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Recommended tools for creating/extending a sip test suite?
by Mike Fedyk 15 Apr '09

15 Apr '09

Hi all, I'm looking for suggestions on which open source tools to use for creating (or extending if there is already a project for this) a sip test suite. I have already heard of sipp, but I want to know what others are using and how they go about this before starting from scratch myself. Some things I'd like to do: - Dialplan/ voice menu/provider/did testing: Call number, press 1, expect to receive call on another extension. (kinda like expect) - Load testing Basically I want to be able to automate how a human may interact with my installation to reproduce bugs and make sure they don't come back. That way I can make sure my changes (wherever they may be in my stack, dialplan, freeswitch, openser/kamailio/opensips, etc.). Any pointers and/or tips will be much appreciated.

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Help with play_and_get_digits from mod_dptools
by Diego Viola 14 Apr '09

14 Apr '09

Hi everyone, I have a question... I have this on my dialplan: <extension name="outbound2"> <condition field="destination_number" expression="^(1888)$"> <action application="play_and_get_digits" data="2 5 3 7000 # phrase:rac_demo /invalid.wav foobar \\d+"/> <action application="transfer" data="${foobar} XML default"/> </condition> </extension> What I want to do is play and read some digits and as soon as I get those digits, transfer to that extension... but this never happens, even if I terminate with a #. I do the same thing with Lua and it works with Lua, but I need it to work with play_and_get_digits from mod_dptools, because I plan to use this with event socket outbound, with an application which I'm currently working on. Any ideas? Thanks, Diego

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ANNOUNCEMENT: FreeSWITCH 1.0.4pre4 Now Available
by Michael Collins 14 Apr '09

14 Apr '09

The FreeSWITCH team is pleased to announce the immediate availability of version 1.0.4pre4. Details are available here: http://www.freeswitch.org/node/173 All are encouraged to upgrade as soon as possible. Thanks to everyone for their feedback, ideas, and bug reports. Please keep them coming. -Michael http://www.cluecon.com

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Use of loopback channels and bridge() in scripts...
by Peter Olsson 14 Apr '09

14 Apr '09

I have two problems that I haven't been able to solve. I've done the same tests in both javascript, and in .NET. The two scripts are pretty simple, they just answer an incomming call, creates a new session, wait for an answer on the second call leg, and then bridge the two channels together. In both cases everything works just fine, but the audio is distorted. The destination I'm calling is "loopback/5000" - the sample IVR application included in FreeSWITCH. I first thought it was a codec issue, but even after trying to switch to different codecs the problem was the same. It more sounds like it's a timestamping issue - the voice is not distorted enough to be a bad codec, but it reads way to fast (mayby twice the "normal" speed). When doing a direct transfer() to the other destination this works just fine, but I need to be able to have some extra logic to tell if the destination is available or not. The second problem occurs only in .NET. After doing this sample there is as loopback channel still hanging around. It seems like the call creates a loopback-a and loopback-b, the loopback-b dissapears as it should (when the call has been disconnected), but the other one stays there. When doing the same in javascript this doesn't seem to occur. I'm using the latest SVN trunk, and my OS is Windows XP. I found bug FSCORE-349 in Jira, which seems to point in to the direction that there might be a bug with the loopback channels in some cases, but I could not find anything about the audio which plays too fast. Has anyone else experienced this? Regards, Peter Olsson

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skypiax Round Robin interface
by dujinfang 14 Apr '09

14 Apr '09

Hi, I made a patch, so skypiax is possible to do a RR hunt besides the sequential interface ANY. Usage: originate skypiax/RR/other_skype_name sk list http://jira.freeswitch.org/browse/MODENDP-211

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Google Talk Integration...
by Moiz Chinoy 14 Apr '09

14 Apr '09

Hi, I have tried google talk integration with FS and is working fine. Great Work! Is it possible to have multiple concurrent incoming calls on the same gmail account? -- Regards, Moiz Chinoy.

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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

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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

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