Re: [Freeswitch-users] uuid_bridge kills both channels if they are executing java app
I've sent deep-breath message to the dev list.
Just-in-case, here is a "cross-post":
Hi there!
This message is a forward from user-mail-list.
I'm trying to fix such a problem:
FreSwithch compiled from SVN-trunk, date = 11/02/2009.
What is need: connect two users, initially one is on the home-grown
java-based IVR and other party is off hook.
What is done/got:
User1 is on the java application, it represents simple IVR system, and the
most used FS API operation is "streamFile".
User2 is off hook.
next:
(mod_socket) create_uuid
bgapi originate
{origination_caller_id_name=User1}[origination_uuid=uuid_x]User1 &park()
uuid_bridge uuid_User1 uuid_User2
FS log is here: http://pastebin.freeswitch.org/11380
Thank you much for any help,
Artem
On Wed, Dec 2, 2009 at 10:24 PM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:
> you should be working on SVN trunk if you are doing development, we are so
> far forward from 1.0.4 we can't do debugging very easily.
>
> I don't know all of the details of what you are trying to do but you are
> hitting some race conditions because of the async nature of the socket
> connection and the way you are using it.
>
>
>
>
> On Wed, Dec 2, 2009 at 1:08 PM, Artem Shiyanov wrote:
>
>> I'm back again with the same issue.
>> Now it is became worse: it reproduces occasionally.
>> [FS version is 1.04, test_load = 2 active calls]
>>
>> I've got 2 logs: successful and not.
>> Here is a bad_case:
>>
>> 2009-12-02 13:27:55.159931 [NOTICE] switch_core_session.c:1576 Execute
>> java(/usr/local/freeswitch/scripts/fs2agi.jar
>> org.starpound.fs2agi.Translator
>> ${agi_url})
>> Dec 2, 2009 1:27:55 PM org.starpound.fs2agi.Translator run
>> INFO: ***
>> Dec 2, 2009 1:27:55 PM org.starpound.fs2agi.Translator run
>> INFO: Run AGI application agi://localhost:4573/hello.agi?callId=929 for
>> session
>> 2898ad41-4ec1-4628-89fd-651a93a7221d
>> 2009-12-02 13:27:55.169841 [NOTICE] switch_cpp.cpp:1130 Run AGI
>> application
>> agi://localhost:4573/hello.agi?callId=929
>> 2009-12-02 13:28:02.31 [CRIT] mod_local_stream.c:234 Leaking stream
>> handle!
>>
>> [switch_ivr_play_file() src/switch_ivr_play_say.c:1026]
>> 2009-12-02 13:28:04.799806 [NOTICE] switch_channel.c:602 New Channel
>> sofia/internal/2001 [76d2c0e9-16a4-4098-92c0-5977cb482e17]
>> 2009-12-02 13:28:05.148834 [NOTICE] sofia.c:3353 Ring-Ready
>> sofia/internal/2001!
>> 2009-12-02 13:28:05.855093 [NOTICE] sofia.c:3794 Channel
>> [sofia/internal/2001] has
>> been answered
>> Dec 2, 2009 1:28:05 PM org.starpound.fs2agi.Translator tellAllWeCrashed
>> INFO: AGI application agi://localhost:4573/hello.agi?callId=929 for
>> session
>> 2898ad41-4ec1-4628-89fd-651a93a7221d crashed. Exception is:
>> java.lang.Exception: Internal FreeSwitch failure while streamming file,
>> see
>> FreeSwitch logs for details
>> at
>>
>> org.starpound.fs2agi.agicommands.StreamFileCommand.execute(StreamFileCommand.java:36)
>> at org.starpound.fs2agi.AgiConnection.run(AgiConnection.java:48)
>> at org.starpound.fs2agi.Translator.run(Translator.java:56)
>> at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method)
>> at
>>
>> sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39)
>> at
>>
>> sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25)
>> at java.lang.reflect.Method.invoke(Method.java:597)
>> at org.freeswitch.Launcher.launch(Launcher.java:80)
>> 2009-12-02 13:28:05.870185 [NOTICE] switch_core_state_machine.c:179 Hangup
>> sofia/internal/2001 [CS_EXECUTE] [NORMAL_CLEARING]
>> 2009-12-02 13:28:05.878807 [INFO] switch_cpp.cpp:1130 AGI application
>> agi://localhost:4573/hello.agi?callId=929 crashed. See FS2AGI log for
>> details.
