March 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Call Hanging up on receiving 302 from SIP endpoint
by Akhil Chandra 04 Mar '14

04 Mar '14

Hi Brian, I am a little confused. Shouldn't the profile for the extension forwarding (sending a 302 message) be loaded in the flow. For eg: In case the extension is not authorized to make international calls, and I set a default profile(which would be a blanket profile for all extensions) as sip_redirect_profile, the call could go through. I still haven't tested your suggestion, but will do it at earliest. Regards, Akhil

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Call Without Authorization
by Shahzad Bhatti 04 Mar '14

04 Mar '14

Hi Everybody, i am rephrasing my question that i got a legal registered sip account 1001 on freeswitch but some hacker who is not registered on my freeswitch but use same 1001 account and make call. i put condition in xml_dialplan to verify and allow only register sip accounts to call as *<condition field=* *"${sofia_contact */1001@freeswitchIP}" expression="^[^@]+@(.+)">>*but hacker find someway to pass the regex through some back whole in my script and make calls *dialplan xml is * http://pastebin.freeswitch.org/22054 *fs_cli log as * http://pastebin.freeswitch.org/22050 *xml_cdr is* http://pastebin.freeswitch.org/22052 i also try to generate the scenario but got no success, but now want to know how hacker made successful call in the above scenario and what is the best way to prevent from hacking in future Regards Shahzad Bhatti ---------- Forwarded message ---------- From: Shahzad Bhatti <shahzad.bhatti(a)g-r-v.com> Date: Fri, Feb 28, 2014 at 11:51 PM Subject: Call Without Authorization To: freeswitch-users(a)lists.freeswitch.org Hi everybody, i create my xml_curl script as that don't allow unregistered calls with the following condition *<condition field=\"\${sofia_contact */{$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

3 2

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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test -- ignore
by Terry Barnum 04 Mar '14

04 Mar '14

2 1

Reasons for uuid_bridge not appear not to work?
by Alex Lake 04 Mar '14

04 Mar '14

Got a slightly troublesome Lua script/dialplan combo that is not bridging calls properly. I am thinking that there must be some subtlety concerning when bridging of ACTIVE channels is not possible. The scenario is in an attended transfer where A calls B, B calls C, and then B ducks out. bridging A&C. As far as I can tell, A&C are NOT terminated, but something funny happens that prevents the bridge establishing a call path. Any ideas?!

2 1

mod_g723_1
by Veerabhadra Rao 04 Mar '14

04 Mar '14

Hi, I am currently working on freeswitch-1.2.12 version. how can i convert wav to G723 format using fs_encode binary. After installing the module mod_g729_1, I run the command as follows "./fs_encode -l mod_g723_1 temp.PCMU temp.G723" then it is raising the error as " Couldn't initialize codec for G723@8000h@20i ". Thanks in advance.

3 2

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 Hanging up on receiving 302 from SIP endpoint
by Akhil Chandra 03 Mar '14

03 Mar '14

Hi Brian, I am a little confused. Shouldn't the profile for the extension forwarding (sending a 302 message) be loaded in the flow. For eg: In case the extension is not authorized to make international calls, and I set a default profile(which would be a blanket profile for all extensions) as sip_redirect_profile, the call could go through. I still haven't tested your suggestion, but will do it at earliest. Regards, Akhil

1 0

[fs_ivrd] <defunct>
by Ali Pey 03 Mar '14

03 Mar '14

Hello, I use fs_ivrd to call a perl script for every new call. In the script I have a bridge. If a-leg hangs up first, sometimes I endup with [fs_ivrd] <defunct> processes that stays there forever. The process doesn't get cleaned up. Any suggestions or guidance would be greatly appreciated. Thanks, Ali Pey

1 0

Call Without Authorization
by Shahzad Bhatti 03 Mar '14

03 Mar '14

Catch the scenario here the hacker use external profile using port 5080 where sofia_contact is not checking and allow to pass the regex if user is also register on internal profile hence TRUE the regex also and call proceeds but in external profile configuration file when i make auth-calls=true call from hacker ip is not allowed but now i want to know is there any better or professional way to avoid hacker calls. if any one have any suggestion do reply me. thanks in advance Regards Shahzad Bhatti ---------- Forwarded message ---------- From: Shahzad Bhatti <shahzad.bhatti(a)g-r-v.com> Date: Fri, Feb 28, 2014 at 11:51 PM Subject: Call Without Authorization To: freeswitch-users(a)lists.freeswitch.org 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

1 0