date:20091203

First off, maybe this conversation is better suited to the dev list, and second 
off, the current setup of where we do timers, where we poll, polling frequency 
and architecture is the result of 4+ years of ongoing testing and optimization. 
 We have tried all different methods throughout.  Sometimes what we found to be 
most efficient is not what we thought at first would be, but testing showed 
otherwise.  We have always optimized the general case as to if there are many 
calls, and no suggestion would be implemented that hurts this case.  That being 
said, if you could really come up with a way for this to be more efficient in 
any case, without sacrificing performance int he other cases, you are able to 
prove this with extensive test results, and you are able to prove that it does 
not impact for example call quality in any of the hundreds of edge cases that 
have led us to the point we are now, then we may be interested in taking such a 
patch.  

Mike



On Dec 2, 2009, at 11:58 PM, eaf wrote:

 
 As I see it, switch_cond_next() currently is just a do_sleep(1000). Yes, it
 could be mapped to a 1ms timer, but #define DISABLE_1MS_COND overrides
 that.
 
 Yeah, there is a global timestamp... It's easy to workaround that for RTP
 who calls switch_micro_time_now()... But if somebody accesses
 runtime.timestamp directly, it's gonna be tough to grep for that. If only
 this was C++...
 
 I'll play around. Never liked polling too much. Never could've guessed that
 polling could be so useful for scalability ;) My naive implementation
 would've pulled timestamp via system calls and would've done sleeping by
 passing exact interval to select() instead of syncing with a pacing thread.
 Which would be dead-quiet at idle time, but, of course, would stop scaling
 at some point due to excessive number of system calls.
 
 Thanks.
 
 
 Michael Jerris wrote:
 
 In short.  No, you can not for many reasons. The milisecond tic is  
 used throughout the code even when there is not any calls up.  You can  
 grep for switch_cond_next if you would like to see where but it is  
 required to keep our global timestamp and for pacing the scheduler  
 among other services that run all the time.
 
 Mike
 
 On Dec 2, 2009, at 7:31 PM, eaf erandr-j...@usa.net wrote:
 
 
 Can I reduce resolution of that timer thread 10 times? I mean, I  
 glanced
 through the code, and see that among others (are there others?) RTP  
 and IVR
 set up their timers that are subsequently managed by this thread.  
 RTP timers
 should be eliminated by that setting you've suggested. IVR timers  
 are set at
 20ms... So, if the thread is set to wake up every 10ms instead of  
 1ms it
 should be able to wake up those IVR timers just fine. Right?
 
 That's a cool design to have one dedicated thread that maintains  
 accurate
 timing and then broadcasts via condition variables to hundreds of  
 other
 threads events that they can register for. I'm sure it's one of the  
 reasons
 why FS scales so much better than Asterisk. But for poor low-end  
 setups that
 sit in the closet, eat only 6W of power and hardly ever run more  
 than two
 calls at the same time, can I hack it somehow to be more UNIX- 
 friendly? I.e.
 make it stuck in select() or recv() when there is nothing to do, call
 clock_gettime() right from the thread that wants and when it wants  
 to know
 current time?
 
 Say, what if that thread is made to suspend on a condition variable  
 in case
 if there are no timers registered in TIMER_MATRIX? Then, if some other
 thread comes up and adds its timer into the matrix, it could wake up  
 the
 timer thread and enjoy accurate timing as needed, on demand? And in- 
 between
 the calls, when there is no RTP or IVR, it will all go silent? I mean,
 sitting on a wait queue in the kernel is way better than go back and  
 forth
 incrementing counters that nobody even needs at the moment?
 
 
 Anthony Minessale-2 wrote:
 
 idle is a 4 letter word to a realtime application.
 
 The core keeps a single high-priority thread to keep 1ms timing and
 expands
 that broadcasting
 to hundreds or thousand of threads who need accurate timing.
 
 Your choppy audio is caused by linksys lying about the packet len  
 that
 it's
 using and we set our timer
 to the wrong speed.
 
 
 
 -- 
 View this message in context:
 http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html
 Sent from the Freeswitch-users mailing list archive at Nabble.com.
 
 
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Re: [Freeswitch-users] How to run a JS script periodically

You could also use the scheduler to run the jsrun command inside FreeSWITCH.

Mike


On Dec 3, 2009, at 8:31 AM, Rob Forman wrote:

 What about cron?
 
 Create a cron entry like:
 */5 * * * * /usr/local/freeswitch/bin/fs_cli -x jsrun yourscript app()
 
 But if you're just dumping global variables, you could easily retrieve them 
 directly from fs_cli without running an app and process the output however 
 you'd like:
 
 /usr/local/freeswitch/bin/fs_cli -x global_getvar
 
 
 On Thu, Dec 3, 2009 at 6:21 AM, Oscav os...@hotmail.fr wrote:
 
 Hi,
 
 Someone knows how to run periodically a JS script ?? The purpose is to write
 to a db some global informations (Global Variables) about FS like every 5
 minutes.

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Re: [Freeswitch-users] Best way to run originate calls through dial plan

http://wiki.freeswitch.org/wiki/Mod_commands#originate

   Usage: originate call_url exten|application_name(app_args) 
[dialplan] [context] [cid_name] [cid_num] [timeout_sec]

You can do this via shelling out to fs_cli like your example below or using esl 
directly from php:

http://wiki.freeswitch.org/wiki/Esl

Mike

On Dec 2, 2009, at 1:23 PM, eaf wrote:

 
 I need a way to start a call from the PHP script to the originating number,
 tell the party on that number to hold on, start another call to destination
 number, and bridge everything together. On both legs I need to pass custom
 caller ID. I can of course open direct connections to VOIP gateways right
 from PHP, but I want to reuse existing routing rules in the dial plan, hence
 I want to know what's the best way of making originate go through a specific
 context of the dial plan.
 
 As for the number of calls per second, it's going to be only occasionally
 used.
 
 
 mercutioviz wrote:
 
 On Wed, Dec 2, 2009 at 6:47 AM, eaf erandr-j...@usa.net wrote:
 
 
 What would be the best way of making originate() run call through a dial
 plan
 (compared to directly going to a specified VOIP gateway). Would it be
 loopbacks, i.e. smth like this?
 
 /opt/freeswitch/bin/fs_cli -x originate
 
 {ignore_early_media=true,origination_caller_id_number=xx}loopback/yy/default/XML
 'javascript(/opt/freeswitch/conf/dialplan/public/webcall.js
 zz)'
 
 The idea of this is that originate() sets up the first call, then
 webcall.js
 plays back a WAV, and bridges the first call with the second one (also
 set
 up via loopback).
 
 
 Could you describe the problem that you're trying to solve? That would
 make
 it easier to know if what you've come up with is the best solution. How
 many
 calls per second were you wanting to generate with this setup?
 -MC
 
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Re: [Freeswitch-users] Eavesdrop error?

The behavior is probably expected, the unhelpful error is probably undesirable 
but it would make a mess of the dial-plan to clean that up.

Mike

On Dec 2, 2009, at 9:19 PM, Lars Zeb wrote:

 Is this reasonable given it was the only call in FreeSwitch at the time? How
 can this situation be corrected in the future?
 
 From: freeswitch-users-boun...@lists.freeswitch.org
 [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony
 Minessale
 Sent: Wednesday, December 02, 2009 3:35 PM
 To: freeswitch-users@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] Eavesdrop error?
 
 it probably just means the uuid was not retrieved from the db when you
 called the eavesdrop exten which does the lookup on the uuid for the hash
 key based on what ext you hit to retrieve the most recent uuid that called
 that ext.
 
 
 On Wed, Dec 2, 2009 at 5:22 PM, Lars Zeb larc...@yahoo.com wrote:
 Sorry, svn 15753
 
 -Original Message-
 From: freeswitch-users-boun...@lists.freeswitch.org
 [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Lars Zeb
 Sent: Wednesday, December 02, 2009 2:08 PM
 To: freeswitch-users@lists.freeswitch.org
 Subject: [Freeswitch-users] Eavesdrop error?
 
 I tried to use eavesdrop today and it did not work. The error message in the
 log is:
 
 [ERR] mod_dptools.c:334 Usage: [all | uuid]
 
 I simply dialed 881010, trying to eavesdrop on extension 1010. Is this
 incorrect?
 
 http://pastebin.freeswitch.org/11363
 
 Thanks Lars
 
 
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 -- 
 Anthony Minessale II
 
 FreeSWITCH http://www.freeswitch.org/
 ClueCon http://www.cluecon.com/
 Twitter: http://twitter.com/FreeSWITCH_wire
 
 AIM: anthm
 MSN:anthony_miness...@hotmail.com
 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
 IRC: irc.freenode.net #freeswitch
 
 FreeSWITCH Developer Conference
 sip:8...@conference.freeswitch.org
 iax:gu...@conference.freeswitch.org/888
 googletalk:conf+...@conference.freeswitch.org
 pstn:213-799-1400
 
 
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Re: [Freeswitch-users] Choppy sound with PCMU


Oh, it's not just one timer thread... Why, why is sql_thread keeps on
checking for messages every millisecond? Couldn't there be some signalling
implemented that will make the thread suspend on condition variable or a
socket/pipe in between? 

#0  do_sleep (t=1000) at src/switch_time.c:109
#1  0xb7e5e215 in switch_core_sql_thread (thread=0xb7586ae8, obj=0x0) at
src/switch_core_sqldb.c:783

Why does this sofia_profile_worker_thread keeps on looping checking for the
queue? Have a semaphore!

#0  do_sleep (t=1000) at src/switch_time.c:109
#1  0xb73a4701 in sofia_profile_worker_thread_run (thread=0x80f3a30,
obj=0x80f2490) at sofia.c:978

Nothing's happening on the box, but there are three threads that pretend to
be actively busy with smth. Others at least sleep for hundreds of
milliseconds, not for one.

And there is even infrastructure present to do blocking pops: i.e. why
couldn't sqldb thread do queue_pop() instead of queue_trypop() intermixed
with 1ms sleeps? This looping is such a waste...


eaf wrote:
 
 As I see it, switch_cond_next() currently is just a do_sleep(1000). Yes,
 it could be mapped to a 1ms timer, but #define DISABLE_1MS_COND
 overrides that.
 
 Yeah, there is a global timestamp... It's easy to workaround that for RTP
 who calls switch_micro_time_now()... But if somebody accesses
 runtime.timestamp directly, it's gonna be tough to grep for that. If only
 this was C++...
 
 I'll play around. Never liked polling too much. Never could've guessed
 that polling could be so useful for scalability ;) My naive implementation
 would've pulled timestamp via system calls and would've done sleeping by
 passing exact interval to select() instead of syncing with a pacing
 thread. Which would be dead-quiet at idle time, but, of course, would stop
 scaling at some point due to excessive number of system calls.
 
 Thanks.
 
 
 Michael Jerris wrote:
 
 In short.  No, you can not for many reasons. The milisecond tic is  
 used throughout the code even when there is not any calls up.  You can  
 grep for switch_cond_next if you would like to see where but it is  
 required to keep our global timestamp and for pacing the scheduler  
 among other services that run all the time.
 
 Mike
 
 On Dec 2, 2009, at 7:31 PM, eaf erandr-j...@usa.net wrote:
 

 Can I reduce resolution of that timer thread 10 times? I mean, I  
 glanced
 through the code, and see that among others (are there others?) RTP  
 and IVR
 set up their timers that are subsequently managed by this thread.  
 RTP timers
 should be eliminated by that setting you've suggested. IVR timers  
 are set at
 20ms... So, if the thread is set to wake up every 10ms instead of  
 1ms it
 should be able to wake up those IVR timers just fine. Right?

 That's a cool design to have one dedicated thread that maintains  
 accurate
 timing and then broadcasts via condition variables to hundreds of  
 other
 threads events that they can register for. I'm sure it's one of the  
 reasons
 why FS scales so much better than Asterisk. But for poor low-end  
 setups that
 sit in the closet, eat only 6W of power and hardly ever run more  
 than two
 calls at the same time, can I hack it somehow to be more UNIX- 
 friendly? I.e.
 make it stuck in select() or recv() when there is nothing to do, call
 clock_gettime() right from the thread that wants and when it wants  
 to know
 current time?

 Say, what if that thread is made to suspend on a condition variable  
 in case
 if there are no timers registered in TIMER_MATRIX? Then, if some other
 thread comes up and adds its timer into the matrix, it could wake up  
 the
 timer thread and enjoy accurate timing as needed, on demand? And in- 
 between
 the calls, when there is no RTP or IVR, it will all go silent? I mean,
 sitting on a wait queue in the kernel is way better than go back and  
 forth
 incrementing counters that nobody even needs at the moment?


 Anthony Minessale-2 wrote:

 idle is a 4 letter word to a realtime application.

 The core keeps a single high-priority thread to keep 1ms timing and
 expands
 that broadcasting
 to hundreds or thousand of threads who need accurate timing.

 Your choppy audio is caused by linksys lying about the packet len  
 that
 it's
 using and we set our timer
 to the wrong speed.



 -- 
 View this message in context:
 http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html
 Sent from the Freeswitch-users mailing list archive at Nabble.com.


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Re: [Freeswitch-users] Bridging/Connecting Freeswitch servers

2009-12-03 Thread Otis

Michael Collins wrote:


 On Wed, Dec 2, 2009 at 9:58 AM, Frank Carmickle fr...@carmickle.com 
 mailto:fr...@carmickle.com wrote:

 On Wed, Dec 02, Otis wrote:
 Snip...

  Thanks.
 
  I would like all extensions on say server A  to be contactable
  by those
  on server B and vice versa.

 The example I gave you should get you started.  Let us know how
 you get along.  Have a read through the wiki pages like

 http://wiki.freeswitch.org/wiki/Dialplan_XML
 http://wiki.freeswitch.org/wiki/Mod_dptools#Applications
 http://wiki.freeswitch.org/wiki/Sofia

 --FC


 Remember, too, that gateways are useful for doing auth/reg so having a 
 gateway on each box that registers to the other box is pretty handy. 
 If you run into any trouble trying to set it up you can ask here or 
 join us in #freeswitch on irc.freenode.net http://irc.freenode.net.
 -MC


Hi FC

I used your code :

extension name=fjc-pbx-inbound
  condition field=network_addr 
expression=^2001\:470\:1f..\:6..\:.e0\:.1f.\:fe34\:b29d$/
condition field=destination_number expression=^(.*)$
  action application=transfer data=$1 xml default/
/condition
  /extension
replacing with my box's ip address. I have received any errors in the 
fs_cli console neither is there any reference to my box'x ipddress. Any 
way to check all is well ?

And how do I join join us in #freeswitch on irc.freenode.net 
http://irc.freenode.net. ? Went to the freenode.net site and got lost. 
Will persevere.


Thanks


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Re: [Freeswitch-users] Bridging/Connecting Freeswitch servers

2009-12-03 Thread William Suffill

http://www.freeswitch.org/

On the right side. Join IRC

Just fill in a nickname and click JOIN IRC



-- W

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Re: [Freeswitch-users] Choppy sound with PCMU


Btw, I have these popping up in my logs from time to time:

2009-12-03 09:42:06.035294 [DEBUG] switch_core_state_machine.c:314
(sofia/external/xx...@4.68.250.148) Running State Change CS_HANGUP
2009-12-03 09:42:06.035294 [CRIT] switch_time.c:473 Virtual Migration
Detected! Syncing Clock

In this case an incoming call rang to both FS and Asterisk, Asterisk picked
up, but the surge of activity made FS timer thread miss a beat or two.


eaf wrote:
 
 Oh, it's not just one timer thread... Why, why is sql_thread keeps on
 checking for messages every millisecond? Couldn't there be some signalling
 implemented that will make the thread suspend on condition variable or a
 socket/pipe in between? 
 