>> 2009-12-02 13:28:05.894422 [NOTICE] switch_ivr_bridge.c:667 Hangup
>> sofia/external/6786081...@66.19.38.143 [CS_SOFT_EXECUTE]
>> [DESTINATION_OUT_OF_ORDER]
>> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1086 Session 17
>> (sofia/external/6786081...@66.19.38.143) Ended
>> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1088 Close
>> Channel
>> sofia/external/6786081...@66.19.38.143 [CS_DESTROY]
>> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1086 Session 18
>> (sofia/internal/2001) Ended
>> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1088 Close
>> Channel
>> sofia/internal/2001 [CS_DE
Re: [Freeswitch-users] uuid_bridge kills both channels if they are executing java app
e Method) at sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39) at sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25) at java.lang.reflect.Method.invoke(Method.java:597) at org.freeswitch.Launcher.launch(Launcher.java:80) 2009-12-02 13:32:00.149080 [INFO] switch_cpp.cpp:1130 AGI application agi://localhost:4573/hello.agi?callId=932 crashed. See FS2AGI log for details. 2009-12-02 13:32:00.388838 [INFO] switch_rtp.c:1869 Auto Changing port from 172.26.10.39:26402 to 91.190.120.190:26402 Suggestions? On Sat, Nov 21, 2009 at 12:58 PM, Artem Shiyanov wrote: > Anthony, > > >>As soon as you call uuid_bridge you are transferring both legs of the > call to bridge to each other. > >>This means your java app must exit so the channels can connect to each > other. > > I didn't know that. Now my java app is exiting upon the onHangup() call so > everything has become "ok". Thank you much. > I'll add note to the wiki about this issue. > > Artem > > > > > On Fri, Nov 20, 2009 at 5:49 AM, Anthony Minessale < > anthony.miness...@gmail.com> wrote: > >> Your "annoying behaviour" is the exact behavior you should be getting >> considering what you told FS to do. >> >> As soon as you call uuid_bridge you are transferring both legs of the call >> to bridge to each other. >> This means your java app must exit so the channels can connect to each >> other. >> >> remember that you hangup hook can be called when the channel is >> transferred not only when it hangs up. >> you have to test which is happening based on the input to your callback. >> >> >> On Thu, Nov 19, 2009 at 1:46 PM, Artem Shiyanov wrote: >> >>> Hi there! >>> >>> I've got annoying FS behavior: >>> There are 2 channels executing the same Java application (application >>> itself is an IVR). If I try to bridge them with uuid_bridged then both >>> channels are killed. Here is a log from FS console: >>> uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2 >>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de >>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165 >>> (sofia/internal/1...@192.168.147.130) State Change CS_EXECUTE -> >>> CS_HIBERNATE >>> 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook called >>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done >>> playing file >>> 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done >>> playing file >>> 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send signal >>> sofia/internal/1...@192.168.147.130 [BREAK] >>> 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167 >>> (sofia/internal/1...@master.agent.starpoundtech.net) State Change >>> CS_EXECUTE -> CS_HIBERNATE >>> 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook called >>> API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2 >>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output: >>> +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de >>> >>> freeswi...@localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG] >>> switch_core_session.c:933 Send signal >>> sofia/internal/1...@master.agent.starpoundtec >>> 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send signal >>> sofia/internal/1...@192.168.147.130 [BREAK] >>> >>> 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream >>> handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] >>> 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send signal >>> sofia/internal/1...@master.agent.starpoundtech.net [BREAK] >>> >>> (FS version is 1.0.4) >>> >>> Any thoughts? >>> >>> >>> Artem >>> >>> >>> >>> ___ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_miness...@hotmail.com >> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:8...@conference.freeswitch.org >> iax:gu...@conference.freeswitch.org/888 >> googletalk:conf+...@conference.freeswitch.org >> pstn:213-799-1400 >> >> ___ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] call barge in
1 - config 2 - I've done this with programming 3 - suppose programming would be needed Here is a bunch of code, search there ''barge" Artem On Wed, Dec 2, 2009 at 11:34 AM, Nikolay Kondratyev wrote: > Hi all, > > > > I’m evaluating FS for our organization. > > I must fulfill the following requirements: > > 1. Call recording: All (or selected) calls to the secretary must be > recorded. > > 2. Call barge in: Assume that two subscribers are talking to each other. > Secretary makes “emergency” (for example, an extension with emergency > prefix) call to one of these subscribers -> Secretary barges in the > established call (conference). > > 3. Call drop when emergency call arrives: the same as above, but > established call is dropped end emergency call is established. > > > > Can anybody please advise if this is possible with FS? > > If yes, is it just a configuration task, or additional programming will be > needed? > > > > Thanks in advance, > > Nikolay. > > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Clarification about channel variables please.
both types of variables are mutable On Sun, Nov 22, 2009 at 2:25 PM, Lon Baker wrote: > Are either global or regular channel variable mutable during a call? > Or can they only be set before and after? > > Any clarification would help, since the existing wiki doesn't make it > clear. > > Lon > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] uuid_bridge kills both channels if they are executing java app
Anthony, >>As soon as you call uuid_bridge you are transferring both legs of the call to bridge to each other. >>This means your java app must exit so the channels can connect to each other. I didn't know that. Now my java app is exiting upon the onHangup() call so everything has become "ok". Thank you much. I'll add note to the wiki about this issue. Artem On Fri, Nov 20, 2009 at 5:49 AM, Anthony Minessale < anthony.miness...@gmail.com> wrote: > Your "annoying behaviour" is the exact behavior you should be getting > considering what you told FS to do. > > As soon as you call uuid_bridge you are transferring both legs of the call > to bridge to each other. > This means your java app must exit so the channels can connect to each > other. > > remember that you hangup hook can be called when the channel is transferred > not only when it hangs up. > you have to test which is happening based on the input to your callback. > > > On Thu, Nov 19, 2009 at 1:46 PM, Artem Shiyanov wrote: > >> Hi there! >> >> I've got annoying FS behavior: >> There are 2 channels executing the same Java application (application >> itself is an IVR). If I try to bridge them with uuid_bridged then both >> channels are killed. Here is a log from FS console: >> uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2 >> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de >> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165 >> (sofia/internal/1...@192.168.147.130) State Change CS_EXECUTE -> >> CS_HIBERNATE >> 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook called >> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done playing >> file >> 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done playing >> file >> 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send signal >> sofia/internal/1...@192.168.147.130 [BREAK] >> 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167 >> (sofia/internal/1...@master.agent.starpoundtech.net) State Change >> CS_EXECUTE -> CS_HIBERNATE >> 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook called >> API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2 >> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output: >> +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de >> >> freeswi...@localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG] >> switch_core_session.c:933 Send signal >> sofia/internal/1...@master.agent.starpoundtec >> 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send signal >> sofia/internal/1...@192.168.147.130 [BREAK] >> >> 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream >> handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] >> 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send signal >> sofia/internal/1...@master.agent.starpoundtech.net [BREAK] >> >> (FS version is 1.0.4) >> >> Any thoughts? >> >> >> Artem >> >> >> >> ___ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_miness...@hotmail.com > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:8...@conference.freeswitch.org > iax:gu...@conference.freeswitch.org/888 > googletalk:conf+...@conference.freeswitch.org > pstn:213-799-1400 > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] uuid_bridge kills both channels if they are executing java app
