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Re: [Freeswitch-users] openzap / dialplan / Sangoma A204 questions
by John Wehle 02 Jul '08

02 Jul '08

> Can you try upgrading to the latest svn build of FS. > There are several fixes to openzap in there that I know will fix your issue. Note I do get the following messages: [WARNING] zap_zt.c:356 zt_open() Echo training not available for 1:2 [WARNING] zap_zt.c:642 zt_next_event() Unhandled event 17 Any idea what they mean? Should I be concern? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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How to dial out via Voipbuster ?
by Henk Oegema 01 Jul '08

01 Jul '08

I have created a gateway to Voipbuster in conf/sip_profiles/external/voipbuster.xml: <include> <gateway name="voipbuster"> <param name="username" value="henkoegema"/> <param name="realm" value="voipbuster.com"/> <param name="from-user" value="003216980xxx"/> <param name="password" value="xxxxxx"/> <param name="extension" value="1000"/> </gateway> </include> I suppose this is correct since I can register with Voipbuster: freeswitch@ubuntu> sofia status API CALL [sofia(status)] output: Name Type Data State =============================================== internal profile sip:mod_sofia@192.168.1.101:5060 RUNNING (0) external profile sip:mod_sofia@78.21.64.194:5080 RUNNING (0) voipbuster gateway sip:henkoegema@voipbuster.com REGED nat profile sip:mod_sofia@78.21.64.194:5070 RUNNING (0) default alias internal ALIASED 192.168.1.101 alias internal ALIASED outbound alias external ALIASED ================================================ 3 profiles 3 aliases Then I added the following lines to conf/dialplan/default.xml  <extension name="voipbuster"> <condition field="destination_number" expression="^(0031\d*})$"> <action application="bridge" data="sofia/gateway/voipbuster/$1(a)voipbuster.com"/> </condition> </extension> Something is still not correct: Get always busy tone when I dial 003134464135x (?) No Route !! 2008-07-01 17:47:21 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting What am I doing wrong ? Thanks Henk

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openzap / dialplan / Sangoma A204 questions
by John Wehle 01 Jul '08

01 Jul '08

I have a A204 with hardware echo cancelling and two FXO modules on FreeBSD 6.2 connected to tip-ring lines from a PBX. ztcfg reports: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) Hooking a plain old telephone to the tip-ring lines from the PBX works fine. On startup freeswitch reports: [DEBUG] zap_io.c:1951 load_config() found config for span [DEBUG] zap_io.c:1978 load_config() created span 1 of type zt [DEBUG] zap_io.c:1991 load_config() span 1 [name]=[OpenZAP] [DEBUG] zap_io.c:1991 load_config() span 1 [number]=[551] [DEBUG] zap_io.c:1991 load_config() span 1 [fxo-channel]=[1] [DEBUG] zap_io.c:2020 load_config() setting trunk type to 'FXO' start(KEWL) [WARNING] zap_zt.c:135 zt_open_range() this ioctl fails on older zaptel but is harmless if you used ztcfg [device /dev/zap/channel chan 1 fd 26 (Inappropriate ioctl for device)] [INFO] zap_zt.c:170 zt_open_range() configuring device /dev/zap/channel channel 1 as OpenZAP device 1:1 fd:25 Ultimately I want freeswitch to run a script when any of the FXO lines receive a call. Playing around produced some questions: 1) I have a dialplan of: <extension name="outgoing-fxo"> <condition field="destination_number" expression="^55[1-4]$"> <action application="set" data="dialed_ext=482"/> <action application="bridge" data="openzap/1/1/${dialed_ext}"/> </condition> </extension> which I'm assuming will cause freeswitch to use the fxo to dial 482 on the PBX when routing a call to 551. When I dial 551 from a VoIP phone I see: [NOTICE] switch_channel.c:533 switch_channel_set_name() New Channel OpenZAP/1:1/482 [edf17e96-0247-dd11-9800-001fc6ab49e2] [WARNING] zap_analog.c:52 analog_fxo_outgoing_call() VETO Changing state on 1:1 from DOWN to DIALING [WARNING] zap_zt.c:356 zt_open() Echo training not available for 1:1 however I don't hear anything on the VoIP phone (i.e. no ringing) and extension 482 which is right next to the VoIP doesn't ring. 2) What would my dialplan look like so that dialing 551 bridges the call to the FXO with the FXO just going off hook ... not dialing? I.e. dialing 551 just gets me a PBX line with dialtone. 3) What condition would I use in my dialplan to match an FXO line ringing? I.e. when the FXO line rings I want to invoke javascript. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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Conference call 5PM Central UTC-6 on Wednesday
by Brian West 01 Jul '08

