[Freeswitch-users] How to bridge a call to an extension defined in dialplan

x.liu x.liu at hw.ac.uk
Tue Feb 28 17:08:26 MSK 2012 

Hi Hector,

Many thanks for your advice!

I had a try but without success yet. What I am doing is

1. call to an extension: 5801, it will execute my ESL Java app. I call 
it a-leg
<extension name="myEslNew5801">
<condition field="destination_number" expression="^5801$">
<action application="limit" data="hash outgoing 5801 4" />
<action application="socket" data="127.0.0.1:9981 async full"/>
</condition>
</extension>

2. From my ESL Java app, I originate b-leg to my softphoe and eavesdrop 
on extension 5801 by
     response = sendSyncSingleLineCommand(ctx.getChannel(), "api 
originate sofia/internal/1003%x.x.x.x 1003 &eavesdrop " + 
this.channelUniqueID);
     the channelUniqueID is the UUID of my a-leg session.

     So I hope my app do the recording and talking to the user and I 
listen it from the softphone 1003 via eavesdrop.

      At the moment I got message: Hangup on 1003 with [CS-ROUTING] 
[NO_ROUTE_DESTINATION]

     I am not quite sure if I understand correctly how the eavesdrop works:

     Which UUID should I use in "&eavesdrop UUID", the existing one of 
a-leg or I need to create a new one for b-leg?
     Do I need to do the bridge, in extension 5801 or in my app?


