August 2006 - FreeSWITCH-users - FreeSWITCH Mailing Lists

[Freeswitch-users] RE: [Freeswitch-users] 這裡有使用中文的朋友嗎?
by Gary Lin 28 Aug '06

28 Aug '06

有, 從台灣來的現在LA. Gary Lin -----Original Message----- From: will [mailto:will@camangi.com] Sent: Sunday, August 27, 2006 8:49 PM To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] 這裡有使用中文的朋友嗎? 這裡有使用中文的朋友嗎? 先跟大家問好一下，因為這是很難得的事情！ Will Lien

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No Dialplan, Aborting
by Naldo Alcalde H. 22 Aug '06

22 Aug '06

Hi, I installed FreeSwitch successfully on Windows 2000 SP4, enabled portaudio module and set my dialplan param to "Regex/XML Dialplan". However, when I make a call to a tollfree number ("pacall 18002275387" for example), I got the message "No Dialplan, Aborting". The output is below. Please give me a clue about the solution. Thanks in advance, Naldo Alcalde freeswitch@nalcalde> pacall 18002275387 2006-08-22 14:12:30 [NOTICE] switch_channel.c:327 switch_channel_set_name() New Chan PortAudio/18002275387-18be [5d340bd9-6cdd-f646-b950-17121b321dcd] 2006-08-22 14:12:30 [INFO] mod_portaudio.c:786 engage_device() Loaded codec L16 8000hz 20ms on PortAudio/18002275387-18be 2006-08-22 14:12:30 [DEBUG] mod_portaudio.c:869 place_call() PortAudio/180022753 87-18be State Change CS_NEW -> CS_INIT API CALL [pacall(18002275387)] output: FAILSUCCESS:2:5d340bd9-6cdd-f646-b950-17121b321dcd freeswitch@nalcalde> 2006-08-22 14:12:30 [DEBUG] switch_core.c:2206 switch_core_ session_run() (PortAudio/18002275387-18be) State INIT 2006-08-22 14:12:30 [DEBUG] mod_portaudio.c:185 channel_on_init() PortAudio/1800 2275387-18be State Change CS_INIT -> CS_RING 2006-08-22 14:12:30 [DEBUG] switch_core.c:2242 switch_core_session_run() (PortAu dio/18002275387-18be) State RING 2006-08-22 14:12:30 [DEBUG] mod_portaudio.c:202 channel_on_ring() PortAudio/1800 2275387-18be CHANNEL RING 2006-08-22 14:12:30 [DEBUG] switch_core.c:1934 switch_core_standard_on_ring() St andard RING PortAudio/18002275387-18be 2006-08-22 14:12:30 [INFO] switch_core.c:1950 switch_core_standard_on_ring() No Dialplan, Aborting 2006-08-22 14:12:30 [NOTICE] switch_core.c:1951 switch_core_standard_on_ring() H angup PortAudio/18002275387-18be [CS_RING] [DESTINATION_OUT_OF_ORDER] 2006-08-22 14:12:30 [INFO] switch_channel.c:838 switch_channel_perform_hangup() Kill PortAudio/18002275387-18be [1] 2006-08-22 14:12:31 [DEBUG] mod_portaudio.c:278 channel_kill_channel() PortAudio /18002275387-18be CHANNEL KILL 2006-08-22 14:12:31 [DEBUG] switch_core.c:2167 switch_core_session_run() (PortAu dio/18002275387-18be) State HANGUP 2006-08-22 14:12:31 [DEBUG] mod_portaudio.c:258 channel_on_hangup() PortAudio/18 002275387-18be CHANNEL HANGUP 2006-08-22 14:12:31 [DEBUG] switch_core.c:1924 switch_core_standard_on_hangup() Standard HANGUP PortAudio/18002275387-18be 2006-08-22 14:12:31 [DEBUG] switch_core.c:2202 switch_core_session_run() PortAud io/18002275387-18be State Change CS_HANGUP -> CS_DONE 2006-08-22 14:12:31 [DEBUG] switch_core.c:2560 switch_core_session_thread() Sess ion 3 (PortAudio/18002275387-18be) Locked, Waiting on external entities 2006-08-22 14:12:31 [INFO] switch_core.c:2567 switch_core_session_thread() Sessi on 3 (PortAudio/18002275387-18be) Ended 2006-08-22 14:12:31 [NOTICE] switch_core.c:2447 switch_core_session_destroy() Cl ose Channel PortAudio/18002275387-18be freeswitch@nalcalde>

