December 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Bridging through gateway
by Jonas Gauffin 16 Dec '08

16 Dec '08

I'm trying to bridge using a non-registered gateway. And I get MANDATORY_IE_MISSING back. Why is that? 2008-12-15 11:06:49 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/u1000044@192.168.1.112:5070Execute bridge(sofia/default/ 0236661201(a)sip-corporate2.tele2.se) [.....] 2008-12-15 11:06:49 [DEBUG] sofia.c:2511 sofia_handle_sip_i_state() Channel sofia/internal/0236661201(a)sip-corporate2.tele2.se entering state [calling] 2008-12-15 11:06:49 [ERR] sofia_reg.c:1312 sofia_reg_handle_sip_r_challenge() No Matching gateway found 2008-12-15 11:06:49 [NOTICE] sofia_reg.c:1333 sofia_reg_handle_sip_r_challenge() Hangup sofia/internal/ 0236661201(a)sip-corporate2.tele2.se [CS_CONSUME_MEDIA] [MANDATORY_IE_MISSING] 2008-12-15 11:06:49 [DEBUG] switch_channel.c:1478 switch_channel_perform_hangup() Send signal sofia/internal/ 0236661201(a)sip-corporate2.tele2.se [KILL]

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running custom script with bind_meta_app
by Stephen Crosby 16 Dec '08

16 Dec '08

I'm Stephen Crosby, and I've just started working with freeswitch. It's been great so far. I want to run a custom script inside a conference when a DTMF sequence is entered. I found bind_meta_app and thought it would be perfect, but I can't seem to get it to work. When I dial-in and press *8, I get no debugging output at all. When I press another sequence like *9 for instance, I get: [WARNING] switch_ivr_async.c:1429 meta_on_dtmf() sofia/external/5593495805(a)sip.gafachi.com Ignoring meta digit '9' not mapped. The script I wrote has been tested with "jsrun script.js" from the command line and it does work. I've got the debugging level all the way up and there's just not much for me to go on. Any help would be greatly appreciated. <extension name="gafachi-DID1"> <condition field="destination_number" expression="^xxxxxxxxxxx$"> <action application="answer"/> <action application="bind_meta_app" data="8 ab s javascript::$${base_dir}/scripts/script.js"/> <action application="conference" data="3000@default"/> </condition> </extension> --Stephen

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Building a web config/billing gui
by ccav 16 Dec '08

16 Dec '08

For anyone interested, I'm in the process of building a web based config/billing gui. I could use some help though. Anyone who has php experience and knows the xml_curl interface pretty well and some spare time to do some development would be a useful partner. Also, if there's a resource online that defines all the params/variables and subobjects for each of the fs object types, like directories, dialplans etc, it would help me fill my parm database a lot faster. For anyone interested in monitoring the development, it's at www.sparkz.tv/smfs login is demo:demo. Feedback on features and development direction is invited. I want to get this done quickly. I think FS is awesome but no config/billing interface is going to stand in the way of it's adoption so I'm burning the midnight oil to get this done. -- View this message in context: http://www.nabble.com/Building-a-web-config-billing-gui-tp21027515p21027515… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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(no subject)
by martin joseph 15 Dec '08

15 Dec '08

I am wondering if anyone running OSX 10.4.11 is able to successfully build FS Trunk? If so, what versions does build.sh report? Please? I get the following and have issues making it... Updated to revision 10753. mail:/usr/src/freeswitch/trunk root# ./bootstrap.sh bootstrap: checking installation... bootstrap: autoconf version 2.61 (ok) bootstrap: automake version 1.10 (ok) bootstrap: libtool version 1.5.24 (ok) Thanks for the info, Marty

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Where FreeSWITCH writes some files
by Andy Spitzer 15 Dec '08

15 Dec '08

Woof! It appears that FreeSWITCH writes freeswitch.history freeswitch.log freeswitch.pid freeswitch.xml.fsxml to the -log directory. Is there a way to put the files other than freeswitch.log into the -db directory instead? In my environment we archive and rotate everything in the log directory (which includes logs beside FreeSWITCH's), and these other FreeSWITCH files are getting rotated. Yeah, I can explicitly exclude them, but to me it seems those really belong in the -db directory anyway, as they are inherently data needed for the current executable of FreeSWITCH, and not logs. --Woof!

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how to handle returned sip 302 dialplan
by Chav Paskov 15 Dec '08

