June 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Forked dialing with different channelvars set...
by Rupa Schomaker (lists) 06 Jun '08

06 Jun '08

I have a seemingly simple problem that I can't figure out how to implement. Simplified Problem: I'd like to have my cell phone and my deskphone ring at the same time. This seems simple, but there is a small caveat. I need to have a channel variable set differently for my cellphone. Specifically: voicepulse requires you to have the caller id number and caller id name set to the number in order to pass it on to the PSTN. I want to see the callerid name on my deskphone however. === Ok, normally, I'd just do a: <action application="bridge" data="sofia/internal/101%$${domain},sofia/gateway/voicepulse/$${cellnumber}"/> but then my cell phone shows Unknown or Blocked when receiving the call. If prior to that I do: <action application="set" data="effective_caller_id_name=${caller_id_number}"/> Then I get the correct callerid on my cell, but only the callerid number on my deskphone. I asked on #freeswitch and had some hints, but they either went over my head or I didn't communicate my problem right. Using multiple extension entries with continue="yes" doesn't seem to execute them in parallel (so the deskphone and the cell don't ring at the same time). The "obvious" answer is how I implemented this in asterisk -- and maybe that is my problem, thinking in terms of asterisk. The bridge equivalent would be the dial() app. This allows me to dial back into the dialplan if I use the "local" channel. This gives a level of indirection that would then let me modify the channelvars as necessary. The only way I can see to "call" back into the dialplan is to use the transfer application. But that can't do multiple extensions in parallel so doesn't solve my problem. What am I missing? -Rupa

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g729
by Nicolas Brenner 06 Jun '08

06 Jun '08

Hi, I'd like to know what's needed to add support for G729, I know there's a bounty, but I couldn't make sense out of what's posted on the wiki. I'm really interested in this, as one of my current VoIP providers restricts me to using only this codec, which limits me to using Asterisk, hence I can't fully move to FS without G729 support. What would it take to make it happen? Thanks, -- Nicolás Brenner Medularis SpA

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trixswitch 0.4
by andrew 06 Jun '08

06 Jun '08

I released the latest version of trixswitch today. Its a fast way to get FreeSWITCH 1.0 running. trixswitch.org

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FS installed, but no external calls
by Brian B 05 Jun '08

05 Jun '08

FS is definitely up and running on my Lylix VPS, which is exciting. X-Lite is dialed in and can talk to the IVR etc. But no inbound or outbound calls. I tried configuring les.net (which I got working in * on the same box) but couldn't get that working, and sofia status is NOREG. I got gizmo to where status is REGED but from XLite when I dial a 10 digit number the console shows [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting Any tips ...or is there an affordable contractor out there who can get a "hello world" call going? sofia status API CALL [sofia(status)] output: Name Type Data State ================================================================================================= internal profile sip:mod_sofia@67.228.218.111:5060 RUNNING (0) nat profile sip:mod_sofia@67.228.218.111:5070 RUNNING (0) default alias internal ALIASED external profile sip:mod_sofia@67.228.218.111:5080 RUNNING (0) gizmo gateway sip:17476481805@proxy01.sipphone.com<sip%3A17476481805(a)proxy01.sipphone.com> REGED lesnet gateway sip:16643@did.voip.les.net<sip%3A16643(a)did.voip.les.net> NOREG 67.228.218.111 alias internal ALIASED outbound alias external ALIASED ================================================================================================= Thanks for any tips or suggestions! -Brian B

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CLI on TrixSwitch 0.4
by Simon Shaw 05 Jun '08

05 Jun '08

How can I open the CLI on Trixswitch 0.4? Thanks, Simon

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Re: [Freeswitch-users] FS installed, but no external calls
by Brian B 04 Jun '08

04 Jun '08

Thanks Brian, Klaus, and Peder for your help! Here's the les2.xml file which is now showing REGED. (Thanks Peder! For any potential Les.net users: your username is the longer peer name, not your shorter les username). I tried to figure out dialplan and what to do or even show there, but can't make out the documentation on that, and am not sure the best way to see if it can dial in. I'll try to put in some clarifying edits on the Wiki when I get over this hurdle. Let me know what to do next! <include> <gateway name="did.voip.les.net"> <param name="username" value="1664329999"/> <param name="realm" value="did.voip.les.net"/> <param name="password" value="mypwhere"/> </gateway> </include> What other files/stats should I send? On Wed, Jun 4, 2008 at 1:36 PM, Brian West <brian(a)freeswitch.org> wrote: > Brian, Just registering doesn't input the routing info into the dialplan. > You'll need to add it to the dialplan in addition to registration. Can you > clarify how you did your registration? Maybe post your XML without > user/pass in them so we can point out why it failed? > > /b > > On Jun 4, 2008, at 2:37 PM, Brian B wrote: > > FS is definitely up and running on my Lylix VPS, which is exciting. X-Lite > is dialed in and can talk to the IVR etc. But no inbound or outbound calls. > > I tried configuring les.net (which I got working in * on the same box) but > couldn't get that working, and sofia status is NOREG. > > I got gizmo to where status is REGED but from XLite when I dial a 10 digit > number the console shows > [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No > Route, Aborting > > > Any tips ...or is there an affordable contractor out there who can get a > "hello world" call going? > > > > sofia status > API CALL [sofia(status)] output: > Name Type > Data State > > ================================================================================================= > internal profile > sip:mod_sofia@67.228.218.111:5060 RUNNING (0) > nat profile > sip:mod_sofia@67.228.218.111:5070 RUNNING (0) > default alias > internal ALIASED > external profile > sip:mod_sofia@67.228.218.111:5080 RUNNING (0) > gizmo gateway > sip:17476481805@proxy01.sipphone.com<sip%3A17476481805(a)proxy01.sipphone.com> > REGED > lesnet gateway sip:16643@did.voip.les.net<sip%3A16643(a)did.voip.les.net> > NOREG > 67.228.218.111 alias > internal ALIASED > outbound alias > external ALIASED > > ================================================================================================= > > > Thanks for any tips or suggestions! > > -Brian B > _______________________________________________ > 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 > >

