Re: [Freeswitch-users] Hold is broken in trunk 16055

[Anthony Minessale]

We now disable sofia SOA mode during proxy calls.
This means that sofia will not try to get involved in the media negotiation
at all which is the optimal behavior.
Previous versions would butt in and try to fix the error but now it just
stays out of the way.

You can see in your trace that the device sends a packet with no SDP
therefore so does sofia.

You can either turn off  proxy-media or post a bounty for me to go hack a
workaround into the patch I spent many hours on getting things to work
right.  Whatever you experienced with 1.0.4 was a happy coincidence where
sofia was fixing a bug in your phone for you.



On Tue, Dec 29, 2009 at 10:08 AM, Michael Jerris  wrote:

> This means there was no sdp sent.  Did you confirm this with siptrace?
>
> On Dec 29, 2009, at 10:37 AM, Lei Tang  wrote:
>
> Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, following
> code in sofia.c send the 200ok response
> sofia.c
> function sofia_handle_sip_i_state 
>.
> switch(ss_state)
>  
> case nua_callstate_received:
>  .
>  else if (tech_pvt && sofia_test_flag(tech_pvt, TFLAG_SDP) &&
> !r_sdp) {
>   nua_respond(tech_pvt->nh, SIP_200_OK, TAG_END());
>   sofia_set_flag_locked(tech_pvt, TFLAG_NOSDP_REINVITE);
>   goto done;
>  }
>
> The cause is r_sdp is null, but I don't known why tl_gets don't return
> remote sdp tag, it's quite strange.
>
> 2009/12/29 Brian West < br...@freeswitch.org>
>
>> the 200ok is not from FS.. its from the end point... so its not us thats
>> not putting the SDP into the 200ok but the device you're talking to because
>> in proxy media they are passed as is.
>>
>> /b
>>
>> On Dec 29, 2009, at 8:53 AM, Lei Tang wrote:
>>
>> > Hi Brian, thanks for your help, I am using FS in proxy media mode. the
>> sip agent I'm using is x-lite and wxCommunicator.
>> > I will test if trunk 16055 work when I set proxy media mode to false
>> tomorrow.
>>
>>
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>
>
>
> --
> Lei.Tang
> lei.tl...@gmail.com
>
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Re: [Freeswitch-users] PSTN-to-Internal Call Does Not Get Routed to Voice Mail

[Anthony Minessale]

you have to update the sangoma driver and probably FreeSWITCH for good
measure.
Its a known bug in the sangoma driver that has been fixed it the latest
release.



On Mon, Dec 28, 2009 at 5:19 PM, Jerry Richards
wrote:

> Hello All,
>
> I posted a FS log into the Pastebin at
> http://pastebin.freeswitch.org/11644.
>
> I am still having the problem where a PSTN-to-Internal call via a Sangoma
> A101D card stops ringing the internal phone after about 10 seconds.  It
> should be ringing for 30 seconds and then go to Voice Mail (as an
> Internal-to-Internal call does).
>
> Best Regards,
> Jerry
>
>
> -Original Message-
> From: Jerry Richards [mailto:jerry.richa...@teotech.com]
> Sent: Tuesday, December 22, 2009 8:02 AM
> To: 'freeswitch-users@lists.freeswitch.org'
> Subject: PSTN-to-Internal Call Does Not Get Routed to Voice Mail
>
>
> I have a Freeswitch PBX server with an installed Sangoma A101D card
> connected to a PRI.  Most everything works okay, however when I get an
> inbound call from the PSTN, if the call is not answered within about 12
> seconds, the call ends (so it doesn't go to voice mail).  If I make a call
> from one internal phone to another, then it will go to voice mail after 30
> seconds.  How can I get the external call to route to voice mail after 30
> seconds?
>
> I put a new 11595 log into the pastebin.  Do you know any Freeswitch
> setting
> that might cause this?
>
> If this issue has been addressed before, what string should I use to search
> for it, because I can't find it.
>
> Thanks,
> Jerry
>
>
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Re: [Freeswitch-users] [CRIT] mod_event_socket.c:337 Lost 8456 events!

[Anthony Minessale]

most likely cause would be connecting a socket then not regularly reading
from it causing the buffer to fill up.
any event socket connection must select on the socket and do regular read
attempts or all the events will accumulate on the server side until some
sanity check is reached and it begins to throw them away, the fist time
there is room in this buffer again (when you consume some from the socket
leaving space in the queue) it will report how many have been lost since the
last read.

One way to cause this would be suspend fs_cli with ctl-z and bring it back
to the foreground after some time.


On Thu, Dec 24, 2009 at 7:05 AM, Nicolas Brenner wrote:

> I just got into the fs cli and when I ran a 'show calls' I got the
> following message:
>
> 2009-12-24 09:58:20.058365 [CRIT] mod_event_socket.c:337 Lost 8456 events!
>
>
> What does this mean? does it mean the event_socket did not report 8456
> events? Why could this happen?
>
> The answer to this is pretty critical to me, as I make and monitor
> calls through the socket.
>
>
> Thanks for your help!
>
>
> Nicolas
>
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Re: [Freeswitch-users] Choosing a Codec.

[Anthony Minessale]

It's more than highly likely you have some other problem like jitter or a
bad network connection.
Not many people would be able to tell the difference between the sound of an
8k PCM file and the same file encoded to G711 just by listening to it unless
there was a severe problem somewhere.  Since you are behind NAT you are even
more likely to experience drops etc.

Record your files as 8k raw 16 bit PCM to get the best out of the file
playback in FS and look elsewhere for your audio issues.

You can always make sure you are using the latest build of FS to rule out
any temporary issues in the code.



On Wed, Dec 23, 2009 at 8:56 AM, Brian West  wrote:

> VMD will force a transcode anyway too.
>
> /b
>
> On Dec 23, 2009, at 1:08 AM, Vinuth Madinur wrote:
>
> > My setup is as follows:
> >
> > FreeSWITCH -> SIP Trunk -> PSTN.
> >
> > From freeswitch, I'm making outbound calls using event socket via the
> "external" profile. Except for the ext_rtp_ip and ext_sip_ip, everything is
> default settings. Using "playback" application, I'm playing a mu-law audio.
> I'm also starting the "vmd" application, so that I can replay the message on
> beep.
> >
> > Thanks for your suggestion on native format. I'll try it.
> >
> > Thanks,
> > Vinuth.
>
>
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Re: [Freeswitch-users] Skypiax: not able to detect "Inband" dtmf tones from pstn call?

[Anthony Minessale]

add "start_dtmf" app to your dialplan before bridge to start the inband dtmf
detector.


On Tue, Dec 22, 2009 at 8:57 AM, Scott Torr
wrote:

> ubuntu-8.04.3-server-amd64.iso (update/upgrade)
> FreeSWITCH Version 1.0.trunk (15787)
> skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb
> mod_skypiax
>
> (POTS)-->(PSTN)-->(skypeIN)-->(skype_client)-->(skypiax)-->(fs)
>
> 
>  
>
>
>  
> data="/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>
>
>  
> 
>
>
> fs>console loglevel 7
>
>
> If I dial 501 from from a sip phone using "inband" dtmf I can see the
> dtmf tones being detected and decoded by fs in the debug log.
>
>
> If however I use a pstn phone and dial my skypeIN telephone number the
> call comes into fs via skypiax but when I generate dtmf tones on the
> phone they are not detected or decoded by fs.
>
> If I take the record_session file and spectrum analyze the recorded
> tones appear to be within spec.
>
>
> Can anybody suggest why this is not working for me?
>
>
> Is the correct sample rate being used in libteletone_detect.c?
> Does the Goertzel algorithm work for other sample rates other than
> 8000hz?
>
>
> I'm not sure why I can not get this to work?
>
>
>
> regards,
> Scott Torr
>
>
>
>
>
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Re: [Freeswitch-users] Adding H263 Video to Existing Call Fails First Time

[Anthony Minessale]

Can you repeat that same trace with latest trunk?


On Mon, Dec 21, 2009 at 6:44 PM, Jerry Richards
wrote:

>
> After establishing an audio call between two Bria softphones, and then
> starting video at the caller phone, FS replies to the re-INVITE with a 200
> OK with only the PCMU codec.  This looks incorrect.  The audio call
> previously negotiated to the speex/16000 codec, and the re-INVITE from the
> caller added the H263-1998 codec.  If I re-attempt to start video at the
> caller, then it is successful.
>
> I put a Freeswitch log 11596 into the pastebin that contains the complete
> scenario: establishing audio call, first failed start video attempt, and
> second successful start video attempt.
>
> Best Regards,
> Jerry
>
>
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Re: [Freeswitch-users] mod_xml_curl and gateways

[Anthony Minessale]

same exact syntax only put the  in the sofia profile

On Mon, Dec 21, 2009 at 10:13 AM, Jon Bruel  wrote:

>  I wonder if it is possible to define common gateways (not user specific
> gateways) by xml_curl, and if so, the bindings and syntax to use?
>
>
>
> All the best /Jon
>
>
>
>
>
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Re: [Freeswitch-users] How can I detect an execute failure using ESL?

[Anthony Minessale]

the latest version returns an event with that data in it similar to the api
method.


On Mon, Dec 21, 2009 at 2:48 AM, Ron McLeod wrote:

>  When I try and perform an operation on a channel which has gone, an error
> is returned.  How can I detect this using the ESL?  execute() and sendRecv()
> always return 0 (zero) regardless of whether the command returns *+OK* or
> *–ERR*.
>
>
>
> sendmsg 5d09753c-ede7-11de-85c6-27ab474dd533
>
> call-command: execute
>
> execute-app-name: hangup
>
> execute-app-arg: UNALLOCATED_NUMBER
>
>
>
> Content-Type: command/reply
>
> *Reply-Text: -ERR invalid session id
> [5d09753c-ede7-11de-85c6-27ab474dd533]*
>
>
>
> Thanks,
>
> Ron
>
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Re: [Freeswitch-users] Difference between ESL execute() andexecuteAsync()

[Anthony Minessale]

if you run the socket in async mode, every call to execute is async
if you don't specify async in the socket app in FS all calls are synchronous
but you can send async calls with te asyncExecute


On Sat, Dec 19, 2009 at 9:16 PM, Ron McLeod wrote:

> Here's the ES network trace:
>
> Content-Length: 1502
> Content-Type: text/event-plain
> Event-Name: CHANNEL_STATE
> Core-UUID: bb9ea62a-ed02-11de-91b1-8b7cb185f66f
> FreeSWITCH-Hostname: ron-laptop
> FreeSWITCH-IPv4: 192.168.100.132
> FreeSWITCH-IPv6: %3A%3A1
> Event-Date-Local: 2009-12-19%2019%3A12%3A09
> Event-Date-GMT: Sun,%2020%20Dec%202009%2003%3A12%3A09%20GMT
> Event-Date-Timestamp: 1261278729767397
> Event-Calling-File: switch_channel.c
> Event-Calling-Function: switch_channel_perform_set_running_state
> Event-Calling-Line-Number: 1024
> Channel-State: CS_ROUTING
> Channel-State-Number: 2
> Channel-Name: sofia/internal/699%40192.168.100.132
> Unique-ID: 76021ab2-ed15-11de-91b1-8b7cb185f66f
> Call-Direction: inbound
> Presence-Call-Direction: inbound
> Answer-State: ringing
> Channel-Read-Codec-Name: PCMU
> Channel-Read-Codec-Rate: 8000
> Channel-Write-Codec-Name: PCMU
> Channel-Write-Codec-Rate: 8000
> Caller-Username: 699
> Caller-Dialplan: XML
> Caller-Caller-ID-Name: Ron%20Soft%20Phone
> Caller-Caller-ID-Number: 699
> Caller-Network-Addr: 192.168.100.3
> Caller-Destination-Number: 444
> Caller-Unique-ID: 76021ab2-ed15-11de-91b1-8b7cb185f66f
> Caller-Source: mod_sofia
> Caller-Context: mytest
> Caller-Channel-Name: sofia/internal/699%40192.168.100.132
> Caller-Profile-Index: 1
> Caller-Profile-Created-Time: 1261278729764077
> Caller-Channel-Created-Time: 1261278729764077
> Caller-Channel-Answered-Time: 0
> Caller-Channel-Progress-Time: 0
> Caller-Channel-Progress-Media-Time: 0
> Caller-Channel-Hangup-Time: 0
> Caller-Channel-Transfer-Time: 0
> Caller-Screen-Bit: true
> Caller-Privacy-Hide-Name: false
> Caller-Privacy-Hide-Number: false
>
>
> sendmsg 76021ab2-ed15-11de-91b1-8b7cb185f66f
> call-command: execute
> execute-app-name: answer
> execute-app-arg:
>
>
> Content-Type: command/reply
> Reply-Text: +OK
>
>
> sendmsg 76021ab2-ed15-11de-91b1-8b7cb185f66f
> call-command: execute
> execute-app-name: playback
> execute-app-arg: /tmp/ann.wav
>
>
> Content-Type: command/reply
> Reply-Text: +OK
>
>
> > -Original Message-
> > From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-
> > users-boun...@lists.freeswitch.org] On Behalf Of Ron McLeod
> > Sent: Saturday, December 19, 2009 5:30 PM
> > To: freeswitch-users@lists.freeswitch.org
> > Subject: [Freeswitch-users] Difference between ESL execute()
> > andexecuteAsync()
> >
> > I don't notice any different in behavior between execute() and
> > executeAsync().  I was expecting that executeAsync() would return
> > right-away, and that execute() would only return after the specified
> > application runs to completion (CHANNEL_EXECUTE_COMPLETE event).
> >
> > Running the sample app below, I see the "About to call execute(playback)"
> > and "returned" displayed one right-after the other, even though the file
> > being played takes about 4 minutes to play-out.
> >
> > Do I have this wrong, or is there something incorrect in my app?
> >
> > APP:
> > #!/usr/bin/php
> >  > require_once "ESL.php";
> >
> > $eventSocket = New ESLconnection('192.168.100.132', '8021', 'ClueCon');
> > $eventSocket->events('plain', 'CHANNEL_STATE');
> > $eventSocket->filter('channel-state', 'CS_ROUTING');
> >
> > // Wait for new call attempts
> > while($eventSocket->connected()){
> > $event = $eventSocket->recvEvent();
> > $serializedBody = $event->serialize();
> > $listOfLines = toArrayOfLines($serializedBody);
> > $nameValuePairs = toArrayOfNameValuePairs($listOfLines);
> >
> > $uuid = $nameValuePairs['Caller-Unique-ID'];
> > printf("New call from uuid: $uuid\n");
> >
> > // answer the caller and play announcement
> > $eventSocket->execute('answer', Null ,$uuid);
> >
> > printf("About to call execute(playback)\n");
> > $eventSocket->execute('playback', '/tmp/ann.wav', $uuid);
> > printf("returned\n");
> > }
> > ?>
> >
> >
> > DIALPLAN:
> > 
> > 
> >   
> > 
> >   
> > 
> >   
> > 
> >

Re: [Freeswitch-users] RTP problems in recent revisions?

[Anthony Minessale]

I tried a patch out of pure deduction and speculation from your post.
Can you update and test it for me please?


On Sat, Dec 19, 2009 at 9:19 AM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:

> Also retest with no zrtp
> send a full console debug log with sip trace
>
> On Dec 19, 2009 8:33 AM, "Michael Jerris"  wrote:
>
> The best help to track this down is to try to identify the specific
> svn revision that caused the issue and to supply a full freeswitch
> debug with sip trace.
>
> Mike
>
> On Dec 19, 2009, at 3:31 AM, Jason White  wrote: >
> Revision 15904 is fine, but...
>
>


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Re: [Freeswitch-users] Park with Pre Answer

[Anthony Minessale]

how are you parking it?
do you have a debug log showing it happen?


On Fri, Dec 18, 2009 at 11:42 PM, Ron McLeod
wrote:

>  Is there any way to park a channel without causing pre-answer (resulting
> is a SIP 183 Session Progress)?
>
>
>
> Thanks,
>
> Ron
>
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Re: [Freeswitch-users] RTP problems in recent revisions?

[Anthony Minessale]

Also retest with no zrtp
send a full console debug log with sip trace

On Dec 19, 2009 8:33 AM, "Michael Jerris"  wrote:

The best help to track this down is to try to identify the specific
svn revision that caused the issue and to supply a full freeswitch
debug with sip trace.

Mike

On Dec 19, 2009, at 3:31 AM, Jason White  wrote: >
Revision 15904 is fine, but...
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Re: [Freeswitch-users] mod_conference scalability

[Anthony Minessale]

I am more than sure there is probably plenty of room for conference
optimizations it's just a big task.
We don't have a test labbed up and an urgency to work on it.  If you really
want us to pursue trying to improve the performance perhaps you can contact
us at consult...@freeswitch.org and provide us with access your test
environment and let us investigate the possibility of making improvements.