Hi there! I've got annoying FS behavior: There are 2 channels executing the same Java application (application itself is an IVR). If I try to bridge them with uuid_bridged then both channels are killed. Here is a log from FS console: uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2 7d6c08fc-62bf-4a6c-a9ae-763d607e43de 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165 (sofia/internal/ 1...@192.168.147.130) State Change CS_EXECUTE -> CS_HIBERNATE 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook called 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1...@192.168.147.130 [BREAK] 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167 (sofia/internal/ 1...@master.agent.starpoundtech.net) State Change CS_EXECUTE -> CS_HIBERNATE 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook called API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output: +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de freeswi...@localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1...@master.agent.starpoundtec 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send signal sofia/internal/1...@192.168.147.130 [BREAK] 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send signal sofia/internal/1...@master.agent.starpoundtech.net [BREAK] (FS version is 1.0.4) Any thoughts? Artem ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod dptools record problem - hangup channel with invalid file path
I had almost the same problem- it was needed to record everything, even if the record path doesn't exist - it was requested to create the needed path. For this purpose I've used event_socket command "api system ...", precisely, api system mkdir -p path And after this command I've started recording. So, you may the same approach. On Wed, Nov 18, 2009 at 11:26 PM, William Kendi ... < william.nis...@voicetechnology.com.br> wrote: > Actually, I am integrating FreeSWITCH with a weird IVR Framework, and the > current behaviour of the "mod dptools record" application breaks some rules > of the weird IVR Framework that must be integrated with FreeSWITCH. > In order to integrate FreeSWITCH with the weird IVR Framework, the "mod > dptools record" application mustn't terminate the call when invalid file > paths are passed, and a notification of the invalid file path through the > event system of FreeSWITCH should be enough for me, like the behaviour of > the "mod dptools playback" application when invalid file paths are passed. > > Thanks in advance. > > ** > 2009/11/18 Michael Jerris > > Okay, I'll ask the obvious question. Why are you passing record invalid >> file paths and why should it not fail if you do? >> >> Mike >> >> On Nov 18, 2009, at 2:38 PM, William Kendi ... wrote: >> >> > While I was testing the "mod dptools record" application using invalid >> file paths, i noted that the "mod dptools record" application terminated the >> call. >> > I am currently looking for a way to change this behaviour. >> > Any suggestions? Can this be done? >> >> >> ___ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Dialpan: try.. finally analogs
Closed. As (almost) usual the reason was me. Anthony's hint works perfectly: api uuid_transfer bridge:sofia/gateway// inline Sorry for bothering! On Sat, Nov 7, 2009 at 3:08 AM, Anthony Minessale < anthony.miness...@gmail.com> wrote: > If you know the reason, why are you so puzzled by it? > I think you should not assume you understand what is happening unless you > really do. > > I think you need to provide an exact description of what you are doing so I > can explain to you where you are making the mistake. > > Make sure you are on latest SVN and reproduce this in a console log for us > and add an exact description of what you are doing in detail. > > > On Thu, Nov 5, 2009 at 11:44 AM, Artem Shiyanov wrote: > >> Hello! >> >> I have to deal with classic problem: "Leaking stream handle" in FS >> console. I also know the reason - firstly channel is sent to the extension >> with "playback" and later it is transfered to another extensions with >> "execute_extension" or, another trouble-case - channel is bridged to some >> addres. >> I do not ask (but I wish to) why FS doesn't close stream automatically >> when channel is gone. >> I ask whether it is possible to use some "try.. finally" construction in >> diaplan? If "yes" then I can simply stop playback in the "finally" block.. >> >> Any thoughs? >> >> ___ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_miness...@hotmail.com > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:8...@conference.freeswitch.org > iax:gu...@conference.freeswitch.org/888 > googletalk:conf+...@conference.freeswitch.org > pstn:213-799-1400 > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Java example
Here is rather big and, let's say, complete example of mod_java usage: https://starpound.svn.sourceforge.net/svnroot/starpound/trunk/src/fs2agi The goal of this project is to be a proxy between FreeSwitch and server application which knows Asterisk AGI. On Mon, Nov 2, 2009 at 2:53 PM, wrote: > > Hi, > > Can you please paste me your sample java dialplan code that work for you ? > ..coz m also facing the same problem. > > My mod_java is loaded properly. > Also /usr/lib/jvm/java-1.5.0-gcj-4.3-1.5.0.0/jre/lib/i386/client/libjvm.so > and freeswitch.jar in java.conf.xml is specified properly. > > I have written a java code to print HII on the console but its not > printing. Level mentioned is INFO. > > on FS console it just prints > EXECUTE sofia/internal/1...@192.168.1.144:5061 java(testing.class) > > > I am attaching my java code herewith. > Can u just tell me where more i should do the modification to get my > dialplan work. > > Waiting for your kind reply. > > Thanks & Regards, > Dipen Velani > > On Fri 19/12/08 4:09 AM , kriko wrote: > > > Seems like my dialplan was a bit problematic, it works now. > > Thanks. > > On Thu, 18 Dec 2008 15:19:22 +0100, Anthony Minessale > > wrote: > > > did you turn up your console log level high enough to see it? The > > default > > > level is "INFO" > > > > > > > > >> > > >> ___ > > >> Freeswitch-users mailing list > > >> > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> http://www.freeswitch.org > > >> > > > > > > > > > > > -- > > kriko > > ___ > > Freeswitch-users mailing list > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > Msg sent via @Mail - http://atmail.com/ > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] [mod_socket] Can't set channel variable SOMETIMES
Hello!
I've got strange problem:
In my app which talks to FreeSwitch via mod_socket there is such logic:
for (channel in {array of channel1, channel2, channel3}) {
set value for the channel variable X
}
uuid_transfer channel1 extension Y
uuid_transfer -both channel2 extension Y
In extension Y there is a logic based on the channel variable X.
The problem itself: every second launching falls, the reason - channel
variable X hasn't been set for one of channel1..3.
Sniffer shows that commands are sent (and confirmated by FreeSwitch)
consiquently, FreeSwitch sends OK for each command in time.
Temporary sollution that I found is to wait 0.5 sec after 'set' commands
have been sent and before transfer any channel to extension Y.
Maybe some already faced with similar problem?
Thanks,
Artem
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Re: [Freeswitch-users] Mod_socket: custom caller sip domain in originate command
Tested- it works!
