November 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Javascript File
by Cavalera Claudio Luigi 17 Nov '08

17 Nov '08

Hello guys, I'm looking for some hints to debug javascript file handling with fs I've written a simple script console_log("info","hello\n"); fd = new File("/tmp/blah"); fd.open("read,write,create"); var buff = fd.readAll(); console_log("info",buff); but it does not output the content of /tmp/blah I've read this http://wiki.freeswitch.org/wiki/File I have done a svn update but can't find the file mozilla.build.sh where is it? any hint on how to track down the problem? Thanks, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. -----------------------------------------------------------------------------------------------------

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Re: [Freeswitch-users] Putting javascript to sleep
by Jon Bruel 17 Nov '08

17 Nov '08

Thanks Anthony, the msleep function is really nice, but I have some problems getting it to work. Somehow it fucks up the system, which becomes unresponsive after I have run it in a javascript. Should I just wait for the next release, or do you want a more detailed error report through the appropriate channels? /Jon

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Re: [Freeswitch-users] mod_openmrcp
by Arsen Chaloyan 17 Nov '08

17 Nov '08

OpenMRCP project is not supported for a while, probably neither is mod_openmrcp. An open source project maintenance is not a trivial and sometimes thankless job. I was thinking either to give up my activity in open source or just to start over last year. As a result I have invested a lot of time and proved my commitment to UniMRCP project. I would like to avoid the discussion over the move as this is meaningless task, instead if you see the use case in MRCP module probably we can think about mod_unimrcp. ________________________________ From: Anthony Minessale <anthony.minessale(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Cc: achaloyan(a)yahoo.com Sent: Sunday, November 16, 2008 11:09:54 PM Subject: Re: [Freeswitch-users] mod_openmrcp We did not write the openmrcp so we are at a disadvantage. I have CC'd the author of the code in case he has any idea. Maybe he can chime in. On Sun, Nov 16, 2008 at 12:34 PM, David Knell <dave(a)3c.co.uk> wrote: Hi Mike, Sure. I've a specific issue, which is a segfault the second time I try to invoke a TTS session (see MODASRTTS-8), and I'm interested in generalities about its stability, throughput, etc. Cheers -- Dave Do you have any specific questions? Mike On Nov 16, 2008, at 5:54 AM, David Knell wrote: Hi - Is anyone out there using mod_openmrcp? Be interested in finding someone further down the learning curve than I am at the moment.. _______________________________________________ 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 -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk _______________________________________________ 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 -- 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

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fifo in on connect
by Jon Bruel 17 Nov '08

17 Nov '08

Is there a way to activate an action when a call has entered the "fifo in"? I'm thinking of something similar as setting the channel variable execute_on_aswer. /Jon

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Re: [Freeswitch-users] E-mail as user id and bridging
by Jan Kubr 17 Nov '08

17 Nov '08

> I'll stick to what I'm doing now which is that I replace the at-sign > with a dot and take that as a username. > Sorry for the newbie question and thanks all for the answers. > Its better to use a dash... a dot is valid in the username part of an > email address... so what dot do you split on? > > Example > > brian.west(a)domain.com > brian(a)west.domain.com > > So which do you split on :) Not very clear is it? It is not, but I don't need to do that. I don't need to guess the user's e-mail from their SIP name; I put their ID in the directory entry as a call variable: <variable name="user_id" value="4"/> This ID is from the users database where I can access all the info about the user, including her e-mail address. My motivation was making it as simply as possible for the end users to connect to Freeswitch with their SIP phones. Since they already use their e-mails to log in to other systems, it would be consistent to use it as their SIP name, too. You showed me it is not possible, so I'll just tell them the username is their e-mail with @ replaced with a dot. For now that is good enough. (BTW dashes are allowed in the username part of e-mails as well, see atext definition in RFC822) Jan PS: How do I not start a new thread on each reply? I'm getting only the digest e-mail and need to start a new message when I want to reply.

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Sending messages with mod_dingaling
by Gonzalo Servat 17 Nov '08

