- FreeSWITCH-users - FreeSWITCH Mailing Lists

unable to make outbound fax
by Louie Liu 07 Feb '12

07 Feb '12

Hi, I've got problem with outbound fax, the fax machine is connected to FS server via ATA device in a pass through mode, inbound fax works fine, but could get outbound fax to work, here is the dialplan for it, do I miss any parameters? What's wrong with the dialplan? Fax_A (4919)->ATA_A->SIP->FreeSWITCH->SIP provider->PSTN->Fax B <extension name="t38_passthru"> <condition field="destination_number" expression="^61000$"> <action application="export" data="t38_passthru=true"/> <action application="bridge" data="{sip_invite_domain=$${domain}}user/${dialed_extension}@${domain_name}" /> </condition> </extension> Cheers, Louie

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Setting ptime in inbound and outbound call
by Arafath-uz-zaman khan 07 Feb '12

07 Feb '12

Hello Is that possible the given diagram in freeswitch. G729 40 ptime -----> G729 -----> 20 ptime User Agent ------------------------------------ Fresswitch ------------------------------------ PSTN GW G729 60 ptime <----- G729 <----- 20 ptime If it is possible to do that can any one guide me how can i do that. With Regards -- Arafath-uz-zaman khan

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One way Audio when dialing from one end.
by Tariq Mahmood 07 Feb '12

07 Feb '12

Hi, I hope you guys are doing great. I am having an issue with my number, When i dial the number with my asterisk server, the callee can hear me but i cant hear him. But if the same guy call me, the conversation goes great. Any Help??? -- Best Regards Tareq Khan

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I'm Alive!
by Brian West 07 Feb '12

07 Feb '12

I would like to thank everyone that has donated money to help me pay for all the expense of having a stay in the hospital. It was a pretty scary week for me last week. If it were not for the morphine last week would have sucked even more! I have missed out on the annual Developer meeting in Milwaukee this year due to this event. So please make sure you guys donate some money for the dinner for those guys. Again Thank you all for your support. -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian(a)freeswitch.org http://www.freeswitch.org

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Freeswitch sharing same local host with sip server
by Wayne 07 Feb '12

07 Feb '12

Hello all, I am using Freeswitch for a Voicemail server as a part of a larger project. It is sharing the same local host with a different SIP server. The other server is setting on port 5070. Freeswitch is sending calls from 5080 to 5070 the call gets connected and works fine. Currently I am testing with sip phones registered on both systems. When I hang up on the phone registered with freeswitch the call does not get torn down on the other sip server. A whireshark trace shows the hang up going to port 5060 not 5070. So Freeswitch is sending it's self a disconnect message on port 5060 and replies with a 481 call does not exist. What am I missing? Wayne

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Re: [Freeswitch-users] xml_curl dialplan and static files
by Kyle Haefner 06 Feb '12

06 Feb '12

Hi Michael, Awesome.. I was actually very close, I was trying the "not found" route but had the section name as "dialplan", changing it to "result" made it work. thanks! At CSU we use Freeswitch as part of sipXecs and BigBlueButton production deployments. Really, though, Freeswitch is our go-to for anything that needs to be customized or scripted. The most recent example of this is using the xml_curl function to build a pre-set conference for emergency situations. Freeswitch dials out to a group of high-level administrators and once they answer and prove they are human by pressing a number key they are bridged into a conference. This a good way to get groups of people like our Public Safety Team together. Thanks! Kyle On Mon, Feb 6, 2012 at 2:36 PM, Michael Collins <msc(a)freeswitch.org> wrote: > Kyle, > > Welcome to FreeSWITCH! We'll be happy to help answer your question. In return we would just like to know what you guys are up to over there at CSU. :) Are you using FreeSWITCH for anything in particular? > > As far as your question goes, you'll most likely need to employ the "dynamic config w/ static fallback" setup. The way it works is that if your xml curl program returns a special "not found" config then FreeSWITCH will go looking in the static XML for the relevant information. The trick for you will be to return the "not found" message any time that you want the static XML to effect. The "special" config is this: > > <?xml version="1.0" encoding="UTF-8" standalone="no"?> > <document type="freeswitch/xml"> > <section name="result"> > <result status="not found"/> > </section> > </document> > > So, you'll need your xml curl script to recognize which dialplan contexts to handle and which to send back w/ the "not found" message. > > Let us know how it goes. > -MC > > On Mon, Feb 6, 2012 at 11:58 AM, Kyle Haefner <Kyle.Haefner(a)colostate.edu<mailto:Kyle.Haefner@colostate.edu>> wrote: > Hi All, > > I have not been able to glean enough from the wiki and Google searches > to figure this out. > > I have freeswitch xml_curl fetching a dial plan. What I would *like* > to do is have one context fetched > dynamically while another context is from a static file (or fail-back > to a static file). > > I would appreciate any help. > > Thanks! > > > > > -- > Kyle Haefner, M.S. > Communication Systems Programmer > Colorado State University > Fort Collins, CO > Phone: 970-491-1012<tel:970-491-1012> > Email: kyle.haefner(a)colostate.edu<mailto:kyle.haefner@colostate.edu> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kyle Haefner, M.S. Communication Systems Programmer Colorado State University Fort Collins, CO Phone: 970-491-1012 Email: kyle.haefner(a)colostate.edu

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xml_curl dialplan and static files
by Kyle Haefner 06 Feb '12

06 Feb '12

Hi All, I have not been able to glean enough from the wiki and Google searches to figure this out. I have freeswitch xml_curl fetching a dial plan. What I would *like* to do is have one context fetched dynamically while another context is from a static file (or fail-back to a static file). I would appreciate any help. Thanks! -- Kyle Haefner, M.S. Communication Systems Programmer Colorado State University Fort Collins, CO Phone: 970-491-1012 Email: kyle.haefner(a)colostate.edu

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FreeSwtich DB
by Joe Flemmings 06 Feb '12

06 Feb '12

Is their a reason why freeswitch still creates DB files in /usr/local/freeswitch/db/ even when using ODBC at the core as decsribed here http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core

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No outbound audio on new fs build
by Ira Tessler 06 Feb '12

06 Feb '12

The following are two pcap files that have one outbound call in them. Everything is identical except the version of Freeswitch. With the later version of Freeswitch, I am getting one way audio on the outbound call. On the later version of Freeswitch, there is an Update sip message that is not in the older version. Can anyone help me and let me know how I can fix the one way audio? I only get the one way audio when using Yealink phones behind an Edgemarc router. Old version of Freeswitch pcap: http://dl.dropbox.com/u/47191639/lan_side_yealink_t28_good_audio_edgemarc_o… New Version of Freeswitch pcap: http://dl.dropbox.com/u/47191639/lan_side_yealink_t28_no_audio_edgemarc.pcap Thanks, Ira Tessler ConnectMe (732) 490-9007 x2 www.connectmevoice.com

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Re: [Freeswitch-users] Getting USER_NOT_REGISTERED for no reason ?
by Bob Smith 06 Feb '12

06 Feb '12

Welcome back by the way Brian... ;-) If only it were that easy...... in conf/directory/default.xml I changed.... <param name="dial-string" value="{sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> to read <param name="dial-string" value="{sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain}),sofia/other_profile/${dialed_user}}"/> Then the OpenVPN phone rings, but the internally registered phone does not ? > You'll have to modify the dial-string for the user 1000 to make sure it searches this new profile ...

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