Thanks a lot!!
On Mon, Oct 26, 2009 at 6:32 PM, mayamatakeshi wrote:
>
>
> On Tue, Oct 27, 2009 at 12:20 AM, Artem Shiyanov wrote:
>
>> Hi there!
>>
>> Please, suggest how to specify custom caller sip domain (logical) in
>> originate command.
>> I've been trying several alternatives but no one worked:
>> 1) specify full sip address in
>> origination_caller_id_number=1...@uat.pbx.mblagov.starpoundtech.net - FS
>> adds its IP address so the
>> result From address would be 1...@uat.pbx.mblagov.starpoundtech.net@
>> 172.26.200.250
>> 2) specify sip_from_uri varuable like follows:
>> bgapi originate {origination_caller_id_number=1000,
>> sip_from_uri=1...@uat.pbx.mblagov.starpoundtech.net,originate_timeout=10}[origination_uuid=bf8e9778-d410-4504-988d-2d405303183c]sofia/internal/1002%172.26.200.250
>> &park()
>> FS doesn't even dial the called endpoint.
>>
>
> I use
> sip_invite_domain
> on bridge. I suppose it works with originate too.
>
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>
>
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[Freeswitch-users] Mod_socket: custom caller sip domain in originate command
Hi there!
Please, suggest how to specify custom caller sip domain (logical) in
originate command.
I've been trying several alternatives but no one worked:
1) specify full sip address in
origination_caller_id_number=1...@uat.pbx.mblagov.starpoundtech.net - FS
adds its IP address so the
result From address would be 1...@uat.pbx.mblagov.starpoundtech.net@
172.26.200.250
2) specify sip_from_uri varuable like follows:
bgapi originate {origination_caller_id_number=1000,
sip_from_uri=1...@uat.pbx.mblagov.starpoundtech.net,originate_timeout=10}[origination_uuid=bf8e9778-d410-4504-988d-2d405303183c]sofia/internal/1002%172.26.200.250
&park()
FS doesn't even dial the called endpoint.
Thanks,
Artem
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Re: [Freeswitch-users] mod_socket: custom timeout for HEARTBEAT event
Finally!! Thank you Michael, I didn't know about "status" app. It satisfies all my desires. And again, thanks for all the community for the strong support! Artem On Tue, Oct 13, 2009 at 10:48 PM, Michael Collins wrote: > > > On Tue, Oct 13, 2009 at 8:32 AM, Artem Shiyanov wrote: > >> Sorry for my foolishness but I stil can't grasp it. >> I'm developing app based on inbound mod_event_socket and I don't know how >> to run "enable_heartbeat" or " sched_heartbeat" without specifying any alive >> session uuid. I tried to use "create_uuid" and send mentioned commands to >> the created uuid but this approach doesn't work. I do need to monitor the >> FS state itself rather then any particular call. >> >> Please, enlighten me! >> Artem >> > > So if I understand correctly, you want to see the system heartbeat more > frequently than every 20 seconds? What, exactly, is in the heartbeat that > you need? You can get the same information by sending the API "status" and > reading the results. > > If I read the source code correctly it looks like the system heartbeat is > hard-coded at 20 seconds. See switch_core.c, line 155. However, before you > go messing with a core file I would make sure that you can't already use the > tools that are available, like sending a "status" API every 5 seconds (or > whatever interval you need). > > -MC > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_socket: custom timeout for HEARTBEAT event
Sorry for my foolishness but I stil can't grasp it. I'm developing app based on inbound mod_event_socket and I don't know how to run "enable_heartbeat" or " sched_heartbeat" without specifying any alive session uuid. I tried to use "create_uuid" and send mentioned commands to the created uuid but this approach doesn't work. I do need to monitor the FS state itself rather then any particular call. Please, enlighten me! Artem On Mon, Oct 12, 2009 at 6:26 PM, Anthony Minessale < anthony.miness...@gmail.com> wrote: > it works in either case with or without media > the syntax for setting the frequency was answered above. > > > > On Mon, Oct 12, 2009 at 12:38 AM, Artem Shiyanov wrote: > >> Michael, Diego, >> thanks for the rapid answers! >> >> As far as I understand, "enable_heartbeat" app is launching >> SESSION_HEARTBEAT events that will stop when the call will be cleared. Also >> I "heard" that "enable_heartbeat" works only for calls with proxied media. >> >> What I want is to monitor FreeSwitch status: is it alive and what is the >> system status message. This info is provided in HEARTBEAT event gracefully >> but in constant time period = 20 sec. So the main question is- how to >> customize this period? >> >> >> Artem >> >> >> >> >> On Mon, Oct 12, 2009 at 2:52 AM, Michael Jerris wrote: >> >>> >>> On Oct 11, 2009, at 5:44 PM, Diego Viola wrote: >>> >>> > Maybe you want enable_heartbeat or sched_heartbeat from mod_dptools? >>> > >>> > You can pass your parameters in second to these two. >>> > >>> > Example: >>> > >>> > >>> > >>> > >>> > Where 1 in this case is the number of heartbeats per seconds. >>> > >>> >>> Number of seconds between hearbeats, not hearbeats per second. >>> >>> >>> > You can use that example on the Dialplan XML but you can also use it >>> > on mod_event_socket outbound, etc. >>> >>> >>> ___ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> ___ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_miness...@hotmail.com > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:8...@conference.freeswitch.org > iax:gu...@conference.freeswitch.org/888 > googletalk:conf+...@conference.freeswitch.org > pstn:213-799-1400 > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_socket: custom timeout for HEARTBEAT event
Michael, Diego, thanks for the rapid answers! As far as I understand, "enable_heartbeat" app is launching SESSION_HEARTBEAT events that will stop when the call will be cleared. Also I "heard" that "enable_heartbeat" works only for calls with proxied media. What I want is to monitor FreeSwitch status: is it alive and what is the system status message. This info is provided in HEARTBEAT event gracefully but in constant time period = 20 sec. So the main question is- how to customize this period? Artem On Mon, Oct 12, 2009 at 2:52 AM, Michael Jerris wrote: > > On Oct 11, 2009, at 5:44 PM, Diego Viola wrote: > > > Maybe you want enable_heartbeat or sched_heartbeat from mod_dptools? > > > > You can pass your parameters in second to these two. > > > > Example: > > > > > > > > > > Where 1 in this case is the number of heartbeats per seconds. > > > > Number of seconds between hearbeats, not hearbeats per second. > > > > You can use that example on the Dialplan XML but you can also use it > > on mod_event_socket outbound, etc. > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] mod_socket: custom timeout for HEARTBEAT event
