[Freeswitch-users] FreeSWITCH-users Digest, Vol 67, Issue 52

Bharat Lalcheta bharatlalcheta at gmail.com
Fri Jan 6 18:46:11 MSK 2012 

Can you please provide a simple example to use different codec in both leg
using dialplan

Regards,

Bharat Lalcheta

On Fri, Jan 6, 2012 at 9:10 PM, <
freeswitch-users-request at lists.freeswitch.org> wrote:

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> Today's Topics:
>
>   1. Re: Codec Preferance (Kristian Kielhofner)
>   2. Re: Codec Preferance (curriegrad2004)
>   3. Re: FreeSWITCH-users Digest, Vol 67, Issue 51 (Bharat Lalcheta)
>
>
> ---------- Forwarded message ----------
> From: Kristian Kielhofner <kris at kriskinc.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Fri, 6 Jan 2012 10:24:01 -0500
> Subject: Re: [Freeswitch-users] Codec Preferance
> inbound_codec_prefs
> outbound_codec_prefs
>
> These are Sofia profile configuration options.  They are NOT valid
> options for the directory UNLESS you're setting them as variable to
> use in your dialplan later.
>
> If you want to control codes on a per-user basis you have to set late
> negotiation and use the dialplan.
>
> On Fri, Jan 6, 2012 at 8:18 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
> >
>
>
>
> --
> Kristian Kielhofner
>
>
>
>
> ---------- Forwarded message ----------
> From: curriegrad2004 <curriegrad2004 at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Fri, 6 Jan 2012 07:37:06 -0800
> Subject: Re: [Freeswitch-users] Codec Preferance
>
> In theory he could use those variable names but it would involve some
> extra work ;)
> On 2012-01-06 7:24 AM, "Kristian Kielhofner" <kris at kriskinc.com> wrote:
>
>> inbound_codec_prefs
>> outbound_codec_prefs
>>
>> These are Sofia profile configuration options.  They are NOT valid
>> options for the directory UNLESS you're setting them as variable to
>> use in your dialplan later.
>>
>> If you want to control codes on a per-user basis you have to set late
>> negotiation and use the dialplan.
>>
>> On Fri, Jan 6, 2012 at 8:18 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
>> >
>>
>>
>>
>> --
>> Kristian Kielhofner
>>
>> _________________________________________________________________________
>> 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: Bharat Lalcheta <bharatlalcheta at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Cc:
> Date: Fri, 6 Jan 2012 21:10:01 +0530
> Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 67, Issue 51
> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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