 #0  do_sleep (t=1000) at src/switch_time.c:109
 #1  0xb7e5e215 in switch_core_sql_thread (thread=0xb7586ae8, obj=0x0) at
 src/switch_core_sqldb.c:783
 
 Why does this sofia_profile_worker_thread keeps on looping checking for
 the queue? Have a semaphore!
 
 #0  do_sleep (t=1000) at src/switch_time.c:109
 #1  0xb73a4701 in sofia_profile_worker_thread_run (thread=0x80f3a30,
 obj=0x80f2490) at sofia.c:978
 
 Nothing's happening on the box, but there are three threads that pretend
 to be actively busy with smth. Others at least sleep for hundreds of
 milliseconds, not for one.
 
 And there is even infrastructure present to do blocking pops: i.e. why
 couldn't sqldb thread do queue_pop() instead of queue_trypop() intermixed
 with 1ms sleeps? This looping is such a waste...
 
 
 eaf wrote:
 
 As I see it, switch_cond_next() currently is just a do_sleep(1000). Yes,
 it could be mapped to a 1ms timer, but #define DISABLE_1MS_COND
 overrides that.
 
 Yeah, there is a global timestamp... It's easy to workaround that for RTP
 who calls switch_micro_time_now()... But if somebody accesses
 runtime.timestamp directly, it's gonna be tough to grep for that. If only
 this was C++...
 
 I'll play around. Never liked polling too much. Never could've guessed
 that polling could be so useful for scalability ;) My naive
 implementation would've pulled timestamp via system calls and would've
 done sleeping by passing exact interval to select() instead of syncing
 with a pacing thread. Which would be dead-quiet at idle time, but, of
 course, would stop scaling at some point due to excessive number of
 system calls.
 
 Thanks.
 
 
 Michael Jerris wrote:
 
 In short.  No, you can not for many reasons. The milisecond tic is  
 used throughout the code even when there is not any calls up.  You can  
 grep for switch_cond_next if you would like to see where but it is  
 required to keep our global timestamp and for pacing the scheduler  
 among other services that run all the time.
 
 Mike
 
 On Dec 2, 2009, at 7:31 PM, eaf erandr-j...@usa.net wrote:
 

 Can I reduce resolution of that timer thread 10 times? I mean, I  
 glanced
 through the code, and see that among others (are there others?) RTP  
 and IVR
 set up their timers that are subsequently managed by this thread.  
 RTP timers
 should be eliminated by that setting you've suggested. IVR timers  
 are set at
 20ms... So, if the thread is set to wake up every 10ms instead of  
 1ms it
 should be able to wake up those IVR timers just fine. Right?

 That's a cool design to have one dedicated thread that maintains  
 accurate
 timing and then broadcasts via condition variables to hundreds of  
 other
 threads events that they can register for. I'm sure it's one of the  
 reasons
 why FS scales so much better than Asterisk. But for poor low-end  
 setups that
 sit in the closet, eat only 6W of power and hardly ever run more  
 than two
 calls at the same time, can I hack it somehow to be more UNIX- 
 friendly? I.e.
 make it stuck in select() or recv() when there is nothing to do, call
 clock_gettime() right from the thread that wants and when it wants  
 to know
 current time?

 Say, what if that thread is made to suspend on a condition variable  
 in case
 if there are no timers registered in TIMER_MATRIX? Then, if some other
 thread comes up and adds its timer into the matrix, it could wake up  
 the
 timer thread and enjoy accurate timing as needed, on demand? And in- 
 between
 the calls, when there is no RTP or IVR, it will all go silent? I mean,
 sitting on a wait queue in the kernel is way better than go back and  
 forth
 incrementing counters that nobody even needs at the moment?


 Anthony Minessale-2 wrote:

 idle is a 4 letter word to a realtime application.

 The core keeps a single high-priority thread to keep 1ms timing and
 expands
 that broadcasting
 to hundreds or thousand of threads who need accurate timing.

 Your choppy audio is caused by linksys lying about the packet len  
 that
 it's
 using and we set our timer
 to the wrong speed.



 -- 
 View this message in context:
 http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html
 Sent from the Freeswitch-users mailing list archive at Nabble.com.


 ___

Re: [Freeswitch-users] How to run a JS script periodically

2009-12-03 Thread Rupa Schomaker

If doing this, I'd suggest checking for a global var to see if the script
should terminate itself.  Otherwise, you'll have to bring down the whole
freeswitch to stop this script.

On Thu, Dec 3, 2009 at 7:28 AM, Seven Du dujinf...@gmail.com wrote:

 Not sure about js, but in lua, you can use luarun to run a
 long-running script like


 loop
 do sth.
 sleep 5min
 end

 and also it can be set to start with freeswitch in lua.conf.xml

 I guess you can also use jsrun to run js.

 And, if you run every 5 min, why not use crontab?

 fs_cli -x jsrun xx.js


 2009/12/3 Oscav os...@hotmail.fr:
 
  Hi,
 
  Someone knows how to run periodically a JS script ?? The purpose is to
 write
  to a db some global informations (Global Variables) about FS like every 5
  minutes.
 
  Thanks.
 
 
  --
  View this message in context:
 http://old.nabble.com/How-to-run-a-JS-script-periodically-tp26625147p26625147.html
  Sent from the Freeswitch-users mailing list archive at Nabble.com.
 
 
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-- 
-Rupa
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[Freeswitch-users] Call transfer got broken for me

2009-12-03 Thread Milena

Hello,

It was all ok until yesterday when i updated to svn 15761(last update before
that was about 4 days ago), Now I have this issue:

someone from the pstn (555) calls through my FXO gw (10.1.1.90) to ext
200
200 picks up, then 200 transfers the call to 205
call gets lost (it used to transfer normal until the moment I updated)

Today I updated to 15771 and the issue is still there.
Can anyone help me figure out what is going on?

Call log: http://pastebin.freeswitch.org/11374

thank you
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Re: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted

Try trunk again

On Wed, Dec 2, 2009 at 5:33 PM, Anthony Minessale 
anthony.miness...@gmail.com wrote:

 I am not sure what you are sending over the socket but you have a queued
 hangup being processed on line 640 of your pastebin
 are you executing any commands with a ! character in it by any chance or
 executing the hangup app on purpose?




 On Wed, Dec 2, 2009 at 2:16 PM, Kristian Kielhofner 
 kristian.kielhof...@gmail.com wrote:

 Tony,

  Thanks for that but now it appears that the call just gets hung up
 on when the caller takes the callee off hold.  Debug here:

 http://pastebin.freeswitch.org/11359

  Thanks again!

 On Wed, Dec 2, 2009 at 1:13 PM, Anthony Minessale
 anthony.miness...@gmail.com wrote:
  I decided to just change the code so its more elegant to handle
 recursive
  broadcasting so you can try again and see if that helps.
 
 

 --
 Kristian Kielhofner
 http://www.astlinux.org
 http://blog.krisk.org
 http://www.star2star.com
 http://www.submityoursip.com
 http://www.voalte.com

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Re: [Freeswitch-users] HA questions.

The easiest place to do this is at the point you send the calls to FreeSWITCH.  
How are the calls coming in?

Mike

On Dec 2, 2009, at 7:49 PM, Tim Uckun wrote:

 I have read some of the archived emails about HA, loadbalancing,
 failover etc and I am still a bit confused about how I could set up
 some sort of resiliency with freeswitch.
 
 My situation is much less complex than the scenarios people were
 talking about and I hoping the solution is similarly much less
 complex.
 
 I have two machines. Both will run freeswitch and also an IVR
 application with local databases.  I will take care of the database,
 application and configuration synchronization between the two
 machines.  Ideally the calls would be load balanced between the
 machines and if any application falls down then the calls should go to
 the other machine. Same if I take a machine down for whatever reason.
 
 If a machine goes down I am willing to lose those people who were
 making a call at the time. I do have a flag in the application which
 will stop answering the calls while processing the existing calls for
 a graceful shutdown and hopefully the load balancer would shuttle the
 calls to the other machine while this is happening.
 
 At this stage everything is done via SIP.
 
 My questions are...
 
 Do I have to have a sip proxy? If the answer is yes it seems like I
 have to set up two sip proxies so I don't have another single point of
 failure. Can I load the sip proxies on the same machine? Do I need two
 more machines?
 
 If I take load balancing out of the picture would it be possible to do
 a simple linux HA or a windows built in ip failover solution? Would a
 simple IP failover work over UDP or would I have to use IAX and tcp/ip
 ?
 
 Is it better to go the virtualization route?
 
 Sorry if these are dumb questions. I am just trying to get my head
 wrapped around this. I don't need five nines (although that would be
 awesome), I just want a reasonable degree of assurance that my app can
 keep taking calls in case something weird happens.


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Re: [Freeswitch-users] can't register Inphonex

You can turn up the full freeswitch debug or enable the siptrace on the sip 
profile to get more information about this.  This looks like a nat related 
issue getting no response from the provider.  A sip trace is probably the best 
tool to figure this one out.  

sofia profile internal siptrace on

Mike

On Dec 2, 2009, at 10:35 PM, John Lalande wrote:

 I am new to FS having ditched Asterisk a few weeks ago.  I have iptel 
 registered but I cannot get Inphonex to work.  I am using the settings 
 fromhttp://wiki.freeswitch.org/wiki/Provider_Configuration:_Inphonex to no 
 avail.
  
 The error displayed in the console is 2009-12-02 21:32:55.243917 [ERR] 
 sofia_reg.c:1442 inphonex Registration Failed with status Request Timeout 
 [408].
  
 Is there some way to debug this?  sofia status displays:
  
  Name  Type   Data
   State
 =
  external   profile   sip:mod_so...@192.168.125.15:5080   
   RUNNING (0)
   example.com   gatewaysip:joeu...@example.com
   NOREG
  inphonex   gateway   sip:5285...@sip.inphonex.com
   FAILED (retry: 28s)
 iptel   gateway sip:jlala...@sip.iptel.org
   REGED
  internal   profile   sip:mod_so...@192.168.125.15:5060   
   RUNNING (0)
 internal-ipv6   profile   sip:mod_so...@[::1]:5060
   RUNNING (0)
192.168.125.15 alias   internal
   ALIASED
 =
  
  
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Re: [Freeswitch-users] Gateway issue with no audio

You may want to try this again with latest svn trunk.  We have done quite a lot 
of work to make nat support much better sense 1.0.4 

Mike

p.s. I can't comment about version 1.4 due to broken flux capacitor.


On Dec 3, 2009, at 4:36 AM, Henry Huang wrote:

 My freeswitch is using public IP. I setup a gateway registering to voipstunt, 
 and put it under internal profile. I tried to make call, and I got no RTP 
 back from the provider... Tried treating NAT issue by changing IP address, 
 internal IP, external IP. But no use, still getting no audio. 
 
 Finally, I gave up play around with the internal profile and put the gateway 
 settings under external profile. And magically, it worked. I am getting audio 
 now. But it leads me to wonders, what's the core difference between external 
 profile and internal profile. Even if I set the external SIP IP and exteranl 
 RTP IP to the public IP in internal profile, I am still getting no audio. Can 
 anyone clear the concept for me here?
 
 by the way, I am using freeswitch 1.4 stable version. 
 
 
 
 -- 
 Henry Huang
 
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Re: [Freeswitch-users] IAX? Issues connecting road warriors with SIP?

with the right clients, it nearly always works well.  with a client that does 
not support stun or at least rfc 3581 the results are much more sketchy and 
require more hacks on the server side, but with enough effort can almost always 
be made to work.

Mike

On Dec 3, 2009, at 7:17 AM, Fred-145 wrote:

 
 Hello
 
 In a thread back in March, I read that support for IAX in FreeSwitch is a
 bit of kludge and since there's not much demand for it, chances are it won't
 improve in the foreseeable future.
 
 So I'd like some feedback from users who routinely connect to a FreeSwitch
 server from various venues, ie. wifi hotspots at McD, Ethernet LAN in
 hotels, etc. (in my case, the FreeSwitch server is located in a private
 network behind a NAT router with SIP/RTP ports statically mapped.)
 
 Do you sometimes/often get issues where SIP (UDP5060) or RTP (UDPwhatever)
 ports fail being opened dynamically to work properly, or does SIP today
 really work well over NAT firewalls?
 
 Thank you.
 -- 
 View this message in context: 
 http://old.nabble.com/IAX--Issues-connecting-road-warriors-with-SIP--tp26625105p26625105.html
 Sent from the Freeswitch-users mailing list archive at Nabble.com.
 
 
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Re: [Freeswitch-users] Choppy sound with PCMU

If you see that message then your machine/os/combo is having some problems
keeping up.
It's not the timer missing anything its the monotonic clock detecting a 1
second or more differential from what its next prediction for the time
should be.  The best way to trigger this would be to suspend FS with
control-z or attach to it with gdb blocking the entire process,  that 1ms
thread would have to miss 1000 iterations to trigger that warning.

Btw, that error message is at line 471 not 473 so you are using modified
code.

Its possible your box has a bad monotonic timer, you can set

param name=disable-monotonic-timing value=true/

under settings in switch.conf.xml

We are now starting to guess you are using some small embedded type platform
perhaps?
I've run FS even on a nokia n810 and never caused that message to fire.

if 1 call can interrupt the cpu enough to  cause noticeable issues you might
want to consider running the process at a
greater priority by using the -hp command line arg or at least nice it

Why don't you tell us the whole story about what OS/platform you are using
here rather that form conjectures about what is wrong with our code that
thousands of people are happy with.







On Thu, Dec 3, 2009 at 8:55 AM, eaf erandr-j...@usa.net wrote:


 Btw, I have these popping up in my logs from time to time:

 2009-12-03 09:42:06.035294 [DEBUG] switch_core_state_machine.c:314
 (sofia/external/xx...@4.68.250.148) Running State Change CS_HANGUP
 2009-12-03 09:42:06.035294 [CRIT] switch_time.c:473 Virtual Migration
 Detected! Syncing Clock

 In this case an incoming call rang to both FS and Asterisk, Asterisk picked
 up, but the surge of activity made FS timer thread miss a beat or two.


 eaf wrote:
 
  Oh, it's not just one timer thread... Why, why is sql_thread keeps on
  checking for messages every millisecond? Couldn't there be some
 signalling
  implemented that will make the thread suspend on condition variable or a
  socket/pipe in between?
 
  #0  do_sleep (t=1000) at src/switch_time.c:109
  #1  0xb7e5e215 in switch_core_sql_thread (thread=0xb7586ae8, obj=0x0) at
  src/switch_core_sqldb.c:783
 
  Why does this sofia_profile_worker_thread keeps on looping checking for
  the queue? Have a semaphore!
 
  #0  do_sleep (t=1000) at src/switch_time.c:109
  #1  0xb73a4701 in sofia_profile_worker_thread_run (thread=0x80f3a30,
  obj=0x80f2490) at sofia.c:978
 
  Nothing's happening on the box, but there are three threads that pretend
  to be actively busy with smth. Others at least sleep for hundreds of
  milliseconds, not for one.
 
  And there is even infrastructure present to do blocking pops: i.e. why
  couldn't sqldb thread do queue_pop() instead of queue_trypop() intermixed
  with 1ms sleeps? This looping is such a waste...
 
 
  eaf wrote:
 
  As I see it, switch_cond_next() currently is just a do_sleep(1000). Yes,
  it could be mapped to a 1ms timer, but #define DISABLE_1MS_COND
  overrides that.
 