On Fri, Dec 18, 2009 at 2:16 PM, Brian  wrote:

>  Hi  Michael,
>
>
>
> Thanks for the invite, but I can’t make it on the call. Anyway, I’m not
> sure if discussing my specific case is meant for that type of call, is it?
>
>
>
> After Brian’s suggestion to use shoutcast and local streams, I was looking
> at the code for those modules. I’m not familiar with shoutcast or icecast
> capabilities, so I don’t know if they can just pass though my audio stream
> unchanged (as uLaw packets). I want to avoid converting from uLaw to mp3 on
> the source server, and then back from mp3 to uLaw (or whatever phone codec)
> on the other server.
>
>
>
> I was wondering if maybe there was a way to make a stream out of an
> existing channel, and have all the other channels just listen to that
> stream. It would be sort of halfway between conference and shoutcast. I
> would call in to the secondary server like I already do, but only instead of
> entering into a conference as a speaker, the channel would just start
> producing a local audio stream for the listener channels to tap into. It
> would avoid the need to have another piece of software to manage (shoutcast
> or icecast), and my support team would be happier...
>
>
>
> However, I would still need to do tests for the streaming idea to see how
> that scales...
>
>
>
> Brian.
>
>
>
>
>
> *From:* Michael Collins [mailto:m...@freeswitch.org]
> *Sent:* Friday, December 18, 2009 2:33 PM
>
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] mod_conference scalability
>
>
>
>
>
> On Fri, Dec 18, 2009 at 11:14 AM, Brian  wrote:
>
> I was evaluating the technologies available, and I thought you would be
> interested in my results. However, almost every other reply I get from you
> to my posts, rather than being helpful, has been hostile and insulting.
>
> Thanks for your input. Just so you know, Tony deals with people on a near
> daily basis who want to spend time doing crazy schemes under the guise of
> "load testing" or "researching a new solution" which are not grounded in
> reality. At first blush this scenario sounded like one of those schemes.
> However it definitely looks like you've built a test scenario that mimics
> reality better than most. I think we can give you a pass for not being able
> to get 500 people all at once to call in every time you need to test. :)
>
>
>
> My scenario is not a hypothetical one of “having robots call the conference
> in a way that probably does not match reality”. In fact, this will very much
> reflect the reality of the application I’m building. Only instead of 300
> listeners, I need to scale to over 2000 listeners minimum – per event, with
> possibly more than one concurrent event. I want to pack as many listeners on
> one server as I can. I’m trying to find a real solution to a real problem.
>
>  That kind of volume suggests that the icecast style solution would be
> best. It takes much less resources to send audio in one direction than it
> does to mix audio from multiple parties.  I like bkw's initial suggestion of
> transferring a caller to the conference only when he/she needs to speak,
> such as to ask a question. Like Tony mentioned, his focus is on quality not
> quantity, so mod_conference probably isn't the best tool for this scenario.
>
>
>
> I work with other open source projects and fund enhancements or fixes I
> need. FreeSWITCH would be no different.
>
>
>
>  Excellent! It looks like we don't already have a canned solution,
> obviously, but as bkw likes to say, all the Lego bricks are there to build
> the solution.  Hop on IRC (#freeswitch in irc.freenode.net) or join the
> weekly conference which is going on right now and you might catch some of
> the devs and leading community members and you can chat in real-time about
> your challenges. (http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14)
>
> -Michael
>
>
>
> ___
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>


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Re: [Freeswitch-users] Ringing after call has been rejected

[Anthony Minessale]

that will only work if you have not answered yet.
if you already have, you would need to indicate the tones inband like I
mentioned.


On Fri, Dec 18, 2009 at 2:16 PM, Yehavi Bourvine
wrote:

> Try the following:
>
> 
> I don't know whether it will work in your case, but here we use it to
> reject a call while we want to signal that the remote party is busy.
>
>  Regards, __Yehavi:
>
>
>
> 2009/12/18 bcxml 
>
>
>> I have an incomming call being answered by FreeSwitch and passed to IVR
>> application which rejects the call.
>>
>> The call is never answered by FreeSwitch, but instead of hearing a busy
>> signal, the caller hears ringing.
>>
>> Can anyone advise how I can get the user to hear a busy signal after call
>> rejection instead of ringing.
>>
>> Here is the debug trace
>>
>> http://pastebin.freeswitch.org/11558
>>
>> Thanks
>>
>>
>> Brian
>>
>> --
>> View this message in context:
>> http://old.nabble.com/Ringing-after-call-has-been-rejected-tp26842055p26842055.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
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>


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Re: [Freeswitch-users] Allow/Deny REGISTER Request Based on User-Agent Header

[Anthony Minessale]

could be possible with a code change, open a bounty on jira and someone may
do it


On Fri, Dec 18, 2009 at 12:35 PM, Jerry Richards  wrote:

> Is it possible to allow/deny REGISTER requests based on the User-Agent
> header?  I need to know/manage what devices are registering.
>
> Best Regards,
> Jerry
>
>
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Re: [Freeswitch-users] mod_conference scalability

[Anthony Minessale]

Brian, there was not one insulting word in anything I have said and as this
is a community mailing list my replies are always voiced to address the
public in general not you specifically, like I already mentioned in my last
post.

If you open a public forum on a FAQ be prepared to hear our policy.

Indeed many people do unrealistic load testing and most people with strong
will find it insulting when a group of people have a set of standard policy
by which they try to deal with making a penny jar for all the 2 cents worth
of input we get on a daily basis.  I can't begin to iterate over all the
cases we endure on a weekly basis.

additionally 90% of bug reports are on older releases and we always make
people reproduce their issues on SVN trunk because 3 core devs and a handful
of helpers can't maintain 20 versions of the code.

I gave you some really suggestions yesterday let me repaste it, I fail to
see any insults:

---
What exactly is your test process?

you should try increasing the interval in the conference profile to a bigger
time slice maybe 30 40 or 60ms
you could also increase the ptime to match as well.


like brian said you could use mod_shout to broadcast the single speaker to
icecast and let people listen with itunes/winamp

---

I have to get in these "fights" with people constantly so I guess that is
part of my job and my biggest mistake is spending so much time trying to
explain myself.



- Show quoted text -




On Fri, Dec 18, 2009 at 1:33 PM, Michael Collins  wrote:

>
>
> On Fri, Dec 18, 2009 at 11:14 AM, Brian  wrote:
>
>>  I was evaluating the technologies available, and I thought you would be
>> interested in my results. However, almost every other reply I get from you
>> to my posts, rather than being helpful, has been hostile and insulting.
>>
> Thanks for your input. Just so you know, Tony deals with people on a near
> daily basis who want to spend time doing crazy schemes under the guise of
> "load testing" or "researching a new solution" which are not grounded in
> reality. At first blush this scenario sounded like one of those schemes.
> However it definitely looks like you've built a test scenario that mimics
> reality better than most. I think we can give you a pass for not being able
> to get 500 people all at once to call in every time you need to test. :)
>
>>
>>
>> My scenario is not a hypothetical one of “having robots call the
>> conference in a way that probably does not match reality”. In fact, this
>> will very much reflect the reality of the application I’m building. Only
>> instead of 300 listeners, I need to scale to over 2000 listeners minimum –
>> per event, with possibly more than one concurrent event. I want to pack as
>> many listeners on one server as I can. I’m trying to find a real solution to
>> a real problem.
>>
> That kind of volume suggests that the icecast style solution would be best.
> It takes much less resources to send audio in one direction than it does to
> mix audio from multiple parties.  I like bkw's initial suggestion of
> transferring a caller to the conference only when he/she needs to speak,
> such as to ask a question. Like Tony mentioned, his focus is on quality not
> quantity, so mod_conference probably isn't the best tool for this scenario.
>
>>
>>
>> I work with other open source projects and fund enhancements or fixes I
>> need. FreeSWITCH would be no different.
>>
>>
>>
> Excellent! It looks like we don't already have a canned solution,
> obviously, but as bkw likes to say, all the Lego bricks are there to build
> the solution.  Hop on IRC (#freeswitch in irc.freenode.net) or join the
> weekly conference which is going on right now and you might catch some of
> the devs and leading community members and you can chat in real-time about
> your challenges. (http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14)
>
> -Michael
>
>
> _______
> FreeSWITCH-users mailing list
> FreeSWITCH-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


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Re: [Freeswitch-users] mod_conference scalability

[Anthony Minessale]

yes, I understand.
My reply was to the thread in general not directed at you =p


On Fri, Dec 18, 2009 at 11:41 AM, François Delawarde <
fdelawa...@wirelessmundi.com> wrote:

> It was of course just bad humor, I love both projects for what they are,
> and I agree that both have their own advantages and inconvenients.
>
> For example, accessing that same conference from a dahdi card could be
> another goal where Asterisk would be at an advantage, as chan_dahdi is
> still superior (in the more tested sense) than openzap+mod_openzap.
>
> I just use both projects separately or together depending on what's
> needed!
>
> I'm no banker nor do I understand the code, but many thanks for all
> those unpaid contributions providing an excellent alternative for free
> telephony. Your names really deserve being engraved in google's cache
> for eternity. :-)
>
> But still, I would like to see those numbers...
>
> François.
>
>
> On Fri, 2009-12-18 at 10:34 -0600, Anthony Minessale wrote:
> > Conferencing is hardly the best place to judge performance.
> > Quality is a far more important goal to me in conferencing.
> >
> > Lets compare who can do 48khz conferences with several 32k siren
> > callers on a polycom 6000, several more using G722 at 16khz and
> > another handful of people on g711 ulaw all at different rates and
> > ptimes talking in near-real time with low delay and low echo.  The
> > fact that you can broadcast the conferences to icecast, control it
> > from an external application and play files etc, and oh yeah, it can
> > stream video.
> >
> > Frankly, considering this is a free software project and so many
> > people benefit, i would rather focus on quality than what numbers i
> > can get from having robots call the conference in some way that
> > probably does not match reality.  I would love for someone to sponsor
> > the effort to add features to the conference module, but of course, I
> > do not hold my breath, instead I continue to improve it for free when
> > I find time.  This is one of many reasons I do not enjoy performance
> > discussions unless I am talking to an engineer who understands the
> > code or a banker ready to pay for improvements.  That is not my way of
> > saying pay me or forget it as you can clearly see the conference
> > module has made it to where it is today with no financial support at
> > all.  Just the efforts of myself and several brave volunteers over the
> > years who have contributed to it.
> >
> > BTW,
> >
> > We have a weekly call, there is one today in 30 minutes.
> > Drop by 
> > sip:8...@conference.freeswitch.orgThis 
> > is just an openVZ
> > instance mind you running at 48khz waiting for anyone to call in and
> > say hi.
> >
> >
> >
> >
> >
> > On Fri, Dec 18, 2009 at 10:12 AM, François Delawarde
> >  wrote:
> > Hearing that Asterisk (1.4) scales 2x like FS is not common,
> > sounds like
> > a configuration error.
> >
> > If not, I already see the title of the next Digium blog entry:
> > "FreeSwitch scalability myth finally ends: The worst Asterisk
> > version
> > ever (1.4) beating the crap of the best and latest FS."
> >
> > Anyway, you should compare FS trunk to Asterisk 1.6.2 to see
> > who wins
> > the final conference battle! :-)
> >
> > François.
> >
> >
> >
> > On Thu, 2009-12-17 at 16:41 -0500, Brian wrote:
> > > I did a test with the trunk version for the one conference
> > case, and
> > > it is the same results as for 1.0.4. The audio failed at
> > around 300
> > > listeners. Oddly though, it consumed less %CPU (240% instead
> > of 300%),
> > > and yet the audio still failed at the same number of
> > listeners.
> > >
> > >
> > >
> > > Brian.
> > >
> > >
> > >
> > > From: Anthony Minessale [mailto:anthony.miness...@gmail.com]
> > > Sent: Thursday, December 17, 2009 3:49 PM
> > > To: freeswitch-users@lists.freeswitch.org
> > > Subject: Re: [Freeswitch-users] mod_conference scalability
> > >
> > >
> > >
> > >
> > > We didn't post it anywhere but we just get overwhelmed with
> > them and
> > > many of t

Re: [Freeswitch-users] Ringing after call has been rejected

[Anthony Minessale]

not answering it would be the best way.
if you want to generate fake congestion you can use tone_stream:// or
gentones


On Fri, Dec 18, 2009 at 5:16 AM, bcxml  wrote:

>
> I have an incomming call being answered by FreeSwitch and passed to IVR
> application which rejects the call.
>
> The call is never answered by FreeSwitch, but instead of hearing a busy
> signal, the caller hears ringing.
>
> Can anyone advise how I can get the user to hear a busy signal after call
> rejection instead of ringing.
>
> Here is the debug trace
>
> http://pastebin.freeswitch.org/11558
>
> Thanks
>
>
> Brian
>
> --
> View this message in context:
> http://old.nabble.com/Ringing-after-call-has-been-rejected-tp26842055p26842055.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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Re: [Freeswitch-users] mod_conference scalability

[Anthony Minessale]

Conferencing is hardly the best place to judge performance.
Quality is a far more important goal to me in conferencing.

Lets compare who can do 48khz conferences with several 32k siren callers on
a polycom 6000, several more using G722 at 16khz and another handful of
people on g711 ulaw all at different rates and ptimes talking in near-real
time with low delay and low echo.  The fact that you can broadcast the
conferences to icecast, control it from an external application and play
files etc, and oh yeah, it can stream video.

Frankly, considering this is a free software project and so many people
benefit, i would rather focus on quality than what numbers i can get from
having robots call the conference in some way that probably does not match
reality.  I would love for someone to sponsor the effort to add features to
the conference module, but of course, I do not hold my breath, instead I
continue to improve it for free when I find time.  This is one of many
reasons I do not enjoy performance discussions unless I am talking to an
engineer who understands the code or a banker ready to pay for
improvements.  That is not my way of saying pay me or forget it as you can
clearly see the conference module has made it to where it is today with no
financial support at all.  Just the efforts of myself and several brave
volunteers over the years who have contributed to it.

BTW,

We have a weekly call, there is one today in 30 minutes.
Drop by 
sip:8...@conference.freeswitch.orgThis
is just an openVZ instance mind you running at 48khz waiting for
anyone
to call in and say hi.





On Fri, Dec 18, 2009 at 10:12 AM, François Delawarde <
fdelawa...@wirelessmundi.com> wrote:

> Hearing that Asterisk (1.4) scales 2x like FS is not common, sounds like
> a configuration error.
>
> If not, I already see the title of the next Digium blog entry:
> "FreeSwitch scalability myth finally ends: The worst Asterisk version
> ever (1.4) beating the crap of the best and latest FS."
>
> Anyway, you should compare FS trunk to Asterisk 1.6.2 to see who wins
> the final conference battle! :-)
>
> François.
>
>
> On Thu, 2009-12-17 at 16:41 -0500, Brian wrote:
> > I did a test with the trunk version for the one conference case, and
> > it is the same results as for 1.0.4. The audio failed at around 300
> > listeners. Oddly though, it consumed less %CPU (240% instead of 300%),
> > and yet the audio still failed at the same number of listeners.
> >
> >
> >
> > Brian.
> >
> >
> >
> > From: Anthony Minessale [mailto:anthony.miness...@gmail.com]
> > Sent: Thursday, December 17, 2009 3:49 PM
> > To: freeswitch-users@lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] mod_conference scalability
> >
> >
> >
> >
> > We didn't post it anywhere but we just get overwhelmed with them and
> > many of them are unfounded and take up a lot of time to track down.
> > That does not mean you have not found a real problem but the first
> > step is trying trunk.
> >
> >
> >
> >
> > On Thu, Dec 17, 2009 at 2:32 PM, Brian 
> > wrote:
> >
> > I didn’t realize there was a policy about load testing questions. What
> > forum should I have used for this?
> >
> >
> >
> > I didn’t get the chance to test on FS trunk yet, but when I do I will
> > provide you with the feedback when I do. Just let me know what forum
> > to use for this topic from now on.
> >
> >
> >
> > Thanks,
> >
> >
> >
> > Brian.
> >
> >
> >
> > From: Anthony Minessale [mailto:anthony.miness...@gmail.com]
> > Sent: Thursday, December 17, 2009 2:42 PM
> >
> >
> > To: freeswitch-users@lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] mod_conference scalability
> >
> >
> >
> >
> > One man's stable release is another man's 6 month old release with
> > hundreds of known fixed bugs.
> > If one of the core developers tells you to try it, you may as well
> > take the time to try it now that you have opened a forum questioning
> > the scalability.
> >
> > When you tested asterisk did you actually use 600 phones and verify
> > that each one can hear the audio perfectly and in time with what the
> > speaker was saying?  Did you try same on FS?
> >
> > Did you optimize your dialplan on FS to deal with a load test or
> > follow any of the recommended performance tuning page.
> >
> > All of the answers to these questions are really moot because we have
> > a policy against entertaining load testing questions but if you like
> > asterisk, by all means, use it, and good luck to you if th

Re: [Freeswitch-users] Voicemail->Email

[Anthony Minessale]

oh really,
sendmail segfaults?

if another application is crashing you need to figure that out, whatever
used to work doesnt now so you need to figure out what it was and let us
know.


