August 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

BLF and Registration State
by Kevin Wormington 15 Aug '13

15 Aug '13

Hi, I'm doing some testing of BLF with Freeswitch and Yealink T28 phones. It works fine as far as flashing the indicator when a monitored extension is ringing, solid red when in use, etc. However, the indicator is green even when an extension is not registered. I haven't found much documentation on this, but it seems the indicator should be unlit if the extension is not registered. If I manually clear the registration with sip tracing on I see that FS sends the below content in the NOTIFY and gets a 200 OK back from the T28: Subscription-State: active;expires=1175 Content-Type: application/dialog-info+xml Content-Length: 155 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="13" state="partial" entity="sip:1102@x.x.x.x"> </dialog-info> I assume this is not working because the phone is expecting something else in order to turn off the light. Is this something that is configurable or would it need code changes? Also, is it possible to use an OPTIONS ping or otherwise to determine if a registered endpoint is available rather than wait for the expiration? I know this can be done on un-registered gateways. Thanks, Kevin

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deny_refer_requests failure
by Ryan Gard 15 Aug '13

15 Aug '13

Hey, Running into issues with deny_refer_requests. I have verified that it is setting the variable appropriately on both the a-leg and b-leg of the call, but it still processes the refer request without a second thought upon receiving it. Is there anything specific on top of setting the variable to 'true' I should be keeping tabs on? It doesn't seem to have too much in the way of documentation, but the references I have seen say that it should work without issue as long as one of the legs (both legs have it set) has it set to true. Thanks :) -- Ryan Gard

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Potential FreeSWITCH support for SMPP
by William King 15 Aug '13

15 Aug '13

Is anyone on this list interested in FreeSWITCH support for SMPP for SMS messages? For more information about the SMPP protocol checkout: http://en.wikipedia.org/wiki/Short_Message_Peer-to-Peer If so feel free to contact me on or off this list. -- William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king(a)quentustech.com

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Handle individual SIP messages
by Grant Bagdasarian 15 Aug '13

15 Aug '13

Hello, Is there a way to process individual SIP messages in the dialplan or somewhere else? When receiving a 200 OK from our Carrier, I'd like to process (add some custom headers) the 200 OK manually before I send it off to the next hop (SIP Proxy server). Regards, Grant

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garbled audio with G726-32, other codecs are fine
by Ivan Mitev 15 Aug '13

15 Aug '13

Hello I'm migrating an office setup from asterisk to FS and in the process I was considering using G726-32 for some bandwidth starved remote endpoints. However I only get metallic/garbled audio with that codec even when simply playing moh to the endpoint, while other codecs work fine (G711U/A, G722, GSM). G732-16 is inaudible, G732-40 sounds marginally better but still garbled and really worse than G711. The setup is FS 1.2.12 from FS' yum repo on a centos6 64bit KVM guest (centos6 64bit host). But please don't shoot ! :) - I know about virtual environment limitations but for these tests the host is only lightly loaded, there aren't any calls to the FS instance except my tests, and the fact that it works with other codecs makes me think that virtualization is not the issue here. I may be wrong though. Is there any guide for debugging that kind of problem before reverting to a fresh install on bare-metal with the latest HEAD ? Until now I've tried: - improving timers ; but the default soft timer (which I guess uses timerd) works best. The time interval between sent packets on a tcpdump trace looks identical to the output of "timer_test", so that doesn't seem to be a network/jitter problem. And there's no problem with other codecs, but maybe G726-XX is specific. For info the guest's clocksource is kvm_clock, while the host uses tsc. - using different endpoints: the production ones are Linksys PAP2 ("fixed" for 20ms psize, and G726-32 SDP type indentification), but the same thing happens with linphone on a fedora 19 laptop. A call with rtp media going through FS without transcoding - G726-32 to G726-32 - works perfectly (I can't hear the difference with G711). The problem is only when there's transcoding to G726 (from wav for moh, or from any other codec when bridging). I've looked at the wiki, posts, changelogs, jira, ..., but am a bit at a loss now. Any pointers ? Except that little problem, FS rocks, and I'm happy I can finally ditch asterisk. Kudos to the core devs and contributors. Ivan

