com>

I'm learning to use ESL in C++ to do a call control and I want to share my
code with the community.

I had a little difficult at startup so I hope that this code can help
someone.

The code is: http://pastebin.freeswitch.org/15418

Using "esl_oop" module, I have implemented a basic ESL Channel class (that
can be extended) so the "main" event handling loop can be very simple, for
example for an unuseful esl bridge application, can be:

    ESLchannel legA(&connection, inbound_uuid);

    ESLchannel legB(&connection);
    legB.Originate(dialstring)

    while (1)
    {
        ESLevent * event = connection.recvEventTimed(1000);

        if (event)
        {
            const char * ename = event->getHeader("Event-Name");
            const char * event_uuid = event->getHeader("unique-id");

            // It's an event for LegB ?
            //-------------------------
            if (legB.Uuid() == event_uuid)
            {
                if (!strcmp(ename, "CHANNEL_PROGRESS_MEDIA"))
                {
                    legA.PreAnswer();
                    legA.BridgeTo(legB);
                }
                else if (!strcmp(ename, "CHANNEL_ANSWER"))
                {
                    legA.Answer();
                    legA.BridgeTo(legB);
                }
                else if (!strcmp(ename,"CHANNEL_HANGUP"))
                {
                    const char * cause = event->getHeader("Hangup-Cause");

                    legA.Hangup(cause);
                }
            }
            // It's an event for LegA ?
            //-------------------------
            else if (legA.Uuid() == event_uuid)
            {
                if (!strcmp(ename,"CHANNEL_HANGUP"))
                {
                    const char * cause = event->getHeader("Hangup-Cause");

                    legB.Hangup(cause);
                }
            }

            delete event;
        }
    }
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From lloydie.t at gmail.com  Fri Feb 18 20:48:28 2011
From: lloydie.t at gmail.com (lloyd thomas)
Date: Fri, 18 Feb 2011 17:48:28 +0000
Subject: [Freeswitch-users] FS on Public IP adrress
Message-ID: <AANLkTin57u9RgQQdB5bd140bmeP0=rHsWfC0=f4Dr2+d@mail.gmail.com>

I have put my FS box on a public IP address so that it can serve phones at
remote sites. I have not set up any providers on it yet.
I would like to make an internal call between two extensions, but I get the
following error when I try to call

freeswitch at internal> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV
HEADER [Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Content-Length]
= [121]
[DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-Level] =
[7]
[DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Text-Channel] =
[0]
[DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-File] =
[sofia.c]
[DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-Func] =
[sofia_handle_sip_i_invite]
[DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-Line] =
[6081]
[DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [User-Data] = []
[DEBUG] libs/esl/src/esl.c:1111 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 121
Log-Level: 7
Text-Channel: 0
Log-File: sofia.c
Log-Func: sofia_handle_sip_i_invite
Log-Line: 6081
User-Data: _undef_

2011-02-18 17:41:40.638553 [DEBUG] sofia.c:6081 IP 188.221.84.33 Rejected by
acl "domains". Falling back to Digest auth.

2011-02-18 17:41:40.638553 [DEBUG] sofia.c:6081 IP 188.221.84.33 Rejected by
acl "domains". Falling back to Digest auth.


I am using two polycom 550's to test this.
Any Ideas?

Thanks Lloydie T
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From lloyd.aloysius at gmail.com  Fri Feb 18 22:03:33 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 18 Feb 2011 14:03:33 -0500
Subject: [Freeswitch-users] FS on Public IP adrress
In-Reply-To: <AANLkTin57u9RgQQdB5bd140bmeP0=rHsWfC0=f4Dr2+d@mail.gmail.com>
References: <AANLkTin57u9RgQQdB5bd140bmeP0=rHsWfC0=f4Dr2+d@mail.gmail.com>
Message-ID: <AANLkTi===9TMHYPc-w3TdxyL7ffK18UfEcAMpXY+gy-g@mail.gmail.com>

1. Can you dial 9664 - MOH

2. Two Polycom Phones Behind the NAT

Did you enable the  <param name="NDLB-force-rport" value="safe"/> in the
internal.xml

I find for two or more polycom phones behind the NAT .... you need to enable
the above.

Can you post your registration status.

Thanks
Lloyd


On Fri, Feb 18, 2011 at 12:48 PM, lloyd thomas <lloydie.t at gmail.com> wrote:

> I have put my FS box on a public IP address so that it can serve phones at
> remote sites. I have not set up any providers on it yet.
> I would like to make an internal call between two extensions, but I get the
> following error when I try to call
>
> freeswitch at internal> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV
> HEADER [Content-Type] = [log/data]
> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER
> [Content-Length] = [121]
> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Text-Channel]
> = [0]
> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-File] =
> [sofia.c]
> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-Func] =
> [sofia_handle_sip_i_invite]
> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-Line] =
> [6081]
> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [User-Data] =
> []
> [DEBUG] libs/esl/src/esl.c:1111 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 121
> Log-Level: 7
> Text-Channel: 0
> Log-File: sofia.c
> Log-Func: sofia_handle_sip_i_invite
> Log-Line: 6081
> User-Data: _undef_
>
> 2011-02-18 17:41:40.638553 [DEBUG] sofia.c:6081 IP 188.221.84.33 Rejected
> by acl "domains". Falling back to Digest auth.
>
> 2011-02-18 17:41:40.638553 [DEBUG] sofia.c:6081 IP 188.221.84.33 Rejected
> by acl "domains". Falling back to Digest auth.
>
>
> I am using two polycom 550's to test this.
> Any Ideas?
>
> Thanks Lloydie T
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 18 22:27:49 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 18 Feb 2011 19:27:49 +0000
Subject: [Freeswitch-users] FS on Public IP adrress
In-Reply-To: <AANLkTin57u9RgQQdB5bd140bmeP0=rHsWfC0=f4Dr2+d@mail.gmail.com>
References: <AANLkTin57u9RgQQdB5bd140bmeP0=rHsWfC0=f4Dr2+d@mail.gmail.com>
Message-ID: <AANLkTimndpPn7d8jLz4B2m2oq8ZHWyT5WRYJy=9dmOBo@mail.gmail.com>

What's the error?

"Rejected by acl "domains". Falling back to Digest auth."?

That means that the IP wasn't in the ACL, so it's going to do a Digest
password authentication. It rejects the 1st INVITE with a 407 which contains
a challenge which allows the client to send another INVITE with the digest
of the password to authenticate by password.

If you're authenticating by password, then that message is completely
normal.

How do you want to authenticate the phones? By IP or by password?

-Steve



On 18 February 2011 17:48, lloyd thomas <lloydie.t at gmail.com> wrote:

> I have put my FS box on a public IP address so that it can serve phones at
> remote sites. I have not set up any providers on it yet.
> I would like to make an internal call between two extensions, but I get the
> following error when I try to call
>
> freeswitch at internal> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV
> HEADER [Content-Type] = [log/data]
> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER
> [Content-Length] = [121]
> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Text-Channel]
> = [0]
> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-File] =
> [sofia.c]
> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-Func] =
> [sofia_handle_sip_i_invite]
> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-Line] =
> [6081]
> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [User-Data] =
> []
> [DEBUG] libs/esl/src/esl.c:1111 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 121
> Log-Level: 7
> Text-Channel: 0
> Log-File: sofia.c
> Log-Func: sofia_handle_sip_i_invite
> Log-Line: 6081
> User-Data: _undef_
>
> 2011-02-18 17:41:40.638553 [DEBUG] sofia.c:6081 IP 188.221.84.33 Rejected
> by acl "domains". Falling back to Digest auth.
>
> 2011-02-18 17:41:40.638553 [DEBUG] sofia.c:6081 IP 188.221.84.33 Rejected
> by acl "domains". Falling back to Digest auth.
>
>
> I am using two polycom 550's to test this.
> Any Ideas?
>
> Thanks Lloydie T
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From george.niculae79 at gmail.com  Fri Feb 18 23:13:47 2011
From: george.niculae79 at gmail.com (George Niculae)
Date: Fri, 18 Feb 2011 22:13:47 +0200
Subject: [Freeswitch-users] big audio delay in bridged conference
In-Reply-To: <AANLkTinh-EAyoyjMY+RCVYS1XGoME0uMRsy4UKWh09LR@mail.gmail.com>
References: <AANLkTikNn40pM56bYb6okRipL=0tivvUEEy7apSLEbkn@mail.gmail.com>
	<AANLkTinh-EAyoyjMY+RCVYS1XGoME0uMRsy4UKWh09LR@mail.gmail.com>
Message-ID: <AANLkTikzNRsYxBJ43N6fomhd=NkHXsY2vqzwRV2a=F4Y@mail.gmail.com>

Updating to latest revision fixed it, thanks so much!

George


2011/2/18 Anthony Minessale <anthony.minessale at gmail.com>

> try updating again, there are 3 or 4 more commits to mod_loopback
> since that one.
>
>
> On Thu, Feb 17, 2011 at 4:37 PM, George Niculae
> <george.niculae79 at gmail.com> wrote:
> >  Hi All,
> >
> > I have two conferences set in my dialplan - one is bridged and is using
> > mod_loopback (as advised in http://jira.freeswitch.org/browse/FS-2741 )
> the
> > other is a normal one. Extensions in dialplan looks like:
> >
> >  <extension name="3202">
> >     <condition field="destination_number" expression="^202-conference$">
> >       <action application="bridge"
> > data="loopback/app=conference:202-conference at 3202"/>
> >     </condition>
> >   </extension>
> >   <extension name="3203">
> >     <condition field="destination_number" expression="^203-conference$">
> >       <action application="conference" data="203-conference at 3203"/>
> >     </condition>
> >   </extension>
> >
> > With the bridged conference I hit weird behaviour: there is a delay in
> audio
> > that grows in time and can even touch 9 seconds in about 10 minutes. The
> > other conference works just fine tested in same conditions as the first
> one.
> > I have monitored FS console during both conferences but noticed nothing
> > special. (running HEAD, commit 49a5effcdf2cea9e0ddcf146cf3fe85d1872e654)
> >
> > Any pointer greatly appreciated,
> >
> > 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
>
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From w8hdkim at gmail.com  Fri Feb 18 23:17:21 2011
From: w8hdkim at gmail.com (Kim Culhan)
Date: Fri, 18 Feb 2011 15:17:21 -0500
Subject: [Freeswitch-users] Build on solaris fails in libs/esl
Message-ID: <AANLkTimGtXWdRPAv8OJYWeu8VYtCF8ZfQt0osGVgU7G=@mail.gmail.com>

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

--
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From k-b-24 at live.com  Fri Feb 18 23:35:49 2011
From: k-b-24 at live.com (Jason b.a)
Date: Fri, 18 Feb 2011 20:35:49 +0000
Subject: [Freeswitch-users] IVR application
Message-ID: <COL111-W38C088A856EBAA7D204556ACD40@phx.gbl>


i am not going to use the embedded dialplan,  do the application interface has  SIP API such as  " onInvite " for example , or no !  		 	   		  
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From lloyd.aloysius at gmail.com  Fri Feb 18 23:43:21 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 18 Feb 2011 15:43:21 -0500
Subject: [Freeswitch-users] Aastra phone registration lost
In-Reply-To: <AANLkTik5pgWLDqg3Q2FDsc3i+NJ9vo77VGjrX7v-2bK6@mail.gmail.com>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>
	<4D52B987.7050002@communicatefreely.net>
	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>
	<4D534AFD.3020401@communicatefreely.net>
	<AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>
	<60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>
	<AANLkTimux4MBM1K_7itE4GnZQHjy_vR2Y6sr_z+u3gkR@mail.gmail.com>
	<4D5577CE.1000603@communicatefreely.net>
	<AANLkTinJK+e9hZJbk4ur_OQwGV5B7tRLrpPMH=TU5MOo@mail.gmail.com>
	<4D59C085.9040008@communicatefreely.net>
	<AANLkTikZbWPqf7N1fT-9fF8Fvydi85mJVZ+o8_J=RLjE@mail.gmail.com>
	<4D5A9BEA.7070302@communicatefreely.net>
	<AANLkTikuozcozSiFFL8-6+s3iRP4Lfy9DT2xFi9TVBOd@mail.gmail.com>
	<16242449-AADE-4068-8523-C1727F66FDF8@freeswitch.org>
	<4D5BF2DC.20409@communicatefreely.net>
	<AANLkTik5pgWLDqg3Q2FDsc3i+NJ9vo77VGjrX7v-2bK6@mail.gmail.com>
Message-ID: <AANLkTi=6xtWFSNV3qOHiW9Sedy8xq2FiO7g03qEX0qSk@mail.gmail.com>

I did some traces and figure it out the problem losing the Registrations.

1. FreeSWITCH Send the response to a wrong port . Is this a Bug?

2. Why FreeSWITCH + Aastra Registration contact not have the port number ?

3. See the below ...

203 & 202 Aastra Phones on Domain - *aastra.mydomain.com*

But user Agent Shows Polycom. BTW in the same LAN I have two polycom phones
on different domain *foo.mydomain.com*

==============

SIP/2.0 200 OK
Via: SIP/2.0/UDP 173.230.136.12;rport;branch=z9hG4bKZ6DDSDNZ11gvS
*From: <sip:203 at aastra.mydomain.com>;tag=3jFNS2KBvNX0S*
*To: "Ext 202" <sip:203 at aastra.mydomain.com>;tag=B550915E-F491D63B*
CSeq: 8639912 NOTIFY
Call-ID: cdad5068-b542-122e-ef86-fefdade6880c
*Contact: <sip:202 at 192.168.240.128:5060>*
Event: message-summary
*User-Agent: PolycomSoundPointIP-SPIP_450-UA/3.3.1.0769*
Accept-Language: en
Content-Length: 0

=============

Any help on this why sending wrong ports?

Thanks
Lloyd
On Wed, Feb 16, 2011 at 10:34 PM, Aloysius Lloyd
<lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
>
> I stay away with Aastra phone for a long time  and today I did some tests.
> All of my test ... the phones not reliable with FreeSWITCH
>
> I try both 6731i and 57i with the most recent firmware
>
> Here is the configuration files
>
> *aastra.cfg*
>
> dhcp: 1
> sip digit timeout: 3
> sip dial plan:
> "x+#|xx+*|[2-9]XX[2-9]XXXXXX|1[2-9]XX[2-9]XXXXXX|1XXXXXXXXXX|[2-3]XX|67[2-9]XX[2-9]XXXXXX"
> sip rport: 1
> sip customized
> codec:payload=9;ptime=20;silsupp=off,payload=0;ptime=20;silsupp=off,payload=18;ptime=20;silsupp=off
> #sip registration period: 120
> #sip registration renewal timer: 15
> headset tx gain: -3
> headset sidetone gain: -3
> handset tx gain: -3
> handset sidetone gain: -3
> handsfree tx gain: 0
> handset volume: 5     #RX volumes - user adjustable, so easily changed
> speaker volume: 5
> ringer volume: 3
> web interface enabled: 1
> live dialpad: 1
> missed calls indicator disabled: 1
> suppress dtmf playback: 0
> #audio mode: 2 #0 = speaker (default)1 = headset 2 = speaker/headset 3 =
> headset/speaker
> time server disabled: 0
> time server1: pool.ntp.org
> #directory
> directory 1: internal_list.csv
> directory 2: external_list.csv
>
> *mac.cfg*
>
> directed call pickup: 1
> directed call pickup prefix: **
> #
> sip line1 screen name: Ext 203
> sip line1 display name: Ext 203
> sip line1 auth name: 203
> sip line1 user name: 203
> sip line1 password: *********
> sip line1 vmail: *97
> sip line1 mode: 0
> #
> sip line1 proxy ip: aastra.mydomain.com
> #sip line1 proxy port: 5060
> sip line1 registrar ip: aastra.mydomain.com
> sip line1 registration period: 300
> #sip line1 registrar port: 5060
>
> ------------
>
> 1. FreeSWITCH Registration shows two entires in the internal profile and
> when I try to call to the extension two lines on the phones rings .... no
> idea why this is happen?
>
> Call-ID:        d8fd4c8795801cd2
> User:           203 at aastra.mydomain.com
> Contact:        "Ext 203" <sip:203 at 192.168.240.106:5060
> ;transport=udp;fs_nat=yes;fs_path=sip%3A203%40173.33.178.49%3A1627%3Btransport%3Dudp>
> Agent:          Aastra 6731i/2.6.0.2010
> Status:         Registered(UDP-NAT)(unknown) EXP(2011-02-16 22:24:36)
> EXPSECS(310)
> Host:           li176-12
> IP:             173.33.178.49
> Port:           1627
> Auth-User:      203
> Auth-Realm:     aastra.mydomain.com
> MWI-Account:    203 at aastra.mydomain.com
>
> Call-ID:        d8fd4c8795801cd2
> User:           203 at aastra.mydomain.com
> Contact:        "Ext 203" <sip:203 at 173.33.178.49:1627;transport=udp>
> Agent:          Aastra 6731i/2.6.0.2010
> Status:         Registered(UDP)(unknown) EXP(2011-02-16 22:24:36)
> EXPSECS(310)
> Host:           li176-12
> IP:             173.33.178.49
> Port:           1627
> Auth-User:      203
> Auth-Realm:     aastra.mydomain.com
> MWI-Account:    203 at aastra.mydomain.com
>
>
> 2. After first registration expires .... FreeSWITCH internal registration
> status shows the following entry. When I dial the extension now there is a
> long delay ... FreeSWITCH dialing and waiting then goes to voicemail.
>
> Call-ID:        d8fd4c8795801cd2
> User:           203 at aastra.mydomain.com
> Contact:        "Ext 203" <sip:203 at 173.33.178.49:1627;transport=udp>
> Agent:          Aastra 6731i/2.6.0.2010
> Status:         Registered(UDP)(unknown) EXP(2011-02-16 22:31:25)
> EXPSECS(266)
> Host:           li176-12
> IP:             173.33.178.49
> Port:           1631
> Auth-User:      203
> Auth-Realm:     aastra.mydomain.com
> MWI-Account:    203 at aastra.mydomain.com
>
> 3. Phones Goes randomly "No Service" .... I see this problem several times
> in the past.
>
> -------------
>
> At the same time in my lab Linksys and Polycom Phones working without any
> issues with the default settings.
>
> -------------
>
> my question what is Aastra Doing differently?
>
> Thanks
> Lloyd
>
> On Wed, Feb 16, 2011 at 10:53 AM, Tim St. Pierre <
> fs-list at communicatefreely.net> wrote:
>
>> I have been setting my expires to 600.
>>
>> Brian West wrote:
>> > Are you setting your expires to > 300 seconds?
>> >
>> > /b
>> >
>> > On Feb 15, 2011, at 9:42 AM, Aloysius Lloyd wrote:
>> >
>> >>
>> >>
>> >> I have the same issue, around 275 phones in the field. I want the 275
>> >> phones work with FreeSWITCH.
>> >>
>> >>
>> >> 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
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From lloyd.aloysius at sunteltech.ca  Fri Feb 18 23:49:54 2011
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Fri, 18 Feb 2011 15:49:54 -0500
Subject: [Freeswitch-users] FS on Public IP adrress
In-Reply-To: <AANLkTimndpPn7d8jLz4B2m2oq8ZHWyT5WRYJy=9dmOBo@mail.gmail.com>
References: <AANLkTin57u9RgQQdB5bd140bmeP0=rHsWfC0=f4Dr2+d@mail.gmail.com>
	<AANLkTimndpPn7d8jLz4B2m2oq8ZHWyT5WRYJy=9dmOBo@mail.gmail.com>
Message-ID: <AANLkTi=3m2XEkEMkPFP+Nj58rsA2d7YqCZb6n7ac5cCZ@mail.gmail.com>

Lloyd Thomas,

I just noticed you reply me directly. Please post the answer to the mailing
list. This will help someone in the future.

===========

Call-ID:        2d9a6489-7762dc9c-a2fb31a7 at 192.168.101.11
User:           1000 at 188.222.85.33
Contact:        "user" <sip:1000 at 192.168.101.11:5060>
Agent:          PolycomSoundPointIP-SPIP_550-UA/3.2.3.1734
Status:         Registered(UDP)(unknown) EXP(2011-02-18 21:33:10)
EXPSECS(6838)
Host:           phiFS1
IP:             188.222.85.32
Port:           38532
Auth-User:      1000
Auth-Realm:     188.222.85.33
MWI-Account:    1000 at 188.222.85.33

Call-ID:        70202574-4ecbe83f-71727f5e at 192.168.101.12
User:           1001 at 188.222.85.33
Contact:        "user" <sip:1001 at 192.168.101.12:5060>
Agent:          PolycomSoundPointIP-SPIP_550-UA/3.2.3.1734
Status:         Registered(UDP)(unknown) EXP(2011-02-18 21:35:24)
EXPSECS(6972)
Host:           phiFS1
IP:             188.222.85.32
Port:           38596
Auth-User:      1001
Auth-Realm:     188.222.85.33
MWI-Account:    1001 at 188.222.85.33

Total items returned: 2

=================

Here is the Solution for the problem...


1. Add the following to the user 1000.xml and 1001.xml

<param name="sip-force-contact" value="NDLB-connectile-dysfunction"/>

2.sip_profiles/internal.xml make sure the following line enable.

<param name="NDLB-force-rport" value="safe"/>

3. Reload the profiles / Reload settings

4. Reboot the phone

---

in your registration the following not correct.  see both NAT show 5060

Contact:        "user" <sip:1000 at 192.168.101.11:5060>

Contact:        "user" <sip:1001 at 192.168.101.12:5060>


Thanks
Lloyd
On Fri, Feb 18, 2011 at 2:27 PM, Steven Ayre <steveayre at gmail.com> wrote:

> What's the error?
>
> "Rejected by acl "domains". Falling back to Digest auth."?
>
> That means that the IP wasn't in the ACL, so it's going to do a Digest
> password authentication. It rejects the 1st INVITE with a 407 which contains
> a challenge which allows the client to send another INVITE with the digest
> of the password to authenticate by password.
>
> If you're authenticating by password, then that message is completely
> normal.
>
> How do you want to authenticate the phones? By IP or by password?
>
> -Steve
>
>
>
> On 18 February 2011 17:48, lloyd thomas <lloydie.t at gmail.com> wrote:
>
>> I have put my FS box on a public IP address so that it can serve phones at
>> remote sites. I have not set up any providers on it yet.
>> I would like to make an internal call between two extensions, but I get
>> the following error when I try to call
>>
>> freeswitch at internal> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV
>> HEADER [Content-Type] = [log/data]
>> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER
>> [Content-Length] = [121]
>> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-Level] =
>> [7]
>> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Text-Channel]
>> = [0]
>> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-File] =
>> [sofia.c]
>> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-Func] =
>> [sofia_handle_sip_i_invite]
>> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-Line] =
>> [6081]
>> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [User-Data] =
>> []
>> [DEBUG] libs/esl/src/esl.c:1111 esl_recv_event() RECV MESSAGE
>> Event-Name: SOCKET_DATA
>> Content-Type: log/data
>> Content-Length: 121
>> Log-Level: 7
>> Text-Channel: 0
>> Log-File: sofia.c
>> Log-Func: sofia_handle_sip_i_invite
>> Log-Line: 6081
>> User-Data: _undef_
>>
>> 2011-02-18 17:41:40.638553 [DEBUG] sofia.c:6081 IP 188.221.84.33 Rejected
>> by acl "domains". Falling back to Digest auth.
>>
>> 2011-02-18 17:41:40.638553 [DEBUG] sofia.c:6081 IP 188.221.84.33 Rejected
>> by acl "domains". Falling back to Digest auth.
>>
>>
>> I am using two polycom 550's to test this.
>> Any Ideas?
>>
>> Thanks Lloydie T
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloydie.t at gmail.com  Sat Feb 19 00:58:37 2011
From: lloydie.t at gmail.com (lloyd thomas)
Date: Fri, 18 Feb 2011 21:58:37 +0000
Subject: [Freeswitch-users] FS on Public IP adrress
In-Reply-To: <AANLkTi=3m2XEkEMkPFP+Nj58rsA2d7YqCZb6n7ac5cCZ@mail.gmail.com>
References: <AANLkTin57u9RgQQdB5bd140bmeP0=rHsWfC0=f4Dr2+d@mail.gmail.com>
	<AANLkTimndpPn7d8jLz4B2m2oq8ZHWyT5WRYJy=9dmOBo@mail.gmail.com>
	<AANLkTi=3m2XEkEMkPFP+Nj58rsA2d7YqCZb6n7ac5cCZ@mail.gmail.com>
Message-ID: <AANLkTimxN4bNwxtx3Z+G3xwDZzbo26eX+5bH8JZAZZos@mail.gmail.com>

Sorry about that. Just press reply on gmail

On 18 February 2011 20:49, Aloysius Lloyd <lloyd.aloysius at sunteltech.ca>wrote:

> Lloyd Thomas,
>
> I just noticed you reply me directly. Please post the answer to the mailing
> list. This will help someone in the future.
>
> ===========
>
> Call-ID:        2d9a6489-7762dc9c-a2fb31a7 at 192.168.101.11
> User:           1000 at 188.222.85.33
> Contact:        "user" <sip:1000 at 192.168.101.11:5060>
> Agent:          PolycomSoundPointIP-SPIP_550-UA/3.2.3.1734
> Status:         Registered(UDP)(unknown) EXP(2011-02-18 21:33:10)
> EXPSECS(6838)
> Host:           phiFS1
> IP:             188.222.85.32
> Port:           38532
> Auth-User:      1000
> Auth-Realm:     188.222.85.33
> MWI-Account:    1000 at 188.222.85.33
>
> Call-ID:        70202574-4ecbe83f-71727f5e at 192.168.101.12
> User:           1001 at 188.222.85.33
> Contact:        "user" <sip:1001 at 192.168.101.12:5060>
> Agent:          PolycomSoundPointIP-SPIP_550-UA/3.2.3.1734
> Status:         Registered(UDP)(unknown) EXP(2011-02-18 21:35:24)
> EXPSECS(6972)
> Host:           phiFS1
> IP:             188.222.85.32
> Port:           38596
> Auth-User:      1001
> Auth-Realm:     188.222.85.33
> MWI-Account:    1001 at 188.222.85.33
>
> Total items returned: 2
>
> =================
>
> Here is the Solution for the problem...
>
>
> 1. Add the following to the user 1000.xml and 1001.xml
>
> <param name="sip-force-contact" value="NDLB-connectile-dysfunction"/>
>
> 2.sip_profiles/internal.xml make sure the following line enable.
>
> <param name="NDLB-force-rport" value="safe"/>
>
> 3. Reload the profiles / Reload settings
>
> 4. Reboot the phone
>
> ---
>
> in your registration the following not correct.  see both NAT show 5060
>
> Contact:        "user" <sip:1000 at 192.168.101.11:5060>
>
> Contact:        "user" <sip:1001 at 192.168.101.12:5060>
>
>
> Thanks
> Lloyd
>
> On Fri, Feb 18, 2011 at 2:27 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> What's the error?
>>
>> "Rejected by acl "domains". Falling back to Digest auth."?
>>
>> That means that the IP wasn't in the ACL, so it's going to do a Digest
>> password authentication. It rejects the 1st INVITE with a 407 which contains
>> a challenge which allows the client to send another INVITE with the digest
>> of the password to authenticate by password.
>>
>> If you're authenticating by password, then that message is completely
>> normal.
>>
>> How do you want to authenticate the phones? By IP or by password?
>>
>> -Steve
>>
>>
>>
>> On 18 February 2011 17:48, lloyd thomas <lloydie.t at gmail.com> wrote:
>>
>>> I have put my FS box on a public IP address so that it can serve phones
>>> at remote sites. I have not set up any providers on it yet.
>>> I would like to make an internal call between two extensions, but I get
>>> the following error when I try to call
>>>
>>> freeswitch at internal> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event()
>>> RECV HEADER [Content-Type] = [log/data]
>>> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER
>>> [Content-Length] = [121]
>>> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-Level] =
>>> [7]
>>> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER
>>> [Text-Channel] = [0]
>>> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-File] =
>>> [sofia.c]
>>> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-Func] =
>>> [sofia_handle_sip_i_invite]
>>> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-Line] =
>>> [6081]
>>> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [User-Data] =
>>> []
>>> [DEBUG] libs/esl/src/esl.c:1111 esl_recv_event() RECV MESSAGE
>>> Event-Name: SOCKET_DATA
>>> Content-Type: log/data
>>> Content-Length: 121
>>> Log-Level: 7
>>> Text-Channel: 0
>>> Log-File: sofia.c
>>> Log-Func: sofia_handle_sip_i_invite
>>> Log-Line: 6081
>>> User-Data: _undef_
>>>
>>> 2011-02-18 17:41:40.638553 [DEBUG] sofia.c:6081 IP 188.221.84.33 Rejected
>>> by acl "domains". Falling back to Digest auth.
>>>
>>> 2011-02-18 17:41:40.638553 [DEBUG] sofia.c:6081 IP 188.221.84.33 Rejected
>>> by acl "domains". Falling back to Digest auth.
>>>
>>>
>>> I am using two polycom 550's to test this.
>>> Any Ideas?
>>>
>>> Thanks Lloydie T
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fraserredmond at gmail.com  Sat Feb 19 03:15:28 2011
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 18 Feb 2011 19:15:28 -0500
Subject: [Freeswitch-users] loss of audio for 30sec in middle of call
Message-ID: <AANLkTi=M+vs-RN2eA83dT+4h=FxpUPCNo5xupye55OMX@mail.gmail.com>

We've got 10 callers using our FS server (outbound call-center), and they're
reporting that during some calls they lose audio for a while (it varies, but
they've said half a minute is typical).

It's calling from softphone -> FS -> gateway

It's intermittent though - happens to all the users, but only some calls,
and it happens frequently for a while, then goes away for a while.

I can't think where to start looking to find the problem - still trying to
arrange for them to keep a record of affected calls so I can look in the
logs, but I have looked at a couple of examples already with no obvious
problems there. (Besides, the logs are generally quiet during the call
anyway.)

Any suggestions of where to look or what else to try?

In the wiki it talks about some new sofia debug commands, but doesn't
explain them at all - anyone know if they'd help?
http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP

sofia global debug sla
sofia global debug presence

Or do I need to packet-trace everything, then troll through that?

Cheers,
Fraser
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From anthony.minessale at gmail.com  Sat Feb 19 03:41:01 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 18 Feb 2011 18:41:01 -0600
Subject: [Freeswitch-users] loss of audio for 30sec in middle of call
In-Reply-To: <AANLkTi=M+vs-RN2eA83dT+4h=FxpUPCNo5xupye55OMX@mail.gmail.com>
References: <AANLkTi=M+vs-RN2eA83dT+4h=FxpUPCNo5xupye55OMX@mail.gmail.com>
Message-ID: <AANLkTimNZpNqwJfRCFUVyV9fy2yC7wgo4dyGfF=sNRa+@mail.gmail.com>

The only thing that would be relevant is pcaps.

there is a tool that will put all of your calls into individual pcaps

http://wiki.freeswitch.org/wiki/Packet_Capture#pcapsipdump


Most likely its downed route or router rebooting

you could also try pinging or mtr the box from one of the host sites
and look for loss.





On Fri, Feb 18, 2011 at 6:15 PM, Fraser Redmond <fraserredmond at gmail.com> wrote:
> We've got 10 callers using our FS server (outbound call-center), and they're
> reporting that during some calls they lose audio for a while (it varies, but
> they've said half a minute is typical).
>
> It's calling from softphone -> FS -> gateway
>
> It's intermittent though - happens to all the users, but only some calls,
> and it happens frequently for a while, then goes away for a while.
>
> I can't think where to start looking to find the problem - still trying to
> arrange for them to keep a record of affected calls so I can look in the
> logs, but I have looked at a couple of examples already with no obvious
> problems there. (Besides, the logs are generally quiet during the call
> anyway.)
>
> Any suggestions of where to look or what else to try?
>
> In the wiki it talks about some new sofia debug commands, but doesn't
> explain them at all - anyone know if they'd help?
> http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP
>
> sofia global debug sla
> sofia global debug presence
>
> Or do I need to packet-trace everything, then troll through that?
>
> Cheers,
> Fraser
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 hareem.haque at gmail.com  Fri Feb 18 19:47:37 2011
From: hareem.haque at gmail.com (Hareem Haque)
Date: Fri, 18 Feb 2011 11:47:37 -0500
Subject: [Freeswitch-users] Need some help. Novice freeswitch setup gone
	wrong
Message-ID: <AANLkTi=Hj=b5gKm7BdgYCjnh801HVY-0Lwfr__=v76wZ@mail.gmail.com>

I have a little problem i setup the blue.box app it runs very well. Now i am
having some problems.

A. How can i stop using the xml cdr and just use text file only cdr which
are dumped into a folder called /mnt/fscdr

B. How can i setup the pbx as a multi tenant system and keep track of all
the users we have 6 departments with 20 employees each.

C. Can i setup trunks in a way that SIP trunk A can only be used by
Department A.

Any help is greatly appreciated.

Regards
Hareem Haque


Best Regards
Hareem. Haque
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From mitch.johnson7 at gmail.com  Fri Feb 18 20:02:06 2011
From: mitch.johnson7 at gmail.com (Mitch Johnson)
Date: Fri, 18 Feb 2011 12:02:06 -0500
Subject: [Freeswitch-users] CallManager trunk to FS
Message-ID: <6242B672-57F7-4345-ACF6-14DF821055E5@gmail.com>

I'm having a few issues where the freeswitch 1.6 is not registering with the callmanager 8.  

[ERR] sofia_reg c:1765 CM8 Registration Failed with status DNS Error [503].

<include>
  <gateway name="CM8">
  <param name="username" value="xxxx"/>
  <param name="password" value="xxxx"/>
  <param name="realm" value="CM8.localhost.local"/>
  </gateway>
</include>
I added the CM8.localhost.local to the hosts file as the CM is on the same network and doesn't need a DNS entry.  Can I replace the value in the realm field with an IP Address?

I did check the box MTP required on the callmanager. 

I've reloaded the xml file using the sofia profile external ...

Thanks so much,

Mitch



From spencer at 5ninesolutions.com  Fri Feb 18 20:12:34 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Fri, 18 Feb 2011 09:12:34 -0800
Subject: [Freeswitch-users] NAT issue
Message-ID: <1CF42F8F-61A9-47BB-924B-4E466EDDBF6E@5ninesolutions.com>

Hello all,
I'm having a strange NAT problem.  My Freeswitch server has a public  
ip and the endpoints which are Polycom IP 650s are all behind NAT with  
mainly Netgear SRX 5308 routers at multiple locations.  The server is  
on a public class C network and one of the locations has its router on  
the same class C.  I.e x.x.x.7/24-router, x.x.x.240/24-server.  I'm  
only experiencing this with 2 of about 30 extensions.  The first  
endpoint to register from this location gets detected as NAT but the  
others with the ports remapped do not. This maybe explains it better:

Call-ID:    	4feea796-bd96e2ed-e16d27ac at 10.6.71.103
User:       	1103 at x.x.x.240
Contact:    	"user" <sip:1103 at 10.6.71.103;received=x.x.x. 
7:5060;fs_nat=yes;fs_path=sip%3A1103%40x.x.x.7%3A5060>
Agent:      	PolycomSoundPointIP-SPIP_650-UA/3.3.1.0769
Status:     	Registered(UDP-NAT)(unknown) EXP(2011-02-18 11:15:47)  
EXPSECS(1143)
Host:       	pbx10.5ninesolutions.com
IP:         	x.x.x.7
Port:       	5060
Auth-User:  	1103
Auth-Realm: 	pbx10.5ninesolutions.com
MWI-Account:	1103 at x.x.x.x

Call-ID:    	d21b71b2-75baa80d-768c0b08 at 10.6.71.104
User:       	1101 at x.x.x.240
Contact:    	"user" <sip:1101 at 10.6.71.104>
Agent:      	PolycomSoundPointIP-SPIP_650-UA/3.3.1.0769
Status:     	Registered(UDP)(unknown) EXP(2011-02-18 11:16:48)  
EXPSECS(1204)
Host:       	pbx10.5ninesolutions.com
IP:         	x.x.x.7
Port:       	1036
Auth-User:  	1101
Auth-Realm: 	pbx10.5ninesolutions.com
MWI-Account:	1101 at x.x.x.x

Call-ID:    	a74e61a6-1863ea7d-789686c4 at 10.6.71.105
User:       	1102 at x.x.x.240
Contact:    	"user" <sip:1102 at 10.6.71.105>
Agent:      	PolycomSoundPointIP-SPIP_650-UA/3.3.1.0769
Status:     	Registered(UDP)(unknown) EXP(2011-02-18 11:17:23)  
EXPSECS(1239)
Host:       	pbx10.5ninesolutions.com
IP:         	x.x.x.7
Port:       	1036
Auth-User:  	1102
Auth-Realm: 	pbx10.5ninesolutions.com
MWI-Account:	1102 at x.x.x.x

Call-ID:    	100b5148-6a47060a-13c4-50029-c844-22fc4dd3-c844
User:       	1801 at 97.67.110.240
Contact:    	"user" <sip:1801 at 10.6.71.106:5060>
Agent:      	MP201 B 1FXS/3.0.1_build_19
Status:     	Registered(UDP)(unknown) EXP(2011-02-18 11:26:23)  
EXPSECS(1779)
Host:       	pbx10.5ninesolutions.com
IP:         	97.67.110.7
Port:       	1036
Auth-User:  	1801
Auth-Realm: 	pbx10.5ninesolutions.com
MWI-Account:	1801 at x.x.x.x

-------  Another Location ----

Call-ID:    	dca2a276-4eb94f97-c55177b0 at 10.1.10.186
User:       	4204 at x.x.x.240
Contact:    	"user" <sip:4204 at 10.1.10.186;received=x.x.x.x: 
5060;fs_nat=yes;fs_path=sip%3A4204%40x.x.x.x%3A5060>
Agent:      	PolycomSoundStationIP-SSIP_5000-UA/3.3.1.0769
Status:     	Registered(UDP-NAT)(unknown) EXP(2011-02-18 11:28:04)  
EXPSECS(1880)
Host:       	pbx10.5ninesolutions.com
IP:         	x.x.x.x
Port:       	5060
Auth-User:  	4204
Auth-Realm: 	pbx10.5ninesolutions.com
MWI-Account:	4204 at x.x.x.x

The port and IP are detected correctly but the contact is a private  
IP.  Any clue as to why the two extension 1101 and 1102 are any  
different?

Thanks,
Spencer







From sunwood360 at gmail.com  Sat Feb 19 02:14:52 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Fri, 18 Feb 2011 15:14:52 -0800
Subject: [Freeswitch-users] flite build error
Message-ID: <AANLkTi=220qvzpdnG4OZkMfBVchmgvpBFEzQJ578n29h@mail.gmail.com>

making install mod_flite
making in ...
making in include ...
making in src ...
making in src/audio ...
making in src/utils ...
making in src/regex ...
making in src/hrg ...
making in src/stats ...
making in src/speech ...
making in src/lexicon ...
making in src/synth ...
making in src/wavesynth ...
making in src/cg ...
making in lang ...
making in lang/cmulex ...
making in lang/usenglish ...
making in lang/cmu_us_kal ...
gcc: Internal error: Killed (program cc1)
Please submit a full bug report.
See <file:///usr/share/doc/gcc-4.4/README.Bugs> for instructions.
make[6]: ***
[../../build/armv5tel-linux-gnu/obj/lang/cmu_us_kal/cmu_us_kal_res.o] Error
1
make[5]: *** [../build/armv5tel-linux-gnu/obj/lang/.make_build_dirs] Error 2
make[4]: *** [build/armv5tel-linux-gnu/obj//.make_build_dirs] Error 2
make[3]: ***
[/root/fs/freeswitch-1.0.7/libs/flite-1.3.99/build/libs/libflite_cmu_us_awb.a]
Error 2
make[2]: *** [install] Error 1
make[1]: *** [mod_flite-install] Error 1
make: *** [mod_flite-install] Error 2


my system is debian.

thanks!
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From sunwood360 at gmail.com  Sat Feb 19 02:47:45 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Fri, 18 Feb 2011 15:47:45 -0800
Subject: [Freeswitch-users] flite build error
In-Reply-To: <AANLkTi=220qvzpdnG4OZkMfBVchmgvpBFEzQJ578n29h@mail.gmail.com>
References: <AANLkTi=220qvzpdnG4OZkMfBVchmgvpBFEzQJ578n29h@mail.gmail.com>
Message-ID: <AANLkTimC+sZiJM_Ka0_d_UKKStT-oJTMJS-=0B0vaOFz@mail.gmail.com>

fixed the issue. it was run out of memory. for some reason, Debian didn't
pickup the swap space.



On Fri, Feb 18, 2011 at 3:14 PM, envelopes envelopes
<sunwood360 at gmail.com>wrote:

> making install mod_flite
> making in ...
> making in include ...
> making in src ...
> making in src/audio ...
> making in src/utils ...
> making in src/regex ...
> making in src/hrg ...
> making in src/stats ...
> making in src/speech ...
> making in src/lexicon ...
> making in src/synth ...
> making in src/wavesynth ...
> making in src/cg ...
> making in lang ...
> making in lang/cmulex ...
> making in lang/usenglish ...
> making in lang/cmu_us_kal ...
> gcc: Internal error: Killed (program cc1)
> Please submit a full bug report.
> See <file:///usr/share/doc/gcc-4.4/README.Bugs> for instructions.
> make[6]: ***
> [../../build/armv5tel-linux-gnu/obj/lang/cmu_us_kal/cmu_us_kal_res.o] Error
> 1
> make[5]: *** [../build/armv5tel-linux-gnu/obj/lang/.make_build_dirs] Error
> 2
> make[4]: *** [build/armv5tel-linux-gnu/obj//.make_build_dirs] Error 2
> make[3]: ***
> [/root/fs/freeswitch-1.0.7/libs/flite-1.3.99/build/libs/libflite_cmu_us_awb.a]
> Error 2
> make[2]: *** [install] Error 1
> make[1]: *** [mod_flite-install] Error 1
> make: *** [mod_flite-install] Error 2
>
>
> my system is debian.
>
> thanks!
>
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From msc at freeswitch.org  Sat Feb 19 04:09:05 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 18 Feb 2011 17:09:05 -0800
Subject: [Freeswitch-users] IVR application
In-Reply-To: <COL111-W38C088A856EBAA7D204556ACD40@phx.gbl>
References: <COL111-W38C088A856EBAA7D204556ACD40@phx.gbl>
Message-ID: <AANLkTiku7SS0OgehxB+KvhiC4QDQbxdRarY_e-7cHes=@mail.gmail.com>

Given your limitations I'd say the only choice you have is to use the event
socket. You should probably read up on ESL - the event socket library. It's
available for multiple programming languages.

-MC

On Fri, Feb 18, 2011 at 12:35 PM, Jason b.a <k-b-24 at live.com> wrote:

>  i am not going to use the embedded dialplan,  do the application interface
> has  SIP API such as  " onInvite " for example , or no !
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 19 04:16:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 18 Feb 2011 17:16:36 -0800
Subject: [Freeswitch-users]
	=?iso-8859-1?q?Newbie=B4s_question_about_FreeS?=
	=?iso-8859-1?q?witch=2E=2E=2E?=
In-Reply-To: <AANLkTinf=z4JEp4G2xK_0n+u8v0OhR++GBvPzzbT1Y_-@mail.gmail.com>
References: <AANLkTinf=z4JEp4G2xK_0n+u8v0OhR++GBvPzzbT1Y_-@mail.gmail.com>
Message-ID: <AANLkTi=L4NjstM7th7aXea=N=BM-C66N=3TLCic_QNQz@mail.gmail.com>

Bienvenido a FreeSWITCH!

Yes, there are lots of things you can do. You probably need to consider a
GUI front-end to your FreeSWITCH unless you have experience (or time to
learn) how to configure and maintain FreeSWITCH.

>From my experience, you can probably save some money, but migrating from an
old analog PBX to a new VoIP system will still cost some money. Your old
phones won't be useful on the new system without buying ATA's. You are
better off buying Polycom IP550 or Cisco SPA504G phones. You will also need
a server on which to run your VoIP system, possibly some network cabling in
your office, and most likely some professional assistance with getting
everything set up. So, FreeSWITCH itself is *gratis*, but there is still an
investment in everything else. That said, I'm sure you can get everything
you need for well under $4500.

-MC

On Thu, Feb 17, 2011 at 5:42 PM, Francisco Javier Cintr?n Olgu?n <
fcintrono at gmail.com> wrote:

> Hi, My name is Francisco from M?xico.
>
> Here, in my work we have a very very old panasonic PBX(12 years old). We
> are growing and we need to increase our external lines(from 3 to 4) and our
> internal lines(from 6 to 10). Besides we need voice mail and voice menu too.
>
>
> We asked for a quote to our panasonic dealer. The whole thing cost about
> 4,500 dollars.
>
> My boss just saw a thing called FreeSwitch this morning looking for options
> in Google. He asked my to investigate what this thing called FreeSwitch is
> and if we could save some money using it instead of the panasonic solution.
> So, here I am.
>
> I have some experience as linux sysadmin(we have 1 oracle linux server and
> 1 linux print server) nevertheless I don?t have any idea where and how to
> start this evaluation?
>
>
> Please
> Would you give us a clue where to see If FreeSwitch could work for us?
>
> Thanks for your kind help.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 19 04:27:26 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 18 Feb 2011 17:27:26 -0800
Subject: [Freeswitch-users] ESL socket outbound in C++
In-Reply-To: <AANLkTi=+LAue0ZEiJ2zGg4Kx_KBBSrcXEyKMunw9m=kX@mail.gmail.com>
References: <AANLkTi=+LAue0ZEiJ2zGg4Kx_KBBSrcXEyKMunw9m=kX@mail.gmail.com>
Message-ID: <AANLkTinYwvATpqQn38OX2F4eoXkhCuaCH3i1KWnbjuWF@mail.gmail.com>

We appreciate you sharing! Feel free to add this to the wiki as well.
-MC

On Fri, Feb 18, 2011 at 8:25 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> I'm learning to use ESL in C++ to do a call control and I want to share my
> code with the community.
>
> I had a little difficult at startup so I hope that this code can help
> someone.
>
> The code is: http://pastebin.freeswitch.org/15418
>
> Using "esl_oop" module, I have implemented a basic ESL Channel class (that
> can be extended) so the "main" event handling loop can be very simple, for
> example for an unuseful esl bridge application, can be:
>
>     ESLchannel legA(&connection, inbound_uuid);
>
>     ESLchannel legB(&connection);
>     legB.Originate(dialstring)
>
>     while (1)
>     {
>         ESLevent * event = connection.recvEventTimed(1000);
>
>         if (event)
>         {
>             const char * ename = event->getHeader("Event-Name");
>             const char * event_uuid = event->getHeader("unique-id");
>
>             // It's an event for LegB ?
>             //-------------------------
>             if (legB.Uuid() == event_uuid)
>             {
>                 if (!strcmp(ename, "CHANNEL_PROGRESS_MEDIA"))
>                 {
>                     legA.PreAnswer();
>                     legA.BridgeTo(legB);
>                 }
>                 else if (!strcmp(ename, "CHANNEL_ANSWER"))
>                 {
>                     legA.Answer();
>                     legA.BridgeTo(legB);
>                 }
>                 else if (!strcmp(ename,"CHANNEL_HANGUP"))
>                 {
>                     const char * cause = event->getHeader("Hangup-Cause");
>
>                     legA.Hangup(cause);
>                 }
>             }
>             // It's an event for LegA ?
>             //-------------------------
>             else if (legA.Uuid() == event_uuid)
>             {
>                 if (!strcmp(ename,"CHANNEL_HANGUP"))
>                 {
>                     const char * cause = event->getHeader("Hangup-Cause");
>
>                     legB.Hangup(cause);
>                 }
>             }
>
>             delete event;
>         }
>     }
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at sunteltech.ca  Sat Feb 19 04:28:46 2011
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Fri, 18 Feb 2011 20:28:46 -0500
Subject: [Freeswitch-users] NAT issue
In-Reply-To: <1CF42F8F-61A9-47BB-924B-4E466EDDBF6E@5ninesolutions.com>
References: <1CF42F8F-61A9-47BB-924B-4E466EDDBF6E@5ninesolutions.com>
Message-ID: <AANLkTin0PCdP5+TnA39CjAQ9JwVexUzOyno23nPsbyrh@mail.gmail.com>

May be the Two phones using different firmware.

Try the following parameter in the extension

<param name="sip-force-contact" value="NDLB-connectile-dysfunction"/>

Thanks
Lloyd


On Fri, Feb 18, 2011 at 12:12 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> Hello all,
> I'm having a strange NAT problem.  My Freeswitch server has a public
> ip and the endpoints which are Polycom IP 650s are all behind NAT with
> mainly Netgear SRX 5308 routers at multiple locations.  The server is
> on a public class C network and one of the locations has its router on
> the same class C.  I.e x.x.x.7/24-router, x.x.x.240/24-server.  I'm
> only experiencing this with 2 of about 30 extensions.  The first
> endpoint to register from this location gets detected as NAT but the
> others with the ports remapped do not. This maybe explains it better:
>
> Call-ID:        4feea796-bd96e2ed-e16d27ac at 10.6.71.103
> User:           1103 at x.x.x.240
> Contact:        "user" <sip:1103 at 10.6.71.103;received=x.x.x.
> 7:5060;fs_nat=yes;fs_path=sip%3A1103%40x.x.x.7%3A5060>
> Agent:          PolycomSoundPointIP-SPIP_650-UA/3.3.1.0769
> Status:         Registered(UDP-NAT)(unknown) EXP(2011-02-18 11:15:47)
> EXPSECS(1143)
> Host:           pbx10.5ninesolutions.com
> IP:             x.x.x.7
> Port:           5060
> Auth-User:      1103
> Auth-Realm:     pbx10.5ninesolutions.com
> MWI-Account:    1103 at x.x.x.x
>
> Call-ID:        d21b71b2-75baa80d-768c0b08 at 10.6.71.104
> User:           1101 at x.x.x.240
> Contact:        "user" <sip:1101 at 10.6.71.104>
> Agent:          PolycomSoundPointIP-SPIP_650-UA/3.3.1.0769
> Status:         Registered(UDP)(unknown) EXP(2011-02-18 11:16:48)
> EXPSECS(1204)
> Host:           pbx10.5ninesolutions.com
> IP:             x.x.x.7
> Port:           1036
> Auth-User:      1101
> Auth-Realm:     pbx10.5ninesolutions.com
> MWI-Account:    1101 at x.x.x.x
>
> Call-ID:        a74e61a6-1863ea7d-789686c4 at 10.6.71.105
> User:           1102 at x.x.x.240
> Contact:        "user" <sip:1102 at 10.6.71.105>
> Agent:          PolycomSoundPointIP-SPIP_650-UA/3.3.1.0769
> Status:         Registered(UDP)(unknown) EXP(2011-02-18 11:17:23)
> EXPSECS(1239)
> Host:           pbx10.5ninesolutions.com
> IP:             x.x.x.7
> Port:           1036
> Auth-User:      1102
> Auth-Realm:     pbx10.5ninesolutions.com
> MWI-Account:    1102 at x.x.x.x
>
> Call-ID:        100b5148-6a47060a-13c4-50029-c844-22fc4dd3-c844
> User:           1801 at 97.67.110.240
> Contact:        "user" <sip:1801 at 10.6.71.106:5060>
> Agent:          MP201 B 1FXS/3.0.1_build_19
> Status:         Registered(UDP)(unknown) EXP(2011-02-18 11:26:23)
> EXPSECS(1779)
> Host:           pbx10.5ninesolutions.com
> IP:             97.67.110.7
> Port:           1036
> Auth-User:      1801
> Auth-Realm:     pbx10.5ninesolutions.com
> MWI-Account:    1801 at x.x.x.x
>
> -------  Another Location ----
>
> Call-ID:        dca2a276-4eb94f97-c55177b0 at 10.1.10.186
> User:           4204 at x.x.x.240
> Contact:        "user" <sip:4204 at 10.1.10.186;received=x.x.x.x:
> 5060;fs_nat=yes;fs_path=sip%3A4204%40x.x.x.x%3A5060>
> Agent:          PolycomSoundStationIP-SSIP_5000-UA/3.3.1.0769
> Status:         Registered(UDP-NAT)(unknown) EXP(2011-02-18 11:28:04)
> EXPSECS(1880)
> Host:           pbx10.5ninesolutions.com
> IP:             x.x.x.x
> Port:           5060
> Auth-User:      4204
> Auth-Realm:     pbx10.5ninesolutions.com
> MWI-Account:    4204 at x.x.x.x
>
> The port and IP are detected correctly but the contact is a private
> IP.  Any clue as to why the two extension 1101 and 1102 are any
> different?
>
> Thanks,
> 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
>
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From msc at freeswitch.org  Sat Feb 19 04:29:33 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 18 Feb 2011 17:29:33 -0800
Subject: [Freeswitch-users] Aastra phone registration lost
In-Reply-To: <AANLkTi=6xtWFSNV3qOHiW9Sedy8xq2FiO7g03qEX0qSk@mail.gmail.com>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>
	<4D52B987.7050002@communicatefreely.net>
	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>
	<4D534AFD.3020401@communicatefreely.net>
	<AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>
	<60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>
	<AANLkTimux4MBM1K_7itE4GnZQHjy_vR2Y6sr_z+u3gkR@mail.gmail.com>
	<4D5577CE.1000603@communicatefreely.net>
	<AANLkTinJK+e9hZJbk4ur_OQwGV5B7tRLrpPMH=TU5MOo@mail.gmail.com>
	<4D59C085.9040008@communicatefreely.net>
	<AANLkTikZbWPqf7N1fT-9fF8Fvydi85mJVZ+o8_J=RLjE@mail.gmail.com>
	<4D5A9BEA.7070302@communicatefreely.net>
	<AANLkTikuozcozSiFFL8-6+s3iRP4Lfy9DT2xFi9TVBOd@mail.gmail.com>
	<16242449-AADE-4068-8523-C1727F66FDF8@freeswitch.org>
	<4D5BF2DC.20409@communicatefreely.net>
	<AANLkTik5pgWLDqg3Q2FDsc3i+NJ9vo77VGjrX7v-2bK6@mail.gmail.com>
	<AANLkTi=6xtWFSNV3qOHiW9Sedy8xq2FiO7g03qEX0qSk@mail.gmail.com>
Message-ID: <AANLkTimqVbncDnXmh99tYvxkJtKHxGyfYaU8R98iEJ5O@mail.gmail.com>

Get the SIP trace of the phone doing its registration. Also, is the phone on
the same LAN as FreeSWITCH? Is there any NAT involved?
-MC

On Fri, Feb 18, 2011 at 12:43 PM, Aloysius Lloyd
<lloyd.aloysius at gmail.com>wrote:

> I did some traces and figure it out the problem losing the Registrations.
>
> 1. FreeSWITCH Send the response to a wrong port . Is this a Bug?
>
> 2. Why FreeSWITCH + Aastra Registration contact not have the port number ?
>
> 3. See the below ...
>
> 203 & 202 Aastra Phones on Domain - *aastra.mydomain.com*
>
> But user Agent Shows Polycom. BTW in the same LAN I have two polycom phones
> on different domain *foo.mydomain.com*
>
> ==============
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 173.230.136.12;rport;branch=z9hG4bKZ6DDSDNZ11gvS
> *From: <sip:203 at aastra.mydomain.com>;tag=3jFNS2KBvNX0S*
> *To: "Ext 202" <sip:203 at aastra.mydomain.com>;tag=B550915E-F491D63B*
> CSeq: 8639912 NOTIFY
> Call-ID: cdad5068-b542-122e-ef86-fefdade6880c
> *Contact: <sip:202 at 192.168.240.128:5060>*
> Event: message-summary
> *User-Agent: PolycomSoundPointIP-SPIP_450-UA/3.3.1.0769*
> Accept-Language: en
> Content-Length: 0
>
> =============
>
> Any help on this why sending wrong ports?
>
> Thanks
> Lloyd
>
> On Wed, Feb 16, 2011 at 10:34 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com
> > wrote:
>
>> Hi All,
>>
>>
>> I stay away with Aastra phone for a long time  and today I did some tests.
>> All of my test ... the phones not reliable with FreeSWITCH
>>
>> I try both 6731i and 57i with the most recent firmware
>>
>> Here is the configuration files
>>
>> *aastra.cfg*
>>
>> dhcp: 1
>> sip digit timeout: 3
>> sip dial plan:
>> "x+#|xx+*|[2-9]XX[2-9]XXXXXX|1[2-9]XX[2-9]XXXXXX|1XXXXXXXXXX|[2-3]XX|67[2-9]XX[2-9]XXXXXX"
>> sip rport: 1
>> sip customized
>> codec:payload=9;ptime=20;silsupp=off,payload=0;ptime=20;silsupp=off,payload=18;ptime=20;silsupp=off
>> #sip registration period: 120
>> #sip registration renewal timer: 15
>> headset tx gain: -3
>> headset sidetone gain: -3
>> handset tx gain: -3
>> handset sidetone gain: -3
>> handsfree tx gain: 0
>> handset volume: 5     #RX volumes - user adjustable, so easily changed
>> speaker volume: 5
>> ringer volume: 3
>> web interface enabled: 1
>> live dialpad: 1
>> missed calls indicator disabled: 1
>> suppress dtmf playback: 0
>> #audio mode: 2 #0 = speaker (default)1 = headset 2 = speaker/headset 3 =
>> headset/speaker
>> time server disabled: 0
>> time server1: pool.ntp.org
>> #directory
>> directory 1: internal_list.csv
>> directory 2: external_list.csv
>>
>> *mac.cfg*
>>
>> directed call pickup: 1
>> directed call pickup prefix: **
>> #
>> sip line1 screen name: Ext 203
>> sip line1 display name: Ext 203
>> sip line1 auth name: 203
>> sip line1 user name: 203
>> sip line1 password: *********
>> sip line1 vmail: *97
>> sip line1 mode: 0
>> #
>> sip line1 proxy ip: aastra.mydomain.com
>> #sip line1 proxy port: 5060
>> sip line1 registrar ip: aastra.mydomain.com
>> sip line1 registration period: 300
>> #sip line1 registrar port: 5060
>>
>> ------------
>>
>> 1. FreeSWITCH Registration shows two entires in the internal profile and
>> when I try to call to the extension two lines on the phones rings .... no
>> idea why this is happen?
>>
>> Call-ID:        d8fd4c8795801cd2
>> User:           203 at aastra.mydomain.com
>> Contact:        "Ext 203" <sip:203 at 192.168.240.106:5060
>> ;transport=udp;fs_nat=yes;fs_path=sip%3A203%40173.33.178.49%3A1627%3Btransport%3Dudp>
>> Agent:          Aastra 6731i/2.6.0.2010
>> Status:         Registered(UDP-NAT)(unknown) EXP(2011-02-16 22:24:36)
>> EXPSECS(310)
>> Host:           li176-12
>> IP:             173.33.178.49
>> Port:           1627
>> Auth-User:      203
>> Auth-Realm:     aastra.mydomain.com
>> MWI-Account:    203 at aastra.mydomain.com
>>
>> Call-ID:        d8fd4c8795801cd2
>> User:           203 at aastra.mydomain.com
>> Contact:        "Ext 203" <sip:203 at 173.33.178.49:1627;transport=udp>
>> Agent:          Aastra 6731i/2.6.0.2010
>> Status:         Registered(UDP)(unknown) EXP(2011-02-16 22:24:36)
>> EXPSECS(310)
>> Host:           li176-12
>> IP:             173.33.178.49
>> Port:           1627
>> Auth-User:      203
>> Auth-Realm:     aastra.mydomain.com
>> MWI-Account:    203 at aastra.mydomain.com
>>
>>
>> 2. After first registration expires .... FreeSWITCH internal registration
>> status shows the following entry. When I dial the extension now there is a
>> long delay ... FreeSWITCH dialing and waiting then goes to voicemail.
>>
>> Call-ID:        d8fd4c8795801cd2
>> User:           203 at aastra.mydomain.com
>> Contact:        "Ext 203" <sip:203 at 173.33.178.49:1627;transport=udp>
>> Agent:          Aastra 6731i/2.6.0.2010
>> Status:         Registered(UDP)(unknown) EXP(2011-02-16 22:31:25)
>> EXPSECS(266)
>> Host:           li176-12
>> IP:             173.33.178.49
>> Port:           1631
>> Auth-User:      203
>> Auth-Realm:     aastra.mydomain.com
>> MWI-Account:    203 at aastra.mydomain.com
>>
>> 3. Phones Goes randomly "No Service" .... I see this problem several times
>> in the past.
>>
>> -------------
>>
>> At the same time in my lab Linksys and Polycom Phones working without any
>> issues with the default settings.
>>
>> -------------
>>
>> my question what is Aastra Doing differently?
>>
>> Thanks
>> Lloyd
>>
>> On Wed, Feb 16, 2011 at 10:53 AM, Tim St. Pierre <
>> fs-list at communicatefreely.net> wrote:
>>
>>> I have been setting my expires to 600.
>>>
>>> Brian West wrote:
>>> > Are you setting your expires to > 300 seconds?
>>> >
>>> > /b
>>> >
>>> > On Feb 15, 2011, at 9:42 AM, Aloysius Lloyd wrote:
>>> >
>>> >>
>>> >>
>>> >> I have the same issue, around 275 phones in the field. I want the 275
>>> >> phones work with FreeSWITCH.
>>> >>
>>> >>
>>> >> 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
>>> >
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sat Feb 19 04:34:41 2011
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 18 Feb 2011 19:34:41 -0600
Subject: [Freeswitch-users] IVR application
In-Reply-To: <COL111-W38C088A856EBAA7D204556ACD40@phx.gbl>
Message-ID: <C9847A51.6EC5%krice@freeswitch.org>

No it does not... FreeSWITCH does not expose that level of sip up into the
dialplan... Its all handled by the SIP stack... Please Note FreeSWITCH
itself can be your application server... If you are using SIP to control
your media playback and pass prompt responses, there are probably 100 other
ways to do this more efficiently and much easier.

Ken


On 2/18/11 2:35 PM, "Jason b.a" <k-b-24 at live.com> wrote:

> i am not going to use the embedded dialplan,  do the application interface has
> SIP API such as  " onInvite " for example , or no !
>        
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From lloyd.aloysius at sunteltech.ca  Sat Feb 19 04:39:40 2011
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Fri, 18 Feb 2011 20:39:40 -0500
Subject: [Freeswitch-users] Aastra phone registration lost
In-Reply-To: <AANLkTimqVbncDnXmh99tYvxkJtKHxGyfYaU8R98iEJ5O@mail.gmail.com>
References: <AANLkTinR4gtAFB2rNjGkoKsOOgaQpeXGDE1E-funpdR_@mail.gmail.com>
	<4D52B987.7050002@communicatefreely.net>
	<AANLkTik7UxBLs0=HHdNZVseht-Vd6qiptY__AwnabB5U@mail.gmail.com>
	<4D534AFD.3020401@communicatefreely.net>
	<AANLkTimaF-OMZsNd1ydbG9h6McdRVSs0Jc5ihmoq8y40@mail.gmail.com>
	<60853337-266B-4400-B58A-7376F169D88D@ipeva.fr>
	<AANLkTimux4MBM1K_7itE4GnZQHjy_vR2Y6sr_z+u3gkR@mail.gmail.com>
	<4D5577CE.1000603@communicatefreely.net>
	<AANLkTinJK+e9hZJbk4ur_OQwGV5B7tRLrpPMH=TU5MOo@mail.gmail.com>
	<4D59C085.9040008@communicatefreely.net>
	<AANLkTikZbWPqf7N1fT-9fF8Fvydi85mJVZ+o8_J=RLjE@mail.gmail.com>
	<4D5A9BEA.7070302@communicatefreely.net>
	<AANLkTikuozcozSiFFL8-6+s3iRP4Lfy9DT2xFi9TVBOd@mail.gmail.com>
	<16242449-AADE-4068-8523-C1727F66FDF8@freeswitch.org>
	<4D5BF2DC.20409@communicatefreely.net>
	<AANLkTik5pgWLDqg3Q2FDsc3i+NJ9vo77VGjrX7v-2bK6@mail.gmail.com>
	<AANLkTi=6xtWFSNV3qOHiW9Sedy8xq2FiO7g03qEX0qSk@mail.gmail.com>
	<AANLkTimqVbncDnXmh99tYvxkJtKHxGyfYaU8R98iEJ5O@mail.gmail.com>
Message-ID: <AANLkTim21=aKFygL6VJMc+OO5ND50o2STkVhhb-DEBZ+@mail.gmail.com>

FreeSWITCH is on public IP and Phones Behind the NAT. I see sometimes
FreeSWITCH send the reply to a wrong port or not responses for the
registration request.

Please see the following contact

Call-ID:        1bb9e78e1cba12f8
User:           202 at aastra.mydomain.com
*Contact:        "Ext 202" <sip:202 at 173.33.178.49:1287;transport=udp>*
Agent:          Aastra 6731i/2.6.0.2010
Status:         Registered(UDP)(unknown) EXP(2011-02-18 20:42:53)
EXPSECS(358)
Host:           li176-12
IP:             173.33.178.49
Port:           1287
Auth-User:      202
Auth-Realm:     aastra.mydomain.com
MWI-Account:    202 at aastra.mydomain.com


---------

I solve the problem using the <param name="sip-force-contact" value="
NDLB-connectile-dysfunction"/> in the user directory.

But why this is happening to Aastra Phones only? Please let me know.

Thanks
Lloyd
On Fri, Feb 18, 2011 at 8:29 PM, Michael Collins <msc at freeswitch.org> wrote:

> Get the SIP trace of the phone doing its registration. Also, is the phone
> on the same LAN as FreeSWITCH? Is there any NAT involved?
> -MC
>
>
> On Fri, Feb 18, 2011 at 12:43 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com
> > wrote:
>
>> I did some traces and figure it out the problem losing the Registrations.
>>
>> 1. FreeSWITCH Send the response to a wrong port . Is this a Bug?
>>
>> 2. Why FreeSWITCH + Aastra Registration contact not have the port number ?
>>
>> 3. See the below ...
>>
>> 203 & 202 Aastra Phones on Domain - *aastra.mydomain.com*
>>
>> But user Agent Shows Polycom. BTW in the same LAN I have two polycom
>> phones on different domain *foo.mydomain.com*
>>
>> ==============
>>
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP 173.230.136.12;rport;branch=z9hG4bKZ6DDSDNZ11gvS
>> *From: <sip:203 at aastra.mydomain.com>;tag=3jFNS2KBvNX0S*
>> *To: "Ext 202" <sip:203 at aastra.mydomain.com>;tag=B550915E-F491D63B*
>> CSeq: 8639912 NOTIFY
>> Call-ID: cdad5068-b542-122e-ef86-fefdade6880c
>> *Contact: <sip:202 at 192.168.240.128:5060>*
>> Event: message-summary
>> *User-Agent: PolycomSoundPointIP-SPIP_450-UA/3.3.1.0769*
>>  Accept-Language: en
>> Content-Length: 0
>>
>> =============
>>
>> Any help on this why sending wrong ports?
>>
>> Thanks
>> Lloyd
>>
>> On Wed, Feb 16, 2011 at 10:34 PM, Aloysius Lloyd <
>> lloyd.aloysius at gmail.com> wrote:
>>
>>> Hi All,
>>>
>>>
>>> I stay away with Aastra phone for a long time  and today I did some
>>> tests. All of my test ... the phones not reliable with FreeSWITCH
>>>
>>> I try both 6731i and 57i with the most recent firmware
>>>
>>> Here is the configuration files
>>>
>>> *aastra.cfg*
>>>
>>> dhcp: 1
>>> sip digit timeout: 3
>>> sip dial plan:
>>> "x+#|xx+*|[2-9]XX[2-9]XXXXXX|1[2-9]XX[2-9]XXXXXX|1XXXXXXXXXX|[2-3]XX|67[2-9]XX[2-9]XXXXXX"
>>> sip rport: 1
>>> sip customized
>>> codec:payload=9;ptime=20;silsupp=off,payload=0;ptime=20;silsupp=off,payload=18;ptime=20;silsupp=off
>>> #sip registration period: 120
>>> #sip registration renewal timer: 15
>>> headset tx gain: -3
>>> headset sidetone gain: -3
>>> handset tx gain: -3
>>> handset sidetone gain: -3
>>> handsfree tx gain: 0
>>> handset volume: 5     #RX volumes - user adjustable, so easily changed
>>> speaker volume: 5
>>> ringer volume: 3
>>> web interface enabled: 1
>>> live dialpad: 1
>>> missed calls indicator disabled: 1
>>> suppress dtmf playback: 0
>>> #audio mode: 2 #0 = speaker (default)1 = headset 2 = speaker/headset 3 =
>>> headset/speaker
>>> time server disabled: 0
>>> time server1: pool.ntp.org
>>> #directory
>>> directory 1: internal_list.csv
>>> directory 2: external_list.csv
>>>
>>> *mac.cfg*
>>>
>>> directed call pickup: 1
>>> directed call pickup prefix: **
>>> #
>>> sip line1 screen name: Ext 203
>>> sip line1 display name: Ext 203
>>> sip line1 auth name: 203
>>> sip line1 user name: 203
>>> sip line1 password: *********
>>> sip line1 vmail: *97
>>> sip line1 mode: 0
>>> #
>>> sip line1 proxy ip: aastra.mydomain.com
>>> #sip line1 proxy port: 5060
>>> sip line1 registrar ip: aastra.mydomain.com
>>> sip line1 registration period: 300
>>> #sip line1 registrar port: 5060
>>>
>>> ------------
>>>
>>> 1. FreeSWITCH Registration shows two entires in the internal profile and
>>> when I try to call to the extension two lines on the phones rings .... no
>>> idea why this is happen?
>>>
>>> Call-ID:        d8fd4c8795801cd2
>>> User:           203 at aastra.mydomain.com
>>> Contact:        "Ext 203" <sip:203 at 192.168.240.106:5060
>>> ;transport=udp;fs_nat=yes;fs_path=sip%3A203%40173.33.178.49%3A1627%3Btransport%3Dudp>
>>> Agent:          Aastra 6731i/2.6.0.2010
>>> Status:         Registered(UDP-NAT)(unknown) EXP(2011-02-16 22:24:36)
>>> EXPSECS(310)
>>> Host:           li176-12
>>> IP:             173.33.178.49
>>> Port:           1627
>>> Auth-User:      203
>>> Auth-Realm:     aastra.mydomain.com
>>> MWI-Account:    203 at aastra.mydomain.com
>>>
>>> Call-ID:        d8fd4c8795801cd2
>>> User:           203 at aastra.mydomain.com
>>> Contact:        "Ext 203" <sip:203 at 173.33.178.49:1627;transport=udp>
>>> Agent:          Aastra 6731i/2.6.0.2010
>>> Status:         Registered(UDP)(unknown) EXP(2011-02-16 22:24:36)
>>> EXPSECS(310)
>>> Host:           li176-12
>>> IP:             173.33.178.49
>>> Port:           1627
>>> Auth-User:      203
>>> Auth-Realm:     aastra.mydomain.com
>>> MWI-Account:    203 at aastra.mydomain.com
>>>
>>>
>>> 2. After first registration expires .... FreeSWITCH internal registration
>>> status shows the following entry. When I dial the extension now there is a
>>> long delay ... FreeSWITCH dialing and waiting then goes to voicemail.
>>>
>>> Call-ID:        d8fd4c8795801cd2
>>> User:           203 at aastra.mydomain.com
>>> Contact:        "Ext 203" <sip:203 at 173.33.178.49:1627;transport=udp>
>>> Agent:          Aastra 6731i/2.6.0.2010
>>> Status:         Registered(UDP)(unknown) EXP(2011-02-16 22:31:25)
>>> EXPSECS(266)
>>> Host:           li176-12
>>> IP:             173.33.178.49
>>> Port:           1631
>>> Auth-User:      203
>>> Auth-Realm:     aastra.mydomain.com
>>> MWI-Account:    203 at aastra.mydomain.com
>>>
>>> 3. Phones Goes randomly "No Service" .... I see this problem several
>>> times in the past.
>>>
>>> -------------
>>>
>>> At the same time in my lab Linksys and Polycom Phones working without any
>>> issues with the default settings.
>>>
>>> -------------
>>>
>>> my question what is Aastra Doing differently?
>>>
>>> Thanks
>>> Lloyd
>>>
>>> On Wed, Feb 16, 2011 at 10:53 AM, Tim St. Pierre <
>>> fs-list at communicatefreely.net> wrote:
>>>
>>>> I have been setting my expires to 600.
>>>>
>>>> Brian West wrote:
>>>> > Are you setting your expires to > 300 seconds?
>>>> >
>>>> > /b
>>>> >
>>>> > On Feb 15, 2011, at 9:42 AM, Aloysius Lloyd wrote:
>>>> >
>>>> >>
>>>> >>
>>>> >> I have the same issue, around 275 phones in the field. I want the 275
>>>> >> phones work with FreeSWITCH.
>>>> >>
>>>> >>
>>>> >> 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
>>>> >
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at sunteltech.ca  Sat Feb 19 04:53:40 2011
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Fri, 18 Feb 2011 20:53:40 -0500
Subject: [Freeswitch-users] Need some help. Novice freeswitch setup gone
	wrong
In-Reply-To: <AANLkTi=Hj=b5gKm7BdgYCjnh801HVY-0Lwfr__=v76wZ@mail.gmail.com>
References: <AANLkTi=Hj=b5gKm7BdgYCjnh801HVY-0Lwfr__=v76wZ@mail.gmail.com>
Message-ID: <AANLkTi=yRgpU6FpcUEQ=bw5KLUY6mjxR19zOnbBuj8q4@mail.gmail.com>

Do not know the blue.box app. But I assume installation on the default
directory.

A.stop xml_cdr .

Comment the following line <load module="mod_xml_cdr"/> in the
/usr/local/freeswitch/conf/autoload_configs/modules.conf.xml file

reloadxml

unload mod_xml_cdr

Enable CSV

 <load module="mod_xml_cdr"/>

Enable CSV text file

add the following  in the
/usr/local/freeswitch/conf/autoload_configs/modules.conf.xml file
<load module="mod_cdr_csv"/>

reloadxml
load mod_cdr_csv

B. Multi tenant Read the following
http://wiki.freeswitch.org/wiki/Multi-tenant

<http://wiki.freeswitch.org/wiki/Multi-tenant>C. Once you understand How
multitenant works then you know the answer for this question.


LLoyd

On Fri, Feb 18, 2011 at 11:47 AM, Hareem Haque <hareem.haque at gmail.com>wrote:

>  I have a little problem i setup the blue.box app it runs very well. Now i
> am having some problems.
>
> A. How can i stop using the xml cdr and just use text file only cdr which
> are dumped into a folder called /mnt/fscdr
>
> B. How can i setup the pbx as a multi tenant system and keep track of all
> the users we have 6 departments with 20 employees each.
>
> C. Can i setup trunks in a way that SIP trunk A can only be used by
> Department A.
>
> Any help is greatly appreciated.
>
> Regards
> Hareem Haque
>
>
> Best Regards
> Hareem. Haque
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at sunteltech.ca  Sat Feb 19 05:48:09 2011
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Fri, 18 Feb 2011 21:48:09 -0500
Subject: [Freeswitch-users] Fwd:  NAT issue
In-Reply-To: <AANLkTin0PCdP5+TnA39CjAQ9JwVexUzOyno23nPsbyrh@mail.gmail.com>
References: <AANLkTin0PCdP5+TnA39CjAQ9JwVexUzOyno23nPsbyrh@mail.gmail.com>
Message-ID: <AANLkTikvgh9vhL2Za2avzxMz-BFbt2ud+wE+kgrWBkur@mail.gmail.com>

---------- Forwarded message ----------
From: Spencer Thomason <spencer at 5ninesolutions.com>
Date: Fri, Feb 18, 2011 at 8:46 PM
Subject: Re: [Freeswitch-users] NAT issue
To: Aloysius Lloyd <lloyd.aloysius at sunteltech.ca>


I did and that resolved the issue.  I was more or less just curious as to
why Freeswith detected all the others at NAT and not those.  There isn't any
real network topology differences between the endpoints. All are running
Polycom firmware v 3.3.1.

Thanks,
Spencer

Aloysius Lloyd <lloyd.aloysius at sunteltech.ca> wrote:

>May be the Two phones using different firmware.
>
>Try the following parameter in the extension
>
><param name="sip-force-contact" value="NDLB-connectile-dysfunction"/>
>
>Thanks
>Lloyd
>
>
>On Fri, Feb 18, 2011 at 12:12 PM, Spencer Thomason <
>spencer at 5ninesolutions.com> wrote:
>
>> Hello all,
>> I'm having a strange NAT problem.  My Freeswitch server has a public
>> ip and the endpoints which are Polycom IP 650s are all behind NAT with
>> mainly Netgear SRX 5308 routers at multiple locations.  The server is
>> on a public class C network and one of the locations has its router on
>> the same class C.  I.e x.x.x.7/24-router, x.x.x.240/24-server.  I'm
>> only experiencing this with 2 of about 30 extensions.  The first
>> endpoint to register from this location gets detected as NAT but the
>> others with the ports remapped do not. This maybe explains it better:
>>
>> Call-ID:        4feea796-bd96e2ed-e16d27ac at 10.6.71.103
>> User:           1103 at x.x.x.240
>> Contact:        "user" <sip:1103 at 10.6.71.103;received=x.x.x.
>> 7:5060;fs_nat=yes;fs_path=sip%3A1103%40x.x.x.7%3A5060>
>> Agent:          PolycomSoundPointIP-SPIP_650-UA/3.3.1.0769
>> Status:         Registered(UDP-NAT)(unknown) EXP(2011-02-18 11:15:47)
>> EXPSECS(1143)
>> Host:           pbx10.5ninesolutions.com
>> IP:             x.x.x.7
>> Port:           5060
>> Auth-User:      1103
>> Auth-Realm:     pbx10.5ninesolutions.com
>> MWI-Account:    1103 at x.x.x.x
>>
>> Call-ID:        d21b71b2-75baa80d-768c0b08 at 10.6.71.104
>> User:           1101 at x.x.x.240
>> Contact:        "user" <sip:1101 at 10.6.71.104>
>> Agent:          PolycomSoundPointIP-SPIP_650-UA/3.3.1.0769
>> Status:         Registered(UDP)(unknown) EXP(2011-02-18 11:16:48)
>> EXPSECS(1204)
>> Host:           pbx10.5ninesolutions.com
>> IP:             x.x.x.7
>> Port:           1036
>> Auth-User:      1101
>> Auth-Realm:     pbx10.5ninesolutions.com
>> MWI-Account:    1101 at x.x.x.x
>>
>> Call-ID:        a74e61a6-1863ea7d-789686c4 at 10.6.71.105
>> User:           1102 at x.x.x.240
>> Contact:        "user" <sip:1102 at 10.6.71.105>
>> Agent:          PolycomSoundPointIP-SPIP_650-UA/3.3.1.0769
>> Status:         Registered(UDP)(unknown) EXP(2011-02-18 11:17:23)
>> EXPSECS(1239)
>> Host:           pbx10.5ninesolutions.com
>> IP:             x.x.x.7
>> Port:           1036
>> Auth-User:      1102
>> Auth-Realm:     pbx10.5ninesolutions.com
>> MWI-Account:    1102 at x.x.x.x
>>
>> Call-ID:        100b5148-6a47060a-13c4-50029-c844-22fc4dd3-c844
>> User:           1801 at 97.67.110.240
>> Contact:        "user" <sip:1801 at 10.6.71.106:5060>
>> Agent:          MP201 B 1FXS/3.0.1_build_19
>> Status:         Registered(UDP)(unknown) EXP(2011-02-18 11:26:23)
>> EXPSECS(1779)
>> Host:           pbx10.5ninesolutions.com
>> IP:             97.67.110.7
>> Port:           1036
>> Auth-User:      1801
>> Auth-Realm:     pbx10.5ninesolutions.com
>> MWI-Account:    1801 at x.x.x.x
>>
>> -------  Another Location ----
>>
>> Call-ID:        dca2a276-4eb94f97-c55177b0 at 10.1.10.186
>> User:           4204 at x.x.x.240
>> Contact:        "user" <sip:4204 at 10.1.10.186;received=x.x.x.x:
>> 5060;fs_nat=yes;fs_path=sip%3A4204%40x.x.x.x%3A5060>
>> Agent:          PolycomSoundStationIP-SSIP_5000-UA/3.3.1.0769
>> Status:         Registered(UDP-NAT)(unknown) EXP(2011-02-18 11:28:04)
>> EXPSECS(1880)
>> Host:           pbx10.5ninesolutions.com
>> IP:             x.x.x.x
>> Port:           5060
>> Auth-User:      4204
>> Auth-Realm:     pbx10.5ninesolutions.com
>> MWI-Account:    4204 at x.x.x.x
>>
>> The port and IP are detected correctly but the contact is a private
>> IP.  Any clue as to why the two extension 1101 and 1102 are any
>> different?
>>
>> Thanks,
>> 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
>>
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From sunwood360 at gmail.com  Sat Feb 19 06:55:21 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Fri, 18 Feb 2011 19:55:21 -0800
Subject: [Freeswitch-users] core dump in debian
Message-ID: <AANLkTi=YopVmLSTi9MwhpXEx-wpJ-nLmiB4jeBSszAz1@mail.gmail.com>

#0  vxprintf (arg=0x40ac9be0, fmt=0x40249412 "q'", ap=<value optimized out>,
useExtended=<value optimized out>, func=<value optimized out>)
    at src/switch_mprintf.c:686
#1  0x40130774 in base_vprintf (xRealloc=0x4012f328 <printf_realloc>,
zInitBuf=0xb <Address 0xb out of bounds>, nInitBuf=<value optimized out>,
    zFormat=0xa <Address 0xa out of bounds>, ap=..., useInternal=<value
optimized out>) at src/switch_mprintf.c:849
#2  0x40130880 in switch_mprintf (zFormat=0x4027671c "debian") at
src/switch_mprintf.c:892
#3  0x4013ba48 in switch_core_expire_registration (force=0) at
src/switch_core_sqldb.c:1726
#4  0x4013bf3c in switch_core_sql_db_thread (thread=<value optimized out>,
obj=0x0) at src/switch_core_sqldb.c:926
#5  0x401c93dc in dummy_worker (opaque=0x402493d4) at
threadproc/unix/thread.c:138
#6  0x402e48cc in start_thread () from /lib/libpthread.so.0
#7  0x4058bbdc in clone () from /lib/libc.so.6
#8  0x4058bbdc in clone () from /lib/libc.so.6


FS crashed right after start!
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From jmesquita at freeswitch.org  Sat Feb 19 07:11:46 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 19 Feb 2011 01:11:46 -0300
Subject: [Freeswitch-users] ESL socket outbound in C++
In-Reply-To: <AANLkTinYwvATpqQn38OX2F4eoXkhCuaCH3i1KWnbjuWF@mail.gmail.com>
References: <AANLkTi=+LAue0ZEiJ2zGg4Kx_KBBSrcXEyKMunw9m=kX@mail.gmail.com>
	<AANLkTinYwvATpqQn38OX2F4eoXkhCuaCH3i1KWnbjuWF@mail.gmail.com>
Message-ID: <AANLkTi=6Wsrxkz_3mR2LJUX5ZGNZiTA3D8_vPx54YkjS@mail.gmail.com>

There is some C implementations of ESL stuff on my contrib dir as well if
anyone is interested...

Also, FSGui is all ESL with Qt and can also be found on my contrib dir.

Regards,
Jo?o Mesquita



On Fri, Feb 18, 2011 at 10:27 PM, Michael Collins <msc at freeswitch.org>wrote:

> We appreciate you sharing! Feel free to add this to the wiki as well.
> -MC
>
> On Fri, Feb 18, 2011 at 8:25 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> I'm learning to use ESL in C++ to do a call control and I want to share my
>> code with the community.
>>
>> I had a little difficult at startup so I hope that this code can help
>> someone.
>>
>> The code is: http://pastebin.freeswitch.org/15418
>>
>> Using "esl_oop" module, I have implemented a basic ESL Channel class (that
>> can be extended) so the "main" event handling loop can be very simple, for
>> example for an unuseful esl bridge application, can be:
>>
>>     ESLchannel legA(&connection, inbound_uuid);
>>
>>     ESLchannel legB(&connection);
>>     legB.Originate(dialstring)
>>
>>     while (1)
>>      {
>>         ESLevent * event = connection.recvEventTimed(1000);
>>
>>         if (event)
>>         {
>>             const char * ename = event->getHeader("Event-Name");
>>             const char * event_uuid = event->getHeader("unique-id");
>>
>>             // It's an event for LegB ?
>>             //-------------------------
>>             if (legB.Uuid() == event_uuid)
>>             {
>>                 if (!strcmp(ename, "CHANNEL_PROGRESS_MEDIA"))
>>                 {
>>                     legA.PreAnswer();
>>                     legA.BridgeTo(legB);
>>                 }
>>                 else if (!strcmp(ename, "CHANNEL_ANSWER"))
>>                 {
>>                     legA.Answer();
>>                     legA.BridgeTo(legB);
>>                 }
>>                 else if (!strcmp(ename,"CHANNEL_HANGUP"))
>>                 {
>>                     const char * cause = event->getHeader("Hangup-Cause");
>>
>>                     legA.Hangup(cause);
>>                 }
>>             }
>>             // It's an event for LegA ?
>>             //-------------------------
>>             else if (legA.Uuid() == event_uuid)
>>             {
>>                 if (!strcmp(ename,"CHANNEL_HANGUP"))
>>                 {
>>                     const char * cause = event->getHeader("Hangup-Cause");
>>
>>                     legB.Hangup(cause);
>>                 }
>>             }
>>
>>             delete event;
>>         }
>>     }
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From k-b-24 at live.com  Sat Feb 19 08:47:11 2011
From: k-b-24 at live.com (Jason b.a)
Date: Sat, 19 Feb 2011 05:47:11 +0000
Subject: [Freeswitch-users] IVR application
Message-ID: <col111-w45D6C4A3E561CD002A7306ACD70@phx.gbl>


aha thx MC ! 
Ken give me ur suggestions pls , if i need a ivr application that controlled the media server using SIP and an embedded control language !  		 	   		  
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From edward.dejong at voicecarrier.com  Sat Feb 19 11:04:21 2011
From: edward.dejong at voicecarrier.com (Edward de Jong)
Date: Sat, 19 Feb 2011 00:04:21 -0800
Subject: [Freeswitch-users] newbies is stuck / windows version of freeswitch
	won't start up after reboot
Message-ID: <3253D3FE-97A0-4884-A13A-5D9164EC794C@voicecarrier.com>

I installed Freeswitch (by downloading the .EXE version for windows x64) and was running it okay on Windows 7 Professional, 64 bit edition, but after a restart, when I try to run fs_cli it gives me a socket error and fails.  I am not really running anything else much on my machine, and i figure somehow that port has gotten blocked. I turned off windows firewall, but still it won't run, i can't figure out why it was running so well, but now on a clean restart it won't start up.

I didn't install it as a startup program, I was manually running fs_cli and freeswitchconsole, which are the two freeswitch programs, yes?  

thanks in advance.

edj



From mitch.capper at gmail.com  Sat Feb 19 11:19:56 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Sat, 19 Feb 2011 00:19:56 -0800
Subject: [Freeswitch-users] newbies is stuck / windows version of
 freeswitch won't start up after reboot
In-Reply-To: <3253D3FE-97A0-4884-A13A-5D9164EC794C@voicecarrier.com>
References: <3253D3FE-97A0-4884-A13A-5D9164EC794C@voicecarrier.com>
Message-ID: <AANLkTinqSeMNyJ36JzQAGtnbUrNrkYurMGHqFN6=9aub@mail.gmail.com>

fs_cli connects to the event socket created by freeswitch console.
Assuming freeswitch console is spinning up correctly look in the log
file for a line like:
2011-02-18 11:01:46.009160 [DEBUG] mod_event_socket.c:2722 Socket up
listening on 127.0.0.1:8021
that should show the event socket binding on the ip and port. then use
fs_cli -P [port] -h [ip]
and it should connect.

~Mitch

On Sat, Feb 19, 2011 at 12:04 AM, Edward de Jong
<edward.dejong at voicecarrier.com> wrote:
> I installed Freeswitch (by downloading the .EXE version for windows x64) and was running it okay on Windows 7 Professional, 64 bit edition, but after a restart, when I try to run fs_cli it gives me a socket error and fails. ?I am not really running anything else much on my machine, and i figure somehow that port has gotten blocked. I turned off windows firewall, but still it won't run, i can't figure out why it was running so well, but now on a clean restart it won't start up.
>
> I didn't install it as a startup program, I was manually running fs_cli and freeswitchconsole, which are the two freeswitch programs, yes?
>
> thanks in advance.
>
> edj
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 edward.dejong at voicecarrier.com  Sat Feb 19 11:25:09 2011
From: edward.dejong at voicecarrier.com (Edward de Jong)
Date: Sat, 19 Feb 2011 00:25:09 -0800
Subject: [Freeswitch-users] newbies is stuck / windows version of
	freeswitch won't start up after reboot
In-Reply-To: <AANLkTinqSeMNyJ36JzQAGtnbUrNrkYurMGHqFN6=9aub@mail.gmail.com>
References: <3253D3FE-97A0-4884-A13A-5D9164EC794C@voicecarrier.com>
	<AANLkTinqSeMNyJ36JzQAGtnbUrNrkYurMGHqFN6=9aub@mail.gmail.com>
Message-ID: <EDBAC4F4-29CC-421D-8957-D17488ABC535@voicecarrier.com>

No, my freeswitch console is not starting up. I am using the standard port 5060 but no go... can't figure out why the console program won't start up now. it did when I installed the program the first time.

edj

On Feb 19, 2011, at 12:19 AM, Mitch Capper wrote:

> fs_cli connects to the event socket created by freeswitch console.
> Assuming freeswitch console is spinning up correctly look in the log
> file for a line like:
> 2011-02-18 11:01:46.009160 [DEBUG] mod_event_socket.c:2722 Socket up
> listening on 127.0.0.1:8021
> that should show the event socket binding on the ip and port. then use
> fs_cli -P [port] -h [ip]
> and it should connect.
> 
> ~Mitch
> 
> On Sat, Feb 19, 2011 at 12:04 AM, Edward de Jong
> <edward.dejong at voicecarrier.com> wrote:
>> I installed Freeswitch (by downloading the .EXE version for windows x64) and was running it okay on Windows 7 Professional, 64 bit edition, but after a restart, when I try to run fs_cli it gives me a socket error and fails.  I am not really running anything else much on my machine, and i figure somehow that port has gotten blocked. I turned off windows firewall, but still it won't run, i can't figure out why it was running so well, but now on a clean restart it won't start up.
>> 
>> I didn't install it as a startup program, I was manually running fs_cli and freeswitchconsole, which are the two freeswitch programs, yes?
>> 
>> thanks in advance.
>> 
>> edj
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 b_ball_henry at hotmail.com  Sat Feb 19 11:30:48 2011
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Sat, 19 Feb 2011 16:30:48 +0800
Subject: [Freeswitch-users] Skype 2.0.72 for Linux
In-Reply-To: <1297996932418-6038482.post@n2.nabble.com>
References: <AANLkTinYLAuoB9wy_vjbZwbO38PthoFzSia-2bS9yCwf@mail.gmail.com>
	<1297996932418-6038482.post@n2.nabble.com>
Message-ID: <AANLkTikSGshibahhc3aw1=K0O8WFdeNYgO-R=nxO34jm@mail.gmail.com>

Does this work on all distribution of Linux as long as we run the disrpm
utility?

Thanks,

Henry

On Fri, Feb 18, 2011 at 10:42 AM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
> You can download it from
>
> ftp://ftp.isu.edu.tw/pub/Linux/PLD/dists/2.0/test/SRPMS/skype-2.0.0.27-1.src.rpm
> here . Then, use this shell scripts
> http://www.mombu.com/gnu_linux/red-hat/t-disrpm-shell-script-8907760.html
> disrpm  utility to extract it.
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Skype-2-0-72-for-Linux-tp6038235p6038482.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 Nabble at slickdeals.endjunk.com  Sat Feb 19 15:11:31 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sat, 19 Feb 2011 04:11:31 -0800 (PST)
Subject: [Freeswitch-users] Skype 2.0.72 for Linux
In-Reply-To: <AANLkTikSGshibahhc3aw1=K0O8WFdeNYgO-R=nxO34jm@mail.gmail.com>
References: <AANLkTinYLAuoB9wy_vjbZwbO38PthoFzSia-2bS9yCwf@mail.gmail.com>
	<1297996932418-6038482.post@n2.nabble.com>
	<AANLkTikSGshibahhc3aw1=K0O8WFdeNYgO-R=nxO34jm@mail.gmail.com>
Message-ID: <1298117491354-6043071.post@n2.nabble.com>



Henry Huang-2 wrote:
> 
> Does this work on all distribution of Linux as long as we run the disrpm
> utility?
disrpm is a shell script file that should run on any Linux/Un*x system.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Skype-2-0-72-for-Linux-tp6038235p6043071.html
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From neilp at cs.stanford.edu  Sat Feb 19 15:55:22 2011
From: neilp at cs.stanford.edu (Neil Patel)
Date: Sat, 19 Feb 2011 18:25:22 +0530
Subject: [Freeswitch-users] bridging?
In-Reply-To: <AANLkTimCeaoCc5pTs+a2vhhyQkDnuFX2dstB1uNy3uX_@mail.gmail.com>
References: <AANLkTikkWZtatkiLRLdvu9-T2-VdsNRtyAk7ZCMCcF+4@mail.gmail.com>
	<AANLkTikWBopbR5yk6xAy-y6Ueqm+4wS+=_MNVah9bqD1@mail.gmail.com>
	<AANLkTimCeaoCc5pTs+a2vhhyQkDnuFX2dstB1uNy3uX_@mail.gmail.com>
Message-ID: <AANLkTikFuFg1mDDgvHpz_VM2QSKZ0mPLTpee_K2-arJD@mail.gmail.com>

Hi Michael,

Using transfer seems to do the trick:

session:setAutoHangup(false);
session:transfer("7777", "XML", "default");

Any downsides to this?
Neil

On Thu, Feb 17, 2011 at 1:48 PM, Neil Patel <neilp at cs.stanford.edu> wrote:

> Can I use this from within a live outbound (FS -> 123456789) call initiated
> through a lua app? Once this command is executed, what happens to that
> original call?
>
>
> On Thu, Feb 17, 2011 at 1:26 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Can you use originate?
>>
>> originate sofia/gateway/gwname/123456789 7777
>>
>> that will create the outbound call to the destination number (1233456789)
>> and then bridge it to 7777.
>>
>> -MC
>>
>> On Wed, Feb 16, 2011 at 10:33 AM, Neil Patel <neilp at cs.stanford.edu>wrote:
>>
>>> Hi Folks, newbie question:
>>>
>>> I have a simple dialplan that takes calls to an extension and executes a
>>> lua app:
>>>
>>> *<extension name="incoming">*
>>> *        <condition field="destination_number" expression="^7777$">*
>>> *            <action application="lua" data="lua/app.lua" />*
>>> *        </condition>*
>>> * </extension>*
>>> *
>>> *
>>>
>>> From the same extension I am initiating calls from FS to endpoints and
>>> executing another lua app. What I want to do is connect (bridge?) those
>>> outbound calls to app.lua. From FS perspective this call should now look
>>> like it was initiated by the caller to 7777.
>>>
>>> Is this what call bridging is? What commands/wiki page should I be
>>> looking at to do this?
>>>
>>> Thanks in advance,
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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  Sat Feb 19 17:44:36 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sat, 19 Feb 2011 06:44:36 -0800 (PST)
Subject: [Freeswitch-users] newbies is stuck / windows version of
 freeswitch won't start up after reboot
In-Reply-To: <3253D3FE-97A0-4884-A13A-5D9164EC794C@voicecarrier.com>
References: <3253D3FE-97A0-4884-A13A-5D9164EC794C@voicecarrier.com>
Message-ID: <1298126676111-6043351.post@n2.nabble.com>


Does it start when you run it manually? You can also run fs as a service if
you want it to run all the time when your machine is booted.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/newbies-is-stuck-windows-version-of-freeswitch-won-t-start-up-after-reboot-tp6042687p6043351.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Sat Feb 19 18:32:22 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sat, 19 Feb 2011 07:32:22 -0800 (PST)
Subject: [Freeswitch-users]
	=?utf-8?q?Newbie=C2=B4s_question_about_FreeSwi?=
	=?utf-8?b?dGNoLi4u?=
In-Reply-To: <AANLkTinf=z4JEp4G2xK_0n+u8v0OhR++GBvPzzbT1Y_-@mail.gmail.com>
References: <AANLkTinf=z4JEp4G2xK_0n+u8v0OhR++GBvPzzbT1Y_-@mail.gmail.com>
Message-ID: <1298129542465-6043436.post@n2.nabble.com>


For your usage, it is possible to use your existing server to host
freeswitch. My usage is pretty low, i.e. average of two lines usage, and my
Seagate DockStar (used as a host to my freeswitch) only shows a usage of
less than 10% of CPU resources. Assuming your usage is what you said and
your existing server load is no more than 50%, all the expenses you will
incur is probably to buy additional two ATA devices where each will have two
FXS ports, i.e. Linksys/Cisco PAP2T (beware of a counterfeit PAP2T sold on
e-bay), the new Obi products, etc. Personally, my preference is the Obi 110
which comes with 1 FXO and 1 FXS port. The FXO port can be connected
directly to your existing PBX system as an extension. This device also
supports XMPP to use with Google Voice and is sold @US$49.99/each (MSRP
price). Do a Google search to find out more information on this product.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Newbie-s-question-about-FreeSwitch-tp6038911p6043436.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From dmitry.bely at gmail.com  Sat Feb 19 20:05:02 2011
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Sat, 19 Feb 2011 20:05:02 +0300
Subject: [Freeswitch-users] Enterprise bridge,
	groups and "follow me" dialplan
Message-ID: <AANLkTintRxtrmCdKi8MMYsvP5hNZEu7Nutj96+LCKE33@mail.gmail.com>

I am trying to implement the following bridging scenario:

<action application="set" data="call_timeout=30"/>
<action application="bridge"
data="group/group1 at domain,[leg_delay_start=10]group/group2 at domain"/>
...

i.e. all phones in group1 should ring for 10 seconds and if nobody
answers group1+group2 rings for additional 20 seconds.

That almost works, but only the first phone rings in either group:
[WARNING] switch_ivr_originate.c:2339 Only calling the first element
in the list in this mode.

Surely I need enterprise bridge:

<action application="bridge"
data="group/group1 at domain:_:[leg_delay_start=10]group/group2 at domain"/>

but now there is no correct caller id and no leg delay. Caller id can
be restored with:

<action application="bridge" data="
  <origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number}>
  group/group1 at domain:_:[leg_delay_start=10]group/group2 at domain"/>

(is this a bug or design decision?), but what to do with leg delay?
[leg_delay_start=10] is simply ignored there.

- Dmitry Bely


From anthony.minessale at gmail.com  Sat Feb 19 21:20:32 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 19 Feb 2011 12:20:32 -0600
Subject: [Freeswitch-users] core dump in debian
In-Reply-To: <AANLkTi=YopVmLSTi9MwhpXEx-wpJ-nLmiB4jeBSszAz1@mail.gmail.com>
References: <AANLkTi=YopVmLSTi9MwhpXEx-wpJ-nLmiB4jeBSszAz1@mail.gmail.com>
Message-ID: <AANLkTimq=bHNgLK=NROOJz7ce76oA4cS1gM1W2Fk6ehj@mail.gmail.com>

Do a fresh checkout and remove previous binaries.
 On Feb 18, 2011 9:57 PM, "envelopes envelopes" <sunwood360 at gmail.com>
wrote:
> #0 vxprintf (arg=0x40ac9be0, fmt=0x40249412 "q'", ap=<value optimized
out>,
> useExtended=<value optimized out>, func=<value optimized out>)
> at src/switch_mprintf.c:686
> #1 0x40130774 in base_vprintf (xRealloc=0x4012f328 <printf_realloc>,
> zInitBuf=0xb <Address 0xb out of bounds>, nInitBuf=<value optimized out>,
> zFormat=0xa <Address 0xa out of bounds>, ap=..., useInternal=<value
> optimized out>) at src/switch_mprintf.c:849
> #2 0x40130880 in switch_mprintf (zFormat=0x4027671c "debian") at
> src/switch_mprintf.c:892
> #3 0x4013ba48 in switch_core_expire_registration (force=0) at
> src/switch_core_sqldb.c:1726
> #4 0x4013bf3c in switch_core_sql_db_thread (thread=<value optimized out>,
> obj=0x0) at src/switch_core_sqldb.c:926
> #5 0x401c93dc in dummy_worker (opaque=0x402493d4) at
> threadproc/unix/thread.c:138
> #6 0x402e48cc in start_thread () from /lib/libpthread.so.0
> #7 0x4058bbdc in clone () from /lib/libc.so.6
> #8 0x4058bbdc in clone () from /lib/libc.so.6
>
>
> FS crashed right after start!
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From lloyd.aloysius at gmail.com  Sat Feb 19 22:22:34 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sat, 19 Feb 2011 14:22:34 -0500
Subject: [Freeswitch-users] DTMF not working properly
Message-ID: <AANLkTimdFsRbQXBXZD8JgLCUXiJbsLSMGXfY6ob9cdgU@mail.gmail.com>

Hi All,

I have a DTMF  problem in version FreeSWITCH Version 1.0.head (git-39ff78b
2011-02-18 20-16-11 -0600)

Phone - Polycom 335 Connected to the FreeSWITCH. Dial out a External Number
. Then dial a Extension on the Destination IVR ( Destination is not
FreeSWITCH) ... most of the time FreeSWITCH not sending the correct digits.

I can see from the Console only the First Digit Captured by FreeSWITCH not
rest of the Digits this is happen randomly

I try with different SIP Trunk from different provider.. Same behavior

--- here is partial of DTMF first digit then no digits shows up on the CLI

2011-02-19 14:18:20.688643 [DEBUG] switch_rtp.c:2294 Send start packet for
[2] ts=1463986596 dur=160/160/2080 seq=54483
2011-02-19 14:18:20.708130 [DEBUG] switch_rtp.c:2230 Send middle packet for
[2] ts=1463986596 dur=320/320/2080 seq=54484
2011-02-19 14:18:20.728688 [DEBUG] switch_rtp.c:2230 Send middle packet for
[2] ts=1463986596 dur=480/480/2080 seq=54485
2011-02-19 14:18:20.748147 [DEBUG] switch_rtp.c:2230 Send middle packet for
[2] ts=1463986596 dur=640/640/2080 seq=54486
2011-02-19 14:18:20.768727 [DEBUG] switch_rtp.c:2230 Send middle packet for
[2] ts=1463986596 dur=800/800/2080 seq=54487
2011-02-19 14:18:20.788050 [DEBUG] switch_rtp.c:2230 Send middle packet for
[2] ts=1463986596 dur=960/960/2080 seq=54488
2011-02-19 14:18:20.808506 [DEBUG] switch_rtp.c:2230 Send middle packet for
[2] ts=1463986596 dur=1120/1120/2080 seq=54489
2011-02-19 14:18:20.828043 [DEBUG] switch_rtp.c:2230 Send middle packet for
[2] ts=1463986596 dur=1280/1280/2080 seq=54490
2011-02-19 14:18:20.848501 [DEBUG] switch_rtp.c:2230 Send middle packet for
[2] ts=1463986596 dur=1440/1440/2080 seq=54491
2011-02-19 14:18:20.867984 [DEBUG] switch_rtp.c:2230 Send middle packet for
[2] ts=1463986596 dur=1600/1600/2080 seq=54492
2011-02-19 14:18:20.888589 [DEBUG] switch_rtp.c:2230 Send middle packet for
[2] ts=1463986596 dur=1760/1760/2080 seq=54493
2011-02-19 14:18:20.908531 [DEBUG] switch_rtp.c:2230 Send middle packet for
[2] ts=1463986596 dur=1920/1920/2080 seq=54494
2011-02-19 14:18:20.927937 [DEBUG] switch_rtp.c:2230 Send end packet for [2]
ts=1463986596 dur=2080/2080/2080 seq=54495
2011-02-19 14:18:20.927937 [DEBUG] switch_rtp.c:2230 Send end packet for [2]
ts=1463986596 dur=2080/2080/2080 seq=54496
2011-02-19 14:18:20.927937 [DEBUG] switch_rtp.c:2230 Send end packet for [2]
ts=1463986596 dur=2080/2080/2080 seq=54497

---

Please let me know how to fix this or is it a Bug?


Thanks
Lloyd
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From sunwood360 at gmail.com  Sun Feb 20 01:07:08 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Sat, 19 Feb 2011 14:07:08 -0800
Subject: [Freeswitch-users] core dump in debian
In-Reply-To: <AANLkTimq=bHNgLK=NROOJz7ce76oA4cS1gM1W2Fk6ehj@mail.gmail.com>
References: <AANLkTi=YopVmLSTi9MwhpXEx-wpJ-nLmiB4jeBSszAz1@mail.gmail.com>
	<AANLkTimq=bHNgLK=NROOJz7ce76oA4cS1gM1W2Fk6ehj@mail.gmail.com>
Message-ID: <AANLkTik7J61+xbmn+-ExnDJCA=0MSM=_fQAQR3EbnLcO@mail.gmail.com>

I tried two times. one is 1.0.7 release, one is release from Git source. the
same issue of crash

my FS is running on dockstar.


On Sat, Feb 19, 2011 at 10:20 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Do a fresh checkout and remove previous binaries.
>  On Feb 18, 2011 9:57 PM, "envelopes envelopes" <sunwood360 at gmail.com>
> wrote:
> > #0 vxprintf (arg=0x40ac9be0, fmt=0x40249412 "q'", ap=<value optimized
> out>,
> > useExtended=<value optimized out>, func=<value optimized out>)
> > at src/switch_mprintf.c:686
> > #1 0x40130774 in base_vprintf (xRealloc=0x4012f328 <printf_realloc>,
> > zInitBuf=0xb <Address 0xb out of bounds>, nInitBuf=<value optimized out>,
> > zFormat=0xa <Address 0xa out of bounds>, ap=..., useInternal=<value
> > optimized out>) at src/switch_mprintf.c:849
> > #2 0x40130880 in switch_mprintf (zFormat=0x4027671c "debian") at
> > src/switch_mprintf.c:892
> > #3 0x4013ba48 in switch_core_expire_registration (force=0) at
> > src/switch_core_sqldb.c:1726
> > #4 0x4013bf3c in switch_core_sql_db_thread (thread=<value optimized out>,
> > obj=0x0) at src/switch_core_sqldb.c:926
> > #5 0x401c93dc in dummy_worker (opaque=0x402493d4) at
> > threadproc/unix/thread.c:138
> > #6 0x402e48cc in start_thread () from /lib/libpthread.so.0
> > #7 0x4058bbdc in clone () from /lib/libc.so.6
> > #8 0x4058bbdc in clone () from /lib/libc.so.6
> >
> >
> > FS crashed right after start!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sun Feb 20 03:30:30 2011
From: me at nevian.org (Serge Yuriev)
Date: Sun, 20 Feb 2011 03:30:30 +0300
Subject: [Freeswitch-users] cdr fields
In-Reply-To: <428041297202435@web1.yandex.ru>
References: <428041297202435@web1.yandex.ru>
Message-ID: <1017141298161830@web35.yandex.ru>

Hello,

Anyone?

09.02.2011, 01:00, "Serge Yuriev" <me at nevian.org>:
> Hello
>
> I noticed difference in cause codes written in csv CDR and RADIUS. Perhaps I need to change something in template?
> Current template is
> <template name="lanbilling">
> ${direction},${sip_network_ip},${user_name},${caller_id_number},${destination_number},${start_stamp},${answer_stamp},${end_stamp},${billsec},${hangup_cause_q850},${hangup_cause},${uuid},${bleg_uuid},${accountcode},${read_codec},${write_codec}
> </template>
>
> This writes
> inbound,109.173.67.229,nevian,nevian,79645835822,2011-02-09 00:18:52,,2011-02-09 00:20:00,0,16,NORMAL_CLEARING,a4da79c4-5356-43b8-894b-78627fb5e243,,nevian,G729,G729
>
> As you can see cause is 16 but in RADiUS is 27 and it's more accurate
>
> In mod_radius_cdr cause got from switch_channel_get_cause(channel)
>
> Please advice.
>
> --
> 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

-- 
wbr,
Serge


From me at nevian.org  Sun Feb 20 03:34:19 2011
From: me at nevian.org (Serge Yuriev)
Date: Sun, 20 Feb 2011 03:34:19 +0300
Subject: [Freeswitch-users] destination number variable
In-Reply-To: <AANLkTinNzpwDiiZHX8F_vCm2yf_mPMiQ+Y4DrC0w75wp@mail.gmail.com>
References: <283881297203422@web100.yandex.ru>
	<AANLkTinNzpwDiiZHX8F_vCm2yf_mPMiQ+Y4DrC0w75wp@mail.gmail.com>
Message-ID: <1171731298162059@web37.yandex.ru>

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From chat2jesse at gmail.com  Sun Feb 20 04:11:11 2011
From: chat2jesse at gmail.com (jesse)
Date: Sat, 19 Feb 2011 17:11:11 -0800
Subject: [Freeswitch-users]
	=?iso-8859-1?q?Newbie=B4s_question_about_FreeS?=
	=?iso-8859-1?q?witch=2E=2E=2E?=
In-Reply-To: <1298129542465-6043436.post@n2.nabble.com>
References: <AANLkTinf=z4JEp4G2xK_0n+u8v0OhR++GBvPzzbT1Y_-@mail.gmail.com>
	<1298129542465-6043436.post@n2.nabble.com>
Message-ID: <AANLkTinD08hCZ74SncUbR318agQnO1=JB_Oebc17EPn6@mail.gmail.com>

mazilo:

 have you tried to install FS on Debian of your DockStar? my FS (1.0.7 or
latest source)
always crashed right after start.

I installed your openwrt/FS image, it worked. but FS is 1.0.6,

  thanks


On Sat, Feb 19, 2011 at 7:32 AM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
> For your usage, it is possible to use your existing server to host
> freeswitch. My usage is pretty low, i.e. average of two lines usage, and my
> Seagate DockStar (used as a host to my freeswitch) only shows a usage of
> less than 10% of CPU resources. Assuming your usage is what you said and
> your existing server load is no more than 50%, all the expenses you will
> incur is probably to buy additional two ATA devices where each will have
> two
> FXS ports, i.e. Linksys/Cisco PAP2T (beware of a counterfeit PAP2T sold on
> e-bay), the new Obi products, etc. Personally, my preference is the Obi 110
> which comes with 1 FXO and 1 FXS port. The FXO port can be connected
> directly to your existing PBX system as an extension. This device also
> supports XMPP to use with Google Voice and is sold @US$49.99/each (MSRP
> price). Do a Google search to find out more information on this product.
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Newbie-s-question-about-FreeSwitch-tp6038911p6043436.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 me at nevian.org  Sun Feb 20 04:33:00 2011
From: me at nevian.org (Serge Yuriev)
Date: Sun, 20 Feb 2011 04:33:00 +0300
Subject: [Freeswitch-users] Codec negotiation, again
Message-ID: <1078011298165580@web71.yandex.ru>

Hello,

A offers g729, g711a, g711u
FS allows all of them
B offers g711a, g711u

Call fails with INCOMPATIBLE DESTINATION.

Cant understand why FS offers only ONE codec (g729) to B - wiki says it should offer ordered LIST of..
GIT few days old, default config

-- 
wbr,
Serge


From me at nevian.org  Sun Feb 20 04:41:10 2011
From: me at nevian.org (Serge Yuriev)
Date: Sun, 20 Feb 2011 04:41:10 +0300
Subject: [Freeswitch-users] How to reread GW config?
Message-ID: <265301298166070@web108.yandex.ru>

Hello,

I have changed few parameters on gw xml.
reloadxml - no success
sofia profile external rescan - no success
sofia profile external register <gw> - no success
sofia profile external restart - made things work

I think it's too hard to restart profile for committing configs..
I'm doing something wrong?

-- 
wbr,
Serge


From lloyd.aloysius at gmail.com  Sun Feb 20 05:12:47 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sat, 19 Feb 2011 21:12:47 -0500
Subject: [Freeswitch-users] How to reread GW config?
In-Reply-To: <265301298166070@web108.yandex.ru>
References: <265301298166070@web108.yandex.ru>
Message-ID: <AANLkTi=3n25A4-9AV58A2gScshemBRn7_ms68sNLj+Se@mail.gmail.com>

sofia profile external rescan reloadxml

Lloyd

On Sat, Feb 19, 2011 at 8:41 PM, Serge Yuriev <me at nevian.org> wrote:

> Hello,
>
> I have changed few parameters on gw xml.
> reloadxml - no success
> sofia profile external rescan - no success
> sofia profile external register <gw> - no success
> sofia profile external restart - made things work
>
> I think it's too hard to restart profile for committing configs..
> I'm doing something wrong?
>
> --
> 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 Nabble at slickdeals.endjunk.com  Sun Feb 20 05:34:28 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sat, 19 Feb 2011 18:34:28 -0800 (PST)
Subject: [Freeswitch-users]
	=?utf-8?q?Newbie=C2=B4s_question_about_FreeSwi?=
	=?utf-8?b?dGNoLi4u?=
In-Reply-To: <AANLkTinD08hCZ74SncUbR318agQnO1=JB_Oebc17EPn6@mail.gmail.com>
References: <AANLkTinf=z4JEp4G2xK_0n+u8v0OhR++GBvPzzbT1Y_-@mail.gmail.com>
	<1298129542465-6043436.post@n2.nabble.com>
	<AANLkTinD08hCZ74SncUbR318agQnO1=JB_Oebc17EPn6@mail.gmail.com>
Message-ID: <1298169268414-6044831.post@n2.nabble.com>



jesse zhao wrote:
> 
> mazilo:
> 
>  have you tried to install FS on Debian of your DockStar? my FS (1.0.7 or
> latest source) always crashed right after start.
Please read this  http://www.dslreports.com/forum/r25389970- post  and
ignore my posts after that one. It has something to do with the GCC word
alignment for ARM platform. I hope this will fix the problem you are
experiencing.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Newbie-s-question-about-FreeSwitch-tp6038911p6044831.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From yehavi.bourvine at gmail.com  Sun Feb 20 10:31:51 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 20 Feb 2011 09:31:51 +0200
Subject: [Freeswitch-users] Any experience with DTMF from FreeSwitch to
	Sonus with Vega ATAs?
Message-ID: <AANLkTin0UDw7Yyg-RWM-f6w3+or56qRwg_Y7AJxWL8h5@mail.gmail.com>

Hello,

  Anyone has an experience with the above configuration? All sides are
marked as RFC-2833 and the Sonus recognises most of the DTMF's twice.
The Vega sends both RFC-2833 events and about 60msec of the DTMF tone
(according to Vega's engineers it takes it about 60 msec to detect the DTMF
tones).

According to the RFC this is acceptable. The service provider I am working
with says that it shouldn't be like that. Before I waste time in wars, has
anybody had this issue and knows whether it is a matter of configuration at
the Sonus side?

                                   Thanks! __Yehavi:
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From krice at freeswitch.org  Sun Feb 20 10:44:53 2011
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 20 Feb 2011 01:44:53 -0600
Subject: [Freeswitch-users] Any experience with DTMF from FreeSwitch to
 Sonus with Vega ATAs?
In-Reply-To: <AANLkTin0UDw7Yyg-RWM-f6w3+or56qRwg_Y7AJxWL8h5@mail.gmail.com>
Message-ID: <C9862295.120E4%krice@freeswitch.org>

Sonus is known to do stupid crap with DTMF... Can you get a good pcap with a
full end to end sip and rtp capture?

K


On 2/20/11 1:31 AM, "Yehavi Bourvine" <yehavi.bourvine at gmail.com> wrote:

> Hello,
> ?
> ? Anyone has an experience with the above configuration? All sides are marked
> as RFC-2833 and the Sonus recognises?most of the DTMF's twice.
> The?Vega sends both RFC-2833 events and about 60msec of the DTMF?tone
> (according to Vega's engineers?it takes it about 60 msec to detect the DTMF
> tones).
> ?
> According to the RFC this is acceptable. The service provider I am working
> with?says that it shouldn't be like that. Before I waste time in wars, has
> anybody had this issue and knows whether it is a matter of configuration at
> the Sonus side?
> ?
> ?????????????????????????????????? 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  Sun Feb 20 13:04:59 2011
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 20 Feb 2011 18:04:59 +0800
Subject: [Freeswitch-users] Any experience with DTMF from FreeSwitch to
 Sonus with Vega ATAs?
In-Reply-To: <AANLkTin0UDw7Yyg-RWM-f6w3+or56qRwg_Y7AJxWL8h5@mail.gmail.com>
References: <AANLkTin0UDw7Yyg-RWM-f6w3+or56qRwg_Y7AJxWL8h5@mail.gmail.com>
Message-ID: <4D60E74B.102@coppice.org>

On 02/20/2011 03:31 PM, Yehavi Bourvine wrote:
> Hello,
>   Anyone has an experience with the above configuration? All sides are 
> marked as RFC-2833 and the Sonus recognises most of the DTMF's twice.
> The Vega sends both RFC-2833 events and about 60msec of the DTMF tone 
> (according to Vega's engineers it takes it about 60 msec to detect the 
> DTMF tones).
> According to the RFC this is acceptable. The service provider I am 
> working with says that it shouldn't be like that. Before I waste time 
> in wars, has anybody had this issue and knows whether it is a matter 
> of configuration at the Sonus side?
>                                    Thanks! __Yehavi:
Are you saying the Vega sends 60ms of DTMF as audio, then sends RFC2833 
DTMF packets and mutes the audio? In what reading of the RFC is that 
considered appropriate action? In addition, a DTMF detector should 
detect is more like 40ms. 60ms is rather long.

Steve



From yehavi.bourvine at gmail.com  Sun Feb 20 13:33:59 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 20 Feb 2011 12:33:59 +0200
Subject: [Freeswitch-users] Any experience with DTMF from FreeSwitch to
 Sonus with Vega ATAs?
In-Reply-To: <4D60E74B.102@coppice.org>
References: <AANLkTin0UDw7Yyg-RWM-f6w3+or56qRwg_Y7AJxWL8h5@mail.gmail.com>
	<4D60E74B.102@coppice.org>
Message-ID: <AANLkTinewBVZUahRCu_MQFkTXQ_jYWLrAg4AP3K+SU0g@mail.gmail.com>

Hello Steve,

 I uderstand from paragraph 3.2 in RFC-2833 that the sender may send some
audio of the DTMF while sending the events, until it revognises that this is
a DTMF tone. The 60mSec is what I've been told by Vega engineers.

                               Thanks! __Yehavi:

2011/2/20 Steve Underwood <steveu at coppice.org>

>  On 02/20/2011 03:31 PM, Yehavi Bourvine wrote:
> > Hello,
> >   Anyone has an experience with the above configuration? All sides are
> > marked as RFC-2833 and the Sonus recognises most of the DTMF's twice.
> > The Vega sends both RFC-2833 events and about 60msec of the DTMF tone
> > (according to Vega's engineers it takes it about 60 msec to detect the
> > DTMF tones).
> > According to the RFC this is acceptable. The service provider I am
> > working with says that it shouldn't be like that. Before I waste time
> > in wars, has anybody had this issue and knows whether it is a matter
> > of configuration at the Sonus side?
> >                                    Thanks! __Yehavi:
> Are you saying the Vega sends 60ms of DTMF as audio, then sends RFC2833
> DTMF packets and mutes the audio? In what reading of the RFC is that
> considered appropriate action? In addition, a DTMF detector should
> detect is more like 40ms. 60ms is rather long.
>
> 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 dujinfang at gmail.com  Sun Feb 20 14:20:09 2011
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 20 Feb 2011 19:20:09 +0800
Subject: [Freeswitch-users] escape in originate
Message-ID: <AANLkTi=EGjfNFKPYgsCYrw_-OyXAj+OpZOi4QSQWROTp@mail.gmail.com>

I want to put tone_stream in dialstring but don't know how to escape the comma.


originate  {some_var='tone_stream://%(2000,4000,440.0,480.0);loops=1'}portaudio/auto_answer
&echo
2011-02-20 19:13:09.516139 [DEBUG] switch_ivr_originate.c:1971
variable string 0 = [some_var=tone_stream://%(2000]
2011-02-20 19:13:09.516139 [DEBUG] switch_ivr_originate.c:1971
variable string 1 = [4000]
2011-02-20 19:13:09.516139 [DEBUG] switch_ivr_originate.c:1971
variable string 2 = [440.0]
2011-02-20 19:13:09.516139 [DEBUG] switch_ivr_originate.c:1971
variable string 3 = [480.0);loops=1]

There's a work around but it's not elegant.

http://lists.freeswitch.org/pipermail/freeswitch-users/2010-September/062753.html

Actually we may not need to escape the comma, the parser should keep
reading until found the second quote. Would this need a patch?

Thanks.

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


From mariusz_kolo at wp.pl  Sun Feb 20 14:32:56 2011
From: mariusz_kolo at wp.pl (=?ISO-8859-2?Q?Mariusz_Ko=B3odziejczyk?=)
Date: Sun, 20 Feb 2011 12:32:56 +0100
Subject: [Freeswitch-users] Strange warning in ftmod_sangoma_isdn
Message-ID: <4D60FBE8.4030504@wp.pl>

Hi,

I have sometimes strange warning in module ftmod_sangoma_isdn

....
2011-02-20 11:31:35.190942 [WARNING] ftmod_wanpipe.c:946 [s1c31][1:16] 
Failed to read from sangoma device: No buffer space available (-65)
2011-02-20 11:31:35.190942 [WARNING] ftdm_io.c:3756 [s1c31][1:16] raw 
I/O read filed
2011-02-20 11:31:35.190942 [WARNING] ftmod_sangoma_isdn.c:386 
[s1c31][1:16] Failed to read from channel
2011-02-20 11:31:36.002943 [WARNING] ftmod_wanpipe.c:946 [s1c31][1:16] 
Failed to read from sangoma device: No buffer space available (-65)
2011-02-20 11:31:36.002943 [WARNING] ftdm_io.c:3756 [s1c31][1:16] raw 
I/O read filed
2011-02-20 11:31:36.002943 [WARNING] ftmod_sangoma_isdn.c:386 
[s1c31][1:16] Failed to read from channel
2011-02-20 11:31:36.390942 [WARNING] ftmod_wanpipe.c:946 [s1c31][1:16] 
Failed to read from sangoma device: No buffer space available (-65)
2011-02-20 11:31:36.390942 [WARNING] ftdm_io.c:3756 [s1c31][1:16] raw 
I/O read filed
2011-02-20 11:31:36.390942 [WARNING] ftmod_sangoma_isdn.c:386 
[s1c31][1:16] Failed to read from channel
2011-02-20 11:31:36.998941 [INFO] ftmod_sangoma_isdn_stack_rcv.c:688 
[SNGISDN Q921] wp1: Protocol: T200 expired N200 times(15): T200 Timeout(260)
2011-02-20 11:31:36.998941 [INFO] ftmod_sangoma_isdn_stack_rcv.c:688 
[SNGISDN Q921] wp1: Protocol: SABME reset started(17): SABME tx to 
initate link reset(264)
2011-02-20 11:31:37.002947 [WARNING] ftmod_wanpipe.c:946 [s1c31][1:16] 
Failed to read from sangoma device: No buffer space available (-65)
2011-02-20 11:31:37.002947 [WARNING] ftdm_io.c:3756 [s1c31][1:16] raw 
I/O read filed
2011-02-20 11:31:37.002947 [WARNING] ftmod_sangoma_isdn.c:386 
[s1c31][1:16] Failed to read from channel
2011-02-20 11:31:38.002947 [WARNING] ftmod_wanpipe.c:946 [s1c31][1:16] 
Failed to read from sangoma device: No buffer space available (-65)
2011-02-20 11:31:38.002947 [WARNING] ftdm_io.c:3756 [s1c31][1:16] raw 
I/O read filed
2011-02-20 11:31:38.002947 [WARNING] ftmod_sangoma_isdn.c:386 
[s1c31][1:16] Failed to read from channel
2011-02-20 11:31:39.002947 [WARNING] ftmod_wanpipe.c:946 [s1c31][1:16] 
Failed to read from sangoma device: No buffer space available (-65)
2011-02-20 11:31:39.002947 [WARNING] ftdm_io.c:3756 [s1c31][1:16] raw 
I/O read filed
2011-02-20 11:31:39.002947 [WARNING] ftmod_sangoma_isdn.c:386 
[s1c31][1:16] Failed to read from channel
2011-02-20 11:31:40.002947 [WARNING] ftmod_wanpipe.c:946 [s1c31][1:16] 
Failed to read from sangoma device: No buffer space available (-65)
2011-02-20 11:31:40.002947 [WARNING] ftdm_io.c:3756 [s1c31][1:16] raw 
I/O read filed
2011-02-20 11:31:40.002947 [WARNING] ftmod_sangoma_isdn.c:386 
[s1c31][1:16] Failed to read from channel
2011-02-20 11:31:40.998941 [INFO] ftmod_sangoma_isdn_stack_rcv.c:688 
[SNGISDN Q921] wp1: Protocol: T200 expired N200 times(15): T200 Timeout(260)
2011-02-20 11:31:40.998941 [INFO] ftmod_sangoma_isdn_stack_rcv.c:688 
[SNGISDN Q921] wp1: Protocol: Data Link connection DN(4): Disconnect 
initiated(263)
....

this series of warning appearances almost every half hour, sometimes as 
single warning. this is'nt related to any monitoring params in system 
(for example 'vmstat 1' shows nothing special in logs just before i get 
this warning message, during and a bit after, for warning above vmstat 
shows (i added time stamp to the end every each rows):

procs -----------memory---------- ---swap-- -----io---- -system-- 
----cpu----
r  b   swpd   free   buff  cache   si   so    bi    bo   in   cs us sy id wa
0  0      0 4631820 167932 1087348    0    0     0     0  100 2697  0  0 
100  0 02-20-2011 11:31:30
0  0      0 4631944 167932 1087348    0    0     0     0  100 2693  0  0 
100  0 02-20-2011 11:31:31
0  0      0 4631820 167932 1087348    0    0     0     0  100 2687  0  0 
100  0 02-20-2011 11:31:32
0  0      0 4631944 167932 1087348    0    0     0    72  121 2711  0  0 
100  0 02-20-2011 11:31:33
0  0      0 4631944 167932 1087348    0    0     0     0  104 2713  0  0 
100  0 02-20-2011 11:31:34
0  0      0 4631944 167932 1087352    0    0     0     0  104 2703  0  0 
100  0 02-20-2011 11:31:35
0  0      0 4631944 167932 1087352    0    0     0     0  104 2716  0  0 
100  0 02-20-2011 11:31:36
0  0      0 4631944 167932 1087352    0    0     0   152  137 2773  0  0 
100  0 02-20-2011 11:31:37
0  0      0 4631944 167932 1087356    0    0     0     0  107 2708  0  0 
100  0 02-20-2011 11:31:38
0  0      0 4631820 167932 1087356    0    0     0     0  104 2700  0  0 
100  0 02-20-2011 11:31:39
0  0      0 4631820 167932 1087360    0    0     0     0  104 2701  0  0 
100  0 02-20-2011 11:31:40
0  0      0 4631820 167932 1087360    0    0     0     0  104 2719  0  0 
100  0 02-20-2011 11:31:41
0  0      0 4631820 167932 1087360    0    0     0   156  124 2736  0  0 
100  0 02-20-2011 11:31:42
0  0      0 4631696 167932 1087360    0    0     0     8  109 2697  0  0 
100  0 02-20-2011 11:31:43
0  0      0 4631696 167932 1087360    0    0     0     0  103 2695  0  0 
100  0 02-20-2011 11:31:44
0  0      0 4631696 167932 1087360    0    0     0     0  101 2684  0  0 
100  0 02-20-2011 11:31:45


When i've tested system i can make outgoing and incoming calls without 
any problem (max 3 calls simultaneously with different configuration, 
but i'm affraid if i use this in production environment (about 30-50 
calls via isdn) this warning may cause addiction problems.

I have installed:

freeswitch : FreeSWITCH Version 1.0.head (git-a3b18e5 2011-02-19 
16-47-19 +0100)
linux (Debian) : Linux PBX 2.6.26-2-amd64 #1 SMP Thu Sep 16 15:56:38 UTC 
2010 x86_64 GNU/Linux
wanrouter: WANPIPE Release: 3.5.18
libsng_isdn : libsng_isdn-7.3.0.x86_64
sangoma card: A102DE

Is anybody have the same warning ?

Thanks



From dmitry.bely at gmail.com  Sun Feb 20 15:17:48 2011
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Sun, 20 Feb 2011 15:17:48 +0300
Subject: [Freeswitch-users] local-network-acl parameter
Message-ID: <AANLkTi=+8QqhHzuO+ThNPRJdpc8c_4mB+jtVuHqd-DfX@mail.gmail.com>

Can you explain what objects is associated with this ACL? What does it
allow or deny? http://wiki.freeswitch.org/wiki/Acl is silent on that.

- Dmitry Bely


From steveu at coppice.org  Sun Feb 20 15:23:24 2011
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 20 Feb 2011 20:23:24 +0800
Subject: [Freeswitch-users] Any experience with DTMF from FreeSwitch to
 Sonus with Vega ATAs?
In-Reply-To: <AANLkTinewBVZUahRCu_MQFkTXQ_jYWLrAg4AP3K+SU0g@mail.gmail.com>
References: <AANLkTin0UDw7Yyg-RWM-f6w3+or56qRwg_Y7AJxWL8h5@mail.gmail.com>	<4D60E74B.102@coppice.org>
	<AANLkTinewBVZUahRCu_MQFkTXQ_jYWLrAg4AP3K+SU0g@mail.gmail.com>
Message-ID: <4D6107BC.4090906@coppice.org>

Hi Yehavi,

Section 3.2 of RFC2833 is poorly worded, and causes a lot of issues in 
the real world. The first paragraph says you can choose to send both 
audio and events, but doesn't clarify how. Presumably RFC2198 should be 
used. Is your system using RFC2198 encoding? I doubt it. The second 
paragraph talks about tone onset getting through as audio. Valid DTMF 
can be as short as 45ms, so 60ms is far more than anything which might 
be termed onset. Because they didn't specify anything about what is 
tolerable as tone onset passing through the channel as audio we have 
chaos today.

RFC2833 is an obsolete spec, and we should be talking about RFC4733 
today. Section 2.5.1.3.1 of RFC4733 does clearly call for combined 
payloads to be sent as RFC2198 packets. RFC4733 is even vaguer about the 
onset issue, though.

Steve


On 02/20/2011 06:33 PM, Yehavi Bourvine wrote:
> Hello Steve,
>  I uderstand from paragraph 3.2 in RFC-2833 that the sender may send 
> some audio of the DTMF while sending the events, until it revognises 
> that this is a DTMF tone. The 60mSec is what I've been told by Vega 
> engineers.
>                                Thanks! __Yehavi:
>
> 2011/2/20 Steve Underwood <steveu at coppice.org <mailto:steveu at coppice.org>>
>
>     On 02/20/2011 03:31 PM, Yehavi Bourvine wrote:
>     > Hello,
>     >   Anyone has an experience with the above configuration? All
>     sides are
>     > marked as RFC-2833 and the Sonus recognises most of the DTMF's
>     twice.
>     > The Vega sends both RFC-2833 events and about 60msec of the DTMF
>     tone
>     > (according to Vega's engineers it takes it about 60 msec to
>     detect the
>     > DTMF tones).
>     > According to the RFC this is acceptable. The service provider I am
>     > working with says that it shouldn't be like that. Before I waste
>     time
>     > in wars, has anybody had this issue and knows whether it is a matter
>     > of configuration at the Sonus side?
>     >                                    Thanks! __Yehavi:
>     Are you saying the Vega sends 60ms of DTMF as audio, then sends
>     RFC2833
>     DTMF packets and mutes the audio? In what reading of the RFC is that
>     considered appropriate action? In addition, a DTMF detector should
>     detect is more like 40ms. 60ms is rather long.
>
>     Steve
>
>
>     _______________________________________________
>     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
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From lloydie.t at gmail.com  Sun Feb 20 15:31:08 2011
From: lloydie.t at gmail.com (lloyd thomas)
Date: Sun, 20 Feb 2011 12:31:08 +0000
Subject: [Freeswitch-users] Mod_callcenter and Multi-Tenancy
Message-ID: <AANLkTinUJyWTM2J3hWxaob=AbN0H8SfqNt07TtScSGNA@mail.gmail.com>

Hi,
 Can mod_callcenter be used in a multi-tenancy enviroment and with
mod_xml_curl?
Can you clarify what the following line means (Whether Multi-Tenancy or
Multiple FS servers)?
<!-- WARNING: Agents and Tiers XML config shouldn't be used in a multi FS
shared DB setup. -->
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From steveayre at gmail.com  Sun Feb 20 15:33:48 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 20 Feb 2011 12:33:48 +0000
Subject: [Freeswitch-users] How to reread GW config?
In-Reply-To: <265301298166070@web108.yandex.ru>
References: <265301298166070@web108.yandex.ru>
Message-ID: <0A4BCDDD-A478-41E3-90AE-85B01F654DA4@gmail.com>

First remove the gateway:
sofia profile profilename killgw gwname

Wait a few seconds until the gateway is deleted (there's a log entry).

Then reload the profile which creates any previously unknown gateways:
sofia profile profilename rescan reloadxml

Steve on iPhone


On 20 Feb 2011, at 01:41, Serge Yuriev <me at nevian.org> wrote:

> Hello,
> 
> I have changed few parameters on gw xml.
> reloadxml - no success
> sofia profile external rescan - no success
> sofia profile external register <gw> - no success
> sofia profile external restart - made things work
> 
> I think it's too hard to restart profile for committing configs..
> I'm doing something wrong?
> 
> -- 
> 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 steveayre at gmail.com  Sun Feb 20 15:36:45 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 20 Feb 2011 12:36:45 +0000
Subject: [Freeswitch-users] Codec negotiation, again
In-Reply-To: <1078011298165580@web71.yandex.ru>
References: <1078011298165580@web71.yandex.ru>
Message-ID: <7DBE786F-4C0A-406F-AEE6-186D9112BC5A@gmail.com>

In default behaviour, FS will negotiate the codec for the aleg, then do the dialplan, then do the bleg. As a result it can have already picked the 729 codec by the time it hits the dialplan. With transcoding that's fine but with 729 not.

The workaround is the late-negotiation profile parameter. It delays picking the aleg codec until the bleg picks one, then uses that if possible for the aleg.

