March 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeTDM Presence
by Nandy Dagondon 19 Mar '14

19 Mar '14

Is it possible to use SIP PRESENCE on IP phone BLF indicators to monitor FreeTDM FXO channels? Any hint how to approach it? Thanks. /Nandy

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Unable to access variables set in directory from the dialplan
by Iskren Hadzhinedev 19 Mar '14

19 Mar '14

Greetings! I'm unable to access user variables that are set in the directory from the dialplan. The setup is quite simple, the user is: -- <user id="611"> <params> <param name="a1-hash" value="693c2978ddbe62f7129b7e551bcf7d49"/> <param name="vm-enabled" value="true"/> <param name="vm-a1-hash" value="126003cf36e8e6a69ecdbfd1bba121c8"/> </params> <variables> <variable name="toll_allow" value="local,international"/> <variable name="user_context" value="sip.test.com"/> <variable name="effective_caller_id_name" value="Tester"/> <variable name="effective_caller_id_number" value="611"/> </variables> </user> -- And the extension in the dialplan is: -- <extension name="Outgoing Test"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^([0-9]{6})$"> <action application="bridge" data="sofia/gateway/testgw/$1"/> </condition> </extension> -- The toll_allow always gets evaluated to an empty string, and when I actually run the info application I don't see any of the variables that are set. If I replace the ${toll_allow} with ${user_data(${sip_from_uri} var toll_allow}, everything is working, although the examples in the wiki are showing that you can use those variables just fine without invoking user_data. What am I doing wrong? Thanks! ---- Kind regards, Iskren

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Frequent connect/disconnect status on PRI line
by Ravi 19 Mar '14

19 Mar '14

Hello Everyone ! I have been running Freeswitch for about 7 months. So far, it had been running so well. The last few days, I noticed that the status changes from up to down frequently, disrupting the incoming/outgoing calls. I am not able to figure out why this is happening. I went through the user list and found that some one had a similar problem on Feb 20, 2014, but no resolution in the emails. Please find below details on my freeswitch system: WANPIPE Release: 7.0.5 FreeSWITCH Version 1.5.5b+git~20130722T104835Z~cb4e31b6cf (git cb4e31b 2013-07-22 10:48:35Z) libsng_isdn.so.7.27.2 CentOS release 6.4 (Final) Linux bfree-server 2.6.32-358.14.1.el6.x86_64 #1 SMP Tue Jul 16 23:51:20 UTC 2013 x86_64 x86_64 x86_64 GNU/Linux Also attached a log file that has the status for reference. Any help is appreciated. Thanks. Ravi

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luasql is not compiled in current version
by Shahzad Bhatti 19 Mar '14

19 Mar '14

i am trying to compile luasql on new server with current version of freeswitch but found that here i got lua 5.2 and it didn't allow me to get luasql working and then i use git for v1.2 stable and here luasql is compile on the same server but luasql is not working here i want to know how i can compile luasql on latest version with lua 5.2 here is my current version of freeswitch FreeSWITCH Version 1.5.11b+git~20140312T014059Z~60bb1602d2~64bit (git 60bb160 2014-03-12 01:40:59Z 64bit) Regards Shahzad Bhatti

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Solaris / SmartOS (FS-6375)
by Brian West 18 Mar '14

18 Mar '14

FreeSWITCHers, Solaris 11.1 is working with minimal hacks now based on the work MikeJ has done with automake with all the modules in tree. I’ve put this up so others can try to replicate the process from a SmartOS if possible and report back to the JIRA. https://jira.freeswitch.org/browse/FS-6375 Testers wanted ;) Apply within! -- 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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Eavesdrop application doesn't work for the WebRTC clients
by Yan Brenman 18 Mar '14

18 Mar '14

Hello, I was trying to get the eavesdrop application work on the 1.4.beta branch for the WebRTC clients (works great for all other types of clients) and could not. Neither audio nor video (I merged the code from the video-media-bug branch for the video) is going through. For the client I tried both - sipml5 and jssip clients. Please let me know if there is any other piece of information I can provide. I will be more then happy to do any kind of code changes and testing - just need some starting points. Especially considering that I am very new to the WebRTC. Every piece of advice is greatly appreciated. Thanks