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Httapi: getting digits in playback
by Denis Gasparin 15 Aug '13

15 Aug '13

Hi. After playback I ask to the user to digit a number between 1 and 9999 with or without terminating the input with '#'. The first httapi xml I tried was the following: <document type="text/freeswitch-httapi"> <work> <playback file="ivr/please_enter_number_between_1_9999.wav" digit-timeout="5000" name="user_input" terminators="#"> <bind>~\d{1,4}#{0,1}</bind> </playback> </work> </document> Freeswitch calls my action as soon as the user input the first digit. So I modified the xml in order to use two bindings: <document type="text/freeswitch-httapi"> <work> <playback file="ivr/please_enter_number_between_1_9999.wav" digit-timeout="5000" name="user_input" terminators="#"> <bind>~\d{1,3}#{0,1}</bind> <bind>~\d{4}#{0,1}</bind> </playback> </work> </document> Using this xml I get the user input only if the user press "#" after inserting the number. If the user presses 123 and then waits for timeout, Freeswitch doesn't send any digits to my action (but sometimes it does). Why? >From the logs I see that the digits are always received from Freeswitch. Thank you in advance for your help. Denis Gasparin

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Re: [Freeswitch-users] Recording conference only when a particular user is connected
by Matthew Cordes 15 Aug '13

15 Aug '13

Hi Stanislav, For a) (below) how do I stop the recording? Will that happen automatically when the user hangs up even though there are others still in the conference? If so, this sounds like it could work. Thanks, -Matt On Wed, Aug 14, 2013 at 7:19 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > ---------- Forwarded message ---------- > From: Stanislav Sinyagin <ssinyagin(a)yahoo.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 14 Aug 2013 15:26:12 -0700 (PDT) > Subject: Re: [Freeswitch-users] Recording conference only when a > particular user is connected > some possible scenarios, need further digging: > > a) for that user, execute the "record" dialplan application, then transfer > to the conference extension. It can be achieved by a1) setting a variable > in the user directory entry, and then matching the variable in the > dialplan; a2) defining a separate context for the user. > > b) an external script would monitor the ESL events and execute uuid_record > when needed >

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Recording conference only when a particular user is connected
by Matthew Cordes 14 Aug '13

14 Aug '13

Hello, I'd like to record a conference call, but only when a particular user is part of the call and even when the conference just contains that user. When that user leaves the conference, even if there are other users in the conference I want to stop recording. Any recommendations on the best approach? -Matt

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FreeSWITCH Security Topics...
by Brian West 14 Aug '13

14 Aug '13

We need to start the security discussion in terms of FreeSWITCH and security related topics. It was a very popular topic at ClueCon 2013 and I would like to continue the discussion. freeswitch-sec(a)lists.freeswitch.org for those wanting to join in. Thanks, -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

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loopback problem with lua and rxfax
by Babak Yakhchali 14 Aug '13

14 Aug '13

Hi I'm using these 2 extensions for receiving faxes. to receive a fax first call hits "faxin" extension and then is bridged to "dofaxin". I'm using loopback to be able to handle attended transfers from ivrs that include fax. the problem is if I send a call directly to 704 everything works fine. but if I send it to 705 rxfax (which is the main case)it is not working as expected and after a long waiting rxfax fails with " The call dropped prematurely". <extension name="dofaxin"> <condition field="destination_number" expression="^704$"> <action application="answer"/> <action application="playback" data="silence_stream://2000"/> <action application="set" data="fax_enable_t38_request=true"/> <action application="set" data="fax_enable_t38=true"/> <action application="lua" data="lua/applications/fax.lua"/>  </condition> </extension> <extension name="faxin"> <condition field="destination_number" expression="^705$"> <action application="bridge" data="loopback/704/public/XML"/> </condition> </extension> and the simplified version of file "lua/applications/fax.lua" is: ... session:execute('rxfax','/tmp/test.tiff') ....

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