March 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Check if UA is still there?
by Peter Steinbach 03 May '13

03 May '13

Hello, is there a way to check if a UA is still available? The reason is, that sometimes a channel hangs and is not being terminated because the UA was disconnected due to a network failure. As the media is outside Freeswitch (bypass media), it is difficult for us to determine, if the call is still active. Meanwhile the B leg to PSTN is also not hungup and produces costs. Is there a way Freeswitch can detect this? Or is there a way Freeswitch may send a SIP message to the UA by an external process and wait for an OK which can be captured e.g. by event socket interface? An help is welcome. -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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page application
by Abaci 25 Apr '13

25 Apr '13

There is an application "page" (in mod_dptools), I didn't find any documentation for it on the wiki so I'm sending here if someone can provide details how to use it. see http://fisheye.freeswitch.org/browse/freeswitch.git/src/mod/applications/mo… Thanks

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SwitchPi - FreeSWITCH Meets the Raspberry Pi with some small Open Hardware Addons
by Ken Rice 22 Apr '13

22 Apr '13

Hey Guys, May of you know I have been working to get FreeSWITCH crammed on to the RaspberryPi. And a few of you have noticed where I have been blogging about it at http://www.switchpi.org/ well its come time to ask for a little help... If you have an EE background and have recently sent boards sent off to a short run proto fab and want to assist in a little refinement of the PCB, I think its time to start heading down that road. Please contact me off list with how you can help. Where am I currently? A) FreeSWITCH Building cleaning on Raspbian (I¹ll be making an image with FS precompiled and installed shortly), B) breadboard hardware working with Python and limited C sources. Eagle Files and Source code is on Github (Linked off the blog for easy reference) K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org http://www.switchpi.org irc.freenode.net #freeswitch

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default timeout
by Stephen Thwaites 15 Apr '13

15 Apr '13

Hi All, Apologies for the simple question but I can't find the answer anywhere in the books, wiki, or our friend google. If somebody calls us, FS does an ORIGINATE_CANCEL after 60s but the follow-me scheme we have configured is100s. How can I increase the default call_timeout of 60s to 100s? Or maybe I am just doing something wrong! I have tried leg_timeouts, originate_timeouts both on the transfer and the bridge as well to no avail? Would be very grateful for any help or advise. Regards, Steve Some Details: - External call comes in on external profile from our voip provider. - Dialplan in the public context does a transfer to a follow-me extension 7777 in context creche-babys <action application="transfer" data="7777 XML creche-babys "/> - The 7777 extension is as follows and the call hangs up part way through the third step if nobody picks up (after 60s total). <extension name="creche-fm"> <condition field="destination_number" expression="^7777$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="ringback=${nl-ring}"/> <action application="set" data="call_timeout=25"/> <action application="bridge" data="{ignore_early_media=true}user/21@${domain_name},user/20@${domain_name}"/> <action application="set" data="call_timeout=25"/> <action application="bridge" data="{ignore_early_media=true}user/22@${domain_name}"/> <action application="set" data="call_timeout=25"/> <action application="bridge" data={ignore_early_media=true}user/23@${domain_name},user/24@${domain_name},user/26@${domain_name},user/27@${domain_name},user/28@${domain_name}"/> <action application="set" data="ringback=${nl-ring}"/> <action application="set" data="call_timeout=25"/> <action application="bridge" data="{ignore_early_media=true}sofia/gateway/3120 ...voipprovidergateway/06 ...mobile number"/> </condition> </extension>

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MPL v2
by Puskás Zsolt 12 Apr '13

12 Apr '13

Hi. A new version of the MPL license just came out ( http://www.opensource.org/licenses/MPL-2.0 ). Is the new version better than the last one? I'm trying to find the differences but this kind of text is to hard for me :) Any plans for changing FreeSWITCH license to the new one ? Zsolt

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Unloop extension
by Ержан Туле пов 11 Apr '13

11 Apr '13

What unloop extension is used for? BTW: vars.xml pre-(process)-sets unroll_loops, whereas DPs (public and default) are using channel variable syntax? Is this ok? Y.