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How to start with js
by Tamas 21 Aug '06

21 Aug '06

Hi, I'm experimenting with freeswitch and want to build simple IVR in js, however I don't know how to set up freeswitch for it. I have mod_spidermonkey compiled already. Could anybody tell me what to do next? [I know, there are some examples scripts at http://www.freeswitch.org/eg/js/] Thanks in advance, Tamas

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IAX2 'redirects'
by JP Hindin 11 Aug '06

11 Aug '06

Greetings; I was wondering if FreeSwitch (or, rather, libiax I suppose) had the ability to do a redirect in such a way that FreeSwitch would drop out of the media stream. In other words, on a bridge, will FreeSwitch let the source send the media stream direct to the destination service, without having to proxy the packets - much in the same way a SIP 302 Redirect works. Thanks; JP

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Re: [Freeswitch-users] How to interface with google talk
by danish.samad＠vocalseeds.com 10 Aug '06

10 Aug '06

Hi, Mike, thanks for showing your concern. I was able to successfully bridge a call between a google talk client and asterisk(SIP). Although there were times when asterisk received and accepted the call while the jingle session was dropped. I did not really delve into the issue, so I am not really sure what was causing the issue. There are certain things I noticed, that I would like to know more about. Firstly, when calling from jinlge to sip,I can only map a single jabber account with a SIP account. Is there any way I can bridge multiple calls between Jingle and SIP endpoints, using the current configuration. If not, are you guys planning to implement this feature. Secondly, how can I make calls from sip endpoints to Jingle endpoints. Any configuration tips will be helpful. Regards, Danish > Danish, did you ever get a response to this message that resolved your > issue? We were all at our conference last week, so this may have been > missed. > > Mike > > -----Original Message----- > From: danish.samad(a)vocalseeds.com [mailto:danish.samad@vocalseeds.com] > Sent: Tuesday, August 01, 2006 6:37 AM > To: freeswitch-users(a)lists.freeswitch.org > Subject: [Freeswitch-users] How to interface with google talk > > Hi, > > I am trying to interface with google talk using the following setup > > Freeswitch<->JabberServer<->GoogleTalkServer<->Googletalk client > > Freeswitch will register with Jabberserver using a valid jabber account > on > the Jabber server. Using this account I should be able to login and add > any valid gmail id in its userlist and perform standard operations > (using > the jingle protocol) such as chat, voice, etc. Eventually I intend to > use > Freeswitch as a gateway between SIP and google talk clients, but I first > need to validate the above setup. > > To test the setup out, I created an account on jabber.anywise.com. I > was > able to successfully log into the server using PSI and adding a valid > gmail accounts in my roster. Now when I specify the account in > freeswitch.xml I am not able to register as I donot see the user coming > online in my google talk user list. I just see the following XML > messages > being dumped on the console. > > freeswitch> SEND[<?xml version='1.0'?><stream:stream xmlns:stream='h > ttp://etherx.jabber.org/streams' xmlns='jabber:client' > to='jabber.anywise.com' version='1.0'>] > > RECV[<?xml version='1.0'?><stream:stream > xmlns:stream='http://etherx.jabber.org/streams' id='44CF2BF7' > xmlns='jabber:client' from='jabber.anywise.com' version='1.0'>] > > RECV[<stream:features><register > xmlns='http://jabber.org/features/iq-register'/><auth > xmlns='http://jabber.org/features/iq-auth'/></stream:features>] > > For reference I am attaching related excerpts from the freeswitch.xml > file > > <load module="mod_dingaling"/> > .... > <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> > <settings> > <param name="debug" value="1"/> > <param name="codec-prefs" value="PCMU"/> > </settings> > <interface> > <param name="name" value="jingle"/> > <param name="login" value="user(a)jabber.anywise.com/talk"/> > <param name="password" value="pwd"/> > <param name="dialplan" value="XML"/> > <param name="message" value="Jingle all the way"/> > <param name="rtp-ip" value="vostroip"/> > <param name="auto-login" value="true"/> > <param name="use-rtp-timer" value="true"/> > <param name="exten" value="7777"/> > <param name="vad" value="both"/> > </interface> > </configuration> > ... > > <extension name="7777"> > <condition field="destination_number" expression="^7777$"> > <action application="bridge" data="playback > /var/lib/asterisk/sounds/test.wav"/> > </condition> > </extension> > > Please suggest what I need to do to achieve the required setup. Any help > will be appreciated. > > Regards, > Danish > > _______________________________________________ > 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 > > > _______________________________________________ > 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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the Compile Failure for FreeSWITCH
by Mizuki Tashiro 10 Aug '06

