- FreeSWITCH-users - FreeSWITCH Mailing Lists

IuUP support question
by Anya 11 Apr '08

11 Apr '08

Hi all, can someone tell me if this project supports IuUP interface? I am mainly interested in support for Iu-CS. I am looking for a small MSC that supports A and IuUP interfaces for 2g & 3g networks respectively. thank you, Anna

2 4

Dingaling In Component Mode "How To Call Jingle To SIP"
by Ali Jawad 11 Apr '08

11 Apr '08

Dear All First I would like to state that dingaling is working in client mode with gtalk perfectly. I can call from gtalk to my sip extension "1000". I have setup component mode now I am using FS with Openfire, open does show the following: 1 fs.jabber2.mydomain.net LibDingaLing gateway However stats on the jabber server are Session Statistics: Packets Received/Sent: 1/1 So the component is REGISTERED, so far so good my problem now is how to initiate a call from XMPP to a SIP extension. Using client mode I would do the following: 1. Setup GTALK account and assign extension to it 2. Login with another gtalk account and dial the first gtalk account 3. The assigned extension rings I don't know how to do the same using component mode. Adding sip+1000@mydomain to the jabber client "Jabbin" does not popup an "add contact" on XTEN. Anything I missed here ? Thanks for any hints or suggestions The Freeswitch server logs are below: 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND: ------------------------------------------------------------------------ ------- <stream:stream xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:component:accept" to="fs" version="1.0"></stream:stream> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV: ------------------------------------------------------------------------ ------- <stream:stream xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:component:accept" from="fs.jabber2.MyDomain" id="69fcaa7b"></stream:stream> 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND: ------------------------------------------------------------------------ ------- <handshake>72dac833190326bb82dddedae558b2b794df5e1b</handshake> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV: ------------------------------------------------------------------------ ------- <handshake></handshake> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV: ------------------------------------------------------------------------ ------- <iq type="get" id="970-504" from="component.jabber2.MyDomain" to="fs.jabber2.MyDomain"> <query xmlns="http://jabber.org/protocol/disco#info"></query> </iq> 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND: ------------------------------------------------------------------------ ------- <iq from="fs" to="component.jabber2.MyDomain" id="970-504" type="result"> <query xmlns="http://jabber.org/protocol/disco#info"> <identity category="gateway" name="LibDingaLing"></identity> <feature var="http://jabber.org/protocol/disco"></feature> <feature var="http://jabber.org/protocol/disco#info"></feature> <feature var="jabber:iq:version"></feature> <feature var="vcard-temp"></feature> <feature var="http://www.google.com/xmpp/protocol/voice/v1"></feature> <feature var="jabber:iq:last"></feature> </query> </iq>

1 0

Freeswitch and Openser
by Pete Kay 11 Apr '08

11 Apr '08

Hi all, I am currently evaluating either using Freeswitch or Openser for my SIP server. I am trying to look for information that can tell me the difference between the two in terms of ease-of-maintain, ease-of-implement, and scalability. If you have such kind of information, would you please share with me? Thank you very much for your input. Best Regards, Pete

