[Freeswitch-users] FreeSWITCH-users Digest, Vol 67, Issue 51

Bharat Lalcheta bharatlalcheta at gmail.com
Fri Jan 6 18:40:01 MSK 2012 

can you please explain in details what you want to tell ?




> ---------- Forwarded message ----------
> From: curriegrad2004 <curriegrad2004 at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Fri, 6 Jan 2012 07:11:52 -0800
> Subject: Re: [Freeswitch-users] Codec Preferance
>
> I highly would recommend that you change the name of those codecs to
> something else because you might be making matters worse later down the
> road
> On 2012-01-06 5:21 AM, "Bharat Lalcheta" <bharatlalcheta at gmail.com> wrote:
>
>> Hiii,
>>
>> I am new to freeswitch. Prior to freeswitch i was using asterisk.
>>
>> I have 200 extensions working in my office and want to move all to
>> freeswitch from asterisk.
>>
>> In asterisk, i can give codec selection and preferance in sip.conf to all
>> extensions. In the same way i created 200 extensions under internal profile
>> in freeswitch.
>>
>> Follwing is one example....
>> -----------------------------------------------------------------
>> <include>
>>   <user id="590">
>>     <params>
>>       <param name="password" value="590"/>
>>       <param name="vm-password" value=""/>
>>       <param name="vm-enabled" value="true"/>
>>       <param name="inbound_codec_prefs" value="PCMA,H264"/>
>>       <param name="outbound_codec_prefs" value="PCMA,H264"/>
>>     </params>
>>     <variables>
>>       <variable name="accountcode" value=""/>
>>       <variable name="user_context" value="default"/>
>>       <variable name="max-calls" value="2"/>
>>       <variable name="bypass_media_after_bridge" value="no"/>
>>     </variables>
>>   </user>
>> </include>
>> ----------------------------------------------------------
>>
>> Now when ever i called to 590 freeswitch sends all codecs to 590 sip
>> phone other than defined in 590.xml. It is seding codes which is mentioned
>> in my conf/sip_profiles/internal.xml and codec negotiation done on whatever
>> codec my sip phone having.
>>
>> I want to use different codecs for different extensions.
>>
>> Is it common behaviour of Freeswitch ? Should i override codec prerfrance
>> in my extension list from my internal profile or not ?
>>
>> If no, then is it that i have to create 200 profiles in freeswitch to
>> solve this problem ?
>>
>> Please guide me and provide solution for the same
>>
>>
>> Thanks in advance
>>
>> Bharat Lalcheta
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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: Peter Olsson <peter.olsson at visionutveckling.se>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Fri, 6 Jan 2012 15:21:00 +0000
> Subject: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR
> Are you still using ignore_early_media=true - this must be set for this to
> work correctly.
>
> You will see a EXECUTE log line when FS executes the application, with
> ignore_early_media enabled it shouldn't execute until the call has been
> answered. I just tried it myself, and it works as expected.
>
> Example "originate {ignore_early_media=true}sofia/internal/number at host&park()"
>
> Park application is only executed after the call was answered.
>
> /Peter
>
> ________________________________________
> Från: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Oliver Schenk [
> olimonkey at gmail.com]
> Skickat: den 6 januari 2012 12:04
> Till: FreeSWITCH Users Help
> Ämne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR
>
> Because I'm using an FXO card with voice, I added something to my
> CISCO conf. Many others had the same thing:
>
>
> voice-port 0/3/0
>   ...
>   supervisory disconnect dualtone mid-call
>   supervisory answer dualtone    <---- ADDED THIS ONE
>   ...
>
>
>
> Once I added this, the FS output now just showed the following while
> the phone was ringing:
>
> 2012-01-05 16:19:31.644440 [NOTICE] switch_channel.c:816 New Channel
> sofia/internal/109212xxxx at 192.168.x.x
> [69e3f13d-1e2a-409e-97a4-b5526ea6e4ec]
> 2012-01-05 16:19:35.124882 [INFO] sofia.c:740
> sofia/internal/109212xxxx at 192.168.x.x Update Callee ID to "Outbound
> Call" <109212xxxx>
> 2012-01-05 16:19:35.126883 [NOTICE] sofia_glue.c:3793 Pre-Answer
> sofia/internal/109212xxxx at 192.168.x.x!