Steve on iPhone


On 20 Feb 2011, at 01:33, Serge Yuriev <me at nevian.org> wrote:

> Hello,
> 
> A offers g729, g711a, g711u
> FS allows all of them
> B offers g711a, g711u
> 
> Call fails with INCOMPATIBLE DESTINATION.
> 
> Cant understand why FS offers only ONE codec (g729) to B - wiki says it should offer ordered LIST of..
> GIT few days old, default config
> 
> -- 
> 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


From yehavi.bourvine at gmail.com  Sun Feb 20 15:39:18 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 20 Feb 2011 14:39:18 +0200
Subject: [Freeswitch-users] Any experience with DTMF from FreeSwitch to
 Sonus with Vega ATAs?
In-Reply-To: <4D6107BC.4090906@coppice.org>
References: <AANLkTin0UDw7Yyg-RWM-f6w3+or56qRwg_Y7AJxWL8h5@mail.gmail.com>
	<4D60E74B.102@coppice.org>
	<AANLkTinewBVZUahRCu_MQFkTXQ_jYWLrAg4AP3K+SU0g@mail.gmail.com>
	<4D6107BC.4090906@coppice.org>
Message-ID: <AANLkTimcgsTVZfo4CWZQt1pNg3Z+hybPhTW2OGHqm-qT@mail.gmail.com>

Hello Steve,

  Thanks for clarifying it.

                     __Yehavi:

2011/2/20 Steve Underwood <steveu at coppice.org>

> Hi Yehavi,
>
> Section 3.2 of RFC2833 is poorly worded, and causes a lot of issues in
> the real world. The first paragraph says you can choose to send both
> audio and events, but doesn't clarify how. Presumably RFC2198 should be
> used. Is your system using RFC2198 encoding? I doubt it. The second
> paragraph talks about tone onset getting through as audio. Valid DTMF
> can be as short as 45ms, so 60ms is far more than anything which might
> be termed onset. Because they didn't specify anything about what is
> tolerable as tone onset passing through the channel as audio we have
> chaos today.
>
> RFC2833 is an obsolete spec, and we should be talking about RFC4733
> today. Section 2.5.1.3.1 of RFC4733 does clearly call for combined
> payloads to be sent as RFC2198 packets. RFC4733 is even vaguer about the
> onset issue, though.
>
> Steve
>
>
> On 02/20/2011 06:33 PM, Yehavi Bourvine wrote:
> > Hello Steve,
> >  I uderstand from paragraph 3.2 in RFC-2833 that the sender may send
> > some audio of the DTMF while sending the events, until it revognises
> > that this is a DTMF tone. The 60mSec is what I've been told by Vega
> > engineers.
> >                                Thanks! __Yehavi:
> >
> > 2011/2/20 Steve Underwood <steveu at coppice.org <mailto:steveu at coppice.org
> >>
>  >
> >     On 02/20/2011 03:31 PM, Yehavi Bourvine wrote:
> >     > Hello,
> >     >   Anyone has an experience with the above configuration? All
> >     sides are
> >     > marked as RFC-2833 and the Sonus recognises most of the DTMF's
> >     twice.
> >     > The Vega sends both RFC-2833 events and about 60msec of the DTMF
> >     tone
> >     > (according to Vega's engineers it takes it about 60 msec to
> >     detect the
> >     > DTMF tones).
> >     > According to the RFC this is acceptable. The service provider I am
> >     > working with says that it shouldn't be like that. Before I waste
> >     time
> >     > in wars, has anybody had this issue and knows whether it is a
> matter
> >     > of configuration at the Sonus side?
> >     >                                    Thanks! __Yehavi:
> >     Are you saying the Vega sends 60ms of DTMF as audio, then sends
> >     RFC2833
> >     DTMF packets and mutes the audio? In what reading of the RFC is that
> >     considered appropriate action? In addition, a DTMF detector should
> >     detect is more like 40ms. 60ms is rather long.
> >
> >     Steve
> >
> >
> >     _______________________________________________
> >     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
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From hareem.haque at gmail.com  Sun Feb 20 19:08:08 2011
From: hareem.haque at gmail.com (Hareem Haque)
Date: Sun, 20 Feb 2011 11:08:08 -0500
Subject: [Freeswitch-users] Need help with CDR setup and call routing
	patterns
Message-ID: <AANLkTinZUd=YnvkqjO9Bx44isreeCCzBHB26yf_rkzo=@mail.gmail.com>

Thank you very much for helping me out with the initial switch setup. I
really appreciate it. However, I am stuck with the cdr and call routing
patterns setup.


A. I have attached my cdr_csv file and its cdr log file. As you may see in
the conf xml that i have 22 items that i need posted onto my cdr logs.
However, the csv file only shows 15 items. How can i make this work

B. I have setup a few sip trunks. Now how do i set them up so that if i dial
01192 that call goes to gateway A and if i dial 1XXX then that call goes to
gateway B

Many thanks for all the help. I really appreciate it

Best Regards
Hareem. Haque
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From steveayre at gmail.com  Sun Feb 20 20:01:47 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 20 Feb 2011 17:01:47 +0000
Subject: [Freeswitch-users] Need help with CDR setup and call routing
	patterns
In-Reply-To: <AANLkTinZUd=YnvkqjO9Bx44isreeCCzBHB26yf_rkzo=@mail.gmail.com>
References: <AANLkTinZUd=YnvkqjO9Bx44isreeCCzBHB26yf_rkzo=@mail.gmail.com>
Message-ID: <01D5BE07-B144-4A68-86B6-7E0FFD6FB898@gmail.com>

After changing the config, did you reload the config?

reloadxml
reload mod_cdr_csv

Steve on iPhone



On 20 Feb 2011, at 16:08, Hareem Haque <hareem.haque at gmail.com> wrote:

> Thank you very much for helping me out with the initial switch setup. I really appreciate it. However, I am stuck with the cdr and call routing patterns setup. 
> 
> 
> A. I have attached my cdr_csv file and its cdr log file. As you may see in the conf xml that i have 22 items that i need posted onto my cdr logs. However, the csv file only shows 15 items. How can i make this work
> 
> B. I have setup a few sip trunks. Now how do i set them up so that if i dial 01192 that call goes to gateway A and if i dial 1XXX then that call goes to gateway B
> 
> Many thanks for all the help. I really appreciate it
> 
> Best Regards
> Hareem. Haque
> 
> 
> <cdr_csv.conf.xml>
> <hareem.csv>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Sun Feb 20 20:10:34 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 20 Feb 2011 17:10:34 +0000
Subject: [Freeswitch-users] Need help with CDR setup and call routing
	patterns
In-Reply-To: <AANLkTinZUd=YnvkqjO9Bx44isreeCCzBHB26yf_rkzo=@mail.gmail.com>
References: <AANLkTinZUd=YnvkqjO9Bx44isreeCCzBHB26yf_rkzo=@mail.gmail.com>
Message-ID: <AANLkTinBFEsafzBa7y_7ffNZPKvqkdZ40cx4cWS8SO77@mail.gmail.com>

>
> B. I have setup a few sip trunks. Now how do i set them up so that if i
> dial 01192 that call goes to gateway A and if i dial 1XXX then that call
> goes to gateway B
>

Handle each in a different extension in the dialplan:

<extension name="route1">
   <condition field="destination_number" expression="^(01192)$">
      <action application="bridge" data="sofia/gateway/gatewaya/$1"/>
   </condition>
</extension>
<extension name="route2">
   <condition field="destination_number" expression="^(1\d\d\d)$">
      <action application="bridge" data="sofia/gateway/gatewayb/$1"/>
   </condition>
</extension>

Adjust the expressions to suit the numbers you want to handle better. The
contents of the () brackets are placed in variable $1, which is being used
in the bridge to dial that number via the gateway. You can add prefixes etc
there if you need them. ^ and $ match the start and end of the string, not
actual characters. They're called regular expressions if you want to read up
on them, and are more flexible than just matching absolute strings because
you match patterns instead.

An example of changing a prefix, should your gateway need that (for example
converting a local number to an international prefix):
   <condition field="destination_number" expression="^0(1192)$">
      <action application="bridge" data="sofia/gateway/gatewaya/44$1"/>
   </condition>
01192 would be dialed as 441192 through gateway a.

-Steve




On 20 February 2011 16:08, Hareem Haque <hareem.haque at gmail.com> wrote:

> Thank you very much for helping me out with the initial switch setup. I
> really appreciate it. However, I am stuck with the cdr and call routing
> patterns setup.
>
>
> A. I have attached my cdr_csv file and its cdr log file. As you may see in
> the conf xml that i have 22 items that i need posted onto my cdr logs.
> However, the csv file only shows 15 items. How can i make this work
>
> B. I have setup a few sip trunks. Now how do i set them up so that if i
> dial 01192 that call goes to gateway A and if i dial 1XXX then that call
> goes to gateway B
>
> Many thanks for all the help. I really appreciate it
>
> Best Regards
> Hareem. Haque
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sun Feb 20 21:41:10 2011
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Sun, 20 Feb 2011 19:41:10 +0100
Subject: [Freeswitch-users] ODBC error STATE: HY000 CODE 21 ERROR
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE07872CFA89@FS1.wvds.local>

Hi guys,



At the moment I'm doing some testing with mod_cidlookup. My intention is to match the incoming numbers against a local database. Right now im trying to set it up with SQLite.

I've installed SQLite, unixODBC and created a DSN (freeswitch is also compiled with odbc support). When testing the connection to the SQLite database with the isql utility everything works:



+---------------------------------------+
| Connected!                            |
|                                       |
| sql-statement                         |
| help [tablename]                      |
| quit                                  |
|                                       |
+---------------------------------------+
SQL>





Unfortunately when starting freeswitch I get this error in the console:



2011-02-20 19:28:44.203665 [ERR] switch_odbc.c:365 STATE: HY000 CODE 21 ERROR: [unixODBC][SQLite]connect failed

2011-02-20 19:28:44.203672 [CRIT] switch_core_sqldb.c:386 Failure!
2011-02-20 19:28:44.203677 [CRIT] mod_cidlookup.c:137 Cannot Open ODBC Database!





Here's my config from /conf/autoload_configs/cidlookup.conf.xml:



<configuration name="cidlookup.conf" description="cidlookup Configuration">
  <settings>
    <param name="cache" value="false"/>
    <!-- expire is in seconds -->
    <param name="cache-expire" value="86400"/>
    <param name="odbc-dsn" value="mysqlitedb"/>

    <!-- comment out sql to not setup a database (directory) lookup -->
    <param name="sql" value="
     SELECT name||' ('||type||')' AS name
      FROM phonebook p JOIN numbers n ON p.id = n.phonebook_id
      WHERE n.number='${caller_id_number}'
      LIMIT 1
      "/>
  </settings>
</configuration>


I someone could point me into the right direction that would be great.

Kind regards,

Erik Dekkers (wvds-nl)
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From mbsip at gazeta.pl  Sun Feb 20 21:51:47 2011
From: mbsip at gazeta.pl (Mac)
Date: Sun, 20 Feb 2011 19:51:47 +0100
Subject: [Freeswitch-users] RTP Proxy mode, DTMF conversion problem
Message-ID: <AANLkTimUvhO65MTg7DaMRbbzO1M5bz4oKg5JfnR0_6uC@mail.gmail.com>

Dear ALL,

I am struggling for some time with DTMF issue.
Let me introduce you my configuration.

The main task is to configure RTP Proxy with full topology hiding -
OPERATOR_1 ------- Freeswitch ------- OPERATOR_2
Here is a prt of my config:
- sip profile.
I have one new sip profile with two gateways (OPERATOR_1 and OPERATOR_2)
<include>
  <gateway name="OP1">
    <param name="username" value="X1.Y1.V1.Z1"/>
    <param name="password" value="09321234"/>
    <param name="proxy" value="X1.Y1.V1.Z1"/>
    <param name="from-domain" value="test.com"/>
    <param name="register" value="false"/>
    <param name="caller-id-in-from" value="true"/>
    <param name="extension" value="test"/>
    <param name="extension-in-contact" value="true"/>
    <param name="outbound-codec-prefs" value="$${outbound_codec_
prefs}"/>
    <param name="disable-transcoding" value="true"/>
    <param name="inbound-late-negotiation" value="true"/>
  </gateway>
</include>

- dialplan
<include>
  <context name="public">
        <extension name="term">
                <condition field="network_addr"
expression="^(X1\.Y1\.V1\.Z1|X2\.Y2\.V1\.Z1)$" break="on-false"/>
                <condition field="destination_number" expression="^46(\d{9})$">
                <action application="set" data="proxy_media=true"/>
                <action application="set" data="bypass_media=false"/>
                <action application="start_dtmf"/>
                <action application="bridge" data="sofia/gateway/OP1/$0"/>
                </condition>
        </extension>
  </context>
</include>

Everything is fine, but i have problem with DTMF conversion from
RFC2833 to inband.
Refering to http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf
and http://wiki.freeswitch.org/wiki/Proxy_Media i have done all the
necessary things.

vars.xml
  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,G.729,PCMU,GSM"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA,G.729,PCMU,GSM"/>

The after-effect is following output:
[ERR] switch_core_io.c:1055 Codec PROXY PASS-THROUGH encoder error!

Placing  <param name="dtmf-type" value="none"/> does not help anyway.

Could sb point me where the problem is located?

Thanks in advance,
Mac


From steveayre at gmail.com  Sun Feb 20 22:46:16 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 20 Feb 2011 19:46:16 +0000
Subject: [Freeswitch-users] RTP Proxy mode, DTMF conversion problem
In-Reply-To: <AANLkTimUvhO65MTg7DaMRbbzO1M5bz4oKg5JfnR0_6uC@mail.gmail.com>
References: <AANLkTimUvhO65MTg7DaMRbbzO1M5bz4oKg5JfnR0_6uC@mail.gmail.com>
Message-ID: <12CA5F58-378A-4231-9E1E-3F89DD59049B@gmail.com>

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

As indicated by the error, this is the problem. "proxy passthrough". In proxy mode you pass the media straight through (passthrough). You can't process passthrough media, such as is needed to mix in inband dtmf.

Use proxy_media=false and bypass_media=false. That's actually the default so unless you're setting either to true in the sip profile, you can remove those lines from the dialplan completely.

Steve on iPhone

On 20 Feb 2011, at 18:51, Mac <mbsip at gazeta.pl> wrote:

> action application="set" data="proxy_media=true"/>
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From essobi at gmail.com  Sun Feb 20 23:40:45 2011
From: essobi at gmail.com (Essobi)
Date: Sun, 20 Feb 2011 15:40:45 -0500
Subject: [Freeswitch-users] (no subject)
Message-ID: <AANLkTikUsVJbB+bwsfe6n60jw5mOCja=WnnCMzbosJNJ@mail.gmail.com>

 http://hiawatha-von-hesperus.com/cool01.11.php?ID=731


From mthakershi at gmail.com  Sun Feb 20 23:51:33 2011
From: mthakershi at gmail.com (Malay Thakershi)
Date: Sun, 20 Feb 2011 14:51:33 -0600
Subject: [Freeswitch-users] Using 16 KHz sounds
Message-ID: <AANLkTim5CsCL4unpWh1OvH13Xx4HrKX2KF7Jg6UQYAZR@mail.gmail.com>

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
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From mbsip at gazeta.pl  Mon Feb 21 00:01:31 2011
From: mbsip at gazeta.pl (Mac)
Date: Sun, 20 Feb 2011 22:01:31 +0100
Subject: [Freeswitch-users] RTP Proxy mode, DTMF conversion problem
In-Reply-To: <12CA5F58-378A-4231-9E1E-3F89DD59049B@gmail.com>
References: <AANLkTimUvhO65MTg7DaMRbbzO1M5bz4oKg5JfnR0_6uC@mail.gmail.com>
	<12CA5F58-378A-4231-9E1E-3F89DD59049B@gmail.com>
Message-ID: <AANLkTikABFQ4hwj9+dkUyKGTU02+dL4gY8DYdRH8RNhz@mail.gmail.com>

Hi Steven,

There is much better :)
I have different SDPs right now. The one is with rtpmap:101
telephone-event/8000 (leg that is working with RFC2833), the opposite
one incorporates a=rtpmap:101 telephone-event/8000 in INVITE SDP and
pure codec rtpmaps.
There is rtp.p_type == 101 working only on the left side. I cannot
find any rtp.p_type == 101 on thark in opposide side which is okay.
But the problem still persists.

Once setting my UA to work with Inband DTMF everything works fine.

Thanks,
Mac

2011/2/20 Steven Ayre <steveayre at gmail.com>:
> <action application="set" data="proxy_media=true"/>
>
> As indicated by the error, this is the problem. "proxy passthrough". In
> proxy mode you pass the media straight through (passthrough). You can't
> process passthrough media, such as is needed to mix in inband dtmf.
> Use proxy_media=false and bypass_media=false. That's actually the default so
> unless you're setting either to true in the sip profile, you can remove
> those lines from the dialplan completely.
> Steve on iPhone
> On 20 Feb 2011, at 18:51, Mac <mbsip at gazeta.pl> wrote:
>
> action application="set" data="proxy_media=true"/>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 essobi at gmail.com  Mon Feb 21 00:28:56 2011
From: essobi at gmail.com (Essobi)
Date: Sun, 20 Feb 2011 16:28:56 -0500
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <AANLkTikUsVJbB+bwsfe6n60jw5mOCja=WnnCMzbosJNJ@mail.gmail.com>
References: <AANLkTikUsVJbB+bwsfe6n60jw5mOCja=WnnCMzbosJNJ@mail.gmail.com>
Message-ID: <AANLkTinOHXu+y-jq62zK=6griDF5P0Ra8cQyuoNokhYK@mail.gmail.com>

Someone hacked my gmail account.  It's a virus, don't click it.  The login
was from Brazil, I'm reporting it to authorities and gmail now.


On Sun, Feb 20, 2011 at 3:40 PM, Essobi <essobi at gmail.com> wrote:

>  http://hiawatha-von-hesperus.com/cool01.11.php?ID=731
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 21 00:34:23 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 20 Feb 2011 13:34:23 -0800
Subject: [Freeswitch-users] ODBC error STATE: HY000 CODE 21 ERROR
In-Reply-To: <CDF5E00590C9BA4AB96BE7CDD28C86AE07872CFA89@FS1.wvds.local>
References: <CDF5E00590C9BA4AB96BE7CDD28C86AE07872CFA89@FS1.wvds.local>
Message-ID: <AANLkTikC16qswxL-PnrUAqmrLKBKigBULG2TmjE7PZr7@mail.gmail.com>

sqlite should be built in freeswitch without needing unixODBC. There's
no need to use unixODBC unless you're trying to use MySQL or MSSQL
with FreeSwitch, iirc.

On Sun, Feb 20, 2011 at 10:41 AM, Erik Dekkers <erik.dekkers at wvds.nl> wrote:
> Hi guys,
>
>
>
> At the moment I'm doing some testing with mod_cidlookup. My intention is to
> match the incoming numbers against a local database. Right now im trying to
> set it up with SQLite.
>
> I've installed SQLite, unixODBC and created a DSN (freeswitch is also
> compiled with odbc support). When testing the connection to the SQLite
> database with the isql utility everything works:
>
>
>
> +---------------------------------------+
> | Connected!??????????????????????????? |
> |?????????????????????????????????????? |
> | sql-statement???????????????????????? |
> | help [tablename]????????????????????? |
> | quit????????????????????????????????? |
> |?????????????????????????????????????? |
> +---------------------------------------+
> SQL>
>
>
>
>
>
> Unfortunately when starting freeswitch I get this error in the console:
>
>
>
> 2011-02-20 19:28:44.203665 [ERR] switch_odbc.c:365 STATE: HY000 CODE 21
> ERROR: [unixODBC][SQLite]connect failed
>
> 2011-02-20 19:28:44.203672 [CRIT] switch_core_sqldb.c:386 Failure!
> 2011-02-20 19:28:44.203677 [CRIT] mod_cidlookup.c:137 Cannot Open ODBC
> Database!
>
>
>
>
>
> Here's my config from /conf/autoload_configs/cidlookup.conf.xml:
>
>
>
> <configuration name="cidlookup.conf" description="cidlookup Configuration">
> ? <settings>
> ??? <param name="cache" value="false"/>
> ??? <!-- expire is in seconds -->
> ??? <param name="cache-expire" value="86400"/>
> ??? <param name="odbc-dsn" value="mysqlitedb"/>
>
> ??? <!-- comment out sql to not setup a database (directory) lookup -->
> ??? <param name="sql" value="
> ???? SELECT name||' ('||type||')' AS name
> ????? FROM phonebook p JOIN numbers n ON p.id = n.phonebook_id
> ????? WHERE n.number='${caller_id_number}'
> ????? LIMIT 1
> ????? "/>
> ? </settings>
> </configuration>
>
> I someone could point me into the right direction that would be great.
>
> Kind regards,
>
> Erik Dekkers (wvds-nl)
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 nazim.aghabayov at gmail.com  Mon Feb 21 00:42:50 2011
From: nazim.aghabayov at gmail.com (Nazim Aghabayov)
Date: Mon, 21 Feb 2011 01:42:50 +0400
Subject: [Freeswitch-users] cdr fields
In-Reply-To: <1017141298161830@web35.yandex.ru>
References: <428041297202435@web1.yandex.ru> <1017141298161830@web35.yandex.ru>
Message-ID: <4D618ADA.2050607@gmail.com>

Hello,
I'm using mod_xml_cdr and xmlcdrd: http://wiki.freeswitch.org/wiki/Xmlcdrd .
You would not get the differences with it, because it's parsing the 
"standard" cdr.
Xmlcdrd finds necessary fields inside the cdr given the Xpath expression 
and then packs these fields into a radius stop packet.
Optionally, fields could be inserted into a mysql table or fed to a Lua 
script.
At the moment I'm using it in production and it works fine for me.
That's not the answer for your question, just a workaround which may (or 
may not) help.

Regards,
Nazim


On 02/20/2011 04:30 AM, Serge Yuriev wrote:
> Hello,
>
> Anyone?
>
> 09.02.2011, 01:00, "Serge Yuriev"<me at nevian.org>:
>> Hello
>>
>> I noticed difference in cause codes written in csv CDR and RADIUS. Perhaps I need to change something in template?
>> Current template is
>> <template name="lanbilling">
>> ${direction},${sip_network_ip},${user_name},${caller_id_number},${destination_number},${start_stamp},${answer_stamp},${end_stamp},${billsec},${hangup_cause_q850},${hangup_cause},${uuid},${bleg_uuid},${accountcode},${read_codec},${write_codec}
>> </template>
>>
>> This writes
>> inbound,109.173.67.229,nevian,nevian,79645835822,2011-02-09 00:18:52,,2011-02-09 00:20:00,0,16,NORMAL_CLEARING,a4da79c4-5356-43b8-894b-78627fb5e243,,nevian,G729,G729
>>
>> As you can see cause is 16 but in RADiUS is 27 and it's more accurate
>>
>> In mod_radius_cdr cause got from switch_channel_get_cause(channel)
>>
>> Please advice.
>>
>> --
>> wbr,
>> Serge


From steveayre at gmail.com  Mon Feb 21 03:02:44 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 21 Feb 2011 00:02:44 +0000
Subject: [Freeswitch-users] cdr fields
In-Reply-To: <1017141298161830@web35.yandex.ru>
References: <428041297202435@web1.yandex.ru> <1017141298161830@web35.yandex.ru>
Message-ID: <0A24684A-98CF-44BD-B76F-496845CEE745@gmail.com>

Is it possible one cdr is for the aleg and the other for the bleg? If they're the same channel they should both be the same value...

Steve on iPhone

On 20 Feb 2011, at 00:30, Serge Yuriev <me at nevian.org> wrote:

> Hello,
> 
> Anyone?
> 
> 09.02.2011, 01:00, "Serge Yuriev" <me at nevian.org>:
>> Hello
>> 
>> I noticed difference in cause codes written in csv CDR and RADIUS. Perhaps I need to change something in template?
>> Current template is
>> <template name="lanbilling">
>> ${direction},${sip_network_ip},${user_name},${caller_id_number},${destination_number},${start_stamp},${answer_stamp},${end_stamp},${billsec},${hangup_cause_q850},${hangup_cause},${uuid},${bleg_uuid},${accountcode},${read_codec},${write_codec}
>> </template>
>> 
>> This writes
>> inbound,109.173.67.229,nevian,nevian,79645835822,2011-02-09 00:18:52,,2011-02-09 00:20:00,0,16,NORMAL_CLEARING,a4da79c4-5356-43b8-894b-78627fb5e243,,nevian,G729,G729
>> 
>> As you can see cause is 16 but in RADiUS is 27 and it's more accurate
>> 
>> In mod_radius_cdr cause got from switch_channel_get_cause(channel)
>> 
>> Please advice.
>> 
>> --
>> 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
> 
> -- 
> 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


From steveayre at gmail.com  Mon Feb 21 03:03:55 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 21 Feb 2011 00:03:55 +0000
Subject: [Freeswitch-users] ODBC error STATE: HY000 CODE 21 ERROR
In-Reply-To: <AANLkTikC16qswxL-PnrUAqmrLKBKigBULG2TmjE7PZr7@mail.gmail.com>
References: <CDF5E00590C9BA4AB96BE7CDD28C86AE07872CFA89@FS1.wvds.local>
	<AANLkTikC16qswxL-PnrUAqmrLKBKigBULG2TmjE7PZr7@mail.gmail.com>
Message-ID: <533D2543-EC11-4945-9100-45E9E6520AE6@gmail.com>

Correct. Sqlite doesn't use odbc, you only need unixodbc if you want to connect via odbc to mysql, mssql, postgresql etc.

Steve on iPhone

On 20 Feb 2011, at 21:34, curriegrad2004 <curriegrad2004 at gmail.com> wrote:

> sqlite should be built in freeswitch without needing unixODBC. There's
> no need to use unixODBC unless you're trying to use MySQL or MSSQL
> with FreeSwitch, iirc.
> 
> On Sun, Feb 20, 2011 at 10:41 AM, Erik Dekkers <erik.dekkers at wvds.nl> wrote:
>> Hi guys,
>> 
>> 
>> 
>> At the moment I'm doing some testing with mod_cidlookup. My intention is to
>> match the incoming numbers against a local database. Right now im trying to
>> set it up with SQLite.
>> 
>> I've installed SQLite, unixODBC and created a DSN (freeswitch is also
>> compiled with odbc support). When testing the connection to the SQLite
>> database with the isql utility everything works:
>> 
>> 
>> 
>> +---------------------------------------+
>> | Connected!                            |
>> |                                       |
>> | sql-statement                         |
>> | help [tablename]                      |
>> | quit                                  |
>> |                                       |
>> +---------------------------------------+
>> SQL>
>> 
>> 
>> 
>> 
>> 
>> Unfortunately when starting freeswitch I get this error in the console:
>> 
>> 
>> 
>> 2011-02-20 19:28:44.203665 [ERR] switch_odbc.c:365 STATE: HY000 CODE 21
>> ERROR: [unixODBC][SQLite]connect failed
>> 
>> 2011-02-20 19:28:44.203672 [CRIT] switch_core_sqldb.c:386 Failure!
>> 2011-02-20 19:28:44.203677 [CRIT] mod_cidlookup.c:137 Cannot Open ODBC
>> Database!
>> 
>> 
>> 
>> 
>> 
>> Here's my config from /conf/autoload_configs/cidlookup.conf.xml:
>> 
>> 
>> 
>> <configuration name="cidlookup.conf" description="cidlookup Configuration">
>>   <settings>
>>     <param name="cache" value="false"/>
>>     <!-- expire is in seconds -->
>>     <param name="cache-expire" value="86400"/>
>>     <param name="odbc-dsn" value="mysqlitedb"/>
>> 
>>     <!-- comment out sql to not setup a database (directory) lookup -->
>>     <param name="sql" value="
>>      SELECT name||' ('||type||')' AS name
>>       FROM phonebook p JOIN numbers n ON p.id = n.phonebook_id
>>       WHERE n.number='${caller_id_number}'
>>       LIMIT 1
>>       "/>
>>   </settings>
>> </configuration>
>> 
>> I someone could point me into the right direction that would be great.
>> 
>> Kind regards,
>> 
>> Erik Dekkers (wvds-nl)
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 christian at yellox.de  Mon Feb 21 03:29:56 2011
From: christian at yellox.de (Christian Hiller)
Date: Mon, 21 Feb 2011 01:29:56 +0100
Subject: [Freeswitch-users] javascript hanguphook
In-Reply-To: <CDF5E00590C9BA4AB96BE7CDD28C86AE07873018D3@FS1.wvds.local>
References: <4D5AA03F.8030307@yellox.de>
	<CDF5E00590C9BA4AB96BE7CDD28C86AE07873018D3@FS1.wvds.local>
Message-ID: <4D61B204.4060102@yellox.de>

hello Erik,

i have set

<action application="set" data="my_var=123"/>
<action application="set" data="session_in_hangup_hook=true"/>
<action application="set" data="api_hangup_hook=jsrun cleanup.js 
${my_var}"/>
<action application="javascript" data="routing.js"/>

The channel_variable ${my_var} gets updated in routing.js but whenever 
cleanup.js is executed, then ${my_var} still has the old value of 123. 
Why that?

Kind regards

Am 20:59, schrieb Erik Dekkers:
> Hi Christian,
>
> The hangup hook api should be set in the dialplan:
> <action application=et" data="api_hangup_hook=jsrun cleanup.js ${uuid}"/>
>
> This will trigger the hangup api once the session is destroyed.
>
> Regards,
>
> Erik
>
> -----Oorspronkelijk bericht-----
> Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Christian Hiller
> Verzonden: dinsdag 15 februari 2011 16:48
> Aan: freeswitch-users at lists.freeswitch.org
> Onderwerp: [Freeswitch-users] javascript hanguphook
>
> Hello,
>
> i have a javscript that is called from xml-dialplan ...
> function on_hangup(hup_session, how)
> {
>       console_log("info","got hungup");
>       exit();
> }
>
> session.execute('bridge','sofia/internal....');
> session.setHangupHook(on_hangup);
> ...
>
> Now i experience, that this function is only called if the session is not answered yet.
> Once its answered and then got hung up, the function on_hangup is not called anymore.
> Any ideas ?
>
> Kind regards
>
> Christian Hiller
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 philippe at ppmt.org  Sun Feb 20 01:54:01 2011
From: philippe at ppmt.org (Philippe Le Toquin)
Date: Sat, 19 Feb 2011 17:54:01 -0500
Subject: [Freeswitch-users] core dump in debian
In-Reply-To: <AANLkTik7J61+xbmn+-ExnDJCA=0MSM=_fQAQR3EbnLcO@mail.gmail.com>
References: <AANLkTi=YopVmLSTi9MwhpXEx-wpJ-nLmiB4jeBSszAz1@mail.gmail.com>	<AANLkTimq=bHNgLK=NROOJz7ce76oA4cS1gM1W2Fk6ehj@mail.gmail.com>
	<AANLkTik7J61+xbmn+-ExnDJCA=0MSM=_fQAQR3EbnLcO@mail.gmail.com>
Message-ID: <4D604A09.4060502@ppmt.org>

Interesting

I have compiled Freeswitch on my Guruplug and it crashes with a core
dump as well

I thought it was me doing something but there might be more to it.

I am about to recompile and will see if it still happens

So another question could be: Has anyone ever managed to get Freeswitch
working on a ARM based plug?

/Philippe

On 11-02-19 05:07 PM, envelopes envelopes wrote:
> I tried two times. one is 1.0.7 release, one is release from Git
> source. the same issue of crash
>
> my FS is running on dockstar.
>
>
> On Sat, Feb 19, 2011 at 10:20 AM, Anthony Minessale
> <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
>
>     Do a fresh checkout and remove previous binaries.
>
>     On Feb 18, 2011 9:57 PM, "envelopes envelopes"
>     <sunwood360 at gmail.com <mailto:sunwood360 at gmail.com>> wrote:
>     > #0 vxprintf (arg=0x40ac9be0, fmt=0x40249412 "q'", ap=<value
>     optimized out>,
>     > useExtended=<value optimized out>, func=<value optimized out>)
>     > at src/switch_mprintf.c:686
>     > #1 0x40130774 in base_vprintf (xRealloc=0x4012f328 <printf_realloc>,
>     > zInitBuf=0xb <Address 0xb out of bounds>, nInitBuf=<value
>     optimized out>,
>     > zFormat=0xa <Address 0xa out of bounds>, ap=..., useInternal=<value
>     > optimized out>) at src/switch_mprintf.c:849
>     > #2 0x40130880 in switch_mprintf (zFormat=0x4027671c "debian") at
>     > src/switch_mprintf.c:892
>     > #3 0x4013ba48 in switch_core_expire_registration (force=0) at
>     > src/switch_core_sqldb.c:1726
>     > #4 0x4013bf3c in switch_core_sql_db_thread (thread=<value
>     optimized out>,
>     > obj=0x0) at src/switch_core_sqldb.c:926
>     > #5 0x401c93dc in dummy_worker (opaque=0x402493d4) at
>     > threadproc/unix/thread.c:138
>     > #6 0x402e48cc in start_thread () from /lib/libpthread.so.0
>     > #7 0x4058bbdc in clone () from /lib/libc.so.6
>     > #8 0x4058bbdc in clone () from /lib/libc.so.6
>     >
>     >
>     > FS crashed right after start!
>
>     _______________________________________________
>     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 validator77 at gmail.com  Sun Feb 20 01:51:52 2011
From: validator77 at gmail.com (Mac)
Date: Sat, 19 Feb 2011 23:51:52 +0100
Subject: [Freeswitch-users] RTP Proxy config and DTMF conversion problem
Message-ID: <AANLkTi=Fz6_1riAh2RrME-k_Dcodk6M6W6Q3zF6c1aJp@mail.gmail.com>

Dear ALL,

Thats my first post here. I am struggling for some time with DTMF issue.
Let me introduce you my configuration.

The main task is to configure RTP Proxy with full topology hiding -
OPERATOR_1 ------- Freeswitch ------- OPERATOR_2
Here is a prt of my config:
- sip profile.
I have one new sip profile with two gateways (OPERATOR_1 and OPERATOR_2)
<include>
  <gateway name="OP1">
    <param name="username" value="X1.Y1.V1.Z1"/>
    <param name="password" value="09321234"/>
    <param name="proxy" value="X1.Y1.V1.Z1"/>
    <param name="from-domain" value="test.com"/>
    <param name="register" value="false"/>
    <param name="caller-id-in-from" value="true"/>
    <param name="extension" value="test"/>
    <param name="extension-in-contact" value="true"/>
    <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
    <param name="disable-transcoding" value="true"/>
    <param name="inbound-late-negotiation" value="true"/>
  </gateway>
</include>

- dialplan
<include>
  <context name="public">
        <extension name="term">
                <condition field="network_addr"
expression="^(X1\.Y1\.V1\.Z1|X2\.Y2\.V1\.Z1)$" break="on-false"/>
                <condition field="destination_number"
expression="^46(\d{9})$">
                <action application="set" data="proxy_media=true"/>
                <action application="set" data="bypass_media=false"/>
                <action application="start_dtmf"/>
                <action application="bridge" data="sofia/gateway/OP1/$0"/>
                </condition>
        </extension>
  </context>
</include>

Everything is fine, but i have problem with DTMF conversion from RFC2833 to
inband.
Refering to http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmfand
http://wiki.freeswitch.org/wiki/Proxy_Media i have done all the necessary
things.

vars.xml
  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,G.729,PCMU,GSM"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA,G.729,PCMU,GSM"/>

Placing  <param name="dtmf-type" value="none"/> does not help anyway.

Could sb point me where the problem is located?

Thanks in advance,
Mac
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From validator77 at gmail.com  Sun Feb 20 20:46:30 2011
From: validator77 at gmail.com (Mac)
Date: Sun, 20 Feb 2011 18:46:30 +0100
Subject: [Freeswitch-users] RTP Proxy config and DTMF conversion problem
Message-ID: <AANLkTimW23Uf61OU2RVJWD6HAM9i_towb2GocwYsi7aC@mail.gmail.com>

Dear ALL,

Thats my first post here. I am struggling for some time with DTMF issue.
Let me introduce you my configuration.

The main task is to configure RTP Proxy with full topology hiding -
OPERATOR_1 ------- Freeswitch ------- OPERATOR_2
Here is a prt of my config:
- sip profile.
I have one new sip profile with two gateways (OPERATOR_1 and OPERATOR_2)
<include>
  <gateway name="OP1">
    <param name="username" value="X1.Y1.V1.Z1"/>
    <param name="password" value="09321234"/>
    <param name="proxy" value="X1.Y1.V1.Z1"/>
    <param name="from-domain" value="test.com"/>
    <param name="register" value="false"/>
    <param name="caller-id-in-from" value="true"/>
    <param name="extension" value="test"/>
    <param name="extension-in-contact" value="true"/>
    <param name="outbound-codec-prefs" value="$${outbound_codec_
prefs}"/>
    <param name="disable-transcoding" value="true"/>
    <param name="inbound-late-negotiation" value="true"/>
  </gateway>
</include>

- dialplan
<include>
  <context name="public">
        <extension name="term">
                <condition field="network_addr"
expression="^(X1\.Y1\.V1\.Z1|X2\.Y2\.V1\.Z1)$" break="on-false"/>
                <condition field="destination_number"
expression="^46(\d{9})$">
                <action application="set" data="proxy_media=true"/>
                <action application="set" data="bypass_media=false"/>
                <action application="start_dtmf"/>
                <action application="bridge" data="sofia/gateway/OP1/$0"/>
                </condition>
        </extension>
  </context>
</include>

Everything is fine, but i have problem with DTMF conversion from RFC2833 to
inband.
Refering to http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmfand
http://wiki.freeswitch.org/wiki/Proxy_Media i have done all the necessary
things.

vars.xml
  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,G.729,PCMU,GSM"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA,G.729,PCMU,GSM"/>

The after-effect is following output:
[ERR] switch_core_io.c:1055 Codec PROXY PASS-THROUGH encoder error!

Placing  <param name="dtmf-type" value="none"/> does not help anyway.

Could sb point me where the problem is located?

Thanks in advance,
Mac
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From sgk at ppona.com  Sun Feb 20 19:32:29 2011
From: sgk at ppona.com (Shigeru Kanemoto)
Date: Mon, 21 Feb 2011 01:32:29 +0900
Subject: [Freeswitch-users] FS gateway which registers into FS
Message-ID: <AANLkTi=fKGKAJJyAVzVaQwKh2y8cMPwrSWhcBu9g8HH=@mail.gmail.com>

Hello,

I am new to FreeSWITCH and having a headache configuring it. My
experiences to Asterisk do not work completely. I appreciate your
suggestion.

I am trying to setup a central SIP server which accepts registers from
subsidiary SIP servers which is behind NAT and has no fixed IP
address. Each subsidiary servers have extension number ranges. I have
to route such extension ranges to each SIP servers.

FS central server
   <--- (NAT) --- Server A (ext. 1000 - 1099)
   <--- (NAT) --- Server B (ext. 1100 - 1199)
   <--- (NAT) --- Phone ext. 1201
   <--- (NAT) --- Phone ext. 1202

According to the wiki page
http://wiki.freeswitch.org/wiki/Clarification:gateways , I created an
XML file under "conf/directory/default" for a gateway setting inside a
<user> block. In my configuration, the gateway setting is parsed in
the profile "internal".

My current problem is the dialplan "bridge" application does not work
to bridge to the destination like "sofia/gateway/server-a/1001". The
call terminates after printing a log message like "[NOTICE]
sofia.c:5082 Hangup sofia/internal/1001 [CS_CONSUME_MEDIA]
[NORMAL_TEMPORARY_FAILURE]".

What is my mistake? Is it wrong to use the "gateway" feature for such purpose?

sgk


From edpimentl at gmail.com  Mon Feb 21 06:05:11 2011
From: edpimentl at gmail.com (EdPimentl)
Date: Sun, 20 Feb 2011 22:05:11 -0500
Subject: [Freeswitch-users] Easy IVR?
Message-ID: <AANLkTin5VqM7Xt7VU3Abs+X0Y3FqGQ9tNS3Y7aHgCOmq@mail.gmail.com>

Is this not a good example of how easy IVR design/creation should be?
http://asymmetrical-view.com/2011/02/20/twilio-in-ten-minutes.html
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From hareem.haque at gmail.com  Mon Feb 21 06:40:28 2011
From: hareem.haque at gmail.com (Hareem Haque)
Date: Sun, 20 Feb 2011 22:40:28 -0500
Subject: [Freeswitch-users] How to limit a certain gateway/extension
Message-ID: <AANLkTimE+TqWkX4tEiiZHqi9rKO=q1+s21U0KZumnn4O@mail.gmail.com>

Hello Everyone

Many thanks for helping me out. I got most of the switch setup. The mod_cdr
works well now. I got some ideas of the routing patterns. I have1 question
and would greatly appreciate your help in resolving this issue.


 How can i limit a certain extension/gateway to only able to initiate a
predefined number of calls. For example extension 1000 connects to the
switch. And now that extension can only make 24 concurrent calls. How can i
set this up.




Best Regards
Hareem. Haque
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From hareem.haque at gmail.com  Mon Feb 21 06:55:09 2011
From: hareem.haque at gmail.com (Hareem Haque)
Date: Sun, 20 Feb 2011 22:55:09 -0500
Subject: [Freeswitch-users] Question regardin freeswitch startup info
Message-ID: <AANLkTikDDVXFjdeeAg6++jisFUkB+aaFCfSY3cPX7Pb4@mail.gmail.com>

 i start freeswitch it says max sessions 1000
and session rate 30.. what exact do these numbers mean and how can i
increase them
Your help is greatly appreciated.

Best Regards
Hareem. Haque



On Sun, Feb 20, 2011 at 12:11 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: Codec negotiation, again (Steven Ayre)
>   2. Re: Any experience with DTMF from FreeSwitch to Sonus with
>      Vega ATAs? (Yehavi Bourvine)
>   3. Need help with CDR setup and call routing patterns (Hareem Haque)
>   4. Re: Need help with CDR setup and call routing     patterns
>      (Steven Ayre)
>   5. Re: Need help with CDR setup and call routing     patterns
>      (Steven Ayre)
>
>
> ---------- Forwarded message ----------
> From: Steven Ayre <steveayre at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Sun, 20 Feb 2011 12:36:45 +0000
> Subject: Re: [Freeswitch-users] Codec negotiation, again
> In default behaviour, FS will negotiate the codec for the aleg, then do the
> dialplan, then do the bleg. As a result it can have already picked the 729
> codec by the time it hits the dialplan. With transcoding that's fine but
> with 729 not.
>
> The workaround is the late-negotiation profile parameter. It delays picking
> the aleg codec until the bleg picks one, then uses that if possible for the
> aleg.
>
> Steve on iPhone
>
>
> On 20 Feb 2011, at 01:33, Serge Yuriev <me at nevian.org> wrote:
>
> > Hello,
> >
> > A offers g729, g711a, g711u
> > FS allows all of them
> > B offers g711a, g711u
> >
> > Call fails with INCOMPATIBLE DESTINATION.
> >
> > Cant understand why FS offers only ONE codec (g729) to B - wiki says it
> should offer ordered LIST of..
> > GIT few days old, default config
> >
> > --
> > 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
>
>
>
>
> ---------- Forwarded message ----------
> From: Yehavi Bourvine <yehavi.bourvine at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Sun, 20 Feb 2011 14:39:18 +0200
> Subject: Re: [Freeswitch-users] Any experience with DTMF from FreeSwitch to
> Sonus with Vega ATAs?
> Hello Steve,
>
>   Thanks for clarifying it.
>
>                      __Yehavi:
>
> 2011/2/20 Steve Underwood <steveu at coppice.org>
>
>> Hi Yehavi,
>>
>> Section 3.2 of RFC2833 is poorly worded, and causes a lot of issues in
>> the real world. The first paragraph says you can choose to send both
>> audio and events, but doesn't clarify how. Presumably RFC2198 should be
>> used. Is your system using RFC2198 encoding? I doubt it. The second
>> paragraph talks about tone onset getting through as audio. Valid DTMF
>> can be as short as 45ms, so 60ms is far more than anything which might
>> be termed onset. Because they didn't specify anything about what is
>> tolerable as tone onset passing through the channel as audio we have
>> chaos today.
>>
>> RFC2833 is an obsolete spec, and we should be talking about RFC4733
>> today. Section 2.5.1.3.1 of RFC4733 does clearly call for combined
>> payloads to be sent as RFC2198 packets. RFC4733 is even vaguer about the
>> onset issue, though.
>>
>> Steve
>>
>>
>> On 02/20/2011 06:33 PM, Yehavi Bourvine wrote:
>> > Hello Steve,
>> >  I uderstand from paragraph 3.2 in RFC-2833 that the sender may send
>> > some audio of the DTMF while sending the events, until it revognises
>> > that this is a DTMF tone. The 60mSec is what I've been told by Vega
>> > engineers.
>> >                                Thanks! __Yehavi:
>> >
>> > 2011/2/20 Steve Underwood <steveu at coppice.org <mailto:
>> steveu at coppice.org>>
>>  >
>> >     On 02/20/2011 03:31 PM, Yehavi Bourvine wrote:
>> >     > Hello,
>> >     >   Anyone has an experience with the above configuration? All
>> >     sides are
>> >     > marked as RFC-2833 and the Sonus recognises most of the DTMF's
>> >     twice.
>> >     > The Vega sends both RFC-2833 events and about 60msec of the DTMF
>> >     tone
>> >     > (according to Vega's engineers it takes it about 60 msec to
>> >     detect the
>> >     > DTMF tones).
>> >     > According to the RFC this is acceptable. The service provider I am
>> >     > working with says that it shouldn't be like that. Before I waste
>> >     time
>> >     > in wars, has anybody had this issue and knows whether it is a
>> matter
>> >     > of configuration at the Sonus side?
>> >     >                                    Thanks! __Yehavi:
>> >     Are you saying the Vega sends 60ms of DTMF as audio, then sends
>> >     RFC2833
>> >     DTMF packets and mutes the audio? In what reading of the RFC is that
>> >     considered appropriate action? In addition, a DTMF detector should
>> >     detect is more like 40ms. 60ms is rather long.
>> >
>> >     Steve
>> >
>> >
>> >     _______________________________________________
>> >     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
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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: Hareem Haque <hareem.haque at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Sun, 20 Feb 2011 11:08:08 -0500
> Subject: [Freeswitch-users] Need help with CDR setup and call routing
> patterns
> Thank you very much for helping me out with the initial switch setup. I
> really appreciate it. However, I am stuck with the cdr and call routing
> patterns setup.
>
>
> A. I have attached my cdr_csv file and its cdr log file. As you may see in
> the conf xml that i have 22 items that i need posted onto my cdr logs.
> However, the csv file only shows 15 items. How can i make this work
>
> B. I have setup a few sip trunks. Now how do i set them up so that if i
> dial 01192 that call goes to gateway A and if i dial 1XXX then that call
> goes to gateway B
>
> Many thanks for all the help. I really appreciate it
>
> Best Regards
> Hareem. Haque
>
>
>
>
> ---------- Forwarded message ----------
> From: Steven Ayre <steveayre at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Sun, 20 Feb 2011 17:01:47 +0000
> Subject: Re: [Freeswitch-users] Need help with CDR setup and call routing
> patterns
> After changing the config, did you reload the config?
>
> reloadxml
> reload mod_cdr_csv
>
> Steve on iPhone
>
>
>
> On 20 Feb 2011, at 16:08, Hareem Haque <hareem.haque at gmail.com> wrote:
>
> > Thank you very much for helping me out with the initial switch setup. I
> really appreciate it. However, I am stuck with the cdr and call routing
> patterns setup.
> >
> >
> > A. I have attached my cdr_csv file and its cdr log file. As you may see
> in the conf xml that i have 22 items that i need posted onto my cdr logs.
> However, the csv file only shows 15 items. How can i make this work
> >
> > B. I have setup a few sip trunks. Now how do i set them up so that if i
> dial 01192 that call goes to gateway A and if i dial 1XXX then that call
> goes to gateway B
> >
> > Many thanks for all the help. I really appreciate it
> >
> > Best Regards
> > Hareem. Haque
> >
> >
> > <cdr_csv.conf.xml>
> > <hareem.csv>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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: Steven Ayre <steveayre at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Sun, 20 Feb 2011 17:10:34 +0000
> Subject: Re: [Freeswitch-users] Need help with CDR setup and call routing
> patterns
>
>> B. I have setup a few sip trunks. Now how do i set them up so that if i
>> dial 01192 that call goes to gateway A and if i dial 1XXX then that call
>> goes to gateway B
>>
>
> Handle each in a different extension in the dialplan:
>
> <extension name="route1">
>    <condition field="destination_number" expression="^(01192)$">
>       <action application="bridge" data="sofia/gateway/gatewaya/$1"/>
>    </condition>
> </extension>
> <extension name="route2">
>    <condition field="destination_number" expression="^(1\d\d\d)$">
>       <action application="bridge" data="sofia/gateway/gatewayb/$1"/>
>    </condition>
> </extension>
>
> Adjust the expressions to suit the numbers you want to handle better. The
> contents of the () brackets are placed in variable $1, which is being used
> in the bridge to dial that number via the gateway. You can add prefixes etc
> there if you need them. ^ and $ match the start and end of the string, not
> actual characters. They're called regular expressions if you want to read up
> on them, and are more flexible than just matching absolute strings because
> you match patterns instead.
>
> An example of changing a prefix, should your gateway need that (for example
> converting a local number to an international prefix):
>    <condition field="destination_number" expression="^0(1192)$">
>       <action application="bridge" data="sofia/gateway/gatewaya/44$1"/>
>    </condition>
> 01192 would be dialed as 441192 through gateway a.
>
> -Steve
>
>
>
>
> On 20 February 2011 16:08, Hareem Haque <hareem.haque at gmail.com> wrote:
>
>> Thank you very much for helping me out with the initial switch setup. I
>> really appreciate it. However, I am stuck with the cdr and call routing
>> patterns setup.
>>
>>
>> A. I have attached my cdr_csv file and its cdr log file. As you may see in
>> the conf xml that i have 22 items that i need posted onto my cdr logs.
>> However, the csv file only shows 15 items. How can i make this work
>>
>> B. I have setup a few sip trunks. Now how do i set them up so that if i
>> dial 01192 that call goes to gateway A and if i dial 1XXX then that call
>> goes to gateway B
>>
>> Many thanks for all the help. I really appreciate it
>>
>> Best Regards
>> Hareem. Haque
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Mon Feb 21 07:30:26 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 21 Feb 2011 10:00:26 +0530
Subject: [Freeswitch-users] mod_odbc_query
Message-ID: <AANLkTimZbs8S5sgkS7z=6+BDjW3f1U=LmO03203WZYUC@mail.gmail.com>

Hello,

How to download mod_odbc_query source, there is not such source in 1.0.6 or
1.0.7 .

Regds
Sam
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From edward.dejong at voicecarrier.com  Mon Feb 21 07:33:10 2011
From: edward.dejong at voicecarrier.com (Edward de Jong)
Date: Sun, 20 Feb 2011 20:33:10 -0800
Subject: [Freeswitch-users] Newbie question / what is the best text to
	speech module out there?
Message-ID: <2D36A10A-9E60-49B5-BFFE-20320448F329@voicecarrier.com>

I see Cepstra and some other text to speech modules. Which is the best of the free ones? for my experiments I can't afford a fancy one like Cepstra.

edj



From spencer at 5ninesolutions.com  Mon Feb 21 08:07:23 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 20 Feb 2011 21:07:23 -0800
Subject: [Freeswitch-users] INVITE DoS Prevention
Message-ID: <6461E11A-84B5-40AF-96BE-C68F3DF91D8E@5ninesolutions.com>

Hi,
We run hosted Freeswitch instances in VMs with the internal profile on  
port 5060 connecting to clients mostly behind NAT and then the  
external profile connecting to our proxies only.  Protecting the  
external profile its straightforward.. we only allow traffic to/from  
our proxies at the firewall level.  But protecting the internal  
profile seems to be a bit more difficult because the UACs could be  
theoretically anywhere on the network.

I'm currently using Fail2Ban to prevent brute force registration and  
INVITEs on auth failures, e.g.:
failregex = \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(REGISTER\)  
on sofia profile \'\w+\' for \[.*\] from ip <HOST>
             \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(INVITE\)  
on sofia profile \'\w+\' for \[.*\] from ip <HOST>

My question is, since its part of a normal SIP dialog to challenge the  
INVITE, is there any way to prevent a possible DoS from just sheer  
volume of incoming INVITEs on an Internet facing server  
automatically.  I.e., If you block the logged challenge, you'd block  
all legitimate INVITEs and registrations.  Since its UDP traffic I  
couldn't come up with a way to do it automatically at the iptables  
level. i.e. number of concurrent connections.  Is there some option to  
just not respond if a client is sending a number of requests over a  
certain threshold?  It might not stop them from sending the traffic  
but pretty soon they'd get the idea that it wasn't going to go  
anywhere.  My concern is say there are 50 Freeswitch instances on a  
box (albeit 8 core, 32GB ram, 8 15K raid 10 storage) and someone  
starts sending thousands of rouge INVITEs to every VM on a physical  
box that the CPU load from just challenging the incoming INVITEs would  
create a DoS.  We the logs regularly to try to catch people doing this  
sort of thing and drop them at a router upstream of the core network,  
but I'd like to have it happen without human intervention.  Have I  
completely over thought this and am missing something obvious?

Thanks,
Spencer



From edward.dejong at voicecarrier.com  Mon Feb 21 08:08:10 2011
From: edward.dejong at voicecarrier.com (Edward de Jong)
Date: Sun, 20 Feb 2011 21:08:10 -0800
Subject: [Freeswitch-users] newbie question: how do you set the folder that
	freeswitch uses for sounds?
Message-ID: <691770B6-DEC9-4F09-BE93-6B3DC6F9D6D3@voicecarrier.com>

I see that in the default config, freeswitch uses sounds from the en/us/callie folder.
so when you ask for playback of hello.wav, it goes to that folder by default.
how does one change the folder used by freeswitch?
i want to let the user pick between different voices, which of the many config files sets up which voice to use?

and what if I want to have better voice on my user's phone? can I go to 16000 sample rate? if the user has HD capability on their phone, shouldn't I be using better voice samples? and how can I choose the voice folder depending on the phone connected to the PBX at that moment? In other words, if the user has an HD phone can I connect with them at the better codec? obviously if we connect to some other lesser device the dreaded transcoding will occur, but i would think for internal use, for example when the user is navigating menus, they would appreciate a better voice. Or should i just use all waveforms at 16000 and let it drop them down...

thanks in advance.




From chat2jesse at gmail.com  Mon Feb 21 08:42:01 2011
From: chat2jesse at gmail.com (jesse)
Date: Sun, 20 Feb 2011 21:42:01 -0800
Subject: [Freeswitch-users]
	=?iso-8859-1?q?Newbie=B4s_question_about_FreeS?=
	=?iso-8859-1?q?witch=2E=2E=2E?=
In-Reply-To: <1298169268414-6044831.post@n2.nabble.com>
References: <AANLkTinf=z4JEp4G2xK_0n+u8v0OhR++GBvPzzbT1Y_-@mail.gmail.com>
	<1298129542465-6043436.post@n2.nabble.com>
	<AANLkTinD08hCZ74SncUbR318agQnO1=JB_Oebc17EPn6@mail.gmail.com>
	<1298169268414-6044831.post@n2.nabble.com>
Message-ID: <AANLkTikE=yjS=ku914rpO5+MoJzWikbw0md+dTMDiM07@mail.gmail.com>

no. the trick didn't work. it crashed right after FS was restarted on
Debian.

btw, the current openwrt trunk has FS 1.0.6,  I downloaded 1.0.78 source
code and made symbol link to the new src from feeds/packages.  but openwrt
still built old 1.0.6. is there a way to enforce openwrt build system pick
up the new code base?


thanks!

On Sat, Feb 19, 2011 at 6:34 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
>
> jesse zhao wrote:
> >
> > mazilo:
> >
> >  have you tried to install FS on Debian of your DockStar? my FS (1.0.7 or
> > latest source) always crashed right after start.
> Please read this  http://www.dslreports.com/forum/r25389970- post  and
> ignore my posts after that one. It has something to do with the GCC word
> alignment for ARM platform. I hope this will fix the problem you are
> experiencing.
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Newbie-s-question-about-FreeSwitch-tp6038911p6044831.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 sunwood360 at gmail.com  Mon Feb 21 09:26:07 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Sun, 20 Feb 2011 22:26:07 -0800
Subject: [Freeswitch-users] core dump in debian
In-Reply-To: <4D604A09.4060502@ppmt.org>
References: <AANLkTi=YopVmLSTi9MwhpXEx-wpJ-nLmiB4jeBSszAz1@mail.gmail.com>
	<AANLkTimq=bHNgLK=NROOJz7ce76oA4cS1gM1W2Fk6ehj@mail.gmail.com>
	<AANLkTik7J61+xbmn+-ExnDJCA=0MSM=_fQAQR3EbnLcO@mail.gmail.com>
	<4D604A09.4060502@ppmt.org>
Message-ID: <AANLkTin6i-GWLzQV+GB8gtB=f3XbGTuSNSoLg4o_YLQt@mail.gmail.com>

Openwrt based fs 1.0.6 definitely works on arm.

But the crash of fs on arm debian made me crazy. CPU alignment seems not the
cause.  Maybe you can try 1.0.6
On Feb 20, 2011 6:05 PM, "Philippe Le Toquin" <philippe at ppmt.org> wrote:
> Interesting
>
> I have compiled Freeswitch on my Guruplug and it crashes with a core
> dump as well
>
> I thought it was me doing something but there might be more to it.
>
> I am about to recompile and will see if it still happens
>
> So another question could be: Has anyone ever managed to get Freeswitch
> working on a ARM based plug?
>
> /Philippe
>
> On 11-02-19 05:07 PM, envelopes envelopes wrote:
>> I tried two times. one is 1.0.7 release, one is release from Git
>> source. the same issue of crash
>>
>> my FS is running on dockstar.
>>
>>
>> On Sat, Feb 19, 2011 at 10:20 AM, Anthony Minessale
>> <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
>>
>> Do a fresh checkout and remove previous binaries.
>>
>> On Feb 18, 2011 9:57 PM, "envelopes envelopes"
>> <sunwood360 at gmail.com <mailto:sunwood360 at gmail.com>> wrote:
>> > #0 vxprintf (arg=0x40ac9be0, fmt=0x40249412 "q'", ap=<value
>> optimized out>,
>> > useExtended=<value optimized out>, func=<value optimized out>)
>> > at src/switch_mprintf.c:686
>> > #1 0x40130774 in base_vprintf (xRealloc=0x4012f328 <printf_realloc>,
>> > zInitBuf=0xb <Address 0xb out of bounds>, nInitBuf=<value
>> optimized out>,
>> > zFormat=0xa <Address 0xa out of bounds>, ap=..., useInternal=<value
>> > optimized out>) at src/switch_mprintf.c:849
>> > #2 0x40130880 in switch_mprintf (zFormat=0x4027671c "debian") at
>> > src/switch_mprintf.c:892
>> > #3 0x4013ba48 in switch_core_expire_registration (force=0) at
>> > src/switch_core_sqldb.c:1726
>> > #4 0x4013bf3c in switch_core_sql_db_thread (thread=<value
>> optimized out>,
>> > obj=0x0) at src/switch_core_sqldb.c:926
>> > #5 0x401c93dc in dummy_worker (opaque=0x402493d4) at
>> > threadproc/unix/thread.c:138
>> > #6 0x402e48cc in start_thread () from /lib/libpthread.so.0
>> > #7 0x4058bbdc in clone () from /lib/libc.so.6
>> > #8 0x4058bbdc in clone () from /lib/libc.so.6
>> >
>> >
>> > FS crashed right after start!
>>
>> _______________________________________________
>> 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 krice at freeswitch.org  Mon Feb 21 09:39:12 2011
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 21 Feb 2011 00:39:12 -0600
Subject: [Freeswitch-users] INVITE DoS Prevention
In-Reply-To: <6461E11A-84B5-40AF-96BE-C68F3DF91D8E@5ninesolutions.com>
Message-ID: <C98764B0.11627%krice@freeswitch.org>

Fail2Ban ... This is block an IP with too many failed attempts from
something like SipVicious pretty quickly


On 2/20/11 11:07 PM, "Spencer Thomason" <spencer at 5ninesolutions.com> wrote:

> Hi,
> We run hosted Freeswitch instances in VMs with the internal profile on
> port 5060 connecting to clients mostly behind NAT and then the
> external profile connecting to our proxies only.  Protecting the
> external profile its straightforward.. we only allow traffic to/from
> our proxies at the firewall level.  But protecting the internal
> profile seems to be a bit more difficult because the UACs could be
> theoretically anywhere on the network.
> 
> I'm currently using Fail2Ban to prevent brute force registration and
> INVITEs on auth failures, e.g.:
> failregex = \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(REGISTER\)
> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
>              \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(INVITE\)
> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
> 
> My question is, since its part of a normal SIP dialog to challenge the
> INVITE, is there any way to prevent a possible DoS from just sheer
> volume of incoming INVITEs on an Internet facing server
> automatically.  I.e., If you block the logged challenge, you'd block
> all legitimate INVITEs and registrations.  Since its UDP traffic I
> couldn't come up with a way to do it automatically at the iptables
> level. i.e. number of concurrent connections.  Is there some option to
> just not respond if a client is sending a number of requests over a
> certain threshold?  It might not stop them from sending the traffic
> but pretty soon they'd get the idea that it wasn't going to go
> anywhere.  My concern is say there are 50 Freeswitch instances on a
> box (albeit 8 core, 32GB ram, 8 15K raid 10 storage) and someone
> starts sending thousands of rouge INVITEs to every VM on a physical
> box that the CPU load from just challenging the incoming INVITEs would
> create a DoS.  We the logs regularly to try to catch people doing this
> sort of thing and drop them at a router upstream of the core network,
> but I'd like to have it happen without human intervention.  Have I
> completely over thought this and am missing something obvious?
> 
> Thanks,
> 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 brent at overthewire.com.au  Mon Feb 21 09:43:36 2011
From: brent at overthewire.com.au (Brent Paddon)
Date: Mon, 21 Feb 2011 16:43:36 +1000
Subject: [Freeswitch-users] INVITE DoS Prevention
In-Reply-To: <6461E11A-84B5-40AF-96BE-C68F3DF91D8E@5ninesolutions.com>
References: <6461E11A-84B5-40AF-96BE-C68F3DF91D8E@5ninesolutions.com>
Message-ID: <AANLkTimhDcBoyEtVzJ4sVX-opQBe4SaYL8_gjtUUvPSY@mail.gmail.com>

We run a somewhat similar sounding setup.  We wrote some code that grabs the
/32 from the fail2ban instance running on each VM, and automatically puts
that /32 into a BGP blackhole sink which stops traffic to the entire network
for that /32.

You could look to do something similar, either BGP blackholing it or if you
have a single upstream, use something like expect to insert a firewall rule
??

With that said, it could be nice for some of this to exist in FS itself (as
you say, slowing down responses over certain thresholds).

Brent

On Mon, Feb 21, 2011 at 3:07 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> Hi,
> We run hosted Freeswitch instances in VMs with the internal profile on
> port 5060 connecting to clients mostly behind NAT and then the
> external profile connecting to our proxies only.  Protecting the
> external profile its straightforward.. we only allow traffic to/from
> our proxies at the firewall level.  But protecting the internal
> profile seems to be a bit more difficult because the UACs could be
> theoretically anywhere on the network.
>
> I'm currently using Fail2Ban to prevent brute force registration and
> INVITEs on auth failures, e.g.:
> failregex = \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(REGISTER\)
> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
>             \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(INVITE\)
> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
>
> My question is, since its part of a normal SIP dialog to challenge the
> INVITE, is there any way to prevent a possible DoS from just sheer
> volume of incoming INVITEs on an Internet facing server
> automatically.  I.e., If you block the logged challenge, you'd block
> all legitimate INVITEs and registrations.  Since its UDP traffic I
> couldn't come up with a way to do it automatically at the iptables
> level. i.e. number of concurrent connections.  Is there some option to
> just not respond if a client is sending a number of requests over a
> certain threshold?  It might not stop them from sending the traffic
> but pretty soon they'd get the idea that it wasn't going to go
> anywhere.  My concern is say there are 50 Freeswitch instances on a
> box (albeit 8 core, 32GB ram, 8 15K raid 10 storage) and someone
> starts sending thousands of rouge INVITEs to every VM on a physical
> box that the CPU load from just challenging the incoming INVITEs would
> create a DoS.  We the logs regularly to try to catch people doing this
> sort of thing and drop them at a router upstream of the core network,
> but I'd like to have it happen without human intervention.  Have I
> completely over thought this and am missing something obvious?
>
> Thanks,
> 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
>



-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From spencer at 5ninesolutions.com  Mon Feb 21 10:10:04 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 20 Feb 2011 23:10:04 -0800
Subject: [Freeswitch-users] INVITE DoS Prevention
In-Reply-To: <C98764B0.11627%krice@freeswitch.org>
References: <C98764B0.11627%krice@freeswitch.org>
Message-ID: <A624F826-AF73-40BD-8DCD-10F39551D924@5ninesolutions.com>

Yes, that works great if they respond to the challenge with a failed  
auth. But the scenario I'm trying to prevent is if they just send the  
INVITE and never respond to the challenge.  Fail2Ban will not work as  
every endpoint will initially send an INVITE and receive a challenge.   
Legit calls will then respond correctly and not be logged as a SIP  
auth failure but every call that is challenged will show up as SIP  
auth challenge in the logs so there is no regex to differentiate  
between legit an non legit traffic.

Spencer

On Feb 20, 2011, at 10:39 PM, Ken Rice wrote:

> Fail2Ban ... This is block an IP with too many failed attempts from
> something like SipVicious pretty quickly
>
>
> On 2/20/11 11:07 PM, "Spencer Thomason" <spencer at 5ninesolutions.com>  
> wrote:
>
>> Hi,
>> We run hosted Freeswitch instances in VMs with the internal profile  
>> on
>> port 5060 connecting to clients mostly behind NAT and then the
>> external profile connecting to our proxies only.  Protecting the
>> external profile its straightforward.. we only allow traffic to/from
>> our proxies at the firewall level.  But protecting the internal
>> profile seems to be a bit more difficult because the UACs could be
>> theoretically anywhere on the network.
>>
>> I'm currently using Fail2Ban to prevent brute force registration and
>> INVITEs on auth failures, e.g.:
>> failregex = \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(REGISTER\)
>> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
>>             \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(INVITE\)
>> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
>>
>> My question is, since its part of a normal SIP dialog to challenge  
>> the
>> INVITE, is there any way to prevent a possible DoS from just sheer
>> volume of incoming INVITEs on an Internet facing server
>> automatically.  I.e., If you block the logged challenge, you'd block
>> all legitimate INVITEs and registrations.  Since its UDP traffic I
>> couldn't come up with a way to do it automatically at the iptables
>> level. i.e. number of concurrent connections.  Is there some option  
>> to
>> just not respond if a client is sending a number of requests over a
>> certain threshold?  It might not stop them from sending the traffic
>> but pretty soon they'd get the idea that it wasn't going to go
>> anywhere.  My concern is say there are 50 Freeswitch instances on a
>> box (albeit 8 core, 32GB ram, 8 15K raid 10 storage) and someone
>> starts sending thousands of rouge INVITEs to every VM on a physical
>> box that the CPU load from just challenging the incoming INVITEs  
>> would
>> create a DoS.  We the logs regularly to try to catch people doing  
>> this
>> sort of thing and drop them at a router upstream of the core network,
>> but I'd like to have it happen without human intervention.  Have I
>> completely over thought this and am missing something obvious?
>>
>> Thanks,
>> 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
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 jaybinks at gmail.com  Mon Feb 21 10:22:16 2011
From: jaybinks at gmail.com (jay binks)
Date: Mon, 21 Feb 2011 17:22:16 +1000
Subject: [Freeswitch-users] INVITE DoS Prevention
In-Reply-To: <A624F826-AF73-40BD-8DCD-10F39551D924@5ninesolutions.com>
References: <C98764B0.11627%krice@freeswitch.org>
	<A624F826-AF73-40BD-8DCD-10F39551D924@5ninesolutions.com>
Message-ID: <AANLkTi=+f0GqChj4PeW=XFjQW_mYTtJYPuWEj47gw8r+@mail.gmail.com>

Howdy

   mmm I just had a quick look at mod_sofia , and unfortunatly it looks like
the channel is hungup after the challenge is sent.   But my guess is that
there HAS To be a list or similar somewhere that tracks endpoints waiting
for challenge responses.

Ill see if can track it down and submit a patch, because I too would like to
see this at least logged.

as for rate-limiting responses you can have iptables drop packets over X
number of invites per sec ...
( But they are dropped silently at the Firewall - Kristians SIP Ratelimiter
has this in it )

or you can use mod_limit to allow X number of invites per sec also,  you
would want to tell FS to hit the dialplan before codec negotiation though to
do that nice and early in the piece.

Jay

On Mon, Feb 21, 2011 at 5:10 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> Yes, that works great if they respond to the challenge with a failed
> auth. But the scenario I'm trying to prevent is if they just send the
> INVITE and never respond to the challenge.  Fail2Ban will not work as
> every endpoint will initially send an INVITE and receive a challenge.
> Legit calls will then respond correctly and not be logged as a SIP
> auth failure but every call that is challenged will show up as SIP
> auth challenge in the logs so there is no regex to differentiate
> between legit an non legit traffic.
>
> Spencer
>
> On Feb 20, 2011, at 10:39 PM, Ken Rice wrote:
>
> > Fail2Ban ... This is block an IP with too many failed attempts from
> > something like SipVicious pretty quickly
> >
> >
> > On 2/20/11 11:07 PM, "Spencer Thomason" <spencer at 5ninesolutions.com>
> > wrote:
> >
> >> Hi,
> >> We run hosted Freeswitch instances in VMs with the internal profile
> >> on
> >> port 5060 connecting to clients mostly behind NAT and then the
> >> external profile connecting to our proxies only.  Protecting the
> >> external profile its straightforward.. we only allow traffic to/from
> >> our proxies at the firewall level.  But protecting the internal
> >> profile seems to be a bit more difficult because the UACs could be
> >> theoretically anywhere on the network.
> >>
> >> I'm currently using Fail2Ban to prevent brute force registration and
> >> INVITEs on auth failures, e.g.:
> >> failregex = \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(REGISTER\)
> >> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
> >>             \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(INVITE\)
> >> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
> >>
> >> My question is, since its part of a normal SIP dialog to challenge
> >> the
> >> INVITE, is there any way to prevent a possible DoS from just sheer
> >> volume of incoming INVITEs on an Internet facing server
> >> automatically.  I.e., If you block the logged challenge, you'd block
> >> all legitimate INVITEs and registrations.  Since its UDP traffic I
> >> couldn't come up with a way to do it automatically at the iptables
> >> level. i.e. number of concurrent connections.  Is there some option
> >> to
> >> just not respond if a client is sending a number of requests over a
> >> certain threshold?  It might not stop them from sending the traffic
> >> but pretty soon they'd get the idea that it wasn't going to go
> >> anywhere.  My concern is say there are 50 Freeswitch instances on a
> >> box (albeit 8 core, 32GB ram, 8 15K raid 10 storage) and someone
> >> starts sending thousands of rouge INVITEs to every VM on a physical
> >> box that the CPU load from just challenging the incoming INVITEs
> >> would
> >> create a DoS.  We the logs regularly to try to catch people doing
> >> this
> >> sort of thing and drop them at a router upstream of the core network,
> >> but I'd like to have it happen without human intervention.  Have I
> >> completely over thought this and am missing something obvious?
> >>
> >> Thanks,
> >> 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
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 spencer at 5ninesolutions.com  Mon Feb 21 10:58:45 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 20 Feb 2011 23:58:45 -0800
Subject: [Freeswitch-users] INVITE DoS Prevention
In-Reply-To: <AANLkTi=iUr64By-HgxDt9SnZzLY3FNqvEaOz5zSyocb8@mail.gmail.com>
References: <6461E11A-84B5-40AF-96BE-C68F3DF91D8E@5ninesolutions.com>
	<AANLkTi=iUr64By-HgxDt9SnZzLY3FNqvEaOz5zSyocb8@mail.gmail.com>
Message-ID: <C67ED94E-D615-471A-BFA0-DDCE16F7640A@5ninesolutions.com>

Hi R?gis,
We currently use openvz.  Previously we used Xen and then BSD jails.   
Every one has their ups and downs.  As many people will tell you,  
timing is critical in VoIP applications and by using OpenVZ there is  
only one kernel loaded and the VMs all run bare metal.  This is a much  
more efficient way of running several VMs if they are all of a similar  
OS because you don't need to waste a bunch of RAM and CPU cycles just  
loading 30 copied of the kernel into memory and dealing with paging  
etc.. Also this makes administration much easier because of separate  
filesystems for each VM, the root of each VM is located somewhere in  
the host's filesystem.  This is a big plus when you go to back up your  
hardware node.  What we do is put all the VMs on an LVM partition,  
make a snapshot and use rsync to back up just the changes.  It was a  
major pain with several independent LVM partitions or "virtual" file  
based partitions in Xen. KVM or any true virt platform would require  
the same, a dedicated partition for each VM.  And CPU scaling works  
well with openvz allowing us to save on the electric bill and cooling  
costs.  The downside is if there is a kernel panic, its one kernel and  
the whole world goes with it.  BSD Jails was cool but didn't allow the  
per client resource control we were looking for.  Our experience with  
openvz has been solid with a couple of machines up well over a year.   
We currently use Centos but we've modified it to the point where its  
basically our own in house distro with updates to some of the core  
packages since RH stays so far behind. (for good measure :-))  This  
allows us to pick and choose which packages we'd like to update and we  
make available our own repo which a simple yum update pulls down  
everything from a local mirror.  If you are interested: http://repo.5ninesolutions.com 
.  Having said that, there is a lot of time that goes into maintaining  
some of the out of distro stuff so we are currently looking at  
something that stays a little more current.. I.e. RH 6 has been out  
since Nov and I haven't seen a beta for Centos 6.. but thats a whole  
can of worms..

Spencer


On Feb 20, 2011, at 11:35 PM, R?gis MARTIN wrote:

> Hi Spencer,
>
> >We run hosted Freeswitch instances in VMs
> Nothing related with your question, but can you say me what  
> hypervisor do you use to run freeswitch virtualized (VMware ? KVM ?  
> Xen? OpenVZ ? other .. ?). ?
> What's is your linux distrib under too ?
>
> I'm looking to virtualize some of mine under KVM and I wonder if it  
> works like standard host and looking to similar experience.
>
> Thanks in advance if you could confirm that to me.
>
> Regards,
>
>
> 2011/2/21 Spencer Thomason <spencer at 5ninesolutions.com>
> Hi,
> We run hosted Freeswitch instances in VMs with the internal profile on
> port 5060 connecting to clients mostly behind NAT and then the
> external profile connecting to our proxies only.  Protecting the
> external profile its straightforward.. we only allow traffic to/from
> our proxies at the firewall level.  But protecting the internal
> profile seems to be a bit more difficult because the UACs could be
> theoretically anywhere on the network.
>
> I'm currently using Fail2Ban to prevent brute force registration and
> INVITEs on auth failures, e.g.:
> failregex = \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(REGISTER\)
> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
>             \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(INVITE\)
> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
>
> My question is, since its part of a normal SIP dialog to challenge the
> INVITE, is there any way to prevent a possible DoS from just sheer
> volume of incoming INVITEs on an Internet facing server
> automatically.  I.e., If you block the logged challenge, you'd block
> all legitimate INVITEs and registrations.  Since its UDP traffic I
> couldn't come up with a way to do it automatically at the iptables
> level. i.e. number of concurrent connections.  Is there some option to
> just not respond if a client is sending a number of requests over a
> certain threshold?  It might not stop them from sending the traffic
> but pretty soon they'd get the idea that it wasn't going to go
> anywhere.  My concern is say there are 50 Freeswitch instances on a
> box (albeit 8 core, 32GB ram, 8 15K raid 10 storage) and someone
> starts sending thousands of rouge INVITEs to every VM on a physical
> box that the CPU load from just challenging the incoming INVITEs would
> create a DoS.  We the logs regularly to try to catch people doing this
> sort of thing and drop them at a router upstream of the core network,
> but I'd like to have it happen without human intervention.  Have I
> completely over thought this and am missing something obvious?
>
> Thanks,
> 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
>

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From spencer at 5ninesolutions.com  Mon Feb 21 11:04:22 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Mon, 21 Feb 2011 00:04:22 -0800
Subject: [Freeswitch-users] INVITE DoS Prevention
In-Reply-To: <AANLkTi=+f0GqChj4PeW=XFjQW_mYTtJYPuWEj47gw8r+@mail.gmail.com>
References: <C98764B0.11627%krice@freeswitch.org>
	<A624F826-AF73-40BD-8DCD-10F39551D924@5ninesolutions.com>
	<AANLkTi=+f0GqChj4PeW=XFjQW_mYTtJYPuWEj47gw8r+@mail.gmail.com>
Message-ID: <ECB69E6D-F45D-4992-90B3-58911978BC91@5ninesolutions.com>

Thanks for the info!  Yes it would be great if Freeswitch logged this  
info somewhere but this firewall script was was I was trying to  
accomplish. :-)  I was trying to do this with the minimal amount of  
overhead and I think that might be the ticket.  The cool thing about  
having Freeswitch log it and then pick it up with Fail2ban would be  
that you could define the limit per user/IP.  Say for instance you are  
using Freeswitch as an SBC and if the invite matches a user or an IP  
that is known, that limit could be much higher than an unknown IP or  
user.  That would keep you from having to define a limit that is  
overly "safe" to prevent blocking real traffic especially if you are  
talking about a lot of traffic.

Thanks again,
Spencer

On Feb 20, 2011, at 11:22 PM, jay binks wrote:

> Howdy
>
>    mmm I just had a quick look at mod_sofia , and unfortunatly it  
> looks like the channel is hungup after the challenge is sent.   But  
> my guess is that there HAS To be a list or similar somewhere that  
> tracks endpoints waiting for challenge responses.
>
> Ill see if can track it down and submit a patch, because I too would  
> like to see this at least logged.
>
> as for rate-limiting responses you can have iptables drop packets  
> over X number of invites per sec ...
> ( But they are dropped silently at the Firewall - Kristians SIP  
> Ratelimiter has this in it )
>
> or you can use mod_limit to allow X number of invites per sec also,   
> you would want to tell FS to hit the dialplan before codec  
> negotiation though to do that nice and early in the piece.
>
> Jay
>
> On Mon, Feb 21, 2011 at 5:10 PM, Spencer Thomason <spencer at 5ninesolutions.com 
> > wrote:
> Yes, that works great if they respond to the challenge with a failed
> auth. But the scenario I'm trying to prevent is if they just send the
> INVITE and never respond to the challenge.  Fail2Ban will not work as
> every endpoint will initially send an INVITE and receive a challenge.
> Legit calls will then respond correctly and not be logged as a SIP
> auth failure but every call that is challenged will show up as SIP
> auth challenge in the logs so there is no regex to differentiate
> between legit an non legit traffic.
>
> Spencer
>
> On Feb 20, 2011, at 10:39 PM, Ken Rice wrote:
>
> > Fail2Ban ... This is block an IP with too many failed attempts from
> > something like SipVicious pretty quickly
> >
> >
> > On 2/20/11 11:07 PM, "Spencer Thomason" <spencer at 5ninesolutions.com>
> > wrote:
> >
> >> Hi,
> >> We run hosted Freeswitch instances in VMs with the internal profile
> >> on
> >> port 5060 connecting to clients mostly behind NAT and then the
> >> external profile connecting to our proxies only.  Protecting the
> >> external profile its straightforward.. we only allow traffic to/ 
> from
> >> our proxies at the firewall level.  But protecting the internal
> >> profile seems to be a bit more difficult because the UACs could be
> >> theoretically anywhere on the network.
> >>
> >> I'm currently using Fail2Ban to prevent brute force registration  
> and
> >> INVITEs on auth failures, e.g.:
> >> failregex = \[WARNING\] sofia_reg.c:\d+ SIP auth failure \ 
> (REGISTER\)
> >> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
> >>             \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(INVITE\)
> >> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
> >>
> >> My question is, since its part of a normal SIP dialog to challenge
> >> the
> >> INVITE, is there any way to prevent a possible DoS from just sheer
> >> volume of incoming INVITEs on an Internet facing server
> >> automatically.  I.e., If you block the logged challenge, you'd  
> block
> >> all legitimate INVITEs and registrations.  Since its UDP traffic I
> >> couldn't come up with a way to do it automatically at the iptables
> >> level. i.e. number of concurrent connections.  Is there some option
> >> to
> >> just not respond if a client is sending a number of requests over a
> >> certain threshold?  It might not stop them from sending the traffic
> >> but pretty soon they'd get the idea that it wasn't going to go
> >> anywhere.  My concern is say there are 50 Freeswitch instances on a
> >> box (albeit 8 core, 32GB ram, 8 15K raid 10 storage) and someone
> >> starts sending thousands of rouge INVITEs to every VM on a physical
> >> box that the CPU load from just challenging the incoming INVITEs
> >> would
> >> create a DoS.  We the logs regularly to try to catch people doing
> >> this
> >> sort of thing and drop them at a router upstream of the core  
> network,
> >> but I'd like to have it happen without human intervention.  Have I
> >> completely over thought this and am missing something obvious?
> >>
> >> Thanks,
> >> 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
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 erik.dekkers at wvds.nl  Mon Feb 21 11:18:29 2011
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Mon, 21 Feb 2011 09:18:29 +0100
Subject: [Freeswitch-users] ODBC error STATE: HY000 CODE 21 ERROR
In-Reply-To: <533D2543-EC11-4945-9100-45E9E6520AE6@gmail.com>
References: <CDF5E00590C9BA4AB96BE7CDD28C86AE07872CFA89@FS1.wvds.local>
	<AANLkTikC16qswxL-PnrUAqmrLKBKigBULG2TmjE7PZr7@mail.gmail.com>
	<533D2543-EC11-4945-9100-45E9E6520AE6@gmail.com>
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE07873018F2@FS1.wvds.local>

Thnx for the answer,

Mod_cidlookup requires that you set a DSN so it knows where to get the data right? How would I connect mod_cidlookup to a db without setting a DSN?

Regards 

Erik

-----Oorspronkelijk bericht-----
Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Steven Ayre
Verzonden: maandag 21 februari 2011 1:04
Aan: FreeSWITCH Users Help
Onderwerp: Re: [Freeswitch-users] ODBC error STATE: HY000 CODE 21 ERROR

Correct. Sqlite doesn't use odbc, you only need unixodbc if you want to connect via odbc to mysql, mssql, postgresql etc.

Steve on iPhone

On 20 Feb 2011, at 21:34, curriegrad2004 <curriegrad2004 at gmail.com> wrote:

> sqlite should be built in freeswitch without needing unixODBC. There's 
> no need to use unixODBC unless you're trying to use MySQL or MSSQL 
> with FreeSwitch, iirc.
> 
> On Sun, Feb 20, 2011 at 10:41 AM, Erik Dekkers <erik.dekkers at wvds.nl> wrote:
>> Hi guys,
>> 
>> 
>> 
>> At the moment I'm doing some testing with mod_cidlookup. My intention 
>> is to match the incoming numbers against a local database. Right now 
>> im trying to set it up with SQLite.
>> 
>> I've installed SQLite, unixODBC and created a DSN (freeswitch is also 
>> compiled with odbc support). When testing the connection to the 
>> SQLite database with the isql utility everything works:
>> 
>> 
>> 
>> +---------------------------------------+
>> | Connected!                            |
>> |                                       |
>> | sql-statement                         |
>> | help [tablename]                      |
>> | quit                                  |
>> |                                       |
>> +---------------------------------------+
>> SQL>
>> 
>> 
>> 
>> 
>> 
>> Unfortunately when starting freeswitch I get this error in the console:
>> 
>> 
>> 
>> 2011-02-20 19:28:44.203665 [ERR] switch_odbc.c:365 STATE: HY000 CODE 
>> 21
>> ERROR: [unixODBC][SQLite]connect failed
>> 
>> 2011-02-20 19:28:44.203672 [CRIT] switch_core_sqldb.c:386 Failure!
>> 2011-02-20 19:28:44.203677 [CRIT] mod_cidlookup.c:137 Cannot Open 
>> ODBC Database!
>> 
>> 
>> 
>> 
>> 
>> Here's my config from /conf/autoload_configs/cidlookup.conf.xml:
>> 
>> 
>> 
>> <configuration name="cidlookup.conf" description="cidlookup Configuration">
>>   <settings>
>>     <param name="cache" value="false"/>
>>     <!-- expire is in seconds -->
>>     <param name="cache-expire" value="86400"/>
>>     <param name="odbc-dsn" value="mysqlitedb"/>
>> 
>>     <!-- comment out sql to not setup a database (directory) lookup -->
>>     <param name="sql" value="
>>      SELECT name||' ('||type||')' AS name
>>       FROM phonebook p JOIN numbers n ON p.id = n.phonebook_id
>>       WHERE n.number='${caller_id_number}'
>>       LIMIT 1
>>       "/>
>>   </settings>
>> </configuration>
>> 
>> I someone could point me into the right direction that would be great.
>> 
>> Kind regards,
>> 
>> Erik Dekkers (wvds-nl)
>> _______________________________________________
>> FreeSWITCH-users 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
>> ers
>> http://www.freeswitch.org
>> 
>> 
> 
> _______________________________________________
> FreeSWITCH-users 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 erik.dekkers at wvds.nl  Mon Feb 21 11:32:20 2011
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Mon, 21 Feb 2011 09:32:20 +0100
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <AANLkTikUsVJbB+bwsfe6n60jw5mOCja=WnnCMzbosJNJ@mail.gmail.com>
References: <AANLkTikUsVJbB+bwsfe6n60jw5mOCja=WnnCMzbosJNJ@mail.gmail.com>
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE07873018F3@FS1.wvds.local>

Hi Essobi,

Thnx for the viagra. The girlfriend really liked it!  :)

Regards

Erik (wvds-nl)

-----Oorspronkelijk bericht-----
Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Essobi
Verzonden: zondag 20 februari 2011 21:41
Aan: changstephens at yahoo.com; piscescdt at gmail.com; files at quickofficeconnect.com; freeswitch-users at lists.freeswitch.org; iansgh at gmail.com; iunna at carrionfields.com; jasonpittman502 at gmail.com
Onderwerp: [Freeswitch-users] (no subject)

 http://hiawatha-von-hesperus.com/cool01.11.php?ID=731

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at 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 Feb 21 11:37:15 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 21 Feb 2011 08:37:15 +0000
Subject: [Freeswitch-users] ODBC error STATE: HY000 CODE 21 ERROR
In-Reply-To: <CDF5E00590C9BA4AB96BE7CDD28C86AE07873018F2@FS1.wvds.local>
References: <CDF5E00590C9BA4AB96BE7CDD28C86AE07872CFA89@FS1.wvds.local>
	<AANLkTikC16qswxL-PnrUAqmrLKBKigBULG2TmjE7PZr7@mail.gmail.com>
	<533D2543-EC11-4945-9100-45E9E6520AE6@gmail.com>
	<CDF5E00590C9BA4AB96BE7CDD28C86AE07873018F2@FS1.wvds.local>
Message-ID: <AANLkTimL18vK4UAdnQkQz=i_qtVD+PV6oWBwU-=8StRo@mail.gmail.com>

While ODBC isn't required for the core for a few modules such as mod_sofia,
most modules that use a database do need ODBC - mod_cidlookup is one of
them.

-Steve


On 21 February 2011 08:18, Erik Dekkers <erik.dekkers at wvds.nl> wrote:

> Thnx for the answer,
>
> Mod_cidlookup requires that you set a DSN so it knows where to get the data
> right? How would I connect mod_cidlookup to a db without setting a DSN?
>
> Regards
>
> Erik
>
> -----Oorspronkelijk bericht-----
> Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] Namens Steven Ayre
> Verzonden: maandag 21 februari 2011 1:04
> Aan: FreeSWITCH Users Help
> Onderwerp: Re: [Freeswitch-users] ODBC error STATE: HY000 CODE 21 ERROR
>
> Correct. Sqlite doesn't use odbc, you only need unixodbc if you want to
> connect via odbc to mysql, mssql, postgresql etc.
>
> Steve on iPhone
>
> On 20 Feb 2011, at 21:34, curriegrad2004 <curriegrad2004 at gmail.com> wrote:
>
> > sqlite should be built in freeswitch without needing unixODBC. There's
> > no need to use unixODBC unless you're trying to use MySQL or MSSQL
> > with FreeSwitch, iirc.
> >
> > On Sun, Feb 20, 2011 at 10:41 AM, Erik Dekkers <erik.dekkers at wvds.nl>
> wrote:
> >> Hi guys,
> >>
> >>
> >>
> >> At the moment I'm doing some testing with mod_cidlookup. My intention
> >> is to match the incoming numbers against a local database. Right now
> >> im trying to set it up with SQLite.
> >>
> >> I've installed SQLite, unixODBC and created a DSN (freeswitch is also
> >> compiled with odbc support). When testing the connection to the
> >> SQLite database with the isql utility everything works:
> >>
> >>
> >>
> >> +---------------------------------------+
> >> | Connected!                            |
> >> |                                       |
> >> | sql-statement                         |
> >> | help [tablename]                      |
> >> | quit                                  |
> >> |                                       |
> >> +---------------------------------------+
> >> SQL>
> >>
> >>
> >>
> >>
> >>
> >> Unfortunately when starting freeswitch I get this error in the console:
> >>
> >>
> >>
> >> 2011-02-20 19:28:44.203665 [ERR] switch_odbc.c:365 STATE: HY000 CODE
> >> 21
> >> ERROR: [unixODBC][SQLite]connect failed
> >>
> >> 2011-02-20 19:28:44.203672 [CRIT] switch_core_sqldb.c:386 Failure!
> >> 2011-02-20 19:28:44.203677 [CRIT] mod_cidlookup.c:137 Cannot Open
> >> ODBC Database!
> >>
> >>
> >>
> >>
> >>
> >> Here's my config from /conf/autoload_configs/cidlookup.conf.xml:
> >>
> >>
> >>
> >> <configuration name="cidlookup.conf" description="cidlookup
> Configuration">
> >>   <settings>
> >>     <param name="cache" value="false"/>
> >>     <!-- expire is in seconds -->
> >>     <param name="cache-expire" value="86400"/>
> >>     <param name="odbc-dsn" value="mysqlitedb"/>
> >>
> >>     <!-- comment out sql to not setup a database (directory) lookup -->
> >>     <param name="sql" value="
> >>      SELECT name||' ('||type||')' AS name
> >>       FROM phonebook p JOIN numbers n ON p.id = n.phonebook_id
> >>       WHERE n.number='${caller_id_number}'
> >>       LIMIT 1
> >>       "/>
> >>   </settings>
> >> </configuration>
> >>
> >> I someone could point me into the right direction that would be great.
> >>
> >> Kind regards,
> >>
> >> Erik Dekkers (wvds-nl)
> >> _______________________________________________
> >> FreeSWITCH-users 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
> >> ers
> >> http://www.freeswitch.org
> >>
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 21 11:38:46 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 21 Feb 2011 08:38:46 +0000
Subject: [Freeswitch-users] How to limit a certain gateway/extension
In-Reply-To: <AANLkTimE+TqWkX4tEiiZHqi9rKO=q1+s21U0KZumnn4O@mail.gmail.com>
References: <AANLkTimE+TqWkX4tEiiZHqi9rKO=q1+s21U0KZumnn4O@mail.gmail.com>
Message-ID: <AANLkTimdU1m_mN8m91ukJmOiYN_cPHahkZiAFfbc0Rfn@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Limit



On 21 February 2011 03:40, Hareem Haque <hareem.haque at gmail.com> wrote:

> Hello Everyone
>
> Many thanks for helping me out. I got most of the switch setup. The mod_cdr
> works well now. I got some ideas of the routing patterns. I have1 question
> and would greatly appreciate your help in resolving this issue.
>
>
>  How can i limit a certain extension/gateway to only able to initiate a
> predefined number of calls. For example extension 1000 connects to the
> switch. And now that extension can only make 24 concurrent calls. How can i
> set this up.
>
>
>
>
> Best Regards
> Hareem. Haque
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Mon Feb 21 11:41:54 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Mon, 21 Feb 2011 00:41:54 -0800
Subject: [Freeswitch-users] INVITE DoS Prevention
In-Reply-To: <ECB69E6D-F45D-4992-90B3-58911978BC91@5ninesolutions.com>
References: <C98764B0.11627%krice@freeswitch.org>
	<A624F826-AF73-40BD-8DCD-10F39551D924@5ninesolutions.com>
	<AANLkTi=+f0GqChj4PeW=XFjQW_mYTtJYPuWEj47gw8r+@mail.gmail.com>
	<ECB69E6D-F45D-4992-90B3-58911978BC91@5ninesolutions.com>
Message-ID: <929FDF0C-A800-4D2B-865A-136FDA18CAC4@5ninesolutions.com>

BTW.. here is a link :-)

http://etel.wiki.oreilly.com/wiki/index.php/SIP_DoS/DDoS_Mitigation


On Feb 21, 2011, at 12:04 AM, Spencer Thomason wrote:

> Thanks for the info!  Yes it would be great if Freeswitch logged  
> this info somewhere but this firewall script was was I was trying to  
> accomplish. :-)  I was trying to do this with the minimal amount of  
> overhead and I think that might be the ticket.  The cool thing about  
> having Freeswitch log it and then pick it up with Fail2ban would be  
> that you could define the limit per user/IP.  Say for instance you  
> are using Freeswitch as an SBC and if the invite matches a user or  
> an IP that is known, that limit could be much higher than an unknown  
> IP or user.  That would keep you from having to define a limit that  
> is overly "safe" to prevent blocking real traffic especially if you  
> are talking about a lot of traffic.
>
> Thanks again,
> Spencer
>
> On Feb 20, 2011, at 11:22 PM, jay binks wrote:
>
>> Howdy
>>
>>    mmm I just had a quick look at mod_sofia , and unfortunatly it  
>> looks like the channel is hungup after the challenge is sent.   But  
>> my guess is that there HAS To be a list or similar somewhere that  
>> tracks endpoints waiting for challenge responses.
>>
>> Ill see if can track it down and submit a patch, because I too  
>> would like to see this at least logged.
>>
>> as for rate-limiting responses you can have iptables drop packets  
>> over X number of invites per sec ...
>> ( But they are dropped silently at the Firewall - Kristians SIP  
>> Ratelimiter has this in it )
>>
>> or you can use mod_limit to allow X number of invites per sec  
>> also,  you would want to tell FS to hit the dialplan before codec  
>> negotiation though to do that nice and early in the piece.
>>
>> Jay
>>
>> On Mon, Feb 21, 2011 at 5:10 PM, Spencer Thomason <spencer at 5ninesolutions.com 
>> > wrote:
>> Yes, that works great if they respond to the challenge with a failed
>> auth. But the scenario I'm trying to prevent is if they just send the
>> INVITE and never respond to the challenge.  Fail2Ban will not work as
>> every endpoint will initially send an INVITE and receive a challenge.
>> Legit calls will then respond correctly and not be logged as a SIP
>> auth failure but every call that is challenged will show up as SIP
>> auth challenge in the logs so there is no regex to differentiate
>> between legit an non legit traffic.
>>
>> Spencer
>>
>> On Feb 20, 2011, at 10:39 PM, Ken Rice wrote:
>>
>> > Fail2Ban ... This is block an IP with too many failed attempts from
>> > something like SipVicious pretty quickly
>> >
>> >
>> > On 2/20/11 11:07 PM, "Spencer Thomason"  
>> <spencer at 5ninesolutions.com>
>> > wrote:
>> >
>> >> Hi,
>> >> We run hosted Freeswitch instances in VMs with the internal  
>> profile
>> >> on
>> >> port 5060 connecting to clients mostly behind NAT and then the
>> >> external profile connecting to our proxies only.  Protecting the
>> >> external profile its straightforward.. we only allow traffic to/ 
>> from
>> >> our proxies at the firewall level.  But protecting the internal
>> >> profile seems to be a bit more difficult because the UACs could be
>> >> theoretically anywhere on the network.
>> >>
>> >> I'm currently using Fail2Ban to prevent brute force registration  
>> and
>> >> INVITEs on auth failures, e.g.:
>> >> failregex = \[WARNING\] sofia_reg.c:\d+ SIP auth failure \ 
>> (REGISTER\)
>> >> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
>> >>             \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(INVITE 
>> \)
>> >> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
>> >>
>> >> My question is, since its part of a normal SIP dialog to challenge
>> >> the
>> >> INVITE, is there any way to prevent a possible DoS from just sheer
>> >> volume of incoming INVITEs on an Internet facing server
>> >> automatically.  I.e., If you block the logged challenge, you'd  
>> block
>> >> all legitimate INVITEs and registrations.  Since its UDP traffic I
>> >> couldn't come up with a way to do it automatically at the iptables
>> >> level. i.e. number of concurrent connections.  Is there some  
>> option
>> >> to
>> >> just not respond if a client is sending a number of requests  
>> over a
>> >> certain threshold?  It might not stop them from sending the  
>> traffic
>> >> but pretty soon they'd get the idea that it wasn't going to go
>> >> anywhere.  My concern is say there are 50 Freeswitch instances  
>> on a
>> >> box (albeit 8 core, 32GB ram, 8 15K raid 10 storage) and someone
>> >> starts sending thousands of rouge INVITEs to every VM on a  
>> physical
>> >> box that the CPU load from just challenging the incoming INVITEs
>> >> would
>> >> create a DoS.  We the logs regularly to try to catch people doing
>> >> this
>> >> sort of thing and drop them at a router upstream of the core  
>> network,
>> >> but I'd like to have it happen without human intervention.  Have I
>> >> completely over thought this and am missing something obvious?
>> >>
>> >> Thanks,
>> >> 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
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 erik.dekkers at wvds.nl  Mon Feb 21 11:49:30 2011
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Mon, 21 Feb 2011 09:49:30 +0100
Subject: [Freeswitch-users] ODBC error STATE: HY000 CODE 21 ERROR
In-Reply-To: <AANLkTimL18vK4UAdnQkQz=i_qtVD+PV6oWBwU-=8StRo@mail.gmail.com>
References: <CDF5E00590C9BA4AB96BE7CDD28C86AE07872CFA89@FS1.wvds.local>
	<AANLkTikC16qswxL-PnrUAqmrLKBKigBULG2TmjE7PZr7@mail.gmail.com>
	<533D2543-EC11-4945-9100-45E9E6520AE6@gmail.com>
	<CDF5E00590C9BA4AB96BE7CDD28C86AE07873018F2@FS1.wvds.local>
	<AANLkTimL18vK4UAdnQkQz=i_qtVD+PV6oWBwU-=8StRo@mail.gmail.com>
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE07873018F6@FS1.wvds.local>

Steve,

Thnx for this explanation! Now it's clear to me.
Still having the connection failure. Any idea's how to fix it? The errorcodes doesn't provide detailed information of the problem and how to fix it.

Thnx

Erik

Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Steven Ayre
Verzonden: maandag 21 februari 2011 9:37
Aan: FreeSWITCH Users Help
Onderwerp: Re: [Freeswitch-users] ODBC error STATE: HY000 CODE 21 ERROR

While ODBC isn't required for the core for a few modules such as mod_sofia, most modules that use a database do need ODBC - mod_cidlookup is one of them.

-Steve

On 21 February 2011 08:18, Erik Dekkers <erik.dekkers at wvds.nl<mailto:erik.dekkers at wvds.nl>> wrote:
Thnx for the answer,

Mod_cidlookup requires that you set a DSN so it knows where to get the data right? How would I connect mod_cidlookup to a db without setting a DSN?

Regards

Erik

-----Oorspronkelijk bericht-----
Van: 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>] Namens Steven Ayre
Verzonden: maandag 21 februari 2011 1:04
Aan: FreeSWITCH Users Help
Onderwerp: Re: [Freeswitch-users] ODBC error STATE: HY000 CODE 21 ERROR

Correct. Sqlite doesn't use odbc, you only need unixodbc if you want to connect via odbc to mysql, mssql, postgresql etc.

Steve on iPhone

On 20 Feb 2011, at 21:34, curriegrad2004 <curriegrad2004 at gmail.com<mailto:curriegrad2004 at gmail.com>> wrote:

> sqlite should be built in freeswitch without needing unixODBC. There's
> no need to use unixODBC unless you're trying to use MySQL or MSSQL
> with FreeSwitch, iirc.
>
> On Sun, Feb 20, 2011 at 10:41 AM, Erik Dekkers <erik.dekkers at wvds.nl<mailto:erik.dekkers at wvds.nl>> wrote:
>> Hi guys,
>>
>>
>>
>> At the moment I'm doing some testing with mod_cidlookup. My intention
>> is to match the incoming numbers against a local database. Right now
>> im trying to set it up with SQLite.
>>
>> I've installed SQLite, unixODBC and created a DSN (freeswitch is also
>> compiled with odbc support). When testing the connection to the
>> SQLite database with the isql utility everything works:
>>
>>
>>
>> +---------------------------------------+
>> | Connected!                            |
>> |                                       |
>> | sql-statement                         |
>> | help [tablename]                      |
>> | quit                                  |
>> |                                       |
>> +---------------------------------------+
>> SQL>
>>
>>
>>
>>
>>
>> Unfortunately when starting freeswitch I get this error in the console:
>>
>>
>>
>> 2011-02-20 19:28:44.203665 [ERR] switch_odbc.c:365 STATE: HY000 CODE
>> 21
>> ERROR: [unixODBC][SQLite]connect failed
>>
>> 2011-02-20 19:28:44.203672 [CRIT] switch_core_sqldb.c:386 Failure!
>> 2011-02-20 19:28:44.203677 [CRIT] mod_cidlookup.c:137 Cannot Open
>> ODBC Database!
>>
>>
>>
>>
>>
>> Here's my config from /conf/autoload_configs/cidlookup.conf.xml:
>>
>>
>>
>> <configuration name="cidlookup.conf" description="cidlookup Configuration">
>>   <settings>
>>     <param name="cache" value="false"/>
>>     <!-- expire is in seconds -->
>>     <param name="cache-expire" value="86400"/>
>>     <param name="odbc-dsn" value="mysqlitedb"/>
>>
>>     <!-- comment out sql to not setup a database (directory) lookup -->
>>     <param name="sql" value="
>>      SELECT name||' ('||type||')' AS name
>>       FROM phonebook p JOIN numbers n ON p.id<http://p.id> = n.phonebook_id
>>       WHERE n.number='${caller_id_number}'
>>       LIMIT 1
>>       "/>
>>   </settings>
>> </configuration>
>>
>> I someone could point me into the right direction that would be great.
>>
>> Kind regards,
>>
>> Erik Dekkers (wvds-nl)
>> _______________________________________________
>> 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-us
>> ers
>> 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-use
> rs
> 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
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From steveayre at gmail.com  Mon Feb 21 12:17:46 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 21 Feb 2011 09:17:46 +0000
Subject: [Freeswitch-users] ODBC error STATE: HY000 CODE 21 ERROR
In-Reply-To: <CDF5E00590C9BA4AB96BE7CDD28C86AE07873018F6@FS1.wvds.local>
References: <CDF5E00590C9BA4AB96BE7CDD28C86AE07872CFA89@FS1.wvds.local>
	<AANLkTikC16qswxL-PnrUAqmrLKBKigBULG2TmjE7PZr7@mail.gmail.com>
	<533D2543-EC11-4945-9100-45E9E6520AE6@gmail.com>
	<CDF5E00590C9BA4AB96BE7CDD28C86AE07873018F2@FS1.wvds.local>
	<AANLkTimL18vK4UAdnQkQz=i_qtVD+PV6oWBwU-=8StRo@mail.gmail.com>
	<CDF5E00590C9BA4AB96BE7CDD28C86AE07873018F6@FS1.wvds.local>
Message-ID: <AANLkTikfitq+SPPekWxJTQtEiaJNgQD0TyWJeid4QuDm@mail.gmail.com>

mysqlitedb is your dsn? Can you can share your unixodbc config files -
perhaps there's a problem in there.

-Steve


On 21 February 2011 08:49, Erik Dekkers <erik.dekkers at wvds.nl> wrote:

> Steve,
>
>
>
> Thnx for this explanation! Now it?s clear to me.
>
> Still having the connection failure. Any idea?s how to fix it? The
> errorcodes doesn?t provide detailed information of the problem and how to
> fix it.
>
>
>
> Thnx
>
>
>
> Erik
>
>
>
> *Van:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *N**amens *
> Steven Ayre
> *Verzonden:* maandag 21 februari 2011 9:37
>
> *Aan:* FreeSWITCH Users Help
> *Onderwerp:* Re: [Freeswitch-users] ODBC error STATE: HY000 CODE 21 ERROR
>
>
>
> While ODBC isn't required for the core for a few modules such as mod_sofia,
> most modules that use a database do need ODBC - mod_cidlookup is one of
> them.
>
> -Steve
>
> On 21 February 2011 08:18, Erik Dekkers <erik.dekkers at wvds.nl> wrote:
>
> Thnx for the answer,
>
> Mod_cidlookup requires that you set a DSN so it knows where to get the data
> right? How would I connect mod_cidlookup to a db without setting a DSN?
>
> Regards
>
> Erik
>
> -----Oorspronkelijk bericht-----
> Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] Namens Steven Ayre
> Verzonden: maandag 21 februari 2011 1:04
> Aan: FreeSWITCH Users Help
> Onderwerp: Re: [Freeswitch-users] ODBC error STATE: HY000 CODE 21 ERROR
>
>
> Correct. Sqlite doesn't use odbc, you only need unixodbc if you want to
> connect via odbc to mysql, mssql, postgresql etc.
>
> Steve on iPhone
>
> On 20 Feb 2011, at 21:34, curriegrad2004 <curriegrad2004 at gmail.com> wrote:
>
> > sqlite should be built in freeswitch without needing unixODBC. There's
> > no need to use unixODBC unless you're trying to use MySQL or MSSQL
> > with FreeSwitch, iirc.
> >
> > On Sun, Feb 20, 2011 at 10:41 AM, Erik Dekkers <erik.dekkers at wvds.nl>
> wrote:
> >> Hi guys,
> >>
> >>
> >>
> >> At the moment I'm doing some testing with mod_cidlookup. My intention
> >> is to match the incoming numbers against a local database. Right now
> >> im trying to set it up with SQLite.
> >>
> >> I've installed SQLite, unixODBC and created a DSN (freeswitch is also
> >> compiled with odbc support). When testing the connection to the
> >> SQLite database with the isql utility everything works:
> >>
> >>
> >>
> >> +---------------------------------------+
> >> | Connected!                            |
> >> |                                       |
> >> | sql-statement                         |
> >> | help [tablename]                      |
> >> | quit                                  |
> >> |                                       |
> >> +---------------------------------------+
> >> SQL>
> >>
> >>
> >>
> >>
> >>
> >> Unfortunately when starting freeswitch I get this error in the console:
> >>
> >>
> >>
> >> 2011-02-20 19:28:44.203665 [ERR] switch_odbc.c:365 STATE: HY000 CODE
> >> 21
> >> ERROR: [unixODBC][SQLite]connect failed
> >>
> >> 2011-02-20 19:28:44.203672 [CRIT] switch_core_sqldb.c:386 Failure!
> >> 2011-02-20 19:28:44.203677 [CRIT] mod_cidlookup.c:137 Cannot Open
> >> ODBC Database!
> >>
> >>
> >>
> >>
> >>
> >> Here's my config from /conf/autoload_configs/cidlookup.conf.xml:
> >>
> >>
> >>
> >> <configuration name="cidlookup.conf" description="cidlookup
> Configuration">
> >>   <settings>
> >>     <param name="cache" value="false"/>
> >>     <!-- expire is in seconds -->
> >>     <param name="cache-expire" value="86400"/>
> >>     <param name="odbc-dsn" value="mysqlitedb"/>
> >>
> >>     <!-- comment out sql to not setup a database (directory) lookup -->
> >>     <param name="sql" value="
> >>      SELECT name||' ('||type||')' AS name
> >>       FROM phonebook p JOIN numbers n ON p.id = n.phonebook_id
> >>       WHERE n.number='${caller_id_number}'
> >>       LIMIT 1
> >>       "/>
> >>   </settings>
> >> </configuration>
> >>
> >> I someone could point me into the right direction that would be great.
> >>
> >> Kind regards,
> >>
> >> Erik Dekkers (wvds-nl)
> >> _______________________________________________
> >> FreeSWITCH-users 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
> >> ers
> >> http://www.freeswitch.org
> >>
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 21 12:19:54 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 21 Feb 2011 09:19:54 +0000
Subject: [Freeswitch-users] ODBC error STATE: HY000 CODE 21 ERROR
In-Reply-To: <AANLkTikfitq+SPPekWxJTQtEiaJNgQD0TyWJeid4QuDm@mail.gmail.com>
References: <CDF5E00590C9BA4AB96BE7CDD28C86AE07872CFA89@FS1.wvds.local>
	<AANLkTikC16qswxL-PnrUAqmrLKBKigBULG2TmjE7PZr7@mail.gmail.com>
	<533D2543-EC11-4945-9100-45E9E6520AE6@gmail.com>
	<CDF5E00590C9BA4AB96BE7CDD28C86AE07873018F2@FS1.wvds.local>
	<AANLkTimL18vK4UAdnQkQz=i_qtVD+PV6oWBwU-=8StRo@mail.gmail.com>
	<CDF5E00590C9BA4AB96BE7CDD28C86AE07873018F6@FS1.wvds.local>
	<AANLkTikfitq+SPPekWxJTQtEiaJNgQD0TyWJeid4QuDm@mail.gmail.com>
Message-ID: <AANLkTikXEbfG2Yeq7Rcqg8gbF0n7p31n-TquxERy+bTe@mail.gmail.com>

Error 21 is Is A Directory, but I don't know whether that's the same as a
Linux error code...

Do you have the ODBC configuration pointing at the db directory? Sqllite
databases are the .db files in the db directory, not the directory itself.

-Steve

On 21 February 2011 09:17, Steven Ayre <steveayre at gmail.com> wrote:

> mysqlitedb is your dsn? Can you can share your unixodbc config files -
> perhaps there's a problem in there.
>
> -Steve
>
>
> On 21 February 2011 08:49, Erik Dekkers <erik.dekkers at wvds.nl> wrote:
>
>> Steve,
>>
>>
>>
>> Thnx for this explanation! Now it?s clear to me.
>>
>> Still having the connection failure. Any idea?s how to fix it? The
>> errorcodes doesn?t provide detailed information of the problem and how to
>> fix it.
>>
>>
>>
>> Thnx
>>
>>
>>
>> Erik
>>
>>
>>
>> *Van:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *N**amens *
>> Steven Ayre
>> *Verzonden:* maandag 21 februari 2011 9:37
>>
>> *Aan:* FreeSWITCH Users Help
>> *Onderwerp:* Re: [Freeswitch-users] ODBC error STATE: HY000 CODE 21 ERROR
>>
>>
>>
>> While ODBC isn't required for the core for a few modules such as
>> mod_sofia, most modules that use a database do need ODBC - mod_cidlookup is
>> one of them.
>>
>> -Steve
>>
>> On 21 February 2011 08:18, Erik Dekkers <erik.dekkers at wvds.nl> wrote:
>>
>> Thnx for the answer,
>>
>> Mod_cidlookup requires that you set a DSN so it knows where to get the
>> data right? How would I connect mod_cidlookup to a db without setting a DSN?
>>
>> Regards
>>
>> Erik
>>
>> -----Oorspronkelijk bericht-----
>> Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] Namens Steven Ayre
>> Verzonden: maandag 21 februari 2011 1:04
>> Aan: FreeSWITCH Users Help
>> Onderwerp: Re: [Freeswitch-users] ODBC error STATE: HY000 CODE 21 ERROR
>>
>>
>> Correct. Sqlite doesn't use odbc, you only need unixodbc if you want to
>> connect via odbc to mysql, mssql, postgresql etc.
>>
>> Steve on iPhone
>>
>> On 20 Feb 2011, at 21:34, curriegrad2004 <curriegrad2004 at gmail.com>
>> wrote:
>>
>> > sqlite should be built in freeswitch without needing unixODBC. There's
>> > no need to use unixODBC unless you're trying to use MySQL or MSSQL
>> > with FreeSwitch, iirc.
>> >
>> > On Sun, Feb 20, 2011 at 10:41 AM, Erik Dekkers <erik.dekkers at wvds.nl>
>> wrote:
>> >> Hi guys,
>> >>
>> >>
>> >>
>> >> At the moment I'm doing some testing with mod_cidlookup. My intention
>> >> is to match the incoming numbers against a local database. Right now
>> >> im trying to set it up with SQLite.
>> >>
>> >> I've installed SQLite, unixODBC and created a DSN (freeswitch is also
>> >> compiled with odbc support). When testing the connection to the
>> >> SQLite database with the isql utility everything works:
>> >>
>> >>
>> >>
>> >> +---------------------------------------+
>> >> | Connected!                            |
>> >> |                                       |
>> >> | sql-statement                         |
>> >> | help [tablename]                      |
>> >> | quit                                  |
>> >> |                                       |
>> >> +---------------------------------------+
>> >> SQL>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >> Unfortunately when starting freeswitch I get this error in the console:
>> >>
>> >>
>> >>
>> >> 2011-02-20 19:28:44.203665 [ERR] switch_odbc.c:365 STATE: HY000 CODE
>> >> 21
>> >> ERROR: [unixODBC][SQLite]connect failed
>> >>
>> >> 2011-02-20 19:28:44.203672 [CRIT] switch_core_sqldb.c:386 Failure!
>> >> 2011-02-20 19:28:44.203677 [CRIT] mod_cidlookup.c:137 Cannot Open
>> >> ODBC Database!
>> >>
>> >>
>> >>
>> >>
>> >>
>> >> Here's my config from /conf/autoload_configs/cidlookup.conf.xml:
>> >>
>> >>
>> >>
>> >> <configuration name="cidlookup.conf" description="cidlookup
>> Configuration">
>> >>   <settings>
>> >>     <param name="cache" value="false"/>
>> >>     <!-- expire is in seconds -->
>> >>     <param name="cache-expire" value="86400"/>
>> >>     <param name="odbc-dsn" value="mysqlitedb"/>
>> >>
>> >>     <!-- comment out sql to not setup a database (directory) lookup -->
>> >>     <param name="sql" value="
>> >>      SELECT name||' ('||type||')' AS name
>> >>       FROM phonebook p JOIN numbers n ON p.id = n.phonebook_id
>> >>       WHERE n.number='${caller_id_number}'
>> >>       LIMIT 1
>> >>       "/>
>> >>   </settings>
>> >> </configuration>
>> >>
>> >> I someone could point me into the right direction that would be great.
>> >>
>> >> Kind regards,
>> >>
>> >> Erik Dekkers (wvds-nl)
>> >> _______________________________________________
>> >> FreeSWITCH-users 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
>> >> ers
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> > _______________________________________________
>> > FreeSWITCH-users 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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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  Mon Feb 21 12:31:18 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Mon, 21 Feb 2011 04:31:18 -0500
Subject: [Freeswitch-users] INVITE DoS Prevention
In-Reply-To: <A624F826-AF73-40BD-8DCD-10F39551D924@5ninesolutions.com>
References: <C98764B0.11627%krice@freeswitch.org>
	<A624F826-AF73-40BD-8DCD-10F39551D924@5ninesolutions.com>
Message-ID: <9922.1298280678@ccs.covici.com>

I would change sip auth failure to challenge and then have sufficient
times to only block if there are too many challenges in a certain time.
I am not even sure the failure works any more in recent gits.

Spencer Thomason <spencer at 5ninesolutions.com> wrote:

> Yes, that works great if they respond to the challenge with a failed  
> auth. But the scenario I'm trying to prevent is if they just send the  
> INVITE and never respond to the challenge.  Fail2Ban will not work as  
> every endpoint will initially send an INVITE and receive a challenge.   
> Legit calls will then respond correctly and not be logged as a SIP  
> auth failure but every call that is challenged will show up as SIP  
> auth challenge in the logs so there is no regex to differentiate  
> between legit an non legit traffic.
> 
> Spencer
> 
> On Feb 20, 2011, at 10:39 PM, Ken Rice wrote:
> 
> > Fail2Ban ... This is block an IP with too many failed attempts from
> > something like SipVicious pretty quickly
> >
> >
> > On 2/20/11 11:07 PM, "Spencer Thomason" <spencer at 5ninesolutions.com>  
> > wrote:
> >
> >> Hi,
> >> We run hosted Freeswitch instances in VMs with the internal profile  
> >> on
> >> port 5060 connecting to clients mostly behind NAT and then the
> >> external profile connecting to our proxies only.  Protecting the
> >> external profile its straightforward.. we only allow traffic to/from
> >> our proxies at the firewall level.  But protecting the internal
> >> profile seems to be a bit more difficult because the UACs could be
> >> theoretically anywhere on the network.
> >>
> >> I'm currently using Fail2Ban to prevent brute force registration and
> >> INVITEs on auth failures, e.g.:
> >> failregex = \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(REGISTER\)
> >> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
> >>             \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(INVITE\)
> >> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
> >>
> >> My question is, since its part of a normal SIP dialog to challenge  
> >> the
> >> INVITE, is there any way to prevent a possible DoS from just sheer
> >> volume of incoming INVITEs on an Internet facing server
> >> automatically.  I.e., If you block the logged challenge, you'd block
> >> all legitimate INVITEs and registrations.  Since its UDP traffic I
> >> couldn't come up with a way to do it automatically at the iptables
> >> level. i.e. number of concurrent connections.  Is there some option  
> >> to
> >> just not respond if a client is sending a number of requests over a
> >> certain threshold?  It might not stop them from sending the traffic
> >> but pretty soon they'd get the idea that it wasn't going to go
> >> anywhere.  My concern is say there are 50 Freeswitch instances on a
> >> box (albeit 8 core, 32GB ram, 8 15K raid 10 storage) and someone
> >> starts sending thousands of rouge INVITEs to every VM on a physical
> >> box that the CPU load from just challenging the incoming INVITEs  
> >> would
> >> create a DoS.  We the logs regularly to try to catch people doing  
> >> this
> >> sort of thing and drop them at a router upstream of the core network,
> >> but I'd like to have it happen without human intervention.  Have I
> >> completely over thought this and am missing something obvious?
> >>
> >> Thanks,
> >> 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
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 avi at avimarcus.net  Mon Feb 21 13:17:25 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 21 Feb 2011 12:17:25 +0200
Subject: [Freeswitch-users] mod_odbc_query
In-Reply-To: <AANLkTimZbs8S5sgkS7z=6+BDjW3f1U=LmO03203WZYUC@mail.gmail.com>
References: <AANLkTimZbs8S5sgkS7z=6+BDjW3f1U=LmO03203WZYUC@mail.gmail.com>
Message-ID: <AANLkTinxvwoptnvXmqFEokvQikHNzspTcXvJLndFJLjv@mail.gmail.com>

You can find the download / compile instructions on the wiki:
http://wiki.freeswitch.org/wiki/Mod_odbc_query

<http://wiki.freeswitch.org/wiki/Mod_odbc_query>-Avi

On Mon, Feb 21, 2011 at 6:30 AM, Sam <u2nsam at gmail.com> wrote:

> Hello,
>
> How to download mod_odbc_query source, there is not such source in 1.0.6 or
> 1.0.7 .
>
> 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 jaybinks at gmail.com  Mon Feb 21 13:24:05 2011
From: jaybinks at gmail.com (Jay Binks)
Date: Mon, 21 Feb 2011 20:24:05 +1000
Subject: [Freeswitch-users] INVITE DoS Prevention
In-Reply-To: <9922.1298280678@ccs.covici.com>
References: <C98764B0.11627%krice@freeswitch.org>
	<A624F826-AF73-40BD-8DCD-10F39551D924@5ninesolutions.com>
	<9922.1298280678@ccs.covici.com>
Message-ID: <1317EEC0-912C-4E6C-9737-1B19B38B5B88@gmail.com>

Auth failure has been working fine for me I the last 2 weeks.  What makes you this it's not been working ??



On 21/02/2011, at 7:31 PM, covici at ccs.covici.com wrote:

> I would change sip auth failure to challenge and then have sufficient
> times to only block if there are too many challenges in a certain time.
> I am not even sure the failure works any more in recent gits.
> 
> Spencer Thomason <spencer at 5ninesolutions.com> wrote:
> 
>> Yes, that works great if they respond to the challenge with a failed  
>> auth. But the scenario I'm trying to prevent is if they just send the  
>> INVITE and never respond to the challenge.  Fail2Ban will not work as  
>> every endpoint will initially send an INVITE and receive a challenge.   
>> Legit calls will then respond correctly and not be logged as a SIP  
>> auth failure but every call that is challenged will show up as SIP  
>> auth challenge in the logs so there is no regex to differentiate  
>> between legit an non legit traffic.
>> 
>> Spencer
>> 
>> On Feb 20, 2011, at 10:39 PM, Ken Rice wrote:
>> 
>>> Fail2Ban ... This is block an IP with too many failed attempts from
>>> something like SipVicious pretty quickly
>>> 
>>> 
>>> On 2/20/11 11:07 PM, "Spencer Thomason" <spencer at 5ninesolutions.com>  
>>> wrote:
>>> 
>>>> Hi,
>>>> We run hosted Freeswitch instances in VMs with the internal profile  
>>>> on
>>>> port 5060 connecting to clients mostly behind NAT and then the
>>>> external profile connecting to our proxies only.  Protecting the
>>>> external profile its straightforward.. we only allow traffic to/from
>>>> our proxies at the firewall level.  But protecting the internal
>>>> profile seems to be a bit more difficult because the UACs could be
>>>> theoretically anywhere on the network.
>>>> 
>>>> I'm currently using Fail2Ban to prevent brute force registration and
>>>> INVITEs on auth failures, e.g.:
>>>> failregex = \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(REGISTER\)
>>>> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
>>>>            \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(INVITE\)
>>>> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
>>>> 
>>>> My question is, since its part of a normal SIP dialog to challenge  
>>>> the
>>>> INVITE, is there any way to prevent a possible DoS from just sheer
>>>> volume of incoming INVITEs on an Internet facing server
>>>> automatically.  I.e., If you block the logged challenge, you'd block
>>>> all legitimate INVITEs and registrations.  Since its UDP traffic I
>>>> couldn't come up with a way to do it automatically at the iptables
>>>> level. i.e. number of concurrent connections.  Is there some option  
>>>> to
>>>> just not respond if a client is sending a number of requests over a
>>>> certain threshold?  It might not stop them from sending the traffic
>>>> but pretty soon they'd get the idea that it wasn't going to go
>>>> anywhere.  My concern is say there are 50 Freeswitch instances on a
>>>> box (albeit 8 core, 32GB ram, 8 15K raid 10 storage) and someone
>>>> starts sending thousands of rouge INVITEs to every VM on a physical
>>>> box that the CPU load from just challenging the incoming INVITEs  
>>>> would
>>>> create a DoS.  We the logs regularly to try to catch people doing  
>>>> this
>>>> sort of thing and drop them at a router upstream of the core network,
>>>> but I'd like to have it happen without human intervention.  Have I
>>>> completely over thought this and am missing something obvious?
>>>> 
>>>> Thanks,
>>>> 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
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 spencer at 5ninesolutions.com  Mon Feb 21 13:28:25 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Mon, 21 Feb 2011 02:28:25 -0800
Subject: [Freeswitch-users] INVITE DoS Prevention
In-Reply-To: <9922.1298280678@ccs.covici.com>
References: <C98764B0.11627%krice@freeswitch.org>
	<A624F826-AF73-40BD-8DCD-10F39551D924@5ninesolutions.com>
	<9922.1298280678@ccs.covici.com>
Message-ID: <B706C824-2C91-4F74-9D68-87FF817EA969@5ninesolutions.com>

After tinkering with it, I think that might be the best way.  The  
iptables method is cool but I'd like to have more dynamic control and  
with Fail2Ban looking at the challenges you could specifically ignore  
certain high traffic IPs and block others.  What would be very cool is  
if instead of logging every challenge, a log entry was written if  
there was a high number from a specific IP, then you could decide what  
to do about it with fail2ban, similar to the pike module for opensips  
does.


On Feb 21, 2011, at 1:31 AM, covici at ccs.covici.com wrote:

> I would change sip auth failure to challenge and then have sufficient
> times to only block if there are too many challenges in a certain  
> time.
> I am not even sure the failure works any more in recent gits.
>
> Spencer Thomason <spencer at 5ninesolutions.com> wrote:
>
>> Yes, that works great if they respond to the challenge with a failed
>> auth. But the scenario I'm trying to prevent is if they just send the
>> INVITE and never respond to the challenge.  Fail2Ban will not work as
>> every endpoint will initially send an INVITE and receive a challenge.
>> Legit calls will then respond correctly and not be logged as a SIP
>> auth failure but every call that is challenged will show up as SIP
>> auth challenge in the logs so there is no regex to differentiate
>> between legit an non legit traffic.
>>
>> Spencer
>>
>> On Feb 20, 2011, at 10:39 PM, Ken Rice wrote:
>>
>>> Fail2Ban ... This is block an IP with too many failed attempts from
>>> something like SipVicious pretty quickly
>>>
>>>
>>> On 2/20/11 11:07 PM, "Spencer Thomason" <spencer at 5ninesolutions.com>
>>> wrote:
>>>
>>>> Hi,
>>>> We run hosted Freeswitch instances in VMs with the internal profile
>>>> on
>>>> port 5060 connecting to clients mostly behind NAT and then the
>>>> external profile connecting to our proxies only.  Protecting the
>>>> external profile its straightforward.. we only allow traffic to/ 
>>>> from
>>>> our proxies at the firewall level.  But protecting the internal
>>>> profile seems to be a bit more difficult because the UACs could be
>>>> theoretically anywhere on the network.
>>>>
>>>> I'm currently using Fail2Ban to prevent brute force registration  
>>>> and
>>>> INVITEs on auth failures, e.g.:
>>>> failregex = \[WARNING\] sofia_reg.c:\d+ SIP auth failure \ 
>>>> (REGISTER\)
>>>> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
>>>>            \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(INVITE\)
>>>> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
>>>>
>>>> My question is, since its part of a normal SIP dialog to challenge
>>>> the
>>>> INVITE, is there any way to prevent a possible DoS from just sheer
>>>> volume of incoming INVITEs on an Internet facing server
>>>> automatically.  I.e., If you block the logged challenge, you'd  
>>>> block
>>>> all legitimate INVITEs and registrations.  Since its UDP traffic I
>>>> couldn't come up with a way to do it automatically at the iptables
>>>> level. i.e. number of concurrent connections.  Is there some option
>>>> to
>>>> just not respond if a client is sending a number of requests over a
>>>> certain threshold?  It might not stop them from sending the traffic
>>>> but pretty soon they'd get the idea that it wasn't going to go
>>>> anywhere.  My concern is say there are 50 Freeswitch instances on a
>>>> box (albeit 8 core, 32GB ram, 8 15K raid 10 storage) and someone
>>>> starts sending thousands of rouge INVITEs to every VM on a physical
>>>> box that the CPU load from just challenging the incoming INVITEs
>>>> would
>>>> create a DoS.  We the logs regularly to try to catch people doing
>>>> this
>>>> sort of thing and drop them at a router upstream of the core  
>>>> network,
>>>> but I'd like to have it happen without human intervention.  Have I
>>>> completely over thought this and am missing something obvious?
>>>>
>>>> Thanks,
>>>> 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
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 saeedahmad1981 at gmail.com  Mon Feb 21 15:49:19 2011
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Mon, 21 Feb 2011 13:49:19 +0100
Subject: [Freeswitch-users] MOH not working in G729 passthru with native
 sound files
In-Reply-To: <AANLkTi=41CbE_tU8LsW52rZCaD0Z+Y8NbE7WXh+bxTn1@mail.gmail.com>
References: <AANLkTi=41CbE_tU8LsW52rZCaD0Z+Y8NbE7WXh+bxTn1@mail.gmail.com>
Message-ID: <AANLkTim2=nnHfXOPjMZCeJ1bmLOLYPSi0zspky9smZeD@mail.gmail.com>

how did you record in G729?

On Thu, Feb 10, 2011 at 3:41 AM, Marcin Wojtowicz <marcin321 at gmail.com>wrote:

> I'm trying to enable MOH when both legs of the call are using G729 (FS is
> in passthru). I converted an edited sample wave file to G729 and put in the
> appropriate folder, and FS loads it correctly because this is the message
> that keeps popping up in the console:
> 2011-02-09 21:10:25.609375 [INFO] mod_native_file.c:94 Opening File
> [C:\freeswitch\sounds/music/8000\sample.G729] 8000hz
>
> I establish a call, and everything is fine, but when I press hold on my
> handset I see an error message that says that G729 is only useable in
> passthru (here is the debug message):
>
> 2011-02-09 21:14:01.828125 [DEBUG] switch_channel.c:1376 (sofia/internal/
> sip:M9jdt73ig0oOJSbt6Uyy at 192.168.1.50:5060) Callstate Change ACTIVE ->
> HELD
> 2011-02-09 21:14:01.828125 [DEBUG] switch_core_session.c:954 Send signal
> sofia/external/MYHOME#@74.63.41.218 [BREAK]
> 2011-02-09 21:14:01.859375 [DEBUG] switch_core_session.c:709 Send signal
> sofia/external/MYHOME#@74.63.41.218 [BREAK]
> 2011-02-09 21:14:02.000000 [DEBUG] switch_ivr.c:563 sofia/external/MYHOME#@
> 74.63.41.218 Command Execute playback(local_stream://moh/8000)
> EXECUTE sofia/external/MYHOME#@74.63.41.218playback(local_stream://moh/8000)
> 2011-02-09 21:14:02.000000 [DEBUG] mod_local_stream.c:421 Opening Stream
> [moh/8000] 8000hz
> 2011-02-09 21:14:02.000000 [DEBUG] switch_ivr_play_say.c:1244 Codec
> Activated L16 at 8000hz 1 channels 20ms
> 2011-02-09 21:14:02.015625 [ERR] mod_g729.c:102 This codec is only usable
> in passthrough mode!
> 2011-02-09 21:14:02.015625 [ERR] switch_core_io.c:1042 Codec G.729 encoder
> error!
> 2011-02-09 21:14:02.015625 [DEBUG] switch_ivr_play_say.c:1581 done playing
> file
>
> I don't understand why that would be, since my music file is in G729 so I'm
> not asking freeswitch to convert, only stream. My custom ringback (before a
> call is established) works just fine using a similar method, so could anyone
> explain me why what I want to do is not permitted?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Mon Feb 21 15:52:30 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 21 Feb 2011 18:22:30 +0530
Subject: [Freeswitch-users] mod_odbc_query
In-Reply-To: <AANLkTinxvwoptnvXmqFEokvQikHNzspTcXvJLndFJLjv@mail.gmail.com>
References: <AANLkTimZbs8S5sgkS7z=6+BDjW3f1U=LmO03203WZYUC@mail.gmail.com>
	<AANLkTinxvwoptnvXmqFEokvQikHNzspTcXvJLndFJLjv@mail.gmail.com>
Message-ID: <AANLkTi=+N4azher3v4u-AxFP_34OXZYiC+osHZhUc7fK@mail.gmail.com>

Do i need to pull down git for it ?

regds
Sam

On Mon, Feb 21, 2011 at 3:47 PM, Avi Marcus <avi at avimarcus.net> wrote:

> You can find the download / compile instructions on the wiki:
> http://wiki.freeswitch.org/wiki/Mod_odbc_query
>
> <http://wiki.freeswitch.org/wiki/Mod_odbc_query>-Avi
>
> On Mon, Feb 21, 2011 at 6:30 AM, Sam <u2nsam at gmail.com> wrote:
>
>> Hello,
>>
>> How to download mod_odbc_query source, there is not such source in 1.0.6
>> or 1.0.7 .
>>
>> 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 avi at avimarcus.net  Mon Feb 21 16:06:35 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 21 Feb 2011 15:06:35 +0200
Subject: [Freeswitch-users] mod_odbc_query
In-Reply-To: <AANLkTi=+N4azher3v4u-AxFP_34OXZYiC+osHZhUc7fK@mail.gmail.com>
References: <AANLkTimZbs8S5sgkS7z=6+BDjW3f1U=LmO03203WZYUC@mail.gmail.com>
	<AANLkTinxvwoptnvXmqFEokvQikHNzspTcXvJLndFJLjv@mail.gmail.com>
	<AANLkTi=+N4azher3v4u-AxFP_34OXZYiC+osHZhUc7fK@mail.gmail.com>
Message-ID: <AANLkTimcO+i0qSJZH4CB3vnEM3mt0noi5JB0Ph0pbKH9@mail.gmail.com>

Like the wiki says, you need to pull the git CONTRIB. There's a link in that
wiki page..
-Avi

On Mon, Feb 21, 2011 at 2:52 PM, Sam <u2nsam at gmail.com> wrote:

> Do i need to pull down git for it ?
>
> regds
> Sam
>
>
> On Mon, Feb 21, 2011 at 3:47 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> You can find the download / compile instructions on the wiki:
>> http://wiki.freeswitch.org/wiki/Mod_odbc_query
>>
>> <http://wiki.freeswitch.org/wiki/Mod_odbc_query>-Avi
>>
>> On Mon, Feb 21, 2011 at 6:30 AM, Sam <u2nsam at gmail.com> wrote:
>>
>>> Hello,
>>>
>>> How to download mod_odbc_query source, there is not such source in 1.0.6
>>> or 1.0.7 .
>>>
>>> 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
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 21 17:10:03 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 21 Feb 2011 06:10:03 -0800 (PST)
Subject: [Freeswitch-users]
	=?utf-8?q?Newbie=C2=B4s_question_about_FreeSwi?=
	=?utf-8?b?dGNoLi4u?=
In-Reply-To: <AANLkTikE=yjS=ku914rpO5+MoJzWikbw0md+dTMDiM07@mail.gmail.com>
References: <AANLkTinf=z4JEp4G2xK_0n+u8v0OhR++GBvPzzbT1Y_-@mail.gmail.com>
	<1298129542465-6043436.post@n2.nabble.com>
	<AANLkTinD08hCZ74SncUbR318agQnO1=JB_Oebc17EPn6@mail.gmail.com>
	<1298169268414-6044831.post@n2.nabble.com>
	<AANLkTikE=yjS=ku914rpO5+MoJzWikbw0md+dTMDiM07@mail.gmail.com>
Message-ID: <1298297403437-6048821.post@n2.nabble.com>



jesse zhao wrote:
> btw, the current openwrt trunk has FS 1.0.6,  I downloaded 1.0.78 source
> code and made symbol link to the new src from feeds/packages.  but openwrt
> still built old 1.0.6. is there a way to enforce openwrt build system pick
> up the new code base?
For OpenWRT, you are better off heading to OpenWRT  http://forum.openwrt.org
Forum  or #openwrt (IRC) to ask such a question.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Newbie-s-question-about-FreeSwitch-tp6038911p6048821.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From fdelawarde at wirelessmundi.com  Mon Feb 21 17:19:13 2011
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Mon, 21 Feb 2011 15:19:13 +0100
Subject: [Freeswitch-users] blind or attended transfer detection
Message-ID: <1298297953.12648.167.camel@luna.tc.commsmundi.com>

Hi,

Is there an easy way to detect if a channel is being or has just been
transfered (blind or attended)?