Thanks!
Xing



On 02/27/2012 08:59 PM, Hector Geraldino wrote:
> You can: bridge two endpoints (A ->  B), record the conversation using uuid_record (or the record application), and have a 3rd extension hearing the conversation using the eavesdrop app. Here's how I've done this before:
>
> Incoming call (A-Party)
> Bridge call to B-Party (bridge 1xxx at example.domain)
> Record the conversation (uuid_record aleg_uuid /path/to/audio.file)
> A C-Party listen using eavesdrop (originate 2xxx at example.domain&eavesdrop aleg_uuid)
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of x.liu
> Sent: Monday, February 27, 2012 2:52 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] How to bridge a call to an extension defined in dialplan
>
> Hi Peter,
>
> Thanks for your reply!
>
> You are right, that is what I have been experiencing but not I wanted.
>
> Actually my goal is simply to record the user call by my app (js or java
> via ESL) and at the same time
> I want to hear what the user is saying through my computer's speaker.
>
> The recording app seems to be ok ( I tried recording via dialplan
> commands as well as ESL Java app).
> What I am doing now is to bridge the call to my app for recording and to
> a softphone for hearing user's voice.
>
> I thought it would work as the wiki says the bridge can bridge the call
> to multiple endpoints at the same time,
> but unfortunately it won't work for me.
>
> Any suggestions to achieve my goal please: recording and hearing user's
> voice?
>     (Note: my app will do the recording and also allow me send TTS to the
> user)
>
> Best,
> Xing
>
>
>
> On 02/27/2012 06:33 PM, Peter Olsson wrote:
>> Can you explain what you're trying to achieve by doing this?
>>
>> If you bridge to multiple destinations, the first destination that answers will create the actual call - all other legs are terminated.
>>
>> /Peter
>>
>> Från: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för x.liu [x.liu at hw.ac.uk]
>> Skickat: den 27 februari 2012 19:11
>> Till: freeswitch-users at lists.freeswitch.org
>> Ämne: Re: [Freeswitch-users] How to bridge a call to an extension defined in dialplan
>>
>> Hi again,
>>
>> It looks like there is something wrong relating to the [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL],
>> but no idea how this happened.
>>
>> A bit desperately to solve this problem ASAP as we will have a demo in a few days.
>>
>> Please could anybody give any clue?
>>
>> Many thanks!
>>
>> On 02/27/2012 11:34 AM, x.liu wrote:
>> Now I tried it at a different machine with different version of FS.
>> The behaviour is slightly different. This time the softphone is ringing but the recording extension is not correctly executed.
>> The version for this test is "FreeSWITCH Version 1.0.head (git-e566057 2011-12-14 10-27-31 -0500)"
>>    (The FS version for previous tries is "FreeSWITCH Version 1.0.head (git-54ddef0 2011-12-06 21-53-45 -0600)" )
>>
>> I copy part of terminal message here, hope it is not too much message for the email list.
>>
>> --------------------------------------------------
>> 09:04.180223 [NOTICE] switch_channel.c:920 New Channel sofia/external/anonymous at sipgate.co.uk<mailto:sofia/external/anonymous at sipgate.co.uk>   [7596ad60-6133-11e1-9141-8b50a5b8b13b]
>> 09:04.180223 [INFO] mod_dialplan_xml.c:481 Processing anonymous<anonymous>->74997 in context public
>> 09:04.180223 [NOTICE] switch_channel.c:920 New Channel sofia/internal/1003 at x.x.x.x<mailto:sofia/internal/1003 at x.x.x.x>   [75976d18-6133-11e1-9145-8b50a5b8b13b]
>> 09:04.180223 [NOTICE] switch_channel.c:920 New Channel sofia/internal/74999 at x.x.x.x<mailto:sofia/internal/74999 at x.x.x.x>   [75977f24-6133-11e1-914a-8b50a5b8b13b]
>> 09:04.180223 [NOTICE] switch_channel.c:920 New Channel sofia/internal/anonymous at x.x.x.x<mailto:sofia/internal/anonymous at x.x.x.x>   [7597aab2-6133-11e1-914e-8b50a5b8b13b]
>> 09:04.180223 [INFO] mod_dialplan_xml.c:481 Processing anonymous<anonymous>->1003 in context public
>> 09:04.180223 [NOTICE] switch_channel.c:920 New Channel sofia/internal/anonymous at x.x.x.x<mailto:sofia/internal/anonymous at x.x.x.x>   [75980192-6133-11e1-9150-8b50a5b8b13b]
>> 09:04.180223 [NOTICE] switch_ivr.c:1711 Transfer sofia/external/anonymous at sipgate.co.uk<mailto:sofia/external/anonymous at sipgate.co.uk>   to XML[1003 at public]
>> 09:04.180223 [NOTICE] switch_ivr_originate.c:3182 Hangup sofia/internal/1003 at x.x.x.x<mailto:sofia/internal/1003 at x.x.x.x>   [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
>> 09:04.180223 [NOTICE] switch_ivr_originate.c:3182 Hangup sofia/internal/74999 at x.x.x.x<mailto:sofia/internal/74999 at x.x.x.x>   [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
>> 09:04.180223 [INFO] mod_dptools.c:2897 Originate Failed.  Cause: ORIGINATOR_CANCEL
>> 09:04.180223 [INFO] mod_dialplan_xml.c:481 Processing anonymous<anonymous>->1003 in context public
>> 09:04.180223 [NOTICE] switch_core_session.c:1397 Session 2 (sofia/internal/1003 at x.x.x.x<mailto:sofia/internal/1003 at x.x.x.x>) Ended
>> 09:04.180223 [NOTICE] switch_core_session.c:1399 Close Channel sofia/internal/1003 at x.x.x.x<mailto:sofia/internal/1003 at x.x.x.x>   [CS_DESTROY]
>> 09:04.180223 [NOTICE] switch_ivr.c:1711 Transfer sofia/external/anonymous at sipgate.co.uk<mailto:sofia/external/anonymous at sipgate.co.uk>   to XML[1003 at default]
>> 09:04.180223 [INFO] mod_dialplan_xml.c:481 Processing anonymous<anonymous>->1003 in context default
>> 09:04.180223 [NOTICE] sofia.c:427 Hangup sofia/internal/anonymous at x.x.x.x<mailto:sofia/internal/anonymous at x.x.x.x>   [CS_EXECUTE] [BLIND_TRANSFER]