01 Jul '08

Everyone is invited... we need more people to get involved in the wiki/ doc project. I would also like to go over usability issues with the website... Download, getting started guide to make sure everyone can get started with minimal effort. We also need to have someone monitor and update the FAQ. I add questions and answers but we need to pay closer attention and make sure we get them all on there. I hope to see everyone there... http://wiki.freeswitch.org/wiki/Wiki_meet_2008_07_02 Please add to this page anything you wish to discuss on the agenda... ASAP. Thanks, Brian West

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polyphase lowpass filter
by e schmidbauer 30 Jun '08

30 Jun '08

can someone direct me to the line in the mod_shout.c that enables polyphase lowpass filter? i'd like to enable it when sending to an icecast stream. thanks.

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How to Pass the CLID
by Klaus Teller 30 Jun '08

30 Jun '08

Hi Guys, Forget about the YAML guy and continue with the impressive work. I got one question: when I bridge a leg A, to a number B. How do I pass the number of the caller A to the B. That is, what do I need to do to have B see the number of A on the display? Right now, despite setting the caller_id_num variable (using both sesstion.setVariable, and session.set), I still see a very strange number (area code 690) on B's display. Is the problem lying with the termination VOIP provider? Klaus L'Ivrogne. -- Der GMX SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen! Ideal für Modem und ISDN: http://www.gmx.net/de/go/smartsurfer

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Sphinx integration
by imthiyaz＠peopletech.co.in 30 Jun '08

