November 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Call from Secure RTP to non-secure RTP
by Mark Campbell-Smith 19 Nov '09

19 Nov '09

Hi! How do I setup FS so that placing a call from an extension that only support SRTP (1002) to an extension that only supports RTP (1000)? I put this dialstring, from the wiki http://wiki.freeswitch.org/wiki/Tls, into the users xml file under directory/default <param name="dial-string" value="{sip_secure_media=${regex(${sofia_contact(${dialed_user}@${dialed_domain})}|transport=tls)}, presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" /> I have also put a <action application="export" data="sip_secure_media=true"/> when 1000 is dialing 1002. <condition field="destination_number" expression="^(1002)$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="export" data="sip_secure_media=true"/> <action application="bridge" data="user/${dialed_extension}@${domain}"/> However I never see crytpo sent in the RTP to 1002 and it responds with Bad Security Level What have I missed? Thanks

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mod_fifo and multi-tenancy
by mayamatakeshi 19 Nov '09

19 Nov '09

About mod_fifo, it would be safe to use it in multi-tenancy scenarios where domains are created and deleted all the time and in consequence, fifos are created all the time? I mean, fifos are eventually destroyed by mod_fifo itself. Is this correct? br, takeshi

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Anybody interested in helping fix the -srcdir option?
by Robert Hadley 18 Nov '09

18 Nov '09

Hi All, Anybody interested in helping fix the -srcdir option? I am trying to build in a subdirectory off the Freeswitch source. I am working on it and finding issues. However, being a newbie at autoconf/automake and shell scripting I sometimes struggle at finding fixes. For example, the script command below is in bootstrap.sh, but might need to be moved or duplicated in configure.* to support using configure -srcdir option, as the modules.conf file also needs to be to the build destination folder. if [ ! -f modules.conf ]; then cp build/modules.conf.in modules.conf fi Thanks, Robert

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Build FS without spandsp or libtiff
by Kristian Kielhofner 18 Nov '09

18 Nov '09

Hello everyone, I'm trying to keep my build as small as possible (for AstLinux). Is there anyway to build without spandsp or libtiff? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

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Registration Error - 408 timeout
by Samuel Abekah-Mensah 18 Nov '09

18 Nov '09

Hello Please pardon me if the solution to this is somewhere already that I have been unable to locate. I have just got a straight out-of-the-box build of FS. According to the wiki, I should be able to test using user IDs 1001 and 1002. However, I am get the above error. If I, however, un-tick register with domain I do net get the error but does not communicate either. Is there a conf that I should have done ? I am using X-lite3 Thanks in advance. Abeka

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Compilation problem
by Christopher Z. 18 Nov '09

18 Nov '09

Hi, I've got this error after make: http://pastebin.freeswitch.org/11145 Any idea how to fix this error ? Thanks.

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acl configuration
by Jonas Gauffin 18 Nov '09

18 Nov '09

Hello, What should my acl.conf.xml look like if I want to allow ALL calls on the external profile and use only digest authentication on all other profiles? Thanks, Jonas

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change event value
by Eli Hayun 18 Nov '09

18 Nov '09

Hi Is there is a way to intercept an event (for example : REGISTER) and change one of its parameters (for example: the extension number) and fire up the corrected event? I need it to set the speedial of the phone value to be "**xxxxx" but to make it register as "xxxxx" Thanks Eli

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Rewriting SDP with switch_r_sdp
by Apostolos Pantsiopoulos 18 Nov '09

18 Nov '09

I am trying to use switch_r_sdp to rewrite the SDP. The problem I am facing has to do with the way of doing it. Let's say I have: v=0 o=- 1258463684 1258463684 IN IP4 xxx.xxx.xxx.xxx s=Opal SIP Session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 5144 RTP/AVP 18 3 101 120 c=IN IP4 xxx.xxx.xxx.xxx a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:3 gsm/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16,32,36 a=rtpmap:120 NSE/8000 a=fmtp:120 192-193 who to I set the switch_r_sdp variable in xml? Obviously this doesn't work : <action application="set" data="switch_r_sdp=v=0 o=- 1258463842 1258463842 IN IP4 xxx.xxx.xxx.xxx s=Opal SIP Session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 5148 RTP/AVP 18 3 101 120 c=IN IP4 xxx.xxx.xxx.xxx a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:3 gsm/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16,32,36 a=rtpmap:120 NSE/8000 a=fmtp:120 192-193 "/> Do I have to escape any special characters and how? I tried using escaped quotes, escaped spaces, escaped tabs etc. Nothing worked. Any suggestions? -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs(a)kinetix.gr -------------------------------------------

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application="info"
by Mark Campbell-Smith 18 Nov '09

18 Nov '09

HI All, pretty basic question and I feel a bit stupid asking this, but what are the prerequisites for the INFO to be displayed when <action application="info"/> is called in a dialplan? ie are there requirements on the loglevel, does the INFO command have to be put at a certain place in the dialplan etc? The reason i ask is that I have a dialplan and the <action application="info"/> is not getting triggered on the fs_cli output. Is there some other debbugging level that needs to be set? Thanks!

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