On Fri, Dec 18, 2009 at 3:51 AM, François Legal wrote:

> I get the same result with sendmail. This used to work in 1.0.3 , and after
> upgrading to 1.0.4 (then some snapshots) then 1.0.5pre8 and the problem is
> still there.
>
>
>
> François
>
>
>
> On Thu, 17 Dec 2009 17:33:58 +0100, Oliver Schönbeck wrote:
>
>  Currently it is Version 1.0.trunk (15982)
>
>
>
> *Von:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
> freeswitch-users-boun...@lists.freeswitch.org] *Im Auftrag von *Brian West
> *Gesendet:* Donnerstag, 17. Dezember 2009 17:17
> *An:* freeswitch-users@lists.freeswitch.org
> *Betreff:* Re: [Freeswitch-users] Voicemail->Email
>
>
>
> What SVN rev. exactly?
>
>
>
> /b
>
>
>
> On Dec 17, 2009, at 10:13 AM, Oliver Schönbeck wrote:
>
>
>
>   Hello,
>
>
>
> we are running freeswitch 1.0.trunk and are currently trying to get the
> mod_voicemail to send the received messages to the user by using exim4 on a
> debian machine.
>
>
>
> So far we followed  the instructions in the wiki article (
> http://wiki.freeswitch.org/wiki/Mod_voicemail ).
>
>
>
> I added some lines to the bash script to enable some kind of logging:
> #! /bin/bash
>
> typeset LOG="/tmp/${0##*/}.out"
>
> mv $LOG ${LOG}.old >/dev/null 2>&1
>
> [[ -t 1 ]] && echo "Writing to logfile '$LOG'."
>
> exec > $LOG 2>&1
>
> exim4 -t -v >> $LOG
>
>
>
> If I run the script from the command line everything is working as
> expected. If the script gets called by freeswitch I get the following result
> in my logfile:
>
> /usr/local/freeswitch/scripts/exec_exim.sh: line 6:  4920 Segmentation
> fault  (core dumped) exim4 -t -v >> $LOG
>
>
>
> Has anybody seen similar effects before?
>
>
>
> Any advice whats going wrong is heavily appreciated.
>
>
>
> Thanks
>
>Oliver
>
>
>
>
>
> ___
> FreeSWITCH-users mailing list
> FreeSWITCH-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> ___
> FreeSWITCH-users mailing list
> FreeSWITCH-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
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 
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pstn:+19193869900
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Re: [Freeswitch-users] mod_conference scalability

[Anthony Minessale]

What exactly is your test process?

you should try increasing the interval in the conference profile to a bigger
time slice maybe 30 40 or 60ms
you could also increase the ptime to match as well.


like brian said you could use mod_shout to broadcast the single speaker to
icecast and let people listen with itunes/winamp


On Thu, Dec 17, 2009 at 3:41 PM, Brian  wrote:

>  I did a test with the trunk version for the one conference case, and it
> is the same results as for 1.0.4. The audio failed at around 300 listeners.
> Oddly though, it consumed less %CPU (240% instead of 300%), and yet the
> audio still failed at the same number of listeners.
>
>
>
> Brian.
>
>
>
> *From:* Anthony Minessale [mailto:anthony.miness...@gmail.com]
> *Sent:* Thursday, December 17, 2009 3:49 PM
>
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] mod_conference scalability
>
>
>
> We didn't post it anywhere but we just get overwhelmed with them and many
> of them are unfounded and take up a lot of time to track down.  That does
> not mean you have not found a real problem but the first step is trying
> trunk.
>
>
>  On Thu, Dec 17, 2009 at 2:32 PM, Brian  wrote:
>
> I didn’t realize there was a policy about load testing questions. What
> forum should I have used for this?
>
>
>
> I didn’t get the chance to test on FS trunk yet, but when I do I will
> provide you with the feedback when I do. Just let me know what forum to use
> for this topic from now on.
>
>
>
> Thanks,
>
>
>
> Brian.
>
>
>
> *From:* Anthony Minessale [mailto:anthony.miness...@gmail.com]
> *Sent:* Thursday, December 17, 2009 2:42 PM
>
>
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] mod_conference scalability
>
>
>
> One man's stable release is another man's 6 month old release with hundreds
> of known fixed bugs.
> If one of the core developers tells you to try it, you may as well take the
> time to try it now that you have opened a forum questioning the scalability.
>
> When you tested asterisk did you actually use 600 phones and verify that
> each one can hear the audio perfectly and in time with what the speaker was
> saying?  Did you try same on FS?
>
> Did you optimize your dialplan on FS to deal with a load test or follow any
> of the recommended performance tuning page.
>
> All of the answers to these questions are really moot because we have a
> policy against entertaining load testing questions but if you like asterisk,
> by all means, use it, and good luck to you if those numbers you are testing
> at are what you plan to put in real production.
>
> On Thu, Dec 17, 2009 at 1:29 PM, Brian  wrote:
>
> Hi Mike,
>
>
>
> I didn’t get around to testing on the FreeSWITCH trunk yet. Are there
> substantial fixes to mod_conference in the FreeSWITCH trunk that might
> increase capacity for my scenario of one speaker and many listeners? If I
> want to put this into a production environment, I would need a stable
> version, which as far as I know is the 1.0.4 version.
>
>
>
> However, I did test on Asterisk 1.4 using app_conference, and doing the
> same scenario was able to get 1 speaker and 600 listeners on a single
> conference with no audio issues. The CPU at that point was just over 300%,
> same as where the single conference scenario failed on FreeSWITCH with 300
> listeners.  I was able to push it to over 700 listeners before I reached
> 400% CPU usage (I guess maxing out my quad-core processors), and asterisk
> finally crashed. But up until that point, there were no audio problems.
>
>
>
> I’ve read a lot about how FreeSWITCH is supposed to be more scalable than
> Asterisk, but unless there is something wrong with my FreeSWITCH setup,
> Asterisk was clearly the winner in this test – more than doubling FreeSWITCH
> capacity in this case. Again, maybe there is something on the FreeSWITCH
> side that I’m doing wrong, but I don’t see what it could be.
>
>
>
> Brian.
>
>
>
>
>
> *From:* Michael Jerris [mailto:m...@jerris.com]
> *Sent:* Thursday, December 17, 2009 10:18 AM
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] mod_conference scalability
>
>
>
> I would be curious what the same tests produce with svn trunk of
> FreeSWITCH.
>
>
>
> Mike
>
>
>
> On Dec 16, 2009, at 4:49 PM, Brian wrote:
>
>
>
> Hi,
>
>
>
> I’m new to FreeSWITCH and I’m testing the scalability of mod_conference to
> see if it will scale better that other solutions. My scenario is to have one
> speaker, and many listeners (mute). Since I have only one speaker, I 

Re: [Freeswitch-users] Voicemail->Email

[Anthony Minessale]

yah it's exim segfaulting because you have to configure it to emulate
sendmail per the wiki page.


On Thu, Dec 17, 2009 at 4:17 PM, Peter P GMX  wrote:

> Hello Oliver,
>
> I have the same on Ubuntu wth newest trunk.
>
> Best regards
> Peter
>
> Oliver Schönbeck schrieb:
> >
> > Hello,
> >
> >
> >
> > we are running freeswitch 1.0.trunk and are currently trying to get
> > the mod_voicemail to send the received messages to the user by using
> > exim4 on a debian machine.
> >
> >
> >
> > So far we followed  the instructions in the wiki article (
> > http://wiki.freeswitch.org/wiki/Mod_voicemail ).
> >
> >
> >
> > I added some lines to the bash script to enable some kind of logging:
> > #! /bin/bash
> >
> > typeset LOG="/tmp/${0##*/}.out"
> >
> > mv $LOG ${LOG}.old >/dev/null 2>&1
> >
> > [[ -t 1 ]] && echo "Writing to logfile '$LOG'."
> >
> > exec > $LOG 2>&1
> >
> > exim4 -t -v >> $LOG
> >
> >
> >
> > If I run the script from the command line everything is working as
> > expected. If the script gets called by freeswitch I get the following
> > result in my logfile:
> >
> > /usr/local/freeswitch/scripts/exec_exim.sh: line 6:  4920 Segmentation
> > fault  (core dumped) exim4 -t -v >> $LOG
> >
> >
> >
> > Has anybody seen similar effects before?
> >
> >
> >
> > Any advice whats going wrong is heavily appreciated.
> >
> >
> >
> > Thanks
> >
> >Oliver
> >
> >
> >
> >
> >
> > 
> >
> > ___
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users@lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> ___
> FreeSWITCH-users mailing list
> FreeSWITCH-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
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:+19193869900
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Re: [Freeswitch-users] Handling REFER...

[Anthony Minessale]

The calls inherit the context from the parent, I think there is a var you
can set on the chan to pick what context to use in a transfer like
transfer_context or something grep the code for it

On Dec 17, 2009 1:07 PM, "Kristian Kielhofner" <
kristian.kielhof...@gmail.com> wrote:

Hello everyone,

I've got two profiles running: s2s and trunk.  The context for s2s is
defined as s2s-in.  The context for trunk is defined as trunk-in.
trunk is bound to 192.168.168.3.

recv 481 bytes from udp/[192.168.168.76]:5065 at 18:43:37.309706:
  
  REFER sip:mod_so...@192.168.168.3:5060 SIP/2.0
  Via: SIP/2.0/UDP 192.168.168.76:5065;branch=z9hG4bK__7539073431157335561_9
  To: "NONAME" 
>;tag=BagvZeKSrj7yH
  From: ;tag=203332153_1430350929_10
  Call-ID: e505f332-65de-122d-d183-eb12ad0ec1ac
  CSeq: 2 REFER
  Max-Forwards: 70
  Refer-To: >
  Contact: 
  Content-Length: 0

  
send 592 bytes to udp/[192.168.168.76]:5065 at 18:43:37.316093:
  
  SIP/2.0 202 Accepted
  Via: SIP/2.0/UDP 192.168.168.76:5065;branch=z9hG4bK__7539073431157335561_9
  From: ;tag=203332153_1430350929_10
  To: "NONAME" 
>;tag=BagvZeKSrj7yH
  Call-ID: e505f332-65de-122d-d183-eb12ad0ec1ac
  CSeq: 2 REFER
  Contact: 
  User-Agent: FreeSWITCH
  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,
SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
  Supported: precondition, path, replaces
  Allow-Events: talk, refer
  Content-Length: 0

 FS routed this to the s2s-in context, even though it was sent to the
trunk profile.  Shouldn't it have ended up in trunk-in?  For the time
being I wrote some crazy dialplan for s2s-in to transfer the call to
trunk-in but I'm wondering what could be going on here.

--
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] mod_conference scalability

[Anthony Minessale]

We didn't post it anywhere but we just get overwhelmed with them and many of
them are unfounded and take up a lot of time to track down.  That does not
mean you have not found a real problem but the first step is trying trunk.



On Thu, Dec 17, 2009 at 2:32 PM, Brian  wrote:

>  I didn’t realize there was a policy about load testing questions. What
> forum should I have used for this?
>
>
>
> I didn’t get the chance to test on FS trunk yet, but when I do I will
> provide you with the feedback when I do. Just let me know what forum to use
> for this topic from now on.
>
>
>
> Thanks,
>
>
>
> Brian.
>
>
>
> *From:* Anthony Minessale [mailto:anthony.miness...@gmail.com]
> *Sent:* Thursday, December 17, 2009 2:42 PM
>
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] mod_conference scalability
>
>
>
> One man's stable release is another man's 6 month old release with hundreds
> of known fixed bugs.
> If one of the core developers tells you to try it, you may as well take the
> time to try it now that you have opened a forum questioning the scalability.
>
> When you tested asterisk did you actually use 600 phones and verify that
> each one can hear the audio perfectly and in time with what the speaker was
> saying?  Did you try same on FS?
>
> Did you optimize your dialplan on FS to deal with a load test or follow any
> of the recommended performance tuning page.
>
> All of the answers to these questions are really moot because we have a
> policy against entertaining load testing questions but if you like asterisk,
> by all means, use it, and good luck to you if those numbers you are testing
> at are what you plan to put in real production.
>
>  On Thu, Dec 17, 2009 at 1:29 PM, Brian  wrote:
>
> Hi Mike,
>
>
>
> I didn’t get around to testing on the FreeSWITCH trunk yet. Are there
> substantial fixes to mod_conference in the FreeSWITCH trunk that might
> increase capacity for my scenario of one speaker and many listeners? If I
> want to put this into a production environment, I would need a stable
> version, which as far as I know is the 1.0.4 version.
>
>
>
> However, I did test on Asterisk 1.4 using app_conference, and doing the
> same scenario was able to get 1 speaker and 600 listeners on a single
> conference with no audio issues. The CPU at that point was just over 300%,
> same as where the single conference scenario failed on FreeSWITCH with 300
> listeners.  I was able to push it to over 700 listeners before I reached
> 400% CPU usage (I guess maxing out my quad-core processors), and asterisk
> finally crashed. But up until that point, there were no audio problems.
>
>
>
> I’ve read a lot about how FreeSWITCH is supposed to be more scalable than
> Asterisk, but unless there is something wrong with my FreeSWITCH setup,
> Asterisk was clearly the winner in this test – more than doubling FreeSWITCH
> capacity in this case. Again, maybe there is something on the FreeSWITCH
> side that I’m doing wrong, but I don’t see what it could be.
>
>
>
> Brian.
>
>
>
>
>
> *From:* Michael Jerris [mailto:m...@jerris.com]
> *Sent:* Thursday, December 17, 2009 10:18 AM
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] mod_conference scalability
>
>
>
> I would be curious what the same tests produce with svn trunk of
> FreeSWITCH.
>
>
>
> Mike
>
>
>
> On Dec 16, 2009, at 4:49 PM, Brian wrote:
>
>
>
> Hi,
>
>
>
> I’m new to FreeSWITCH and I’m testing the scalability of mod_conference to
> see if it will scale better that other solutions. My scenario is to have one
> speaker, and many listeners (mute). Since I have only one speaker, I was
> expecting this to scale well because there is no audio mixing required, just
> send each frame of the single speaker to each listener. Unfortunately, my
> testing was disappointing, and it didn’t scale nearly as well as I’d hoped
> (based on what I’ve read on how FreeSWITCH is supposed to be generally very
> scalable).
>
>
>
> Here’s my server setup is this:
>
>
>
> FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of
> RAM. I’ve set file logging to “notice” level. My conference profile is
> configured to suppress several events, hoping that it would improve
> performance.
>
>
>
> Here are a few scenarios I tested, and roughly where I reached the point of
> audio failure on the conferences:
>
>
>
> Scenario 1:
>
> 1 conference, 1 speaker, audio failed at approx 300 listeners (mute)
>
>
>
> Scenario 2:
>
> 4 conferences, 1 speaker per conference, audio failed

Re: [Freeswitch-users] mod_conference scalability

[Anthony Minessale]

One man's stable release is another man's 6 month old release with hundreds
of known fixed bugs.
If one of the core developers tells you to try it, you may as well take the
time to try it now that you have opened a forum questioning the scalability.

When you tested asterisk did you actually use 600 phones and verify that
each one can hear the audio perfectly and in time with what the speaker was
saying?  Did you try same on FS?

Did you optimize your dialplan on FS to deal with a load test or follow any
of the recommended performance tuning page.

All of the answers to these questions are really moot because we have a
policy against entertaining load testing questions but if you like asterisk,
by all means, use it, and good luck to you if those numbers you are testing
at are what you plan to put in real production.


On Thu, Dec 17, 2009 at 1:29 PM, Brian  wrote:

>  Hi Mike,
>
>
>
> I didn’t get around to testing on the FreeSWITCH trunk yet. Are there
> substantial fixes to mod_conference in the FreeSWITCH trunk that might
> increase capacity for my scenario of one speaker and many listeners? If I
> want to put this into a production environment, I would need a stable
> version, which as far as I know is the 1.0.4 version.
>
>
>
> However, I did test on Asterisk 1.4 using app_conference, and doing the
> same scenario was able to get 1 speaker and 600 listeners on a single
> conference with no audio issues. The CPU at that point was just over 300%,
> same as where the single conference scenario failed on FreeSWITCH with 300
> listeners.  I was able to push it to over 700 listeners before I reached
> 400% CPU usage (I guess maxing out my quad-core processors), and asterisk
> finally crashed. But up until that point, there were no audio problems.
>
>
>
> I’ve read a lot about how FreeSWITCH is supposed to be more scalable than
> Asterisk, but unless there is something wrong with my FreeSWITCH setup,
> Asterisk was clearly the winner in this test – more than doubling FreeSWITCH
> capacity in this case. Again, maybe there is something on the FreeSWITCH
> side that I’m doing wrong, but I don’t see what it could be.
>
>
>
> Brian.
>
>
>
>
>
> *From:* Michael Jerris [mailto:m...@jerris.com]
> *Sent:* Thursday, December 17, 2009 10:18 AM
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] mod_conference scalability
>
>
>
> I would be curious what the same tests produce with svn trunk of
> FreeSWITCH.
>
>
>
> Mike
>
>
>
> On Dec 16, 2009, at 4:49 PM, Brian wrote:
>
>
>
>   Hi,
>
>
>
> I’m new to FreeSWITCH and I’m testing the scalability of mod_conference to
> see if it will scale better that other solutions. My scenario is to have one
> speaker, and many listeners (mute). Since I have only one speaker, I was
> expecting this to scale well because there is no audio mixing required, just
> send each frame of the single speaker to each listener. Unfortunately, my
> testing was disappointing, and it didn’t scale nearly as well as I’d hoped
> (based on what I’ve read on how FreeSWITCH is supposed to be generally very
> scalable).
>
>
>
> Here’s my server setup is this:
>
>
>
> FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of
> RAM. I’ve set file logging to “notice” level. My conference profile is
> configured to suppress several events, hoping that it would improve
> performance.
>
>
>
> Here are a few scenarios I tested, and roughly where I reached the point of
> audio failure on the conferences:
>
>
>
> Scenario 1:
>
> 1 conference, 1 speaker, audio failed at approx 300 listeners (mute)
>
>
>
> Scenario 2:
>
> 4 conferences, 1 speaker per conference, audio failed approx 110 listeners
> per conference (so just over 400 total channels on the system).
>
>
>
> Scenario 3:
>
> 16 conferences, 1 speaker per conference, audio failed at 32 listeners per
> conference (so just over 500 total channels on the system).
>
>
>
>
>
> Looking at the output from “top”, it seems that in all 3 scenarios, the
> audio quality failed when the % CPU for the FreeSWITCH process exceeded
> 300%.
>
>
>
> I was hoping maybe someone else might have done similar testing, or maybe
> has suggestions on how to improve the performance. Or perhaps an alternate
> solution to the one speaker, many listener case?
>
>
>
> Thanks,
>
>
>
> Brian.
>
>
>
> ___
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Re: [Freeswitch-users] Small delay in registration validity

[Anthony Minessale]

The sql is sorted into transactions to boost performance so it waits for
either 500 statements to execute or 500ms to elapse to accumulate as many
sql stmts as possible into the transaction.

set sql-in-transactions to false in your profile or make a patch to make the
500ms configurable




On Thu, Dec 17, 2009 at 9:53 AM, mayamatakeshi wrote:

> It seems to me, in previous revisions of FS, we could successfully call a
> registered user as soon as his terminal gets 200 OK for REGISTER.
> But after testing recent revisions, it seems we must wait a little (I wait
> 1 second) otherwise a call to bridge would end with this:
>
> 2009-12-17 22:46:14.155163 [ERR] switch_ivr_originate.c:2249 Cannot create
> outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED]
>
> Similar thing is happening when the terminal unregisters: after
> unregistration an immediate call to bridge sofia/profile/user%domain will
> succeed.
>
> Has anything changed recently in the way registration works that could
> explain this?
>
> br,
> takeshi
>
> ___
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>


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Re: [Freeswitch-users] detecting rtp packet for zombie channels

[Anthony Minessale]

sip session timers is the standardized way to handle this.


