March 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_mp4
by Lawrence Conroy 04 Mar '14

04 Mar '14

Hi Folks, before I raise a JIRA (but I will if it's not user error), does anyone use mod_mp4 on OS X Mavericks? When I added into modules.conf mod_mp4 and mod_mp4v, I note that my build on Mavericks reported a make error in mp4_helper.http, via mp4_helper.cpp. This is because it doesn't seem to know about type u_int32_t. Adding #include <sys/types.h> @ line 28 of mp4_helper.hpp fixed that error. This did not get reported in make on SL or ML, so it looks like yet another entertainment apple had added with Mavericks. So ... Q: does anyone use this on Mavericks, and if so, has anyone else had a successful build? I'm pretty sure there are a whole host of these little niggles that will get exposed only as people use the different mod_s on this fine operating system; hence ... Q: should this be raised as a generic JIRA for "fS doesn't make on Mavericks", or be specific to this particular lacuna? [there's another one already for portaudio, and no doubt there will be others :] all the best, Lawrence

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NO_USER_RESPONSE
by Web Developer 04 Mar '14

04 Mar '14

Hello, Why am I getting this error. switch_ivr_originate.c:2430 Cannot create outgoing channel of type [dingaling] cause: [NO_USER_RESPONSE] Thanks

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Call Without Authorization
by Shahzad Bhatti 04 Mar '14

04 Mar '14

> > Thanks for such support and reply. > Regards Shahzad Bhatti > > ---------- Forwarded message ---------- > From: Donny Hardyanto <hardyanto.donny(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 4 Mar 2014 10:21:51 +0700 > Subject: Re: [Freeswitch-users] Call Without Authorization > Hi Shahzad, > > 1. Dont expose you SIP port on internet. If only for testing or local use, > dont open it on internet. If you have to, use VPN. > > 2. Make sure all SIP/H.323 ALG on all Routers (they have one!) in your > network is turn off. They can mask the intruder from outside and look like > from local IP. > > 3. Change your default password! The one on vars.xml and every password on > directory! > > 4. If you in control of your SIP client/ip phone, it good to change the > default port 5060 and 5080 to some thing random (like 61351 etc). It make > it harder for hacker to find your ports. Usually the SIP hacker is time and > money oriented (not achivement-oriented), so most of the time it does not > bother to find SIP port other than the default ports. They will quickly > find another server IP to probe. They have organization behind them that > can stream international phone call to the hacked servers. They usually > test the softswitch first by sending alot of registration (some sip id is > John etc). I think some show they can break the password using this because > SIP authorization only using hash to check. So because hash has collision, > they can calculate your password. > > 5. They usually try to break on weekdays (check your cdr) and use your > hacked line on weekends! Please be carefull if you connected to PSTN line > or operator! You can lose thousand of dollars in 1 day! > > 6. You can use SBC but you need very through on configuring. Unconfigured > SBC same as no protection at all. > > Lastly, the SIP world is VERY CRUEL. Honest mistake can destroy your life. > Be EXTRA careful. > > Donny > > > > On Sun, Mar 2, 2014 at 9:25 AM, Shahzad Bhatti <shahzad.bhatti(a)g-r-v.com>wrote: > >> 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 >> >>

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General presence question
by Darcy Primrose 04 Mar '14

04 Mar '14

A general question, I send a presence_in or presence_out message to light a lamp for a non registered entity, such as a nite mode key or a registration on another freeswitch, should the freeswitch keep this status so the next time a set sends a subscription, it returns the status I have sent. This is not happening and I just want to know if I am chasing something that is not supposed to happen. Thanks Darcy Primrose voice2net.ca

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How to suppress the SIP Registration trace on sofia global siptrace on
by Donny Hardyanto 04 Mar '14

04 Mar '14

Hi, Is there any way to suppress SIP Registration trace on sofia global siptrace? I only need to trace the INVITE but because my system has a lot of clients doing SIP Registers, I having a hard time finding the SIP INVITE dump because the screen keep on scrolling. Thanks, Regards, Donny

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How to propagate destination number from A-leg to the (B-)leg created by mod_fifo ?
by Thorleif Stene 04 Mar '14

04 Mar '14

I’m looking for a way to propagate the A-leg called number to the B-leg setup by mod_fifo to the member. (on-hook agents via loopback ) Legs involved: A-leg -> loopback-A > loopback-B —> B-leg I’d like to control the caller_id for the B-leg. Sometimes showing the A-leg caller number (A-number) and sometimes the A-leg called number (the B number) Any wizards out there with a quick answer ? Thorleif

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conference_auto_outcall_maxwait not disconnecting endpoints
by Mike Kane 04 Mar '14

04 Mar '14

Good morning, I'm having an issue with the "conference_ auto_outcall_maxwait" variable. The variable executes but the result is an audible beep when the timer is met and the ringing endpoint does not disconnect. Any help would be greatly appreciated. Pastebin: http://pastebin.freeswitch.org/22070

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Interpreting "top" output
by Stanislav Sinyagin 04 Mar '14

04 Mar '14

hi, "top" shows the freeswitch process with up to 25-30% CPU usage (5 channels in a conference, 1x Speex, 1x GSM, 2xPCMA, 1xPCMU), while total userland usage is shown at 0.1%. Also mpstat shows almost 100% idle. Is it a bug in the kernel? or any other ideas? Intel Atom N570, 2 physical cores with hyperthreading, /proc/cpuinfo shows 4 CPU's. Debian 7.4 i386 3.2.0-4-686-pae #1 SMP Debian 3.2.54-2 i686 GNU/Linux FreeSWITCH 1.2.22 from debs Details here: https://gist.github.com/ssinyagin/9302079 regards, stanislav

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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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