July 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

xml_curl and gateways
by Peter P GMX 21 Feb '11

21 Feb '11

According to http://wiki.freeswitch.org/wiki/Mod_xml_curl I'v tried to set up gateways dynamically, but I do net get it to work: I always get : "Invalid profile" My assumptions for a right xml answer back to FS are as follows, but I think at least one of it is false: 1. ) I start with <document type='freeswitch/xml'> <section name='configuration'> <configuration name='sofia.conf' description='RailsAdditionalConf'> 2.) I do not want to change settings, so I do NOT add a <settings> tag 3.) I want to enhance an existing profile, so I add a profile tag (<profile name='external'>) 4.) I assume that I can have an existing profile "external" in the xml conf files and can mix additional parts dynamically into the same profile 5.) Within the profile I want to add gateways, so I add a gateway tag (<gateways>) 6.) Then I add the gateways 7.) I add the closing tags Which one of the assumptions if wrong? The request I receive is as follows: {"key_name"=>"name", "action"=>"directory", "tag_name"=>"configuration", "key_value"=>"sofia.conf", "controller"=>"xml_curls", "hostname"=>"freeswitch", "section"=>"configuration"} I only answer once at startup with the follwing: <document type='freeswitch/xml'> <section name='configuration'> <configuration name='sofia.conf' description='RailsAdditionalConf'> <profile name='external'> <gateways> <gateway name="freeswitch2"> <param name="username" value="1007"/> <param name="realm" value="192.168.0.56:5061"/> <param name="password" value="xxxx"/> <param name="proxy" value="192.168.0.56"/> <param name="expire-seconds" value="30"/> <param name="register" value="true"/> <param name="register-transport" value="tls"/> <param name="retry-seconds" value="30"/> </gateway> </gateways> </profile> </configuration> </section> </document> Does anybody have a correct XML answer for a gateway and can post it? Best regards Peter

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Stream audio file/live to multiple SIP endpoints with IP multicast
by Sluschny, Thomas 30 Aug '10

30 Aug '10

Hi, i want to stream audio (from file and live) from FreeSwitch to multiple SIP endpoints (these are proprietary). Because of bandwith limitations we have to use IP multicast for audio data. Now the question: how can i do this smartly in FreeSwitch? I have read some Wiki entries: - mod_esf: seams to use multicast, but i couldn't find usage information - mod_shout: streaming, but from internet ? MP3, but i want wav-files to stream - mod_local_stream: seams to play files in a loop, i want to play a whole file once. - mod_conference: seams not to support IP multicast Im searching for a simple way to send a SIP INVITE to all my endpoints and then stream audio (live from soundcard or from file) over IP multicast to all these endpoints. Thanks in advance, Thomas

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Direct inward dialling
by Peter P GMX 07 Jun '10

07 Jun '10

Dear all, when I get an incoming call from my SIP provider I do receive an invite on the trunk number, e.g. in Germany 0xxxxx9. However I have an extension block of 0xxxxx90 to 0xxxxx999. In the dialplan I checked the condition field="destination_number" but this compares to the number of the trunk 0xxxxx9, so I do not get the direct inward dialling extensions. The SIP to-header contains the direct inward dialling number 0xxxxx90 to 0xxxxx999 but I have no success in accessing it in the dialplan. I tried sip_to_user, sip_to_uri, sip_req_user, sip_req_uri, but no success. They are all empty. Any idea which variable to parse? Best regards Peter

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How can I join two freeswitch on two servers?
by Edmar Cruz 18 Dec '09

18 Dec '09

I like to connect two freeswitch, call each other, communicate and vice versa. Can you give me an example for that? Thanks -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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OSP Interop w/ Trans Nexus
by Gregory Boehnlein 02 Dec '09

02 Dec '09

Anyone know if Freeswitch can work w/ OSP and use something like Transnexus for CDR/Rating?