On Thu, Dec 17, 2009 at 10:00 AM, Mathieu Rene  wrote:

> Are you doing proxy or bypass meda?
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mr...@avgs.ca
>
>
>
>
> On 17-Dec-09, at 12:53 AM, Juan Backson wrote:
>
> Hi
>
> I have rtp-timeout-sec set to 300 s but I am still getting calls with
> duration of 1 day long.
>
> Is there any other ways to check for zombie channels?
>
> jb
>
> On Wed, Dec 16, 2009 at 10:52 PM, Brian West  wrote:
>
>> Why not just set rtp-timeout-sec on the sofia profile and it'll do
>> that for you.
>>
>> Unless something else is going on.
>>
>> /b
>>
>> On Dec 16, 2009, at 6:33 AM, Juan Backson wrote:
>>
>> > Hi,
>> >
>> > I am having problem with around 1 % of the channels always get
>> > zombilized.
>> >
>> > What I want to do is to have a background thread that regularly
>> > check all the channels that have been in existance for like > 1 hr,
>> > and then check to see if there is any RTP coming in and going out.
>> > If there is no RTP, then I just hangup that channel.  Does anyone
>> > know if there is anyway to do that in a freeswitch module?  Which
>> > API can I use to accomplish this purpose?  Alternatively, is there
>> > anyway to configure freeswitch so that it will hangup the calls
>> > where there is no media in and out for so many seconds?
>> >
>> > Thanks,
>> > jb
>> > ___
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>> > users
>> > http://www.freeswitch.org
>>
>>
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>
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Re: [Freeswitch-users] SIP Re-invite

[Anthony Minessale]

The question was:

Are you doing the packet capture on the actual FS box using tshark or
tcpdump?


On Thu, Dec 17, 2009 at 9:48 AM, DJB  wrote:

> Anthony,
>
> I have pasted the invite sip trace here:
> http://pastebin.freeswitch.org/11536
> Please advise if you need further info.
>
> Thank you.
>
>  ------
> *From:* Anthony Minessale 
> *To:* freeswitch-users@lists.freeswitch.org
> *Sent:* Wed, December 16, 2009 3:42:48 PM
> *Subject:* Re: [Freeswitch-users] SIP Re-invite
>
> that means the invite is not matching the call dialog
> compare the via tags and call-id etc
>
>
> On Wed, Dec 16, 2009 at 5:29 PM, DJB  wrote:
>
>>   We have a customer that we are sending calls to off the FS and here is
>> the issue:
>>
>>
>>
>> Call is initially setup fine and they send a first re-invite with media
>> 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first
>> re-invite fine
>>
>>
>>
>> They then send a second re-invite with their media IP to cut through media
>> and the FS sends a 200 OK to this fine. At this point the call is fine
>>
>>
>>
>> 30 minutes later they send a third re-invite because according to them it
>> is strictly for the purpose of “keep alive” per RFC 4028. This third
>> re-invite has the exact same media IP and UDP pot information as the second
>> re-invite does. The problem is FS does not respond to this third re-invite
>> AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the
>> call to be dropped as the other end does not recieve a response from FS.
>>
>>
>> One more thing, we did not see the third re-invite in sofia siptrace, but
>> we do see it in ethereal, which is kind of odds.
>>
>>
>> We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode.
>>
>>
>> Thank you very much.
>>
>>
>> _______
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users@lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
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>
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>
> FreeSWITCH Developer Conference
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> iax:gu...@conference.freeswitch.org/888
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> pstn:+19193869900
>
>
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>
>


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Re: [Freeswitch-users] SIP Re-invite

[Anthony Minessale]

Is the packet capture running on the FS box itself?


On Thu, Dec 17, 2009 at 9:36 AM, Michael Jerris  wrote:

> if you don't see it in sofia siptrace but do see it in tcpdump capture then
> something very ugly is going on.  Either sofia has hung up completely and is
> not listening on that port anymore (can other calls go through?) or the
> packet you see in tcpdump is not really going to the right port.  Can you
> confirm which one?
>
> Mike
>
> On Dec 16, 2009, at 6:29 PM, DJB wrote:
>
> We have a customer that we are sending calls to off the FS and here is the
> issue:
>
>
>
> Call is initially setup fine and they send a first re-invite with media
> 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first
> re-invite fine
>
>
>
> They then send a second re-invite with their media IP to cut through media
> and the FS sends a 200 OK to this fine. At this point the call is fine
>
>
>
> 30 minutes later they send a third re-invite because according to them it
> is strictly for the purpose of “keep alive” per RFC 4028. This third
> re-invite has the exact same media IP and UDP pot information as the second
> re-invite does. The problem is FS does not respond to this third re-invite
> AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the
> call to be dropped as the other end does not recieve a response from FS.
>
>
> One more thing, we did not see the third re-invite in sofia siptrace, but
> we do see it in ethereal, which is kind of odds.
>
>
> We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode.
>
>
>
> ___
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>


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Re: [Freeswitch-users] SIP Re-invite

[Anthony Minessale]

that means the invite is not matching the call dialog
compare the via tags and call-id etc


On Wed, Dec 16, 2009 at 5:29 PM, DJB  wrote:

> We have a customer that we are sending calls to off the FS and here is the
> issue:
>
>
>
> Call is initially setup fine and they send a first re-invite with media
> 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first
> re-invite fine
>
>
>
> They then send a second re-invite with their media IP to cut through media
> and the FS sends a 200 OK to this fine. At this point the call is fine
>
>
>
> 30 minutes later they send a third re-invite because according to them it
> is strictly for the purpose of “keep alive” per RFC 4028. This third
> re-invite has the exact same media IP and UDP pot information as the second
> re-invite does. The problem is FS does not respond to this third re-invite
> AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the
> call to be dropped as the other end does not recieve a response from FS.
>
>
> One more thing, we did not see the third re-invite in sofia siptrace, but
> we do see it in ethereal, which is kind of odds.
>
>
> We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode.
>
>
> Thank you very much.
>
>
> ___
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>


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Re: [Freeswitch-users] Where is that codec list coming from?

[Anthony Minessale]

np

On Wed, Dec 16, 2009 at 1:59 PM, Kristian Kielhofner <
kristian.kielhof...@gmail.com> wrote:

> Anthony,
>
>  As always, thanks.  I thought that might be it but I wanted to make sure.
>
> Thanks again!
>
> On Wed, Dec 16, 2009 at 2:48 PM, Anthony Minessale
>  wrote:
> > yah so the codec chosen by the inbound leg is always offered in the
> outbound
> > sdp to try and prevent transcoding.
> > if you set {absolute_codec_string=G722} in the bridge string you will
> bypass
> > this feature.
> >
>
> --
> 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] Where is that codec list coming from?

[Anthony Minessale]

yah so the codec chosen by the inbound leg is always offered in the outbound
sdp to try and prevent transcoding.
if you set {absolute_codec_string=G722} in the bridge string you will bypass
this feature.


On Wed, Dec 16, 2009 at 1:41 PM, Kristian Kielhofner <
kristian.kielhof...@gmail.com> wrote:

> Sure...  The call comes up as PCMU:
>
>   INVITE sip:5...@10.70.0.99  SIP/2.0
>   Call-ID: 80ea31a017f6de1d53e4a9c52f00
>   CSeq: 1 INVITE
>   From: sip:9413122...@smh.sip.local;tag=80ea31a017f6de1d43e4a9c52f00
>   Record-Route: ,
>   To: "5888" >
>   Via: SIP/2.0/UDP
> 10.70.0.65:5060;branch=z9hG4bK838383030303565656105e9.0,SIP/2.0/TCP
>
> 10.70.0.69;psrrposn=2;received=10.70.0.69;branch=z9hG4bK80ea31a017f6de1d63e4a9c52f00
>   Content-Length: 206
>   Content-Type: application/sdp
>   Contact: 
> ;transport=tcp>
>   Max-Forwards: 70
>   User-Agent: Avaya CM/R015x.02.0.947.3
>   Allow:
> INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH
>   Supported: timer,replaces,join,histinfo,100rel
>   Alert-Info: 
> >;avaya-cm-alert-type=external
>   Min-SE: 1200
>   Session-Expires: 1200;refresher=uac
>   P-Asserted-Identity: sip:9413122...@smh.sip.local
>   P-Charging-Vector: icid-value="AAS:13283-a031ea801def6179c4a3ed3f52"
>   History-Info: 
> >;index=1,"5888"
> >;index=1.1
>
>   v=0
>   o=- 1 1 IN IP4 10.70.0.69
>   s=-
>   c=IN IP4 10.70.0.22
>   b=AS:64
>   t=0 0
>   m=audio 2176 RTP/AVP 18 0 101
>   a=rtpmap:18 G729/8000
>   a=fmtp:18 annexb=no
>   a=rtpmap:0 PCMU/8000
>   a=rtpmap:101 telephone-event/8000
>
>  We don't support G729 so this call comes up as PCMU when we answer
> and then that codec is first in the codec list...
>
> On Wed, Dec 16, 2009 at 2:26 PM, Anthony Minessale
>  wrote:
> > can you do another trace to show the inbound invite too?
> >
>
> --
> 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] Where is that codec list coming from?

[Anthony Minessale]

can you do another trace to show the inbound invite too?

On Wed, Dec 16, 2009 at 1:13 PM, Kristian Kielhofner <
kristian.kielhof...@gmail.com> wrote:

> Hello everyone,
>
>  Pastebin here:
>
> http://pastebin.freeswitch.org/11525
>
>  I've got my pjsip profile configured for G722 only:
>
>
>
>  Yet whenever I send calls using that profile it (mysteriously)
> indicates support for PCMU in the INVITE.  The pastebin includes both
> the INVITE and "sofia status profile pjsip" to show that only G722 has
> been enabled.  Where is PCMU coming from?
>
> --
> 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] PlayAndGetDigits multiple WAV files

[Anthony Minessale]

make sure you are using latest trunk because it should not be sounding
choppy.


On Tue, Dec 15, 2009 at 4:22 PM, Dan Le  wrote:

> Or using mod file string: http://wiki.freeswitch.org/wiki/Mod_file_string
>
> <http://wiki.freeswitch.org/wiki/Mod_file_string>Dan
>
> On Tue, Dec 15, 2009 at 5:05 PM, Michael Jerris  wrote:
>
>> You can do that with phrase macros.
>>
>> Mike
>>
>> On Dec 15, 2009, at 2:56 PM, Malay Thakershi wrote:
>>
>>  Hello, I create one WAV file that has:
>>
>> Question + Option 1 + Option 2 + Option 3 + …
>>
>> I noticed towards end of the file Cepstral Allison starts chopping and
>> speeding up.
>>
>> So my question text that gets converted to WAV file using swift EXE looks
>> like:
>>
>> Which is the biggest mammal on land?
>> Select one of the following choices.> strength='weak'/>Or press star to skip the question
>> 1  Parrot
>>  2  Elephant
>>  3  T-Rex
>>  4  Blue Whale
>> 
>>
>> And my csharp code looks like:
>> pStrRetID = mObjMainSession.PlayAndGetDigits(1, 1, 3,
>> 5000, "*#",
>> 
>> @"C:\FreeSWITCH\sounds\en\us\chAsmt\Student_1222\Q1.WAV
>> ",
>>
>> @"C:\FreeSWITCH\sounds\en\us\chAsmt\static\error\invalid_choice.wav",
>> "^\\d", "");
>>
>>
>> What happens is, the voice just starts chopping and speeding up between
>> options. Even though I am not able to say that it only does that towards the
>> end, I think so.
>>
>> I thought, if I break each file into individual WAV instead of 1 big WAV,
>> it may help?
>>
>> Is there a way to play multiple (separate) WAV files in PlayAndGetDigits
>> function?
>>
>>
>>
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Re: [Freeswitch-users] Event Socket outbound in PHP

[Anthony Minessale]

there is a lua esl wrapper too the are all based on the same C code for the
ESL obj.
same rule applies get file number of stdin and pass to constructor.
I like perl the best for ESL but that's just me.


On Tue, Dec 15, 2009 at 7:38 AM, Dome Charoenyost  wrote:

>
>
> 2009/12/15 Anthony Minessale 
>
> you need to get the fd number of stdin however you do it in sdp and pass it
>> as the constructor to the esl obj
>>
>> It's work. Thanks. but i found PHP not good for this case. PHP need more
> resource. LUA look better.
> Now i'm testing by mod_lua but i plan to mover LUA work with outbound
> socket. but not found about lua outbounf socket in WIKI
>
>
> Best Regards.
>
> Dome C.
>
>
>
>
>>
>>
>> On Sun, Dec 13, 2009 at 8:41 PM, Dome Charoenyost  wrote:
>>
>>> Dear All,
>>>  Now i use php for ESL outbound. i get variable from stdin and
>>> process. (i use xinetd for handle socket)
>>>  $in = fopen("php://stdin", "r");
>>>  Problem is when i use read command  for get input from DTMF. i
>>> can't get variable. So now i use 2 php script. and use read appliction in
>>> XML DIalplan for solve this problem.
>>>  I plan to use php handle socket like a perl in
>>> http://svn.freeswitch.org/svn/freeswitch/trunk/libs/esl/perl/server2.pl
>>>  But i want to know how PHP work like this example ?
>>>
>>>  my $host = $new_sock->sockhost();
>>>
>>>  my $fd = fileno($new_sock);
>>>  my $con = new ESL::ESLconnection($fd);
>>>
>>>  my $info = $con->getInfo();
>>>
>>>
>>> Can someoue help me ?
>>>
>>>
>>> Best Regards.
>>>
>>> Dome C.
>>>
>>>
>>>
>>>
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>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
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>>
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>>
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>>
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Re: [Freeswitch-users] Event Socket outbound in PHP

[Anthony Minessale]

you need to get the fd number of stdin however you do it in sdp and pass it
as the constructor to the esl obj



On Sun, Dec 13, 2009 at 8:41 PM, Dome Charoenyost  wrote:

> Dear All,
>  Now i use php for ESL outbound. i get variable from stdin and
> process. (i use xinetd for handle socket)
>  $in = fopen("php://stdin", "r");
>  Problem is when i use read command  for get input from DTMF. i
> can't get variable. So now i use 2 php script. and use read appliction in
> XML DIalplan for solve this problem.
>  I plan to use php handle socket like a perl in
> http://svn.freeswitch.org/svn/freeswitch/trunk/libs/esl/perl/server2.pl
>  But i want to know how PHP work like this example ?
>
>  my $host = $new_sock->sockhost();
>
>  my $fd = fileno($new_sock);
>  my $con = new ESL::ESLconnection($fd);
>
>  my $info = $con->getInfo();
>
>
> Can someoue help me ?
>
>
> Best Regards.
>
> Dome C.
>
>
>
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Re: [Freeswitch-users] Link between User context and dialplan

[Anthony Minessale]

a dial plan is a another level of indirection on top of contexts it denotes
a specific module which implements the entire universe of dialing with room
for as many contexts as you have room for.  There is an XML dialplan, an
ENUM dialplan etc.  You can write your own dialplan and send calls to it.