  Yeah, there is a global timestamp... It's easy to workaround that for
 RTP
  who calls switch_micro_time_now()... But if somebody accesses
  runtime.timestamp directly, it's gonna be tough to grep for that. If
 only
  this was C++...
 
  I'll play around. Never liked polling too much. Never could've guessed
  that polling could be so useful for scalability ;) My naive
  implementation would've pulled timestamp via system calls and would've
  done sleeping by passing exact interval to select() instead of syncing
  with a pacing thread. Which would be dead-quiet at idle time, but, of
  course, would stop scaling at some point due to excessive number of
  system calls.
 
  Thanks.
 
 
  Michael Jerris wrote:
 
  In short.  No, you can not for many reasons. The milisecond tic is
  used throughout the code even when there is not any calls up.  You can
  grep for switch_cond_next if you would like to see where but it is
  required to keep our global timestamp and for pacing the scheduler
  among other services that run all the time.
 
  Mike
 
  On Dec 2, 2009, at 7:31 PM, eaf erandr-j...@usa.net wrote:
 
 
  Can I reduce resolution of that timer thread 10 times? I mean, I
  glanced
  through the code, and see that among others (are there others?) RTP
  and IVR
  set up their timers that are subsequently managed by this thread.
  RTP timers
  should be eliminated by that setting you've suggested. IVR timers
  are set at
  20ms... So, if the thread is set to wake up every 10ms instead of
  1ms it
  should be able to wake up those IVR timers just fine. Right?
 
  That's a cool design to have one dedicated thread that maintains
  accurate
  timing and then broadcasts via condition variables to hundreds of
  other
  threads events that they can register for. I'm sure it's one of the
  reasons
  why FS scales so much better than Asterisk. But for poor low-end
  setups that
  sit in the closet, eat only 6W of power and hardly ever run more
  than two
  calls at the same time, can I hack it somehow to be more

Re: [Freeswitch-users] Call transfer got broken for me

what revision were you at prior to upgrade or can you narrow the range of 
versions that broke this any more (or even better the exact version that broke 
this).  Please post this bug to http://jira.freeswitch.org.

Mike

On Dec 3, 2009, at 10:30 AM, Milena wrote:

 Hello,
 
 It was all ok until yesterday when i updated to svn 15761(last update before 
 that was about 4 days ago), Now I have this issue:
 
 someone from the pstn (555) calls through my FXO gw (10.1.1.90) to ext 200
 200 picks up, then 200 transfers the call to 205
 call gets lost (it used to transfer normal until the moment I updated)
 
 Today I updated to 15771 and the issue is still there.
 Can anyone help me figure out what is going on?
 
 Call log: http://pastebin.freeswitch.org/11374
 
 thank you 
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Re: [Freeswitch-users] Call transfer got broken for me

2009-12-03 Thread Milena

This got fixed in version 15773, thank you very much

2009/12/3 Michael Jerris m...@jerris.com

 what revision were you at prior to upgrade or can you narrow the range of
 versions that broke this any more (or even better the exact version that
 broke this).  Please post this bug to http://jira.freeswitch.org.

 Mike

 On Dec 3, 2009, at 10:30 AM, Milena wrote:

 Hello,

 It was all ok until yesterday when i updated to svn 15761(last update
 before that was about 4 days ago), Now I have this issue:

 someone from the pstn (555) calls through my FXO gw (10.1.1.90) to ext
 200
 200 picks up, then 200 transfers the call to 205
 call gets lost (it used to transfer normal until the moment I updated)

 Today I updated to 15771 and the issue is still there.
 Can anyone help me figure out what is going on?

 Call log: http://pastebin.freeswitch.org/11374

 thank you
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Re: [Freeswitch-users] Call transfer got broken for me

to late it's fixed now.


On Thu, Dec 3, 2009 at 10:21 AM, Michael Jerris m...@jerris.com wrote:

 what revision were you at prior to upgrade or can you narrow the range of
 versions that broke this any more (or even better the exact version that
 broke this).  Please post this bug to http://jira.freeswitch.org.

 Mike

 On Dec 3, 2009, at 10:30 AM, Milena wrote:

 Hello,

 It was all ok until yesterday when i updated to svn 15761(last update
 before that was about 4 days ago), Now I have this issue:

 someone from the pstn (555) calls through my FXO gw (10.1.1.90) to ext
 200
 200 picks up, then 200 transfers the call to 205
 call gets lost (it used to transfer normal until the moment I updated)

 Today I updated to 15771 and the issue is still there.
 Can anyone help me figure out what is going on?

 Call log: http://pastebin.freeswitch.org/11374

 thank you
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-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
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FreeSWITCH Developer Conference
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Re: [Freeswitch-users] Eavesdrop error?

you could check if the uuid is blank with an expression and playback an
audio warning that it's an invalid call.


On Thu, Dec 3, 2009 at 8:08 AM, Michael Jerris m...@jerris.com wrote:

 The behavior is probably expected, the unhelpful error is probably
 undesirable but it would make a mess of the dial-plan to clean that up.

 Mike

 On Dec 2, 2009, at 9:19 PM, Lars Zeb wrote:

  Is this reasonable given it was the only call in FreeSwitch at the time?
 How
  can this situation be corrected in the future?
 
  From: freeswitch-users-boun...@lists.freeswitch.org
  [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
 Anthony
  Minessale
  Sent: Wednesday, December 02, 2009 3:35 PM
  To: freeswitch-users@lists.freeswitch.org
  Subject: Re: [Freeswitch-users] Eavesdrop error?
 
  it probably just means the uuid was not retrieved from the db when you
  called the eavesdrop exten which does the lookup on the uuid for the hash
  key based on what ext you hit to retrieve the most recent uuid that
 called
  that ext.
 
 
  On Wed, Dec 2, 2009 at 5:22 PM, Lars Zeb larc...@yahoo.com wrote:
  Sorry, svn 15753
 
  -Original Message-
  From: freeswitch-users-boun...@lists.freeswitch.org
  [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Lars
 Zeb
  Sent: Wednesday, December 02, 2009 2:08 PM
  To: freeswitch-users@lists.freeswitch.org
  Subject: [Freeswitch-users] Eavesdrop error?
 
  I tried to use eavesdrop today and it did not work. The error message in
 the
  log is:
 
  [ERR] mod_dptools.c:334 Usage: [all | uuid]
 
  I simply dialed 881010, trying to eavesdrop on extension 1010. Is this
  incorrect?
 
  http://pastebin.freeswitch.org/11363
 
  Thanks Lars
 
 
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  googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
  pstn:213-799-1400
 
 
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-- 
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FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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Re: [Freeswitch-users] HA questions.

2009-12-03 Thread Adam Ford

Have you checked out Redfone? While I haven't attempted to implement it yet,
my Redfone foneBridge2 claims to be able to handle load balancing and
failover between two Asterisk/Freeswitch servers.

-AF

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Tim
Uckun
Sent: Wednesday, December 02, 2009 5:50 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] HA questions.

I have read some of the archived emails about HA, loadbalancing,
failover etc and I am still a bit confused about how I could set up
some sort of resiliency with freeswitch.

My situation is much less complex than the scenarios people were
talking about and I hoping the solution is similarly much less
complex.

I have two machines. Both will run freeswitch and also an IVR
application with local databases.  I will take care of the database,
application and configuration synchronization between the two
machines.  Ideally the calls would be load balanced between the
machines and if any application falls down then the calls should go to
the other machine. Same if I take a machine down for whatever reason.

If a machine goes down I am willing to lose those people who were
making a call at the time. I do have a flag in the application which
will stop answering the calls while processing the existing calls for
a graceful shutdown and hopefully the load balancer would shuttle the
calls to the other machine while this is happening.

At this stage everything is done via SIP.

My questions are...

Do I have to have a sip proxy? If the answer is yes it seems like I
have to set up two sip proxies so I don't have another single point of
failure. Can I load the sip proxies on the same machine? Do I need two
more machines?

If I take load balancing out of the picture would it be possible to do
a simple linux HA or a windows built in ip failover solution? Would a
simple IP failover work over UDP or would I have to use IAX and tcp/ip
?

Is it better to go the virtualization route?

Sorry if these are dumb questions. I am just trying to get my head
wrapped around this. I don't need five nines (although that would be
awesome), I just want a reasonable degree of assurance that my app can
keep taking calls in case something weird happens.

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Re: [Freeswitch-users] Choppy sound with PCMU


I'm sorry if I sounded that way. Did mean to. :)

Yes, it's an embedded platform. It's an ALIX board with AMD Geode LX800 chip
and 256MB of RAM. http://www.pcengines.ch/alix2d3.htm

Line offset difference is due to some minor logging changes I made to see
who's allocating timers and how often. This way I found MOH streaming and
that RTP still allocates timers even when it's set to none in the profile.

I feel that this platform turned out to be underpowered for FS because it
cannot meet its scheduling expectations. I guess, some degree of kernel
tweaking or setting priorities will fix that. Meanwhile I just got rid of
the SQLDB 1ms thread via -nosql command line option, split sofia worker 1ms
thread in two (one blocked and waiting for new commands in the SQL queue,
the other one checking registrations and gateways with 1sec interval), and
don't know yet what to do about the timer thread.

Again, I apologize for stupid or accusing questions, I'm just trying to see
how FS can be made friendlier to this board. Or the board be made friendlier
to FS ;)


Anthony Minessale-2 wrote:
 
 If you see that message then your machine/os/combo is having some problems
 keeping up.
 It's not the timer missing anything its the monotonic clock detecting a 1
 second or more differential from what its next prediction for the time
 should be.  The best way to trigger this would be to suspend FS with
 control-z or attach to it with gdb blocking the entire process,  that 1ms
 thread would have to miss 1000 iterations to trigger that warning.
 
 Btw, that error message is at line 471 not 473 so you are using modified
 code.
 
 Its possible your box has a bad monotonic timer, you can set
 
 
 
 under settings in switch.conf.xml
 
 We are now starting to guess you are using some small embedded type
 platform
 perhaps?
 I've run FS even on a nokia n810 and never caused that message to fire.
 
 if 1 call can interrupt the cpu enough to  cause noticeable issues you
 might
 want to consider running the process at a
 greater priority by using the -hp command line arg or at least nice it
 
 Why don't you tell us the whole story about what OS/platform you are using
 here rather that form conjectures about what is wrong with our code that
 thousands of people are happy with.
 
 
 
 
 
 
 
 On Thu, Dec 3, 2009 at 8:55 AM, eaf erandr-j...@usa.net wrote:
 

 Btw, I have these popping up in my logs from time to time:

 2009-12-03 09:42:06.035294 [DEBUG] switch_core_state_machine.c:314
 (sofia/external/xx...@4.68.250.148) Running State Change CS_HANGUP
 2009-12-03 09:42:06.035294 [CRIT] switch_time.c:473 Virtual Migration
 Detected! Syncing Clock

 In this case an incoming call rang to both FS and Asterisk, Asterisk
 picked
 up, but the surge of activity made FS timer thread miss a beat or two.


 eaf wrote:
 
  Oh, it's not just one timer thread... Why, why is sql_thread keeps on
  checking for messages every millisecond? Couldn't there be some
 signalling
  implemented that will make the thread suspend on condition variable or
 a
  socket/pipe in between?
 
  #0  do_sleep (t=1000) at src/switch_time.c:109
  #1  0xb7e5e215 in switch_core_sql_thread (thread=0xb7586ae8, obj=0x0)
 at
  src/switch_core_sqldb.c:783
 
  Why does this sofia_profile_worker_thread keeps on looping checking for
  the queue? Have a semaphore!
 
  #0  do_sleep (t=1000) at src/switch_time.c:109
  #1  0xb73a4701 in sofia_profile_worker_thread_run (thread=0x80f3a30,
  obj=0x80f2490) at sofia.c:978
 
  Nothing's happening on the box, but there are three threads that
 pretend
  to be actively busy with smth. Others at least sleep for hundreds of
  milliseconds, not for one.
 
  And there is even infrastructure present to do blocking pops: i.e. why
  couldn't sqldb thread do queue_pop() instead of queue_trypop()
 intermixed
  with 1ms sleeps? This looping is such a waste...
 
 
  eaf wrote:
 
  As I see it, switch_cond_next() currently is just a do_sleep(1000).
 Yes,
  it could be mapped to a 1ms timer, but #define DISABLE_1MS_COND
  overrides that.
 
  Yeah, there is a global timestamp... It's easy to workaround that for
 RTP
  who calls switch_micro_time_now()... But if somebody accesses
  runtime.timestamp directly, it's gonna be tough to grep for that. If
 only
  this was C++...
 
  I'll play around. Never liked polling too much. Never could've guessed
  that polling could be so useful for scalability ;) My naive
  implementation would've pulled timestamp via system calls and would've
  done sleeping by passing exact interval to select() instead of syncing
  with a pacing thread. Which would be dead-quiet at idle time, but, of
  course, would stop scaling at some point due to excessive number of
  system calls.
 
  Thanks.
 
 
  Michael Jerris wrote:
 
  In short.  No, you can not for many reasons. The milisecond tic is
  used throughout the code even when there is not any calls up.  You
 can
  grep for switch_cond_next if you would like to see where but it is
  required

Re: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted

2009-12-03 Thread Kristian Kielhofner

Tony,

  The call no longer hangs up but we still only get hold music in one
direction - if the callee places the caller on hold there is no music.

PB here:
http://pastebin.freeswitch.org/11378

  This was on rev 15773.

Thanks again Tony!

On Thu, Dec 3, 2009 at 10:56 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
 Try trunk again


-- 
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http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com

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Re: [Freeswitch-users] Choppy sound with PCMU

2009-12-03 Thread Kristian Kielhofner

I don't think it's the board itself...

We have extensively tested FreeSwitch (no modifications) on that exact
board with AstLinux and have it running at multiple customer
locations.

No timing errors, no warnings or errors of any kind.  Pretty standard
really just don't expect too much from the LX800 (transcoding,
resampling, massive numbers of calls, etc).

On Thu, Dec 3, 2009 at 12:29 PM, eaf erandr-j...@usa.net wrote:

 I'm sorry if I sounded that way. Did mean to. :)

 Yes, it's an embedded platform. It's an ALIX board with AMD Geode LX800 chip
 and 256MB of RAM. http://www.pcengines.ch/alix2d3.htm

 Line offset difference is due to some minor logging changes I made to see
 who's allocating timers and how often. This way I found MOH streaming and
 that RTP still allocates timers even when it's set to none in the profile.

 I feel that this platform turned out to be underpowered for FS because it
 cannot meet its scheduling expectations. I guess, some degree of kernel
 tweaking or setting priorities will fix that. Meanwhile I just got rid of
 the SQLDB 1ms thread via -nosql command line option, split sofia worker 1ms
 thread in two (one blocked and waiting for new commands in the SQL queue,
 the other one checking registrations and gateways with 1sec interval), and
 don't know yet what to do about the timer thread.

 Again, I apologize for stupid or accusing questions, I'm just trying to see
 how FS can be made friendlier to this board. Or the board be made friendlier
 to FS ;)



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com

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[Freeswitch-users] Cannot Do this Basic thing

I have copied 1001.xml in directory/default to a test user 2319.xm 
changing or instances of 1001 in the file to 2319. I then went into 
default.xml  in directory folder and in one of the groups  just mimicked 
1001 details by changing 1001 to 2319.

Connecting  to FS gives Forbidden message. However 1001 connects without 
a problem.  What have I missed ?

Is there a place that just puts things in do this and that and that to 
create a new user ?

Thanks

 

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Re: [Freeswitch-users] Choppy sound with PCMU

I know people with hardware out there in production based on arm11 and those 
are pretty small processors, not sure how they compare to this.  In regards to 
the DISABLE_1MS_COND, try getting rid of that, it did increase performance on 
the high end but may be better for you on the low end with lower compute on 
idle busy loops.