The objective would be to make loopback channels bowout on transfer
(adding "loopback_bowout_on_transfer" variable), for mod_callcenter to
"release" a loopback agent (mark it available for new calls) if it
transfers the call to another destination.

It might also be useful to have some type of "on_transfer" hook for
other purposes (log transfer attempts, authorize/deny transfer, ...).

Thanks,
Fran?ois.



From david.villasmil.work at gmail.com  Mon Feb 21 17:47:07 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 21 Feb 2011 15:47:07 +0100
Subject: [Freeswitch-users] XML_CURL & gateways
Message-ID: <AANLkTikJ+BEq6BW3MqRnpUuD9bz=hnvf-m_=ytFBsOVP@mail.gmail.com>

Hello All,

I'm trying to have my gateways defined dynamically with xml_curl.
According to the logs, the xml is requested and relivered to fs but fs
doesn't seem to load it as i'm getting "invalid gateway" when i try to
use them.

this is my http://curl-gateway/gateways.php file:



<document type="freeswitch/xml">
     <configuration name="sofia.conf" description="Gateways configuration">

          <profile name="external">
                 <!--
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
                 <!-- This profile is only for outbound registrations
to providers -->
                 <gateways>

                    <gateway name="gw_1">
                         <param name="realm" value="1.2.3.4:5060"/>
                         <param name="username" value="foo"/>
                         <param name="password" value="1234"/>
                         <param name="register" value="false"/>
                         <param name="retry-seconds" value="30"/>
                    </gateway>

                    <gateway name="gw_2">
                         <param name="realm" value="3.4.5.6:5060"/>
                         <param name="username" value="foo"/>
                         <param name="password" value="pass"/>
                         <param name="register" value="false"/>
                         <param name="retry-seconds" value="30"/>
                    </gateway>

                 </gateways>

          </profile>

     </configuration>
</document>


and i get:


freeswitch at internal> sofia xmlstatus gateway gw_1
Invalid Gateway!


Is my file correct?

It is not possible to find xml_curl gateways samples anywhere.



Thanks

David


From steveayre at gmail.com  Mon Feb 21 17:55:57 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 21 Feb 2011 14:55:57 +0000
Subject: [Freeswitch-users] MOH not working in G729 passthru with native
 sound files
In-Reply-To: <AANLkTim2=nnHfXOPjMZCeJ1bmLOLYPSi0zspky9smZeD@mail.gmail.com>
References: <AANLkTi=41CbE_tU8LsW52rZCaD0Z+Y8NbE7WXh+bxTn1@mail.gmail.com>
	<AANLkTim2=nnHfXOPjMZCeJ1bmLOLYPSi0zspky9smZeD@mail.gmail.com>
Message-ID: <AANLkTimaCpu6+SEOppgXpUzzBVoaFm9+70=PyXbEKjG4@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Mod_native_file#Script_to_convert_a_sound_file_to_specific_formats_to_avoid_transcoding



On 21 February 2011 12:49, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:

> how did you record in G729?
>
> On Thu, Feb 10, 2011 at 3:41 AM, Marcin Wojtowicz <marcin321 at gmail.com>wrote:
>
>> I'm trying to enable MOH when both legs of the call are using G729 (FS is
>> in passthru). I converted an edited sample wave file to G729 and put in the
>> appropriate folder, and FS loads it correctly because this is the message
>> that keeps popping up in the console:
>> 2011-02-09 21:10:25.609375 [INFO] mod_native_file.c:94 Opening File
>> [C:\freeswitch\sounds/music/8000\sample.G729] 8000hz
>>
>> I establish a call, and everything is fine, but when I press hold on my
>> handset I see an error message that says that G729 is only useable in
>> passthru (here is the debug message):
>>
>> 2011-02-09 21:14:01.828125 [DEBUG] switch_channel.c:1376 (sofia/internal/
>> sip:M9jdt73ig0oOJSbt6Uyy at 192.168.1.50:5060) Callstate Change ACTIVE ->
>> HELD
>> 2011-02-09 21:14:01.828125 [DEBUG] switch_core_session.c:954 Send signal
>> sofia/external/MYHOME#@74.63.41.218 [BREAK]
>> 2011-02-09 21:14:01.859375 [DEBUG] switch_core_session.c:709 Send signal
>> sofia/external/MYHOME#@74.63.41.218 [BREAK]
>> 2011-02-09 21:14:02.000000 [DEBUG] switch_ivr.c:563
>> sofia/external/MYHOME#@74.63.41.218 Command Execute
>> playback(local_stream://moh/8000)
>> EXECUTE sofia/external/MYHOME#@74.63.41.218playback(local_stream://moh/8000)
>> 2011-02-09 21:14:02.000000 [DEBUG] mod_local_stream.c:421 Opening Stream
>> [moh/8000] 8000hz
>> 2011-02-09 21:14:02.000000 [DEBUG] switch_ivr_play_say.c:1244 Codec
>> Activated L16 at 8000hz 1 channels 20ms
>> 2011-02-09 21:14:02.015625 [ERR] mod_g729.c:102 This codec is only usable
>> in passthrough mode!
>> 2011-02-09 21:14:02.015625 [ERR] switch_core_io.c:1042 Codec G.729 encoder
>> error!
>> 2011-02-09 21:14:02.015625 [DEBUG] switch_ivr_play_say.c:1581 done playing
>> file
>>
>> I don't understand why that would be, since my music file is in G729 so
>> I'm not asking freeswitch to convert, only stream. My custom ringback
>> (before a call is established) works just fine using a similar method, so
>> could anyone explain me why what I want to do is not permitted?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Mon Feb 21 17:56:50 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 21 Feb 2011 15:56:50 +0100
Subject: [Freeswitch-users] xml_curl and gateways
In-Reply-To: <48D58B11.3070704@gmx.net>
References: <48D17932.4000302@gmx.net> <48D26646.5030407@freeswitch.org>
	<48D58B11.3070704@gmx.net>
Message-ID: <AANLkTik-4jXeoCAwBiL7UkoppPw2M01VhHMb5UaTOZAa@mail.gmail.com>

Hello,

care to paste a sample? i can't seem to find any... I'm trying to load
only the <gateways>, i'm using it like this:

<document type="freeswitch/xml">
     <configuration name="sofia.conf" description="Gateways configuration">

          <profile name="external">

                 <gateways>

                    <gateway name="asterisk_1">
                         <param name="realm" value="1.2.3.4:5060"/>
                         <param name="username" value="foo"/>
                         <param name="password" value="1234"/>
                         <param name="register" value="false"/>
                         <param name="retry-seconds" value="30"/>
                    </gateway>

                    <gateway name="opensips_1">
                         <param name="realm" value="5.6.7.8:5060"/>
                         <param name="username" value="foo"/>
                         <param name="password" value="pass"/>
                         <param name="register" value="false"/>
                         <param name="retry-seconds" value="30"/>
                    </gateway>

                 </gateways>

          </profile>

     </configuration>
</document>


FS dumps the files correctly when i set debug on on xmo_curl... but it
doesn't load the gateways...

Thanks


David

On Sun, Sep 21, 2008 at 1:45 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> I now serve the whole sofia config from my Ruby on Rails app and the
> gateways now work dynamically from my database entries.
>
> Thanks to all who contributed. I updated the wiki with my experience on
> the issues I had on this topic.
>
> Freeswitch is so great!
>
> Best regards
> Peter
>
> Raymond Chandler schrieb:
>> Peter P GMX wrote:
>>
>>> According to http://wiki.freeswitch.org/wiki/Mod_xml_curl I'v tried to
>>> set up gateways dynamically, but I do net get it to work:
>>> I always get : "Invalid profile"
>>>
>>> My assumptions for a right xml answer back to FS are as follows, but I
>>> think at least one of it is false:
>>> 1. ) I start with
>>> <document type='freeswitch/xml'>
>>> ? <section name='configuration'>
>>> ? ? <configuration name='sofia.conf' description='RailsAdditionalConf'>
>>> 2.) I do not want to change settings, so I do NOT add a <settings> tag
>>> 3.) I want to enhance an existing profile, so I add a profile tag
>>> (<profile name='external'>)
>>> 4.) I assume that I can have an existing profile "external" in the xml
>>> conf files and can mix additional parts dynamically into the same profile
>>> 5.) Within the profile I want to add gateways, so I add a gateway tag
>>> (<gateways>)
>>> 6.) Then I add the gateways
>>> 7.) I add the closing tags
>>>
>>> Which one of the assumptions if wrong?
>>>
>>>
>>>
>> I think #2 and #3 are wrong, if I understand what you're saying.
>>
>> Your response will override the static XML files, NOT add to them or
>> "enhance" them
>>
>>> The request I receive is as follows:
>>> {"key_name"=>"name", "action"=>"directory", "tag_name"=>"configuration",
>>> "key_value"=>"sofia.conf", "controller"=>"xml_curls",
>>> "hostname"=>"freeswitch", "section"=>"configuration"}
>>>
>>> I only answer once at startup with the follwing:
>>> <document type='freeswitch/xml'>
>>> ? <section name='configuration'>
>>> ? ? <configuration name='sofia.conf' description='RailsAdditionalConf'>
>>> ? ? ? ? <profile name='external'>
>>> ? ? ? ? ? <gateways>
>>> ? <gateway name="freeswitch2">
>>> ? <param name="username" value="1007"/>
>>> ? <param name="realm" value="192.168.0.56:5061"/>
>>> ? <param name="password" value="xxxx"/>
>>> ? <param name="proxy" value="192.168.0.56"/>
>>> ? <param name="expire-seconds" value="30"/>
>>> ? <param name="register" value="true"/>
>>> ? <param name="register-transport" value="tls"/>
>>> ? <param name="retry-seconds" value="30"/>
>>> ? </gateway>
>>> ? ? ? ? ? </gateways>
>>> ? ? ? ? </profile>
>>> ? ? </configuration>
>>> ? </section>
>>> </document>
>>>
>>> Does anybody have a correct XML answer for a gateway and can post it?
>>>
>>>
>>
>> I wrote quite a bit of stuff that does most of the xml_curl files.
>> sofia.conf was definitely on my list. you can find the stuff that I
>> wrote in ${SVNROOT}/scripts/contrib/intralanman/PHP/fs_curl/
>>
>> yes, it's in PHP, but it should give you enough of an idea of how to
>> send your responses.
>>
>>
>> -Ray
>>
>>
>>
>>
>>> Best regards
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 rajesh.npnr at yahoo.com  Mon Feb 21 18:00:57 2011
From: rajesh.npnr at yahoo.com (rex.alex)
Date: Mon, 21 Feb 2011 07:00:57 -0800 (PST)
Subject: [Freeswitch-users] The call dropped prematurely error on txfax
In-Reply-To: <AANLkTinTX+vJj8qibEVsfyrSMVK=jPHbYksPkD20O=Z_@mail.gmail.com>
References: <1297885773534-6033268.post@n2.nabble.com>
	<AANLkTiku+q9B_FeSErvMuXSog6DQ4mMtDmxn_TjtG9Zw@mail.gmail.com>
	<1297954260038-6036197.post@n2.nabble.com>
	<140D740BF92640F7A514BD35BD4BD1AA@e1705>
	<1297958857213-6036503.post@n2.nabble.com>
	<C0AB523AE4774FA5837E88F48243F970@e1705>
	<1297961826262-6036700.post@n2.nabble.com>
	<AANLkTikmEvZB5tF6qiT2Jf4Ca6VzBB8bqQ_k=O59=eO=@mail.gmail.com>
	<1298030300075-6039655.post@n2.nabble.com>
	<AANLkTinTX+vJj8qibEVsfyrSMVK=jPHbYksPkD20O=Z_@mail.gmail.com>
Message-ID: <1298300457397-6048943.post@n2.nabble.com>


Hi,

I did the same but got "Reinvite Codec Error!" with result text "Timed out
waiting for initial communication".

Please see the pastebin in http://pastebin.freeswitch.org/15435

Thanks,
Rex
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/The-call-dropped-prematurely-error-on-txfax-tp6032194p6048943.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Mon Feb 21 18:09:26 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 21 Feb 2011 15:09:26 +0000
Subject: [Freeswitch-users] XML_CURL & gateways
In-Reply-To: <AANLkTikJ+BEq6BW3MqRnpUuD9bz=hnvf-m_=ytFBsOVP@mail.gmail.com>
References: <AANLkTikJ+BEq6BW3MqRnpUuD9bz=hnvf-m_=ytFBsOVP@mail.gmail.com>
Message-ID: <AANLkTin6DEWodK=Am2Gj=eGaL5jac4Y-c4Y9-E+nekY5@mail.gmail.com>

"sofia status" will show you which profiles & gateways are defined.

Are there any errors in FreeSWITCH's logfile when the XML is loaded? Perhaps
there's an XML parsing error, or some other error which gets logged.

Gateways are only loaded when you load the configuration (when mod_sofia
loads, or on sofia profile name rescan). Existing gateways must be deleted
and then rescanned to pick up configuration changes. Did you do that after
making the changes?

The gateway XML looks ok and I can confirm you can serve the config via
mod_xml_curl (I do). Your profile looks incomplete though, for instance
there's no sip-ip to bind to. Did you remove the profile params when posting
to the list, or is that your entire profile?

Regards,
-Steve


On 21 February 2011 14:47, David Villasmil
<david.villasmil.work at gmail.com>wrote:

> Hello All,
>
> I'm trying to have my gateways defined dynamically with xml_curl.
> According to the logs, the xml is requested and relivered to fs but fs
> doesn't seem to load it as i'm getting "invalid gateway" when i try to
> use them.
>
> this is my http://curl-gateway/gateways.php file:
>
>
>
> <document type="freeswitch/xml">
>     <configuration name="sofia.conf" description="Gateways configuration">
>
>          <profile name="external">
>                 <!--
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
>                 <!-- This profile is only for outbound registrations
> to providers -->
>                 <gateways>
>
>                    <gateway name="gw_1">
>                         <param name="realm" value="1.2.3.4:5060"/>
>                         <param name="username" value="foo"/>
>                         <param name="password" value="1234"/>
>                         <param name="register" value="false"/>
>                         <param name="retry-seconds" value="30"/>
>                    </gateway>
>
>                    <gateway name="gw_2">
>                         <param name="realm" value="3.4.5.6:5060"/>
>                         <param name="username" value="foo"/>
>                         <param name="password" value="pass"/>
>                         <param name="register" value="false"/>
>                         <param name="retry-seconds" value="30"/>
>                    </gateway>
>
>                 </gateways>
>
>          </profile>
>
>     </configuration>
> </document>
>
>
> and i get:
>
>
> freeswitch at internal> sofia xmlstatus gateway gw_1
> Invalid Gateway!
>
>
> Is my file correct?
>
> It is not possible to find xml_curl gateways samples anywhere.
>
>
>
> 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 david.villasmil.work at gmail.com  Mon Feb 21 18:40:01 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 21 Feb 2011 16:40:01 +0100
Subject: [Freeswitch-users] XML_CURL & gateways
In-Reply-To: <AANLkTin6DEWodK=Am2Gj=eGaL5jac4Y-c4Y9-E+nekY5@mail.gmail.com>
References: <AANLkTikJ+BEq6BW3MqRnpUuD9bz=hnvf-m_=ytFBsOVP@mail.gmail.com>
	<AANLkTin6DEWodK=Am2Gj=eGaL5jac4Y-c4Y9-E+nekY5@mail.gmail.com>
Message-ID: <AANLkTinCMf4+OW52CZ1dj-hyLyLk3Fh2dmYC2q9vgJ=2@mail.gmail.com>

Thanks Steven,

The sofia profile loads statically. Only the gateway part is passed
via curl, as I only want to load that dynamically.

I see no errors anywhere:

 sofia profile external rescan
+OK scan complete

freeswitch at internal>

HOw do i remove all gateways before rescanning them?

Thanks

David

On Mon, Feb 21, 2011 at 4:09 PM, Steven Ayre <steveayre at gmail.com> wrote:
> "sofia status" will show you which profiles & gateways are defined.
>
> Are there any errors in FreeSWITCH's logfile when the XML is loaded? Perhaps
> there's an XML parsing error, or some other error which gets logged.
>
> Gateways are only loaded when you load the configuration (when mod_sofia
> loads, or on sofia profile name rescan). Existing gateways must be deleted
> and then rescanned to pick up configuration changes. Did you do that after
> making the changes?
>
> The gateway XML looks ok and I can confirm you can serve the config via
> mod_xml_curl (I do). Your profile looks incomplete though, for instance
> there's no sip-ip to bind to. Did you remove the profile params when posting
> to the list, or is that your entire profile?
>
> Regards,
> -Steve
>
>
> On 21 February 2011 14:47, David Villasmil <david.villasmil.work at gmail.com>
> wrote:
>>
>> Hello All,
>>
>> I'm trying to have my gateways defined dynamically with xml_curl.
>> According to the logs, the xml is requested and relivered to fs but fs
>> doesn't seem to load it as i'm getting "invalid gateway" when i try to
>> use them.
>>
>> this is my http://curl-gateway/gateways.php file:
>>
>>
>>
>> <document type="freeswitch/xml">
>> ? ? <configuration name="sofia.conf" description="Gateways configuration">
>>
>> ? ? ? ? ?<profile name="external">
>> ? ? ? ? ? ? ? ? <!--
>> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
>> ? ? ? ? ? ? ? ? <!-- This profile is only for outbound registrations
>> to providers -->
>> ? ? ? ? ? ? ? ? <gateways>
>>
>> ? ? ? ? ? ? ? ? ? ?<gateway name="gw_1">
>> ? ? ? ? ? ? ? ? ? ? ? ? <param name="realm" value="1.2.3.4:5060"/>
>> ? ? ? ? ? ? ? ? ? ? ? ? <param name="username" value="foo"/>
>> ? ? ? ? ? ? ? ? ? ? ? ? <param name="password" value="1234"/>
>> ? ? ? ? ? ? ? ? ? ? ? ? <param name="register" value="false"/>
>> ? ? ? ? ? ? ? ? ? ? ? ? <param name="retry-seconds" value="30"/>
>> ? ? ? ? ? ? ? ? ? ?</gateway>
>>
>> ? ? ? ? ? ? ? ? ? ?<gateway name="gw_2">
>> ? ? ? ? ? ? ? ? ? ? ? ? <param name="realm" value="3.4.5.6:5060"/>
>> ? ? ? ? ? ? ? ? ? ? ? ? <param name="username" value="foo"/>
>> ? ? ? ? ? ? ? ? ? ? ? ? <param name="password" value="pass"/>
>> ? ? ? ? ? ? ? ? ? ? ? ? <param name="register" value="false"/>
>> ? ? ? ? ? ? ? ? ? ? ? ? <param name="retry-seconds" value="30"/>
>> ? ? ? ? ? ? ? ? ? ?</gateway>
>>
>> ? ? ? ? ? ? ? ? </gateways>
>>
>> ? ? ? ? ?</profile>
>>
>> ? ? </configuration>
>> </document>
>>
>>
>> and i get:
>>
>>
>> freeswitch at internal> sofia xmlstatus gateway gw_1
>> Invalid Gateway!
>>
>>
>> Is my file correct?
>>
>> It is not possible to find xml_curl gateways samples anywhere.
>>
>>
>>
>> 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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Mon Feb 21 18:43:42 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 21 Feb 2011 09:43:42 -0600
Subject: [Freeswitch-users] xml_curl and gateways
In-Reply-To: <AANLkTik-4jXeoCAwBiL7UkoppPw2M01VhHMb5UaTOZAa@mail.gmail.com>
References: <48D17932.4000302@gmx.net> <48D26646.5030407@freeswitch.org>
	<48D58B11.3070704@gmx.net>
	<AANLkTik-4jXeoCAwBiL7UkoppPw2M01VhHMb5UaTOZAa@mail.gmail.com>
Message-ID: <C5AA9686-A623-4CE5-B562-5F53E81AC2AE@freeswitch.org>

I'm currently not aware of any magical xml fairy that can fill out the rest of the required config for you... so if you would see the default external profile and provide the proper settings for things like rtp-ip and sip-ip... ;)  Cuz you're missing a lot of options here so I'm guessing that its not even starting the profile due to that.

/b

On Feb 21, 2011, at 8:56 AM, David Villasmil wrote:

> <document type="freeswitch/xml">
>     <configuration name="sofia.conf" description="Gateways configuration">
> 
>          <profile name="external">
> 
>                 <gateways>
> 
>                    <gateway name="asterisk_1">
>                         <param name="realm" value="1.2.3.4:5060"/>
>                         <param name="username" value="foo"/>
>                         <param name="password" value="1234"/>
>                         <param name="register" value="false"/>
>                         <param name="retry-seconds" value="30"/>
>                    </gateway>
> 
>                    <gateway name="opensips_1">
>                         <param name="realm" value="5.6.7.8:5060"/>
>                         <param name="username" value="foo"/>
>                         <param name="password" value="pass"/>
>                         <param name="register" value="false"/>
>                         <param name="retry-seconds" value="30"/>
>                    </gateway>
> 
>                 </gateways>
> 
>          </profile>
> 
>     </configuration>
> </document>



From brian at freeswitch.org  Mon Feb 21 18:44:39 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 21 Feb 2011 09:44:39 -0600
Subject: [Freeswitch-users] XML_CURL & gateways
In-Reply-To: <AANLkTinCMf4+OW52CZ1dj-hyLyLk3Fh2dmYC2q9vgJ=2@mail.gmail.com>
References: <AANLkTikJ+BEq6BW3MqRnpUuD9bz=hnvf-m_=ytFBsOVP@mail.gmail.com>
	<AANLkTin6DEWodK=Am2Gj=eGaL5jac4Y-c4Y9-E+nekY5@mail.gmail.com>
	<AANLkTinCMf4+OW52CZ1dj-hyLyLk3Fh2dmYC2q9vgJ=2@mail.gmail.com>
Message-ID: <A87BE301-1FF1-4C2B-B996-E5AEEE826031@freeswitch.org>

sofia profile xxx killgw xxx

/b

On Feb 21, 2011, at 9:40 AM, David Villasmil wrote:

> Thanks Steven,
> 
> The sofia profile loads statically. Only the gateway part is passed
> via curl, as I only want to load that dynamically.
> 
> I see no errors anywhere:
> 
> sofia profile external rescan
> +OK scan complete
> 
> freeswitch at internal>
> 
> HOw do i remove all gateways before rescanning them?
> 
> Thanks
> 
> David



From david.villasmil.work at gmail.com  Mon Feb 21 18:50:22 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 21 Feb 2011 16:50:22 +0100
Subject: [Freeswitch-users] xml_curl and gateways
In-Reply-To: <C5AA9686-A623-4CE5-B562-5F53E81AC2AE@freeswitch.org>
References: <48D17932.4000302@gmx.net> <48D26646.5030407@freeswitch.org>
	<48D58B11.3070704@gmx.net>
	<AANLkTik-4jXeoCAwBiL7UkoppPw2M01VhHMb5UaTOZAa@mail.gmail.com>
	<C5AA9686-A623-4CE5-B562-5F53E81AC2AE@freeswitch.org>
Message-ID: <AANLkTi=85E16XR-B4GRgE2F6ikP40KPAAWgLjrvY05f0@mail.gmail.com>

Hello Brian,

It seems i didn't explain myself correctly, all sofia.xml is in xml
files, except the <gateways> part. That's the only section i want
loaded dynamically.

os it is basically like this:

FIles:

sofia.conf.xml

<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>


--------------------------------------------------------------------------------

and the in sip_profiles i only have external.conf.xml:

--------------------------------------------------------------------------------

<profile name="external">
  <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
  <!-- This profile is only for outbound registrations to providers -->

  <aliases>
    <!--
    <alias name="outbound"/>
    <alias name="nat"/>
    -->
  </aliases>

  <domains>
    <domain name="all" alias="false" parse="true"/>
  </domains>

  <settings>
    <param name="debug" value="0"/>
	<!-- If you want FreeSWITCH to shutdown if this profile fails to
load, uncomment the next line. -->
	<!-- <param name="shutdown-on-fail" value="true"/> -->
    <param name="sip-trace" value="no"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="sip-port" value="$${external_sip_port}"/>
<!--    <param name="sip-port" value="5061"/>                         -->
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="dtmf-duration" value="2000"/>
    <param name="inbound-codec-prefs" value="$${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="enable-100rel" value="true"/>-->
    <!-- This could be set to "passive" -->
    <param name="local-network-acl" value="localnet.auto"/>
    <param name="manage-presence" value="false"/>

    <!-- used to share presence info across sofia profiles
	 manage-presence needs to be set to passive on this profile
	 if you want it to behave as if it were the internal profile
	 for presence.
    -->
    <!-- Name of the db to use for this profile -->
    <!--<param name="dbname" value="share_presence"/>-->
    <!--<param name="presence-hosts" value="$${domain}"/>-->
    <!--<param name="force-register-domain" value="$${domain}"/>-->
    <!--all inbound reg will stored in the db using this domain -->
    <!--<param name="force-register-db-domain" value="$${domain}"/>-->
    <!-- ************************************************* -->

    <!--<param name="aggressive-nat-detection" value="true"/>-->
    <param name="inbound-codec-negotiation" value="generous"/>
    <param name="nonce-ttl" value="60"/>
    <param name="auth-calls" value="false"/>
    <!--
	DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
    -->
    <param name="rtp-ip" value="my_ip"/>
    <param name="sip-ip" value="my_ip"/>
    <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="enable-3pcc" value="true"/>-->

    <!-- TLS: disabled by default, set to "true" to enable -->
    <param name="tls" value="$${external_ssl_enable}"/>
    <!-- additional bind parameters for TLS -->
    <param name="tls-bind-params" value="transport=tls"/>
    <!-- Port to listen on for TLS requests. (5081 will be used if
unspecified) -->
    <param name="tls-sip-port" value="$${external_tls_port}"/>
    <!-- Location of the agent.pem and cafile.pem ssl certificates
(needed for TLS server) -->
    <param name="tls-cert-dir" value="$${external_ssl_dir}"/>
    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may
not work with TLSv1 -->
    <param name="tls-version" value="$${sip_tls_version}"/>

  </settings>
</profile>


--------------------------------------------------------------------------------



And then the gateways.php should be loaded (if i'm correct) from the
xml_curl. At lease i see the requests and responses from the
curl-gateway.

Thanks


David

On Mon, Feb 21, 2011 at 4:43 PM, Brian West <brian at freeswitch.org> wrote:
> I'm currently not aware of any magical xml fairy that can fill out the rest of the required config for you... so if you would see the default external profile and provide the proper settings for things like rtp-ip and sip-ip... ;) ?Cuz you're missing a lot of options here so I'm guessing that its not even starting the profile due to that.
>
> /b
>
> On Feb 21, 2011, at 8:56 AM, David Villasmil wrote:
>
>> <document type="freeswitch/xml">
>> ? ? <configuration name="sofia.conf" description="Gateways configuration">
>>
>> ? ? ? ? ?<profile name="external">
>>
>> ? ? ? ? ? ? ? ? <gateways>
>>
>> ? ? ? ? ? ? ? ? ? ?<gateway name="asterisk_1">
>> ? ? ? ? ? ? ? ? ? ? ? ? <param name="realm" value="1.2.3.4:5060"/>
>> ? ? ? ? ? ? ? ? ? ? ? ? <param name="username" value="foo"/>
>> ? ? ? ? ? ? ? ? ? ? ? ? <param name="password" value="1234"/>
>> ? ? ? ? ? ? ? ? ? ? ? ? <param name="register" value="false"/>
>> ? ? ? ? ? ? ? ? ? ? ? ? <param name="retry-seconds" value="30"/>
>> ? ? ? ? ? ? ? ? ? ?</gateway>
>>
>> ? ? ? ? ? ? ? ? ? ?<gateway name="opensips_1">
>> ? ? ? ? ? ? ? ? ? ? ? ? <param name="realm" value="5.6.7.8:5060"/>
>> ? ? ? ? ? ? ? ? ? ? ? ? <param name="username" value="foo"/>
>> ? ? ? ? ? ? ? ? ? ? ? ? <param name="password" value="pass"/>
>> ? ? ? ? ? ? ? ? ? ? ? ? <param name="register" value="false"/>
>> ? ? ? ? ? ? ? ? ? ? ? ? <param name="retry-seconds" value="30"/>
>> ? ? ? ? ? ? ? ? ? ?</gateway>
>>
>> ? ? ? ? ? ? ? ? </gateways>
>>
>> ? ? ? ? ?</profile>
>>
>> ? ? </configuration>
>> </document>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb 21 18:52:19 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 21 Feb 2011 15:52:19 +0000
Subject: [Freeswitch-users] XML_CURL & gateways
In-Reply-To: <AANLkTinCMf4+OW52CZ1dj-hyLyLk3Fh2dmYC2q9vgJ=2@mail.gmail.com>
References: <AANLkTikJ+BEq6BW3MqRnpUuD9bz=hnvf-m_=ytFBsOVP@mail.gmail.com>
	<AANLkTin6DEWodK=Am2Gj=eGaL5jac4Y-c4Y9-E+nekY5@mail.gmail.com>
	<AANLkTinCMf4+OW52CZ1dj-hyLyLk3Fh2dmYC2q9vgJ=2@mail.gmail.com>
Message-ID: <AANLkTi=ee+HR868-1D8ZMWe1AxB4-ndffEZwEoo=SpnG@mail.gmail.com>

As Brian said, "sofia profile xxx killgw xxx" - you have to do them one at a
time.

If you load the profile statically at start and rescan later from
mod_xml_curl, it should add any missing gateways at that time.

-Steve


On 21 February 2011 15:40, David Villasmil
<david.villasmil.work at gmail.com>wrote:

> Thanks Steven,
>
> The sofia profile loads statically. Only the gateway part is passed
> via curl, as I only want to load that dynamically.
>
> I see no errors anywhere:
>
>  sofia profile external rescan
> +OK scan complete
>
> freeswitch at internal>
>
> HOw do i remove all gateways before rescanning them?
>
> Thanks
>
> David
>
> On Mon, Feb 21, 2011 at 4:09 PM, Steven Ayre <steveayre at gmail.com> wrote:
> > "sofia status" will show you which profiles & gateways are defined.
> >
> > Are there any errors in FreeSWITCH's logfile when the XML is loaded?
> Perhaps
> > there's an XML parsing error, or some other error which gets logged.
> >
> > Gateways are only loaded when you load the configuration (when mod_sofia
> > loads, or on sofia profile name rescan). Existing gateways must be
> deleted
> > and then rescanned to pick up configuration changes. Did you do that
> after
> > making the changes?
> >
> > The gateway XML looks ok and I can confirm you can serve the config via
> > mod_xml_curl (I do). Your profile looks incomplete though, for instance
> > there's no sip-ip to bind to. Did you remove the profile params when
> posting
> > to the list, or is that your entire profile?
> >
> > Regards,
> > -Steve
> >
> >
> > On 21 February 2011 14:47, David Villasmil <
> david.villasmil.work at gmail.com>
> > wrote:
> >>
> >> Hello All,
> >>
> >> I'm trying to have my gateways defined dynamically with xml_curl.
> >> According to the logs, the xml is requested and relivered to fs but fs
> >> doesn't seem to load it as i'm getting "invalid gateway" when i try to
> >> use them.
> >>
> >> this is my http://curl-gateway/gateways.php file:
> >>
> >>
> >>
> >> <document type="freeswitch/xml">
> >>     <configuration name="sofia.conf" description="Gateways
> configuration">
> >>
> >>          <profile name="external">
> >>                 <!--
> >> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
> >>                 <!-- This profile is only for outbound registrations
> >> to providers -->
> >>                 <gateways>
> >>
> >>                    <gateway name="gw_1">
> >>                         <param name="realm" value="1.2.3.4:5060"/>
> >>                         <param name="username" value="foo"/>
> >>                         <param name="password" value="1234"/>
> >>                         <param name="register" value="false"/>
> >>                         <param name="retry-seconds" value="30"/>
> >>                    </gateway>
> >>
> >>                    <gateway name="gw_2">
> >>                         <param name="realm" value="3.4.5.6:5060"/>
> >>                         <param name="username" value="foo"/>
> >>                         <param name="password" value="pass"/>
> >>                         <param name="register" value="false"/>
> >>                         <param name="retry-seconds" value="30"/>
> >>                    </gateway>
> >>
> >>                 </gateways>
> >>
> >>          </profile>
> >>
> >>     </configuration>
> >> </document>
> >>
> >>
> >> and i get:
> >>
> >>
> >> freeswitch at internal> sofia xmlstatus gateway gw_1
> >> Invalid Gateway!
> >>
> >>
> >> Is my file correct?
> >>
> >> It is not possible to find xml_curl gateways samples anywhere.
> >>
> >>
> >>
> >> 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
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Mon Feb 21 18:54:42 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 21 Feb 2011 09:54:42 -0600
Subject: [Freeswitch-users] The call dropped prematurely error on txfax
In-Reply-To: <1298300457397-6048943.post@n2.nabble.com>
References: <1297885773534-6033268.post@n2.nabble.com>
	<AANLkTiku+q9B_FeSErvMuXSog6DQ4mMtDmxn_TjtG9Zw@mail.gmail.com>
	<1297954260038-6036197.post@n2.nabble.com>
	<140D740BF92640F7A514BD35BD4BD1AA@e1705>
	<1297958857213-6036503.post@n2.nabble.com>
	<C0AB523AE4774FA5837E88F48243F970@e1705>
	<1297961826262-6036700.post@n2.nabble.com>
	<AANLkTikmEvZB5tF6qiT2Jf4Ca6VzBB8bqQ_k=O59=eO=@mail.gmail.com>
	<1298030300075-6039655.post@n2.nabble.com>
	<AANLkTinTX+vJj8qibEVsfyrSMVK=jPHbYksPkD20O=Z_@mail.gmail.com>
	<1298300457397-6048943.post@n2.nabble.com>
Message-ID: <AANLkTim4Vsy3F-mBRnFuHK4oPvpQyzPB6CLy9akjxREG@mail.gmail.com>

There is no m line in this sdp
m=image is missing.
FreeSWITCH is refusing the re-invite because its not a valid SDP.


   v=0
   o=Sonus_UAC 25105 9552 IN IP4 X.X.X.X
   s=SIP Media Capabilities
   c=IN IP4 X.X.X.X
   t=0 0
   a=T38FaxVersion:0
   a=T38MaxBitRate:14400
   a=T38FaxRateManagement:transferredTCF
   a=T38FaxMaxBuffer:262
   a=T38FaxMaxDatagram:90
   a=T38FaxUdpEC:t38UDPRedundancy
   a=sendrecv


On Mon, Feb 21, 2011 at 9:00 AM, rex.alex <rajesh.npnr at yahoo.com> wrote:
>
> Hi,
>
> I did the same but got "Reinvite Codec Error!" with result text "Timed out
> waiting for initial communication".
>
> Please see the pastebin in http://pastebin.freeswitch.org/15435
>
> Thanks,
> Rex
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/The-call-dropped-prematurely-error-on-txfax-tp6032194p6048943.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 david.villasmil.work at gmail.com  Mon Feb 21 18:56:06 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 21 Feb 2011 16:56:06 +0100
Subject: [Freeswitch-users] XML_CURL & gateways
In-Reply-To: <AANLkTi=ee+HR868-1D8ZMWe1AxB4-ndffEZwEoo=SpnG@mail.gmail.com>
References: <AANLkTikJ+BEq6BW3MqRnpUuD9bz=hnvf-m_=ytFBsOVP@mail.gmail.com>
	<AANLkTin6DEWodK=Am2Gj=eGaL5jac4Y-c4Y9-E+nekY5@mail.gmail.com>
	<AANLkTinCMf4+OW52CZ1dj-hyLyLk3Fh2dmYC2q9vgJ=2@mail.gmail.com>
	<AANLkTi=ee+HR868-1D8ZMWe1AxB4-ndffEZwEoo=SpnG@mail.gmail.com>
Message-ID: <AANLkTi=NTm+GPrFPuifvbUcnTOA+3MRJErPwfW-N7FU7@mail.gmail.com>

Thanks Steven,

> If you load the profile statically at start and rescan later from
> mod_xml_curl, it should add any missing gateways at that time.
>

Doesn't do it though... that's why i was wondering if I had something
wrong in my gateways section. I just posted my config in thi email
thread before this email. Hope you see something wrong with it.

Thanks

David

On Mon, Feb 21, 2011 at 4:52 PM, Steven Ayre <steveayre at gmail.com> wrote:
> As Brian said, "sofia profile xxx killgw xxx" - you have to do them one at a
> time.
>
> If you load the profile statically at start and rescan later from
> mod_xml_curl, it should add any missing gateways at that time.
>
> -Steve
>
>
> On 21 February 2011 15:40, David Villasmil <david.villasmil.work at gmail.com>
> wrote:
>>
>> Thanks Steven,
>>
>> The sofia profile loads statically. Only the gateway part is passed
>> via curl, as I only want to load that dynamically.
>>
>> I see no errors anywhere:
>>
>> ?sofia profile external rescan
>> +OK scan complete
>>
>> freeswitch at internal>
>>
>> HOw do i remove all gateways before rescanning them?
>>
>> Thanks
>>
>> David
>>
>> On Mon, Feb 21, 2011 at 4:09 PM, Steven Ayre <steveayre at gmail.com> wrote:
>> > "sofia status" will show you which profiles & gateways are defined.
>> >
>> > Are there any errors in FreeSWITCH's logfile when the XML is loaded?
>> > Perhaps
>> > there's an XML parsing error, or some other error which gets logged.
>> >
>> > Gateways are only loaded when you load the configuration (when mod_sofia
>> > loads, or on sofia profile name rescan). Existing gateways must be
>> > deleted
>> > and then rescanned to pick up configuration changes. Did you do that
>> > after
>> > making the changes?
>> >
>> > The gateway XML looks ok and I can confirm you can serve the config via
>> > mod_xml_curl (I do). Your profile looks incomplete though, for instance
>> > there's no sip-ip to bind to. Did you remove the profile params when
>> > posting
>> > to the list, or is that your entire profile?
>> >
>> > Regards,
>> > -Steve
>> >
>> >
>> > On 21 February 2011 14:47, David Villasmil
>> > <david.villasmil.work at gmail.com>
>> > wrote:
>> >>
>> >> Hello All,
>> >>
>> >> I'm trying to have my gateways defined dynamically with xml_curl.
>> >> According to the logs, the xml is requested and relivered to fs but fs
>> >> doesn't seem to load it as i'm getting "invalid gateway" when i try to
>> >> use them.
>> >>
>> >> this is my http://curl-gateway/gateways.php file:
>> >>
>> >>
>> >>
>> >> <document type="freeswitch/xml">
>> >> ? ? <configuration name="sofia.conf" description="Gateways
>> >> configuration">
>> >>
>> >> ? ? ? ? ?<profile name="external">
>> >> ? ? ? ? ? ? ? ? <!--
>> >> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
>> >> ? ? ? ? ? ? ? ? <!-- This profile is only for outbound registrations
>> >> to providers -->
>> >> ? ? ? ? ? ? ? ? <gateways>
>> >>
>> >> ? ? ? ? ? ? ? ? ? ?<gateway name="gw_1">
>> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="realm" value="1.2.3.4:5060"/>
>> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="username" value="foo"/>
>> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="password" value="1234"/>
>> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="register" value="false"/>
>> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="retry-seconds" value="30"/>
>> >> ? ? ? ? ? ? ? ? ? ?</gateway>
>> >>
>> >> ? ? ? ? ? ? ? ? ? ?<gateway name="gw_2">
>> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="realm" value="3.4.5.6:5060"/>
>> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="username" value="foo"/>
>> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="password" value="pass"/>
>> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="register" value="false"/>
>> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="retry-seconds" value="30"/>
>> >> ? ? ? ? ? ? ? ? ? ?</gateway>
>> >>
>> >> ? ? ? ? ? ? ? ? </gateways>
>> >>
>> >> ? ? ? ? ?</profile>
>> >>
>> >> ? ? </configuration>
>> >> </document>
>> >>
>> >>
>> >> and i get:
>> >>
>> >>
>> >> freeswitch at internal> sofia xmlstatus gateway gw_1
>> >> Invalid Gateway!
>> >>
>> >>
>> >> Is my file correct?
>> >>
>> >> It is not possible to find xml_curl gateways samples anywhere.
>> >>
>> >>
>> >>
>> >> 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
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb 21 19:07:45 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 21 Feb 2011 16:07:45 +0000
Subject: [Freeswitch-users] XML_CURL & gateways
In-Reply-To: <AANLkTi=NTm+GPrFPuifvbUcnTOA+3MRJErPwfW-N7FU7@mail.gmail.com>
References: <AANLkTikJ+BEq6BW3MqRnpUuD9bz=hnvf-m_=ytFBsOVP@mail.gmail.com>
	<AANLkTin6DEWodK=Am2Gj=eGaL5jac4Y-c4Y9-E+nekY5@mail.gmail.com>
	<AANLkTinCMf4+OW52CZ1dj-hyLyLk3Fh2dmYC2q9vgJ=2@mail.gmail.com>
	<AANLkTi=ee+HR868-1D8ZMWe1AxB4-ndffEZwEoo=SpnG@mail.gmail.com>
	<AANLkTi=NTm+GPrFPuifvbUcnTOA+3MRJErPwfW-N7FU7@mail.gmail.com>
Message-ID: <AANLkTi=31r4-Lainyq+K4XKyi2o+cQA3DuUY4MoJy+NA@mail.gmail.com>

Try (re)including the entire profile config. FS might be ignoring it because
of the missing profile params.

-Steve


On 21 February 2011 15:56, David Villasmil
<david.villasmil.work at gmail.com>wrote:

> Thanks Steven,
>
> > If you load the profile statically at start and rescan later from
> > mod_xml_curl, it should add any missing gateways at that time.
> >
>
> Doesn't do it though... that's why i was wondering if I had something
> wrong in my gateways section. I just posted my config in thi email
> thread before this email. Hope you see something wrong with it.
>
> Thanks
>
> David
>
> On Mon, Feb 21, 2011 at 4:52 PM, Steven Ayre <steveayre at gmail.com> wrote:
> > As Brian said, "sofia profile xxx killgw xxx" - you have to do them one
> at a
> > time.
> >
> > If you load the profile statically at start and rescan later from
> > mod_xml_curl, it should add any missing gateways at that time.
> >
> > -Steve
> >
> >
> > On 21 February 2011 15:40, David Villasmil <
> david.villasmil.work at gmail.com>
> > wrote:
> >>
> >> Thanks Steven,
> >>
> >> The sofia profile loads statically. Only the gateway part is passed
> >> via curl, as I only want to load that dynamically.
> >>
> >> I see no errors anywhere:
> >>
> >>  sofia profile external rescan
> >> +OK scan complete
> >>
> >> freeswitch at internal>
> >>
> >> HOw do i remove all gateways before rescanning them?
> >>
> >> Thanks
> >>
> >> David
> >>
> >> On Mon, Feb 21, 2011 at 4:09 PM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >> > "sofia status" will show you which profiles & gateways are defined.
> >> >
> >> > Are there any errors in FreeSWITCH's logfile when the XML is loaded?
> >> > Perhaps
> >> > there's an XML parsing error, or some other error which gets logged.
> >> >
> >> > Gateways are only loaded when you load the configuration (when
> mod_sofia
> >> > loads, or on sofia profile name rescan). Existing gateways must be
> >> > deleted
> >> > and then rescanned to pick up configuration changes. Did you do that
> >> > after
> >> > making the changes?
> >> >
> >> > The gateway XML looks ok and I can confirm you can serve the config
> via
> >> > mod_xml_curl (I do). Your profile looks incomplete though, for
> instance
> >> > there's no sip-ip to bind to. Did you remove the profile params when
> >> > posting
> >> > to the list, or is that your entire profile?
> >> >
> >> > Regards,
> >> > -Steve
> >> >
> >> >
> >> > On 21 February 2011 14:47, David Villasmil
> >> > <david.villasmil.work at gmail.com>
> >> > wrote:
> >> >>
> >> >> Hello All,
> >> >>
> >> >> I'm trying to have my gateways defined dynamically with xml_curl.
> >> >> According to the logs, the xml is requested and relivered to fs but
> fs
> >> >> doesn't seem to load it as i'm getting "invalid gateway" when i try
> to
> >> >> use them.
> >> >>
> >> >> this is my http://curl-gateway/gateways.php file:
> >> >>
> >> >>
> >> >>
> >> >> <document type="freeswitch/xml">
> >> >>     <configuration name="sofia.conf" description="Gateways
> >> >> configuration">
> >> >>
> >> >>          <profile name="external">
> >> >>                 <!--
> >> >> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
> >> >>                 <!-- This profile is only for outbound registrations
> >> >> to providers -->
> >> >>                 <gateways>
> >> >>
> >> >>                    <gateway name="gw_1">
> >> >>                         <param name="realm" value="1.2.3.4:5060"/>
> >> >>                         <param name="username" value="foo"/>
> >> >>                         <param name="password" value="1234"/>
> >> >>                         <param name="register" value="false"/>
> >> >>                         <param name="retry-seconds" value="30"/>
> >> >>                    </gateway>
> >> >>
> >> >>                    <gateway name="gw_2">
> >> >>                         <param name="realm" value="3.4.5.6:5060"/>
> >> >>                         <param name="username" value="foo"/>
> >> >>                         <param name="password" value="pass"/>
> >> >>                         <param name="register" value="false"/>
> >> >>                         <param name="retry-seconds" value="30"/>
> >> >>                    </gateway>
> >> >>
> >> >>                 </gateways>
> >> >>
> >> >>          </profile>
> >> >>
> >> >>     </configuration>
> >> >> </document>
> >> >>
> >> >>
> >> >> and i get:
> >> >>
> >> >>
> >> >> freeswitch at internal> sofia xmlstatus gateway gw_1
> >> >> Invalid Gateway!
> >> >>
> >> >>
> >> >> Is my file correct?
> >> >>
> >> >> It is not possible to find xml_curl gateways samples anywhere.
> >> >>
> >> >>
> >> >>
> >> >> 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
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Mon Feb 21 19:19:19 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 21 Feb 2011 17:19:19 +0100
Subject: [Freeswitch-users] XML_CURL & gateways
In-Reply-To: <AANLkTi=31r4-Lainyq+K4XKyi2o+cQA3DuUY4MoJy+NA@mail.gmail.com>
References: <AANLkTikJ+BEq6BW3MqRnpUuD9bz=hnvf-m_=ytFBsOVP@mail.gmail.com>
	<AANLkTin6DEWodK=Am2Gj=eGaL5jac4Y-c4Y9-E+nekY5@mail.gmail.com>
	<AANLkTinCMf4+OW52CZ1dj-hyLyLk3Fh2dmYC2q9vgJ=2@mail.gmail.com>
	<AANLkTi=ee+HR868-1D8ZMWe1AxB4-ndffEZwEoo=SpnG@mail.gmail.com>
	<AANLkTi=NTm+GPrFPuifvbUcnTOA+3MRJErPwfW-N7FU7@mail.gmail.com>
	<AANLkTi=31r4-Lainyq+K4XKyi2o+cQA3DuUY4MoJy+NA@mail.gmail.com>
Message-ID: <AANLkTinKvJk7XmoSLwhqEczcOXNHy1YRP=NwPW3J1ooY@mail.gmail.com>

Yeah, that's what i was trying, but now sofia is not starting...

could it be my xml_curl is wrong? here it is:

trying to load all sofia config:

<configuration name="xml_curl.conf" description="cURL XML Gateway">
  <bindings>
    <binding name="sofia.conf">
      <param name="gateway-url"
value="http://server_ip/fsxml/gateways.php" bindings="sofia.conf"/>
-->
    </binding>
  </bindings>
</configuration>

this: http://server_ip/fsxml/gateways.php returns all the sofia
config, and it is ok because I copied the result into sofia.conf.xml
and it starts prefectly...

is my "bindings" ok? I've tried: sofia.conf.xml, sofia.conf and
sofia... none seems to load it, i always get:

2011-02-21 17:08:58.277368 [ERR] sofia.c:2646 Open of sofia.conf failed
2011-02-21 17:08:58.277757 [CRIT] switch_loadable_module.c:882 Error
Loading module /usr/local/freeswitch/mod/mod_sofia.so
**Module load routine returned an error**

I removed the sofia.conf.xml from the autoload_configs... not sure
whether i should leave it or what...

David


On Mon, Feb 21, 2011 at 5:07 PM, Steven Ayre <steveayre at gmail.com> wrote:
> Try (re)including the entire profile config. FS might be ignoring it because
> of the missing profile params.
>
> -Steve
>
>
> On 21 February 2011 15:56, David Villasmil <david.villasmil.work at gmail.com>
> wrote:
>>
>> Thanks Steven,
>>
>> > If you load the profile statically at start and rescan later from
>> > mod_xml_curl, it should add any missing gateways at that time.
>> >
>>
>> Doesn't do it though... that's why i was wondering if I had something
>> wrong in my gateways section. I just posted my config in thi email
>> thread before this email. Hope you see something wrong with it.
>>
>> Thanks
>>
>> David
>>
>> On Mon, Feb 21, 2011 at 4:52 PM, Steven Ayre <steveayre at gmail.com> wrote:
>> > As Brian said, "sofia profile xxx killgw xxx" - you have to do them one
>> > at a
>> > time.
>> >
>> > If you load the profile statically at start and rescan later from
>> > mod_xml_curl, it should add any missing gateways at that time.
>> >
>> > -Steve
>> >
>> >
>> > On 21 February 2011 15:40, David Villasmil
>> > <david.villasmil.work at gmail.com>
>> > wrote:
>> >>
>> >> Thanks Steven,
>> >>
>> >> The sofia profile loads statically. Only the gateway part is passed
>> >> via curl, as I only want to load that dynamically.
>> >>
>> >> I see no errors anywhere:
>> >>
>> >> ?sofia profile external rescan
>> >> +OK scan complete
>> >>
>> >> freeswitch at internal>
>> >>
>> >> HOw do i remove all gateways before rescanning them?
>> >>
>> >> Thanks
>> >>
>> >> David
>> >>
>> >> On Mon, Feb 21, 2011 at 4:09 PM, Steven Ayre <steveayre at gmail.com>
>> >> wrote:
>> >> > "sofia status" will show you which profiles & gateways are defined.
>> >> >
>> >> > Are there any errors in FreeSWITCH's logfile when the XML is loaded?
>> >> > Perhaps
>> >> > there's an XML parsing error, or some other error which gets logged.
>> >> >
>> >> > Gateways are only loaded when you load the configuration (when
>> >> > mod_sofia
>> >> > loads, or on sofia profile name rescan). Existing gateways must be
>> >> > deleted
>> >> > and then rescanned to pick up configuration changes. Did you do that
>> >> > after
>> >> > making the changes?
>> >> >
>> >> > The gateway XML looks ok and I can confirm you can serve the config
>> >> > via
>> >> > mod_xml_curl (I do). Your profile looks incomplete though, for
>> >> > instance
>> >> > there's no sip-ip to bind to. Did you remove the profile params when
>> >> > posting
>> >> > to the list, or is that your entire profile?
>> >> >
>> >> > Regards,
>> >> > -Steve
>> >> >
>> >> >
>> >> > On 21 February 2011 14:47, David Villasmil
>> >> > <david.villasmil.work at gmail.com>
>> >> > wrote:
>> >> >>
>> >> >> Hello All,
>> >> >>
>> >> >> I'm trying to have my gateways defined dynamically with xml_curl.
>> >> >> According to the logs, the xml is requested and relivered to fs but
>> >> >> fs
>> >> >> doesn't seem to load it as i'm getting "invalid gateway" when i try
>> >> >> to
>> >> >> use them.
>> >> >>
>> >> >> this is my http://curl-gateway/gateways.php file:
>> >> >>
>> >> >>
>> >> >>
>> >> >> <document type="freeswitch/xml">
>> >> >> ? ? <configuration name="sofia.conf" description="Gateways
>> >> >> configuration">
>> >> >>
>> >> >> ? ? ? ? ?<profile name="external">
>> >> >> ? ? ? ? ? ? ? ? <!--
>> >> >> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
>> >> >> ? ? ? ? ? ? ? ? <!-- This profile is only for outbound registrations
>> >> >> to providers -->
>> >> >> ? ? ? ? ? ? ? ? <gateways>
>> >> >>
>> >> >> ? ? ? ? ? ? ? ? ? ?<gateway name="gw_1">
>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="realm" value="1.2.3.4:5060"/>
>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="username" value="foo"/>
>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="password" value="1234"/>
>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="register" value="false"/>
>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="retry-seconds" value="30"/>
>> >> >> ? ? ? ? ? ? ? ? ? ?</gateway>
>> >> >>
>> >> >> ? ? ? ? ? ? ? ? ? ?<gateway name="gw_2">
>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="realm" value="3.4.5.6:5060"/>
>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="username" value="foo"/>
>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="password" value="pass"/>
>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="register" value="false"/>
>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="retry-seconds" value="30"/>
>> >> >> ? ? ? ? ? ? ? ? ? ?</gateway>
>> >> >>
>> >> >> ? ? ? ? ? ? ? ? </gateways>
>> >> >>
>> >> >> ? ? ? ? ?</profile>
>> >> >>
>> >> >> ? ? </configuration>
>> >> >> </document>
>> >> >>
>> >> >>
>> >> >> and i get:
>> >> >>
>> >> >>
>> >> >> freeswitch at internal> sofia xmlstatus gateway gw_1
>> >> >> Invalid Gateway!
>> >> >>
>> >> >>
>> >> >> Is my file correct?
>> >> >>
>> >> >> It is not possible to find xml_curl gateways samples anywhere.
>> >> >>
>> >> >>
>> >> >>
>> >> >> 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
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 david.villasmil.work at gmail.com  Mon Feb 21 19:44:23 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 21 Feb 2011 17:44:23 +0100
Subject: [Freeswitch-users] XML_CURL & gateways
In-Reply-To: <AANLkTinKvJk7XmoSLwhqEczcOXNHy1YRP=NwPW3J1ooY@mail.gmail.com>
References: <AANLkTikJ+BEq6BW3MqRnpUuD9bz=hnvf-m_=ytFBsOVP@mail.gmail.com>
	<AANLkTin6DEWodK=Am2Gj=eGaL5jac4Y-c4Y9-E+nekY5@mail.gmail.com>
	<AANLkTinCMf4+OW52CZ1dj-hyLyLk3Fh2dmYC2q9vgJ=2@mail.gmail.com>
	<AANLkTi=ee+HR868-1D8ZMWe1AxB4-ndffEZwEoo=SpnG@mail.gmail.com>
	<AANLkTi=NTm+GPrFPuifvbUcnTOA+3MRJErPwfW-N7FU7@mail.gmail.com>
	<AANLkTi=31r4-Lainyq+K4XKyi2o+cQA3DuUY4MoJy+NA@mail.gmail.com>
	<AANLkTinKvJk7XmoSLwhqEczcOXNHy1YRP=NwPW3J1ooY@mail.gmail.com>
Message-ID: <AANLkTimA7DQ24RN+fLq1iv-BKmfU6MoBPmZEMBOkFL0=@mail.gmail.com>

Hello,

I changed sofia.conf.xml 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>

</configuration>

it seems to load now, but it doesn't load any profiles from the curl
server, this is my xml_curl:

<configuration name="xml_curl.conf" description="cURL XML Gateway">
  <bindings>
    <binding name="sofia">
      <param name="gateway-url"
value="http://93.90.20.99/fsxml/gateways.php"
bindings="configuration"/> -->
    </binding>
  </bindings>
</configuration>


I don't what binding name and bindings should be to load the rest of
sofia from the curl server...


any help?

Thanks!


David



On Mon, Feb 21, 2011 at 5:19 PM, David Villasmil
<david.villasmil.work at gmail.com> wrote:
> Yeah, that's what i was trying, but now sofia is not starting...
>
> could it be my xml_curl is wrong? here it is:
>
> trying to load all sofia config:
>
> <configuration name="xml_curl.conf" description="cURL XML Gateway">
> ?<bindings>
> ? ?<binding name="sofia.conf">
> ? ? ?<param name="gateway-url"
> value="http://server_ip/fsxml/gateways.php" bindings="sofia.conf"/>
> -->
> ? ?</binding>
> ?</bindings>
> </configuration>
>
> this: http://server_ip/fsxml/gateways.php returns all the sofia
> config, and it is ok because I copied the result into sofia.conf.xml
> and it starts prefectly...
>
> is my "bindings" ok? I've tried: sofia.conf.xml, sofia.conf and
> sofia... none seems to load it, i always get:
>
> 2011-02-21 17:08:58.277368 [ERR] sofia.c:2646 Open of sofia.conf failed
> 2011-02-21 17:08:58.277757 [CRIT] switch_loadable_module.c:882 Error
> Loading module /usr/local/freeswitch/mod/mod_sofia.so
> **Module load routine returned an error**
>
> I removed the sofia.conf.xml from the autoload_configs... not sure
> whether i should leave it or what...
>
> David
>
>
> On Mon, Feb 21, 2011 at 5:07 PM, Steven Ayre <steveayre at gmail.com> wrote:
>> Try (re)including the entire profile config. FS might be ignoring it because
>> of the missing profile params.
>>
>> -Steve
>>
>>
>> On 21 February 2011 15:56, David Villasmil <david.villasmil.work at gmail.com>
>> wrote:
>>>
>>> Thanks Steven,
>>>
>>> > If you load the profile statically at start and rescan later from
>>> > mod_xml_curl, it should add any missing gateways at that time.
>>> >
>>>
>>> Doesn't do it though... that's why i was wondering if I had something
>>> wrong in my gateways section. I just posted my config in thi email
>>> thread before this email. Hope you see something wrong with it.
>>>
>>> Thanks
>>>
>>> David
>>>
>>> On Mon, Feb 21, 2011 at 4:52 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>> > As Brian said, "sofia profile xxx killgw xxx" - you have to do them one
>>> > at a
>>> > time.
>>> >
>>> > If you load the profile statically at start and rescan later from
>>> > mod_xml_curl, it should add any missing gateways at that time.
>>> >
>>> > -Steve
>>> >
>>> >
>>> > On 21 February 2011 15:40, David Villasmil
>>> > <david.villasmil.work at gmail.com>
>>> > wrote:
>>> >>
>>> >> Thanks Steven,
>>> >>
>>> >> The sofia profile loads statically. Only the gateway part is passed
>>> >> via curl, as I only want to load that dynamically.
>>> >>
>>> >> I see no errors anywhere:
>>> >>
>>> >> ?sofia profile external rescan
>>> >> +OK scan complete
>>> >>
>>> >> freeswitch at internal>
>>> >>
>>> >> HOw do i remove all gateways before rescanning them?
>>> >>
>>> >> Thanks
>>> >>
>>> >> David
>>> >>
>>> >> On Mon, Feb 21, 2011 at 4:09 PM, Steven Ayre <steveayre at gmail.com>
>>> >> wrote:
>>> >> > "sofia status" will show you which profiles & gateways are defined.
>>> >> >
>>> >> > Are there any errors in FreeSWITCH's logfile when the XML is loaded?
>>> >> > Perhaps
>>> >> > there's an XML parsing error, or some other error which gets logged.
>>> >> >
>>> >> > Gateways are only loaded when you load the configuration (when
>>> >> > mod_sofia
>>> >> > loads, or on sofia profile name rescan). Existing gateways must be
>>> >> > deleted
>>> >> > and then rescanned to pick up configuration changes. Did you do that
>>> >> > after
>>> >> > making the changes?
>>> >> >
>>> >> > The gateway XML looks ok and I can confirm you can serve the config
>>> >> > via
>>> >> > mod_xml_curl (I do). Your profile looks incomplete though, for
>>> >> > instance
>>> >> > there's no sip-ip to bind to. Did you remove the profile params when
>>> >> > posting
>>> >> > to the list, or is that your entire profile?
>>> >> >
>>> >> > Regards,
>>> >> > -Steve
>>> >> >
>>> >> >
>>> >> > On 21 February 2011 14:47, David Villasmil
>>> >> > <david.villasmil.work at gmail.com>
>>> >> > wrote:
>>> >> >>
>>> >> >> Hello All,
>>> >> >>
>>> >> >> I'm trying to have my gateways defined dynamically with xml_curl.
>>> >> >> According to the logs, the xml is requested and relivered to fs but
>>> >> >> fs
>>> >> >> doesn't seem to load it as i'm getting "invalid gateway" when i try
>>> >> >> to
>>> >> >> use them.
>>> >> >>
>>> >> >> this is my http://curl-gateway/gateways.php file:
>>> >> >>
>>> >> >>
>>> >> >>
>>> >> >> <document type="freeswitch/xml">
>>> >> >> ? ? <configuration name="sofia.conf" description="Gateways
>>> >> >> configuration">
>>> >> >>
>>> >> >> ? ? ? ? ?<profile name="external">
>>> >> >> ? ? ? ? ? ? ? ? <!--
>>> >> >> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
>>> >> >> ? ? ? ? ? ? ? ? <!-- This profile is only for outbound registrations
>>> >> >> to providers -->
>>> >> >> ? ? ? ? ? ? ? ? <gateways>
>>> >> >>
>>> >> >> ? ? ? ? ? ? ? ? ? ?<gateway name="gw_1">
>>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="realm" value="1.2.3.4:5060"/>
>>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="username" value="foo"/>
>>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="password" value="1234"/>
>>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="register" value="false"/>
>>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="retry-seconds" value="30"/>
>>> >> >> ? ? ? ? ? ? ? ? ? ?</gateway>
>>> >> >>
>>> >> >> ? ? ? ? ? ? ? ? ? ?<gateway name="gw_2">
>>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="realm" value="3.4.5.6:5060"/>
>>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="username" value="foo"/>
>>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="password" value="pass"/>
>>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="register" value="false"/>
>>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="retry-seconds" value="30"/>
>>> >> >> ? ? ? ? ? ? ? ? ? ?</gateway>
>>> >> >>
>>> >> >> ? ? ? ? ? ? ? ? </gateways>
>>> >> >>
>>> >> >> ? ? ? ? ?</profile>
>>> >> >>
>>> >> >> ? ? </configuration>
>>> >> >> </document>
>>> >> >>
>>> >> >>
>>> >> >> and i get:
>>> >> >>
>>> >> >>
>>> >> >> freeswitch at internal> sofia xmlstatus gateway gw_1
>>> >> >> Invalid Gateway!
>>> >> >>
>>> >> >>
>>> >> >> Is my file correct?
>>> >> >>
>>> >> >> It is not possible to find xml_curl gateways samples anywhere.
>>> >> >>
>>> >> >>
>>> >> >>
>>> >> >> 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
>>> >> >
>>> >> >
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >
>>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> > http://www.freeswitch.org
>>> >> >
>>> >> >
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 Feb 21 20:01:37 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 21 Feb 2011 17:01:37 +0000
Subject: [Freeswitch-users] XML_CURL & gateways
In-Reply-To: <AANLkTimA7DQ24RN+fLq1iv-BKmfU6MoBPmZEMBOkFL0=@mail.gmail.com>
References: <AANLkTikJ+BEq6BW3MqRnpUuD9bz=hnvf-m_=ytFBsOVP@mail.gmail.com>
	<AANLkTin6DEWodK=Am2Gj=eGaL5jac4Y-c4Y9-E+nekY5@mail.gmail.com>
	<AANLkTinCMf4+OW52CZ1dj-hyLyLk3Fh2dmYC2q9vgJ=2@mail.gmail.com>
	<AANLkTi=ee+HR868-1D8ZMWe1AxB4-ndffEZwEoo=SpnG@mail.gmail.com>
	<AANLkTi=NTm+GPrFPuifvbUcnTOA+3MRJErPwfW-N7FU7@mail.gmail.com>
	<AANLkTi=31r4-Lainyq+K4XKyi2o+cQA3DuUY4MoJy+NA@mail.gmail.com>
	<AANLkTinKvJk7XmoSLwhqEczcOXNHy1YRP=NwPW3J1ooY@mail.gmail.com>
	<AANLkTimA7DQ24RN+fLq1iv-BKmfU6MoBPmZEMBOkFL0=@mail.gmail.com>
Message-ID: <AANLkTikN+O-+rNe9+S1vkrqydQaStWTpayVv15P=67kN@mail.gmail.com>

Can you post the entire file that is being sent to FS? It's almost certainly
a syntax problem.

Your mod_xml_curl config is fine. "configuration" is all you need. However
you should know that after mod_xml_curl loads *all* attempts to read a
config file will go to that URL. You can't make it specific to a single
config file. The request contains parameters that tell your script which
config file it's looking for.

You would be best off serving all of your configuration through
mod_xml_curl, except some of the more essential modules such as mod_logfile,
mod_event_socket etc.

-Steve



On 21 February 2011 16:44, David Villasmil
<david.villasmil.work at gmail.com>wrote:

> Hello,
>
> I changed sofia.conf.xml 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>
>
> </configuration>
>
> it seems to load now, but it doesn't load any profiles from the curl
> server, this is my xml_curl:
>
> <configuration name="xml_curl.conf" description="cURL XML Gateway">
>  <bindings>
>    <binding name="sofia">
>       <param name="gateway-url"
> value="http://93.90.20.99/fsxml/gateways.php"
> bindings="configuration"/> -->
>    </binding>
>  </bindings>
> </configuration>
>
>
> I don't what binding name and bindings should be to load the rest of
> sofia from the curl server...
>
>
> any help?
>
> Thanks!
>
>
> David
>
>
>
> On Mon, Feb 21, 2011 at 5:19 PM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
> > Yeah, that's what i was trying, but now sofia is not starting...
> >
> > could it be my xml_curl is wrong? here it is:
> >
> > trying to load all sofia config:
> >
> > <configuration name="xml_curl.conf" description="cURL XML Gateway">
> >  <bindings>
> >    <binding name="sofia.conf">
> >      <param name="gateway-url"
> > value="http://server_ip/fsxml/gateways.php" bindings="sofia.conf"/>
> > -->
> >    </binding>
> >  </bindings>
> > </configuration>
> >
> > this: http://server_ip/fsxml/gateways.php returns all the sofia
> > config, and it is ok because I copied the result into sofia.conf.xml
> > and it starts prefectly...
> >
> > is my "bindings" ok? I've tried: sofia.conf.xml, sofia.conf and
> > sofia... none seems to load it, i always get:
> >
> > 2011-02-21 17:08:58.277368 [ERR] sofia.c:2646 Open of sofia.conf failed
> > 2011-02-21 17:08:58.277757 [CRIT] switch_loadable_module.c:882 Error
> > Loading module /usr/local/freeswitch/mod/mod_sofia.so
> > **Module load routine returned an error**
> >
> > I removed the sofia.conf.xml from the autoload_configs... not sure
> > whether i should leave it or what...
> >
> > David
> >
> >
> > On Mon, Feb 21, 2011 at 5:07 PM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >> Try (re)including the entire profile config. FS might be ignoring it
> because
> >> of the missing profile params.
> >>
> >> -Steve
> >>
> >>
> >> On 21 February 2011 15:56, David Villasmil <
> david.villasmil.work at gmail.com>
> >> wrote:
> >>>
> >>> Thanks Steven,
> >>>
> >>> > If you load the profile statically at start and rescan later from
> >>> > mod_xml_curl, it should add any missing gateways at that time.
> >>> >
> >>>
> >>> Doesn't do it though... that's why i was wondering if I had something
> >>> wrong in my gateways section. I just posted my config in thi email
> >>> thread before this email. Hope you see something wrong with it.
> >>>
> >>> Thanks
> >>>
> >>> David
> >>>
> >>> On Mon, Feb 21, 2011 at 4:52 PM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >>> > As Brian said, "sofia profile xxx killgw xxx" - you have to do them
> one
> >>> > at a
> >>> > time.
> >>> >
> >>> > If you load the profile statically at start and rescan later from
> >>> > mod_xml_curl, it should add any missing gateways at that time.
> >>> >
> >>> > -Steve
> >>> >
> >>> >
> >>> > On 21 February 2011 15:40, David Villasmil
> >>> > <david.villasmil.work at gmail.com>
> >>> > wrote:
> >>> >>
> >>> >> Thanks Steven,
> >>> >>
> >>> >> The sofia profile loads statically. Only the gateway part is passed
> >>> >> via curl, as I only want to load that dynamically.
> >>> >>
> >>> >> I see no errors anywhere:
> >>> >>
> >>> >>  sofia profile external rescan
> >>> >> +OK scan complete
> >>> >>
> >>> >> freeswitch at internal>
> >>> >>
> >>> >> HOw do i remove all gateways before rescanning them?
> >>> >>
> >>> >> Thanks
> >>> >>
> >>> >> David
> >>> >>
> >>> >> On Mon, Feb 21, 2011 at 4:09 PM, Steven Ayre <steveayre at gmail.com>
> >>> >> wrote:
> >>> >> > "sofia status" will show you which profiles & gateways are
> defined.
> >>> >> >
> >>> >> > Are there any errors in FreeSWITCH's logfile when the XML is
> loaded?
> >>> >> > Perhaps
> >>> >> > there's an XML parsing error, or some other error which gets
> logged.
> >>> >> >
> >>> >> > Gateways are only loaded when you load the configuration (when
> >>> >> > mod_sofia
> >>> >> > loads, or on sofia profile name rescan). Existing gateways must be
> >>> >> > deleted
> >>> >> > and then rescanned to pick up configuration changes. Did you do
> that
> >>> >> > after
> >>> >> > making the changes?
> >>> >> >
> >>> >> > The gateway XML looks ok and I can confirm you can serve the
> config
> >>> >> > via
> >>> >> > mod_xml_curl (I do). Your profile looks incomplete though, for
> >>> >> > instance
> >>> >> > there's no sip-ip to bind to. Did you remove the profile params
> when
> >>> >> > posting
> >>> >> > to the list, or is that your entire profile?
> >>> >> >
> >>> >> > Regards,
> >>> >> > -Steve
> >>> >> >
> >>> >> >
> >>> >> > On 21 February 2011 14:47, David Villasmil
> >>> >> > <david.villasmil.work at gmail.com>
> >>> >> > wrote:
> >>> >> >>
> >>> >> >> Hello All,
> >>> >> >>
> >>> >> >> I'm trying to have my gateways defined dynamically with xml_curl.
> >>> >> >> According to the logs, the xml is requested and relivered to fs
> but
> >>> >> >> fs
> >>> >> >> doesn't seem to load it as i'm getting "invalid gateway" when i
> try
> >>> >> >> to
> >>> >> >> use them.
> >>> >> >>
> >>> >> >> this is my http://curl-gateway/gateways.php file:
> >>> >> >>
> >>> >> >>
> >>> >> >>
> >>> >> >> <document type="freeswitch/xml">
> >>> >> >>     <configuration name="sofia.conf" description="Gateways
> >>> >> >> configuration">
> >>> >> >>
> >>> >> >>          <profile name="external">
> >>> >> >>                 <!--
> >>> >> >> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
> >>> >> >>                 <!-- This profile is only for outbound
> registrations
> >>> >> >> to providers -->
> >>> >> >>                 <gateways>
> >>> >> >>
> >>> >> >>                    <gateway name="gw_1">
> >>> >> >>                         <param name="realm" value="1.2.3.4:5060
> "/>
> >>> >> >>                         <param name="username" value="foo"/>
> >>> >> >>                         <param name="password" value="1234"/>
> >>> >> >>                         <param name="register" value="false"/>
> >>> >> >>                         <param name="retry-seconds" value="30"/>
> >>> >> >>                    </gateway>
> >>> >> >>
> >>> >> >>                    <gateway name="gw_2">
> >>> >> >>                         <param name="realm" value="3.4.5.6:5060
> "/>
> >>> >> >>                         <param name="username" value="foo"/>
> >>> >> >>                         <param name="password" value="pass"/>
> >>> >> >>                         <param name="register" value="false"/>
> >>> >> >>                         <param name="retry-seconds" value="30"/>
> >>> >> >>                    </gateway>
> >>> >> >>
> >>> >> >>                 </gateways>
> >>> >> >>
> >>> >> >>          </profile>
> >>> >> >>
> >>> >> >>     </configuration>
> >>> >> >> </document>
> >>> >> >>
> >>> >> >>
> >>> >> >> and i get:
> >>> >> >>
> >>> >> >>
> >>> >> >> freeswitch at internal> sofia xmlstatus gateway gw_1
> >>> >> >> Invalid Gateway!
> >>> >> >>
> >>> >> >>
> >>> >> >> Is my file correct?
> >>> >> >>
> >>> >> >> It is not possible to find xml_curl gateways samples anywhere.
> >>> >> >>
> >>> >> >>
> >>> >> >>
> >>> >> >> 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
> >>> >> >
> >>> >> >
> >>> >> > _______________________________________________
> >>> >> > FreeSWITCH-users mailing list
> >>> >> > FreeSWITCH-users at lists.freeswitch.org
> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> >
> >>> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> > http://www.freeswitch.org
> >>> >> >
> >>> >> >
> >>> >>
> >>> >> _______________________________________________
> >>> >> FreeSWITCH-users mailing list
> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>
> >>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> http://www.freeswitch.org
> >>> >
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Mon Feb 21 20:07:05 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 21 Feb 2011 18:07:05 +0100
Subject: [Freeswitch-users] XML_CURL & gateways
In-Reply-To: <AANLkTikN+O-+rNe9+S1vkrqydQaStWTpayVv15P=67kN@mail.gmail.com>
References: <AANLkTikJ+BEq6BW3MqRnpUuD9bz=hnvf-m_=ytFBsOVP@mail.gmail.com>
	<AANLkTin6DEWodK=Am2Gj=eGaL5jac4Y-c4Y9-E+nekY5@mail.gmail.com>
	<AANLkTinCMf4+OW52CZ1dj-hyLyLk3Fh2dmYC2q9vgJ=2@mail.gmail.com>
	<AANLkTi=ee+HR868-1D8ZMWe1AxB4-ndffEZwEoo=SpnG@mail.gmail.com>
	<AANLkTi=NTm+GPrFPuifvbUcnTOA+3MRJErPwfW-N7FU7@mail.gmail.com>
	<AANLkTi=31r4-Lainyq+K4XKyi2o+cQA3DuUY4MoJy+NA@mail.gmail.com>
	<AANLkTinKvJk7XmoSLwhqEczcOXNHy1YRP=NwPW3J1ooY@mail.gmail.com>
	<AANLkTimA7DQ24RN+fLq1iv-BKmfU6MoBPmZEMBOkFL0=@mail.gmail.com>
	<AANLkTikN+O-+rNe9+S1vkrqydQaStWTpayVv15P=67kN@mail.gmail.com>
Message-ID: <AANLkTincfAPoz4BE6ieoud1epMGywK82Ks55=azh+XzQ@mail.gmail.com>

Hello Steven,

Layer 8 error!

My mistake, i was missing the most obvious thing:

<document type=\"freeswitch/xml\">
  <section name=\"configuration\">

duh!!!


It's now loading the whole sofia config from the server! Great!

Thanks, man.

David

On Mon, Feb 21, 2011 at 6:01 PM, Steven Ayre <steveayre at gmail.com> wrote:
> Can you post the entire file that is being sent to FS? It's almost certainly
> a syntax problem.
>
> Your mod_xml_curl config is fine. "configuration" is all you need. However
> you should know that after mod_xml_curl loads *all* attempts to read a
> config file will go to that URL. You can't make it specific to a single
> config file. The request contains parameters that tell your script which
> config file it's looking for.
>
> You would be best off serving all of your configuration through
> mod_xml_curl, except some of the more essential modules such as mod_logfile,
> mod_event_socket etc.
>
> -Steve
>
>
>
> On 21 February 2011 16:44, David Villasmil <david.villasmil.work at gmail.com>
> wrote:
>>
>> Hello,
>>
>> I changed sofia.conf.xml 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>
>>
>> </configuration>
>>
>> it seems to load now, but it doesn't load any profiles from the curl
>> server, this is my xml_curl:
>>
>> <configuration name="xml_curl.conf" description="cURL XML Gateway">
>> ?<bindings>
>> ? ?<binding name="sofia">
>> ? ? ?<param name="gateway-url"
>> value="http://93.90.20.99/fsxml/gateways.php"
>> bindings="configuration"/> -->
>> ? ?</binding>
>> ?</bindings>
>> </configuration>
>>
>>
>> I don't what binding name and bindings should be to load the rest of
>> sofia from the curl server...
>>
>>
>> any help?
>>
>> Thanks!
>>
>>
>> David
>>
>>
>>
>> On Mon, Feb 21, 2011 at 5:19 PM, David Villasmil
>> <david.villasmil.work at gmail.com> wrote:
>> > Yeah, that's what i was trying, but now sofia is not starting...
>> >
>> > could it be my xml_curl is wrong? here it is:
>> >
>> > trying to load all sofia config:
>> >
>> > <configuration name="xml_curl.conf" description="cURL XML Gateway">
>> > ?<bindings>
>> > ? ?<binding name="sofia.conf">
>> > ? ? ?<param name="gateway-url"
>> > value="http://server_ip/fsxml/gateways.php" bindings="sofia.conf"/>
>> > -->
>> > ? ?</binding>
>> > ?</bindings>
>> > </configuration>
>> >
>> > this: http://server_ip/fsxml/gateways.php returns all the sofia
>> > config, and it is ok because I copied the result into sofia.conf.xml
>> > and it starts prefectly...
>> >
>> > is my "bindings" ok? I've tried: sofia.conf.xml, sofia.conf and
>> > sofia... none seems to load it, i always get:
>> >
>> > 2011-02-21 17:08:58.277368 [ERR] sofia.c:2646 Open of sofia.conf failed
>> > 2011-02-21 17:08:58.277757 [CRIT] switch_loadable_module.c:882 Error
>> > Loading module /usr/local/freeswitch/mod/mod_sofia.so
>> > **Module load routine returned an error**
>> >
>> > I removed the sofia.conf.xml from the autoload_configs... not sure
>> > whether i should leave it or what...
>> >
>> > David
>> >
>> >
>> > On Mon, Feb 21, 2011 at 5:07 PM, Steven Ayre <steveayre at gmail.com>
>> > wrote:
>> >> Try (re)including the entire profile config. FS might be ignoring it
>> >> because
>> >> of the missing profile params.
>> >>
>> >> -Steve
>> >>
>> >>
>> >> On 21 February 2011 15:56, David Villasmil
>> >> <david.villasmil.work at gmail.com>
>> >> wrote:
>> >>>
>> >>> Thanks Steven,
>> >>>
>> >>> > If you load the profile statically at start and rescan later from
>> >>> > mod_xml_curl, it should add any missing gateways at that time.
>> >>> >
>> >>>
>> >>> Doesn't do it though... that's why i was wondering if I had something
>> >>> wrong in my gateways section. I just posted my config in thi email
>> >>> thread before this email. Hope you see something wrong with it.
>> >>>
>> >>> Thanks
>> >>>
>> >>> David
>> >>>
>> >>> On Mon, Feb 21, 2011 at 4:52 PM, Steven Ayre <steveayre at gmail.com>
>> >>> wrote:
>> >>> > As Brian said, "sofia profile xxx killgw xxx" - you have to do them
>> >>> > one
>> >>> > at a
>> >>> > time.
>> >>> >
>> >>> > If you load the profile statically at start and rescan later from
>> >>> > mod_xml_curl, it should add any missing gateways at that time.
>> >>> >
>> >>> > -Steve
>> >>> >
>> >>> >
>> >>> > On 21 February 2011 15:40, David Villasmil
>> >>> > <david.villasmil.work at gmail.com>
>> >>> > wrote:
>> >>> >>
>> >>> >> Thanks Steven,
>> >>> >>
>> >>> >> The sofia profile loads statically. Only the gateway part is passed
>> >>> >> via curl, as I only want to load that dynamically.
>> >>> >>
>> >>> >> I see no errors anywhere:
>> >>> >>
>> >>> >> ?sofia profile external rescan
>> >>> >> +OK scan complete
>> >>> >>
>> >>> >> freeswitch at internal>
>> >>> >>
>> >>> >> HOw do i remove all gateways before rescanning them?
>> >>> >>
>> >>> >> Thanks
>> >>> >>
>> >>> >> David
>> >>> >>
>> >>> >> On Mon, Feb 21, 2011 at 4:09 PM, Steven Ayre <steveayre at gmail.com>
>> >>> >> wrote:
>> >>> >> > "sofia status" will show you which profiles & gateways are
>> >>> >> > defined.
>> >>> >> >
>> >>> >> > Are there any errors in FreeSWITCH's logfile when the XML is
>> >>> >> > loaded?
>> >>> >> > Perhaps
>> >>> >> > there's an XML parsing error, or some other error which gets
>> >>> >> > logged.
>> >>> >> >
>> >>> >> > Gateways are only loaded when you load the configuration (when
>> >>> >> > mod_sofia
>> >>> >> > loads, or on sofia profile name rescan). Existing gateways must
>> >>> >> > be
>> >>> >> > deleted
>> >>> >> > and then rescanned to pick up configuration changes. Did you do
>> >>> >> > that
>> >>> >> > after
>> >>> >> > making the changes?
>> >>> >> >
>> >>> >> > The gateway XML looks ok and I can confirm you can serve the
>> >>> >> > config
>> >>> >> > via
>> >>> >> > mod_xml_curl (I do). Your profile looks incomplete though, for
>> >>> >> > instance
>> >>> >> > there's no sip-ip to bind to. Did you remove the profile params
>> >>> >> > when
>> >>> >> > posting
>> >>> >> > to the list, or is that your entire profile?
>> >>> >> >
>> >>> >> > Regards,
>> >>> >> > -Steve
>> >>> >> >
>> >>> >> >
>> >>> >> > On 21 February 2011 14:47, David Villasmil
>> >>> >> > <david.villasmil.work at gmail.com>
>> >>> >> > wrote:
>> >>> >> >>
>> >>> >> >> Hello All,
>> >>> >> >>
>> >>> >> >> I'm trying to have my gateways defined dynamically with
>> >>> >> >> xml_curl.
>> >>> >> >> According to the logs, the xml is requested and relivered to fs
>> >>> >> >> but
>> >>> >> >> fs
>> >>> >> >> doesn't seem to load it as i'm getting "invalid gateway" when i
>> >>> >> >> try
>> >>> >> >> to
>> >>> >> >> use them.
>> >>> >> >>
>> >>> >> >> this is my http://curl-gateway/gateways.php file:
>> >>> >> >>
>> >>> >> >>
>> >>> >> >>
>> >>> >> >> <document type="freeswitch/xml">
>> >>> >> >> ? ? <configuration name="sofia.conf" description="Gateways
>> >>> >> >> configuration">
>> >>> >> >>
>> >>> >> >> ? ? ? ? ?<profile name="external">
>> >>> >> >> ? ? ? ? ? ? ? ? <!--
>> >>> >> >> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
>> >>> >> >> ? ? ? ? ? ? ? ? <!-- This profile is only for outbound
>> >>> >> >> registrations
>> >>> >> >> to providers -->
>> >>> >> >> ? ? ? ? ? ? ? ? <gateways>
>> >>> >> >>
>> >>> >> >> ? ? ? ? ? ? ? ? ? ?<gateway name="gw_1">
>> >>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="realm"
>> >>> >> >> value="1.2.3.4:5060"/>
>> >>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="username" value="foo"/>
>> >>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="password" value="1234"/>
>> >>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="register" value="false"/>
>> >>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="retry-seconds" value="30"/>
>> >>> >> >> ? ? ? ? ? ? ? ? ? ?</gateway>
>> >>> >> >>
>> >>> >> >> ? ? ? ? ? ? ? ? ? ?<gateway name="gw_2">
>> >>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="realm"
>> >>> >> >> value="3.4.5.6:5060"/>
>> >>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="username" value="foo"/>
>> >>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="password" value="pass"/>
>> >>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="register" value="false"/>
>> >>> >> >> ? ? ? ? ? ? ? ? ? ? ? ? <param name="retry-seconds" value="30"/>
>> >>> >> >> ? ? ? ? ? ? ? ? ? ?</gateway>
>> >>> >> >>
>> >>> >> >> ? ? ? ? ? ? ? ? </gateways>
>> >>> >> >>
>> >>> >> >> ? ? ? ? ?</profile>
>> >>> >> >>
>> >>> >> >> ? ? </configuration>
>> >>> >> >> </document>
>> >>> >> >>
>> >>> >> >>
>> >>> >> >> and i get:
>> >>> >> >>
>> >>> >> >>
>> >>> >> >> freeswitch at internal> sofia xmlstatus gateway gw_1
>> >>> >> >> Invalid Gateway!
>> >>> >> >>
>> >>> >> >>
>> >>> >> >> Is my file correct?
>> >>> >> >>
>> >>> >> >> It is not possible to find xml_curl gateways samples anywhere.
>> >>> >> >>
>> >>> >> >>
>> >>> >> >>
>> >>> >> >> 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
>> >>> >> >
>> >>> >> >
>> >>> >> > _______________________________________________
>> >>> >> > FreeSWITCH-users mailing list
>> >>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >> >
>> >>> >> >
>> >>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >> > http://www.freeswitch.org
>> >>> >> >
>> >>> >> >
>> >>> >>
>> >>> >> _______________________________________________
>> >>> >> FreeSWITCH-users mailing list
>> >>> >> FreeSWITCH-users at lists.freeswitch.org
>> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>
>> >>> >>
>> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >> http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >
>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 massimiliano.ravelli at gmail.com  Mon Feb 21 20:12:41 2011
From: massimiliano.ravelli at gmail.com (Massimiliano Ravelli)
Date: Mon, 21 Feb 2011 18:12:41 +0100
Subject: [Freeswitch-users] xml_curl and gateways
In-Reply-To: <AANLkTi=85E16XR-B4GRgE2F6ikP40KPAAWgLjrvY05f0@mail.gmail.com>
References: <48D17932.4000302@gmx.net> <48D26646.5030407@freeswitch.org>
	<48D58B11.3070704@gmx.net>
	<AANLkTik-4jXeoCAwBiL7UkoppPw2M01VhHMb5UaTOZAa@mail.gmail.com>
	<C5AA9686-A623-4CE5-B562-5F53E81AC2AE@freeswitch.org>
	<AANLkTi=85E16XR-B4GRgE2F6ikP40KPAAWgLjrvY05f0@mail.gmail.com>
Message-ID: <AANLkTind8aJuCheMJzexrfMjDU+4nmAtsrOaGk_Xf5oy@mail.gmail.com>

2011/2/21 David Villasmil <david.villasmil.work at gmail.com>

> Hello Brian,
>
> It seems i didn't explain myself correctly, all sofia.xml is in xml
> files, except the <gateways> part. That's the only section i want
> loaded dynamically.
>
>
I'm using xml curl too and AFAIK you cannot load some parts of sofia config
from files and some other from xml curl request
If using curl you should provide all sofia configuration through http
response or (as fallback) entire configuration in static files.

Massimiliano
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From anthony.minessale at gmail.com  Mon Feb 21 20:15:36 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 21 Feb 2011 11:15:36 -0600
Subject: [Freeswitch-users] MOH not working in G729 passthru with native
 sound files
In-Reply-To: <AANLkTim2=nnHfXOPjMZCeJ1bmLOLYPSi0zspky9smZeD@mail.gmail.com>
References: <AANLkTi=41CbE_tU8LsW52rZCaD0Z+Y8NbE7WXh+bxTn1@mail.gmail.com>
	<AANLkTim2=nnHfXOPjMZCeJ1bmLOLYPSi0zspky9smZeD@mail.gmail.com>
Message-ID: <AANLkTimOvSfT+3xmLOZ8EEUiPgmqD=cHBZMoOY25aoc0@mail.gmail.com>

local_stream is not native its a stream running from a mixer.
There is no native file streamer you must put absolute paths to a
.G729 file omitting the extension.



On Mon, Feb 21, 2011 at 6:49 AM, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
> how did you record in G729?
>
> On Thu, Feb 10, 2011 at 3:41 AM, Marcin Wojtowicz <marcin321 at gmail.com>
> wrote:
>>
>> I'm trying to enable MOH when both legs of the call are using G729 (FS is
>> in passthru). I converted an edited sample wave file to G729 and put in the
>> appropriate folder, and FS loads it correctly because this is the message
>> that keeps popping up in the console:
>> 2011-02-09 21:10:25.609375 [INFO] mod_native_file.c:94 Opening File
>> [C:\freeswitch\sounds/music/8000\sample.G729] 8000hz
>>
>> I establish a call, and everything is fine, but when I press hold on my
>> handset I see an error message that says that G729 is only useable in
>> passthru (here is the debug message):
>>
>> 2011-02-09 21:14:01.828125 [DEBUG] switch_channel.c:1376
>> (sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy at 192.168.1.50:5060) Callstate Change
>> ACTIVE -> HELD
>> 2011-02-09 21:14:01.828125 [DEBUG] switch_core_session.c:954 Send signal
>> sofia/external/MYHOME#@74.63.41.218 [BREAK]
>> 2011-02-09 21:14:01.859375 [DEBUG] switch_core_session.c:709 Send signal
>> sofia/external/MYHOME#@74.63.41.218 [BREAK]
>> 2011-02-09 21:14:02.000000 [DEBUG] switch_ivr.c:563
>> sofia/external/MYHOME#@74.63.41.218 Command Execute
>> playback(local_stream://moh/8000)
>> EXECUTE sofia/external/MYHOME#@74.63.41.218
>> playback(local_stream://moh/8000)
>> 2011-02-09 21:14:02.000000 [DEBUG] mod_local_stream.c:421 Opening Stream
>> [moh/8000] 8000hz
>> 2011-02-09 21:14:02.000000 [DEBUG] switch_ivr_play_say.c:1244 Codec
>> Activated L16 at 8000hz 1 channels 20ms
>> 2011-02-09 21:14:02.015625 [ERR] mod_g729.c:102 This codec is only usable
>> in passthrough mode!
>> 2011-02-09 21:14:02.015625 [ERR] switch_core_io.c:1042 Codec G.729 encoder
>> error!
>> 2011-02-09 21:14:02.015625 [DEBUG] switch_ivr_play_say.c:1581 done playing
>> file
>>
>> I don't understand why that would be, since my music file is in G729 so
>> I'm not asking freeswitch to convert, only stream. My custom ringback
>> (before a call is established) works just fine using a similar method, so
>> could anyone explain me why what I want to do is not permitted?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 massimiliano.ravelli at gmail.com  Mon Feb 21 20:21:19 2011
From: massimiliano.ravelli at gmail.com (Massimiliano Ravelli)
Date: Mon, 21 Feb 2011 18:21:19 +0100
Subject: [Freeswitch-users] MOH not working in G729 passthru with native
 sound files
In-Reply-To: <AANLkTimaCpu6+SEOppgXpUzzBVoaFm9+70=PyXbEKjG4@mail.gmail.com>
References: <AANLkTi=41CbE_tU8LsW52rZCaD0Z+Y8NbE7WXh+bxTn1@mail.gmail.com>
	<AANLkTim2=nnHfXOPjMZCeJ1bmLOLYPSi0zspky9smZeD@mail.gmail.com>
	<AANLkTimaCpu6+SEOppgXpUzzBVoaFm9+70=PyXbEKjG4@mail.gmail.com>
Message-ID: <AANLkTi=1Y2PYdNSB5wusauzgp4tq9CKuEnq4Gg8bE6CR@mail.gmail.com>

2011/2/21 Steven Ayre <steveayre at gmail.com>

>
> http://wiki.freeswitch.org/wiki/Mod_native_file#Script_to_convert_a_sound_file_to_specific_formats_to_avoid_transcoding
>
>
I saw an fs_encode binary compiling a recent version of freeswitch.
Does anyone know if it can be used to achieve the same result without
installing asterisk ?
And without buying g729 codecs for asterisk too !  ;-)

Massimiliano
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From andre.netvision.com.br at gmail.com  Mon Feb 21 05:09:43 2011
From: andre.netvision.com.br at gmail.com (=?UTF-8?Q?Andr=C3=A9_Luiz_dos_Santos?=)
Date: Sun, 20 Feb 2011 23:09:43 -0300
Subject: [Freeswitch-users] "Sofia SIP stack may stop responding"
Message-ID: <AANLkTintFQpqFo58Po60e2=pLqj=hQ1A0ykDFfgi__tG@mail.gmail.com>

>From http://wiki.freeswitch.org/wiki/Sofia:
"Sometimes, in extremely rare edge cases, the Sofia SIP stack may stop
responding. ..."

Is there more documentation on this?  Specifically, is there something
I should avoid doing to lower my chances of hitting those rare edge
cases?


From mthakershi at gmail.com  Mon Feb 21 23:22:28 2011
From: mthakershi at gmail.com (Malay Thakershi)
Date: Mon, 21 Feb 2011 14:22:28 -0600
Subject: [Freeswitch-users] FS with Microsoft Forefront TMG
Message-ID: <AANLkTikkkM1S=F9u3UoZOO4oUzwDh28KUr3+8B+OGKWc@mail.gmail.com>

Hello,

Anyone has tried using FS successfully behind Microsoft Forefront Threat
Management Gateway?

Thank you for any direction.

Malay
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From msc at freeswitch.org  Mon Feb 21 23:35:54 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 21 Feb 2011 12:35:54 -0800
Subject: [Freeswitch-users] Using 16 KHz sounds
In-Reply-To: <AANLkTim5CsCL4unpWh1OvH13Xx4HrKX2KF7Jg6UQYAZR@mail.gmail.com>
References: <AANLkTim5CsCL4unpWh1OvH13Xx4HrKX2KF7Jg6UQYAZR@mail.gmail.com>
Message-ID: <AANLkTin9tYDVJC0KUncV0h=g0+u=833QcRtp49OfU2XC@mail.gmail.com>

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
>
>
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From curriegrad2004 at gmail.com  Tue Feb 22 00:28:35 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 21 Feb 2011 13:28:35 -0800
Subject: [Freeswitch-users] FS with Microsoft Forefront TMG
In-Reply-To: <AANLkTikkkM1S=F9u3UoZOO4oUzwDh28KUr3+8B+OGKWc@mail.gmail.com>
References: <AANLkTikkkM1S=F9u3UoZOO4oUzwDh28KUr3+8B+OGKWc@mail.gmail.com>
Message-ID: <AANLkTimkZ-ogeJKLhegt=rYnVpJH9JuLBBm=EJuayPU+@mail.gmail.com>

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
>
>


From mthakershi at gmail.com  Tue Feb 22 01:03:15 2011
From: mthakershi at gmail.com (Malay Thakershi)
Date: Mon, 21 Feb 2011 16:03:15 -0600
Subject: [Freeswitch-users] FS with Microsoft Forefront TMG
In-Reply-To: <AANLkTimkZ-ogeJKLhegt=rYnVpJH9JuLBBm=EJuayPU+@mail.gmail.com>
References: <AANLkTikkkM1S=F9u3UoZOO4oUzwDh28KUr3+8B+OGKWc@mail.gmail.com>
	<AANLkTimkZ-ogeJKLhegt=rYnVpJH9JuLBBm=EJuayPU+@mail.gmail.com>
Message-ID: <AANLkTim-=bO-q+ed+ub_xOZDKF17BGtzjAmdTtLdc_Y8@mail.gmail.com>

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
>
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From william.nishio at voicetechnology.com.br  Tue Feb 22 01:21:26 2011
From: william.nishio at voicetechnology.com.br (William Kendi ...)
Date: Mon, 21 Feb 2011 19:21:26 -0300
Subject: [Freeswitch-users] mod_fsv, record and playback options
In-Reply-To: <AANLkTinAwkmV4R=RLrhASWkymL8Crb7MRn5aJWoTM1tY@mail.gmail.com>
References: <AANLkTink1-sZBh9zO9p8_he8XfaH+NdLHRHwZi5n_BE5@mail.gmail.com>
	<AANLkTi=e_n9T3NFaB=Y1BE6_NAtFhAAan7m96WoKOM6t@mail.gmail.com>
	<AANLkTim0Zm1GLpbU5q2dvsm=nguwuMr6xY4ZUQAMWGX0@mail.gmail.com>
	<AANLkTinAwkmV4R=RLrhASWkymL8Crb7MRn5aJWoTM1tY@mail.gmail.com>
Message-ID: <AANLkTi=f-+Y8XqY6pwD=c4hD0VCLQXQHqib6yayZEjww@mail.gmail.com>

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
>
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From jaybinks at gmail.com  Tue Feb 22 01:33:16 2011
From: jaybinks at gmail.com (jay binks)
Date: Tue, 22 Feb 2011 08:33:16 +1000
Subject: [Freeswitch-users] INVITE DoS Prevention
In-Reply-To: <B706C824-2C91-4F74-9D68-87FF817EA969@5ninesolutions.com>
References: <C98764B0.11627%krice@freeswitch.org>
	<A624F826-AF73-40BD-8DCD-10F39551D924@5ninesolutions.com>
	<9922.1298280678@ccs.covici.com>
	<B706C824-2C91-4F74-9D68-87FF817EA969@5ninesolutions.com>
Message-ID: <AANLkTi=FT5ZC5A8eUnucoWEm=xncWMWHiyTtyVA6mb8F@mail.gmail.com>

could you not just modify your fail2ban regex and set the threshold for
Register Auths in fail2ban ?
( you would not want to do it on invite auths, because it will include GOOD
auths for calls )

I still like the idea of loggin sip_authentication timeouts ..
I might play with that a little today.

J

On Mon, Feb 21, 2011 at 8:28 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> After tinkering with it, I think that might be the best way.  The
> iptables method is cool but I'd like to have more dynamic control and
> with Fail2Ban looking at the challenges you could specifically ignore
> certain high traffic IPs and block others.  What would be very cool is
> if instead of logging every challenge, a log entry was written if
> there was a high number from a specific IP, then you could decide what
> to do about it with fail2ban, similar to the pike module for opensips
> does.
>
>
> On Feb 21, 2011, at 1:31 AM, covici at ccs.covici.com wrote:
>
> > I would change sip auth failure to challenge and then have sufficient
> > times to only block if there are too many challenges in a certain
> > time.
> > I am not even sure the failure works any more in recent gits.
> >
> > Spencer Thomason <spencer at 5ninesolutions.com> wrote:
> >
> >> Yes, that works great if they respond to the challenge with a failed
> >> auth. But the scenario I'm trying to prevent is if they just send the
> >> INVITE and never respond to the challenge.  Fail2Ban will not work as
> >> every endpoint will initially send an INVITE and receive a challenge.
> >> Legit calls will then respond correctly and not be logged as a SIP
> >> auth failure but every call that is challenged will show up as SIP
> >> auth challenge in the logs so there is no regex to differentiate
> >> between legit an non legit traffic.
> >>
> >> Spencer
> >>
> >> On Feb 20, 2011, at 10:39 PM, Ken Rice wrote:
> >>
> >>> Fail2Ban ... This is block an IP with too many failed attempts from
> >>> something like SipVicious pretty quickly
> >>>
> >>>
> >>> On 2/20/11 11:07 PM, "Spencer Thomason" <spencer at 5ninesolutions.com>
> >>> wrote:
> >>>
> >>>> Hi,
> >>>> We run hosted Freeswitch instances in VMs with the internal profile
> >>>> on
> >>>> port 5060 connecting to clients mostly behind NAT and then the
> >>>> external profile connecting to our proxies only.  Protecting the
> >>>> external profile its straightforward.. we only allow traffic to/
> >>>> from
> >>>> our proxies at the firewall level.  But protecting the internal
> >>>> profile seems to be a bit more difficult because the UACs could be
> >>>> theoretically anywhere on the network.
> >>>>
> >>>> I'm currently using Fail2Ban to prevent brute force registration
> >>>> and
> >>>> INVITEs on auth failures, e.g.:
> >>>> failregex = \[WARNING\] sofia_reg.c:\d+ SIP auth failure \
> >>>> (REGISTER\)
> >>>> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
> >>>>            \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(INVITE\)
> >>>> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
> >>>>
> >>>> My question is, since its part of a normal SIP dialog to challenge
> >>>> the
> >>>> INVITE, is there any way to prevent a possible DoS from just sheer
> >>>> volume of incoming INVITEs on an Internet facing server
> >>>> automatically.  I.e., If you block the logged challenge, you'd
> >>>> block
> >>>> all legitimate INVITEs and registrations.  Since its UDP traffic I
> >>>> couldn't come up with a way to do it automatically at the iptables
> >>>> level. i.e. number of concurrent connections.  Is there some option
> >>>> to
> >>>> just not respond if a client is sending a number of requests over a
> >>>> certain threshold?  It might not stop them from sending the traffic
> >>>> but pretty soon they'd get the idea that it wasn't going to go
> >>>> anywhere.  My concern is say there are 50 Freeswitch instances on a
> >>>> box (albeit 8 core, 32GB ram, 8 15K raid 10 storage) and someone
> >>>> starts sending thousands of rouge INVITEs to every VM on a physical
> >>>> box that the CPU load from just challenging the incoming INVITEs
> >>>> would
> >>>> create a DoS.  We the logs regularly to try to catch people doing
> >>>> this
> >>>> sort of thing and drop them at a router upstream of the core
> >>>> network,
> >>>> but I'd like to have it happen without human intervention.  Have I
> >>>> completely over thought this and am missing something obvious?
> >>>>
> >>>> Thanks,
> >>>> 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
> >>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at 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
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 k-b-24 at live.com  Tue Feb 22 01:51:31 2011
From: k-b-24 at live.com (Jason b.a)
Date: Mon, 21 Feb 2011 22:51:31 +0000
Subject: [Freeswitch-users] Sip proxy!
Message-ID: <col111-w21E9DE4BF84C471A1AB45ACD90@phx.gbl>


Guys is there any free open source Sip Proxy that runs on windows ! 
 
Regards 		 	   		  
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From spencer at 5ninesolutions.com  Tue Feb 22 01:55:57 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Mon, 21 Feb 2011 14:55:57 -0800
Subject: [Freeswitch-users] INVITE DoS Prevention
In-Reply-To: <AANLkTi=FT5ZC5A8eUnucoWEm=xncWMWHiyTtyVA6mb8F@mail.gmail.com>
References: <C98764B0.11627%krice@freeswitch.org>
	<A624F826-AF73-40BD-8DCD-10F39551D924@5ninesolutions.com>
	<9922.1298280678@ccs.covici.com>
	<B706C824-2C91-4F74-9D68-87FF817EA969@5ninesolutions.com>
	<AANLkTi=FT5ZC5A8eUnucoWEm=xncWMWHiyTtyVA6mb8F@mail.gmail.com>
Message-ID: <57CD944B-1B4C-4CCC-A698-97206AED588F@5ninesolutions.com>

Yes, and that works well.  Initially I was trying to guard against a  
bunch of INVITEs without auth replys.  For the time being I've set up  
a separate fail2ban filter that looks at the invite challenges and  
only blocks someone if its extremely high in a short period of time.    
I still have to figure out how many of these INVITEs without auths the  
systems can handle because when you've got several instances going  
just the disk activity from the logs becomes problematic if there is a  
massive spike.  Logging the auth timeouts seems to be the ideal  
solution because you could then drop the timeouts without affecting  
legit traffic.

Spencer


On Feb 21, 2011, at 2:33 PM, jay binks wrote:

> could you not just modify your fail2ban regex and set the threshold  
> for Register Auths in fail2ban ?
> ( you would not want to do it on invite auths, because it will  
> include GOOD auths for calls )
>
> I still like the idea of loggin sip_authentication timeouts ..
> I might play with that a little today.
>
> J
>
> On Mon, Feb 21, 2011 at 8:28 PM, Spencer Thomason <spencer at 5ninesolutions.com 
> > wrote:
> After tinkering with it, I think that might be the best way.  The
> iptables method is cool but I'd like to have more dynamic control and
> with Fail2Ban looking at the challenges you could specifically ignore
> certain high traffic IPs and block others.  What would be very cool is
> if instead of logging every challenge, a log entry was written if
> there was a high number from a specific IP, then you could decide what
> to do about it with fail2ban, similar to the pike module for opensips
> does.
>
>
> On Feb 21, 2011, at 1:31 AM, covici at ccs.covici.com wrote:
>
> > I would change sip auth failure to challenge and then have  
> sufficient
> > times to only block if there are too many challenges in a certain
> > time.
> > I am not even sure the failure works any more in recent gits.
> >
> > Spencer Thomason <spencer at 5ninesolutions.com> wrote:
> >
> >> Yes, that works great if they respond to the challenge with a  
> failed
> >> auth. But the scenario I'm trying to prevent is if they just send  
> the
> >> INVITE and never respond to the challenge.  Fail2Ban will not  
> work as
> >> every endpoint will initially send an INVITE and receive a  
> challenge.
> >> Legit calls will then respond correctly and not be logged as a SIP
> >> auth failure but every call that is challenged will show up as SIP
> >> auth challenge in the logs so there is no regex to differentiate
> >> between legit an non legit traffic.
> >>
> >> Spencer
> >>
> >> On Feb 20, 2011, at 10:39 PM, Ken Rice wrote:
> >>
> >>> Fail2Ban ... This is block an IP with too many failed attempts  
> from
> >>> something like SipVicious pretty quickly
> >>>
> >>>
> >>> On 2/20/11 11:07 PM, "Spencer Thomason" <spencer at 5ninesolutions.com 
> >
> >>> wrote:
> >>>
> >>>> Hi,
> >>>> We run hosted Freeswitch instances in VMs with the internal  
> profile
> >>>> on
> >>>> port 5060 connecting to clients mostly behind NAT and then the
> >>>> external profile connecting to our proxies only.  Protecting the
> >>>> external profile its straightforward.. we only allow traffic to/
> >>>> from
> >>>> our proxies at the firewall level.  But protecting the internal
> >>>> profile seems to be a bit more difficult because the UACs could  
> be
> >>>> theoretically anywhere on the network.
> >>>>
> >>>> I'm currently using Fail2Ban to prevent brute force registration
> >>>> and
> >>>> INVITEs on auth failures, e.g.:
> >>>> failregex = \[WARNING\] sofia_reg.c:\d+ SIP auth failure \
> >>>> (REGISTER\)
> >>>> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
> >>>>            \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(INVITE 
> \)
> >>>> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
> >>>>
> >>>> My question is, since its part of a normal SIP dialog to  
> challenge
> >>>> the
> >>>> INVITE, is there any way to prevent a possible DoS from just  
> sheer
> >>>> volume of incoming INVITEs on an Internet facing server
> >>>> automatically.  I.e., If you block the logged challenge, you'd
> >>>> block
> >>>> all legitimate INVITEs and registrations.  Since its UDP  
> traffic I
> >>>> couldn't come up with a way to do it automatically at the  
> iptables
> >>>> level. i.e. number of concurrent connections.  Is there some  
> option
> >>>> to
> >>>> just not respond if a client is sending a number of requests  
> over a
> >>>> certain threshold?  It might not stop them from sending the  
> traffic
> >>>> but pretty soon they'd get the idea that it wasn't going to go
> >>>> anywhere.  My concern is say there are 50 Freeswitch instances  
> on a
> >>>> box (albeit 8 core, 32GB ram, 8 15K raid 10 storage) and someone
> >>>> starts sending thousands of rouge INVITEs to every VM on a  
> physical
> >>>> box that the CPU load from just challenging the incoming INVITEs
> >>>> would
> >>>> create a DoS.  We the logs regularly to try to catch people doing
> >>>> this
> >>>> sort of thing and drop them at a router upstream of the core
> >>>> network,
> >>>> but I'd like to have it happen without human intervention.   
> Have I
> >>>> completely over thought this and am missing something obvious?
> >>>>
> >>>> Thanks,
> >>>> 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
> >>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at 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
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 curriegrad2004 at gmail.com  Tue Feb 22 02:29:33 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 21 Feb 2011 15:29:33 -0800
Subject: [Freeswitch-users] FS with Microsoft Forefront TMG
In-Reply-To: <AANLkTim-=bO-q+ed+ub_xOZDKF17BGtzjAmdTtLdc_Y8@mail.gmail.com>
References: <AANLkTikkkM1S=F9u3UoZOO4oUzwDh28KUr3+8B+OGKWc@mail.gmail.com>
	<AANLkTimkZ-ogeJKLhegt=rYnVpJH9JuLBBm=EJuayPU+@mail.gmail.com>
	<AANLkTim-=bO-q+ed+ub_xOZDKF17BGtzjAmdTtLdc_Y8@mail.gmail.com>
Message-ID: <AANLkTin4e_re8OgGOhRMRLhY=E320-rJhibtbdds_tHj@mail.gmail.com>

I'd say uncomment the rtp-start-port and rtp-end-port directives just
to make sure. Afiak, just publish it like a normal service or
something like that, but remember, that was done under ISA 2006, not
the newer TMG

On Mon, Feb 21, 2011 at 2:03 PM, Malay Thakershi <mthakershi at gmail.com> wrote:
> 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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 jaybinks at gmail.com  Tue Feb 22 03:06:00 2011
From: jaybinks at gmail.com (jay binks)
Date: Tue, 22 Feb 2011 10:06:00 +1000
Subject: [Freeswitch-users] INVITE DoS Prevention
In-Reply-To: <57CD944B-1B4C-4CCC-A698-97206AED588F@5ninesolutions.com>
References: <C98764B0.11627%krice@freeswitch.org>
	<A624F826-AF73-40BD-8DCD-10F39551D924@5ninesolutions.com>
	<9922.1298280678@ccs.covici.com>
	<B706C824-2C91-4F74-9D68-87FF817EA969@5ninesolutions.com>
	<AANLkTi=FT5ZC5A8eUnucoWEm=xncWMWHiyTtyVA6mb8F@mail.gmail.com>
	<57CD944B-1B4C-4CCC-A698-97206AED588F@5ninesolutions.com>
Message-ID: <AANLkTimUnz6w3rWWm=NchZuQh-SH3AsvBkh73XBvSouk@mail.gmail.com>

maybe put your logs on TmpFS also then ?
I assume you have your DB on TmpFS already ?

Jay


On Tue, Feb 22, 2011 at 8:55 AM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> Yes, and that works well.  Initially I was trying to guard against a bunch
> of INVITEs without auth replys.  For the time being I've set up a separate
> fail2ban filter that looks at the invite challenges and only blocks someone
> if its extremely high in a short period of time.   I still have to figure
> out how many of these INVITEs without auths the systems can handle because
> when you've got several instances going just the disk activity from the logs
> becomes problematic if there is a massive spike.  Logging the auth timeouts
> seems to be the ideal solution because you could then drop the timeouts
> without affecting legit traffic.
>
> Spencer
>
>
> On Feb 21, 2011, at 2:33 PM, jay binks wrote:
>
> could you not just modify your fail2ban regex and set the threshold for
> Register Auths in fail2ban ?
> ( you would not want to do it on invite auths, because it will include GOOD
> auths for calls )
>
> I still like the idea of loggin sip_authentication timeouts ..
> I might play with that a little today.
>
> J
>
> On Mon, Feb 21, 2011 at 8:28 PM, Spencer Thomason <
> spencer at 5ninesolutions.com> wrote:
>
>> After tinkering with it, I think that might be the best way.  The
>> iptables method is cool but I'd like to have more dynamic control and
>> with Fail2Ban looking at the challenges you could specifically ignore
>> certain high traffic IPs and block others.  What would be very cool is
>> if instead of logging every challenge, a log entry was written if
>> there was a high number from a specific IP, then you could decide what
>> to do about it with fail2ban, similar to the pike module for opensips
>> does.
>>
>>
>> On Feb 21, 2011, at 1:31 AM, covici at ccs.covici.com wrote:
>>
>> > I would change sip auth failure to challenge and then have sufficient
>> > times to only block if there are too many challenges in a certain
>> > time.
>> > I am not even sure the failure works any more in recent gits.
>> >
>> > Spencer Thomason <spencer at 5ninesolutions.com> wrote:
>> >
>> >> Yes, that works great if they respond to the challenge with a failed
>> >> auth. But the scenario I'm trying to prevent is if they just send the
>> >> INVITE and never respond to the challenge.  Fail2Ban will not work as
>> >> every endpoint will initially send an INVITE and receive a challenge.
>> >> Legit calls will then respond correctly and not be logged as a SIP
>> >> auth failure but every call that is challenged will show up as SIP
>> >> auth challenge in the logs so there is no regex to differentiate
>> >> between legit an non legit traffic.
>> >>
>> >> Spencer
>> >>
>> >> On Feb 20, 2011, at 10:39 PM, Ken Rice wrote:
>> >>
>> >>> Fail2Ban ... This is block an IP with too many failed attempts from
>> >>> something like SipVicious pretty quickly
>> >>>
>> >>>
>> >>> On 2/20/11 11:07 PM, "Spencer Thomason" <spencer at 5ninesolutions.com>
>> >>> wrote:
>> >>>
>> >>>> Hi,
>> >>>> We run hosted Freeswitch instances in VMs with the internal profile
>> >>>> on
>> >>>> port 5060 connecting to clients mostly behind NAT and then the
>> >>>> external profile connecting to our proxies only.  Protecting the
>> >>>> external profile its straightforward.. we only allow traffic to/
>> >>>> from
>> >>>> our proxies at the firewall level.  But protecting the internal
>> >>>> profile seems to be a bit more difficult because the UACs could be
>> >>>> theoretically anywhere on the network.
>> >>>>
>> >>>> I'm currently using Fail2Ban to prevent brute force registration
>> >>>> and
>> >>>> INVITEs on auth failures, e.g.:
>> >>>> failregex = \[WARNING\] sofia_reg.c:\d+ SIP auth failure \
>> >>>> (REGISTER\)
>> >>>> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
>> >>>>            \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(INVITE\)
>> >>>> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
>> >>>>
>> >>>> My question is, since its part of a normal SIP dialog to challenge
>> >>>> the
>> >>>> INVITE, is there any way to prevent a possible DoS from just sheer
>> >>>> volume of incoming INVITEs on an Internet facing server
>> >>>> automatically.  I.e., If you block the logged challenge, you'd
>> >>>> block
>> >>>> all legitimate INVITEs and registrations.  Since its UDP traffic I
>> >>>> couldn't come up with a way to do it automatically at the iptables
>> >>>> level. i.e. number of concurrent connections.  Is there some option
>> >>>> to
>> >>>> just not respond if a client is sending a number of requests over a
>> >>>> certain threshold?  It might not stop them from sending the traffic
>> >>>> but pretty soon they'd get the idea that it wasn't going to go
>> >>>> anywhere.  My concern is say there are 50 Freeswitch instances on a
>> >>>> box (albeit 8 core, 32GB ram, 8 15K raid 10 storage) and someone
>> >>>> starts sending thousands of rouge INVITEs to every VM on a physical
>> >>>> box that the CPU load from just challenging the incoming INVITEs
>> >>>> would
>> >>>> create a DoS.  We the logs regularly to try to catch people doing
>> >>>> this
>> >>>> sort of thing and drop them at a router upstream of the core
>> >>>> network,
>> >>>> but I'd like to have it happen without human intervention.  Have I
>> >>>> completely over thought this and am missing something obvious?
>> >>>>
>> >>>> Thanks,
>> >>>> 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
>> >>>
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at 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
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
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From wstephen80 at gmail.com  Tue Feb 22 02:39:06 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 22 Feb 2011 00:39:06 +0100
Subject: [Freeswitch-users] ESL socket outbound: the execution of answer has
	not effect
Message-ID: <AANLkTimY7t8iBSZUGuCLx2f_e=3R+taGfW4s-NaT2Evg@mail.gmail.com>

I'm running an esl socket outbound application and I have a systematic
problem: in a particular situation, the execution of answer on incoming call
has no effect.

This happens always when (and only in this case):

1. an incoming call is received (legA) and via dialplan I send the call to
my socket outbound application
2. I make an outbound call with originate (legB)
3. I receive a CHANNEL_PROGRESS_MEDIA on legB
4. I do a "pre_answer" on legA
5, I do a "uuid_bridge" between legA and legB

in this phase, legA can hear the progress tone coming from legB (an in-band
ringback)

5. legB answer the call and I receive a CHANNEL_ANSWER on legB
6. I do an execute of "answer" on legA

This "answer" has no effect!

I do both pre_answer and answer specifying the UUID parameter (sendmsg
<uuid>), obviously using legA uuid .

I do the bridge specifying inbound uuid as first parameter and outbound uuid
as second parameter: uuid_bridge <inbound_uuid> <outbound_uuid>

In fs_cli I see:  [WARNING] switch_ivr_bridge.c:1412 reversing order of
channels so this will work!

To me it seems that in uuid_bridge is done an inversion with my
inbound/outbound channels and because I do an answer to the incoming uuid,
probably (due to the inversion) is treated as outbound so the answer is
ignored.

Attached to this email the .pcap file related to tcp traffic between
Freeswitch and my socket outbound application where at time 8.562639 there
is my sendmsg with the answer.

Stephen
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From msc at freeswitch.org  Tue Feb 22 03:18:57 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 21 Feb 2011 16:18:57 -0800
Subject: [Freeswitch-users] Sip proxy!
In-Reply-To: <col111-w21E9DE4BF84C471A1AB45ACD90@phx.gbl>
References: <col111-w21E9DE4BF84C471A1AB45ACD90@phx.gbl>
Message-ID: <AANLkTimBsZOpCHOKWZ8BdpV-QLqzb58=gLQmL=GwnCgD@mail.gmail.com>

http://sourceforge.net/projects/winsip/

On Mon, Feb 21, 2011 at 2:51 PM, Jason b.a <k-b-24 at live.com> wrote:

>  Guys is there any free open source Sip Proxy that runs on windows !
>
> 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  Tue Feb 22 03:33:14 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 21 Feb 2011 16:33:14 -0800
Subject: [Freeswitch-users] bridging?
In-Reply-To: <AANLkTikFuFg1mDDgvHpz_VM2QSKZ0mPLTpee_K2-arJD@mail.gmail.com>
References: <AANLkTikkWZtatkiLRLdvu9-T2-VdsNRtyAk7ZCMCcF+4@mail.gmail.com>
	<AANLkTikWBopbR5yk6xAy-y6Ueqm+4wS+=_MNVah9bqD1@mail.gmail.com>
	<AANLkTimCeaoCc5pTs+a2vhhyQkDnuFX2dstB1uNy3uX_@mail.gmail.com>
	<AANLkTikFuFg1mDDgvHpz_VM2QSKZ0mPLTpee_K2-arJD@mail.gmail.com>
Message-ID: <AANLkTincMLaEJBCQbetimLZJBnpd0gDqMDAaV14U8cVC@mail.gmail.com>

On Sat, Feb 19, 2011 at 4:55 AM, Neil Patel <neilp at cs.stanford.edu> wrote:

> Hi Michael,
>
> Using transfer seems to do the trick:
>
> session:setAutoHangup(false);
> session:transfer("7777", "XML", "default");
>
> Any downsides to this?
> Neil
>

Not at all!
-MC
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From msc at freeswitch.org  Tue Feb 22 03:39:20 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 21 Feb 2011 16:39:20 -0800
Subject: [Freeswitch-users] Enterprise bridge,
	groups and "follow me" dialplan
In-Reply-To: <AANLkTintRxtrmCdKi8MMYsvP5hNZEu7Nutj96+LCKE33@mail.gmail.com>
References: <AANLkTintRxtrmCdKi8MMYsvP5hNZEu7Nutj96+LCKE33@mail.gmail.com>
Message-ID: <AANLkTik9OaWak7cq3SO6Mw7hdjHnZWwwLbmJLnvGdf+C@mail.gmail.com>

How are you defining what is in each group? Are these simply locally
registered users? External phone numbers? Both? Those questions will help
determine how to proceed.

-MC

On Sat, Feb 19, 2011 at 9:05 AM, Dmitry Bely <dmitry.bely at gmail.com> wrote:

> I am trying to implement the following bridging scenario:
>
> <action application="set" data="call_timeout=30"/>
> <action application="bridge"
> data="group/group1 at domain,[leg_delay_start=10]group/group2 at domain"/>
> ...
>
> i.e. all phones in group1 should ring for 10 seconds and if nobody
> answers group1+group2 rings for additional 20 seconds.
>
> That almost works, but only the first phone rings in either group:
> [WARNING] switch_ivr_originate.c:2339 Only calling the first element
> in the list in this mode.
>
> Surely I need enterprise bridge:
>
> <action application="bridge"
> data="group/group1 at domain:_:[leg_delay_start=10]group/group2 at domain"/>
>
> but now there is no correct caller id and no leg delay. Caller id can
> be restored with:
>
> <action application="bridge" data="
>
>  <origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number}>
>  group/group1 at domain:_:[leg_delay_start=10]group/group2 at domain"/>
>
> (is this a bug or design decision?), but what to do with leg delay?
> [leg_delay_start=10] is simply ignored there.
>
> - 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 mbsip at gazeta.pl  Tue Feb 22 04:16:13 2011
From: mbsip at gazeta.pl (Mac)
Date: Tue, 22 Feb 2011 02:16:13 +0100
Subject: [Freeswitch-users] RTP Proxy mode, DTMF conversion problem
In-Reply-To: <AANLkTikABFQ4hwj9+dkUyKGTU02+dL4gY8DYdRH8RNhz@mail.gmail.com>
References: <AANLkTimUvhO65MTg7DaMRbbzO1M5bz4oKg5JfnR0_6uC@mail.gmail.com>
	<12CA5F58-378A-4231-9E1E-3F89DD59049B@gmail.com>
	<AANLkTikABFQ4hwj9+dkUyKGTU02+dL4gY8DYdRH8RNhz@mail.gmail.com>
Message-ID: <AANLkTikO+U5=A9yLepzeA8-LxggWM2SLcHRDfzfr4ieg@mail.gmail.com>

Hello,

Could somebody give me a hint?

Thanks,
Mac

> Hi Steven,
>
> There is much better :)
> I have different SDPs right now. The one is with rtpmap:101
> telephone-event/8000 (leg that is working with RFC2833), the opposite
> one incorporates a=rtpmap:101 telephone-event/8000 in INVITE SDP and
> pure codec rtpmaps.
> There is rtp.p_type == 101 working only on the left side. I cannot
> find any rtp.p_type == 101 on thark in opposide side which is okay.
> But the problem still persists.
>
> Once setting my UA to work with Inband DTMF everything works fine.
>
> Thanks,
> Mac
>
> 2011/2/20 Steven Ayre <steveayre at gmail.com>:
>> <action application="set" data="proxy_media=true"/>
>>
>> As indicated by the error, this is the problem. "proxy passthrough". In
>> proxy mode you pass the media straight through (passthrough). You can't
>> process passthrough media, such as is needed to mix in inband dtmf.
>> Use proxy_media=false and bypass_media=false. That's actually the default so
>> unless you're setting either to true in the sip profile, you can remove
>> those lines from the dialplan completely.
>> Steve on iPhone
>> On 20 Feb 2011, at 18:51, Mac <mbsip at gazeta.pl> wrote:
>>
>> action application="set" data="proxy_media=true"/>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 Feb 22 04:20:37 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 21 Feb 2011 17:20:37 -0800
Subject: [Freeswitch-users] escape in originate
In-Reply-To: <AANLkTi=EGjfNFKPYgsCYrw_-OyXAj+OpZOi4QSQWROTp@mail.gmail.com>
References: <AANLkTi=EGjfNFKPYgsCYrw_-OyXAj+OpZOi4QSQWROTp@mail.gmail.com>
Message-ID: <AANLkTim+r5gvZaxfbYrop5awZLv5ynPu1yF3KgqX+3ML@mail.gmail.com>

This is indeed the case. I'm looking switch_utils.c and it surely looks like
there is no means for escaping the commas. If you use a backslash then both
the backslash and the comma get passed through as-is.

Any suggestions on how to send through strings that contain commas?
-MC

On Sun, Feb 20, 2011 at 3:20 AM, Seven Du <dujinfang at gmail.com> wrote:

> I want to put tone_stream in dialstring but don't know how to escape the
> comma.
>
>
> originate
>  {some_var='tone_stream://%(2000,4000,440.0,480.0);loops=1'}portaudio/auto_answer
> &echo
> 2011-02-20 19:13:09.516139 [DEBUG] switch_ivr_originate.c:1971
> variable string 0 = [some_var=tone_stream://%(2000]
> 2011-02-20 19:13:09.516139 [DEBUG] switch_ivr_originate.c:1971
> variable string 1 = [4000]
> 2011-02-20 19:13:09.516139 [DEBUG] switch_ivr_originate.c:1971
> variable string 2 = [440.0]
> 2011-02-20 19:13:09.516139 [DEBUG] switch_ivr_originate.c:1971
> variable string 3 = [480.0);loops=1]
>
> There's a work around but it's not elegant.
>
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-September/062753.html
>
> Actually we may not need to escape the comma, the parser should keep
> reading until found the second quote. Would this need a patch?
>
> Thanks.
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From freeswitch at cartissolutions.com  Tue Feb 22 04:45:58 2011
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Mon, 21 Feb 2011 19:45:58 -0600
Subject: [Freeswitch-users] How to create a C++ application (project),
 that can code FS C code function?
In-Reply-To: <AANLkTikfxCrHsC5S-DZVfQejSWiWtF0HRM15MXksX-qK@mail.gmail.com>
References: <AANLkTikfxCrHsC5S-DZVfQejSWiWtF0HRM15MXksX-qK@mail.gmail.com>
Message-ID: <4D631556.7060307@cartissolutions.com>

Once upon a time I wrote the first C++ module that used to be in the 
tree.  Changes to the core obsoleted the module, but I can tell you that 
the process between making an application module is basically setting 
your module to compile with the c++ compiler.  The only difficulty comes 
into play when you want to expose functionality to C code.

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


On 02/14/2011 03:30 AM, Frankie Yiu wrote:
> Hi there,
> I am new to FreeSwitch and I would like to implement a C++ application 
> / mod that would do some analysis of RTP packets.  I would like to ask 
> what mod I should use as a template to create my C++ project so that I 
> can access the core FreeSwitch APIs / functions, since they are in C 
> language?
> I know under Endpoints, there are 2 MODs that are using C++ code 
> (mod_h323 and mod_opal), but I don't know if I should implement the 
> same way if I am creating an "application" instead of "endpoints".
> Please kindly let me know if there is any special / required function 
> that I need to implemented in C++ code.
>
> Thanks in advance.
> 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 anthony.minessale at gmail.com  Tue Feb 22 05:42:49 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 21 Feb 2011 20:42:49 -0600
Subject: [Freeswitch-users] ESL socket outbound: the execution of answer
 has not effect
In-Reply-To: <AANLkTimY7t8iBSZUGuCLx2f_e=3R+taGfW4s-NaT2Evg@mail.gmail.com>
References: <AANLkTimY7t8iBSZUGuCLx2f_e=3R+taGfW4s-NaT2Evg@mail.gmail.com>
Message-ID: <AANLkTim5AQSRCij70F1fmsyEzV2jphXnTqP6YA0Y6+X3@mail.gmail.com>

If you bridge them, the bridge app will already pass the  answer for
you when the B leg is answered.
If you are seeing that message it means you are supplying the uuid in
reverse order where the A leg is not the one with media.

That reversing the order message can only be true if you pass a leg
with media as the 2nd leg and one without as the 1st to uuid_bridge.

When you pre_answer the A leg, do you wait for it to actually happen
before you call bridge?
sendmsg execute only queues the app to be executed, you need to get
the event confirming it actually happened before you can be sure it
did.

Try adding the header

async: true

to your sendmsg to indicate you want to execute the app instantly and
wait for it to finish.



On Mon, Feb 21, 2011 at 5:39 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:
> I'm running an esl socket outbound application and I have a systematic
> problem: in a particular situation, the execution of answer on incoming call
> has no effect.
> This happens always when (and only in this case):
> 1. an incoming call is received?(legA)?and via dialplan I send the call to
> my socket outbound application
> 2. I make an outbound call with originate (legB)
> 3. I receive a CHANNEL_PROGRESS_MEDIA on legB
> 4. I do a "pre_answer" on legA
> 5, I do a "uuid_bridge" between legA and legB
> in this phase, legA can hear the progress tone coming from legB (an in-band
> ringback)
> 5. legB answer the call and I receive a CHANNEL_ANSWER on legB
> 6. I do an execute of "answer" on legA
> This "answer" has no effect!
> I do both pre_answer and answer specifying the UUID parameter (sendmsg
> <uuid>), obviously using legA uuid .
> I do the bridge specifying inbound uuid as first parameter and outbound uuid
> as second parameter: uuid_bridge <inbound_uuid> <outbound_uuid>
> In fs_cli I see:??[WARNING] switch_ivr_bridge.c:1412 reversing order of
> channels so this will work!
> To me it seems that in uuid_bridge is done an inversion with my
> inbound/outbound channels and because I do an answer to the incoming uuid,
> probably (due to the inversion) is treated as outbound so the answer is
> ignored.
> Attached to this email the .pcap file related to tcp traffic between
> Freeswitch and my socket outbound application where at time 8.562639 there
> is my sendmsg with the answer.
> 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
>
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 dmitry.bely at gmail.com  Tue Feb 22 10:19:22 2011
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Tue, 22 Feb 2011 10:19:22 +0300
Subject: [Freeswitch-users] Enterprise bridge,
	groups and "follow me" dialplan
In-Reply-To: <AANLkTik9OaWak7cq3SO6Mw7hdjHnZWwwLbmJLnvGdf+C@mail.gmail.com>
References: <AANLkTintRxtrmCdKi8MMYsvP5hNZEu7Nutj96+LCKE33@mail.gmail.com>
	<AANLkTik9OaWak7cq3SO6Mw7hdjHnZWwwLbmJLnvGdf+C@mail.gmail.com>
Message-ID: <AANLkTikTBJK7sm4uWSc2wHpTAw__1hU2kd5p=VrmS7vL@mail.gmail.com>

On Tue, Feb 22, 2011 at 3:39 AM, Michael Collins <msc at freeswitch.org> wrote:
> How are you defining what is in each group? Are these simply locally
> registered users? External phone numbers? Both? Those questions will help
> determine how to proceed.

Yes, they are mostly locally registered users. The groups are defined like

      <group name="group1">
        <users>
          <user id="1001" type="pointer"/>
          <user id="1002" type="pointer"/>
        </users>
      </group>

Some users are associated with external numbers:

<include>
  <user id="external_user">
    <params>
      <param name="dial-string"
value="[toll_allow=local]loopback/XXXXXXX/default/XML"/>
    </params>
    <variables>
      <variable name="user_context" value="default"/>
    </variables>
  </user>
</include>

if that matters.

> On Sat, Feb 19, 2011 at 9:05 AM, Dmitry Bely <dmitry.bely at gmail.com> wrote:
>>
>> I am trying to implement the following bridging scenario:
>>
>> <action application="set" data="call_timeout=30"/>
>> <action application="bridge"
>> data="group/group1 at domain,[leg_delay_start=10]group/group2 at domain"/>
>> ...
>>
>> i.e. all phones in group1 should ring for 10 seconds and if nobody
>> answers group1+group2 rings for additional 20 seconds.
>>
>> That almost works, but only the first phone rings in either group:
>> [WARNING] switch_ivr_originate.c:2339 Only calling the first element
>> in the list in this mode.
>>
>> Surely I need enterprise bridge:
>>
>> <action application="bridge"
>> data="group/group1 at domain:_:[leg_delay_start=10]group/group2 at domain"/>
>>
>> but now there is no correct caller id and no leg delay. Caller id can
>> be restored with:
>>
>> <action application="bridge" data="
>>
>> ?<origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number}>
>> ?group/group1 at domain:_:[leg_delay_start=10]group/group2 at domain"/>
>>
>> (is this a bug or design decision?), but what to do with leg delay?
>> [leg_delay_start=10] is simply ignored there.

- Dmitry Bely


From mayamatakeshi at gmail.com  Tue Feb 22 10:23:57 2011
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Tue, 22 Feb 2011 16:23:57 +0900
Subject: [Freeswitch-users] Updating callee info when using application
	intercept
Message-ID: <AANLkTi=7O9JXWkXzT88sGiVMKeOmAU=PX4W+WC-S3RsY@mail.gmail.com>

Is there any way to update the callee number/name that would show up in a
terminal that gets a dialplan to intercept a channel?
I am trying with this:

        <action application='set' data='sip_callee_id_number=12341234'/>
        <action application='set' data='sip_callee_id_name=abcdabcd'/>
        <action application='send_display' data='fffff'/>
        <action application='answer'/>
        <action application='intercept' data='Some-UUID'/>"

But I don't see any changes (I was expecting that it might change the
username in the header To or cause FS to send some extra SIP message before
or after "200 OK").

r,
takeshi
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From jmesquita at freeswitch.org  Mon Feb 21 16:08:45 2011
From: jmesquita at freeswitch.org (Jmesquita@freeswitch.org)
Date: Mon, 21 Feb 2011 10:08:45 -0300
Subject: [Freeswitch-users] INVITE DoS Prevention
In-Reply-To: <B706C824-2C91-4F74-9D68-87FF817EA969@5ninesolutions.com>
References: <C98764B0.11627%krice@freeswitch.org>
	<A624F826-AF73-40BD-8DCD-10F39551D924@5ninesolutions.com>
	<9922.1298280678@ccs.covici.com>
	<B706C824-2C91-4F74-9D68-87FF817EA969@5ninesolutions.com>
Message-ID: <4430D9CC-43AC-460F-AF31-348008044A67@freeswitch.org>

Just an idea for the other developers. Wouldn't be cool to have and event thrown by each module that does authentication so that other application modules are able to listen in those events and do whatever with it? Some people might even like to have that running an eel daemon and throw some snmp traps based on that?

Just an idea that might be completely ridiculous or might raise some interest. Events are pretty cheap so there wouldn't be a lot of overhead involved.