>
>
> Where as previous it would show the above and also show the following:
>
> 2012-01-05 16:19:35.127883 [INFO] switch_channel.c:2456
> sofia/internal/109212xxxx at 192.168.x.x Flipping CID from ""
> <0000000000> to "Outbound Call" <109212xxxx>
> 2012-01-05 16:19:35.137384 [INFO] sofia.c:740
> sofia/internal/109212xxxx at 192.168.x.x Update Callee ID to "109212xxxx"
> <1092122856>
> 2012-01-05 16:19:35.138384 [NOTICE] sofia.c:5296 Channel
> [sofia/internal/109212xxxx at 192.168.x.x] has been answered
>
>
>
> BUT, the IVR still started playing even before I pick up the phone.
> Hmmmm.....so why is FS still starting the managed application when the
> call has not been answered yet. Are we all sure that the managed
> application should not be executed until the call "has been answered"
> shows up in the log file?
>
>
> Will have to keep testing on monday as I don't have access to the
> CISCO from where i am now. I'll have to see whether the CISCO changes
> had any impact on the times at which the SIP messages are sent back
> and forth. Especially the 200 OK message.
>
>
> Thanks again for help, maybe getting somewhere now......
>
> Oliver
>
>
>
>
> On Fri, Jan 6, 2012 at 4:20 PM, Peter Olsson
> <peter.olsson at visionutveckling.se> wrote:
> > If it sends 200 OK right after 183, this IS the problem.
> >
> > 200 OK means that the call was answered, it should not be sent until the
> call was actually picked up in the remote end. When 200 OK arrives to FS it
> will execute your app, and you will start playing the files.
> >
> > Seems to me there is something broken in the Cisco.
> >
> > /Peter
> >
> > ________________________________________
> > Från: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Oliver Schenk [
> olimonkey at gmail.com]
> > Skickat: den 6 januari 2012 06:55
> > Till: FreeSWITCH Users Help
> > Ämne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR
> >
> > I've tried looking at disable-early-media configuration command, but
> > that didn't work and I doubt that has anything to do with the CISCO
> > sending a 200 OK right after a 183 SESSION PROGRESS.
> >
> >
> >
> >
> > On Fri, Jan 6, 2012 at 9:20 AM, Brian West <brian at freeswitch.org> wrote:
> >> Thats what the 183 is.. 180 vs 183 are kinda sketchy in some devices..
> 180
> >> is usually RINGING (generate ringback locally) while a 183 has media...
> aka
> >> early media and usually provides ringback inband.
> >>
> >> /b
> >>
> >> On Jan 5, 2012, at 7:13 PM, Oliver Schenk wrote:
> >>
> >> Shouldn't there be a  180 RINGING  somewhere in there?
> >>
> >>
> >>
> >>
> >> On Fri, Jan 6, 2012 at 8:25 AM, Oliver Schenk <olimonkey at gmail.com>
> wrote:
> >>
> >> I just noticed something else, if I don't pick up the phone at all.
> >>
> >> The IVR just keeps playing until the menu timeout kicks in.
> >>
> >>
> >> So here is a CISCO SIP log:
> >>
> >> http://pastebin.com/Y9sYkuxi
> >>
> >>
> >> The FS server is 192.168.x.50 and the CISCO is 192.168.x.1.
> >>
> >> I hope the CISCO log is readable, it's a bit long because I just did
> >>
> >> "debug ccsip all".
> >>
> >>
> >>
> >>
> >> In this test I didn't bother picking up the phone at all, but I can
> >>
> >> see that FS answered anyway and the IVR kept playing until it timed
> >>
> >> out.
> >>
> >> I'm not an expert, but here is what I picked out of it:
> >>
> >>
> >> At 00:08:10 we get a
> >>
> >> Received: "INVITE sip:109212xxxx at 192.168.x.1 SIP/2.0"
> >>
> >>
> >> the further down at the same timestamp we get
> >>
> >> Sent: "SIP/2.0 100 Trying"
> >>
> >>
> >> At 00:08:13 we get a
> >>
> >> Sent: "SIP/2.0 183 Session Progress"
> >>
> >>
> >> At 00:18:13 we get a
> >>
> >> Sent: "SIP/2.0 200 OK"
> >>
> >>
> >> Then at the same timestamp we get:
> >>
> >> Received: "ACK sip:109212xxxx at 192.168.x.1:5060 SIP/2.0"
> >>
> >>
> >>
> >>
> >> Once the IVR times out at 00:09:16 we get
> >>
> >> Received: "BYE sip:109212xxxx at 192.168.x.1:5060 SIP/2.0"
> >>
> >>
> >> And then the reply right after
> >>
> >> Sent: "SIP/2.0 200 OK"
> >>
> >>
> >>
> >>
> >> So I think you were right, the CISCO is sending back an "OK" 3 seconds
> >>
> >> after the "INVITE" is received.