30 Jun '08

Hi I am tring to get some opensource voice recognization and i found Sphinx and Sphinx latest versions supports only 16KHz samples.( which asterisk doesn't http://www.voip-info.org/wiki-Sphinx) I planing to test freeswitch ( i think it supports )with Sphinx , any inputs and support information will be appreciated. Thank you Imthiyaz Original Message: ----------------- From: Michael Collins mcollins(a)fcnetwork.com Date: Sun, 29 Jun 2008 19:46:55 -0700 To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] [Freeswitch-dev] YAML support asanalternative of XML for configuration > You said: "If XML is the single reason you won't be using FreeSWITCH > then by all means stick with Asterisk, because we aren't going to beg > you to use FreeSWITCH." > > This is sad, you claim that FreeSWITCH is more advanced in terms of > features and innovation than Asterisk, but all I see with your > attitude is that it's more limited and closed-minded. > Please don't be so quick to judge Brian by this response. Or better yet, please don't be so quick to judge FreeSWITCH by Brian's response. The simple fact of the matter is that FS *is* more advanced than Asterisk, regardless of whether or not the devs are willing or able to swap out XML for YAML. > I'm afraid that I will have to stick with Asterisk, at least they > aren't arrogant and closed-minded when you ask for a feature, all I I wonder what the asterisk devs would say if you asked them to switch the conf files out for YAML files? Let's just say that their response would be something similar to: "Uh... no." > want is a more effective way (for me) to configure FreeSWITCH, this is Actually, technically, there's nothing preventing someone from writing a script/module/program that would convert YAML to XML, and even doing it on the fly. Can you honestly say that can be done with Asterisk? Even if it could be done it would be much more difficult than doing it with FS. FS has hooks built into it that allow for this kind of flexibility. Tony knew from the get go that not everyone would be willing/able to hand edit all the XML files. So an important decision was made early: make the XML be both human- and machine-readable. > not going to hurt you or steal your freedom, or change the way you do > things... it's only going to allow me to do things in the way I'm used > to and I like. > > This is exactly why I love Free Software & Open Source, because it > gives me the freedom to do things exactly the way I want, but I see I > wont change your mind... so, good luck with FS. I don't know if changing Brian's mind is actually relevant anyway. The best thing to do would be to write a parser that converts XML to YAML and YAML to XML and demonstrate the coolness/value/etc. of a YAML-based configuration. If YAML is indeed as great as you feel it is then I'm certain that there will be others who would agree and would be willing to support the effort to make a YAML-based config module/parser/thingy. So let me rephrase Brian's last words: If the XML-based nature of FreeSWITCH is the *only* thing that is preventing you from using it then please consider the possibility that you might be missing out on something very cool and very powerful. You aren't forced to use XML, but it is the easiest of the alternatives, at least at the start. Once you get hooked on FS you'll realize that it's a better choice for a switching platform. The choice certainly is yours, and no one here will begrudge you your right to choose. My parting advice: choose wisely. -MC > > Diego > > On Thu, Jun 26, 2008 at 11:15 PM, Brian West <brian(a)freeswitch.org> wrote: > > Diego, > > I can already tell you Its not going to happen. The choice was > made > > early on and it was the correct one. It's not something that can be > > changed or modular. If XML is the single reason you won't be using > > FreeSWITCH then by all means stick with Asterisk, because we aren't > > going to beg you to use FreeSWITCH. > > > > Read this for more info http://www.freeswitch.org/node/123 > > > > /b > > > > On Jun 26, 2008, at 9:14 PM, Diego Viola wrote: > > > >> Hi everyone - > >> > >> I really love this project... I love the design you guys took with FS, > >> I love the stability, the features, the openness of it, it's the VoIP > >> platform I always dreamed of it... FreeSWITCH is what Asterisk should > >> have been. > >> > >> But I just don't like one thing: XML. > >> > >> I also don't intend to create a flame war or troll with this thread, > >> please take this as a suggestion of someone that really wants to use > >> FS... XML is the only reason why I still stay with Asterisk, I just > >> can't like XML, and I don't like Asterisk as well, I want FreeSWITCH > >> stability, features, etc, but I can't like XML, not even if I try to, > >> it's too slow to use/write, verbose and complex. > >> > >> As a sysadmin point of view, I prefer parameter=value but I realize > >> the advantages of XML (data serialization, etc), this is also why I > >> don't use mod_dialplan_asterisk, because I think it's time for > >> something better than INI files, but XML isn't for me, I'm a > >> Linux/UNIX guy and I don't want a GUI either, I love CLI, but I want > >> something simple to use, read and write, and I think YAML is the best > >> of the both worlds. > >> > >> I would like to have more than one option to configure FreeSWITCH, > >> also, not everyone likes to do the same thing the same way and it > >> would be really cool if FS gives this freedom to people... > >> > >> I would really like to see YAML as a configuration option in > >> FreeSWITCH, give me this any day and I swear I don't toch Asterisk > >> _ever again_, YAML can be used for data serialization and it's more > >> simple than XML, Ruby projects like Rails uses it. > >> > >> Take a look here: > >> > >> http://yaml.org/spec/1.2/ > >> http://yaml.org/ > >> > >> "YAML(tm) (rhymes with "camel") is a human-friendly, cross language, > >> Unicode based data serialization language designed around the common > >> native data types of agile programming languages. It is broadly useful > >> for programming needs ranging from configuration files to Internet > >> messaging to object persistence to data auditing. Together with the > >> Unicode standard for characters, this specification provides all the > >> information necessary to understand YAML Version 1.2 and to create > >> programs that process YAML information. " > >> > >> Please consider it, I'm also willing to pay for a mod_yaml, other than > >> this I really love FreeSWITCH and I think it's great, I will switch in > >> a heartbeat if you guys add support for it ;-) > >> > >> Best regards, > >> > >> Diego > >> > >> _______________________________________________ > >> 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 > > > > > > _______________________________________________ > > Freeswitch-dev mailing list > > Freeswitch-dev(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > > > _______________________________________________ > 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 _______________________________________________ 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 -------------------------------------------------------------------- mail2web.com What can On Demand Business Solutions do for you? http://link.mail2web.com/Business/SharePoint