Regards,

Jo?o Mesquita

On 21/02/2011, at 07:28, Spencer Thomason <spencer at 5ninesolutions.com> wrote:

> After tinkering with it, I think that might be the best way.  The  
> iptables method is cool but I'd like to have more dynamic control and  
> with Fail2Ban looking at the challenges you could specifically ignore  
> certain high traffic IPs and block others.  What would be very cool is  
> if instead of logging every challenge, a log entry was written if  
> there was a high number from a specific IP, then you could decide what  
> to do about it with fail2ban, similar to the pike module for opensips  
> does.
> 
> 
> On Feb 21, 2011, at 1:31 AM, covici at ccs.covici.com wrote:
> 
>> I would change sip auth failure to challenge and then have sufficient
>> times to only block if there are too many challenges in a certain  
>> time.
>> I am not even sure the failure works any more in recent gits.
>> 
>> Spencer Thomason <spencer at 5ninesolutions.com> wrote:
>> 
>>> Yes, that works great if they respond to the challenge with a failed
>>> auth. But the scenario I'm trying to prevent is if they just send the
>>> INVITE and never respond to the challenge.  Fail2Ban will not work as
>>> every endpoint will initially send an INVITE and receive a challenge.
>>> Legit calls will then respond correctly and not be logged as a SIP
>>> auth failure but every call that is challenged will show up as SIP
>>> auth challenge in the logs so there is no regex to differentiate
>>> between legit an non legit traffic.
>>> 
>>> Spencer
>>> 
>>> On Feb 20, 2011, at 10:39 PM, Ken Rice wrote:
>>> 
>>>> Fail2Ban ... This is block an IP with too many failed attempts from
>>>> something like SipVicious pretty quickly
>>>> 
>>>> 
>>>> On 2/20/11 11:07 PM, "Spencer Thomason" <spencer at 5ninesolutions.com>
>>>> wrote:
>>>> 
>>>>> Hi,
>>>>> We run hosted Freeswitch instances in VMs with the internal profile
>>>>> on
>>>>> port 5060 connecting to clients mostly behind NAT and then the
>>>>> external profile connecting to our proxies only.  Protecting the
>>>>> external profile its straightforward.. we only allow traffic to/ 
>>>>> from
>>>>> our proxies at the firewall level.  But protecting the internal
>>>>> profile seems to be a bit more difficult because the UACs could be
>>>>> theoretically anywhere on the network.
>>>>> 
>>>>> I'm currently using Fail2Ban to prevent brute force registration  
>>>>> and
>>>>> INVITEs on auth failures, e.g.:
>>>>> failregex = \[WARNING\] sofia_reg.c:\d+ SIP auth failure \ 
>>>>> (REGISTER\)
>>>>> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
>>>>>           \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(INVITE\)
>>>>> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
>>>>> 
>>>>> My question is, since its part of a normal SIP dialog to challenge
>>>>> the
>>>>> INVITE, is there any way to prevent a possible DoS from just sheer
>>>>> volume of incoming INVITEs on an Internet facing server
>>>>> automatically.  I.e., If you block the logged challenge, you'd  
>>>>> block
>>>>> all legitimate INVITEs and registrations.  Since its UDP traffic I
>>>>> couldn't come up with a way to do it automatically at the iptables
>>>>> level. i.e. number of concurrent connections.  Is there some option
>>>>> to
>>>>> just not respond if a client is sending a number of requests over a
>>>>> certain threshold?  It might not stop them from sending the traffic
>>>>> but pretty soon they'd get the idea that it wasn't going to go
>>>>> anywhere.  My concern is say there are 50 Freeswitch instances on a
>>>>> box (albeit 8 core, 32GB ram, 8 15K raid 10 storage) and someone
>>>>> starts sending thousands of rouge INVITEs to every VM on a physical
>>>>> box that the CPU load from just challenging the incoming INVITEs
>>>>> would
>>>>> create a DoS.  We the logs regularly to try to catch people doing
>>>>> this
>>>>> sort of thing and drop them at a router upstream of the core  
>>>>> network,
>>>>> but I'd like to have it happen without human intervention.  Have I
>>>>> completely over thought this and am missing something obvious?
>>>>> 
>>>>> Thanks,
>>>>> 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
>>>> 
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mayamatakeshi at gmail.com  Tue Feb 22 10:29:22 2011
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Tue, 22 Feb 2011 16:29:22 +0900
Subject: [Freeswitch-users] Updating callee info when using application
	intercept
In-Reply-To: <AANLkTi=7O9JXWkXzT88sGiVMKeOmAU=PX4W+WC-S3RsY@mail.gmail.com>
References: <AANLkTi=7O9JXWkXzT88sGiVMKeOmAU=PX4W+WC-S3RsY@mail.gmail.com>
Message-ID: <AANLkTiksFhBj8Ko_aLE3HtowcAaSxjQ4-aThxydnPvn=@mail.gmail.com>

On Tue, Feb 22, 2011 at 4:23 PM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

> Is there any way to update the callee number/name that would show up in a
> terminal that gets a dialplan to intercept a channel?
> I am trying with this:
>
>         <action application='set' data='sip_callee_id_number=12341234'/>
>         <action application='set' data='sip_callee_id_name=abcdabcd'/>
>         <action application='send_display' data='fffff'/>
>         <action application='answer'/>
>         <action application='intercept' data='Some-UUID'/>"
>
> But I don't see any changes (I was expecting that it might change the
> username in the header To or cause FS to send some extra SIP message before
> or after "200 OK").
>

Sorry,
nevermind.

It escaped my attention that a header was added to the "200 OK" response:

Remote-Party-ID: "abcdabcd" <sip:12341234 at 127.0.0.1
>;party=calling;privacy=off;screen=no

So I believe my terminal doesn't support the above.
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From steveayre at gmail.com  Tue Feb 22 10:48:12 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 22 Feb 2011 07:48:12 +0000
Subject: [Freeswitch-users] RTP Proxy mode, DTMF conversion problem
In-Reply-To: <AANLkTikO+U5=A9yLepzeA8-LxggWM2SLcHRDfzfr4ieg@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>
Message-ID: <AANLkTikH6EKOMzYriW=egWA-dFo0-KckRY=Qsxn-FgoL@mail.gmail.com>

Get a debug-level log with siptrace enabled and paste it so we can see
what's going on.

-Steve


On 22 February 2011 01:16, Mac <mbsip at gazeta.pl> wrote:

> Hello,
>
> Could somebody give me a hint?
>
> Thanks,
> Mac
>
> > Hi Steven,
> >
> > There is much better :)
> > I have different SDPs right now. The one is with rtpmap:101
> > telephone-event/8000 (leg that is working with RFC2833), the opposite
> > one incorporates a=rtpmap:101 telephone-event/8000 in INVITE SDP and
> > pure codec rtpmaps.
> > There is rtp.p_type == 101 working only on the left side. I cannot
> > find any rtp.p_type == 101 on thark in opposide side which is okay.
> > But the problem still persists.
> >
> > Once setting my UA to work with Inband DTMF everything works fine.
> >
> > Thanks,
> > Mac
> >
> > 2011/2/20 Steven Ayre <steveayre at gmail.com>:
> >> <action application="set" data="proxy_media=true"/>
> >>
> >> As indicated by the error, this is the problem. "proxy passthrough". In
> >> proxy mode you pass the media straight through (passthrough). You can't
> >> process passthrough media, such as is needed to mix in inband dtmf.
> >> Use proxy_media=false and bypass_media=false. That's actually the
> default so
> >> unless you're setting either to true in the sip profile, you can remove
> >> those lines from the dialplan completely.
> >> Steve on iPhone
> >> On 20 Feb 2011, at 18:51, Mac <mbsip at gazeta.pl> wrote:
> >>
> >> action application="set" data="proxy_media=true"/>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ce at kapper.net  Tue Feb 22 04:50:56 2011
From: ce at kapper.net (Clemens Ebentheuer)
Date: Tue, 22 Feb 2011 02:50:56 +0100
Subject: [Freeswitch-users] t38_gateway self - no re-Invite?
Message-ID: <1B19ABD72889C245AE8EEE08AC24103A28C1AFC00E@exmachina.office.kapper.net>

Hi to all from austria,

I?m very new to freeswitch and I?m trying to get t38_gateway self working - but no success.
for testing I use zoiper or spa2102 as t38 "device" - i have troubles in outbound direction:

t38 device(internal) -> FS:t38_gateway self -> sip-provider (external - no t38 support)

freeswitch sends no re-invite, I guess
and Zoiper BYEs with Reason: SIP;description="Stale re-Invite"

also see:
http://pastebin.freeswitch.org/15439

inbound (t38_gateway peer) works fine - also when I do 

t38 device -> FS:rxfax

my simple dialplan for the outbound call:

<extension name="scpa" >
   <condition field="destination_number" expression="^1(.*)$">
       <action application="set" data="fax_enable_t38_request=true"/>
       <action application="set" data="fax_enable_t38=true"/>
       <action application="set" data="execute_on_answer=t38_gateway self"/>
       <action application="bridge" data="sofia/gateway/scpa567/$1"/>
   </condition>
</extension>

Maybe I have to mention that I?m using Fusionpbx?!
Thanks for any advice! I allready tried different examples I found, but no luck...

Clemens



From delorenzodesign at gmail.com  Tue Feb 22 10:22:31 2011
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Tue, 22 Feb 2011 02:22:31 -0500
Subject: [Freeswitch-users] Message Playback Stops
Message-ID: <AANLkTi=EMQsSkH-SZDq2rUzvh2-7nYK=NpKHAXXEwc-W@mail.gmail.com>

Hello,

I have a Freeswitch installation that is intended to make many calls
(thousands) and playback a single wav file.  The calls are successfully
processed (the recipient's phone rings), but the call almost immediately
disconnects, after about 1s.  Everything seems to work fine if I'm only
pushing one or two calls through the Freeswitch instance, but as soon as I
turn up the call rate (I'm still only doing about 50 concurrent sessions)
the playback begins to fail.

I've watched the calls go out from the console and nothing looks out of the
ordinary, except that the calls are disconnected with NORMAL CLEARING prior
to completion.

Here's the Lua script I'm using...

profile_id = argv[1];
account_code = argv[2];
client_id = argv[3];
caller_id_name = argv[4];
caller_id = argv[5];
dial_id = argv[6];
number_to_call = argv[7];
message_to_play = argv[8];
max_retries = argv[9];
retry_interval = argv[10];

local human_detected = false
local voicemail_detected = false;
local message_played = false;

recordings_directory = "/usr/local/freeswitch/recordings/messages/";

function setDialVariables(set_as_session_variables)
    local s = "profile_id=" .. profile_id;
    s = s .. ",account_code=" .. account_code;
    s = s .. ",client_id=" .. client_id;
    s = s .. ",caller_id_name=" .. caller_id_name;
    s = s .. ",caller_id=" .. caller_id;
    s = s .. ",dial_id=" .. dial_id;
    s = s .. ",number_to_call=" .. number_to_call;
    s = s .. ",message_to_play=" .. message_to_play;

    freeswitch.consoleLog("notice", s .. "\n");

    return s
end

function printSessionVariables()
    freeswitch.consoleLog("notice", "******* PRINTING SESSION VARIABLES
**********\n");
    -- ommitted
    freeswitch.consoleLog("notice",
"**********************************************\n");
end

function onInput(s, type, obj, arg)
    if(type == "event" and voicemail_detected == false) then
        freeswitch.consoleLog("debug","************ VOICE MAIL/ANSWERING
MACHINE DETECTED *************\n");
        voicemail_detected = true;
        return "break";
    end
    return true;
end

function playbackMessage(sleepTime)
    message_played = false;
    session:sleep(sleepTime);
    -- play a file
    message_file = recordings_directory .. message_to_play;
    freeswitch.consoleLog("notice", "Playing file: " .. message_file ..
"\n");
    session:streamFile(message_file);
    freeswitch.consoleLog("notice", "!!!!! Finished playing the file
!!!!!\n");
    message_played = true;
end

session = freeswitch.Session("{" .. setDialVariables(false) ..
",ignore_early_media=true,origination_caller_id_name=" .. caller_id_name ..
",origination_caller_id_number=+1" .. caller_id .. "}sofia/gateway/gateway_"
.. profile_id .. "/" .. number_to_call);

while(session:ready()) do
   setDialVariables(true)
   session:answer();

   -- session:execute("continue_on_fail","true");
   session:setInputCallback("onInput","true");
   session:execute("avmd","start");

   playbackMessage(200);

   vm_status = voicemail_detected == true and "yes" or "no"
   freeswitch.consoleLog("info", "Was VM detected? " .. vm_status .. "\n");
   if(voicemail_detected) then
      return "break";
   end

   freeswitch.consoleLog("notice", "Played the message at least once and
checked for VM, we should be exiting the loop.\n")
end

if (voicemail_detected) then
    freeswitch.consoleLog("info", "Playback for voicemail.\n");
    session:execute("avmd","stop");
    playbackMessage(5000);
end

freeswitch.consoleLog("info", "All finished, hanging up the session.\n");
session:hangup();

Any help would be greatly appreciated.

Thank you,

Michael
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From spencer at 5ninesolutions.com  Tue Feb 22 12:02:36 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Tue, 22 Feb 2011 01:02:36 -0800
Subject: [Freeswitch-users] VM to email error
Message-ID: <817664CE-FEA5-430C-A0C2-549F0FD510C7@5ninesolutions.com>

Hello all,
I keep seeing an error in the logs similar to the following when a VM  
is emailed.  Everything seems to be working just fine so I'm trying to  
decipher the log entry.

2011-02-21 19:57:48.288269 [ERR] switch_utils.c:699 Unable to execute  
command: /bin/cat /tmp/mail.12983398676f47 | /usr/sbin/sendmail -f fromuser at fromdomain.com 
  -t touser at todomain.com

/bin/cat works as does /usr/sbin/sendmail.

This is on Centos 5.5 w/ Postfix instead of sendmail and /var/log/ 
maillog clearly shows the message being delivered.  Do I have  
something misconfigured?

Thanks,
Spencer



From helmut.kuper at ewetel.de  Tue Feb 22 12:28:32 2011
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 22 Feb 2011 10:28:32 +0100
Subject: [Freeswitch-users] t38_gateway self - no re-Invite?
In-Reply-To: <1B19ABD72889C245AE8EEE08AC24103A28C1AFC00E@exmachina.office.kapper.net>
References: <1B19ABD72889C245AE8EEE08AC24103A28C1AFC00E@exmachina.office.kapper.net>
Message-ID: <4D6381C0.7040408@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello Clemens,

I'm using t38 a few month now. And I have had also big problems to get
FS and mod_spandsp as a t38-Gateway up and running. The only way I found
to get it work in most cases was to change the code in mod_spandsp.c

I changed line 99 from
switch_ivr_tone_detect_session(session, "t38", "1100", "rw", timeout, 1,
data, NULL, t38_gateway_start)

to
switch_ivr_tone_detect_session(session, "t38", "2100", "rw", timeout, 1,
data, NULL, t38_gateway_start)

The recompile mod_spandsp and reload it into FS.

This could affected the inbound way. So you have to play with the
frequency to detect. Maybe you have to listen to both 1100Hz and 2100Hz.

t38 is not really easy with mod_spandsp.



Am 22.02.2011 02:50, schrieb Clemens Ebentheuer:
> Hi to all from austria,
> 
> I?m very new to freeswitch and I?m trying to get t38_gateway self working - but no success.
> for testing I use zoiper or spa2102 as t38 "device" - i have troubles in outbound direction:
> 
> t38 device(internal) -> FS:t38_gateway self -> sip-provider (external - no t38 support)
> 
> freeswitch sends no re-invite, I guess
> and Zoiper BYEs with Reason: SIP;description="Stale re-Invite"
> 
> also see:
> http://pastebin.freeswitch.org/15439
> 
> inbound (t38_gateway peer) works fine - also when I do 
> 
> t38 device -> FS:rxfax
> 
> my simple dialplan for the outbound call:
> 
> <extension name="scpa" >
>    <condition field="destination_number" expression="^1(.*)$">
>        <action application="set" data="fax_enable_t38_request=true"/>
>        <action application="set" data="fax_enable_t38=true"/>
>        <action application="set" data="execute_on_answer=t38_gateway self"/>
>        <action application="bridge" data="sofia/gateway/scpa567/$1"/>
>    </condition>
> </extension>
> 
> Maybe I have to mention that I?m using Fusionpbx?!
> Thanks for any advice! I allready tried different examples I found, but no luck...
> 
> Clemens
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iEYEARECAAYFAk1jgcAACgkQ4tZeNddg3dx9ZgCfZQ6SgFv87Jag5SVaXX5E3QvV
jkAAoK0prDieLE2onX2t3QfJk06BsjEs
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From rajesh.npnr at yahoo.com  Tue Feb 22 14:53:45 2011
From: rajesh.npnr at yahoo.com (rex.alex)
Date: Tue, 22 Feb 2011 03:53:45 -0800 (PST)
Subject: [Freeswitch-users] The call dropped prematurely error on txfax
In-Reply-To: <AANLkTim4Vsy3F-mBRnFuHK4oPvpQyzPB6CLy9akjxREG@mail.gmail.com>
References: <1297954260038-6036197.post@n2.nabble.com>
	<140D740BF92640F7A514BD35BD4BD1AA@e1705>
	<1297958857213-6036503.post@n2.nabble.com>
	<C0AB523AE4774FA5837E88F48243F970@e1705>
	<1297961826262-6036700.post@n2.nabble.com>
	<AANLkTikmEvZB5tF6qiT2Jf4Ca6VzBB8bqQ_k=O59=eO=@mail.gmail.com>
	<1298030300075-6039655.post@n2.nabble.com>
	<AANLkTinTX+vJj8qibEVsfyrSMVK=jPHbYksPkD20O=Z_@mail.gmail.com>
	<1298300457397-6048943.post@n2.nabble.com>
	<AANLkTim4Vsy3F-mBRnFuHK4oPvpQyzPB6CLy9akjxREG@mail.gmail.com>
Message-ID: <1298375625958-6052070.post@n2.nabble.com>


Hi,

Is it an issue from receiving end or from my gateway end? 

If it is from my gateway end, then could you please prefer some device which
gives at least 90% success in faxing as the current system just gives 30%
success in faxing and rest are failing with the  "The call dropped
prematurely" error.

Thanks,
Rex
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/The-call-dropped-prematurely-error-on-txfax-tp6032194p6052070.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From ce at kapper.net  Tue Feb 22 15:31:49 2011
From: ce at kapper.net (Clemens Ebentheuer)
Date: Tue, 22 Feb 2011 13:31:49 +0100
Subject: [Freeswitch-users] t38_gateway self - no re-Invite?
In-Reply-To: <4D6381C0.7040408@ewetel.de>
References: <1B19ABD72889C245AE8EEE08AC24103A28C1AFC00E@exmachina.office.kapper.net>
	<4D6381C0.7040408@ewetel.de>
Message-ID: <1B19ABD72889C245AE8EEE08AC24103A28C1AFC015@exmachina.office.kapper.net>

Thx Helmut,

I?ve allready tried this:
> switch_ivr_tone_detect_session(session, "t38", "2100", "rw", timeout,
> 1,
> data, NULL, t38_gateway_start)
> 
> The recompile mod_spandsp and reload it into FS.

But no luck, I?ll try different frequencies.

Is it because FS doesn?t recognize it as a t38 fax? If so, I?m wondering why rxfax in t38 mode is working fine then - with re-invite.
Any other ideas?

Clemens



From dmitry.bely at gmail.com  Tue Feb 22 15:58:52 2011
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Tue, 22 Feb 2011 15:58:52 +0300
Subject: [Freeswitch-users] Auto-nat affects all profiles?
Message-ID: <AANLkTi=uw63jzwqvWwnpBwVgbDm8CO9CzTgg8K2oa+nF@mail.gmail.com>

I have enabled auto-nat only for the external profile (port 5080), but
FreeSWITCH also maps SIP port for the internal one:

freeswitch at internal> show nat_map
port,proto,proto_num,sticky
5060,udp,0,0
5060,tcp,1,0
5080,udp,0,0
5080,tcp,1,0

4 total.

Is this intended or just a bug?

- Dmitry Bely


From wstephen80 at gmail.com  Tue Feb 22 17:18:15 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 22 Feb 2011 15:18:15 +0100
Subject: [Freeswitch-users] ESL socket outbound: the execution of answer
 has not effect
In-Reply-To: <AANLkTim5AQSRCij70F1fmsyEzV2jphXnTqP6YA0Y6+X3@mail.gmail.com>
References: <AANLkTimY7t8iBSZUGuCLx2f_e=3R+taGfW4s-NaT2Evg@mail.gmail.com>
	<AANLkTim5AQSRCij70F1fmsyEzV2jphXnTqP6YA0Y6+X3@mail.gmail.com>
Message-ID: <AANLkTi=HJOuMbZO-z6X9CLtT84kgx_c-5Zjr-QyHU31G@mail.gmail.com>

Than you Antony,
waiting the completition of "pre_anser" before bridging
(CHANNEL_PROGRESSING_MEDIA event from legA) and with a correct uuid order in
the "uuid_bridge" all works fine.

Stephen

On Tue, Feb 22, 2011 at 3:42 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> If you bridge them, the bridge app will already pass the  answer for
> you when the B leg is answered.
> If you are seeing that message it means you are supplying the uuid in
> reverse order where the A leg is not the one with media.
>
> That reversing the order message can only be true if you pass a leg
> with media as the 2nd leg and one without as the 1st to uuid_bridge.
>
> When you pre_answer the A leg, do you wait for it to actually happen
> before you call bridge?
> sendmsg execute only queues the app to be executed, you need to get
> the event confirming it actually happened before you can be sure it
> did.
>
> Try adding the header
>
> async: true
>
> to your sendmsg to indicate you want to execute the app instantly and
> wait for it to finish.
>
>
>
> On Mon, Feb 21, 2011 at 5:39 PM, Stephen Wilde <wstephen80 at gmail.com>
> wrote:
> > I'm running an esl socket outbound application and I have a systematic
> > problem: in a particular situation, the execution of answer on incoming
> call
> > has no effect.
> > This happens always when (and only in this case):
> > 1. an incoming call is received (legA) and via dialplan I send the call
> to
> > my socket outbound application
> > 2. I make an outbound call with originate (legB)
> > 3. I receive a CHANNEL_PROGRESS_MEDIA on legB
> > 4. I do a "pre_answer" on legA
> > 5, I do a "uuid_bridge" between legA and legB
> > in this phase, legA can hear the progress tone coming from legB (an
> in-band
> > ringback)
> > 5. legB answer the call and I receive a CHANNEL_ANSWER on legB
> > 6. I do an execute of "answer" on legA
> > This "answer" has no effect!
> > I do both pre_answer and answer specifying the UUID parameter (sendmsg
> > <uuid>), obviously using legA uuid .
> > I do the bridge specifying inbound uuid as first parameter and outbound
> uuid
> > as second parameter: uuid_bridge <inbound_uuid> <outbound_uuid>
> > In fs_cli I see:  [WARNING] switch_ivr_bridge.c:1412 reversing order of
> > channels so this will work!
> > To me it seems that in uuid_bridge is done an inversion with my
> > inbound/outbound channels and because I do an answer to the incoming
> uuid,
> > probably (due to the inversion) is treated as outbound so the answer is
> > ignored.
> > Attached to this email the .pcap file related to tcp traffic between
> > Freeswitch and my socket outbound application where at time 8.562639
> there
> > is my sendmsg with the answer.
> > 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
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 helmut.kuper at ewetel.de  Tue Feb 22 17:26:24 2011
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 22 Feb 2011 15:26:24 +0100
Subject: [Freeswitch-users] t38_gateway self - no re-Invite?
In-Reply-To: <1B19ABD72889C245AE8EEE08AC24103A28C1AFC015@exmachina.office.kapper.net>
References: <1B19ABD72889C245AE8EEE08AC24103A28C1AFC00E@exmachina.office.kapper.net>	<4D6381C0.7040408@ewetel.de>
	<1B19ABD72889C245AE8EEE08AC24103A28C1AFC015@exmachina.office.kapper.net>
Message-ID: <4D63C790.7000306@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Clemens,

Am 22.02.2011 13:31, schrieb Clemens Ebentheuer:
> Thx Helmut,
> 
> I?ve allready tried this:
>> switch_ivr_tone_detect_session(session, "t38", "2100", "rw", timeout,
>> 1,
>> data, NULL, t38_gateway_start)
>>
>> The recompile mod_spandsp and reload it into FS.
> 
> But no luck, I?ll try different frequencies.
> 
> Is it because FS doesn?t recognize it as a t38 fax? 

Yes, no ReINVITE from FS means no detection of CNG or CED.

AFAIK t38 works like this:
a) establish a call to FAX
b) exchange CNG and CED via G.711
c) Listen to CNG and/or CED and when present, switch to t.38 via ReINVITE

More about FAX process is here: http://telecom.tbi.net/fax-call.htm

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.

Just trace the sending FAX-Device for SIP-Signalling and you should se a
ReINVITE from FAX-device to FS, if this is true.

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.

regards
helmut
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From saeedahmad1981 at gmail.com  Tue Feb 22 20:33:03 2011
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Tue, 22 Feb 2011 18:33:03 +0100
Subject: [Freeswitch-users] MOH not working in G729 passthru with native
 sound files
In-Reply-To: <AANLkTimaCpu6+SEOppgXpUzzBVoaFm9+70=PyXbEKjG4@mail.gmail.com>
References: <AANLkTi=41CbE_tU8LsW52rZCaD0Z+Y8NbE7WXh+bxTn1@mail.gmail.com>
	<AANLkTim2=nnHfXOPjMZCeJ1bmLOLYPSi0zspky9smZeD@mail.gmail.com>
	<AANLkTimaCpu6+SEOppgXpUzzBVoaFm9+70=PyXbEKjG4@mail.gmail.com>
Message-ID: <AANLkTimD0xcvc7TsWQT=6cEHdV+tnPAZq7eKkSdwRRK8@mail.gmail.com>

Thanks Steve

On Mon, Feb 21, 2011 at 3:55 PM, Steven Ayre <steveayre at gmail.com> wrote:

>
> http://wiki.freeswitch.org/wiki/Mod_native_file#Script_to_convert_a_sound_file_to_specific_formats_to_avoid_transcoding
>
>
>
>
> On 21 February 2011 12:49, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
>
>> how did you record in G729?
>>
>> On Thu, Feb 10, 2011 at 3:41 AM, Marcin Wojtowicz <marcin321 at gmail.com>wrote:
>>
>>> I'm trying to enable MOH when both legs of the call are using G729 (FS is
>>> in passthru). I converted an edited sample wave file to G729 and put in the
>>> appropriate folder, and FS loads it correctly because this is the message
>>> that keeps popping up in the console:
>>> 2011-02-09 21:10:25.609375 [INFO] mod_native_file.c:94 Opening File
>>> [C:\freeswitch\sounds/music/8000\sample.G729] 8000hz
>>>
>>> I establish a call, and everything is fine, but when I press hold on my
>>> handset I see an error message that says that G729 is only useable in
>>> passthru (here is the debug message):
>>>
>>> 2011-02-09 21:14:01.828125 [DEBUG] switch_channel.c:1376 (sofia/internal/
>>> sip:M9jdt73ig0oOJSbt6Uyy at 192.168.1.50:5060) Callstate Change ACTIVE ->
>>> HELD
>>> 2011-02-09 21:14:01.828125 [DEBUG] switch_core_session.c:954 Send signal
>>> sofia/external/MYHOME#@74.63.41.218 [BREAK]
>>> 2011-02-09 21:14:01.859375 [DEBUG] switch_core_session.c:709 Send signal
>>> sofia/external/MYHOME#@74.63.41.218 [BREAK]
>>> 2011-02-09 21:14:02.000000 [DEBUG] switch_ivr.c:563
>>> sofia/external/MYHOME#@74.63.41.218 Command Execute
>>> playback(local_stream://moh/8000)
>>> EXECUTE sofia/external/MYHOME#@74.63.41.218playback(local_stream://moh/8000)
>>> 2011-02-09 21:14:02.000000 [DEBUG] mod_local_stream.c:421 Opening Stream
>>> [moh/8000] 8000hz
>>> 2011-02-09 21:14:02.000000 [DEBUG] switch_ivr_play_say.c:1244 Codec
>>> Activated L16 at 8000hz 1 channels 20ms
>>> 2011-02-09 21:14:02.015625 [ERR] mod_g729.c:102 This codec is only usable
>>> in passthrough mode!
>>> 2011-02-09 21:14:02.015625 [ERR] switch_core_io.c:1042 Codec G.729
>>> encoder error!
>>> 2011-02-09 21:14:02.015625 [DEBUG] switch_ivr_play_say.c:1581 done
>>> playing file
>>>
>>> I don't understand why that would be, since my music file is in G729 so
>>> I'm not asking freeswitch to convert, only stream. My custom ringback
>>> (before a call is established) works just fine using a similar method, so
>>> could anyone explain me why what I want to do is not permitted?
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mthakershi at gmail.com  Tue Feb 22 21:25:45 2011
From: mthakershi at gmail.com (Malay Thakershi)
Date: Tue, 22 Feb 2011 12:25:45 -0600
Subject: [Freeswitch-users] Using 16 KHz sounds
In-Reply-To: <AANLkTin9tYDVJC0KUncV0h=g0+u=833QcRtp49OfU2XC@mail.gmail.com>
References: <AANLkTim5CsCL4unpWh1OvH13Xx4HrKX2KF7Jg6UQYAZR@mail.gmail.com>
	<AANLkTin9tYDVJC0KUncV0h=g0+u=833QcRtp49OfU2XC@mail.gmail.com>
Message-ID: <AANLkTi=YkDwu5ANjYFFWGDR2pmuscsJUDcK09QL92Kqo@mail.gmail.com>

I found I am already using 16 KHz profile. .SetTtsParameters("cepstral",
"Allison-16kHz");

I read this under FS wiki on Cepstral under 'Gotchas':
-------------
Using a 16khz voice and 0.03 for RTP Packet Size (Sipura Setting), it will
sound horrible. Workaround: modify RTP Packet Size to 0.02 in Sipura config,
under Advanced/SIP section.
-------------

Do you think that is my problem? Is this to be done in FS configuration?

Malay

On Mon, Feb 21, 2011 at 2:35 PM, Michael Collins <msc at freeswitch.org> wrote:

> 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
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From k-b-24 at live.com  Tue Feb 22 22:35:55 2011
From: k-b-24 at live.com (Jason b.a)
Date: Tue, 22 Feb 2011 19:35:55 +0000
Subject: [Freeswitch-users] OpenSER interface to FreeSWITCH
Message-ID: <COL111-W101838B312AE655465A34CACD80@phx.gbl>


Hi,
 
is it possible to implements is an interface to OpenSER over a TCP connection, so the design will be : 
 
Caller----IVR application----OpenSER-----Freeswitch
                   |
                   |
               Callee
 
without the need to use the Sip sophia module, only openSER control the freeswitch for media processing using the socket interface.
 
Thx  		 	   		  
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From jjj at 3js.de  Tue Feb 22 22:30:08 2011
From: jjj at 3js.de (Johannes Jakob)
Date: Tue, 22 Feb 2011 20:30:08 +0100
Subject: [Freeswitch-users] Problems getting asterisk registered with FS sbc
Message-ID: <D4534174-491C-45D2-A3BF-E29EDE85DF2A@3js.de>

Fellow FreeSWITCH Admins,

I'm having a hard time, getting a Trixbox 2.8 box to register with our FreeSWITCH SBCs.

The FreeSWITCHes are running FreeSWITCH-mod_sofia/1.0.head-git-7847289, the asterisk on the trixbox is Asterisk 1.6.0.22-samy-r60.


The user's directory entry looks like this:


<include>
  <user id="748732" cidr="10.16.153.163/32">
    <params>
<!--
      <param name="password" value="Idsd67Hsa"/>
-->
      <param name="t38-passthru" value="true"/>
    </params>
    <variables>
      <variable name="client_asserted_identity" value="03023671836"/>
      <variable name="accountcode" value="748732"/>
      <variable name="contract" value="2253"/>
      <variable name="nibble_rate" value="1"/>
      <variable name="nibble_account" value="2253"/>

      <variable name="proxy_media" value="true"/>
      <variable name="bypass_media" value="false"/>
      <variable name="user_context" value="mysip.net"/>
      <variable name="default_language" value="de"/>
      <variable name="language" value="de"/>
    </variables>
  </user>
</include>


Asterisk's register string: 748732 at mysip.net@sbc1.mysip.net/748732


I'm getting the "normal" FS errors for wrong credentials:

2011-02-22 18:03:57.484939 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [748732 at mysip.net] from ip 10.16.153.163
2011-02-22 18:03:57.491471 [WARNING] sofia_reg.c:1204 SIP auth failure (REGISTER) on sofia profile 'internal' for [748732 at mysip.net] from ip 10.16.153.163


but why am I getting these? I specified the right address in the cidr statement! Why is it even bothering with anything else but the right user at domain and IP-address?


There are some other asterisk boxes (> 1.8.2) registering to this SBC with equal settings just fine, what's wrong with this little trixbox system? ;)



Of course I did get you some SIP traces as well:


18:00:37.063410 IP 10.16.153.163.5060 > 10.16.133.66.5060: UDP, length: 419
E`..f...>.7.^...^..B.......-REGISTER sip:mysip.net SIP/2.0
Via: SIP/2.0/UDP 10.16.153.163:5060;branch=z9hG4bK3e70680b;rport
Max-Forwards: 70
From: <sip:748732 at mysip.net>;tag=as77c8852d
To: <sip:748732 at mysip.net>
Call-ID: 53d04cc277cfe60301bddb6d79033420 at 10.16.153.163
CSeq: 102 REGISTER
User-Agent: Asterisk PBX 1.6.0.22-samy-r60
Expires: 1800
Contact: <sip:748732 at 10.16.153.163>
Event: registration
Content-Length: 0



18:00:37.074085 IP 10.16.133.66.5060 > 10.16.153.163.5060: UDP, length: 657
E...F...?.Vc^..B^...........SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.16.153.163:5060;branch=z9hG4bK3e70680b;rport=5060
From: <sip:748732 at mysip.net>;tag=as77c8852d
To: <sip:748732 at mysip.net>;tag=5jD9Qcg3N9S6p
Call-ID: 53d04cc277cfe60301bddb6d79033420 at 10.16.153.163
CSeq: 102 REGISTER
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
WWW-Authenticate: Digest realm="mysip.net", nonce="ce2bccbf-a27b-43c8-b7b0-a89ab429d8a7", algorithm=MD5, qop="auth"
Content-Length: 0



18:00:37.074969 IP 10.16.153.163.5060 > 10.16.133.66.5060: UDP, length: 672
E`..f...>.6.^...^..B........REGISTER sip:mysip.net SIP/2.0
Via: SIP/2.0/UDP 10.16.153.163:5060;branch=z9hG4bK30df5010;rport
Max-Forwards: 70
From: <sip:748732 at mysip.net>;tag=as03431ba4
To: <sip:748732 at mysip.net>
Call-ID: 53d04cc277cfe60301bddb6d79033420 at 10.16.153.163
CSeq: 103 REGISTER
User-Agent: Asterisk PBX 1.6.0.22-samy-r60
Authorization: Digest username="748732 at mysip.net", realm="mysip.net", algorithm=MD5, uri="sip:mysip.net", nonce="ce2bccbf-a27b-43c8-b7b0-a89ab429d8a7", response="133a0ba843fe9f5afba67d1377fa8c11", qop=auth, cnonce="119cf18c", nc=00000001
Expires: 1800
Contact: <sip:748732 at 10.16.153.163>
Event: registration
Content-Length: 0


18:00:37.081517 IP 10.16.133.66.5060 > 10.16.153.163.5060: UDP, length: 532
E..0F...?.V.^..B^.........1.SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.16.153.163:5060;branch=z9hG4bK30df5010;rport=5060
From: <sip:748732 at mysip.net>;tag=as03431ba4
To: <sip:748732 at mysip.net>;tag=6U61S706jjgSj
Call-ID: 53d04cc277cfe60301bddb6d79033420 at 10.16.153.163
CSeq: 103 REGISTER
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-Length: 0


Can somebody point me in the right direction?


Thanks and best regards,

  John



From msc at freeswitch.org  Tue Feb 22 22:46:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Feb 2011 11:46:36 -0800
Subject: [Freeswitch-users] OpenSER interface to FreeSWITCH
In-Reply-To: <COL111-W101838B312AE655465A34CACD80@phx.gbl>
References: <COL111-W101838B312AE655465A34CACD80@phx.gbl>
Message-ID: <AANLkTin4Z5SMfwnshuWSMu3iZ9WBfoRxHjrFEBire1Q=@mail.gmail.com>

I doubt you'd want to do this - you'd be adding yet another layer of
complexity. I suppose the only benefit is that having a proxy (OpenSER) in
front of FreeSWITCH means you could have multiple FS servers for redundancy.
In your case, though, I don't know how valuable that is.

Can you remind us again of what you're building? I think we're all having a
hard time understanding why you need so many pieces. If you could draw the
audio and signaling paths as well that would really help us understand.

-MC

On Tue, Feb 22, 2011 at 11:35 AM, Jason b.a <k-b-24 at live.com> wrote:

>  Hi,
>
> is it possible to implements is an interface to OpenSER over a TCP
> connection, so the design will be :
>
> Caller----IVR application----OpenSER-----Freeswitch
>                    |
>                    |
>                Callee
>
> without the need to use the Sip sophia module, only openSER control the
> freeswitch for media processing using the socket interface.
>
> Thx
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bernhard.suttner at winet.ch  Tue Feb 22 22:47:57 2011
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Tue, 22 Feb 2011 20:47:57 +0100
Subject: [Freeswitch-users] OpenSER interface to FreeSWITCH
Message-ID: <20110222204757.096c2598@mail.winet.ch>


currently not but might be a cool idea ...


----- Original Message -----
From: Jason b.a [mailto:k-b-24 at live.com]
To: freeswitch-users at lists.freeswitch.org
Sent: Tue, 22 Feb 2011 20:35:55 +0100
Subject: [Freeswitch-users] OpenSER interface to FreeSWITCH


> 
> Hi,
>  
> is it possible to implements is an interface to OpenSER over a TCP
> connection, so the design will be : 
>  
> Caller----IVR application----OpenSER-----Freeswitch
>                    |
>                    |
>                Callee
>  
> without the need to use the Sip sophia module, only openSER control the
> freeswitch for media processing using the socket interface.
>  
> Thx  		 	   		  


From msc at freeswitch.org  Tue Feb 22 22:57:34 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Feb 2011 11:57:34 -0800
Subject: [Freeswitch-users] javascript hanguphook
In-Reply-To: <4D61B204.4060102@yellox.de>
References: <4D5AA03F.8030307@yellox.de>
	<CDF5E00590C9BA4AB96BE7CDD28C86AE07873018D3@FS1.wvds.local>
	<4D61B204.4060102@yellox.de>
Message-ID: <AANLkTikmzVDWmaOWP-56QAUSXWp7SXxAPFGQgTOz0ZAR@mail.gmail.com>

On Sun, Feb 20, 2011 at 4:29 PM, Christian Hiller <christian at yellox.de>wrote:

> hello Erik,
>
> i have set
>
> <action application="set" data="my_var=123"/>
> <action application="set" data="session_in_hangup_hook=true"/>
> <action application="set" data="api_hangup_hook=jsrun cleanup.js
> ${my_var}"/>
> <action application="javascript" data="routing.js"/>
>
> The channel_variable ${my_var} gets updated in routing.js but whenever
> cleanup.js is executed, then ${my_var} still has the old value of 123.
> Why that?
>
>
At the time that the api_hangup_hook=jsrun cleanup.js ${my_var} action is
parsed, the value of ${my_var} is "123" so that is what is set. The good
news is that if you have session_in_hangup_hook set to true you should be
able to see the updated value of my_var. I'm afraid I don't see any wiki
examples of how to do this in Javascript. Does anyone have a code sample of
how to access the channel variables in a js hangup hook when
session_in_hangup_hook is true?

For reference, the Lua method is shown here:
http://wiki.freeswitch.org/wiki/Lua#Special_Case:_env_object

-MC
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From k-b-24 at live.com  Wed Feb 23 00:40:51 2011
From: k-b-24 at live.com (Jason b.a)
Date: Tue, 22 Feb 2011 21:40:51 +0000
Subject: [Freeswitch-users] OpenSER interface to FreeSWITCH
Message-ID: <COL111-W3613A97BE7ABA842EA27E0ACD80@phx.gbl>


1- i need an IVR application separated from the media server
2- the application server should control the media server using MSCML , VXML or any control protocol
 
i know it is easier to use the embedded IVR in freeswitch, but this is not my case.
 
Beside by using openSER between the application and the media server ==> openSER can interface with Freeswitch through socket interface,to control it ! 
 
i dont know if the sip sophia interface can handle the request from the application , and pass it to the core for processing, and generate a responce and
pass it through the same interface, so here no need for OpenSER
 
if we took this example:
1-Caller call IVR application 
2-Application as media server to play welcome message and to collect digit
3-caller enter digits
4-media server pass the digits to the application
5-application bridge caller and callee together  		 	   		  
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From brian at freeswitch.org  Wed Feb 23 01:01:15 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 22 Feb 2011 16:01:15 -0600
Subject: [Freeswitch-users] Auto-nat affects all profiles?
In-Reply-To: <AANLkTi=uw63jzwqvWwnpBwVgbDm8CO9CzTgg8K2oa+nF@mail.gmail.com>
References: <AANLkTi=uw63jzwqvWwnpBwVgbDm8CO9CzTgg8K2oa+nF@mail.gmail.com>
Message-ID: <B43954E4-8AF0-48F3-A04B-BAA0F8D7E1F0@freeswitch.org>

only works on one profile.  Not designed to work with more than one.

/b

On Feb 22, 2011, at 6:58 AM, Dmitry Bely wrote:

> I have enabled auto-nat only for the external profile (port 5080), but
> FreeSWITCH also maps SIP port for the internal one:
> 
> freeswitch at internal> show nat_map
> port,proto,proto_num,sticky
> 5060,udp,0,0
> 5060,tcp,1,0
> 5080,udp,0,0
> 5080,tcp,1,0
> 
> 4 total.
> 
> Is this intended or just a bug?
> 
> - Dmitry Bely



From steveayre at gmail.com  Wed Feb 23 02:03:45 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 22 Feb 2011 23:03:45 +0000
Subject: [Freeswitch-users] OpenSER interface to FreeSWITCH
In-Reply-To: <COL111-W101838B312AE655465A34CACD80@phx.gbl>
References: <COL111-W101838B312AE655465A34CACD80@phx.gbl>
Message-ID: <AANLkTimmWknkNLS2aXQEenGOg+79r54ELED8pzUuLN_V@mail.gmail.com>

You need a protocol to control the media processing though... that's what
SIP gives you.

-Steve


On 22 February 2011 19:35, Jason b.a <k-b-24 at live.com> wrote:

>  Hi,
>
> is it possible to implements is an interface to OpenSER over a TCP
> connection, so the design will be :
>
> Caller----IVR application----OpenSER-----Freeswitch
>                    |
>                    |
>                Callee
>
> without the need to use the Sip sophia module, only openSER control the
> freeswitch for media processing using the socket interface.
>
> Thx
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Feb 23 02:21:45 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 22 Feb 2011 18:21:45 -0500
Subject: [Freeswitch-users] auth-all-packets
Message-ID: <DB3816F6776C4FA6A28840E376B0500B@e1705>

Is auth-all-packets to true decrease performance ?

Thanks
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From infos at madovsky.org  Wed Feb 23 02:26:33 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 22 Feb 2011 18:26:33 -0500
Subject: [Freeswitch-users] risky hangup
Message-ID: <41B940C7439741069AFFEF360F06744B@e1705>

on last git I noticed that every hangup
after TTS is made with a dfferent delay.
sometimes it cuts the voice, sometimes it cuts suddenly before the voice starts
and sometimes it cuts the voice 2/3 seconds before.
after 10 tests all hangups are done with a different delay...
please advice

thanks
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From richocet2 at hotmail.com  Wed Feb 23 00:56:08 2011
From: richocet2 at hotmail.com (Dave Bracken)
Date: Tue, 22 Feb 2011 15:56:08 -0600
Subject: [Freeswitch-users] ESL in Delphi/Pascal
Message-ID: <SNT144-w572265F9927858BC1E69EFF0D80@phx.gbl>


Can anyone help me to figure out how to use the ESL in Delphi? We have tried compiling the files in VS2008, and it produces a .lib file and some .obj files, but none of which can be linked into delphi. I do not know hardly anything about C, so i am kinda of flying blind here. Any help would be very greatly appreciated.
DelphiGuy 		 	   		  
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From msc at freeswitch.org  Wed Feb 23 02:42:32 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Feb 2011 15:42:32 -0800
Subject: [Freeswitch-users] risky hangup
In-Reply-To: <41B940C7439741069AFFEF360F06744B@e1705>
References: <41B940C7439741069AFFEF360F06744B@e1705>
Message-ID: <AANLkTimnJ2MOXGVWKSQBZckfQnimT3CT=E03aZ4+Hfhy@mail.gmail.com>

please give some context... how are you testing this? What is calling what,
etc.?
-MC

On Tue, Feb 22, 2011 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:

>  on last git I noticed that every hangup
> after TTS is made with a dfferent delay.
> sometimes it cuts the voice, sometimes it cuts suddenly before the voice
> starts
> and sometimes it cuts the voice 2/3 seconds before.
> after 10 tests all hangups are done with a different delay...
> please advice
>
> thanks
>
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From infos at madovsky.org  Wed Feb 23 03:06:35 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 22 Feb 2011 19:06:35 -0500
Subject: [Freeswitch-users] risky hangup
References: <41B940C7439741069AFFEF360F06744B@e1705>
	<AANLkTimnJ2MOXGVWKSQBZckfQnimT3CT=E03aZ4+Hfhy@mail.gmail.com>
Message-ID: <3CD6B60400ED4026BBD40CE29489339E@e1705>

Example:
I make an external call without credits
so the call is transferred (by nibblebill)
to a "nofunds" extension with plaback wav file
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, February 22, 2011 6:42 PM
  Subject: Re: [Freeswitch-users] risky hangup


  please give some context... how are you testing this? What is calling what, etc.?
  -MC



  On Tue, Feb 22, 2011 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:

    on last git I noticed that every hangup
    after TTS is made with a dfferent delay.
    sometimes it cuts the voice, sometimes it cuts suddenly before the voice starts
    and sometimes it cuts the voice 2/3 seconds before.
    after 10 tests all hangups are done with a different delay...
    please advice

    thanks


------------------------------------------------------------------------------


  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
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  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From msc at freeswitch.org  Wed Feb 23 03:06:28 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Feb 2011 16:06:28 -0800
Subject: [Freeswitch-users] OpenSER interface to FreeSWITCH
In-Reply-To: <COL111-W3613A97BE7ABA842EA27E0ACD80@phx.gbl>
References: <COL111-W3613A97BE7ABA842EA27E0ACD80@phx.gbl>
Message-ID: <AANLkTimwWw3d9N50s2Duf8PpWaqNQwwiBbr0HZgHjs+7@mail.gmail.com>

Comments inline...

On Tue, Feb 22, 2011 at 1:40 PM, Jason b.a <k-b-24 at live.com> wrote:

>  1- i need an IVR application separated from the media server
> 2- the application server should control the media server using MSCML ,
> VXML or any control protocol
>
> i know it is easier to use the embedded IVR in freeswitch, but this is not
> my case.
>


>
>
This is problem is screaming "event socket" all over


> Beside by using openSER between the application and the media server ==>
> openSER can interface with Freeswitch through socket interface,to control it
> !
>

Yes, but is there any reason YOUR application can't talk to FreeSWITCH over
the event socket?


> i dont know if the sip sophia interface can handle the request from the
> application , and pass it to the core for processing, and generate a
> responce and
> pass it through the same interface, so here no need for OpenSER
>

Again, the event socket lets you have this kind of control.


>
> if we took this example:
> 1-Caller call IVR application
>
When you say that the caller calls your IVR application, what exactly does
that mean? Do you do SIP handling in your application? Do NEED to do SIP
handling? You could have the call terminate and FreeSWITCH and let
FreeSWITCH do an outbound event socket connection to your program - at that
point your program has complete control of this call - you can play TTS,
collect digits, transfer caller elsewhere and hangup. You can even create an
outbound call leg to connect to the callee and do all sorts of crazy things.


> 2-Application as media server to play welcome message and to collect digit
> 3-caller enter digits
> 4-media server pass the digits to the application
> 5-application bridge caller and callee together
>

I recommend you look at the outbound event socket on the wiki. Also, look at
the fs_ivrd program. It's a handy way to organize calls to a socket-based
application.

-MC
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From msc at freeswitch.org  Wed Feb 23 03:09:52 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Feb 2011 16:09:52 -0800
Subject: [Freeswitch-users] RTP Proxy config and DTMF conversion problem
In-Reply-To: <AANLkTimW23Uf61OU2RVJWD6HAM9i_towb2GocwYsi7aC@mail.gmail.com>
References: <AANLkTimW23Uf61OU2RVJWD6HAM9i_towb2GocwYsi7aC@mail.gmail.com>
Message-ID: <AANLkTimSkWWvVvjKsu1_kSOfYRxR9RQCg4vHq5tuUnKf@mail.gmail.com>

So are you trying to detect in-band dtmf on one leg and convert it to 2833
going to the other leg?
-MC

On Sun, Feb 20, 2011 at 9:46 AM, Mac <validator77 at gmail.com> wrote:

> Dear ALL,
>
> Thats my first post here. I am struggling for some time with DTMF issue.
> Let me introduce you my configuration.
>
> The main task is to configure RTP Proxy with full topology hiding -
> OPERATOR_1 ------- Freeswitch ------- OPERATOR_2
> Here is a prt of my config:
> - sip profile.
> I have one new sip profile with two gateways (OPERATOR_1 and OPERATOR_2)
> <include>
>   <gateway name="OP1">
>     <param name="username" value="X1.Y1.V1.Z1"/>
>     <param name="password" value="09321234"/>
>     <param name="proxy" value="X1.Y1.V1.Z1"/>
>     <param name="from-domain" value="test.com"/>
>     <param name="register" value="false"/>
>     <param name="caller-id-in-from" value="true"/>
>     <param name="extension" value="test"/>
>     <param name="extension-in-contact" value="true"/>
>     <param name="outbound-codec-prefs" value="$${outbound_codec_
> prefs}"/>
>     <param name="disable-transcoding" value="true"/>
>     <param name="inbound-late-negotiation" value="true"/>
>   </gateway>
> </include>
>
> - dialplan
> <include>
>   <context name="public">
>         <extension name="term">
>                 <condition field="network_addr"
> expression="^(X1\.Y1\.V1\.Z1|X2\.Y2\.V1\.Z1)$" break="on-false"/>
>                 <condition field="destination_number"
> expression="^46(\d{9})$">
>                 <action application="set" data="proxy_media=true"/>
>                 <action application="set" data="bypass_media=false"/>
>                 <action application="start_dtmf"/>
>                 <action application="bridge" data="sofia/gateway/OP1/$0"/>
>                 </condition>
>         </extension>
>   </context>
> </include>
>
> Everything is fine, but i have problem with DTMF conversion from RFC2833 to
> inband.
> Refering to
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf and
> http://wiki.freeswitch.org/wiki/Proxy_Media i have done all the necessary
> things.
>
> vars.xml
>   <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,G.729,PCMU,GSM"/>
>   <X-PRE-PROCESS cmd="set"
> data="outbound_codec_prefs=PCMA,G.729,PCMU,GSM"/>
>
> The after-effect is following output:
> [ERR] switch_core_io.c:1055 Codec PROXY PASS-THROUGH encoder error!
>
> Placing  <param name="dtmf-type" value="none"/> does not help anyway.
>
> Could sb point me where the problem is located?
>
> Thanks in advance,
> Mac
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 23 03:16:56 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 22 Feb 2011 18:16:56 -0600
Subject: [Freeswitch-users] RTP Proxy config and DTMF conversion problem
In-Reply-To: <AANLkTimW23Uf61OU2RVJWD6HAM9i_towb2GocwYsi7aC@mail.gmail.com>
References: <AANLkTimW23Uf61OU2RVJWD6HAM9i_towb2GocwYsi7aC@mail.gmail.com>
Message-ID: <AANLkTikyYbkmUOCUOk90tb=Fq0p_BewoAK4dtzR5L-fa@mail.gmail.com>

just because you want to make a proxy and there is a mode with the
name proxy in it, you do not have to use it.
proxy_media mode is not going to work with inband detection because
then you are looking at the media and not really proxying it.



On Sun, Feb 20, 2011 at 11:46 AM, Mac <validator77 at gmail.com> wrote:
> Dear ALL,
>
> Thats my first post here. I am struggling for some time with DTMF issue.
> Let me introduce you my configuration.
>
> The main task is to configure RTP Proxy with full topology hiding -
> OPERATOR_1 ------- Freeswitch ------- OPERATOR_2
> Here is a prt of my config:
> - sip profile.
> I have one new sip profile with two gateways (OPERATOR_1 and OPERATOR_2)
> <include>
> ? <gateway name="OP1">
> ??? <param name="username" value="X1.Y1.V1.Z1"/>
> ??? <param name="password" value="09321234"/>
> ??? <param name="proxy" value="X1.Y1.V1.Z1"/>
> ??? <param name="from-domain" value="test.com"/>
> ??? <param name="register" value="false"/>
> ??? <param name="caller-id-in-from" value="true"/>
> ??? <param name="extension" value="test"/>
> ??? <param name="extension-in-contact" value="true"/>
> ??? <param name="outbound-codec-prefs" value="$${outbound_codec_
> prefs}"/>
> ??? <param name="disable-transcoding" value="true"/>
> ??? <param name="inbound-late-negotiation" value="true"/>
> ? </gateway>
> </include>
>
> - dialplan
> <include>
> ? <context name="public">
> ??????? <extension name="term">
> ??????????????? <condition field="network_addr"
> expression="^(X1\.Y1\.V1\.Z1|X2\.Y2\.V1\.Z1)$" break="on-false"/>
> ??????????????? <condition field="destination_number"
> expression="^46(\d{9})$">
> ??????????????? <action application="set" data="proxy_media=true"/>
> ??????????????? <action application="set" data="bypass_media=false"/>
> ??????????????? <action application="start_dtmf"/>
> ??????????????? <action application="bridge" data="sofia/gateway/OP1/$0"/>
> ??????????????? </condition>
> ??????? </extension>
> ? </context>
> </include>
>
> Everything is fine, but i have problem with DTMF conversion from RFC2833 to
> inband.
> Refering to http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf
> and http://wiki.freeswitch.org/wiki/Proxy_Media i have done all the
> necessary things.
>
> vars.xml
> ? <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,G.729,PCMU,GSM"/>
> ? <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA,G.729,PCMU,GSM"/>
>
> The after-effect is following output:
> [ERR] switch_core_io.c:1055 Codec PROXY PASS-THROUGH encoder error!
>
> Placing? <param name="dtmf-type" value="none"/> does not help anyway.
>
> Could sb point me where the problem is located?
>
> Thanks in advance,
> Mac
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Wed Feb 23 03:21:53 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Feb 2011 16:21:53 -0800
Subject: [Freeswitch-users] FS gateway which registers into FS
In-Reply-To: <AANLkTi=fKGKAJJyAVzVaQwKh2y8cMPwrSWhcBu9g8HH=@mail.gmail.com>
References: <AANLkTi=fKGKAJJyAVzVaQwKh2y8cMPwrSWhcBu9g8HH=@mail.gmail.com>
Message-ID: <AANLkTin1Tabs=J+JersLG+EuOeKuybTyQGQdztXPFGeu@mail.gmail.com>

Could you re-draw your layout? I think my email reader munged the diagram so
it makes it hard to know exactly how everything is laid out.

As far as gateways are concerned, those are primarily for handling either
registrations to other servers or to answer auth challenges that come from
other servers. So if FreeSWITCH will register to Server A (or Server B) then
you need a gateway. You would also need a gateway if Server A (or B) will
send an auth challenge whenever FreeSWITCH sends a call to it. (Hope that
made sense.)

-MC

On Sun, Feb 20, 2011 at 8:32 AM, Shigeru Kanemoto <sgk at ppona.com> wrote:

> Hello,
>
> I am new to FreeSWITCH and having a headache configuring it. My
> experiences to Asterisk do not work completely. I appreciate your
> suggestion.
>
> I am trying to setup a central SIP server which accepts registers from
> subsidiary SIP servers which is behind NAT and has no fixed IP
> address. Each subsidiary servers have extension number ranges. I have
> to route such extension ranges to each SIP servers.
>
> FS central server
>   <--- (NAT) --- Server A (ext. 1000 - 1099)
>   <--- (NAT) --- Server B (ext. 1100 - 1199)
>   <--- (NAT) --- Phone ext. 1201
>   <--- (NAT) --- Phone ext. 1202
>
> According to the wiki page
> http://wiki.freeswitch.org/wiki/Clarification:gateways , I created an
> XML file under "conf/directory/default" for a gateway setting inside a
> <user> block. In my configuration, the gateway setting is parsed in
> the profile "internal".
>
> My current problem is the dialplan "bridge" application does not work
> to bridge to the destination like "sofia/gateway/server-a/1001". The
> call terminates after printing a log message like "[NOTICE]
> sofia.c:5082 Hangup sofia/internal/1001 [CS_CONSUME_MEDIA]
> [NORMAL_TEMPORARY_FAILURE]".
>
> What is my mistake? Is it wrong to use the "gateway" feature for such
> purpose?
>
> sgk
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 23 03:34:59 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Feb 2011 16:34:59 -0800
Subject: [Freeswitch-users] Easy IVR?
In-Reply-To: <AANLkTin5VqM7Xt7VU3Abs+X0Y3FqGQ9tNS3Y7aHgCOmq@mail.gmail.com>
References: <AANLkTin5VqM7Xt7VU3Abs+X0Y3FqGQ9tNS3Y7aHgCOmq@mail.gmail.com>
Message-ID: <AANLkTimZcETCP4L8h1v2G8mzeYsiJsQjH2NFtnmmBVMN@mail.gmail.com>

It is if you are a Rails person. :)
-MC

On Sun, Feb 20, 2011 at 7:05 PM, EdPimentl <edpimentl at gmail.com> wrote:

> Is this not a good example of how easy IVR design/creation should be?
> http://asymmetrical-view.com/2011/02/20/twilio-in-ten-minutes.html
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 23 03:37:28 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Feb 2011 16:37:28 -0800
Subject: [Freeswitch-users] Question regardin freeswitch startup info
In-Reply-To: <AANLkTikDDVXFjdeeAg6++jisFUkB+aaFCfSY3cPX7Pb4@mail.gmail.com>
References: <AANLkTikDDVXFjdeeAg6++jisFUkB+aaFCfSY3cPX7Pb4@mail.gmail.com>
Message-ID: <AANLkTi=VQ5WqK8gJ4onuD8MoB-xSDP_8ST-fadBFF0om@mail.gmail.com>

On Sun, Feb 20, 2011 at 7:55 PM, Hareem Haque <hareem.haque at gmail.com>wrote:

>  i start freeswitch it says max sessions 1000
> and session rate 30.. what exact do these numbers mean and how can i
> increase them
> Your help is greatly appreciated.
>
>
Max sessions is the total number of sessions (i.e. call legs) that can be
active at any one time. A single bridged call is two session (each person is
a single session). A conference with 10 people is, naturally, 10 sessions.

The session rate, also known as SPS (sessions per second) is just that - how
quickly can new sessions be created. These are default limits that can be
changed, however they were put there for your own protection, so please use
caution. (Do you REALLY need more than 30 sessions per second and 1000
concurrent sessions?)

-MC

Best Regards
> Hareem. Haque
>
>
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From msc at freeswitch.org  Wed Feb 23 03:38:52 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Feb 2011 16:38:52 -0800
Subject: [Freeswitch-users] Newbie question / what is the best text to
 speech module out there?
In-Reply-To: <2D36A10A-9E60-49B5-BFFE-20320448F329@voicecarrier.com>
References: <2D36A10A-9E60-49B5-BFFE-20320448F329@voicecarrier.com>
Message-ID: <AANLkTimDa9eR+1KT_DhcfHy-6Df=O37f0YKb6Yhcb6V1@mail.gmail.com>

Look at mod_tts_commandline and the svox TTS engine.
http://wiki.freeswitch.org/wiki/Mod_tts_commandline

-MC

On Sun, Feb 20, 2011 at 8:33 PM, Edward de Jong <
edward.dejong at voicecarrier.com> wrote:

> I see Cepstra and some other text to speech modules. Which is the best of
> the free ones? for my experiments I can't afford a fancy one like Cepstra.
>
> edj
>
>
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From anthony.minessale at gmail.com  Wed Feb 23 03:43:57 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 22 Feb 2011 18:43:57 -0600
Subject: [Freeswitch-users] Easy IVR?
In-Reply-To: <AANLkTimZcETCP4L8h1v2G8mzeYsiJsQjH2NFtnmmBVMN@mail.gmail.com>
References: <AANLkTin5VqM7Xt7VU3Abs+X0Y3FqGQ9tNS3Y7aHgCOmq@mail.gmail.com>
	<AANLkTimZcETCP4L8h1v2G8mzeYsiJsQjH2NFtnmmBVMN@mail.gmail.com>
Message-ID: <AANLkTimqmmT_gBmJ6uOhXtiUmLrdGK9_U1nkkwqyiagS@mail.gmail.com>

its kind of like saying, have your own softswitch in 5 lines of code:

#!/bin/sh
git clone git://git.freeswitch.org/freeswitch.git
cd freeswitch.git
./configure
make install
./freeswitch

The complexity is hiding somewhere.
That kinda felt like an ad.



On Tue, Feb 22, 2011 at 6:34 PM, Michael Collins <msc at freeswitch.org> wrote:
> It is if you are a Rails person. :)
> -MC
>
> On Sun, Feb 20, 2011 at 7:05 PM, EdPimentl <edpimentl at gmail.com> wrote:
>>
>> Is this not a good example of how easy IVR design/creation should be?
>> http://asymmetrical-view.com/2011/02/20/twilio-in-ten-minutes.html
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Wed Feb 23 03:44:27 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Feb 2011 16:44:27 -0800
Subject: [Freeswitch-users] newbie question: how do you set the folder
 that freeswitch uses for sounds?
In-Reply-To: <691770B6-DEC9-4F09-BE93-6B3DC6F9D6D3@voicecarrier.com>
References: <691770B6-DEC9-4F09-BE93-6B3DC6F9D6D3@voicecarrier.com>
Message-ID: <AANLkTimwvh_v0b+eYHS9UaAV8d8UJFSMfnf1cbrybUuR@mail.gmail.com>

Welcome to FreeSWITCH, btw! :)

These are all good questions that are answered on the wiki and in the
awesome FreeSWITCH book. For brevity's sake here's a quick overview:

When you specify an absolute path to a file, FreeSWITCH will always look in
the exact path that you specify, no questions asked.

When you specify a relative path then FreeSWITCH will be smart and look in
the sounds directory that correlates to the sampling rate of of the channel
where the file is being played. This means you should just install all the
sound files by issuing "make cd-sounds-install" and be done with it. (This
make command will install 8k, 16k, 32k, and 48k sound files.) You can play
around by using a phone that does 16k audio like most newer polycoms or even
x-lite.

Hope this helps.

-MC

On Sun, Feb 20, 2011 at 9:08 PM, Edward de Jong <
edward.dejong at voicecarrier.com> wrote:

> I see that in the default config, freeswitch uses sounds from the
> en/us/callie folder.
> so when you ask for playback of hello.wav, it goes to that folder by
> default.
> how does one change the folder used by freeswitch?
> i want to let the user pick between different voices, which of the many
> config files sets up which voice to use?
>
> and what if I want to have better voice on my user's phone? can I go to
> 16000 sample rate? if the user has HD capability on their phone, shouldn't I
> be using better voice samples? and how can I choose the voice folder
> depending on the phone connected to the PBX at that moment? In other words,
> if the user has an HD phone can I connect with them at the better codec?
> obviously if we connect to some other lesser device the dreaded transcoding
> will occur, but i would think for internal use, for example when the user is
> navigating menus, they would appreciate a better voice. Or should i just use
> all waveforms at 16000 and let it drop them down...
>
> 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 anthony.minessale at gmail.com  Wed Feb 23 03:45:48 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 22 Feb 2011 18:45:48 -0600
Subject: [Freeswitch-users] Easy IVR?
In-Reply-To: <AANLkTimqmmT_gBmJ6uOhXtiUmLrdGK9_U1nkkwqyiagS@mail.gmail.com>
References: <AANLkTin5VqM7Xt7VU3Abs+X0Y3FqGQ9tNS3Y7aHgCOmq@mail.gmail.com>
	<AANLkTimZcETCP4L8h1v2G8mzeYsiJsQjH2NFtnmmBVMN@mail.gmail.com>
	<AANLkTimqmmT_gBmJ6uOhXtiUmLrdGK9_U1nkkwqyiagS@mail.gmail.com>
Message-ID: <AANLkTimMZa-J9MXvJXtrK=PY6in15j6MY708qo-vF+gZ@mail.gmail.com>

oh and there is a bug in my script

#!/bin/sh
git clone git://git.freeswitch.org/freeswitch.git
cd freeswitch.git
./configure
make install
exec /usr/local/freeswitch/bin/freeswitch -nonat






On Tue, Feb 22, 2011 at 6:43 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> its kind of like saying, have your own softswitch in 5 lines of code:
>
> #!/bin/sh
> git clone git://git.freeswitch.org/freeswitch.git
> cd freeswitch.git
> ./configure
> make install
> ./freeswitch
>
> The complexity is hiding somewhere.
> That kinda felt like an ad.
>
>
>
> On Tue, Feb 22, 2011 at 6:34 PM, Michael Collins <msc at freeswitch.org> wrote:
>> It is if you are a Rails person. :)
>> -MC
>>
>> On Sun, Feb 20, 2011 at 7:05 PM, EdPimentl <edpimentl at gmail.com> wrote:
>>>
>>> Is this not a good example of how easy IVR design/creation should be?
>>> http://asymmetrical-view.com/2011/02/20/twilio-in-ten-minutes.html
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 infos at madovsky.org  Wed Feb 23 03:47:22 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 22 Feb 2011 19:47:22 -0500
Subject: [Freeswitch-users] risky hangup
Message-ID: <6DB309CE8CDA497880601214A1FB1D03@e1705>

apparently I found the problem,
there were 2 instances of FS running...
btw, is there any trick (other than killall -9 freeswitch before start the fs startup script)
to avoid to run 2 instances of FS ?

  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, February 22, 2011 7:06 PM
  Subject: Re: [Freeswitch-users] risky hangup


  Example:
  I make an external call without credits
  so the call is transferred (by nibblebill)
  to a "nofunds" extension with plaback wav file
    ----- Original Message ----- 
    From: Michael Collins 
    To: FreeSWITCH Users Help 
    Sent: Tuesday, February 22, 2011 6:42 PM
    Subject: Re: [Freeswitch-users] risky hangup


    please give some context... how are you testing this? What is calling what, etc.? 
    -MC



    On Tue, Feb 22, 2011 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:

      on last git I noticed that every hangup
      after TTS is made with a dfferent delay.
      sometimes it cuts the voice, sometimes it cuts suddenly before the voice starts
      and sometimes it cuts the voice 2/3 seconds before.
      after 10 tests all hangups are done with a different delay...
      please advice

      thanks


----------------------------------------------------------------------------


    _______________________________________________
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    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From edpimentl at gmail.com  Wed Feb 23 03:47:25 2011
From: edpimentl at gmail.com (EdPimentl)
Date: Tue, 22 Feb 2011 19:47:25 -0500
Subject: [Freeswitch-users] Easy IVR?
In-Reply-To: <AANLkTimZcETCP4L8h1v2G8mzeYsiJsQjH2NFtnmmBVMN@mail.gmail.com>
References: <AANLkTin5VqM7Xt7VU3Abs+X0Y3FqGQ9tNS3Y7aHgCOmq@mail.gmail.com>
	<AANLkTimZcETCP4L8h1v2G8mzeYsiJsQjH2NFtnmmBVMN@mail.gmail.com>
Message-ID: <AANLkTi=xuG_LEY2Wdo1Dd2eFKtODwajryra7zhW3HNi-@mail.gmail.com>

True....
Would consider putting a bounty to get similar functionality in FS.


-E





On Tue, Feb 22, 2011 at 7:34 PM, Michael Collins <msc at freeswitch.org> wrote:

> It is if you are a Rails person. :)
> -MC
>
> On Sun, Feb 20, 2011 at 7:05 PM, EdPimentl <edpimentl at gmail.com> wrote:
>
>> Is this not a good example of how easy IVR design/creation should be?
>> http://asymmetrical-view.com/2011/02/20/twilio-in-ten-minutes.html
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 23 03:49:31 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 22 Feb 2011 18:49:31 -0600
Subject: [Freeswitch-users] risky hangup
In-Reply-To: <6DB309CE8CDA497880601214A1FB1D03@e1705>
References: <6DB309CE8CDA497880601214A1FB1D03@e1705>
Message-ID: <AANLkTikpg5fJp6uEgg7RZHLur_RvyO5k4h-QNvP+KOTc@mail.gmail.com>

it should not be able to run more than once unless you have a problem
with your filesystem.
it refuses to start when its already running unless you don't have or
have permssion problems on the pid file

usually it would say:

Cannot lock pid file /usr/local/freeswitch/run/freeswitch.pid.



On Tue, Feb 22, 2011 at 6:47 PM, Madovsky <infos at madovsky.org> wrote:
> apparently I found the problem,
> there were 2 instances of FS running...
> btw, is there any trick (other than killall -9 freeswitch before start the
> fs startup script)
> to avoid to run 2 instances of FS ?
>
>
> ----- Original Message -----
> From: Madovsky
> To: FreeSWITCH Users Help
> Sent: Tuesday, February 22, 2011 7:06 PM
> Subject: Re: [Freeswitch-users] risky hangup
> Example:
> I make an external call without credits
> so the call is transferred (by nibblebill)
> to a "nofunds" extension with plaback wav file
>
> ----- Original Message -----
> From: Michael Collins
> To: FreeSWITCH Users Help
> Sent: Tuesday, February 22, 2011 6:42 PM
> Subject: Re: [Freeswitch-users] risky hangup
> please give some context... how are you testing this? What is calling what,
> etc.?
> -MC
>
> On Tue, Feb 22, 2011 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:
>>
>> on last git I noticed that every hangup
>> after TTS is made with a dfferent delay.
>> sometimes it cuts the voice, sometimes it cuts suddenly before the voice
>> starts
>> and sometimes it cuts the voice 2/3 seconds before.
>> after 10 tests all hangups are done with a different delay...
>> please advice
>>
>> 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  Wed Feb 23 03:54:55 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 22 Feb 2011 18:54:55 -0600
Subject: [Freeswitch-users] Easy IVR?
In-Reply-To: <AANLkTi=xuG_LEY2Wdo1Dd2eFKtODwajryra7zhW3HNi-@mail.gmail.com>
References: <AANLkTin5VqM7Xt7VU3Abs+X0Y3FqGQ9tNS3Y7aHgCOmq@mail.gmail.com>
	<AANLkTimZcETCP4L8h1v2G8mzeYsiJsQjH2NFtnmmBVMN@mail.gmail.com>
	<AANLkTi=xuG_LEY2Wdo1Dd2eFKtODwajryra7zhW3HNi-@mail.gmail.com>
Message-ID: <AANLkTin-JM4_fPc8ag8_Mz=_5WUmZJTKQALr6a0gXbWw@mail.gmail.com>

What exactly is the functionality?  We pretty much have all the API
hooks you can imagine to build apps.


On Tue, Feb 22, 2011 at 6:47 PM, EdPimentl <edpimentl at gmail.com> wrote:
> True....
> Would consider putting a bounty to get similar functionality in FS.
>
>
> -E
>
>
>
>
>
> On Tue, Feb 22, 2011 at 7:34 PM, Michael Collins <msc at freeswitch.org> wrote:
>>
>> It is if you are a Rails person. :)
>> -MC
>>
>> On Sun, Feb 20, 2011 at 7:05 PM, EdPimentl <edpimentl at gmail.com> wrote:
>>>
>>> Is this not a good example of how easy IVR design/creation should be?
>>> http://asymmetrical-view.com/2011/02/20/twilio-in-ten-minutes.html
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Wed Feb 23 03:56:46 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 22 Feb 2011 19:56:46 -0500
Subject: [Freeswitch-users] Easy IVR?
References: <AANLkTin5VqM7Xt7VU3Abs+X0Y3FqGQ9tNS3Y7aHgCOmq@mail.gmail.com><AANLkTimZcETCP4L8h1v2G8mzeYsiJsQjH2NFtnmmBVMN@mail.gmail.com><AANLkTimqmmT_gBmJ6uOhXtiUmLrdGK9_U1nkkwqyiagS@mail.gmail.com>
	<AANLkTimMZa-J9MXvJXtrK=PY6in15j6MY708qo-vF+gZ@mail.gmail.com>
Message-ID: <CA2A526965ED4DABAA53CCB0573747F6@e1705>

not sure this twilio in ten minutes can be done iin ten minutes.
Even Anthony can't do it I'm almost sure :D

----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, February 22, 2011 7:45 PM
Subject: Re: [Freeswitch-users] Easy IVR?


> oh and there is a bug in my script
>
> #!/bin/sh
> git clone git://git.freeswitch.org/freeswitch.git
> cd freeswitch.git
> ./configure
> make install
> exec /usr/local/freeswitch/bin/freeswitch -nonat
>
>
>
>
>
>
> On Tue, Feb 22, 2011 at 6:43 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> its kind of like saying, have your own softswitch in 5 lines of code:
>>
>> #!/bin/sh
>> git clone git://git.freeswitch.org/freeswitch.git
>> cd freeswitch.git
>> ./configure
>> make install
>> ./freeswitch
>>
>> The complexity is hiding somewhere.
>> That kinda felt like an ad.
>>
>>
>>
>> On Tue, Feb 22, 2011 at 6:34 PM, Michael Collins <msc at freeswitch.org> 
>> wrote:
>>> It is if you are a Rails person. :)
>>> -MC
>>>
>>> On Sun, Feb 20, 2011 at 7:05 PM, EdPimentl <edpimentl at gmail.com> wrote:
>>>>
>>>> Is this not a good example of how easy IVR design/creation should be?
>>>> http://asymmetrical-view.com/2011/02/20/twilio-in-ten-minutes.html
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 



From infos at madovsky.org  Wed Feb 23 03:59:00 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 22 Feb 2011 19:59:00 -0500
Subject: [Freeswitch-users] risky hangup
References: <6DB309CE8CDA497880601214A1FB1D03@e1705>
	<AANLkTikpg5fJp6uEgg7RZHLur_RvyO5k4h-QNvP+KOTc@mail.gmail.com>
Message-ID: <D53AA90F645442FB938594292D4D4454@e1705>

mmhm weird...
in fact sometimes I work/recompile/reinstall FS manually
and in same time there is a cron task that check if FS is running,
and sometimes cron istarts when I restart FS manually...

----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, February 22, 2011 7:49 PM
Subject: Re: [Freeswitch-users] risky hangup


> it should not be able to run more than once unless you have a problem
> with your filesystem.
> it refuses to start when its already running unless you don't have or
> have permssion problems on the pid file
>
> usually it would say:
>
> Cannot lock pid file /usr/local/freeswitch/run/freeswitch.pid.
>
>
>
> On Tue, Feb 22, 2011 at 6:47 PM, Madovsky <infos at madovsky.org> wrote:
>> apparently I found the problem,
>> there were 2 instances of FS running...
>> btw, is there any trick (other than killall -9 freeswitch before start 
>> the
>> fs startup script)
>> to avoid to run 2 instances of FS ?
>>
>>
>> ----- Original Message -----
>> From: Madovsky
>> To: FreeSWITCH Users Help
>> Sent: Tuesday, February 22, 2011 7:06 PM
>> Subject: Re: [Freeswitch-users] risky hangup
>> Example:
>> I make an external call without credits
>> so the call is transferred (by nibblebill)
>> to a "nofunds" extension with plaback wav file
>>
>> ----- Original Message -----
>> From: Michael Collins
>> To: FreeSWITCH Users Help
>> Sent: Tuesday, February 22, 2011 6:42 PM
>> Subject: Re: [Freeswitch-users] risky hangup
>> please give some context... how are you testing this? What is calling 
>> what,
>> etc.?
>> -MC
>>
>> On Tue, Feb 22, 2011 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:
>>>
>>> on last git I noticed that every hangup
>>> after TTS is made with a dfferent delay.
>>> sometimes it cuts the voice, sometimes it cuts suddenly before the voice
>>> starts
>>> and sometimes it cuts the voice 2/3 seconds before.
>>> after 10 tests all hangups are done with a different delay...
>>> please advice
>>>
>>> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 edpimentl at gmail.com  Wed Feb 23 04:06:45 2011
From: edpimentl at gmail.com (EdPimentl)
Date: Tue, 22 Feb 2011 20:06:45 -0500
Subject: [Freeswitch-users] Easy IVR?
In-Reply-To: <AANLkTin-JM4_fPc8ag8_Mz=_5WUmZJTKQALr6a0gXbWw@mail.gmail.com>
References: <AANLkTin5VqM7Xt7VU3Abs+X0Y3FqGQ9tNS3Y7aHgCOmq@mail.gmail.com>
	<AANLkTimZcETCP4L8h1v2G8mzeYsiJsQjH2NFtnmmBVMN@mail.gmail.com>
	<AANLkTi=xuG_LEY2Wdo1Dd2eFKtODwajryra7zhW3HNi-@mail.gmail.com>
	<AANLkTin-JM4_fPc8ag8_Mz=_5WUmZJTKQALr6a0gXbWw@mail.gmail.com>
Message-ID: <AANLkTikC3bHLhDn8Q3FHJfKt8kzfdZiBNF1huTX5qoV5@mail.gmail.com>

I would be glad to provide a HTML5 / CSS3 form.

My thinking is do just cover the very basics....


Other can then enhance it for their specific use case.


Sincerely,
-E




On Tue, Feb 22, 2011 at 7:54 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> What exactly is the functionality?  We pretty much have all the API
> hooks you can imagine to build apps.
>
>
> On Tue, Feb 22, 2011 at 6:47 PM, EdPimentl <edpimentl at gmail.com> wrote:
> > True....
> > Would consider putting a bounty to get similar functionality in FS.
> >
> >
> > -E
> >
> >
> >
> >
> >
> > On Tue, Feb 22, 2011 at 7:34 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> >>
> >> It is if you are a Rails person. :)
> >> -MC
> >>
> >> On Sun, Feb 20, 2011 at 7:05 PM, EdPimentl <edpimentl at gmail.com> wrote:
> >>>
> >>> Is this not a good example of how easy IVR design/creation should be?
> >>> http://asymmetrical-view.com/2011/02/20/twilio-in-ten-minutes.html
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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  Wed Feb 23 04:15:19 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Feb 2011 17:15:19 -0800
Subject: [Freeswitch-users] Message Playback Stops
In-Reply-To: <AANLkTi=EMQsSkH-SZDq2rUzvh2-7nYK=NpKHAXXEwc-W@mail.gmail.com>
References: <AANLkTi=EMQsSkH-SZDq2rUzvh2-7nYK=NpKHAXXEwc-W@mail.gmail.com>
Message-ID: <AANLkTin6Y5NBf8jzV8cyQq=9UZWdTOqGXD32X3O_4TLd@mail.gmail.com>

I think this could be filed under "abusing Lua" :)

The amount of control you are doing on these calls tells me that you should
absolutely be doing most of this logic outside of FreeSWITCH. I would
recommend a socket-based application that can build the dialstrings and send
them to FS as originate commands. I've personally tinkered around and gotten
50-60 cps with a perl script that does this kind of thing. (Haven't done
much w/ it lately.)

-MC

On Mon, Feb 21, 2011 at 11:22 PM, Michael De Lorenzo <
delorenzodesign at gmail.com> wrote:

> Hello,
>
> I have a Freeswitch installation that is intended to make many calls
> (thousands) and playback a single wav file.  The calls are successfully
> processed (the recipient's phone rings), but the call almost immediately
> disconnects, after about 1s.  Everything seems to work fine if I'm only
> pushing one or two calls through the Freeswitch instance, but as soon as I
> turn up the call rate (I'm still only doing about 50 concurrent sessions)
> the playback begins to fail.
>
> I've watched the calls go out from the console and nothing looks out of the
> ordinary, except that the calls are disconnected with NORMAL CLEARING prior
> to completion.
>
> Here's the Lua script I'm using...
>
> profile_id = argv[1];
> account_code = argv[2];
> client_id = argv[3];
> caller_id_name = argv[4];
> caller_id = argv[5];
> dial_id = argv[6];
> number_to_call = argv[7];
> message_to_play = argv[8];
> max_retries = argv[9];
> retry_interval = argv[10];
>
> local human_detected = false
> local voicemail_detected = false;
> local message_played = false;
>
> recordings_directory = "/usr/local/freeswitch/recordings/messages/";
>
> function setDialVariables(set_as_session_variables)
>     local s = "profile_id=" .. profile_id;
>     s = s .. ",account_code=" .. account_code;
>     s = s .. ",client_id=" .. client_id;
>     s = s .. ",caller_id_name=" .. caller_id_name;
>     s = s .. ",caller_id=" .. caller_id;
>     s = s .. ",dial_id=" .. dial_id;
>     s = s .. ",number_to_call=" .. number_to_call;
>     s = s .. ",message_to_play=" .. message_to_play;
>
>     freeswitch.consoleLog("notice", s .. "\n");
>
>     return s
> end
>
> function printSessionVariables()
>     freeswitch.consoleLog("notice", "******* PRINTING SESSION VARIABLES
> **********\n");
>     -- ommitted
>     freeswitch.consoleLog("notice",
> "**********************************************\n");
> end
>
> function onInput(s, type, obj, arg)
>     if(type == "event" and voicemail_detected == false) then
>         freeswitch.consoleLog("debug","************ VOICE MAIL/ANSWERING
> MACHINE DETECTED *************\n");
>         voicemail_detected = true;
>         return "break";
>     end
>     return true;
> end
>
> function playbackMessage(sleepTime)
>     message_played = false;
>     session:sleep(sleepTime);
>     -- play a file
>     message_file = recordings_directory .. message_to_play;
>     freeswitch.consoleLog("notice", "Playing file: " .. message_file ..
> "\n");
>     session:streamFile(message_file);
>     freeswitch.consoleLog("notice", "!!!!! Finished playing the file
> !!!!!\n");
>     message_played = true;
> end
>
> session = freeswitch.Session("{" .. setDialVariables(false) ..
> ",ignore_early_media=true,origination_caller_id_name=" .. caller_id_name ..
> ",origination_caller_id_number=+1" .. caller_id .. "}sofia/gateway/gateway_"
> .. profile_id .. "/" .. number_to_call);
>
> while(session:ready()) do
>    setDialVariables(true)
>    session:answer();
>
>    -- session:execute("continue_on_fail","true");
>    session:setInputCallback("onInput","true");
>    session:execute("avmd","start");
>
>    playbackMessage(200);
>
>    vm_status = voicemail_detected == true and "yes" or "no"
>    freeswitch.consoleLog("info", "Was VM detected? " .. vm_status .. "\n");
>    if(voicemail_detected) then
>       return "break";
>    end
>
>    freeswitch.consoleLog("notice", "Played the message at least once and
> checked for VM, we should be exiting the loop.\n")
> end
>
> if (voicemail_detected) then
>     freeswitch.consoleLog("info", "Playback for voicemail.\n");
>     session:execute("avmd","stop");
>     playbackMessage(5000);
> end
>
> freeswitch.consoleLog("info", "All finished, hanging up the session.\n");
> session:hangup();
>
> Any help would be greatly appreciated.
>
> Thank you,
>
> 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 msc at freeswitch.org  Wed Feb 23 04:17:12 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Feb 2011 17:17:12 -0800
Subject: [Freeswitch-users] VM to email error
In-Reply-To: <817664CE-FEA5-430C-A0C2-549F0FD510C7@5ninesolutions.com>
References: <817664CE-FEA5-430C-A0C2-549F0FD510C7@5ninesolutions.com>
Message-ID: <AANLkTim6TcKnKWiRFiibH75gMtEEvyqhWPXkC3smpzBa@mail.gmail.com>

What user is FS running as and did you test as that user?
-MC

On Tue, Feb 22, 2011 at 1:02 AM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> Hello all,
> I keep seeing an error in the logs similar to the following when a VM
> is emailed.  Everything seems to be working just fine so I'm trying to
> decipher the log entry.
>
> 2011-02-21 19:57:48.288269 [ERR] switch_utils.c:699 Unable to execute
> command: /bin/cat /tmp/mail.12983398676f47 | /usr/sbin/sendmail -f
> fromuser at fromdomain.com
>  -t touser at todomain.com
>
> /bin/cat works as does /usr/sbin/sendmail.
>
> This is on Centos 5.5 w/ Postfix instead of sendmail and /var/log/
> maillog clearly shows the message being delivered.  Do I have
> something misconfigured?
>
> Thanks,
> 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
>
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From anthony.minessale at gmail.com  Wed Feb 23 04:20:26 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 22 Feb 2011 19:20:26 -0600
Subject: [Freeswitch-users] Message Playback Stops
In-Reply-To: <AANLkTi=EMQsSkH-SZDq2rUzvh2-7nYK=NpKHAXXEwc-W@mail.gmail.com>
References: <AANLkTi=EMQsSkH-SZDq2rUzvh2-7nYK=NpKHAXXEwc-W@mail.gmail.com>
Message-ID: <AANLkTinpDt+NrcoK8pwWuuwMufut+DTC7GSrnAdWYhJ5@mail.gmail.com>

do you have any console logs of it failing or any error messages etc?


On Tue, Feb 22, 2011 at 1:22 AM, Michael De Lorenzo
<delorenzodesign at gmail.com> wrote:
> Hello,
>
> I have a Freeswitch installation that is intended to make many calls
> (thousands) and playback a single wav file.? The calls are successfully
> processed (the recipient's phone rings), but the call almost immediately
> disconnects, after about 1s.? Everything seems to work fine if I'm only
> pushing one or two calls through the Freeswitch instance, but as soon as I
> turn up the call rate (I'm still only doing about 50 concurrent sessions)
> the playback begins to fail.
>
> I've watched the calls go out from the console and nothing looks out of the
> ordinary, except that the calls are disconnected with NORMAL CLEARING prior
> to completion.
>
> Here's the Lua script I'm using...
>
> profile_id = argv[1];
> account_code = argv[2];
> client_id = argv[3];
> caller_id_name = argv[4];
> caller_id = argv[5];
> dial_id = argv[6];
> number_to_call = argv[7];
> message_to_play = argv[8];
> max_retries = argv[9];
> retry_interval = argv[10];
>
> local human_detected = false
> local voicemail_detected = false;
> local message_played = false;
>
> recordings_directory = "/usr/local/freeswitch/recordings/messages/";
>
> function setDialVariables(set_as_session_variables)
> ??? local s = "profile_id=" .. profile_id;
> ??? s = s .. ",account_code=" .. account_code;
> ??? s = s .. ",client_id=" .. client_id;
> ??? s = s .. ",caller_id_name=" .. caller_id_name;
> ??? s = s .. ",caller_id=" .. caller_id;
> ??? s = s .. ",dial_id=" .. dial_id;
> ??? s = s .. ",number_to_call=" .. number_to_call;
> ??? s = s .. ",message_to_play=" .. message_to_play;
>
> ??? freeswitch.consoleLog("notice", s .. "\n");
>
> ??? return s
> end
>
> function printSessionVariables()
> ??? freeswitch.consoleLog("notice", "******* PRINTING SESSION VARIABLES
> **********\n");
> ??? -- ommitted
> ??? freeswitch.consoleLog("notice",
> "**********************************************\n");
> end
>
> function onInput(s, type, obj, arg)
> ??? if(type == "event" and voicemail_detected == false) then
> ??????? freeswitch.consoleLog("debug","************ VOICE MAIL/ANSWERING
> MACHINE DETECTED *************\n");
> ??????? voicemail_detected = true;
> ??????? return "break";
> ??? end
> ??? return true;
> end
>
> function playbackMessage(sleepTime)
> ??? message_played = false;
> ??? session:sleep(sleepTime);
> ??? -- play a file
> ??? message_file = recordings_directory .. message_to_play;
> ??? freeswitch.consoleLog("notice", "Playing file: " .. message_file ..
> "\n");
> ??? session:streamFile(message_file);
> ??? freeswitch.consoleLog("notice", "!!!!! Finished playing the file
> !!!!!\n");
> ??? message_played = true;
> end
>
> session = freeswitch.Session("{" .. setDialVariables(false) ..
> ",ignore_early_media=true,origination_caller_id_name=" .. caller_id_name ..
> ",origination_caller_id_number=+1" .. caller_id .. "}sofia/gateway/gateway_"
> .. profile_id .. "/" .. number_to_call);
>
> while(session:ready()) do
> ?? setDialVariables(true)
> ?? session:answer();
>
> ?? -- session:execute("continue_on_fail","true");
> ?? session:setInputCallback("onInput","true");
> ?? session:execute("avmd","start");
>
> ?? playbackMessage(200);
>
> ?? vm_status = voicemail_detected == true and "yes" or "no"
> ?? freeswitch.consoleLog("info", "Was VM detected? " .. vm_status .. "\n");
> ?? if(voicemail_detected) then
> ????? return "break";
> ?? end
>
> ?? freeswitch.consoleLog("notice", "Played the message at least once and
> checked for VM, we should be exiting the loop.\n")
> end
>
> if (voicemail_detected) then
> ??? freeswitch.consoleLog("info", "Playback for voicemail.\n");
> ??? session:execute("avmd","stop");
> ??? playbackMessage(5000);
> end
>
> freeswitch.consoleLog("info", "All finished, hanging up the session.\n");
> session:hangup();
>
> Any help would be greatly appreciated.
>
> Thank you,
>
> 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
>
>



-- 
Anthony Minessale II

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

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

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


From msc at freeswitch.org  Wed Feb 23 05:13:01 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Feb 2011 18:13:01 -0800
Subject: [Freeswitch-users] Problems getting asterisk registered with FS
	sbc
In-Reply-To: <D4534174-491C-45D2-A3BF-E29EDE85DF2A@3js.de>
References: <D4534174-491C-45D2-A3BF-E29EDE85DF2A@3js.de>
Message-ID: <AANLkTikmJscHtGUUZcqY8CD4ER0-UVMprFkQPUsXDO1+@mail.gmail.com>

I don't believe the ACL works for registrations, only for phone calls.
You'll still need to auth for the registration part. For the ACL, though,
you can do "reloadacl" and confirm that your CIDR is getting added. When you
send calls from TB to FS they should be let in via the ACL without an auth
challenge.

-MC

On Tue, Feb 22, 2011 at 11:30 AM, Johannes Jakob <jjj at 3js.de> wrote:

> Fellow FreeSWITCH Admins,
>
> I'm having a hard time, getting a Trixbox 2.8 box to register with our
> FreeSWITCH SBCs.
>
> The FreeSWITCHes are running FreeSWITCH-mod_sofia/1.0.head-git-7847289, the
> asterisk on the trixbox is Asterisk 1.6.0.22-samy-r60.
>
>
> The user's directory entry looks like this:
>
>
> <include>
>  <user id="748732" cidr="10.16.153.163/32">
>    <params>
> <!--
>      <param name="password" value="Idsd67Hsa"/>
> -->
>      <param name="t38-passthru" value="true"/>
>    </params>
>    <variables>
>      <variable name="client_asserted_identity" value="03023671836"/>
>      <variable name="accountcode" value="748732"/>
>      <variable name="contract" value="2253"/>
>      <variable name="nibble_rate" value="1"/>
>      <variable name="nibble_account" value="2253"/>
>
>      <variable name="proxy_media" value="true"/>
>      <variable name="bypass_media" value="false"/>
>      <variable name="user_context" value="mysip.net"/>
>      <variable name="default_language" value="de"/>
>      <variable name="language" value="de"/>
>    </variables>
>  </user>
> </include>
>
>
> Asterisk's register string: 748732 at mysip.net@sbc1.mysip.net/748732
>
>
> I'm getting the "normal" FS errors for wrong credentials:
>
> 2011-02-22 18:03:57.484939 [WARNING] sofia_reg.c:1246 SIP auth challenge
> (REGISTER) on sofia profile 'internal' for [748732 at mysip.net] from ip
> 10.16.153.163
> 2011-02-22 18:03:57.491471 [WARNING] sofia_reg.c:1204 SIP auth failure
> (REGISTER) on sofia profile 'internal' for [748732 at mysip.net] from ip
> 10.16.153.163
>
>
> but why am I getting these? I specified the right address in the cidr
> statement! Why is it even bothering with anything else but the right
> user at domain and IP-address?
>
>
> There are some other asterisk boxes (> 1.8.2) registering to this SBC with
> equal settings just fine, what's wrong with this little trixbox system? ;)
>
>
>
> Of course I did get you some SIP traces as well:
>
>
> 18:00:37.063410 IP 10.16.153.163.5060 > 10.16.133.66.5060: UDP, length: 419
> E`..f...>.7.^...^..B.......-REGISTER sip:mysip.net SIP/2.0
> Via: SIP/2.0/UDP 10.16.153.163:5060;branch=z9hG4bK3e70680b;rport
> Max-Forwards: 70
> From: <sip:748732 at mysip.net>;tag=as77c8852d
> To: <sip:748732 at mysip.net>
> Call-ID: 53d04cc277cfe60301bddb6d79033420 at 10.16.153.163
> CSeq: 102 REGISTER
> User-Agent: Asterisk PBX 1.6.0.22-samy-r60
> Expires: 1800
> Contact: <sip:748732 at 10.16.153.163>
> Event: registration
> Content-Length: 0
>
>
>
> 18:00:37.074085 IP 10.16.133.66.5060 > 10.16.153.163.5060: UDP, length: 657
> E...F...?.Vc^..B^...........SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 10.16.153.163:5060;branch=z9hG4bK3e70680b;rport=5060
> From: <sip:748732 at mysip.net>;tag=as77c8852d
> To: <sip:748732 at mysip.net>;tag=5jD9Qcg3N9S6p
> Call-ID: 53d04cc277cfe60301bddb6d79033420 at 10.16.153.163
> CSeq: 102 REGISTER
> 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
> WWW-Authenticate: Digest realm="mysip.net",
> nonce="ce2bccbf-a27b-43c8-b7b0-a89ab429d8a7", algorithm=MD5, qop="auth"
> Content-Length: 0
>
>
>
> 18:00:37.074969 IP 10.16.153.163.5060 > 10.16.133.66.5060: UDP, length: 672
> E`..f...>.6.^...^..B........REGISTER sip:mysip.net SIP/2.0
> Via: SIP/2.0/UDP 10.16.153.163:5060;branch=z9hG4bK30df5010;rport
> Max-Forwards: 70
> From: <sip:748732 at mysip.net>;tag=as03431ba4
> To: <sip:748732 at mysip.net>
> Call-ID: 53d04cc277cfe60301bddb6d79033420 at 10.16.153.163
> CSeq: 103 REGISTER
> User-Agent: Asterisk PBX 1.6.0.22-samy-r60
> Authorization: Digest username="748732 at mysip.net", realm="mysip.net",
> algorithm=MD5, uri="sip:mysip.net",
> nonce="ce2bccbf-a27b-43c8-b7b0-a89ab429d8a7",
> response="133a0ba843fe9f5afba67d1377fa8c11", qop=auth, cnonce="119cf18c",
> nc=00000001
> Expires: 1800
> Contact: <sip:748732 at 10.16.153.163>
> Event: registration
> Content-Length: 0
>
>
> 18:00:37.081517 IP 10.16.133.66.5060 > 10.16.153.163.5060: UDP, length: 532
> E..0F...?.V.^..B^.........1.SIP/2.0 403 Forbidden
> Via: SIP/2.0/UDP 10.16.153.163:5060;branch=z9hG4bK30df5010;rport=5060
> From: <sip:748732 at mysip.net>;tag=as03431ba4
> To: <sip:748732 at mysip.net>;tag=6U61S706jjgSj
> Call-ID: 53d04cc277cfe60301bddb6d79033420 at 10.16.153.163
> CSeq: 103 REGISTER
> 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-Length: 0
>
>
> Can somebody point me in the right direction?
>
>
> Thanks and best regards,
>
>  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
>
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From k-b-24 at live.com  Wed Feb 23 05:35:36 2011
From: k-b-24 at live.com (Jason b.a)
Date: Wed, 23 Feb 2011 02:35:36 +0000
Subject: [Freeswitch-users] OpenSER interface to FreeSWITCH
Message-ID: <COL111-W442B2A5D707F79AF2A8338ACDB0@phx.gbl>


yeh sure i have a SIP endpoints , can the application do the sip handling and have a socket interface with freeswitch in the same time without using OpenSER.
in this case i need Sip servelet plugin in my application , is it possible.
I went through the outbound event socket , seems to be helpful thx , also i saw the options for using java to connect to freeswitch, i think using ESL , i can build a java IVR application .
  		 	   		  
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From brian at freeswitch.org  Wed Feb 23 05:38:50 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 22 Feb 2011 20:38:50 -0600
Subject: [Freeswitch-users] OpenSER interface to FreeSWITCH
In-Reply-To: <COL111-W442B2A5D707F79AF2A8338ACDB0@phx.gbl>
References: <COL111-W442B2A5D707F79AF2A8338ACDB0@phx.gbl>
Message-ID: <48415C45-67AE-4E4F-8F17-3C08FBB5E10F@freeswitch.org>

HUH?  Are you just trying to use FreeSWITCH and a SIP app server?

/b

On Feb 22, 2011, at 8:35 PM, Jason b.a wrote:

> yeh sure i have a SIP endpoints , can the application do the sip handling and have a socket interface with freeswitch in the same time without using OpenSER.
> in this case i need Sip servelet plugin in my application , is it possible.
> I went through the outbound event socket , seems to be helpful thx , also i saw the options for using java to connect to freeswitch, i think using ESL , i can build a java IVR 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

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From Nabble at slickdeals.endjunk.com  Wed Feb 23 06:16:26 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 22 Feb 2011 19:16:26 -0800 (PST)
Subject: [Freeswitch-users] risky hangup
In-Reply-To: <6DB309CE8CDA497880601214A1FB1D03@e1705>
References: <41B940C7439741069AFFEF360F06744B@e1705>
	<6DB309CE8CDA497880601214A1FB1D03@e1705>
Message-ID: <1298430986573-6055032.post@n2.nabble.com>



Madovsky wrote:
> 
> apparently I found the problem,
> there were 2 instances of FS running...
Hmm ..., what exactly did you mean by instances? I don't suppose you meant
instances = processes, right?

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/risky-hangup-tp6054481p6055032.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From infos at madovsky.org  Wed Feb 23 06:26:26 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 22 Feb 2011 22:26:26 -0500
Subject: [Freeswitch-users] risky hangup
References: <41B940C7439741069AFFEF360F06744B@e1705><6DB309CE8CDA497880601214A1FB1D03@e1705>
	<1298430986573-6055032.post@n2.nabble.com>
Message-ID: <B0A5ECC84C31496CBB14F008B872B7C9@e1705>

yes

----- Original Message ----- 
From: "mazilo" <Nabble at slickdeals.endjunk.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, February 22, 2011 10:16 PM
Subject: Re: [Freeswitch-users] risky hangup


>
>
> Madovsky wrote:
>>
>> apparently I found the problem,
>> there were 2 instances of FS running...
> Hmm ..., what exactly did you mean by instances? I don't suppose you meant
> instances = processes, right?
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> -- 
> View this message in context: 
> http://freeswitch-users.2379917.n2.nabble.com/risky-hangup-tp6054481p6055032.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 infos at madovsky.org  Wed Feb 23 07:02:50 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 22 Feb 2011 23:02:50 -0500
Subject: [Freeswitch-users] risky hangup
Message-ID: <B19CF4E8F63F444EA70EB8D4F8A0505A@e1705>

ok apparently I was wrong...
I restarted the machine and the issue is still there :((

----- Original Message ----- 
From: "Madovsky" <infos at madovsky.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, February 22, 2011 10:26 PM
Subject: Re: [Freeswitch-users] risky hangup


> yes
>
> ----- Original Message ----- 
> From: "mazilo" <Nabble at slickdeals.endjunk.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, February 22, 2011 10:16 PM
> Subject: Re: [Freeswitch-users] risky hangup
>
>
>>
>>
>> Madovsky wrote:
>>>
>>> apparently I found the problem,
>>> there were 2 instances of FS running...
>> Hmm ..., what exactly did you mean by instances? I don't suppose you 
>> meant
>> instances = processes, right?
>>
>> -----
>> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
>> -- 
>> View this message in context: 
>> http://freeswitch-users.2379917.n2.nabble.com/risky-hangup-tp6054481p6055032.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 spencer at 5ninesolutions.com  Wed Feb 23 07:06:30 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Tue, 22 Feb 2011 22:06:30 -0600
Subject: [Freeswitch-users] VM to email error
In-Reply-To: <AANLkTim6TcKnKWiRFiibH75gMtEEvyqhWPXkC3smpzBa@mail.gmail.com>
References: <817664CE-FEA5-430C-A0C2-549F0FD510C7@5ninesolutions.com>
	<AANLkTim6TcKnKWiRFiibH75gMtEEvyqhWPXkC3smpzBa@mail.gmail.com>
Message-ID: <927246DC-3CD8-44E7-9AF0-228B229AA3FB@5ninesolutions.com>

Freeswitch is running as freeswitch, uid 101.  I did and yes both /bin/ 
cat and /usr/sbin/sendmail are executable as the freeswitch user.  I  
checked perms and then verified with sudo there wasn't anything stupid  
somewhere in the path.  The email does get delivered.. it seems if  
this command didn't get executed the email wouldn't get sent.

[admin at pbx10 /]$ ps aux | grep freeswitch
101       5570  0.6  6.4 543132 33976 ?        Ssl  Feb19  30:16 /opt/ 
freeswitch/bin/freeswitch -ncwait -nonat


Thanks,
Spencer

On Feb 22, 2011, at 7:17 PM, Michael Collins wrote:

> What user is FS running as and did you test as that user?
> -MC
>
> On Tue, Feb 22, 2011 at 1:02 AM, Spencer Thomason <spencer at 5ninesolutions.com 
> > wrote:
> Hello all,
> I keep seeing an error in the logs similar to the following when a VM
> is emailed.  Everything seems to be working just fine so I'm trying to
> decipher the log entry.
>
> 2011-02-21 19:57:48.288269 [ERR] switch_utils.c:699 Unable to execute
> command: /bin/cat /tmp/mail.12983398676f47 | /usr/sbin/sendmail -f fromuser at fromdomain.com
>  -t touser at todomain.com
>
> /bin/cat works as does /usr/sbin/sendmail.
>
> This is on Centos 5.5 w/ Postfix instead of sendmail and /var/log/
> maillog clearly shows the message being delivered.  Do I have
> something misconfigured?
>
> Thanks,
> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From delorenzodesign at gmail.com  Wed Feb 23 09:00:14 2011
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Wed, 23 Feb 2011 01:00:14 -0500
Subject: [Freeswitch-users] Message Playback Stops
In-Reply-To: <AANLkTinpDt+NrcoK8pwWuuwMufut+DTC7GSrnAdWYhJ5@mail.gmail.com>
References: <AANLkTi=EMQsSkH-SZDq2rUzvh2-7nYK=NpKHAXXEwc-W@mail.gmail.com>
	<AANLkTinpDt+NrcoK8pwWuuwMufut+DTC7GSrnAdWYhJ5@mail.gmail.com>
Message-ID: <AANLkTi=O2R3oqcSDOUxh3g-mqKJ_E-rgrLZ210a_SeJi@mail.gmail.com>

I'm not sure where to check, but there's entries like this in
freeswitch.log.2011-02-21-23-40-11.1

2011-02-21 17:22:53.157390 [NOTICE] mod_logfile.c:158 New log started.
2011-02-21 17:22:53.157390 [NOTICE] switch_cpp.cpp:1181 Played the message
at least once and checked for VM, we should be exiting the loop.
2011-02-21 17:22:53.157390 [NOTICE] switch_cpp.cpp:1181 *************
SETTING SESSION VARIABLES ***************
2011-02-21 17:22:53.157390 [NOTICE] switch_cpp.cpp:1181 Session
Variable[profile_id]: 1
2011-02-21 17:22:53.157390 [NOTICE] switch_cpp.cpp:1181 ************* DONE
SETTING SESSION VARIABLES ***********
2011-02-21 17:22:53.377395 [NOTICE] switch_cpp.cpp:1181 Playing file:
/usr/local/freeswitch/recordings/messages/16c0f890_c35e33c0_777973.wav
2011-02-21 17:22:53.377395 [DEBUG] switch_ivr_play_say.c:1186 Codec
Activated L16 at 8000hz 1 channels 20ms
2011-02-21 17:23:03.717511 [DEBUG] switch_ivr_play_say.c:1515 done playing
file
2011-02-21 17:23:03.717511 [NOTICE] switch_cpp.cpp:1181 !!!!! Finished
playing the file !!!!!
2011-02-21 17:23:03.717511 [INFO] switch_cpp.cpp:1181 Was VM detected? no
2011-02-21 17:23:03.717511 [NOTICE] switch_cpp.cpp:1181 Played the message
at least once and checked for VM, we should be exiting the loop.
2011-02-21 17:23:03.717511 [NOTICE] switch_cpp.cpp:1181 *************
SETTING SESSION VARIABLES ***************
2011-02-21 17:23:03.717511 [NOTICE] switch_cpp.cpp:1181 Session
Variable[profile_id]: 1
2011-02-21 17:23:03.717511 [NOTICE] switch_cpp.cpp:1181 ************* DONE
SETTING SESSION VARIABLES ***********
2011-02-21 17:23:03.937661 [NOTICE] switch_cpp.cpp:1181 Playing file:
/usr/local/freeswitch/recordings/messages/16c0f890_c35e33c0_777973.wav
2011-02-21 17:23:03.937661 [DEBUG] switch_ivr_play_say.c:1186 Codec
Activated L16 at 8000hz 1 channels 20ms
2011-02-21 17:23:08.237260 [DEBUG] switch_ivr_play_say.c:1515 done playing
file
2011-02-21 17:23:08.237260 [NOTICE] switch_cpp.cpp:1181 !!!!! Finished
playing the file !!!!!
2011-02-21 17:23:08.237260 [INFO] switch_cpp.cpp:1181 Was VM detected? no
2011-02-21 17:23:08.237260 [NOTICE] switch_cpp.cpp:1181 Played the message
at least once and checked for VM, we should be exiting the loop.
2011-02-21 17:23:08.237260 [NOTICE] switch_cpp.cpp:1181 *************
SETTING SESSION VARIABLES ***************
2011-02-21 17:23:08.237260 [NOTICE] switch_cpp.cpp:1181 Session
Variable[profile_id]: 1
2011-02-21 17:23:08.237260 [NOTICE] switch_cpp.cpp:1181 ************* DONE
SETTING SESSION VARIABLES ***********
2011-02-21 17:23:08.437885 [NOTICE] switch_cpp.cpp:1181 Playing file:
/usr/local/freeswitch/recordings/messages/16c0f890_c35e33c0_777973.wav
2011-02-21 17:23:08.437885 [DEBUG] switch_ivr_play_say.c:1186 Codec
Activated L16 at 8000hz 1 channels 20ms
2011-02-21 17:23:08.857270 [DEBUG] switch_ivr_play_say.c:1515 done playing
file
2011-02-21 17:23:08.857270 [NOTICE] switch_cpp.cpp:1181 !!!!! Finished
playing the file !!!!!
2011-02-21 17:23:08.857270 [INFO] switch_cpp.cpp:1181 Was VM detected? no
2011-02-21 17:23:08.857270 [NOTICE] switch_cpp.cpp:1181 Played the message
at least once and checked for VM, we should be exiting the loop.
2011-02-21 17:23:08.857270 [NOTICE] switch_cpp.cpp:1181 *************
SETTING SESSION VARIABLES ***************
2011-02-21 17:23:08.857270 [NOTICE] switch_cpp.cpp:1181 Session
Variable[profile_id]: 1
2011-02-21 17:23:08.857270 [NOTICE] switch_cpp.cpp:1181 ************* DONE
SETTING SESSION VARIABLES ***********
2011-02-21 17:23:09.077014 [NOTICE] switch_cpp.cpp:1181 Playing file:
/usr/local/freeswitch/recordings/messages/16c0f890_c35e33c0_777973.wav
2011-02-21 17:23:09.077014 [DEBUG] switch_ivr_play_say.c:1186 Codec
Activated L16 at 8000hz 1 channels 20ms
2011-02-21 17:23:19.417631 [DEBUG] switch_ivr_play_say.c:1515 done playing
file
2011-02-21 17:23:19.417631 [NOTICE] switch_cpp.cpp:1181 !!!!! Finished
playing the file !!!!!
2011-02-21 17:23:19.417631 [INFO] switch_cpp.cpp:1181 Was VM detected? no
2011-02-21 17:23:19.417631 [NOTICE] switch_cpp.cpp:1181 Played the message
at least once and checked for VM, we should be exiting the loop.

I also have the Freeswitch log from the date too, but I'm not sure what I
should post here that would be useful.



On Tue, Feb 22, 2011 at 8:20 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> do you have any console logs of it failing or any error messages etc?
>
>
> On Tue, Feb 22, 2011 at 1:22 AM, Michael De Lorenzo
> <delorenzodesign at gmail.com> wrote:
> > Hello,
> >
> > I have a Freeswitch installation that is intended to make many calls
> > (thousands) and playback a single wav file.  The calls are successfully
> > processed (the recipient's phone rings), but the call almost immediately
> > disconnects, after about 1s.  Everything seems to work fine if I'm only
> > pushing one or two calls through the Freeswitch instance, but as soon as
> I
> > turn up the call rate (I'm still only doing about 50 concurrent sessions)
> > the playback begins to fail.
> >
> > I've watched the calls go out from the console and nothing looks out of
> the
> > ordinary, except that the calls are disconnected with NORMAL CLEARING
> prior
> > to completion.
> >
> > Here's the Lua script I'm using...
> >
> > profile_id = argv[1];
> > account_code = argv[2];
> > client_id = argv[3];
> > caller_id_name = argv[4];
> > caller_id = argv[5];
> > dial_id = argv[6];
> > number_to_call = argv[7];
> > message_to_play = argv[8];
> > max_retries = argv[9];
> > retry_interval = argv[10];
> >
> > local human_detected = false
> > local voicemail_detected = false;
> > local message_played = false;
> >
> > recordings_directory = "/usr/local/freeswitch/recordings/messages/";
> >
> > function setDialVariables(set_as_session_variables)
> >     local s = "profile_id=" .. profile_id;
> >     s = s .. ",account_code=" .. account_code;
> >     s = s .. ",client_id=" .. client_id;
> >     s = s .. ",caller_id_name=" .. caller_id_name;
> >     s = s .. ",caller_id=" .. caller_id;
> >     s = s .. ",dial_id=" .. dial_id;
> >     s = s .. ",number_to_call=" .. number_to_call;
> >     s = s .. ",message_to_play=" .. message_to_play;
> >
> >     freeswitch.consoleLog("notice", s .. "\n");
> >
> >     return s
> > end
> >
> > function printSessionVariables()
> >     freeswitch.consoleLog("notice", "******* PRINTING SESSION VARIABLES
> > **********\n");
> >     -- ommitted
> >     freeswitch.consoleLog("notice",
> > "**********************************************\n");
> > end
> >
> > function onInput(s, type, obj, arg)
> >     if(type == "event" and voicemail_detected == false) then
> >         freeswitch.consoleLog("debug","************ VOICE MAIL/ANSWERING
> > MACHINE DETECTED *************\n");
> >         voicemail_detected = true;
> >         return "break";
> >     end
> >     return true;
> > end
> >
> > function playbackMessage(sleepTime)
> >     message_played = false;
> >     session:sleep(sleepTime);
> >     -- play a file
> >     message_file = recordings_directory .. message_to_play;
> >     freeswitch.consoleLog("notice", "Playing file: " .. message_file ..
> > "\n");
> >     session:streamFile(message_file);
> >     freeswitch.consoleLog("notice", "!!!!! Finished playing the file
> > !!!!!\n");
> >     message_played = true;
> > end
> >
> > session = freeswitch.Session("{" .. setDialVariables(false) ..
> > ",ignore_early_media=true,origination_caller_id_name=" .. caller_id_name
> ..
> > ",origination_caller_id_number=+1" .. caller_id ..
> "}sofia/gateway/gateway_"
> > .. profile_id .. "/" .. number_to_call);
> >
> > while(session:ready()) do
> >    setDialVariables(true)
> >    session:answer();
> >
> >    -- session:execute("continue_on_fail","true");
> >    session:setInputCallback("onInput","true");
> >    session:execute("avmd","start");
> >
> >    playbackMessage(200);
> >
> >    vm_status = voicemail_detected == true and "yes" or "no"
> >    freeswitch.consoleLog("info", "Was VM detected? " .. vm_status ..
> "\n");
> >    if(voicemail_detected) then
> >       return "break";
> >    end
> >
> >    freeswitch.consoleLog("notice", "Played the message at least once and
> > checked for VM, we should be exiting the loop.\n")
> > end
> >
> > if (voicemail_detected) then
> >     freeswitch.consoleLog("info", "Playback for voicemail.\n");
> >     session:execute("avmd","stop");
> >     playbackMessage(5000);
> > end
> >
> > freeswitch.consoleLog("info", "All finished, hanging up the session.\n");
> > session:hangup();
> >
> > Any help would be greatly appreciated.
> >
> > Thank you,
> >
> > 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
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 u2nsam at gmail.com  Wed Feb 23 09:41:18 2011
From: u2nsam at gmail.com (Sam)
Date: Wed, 23 Feb 2011 12:11:18 +0530
Subject: [Freeswitch-users] mod_odbc_query
In-Reply-To: <AANLkTimcO+i0qSJZH4CB3vnEM3mt0noi5JB0Ph0pbKH9@mail.gmail.com>
References: <AANLkTimZbs8S5sgkS7z=6+BDjW3f1U=LmO03203WZYUC@mail.gmail.com>
	<AANLkTinxvwoptnvXmqFEokvQikHNzspTcXvJLndFJLjv@mail.gmail.com>
	<AANLkTi=+N4azher3v4u-AxFP_34OXZYiC+osHZhUc7fK@mail.gmail.com>
	<AANLkTimcO+i0qSJZH4CB3vnEM3mt0noi5JB0Ph0pbKH9@mail.gmail.com>
Message-ID: <AANLkTi==ifxSByttmvT9BQj0MJCYxESS4BnLN8egD9e+@mail.gmail.com>

I am getting this error while loading mod_odbc_query


2011-02-23 12:01:58.487072 [ERR] mod_odbc_query.c:489 No core ODBC
available!
2011-02-23 12:01:58.487072 [CRIT] switch_loadable_module.c:928 Error Loading
module /usr/local/fs/novanet/mod//mod_odbc_query.so
**Module load routine returned an error**

Regds
Sam




On Mon, Feb 21, 2011 at 6:36 PM, Avi Marcus <avi at avimarcus.net> wrote:

> Like the wiki says, you need to pull the git CONTRIB. There's a link in
> that wiki page..
> -Avi
>
>
> On Mon, Feb 21, 2011 at 2:52 PM, Sam <u2nsam at gmail.com> wrote:
>
>> Do i need to pull down git for it ?
>>
>> regds
>> Sam
>>
>>
>> On Mon, Feb 21, 2011 at 3:47 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> You can find the download / compile instructions on the wiki:
>>> http://wiki.freeswitch.org/wiki/Mod_odbc_query
>>>
>>> <http://wiki.freeswitch.org/wiki/Mod_odbc_query>-Avi
>>>
>>> On Mon, Feb 21, 2011 at 6:30 AM, Sam <u2nsam at gmail.com> wrote:
>>>
>>>> Hello,
>>>>
>>>> How to download mod_odbc_query source, there is not such source in 1.0.6
>>>> or 1.0.7 .
>>>>
>>>> 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
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Feb 23 09:55:31 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 23 Feb 2011 07:55:31 +0100
Subject: [Freeswitch-users] mod_odbc_query
Message-ID: <7F45D19B-E224-4AAC-94A7-B3C1D11FA293@visionutveckling.se>

Rebuild FS with ODBC support, you were probably missing some ODBC devel files when configuring and building FS.

/Peter

----- Reply message -----
Fr?n: "Sam" <u2nsam at gmail.com>
Datum: ons, feb 23, 2011 07:48
Rubrik: [Freeswitch-users] mod_odbc_query
Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

I am getting this error while loading mod_odbc_query


2011-02-23 12:01:58.487072 [ERR] mod_odbc_query.c:489 No core ODBC available!
2011-02-23 12:01:58.487072 [CRIT] switch_loadable_module.c:928 Error Loading module /usr/local/fs/novanet/mod//mod_odbc_query.so
**Module load routine returned an error**

Regds
Sam




On Mon, Feb 21, 2011 at 6:36 PM, Avi Marcus <avi at avimarcus.net<mailto:avi at avimarcus.net>> wrote:
Like the wiki says, you need to pull the git CONTRIB. There's a link in that wiki page..
-Avi


On Mon, Feb 21, 2011 at 2:52 PM, Sam <u2nsam at gmail.com<mailto:u2nsam at gmail.com>> wrote:
Do i need to pull down git for it ?

regds
Sam


On Mon, Feb 21, 2011 at 3:47 PM, Avi Marcus <avi at avimarcus.net<mailto:avi at avimarcus.net>> wrote:
You can find the download / compile instructions on the wiki:
http://wiki.freeswitch.org/wiki/Mod_odbc_query

<http://wiki.freeswitch.org/wiki/Mod_odbc_query>-Avi

On Mon, Feb 21, 2011 at 6:30 AM, Sam <u2nsam at gmail.com<mailto:u2nsam at gmail.com>> wrote:
Hello,

How to download mod_odbc_query source, there is not such source in 1.0.6 or 1.0.7 .

Regds
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


!DSPAM:4d64ad4632761097013501!


From jaybinks at gmail.com  Wed Feb 23 09:58:01 2011
From: jaybinks at gmail.com (jay binks)
Date: Wed, 23 Feb 2011 16:58:01 +1000
Subject: [Freeswitch-users]  SIP DoS Prevention - Patch
Message-ID: <AANLkTik7V5cRe=SNQW9Xq7JdugWeONn5mubM=zPdgNuQ@mail.gmail.com>

check out my patch Ive just posted to jira.
http://jira.freeswitch.org/browse/FS-3094

you can subscribe to "Auth" events now - SWITCH_EVENT_AUTH
which are generic for Register & Invite ( and any other auth'd packet I
guess )
and there is an auth_state which will tell you if its an auth fail or a
expire etc..
you can get the to_user, to_host, useragent, sip profile and network_ip

Auth Expires also have a Log messages similar to Auth Failures ( fail2ban
compatible )

PLEASE guys give this some testing.  I only banged this together in a few
hours today in my currently sleep deprived state.  So im sure its going to
have things that can be improved or work better, but at least the bones of
it are there now.

let me know your thoughts.

Jay



On Mon, Feb 21, 2011 at 11:08 PM, Jmesquita at freeswitch.org <
jmesquita at freeswitch.org> wrote:

> Just an idea for the other developers. Wouldn't be cool to have and event
> thrown by each module that does authentication so that other application
> modules are able to listen in those events and do whatever with it? Some
> people might even like to have that running an eel daemon and throw some
> snmp traps based on that?
>
> Just an idea that might be completely ridiculous or might raise some
> interest. Events are pretty cheap so there wouldn't be a lot of overhead
> involved.
>
> Regards,
>
> Jo?o Mesquita
>
> On 21/02/2011, at 07:28, Spencer Thomason <spencer at 5ninesolutions.com>
> wrote:
>
> > After tinkering with it, I think that might be the best way.  The
> > iptables method is cool but I'd like to have more dynamic control and
> > with Fail2Ban looking at the challenges you could specifically ignore
> > certain high traffic IPs and block others.  What would be very cool is
> > if instead of logging every challenge, a log entry was written if
> > there was a high number from a specific IP, then you could decide what
> > to do about it with fail2ban, similar to the pike module for opensips
> > does.
> >
> >
> > On Feb 21, 2011, at 1:31 AM, covici at ccs.covici.com wrote:
> >
> >> I would change sip auth failure to challenge and then have sufficient
> >> times to only block if there are too many challenges in a certain
> >> time.
> >> I am not even sure the failure works any more in recent gits.
> >>
> >> Spencer Thomason <spencer at 5ninesolutions.com> wrote:
> >>
> >>> Yes, that works great if they respond to the challenge with a failed
> >>> auth. But the scenario I'm trying to prevent is if they just send the
> >>> INVITE and never respond to the challenge.  Fail2Ban will not work as
> >>> every endpoint will initially send an INVITE and receive a challenge.
> >>> Legit calls will then respond correctly and not be logged as a SIP
> >>> auth failure but every call that is challenged will show up as SIP
> >>> auth challenge in the logs so there is no regex to differentiate
> >>> between legit an non legit traffic.
> >>>
> >>> Spencer
> >>>
> >>> On Feb 20, 2011, at 10:39 PM, Ken Rice wrote:
> >>>
> >>>> Fail2Ban ... This is block an IP with too many failed attempts from
> >>>> something like SipVicious pretty quickly
> >>>>
> >>>>
> >>>> On 2/20/11 11:07 PM, "Spencer Thomason" <spencer at 5ninesolutions.com>
> >>>> wrote:
> >>>>
> >>>>> Hi,
> >>>>> We run hosted Freeswitch instances in VMs with the internal profile
> >>>>> on
> >>>>> port 5060 connecting to clients mostly behind NAT and then the
> >>>>> external profile connecting to our proxies only.  Protecting the
> >>>>> external profile its straightforward.. we only allow traffic to/
> >>>>> from
> >>>>> our proxies at the firewall level.  But protecting the internal
> >>>>> profile seems to be a bit more difficult because the UACs could be
> >>>>> theoretically anywhere on the network.
> >>>>>
> >>>>> I'm currently using Fail2Ban to prevent brute force registration
> >>>>> and
> >>>>> INVITEs on auth failures, e.g.:
> >>>>> failregex = \[WARNING\] sofia_reg.c:\d+ SIP auth failure \
> >>>>> (REGISTER\)
> >>>>> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
> >>>>>           \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(INVITE\)
> >>>>> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
> >>>>>
> >>>>> My question is, since its part of a normal SIP dialog to challenge
> >>>>> the
> >>>>> INVITE, is there any way to prevent a possible DoS from just sheer
> >>>>> volume of incoming INVITEs on an Internet facing server
> >>>>> automatically.  I.e., If you block the logged challenge, you'd
> >>>>> block
> >>>>> all legitimate INVITEs and registrations.  Since its UDP traffic I
> >>>>> couldn't come up with a way to do it automatically at the iptables
> >>>>> level. i.e. number of concurrent connections.  Is there some option
> >>>>> to
> >>>>> just not respond if a client is sending a number of requests over a
> >>>>> certain threshold?  It might not stop them from sending the traffic
> >>>>> but pretty soon they'd get the idea that it wasn't going to go
> >>>>> anywhere.  My concern is say there are 50 Freeswitch instances on a
> >>>>> box (albeit 8 core, 32GB ram, 8 15K raid 10 storage) and someone
> >>>>> starts sending thousands of rouge INVITEs to every VM on a physical
> >>>>> box that the CPU load from just challenging the incoming INVITEs
> >>>>> would
> >>>>> create a DoS.  We the logs regularly to try to catch people doing
> >>>>> this
> >>>>> sort of thing and drop them at a router upstream of the core
> >>>>> network,
> >>>>> but I'd like to have it happen without human intervention.  Have I
> >>>>> completely over thought this and am missing something obvious?
> >>>>>
> >>>>> Thanks,
> >>>>> 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
> >>>>
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at 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
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 herman.griffin at gmail.com  Wed Feb 23 10:39:36 2011
From: herman.griffin at gmail.com (Herman Griffin)
Date: Tue, 22 Feb 2011 23:39:36 -0800
Subject: [Freeswitch-users] Using outbound Event Sockets versus using embed
	language scripts
Message-ID: <AANLkTimyEsRcK02wzbKqWfjbXz-YJXHjSCTi6kXawufU@mail.gmail.com>

Unless I'm complicating things, it seems like a drop to write outbound event
sockets apps versus writing embedded language apps. However, I'm interested
in writing event sockets apps because I can see an advantage in being able
to load balance the app by running you app behind a load balancer. One
simple thing that I am trying to accomplish is collecting digits and doing
something useful with string of digits that have been collected.

My first attempt at this is to use the play_and_get_digits dptool, but I
don't now how to pull data from stored variable so that I can use in the
script. I slightly modified the
freeswitch.git/libs/esl/perl/server2.plscript by adding this line:

$con->execute('play_and_get_digits', '2 5 3 7000 #
${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav
foobar \d+' );

The next thing that I'd like to do is grab the value in foobar is use it in
the perl logic.

Can someone lead me to the next step?
Does anyone with experience with event socket apps and embedded language
apps have some useful information about their preferred path?

Thanks,
Herman
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From steveayre at gmail.com  Wed Feb 23 11:56:27 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 23 Feb 2011 08:56:27 +0000
Subject: [Freeswitch-users] RTP Proxy config and DTMF conversion problem
In-Reply-To: <AANLkTikyYbkmUOCUOk90tb=Fq0p_BewoAK4dtzR5L-fa@mail.gmail.com>
References: <AANLkTimW23Uf61OU2RVJWD6HAM9i_towb2GocwYsi7aC@mail.gmail.com>
	<AANLkTikyYbkmUOCUOk90tb=Fq0p_BewoAK4dtzR5L-fa@mail.gmail.com>
Message-ID: <AANLkTikKufEb=ag4smQgKpSYVUiO2kT52cEr_FU-6R+c@mail.gmail.com>

Exactly,

Mac - I already answered this and pointed out this was your problem in your
previous thread.

There are three media modes:
  Standard - Media flows through FS, full processing options (the default)
  Proxy - Media flows through FS *no* processing options, just passed
straight through
  Bypass - Media flows around FS

If you want to do anything that requires making a change to the media you
have to use Standard.

This is equivalent to:
<action application="set" data="proxy_media=*false*"/>
<action application="set" data="bypass_media=*false*"/>

Since it's the default those lines are *not* required (leave them out
completely) unless you've set either of those modes on your SIP profile in
which case it would override the profile's parameter.

Steve



On 23 February 2011 00:16, Anthony Minessale <anthony.minessale at gmail.com>wrote:

> just because you want to make a proxy and there is a mode with the
> name proxy in it, you do not have to use it.
> proxy_media mode is not going to work with inband detection because
> then you are looking at the media and not really proxying it.
>
>
>
> On Sun, Feb 20, 2011 at 11:46 AM, Mac <validator77 at gmail.com> wrote:
> > Dear ALL,
> >
> > Thats my first post here. I am struggling for some time with DTMF issue.
> > Let me introduce you my configuration.
> >
> > The main task is to configure RTP Proxy with full topology hiding -
> > OPERATOR_1 ------- Freeswitch ------- OPERATOR_2
> > Here is a prt of my config:
> > - sip profile.
> > I have one new sip profile with two gateways (OPERATOR_1 and OPERATOR_2)
> > <include>
> >   <gateway name="OP1">
> >     <param name="username" value="X1.Y1.V1.Z1"/>
> >     <param name="password" value="09321234"/>
> >     <param name="proxy" value="X1.Y1.V1.Z1"/>
> >     <param name="from-domain" value="test.com"/>
> >     <param name="register" value="false"/>
> >     <param name="caller-id-in-from" value="true"/>
> >     <param name="extension" value="test"/>
> >     <param name="extension-in-contact" value="true"/>
> >     <param name="outbound-codec-prefs" value="$${outbound_codec_
> > prefs}"/>
> >     <param name="disable-transcoding" value="true"/>
> >     <param name="inbound-late-negotiation" value="true"/>
> >   </gateway>
> > </include>
> >
> > - dialplan
> > <include>
> >   <context name="public">
> >         <extension name="term">
> >                 <condition field="network_addr"
> > expression="^(X1\.Y1\.V1\.Z1|X2\.Y2\.V1\.Z1)$" break="on-false"/>
> >                 <condition field="destination_number"
> > expression="^46(\d{9})$">
> >                 <action application="set" data="proxy_media=true"/>
> >                 <action application="set" data="bypass_media=false"/>
> >                 <action application="start_dtmf"/>
> >                 <action application="bridge"
> data="sofia/gateway/OP1/$0"/>
> >                 </condition>
> >         </extension>
> >   </context>
> > </include>
> >
> > Everything is fine, but i have problem with DTMF conversion from RFC2833
> to
> > inband.
> > Refering to
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf
> > and http://wiki.freeswitch.org/wiki/Proxy_Media i have done all the
> > necessary things.
> >
> > vars.xml
> >   <X-PRE-PROCESS cmd="set"
> data="global_codec_prefs=PCMA,G.729,PCMU,GSM"/>
> >   <X-PRE-PROCESS cmd="set"
> data="outbound_codec_prefs=PCMA,G.729,PCMU,GSM"/>
> >
> > The after-effect is following output:
> > [ERR] switch_core_io.c:1055 Codec PROXY PASS-THROUGH encoder error!
> >
> > Placing  <param name="dtmf-type" value="none"/> does not help anyway.
> >
> > Could sb point me where the problem is located?
> >
> > Thanks in advance,
> > Mac
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 steveayre at gmail.com  Wed Feb 23 12:04:10 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 23 Feb 2011 09:04:10 +0000
Subject: [Freeswitch-users] RTP Proxy config and DTMF conversion problem
In-Reply-To: <AANLkTikKufEb=ag4smQgKpSYVUiO2kT52cEr_FU-6R+c@mail.gmail.com>
References: <AANLkTimW23Uf61OU2RVJWD6HAM9i_towb2GocwYsi7aC@mail.gmail.com>
	<AANLkTikyYbkmUOCUOk90tb=Fq0p_BewoAK4dtzR5L-fa@mail.gmail.com>
	<AANLkTikKufEb=ag4smQgKpSYVUiO2kT52cEr_FU-6R+c@mail.gmail.com>
Message-ID: <AANLkTikA++2sFyQ98V=kzwf4mVhqFthOkNJYXBC3Pb0K@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Proxy_Media#Why_you_almost_certainly_don.27t_want_to_use_it


On 23 February 2011 08:56, Steven Ayre <steveayre at gmail.com> wrote:

> Exactly,
>
> Mac - I already answered this and pointed out this was your problem in your
> previous thread.
>
> There are three media modes:
>   Standard - Media flows through FS, full processing options (the default)
>   Proxy - Media flows through FS *no* processing options, just passed
> straight through
>   Bypass - Media flows around FS
>
> If you want to do anything that requires making a change to the media you
> have to use Standard.
>
> This is equivalent to:
> <action application="set" data="proxy_media=*false*"/>
>
> <action application="set" data="bypass_media=*false*"/>
>
> Since it's the default those lines are *not* required (leave them out
> completely) unless you've set either of those modes on your SIP profile in
> which case it would override the profile's parameter.
>
> Steve
>
>
>
>
> On 23 February 2011 00:16, Anthony Minessale <anthony.minessale at gmail.com>wrote:
>
>> just because you want to make a proxy and there is a mode with the
>> name proxy in it, you do not have to use it.
>> proxy_media mode is not going to work with inband detection because
>> then you are looking at the media and not really proxying it.
>>
>>
>>
>> On Sun, Feb 20, 2011 at 11:46 AM, Mac <validator77 at gmail.com> wrote:
>> > Dear ALL,
>> >
>> > Thats my first post here. I am struggling for some time with DTMF issue.
>> > Let me introduce you my configuration.
>> >
>> > The main task is to configure RTP Proxy with full topology hiding -
>> > OPERATOR_1 ------- Freeswitch ------- OPERATOR_2
>> > Here is a prt of my config:
>> > - sip profile.
>> > I have one new sip profile with two gateways (OPERATOR_1 and OPERATOR_2)
>> > <include>
>> >   <gateway name="OP1">
>> >     <param name="username" value="X1.Y1.V1.Z1"/>
>> >     <param name="password" value="09321234"/>
>> >     <param name="proxy" value="X1.Y1.V1.Z1"/>
>> >     <param name="from-domain" value="test.com"/>
>> >     <param name="register" value="false"/>
>> >     <param name="caller-id-in-from" value="true"/>
>> >     <param name="extension" value="test"/>
>> >     <param name="extension-in-contact" value="true"/>
>> >     <param name="outbound-codec-prefs" value="$${outbound_codec_
>> > prefs}"/>
>> >     <param name="disable-transcoding" value="true"/>
>> >     <param name="inbound-late-negotiation" value="true"/>
>> >   </gateway>
>> > </include>
>> >
>> > - dialplan
>> > <include>
>> >   <context name="public">
>> >         <extension name="term">
>> >                 <condition field="network_addr"
>> > expression="^(X1\.Y1\.V1\.Z1|X2\.Y2\.V1\.Z1)$" break="on-false"/>
>> >                 <condition field="destination_number"
>> > expression="^46(\d{9})$">
>> >                 <action application="set" data="proxy_media=true"/>
>> >                 <action application="set" data="bypass_media=false"/>
>> >                 <action application="start_dtmf"/>
>> >                 <action application="bridge"
>> data="sofia/gateway/OP1/$0"/>
>> >                 </condition>
>> >         </extension>
>> >   </context>
>> > </include>
>> >
>> > Everything is fine, but i have problem with DTMF conversion from RFC2833
>> to
>> > inband.
>> > Refering to
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf
>> > and http://wiki.freeswitch.org/wiki/Proxy_Media i have done all the
>> > necessary things.
>> >
>> > vars.xml
>> >   <X-PRE-PROCESS cmd="set"
>> data="global_codec_prefs=PCMA,G.729,PCMU,GSM"/>
>> >   <X-PRE-PROCESS cmd="set"
>> data="outbound_codec_prefs=PCMA,G.729,PCMU,GSM"/>
>> >
>> > The after-effect is following output:
>> > [ERR] switch_core_io.c:1055 Codec PROXY PASS-THROUGH encoder error!
>> >
>> > Placing  <param name="dtmf-type" value="none"/> does not help anyway.
>> >
>> > Could sb point me where the problem is located?
>> >
>> > Thanks in advance,
>> > Mac
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at 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 steveayre at gmail.com  Wed Feb 23 12:17:05 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 23 Feb 2011 09:17:05 +0000
Subject: [Freeswitch-users] OpenSER interface to FreeSWITCH
In-Reply-To: <COL111-W442B2A5D707F79AF2A8338ACDB0@phx.gbl>
References: <COL111-W442B2A5D707F79AF2A8338ACDB0@phx.gbl>
Message-ID: <AANLkTik6b9ULcxYRFf_bbLp13gx6evZ8SGPex-TPKfMU@mail.gmail.com>

Jason, you seem to be misunderstanding what freeswitch is.

It does not purely handle media. mediaproxy is the closest thing that would
do that, and I believe it does that by watching the sip traffic flowing
through the machine and grabbing the IP/ports to set up the flow, but with
no options to do any processing.

FreeSWITCH is a B2BUA. You make a call into FS via SIP (or other supported
protocol). The application then runs within FS (or via ESL). FS handles all
signalling and media.

Your application should never handle SIP. The only reason to use a SIP proxy
such as OpenSER in front of FS would be load balancing.

-Steve



On 23 February 2011 02:35, Jason b.a <k-b-24 at live.com> wrote:

>  yeh sure i have a SIP endpoints , can the application do the sip handling
> and have a socket interface with freeswitch in the same time without using
> OpenSER.
> in this case i need Sip servelet plugin in my application , is it possible.
> I went through the outbound event socket , seems to be helpful thx , also i
> saw the options for using java to connect to freeswitch, i think using ESL ,
> i can build a java IVR 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
>
>
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From dmitry.bely at gmail.com  Wed Feb 23 13:29:25 2011
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Wed, 23 Feb 2011 13:29:25 +0300
Subject: [Freeswitch-users] Auto-nat affects all profiles?
In-Reply-To: <B43954E4-8AF0-48F3-A04B-BAA0F8D7E1F0@freeswitch.org>
References: <AANLkTi=uw63jzwqvWwnpBwVgbDm8CO9CzTgg8K2oa+nF@mail.gmail.com>
	<B43954E4-8AF0-48F3-A04B-BAA0F8D7E1F0@freeswitch.org>
Message-ID: <AANLkTi=czS_fQo=tU-Greg0eSaneqe1xw2x4JAwLpGT4@mail.gmail.com>

On Wed, Feb 23, 2011 at 1:01 AM, Brian West <brian at freeswitch.org> wrote:
> only works on one profile. ?Not designed to work with more than one.

It's a bit strange as FreeSWITCH default configs contain 2 profiles
(internal and external), Auto-nat is not designed for the default
configuration?