Mike

On Dec 3, 2009, at 12:29 PM, eaf wrote:

 
 I'm sorry if I sounded that way. Did mean to. :)
 
 Yes, it's an embedded platform. It's an ALIX board with AMD Geode LX800 chip
 and 256MB of RAM. http://www.pcengines.ch/alix2d3.htm
 
 Line offset difference is due to some minor logging changes I made to see
 who's allocating timers and how often. This way I found MOH streaming and
 that RTP still allocates timers even when it's set to none in the profile.
 
 I feel that this platform turned out to be underpowered for FS because it
 cannot meet its scheduling expectations. I guess, some degree of kernel
 tweaking or setting priorities will fix that. Meanwhile I just got rid of
 the SQLDB 1ms thread via -nosql command line option, split sofia worker 1ms
 thread in two (one blocked and waiting for new commands in the SQL queue,
 the other one checking registrations and gateways with 1sec interval), and
 don't know yet what to do about the timer thread.
 
 Again, I apologize for stupid or accusing questions, I'm just trying to see
 how FS can be made friendlier to this board. Or the board be made friendlier
 to FS ;)
 
 
 Anthony Minessale-2 wrote:
 
 If you see that message then your machine/os/combo is having some problems
 keeping up.
 It's not the timer missing anything its the monotonic clock detecting a 1
 second or more differential from what its next prediction for the time
 should be.  The best way to trigger this would be to suspend FS with
 control-z or attach to it with gdb blocking the entire process,  that 1ms
 thread would have to miss 1000 iterations to trigger that warning.
 
 Btw, that error message is at line 471 not 473 so you are using modified
 code.
 
 Its possible your box has a bad monotonic timer, you can set
 
 
 
 under settings in switch.conf.xml
 
 We are now starting to guess you are using some small embedded type
 platform
 perhaps?
 I've run FS even on a nokia n810 and never caused that message to fire.
 
 if 1 call can interrupt the cpu enough to  cause noticeable issues you
 might
 want to consider running the process at a
 greater priority by using the -hp command line arg or at least nice it
 
 Why don't you tell us the whole story about what OS/platform you are using
 here rather that form conjectures about what is wrong with our code that
 thousands of people are happy with.
 
 
 
 
 
 
 
 On Thu, Dec 3, 2009 at 8:55 AM, eaf erandr-j...@usa.net wrote:
 
 
 Btw, I have these popping up in my logs from time to time:
 
 2009-12-03 09:42:06.035294 [DEBUG] switch_core_state_machine.c:314
 (sofia/external/xx...@4.68.250.148) Running State Change CS_HANGUP
 2009-12-03 09:42:06.035294 [CRIT] switch_time.c:473 Virtual Migration
 Detected! Syncing Clock
 
 In this case an incoming call rang to both FS and Asterisk, Asterisk
 picked
 up, but the surge of activity made FS timer thread miss a beat or two.
 
 
 eaf wrote:
 
 Oh, it's not just one timer thread... Why, why is sql_thread keeps on
 checking for messages every millisecond? Couldn't there be some
 signalling
 implemented that will make the thread suspend on condition variable or
 a
 socket/pipe in between?
 
 #0  do_sleep (t=1000) at src/switch_time.c:109
 #1  0xb7e5e215 in switch_core_sql_thread (thread=0xb7586ae8, obj=0x0)
 at
 src/switch_core_sqldb.c:783
 
 Why does this sofia_profile_worker_thread keeps on looping checking for
 the queue? Have a semaphore!
 
 #0  do_sleep (t=1000) at src/switch_time.c:109
 #1  0xb73a4701 in sofia_profile_worker_thread_run (thread=0x80f3a30,
 obj=0x80f2490) at sofia.c:978
 
 Nothing's happening on the box, but there are three threads that
 pretend
 to be actively busy with smth. Others at least sleep for hundreds of
 milliseconds, not for one.
 
 And there is even infrastructure present to do blocking pops: i.e. why
 couldn't sqldb thread do queue_pop() instead of queue_trypop()
 intermixed
 with 1ms sleeps? This looping is such a waste...
 
 
 eaf wrote:
 
 As I see it, switch_cond_next() currently is just a do_sleep(1000).
 Yes,
 it could be mapped to a 1ms timer, but #define DISABLE_1MS_COND
 overrides that.
 
 Yeah, there is a global timestamp... It's easy to workaround that for
 RTP
 who calls switch_micro_time_now()... But if somebody accesses
 runtime.timestamp directly, it's gonna be tough to grep for that. If
 only
 this was C++...
 
 I'll play around. Never liked polling too much. Never could've guessed
 that polling could be so useful for scalability ;) My naive
 implementation would've pulled timestamp via system calls and would've
 done sleeping by passing exact interval to select() instead of syncing
 with a pacing thread. Which would be

Re: [Freeswitch-users] Cannot Do this Basic thing

2009-12-03 Thread Michael Collins

On Thu, Dec 3, 2009 at 9:46 AM, Samuel Abekah-Mensah ab...@greatiam.comwrote:

 I have copied 1001.xml in directory/default to a test user 2319.xm
 changing or instances of 1001 in the file to 2319. I then went into
 default.xml  in directory folder and in one of the groups  just mimicked
 1001 details by changing 1001 to 2319.

 Connecting  to FS gives Forbidden message. However 1001 connects without
 a problem.  What have I missed ?

 Is there a place that just puts things in do this and that and that to
 create a new user ?


Did you execute reloadxml from the fs cli before trying to connect with
2319? Also I'm assuming that 2319.xm is a typo and you actually created
2319.xml in the default/directory subdir.
-MC
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Re: [Freeswitch-users] Choppy sound with PCMU

What about the things I spent time suggesting in my last email?
Did you try them because I was actually curious if they made any impact.


On Thu, Dec 3, 2009 at 11:29 AM, eaf erandr-j...@usa.net wrote:


 I'm sorry if I sounded that way. Did mean to. :)

 Yes, it's an embedded platform. It's an ALIX board with AMD Geode LX800
 chip
 and 256MB of RAM. http://www.pcengines.ch/alix2d3.htm

 Line offset difference is due to some minor logging changes I made to see
 who's allocating timers and how often. This way I found MOH streaming and
 that RTP still allocates timers even when it's set to none in the profile.

 I feel that this platform turned out to be underpowered for FS because it
 cannot meet its scheduling expectations. I guess, some degree of kernel
 tweaking or setting priorities will fix that. Meanwhile I just got rid of
 the SQLDB 1ms thread via -nosql command line option, split sofia worker 1ms
 thread in two (one blocked and waiting for new commands in the SQL queue,
 the other one checking registrations and gateways with 1sec interval), and
 don't know yet what to do about the timer thread.

 Again, I apologize for stupid or accusing questions, I'm just trying to see
 how FS can be made friendlier to this board. Or the board be made
 friendlier
 to FS ;)


 Anthony Minessale-2 wrote:
 
  If you see that message then your machine/os/combo is having some
 problems
  keeping up.
  It's not the timer missing anything its the monotonic clock detecting a 1
  second or more differential from what its next prediction for the time
  should be.  The best way to trigger this would be to suspend FS with
  control-z or attach to it with gdb blocking the entire process,  that 1ms
  thread would have to miss 1000 iterations to trigger that warning.
 
  Btw, that error message is at line 471 not 473 so you are using modified
  code.
 
  Its possible your box has a bad monotonic timer, you can set
 
 
 
  under settings in switch.conf.xml
 
  We are now starting to guess you are using some small embedded type
  platform
  perhaps?
  I've run FS even on a nokia n810 and never caused that message to fire.
 
  if 1 call can interrupt the cpu enough to  cause noticeable issues you
  might
  want to consider running the process at a
  greater priority by using the -hp command line arg or at least nice it
 
  Why don't you tell us the whole story about what OS/platform you are
 using
  here rather that form conjectures about what is wrong with our code that
  thousands of people are happy with.
 
 
 
 
 
 
 
  On Thu, Dec 3, 2009 at 8:55 AM, eaf erandr-j...@usa.net wrote:
 
 
  Btw, I have these popping up in my logs from time to time:
 
  2009-12-03 09:42:06.035294 [DEBUG] switch_core_state_machine.c:314
  (sofia/external/xx...@4.68.250.148) Running State Change CS_HANGUP
  2009-12-03 09:42:06.035294 [CRIT] switch_time.c:473 Virtual Migration
  Detected! Syncing Clock
 
  In this case an incoming call rang to both FS and Asterisk, Asterisk
  picked
  up, but the surge of activity made FS timer thread miss a beat or two.
 
 
  eaf wrote:
  
   Oh, it's not just one timer thread... Why, why is sql_thread keeps on
   checking for messages every millisecond? Couldn't there be some
  signalling
   implemented that will make the thread suspend on condition variable or
  a
   socket/pipe in between?
  
   #0  do_sleep (t=1000) at src/switch_time.c:109
   #1  0xb7e5e215 in switch_core_sql_thread (thread=0xb7586ae8, obj=0x0)
  at
   src/switch_core_sqldb.c:783
  
   Why does this sofia_profile_worker_thread keeps on looping checking
 for
   the queue? Have a semaphore!
  
   #0  do_sleep (t=1000) at src/switch_time.c:109
   #1  0xb73a4701 in sofia_profile_worker_thread_run (thread=0x80f3a30,
   obj=0x80f2490) at sofia.c:978
  
   Nothing's happening on the box, but there are three threads that
  pretend
   to be actively busy with smth. Others at least sleep for hundreds of
   milliseconds, not for one.
  
   And there is even infrastructure present to do blocking pops: i.e. why
   couldn't sqldb thread do queue_pop() instead of queue_trypop()
  intermixed
   with 1ms sleeps? This looping is such a waste...
  
  
   eaf wrote:
  
   As I see it, switch_cond_next() currently is just a do_sleep(1000).
  Yes,
   it could be mapped to a 1ms timer, but #define DISABLE_1MS_COND
   overrides that.
  
   Yeah, there is a global timestamp... It's easy to workaround that for
  RTP
   who calls switch_micro_time_now()... But if somebody accesses
   runtime.timestamp directly, it's gonna be tough to grep for that. If
  only
   this was C++...
  
   I'll play around. Never liked polling too much. Never could've
 guessed
   that polling could be so useful for scalability ;) My naive
   implementation would've pulled timestamp via system calls and
 would've
   done sleeping by passing exact interval to select() instead of
 syncing
   with a pacing thread. Which would be dead-quiet at idle time, but, of
   course, would stop scaling at

Re: [Freeswitch-users] Cisco IOS gateway: command to send connected line name

2009-12-03 Thread Metik

Yehavi,

There are a few variations of transmitting this information... If you 
have already enabled a supplemental isdn service profile, try adding the 
following to the PRI you are using:

(config-if)#isdn outgoing ie facility
(config-if)#iisdn outgoing ie extended-facility 
(config-if)#isdn outgoing display-ie
(config-if)#isdn outgoing ie caller-number
(config-if)#isdn outgoing ie called-number

-metik

Yehavi Bourvine wrote:
 Hello,
  
   We have a Cisco running IOS 12.4T used as our SIP-PRI gateway. On 
 the PRI there is a Nortel with Q.Sig. After a lot of configuration 
 trials I've managed to set it to send back the connected name over the 
 SIP (i.e. when a call goes from SIP to PRI, the PRI sends back the 
 connected name and then the Cisco adds it as a Remote-Party-ID). 
 However, I did not save it and a power outage cleared this config. In 
 my age I don't remember what I've done...
  
 Anyone knows the correct config?
  
Thanks! __Yehavi:
 

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Re: [Freeswitch-users] uuid_bridge kills both channels if they are executing java app

2009-12-03 Thread Artem Shiyanov

I've sent deep-breath message to the dev list.
Just-in-case, here is a cross-post:




Hi there!

This message is a forward from user-mail-list.
I'm trying to fix such a problem:
FreSwithch compiled from SVN-trunk, date = 11/02/2009.

What is need: connect two users, initially one is on the home-grown
java-based IVR and other party is off hook.

What is done/got:
User1 is on the java application, it represents simple IVR system, and the
most used FS API operation is streamFile.
User2 is off hook.
next:
(mod_socket) create_uuid
get uuid_x
bgapi originate
{origination_caller_id_name=User1}[origination_uuid=uuid_x]User1 park()
User1 answers incomming call
get event channel_User1 answered
get event channel_User1 parked
uuid_bridge uuid_User1 uuid_User2
get event channel_User1 hangup, cause=NORMAL_CLEARING
get event channel_User2 hangup, cause=*DESTINATION_OUT_OF_ORDER*
both channel_User1 and channel_User2 are down


FS log is here: http://pastebin.freeswitch.org/11380


Thank you much for any help,
Artem








On Wed, Dec 2, 2009 at 10:24 PM, Anthony Minessale 
anthony.miness...@gmail.com wrote:

 you should be working on SVN trunk if you are doing development, we are so
 far forward from 1.0.4 we can't do debugging very easily.

 I don't know all of the details of what you are trying to do but you are
 hitting some race conditions because of the async nature of the socket
 connection and the way you are using it.




 On Wed, Dec 2, 2009 at 1:08 PM, Artem Shiyanov shiya...@gmail.com wrote:

 I'm back again with the same issue.
 Now it is became worse: it reproduces occasionally.
 [FS version is 1.04, test_load = 2 active calls]

 I've got 2 logs: successful and not.
 Here is a bad_case:

 2009-12-02 13:27:55.159931 [NOTICE] switch_core_session.c:1576 Execute
 java(/usr/local/freeswitch/scripts/fs2agi.jar
 org.starpound.fs2agi.Translator
 ${agi_url})
 Dec 2, 2009 1:27:55 PM org.starpound.fs2agi.Translator run
 INFO: ***
 Dec 2, 2009 1:27:55 PM org.starpound.fs2agi.Translator run
 INFO: Run AGI application agi://localhost:4573/hello.agi?callId=929 for
 session
 2898ad41-4ec1-4628-89fd-651a93a7221d
 2009-12-02 13:27:55.169841 [NOTICE] switch_cpp.cpp:1130 Run AGI
 application
 agi://localhost:4573/hello.agi?callId=929
 2009-12-02 13:28:02.31 [CRIT] mod_local_stream.c:234 Leaking stream
 handle!