Hi all! As it stays in wiki: ... HEARTBEAT Status information for freeswitch trigerred by freeswitch's heartbeat every 20 seconds. ... Is there any way to customize timeout of HEARTBEAT events? Thanks in advance, Artem ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] "barge in" implementation with mod_socket and eavesdrop
Anthony, thank you much, "three_way" is a powerful app! I've added small description in wiki, if you don't mind. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_three_way Meanwhile, another question has been evolved: In words of the scenario below, now if A hangs up the call then all other (B, C) channels also being hanged up automatically by FreeSwitch. Is there any way to save the call B <--> C if A has hanged up the phone? Again, the whole scenario is the same as it is described in my first message. Artem On Tue, Sep 15, 2009 at 7:08 PM, Anthony Minessale < anthony.miness...@gmail.com> wrote: > yes call the app as "three_way" like i said in the other thread. > > > On Tue, Sep 15, 2009 at 9:22 AM, Artem Shiyanov wrote: > >> Hello! >> >> I'm trying to implement "barge in" functionality (see >> http://www.yourdictionary.com/telecom/barge-in) with "eavesdrop" but >> still with no success. >> >> The situation is: >> - Person A calls to the extension: >> >> >> >> >> >> >> >> - I bridge him with person B with help of mod_socket: >> SendMsg >> call-command: execute >> execute-app-name: bridge >> execute-app-arg: >> >> - A and B talks >> >> - Person C decides to barge in the call A<-->B (to become a third >> participator in the call) >> a) I send (mod_socket): >> SendMsg >> call-command: execute >> execute-app-name: eavesdrop >> execute-app-arg: >> >> b) Then, as the spec says ( >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop) >> I sent DTMF "3" with >> api uuid_send_dtmf 3 >> but it doesn't work. I mean that A can hear B and vice verse, but both A >> and B can't hear C. C also doesn't hear neither A nor B. >> >> If I press "3" on the C's softphone (latest X-Lite) then, really, C >> becomes a full-capabilities participator of the call. >> Instead of "uuid_send_dtmf" I tried: >> 1) >> sendevent DTMF >> Unique-ID: >> DTMF-Digit: 3 >> DTMF-Duration: 2000 >> >> 2) first make queue_dtmf for the , and then eavesdrop >> >> 3) >> SendMsg >> call-command: execute >> execute-app-name: gentones >> execute-app-arg: 3 >> >> 4) >> SendMsg >> call-command: execute >> execute-app-name: send_dtmf >> execute-app-arg: 3 >> >> And none of these methods leads to the "barged in" call. >> >> Anyone knows how to press "3" programmatically on behalf of the given >> channel with mod_socket?! >> >> >> Artem >> >> ___ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_miness...@hotmail.com > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:8...@conference.freeswitch.org > iax:gu...@conference.freeswitch.org/888 > googletalk:conf+...@conference.freeswitch.org > pstn:213-799-1400 > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] "barge in" implementation with mod_socket and eavesdrop
Hello! I'm trying to implement "barge in" functionality (see http://www.yourdictionary.com/telecom/barge-in) with "eavesdrop" but still with no success. The situation is: - Person A calls to the extension: - I bridge him with person B with help of mod_socket: SendMsg call-command: execute execute-app-name: bridge execute-app-arg: - A and B talks - Person C decides to barge in the call A<-->B (to become a third participator in the call) a) I send (mod_socket): SendMsg call-command: execute execute-app-name: eavesdrop execute-app-arg: b) Then, as the spec says ( http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop) I sent DTMF "3" with api uuid_send_dtmf 3 but it doesn't work. I mean that A can hear B and vice verse, but both A and B can't hear C. C also doesn't hear neither A nor B. If I press "3" on the C's softphone (latest X-Lite) then, really, C becomes a full-capabilities participator of the call. Instead of "uuid_send_dtmf" I tried: 1) sendevent DTMF Unique-ID: DTMF-Digit: 3 DTMF-Duration: 2000 2) first make queue_dtmf for the , and then eavesdrop 3) SendMsg call-command: execute execute-app-name: gentones execute-app-arg: 3 4) SendMsg call-command: execute execute-app-name: send_dtmf execute-app-arg: 3 And none of these methods leads to the "barged in" call. Anyone knows how to press "3" programmatically on behalf of the given channel with mod_socket?! Artem ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] problem: no audio for one of the person in conference
Hi there! The situation is: - Person A calls to the extension: - I bridge him with person B with help of mod_socket: SendMsg call-command: execute execute-app-name: bridge execute-app-arg: - A and B talks - Person C decides to barge in the call A<-->B (to become a third participator in the call) a) I send (mod_socket): api originate user/ &park() b) then I move A, B, C to the extension: conference profile "my_profile" is: The "moving" itself is done by sending this for each (A,B,C) channel SendMsg call-command: execute execute-app-name: execute_extension execute-app-arg: barge_in - Result: A, B, C are in the same conference with name "my_confname", A can hear B and vice verse, but both A and B can't hear C. C also doesn't hear neither A nor B. I also tried the "moving" to conference with api uuid_transfer -both barge_in api uuid_transfer barge_in but result is the same. Maybe someone already faced with such issue? Thanks, Artem ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] "mute" channel programmatically with mod_event_socket
