December 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

conference configured to call automatically the attended does not work
by Carole O. 08 Dec '08

08 Dec '08

Hello, I have got some problems for the configuration of a simple conference which should be established by calling an extension and automatically inviting 2 people. Actually, this is based on the default configuration of Freeswitch (extension 0911). I have changed it a little: <extension name="conference_test"> <condition field="destination_number" expression="^0911$">  <action application="set" data="conference_auto_outcall_caller_id_name=telephoneX" /> <action application="set" data="conference_auto_outcall_id_number=0911" /> <action application="set" data="conference_auto_outcall_timeout=60" /> <action application="set" data="conference_auto_outcall_flags=none" />  <action application="conference_set_auto_outcall" data="sofia/internal/1010@$${domain}" /> <action application="conference_set_auto_outcall" data="sofia/internal/1002@$${domain}" /> <action application="conference" data="conferenceTest@default" /> </condition> </extension> I have attached a file with the console errors. There are some errors (moh errors) but since these were also present for room conference and it did not prevent it for working, I guess this is not the fundamental reason for the previous problem. I have an additional question. I have installed freeswitch from opensuse.org, there is a simple "one-click installation" but I am not sure this was a good idea, it seems to be light isn't? Thanks for your help, Carole http://www.nabble.com/file/p20856465/error_console.txt error_console.txt -- View this message in context: http://www.nabble.com/conference-configured-to-call-automatically-the-atten… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Autoanswer
by Jonas Gauffin 08 Dec '08

08 Dec '08

Hello I send an API command through the event socket that looks like this (the first two variables are used by our server): api originate {gate_user_id=44,gate_site_id=1,sip_invite_params=intercom=true,sip_h_Call-Info=<sip:$${domain}>;answer-after=0,sip_auto_answer=true,origination_caller_id_name='Jonas Gauffin',origination_caller_id_number=+4623666XXXX,sip_auto_answer=true}sofia/localdomain/u1000044% 192.168.1.112 '&execute_extension(8901 XML)' The command works just as it should. The problem is that the auto-answer variables seems to stick to the b-leg (execute extensions), which means that both calls gets auto answered. What I want is that only the first call gets answered. (A logged in user presses the "call" icon in our webdirectory, which makes freeswitch use his phone to call 8901) //Jonas

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Re: [Freeswitch-users] Handling directory of sound files
by Michael S Collins 07 Dec '08

07 Dec '08

Check out mod_localstream on the wiki and see if that sounds like what you need. I'm still learning it all myself but I believe that's where you should start. Please report back with any questions and we will take it from there! -MC On Dec 5, 2008, at 3:48 AM, Faisal Maqsoodi <faisalmaqsoodi(a)yahoo.com> wrote: > Its not without music on hold completely. Say, e.g, moh is being > played but when i press 1 it should start playing files contained in > a specific directory sequentially or randomly. I havent got any > solution to this problem yet. Can anyone plz guide me to some > documentation or anything else regarding this matter. > > > > > > > > > > > > > > > > > Faisal > > _______________________________________________ > 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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VoIP Product/Service and Man/Woman of the year 2008
by UV 07 Dec '08

07 Dec '08

Biz-news.com <http://www.linkedin.com/redirect?url=http%3A%2F%2FBiz-news%2Ecom&urlhash=oN Bl&_t=disc_detail_link> is conducting a survey to discover the leading VoIP man/woman of the year 2008, and the best VoIP product or service also for the year 2008 . The winner will be selected based on the opinions of professionals and technology enthusiasts in the industry. A) best VoIP product or service of 2008 form: http://voip.biz-news.com/forms/py2008 <http://www.linkedin.com/redirect?url=http%3A%2F%2Fvoip%2Ebiz-news%2Ecom%2Ff orms%2Fpy2008&urlhash=dspi&_t=disc_detail_link> B) VoIP Man/Woman of the year 2008 form at: http://voip.biz-news.com/forms/my2008 <http://www.linkedin.com/redirect?url=http%3A%2F%2Fvoip%2Ebiz-news%2Ecom%2Ff orms%2Fmy2008&urlhash=eDrg&_t=disc_detail_link> The results will be published early 2009 and share the raw data with the community.

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Playing a file again and again
by Faisal Maqsoodi 06 Dec '08

06 Dec '08

Hi, Is there any built-in function, like playback, which plays a file again and again unless interrupted. I want to use a simple function not FIFO. Faisal

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Windows is slow?
by Per Møller 06 Dec '08

06 Dec '08

I have installed FS 1.0.0 on a Mac using the precompiled .dmg and FS 1.0.1 on a Windows Vista machine using the precompiled .msi - actually the same machine). Using the default configuration files, and using 2 Snom 360 phones I dialed from extension 1000 to extension 1001. On the Mac, 1001 starts ringing instantly, but under Windows it takes 1-2 seconds before it starts ringing. It seems to be in the dialplan the time is spent. From the time I see this line on the console: [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000->1000 in context default Until the next thing happens it always takes at least 1 full second, but on the Mac it happens instantly. Why is the Windows build this much slower? Is it a known problem? I get the feeling that the majority of the FS community is Unix based, which is fine by me, but I would really like to know just how well supported and stable the Win32 build is and if this is currently a viable way to go, or if I should stick to Linux/BSD/Mac for production use? // Per

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How to setup TLS between two Freeswitch servers
by shehzad p 06 Dec '08

06 Dec '08

I am wondering how to setup two freeswitch servers to route call with TLS configured between them. As shown in wiki http://wiki.freeswitch.org/wiki/SIP_TLS, I created two certificates on one freeswitch, and changed SIP profile by enabling tls in it, then Starting freeswitch it just opens port 5061 (for TLS ), But when i route the call from that FS server, it uses the its general ports (5060 and 5080) for call. Where i am missing something?, A doubt is about where to place which certificate. -- View this message in context: http://www.nabble.com/How-to-setup-TLS-between-two-Freeswitch-servers-tp208… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Provider: Junction Networks
by mehdix 06 Dec '08

06 Dec '08

I've got a problem with configuring a SIP trunk from Junction Networks with FS: it only works for a few minutes then the line is dropped. I tried Unlimitel with no problem. Any Ideas? Thanks -- View this message in context: http://www.nabble.com/Provider%3A-Junction-Networks-tp20859688p20859688.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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How to force FS using the hosts entry in outgoing gateway calls
by Jon Bruel 06 Dec '08

06 Dec '08

For the configuration of a gateway I need to use a specific proxy domain name before the server (Covergence SBC with a BroadWorks Application Server behind) accepts calls. The twist is that the right proxy name points the wrong IP-address (the voicemail server for the account). I have tried to overrule this by adding a host entry (Linux). When I ping to the domain name I get the right address (the one from the host table), but the FS uses the address from the DNS lookup, not the address from the host table. What can I do to force the FS using the entry from the host table? Thanks /Jon.

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Re: [Freeswitch-users] How to force FS using the hosts entry in outgoing gateway calls
by Jon Bruel 05 Dec '08

05 Dec '08

Thanks Anthony. Using the parameters: <param name="proxy" value="domain name"/> <param name="register-proxy" value="ip address"/> Returns error 900, and a 'ngrep port port-number' indicates that its doesn't try to register at all. I have now let the server look at a local DNS where I have added a "wrong" A-record. That solves the issue, but your solution would be cleaner. The version is: trunk 10220. /Jon

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