 [switch_ivr_play_file() src/switch_ivr_play_say.c:1026]
 2009-12-02 13:28:04.799806 [NOTICE] switch_channel.c:602 New Channel
 sofia/internal/2001 [76d2c0e9-16a4-4098-92c0-5977cb482e17]
 2009-12-02 13:28:05.148834 [NOTICE] sofia.c:3353 Ring-Ready
 sofia/internal/2001!
 2009-12-02 13:28:05.855093 [NOTICE] sofia.c:3794 Channel
 [sofia/internal/2001] has
 been answered
 Dec 2, 2009 1:28:05 PM org.starpound.fs2agi.Translator tellAllWeCrashed
 INFO: AGI application agi://localhost:4573/hello.agi?callId=929 for
 session
 2898ad41-4ec1-4628-89fd-651a93a7221d crashed. Exception is:
 java.lang.Exception: Internal FreeSwitch failure while streamming file,
 see
 FreeSwitch logs for details
 at

 org.starpound.fs2agi.agicommands.StreamFileCommand.execute(StreamFileCommand.java:36)
 at org.starpound.fs2agi.AgiConnection.run(AgiConnection.java:48)
 at org.starpound.fs2agi.Translator.run(Translator.java:56)
 at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method)
 at

 sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39)
 at

 sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25)
 at java.lang.reflect.Method.invoke(Method.java:597)
 at org.freeswitch.Launcher.launch(Launcher.java:80)
 2009-12-02 13:28:05.870185 [NOTICE] switch_core_state_machine.c:179 Hangup
 sofia/internal/2001 [CS_EXECUTE] [NORMAL_CLEARING]
 2009-12-02 13:28:05.878807 [INFO] switch_cpp.cpp:1130 AGI application
 agi://localhost:4573/hello.agi?callId=929 crashed. See FS2AGI log for
 details.
 2009-12-02 13:28:05.894422 [NOTICE] switch_ivr_bridge.c:667 Hangup
 sofia/external/6786081...@66.19.38.143 [CS_SOFT_EXECUTE]
 [DESTINATION_OUT_OF_ORDER]
 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1086 Session 17
 (sofia/external/6786081...@66.19.38.143) Ended
 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1088 Close
 Channel
 sofia/external/6786081...@66.19.38.143 [CS_DESTROY]
 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1086 Session 18
 (sofia/internal/2001) Ended
 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1088 Close
 Channel
 sofia/internal/2001 [CS_DESTROY]



 Message
 Dec 2, 2009 1:28:05 PM org.starpound.fs2agi.Translator tellAllWeCrashed
 INFO: AGI application agi://localhost:4573/hello.agi?callId=929 for
 session
 2898ad41-4ec1-4628-89fd-651a93a7221d crashed. Exception is:
 ...
 is sent from my app upon the onHangup().`

 And here is a good_case:

 2009-12-02 13:31:45.959813 [NOTICE] switch_core_session.c:1576 Execute

[Freeswitch-users] Dialplan behavior

2009-12-03 Thread David Laperle

Hi guys,

i have a weird problem with my dialplans. For the moment, i have only 2
«usable» extensions. They were working #1 yesterday, but this morning i
realize i forgot to compile mod_python, so i go back into my source
folder and modify the modules.conf to uncomment mod_python, did a make
and make install (i did a backup of my conf folder before)! The make and
make install worked flawlessly. Then i put back my bkp of conf
directory.

I restarted the freeswitch service, created my python test dialplan and
entered into cli to see what's gonna happen! To my surprise, the call
didn't processed to the extension i was dialing.

i tried all the other extensions i had, they were all not working

After that i realized that the .xml in freeswitch/dialplan/default/
weren't imported into configuration at startup ... 

I have read all the documentation about difference between public and
default dialplan and i understand them correctly, in public if i include
all default folder, it's working again (i can reach all my extensions in
default.

My extensions are in the correct user_context ... i did nothing since
yesterday other than a make  make install after enabling python ...

Any other user have an idea why the default/*.xml aren't processed
automatically? What could i have done wrong so they are no longer
processed?

Thanks a lot,

David Laperle 
Administrateur réseau / Network administrator
(514) 393-7647 
dlape...@rsslex.com

Robinson Sheppard Shapiro s.e.n.c.r.l/LLP
Avocats / Barristers  Solicitors
4600 - 800 Place Victoria
Montréal Qc H4Z 1H6
T (514) 878-2631 F (514) 878-1865
www.rsslex.com et/and www.rsscanadaimmigration.com 






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Re: [Freeswitch-users] Cannot Do this Basic thing

Hi

Sorry .xm is a typo. I actually shut down the server and restarted. The 
log says I need to create a domain of aaa.bbb.ccc.ddd (which is the 
server IP address ) and then put the user in that domain.  Isn't the 
default domain that of the server FS is running on ?
2319.xml is in /usr/local/freeswitch/conf/directory/default/

Thanks for your time





Michael Collins wrote:


 On Thu, Dec 3, 2009 at 9:46 AM, Samuel Abekah-Mensah 
 ab...@greatiam.com mailto:ab...@greatiam.com wrote:

 I have copied 1001.xml in directory/default to a test user 2319.xm
 changing or instances of 1001 in the file to 2319. I then went into
 default.xml  in directory folder and in one of the groups  just
 mimicked
 1001 details by changing 1001 to 2319.

 Connecting  to FS gives Forbidden message. However 1001 connects
 without
 a problem.  What have I missed ?

 Is there a place that just puts things in do this and that and that to
 create a new user ?


 Did you execute reloadxml from the fs cli before trying to connect 
 with 2319? Also I'm assuming that 2319.xm is a typo and you actually 
 created 2319.xml in the default/directory subdir.
 -MC


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Re: [Freeswitch-users] Cannot Do this Basic thing

Hi

Sorry .xm is a typo. I actually shut down the server and restarted. The 
log says I need to create a domain of aaa.bbb.ccc.ddd (which is the 
server IP address ) and then put the user in that domain.  Isn't the 
default domain that of the server FS is running on ?
2319.xml is in /usr/local/freeswitch/conf/directory/default/

Thanks for your time





Michael Collins wrote:


 On Thu, Dec 3, 2009 at 9:46 AM, Samuel Abekah-Mensah 
 ab...@greatiam.com mailto:ab...@greatiam.com wrote:

 I have copied 1001.xml in directory/default to a test user 2319.xm
 changing or instances of 1001 in the file to 2319. I then went into
 default.xml  in directory folder and in one of the groups  just
 mimicked
 1001 details by changing 1001 to 2319.

 Connecting  to FS gives Forbidden message. However 1001 connects
 without
 a problem.  What have I missed ?

 Is there a place that just puts things in do this and that and that to
 create a new user ?


 Did you execute reloadxml from the fs cli before trying to connect 
 with 2319? Also I'm assuming that 2319.xm is a typo and you actually 
 created 2319.xml in the default/directory subdir.
 -MC


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Re: [Freeswitch-users] Cannot Do this Basic thing

Hi

Sorry .xm is a typo. I actually shut down the server and restarted. The 
log says I need to create a domain of aaa.bbb.ccc.ddd (which is the 
server IP address ) and then put the user in that domain.  Isn't the 
default domain that of the server FS is running on ?
2319.xml is in /usr/local/freeswitch/conf/directory/default/

Thanks for your time



Michael Collins wrote:


 On Thu, Dec 3, 2009 at 9:46 AM, Samuel Abekah-Mensah 
 ab...@greatiam.com mailto:ab...@greatiam.com wrote:

 I have copied 1001.xml in directory/default to a test user 2319.xm
 changing or instances of 1001 in the file to 2319. I then went into
 default.xml  in directory folder and in one of the groups  just
 mimicked
 1001 details by changing 1001 to 2319.

 Connecting  to FS gives Forbidden message. However 1001 connects
 without
 a problem.  What have I missed ?

 Is there a place that just puts things in do this and that and that to
 create a new user ?


 Did you execute reloadxml from the fs cli before trying to connect 
 with 2319? Also I'm assuming that 2319.xm is a typo and you actually 
 created 2319.xml in the default/directory subdir.
 -MC


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Re: [Freeswitch-users] Choppy sound with PCMU


You mean, upgrading to the trunk and disabling RTP timers? Yes, I did. I
thought I responded back. Perhaps it didn't make through though, as I just
emailed back to the list instead of using nabble.com...

Anyway, upgrading to the trunk didn't change much, forcing SPA to 30ms went
w/o any effect either, but disabling RTP timers did the trick. I don't have
the original choppy sound with PCMU problem any more, thanks a lot for the
quick turnaround on that question.

But your suggestions made me look, into logs, strace, code, etc, so now I'm
just checking on how to quiet down those busy loops a little and how to get
rid of periodic CRIT messages about Virtual Machine Migration.


Anthony Minessale-2 wrote:
 
 What about the things I spent time suggesting in my last email?
 Did you try them because I was actually curious if they made any impact.
 
 
 On Thu, Dec 3, 2009 at 11:29 AM, eaf erandr-j...@usa.net wrote:
 

 I'm sorry if I sounded that way. Did mean to. :)

 Yes, it's an embedded platform. It's an ALIX board with AMD Geode LX800
 chip
 and 256MB of RAM. http://www.pcengines.ch/alix2d3.htm

 Line offset difference is due to some minor logging changes I made to see
 who's allocating timers and how often. This way I found MOH streaming and
 that RTP still allocates timers even when it's set to none in the
 profile.

 I feel that this platform turned out to be underpowered for FS because it
 cannot meet its scheduling expectations. I guess, some degree of kernel
 tweaking or setting priorities will fix that. Meanwhile I just got rid of
 the SQLDB 1ms thread via -nosql command line option, split sofia worker
 1ms
 thread in two (one blocked and waiting for new commands in the SQL queue,
 the other one checking registrations and gateways with 1sec interval),
 and
 don't know yet what to do about the timer thread.

 Again, I apologize for stupid or accusing questions, I'm just trying to
 see
 how FS can be made friendlier to this board. Or the board be made
 friendlier
 to FS ;)


 Anthony Minessale-2 wrote:
 
  If you see that message then your machine/os/combo is having some
 problems
  keeping up.
  It's not the timer missing anything its the monotonic clock detecting a
 1
  second or more differential from what its next prediction for the time
  should be.  The best way to trigger this would be to suspend FS with
  control-z or attach to it with gdb blocking the entire process,  that
 1ms
  thread would have to miss 1000 iterations to trigger that warning.
 
  Btw, that error message is at line 471 not 473 so you are using
 modified
  code.
 
  Its possible your box has a bad monotonic timer, you can set
 
 
 
  under settings in switch.conf.xml
 
  We are now starting to guess you are using some small embedded type
  platform
  perhaps?
  I've run FS even on a nokia n810 and never caused that message to fire.
 
  if 1 call can interrupt the cpu enough to  cause noticeable issues you
  might
  want to consider running the process at a
  greater priority by using the -hp command line arg or at least nice it
 
  Why don't you tell us the whole story about what OS/platform you are
 using
  here rather that form conjectures about what is wrong with our code
 that
  thousands of people are happy with.
 
 
 
 
 
 
 
  On Thu, Dec 3, 2009 at 8:55 AM, eaf erandr-j...@usa.net wrote:
 
 
  Btw, I have these popping up in my logs from time to time:
 
  2009-12-03 09:42:06.035294 [DEBUG] switch_core_state_machine.c:314
  (sofia/external/xx...@4.68.250.148) Running State Change CS_HANGUP
  2009-12-03 09:42:06.035294 [CRIT] switch_time.c:473 Virtual Migration
  Detected! Syncing Clock
 
  In this case an incoming call rang to both FS and Asterisk, Asterisk
  picked
  up, but the surge of activity made FS timer thread miss a beat or two.
 
 
  eaf wrote:
  
   Oh, it's not just one timer thread... Why, why is sql_thread keeps
 on
   checking for messages every millisecond? Couldn't there be some
  signalling
   implemented that will make the thread suspend on condition variable
 or
  a
   socket/pipe in between?
  
   #0  do_sleep (t=1000) at src/switch_time.c:109
   #1  0xb7e5e215 in switch_core_sql_thread (thread=0xb7586ae8,
 obj=0x0)
  at
   src/switch_core_sqldb.c:783
  
   Why does this sofia_profile_worker_thread keeps on looping checking
 for
   the queue? Have a semaphore!
  
   #0  do_sleep (t=1000) at src/switch_time.c:109
   #1  0xb73a4701 in sofia_profile_worker_thread_run (thread=0x80f3a30,
   obj=0x80f2490) at sofia.c:978
  
   Nothing's happening on the box, but there are three threads that
  pretend
   to be actively busy with smth. Others at least sleep for hundreds of
   milliseconds, not for one.
  
   And there is even infrastructure present to do blocking pops: i.e.
 why
   couldn't sqldb thread do queue_pop() instead of queue_trypop()
  intermixed
   with 1ms sleeps? This looping is such a waste...
  
  
   eaf wrote:
  
   As I see it, switch_cond_next() currently is just a do_sleep(1000).

Re: [Freeswitch-users] Dialplan behavior

2009-12-03 Thread Ghulam Mustafa

other than configuration/syntax problem it could be a simple character/file
encoding problem or may be improper file
permissions!



On Thu, Dec 3, 2009 at 11:29 PM, David Laperle dlape...@rsslex.com wrote:

  Hi guys,

 i have a weird problem with my dialplans. For the moment, i have only 2
 «usable» extensions. They were working #1 yesterday, but this morning i
 realize i forgot to compile mod_python, so i go back into my source folder
 and modify the modules.conf to uncomment mod_python, did a make and make
 install (i did a backup of my conf folder before)! The make and make install
 worked flawlessly. Then i put back my bkp of conf directory.

 I restarted the freeswitch service, created my python test dialplan and
 entered into cli to see what's gonna happen! To my surprise, the call didn't
 processed to the extension i was dialing.

 i tried all the other extensions i had, they were all not working

 After that i realized that the .xml in freeswitch/dialplan/default/ weren't
 imported into configuration at startup ...

 I have read all the documentation about difference between public and
 default dialplan and i understand them correctly, in public if i include all
 default folder, it's working again (i can reach all my extensions in
 default.

 My extensions are in the correct user_context ... i did nothing since
 yesterday other than a make  make install after enabling python ...

 Any other user have an idea why the default/*.xml aren't processed
 automatically? What could i have done wrong so they are no longer processed?

 Thanks a lot,

 *David Laperle *
 Administrateur réseau / Network administrator
 (514) 393-7647
 *dlape...@rsslex.com*

 *Robinson Sheppard Shapiro *s.e.n.c.r.l/LLP
 Avocats / Barristers  Solicitors
 4600 - 800 Place Victoria
 Montréal Qc H4Z 1H6
 T (514) 878-2631 F (514) 878-1865
 www.rsslex.com et/and www.rsscanadaimmigration.com




   *
 --
 **http://www.rsslex.com** *

 *AVIS:* Ce courriel privilégié et confidentiel est destiné à la seule
 personne ou entité à laquelle il est adressé. Pour toute autre personne,
 toute action prise en rapport à ce courriel ainsi que toute lecture,
 reproduction, transmission et/ou divulgation d'une partie ou de l'ensemble
 de celui-ci est interdite. Si vous n'êtes pas la personne autorisée à
 recevoir ce courriel, S.V.P. le retourner à l'expéditeur et le détruire.
 Bien que ce courriel ait été traité contre les virus, il est de la
 responsabilité du destinataire de s'assurer que l'envoi en est exempt. Nos
 communications avec vous peuvent contenir des renseignements confidentiels
 ou protégés par le secret professionnel. Si vous désirez que nous
 communiquions avec vous par un autre moyen de transmission que le courrier
 électronique ordinaire non sécurisé, veuillez nous en aviser.

 *NOTICE:* This privileged and confidential email is intended only for the
 individual or entity to whom it is addressed. With regard to all others, any
 action related with this email as well as any reading, reproduction,
 transmission and/or dissemination in whole or in part of the information
 included in this email is prohibited. If you are not the addressee,
 immediately return the email to sender prior to destroying all copies. Even
 if this email is believed to be free from any virus, it is the
 responsibility of the recipient to make sure that it is virus exempt. Our
 communications to you may contain confidential information or information
 protected under solicitor-client privilege. Please advise if you wish us to
 use a mode of communication other than regular, unsecured e-mail in our
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-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyren...@ekiga.net
mail: mustafa...@gmail.com
web: cyrenity.wordpress.com
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Re: [Freeswitch-users] can't register Inphonex

2009-12-03 Thread bakko

From de console:

sofia profile external siptrace on

or

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Re: [Freeswitch-users] Cisco IOS gateway: command to send connected line name

2009-12-03 Thread Yehavi Bourvine

Unfortunately this didn't help... Incoming calls from ISDN to SIP sends back
to ISDN the name of the destination, but not the other way around...