On Sat, Dec 12, 2009 at 5:17 PM, Otis  wrote:

> Hi folks
>
> I am so sorry if this is such a basic thing.
>
> well, when a user/extension  eg  is created in with say a user
> context - SWAHILI-SPEAKERS  Please hear are my questions:
>
>   1. What dialplan will that user/extn use.
>   2. I guess I have to create a dialplan  Should the dial-plan also be
>  called WAHILI-SPEAKERS (is the case relevant ) ? Or  could it be
>  any name ?
>   3. And how does FS know to load that dialplan for that user.
>   4. Where should that xml file be stored ?
>   5. Is there a means of determinig which dialplan  was used for a call ?
>
> Thanks I think I have demonstrated enough thickness for now
>
>
>
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Re: [Freeswitch-users] Context vs. profile?

[Anthony Minessale]

Profile is a collection of preferences uses by conferences etc.
In the case of SIP a profile is also the name for the resulting SIP UA
created by a particular profile.

Context is a narrowed down view of something, in the case of the dialplan a
context is a set of extensions.  It's like having a dedicated set of
extensions per distinct context name like parallel universes.  both the foo
context and the bar context can have extension 2001.




On Mon, Dec 14, 2009 at 7:53 AM, Fred-145  wrote:

>
> Hello
>
> I'm a bit confused at the difference between those two concepts. Contexts
> are created in the /dialplan, and are refered to by items in /SIP_profiles
> and extensions in /directory.
>
> What purpose do contexts and profiles play?
>
> Thank you.
> --
> View this message in context:
> http://old.nabble.com/Context-vs.-profile--tp26778101p26778101.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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Re: [Freeswitch-users] What are the solutions for G729 support ?

[Anthony Minessale]

Software G729 will be available by the end of the month.
As for, G723 we are not currently working on it.


On Mon, Dec 14, 2009 at 6:45 AM, Oscav  wrote:

>
> Hi,
>
> What are the solutions to support the G729/G723 codec within FreeSwitch ?
>
> Thanks
>
>
> --
> View this message in context:
> http://old.nabble.com/What-are-the-solutions-for-G729-support---tp26777181p26777181.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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Re: [Freeswitch-users] Java ESL

[Anthony Minessale]

In the libs/esl there is already swigged java but I don't know how to load
it etc try make javamod in esl

On Dec 13, 2009 12:47 PM, "Niall Crosby"  wrote:


Where can I find the Java swig ESL?

I like Java so am happy to put time towards generating a pure Java ECL,
however I haven't programmed C in 10+ years so feel like swig would be to
much in the deep end for me.


2009/12/13 Anthony Minessale 

> > The swig java one is almost done we need someone who likes java to
finish it but as you can see...



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The information transmitted is intended only for the person or entity to
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Re: [Freeswitch-users] Sofia performance

[Anthony Minessale]

Sep processes does better than sep profiles.  We need to push the sofia devs
to work on a better concurrancy scheme but they are too busy with other
nokia duties these days so were stuck with what we got for now.  About
400cps on a good day

On Dec 13, 2009 4:05 PM, "Jay Binks"  wrote:

I'm interested in what the upper limit would be,  when expecting a
performance improvement with sofia profiles.

For example let's say I were to direct connect to customers ( layer 2 ) with
a .1q trunk coming in to fs and a Sofia profile for each customer.   Am I
going to hit a bottleneck at 20,50,100,500 ???

Guess it's hardware limited ,  but any thoughts ?

J

On 14/12/2009, at 4:36, Anthony Minessale 
wrote: > Here is my standa...

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Re: [Freeswitch-users] Sofia performance

[Anthony Minessale]

Here is my standard asvice on people worrying about performance before even
trying fs.

Rule of thumb, if you have ever used asterisk, multiply everything by 10 so
if there is a performance concern assume it will not arise unless you get at
least 10 asterisks worth of performance first.

People do thosands of channels with media and tens of thousands with no
media, try it first before freting about imaginary load concerns.

On Dec 13, 2009 12:28 PM, "Yehavi Bourvine" 
wrote:

We are still on  a small "proof of concept" system, but I am looking at the
future...

  Thanks, __Yehavi:

2009/12/13 Frank Carmickle 

> > On Sun, Dec 13, Yehavi Bourvine wrote: > > I would like all phones have
the same general config...

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Re: [Freeswitch-users] Java ESL

[Anthony Minessale]

The swig java one is almost done we need someone who likes java to finish it
but as you can see most java ppl seem to always want to do it "their way"

On Dec 13, 2009 9:56 AM, "Niall Crosby"  wrote:


Pure Java is my preference - am looking to build apps that are portable.

N.

2009/12/13 João Mesquita 

> > Can't we just swig it to Java? > > JM > > On Sun, Dec 13, 2009 at 1:05
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Re: [Freeswitch-users] bridge doesn't respect bypass_media=true over the socket

[Anthony Minessale]

Hey,

You can't set bypass_media=true in {} or it will not take effect unless that
b leg itself becomes an a leg some day.
you need to execute set on bypass_media=true on the leg before you call
bridge to trigger it.

Alternatively you could set {bypass_media_after_bridge=true} or set it on A
leg as described above on either leg and it will do the bypass once the
audio is flowing.


On Fri, Dec 11, 2009 at 11:14 AM, Kristian Kielhofner <
kristian.kielhof...@gmail.com> wrote:

> Hello everyone,
>
>  PB here:
>
> http://pastebin.freeswitch.org/11482
>
> FS rev 15909.  The relevant bits from the log are here (starting
> around line 135):
>
> #
> 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr.c:548
> sofia/pjsip/nob...@192.168.4.253 Command Execute
>
> bridge({originate_timeout=30,bypass_media=true,origination_caller_id_number=,origination_caller_id_name=Tara
> Ousley}sofia/voalte/hut...@192.168.4.17)
> #
> EXECUTE sofia/pjsip/nob...@192.168.4.253
>
> bridge({originate_timeout=30,bypass_media=true,origination_caller_id_number=,origination_caller_id_name=Tara
> Ousley}sofia/voalte/hut...@192.168.4.17)
> #
> 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735
> variable string 0 = [originate_timeout=30]
> #
> 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735
> variable string 1 = [bypass_media=true]
> #
> 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735
> variable string 2 = [origination_caller_id_number=]
> #
> 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735
> variable string 3 = [origination_caller_id_name=Tara Ousley]
>
> bypass_media=true yet the SDP of the outgoing INVITE looks like this:
>
> #
> send 1032 bytes to udp/[192.168.4.17]:5060 at 12:06:12.876994:
> #
>   
> #
>   INVITE sip:hut...@192.168.4.17  SIP/2.0
> #
>   Via: SIP/2.0/UDP 192.168.2.10:5062;rport;branch=z9hG4bK1K6mc3NcmmaNr
> #
>   Max-Forwards: 69
> #
>   From: "Tara Ousley" 
> >;tag=0tU8SN9pvejNK
> #
>   To: >
> #
>   Call-ID: 802f4045-4215-42a2-91a6-ff9cf18b1aa8
> #
>   CSeq: 124148250 INVITE
> #
>   Contact: 
> #
>   User-Agent: Voalte Voice
> #
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> #
>   Supported: timer, precondition, path, replaces
> #
>   Allow-Events: talk, refer
> #
>   Content-Type: application/sdp
> #
>   Content-Disposition: session
> #
>   Content-Length: 271
> #
>   X-voalte-call-id: 898ef33c-50f2-487c-9e8d-8c6fcee15ab8
> #
>   Remote-Party-ID: "Tara Ousley"
> 
> >;party=calling;screen=yes;privacy=off
> #
>
> #
>   v=0
> #
>   o=FreeSWITCH 1260508000 1260508001 IN IP4 192.168.2.10
> #
>   s=FreeSWITCH
> #
>   c=IN IP4 192.168.2.10
> #
>   t=0 0
> #
>   m=audio 25172 RTP/AVP 9 0 101
> #
>   a=rtpmap:9 G722/8000
> #
>   a=rtpmap:0 PCMU/8000
> #
>   a=rtpmap:101 telephone-event/8000
> #
>   a=fmtp:101 0-16
> #
>   a=silenceSupp:off - - - -
> #
>   a=ptime:20
>
> 192.168.2.10 is the address of my FS box...
>
> --
> 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] Passing user variables to mod_voicemail

[Anthony Minessale]

That wont work.
I'm not sure if there is a way, I cant think of one off the top of my head.


On Thu, Dec 10, 2009 at 10:10 PM, Mark Campbell-Smith <
mcampbellsm...@gmail.com> wrote:

> Hi!
>
> My voip provider provides a SOAP interface to be able to send SMS's,
> so after a voicemail is left, I want to execute a 'send sms' script.
> I don't want a separate statement in the dialplan after the voicemail
> statement because I only want to send sms's when a voicemail is
> actually left.
>
> The way I was going to do this was to modify the mailer-app to point
> to a shell script and modify the mailer-app-args to include some user
> defined variables (in conf/directory/default/*.xml).
>
> value="/usr/local/freeswitch/scripts/emailvm.sh"/>
>
>
> The shell script would do the following:
>
> emailvm.sh
>
> #$1 $2 $3 = smsaccount smspassword textmessage
> tee /tmp/vmmail | /usr/sbin/sendmail -t
> exec /usr/local/freeswitch/scripts/sendsms.pl $1 $2 $3
> #echo $1 $2 $3 $4 $5 $6 >> /usr/local/freeswitch/scripts/log.log
>
> However, if I uncomment the last line, I never see the user variables
> being passed to the shell script.  The email is sucessfully sent, but
> the sms script doesnt work.  If fact, the output of log.log is (for
> example):
>
> -f 1...@192.168.1.120 email_addr...@domain.com
>
> Any ideas if it is possible to pass user variables via mod_voicemail
> in this way?
>
> Thanks
>
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Re: [Freeswitch-users] Invite local number into a conference - codec problem

[Anthony Minessale]

set absolute_codec_string to whatever codec you want to offer in the {} on
the bridge string


On Thu, Dec 10, 2009 at 8:26 AM, Peter P GMX  wrote:

> Hello,
>
> I try to invite a user into a conference by
> loopback/255 8000 Conference
> 255 is the user, I invite the user via loopback as that way I can also
> invite external numbers.
>
> It processes the user's local dialplan correctly (as if the user was
> normally dialled), however it only offers L16 codec, so the Phone fails.
> I can see no codec negociation on the debug console.
> If I call the phone from another phone, then codec negociation is taking
> place.
> If I invite an external PSTN user into the conference then codecs are
> set correctly (L16+PCMA+PCMU etc)
>
> Is there a way to explicitely set the codec for the conference?
>
>  is not set is,
> still commented in the internal profile.
> In vars.conf.xml only only PCMA and PCMU are set.
>
> Best regards
> Peter
>
>
>
>
>
>
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Re: [Freeswitch-users] FS Rocks!!!!!!!!!

[Anthony Minessale]

Don't worry.
I was an asterisk developer/volunteer in 2003.  I still managed to figure it
out. ;)


On Thu, Dec 10, 2009 at 4:13 AM, Julian Lyndon-Smith wrote:

> Sometime next week I hopefully am going to start a document that
> follows my progress in setting up a FS system from scratch, with all
> the pitfalls and successes. A kinds of "warts and all" story.
> Alongside this "blog" (for want of a better word) I will also then
> document the steps needed to get it working (a howto guide,
> effectively).
>
> I am a long time * user (2004), so my mindset is kind of skewed - but
> perhaps that would be beneficial for other * users looking at
> implementing FS.
>
> Most of our config and dialplan is generated by using res_config_curl,
> and we use things like call listening, conferencing, parking and
> queues. We do use queues in a slightly odd manner (we add 1 agent, and
> call a local channel). When this channels is called, we use curl to
> get our application to return the most appropriate agent to actually
> call).
>
> We also use * as a power dialler, making upwards of 400,000 call
> attempts per month. Not massive, but not tiny either.
>
> Hopefully, this will be of use to both FS and * users. What would be
> great is that if other people follow my progress, and make suggestions
> as and when I hit a brick wall :)
>
> What would be best for this ? A blog ? Or a wiki page ?
>
> Julian
>
> 2009/12/9 Brian West :
> > That is what is nice about our community I'm more than willing to answer
> the
> > questions if you document them... as are many others in the core
> team...we
> > just have a lot to do and I think the best repayment is documentation! ;)
> > /b
> > On Dec 9, 2009, at 4:39 PM, Tim Uckun wrote:
> >
> > On Thu, Dec 10, 2009 at 11:07 AM, Brian West 
> wrote:
> >
> > Visit the friday meetings and we can help if you document it.  ;)
> >
> >
> > I would be willing to lend a hand with the documentation but I know so
> > little (a complete freeswitch noob). For example I was trying to
> > figure out how to tell if an extension was set up "show dialplan in
> > asterisk".  I could not find this anywhere. If I find out I would be
> > happy to add it to the rosetta stone.
> >
> > I am currently working on getting outbound socket working. Once I get
> > it going I would be happy to add it to the relevant section of the
> > wiki (in this case ruby).
> >
> >
> > ___
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> >
> >
>
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Re: [Freeswitch-users] Even socket question.

[Anthony Minessale]

the dialplan is dynamic there is no such thing
you have to look in your dialplan xml files because it's served up live.
FS has a different paradigm than asterisk.


On Wed, Dec 9, 2009 at 8:00 PM, Tim Uckun  wrote:

> On Thu, Dec 10, 2009 at 2:46 PM, Anthony Minessale
>  wrote:
> > do you have something listening on 8084 ?
> >
>
> Yes.
>
> I figured out the problem. There was already an extension called 8084
> and it overwrote the extension I defined.
>
> Which brings me back to a question I had earlier.
>
> Where is the equivalent of the "show dialplan" command? How can I list
> all the extensions and their definitions?
>
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Re: [Freeswitch-users] Even socket question.

[Anthony Minessale]

hine.c:348
> (sofia/internal/1...@192.168.56.3) State EXECUTE going to sleep
> 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/1...@192.168.56.3) Running State Change CS_HANGUP
> 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:465
> sofia/internal/1...@192.168.56.3 handler already called, skipping
> state handler.
> 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/1...@192.168.56.3) State Change CS_HANGUP ->
> CS_REPORTING
> 2009-12-09 14:31:54.430526 [DEBUG] switch_core_session.c:999 Send
> signal sofia/internal/1...@192.168.56.3 [BREAK]
> 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/1...@192.168.56.3) Running State Change CS_REPORTING
> 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:579
> (sofia/internal/1...@192.168.56.3) State REPORTING
> 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/1...@192.168.56.3 Standard REPORTING, cause:
> NORMAL_CLEARING
> 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:579
> (sofia/internal/1...@192.168.56.3) State REPORTING going to sleep
> 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:327
> (sofia/internal/1...@192.168.56.3) State Change CS_REPORTING ->
> CS_DESTROY
> 2009-12-09 14:31:54.430526 [DEBUG] switch_core_session.c:999 Send
> signal sofia/internal/1...@192.168.56.3 [BREAK]
> 2009-12-09 14:31:54.430526 [DEBUG] switch_core_session.c:1136 Session
> 6 (sofia/internal/1...@192.168.56.3) Locked, Waiting on external
> entities
> 2009-12-09 14:31:54.430526 [NOTICE] switch_core_session.c:1154 Session
> 6 (sofia/internal/1...@192.168.56.3) Ended
> 2009-12-09 14:31:54.430526 [NOTICE] switch_core_session.c:1156 Close
> Channel sofia/internal/1...@192.168.56.3 [CS_DESTROY]
> 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:423
> (sofia/internal/1...@192.168.56.3) Running State Change CS_DESTROY
> 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:434
> (sofia/internal/1...@192.168.56.3) State DESTROY
> 2009-12-09 14:31:54.430526 [DEBUG] mod_sofia.c:293
> sofia/internal/1...@192.168.56.3 SOFIA DESTROY
> 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/1...@192.168.56.3 Standard DESTROY
> 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:434
> (sofia/internal/1...@192.168.56.3) State DESTROY going to sleep
>
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Re: [Freeswitch-users] FXO PCI card with Pfsense Freeswitch.

[Anthony Minessale]

I dont think there are any supported hw for bsd, there are legacy sangoma
and zaptel drivers floating around but they are not supported by the
vendors.


On Tue, Dec 8, 2009 at 3:25 PM, Orien Love  wrote:

> I am looking for a 4 port FXO card to use with my PfSense installation
> of freeswitch. does anybody know if the
> Asterisk Trixbox 4 FXO Voip Digium TDM410 will work on pfsense?
> or could somebody recommend one that would.
>
> Thank You
>   Orien
>
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Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP

[Anthony Minessale]

are you using more than one profile here?
if so you have to repeat the siptrace on command for each one.

This trace makes little sense to me because I think half of it is missing.
but you can see several packets coming in like 20 times each which means you
have some kind of nat or network problem causing the other end of this call
to send retries on all the packets.