>> 09:04.201188 [NOTICE] switch_core_session.c:1397 Session 4 (sofia/internal/anonymous at x.x.x.x<mailto:sofia/internal/anonymous at x.x.x.x>) Ended
>> 09:04.201188 [NOTICE] switch_core_session.c:1399 Close Channel sofia/internal/anonymous at x.x.x.x<mailto:sofia/internal/anonymous at x.x.x.x>   [CS_DESTROY]
>> 09:04.201188 [INFO] switch_ivr_async.c:3164 Bound B-Leg: *1 execute_extension::dx XML features
>> 09:04.201188 [INFO] mod_dialplan_xml.c:481 Processing anonymous<anonymous>->74999 in context public
>> 09:04.201188 [INFO] switch_ivr_async.c:3164 Bound B-Leg: *2 record_session:: recordings/anonymous.2012-02-27-11-09-04.wav
>> 09:04.201188 [NOTICE] switch_core_session.c:1397 Session 3 (sofia/internal/74999 at x.x.x.x<mailto:sofia/internal/74999 at x.x.x.x>) Ended
>> 09:04.201188 [NOTICE] switch_core_session.c:1399 Close Channel sofia/internal/74999 at x.x.x.x<mailto:sofia/internal/74999 at x.x.x.x>   [CS_DESTROY]
>> 09:04.201188 [INFO] switch_ivr_async.c:3164 Bound B-Leg: *3 execute_extension::cf XML features
>> 09:04.201188 [NOTICE] sofia.c:6134 Hangup sofia/internal/anonymous at x.x.x.x<mailto:sofia/internal/anonymous at x.x.x.x>   [CS_EXECUTE] [ORIGINATOR_CANCEL]
>> 09:04.201188 [INFO] switch_ivr_async.c:3164 Bound B-Leg: *4 execute_extension::att_xfer XML features
>> 09:04.201188 [NOTICE] switch_core_session.c:1397 Session 5 (sofia/internal/anonymous at x.x.x.x<mailto:sofia/internal/anonymous at x.x.x.x>) Ended
>> 09:04.201188 [NOTICE] switch_core_session.c:1399 Close Channel sofia/internal/anonymous at x.x.x.x<mailto:sofia/internal/anonymous at x.x.x.x>   [CS_DESTROY]
>> 09:04.201188 [NOTICE] switch_channel.c:920 New Channel sofia/internal/sip:1003 at y.y.y.y:5062 [759a6630-6133-11e1-9158-8b50a5b8b13b]
>> 09:04.740223 [NOTICE] sofia.c:5462 Ring-Ready sofia/internal/sip:1003 at y.y.y.y:5062!
>> 09:04.740223 [NOTICE] mod_sofia.c:2470 Ring-Ready sofia/external/anonymous at sipgate.co.uk<mailto:sofia/external/anonymous at sipgate.co.uk>!
>> 09:04.740223 [NOTICE] switch_ivr_originate.c:483 Ring Ready sofia/external/anonymous at sipgate.co.uk<mailto:sofia/external/anonymous at sipgate.co.uk>!
>> 09:07.700221 [NOTICE] sofia.c:6040 Channel [sofia/internal/sip:1003 at y.y.y.y:5062] has been answered
>> 09:07.700221 [NOTICE] switch_ivr_originate.c:483 Ring Ready sofia/external/anonymous at sipgate.co.uk<mailto:sofia/external/anonymous at sipgate.co.uk>!
>> -------------------------------------------------
>>
>>
>> Hi,
>>
>> H tried this way
>>
>> <action application="bridge" data="sofia/internal/1003%${sip_profile},sofia/internal/74991@${domain}"/>
>>
>> 74991 is reached and I heard the prompt. The 1003 softphone is terminated immediately after it is reached.
>> then extension 74991 hung up itself as well.
>>
>> I saw the terminal message:
>> mod_dptools.c:2897 Originate Failed.  Cause: ORIGINATOR_CANCEL
>>
>> What would be the reason for that?
>>
>> Thanks,
>> Xing
>>
>>
>>
>> On 02/26/2012 10:29 PM, Liu, Xingkun wrote:
>>
>>
>> Hello,
>>
>> I'd like to bridge an incoming to two endpoints simultaneuously:
>> one is a softphone which is registered to FS (to user 1003),
>> another is just an extension which is defined in conf/dialplan/public.xml
>>
>> It looks like:
>>
>> <extension name="testRecording_bridge">
>>     <condition field="destination_number" expression="^(74990)$">
>>       <action application="bridge" data="sofia/internal/1003%${sip_profile},sofia/internal/74991%${sip_profile}"/>
>>     </condition>
>> </extension>
>>
>> <extension name="testRecordingOnly">
>>       <condition field="destination_number" expression="^74991$">
>>         <action application="set" data="record_waste_resources=true"/>
>>         <action application="javascript" data="testRecordFile.js"/>
>>       </condition>
>>     </extension>
>>
>> The incoming call dial to extension 74990, from there I want to bridge the call to the sofphone at 1003 and
>> to the extension 74991.
>>
>> The problem is that FS can not locate user 74991. So my question is what else I need to modify in order
>> to bridge the call also to extension 74991 at the same time?
>>
>> I did lots of Googling, checked the wiki and tried various ways but still without success.
>>
>> Any advices please?
>>
>> Many Thanks!
>> Xing
>>
>> ________________________________
>>
>> Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012.
>>
>> Heriot-Watt University is a Scottish charity registered under charity number SC000278.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
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>>
>>
>> ________________________________
>>
>> Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012.
>>
>> Heriot-Watt University is a Scottish charity registered under charity number SC000278.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
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>>
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>>
>>
>> ________________________________
>>
>> Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012.
>>
>> Heriot-Watt University is a Scottish charity registered under charity number SC000278.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org<mailto: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
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>>
>>
>>
>> ________________________________
>>
>> Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012.
>>
>> Heriot-Watt University is a Scottish charity registered under charity number SC000278.
>> !DSPAM:4f4bc71e32767806511310!
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
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>>
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>
>



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