- FreeSWITCH-users - FreeSWITCH Mailing Lists

auth-all-packets
by Madovsky 22 Feb '11

22 Feb '11

Is auth-all-packets to true decrease performance ? Thanks

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OpenSER interface to FreeSWITCH
by Jason b.a 22 Feb '11

22 Feb '11

Hi, is it possible to implements is an interface to OpenSER over a TCP connection, so the design will be : Caller----IVR application----OpenSER-----Freeswitch | | Callee without the need to use the Sip sophia module, only openSER control the freeswitch for media processing using the socket interface. Thx

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javascript hanguphook
by Christian Hiller 22 Feb '11

22 Feb '11

Hello, i have a javscript that is called from xml-dialplan ... function on_hangup(hup_session, how) { console_log("info","got hungup"); exit(); } session.execute('bridge','sofia/internal....'); session.setHangupHook(on_hangup); ... Now i experience, that this function is only called if the session is not answered yet. Once its answered and then got hung up, the function on_hangup is not called anymore. Any ideas ? Kind regards Christian Hiller

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Re: [Freeswitch-users] OpenSER interface to FreeSWITCH
by Bernhard Suttner 22 Feb '11

22 Feb '11

currently not but might be a cool idea ... ----- Original Message ----- From: Jason b.a [mailto:k-b-24@live.com] To: freeswitch-users(a)lists.freeswitch.org Sent: Tue, 22 Feb 2011 20:35:55 +0100 Subject: [Freeswitch-users] OpenSER interface to FreeSWITCH > > Hi, > > is it possible to implements is an interface to OpenSER over a TCP > connection, so the design will be : > > Caller----IVR application----OpenSER-----Freeswitch > | > | > Callee > > without the need to use the Sip sophia module, only openSER control the > freeswitch for media processing using the socket interface. > > Thx

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MOH not working in G729 passthru with native sound files
by Marcin Wojtowicz 22 Feb '11

22 Feb '11

I'm trying to enable MOH when both legs of the call are using G729 (FS is in passthru). I converted an edited sample wave file to G729 and put in the appropriate folder, and FS loads it correctly because this is the message that keeps popping up in the console: 2011-02-09 21:10:25.609375 [INFO] mod_native_file.c:94 Opening File [C:\freeswitch\sounds/music/8000\sample.G729] 8000hz I establish a call, and everything is fine, but when I press hold on my handset I see an error message that says that G729 is only useable in passthru (here is the debug message): 2011-02-09 21:14:01.828125 [DEBUG] switch_channel.c:1376 (sofia/internal/ sip:M9jdt73ig0oOJSbt6Uyy@192.168.1.50:5060) Callstate Change ACTIVE -> HELD 2011-02-09 21:14:01.828125 [DEBUG] switch_core_session.c:954 Send signal sofia/external/MYHOME#(a)74.63.41.218 [BREAK] 2011-02-09 21:14:01.859375 [DEBUG] switch_core_session.c:709 Send signal sofia/external/MYHOME#(a)74.63.41.218 [BREAK] 2011-02-09 21:14:02.000000 [DEBUG] switch_ivr.c:563 sofia/external/MYHOME#@ 74.63.41.218 Command Execute playback(local_stream://moh/8000) EXECUTE sofia/external/MYHOME#@74.63.41.218playback(local_stream://moh/8000) 2011-02-09 21:14:02.000000 [DEBUG] mod_local_stream.c:421 Opening Stream [moh/8000] 8000hz 2011-02-09 21:14:02.000000 [DEBUG] switch_ivr_play_say.c:1244 Codec Activated L16@8000hz 1 channels 20ms 2011-02-09 21:14:02.015625 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode! 2011-02-09 21:14:02.015625 [ERR] switch_core_io.c:1042 Codec G.729 encoder error! 2011-02-09 21:14:02.015625 [DEBUG] switch_ivr_play_say.c:1581 done playing file I don't understand why that would be, since my music file is in G729 so I'm not asking freeswitch to convert, only stream. My custom ringback (before a call is established) works just fine using a similar method, so could anyone explain me why what I want to do is not permitted?