On Tue, Dec 8, 2009 at 2:57 PM, Jerry Richards
wrote:

>  Here is the Pastebin Link: http://pastebin.freeswitch.org/11432
>
> Thanks,
> Jerry
>
>  --
> *From:* Jerry Richards [mailto:jerry.richa...@teotech.com]
> *Sent:* Tuesday, December 08, 2009 12:35 PM
>
> *To:* 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org'
> *Subject:* RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION
> UNREACHABLE When Gateway Sends RTP
>
>  Anthony and Michael,
>
> I downloaded the latest trunk, rebuilt it, and re-ran the test with the
> logs that Anthony told me to turn on.  I put the results up in the PasteBin.
>
> Best Regards,
> Jerry
>
>  --
> *From:* Jerry Richards [mailto:jerry.richa...@teotech.com]
> *Sent:* Monday, December 07, 2009 10:49 AM
> *To:* 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org'
> *Subject:* RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION
> UNREACHABLE When Gateway Sends RTP
>
>  When I got the latest trunk the make gets an error.  Should I perhaps
> disable the mod_amr?
>
> making all mod_amr
> make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'.
> Stop
>
> The method I used to get the latest trunk follows:
>
> svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch
>
> Best Regards,
> Jerry
>
>  --
> *From:* Jerry Richards [mailto:jerry.richa...@teotech.com]
> *Sent:* Monday, December 07, 2009 7:44 AM
> *To:* 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org'
> *Subject:* RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION
> UNREACHABLE When Gateway Sends RTP
>
>  I am changing the 3pcc setting because one of my gateways sends INVITEs
> without SDP.  I will try to update to the latest trunk today and capture
> traces as Anthony described.  If I can't do it today, it might be at the end
> of the week.
>
> Best Regards,
> Jerry
>
>
>  --
> *From:* Michael Jerris [mailto:m...@jerris.com]
> *Sent:* Saturday, December 05, 2009 7:30 PM
> *To:* Jerry Richards
> *Subject:* Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION
> UNREACHABLE When Gateway Sends RTP
>
> Jerry-
>
> Any update on this?
>
> Mike
>
>  On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote:
>
> Why are you changing the 3pcc setting, is this an invite with no sdp?
> you need to take a trace from FS.
>
> 1) update to latest trunk first so line number match up.
> 2) issue these commands
>
> sofia profile internal siptrace on
> console loglevel debug
>
> save the output and put it on pastebin http://pastebin.freeswitch.org
>
>
>
>
> On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards  > wrote:
>
>>
>> I have  Mediant 1000 gateway, and for some reason, when I make an outbound
>> call, FS enters the CS_CONSUME_MEDIA state and never connects the call.  A
>> Wireshark trace shows that FS is replying to the gateway's inbound RTP
>> packets with ICMP DESTINATION UNREACHABLE.  But the gateway is sending RTP
>> packets to the same port that FS specified in the outbound INVITE.  It
>> appears in the log that FS is discarding the 200 OK from the gateway.
>>
>> I disabled the Firewall and SELinux on the Freeswitch machine.  I tried
>> changing  to "true" and also "proxy", but it has no effect.
>>
>> Anyone know what could be the issue?  I posted the Freeswitch log in the
>> pastebin.
>>
>> Best Regards,
>> Jerry
>>
>>
>> ___
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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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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 Conf

Re: [Freeswitch-users] FreeSWITCH 1.0.5 is (almost) here!

[Anthony Minessale]

Let's see if we can beat Duke Nukem Forever!


On Tue, Dec 8, 2009 at 12:19 PM, Michael Collins  wrote:

> Greetings,
>
> The FreeSWITCH developers have uploaded the latest and greatest FreeSWITCH
> 1.0.5 pre-release version. Please check out the release 
> announcement<http://www.freeswitch.org/node/220>.
> Let's all get updated as soon as possible. Also, please report bugs right
> away and follow up when the developers need further information. We have had
> to close out some bugs due to lack of information from the one reporting.
>
> Of course, those running SVN trunk are asked to do a "make current" as soon
> as reasonably possible. The devs love it when you are on the latest trunk.
> :)
>
> Thanks again for all of your help! Let's keep up the good work and we'll
> have 1.0.5 available in no time.
>
> -Michael
>
> ___
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>


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Re: [Freeswitch-users] 100% CPU Usage when Asterisk 1.6.0.19 Registers

[Anthony Minessale]

would not be able to even guess without some data to examine.


On Tue, Dec 8, 2009 at 12:14 PM, Spencer Thomason <
spen...@5ninesolutions.com> wrote:

> Hmm.. It doesn't seem to be a problem with Asterisk < 1.6.0.13.  Asterisk
> 1.6.0.15-18 doesn't work because of Asterisk bugs and I only noticed this
> after an upgrade to 1.6.0.19.  We're using xen on all our machines with
> 250hz timers.  Could that be a problem?  When I get a change I'll try to
> recreate this with a few more virtual machines to try to debug it.
>
> Spencer
>
> On Dec 8, 2009, at 8:22 AM, Anthony Minessale wrote:
>
> We could check it out for you if you want to contact me and give me ssh
> access.
> Or I can provide the instructions
>
> get it into the 100% cpu usage state then do the following without stopping
> FS.
>
> 1) run top -H and sort so all the FS threads are at the top and screen cap
> it so we can see which thread id is using the most cpu.
> 2) make sure you have gdb installed and issue this command from the build
> root
> ./support-d/fscore_pb gcore cpu_race_issue
>
> then we can compare the thread using the most cpu with the trace and locate
> your problem.
>
>
>
> On Mon, Dec 7, 2009 at 3:34 PM, Spencer Thomason <
> spen...@5ninesolutions.com> wrote:
>
>> Hi I'm trying to use Freeswitch as a SBC for a few Asterisk boxes.
>> The Asterisk boxes are individual hosted PBXs but they are configured
>> with identical software.  This a x86_64 CentOS 5.4 system.  I've tried
>> 1.0.4 and the latest svn with the same results.  Basically Freeswitch
>> registers with outbound providers and I can send and receive test
>> calls.  Then without warning, i.e. the Asterisk boxes are all idle and
>> there are no calls, the Freeswitch process starts using 100% of the cpu.
>>
>>
>>
>> ___
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>> FreeSWITCH-users@lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> 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
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> iax:gu...@conference.freeswitch.org/888
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
>
>
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>
>


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

[Anthony Minessale]

One last bit of free consulting advice for you:

You are again being rude because you want us to work for you for free.
The code is free sir, the support here is voluntary and based on our
willingness to help and comments like that are all it takes to get us to
ignore you completely.


On Tue, Dec 8, 2009 at 5:46 AM, Jon Bruel  wrote:

>  I got the combination Lua with direct access to the core Sqlite database
> to work. Hurray, maybe I’m not as stupid as A.M II hints…
>
> The problem was that Lua did not “like”:
>
>
>
> require "luasql.sqlite"
>
> env = luasql.sqlite()
>
> con = assert(env:connect("/usr/local/freeswitch/db/core.db"))
>
>
>
> After changing it to
>
>
>
> require "luasql.sqlite3"
>
> env = luasql.sqlite3()
>
> con = assert(env:connect("/usr/local/freeswitch/db/core.db"))
>
>
>
> And seeing that there was a symlink in one of the right directories called
> with the name: sqlite3.so, it worked.
>
>
>
> Changing the core db into a MySQL via ODBC caused some problems even after
> it seemed to work. For instance, console help caused an error with an error
> description indicating that a SQL SELECT query including the reserved word
> key has been fired.
>
>
>
> It this problem likely to be solved if I used another version of the MySQL?
>
>
>
> *Jon Brüel*
> Consiglia Telecommunications
>
> DK-2960 Rungsted Kyst
> Tel: +45 45 16 1000
> Mob: +45 26 15 30 60
>
> CVR: 27047882
>
>
>
>
>
> ___
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>
>


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Re: [Freeswitch-users] 100% CPU Usage when Asterisk 1.6.0.19 Registers

[Anthony Minessale]

We could check it out for you if you want to contact me and give me ssh
access.
Or I can provide the instructions

get it into the 100% cpu usage state then do the following without stopping
FS.

1) run top -H and sort so all the FS threads are at the top and screen cap
it so we can see which thread id is using the most cpu.
2) make sure you have gdb installed and issue this command from the build
root
./support-d/fscore_pb gcore cpu_race_issue

then we can compare the thread using the most cpu with the trace and locate
your problem.



On Mon, Dec 7, 2009 at 3:34 PM, Spencer Thomason  wrote:

> Hi I'm trying to use Freeswitch as a SBC for a few Asterisk boxes.
> The Asterisk boxes are individual hosted PBXs but they are configured
> with identical software.  This a x86_64 CentOS 5.4 system.  I've tried
> 1.0.4 and the latest svn with the same results.  Basically Freeswitch
> registers with outbound providers and I can send and receive test
> calls.  Then without warning, i.e. the Asterisk boxes are all idle and
> there are no calls, the Freeswitch process starts using 100% of the cpu.
>
>
>
> ___
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> FreeSWITCH-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
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Re: [Freeswitch-users] Zombie Records in core db

[Anthony Minessale]

For starters, try using the latest svn snapshot.  Your version is 6 months
old and several thousand revs old.

On Dec 7, 2009 8:34 PM, "DJB"  wrote:

We have FreeSWITCH Version 1.0.4 (exported) running at a high volume
traffic.  I normally check the concurrent calls by looking at the number of
sessions from status command.  However, the number of concurrent calls in FS
is normally higher than it's supposed to be after we ran traffic for about a
week.  Thus, I routed the traffic away from the FS and found out from "show
calls" that there were so many old calls from previous days.  We are running
a pass-thru traffic in signaling only.  I wonder whether there is a way to
have those "zombied" records clean up automatically.  Also, what should I do
to prevent this problem?

Thank you,
Dorn B.




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Re: [Freeswitch-users] Trapping dtmf on bridged call

[Anthony Minessale]

did you set the inputcallback too?


On Mon, Dec 7, 2009 at 4:59 PM, Nik Middleton <
nik.middle...@noblesolutions.co.uk> wrote:

>  Can this be done in an lua script?
>
>
>
> Regards,
>
>
>  --
>
> *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
> freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* 07 December 2009 22:18
>
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Trapping dtmf on bridged call
>
>
>
>
>
> On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton <
> nik.middle...@noblesolutions.co.uk> wrote:
>
> Hi
>
>
>
> Is it possible to trap on DTMF on a bridged call within an LUA script?
> I’ve tried setting the gateway to use inband, but no joy.  It looks like I
> could use start_dtmf, but I can’t see how to launch this within LUA
>
> Perhaps you could use bind-meta-app to bind a key combo like *1 to whatever
> you want to have happen. Check it out:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app
>
> The Local_Extension in the default.xml dialplan file has a few examples of
> using this tool.
> -MC
>
>
>
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>


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Re: [Freeswitch-users] Trapping dtmf on bridged call

[Anthony Minessale]

session:execute("start_dtmf");

this app captures inband audio tone dtmf and interprets them aka calls your
callback etc.


On Mon, Dec 7, 2009 at 4:02 PM, Nik Middleton <
nik.middle...@noblesolutions.co.uk> wrote:

>  Hi
>
>
>
> Is it possible to trap on DTMF on a bridged call within an LUA script?
> I’ve tried setting the gateway to use inband, but no joy.  It looks like I
> could use start_dtmf, but I can’t see how to launch this within LUA
>
>
>
> Regards,
>
> ___
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>


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Re: [Freeswitch-users] Choppy sound with PCMU

[Anthony Minessale]

Both,
if it always sounds ok then I guess CPU usage.



On Mon, Dec 7, 2009 at 2:58 PM, eaf  wrote:

>
> What do you want me to check while running these tests? Sound quality (it's
> good now even with original 1.0.4). Or CPU utilization?
>
> It's Debian 4.
>
>
> Anthony Minessale-2 wrote:
> >
> > Did you do each thing alone too to tell the difference?
> > -hp alone, disable monotonic alone (i did not see you mention the disable
> > monotonic)
> >
> > as for your 4ms thing, yes we require high resolution timing, if we ask
> to
> > sleep 1000 microseconds that is what we need it to sleep for or at least
> > as
> > close as possible, and the main reason that thread is never sleeping is
> > because you can't actually count on it to run every 1ms but you mostly
> > can.
> > Hence the whole philosophy on only making 1 thread run hot all the time
> to
> > ensure that the rest don't have to repeat the same algorithm.  We focus
> on
> > high end performance this was the point of your experimentation because
> we
> > will need to use a compile time defines and other logic to make it more
> > efficient on your platform, a platform which we are not using.  I am
> > curious
> > what would happen if you install Kristian's astlinux on one of your
> > devices,
> > i think you should also compare the kernel versions.
> >
> >
> > What OS are you running anyway?
> >
> > Here are some more things to try (running plain trunk with no mods) do
> > these
> > systematically each alone and all together with/without -hp or disable
> > monotonic etc to see what different combos create
> >
> > comment out this line (line 10)
> > #define DISABLE_1MS_COND
> >
> > rebuild, this tells it to run a conditional at 1ms in the same timer
> > thread
> > which will make all the switch_cond_next share a 1ms conditional instead
> > of
> > doing microsleeps
> >
> > next
> >
> > some kernels/devices work better using select(0) for sleep where others
> > work
> > better using usleep.
> > comment out line 109
> > apr_sleep(t);
> >
> > and try
> > usleep(t)
> >
> > also mac works better using nanosleep so you could try changing it so it
> > uses the code starting at 101 instead.
> >
> >
> > also your claim about JS should be investigated because I do not think it
> > should be the case.
> > but you may want to move this to a jira http://jira.freeswitch.org
> >
> > As for the asterisk comparison,
> > not sure how to answer you, that's your decision.
> >
> >
> >
> > On Mon, Dec 7, 2009 at 9:28 AM, eaf  wrote:
> >
> >>
> >> Here is what I found...
> >>
> >> I tried high-priority scheduling as per your suggestion, reniced the
> >> program
> >> explicitly, rewrote timer thread to sleep on cond. variable and activate
> >> only when there are timers and only when the timer actually had to be
> >> clicked, turned off SQL thread and removed polling from sofia profile
> >> thread.
> >>
> >> That pretty much eliminated all idle 1ms sleepers that were there except
> >> for
> >> three in sofia itself (su_epoll_port). And when I was about to be happy,
> >> I
> >> found that two outgoing calls through my VOIP providers when bridged
> >> together showed terrible distortions. I undid all my changes, tried
> >> 1.0.4,
> >> trunk (noticed btw that when I bridge two calls via loopback in JS in
> the
> >> trunk I must keep JS running, or the calls get terminated - NOT the same
> >> as
> >> in 1.0.4 where exitting JS left calls running), got pretty much the same
> >> sad
> >> results. At the same time calls bridged by freeswitch between LAN and
> any
> >> of
> >> the VOIP providers behaved just fine. And calls bridged by Asterisk any
> >> way
> >> were fine too. So that pretty much looked like the end of the freeswitch
> >> trials for me.
> >>
> >> But then I timed your code, mine and found that all those 1ms sleeps
> that
> >> your timer thread was doing (and all those pollers were doing as well)
> >> were
> >> actually 4ms sleeps because you know what unless kernel is configured
> >> with
> >> HZ=1000, you can't sleep for less than 4ms (HZ=250) or perhaps even 10ms
> >> (HZ=100). Mine was 250.
> >>
> >> This actually meant that the original timer thread was firing once,
> >

Re: [Freeswitch-users] esl for Mac OS X 10.4

[Anthony Minessale]

also,
don't use 1.0.4, please us the latest SVN or last svn snapshot at the very
least.


