[Freeswitch-users] Codec Preferance

Bharat Lalcheta bharatlalcheta at gmail.com
Fri Jan 6 19:13:52 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:
>
>> 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 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
>



-- 
Bharat Lalcheta
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/3ab0406d/attachment-0001.html

Join us at ClueCon 2011 Aug 9-11, 2011
More information about the FreeSWITCH-users mailing list