4 4

Incoming fax problems
by Josip Djuricic 11 Apr '08

11 Apr '08

Hi there I wanted to ask for advice on my incoming fax receive problem. When I send the fax to a Patton smartnode FXS gateway behind FS (and the call goes PSTN < -DSS1- > AS5300 <-h323> Yate <-sip-> FS <-sip-> PATTON). I needed to add yate to act as h323 to sip converter because my AS5300 had problems with sip (not only to FS, but to almost every device, seams like a sw issue). So now my fax behind FS answers, but the call fails after a while. What I've noticed on FS output, though not sure if it's related is: [DEBUG] switch_ivr_bridge.c:275 audio_bridge_thread() read: sofia/default/38514001702 Bad Frame.... Bubye! I'm including entire FS debug output of that call. I need to also notice that outgoing fax sending works just perfect. Thank you all, Josip #irc jdjurici 2008-04-11 13:46:46 [DEBUG] sofia.c:1595 sofia_handle_sip_i_state() Channel sofia/default/38514001702 entering state [ready] 2008-04-11 13:46:46 [DEBUG] sofia.c:1599 sofia_handle_sip_i_state() Remote SDP: v=0 o=MxSIP 0 81 IN IP4 89.18.37.100 s=SIP Call c=IN IP4 89.18.37.100 t=0 0 m=audio 5024 RTP/AVP 8 a=rtpmap:8 PCMA/8000 2008-04-11 13:46:46 [DEBUG] sofia_glue.c:2011 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMA:8:8000]/[PCMA:8:8000] 2008-04-11 13:46:46 [DEBUG] sofia_glue.c:1333 sofia_glue_tech_set_codec() Set Codec sofia/default/38514001702 PCMA/8000 20 ms 160 samples 2008-04-11 13:46:46 [DEBUG] sofia_glue.c:1523 sofia_glue_activate_rtp() AUDIO RTP [sofia/default/38514001702] 89.18.32.19:25440->89.18.37.100:5024 codec: 8 ms: 20 2008-04-11 13:46:46 [DEBUG] switch_rtp.c:716 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms 2008-04-11 13:46:46 [DEBUG] switch_channel.c:1506 switch_channel_perform_mark_answered() Kill sofia/outbound/38516184904(a)10.0.3.220 [BREAK] 2008-04-11 13:46:46 [NOTICE] sofia.c:1945 sofia_handle_sip_i_state() Channel [sofia/default/38514001702] has been answered 2008-04-11 13:46:46 [DEBUG] sofia_glue.c:1523 sofia_glue_activate_rtp() AUDIO RTP [sofia/outbound/38516184904(a)10.0.3.220] 89.18.32.19:29046->10.0.3.220:17396 codec: 8 ms: 20 2008-04-11 13:46:46 [DEBUG] switch_rtp.c:716 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms 2008-04-11 13:46:46 [DEBUG] mod_sofia.c:376 sofia_answer_channel() Local SDP sofia/outbound/38516184904(a)10.0.3.220: v=0 o=FreeSWITCH 1207885360 1207885361 IN IP4 89.18.32.19 s=FreeSWITCH c=IN IP4 89.18.32.19 t=0 0 a=sendrecv m=audio 29046 RTP/AVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 2008-04-11 13:46:46 [DEBUG] switch_core_session.c:383 switch_core_session_receive_message() Kill sofia/outbound/38516184904(a)10.0.3.220 [BREAK] 2008-04-11 13:46:46 [NOTICE] switch_ivr_originate.c:1171 switch_ivr_originate() Channel [sofia/outbound/38516184904(a)10.0.3.220] has been answered 2008-04-11 13:46:46 [DEBUG] switch_ivr_originate.c:1205 switch_ivr_originate() Originate Resulted in Success: [sofia/default/38514001702] 2008-04-11 13:46:46 [DEBUG] switch_core_session.c:383 switch_core_session_receive_message() Kill sofia/default/38514001702 [BREAK] 2008-04-11 13:46:46 [DEBUG] switch_core_session.c:383 switch_core_session_receive_message() Kill sofia/outbound/38516184904(a)10.0.3.220 [BREAK] 2008-04-11 13:46:46 [DEBUG] switch_ivr_bridge.c:756 switch_ivr_multi_threaded_bridge() sofia/default/38514001702 State Change CS_HOLD -> CS_LOOPBACK 2008-04-11 13:46:46 [DEBUG] switch_core_session.c:670 switch_core_session_signal_state_change() Kill sofia/default/38514001702 [BREAK] send 897 bytes to udp/[10.0.3.220]:5090 at 11:46:46.446911: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.3.220:5090;rport=5090;branch=z9hG4bK293463607 From: "10.0.3.10" <sip:38516184904@10.0.3.220>;tag=1005463975 To: <sip:38514001702@89.18.32.19:5080>;tag=2Sgt80aptejjg Call-ID: 1414260367(a)10.0.3.220 CSeq: 19 INVITE Contact: <sip:mod_sofia@89.18.32.19:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.pre4-8058 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition, timer Allow-Events: talk Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 272 v=0 o=FreeSWITCH 6013977101500251822 7042482777030961138 IN IP4 89.18.32.19 s=FreeSWITCH c=IN IP4 89.18.32.19 t=0 0 a=sendrecv m=audio 29046 RTP/AVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 ------------------------------------------------------------------------ 2008-04-11 13:46:46 [DEBUG] sofia.c:1595 sofia_handle_sip_i_state() Channel sofia/outbound/38516184904(a)10.0.3.220 entering state [completed] recv 404 bytes from udp/[10.0.3.220]:5090 at 11:46:46.459555: ------------------------------------------------------------------------ ACK sip:mod_sofia@89.18.32.19:5080;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.3.220:5090;rport;branch=z9hG4bK1807442927 From: "10.0.3.10" <sip:38516184904@10.0.3.220>;tag=1005463975 To: <sip:38514001702@89.18.32.19:5080>;tag=2Sgt80aptejjg Call-ID: 1414260367(a)10.0.3.220 CSeq: 19 ACK Max-Forwards: 20 Contact: <sip:38516184904@10.0.3.220:5090> User-Agent: sip-to-h323/0.9.0 Content-Length: 0 ------------------------------------------------------------------------ 2008-04-11 13:46:46 [DEBUG] sofia.c:1595 sofia_handle_sip_i_state() Channel sofia/outbound/38516184904(a)10.0.3.220 entering state [ready] 2008-04-11 13:46:46 [DEBUG] switch_core_state_machine.c:440 switch_core_session_run() (sofia/default/38514001702) State HOLD going to sleep 2008-04-11 13:46:46 [DEBUG] switch_core_state_machine.c:364 switch_core_session_run() sofia/default/38514001702 Running State Change CS_LOOPBACK 2008-04-11 13:46:46 [DEBUG] switch_core_state_machine.c:431 switch_core_session_run() (sofia/default/38514001702) State LOOPBACK 2008-04-11 13:46:46 [DEBUG] mod_sofia.c:314 sofia_on_loopback() SOFIA LOOPBACK freeswitch@softsw00> freeswitch@softsw00> recv 440 bytes from udp/[89.18.37.100]:5062 at 11:47:25.493537: ------------------------------------------------------------------------ BYE sip:mod_sofia@89.18.32.19:5060 SIP/2.0 Via: SIP/2.0/UDP 89.18.37.100:5062;branch=z9hG4bKc22903e25 Max-Forwards: 70 Content-Length: 0 To: "10.0.3.10" <sip:38516184904@89.18.32.19>;tag=KtFgg4Z384S5j From: <sip:38514001702@89.18.37.100:5062>;tag=01ec8a090df17a1 Call-ID: cc981044-825f-122b-1ca9-001e0b83659c CSeq: 802101929 BYE Supported: timer Supported: replaces User-Agent: Patton SN4112 JS EUI MxSF v3.2.8.45 00A0BA02B814 ------------------------------------------------------------------------ send 498 bytes to udp/[89.18.37.100]:5062 at 11:47:25.494066: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 89.18.37.100:5062;branch=z9hG4bKc22903e25 From: <sip:38514001702@89.18.37.100:5062>;tag=01ec8a090df17a1 To: "10.0.3.10" <sip:38516184904@89.18.32.19>;tag=KtFgg4Z384S5j Call-ID: cc981044-825f-122b-1ca9-001e0b83659c CSeq: 802101929 BYE User-Agent: FreeSWITCH-mod_sofia/1.0.pre4-8058 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, precondition, timer Content-Length: 0 ------------------------------------------------------------------------ 2008-04-11 13:47:25 [DEBUG] sofia.c:1595 sofia_handle_sip_i_state() Channel sofia/default/38514001702 entering state [terminated] 2008-04-11 13:47:25 [NOTICE] sofia.c:1992 sofia_handle_sip_i_state() Hangup sofia/default/38514001702 [CS_LOOPBACK] [NORMAL_CLEARING] 2008-04-11 13:47:25 [DEBUG] switch_channel.c:1355 switch_channel_perform_hangup() Kill sofia/default/38514001702 [KILL] 2008-04-11 13:47:25 [DEBUG] switch_core_session.c:670 switch_core_session_signal_state_change() Kill sofia/default/38514001702 [BREAK] 2008-04-11 13:47:25 [DEBUG] switch_ivr_bridge.c:275 audio_bridge_thread() read: sofia/default/38514001702 Bad Frame.... Bubye! 2008-04-11 13:47:25 [DEBUG] switch_ivr_bridge.c:344 audio_bridge_thread() Kill sofia/outbound/38516184904(a)10.0.3.220 [BREAK] 2008-04-11 13:47:25 [DEBUG] switch_ivr_bridge.c:347 audio_bridge_thread() BRIDGE THREAD DONE [sofia/default/38514001702] 2008-04-11 13:47:25 [DEBUG] switch_core_state_machine.c:431 switch_core_session_run() (sofia/default/38514001702) State LOOPBACK going to sleep 2008-04-11 13:47:25 [DEBUG] switch_core_state_machine.c:364 switch_core_session_run() sofia/default/38514001702 Running State Change CS_HANGUP 2008-04-11 13:47:25 [DEBUG] switch_core_state_machine.c:392 switch_core_session_run() (sofia/default/38514001702) State HANGUP 2008-04-11 13:47:25 [DEBUG] mod_sofia.c:255 sofia_on_hangup() Channel sofia/default/38514001702 hanging up, cause: NORMAL_CLEARING 2008-04-11 13:47:25 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() Standard HANGUP sofia/default/38514001702, cause: NORMAL_CLEARING 2008-04-11 13:47:25 [DEBUG] switch_core_state_machine.c:392 