Thanks! __Yehavi:

2009/12/3 Metik freeswitch-users-l...@metik.com

 Yehavi,

 There are a few variations of transmitting this information... If you
 have already enabled a supplemental isdn service profile, try adding the
 following to the PRI you are using:

 (config-if)#isdn outgoing ie facility
 (config-if)#iisdn outgoing ie extended-facility
 (config-if)#isdn outgoing display-ie
 (config-if)#isdn outgoing ie caller-number
 (config-if)#isdn outgoing ie called-number

 -metik

 Yehavi Bourvine wrote:
  Hello,
 
We have a Cisco running IOS 12.4T used as our SIP-PRI gateway. On
  the PRI there is a Nortel with Q.Sig. After a lot of configuration
  trials I've managed to set it to send back the connected name over the
  SIP (i.e. when a call goes from SIP to PRI, the PRI sends back the
  connected name and then the Cisco adds it as a Remote-Party-ID).
  However, I did not save it and a power outage cleared this config. In
  my age I don't remember what I've done...
 
  Anyone knows the correct config?
 
 Thanks! __Yehavi:
  
 
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Re: [Freeswitch-users] change the remote RTP port after sample rate doesnot match

2009-12-03 Thread Erwin Davis

Hi, Anthony and Mike,

With the latest version from SVN, I was able to remove the warning sample
rate not matching. But the remote RTP port was still changed after after
playing the vm greeting. See below,
2009-12-03 13:44:46.901216 [INFO] switch_rtp.c:1975 Auto Changing port from
XXX.YYY.ZZZ.39:10002 to XXX.YYY.ZZZ.39:3335

Any clue?

I looked at the source code in switch_rtp.c:1975, it shows that if
rtp_session-autoadj_tally = 10, then a rtp port change will happen. Any
idea about autoadj_tally and what cause the increase of autoadj_tally ?
Thanks,




On 12/2/09, Erwin Davis davis.er...@gmail.com wrote:

 Hi, Anthony and Mike,

 Thanks for your reply. The problem still exists even after I ran make
 hd-sounds install.
 I will try the latest version from the SVN to see if the problem will go
 away. I will let you know.
 Thanks folks,

 Regards,

 On 12/2/09, Michael Collins m...@freeswitch.org wrote:



 On Wed, Dec 2, 2009 at 12:08 PM, Erwin Davis davis.er...@gmail.comwrote:

 Hi, Anthony,

 Thanks for your reply.

 When I type the command below, I got the error,
 Unknown target hd-sound-install
 make[1]: *** [hd-sound-install] Error 1
 make: *** [hd-sound-install] Error 2

 I found out that under
 /usr/local/freeswitch/sounds/en/us/callie/voicemail, there are directories,
 8000, 16000, 32000, 48000 for recorded voicemail greetings. It should
 explain why at first FS played in right sample rate. But after playing
 serveral time, FS complained about sample rate not matching.  Any clue?
 Thanks,


 Erwin,

 As Tony said you've actually got a pretty old installation. If this is in
 production then I would recommend getting a sandbox machine, install trunk
 using the quick-and-dirty install, and then update the default config to you
 specific configuration. Test to make sure it works before you put it into
 production. :)

 Feel free to join us on IRC (#freeswitch on irc.freenode.net) if you run
 into any issues that require more real-time conversation.
 -MC


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Re: [Freeswitch-users] Cannot Do this Basic thing

2009-12-03 Thread Michael Collins

On Thu, Dec 3, 2009 at 10:34 AM, Samuel Abekah-Mensah ab...@greatiam.comwrote:

 Hi

 Sorry .xm is a typo. I actually shut down the server and restarted. The
 log says I need to create a domain of aaa.bbb.ccc.ddd (which is the
 server IP address ) and then put the user in that domain.  Isn't the
 default domain that of the server FS is running on ?
 2319.xml is in /usr/local/freeswitch/conf/directory/default/

 Thanks for your time


Okay, here's exactly what I did:
cd /usr/local/freeswitch/conf/directory/default
cp 1001.xml 2319.xml
perl -pi -e 's/1001/2319/g' 2319.xml
cat 2319.xml

include
  user id=2319
params
  param name=password value=$${default_password}/
  param name=vm-password value=2319/
/params
variables
  variable name=toll_allow value=domestic,international,local/
  variable name=accountcode value=2319/
  variable name=user_context value=default/
  variable name=effective_caller_id_name value=Extension 2319/
  variable name=effective_caller_id_number value=2319/
  variable name=outbound_caller_id_name
value=$${outbound_caller_name}/
  variable name=outbound_caller_id_number
value=$${outbound_caller_id}/
  variable name=callgroup value=techsupport/
/variables
  /user
/include

Then I logged into fs_cli, pressed F6 (which does reloadxml) and then I
set up my x-lite:
Display Name: Test
User name: 2319
Password: 1234
Authorization user name: 2319
Domain: 10.15.0.91

It registered just fine as can be seen by the output of sofia status
profile internal:
snip
Call-ID:MzRiOGI4NTA2YjA0ZTkzMDYwZjA3MTlkZGQ3ZjNhMjg.
User:   2...@10.15.0.91
Contact:Test sip:2...@10.15.0.124:41680
;rinstance=09c51f8aa23d6738
Agent:  X-Lite release 1014k stamp 47051
Status: Registered(UDP)(unknown) EXP(2009-12-03 13:41:38)
Host:   freeswitch1.yt
IP: 10.15.0.124
Port:   41680
Auth-User:  2319
Auth-Realm: 10.15.0.91
MWI-Account:2...@10.15.0.91

So, most likely you've got an issue with the XML file itself or the
configuration on your SIP device. Double check the username and auth
username values. If need be delete your 2319.xml file and start over.

-MC
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Re: [Freeswitch-users] Dialplan behavior

2009-12-03 Thread Michael Collins

On Thu, Dec 3, 2009 at 10:29 AM, David Laperle dlape...@rsslex.com wrote:

  Hi guys,

 i have a weird problem with my dialplans. For the moment, i have only 2
 «usable» extensions. They were working #1 yesterday, but this morning i
 realize i forgot to compile mod_python, so i go back into my source folder
 and modify the modules.conf to uncomment mod_python, did a make and make
 install (i did a backup of my conf folder before)! The make and make install
 worked flawlessly. Then i put back my bkp of conf directory.

 I restarted the freeswitch service, created my python test dialplan and
 entered into cli to see what's gonna happen! To my surprise, the call didn't
 processed to the extension i was dialing.

 i tried all the other extensions i had, they were all not working

 After that i realized that the .xml in freeswitch/dialplan/default/ weren't
 imported into configuration at startup ...

 I have read all the documentation about difference between public and
 default dialplan and i understand them correctly, in public if i include all
 default folder, it's working again (i can reach all my extensions in
 default.

 My extensions are in the correct user_context ... i did nothing since
 yesterday other than a make  make install after enabling python ...

 Any other user have an idea why the default/*.xml aren't processed
 automatically? What could i have done wrong so they are no longer processed?


double-check for the existence of conf/dialplan/default.xml - I've seen on
rare occasion where that file simple goes away for no apparent reason. Since
I never change that file - and I recommend that you never change it either
;) - you can go to your FS source directory and issue make samples and it
will re-create any missing default config files without overwriting you
existing config files.
-MC
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[Freeswitch-users] Lua and database access to core_db

2009-12-03 Thread Jon Bruel

I am trying to rewrite all my javascript scripts into Lua scripts. I have run 
into the problem of core_db access. This can be achieved with Spidermonkey, but 
apparently not with Lua. I have tried to get the binary for Lua (using apt-get) 
but I get an error when I require the sqlite.so: undefined symbol: 
luaopen_luasql_sqlite, so I'm stuck. So what is a feasible way to manipulate 
the core database from Lua?
I may mention that access to MySQL works perfectly from Lua.
Regards Jon

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Re: [Freeswitch-users] Lua and database access to core_db

In latest trunk you can run the core db in your same mysql db.
other than that we would need to create an object from our lua module
similar to how it was done in js.


On Thu, Dec 3, 2009 at 2:05 PM, Jon Bruel j...@consiglia.dk wrote:

  I am trying to rewrite all my javascript scripts into Lua scripts. I have
 run into the problem of core_db access. This can be achieved with
 Spidermonkey, but apparently not with Lua. I have tried to get the binary
 for Lua (using apt-get) but I get an error when I require the sqlite.so:
 undefined symbol: luaopen_luasql_sqlite, so I’m stuck. So what is a feasible
 way to manipulate the core database from Lua?

 I may mention that access to MySQL works perfectly from Lua.

 Regards Jon



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Twitter: http://twitter.com/FreeSWITCH_wire

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MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
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Re: [Freeswitch-users] HA questions.

2009-12-03 Thread Tim Uckun

On Fri, Dec 4, 2009 at 4:59 AM, Michael Jerris m...@jerris.com wrote:
 The easiest place to do this is at the point you send the calls to 
 FreeSWITCH.  How are the calls coming in?


From an as of now unkown SIP trunk provider (we are still in
negotiations with a couple of companies).

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Re: [Freeswitch-users] change the remote RTP port after sample rate doesnot match

2009-12-03 Thread Erwin Davis

Hi, I solved this issue. the reason is because of the different port number
between the the one in SDP and the one in real RTP stream. This is very nice
feature.

e

On 12/2/09, Erwin Davis davis.er...@gmail.com wrote:

 Hi, I got a weird issue when I dialed an extension and listen to a recorded
 voice mail greeting message.
 After playing a couple of time of the greeting, the FS printed the warning
 of sample rate not matching, then
 send the audio to a different remote RTP port. See the log below,


 2009-12-02 12:42:27.109529 [DEBUG] switch_ivr_play_say.c:1097 Codec
 Activated l...@16000hz 1 channels 20ms
 2009-12-02 12:42:27.112038 [DEBUG] switch_core_io.c:649
 sofia/internal/1...@xxx.yyy.zzz.31 receive message [TRANSCODING_NECESSARY]
 2009-12-02 12:42:28.86119 [DEBUG] switch_ivr_play_say.c:1391 done playing
 file
 2009-12-02 12:42:28.205975 [DEBUG] switch_ivr_play_say.c:118 No language
 specified - Using [en]
 2009-12-02 12:42:28.220967 [DEBUG] switch_ivr_play_say.c:273 Handle
 play-file:[voicemail/vm-record_message.wav] (en:en)
 2009-12-02 12:42:28.222971 [DEBUG] switch_ivr_play_say.c:1097 Codec
 Activated l...@16000hz 1 channels 20ms
 2009-12-02 12:42:28.222971 [DEBUG] switch_core_io.c:649
 sofia/internal/1...@xxx.yyy.zzz.31 receive message [TRANSCODING_NECESSARY]
 2009-12-02 12:42:32.804858 [DEBUG] switch_ivr_play_say.c:1391 done playing
 file
 2009-12-02 12:42:32.944554 [DEBUG] switch_core_io.c:649
 sofia/internal/1...@xxx.yyy.zzz.31 receive message [TRANSCODING_NECESSARY]
 2009-12-02 12:42:33.946969 [WARNING] switch_core_file.c:120 Sample rate
 doesn't match
 2009-12-02 12:42:33.950231 [DEBUG] switch_ivr_play_say.c:549 Raw Codec
 Activated
 2009-12-02 12:42:34.124584 [INFO] switch_rtp.c:1869 Auto Changing port from
 xxx.yyy.zzz.39:10002 to xxx.yyy.zzz.39:1748
 2009-12-02 12:42:34.944913 [DEBUG] switch_core_codec.c:122 Restore original
 codec.
 2009-12-02 12:42:34.947918 [DEBUG] mod_voicemail.c:1162 Message is less
 than minimum record length: 3, discarding it.
 2009-12-02 12:42:34.947918 [DEBUG] switch_ivr_play_say.c:118 No language
 specified - Using [en]
 2009-12-02 12:42:34.960002 [DEBUG] switch_ivr_play_say.c:273 Handle
 play-file:[voicemail/vm-too-small.wav] (en:en)
 2009-12-02 12:42:34.960850 [DEBUG] switch_ivr_play_say.c:1097 Codec
 Activated l...@16000hz 1 channels 20ms
 2009-12-02 12:42:34.960850 [DEBUG] switch_core_io.c:649
 sofia/internal/1...@xxx.yyy.zzz.31 receive message [


 the original codec is wideband 16kHz Speex and the wireshark shows that the
 FS used the same codec. I used FS 1.04 in fedora 8.
 I have two questions here,
 (1) why does FS report Sample rate doesn't match? is it a bug or
 configuration issue?
 (2) Why does FS change the RTP port ? how to fix it?

 Thanks,

 Regards,


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Re: [Freeswitch-users] HA questions.

2009-12-03 Thread Tim Uckun

On Fri, Dec 4, 2009 at 5:56 AM, Adam Ford li...@redbonez.net wrote:
 Have you checked out Redfone? While I haven't attempted to implement it yet,
 my Redfone foneBridge2 claims to be able to handle load balancing and
 failover between two Asterisk/Freeswitch servers.



That would be my choice for incoming E1 lines. Right now I am looking
for a SIP solution.

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Re: [Freeswitch-users] Choppy sound with PCMU

no,

I mean the one after that that you must have completely skipped with a
command line option to try and a param to set in the config. It somewhat
annoys me for taking the time to compose it now.  I wrote all of the code
you are talking about myself and I was trying to give you some
suggestions

Well, actually,  you did answer my question about the platform so you must
have seen it.

The loops are not the cause of that migration message, something wrong with
the hardware or the kernel is.
Another guy just told you he does not see that problem on the same exact
hardware.

Even if you have a point about the sql threads, you could make a patch to
slow them down but you cant slow down too much or you will not be able to
handle 400 cps all asking to send updates to transactions in batches of
thousands of sql stmts.  Every line of that code is carefully designed so I
don't know what else to tell you but to stop being so arrogant and re-read
this thread for all the advice you have totally ignored.  I started out
trying to help you but I have a lot of work to do.  I thoroughly explained
it to you and you are choosing to ignore me so I guess I'm done.
You can do whatever you want with your working copy, i'll see you in 3 or 4
years when you get up to speed with the rest of us






On Thu, Dec 3, 2009 at 12:43 PM, eaf erandr-j...@usa.net wrote:


 You mean, upgrading to the trunk and disabling RTP timers? Yes, I did. I
 thought I responded back. Perhaps it didn't make through though, as I just
 emailed back to the list instead of using nabble.com...

 Anyway, upgrading to the trunk didn't change much, forcing SPA to 30ms went
 w/o any effect either, but disabling RTP timers did the trick. I don't have
 the original choppy sound with PCMU problem any more, thanks a lot for
 the
 quick turnaround on that question.

 But your suggestions made me look, into logs, strace, code, etc, so now I'm
 just checking on how to quiet down those busy loops a little and how to get
 rid of periodic CRIT messages about Virtual Machine Migration.


 Anthony Minessale-2 wrote:
 
  What about the things I spent time suggesting in my last email?
  Did you try them because I was actually curious if they made any impact.
 
 
  On Thu, Dec 3, 2009 at 11:29 AM, eaf erandr-j...@usa.net wrote:
 
 
  I'm sorry if I sounded that way. Did mean to. :)
 
  Yes, it's an embedded platform. It's an ALIX board with AMD Geode LX800
  chip
  and 256MB of RAM. http://www.pcengines.ch/alix2d3.htm
 
  Line offset difference is due to some minor logging changes I made to
 see
  who's allocating timers and how often. This way I found MOH streaming
 and
  that RTP still allocates timers even when it's set to none in the
  profile.
 
