- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] RESTful Bounty ..... Putting the competition to REST with RestFul VoIP2.x services
by Ward Mundy 28 May '08

28 May '08

Count me in for $100. Great idea! --Ward Mundy nerdvittles.com

4 3

Sip trunk aka gateway configuration
by Cavalera Claudio Luigi 27 May '08

27 May '08

Hello, I have got a class 4 softswitch connected to public network via E1, ISUP etc... My softswitch already accepts SIP Invites arriving from IP address of my freeswitch installation (1.0 "phoenix"). I would like to configure a SIP trunk class 4 between freeswitch and my softswitch; I've tried following instructions http://wiki.freeswitch.org/wiki/Tested_Phone_Providers_Listing but I think these are not the same as they are not class 4. Could you please point me to the right instructions on the wiki? Thanks a lot, 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. -----------------------------------------------------------------------------------------------------

2 6

XML based IVR
by Brian Snipes 27 May '08

27 May '08

Is it possible to have an xml based ivr play the greeting message once and if no response within a timeout period go ahead and tranfer to an internal extension in place of repeating the greeting? Also, is menu-exec-api and menu-exec-app when used in an xml ivr the same thing? Thanks, Brian

7 18

Newbie need help: no sound when dialing test numbers
by Klaus Teller 27 May '08

27 May '08

Hi Folks, I'm trying to find my way through Freeswitch. I installed it a while ago on my notebook and everything went really well. Yesterday, i aslo installed it on a dedicated server. But here I seem to run into some troubles that I trust you can help me solve. Essentially, i can connect to FS with X-Lite without problem (user: 1000, pass: 1234). Now when i dial all test numbers 1000, 1001, 9998, 9999, I simply hear nothing (no tetris, no voicemail, blank silence). I see on the X-Lite display that the call is established. I aslo see on the FS console that the call is being processed, but I can't hear anything. Any idea what might cause this? I thought at some point that sounds haven't been installed. But then there is no difference between the content of the sounds folder on my notebook and on the remote server. What am i doing wrong? To complete the picture, when I installed the sounds (make uhd-sounds-install, make uhd-moh-install, make hd-sounds-install, make hd-moh-install, make sounds-install, make moh-install) on the remote server, I repeatedly got the following message which I'm not sure if it means that sounds were successfully installed. Thanks for the great piece of work. I look forward to your support. Klaus. =========================FS Sound Install Result======== /bin/sh /home/freeswitch/quiet_libtool --mode=install /usr/bin/install -c 'libfreeswitch.la' '/usr/local/freeswitch/lib/libfreeswitch.la' /usr/bin/install -c .libs/libfreeswitch.so.1.0.0 /usr/local/freeswitch/lib/libfreeswitch.so.1.0.0 (cd /usr/local/freeswitch/lib && { ln -s -f libfreeswitch.so.1.0.0 libfreeswitch.so.1 || { rm -f libfreeswitch.so.1 && ln -s libfreeswitch.so.1.0.0 libfreeswitch.so.1; }; }) (cd /usr/local/freeswitch/lib && { ln -s -f libfreeswitch.so.1.0.0 libfreeswitch.so || { rm -f libfreeswitch.so && ln -s libfreeswitch.so.1.0.0 libfreeswitch.so; }; }) /usr/bin/install -c .libs/libfreeswitch.lai /usr/local/freeswitch/lib/libfreeswitch.la /usr/bin/install -c .libs/libfreeswitch.a /usr/local/freeswitch/lib/libfreeswitch.a chmod 644 /usr/local/freeswitch/lib/libfreeswitch.a ranlib /usr/local/freeswitch/lib/libfreeswitch.a PATH="$PATH:/sbin" ldconfig -n /usr/local/freeswitch/lib ---------------------------------------------------------------------- Libraries have been installed in: /usr/local/freeswitch/lib If you ever happen to want to link against installed libraries in a given directory, LIBDIR, you must either use libtool, and specify the full pathname of the library, or use the `-LLIBDIR' flag during linking and do at least one of the following: - add LIBDIR to the `LD_LIBRARY_PATH' environment variable during execution - add LIBDIR to the `LD_RUN_PATH' environment variable during linking - use the `-Wl,--rpath -Wl,LIBDIR' linker flag - have your system administrator add LIBDIR to `/etc/ld.so.conf' See any operating system documentation about shared libraries for more information, such as the ld(1) and ld.so(8) manual pages. ---------------------------------------------------------------------- -- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger

3 3

LCR
by Max Clark 27 May '08

27 May '08

Hello, I'm looking for LCR implementations for FreeSWITCH - are there any out there? Thanks, Max

4 3

play file on two session bridge
by mashudi 27 May '08

27 May '08

Dear all, In case bridging two session, is it possible to play a voice file so the caller and called can hear the same voice with low volume? How ?

