April 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

running on visual studio and elaborative documentation
by kunal rao 14 Apr '09

14 Apr '09

Hi even I have downloaded FreeSWITCH and using MS Visual Studio 2008. It is building properly. I now want to configure it properly. Can you please give me directions and also some links for good detailed comparisons between Asterisk and FreeSWITCH and elaborative documentation for the same.. --- On Wed, 1/4/09, freeswitch-users-request(a)lists.freeswitch.org <freeswitch-users-request(a)lists.freeswitch.org> wrote: From: freeswitch-users-request(a)lists.freeswitch.org <freeswitch-users-request(a)lists.freeswitch.org> Subject: Freeswitch-users Digest, Vol 34, Issue 3 To: freeswitch-users(a)lists.freeswitch.org Date: Wednesday, 1 April, 2009, 7:29 PM Send Freeswitch-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Freeswitch-users digest..." Today's Topics: 1. Re: Originate and Conference (Peter P GMX) 2. Compiler error for Windows XP (SP2) (Lewis Liu) 3. Re: Call For Help: Janitor Projects (Anthony Minessale) 4. Re: Compiler error for Windows XP (SP2) (Michael Jerris) 5. Re: Call For Help: Janitor Projects (Raymond Chandler) ---------------------------------------------------------------------- Message: 1 Date: Wed, 01 Apr 2009 13:41:06 +0200 From: Peter P GMX <Prometheus001(a)gmx.net> Subject: Re: [Freeswitch-users] Originate and Conference To: freeswitch-users(a)lists.freeswitch.org Message-ID: <49D352D2.3070303(a)gmx.net> Content-Type: text/plain; charset=ISO-8859-15 Hello Brian, I tried this (on trunk 12862), but still the same behaviour. It does not aks for a PIN. Neither when transfering directly to the conference nor by transfering to the dialplan extension where conference is handled. Best regards Peter Brian West schrieb: > Update again to svn trunk... btw 1.0.4 pre3 is out on > files.freeswitch.org > > /b > > On Mar 30, 2009, at 6:44 PM, Dan Le wrote: > >> I get similar behavior as Peter when trying to enter a locked >> conference. >> >> If I am just dialing from a phone to a conference (on a dialplan), it >> will properly lock me out. But if I do an originate command >> (originate sofia/internal/1001 &conference(3000)), it will drop me >> into the conference, even though it is suppose to be locked. >> >> I am using the released 1.0.3 tag. >> > > Brian West > brian(a)freeswitch.org <mailto:brian@freeswitch.org> > > -- Meet us a ClueCon! http://www.cluecon.com <http://www.cluecon.com/> > > > > ------------------------------------------------------------------------ > > _______________________________________________ > 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 > ------------------------------ Message: 2 Date: Wed, 1 Apr 2009 18:41:44 +0800 From: Lewis Liu <lewisppp(a)gmail.com> Subject: [Freeswitch-users] Compiler error for Windows XP (SP2) To: freeswitch-users(a)lists.freeswitch.org Message-ID: <814e59990904010341y61b920c2h24bb8a50c8ae2f44(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" We download FreeSWITCH from SVN Trunk and want to build it on MS Visual Studio 2008 with platform. But we got one error message when we build it. FreeSWITCH\libs\win32\sofia\debug\libsofia_sip_ua_static.lib is built fail. So many files are lost, such as mod_sofia.dll..... Could you help me me for this, Please?? Whether something is lost in MS Visual Studio 2008 ?? Thanks a lot!! Lewis

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Can't call registered user in internal-ipv6 profile
by Bernhard Schmidt 13 Apr '09

13 Apr '09

Hi, probably a pretty easy problem, but I can't figure it out nevertheless. I'm still experimenting with FreeSwitch (SVN trunk, about two weeks old), mainly due to the IPv6 SIP support. I'm pretty much running the default configuration included in the sourcetree. Now I've hit the following problem: I have two phones registered, one Snom with extension 1000 on IPv4 (profile internal), one SIP Communicator with extension 1002 on IPv6 (profile internal-ipv6). I can call the Snom just fine (from the SIP Communicator as well as from outside or the CLI), but not the SIP Communicator. EXECUTE sofia/internal/1000(a)obelix.oms16.birkenwald.de bridge(user/1002(a)172.16.1.69) 2009-04-13 18:23:45 [DEBUG] switch_ivr_originate.c:1077 switch_ivr_originate() variable string 0 = [presence_id=1002(a)172.16.1.69] 2009-04-13 18:23:45 [ERR] switch_ivr_originate.c:1486 switch_ivr_originate() Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2009-04-13 18:23:45 [DEBUG] switch_ivr_originate.c:2084 switch_ivr_originate() Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2009-04-13 18:23:45 [ERR] switch_ivr_originate.c:1486 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] 2009-04-13 18:23:45 [DEBUG] switch_ivr_originate.c:2084 switch_ivr_originate() Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2009-04-13 18:23:45 [INFO] mod_dptools.c:2051 audio_bridge_function() Originate Failed. Cause: USER_NOT_REGISTERED It works when I explicitly specify the internal-ipv6 profile for the outgoing call like this: <action application="bridge" data="sofia/internal-ipv6/${dialed_extension}%172.16.1.69"/> but that isn't really what I want. What are the quirks I need to add to have "user/<extension>@<domain>" search both profiles? I already set force-register-domain on the profile, but I don't think that is what I'm looking for. Bernhard