I still think it should be fixed. Then you have

internal profile:
<param name="ext-sip-ip" value="auto"/>
<param name="sip-port" value="5060"/>

external profile:
<param name="ext-sip-ip" value="auto-nat"/>
<param name="sip-port" value="5080"/>

you don't expect that port 5060 is silently exposed to the Internet.

> On Feb 22, 2011, at 6:58 AM, Dmitry Bely wrote:
>
>> I have enabled auto-nat only for the external profile (port 5080), but
>> FreeSWITCH also maps SIP port for the internal one:
>>
>> freeswitch at internal> show nat_map
>> port,proto,proto_num,sticky
>> 5060,udp,0,0
>> 5060,tcp,1,0
>> 5080,udp,0,0
>> 5080,tcp,1,0
>>
>> 4 total.
>>
>> Is this intended or just a bug?

- Dmitry Bely


From dujinfang at gmail.com  Wed Feb 23 15:00:16 2011
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 23 Feb 2011 20:00:16 +0800
Subject: [Freeswitch-users] outbound-caller-name missing?
Message-ID: <AANLkTimZnnZYrHmaHVX3dW4KA4kfksEX=zaaokuwLYb-@mail.gmail.com>

Hi,


I tried

FreeSWITCH Version 1.0.head (git-f5dafc9 2011-01-21 23-42-30 +0100)
FreeSWITCH Version 1.0.head (git-06988e1 2011-02-20 15-10-02 -0600)


originate user/1000 &echo got this:

   INVITE sip:tbskvnyi at 192.168.7.2:61901 SIP/2.0
   Via: SIP/2.0/UDP 192.168.7.2;rport;branch=z9hG4bK0USe34K3r46gF
   Max-Forwards: 70
   From: "" <sip:0000000000 at 192.168.7.2>;tag=cZt7ZSBZjQK3H
   Remote-Party-ID:
<sip:0000000000 at 192.168.7.2>;party=calling;screen=yes;privacy=off


And some versions of Xlite4 fails to respond to this

originate {origination_caller_id_name=xxx}user/1000 works.

I haven't got time to test git head (02-20 almost head),  just want to
ask here anyone noticed this, let me know if need a jira.

btw, vars.xml is untouched:

  <X-PRE-PROCESS cmd="set" data="outbound_caller_name=FreeSWITCH"/>
  <X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/>

Thanks.

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


From david.villasmil.work at gmail.com  Wed Feb 23 17:03:22 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 23 Feb 2011 15:03:22 +0100
Subject: [Freeswitch-users] Weird CDR ERROR...
Message-ID: <AANLkTikVW7K=tvsrFa7azTEtU8CKso+fTiSj6sw6yhAf@mail.gmail.com>

Hello All,

I have an XML_CDR cdr that is not correct, i made some test and on one cdr
(that i could find) which show endpoint_disposition as ANSWER when the call
was NOT answered! The call didn't even go out of the box...

here's the CDR:

note that billsec is 0, but disposition is ANSWER... weird, i got more than
one, also...



<cdr>
 <channel_data>
   <state>CS_REPORTING</state>
   <direction>inbound</direction>
   <state_number>11</state_number>
   <flags>0=1;1=1;3=1;35=1;36=1;38=1;46=1;51=1</flags>
   <caps>1=1;2=1;3=1</caps>
 </channel_data>
 <variables>
   <uuid>5d75e976-3ed9-11e0-a60b-fba7243e1308</uuid>
   <sip_local_network_addr>1.2.3.4</sip_local_network_addr>
   <sip_network_ip>1.2.3.4</sip_network_ip>
   <sip_network_port>9774</sip_network_port>
   <sip_received_ip>1.2.3.4</sip_received_ip>
   <sip_received_port>9774</sip_received_port>
   <sip_via_protocol>udp</sip_via_protocol>
   <sip_from_user>8889990</sip_from_user>
   <sip_from_port>5060</sip_from_port>
   <sip_from_uri>8889990%1.2.3.4%3A5060</sip_from_uri>
   <sip_from_host>1.2.3.4</sip_from_host>
   <sip_from_user_stripped>8889990</sip_from_user_stripped>
   <sip_from_tag>2e41ba6e</sip_from_tag>
   <sofia_profile_name>external</sofia_profile_name>

 <sip_full_via>SIP/2.0/UDP%1.2.3.4%3A9774%3Bbranch%3Dz9hG4bK-d8754z-c551a41eb93bd648-1---d8754z-%3Brport%3D9774%3Breceived%3D1.2.3.4</sip_full_via>
   <sip_from_display>david</sip_from_display>

 <sip_full_from>%22david%22%20%3Csip%3A8889990%1.2.3.4%3A5060%3E%3Btag%3D2e41ba6e</sip_full_from>
   <sip_to_display>0013058883456</sip_to_display>

 <sip_full_to>%220013058883456%22%20%3Csip%3A0013058883456%1.2.3.4%3A5060%3E</sip_full_to>
   <sip_req_user>0013058883456</sip_req_user>
   <sip_req_port>5060</sip_req_port>
   <sip_req_uri>0013058883456%1.2.3.4%3A5060</sip_req_uri>
   <sip_req_host>1.2.3.4</sip_req_host>
   <sip_to_user>0013058883456</sip_to_user>
   <sip_to_port>5060</sip_to_port>
   <sip_to_uri>0013058883456%1.2.3.4%3A5060</sip_to_uri>
   <sip_to_host>1.2.3.4</sip_to_host>
   <sip_contact_user>8889990</sip_contact_user>
   <sip_contact_port>9774</sip_contact_port>
   <sip_contact_uri>8889990%1.2.3.4%3A9774</sip_contact_uri>
   <sip_contact_host>1.2.3.4</sip_contact_host>
   <channel_name>sofia/external/8889990%1.2.3.4%3A5060</channel_name>
   <sip_call_id>OGVkMmMyMmQ5MDEyOTU2MWUyN2M2NzBhYjk5ZmQzOTM.</sip_call_id>

 <sip_user_agent>eyeBeam%20release%201102q%20stamp%2051814</sip_user_agent>
   <sip_via_host>1.2.3.4</sip_via_host>
   <sip_via_port>9774</sip_via_port>
   <sip_via_rport>9774</sip_via_rport>
   <max_forwards>70</max_forwards>

 <switch_r_sdp>v%3D0%0D%0Ao%3D-%207%202%20IN%20IP4%1.2.3.4%0D%0As%3DCounterPath%20eyeBeam%201.5%0D%0Ac%3DIN%20IP4%1.2.3.4%0D%0At%3D0%200%0D%0Am%3Daudio%2023746%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0Aa%3Dalt%3A1%203%20%3A%20L2qFFNvS%20HwniVnkK%1.2.3.4%2023746%0D%0Aa%3Dalt%3A2%202%20%3A%20HWrMm7HE%20Wk82PRBC%1.2.3.4%2023746%0D%0Aa%3Dalt%3A3%201%20%3A%20virMBT30%20GCcqsUuh%1.2.3.4%2023746%0D%0A</switch_r_sdp>
   <sip_use_codec_name>PCMU</sip_use_codec_name>
   <sip_use_codec_rate>8000</sip_use_codec_rate>
   <sip_use_codec_ptime>20</sip_use_codec_ptime>
   <read_codec>PCMU</read_codec>
   <read_rate>8000</read_rate>
   <write_codec>PCMU</write_codec>
   <write_rate>8000</write_rate>
   <outside_call>true</outside_call>
   <hangup_after_bridge>true</hangup_after_bridge>

 <continue_on_fail>NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407</continue_on_fail>

 <current_application_data>sofia/gateway/OPENSIP/0013058883456</current_application_data>
   <current_application>bridge</current_application>
   <local_media_ip>1.2.3.4</local_media_ip>
   <local_media_port>20342</local_media_port>
   <sip_use_pt>0</sip_use_pt>
   <rtp_use_ssrc>820599639</rtp_use_ssrc>
   <remote_media_ip>1.2.3.4</remote_media_ip>
   <remote_media_port>23746</remote_media_port>
   <originate_disposition>SUCCESS</originate_disposition>
   <bridge_channel>sofia/external/0013058883456</bridge_channel>
   <bridge_uuid>5d7e9d6e-3ed9-11e0-a60c-fba7243e1308</bridge_uuid>
   <signal_bond>5d7e9d6e-3ed9-11e0-a60c-fba7243e1308</signal_bond>

 <switch_m_sdp>v%3D0%0D%0Ao%3Droot%2023193%2023194%20IN%20IP4%1.2.3.4%0D%0As%3Dsession%0D%0Ac%3DIN%20IP4%1.2.3.4%0D%0At%3D0%200%0D%0Am%3Daudio%2021586%20RTP/AVP%2018%203%200%208%20101%0D%0Aa%3Drtpmap%3A18%20G729/8000%0D%0Aa%3Dfmtp%3A18%20annexb%3Dno%0D%0Aa%3Drtpmap%3A3%20GSM/8000%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-16%0D%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0D%0Aa%3Dptime%3A20%0D%0A</switch_m_sdp>

 <sip_local_sdp_str>v%3D0%0Ao%3DFreeSWITCH%201298396074%201298396076%20IN%20IP4%1.2.3.4%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%1.2.3.4%0At%3D0%200%0Am%3Daudio%2020342%20RTP/AVP%200%20101%0Aa%3Drtpmap%3A0%20PCMU/8000%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A</sip_local_sdp_str>
   <endpoint_disposition>ANSWER</endpoint_disposition>
   <bridge_hangup_cause>NORMAL_CLEARING</bridge_hangup_cause>
   <hangup_cause>NORMAL_CLEARING</hangup_cause>
   <hangup_cause_q850>16</hangup_cause_q850>
   <digits_dialed>none</digits_dialed>
   <start_stamp>2011-02-23%2000%3A13%3A31</start_stamp>
   <profile_start_stamp>2011-02-23%2000%3A13%3A31</profile_start_stamp>
   <answer_stamp>2011-02-23%2000%3A13%3A36</answer_stamp>
   <progress_media_stamp>2011-02-23%2000%3A13%3A36</progress_media_stamp>
   <end_stamp>2011-02-23%2000%3A13%3A36</end_stamp>
   <start_epoch>1298416411</start_epoch>
   <start_uepoch>1298416411795887</start_uepoch>
   <profile_start_epoch>1298416411</profile_start_epoch>
   <profile_start_uepoch>1298416411795887</profile_start_uepoch>
   <answer_epoch>1298416416</answer_epoch>
   <answer_uepoch>1298416416500657</answer_uepoch>
   <resurrect_epoch>0</resurrect_epoch>
   <resurrect_uepoch>0</resurrect_uepoch>
   <progress_epoch>0</progress_epoch>
   <progress_uepoch>0</progress_uepoch>
   <progress_media_epoch>1298416416</progress_media_epoch>
   <progress_media_uepoch>1298416416470740</progress_media_uepoch>
   <end_epoch>1298416416</end_epoch>
   <end_uepoch>1298416416562577</end_uepoch>
   <last_app>bridge</last_app>
   <last_arg>sofia/gateway/OPENSIP/0013058883456</last_arg>
   <caller_id>%22david%22%20%3C8889990%3E</caller_id>
   <duration>5</duration>
   <billsec>0</billsec>
   <progresssec>0</progresssec>
   <answersec>5</answersec>
   <progress_mediasec>5</progress_mediasec>
   <flow_billsec>5</flow_billsec>
   <mduration>4767</mduration>
   <billmsec>62</billmsec>
   <progressmsec>0</progressmsec>
   <answermsec>4705</answermsec>
   <progress_mediamsec>0</progress_mediamsec>
   <flow_billmsec>4767</flow_billmsec>
   <uduration>4766690</uduration>
   <billusec>61920</billusec>
   <progressusec>0</progressusec>
   <answerusec>4704770</answerusec>
   <progress_mediausec>4674853</progress_mediausec>
   <flow_billusec>4766690</flow_billusec>
   <sip_hangup_disposition>send_bye</sip_hangup_disposition>
   <rtp_audio_in_raw_bytes>0</rtp_audio_in_raw_bytes>
   <rtp_audio_in_media_bytes>0</rtp_audio_in_media_bytes>
   <rtp_audio_in_packet_count>0</rtp_audio_in_packet_count>
   <rtp_audio_in_media_packet_count>0</rtp_audio_in_media_packet_count>
   <rtp_audio_in_skip_packet_count>5</rtp_audio_in_skip_packet_count>
   <rtp_audio_in_jb_packet_count>0</rtp_audio_in_jb_packet_count>
   <rtp_audio_in_dtmf_packet_count>0</rtp_audio_in_dtmf_packet_count>
   <rtp_audio_in_cng_packet_count>0</rtp_audio_in_cng_packet_count>
   <rtp_audio_in_flush_packet_count>0</rtp_audio_in_flush_packet_count>
   <rtp_audio_out_raw_bytes>0</rtp_audio_out_raw_bytes>
   <rtp_audio_out_media_bytes>0</rtp_audio_out_media_bytes>
   <rtp_audio_out_packet_count>0</rtp_audio_out_packet_count>
   <rtp_audio_out_media_packet_count>0</rtp_audio_out_media_packet_count>
   <rtp_audio_out_skip_packet_count>0</rtp_audio_out_skip_packet_count>
   <rtp_audio_out_dtmf_packet_count>0</rtp_audio_out_dtmf_packet_count>
   <rtp_audio_out_cng_packet_count>0</rtp_audio_out_cng_packet_count>
 </variables>
 <app_log>
   <application app_name="set" app_data="outside_call=true"></application>
   <application app_name="info" app_data=""></application>
   <application app_name="set"
app_data="hangup_after_bridge=true"></application>
   <application app_name="set"
app_data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407"></application>
   <application app_name="bridge"
app_data="sofia/gateway/OPENSIP/0013058883456"></application>
 </app_log>
 <callflow dialplan="XML" profile_index="1">
   <extension name="outside_call" number="0013058883456"
current_app="bridge">
     <application app_name="set" app_data="outside_call=true"></application>
     <application app_name="info" app_data=""></application>
     <application app_name="set"
app_data="hangup_after_bridge=true"></application>
     <application app_name="set"
app_data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407"></application>
     <application app_name="bridge"
app_data="sofia/gateway/${distributor(test)}13058883456"></application>
     <application last_executed="true" app_name="bridge"
app_data="sofia/gateway/${distributor(test)}13058883456"></application>
   </extension>
   <caller_profile>
     <username>8889990</username>
     <dialplan>XML</dialplan>
     <caller_id_name>david</caller_id_name>
     <ani>8889990</ani>
     <aniii></aniii>
     <caller_id_number>8889990</caller_id_number>
     <network_addr>1.2.3.4</network_addr>
     <rdnis></rdnis>
     <destination_number>0013058883456</destination_number>
     <uuid>5d75e976-3ed9-11e0-a60b-fba7243e1308</uuid>
     <source>mod_sofia</source>
     <context>public</context>
     <chan_name>sofia/external/8889990 at 1.2.3.4:5060</chan_name>
     <originatee>
       <originatee_caller_profile>
         <username>8889990</username>
         <dialplan>XML</dialplan>
         <caller_id_name>david</caller_id_name>
         <ani>8889990</ani>
         <aniii></aniii>
         <caller_id_number>8889990</caller_id_number>
         <network_addr>1.2.3.4</network_addr>
         <rdnis></rdnis>
         <destination_number>0013058883456</destination_number>
         <uuid>5d7e9d6e-3ed9-11e0-a60c-fba7243e1308</uuid>
         <source>mod_sofia</source>
         <context>public</context>
         <chan_name>sofia/external/0013058883456</chan_name>
       </originatee_caller_profile>
     </originatee>
   </caller_profile>
   <times>
     <created_time>1298416411795887</created_time>
     <profile_created_time>1298416411795887</profile_created_time>
     <progress_time>0</progress_time>
     <progress_media_time>1298416416470740</progress_media_time>
     <answered_time>1298416416500657</answered_time>
     <hangup_time>1298416416562577</hangup_time>
     <resurrect_time>0</resurrect_time>
     <transfer_time>0</transfer_time>
   </times>
 </callflow>
</cdr>


print_r

Array
(
   [] => Array
       (
           [channel_data] =>

           [state] => CS_REPORTING
           [direction] => inbound
           [state_number] => 11
           [flags] => 0=1;1=1;3=1;35=1;36=1;38=1;46=1;51=1
           [caps] => 1=1;2=1;3=1
       )

   [variables] => Array
       (
           [uuid] => 5d75e976-3ed9-11e0-a60b-fba7243e1308
           [sip_local_network_addr] => 1.2.3.4
           [sip_network_ip] => 1.2.3.4
           [sip_network_port] => 9774
           [sip_received_ip] => 1.2.3.4
           [sip_received_port] => 9774
           [sip_via_protocol] => udp
           [sip_from_user] => 8889990
           [sip_from_port] => 5060
           [sip_from_uri] => 8889990%1.2.3.4%3A5060
           [sip_from_host] => 1.2.3.4
           [sip_from_user_stripped] => 8889990
           [sip_from_tag] => 2e41ba6e
           [sofia_profile_name] => external
           [sip_full_via] =>
SIP/2.0/UDP%1.2.3.4%3A9774%3Bbranch%3Dz9hG4bK-d8754z-c551a41eb93bd648-1---d8754z-%3Brport%3D9774%3Breceived%3D1.2.3.4
           [sip_from_display] => david
           [sip_full_from] =>
%22david%22%20%3Csip%3A8889990%1.2.3.4%3A5060%3E%3Btag%3D2e41ba6e
           [sip_to_display] => 0013058883456
           [sip_full_to] =>
%220013058883456%22%20%3Csip%3A0013058883456%1.2.3.4%3A5060%3E
           [sip_req_user] => 0013058883456
           [sip_req_port] => 5060
           [sip_req_uri] => 0013058883456%1.2.3.4%3A5060
           [sip_req_host] => 1.2.3.4
           [sip_to_user] => 0013058883456
           [sip_to_port] => 5060
           [sip_to_uri] => 0013058883456%1.2.3.4%3A5060
           [sip_to_host] => 1.2.3.4
           [sip_contact_user] => 8889990
           [sip_contact_port] => 9774
           [sip_contact_uri] => 8889990%1.2.3.4%3A9774
           [sip_contact_host] => 1.2.3.4
           [channel_name] => sofia/external/8889990%1.2.3.4%3A5060
           [sip_call_id] => OGVkMmMyMmQ5MDEyOTU2MWUyN2M2NzBhYjk5ZmQzOTM.
           [sip_user_agent] => eyeBeam%20release%201102q%20stamp%2051814
           [sip_via_host] => 1.2.3.4
           [sip_via_port] => 9774
           [sip_via_rport] => 9774
           [max_forwards] => 70
           [switch_r_sdp] =>
v%3D0%0D%0Ao%3D-%207%202%20IN%20IP4%1.2.3.4%0D%0As%3DCounterPath%20eyeBeam%201.5%0D%0Ac%3DIN%20IP4%1.2.3.4%0D%0At%3D0%200%0D%0Am%3Daudio%2023746%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0Aa%3Dalt%3A1%203%20%3A%20L2qFFNvS%20HwniVnkK%1.2.3.4%2023746%0D%0Aa%3Dalt%3A2%202%20%3A%20HWrMm7HE%20Wk82PRBC%1.2.3.4%2023746%0D%0Aa%3Dalt%3A3%201%20%3A%20virMBT30%20GCcqsUuh%1.2.3.4%2023746%0D%0A
           [sip_use_codec_name] => PCMU
           [sip_use_codec_rate] => 8000
           [sip_use_codec_ptime] => 20
           [read_codec] => PCMU
           [read_rate] => 8000
           [write_codec] => PCMU
           [write_rate] => 8000
           [outside_call] => true
           [hangup_after_bridge] => true
           [continue_on_fail] =>
NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407
           [current_application_data] => sofia/gateway/OPENSIP/0013058883456
           [current_application] => bridge
           [local_media_ip] => 1.2.3.4
           [local_media_port] => 20342
           [sip_use_pt] => 0
           [rtp_use_ssrc] => 820599639
           [remote_media_ip] => 1.2.3.4
           [remote_media_port] => 23746
           [originate_disposition] => SUCCESS
           [bridge_channel] => sofia/external/0013058883456
           [bridge_uuid] => 5d7e9d6e-3ed9-11e0-a60c-fba7243e1308
           [signal_bond] => 5d7e9d6e-3ed9-11e0-a60c-fba7243e1308
           [switch_m_sdp] =>
v%3D0%0D%0Ao%3Droot%2023193%2023194%20IN%20IP4%1.2.3.4%0D%0As%3Dsession%0D%0Ac%3DIN%20IP4%1.2.3.4%0D%0At%3D0%200%0D%0Am%3Daudio%2021586%20RTP/AVP%2018%203%200%208%20101%0D%0Aa%3Drtpmap%3A18%20G729/8000%0D%0Aa%3Dfmtp%3A18%20annexb%3Dno%0D%0Aa%3Drtpmap%3A3%20GSM/8000%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-16%0D%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0D%0Aa%3Dptime%3A20%0D%0A
           [sip_local_sdp_str] =>
v%3D0%0Ao%3DFreeSWITCH%201298396074%201298396076%20IN%20IP4%1.2.3.4%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%1.2.3.4%0At%3D0%200%0Am%3Daudio%2020342%20RTP/AVP%200%20101%0Aa%3Drtpmap%3A0%20PCMU/8000%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A
           [endpoint_disposition] => ANSWER
           [bridge_hangup_cause] => NORMAL_CLEARING
           [hangup_cause] => NORMAL_CLEARING
           [hangup_cause_q850] => 16
           [digits_dialed] => none
           [start_stamp] => 2011-02-23%2000%3A13%3A31
           [profile_start_stamp] => 2011-02-23%2000%3A13%3A31
           [answer_stamp] => 2011-02-23%2000%3A13%3A36
           [progress_media_stamp] => 2011-02-23%2000%3A13%3A36
           [end_stamp] => 2011-02-23%2000%3A13%3A36
           [start_epoch] => 1298416411
           [start_uepoch] => 1298416411795887
           [profile_start_epoch] => 1298416411
           [profile_start_uepoch] => 1298416411795887
           [answer_epoch] => 1298416416
           [answer_uepoch] => 1298416416500657
           [resurrect_epoch] => 0
           [resurrect_uepoch] => 0
           [progress_epoch] => 0
           [progress_uepoch] => 0
           [progress_media_epoch] => 1298416416
           [progress_media_uepoch] => 1298416416470740
           [end_epoch] => 1298416416
           [end_uepoch] => 1298416416562577
           [last_app] => bridge
           [last_arg] => sofia/gateway/OPENSIP/0013058883456
           [caller_id] => %22david%22%20%3C8889990%3E
           [duration] => 5
           [billsec] => 0
           [progresssec] => 0
           [answersec] => 5
           [progress_mediasec] => 5
           [flow_billsec] => 5
           [mduration] => 4767
           [billmsec] => 62
           [progressmsec] => 0
           [answermsec] => 4705
           [progress_mediamsec] => 0
           [flow_billmsec] => 4767
           [uduration] => 4766690
           [billusec] => 61920
           [progressusec] => 0
           [answerusec] => 4704770
           [progress_mediausec] => 4674853
           [flow_billusec] => 4766690
           [sip_hangup_disposition] => send_bye
           [rtp_audio_in_raw_bytes] => 0
           [rtp_audio_in_media_bytes] => 0
           [rtp_audio_in_packet_count] => 0
           [rtp_audio_in_media_packet_count] => 0
           [rtp_audio_in_skip_packet_count] => 5
           [rtp_audio_in_jb_packet_count] => 0
           [rtp_audio_in_dtmf_packet_count] => 0
           [rtp_audio_in_cng_packet_count] => 0
           [rtp_audio_in_flush_packet_count] => 0
           [rtp_audio_out_raw_bytes] => 0
           [rtp_audio_out_media_bytes] => 0
           [rtp_audio_out_packet_count] => 0
           [rtp_audio_out_media_packet_count] => 0
           [rtp_audio_out_skip_packet_count] => 0
           [rtp_audio_out_dtmf_packet_count] => 0
           [rtp_audio_out_cng_packet_count] => 0
       )

   [app_log] => Array
       (
           [0] => Array
               (
                   [app_name] => set
                   [app_data] => outside_call=true
               )

           [1] => Array
               (
                   [app_name] => info
                   [app_data] =>
               )

           [2] => Array
               (
                   [app_name] => set
                   [app_data] => hangup_after_bridge=true
               )

           [3] => Array
               (
                   [app_name] => set
                   [app_data] =>
continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407
               )

           [4] => Array
               (
                   [app_name] => bridge
                   [app_data] => sofia/gateway/OPENSIP/0013058883456
               )

       )

   [callflow] => Array
       (
           [0] => Array
               (
                   [dialplan] => XML
                   [profile_index] => 1
                   [extension] => Array
                       (
                           [name] => outside_call
                           [number] => 0013058883456
                           [current_app] => bridge
                           [application] => Array
                               (
                                   [0] => Array
                                       (
                                           [app_name] => set
                                           [app_data] => outside_call=true
                                       )

                                   [1] => Array
                                       (
                                           [app_name] => info
                                           [app_data] =>
                                       )

                                   [2] => Array
                                       (
                                           [app_name] => set
                                           [app_data] =>
hangup_after_bridge=true
                                       )

                                   [3] => Array
                                       (
                                           [app_name] => set
                                           [app_data] =>
continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407
                                       )

                                   [4] => Array
                                       (
                                           [app_name] => bridge
                                           [app_data] =>
sofia/gateway/${distributor(test)}13058883456
                                       )

                                   [5] => Array
                                       (
                                           [last_executed] => true
                                           [app_name] => bridge
                                           [app_data] =>
sofia/gateway/${distributor(test)}13058883456
                                       )

                               )

                       )

                   [caller_profile] => Array
                       (
                           [username] => 8889990
                           [dialplan] => XML
                           [caller_id_name] => david
                           [ani] => 8889990
                           [caller_id_number] => 8889990
                           [network_addr] => 1.2.3.4
                           [destination_number] => 0013058883456
                           [uuid] => 5d7e9d6e-3ed9-11e0-a60c-fba7243e1308
                           [source] => mod_sofia
                           [context] => public
                           [chan_name] => sofia/external/0013058883456
                           [originatee] =>

                           [originatee_caller_profile] =>

                       )

                   [times] => Array
                       (
                           [created_time] => 1298416411795887
                           [profile_created_time] => 1298416411795887
                           [progress_time] => 0
                           [progress_media_time] => 1298416416470740
                           [answered_time] => 1298416416500657
                           [hangup_time] => 1298416416562577
                           [resurrect_time] => 0
                           [transfer_time] => 0
                       )

               )

       )

)
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From anthony.minessale at gmail.com  Wed Feb 23 17:08:02 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 23 Feb 2011 08:08:02 -0600
Subject: [Freeswitch-users] SIP DoS Prevention - Patch
In-Reply-To: <AANLkTik7V5cRe=SNQW9Xq7JdugWeONn5mubM=zPdgNuQ@mail.gmail.com>
References: <AANLkTik7V5cRe=SNQW9Xq7JdugWeONn5mubM=zPdgNuQ@mail.gmail.com>
Message-ID: <AANLkTimrGfUEYOZqWBFoq-qjA=HYmG14pc7QgrKk7g9v@mail.gmail.com>

Keep in mind there is also a register attempt event but this is probably an
ok addition.
On Feb 23, 2011 12:59 AM, "jay binks" <jaybinks at gmail.com> wrote:
> check out my patch Ive just posted to jira.
> http://jira.freeswitch.org/browse/FS-3094
>
> you can subscribe to "Auth" events now - SWITCH_EVENT_AUTH
> which are generic for Register & Invite ( and any other auth'd packet I
> guess )
> and there is an auth_state which will tell you if its an auth fail or a
> expire etc..
> you can get the to_user, to_host, useragent, sip profile and network_ip
>
> Auth Expires also have a Log messages similar to Auth Failures ( fail2ban
> compatible )
>
> PLEASE guys give this some testing. I only banged this together in a few
> hours today in my currently sleep deprived state. So im sure its going to
> have things that can be improved or work better, but at least the bones of
> it are there now.
>
> let me know your thoughts.
>
> Jay
>
>
>
> On Mon, Feb 21, 2011 at 11:08 PM, Jmesquita at freeswitch.org <
> jmesquita at freeswitch.org> wrote:
>
>> Just an idea for the other developers. Wouldn't be cool to have and event
>> thrown by each module that does authentication so that other application
>> modules are able to listen in those events and do whatever with it? Some
>> people might even like to have that running an eel daemon and throw some
>> snmp traps based on that?
>>
>> Just an idea that might be completely ridiculous or might raise some
>> interest. Events are pretty cheap so there wouldn't be a lot of overhead
>> involved.
>>
>> Regards,
>>
>> Jo?o Mesquita
>>
>> On 21/02/2011, at 07:28, Spencer Thomason <spencer at 5ninesolutions.com>
>> wrote:
>>
>> > After tinkering with it, I think that might be the best way. The
>> > iptables method is cool but I'd like to have more dynamic control and
>> > with Fail2Ban looking at the challenges you could specifically ignore
>> > certain high traffic IPs and block others. What would be very cool is
>> > if instead of logging every challenge, a log entry was written if
>> > there was a high number from a specific IP, then you could decide what
>> > to do about it with fail2ban, similar to the pike module for opensips
>> > does.
>> >
>> >
>> > On Feb 21, 2011, at 1:31 AM, covici at ccs.covici.com wrote:
>> >
>> >> I would change sip auth failure to challenge and then have sufficient
>> >> times to only block if there are too many challenges in a certain
>> >> time.
>> >> I am not even sure the failure works any more in recent gits.
>> >>
>> >> Spencer Thomason <spencer at 5ninesolutions.com> wrote:
>> >>
>> >>> Yes, that works great if they respond to the challenge with a failed
>> >>> auth. But the scenario I'm trying to prevent is if they just send the
>> >>> INVITE and never respond to the challenge. Fail2Ban will not work as
>> >>> every endpoint will initially send an INVITE and receive a challenge.
>> >>> Legit calls will then respond correctly and not be logged as a SIP
>> >>> auth failure but every call that is challenged will show up as SIP
>> >>> auth challenge in the logs so there is no regex to differentiate
>> >>> between legit an non legit traffic.
>> >>>
>> >>> Spencer
>> >>>
>> >>> On Feb 20, 2011, at 10:39 PM, Ken Rice wrote:
>> >>>
>> >>>> Fail2Ban ... This is block an IP with too many failed attempts from
>> >>>> something like SipVicious pretty quickly
>> >>>>
>> >>>>
>> >>>> On 2/20/11 11:07 PM, "Spencer Thomason" <spencer at 5ninesolutions.com>
>> >>>> wrote:
>> >>>>
>> >>>>> Hi,
>> >>>>> We run hosted Freeswitch instances in VMs with the internal profile
>> >>>>> on
>> >>>>> port 5060 connecting to clients mostly behind NAT and then the
>> >>>>> external profile connecting to our proxies only. Protecting the
>> >>>>> external profile its straightforward.. we only allow traffic to/
>> >>>>> from
>> >>>>> our proxies at the firewall level. But protecting the internal
>> >>>>> profile seems to be a bit more difficult because the UACs could be
>> >>>>> theoretically anywhere on the network.
>> >>>>>
>> >>>>> I'm currently using Fail2Ban to prevent brute force registration
>> >>>>> and
>> >>>>> INVITEs on auth failures, e.g.:
>> >>>>> failregex = \[WARNING\] sofia_reg.c:\d+ SIP auth failure \
>> >>>>> (REGISTER\)
>> >>>>> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
>> >>>>> \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(INVITE\)
>> >>>>> on sofia profile \'\w+\' for \[.*\] from ip <HOST>
>> >>>>>
>> >>>>> My question is, since its part of a normal SIP dialog to challenge
>> >>>>> the
>> >>>>> INVITE, is there any way to prevent a possible DoS from just sheer
>> >>>>> volume of incoming INVITEs on an Internet facing server
>> >>>>> automatically. I.e., If you block the logged challenge, you'd
>> >>>>> block
>> >>>>> all legitimate INVITEs and registrations. Since its UDP traffic I
>> >>>>> couldn't come up with a way to do it automatically at the iptables
>> >>>>> level. i.e. number of concurrent connections. Is there some option
>> >>>>> to
>> >>>>> just not respond if a client is sending a number of requests over a
>> >>>>> certain threshold? It might not stop them from sending the traffic
>> >>>>> but pretty soon they'd get the idea that it wasn't going to go
>> >>>>> anywhere. My concern is say there are 50 Freeswitch instances on a
>> >>>>> box (albeit 8 core, 32GB ram, 8 15K raid 10 storage) and someone
>> >>>>> starts sending thousands of rouge INVITEs to every VM on a physical
>> >>>>> box that the CPU load from just challenging the incoming INVITEs
>> >>>>> would
>> >>>>> create a DoS. We the logs regularly to try to catch people doing
>> >>>>> this
>> >>>>> sort of thing and drop them at a router upstream of the core
>> >>>>> network,
>> >>>>> but I'd like to have it happen without human intervention. Have I
>> >>>>> completely over thought this and am missing something obvious?
>> >>>>>
>> >>>>> Thanks,
>> >>>>> 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
>> >>>>
>> >>>>
>> >>>>
>> >>>> _______________________________________________
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>>
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at 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
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 sos at sokhapkin.dyndns.org  Wed Feb 23 17:11:47 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 23 Feb 2011 09:11:47 -0500
Subject: [Freeswitch-users] Weird CDR ERROR...
In-Reply-To: <AANLkTikVW7K=tvsrFa7azTEtU8CKso+fTiSj6sw6yhAf@mail.gmail.com>
References: <AANLkTikVW7K=tvsrFa7azTEtU8CKso+fTiSj6sw6yhAf@mail.gmail.com>
Message-ID: <201102230911.47818.sos@sokhapkin.dyndns.org>

<billmsec>62</billmsec>

The call was answered for less than 1 second.

On Wednesday 23 February 2011, David Villasmil wrote:
> Hello All,
> 
> I have an XML_CDR cdr that is not correct, i made some test and on one cdr
> (that i could find) which show endpoint_disposition as ANSWER when the call
> was NOT answered! The call didn't even go out of the box...
> 
> here's the CDR:
> 
> note that billsec is 0, but disposition is ANSWER... weird, i got more than
> one, also...
> 
> 
> 
> <cdr>
>  <channel_data>
>    <state>CS_REPORTING</state>
>    <direction>inbound</direction>
>    <state_number>11</state_number>
>    <flags>0=1;1=1;3=1;35=1;36=1;38=1;46=1;51=1</flags>
>    <caps>1=1;2=1;3=1</caps>
>  </channel_data>
>  <variables>
>    <uuid>5d75e976-3ed9-11e0-a60b-fba7243e1308</uuid>
>    <sip_local_network_addr>1.2.3.4</sip_local_network_addr>
>    <sip_network_ip>1.2.3.4</sip_network_ip>
>    <sip_network_port>9774</sip_network_port>
>    <sip_received_ip>1.2.3.4</sip_received_ip>
>    <sip_received_port>9774</sip_received_port>
>    <sip_via_protocol>udp</sip_via_protocol>
>    <sip_from_user>8889990</sip_from_user>
>    <sip_from_port>5060</sip_from_port>
>    <sip_from_uri>8889990%1.2.3.4%3A5060</sip_from_uri>
>    <sip_from_host>1.2.3.4</sip_from_host>
>    <sip_from_user_stripped>8889990</sip_from_user_stripped>
>    <sip_from_tag>2e41ba6e</sip_from_tag>
>    <sofia_profile_name>external</sofia_profile_name>
> 
>  <sip_full_via>SIP/2.0/UDP%1.2.3.4%3A9774%3Bbranch%3Dz9hG4bK-d8754z-c551a41
> eb93bd648-1---d8754z-%3Brport%3D9774%3Breceived%3D1.2.3.4</sip_full_via>
> <sip_from_display>david</sip_from_display>
> 
>  <sip_full_from>%22david%22%20%3Csip%3A8889990%1.2.3.4%3A5060%3E%3Btag%3D2e
> 41ba6e</sip_full_from> <sip_to_display>0013058883456</sip_to_display>
> 
>  <sip_full_to>%220013058883456%22%20%3Csip%3A0013058883456%1.2.3.4%3A5060%3
> E</sip_full_to> <sip_req_user>0013058883456</sip_req_user>
>    <sip_req_port>5060</sip_req_port>
>    <sip_req_uri>0013058883456%1.2.3.4%3A5060</sip_req_uri>
>    <sip_req_host>1.2.3.4</sip_req_host>
>    <sip_to_user>0013058883456</sip_to_user>
>    <sip_to_port>5060</sip_to_port>
>    <sip_to_uri>0013058883456%1.2.3.4%3A5060</sip_to_uri>
>    <sip_to_host>1.2.3.4</sip_to_host>
>    <sip_contact_user>8889990</sip_contact_user>
>    <sip_contact_port>9774</sip_contact_port>
>    <sip_contact_uri>8889990%1.2.3.4%3A9774</sip_contact_uri>
>    <sip_contact_host>1.2.3.4</sip_contact_host>
>    <channel_name>sofia/external/8889990%1.2.3.4%3A5060</channel_name>
>    <sip_call_id>OGVkMmMyMmQ5MDEyOTU2MWUyN2M2NzBhYjk5ZmQzOTM.</sip_call_id>
> 
>  <sip_user_agent>eyeBeam%20release%201102q%20stamp%2051814</sip_user_agent>
>    <sip_via_host>1.2.3.4</sip_via_host>
>    <sip_via_port>9774</sip_via_port>
>    <sip_via_rport>9774</sip_via_rport>
>    <max_forwards>70</max_forwards>
> 
>  <switch_r_sdp>v%3D0%0D%0Ao%3D-%207%202%20IN%20IP4%1.2.3.4%0D%0As%3DCounter
> Path%20eyeBeam%201.5%0D%0Ac%3DIN%20IP4%1.2.3.4%0D%0At%3D0%200%0D%0Am%3Daudi
> o%2023746%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-eve
> nt/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0Aa%3Dalt%3A1%203%20%3A%20L2qFFNvS%20
> HwniVnkK%1.2.3.4%2023746%0D%0Aa%3Dalt%3A2%202%20%3A%20HWrMm7HE%20Wk82PRBC%1
> .2.3.4%2023746%0D%0Aa%3Dalt%3A3%201%20%3A%20virMBT30%20GCcqsUuh%1.2.3.4%202
> 3746%0D%0A</switch_r_sdp> <sip_use_codec_name>PCMU</sip_use_codec_name>
>    <sip_use_codec_rate>8000</sip_use_codec_rate>
>    <sip_use_codec_ptime>20</sip_use_codec_ptime>
>    <read_codec>PCMU</read_codec>
>    <read_rate>8000</read_rate>
>    <write_codec>PCMU</write_codec>
>    <write_rate>8000</write_rate>
>    <outside_call>true</outside_call>
>    <hangup_after_bridge>true</hangup_after_bridge>
> 
>  <continue_on_fail>NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UN
> ALLOCATED_NUMBER,407</continue_on_fail>
> 
>  <current_application_data>sofia/gateway/OPENSIP/0013058883456</current_app
> lication_data> <current_application>bridge</current_application>
>    <local_media_ip>1.2.3.4</local_media_ip>
>    <local_media_port>20342</local_media_port>
>    <sip_use_pt>0</sip_use_pt>
>    <rtp_use_ssrc>820599639</rtp_use_ssrc>
>    <remote_media_ip>1.2.3.4</remote_media_ip>
>    <remote_media_port>23746</remote_media_port>
>    <originate_disposition>SUCCESS</originate_disposition>
>    <bridge_channel>sofia/external/0013058883456</bridge_channel>
>    <bridge_uuid>5d7e9d6e-3ed9-11e0-a60c-fba7243e1308</bridge_uuid>
>    <signal_bond>5d7e9d6e-3ed9-11e0-a60c-fba7243e1308</signal_bond>
> 
>  <switch_m_sdp>v%3D0%0D%0Ao%3Droot%2023193%2023194%20IN%20IP4%1.2.3.4%0D%0A
> s%3Dsession%0D%0Ac%3DIN%20IP4%1.2.3.4%0D%0At%3D0%200%0D%0Am%3Daudio%2021586
> %20RTP/AVP%2018%203%200%208%20101%0D%0Aa%3Drtpmap%3A18%20G729/8000%0D%0Aa%3
> Dfmtp%3A18%20annexb%3Dno%0D%0Aa%3Drtpmap%3A3%20GSM/8000%0D%0Aa%3Drtpmap%3A0
> %20PCMU/8000%0D%0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Drtpmap%3A101%20teleph
> one-event/8000%0D%0Aa%3Dfmtp%3A101%200-16%0D%0Aa%3DsilenceSupp%3Aoff%20-%20
> -%20-%20-%0D%0Aa%3Dptime%3A20%0D%0A</switch_m_sdp>
> 
>  <sip_local_sdp_str>v%3D0%0Ao%3DFreeSWITCH%201298396074%201298396076%20IN%2
> 0IP4%1.2.3.4%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%1.2.3.4%0At%3D0%200%0Am%3Daudi
> o%2020342%20RTP/AVP%200%20101%0Aa%3Drtpmap%3A0%20PCMU/8000%0Aa%3Drtpmap%3A1
> 01%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3DsilenceSupp%3Aoff%2
> 0-%20-%20-%20-%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A</sip_local_sdp_str>
> <endpoint_disposition>ANSWER</endpoint_disposition>
>    <bridge_hangup_cause>NORMAL_CLEARING</bridge_hangup_cause>
>    <hangup_cause>NORMAL_CLEARING</hangup_cause>
>    <hangup_cause_q850>16</hangup_cause_q850>
>    <digits_dialed>none</digits_dialed>
>    <start_stamp>2011-02-23%2000%3A13%3A31</start_stamp>
>    <profile_start_stamp>2011-02-23%2000%3A13%3A31</profile_start_stamp>
>    <answer_stamp>2011-02-23%2000%3A13%3A36</answer_stamp>
>    <progress_media_stamp>2011-02-23%2000%3A13%3A36</progress_media_stamp>
>    <end_stamp>2011-02-23%2000%3A13%3A36</end_stamp>
>    <start_epoch>1298416411</start_epoch>
>    <start_uepoch>1298416411795887</start_uepoch>
>    <profile_start_epoch>1298416411</profile_start_epoch>
>    <profile_start_uepoch>1298416411795887</profile_start_uepoch>
>    <answer_epoch>1298416416</answer_epoch>
>    <answer_uepoch>1298416416500657</answer_uepoch>
>    <resurrect_epoch>0</resurrect_epoch>
>    <resurrect_uepoch>0</resurrect_uepoch>
>    <progress_epoch>0</progress_epoch>
>    <progress_uepoch>0</progress_uepoch>
>    <progress_media_epoch>1298416416</progress_media_epoch>
>    <progress_media_uepoch>1298416416470740</progress_media_uepoch>
>    <end_epoch>1298416416</end_epoch>
>    <end_uepoch>1298416416562577</end_uepoch>
>    <last_app>bridge</last_app>
>    <last_arg>sofia/gateway/OPENSIP/0013058883456</last_arg>
>    <caller_id>%22david%22%20%3C8889990%3E</caller_id>
>    <duration>5</duration>
>    <billsec>0</billsec>
>    <progresssec>0</progresssec>
>    <answersec>5</answersec>
>    <progress_mediasec>5</progress_mediasec>
>    <flow_billsec>5</flow_billsec>
>    <mduration>4767</mduration>
>    <billmsec>62</billmsec>
>    <progressmsec>0</progressmsec>
>    <answermsec>4705</answermsec>
>    <progress_mediamsec>0</progress_mediamsec>
>    <flow_billmsec>4767</flow_billmsec>
>    <uduration>4766690</uduration>
>    <billusec>61920</billusec>
>    <progressusec>0</progressusec>
>    <answerusec>4704770</answerusec>
>    <progress_mediausec>4674853</progress_mediausec>
>    <flow_billusec>4766690</flow_billusec>
>    <sip_hangup_disposition>send_bye</sip_hangup_disposition>
>    <rtp_audio_in_raw_bytes>0</rtp_audio_in_raw_bytes>
>    <rtp_audio_in_media_bytes>0</rtp_audio_in_media_bytes>
>    <rtp_audio_in_packet_count>0</rtp_audio_in_packet_count>
>    <rtp_audio_in_media_packet_count>0</rtp_audio_in_media_packet_count>
>    <rtp_audio_in_skip_packet_count>5</rtp_audio_in_skip_packet_count>
>    <rtp_audio_in_jb_packet_count>0</rtp_audio_in_jb_packet_count>
>    <rtp_audio_in_dtmf_packet_count>0</rtp_audio_in_dtmf_packet_count>
>    <rtp_audio_in_cng_packet_count>0</rtp_audio_in_cng_packet_count>
>    <rtp_audio_in_flush_packet_count>0</rtp_audio_in_flush_packet_count>
>    <rtp_audio_out_raw_bytes>0</rtp_audio_out_raw_bytes>
>    <rtp_audio_out_media_bytes>0</rtp_audio_out_media_bytes>
>    <rtp_audio_out_packet_count>0</rtp_audio_out_packet_count>
>    <rtp_audio_out_media_packet_count>0</rtp_audio_out_media_packet_count>
>    <rtp_audio_out_skip_packet_count>0</rtp_audio_out_skip_packet_count>
>    <rtp_audio_out_dtmf_packet_count>0</rtp_audio_out_dtmf_packet_count>
>    <rtp_audio_out_cng_packet_count>0</rtp_audio_out_cng_packet_count>
>  </variables>
>  <app_log>
>    <application app_name="set" app_data="outside_call=true"></application>
>    <application app_name="info" app_data=""></application>
>    <application app_name="set"
> app_data="hangup_after_bridge=true"></application>
>    <application app_name="set"
> app_data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTIN
> ATION,UNALLOCATED_NUMBER,407"></application> <application app_name="bridge"
> app_data="sofia/gateway/OPENSIP/0013058883456"></application>
>  </app_log>
>  <callflow dialplan="XML" profile_index="1">
>    <extension name="outside_call" number="0013058883456"
> current_app="bridge">
>      <application app_name="set"
> app_data="outside_call=true"></application> <application app_name="info"
> app_data=""></application>
>      <application app_name="set"
> app_data="hangup_after_bridge=true"></application>
>      <application app_name="set"
> app_data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTIN
> ATION,UNALLOCATED_NUMBER,407"></application> <application app_name="bridge"
> app_data="sofia/gateway/${distributor(test)}13058883456"></application>
>      <application last_executed="true" app_name="bridge"
> app_data="sofia/gateway/${distributor(test)}13058883456"></application>
>    </extension>
>    <caller_profile>
>      <username>8889990</username>
>      <dialplan>XML</dialplan>
>      <caller_id_name>david</caller_id_name>
>      <ani>8889990</ani>
>      <aniii></aniii>
>      <caller_id_number>8889990</caller_id_number>
>      <network_addr>1.2.3.4</network_addr>
>      <rdnis></rdnis>
>      <destination_number>0013058883456</destination_number>
>      <uuid>5d75e976-3ed9-11e0-a60b-fba7243e1308</uuid>
>      <source>mod_sofia</source>
>      <context>public</context>
>      <chan_name>sofia/external/8889990 at 1.2.3.4:5060</chan_name>
>      <originatee>
>        <originatee_caller_profile>
>          <username>8889990</username>
>          <dialplan>XML</dialplan>
>          <caller_id_name>david</caller_id_name>
>          <ani>8889990</ani>
>          <aniii></aniii>
>          <caller_id_number>8889990</caller_id_number>
>          <network_addr>1.2.3.4</network_addr>
>          <rdnis></rdnis>
>          <destination_number>0013058883456</destination_number>
>          <uuid>5d7e9d6e-3ed9-11e0-a60c-fba7243e1308</uuid>
>          <source>mod_sofia</source>
>          <context>public</context>
>          <chan_name>sofia/external/0013058883456</chan_name>
>        </originatee_caller_profile>
>      </originatee>
>    </caller_profile>
>    <times>
>      <created_time>1298416411795887</created_time>
>      <profile_created_time>1298416411795887</profile_created_time>
>      <progress_time>0</progress_time>
>      <progress_media_time>1298416416470740</progress_media_time>
>      <answered_time>1298416416500657</answered_time>
>      <hangup_time>1298416416562577</hangup_time>
>      <resurrect_time>0</resurrect_time>
>      <transfer_time>0</transfer_time>
>    </times>
>  </callflow>
> </cdr>
> 
> 
> print_r
> 
> Array
> (
>    [] => Array
>        (
>            [channel_data] =>
> 
>            [state] => CS_REPORTING
>            [direction] => inbound
>            [state_number] => 11
>            [flags] => 0=1;1=1;3=1;35=1;36=1;38=1;46=1;51=1
>            [caps] => 1=1;2=1;3=1
>        )
> 
>    [variables] => Array
>        (
>            [uuid] => 5d75e976-3ed9-11e0-a60b-fba7243e1308
>            [sip_local_network_addr] => 1.2.3.4
>            [sip_network_ip] => 1.2.3.4
>            [sip_network_port] => 9774
>            [sip_received_ip] => 1.2.3.4
>            [sip_received_port] => 9774
>            [sip_via_protocol] => udp
>            [sip_from_user] => 8889990
>            [sip_from_port] => 5060
>            [sip_from_uri] => 8889990%1.2.3.4%3A5060
>            [sip_from_host] => 1.2.3.4
>            [sip_from_user_stripped] => 8889990
>            [sip_from_tag] => 2e41ba6e
>            [sofia_profile_name] => external
>            [sip_full_via] =>
> SIP/2.0/UDP%1.2.3.4%3A9774%3Bbranch%3Dz9hG4bK-d8754z-c551a41eb93bd648-1---d
> 8754z-%3Brport%3D9774%3Breceived%3D1.2.3.4 [sip_from_display] => david
>            [sip_full_from] =>
> %22david%22%20%3Csip%3A8889990%1.2.3.4%3A5060%3E%3Btag%3D2e41ba6e
>            [sip_to_display] => 0013058883456
>            [sip_full_to] =>
> %220013058883456%22%20%3Csip%3A0013058883456%1.2.3.4%3A5060%3E
>            [sip_req_user] => 0013058883456
>            [sip_req_port] => 5060
>            [sip_req_uri] => 0013058883456%1.2.3.4%3A5060
>            [sip_req_host] => 1.2.3.4
>            [sip_to_user] => 0013058883456
>            [sip_to_port] => 5060
>            [sip_to_uri] => 0013058883456%1.2.3.4%3A5060
>            [sip_to_host] => 1.2.3.4
>            [sip_contact_user] => 8889990
>            [sip_contact_port] => 9774
>            [sip_contact_uri] => 8889990%1.2.3.4%3A9774
>            [sip_contact_host] => 1.2.3.4
>            [channel_name] => sofia/external/8889990%1.2.3.4%3A5060
>            [sip_call_id] => OGVkMmMyMmQ5MDEyOTU2MWUyN2M2NzBhYjk5ZmQzOTM.
>            [sip_user_agent] => eyeBeam%20release%201102q%20stamp%2051814
>            [sip_via_host] => 1.2.3.4
>            [sip_via_port] => 9774
>            [sip_via_rport] => 9774
>            [max_forwards] => 70
>            [switch_r_sdp] =>
> v%3D0%0D%0Ao%3D-%207%202%20IN%20IP4%1.2.3.4%0D%0As%3DCounterPath%20eyeBeam%
> 201.5%0D%0Ac%3DIN%20IP4%1.2.3.4%0D%0At%3D0%200%0D%0Am%3Daudio%2023746%20RTP
> /AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%
> 3Dfmtp%3A101%200-15%0D%0Aa%3Dalt%3A1%203%20%3A%20L2qFFNvS%20HwniVnkK%1.2.3.
> 4%2023746%0D%0Aa%3Dalt%3A2%202%20%3A%20HWrMm7HE%20Wk82PRBC%1.2.3.4%2023746%
> 0D%0Aa%3Dalt%3A3%201%20%3A%20virMBT30%20GCcqsUuh%1.2.3.4%2023746%0D%0A
> [sip_use_codec_name] => PCMU
>            [sip_use_codec_rate] => 8000
>            [sip_use_codec_ptime] => 20
>            [read_codec] => PCMU
>            [read_rate] => 8000
>            [write_codec] => PCMU
>            [write_rate] => 8000
>            [outside_call] => true
>            [hangup_after_bridge] => true
>            [continue_on_fail] =>
> NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,40
> 7 [current_application_data] => sofia/gateway/OPENSIP/0013058883456
> [current_application] => bridge
>            [local_media_ip] => 1.2.3.4
>            [local_media_port] => 20342
>            [sip_use_pt] => 0
>            [rtp_use_ssrc] => 820599639
>            [remote_media_ip] => 1.2.3.4
>            [remote_media_port] => 23746
>            [originate_disposition] => SUCCESS
>            [bridge_channel] => sofia/external/0013058883456
>            [bridge_uuid] => 5d7e9d6e-3ed9-11e0-a60c-fba7243e1308
>            [signal_bond] => 5d7e9d6e-3ed9-11e0-a60c-fba7243e1308
>            [switch_m_sdp] =>
> v%3D0%0D%0Ao%3Droot%2023193%2023194%20IN%20IP4%1.2.3.4%0D%0As%3Dsession%0D%
> 0Ac%3DIN%20IP4%1.2.3.4%0D%0At%3D0%200%0D%0Am%3Daudio%2021586%20RTP/AVP%2018
> %203%200%208%20101%0D%0Aa%3Drtpmap%3A18%20G729/8000%0D%0Aa%3Dfmtp%3A18%20an
> nexb%3Dno%0D%0Aa%3Drtpmap%3A3%20GSM/8000%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D
> %0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%
> 0D%0Aa%3Dfmtp%3A101%200-16%0D%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0D%0A
> a%3Dptime%3A20%0D%0A [sip_local_sdp_str] =>
> v%3D0%0Ao%3DFreeSWITCH%201298396074%201298396076%20IN%20IP4%1.2.3.4%0As%3DF
> reeSWITCH%0Ac%3DIN%20IP4%1.2.3.4%0At%3D0%200%0Am%3Daudio%2020342%20RTP/AVP%
> 200%20101%0Aa%3Drtpmap%3A0%20PCMU/8000%0Aa%3Drtpmap%3A101%20telephone-event
> /8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0Aa%3
> Dptime%3A20%0Aa%3Dsendrecv%0A [endpoint_disposition] => ANSWER
>            [bridge_hangup_cause] => NORMAL_CLEARING
>            [hangup_cause] => NORMAL_CLEARING
>            [hangup_cause_q850] => 16
>            [digits_dialed] => none
>            [start_stamp] => 2011-02-23%2000%3A13%3A31
>            [profile_start_stamp] => 2011-02-23%2000%3A13%3A31
>            [answer_stamp] => 2011-02-23%2000%3A13%3A36
>            [progress_media_stamp] => 2011-02-23%2000%3A13%3A36
>            [end_stamp] => 2011-02-23%2000%3A13%3A36
>            [start_epoch] => 1298416411
>            [start_uepoch] => 1298416411795887
>            [profile_start_epoch] => 1298416411
>            [profile_start_uepoch] => 1298416411795887
>            [answer_epoch] => 1298416416
>            [answer_uepoch] => 1298416416500657
>            [resurrect_epoch] => 0
>            [resurrect_uepoch] => 0
>            [progress_epoch] => 0
>            [progress_uepoch] => 0
>            [progress_media_epoch] => 1298416416
>            [progress_media_uepoch] => 1298416416470740
>            [end_epoch] => 1298416416
>            [end_uepoch] => 1298416416562577
>            [last_app] => bridge
>            [last_arg] => sofia/gateway/OPENSIP/0013058883456
>            [caller_id] => %22david%22%20%3C8889990%3E
>            [duration] => 5
>            [billsec] => 0
>            [progresssec] => 0
>            [answersec] => 5
>            [progress_mediasec] => 5
>            [flow_billsec] => 5
>            [mduration] => 4767
>            [billmsec] => 62
>            [progressmsec] => 0
>            [answermsec] => 4705
>            [progress_mediamsec] => 0
>            [flow_billmsec] => 4767
>            [uduration] => 4766690
>            [billusec] => 61920
>            [progressusec] => 0
>            [answerusec] => 4704770
>            [progress_mediausec] => 4674853
>            [flow_billusec] => 4766690
>            [sip_hangup_disposition] => send_bye
>            [rtp_audio_in_raw_bytes] => 0
>            [rtp_audio_in_media_bytes] => 0
>            [rtp_audio_in_packet_count] => 0
>            [rtp_audio_in_media_packet_count] => 0
>            [rtp_audio_in_skip_packet_count] => 5
>            [rtp_audio_in_jb_packet_count] => 0
>            [rtp_audio_in_dtmf_packet_count] => 0
>            [rtp_audio_in_cng_packet_count] => 0
>            [rtp_audio_in_flush_packet_count] => 0
>            [rtp_audio_out_raw_bytes] => 0
>            [rtp_audio_out_media_bytes] => 0
>            [rtp_audio_out_packet_count] => 0
>            [rtp_audio_out_media_packet_count] => 0
>            [rtp_audio_out_skip_packet_count] => 0
>            [rtp_audio_out_dtmf_packet_count] => 0
>            [rtp_audio_out_cng_packet_count] => 0
>        )
> 
>    [app_log] => Array
>        (
>            [0] => Array
>                (
>                    [app_name] => set
>                    [app_data] => outside_call=true
>                )
> 
>            [1] => Array
>                (
>                    [app_name] => info
>                    [app_data] =>
>                )
> 
>            [2] => Array
>                (
>                    [app_name] => set
>                    [app_data] => hangup_after_bridge=true
>                )
> 
>            [3] => Array
>                (
>                    [app_name] => set
>                    [app_data] =>
> continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNAL
> LOCATED_NUMBER,407 )
> 
>            [4] => Array
>                (
>                    [app_name] => bridge
>                    [app_data] => sofia/gateway/OPENSIP/0013058883456
>                )
> 
>        )
> 
>    [callflow] => Array
>        (
>            [0] => Array
>                (
>                    [dialplan] => XML
>                    [profile_index] => 1
>                    [extension] => Array
>                        (
>                            [name] => outside_call
>                            [number] => 0013058883456
>                            [current_app] => bridge
>                            [application] => Array
>                                (
>                                    [0] => Array
>                                        (
>                                            [app_name] => set
>                                            [app_data] => outside_call=true
>                                        )
> 
>                                    [1] => Array
>                                        (
>                                            [app_name] => info
>                                            [app_data] =>
>                                        )
> 
>                                    [2] => Array
>                                        (
>                                            [app_name] => set
>                                            [app_data] =>
> hangup_after_bridge=true
>                                        )
> 
>                                    [3] => Array
>                                        (
>                                            [app_name] => set
>                                            [app_data] =>
> continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNAL
> LOCATED_NUMBER,407 )
> 
>                                    [4] => Array
>                                        (
>                                            [app_name] => bridge
>                                            [app_data] =>
> sofia/gateway/${distributor(test)}13058883456
>                                        )
> 
>                                    [5] => Array
>                                        (
>                                            [last_executed] => true
>                                            [app_name] => bridge
>                                            [app_data] =>
> sofia/gateway/${distributor(test)}13058883456
>                                        )
> 
>                                )
> 
>                        )
> 
>                    [caller_profile] => Array
>                        (
>                            [username] => 8889990
>                            [dialplan] => XML
>                            [caller_id_name] => david
>                            [ani] => 8889990
>                            [caller_id_number] => 8889990
>                            [network_addr] => 1.2.3.4
>                            [destination_number] => 0013058883456
>                            [uuid] => 5d7e9d6e-3ed9-11e0-a60c-fba7243e1308
>                            [source] => mod_sofia
>                            [context] => public
>                            [chan_name] => sofia/external/0013058883456
>                            [originatee] =>
> 
>                            [originatee_caller_profile] =>
> 
>                        )
> 
>                    [times] => Array
>                        (
>                            [created_time] => 1298416411795887
>                            [profile_created_time] => 1298416411795887
>                            [progress_time] => 0
>                            [progress_media_time] => 1298416416470740
>                            [answered_time] => 1298416416500657
>                            [hangup_time] => 1298416416562577
>                            [resurrect_time] => 0
>                            [transfer_time] => 0
>                        )
> 
>                )
> 
>        )
> 
> )



From Nabble at slickdeals.endjunk.com  Wed Feb 23 18:10:08 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 23 Feb 2011 07:10:08 -0800 (PST)
Subject: [Freeswitch-users] INVITE DoS Prevention
In-Reply-To: <AANLkTi=+f0GqChj4PeW=XFjQW_mYTtJYPuWEj47gw8r+@mail.gmail.com>
References: <6461E11A-84B5-40AF-96BE-C68F3DF91D8E@5ninesolutions.com>
	<C98764B0.11627%krice@freeswitch.org>
	<A624F826-AF73-40BD-8DCD-10F39551D924@5ninesolutions.com>
	<AANLkTi=+f0GqChj4PeW=XFjQW_mYTtJYPuWEj47gw8r+@mail.gmail.com>
Message-ID: <1298473808528-6056642.post@n2.nabble.com>



jay binks wrote:
> as for rate-limiting responses you can have iptables drop packets over X
> number of invites per sec ...
Just a thought. Perhaps, we should contemplate to add a feature on FS to set
maximum of invites/sec/host. When the invites max out, add some sleep to
slow down the response to the requested host. This will probably slow down
the bot, especially if the bot is trying to hit a lot of FS servers out
there.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/INVITE-DoS-Prevention-tp6047615p6056642.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From david.villasmil.work at gmail.com  Wed Feb 23 18:36:02 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 23 Feb 2011 16:36:02 +0100
Subject: [Freeswitch-users] Weird CDR ERROR...
In-Reply-To: <201102230911.47818.sos@sokhapkin.dyndns.org>
References: <AANLkTikVW7K=tvsrFa7azTEtU8CKso+fTiSj6sw6yhAf@mail.gmail.com>
	<201102230911.47818.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTi=C8xdPAgsLo9_UB-_5KpkmQ8hCr8eHCA+dj9kv@mail.gmail.com>

Hello Sergey and thanks for replying.

The call was NOT answered. It actually failed as there was no context for
that call... that is the weird part... i will try to reproduce it.

Thanks again

David

On Wed, Feb 23, 2011 at 3:11 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> <billmsec>62</billmsec>
>
> The call was answered for less than 1 second.
>
> On Wednesday 23 February 2011, David Villasmil wrote:
> > Hello All,
> >
> > I have an XML_CDR cdr that is not correct, i made some test and on one
> cdr
> > (that i could find) which show endpoint_disposition as ANSWER when the
> call
> > was NOT answered! The call didn't even go out of the box...
> >
> > here's the CDR:
> >
> > note that billsec is 0, but disposition is ANSWER... weird, i got more
> than
> > one, also...
> >
> >
> >
> > <cdr>
> >  <channel_data>
> >    <state>CS_REPORTING</state>
> >    <direction>inbound</direction>
> >    <state_number>11</state_number>
> >    <flags>0=1;1=1;3=1;35=1;36=1;38=1;46=1;51=1</flags>
> >    <caps>1=1;2=1;3=1</caps>
> >  </channel_data>
> >  <variables>
> >    <uuid>5d75e976-3ed9-11e0-a60b-fba7243e1308</uuid>
> >    <sip_local_network_addr>1.2.3.4</sip_local_network_addr>
> >    <sip_network_ip>1.2.3.4</sip_network_ip>
> >    <sip_network_port>9774</sip_network_port>
> >    <sip_received_ip>1.2.3.4</sip_received_ip>
> >    <sip_received_port>9774</sip_received_port>
> >    <sip_via_protocol>udp</sip_via_protocol>
> >    <sip_from_user>8889990</sip_from_user>
> >    <sip_from_port>5060</sip_from_port>
> >    <sip_from_uri>8889990%1.2.3.4%3A5060</sip_from_uri>
> >    <sip_from_host>1.2.3.4</sip_from_host>
> >    <sip_from_user_stripped>8889990</sip_from_user_stripped>
> >    <sip_from_tag>2e41ba6e</sip_from_tag>
> >    <sofia_profile_name>external</sofia_profile_name>
> >
> >
>  <sip_full_via>SIP/2.0/UDP%1.2.3.4%3A9774%3Bbranch%3Dz9hG4bK-d8754z-c551a41
> > eb93bd648-1---d8754z-%3Brport%3D9774%3Breceived%3D1.2.3.4</sip_full_via>
> > <sip_from_display>david</sip_from_display>
> >
> >
>  <sip_full_from>%22david%22%20%3Csip%3A8889990%1.2.3.4%3A5060%3E%3Btag%3D2e
> > 41ba6e</sip_full_from> <sip_to_display>0013058883456</sip_to_display>
> >
> >
>  <sip_full_to>%220013058883456%22%20%3Csip%3A0013058883456%1.2.3.4%3A5060%3
> > E</sip_full_to> <sip_req_user>0013058883456</sip_req_user>
> >    <sip_req_port>5060</sip_req_port>
> >    <sip_req_uri>0013058883456%1.2.3.4%3A5060</sip_req_uri>
> >    <sip_req_host>1.2.3.4</sip_req_host>
> >    <sip_to_user>0013058883456</sip_to_user>
> >    <sip_to_port>5060</sip_to_port>
> >    <sip_to_uri>0013058883456%1.2.3.4%3A5060</sip_to_uri>
> >    <sip_to_host>1.2.3.4</sip_to_host>
> >    <sip_contact_user>8889990</sip_contact_user>
> >    <sip_contact_port>9774</sip_contact_port>
> >    <sip_contact_uri>8889990%1.2.3.4%3A9774</sip_contact_uri>
> >    <sip_contact_host>1.2.3.4</sip_contact_host>
> >    <channel_name>sofia/external/8889990%1.2.3.4%3A5060</channel_name>
> >
>  <sip_call_id>OGVkMmMyMmQ5MDEyOTU2MWUyN2M2NzBhYjk5ZmQzOTM.</sip_call_id>
> >
> >
>  <sip_user_agent>eyeBeam%20release%201102q%20stamp%2051814</sip_user_agent>
> >    <sip_via_host>1.2.3.4</sip_via_host>
> >    <sip_via_port>9774</sip_via_port>
> >    <sip_via_rport>9774</sip_via_rport>
> >    <max_forwards>70</max_forwards>
> >
> >
>  <switch_r_sdp>v%3D0%0D%0Ao%3D-%207%202%20IN%20IP4%1.2.3.4%0D%0As%3DCounter
> >
> Path%20eyeBeam%201.5%0D%0Ac%3DIN%20IP4%1.2.3.4%0D%0At%3D0%200%0D%0Am%3Daudi
> >
> o%2023746%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-eve
> >
> nt/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0Aa%3Dalt%3A1%203%20%3A%20L2qFFNvS%20
> >
> HwniVnkK%1.2.3.4%2023746%0D%0Aa%3Dalt%3A2%202%20%3A%20HWrMm7HE%20Wk82PRBC%1
> >
> .2.3.4%2023746%0D%0Aa%3Dalt%3A3%201%20%3A%20virMBT30%20GCcqsUuh%1.2.3.4%202
> > 3746%0D%0A</switch_r_sdp> <sip_use_codec_name>PCMU</sip_use_codec_name>
> >    <sip_use_codec_rate>8000</sip_use_codec_rate>
> >    <sip_use_codec_ptime>20</sip_use_codec_ptime>
> >    <read_codec>PCMU</read_codec>
> >    <read_rate>8000</read_rate>
> >    <write_codec>PCMU</write_codec>
> >    <write_rate>8000</write_rate>
> >    <outside_call>true</outside_call>
> >    <hangup_after_bridge>true</hangup_after_bridge>
> >
> >
>  <continue_on_fail>NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UN
> > ALLOCATED_NUMBER,407</continue_on_fail>
> >
> >
>  <current_application_data>sofia/gateway/OPENSIP/0013058883456</current_app
> > lication_data> <current_application>bridge</current_application>
> >    <local_media_ip>1.2.3.4</local_media_ip>
> >    <local_media_port>20342</local_media_port>
> >    <sip_use_pt>0</sip_use_pt>
> >    <rtp_use_ssrc>820599639</rtp_use_ssrc>
> >    <remote_media_ip>1.2.3.4</remote_media_ip>
> >    <remote_media_port>23746</remote_media_port>
> >    <originate_disposition>SUCCESS</originate_disposition>
> >    <bridge_channel>sofia/external/0013058883456</bridge_channel>
> >    <bridge_uuid>5d7e9d6e-3ed9-11e0-a60c-fba7243e1308</bridge_uuid>
> >    <signal_bond>5d7e9d6e-3ed9-11e0-a60c-fba7243e1308</signal_bond>
> >
> >
>  <switch_m_sdp>v%3D0%0D%0Ao%3Droot%2023193%2023194%20IN%20IP4%1.2.3.4%0D%0A
> >
> s%3Dsession%0D%0Ac%3DIN%20IP4%1.2.3.4%0D%0At%3D0%200%0D%0Am%3Daudio%2021586
> >
> %20RTP/AVP%2018%203%200%208%20101%0D%0Aa%3Drtpmap%3A18%20G729/8000%0D%0Aa%3
> >
> Dfmtp%3A18%20annexb%3Dno%0D%0Aa%3Drtpmap%3A3%20GSM/8000%0D%0Aa%3Drtpmap%3A0
> >
> %20PCMU/8000%0D%0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Drtpmap%3A101%20teleph
> >
> one-event/8000%0D%0Aa%3Dfmtp%3A101%200-16%0D%0Aa%3DsilenceSupp%3Aoff%20-%20
> > -%20-%20-%0D%0Aa%3Dptime%3A20%0D%0A</switch_m_sdp>
> >
> >
>  <sip_local_sdp_str>v%3D0%0Ao%3DFreeSWITCH%201298396074%201298396076%20IN%2
> >
> 0IP4%1.2.3.4%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%1.2.3.4%0At%3D0%200%0Am%3Daudi
> >
> o%2020342%20RTP/AVP%200%20101%0Aa%3Drtpmap%3A0%20PCMU/8000%0Aa%3Drtpmap%3A1
> >
> 01%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3DsilenceSupp%3Aoff%2
> > 0-%20-%20-%20-%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A</sip_local_sdp_str>
> > <endpoint_disposition>ANSWER</endpoint_disposition>
> >    <bridge_hangup_cause>NORMAL_CLEARING</bridge_hangup_cause>
> >    <hangup_cause>NORMAL_CLEARING</hangup_cause>
> >    <hangup_cause_q850>16</hangup_cause_q850>
> >    <digits_dialed>none</digits_dialed>
> >    <start_stamp>2011-02-23%2000%3A13%3A31</start_stamp>
> >    <profile_start_stamp>2011-02-23%2000%3A13%3A31</profile_start_stamp>
> >    <answer_stamp>2011-02-23%2000%3A13%3A36</answer_stamp>
> >    <progress_media_stamp>2011-02-23%2000%3A13%3A36</progress_media_stamp>
> >    <end_stamp>2011-02-23%2000%3A13%3A36</end_stamp>
> >    <start_epoch>1298416411</start_epoch>
> >    <start_uepoch>1298416411795887</start_uepoch>
> >    <profile_start_epoch>1298416411</profile_start_epoch>
> >    <profile_start_uepoch>1298416411795887</profile_start_uepoch>
> >    <answer_epoch>1298416416</answer_epoch>
> >    <answer_uepoch>1298416416500657</answer_uepoch>
> >    <resurrect_epoch>0</resurrect_epoch>
> >    <resurrect_uepoch>0</resurrect_uepoch>
> >    <progress_epoch>0</progress_epoch>
> >    <progress_uepoch>0</progress_uepoch>
> >    <progress_media_epoch>1298416416</progress_media_epoch>
> >    <progress_media_uepoch>1298416416470740</progress_media_uepoch>
> >    <end_epoch>1298416416</end_epoch>
> >    <end_uepoch>1298416416562577</end_uepoch>
> >    <last_app>bridge</last_app>
> >    <last_arg>sofia/gateway/OPENSIP/0013058883456</last_arg>
> >    <caller_id>%22david%22%20%3C8889990%3E</caller_id>
> >    <duration>5</duration>
> >    <billsec>0</billsec>
> >    <progresssec>0</progresssec>
> >    <answersec>5</answersec>
> >    <progress_mediasec>5</progress_mediasec>
> >    <flow_billsec>5</flow_billsec>
> >    <mduration>4767</mduration>
> >    <billmsec>62</billmsec>
> >    <progressmsec>0</progressmsec>
> >    <answermsec>4705</answermsec>
> >    <progress_mediamsec>0</progress_mediamsec>
> >    <flow_billmsec>4767</flow_billmsec>
> >    <uduration>4766690</uduration>
> >    <billusec>61920</billusec>
> >    <progressusec>0</progressusec>
> >    <answerusec>4704770</answerusec>
> >    <progress_mediausec>4674853</progress_mediausec>
> >    <flow_billusec>4766690</flow_billusec>
> >    <sip_hangup_disposition>send_bye</sip_hangup_disposition>
> >    <rtp_audio_in_raw_bytes>0</rtp_audio_in_raw_bytes>
> >    <rtp_audio_in_media_bytes>0</rtp_audio_in_media_bytes>
> >    <rtp_audio_in_packet_count>0</rtp_audio_in_packet_count>
> >    <rtp_audio_in_media_packet_count>0</rtp_audio_in_media_packet_count>
> >    <rtp_audio_in_skip_packet_count>5</rtp_audio_in_skip_packet_count>
> >    <rtp_audio_in_jb_packet_count>0</rtp_audio_in_jb_packet_count>
> >    <rtp_audio_in_dtmf_packet_count>0</rtp_audio_in_dtmf_packet_count>
> >    <rtp_audio_in_cng_packet_count>0</rtp_audio_in_cng_packet_count>
> >    <rtp_audio_in_flush_packet_count>0</rtp_audio_in_flush_packet_count>
> >    <rtp_audio_out_raw_bytes>0</rtp_audio_out_raw_bytes>
> >    <rtp_audio_out_media_bytes>0</rtp_audio_out_media_bytes>
> >    <rtp_audio_out_packet_count>0</rtp_audio_out_packet_count>
> >    <rtp_audio_out_media_packet_count>0</rtp_audio_out_media_packet_count>
> >    <rtp_audio_out_skip_packet_count>0</rtp_audio_out_skip_packet_count>
> >    <rtp_audio_out_dtmf_packet_count>0</rtp_audio_out_dtmf_packet_count>
> >    <rtp_audio_out_cng_packet_count>0</rtp_audio_out_cng_packet_count>
> >  </variables>
> >  <app_log>
> >    <application app_name="set"
> app_data="outside_call=true"></application>
> >    <application app_name="info" app_data=""></application>
> >    <application app_name="set"
> > app_data="hangup_after_bridge=true"></application>
> >    <application app_name="set"
> >
> app_data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTIN
> > ATION,UNALLOCATED_NUMBER,407"></application> <application
> app_name="bridge"
> > app_data="sofia/gateway/OPENSIP/0013058883456"></application>
> >  </app_log>
> >  <callflow dialplan="XML" profile_index="1">
> >    <extension name="outside_call" number="0013058883456"
> > current_app="bridge">
> >      <application app_name="set"
> > app_data="outside_call=true"></application> <application app_name="info"
> > app_data=""></application>
> >      <application app_name="set"
> > app_data="hangup_after_bridge=true"></application>
> >      <application app_name="set"
> >
> app_data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTIN
> > ATION,UNALLOCATED_NUMBER,407"></application> <application
> app_name="bridge"
> > app_data="sofia/gateway/${distributor(test)}13058883456"></application>
> >      <application last_executed="true" app_name="bridge"
> > app_data="sofia/gateway/${distributor(test)}13058883456"></application>
> >    </extension>
> >    <caller_profile>
> >      <username>8889990</username>
> >      <dialplan>XML</dialplan>
> >      <caller_id_name>david</caller_id_name>
> >      <ani>8889990</ani>
> >      <aniii></aniii>
> >      <caller_id_number>8889990</caller_id_number>
> >      <network_addr>1.2.3.4</network_addr>
> >      <rdnis></rdnis>
> >      <destination_number>0013058883456</destination_number>
> >      <uuid>5d75e976-3ed9-11e0-a60b-fba7243e1308</uuid>
> >      <source>mod_sofia</source>
> >      <context>public</context>
> >      <chan_name>sofia/external/8889990 at 1.2.3.4:5060</chan_name>
> >      <originatee>
> >        <originatee_caller_profile>
> >          <username>8889990</username>
> >          <dialplan>XML</dialplan>
> >          <caller_id_name>david</caller_id_name>
> >          <ani>8889990</ani>
> >          <aniii></aniii>
> >          <caller_id_number>8889990</caller_id_number>
> >          <network_addr>1.2.3.4</network_addr>
> >          <rdnis></rdnis>
> >          <destination_number>0013058883456</destination_number>
> >          <uuid>5d7e9d6e-3ed9-11e0-a60c-fba7243e1308</uuid>
> >          <source>mod_sofia</source>
> >          <context>public</context>
> >          <chan_name>sofia/external/0013058883456</chan_name>
> >        </originatee_caller_profile>
> >      </originatee>
> >    </caller_profile>
> >    <times>
> >      <created_time>1298416411795887</created_time>
> >      <profile_created_time>1298416411795887</profile_created_time>
> >      <progress_time>0</progress_time>
> >      <progress_media_time>1298416416470740</progress_media_time>
> >      <answered_time>1298416416500657</answered_time>
> >      <hangup_time>1298416416562577</hangup_time>
> >      <resurrect_time>0</resurrect_time>
> >      <transfer_time>0</transfer_time>
> >    </times>
> >  </callflow>
> > </cdr>
> >
> >
> > print_r
> >
> > Array
> > (
> >    [] => Array
> >        (
> >            [channel_data] =>
> >
> >            [state] => CS_REPORTING
> >            [direction] => inbound
> >            [state_number] => 11
> >            [flags] => 0=1;1=1;3=1;35=1;36=1;38=1;46=1;51=1
> >            [caps] => 1=1;2=1;3=1
> >        )
> >
> >    [variables] => Array
> >        (
> >            [uuid] => 5d75e976-3ed9-11e0-a60b-fba7243e1308
> >            [sip_local_network_addr] => 1.2.3.4
> >            [sip_network_ip] => 1.2.3.4
> >            [sip_network_port] => 9774
> >            [sip_received_ip] => 1.2.3.4
> >            [sip_received_port] => 9774
> >            [sip_via_protocol] => udp
> >            [sip_from_user] => 8889990
> >            [sip_from_port] => 5060
> >            [sip_from_uri] => 8889990%1.2.3.4%3A5060
> >            [sip_from_host] => 1.2.3.4
> >            [sip_from_user_stripped] => 8889990
> >            [sip_from_tag] => 2e41ba6e
> >            [sofia_profile_name] => external
> >            [sip_full_via] =>
> >
> SIP/2.0/UDP%1.2.3.4%3A9774%3Bbranch%3Dz9hG4bK-d8754z-c551a41eb93bd648-1---d
> > 8754z-%3Brport%3D9774%3Breceived%3D1.2.3.4 [sip_from_display] => david
> >            [sip_full_from] =>
> > %22david%22%20%3Csip%3A8889990%1.2.3.4%3A5060%3E%3Btag%3D2e41ba6e
> >            [sip_to_display] => 0013058883456
> >            [sip_full_to] =>
> > %220013058883456%22%20%3Csip%3A0013058883456%1.2.3.4%3A5060%3E
> >            [sip_req_user] => 0013058883456
> >            [sip_req_port] => 5060
> >            [sip_req_uri] => 0013058883456%1.2.3.4%3A5060
> >            [sip_req_host] => 1.2.3.4
> >            [sip_to_user] => 0013058883456
> >            [sip_to_port] => 5060
> >            [sip_to_uri] => 0013058883456%1.2.3.4%3A5060
> >            [sip_to_host] => 1.2.3.4
> >            [sip_contact_user] => 8889990
> >            [sip_contact_port] => 9774
> >            [sip_contact_uri] => 8889990%1.2.3.4%3A9774
> >            [sip_contact_host] => 1.2.3.4
> >            [channel_name] => sofia/external/8889990%1.2.3.4%3A5060
> >            [sip_call_id] => OGVkMmMyMmQ5MDEyOTU2MWUyN2M2NzBhYjk5ZmQzOTM.
> >            [sip_user_agent] => eyeBeam%20release%201102q%20stamp%2051814
> >            [sip_via_host] => 1.2.3.4
> >            [sip_via_port] => 9774
> >            [sip_via_rport] => 9774
> >            [max_forwards] => 70
> >            [switch_r_sdp] =>
> >
> v%3D0%0D%0Ao%3D-%207%202%20IN%20IP4%1.2.3.4%0D%0As%3DCounterPath%20eyeBeam%
> >
> 201.5%0D%0Ac%3DIN%20IP4%1.2.3.4%0D%0At%3D0%200%0D%0Am%3Daudio%2023746%20RTP
> >
> /AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%
> >
> 3Dfmtp%3A101%200-15%0D%0Aa%3Dalt%3A1%203%20%3A%20L2qFFNvS%20HwniVnkK%1.2.3.
> >
> 4%2023746%0D%0Aa%3Dalt%3A2%202%20%3A%20HWrMm7HE%20Wk82PRBC%1.2.3.4%2023746%
> > 0D%0Aa%3Dalt%3A3%201%20%3A%20virMBT30%20GCcqsUuh%1.2.3.4%2023746%0D%0A
> > [sip_use_codec_name] => PCMU
> >            [sip_use_codec_rate] => 8000
> >            [sip_use_codec_ptime] => 20
> >            [read_codec] => PCMU
> >            [read_rate] => 8000
> >            [write_codec] => PCMU
> >            [write_rate] => 8000
> >            [outside_call] => true
> >            [hangup_after_bridge] => true
> >            [continue_on_fail] =>
> >
> NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,40
> > 7 [current_application_data] => sofia/gateway/OPENSIP/0013058883456
> > [current_application] => bridge
> >            [local_media_ip] => 1.2.3.4
> >            [local_media_port] => 20342
> >            [sip_use_pt] => 0
> >            [rtp_use_ssrc] => 820599639
> >            [remote_media_ip] => 1.2.3.4
> >            [remote_media_port] => 23746
> >            [originate_disposition] => SUCCESS
> >            [bridge_channel] => sofia/external/0013058883456
> >            [bridge_uuid] => 5d7e9d6e-3ed9-11e0-a60c-fba7243e1308
> >            [signal_bond] => 5d7e9d6e-3ed9-11e0-a60c-fba7243e1308
> >            [switch_m_sdp] =>
> >
> v%3D0%0D%0Ao%3Droot%2023193%2023194%20IN%20IP4%1.2.3.4%0D%0As%3Dsession%0D%
> >
> 0Ac%3DIN%20IP4%1.2.3.4%0D%0At%3D0%200%0D%0Am%3Daudio%2021586%20RTP/AVP%2018
> >
> %203%200%208%20101%0D%0Aa%3Drtpmap%3A18%20G729/8000%0D%0Aa%3Dfmtp%3A18%20an
> >
> nexb%3Dno%0D%0Aa%3Drtpmap%3A3%20GSM/8000%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D
> >
> %0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%
> >
> 0D%0Aa%3Dfmtp%3A101%200-16%0D%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0D%0A
> > a%3Dptime%3A20%0D%0A [sip_local_sdp_str] =>
> >
> v%3D0%0Ao%3DFreeSWITCH%201298396074%201298396076%20IN%20IP4%1.2.3.4%0As%3DF
> >
> reeSWITCH%0Ac%3DIN%20IP4%1.2.3.4%0At%3D0%200%0Am%3Daudio%2020342%20RTP/AVP%
> >
> 200%20101%0Aa%3Drtpmap%3A0%20PCMU/8000%0Aa%3Drtpmap%3A101%20telephone-event
> >
> /8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0Aa%3
> > Dptime%3A20%0Aa%3Dsendrecv%0A [endpoint_disposition] => ANSWER
> >            [bridge_hangup_cause] => NORMAL_CLEARING
> >            [hangup_cause] => NORMAL_CLEARING
> >            [hangup_cause_q850] => 16
> >            [digits_dialed] => none
> >            [start_stamp] => 2011-02-23%2000%3A13%3A31
> >            [profile_start_stamp] => 2011-02-23%2000%3A13%3A31
> >            [answer_stamp] => 2011-02-23%2000%3A13%3A36
> >            [progress_media_stamp] => 2011-02-23%2000%3A13%3A36
> >            [end_stamp] => 2011-02-23%2000%3A13%3A36
> >            [start_epoch] => 1298416411
> >            [start_uepoch] => 1298416411795887
> >            [profile_start_epoch] => 1298416411
> >            [profile_start_uepoch] => 1298416411795887
> >            [answer_epoch] => 1298416416
> >            [answer_uepoch] => 1298416416500657
> >            [resurrect_epoch] => 0
> >            [resurrect_uepoch] => 0
> >            [progress_epoch] => 0
> >            [progress_uepoch] => 0
> >            [progress_media_epoch] => 1298416416
> >            [progress_media_uepoch] => 1298416416470740
> >            [end_epoch] => 1298416416
> >            [end_uepoch] => 1298416416562577
> >            [last_app] => bridge
> >            [last_arg] => sofia/gateway/OPENSIP/0013058883456
> >            [caller_id] => %22david%22%20%3C8889990%3E
> >            [duration] => 5
> >            [billsec] => 0
> >            [progresssec] => 0
> >            [answersec] => 5
> >            [progress_mediasec] => 5
> >            [flow_billsec] => 5
> >            [mduration] => 4767
> >            [billmsec] => 62
> >            [progressmsec] => 0
> >            [answermsec] => 4705
> >            [progress_mediamsec] => 0
> >            [flow_billmsec] => 4767
> >            [uduration] => 4766690
> >            [billusec] => 61920
> >            [progressusec] => 0
> >            [answerusec] => 4704770
> >            [progress_mediausec] => 4674853
> >            [flow_billusec] => 4766690
> >            [sip_hangup_disposition] => send_bye
> >            [rtp_audio_in_raw_bytes] => 0
> >            [rtp_audio_in_media_bytes] => 0
> >            [rtp_audio_in_packet_count] => 0
> >            [rtp_audio_in_media_packet_count] => 0
> >            [rtp_audio_in_skip_packet_count] => 5
> >            [rtp_audio_in_jb_packet_count] => 0
> >            [rtp_audio_in_dtmf_packet_count] => 0
> >            [rtp_audio_in_cng_packet_count] => 0
> >            [rtp_audio_in_flush_packet_count] => 0
> >            [rtp_audio_out_raw_bytes] => 0
> >            [rtp_audio_out_media_bytes] => 0
> >            [rtp_audio_out_packet_count] => 0
> >            [rtp_audio_out_media_packet_count] => 0
> >            [rtp_audio_out_skip_packet_count] => 0
> >            [rtp_audio_out_dtmf_packet_count] => 0
> >            [rtp_audio_out_cng_packet_count] => 0
> >        )
> >
> >    [app_log] => Array
> >        (
> >            [0] => Array
> >                (
> >                    [app_name] => set
> >                    [app_data] => outside_call=true
> >                )
> >
> >            [1] => Array
> >                (
> >                    [app_name] => info
> >                    [app_data] =>
> >                )
> >
> >            [2] => Array
> >                (
> >                    [app_name] => set
> >                    [app_data] => hangup_after_bridge=true
> >                )
> >
> >            [3] => Array
> >                (
> >                    [app_name] => set
> >                    [app_data] =>
> >
> continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNAL
> > LOCATED_NUMBER,407 )
> >
> >            [4] => Array
> >                (
> >                    [app_name] => bridge
> >                    [app_data] => sofia/gateway/OPENSIP/0013058883456
> >                )
> >
> >        )
> >
> >    [callflow] => Array
> >        (
> >            [0] => Array
> >                (
> >                    [dialplan] => XML
> >                    [profile_index] => 1
> >                    [extension] => Array
> >                        (
> >                            [name] => outside_call
> >                            [number] => 0013058883456
> >                            [current_app] => bridge
> >                            [application] => Array
> >                                (
> >                                    [0] => Array
> >                                        (
> >                                            [app_name] => set
> >                                            [app_data] =>
> outside_call=true
> >                                        )
> >
> >                                    [1] => Array
> >                                        (
> >                                            [app_name] => info
> >                                            [app_data] =>
> >                                        )
> >
> >                                    [2] => Array
> >                                        (
> >                                            [app_name] => set
> >                                            [app_data] =>
> > hangup_after_bridge=true
> >                                        )
> >
> >                                    [3] => Array
> >                                        (
> >                                            [app_name] => set
> >                                            [app_data] =>
> >
> continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNAL
> > LOCATED_NUMBER,407 )
> >
> >                                    [4] => Array
> >                                        (
> >                                            [app_name] => bridge
> >                                            [app_data] =>
> > sofia/gateway/${distributor(test)}13058883456
> >                                        )
> >
> >                                    [5] => Array
> >                                        (
> >                                            [last_executed] => true
> >                                            [app_name] => bridge
> >                                            [app_data] =>
> > sofia/gateway/${distributor(test)}13058883456
> >                                        )
> >
> >                                )
> >
> >                        )
> >
> >                    [caller_profile] => Array
> >                        (
> >                            [username] => 8889990
> >                            [dialplan] => XML
> >                            [caller_id_name] => david
> >                            [ani] => 8889990
> >                            [caller_id_number] => 8889990
> >                            [network_addr] => 1.2.3.4
> >                            [destination_number] => 0013058883456
> >                            [uuid] => 5d7e9d6e-3ed9-11e0-a60c-fba7243e1308
> >                            [source] => mod_sofia
> >                            [context] => public
> >                            [chan_name] => sofia/external/0013058883456
> >                            [originatee] =>
> >
> >                            [originatee_caller_profile] =>
> >
> >                        )
> >
> >                    [times] => Array
> >                        (
> >                            [created_time] => 1298416411795887
> >                            [profile_created_time] => 1298416411795887
> >                            [progress_time] => 0
> >                            [progress_media_time] => 1298416416470740
> >                            [answered_time] => 1298416416500657
> >                            [hangup_time] => 1298416416562577
> >                            [resurrect_time] => 0
> >                            [transfer_time] => 0
> >                        )
> >
> >                )
> >
> >        )
> >
> > )
>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
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From steveayre at gmail.com  Wed Feb 23 18:49:25 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 23 Feb 2011 15:49:25 +0000
Subject: [Freeswitch-users] Weird CDR ERROR...
In-Reply-To: <AANLkTi=C8xdPAgsLo9_UB-_5KpkmQ8hCr8eHCA+dj9kv@mail.gmail.com>
References: <AANLkTikVW7K=tvsrFa7azTEtU8CKso+fTiSj6sw6yhAf@mail.gmail.com>
	<201102230911.47818.sos@sokhapkin.dyndns.org>
	<AANLkTi=C8xdPAgsLo9_UB-_5KpkmQ8hCr8eHCA+dj9kv@mail.gmail.com>
Message-ID: <AANLkTi=Mit+Z8Lzy9ovf455YWUhH6qXS9YgPn4DdTSsM@mail.gmail.com>

 <app_log>
   <application app_name="set" app_data="outside_call=true"><
/application>
   <application app_name="info" app_data=""></application>
   <application app_name="set"
app_data="hangup_after_bridge=true"></application>
   <application app_name="set"
app_data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407"></application>
   <application app_name="bridge"
app_data="sofia/gateway/OPENSIP/0013058883456"></application>
 </app_log>

This is what it executed... you'd need a sip trace to show that the OPENSIP
gateway hadn't answered the call.

-Steve



On 23 February 2011 15:36, David Villasmil
<david.villasmil.work at gmail.com>wrote:

> Hello Sergey and thanks for replying.
>
> The call was NOT answered. It actually failed as there was no context for
> that call... that is the weird part... i will try to reproduce it.
>
> Thanks again
>
> David
>
>
> On Wed, Feb 23, 2011 at 3:11 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org
> > wrote:
>
>> <billmsec>62</billmsec>
>>
>> The call was answered for less than 1 second.
>>
>> On Wednesday 23 February 2011, David Villasmil wrote:
>> > Hello All,
>> >
>> > I have an XML_CDR cdr that is not correct, i made some test and on one
>> cdr
>> > (that i could find) which show endpoint_disposition as ANSWER when the
>> call
>> > was NOT answered! The call didn't even go out of the box...
>> >
>> > here's the CDR:
>> >
>> > note that billsec is 0, but disposition is ANSWER... weird, i got more
>> than
>> > one, also...
>> >
>> >
>> >
>> > <cdr>
>> >  <channel_data>
>> >    <state>CS_REPORTING</state>
>> >    <direction>inbound</direction>
>> >    <state_number>11</state_number>
>> >    <flags>0=1;1=1;3=1;35=1;36=1;38=1;46=1;51=1</flags>
>> >    <caps>1=1;2=1;3=1</caps>
>> >  </channel_data>
>> >  <variables>
>> >    <uuid>5d75e976-3ed9-11e0-a60b-fba7243e1308</uuid>
>> >    <sip_local_network_addr>1.2.3.4</sip_local_network_addr>
>> >    <sip_network_ip>1.2.3.4</sip_network_ip>
>> >    <sip_network_port>9774</sip_network_port>
>> >    <sip_received_ip>1.2.3.4</sip_received_ip>
>> >    <sip_received_port>9774</sip_received_port>
>> >    <sip_via_protocol>udp</sip_via_protocol>
>> >    <sip_from_user>8889990</sip_from_user>
>> >    <sip_from_port>5060</sip_from_port>
>> >    <sip_from_uri>8889990%1.2.3.4%3A5060</sip_from_uri>
>> >    <sip_from_host>1.2.3.4</sip_from_host>
>> >    <sip_from_user_stripped>8889990</sip_from_user_stripped>
>> >    <sip_from_tag>2e41ba6e</sip_from_tag>
>> >    <sofia_profile_name>external</sofia_profile_name>
>> >
>> >
>>  <sip_full_via>SIP/2.0/UDP%1.2.3.4%3A9774%3Bbranch%3Dz9hG4bK-d8754z-c551a41
>> > eb93bd648-1---d8754z-%3Brport%3D9774%3Breceived%3D1.2.3.4</sip_full_via>
>> > <sip_from_display>david</sip_from_display>
>> >
>> >
>>  <sip_full_from>%22david%22%20%3Csip%3A8889990%1.2.3.4%3A5060%3E%3Btag%3D2e
>> > 41ba6e</sip_full_from> <sip_to_display>0013058883456</sip_to_display>
>> >
>> >
>>  <sip_full_to>%220013058883456%22%20%3Csip%3A0013058883456%1.2.3.4%3A5060%3
>> > E</sip_full_to> <sip_req_user>0013058883456</sip_req_user>
>> >    <sip_req_port>5060</sip_req_port>
>> >    <sip_req_uri>0013058883456%1.2.3.4%3A5060</sip_req_uri>
>> >    <sip_req_host>1.2.3.4</sip_req_host>
>> >    <sip_to_user>0013058883456</sip_to_user>
>> >    <sip_to_port>5060</sip_to_port>
>> >    <sip_to_uri>0013058883456%1.2.3.4%3A5060</sip_to_uri>
>> >    <sip_to_host>1.2.3.4</sip_to_host>
>> >    <sip_contact_user>8889990</sip_contact_user>
>> >    <sip_contact_port>9774</sip_contact_port>
>> >    <sip_contact_uri>8889990%1.2.3.4%3A9774</sip_contact_uri>
>> >    <sip_contact_host>1.2.3.4</sip_contact_host>
>> >    <channel_name>sofia/external/8889990%1.2.3.4%3A5060</channel_name>
>> >
>>  <sip_call_id>OGVkMmMyMmQ5MDEyOTU2MWUyN2M2NzBhYjk5ZmQzOTM.</sip_call_id>
>> >
>> >
>>  <sip_user_agent>eyeBeam%20release%201102q%20stamp%2051814</sip_user_agent>
>> >    <sip_via_host>1.2.3.4</sip_via_host>
>> >    <sip_via_port>9774</sip_via_port>
>> >    <sip_via_rport>9774</sip_via_rport>
>> >    <max_forwards>70</max_forwards>
>> >
>> >
>>  <switch_r_sdp>v%3D0%0D%0Ao%3D-%207%202%20IN%20IP4%1.2.3.4%0D%0As%3DCounter
>> >
>> Path%20eyeBeam%201.5%0D%0Ac%3DIN%20IP4%1.2.3.4%0D%0At%3D0%200%0D%0Am%3Daudi
>> >
>> o%2023746%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-eve
>> >
>> nt/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0Aa%3Dalt%3A1%203%20%3A%20L2qFFNvS%20
>> >
>> HwniVnkK%1.2.3.4%2023746%0D%0Aa%3Dalt%3A2%202%20%3A%20HWrMm7HE%20Wk82PRBC%1
>> >
>> .2.3.4%2023746%0D%0Aa%3Dalt%3A3%201%20%3A%20virMBT30%20GCcqsUuh%1.2.3.4%202
>> > 3746%0D%0A</switch_r_sdp> <sip_use_codec_name>PCMU</sip_use_codec_name>
>> >    <sip_use_codec_rate>8000</sip_use_codec_rate>
>> >    <sip_use_codec_ptime>20</sip_use_codec_ptime>
>> >    <read_codec>PCMU</read_codec>
>> >    <read_rate>8000</read_rate>
>> >    <write_codec>PCMU</write_codec>
>> >    <write_rate>8000</write_rate>
>> >    <outside_call>true</outside_call>
>> >    <hangup_after_bridge>true</hangup_after_bridge>
>> >
>> >
>>  <continue_on_fail>NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UN
>> > ALLOCATED_NUMBER,407</continue_on_fail>
>> >
>> >
>>  <current_application_data>sofia/gateway/OPENSIP/0013058883456</current_app
>> > lication_data> <current_application>bridge</current_application>
>> >    <local_media_ip>1.2.3.4</local_media_ip>
>> >    <local_media_port>20342</local_media_port>
>> >    <sip_use_pt>0</sip_use_pt>
>> >    <rtp_use_ssrc>820599639</rtp_use_ssrc>
>> >    <remote_media_ip>1.2.3.4</remote_media_ip>
>> >    <remote_media_port>23746</remote_media_port>
>> >    <originate_disposition>SUCCESS</originate_disposition>
>> >    <bridge_channel>sofia/external/0013058883456</bridge_channel>
>> >    <bridge_uuid>5d7e9d6e-3ed9-11e0-a60c-fba7243e1308</bridge_uuid>
>> >    <signal_bond>5d7e9d6e-3ed9-11e0-a60c-fba7243e1308</signal_bond>
>> >
>> >
>>  <switch_m_sdp>v%3D0%0D%0Ao%3Droot%2023193%2023194%20IN%20IP4%1.2.3.4%0D%0A
>> >
>> s%3Dsession%0D%0Ac%3DIN%20IP4%1.2.3.4%0D%0At%3D0%200%0D%0Am%3Daudio%2021586
>> >
>> %20RTP/AVP%2018%203%200%208%20101%0D%0Aa%3Drtpmap%3A18%20G729/8000%0D%0Aa%3
>> >
>> Dfmtp%3A18%20annexb%3Dno%0D%0Aa%3Drtpmap%3A3%20GSM/8000%0D%0Aa%3Drtpmap%3A0
>> >
>> %20PCMU/8000%0D%0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Drtpmap%3A101%20teleph
>> >
>> one-event/8000%0D%0Aa%3Dfmtp%3A101%200-16%0D%0Aa%3DsilenceSupp%3Aoff%20-%20
>> > -%20-%20-%0D%0Aa%3Dptime%3A20%0D%0A</switch_m_sdp>
>> >
>> >
>>  <sip_local_sdp_str>v%3D0%0Ao%3DFreeSWITCH%201298396074%201298396076%20IN%2
>> >
>> 0IP4%1.2.3.4%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%1.2.3.4%0At%3D0%200%0Am%3Daudi
>> >
>> o%2020342%20RTP/AVP%200%20101%0Aa%3Drtpmap%3A0%20PCMU/8000%0Aa%3Drtpmap%3A1
>> >
>> 01%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3DsilenceSupp%3Aoff%2
>> > 0-%20-%20-%20-%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A</sip_local_sdp_str>
>> > <endpoint_disposition>ANSWER</endpoint_disposition>
>> >    <bridge_hangup_cause>NORMAL_CLEARING</bridge_hangup_cause>
>> >    <hangup_cause>NORMAL_CLEARING</hangup_cause>
>> >    <hangup_cause_q850>16</hangup_cause_q850>
>> >    <digits_dialed>none</digits_dialed>
>> >    <start_stamp>2011-02-23%2000%3A13%3A31</start_stamp>
>> >    <profile_start_stamp>2011-02-23%2000%3A13%3A31</profile_start_stamp>
>> >    <answer_stamp>2011-02-23%2000%3A13%3A36</answer_stamp>
>> >
>>  <progress_media_stamp>2011-02-23%2000%3A13%3A36</progress_media_stamp>
>> >    <end_stamp>2011-02-23%2000%3A13%3A36</end_stamp>
>> >    <start_epoch>1298416411</start_epoch>
>> >    <start_uepoch>1298416411795887</start_uepoch>
>> >    <profile_start_epoch>1298416411</profile_start_epoch>
>> >    <profile_start_uepoch>1298416411795887</profile_start_uepoch>
>> >    <answer_epoch>1298416416</answer_epoch>
>> >    <answer_uepoch>1298416416500657</answer_uepoch>
>> >    <resurrect_epoch>0</resurrect_epoch>
>> >    <resurrect_uepoch>0</resurrect_uepoch>
>> >    <progress_epoch>0</progress_epoch>
>> >    <progress_uepoch>0</progress_uepoch>
>> >    <progress_media_epoch>1298416416</progress_media_epoch>
>> >    <progress_media_uepoch>1298416416470740</progress_media_uepoch>
>> >    <end_epoch>1298416416</end_epoch>
>> >    <end_uepoch>1298416416562577</end_uepoch>
>> >    <last_app>bridge</last_app>
>> >    <last_arg>sofia/gateway/OPENSIP/0013058883456</last_arg>
>> >    <caller_id>%22david%22%20%3C8889990%3E</caller_id>
>> >    <duration>5</duration>
>> >    <billsec>0</billsec>
>> >    <progresssec>0</progresssec>
>> >    <answersec>5</answersec>
>> >    <progress_mediasec>5</progress_mediasec>
>> >    <flow_billsec>5</flow_billsec>
>> >    <mduration>4767</mduration>
>> >    <billmsec>62</billmsec>
>> >    <progressmsec>0</progressmsec>
>> >    <answermsec>4705</answermsec>
>> >    <progress_mediamsec>0</progress_mediamsec>
>> >    <flow_billmsec>4767</flow_billmsec>
>> >    <uduration>4766690</uduration>
>> >    <billusec>61920</billusec>
>> >    <progressusec>0</progressusec>
>> >    <answerusec>4704770</answerusec>
>> >    <progress_mediausec>4674853</progress_mediausec>
>> >    <flow_billusec>4766690</flow_billusec>
>> >    <sip_hangup_disposition>send_bye</sip_hangup_disposition>
>> >    <rtp_audio_in_raw_bytes>0</rtp_audio_in_raw_bytes>
>> >    <rtp_audio_in_media_bytes>0</rtp_audio_in_media_bytes>
>> >    <rtp_audio_in_packet_count>0</rtp_audio_in_packet_count>
>> >    <rtp_audio_in_media_packet_count>0</rtp_audio_in_media_packet_count>
>> >    <rtp_audio_in_skip_packet_count>5</rtp_audio_in_skip_packet_count>
>> >    <rtp_audio_in_jb_packet_count>0</rtp_audio_in_jb_packet_count>
>> >    <rtp_audio_in_dtmf_packet_count>0</rtp_audio_in_dtmf_packet_count>
>> >    <rtp_audio_in_cng_packet_count>0</rtp_audio_in_cng_packet_count>
>> >    <rtp_audio_in_flush_packet_count>0</rtp_audio_in_flush_packet_count>
>> >    <rtp_audio_out_raw_bytes>0</rtp_audio_out_raw_bytes>
>> >    <rtp_audio_out_media_bytes>0</rtp_audio_out_media_bytes>
>> >    <rtp_audio_out_packet_count>0</rtp_audio_out_packet_count>
>> >
>>  <rtp_audio_out_media_packet_count>0</rtp_audio_out_media_packet_count>
>> >    <rtp_audio_out_skip_packet_count>0</rtp_audio_out_skip_packet_count>
>> >    <rtp_audio_out_dtmf_packet_count>0</rtp_audio_out_dtmf_packet_count>
>> >    <rtp_audio_out_cng_packet_count>0</rtp_audio_out_cng_packet_count>
>> >  </variables>
>> >  <app_log>
>> >    <application app_name="set"
>> app_data="outside_call=true"></application>
>> >    <application app_name="info" app_data=""></application>
>> >    <application app_name="set"
>> > app_data="hangup_after_bridge=true"></application>
>> >    <application app_name="set"
>> >
>> app_data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTIN
>> > ATION,UNALLOCATED_NUMBER,407"></application> <application
>> app_name="bridge"
>> > app_data="sofia/gateway/OPENSIP/0013058883456"></application>
>> >  </app_log>
>> >  <callflow dialplan="XML" profile_index="1">
>> >    <extension name="outside_call" number="0013058883456"
>> > current_app="bridge">
>> >      <application app_name="set"
>> > app_data="outside_call=true"></application> <application app_name="info"
>> > app_data=""></application>
>> >      <application app_name="set"
>> > app_data="hangup_after_bridge=true"></application>
>> >      <application app_name="set"
>> >
>> app_data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTIN
>> > ATION,UNALLOCATED_NUMBER,407"></application> <application
>> app_name="bridge"
>> > app_data="sofia/gateway/${distributor(test)}13058883456"></application>
>> >      <application last_executed="true" app_name="bridge"
>> > app_data="sofia/gateway/${distributor(test)}13058883456"></application>
>> >    </extension>
>> >    <caller_profile>
>> >      <username>8889990</username>
>> >      <dialplan>XML</dialplan>
>> >      <caller_id_name>david</caller_id_name>
>> >      <ani>8889990</ani>
>> >      <aniii></aniii>
>> >      <caller_id_number>8889990</caller_id_number>
>> >      <network_addr>1.2.3.4</network_addr>
>> >      <rdnis></rdnis>
>> >      <destination_number>0013058883456</destination_number>
>> >      <uuid>5d75e976-3ed9-11e0-a60b-fba7243e1308</uuid>
>> >      <source>mod_sofia</source>
>> >      <context>public</context>
>> >      <chan_name>sofia/external/8889990 at 1.2.3.4:5060</chan_name>
>> >      <originatee>
>> >        <originatee_caller_profile>
>> >          <username>8889990</username>
>> >          <dialplan>XML</dialplan>
>> >          <caller_id_name>david</caller_id_name>
>> >          <ani>8889990</ani>
>> >          <aniii></aniii>
>> >          <caller_id_number>8889990</caller_id_number>
>> >          <network_addr>1.2.3.4</network_addr>
>> >          <rdnis></rdnis>
>> >          <destination_number>0013058883456</destination_number>
>> >          <uuid>5d7e9d6e-3ed9-11e0-a60c-fba7243e1308</uuid>
>> >          <source>mod_sofia</source>
>> >          <context>public</context>
>> >          <chan_name>sofia/external/0013058883456</chan_name>
>> >        </originatee_caller_profile>
>> >      </originatee>
>> >    </caller_profile>
>> >    <times>
>> >      <created_time>1298416411795887</created_time>
>> >      <profile_created_time>1298416411795887</profile_created_time>
>> >      <progress_time>0</progress_time>
>> >      <progress_media_time>1298416416470740</progress_media_time>
>> >      <answered_time>1298416416500657</answered_time>
>> >      <hangup_time>1298416416562577</hangup_time>
>> >      <resurrect_time>0</resurrect_time>
>> >      <transfer_time>0</transfer_time>
>> >    </times>
>> >  </callflow>
>> > </cdr>
>> >
>> >
>> > print_r
>> >
>> > Array
>> > (
>> >    [] => Array
>> >        (
>> >            [channel_data] =>
>> >
>> >            [state] => CS_REPORTING
>> >            [direction] => inbound
>> >            [state_number] => 11
>> >            [flags] => 0=1;1=1;3=1;35=1;36=1;38=1;46=1;51=1
>> >            [caps] => 1=1;2=1;3=1
>> >        )
>> >
>> >    [variables] => Array
>> >        (
>> >            [uuid] => 5d75e976-3ed9-11e0-a60b-fba7243e1308
>> >            [sip_local_network_addr] => 1.2.3.4
>> >            [sip_network_ip] => 1.2.3.4
>> >            [sip_network_port] => 9774
>> >            [sip_received_ip] => 1.2.3.4
>> >            [sip_received_port] => 9774
>> >            [sip_via_protocol] => udp
>> >            [sip_from_user] => 8889990
>> >            [sip_from_port] => 5060
>> >            [sip_from_uri] => 8889990%1.2.3.4%3A5060
>> >            [sip_from_host] => 1.2.3.4
>> >            [sip_from_user_stripped] => 8889990
>> >            [sip_from_tag] => 2e41ba6e
>> >            [sofia_profile_name] => external
>> >            [sip_full_via] =>
>> >
>> SIP/2.0/UDP%1.2.3.4%3A9774%3Bbranch%3Dz9hG4bK-d8754z-c551a41eb93bd648-1---d
>> > 8754z-%3Brport%3D9774%3Breceived%3D1.2.3.4 [sip_from_display] => david
>> >            [sip_full_from] =>
>> > %22david%22%20%3Csip%3A8889990%1.2.3.4%3A5060%3E%3Btag%3D2e41ba6e
>> >            [sip_to_display] => 0013058883456
>> >            [sip_full_to] =>
>> > %220013058883456%22%20%3Csip%3A0013058883456%1.2.3.4%3A5060%3E
>> >            [sip_req_user] => 0013058883456
>> >            [sip_req_port] => 5060
>> >            [sip_req_uri] => 0013058883456%1.2.3.4%3A5060
>> >            [sip_req_host] => 1.2.3.4
>> >            [sip_to_user] => 0013058883456
>> >            [sip_to_port] => 5060
>> >            [sip_to_uri] => 0013058883456%1.2.3.4%3A5060
>> >            [sip_to_host] => 1.2.3.4
>> >            [sip_contact_user] => 8889990
>> >            [sip_contact_port] => 9774
>> >            [sip_contact_uri] => 8889990%1.2.3.4%3A9774
>> >            [sip_contact_host] => 1.2.3.4
>> >            [channel_name] => sofia/external/8889990%1.2.3.4%3A5060
>> >            [sip_call_id] => OGVkMmMyMmQ5MDEyOTU2MWUyN2M2NzBhYjk5ZmQzOTM.
>> >            [sip_user_agent] => eyeBeam%20release%201102q%20stamp%2051814
>> >            [sip_via_host] => 1.2.3.4
>> >            [sip_via_port] => 9774
>> >            [sip_via_rport] => 9774
>> >            [max_forwards] => 70
>> >            [switch_r_sdp] =>
>> >
>> v%3D0%0D%0Ao%3D-%207%202%20IN%20IP4%1.2.3.4%0D%0As%3DCounterPath%20eyeBeam%
>> >
>> 201.5%0D%0Ac%3DIN%20IP4%1.2.3.4%0D%0At%3D0%200%0D%0Am%3Daudio%2023746%20RTP
>> >
>> /AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%
>> >
>> 3Dfmtp%3A101%200-15%0D%0Aa%3Dalt%3A1%203%20%3A%20L2qFFNvS%20HwniVnkK%1.2.3.
>> >
>> 4%2023746%0D%0Aa%3Dalt%3A2%202%20%3A%20HWrMm7HE%20Wk82PRBC%1.2.3.4%2023746%
>> > 0D%0Aa%3Dalt%3A3%201%20%3A%20virMBT30%20GCcqsUuh%1.2.3.4%2023746%0D%0A
>> > [sip_use_codec_name] => PCMU
>> >            [sip_use_codec_rate] => 8000
>> >            [sip_use_codec_ptime] => 20
>> >            [read_codec] => PCMU
>> >            [read_rate] => 8000
>> >            [write_codec] => PCMU
>> >            [write_rate] => 8000
>> >            [outside_call] => true
>> >            [hangup_after_bridge] => true
>> >            [continue_on_fail] =>
>> >
>> NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,40
>> > 7 [current_application_data] => sofia/gateway/OPENSIP/0013058883456
>> > [current_application] => bridge
>> >            [local_media_ip] => 1.2.3.4
>> >            [local_media_port] => 20342
>> >            [sip_use_pt] => 0
>> >            [rtp_use_ssrc] => 820599639
>> >            [remote_media_ip] => 1.2.3.4
>> >            [remote_media_port] => 23746
>> >            [originate_disposition] => SUCCESS
>> >            [bridge_channel] => sofia/external/0013058883456
>> >            [bridge_uuid] => 5d7e9d6e-3ed9-11e0-a60c-fba7243e1308
>> >            [signal_bond] => 5d7e9d6e-3ed9-11e0-a60c-fba7243e1308
>> >            [switch_m_sdp] =>
>> >
>> v%3D0%0D%0Ao%3Droot%2023193%2023194%20IN%20IP4%1.2.3.4%0D%0As%3Dsession%0D%
>> >
>> 0Ac%3DIN%20IP4%1.2.3.4%0D%0At%3D0%200%0D%0Am%3Daudio%2021586%20RTP/AVP%2018
>> >
>> %203%200%208%20101%0D%0Aa%3Drtpmap%3A18%20G729/8000%0D%0Aa%3Dfmtp%3A18%20an
>> >
>> nexb%3Dno%0D%0Aa%3Drtpmap%3A3%20GSM/8000%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D
>> >
>> %0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%
>> >
>> 0D%0Aa%3Dfmtp%3A101%200-16%0D%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0D%0A
>> > a%3Dptime%3A20%0D%0A [sip_local_sdp_str] =>
>> >
>> v%3D0%0Ao%3DFreeSWITCH%201298396074%201298396076%20IN%20IP4%1.2.3.4%0As%3DF
>> >
>> reeSWITCH%0Ac%3DIN%20IP4%1.2.3.4%0At%3D0%200%0Am%3Daudio%2020342%20RTP/AVP%
>> >
>> 200%20101%0Aa%3Drtpmap%3A0%20PCMU/8000%0Aa%3Drtpmap%3A101%20telephone-event
>> >
>> /8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0Aa%3
>> > Dptime%3A20%0Aa%3Dsendrecv%0A [endpoint_disposition] => ANSWER
>> >            [bridge_hangup_cause] => NORMAL_CLEARING
>> >            [hangup_cause] => NORMAL_CLEARING
>> >            [hangup_cause_q850] => 16
>> >            [digits_dialed] => none
>> >            [start_stamp] => 2011-02-23%2000%3A13%3A31
>> >            [profile_start_stamp] => 2011-02-23%2000%3A13%3A31
>> >            [answer_stamp] => 2011-02-23%2000%3A13%3A36
>> >            [progress_media_stamp] => 2011-02-23%2000%3A13%3A36
>> >            [end_stamp] => 2011-02-23%2000%3A13%3A36
>> >            [start_epoch] => 1298416411
>> >            [start_uepoch] => 1298416411795887
>> >            [profile_start_epoch] => 1298416411
>> >            [profile_start_uepoch] => 1298416411795887
>> >            [answer_epoch] => 1298416416
>> >            [answer_uepoch] => 1298416416500657
>> >            [resurrect_epoch] => 0
>> >            [resurrect_uepoch] => 0
>> >            [progress_epoch] => 0
>> >            [progress_uepoch] => 0
>> >            [progress_media_epoch] => 1298416416
>> >            [progress_media_uepoch] => 1298416416470740
>> >            [end_epoch] => 1298416416
>> >            [end_uepoch] => 1298416416562577
>> >            [last_app] => bridge
>> >            [last_arg] => sofia/gateway/OPENSIP/0013058883456
>> >            [caller_id] => %22david%22%20%3C8889990%3E
>> >            [duration] => 5
>> >            [billsec] => 0
>> >            [progresssec] => 0
>> >            [answersec] => 5
>> >            [progress_mediasec] => 5
>> >            [flow_billsec] => 5
>> >            [mduration] => 4767
>> >            [billmsec] => 62
>> >            [progressmsec] => 0
>> >            [answermsec] => 4705
>> >            [progress_mediamsec] => 0
>> >            [flow_billmsec] => 4767
>> >            [uduration] => 4766690
>> >            [billusec] => 61920
>> >            [progressusec] => 0
>> >            [answerusec] => 4704770
>> >            [progress_mediausec] => 4674853
>> >            [flow_billusec] => 4766690
>> >            [sip_hangup_disposition] => send_bye
>> >            [rtp_audio_in_raw_bytes] => 0
>> >            [rtp_audio_in_media_bytes] => 0
>> >            [rtp_audio_in_packet_count] => 0
>> >            [rtp_audio_in_media_packet_count] => 0
>> >            [rtp_audio_in_skip_packet_count] => 5
>> >            [rtp_audio_in_jb_packet_count] => 0
>> >            [rtp_audio_in_dtmf_packet_count] => 0
>> >            [rtp_audio_in_cng_packet_count] => 0
>> >            [rtp_audio_in_flush_packet_count] => 0
>> >            [rtp_audio_out_raw_bytes] => 0
>> >            [rtp_audio_out_media_bytes] => 0
>> >            [rtp_audio_out_packet_count] => 0
>> >            [rtp_audio_out_media_packet_count] => 0
>> >            [rtp_audio_out_skip_packet_count] => 0
>> >            [rtp_audio_out_dtmf_packet_count] => 0
>> >            [rtp_audio_out_cng_packet_count] => 0
>> >        )
>> >
>> >    [app_log] => Array
>> >        (
>> >            [0] => Array
>> >                (
>> >                    [app_name] => set
>> >                    [app_data] => outside_call=true
>> >                )
>> >
>> >            [1] => Array
>> >                (
>> >                    [app_name] => info
>> >                    [app_data] =>
>> >                )
>> >
>> >            [2] => Array
>> >                (
>> >                    [app_name] => set
>> >                    [app_data] => hangup_after_bridge=true
>> >                )
>> >
>> >            [3] => Array
>> >                (
>> >                    [app_name] => set
>> >                    [app_data] =>
>> >
>> continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNAL
>> > LOCATED_NUMBER,407 )
>> >
>> >            [4] => Array
>> >                (
>> >                    [app_name] => bridge
>> >                    [app_data] => sofia/gateway/OPENSIP/0013058883456
>> >                )
>> >
>> >        )
>> >
>> >    [callflow] => Array
>> >        (
>> >            [0] => Array
>> >                (
>> >                    [dialplan] => XML
>> >                    [profile_index] => 1
>> >                    [extension] => Array
>> >                        (
>> >                            [name] => outside_call
>> >                            [number] => 0013058883456
>> >                            [current_app] => bridge
>> >                            [application] => Array
>> >                                (
>> >                                    [0] => Array
>> >                                        (
>> >                                            [app_name] => set
>> >                                            [app_data] =>
>> outside_call=true
>> >                                        )
>> >
>> >                                    [1] => Array
>> >                                        (
>> >                                            [app_name] => info
>> >                                            [app_data] =>
>> >                                        )
>> >
>> >                                    [2] => Array
>> >                                        (
>> >                                            [app_name] => set
>> >                                            [app_data] =>
>> > hangup_after_bridge=true
>> >                                        )
>> >
>> >                                    [3] => Array
>> >                                        (
>> >                                            [app_name] => set
>> >                                            [app_data] =>
>> >
>> continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNAL
>> > LOCATED_NUMBER,407 )
>> >
>> >                                    [4] => Array
>> >                                        (
>> >                                            [app_name] => bridge
>> >                                            [app_data] =>
>> > sofia/gateway/${distributor(test)}13058883456
>> >                                        )
>> >
>> >                                    [5] => Array
>> >                                        (
>> >                                            [last_executed] => true
>> >                                            [app_name] => bridge
>> >                                            [app_data] =>
>> > sofia/gateway/${distributor(test)}13058883456
>> >                                        )
>> >
>> >                                )
>> >
>> >                        )
>> >
>> >                    [caller_profile] => Array
>> >                        (
>> >                            [username] => 8889990
>> >                            [dialplan] => XML
>> >                            [caller_id_name] => david
>> >                            [ani] => 8889990
>> >                            [caller_id_number] => 8889990
>> >                            [network_addr] => 1.2.3.4
>> >                            [destination_number] => 0013058883456
>> >                            [uuid] =>
>> 5d7e9d6e-3ed9-11e0-a60c-fba7243e1308
>> >                            [source] => mod_sofia
>> >                            [context] => public
>> >                            [chan_name] => sofia/external/0013058883456
>> >                            [originatee] =>
>> >
>> >                            [originatee_caller_profile] =>
>> >
>> >                        )
>> >
>> >                    [times] => Array
>> >                        (
>> >                            [created_time] => 1298416411795887
>> >                            [profile_created_time] => 1298416411795887
>> >                            [progress_time] => 0
>> >                            [progress_media_time] => 1298416416470740
>> >                            [answered_time] => 1298416416500657
>> >                            [hangup_time] => 1298416416562577
>> >                            [resurrect_time] => 0
>> >                            [transfer_time] => 0
>> >                        )
>> >
>> >                )
>> >
>> >        )
>> >
>> > )
>>
>>
>> _______________________________________________
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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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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From steveayre at gmail.com  Wed Feb 23 19:02:27 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 23 Feb 2011 16:02:27 +0000
Subject: [Freeswitch-users] INVITE DoS Prevention
In-Reply-To: <1298473808528-6056642.post@n2.nabble.com>
References: <6461E11A-84B5-40AF-96BE-C68F3DF91D8E@5ninesolutions.com>
	<C98764B0.11627%krice@freeswitch.org>
	<A624F826-AF73-40BD-8DCD-10F39551D924@5ninesolutions.com>
	<AANLkTi=+f0GqChj4PeW=XFjQW_mYTtJYPuWEj47gw8r+@mail.gmail.com>
	<1298473808528-6056642.post@n2.nabble.com>
Message-ID: <AANLkTi=uN_TAqkA0Hbq27kQ0HVWrOxdwfTMzq6otc07v@mail.gmail.com>

That's be within the Sofia stack and it had to acknowledge the INVITE with a
100 Trying otherwise the INVITE either resends or gives up the same as a
packet drop. Sleeping would mean keeping the INVITE in memory for longer
while starting the session before accepting/rejecting it, increasing memory
usage under a DOS attack and therefore making FS fall over faster. It'd also
increase the complexity of the code starting up a session while it puts new
invites aside and schedules them to be processed shortly afterwards. You
couldn't just do a sleep as it'd probably lock Sofia up. Doesn't seem worth
it.

Iptables is a much better way of handling it. You can rate limit per-host.

-Steve


On 23 February 2011 15:10, mazilo <Nabble at slickdeals.endjunk.com> wrote:

>
>
> jay binks wrote:
> > as for rate-limiting responses you can have iptables drop packets over X
> > number of invites per sec ...
> Just a thought. Perhaps, we should contemplate to add a feature on FS to
> set
> maximum of invites/sec/host. When the invites max out, add some sleep to
> slow down the response to the requested host. This will probably slow down
> the bot, especially if the bot is trying to hit a lot of FS servers out
> there.
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/INVITE-DoS-Prevention-tp6047615p6056642.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
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From anthony.minessale at gmail.com  Wed Feb 23 19:20:52 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 23 Feb 2011 10:20:52 -0600
Subject: [Freeswitch-users] outbound-caller-name missing?
In-Reply-To: <AANLkTimZnnZYrHmaHVX3dW4KA4kfksEX=zaaokuwLYb-@mail.gmail.com>
References: <AANLkTimZnnZYrHmaHVX3dW4KA4kfksEX=zaaokuwLYb-@mail.gmail.com>
Message-ID: <AANLkTim3ep_yzfesqzoB7exZVWQJJB7yRNVqjvtqRE9Z@mail.gmail.com>

That is normal.

All that stuff in vars.xml is just example config its not code in FS.
Try git grep outbound_caller_id

You are, and always will be, required to specify the caller_id on 1
legged outbound calls.


On Wed, Feb 23, 2011 at 6:00 AM, Seven Du <dujinfang at gmail.com> wrote:
> Hi,
>
>
> I tried
>
> FreeSWITCH Version 1.0.head (git-f5dafc9 2011-01-21 23-42-30 +0100)
> FreeSWITCH Version 1.0.head (git-06988e1 2011-02-20 15-10-02 -0600)
>
>
> originate user/1000 &echo got this:
>
> ? INVITE sip:tbskvnyi at 192.168.7.2:61901 SIP/2.0
> ? Via: SIP/2.0/UDP 192.168.7.2;rport;branch=z9hG4bK0USe34K3r46gF
> ? Max-Forwards: 70
> ? From: "" <sip:0000000000 at 192.168.7.2>;tag=cZt7ZSBZjQK3H
> ? Remote-Party-ID:
> <sip:0000000000 at 192.168.7.2>;party=calling;screen=yes;privacy=off
>
>
> And some versions of Xlite4 fails to respond to this
>
> originate {origination_caller_id_name=xxx}user/1000 works.
>
> I haven't got time to test git head (02-20 almost head), ?just want to
> ask here anyone noticed this, let me know if need a jira.
>
> btw, vars.xml is untouched:
>
> ?<X-PRE-PROCESS cmd="set" data="outbound_caller_name=FreeSWITCH"/>
> ?<X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/>
>
> Thanks.
>
> --
> 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
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 helmut.kuper at ewetel.de  Wed Feb 23 19:21:23 2011
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 23 Feb 2011 17:21:23 +0100
Subject: [Freeswitch-users] intercom in lua dialplan
In-Reply-To: <4D497437.20105@ewetel.de>
References: <4D3EF036.80701@ewetel.de>	<AANLkTik8jrN_hzom=VEQZfx39YqMgZ15OE8kzZWqJ8iO@mail.gmail.com>	<4D3FD1D9.2@ewetel.de>
	<4D42DC25.4090407@ewetel.de> <4D497437.20105@ewetel.de>
Message-ID: <4D653403.9090201@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Michael,

I fixed this. Was my fault. I tried to control a session directly in lua
when lua was directly called as a dialplan from sofia profile. In this
mode FS seems to expects a table (ACTIONS) filled with action to be
executed. If this is not filled, FS hangs up the call after lua script
has returned.

Now I use a minimized XML dialplan which allways calls the lua script
and voila it works as expected.

Sorry for the trouble.

regards
helmut
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.10 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/

iEYEARECAAYFAk1lNAMACgkQ4tZeNddg3dyoWACgh91ocgT7vj/v+JNoApYIX6Yf
aG4AoJMqYTzoTcTH9qxcHiPNfgVlzsZ9
=06AT
-----END PGP SIGNATURE-----


From anthony.minessale at gmail.com  Wed Feb 23 19:26:14 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 23 Feb 2011 10:26:14 -0600
Subject: [Freeswitch-users] Using outbound Event Sockets versus using
 embed language scripts
In-Reply-To: <AANLkTimyEsRcK02wzbKqWfjbXz-YJXHjSCTi6kXawufU@mail.gmail.com>
References: <AANLkTimyEsRcK02wzbKqWfjbXz-YJXHjSCTi6kXawufU@mail.gmail.com>
Message-ID: <AANLkTi=m99i4r3Osud9x=dUv4PG0fE4w03n=JBVWvuNn@mail.gmail.com>

my $e;
my $foobar;

$e = $con->api("uuid_getvar", "$uuid foobar");

if ($e) {
    $foobar = $e->getBody();
}


On Wed, Feb 23, 2011 at 1:39 AM, Herman Griffin
<herman.griffin at gmail.com> wrote:
> Unless I'm complicating things, it seems like a drop to write outbound event
> sockets apps versus writing embedded language apps. However, I'm interested
> in writing event sockets apps because I can see an advantage in being able
> to load balance the app by running you app behind a load balancer. One
> simple thing that I am trying to accomplish is collecting digits and doing
> something useful with string of digits that have been collected.
>
> My first attempt at this is to use the play_and_get_digits dptool, but I
> don't now how to pull data from stored variable so that I can use in the
> script. I slightly modified the freeswitch.git/libs/esl/perl/server2.pl
> script by adding this line:
>
> $con->execute('play_and_get_digits', '2 5 3 7000 #
> ${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav
> foobar \d+' );
>
> The next thing that I'd like to do is grab the value in foobar is use it in
> the perl logic.
>
> Can someone lead me to the next step?
> Does anyone with experience with event socket apps and embedded language
> apps have some useful information about their preferred path?
>
> Thanks,
> Herman
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Wed Feb 23 12:33:27 2011
From: jjj at 3js.de (Johannes Jakob)
Date: Wed, 23 Feb 2011 10:33:27 +0100
Subject: [Freeswitch-users] Problems getting asterisk registered with FS
	sbc
In-Reply-To: <AANLkTikmJscHtGUUZcqY8CD4ER0-UVMprFkQPUsXDO1+@mail.gmail.com>
References: <D4534174-491C-45D2-A3BF-E29EDE85DF2A@3js.de>
	<AANLkTikmJscHtGUUZcqY8CD4ER0-UVMprFkQPUsXDO1+@mail.gmail.com>
Message-ID: <B9B4B93F-CDED-433B-BD89-92077E6BEEFC@3js.de>

Hi Michael,

I'd love to use authentication for this asterisk, but I didn't get it working!
You'll find the testcase attached with secret set, register string updated and the corresponding directory entry, if you are willing to check it.


Thanks to you for the hint with the ACLs! There haven't been any acls added for none of the asterisk boxes, because I didn't specify
      <node type="allow" domain="mysip.net"/>
in acl.conf.xml, now I did and there are now acls for every asterisk's IP address. Thanks Michael!


BUT: Why are the other asterisk boxes, running asterisk >1.8 working? Why have they been able to connect without password, just by adding the cidr parameter and this not even being evaluated because of my stupidity? Why is everything working smoothly for those boxes and not for the "new" asterisk 1.6 with equivalent settings?



Well... can somebody tell me, what settings I need to make an asterisk register cleanly with my freeswitch box and what might be the problem with my current settings?