> >>
> >>
> >>
> >>
> >> The part that is beyond my field of expertise so far is WHY?
> >>
> >>
> >>
> >>
> >> Thanks,
> >>
> >>
> >>
> >> Oliver
> >>
> >>
> >>
> >>
> >> On Fri, Jan 6, 2012 at 8:04 AM, Oliver Schenk <olimonkey at gmail.com>
> wrote:
> >>
> >> By the way:
> >>
> >>
> >> I tried {ignore_early_media=true} as well, but as I think we
> >>
> >> determined, my problem is probably with the CISCO telling FS that the
> >>
> >> call has been answered when really it hasn't yet.
> >>
> >>
> >>
> >>
> >> On Fri, Jan 6, 2012 at 8:01 AM, Oliver Schenk <olimonkey at gmail.com>
> wrote:
> >>
> >> Thanks for the help so far.
> >>
> >>
> >>
> >> Here is a pastebin of FreeSWITCH output:
> >>
> >> http://pastebin.com/i6Qgc7ws
> >>
> >>
> >> Notice how the "has been answered" log message comes immediately
> >>
> >> (within a few milliseconds) after the call was originated. I think
> >>
> >> this would suggest that the CISCO is immediately sending a 200 OK, as
> >>
> >> you suggested. I also turned on CISCO debugging, but I'm just trying
> >>
> >> to figure out how to get the information regarding SIP messages back
> >>
> >> to Freeswitch. I'll run the test again and see if I can get some
> >>
> >> useful CISCO debug.
> >>
> >>
> >> Which "debug ccsip" commands are relevant to what I want for the CISCO
> >>
> >> SIP debugging?
> >>
> >>
> >>
> >> Thanks!
> >>
> >>
> >>
> >>
> >>
> >> 2012/1/6 Gustavo Mársico <gustavomarsico at gmail.com>:
> >>
> >> I think I've a similar problem related to callcenter app. When I made an
> >> originate like this:
> >>
> >>
> >> originate loopback/2500/default/XML &bridge(user/2001)
> >>
> >>
> >> 2500 is an extension that leads to a callcenter application. In this
> case,
> >> we dial first to the queue and when an agent answered we call to the
> >> customer. As far as I know
> >>
> >> When the A-leg reaches to the queue, without selecting an agent, the
> call is
> >> automatically sent to the B-leg. As far as I see, there is a pre-answer
> >> method that fs needs to send the media to A-leg.
> >>
> >> In order to try to avoid this, I tried using ignore_early_media=true as
> part
> >> of the originate in A-leg and/or B-leg, with no luck.
> >>
> >>
> >> originate {ignore_early_media=true}loopback/2500/default/XML
> >> &bridge({ignore_early_media=true}user/2001)
> >>
> >>
> >> Dialplan: loopback/2500-b Regex (PASS) [CallCenter_Click2Call]
> >> destination_number(2500) =~ /^(2500)$/ break=on-false
> >>
> >> Dialplan: loopback/2500-b Action set(ignore_early_media=true)
> >>
> >> Dialplan: loopback/2500-b Action callcenter(click2call)
> >>
> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:154
> >> (loopback/2500-b) State Change CS_ROUTING -> CS_EXECUTE
> >>
> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:1180 Send
> signal
> >> loopback/2500-b [BREAK]
> >>
> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b
> >> CHANNEL KILL
> >>
> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:410
> >> (loopback/2500-b) State ROUTING going to sleep
> >>
> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:362
> >> (loopback/2500-b) Running State Change CS_EXECUTE
> >>
> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:417
> >> (loopback/2500-b) State EXECUTE
> >>
> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:375 loopback/2500-b
> >> CHANNEL EXECUTE
> >>
> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:192
> >> loopback/2500-b Standard EXECUTE
> >>
> >> EXECUTE loopback/2500-b set(open=true)
> >>
> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b
> SET
> >> [open]=[true]
> >>
> >> EXECUTE loopback/2500-b
> >>
> hash(insert/10.8.0.70-spymap/0000000000/fef7d864-b3c7-407c-a6e1-94386642bfbb)
> >>
> >> EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/0000000000/2500)
> >>
> >> EXECUTE loopback/2500-b
> >>
> hash(insert/10.8.0.70-last_dial/global/fef7d864-b3c7-407c-a6e1-94386642bfbb)
> >>
> >> EXECUTE loopback/2500-b set(RFC2822_DATE=Thu, 05 Jan 2012 13:36:08
> -0300)
> >>
> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b
> SET
> >> [RFC2822_DATE]=[Thu, 05 Jan 2012 13:36:08 -0300]
> >>
> >> EXECUTE loopback/2500-b set(ignore_early_media=true)
> >>
> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b
> SET
> >> [ignore_early_media]=[true]
> >>
> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:2133
> Application
> >> callcenter Requires media! pre_answering channel loopback/2500-b
> >>
> >> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:760 Pre-Answer
> >> loopback/2500-a!