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Re: [Freeswitch-users] problems with freeswitch + zrtp in proxy-media mode
by Eli Burke 09 Apr '13

09 Apr '13

Yes, inbound-zrtp-passthru is true. I think it gets set automatically when you turn on ZRTP. The config I'm testing with is a stock 'enable_zrtp=yes' build of HEAD circa last week, with proxy-media=true added to Local_Extension in the default dial plan. -Eli On Feb 1, 2013, at 6:00 AM, royj(a)yandex.ru wrote: > Is there <param name="inbound-zrtp-passthru" value="true"/> in profile > > On Thu, 31 Jan 2013 17:43:34 -0500 > Eli Burke <eburke(a)edge-net.net> wrote: > >> I'm trying to use Freeswitch with an open source voip client called Linphone, but cannot manage to establish functioning ZRTP sessions. The problem *seems* to be related to the RTP stream's SSRC. I'm trying to conclusively determine if the problem lies with Linphone, Freeswitch, or one of the underlying zrtp implementations (libzrtp and zrtpcpp). >> >> Here's what I've determined so far: >> 1) ZRTP support is compiled and functioning. I can establish ZRTP sessions between two iPhones running Groundwire. >> 2) linphone + ZRTP works correctly if I make a one legged call to a Freeswitch echo extension >> 3) linphone + ZRTP works correctly if I turn on bypass-media >> 4) linphone gets a CRC error when trying to verify the ZRTP packet on any two-legged call >> 5) disabling the CRC check results in a successful ZRTP session, but the audio is extremely garbled >> >> Is Freeswitch modifying the ZRTP packets when running in proxy mode? If so, how is it that some clients are able to handle this situation?

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zrtp passthrough
by Kosztolányi Péter 09 Apr '13

09 Apr '13

I need to set up a passthrough zrtp in freeswitch. I followed this wiki: http://wiki.freeswitch.org/wiki/ZRTP and setup two Jitsi softphones. MiTM: works well but with different SAS in the softphones. It is ok because of a missing zrtp enrollment process. Passthrough: no SAS at all and Jitsi says the media is not encrypted I tried to achieve passthrough by uncommenting sip_profiles/internal.xml: <param name="inbound-proxy-media" value="true"/> What is the proper way to set up passthrough zrtp and having an end-to-end zrtp session with the same SAS on both softphones? Thank you, Peter

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FAX problems
by Steven Schoch 08 Apr '13

08 Apr '13

Thank you guys, I just sorted out the multiple sofia profiles and have FS accepting SIP registrations on our LAN. I plugged a phone into the LAN and it works. Now I'm trying to get this VoIP Plug-In for Microsoft Fax to work, that I downloaded from www.faxback.com. I configured it with a directory number I created, the dialplan seems to work, but it fails when sending the test FAX. The FAX number I'm calling is a POTS line connected to a Fax modem, which is answered by Hylafax. This works and is used everyday. The faxback software is installed on a Windows 7 machine, and configured by putting in a proxy address of 192.168.1.17, a username and password, and address of user@domain. The test call rings the fax modem, it answers, but then after some handshaking, it hangs up and the window says the call was hung up. Hylafax says: "No answer (T.30 T1 timeout)" -- Steve

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adding In-Reply-To in sip header not working
by Mario G 08 Apr '13

08 Apr '13

I never did this so I must be missing something, I tried both below but the bridge then fails. Can anyone shed some light on what I am doing wrong. My ITSP now supports in-reply-to so I can pass the caller ID to a forwarded call from FS. Mario G <action application="set" data="sip_rh_In-Reply-To=${caller_id_number}"/> <action application="set"><![CDATA[sip_h_In-Reply-To=<sip:${caller_id_number}>]]></action>

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