Mathieu, have to confess- you are right! uuid_bridge works as expected. Usual saying - is didn't work last time I tried! Anyway, thank you much! Artem On Wed, Aug 19, 2009 at 5:02 PM, Mathieu Rene wrote: > Hi, > > Eavesdrop kind of works yeah, you can use the intercept application to > re-bridge the channels together, like: > > SendMsg > call-command: execute > execute-app-name: intercept > execute-app-arg: > > The same can be done with the uuid_bridge api. > > api uuid_bridge > > If you want the > cleaner way you could implement a media bug that replaces all the audio of > the channel by silence, but that'd require some C coding. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mr...@avgs.ca > > > > > On 19-Aug-09, at 7:13 AM, Artem Shiyanov wrote: > > The point is - a simple call flow is desired- if I have a ordinary 1-to-1 > call and one of the participators decides to mute call - I don't want to put > both channels into a conference room but it looks like I have no other > choices. > > BUT: > I found brilliant app - eavesdrop! If I do this for one-to-one call - mute > works! > SendMsg > call-command: execute > execute-app-name: eavesdrop > execute-app-arg: > > But the problem appears when I want to unmute.. the call! I've tried to > re-bridge channels, intercept them- nothing happens- one channel (muted one) > doesn't hear the participator. And CLI command 'show channels' shows that > channel with uui= still process eavesdrop app. > Maybe someone know how to switch off eavesdrop app? > > > Artem > > > On Wed, Aug 19, 2009 at 1:17 PM, Rudolf Denert wrote: > >> You can use the caller controlls in the conference.conf.xml to implement >> your own features something like mute or kick. Or do you want to mute mute >> other conference members like a moderator can do this. >> >> BR >> >> - Ursprüngliche Mail - >> Von: "Artem Shiyanov" >> An: freeswitch-users@lists.freeswitch.org >> Gesendet: Dienstag, 18. August 2009 14:54:04 GMT +01:00 >> Amsterdam/Berlin/Bern/Rom/Stockholm/Wien >> Betreff: [Freeswitch-users] "mute" channel programmatically with >> mod_event_socket >> >> >> Hello all! >> >> I'm trying to implement "mute" feature with mod_event_socket: I want >> programmatically mute/unmute some channel in a call.. And I don't see any >> other ways except to use conference room with special rule "mute". >> Anybody knows the better way? >> >> >> Thanks, >> Artem >> >> ___ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> ___ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] "mute" channel programmatically with mod_event_socket
The point is - a simple call flow is desired- if I have a ordinary 1-to-1 call and one of the participators decides to mute call - I don't want to put both channels into a conference room but it looks like I have no other choices. BUT: I found brilliant app - eavesdrop! If I do this for one-to-one call - mute works! SendMsg call-command: execute execute-app-name: eavesdrop execute-app-arg: But the problem appears when I want to unmute.. the call! I've tried to re-bridge channels, intercept them- nothing happens- one channel (muted one) doesn't hear the participator. And CLI command 'show channels' shows that channel with uui= still process eavesdrop app. Maybe someone know how to switch off eavesdrop app? Artem On Wed, Aug 19, 2009 at 1:17 PM, Rudolf Denert wrote: > You can use the caller controlls in the conference.conf.xml to implement > your own features something like mute or kick. Or do you want to mute mute > other conference members like a moderator can do this. > > BR > > - Ursprüngliche Mail - > Von: "Artem Shiyanov" > An: freeswitch-users@lists.freeswitch.org > Gesendet: Dienstag, 18. August 2009 14:54:04 GMT +01:00 > Amsterdam/Berlin/Bern/Rom/Stockholm/Wien > Betreff: [Freeswitch-users] "mute" channel programmatically with > mod_event_socket > > > Hello all! > > I'm trying to implement "mute" feature with mod_event_socket: I want > programmatically mute/unmute some channel in a call.. And I don't see any > other ways except to use conference room with special rule "mute". > Anybody knows the better way? > > > Thanks, > Artem > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] "mute" channel programmatically with mod_event_socket
Hello all! I'm trying to implement "mute" feature with mod_event_socket: I want programmatically mute/unmute some channel in a call.. And I don't see any other ways except to use conference room with special rule "mute". Anybody knows the better way? Thanks, Artem ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Can't mute SIP channel with "receiveonly" in SDP
Hy all! With Asterisk I can mute SIP channel using re-INVITE with "a=receiveonly" in media description. But this feature doesn't work with Freeswitch. For sure, there is old good method: transfer both legs to the conference room where one leg is able to listen/talk, the other one - only to listen, but this is unwanted workaround for me.. So I wonder: is there any other (preferably through the SIP) way to "mute" given SIP channel with Freeswitch? Thanks for all, Artem Shiyanov ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] one-way audio after playback+bridge
Update again: FS debug logs of the problematic part http://pastebin.freeswitch.org/pastebin.php?dl=9512 Artem On Mon, Jun 29, 2009 at 7:07 PM, Artem Shiyanov wrote: > I've tried with the snapshot (06.26.2009) - and situation had become even > worse - now both agents hear nothing.. > Maybe problem is in my sip_profiles? > Here they are: > http://pastebin.freeswitch.org/pastebin.php?dl=9510 > http://pastebin.freeswitch.org/pastebin.php?dl=9511 > > > > > On Fri, Jun 26, 2009 at 10:03 PM, Brian West wrote: > >> Can you try svn trunk... I know the remote FreeSWITCH isn't on trunk >> also... due to the lines below. >> >> /b >> >> On Jun 26, 2009, at 12:59 PM, Artem Shiyanov wrote: >> >> > o=FreeSWITCH 3898736844884231 105343281763004076 IN IP4 >> > 192.168.147.130 >> > s=FreeSWITCH >> >> >> ___ >> Freeswitch-users mailing list >> Freeswitch-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] one-way audio after playback+bridge
I've tried with the snapshot (06.26.2009) - and situation had become even worse - now both agents hear nothing.. Maybe problem is in my sip_profiles? Here they are: http://pastebin.freeswitch.org/pastebin.php?dl=9510 http://pastebin.freeswitch.org/pastebin.php?dl=9511 On Fri, Jun 26, 2009 at 10:03 PM, Brian West wrote: > Can you try svn trunk... I know the remote FreeSWITCH isn't on trunk > also... due to the lines below. > > /b > > On Jun 26, 2009, at 12:59 PM, Artem Shiyanov wrote: > > > o=FreeSWITCH 3898736844884231 105343281763004076 IN IP4 > > 192.168.147.130 > > s=FreeSWITCH > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] one-way audio after playback+bridge
Updates:
1. One-way audio is in 95% tries. But how the rest 5% works??
2. Strange FS logging after the channels are bridged (user A talk to user B)
2009-06-26 02:16:07 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel
sofia/external/1...@192.168.147.1 entering state [ready]
2009-06-26 02:16:07 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote
SDP:
v=0
o=FreeSWITCH 3898736844884231 105343281763004076 IN IP4 192.168.147.130
s=FreeSWITCH
c=IN IP4 192.168.147.130
t=0 0
m=audio 31134 RTP/AVP 0 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
m=video 0 RTP/AVP 34
a=rtpmap:34 H263/9
2009-06-26 02:16:07 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel
sofia/external/100...@192.168.147.130:5060 entering state [ready]
freeswi...@localhost.localdomain> 2009-06-26 02:17:09 [DEBUG] sofia.c:2728
sofia_handle_sip_i_state() Channel sofia/external/
1...@uat.pbx.starpoundtech.net entering state [calling]
2009-06-26 02:17:09 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel
sofia/external/1...@uat.pbx.starpoundtech.net entering state [ready]
2009-06-26 02:17:09 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote
SDP:
v=0
o=- 1 3 IN IP4 192.168.147.1
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.147.1
t=0 0
m=audio 47590 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
freeswi...@localhost.localdomain> show calls
API CALL [show(calls)] output:
created,created_epoch,function,caller_cid_name,caller_cid_num,caller_dest_num,caller_chan_name,caller_uuid,callee_cid_name,callee_cid_num,callee_dest_num,callee_chan_name,callee_uuid
2009-06-26
02:16:05,1245968165,switch_ivr_multi_threaded_bridge,1005,1005,inbound10,sofia/external/
1...@uat.pbx.starpoundtech.net
,4fa86434-b542-4066-99af-5924c78ddab7,1005,1005,
100...@192.168.147.130:5060,sofia/external/
100...@192.168.147.130:5060,73df8735-fee2-464d-aec0-fda886ba2cba
2009-06-26
02:16:07,1245968167,switch_ivr_multi_threaded_bridge,1005,1005,1001,sofia/external/
1...@192.168.147.1
,1c2c5f6d-669f-4432-ad04-35a64dbc8a14,1005,1005,sip:1...@192.168.147.1:5060
;fs_nat=yes,sofia/doublenat5090/sip:1...@192.168.147.1:5060
;fs_nat=yes,66895f68-70bf-410a-bff7-cda9549c102d
2 total.
freeswi...@localhost.localdomain> 2009-06-26 02:18:09 [DEBUG] sofia.c:2728
sofia_handle_sip_i_state() Channel sofia/external/
1...@uat.pbx.starpoundtech.net entering state [calling]
2009-06-26 02:18:10 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel
sofia/external/1...@uat.pbx.starpoundtech.net entering state [ready]
2009-06-26 02:18:10 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote
SDP:
v=0
o=- 1 3 IN IP4 192.168.147.1
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.147.1
t=0 0
m=audio 47590 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
2009-06-26 02:19:07 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel
sofia/external/1...@uat.pbx.starpoundtech.net entering state [calling]
2009-06-26 02:19:08 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel
sofia/external/1...@uat.pbx.starpoundtech.net entering state [ready]
2009-06-26 02:19:08 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote
SDP:
v=0
o=- 1 3 IN IP4 192.168.147.1
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.147.1
t=0 0
m=audio 47590 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Artem
On Fri, Jun 26, 2009 at 9:25 PM, Artem Shiyanov wrote:
> Hello!
>
> I got a problem with one way audio, symptoms are:
> firstly play audio file to channel A (A is hears sound)
> secondly bridge channel B with A (A doesn't hear B).
>
> Environment:
> - no NAT
> - User Agents being used X-Lite, EyeBeam, SJphone - same result for all of
> them- no audio, Wireshark shows that there is no RTP-flow to A from
> FreeSwitch
> - dialplan:
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>expression="^${caller_id_number}$">
>
> data="transfer_ringback=${us-ring}"/>
>
>
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED"/>
>
>
> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
> data="called_party_callgroup=${user_data(${dialed_extensi...@${domain_name}
> var callgroup)}"/>
> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>
>
> data="sip_h_X-SPFrom="e;${sip_from_user}"e;<${sip_from_uri}>"/>
> data="sip_h_X-SPTo=<${sip_to_uri}>&quo
[Freeswitch-users] one-way audio after playback+bridge
Hello! I got a problem with one way audio, symptoms are: firstly play audio file to channel A (A is hears sound) secondly bridge channel B with A (A doesn't hear B). Environment: - no NAT - User Agents being used X-Lite, EyeBeam, SJphone - same result for all of them- no audio, Wireshark shows that there is no RTP-flow to A from FreeSwitch - dialplan: - Call routing scheme: user calls to FS, FS calls to my B2BUA which manage call with SIP 3pcc Exact description what's going on is: user A -> FS -(bridge)-> my B2BUA Then my B2BUA transfers (using re-INVITE's), on behalf of user, call to extension "playback_media_file" . After a while B2BUA transfer (re-Inviting) user to extension "Local_Extension_from_SP". This should create a new call to user B. As a result - A doesn't hear B, but B- is OK. On the contrary, if a call is routed (by B2BUA) to the "Local_Extension_from_SP" extension (ommiting "playback_media_file" ext) - everything is OK. What I've tried: - set parameter "inbound-proxy-media" to "true" in Sofia profile - set parameter "disable_rtp_auto_adjust to "true" in Sofia profile Nothing helps. Any help or thoughts would be MUCH appreciated! Artem ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Problem: re-invite with 'inactive' SDP and 'bridge' function
