February 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

I want to call unregistr user.
by 宮崎武 06 Mar '14

06 Mar '14

Hi, I want to call unregistr user. after regist ,start calling. A:unregisterd B:registerd 1.B call to A 2.B is calling 3.A regist 4.A ring 5.A and B talk start Please, help. Thanks.

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Audio Handshake Failure 1 in switch_rtp.c
by Kshitij Saxena 04 Mar '14

04 Mar '14

Summary: SIPml / JSSIP calls over WS to FreeSWITCH are terminated with "Audio Handshake Failure 1" in switch_rtp.c I am running FreeSWITCH Version 1.5.8b on an Ubuntu 12.04 server with a Sangoma A101DE PRI card running on Wanpipe 7.0.10. The PRI card is connected to the ISP's network. When I try connecting to FreeSWITCH from the SIPml or JSSIP demo pages, (over websocket, RTCWebBreaker enabled, port 5066 open on server for tcp/udp along with 5060-5081 and 16383-32768), the client is successfully registered, but any attempt to make a call is failing. In fs_cli I can see an error "Audio Handshake Failure 1" in switch_rtp.c as soon as a call is attempted from the client. When using X-lite, the calls are successful. Originate from within fs_cli is also successful. I have been splitting my hair for 2 days but cannot begin to decipher what has gone wrong! What is worse is that I have used the exact same configuration successfully at 2 previous locations! Any help would be very greatly appreciated. Full Version: FreeSWITCH Version 1.5.8b+git~20140208T085053Z~4fa68fcd75~64bit (git 4fa68fc 2014-02-08 08:50:53Z 64bit) Originate: originate freetdm/1/a/99xxxxxxxx &bridge(freetdm/1/a/99xxxxxxxx) Regards, Kshitij Saxena

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On-conference distortion for FreeTDM channels, but not SIP channels
by Melanie Treitinger 03 Mar '14

03 Mar '14

Nobody has an idea on this? Here again Bob's Message from 2013-09-20: I've tried everything I can think of on this, so now I'm reaching out to you good people. I have a FreeSWITCH system that has the latest version of FreeTDM, Dahdi, libpri, and a Rhino R4T1 quad T1 board with up-to-date firmware. When I call into one of the PRI channels, a "playback"-played file sounds clean, but when I go into conference, the conference audio is distorted. This does not happen with a SIP channel. I would classify the distortion as a "crackle", but it is very regularly periodic. Repeating ticks at (roughly) 10-20 Hz repeat rate, I would say. I can only hear them when audio is present, i.e. the audio seems to be modulated with the ticks. And that conference audio can be the MOH guitar recording. I've eliminated the following: - T1 clocking - Echo Cancellers (I've turned on and off all combinations of hardware and software EC) - T1 board firmware (it's the latest, and the problem only happens on-conference). - Packet length - it's the same (20 mS) for the conference engine, FreeTDM, and Dahdi - mu-law vs. a-law encoding - behaves identically. - Comfort noise - happens with and without. - conference sampling rate (tried 8000 and 16000, even though I'm using TDM, which is 8000). - FreeSWITCH version - happens on 1.2 stable, the latest master branch, and an older version that we were using. If you haven't heard of anything like this, could you please give me your thoughts on why this could happen only on-conference, not while playing WAV files, and only with FreeTDM channels? Thanks! Bob

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Call Without Authorization
by Shahzad Bhatti 02 Mar '14

02 Mar '14

Hi everybody, i create my xml_curl script as that don't allow unregistered calls with the following condition *<condition field=\"\${sofia_contact */{$sipuser}@$domain}\" expression=\"^[^@]+@(.+)\">* and its working but yesterday a call is originated from having *fs_cli log as * http://pastebin.freeswitch.org/22050 *xml_cdr is* http://pastebin.freeswitch.org/22052 *dialplan xml is * http://pastebin.freeswitch.org/22054 this is only example that how the hacker breached i want to know that *1. how it is possible that this call is originated as i check condition that allow to call only registered sip accounts.* *2. how to prevent that this would not happened in future. * *3. if there any better way to do that do inform me;* i check about 500 calls placed under the given scenario and many of them also answered Regards Shahzad Bhatti

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Dialplan Issues with Nortel CS1000
by Joel White 01 Mar '14

01 Mar '14

> I have modded the default dialplan to allow 5 digit local extensions. I have the Nortel NRS sending the calls to the public context. I have created the xml to route the incoming call to a local extension. > > Question is, why do the calls go straight to voicemail without ringing the extension. The extension is registered and can make and recieve calls locally on the FS. > > The incoming call from the Nortel CS 1000 reaches the XML parsing for "public" and executes the XML in the public folder relating to the Nortel. The call also makes its way to the "default" context and executes the LOCAL 5 Digit Extension XML config. What I am not understanding is why, if it makes it through the variables and expressions and ends up at the 5 digit local dialing, that it would go straight to voicemail without ringing the phone. > > > > If anyone has any insight into why this behavior would occur.... please point me in the right direction. > > > Thank you in advance > > Joel

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Re: [Freeswitch-users] Performance
by Keith 01 Mar '14

01 Mar '14

Hi Yossi, Yeah I appreciate that and I don't expect any performance tuning tips above what is in thw wiki. I suppose what I really need to understand is why in 1.2.12 I can load 1000 concurrent calls but in 1.2.22 I can only load 100 without audio delay etc. I appreciate all comments and help with this. Thanks Keith

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Working on new vanilla config
by Brian West 28 Feb '14

28 Feb '14

I’ve filed FS-6277 to start the planning process of a new vanilla config set. Input is welcome by all if you have an idea please post them on the JIRA and we’ll touch on this tomorrow during our friday call so we can get the ball rolling on this task. http://jira.freeswitch.org/browse/FS-6277 Thanks, -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest 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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You guys are terrible
by Jose Suero 28 Feb '14

28 Feb '14

I'm swimming in a freeswitch log looking for a problem when all of the sudden this comes up: 2013-08-21 19:51:08.002385 [DEBUG] sofia.c:6928 1 .. 2 .. Freddie's commin' for you... Didn't find the freeswitch problem, but now have a machete masked fellow on his way :-) Made me smile

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SSD Suggestions
by Nandy Dagondon 28 Feb '14

28 Feb '14

Hi! I'm interested to implement SSD. I have already read in the Wiki Tuning SSD on Linux. Very helpful. Can you suggest which brand/model to try? Thanks. /Nandy

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Issues with stripping Nortel extra SIP data
by Joel E White 28 Feb '14

28 Feb '14

Hi I am wondering how I can strip the extra information being sent from the nortel and passing only digits along. The string from nortel is 23673;phone-context=location.systemname I have substituted the data with fictional information, but this is the gist of it What I need freeswitch to do is remove everything after the 5 numbers I have been unsuccessful in doing so using regex (probably because of my lack of knowledge on the subject) I have been able to accomplish this using asterisk, then passing the data along to freeswitch exten => _X.,Dial(SIP/${EXTEN:0:-33}@freeswitch Can someone show me how to accomplish the same using regex and variables like $1? Thank you, Joel Sent from my iPad

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