- FreeSWITCH-users - FreeSWITCH Mailing Lists

internal profile question
by daemonserj TVC 23 Mar '13

23 Mar '13

Why out of the box "internal" profile bound to "public" dialplan instead of "default" ?

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FreeSWITCH-Portal with ember.js - Another GUI for FS fans
by Seven Du 22 Mar '13

22 Mar '13

Hi, I'm learning ember.js and wondering if anyone here has experience on that and has interest for another open source FS GUI. http://emberjs.com/ said it's a framework for creating ambitious web applications. I had used backbone.js and it sames that ember.js is more complicated and short of document so I had a hard time on that. Anyway, back to the topic. I started this project as a way of learning ember.js and hope it would be useful for FreeSWITCH. Make another FS GUI is not to reinvent a wheel but, sometimes you just found you have a hard time to get fusionPBX or blue.box runing. So the FS-Portal project is not aims to replace them but sometimes you just need a GUI that is simple and can be used out of the box. Take a look at the code https://github.com/seven1240/FreeSWITCH-Portal. It's just static html and js files you can put in your htdocs dir and it only depends on mod_xml_rpc right now. To make it more fun Websocket is more helpful and I patched mod_event_socket to use the libwebsockets lib. It is kind of working, just I'm not sure if it's a good idea to add the code into mod_event_socket or make a new module. Or should we wait the new WebRTC out which might use Websocket for signalling. Anyway, it's working w/o websocket now and please try and give me some feedback if you thinks it's helpful. Well, I do have something more to discuss here. mod_xml_rpc implemented some interfaces like /api, /txtapi, /webapi, and some commands implemented "show xx as xml", "show xx as json" and "sofia xmlstatus" etc. Is there a easy way or interest to make this more consistent to respond to different Content-Type? e.g. api, txtapi, xmlapi, jsonapi etc? Thanks. -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig)

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Friday Free For All Is GO
by Ken Rice 22 Mar '13

22 Mar '13

Join us on 888 Today¹s topic? You tell us -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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how many conferences?
by veerabhadrarao` 22 Mar '13

22 Mar '13

hi, I am working on Freeswitch Conference functionality. How many conferences can we create in freeswitch?and how to control the Creation of number of conferences in freeswitch? thanks in advance

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Alternative Debian package builder
by Cal Leeming [Simplicity Media Ltd] 22 Mar '13

22 Mar '13

Hello, I've recently released an alternative Debian package builder for FreeSWITCH. https://github.com/foxx/freeswitch-debian Although FreeSWITCH does already have suitable Debian packages (and a builder), it might not be suitable for your needs (and in our specific use case, we required an alternative approach). Some of the reasons for this might be; * Build your own packages with custom patches applied * Your build system requires an easy to use, 1 command buider * Building your own source packages from GIT for security reasons * Have a single Debian package to install rather than 100+ It supports the following features; * Uses 'get-packaged-orig-source' to fetch original source from FreeSWITCH git * Builds as non-native, all arch package using quilt 3.0 patching (in accordance with Debian guidelines) * Uses start-stop-daemon * Uses pbuilder to ensure a clean build * Creates 'freeswitch' system user and enforces permissions on FreeSWITCH files * Installs into /opt/freeswitch, rather than system dirs * Removing/purging package will NOT remove data/logs dir or delete 'freeswitch' system user (in accordance with Debian guidelines) * Enforces all necessary dependancies Usage: # Replace GIT_REF with the ref from GIT you wish to compile against # Replace FS_VERSION with the version of FreeSWITCH we are compiling $ apt-get install git $ git clone https://github.com/foxx/freeswitch-debian.git $ cd freeswitch-debian $ GIT_REF=master FS_VERSION=1.3.16 make Hope this helps someone else! Cal

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Friday Free For All?
by Jeff Leung 22 Mar '13

22 Mar '13

There's 2 people in there right now, so when are we going to start? ;)

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Regexp question
by Steven Schoch 22 Mar '13

22 Mar '13

I'm getting this error: 2013-03-21 09:52:46.921136 [ERR] switch_regex.c:101 COMPILE ERROR: 1 [nothing to repeat][^+12243336164$] It comes from this condition: <condition field="caller_id_number" expression="^\+?(12243336164|12025551212)$"> (The corresponding action routes the call directly to my extension, but that's not important.) The reason I have "\+?" is I didn't know if the plus sign was included in the caller-ID. I have since learned that it is, so I can remove the "?", but I still want to know what I did wrong. -- Steve

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java-esl-client
by Joel Dodson 22 Mar '13

