January 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Blind dialed number
by Edward Q. 29 Jan '10

29 Jan '10

Hi guys. Sorry for been a newbie on all this. And thank you for your help in advanced. Here is the scenario I am trying to accomplish. I would like to have the phone number inside a database not in the extension.xml file. Here is what I am trying to do. Create a let's say 1001.xml file ( if needed ) but that file instead of having the phone number to be dialed, go to MySQL and look for the phone number there. Or when freeswitch is told to call that extension instead of looking for the 1001.xml file go directly to mysql to look for the phone number associated to that extension to dial it out. That way the phone numbers are always kept private at all times. What we are doing is. >From the webserver customer clicks on a link -> link dials the phone number in the xml file automatically but I would like to keep the phone number on the DB server not in the xml file. Can anyone point me please in the right direction? Really appreciated. Ed

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FreeSWITCH Conference Call Agenda
by Michael Collins 29 Jan '10

29 Jan '10

Greetings, This week's conference call agenda is posted here: http://wiki.freeswitch.org/wiki/FS_weekly_2010_01_29 Please add your items as the agenda is very light this week. We do have a few things to discuss, though, so please hop on and bring a friend! Talk to you all soon, Michael

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Set an gateway parameter for external gateway (skype)
by Peter P GMX 29 Jan '10

29 Jan '10

Hello, I have a main freeswitch server and a separate Freeswitch/Skype server with mod_skypiax. I want main freeswitch server to tell the Freeswitch/Skype server to use a dedicated Skype interface(interface1, interface2 etc). What is the best way to pass this variable? Some ideas from my side * set caller_id_number or caller_id_number_name, as these are overwritten by Skype anyway. But this is not a clean solution * use another UDP port for each gateway, but this is a huge effort if the number of gateways becomes larger * set a dedicated sip header for this. Reading sip-header is easy (documented) but how to set my own sip header entry in FS? * any other idea? Best regards Peter

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min_dtmf_duration has not changed
by Magesh R 29 Jan '10

29 Jan '10

Dear All, I am trying to change the min_dtmf_duration value by using fsctl. But it didn't changed. freeswitch@debian> fsctl min_dtmf_duration 800 +OK min dtmf duration: 400 Is there any thing need to be set before changing this? Thanks, Mag.

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Re: [Freeswitch-users] ERR root tag missing
by Milena 29 Jan '10

29 Jan '10

Hello :) First, do not hijack threads, if you want to post about a different problem, do not click reply and then change the subject, please create a NEW MESSAGE. About your question, are you using xml_curl? or any other dialplan seekers? -Milena 2010/1/28 Troy Anderson <troy(a)tlainvestments.com> > I'm seeing this error quite often on my systems: > 2010-01-28 12:35:46.703112 [ERR] switch_xml.c:1571 Error[[error near > line 1]: root tag missing] > > I've looked at freeswitch.xml.fsxml to see if I could find some kind of > malformed XML, but with no luck. Which Is line 1is it referring to? > > Line 1 of freeswitch.xml.fsxml is a comment, with the first actual XML line > like: > <document type="freeswitch/xml"> > > This error always happens right after after a mod_dialplan_xml.c:408 log > message, so I'm led to believe my dialplan XML is messed up, but I cannot > see where. > > In freeswitch.xml.fsxml near the dialplan section, this is what I have: > > ... > <section name="configuration" description="Various Configuration"> >  > </section> > <section name="dialplan" description="Regex/XML Dialplan"> > <context name="my_office"> > <extension name="unloop"> > <condition field="${unroll_loops}" > expression="^true$"/> > <condition field="${sip_looped_call}" > expression="^true$"> > <action application="deflect" > data="${destination_number}"/> > </condition> > </extension> > <extension name="set_domain" continue="true"> > <condition field="${domain_name}" expression="^$"/> > <condition field="source" expression="mod_sofia"/> > <condition field="${sip_auth_realm}" > expression="^$"> > <action application="set" > data="domain_name=10.0.0.120"/> > <anti-action application="set" > data="domain_name=${sip_auth_realm}"/> > </condition> > </extension> > ... > </section> > ... > > Thanks for any ideas! > > -Troy > > > _______________________________________________ > 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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prefix on exten
by FERNANDO VILLARROEL 28 Jan '10

28 Jan '10

Dear. If i receive a call from a customer with some prefx like 1234 How i can do in order to forward this call with out prefix like Asterisk {ENTEN:4} Regards.

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Voicemail in MP3
by John 28 Jan '10

28 Jan '10

I have installed mod_shout and edited the modules.conf.xml to installed it as well. I have updated the user.xml to include <variable name="vm_message_ext" value="mp3"/>.... Messages are still being saved in WAV format. Missing Step? Thanks

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Call hanging up after 30 minutes (exactly)
by Jonas Gauffin 28 Jan '10

28 Jan '10

Hello, I have problems with calls being hung up after 30 minutes. I do not know if the problem is with FreeSWITCH or my sip provider. I've got a log here: http://pastebin.freeswitch.org/11962 Can you please point me in the right direction? Also, why is a second invite sent after 15 minutes? Regards, Jonas

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[fs_cli] Only display SIP messages?
by Fred-145 28 Jan '10

28 Jan '10

Hello Before I learn how to use Wireshark and filter stuff... can the fs_cli console be configured so that only SIP messages are displayed? I'd like to do this so I can learn more about what goes on when I play with SIP clients. Thank you.

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[fs_cli] Only display SIP messages?
by Fred 28 Jan '10

28 Jan '10

At 17:17 28/01/2010, Diego Viola <diego.viola(a)gmail.com> wrote: >You could try this from fs_cli: > >fsctl loglevel 0 >sofia profile internal siptrace on Works great. Thank you.

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