10 Aug '06

Dear FreeSWITCH guys! I tried to compile the FreeSWITCH, but failed. I compiled as the below URL instruction. http://www.freeswitch.org/gettingstarted.html At the end of compiling, the following message has been displayed. ********************************************************* creating freeswitch creating fsxs cd . && /bin/sh /usr/local/snapshot_freeswitch/build/config/missing --run autoheader configure.in:7: error: Autoconf version 2.58 or higher is required aclocal.m4:6426: AM_INIT_AUTOMAKE is expanded from... configure.in:7: the top level autom4te: /usr/bin/m4 failed with exit status: 1 autoheader: /usr/bin/autom4te failed with exit status: 1 gmake[2]: *** [src/include/switch_am_config.h.in] Error 1 gmake[2]: Exit Directory `/usr/local/snapshot_freeswitch' gmake[1]: *** [install-recursive] Error 1 gmake[1]: Exit Directory `/usr/local/snapshot_freeswitch' make: *** [install] Error 2 ********************************************************* I guess that the very primary error has been occurred on the final step for the creating freeswitch binary. Could you please advise me for fixing it? Thanks in advance and Best Regards, Mizuki Tashiro

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Call recording
by Bruno Impens 09 Aug '06

09 Aug '06

For telephone training, I need to create a fairly simple system that allows a call between two (sip) phones to be recorded and maybe to be transferred to a different sip number. At the customers request, this has has to be done on windows do no asterisk. Can freeswitch already do this ? Before I spend a lot of time trying to get it working only to find out that you cannot record. Suggestions on how to do it are welcome of course. Bruno Impens

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How to interface with google talk
by danish.samad＠vocalseeds.com 03 Aug '06

03 Aug '06

Hi, I am trying to interface with google talk using the following setup Freeswitch<->JabberServer<->GoogleTalkServer<->Googletalk client Freeswitch will register with Jabberserver using a valid jabber account on the Jabber server. Using this account I should be able to login and add any valid gmail id in its userlist and perform standard operations (using the jingle protocol) such as chat, voice, etc. Eventually I intend to use Freeswitch as a gateway between SIP and google talk clients, but I first need to validate the above setup. To test the setup out, I created an account on jabber.anywise.com. I was able to successfully log into the server using PSI and adding a valid gmail accounts in my roster. Now when I specify the account in freeswitch.xml I am not able to register as I donot see the user coming online in my google talk user list. I just see the following XML messages being dumped on the console. freeswitch> SEND[<?xml version='1.0'?><stream:stream xmlns:stream='h ttp://etherx.jabber.org/streams' xmlns='jabber:client' to='jabber.anywise.com' version='1.0'>] RECV[<?xml version='1.0'?><stream:stream xmlns:stream='http://etherx.jabber.org/streams' id='44CF2BF7' xmlns='jabber:client' from='jabber.anywise.com' version='1.0'>] RECV[<stream:features><register xmlns='http://jabber.org/features/iq-register'/><auth xmlns='http://jabber.org/features/iq-auth'/></stream:features>] For reference I am attaching related excerpts from the freeswitch.xml file <load module="mod_dingaling"/> .... <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="1"/> <param name="codec-prefs" value="PCMU"/> </settings> <interface> <param name="name" value="jingle"/> <param name="login" value="user(a)jabber.anywise.com/talk"/> <param name="password" value="pwd"/> <param name="dialplan" value="XML"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="vostroip"/> <param name="auto-login" value="true"/> <param name="use-rtp-timer" value="true"/> <param name="exten" value="7777"/> <param name="vad" value="both"/> </interface> </configuration> ... <extension name="7777"> <condition field="destination_number" expression="^7777$"> <action application="bridge" data="playback /var/lib/asterisk/sounds/test.wav"/> </condition> </extension> Please suggest what I need to do to achieve the required setup. Any help will be appreciated. Regards, Danish

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