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Re: [Freeswitch-users] Can't register with Linksys SPA 3000
by Henk Oegema 30 Jun '08

30 Jun '08

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YAML support as an alternative of XML for configuration
by Diego Viola 30 Jun '08

30 Jun '08

Hi everyone - I really love this project... I love the design you guys took with FS, I love the stability, the features, the openness of it, it's the VoIP platform I always dreamed of it... FreeSWITCH is what Asterisk should have been. But I just don't like one thing: XML. I also don't intend to create a flame war or troll with this thread, please take this as a suggestion of someone that really wants to use FS... XML is the only reason why I still stay with Asterisk, I just can't like XML, and I don't like Asterisk as well, I want FreeSWITCH stability, features, etc, but I can't like XML, not even if I try to, it's too slow to use/write, verbose and complex. As a sysadmin point of view, I prefer parameter=value but I realize the advantages of XML (data serialization, etc), this is also why I don't use mod_dialplan_asterisk, because I think it's time for something better than INI files, but XML isn't for me, I'm a Linux/UNIX guy and I don't want a GUI either, I love CLI, but I want something simple to use, read and write, and I think YAML is the best of the both worlds. I would like to have more than one option to configure FreeSWITCH, also, not everyone likes to do the same thing the same way and it would be really cool if FS gives this freedom to people... I would really like to see YAML as a configuration option in FreeSWITCH, give me this any day and I swear I don't toch Asterisk _ever again_, YAML can be used for data serialization and it's more simple than XML, Ruby projects like Rails uses it. Take a look here: http://yaml.org/spec/1.2/ http://yaml.org/ "YAML™ (rhymes with "camel") is a human-friendly, cross language, Unicode based data serialization language designed around the common native data types of agile programming languages. It is broadly useful for programming needs ranging from configuration files to Internet messaging to object persistence to data auditing. Together with the Unicode standard for characters, this specification provides all the information necessary to understand YAML Version 1.2 and to create programs that process YAML information. " Please consider it, I'm also willing to pay for a mod_yaml, other than this I really love FreeSWITCH and I think it's great, I will switch in a heartbeat if you guys add support for it ;-) Best regards, Diego

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Re: [Freeswitch-users] Can't register with Linksys SPA 3000
by Henk Oegema 29 Jun '08

29 Jun '08

On Sunday 29 June 2008 17:44:29 Михаил Кривушин wrote: > В сообщении от 29 июня 2008 вы написали: > HO> On Sunday 29 June 2008 15:38:25 Михаил Кривушин wrote: > HO> > Are you try to read log file? Can you post lines with error? > HO> > HO> > HO> Many lines of this message: > HO> > HO> 2008-06-29 17:14:28 [DEBUG] sofia.c:190 sofia_event_callback() event > HO> [nua_r_options] status [405][Method Not Allowed] session: n/a > HO> 2008-06-29 17:14:59 [DEBUG] sofia.c:190 sofia_event_callback() event > HO> [nua_r_options] status [405][Method Not Allowed] session: n/a > HO> 2008-06-29 17:15:29 [DEBUG] sofia.c:190 sofia_event_callback() event > HO> [nua_r_options] status [405][Method Not Allowed] session: n/a > HO> 2008-06-29 17:15:59 [DEBUG] sofia.c:190 sofia_event_callback() event > HO> [nua_r_options] status [405][Method Not Allowed] session: n/a I found that the messages above are caused by SJPhone (which works perfectly with FS) and not by Linksys. > I think <param name="disable-register" value="true"/> in one of your > profiles does not allow registration (I see such message when set this > param). May be Linksys try to connect to other than 5060 port? The port I specified is 5060.

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