- FreeSWITCH-users - FreeSWITCH Mailing Lists

presence_in
by fs 04 Mar '14

04 Mar '14

Hi. I am trying to user event_sockets to light a lamp on another freeswitch. Running version 1.5.8b+git~20140101T180840Z~e75497c98d~32bit (git e75497c 2014-01-01 18:08:40Z 32bit) I send the following to the remote freeswitch but it does not affect the lamps. The following does reach the freeswitch and the phone does receive data, it is below. sendevent PRESENCE_IN proto: sip from: 25(a)group1.sipline.ca login: 25(a)group1.sipline.ca event_type: presence alt_event_type: dialog Presence-Call-Direction: inbound answer-state: confirmed from the snom 370 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="1" state="full" entity="sip:25@group1.sipline.ca"> </dialog-info> Any ideas what I might be doing wrong Darcy Primrose voice2net

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ft_mod_misdn: Failed to bind mISDN socket [0:1]: Device or resource busy
by Doug Warren 04 Mar '14

04 Mar '14

Hi! I got on my machine a 4 port ISDN board, which I'd love to use with my FS machine. I followed this guide, one of your developers Stefan Knoblich, had written once. I followed this guide: http://lists.freeswitch.org/pipermail/freeswitch-users/2012-June/084633.html but I am not getting really smart. Somewhere in the output I see the error message: http://pastebin.com/YucJ7rPf Failed to bind mISDN socket [0:1]: Device or resource busy Is somebody of you guys familiar with mISDN with freeswitch ?! Or do you advise that I should keep the fingers away, and switch to asterisk to make use of ISDN with a pbx ?! if you guys have any ideas, would be cool! Doug

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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

3 3

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