22 Mar '13

Hi, I'm looking at the various ways to control a freeswitch instance from a java program. The event-socket interface looks like it does what we need. And I've looked into using the Java-ESL-Client. I haven't found a whole lot of documentation or recent discussion around the ESL-Client (looked on the freeswitch wiki, googled it and the mail list archives), and the last checkin to freeswitch-contrib was in Aug, 2010. Is the ESL-Client widely used by java developers needing to interface with freeswitch? I've downloaded the code from git.freeswitch.org and it's nice looking code and decently commented. Is the lack of activity an indication it's really stable and the event-socket interface hasn't changed much recently? Or has the project lost steam? thanks, Joel

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INVITE routing problem
by daemonserj TVC 22 Mar '13

22 Mar '13

Hello! I have experienced problem with routing incoming call from gateway to specified user. My configuration is one sofia endpoint with public context, one user, one dialplan extension. FS version is FreeSWITCH Version 1.2.7+git~20130307T181046Z~93e2a38efd (git 93e2a38 2013-03-07 18:10:46Z) User dial string is <param name="dial-string" value="${sofia_contact(${dialed_user}@ ${dialed_domain})}"/> but FS route INVITE to user through gateway instead of registered user network address. My gateway is <gateway name="multifon.ru"> <param name="username" value="79231382196"/> <param name="password" value="xxxxxxx"/> <param name="realm" value="multifon.ru"/> <param name="from-domain" value="multifon.ru"/> <param name="proxy" value="sbc.megafon.ru"/> <param name="register" value="true"/>  </gateway> nslookup sbc.megafon.ru Name: sbc.megafon.ru Address: *193.201.229.35* sofia_contact shows sofia_contact user/79231382196(a)multifon.ru *sofia/main_sip_profile/sip:79231382196@213.87.121.81:13820;ob* that corresponding to dial-string Here is INVITE to user while bridging: send 1146 bytes to tcp/[*193.201.229.35*]:5060 at 07:59:21.699359: ------------------------------------------------------------------------ INVITE sip:79231382196@213.87.121.81:13820;ob SIP/2.0 Via: SIP/2.0/TCP 37.143.10.90:53774;branch=z9hG4bKrXprN61vgyS9K Route: <sip:79231382196@*193.201.229.35* :5060;transport=tcp;lr>;tport=tcp;gw=multifon.ru Max-Forwards: 67 From: "79137519015" <sip:79137519015@37.143.10.90>;tag=yUX4QvQKmBmHQ To: <sip:79231382196@213.87.121.81:13820;ob> RequesURI and Route header shows destination is GW proxy. Why is this happens? I think it is able to play with sip_invite_req_uri and/or sip_invite_route_uri variables and so on but i think it is a wrong way... Dialplan is <include> <context name="public"> <extension name="from_multifon"> <condition field="destination_number" expression="^79231382196$"> <action application="set" data="ringback=${ru-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="user/79231382196(a)multifon.ru"/> </condition> </extension> </context> </include> User description is <domain name="multifon.ru"> <params> <param name="dial-string" value="${sofia_contact(${dialed_user}@ ${dialed_domain})}"/> </params> <variables> <variable name="user_context" value="public"/> </variables> <groups> <group name="default"> <users> <user id="79231382196"> <params> <param name="password" value="xxxxxxxx"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="79231382196"/> <variable name="effective_caller_id_name" value="79231382196"/> <variable name="effective_caller_id_number" value="79231382196"/> <variable name="outbound_caller_id_name" value="79231382196"/> <variable name="outbound_caller_id_number" value="79231382196"/> </variables> </users> </group> </groups> </domain>

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ERROR! unable to open ZRTP cache file
by Vbvbrj 22 Mar '13

22 Mar '13

Hello. Regularly in FS log I see: 09:03:54.044779 [DEBUG] switch_rtp.c:915 [ zrtp cache]: Storing ZRTP Cache... 09:03:54.044779 [DEBUG] switch_rtp.c:915 [ zrtp cache]: Storing ZRTP cache to <... 09:03:54.044779 [DEBUG] switch_rtp.c:915 [ zrtp cache]: ERROR! unable to open ZRTP cache file <. 09:03:54.044779 [DEBUG] switch_rtp.c:915 [ zrtp cache]: Storing ZRTP Cache - DONE. 09:03:54.044779 [DEBUG] switch_rtp.c:776 Saving ZRTP cache: OK Why is this logged, and why an error? There is no file name to search for permission or access. Note the missing ">" where file name must be. -- Mimiko desu.

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