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Issues with blind transfers involving skinny phones
by Nathan Neulinger 26 Mar '13

26 Mar '13

Noticed some issues with blind transfers during some testing today, current head. SIP ext -> * Blind transfer to another SIP ext or external number works fine SIP ext -> * Blind transfer to Skinny ext drops the other leg as soon as I push the transfer button the second time - and the target extension keeps ringing Skinny ext -> SIP ext Blind transfer to Skinny ext acts like it was non-blind until the target extension answers - i.e. first two show with an active call, third is ringining. I'm wondering if this might have anything to do with the ring-ready changes that were done in FS-5180 / 84709b8b610f71aec4afec8f38f3cbcd813c01ec -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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DTMF does not work from PSTN(cellphone or your home phone)
by Sayyed Mohammad Emami Razavi 26 Mar '13

26 Mar '13

No digits are passed from cellphone or your home phone but when you press any key from your registered IP phone on local lan, digits are passed very well, what is the problem? This problem is new and beforehand i had no problem with dtmf from cellphone! Every thing is good, all logs are good, my IVR that gets digits and transfer calls are good and no problem exists in these aspects. May my trunk kill or resolve or delete all DTMF signals on sofia sip?! Is problem from some configuration in sofia?! any idea?

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refer doesn't always creates a new channel
by Michel Brabants 26 Mar '13

26 Mar '13

Hello, I've noticed that in 1/2-cases refer doesn't create a new channel. It just doesn't do anything ... I also noticed that there were dead channels listed in "show channels" while no calls were present anymore. This is in freeswitch version 1.2.5.3. If it never worked, I would "understand" it, but this seems to be a bug or misconfiguration ..., but if it really was misconfigured I suppose that it shouldn't work at all ... I can see the bridge-command being listed as to be executed after the refer-message, but it doesn't create a new channel (all the rest, sets, exports, is executed). There's just nothing related to the new channel. The only thing I can see if the old channel/leg being destroyed by the refer-message. In the end I can see I'm trying to update to 1.2.7, but I don't really like to do this as I performed all my earlier tests on 1.2.5.3. Anyway, if somebody could tell me whether I should really pay attention to some things related to refer, it may help. Thanks, Michel

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Integrate with a media server
by Jimmy Chang 26 Mar '13

26 Mar '13

Hi, We plan to bridge calls to a application (media) server that acts as a IVVR server. After that, the ap server transfers back the call to a queue (callcenter). The question is what's the role of the ap server in the freeSWITCH? We try to register it as a UA, seems not work. Any advice? Regards, Jimmy

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migrating voicemail including custom greetings to updated fs server
by Jason Holden 26 Mar '13

26 Mar '13

Is their a way to accomplish this?

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freeswitch is hanging
by Archana Venugopan 26 Mar '13

26 Mar '13

Hi, At times freeswitch hangs inspite of freeswitch is running in background. The version I am using is FreeSWITCH Version 1.2.3 (1.2.3). At the time of freeswitch hang I see below log messages. I feel that 'MSG Thread' is making it hang not sure though. 2013-03-25 15:16:00.194109 [WARNING] sofia_reg.c:1484 SIP auth challenge (REGISTER) on sofia profile 'internal' for [gcmjanssen1(a)testcloud.com] from ip 86.83.240.9 2013-03-25 15:16:52.074111 [WARNING] switch_event.c:554 Create additional event dispatch thread 2 2013-03-25 15:16:52.374112 [WARNING] switch_event.c:554 Create additional event dispatch thread 3 2013-03-25 15:24:26.164449 [CONSOLE] sofia.c:1602 MSG Thread 2 Started 2013-03-25 15:24:26.164449 [CONSOLE] sofia.c:1602 MSG Thread 1 Started 2013-03-25 15:39:18.254110 [CONSOLE] sofia.c:1602 MSG Thread 3 Started 2013-03-25 15:39:18.254110 [CONSOLE] sofia.c:1602 MSG Thread 4 Started Can anyone please let me know why is it hanging at times and what can be done to resolve it? Many thanks

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How to generate message waiting dialtone with pots
by Trever L. Adams 26 Mar '13

26 Mar '13

Hello everyone, With some recent versions of FreeSWITCH I have been successful in getting some of my configurations to work finally. However, I still have a few remaining questions. The one I am trying to figure out now is this: With all of the POTS providers I have had, if there is a message waiting on their system, you get a waivering dialtone and a flashing light (if the phone supports it). I do not believe my board supports creating the flashing light signal to the phone. Can FreeSWITCH generate the waivering tone? How would I go about configuring it? I imagine there would be a check on voicemail to see if there are NEW messages and then setting some things in the tone file. I haven't a clue what any of those things are and would greatly appreciate help. Thank you, Trever -- "All this technology has somehow made you a stranger in your own land." -- Robert M. Pirsig

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record_session not working
by freeswitch＠tcowan.net 26 Mar '13

26 Mar '13

I am trying to add a record_session to record all my incoming calls. But it just doesnt work. There is nothing in the debug logs at all too. How do I diagnose what is wrong? Does something have to be compiled in for this to work?

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SAY email
by Amit Kumar 25 Mar '13

25 Mar '13

I am trying to make this work session:say("findme(a)itsamit.com", "en", "email_address", "pronounced") but I keep getting the error 2013-03-25 13:19:29.785056 [ERR] mod_say_en.c:551 Unknown Say type=[13] What am I doing wrong?

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FreeSWITCH Weekly News and Notes
by Michael Collins 25 Mar '13

25 Mar '13

Greetings! We are back after a brief hiatus. Our friend and colleague Brian West took some much-deserved time off last week so the rest of us were a bit busier than usual. Brian does an amazing amount of work on FreeSWITCH, CudaTel, and ClueCon, so we're definitely glad he's back to work! On last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_20>we talked about Debian packaging with a little VoIP security and ZRTP thrown in for good measure. Interestingly, a number of people stayed on the call until 6pm EST! You are welcome to stay in the public conference room for as long as you like. This week<http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_27>we have Mark Crane scheduled to give us an update on FusionPBX. It's been about 12 months since we last heard from him. We are looking forward to seeing the improvements that have been added in the past year. We are also gearing up for ClueCon 2013 <http://www.cluecon.com> so save the date: August 6-8, 2013. If you have any questions about being a speaker, sponsor, or attendee then by all means contact us at this email address. Remember that WebRTC is a big topic in the news right now and we are looking for folks to talk about WebRTC in their open source telephony projects and solutions. Have a great week and we'll talk to you again in April! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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