I've checked again (thanks for your hint) and, really, FreeSwitch is so wise that he plays MOH for the holded person and and silence for the hold initiator! I'm going to is this work_flow for my B2BUA. Thanks! On Wed, May 27, 2009 at 8:47 PM, Brian West wrote: > Yes you won't hear anything if you press HOLD... the other caller you were > talking to will hear music. > /b > > On May 27, 2009, at 11:45 AM, Artem Shiyanov wrote: > > Probably mentioned sopftphones simply do not play incomming media when the > call is holded? > In general, is how should I hold a call using FS? > > > Brian West > br...@freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Problem: re-invite with 'inactive' SDP and 'bridge' function
I've just tried native "hold" X-Lite hold with FreeSwitch - it works. When you hold call, X-Lite (eyeBeam) sends re-INVITE with SDP v=0 o=- 8 3 IN IP4 172.16.0.6 s=CounterPath eyeBeam 1.5 c=IN IP4 0.0.0.0 t=0 0 m=audio 43362 RTP/AVP 3 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendonly SJPhone does the same way. I also tried to send the same - own SDP with c=IN IP4 0.0.0.0 a=sendonly but no luck. Probably mentioned sopftphones simply do not play incomming media when the call is holded? In general, is how should I hold a call using FS? On Wed, May 27, 2009 at 6:34 PM, Artem Shiyanov wrote: > Thanks for the answer! > Maybe you can advise me another scheme how to accomlish "mute" or "hold" > functionality? I just wonder if mute/hold in X-Lite works with FreeSwitch.. > I'll try and send the results. > > > > On Wed, May 27, 2009 at 6:22 PM, Brian West wrote: > >> Try not using RFC2543 HOLD since we do not support it. >> /b >> >> On May 27, 2009, at 7:38 AM, Artem Shiyanov wrote: >> >> Hi everyone! >> >> In my environment I use FreeSwitch as media server and session border >> controller. SIP routing is mostly done with my private B2BUA. The problem >> itself is in my "hold" functionality. In details: A is calling to B: >> >> >> Brian West >> br...@freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> ___ >> Freeswitch-users mailing list >> Freeswitch-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Problem: re-invite with 'inactive' SDP and 'bridge' function
Thanks for the answer! Maybe you can advise me another scheme how to accomlish "mute" or "hold" functionality? I just wonder if mute/hold in X-Lite works with FreeSwitch.. I'll try and send the results. On Wed, May 27, 2009 at 6:22 PM, Brian West wrote: > Try not using RFC2543 HOLD since we do not support it. > /b > > On May 27, 2009, at 7:38 AM, Artem Shiyanov wrote: > > Hi everyone! > > In my environment I use FreeSwitch as media server and session border > controller. SIP routing is mostly done with my private B2BUA. The problem > itself is in my "hold" functionality. In details: A is calling to B: > > > Brian West > br...@freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Problem: re-invite with 'inactive' SDP and 'bridge' function
Hi everyone! In my environment I use FreeSwitch as media server and session border controller. SIP routing is mostly done with my private B2BUA. The problem itself is in my "hold" functionality. In details: A is calling to B: Call being answered. Then B wants to hold a call (A should hear music-on-hold, B should hear nothing but still have a call). To accomplish this I: 1. on behalf of B2BUA make call to FreeSwitch "hold" extension, get SDP for music-on-hold (moh_sdp) 2. re-invite A with SDP moh_sdp 3. re-invite B with it's own sdp and attribute 'inactive' (also tried re-invite with A's SDP and 'inactive' and with 0.0.0.0 IP in SDP and 'inactive' - same results) Result: both A and B hear music-on-hold. I suspect that the reason is in 'bridge' function.. it seems, that it still bridges media against the SIP. Here is a snippet from FreeSwitch log related to "strange" (from my point of view) behaviour - when FreeSwitch change media for B to the incorrect one. Here 1004 is B. 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2939 sofia_glue_negotiate_sdp() Audio Codec Compare [GSM:3:8000:20]/[GSM:3:8000:20] 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:1863 sofia_glue_tech_set_codec() Already using GSM 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2899 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2908 sofia_glue_negotiate_sdp() Set comfort noise payload to 13 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2139 sofia_glue_activate_rtp() Audio params changed for sofia/external/6782886...@10.249.194.9:5060 from 75.101.166.29:27426 to 75.101.166.29:22922 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2146 sofia_glue_activate_rtp() AUDIO RTP [sofia/external/6782886...@10.249.194.9:5060] 10.249.194.9 port 28594 -> 75.101.166.29 port 22922 codec: 3 ms: 20 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2165 sofia_glue_activate_rtp() AUDIO RTP CHANGING DEST TO: [75.101.166.29:22922] 2009-05-26 13:19:14 [DEBUG] sofia.c:2996 sofia_handle_sip_i_state() Channel sofia/external/1...@10.249.194.9 entering state [ready][200] 2009-05-26 13:19:15 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() sofia/doublenat5090/sip:1...@172.16.0.6 ;fs_nat=yes;fs_path=sip%3A1004%40193.33.170.63%3A5060 Command Execute playback(local_stream://moh) EXECUTE sofia/doublenat5090/sip:1...@172.16.0.6 ;fs_nat=yes;fs_path=sip%3A1004%40193.33.170.63%3A5060 playback(local_stream://moh) 2009-05-26 13:19:15 [DEBUG] mod_local_stream.c:346 local_stream_file_open() Opening Stream [moh/8000] 8000hz 2009-05-26 13:19:15 [DEBUG] switch_ivr_play_say.c:1084 switch_ivr_play_file() Codec Activated l...@8000hz 1 channels 20ms 2009-05-26 13:19:15 [DEBUG] switch_core_io.c:649 switch_core_session_write_frame() sofia/doublenat5090/sip:1...@172.16.0.6 ;fs_nat=yes;fs_path=sip%3A1004%40193.33.170.63%3A5060 receive message [TRANSCODING_NECESSARY] Any thoughts would be really appreciated! Artem ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