DEBUG for asterisk 1.6 trying to auth with secret:


Asterisk TRUNK:
host=sbc1.mysip.net
username=748732
fromuser=748732
secret=Idsd67Hsa
fromdomain=mysip.net
type=peer
t38pt_udptl=yes,redundancy,maxdatagram=400
directmedia=no

Register String: 748732 at mysip.net:UB9aizimo9 at sbc1.mysip.net/748732

FreeSWITCH User entry in directory

<include>
  <user id="748732" cidr="10.16.153.163/32">
    <params>
      <param name="password" value="Idsd67Hsa"/>
<!--
      <param name="apply-inbound-acl" value="domains"/>
      <param name="apply-register-acl" value="domains"/>
-->
      <param name="t38-passthru" value="true"/>
    </params>
    <variables>
      <variable name="client_asserted_identity" value="03023671836"/>
      <variable name="accountcode" value="748732"/>
      <variable name="contract" value="2253"/>
      <variable name="nibble_rate" value="1"/>
      <variable name="nibble_account" value="2253"/>

      <variable name="proxy_media" value="true"/>
      <variable name="bypass_media" value="false"/>
      <variable name="user_context" value="mysip.net"/>
      <variable name="default_language" value="de"/>
      <variable name="language" value="de"/>
    </variables>
  </user>
</include>


2011-02-23 09:53:59.948696 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [748732 at mysip.net] from ip 10.16.153.163
2011-02-23 09:53:59.956363 [WARNING] sofia_reg.c:1204 SIP auth failure (REGISTER) on sofia profile 'internal' for [748732 at mysip.net] from ip 10.16.153.163


09:52:41.577316 IP 10.16.153.163.5060 > 10.16.133.66.5060: UDP, length: 419
E`...f..>...^...^..B......[@REGISTER sip:mysip.net SIP/2.0
Via: SIP/2.0/UDP 10.16.153.163:5060;branch=z9hG4bK281763ac;rport
Max-Forwards: 70
From: <sip:748732 at mysip.net>;tag=as3d3f47c7
To: <sip:748732 at mysip.net>
Call-ID: 312825e32715888c5626ef57287dcf4d at 10.16.153.163
CSeq: 102 REGISTER
User-Agent: Asterisk PBX 1.6.0.22-samy-r60
Expires: 1800
Contact: <sip:748732 at 10.16.153.163>
Event: registration
Content-Length: 0

09:52:41.587730 IP 10.16.133.66.5060 > 10.16.153.163.5060: UDP, length: 657
E....x..?..m^..B^...........SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.16.153.163:5060;branch=z9hG4bK281763ac;rport=5060
From: <sip:748732 at mysip.net>;tag=as3d3f47c7
To: <sip:748732 at mysip.net>;tag=KQS7yprQrS8Kr
Call-ID: 312825e32715888c5626ef57287dcf4d at 10.16.153.163
CSeq: 102 REGISTER
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
WWW-Authenticate: Digest realm="mysip.net", nonce="25994bae-8015-421c-99c3-94a796e31fd2", algorithm=MD5, qop="auth"
Content-Length: 0



09:52:41.588819 IP 10.16.153.163.5060 > 10.16.133.66.5060: UDP, length: 672
E`...g..>...^...^..B........REGISTER sip:mysip.net SIP/2.0
Via: SIP/2.0/UDP 10.16.153.163:5060;branch=z9hG4bK00c12e44;rport
Max-Forwards: 70
From: <sip:748732 at mysip.net>;tag=as705bdca9
To: <sip:748732 at mysip.net>
Call-ID: 312825e32715888c5626ef57287dcf4d at 10.16.153.163
CSeq: 103 REGISTER
User-Agent: Asterisk PBX 1.6.0.22-samy-r60
Authorization: Digest username="748732 at mysip.net", realm="mysip.net", algorithm=MD5, uri="sip:mysip.net", nonce="25994bae-8015-421c-99c3-94a796e31fd2", response="15d31712a3e0ff91e90b1ee5185ea166", qop=auth, cnonce="0fb9367b", nc=00000001
Expires: 1800
Contact: <sip:748732 at 10.16.153.163>
Event: registration
Content-Length: 0


09:52:41.595116 IP 10.16.133.66.5060 > 10.16.153.163.5060: UDP, length: 532
E..0.y..?...^..B^.........  .SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.16.153.163:5060;branch=z9hG4bK00c12e44;rport=5060
From: <sip:748732 at mysip.net>;tag=as705bdca9
To: <sip:748732 at mysip.net>;tag=m0j00H9tN2y6K
Call-ID: 312825e32715888c5626ef57287dcf4d at 10.16.153.163
CSeq: 103 REGISTER
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-Length: 0







Registration of an asterisk 1.8 that works:




10:26:35.024214 IP 10.16.139.29.5060 > 10.16.133.66.5060: SIP, length: 389
E`......?...^...^..B......
.REGISTER sip:mysip.net SIP/2.0
Via: SIP/2.0/UDP 10.16.139.29:5060;branch=z9hG4bK52d214d2
Max-Forwards: 70
From: <sip:742432-2 at mysip.net>;tag=as08f7bdb5
To: <sip:742432-2 at mysip.net>
Call-ID: 68e6f0e5360a42a56cce41700db4f1d6 at 10.16.139.29
CSeq: 102 REGISTER
User-Agent: FPBX-2.8.1(1.8.2.3)
Expires: 3600
Contact: <sip:742432-2 at 10.16.139.29:5060>
Content-Length: 0


10:26:35.040801 IP 10.16.133.66.5060 > 10.16.139.29.5060: SIP, length: 648
E...<... at .nm^..B^..........uSIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.16.139.29:5060;branch=z9hG4bK52d214d2
From: <sip:742432-2 at mysip.net>;tag=as08f7bdb5
To: <sip:742432-2 at mysip.net>;tag=ytFrj0HyrUX1Q
Call-ID: 68e6f0e5360a42a56cce41700db4f1d6 at 10.16.139.29
CSeq: 102 REGISTER
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
WWW-Authenticate: Digest realm="mysip.net", nonce="75c922c2-1bcf-4ae6-95e6-094f38aacc4a", algorithm=MD5, qop="auth"
Content-Length: 0


10:26:35.045018 IP 10.16.139.29.5060 > 10.16.133.66.5060: SIP, length: 629
E`......?...^...^..B.....}.oREGISTER sip:mysip.net SIP/2.0
Via: SIP/2.0/UDP 10.16.139.29:5060;branch=z9hG4bK03be27e5
Max-Forwards: 70
From: <sip:742432-2 at mysip.net>;tag=as48a1a645
To: <sip:742432-2 at mysip.net>
Call-ID: 68e6f0e5360a42a56cce41700db4f1d6 at 10.16.139.29
CSeq: 103 REGISTER
User-Agent: FPBX-2.8.1(1.8.2.3)
Authorization: Digest username="742432-2", realm="mysip.net", algorithm=MD5, uri="sip:mysip.net", nonce="75c922c2-1bcf-4ae6-95e6-094f38aacc4a", response="bae2f45c0f2b47cd8e3a3cd13d9b60b6", qop=auth, cnonce="3cf8c9c5", nc=00000001
Expires: 3600
Contact: <sip:742432-2 at 10.16.139.29:5060>
Content-Length: 0


10:26:35.076972 IP 10.16.133.66.5060 > 10.16.139.29.5060: SIP, length: 609
E..}< .. at .n.^..B^........i.NSIP/2.0 200 OK
Via: SIP/2.0/UDP 10.16.139.29:5060;branch=z9hG4bK03be27e5
From: <sip:742432-2 at mysip.net>;tag=as48a1a645
To: <sip:742432-2 at mysip.net>;tag=Z38gmU21N4KmK
Call-ID: 68e6f0e5360a42a56cce41700db4f1d6 at 10.16.139.29
CSeq: 103 REGISTER
Contact: <sip:742432-2 at 10.16.139.29:5060>;expires=3600
Date: Wed, 23 Feb 2011 09:26:35 GMT 
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-Length: 0

2011-02-23 10:26:35.040164 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [742432-2 at mysip.net] from ip 10.16.139.29


does that mean, I'm just authenticating with an empty password? ;)







On 23.02.2011, at 03:13, Michael Collins wrote:

> I don't believe the ACL works for registrations, only for phone calls. You'll still need to auth for the registration part. For the ACL, though, you can do "reloadacl" and confirm that your CIDR is getting added. When you send calls from TB to FS they should be let in via the ACL without an auth challenge.
> 
> -MC
> 
> On Tue, Feb 22, 2011 at 11:30 AM, Johannes Jakob <jjj at 3js.de> wrote:
> Fellow FreeSWITCH Admins,
> 
> I'm having a hard time, getting a Trixbox 2.8 box to register with our FreeSWITCH SBCs.
> 
> The FreeSWITCHes are running FreeSWITCH-mod_sofia/1.0.head-git-7847289, the asterisk on the trixbox is Asterisk 1.6.0.22-samy-r60.
> 
> 
> The user's directory entry looks like this:
> 
> 
> <include>
>  <user id="748732" cidr="10.16.153.163/32">
>    <params>
> <!--
>      <param name="password" value="Idsd67Hsa"/>
> -->
>      <param name="t38-passthru" value="true"/>
>    </params>
>    <variables>
>      <variable name="client_asserted_identity" value="03023671836"/>
>      <variable name="accountcode" value="748732"/>
>      <variable name="contract" value="2253"/>
>      <variable name="nibble_rate" value="1"/>
>      <variable name="nibble_account" value="2253"/>
> 
>      <variable name="proxy_media" value="true"/>
>      <variable name="bypass_media" value="false"/>
>      <variable name="user_context" value="mysip.net"/>
>      <variable name="default_language" value="de"/>
>      <variable name="language" value="de"/>
>    </variables>
>  </user>
> </include>
> 
> 
> Asterisk's register string: 748732 at mysip.net@sbc1.mysip.net/748732
> 
> 
> I'm getting the "normal" FS errors for wrong credentials:
> 
> 2011-02-22 18:03:57.484939 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [748732 at mysip.net] from ip 10.16.153.163
> 2011-02-22 18:03:57.491471 [WARNING] sofia_reg.c:1204 SIP auth failure (REGISTER) on sofia profile 'internal' for [748732 at mysip.net] from ip 10.16.153.163
> 
> 
> but why am I getting these? I specified the right address in the cidr statement! Why is it even bothering with anything else but the right user at domain and IP-address?
> 
> 
> There are some other asterisk boxes (> 1.8.2) registering to this SBC with equal settings just fine, what's wrong with this little trixbox system? ;)
> 
> 
> 
> Of course I did get you some SIP traces as well:
> 
> 
> 18:00:37.063410 IP 10.16.153.163.5060 > 10.16.133.66.5060: UDP, length: 419
> E`..f...>.7.^...^..B.......-REGISTER sip:mysip.net SIP/2.0
> Via: SIP/2.0/UDP 10.16.153.163:5060;branch=z9hG4bK3e70680b;rport
> Max-Forwards: 70
> From: <sip:748732 at mysip.net>;tag=as77c8852d
> To: <sip:748732 at mysip.net>
> Call-ID: 53d04cc277cfe60301bddb6d79033420 at 10.16.153.163
> CSeq: 102 REGISTER
> User-Agent: Asterisk PBX 1.6.0.22-samy-r60
> Expires: 1800
> Contact: <sip:748732 at 10.16.153.163>
> Event: registration
> Content-Length: 0
> 
> 
> 
> 18:00:37.074085 IP 10.16.133.66.5060 > 10.16.153.163.5060: UDP, length: 657
> E...F...?.Vc^..B^...........SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 10.16.153.163:5060;branch=z9hG4bK3e70680b;rport=5060
> From: <sip:748732 at mysip.net>;tag=as77c8852d
> To: <sip:748732 at mysip.net>;tag=5jD9Qcg3N9S6p
> Call-ID: 53d04cc277cfe60301bddb6d79033420 at 10.16.153.163
> CSeq: 102 REGISTER
> 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
> WWW-Authenticate: Digest realm="mysip.net", nonce="ce2bccbf-a27b-43c8-b7b0-a89ab429d8a7", algorithm=MD5, qop="auth"
> Content-Length: 0
> 
> 
> 
> 18:00:37.074969 IP 10.16.153.163.5060 > 10.16.133.66.5060: UDP, length: 672
> E`..f...>.6.^...^..B........REGISTER sip:mysip.net SIP/2.0
> Via: SIP/2.0/UDP 10.16.153.163:5060;branch=z9hG4bK30df5010;rport
> Max-Forwards: 70
> From: <sip:748732 at mysip.net>;tag=as03431ba4
> To: <sip:748732 at mysip.net>
> Call-ID: 53d04cc277cfe60301bddb6d79033420 at 10.16.153.163
> CSeq: 103 REGISTER
> User-Agent: Asterisk PBX 1.6.0.22-samy-r60
> Authorization: Digest username="748732 at mysip.net", realm="mysip.net", algorithm=MD5, uri="sip:mysip.net", nonce="ce2bccbf-a27b-43c8-b7b0-a89ab429d8a7", response="133a0ba843fe9f5afba67d1377fa8c11", qop=auth, cnonce="119cf18c", nc=00000001
> Expires: 1800
> Contact: <sip:748732 at 10.16.153.163>
> Event: registration
> Content-Length: 0
> 
> 
> 18:00:37.081517 IP 10.16.133.66.5060 > 10.16.153.163.5060: UDP, length: 532
> E..0F...?.V.^..B^.........1.SIP/2.0 403 Forbidden
> Via: SIP/2.0/UDP 10.16.153.163:5060;branch=z9hG4bK30df5010;rport=5060
> From: <sip:748732 at mysip.net>;tag=as03431ba4
> To: <sip:748732 at mysip.net>;tag=6U61S706jjgSj
> Call-ID: 53d04cc277cfe60301bddb6d79033420 at 10.16.153.163
> CSeq: 103 REGISTER
> 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-Length: 0
> 
> 
> Can somebody point me in the right direction?
> 
> 
> Thanks and best regards,
> 
>  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



From misi at niif.hu  Wed Feb 23 16:14:11 2011
From: misi at niif.hu (=?ISO-8859-2?Q?M=C9SZ=C1ROS_Mih=E1ly?=)
Date: Wed, 23 Feb 2011 14:14:11 +0100
Subject: [Freeswitch-users] Freeswtich as a media proxy between ipv4<=>ipv6
 using Polycom HDX8006 SIP UA-s
Message-ID: <4D650823.5050305@niif.hu>

Hi,

   1. 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)
   2. 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?
   3. 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
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From anthony.minessale at gmail.com  Wed Feb 23 19:55:47 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 23 Feb 2011 10:55:47 -0600
Subject: [Freeswitch-users] Using 16 KHz sounds
In-Reply-To: <AANLkTi=YkDwu5ANjYFFWGDR2pmuscsJUDcK09QL92Kqo@mail.gmail.com>
References: <AANLkTim5CsCL4unpWh1OvH13Xx4HrKX2KF7Jg6UQYAZR@mail.gmail.com>
	<AANLkTin9tYDVJC0KUncV0h=g0+u=833QcRtp49OfU2XC@mail.gmail.com>
	<AANLkTi=YkDwu5ANjYFFWGDR2pmuscsJUDcK09QL92Kqo@mail.gmail.com>
Message-ID: <AANLkTi=nUC=XEtb7rod7TjHwfaXwbK=c3=tz4MVrxrHz@mail.gmail.com>

Depends, are you using a sipura? if so, try it, the setting is on the
web ui of the phone/device not in FS.


On Tue, Feb 22, 2011 at 12:25 PM, Malay Thakershi <mthakershi at gmail.com> wrote:
> I found I am already using 16 KHz profile.?.SetTtsParameters("cepstral",
> "Allison-16kHz");
> I read this under FS wiki on Cepstral under 'Gotchas':
> -------------
> Using a 16khz voice and 0.03 for RTP Packet Size (Sipura Setting), it will
> sound horrible. Workaround: modify RTP Packet Size to 0.02 in Sipura config,
> under Advanced/SIP section.
> -------------
> Do you think that is my problem? Is this to be done in FS configuration?
> Malay
> On Mon, Feb 21, 2011 at 2:35 PM, Michael Collins <msc at freeswitch.org> wrote:
>>
>> 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
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb 23 20:08:50 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 23 Feb 2011 11:08:50 -0600
Subject: [Freeswitch-users] risky hangup
In-Reply-To: <B19CF4E8F63F444EA70EB8D4F8A0505A@e1705>
References: <B19CF4E8F63F444EA70EB8D4F8A0505A@e1705>
Message-ID: <AANLkTikJskMjnoUNr4vTpYxY7LSY68JsaYWDzEosfOfG@mail.gmail.com>

this is like the 12th email and the exact issue is hard to follow.

Are you saying it plays a tts like "you are out of money goodbye" and
you don't always hear it all?
You probably have a delay?
Try adding like sleep 3 seconds after you play the tts.



On Tue, Feb 22, 2011 at 10:02 PM, Madovsky <infos at madovsky.org> wrote:
> ok apparently I was wrong...
> I restarted the machine and the issue is still there :((
>
> ----- Original Message -----
> From: "Madovsky" <infos at madovsky.org>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, February 22, 2011 10:26 PM
> Subject: Re: [Freeswitch-users] risky hangup
>
>
>> yes
>>
>> ----- Original Message -----
>> From: "mazilo" <Nabble at slickdeals.endjunk.com>
>> To: <freeswitch-users at lists.freeswitch.org>
>> Sent: Tuesday, February 22, 2011 10:16 PM
>> Subject: Re: [Freeswitch-users] risky hangup
>>
>>
>>>
>>>
>>> Madovsky wrote:
>>>>
>>>> apparently I found the problem,
>>>> there were 2 instances of FS running...
>>> Hmm ..., what exactly did you mean by instances? I don't suppose you
>>> meant
>>> instances = processes, right?
>>>
>>> -----
>>> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
>>> --
>>> View this message in context:
>>> http://freeswitch-users.2379917.n2.nabble.com/risky-hangup-tp6054481p6055032.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


From mthakershi at gmail.com  Wed Feb 23 20:31:10 2011
From: mthakershi at gmail.com (Malay Thakershi)
Date: Wed, 23 Feb 2011 11:31:10 -0600
Subject: [Freeswitch-users] Using 16 KHz sounds
In-Reply-To: <AANLkTi=nUC=XEtb7rod7TjHwfaXwbK=c3=tz4MVrxrHz@mail.gmail.com>
References: <AANLkTim5CsCL4unpWh1OvH13Xx4HrKX2KF7Jg6UQYAZR@mail.gmail.com>
	<AANLkTin9tYDVJC0KUncV0h=g0+u=833QcRtp49OfU2XC@mail.gmail.com>
	<AANLkTi=YkDwu5ANjYFFWGDR2pmuscsJUDcK09QL92Kqo@mail.gmail.com>
	<AANLkTi=nUC=XEtb7rod7TjHwfaXwbK=c3=tz4MVrxrHz@mail.gmail.com>
Message-ID: <AANLkTim7C9ATOtyEehBVXub-DSrUXQsRr25fKDDHdsaq@mail.gmail.com>

I don't use Sipura. I use FS to make / receive calls from mobile phones /
regular land line phones.

Unlike what I said in my previous email, I am still using Allison-8kHz
voice. Somehow in managed code I had Allison-16kHz specified.

I created three WAV files using Cepstral SWIFT command with 8000, 16000,
22000 Hz.  When I play each file, the later two give me message at the FS
console "Sample rates don't match".

Is there a setting where I can ask FS to sample at a higher rate that would
help me with sound quality issues? Is having a good sound card on the server
a good practice?

Thank you for replies.

Malay


On Wed, Feb 23, 2011 at 10:55 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Depends, are you using a sipura? if so, try it, the setting is on the
> web ui of the phone/device not in FS.
>
>
> On Tue, Feb 22, 2011 at 12:25 PM, Malay Thakershi <mthakershi at gmail.com>
> wrote:
> > I found I am already using 16 KHz profile. .SetTtsParameters("cepstral",
> > "Allison-16kHz");
> > I read this under FS wiki on Cepstral under 'Gotchas':
> > -------------
> > Using a 16khz voice and 0.03 for RTP Packet Size (Sipura Setting), it
> will
> > sound horrible. Workaround: modify RTP Packet Size to 0.02 in Sipura
> config,
> > under Advanced/SIP section.
> > -------------
> > Do you think that is my problem? Is this to be done in FS configuration?
> > Malay
> > On Mon, Feb 21, 2011 at 2:35 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> >>
> >> 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
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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  Wed Feb 23 20:32:35 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 23 Feb 2011 12:32:35 -0500
Subject: [Freeswitch-users] risky hangup
References: <B19CF4E8F63F444EA70EB8D4F8A0505A@e1705>
	<AANLkTikJskMjnoUNr4vTpYxY7LSY68JsaYWDzEosfOfG@mail.gmail.com>
Message-ID: <6B6CF3ECE1FC417B86E838CC9C53F8D1@e1705>

I have this in the extension :

                <condition break="always">
                        <action application="answer"/>
                        <action application="sleep" data="1000"/>
                        <action application="playback" 
data="$${sounds_dir}/nibbleBill/no_more_funds.wav"/>
                        <action application="hangup" 
data="OUTGOING_CALL_BARRED"/>
                </condition>

the hangup doesn't respect the end of the wav file.
it can do it at the start or end or middle of while the playback is playing.
I noticed that since last git and yesterday


----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, February 23, 2011 12:08 PM
Subject: Re: [Freeswitch-users] risky hangup


> this is like the 12th email and the exact issue is hard to follow.
>
> Are you saying it plays a tts like "you are out of money goodbye" and
> you don't always hear it all?
> You probably have a delay?
> Try adding like sleep 3 seconds after you play the tts.
>
>
>
> On Tue, Feb 22, 2011 at 10:02 PM, Madovsky <infos at madovsky.org> wrote:
>> ok apparently I was wrong...
>> I restarted the machine and the issue is still there :((
>>
>> ----- Original Message -----
>> From: "Madovsky" <infos at madovsky.org>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Tuesday, February 22, 2011 10:26 PM
>> Subject: Re: [Freeswitch-users] risky hangup
>>
>>
>>> yes
>>>
>>> ----- Original Message -----
>>> From: "mazilo" <Nabble at slickdeals.endjunk.com>
>>> To: <freeswitch-users at lists.freeswitch.org>
>>> Sent: Tuesday, February 22, 2011 10:16 PM
>>> Subject: Re: [Freeswitch-users] risky hangup
>>>
>>>
>>>>
>>>>
>>>> Madovsky wrote:
>>>>>
>>>>> apparently I found the problem,
>>>>> there were 2 instances of FS running...
>>>> Hmm ..., what exactly did you mean by instances? I don't suppose you
>>>> meant
>>>> instances = processes, right?
>>>>
>>>> -----
>>>> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
>>>> --
>>>> View this message in context:
>>>> http://freeswitch-users.2379917.n2.nabble.com/risky-hangup-tp6054481p6055032.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 



From cmrienzo at gmail.com  Wed Feb 23 20:55:04 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Wed, 23 Feb 2011 12:55:04 -0500
Subject: [Freeswitch-users] OpenSER interface to FreeSWITCH
In-Reply-To: <COL111-W442B2A5D707F79AF2A8338ACDB0@phx.gbl>
References: <COL111-W442B2A5D707F79AF2A8338ACDB0@phx.gbl>
Message-ID: <AANLkTikCH+6L2uC8LFfcJvbihkR2if46e8n9bct+inbB@mail.gmail.com>

I implemented something sort of like RFC 4722 (MSCML), except I scrapped the
SIP INFO stuff and just used event socket.

1. Application server receives incoming phone call
2. Application server calls FS to get IVR session.  FS handles all media
processing on incoming phone call
3. Application server executes IVR script, sending commands to FS over event
socket.

Think over your requirements carefully and use what FS already offers.  Your
requirements may be met with a simpler solution.


On Tue, Feb 22, 2011 at 9:35 PM, Jason b.a <k-b-24 at live.com> wrote:

>  yeh sure i have a SIP endpoints , can the application do the sip handling
> and have a socket interface with freeswitch in the same time without using
> OpenSER.
> in this case i need Sip servelet plugin in my application , is it possible.
> I went through the outbound event socket , seems to be helpful thx , also i
> saw the options for using java to connect to freeswitch, i think using ESL ,
> i can build a java IVR 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
>
>
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From msc at freeswitch.org  Wed Feb 23 20:54:22 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 23 Feb 2011 09:54:22 -0800
Subject: [Freeswitch-users] Using 16 KHz sounds
In-Reply-To: <AANLkTim7C9ATOtyEehBVXub-DSrUXQsRr25fKDDHdsaq@mail.gmail.com>
References: <AANLkTim5CsCL4unpWh1OvH13Xx4HrKX2KF7Jg6UQYAZR@mail.gmail.com>
	<AANLkTin9tYDVJC0KUncV0h=g0+u=833QcRtp49OfU2XC@mail.gmail.com>
	<AANLkTi=YkDwu5ANjYFFWGDR2pmuscsJUDcK09QL92Kqo@mail.gmail.com>
	<AANLkTi=nUC=XEtb7rod7TjHwfaXwbK=c3=tz4MVrxrHz@mail.gmail.com>
	<AANLkTim7C9ATOtyEehBVXub-DSrUXQsRr25fKDDHdsaq@mail.gmail.com>
Message-ID: <AANLkTinKGfnq+escY62cAdWvM4vaut8La5e5EtzpoqGo@mail.gmail.com>

It sounds very much like you have standard 8k calls. Increasing the sampling
rate won't help since FS would have to downsample to 8k on the call leg
anyway.

It's time to go back to the original issue. You are having sound quality
issues, correct? It's time to roll up your sleeves and do some detective
work:

What kind of network are you running? What routers, switches, NAT devices,
and other computers are using the network? What kind of system is FS running
on? Any virtualization being used? What OS?

Are you running the latest git of FS?

What SIP clients have you tried? Can you reproduce the sound quality issues
on all of your SIP clients? Can you reproduce with different SIP clients on
a different computer? Do you have a hard phone and do the symptoms persist
there?

Are you having sound quality issues in one direction or both directions?
Have you done a tcpdump of the traffic and analyzed in wireshark?

Those are all questions worth pursuing. The idea is to narrow the symptoms
down as much as possible. I know it's not fun, but then again, this is
telephony. :)

-MC

On Wed, Feb 23, 2011 at 9:31 AM, Malay Thakershi <mthakershi at gmail.com>wrote:

> I don't use Sipura. I use FS to make / receive calls from mobile phones /
> regular land line phones.
>
> Unlike what I said in my previous email, I am still using Allison-8kHz
> voice. Somehow in managed code I had Allison-16kHz specified.
>
> I created three WAV files using Cepstral SWIFT command with 8000, 16000,
> 22000 Hz.  When I play each file, the later two give me message at the FS
> console "Sample rates don't match".
>
> Is there a setting where I can ask FS to sample at a higher rate that would
> help me with sound quality issues? Is having a good sound card on the server
> a good practice?
>
> Thank you for replies.
>
> Malay
>
>
> On Wed, Feb 23, 2011 at 10:55 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Depends, are you using a sipura? if so, try it, the setting is on the
>> web ui of the phone/device not in FS.
>>
>>
>> On Tue, Feb 22, 2011 at 12:25 PM, Malay Thakershi <mthakershi at gmail.com>
>> wrote:
>> > I found I am already using 16 KHz profile. .SetTtsParameters("cepstral",
>> > "Allison-16kHz");
>> > I read this under FS wiki on Cepstral under 'Gotchas':
>> > -------------
>> > Using a 16khz voice and 0.03 for RTP Packet Size (Sipura Setting), it
>> will
>> > sound horrible. Workaround: modify RTP Packet Size to 0.02 in Sipura
>> config,
>> > under Advanced/SIP section.
>> > -------------
>> > Do you think that is my problem? Is this to be done in FS configuration?
>> > Malay
>> > On Mon, Feb 21, 2011 at 2:35 PM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> >>
>> >> 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
>> >>>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at 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 santiagosoares at gmail.com  Wed Feb 23 21:01:38 2011
From: santiagosoares at gmail.com (Santiago Soares)
Date: Wed, 23 Feb 2011 15:01:38 -0300
Subject: [Freeswitch-users] Second transfer
Message-ID: <AANLkTi=pB-Rt7j1cvNGgSs2s3F4h6jse1nTqTqRr8kKJ@mail.gmail.com>

Hello,

I'm using att_xfer to make call transfer, like that:
A calls B
B calls C
B hangup and A is bridged do C

The thing is, the way I am trying to do it, C is able to transfer the call
again.
I don't want that C be able to transfer the call again.
The call should be transfered only once.
Is it possible?

This is my dialplan:

<include>

   <extension name="test_extension">
      <condition field="destination_number" expression="^numberB$">
         <action application="set" data="bind_meta_key=*"/>
         <action application="bind_meta_app" data="3 b s
execute_extension::transferencia XML public"/>
         <action application="bridge" data="sofia/gateway/test/loginB"/>
      </condition>
   </extension>

   <extension name="transferencia">
      <condition field="destination_number" expression="^transferencia$">
         <action application="read" data="3 3 base256/16000/sound.wav
attxfer_callthis 30000 #"/>
         <action application="execute_extension" data="transferencia2"/>
      </condition>
   </extension>

   <extension name="transferencia2">
      <condition field="${attxfer_callthis}" expression="^numberC$">
         <action application="att_xfer" data="sofia/gateway/test/loginC"/>
      </condition>
      <condition field="${attxfer_callthis}" expression="^numberD$">
         <action application="att_xfer" data="sofia/gateway/test/loginD"/>
      </condition>
   </extension>

</include>


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

Hey folks, feel free to hop on!

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

Talk to you shortly,
MC
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From anthony.minessale at gmail.com  Wed Feb 23 21:05:11 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 23 Feb 2011 12:05:11 -0600
Subject: [Freeswitch-users] risky hangup
In-Reply-To: <6B6CF3ECE1FC417B86E838CC9C53F8D1@e1705>
References: <B19CF4E8F63F444EA70EB8D4F8A0505A@e1705>
	<AANLkTikJskMjnoUNr4vTpYxY7LSY68JsaYWDzEosfOfG@mail.gmail.com>
	<6B6CF3ECE1FC417B86E838CC9C53F8D1@e1705>
Message-ID: <AANLkTinq9PDS5T8wpAOfzZN4HDZmyVKB70dgf6FrKUZf@mail.gmail.com>

Are you saying this is something that is suddenly happening and was not before?
Can you get a pcap of it?


On Wed, Feb 23, 2011 at 11:32 AM, Madovsky <infos at madovsky.org> wrote:
> I have this in the extension :
>
> ? ? ? ? ? ? ? ?<condition break="always">
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="answer"/>
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="sleep" data="1000"/>
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="playback"
> data="$${sounds_dir}/nibbleBill/no_more_funds.wav"/>
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="hangup"
> data="OUTGOING_CALL_BARRED"/>
> ? ? ? ? ? ? ? ?</condition>
>
> the hangup doesn't respect the end of the wav file.
> it can do it at the start or end or middle of while the playback is playing.
> I noticed that since last git and yesterday
>
>
> ----- Original Message -----
> From: "Anthony Minessale" <anthony.minessale at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, February 23, 2011 12:08 PM
> Subject: Re: [Freeswitch-users] risky hangup
>
>
>> this is like the 12th email and the exact issue is hard to follow.
>>
>> Are you saying it plays a tts like "you are out of money goodbye" and
>> you don't always hear it all?
>> You probably have a delay?
>> Try adding like sleep 3 seconds after you play the tts.
>>
>>
>>
>> On Tue, Feb 22, 2011 at 10:02 PM, Madovsky <infos at madovsky.org> wrote:
>>> ok apparently I was wrong...
>>> I restarted the machine and the issue is still there :((
>>>
>>> ----- Original Message -----
>>> From: "Madovsky" <infos at madovsky.org>
>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>> Sent: Tuesday, February 22, 2011 10:26 PM
>>> Subject: Re: [Freeswitch-users] risky hangup
>>>
>>>
>>>> yes
>>>>
>>>> ----- Original Message -----
>>>> From: "mazilo" <Nabble at slickdeals.endjunk.com>
>>>> To: <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Tuesday, February 22, 2011 10:16 PM
>>>> Subject: Re: [Freeswitch-users] risky hangup
>>>>
>>>>
>>>>>
>>>>>
>>>>> Madovsky wrote:
>>>>>>
>>>>>> apparently I found the problem,
>>>>>> there were 2 instances of FS running...
>>>>> Hmm ..., what exactly did you mean by instances? I don't suppose you
>>>>> meant
>>>>> instances = processes, right?
>>>>>
>>>>> -----
>>>>> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
>>>>> --
>>>>> View this message in context:
>>>>> http://freeswitch-users.2379917.n2.nabble.com/risky-hangup-tp6054481p6055032.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 infos at madovsky.org  Wed Feb 23 21:09:27 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 23 Feb 2011 13:09:27 -0500
Subject: [Freeswitch-users] risky hangup
References: <B19CF4E8F63F444EA70EB8D4F8A0505A@e1705><AANLkTikJskMjnoUNr4vTpYxY7LSY68JsaYWDzEosfOfG@mail.gmail.com><6B6CF3ECE1FC417B86E838CC9C53F8D1@e1705>
	<AANLkTinq9PDS5T8wpAOfzZN4HDZmyVKB70dgf6FrKUZf@mail.gmail.com>
Message-ID: <4B153C5078624B7D8608DC62B8000952@e1705>

yes...
please remind me the pcap command line for linux
and will send to you

----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, February 23, 2011 1:05 PM
Subject: Re: [Freeswitch-users] risky hangup


Are you saying this is something that is suddenly happening and was not 
before?
Can you get a pcap of it?


On Wed, Feb 23, 2011 at 11:32 AM, Madovsky <infos at madovsky.org> wrote:
> I have this in the extension :
>
> <condition break="always">
> <action application="answer"/>
> <action application="sleep" data="1000"/>
> <action application="playback"
> data="$${sounds_dir}/nibbleBill/no_more_funds.wav"/>
> <action application="hangup"
> data="OUTGOING_CALL_BARRED"/>
> </condition>
>
> the hangup doesn't respect the end of the wav file.
> it can do it at the start or end or middle of while the playback is 
> playing.
> I noticed that since last git and yesterday
>
>
> ----- Original Message -----
> From: "Anthony Minessale" <anthony.minessale at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, February 23, 2011 12:08 PM
> Subject: Re: [Freeswitch-users] risky hangup
>
>
>> this is like the 12th email and the exact issue is hard to follow.
>>
>> Are you saying it plays a tts like "you are out of money goodbye" and
>> you don't always hear it all?
>> You probably have a delay?
>> Try adding like sleep 3 seconds after you play the tts.
>>
>>
>>
>> On Tue, Feb 22, 2011 at 10:02 PM, Madovsky <infos at madovsky.org> wrote:
>>> ok apparently I was wrong...
>>> I restarted the machine and the issue is still there :((
>>>
>>> ----- Original Message -----
>>> From: "Madovsky" <infos at madovsky.org>
>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>> Sent: Tuesday, February 22, 2011 10:26 PM
>>> Subject: Re: [Freeswitch-users] risky hangup
>>>
>>>
>>>> yes
>>>>
>>>> ----- Original Message -----
>>>> From: "mazilo" <Nabble at slickdeals.endjunk.com>
>>>> To: <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Tuesday, February 22, 2011 10:16 PM
>>>> Subject: Re: [Freeswitch-users] risky hangup
>>>>
>>>>
>>>>>
>>>>>
>>>>> Madovsky wrote:
>>>>>>
>>>>>> apparently I found the problem,
>>>>>> there were 2 instances of FS running...
>>>>> Hmm ..., what exactly did you mean by instances? I don't suppose you
>>>>> meant
>>>>> instances = processes, right?
>>>>>
>>>>> -----
>>>>> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
>>>>> --
>>>>> View this message in context:
>>>>> http://freeswitch-users.2379917.n2.nabble.com/risky-hangup-tp6054481p6055032.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
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FreeSWITCH Developer Conference
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From infos at madovsky.org  Wed Feb 23 21:59:36 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 23 Feb 2011 13:59:36 -0500
Subject: [Freeswitch-users] risky hangup
Message-ID: <AA88D341A8A14329BB10CC46DB609998@e1705>

Anthony,

I think I found the problem (it's not  a bug)
is FS more sensitive about DTMF packet ?

thanks


----- Original Message ----- 
From: "Madovsky" <infos at madovsky.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, February 23, 2011 1:09 PM
Subject: Re: [Freeswitch-users] risky hangup


> yes...
> please remind me the pcap command line for linux
> and will send to you
>
> ----- Original Message ----- 
> From: "Anthony Minessale" <anthony.minessale at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, February 23, 2011 1:05 PM
> Subject: Re: [Freeswitch-users] risky hangup
>
>
> Are you saying this is something that is suddenly happening and was not 
> before?
> Can you get a pcap of it?
>
>
> On Wed, Feb 23, 2011 at 11:32 AM, Madovsky <infos at madovsky.org> wrote:
>> I have this in the extension :
>>
>> <condition break="always">
>> <action application="answer"/>
>> <action application="sleep" data="1000"/>
>> <action application="playback"
>> data="$${sounds_dir}/nibbleBill/no_more_funds.wav"/>
>> <action application="hangup"
>> data="OUTGOING_CALL_BARRED"/>
>> </condition>
>>
>> the hangup doesn't respect the end of the wav file.
>> it can do it at the start or end or middle of while the playback is 
>> playing.
>> I noticed that since last git and yesterday
>>
>>
>> ----- Original Message -----
>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Wednesday, February 23, 2011 12:08 PM
>> Subject: Re: [Freeswitch-users] risky hangup
>>
>>
>>> this is like the 12th email and the exact issue is hard to follow.
>>>
>>> Are you saying it plays a tts like "you are out of money goodbye" and
>>> you don't always hear it all?
>>> You probably have a delay?
>>> Try adding like sleep 3 seconds after you play the tts.
>>>
>>>
>>>
>>> On Tue, Feb 22, 2011 at 10:02 PM, Madovsky <infos at madovsky.org> wrote:
>>>> ok apparently I was wrong...
>>>> I restarted the machine and the issue is still there :((
>>>>
>>>> ----- Original Message -----
>>>> From: "Madovsky" <infos at madovsky.org>
>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Tuesday, February 22, 2011 10:26 PM
>>>> Subject: Re: [Freeswitch-users] risky hangup
>>>>
>>>>
>>>>> yes
>>>>>
>>>>> ----- Original Message -----
>>>>> From: "mazilo" <Nabble at slickdeals.endjunk.com>
>>>>> To: <freeswitch-users at lists.freeswitch.org>
>>>>> Sent: Tuesday, February 22, 2011 10:16 PM
>>>>> Subject: Re: [Freeswitch-users] risky hangup
>>>>>
>>>>>
>>>>>>
>>>>>>
>>>>>> Madovsky wrote:
>>>>>>>
>>>>>>> apparently I found the problem,
>>>>>>> there were 2 instances of FS running...
>>>>>> Hmm ..., what exactly did you mean by instances? I don't suppose you
>>>>>> meant
>>>>>> instances = processes, right?
>>>>>>
>>>>>> -----
>>>>>> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
>>>>>> --
>>>>>> View this message in context:
>>>>>> http://freeswitch-users.2379917.n2.nabble.com/risky-hangup-tp6054481p6055032.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 


-- 
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.



From Nabble at slickdeals.endjunk.com  Wed Feb 23 22:49:44 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 23 Feb 2011 11:49:44 -0800 (PST)
Subject: [Freeswitch-users] INVITE DoS Prevention
In-Reply-To: <AANLkTi=uN_TAqkA0Hbq27kQ0HVWrOxdwfTMzq6otc07v@mail.gmail.com>
References: <6461E11A-84B5-40AF-96BE-C68F3DF91D8E@5ninesolutions.com>
	<C98764B0.11627%krice@freeswitch.org>
	<A624F826-AF73-40BD-8DCD-10F39551D924@5ninesolutions.com>
	<AANLkTi=+f0GqChj4PeW=XFjQW_mYTtJYPuWEj47gw8r+@mail.gmail.com>
	<1298473808528-6056642.post@n2.nabble.com>
	<AANLkTi=uN_TAqkA0Hbq27kQ0HVWrOxdwfTMzq6otc07v@mail.gmail.com>
Message-ID: <1298490584118-6057618.post@n2.nabble.com>



Steven Ayre wrote:
> 
> That's be within the Sofia stack and it had to acknowledge the INVITE with
> a
> 100 Trying otherwise the INVITE either resends or gives up the same as a
> packet drop. Sleeping would mean keeping the INVITE in memory for longer
> while starting the session before accepting/rejecting it, increasing
> memory
> usage under a DOS attack and therefore making FS fall over faster. It'd
> also
> increase the complexity of the code starting up a session while it puts
> new
> invites aside and schedules them to be processed shortly afterwards. You
> couldn't just do a sleep as it'd probably lock Sofia up. Doesn't seem
> worth
> it.
When in sleep mode, check the new INVITEs against the same host. If so, then
send them to sleep, too. If invites in sleep reaches Y amount, we can flag
the host to ignore further requests and release/junk all the sleeping
INVITEs. A database needs be setup to keep a tab of all rogue hosts.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/INVITE-DoS-Prevention-tp6047615p6057618.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From k-b-24 at live.com  Wed Feb 23 22:57:47 2011
From: k-b-24 at live.com (Jason b.a)
Date: Wed, 23 Feb 2011 19:57:47 +0000
Subject: [Freeswitch-users] OpenSER interface to FreeSWITCH
Message-ID: <COL111-W31211C1CADAECCC546428ACDB0@phx.gbl>


Christopher this is same as my scenario ! but how did u use MSCML to control FS!  		 	   		  
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From kemen04 at gmail.com  Thu Feb 24 01:06:02 2011
From: kemen04 at gmail.com (Travis Kemen)
Date: Wed, 23 Feb 2011 16:06:02 -0600
Subject: [Freeswitch-users] SIP SUBSCRIBE for as-feature-event events
In-Reply-To: <4D5D23AC.7090804@ewetel.de>
References: <4D5D23AC.7090804@ewetel.de>
Message-ID: <AANLkTi=_QvqNyUkjxTNUv+yqb=D9hnfAgGu-g5OzRzju@mail.gmail.com>

Polycom uses the same thing. http://jira.freeswitch.org/browse/FS-2731

Travis

On Thu, Feb 17, 2011 at 7:33 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> Snom is supporting subscriptions to as-feature-event Events.
> Unfortunately FS replys with "Bad Event" to this.
>
> Are there plans to support such subscriptions? Maybe the headers and
> body can be passed to an application? Or the whole thing could be
> forwared to an application server to work on it.
>
> The SIP message looks like this:
>
>
> Session Initiation Protocol
>    Request-Line: SUBSCRIBE sip:2850 at a.a.a.a SIP/2.0
>    Message Header
>        Via: SIP/2.0/UDP b.b.b.b:5060;branch=z9hG4bK-b3v8fu4shagb;rport
>        From: <sip:2850 at a.a.a.a>;tag=dq168o7qj9
>        To: "Helmut Kuper" <sip:2850 at a.a.a.a>
>        Call-ID: 3c26705696f0-6ntjya2y7b3s
>        CSeq: 2 SUBSCRIBE
>        Max-Forwards: 70
>        Contact: <sip:2850 at b.b.b.b:5060>;reg-id=1
>        Event: as-feature-event
>        User-Agent: snom370/8.4.27
>        Expires: 3600
>        Content-Type: application/x-as-feature-event+xml
>        Content-Length: 185
>    Message Body
>        <?xml version="1.0" encoding="ISO-8859-1"?><SetDoNotDisturb
> xmlns="http://www.ecma-international.org/standards/ecma-323/csta/ed3
> "><doNotDisturbOn>true</doNotDisturbOn></SetDoNotDisturb>
>
> Best regards
> Helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.10 (MingW32)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/
>
> iEYEARECAAYFAk1dI6wACgkQ4tZeNddg3dx0LQCdGPIHZWb5tUEaP4SOMl7cWJrE
> f38AoK1NEd9VeOUJ6Ut+XAyZgS6LdcZz
> =JLAn
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From cmrienzo at gmail.com  Thu Feb 24 01:49:10 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Wed, 23 Feb 2011 17:49:10 -0500
Subject: [Freeswitch-users] OpenSER interface to FreeSWITCH
In-Reply-To: <COL111-W31211C1CADAECCC546428ACDB0@phx.gbl>
References: <COL111-W31211C1CADAECCC546428ACDB0@phx.gbl>
Message-ID: <AANLkTintuX+ENxzKcGY680gZ6G8b+Wq36Z9iBdTMO+Q=@mail.gmail.com>

I didn't use MSCML.  I said it was sort of like it.  Basically, I was told
to implement MSCML.  I investigated and decided it was a waste of time to do
that.  So, I used event socket instead to control the calls.  You can
accomplish what you need over event socket with the APPs in mod_dptools.



On Wed, Feb 23, 2011 at 2:57 PM, Jason b.a <k-b-24 at live.com> wrote:

>  Christopher this is same as my scenario ! but how did u use MSCML to
> control FS!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 24 02:20:17 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 23 Feb 2011 15:20:17 -0800
Subject: [Freeswitch-users] Second transfer
In-Reply-To: <AANLkTi=pB-Rt7j1cvNGgSs2s3F4h6jse1nTqTqRr8kKJ@mail.gmail.com>
References: <AANLkTi=pB-Rt7j1cvNGgSs2s3F4h6jse1nTqTqRr8kKJ@mail.gmail.com>
Message-ID: <AANLkTi=fvkp_8e4gXthWtYQTLAyxj+bbUMtdpAhXQamQ@mail.gmail.com>

I think you can make another call to bind_meta_app and disable the *3 prior
to the att_xfer. You could make *3 do something completely different (like
trigger a log entry) and then you could see how many times your users tried
to use *3 when they weren't supposed to. :)

-MC

On Wed, Feb 23, 2011 at 10:01 AM, Santiago Soares
<santiagosoares at gmail.com>wrote:

> Hello,
>
> I'm using att_xfer to make call transfer, like that:
> A calls B
> B calls C
> B hangup and A is bridged do C
>
> The thing is, the way I am trying to do it, C is able to transfer the call
> again.
> I don't want that C be able to transfer the call again.
> The call should be transfered only once.
> Is it possible?
>
> This is my dialplan:
>
> <include>
>
>    <extension name="test_extension">
>       <condition field="destination_number" expression="^numberB$">
>          <action application="set" data="bind_meta_key=*"/>
>          <action application="bind_meta_app" data="3 b s
> execute_extension::transferencia XML public"/>
>          <action application="bridge" data="sofia/gateway/test/loginB"/>
>       </condition>
>    </extension>
>
>    <extension name="transferencia">
>       <condition field="destination_number" expression="^transferencia$">
>          <action application="read" data="3 3 base256/16000/sound.wav
> attxfer_callthis 30000 #"/>
>          <action application="execute_extension" data="transferencia2"/>
>       </condition>
>    </extension>
>
>    <extension name="transferencia2">
>       <condition field="${attxfer_callthis}" expression="^numberC$">
>          <action application="att_xfer" data="sofia/gateway/test/loginC"/>
>       </condition>
>       <condition field="${attxfer_callthis}" expression="^numberD$">
>          <action application="att_xfer" data="sofia/gateway/test/loginD"/>
>       </condition>
>    </extension>
>
> </include>
>
>
> Thank you,
> Santiago Soares
>
>
> _______________________________________________
> FreeSWITCH-users mailing 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 dujinfang at gmail.com  Thu Feb 24 02:56:05 2011
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 24 Feb 2011 07:56:05 +0800
Subject: [Freeswitch-users] outbound-caller-name missing?
In-Reply-To: <AANLkTim3ep_yzfesqzoB7exZVWQJJB7yRNVqjvtqRE9Z@mail.gmail.com>
References: <AANLkTimZnnZYrHmaHVX3dW4KA4kfksEX=zaaokuwLYb-@mail.gmail.com>
	<AANLkTim3ep_yzfesqzoB7exZVWQJJB7yRNVqjvtqRE9Z@mail.gmail.com>
Message-ID: <AANLkTi=puTetTGM4iQCcdem4G=-E6pjYzgc3u+ZO-BZ_@mail.gmail.com>

Thanks and sorry I didn't look that hard.

On Thu, Feb 24, 2011 at 12:20 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> That is normal.
>
> All that stuff in vars.xml is just example config its not code in FS.
> Try git grep outbound_caller_id
>
> You are, and always will be, required to specify the caller_id on 1
> legged outbound calls.
>
>
> On Wed, Feb 23, 2011 at 6:00 AM, Seven Du <dujinfang at gmail.com> wrote:
>> Hi,
>>
>>
>> I tried
>>
>> FreeSWITCH Version 1.0.head (git-f5dafc9 2011-01-21 23-42-30 +0100)
>> FreeSWITCH Version 1.0.head (git-06988e1 2011-02-20 15-10-02 -0600)
>>
>>
>> originate user/1000 &echo got this:
>>
>> ? INVITE sip:tbskvnyi at 192.168.7.2:61901 SIP/2.0
>> ? Via: SIP/2.0/UDP 192.168.7.2;rport;branch=z9hG4bK0USe34K3r46gF
>> ? Max-Forwards: 70
>> ? From: "" <sip:0000000000 at 192.168.7.2>;tag=cZt7ZSBZjQK3H
>> ? Remote-Party-ID:
>> <sip:0000000000 at 192.168.7.2>;party=calling;screen=yes;privacy=off
>>
>>
>> And some versions of Xlite4 fails to respond to this
>>
>> originate {origination_caller_id_name=xxx}user/1000 works.
>>
>> I haven't got time to test git head (02-20 almost head), ?just want to
>> ask here anyone noticed this, let me know if need a jira.
>>
>> btw, vars.xml is untouched:
>>
>> ?<X-PRE-PROCESS cmd="set" data="outbound_caller_name=FreeSWITCH"/>
>> ?<X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/>
>>
>> Thanks.
>>
>> --
>> 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
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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 infos at madovsky.org  Thu Feb 24 06:23:08 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 23 Feb 2011 22:23:08 -0500
Subject: [Freeswitch-users] excute app from CLI with vars on answered channel
Message-ID: <3F3D30233AF74DEFB8977A1B3AD5AEBA@e1705>

I'd like to execute an app while from CLI
on an answered channel (I know the uuid) and pass vars on it like

{exec?} <app with vars and params> <uuid>

or if it's not possible execute an extension from CLI with a  uuid 


Thanks
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From herman.griffin at gmail.com  Thu Feb 24 06:49:37 2011
From: herman.griffin at gmail.com (Herman Griffin)
Date: Wed, 23 Feb 2011 19:49:37 -0800
Subject: [Freeswitch-users] Using outbound Event Sockets versus using
 embed language scripts
In-Reply-To: <AANLkTi=m99i4r3Osud9x=dUv4PG0fE4w03n=JBVWvuNn@mail.gmail.com>
References: <AANLkTimyEsRcK02wzbKqWfjbXz-YJXHjSCTi6kXawufU@mail.gmail.com>
	<AANLkTi=m99i4r3Osud9x=dUv4PG0fE4w03n=JBVWvuNn@mail.gmail.com>
Message-ID: <AANLkTi=vZzEZW2T6FqmKgDcABTt=Tw3NdcYrhzxnn-Wx@mail.gmail.com>

I notice that there are DTMF events that come over the event socket. Is it
best practice to use the play_and_get_digits dialplan app to collect digits
or is collecting the digits using the DTMF events a better way?

On Wed, Feb 23, 2011 at 8:26 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> my $e;
> my $foobar;
>
> $e = $con->api("uuid_getvar", "$uuid foobar");
>
> if ($e) {
>    $foobar = $e->getBody();
> }
>
>
> On Wed, Feb 23, 2011 at 1:39 AM, Herman Griffin
> <herman.griffin at gmail.com> wrote:
> > Unless I'm complicating things, it seems like a drop to write outbound
> event
> > sockets apps versus writing embedded language apps. However, I'm
> interested
> > in writing event sockets apps because I can see an advantage in being
> able
> > to load balance the app by running you app behind a load balancer. One
> > simple thing that I am trying to accomplish is collecting digits and
> doing
> > something useful with string of digits that have been collected.
> >
> > My first attempt at this is to use the play_and_get_digits dptool, but I
> > don't now how to pull data from stored variable so that I can use in the
> > script. I slightly modified the freeswitch.git/libs/esl/perl/server2.pl
> > script by adding this line:
> >
> > $con->execute('play_and_get_digits', '2 5 3 7000 #
> > ${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav
> > foobar \d+' );
> >
> > The next thing that I'd like to do is grab the value in foobar is use it
> in
> > the perl logic.
> >
> > Can someone lead me to the next step?
> > Does anyone with experience with event socket apps and embedded language
> > apps have some useful information about their preferred path?
> >
> > Thanks,
> > Herman
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 delorenzodesign at gmail.com  Thu Feb 24 06:50:11 2011
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Wed, 23 Feb 2011 22:50:11 -0500
Subject: [Freeswitch-users] Message Playback Stops
In-Reply-To: <AANLkTikvwZR5t8PaPnHVwNMqDRpipYndc+XbSzYzvjRH@mail.gmail.com>
References: <AANLkTi=EMQsSkH-SZDq2rUzvh2-7nYK=NpKHAXXEwc-W@mail.gmail.com>
	<AANLkTinpDt+NrcoK8pwWuuwMufut+DTC7GSrnAdWYhJ5@mail.gmail.com>
	<AANLkTi=O2R3oqcSDOUxh3g-mqKJ_E-rgrLZ210a_SeJi@mail.gmail.com>
	<AANLkTikSap1zEH58kVWVzpw_kWoWme26WQ1Y6CB3vnxm@mail.gmail.com>
	<AANLkTikCs4Uh1uuTwut=mvt4m7KLzwC-Y7bRedvrO-Te@mail.gmail.com>
	<AANLkTikDvE_Y93owBZ6_DHz75pNCdrKoocy6ywjjWLcF@mail.gmail.com>
	<AANLkTikvwZR5t8PaPnHVwNMqDRpipYndc+XbSzYzvjRH@mail.gmail.com>
Message-ID: <AANLkTik9tSO5gWA_bUEQKEbdD6YxG6GAvedWtBs21sxZ@mail.gmail.com>

Is there some place where I can get more detailed log information on this?
I thought that streamFile was a blocking operation (the next statement
wouldn't be executed until playback was completed), is that not the case?

Is there a difference in terms of performance with Freeswitch for Lua versus
SpiderMonkey?


Am I misreading the log?
>
>
> *2011-02-21 17:23:09.077014 [NOTICE] switch_cpp.cpp:1181 Playing file:
> /usr/local/freeswitch/*
> *recordings/messages/16c0f890_**c35e33c0_777973.wav*
> 2011-02-21 17:23:09.077014 [DEBUG] switch_ivr_play_say.c:1186 Codec
> Activated L16 at 8000hz 1 channels 20ms
> 2011-02-21 17:23:19.417631 [DEBUG] switch_ivr_play_say.c:1515 done playing
> file
> 2011-02-21 17:23:19.417631 [NOTICE] switch_cpp.cpp:1181 !!!!! Finished
> playing the file !!!!!
> 2011-02-21 17:23:19.417631 [INFO] switch_cpp.cpp:1181 Was VM detected? no
> 2011-02-21 17:23:19.417631 [NOTICE] switch_cpp.cpp:1181 Played the message
> at least once and checked for VM, we should be exiting the loop
>
> The messages like "finished playing" are log commands that are only
> available in the script after it's answered.
>
>
>
>
> On Wed, Feb 23, 2011 at 11:48 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Your logs also don't show anything else like the call being answered
>> or hungup or any other information.
>>
>> you probably need to add {ignore_early_media=true} to your dial string
>> so the message does not play while the call is not answered.
>>
>> if you are calling sofia/internal/foo at bar.com change it to
>> {ignore_early_media=true}sofia/internal/foo at bar.com
>>
>>
>>
>>
>> On Wed, Feb 23, 2011 at 10:39 AM, Michael De Lorenzo
>> <delorenzodesign at gmail.com> wrote:
>> > Maybe failure isn't the correct word.  Here's what happens:
>> >
>> > Call gets placed, recipient phone rings
>> > Call answered
>> > Check for VM
>> > If not VM, play the message back after short delay  <-- the message
>> doesn't
>> > play, the call hangs up (step 5), but the logs indicate that the message
>> > played
>> > Hangup
>> >
>> > As the log shows, it doesn't indicate that the message playback fails.
>> >
>> > On Wed, Feb 23, 2011 at 11:14 AM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> I am asking for description of what happens when it fails, How do you
>> >> know it's failing?
>> >> If it fails it would be logging errors.
>> >>
>> >>
>> >>
>> >>
>> >> On Wed, Feb 23, 2011 at 12:00 AM, Michael De Lorenzo
>> >> <delorenzodesign at gmail.com> wrote:
>> >> > I'm not sure where to check, but there's entries like this in
>> >> > freeswitch.log.2011-02-21-23-40-11.1
>> >> >
>> >> > 2011-02-21 17:22:53.157390 [NOTICE] mod_logfile.c:158 New log
>> started.
>> >> > 2011-02-21 17:22:53.157390 [NOTICE] switch_cpp.cpp:1181 Played the
>> >> > message
>> >> > at least once and checked for VM, we should be exiting the loop.
>> >> > 2011-02-21 17:22:53.157390 [NOTICE] switch_cpp.cpp:1181 *************
>> >> > SETTING SESSION VARIABLES ***************
>> >> > 2011-02-21 17:22:53.157390 [NOTICE] switch_cpp.cpp:1181 Session
>> >> > Variable[profile_id]: 1
>> >> > 2011-02-21 17:22:53.157390 [NOTICE] switch_cpp.cpp:1181 *************
>> >> > DONE
>> >> > SETTING SESSION VARIABLES ***********
>> >> > 2011-02-21 17:22:53.377395 [NOTICE] switch_cpp.cpp:1181 Playing file:
>> >> >
>> /usr/local/freeswitch/recordings/messages/16c0f890_c35e33c0_777973.wav
>> >> > 2011-02-21 17:22:53.377395 [DEBUG] switch_ivr_play_say.c:1186 Codec
>> >> > Activated L16 at 8000hz 1 channels 20ms
>> >> > 2011-02-21 17:23:03.717511 [DEBUG] switch_ivr_play_say.c:1515 done
>> >> > playing
>> >> > file
>> >> > 2011-02-21 17:23:03.717511 [NOTICE] switch_cpp.cpp:1181 !!!!!
>> Finished
>> >> > playing the file !!!!!
>> >> > 2011-02-21 17:23:03.717511 [INFO] switch_cpp.cpp:1181 Was VM
>> detected?
>> >> > no
>> >> > 2011-02-21 17:23:03.717511 [NOTICE] switch_cpp.cpp:1181 Played the
>> >> > message
>> >> > at least once and checked for VM, we should be exiting the loop.
>> >> > 2011-02-21 17:23:03.717511 [NOTICE] switch_cpp.cpp:1181 *************
>> >> > SETTING SESSION VARIABLES ***************
>> >> > 2011-02-21 17:23:03.717511 [NOTICE] switch_cpp.cpp:1181 Session
>> >> > Variable[profile_id]: 1
>> >> > 2011-02-21 17:23:03.717511 [NOTICE] switch_cpp.cpp:1181 *************
>> >> > DONE
>> >> > SETTING SESSION VARIABLES ***********
>> >> > 2011-02-21 17:23:03.937661 [NOTICE] switch_cpp.cpp:1181 Playing file:
>> >> >
>> /usr/local/freeswitch/recordings/messages/16c0f890_c35e33c0_777973.wav
>> >> > 2011-02-21 17:23:03.937661 [DEBUG] switch_ivr_play_say.c:1186 Codec
>> >> > Activated L16 at 8000hz 1 channels 20ms
>> >> > 2011-02-21 17:23:08.237260 [DEBUG] switch_ivr_play_say.c:1515 done
>> >> > playing
>> >> > file
>> >> > 2011-02-21 17:23:08.237260 [NOTICE] switch_cpp.cpp:1181 !!!!!
>> Finished
>> >> > playing the file !!!!!
>> >> > 2011-02-21 17:23:08.237260 [INFO] switch_cpp.cpp:1181 Was VM
>> detected?
>> >> > no
>> >> > 2011-02-21 17:23:08.237260 [NOTICE] switch_cpp.cpp:1181 Played the
>> >> > message
>> >> > at least once and checked for VM, we should be exiting the loop.
>> >> > 2011-02-21 17:23:08.237260 [NOTICE] switch_cpp.cpp:1181 *************
>> >> > SETTING SESSION VARIABLES ***************
>> >> > 2011-02-21 17:23:08.237260 [NOTICE] switch_cpp.cpp:1181 Session
>> >> > Variable[profile_id]: 1
>> >> > 2011-02-21 17:23:08.237260 [NOTICE] switch_cpp.cpp:1181 *************
>> >> > DONE
>> >> > SETTING SESSION VARIABLES ***********
>> >> > 2011-02-21 17:23:08.437885 [NOTICE] switch_cpp.cpp:1181 Playing file:
>> >> >
>> /usr/local/freeswitch/recordings/messages/16c0f890_c35e33c0_777973.wav
>> >> > 2011-02-21 17:23:08.437885 [DEBUG] switch_ivr_play_say.c:1186 Codec
>> >> > Activated L16 at 8000hz 1 channels 20ms
>> >> > 2011-02-21 17:23:08.857270 [DEBUG] switch_ivr_play_say.c:1515 done
>> >> > playing
>> >> > file
>> >> > 2011-02-21 17:23:08.857270 [NOTICE] switch_cpp.cpp:1181 !!!!!
>> Finished
>> >> > playing the file !!!!!
>> >> > 2011-02-21 17:23:08.857270 [INFO] switch_cpp.cpp:1181 Was VM
>> detected?
>> >> > no
>> >> > 2011-02-21 17:23:08.857270 [NOTICE] switch_cpp.cpp:1181 Played the
>> >> > message
>> >> > at least once and checked for VM, we should be exiting the loop.
>> >> > 2011-02-21 17:23:08.857270 [NOTICE] switch_cpp.cpp:1181 *************
>> >> > SETTING SESSION VARIABLES ***************
>> >> > 2011-02-21 17:23:08.857270 [NOTICE] switch_cpp.cpp:1181 Session
>> >> > Variable[profile_id]: 1
>> >> > 2011-02-21 17:23:08.857270 [NOTICE] switch_cpp.cpp:1181 *************
>> >> > DONE
>> >> > SETTING SESSION VARIABLES ***********
>> >> > 2011-02-21 17:23:09.077014 [NOTICE] switch_cpp.cpp:1181 Playing file:
>> >> >
>> /usr/local/freeswitch/recordings/messages/16c0f890_c35e33c0_777973.wav
>> >> > 2011-02-21 17:23:09.077014 [DEBUG] switch_ivr_play_say.c:1186 Codec
>> >> > Activated L16 at 8000hz 1 channels 20ms
>> >> > 2011-02-21 17:23:19.417631 [DEBUG] switch_ivr_play_say.c:1515 done
>> >> > playing
>> >> > file
>> >> > 2011-02-21 17:23:19.417631 [NOTICE] switch_cpp.cpp:1181 !!!!!
>> Finished
>> >> > playing the file !!!!!
>> >> > 2011-02-21 17:23:19.417631 [INFO] switch_cpp.cpp:1181 Was VM
>> detected?
>> >> > no
>> >> > 2011-02-21 17:23:19.417631 [NOTICE] switch_cpp.cpp:1181 Played the
>> >> > message
>> >> > at least once and checked for VM, we should be exiting the loop.
>> >> >
>> >> > I also have the Freeswitch log from the date too, but I'm not sure
>> what
>> >> > I
>> >> > should post here that would be useful.
>> >> >
>> >> >
>> >> >
>> >> > On Tue, Feb 22, 2011 at 8:20 PM, Anthony Minessale
>> >> > <anthony.minessale at gmail.com> wrote:
>> >> >>
>> >> >> do you have any console logs of it failing or any error messages
>> etc?
>> >> >>
>> >> >>
>> >> >> On Tue, Feb 22, 2011 at 1:22 AM, Michael De Lorenzo
>> >> >> <delorenzodesign at gmail.com> wrote:
>> >> >> > Hello,
>> >> >> >
>> >> >> > I have a Freeswitch installation that is intended to make many
>> calls
>> >> >> > (thousands) and playback a single wav file.  The calls are
>> >> >> > successfully
>> >> >> > processed (the recipient's phone rings), but the call almost
>> >> >> > immediately
>> >> >> > disconnects, after about 1s.  Everything seems to work fine if I'm
>> >> >> > only
>> >> >> > pushing one or two calls through the Freeswitch instance, but as
>> soon
>> >> >> > as
>> >> >> > I
>> >> >> > turn up the call rate (I'm still only doing about 50 concurrent
>> >> >> > sessions)
>> >> >> > the playback begins to fail.
>> >> >> >
>> >> >> > I've watched the calls go out from the console and nothing looks
>> out
>> >> >> > of
>> >> >> > the
>> >> >> > ordinary, except that the calls are disconnected with NORMAL
>> CLEARING
>> >> >> > prior
>> >> >> > to completion.
>> >> >> >
>> >> >> > Here's the Lua script I'm using...
>> >> >> >
>> >> >> > profile_id = argv[1];
>> >> >> > account_code = argv[2];
>> >> >> > client_id = argv[3];
>> >> >> > caller_id_name = argv[4];
>> >> >> > caller_id = argv[5];
>> >> >> > dial_id = argv[6];
>> >> >> > number_to_call = argv[7];
>> >> >> > message_to_play = argv[8];
>> >> >> > max_retries = argv[9];
>> >> >> > retry_interval = argv[10];
>> >> >> >
>> >> >> > local human_detected = false
>> >> >> > local voicemail_detected = false;
>> >> >> > local message_played = false;
>> >> >> >
>> >> >> > recordings_directory =
>> "/usr/local/freeswitch/recordings/messages/";
>> >> >> >
>> >> >> > function setDialVariables(set_as_session_variables)
>> >> >> >     local s = "profile_id=" .. profile_id;
>> >> >> >     s = s .. ",account_code=" .. account_code;
>> >> >> >     s = s .. ",client_id=" .. client_id;
>> >> >> >     s = s .. ",caller_id_name=" .. caller_id_name;
>> >> >> >     s = s .. ",caller_id=" .. caller_id;
>> >> >> >     s = s .. ",dial_id=" .. dial_id;
>> >> >> >     s = s .. ",number_to_call=" .. number_to_call;
>> >> >> >     s = s .. ",message_to_play=" .. message_to_play;
>> >> >> >
>> >> >> >     freeswitch.consoleLog("notice", s .. "\n");
>> >> >> >
>> >> >> >     return s
>> >> >> > end
>> >> >> >
>> >> >> > function printSessionVariables()
>> >> >> >     freeswitch.consoleLog("notice", "******* PRINTING SESSION
>> >> >> > VARIABLES
>> >> >> > **********\n");
>> >> >> >     -- ommitted
>> >> >> >     freeswitch.consoleLog("notice",
>> >> >> > "**********************************************\n");
>> >> >> > end
>> >> >> >
>> >> >> > function onInput(s, type, obj, arg)
>> >> >> >     if(type == "event" and voicemail_detected == false) then
>> >> >> >         freeswitch.consoleLog("debug","************ VOICE
>> >> >> > MAIL/ANSWERING
>> >> >> > MACHINE DETECTED *************\n");
>> >> >> >         voicemail_detected = true;
>> >> >> >         return "break";
>> >> >> >     end
>> >> >> >     return true;
>> >> >> > end
>> >> >> >
>> >> >> > function playbackMessage(sleepTime)
>> >> >> >     message_played = false;
>> >> >> >     session:sleep(sleepTime);
>> >> >> >     -- play a file
>> >> >> >     message_file = recordings_directory .. message_to_play;
>> >> >> >     freeswitch.consoleLog("notice", "Playing file: " ..
>> message_file
>> >> >> > ..
>> >> >> > "\n");
>> >> >> >     session:streamFile(message_file);
>> >> >> >     freeswitch.consoleLog("notice", "!!!!! Finished playing the
>> file
>> >> >> > !!!!!\n");
>> >> >> >     message_played = true;
>> >> >> > end
>> >> >> >
>> >> >> > session = freeswitch.Session("{" .. setDialVariables(false) ..
>> >> >> > ",ignore_early_media=true,origination_caller_id_name=" ..
>> >> >> > caller_id_name
>> >> >> > ..
>> >> >> > ",origination_caller_id_number=+1" .. caller_id ..
>> >> >> > "}sofia/gateway/gateway_"
>> >> >> > .. profile_id .. "/" .. number_to_call);
>> >> >> >
>> >> >> > while(session:ready()) do
>> >> >> >    setDialVariables(true)
>> >> >> >    session:answer();
>> >> >> >
>> >> >> >    -- session:execute("continue_on_fail","true");
>> >> >> >    session:setInputCallback("onInput","true");
>> >> >> >    session:execute("avmd","start");
>> >> >> >
>> >> >> >    playbackMessage(200);
>> >> >> >
>> >> >> >    vm_status = voicemail_detected == true and "yes" or "no"
>> >> >> >    freeswitch.consoleLog("info", "Was VM detected? " .. vm_status
>> ..
>> >> >> > "\n");
>> >> >> >    if(voicemail_detected) then
>> >> >> >       return "break";
>> >> >> >    end
>> >> >> >
>> >> >> >    freeswitch.consoleLog("notice", "Played the message at least
>> once
>> >> >> > and
>> >> >> > checked for VM, we should be exiting the loop.\n")
>> >> >> > end
>> >> >> >
>> >> >> > if (voicemail_detected) then
>> >> >> >     freeswitch.consoleLog("info", "Playback for voicemail.\n");
>> >> >> >     session:execute("avmd","stop");
>> >> >> >     playbackMessage(5000);
>> >> >> > end
>> >> >> >
>> >> >> > freeswitch.consoleLog("info", "All finished, hanging up the
>> >> >> > session.\n");
>> >> >> > session:hangup();
>> >> >> >
>> >> >> > Any help would be greatly appreciated.
>> >> >> >
>> >> >> > Thank you,
>> >> >> >
>> >> >> > 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
>> >> >> >
>> >> >> >
>> >> >>
>> >> >>
>> >> >>
>> >> >> --
>> >> >> Anthony Minessale II
>> >> >>
>> >> >> FreeSWITCH http://www.freeswitch.org/
>> >> >> ClueCon http://www.cluecon.com/
>> >> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >>
>> >> >> AIM: anthm
>> >> >> MSN:anthony_minessale at hotmail.com
>> >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.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
>>
>
>
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From steveayre at gmail.com  Thu Feb 24 12:25:14 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 24 Feb 2011 09:25:14 +0000
Subject: [Freeswitch-users] excute app from CLI with vars on answered
	channel
In-Reply-To: <3F3D30233AF74DEFB8977A1B3AD5AEBA@e1705>
References: <3F3D30233AF74DEFB8977A1B3AD5AEBA@e1705>
Message-ID: <AANLkTiksf7UOLGx0fz8DLkyMhxa7=xeANW+Pes+pWwO5@mail.gmail.com>

Use uuid_broadcast:
http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast

Usage: uuid_broadcast <uuid> app[![hangup_cause]]::args [aleg|bleg|both]

Hangup is after the application runs is optional.

-Steve



On 24 February 2011 03:23, Madovsky <infos at madovsky.org> wrote:

>  I'd like to execute an app while from CLI
> on an answered channel (I know the uuid) and pass vars on it like
>
> {exec?} <app with vars and params> <uuid>
>
> or if it's not possible execute an extension from CLI with a  uuid
>
>
> 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 david.villasmil.work at gmail.com  Thu Feb 24 14:39:18 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 24 Feb 2011 12:39:18 +0100
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
Message-ID: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>

Hello Guys,

I'm finishing a "complete" wholesale application created on freeswitch and I
was wondering whether it would be a good idea to put it up on the wiki. I
just don't know how.

Features include all the following parameters configurable via web
interface:

- Profile creation based on server IP where traffic is received. You can
have multiple IPs, system will create multiple profiles/diaplans so it can
differentiate.
     - i.e. offer to the same customer a "gold" routing on IP1 and cheap
routing on IP2

- Customer add/modify/delete
     - IP source
     - Rates for client routes based on areacode
     - Prepaid or postpaid.
          - When cutomer balance is 0, no more calls are allowed.
     - limit max channels
     - Media by-pass
          - When by-passed, customer and provider will exchange RTPs
directly. Else, server will be in the middle.

- Provider add/modify/delete
     - costs for provider routes based on areacode
     - limit max channels

- Routing based on areacode, gives great granularity.

- Routes can be assigned multiple gateways/providers which can in turn be
distributed based on weigth. Includes overflow to next configured GW.

- Basic financial report generation (totals) by customer/provider

- Basic traffic ASR/ACD report (totals) by cutomer/provider

- Basic user administration. (No access level, only total access)

- CDR export to csv file.




I also have a prepaid card app... no web interface on that one though...

Thanks all


David
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From avi at avimarcus.net  Thu Feb 24 14:54:21 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 24 Feb 2011 13:54:21 +0200
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
Message-ID: <AANLkTi=nJ3yHO7xLFrXETULmxpZz-+KK1HJnO1Phx-j+@mail.gmail.com>

I'd be interested in such a thing.
There's a separate GIT for contributions which you can get access to,
otherwise you could throw it up on a free github account. Then, just make a
link to it on the wiki.
Some, e.g. grnvoip, differentiate grey/premium routes by a brand/tech prefix
rather than just by IP. You want it by different IP though?

Also, for calling card, how does it handle the billing? I'd be interested in
seeing it.

-Avi




On Thu, Feb 24, 2011 at 1:39 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello Guys,
>
> I'm finishing a "complete" wholesale application created on freeswitch and
> I was wondering whether it would be a good idea to put it up on the wiki. I
> just don't know how.
>
> Features include all the following parameters configurable via web
> interface:
>
> - Profile creation based on server IP where traffic is received. You can
> have multiple IPs, system will create multiple profiles/diaplans so it can
> differentiate.
>      - i.e. offer to the same customer a "gold" routing on IP1 and cheap
> routing on IP2
>
> - Customer add/modify/delete
>      - IP source
>      - Rates for client routes based on areacode
>      - Prepaid or postpaid.
>           - When cutomer balance is 0, no more calls are allowed.
>      - limit max channels
>      - Media by-pass
>           - When by-passed, customer and provider will exchange RTPs
> directly. Else, server will be in the middle.
>
> - Provider add/modify/delete
>      - costs for provider routes based on areacode
>      - limit max channels
>
> - Routing based on areacode, gives great granularity.
>
> - Routes can be assigned multiple gateways/providers which can in turn be
> distributed based on weigth. Includes overflow to next configured GW.
>
> - Basic financial report generation (totals) by customer/provider
>
> - Basic traffic ASR/ACD report (totals) by cutomer/provider
>
> - Basic user administration. (No access level, only total access)
>
> - CDR export to csv file.
>
>
>
>
> I also have a prepaid card app... no web interface on that one though...
>
> 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 david.villasmil.work at gmail.com  Thu Feb 24 15:31:11 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 24 Feb 2011 13:31:11 +0100
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <AANLkTi=nJ3yHO7xLFrXETULmxpZz-+KK1HJnO1Phx-j+@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTi=nJ3yHO7xLFrXETULmxpZz-+KK1HJnO1Phx-j+@mail.gmail.com>
Message-ID: <AANLkTi=n=5Rq3KHWxch8sPHXy+Umg2FHLa9a70hZuknf@mail.gmail.com>

Hello AviMarcus,

The calling card works the following way:

- You can have multiple card access numbers associated to a card "name"
- Each card is associated with a rate table.
- Each areacode in the rate table has a route to use, which in turn is a
distributor gw list.
- Every gateway is actually a provider which also has a cost table
associated with it.
- After the call duration is calculated, the app sets a shcedule-api to
disconnect the call.
- When the call is hung up, a CDR is posted to the webserver via xml_cdr.
- This CDR contains all information regarding the call. i.e. rate, cost,
clgnum, cldnum, gw_used, balance_before, balance_after, ratetable,
costtable, etc...


David


On Thu, Feb 24, 2011 at 12:54 PM, Avi Marcus <avi at avimarcus.net> wrote:

> I'd be interested in such a thing.
> There's a separate GIT for contributions which you can get access to,
> otherwise you could throw it up on a free github account. Then, just make a
> link to it on the wiki.
> Some, e.g. grnvoip, differentiate grey/premium routes by a brand/tech
> prefix rather than just by IP. You want it by different IP though?
>
> Also, for calling card, how does it handle the billing? I'd be interested
> in seeing it.
>
> -Avi
>
>
>
>
> On Thu, Feb 24, 2011 at 1:39 PM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Hello Guys,
>>
>> I'm finishing a "complete" wholesale application created on freeswitch and
>> I was wondering whether it would be a good idea to put it up on the wiki. I
>> just don't know how.
>>
>> Features include all the following parameters configurable via web
>> interface:
>>
>> - Profile creation based on server IP where traffic is received. You can
>> have multiple IPs, system will create multiple profiles/diaplans so it can
>> differentiate.
>>      - i.e. offer to the same customer a "gold" routing on IP1 and cheap
>> routing on IP2
>>
>> - Customer add/modify/delete
>>      - IP source
>>      - Rates for client routes based on areacode
>>      - Prepaid or postpaid.
>>           - When cutomer balance is 0, no more calls are allowed.
>>      - limit max channels
>>      - Media by-pass
>>           - When by-passed, customer and provider will exchange RTPs
>> directly. Else, server will be in the middle.
>>
>> - Provider add/modify/delete
>>      - costs for provider routes based on areacode
>>      - limit max channels
>>
>> - Routing based on areacode, gives great granularity.
>>
>> - Routes can be assigned multiple gateways/providers which can in turn be
>> distributed based on weigth. Includes overflow to next configured GW.
>>
>> - Basic financial report generation (totals) by customer/provider
>>
>> - Basic traffic ASR/ACD report (totals) by cutomer/provider
>>
>> - Basic user administration. (No access level, only total access)
>>
>> - CDR export to csv file.
>>
>>
>>
>>
>> I also have a prepaid card app... no web interface on that one though...
>>
>> 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 jjj at 3js.de  Thu Feb 24 19:44:06 2011
From: jjj at 3js.de (Johannes Jakob)
Date: Thu, 24 Feb 2011 17:44:06 +0100
Subject: [Freeswitch-users] T.38 Issues with passthrough handshaking
Message-ID: <640F91CF-9489-4F74-96B1-A6398CD84F07@3js.de>

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

From infos at madovsky.org  Thu Feb 24 20:16:07 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 24 Feb 2011 12:16:07 -0500
Subject: [Freeswitch-users] excute app from CLI with vars on
	answeredchannel
References: <3F3D30233AF74DEFB8977A1B3AD5AEBA@e1705>
	<AANLkTiksf7UOLGx0fz8DLkyMhxa7=xeANW+Pes+pWwO5@mail.gmail.com>
Message-ID: <DBFCCA7E1D814287A5E79432AAB04E39@e1705>

of course, forgot this one . .
thanks Steve
  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Thursday, February 24, 2011 4:25 AM
  Subject: Re: [Freeswitch-users] excute app from CLI with vars on answeredchannel


  Use uuid_broadcast:
  http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast

  Usage: uuid_broadcast <uuid> app[![hangup_cause]]::args [aleg|bleg|both]

  Hangup is after the application runs is optional.

  -Steve






  On 24 February 2011 03:23, Madovsky <infos at madovsky.org> wrote:

    I'd like to execute an app while from CLI
    on an answered channel (I know the uuid) and pass vars on it like

    {exec?} <app with vars and params> <uuid>

    or if it's not possible execute an extension from CLI with a  uuid 


    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 k-b-24 at live.com  Thu Feb 24 21:08:05 2011
From: k-b-24 at live.com (Jason b.a)
Date: Thu, 24 Feb 2011 18:08:05 +0000
Subject: [Freeswitch-users] play media
Message-ID: <COL111-W41D3485B4DE87A09C11051ACDA0@phx.gbl>


guys in this case: 
 
sip caller -------Sip-------- Application server -------socket------- FreeSWITCH
 
in this case how freeswitch is going to play media (RTP) for the sip caller, if i am only using the socket interface of freeswitch.
 
Thx in advance ! 
  		 	   		  
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From santiagosoares at gmail.com  Thu Feb 24 21:22:55 2011
From: santiagosoares at gmail.com (Santiago Soares)
Date: Thu, 24 Feb 2011 15:22:55 -0300
Subject: [Freeswitch-users] Second transfer
In-Reply-To: <AANLkTi=fvkp_8e4gXthWtYQTLAyxj+bbUMtdpAhXQamQ@mail.gmail.com>
References: <AANLkTi=pB-Rt7j1cvNGgSs2s3F4h6jse1nTqTqRr8kKJ@mail.gmail.com>
	<AANLkTi=fvkp_8e4gXthWtYQTLAyxj+bbUMtdpAhXQamQ@mail.gmail.com>
Message-ID: <AANLkTinw4CWxe+GW31AcMnzWWNsQU=W7WGSmsr2ky9zg@mail.gmail.com>

Thanks for your answer, but it doesn't work.
I tried to assign *3 to log, like this:

   <extension name="transferencia2">
      <condition field="${attxfer_callthis}" expression="^numberC$">
         <action application="bind_meta_app" data="3 b s log::INFO
Transfer"/>
         <action application="att_xfer" data="sofia/gateway/test/loginC"/>
      </condition>
   </extension>

But *3 still calls the transferencia extension.
Any ideas?

Thank you very much!

Santiago Soares
Fone: (41) 8488-0537


On Wed, Feb 23, 2011 at 8:20 PM, Michael Collins <msc at freeswitch.org> wrote:

> I think you can make another call to bind_meta_app and disable the *3 prior
> to the att_xfer. You could make *3 do something completely different (like
> trigger a log entry) and then you could see how many times your users tried
> to use *3 when they weren't supposed to. :)
>
> -MC
>
> On Wed, Feb 23, 2011 at 10:01 AM, Santiago Soares <
> santiagosoares at gmail.com> wrote:
>
>> Hello,
>>
>> I'm using att_xfer to make call transfer, like that:
>> A calls B
>> B calls C
>> B hangup and A is bridged do C
>>
>> The thing is, the way I am trying to do it, C is able to transfer the call
>> again.
>> I don't want that C be able to transfer the call again.
>> The call should be transfered only once.
>> Is it possible?
>>
>> This is my dialplan:
>>
>> <include>
>>
>>    <extension name="test_extension">
>>       <condition field="destination_number" expression="^numberB$">
>>          <action application="set" data="bind_meta_key=*"/>
>>          <action application="bind_meta_app" data="3 b s
>> execute_extension::transferencia XML public"/>
>>          <action application="bridge" data="sofia/gateway/test/loginB"/>
>>       </condition>
>>    </extension>
>>
>>    <extension name="transferencia">
>>       <condition field="destination_number" expression="^transferencia$">
>>          <action application="read" data="3 3 base256/16000/sound.wav
>> attxfer_callthis 30000 #"/>
>>          <action application="execute_extension" data="transferencia2"/>
>>       </condition>
>>    </extension>
>>
>>    <extension name="transferencia2">
>>       <condition field="${attxfer_callthis}" expression="^numberC$">
>>          <action application="att_xfer" data="sofia/gateway/test/loginC"/>
>>       </condition>
>>       <condition field="${attxfer_callthis}" expression="^numberD$">
>>          <action application="att_xfer" data="sofia/gateway/test/loginD"/>
>>       </condition>
>>    </extension>
>>
>> </include>
>>
>>
>> Thank you,
>> Santiago Soares
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cmrienzo at gmail.com  Thu Feb 24 21:59:31 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Thu, 24 Feb 2011 13:59:31 -0500
Subject: [Freeswitch-users] play media
In-Reply-To: <COL111-W41D3485B4DE87A09C11051ACDA0@phx.gbl>
References: <COL111-W41D3485B4DE87A09C11051ACDA0@phx.gbl>
Message-ID: <AANLkTik9t09NY_CfpxJ1MVAvGMDG8=LF+30N-bjUvDpi@mail.gmail.com>

Why not have the call go straight to freeswitch and use outbound event
socket instead?

On Feb 24, 2011 1:10 PM, "Jason b.a" <k-b-24 at live.com> wrote:

 guys in this case:

*sip caller *-------Sip--------* Application server *-------socket------- *
FreeSWITCH*

in this case how freeswitch is going to play media (RTP) for the sip caller,
if i am only using the socket interface of freeswitch.

Thx 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
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From k-b-24 at live.com  Thu Feb 24 22:22:47 2011
From: k-b-24 at live.com (Jason b.a)
Date: Thu, 24 Feb 2011 19:22:47 +0000
Subject: [Freeswitch-users] play media
Message-ID: <COL111-W546625C8844D550C9EDAABACDA0@phx.gbl>


coz i need freeswitch only for media processing, i dont want to have a direct SIP connection between user and Freeswitch,
so the application server can handle the call using SIP, and using the event socket, it can ask freeswitch to play prompt, or collect digit and 
return it to the application so the application according to the digits can bridge the call. is it possible ? and how freeswitch can play RTP to the caller ? 		 	   		  
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From matzemuc86 at gmail.com  Thu Feb 24 20:22:38 2011
From: matzemuc86 at gmail.com (MatzeMuc86)
Date: Thu, 24 Feb 2011 18:22:38 +0100
Subject: [Freeswitch-users] implementing stereo (e.g. portaudio to test it
	easily)
Message-ID: <AANLkTikL9SQPdsJ6XF-AUZuLQ3Z-AG7Eqgj2BsL=CZ4S@mail.gmail.com>

hello FreeSWITCH community,

actually it is my first time to use lists like this. I hope I do everything
fine.
For my project I try to get stereo (and some special features later)
integrated in freeSWITCH, especially SIP.
TO start with this idea I need to know where I have to activate or implement
stereo.
In the second I try to play a bit with the portaudio module as I do not have
to focus on SDP problems transporting SIP and the need of a SIP client which
is able to receive stereo.
So I set up some custom moh local media stuff which works finde - using
mono.
Changing the channels to 2 makes the console telling me that the sound file
will be mixed to mono and the sound is quite strange.
I would be happy about getting some useful information which modules or
configs are the important ones for my idea. Unfurtun ately I have only found
some recording stuff about stereo which is not what I want to do.

Thank you very much.
MatzeMuc86
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From mauritz.lovgren at hotmail.com  Thu Feb 24 22:01:27 2011
From: mauritz.lovgren at hotmail.com (=?utf-8?Q?Mauritz_L=C3=B8vgren?=)
Date: Thu, 24 Feb 2011 20:01:27 +0100
Subject: [Freeswitch-users] Receive order guarantee for bgapi command
	replies over ESL?
Message-ID: <SNT107-DS120CD9CBAFDAA80E16A6CDE6DA0@phx.gbl>

Hi,

Are replies for bgapi commands sent back to an inbound socket connection in the same order as the commands were sent?

Example: sending 10 bgapi messages from client to freeswitch socket with no delay inbetween.
Will the response for those command messages be sent back by freeswitch in the exact same order as their commands were received, or should one wait for each reply before sending a new bgapi command to be sure one gets a reply for the correct command?

If there was an option of providing a UUID (or sequenceId) for the command, it would be easier to match the reply with the request upon receival, but this doesn?t seem to be possible with the current implementation?

Regards,
Mauritz Lovgren
Systems Architect
IPLink Inc.
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From brad at tritelcomm.com  Thu Feb 24 22:31:19 2011
From: brad at tritelcomm.com (Brad Mina)
Date: Thu, 24 Feb 2011 11:31:19 -0800
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <AANLkTi=n=5Rq3KHWxch8sPHXy+Umg2FHLa9a70hZuknf@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTi=nJ3yHO7xLFrXETULmxpZz-+KK1HJnO1Phx-j+@mail.gmail.com>
	<AANLkTi=n=5Rq3KHWxch8sPHXy+Umg2FHLa9a70hZuknf@mail.gmail.com>
Message-ID: <AANLkTimAGkSQ-KkGwqFCkXx29-kVNW5MOX0orGT9DoNw@mail.gmail.com>

I'm very interested as well. This sounds very useful.

On Thu, Feb 24, 2011 at 4:31 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello AviMarcus,
>
> The calling card works the following way:
>
> - You can have multiple card access numbers associated to a card "name"
> - Each card is associated with a rate table.
> - Each areacode in the rate table has a route to use, which in turn is a
> distributor gw list.
> - Every gateway is actually a provider which also has a cost table
> associated with it.
> - After the call duration is calculated, the app sets a shcedule-api to
> disconnect the call.
> - When the call is hung up, a CDR is posted to the webserver via xml_cdr.
> - This CDR contains all information regarding the call. i.e. rate, cost,
> clgnum, cldnum, gw_used, balance_before, balance_after, ratetable,
> costtable, etc...
>
>
> David
>
>
> On Thu, Feb 24, 2011 at 12:54 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> I'd be interested in such a thing.
>> There's a separate GIT for contributions which you can get access to,
>> otherwise you could throw it up on a free github account. Then, just make a
>> link to it on the wiki.
>> Some, e.g. grnvoip, differentiate grey/premium routes by a brand/tech
>> prefix rather than just by IP. You want it by different IP though?
>>
>> Also, for calling card, how does it handle the billing? I'd be interested
>> in seeing it.
>>
>> -Avi
>>
>>
>>
>>
>> On Thu, Feb 24, 2011 at 1:39 PM, David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> Hello Guys,
>>>
>>> I'm finishing a "complete" wholesale application created on freeswitch
>>> and I was wondering whether it would be a good idea to put it up on the
>>> wiki. I just don't know how.
>>>
>>> Features include all the following parameters configurable via web
>>> interface:
>>>
>>> - Profile creation based on server IP where traffic is received. You can
>>> have multiple IPs, system will create multiple profiles/diaplans so it can
>>> differentiate.
>>>      - i.e. offer to the same customer a "gold" routing on IP1 and cheap
>>> routing on IP2
>>>
>>> - Customer add/modify/delete
>>>      - IP source
>>>      - Rates for client routes based on areacode
>>>      - Prepaid or postpaid.
>>>           - When cutomer balance is 0, no more calls are allowed.
>>>      - limit max channels
>>>      - Media by-pass
>>>           - When by-passed, customer and provider will exchange RTPs
>>> directly. Else, server will be in the middle.
>>>
>>> - Provider add/modify/delete
>>>      - costs for provider routes based on areacode
>>>      - limit max channels
>>>
>>> - Routing based on areacode, gives great granularity.
>>>
>>> - Routes can be assigned multiple gateways/providers which can in turn be
>>> distributed based on weigth. Includes overflow to next configured GW.
>>>
>>> - Basic financial report generation (totals) by customer/provider
>>>
>>> - Basic traffic ASR/ACD report (totals) by cutomer/provider
>>>
>>> - Basic user administration. (No access level, only total access)
>>>
>>> - CDR export to csv file.
>>>
>>>
>>>
>>>
>>> I also have a prepaid card app... no web interface on that one though...
>>>
>>> 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
>
>
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From infos at madovsky.org  Thu Feb 24 22:35:13 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 24 Feb 2011 14:35:13 -0500
Subject: [Freeswitch-users] one leg call and channel uuid
Message-ID: <5CB1934490CF4E55977717E2B6455962@e1705>

when there is one leg call,
if I want to use uuid_broadcast or uuid_transfer,
should I use the uuid of leg A ?

Thanks
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From cmrienzo at gmail.com  Thu Feb 24 22:51:17 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Thu, 24 Feb 2011 14:51:17 -0500
Subject: [Freeswitch-users] play media
In-Reply-To: <COL111-W546625C8844D550C9EDAABACDA0@phx.gbl>
References: <COL111-W546625C8844D550C9EDAABACDA0@phx.gbl>
Message-ID: <AANLkTikmL+YeOwu932spb1u4FNLomA0tos=C4gNO11kt@mail.gmail.com>

You'll have to pick a call flow from RFC 3275, then.



On Thu, Feb 24, 2011 at 2:22 PM, Jason b.a <k-b-24 at live.com> wrote:

>  coz i need freeswitch only for media processing, i dont want to have a
> direct SIP connection between user and Freeswitch,
> so the application server can handle the call using SIP, and using the
> event socket, it can ask freeswitch to play prompt, or collect digit and
> return it to the application so the application according to the digits can
> bridge the call. is it possible ? and how freeswitch can play RTP to the
> caller ?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cmrienzo at gmail.com  Thu Feb 24 23:00:30 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Thu, 24 Feb 2011 15:00:30 -0500
Subject: [Freeswitch-users] play media
In-Reply-To: <AANLkTikmL+YeOwu932spb1u4FNLomA0tos=C4gNO11kt@mail.gmail.com>
References: <COL111-W546625C8844D550C9EDAABACDA0@phx.gbl>
	<AANLkTikmL+YeOwu932spb1u4FNLomA0tos=C4gNO11kt@mail.gmail.com>
Message-ID: <AANLkTikcWm5Cj-ov8L+iOqp697NB-rBGDf3BNdA+QkV1@mail.gmail.com>

Sorry, RFC 3725.


On Thu, Feb 24, 2011 at 2:51 PM, Christopher Rienzo <cmrienzo at gmail.com>wrote:

> You'll have to pick a call flow from RFC 3275, then.
>
>
>
> On Thu, Feb 24, 2011 at 2:22 PM, Jason b.a <k-b-24 at live.com> wrote:
>
>>  coz i need freeswitch only for media processing, i dont want to have a
>> direct SIP connection between user and Freeswitch,
>> so the application server can handle the call using SIP, and using the
>> event socket, it can ask freeswitch to play prompt, or collect digit and
>> return it to the application so the application according to the digits
>> can bridge the call. is it possible ? and how freeswitch can play RTP to the
>> caller ?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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  Thu Feb 24 22:47:17 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Thu, 24 Feb 2011 19:47:17 +0000
Subject: [Freeswitch-users] Mitel 3300 ICP "Multicall" Equivalent
Message-ID: <4D66B5C5.7000005@integrafin.co.uk>

Hello list,

Our company is considering a migration from Mitel 3300 ICP to an 
alternative platform, and of course FreeSwitch is a major contender, 
being as we are not enamoured with "black boxes" and proprietary protocols.

There is a function in the Mitel that ideally we'd like to replicate (on 
whatever hardphones are most appropriate). "Multicall" allows you to 
assign one number to a button (other than the "main line" or default 
extension) on several phones (up to 31 on the 3300) such that a call (or 
any number of concurrent calls up to the number of phones configured 
with that button)  to that number can be signalled on all phones and 
picked up by anyone with that button configured on that phone. We use 
that for calling direct to regional teams. No barging is allowed, and 
the button only changes state (from off to flashing) when a call needs 
answering on the multicall number. Said button assignment can be set to 
either ring or not. Once the call has been answered by one user of the 
multicall the light goes off on other configured extensions and another 
incoming call can be answered in the same way by another extension, 
regardless of the status of all other extensions with the same button 
configured.

I was looking at the SCA/SLA stuff on the wiki, which is a close as I 
could find to this behaviour, but barging is unwanted and we require the 
line appearance, ringing and answering to be as described above.

My question, given the above, is:

1. If SCA is able to do this, what is the best phone to support it 
(Polycom seems good from the wiki)
2. If SCA is not appropriate, I have the feeling that queues would be 
the next best option - if these can be assigned to a button on the phone 
so much the better.
3. BLF doesn't look right as it seems it needs at least one extension 
with that as its "primary" number.

There is some flexibilty here, but to ease transition it would be best 
to keep the functionality as close as possible.

IMHO the Mitel way is really odd but it's got a lot of traction having 
been used here for 7 years+.

Thanks

Alex

-- 
This message is intended only for the addressee and may contain
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the message along with any attachments and notify us immediately.

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From msc at freeswitch.org  Thu Feb 24 23:06:52 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 24 Feb 2011 12:06:52 -0800
Subject: [Freeswitch-users] play media
In-Reply-To: <COL111-W546625C8844D550C9EDAABACDA0@phx.gbl>
References: <COL111-W546625C8844D550C9EDAABACDA0@phx.gbl>
Message-ID: <AANLkTikPm7dc=gPMMhhZWk3W51rsNMHwJHD-1qsw6NM4@mail.gmail.com>

On Thu, Feb 24, 2011 at 11:22 AM, Jason b.a <k-b-24 at live.com> wrote:

>  coz i need freeswitch only for media processing, i dont want to have a
> direct SIP connection between user and Freeswitch,
> so the application server can handle the call using SIP, and using the
> event socket, it can ask freeswitch to play prompt, or collect digit and
> return it to the application so the application according to the digits can
> bridge the call. is it possible ? and how freeswitch can play RTP to the
> caller ?
>

I recommend that you contact support at freeswitch.org and ask for professional
assistance. It is not possible to answer these questions via mailing list.
The subject matter is too deep for emailing back and forth. You need
professional help (uh, so to speak).

You can also join IRC: #freeswitch on irc.freenode.net. We have a number of
community members who hang out there and you might get lucky and find
someone who can help you.

-MC
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From msc at freeswitch.org  Thu Feb 24 23:09:51 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 24 Feb 2011 12:09:51 -0800
Subject: [Freeswitch-users] implementing stereo (e.g. portaudio to test
	it easily)
In-Reply-To: <AANLkTikL9SQPdsJ6XF-AUZuLQ3Z-AG7Eqgj2BsL=CZ4S@mail.gmail.com>
References: <AANLkTikL9SQPdsJ6XF-AUZuLQ3Z-AG7Eqgj2BsL=CZ4S@mail.gmail.com>
Message-ID: <AANLkTinowzmzt8NxB-FtifkZEQZ8s3OQAWLuC18PN60Q@mail.gmail.com>

Sorry, FreeSWITCH is not set up to do stereo... perhaps one day, but it
isn't something that is high on the priority list.

-MC

On Thu, Feb 24, 2011 at 9:22 AM, MatzeMuc86 <matzemuc86 at gmail.com> wrote:

> hello FreeSWITCH community,
>
> actually it is my first time to use lists like this. I hope I do everything
> fine.
> For my project I try to get stereo (and some special features later)
> integrated in freeSWITCH, especially SIP.
> TO start with this idea I need to know where I have to activate or
> implement stereo.
> In the second I try to play a bit with the portaudio module as I do not
> have to focus on SDP problems transporting SIP and the need of a SIP client
> which is able to receive stereo.
> So I set up some custom moh local media stuff which works finde - using
> mono.
> Changing the channels to 2 makes the console telling me that the sound file
> will be mixed to mono and the sound is quite strange.
> I would be happy about getting some useful information which modules or
> configs are the important ones for my idea. Unfurtun ately I have only found
> some recording stuff about stereo which is not what I want to do.
>
> Thank you very much.
> MatzeMuc86
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 24 23:23:13 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 24 Feb 2011 12:23:13 -0800
Subject: [Freeswitch-users] one leg call and channel uuid
In-Reply-To: <5CB1934490CF4E55977717E2B6455962@e1705>
References: <5CB1934490CF4E55977717E2B6455962@e1705>
Message-ID: <AANLkTimVWtHEh2K_6TdVftLpsW6TD7aUPZxJrqQtgd6f@mail.gmail.com>

On Thu, Feb 24, 2011 at 11:35 AM, Madovsky <infos at madovsky.org> wrote:

>  when there is one leg call,
> if I want to use uuid_broadcast or uuid_transfer,
> should I use the uuid of leg A ?
>
> What is the scenario? What is your one-legged call?
-MC
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From nazim.aghabayov at gmail.com  Thu Feb 24 23:29:57 2011
From: nazim.aghabayov at gmail.com (Nazim Aghabayov)
Date: Fri, 25 Feb 2011 00:29:57 +0400
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <AANLkTimAGkSQ-KkGwqFCkXx29-kVNW5MOX0orGT9DoNw@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>	<AANLkTi=nJ3yHO7xLFrXETULmxpZz-+KK1HJnO1Phx-j+@mail.gmail.com>	<AANLkTi=n=5Rq3KHWxch8sPHXy+Umg2FHLa9a70hZuknf@mail.gmail.com>
	<AANLkTimAGkSQ-KkGwqFCkXx29-kVNW5MOX0orGT9DoNw@mail.gmail.com>
Message-ID: <4D66BFC5.6090405@gmail.com>

Thanks David! It would be great to see you app on freeswith-contrib.

On 02/24/2011 11:31 PM, Brad Mina wrote:
> I'm very interested as well. This sounds very useful.
>
> On Thu, Feb 24, 2011 at 4:31 AM, David Villasmil<
> david.villasmil.work at gmail.com>  wrote:
>
>> Hello AviMarcus,
>>
>> The calling card works the following way:
>>
>> - You can have multiple card access numbers associated to a card "name"
>> - Each card is associated with a rate table.
>> - Each areacode in the rate table has a route to use, which in turn is a
>> distributor gw list.
>> - Every gateway is actually a provider which also has a cost table
>> associated with it.
>> - After the call duration is calculated, the app sets a shcedule-api to
>> disconnect the call.
>> - When the call is hung up, a CDR is posted to the webserver via xml_cdr.
>> - This CDR contains all information regarding the call. i.e. rate, cost,
>> clgnum, cldnum, gw_used, balance_before, balance_after, ratetable,
>> costtable, etc...
>>
>>
>> David
>>
>>
>> On Thu, Feb 24, 2011 at 12:54 PM, Avi Marcus<avi at avimarcus.net>  wrote:
>>
>>> I'd be interested in such a thing.
>>> There's a separate GIT for contributions which you can get access to,
>>> otherwise you could throw it up on a free github account. Then, just make a
>>> link to it on the wiki.
>>> Some, e.g. grnvoip, differentiate grey/premium routes by a brand/tech
>>> prefix rather than just by IP. You want it by different IP though?
>>>
>>> Also, for calling card, how does it handle the billing? I'd be interested
>>> in seeing it.
>>>
>>> -Avi
>>>
>>>
>>>
>>>
>>> On Thu, Feb 24, 2011 at 1:39 PM, David Villasmil<
>>> david.villasmil.work at gmail.com>  wrote:
>>>
>>>> Hello Guys,
>>>>
>>>> I'm finishing a "complete" wholesale application created on freeswitch
>>>> and I was wondering whether it would be a good idea to put it up on the
>>>> wiki. I just don't know how.
>>>>
>>>> Features include all the following parameters configurable via web
>>>> interface:
>>>>
>>>> - Profile creation based on server IP where traffic is received. You can
>>>> have multiple IPs, system will create multiple profiles/diaplans so it can
>>>> differentiate.
>>>>       - i.e. offer to the same customer a "gold" routing on IP1 and cheap
>>>> routing on IP2
>>>>
>>>> - Customer add/modify/delete
>>>>       - IP source
>>>>       - Rates for client routes based on areacode
>>>>       - Prepaid or postpaid.
>>>>            - When cutomer balance is 0, no more calls are allowed.
>>>>       - limit max channels
>>>>       - Media by-pass
>>>>            - When by-passed, customer and provider will exchange RTPs
>>>> directly. Else, server will be in the middle.
>>>>
>>>> - Provider add/modify/delete
>>>>       - costs for provider routes based on areacode
>>>>       - limit max channels
>>>>
>>>> - Routing based on areacode, gives great granularity.
>>>>
>>>> - Routes can be assigned multiple gateways/providers which can in turn be
>>>> distributed based on weigth. Includes overflow to next configured GW.
>>>>
>>>> - Basic financial report generation (totals) by customer/provider
>>>>
>>>> - Basic traffic ASR/ACD report (totals) by cutomer/provider
>>>>
>>>> - Basic user administration. (No access level, only total access)
>>>>
>>>> - CDR export to csv file.
>>>>
>>>>
>>>>
>>>>
>>>> I also have a prepaid card app... no web interface on that one though...
>>>>
>>>> Thanks all
>>>>
>>>>
>>>> David


From msc at freeswitch.org  Thu Feb 24 23:26:28 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 24 Feb 2011 12:26:28 -0800
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
Message-ID: <AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>

Very cool-sounding stuff. How's the documentation? I'd recommend putting it
in freeswitch-contrib and maintaining it there or doing a github as Avi
suggested. (You can even do both and just have one be sync'd to the other.)
If you have a github repo then you have additional infrastructure like docs
and bug tracking specifically for your project.

In any case, very well done! If you want freeswitch-contrib access then
contact intralanman at freeswitch.org and he'll get you all set up.

-MC

On Thu, Feb 24, 2011 at 3:39 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello Guys,
>
> I'm finishing a "complete" wholesale application created on freeswitch and
> I was wondering whether it would be a good idea to put it up on the wiki. I
> just don't know how.
>
> Features include all the following parameters configurable via web
> interface:
>
> - Profile creation based on server IP where traffic is received. You can
> have multiple IPs, system will create multiple profiles/diaplans so it can
> differentiate.
>      - i.e. offer to the same customer a "gold" routing on IP1 and cheap
> routing on IP2
>
> - Customer add/modify/delete
>      - IP source
>      - Rates for client routes based on areacode
>      - Prepaid or postpaid.
>           - When cutomer balance is 0, no more calls are allowed.
>      - limit max channels
>      - Media by-pass
>           - When by-passed, customer and provider will exchange RTPs
> directly. Else, server will be in the middle.
>
> - Provider add/modify/delete
>      - costs for provider routes based on areacode
>      - limit max channels
>
> - Routing based on areacode, gives great granularity.
>
> - Routes can be assigned multiple gateways/providers which can in turn be
> distributed based on weigth. Includes overflow to next configured GW.
>
> - Basic financial report generation (totals) by customer/provider
>
> - Basic traffic ASR/ACD report (totals) by cutomer/provider
>
> - Basic user administration. (No access level, only total access)
>
> - CDR export to csv file.
>
>
>
>
> I also have a prepaid card app... no web interface on that one though...
>
> 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 msc at freeswitch.org  Thu Feb 24 23:44:28 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 24 Feb 2011 12:44:28 -0800
Subject: [Freeswitch-users] Receive order guarantee for bgapi command
 replies over ESL?
In-Reply-To: <SNT107-DS120CD9CBAFDAA80E16A6CDE6DA0@phx.gbl>
References: <SNT107-DS120CD9CBAFDAA80E16A6CDE6DA0@phx.gbl>
Message-ID: <AANLkTikMhk3C+3Gc8ebf8B_O8kOnrxhBRNS4a6bH4Ux=@mail.gmail.com>

Just curious - what is the use case where sending a stack of bgapi's without
listening for the reply is more desirable than looping through the bgapi
send/reply sequence for each one? I'm no super programmer but it seems to me
that blindly relying on another system to always send stuff in the exact
correct order is dangerous and may cause bugs that are difficult to
diagnose. Personally I would consider it a programming best practice to wait
for the response of the bgapi before sending another one.

I invite other socket programmers to give their input...

-MC

P.S - you may wish to disable the legal notice at the and of your emails
when sending to a public list. (We know that some servers tag outgoing
messages automatically and if that is your case we understand. Go yell at
the I.T. guy! :)

2011/2/24 Mauritz L?vgren <mauritz.lovgren at hotmail.com>

>   Hi,
>
> Are replies for bgapi commands sent back to an inbound socket connection in
> the same order as the commands were sent?
>
> Example: sending 10 bgapi messages from client to freeswitch socket with no
> delay inbetween.
> Will the response for those command messages be sent back by freeswitch in
> the exact same order as their commands were received, or should one wait for
> each reply before sending a new bgapi command to be sure one gets a reply
> for the correct command?
>
> If there was an option of providing a UUID (or sequenceId) for the command,
> it would be easier to match the reply with the request upon receival, but
> this doesn?t seem to be possible with the current implementation?
>
> Regards,
> Mauritz Lovgren
> Systems Architect
> IPLink Inc.
>
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From msc at freeswitch.org  Thu Feb 24 23:46:53 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 24 Feb 2011 12:46:53 -0800
Subject: [Freeswitch-users] Using outbound Event Sockets versus using
 embed language scripts
In-Reply-To: <AANLkTi=vZzEZW2T6FqmKgDcABTt=Tw3NdcYrhzxnn-Wx@mail.gmail.com>
References: <AANLkTimyEsRcK02wzbKqWfjbXz-YJXHjSCTi6kXawufU@mail.gmail.com>
	<AANLkTi=m99i4r3Osud9x=dUv4PG0fE4w03n=JBVWvuNn@mail.gmail.com>
	<AANLkTi=vZzEZW2T6FqmKgDcABTt=Tw3NdcYrhzxnn-Wx@mail.gmail.com>
Message-ID: <AANLkTindk1Bq5U9v2u92emTw0caC=std6QXApff-wcE2@mail.gmail.com>

Using PAGD is the controlled way of collecting digits from the caller. You
could rely on the events from the event socket PAGD does a lot of the dirty
work for you, like handling invalid input, etc.

-MC

On Wed, Feb 23, 2011 at 7:49 PM, Herman Griffin <herman.griffin at gmail.com>wrote:

> I notice that there are DTMF events that come over the event socket. Is it
> best practice to use the play_and_get_digits dialplan app to collect digits
> or is collecting the digits using the DTMF events a better way?
>
> On Wed, Feb 23, 2011 at 8:26 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> my $e;
>> my $foobar;
>>
>> $e = $con->api("uuid_getvar", "$uuid foobar");
>>
>> if ($e) {
>>    $foobar = $e->getBody();
>> }
>>
>>
>> On Wed, Feb 23, 2011 at 1:39 AM, Herman Griffin
>> <herman.griffin at gmail.com> wrote:
>> > Unless I'm complicating things, it seems like a drop to write outbound
>> event
>> > sockets apps versus writing embedded language apps. However, I'm
>> interested
>> > in writing event sockets apps because I can see an advantage in being
>> able
>> > to load balance the app by running you app behind a load balancer. One
>> > simple thing that I am trying to accomplish is collecting digits and
>> doing
>> > something useful with string of digits that have been collected.
>> >
>> > My first attempt at this is to use the play_and_get_digits dptool, but I
>> > don't now how to pull data from stored variable so that I can use in the
>> > script. I slightly modified the freeswitch.git/libs/esl/perl/server2.pl
>> > script by adding this line:
>> >
>> > $con->execute('play_and_get_digits', '2 5 3 7000 #
>> > ${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav
>> /invalid.wav
>> > foobar \d+' );
>> >
>> > The next thing that I'd like to do is grab the value in foobar is use it
>> in
>> > the perl logic.
>> >
>> > Can someone lead me to the next step?
>> > Does anyone with experience with event socket apps and embedded language
>> > apps have some useful information about their preferred path?
>> >
>> > Thanks,
>> > Herman
>>
>
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From errotan at elder.hu  Thu Feb 24 23:52:02 2011
From: errotan at elder.hu (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Thu, 24 Feb 2011 21:52:02 +0100
Subject: [Freeswitch-users] Freeswtich as a media proxy between
	ipv4<=>ipv6 using Polycom HDX8006 SIP UA-s
In-Reply-To: <4D650823.5050305@niif.hu>
References: <4D650823.5050305@niif.hu>
Message-ID: <201102242152.02899.errotan@elder.hu>

Hi.

If you just want a sip proxy use OpenSER or Opensips.


2011. febru?r 23. 14:14:11 d?tummal M?SZ?ROS Mih?ly az al?bbiakat ?rta:
> Hi,
> 
>    1. 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)
>    2. 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?
>    3. 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


From msc at freeswitch.org  Thu Feb 24 23:51:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 24 Feb 2011 12:51:38 -0800
Subject: [Freeswitch-users] Message Playback Stops
In-Reply-To: <AANLkTik9tSO5gWA_bUEQKEbdD6YxG6GAvedWtBs21sxZ@mail.gmail.com>
References: <AANLkTi=EMQsSkH-SZDq2rUzvh2-7nYK=NpKHAXXEwc-W@mail.gmail.com>
	<AANLkTinpDt+NrcoK8pwWuuwMufut+DTC7GSrnAdWYhJ5@mail.gmail.com>
	<AANLkTi=O2R3oqcSDOUxh3g-mqKJ_E-rgrLZ210a_SeJi@mail.gmail.com>
	<AANLkTikSap1zEH58kVWVzpw_kWoWme26WQ1Y6CB3vnxm@mail.gmail.com>
	<AANLkTikCs4Uh1uuTwut=mvt4m7KLzwC-Y7bRedvrO-Te@mail.gmail.com>
	<AANLkTikDvE_Y93owBZ6_DHz75pNCdrKoocy6ywjjWLcF@mail.gmail.com>
	<AANLkTikvwZR5t8PaPnHVwNMqDRpipYndc+XbSzYzvjRH@mail.gmail.com>
	<AANLkTik9tSO5gWA_bUEQKEbdD6YxG6GAvedWtBs21sxZ@mail.gmail.com>
Message-ID: <AANLkTinC72Fn5VAWZDTDhYSwTVckGEVtL4i6p3UzAydf@mail.gmail.com>

On Wed, Feb 23, 2011 at 7:50 PM, Michael De Lorenzo <
delorenzodesign at gmail.com> wrote:

> Is there some place where I can get more detailed log information on this?
> I thought that streamFile was a blocking operation (the next statement
> wouldn't be executed until playback was completed), is that not the case?
>
In all my usage of streamFile it has been blocking. I would write a very
simple Lua script that plays a file that is at least several seconds long
and then put log output before and after the streamFile method is called.


>
> Is there a difference in terms of performance with Freeswitch for Lua
> versus SpiderMonkey?
>
Lua is definitely cleaner, lighter, and faster. There may be more docs on
using JavaScript but that's only because it was adopted early on in the
development of FS and there were more people familiar with it. Lua is
definitely the scripting language of choice when you are calling a script
from the dialplan.

-MC


>
>
> Am I misreading the log?
>>
>>
>> *2011-02-21 17:23:09.077014 [NOTICE] switch_cpp.cpp:1181 Playing file:
>> /usr/local/freeswitch/*
>> *recordings/messages/16c0f890_**c35e33c0_777973.wav*
>> 2011-02-21 17:23:09.077014 [DEBUG] switch_ivr_play_say.c:1186 Codec
>> Activated L16 at 8000hz 1 channels 20ms
>> 2011-02-21 17:23:19.417631 [DEBUG] switch_ivr_play_say.c:1515 done playing
>> file
>> 2011-02-21 17:23:19.417631 [NOTICE] switch_cpp.cpp:1181 !!!!! Finished
>> playing the file !!!!!
>> 2011-02-21 17:23:19.417631 [INFO] switch_cpp.cpp:1181 Was VM detected? no
>> 2011-02-21 17:23:19.417631 [NOTICE] switch_cpp.cpp:1181 Played the message
>> at least once and checked for VM, we should be exiting the loop
>>
>> The messages like "finished playing" are log commands that are only
>> available in the script after it's answered.
>>
>>
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From matzemuc86 at gmail.com  Thu Feb 24 23:48:40 2011
From: matzemuc86 at gmail.com (MatzeMuc86)
Date: Thu, 24 Feb 2011 21:48:40 +0100
Subject: [Freeswitch-users] implementing stereo (e.g. portaudio to test
	it easily)
In-Reply-To: <AANLkTinowzmzt8NxB-FtifkZEQZ8s3OQAWLuC18PN60Q@mail.gmail.com>
References: <AANLkTikL9SQPdsJ6XF-AUZuLQ3Z-AG7Eqgj2BsL=CZ4S@mail.gmail.com>
	<AANLkTinowzmzt8NxB-FtifkZEQZ8s3OQAWLuC18PN60Q@mail.gmail.com>
Message-ID: <AANLkTikDFqS_Y9XS-gnVvnfPhAW1Eri=zy4XB3vAS2N0@mail.gmail.com>

Are you sure?
I had a diskussion at the IRC dev channel and Grmt told me it should be
possible.
It is quite important for me and i do not ask somebody to implement this for
me.

Thanks
MatzeMuc86


2011/2/24 Michael Collins <msc at freeswitch.org>

> Sorry, FreeSWITCH is not set up to do stereo... perhaps one day, but it
> isn't something that is high on the priority list.
>
> -MC
>
> On Thu, Feb 24, 2011 at 9:22 AM, MatzeMuc86 <matzemuc86 at gmail.com> wrote:
>
>> hello FreeSWITCH community,
>>
>> actually it is my first time to use lists like this. I hope I do
>> everything fine.
>> For my project I try to get stereo (and some special features later)
>> integrated in freeSWITCH, especially SIP.
>> TO start with this idea I need to know where I have to activate or
>> implement stereo.
>> In the second I try to play a bit with the portaudio module as I do not
>> have to focus on SDP problems transporting SIP and the need of a SIP client
>> which is able to receive stereo.
>> So I set up some custom moh local media stuff which works finde - using
>> mono.
>> Changing the channels to 2 makes the console telling me that the sound
>> file will be mixed to mono and the sound is quite strange.
>> I would be happy about getting some useful information which modules or
>> configs are the important ones for my idea. Unfurtun ately I have only found
>> some recording stuff about stereo which is not what I want to do.
>>
>> Thank you very much.
>> MatzeMuc86
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing 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 mauritz.lovgren at hotmail.com  Fri Feb 25 01:15:40 2011
From: mauritz.lovgren at hotmail.com (=?Windows-1252?Q?Mauritz_L=F8vgren?=)
Date: Thu, 24 Feb 2011 23:15:40 +0100
Subject: [Freeswitch-users] Receive order guarantee for bgapi command
	replies over ESL?
In-Reply-To: <AANLkTikMhk3C+3Gc8ebf8B_O8kOnrxhBRNS4a6bH4Ux=@mail.gmail.com>
References: <SNT107-DS120CD9CBAFDAA80E16A6CDE6DA0@phx.gbl>
	<AANLkTikMhk3C+3Gc8ebf8B_O8kOnrxhBRNS4a6bH4Ux=@mail.gmail.com>
Message-ID: <SNT107-DS17CDAAB3CB7918E0D98882E6DA0@phx.gbl>

The use case is as follows;

Multi-threaded Java applicaton that sends bgapi commands to several channels in parallel over the same ESL connection.
(bgapi commands are serialized in order to the socket outputstream, but is there any guarantee that we will receive the OK / ERR reply for each bgapi in the correct order from the socket inputstream?)

And, what about sending bgapi commands on the same socket while a ?regular? api command is currently executing? I suspect the bgapi reply will delay until the ?regular? (blocking) api command is finished processing?

- Mauritz

From: Michael Collins 
Sent: Thursday, February 24, 2011 9:44 PM
To: FreeSWITCH Users Help 
Subject: Re: [Freeswitch-users] Receive order guarantee for bgapi command replies over ESL?

Just curious - what is the use case where sending a stack of bgapi's without listening for the reply is more desirable than looping through the bgapi send/reply sequence for each one? I'm no super programmer but it seems to me that blindly relying on another system to always send stuff in the exact correct order is dangerous and may cause bugs that are difficult to diagnose. Personally I would consider it a programming best practice to wait for the response of the bgapi before sending another one. 

I invite other socket programmers to give their input...

-MC

P.S - you may wish to disable the legal notice at the and of your emails when sending to a public list. (We know that some servers tag outgoing messages automatically and if that is your case we understand. Go yell at the I.T. guy! :)


2011/2/24 Mauritz L?vgren <mauritz.lovgren at hotmail.com>

  Hi,

  Are replies for bgapi commands sent back to an inbound socket connection in the same order as the commands were sent?

  Example: sending 10 bgapi messages from client to freeswitch socket with no delay inbetween.
  Will the response for those command messages be sent back by freeswitch in the exact same order as their commands were received, or should one wait for each reply before sending a new bgapi command to be sure one gets a reply for the correct command?

  If there was an option of providing a UUID (or sequenceId) for the command, it would be easier to match the reply with the request upon receival, but this doesn?t seem to be possible with the current implementation?

  Regards,
  Mauritz Lovgren
  Systems Architect
  IPLink Inc.



--------------------------------------------------------------------------------
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From anthony.minessale at gmail.com  Fri Feb 25 01:22:38 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 24 Feb 2011 16:22:38 -0600
Subject: [Freeswitch-users] Message Playback Stops
In-Reply-To: <AANLkTinC72Fn5VAWZDTDhYSwTVckGEVtL4i6p3UzAydf@mail.gmail.com>
References: <AANLkTi=EMQsSkH-SZDq2rUzvh2-7nYK=NpKHAXXEwc-W@mail.gmail.com>
	<AANLkTinpDt+NrcoK8pwWuuwMufut+DTC7GSrnAdWYhJ5@mail.gmail.com>
	<AANLkTi=O2R3oqcSDOUxh3g-mqKJ_E-rgrLZ210a_SeJi@mail.gmail.com>
	<AANLkTikSap1zEH58kVWVzpw_kWoWme26WQ1Y6CB3vnxm@mail.gmail.com>
	<AANLkTikCs4Uh1uuTwut=mvt4m7KLzwC-Y7bRedvrO-Te@mail.gmail.com>
	<AANLkTikDvE_Y93owBZ6_DHz75pNCdrKoocy6ywjjWLcF@mail.gmail.com>
	<AANLkTikvwZR5t8PaPnHVwNMqDRpipYndc+XbSzYzvjRH@mail.gmail.com>
	<AANLkTik9tSO5gWA_bUEQKEbdD6YxG6GAvedWtBs21sxZ@mail.gmail.com>
	<AANLkTinC72Fn5VAWZDTDhYSwTVckGEVtL4i6p3UzAydf@mail.gmail.com>
Message-ID: <AANLkTi=3HCZMeMGasKu8hgOJ7zQiuVq0CYy4GnG=A-43@mail.gmail.com>

did you bother to try what I suggested with ignore_early_media=true?


On Thu, Feb 24, 2011 at 2:51 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Wed, Feb 23, 2011 at 7:50 PM, Michael De Lorenzo
> <delorenzodesign at gmail.com> wrote:
>>
>> Is there some place where I can get more detailed log information on
>> this?? I thought that streamFile was a blocking operation (the next
>> statement wouldn't be executed until playback was completed), is that not
>> the case?
>
> In all my usage of streamFile it has been blocking. I would write a very
> simple Lua script that plays a file that is at least several seconds long
> and then put log output before and after the streamFile method is called.
>
>>
>> Is there a difference in terms of performance with Freeswitch for Lua
>> versus SpiderMonkey?
>
> Lua is definitely cleaner, lighter, and faster. There may be more docs on
> using JavaScript but that's only because it was adopted early on in the
> development of FS and there were more people familiar with it. Lua is
> definitely the scripting language of choice when you are calling a script
> from the dialplan.
> -MC
>
>>
>>> Am I misreading the log?
>>>
>>> 2011-02-21 17:23:09.077014 [NOTICE] switch_cpp.cpp:1181 Playing file:
>>> /usr/local/freeswitch/
>>> recordings/messages/16c0f890_c35e33c0_777973.wav
>>> 2011-02-21 17:23:09.077014 [DEBUG] switch_ivr_play_say.c:1186 Codec
>>> Activated L16 at 8000hz 1 channels 20ms
>>> 2011-02-21 17:23:19.417631 [DEBUG] switch_ivr_play_say.c:1515 done
>>> playing file
>>> 2011-02-21 17:23:19.417631 [NOTICE] switch_cpp.cpp:1181 !!!!! Finished
>>> playing the file !!!!!
>>> 2011-02-21 17:23:19.417631 [INFO] switch_cpp.cpp:1181 Was VM detected? no
>>> 2011-02-21 17:23:19.417631 [NOTICE] switch_cpp.cpp:1181 Played the
>>> message at least once and checked for VM, we should be exiting the loop
>>> The messages like "finished playing" are log commands that are only
>>> available in the script after it's answered.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Fri Feb 25 01:37:08 2011
From: jjj at 3js.de (Johannes Jakob)
Date: Thu, 24 Feb 2011 23:37:08 +0100
Subject: [Freeswitch-users] T.38 Issues with passthrough handshaking
In-Reply-To: <640F91CF-9489-4F74-96B1-A6398CD84F07@3js.de>
References: <640F91CF-9489-4F74-96B1-A6398CD84F07@3js.de>
Message-ID: <A53FD340-169C-4DD4-8AF2-C463F9D9E3DD@3js.de>

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



From mbsip at gazeta.pl  Fri Feb 25 01:58:39 2011
From: mbsip at gazeta.pl (Mac)
Date: Thu, 24 Feb 2011 23:58:39 +0100
Subject: [Freeswitch-users] RTP Proxy mode, DTMF conversion problem
In-Reply-To: <AANLkTikH6EKOMzYriW=egWA-dFo0-KckRY=Qsxn-FgoL@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>
Message-ID: <AANLkTik3=WPdnj_WpE-vK8b8SfEp5NXxyXonL7YT+UN=@mail.gmail.com>

Hi,

Here they are:
http://pastebin.freeswitch.org/15473

Thanks,
Mac

> Get a debug-level log with siptrace enabled and paste it so we can see
> what's going on.
>
> -Steve
>
>
> On 22 February 2011 01:16, Mac <mbsip at gazeta.pl> wrote:
>>
>> Hello,
>>
>> Could somebody give me a hint?
>>
>> Thanks,
>> Mac
>>
>> > Hi Steven,
>> >
>> > There is much better :)
>> > I have different SDPs right now. The one is with rtpmap:101
>> > telephone-event/8000 (leg that is working with RFC2833), the opposite
>> > one incorporates a=rtpmap:101 telephone-event/8000 in INVITE SDP and
>> > pure codec rtpmaps.
>> > There is rtp.p_type == 101 working only on the left side. I cannot
>> > find any rtp.p_type == 101 on thark in opposide side which is okay.
>> > But the problem still persists.
>> >
>> > Once setting my UA to work with Inband DTMF everything works fine.
>> >
>> > Thanks,
>> > Mac
>> >
>> > 2011/2/20 Steven Ayre <steveayre at gmail.com>:
>> >> <action application="set" data="proxy_media=true"/>
>> >>
>> >> As indicated by the error, this is the problem. "proxy passthrough". In
>> >> proxy mode you pass the media straight through (passthrough). You can't
>> >> process passthrough media, such as is needed to mix in inband dtmf.
>> >> Use proxy_media=false and bypass_media=false. That's actually the
>> >> default so
>> >> unless you're setting either to true in the sip profile, you can remove
>> >> those lines from the dialplan completely.
>> >> Steve on iPhone
>> >> On 20 Feb 2011, at 18:51, Mac <mbsip at gazeta.pl> wrote:
>> >>
>> >> action application="set" data="proxy_media=true"/>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Fri Feb 25 02:04:11 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 24 Feb 2011 17:04:11 -0600
Subject: [Freeswitch-users] T.38 Issues with passthrough handshaking
In-Reply-To: <A53FD340-169C-4DD4-8AF2-C463F9D9E3DD@3js.de>
References: <640F91CF-9489-4F74-96B1-A6398CD84F07@3js.de>
	<A53FD340-169C-4DD4-8AF2-C463F9D9E3DD@3js.de>
Message-ID: <AANLkTi=GmF93TkQPNuFf9ko_eQVgvqwujZOo8duN8ZDB@mail.gmail.com>

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

<!-- you don't mix t38_passthru with proxy media get rid of these
completely  -->
<!--     <action application="set" data="proxy_media=true"/>
     <action application="set" data="bypass_media=false"/>-->

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

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.



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


From anthony.minessale at gmail.com  Fri Feb 25 02:07:50 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 24 Feb 2011 17:07:50 -0600
Subject: [Freeswitch-users] RTP Proxy mode, DTMF conversion problem
In-Reply-To: <AANLkTik3=WPdnj_WpE-vK8b8SfEp5NXxyXonL7YT+UN=@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>
Message-ID: <AANLkTin4LE4pUBv0cLsFnm=S86Nsx6Lj9jMEvLGk+dX5@mail.gmail.com>

you need to run the start_dtmf() application on the leg that has inband DTMF

if its the inbound leg, add it to the DP
<action application="start_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"/>


On Thu, Feb 24, 2011 at 4:58 PM, Mac <mbsip at gazeta.pl> wrote:
> Hi,
>
> Here they are:
> http://pastebin.freeswitch.org/15473
>
> Thanks,
> Mac
>
>> Get a debug-level log with siptrace enabled and paste it so we can see
>> what's going on.
>>
>> -Steve
>>
>>
>> On 22 February 2011 01:16, Mac <mbsip at gazeta.pl> wrote:
>>>
>>> Hello,
>>>
>>> Could somebody give me a hint?
>>>
>>> Thanks,
>>> Mac
>>>
>>> > Hi Steven,
>>> >
>>> > There is much better :)
>>> > I have different SDPs right now. The one is with rtpmap:101
>>> > telephone-event/8000 (leg that is working with RFC2833), the opposite
>>> > one incorporates a=rtpmap:101 telephone-event/8000 in INVITE SDP and
>>> > pure codec rtpmaps.
>>> > There is rtp.p_type == 101 working only on the left side. I cannot
>>> > find any rtp.p_type == 101 on thark in opposide side which is okay.
>>> > But the problem still persists.
>>> >
>>> > Once setting my UA to work with Inband DTMF everything works fine.
>>> >
>>> > Thanks,
>>> > Mac
>>> >
>>> > 2011/2/20 Steven Ayre <steveayre at gmail.com>:
>>> >> <action application="set" data="proxy_media=true"/>
>>> >>
>>> >> As indicated by the error, this is the problem. "proxy passthrough". In
>>> >> proxy mode you pass the media straight through (passthrough). You can't
>>> >> process passthrough media, such as is needed to mix in inband dtmf.
>>> >> Use proxy_media=false and bypass_media=false. That's actually the
>>> >> default so
>>> >> unless you're setting either to true in the sip profile, you can remove
>>> >> those lines from the dialplan completely.
>>> >> Steve on iPhone
>>> >> On 20 Feb 2011, at 18:51, Mac <mbsip at gazeta.pl> wrote:
>>> >>
>>> >> action application="set" data="proxy_media=true"/>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >>
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Fri Feb 25 02:10:06 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 24 Feb 2011 17:10:06 -0600
Subject: [Freeswitch-users] Receive order guarantee for bgapi command
 replies over ESL?
In-Reply-To: <SNT107-DS17CDAAB3CB7918E0D98882E6DA0@phx.gbl>
References: <SNT107-DS120CD9CBAFDAA80E16A6CDE6DA0@phx.gbl>
	<AANLkTikMhk3C+3Gc8ebf8B_O8kOnrxhBRNS4a6bH4Ux=@mail.gmail.com>
	<SNT107-DS17CDAAB3CB7918E0D98882E6DA0@phx.gbl>
Message-ID: <AANLkTinMLusDxVJ8bm+DEPB4OqrPj292wNbYR_wWF3j8@mail.gmail.com>

bgapi commands are asynchronous that means when you send the command
it spawns a thread and runs in the background.
When its complete it will send you the event, its completely dependent
on how long the task takes that you asked it to run.
Its no different that doing 12 wgets to a website in the bg, there is
no telling when they will be done.



2011/2/24 Mauritz L?vgren <mauritz.lovgren at hotmail.com>:
> The use case is as follows;
>
> Multi-threaded Java applicaton that sends bgapi commands to several channels
> in parallel over the same ESL connection.
> (bgapi commands are serialized in order to the socket outputstream, but is
> there any guarantee that we will receive the OK / ERR reply for each bgapi
> in the correct order from the socket inputstream?)
>
> And, what about sending bgapi commands on the same socket while a ?regular?
> api command is currently executing? I suspect the bgapi reply will delay
> until the ?regular? (blocking) api command is finished processing?
>
> - Mauritz
>
> From: Michael Collins
> Sent: Thursday, February 24, 2011 9:44 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Receive order guarantee for bgapi command
> replies over ESL?
>
> Just curious - what is the use case where sending a stack of bgapi's without
> listening for the reply is more desirable than looping through the bgapi
> send/reply sequence for each one? I'm no super programmer but it seems to me
> that blindly relying on another system to always send stuff in the exact
> correct order is dangerous and may cause bugs that are difficult to
> diagnose. Personally I would consider it a programming best practice to wait
> for the response of the bgapi before sending another one.
>
> I invite other socket programmers to give their input...
>
> -MC
>
> P.S - you may wish to disable the legal notice at the and of your emails
> when sending to a public list. (We know that some servers tag outgoing
> messages automatically and if that is your case we understand. Go yell at
> the I.T. guy! :)
>
> 2011/2/24 Mauritz L?vgren <mauritz.lovgren at hotmail.com>
>>
>> Hi,
>>
>> Are replies for bgapi commands sent back to an inbound socket connection
>> in the same order as the commands were sent?
>>
>> Example: sending 10 bgapi messages from client to freeswitch socket with
>> no delay inbetween.
>> Will the response for those command messages be sent back by freeswitch in
>> the exact same order as their commands were received, or should one wait for
>> each reply before sending a new bgapi command to be sure one gets a reply
>> for the correct command?
>>
>> If there was an option of providing a UUID (or sequenceId) for the
>> command, it would be easier to match the reply with the request upon
>> receival, but this doesn?t seem to be possible with the current
>> implementation?
>>
>> Regards,
>> Mauritz Lovgren
>> Systems Architect
>> IPLink Inc.
>
>
>
> ________________________________
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb 25 02:11:34 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 24 Feb 2011 17:11:34 -0600
Subject: [Freeswitch-users] Second transfer
In-Reply-To: <AANLkTinw4CWxe+GW31AcMnzWWNsQU=W7WGSmsr2ky9zg@mail.gmail.com>
References: <AANLkTi=pB-Rt7j1cvNGgSs2s3F4h6jse1nTqTqRr8kKJ@mail.gmail.com>
	<AANLkTi=fvkp_8e4gXthWtYQTLAyxj+bbUMtdpAhXQamQ@mail.gmail.com>
	<AANLkTinw4CWxe+GW31AcMnzWWNsQU=W7WGSmsr2ky9zg@mail.gmail.com>
Message-ID: <AANLkTikWMF2rf=ASaTMg6bQ=jw7KQu59wJuXr+fRj0Rg@mail.gmail.com>

add the 1 flag to the flags
ction application="bind_meta_app" data="3 b s1 log::INFO> Transfer"/>

this says it can only work once then detaches itself.



On Thu, Feb 24, 2011 at 12:22 PM, Santiago Soares
<santiagosoares at gmail.com> wrote:
> Thanks for your answer, but it doesn't work.
> I tried to assign *3 to log, like this:
>
> ?? <extension name="transferencia2">
> ????? <condition field="${attxfer_callthis}" expression="^numberC$">
> ???????? <action application="bind_meta_app" data="3 b s log::INFO
> Transfer"/>
> ???????? <action application="att_xfer" data="sofia/gateway/test/loginC"/>
> ????? </condition>
> ?? </extension>
> But *3 still calls the transferencia extension.
> Any ideas?
>
> Thank you very much!
>
> Santiago Soares
> Fone: (41) 8488-0537
>
>
> On Wed, Feb 23, 2011 at 8:20 PM, Michael Collins <msc at freeswitch.org> wrote:
>>
>> I think you can make another call to bind_meta_app and disable the *3
>> prior to the att_xfer. You could make *3 do something completely different
>> (like trigger a log entry) and then you could see how many times your users
>> tried to use *3 when they weren't supposed to. :)
>> -MC
>>
>> On Wed, Feb 23, 2011 at 10:01 AM, Santiago Soares
>> <santiagosoares at gmail.com> wrote:
>>>
>>> Hello,
>>>
>>> I'm using att_xfer to make call transfer, like that:
>>> A calls B
>>> B calls C
>>> B hangup and A is bridged do C
>>>
>>> The thing is, the way I am trying to do it, C is able to transfer the
>>> call again.
>>> I don't want that C be able to transfer the call again.
>>> The call should be transfered only once.
>>> Is it possible?
>>>
>>> This is my dialplan:
>>>
>>> <include>
>>>
>>> ?? <extension name="test_extension">
>>> ????? <condition field="destination_number" expression="^numberB$">
>>> ???????? <action application="set" data="bind_meta_key=*"/>
>>> ???????? <action application="bind_meta_app" data="3 b s
>>> execute_extension::transferencia XML public"/>
>>> ???????? <action application="bridge" data="sofia/gateway/test/loginB"/>
>>> ????? </condition>
>>> ?? </extension>
>>>
>>> ?? <extension name="transferencia">
>>> ????? <condition field="destination_number" expression="^transferencia$">
>>> ???????? <action application="read" data="3 3 base256/16000/sound.wav
>>> attxfer_callthis 30000 #"/>
>>> ???????? <action application="execute_extension" data="transferencia2"/>
>>> ????? </condition>
>>> ?? </extension>
>>>
>>> ?? <extension name="transferencia2">
>>> ????? <condition field="${attxfer_callthis}" expression="^numberC$">
>>> ???????? <action application="att_xfer"
>>> data="sofia/gateway/test/loginC"/>
>>> ????? </condition>
>>> ????? <condition field="${attxfer_callthis}" expression="^numberD$">
>>> ???????? <action application="att_xfer"
>>> data="sofia/gateway/test/loginD"/>
>>> ????? </condition>
>>> ?? </extension>
>>>
>>> </include>
>>>
>>>
>>> Thank you,
>>> Santiago Soares
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mauritz.lovgren at hotmail.com  Fri Feb 25 02:27:48 2011
From: mauritz.lovgren at hotmail.com (=?Windows-1252?Q?Mauritz_L=F8vgren?=)
Date: Fri, 25 Feb 2011 00:27:48 +0100
Subject: [Freeswitch-users] Receive order guarantee for bgapi command
	replies over ESL?
In-Reply-To: <AANLkTinMLusDxVJ8bm+DEPB4OqrPj292wNbYR_wWF3j8@mail.gmail.com>
References: <SNT107-DS120CD9CBAFDAA80E16A6CDE6DA0@phx.gbl><AANLkTikMhk3C+3Gc8ebf8B_O8kOnrxhBRNS4a6bH4Ux=@mail.gmail.com><SNT107-DS17CDAAB3CB7918E0D98882E6DA0@phx.gbl>
	<AANLkTinMLusDxVJ8bm+DEPB4OqrPj292wNbYR_wWF3j8@mail.gmail.com>
Message-ID: <SNT107-DS147A2767EEB9599E895DD7E6DA0@phx.gbl>

Clarification:

Yes, the event returned from the background job is expceted to be fully 
asynchronous and can arrive at any time in the future (independent of when 
the bgapi commands were issued).
The issue is the reply from the bgapi command itself, _before_ it spawns the 
background thread.
If you, let's say, issue 10 bgapi commands in sequence, will the reply (OK 
Job-UUID:.... / ERR) for each bgapi command return in the same sequence as 
issued?

Example:

ESL socket outputstream (writer thread):
client --> bgapi status --> fs
client --> bgapi status --> fs
client --> bgapi status --> fs

ESL socket inputstream (reader thread):
client <-- +OK Job-UUID:..... <-- fs
client <-- +OK Job-UUID:..... <-- fs
client <-- +OK Job-UUID:..... <-- fs

The ESL socket outputstream and inputstream are written to and read using a 
writer and a reader thread. If we have no guarantee that the order of the OK 
replies from the bgapi commands are returned in the same order as issued, we 
will be forced to do the following:

client --> bgapi status --> fs
wait for reply
client <-- +OK Job-UUID:..... <-- fs

client --> bgapi status --> fs
wait for reply
client <-- +OK Job-UUID:..... <-- fs

client --> bgapi status --> fs
wait for reply
client <-- +OK Job-UUID:..... <-- fs

Only then could we securly map the command to the correct Job-UUID for the 
reply.
But if the replies arrive in correct order we can put the issued commands in 
an ordered queue that can be processed with certainty by the socket reader 
thread, and assign replies to the correct commands.. Did this really make it 
any clearer?? :-)

Had there been a message sequence ID (or similar construct) there would be 
no need to be order specific as we would always have an exact match between 
command and reply, but I don't think FS / CLI / ESL supports this at the 
moment (or ever will... ;-)).

- Mauritz


-----Original Message----- 
From: Anthony Minessale
Sent: Friday, February 25, 2011 12:10 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Receive order guarantee for bgapi command 
replies over ESL?

bgapi commands are asynchronous that means when you send the command
it spawns a thread and runs in the background.
When its complete it will send you the event, its completely dependent
on how long the task takes that you asked it to run.
Its no different that doing 12 wgets to a website in the bg, there is
no telling when they will be done.