On Mon, Dec 7, 2009 at 12:34 PM, Kendall Stauffer  wrote:

>  Any direction on where to start would be appreciated. I am trying to get
> freepbx working with this, and everything works (I think) except esl
>
>
>
> *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
> freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Brian West
> *Sent:* Monday, December 07, 2009 1:10 PM
>
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] esl for Mac OS X 10.4
>
>
>
> The build system for libesl and everything below that won't work 100% on
> the mac just yet.  You have to make some changes to how its linked and
> you'll have to compile php yourself to get everything in there properly.
>  The perl one however is much easier to fix.
>
>
>
> -SOLINK=-shared -Xlinker -x
>
> +SOLINK=-dynamiclib -Xlinker -x
>
>
>
>
>
> Thats all you usually fix for the mac.
>
>
>
>
>
> /b
>
>
>
>
>
>
>
> On Dec 7, 2009, at 11:27 AM, Kendall Stauffer wrote:
>
>
>
> I have downloaded and compiled freeswitch, and it runs fine, can
> compile everything without error including spandsp, but can’t get esl to
> compile.  My version is earlier than the snow leopard that is mentioned in
> the general install docs,  and I have tried it with and without the compiler
> flags in the freewswtch installation -> MAC os X.
>
>   I have also googled this, and don’t see what I am doing wrong. Anybody
> there that can help?
>
> applesrv:/usr/src/freeswitch-1.0.4/libs/esl root# make
> phpmod-install
>
> make MYLIB="../libesl.a" SOLINK="-Xlinker -x"
> CFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g
> -ggdb -I../libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror
> -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes"
> CXXFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE
> -g -ggdb -I../libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable"
> CXX_CFLAGS="" -C php
>
> g++ -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/mysql -liodbc
> -lmysqlclient -lldap -llber -lcurl -lm -lcurl -liodbc -o ESL.so -L.
>
> /usr/libexec/gcc/powerpc-apple-darwin8/4.0.1/ld: Undefined symbols:
>
> _main
>
> __convert_to_string
>
> __efree
>
> __emalloc
>
> __estrndup
>
> __zend_get_parameters_array_ex
>
> __zend_list_find
>
> __zval_copy_ctor
>
> _compiler_globals
>
> _convert_to_long
>
> _zend_error
>
> _zend_get_constant
>
> _zend_hash_find
>
> _zend_register_list_destructors_ex
>
> _zend_register_long_constant
>
> _zend_register_resource
>
> _zend_rsrc_list_get_rsrc_type
>
> _zend_wrong_param_count
>
> collect2: ld returned 1 exit status
>
> make[1]: *** [ESL.so] Error 1
>
> make: *** [phpmod] Error 2
>
> ___
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
>
>
> ___
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
>


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Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP

[Anthony Minessale]

try rerunning the ./bootstrap.sh


On Mon, Dec 7, 2009 at 12:49 PM, Jerry Richards
wrote:

>  When I got the latest trunk the make gets an error.  Should I perhaps
> disable the mod_amr?
>
> making all mod_amr
> make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'.
> Stop
>
> The method I used to get the latest trunk follows:
>
> svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch
>
> Best Regards,
> Jerry
>
>  --
> *From:* Jerry Richards [mailto:jerry.richa...@teotech.com]
> *Sent:* Monday, December 07, 2009 7:44 AM
> *To:* 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org'
> *Subject:* RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION
> UNREACHABLE When Gateway Sends RTP
>
>  I am changing the 3pcc setting because one of my gateways sends INVITEs
> without SDP.  I will try to update to the latest trunk today and capture
> traces as Anthony described.  If I can't do it today, it might be at the end
> of the week.
>
> Best Regards,
> Jerry
>
>
>  --
> *From:* Michael Jerris [mailto:m...@jerris.com]
> *Sent:* Saturday, December 05, 2009 7:30 PM
> *To:* Jerry Richards
> *Subject:* Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION
> UNREACHABLE When Gateway Sends RTP
>
> Jerry-
>
> Any update on this?
>
> Mike
>
>  On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote:
>
> Why are you changing the 3pcc setting, is this an invite with no sdp?
> you need to take a trace from FS.
>
> 1) update to latest trunk first so line number match up.
> 2) issue these commands
>
> sofia profile internal siptrace on
> console loglevel debug
>
> save the output and put it on pastebin http://pastebin.freeswitch.org
>
>
>
>
> On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards  > wrote:
>
>>
>> I have  Mediant 1000 gateway, and for some reason, when I make an outbound
>> call, FS enters the CS_CONSUME_MEDIA state and never connects the call.  A
>> Wireshark trace shows that FS is replying to the gateway's inbound RTP
>> packets with ICMP DESTINATION UNREACHABLE.  But the gateway is sending RTP
>> packets to the same port that FS specified in the outbound INVITE.  It
>> appears in the log that FS is discarding the 200 OK from the gateway.
>>
>> I disabled the Firewall and SELinux on the Freeswitch machine.  I tried
>> changing  to "true" and also "proxy", but it has no effect.
>>
>> Anyone know what could be the issue?  I posted the Freeswitch log in the
>> pastebin.
>>
>> Best Regards,
>> Jerry
>>
>>
>> ___
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>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
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> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
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>
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>
>
>
> ___
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>
>


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Re: [Freeswitch-users] lua+sqlite example?

[Anthony Minessale]

yes if you use the lua odbc sql plugin you should be able to use that for
sqlite, they may also have a native one.


On Sat, Dec 5, 2009 at 9:21 PM, Steve Klein  wrote:

>  Greetings. We are attempting to add sqlite access to an IVR application
> we are prototyping. We are using lua for the scripts. Is there an example
> anywhere of a lua + sqlite script? Do we need to install luasql? Any
> help/pointers greatly appreciated.
>
>
>
> --Steve Klein
>
>
>
> ___
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>


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Re: [Freeswitch-users] Choppy sound with PCMU

[Anthony Minessale]

oh and also

use top -H to see which threads are using specific CPU and try to cross
reference them by attaching with gdb and dumping all the thread bt


On Mon, Dec 7, 2009 at 10:16 AM, Michael Jerris  wrote:

> Also I have seen some people reporting that the new tickless timers in
> newer kernels work better.  You may want to try those.
>
> Mike
>
> On Dec 7, 2009, at 11:00 AM, Anthony Minessale wrote:
>
> Did you do each thing alone too to tell the difference?
> -hp alone, disable monotonic alone (i did not see you mention the disable
> monotonic)
>
> as for your 4ms thing, yes we require high resolution timing, if we ask to
> sleep 1000 microseconds that is what we need it to sleep for or at least as
> close as possible, and the main reason that thread is never sleeping is
> because you can't actually count on it to run every 1ms but you mostly can.
> Hence the whole philosophy on only making 1 thread run hot all the time to
> ensure that the rest don't have to repeat the same algorithm.  We focus on
> high end performance this was the point of your experimentation because we
> will need to use a compile time defines and other logic to make it more
> efficient on your platform, a platform which we are not using.  I am curious
> what would happen if you install Kristian's astlinux on one of your devices,
> i think you should also compare the kernel versions.
>
>
> What OS are you running anyway?
>
> Here are some more things to try (running plain trunk with no mods) do
> these systematically each alone and all together with/without -hp or disable
> monotonic etc to see what different combos create
>
> comment out this line (line 10)
> #define DISABLE_1MS_COND
>
> rebuild, this tells it to run a conditional at 1ms in the same timer thread
> which will make all the switch_cond_next share a 1ms conditional instead of
> doing microsleeps
>
> next
>
> some kernels/devices work better using select(0) for sleep where others
> work better using usleep.
> comment out line 109
> apr_sleep(t);
>
> and try
> usleep(t)
>
> also mac works better using nanosleep so you could try changing it so it
> uses the code starting at 101 instead.
>
>
> also your claim about JS should be investigated because I do not think it
> should be the case.
> but you may want to move this to a jira http://jira.freeswitch.org
>
> As for the asterisk comparison,
> not sure how to answer you, that's your decision.
>
>
>
> On Mon, Dec 7, 2009 at 9:28 AM, eaf  wrote:
>
>>
>> Here is what I found...
>>
>> I tried high-priority scheduling as per your suggestion, reniced the
>> program
>> explicitly, rewrote timer thread to sleep on cond. variable and activate
>> only when there are timers and only when the timer actually had to be
>> clicked, turned off SQL thread and removed polling from sofia profile
>> thread.
>>
>> That pretty much eliminated all idle 1ms sleepers that were there except
>> for
>> three in sofia itself (su_epoll_port). And when I was about to be happy, I
>> found that two outgoing calls through my VOIP providers when bridged
>> together showed terrible distortions. I undid all my changes, tried 1.0.4,
>> trunk (noticed btw that when I bridge two calls via loopback in JS in the
>> trunk I must keep JS running, or the calls get terminated - NOT the same
>> as
>> in 1.0.4 where exitting JS left calls running), got pretty much the same
>> sad
>> results. At the same time calls bridged by freeswitch between LAN and any
>> of
>> the VOIP providers behaved just fine. And calls bridged by Asterisk any
>> way
>> were fine too. So that pretty much looked like the end of the freeswitch
>> trials for me.
>>
>> But then I timed your code, mine and found that all those 1ms sleeps that
>> your timer thread was doing (and all those pollers were doing as well)
>> were
>> actually 4ms sleeps because you know what unless kernel is configured with
>> HZ=1000, you can't sleep for less than 4ms (HZ=250) or perhaps even 10ms
>> (HZ=100). Mine was 250.
>>
>> This actually meant that the original timer thread was firing once,
>> sleeping
>> for 4ms, firing 3 more times back-to-back, sleeping for 4ms more, firing 4
>> times back-to-back, etc. It was still firing 20ms timers on time, but 30ms
>> ones of course were not, since 30ms doesn't divide by 4 evenly. Plus
>> whoever
>> relied on runtime.reference or switch_micro_time_now() were kind of
>> screwed
>> because both were running jumpy. Plus whoever assumed that apr_sleep(1000)
>> or cond_yield() was sleeping for 1ms were als

Re: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled.

[Anthony Minessale]

maybe you can try both ways and see if there is a significant difference?
I think -hp would help more if you were doing media than if you were not but
that does not mean it could not still help performance but really the extra
performance would only show up once you had consumed all the resources the
box had to offer without -hp enabled in most cases.



On Mon, Dec 7, 2009 at 11:12 AM, DJB  wrote:

> Anthony,
>
> Thank you for your clear response.  Based on your recommendation, if I want
> to route more calls to the first server, should I take off "-hp", or it's
> better to run with it.  We are running FS for pass-thru traffic with
> signaling only.
>
>
> ------
> *From:* Anthony Minessale 
> *To:* freeswitch-users@lists.freeswitch.org
> *Sent:* Mon, December 7, 2009 8:56:14 AM
> *Subject:* Re: [Freeswitch-users] Question regarding running FreeSWITCH
> with high priority enabled.
>
> One of the properties of -hp is to enable memlockall() which means disable
> swapping.  This causes all memory used by FS to be resident permanently and
> is much more costly in memory usage.  -hp also uses a RR scheduler runs the
> process at a less nice level and increases a few other process ulimits.
> This mode is designed for high end usage and uses more resources when idle
> with a large payout when scaling to many calls.
>
>
>
>
> On Mon, Dec 7, 2009 at 10:42 AM, DJB  wrote:
>
>> One thing that I forgot to mention, these 2 FreeSWITCH servers are getting
>> calls with load balancing from another switch.  Thus, the traffic type are
>> pretty much identical and both FSs have exactly the same on configuration.
>>  Any suggestion would be appreciated.  Thank you.
>>
>> --
>> *From:* DJB 
>> *To:* FREESWITCH-USERS MAILING LIST <
>> freeswitch-users@lists.freeswitch.org>
>> *Sent:* Sun, December 6, 2009 5:17:14 PM
>> *Subject:* [Freeswitch-users] Question regarding running FreeSWITCH with
>> high priority enabled.
>>
>> I have 2 identical Dell PowerEdge 1950 servers running FreeSWITCH Version
>> 1.0.4 (exported) with only one thing difference which is the first one is
>> running with -hp enabled; however, I have noticed that the one with -hp
>> option consumed double in memory usage than the other one.
>>
>> I wonder whether anyone can explain why.  Thank you.
>>
>> Please see below:
>>
>>
>> --
>> top - 01:01:42 up 53 days,  2:45,  1 user,  load average: 0.22, 0.28, 0.29
>> Tasks: 143 total,   1 running, 142 sleeping,   0 stopped,   0 zombie
>> Cpu(s):  0.9%us,  0.2%sy,  0.0%ni, 96.4%id,  2.5%wa,  0.0%hi,  0.0%si,
>>  0.0%st
>> Mem:   8174164k total,  7550092k used,   624072k free,   187568k buffers
>> Swap: 10223608k total,0k used, 10223608k free,  5417524k cached
>>
>>   PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+  COMMAND
>>
>> 30750 root  -2 -10 1823m 1.5g  20m S  8.6 19.8   1153:40 freeswitch
>>
>> 4418 session(s) 14/100
>>
>> root 30750  2.1 *19.9* 1879252 1634300 ? S> /usr/local/freeswitch/bin/freeswitch -nc -hp
>>
>>
>> --
>> top - 01:01:58 up 53 days,  2:45,  1 user,  load average: 0.43, 0.51, 0.49
>> Tasks: 143 total,   1 running, 142 sleeping,   0 stopped,   0 zombie
>> Cpu(s):  0.9%us,  0.3%sy,  0.0%ni, 96.4%id,  2.4%wa,  0.0%hi,  0.0%si,
>>  0.0%st
>> Mem:   8174164k total,  6751260k used,  1422904k free,   203948k buffers
>> Swap: 10223608k total,0k used, 10223608k free,  5432632k cached
>>
>>   PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+  COMMAND
>>
>>  7147 root  15   0 1961m 755m 5164 S  9.0  9.5   1452:26 freeswitch
>>
>>
>> 4478 session(s) 14/100
>>
>> root  7147  1.9  *9.4* 2009392 774848 ?  Sl   Oct15 1452:37
>> /usr/local/freeswitch/bin/freeswitch -nc
>>
>>
>>
>>
>>
>> ___
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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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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

Re: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled.

[Anthony Minessale]

One of the properties of -hp is to enable memlockall() which means disable
swapping.  This causes all memory used by FS to be resident permanently and
is much more costly in memory usage.  -hp also uses a RR scheduler runs the
process at a less nice level and increases a few other process ulimits.
This mode is designed for high end usage and uses more resources when idle
with a large payout when scaling to many calls.




On Mon, Dec 7, 2009 at 10:42 AM, DJB  wrote:

> One thing that I forgot to mention, these 2 FreeSWITCH servers are getting
> calls with load balancing from another switch.  Thus, the traffic type are
> pretty much identical and both FSs have exactly the same on configuration.
>  Any suggestion would be appreciated.  Thank you.
>
> --
> *From:* DJB 
> *To:* FREESWITCH-USERS MAILING LIST  >
> *Sent:* Sun, December 6, 2009 5:17:14 PM
> *Subject:* [Freeswitch-users] Question regarding running FreeSWITCH with
> high priority enabled.
>
> I have 2 identical Dell PowerEdge 1950 servers running FreeSWITCH Version
> 1.0.4 (exported) with only one thing difference which is the first one is
> running with -hp enabled; however, I have noticed that the one with -hp
> option consumed double in memory usage than the other one.
>
> I wonder whether anyone can explain why.  Thank you.
>
> Please see below:
>
>
> --
> top - 01:01:42 up 53 days,  2:45,  1 user,  load average: 0.22, 0.28, 0.29
> Tasks: 143 total,   1 running, 142 sleeping,   0 stopped,   0 zombie
> Cpu(s):  0.9%us,  0.2%sy,  0.0%ni, 96.4%id,  2.5%wa,  0.0%hi,  0.0%si,
>  0.0%st
> Mem:   8174164k total,  7550092k used,   624072k free,   187568k buffers
> Swap: 10223608k total,0k used, 10223608k free,  5417524k cached
>
>   PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+  COMMAND
>
> 30750 root  -2 -10 1823m 1.5g  20m S  8.6 19.8   1153:40 freeswitch
>
> 4418 session(s) 14/100
>
> root 30750  2.1 *19.9* 1879252 1634300 ? S /usr/local/freeswitch/bin/freeswitch -nc -hp
>
>
> --
> top - 01:01:58 up 53 days,  2:45,  1 user,  load average: 0.43, 0.51, 0.49
> Tasks: 143 total,   1 running, 142 sleeping,   0 stopped,   0 zombie
> Cpu(s):  0.9%us,  0.3%sy,  0.0%ni, 96.4%id,  2.4%wa,  0.0%hi,  0.0%si,
>  0.0%st
> Mem:   8174164k total,  6751260k used,  1422904k free,   203948k buffers
> Swap: 10223608k total,0k used, 10223608k free,  5432632k cached
>
>   PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+  COMMAND
>
>  7147 root  15   0 1961m 755m 5164 S  9.0  9.5   1452:26 freeswitch
>
> 4478 session(s) 14/100
>
> root  7147  1.9  *9.4* 2009392 774848 ?  Sl   Oct15 1452:37
> /usr/local/freeswitch/bin/freeswitch -nc
>
>
>
>
>
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>


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Re: [Freeswitch-users] Choppy sound with PCMU

[Anthony Minessale]

t in freeswitch that I
> neglected to change because they were sleeping with 100ms interval, so I
> figured, who cares. Maybe when all things come together (sofia, 100ms*N)
> freeswitch ends up spending 3% of CPU while doing pretty much nothing.
>
> Btw, compared with Asterisk, the latter is not even visible on the first
> top's screen and spends 1% CPU when bridging two G711 calls and recording
> them to disk.
>
> So, at this time I have both original Asterisk and FS setups running. One
> is
> seemless but clumsy in configuration, the other one is neat and stylish but
> too preoccupied with smth... Should I look into sofia epollers? That's kind
> of deep in the code. Or should I just stick with Asterisk?
>
>
>
>
>
> Anthony Minessale-2 wrote:
> >
> > There is another user here with a 300mhz box.  I am willing to
> investigate
> > this improved performance for weak devices but I need to do it in a sane
> > cross-platform way.
> >
> >
> > On Fri, Dec 4, 2009 at 1:32 PM, Yossi Neiman
> >  >> wrote:
> >
> >> A word to the wise to the general FreeSWITCH community:  If Anthony
> >> Minessale suggests that you try to do any number of things, it's a very
> >> good idea to try all those ideas before continuing on.  I've known him,
> >> MikeJ, and bkw for several years, and they almost always have very good
> >> ideas as to troubleshoot a problem in FreeSWITCH.  It's extremely
> >> frustrating to try to help people out who won't try the provided
> >> suggestions first.
> >>
> >> And note directly to "eaf" - bogomips is quite possibly the least
> >> significant bit of data about a cpu that you will get out of
> >> /proc/cpuinfo...  The name itself - bogo, means bogus.
> >> http://en.wikipedia.org/wiki/Bogomips
> >>
> >> -Yossi
> >>
> >> ___
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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
> >
> > GTALK/JABBER/PAYPAL:anthony.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.org
> 
> >
> > pstn:213-799-1400
> >
> > _______
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users@lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> --
> View this message in context:
> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26678873.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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Re: [Freeswitch-users] Lua and database access to core_db

[Anthony Minessale]

Yes, exactly my point.