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SIP switching made simple?
by Martin Fiala 13 Apr '09

13 Apr '09

Hello. I'm trying to use freeswitch, was able to compile it without problems, which is very nice. Then studying the configurations etc., I managed to set up SIP accounts those register properly. But now, if I want to call one registered account from the other one, I get error 404 - not found. I tried to set up a minimalistic dialplan using xml syntax as well as asterisk syntax but neither worked for me. I changed just a few thing, I'll list them later. I'm trying to make calls using ip addresses and ports instead of domain names.. This is the error freeswitch outputs: 2009-04-13 18:35:48 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/02(a)192.168.2.100 [19bad83a-ec9a-4b59-8457-cd76f1eaef65] 2009-04-13 18:35:48 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 02->01 in context default 2009-04-13 18:35:48 [NOTICE] switch_ivr.c:1343 switch_ivr_session_transfer() Transfer sofia/internal/02(a)192.168.2.100 to enum[01@default] 2009-04-13 18:35:50 [INFO] switch_core_state_machine.c:122 switch_core_standard_on_routing() No Route, Aborting 2009-04-13 18:35:50 [NOTICE] switch_core_state_machine.c:123 switch_core_standard_on_routing() Hangup sofia/internal/02(a)192.168.2.100 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2009-04-13 18:35:50 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 1 (sofia/internal/02(a)192.168.2.100) Ended 2009-04-13 18:35:50 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/02(a)192.168.2.100 [CS_HANGUP] My users are added in file users.xml in directory/ : <include> <user id="01" mailbox="01"> <params> <param name="password" value="01"/> <param name="vm-password" value="01"/> </params> <variables> <variable name="accountcode" value="01"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="01"/> <variable name="effective_caller_id_number" value="01"/> </variables> </user> <user id="02" mailbox="02"> <params> <param name="password" value="02"/> <param name="vm-password" value="02"/> </params> <variables> <variable name="accountcode" value="02"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="02"/> <variable name="effective_caller_id_number" value="02"/> </variables> </user> </include> I've added the file dialplan/default/000_default.xml with contents: <extension name="internal"> <condition field="source" expression="mod_sofia" /> <condition field="destination_number" expression="^(4\d+)"> <action application="bridge" data="sofia/internal/$0@192.168.2.100:5060" /> </condition> </extension> That's from sample configs, I wonder, if the IP address can be used like that. I understand it that way, the ip address specified is of registrar server. I've added the port as I'm testing it on local loop and thus am running different sip services on the same ip (freeswitch and calling softfones). Is that ok? extensions.conf I've tried to use: [default] ; Things you're used to.... ;exten => music,n,Dial(SIP/1234(a)conference.freeswitch.org|120) ;exten => _1XXXXX,n,set(cool=${EXTEN}) ;exten => _1XXXXX,n,set(myvar=true) ;exten => _1XXXXX,n,Goto(default|music) ;exten => 2137991400/1000,n,Goto(default|music) ; Some new magic you can do.... ;exten => ~^(18(0{2}|8{2}|7{2}|6{2})\d{7})$,n,enum($1) ;exten => ~^(18(0{2}|8{2}|7{2}|6{2})\d{7})$,n,bridge(${enum_auto_route}) ; instead of exten, put anything about the call you would rather match on. ; either the names of a field in caller_profile or a string of variables to expand. ;caller_id_number => 2137991400,n,Goto(default|music) ;${sip_from_user} => bill,n,Goto(default|music) [pbx] exten => 01,1,Dial(SIP/01,20) exten => 02,1,Dial(SIP/02,20) When using extensions.conf I've changed this line in sip_profiles/internal.xml from: <param name="dialplan" value="XML"/> to <param name="dialplan" value="asterisk,XML"/> I didn't make any other changes in that file. I didn't change anything else. I'm trying to use two sip phones - one using port 6001 (user "01") and the other one 5000 (user "02"). After registration succeeds, calling this sip uri : sip:01@192.168.2.100:5060, where 192.168.2.100:5060 is IP:PORT of freeswitch (the IP is same for softphones.. the same machine). Thanks for any help. Fiala

2 1

Noise for dial out conference
by zhaoxxqq 13 Apr '09

13 Apr '09

Hello, I use below to realize call out conference: conference testconf bgdial {originate_timeout=30}sofia/default/1001(a)192.168.0.72 1234567890 FreeSWITCH_Conference" conference testconf bgdial {originate_timeout=30}sofia/default/1002(a)192.168.0.170 1234567890 FreeSWITCH_Conference" but when the second phone is connected. there are big noise in the phone, Can anyone help me to solve it? zhao xiaoqiang