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Is there any license G729?
by Edmar Cruz 04 Nov '09

04 Nov '09

Hi, Is there any available license G729 for freeswitch? I need it to match G729 of Asterisks? If any please help me and instructions how to install this thing. Thanks -- View this message in context: http://www.nabble.com/Is-there-any-license-G729--tp24251951p24251951.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] 2 voicemail questions
by Mark Campbell-Smith 20 Oct '09

20 Oct '09

Hi! >> 1. Can I email the voicemail message to multiple email addresses? A comma separated list does not work in the extensions.xml file (1000.xml), but it does work if I hard code the email addresses into the notify-voicemail.tpl file. Could this be added to the switch so that it can handle comma separated lists? >> 2. How can I make Freeswitch dial a number AFTER a voicemail is left? >api Hangup hook? i want the 'voicemail' application to appear to call the extension to notify the user that there is a waiting message. This is an extract from my dialplan.xml: <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/>  <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> <condition field="${vm_boxcount(${destination_number}@${domain_name})}" expression="^(1)$"> <action application="log" data="MSGBOX ${vm_boxcount(${dialed_extension}@${domain_name})}"/> <action application="set" data="api_hangup_hook=originate sofia/internal_nat/${dialed_etension}%${domain_name} default default Message 4000 4000 3"/> This only works if the B leg (ie voicemail application) hangs up first. This would be an unusual situation and does not achieve what I want... is there any other way to achieve this? Thanks > Hi! > > I have 2 questions regarding voicemail ... > > 1. Can I email the voicemail message to multiple email addresses? If > so, what format is this in? > <param name="vm-mailto" value="me(a)myemail.com"/> Try a comma sep. list. Not sure if it will work. > > 2. How can I make Freeswitch dial a number AFTER a voicemail is left? api Hangup hook? I g > From: Brian West <brian(a)freeswitch.org> > On Jul 7, 2009, at 9:11 AM, Mark Campbell-Smith wrote: > Hi! > > I have 2 questions regarding voicemail ... > > 1. Can I email the voicemail message to multiple email addresses? If > so, what format is this in? > <param name="vm-mailto" value="me(a)myemail.com"/> > Try a comma sep. list. Not sure if it will work.

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Q931 TE State Timer
by Helmut Kuper 03 Sep '09

03 Sep '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I just want to let you know, that I started some work on Q.931 TE state timers in current openzap ozmod_isdn stack. The current stack has some problems with freeing ressources when far end doesn't follow Q931 state machine cleanly. On my side I mainly miss a RELEASE after sending DISCONNECT and so m more and more channels are wasted over time. Openzap Q931 stack has some preworked timer and state handling (in form of Q931_UX states). Unfortunately it is disabled in source. I enabled it, added some Q931_UX states to Q931StateTE.c, added a timeout handler to ozmod_isdn.c and tested it with my special problem. In lab missed RELEASEs are now detected and the corresponding channel is freed cleanly - - hurray :) I would like to put my work in openzap trunk as soon as it works stable in production and FS board allows me to do so. I can't promise that I will implement all timers for Q931 TE nor I plan to work on Q931 NT timers. As I said: Just for your information. Have a nice weekend! Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKMmbl4tZeNddg3dwRAvWbAJ43Vex8J3LiVMu9mvs4US23eif19gCeO37l 4CGbk8CA+n+/dMRo5y5A+i0= =ZtvM -----END PGP SIGNATURE-----

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Inboud Call Queue
by Saeed Ahmed 23 Aug '09

23 Aug '09

Hi All, In an inbound call center scenario is it possible that customers calls in and calls are distributed between online (who are registered on FS and in idle state) agents. I saw some on going discussion on list where it looks that currently it's not possible but I am newbie so maybe I didn't understand it well. If it's possible then please give me a start point that how can I implement it. Many Thanks

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JAVA ESL
by Jean-Marc Hyppolite 17 Aug '09

17 Aug '09

Hello, I built libesl and JAVA mod. (make and make javamod). But when I try to run a JAVA script with the following code ESLconnection connection = new ESLconnection("127.0.0.1", "9000", ""); ESLevent events = connection.getInfo(); System.out.println(events.toString()); I get the following error message: Exception in thread "main" java.lang.UnsatisfiedLinkError: /usr/lib/libesl.so: /usr/lib/libesl.so: undefined symbol: __gxx_personality_v0 at java.lang.ClassLoader$NativeLibrary.load(Native Method) at java.lang.ClassLoader.loadLibrary0(ClassLoader.java:1778) at java.lang.ClassLoader.loadLibrary(ClassLoader.java:1703) at java.lang.Runtime.loadLibrary0(Runtime.java:823) at java.lang.System.loadLibrary(System.java:1030) at ivr.IVRServer.<clinit>(IVRServer.java:18) Any help would be appreciated. Thanks. __________________________________________________________________ Ask a question on any topic and get answers from real people. Go to Yahoo! Answers and share what you know at http://ca.answers.yahoo.com

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