17 Nov '08

Hi 1) I'm trying to get the hang of mod_dingaling and I noticed that when I send a message from gservat(a)gmail.com to znog(a)gmail.com (the latter being the client profile setup on FS), mod_dingaling tries to send another message again from the same sender/recipient which the google talk server rejects with the error shown below. Any ideas why this is? ------------------------------------------------------------------------------- <message to="znog(a)gmail.com" type="chat" id="4BF947C10A146B61_2" iconset="classic" from="gservat(a)gmail.com/gmail.988E8E36"> <body>testing</body> <met:google-mail-signature xmlns:met="google:metadata">12b790b93955e670</met:google-mail-signature> <cha:active xmlns:cha="http://jabber.org/protocol/chatstates "></cha:active> <nos:x value="disabled" xmlns:nos="google:nosave"></nos:x> <arc:record otr="false" xmlns:arc="http://jabber.org/protocol/archive "></arc:record> </message> 2008-11-17 00:57:40 [NOTICE] libdingaling.c:1276 on_log() SecSEND: ------------------------------------------------------------------------------- <message to="znog(a)gmail.com" type="chat" from=" gservat(a)gmail.com/gmail.988E8E36"> <body>testing</body> </message> 2008-11-17 00:57:41 [INFO] libdingaling.c:1274 on_log() SecRECV: ------------------------------------------------------------------------------- <message to="gservat(a)gmail.com/gmail.988E8E36" type="error" from=" znog(a)gmail.com"> <body>testing</body> <error code="400" type="modify"> <bad-request xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"></bad-request> <text xmlns="urn:ietf:params:xml:ns:xmpp-stanzas">If set, the 'from' attribute must be set to the user's full JID.</text> </error> ------------------------------------------------------------------------------- 2) I was trying to use the "chat jingle" interface to send a message to a contact and it didn't work. I used the following command: FreeSWITCH> chat jingle|znog@gmail.com|gservat@gmail.com|testing ------------------------------------------------------------------------------- <message to="gservat(a)gmail.com" type="chat" from="dp+znog(a)gmail.com<dp%2Bznog(a)gmail.com> "> <body>testing</body> </message> 2008-11-17 01:02:34 [INFO] libdingaling.c:1274 on_log() SecRECV: ------------------------------------------------------------------------------- <message to="dp+znog(a)gmail.com <dp%2Bznog(a)gmail.com>" type="error" from=" gservat(a)gmail.com"> <body>testing</body> <error code="400" type="modify"> <bad-request xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"></bad-request> <text xmlns="urn:ietf:params:xml:ns:xmpp-stanzas">If set, the 'from' attribute must be set to the user's full JID.</text> </error> </message> ------------------------------------------------------------------------------- I tried modifying mod_dingaling.c's chat_send() function to see if the "dp+" part was annoying the Google Talk server (ie. forced it to leave the "from" as is) but it didn't help. Just for the record, my jingle client profile looks like this: <include>  <profile type="client"> <param name="name" value="znog(a)gmail.com"/> <param name="login" value="znog(a)gmail.com/gtalk"/> <param name="password" value="pass-goes-here"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Call me"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="auto-login" value="true"/>   <param name="sasl" value="plain"/>  <param name="server" value="talk.google.com"/>  <param name="tls" value="true"/>  <param name="use-rtp-timer" value="true"/>  <param name="exten" value="inbound_gtalk"/>    <param name="vad" value="both"/>   </profile> </include> Any help would be appreciated and in return I will update the mod_dingaling WIKI page :-) Regards, Gonzalo

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Re: [Freeswitch-users] E-mail as user id and bridging
by Jan Kubr 16 Nov '08

16 Nov '08

> El Domingo, 16 de Noviembre de 2008, Jan Kubr escribi?: >> Basically I want the user to use his e-mail to login with her SIP >> phone. This works fine when the user id is james(a)yahoo.com (in a >> domain that is the IP of the FS box), but then it is not possible to >> bridge a call that user. > > An username part if a SIP or MAILTO URI can't contain @ symbol, it's illegal > as it's, in fact, the separator of the userpart and domain part. > > What you can do is escaping "@" in hexadecimal before creating the SIP URI (I > don't know the code for "@", but "a" == "%61"), so: > > sip:alice@domain.com == sip:%lice@domain.com OK I see now. I didn't realize I'm using a SIP URI here (since I'm only calling the user "locally") whose userinfo part can't contain the at-sign (as found here: http://tools.ietf.org/html/rfc3261#section-25.1) The escaping doesn't do quite what I wanted, but thanks a lot for the effort. I'll stick to what I'm doing now which is that I replace the at-sign with a dot and take that as a username. Sorry for the newbie question and thanks all for the answers. Jan

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Re: [Freeswitch-users] mod_openmrcp
by Anthony Minessale 16 Nov '08

16 Nov '08

We can certainly look into that. Anyone want to help code the new module? On Sun, Nov 16, 2008 at 2:21 PM, Arsen Chaloyan <achaloyan(a)yahoo.com> wrote: > OpenMRCP project is not supported for a while, probably neither is > mod_openmrcp. > > An open source project maintenance is not a trivial and sometimes thankless > job. > I was thinking either to give up my activity in open source or just to > start over last year. As a result I have invested a lot of time and proved > my commitment to UniMRCP project. I would like to avoid the discussion over > the move as this is meaningless task, instead if you see the use case in > MRCP module probably we can think about mod_unimrcp. > > ------------------------------ > *From:* Anthony Minessale <anthony.minessale(a)gmail.com> > *To:* freeswitch-users(a)lists.freeswitch.org > *Cc:* achaloyan(a)yahoo.com > *Sent:* Sunday, November 16, 2008 11:09:54 PM > *Subject:* Re: [Freeswitch-users] mod_openmrcp > > We did not write the openmrcp so we are at a disadvantage. > I have CC'd the author of the code in case he has any idea. > Maybe he can chime in. > > > On Sun, Nov 16, 2008 at 12:34 PM, David Knell <dave(a)3c.co.uk> wrote: > >> Hi Mike, >> >> Sure. I've a specific issue, which is a segfault the second time I try to >> invoke a TTS session (see MODASRTTS-8), and I'm interested in generalities >> about its stability, throughput, etc. >> >> Cheers -- >> >> Dave >> >> >> >> Do you have any specific questions? >> >> Mike >> >> On Nov 16, 2008, at 5:54 AM, David Knell wrote: >> >> Hi - >> >> Is anyone out there using mod_openmrcp? Be interested in finding >> someone further down the learning curve than I am at the moment.. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershtt… >> >> >> >> >> -- >> David Knell, Director, 3C Limited >> T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 >> http://www.3c.co.uk >> >> >> _______________________________________________ >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:213-799-1400

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mod_openmrcp
by David Knell 16 Nov '08

16 Nov '08

Hi - Is anyone out there using mod_openmrcp? Be interested in finding someone further down the learning curve than I am at the moment.. Cheers -- Dave -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk

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How many calls can FreeSwitch bridge simultaeously
by George Howitt 16 Nov '08

16 Nov '08

My FreeSwitch app needs to bridge a lot of calls, and process all the media. I'm using PCMU, and every inbound call gets bridged to another outbound call. There shouldn't be any transcoding. How many concurrent calls can FreeSwitch support under this scenario? Can someone point me to any performance measurements for FreeSwitch? Thanks.

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