switch_core_session_run() (sofia/default/38514001702) State HANGUP going to sleep 2008-04-11 13:47:25 [DEBUG] switch_core_session.c:730 switch_core_session_thread() Session 145 (sofia/default/38514001702) Locked, Waiting on external entities 2008-04-11 13:47:25 [DEBUG] switch_core_session.c:383 switch_core_session_receive_message() Kill sofia/outbound/38516184904(a)10.0.3.220 [BREAK] 2008-04-11 13:47:25 [DEBUG] switch_ivr_bridge.c:344 audio_bridge_thread() Kill sofia/default/38514001702 [BREAK] 2008-04-11 13:47:25 [DEBUG] switch_ivr_bridge.c:347 audio_bridge_thread() BRIDGE THREAD DONE [sofia/outbound/38516184904(a)10.0.3.220] 2008-04-11 13:47:25 [NOTICE] switch_core_state_machine.c:156 switch_core_standard_on_execute() Hangup sofia/outbound/38516184904(a)10.0.3.220 [CS_EXECUTE] [NORMAL_CLEARING] 2008-04-11 13:47:25 [DEBUG] switch_channel.c:1355 switch_channel_perform_hangup() Kill sofia/outbound/38516184904(a)10.0.3.220 [KILL] 2008-04-11 13:47:25 [DEBUG] switch_core_session.c:670 switch_core_session_signal_state_change() Kill sofia/outbound/38516184904(a)10.0.3.220 [BREAK] 2008-04-11 13:47:25 [DEBUG] switch_core_state_machine.c:428 switch_core_session_run() (sofia/outbound/38516184904(a)10.0.3.220) State EXECUTE going to sleep 2008-04-11 13:47:25 [DEBUG] switch_core_state_machine.c:364 switch_core_session_run() sofia/outbound/38516184904(a)10.0.3.220 Running State Change CS_HANGUP 2008-04-11 13:47:25 [DEBUG] switch_core_state_machine.c:392 switch_core_session_run() (sofia/outbound/38516184904(a)10.0.3.220) State HANGUP send 626 bytes to udp/[10.0.3.220]:5090 at 11:47:25.500778: 2008-04-11 13:47:25 [DEBUG] mod_sofia.c:255 sofia_on_hangup() Channel sofia/outbound/38516184904(a)10.0.3.220 hanging up, cause: NORMAL_CLEARING BYE sip:38516184904@10.0.3.220:5090 SIP/2.0 Via: SIP/2.0/UDP 89.18.32.19:5080;rport;branch=z9hG4bKK5Zr11t0jaS2m Max-Forwards: 70 From: <sip:38514001702@89.18.32.19:5080>;tag=2Sgt80aptejjg To: "10.0.3.10" <sip:38516184904@10.0.3.220>;tag=1005463975 Call-ID: 1414260367(a)10.0.3.220 CSeq: 97838886 BYE Contact: <sip:mod_sofia@89.18.32.19:5080;transport=udp>2008-04-11 13:47:25 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Sending BYE to sofia/outbound/38516184904(a)10.0.3.220 2008-04-11 13:47:25 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() Standard HANGUP sofia/outbound/38516184904(a)10.0.3.220, cause: NORMAL_CLEARING User-Agent: FreeSWITCH-mod_sofia/1.0.pre4-8058 2008-04-11 13:47:25 [DEBUG] switch_core_state_machine.c:392 switch_core_session_run() (sofia/outbound/38516184904(a)10.0.3.220) State HANGUP going to sleep Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO 2008-04-11 13:47:25 [DEBUG] switch_core_session.c:730 switch_core_session_thread() Session 144 (sofia/outbound/38516184904(a)10.0.3.220) Locked, Waiting on external entities Supported: 100rel, precondition, timer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 ------------------------------------------------------------------------ 2008-04-11 13:47:25 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 144 (sofia/outbound/38516184904(a)10.0.3.220) Ended 2008-04-11 13:47:25 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel sofia/outbound/38516184904(a)10.0.3.220 [CS_HANGUP] 2008-04-11 13:47:25 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 145 (sofia/default/38514001702) Ended 2008-04-11 13:47:25 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel sofia/default/38514001702 [CS_HANGUP] recv 336 bytes from udp/[10.0.3.220]:5090 at 11:47:25.511122: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 89.18.32.19:5080;rport=5080;branch=z9hG4bKK5Zr11t0jaS2m;received=89.18.32.19 From: <sip:38514001702@89.18.32.19:5080>;tag=2Sgt80aptejjg To: "10.0.3.10" <sip:38516184904@10.0.3.220>;tag=1005463975 Call-ID: 1414260367(a)10.0.3.220 CSeq: 97838886 BYE Server: sip-to-h323/0.9.0 Content-Length: 0 ------------------------------------------------------------------------ recv 390 bytes from udp/[10.0.3.220]:5090 at 11:47:25.530548: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 89.18.32.19:5080;rport=5080;branch=z9hG4bKK5Zr11t0jaS2m;received=89.18.32.19 Call-ID: 1414260367(a)10.0.3.220 CSeq: 97838886 BYE Server: sip-to-h323/0.9.0 Allow: ACK, INVITE, BYE, CANCEL, REGISTER, OPTIONS, INFO Content-Length: 0 ------------------------------------------------------------------------