15 Dec '08

Good morning everybody, I would like to know what would be the best way to process the info in returned back sip 302 message. Let me be a bit more specific. i'm sending INVITE as pasted below to z.z.z.z *INVITE* sip:17402740539@z.z.z.z:5000 SIP/2.0^M Via: SIP/2.0/UDP z.z.z.z:5000;branch=z9hG4bK215b.e6ae7e74.0^M Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bKaa6257694041475e0659eeb0019a0bbc^M *From: "Bob Caller" <sip:6144102508@y.y.y.y:5060>;tag=CCC7.1F37^M To: <sip:17402740539@x.x.x.x:5060>^M *Call-ID: 15821337-3367080415-112396(a)msc-01.voice.example.net^M CSeq: 1 INVITE^M Contact: <sip:6144102508@y.y.y.y:5060>^M Content-Type: application/sdp^M Max-forwards: 69^M Session-expires: 3600;Refresher=uac^M Supported: timer^M Content-Length: 200^M ^M 1192 v=0^M o=iserver 16123 16124 IN IP4 y.y.y.y^M s=sip call^M c=IN IP4 64.112.188.81^M t=0 0^M m=audio 55824 RTP/AVP 18 0 101^M a=sendrecv^M a=fmtp:101 0-15^M a=rtpmap:101 telephone-event/8000^M a=ptime:20^M the reply from z.z.z.z back is sip 302 SIP/2.0 *302 Moved Temporarily*^M Via: SIP/2.0/UDP z.z.z.z:5000;branch=z9hG4bK215b.e6ae7e74.0^M Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bKaa6257694041475e0659eeb0019a0bbc^M From: "Bob Caller" <sip:6144102508@y.y.y.y:5060>;tag=CCC7.1F37^M To: <sip:17402740539@x.x.x.x:5060>;tag=1087415105^M Call-ID: 15821337-3367080415-112396(a)msc-01.voice.example.net^M CSeq: 1 INVITE^M *Contact: <sip:2145551234@64.112.188.84>;npdi^M *User-Agent: eXosip/3.1.0^M Content-Length: my question is: Is it possible to send the call to z.z.z.z , receive the SIP 302 , process the data in Contact field and redirect to the new destination contained in *Contact: <sip:2145551234@64.112.188.84>;npdi^M *without closing the session. i red something about <action application="set" data="continue_on_fail=true"/> but i'm not sure how to use it. Any ideas on this matter will be highly appreciated. Best Regards Chav

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Newbie Questions
by Saeed Ahmed 15 Dec '08

15 Dec '08

Hi, I am very new Freeswitch. Till now I've some experience with Asterisk. Can someone explain me the following things: 1. Can I connect my TDM switch to Freeswitch? My switch can speak Dss1/ss7, SS7 is more preferable 2. Is freeswitch similar to nextone SBC? (http://www.nextpointnetworks.com/) 3. Does freeswtich support codec translation? These are very basic questions at start. When I'll go deeper into it then there could be more questions. Kind Regards Saeed

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Recording Pause
by Robert Clayton 15 Dec '08

15 Dec '08

All, I was thinking since there is no direct functionality for the person being recorded to pause the recording could this be done indirectly. For example if using the functionality to record only when voice is present could the audio stream be preprocessed to evaluate the dtmf where if a specific key was pressed the audio stream would no longer be passed or passed as blank to the recording functionality so an implicit pause would be created. And when the person recording wished to continue pressing a key would signal FS to pass the audio again therefore reestablishing recording to the same file? Bob

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Re: [Freeswitch-users] Multiple mod_portaudio channels
by Giovanni Maruzzelli 15 Dec '08

15 Dec '08

You can try making the lynx appear to Operating System like 8 single soundcards (mono 1in 1out). You can do this on Linux via the /etc/asound.conf Then for each soundcard you created, you can start one portaudio channel Sincerely, Giovanni Maruzzelli ========================================= Company : Celliax Website: www.celliax.org Address : via Pierlombardo 9, 20135 Milano Country/Territory : Italy Business Email: gmaruzz at celliax dot org Cell : 39-347-2665618 Fax : 39-02-87390039 On Mon, Dec 15, 2008 at 11:56 AM, Clifford <freeswitch(a)ptmm.com> wrote: > I need to set up a system that would allow multiple mod_portaudio channels > to be configured. > > The calls would come in over SIP and any calls would be assigned a specific > port on the multi-port sound card. > > I would probably set it up so specific extensions go to specific ports, for > example 1001 would be port 1 and 1008 would be port 8 on the sound card. > > The documentation on the mod_portaudio module is very slim. In fact I would > say it is nearly undocumented. Is it capable of handling multiple audio > device ports (such as an 8-port LYNX sound card)? > > If so how would it be interfaced/configured? > > Thanks, > > Clifford > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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Interrupting read application with DTMF
by Jan Kubr 15 Dec '08

15 Dec '08

Hi, I have been having some troubles with the read application for quite a while which I haven't been able to solve yet. I have Freeswitch connected to a SIP gateway to accept calls from a landline-like number. For the incoming calls I have a simple testing dialplan: <action application="read" data="1 1 en/us/callie/conference/8000/conf-pin.wav res 10000 #"/> <action application="phrase" data="spell,${res}"/> The behavior I have a problem with is that the read app should terminate when I press a digit and the execution should jump to the next action - meaning the playback of the file should be interrupted. The problem is that when I call the public number from my cell phone this works only about 50% of the time. In the other cases I need to wait for the wav file to be played (or press the digit two or three times). When using a SIP phone it always works. Today I tried to convert the wav file the read app plays to the GSM format and found out it fixed the problem! Now I can almost always interrupt the read app with DTMF from my cell phone. Doing the same from my SIP phone doesn't work well though when the file is GSM. Can someone explain me what is going on here and what is the right approach? I'm on revision 10751. I've tried to set a few configuration variables based on suggestions from this list, but it didn't make any difference. Thanks, Jan Kubr

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