  I feel that this platform turned out to be underpowered for FS because
 it
  cannot meet its scheduling expectations. I guess, some degree of kernel
  tweaking or setting priorities will fix that. Meanwhile I just got rid
 of
  the SQLDB 1ms thread via -nosql command line option, split sofia worker
  1ms
  thread in two (one blocked and waiting for new commands in the SQL
 queue,
  the other one checking registrations and gateways with 1sec interval),
  and
  don't know yet what to do about the timer thread.
 
  Again, I apologize for stupid or accusing questions, I'm just trying to
  see
  how FS can be made friendlier to this board. Or the board be made
  friendlier
  to FS ;)
 
 
  Anthony Minessale-2 wrote:
  
   If you see that message then your machine/os/combo is having some
  problems
   keeping up.
   It's not the timer missing anything its the monotonic clock detecting
 a
  1
   second or more differential from what its next prediction for the time
   should be.  The best way to trigger this would be to suspend FS with
   control-z or attach to it with gdb blocking the entire process,  that
  1ms
   thread would have to miss 1000 iterations to trigger that warning.
  
   Btw, that error message is at line 471 not 473 so you are using
  modified
   code.
  
   Its possible your box has a bad monotonic timer, you can set
  
  
  
   under settings in switch.conf.xml
  
   We are now starting to guess you are using some small embedded type
   platform
   perhaps?
   I've run FS even on a nokia n810 and never caused that message to
 fire.
  
   if 1 call can interrupt the cpu enough to  cause noticeable issues you
   might
   want to consider running the process at a
   greater priority by using the -hp command line arg or at least nice it
  
   Why don't you tell us the whole story about what OS/platform you are
  using
   here rather that form conjectures about what is wrong with our code
  that
   thousands of people are happy with.
  
  
  
  
  
  
  
   On Thu, Dec 3, 2009 at 8:55 AM, eaf erandr-j...@usa.net wrote:
  
  
   Btw, I have these popping up in my logs from time to time:
  
   2009-12-03 09:42:06.035294 [DEBUG] switch_core_state_machine.c:314
   (sofia/external/xx...@4.68.250.148) Running

Re: [Freeswitch-users] Choppy sound with PCMU


Oh, you mean giving FS higher priority? Yeah, as a last resort I'll do that.
At the moment, I hope it won't be necessary as I can make those hyper
threads behave, and will see how that goes first. I see where your
implementation could be coming from. There is a queue of SQL queries in
sofia.c processed by the worker thread. There are only two pop functions
available in APR: queue_pop() and queue_trypop(), so alas no option with a
timeout here. You don't want to block the thread in pop() indefinitely
because you chose that same worker needs to do ireg and gw processing once
in a while (separated by tens or hundreds of seconds, btw). You also want to
be able to detect shutdown condition so that the worker doesn't hold up
profile thread. So you chose to poll for events every millisecond instead of
just creating an apr_thread_cond_t for resource friendly signalling.

I agree that the timer thread philosophy is great and was the right choice
for scaling, but I just don't comprehend responses to things like these
other SQL or sofia worker threads. Did somebody even remotely acknowledge
that busy loops at least in those areas that I showed may probably be a bad
idea and could've been eliminated? I've heard suggestions to bump up
priority, I've heard that the code was perfect already, that it's the result
of 4-year effort, that I am arrogant, don't listen and don't understand
squat.

I'm sorry if I gave you impression that I was looking for the bad parts in
the software. I apologized for that already. All I wanted was to have
constructive conversation, perhaps I'm not too good at it. Code is already
perfect according to you? Fine with me.


Anthony Minessale-2 wrote:
 
 no,
 
 I mean the one after that that you must have completely skipped with a
 command line option to try and a param to set in the config. It somewhat
 annoys me for taking the time to compose it now.  I wrote all of the code
 you are talking about myself and I was trying to give you some
 suggestions
 
 Well, actually,  you did answer my question about the platform so you must
 have seen it.
 
 The loops are not the cause of that migration message, something wrong
 with
 the hardware or the kernel is.
 Another guy just told you he does not see that problem on the same exact
 hardware.
 
 Even if you have a point about the sql threads, you could make a patch to
 slow them down but you cant slow down too much or you will not be able to
 handle 400 cps all asking to send updates to transactions in batches of
 thousands of sql stmts.  Every line of that code is carefully designed so
 I
 don't know what else to tell you but to stop being so arrogant and re-read
 this thread for all the advice you have totally ignored.  I started out
 trying to help you but I have a lot of work to do.  I thoroughly explained
 it to you and you are choosing to ignore me so I guess I'm done.
 You can do whatever you want with your working copy, i'll see you in 3 or
 4
 years when you get up to speed with the rest of us
 
 

-- 
View this message in context: 
http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26633739.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


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Re: [Freeswitch-users] ATA that supports TLS/SRTP w FS

Hi All,

I managed to borrow a SPA3102 with the latest firmware and have got it
to register using TLS, but I am still struggling with SRTP.  Has
anyone managed to get SRTP working with the Linksys devices and if so,
can they direct me on how to do this.

I have generated a mini-certificates and SRTP Private Key using the
gen-mc tool found at
http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3.
 However, when ever I initiate a call from the SPA, I can see that the
call is not encrypted.

Help appreciated.

Thanks!


On Sat, Nov 28, 2009 at 6:31 AM, eman e...@chabotel.com wrote:
 Check out the Linksys SPA2102

 On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith
 mcampbellsm...@gmail.com wrote:

 The only ATA mentioned on the WIKI that supports TLS/SRTP is the
 Grandstream HandyTone 503.  But, again according to the wiki, that
 doesn't seem to behave to well with TLS ...

 On Wed, Nov 25, 2009 at 7:14 PM, Jason White ja...@jasonjgw.net wrote:
  Mark Campbell-Smith mcampbellsm...@gmail.com wrote:
  Does the SPA3102 support TLS or only SRTP?
 
  I don't know, but supporting only SRTP would be ridiculous, since the
  keys
  would then be transmitted in the clear and therefore amenable to
  interception.
  SRTP requires the SIP channel to be encrypted by TLS in order to be
  secure.
  ZRTP, on the other hand, doesn't have this limitation: it works entirely
  in
  RTP.
 
  I would be rather surprised were a hardware manufacturer to implement
  SRTP
  without TLS for the SIP traffic. On the other hand, we've seen often in
  this
  forum that some manufacturers are really clueless...
 
 
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Re: [Freeswitch-users] ATA that supports TLS/SRTP w FS

2009-12-03 Thread Gabriel Kuri

AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key
exchange to appropriately support SRTP and FreeSWITCH. They do their
proprietary Sipura key exchange only, not sure if Cisco plans on
upgrading the firmware to ever support SDES on the ATAs. They added
support for SDES to their IP Phones about 1 year ago, but nothing has
happened with the ATAs as of yet.

Gabe


On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
 Hi All,

 I managed to borrow a SPA3102 with the latest firmware and have got it
 to register using TLS, but I am still struggling with SRTP.  Has
 anyone managed to get SRTP working with the Linksys devices and if so,
 can they direct me on how to do this.

 I have generated a mini-certificates and SRTP Private Key using the
 gen-mc tool found at
 http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3.
  However, when ever I initiate a call from the SPA, I can see that the
 call is not encrypted.

 Help appreciated.

 Thanks!


 On Sat, Nov 28, 2009 at 6:31 AM, eman e...@chabotel.com wrote:
 Check out the Linksys SPA2102

 On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith
 mcampbellsm...@gmail.com wrote:

 The only ATA mentioned on the WIKI that supports TLS/SRTP is the
 Grandstream HandyTone 503.  But, again according to the wiki, that
 doesn't seem to behave to well with TLS ...

 On Wed, Nov 25, 2009 at 7:14 PM, Jason White ja...@jasonjgw.net wrote:
  Mark Campbell-Smith mcampbellsm...@gmail.com wrote:
  Does the SPA3102 support TLS or only SRTP?
 
  I don't know, but supporting only SRTP would be ridiculous, since the
  keys
  would then be transmitted in the clear and therefore amenable to
  interception.
  SRTP requires the SIP channel to be encrypted by TLS in order to be
  secure.
  ZRTP, on the other hand, doesn't have this limitation: it works entirely
  in
  RTP.
 
  I would be rather surprised were a hardware manufacturer to implement
  SRTP
  without TLS for the SIP traffic. On the other hand, we've seen often in
  this
  forum that some manufacturers are really clueless...
 
 
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Re: [Freeswitch-users] ATA that supports TLS/SRTP w FS

2009-12-03 Thread Itamar Reis Peixoto

you can try xlite too.



On Thu, Dec 3, 2009 at 8:05 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
 Hi All,

 I managed to borrow a SPA3102 with the latest firmware and have got it
 to register using TLS, but I am still struggling with SRTP.  Has
 anyone managed to get SRTP working with the Linksys devices and if so,
 can they direct me on how to do this.

 I have generated a mini-certificates and SRTP Private Key using the
 gen-mc tool found at
 http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3.
  However, when ever I initiate a call from the SPA, I can see that the
 call is not encrypted.

 Help appreciated.

 Thanks!




Itamar Reis Peixoto

e-mail/msn/google talk/sip: ita...@ispbrasil.com.br
skype: itamarjp
icq: 81053601
+55 11 4063 5033
+55 34 3221 8599

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Re: [Freeswitch-users] ATA that supports TLS/SRTP w FS

Quote: Cisco/Linksys SPA series ATAs do not support SDES key exchange
to appropriately support SRTP and FreeSWITCH

I'll check with Cisco regarding their implementation then and try to
find out when/if they will support standard SRTP encryption.


So, back to my origianal question then.  Are there any ATA's that
support TLS AND SRTP with FreeSwitch?


On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri gk...@ieee.org wrote:
 AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key
 exchange to appropriately support SRTP and FreeSWITCH. They do their
 proprietary Sipura key exchange only, not sure if Cisco plans on
 upgrading the firmware to ever support SDES on the ATAs. They added
 support for SDES to their IP Phones about 1 year ago, but nothing has
 happened with the ATAs as of yet.

 Gabe


 On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith
 mcampbellsm...@gmail.com wrote:
 Hi All,

 I managed to borrow a SPA3102 with the latest firmware and have got it
 to register using TLS, but I am still struggling with SRTP.  Has
 anyone managed to get SRTP working with the Linksys devices and if so,
 can they direct me on how to do this.

 I have generated a mini-certificates and SRTP Private Key using the
 gen-mc tool found at
 http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3.
  However, when ever I initiate a call from the SPA, I can see that the
 call is not encrypted.

 Help appreciated.

 Thanks!


 On Sat, Nov 28, 2009 at 6:31 AM, eman e...@chabotel.com wrote:
 Check out the Linksys SPA2102

 On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith
 mcampbellsm...@gmail.com wrote:

 The only ATA mentioned on the WIKI that supports TLS/SRTP is the
 Grandstream HandyTone 503.  But, again according to the wiki, that
 doesn't seem to behave to well with TLS ...

 On Wed, Nov 25, 2009 at 7:14 PM, Jason White ja...@jasonjgw.net wrote:
  Mark Campbell-Smith mcampbellsm...@gmail.com wrote:
  Does the SPA3102 support TLS or only SRTP?
 
  I don't know, but supporting only SRTP would be ridiculous, since the
  keys
  would then be transmitted in the clear and therefore amenable to
  interception.
  SRTP requires the SIP channel to be encrypted by TLS in order to be
  secure.
  ZRTP, on the other hand, doesn't have this limitation: it works entirely
  in
  RTP.
 
  I would be rather surprised were a hardware manufacturer to implement
  SRTP
  without TLS for the SIP traffic. On the other hand, we've seen often in
  this
  forum that some manufacturers are really clueless...
 
 
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[Freeswitch-users] Generate cdrs

2009-12-03 Thread Mouncif Benniane

is it possible to run a javascript at the end of dialplan to generate cdrs?
because (mod_cdr_csv) is giving me hard time as it rotates Master file on
machine reboots or shutdown signals.
javascript or LUA for preferences?

thank you
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Re: [Freeswitch-users] ATA that supports TLS/SRTP w FS

2009-12-03 Thread Gabriel Kuri

The ATAs I'm aware that claim support for TLS and SRTP w/ SDES are the
Grandstream and Mediatrix devices (although I've never tried either
one with FreeSWITCH).

I've personally never had any good experience with the Grandstream
ATAs. The Mediatrix ATAs are OK devices, but I've never personally
tested them with SRTP w/SDES and FreeSWITCH, but supposedly they
support it (so says their marketing material and docs).

I'd see if Cisco has any plans to add support for it to the ATAs. Next
time I see our Cisco SE, I'll try to poke him about it.

Gabe

On Thu, Dec 3, 2009 at 2:34 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
 Quote: Cisco/Linksys SPA series ATAs do not support SDES key exchange
 to appropriately support SRTP and FreeSWITCH

 I'll check with Cisco regarding their implementation then and try to
 find out when/if they will support standard SRTP encryption.


 So, back to my origianal question then.  Are there any ATA's that
 support TLS AND SRTP with FreeSwitch?


 On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri gk...@ieee.org wrote:
 AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key
 exchange to appropriately support SRTP and FreeSWITCH. They do their
 proprietary Sipura key exchange only, not sure if Cisco plans on
 upgrading the firmware to ever support SDES on the ATAs. They added
 support for SDES to their IP Phones about 1 year ago, but nothing has
 happened with the ATAs as of yet.

 Gabe


 On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith
 mcampbellsm...@gmail.com wrote:
 Hi All,

 I managed to borrow a SPA3102 with the latest firmware and have got it
 to register using TLS, but I am still struggling with SRTP.  Has
 anyone managed to get SRTP working with the Linksys devices and if so,
 can they direct me on how to do this.

 I have generated a mini-certificates and SRTP Private Key using the
 gen-mc tool found at
 http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3.
  However, when ever I initiate a call from the SPA, I can see that the
 call is not encrypted.

 Help appreciated.

 Thanks!


 On Sat, Nov 28, 2009 at 6:31 AM, eman e...@chabotel.com wrote:
 Check out the Linksys SPA2102

 On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith
 mcampbellsm...@gmail.com wrote:

 The only ATA mentioned on the WIKI that supports TLS/SRTP is the
 Grandstream HandyTone 503.  But, again according to the wiki, that
 doesn't seem to behave to well with TLS ...

 On Wed, Nov 25, 2009 at 7:14 PM, Jason White ja...@jasonjgw.net wrote:
  Mark Campbell-Smith mcampbellsm...@gmail.com wrote:
  Does the SPA3102 support TLS or only SRTP?
 
  I don't know, but supporting only SRTP would be ridiculous, since the
  keys
  would then be transmitted in the clear and therefore amenable to
  interception.
  SRTP requires the SIP channel to be encrypted by TLS in order to be
  secure.
  ZRTP, on the other hand, doesn't have this limitation: it works entirely
  in
  RTP.
 
  I would be rather surprised were a hardware manufacturer to implement
  SRTP
  without TLS for the SIP traffic. On the other hand, we've seen often in
  this
  forum that some manufacturers are really clueless...
 
 
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Re: [Freeswitch-users] How to run a JS script periodically

2009-12-03 Thread Oscav


fs_cli looks like a good idea. I will try that. Many thanks Rob


Rob Forman wrote:
 
 What about cron?
 
 Create a cron entry like:
 */5 * * * * /usr/local/freeswitch/bin/fs_cli -x jsrun yourscript app()
 
 But if you're just dumping global variables, you could easily retrieve
 them
 directly from fs_cli without running an app and process the output however
 you'd like:
 
 /usr/local/freeswitch/bin/fs_cli -x global_getvar
 
 
 On Thu, Dec 3, 2009 at 6:21 AM, Oscav os...@hotmail.fr wrote:
 

 Hi,

 Someone knows how to run periodically a JS script ?? The purpose is to
 write
 to a db some global informations (Global Variables) about FS like every 5
 minutes.