2011/2/24 Mauritz L?vgren <mauritz.lovgren at hotmail.com>:
> The use case is as follows;
>
> Multi-threaded Java applicaton that sends bgapi commands to several 
> channels
> in parallel over the same ESL connection.
> (bgapi commands are serialized in order to the socket outputstream, but is
> there any guarantee that we will receive the OK / ERR reply for each bgapi
> in the correct order from the socket inputstream?)
>
> And, what about sending bgapi commands on the same socket while a ?regular?
> api command is currently executing? I suspect the bgapi reply will delay
> until the ?regular? (blocking) api command is finished processing?
>
> - Mauritz
>
> From: Michael Collins
> Sent: Thursday, February 24, 2011 9:44 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Receive order guarantee for bgapi command
> replies over ESL?
>
> Just curious - what is the use case where sending a stack of bgapi's 
> without
> listening for the reply is more desirable than looping through the bgapi
> send/reply sequence for each one? I'm no super programmer but it seems to 
> me
> that blindly relying on another system to always send stuff in the exact
> correct order is dangerous and may cause bugs that are difficult to
> diagnose. Personally I would consider it a programming best practice to 
> wait
> for the response of the bgapi before sending another one.
>
> I invite other socket programmers to give their input...
>
> -MC
>
> P.S - you may wish to disable the legal notice at the and of your emails
> when sending to a public list. (We know that some servers tag outgoing
> messages automatically and if that is your case we understand. Go yell at
> the I.T. guy! :)
>
> 2011/2/24 Mauritz L?vgren <mauritz.lovgren at hotmail.com>
>>
>> Hi,
>>
>> Are replies for bgapi commands sent back to an inbound socket connection
>> in the same order as the commands were sent?
>>
>> Example: sending 10 bgapi messages from client to freeswitch socket with
>> no delay inbetween.
>> Will the response for those command messages be sent back by freeswitch 
>> in
>> the exact same order as their commands were received, or should one wait 
>> for
>> each reply before sending a new bgapi command to be sure one gets a reply
>> for the correct command?
>>
>> If there was an option of providing a UUID (or sequenceId) for the
>> command, it would be easier to match the reply with the request upon
>> receival, but this doesn?t seem to be possible with the current
>> implementation?
>>
>> Regards,
>> Mauritz Lovgren
>> Systems Architect
>> IPLink Inc.
>
>
>
> ________________________________
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Fri Feb 25 02:54:49 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 24 Feb 2011 17:54:49 -0600
Subject: [Freeswitch-users] Receive order guarantee for bgapi command
 replies over ESL?
In-Reply-To: <SNT107-DS147A2767EEB9599E895DD7E6DA0@phx.gbl>
References: <SNT107-DS120CD9CBAFDAA80E16A6CDE6DA0@phx.gbl>
	<AANLkTikMhk3C+3Gc8ebf8B_O8kOnrxhBRNS4a6bH4Ux=@mail.gmail.com>
	<SNT107-DS17CDAAB3CB7918E0D98882E6DA0@phx.gbl>
	<AANLkTinMLusDxVJ8bm+DEPB4OqrPj292wNbYR_wWF3j8@mail.gmail.com>
	<SNT107-DS147A2767EEB9599E895DD7E6DA0@phx.gbl>
Message-ID: <AANLkTimPh0eqMz=JaNxV6hLBrG1oLVtYtE+4czuxV=KF@mail.gmail.com>

When you send bgapi command the ok with the job-uuid will always be
for the job you just submitted.
When you send any command you are required to wait for the reply to
that command, if it's bgapi you will get the job-uuid.
The only problem you face is if you are also listening for events you
may get an event in response to your request which you must save in a
queue and keep reading until you get the reply to the command you
issued.

If you are using the ESL library supplied with FreeSWITCH this is all
done for you.

As for the sequence ID question, you can supply your own job-uuid
header when you send the bgapi-command

bgapi show channels
job-uuid: 1234

This was not available from the script mods when I checked so I added
it to latest build.
example in perl but true in all swigged langs.

my $command = shift;
my $args = join(" ", @ARGV);

my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");

my $e = $con->bgapi($command, $args, "my-job-id");
print $e->serialize("json");


returns:

{
        "Event-Name":   "SOCKET_DATA",
        "Content-Type": "command/reply",
        "Reply-Text":   "+OK Job-UUID: my-job-id",
        "Job-UUID":     "my-job-id"
}









2011/2/24 Mauritz L?vgren <mauritz.lovgren at hotmail.com>:
> Clarification:
>
> Yes, the event returned from the background job is expceted to be fully
> asynchronous and can arrive at any time in the future (independent of when
> the bgapi commands were issued).
> The issue is the reply from the bgapi command itself, _before_ it spawns the
> background thread.
> If you, let's say, issue 10 bgapi commands in sequence, will the reply (OK
> Job-UUID:.... / ERR) for each bgapi command return in the same sequence as
> issued?
>
> Example:
>
> ESL socket outputstream (writer thread):
> client --> bgapi status --> fs
> client --> bgapi status --> fs
> client --> bgapi status --> fs
>
> ESL socket inputstream (reader thread):
> client <-- +OK Job-UUID:..... <-- fs
> client <-- +OK Job-UUID:..... <-- fs
> client <-- +OK Job-UUID:..... <-- fs
>
> The ESL socket outputstream and inputstream are written to and read using a
> writer and a reader thread. If we have no guarantee that the order of the OK
> replies from the bgapi commands are returned in the same order as issued, we
> will be forced to do the following:
>
> client --> bgapi status --> fs
> wait for reply
> client <-- +OK Job-UUID:..... <-- fs
>
> client --> bgapi status --> fs
> wait for reply
> client <-- +OK Job-UUID:..... <-- fs
>
> client --> bgapi status --> fs
> wait for reply
> client <-- +OK Job-UUID:..... <-- fs
>
> Only then could we securly map the command to the correct Job-UUID for the
> reply.
> But if the replies arrive in correct order we can put the issued commands in
> an ordered queue that can be processed with certainty by the socket reader
> thread, and assign replies to the correct commands.. Did this really make it
> any clearer?? :-)
>
> Had there been a message sequence ID (or similar construct) there would be
> no need to be order specific as we would always have an exact match between
> command and reply, but I don't think FS / CLI / ESL supports this at the
> moment (or ever will... ;-)).
>
> - Mauritz
>
>
> -----Original Message-----
> From: Anthony Minessale
> Sent: Friday, February 25, 2011 12:10 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Receive order guarantee for bgapi command
> replies over ESL?
>
> bgapi commands are asynchronous that means when you send the command
> it spawns a thread and runs in the background.
> When its complete it will send you the event, its completely dependent
> on how long the task takes that you asked it to run.
> Its no different that doing 12 wgets to a website in the bg, there is
> no telling when they will be done.
>
>
>
> 2011/2/24 Mauritz L?vgren <mauritz.lovgren at hotmail.com>:
>> The use case is as follows;
>>
>> Multi-threaded Java applicaton that sends bgapi commands to several
>> channels
>> in parallel over the same ESL connection.
>> (bgapi commands are serialized in order to the socket outputstream, but is
>> there any guarantee that we will receive the OK / ERR reply for each bgapi
>> in the correct order from the socket inputstream?)
>>
>> And, what about sending bgapi commands on the same socket while a ?regular?
>> api command is currently executing? I suspect the bgapi reply will delay
>> until the ?regular? (blocking) api command is finished processing?
>>
>> - Mauritz
>>
>> From: Michael Collins
>> Sent: Thursday, February 24, 2011 9:44 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Receive order guarantee for bgapi command
>> replies over ESL?
>>
>> Just curious - what is the use case where sending a stack of bgapi's
>> without
>> listening for the reply is more desirable than looping through the bgapi
>> send/reply sequence for each one? I'm no super programmer but it seems to
>> me
>> that blindly relying on another system to always send stuff in the exact
>> correct order is dangerous and may cause bugs that are difficult to
>> diagnose. Personally I would consider it a programming best practice to
>> wait
>> for the response of the bgapi before sending another one.
>>
>> I invite other socket programmers to give their input...
>>
>> -MC
>>
>> P.S - you may wish to disable the legal notice at the and of your emails
>> when sending to a public list. (We know that some servers tag outgoing
>> messages automatically and if that is your case we understand. Go yell at
>> the I.T. guy! :)
>>
>> 2011/2/24 Mauritz L?vgren <mauritz.lovgren at hotmail.com>
>>>
>>> Hi,
>>>
>>> Are replies for bgapi commands sent back to an inbound socket connection
>>> in the same order as the commands were sent?
>>>
>>> Example: sending 10 bgapi messages from client to freeswitch socket with
>>> no delay inbetween.
>>> Will the response for those command messages be sent back by freeswitch
>>> in
>>> the exact same order as their commands were received, or should one wait
>>> for
>>> each reply before sending a new bgapi command to be sure one gets a reply
>>> for the correct command?
>>>
>>> If there was an option of providing a UUID (or sequenceId) for the
>>> command, it would be easier to match the reply with the request upon
>>> receival, but this doesn?t seem to be possible with the current
>>> implementation?
>>>
>>> Regards,
>>> Mauritz Lovgren
>>> Systems Architect
>>> IPLink Inc.
>>
>>
>>
>> ________________________________
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 mauritz.lovgren at hotmail.com  Fri Feb 25 03:16:46 2011
From: mauritz.lovgren at hotmail.com (=?Windows-1252?Q?Mauritz_L=F8vgren?=)
Date: Fri, 25 Feb 2011 01:16:46 +0100
Subject: [Freeswitch-users] Receive order guarantee for bgapi command
	replies over ESL?
In-Reply-To: <AANLkTimPh0eqMz=JaNxV6hLBrG1oLVtYtE+4czuxV=KF@mail.gmail.com>
References: <SNT107-DS120CD9CBAFDAA80E16A6CDE6DA0@phx.gbl><AANLkTikMhk3C+3Gc8ebf8B_O8kOnrxhBRNS4a6bH4Ux=@mail.gmail.com><SNT107-DS17CDAAB3CB7918E0D98882E6DA0@phx.gbl><AANLkTinMLusDxVJ8bm+DEPB4OqrPj292wNbYR_wWF3j8@mail.gmail.com><SNT107-DS147A2767EEB9599E895DD7E6DA0@phx.gbl>
	<AANLkTimPh0eqMz=JaNxV6hLBrG1oLVtYtE+4czuxV=KF@mail.gmail.com>
Message-ID: <SNT107-DS215A438073C211F4E0241FE6DD0@phx.gbl>

The ability to specify the job-uuid is just what we need!
This simplifies the bgapi command / reply / background event mechanism we 
use.
We are currently using the Java ESL library from David Varnes as we run 
freeswitch on various platforms (Windows, FreeBSD, Mac etc etc) and wanted 
to avoid using JNI in our Java application, but we will consider using the 
swig ESL library instead if this proves to be a better solution.

I am running on the latest build now, so will test using self-made job-uuids 
(will probably use UUIDs since they are really unique :-)).

Thank you for the information and keep up the great work!

Mauritz

-----Original Message----- 
From: Anthony Minessale
Sent: Friday, February 25, 2011 12:54 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Receive order guarantee for bgapi command 
replies over ESL?

When you send bgapi command the ok with the job-uuid will always be
for the job you just submitted.
When you send any command you are required to wait for the reply to
that command, if it's bgapi you will get the job-uuid.
The only problem you face is if you are also listening for events you
may get an event in response to your request which you must save in a
queue and keep reading until you get the reply to the command you
issued.

If you are using the ESL library supplied with FreeSWITCH this is all
done for you.

As for the sequence ID question, you can supply your own job-uuid
header when you send the bgapi-command

bgapi show channels
job-uuid: 1234

This was not available from the script mods when I checked so I added
it to latest build.
example in perl but true in all swigged langs.

my $command = shift;
my $args = join(" ", @ARGV);

my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");

my $e = $con->bgapi($command, $args, "my-job-id");
print $e->serialize("json");


returns:

{
        "Event-Name":   "SOCKET_DATA",
        "Content-Type": "command/reply",
        "Reply-Text":   "+OK Job-UUID: my-job-id",
        "Job-UUID":     "my-job-id"
}









2011/2/24 Mauritz L?vgren <mauritz.lovgren at hotmail.com>:
> Clarification:
>
> Yes, the event returned from the background job is expceted to be fully
> asynchronous and can arrive at any time in the future (independent of when
> the bgapi commands were issued).
> The issue is the reply from the bgapi command itself, _before_ it spawns 
> the
> background thread.
> If you, let's say, issue 10 bgapi commands in sequence, will the reply (OK
> Job-UUID:.... / ERR) for each bgapi command return in the same sequence as
> issued?
>
> Example:
>
> ESL socket outputstream (writer thread):
> client --> bgapi status --> fs
> client --> bgapi status --> fs
> client --> bgapi status --> fs
>
> ESL socket inputstream (reader thread):
> client <-- +OK Job-UUID:..... <-- fs
> client <-- +OK Job-UUID:..... <-- fs
> client <-- +OK Job-UUID:..... <-- fs
>
> The ESL socket outputstream and inputstream are written to and read using 
> a
> writer and a reader thread. If we have no guarantee that the order of the 
> OK
> replies from the bgapi commands are returned in the same order as issued, 
> we
> will be forced to do the following:
>
> client --> bgapi status --> fs
> wait for reply
> client <-- +OK Job-UUID:..... <-- fs
>
> client --> bgapi status --> fs
> wait for reply
> client <-- +OK Job-UUID:..... <-- fs
>
> client --> bgapi status --> fs
> wait for reply
> client <-- +OK Job-UUID:..... <-- fs
>
> Only then could we securly map the command to the correct Job-UUID for the
> reply.
> But if the replies arrive in correct order we can put the issued commands 
> in
> an ordered queue that can be processed with certainty by the socket reader
> thread, and assign replies to the correct commands.. Did this really make 
> it
> any clearer?? :-)
>
> Had there been a message sequence ID (or similar construct) there would be
> no need to be order specific as we would always have an exact match 
> between
> command and reply, but I don't think FS / CLI / ESL supports this at the
> moment (or ever will... ;-)).
>
> - Mauritz
>
>
> -----Original Message-----
> From: Anthony Minessale
> Sent: Friday, February 25, 2011 12:10 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Receive order guarantee for bgapi command
> replies over ESL?
>
> bgapi commands are asynchronous that means when you send the command
> it spawns a thread and runs in the background.
> When its complete it will send you the event, its completely dependent
> on how long the task takes that you asked it to run.
> Its no different that doing 12 wgets to a website in the bg, there is
> no telling when they will be done.
>
>
>
> 2011/2/24 Mauritz L?vgren <mauritz.lovgren at hotmail.com>:
>> The use case is as follows;
>>
>> Multi-threaded Java applicaton that sends bgapi commands to several
>> channels
>> in parallel over the same ESL connection.
>> (bgapi commands are serialized in order to the socket outputstream, but 
>> is
>> there any guarantee that we will receive the OK / ERR reply for each 
>> bgapi
>> in the correct order from the socket inputstream?)
>>
>> And, what about sending bgapi commands on the same socket while a 
>> ?regular?
>> api command is currently executing? I suspect the bgapi reply will delay
>> until the ?regular? (blocking) api command is finished processing?
>>
>> - Mauritz
>>
>> From: Michael Collins
>> Sent: Thursday, February 24, 2011 9:44 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Receive order guarantee for bgapi command
>> replies over ESL?
>>
>> Just curious - what is the use case where sending a stack of bgapi's
>> without
>> listening for the reply is more desirable than looping through the bgapi
>> send/reply sequence for each one? I'm no super programmer but it seems to
>> me
>> that blindly relying on another system to always send stuff in the exact
>> correct order is dangerous and may cause bugs that are difficult to
>> diagnose. Personally I would consider it a programming best practice to
>> wait
>> for the response of the bgapi before sending another one.
>>
>> I invite other socket programmers to give their input...
>>
>> -MC
>>
>> P.S - you may wish to disable the legal notice at the and of your emails
>> when sending to a public list. (We know that some servers tag outgoing
>> messages automatically and if that is your case we understand. Go yell at
>> the I.T. guy! :)
>>
>> 2011/2/24 Mauritz L?vgren <mauritz.lovgren at hotmail.com>
>>>
>>> Hi,
>>>
>>> Are replies for bgapi commands sent back to an inbound socket connection
>>> in the same order as the commands were sent?
>>>
>>> Example: sending 10 bgapi messages from client to freeswitch socket with
>>> no delay inbetween.
>>> Will the response for those command messages be sent back by freeswitch
>>> in
>>> the exact same order as their commands were received, or should one wait
>>> for
>>> each reply before sending a new bgapi command to be sure one gets a 
>>> reply
>>> for the correct command?
>>>
>>> If there was an option of providing a UUID (or sequenceId) for the
>>> command, it would be easier to match the reply with the request upon
>>> receival, but this doesn?t seem to be possible with the current
>>> implementation?
>>>
>>> Regards,
>>> Mauritz Lovgren
>>> Systems Architect
>>> IPLink Inc.
>>
>>
>>
>> ________________________________
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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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> FreeSWITCH-users at 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 



From mbsip at gazeta.pl  Fri Feb 25 03:47:28 2011
From: mbsip at gazeta.pl (Mac)
Date: Fri, 25 Feb 2011 01:47:28 +0100
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: <AANLkTi=NGhUK9pnt1TGH9JKwUnn+r+UFphDx4c-7Wrkh@mail.gmail.com>

Thanks for feedback

Inbound leg has RFC2833, outbound has inband dtmf, so i've changed a
part of public.xml as you mentioned:
                <action application="set" data="proxy_media=false"/>
                <action application="set" data="bypass_media=false"/>
                <!--<action application="start_dtmf"/>-->
                <action application="bridge"
data="{execute_on_answer=start_dtmf}sofia/OPERATOR_2//$0"/>

Unfortunately it doesn't work.

Thx,
Mac


> you need to run the start_dtmf() application on the leg that has inband DTMF
>
> if its the inbound leg, add it to the DP
> <action application="start_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"/>
>
>
> On Thu, Feb 24, 2011 at 4:58 PM, Mac <mbsip at gazeta.pl> wrote:
>> Hi,
>>
>> Here they are:
>> http://pastebin.freeswitch.org/15473
>>
>> Thanks,
>> Mac
>>
>>> Get a debug-level log with siptrace enabled and paste it so we can see
>>> what's going on.
>>>
>>> -Steve
>>>
>>>
>>> On 22 February 2011 01:16, Mac <mbsip at gazeta.pl> wrote:
>>>>
>>>> Hello,
>>>>
>>>> Could somebody give me a hint?
>>>>
>>>> Thanks,
>>>> Mac
>>>>
>>>> > Hi Steven,
>>>> >
>>>> > There is much better :)
>>>> > I have different SDPs right now. The one is with rtpmap:101
>>>> > telephone-event/8000 (leg that is working with RFC2833), the opposite
>>>> > one incorporates a=rtpmap:101 telephone-event/8000 in INVITE SDP and
>>>> > pure codec rtpmaps.
>>>> > There is rtp.p_type == 101 working only on the left side. I cannot
>>>> > find any rtp.p_type == 101 on thark in opposide side which is okay.
>>>> > But the problem still persists.
>>>> >
>>>> > Once setting my UA to work with Inband DTMF everything works fine.
>>>> >
>>>> > Thanks,
>>>> > Mac
>>>> >
>>>> > 2011/2/20 Steven Ayre <steveayre at gmail.com>:
>>>> >> <action application="set" data="proxy_media=true"/>
>>>> >>
>>>> >> As indicated by the error, this is the problem. "proxy passthrough". In
>>>> >> proxy mode you pass the media straight through (passthrough). You can't
>>>> >> process passthrough media, such as is needed to mix in inband dtmf.
>>>> >> Use proxy_media=false and bypass_media=false. That's actually the
>>>> >> default so
>>>> >> unless you're setting either to true in the sip profile, you can remove
>>>> >> those lines from the dialplan completely.
>>>> >> Steve on iPhone
>>>> >> On 20 Feb 2011, at 18:51, Mac <mbsip at gazeta.pl> wrote:
>>>> >>
>>>> >> action application="set" data="proxy_media=true"/>
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>
>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >>
>>>> >>
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 mbsip at gazeta.pl  Fri Feb 25 03:59:28 2011
From: mbsip at gazeta.pl (Mac)
Date: Fri, 25 Feb 2011 01:59:28 +0100
Subject: [Freeswitch-users] RTP Proxy mode, DTMF conversion problem
In-Reply-To: <AANLkTi=NGhUK9pnt1TGH9JKwUnn+r+UFphDx4c-7Wrkh@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>
	<AANLkTi=NGhUK9pnt1TGH9JKwUnn+r+UFphDx4c-7Wrkh@mail.gmail.com>
Message-ID: <AANLkTimb_V1VVs33dY4erbnacb00611fNyLQnTjvhGQf@mail.gmail.com>

Okay i've found sth.

I was trying to achieve that by using
<action application="bridge"
data="{execute_on_answer=start_dtmf}sofia/gateway/OPERATOR_2/$0"/"/>
and it failes to convert dtmf

While using <action application="bridge"
data="{execute_on_answer=start_dtmf}sofia/$0 at IP_ADDR_OF_OPERATOR_2"/>
and it was success.

Is that mean that i my sip profile need to be adjusted somehow?

Thx,
Mac


> Thanks for feedback
>
> Inbound leg has RFC2833, outbound has inband dtmf, so i've changed a
> part of public.xml as you mentioned:
> ? ? ? ? ? ? ? ?<action application="set" data="proxy_media=false"/>
> ? ? ? ? ? ? ? ?<action application="set" data="bypass_media=false"/>
> ? ? ? ? ? ? ? ?<!--<action application="start_dtmf"/>-->
> ? ? ? ? ? ? ? ?<action application="bridge"
> data="{execute_on_answer=start_dtmf}sofia/OPERATOR_2//$0"/>
>
> Unfortunately it doesn't work.
>
> Thx,
> Mac
>
>
>> you need to run the start_dtmf() application on the leg that has inband DTMF
>>
>> if its the inbound leg, add it to the DP
>> <action application="start_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"/>
>>
>>
>> On Thu, Feb 24, 2011 at 4:58 PM, Mac <mbsip at gazeta.pl> wrote:
>>> Hi,
>>>
>>> Here they are:
>>> http://pastebin.freeswitch.org/15473
>>>
>>> Thanks,
>>> Mac
>>>
>>>> Get a debug-level log with siptrace enabled and paste it so we can see
>>>> what's going on.
>>>>
>>>> -Steve
>>>>
>>>>
>>>> On 22 February 2011 01:16, Mac <mbsip at gazeta.pl> wrote:
>>>>>
>>>>> Hello,
>>>>>
>>>>> Could somebody give me a hint?
>>>>>
>>>>> Thanks,
>>>>> Mac
>>>>>
>>>>> > Hi Steven,
>>>>> >
>>>>> > There is much better :)
>>>>> > I have different SDPs right now. The one is with rtpmap:101
>>>>> > telephone-event/8000 (leg that is working with RFC2833), the opposite
>>>>> > one incorporates a=rtpmap:101 telephone-event/8000 in INVITE SDP and
>>>>> > pure codec rtpmaps.
>>>>> > There is rtp.p_type == 101 working only on the left side. I cannot
>>>>> > find any rtp.p_type == 101 on thark in opposide side which is okay.
>>>>> > But the problem still persists.
>>>>> >
>>>>> > Once setting my UA to work with Inband DTMF everything works fine.
>>>>> >
>>>>> > Thanks,
>>>>> > Mac
>>>>> >
>>>>> > 2011/2/20 Steven Ayre <steveayre at gmail.com>:
>>>>> >> <action application="set" data="proxy_media=true"/>
>>>>> >>
>>>>> >> As indicated by the error, this is the problem. "proxy passthrough". In
>>>>> >> proxy mode you pass the media straight through (passthrough). You can't
>>>>> >> process passthrough media, such as is needed to mix in inband dtmf.
>>>>> >> Use proxy_media=false and bypass_media=false. That's actually the
>>>>> >> default so
>>>>> >> unless you're setting either to true in the sip profile, you can remove
>>>>> >> those lines from the dialplan completely.
>>>>> >> Steve on iPhone
>>>>> >> On 20 Feb 2011, at 18:51, Mac <mbsip at gazeta.pl> wrote:
>>>>> >>
>>>>> >> action application="set" data="proxy_media=true"/>
>>>>> >>
>>>>> >> _______________________________________________
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >>
>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> http://www.freeswitch.org
>>>>> >>
>>>>> >>
>>>>> >
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 infos at madovsky.org  Fri Feb 25 04:02:45 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 24 Feb 2011 20:02:45 -0500
Subject: [Freeswitch-users] exec app on originate
Message-ID: <748093B885AD48CE86278841FBEA1159@e1705>

when I do this :
originate /sofia/gateway/blabla/9999999999 &app(hahaha)

is the ap executed at early media or after answer ?

Thanks
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From infos at madovsky.org  Fri Feb 25 04:04:31 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 24 Feb 2011 20:04:31 -0500
Subject: [Freeswitch-users] one leg call and channel uuid
References: <5CB1934490CF4E55977717E2B6455962@e1705>
	<AANLkTimVWtHEh2K_6TdVftLpsW6TD7aUPZxJrqQtgd6f@mail.gmail.com>
Message-ID: <EF9533EB7E2741028474447A1D27125E@e1705>

> What is the scenario? What is your one-legged call?
inbound call that is transferred to FS voicemail for example
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Thursday, February 24, 2011 3:23 PM
  Subject: Re: [Freeswitch-users] one leg call and channel uuid




  On Thu, Feb 24, 2011 at 11:35 AM, Madovsky <infos at madovsky.org> wrote:

    when there is one leg call,
    if I want to use uuid_broadcast or uuid_transfer,
    should I use the uuid of leg A ?


  What is the scenario? What is your one-legged call?
  -MC 


------------------------------------------------------------------------------


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From anthony.minessale at gmail.com  Fri Feb 25 04:04:16 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 24 Feb 2011 19:04:16 -0600
Subject: [Freeswitch-users] exec app on originate
In-Reply-To: <748093B885AD48CE86278841FBEA1159@e1705>
References: <748093B885AD48CE86278841FBEA1159@e1705>
Message-ID: <AANLkTikmsGve+DMf9RzyyB3wBH0dgpT_1gkG4hNuq7du@mail.gmail.com>

early media
if you want after answer do

originate {ignore_early_media=true}sofia/gateway/blabla/9999999999 &app(hahaha)


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


From infos at madovsky.org  Fri Feb 25 04:14:27 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 24 Feb 2011 20:14:27 -0500
Subject: [Freeswitch-users] exec app on originate
References: <748093B885AD48CE86278841FBEA1159@e1705>
	<AANLkTikmsGve+DMf9RzyyB3wBH0dgpT_1gkG4hNuq7du@mail.gmail.com>
Message-ID: <D6D6748ED77B478CB219BB6038C58A86@e1705>

ha ok got it.
so in case of txfax app should I use early_media ?
in the case of the call is already answered can
I run the app from uuid_broadcast or uuid_transfer if
the app is in an extension ?
I tried the last but it makes a fax timeout

thanks

----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, February 24, 2011 8:04 PM
Subject: Re: [Freeswitch-users] exec app on originate


> early media
> if you want after answer do
>
> originate {ignore_early_media=true}sofia/gateway/blabla/9999999999 
> &app(hahaha)
>
>
> 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 



From msc at freeswitch.org  Fri Feb 25 05:58:12 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 24 Feb 2011 18:58:12 -0800
Subject: [Freeswitch-users] Second transfer
In-Reply-To: <AANLkTikWMF2rf=ASaTMg6bQ=jw7KQu59wJuXr+fRj0Rg@mail.gmail.com>
References: <AANLkTi=pB-Rt7j1cvNGgSs2s3F4h6jse1nTqTqRr8kKJ@mail.gmail.com>
	<AANLkTi=fvkp_8e4gXthWtYQTLAyxj+bbUMtdpAhXQamQ@mail.gmail.com>
	<AANLkTinw4CWxe+GW31AcMnzWWNsQU=W7WGSmsr2ky9zg@mail.gmail.com>
	<AANLkTikWMF2rf=ASaTMg6bQ=jw7KQu59wJuXr+fRj0Rg@mail.gmail.com>
Message-ID: <AANLkTin6wdxuB3N4oxOGobY0Q7r3n33=wEz2T49eN9yt@mail.gmail.com>

On Thu, Feb 24, 2011 at 3:11 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> add the 1 flag to the flags
> ction application="bind_meta_app" data="3 b s1 log::INFO> Transfer"/>
>
> this says it can only work once then detaches itself.
>
> Awesome. The I'll get this properly wikified.
-MC
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From msc at freeswitch.org  Fri Feb 25 06:07:08 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 24 Feb 2011 19:07:08 -0800
Subject: [Freeswitch-users] Second transfer
In-Reply-To: <AANLkTin6wdxuB3N4oxOGobY0Q7r3n33=wEz2T49eN9yt@mail.gmail.com>
References: <AANLkTi=pB-Rt7j1cvNGgSs2s3F4h6jse1nTqTqRr8kKJ@mail.gmail.com>
	<AANLkTi=fvkp_8e4gXthWtYQTLAyxj+bbUMtdpAhXQamQ@mail.gmail.com>
	<AANLkTinw4CWxe+GW31AcMnzWWNsQU=W7WGSmsr2ky9zg@mail.gmail.com>
	<AANLkTikWMF2rf=ASaTMg6bQ=jw7KQu59wJuXr+fRj0Rg@mail.gmail.com>
	<AANLkTin6wdxuB3N4oxOGobY0Q7r3n33=wEz2T49eN9yt@mail.gmail.com>
Message-ID: <AANLkTimKPV=LY45dNXzdVZ0SzmnBCMv7UoaRR3_8PDz_@mail.gmail.com>

On Thu, Feb 24, 2011 at 6:58 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Thu, Feb 24, 2011 at 3:11 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> add the 1 flag to the flags
>> ction application="bind_meta_app" data="3 b s1 log::INFO> Transfer"/>
>>
>> this says it can only work once then detaches itself.
>>
>> Awesome. The I'll get this properly wikified.
> -MC
>
> Wiki has been updated to reflect the proper flags that can be set. Also, I
happened to stumble across "unbind_meta_app" which I'm assuming could also
be used.

-MC

P.S. - The i flag says to execute inline... what exactly does that do?
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From msc at freeswitch.org  Fri Feb 25 06:13:17 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 24 Feb 2011 19:13:17 -0800
Subject: [Freeswitch-users] Receive order guarantee for bgapi command
 replies over ESL?
In-Reply-To: <AANLkTimPh0eqMz=JaNxV6hLBrG1oLVtYtE+4czuxV=KF@mail.gmail.com>
References: <SNT107-DS120CD9CBAFDAA80E16A6CDE6DA0@phx.gbl>
	<AANLkTikMhk3C+3Gc8ebf8B_O8kOnrxhBRNS4a6bH4Ux=@mail.gmail.com>
	<SNT107-DS17CDAAB3CB7918E0D98882E6DA0@phx.gbl>
	<AANLkTinMLusDxVJ8bm+DEPB4OqrPj292wNbYR_wWF3j8@mail.gmail.com>
	<SNT107-DS147A2767EEB9599E895DD7E6DA0@phx.gbl>
	<AANLkTimPh0eqMz=JaNxV6hLBrG1oLVtYtE+4czuxV=KF@mail.gmail.com>
Message-ID: <AANLkTinA8n5wiAS1Dm8EhCb=oxo_9ADTeFn5m8OhiRMH@mail.gmail.com>

>
>
> As for the sequence ID question, you can supply your own job-uuid
> header when you send the bgapi-command
>
> bgapi show channels
> job-uuid: 1234
>
> This was not available from the script mods when I checked so I added
> it to latest build.
> example in perl but true in all swigged langs.
>
> my $command = shift;
> my $args = join(" ", @ARGV);
>
> my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
>
> my $e = $con->bgapi($command, $args, "my-job-id");
> print $e->serialize("json");
>
>
> returns:
>
> {
>        "Event-Name":   "SOCKET_DATA",
>        "Content-Type": "command/reply",
>        "Reply-Text":   "+OK Job-UUID: my-job-id",
>        "Job-UUID":     "my-job-id"
> }
>


Awesome! This also has been added to the wiki:
http://wiki.freeswitch.org/wiki/Event_Socket_Library#bgapi

-MC
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From msc at freeswitch.org  Fri Feb 25 06:15:00 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 24 Feb 2011 19:15:00 -0800
Subject: [Freeswitch-users] exec app on originate
In-Reply-To: <D6D6748ED77B478CB219BB6038C58A86@e1705>
References: <748093B885AD48CE86278841FBEA1159@e1705>
	<AANLkTikmsGve+DMf9RzyyB3wBH0dgpT_1gkG4hNuq7du@mail.gmail.com>
	<D6D6748ED77B478CB219BB6038C58A86@e1705>
Message-ID: <AANLkTinorB_WKCxL7noF8j-Trzdbj9neGVypSA+hKykC@mail.gmail.com>

On Thu, Feb 24, 2011 at 5:14 PM, Madovsky <infos at madovsky.org> wrote:

> ha ok got it.
> so in case of txfax app should I use early_media ?
> in the case of the call is already answered can
> I run the app from uuid_broadcast or uuid_transfer if
> the app is in an extension ?
> I tried the last but it makes a fax timeout
>

What exactly did you do that resulted in a fax timeout? Please pastebin
console log w/ the extension in question and whatever API commands you sent.

-MC
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From garmt.noname at gmail.com  Fri Feb 25 02:47:43 2011
From: garmt.noname at gmail.com (Grmt)
Date: Fri, 25 Feb 2011 00:47:43 +0100
Subject: [Freeswitch-users] implementing stereo (e.g. portaudio to test
	it easily)
In-Reply-To: <AANLkTikDFqS_Y9XS-gnVvnfPhAW1Eri=zy4XB3vAS2N0@mail.gmail.com>
References: <AANLkTikL9SQPdsJ6XF-AUZuLQ3Z-AG7Eqgj2BsL=CZ4S@mail.gmail.com>
	<AANLkTinowzmzt8NxB-FtifkZEQZ8s3OQAWLuC18PN60Q@mail.gmail.com>
	<AANLkTikDFqS_Y9XS-gnVvnfPhAW1Eri=zy4XB3vAS2N0@mail.gmail.com>
Message-ID: <4d66ee22.857a0e0a.6c3f.0343@mx.google.com>

If I remember well I think you mentioned that anthm said that it was
possible.

I just gave you a quick intro into FS.

 

Did you buy the book? And read it?

Did you spend at least a week playing with FS?

Did you spend at least that amount of time with your development
environment?

Did you debug the scenario I presented to you and inspect  the source code?

 

If you did, you should be able to make a rough order of magnitude estimation
of the effort needed to make this scenario do stereo. And additionally, you
would know if your skills and tenacity match the task you have set out for
yourself. 

 

Given your background and the amount of time you have, I think you should be
able to do your project.

I would start by looking at the code right there were it tells you that it
converts to mono.

 

And again, given my level of expertise in FS or coding or codecs or
whatever, I cannot help you any further, but I hope to see you become a FS
expert so that I can have a stereo conversation with you in a few months.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
MatzeMuc86
Sent: Thursday, 24 February, 2011 21:49
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] implementing stereo (e.g. portaudio to test
it easily)

 

Are you sure?
I had a diskussion at the IRC dev channel and Grmt told me it should be
possible.
It is quite important for me and i do not ask somebody to implement this for
me.

Thanks
MatzeMuc86



2011/2/24 Michael Collins <msc at freeswitch.org>

Sorry, FreeSWITCH is not set up to do stereo... perhaps one day, but it
isn't something that is high on the priority list.

 

-MC

On Thu, Feb 24, 2011 at 9:22 AM, MatzeMuc86 <matzemuc86 at gmail.com> wrote:

hello FreeSWITCH community,

actually it is my first time to use lists like this. I hope I do everything
fine.
For my project I try to get stereo (and some special features later)
integrated in freeSWITCH, especially SIP.
TO start with this idea I need to know where I have to activate or implement
stereo.
In the second I try to play a bit with the portaudio module as I do not have
to focus on SDP problems transporting SIP and the need of a SIP client which
is able to receive stereo.
So I set up some custom moh local media stuff which works finde - using
mono.
Changing the channels to 2 makes the console telling me that the sound file
will be mixed to mono and the sound is quite strange.
I would be happy about getting some useful information which modules or
configs are the important ones for my idea. Unfurtun ately I have only found
some recording stuff about stereo which is not what I want to do.

Thank you very much.
MatzeMuc86

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

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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 Feb 25 06:18:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 24 Feb 2011 19:18:36 -0800
Subject: [Freeswitch-users] RTP Proxy mode, DTMF conversion problem
In-Reply-To: <AANLkTimb_V1VVs33dY4erbnacb00611fNyLQnTjvhGQf@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>
	<AANLkTi=NGhUK9pnt1TGH9JKwUnn+r+UFphDx4c-7Wrkh@mail.gmail.com>
	<AANLkTimb_V1VVs33dY4erbnacb00611fNyLQnTjvhGQf@mail.gmail.com>
Message-ID: <AANLkTinrssFzaPWoYh_v7v0edUvLUSWWeE_+LZaYyh=L@mail.gmail.com>

Get a pcap of both the working and the failing calls and compare them in
wireshark. Also, compare the console output of the working and the failing
calls. Look for differences. You may see a clue. If you need assistance then
pastebin the output and we'll all have a look.

-MC

On Thu, Feb 24, 2011 at 4:59 PM, Mac <mbsip at gazeta.pl> wrote:

> Okay i've found sth.
>
> I was trying to achieve that by using
> <action application="bridge"
> data="{execute_on_answer=start_dtmf}sofia/gateway/OPERATOR_2/$0"/"/>
> and it failes to convert dtmf
>
> While using <action application="bridge"
> data="{execute_on_answer=start_dtmf}sofia/$0 at IP_ADDR_OF_OPERATOR_2"/>
> and it was success.
>
> Is that mean that i my sip profile need to be adjusted somehow?
>
> Thx,
> Mac
>
>
> > Thanks for feedback
> >
> > Inbound leg has RFC2833, outbound has inband dtmf, so i've changed a
> > part of public.xml as you mentioned:
> >                <action application="set" data="proxy_media=false"/>
> >                <action application="set" data="bypass_media=false"/>
> >                <!--<action application="start_dtmf"/>-->
> >                <action application="bridge"
> > data="{execute_on_answer=start_dtmf}sofia/OPERATOR_2//$0"/>
> >
> > Unfortunately it doesn't work.
> >
> > Thx,
> > Mac
> >
> >
> >> you need to run the start_dtmf() application on the leg that has inband
> DTMF
> >>
> >> if its the inbound leg, add it to the DP
> >> <action application="start_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"/>
> >>
> >>
> >> On Thu, Feb 24, 2011 at 4:58 PM, Mac <mbsip at gazeta.pl> wrote:
> >>> Hi,
> >>>
> >>> Here they are:
> >>> http://pastebin.freeswitch.org/15473
> >>>
> >>> Thanks,
> >>> Mac
> >>>
> >>>> Get a debug-level log with siptrace enabled and paste it so we can see
> >>>> what's going on.
> >>>>
> >>>> -Steve
> >>>>
> >>>>
> >>>> On 22 February 2011 01:16, Mac <mbsip at gazeta.pl> wrote:
> >>>>>
> >>>>> Hello,
> >>>>>
> >>>>> Could somebody give me a hint?
> >>>>>
> >>>>> Thanks,
> >>>>> Mac
> >>>>>
> >>>>> > Hi Steven,
> >>>>> >
> >>>>> > There is much better :)
> >>>>> > I have different SDPs right now. The one is with rtpmap:101
> >>>>> > telephone-event/8000 (leg that is working with RFC2833), the
> opposite
> >>>>> > one incorporates a=rtpmap:101 telephone-event/8000 in INVITE SDP
> and
> >>>>> > pure codec rtpmaps.
> >>>>> > There is rtp.p_type == 101 working only on the left side. I cannot
> >>>>> > find any rtp.p_type == 101 on thark in opposide side which is okay.
> >>>>> > But the problem still persists.
> >>>>> >
> >>>>> > Once setting my UA to work with Inband DTMF everything works fine.
> >>>>> >
> >>>>> > Thanks,
> >>>>> > Mac
> >>>>> >
> >>>>> > 2011/2/20 Steven Ayre <steveayre at gmail.com>:
> >>>>> >> <action application="set" data="proxy_media=true"/>
> >>>>> >>
> >>>>> >> As indicated by the error, this is the problem. "proxy
> passthrough". In
> >>>>> >> proxy mode you pass the media straight through (passthrough). You
> can't
> >>>>> >> process passthrough media, such as is needed to mix in inband
> dtmf.
> >>>>> >> Use proxy_media=false and bypass_media=false. That's actually the
> >>>>> >> default so
> >>>>> >> unless you're setting either to true in the sip profile, you can
> remove
> >>>>> >> those lines from the dialplan completely.
> >>>>> >> Steve on iPhone
> >>>>> >> On 20 Feb 2011, at 18:51, Mac <mbsip at gazeta.pl> wrote:
> >>>>> >>
> >>>>> >> action application="set" data="proxy_media=true"/>
> >>>>> >>
> >>>>> >> _______________________________________________
> >>>>> >> FreeSWITCH-users mailing list
> >>>>> >> FreeSWITCH-users at lists.freeswitch.org
> >>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> >>
> >>>>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> >> http://www.freeswitch.org
> >>>>> >>
> >>>>> >>
> >>>>> >
> >>>>>
> >>>>> _______________________________________________
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at 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 msc at freeswitch.org  Fri Feb 25 06:20:21 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 24 Feb 2011 19:20:21 -0800
Subject: [Freeswitch-users] implementing stereo (e.g. portaudio to test
	it easily)
In-Reply-To: <4d66ee22.857a0e0a.6c3f.0343@mx.google.com>
References: <AANLkTikL9SQPdsJ6XF-AUZuLQ3Z-AG7Eqgj2BsL=CZ4S@mail.gmail.com>
	<AANLkTinowzmzt8NxB-FtifkZEQZ8s3OQAWLuC18PN60Q@mail.gmail.com>
	<AANLkTikDFqS_Y9XS-gnVvnfPhAW1Eri=zy4XB3vAS2N0@mail.gmail.com>
	<4d66ee22.857a0e0a.6c3f.0343@mx.google.com>
Message-ID: <AANLkTimrx5q076f+EAjgMQBh4c4kZYE2Mjq401oV2V7z@mail.gmail.com>

Hehe,

In other words, you didn't tell him: Just flip this switch and you can have
stereo. :) It's a lot of work but an expert could definitely do it.

DISCLAIMER: I am not such an expert. :P

-MC

On Thu, Feb 24, 2011 at 3:47 PM, Grmt <garmt.noname at gmail.com> wrote:

> If I remember well I think you mentioned that anthm said that it was
> possible.
>
> I just gave you a quick intro into FS.
>
>
>
> Did you buy the book? And read it?
>
> Did you spend at least a week playing with FS?
>
> Did you spend at least that amount of time with your development
> environment?
>
> Did you debug the scenario I presented to you and inspect  the source code?
>
>
>
> If you did, you should be able to make a rough order of magnitude
> estimation of the effort needed to make this scenario do stereo. And
> additionally, you would know if your skills and tenacity match the task you
> have set out for yourself.
>
>
>
> Given your background and the amount of time you have, I think you should
> be able to do your project.
>
> I would start by looking at the code right there were it tells you that it
> converts to mono.
>
>
>
> And again, given my level of expertise in FS or coding or codecs or
> whatever, I cannot help you any further, but I hope to see you become a FS
> expert so that I can have a stereo conversation with you in a few months?
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *MatzeMuc86
> *Sent:* Thursday, 24 February, 2011 21:49
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] implementing stereo (e.g. portaudio to
> test it easily)
>
>
>
> Are you sure?
> I had a diskussion at the IRC dev channel and Grmt told me it should be
> possible.
> It is quite important for me and i do not ask somebody to implement this
> for me.
>
> Thanks
> MatzeMuc86
>
> 2011/2/24 Michael Collins <msc at freeswitch.org>
>
> Sorry, FreeSWITCH is not set up to do stereo... perhaps one day, but it
> isn't something that is high on the priority list.
>
>
>
> -MC
>
> On Thu, Feb 24, 2011 at 9:22 AM, MatzeMuc86 <matzemuc86 at gmail.com> wrote:
>
> hello FreeSWITCH community,
>
> actually it is my first time to use lists like this. I hope I do everything
> fine.
> For my project I try to get stereo (and some special features later)
> integrated in freeSWITCH, especially SIP.
> TO start with this idea I need to know where I have to activate or
> implement stereo.
> In the second I try to play a bit with the portaudio module as I do not
> have to focus on SDP problems transporting SIP and the need of a SIP client
> which is able to receive stereo.
> So I set up some custom moh local media stuff which works finde - using
> mono.
> Changing the channels to 2 makes the console telling me that the sound file
> will be mixed to mono and the sound is quite strange.
> I would be happy about getting some useful information which modules or
> configs are the important ones for my idea. Unfurtun ately I have only found
> some recording stuff about stereo which is not what I want to do.
>
> Thank you very much.
> MatzeMuc86
>
> _______________________________________________
> FreeSWITCH-users mailing 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 spencer at 5ninesolutions.com  Fri Feb 25 08:45:16 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Thu, 24 Feb 2011 21:45:16 -0800
Subject: [Freeswitch-users] Polycom G722 negotiation
Message-ID: <AE216221-5892-4337-8985-1889410A866F@5ninesolutions.com>

Hello all,
I have several Polycom IP-650s.  On the external profile I have:
CODECS IN        	G722,PCMU,G729,GSM
CODECS OUT       	PCMU,G729

On the internal profile I have
CODECS IN        	G722,PCMU,G729,GSM
CODECS OUT       	G722,PCMU,G729,GSM

The Polycoms have a similar codec priority set:
v=0
o=- 1167611879 1167611879 IN IP4 10.59.1.243
s=Polycom IP Phone
c=IN IP4 10.59.1.243
t=0 0
a=sendrecv
m=audio 2236 RTP/AVP 9 0 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000

The issue I have is that if I leave the internal profile to generous  
and a user places a call from the PSTN on hold which is PCMU, when the  
user tries to pick up the call, the Polycoms' preference then changes  
the call to G722 and there is no audio.  If I set the profile to  
greedy, since the call is already PCMU Freeswitch keeps it at PCMU and  
audio is fine.  I realize this is the intended behavior for greedy vs  
generous but my question is why is there no audio when switching the  
codecs?  I didn't want to clutter this up any more but I'm happy to  
provide traces however the only real difference is the codec order.

Thanks,
Spencer



--
Spencer Thomason
5Nine Solutions LLC
e. sales at 5ninesolutions.com
p. +1.888.271.7959
f. +1.310.510.6980



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From thisjoy0528 at gmail.com  Fri Feb 25 08:49:52 2011
From: thisjoy0528 at gmail.com (joy this)
Date: Fri, 25 Feb 2011 13:49:52 +0800
Subject: [Freeswitch-users]  problem with voice quality
Message-ID: <AANLkTik15iq+zwivCJWZeVFOmDVvAFB70v7haeK9XE2K@mail.gmail.com>

Dear all:



        The scenario is that, when I use 3CX (PCMA) to call XLite, 3CX will
hear loud noise but XLite is fine. And I found that it would only happen
when XLite supports PCMA and PCMU simultaneously. If XLite only support
PCMA, everything is fine.

Maybe it is because of the old version of Freeswitch, I download the latest
version yesterday and re-test. The result is that 3CX hear nothing instead
of noise. I use Wireshark to survey packages, and the server does not
transfer UDP packages to 3CX. I check the log file, it is as below.



-----3CX hears nothing-----

2011-02-24 20:17:30.724125 [DEBUG] sofia.c:4683 Channel
sofia/internal/1000 at thisjoy.x.x:5060 entering state [completed][200]

2011-02-24 20:17:30.724125 [DEBUG] switch_core_session.c:709 Send signal
sofia/internal/1000 at thisjoy.x.x:5060 [BREAK]

2011-02-24 20:17:30.724125 [DEBUG] switch_channel.c:2790
(sofia/internal/1000 at thisjoy.x.x:5060) Callstate Change EARLY -> ACTIVE

2011-02-24 20:17:30.724125 [NOTICE] switch_ivr_originate.c:3363 Channel
[sofia/internal/1000 at thisjoy.x.x:5060] has been answered

2011-02-24 20:17:30.724125 [DEBUG] switch_ivr_originate.c:3408 Originate
Resulted in Success: [sofia/internal/sip:1001 at 114.24.x.x:56922]

2011-02-24 20:17:30.739750 [DEBUG] switch_ivr_originate.c:3408 Originate
Resulted in Success: [sofia/internal/sip:1001 at 114.24.x.x:56922]

2011-02-24 20:17:30.739750 [DEBUG] switch_core_session.c:709 Send signal
sofia/internal/sip:1001 at 114.24.x.x:56922 [BREAK]

2011-02-24 20:17:30.739750 [DEBUG] switch_core_session.c:709 Send signal
sofia/internal/1000 at thisjoy.x.x:5060 [BREAK]

2011-02-24 20:17:30.739750 [DEBUG] switch_ivr_bridge.c:1234
(sofia/internal/sip:1001 at 114.24.x.x:56922) State Change CS_CONSUME_MEDIA ->
CS_EXCHANGE_MEDIA

2011-02-24 20:17:30.739750 [DEBUG] switch_core_session.c:1116 Send signal
sofia/internal/sip:1001 at 114.24.x.x:56922 [BREAK]

2011-02-24 20:17:30.739750 [DEBUG] switch_core_state_machine.c:320
(sofia/internal/sip:1001 at 114.24.x.x:56922) Running State Change
CS_EXCHANGE_MEDIA

2011-02-24 20:17:30.739750 [DEBUG] switch_core_state_machine.c:369
(sofia/internal/sip:1001 at 114.24.x.x:56922) State EXCHANGE_MEDIA

2011-02-24 20:17:30.739750 [DEBUG] mod_sofia.c:552 SOFIA EXCHANGE_MEDIA

2011-02-24 20:17:30.974125 [DEBUG] switch_rtp.c:2985 Correct ip/port
confirmed.

2011-02-24 20:17:30.974125 [DEBUG] sofia.c:4683 Channel
sofia/internal/1000 at thisjoy.x.x:5060 entering state [ready][200]

2011-02-24 20:17:31.021000 [DEBUG] switch_core_session.c:771 Send signal
sofia/internal/sip:1001 at 114.24.x.x:56922 [BREAK]

2011-02-24 20:17:31.021000 [DEBUG] switch_core_session.c:771 Send signal
sofia/internal/1000 at thisjoy.x.x:5060 [BREAK]

-----log end-----



When XLite only supports PCMA, it would be an additional log:

2011-02-24 20:16:06.661625 [INFO] switch_rtp.c:2968 Auto Changing port from
192.168.1.1:51508 to 114.24.x.x:51508

        And I test it again by using a public VoIP server (
sip.lowratevoip.com), there is no noise or silence situation, everything is
fine.

        Does XLite get bugs? Why the public server can deal with the
situation? Could anyone help me?


Sincerely yours,

thisjoy.
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From infos at madovsky.org  Fri Feb 25 09:35:46 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 25 Feb 2011 01:35:46 -0500
Subject: [Freeswitch-users] exec app on originate
References: <748093B885AD48CE86278841FBEA1159@e1705><AANLkTikmsGve+DMf9RzyyB3wBH0dgpT_1gkG4hNuq7du@mail.gmail.com><D6D6748ED77B478CB219BB6038C58A86@e1705>
	<AANLkTinorB_WKCxL7noF8j-Trzdbj9neGVypSA+hKykC@mail.gmail.com>
Message-ID: <A1FB7D912E1541A4BF86D7EF8A8EE78E@e1705>

ok Mike, after hundred tests I understand more now...
but first I made a mistake by sending broadcast to aleg and not bleg.

Thank you

Franck
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Thursday, February 24, 2011 10:15 PM
  Subject: Re: [Freeswitch-users] exec app on originate





  On Thu, Feb 24, 2011 at 5:14 PM, Madovsky <infos at madovsky.org> wrote:

    ha ok got it.
    so in case of txfax app should I use early_media ?
    in the case of the call is already answered can
    I run the app from uuid_broadcast or uuid_transfer if
    the app is in an extension ?
    I tried the last but it makes a fax timeout



  What exactly did you do that resulted in a fax timeout? Please pastebin console log w/ the extension in question and whatever API commands you sent.


  -MC


------------------------------------------------------------------------------


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From matzemuc86 at gmail.com  Fri Feb 25 13:16:31 2011
From: matzemuc86 at gmail.com (MatzeMuc86)
Date: Fri, 25 Feb 2011 11:16:31 +0100
Subject: [Freeswitch-users] implementing stereo (e.g. portaudio to
	test	it easily)
In-Reply-To: <AANLkTimrx5q076f+EAjgMQBh4c4kZYE2Mjq401oV2V7z@mail.gmail.com>
References: <AANLkTikL9SQPdsJ6XF-AUZuLQ3Z-AG7Eqgj2BsL=CZ4S@mail.gmail.com>	<AANLkTinowzmzt8NxB-FtifkZEQZ8s3OQAWLuC18PN60Q@mail.gmail.com>	<AANLkTikDFqS_Y9XS-gnVvnfPhAW1Eri=zy4XB3vAS2N0@mail.gmail.com>	<4d66ee22.857a0e0a.6c3f.0343@mx.google.com>
	<AANLkTimrx5q076f+EAjgMQBh4c4kZYE2Mjq401oV2V7z@mail.gmail.com>
Message-ID: <000801cbd4d5$13362830$39a27890$@gmail.com>

Hello,

 

OK I'm sure that I am not such an expert. I have done some C, but it's in
the past (which shouldn't be a big problem). C++ is a bigger problem but it
seams to be C, not C++ (?).

 

Of course I read the book, played with freeswitch and did some debugging
(problems with debugging are written below).

 

I created a stream local media extension to play a stereo 32 kHz wav file.

So far it is down mixed to mono in the switch_core_file.c file.

I could find an online documentation about this c file by doxygen:
http://www.freeswitch.org/eg/dox/switch__core__file_8c.html

I checked the code and found some positions where fh->channels have to be >
1.

 

All in all I can try to get these functions ready for stereo, but I have the
feeling that this is only one very small problem regarding to my whole idea.

 

I think about debugging the whole "pa call 1987" progress but therefore I
need to know where to put the first debug point => where does the program
starts executing my custom extension?

 

In addition: Is there some further documentation or is the doxygen, mostly
the same as the pure C code, all I can get for code-documentations (e.g.
what are all the variables, what are the ideas e.g.).

 

Thanks a lot

MatzeMuc86

 

Von: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
Michael Collins
Gesendet: Freitag, 25. Februar 2011 04:20
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] implementing stereo (e.g. portaudio to test
it easily)

 

Hehe,

 

In other words, you didn't tell him: Just flip this switch and you can have
stereo. :) It's a lot of work but an expert could definitely do it.

 

DISCLAIMER: I am not such an expert. :P

 

-MC

On Thu, Feb 24, 2011 at 3:47 PM, Grmt <garmt.noname at gmail.com> wrote:

If I remember well I think you mentioned that anthm said that it was
possible.

I just gave you a quick intro into FS.

 

Did you buy the book? And read it?

Did you spend at least a week playing with FS?

Did you spend at least that amount of time with your development
environment?

Did you debug the scenario I presented to you and inspect  the source code?

 

If you did, you should be able to make a rough order of magnitude estimation
of the effort needed to make this scenario do stereo. And additionally, you
would know if your skills and tenacity match the task you have set out for
yourself. 

 

Given your background and the amount of time you have, I think you should be
able to do your project.

I would start by looking at the code right there were it tells you that it
converts to mono.

 

And again, given my level of expertise in FS or coding or codecs or
whatever, I cannot help you any further, but I hope to see you become a FS
expert so that I can have a stereo conversation with you in a few months.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
MatzeMuc86
Sent: Thursday, 24 February, 2011 21:49
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] implementing stereo (e.g. portaudio to test
it easily)

 

Are you sure?
I had a diskussion at the IRC dev channel and Grmt told me it should be
possible.
It is quite important for me and i do not ask somebody to implement this for
me.

Thanks
MatzeMuc86

2011/2/24 Michael Collins <msc at freeswitch.org>

Sorry, FreeSWITCH is not set up to do stereo... perhaps one day, but it
isn't something that is high on the priority list.

 

-MC

On Thu, Feb 24, 2011 at 9:22 AM, MatzeMuc86 <matzemuc86 at gmail.com> wrote:

hello FreeSWITCH community,

actually it is my first time to use lists like this. I hope I do everything
fine.
For my project I try to get stereo (and some special features later)
integrated in freeSWITCH, especially SIP.
TO start with this idea I need to know where I have to activate or implement
stereo.
In the second I try to play a bit with the portaudio module as I do not have
to focus on SDP problems transporting SIP and the need of a SIP client which
is able to receive stereo.
So I set up some custom moh local media stuff which works finde - using
mono.
Changing the channels to 2 makes the console telling me that the sound file
will be mixed to mono and the sound is quite strange.
I would be happy about getting some useful information which modules or
configs are the important ones for my idea. Unfurtun ately I have only found
some recording stuff about stereo which is not what I want to do.

Thank you very much.
MatzeMuc86

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

 


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

 


_______________________________________________
FreeSWITCH-users mailing 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 david.varnes at gmail.com  Fri Feb 25 14:02:56 2011
From: david.varnes at gmail.com (david varnes)
Date: Fri, 25 Feb 2011 22:02:56 +1100
Subject: [Freeswitch-users] Receive order guarantee for bgapi command
 replies over ESL?
In-Reply-To: <SNT107-DS215A438073C211F4E0241FE6DD0@phx.gbl>
References: <SNT107-DS120CD9CBAFDAA80E16A6CDE6DA0@phx.gbl>
	<AANLkTikMhk3C+3Gc8ebf8B_O8kOnrxhBRNS4a6bH4Ux=@mail.gmail.com>
	<SNT107-DS17CDAAB3CB7918E0D98882E6DA0@phx.gbl>
	<AANLkTinMLusDxVJ8bm+DEPB4OqrPj292wNbYR_wWF3j8@mail.gmail.com>
	<SNT107-DS147A2767EEB9599E895DD7E6DA0@phx.gbl>
	<AANLkTimPh0eqMz=JaNxV6hLBrG1oLVtYtE+4czuxV=KF@mail.gmail.com>
	<SNT107-DS215A438073C211F4E0241FE6DD0@phx.gbl>
Message-ID: <AANLkTi=ko5Hds=ESuLQ_o6u4wDdhc-zRq_x6X17czg9S@mail.gmail.com>

Anthony,
thanks for that post .. that is REALLY useful to know :-)


Mauritz,
I will update the Java client to make built-in use of that
information in the next week or so. I am just heading
away for a couple of days, otherwise it would be
sooner.  I will ping the list when it is updated.

davidv

2011/2/25 Mauritz L?vgren <mauritz.lovgren at hotmail.com>:
> The ability to specify the job-uuid is just what we need!
> This simplifies the bgapi command / reply / background event mechanism we
> use.
> We are currently using the Java ESL library from David Varnes as we run
> freeswitch on various platforms (Windows, FreeBSD, Mac etc etc) and wanted
> to avoid using JNI in our Java application, but we will consider using the
> swig ESL library instead if this proves to be a better solution.
>
> I am running on the latest build now, so will test using self-made job-uuids
> (will probably use UUIDs since they are really unique :-)).
>
> Thank you for the information and keep up the great work!
>
> Mauritz
>
> -----Original Message-----
> From: Anthony Minessale
> Sent: Friday, February 25, 2011 12:54 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Receive order guarantee for bgapi command
> replies over ESL?
>
> When you send bgapi command the ok with the job-uuid will always be
> for the job you just submitted.
> When you send any command you are required to wait for the reply to
> that command, if it's bgapi you will get the job-uuid.
> The only problem you face is if you are also listening for events you
> may get an event in response to your request which you must save in a
> queue and keep reading until you get the reply to the command you
> issued.
>
> If you are using the ESL library supplied with FreeSWITCH this is all
> done for you.
>
> As for the sequence ID question, you can supply your own job-uuid
> header when you send the bgapi-command
>
> bgapi show channels
> job-uuid: 1234
>
> This was not available from the script mods when I checked so I added
> it to latest build.
> example in perl but true in all swigged langs.
>
> my $command = shift;
> my $args = join(" ", @ARGV);
>
> my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
>
> my $e = $con->bgapi($command, $args, "my-job-id");
> print $e->serialize("json");
>
>
> returns:
>
> {
> ? ? ? ?"Event-Name": ? "SOCKET_DATA",
> ? ? ? ?"Content-Type": "command/reply",
> ? ? ? ?"Reply-Text": ? "+OK Job-UUID: my-job-id",
> ? ? ? ?"Job-UUID": ? ? "my-job-id"
> }
>
>
>
>
>
>
>
>
>
> 2011/2/24 Mauritz L?vgren <mauritz.lovgren at hotmail.com>:
>> Clarification:
>>
>> Yes, the event returned from the background job is expceted to be fully
>> asynchronous and can arrive at any time in the future (independent of when
>> the bgapi commands were issued).
>> The issue is the reply from the bgapi command itself, _before_ it spawns
>> the
>> background thread.
>> If you, let's say, issue 10 bgapi commands in sequence, will the reply (OK
>> Job-UUID:.... / ERR) for each bgapi command return in the same sequence as
>> issued?
>>
>> Example:
>>
>> ESL socket outputstream (writer thread):
>> client --> bgapi status --> fs
>> client --> bgapi status --> fs
>> client --> bgapi status --> fs
>>
>> ESL socket inputstream (reader thread):
>> client <-- +OK Job-UUID:..... <-- fs
>> client <-- +OK Job-UUID:..... <-- fs
>> client <-- +OK Job-UUID:..... <-- fs
>>
>> The ESL socket outputstream and inputstream are written to and read using
>> a
>> writer and a reader thread. If we have no guarantee that the order of the
>> OK
>> replies from the bgapi commands are returned in the same order as issued,
>> we
>> will be forced to do the following:
>>
>> client --> bgapi status --> fs
>> wait for reply
>> client <-- +OK Job-UUID:..... <-- fs
>>
>> client --> bgapi status --> fs
>> wait for reply
>> client <-- +OK Job-UUID:..... <-- fs
>>
>> client --> bgapi status --> fs
>> wait for reply
>> client <-- +OK Job-UUID:..... <-- fs
>>
>> Only then could we securly map the command to the correct Job-UUID for the
>> reply.
>> But if the replies arrive in correct order we can put the issued commands
>> in
>> an ordered queue that can be processed with certainty by the socket reader
>> thread, and assign replies to the correct commands.. Did this really make
>> it
>> any clearer?? :-)
>>
>> Had there been a message sequence ID (or similar construct) there would be
>> no need to be order specific as we would always have an exact match
>> between
>> command and reply, but I don't think FS / CLI / ESL supports this at the
>> moment (or ever will... ;-)).
>>
>> - Mauritz
>>
>>
>> -----Original Message-----
>> From: Anthony Minessale
>> Sent: Friday, February 25, 2011 12:10 AM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Receive order guarantee for bgapi command
>> replies over ESL?
>>
>> bgapi commands are asynchronous that means when you send the command
>> it spawns a thread and runs in the background.
>> When its complete it will send you the event, its completely dependent
>> on how long the task takes that you asked it to run.
>> Its no different that doing 12 wgets to a website in the bg, there is
>> no telling when they will be done.
>>
>>
>>
>> 2011/2/24 Mauritz L?vgren <mauritz.lovgren at hotmail.com>:
>>> The use case is as follows;
>>>
>>> Multi-threaded Java applicaton that sends bgapi commands to several
>>> channels
>>> in parallel over the same ESL connection.
>>> (bgapi commands are serialized in order to the socket outputstream, but
>>> is
>>> there any guarantee that we will receive the OK / ERR reply for each
>>> bgapi
>>> in the correct order from the socket inputstream?)
>>>
>>> And, what about sending bgapi commands on the same socket while a
>>> ?regular?
>>> api command is currently executing? I suspect the bgapi reply will delay
>>> until the ?regular? (blocking) api command is finished processing?
>>>
>>> - Mauritz
>>>
>>> From: Michael Collins
>>> Sent: Thursday, February 24, 2011 9:44 PM
>>> To: FreeSWITCH Users Help
>>> Subject: Re: [Freeswitch-users] Receive order guarantee for bgapi command
>>> replies over ESL?
>>>
>>> Just curious - what is the use case where sending a stack of bgapi's
>>> without
>>> listening for the reply is more desirable than looping through the bgapi
>>> send/reply sequence for each one? I'm no super programmer but it seems to
>>> me
>>> that blindly relying on another system to always send stuff in the exact
>>> correct order is dangerous and may cause bugs that are difficult to
>>> diagnose. Personally I would consider it a programming best practice to
>>> wait
>>> for the response of the bgapi before sending another one.
>>>
>>> I invite other socket programmers to give their input...
>>>
>>> -MC
>>>
>>> P.S - you may wish to disable the legal notice at the and of your emails
>>> when sending to a public list. (We know that some servers tag outgoing
>>> messages automatically and if that is your case we understand. Go yell at
>>> the I.T. guy! :)
>>>
>>> 2011/2/24 Mauritz L?vgren <mauritz.lovgren at hotmail.com>
>>>>
>>>> Hi,
>>>>
>>>> Are replies for bgapi commands sent back to an inbound socket connection
>>>> in the same order as the commands were sent?
>>>>
>>>> Example: sending 10 bgapi messages from client to freeswitch socket with
>>>> no delay inbetween.
>>>> Will the response for those command messages be sent back by freeswitch
>>>> in
>>>> the exact same order as their commands were received, or should one wait
>>>> for
>>>> each reply before sending a new bgapi command to be sure one gets a
>>>> reply
>>>> for the correct command?
>>>>
>>>> If there was an option of providing a UUID (or sequenceId) for the
>>>> command, it would be easier to match the reply with the request upon
>>>> receival, but this doesn?t seem to be possible with the current
>>>> implementation?
>>>>
>>>> Regards,
>>>> Mauritz Lovgren
>>>> Systems Architect
>>>> IPLink Inc.
>>>
>>>
>>>
>>> ________________________________
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>



-- 
david varnes

e: david.varnes at gmail.com
p: +61 404 925 633


From peter.olsson at visionutveckling.se  Fri Feb 25 15:25:41 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 25 Feb 2011 13:25:41 +0100
Subject: [Freeswitch-users] ESL in Delphi/Pascal
In-Reply-To: <SNT144-w572265F9927858BC1E69EFF0D80@phx.gbl>
References: <SNT144-w572265F9927858BC1E69EFF0D80@phx.gbl>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58ADF8A9DC@cooper>

I have implemented my own ESL "lib" in Delphi (native). I will release it as open source as soon as I think it's working good enough :) Hopefully within a couple of months or so.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Dave Bracken
Skickat: den 22 februari 2011 22:56
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] ESL in Delphi/Pascal

Can anyone help me to figure out how to use the ESL in Delphi? We have tried compiling the files in VS2008, and it produces a .lib file and some .obj files, but none of which can be linked into delphi. I do not know hardly anything about C, so i am kinda of flying blind here. Any help would be very greatly appreciated.
DelphiGuy !DSPAM:4d644a9432762115986248!
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From kris at livecall.com  Fri Feb 25 16:05:13 2011
From: kris at livecall.com (Kris)
Date: Fri, 25 Feb 2011 05:05:13 -0800
Subject: [Freeswitch-users] problem with voice quality
References: <AANLkTik15iq+zwivCJWZeVFOmDVvAFB70v7haeK9XE2K@mail.gmail.com>
Message-ID: <1CC6662BE5B94BD9A2C7DF20A3FDD577@stor1>

I use X-Lite to call into my local freeswitch for testing and it works fine, 
but when I call into the weekly conference, everyone hears noise so I use 
Linphone or Skype for that. It's strange..

----- Original Message ----- 
From: "joy this" <thisjoy0528 at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, February 24, 2011 9:49 PM
Subject: [Freeswitch-users] problem with voice quality


> Dear all:
>
>
>
>        The scenario is that, when I use 3CX (PCMA) to call XLite, 3CX will
> hear loud noise but XLite is fine. And I found that it would only happen
> when XLite supports PCMA and PCMU simultaneously. If XLite only support
> PCMA, everything is fine.
>
> Maybe it is because of the old version of Freeswitch, I download the 
> latest
> version yesterday and re-test. The result is that 3CX hear nothing instead
> of noise. I use Wireshark to survey packages, and the server does not
> transfer UDP packages to 3CX. I check the log file, it is as below.
>
>
>
> -----3CX hears nothing-----
>
> 2011-02-24 20:17:30.724125 [DEBUG] sofia.c:4683 Channel
> sofia/internal/1000 at thisjoy.x.x:5060 entering state [completed][200]
>
> 2011-02-24 20:17:30.724125 [DEBUG] switch_core_session.c:709 Send signal
> sofia/internal/1000 at thisjoy.x.x:5060 [BREAK]
>
> 2011-02-24 20:17:30.724125 [DEBUG] switch_channel.c:2790
> (sofia/internal/1000 at thisjoy.x.x:5060) Callstate Change EARLY -> ACTIVE
>
> 2011-02-24 20:17:30.724125 [NOTICE] switch_ivr_originate.c:3363 Channel
> [sofia/internal/1000 at thisjoy.x.x:5060] has been answered
>
> 2011-02-24 20:17:30.724125 [DEBUG] switch_ivr_originate.c:3408 Originate
> Resulted in Success: [sofia/internal/sip:1001 at 114.24.x.x:56922]
>
> 2011-02-24 20:17:30.739750 [DEBUG] switch_ivr_originate.c:3408 Originate
> Resulted in Success: [sofia/internal/sip:1001 at 114.24.x.x:56922]
>
> 2011-02-24 20:17:30.739750 [DEBUG] switch_core_session.c:709 Send signal
> sofia/internal/sip:1001 at 114.24.x.x:56922 [BREAK]
>
> 2011-02-24 20:17:30.739750 [DEBUG] switch_core_session.c:709 Send signal
> sofia/internal/1000 at thisjoy.x.x:5060 [BREAK]
>
> 2011-02-24 20:17:30.739750 [DEBUG] switch_ivr_bridge.c:1234
> (sofia/internal/sip:1001 at 114.24.x.x:56922) State Change 
> CS_CONSUME_MEDIA ->
> CS_EXCHANGE_MEDIA
>
> 2011-02-24 20:17:30.739750 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/internal/sip:1001 at 114.24.x.x:56922 [BREAK]
>
> 2011-02-24 20:17:30.739750 [DEBUG] switch_core_state_machine.c:320
> (sofia/internal/sip:1001 at 114.24.x.x:56922) Running State Change
> CS_EXCHANGE_MEDIA
>
> 2011-02-24 20:17:30.739750 [DEBUG] switch_core_state_machine.c:369
> (sofia/internal/sip:1001 at 114.24.x.x:56922) State EXCHANGE_MEDIA
>
> 2011-02-24 20:17:30.739750 [DEBUG] mod_sofia.c:552 SOFIA EXCHANGE_MEDIA
>
> 2011-02-24 20:17:30.974125 [DEBUG] switch_rtp.c:2985 Correct ip/port
> confirmed.
>
> 2011-02-24 20:17:30.974125 [DEBUG] sofia.c:4683 Channel
> sofia/internal/1000 at thisjoy.x.x:5060 entering state [ready][200]
>
> 2011-02-24 20:17:31.021000 [DEBUG] switch_core_session.c:771 Send signal
> sofia/internal/sip:1001 at 114.24.x.x:56922 [BREAK]
>
> 2011-02-24 20:17:31.021000 [DEBUG] switch_core_session.c:771 Send signal
> sofia/internal/1000 at thisjoy.x.x:5060 [BREAK]
>
> -----log end-----
>
>
>
> When XLite only supports PCMA, it would be an additional log:
>
> 2011-02-24 20:16:06.661625 [INFO] switch_rtp.c:2968 Auto Changing port 
> from
> 192.168.1.1:51508 to 114.24.x.x:51508
>
>        And I test it again by using a public VoIP server (
> sip.lowratevoip.com), there is no noise or silence situation, everything 
> is
> fine.
>
>        Does XLite get bugs? Why the public server can deal with the
> situation? Could anyone help me?
>
>
> Sincerely yours,
>
> thisjoy.
> 



From Nabble at slickdeals.endjunk.com  Fri Feb 25 17:13:08 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 25 Feb 2011 06:13:08 -0800 (PST)
Subject: [Freeswitch-users] implementing stereo (e.g. portaudio to
	test	it easily)
In-Reply-To: <000801cbd4d5$13362830$39a27890$@gmail.com>
References: <AANLkTikL9SQPdsJ6XF-AUZuLQ3Z-AG7Eqgj2BsL=CZ4S@mail.gmail.com>
	<AANLkTinowzmzt8NxB-FtifkZEQZ8s3OQAWLuC18PN60Q@mail.gmail.com>
	<AANLkTikDFqS_Y9XS-gnVvnfPhAW1Eri=zy4XB3vAS2N0@mail.gmail.com>
	<4d66ee22.857a0e0a.6c3f.0343@mx.google.com>
	<AANLkTimrx5q076f+EAjgMQBh4c4kZYE2Mjq401oV2V7z@mail.gmail.com>
	<000801cbd4d5$13362830$39a27890$@gmail.com>
Message-ID: <1298643188393-6064582.post@n2.nabble.com>



MatzeMuc86 wrote:
> I checked the code and found some positions where fh->channels have to be
> > 1.
This also depends on when the flags is set to SWITCH_FILE_FLAG_READ, right?
 
BTW, I am just curious what kind of application will you be making that
requires a stereo. Do you care to tell us? Perhaps, once the readers
understand what your application is, there may be some interests.


-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/implementing-stereo-e-g-portaudio-to-test-it-easily-tp6061596p6064582.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From misi at niif.hu  Fri Feb 25 17:19:24 2011
From: misi at niif.hu (=?ISO-8859-1?Q?M=C9SZ=C1ROS_Mih=E1ly?=)
Date: Fri, 25 Feb 2011 15:19:24 +0100
Subject: [Freeswitch-users] Freeswtich as a media proxy
 between	ipv4<=>ipv6 using Polycom HDX8006 SIP UA-s
In-Reply-To: <201102242152.02899.errotan@elder.hu>
References: <4D650823.5050305@niif.hu> <201102242152.02899.errotan@elder.hu>
Message-ID: <4D67BA6C.5050401@niif.hu>

Hi Zsolt,

I need media proxy between IPv6 and IPv4.
Not only a signaling proxy needed, so SIP Express Router is not enough.
I need also a Media proxy what can proxy many (and furthermore any) 
media like BFCP, FECC h.224, as i wrote before.

Thanks,
Misi

2011-02-24 21:52 keltez?ssel, Pusk?s Zsolt ?rta:
> Hi.
>
> If you just want a sip proxy use OpenSER or Opensips.
>
>
> 2011. febru?r 23. 14:14:11 d?tummal M?SZ?ROS Mih?ly az al?bbiakat ?rta:
>    
>> Hi,
>>
>>     1. 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)
>>     2. 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?
>>     3. 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
>    



From Nabble at slickdeals.endjunk.com  Fri Feb 25 17:20:35 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 25 Feb 2011 06:20:35 -0800 (PST)
Subject: [Freeswitch-users] problem with voice quality
In-Reply-To: <AANLkTik15iq+zwivCJWZeVFOmDVvAFB70v7haeK9XE2K@mail.gmail.com>
References: <AANLkTik15iq+zwivCJWZeVFOmDVvAFB70v7haeK9XE2K@mail.gmail.com>
Message-ID: <1298643635222-6064616.post@n2.nabble.com>



joy this wrote:
> The scenario is that, when I use 3CX (PCMA) to call XLite, 3CX will
> hear loud noise but XLite is fine. And I found that it would only happen
> when XLite supports PCMA and PCMU simultaneously. If XLite only support
> PCMA, everything is fine.
I am just curious if this problem persists if X-Lite is configured with ONLY
PCMU and not PCMA?

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/problem-with-voice-quality-tp6063485p6064616.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From thisjoy0528 at gmail.com  Fri Feb 25 17:31:43 2011
From: thisjoy0528 at gmail.com (joy this)
Date: Fri, 25 Feb 2011 22:31:43 +0800
Subject: [Freeswitch-users] problem with voice quality
In-Reply-To: <1CC6662BE5B94BD9A2C7DF20A3FDD577@stor1>
References: <AANLkTik15iq+zwivCJWZeVFOmDVvAFB70v7haeK9XE2K@mail.gmail.com>
	<1CC6662BE5B94BD9A2C7DF20A3FDD577@stor1>
Message-ID: <AANLkTim5US0xk1pYHyomm-_VwA1BQtACJU4p46ahPPV8@mail.gmail.com>

    I found that the problem is cause by the codec. When someone calls XLite
(support PCMA and PCMU simultaneously), and Xlite decide to use PCMA, the
problem occurs. In fact, XLite will send the PCMU packages, and I guess it
makes the noise because Freeswitch does not trans-code. The public server is
without the problem, because it detect XLite does not send PCMA and
trans-code for XLite. Even XLite (support PCMA) calls XLite (support PCMA
and PCMU), a-leg hears noise.
    Does Freeswitch support the function which can trans-code the wrong
codec? Not everyone knows how to set the codec in the XLite phone, and I
want to make sure if Freeswitch can do the same thing like the public server
I mentioned.
Sincerely yours,
thisjoy.
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From thisjoy0528 at gmail.com  Fri Feb 25 17:38:00 2011
From: thisjoy0528 at gmail.com (joy this)
Date: Fri, 25 Feb 2011 22:38:00 +0800
Subject: [Freeswitch-users] problem with voice quality
In-Reply-To: <1298643635222-6064616.post@n2.nabble.com>
References: <AANLkTik15iq+zwivCJWZeVFOmDVvAFB70v7haeK9XE2K@mail.gmail.com>
	<1298643635222-6064616.post@n2.nabble.com>
Message-ID: <AANLkTiniv+p0n+LdRax5QMOmU8omK8gbqkHJ5Veim3q8@mail.gmail.com>

    When a-leg supports only PCMA, and b-leg (XLite) suppourts only PCMU,
everything is fine.

2011/2/25 mazilo <Nabble at slickdeals.endjunk.com>

>
>
> joy this wrote:
> > The scenario is that, when I use 3CX (PCMA) to call XLite, 3CX will
> > hear loud noise but XLite is fine. And I found that it would only happen
> > when XLite supports PCMA and PCMU simultaneously. If XLite only support
> > PCMA, everything is fine.
> I am just curious if this problem persists if X-Lite is configured with
> ONLY
> PCMU and not PCMA?
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/problem-with-voice-quality-tp6063485p6064616.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  Fri Feb 25 17:52:54 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 25 Feb 2011 14:52:54 +0000
Subject: [Freeswitch-users] problem with voice quality
In-Reply-To: <AANLkTim5US0xk1pYHyomm-_VwA1BQtACJU4p46ahPPV8@mail.gmail.com>
References: <AANLkTik15iq+zwivCJWZeVFOmDVvAFB70v7haeK9XE2K@mail.gmail.com>
	<1CC6662BE5B94BD9A2C7DF20A3FDD577@stor1>
	<AANLkTim5US0xk1pYHyomm-_VwA1BQtACJU4p46ahPPV8@mail.gmail.com>
Message-ID: <AANLkTik_zP1oX6Uwrk0MOA32ezXUp57K-DJ82BJdNeon@mail.gmail.com>

FS will always transcode if the a-leg and b-leg are using different
codecs. The only exception is on a small number of passthrough only
codecs (g729,g723.1,amr).

(That is assuming that one of the legs isn't lying about the type of
data it's sending).

Can you reproduce the sound problems on the very latest Git? Can you
hear speech with noise, or is it just garbage? 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.

-Steve




On 25 February 2011 14:31, joy this <thisjoy0528 at gmail.com> wrote:
> ??? I found that the problem is cause by the codec. When someone calls XLite
> (support PCMA and PCMU simultaneously), and Xlite decide to use PCMA, the
> problem occurs. In fact, XLite will send the PCMU packages, and I guess it
> makes the noise because Freeswitch does not trans-code. The public server is
> without the problem, because it detect XLite does not send PCMA and
> trans-code for XLite. Even XLite (support PCMA) calls XLite (support PCMA
> and PCMU), a-leg hears noise.
> ??? Does Freeswitch support the function which can trans-code the wrong
> codec? Not everyone knows how to set the codec in the XLite phone, and I
> want to make sure if Freeswitch can do the same thing like the public server
> I mentioned.
> 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
>
>


From Nabble at slickdeals.endjunk.com  Fri Feb 25 17:58:19 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 25 Feb 2011 06:58:19 -0800 (PST)
Subject: [Freeswitch-users] problem with voice quality
In-Reply-To: <AANLkTiniv+p0n+LdRax5QMOmU8omK8gbqkHJ5Veim3q8@mail.gmail.com>
References: <AANLkTik15iq+zwivCJWZeVFOmDVvAFB70v7haeK9XE2K@mail.gmail.com>
	<1298643635222-6064616.post@n2.nabble.com>
	<AANLkTiniv+p0n+LdRax5QMOmU8omK8gbqkHJ5Veim3q8@mail.gmail.com>
Message-ID: <1298645899558-6064746.post@n2.nabble.com>



joy this wrote:
> When a-leg supports only PCMA, and b-leg (XLite) suppourts only PCMU,
> everything is fine.
So, if X-Lite is configured with either PCMU or PCMA and not both, the
sudden loud voice won't occur, right? The reason I asked this question is I
noticed a Linksys SPA3K also produces a sudden loud noise on the callee side
during a VoIP/PSTN call and the Linksys SPA3K is configured with all CoDecs,
i.e. PCMA, PCMU, G723.1, G726, and G729. Perhaps, by disabling the PCMA or
PCMU CoDec, this will eliminate the sudden loud voice.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/problem-with-voice-quality-tp6063485p6064746.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Fri Feb 25 18:01:19 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 25 Feb 2011 15:01:19 +0000
Subject: [Freeswitch-users] Freeswtich as a media proxy between
 ipv4<=>ipv6 using Polycom HDX8006 SIP UA-s
In-Reply-To: <4D650823.5050305@niif.hu>
References: <4D650823.5050305@niif.hu>
Message-ID: <AANLkTinqtxZ94zKXiwFBWpyKkQPiOqzAPWt6_3EP5pqH@mail.gmail.com>

> 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.

> Further more I need to know that FreeSwitch can function as a real media
> proxy?

Yes, it does.

> 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.

-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
>
>


From matzemuc86 at gmail.com  Fri Feb 25 18:09:30 2011
From: matzemuc86 at gmail.com (MatzeMuc86)
Date: Fri, 25 Feb 2011 16:09:30 +0100
Subject: [Freeswitch-users] implementing stereo (e.g. portaudio
	to	test	it easily)
In-Reply-To: <1298643188393-6064582.post@n2.nabble.com>
References: <AANLkTikL9SQPdsJ6XF-AUZuLQ3Z-AG7Eqgj2BsL=CZ4S@mail.gmail.com>	<AANLkTinowzmzt8NxB-FtifkZEQZ8s3OQAWLuC18PN60Q@mail.gmail.com>	<AANLkTikDFqS_Y9XS-gnVvnfPhAW1Eri=zy4XB3vAS2N0@mail.gmail.com>	<4d66ee22.857a0e0a.6c3f.0343@mx.google.com>	<AANLkTimrx5q076f+EAjgMQBh4c4kZYE2Mjq401oV2V7z@mail.gmail.com>	<000801cbd4d5$13362830$39a27890$@gmail.com>
	<1298643188393-6064582.post@n2.nabble.com>
Message-ID: <000f01cbd4fe$015301f0$03f905d0$@gmail.com>

SWITCH_FILE_FLAG_READ: I'm not sure what this does. It's somewhere else in
the code.

My idea: A conference room for regular sip clients but also special user
clients which can receive stereo. Stereo is needed as those clients will get
a special HRTF sound: different people are placed on different 3D positions
in the room and you can hear them. This is NOT regular stereo sound but real
3D sound. Why is this better: Mixing all together to one mono channel, it is
very difficult to distinguish between different people.
Interesting links:
http://en.wikipedia.org/wiki/Head-related_transfer_function and
http://en.wikipedia.org/wiki/Cocktail_party_effect.
There have been several tests (and there is literature as well) that proofs
that it is much easier to follow conversations which use 3D sound like HRTF.
And I want to do this not by some proprietary software like it is done by
with Mumble by J. Feldmayer but with SIP to be able to connect also regular
SIP clients (and regular phones or whatever I want).

-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von mazilo
Gesendet: Freitag, 25. Februar 2011 15:13
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] implementing stereo (e.g. portaudio to test
it easily)



MatzeMuc86 wrote:
> I checked the code and found some positions where fh->channels have to 
> be
> > 1.
This also depends on when the flags is set to SWITCH_FILE_FLAG_READ, right?
 
BTW, I am just curious what kind of application will you be making that
requires a stereo. Do you care to tell us? Perhaps, once the readers
understand what your application is, there may be some interests.


-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/implementing-stereo-e-g-portau
dio-to-test-it-easily-tp6061596p6064582.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_______________________________________________
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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 Nabble at slickdeals.endjunk.com  Fri Feb 25 18:38:27 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 25 Feb 2011 07:38:27 -0800 (PST)
Subject: [Freeswitch-users] implementing stereo (e.g. portaudio
	to	test	it easily)
In-Reply-To: <000f01cbd4fe$015301f0$03f905d0$@gmail.com>
References: <AANLkTikL9SQPdsJ6XF-AUZuLQ3Z-AG7Eqgj2BsL=CZ4S@mail.gmail.com>
	<AANLkTinowzmzt8NxB-FtifkZEQZ8s3OQAWLuC18PN60Q@mail.gmail.com>
	<AANLkTikDFqS_Y9XS-gnVvnfPhAW1Eri=zy4XB3vAS2N0@mail.gmail.com>
	<4d66ee22.857a0e0a.6c3f.0343@mx.google.com>
	<AANLkTimrx5q076f+EAjgMQBh4c4kZYE2Mjq401oV2V7z@mail.gmail.com>
	<000801cbd4d5$13362830$39a27890$@gmail.com>
	<1298643188393-6064582.post@n2.nabble.com>
	<000f01cbd4fe$015301f0$03f905d0$@gmail.com>
Message-ID: <1298648307640-6064882.post@n2.nabble.com>



MatzeMuc86 wrote:
> My idea: A conference room for regular sip clients but also special user
> clients which can receive stereo. Stereo is needed as those clients will
> get
> a special HRTF sound: different people are placed on different 3D
> positions
> in the room and you can hear them. This is NOT regular stereo sound but
> real
> 3D sound. Why is this better: Mixing all together to one mono channel, it
> is
> very difficult to distinguish between different people.
> Interesting links:
> http://en.wikipedia.org/wiki/Head-related_transfer_function and
> http://en.wikipedia.org/wiki/Cocktail_party_effect.
> There have been several tests (and there is literature as well) that
> proofs
> that it is much easier to follow conversations which use 3D sound like
> HRTF.
> And I want to do this not by some proprietary software like it is done by
> with Mumble by J. Feldmayer but with SIP to be able to connect also
> regular
> SIP clients (and regular phones or whatever I want).
AFAIC, this indeed is a very interesting project. I am no expert nor have
any idea how mathematically complex this will become as the conference room
grows larger. I am hoping you will make the end product from project
available to public. Best of luck.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/implementing-stereo-e-g-portaudio-to-test-it-easily-tp6061596p6064882.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Fri Feb 25 18:53:05 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 25 Feb 2011 15:53:05 +0000
Subject: [Freeswitch-users] implementing stereo (e.g. portaudio to test
	it easily)
In-Reply-To: <000f01cbd4fe$015301f0$03f905d0$@gmail.com>
References: <AANLkTikL9SQPdsJ6XF-AUZuLQ3Z-AG7Eqgj2BsL=CZ4S@mail.gmail.com>
	<AANLkTinowzmzt8NxB-FtifkZEQZ8s3OQAWLuC18PN60Q@mail.gmail.com>
	<AANLkTikDFqS_Y9XS-gnVvnfPhAW1Eri=zy4XB3vAS2N0@mail.gmail.com>
	<4d66ee22.857a0e0a.6c3f.0343@mx.google.com>
	<AANLkTimrx5q076f+EAjgMQBh4c4kZYE2Mjq401oV2V7z@mail.gmail.com>
	<000801cbd4d5$13362830$39a27890$@gmail.com>
	<1298643188393-6064582.post@n2.nabble.com>
	<000f01cbd4fe$015301f0$03f905d0$@gmail.com>
Message-ID: <AANLkTikjch_cPU8XzzF4APjWDAcQNcodvtZU1C3OkKP_@mail.gmail.com>

> SWITCH_FILE_FLAG_READ: I'm not sure what this does. It's somewhere else in
> the code.

It specifies that the file is being opened for reading
(SWITCH_FILE_FLAG_READ) rather than writing (SWITCH_FILE_FLAG_WRITE).
They actually wrap similar values in the APR library.

-Steve



On 25 February 2011 15:09, MatzeMuc86 <matzemuc86 at gmail.com> wrote:
> SWITCH_FILE_FLAG_READ: I'm not sure what this does. It's somewhere else in
> the code.
>
> My idea: A conference room for regular sip clients but also special user
> clients which can receive stereo. Stereo is needed as those clients will get
> a special HRTF sound: different people are placed on different 3D positions
> in the room and you can hear them. This is NOT regular stereo sound but real
> 3D sound. Why is this better: Mixing all together to one mono channel, it is
> very difficult to distinguish between different people.
> Interesting links:
> http://en.wikipedia.org/wiki/Head-related_transfer_function and
> http://en.wikipedia.org/wiki/Cocktail_party_effect.
> There have been several tests (and there is literature as well) that proofs
> that it is much easier to follow conversations which use 3D sound like HRTF.
> And I want to do this not by some proprietary software like it is done by
> with Mumble by J. Feldmayer but with SIP to be able to connect also regular
> SIP clients (and regular phones or whatever I want).
>
> -----Urspr?ngliche Nachricht-----
> Von: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von mazilo
> Gesendet: Freitag, 25. Februar 2011 15:13
> An: freeswitch-users at lists.freeswitch.org
> Betreff: Re: [Freeswitch-users] implementing stereo (e.g. portaudio to test
> it easily)
>
>
>
> MatzeMuc86 wrote:
>> I checked the code and found some positions where fh->channels have to
>> be
>> > 1.
> This also depends on when the flags is set to SWITCH_FILE_FLAG_READ, right?
>
> BTW, I am just curious what kind of application will you be making that
> requires a stereo. Do you care to tell us? Perhaps, once the readers
> understand what your application is, there may be some interests.
>
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/implementing-stereo-e-g-portau
> dio-to-test-it-easily-tp6061596p6064582.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 brian at freeswitch.org  Fri Feb 25 19:18:02 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 25 Feb 2011 10:18:02 -0600
Subject: [Freeswitch-users] Polycom G722 negotiation
In-Reply-To: <AE216221-5892-4337-8985-1889410A866F@5ninesolutions.com>
References: <AE216221-5892-4337-8985-1889410A866F@5ninesolutions.com>
Message-ID: <2867CE13-6EED-4FAF-9DF1-49451D164721@freeswitch.org>

What version of FreeSWITCH are you running?

/b

On Feb 24, 2011, at 11:45 PM, Spencer Thomason wrote:

> Hello all,
> I have several Polycom IP-650s.  On the external profile I have:
> CODECS IN        	G722,PCMU,G729,GSM
> CODECS OUT       	PCMU,G729
> 
> On the internal profile I have
> CODECS IN        	G722,PCMU,G729,GSM
> CODECS OUT       	G722,PCMU,G729,GSM
> 
> The Polycoms have a similar codec priority set:
> v=0
> o=- 1167611879 1167611879 IN IP4 10.59.1.243
> s=Polycom IP Phone
> c=IN IP4 10.59.1.243
> t=0 0
> a=sendrecv
> m=audio 2236 RTP/AVP 9 0 18 101
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=yes
> a=rtpmap:101 telephone-event/8000
> 
> The issue I have is that if I leave the internal profile to generous and a user places a call from the PSTN on hold which is PCMU, when the user tries to pick up the call, the Polycoms' preference then changes the call to G722 and there is no audio.  If I set the profile to greedy, since the call is already PCMU Freeswitch keeps it at PCMU and audio is fine.  I realize this is the intended behavior for greedy vs generous but my question is why is there no audio when switching the codecs?  I didn't want to clutter this up any more but I'm happy to provide traces however the only real difference is the codec order.
> 
> Thanks,
> Spencer



From brian at freeswitch.org  Fri Feb 25 19:18:20 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 25 Feb 2011 10:18:20 -0600
Subject: [Freeswitch-users] Mitel 3300 ICP "Multicall" Equivalent
In-Reply-To: <4D66B5C5.7000005@integrafin.co.uk>
References: <4D66B5C5.7000005@integrafin.co.uk>
Message-ID: <9CCDC306-B282-4CC6-A648-B12A97398D9F@freeswitch.org>

You want Directed Call Pickup right?

/b

On Feb 24, 2011, at 1:47 PM, Alex Crow wrote:

> Hello list,
> 
> Our company is considering a migration from Mitel 3300 ICP to an 
> alternative platform, and of course FreeSwitch is a major contender, 
> being as we are not enamoured with "black boxes" and proprietary protocols.
> 
> There is a function in the Mitel that ideally we'd like to replicate (on 
> whatever hardphones are most appropriate). "Multicall" allows you to 
> assign one number to a button (other than the "main line" or default 
> extension) on several phones (up to 31 on the 3300) such that a call (or 
> any number of concurrent calls up to the number of phones configured 
> with that button)  to that number can be signalled on all phones and 
> picked up by anyone with that button configured on that phone. We use 
> that for calling direct to regional teams. No barging is allowed, and 
> the button only changes state (from off to flashing) when a call needs 
> answering on the multicall number. Said button assignment can be set to 
> either ring or not. Once the call has been answered by one user of the 
> multicall the light goes off on other configured extensions and another 
> incoming call can be answered in the same way by another extension, 
> regardless of the status of all other extensions with the same button 
> configured.
> 
> I was looking at the SCA/SLA stuff on the wiki, which is a close as I 
> could find to this behaviour, but barging is unwanted and we require the 
> line appearance, ringing and answering to be as described above.
> 
> My question, given the above, is:
> 
> 1. If SCA is able to do this, what is the best phone to support it 
> (Polycom seems good from the wiki)
> 2. If SCA is not appropriate, I have the feeling that queues would be 
> the next best option - if these can be assigned to a button on the phone 
> so much the better.
> 3. BLF doesn't look right as it seems it needs at least one extension 
> with that as its "primary" number.
> 
> There is some flexibilty here, but to ease transition it would be best 
> to keep the functionality as close as possible.
> 
> IMHO the Mitel way is really odd but it's got a lot of traction having 
> been used here for 7 years+.
> 
> 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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> 
> _______________________________________________
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org



From anthony.minessale at gmail.com  Fri Feb 25 19:30:46 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 25 Feb 2011 10:30:46 -0600
Subject: [Freeswitch-users] implementing stereo (e.g. portaudio to test
	it easily)
In-Reply-To: <AANLkTikjch_cPU8XzzF4APjWDAcQNcodvtZU1C3OkKP_@mail.gmail.com>
References: <AANLkTikL9SQPdsJ6XF-AUZuLQ3Z-AG7Eqgj2BsL=CZ4S@mail.gmail.com>
	<AANLkTinowzmzt8NxB-FtifkZEQZ8s3OQAWLuC18PN60Q@mail.gmail.com>
	<AANLkTikDFqS_Y9XS-gnVvnfPhAW1Eri=zy4XB3vAS2N0@mail.gmail.com>
	<4d66ee22.857a0e0a.6c3f.0343@mx.google.com>
	<AANLkTimrx5q076f+EAjgMQBh4c4kZYE2Mjq401oV2V7z@mail.gmail.com>
	<000801cbd4d5$13362830$39a27890$@gmail.com>
	<1298643188393-6064582.post@n2.nabble.com>
	<000f01cbd4fe$015301f0$03f905d0$@gmail.com>
	<AANLkTikjch_cPU8XzzF4APjWDAcQNcodvtZU1C3OkKP_@mail.gmail.com>
Message-ID: <AANLkTinrvj7Ye22GVu+CtaE+ApzP3OHGm=JdjsO1B2-i@mail.gmail.com>

Doing stereo codecs and playing stereo files are not the same.

The main reason we do not have stereo completely implemented is there
was not really any codec or devices that could handle it to bother
testing with.

We do record stereo files but as for playing them, there was not any
real motivation since there was very little you can do with it talking
to a mono sip phone.

Its probably possible to make playing files to portaudio work but you
would have to unblock the code that mixes it to mono and add support
also to portaudio to understand the difference in the streams.

If you want to do it yourself, like I believe I told you the first
time, use FreeSWITCH and VoIP the way it already works for a long time
until you completely understand the arena you are in before you try to
add a new side entrance.



On Fri, Feb 25, 2011 at 9:53 AM, Steven Ayre <steveayre at gmail.com> wrote:
>> SWITCH_FILE_FLAG_READ: I'm not sure what this does. It's somewhere else in
>> the code.
>
> It specifies that the file is being opened for reading
> (SWITCH_FILE_FLAG_READ) rather than writing (SWITCH_FILE_FLAG_WRITE).
> They actually wrap similar values in the APR library.
>
> -Steve
>
>
>
> On 25 February 2011 15:09, MatzeMuc86 <matzemuc86 at gmail.com> wrote:
>> SWITCH_FILE_FLAG_READ: I'm not sure what this does. It's somewhere else in
>> the code.
>>
>> My idea: A conference room for regular sip clients but also special user
>> clients which can receive stereo. Stereo is needed as those clients will get
>> a special HRTF sound: different people are placed on different 3D positions
>> in the room and you can hear them. This is NOT regular stereo sound but real
>> 3D sound. Why is this better: Mixing all together to one mono channel, it is
>> very difficult to distinguish between different people.
>> Interesting links:
>> http://en.wikipedia.org/wiki/Head-related_transfer_function and
>> http://en.wikipedia.org/wiki/Cocktail_party_effect.
>> There have been several tests (and there is literature as well) that proofs
>> that it is much easier to follow conversations which use 3D sound like HRTF.
>> And I want to do this not by some proprietary software like it is done by
>> with Mumble by J. Feldmayer but with SIP to be able to connect also regular
>> SIP clients (and regular phones or whatever I want).
>>
>> -----Urspr?ngliche Nachricht-----
>> Von: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von mazilo
>> Gesendet: Freitag, 25. Februar 2011 15:13
>> An: freeswitch-users at lists.freeswitch.org
>> Betreff: Re: [Freeswitch-users] implementing stereo (e.g. portaudio to test
>> it easily)
>>
>>
>>
>> MatzeMuc86 wrote:
>>> I checked the code and found some positions where fh->channels have to
>>> be
>>> > 1.
>> This also depends on when the flags is set to SWITCH_FILE_FLAG_READ, right?
>>
>> BTW, I am just curious what kind of application will you be making that
>> requires a stereo. Do you care to tell us? Perhaps, once the readers
>> understand what your application is, there may be some interests.
>>
>>
>> -----
>> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/implementing-stereo-e-g-portau
>> dio-to-test-it-easily-tp6061596p6064582.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
>



-- 
Anthony Minessale II

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


From brian at freeswitch.org  Fri Feb 25 19:32:51 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 25 Feb 2011 10:32:51 -0600
Subject: [Freeswitch-users] t38_gateway self - no re-Invite?
In-Reply-To: <4D6381C0.7040408@ewetel.de>
References: <1B19ABD72889C245AE8EEE08AC24103A28C1AFC00E@exmachina.office.kapper.net>
	<4D6381C0.7040408@ewetel.de>
Message-ID: <B072DB39-6AED-46B6-A640-4B629FB08D4F@freeswitch.org>

Helmet,
	If you had to change the code why do you not Open a JIRA?  And how are you doing this that it doesn't work?  I have many people using this without a single problem.

/b

On Feb 22, 2011, at 3:28 AM, Helmut Kuper wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
> 
> Hello Clemens,
> 
> I'm using t38 a few month now. And I have had also big problems to get
> FS and mod_spandsp as a t38-Gateway up and running. The only way I found
> to get it work in most cases was to change the code in mod_spandsp.c
> 
> I changed line 99 from
> switch_ivr_tone_detect_session(session, "t38", "1100", "rw", timeout, 1,
> data, NULL, t38_gateway_start)
> 
> to
> switch_ivr_tone_detect_session(session, "t38", "2100", "rw", timeout, 1,
> data, NULL, t38_gateway_start)
> 
> The recompile mod_spandsp and reload it into FS.
> 
> This could affected the inbound way. So you have to play with the
> frequency to detect. Maybe you have to listen to both 1100Hz and 2100Hz.
> 
> t38 is not really easy with mod_spandsp.



From anthony.minessale at gmail.com  Fri Feb 25 19:32:59 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 25 Feb 2011 10:32:59 -0600
Subject: [Freeswitch-users] problem with voice quality
In-Reply-To: <1298645899558-6064746.post@n2.nabble.com>
References: <AANLkTik15iq+zwivCJWZeVFOmDVvAFB70v7haeK9XE2K@mail.gmail.com>
	<1298643635222-6064616.post@n2.nabble.com>
	<AANLkTiniv+p0n+LdRax5QMOmU8omK8gbqkHJ5Veim3q8@mail.gmail.com>
	<1298645899558-6064746.post@n2.nabble.com>
Message-ID: <AANLkTi=4srr7mFx2cAY3v+KL831o8BG_b1MpLbBntSp-@mail.gmail.com>

I think x-lite has had a long standing bug where it does not properly
negotiate PCMA.
The loud noise is what you get when you say you are using PCMA and
sending PCMU instead.


On Fri, Feb 25, 2011 at 8:58 AM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
>
> joy this wrote:
>> When a-leg supports only PCMA, and b-leg (XLite) suppourts only PCMU,
>> everything is fine.
> So, if X-Lite is configured with either PCMU or PCMA and not both, the
> sudden loud voice won't occur, right? The reason I asked this question is I
> noticed a Linksys SPA3K also produces a sudden loud noise on the callee side
> during a VoIP/PSTN call and the Linksys SPA3K is configured with all CoDecs,
> i.e. PCMA, PCMU, G723.1, G726, and G729. Perhaps, by disabling the PCMA or
> PCMU CoDec, this will eliminate the sudden loud voice.
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/problem-with-voice-quality-tp6063485p6064746.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  Fri Feb 25 19:36:31 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 25 Feb 2011 11:36:31 -0500
Subject: [Freeswitch-users] send fax in one leg call
Message-ID: <3505B4B09EB64A1E8BA155FF076E0776@e1705>

I'm trying to send a fax to a one leg call.

for example a registered user wants to send a fax to another registered user
so I have this in my default dialplan

         <extension name="internal_fax">
                <condition field="${sip_to_uri}" expression="^(\+)(\d{10})@$${domain}$">
                        <action application="ring_ready"/>
                        <action application="set" data="execute_on_media=tone_detect fax 1100 r +5000 transfer 'receivefax XML features' 2" />
                        <action application="sleep" data="2000"/>
                        <action application="answer"/>
                        <action application="voicemail" data="default ${domain_name} ${sip_to_user}"/>
                </condition>
        </extension>

for now I have no other way to send the fax from CLI like this:
fs_cli -x "uuid_broadcast <uuid> execute_extension::' sendfax XML features'  <aleg=timeout??> <bleg=no bleg??>"

thanks

Franck
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From anthony.minessale at gmail.com  Fri Feb 25 20:02:03 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 25 Feb 2011 11:02:03 -0600
Subject: [Freeswitch-users] Second transfer
In-Reply-To: <AANLkTimKPV=LY45dNXzdVZ0SzmnBCMv7UoaRR3_8PDz_@mail.gmail.com>
References: <AANLkTi=pB-Rt7j1cvNGgSs2s3F4h6jse1nTqTqRr8kKJ@mail.gmail.com>
	<AANLkTi=fvkp_8e4gXthWtYQTLAyxj+bbUMtdpAhXQamQ@mail.gmail.com>
	<AANLkTinw4CWxe+GW31AcMnzWWNsQU=W7WGSmsr2ky9zg@mail.gmail.com>
	<AANLkTikWMF2rf=ASaTMg6bQ=jw7KQu59wJuXr+fRj0Rg@mail.gmail.com>
	<AANLkTin6wdxuB3N4oxOGobY0Q7r3n33=wEz2T49eN9yt@mail.gmail.com>
	<AANLkTimKPV=LY45dNXzdVZ0SzmnBCMv7UoaRR3_8PDz_@mail.gmail.com>
Message-ID: <AANLkTimNxu60W+uHLm7-9Em+T4BS83RvyYCangcB8T-p@mail.gmail.com>

The 'i' makes it execute the app right there on the spot vs queuing it
to be executed async by the session's thread in a similar way to how
ESL queues an app.



On Thu, Feb 24, 2011 at 9:07 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Thu, Feb 24, 2011 at 6:58 PM, Michael Collins <msc at freeswitch.org> wrote:
>>
>>
>> On Thu, Feb 24, 2011 at 3:11 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>>
>>> add the 1 flag to the flags
>>> ction application="bind_meta_app" data="3 b s1 log::INFO> Transfer"/>
>>>
>>> this says it can only work once then detaches itself.
>>>
>> Awesome. The I'll get this properly wikified.
>> -MC
>
> Wiki has been updated to reflect the proper flags that can be set. Also, I
> happened to stumble across "unbind_meta_app" which I'm assuming could also
> be used.
> -MC
> P.S. - The i flag says to execute inline... what exactly does that do?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Fri Feb 25 20:17:51 2011
From: mthakershi at gmail.com (Malay Thakershi)
Date: Fri, 25 Feb 2011 11:17:51 -0600
Subject: [Freeswitch-users] Using 16 KHz sounds
In-Reply-To: <AANLkTinKGfnq+escY62cAdWvM4vaut8La5e5EtzpoqGo@mail.gmail.com>
References: <AANLkTim5CsCL4unpWh1OvH13Xx4HrKX2KF7Jg6UQYAZR@mail.gmail.com>
	<AANLkTin9tYDVJC0KUncV0h=g0+u=833QcRtp49OfU2XC@mail.gmail.com>
	<AANLkTi=YkDwu5ANjYFFWGDR2pmuscsJUDcK09QL92Kqo@mail.gmail.com>
	<AANLkTi=nUC=XEtb7rod7TjHwfaXwbK=c3=tz4MVrxrHz@mail.gmail.com>
	<AANLkTim7C9ATOtyEehBVXub-DSrUXQsRr25fKDDHdsaq@mail.gmail.com>
	<AANLkTinKGfnq+escY62cAdWvM4vaut8La5e5EtzpoqGo@mail.gmail.com>
Message-ID: <AANLkTi=1LChFsOGv2X-Yxxq7oZLEzUPcUPCDAzufQ2g9@mail.gmail.com>

OK. I am on my way to find out what you've suggested to isolate my issues.

I already tried latest GIT version. I will be deploying it shortly.

I should get back to this post with my findings.

Thank you.

On Wed, Feb 23, 2011 at 11:54 AM, Michael Collins <msc at freeswitch.org>wrote:

> It sounds very much like you have standard 8k calls. Increasing the
> sampling rate won't help since FS would have to downsample to 8k on the call
> leg anyway.
>
> It's time to go back to the original issue. You are having sound quality
> issues, correct? It's time to roll up your sleeves and do some detective
> work:
>
> What kind of network are you running? What routers, switches, NAT devices,
> and other computers are using the network? What kind of system is FS running
> on? Any virtualization being used? What OS?
>
> Are you running the latest git of FS?
>
> What SIP clients have you tried? Can you reproduce the sound quality issues
> on all of your SIP clients? Can you reproduce with different SIP clients on
> a different computer? Do you have a hard phone and do the symptoms persist
> there?
>
> Are you having sound quality issues in one direction or both directions?
> Have you done a tcpdump of the traffic and analyzed in wireshark?
>
> Those are all questions worth pursuing. The idea is to narrow the symptoms
> down as much as possible. I know it's not fun, but then again, this is
> telephony. :)
>
> -MC
>
> On Wed, Feb 23, 2011 at 9:31 AM, Malay Thakershi <mthakershi at gmail.com>wrote:
>
>> I don't use Sipura. I use FS to make / receive calls from mobile phones /
>> regular land line phones.
>>
>> Unlike what I said in my previous email, I am still using Allison-8kHz
>> voice. Somehow in managed code I had Allison-16kHz specified.
>>
>> I created three WAV files using Cepstral SWIFT command with 8000, 16000,
>> 22000 Hz.  When I play each file, the later two give me message at the FS
>> console "Sample rates don't match".
>>
>> Is there a setting where I can ask FS to sample at a higher rate that
>> would help me with sound quality issues? Is having a good sound card on the
>> server a good practice?
>>
>> Thank you for replies.
>>
>> Malay
>>
>>
>> On Wed, Feb 23, 2011 at 10:55 AM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> Depends, are you using a sipura? if so, try it, the setting is on the
>>> web ui of the phone/device not in FS.
>>>
>>>
>>> On Tue, Feb 22, 2011 at 12:25 PM, Malay Thakershi <mthakershi at gmail.com>
>>> wrote:
>>> > I found I am already using 16 KHz
>>> profile. .SetTtsParameters("cepstral",
>>> > "Allison-16kHz");
>>> > I read this under FS wiki on Cepstral under 'Gotchas':
>>> > -------------
>>> > Using a 16khz voice and 0.03 for RTP Packet Size (Sipura Setting), it
>>> will
>>> > sound horrible. Workaround: modify RTP Packet Size to 0.02 in Sipura
>>> config,
>>> > under Advanced/SIP section.
>>> > -------------
>>> > Do you think that is my problem? Is this to be done in FS
>>> configuration?
>>> > Malay
>>> > On Mon, Feb 21, 2011 at 2:35 PM, Michael Collins <msc at freeswitch.org>
>>> wrote:
>>> >>
>>> >> 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
>>> >>>
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at 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 santiagosoares at gmail.com  Fri Feb 25 20:27:13 2011
From: santiagosoares at gmail.com (Santiago Soares)
Date: Fri, 25 Feb 2011 14:27:13 -0300
Subject: [Freeswitch-users] Second transfer
In-Reply-To: <AANLkTimNxu60W+uHLm7-9Em+T4BS83RvyYCangcB8T-p@mail.gmail.com>
References: <AANLkTi=pB-Rt7j1cvNGgSs2s3F4h6jse1nTqTqRr8kKJ@mail.gmail.com>
	<AANLkTi=fvkp_8e4gXthWtYQTLAyxj+bbUMtdpAhXQamQ@mail.gmail.com>
	<AANLkTinw4CWxe+GW31AcMnzWWNsQU=W7WGSmsr2ky9zg@mail.gmail.com>
	<AANLkTikWMF2rf=ASaTMg6bQ=jw7KQu59wJuXr+fRj0Rg@mail.gmail.com>
	<AANLkTin6wdxuB3N4oxOGobY0Q7r3n33=wEz2T49eN9yt@mail.gmail.com>
	<AANLkTimKPV=LY45dNXzdVZ0SzmnBCMv7UoaRR3_8PDz_@mail.gmail.com>
	<AANLkTimNxu60W+uHLm7-9Em+T4BS83RvyYCangcB8T-p@mail.gmail.com>
Message-ID: <AANLkTikdb59NVaSiKhseTgXV=Kz1UEP3evfzxnuFq9Uu@mail.gmail.com>

It worked!
Thank you very much!

Santiago Soares
Fone: (41) 8488-0537


On Fri, Feb 25, 2011 at 2:02 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> The 'i' makes it execute the app right there on the spot vs queuing it
> to be executed async by the session's thread in a similar way to how
> ESL queues an app.
>
>
>
> On Thu, Feb 24, 2011 at 9:07 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> >
> >
> > On Thu, Feb 24, 2011 at 6:58 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> >>
> >>
> >> On Thu, Feb 24, 2011 at 3:11 PM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >>>
> >>> add the 1 flag to the flags
> >>> ction application="bind_meta_app" data="3 b s1 log::INFO> Transfer"/>
> >>>
> >>> this says it can only work once then detaches itself.
> >>>
> >> Awesome. The I'll get this properly wikified.
> >> -MC
> >
> > Wiki has been updated to reflect the proper flags that can be set. Also,
> I
> > happened to stumble across "unbind_meta_app" which I'm assuming could
> also
> > be used.
> > -MC
> > P.S. - The i flag says to execute inline... what exactly does that do?
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 mthakershi at gmail.com  Fri Feb 25 20:34:51 2011
From: mthakershi at gmail.com (Malay Thakershi)
Date: Fri, 25 Feb 2011 11:34:51 -0600
Subject: [Freeswitch-users] PlayAndGetDigits parameters
Message-ID: <AANLkTi=p1c=e+F=nW0hPCXRX4bf3S5zAXMRBPGfVGJkD@mail.gmail.com>

Recently I upgraded to latest GIT version and found there was an extra
parameter to this function.

I have two questions if someone can explain:

--> 4th parameter
* timeout = Number of milliseconds to wait once the file is done playing
before you type a digit.
>From the name, I alway thought this had to do with user's inactivity at the
prompt. So if value is 3000, after 3 seconds of inactivity prompt will
either do a retry or move on.
But I think from wiki description, it seems # of milliseconds for which data
entry will be blocked, am I correct?

--> 10th parameter
* digit_timeout = Number of milliseconds to wait after DTMF ( Added
Here[1]<http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=cfa30468e89788d220ad9fbc07ab1ae3406cf3fa>
 )
What does this do? Is it # ms to wait before second digit can be pressed?

I have following function in mod_managed code.

                        pStrLivePreObInput =
mObjMainSession.PlayAndGetDigits(5, 5, 3, 5000, "",
                        pStrObGreetTextFile.Replace(".txt", ".wav"),

 @"C:\FreeSWITCH\sounds\en\us\chAsmt\static\error\invalid_number.wav",
                        "^\\d", "", 3000);

Does that mean user has to wait 5 seconds before a key entry can be
accepted?
And there has to be 3 seconds gap between each key?

If so, I will need to redo the timing. Please let me know.

Malay
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From acrow at integrafin.co.uk  Fri Feb 25 20:47:22 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Fri, 25 Feb 2011 17:47:22 +0000
Subject: [Freeswitch-users] Mitel 3300 ICP "Multicall" Equivalent
In-Reply-To: <9CCDC306-B282-4CC6-A648-B12A97398D9F@freeswitch.org>
References: <4D66B5C5.7000005@integrafin.co.uk>
	<9CCDC306-B282-4CC6-A648-B12A97398D9F@freeswitch.org>
Message-ID: <4D67EB2A.80409@integrafin.co.uk>

On 25/02/11 16:18, Brian West wrote:
> You want Directed Call Pickup right?
>
> /b
>
> On Feb 24, 2011, at 1:47 PM, Alex Crow wrote:
>
>

Hi Brian,

Thanks for your reply - makes perfect sense - if it's Asterisk style 
**<multicall_number> on a key with a SUBSCRIBE to the same number on the 
phone. I am concerned that one of the phones actually would have to be 
registered to that extension to make it work, or is that not correct? 
This is why I was looking at SLA.

The other part of this is that the <multicall_number> must not go to 
voicemail or busy tone until capacity is reached, ie. all the people 
with the BLF button configured are busy. I tried this with 
TrixBox/Asterisk but as soon as one call was live all further calls to 
the same number gave voicemail or busy.

If both of the above work "just great" in FreeSwitch then you may well 
have found my solution.

I haven't attempted what you describe with FS as I don't have enough 
phones set up at the moment to do it, and I currently only have 1 
Grandstream GXP2000 (needs FreeSwitch code hacking for BLF to work even 
half-properly) and a shedload of Mitel (SIP Mode 5224/5324) where the 
BLF doesn't work at all. Softphones are not under consideration at 
present. I'm quite able to get a couple of Polycoms or Snoms but I'd 
need to justify the expense to management.

Cheers

Alex



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From wstephen80 at gmail.com  Fri Feb 25 21:16:12 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Fri, 25 Feb 2011 19:16:12 +0100
Subject: [Freeswitch-users] ESL socket outbound: clean way to close a
	session?
Message-ID: <AANLkTinPRgxuXRwCrB2WxFb4i5yTNS7MrQ6nVRyzMaJs@mail.gmail.com>

I have problem to clean close an outbound session.

I'm using the "esl_oop.h/esl_oop.cpp" files and my application is developed
in c++ and at startup I do a "linger" command.

My event consuming loop is:

while (run)
{
    ESLevent * event = connection.recvEventTimed(sleeptime);


   ... event processing ...
}

This application handle an inbound call (legA), originate an outbound call
(legB) and bridge the two calls.

The problem is present when the legA hangup the call: sometimes is received
the content-type="text/disconnect-notice" (without receiving
"CHANNEL_HANGUP") and the "recvEventTimed" exits with event=NULL.

In this case it's not possible to receive any event or send any other
message to Freeswitch because the "connection.execute" fails.

In this situation, if I have originated (with '& park') an outbound call,
this call remain in parked state and the application cannot hangup it.

What I'm doing wrong?
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From eddie at eddiecaplan.com  Fri Feb 25 19:11:29 2011
From: eddie at eddiecaplan.com (egc52556)
Date: Fri, 25 Feb 2011 08:11:29 -0800 (PST)
Subject: [Freeswitch-users] LRN/LNP lookups..
In-Reply-To: <f78f09e0-8354-496c-eb8d-fc6baf0b547e@me.com>
References: <AANLkTim-=9v+dMn_5cqtY4-Zae5JQAZ_usoE=XmMW17R@mail.gmail.com>
	<f78f09e0-8354-496c-eb8d-fc6baf0b547e@me.com>
Message-ID: <1298650289888-6064996.post@n2.nabble.com>


I appreciate there is no complete database of current LRN to download.  But
is there a free service where I can do a few lookups, say no more than 10-20
a day?

-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/LRN-LNP-lookups-tp5579895p6064996.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From sos at sokhapkin.dyndns.org  Fri Feb 25 21:38:18 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 25 Feb 2011 13:38:18 -0500
Subject: [Freeswitch-users] LRN/LNP lookups..
In-Reply-To: <1298650289888-6064996.post@n2.nabble.com>
References: <AANLkTim-=9v+dMn_5cqtY4-Zae5JQAZ_usoE=XmMW17R@mail.gmail.com>
	<f78f09e0-8354-496c-eb8d-fc6baf0b547e@me.com>
	<1298650289888-6064996.post@n2.nabble.com>
Message-ID: <201102251338.18446.sos@sokhapkin.dyndns.org>

There is no free service, but LRN lookup price is a fraction of cent per 
query, why not pay a few cents a day?

On Friday 25 February 2011, egc52556 wrote:
> I appreciate there is no complete database of current LRN to download.  But
> is there a free service where I can do a few lookups, say no more than
> 10-20 a day?



From helmut.kuper at ewetel.de  Fri Feb 25 21:46:38 2011
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Fri, 25 Feb 2011 19:46:38 +0100
Subject: [Freeswitch-users] t38_gateway self - no re-Invite?
In-Reply-To: <B072DB39-6AED-46B6-A640-4B629FB08D4F@freeswitch.org>
References: <1B19ABD72889C245AE8EEE08AC24103A28C1AFC00E@exmachina.office.kapper.net>	<4D6381C0.7040408@ewetel.de>
	<B072DB39-6AED-46B6-A640-4B629FB08D4F@freeswitch.org>
Message-ID: <4D67F90E.2090504@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Brian,

I never opened a Jira because, I have the same thoughts as you. The code
should work without a change. It is logical to listen to 1100 instead to
2100.

I have no idea why this problems happens here on my site. Maybe there is
a bug in the code, not very likely I guess. Maybe it is a special
problem with Grandstream ATA HT502. Maybe it is a wrong configuration in
my FS e.g wrong codec sample rate as Anthony mentioned.

The hack just helped to get it working for me.

I'm still lost a bit with my t38 problem.


Am 25.02.2011 17:32, schrieb Brian West:
> Helmet,
> 	If you had to change the code why do you not Open a JIRA?  And how are you doing this that it doesn't work?  I have many people using this without a single problem.
> 
> /b
> 

best regards
helmut
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Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/

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7CkAn2fiLEJFecvhBxce8zHDpr+nNZYI
=UAy8
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From msc at freeswitch.org  Fri Feb 25 21:52:58 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 25 Feb 2011 10:52:58 -0800
Subject: [Freeswitch-users] PlayAndGetDigits parameters
In-Reply-To: <AANLkTi=p1c=e+F=nW0hPCXRX4bf3S5zAXMRBPGfVGJkD@mail.gmail.com>
References: <AANLkTi=p1c=e+F=nW0hPCXRX4bf3S5zAXMRBPGfVGJkD@mail.gmail.com>
Message-ID: <AANLkTi=i8H2uk0mmE2vF6_ugPRcmicq2M5b8FU06vdd=@mail.gmail.com>

On Fri, Feb 25, 2011 at 9:34 AM, Malay Thakershi <mthakershi at gmail.com>wrote:

> Recently I upgraded to latest GIT version and found there was an extra
> parameter to this function.
>
> I have two questions if someone can explain:
>
> --> 4th parameter
> * timeout = Number of milliseconds to wait once the file is done playing
> before you type a digit.
> From the name, I alway thought this had to do with user's inactivity at the
> prompt. So if value is 3000, after 3 seconds of inactivity prompt will
> either do a retry or move on.
> But I think from wiki description, it seems # of milliseconds for which
> data entry will be blocked, am I correct?
>

No, it does not block. This timeout is for how long PAGD will wait until it
does a retry. For example, with 3000 it will play the file, wait for 3 secs
while listening for digits, and if no digits are pressed it will start the
process all over again. When it gets to max tries then it will fail and move
on.


>
> --> 10th parameter
> * digit_timeout = Number of milliseconds to wait after DTMF ( Added Here
> [1]<http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=cfa30468e89788d220ad9fbc07ab1ae3406cf3fa>
>  )
> What does this do? Is it # ms to wait before second digit can be pressed?
>

The digit timeout is like an "inter-digit timer." It basically is the
maximum amount of time the caller can wait in between digits. So if that is
set to 2500, then the caller can wait no more than 2.5 seconds in between
key presses. The idea is to tailor just how long PAGD will wait after
hearing digits to assume that the caller is done entering his/her digits. If
you are still confused by this then I recommend doing two tests, one where
you set this value really low, like 2000 and another where you set it really
high, like 8000. Then call and enter some digits. See how long it takes for
PAGD to assume you're done dialing.

I will revisit the wiki page and make sure I clarify this documentation.

-MC
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From fs-list at communicatefreely.net  Fri Feb 25 21:58:53 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 25 Feb 2011 13:58:53 -0500
Subject: [Freeswitch-users] implementing stereo (e.g. portaudio to test
 it easily)
In-Reply-To: <AANLkTinrvj7Ye22GVu+CtaE+ApzP3OHGm=JdjsO1B2-i@mail.gmail.com>
References: <AANLkTikL9SQPdsJ6XF-AUZuLQ3Z-AG7Eqgj2BsL=CZ4S@mail.gmail.com>	<AANLkTinowzmzt8NxB-FtifkZEQZ8s3OQAWLuC18PN60Q@mail.gmail.com>	<AANLkTikDFqS_Y9XS-gnVvnfPhAW1Eri=zy4XB3vAS2N0@mail.gmail.com>	<4d66ee22.857a0e0a.6c3f.0343@mx.google.com>	<AANLkTimrx5q076f+EAjgMQBh4c4kZYE2Mjq401oV2V7z@mail.gmail.com>	<000801cbd4d5$13362830$39a27890$@gmail.com>	<1298643188393-6064582.post@n2.nabble.com>	<000f01cbd4fe$015301f0$03f905d0$@gmail.com>	<AANLkTikjch_cPU8XzzF4APjWDAcQNcodvtZU1C3OkKP_@mail.gmail.com>
	<AANLkTinrvj7Ye22GVu+CtaE+ApzP3OHGm=JdjsO1B2-i@mail.gmail.com>
Message-ID: <4D67FBED.6060101@communicatefreely.net>

Here's an idea  - it might be a little easier to make interoperate with 
other SIP devices.

What if you built a conferencing endpoint that mixed all the audio 
locally into a 3D matrix?

Each participant in the call could use any of the common codecs, but 
would have an extra SIP header than indicated their position in the 
sound field.

Here's a bit of an overview:

A regular Freeswitch system takes care of registering endpoints, 
routing, etc.  Just like a normal PBX.  The only thing special it would 
do is set add a custom SIP header with a person's sound field position, 
based on a directory variable.

Each conference unit would be a stripped down freeswitch system with a 
custom modified mod_conference.  In addition to mixing audio to mono 
like the regular mod_conference does, it would also generate a stereo 
mix, based on the position information of each participant.  It could 
dump out the stereo audio directly to a named pipe, mod_shout, or 
directly to audio hardware.  You could probably shoehorn a stereo input 
from hardware as well.  Essentially, you are building a stereo 
conference phone.

If you want to join up multiple conference units, just establish two 
sessions between each conference - one panned hard left, the other hard 
right.  They can then use G722 without any modification.

This probably isn't quite as nice as having true stereo codecs and media 
handling, but it may be easier to implement.  After all, telecom is very 
much a mono world.

-Tim
> The main reason we do not have stereo completely implemented is there
> was not really any codec or devices that could handle it to bother
> testing with.
>   



From msc at freeswitch.org  Fri Feb 25 22:02:44 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 25 Feb 2011 11:02:44 -0800
Subject: [Freeswitch-users] send fax in one leg call
In-Reply-To: <3505B4B09EB64A1E8BA155FF076E0776@e1705>
References: <3505B4B09EB64A1E8BA155FF076E0776@e1705>
Message-ID: <AANLkTi=6NQPxYsBP8UP7PAKKYJYUE58W_RCYy47Of-Vn@mail.gmail.com>

What are the conditions under which this will happen? You mention using
uuid_broadcast at the fs_cli - that implies there is already a call in
progress. Is this the case? Is the user doing something to trigger the
sending of the fax?

You are leaving out relevant pieces of information... Like Johnny Number 5
said: Malfunction! Need input! :)

-MC

On Fri, Feb 25, 2011 at 8:36 AM, Madovsky <infos at madovsky.org> wrote:

>  I'm trying to send a fax to a one leg call.
>
> for example a registered user wants to send a fax to another registered
> user
> so I have this in my default dialplan
>
> *         <extension name="internal_fax">
>                 <condition field="${sip_to_uri}"
> expression="^(\+)(\d{10})@$${domain}$">
>                         <action application="ring_ready"/>
>                         <action application="set"
> data="execute_on_media=tone_detect fax 1100 r +5000 transfer 'receivefax XML
> features' 2" />
>                         <action application="sleep" data="2000"/>
>                         <action application="answer"/>
>                         <action application="voicemail" data="default
> ${domain_name} ${sip_to_user}"/>
>                 </condition>
>         </extension>
> *
> for now I have no other way to send the fax from CLI like this:
> *fs_cli -x "uuid_broadcast <uuid> execute_extension::' sendfax XML
> features'  <aleg=timeout??> <bleg=no bleg??>"*
>
> thanks
>
> Franck
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Feb 25 22:14:18 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 25 Feb 2011 14:14:18 -0500
Subject: [Freeswitch-users] send fax in one leg call
References: <3505B4B09EB64A1E8BA155FF076E0776@e1705>
	<AANLkTi=6NQPxYsBP8UP7PAKKYJYUE58W_RCYy47Of-Vn@mail.gmail.com>
Message-ID: <DB05C83788FF470BB3E2B88419D23670@e1705>

ok my question is :
how to send a fax between 2 internal registered users ?
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Friday, February 25, 2011 2:02 PM
  Subject: Re: [Freeswitch-users] send fax in one leg call


  What are the conditions under which this will happen? You mention using uuid_broadcast at the fs_cli - that implies there is already a call in progress. Is this the case? Is the user doing something to trigger the sending of the fax? 


  You are leaving out relevant pieces of information... Like Johnny Number 5 said: Malfunction! Need input! :)


  -MC


  On Fri, Feb 25, 2011 at 8:36 AM, Madovsky <infos at madovsky.org> wrote:

    I'm trying to send a fax to a one leg call.

    for example a registered user wants to send a fax to another registered user
    so I have this in my default dialplan

             <extension name="internal_fax">
                    <condition field="${sip_to_uri}" expression="^(\+)(\d{10})@$${domain}$">
                            <action application="ring_ready"/>
                            <action application="set" data="execute_on_media=tone_detect fax 1100 r +5000 transfer 'receivefax XML features' 2" />
                            <action application="sleep" data="2000"/>
                            <action application="answer"/>
                            <action application="voicemail" data="default ${domain_name} ${sip_to_user}"/>
                    </condition>
            </extension>

    for now I have no other way to send the fax from CLI like this:
    fs_cli -x "uuid_broadcast <uuid> execute_extension::' sendfax XML features'  <aleg=timeout??> <bleg=no bleg??>"

    thanks

    Franck

    _______________________________________________
    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






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From sunwood360 at gmail.com  Fri Feb 25 22:23:55 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Fri, 25 Feb 2011 11:23:55 -0800
Subject: [Freeswitch-users] Fax processing not successful - result (41)
	TIFF/F file cannot be opened.
Message-ID: <AANLkTim+v6WAxhuhkPswYwGEL_R2yUCp5Y6pAzt3o2Ry@mail.gmail.com>

I have the extension in default.xml;

<extension name="fax_receive2">
     <condition field="destination_number" expression="^(9979)$">
       <action application="answer" />
       <action application="playback" data="silence_stream://2000"/>
       <action application="rxfax" data="/tmp/fax-${uuid}.rxfax.tiff"/>
       <action application="hangup"/>
     </condition>
   </extension>

then in fs_cli : originate loopback/9979 &txfax(/tmp/a.tif)

in fact, the file /tmp/a.tif does exist.

/tmp$ ls -l /tmp/a.tif
-rwxrwxrwx 1 root root 179378 2011-02-02 05:30 /tmp/a.tif


2011-02-25 11:17:39.444902 [DEBUG] mod_spandsp_fax.c:1108 Raw write codec
activation Success L16
2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:323
==============================================================================

2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:335 Fax processing not
successful - result (17) Received a DCN while waiting for a DIS.
2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:340 Remote station
id:
2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:341 Local station id:
SpanDSP Fax Ident
2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:342 Pages transferred:
0
2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:344 Total fax pages:
0
2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:345 Image resolution:
0x0
2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:346 Transfer Rate:
14400
2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:348 ECM status
off
2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:349 remote
country:
2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:350 remote
vendor:
2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:351 remote
model:
2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:353
==============================================================================

2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:323
==============================================================================

2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:335 Fax processing not
successful - result (41) TIFF/F file cannot be opened.
2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:340 Remote station id:
SpanDSP Fax Ident
2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:341 Local station id:
SpanDSP Fax Ident


what is going wrong??

thanks
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From infos at madovsky.org  Fri Feb 25 22:30:12 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 25 Feb 2011 14:30:12 -0500
Subject: [Freeswitch-users] Fax processing not successful - result
	(41)TIFF/F file cannot be opened.
References: <AANLkTim+v6WAxhuhkPswYwGEL_R2yUCp5Y6pAzt3o2Ry@mail.gmail.com>
Message-ID: <E4A1DC3E781847CEAC56CD709D7FF567@e1705>

check permissions
  ----- Original Message ----- 
  From: envelopes envelopes 
  To: FreeSWITCH Users Help 
  Sent: Friday, February 25, 2011 2:23 PM
  Subject: [Freeswitch-users] Fax processing not successful - result (41)TIFF/F file cannot be opened.


  I have the extension in default.xml;

  <extension name="fax_receive2">
       <condition field="destination_number" expression="^(9979)$">
         <action application="answer" />
         <action application="playback" data="silence_stream://2000"/>
         <action application="rxfax" data="/tmp/fax-${uuid}.rxfax.tiff"/>
         <action application="hangup"/>
       </condition>
     </extension>

  then in fs_cli : originate loopback/9979 &txfax(/tmp/a.tif)

  in fact, the file /tmp/a.tif does exist.

  /tmp$ ls -l /tmp/a.tif
  -rwxrwxrwx 1 root root 179378 2011-02-02 05:30 /tmp/a.tif


  2011-02-25 11:17:39.444902 [DEBUG] mod_spandsp_fax.c:1108 Raw write codec activation Success L16                                            
  2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:323 ==============================================================================     
  2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:335 Fax processing not successful - result (17) Received a DCN while waiting for a DIS.
  2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:340 Remote station id:                                                                 
  2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:341 Local station id:  SpanDSP Fax Ident                                               
  2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:342 Pages transferred: 0                                                               
  2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:344 Total fax pages:   0                                                               
  2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:345 Image resolution:  0x0                                                             
  2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:346 Transfer Rate:     14400                                                           
  2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:348 ECM status         off                                                             
  2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:349 remote country:                                                                    
  2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:350 remote vendor:                                                                     
  2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:351 remote model:                                                                      
  2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:353 ==============================================================================     
  2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:323 ==============================================================================     
  2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:335 Fax processing not successful - result (41) TIFF/F file cannot be opened.          
  2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:340 Remote station id: SpanDSP Fax Ident                                               
  2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:341 Local station id:  SpanDSP Fax Ident


  what is going wrong??

  thanks




------------------------------------------------------------------------------


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From sunwood360 at gmail.com  Fri Feb 25 22:33:27 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Fri, 25 Feb 2011 11:33:27 -0800
Subject: [Freeswitch-users] Fax processing not successful - result
 (41)TIFF/F file cannot be opened.
In-Reply-To: <E4A1DC3E781847CEAC56CD709D7FF567@e1705>
References: <AANLkTim+v6WAxhuhkPswYwGEL_R2yUCp5Y6pAzt3o2Ry@mail.gmail.com>
	<E4A1DC3E781847CEAC56CD709D7FF567@e1705>
Message-ID: <AANLkTi=X_VJQGX1Y0QEFDNUwgSaguk4S9=oZdkNp-sn7@mail.gmail.com>

I don't see any permission issue here.

/tmp$ ls -l /tmp/a.tif
-rwxrwxrwx 1 root root 179378 2011-02-02 05:30 /tmp/a.tif

receive side also writes to /tmp directroy.

It is really weird.



