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bind meta session suddenly VERY slow response
by Lee JJ 29 Aug '08

29 Aug '08

Hello : I have 2 tested bind_meta_app , after some testing , I found the app either no response or very very slow. Got the [WARNING] already broadcasting 2008-08-29 16:52:22 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing 1529->1526@inter2 2008-08-29 16:52:22 [INFO] switch_ivr_async.c:1474 switch_ivr_bind_dtmf_meta_session() Bound A-Leg: 3 execute_extension::att_xfer XML inter2 2008-08-29 16:52:22 [INFO] switch_ivr_async.c:1474 switch_ivr_bind_dtmf_meta_session() Bound A-Leg: 4 javascript::calltest.js 2008-08-29 16:52:29 [WARNING] switch_ivr_async.c:1983 switch_ivr_broadcast() Channel [sofia/inter2/1529@210.243.126.72][execute_extension::att_xfer XML inter2] already broadcasting...broadcast aborted 2008-08-29 16:52:36 [WARNING] switch_ivr_async.c:1983 switch_ivr_broadcast() Channel [sofia/inter2/1529@210.243.126.72][javascript::calltest.js] already broadcasting...broadcast aborted At first suspecting the DTMF problem , change the rfc2833 and sip info back and forth ... but in vain . freeswitch@localhost> sofia status profile internal CODECS G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 check the sip profiles setting <param name="rfc2833-pt" value="101"/>  BTW, where is the in-band DTMF setting ? thanks !

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bind_meta_app for PARK
by Barray McKee 29 Aug '08

29 Aug '08

Hi, Is there anyway that I can set bind_meta_app for PARKed call? When a call is in PARK mode, I want to be able to intercept the DTMP sent so I can do things like: Press 1# to play weather report Press 2# to play music I want to let user to be able to do "things" while the call is on-hold waiting for another party to bridge to it. Can this be done? Regards, Barray

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Unable to execute Playback with event
by Doug Blacksone 29 Aug '08

29 Aug '08

Hi, I am trying to use event to trigger playback on a parked call and I am having difficulty getting it to work: I am not sure what is wrong with this one. Freeswitch responses with ERR: *SendMsg ae67060e-75e5-11dd-9228-efa2040a932a call-command: execute execute-app-name: playback execute-app-arg: /usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav lead-frames: 5 loops: 1* Content-Type: command/reply Reply-Text: -ERR invalid session id [ae67060e-75e5-11dd-9228-efa2040a932a ] If I just simply send in execute hangup, it is fine: *SendMsg ae67060e-75e5-11dd-9228-efa2040a932a call-command: execute execute-app-name: hangup *I am sure my UUID is right as I already checked with the following:* freeswitch@ser> show channels API CALL [show(channels)] output: uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate ae67060e-75e5-11dd-9228-efa2040a932a,2008-08-30 00:15:22,1220026522,sofia/internal/1005@192.168.1.104,CS_EXECUTE,1005,1005, 192.168.1.104,1,park,,XML,public,PCMU,8000,PCMU,8000 Content-Type: command/reply Reply-Text: +OK* Any idea what is wrong? Thanks, Doug

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request uri
by jocke eriksson 29 Aug '08

29 Aug '08

Hello! Freeswitch users i hava a simple question, how can i obtain a request parameter in my application, i use mod_java. Like this sip:name@address ;param=value Regards Jocke

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Newbie question
by Sunil Singh 29 Aug '08

29 Aug '08

Hi, I am a newbie in this list and I am finding openser a lot along with freeswitch and asterisk. First what is the use of openser with free switch/asterisk, is it required with free switch if yes in what kind of environment. It helps in scaling the solution or if natting than its not clear. The purpose is not clear. Thanks & Regards, Sunil

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Do we have talk_timeout?
by Adeel Ansari 29 Aug '08

29 Aug '08

Hi all, My concern is to terminate a call using a specific timeout value. I have referred to the docs, what I found is, *originate_timeout*, * progress_timeout*, and *call_timeout*. Do we have something like * talk_timeout*? or anything for similar purpose? Thanks. -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari

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Default voicemail file path
by Juan Backson 29 Aug '08

29 Aug '08

Hi, Can the default voicemail file path be configured to another location? Any hint on where to look would be appreciated. Thanks, JB

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directly mix 3 way voice
by Lee JJ 28 Aug '08

28 Aug '08

Hello : Is it possible to directly mix 3 way voice ? Not putting eny leg into holding music. I found the script originate a session is quite different than an agent call. Some channel attributes missing . # calltest.js n_sess = new Session() ; res = n_sess.originate(n_sess, "sofia/inter2/1527%210.243.126.72" ); uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate 56089ae3-134c-4bfc-a697-b08194918bf2,2008-08-27 13:35:33,1219815333,sofia/inter2/1527,CS_SOFT_EXECUTE,FreeSWITCH,0000000000,,1527,,,,,PCMU,8000,PCMU,8000 ps: Can someone kindly provide a "threeway" application sample . Thanks !!

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Loadbalance and failover
by Lee JJ 28 Aug '08

28 Aug '08

Hello : http://wiki.freeswitch.org/wiki/Enterprise_deployment_UltraMonkey I don't really test it . I do have some experience like yours , using openser plus some asterisks, but the openser dispatcher module even I use "hash over Call-Id" , I still get some fail experience , e.g. un-hold after hold . [openser , registrar , and dispatcher] ----> multiple [ Asterisks ] The Asterisk is rearded as a class 5 or VAS servers, and solve the NAT issue provided media relay. Perhaps you can share your expierence.

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Bridging
by Robert Clayton 28 Aug '08

28 Aug '08

All, I am attempting to allow a single line (1005) to be called which simultaneously allows multiple threads (lines 1006 and 1007) to apply simultaneous IVR. I would thing such grouping is common. Using the below it appears that line 1005 will not bridge to 1006 and 1007 since they are in the dialplan but not registered. Yet 1006 and 1007 can be called directly without being registered. Can this bridge be accomplished without them being registered? If such bridging is not allowed in the dialplan is there a better way? The two errors FS errors are: [ERR] switch_ivr_originate.c:951 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED <extension name="1005"> <condition field="destination_number" expression="^1005$"> <action application="bridge" data="sofia/internal/1006% 192.168.1.222,sofia/internal/1007%192.168.1.222"/> </condition> </extension> <extension name="1006"> <condition field="destination_number" expression="^1006$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="ivr" data="demo_ivr"/> </condition> </extension> <extension name="1007"> <condition field="destination_number" expression="^1007$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="ivr" data="demo_ivr"/> </condition> </extension> Thanks, Bob

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