> >>
> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930
> (loopback/2500-a)
> >> Callstate Change RINGING -> EARLY
> >>
> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send
> signal
> >> loopback/2500-b [BREAK]
> >>
> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b
> >> CHANNEL KILL
> >>
> >> 2012-01-05 13:36:08.541517 [NOTICE] switch_core_session.c:2135
> Pre-Answer
> >> loopback/2500-b!
> >>
> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930
> (loopback/2500-b)
> >> Callstate Change RINGING -> EARLY
> >>
> >> EXECUTE loopback/2500-b callcenter(click2call)
> >>
> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188
> (loopback/2500-a)
> >> Callstate Change EARLY -> ACTIVE
> >>
> >> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:755 Channel
> >> [loopback/2500-a] has been answered
> >>
> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send
> signal
> >> loopback/2500-b [BREAK]
> >>
> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b
> >> CHANNEL KILL
> >>
> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_ivr_originate.c:3266
> Originate
> >> Resulted in Success: [loopback/2500-a]
> >>
> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188
> (loopback/2500-b)
> >> Callstate Change EARLY -> ACTIVE
> >>
> >> 2012-01-05 13:36:08.541517 [INFO] switch_channel.c:2708 loopback/2500-a
> >> Flipping CID from "" <0000000000> to "Outbound Call" <XML>
> >>
> >>
> >>
> >>
> >>
> >> On Jan 5, 2012, at 4:17 AM, Oliver Schenk wrote:
> >>
> >>
> >> Also, maybe I should be doing something like this:
> >>
> >>
> >> sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)'
> >>
> >>
> >> instead of:
> >>
> >>
> >> sofia/internal/1091234567 at 192.168.x.x '&managed(ivrAppName)'
> >>
> >>
> >>
> >> but, I don't really have the CISCO configured as a gateway, nor do I
> >>
> >> know how really...probably not on the right track there.
> >>
> >>
> >>
> >>
> >>
> >> On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk <olimonkey at gmail.com>
> wrote:
> >>
> >> *bump*
> >>
> >>
> >>
> >> So I think maybe the way I'm doing the originate is the problem? In my
> >>
> >> call string I'm creating a connection directly from the CISCO
> >>
> >> (192.168.x.x) to the managed application, which may be why it starts
> >>
> >> playing straight away?
> >>
> >>
> >> Maybe I should be originating a call first and then only once I know
> >>
> >> the other side has picked up will I bridge the call to the IVR managed
> >>
> >> application.
> >>
> >>
> >> Problem is I dunno how to tell whether the other person has picked up
> >>
> >> (or even if the cisco is going to tell me) and I don't know how to do
> >>
> >> things to a call once it has been established.
> >>
> >>
> >>
> >> I'm currently reading the Dialplan wiki page, hoping to get something
> >>
> >> out of it there.
> >>
> >>
> >>
> >> Cheers
> >>
> >>
> >> Oliver
> >>
> >>
> >>
> >> On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk <olimonkey at gmail.com>
> wrote:
> >>
> >> I've been battling while creating an IVR using FreeSWITCH mod_managed
> >>
> >> and connecting through a CISCO 2811. Most things now work quite well,
> >>
> >> but I am having a few issues with the way the system answers calls (or
> >>
> >> doesn't answer calls...).
> >>
> >>
> >> I have FreeSWITCH running as a windows service on Windows Server 2008,
> >>
> >> which is connected via LAN to a CISCO 2811 with a 4 port FXO card,
> >>
> >> which is then connected to a POTS phone line.
> >>
> >>
> >>
> >> Take the following scenario:
> >>
> >>
> >> 1. Managed .NET application creates a call string and uses ESL to talk
> >>
> >> to freeswitch and originate a call:
> >>
> >>
> >> string callstring =
> >>
> >>
> "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567 at 192.168.x.x
> >>
> >> '&managed(ivrAppName)'";
> >>
> >> eslConnection.API("originate", callstring);
> >>
> >>
> >> where 192.168.x.x is the CISCO IP.
