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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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Investigating one-way sound?
by Fred-145 28 Jan '10

28 Jan '10

Hello With both a PC+XLite and a Siemens A580IP phone on the same LAN, sound is OK both ways when I call from the XLite application, but when calling from the Siemens, I can't hear sound coming from the XLite application. FWIW, both phones are configured to use G711a and G711u, in this order. To investigate, what kind of error message should I pay attention to in all the messages that scroll through the console? Thank you. -- View this message in context: http://old.nabble.com/Investigating-one-way-sound--tp27341219p27341219.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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compilation error on fs_cli (Windows) (latest version)
by Diego Toro 28 Jan '10

28 Jan '10

Hi, I have compilation error "error C2220" on fs_cli project on Windows using VS2008. FS: latest version (2010/01/26) VS: VS2008 SO: Windows 7 VS2008 Error log: Error 1 error C2220: warning treated as error - no 'object' file generated g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c 106 fs_cli Warning 2 warning C6385: Invalid data: accessing 'global_profile->console_fnkeys', the readable size is '48' bytes, but '-4' bytes might be read: Lines: 86, 88, 90 g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c 90 fs_cli Warning 3 warning C6246: Local declaration of 'p' hides declaration of the same name in outer scope. For additional information, see previous declaration at line '844' of 'g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c': Lines: 844 g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c 895 fs_cli Warning 4 warning C6011: Dereferencing NULL pointer 'cursor': Lines: 839, 840, 841, 842, 843, 844, 846, 849, 853, 857, 868, 870, 871, 884 g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c 884 fs_cli Thank you Diego Toro http://lacarretade.blogspot.com/

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Question about Lua script: How do I detect when someone picks up the phone?
by Mike van Lammeren 27 Jan '10

27 Jan '10

Hello! Some day, I'll probably laugh at asking this question, but today I can't figure it out. I've written a Lua script that listens for a call, then dials a phone number to a second person. It plays a message, then prompts the second person to hit pound to connect. If the second person hits pound, then it bridges the two calls together. All that works great, but I can't figure out how to get the session for the second person to wait until that person answers. I'm using FreeSWITCH 1.0.4, and although there is a *getState* function documented in the wiki, it doesn't seem to exist for me. Any help would be appreciated! Mike van Lammeren

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