2 1

Example xml_curl configuration for user directory
by Carl Mate 27 May '08

27 May '08

I wonder if anybody could provide a complete set of configuration files for a working xml_curl user directory lookup. I have been trying using the default set of configuration files, added the xml_curl module loading and modified the configuration like this: <configuration name="xml_curl.conf" description="cURL XML Gateway"> <bindings> <binding name="dialplan fetcher"> <param name="gateway-url" value=" http://172.16.26.10/apache2-default/dialplan.php" bindings="dialplan"/> </binding> <binding name="directory fetcher"> <param name="gateway-url" value=" http://172.16.26.10/apache2-default/directory.php" bindings="directory"/> </binding> </bindings> </configuration> Both scripts just dump the expected XML result: <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="dialplan" description="example_curl_dialplan"> <context name="default"> <extension name="curl_test"> <condition field="destination_number" expression="^999999999$"> <action application="answer"/> <action application="send_display" data="Echo Test"/> <action application="echo"/> </condition> </extension> </context> </section> </document> <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="directory" description="example_curl_directory"> <domain name="172.26.16.10"> <user id="carl"> <params> <param name="password" value="1234"/> </params> </user> </domain> </section> </document> The sofia-sip parameter "accept-blind-reg" is set to false. When using one of the default users (1000), I can register and the xml_curl dialplan lookup works like expected. However, when trying to register as user carl, I can see that the XML is received by the xml_curl interface, but sofia-sip complains that it cannot find the user. Here is a part of the console output: mod_xml_curl.c:194 xml_url_fetch() XML response is in /tmp/6053a796-2bdb-11dd-9449-17a3c8baef50.tmp.xml sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [carl(a)172.16.26.10] When using the xml_locate cli command, I get a similar result: > xml_locate directory domain name 172.16.26.10 mod_xml_curl.c:194 xml_url_fetch() XML response is in /tmp/93d6019a-2bdb-11dd-9449-17a3c8baef50.tmp.xml API CALL [xml_locate(directory domain name 172.16.26.10)] output: <domain name="172.16.26.10"> ### contents of local XML file (users 1000 and up) </domain> It fetches the XML from the webserver and dumps the contents of the local XML files. Disabling the "directory" section in the default freeswitch.xml file does not help, as I initially thought it might be caused by the duplicate domain names. I hope someone could help out here.

1 0

http://digg.com/software/FreeSwitch_1_0_Released
by Ken Rice 27 May '08

27 May '08

Help Spread the word! http://digg.com/software/FreeSwitch_1_0_Released

1 0

SIP-T or SIP-i
by mashudi 26 May '08

26 May '08

Dear all, is FS support SIP-T or SIP-I ?

4 5

Dialplan and user directory for VM
by Alex Kinch 24 May '08

24 May '08

Hi all, I've been tinkering around with Freeswitch for a few days, and am trying to figure out the best way build a voicemail server using mod_voicemail. There'll be no SIP extension registration for the moment, and all calls will come in via a SIP trunk from the PSTN. I've got so far, but need a little help with a couple of things: - I'd like the config information for each voicemail box to be held in an external database. From what I've figured out each voicemail box needs an entry in the directory, however as there'll be lots of them it'd probably be best to use mod_ldap or mod_xml_curl. As I can't seem to find any info on the wiki about mod_ldap I guess it'll be mod_xml_curl. Does this module work in realtime or will it only request the directory on start or reloadxml? - At present if a voicemail box isn't found, FS will pick up the call, say 'bye' and then hang up. Is it possible to put something in the dialplan that checks whether a box exists, and if not do something different with the call (eg play a message)? Thanks in advance for your help, Alex

1 0