On Fri, Feb 25, 2011 at 11:30 AM, Madovsky <infos at madovsky.org> wrote:

>  check permissions
>
> ----- Original Message -----
> *From:* envelopes envelopes <sunwood360 at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Friday, February 25, 2011 2:23 PM
> *Subject:* [Freeswitch-users] Fax processing not successful - result
> (41)TIFF/F file cannot be opened.
>
> I have the extension in default.xml;
>
> <extension name="fax_receive2">
>      <condition field="destination_number" expression="^(9979)$">
>        <action application="answer" />
>        <action application="playback" data="silence_stream://2000"/>
>        <action application="rxfax" data="/tmp/fax-${uuid}.rxfax.tiff"/>
>        <action application="hangup"/>
>      </condition>
>    </extension>
>
> then in fs_cli : originate loopback/9979 &txfax(/tmp/a.tif)
>
> in fact, the file /tmp/a.tif does exist.
>
> /tmp$ ls -l /tmp/a.tif
> -rwxrwxrwx 1 root root 179378 2011-02-02 05:30 /tmp/a.tif
>
>
> 2011-02-25 11:17:39.444902 [DEBUG] mod_spandsp_fax.c:1108 Raw write codec
> activation Success L16
> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:323
> ==============================================================================
>
> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:335 Fax processing not
> successful - result (17) Received a DCN while waiting for a DIS.
> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:340 Remote station
> id:
> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:341 Local station id:
> SpanDSP Fax Ident
> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:342 Pages transferred:
> 0
> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:344 Total fax pages:
> 0
> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:345 Image resolution:
> 0x0
> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:346 Transfer Rate:
> 14400
> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:348 ECM status
> off
> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:349 remote
> country:
> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:350 remote
> vendor:
> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:351 remote
> model:
> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:353
> ==============================================================================
>
> 2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:323
> ==============================================================================
>
> 2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:335 Fax processing not
> successful - result (41) TIFF/F file cannot be opened.
> 2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:340 Remote station id:
> SpanDSP Fax Ident
> 2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:341 Local station id:
> SpanDSP Fax Ident
>
>
> what is going 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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 aastral.net  Fri Feb 25 23:11:58 2011
From: freeswitch at aastral.net (Bill W.)
Date: Fri, 25 Feb 2011 15:11:58 -0500
Subject: [Freeswitch-users] Bridge app and SIP 503
Message-ID: <1Pt41D-0005Ig-S1@mail.aastral.net>

Hi all,

I'm using a failover bridge string with several gateways.

<action application="bridge"
data="sofia/gateway/primary/dialstring|sofia/gateway/secondary/dialstring"/>

One of the gateways responded to a mis-dialed number with a "503 Service
Unavailable" instead of a "484 Address Incomplete"

My FreeSWITCH servers are behind a load-balancer, and what is happening
is the bridge app is passing that 503 back to the load-balancer which
then marks that gateway as down.

I can't remove the 503 detection from the load balancer, because that's
how I disable a gateway from within freeswitch (fsctl pause, which sends
503).

Does anyone have any recommendations on how to address this issue?

Thanks!





From mthakershi at gmail.com  Fri Feb 25 23:15:19 2011
From: mthakershi at gmail.com (Malay Thakershi)
Date: Fri, 25 Feb 2011 14:15:19 -0600
Subject: [Freeswitch-users] PlayAndGetDigits parameters
In-Reply-To: <AANLkTi=i8H2uk0mmE2vF6_ugPRcmicq2M5b8FU06vdd=@mail.gmail.com>
References: <AANLkTi=p1c=e+F=nW0hPCXRX4bf3S5zAXMRBPGfVGJkD@mail.gmail.com>
	<AANLkTi=i8H2uk0mmE2vF6_ugPRcmicq2M5b8FU06vdd=@mail.gmail.com>
Message-ID: <AANLkTimKN5yZCkmeFCXyzmefL4edinVaRWM6izLVpEL9@mail.gmail.com>

Thank you for clarification. Wiki description of the parameters was a little
confusing to me. Modifying that description would help the community.

On Fri, Feb 25, 2011 at 12:52 PM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Fri, Feb 25, 2011 at 9:34 AM, Malay Thakershi <mthakershi at gmail.com>wrote:
>
>> Recently I upgraded to latest GIT version and found there was an extra
>> parameter to this function.
>>
>> I have two questions if someone can explain:
>>
>> --> 4th parameter
>> * timeout = Number of milliseconds to wait once the file is done playing
>> before you type a digit.
>> From the name, I alway thought this had to do with user's inactivity at
>> the prompt. So if value is 3000, after 3 seconds of inactivity prompt will
>> either do a retry or move on.
>> But I think from wiki description, it seems # of milliseconds for which
>> data entry will be blocked, am I correct?
>>
>
> No, it does not block. This timeout is for how long PAGD will wait until it
> does a retry. For example, with 3000 it will play the file, wait for 3 secs
> while listening for digits, and if no digits are pressed it will start the
> process all over again. When it gets to max tries then it will fail and move
> on.
>
>
>>
>> --> 10th parameter
>> * digit_timeout = Number of milliseconds to wait after DTMF ( Added Here
>> [1]<http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=cfa30468e89788d220ad9fbc07ab1ae3406cf3fa>
>>  )
>> What does this do? Is it # ms to wait before second digit can be pressed?
>>
>
> The digit timeout is like an "inter-digit timer." It basically is the
> maximum amount of time the caller can wait in between digits. So if that is
> set to 2500, then the caller can wait no more than 2.5 seconds in between
> key presses. The idea is to tailor just how long PAGD will wait after
> hearing digits to assume that the caller is done entering his/her digits. If
> you are still confused by this then I recommend doing two tests, one where
> you set this value really low, like 2000 and another where you set it really
> high, like 8000. Then call and enter some digits. See how long it takes for
> PAGD to assume you're done dialing.
>
> I will revisit the wiki page and make sure I clarify this documentation.
>
> -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 acrow at integrafin.co.uk  Sat Feb 26 00:04:50 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Fri, 25 Feb 2011 21:04:50 +0000
Subject: [Freeswitch-users] British sounds
Message-ID: <4D681972.7050400@integrafin.co.uk>

Does anyone know of either of the following:

1. A source (paid or free) of British English sounds (preferably female 
voice) for Freeswitch, preferably covering all the same sounds as those 
supplied.
or
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.

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)



From errotan at elder.hu  Sat Feb 26 00:09:00 2011
From: errotan at elder.hu (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Fri, 25 Feb 2011 22:09:00 +0100
Subject: [Freeswitch-users]
 =?iso-8859-1?q?Freeswtich_as_a_media_proxy_bet?=
 =?iso-8859-1?q?ween=09ipv4_=3C=3D=3Eipv6_using_Polycom_HDX8006_SIP_UA-s?=
In-Reply-To: <4D67BA6C.5050401@niif.hu>
References: <4D650823.5050305@niif.hu> <201102242152.02899.errotan@elder.hu>
	<4D67BA6C.5050401@niif.hu>
Message-ID: <201102252209.00664.errotan@elder.hu>

Sorry I didn't fully read your mail, probably you need some comerical program 
for full functionality. I just checked on Polycom's home page that they 
actually do have that kind of "Management Application", and I bet it is as 
expensive as the HDX8006 itself :)


2011. febru?r 25. 15:19:24 d?tummal M?SZ?ROS Mih?ly az al?bbiakat ?rta:
> Hi Zsolt,
> 
> I need media proxy between IPv6 and IPv4.
> Not only a signaling proxy needed, so SIP Express Router is not enough.
> I need also a Media proxy what can proxy many (and furthermore any)
> media like BFCP, FECC h.224, as i wrote before.
> 
> Thanks,
> Misi
> 
> 2011-02-24 21:52 keltez?ssel, Pusk?s Zsolt ?rta:
> > Hi.
> > 
> > If you just want a sip proxy use OpenSER or Opensips.
> > 
> > 2011. febru?r 23. 14:14:11 d?tummal M?SZ?ROS Mih?ly az al?bbiakat ?rta:
> >> Hi,
> >> 
> >>     1. 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)
> >>     
> >>     2. 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?
> >>     
> >>     3. 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 msc at freeswitch.org  Sat Feb 26 00:28:22 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 25 Feb 2011 13:28:22 -0800
Subject: [Freeswitch-users] PlayAndGetDigits parameters
In-Reply-To: <AANLkTimKN5yZCkmeFCXyzmefL4edinVaRWM6izLVpEL9@mail.gmail.com>
References: <AANLkTi=p1c=e+F=nW0hPCXRX4bf3S5zAXMRBPGfVGJkD@mail.gmail.com>
	<AANLkTi=i8H2uk0mmE2vF6_ugPRcmicq2M5b8FU06vdd=@mail.gmail.com>
	<AANLkTimKN5yZCkmeFCXyzmefL4edinVaRWM6izLVpEL9@mail.gmail.com>
Message-ID: <AANLkTinaKy03k8jKJxD8g0iPkmJT_88EM-tbcyeypwa2@mail.gmail.com>

On Fri, Feb 25, 2011 at 12:15 PM, Malay Thakershi <mthakershi at gmail.com>wrote:

> Thank you for clarification. Wiki description of the parameters was a
> little confusing to me. Modifying that description would help the community.
>
The wiki page for PAGD has been updated and clarified.
-MC
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From wesleyakio at tuntscorp.com  Sat Feb 26 00:29:06 2011
From: wesleyakio at tuntscorp.com (Wesley Akio)
Date: Fri, 25 Feb 2011 18:29:06 -0300
Subject: [Freeswitch-users] audio<->T38 gateway problem
Message-ID: <AANLkTim0CB1HKztxKdrwgeaBaO2bj=+c8LvaZr1uLTsG@mail.gmail.com>

Hi All,

I have a FreeSwitch box running FreeSWITCH version: 1.0.head (git-c6e7988
2010-12-09 14-25-39 -0600) under CentOS 5.5.

I need freeswitch act as a gateway with G711 in one side an T38 in the
other: Caller:G711--->FreeSwitch--->Callee:T38

I tried several variations of dialplans without success and was hoping
someone could help me, the call stablishes in G711 and I could not find a
way to switch to t38.

My current dialplan follows:

<condition field="destination_number" expression=".*" break="on-true">
          <action application="set" data="fax_enable_t38=true"/>
          <action application="set" data="execute_on_answer=t38_gateway
peer"/>
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="progress_timeout=10"/>
          <action application="set" data="effective_caller_id_name=0000"/>
          <action application="set" data="effective_caller_id_number=0000"/>
          <action application="set" data="execute_on_answer=sched_hangup
+9000 alloted_timeout"/>
          <action application="bridge" data="sofia/testfax/1234 at 1.2.3.4"/>
</condition>

Thank you,

Wesley Akio
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From msc at freeswitch.org  Sat Feb 26 00:30:18 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 25 Feb 2011 13:30:18 -0800
Subject: [Freeswitch-users] British sounds
In-Reply-To: <4D681972.7050400@integrafin.co.uk>
References: <4D681972.7050400@integrafin.co.uk>
Message-ID: <AANLkTinGQk18xc8WCap6VkyTVYM1-XHcPi1PhwVTNs8B@mail.gmail.com>

On Fri, Feb 25, 2011 at 1:04 PM, Alex Crow <acrow at integrafin.co.uk> wrote:

> Does anyone know of either of the following:
>
> 1. A source (paid or free) of British English sounds (preferably female
> voice) for Freeswitch, preferably covering all the same sounds as those
> supplied.
> or
> 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
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From msc at freeswitch.org  Sat Feb 26 00:33:15 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 25 Feb 2011 13:33:15 -0800
Subject: [Freeswitch-users] send fax in one leg call
In-Reply-To: <DB05C83788FF470BB3E2B88419D23670@e1705>
References: <3505B4B09EB64A1E8BA155FF076E0776@e1705>
	<AANLkTi=6NQPxYsBP8UP7PAKKYJYUE58W_RCYy47Of-Vn@mail.gmail.com>
	<DB05C83788FF470BB3E2B88419D23670@e1705>
Message-ID: <AANLkTi=Li6cxE4xXiy79OPeWzPSDYrmJMHpqdDykaV3G@mail.gmail.com>

On Fri, Feb 25, 2011 at 11:14 AM, Madovsky <infos at madovsky.org> wrote:

>  ok my question is :
> how to send a fax between 2 internal registered users ?
>

are they both fax machines?
-MC
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From acrow at integrafin.co.uk  Sat Feb 26 00:42:04 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Fri, 25 Feb 2011 21:42:04 +0000
Subject: [Freeswitch-users] British sounds
In-Reply-To: <AANLkTinGQk18xc8WCap6VkyTVYM1-XHcPi1PhwVTNs8B@mail.gmail.com>
References: <4D681972.7050400@integrafin.co.uk>
	<AANLkTinGQk18xc8WCap6VkyTVYM1-XHcPi1PhwVTNs8B@mail.gmail.com>
Message-ID: <4D68222C.8010108@integrafin.co.uk>


>     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)

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From whglee at gmail.com  Sat Feb 26 00:56:14 2011
From: whglee at gmail.com (George Lee)
Date: Fri, 25 Feb 2011 16:56:14 -0500
Subject: [Freeswitch-users] FreeSwithc not handling no SDP INVITE properly
	for video call
Message-ID: <AANLkTinoomTncUKH=xKLSfAaWKeTupOMv_aZuN_CVVF+@mail.gmail.com>

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


From ce at kapper.net  Sat Feb 26 01:11:23 2011
From: ce at kapper.net (Clemens Ebentheuer)
Date: Fri, 25 Feb 2011 23:11:23 +0100
Subject: [Freeswitch-users] t38_gateway self - no re-Invite?
In-Reply-To: <4D63C790.7000306@ewetel.de>
References: <1B19ABD72889C245AE8EEE08AC24103A28C1AFC00E@exmachina.office.kapper.net>
	<4D6381C0.7040408@ewetel.de>
	<1B19ABD72889C245AE8EEE08AC24103A28C1AFC015@exmachina.office.kapper.net>
	<4D63C790.7000306@ewetel.de>
Message-ID: <1B19ABD72889C245AE8EEE08AC24103A28C4231F3E@exmachina.office.kapper.net>

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


From anthony.minessale at gmail.com  Sat Feb 26 01:20:04 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 25 Feb 2011 16:20:04 -0600
Subject: [Freeswitch-users] ESL socket outbound: clean way to close a
	session?
In-Reply-To: <AANLkTinPRgxuXRwCrB2WxFb4i5yTNS7MrQ6nVRyzMaJs@mail.gmail.com>
References: <AANLkTinPRgxuXRwCrB2WxFb4i5yTNS7MrQ6nVRyzMaJs@mail.gmail.com>
Message-ID: <AANLkTim8y_i0Vcb9w_+B0==JVQpKgCdrxgm4uhKWAfVx@mail.gmail.com>

Its not you it's me.

commit 0444626b721fdd3a5400a86883d5668fa8678cd8
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Fri Feb 25 16:13:16 2011 -0600

There was a bug in the client code introduce when we tried doing clean
shutdown by detecting the disconnect message.
It was necessary to check for the linger content disposition and not
auto-disconnect.

Note, when you update and use this feature, it will be up to you to
disconnect the client when you send the linger command.


On Fri, Feb 25, 2011 at 12:16 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:
> I have problem to clean close an outbound session.
> I'm using the "esl_oop.h/esl_oop.cpp" files and my application is developed
> in c++ and at startup I do a "linger" command.
> My event consuming loop is:
> while (run)
> {
> ?? ?ESLevent * event = connection.recvEventTimed(sleeptime);
>
> ?? ... event processing ...
> }
> This application handle an inbound call (legA), originate an outbound call
> (legB) and bridge the two calls.
> The problem is present when the legA hangup the call: sometimes is received
> the content-type="text/disconnect-notice" (without receiving
> "CHANNEL_HANGUP") and the "recvEventTimed"?exits with event=NULL.
> In this case it's not possible to receive any event or send any other
> message to Freeswitch because the "connection.execute" fails.
> In this situation, if I have originated (with '& park') an outbound call,
> this call remain in parked state and the application cannot hangup it.
> What I'm doing wrong?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 ce at kapper.net  Sat Feb 26 01:25:00 2011
From: ce at kapper.net (Clemens Ebentheuer)
Date: Fri, 25 Feb 2011 23:25:00 +0100
Subject: [Freeswitch-users] audio<->T38 gateway problem
In-Reply-To: <AANLkTim0CB1HKztxKdrwgeaBaO2bj=+c8LvaZr1uLTsG@mail.gmail.com>
References: <AANLkTim0CB1HKztxKdrwgeaBaO2bj=+c8LvaZr1uLTsG@mail.gmail.com>
Message-ID: <1B19ABD72889C245AE8EEE08AC24103A28C4231F3F@exmachina.office.kapper.net>

Hi,

if you want g711->FS->t38 you have to export (not set) execute_on_answer=t38_gateway peer

in this direction for me works:

<condition field="destination_number" expression="^(567)$">
       <action application="export" data="fax_enable_t38=true"/>
       <action application="export" data="fax_enable_t38_request=true"/>
       <action application="export" data="execute_on_answer=t38_gateway peer"/>
       <action application="transfer" data="765 XML default"/>
       <action application="hangup"/>
   </condition>

In t38->FS->g711 direction I?m struggling too in another thread ;-)

clemens

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Wesley Akio
Sent: Friday, February 25, 2011 10:29 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] audio<->T38 gateway problem

Hi All,

I have a FreeSwitch box running FreeSWITCH version: 1.0.head (git-c6e7988 2010-12-09 14-25-39 -0600) under CentOS 5.5.

I need freeswitch act as a gateway with G711 in one side an T38 in the other: Caller:G711--->FreeSwitch--->Callee:T38

I tried several variations of dialplans without success and was hoping someone could help me, the call stablishes in G711 and I could not find a way to switch to t38.

My current dialplan follows:

<condition field="destination_number" expression=".*" break="on-true">
          <action application="set" data="fax_enable_t38=true"/>
          <action application="set" data="execute_on_answer=t38_gateway peer"/>
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="progress_timeout=10"/>
          <action application="set" data="effective_caller_id_name=0000"/>
          <action application="set" data="effective_caller_id_number=0000"/>
          <action application="set" data="execute_on_answer=sched_hangup +9000 alloted_timeout"/>
          <action application="bridge" data="sofia/testfax/1234 at 1.2.3.4<mailto:1234 at 1.2.3.4>"/>
</condition>

Thank you,

Wesley Akio
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From anthony.minessale at gmail.com  Sat Feb 26 01:24:39 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 25 Feb 2011 16:24:39 -0600
Subject: [Freeswitch-users] t38_gateway self - no re-Invite?
In-Reply-To: <1B19ABD72889C245AE8EEE08AC24103A28C4231F3E@exmachina.office.kapper.net>
References: <1B19ABD72889C245AE8EEE08AC24103A28C1AFC00E@exmachina.office.kapper.net>
	<4D6381C0.7040408@ewetel.de>
	<1B19ABD72889C245AE8EEE08AC24103A28C1AFC015@exmachina.office.kapper.net>
	<4D63C790.7000306@ewetel.de>
	<1B19ABD72889C245AE8EEE08AC24103A28C4231F3E@exmachina.office.kapper.net>
Message-ID: <AANLkTi=aVGEgz=j7rOeknPSNgqv-LvDv3HAkyLZckU3h@mail.gmail.com>

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


From ce at kapper.net  Sat Feb 26 01:36:23 2011
From: ce at kapper.net (Clemens Ebentheuer)
Date: Fri, 25 Feb 2011 23:36:23 +0100
Subject: [Freeswitch-users] t38_gateway self - no re-Invite?
In-Reply-To: <AANLkTi=aVGEgz=j7rOeknPSNgqv-LvDv3HAkyLZckU3h@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>
Message-ID: <1B19ABD72889C245AE8EEE08AC24103A28C4231F40@exmachina.office.kapper.net>

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


From wstephen80 at gmail.com  Sat Feb 26 01:42:03 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Fri, 25 Feb 2011 23:42:03 +0100
Subject: [Freeswitch-users] ESL socket outbound: clean way to close a
	session?
In-Reply-To: <AANLkTim8y_i0Vcb9w_+B0==JVQpKgCdrxgm4uhKWAfVx@mail.gmail.com>
References: <AANLkTinPRgxuXRwCrB2WxFb4i5yTNS7MrQ6nVRyzMaJs@mail.gmail.com>
	<AANLkTim8y_i0Vcb9w_+B0==JVQpKgCdrxgm4uhKWAfVx@mail.gmail.com>
Message-ID: <AANLkTinFgW5_Pi2z_k3FV+aEV4MSJydVc56g1acNc51U@mail.gmail.com>

Perfect!
Yes, I know that I have to control the disconnect and it's what I expect to
do using linger.
Thank you Anthony!

Stephen


On Fri, Feb 25, 2011 at 11:20 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Its not you it's me.
>
> commit 0444626b721fdd3a5400a86883d5668fa8678cd8
> Author: Anthony Minessale <anthm at freeswitch.org>
> Date:   Fri Feb 25 16:13:16 2011 -0600
>
> There was a bug in the client code introduce when we tried doing clean
> shutdown by detecting the disconnect message.
> It was necessary to check for the linger content disposition and not
> auto-disconnect.
>
> Note, when you update and use this feature, it will be up to you to
> disconnect the client when you send the linger command.
>
>
> On Fri, Feb 25, 2011 at 12:16 PM, Stephen Wilde <wstephen80 at gmail.com>
> wrote:
> > I have problem to clean close an outbound session.
> > I'm using the "esl_oop.h/esl_oop.cpp" files and my application is
> developed
> > in c++ and at startup I do a "linger" command.
> > My event consuming loop is:
> > while (run)
> > {
> >     ESLevent * event = connection.recvEventTimed(sleeptime);
> >
> >    ... event processing ...
> > }
> > This application handle an inbound call (legA), originate an outbound
> call
> > (legB) and bridge the two calls.
> > The problem is present when the legA hangup the call: sometimes is
> received
> > the content-type="text/disconnect-notice" (without receiving
> > "CHANNEL_HANGUP") and the "recvEventTimed" exits with event=NULL.
> > In this case it's not possible to receive any event or send any other
> > message to Freeswitch because the "connection.execute" fails.
> > In this situation, if I have originated (with '& park') an outbound call,
> > this call remain in parked state and the application cannot hangup it.
> > What I'm doing wrong?
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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  Sat Feb 26 01:48:53 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 25 Feb 2011 16:48:53 -0600
Subject: [Freeswitch-users] t38_gateway self - no re-Invite?
In-Reply-To: <1B19ABD72889C245AE8EEE08AC24103A28C4231F40@exmachina.office.kapper.net>
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>
Message-ID: <AANLkTinGaJ35CZ+t13KRqpdnV3g5p=Yo1psPZY2YGmnh@mail.gmail.com>

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


From anthony.minessale at gmail.com  Sat Feb 26 01:52:15 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 25 Feb 2011 16:52:15 -0600
Subject: [Freeswitch-users] FreeSwithc not handling no SDP INVITE
 properly for video call
In-Reply-To: <AANLkTinoomTncUKH=xKLSfAaWKeTupOMv_aZuN_CVVF+@mail.gmail.com>
References: <AANLkTinoomTncUKH=xKLSfAaWKeTupOMv_aZuN_CVVF+@mail.gmail.com>
Message-ID: <AANLkTinxTuAv0ybtguJtZh3Xkc11=EHJo6CQwdRmRphV@mail.gmail.com>

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


From ce at kapper.net  Sat Feb 26 02:54:39 2011
From: ce at kapper.net (Clemens Ebentheuer)
Date: Sat, 26 Feb 2011 00:54:39 +0100
Subject: [Freeswitch-users] t38_gateway self - no re-Invite?
In-Reply-To: <AANLkTinGaJ35CZ+t13KRqpdnV3g5p=Yo1psPZY2YGmnh@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>
Message-ID: <1B19ABD72889C245AE8EEE08AC24103A28C4231F41@exmachina.office.kapper.net>

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


From anthony.minessale at gmail.com  Sat Feb 26 03:13:22 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 25 Feb 2011 18:13:22 -0600
Subject: [Freeswitch-users] t38_gateway self - no re-Invite?
In-Reply-To: <1B19ABD72889C245AE8EEE08AC24103A28C4231F41@exmachina.office.kapper.net>
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>
Message-ID: <AANLkTinSoNO7Bh5rEVe=_PEOfqjBdLROVm1FB-6OZPML@mail.gmail.com>

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


From ayhkor at gmail.com  Sat Feb 26 03:16:58 2011
From: ayhkor at gmail.com (deniro)
Date: Fri, 25 Feb 2011 19:16:58 -0500
Subject: [Freeswitch-users] conference dialing and montly billing
Message-ID: <AANLkTinY_-7zPySJMQ915hKY0RTHZtycXoszw3K3LE0u@mail.gmail.com>

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--
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From ce at kapper.net  Sat Feb 26 03:28:46 2011
From: ce at kapper.net (Clemens Ebentheuer)
Date: Sat, 26 Feb 2011 01:28:46 +0100
Subject: [Freeswitch-users] t38_gateway self - no re-Invite?
In-Reply-To: <AANLkTinSoNO7Bh5rEVe=_PEOfqjBdLROVm1FB-6OZPML@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>
Message-ID: <1B19ABD72889C245AE8EEE08AC24103A28C4231F42@exmachina.office.kapper.net>

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


From infos at madovsky.org  Sat Feb 26 04:34:54 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 25 Feb 2011 20:34:54 -0500
Subject: [Freeswitch-users] send fax in one leg call
References: <3505B4B09EB64A1E8BA155FF076E0776@e1705><AANLkTi=6NQPxYsBP8UP7PAKKYJYUE58W_RCYy47Of-Vn@mail.gmail.com><DB05C83788FF470BB3E2B88419D23670@e1705>
	<AANLkTi=Li6cxE4xXiy79OPeWzPSDYrmJMHpqdDykaV3G@mail.gmail.com>
Message-ID: <5A784AE95F5645E4802EEE56C2AFE50A@e1705>

:) no, it's FS the fax machine....
Nevermind, I just found the trick.
brdige with sofia external

thank you

  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Friday, February 25, 2011 4:33 PM
  Subject: Re: [Freeswitch-users] send fax in one leg call





  On Fri, Feb 25, 2011 at 11:14 AM, Madovsky <infos at madovsky.org> wrote:

    ok my question is :
    how to send a fax between 2 internal registered users ?


  are they both fax machines?
  -MC



------------------------------------------------------------------------------


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From Nabble at slickdeals.endjunk.com  Sat Feb 26 05:43:36 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 25 Feb 2011 18:43:36 -0800 (PST)
Subject: [Freeswitch-users] freeswitch -nc -nosql
Message-ID: <1298688216720-6066718.post@n2.nabble.com>


It had been a while I didn't get a chance to upgrade my FS git until today.
With a newly built FreeSWITCH Version 1.0.head (git-0f7682a 2011-02-25
21-06-22 +0100), at least a new problem surfaces where FS Segmentation fault
as shown below:
2011-02-25 21:30:34.277865 [CONSOLE] switch_core.c:1764 
   _____              ______        _____ _____ ____ _   _  
  |  ___| __ ___  ___/ ___\ \      / /_ _|_   _/ ___| | | | 
  | |_ | '__/ _ \/ _ \___ \\ \ /\ / / | |  | || |   | |_| | 
  |  _|| | |  __/  __/___) |\ 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/        *
************************************************************

2011-02-25 21:30:34.278007 [CONSOLE] switch_core.c:1767 
FreeSWITCH Version 1.0.head (git-0f7682a 2011-02-25 21-06-22 +0100) Started.
Max Sessions[1000]
Session Rate[30]
SQL [Enabled]
Segmentation fault
@DockStar:/#
If I add a -nosql switch, i.e. freeswitch -nosql, it runs OK but it keeps
dumping the following message on the CLI:
2011-02-25 21:36:19.378416 [ERR] switch_core_sqldb.c:1721 Error Opening DB!
So, whatI do I need to tweak to make the above message go away?

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-nc-nosql-tp6066718p6066718.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Sat Feb 26 06:40:36 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 25 Feb 2011 21:40:36 -0600
Subject: [Freeswitch-users] freeswitch -nc -nosql
In-Reply-To: <1298688216720-6066718.post@n2.nabble.com>
References: <1298688216720-6066718.post@n2.nabble.com>
Message-ID: <AANLkTikjkTPk3A2FmoRo32WhKxNFaX5YL7Wxezaa6zqG@mail.gmail.com>

Fixed

commit 1cbb8f221b2e76ec66c72d7f09d7f6ae681764b8
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Fri Feb 25 21:39:37 2011 -0600

    supress errors caused from -nosql as reported on the mailing list
and not on JIRA like it should be http://jira.freeswitch.org




On Fri, Feb 25, 2011 at 8:43 PM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
> It had been a while I didn't get a chance to upgrade my FS git until today.
> With a newly built FreeSWITCH Version 1.0.head (git-0f7682a 2011-02-25
> 21-06-22 +0100), at least a new problem surfaces where FS Segmentation fault
> as shown below:
> 2011-02-25 21:30:34.277865 [CONSOLE] switch_core.c:1764
> ? _____ ? ? ? ? ? ? ?______ ? ? ? ?_____ _____ ____ _ ? _
> ?| ?___| __ ___ ?___/ ___\ \ ? ? ?/ /_ _|_ ? _/ ___| | | |
> ?| |_ | '__/ _ \/ _ \___ \\ \ /\ / / | | ?| || | ? | |_| |
> ?| ?_|| | | ?__/ ?__/___) |\ 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/ ? ? ? ?*
> ************************************************************
>
> 2011-02-25 21:30:34.278007 [CONSOLE] switch_core.c:1767
> FreeSWITCH Version 1.0.head (git-0f7682a 2011-02-25 21-06-22 +0100) Started.
> Max Sessions[1000]
> Session Rate[30]
> SQL [Enabled]
> Segmentation fault
> @DockStar:/#
> If I add a -nosql switch, i.e. freeswitch -nosql, it runs OK but it keeps
> dumping the following message on the CLI:
> 2011-02-25 21:36:19.378416 [ERR] switch_core_sqldb.c:1721 Error Opening DB!
> So, whatI do I need to tweak to make the above message go away?
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-nc-nosql-tp6066718p6066718.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 avi at avimarcus.net  Sat Feb 26 19:29:55 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sat, 26 Feb 2011 18:29:55 +0200
Subject: [Freeswitch-users] LRN/LNP lookups..
In-Reply-To: <201102251338.18446.sos@sokhapkin.dyndns.org>
References: <AANLkTim-=9v+dMn_5cqtY4-Zae5JQAZ_usoE=XmMW17R@mail.gmail.com>
	<f78f09e0-8354-496c-eb8d-fc6baf0b547e@me.com>
	<1298650289888-6064996.post@n2.nabble.com>
	<201102251338.18446.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTinor9pqztARo=oxG-4seo8AZXjfE7K+1JkrWEdj@mail.gmail.com>

Not afaik, but you can use callwithus's api for $0.0003 per query.
I integrated it into my php xml_curl lcr script, which you can grab here:
mod_xml-with-PHP <https://github.com/avimar/FreeSWITCH-mod_xml-with-PHP>

<https://github.com/avimar/FreeSWITCH-mod_xml-with-PHP>-Avi Marcus

On Fri, Feb 25, 2011 at 8:38 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> There is no free service, but LRN lookup price is a fraction of cent per
> query, why not pay a few cents a day?
>
> On Friday 25 February 2011, egc52556 wrote:
> > I appreciate there is no complete database of current LRN to download.
>  But
> > is there a free service where I can do a few lookups, say no more than
> > 10-20 a day?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 26 20:40:54 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 26 Feb 2011 17:40:54 +0000
Subject: [Freeswitch-users] Bridge app and SIP 503
In-Reply-To: <1Pt41D-0005Ig-S1@mail.aastral.net>
References: <1Pt41D-0005Ig-S1@mail.aastral.net>
Message-ID: <C80F6EB7-358E-45BE-8C11-678F22D908D0@gmail.com>

Regarding your balancer marking the gateway as down... 503 doesn't necessarily mean the gateway is out of order, for instance it will be returned by many sip providers for dcc34 normal circuit congestion, when all channels are in use.

Steve on iPhone

On 25 Feb 2011, at 20:11, "Bill W." <freeswitch at aastral.net> wrote:

> Hi all,
> 
> I'm using a failover bridge string with several gateways.
> 
> <action application="bridge"
> data="sofia/gateway/primary/dialstring|sofia/gateway/secondary/dialstring"/>
> 
> One of the gateways responded to a mis-dialed number with a "503 Service
> Unavailable" instead of a "484 Address Incomplete"
> 
> My FreeSWITCH servers are behind a load-balancer, and what is happening
> is the bridge app is passing that 503 back to the load-balancer which
> then marks that gateway as down.
> 
> I can't remove the 503 detection from the load balancer, because that's
> how I disable a gateway from within freeswitch (fsctl pause, which sends
> 503).
> 
> Does anyone have any recommendations on how to address this issue?
> 
> 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 steveayre at gmail.com  Sat Feb 26 20:46:44 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 26 Feb 2011 17:46:44 +0000
Subject: [Freeswitch-users] Bridge app and SIP 503
In-Reply-To: <1Pt41D-0005Ig-S1@mail.aastral.net>
References: <1Pt41D-0005Ig-S1@mail.aastral.net>
Message-ID: <2B9ED35E-9E8A-45B2-81ED-E0970B7B8175@gmail.com>

One possible solution is
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="sip_ignore_remote_cause=true"/>
<action application="bridge" data="..."/>
<action application="hangup" data="..."/>

Failed calls will hit the hangup and hangup with a different reason to the bleg. That will let you return something other than 503. You could instead run a lua script if you want to add some intelligence to return different responses based on what comes back from the gateway or which gateway it is.

A better solution might be to disable it on the balancer before doing fsctl pause.

Steve on iPhone

On 25 Feb 2011, at 20:11, "Bill W." <freeswitch at aastral.net> wrote:

> Hi all,
> 
> I'm using a failover bridge string with several gateways.
> 
> <action application="bridge"
> data="sofia/gateway/primary/dialstring|sofia/gateway/secondary/dialstring"/>
> 
> One of the gateways responded to a mis-dialed number with a "503 Service
> Unavailable" instead of a "484 Address Incomplete"
> 
> My FreeSWITCH servers are behind a load-balancer, and what is happening
> is the bridge app is passing that 503 back to the load-balancer which
> then marks that gateway as down.
> 
> I can't remove the 503 detection from the load balancer, because that's
> how I disable a gateway from within freeswitch (fsctl pause, which sends
> 503).
> 
> Does anyone have any recommendations on how to address this issue?
> 
> 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 steveayre at gmail.com  Sat Feb 26 20:48:47 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 26 Feb 2011 17:48:47 +0000
Subject: [Freeswitch-users] LRN/LNP lookups..
In-Reply-To: <AANLkTinor9pqztARo=oxG-4seo8AZXjfE7K+1JkrWEdj@mail.gmail.com>
References: <AANLkTim-=9v+dMn_5cqtY4-Zae5JQAZ_usoE=XmMW17R@mail.gmail.com>
	<f78f09e0-8354-496c-eb8d-fc6baf0b547e@me.com>
	<1298650289888-6064996.post@n2.nabble.com>
	<201102251338.18446.sos@sokhapkin.dyndns.org>
	<AANLkTinor9pqztARo=oxG-4seo8AZXjfE7K+1JkrWEdj@mail.gmail.com>
Message-ID: <DFE43650-F5C0-484F-B0D9-82C01334F473@gmail.com>

If you often get repeated numbers you could also store them in an odbc database for a few days so that you don't have to pay for repeat lookups. If you're doing a lot of lookups that can save you a lot of money. Your routing won't be 100% accurate, but the lnp providers are probably serving you from a cache too anyway.

Steve on iPhone

On 26 Feb 2011, at 16:29, Avi Marcus <avi at avimarcus.net> wrote:

> Not afaik, but you can use callwithus's api for $0.0003 per query.
> I integrated it into my php xml_curl lcr script, which you can grab here: mod_xml-with-PHP
> 
> -Avi Marcus
> 
> On Fri, Feb 25, 2011 at 8:38 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org> wrote:
> There is no free service, but LRN lookup price is a fraction of cent per
> query, why not pay a few cents a day?
> 
> On Friday 25 February 2011, egc52556 wrote:
> > I appreciate there is no complete database of current LRN to download.  But
> > is there a free service where I can do a few lookups, say no more than
> > 10-20 a day?
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mike.burlingame at me.com  Sat Feb 26 20:53:58 2011
From: mike.burlingame at me.com (Mike Burlingame)
Date: Sat, 26 Feb 2011 09:53:58 -0800
Subject: [Freeswitch-users] LRN/LNP lookups..
In-Reply-To: <DFE43650-F5C0-484F-B0D9-82C01334F473@gmail.com>
References: <AANLkTim-=9v+dMn_5cqtY4-Zae5JQAZ_usoE=XmMW17R@mail.gmail.com>
	<f78f09e0-8354-496c-eb8d-fc6baf0b547e@me.com>
	<1298650289888-6064996.post@n2.nabble.com>
	<201102251338.18446.sos@sokhapkin.dyndns.org>
	<AANLkTinor9pqztARo=oxG-4seo8AZXjfE7K+1JkrWEdj@mail.gmail.com>
	<DFE43650-F5C0-484F-B0D9-82C01334F473@gmail.com>
Message-ID: <98C2B4B1-0EAF-4473-AFBE-6423CB71790F@me.com>

I know in my direct relationship with one of the major LRN providers it's in our contract that we do not cache the results - in my tests with porting numbers from carrier x to carrier y the LRN DB is updated in less then 60 min's after the port has completed and released on our network.

On Feb 26, 2011, at 9:48 AM, Steven Ayre wrote:

> If you often get repeated numbers you could also store them in an odbc database for a few days so that you don't have to pay for repeat lookups. If you're doing a lot of lookups that can save you a lot of money. Your routing won't be 100% accurate, but the lnp providers are probably serving you from a cache too anyway.
> 
> Steve on iPhone
> 
> On 26 Feb 2011, at 16:29, Avi Marcus <avi at avimarcus.net> wrote:
> 
>> Not afaik, but you can use callwithus's api for $0.0003 per query.
>> I integrated it into my php xml_curl lcr script, which you can grab here: mod_xml-with-PHP
>> 
>> -Avi Marcus
>> 
>> On Fri, Feb 25, 2011 at 8:38 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org> wrote:
>> There is no free service, but LRN lookup price is a fraction of cent per
>> query, why not pay a few cents a day?
>> 
>> On Friday 25 February 2011, egc52556 wrote:
>> > I appreciate there is no complete database of current LRN to download.  But
>> > is there a free service where I can do a few lookups, say no more than
>> > 10-20 a day?
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sat Feb 26 20:57:22 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sat, 26 Feb 2011 19:57:22 +0200
Subject: [Freeswitch-users] LRN/LNP lookups..
In-Reply-To: <DFE43650-F5C0-484F-B0D9-82C01334F473@gmail.com>
References: <AANLkTim-=9v+dMn_5cqtY4-Zae5JQAZ_usoE=XmMW17R@mail.gmail.com>
	<f78f09e0-8354-496c-eb8d-fc6baf0b547e@me.com>
	<1298650289888-6064996.post@n2.nabble.com>
	<201102251338.18446.sos@sokhapkin.dyndns.org>
	<AANLkTinor9pqztARo=oxG-4seo8AZXjfE7K+1JkrWEdj@mail.gmail.com>
	<DFE43650-F5C0-484F-B0D9-82C01334F473@gmail.com>
Message-ID: <AANLkTi=AW=9ntdd5yZ_EXe_KQyHyfjTmKm4rYHj4e=Y6@mail.gmail.com>

I think I provided a DB schema, and an adjustable timing for how long to
cache results.
I also have a regex to import results from grnvoip/excall, which shows the
LRN in the cdrs after the call, if someone is interested I can upload that..

-Avi

On Sat, Feb 26, 2011 at 7:48 PM, Steven Ayre <steveayre at gmail.com> wrote:

> If you often get repeated numbers you could also store them in an odbc
> database for a few days so that you don't have to pay for repeat lookups. If
> you're doing a lot of lookups that can save you a lot of money. Your routing
> won't be 100% accurate, but the lnp providers are probably serving you from
> a cache too anyway.
>
> Steve on iPhone
>
> On 26 Feb 2011, at 16:29, Avi Marcus <avi at avimarcus.net> wrote:
>
> Not afaik, but you can use callwithus's api for $0.0003 per query.
> I integrated it into my php xml_curl lcr script, which you can grab here:
> mod_xml-with-PHP <https://github.com/avimar/FreeSWITCH-mod_xml-with-PHP>
>
> <https://github.com/avimar/FreeSWITCH-mod_xml-with-PHP>-Avi Marcus
>
> On Fri, Feb 25, 2011 at 8:38 PM, Sergey Okhapkin <<sos at sokhapkin.dyndns.org>
> sos at sokhapkin.dyndns.org> wrote:
>
>> There is no free service, but LRN lookup price is a fraction of cent per
>> query, why not pay a few cents a day?
>>
>> On Friday 25 February 2011, egc52556 wrote:
>> > I appreciate there is no complete database of current LRN to download.
>>  But
>> > is there a free service where I can do a few lookups, say no more than
>> > 10-20 a day?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>
>> 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
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 26 23:31:06 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 26 Feb 2011 20:31:06 +0000
Subject: [Freeswitch-users] LRN/LNP lookups..
In-Reply-To: <98C2B4B1-0EAF-4473-AFBE-6423CB71790F@me.com>
References: <AANLkTim-=9v+dMn_5cqtY4-Zae5JQAZ_usoE=XmMW17R@mail.gmail.com>
	<f78f09e0-8354-496c-eb8d-fc6baf0b547e@me.com>
	<1298650289888-6064996.post@n2.nabble.com>
	<201102251338.18446.sos@sokhapkin.dyndns.org>
	<AANLkTinor9pqztARo=oxG-4seo8AZXjfE7K+1JkrWEdj@mail.gmail.com>
	<DFE43650-F5C0-484F-B0D9-82C01334F473@gmail.com>
	<98C2B4B1-0EAF-4473-AFBE-6423CB71790F@me.com>
Message-ID: <AANLkTikW2bmZQ3_3u0-43DMg-qhU6K5AhR8yv-a5g0G6@mail.gmail.com>

It does depend on your contact... we used to have contract with a
provider that did allow it, but they cached them themselves (their
response included a rough estimate of how out-of-date it was). I think
we could have got more up-to-date data from them but for a higher
fee...

-Steve

On 26 February 2011 17:53, Mike Burlingame <mike.burlingame at me.com> wrote:
> I know in my direct relationship with one of the major LRN providers it's in
> our contract that we do not cache the results - in my tests with porting
> numbers from carrier x to carrier y the LRN DB is updated in less then 60
> min's after the port has completed and released on our network.
> On Feb 26, 2011, at 9:48 AM, Steven Ayre wrote:
>
> If you often get repeated numbers you could also store them in an odbc
> database for a few days so that you don't have to pay for repeat lookups. If
> you're doing a lot of lookups that can save you a lot of money. Your routing
> won't be 100% accurate, but the lnp providers are probably serving you from
> a cache too anyway.
>
> Steve on iPhone
> On 26 Feb 2011, at 16:29, Avi Marcus <avi at avimarcus.net> wrote:
>
> Not afaik, but you can use callwithus's api for?$0.0003 per query.
> I?integrated?it into my php xml_curl lcr script, which you can grab
> here:?mod_xml-with-PHP
> -Avi Marcus
>
> On Fri, Feb 25, 2011 at 8:38 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org>
> wrote:
>>
>> There is no free service, but LRN lookup price is a fraction of cent per
>> query, why not pay a few cents a day?
>>
>> On Friday 25 February 2011, egc52556 wrote:
>> > I appreciate there is no complete database of current LRN to download.
>> > ?But
>> > is there a free service where I can do a few lookups, say no more than
>> > 10-20 a day?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Sat Feb 26 23:50:25 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 26 Feb 2011 15:50:25 -0500
Subject: [Freeswitch-users] dtmf interval
Message-ID: <BF50A739E93C4B85BF58AD059EF5B94C@e1705>

Where can I find the DTMF interval setting ?

Thanks
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From ce at kapper.net  Sat Feb 26 23:56:08 2011
From: ce at kapper.net (Clemens Ebentheuer)
Date: Sat, 26 Feb 2011 21:56:08 +0100
Subject: [Freeswitch-users] t38_gateway self - no re-Invite?
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> 
Message-ID: <1B19ABD72889C245AE8EEE08AC24103A28C4231F4F@exmachina.office.kapper.net>

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


From avi at avimarcus.net  Sun Feb 27 01:45:19 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 27 Feb 2011 00:45:19 +0200
Subject: [Freeswitch-users] conference dialing and montly billing
In-Reply-To: <AANLkTinY_-7zPySJMQ915hKY0RTHZtycXoszw3K3LE0u@mail.gmail.com>
References: <AANLkTinY_-7zPySJMQ915hKY0RTHZtycXoszw3K3LE0u@mail.gmail.com>
Message-ID: <AANLkTimto_KGwO6j+PNexUSZS5w=jzr-0qVVdOs=-AaA@mail.gmail.com>

Here's a few options:
1) use mod_nibblebill and do the billing live. If you don't need pre payment
however, this adds in complexity and doesn't provide as much of an audit
trail
2) If you don't need pre-billing, set the bill-to user and the billing per
minute on the leg of the call. Use something like xml_cdr to save the data -
perhaps post it to a script that saves the appropriate fields. It multiplies
the billing * billsec (if you want rounding, do it..) and saves the amount.
Then, just sum up the billing amount. Just note: if the xml_cdr fails to
post (mostly if the server dies mid-call) you won't have a record of the
call. Also, this only works for post-payment.

I am available to set this up, please email me privately if you are
interested.
-Avi Marcus


On Sat, Feb 26, 2011 at 2:16 AM, deniro <ayhkor at gmail.com> 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
>
>
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From sunwood360 at gmail.com  Sun Feb 27 08:53:26 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Sat, 26 Feb 2011 21:53:26 -0800
Subject: [Freeswitch-users] Fax processing not successful - result
 (41)TIFF/F file cannot be opened.
In-Reply-To: <AANLkTi=X_VJQGX1Y0QEFDNUwgSaguk4S9=oZdkNp-sn7@mail.gmail.com>
References: <AANLkTim+v6WAxhuhkPswYwGEL_R2yUCp5Y6pAzt3o2Ry@mail.gmail.com>
	<E4A1DC3E781847CEAC56CD709D7FF567@e1705>
	<AANLkTi=X_VJQGX1Y0QEFDNUwgSaguk4S9=oZdkNp-sn7@mail.gmail.com>
Message-ID: <AANLkTim7ovBYhhicK-8HDhS_tQucywoJ050rmQNKWbqt@mail.gmail.com>

no one had similar issue?

On Fri, Feb 25, 2011 at 11:33 AM, envelopes envelopes
<sunwood360 at gmail.com>wrote:

> I don't see any permission issue here.
>
>
> /tmp$ ls -l /tmp/a.tif
> -rwxrwxrwx 1 root root 179378 2011-02-02 05:30 /tmp/a.tif
>
> receive side also writes to /tmp directroy.
>
> It is really weird.
>
>
>
>
> On Fri, Feb 25, 2011 at 11:30 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  check permissions
>>
>> ----- Original Message -----
>> *From:* envelopes envelopes <sunwood360 at gmail.com>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Sent:* Friday, February 25, 2011 2:23 PM
>> *Subject:* [Freeswitch-users] Fax processing not successful - result
>> (41)TIFF/F file cannot be opened.
>>
>> I have the extension in default.xml;
>>
>> <extension name="fax_receive2">
>>      <condition field="destination_number" expression="^(9979)$">
>>        <action application="answer" />
>>        <action application="playback" data="silence_stream://2000"/>
>>        <action application="rxfax" data="/tmp/fax-${uuid}.rxfax.tiff"/>
>>        <action application="hangup"/>
>>      </condition>
>>    </extension>
>>
>> then in fs_cli : originate loopback/9979 &txfax(/tmp/a.tif)
>>
>> in fact, the file /tmp/a.tif does exist.
>>
>> /tmp$ ls -l /tmp/a.tif
>> -rwxrwxrwx 1 root root 179378 2011-02-02 05:30 /tmp/a.tif
>>
>>
>> 2011-02-25 11:17:39.444902 [DEBUG] mod_spandsp_fax.c:1108 Raw write codec
>> activation Success L16
>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:323
>> ==============================================================================
>>
>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:335 Fax processing
>> not successful - result (17) Received a DCN while waiting for a DIS.
>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:340 Remote station
>> id:
>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:341 Local station
>> id:  SpanDSP Fax Ident
>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:342 Pages
>> transferred: 0
>>
>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:344 Total fax
>> pages:   0
>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:345 Image
>> resolution:  0x0
>>
>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:346 Transfer
>> Rate:     14400
>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:348 ECM
>> status
>> off
>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:349 remote
>> country:
>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:350 remote
>> vendor:
>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:351 remote
>> model:
>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:353
>> ==============================================================================
>>
>> 2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:323
>> ==============================================================================
>>
>> 2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:335 Fax processing
>> not successful - result (41) TIFF/F file cannot be opened.
>> 2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:340 Remote station
>> id: SpanDSP Fax Ident
>> 2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:341 Local station
>> id:  SpanDSP Fax Ident
>>
>>
>> what is going 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
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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  Sun Feb 27 09:10:45 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 27 Feb 2011 01:10:45 -0500
Subject: [Freeswitch-users] Fax processing not successful - result
	(41)TIFF/F file cannot be opened.
References: <AANLkTim+v6WAxhuhkPswYwGEL_R2yUCp5Y6pAzt3o2Ry@mail.gmail.com><E4A1DC3E781847CEAC56CD709D7FF567@e1705><AANLkTi=X_VJQGX1Y0QEFDNUwgSaguk4S9=oZdkNp-sn7@mail.gmail.com>
	<AANLkTim7ovBYhhicK-8HDhS_tQucywoJ050rmQNKWbqt@mail.gmail.com>
Message-ID: <C9BA852E5CC7484A84A44B83B0B69614@e1705>

look at the the emailist archive,
I remember that one had the same problem
but don't remember the solution
  ----- Original Message ----- 
  From: envelopes envelopes 
  To: FreeSWITCH Users Help 
  Sent: Sunday, February 27, 2011 12:53 AM
  Subject: Re: [Freeswitch-users] Fax processing not successful - result (41)TIFF/F file cannot be opened.


  no one had similar issue? 


  On Fri, Feb 25, 2011 at 11:33 AM, envelopes envelopes <sunwood360 at gmail.com> wrote:

    I don't see any permission issue here.


    /tmp$ ls -l /tmp/a.tif
    -rwxrwxrwx 1 root root 179378 2011-02-02 05:30 /tmp/a.tif


    receive side also writes to /tmp directroy.

    It is really weird.


     


    On Fri, Feb 25, 2011 at 11:30 AM, Madovsky <infos at madovsky.org> wrote:

      check permissions
        ----- Original Message ----- 
        From: envelopes envelopes 
        To: FreeSWITCH Users Help 
        Sent: Friday, February 25, 2011 2:23 PM
        Subject: [Freeswitch-users] Fax processing not successful - result (41)TIFF/F file cannot be opened.


        I have the extension in default.xml;

        <extension name="fax_receive2">
             <condition field="destination_number" expression="^(9979)$">
               <action application="answer" />
               <action application="playback" data="silence_stream://2000"/>
               <action application="rxfax" data="/tmp/fax-${uuid}.rxfax.tiff"/>
               <action application="hangup"/>
             </condition>
           </extension>

        then in fs_cli : originate loopback/9979 &txfax(/tmp/a.tif)

        in fact, the file /tmp/a.tif does exist.

        /tmp$ ls -l /tmp/a.tif
        -rwxrwxrwx 1 root root 179378 2011-02-02 05:30 /tmp/a.tif


        2011-02-25 11:17:39.444902 [DEBUG] mod_spandsp_fax.c:1108 Raw write codec activation Success L16                                            
        2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:323 ==============================================================================     
        2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:335 Fax processing not successful - result (17) Received a DCN while waiting for a DIS.
        2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:340 Remote station id:                                                                 
        2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:341 Local station id:  SpanDSP Fax Ident                                               
        2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:342 Pages transferred: 0                                                               
        2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:344 Total fax pages:   0                                                               
        2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:345 Image resolution:  0x0                                                             
        2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:346 Transfer Rate:     14400                                                           
        2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:348 ECM status         off                                                             
        2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:349 remote country:                                                                    
        2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:350 remote vendor:                                                                     
        2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:351 remote model:                                                                      
        2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:353 ==============================================================================     
        2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:323 ==============================================================================     
        2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:335 Fax processing not successful - result (41) TIFF/F file cannot be opened.          
        2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:340 Remote station id: SpanDSP Fax Ident                                               
        2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:341 Local station id:  SpanDSP Fax Ident


        what is going 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


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








------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing 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 sunwood360 at gmail.com  Sun Feb 27 09:46:18 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Sat, 26 Feb 2011 22:46:18 -0800
Subject: [Freeswitch-users] Fax processing not successful - result
 (41)TIFF/F file cannot be opened.
In-Reply-To: <C9BA852E5CC7484A84A44B83B0B69614@e1705>
References: <AANLkTim+v6WAxhuhkPswYwGEL_R2yUCp5Y6pAzt3o2Ry@mail.gmail.com>
	<E4A1DC3E781847CEAC56CD709D7FF567@e1705>
	<AANLkTi=X_VJQGX1Y0QEFDNUwgSaguk4S9=oZdkNp-sn7@mail.gmail.com>
	<AANLkTim7ovBYhhicK-8HDhS_tQucywoJ050rmQNKWbqt@mail.gmail.com>
	<C9BA852E5CC7484A84A44B83B0B69614@e1705>
Message-ID: <AANLkTinQ62xWUnOfauf_sqkNbGYbRUSq8d4DRWxc5-cX@mail.gmail.com>

http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg06601.html

in fact, I am running as a root. so the reason in that thread doesn't apply
to my case.

root     22613  9.5  0.7  40064 15724 pts/2    SLl+ 22:35   0:41
./bin/freeswitch

/tmp directory has worldwide write permission:
drwxrwxrwt  20 root root  4096 2011-02-26 22:17 tmp

If this is bug, please fix it.  If there is a catch, please post the
solution and summarize on wiki page.

One of the particular reason that i like FS is prompt support and enthusiasm
from FS developers.
please do not lost your momentum.


thanks!

On Sat, Feb 26, 2011 at 10:10 PM, Madovsky <infos at madovsky.org> wrote:

>  look at the the emailist archive,
> I remember that one had the same problem
> but don't remember the solution
>
> ----- Original Message -----
> *From:* envelopes envelopes <sunwood360 at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Sunday, February 27, 2011 12:53 AM
> *Subject:* Re: [Freeswitch-users] Fax processing not successful - result
> (41)TIFF/F file cannot be opened.
>
> no one had similar issue?
>
> On Fri, Feb 25, 2011 at 11:33 AM, envelopes envelopes <
> sunwood360 at gmail.com> wrote:
>
>> I don't see any permission issue here.
>>
>>
>> /tmp$ ls -l /tmp/a.tif
>> -rwxrwxrwx 1 root root 179378 2011-02-02 05:30 /tmp/a.tif
>>
>> receive side also writes to /tmp directroy.
>>
>> It is really weird.
>>
>>
>>
>>
>> On Fri, Feb 25, 2011 at 11:30 AM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  check permissions
>>>
>>>   ----- Original Message -----
>>> *From:* envelopes envelopes <sunwood360 at gmail.com>
>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> *Sent:* Friday, February 25, 2011 2:23 PM
>>> *Subject:* [Freeswitch-users] Fax processing not successful - result
>>> (41)TIFF/F file cannot be opened.
>>>
>>> I have the extension in default.xml;
>>>
>>> <extension name="fax_receive2">
>>>      <condition field="destination_number" expression="^(9979)$">
>>>        <action application="answer" />
>>>        <action application="playback" data="silence_stream://2000"/>
>>>        <action application="rxfax" data="/tmp/fax-${uuid}.rxfax.tiff"/>
>>>        <action application="hangup"/>
>>>      </condition>
>>>    </extension>
>>>
>>> then in fs_cli : originate loopback/9979 &txfax(/tmp/a.tif)
>>>
>>> in fact, the file /tmp/a.tif does exist.
>>>
>>> /tmp$ ls -l /tmp/a.tif
>>> -rwxrwxrwx 1 root root 179378 2011-02-02 05:30 /tmp/a.tif
>>>
>>>
>>> 2011-02-25 11:17:39.444902 [DEBUG] mod_spandsp_fax.c:1108 Raw write codec
>>> activation Success L16
>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:323
>>> ==============================================================================
>>>
>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:335 Fax processing
>>> not successful - result (17) Received a DCN while waiting for a DIS.
>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:340 Remote station
>>> id:
>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:341 Local station
>>> id:  SpanDSP Fax Ident
>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:342 Pages
>>> transferred: 0
>>>
>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:344 Total fax
>>> pages:   0
>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:345 Image
>>> resolution:  0x0
>>>
>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:346 Transfer
>>> Rate:     14400
>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:348 ECM
>>> status
>>> off
>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:349 remote
>>> country:
>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:350 remote
>>> vendor:
>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:351 remote
>>> model:
>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:353
>>> ==============================================================================
>>>
>>> 2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:323
>>> ==============================================================================
>>>
>>> 2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:335 Fax processing
>>> not successful - result (41) TIFF/F file cannot be opened.
>>> 2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:340 Remote station
>>> id: SpanDSP Fax Ident
>>> 2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:341 Local station
>>> id:  SpanDSP Fax Ident
>>>
>>>
>>> what is going 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
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sun Feb 27 09:57:17 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 27 Feb 2011 01:57:17 -0500
Subject: [Freeswitch-users] Fax processing not successful - result
	(41)TIFF/F file cannot be opened.
References: <AANLkTim+v6WAxhuhkPswYwGEL_R2yUCp5Y6pAzt3o2Ry@mail.gmail.com><E4A1DC3E781847CEAC56CD709D7FF567@e1705><AANLkTi=X_VJQGX1Y0QEFDNUwgSaguk4S9=oZdkNp-sn7@mail.gmail.com><AANLkTim7ovBYhhicK-8HDhS_tQucywoJ050rmQNKWbqt@mail.gmail.com><C9BA852E5CC7484A84A44B83B0B69614@e1705>
	<AANLkTinQ62xWUnOfauf_sqkNbGYbRUSq8d4DRWxc5-cX@mail.gmail.com>
Message-ID: <C8970361188D4966A287B3B6CE17F109@e1705>

> If this is bug, please fix it.  If there is a catch, please post the solution and summarize on wiki page.
hjey, cool man, it's an open source mailing list....
/tmp is usually not a folder to execute anything.
try to change the folder with write permssion...
  ----- Original Message ----- 
  From: envelopes envelopes 
  To: FreeSWITCH Users Help 
  Sent: Sunday, February 27, 2011 1:46 AM
  Subject: Re: [Freeswitch-users] Fax processing not successful - result (41)TIFF/F file cannot be opened.


  http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg06601.html

  in fact, I am running as a root. so the reason in that thread doesn't apply to my case.

  root     22613  9.5  0.7  40064 15724 pts/2    SLl+ 22:35   0:41 ./bin/freeswitch

  /tmp directory has worldwide write permission:
  drwxrwxrwt  20 root root  4096 2011-02-26 22:17 tmp

  If this is bug, please fix it.  If there is a catch, please post the solution and summarize on wiki page.

  One of the particular reason that i like FS is prompt support and enthusiasm from FS developers.
  please do not lost your momentum.


  thanks!


  On Sat, Feb 26, 2011 at 10:10 PM, Madovsky <infos at madovsky.org> wrote:

    look at the the emailist archive,
    I remember that one had the same problem
    but don't remember the solution
      ----- Original Message ----- 
      From: envelopes envelopes 
      To: FreeSWITCH Users Help 
      Sent: Sunday, February 27, 2011 12:53 AM
      Subject: Re: [Freeswitch-users] Fax processing not successful - result (41)TIFF/F file cannot be opened.


      no one had similar issue? 


      On Fri, Feb 25, 2011 at 11:33 AM, envelopes envelopes <sunwood360 at gmail.com> wrote:

        I don't see any permission issue here. 


        /tmp$ ls -l /tmp/a.tif
        -rwxrwxrwx 1 root root 179378 2011-02-02 05:30 /tmp/a.tif


        receive side also writes to /tmp directroy.

        It is really weird. 


         


        On Fri, Feb 25, 2011 at 11:30 AM, Madovsky <infos at madovsky.org> wrote:

          check permissions
            ----- Original Message ----- 
            From: envelopes envelopes 
            To: FreeSWITCH Users Help 
            Sent: Friday, February 25, 2011 2:23 PM
            Subject: [Freeswitch-users] Fax processing not successful - result (41)TIFF/F file cannot be opened.


            I have the extension in default.xml;

            <extension name="fax_receive2">
                 <condition field="destination_number" expression="^(9979)$">
                   <action application="answer" />
                   <action application="playback" data="silence_stream://2000"/>
                   <action application="rxfax" data="/tmp/fax-${uuid}.rxfax.tiff"/>
                   <action application="hangup"/>
                 </condition>
               </extension>

            then in fs_cli : originate loopback/9979 &txfax(/tmp/a.tif)

            in fact, the file /tmp/a.tif does exist.

            /tmp$ ls -l /tmp/a.tif
            -rwxrwxrwx 1 root root 179378 2011-02-02 05:30 /tmp/a.tif


            2011-02-25 11:17:39.444902 [DEBUG] mod_spandsp_fax.c:1108 Raw write codec activation Success L16                                            
            2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:323 ==============================================================================     
            2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:335 Fax processing not successful - result (17) Received a DCN while waiting for a DIS.
            2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:340 Remote station id:                                                                 
            2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:341 Local station id:  SpanDSP Fax Ident                                               
            2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:342 Pages transferred: 0                                                               
            2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:344 Total fax pages:   0                                                               
            2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:345 Image resolution:  0x0                                                             
            2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:346 Transfer Rate:     14400                                                           
            2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:348 ECM status         off                                                             
            2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:349 remote country:                                                                    
            2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:350 remote vendor:                                                                     
            2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:351 remote model:                                                                      
            2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:353 ==============================================================================     
            2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:323 ==============================================================================     
            2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:335 Fax processing not successful - result (41) TIFF/F file cannot be opened.          
            2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:340 Remote station id: SpanDSP Fax Ident                                               
            2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:341 Local station id:  SpanDSP Fax Ident


            what is going wrong??

            thanks




--------------------------------------------------------------------


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            http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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            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


    _______________________________________________
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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 sunwood360 at gmail.com  Sun Feb 27 10:37:03 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Sat, 26 Feb 2011 23:37:03 -0800
Subject: [Freeswitch-users] Fax processing not successful - result
 (41)TIFF/F file cannot be opened.
In-Reply-To: <C8970361188D4966A287B3B6CE17F109@e1705>
References: <AANLkTim+v6WAxhuhkPswYwGEL_R2yUCp5Y6pAzt3o2Ry@mail.gmail.com>
	<E4A1DC3E781847CEAC56CD709D7FF567@e1705>
	<AANLkTi=X_VJQGX1Y0QEFDNUwgSaguk4S9=oZdkNp-sn7@mail.gmail.com>
	<AANLkTim7ovBYhhicK-8HDhS_tQucywoJ050rmQNKWbqt@mail.gmail.com>
	<C9BA852E5CC7484A84A44B83B0B69614@e1705>
	<AANLkTinQ62xWUnOfauf_sqkNbGYbRUSq8d4DRWxc5-cX@mail.gmail.com>
	<C8970361188D4966A287B3B6CE17F109@e1705>
Message-ID: <AANLkTimzCxDgDGxdC0UVKvi62HGOvymbcH-3TjXoF80N@mail.gmail.com>

It seems not  a permission issue.

please look at line 49 & 50 @ pastebin

http://pastebin.com/M2LhT1zD


49. 2011-02-26 23:30:40.964962 [DEBUG] mod_spandsp_fax.c:335 Fax processing
not successful - result (17) Received a DCN while waiting for a DIS.
50. 2011-02-26 23:30:40.964962 [DEBUG] mod_spandsp_fax.c:335 Fax processing
not successful - result (41) TIFF/F file cannot be opened.



On Sat, Feb 26, 2011 at 10:57 PM, Madovsky <infos at madovsky.org> wrote:

>  > If this is bug, please fix it.  If there is a catch, please post the
> solution and summarize on wiki page.
> hjey, cool man, it's an open source mailing list....
> /tmp is usually not a folder to execute anything.
> try to change the folder with write permssion...
>
> ----- Original Message -----
> *From:* envelopes envelopes <sunwood360 at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Sunday, February 27, 2011 1:46 AM
> *Subject:* Re: [Freeswitch-users] Fax processing not successful - result
> (41)TIFF/F file cannot be opened.
>
>
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg06601.html
>
> in fact, I am running as a root. so the reason in that thread doesn't apply
> to my case.
>
> root     22613  9.5  0.7  40064 15724 pts/2    SLl+ 22:35   0:41
> ./bin/freeswitch
>
> /tmp directory has worldwide write permission:
> drwxrwxrwt  20 root root  4096 2011-02-26 22:17 tmp
>
> If this is bug, please fix it.  If there is a catch, please post the
> solution and summarize on wiki page.
>
> One of the particular reason that i like FS is prompt support and
> enthusiasm from FS developers.
> please do not lost your momentum.
>
>
> thanks!
>
> On Sat, Feb 26, 2011 at 10:10 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  look at the the emailist archive,
>> I remember that one had the same problem
>> but don't remember the solution
>>
>>  ----- Original Message -----
>> *From:* envelopes envelopes <sunwood360 at gmail.com>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>   *Sent:* Sunday, February 27, 2011 12:53 AM
>> *Subject:* Re: [Freeswitch-users] Fax processing not successful - result
>> (41)TIFF/F file cannot be opened.
>>
>> no one had similar issue?
>>
>> On Fri, Feb 25, 2011 at 11:33 AM, envelopes envelopes <
>> sunwood360 at gmail.com> wrote:
>>
>>> I don't see any permission issue here.
>>>
>>>
>>> /tmp$ ls -l /tmp/a.tif
>>> -rwxrwxrwx 1 root root 179378 2011-02-02 05:30 /tmp/a.tif
>>>
>>> receive side also writes to /tmp directroy.
>>>
>>> It is really weird.
>>>
>>>
>>>
>>>
>>> On Fri, Feb 25, 2011 at 11:30 AM, Madovsky <infos at madovsky.org> wrote:
>>>
>>>>  check permissions
>>>>
>>>>   ----- Original Message -----
>>>> *From:* envelopes envelopes <sunwood360 at gmail.com>
>>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>> *Sent:* Friday, February 25, 2011 2:23 PM
>>>> *Subject:* [Freeswitch-users] Fax processing not successful - result
>>>> (41)TIFF/F file cannot be opened.
>>>>
>>>> I have the extension in default.xml;
>>>>
>>>> <extension name="fax_receive2">
>>>>      <condition field="destination_number" expression="^(9979)$">
>>>>        <action application="answer" />
>>>>        <action application="playback" data="silence_stream://2000"/>
>>>>        <action application="rxfax" data="/tmp/fax-${uuid}.rxfax.tiff"/>
>>>>        <action application="hangup"/>
>>>>      </condition>
>>>>    </extension>
>>>>
>>>> then in fs_cli : originate loopback/9979 &txfax(/tmp/a.tif)
>>>>
>>>> in fact, the file /tmp/a.tif does exist.
>>>>
>>>> /tmp$ ls -l /tmp/a.tif
>>>> -rwxrwxrwx 1 root root 179378 2011-02-02 05:30 /tmp/a.tif
>>>>
>>>>
>>>> 2011-02-25 11:17:39.444902 [DEBUG] mod_spandsp_fax.c:1108 Raw write
>>>> codec activation Success L16
>>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:323
>>>> ==============================================================================
>>>>
>>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:335 Fax processing
>>>> not successful - result (17) Received a DCN while waiting for a DIS.
>>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:340 Remote station
>>>> id:
>>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:341 Local station
>>>> id:  SpanDSP Fax Ident
>>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:342 Pages
>>>> transferred: 0
>>>>
>>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:344 Total fax
>>>> pages:   0
>>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:345 Image
>>>> resolution:  0x0
>>>>
>>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:346 Transfer
>>>> Rate:     14400
>>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:348 ECM
>>>> status
>>>> off
>>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:349 remote
>>>> country:
>>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:350 remote
>>>> vendor:
>>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:351 remote
>>>> model:
>>>> 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:353
>>>> ==============================================================================
>>>>
>>>> 2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:323
>>>> ==============================================================================
>>>>
>>>> 2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:335 Fax processing
>>>> not successful - result (41) TIFF/F file cannot be opened.
>>>> 2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:340 Remote station
>>>> id: SpanDSP Fax Ident
>>>> 2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:341 Local station
>>>> id:  SpanDSP Fax Ident
>>>>
>>>>
>>>> what is going 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
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>  ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 27 11:40:51 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 27 Feb 2011 00:40:51 -0800
Subject: [Freeswitch-users] Compilation with -march options
In-Reply-To: <1296444815434-5976090.post@n2.nabble.com>
References: <AANLkTi=civOgvpz55hxEifdkcmKWmZABsitj290cCkxN@mail.gmail.com>
	<1296444815434-5976090.post@n2.nabble.com>
Message-ID: <AANLkTinc=5cmFngN5uJbWPGLQHBjQ65+rNh6cnYNnX-E@mail.gmail.com>

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 curriegrad2004 at gmail.com  Sun Feb 27 11:43:22 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 27 Feb 2011 00:43:22 -0800
Subject: [Freeswitch-users] Compilation with -march options
In-Reply-To: <AANLkTinc=5cmFngN5uJbWPGLQHBjQ65+rNh6cnYNnX-E@mail.gmail.com>
References: <AANLkTi=civOgvpz55hxEifdkcmKWmZABsitj290cCkxN@mail.gmail.com>
	<1296444815434-5976090.post@n2.nabble.com>
	<AANLkTinc=5cmFngN5uJbWPGLQHBjQ65+rNh6cnYNnX-E@mail.gmail.com>
Message-ID: <AANLkTinJ2Qf06vcrCK+=uwBX+7NNwLd-SxJrz-fi=6rP@mail.gmail.com>

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 steveu at coppice.org  Sun Feb 27 13:55:36 2011
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 27 Feb 2011 18:55:36 +0800
Subject: [Freeswitch-users] Fax processing not successful - result
 (41)TIFF/F file cannot be opened.
In-Reply-To: <AANLkTimzCxDgDGxdC0UVKvi62HGOvymbcH-3TjXoF80N@mail.gmail.com>
References: <AANLkTim+v6WAxhuhkPswYwGEL_R2yUCp5Y6pAzt3o2Ry@mail.gmail.com>	<E4A1DC3E781847CEAC56CD709D7FF567@e1705>	<AANLkTi=X_VJQGX1Y0QEFDNUwgSaguk4S9=oZdkNp-sn7@mail.gmail.com>	<AANLkTim7ovBYhhicK-8HDhS_tQucywoJ050rmQNKWbqt@mail.gmail.com>	<C9BA852E5CC7484A84A44B83B0B69614@e1705>	<AANLkTinQ62xWUnOfauf_sqkNbGYbRUSq8d4DRWxc5-cX@mail.gmail.com>	<C8970361188D4966A287B3B6CE17F109@e1705>
	<AANLkTimzCxDgDGxdC0UVKvi62HGOvymbcH-3TjXoF80N@mail.gmail.com>
Message-ID: <4D6A2DA8.6050209@coppice.org>

On 02/27/2011 03:37 PM, envelopes envelopes wrote:
> It seems not  a permission issue.
>
> please look at line 49 & 50 @ pastebin
>
> http://pastebin.com/M2LhT1zD
>
>
> 49. 2011-02-26 23:30:40.964962 [DEBUG] mod_spandsp_fax.c:335 Fax 
> processing not successful - result (17) Received a DCN while waiting 
> for a DIS.
> 50. 2011-02-26 23:30:40.964962 [DEBUG] mod_spandsp_fax.c:335 Fax 
> processing not successful - result (41) TIFF/F file cannot be opened.
>
Line 50 says the file to be sent can't be opened. You have three 
possibilities - a non-existent file, permissions issues, or a corrupt file.

Steve



From infos at madovsky.org  Sun Feb 27 16:58:20 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 27 Feb 2011 08:58:20 -0500
Subject: [Freeswitch-users] Fax processing not successful - result
	(41)TIFF/F file cannot be opened.
References: <AANLkTim+v6WAxhuhkPswYwGEL_R2yUCp5Y6pAzt3o2Ry@mail.gmail.com><E4A1DC3E781847CEAC56CD709D7FF567@e1705><AANLkTi=X_VJQGX1Y0QEFDNUwgSaguk4S9=oZdkNp-sn7@mail.gmail.com><AANLkTim7ovBYhhicK-8HDhS_tQucywoJ050rmQNKWbqt@mail.gmail.com><C9BA852E5CC7484A84A44B83B0B69614@e1705><AANLkTinQ62xWUnOfauf_sqkNbGYbRUSq8d4DRWxc5-cX@mail.gmail.com><C8970361188D4966A287B3B6CE17F109@e1705>
	<AANLkTimzCxDgDGxdC0UVKvi62HGOvymbcH-3TjXoF80N@mail.gmail.com>
Message-ID: <AFDB26CC60E54BF88BCE49FD5E2231B5@e1705>

> It seems not  a permission issue.
youor debug says "file cannot be opened".
read below what I said.... "/tmp is usually not a folder to execute anything."...
  ----- Original Message ----- 
  From: envelopes envelopes 
  To: FreeSWITCH Users Help 
  Sent: Sunday, February 27, 2011 2:37 AM
  Subject: Re: [Freeswitch-users] Fax processing not successful - result (41)TIFF/F file cannot be opened.


  It seems not  a permission issue.

  please look at line 49 & 50 @ pastebin

  http://pastebin.com/M2LhT1zD



  49. 2011-02-26 23:30:40.964962 [DEBUG] mod_spandsp_fax.c:335 Fax processing not successful - result (17) Received a DCN while waiting for a DIS.
  50. 2011-02-26 23:30:40.964962 [DEBUG] mod_spandsp_fax.c:335 Fax processing not successful - result (41) TIFF/F file cannot be opened.   




  On Sat, Feb 26, 2011 at 10:57 PM, Madovsky <infos at madovsky.org> wrote:

    > If this is bug, please fix it.  If there is a catch, please post the solution and summarize on wiki page.

    hjey, cool man, it's an open source mailing list....
    /tmp is usually not a folder to execute anything.
    try to change the folder with write permssion...
      ----- Original Message ----- 
      From: envelopes envelopes 
      To: FreeSWITCH Users Help 
      Sent: Sunday, February 27, 2011 1:46 AM
      Subject: Re: [Freeswitch-users] Fax processing not successful - result (41)TIFF/F file cannot be opened.


      http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg06601.html

      in fact, I am running as a root. so the reason in that thread doesn't apply to my case.

      root     22613  9.5  0.7  40064 15724 pts/2    SLl+ 22:35   0:41 ./bin/freeswitch

      /tmp directory has worldwide write permission:
      drwxrwxrwt  20 root root  4096 2011-02-26 22:17 tmp

      If this is bug, please fix it.  If there is a catch, please post the solution and summarize on wiki page.

      One of the particular reason that i like FS is prompt support and enthusiasm from FS developers.
      please do not lost your momentum.


      thanks!


      On Sat, Feb 26, 2011 at 10:10 PM, Madovsky <infos at madovsky.org> wrote:

        look at the the emailist archive,
        I remember that one had the same problem
        but don't remember the solution
          ----- Original Message ----- 
          From: envelopes envelopes 
          To: FreeSWITCH Users Help 
          Sent: Sunday, February 27, 2011 12:53 AM
          Subject: Re: [Freeswitch-users] Fax processing not successful - result (41)TIFF/F file cannot be opened.


          no one had similar issue? 


          On Fri, Feb 25, 2011 at 11:33 AM, envelopes envelopes <sunwood360 at gmail.com> wrote:

            I don't see any permission issue here. 


            /tmp$ ls -l /tmp/a.tif
            -rwxrwxrwx 1 root root 179378 2011-02-02 05:30 /tmp/a.tif


            receive side also writes to /tmp directroy.

            It is really weird. 


             


            On Fri, Feb 25, 2011 at 11:30 AM, Madovsky <infos at madovsky.org> wrote:

              check permissions
                ----- Original Message ----- 
                From: envelopes envelopes 
                To: FreeSWITCH Users Help 
                Sent: Friday, February 25, 2011 2:23 PM
                Subject: [Freeswitch-users] Fax processing not successful - result (41)TIFF/F file cannot be opened.


                I have the extension in default.xml;

                <extension name="fax_receive2">
                     <condition field="destination_number" expression="^(9979)$">
                       <action application="answer" />
                       <action application="playback" data="silence_stream://2000"/>
                       <action application="rxfax" data="/tmp/fax-${uuid}.rxfax.tiff"/>
                       <action application="hangup"/>
                     </condition>
                   </extension>

                then in fs_cli : originate loopback/9979 &txfax(/tmp/a.tif)

                in fact, the file /tmp/a.tif does exist.

                /tmp$ ls -l /tmp/a.tif
                -rwxrwxrwx 1 root root 179378 2011-02-02 05:30 /tmp/a.tif


                2011-02-25 11:17:39.444902 [DEBUG] mod_spandsp_fax.c:1108 Raw write codec activation Success L16                                            
                2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:323 ==============================================================================     
                2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:335 Fax processing not successful - result (17) Received a DCN while waiting for a DIS.
                2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:340 Remote station id:                                                                 
                2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:341 Local station id:  SpanDSP Fax Ident                                               
                2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:342 Pages transferred: 0                                                               
                2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:344 Total fax pages:   0                                                               
                2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:345 Image resolution:  0x0                                                             
                2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:346 Transfer Rate:     14400                                                           
                2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:348 ECM status         off                                                             
                2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:349 remote country:                                                                    
                2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:350 remote vendor:                                                                     
                2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:351 remote model:                                                                      
                2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:353 ==============================================================================     
                2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:323 ==============================================================================     
                2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:335 Fax processing not successful - result (41) TIFF/F file cannot be opened.          
                2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:340 Remote station id: SpanDSP Fax Ident                                               
                2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:341 Local station id:  SpanDSP Fax Ident


                what is going wrong??

                thanks




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From Nabble at slickdeals.endjunk.com  Sun Feb 27 17:05:06 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 27 Feb 2011 06:05:06 -0800 (PST)
Subject: [Freeswitch-users] freeswitch -nc -nosql
In-Reply-To: <AANLkTikjkTPk3A2FmoRo32WhKxNFaX5YL7Wxezaa6zqG@mail.gmail.com>
References: <1298688216720-6066718.post@n2.nabble.com>
	<AANLkTikjkTPk3A2FmoRo32WhKxNFaX5YL7Wxezaa6zqG@mail.gmail.com>
Message-ID: <1298815506369-6070268.post@n2.nabble.com>


Anthony Minessale wrote:
> 
> Fixed
> 
> commit 1cbb8f221b2e76ec66c72d7f09d7f6ae681764b8
> Author: Anthony Minessale <anthm at freeswitch.org>
> Date:   Fri Feb 25 21:39:37 2011 -0600
> 
>     supress errors caused from -nosql as reported on the mailing list
> and not on JIRA like it should be http://jira.freeswitch.org
OK. That nasty message no longer shows up. But, actually having -nosql
switch gives no SQL database. This means, if I issue a help command, the CLI
will, instead of spitting out the help messages, complain with -ERR SQL
DISABLED NO DATA AVAILABLE! as shown below. Unless there is some settings I
missed, it looks like something isn't working right with SQL that causes FS
to crash. 
2011-02-27 08:53:38.602830 [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/        *
************************************************************

2011-02-27 08:53:38.602830 [CONSOLE] switch_core.c:1768 
FreeSWITCH Version 1.0.head (git-fd7bbc0 2011-02-26 11-45-19 +0100) Started.
Max Sessions[1000]
Session Rate[30]
SQL [Disabled]
freeswitch at DockStar> version

FreeSWITCH Version 1.0.head (git-fd7bbc0 2011-02-26 11-45-19 +0100)
freeswitch at DockStar> help


Valid Commands:

-ERR SQL DISABLED NO DATA AVAILABLE!

freeswitch at DockStar>

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-nc-nosql-tp6066718p6070268.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From gmaruzz at gmail.com  Sun Feb 27 17:26:03 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sun, 27 Feb 2011 15:26:03 +0100
Subject: [Freeswitch-users] freeswitch -nc -nosql
In-Reply-To: <1298815506369-6070268.post@n2.nabble.com>
References: <1298688216720-6066718.post@n2.nabble.com>
	<AANLkTikjkTPk3A2FmoRo32WhKxNFaX5YL7Wxezaa6zqG@mail.gmail.com>
	<1298815506369-6070268.post@n2.nabble.com>
Message-ID: <AANLkTin8ZRoFSzFoiaxNKg0VDvXi5YebMe_Z6mpyFRgJ@mail.gmail.com>

On Sun, Feb 27, 2011 at 3:05 PM, mazilo <Nabble at slickdeals.endjunk.com> wrote:

>OK. That nasty message no longer shows up. But, actually having -nosql
> switch gives no SQL database. This means, if I issue a help command, the CLI
> will, instead of spitting out the help messages, complain with -ERR SQL
> DISABLED NO DATA AVAILABLE! as shown below. Unless there is some settings I
> missed, it looks like something isn't working right with SQL that causes FS
> to crash.

there are no crash.
That is the expected behavior.



> 2011-02-27 08:53:38.602830 [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/ ? ? ? ?*
> ************************************************************
>
> 2011-02-27 08:53:38.602830 [CONSOLE] switch_core.c:1768
> FreeSWITCH Version 1.0.head (git-fd7bbc0 2011-02-26 11-45-19 +0100) Started.
> Max Sessions[1000]
> Session Rate[30]
> SQL [Disabled]
> freeswitch at DockStar> version
>
> FreeSWITCH Version 1.0.head (git-fd7bbc0 2011-02-26 11-45-19 +0100)
> freeswitch at DockStar> help
>
>
> Valid Commands:
>
> -ERR SQL DISABLED NO DATA AVAILABLE!
>
> freeswitch at DockStar>
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-nc-nosql-tp6066718p6070268.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 matzemuc86 at gmail.com  Sun Feb 27 18:04:44 2011
From: matzemuc86 at gmail.com (MatzeMuc86)
Date: Sun, 27 Feb 2011 16:04:44 +0100
Subject: [Freeswitch-users] implementing stereo (e.g. portaudio to test
	it easily)
In-Reply-To: <4D67FBED.6060101@communicatefreely.net>
References: <AANLkTikL9SQPdsJ6XF-AUZuLQ3Z-AG7Eqgj2BsL=CZ4S@mail.gmail.com>	<AANLkTinowzmzt8NxB-FtifkZEQZ8s3OQAWLuC18PN60Q@mail.gmail.com>	<AANLkTikDFqS_Y9XS-gnVvnfPhAW1Eri=zy4XB3vAS2N0@mail.gmail.com>	<4d66ee22.857a0e0a.6c3f.0343@mx.google.com>	<AANLkTimrx5q076f+EAjgMQBh4c4kZYE2Mjq401oV2V7z@mail.gmail.com>	<000801cbd4d5$13362830$39a27890$@gmail.com>	<1298643188393-6064582.post@n2.nabble.com>	<000f01cbd4fe$015301f0$03f905d0$@gmail.com>	<AANLkTikjch_cPU8XzzF4APjWDAcQNcodvtZU1C3OkKP_@mail.gmail.com>	<AANLkTinrvj7Ye22GVu+CtaE+ApzP3OHGm=JdjsO1B2-i@mail.gmail.com>
	<4D67FBED.6060101@communicatefreely.net>
Message-ID: <001d01cbd68f$ab70e0c0$0252a240$@gmail.com>

Hello,

sorry for my late reply:

@mazilo: I have not tested the HRTF my own but I know that it is not such an
extreme thing. The point is: 5 people in a conference => every participant
needs 4 other people mixed up => 5x4 = 20 HRTF mixings parallel => n*(n-1)
complexity :-(. If I integrate this stuff in FreeSWITCH I will, of course,
offer it as open source as well (we can try to integrate it or whatever the
main developers want to have). But at first I need to get it working.

@Steve: Thanks. Is there some documentation for which I am too stupid to
find it?

@Anthony:
- Playing vs. Codec: Sounds logical! Codec: What about L16. I think there is
a 1 and a 2 channel version available:
http://en.wikipedia.org/wiki/RTP_audio_video_profile
- I completely understand why there is no stereo until now.
- Portaudio: I'm not sure if I really need this mod for my purpose. It was
only a nice idea to test and debug how media is handled.
- I understood SIP the way as it is only a protocol for "managing" things.
The media flow uses RTP and RTP can use different codecs and different
numbers of channels. The SDP can also handle different codecs for sending
and receiving. Therefore, I thought, sending mono and receiving stereo is
completely SIP regular!??

@tim: I'm not sure if I understood your idea completely in the right way:
Your main idea is to use one sip session for each channel? But mixing would
still be done by the server, not the client? How should I take care that the
left and the right channels, processed by two different SIP + RTP sessions,
are played simultaneously?

Thanks to all your great support!!!
MatzeMuc86

-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Tim
St. Pierre
Gesendet: Freitag, 25. Februar 2011 19:59
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] implementing stereo (e.g. portaudio to test
it easily)

Here's an idea  - it might be a little easier to make interoperate with
other SIP devices.

What if you built a conferencing endpoint that mixed all the audio locally
into a 3D matrix?

Each participant in the call could use any of the common codecs, but would
have an extra SIP header than indicated their position in the sound field.

Here's a bit of an overview:

A regular Freeswitch system takes care of registering endpoints, routing,
etc.  Just like a normal PBX.  The only thing special it would do is set add
a custom SIP header with a person's sound field position, based on a
directory variable.

Each conference unit would be a stripped down freeswitch system with a
custom modified mod_conference.  In addition to mixing audio to mono like
the regular mod_conference does, it would also generate a stereo mix, based
on the position information of each participant.  It could dump out the
stereo audio directly to a named pipe, mod_shout, or directly to audio
hardware.  You could probably shoehorn a stereo input from hardware as well.
Essentially, you are building a stereo conference phone.

If you want to join up multiple conference units, just establish two
sessions between each conference - one panned hard left, the other hard
right.  They can then use G722 without any modification.

This probably isn't quite as nice as having true stereo codecs and media
handling, but it may be easier to implement.  After all, telecom is very
much a mono world.

-Tim
> The main reason we do not have stereo completely implemented is there 
> was not really any codec or devices that could handle it to bother 
> testing with.
>   


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at 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  Sun Feb 27 18:21:19 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 27 Feb 2011 07:21:19 -0800 (PST)
Subject: [Freeswitch-users] freeswitch -nc -nosql
In-Reply-To: <AANLkTin8ZRoFSzFoiaxNKg0VDvXi5YebMe_Z6mpyFRgJ@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>
Message-ID: <1298820079407-6070409.post@n2.nabble.com>


Giovanni Maruzzelli-2 wrote:
> 
> On Sun, Feb 27, 2011 at 3:05 PM, mazilo <Nabble at slickdeals.endjunk.com>
> wrote:
> 
>>OK. That nasty message no longer shows up. But, actually having -nosql
>> switch gives no SQL database. This means, if I issue a help command, the
>> CLI
>> will, instead of spitting out the help messages, complain with -ERR SQL
>> DISABLED NO DATA AVAILABLE! as shown below. Unless there is some settings
>> I
>> missed, it looks like something isn't working right with SQL that causes
>> FS
>> to crash.
> 
> there are no crash.
> That is the expected behavior.
But, I can't get the help menu. OTOH, the -nosql switch disables SQL feature
and that's no good if one depends on SQL.

BTW, I did a freeswitch -waste run and here is the 
http://pastebin.com/eyALThdX screen dump  for your perusal. Line# 914
clearly shows FS Segmentation fault after SQL [Enabled].

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-nc-nosql-tp6066718p6070409.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From spencer at 5ninesolutions.com  Sun Feb 27 18:32:11 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 27 Feb 2011 15:32:11 +0000
Subject: [Freeswitch-users] Polycom G722 negotiation
In-Reply-To: <2867CE13-6EED-4FAF-9DF1-49451D164721@freeswitch.org>
References: <AE216221-5892-4337-8985-1889410A866F@5ninesolutions.com>
	<2867CE13-6EED-4FAF-9DF1-49451D164721@freeswitch.org>
Message-ID: <0ABF546B-4000-4936-AD22-ED6B956277D5@5ninesolutions.com>

freeswitch at internal> version
FreeSWITCH Version 1.0.head (hacked-20110223T050150Z)


On Feb 25, 2011, at 4:18 PM, Brian West wrote:

> What version of FreeSWITCH are you running?
>
> /b
>
> On Feb 24, 2011, at 11:45 PM, Spencer Thomason wrote:
>
>> Hello all,
>> I have several Polycom IP-650s.  On the external profile I have:
>> CODECS IN        	G722,PCMU,G729,GSM
>> CODECS OUT       	PCMU,G729
>>
>> On the internal profile I have
>> CODECS IN        	G722,PCMU,G729,GSM
>> CODECS OUT       	G722,PCMU,G729,GSM
>>
>> The Polycoms have a similar codec priority set:
>> v=0
>> o=- 1167611879 1167611879 IN IP4 10.59.1.243
>> s=Polycom IP Phone
>> c=IN IP4 10.59.1.243
>> t=0 0
>> a=sendrecv
>> m=audio 2236 RTP/AVP 9 0 18 101
>> a=rtpmap:9 G722/8000
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:18 G729/8000
>> a=fmtp:18 annexb=yes
>> a=rtpmap:101 telephone-event/8000
>>
>> The issue I have is that if I leave the internal profile to  
>> generous and a user places a call from the PSTN on hold which is  
>> PCMU, when the user tries to pick up the call, the Polycoms'  
>> preference then changes the call to G722 and there is no audio.  If  
>> I set the profile to greedy, since the call is already PCMU  
>> Freeswitch keeps it at PCMU and audio is fine.  I realize this is  
>> the intended behavior for greedy vs generous but my question is why  
>> is there no audio when switching the codecs?  I didn't want to  
>> clutter this up any more but I'm happy to provide traces however  
>> the only real difference is the codec order.
>>
>> Thanks,
>> 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 jonas.gauffin at gmail.com  Sun Feb 27 20:39:58 2011
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Sun, 27 Feb 2011 18:39:58 +0100
Subject: [Freeswitch-users] Cordless phones
Message-ID: <AANLkTikFAO9_bkpmXvk9AKBRTT34_811r_jSnfA7SO47@mail.gmail.com>

Hi,

I'm looking for IP-dects where a base station can handle 5 phones (which all
should be able to talk at the same time).
All suggestions are welcome.

Regards,
  Jonas
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From mario_fs at mgtech.com  Sun Feb 27 20:59:23 2011
From: mario_fs at mgtech.com (Mario G)
Date: Sun, 27 Feb 2011 09:59:23 -0800
Subject: [Freeswitch-users] Cordless phones
In-Reply-To: <AANLkTikFAO9_bkpmXvk9AKBRTT34_811r_jSnfA7SO47@mail.gmail.com>
References: <AANLkTikFAO9_bkpmXvk9AKBRTT34_811r_jSnfA7SO47@mail.gmail.com>
Message-ID: <C4394984-D23F-40C0-865F-A172ACE96A4B@mgtech.com>

Close: Snom M9 handles 9 phones but only 4 concurrent calls. How about getting two? The rep said they are supposed to be in the US February. I am planning on testing one.

On Feb 27, 2011, at 9:39 AM, Jonas Gauffin wrote:

> Hi,
> 
> I'm looking for IP-dects where a base station can handle 5 phones (which all should be able to talk at the same time).
> All suggestions are welcome.
> 
> Regards,
>   Jonas
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 jonas.gauffin at gmail.com  Sun Feb 27 21:46:30 2011
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Sun, 27 Feb 2011 19:46:30 +0100
Subject: [Freeswitch-users] Cordless phones
In-Reply-To: <C4394984-D23F-40C0-865F-A172ACE96A4B@mgtech.com>
References: <AANLkTikFAO9_bkpmXvk9AKBRTT34_811r_jSnfA7SO47@mail.gmail.com>
	<C4394984-D23F-40C0-865F-A172ACE96A4B@mgtech.com>
Message-ID: <AANLkTikxpkc-MEp12xVaWmEYgXC4JgfsNTTXuWD+RXDy@mail.gmail.com>

Ok, didn't know that M9 supported four calls. One of my resellers are
already using Snom M9 at two customer sites. It is working well, although
both offices are small.

Other suggestions are welcome too.

On Sun, Feb 27, 2011 at 6:59 PM, Mario G <mario_fs at mgtech.com> wrote:

> Close: Snom M9 handles 9 phones but only 4 concurrent calls. How about
> getting two? The rep said they are supposed to be in the US February. I am
> planning on testing one.
>
> On Feb 27, 2011, at 9:39 AM, Jonas Gauffin wrote:
>
> > Hi,
> >
> > I'm looking for IP-dects where a base station can handle 5 phones (which
> all should be able to talk at the same time).
> > All suggestions are welcome.
> >
> > Regards,
> >   Jonas
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sjmudd at pobox.com  Sun Feb 27 22:52:09 2011
From: sjmudd at pobox.com (Simon J Mudd)
Date: Sun, 27 Feb 2011 20:52:09 +0100
Subject: [Freeswitch-users] Applying multiple limits for calls by
	sip_auth_username or by sip_received_ip
Message-ID: <20110227195209.GA29496@mad06.wl0.org>

Hi,

I was pointed to the limit application as a way to control call
usage. This seems quite good for me if I'm paranoid about my FreeSWITCH
setup being attacked.

http://wiki.freeswitch.org/wiki/Mod_limit gives a cuople of examples using slightly
different syntax, one using data="db ..." and the other data="hash ...". However the parameters
are not clearly explained. The Rate Limiting calls / Anti SPIT example is what interests me
but it's not clear:
(1) exactly how the first 2 fields are used, and
(2) if I want to set different rates for the same 2 fields but for different time periods if I can
or one configuration will affect another one.

I had a very quick look at the code but it seems to have changed from v1.0.6 to the current head
and I wasn't sure where to look in the "core" code to figure out the usage.

Basically what I'm looking to do is very similar to the Rate Limiting
example, something like:

in dialplan/default.xml:

...
    <!-- taken from: http://wiki.freeswitch.org/wiki/Mod_limit -->
    <extension name="limit_exceeded">
      <condition field="destination_number" expression="^limit_exceeded$">
        <action application="playback" data="$${CUSTOM_SOUNDS_DIR}/limit_exceeded.wav"/>
        <action application="hangup"/>
      </condition>
    </extension>

    <extension name="limit" continue="true">
      <condition>
        <!-- use the A, B, C prefixes to avoid the different per-period   -->
        <!-- limits interfering with each other                           -->
        <action application="limit" data="hash $${domain} A${sip_received_ip}   $${MAX_CALLS_FROM_IP_PER_SEC}/1" />
        <action application="limit" data="hash $${domain} B${sip_received_ip}   $${MAX_CALLS_FROM_IP_PER_MIN}/60" />
        <action application="limit" data="hash $${domain} C${sip_received_ip}   $${MAX_CALLS_FROM_IP_PER_HOUR}/3600" />
        <action application="limit" data="hash $${domain} A${sip_auth_username} $${MAX_CALLS_BY_AUTH_USER_PER_SEC}/1" />
        <action application="limit" data="hash $${domain} B${sip_auth_username} $${MAX_CALLS_BY_AUTH_USER_PER_MIN}/60" />
        <action application="limit" data="hash $${domain} C${sip_auth_username} $${MAX_CALLS_BY_AUTH_USER_PER_HOUR}/3600" />
      </condition>
    </extension> 
...

Am I right in assuming that it's necessary to add  a prefix when using
multiple limits for different periods at the same time?

Also I'd like to apply something similar for registration attempts
(successful or otherwise), basically similar to the ${sip_received_ip}
lines, but am not sure where that would be applied. Could someone point
me to the right location?

Thanks,

Simon


From saeedahmad1981 at gmail.com  Sun Feb 27 23:05:04 2011
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Sun, 27 Feb 2011 21:05:04 +0100
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
Message-ID: <AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>

Great!

want to see it soon.
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From saeedahmad1981 at gmail.com  Sun Feb 27 23:06:28 2011
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Sun, 27 Feb 2011 21:06:28 +0100
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
Message-ID: <AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>

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.
>
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From sunwood360 at gmail.com  Sun Feb 27 23:14:18 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Sun, 27 Feb 2011 12:14:18 -0800
Subject: [Freeswitch-users] Fax processing not successful - result
 (41)TIFF/F file cannot be opened.
In-Reply-To: <2CE8763A-5E5D-4164-9D06-CBD96DCD581F@3js.de>
References: <AANLkTim+v6WAxhuhkPswYwGEL_R2yUCp5Y6pAzt3o2Ry@mail.gmail.com>
	<E4A1DC3E781847CEAC56CD709D7FF567@e1705>
	<AANLkTi=X_VJQGX1Y0QEFDNUwgSaguk4S9=oZdkNp-sn7@mail.gmail.com>
	<AANLkTim7ovBYhhicK-8HDhS_tQucywoJ050rmQNKWbqt@mail.gmail.com>
	<C9BA852E5CC7484A84A44B83B0B69614@e1705>
	<AANLkTinQ62xWUnOfauf_sqkNbGYbRUSq8d4DRWxc5-cX@mail.gmail.com>
	<C8970361188D4966A287B3B6CE17F109@e1705>
	<AANLkTimzCxDgDGxdC0UVKvi62HGOvymbcH-3TjXoF80N@mail.gmail.com>
	<2CE8763A-5E5D-4164-9D06-CBD96DCD581F@3js.de>
Message-ID: <AANLkTim0Fj873XXtcY=YVh+Tj_U7ibJJh9tsvnJeFtaO@mail.gmail.com>

yes!  that is it.

the tif file that I used was copied from a sample file included in windows 7
installation.  C:\Users\xyz\Documents\Fax\Inbox\WelcomeFax.tif

per your tip, I generated the tif file by myself and it worked now.

1) grab a pdf file.
2) sudo apt-get install imagemagick
3) convert -density 204x98 -resize 1728x1186\! -monochrome -compress Fax
./test.pdf ./test.tiff
4) originate loopback/9979 &txfax(/tmp/test.tiff)

whoever has FS wiki edit permission please update mod_spandsp. don't let
future users frustrated by this damn stupid issue.
I also believe this is a bug in mod_spandsp since it can't handle standard
windows tif file.

Johannes, thanks a lot for your great tip.




On Sun, Feb 27, 2011 at 1:37 AM, Johannes Jakob <jjj at 3js.de> wrote:

> Hi,
>
> I'm quite new to the list and don't feel like an expert, so I'm not going
> to reply to the list, but to you directly.
>
> I had a similar problem some days ago and I think the problem was the tif
> file itself.
> I had converted a normal picture into tiff format and FS couldn't read it
> the way it wanted to.
>
> I then changed to a file that FS had written himself (a received fax) and
> that worked just fine.
>
> Maybe this helps ;)
>
> Greetings,
>  John
>
>
> On 27.02.2011, at 08:37, envelopes envelopes wrote:
>
> > It seems not  a permission issue.
> >
> > please look at line 49 & 50 @ pastebin
> >
> > http://pastebin.com/M2LhT1zD
> >
> >
> > 49. 2011-02-26 23:30:40.964962 [DEBUG] mod_spandsp_fax.c:335 Fax
> processing not successful - result (17) Received a DCN while waiting for a
> DIS.
> > 50. 2011-02-26 23:30:40.964962 [DEBUG] mod_spandsp_fax.c:335 Fax
> processing not successful - result (41) TIFF/F file cannot be opened.
> >
> >
> >
> > On Sat, Feb 26, 2011 at 10:57 PM, Madovsky <infos at madovsky.org> wrote:
> > > If this is bug, please fix it.  If there is a catch, please post the
> solution and summarize on wiki page.
> > hjey, cool man, it's an open source mailing list....
> > /tmp is usually not a folder to execute anything.
> > try to change the folder with write permssion...
> > ----- Original Message -----
> > From: envelopes envelopes
> > To: FreeSWITCH Users Help
> > Sent: Sunday, February 27, 2011 1:46 AM
> > Subject: Re: [Freeswitch-users] Fax processing not successful - result
> (41)TIFF/F file cannot be opened.
> >
> >
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg06601.html
> >
> > in fact, I am running as a root. so the reason in that thread doesn't
> apply to my case.
> >
> > root     22613  9.5  0.7  40064 15724 pts/2    SLl+ 22:35   0:41
> ./bin/freeswitch
> >
> > /tmp directory has worldwide write permission:
> > drwxrwxrwt  20 root root  4096 2011-02-26 22:17 tmp
> >
> > If this is bug, please fix it.  If there is a catch, please post the
> solution and summarize on wiki page.
> >
> > One of the particular reason that i like FS is prompt support and
> enthusiasm from FS developers.
> > please do not lost your momentum.
> >
> >
> > thanks!
> >
> > On Sat, Feb 26, 2011 at 10:10 PM, Madovsky <infos at madovsky.org> wrote:
> > look at the the emailist archive,
> > I remember that one had the same problem
> > but don't remember the solution
> > ----- Original Message -----
> > From: envelopes envelopes
> > To: FreeSWITCH Users Help
> > Sent: Sunday, February 27, 2011 12:53 AM
> > Subject: Re: [Freeswitch-users] Fax processing not successful - result
> (41)TIFF/F file cannot be opened.
> >
> > no one had similar issue?
> >
> > On Fri, Feb 25, 2011 at 11:33 AM, envelopes envelopes <
> sunwood360 at gmail.com> wrote:
> > I don't see any permission issue here.
> >
> >
> > /tmp$ ls -l /tmp/a.tif
> > -rwxrwxrwx 1 root root 179378 2011-02-02 05:30 /tmp/a.tif
> >
> > receive side also writes to /tmp directroy.
> >
> > It is really weird.
> >
> >
> >
> >
> > On Fri, Feb 25, 2011 at 11:30 AM, Madovsky <infos at madovsky.org> wrote:
> > check permissions
> > ----- Original Message -----
> > From: envelopes envelopes
> > To: FreeSWITCH Users Help
> > Sent: Friday, February 25, 2011 2:23 PM
> > Subject: [Freeswitch-users] Fax processing not successful - result
> (41)TIFF/F file cannot be opened.
> >
> > I have the extension in default.xml;
> >
> > <extension name="fax_receive2">
> >      <condition field="destination_number" expression="^(9979)$">
> >        <action application="answer" />
> >        <action application="playback" data="silence_stream://2000"/>
> >        <action application="rxfax" data="/tmp/fax-${uuid}.rxfax.tiff"/>
> >        <action application="hangup"/>
> >      </condition>
> >    </extension>
> >
> > then in fs_cli : originate loopback/9979 &txfax(/tmp/a.tif)
> >
> > in fact, the file /tmp/a.tif does exist.
> >
> > /tmp$ ls -l /tmp/a.tif
> > -rwxrwxrwx 1 root root 179378 2011-02-02 05:30 /tmp/a.tif
> >
> >
> > 2011-02-25 11:17:39.444902 [DEBUG] mod_spandsp_fax.c:1108 Raw write codec
> activation Success L16
> > 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:323
> ==============================================================================
> > 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:335 Fax processing
> not successful - result (17) Received a DCN while waiting for a DIS.
> > 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:340 Remote station
> id:
> > 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:341 Local station
> id:  SpanDSP Fax Ident
> > 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:342 Pages
> transferred: 0
> > 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:344 Total fax pages:
>   0
> > 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:345 Image
> resolution:  0x0
> > 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:346 Transfer Rate:
>   14400
> > 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:348 ECM status
>   off
> > 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:349 remote country:
> > 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:350 remote vendor:
> > 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:351 remote model:
> > 2011-02-25 11:17:46.805168 [DEBUG] mod_spandsp_fax.c:353
> ==============================================================================
> > 2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:323
> ==============================================================================
> > 2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:335 Fax processing
> not successful - result (41) TIFF/F file cannot be opened.
> > 2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:340 Remote station
> id: SpanDSP Fax Ident
> > 2011-02-25 11:17:46.825158 [DEBUG] mod_spandsp_fax.c:341 Local station
> id:  SpanDSP Fax Ident
> >
> >
> > what is going 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
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
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From mel0torme at gmail.com  Mon Feb 28 00:32:56 2011
From: mel0torme at gmail.com (Tom C)
Date: Sun, 27 Feb 2011 13:32:56 -0800
Subject: [Freeswitch-users] Dockstar -- FreeSwitch crashing with latest git
Message-ID: <AANLkTi=wz5QAyFsnxdy3Ei9S9TAxq-efBSv6kc=QSGKj@mail.gmail.com>

I just did a "make current" on my dockstar (Debian Lenny) for the first time
in over a month, and now FreeSwitch crashes at startup.  It gets all the way
through the boot process, past the final banner, and then I get a
segmentation fault.  Output below.

2011-02-27 20:51:54.723908 [CONSOLE] switch_core.c:1768
FreeSWITCH Version 1.0.head (git-fd7bbc0 2011-02-26 11-45-19 +0100) Started.
Max Sessions[1000]
Session Rate[30]
SQL [Enabled]
freeswitch at dockstar> Segmentation fault (core dumped)

I can post the core, but in the meantime, how do I git the original 1.0.7
build?
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From simpot at gmail.com  Sun Feb 27 01:33:41 2011
From: simpot at gmail.com (Dmitry Saratsky)
Date: Sun, 27 Feb 2011 00:33:41 +0200
Subject: [Freeswitch-users] Can't Send emails from within FreeSwitch
Message-ID: <AANLkTimDOuzncGhCtXykrEu8mNq7fY7H-O-ssFNS-qDS@mail.gmail.com>

Hi All,

Unfortunately I can't send email from within freeswitch.
I have wrote some LUA script that should send emails.
The relevant part of the script:
-----------------
freeswitch.email("user at domain.com", "user at domain.com", "subject: Test
subject.\n", "Test body.")
-----------------

When script runs, it logs the following to console:
-----------------
2011-02-26 23:44:23.624644 [DEBUG] switch_utils.c:709 Emailed data to [
user at domain.com]
-----------------

In my switch.conf.xml I have the following mailer-app config:
-----------------
    <param name="mailer-app" value="sendmail"/>
    <param name="mailer-app-args" value="-t"/>
-----------------

I also have tried to use full path to sendmail app:
-----------------
    <param name="mailer-app" value="/usr/sbin/sendmail"/>
-----------------

I'm using:
CentOS release 5.5 (Final)
FreeSWITCH Version 1.0.head (git-cb6f1ed 2011-02-22 20-25-16 -0500)
postfix-2.3.3-2.1.el5_2 (as MTA)

The problem is, that no mail is received and nothing logged in
/var/log/maillog.
However I can successfully send emails from shell (from within freeswitch's
user) with sendmail -t command.

I have also wrote simple bash script that add some text to /tmp/12.txt on
run and pointed mailer-app (in switch.conf.xml) to this script full patch
file - nothing was added to the file.

Any Ideas?

Thanks,
Dmitry.
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From simpot at gmail.com  Sun Feb 27 18:00:03 2011
From: simpot at gmail.com (Dmitry Saratsky)
Date: Sun, 27 Feb 2011 17:00:03 +0200
Subject: [Freeswitch-users] How to run linux shell command from within LUA
	script in Freeswitch?
Message-ID: <AANLkTikKCMFzWBJ59jWZ5xa9DFDSQh4u2qC90uB0_o7c@mail.gmail.com>

Hi all,

I can't find how can i run linux shell command from within LUA script in
Freeswitch.

Anyone can help with this?

Thanks,
Dmitry.
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From cmrienzo at gmail.com  Mon Feb 28 01:51:57 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Sun, 27 Feb 2011 17:51:57 -0500
Subject: [Freeswitch-users] How to run linux shell command from within
 LUA script in Freeswitch?
In-Reply-To: <AANLkTikKCMFzWBJ59jWZ5xa9DFDSQh4u2qC90uB0_o7c@mail.gmail.com>
References: <AANLkTikKCMFzWBJ59jWZ5xa9DFDSQh4u2qC90uB0_o7c@mail.gmail.com>
Message-ID: <AANLkTi=_pBaT3VwChh7Qk=99RyC_UrVJVyyLKNtCYpTT@mail.gmail.com>

session:execute("system", "your shell command");


On Sun, Feb 27, 2011 at 10:00 AM, Dmitry Saratsky <simpot at gmail.com> wrote:

> Hi all,
>
> I can't find how can i run linux shell command from within LUA script in
> Freeswitch.
>
> Anyone can help with this?
>
> 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 ayhkor at gmail.com  Mon Feb 28 03:53:59 2011
From: ayhkor at gmail.com (deniro)
Date: Sun, 27 Feb 2011 19:53:59 -0500
Subject: [Freeswitch-users] fs_cli socket connection error
Message-ID: <AANLkTikCoT9dzLNoD=6TzS0jiMw4afNX94TypsLarBbz@mail.gmail.com>

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--
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From Nabble at slickdeals.endjunk.com  Mon Feb 28 03:59:13 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 27 Feb 2011 16:59:13 -0800 (PST)
Subject: [Freeswitch-users] Dockstar -- FreeSwitch crashing with latest
	git
In-Reply-To: <AANLkTi=wz5QAyFsnxdy3Ei9S9TAxq-efBSv6kc=QSGKj@mail.gmail.com>
References: <AANLkTi=wz5QAyFsnxdy3Ei9S9TAxq-efBSv6kc=QSGKj@mail.gmail.com>
Message-ID: <1298854753392-6071631.post@n2.nabble.com>

Have a look 
http://freeswitch-users.2379917.n2.nabble.com/freeswitch-nc-nosql-tp6066718p6066718.html
here .

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dockstar-FreeSwitch-crashing-with-latest-git-tp6071223p6071631.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From fs-list at communicatefreely.net  Mon Feb 28 04:42:55 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Sun, 27 Feb 2011 20:42:55 -0500
Subject: [Freeswitch-users] Cordless phones
In-Reply-To: <AANLkTikFAO9_bkpmXvk9AKBRTT34_811r_jSnfA7SO47@mail.gmail.com>
References: <AANLkTikFAO9_bkpmXvk9AKBRTT34_811r_jSnfA7SO47@mail.gmail.com>
Message-ID: <4D6AFD9F.3030500@communicatefreely.net>

Aastra MBU 400 - Pretty good build quality, LiPo batteries, can handle 
up to 8 handsets and 8 SIP accounts.  You assign the registrations to 
handsets in a sort of matrix, so they can all share one, or each have 
their own.  Maximum 3 channels at a time though.

Snom M3 - Haven't tried it, but it has a lot of common components with 
the Aastra and a similar feature set.  I think this also has a 3 channel 
limit.

Gigaset S675 - About half the price of the other two, but with a 
significant reduction in what you get.  Only one SIP account across all 
phones, and I'm not sure that it can do 5 handsets.  I think it can do 
at least 3.

Aastra DECT 142 handsets and RFP 32 base stations - This is a campus 
DECT mobility system.  It's a lot more expensive, but a lot more 
scaleable.  Each RFP32 base station can have up to 8 simultaneous 
channels.  You can have up to 512 handsets per system, with up to 256 
base stations set up in a cellular-like system.  I have personally 
tested this system with Freeswitch and it works great.  Very easy to set 
up.  The handsets are also very durable.

-Tim

Jonas Gauffin wrote:
> Hi,
>
> I'm looking for IP-dects where a base station can handle 5 phones 
> (which all should be able to talk at the same time).
> All suggestions are welcome.
>
> Regards,
>   Jonas
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 fs-list at communicatefreely.net  Mon Feb 28 04:49:57 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Sun, 27 Feb 2011 20:49:57 -0500
Subject: [Freeswitch-users] conference dialing and montly billing
In-Reply-To: <AANLkTinY_-7zPySJMQ915hKY0RTHZtycXoszw3K3LE0u@mail.gmail.com>
References: <AANLkTinY_-7zPySJMQ915hKY0RTHZtycXoszw3K3LE0u@mail.gmail.com>
Message-ID: <4D6AFF45.50108@communicatefreely.net>

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
>   



From Nabble at slickdeals.endjunk.com  Mon Feb 28 06:47:09 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 27 Feb 2011 19:47:09 -0800 (PST)
Subject: [Freeswitch-users] freeswitch -nc -nosql
In-Reply-To: <1298820079407-6070409.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>
Message-ID: <1298864829635-6071920.post@n2.nabble.com>

In addition to the above, the following is the excerpt from the last part of
strace freeswitch -waste dump:
access("/usr/share/freeswitch/db/core.db-journal", F_OK) = -1 ENOENT (No
such file or directory)
fstat64(5, {st_mode=S_IFREG|0644, st_size=160768, ...}) = 0
_llseek(5, 0, [0], SEEK_SET)            = 0
read(5, "SQLite format 3\0\4\0\1\1\0@  \0\0\23T\0\0\0\0"..., 1024) = 1024
fcntl64(5, F_SETLK64, {type=F_WRLCK, whence=SEEK_SET, start=1073741825,
len=1}, 0xbee27b70) = 0
open("/usr/share/freeswitch/db/core.db-journal",
O_RDWR|O_CREAT|O_EXCL|O_LARGEFILE, 0644) = 47
fstat64(47, {st_mode=S_IFREG|0644, st_size=0, ...}) = 0
open("/usr/share/freeswitch/db", O_RDONLY|O_LARGEFILE) = 48
_llseek(47, 0, [0], SEEK_SET)           = 0
write(47, "\331\325\5\371
\241c\327\377\377\377\377\254\0\7\242\0\0\0\235\0\0\2\0", 24) = 24
_llseek(47, 511, [511], SEEK_SET)       = 0
+++ killed by SIGSEGV +++
Segmentation fault

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-nc-nosql-tp6066718p6071920.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From sunwood360 at gmail.com  Mon Feb 28 09:30:06 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Sun, 27 Feb 2011 22:30:06 -0800
Subject: [Freeswitch-users] mod_dingaling and google voice
Message-ID: <AANLkTin_dFj9ygx3TAmExeN9-pDFNCuZPGZ_Uc2d7o1W@mail.gmail.com>

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!
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From hareem.haque at gmail.com  Mon Feb 28 09:32:15 2011
From: hareem.haque at gmail.com (Hareem Haque)
Date: Mon, 28 Feb 2011 01:32:15 -0500
Subject: [Freeswitch-users] How to config sip profiles and add them into the
	freeswitch system
Message-ID: <AANLkTimbRgSEegN4O5btU_2RsafufbpRLt97ph5s-cfj@mail.gmail.com>

I understand that sip profiles are managed via xml files.. i can make them..
but when i use sofia profile start command.. it does not pick up the .xml
profile


so how can i setup profiles and add them into the system. Your help is
greatly appreciated


Best Regards
Hareem. Haque
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From sunwood360 at gmail.com  Mon Feb 28 09:54:27 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Sun, 27 Feb 2011 22:54:27 -0800
Subject: [Freeswitch-users] mod_dingaling and google voice
In-Reply-To: <AANLkTin_dFj9ygx3TAmExeN9-pDFNCuZPGZ_Uc2d7o1W@mail.gmail.com>
References: <AANLkTin_dFj9ygx3TAmExeN9-pDFNCuZPGZ_Uc2d7o1W@mail.gmail.com>
Message-ID: <AANLkTimUr3=w_fj_k643ChW93Q_YTm63R_qFyBO9EoQV@mail.gmail.com>

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!
>
>
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From ross at ossiantelecom.co.uk  Mon Feb 28 11:06:16 2011
From: ross at ossiantelecom.co.uk (Ross McKillop)
Date: Mon, 28 Feb 2011 08:06:16 +0000
Subject: [Freeswitch-users] Cordless phones
In-Reply-To: <4D6AFD9F.3030500@communicatefreely.net>
References: <AANLkTikFAO9_bkpmXvk9AKBRTT34_811r_jSnfA7SO47@mail.gmail.com>
	<4D6AFD9F.3030500@communicatefreely.net>
Message-ID: <AC038DD3-EF07-4FDF-B460-FA8A12745527@ossiantelecom.co.uk>

To add a couple more into the mix;

Siemens S450IP - Supports up to 6 accounts and offers a similar matrix 
assignment to the Aastra MBU unit but only for up to 6 SIP accounts with
a maximum of 3 channels but each handset can be registered to more than
one base station.

The KIRK system, now owned by Polycom I believe, is a highly scalable 
DECT system much like the RFP32 - pricey but very scalable and the KIRK
handsets are very reliable also.

Ross

On 28 Feb 2011, at 01:42, Tim St. Pierre wrote:

> Aastra MBU 400 - Pretty good build quality, LiPo batteries, can handle 
> up to 8 handsets and 8 SIP accounts.  You assign the registrations to 
> handsets in a sort of matrix, so they can all share one, or each have 
> their own.  Maximum 3 channels at a time though.
> 
> Snom M3 - Haven't tried it, but it has a lot of common components with 
> the Aastra and a similar feature set.  I think this also has a 3 channel 
> limit.
> 
> Gigaset S675 - About half the price of the other two, but with a 
> significant reduction in what you get.  Only one SIP account across all 
> phones, and I'm not sure that it can do 5 handsets.  I think it can do 
> at least 3.
> 
> Aastra DECT 142 handsets and RFP 32 base stations - This is a campus 
> DECT mobility system.  It's a lot more expensive, but a lot more 
> scaleable.  Each RFP32 base station can have up to 8 simultaneous 
> channels.  You can have up to 512 handsets per system, with up to 256 
> base stations set up in a cellular-like system.  I have personally 
> tested this system with Freeswitch and it works great.  Very easy to set 
> up.  The handsets are also very durable.
> 
> -Tim
> 
> Jonas Gauffin wrote:
>> Hi,
>> 
>> I'm looking for IP-dects where a base station can handle 5 phones 
>> (which all should be able to talk at the same time).
>> All suggestions are welcome.
>> 
>> Regards,
>>  Jonas



From erik.dekkers at wvds.nl  Mon Feb 28 11:21:57 2011
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Mon, 28 Feb 2011 09:21:57 +0100
Subject: [Freeswitch-users] Cordless phones
In-Reply-To: <AANLkTikFAO9_bkpmXvk9AKBRTT34_811r_jSnfA7SO47@mail.gmail.com>
References: <AANLkTikFAO9_bkpmXvk9AKBRTT34_811r_jSnfA7SO47@mail.gmail.com>
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE0787301942@FS1.wvds.local>

I would go for Aastra SIP-DECT. http://www.aastra.com/rfp-32-ip_rfp-34-ip.htm
It's capable for 8 simultaneous voice channels per Basestation. If combined with multiple basestations (up to 256) a maximum of 512 handsets are possible.

It's cheap, powerfull and extendable.

My 2 cents

Erik
Btw. I use those things myself and can tell they're working perfect with freeswitch.

Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Jonas Gauffin
Verzonden: zondag 27 februari 2011 18:40
Aan: FreeSWITCH Users Help
Onderwerp: [Freeswitch-users] Cordless phones

Hi,

I'm looking for IP-dects where a base station can handle 5 phones (which all should be able to talk at the same time).
All suggestions are welcome.

Regards,
  Jonas
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From steveayre at gmail.com  Mon Feb 28 11:35:02 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 28 Feb 2011 08:35:02 +0000
Subject: [Freeswitch-users] fs_cli socket connection error
In-Reply-To: <AANLkTikCoT9dzLNoD=6TzS0jiMw4afNX94TypsLarBbz@mail.gmail.com>
References: <AANLkTikCoT9dzLNoD=6TzS0jiMw4afNX94TypsLarBbz@mail.gmail.com>
Message-ID: <AANLkTi=sENgBB9Cdr-EkLJD8XfmH3A8GSfPnvDHO7qrw@mail.gmail.com>

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> 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
> 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 Feb 28 11:36:09 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 28 Feb 2011 08:36:09 +0000
Subject: [Freeswitch-users] How to config sip profiles and add them into
 the freeswitch system
In-Reply-To: <AANLkTimbRgSEegN4O5btU_2RsafufbpRLt97ph5s-cfj@mail.gmail.com>
References: <AANLkTimbRgSEegN4O5btU_2RsafufbpRLt97ph5s-cfj@mail.gmail.com>
Message-ID: <AANLkTimrruqsB8JebsYyWBDJFFNnYZEY2LRn43-ubz_X@mail.gmail.com>

Try reloadxml first:

> reloadxml
> sofia profile xxx start

-Steve


On 28 February 2011 06:32, Hareem Haque <hareem.haque at gmail.com> wrote:
> I understand that sip profiles are managed via xml files.. i can make them..
> but when i use sofia profile start command.. it does not pick up the .xml
> profile
>
> so how can i setup profiles and add them into the system. Your help is
> greatly appreciated
>
>
> Best Regards
> Hareem. Haque
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 patrick.plattes at niemann-frey.info  Mon Feb 28 15:57:05 2011
From: patrick.plattes at niemann-frey.info (Patrick Plattes)
Date: Mon, 28 Feb 2011 13:57:05 +0100
Subject: [Freeswitch-users] Cordless phones
In-Reply-To: <CDF5E00590C9BA4AB96BE7CDD28C86AE0787301942@FS1.wvds.local>
References: <AANLkTikFAO9_bkpmXvk9AKBRTT34_811r_jSnfA7SO47@mail.gmail.com>
	<CDF5E00590C9BA4AB96BE7CDD28C86AE0787301942@FS1.wvds.local>
Message-ID: <AANLkTikxKy+Mr5DShgUQK+ZMu6xBz2cOfMS1xUcjpVV0@mail.gmail.com>

We are using 10 Aastra RFP L32 IP. Not really cheap, but we never had
any serious problem. Our DECT mobile devices are Aastra 6x0d and some
old Siemens Gigasets.

It's a very powerful and extendable solution - IMHO it's not cheap.

Bye :)


From mauritz.lovgren at hotmail.com  Mon Feb 28 17:26:44 2011
From: mauritz.lovgren at hotmail.com (=?utf-8?Q?Mauritz_L=C3=B8vgren?=)
Date: Mon, 28 Feb 2011 15:26:44 +0100
Subject: [Freeswitch-users] mod_event_socket message pizza received from BSD
	machines
Message-ID: <SNT107-DS1630BE652914A48CC2BA01E6DE0@phx.gbl>

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

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From freeswitch at peely.com  Mon Feb 28 18:34:24 2011
From: freeswitch at peely.com (peely)
Date: Mon, 28 Feb 2011 07:34:24 -0800 (PST)
Subject: [Freeswitch-users] Polycom G722 negotiation
In-Reply-To: <0ABF546B-4000-4936-AD22-ED6B956277D5@5ninesolutions.com>
References: <AE216221-5892-4337-8985-1889410A866F@5ninesolutions.com>
	<2867CE13-6EED-4FAF-9DF1-49451D164721@freeswitch.org>
	<0ABF546B-4000-4936-AD22-ED6B956277D5@5ninesolutions.com>
Message-ID: <1298907264976-6073705.post@n2.nabble.com>

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.


From peter.olsson at visionutveckling.se  Mon Feb 28 18:36:46 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 28 Feb 2011 16:36:46 +0100
Subject: [Freeswitch-users] mod_event_socket message pizza received from
 BSD	machines
In-Reply-To: <SNT107-DS1630BE652914A48CC2BA01E6DE0@phx.gbl>
References: <SNT107-DS1630BE652914A48CC2BA01E6DE0@phx.gbl>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58ADB727D2@cooper>

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!


From spencer at 5ninesolutions.com  Mon Feb 28 18:55:27 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Mon, 28 Feb 2011 15:55:27 +0000
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: <1F4FD420-3CE9-4601-9D44-80C8BEEF9707@5ninesolutions.com>

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
>



From mthakershi at gmail.com  Mon Feb 28 19:24:55 2011
From: mthakershi at gmail.com (Malay Thakershi)
Date: Mon, 28 Feb 2011 10:24:55 -0600
Subject: [Freeswitch-users] disabling ptime warning message
Message-ID: <AANLkTim97P5=MJgKq3ncDmbTiXkP63CBoALRD6o+b-Uw@mail.gmail.com>

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
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From anthony.minessale at gmail.com  Mon Feb 28 20:33:48 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Feb 2011 11:33:48 -0600
Subject: [Freeswitch-users] freeswitch -nc -nosql
In-Reply-To: <1298864829635-6071920.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>
Message-ID: <AANLkTimxG==dquT_oeFSiPnxSN6g5mncQ5Gf1GNg_u6U@mail.gmail.com>

This means you need the sql enabled.

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.


On Sun, Feb 27, 2011 at 9:47 PM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
> In addition to the above, the following is the excerpt from the last part of
> strace freeswitch -waste dump:
> access("/usr/share/freeswitch/db/core.db-journal", F_OK) = -1 ENOENT (No
> such file or directory)
> fstat64(5, {st_mode=S_IFREG|0644, st_size=160768, ...}) = 0
> _llseek(5, 0, [0], SEEK_SET) ? ? ? ? ? ?= 0
> read(5, "SQLite format 3\0\4\0\1\1\0@ ?\0\0\23T\0\0\0\0"..., 1024) = 1024
> fcntl64(5, F_SETLK64, {type=F_WRLCK, whence=SEEK_SET, start=1073741825,
> len=1}, 0xbee27b70) = 0
> open("/usr/share/freeswitch/db/core.db-journal",
> O_RDWR|O_CREAT|O_EXCL|O_LARGEFILE, 0644) = 47
> fstat64(47, {st_mode=S_IFREG|0644, st_size=0, ...}) = 0
> open("/usr/share/freeswitch/db", O_RDONLY|O_LARGEFILE) = 48
> _llseek(47, 0, [0], SEEK_SET) ? ? ? ? ? = 0
> write(47, "\331\325\5\371
> \241c\327\377\377\377\377\254\0\7\242\0\0\0\235\0\0\2\0", 24) = 24
> _llseek(47, 511, [511], SEEK_SET) ? ? ? = 0
> +++ killed by SIGSEGV +++
> Segmentation fault
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-nc-nosql-tp6066718p6071920.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  Mon Feb 28 21:05:32 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Feb 2011 12:05:32 -0600
Subject: [Freeswitch-users] disabling ptime warning message
In-Reply-To: <AANLkTim97P5=MJgKq3ncDmbTiXkP63CBoALRD6o+b-Uw@mail.gmail.com>
References: <AANLkTim97P5=MJgKq3ncDmbTiXkP63CBoALRD6o+b-Uw@mail.gmail.com>
Message-ID: <AANLkTikw3TLW8bo39WRX6wN44vhcGSDBBTS_QffqBTwc@mail.gmail.com>

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


From anthony.minessale at gmail.com  Mon Feb 28 21:08:05 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Feb 2011 12:08:05 -0600
Subject: [Freeswitch-users] mod_event_socket message pizza received from
 BSD machines
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58ADB727D2@cooper>
References: <SNT107-DS1630BE652914A48CC2BA01E6DE0@phx.gbl>
	<549CFEF87AEDE841A38E9D15EAB4C04C58ADB727D2@cooper>
Message-ID: <AANLkTim81z7pa50C8VzNToTY+9=AeK8umpyBr3eeKZoF@mail.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


From anthony.minessale at gmail.com  Mon Feb 28 21:45:25 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Feb 2011 12:45:25 -0600
Subject: [Freeswitch-users] t38_gateway self - no re-Invite?
In-Reply-To: <1B19ABD72889C245AE8EEE08AC24103A28C4231F4F@exmachina.office.kapper.net>
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>
Message-ID: <AANLkTinsvti9Q9K-fLzB8o1Xd1qther1ABRQwHeV7FpR@mail.gmail.com>

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


From mthakershi at gmail.com  Mon Feb 28 21:47:00 2011
From: mthakershi at gmail.com (Malay Thakershi)
Date: Mon, 28 Feb 2011 12:47:00 -0600
Subject: [Freeswitch-users] disabling ptime warning message
In-Reply-To: <AANLkTikw3TLW8bo39WRX6wN44vhcGSDBBTS_QffqBTwc@mail.gmail.com>
References: <AANLkTim97P5=MJgKq3ncDmbTiXkP63CBoALRD6o+b-Uw@mail.gmail.com>
	<AANLkTikw3TLW8bo39WRX6wN44vhcGSDBBTS_QffqBTwc@mail.gmail.com>
Message-ID: <AANLkTi=FGWs2bs3EFvaMTTyG0=0inwe5z2UfByctX-g1@mail.gmail.com>

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
>
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From anthony.minessale at gmail.com  Mon Feb 28 21:50:49 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Feb 2011 12:50:49 -0600
Subject: [Freeswitch-users] disabling ptime warning message
In-Reply-To: <AANLkTi=FGWs2bs3EFvaMTTyG0=0inwe5z2UfByctX-g1@mail.gmail.com>
References: <AANLkTim97P5=MJgKq3ncDmbTiXkP63CBoALRD6o+b-Uw@mail.gmail.com>
	<AANLkTikw3TLW8bo39WRX6wN44vhcGSDBBTS_QffqBTwc@mail.gmail.com>
	<AANLkTi=FGWs2bs3EFvaMTTyG0=0inwe5z2UfByctX-g1@mail.gmail.com>
Message-ID: <AANLkTindiPgTH65=0MwBgdviyWByQzmu1=01MYUy_fXy@mail.gmail.com>

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


From anthony.minessale at gmail.com  Mon Feb 28 22:15:09 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Feb 2011 13:15:09 -0600
Subject: [Freeswitch-users] Polycom G722 negotiation
In-Reply-To: <1F4FD420-3CE9-4601-9D44-80C8BEEF9707@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>
Message-ID: <AANLkTikFCLzNvqpHEO4qtCyFDQOTZ0CG5pOw+SJD=2Yz@mail.gmail.com>

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


From spencer at 5ninesolutions.com  Mon Feb 28 22:22:49 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Mon, 28 Feb 2011 19:22:49 +0000
Subject: [Freeswitch-users] Polycom G722 negotiation
In-Reply-To: <AANLkTikFCLzNvqpHEO4qtCyFDQOTZ0CG5pOw+SJD=2Yz@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>
Message-ID: <5AB8CE7C-5283-4585-8DF8-856A3822F3BD@5ninesolutions.com>

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
>



From mthakershi at gmail.com  Mon Feb 28 22:53:57 2011
From: mthakershi at gmail.com (Malay Thakershi)
Date: Mon, 28 Feb 2011 13:53:57 -0600
Subject: [Freeswitch-users] disabling ptime warning message
In-Reply-To: <AANLkTindiPgTH65=0MwBgdviyWByQzmu1=01MYUy_fXy@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>
Message-ID: <AANLkTi=cFCJssphqmji_8LTXTg3ZycGfswMM77hmkie9@mail.gmail.com>

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
>
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From anthony.minessale at gmail.com  Mon Feb 28 23:04:00 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 28 Feb 2011 14:04:00 -0600
Subject: [Freeswitch-users] disabling ptime warning message
In-Reply-To: <AANLkTi=cFCJssphqmji_8LTXTg3ZycGfswMM77hmkie9@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>
Message-ID: <AANLkTimTKLJQ+cE9r5t2fRR+riDo-DPi+7FQLJecoc9M@mail.gmail.com>

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


From ayhkor at gmail.com  Mon Feb 28 23:42:49 2011
From: ayhkor at gmail.com (deniro)
Date: Mon, 28 Feb 2011 15:42:49 -0500
Subject: [Freeswitch-users] fs_cli socket connection error
In-Reply-To: <AANLkTi=sENgBB9Cdr-EkLJD8XfmH3A8GSfPnvDHO7qrw@mail.gmail.com>
References: <AANLkTikCoT9dzLNoD=6TzS0jiMw4afNX94TypsLarBbz@mail.gmail.com>
	<AANLkTi=sENgBB9Cdr-EkLJD8XfmH3A8GSfPnvDHO7qrw@mail.gmail.com>
Message-ID: <AANLkTin5k=2Z-=QVVzqt91172wv88P7OFkqRqTY9VDcS@mail.gmail.com>

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> 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> 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
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Feb 28 23:59:29 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 28 Feb 2011 21:59:29 +0100
Subject: [Freeswitch-users] fs_cli socket connection error
In-Reply-To: <AANLkTin5k=2Z-=QVVzqt91172wv88P7OFkqRqTY9VDcS@mail.gmail.com>
References: <AANLkTikCoT9dzLNoD=6TzS0jiMw4afNX94TypsLarBbz@mail.gmail.com>
	<AANLkTi=sENgBB9Cdr-EkLJD8XfmH3A8GSfPnvDHO7qrw@mail.gmail.com>,
	<AANLkTin5k=2Z-=QVVzqt91172wv88P7OFkqRqTY9VDcS@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58ADB727D7@cooper>

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!


From megahohol at gmail.com  Mon Feb 28 11:51:07 2011
From: megahohol at gmail.com (Grygoriy Dobrovolskyy)
Date: Mon, 28 Feb 2011 09:51:07 +0100
Subject: [Freeswitch-users] Cordless phones
In-Reply-To: <CDF5E00590C9BA4AB96BE7CDD28C86AE0787301942@FS1.wvds.local>
References: <AANLkTikFAO9_bkpmXvk9AKBRTT34_811r_jSnfA7SO47@mail.gmail.com>
	<CDF5E00590C9BA4AB96BE7CDD28C86AE0787301942@FS1.wvds.local>
Message-ID: <AANLkTiksNvXo+Y=XYDMFuK4PMPSOMgOy_n-is1sRF5g4@mail.gmail.com>

2011/2/28 Erik Dekkers <erik.dekkers at wvds.nl>:
> I would go for Aastra SIP-DECT.
> http://www.aastra.com/rfp-32-ip_rfp-34-ip.htm
>
> It?s capable for 8 simultaneous voice channels per Basestation. If combined
> with multiple basestations (up to 256) a maximum of 512 handsets are
> possible.
>
>
>
> It?s cheap, powerfull and extendable.
>
>
>
> My 2 cents
>
>
>
> Erik
>
> Btw. I use those things myself and can tell they?re working perfect with
> freeswitch.
>
>
>
> Van: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Jonas Gauffin
> Verzonden: zondag 27 februari 2011 18:40
> Aan: FreeSWITCH Users Help
> Onderwerp: [Freeswitch-users] Cordless phones
>
>
>
> Hi,
>
>
>
> I'm looking for IP-dects where a base station can handle 5 phones (which all
> should be able to talk at the same time).
>
> All suggestions are welcome.
>
>
>
> Regards,
>
> ??Jonas
>

Hello, Siemens and M9 are buggy, i do not recommend them.
Go aastra.


From whglee at gmail.com  Mon Feb 28 21:04:48 2011
From: whglee at gmail.com (George Lee)
Date: Mon, 28 Feb 2011 13:04:48 -0500
Subject: [Freeswitch-users] FreeSwithc not handling no SDP INVITE
 properly for video call
In-Reply-To: <AANLkTinxTuAv0ybtguJtZh3Xkc11=EHJo6CQwdRmRphV@mail.gmail.com>
References: <AANLkTinoomTncUKH=xKLSfAaWKeTupOMv_aZuN_CVVF+@mail.gmail.com>
	<AANLkTinxTuAv0ybtguJtZh3Xkc11=EHJo6CQwdRmRphV@mail.gmail.com>
Message-ID: <AANLkTimvG+_vo5O0jH1nnHMaHy-a7UqwFUS2+Dsgs+kK@mail.gmail.com>

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
>


From whglee at gmail.com  Mon Feb 28 21:14:38 2011
From: whglee at gmail.com (George Lee)
Date: Mon, 28 Feb 2011 13:14:38 -0500
Subject: [Freeswitch-users] FreeSwitch Installation Error
Message-ID: <AANLkTinfmLJCs+YyERMmGLWDs4u7uOZruTRe-d+-HdWQ@mail.gmail.com>

Hi,

I changed the installation directory to /opt/freeswitch
> ./configure --prefix=/opt/freeswitch

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


From jesse at cronomagic.com  Mon Feb 28 23:20:30 2011
From: jesse at cronomagic.com (Jesse Cloutier)
Date: Mon, 28 Feb 2011 15:20:30 -0500
Subject: [Freeswitch-users] freetdm with  a400p segfaults
Message-ID: <4D6C038E.7050303@cronomagic.com>

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

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