 Thanks.


 --
 View this message in context:
 http://old.nabble.com/How-to-run-a-JS-script-periodically-tp26625147p26625147.html
 Sent from the Freeswitch-users mailing list archive at Nabble.com.


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Re: [Freeswitch-users] Choppy sound with PCMU

Sigh,

You just took it up a notch in terms of disdain and sarcasm.
Why do people always only apologize sarcastically?

I asked you to try the -hp and turn off the monotonic clock just to gather
the results to help you.  You completely missed it and just went on about
the threads.   Please save the ok fine the code is perfect, blah blah if
you would have just read the email and answered the question I might have
cared more about the status of your problem.

I told you both of those threads need to be on their toes because they try
to balance between a certian number of sql stmts or 500ms whatever comes
first.  When there are thousands of events per second being turned into SQL
statements which are in turn compiled into large sql transactions.

If you want to come up with a way that they can sleep longer until there is
a sign of activity and stay busy for a few seconds then slow down again,
that's probably possible but the process is already idle at 0% cpu so maybe
you can appreciate why we are not rushing to work on it.  Maybe I'll give it
a go just to show you it has nothing to do with your problem.

Please don't mock our comment about several years.  You have no idea how
hard this code was to develop and it's truly insulting.  Its clear to see
you are locked into assuming that the busy threads that are not all that
busy because they are constantly yielding to the scheduler is breaking the
timing code.  I begged you to understand me when i told you that the err is
not normal, most boxes do not see it doing nothing and there has to be a
specific problem on your box or configuration.  So instead of working with
us you want to escalate to snotty comments.  That's pretty normal on the
internet I guess.  If you want to have a constructive conversation about
our core, install FS on a normal box, use it for a few weeks, figure out
everything about how it works then try There was pure speculation and
conjecture in your original emails and I never said a word about it until
you kept pushing.

Kristian mentioned he never sees that on that same hardware did you even
consider following up on why that is?

I don't have your device, but I assume if you get it working well it will
certainly help you more than it helps me so you could at least have the
decency to believe what we are trying to tell you.







On Thu, Dec 3, 2009 at 3:44 PM, eaf erandr-j...@usa.net wrote:


 Oh, you mean giving FS higher priority? Yeah, as a last resort I'll do
 that.
 At the moment, I hope it won't be necessary as I can make those hyper
 threads behave, and will see how that goes first. I see where your
 implementation could be coming from. There is a queue of SQL queries in
 sofia.c processed by the worker thread. There are only two pop functions
 available in APR: queue_pop() and queue_trypop(), so alas no option with a
 timeout here. You don't want to block the thread in pop() indefinitely
 because you chose that same worker needs to do ireg and gw processing once
 in a while (separated by tens or hundreds of seconds, btw). You also want
 to
 be able to detect shutdown condition so that the worker doesn't hold up
 profile thread. So you chose to poll for events every millisecond instead
 of
 just creating an apr_thread_cond_t for resource friendly signalling.

 I agree that the timer thread philosophy is great and was the right choice
 for scaling, but I just don't comprehend responses to things like these
 other SQL or sofia worker threads. Did somebody even remotely acknowledge
 that busy loops at least in those areas that I showed may probably be a bad
 idea and could've been eliminated? I've heard suggestions to bump up
 priority, I've heard that the code was perfect already, that it's the
 result
 of 4-year effort, that I am arrogant, don't listen and don't understand
 squat.

 I'm sorry if I gave you impression that I was looking for the bad parts in
 the software. I apologized for that already. All I wanted was to have
 constructive conversation, perhaps I'm not too good at it. Code is already
 perfect according to you? Fine with me.


 Anthony Minessale-2 wrote:
 
  no,
 
  I mean the one after that that you must have completely skipped with a
  command line option to try and a param to set in the config. It somewhat
  annoys me for taking the time to compose it now.  I wrote all of the code
  you are talking about myself and I was trying to give you some
  suggestions
 
  Well, actually,  you did answer my question about the platform so you
 must
  have seen it.
 
  The loops are not the cause of that migration message, something wrong
  with
  the hardware or the kernel is.
  Another guy just told you he does not see that problem on the same exact
  hardware.
 
  Even if you have a point about the sql threads, you could make a patch to
  slow them down but you cant slow down too much or you will not be able to
  handle 400 cps all asking to send updates to transactions in batches of
  thousands of sql stmts.

Re: [Freeswitch-users] Generate cdrs

2009-12-03 Thread Seven Du

why not try mod_xml_cdr?

2009/12/4 Mouncif Benniane mounci...@gmail.com:
 is it possible to run a javascript at the end of dialplan to generate cdrs?
 because (mod_cdr_csv) is giving me hard time as it rotates Master file on
 machine reboots or shutdown signals.
 javascript or LUA for preferences?

 thank you


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Re: [Freeswitch-users] Cannot Do this Basic thing

2009-12-03 Thread Seven Du

You didn't say the exact error was. was 10.15.0.91 == aaa.bbb.ccc.ddd ?

2009/12/4 Samuel Abekah-Mensah ab...@greatiam.com:
 Hi

 Sorry .xm is a typo. I actually shut down the server and restarted. The
 log says I need to create a domain of aaa.bbb.ccc.ddd (which is the
 server IP address ) and then put the user in that domain.  Isn't the
 default domain that of the server FS is running on ?
 2319.xml is in /usr/local/freeswitch/conf/directory/default/

 Thanks for your time



 Michael Collins wrote:


 On Thu, Dec 3, 2009 at 9:46 AM, Samuel Abekah-Mensah
 ab...@greatiam.com mailto:ab...@greatiam.com wrote:

     I have copied 1001.xml in directory/default to a test user 2319.xm
     changing or instances of 1001 in the file to 2319. I then went into
     default.xml  in directory folder and in one of the groups  just
     mimicked
     1001 details by changing 1001 to 2319.

     Connecting  to FS gives Forbidden message. However 1001 connects
     without
     a problem.  What have I missed ?

     Is there a place that just puts things in do this and that and that to
     create a new user ?


 Did you execute reloadxml from the fs cli before trying to connect
 with 2319? Also I'm assuming that 2319.xm is a typo and you actually
 created 2319.xml in the default/directory subdir.
 -MC


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Re: [Freeswitch-users] IAX? Issues connecting road warriors with SIP?

2009-12-03 Thread Tim Uckun


 Do you sometimes/often get issues where SIP (UDP5060) or RTP (UDPwhatever)
 ports fail being opened dynamically to work properly, or does SIP today
 really work well over NAT firewalls?



Yes I get issues quite a bit with the server being behind a firewall.
IAX is much nicer in this circumstance.

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Re: [Freeswitch-users] IAX? Issues connecting road warriors with SIP?

2009-12-03 Thread Jason White

Tim Uckun timuc...@gmail.com wrote:
 
 Yes I get issues quite a bit with the server being behind a firewall.
 IAX is much nicer in this circumstance.

I just set up an IPv6 over IPv4 tunnel and nat goes away.

I have native IPv6 over ADSL now, as part of a trial that my ISP is
conducting. As a result, one end of the conection doesn't go through a tunnel
provider anymore.

Given the problems I've had (and still have) with nat, I want to be rid of it
as much as possible.

Nevertheless, I agree that in a nat scenario, IAX can be easier to configure
correctly.


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Re: [Freeswitch-users] HA questions.

so your registering to the provider to get the calls?  If so, this gets tricky, 
the provider likely does not support multiple registrations, even if they did 
they probably send the call to both registered endpoints.  With this big 
unknown its not very easy to suggest a good solution.  If I were looking to set 
this up without needing proxies I would want to use srv records and naptr 
records and a provider that would balance using these including failiover.

Mike


On Dec 3, 2009, at 3:40 PM, Tim Uckun wrote:

 On Fri, Dec 4, 2009 at 4:59 AM, Michael Jerris m...@jerris.com wrote:
 The easiest place to do this is at the point you send the calls to 
 FreeSWITCH.  How are the calls coming in?
 
 
 From an as of now unkown SIP trunk provider (we are still in
 negotiations with a couple of companies).
 
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[Freeswitch-users] Playing an rtp stream

2009-12-03 Thread Phillip Jones

Hi there,

It it possible do something like:

extension name=rtp
  condition field=destination_number expression=^2127776252$
action application=answer/
action application=playback data=rtp://192.563.41.246:27378/
  /condition
/extension


Basically I have need to connect to incoming calls listen to an existing rtp
stream - I know the IP and port.

Any hints on achieving this would be much appreciated.

Thanks


Phil
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Re: [Freeswitch-users] ATA that supports TLS/SRTP w FS

Cheers Gabriel.. thanks for the information.

I'll look at the Mediatrix ATA's as an alternative - has anyone had
experience with those and TLS/SRTP?

On Fri, Dec 4, 2009 at 10:25 AM, Gabriel Kuri gk...@ieee.org wrote:
The ATAs I'm aware that claim support for TLS and SRTP w/ SDES are the
Grandstream and Mediatrix devices (although I've never tried either
one with FreeSWITCH).

I've personally never had any good experience with the Grandstream
ATAs. The Mediatrix ATAs are OK devices, but I've never personally
tested them with SRTP w/SDES and FreeSWITCH, but supposedly they
support it (so says their marketing material and docs).

I'd see if Cisco has any plans to add support for it to the ATAs. Next
time I see our Cisco SE, I'll try to poke him about it.

Gabe

On Thu, Dec 3, 2009 at 2:34 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Quote: Cisco/Linksys SPA series ATAs do not support SDES key exchange
to appropriately support SRTP and FreeSWITCH

I'll check with Cisco regarding their implementation then and try to
find out when/if they will support standard SRTP encryption.

So, back to my origianal question then. Are there any ATA's that
support TLS AND SRTP with FreeSwitch?

On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri gk...@ieee.org wrote:
AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key
exchange to appropriately support SRTP and FreeSWITCH. They do their
proprietary Sipura key exchange only, not sure if Cisco plans on
upgrading the firmware to ever support SDES on the ATAs. They added
support for SDES to their IP Phones about 1 year ago, but nothing has
happened with the ATAs as of yet.

Gabe

On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi All,

I managed to borrow a SPA3102 with the latest firmware and have got it
to register using TLS, but I am still struggling with SRTP. Has
anyone managed to get SRTP working with the Linksys devices and if so,
can they direct me on how to do this.

I have generated a mini-certificates and SRTP Private Key using the
gen-mc tool found at
http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3.
However, when ever I initiate a call from the SPA, I can see that the
call is not encrypted.

Help appreciated.

Thanks!

On Sat, Nov 28, 2009 at 6:31 AM, eman e...@chabotel.com wrote:
Check out the Linksys SPA2102

On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:

The only ATA mentioned on the WIKI that supports TLS/SRTP is the
Grandstream HandyTone 503. But, again according to the wiki, that
doesn't seem to behave to well with TLS ...

On Wed, Nov 25, 2009 at 7:14 PM, Jason White ja...@jasonjgw.net wrote:
Mark Campbell-Smith mcampbellsm...@gmail.com wrote:
Does the SPA3102 support TLS or only SRTP?

I don't know, but supporting only SRTP would be ridiculous, since the
keys
would then be transmitted in the clear and therefore amenable to
interception.
SRTP requires the SIP channel to be encrypted by TLS in order to be
secure.
ZRTP, on the other hand, doesn't have this limitation: it works
entirely
in
RTP.

I would be rather surprised were a hardware manufacturer to implement
SRTP
without TLS for the SIP traffic. On the other hand, we've seen often in
this
forum that some manufacturers are really clueless...

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Re: [Freeswitch-users] ATA that supports TLS/SRTP w FS

2009-12-03 Thread Yehavi Bourvine

Hello,

I have AudioCodes MP and Vega ATA adapters. They both support SRTP; they
should support TLS also (will try it next week; up to now I preffered to not
use TLS so I can sniff the traffic and debug things).

Regards, __Yehavi:

2009/12/4 Mark Campbell-Smith mcampbellsm...@gmail.com

Cheers Gabriel.. thanks for the information.

I'll look at the Mediatrix ATA's as an alternative - has anyone had
experience with those and TLS/SRTP?

I'd see if Cisco has any plans to add support for it to the ATAs. Next
time I see our Cisco SE, I'll try to poke him about it.

Gabe

On Thu, Dec 3, 2009 at 2:34 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Quote: Cisco/Linksys SPA series ATAs do not support SDES key exchange
to appropriately support SRTP and FreeSWITCH

I'll check with Cisco regarding their implementation then and try to
find out when/if they will support standard SRTP encryption.

So, back to my origianal question then. Are there any ATA's that
support TLS AND SRTP with FreeSwitch?

Gabe

On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi All,

I have generated a mini-certificates and SRTP Private Key using the
gen-mc tool found at

http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3
.
However, when ever I initiate a call from the SPA, I can see that the
call is not encrypted.

Help appreciated.

Thanks!

On Sat, Nov 28, 2009 at 6:31 AM, eman e...@chabotel.com wrote:
Check out the Linksys SPA2102

On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:

The only ATA mentioned on the WIKI that supports TLS/SRTP is the
Grandstream HandyTone 503. But, again according to the wiki, that
doesn't seem to behave to well with TLS ...

On Wed, Nov 25, 2009 at 7:14 PM, Jason White ja...@jasonjgw.net
wrote:
Mark Campbell-Smith mcampbellsm...@gmail.com wrote:
Does the SPA3102 support TLS or only SRTP?

I don't know, but supporting only SRTP would be ridiculous, since
the
keys
would then be transmitted in the clear and therefore amenable to
interception.
SRTP requires the SIP channel to be encrypted by TLS in order to
be
secure.
ZRTP, on the other hand, doesn't have this limitation: it works
entirely
in
RTP.

I would be rather surprised were a hardware manufacturer to
implement
SRTP
without TLS for the SIP traffic. On the other hand, we've seen
often in
this
forum that some manufacturers are really clueless...

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Re: [Freeswitch-users] Cisco IOS gateway: command to send connected line name

2009-12-03 Thread Yehavi Bourvine

I am taking my words back... The Cisco sends back what I want.

I got confused because the Nortel sends the name only for the connected PBX
and not for the othes ones (although it gets this infomation from them).

  Thanks, __Yehavi:

2009/12/3 Yehavi Bourvine yehavi.bourv...@gmail.com

 Unfortunately this didn't help... Incoming calls from ISDN to SIP sends
 back to ISDN the name of the destination, but not the other way around...

 Thanks! __Yehavi:

 2009/12/3 Metik freeswitch-users-l...@metik.com

 Yehavi,

 There are a few variations of transmitting this information... If you
 have already enabled a supplemental isdn service profile, try adding the
 following to the PRI you are using:

 (config-if)#isdn outgoing ie facility
 (config-if)#iisdn outgoing ie extended-facility
 (config-if)#isdn outgoing display-ie
 (config-if)#isdn outgoing ie caller-number
 (config-if)#isdn outgoing ie called-number

 -metik

 Yehavi Bourvine wrote:
  Hello,
 
We have a Cisco running IOS 12.4T used as our SIP-PRI gateway. On
  the PRI there is a Nortel with Q.Sig. After a lot of configuration
  trials I've managed to set it to send back the connected name over the
  SIP (i.e. when a call goes from SIP to PRI, the PRI sends back the
  connected name and then the Cisco adds it as a Remote-Party-ID).
  However, I did not save it and a power outage cleared this config. In
  my age I don't remember what I've done...
 
  Anyone knows the correct config?
 
 Thanks! __Yehavi:
  
 
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Re: [Freeswitch-users] ATA that supports TLS/SRTP w FS