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Help Wikify Default Config
by Raymond Chandler 04 Jun '08

04 Jun '08

Community, We need your help. We are in the middle of documenting the default installation of FreeSWITCH. The wiki page that we're working on now is located at http://wiki.freeswitch.org/wiki/Default_config ... We need help with documenting ALL of the modules, especially the ones that are red on the afforementioned page. What we need from the community is for people to help document what they know. There are several modules and config files that are listed that need to be documented. Some of the modules are documented elsewhere, but aren't in the "approved standardized naming convention" of mod_<mod name> We're looking for volunteers to pick a module, or multiple modules, and document them. Basic documentation should include the applications the module provides, the api functions that the module provides, and the files which may contain configuration parameters that can influence the way the module works. To avoid having multiple people running over one another, we've started assiging modules to volunteers. Anyone interested in helping with the documentation project is welcome/encouraged to reply to this email with a module of interest. The modules for which we currently have volunteers are listed below. mod_console - Michael Collins mod_fsv - Raymond Chandler mod_h26x - Raymond Chandler Thanks, -Ray

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weekly conference call reminder
by Raymond Chandler 04 Jun '08

04 Jun '08

Everyone remember the call that's coming up in about two hours. The agenda was sent out yesterday and is attached below for your reference. We look forward to hearing from everyone then. Thanks, -Ray * 1 we still need to finish adopting the orphaned pages <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_04#we_still_need_to_finis…> * 2 there are still quite a few dead-end pages that need linkbacks <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_04#there_are_still_quite_…> * 3 the default config page is still lacking some info <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_04#the_default_config_pag…> o 3.1 I think we should have a list of mods and files that are included in the default config and link them to pages that explain them. <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_04#I_think_we_should_have….> o 3.2 the mod pages should have the list of apps that they include and what they do also. <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_04#the_mod_pages_should_h….> o 3.3 the mod pages should probably also have a list of params that affect the way they behave and links to the files in which to add them <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_04#the_mod_pages_should_p…> * 4 we need volunteers that will take ownership of certain areas of the wiki and not take on more than they can actually handle <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_04#we_need_volunteers_tha…> * 5 preparing for 1.0.1 <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_04#preparing_for_1.0.1> * 6 Main Wiki Call Page <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_04#Main_Wiki_Call_Page>

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VoiceXML?
by Larry Edelstein 04 Jun '08

04 Jun '08

Hey all - Is anyone working on connecting a VoiceXML interpreter to Freeswitch? I've seen a mention or two of a mod_vxml, but that's about all. How hard would it be to write a module that bridged the two? Larry Edelstein Principal Software Engineer LiveVox http://www.livevox.com

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VoIP Engineer in Beijing, China
by Jonathan Palley 04 Jun '08

04 Jun '08

Since I know there are a number of people in the community in China or with connections in 中国, I post this here. Apologizes for the spam. We are looking for a lead engineer for our VoIP Systems. Its FreeSWITCH based but with many other components that integrate with our Web based server. We are growing/expanding it rapidly. An interesting and unique technical challenge. The position is in Beijing. Thanks - Jonathan http://idapted.com | http://eqenglish.com ------------------------------------------------------------ Senior VoIP Engineer - Idapted Inc. Want to revolutionize online learning? Want to be known for developing the leading SaaS learning platform of the next century? We are a Silicon Valley based technology company headquartered in Beijing that is changing online learning. We use cutting edge OSS technologies such as Ruby on Rails, Flex and FreeSWITCH. We are VC funded with leading investors including those who invested in Google, Amazon, built Skype and started New Oriental. We are looking for smart, motivated and creative colleagues to join our world class technology team. We work in an international environment using the latest development practices. Key details below. All potential candidates are requested to provide a resume by email to jpalley [at] idapted.com. Position: Senior VoIP Engineer Responsibilities: Lead the development of our international VoIP system. Location: Beijing, HaiDian Compensation: Highly Competitive (options + salary) Requirements: • In depth experience with VoIP technologies. • Advanced knowledge of at least two of the following: o SIP o FreeSWITCH or Asterisk o RTP • Can define and explain the significance of: UDP/TCP, Jitter, PLC, ulaw/g729, daemon/process, etc. • Real experience coding in C/C++ • Experience with Linux • Experience using scripting languages such as Ruby/Python/PHP/Lua • Strong English Skills Bonuses: • Experience building international VoIP systems • Experience building Ruby on Rails/Ruby • Experience with Call Center applications

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