3 3

Re: [Freeswitch-users] [Freeswitch-dev] Ideas for lecture
by Arnaldo de Moraes Pereira 11 Apr '08

11 Apr '08

I'm also sending this message to freeswitch-users' list. On Thu, Apr 10, 2008 at 8:35 PM, Anthony Minessale <anthony.minessale(a)gmail.com> wrote: > sure let us know what you need. Despite FreeSWITCH didn't reach 1.0 yet, would be nice to list some cases of success. I've heard "We've never heard of FreeSWITCH, is anyone using it on production ? Who ?" sometimes, as the main argument against the software adoption. So, does anyone wants to share their deployment experiences ? Maybe it's a good idea to elaborate on http://wiki.freeswitch.org/wiki/Enterprise_deployment, or just create another wiki page for something like that... > > Maybe you can do the same presentation at ClueCon this year > http://www.cluecon.com Most likely I won't be able to be there. :( > > > > > On Thu, Apr 10, 2008 at 6:06 PM, Arnaldo de Moraes Pereira > <egghunt(a)gmail.com> wrote: > > > > > > > > > Hello, > > > > I'll give a lecture, focusing on FreeSWITCH, as well as comparing it > > with other open source pbxes/switches/proxies. The talk will have a > > brief overall of the available software choices for the most common > > cases, but the main goal is to spread the word and show what this > > project is capable of. > > > > That'll happen in "Faculdade Impacta de Tecnologia" (Impacta college > > of Tecnology), which exports technology and teaching methodologies to > > some countries, etc. The contents of the talk aren't defined yet, I > > just told them the idea and it was very well welcomed. I would like to > > show the developers' perspective, as well as some real use cases, > > that's why I ask you guys to say whatever you think it's worth about > > those topics. > > > > There will be live examples of applications, I'd also be grateful > > receiving ideas for that. > > > > Thanks, > > Arnaldo > > > > _______________________________________________ > > Freeswitch-dev mailing list > > Freeswitch-dev(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-dev mailing list > Freeswitch-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > >

2 1

Execute perl file from javascript
by Leonardo Alves 11 Apr '08

11 Apr '08

Is there anyway to execute a perl file from a javascript file. What I am doing is this. I am making a originate command from a perl file passing a javascript file to execute when the call is answered. Like this: $fs->sendmsg({'command' => 'bgapi originate {ignore_early_media}sofia/gateway/asterisk/9192117290 &javascript(test.js)'}); What I want is from this javascript file execute a .pl file. Thanks Leonardo Alves

1 0

Re: [Freeswitch-users] mfcr2
by Mike Murdock 11 Apr '08

11 Apr '08

What is mfcr2 used to interface with? Michael Murdock VP Operations & Development Switchmaxx Product Line 125 N. Market. Suite 1520 Wichita KS. 67202 (316) 932-2693 -----Original Message----- From: Brian West [mailto:brian.west@mac.com] Sent: Tuesday, January 30, 2007 11:46 AM Eastern Standard Time To: Yeruult[ICN]; freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] mfcr2 Do you happen to know of an open source libmfcr2 that is license compatible with FreeSWITCH? /b On Jan 29, 2007, at 7:10 AM, Yeruult[ICN] wrote: > Hello all, > > > > Does Freeswitch supports mfcr2? I know mfcr2 is old protocol but > that doesn't mean ppl don't use it. As I know there is still many > guys from everywhere. As I understood asterisk v1.4 becoming > complicated for Steve Underwood's unicall library. Is there any > other open sources which supports mfcr2? > > > > -- > > Best regards, > > Yeruult[ICN] mailto:yeruult@icn-mn.net > > _______________________________________________ > 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

10 16

No outgoing media on SIP endpoints
by Chris Chen 11 Apr '08

11 Apr '08

I am new to free switch, and I am pretty amazed by the features and potentials of it. After reading a lot from wiki site and lists, I managed to make the freeswitch rc2 up and running on my centos 5, and had the sip extension 1000 on Bria 2.1and sip extension 1001 on xlite 3.0 registered from my XP PC to the freeswitch on centos, also have four extensions 1004-1007 on Grandstream 4004 and two extensions 1010 and 1011 on PAP2-NA all registered on freeswicth, I was able to hear music on hold by dialing 9999 and joined conference 300x, I could call each other and saw presence of the extensions on Bria 2.1 , and the extensions could answer the ringing. But I coudn't hear the remote calling party after answering each of the call, I only heard the voice mail prompt, when I dialed any number 1 or 2 or 3 or any digit, seemed no DTMF detection and no audio being sent. Could you guys shed some light on this and put me to the right direction? Your guidance is much appreciated. Thanks Chris

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windows build problems
by Tamas Cseke 10 Apr '08

10 Apr '08

Hello, I have 2 problems: 1, libvoipcodecs C2143: syntax error : missing ')' before ' I don't know what the problem with brackets, but with the attached modification it complies. Do you have any idea, why? :) 2. freeswitchcorelib acl functions can't find inet_pton symbol. Thanks, Tamas Index: lpc10_decode.c =================================================================== --- lpc10_decode.c (revision 8061) +++ lpc10_decode.c (working copy) @@ -249,6 +249,7 @@ float slope; float uvpit; float xxy; + float tmp; rci_dim1 = LPC10_ORDER; rci_offset = rci_dim1 + 1; @@ -444,7 +445,9 @@ xxy = expf(xxy); rci[j + *nout*rci_dim1 + 1] = (xxy - 1.0f)/(xxy + 1.0f); } - rmsi[*nout - 1] = logf(s->rmso) + prop*(logf(*rms) - logf(s->rmso)); + //rmsi[*nout - 1] = logf(s->rmso) + prop *(logf((float)*rms) - logf(s->rmso)); + tmp = logf(*rms) - logf(s->rmso); + rmsi[*nout - 1] = logf(s->rmso) + prop * tmp ; rmsi[*nout - 1] = expf(rmsi[*nout - 1]); } }

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Dialplan recipes section up on the wiki
by Brian Snipes 10 Apr '08

10 Apr '08

I've started up a Dialplan Recipes section on the wiki - http://wiki.freeswitch.org/wiki/Dialplan_Recipes . Please add working dialplan examples to help yourself and others in the FreeSWITCH adventure. The current example uses mod_limit to emulate a pbx with rollover lines and distinctive rings on the second line. Fix any errors you see and I am looking forward to seeing what others come up with. Brian

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