2 1

mod_opal and DTMF...
by Peter Olsson 13 Apr '09

13 Apr '09

When dialing in to FreeSWITCH from a h323 client, FreeSWITCH doesn't seem to detect DTMF. I'm not sure if this is a setting somewhere in the config files, or if it's a bug. The test scenario is simple - use the default FreeSWITCH config, dial in to voicemail (4000) and try to log in. It doesn't detect any DTMF tones. Regards, Peter Olsson

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Can gateways be configured on a per user basis?
by Eric Chamberlain 13 Apr '09

13 Apr '09

Hello, I'm exploring the capabilities of FreeSwitch and have some questions: Can gateways be configured on a per user basis? A user needs to be limited to only their own gateways and other users can't user their gateway. Something like: User 1 has two gateways - VoIP provider A and VoIP provider B User 2 also has two gateways - VoIP provider A and VoIP provider C User 3 has ten gateways User 4 has one gateway etc. Is it possible to have some gateway settings global and some user specific, say proxy and port info global, but username and password are specific to each user? Is it also possible to specify on a per user basis whether the user's gateway will register for the provider? Using the example above, User 1 and User 2 would both register with VoIP provider A. And can that registration happen even if User 1 or User 2 don't have any SIP endpoints registered with FreeSwitch? -- Eric Chamberlain

2 1

Replace sqlite with couchDB?
by Nicolas Brenner 13 Apr '09

13 Apr '09

Hi, I am not very familiar with FS internals, but I recently found this "new" db engine called couchDB. Looks pretty interesting, and its main focus is scalability. Has anybody played with couchDB? does it make sense to replace sqlite with couchDB in FS? Here's a link to the project homepage: - http://couchdb.apache.org/ And here's a video of a presentation given by one of the lead programmers: - http://www.vimeo.com/1992869

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Question about play_and_get_digits in mod_dptools
by Diego Viola 12 Apr '09

12 Apr '09

Hi all, I want to use play_and_get_digits from mod_dptools and have some questions about it. I have used playAndGetDigits() in Lua but I see the syntax in mod_dptools is a bit different and I got a bit confused. I see the syntax in the play_and_get_digits from the mod_dptools is something like this: switch_play_and_get_digits(session, min_digits, max_digits, max_tries, timeout, valid_terminators, prompt_audio_file, bad_input_audio_file, var_name, digit_buffer, sizeof(digit_buffer), digits_regex} Can you please explain to me what the session parameter is? And will this allow me to use a phrase macro so I can call my IVR instead of calling a regular file? This is how I use the playAndGetDigits in Lua: digits = session:playAndGetDigits(2, 5, 3, 7000, "#", "phrase:rac_demo", "", "\\d+"); I call a phrase macro instead of playing a file, can I do the same with play_and_get_digits from mod_dptools? and please explain me what the session parameter in play_and_get_digits is. Thanks, Diego

1 4

Selecting a particular outgoing gateway ?
by David Robinson 12 Apr '09

12 Apr '09

I have configured two providers in sip_profiles/external/example.xml as follows: <include> <gateway name="voicepulse"> <param name="username" value="09515130"/> <param name="realm" value="sip.faktortel.com.au"/> <param name="password" value="xxxx"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="extension" value="1000"/> </gateway> <gateway name="voicepulse_pt"> <param name="username" value="8889200745"/> <param name="realm" value="sip.pennytel.com"/> <param name="password" value="xxxx"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="extension" value="1000"/> </gateway> </include> I have put this in dialplan/defaul.xml to dial out. Regexp matches the first extension if it's a mobile. I want it to route to Pennytelm. voicepulse_pt. <extension name="Long Distance - voicepulse pt"> <condition field="destination_number" expression="^(04\d+)$"> <action application="set" data="effective_caller_id_number=11111111111"/>   <action application="bridge" data="sofia/gateway/voicepulse_pt/ $1"/> </condition> </extension> <extension name="Long Distance - voicepulse"> <condition field="destination_number" expression="^(\d{4,10})$"> <action application="set" data="effective_caller_id_number=99999999999"/>   <action application="bridge" data="sofia/gateway/voicepulse/$1"/> </condition> </extension> Is this how I should be doing this ? I want to specify a different gateway for a different rexep. Please give me some idea what path I should take. David

3 2

Can't send commands with event socket outbound
by Diego Viola 11 Apr '09

11 Apr '09

Hello, I'm testing event socket outbound and whenever I use this: -bash-3.2# nc -v -l 127.0.0.1 8084 I send a call to the socket from my dialplan: <action application="socket" data="127.0.0.1:8084 async full"/> I can receive just fine but I can't send events from the nc cli. I tried with: connect\n\n sendmsg call-command: execute execute-app-name: answer\n\n But nothing happens, what's wrong? Thanks.

1 1