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Outbound call limit bypass prevention
by Dmitry Sytchev 18 Mar '14

18 Mar '14

Hi all! <intro> If you use FS or other softswitch with to provide services with limited amounts of concurrent calls and have enabled REFER method for your SIP users (you may disable it by *disable-transfer* in profiles, default false), your setup may be vulnerable to call limit bypass attack. For example, someone stoles credentials or bruteforces account password of your SIP user and sends an amount of calls to premium numbers or longdistance directions. These attempts should be at least limited with customer call limit, but there is a method to bypass it. When attacker sees the call limit is set on account, and call limit > 1, he tries to bypass it to increase speed and amount of outbound calls. He makes one outbound call, then makes second. At this moment there is four channels - two inbound channels to FS, and two outbound channels from FS to PSTN or SIP gw. FS sets call limit usage to 2. Now attacker issuing transfer request by sending REFER to bridge two outbound channels with each other. After successful transfer, two inbound channels are destroyed and outbound channels are bridged together. After that, FS decreases call limit, since it destroyed channels limit was set on. </intro> The problem is the method people often use to limit concurrent calls. Something like that: <action application="limit" data="hash OutboundLimit ${pbx_subscriber_customer_id} ${pbx_customer_call_limit} !CALL_REJECTED"/> <action application="bridge" data="{ignore_early_media=false,hangup_after_bridge=false}sofia/gateway/gw/$1"/> Now limit is set on inbound channel. When attacker bridges outbound channels, their A-legs (inbound channels) get destroyed and FS decreases limit due to channel destruction. So attacker have 2 outbound bridged channels, and limit is still 0. Then the process repeats, allowing attacker effectively bypass your call limits. One of possible solutions is to move limits on B-legs after answer, while keeping limits on A-legs too in order prevent exceeding limit by unanswered calls: We set limit on A-LEG, then on answer we decrease limit on A-LEG and "move" it to B-leg by calling limit on outbound channel. <action application="export" data="nolocal:execute_on_answer_1=set api_result=${uuid_limit_release(${uuid} hash OutboundLimit ${pbx_subscriber_customer_id})}"/> <action application="export" data="nolocal:execute_on_answer_2=limit hash OutboundLimit ${pbx_subscriber_customer_id} ${pbx_customer_call_limit} !CALL_REJECTED"/> <action application="limit" data="hash OutboundLimit ${pbx_subscriber_customer_id} ${pbx_customer_call_limit} !CALL_REJECTED"/> <action application="bridge" data="{ignore_early_media=false,hangup_after_bridge=false}sofia/gateway/gw/$1"/> Now limit is correctly evaluated for non-answered outbound calls, and doesn't allow excessive calls. Hope this helps somebody. Maybe you guys know more effective and correct ways to prevent limit bypassing in this scenario? -- Best regards, Dmitry Sytchev, IT Engineer

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Music detection
by Muhammad Shahzad 18 Mar '14

18 Mar '14

Hi, Is there a way we can detect music on an answered call? Basically i want to record the call but there may be music in beginning, during or at the end of call, which i do not want to be recorded. Is it possible? If not, are there any tools that can do that on an already record file? Thank you.

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Codec negotiation with ESL socket?
by Ian McMaster 18 Mar '14

18 Mar '14

Our current Polycom phones send the following in the SDP: a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 We added a 550 phone and now the preferred coded is G722 *a=rtpmap:9 G722/8000* a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 Our IVR application (socket) plays files recorded in PCMU format. When the new phone connects (G722), the play function is now looking for files ending in .G722 (which do not exist). How do you negotiate a codec without a B leg? I'd like the phone to use G722 when connected to other G722 phones, but use PCMU if connected to a socket application. Thank you, Ian.

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Ring timeout
by Steven Schoch 18 Mar '14

18 Mar '14

Today I had to call one of those old-school answering machines on a POTS line that had been turned off remotely, accidentally. To turn it back on, I had to call it and let the line ring 15 times (about one and a half minutes). But when I called through our FreeSWITCH PBX, the call was disconnected. Is this timeout in FreeSWITCH, or is my provider? -- Steve

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