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ESL socket outbound: the execution of answer has not effect
by Stephen Wilde 22 Feb '11

22 Feb '11

I'm running an esl socket outbound application and I have a systematic problem: in a particular situation, the execution of answer on incoming call has no effect. This happens always when (and only in this case): 1. an incoming call is received (legA) and via dialplan I send the call to my socket outbound application 2. I make an outbound call with originate (legB) 3. I receive a CHANNEL_PROGRESS_MEDIA on legB 4. I do a "pre_answer" on legA 5, I do a "uuid_bridge" between legA and legB in this phase, legA can hear the progress tone coming from legB (an in-band ringback) 5. legB answer the call and I receive a CHANNEL_ANSWER on legB 6. I do an execute of "answer" on legA This "answer" has no effect! I do both pre_answer and answer specifying the UUID parameter (sendmsg <uuid>), obviously using legA uuid . I do the bridge specifying inbound uuid as first parameter and outbound uuid as second parameter: uuid_bridge <inbound_uuid> <outbound_uuid> In fs_cli I see: [WARNING] switch_ivr_bridge.c:1412 reversing order of channels so this will work! To me it seems that in uuid_bridge is done an inversion with my inbound/outbound channels and because I do an answer to the incoming uuid, probably (due to the inversion) is treated as outbound so the answer is ignored. Attached to this email the .pcap file related to tcp traffic between Freeswitch and my socket outbound application where at time 8.562639 there is my sendmsg with the answer. Stephen

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The call dropped prematurely error on txfax
by rex.alex 22 Feb '11

22 Feb '11

Hello, I am testing the FreeSWITCH mod_spandsp module. I have got a gateway registered in FreeSWITCH which supports fax and using the following command to send the fax, but It always ended up with the "result (49) The call dropped prematurely" error. Also if I remove fax_enable_t38 variables, it's giving "Reinvite Codec Error!". I have pastebin the entire log in http://pastebin.freeswitch.org/15394. originate {absolute_codec_string='PCMU,PCMA', fax_enable_t38=true,fax_enable_t38_request=true}sofia/gateway/Pulse/1XXXXXXXXXX &txfax(/var/www/html/faxtest/docs/15Feb2011103600.pdf.tiff) Please assist. Thanks, Rex -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/The-call-dropped-prematurely-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Updating callee info when using application intercept
by mayamatakeshi 22 Feb '11

22 Feb '11

Is there any way to update the callee number/name that would show up in a terminal that gets a dialplan to intercept a channel? I am trying with this: <action application='set' data='sip_callee_id_number=12341234'/> <action application='set' data='sip_callee_id_name=abcdabcd'/> <action application='send_display' data='fffff'/> <action application='answer'/> <action application='intercept' data='Some-UUID'/>" But I don't see any changes (I was expecting that it might change the username in the header To or cause FS to send some extra SIP message before or after "200 OK"). r, takeshi

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Enterprise bridge, groups and "follow me" dialplan
by Dmitry Bely 22 Feb '11

22 Feb '11

I am trying to implement the following bridging scenario: <action application="set" data="call_timeout=30"/> <action application="bridge" data="group/group1@domain,[leg_delay_start=10]group/group2@domain"/> ... i.e. all phones in group1 should ring for 10 seconds and if nobody answers group1+group2 rings for additional 20 seconds. That almost works, but only the first phone rings in either group: [WARNING] switch_ivr_originate.c:2339 Only calling the first element in the list in this mode. Surely I need enterprise bridge: <action application="bridge" data="group/group1@domain:_:[leg_delay_start=10]group/group2@domain"/> but now there is no correct caller id and no leg delay. Caller id can be restored with: <action application="bridge" data=" <origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number}> group/group1@domain:_:[leg_delay_start=10]group/group2@domain"/> (is this a bug or design decision?), but what to do with leg delay? [leg_delay_start=10] is simply ignored there. - Dmitry Bely

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How to create a C++ application (project), that can code FS C code function?
by Frankie Yiu 22 Feb '11

22 Feb '11

Hi there, I am new to FreeSwitch and I would like to implement a C++ application / mod that would do some analysis of RTP packets. I would like to ask what mod I should use as a template to create my C++ project so that I can access the core FreeSwitch APIs / functions, since they are in C language? I know under Endpoints, there are 2 MODs that are using C++ code (mod_h323 and mod_opal), but I don't know if I should implement the same way if I am creating an "application" instead of "endpoints". Please kindly let me know if there is any special / required function that I need to implemented in C++ code. Thanks in advance. Frankie

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