> >>
> >>
> >> 2. The CISCO sees that the phone number (1091234567) starts with a 1
> >>
> >> so it uses FXO port 1 and strips the 1 and uses the remaining phone
> >>
> >> number (091234567) to make the call.
> >>
> >>
> >> 3. My phone rings, I pick up and I can hear my IVR playing.
> >>
> >>
> >>
> >>
> >> These are my current problems:
> >>
> >>
> >> - IVR starts playing before I even pick up the phone. This means that
> >>
> >> if the system calls a mobile phone and the person doesn't pick up, the
> >>
> >> IVR will start playing and eventually the mobile phone will divert to
> >>
> >> voice mail. Obviously I then get a missed call and an sms saying I
> >>
> >> have a new voice mail, which is annoying. Instead I would like it to
> >>
> >> KNOW that no one has picked up, but I don't know how to do this.
> >>
> >> Somehow the CISCO needs to be able to tell FreeSWITCH that the call
> >>
> >> has not yet been answered. For some reason however as soon as the
> >>
> >> CISCO starts calling FreeSWITCH thinks the call is already connected.
> >>
> >> It doesn't know that the CISCO is actually still ringing. Maybe I'm
> >>
> >> doing originate the wrong way or something ...
> >>
> >>
> >> - The phone only rings for about 10 seconds before hanging up. I've
> >>
> >> tried "call_timeout", "bridge_answer_timeout". I've also tried setting
> >>
> >> CISCO "ring number". Nothing works, my phone still only rings for
> >>
> >> about 10 seconds. I don't know if this is a FreeSWITCH issue or a
> >>
> >> CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just
> >>
> >> starts playing even if no one answers the phone.
> >>
> >>
> >>
> >>
> >>
> >>
> >> CISCO Config for relevant FXO port:
> >>
> >>
> >> voice service voip
> >>
> >>  allow-connections h323 to h323
> >>
> >>  allow-connections h323 to sip
> >>
> >>  allow-connections sip to h323
> >>
> >>  allow-connections sip to sip
> >>
> >>  no supplementary-service h450.2
> >>
> >>  no supplementary-service h450.3
> >>
> >>  supplementary-service h450.12
> >>
> >>  no supplementary-service sip moved-temporarily
> >>
> >>  no supplementary-service sip refer
> >>
> >>  fax protocol cisco
> >>
> >>  sip
> >>
> >>  registrar server expires max 3600 min 3600
> >>
> >>  no update-callerid
> >>
> >>  no call service stop
> >>
> >>
> >> voice-port 0/3/2
> >>
> >>  output attenuation -3
> >>
> >>  no comfort-noise
> >>
> >>  cptone AU
> >>
> >>  impedance complex1
> >>
> >>  caller-id enable
> >>
> >> !
> >>
> >> dial-peer voice 100 pots
> >>
> >>  preference 1
> >>
> >>  destination-pattern 1T
> >>
> >>  port 0/3/2
> >>
> >> !
> >>
> >>
> >>
> >>
> >> Many Thanks,
> >>
> >>
> >> Oliver
> >>
> >>
> >>
> _________________________________________________________________________
> >>
> >> Professional FreeSWITCH Consulting Services:
> >>
> >> consulting at freeswitch.org
> >>
> >> http://www.freeswitchsolutions.com
> >>
> >>
> >> 
> >>
> >> 
> >>
> >>
> >> Official FreeSWITCH Sites
> >>
> >> http://www.freeswitch.org
> >>
> >> http://wiki.freeswitch.org
> >>
> >> http://www.cluecon.com
> >>
> >>
> >> FreeSWITCH-users mailing list
> >>
> >> FreeSWITCH-users at lists.freeswitch.org
> >>
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >>
> >> Professional FreeSWITCH Consulting Services:
> >>
> >> consulting at freeswitch.org
> >>
> >> http://www.freeswitchsolutions.com
> >>
> >>
> >> 
> >>
> >> 
> >>
> >>
> >> Official FreeSWITCH Sites
> >>
> >> http://www.freeswitch.org
> >>
> >> http://wiki.freeswitch.org
> >>
> >> http://www.cluecon.com
> >>
> >>
> >> FreeSWITCH-users mailing list
> >>
> >> FreeSWITCH-users at lists.freeswitch.org
> >>
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>
> >> http://www.freeswitch.org
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >> --
> >> Brian West
> >> FreeSWITCH Solutions, LLC
> >> Phone: +1 (918) 420-9266
> >> Fax:   +1 (918) 420-9267
> >> brian at freeswitch.org
> >> http://www.freeswitch.org
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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:4f06d49b32762089563979!
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Bharat Lalcheta
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