Like I said you have several choices be paitent till we have time to
code it for free, post a bounty to increase the chance somone will do it
from the community, hire someone to set it up for you or keep trying
yourself.

Did I miss something?

On Dec 6, 2009 3:38 PM, "Jon Bruel"  wrote:

 The MySQL version is 5.1.37. Well I’m not an expert on every field, and I
have no skills in the C, include libraries, and the art of compiling. For
this I have to follow the guidelines. But it wouldn’t harm the FS project if
it generally became more accessible to the race of non-specialists, which I
hereby represent.



*Jon Brüel*

  --

*From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Anthony
Minessale
*Sent:* 6. december 2009 20:42
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Lua and database access to core_db

  Most of this is unfortunatly because you do not have the proper skill to
set it up because, wit...

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

[Anthony Minessale]

Most of this is unfortunatly because you do not have the proper skill to set
it up because, with the proper skills, all of the ways you tried would have
ended sucessfully.  I say that beacause I have had many users use each of
the different methods in your list of failures only they were sucessful.

What you are asking for is possible but would require many hours of coding
just to help solve your problem.
You would have to wait a really long time until someone had the time to do
it for free or post a bounty for it.  Probably about 1k in consulting time.
It may be cheaper for you to pay a consultant to set up one of the ways
known to work.  These are your options as I see it.

 On Dec 6, 2009 12:20 PM, "Lon Baker"  wrote:

Jon,

What version of MySQL are you using?


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Re: [Freeswitch-users] Translating DTMF from RFC2833 to INFO

[Anthony Minessale]

Some more bad news for you, info dtmf spec has expired and has been
abandoned.  Wait till you see what they did accept instead..

On Dec 6, 2009 1:22 PM, "Metik"  wrote:

Unless the IOS you are running is extremely buggy, "debug voip ccapi"
commands should not provide you with that detail, what you really want
to use is "debug voip rtp session named-event".

Normal SIP-to-PSTN calls should use both a pots and voip dial peer but
DTMF relay type is determined by the voip dial peer.

I haven't ran into this issue (i.e. DTMF is ignored when using RFC 2833)
previously in the wild.  Unlike some other SIP feature servers,  I have
not had issues (with RFC 2833) between FS and Cisco IOS gateways.

Although unrelated to FS or any other SIP feature server, I have seen
some issues when multple dtmf relay types are left enabled on a voip
dial peer.  Also, there are some (older) IOS versions that have issues
with DTMF duration which cause digits to be misinterpreted by the
far-end (PSTN/POTS) but not ignored altogether.

-metik

Yehavi Bourvine wrote: > Hello Metik, > > > > 2009/12/6 Metik <
freeswitch-users-l...@metik.com
> >

> > You previously stated that your Cisco gateway has some "bug" that >
prevents you from us...

>  >
> _...
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Re: [Freeswitch-users] caller_id_name + caller_id_number and Snom 360...

[Anthony Minessale]

Or set it to true depending on the case
Also consider using set_profile_var to set the caller id explicitly instead
of using effective.  There is also effective_callee_id name and number you
could set on the a leg.  You'll have to expirement but the one mathieu said
is your best bet.

On Dec 6, 2009 12:47 PM, "Mathieu Rene"  wrote:

Hi Klaus,

Try setting ignore_display_updates=false

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca




On 6-Dec-09, at 1:38 PM, Klaus Hochlehnert wrote:

> Hi, >   > I just checked the SIP traces and it looks like FS sends a
sipfrag message to the phone ...
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Re: [Freeswitch-users] uuid_broadcast with pause and rewind for dictation service?

[Anthony Minessale]

Someone else was asking about this too.
I could probably write a dictaction mod in c like the one I made for
asterisk starting at about $3k depending on the featureset required.

On Dec 6, 2009 10:30 AM, "Peter P GMX"  wrote:

Hello,

I would like to offer a dictation service to a secretary.
Means:

   * the boss is dictating some text on a certain phone number
   * the secretary picks up the recording on the phone and types the
 text into the computer

As the secretary is not able to type in as fastly as heir boss is able
to speak, she needs some kind of pause and rewind button.
1st question: Is there any functionality available for example in
uuid_broadcast?
2nd question: How much would be the effort to implement this
(uuid_broadcast_pause, uuid_broadcast_UNpause,
uuid_broadcast_rewind(sec)) ? I may offer then a bounty for this.

Best regards
Peter

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Re: [Freeswitch-users] Generate cdrs

[Anthony Minessale]

set rotate-on-hup to false in the cdr_csv config file
then it will only rotate when the file gets too big

and also you can get a cdr with

session.generateXmlCdr()  and dig out what you need or get it from variables
but it will not be nearly as reliable as using the C ones because you need
low level access to make sure you write to the disk properly from many
threads etc.


On Thu, Dec 3, 2009 at 4:33 PM, Mouncif Benniane wrote:

> 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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-- 
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Re: [Freeswitch-users] Playing an rtp stream

[Anthony Minessale]

yes this is possible assuming that is a either a multicast address or a
dedicated unicast address you want to listen on that something else is
sending audio to.  it would also require writing a module in C to actually
implement it.


On Thu, Dec 3, 2009 at 7:47 PM, Phillip Jones  wrote:

> Hi there,
>
> It it possible do something like:
>
> 
>   
> 
> 
>   
> 
>
>
> 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] Bridging to a non SIP based system

[Anthony Minessale]

you could make an endpoint module for FS that speaks the special protocol
then use that to call the conference.


On Fri, Dec 4, 2009 at 3:29 PM, Phillip Jones  wrote:

> Hi All,
>
> Every so often you have to ask a question - where you know so little - it's
> hard to even now where to start. This is one of the times. I am not
> expecting an full answer here, just a gentle nudge in right direction to get
> me started.
>
> What I have is a propriety IP based conference system - who want to add the
> ability to have inbound PSTN callers join their conferences. All their
> signaling is propriety - no SIP - but I do have access to that signaling
> schema so can do some translation. Enough to get the IP / Port & CODEC of
> the RTP stream. They use speex rtp sessions over TCP.
>
> So from an architectural point of view I am thinking of having the callers
> enter a FS conference and than bridge that conference to their IP based
> conference room. That would do it.
>
> The problem is that because I can not bridge using SIP (through a Sofia
> gateway) to that IP based conference system I am kind of lost. But it seems
> reasonable that I should be able to get my head round this, because I know
> the IP / Port & CODEC of the RTP stream.
>
> But perhaps I missing a key bit of knowledge/understanding here.
>
> I would be grateful for any advise here.
>
> Thanks a lot,
>
>
> Phil
>
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Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP

[Anthony Minessale]

Why are you changing the 3pcc setting, is this an invite with no sdp?
you need to take a trace from FS.

1) update to latest trunk first so line number match up.
2) issue these commands

sofia profile internal siptrace on
console loglevel debug

save the output and put it on pastebin http://pastebin.freeswitch.org




On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards
wrote:

>
> I have  Mediant 1000 gateway, and for some reason, when I make an outbound
> call, FS enters the CS_CONSUME_MEDIA state and never connects the call.  A
> Wireshark trace shows that FS is replying to the gateway's inbound RTP
> packets with ICMP DESTINATION UNREACHABLE.  But the gateway is sending RTP
> packets to the same port that FS specified in the outbound INVITE.  It
> appears in the log that FS is discarding the 200 OK from the gateway.
>
> I disabled the Firewall and SELinux on the Freeswitch machine.  I tried
> changing  to "true" and also "proxy", but it has no effect.
>
> Anyone know what could be the issue?  I posted the Freeswitch log in the
> pastebin.
>
> Best Regards,
> Jerry
>
>
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Re: [Freeswitch-users] Option to hang-up both legs in a bridge

[Anthony Minessale]

did you see my reply to the other thread?

set the channel variable hangup_after_bridge=true on the a leg

your script must not be checking for the case when b leg hangs up that A leg
does not hangup unless that var is set.


On Fri, Dec 4, 2009 at 2:03 PM, Nik Middleton <
nik.middle...@noblesolutions.co.uk> wrote:

>  Hi,
>
>
>
> Is there an option to hang-up both call legs in a bridge when one leg hangs
> up?
>
>
>
> In my lua script I only ever see the hang-up for the call I’m in, not for
> the bridged b leg.  That said, I can see both a hang-up and un bridge event
> being fired for the B leg.  However my issue is that the A leg is still up,
> and if I’ve called 2 Pots numbers, the phone network will maintain the
> bridge.
>
>
>
> Is my only option to subscribe to the unbridge event and fire a hang-up
> event using the ‘other leg’ UID?
>
>
>
> Regards,
>
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Re: [Freeswitch-users] Voicmail - message only

[Anthony Minessale]

You could file it as a feature request and post a bounty and probably get
the functionality fairly inexpensively maybe $100



On Fri, Dec 4, 2009 at 11:26 AM, Peter P GMX  wrote:

> I would like to manage this in the voicemail menu.
>  "Press 6 to enable recording"
>  "Press 7 to only play announcement"
> or so. So hte user can manage it's settings on his own.
>
> Best regrds
> Peter
>
> Adam Ford schrieb:
> > I am still new to freeswitch, but I would think you could achieve this by
> > just passing the call to an IVR application that plays the message
> instead
> > of passing it to the voicemail application.
> >
> > -AF
> >
> > -Original Message-
> > From: freeswitch-users-boun...@lists.freeswitch.org
> > [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
> Peter P
> > GMX
> > Sent: Friday, December 04, 2009 9:02 AM
> > To: freeswitch-users@lists.freeswitch.org
> > Subject: [Freeswitch-users] Voicmail - message only
> >
> > Hello,
> >
> > is there a chance to have the voicemail system to play announcment #1
> > only and not play announcement and then record the voicemail?
> > Means: Can I switch off the recording part?
> >
> > Best regards
> > Peter
> >
> >
> > ___
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> > FreeSWITCH-users@lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> >
> >
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> >
> >
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Re: [Freeswitch-users] B Leg on bridged call is not hanging up

[Anthony Minessale]

did you set the channel variable hangup_after_bridge=true on the A leg?


On Fri, Dec 4, 2009 at 10:06 AM, Nik Middleton <
nik.middle...@noblesolutions.co.uk> wrote:

>  Hi Guys,
>
> This one has me stumped.
>
> I'm originating a call, playing audio, trapping on DTMF and bridging to
> another endpoint (read phone number)
>
> If the A leg hangs up, then the call is cleared down and all is well.
> However if the B Leg attempts to hang-up, the LUA script that is handling
> the bridge continues to play audio to the a leg, while the B leg is in
> limbo.  It does eventually time out with no RTP.
>
> Running Sofia debug on the cli shows that I'm getting the BYE from the B
> Leg, but that's about as far as I can get.  The hang-up hook is not being
> fired in the lua script.
>
> Anyone give me some pointers as to where I might start looking?
>
> regards
>
>
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Re: [Freeswitch-users] Sporadic call drops

[Anthony Minessale]

we changed that message a long time ago so people would not think that
anymore
We are now 3000 rev beyond the version you are at, I would like it if you
try the lastest trunk.


On Fri, Dec 4, 2009 at 12:16 PM, Luis F Urrea  wrote:

> Hi all,
>
> Guys I know the question could be too vague, but I have a customer that
> just reported frequent failure to place outbound calls though a PSTN gateway
> on the LAN.
>
> I looked at the logs and I seem to be able to confirm that FS fails to
> place the call through the gateway and that the issue resides on the FS side
> since the first channel that s killed is tht of the internal extension
> registered to FS and then FS send the BYE to gw and kills the channel.
>
> What are possible causes of this?
>
> I know you always like to look at complete logs but here's a snip that
> could shed some light on the disconnection. (I can provide full logs if
> required and worthed)
>
> 2009-12-04 11:21:56 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel
> sofia/internal/2...@172.16.3.5 entering state [ready][200]
> 2009-12-04 11:21:59 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel
> sofia/internal/2...@172.16.3.5 entering state [terminated][200]
> 2009-12-04 11:21:59 [NOTICE] sofia.c:3597 sofia_handle_sip_i_state() Hangup
> sofia/internal/2...@172.16.3.5 [CS_EXECUTE] [NORMAL_CLEARING]
> 2009-12-04 11:21:59 [DEBUG] switch_channel.c:1660
> switch_channel_perform_hangup() Send signal 
> sofia/internal/2...@172.16.3.5[kill]
> 2009-12-04 11:21:59 [DEBUG] switch_core_session.c:933
> switch_core_session_signal_state_change() Send signal sofia/internal/
> 2...@172.16.3.5 [BREAK]
> 2009-12-04 11:22:00 [DEBUG] switch_ivr_bridge.c:371 audio_bridge_thread()
> sofia/internal/2...@172.16.3.5 ending bridge by request from write function
> 2009-12-04 11:22:00 [DEBUG] switch_ivr_bridge.c:426 audio_bridge_thread()
> sofia/pstn/22909...@172.16.3.46 receive message [UNBRIDGE]
>
>
> Is the 6th line normal behavior for ending the channel?
>
> FreeSWITCH Version 1.0.trunk (13484M)
>
> TIA
>
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Re: [Freeswitch-users] Choppy sound with PCMU

[Anthony Minessale]

There is another user here with a 300mhz box.  I am willing to investigate
this improved performance for weak devices but I need to do it in a sane
cross-platform way.


On Fri, Dec 4, 2009 at 1:32 PM, Yossi Neiman  wrote:

> A word to the wise to the general FreeSWITCH community:  If Anthony
> Minessale suggests that you try to do any number of things, it's a very
> good idea to try all those ideas before continuing on.  I've known him,
> MikeJ, and bkw for several years, and they almost always have very good
> ideas as to troubleshoot a problem in FreeSWITCH.  It's extremely
> frustrating to try to help people out who won't try the provided
> suggestions first.
>
> And note directly to "eaf" - bogomips is quite possibly the least
> significant bit of data about a cpu that you will get out of
> /proc/cpuinfo...  The name itself - bogo, means bogus.
> http://en.wikipedia.org/wiki/Bogomips
>
> -Yossi
>
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Re: [Freeswitch-users] Lua and database access to core_db

[Anthony Minessale]

That means you mysql is not configured to do transactions so it failed over
back to sqlite.
if you scan for the warning message you will see the option you have to set
and you may possibly have to update your myodbc odbc driver.

To answer you other question about the sqlite, like I said the lua does not
have the object coded like js does so it would be a project to implement
it.  You can also consider using ODBC plugin for lua to access the sqlite.


On Fri, Dec 4, 2009 at 3:24 AM, Jon Bruel  wrote:

>  I have now tested the FS with core db configured using MySql (by
> modifying the switch.conf.xml file). Unfortunately, it does not solve my
> problem because some of the core tables still remain as active SQLite
> tables.
>
>
>
> After restarting the FS in the new configuration (with SQLite database core
> deleted), the following tables are created in MySql and SQLite:
>
>
>
> MySQL: aliases, complete, nat and tasks (database starting with no tables
> prior to FS restart).
>
> SQLite: aliases, calls, channels, interfaces, nat and tasks.
>
>
>
> As I would like to access the channels table using Lua, the change did not
> fix my problem. I have positive verified that the channels table is active
> and populated during calls.
>
>
>
> Are there other places where I should define the usage of the MySql
> database?
>
>
>
>
>
> *Jon Brüel*
> Consiglia Telecommunications
>
> DK-2960 Rungsted Kyst
> Tel: +45 45 16 1000
> Mob: +45 26 15 30 60
>
> CVR: 27047882
>
>
>
>
>
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Re: [Freeswitch-users] Choppy sound with PCMU

[Anthony Minessale]

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  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 hardw

Re: [Freeswitch-users] Choppy sound with PCMU

[Anthony Minessale]

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  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  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
> >> >
> >> >

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

[Anthony Minessale]

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  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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Re: [Freeswitch-users] Choppy sound with PCMU

[Anthony Minessale]

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  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  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  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 

Re: [Freeswitch-users] Eavesdrop error?

[Anthony Minessale]

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  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  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 | ]
> >
> > I simply dialed 881010, trying to eavesdrop on extension 1010. Is this
> > incorrect?
> >
> > http://pastebin.freeswitch.org/11363
> >
> > Thanks Lars
> >
> >
> > ___
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users@lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > ___
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users@lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > --
> > 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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> > FreeSWITCH-users@lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
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Re: [Freeswitch-users] Call transfer got broken for me

[Anthony Minessale]

to late it's fixed now.


On Thu, Dec 3, 2009 at 10:21 AM, Michael Jerris  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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>
>
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>
>


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Re: [Freeswitch-users] Choppy sound with PCMU

[Anthony Minessale]

ere 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.
> >>>>
> >>>>
> >>>> ___
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users@lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >>>> users
> >>>> http://www.freeswitch.org
> >>>
> >>> ___
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> >>> FreeSWITCH-users@lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >>
> >
> >
>
> --
> View this message in context:
> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26627246.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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Re: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted

[Anthony Minessale]

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
>>  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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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
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> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
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> iax:gu...@conference.freeswitch.org/888
> googletalk:conf+...@conference.freeswitch.org
> pstn:213-799-1400
>



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Re: [Freeswitch-users] Eavesdrop error?

[Anthony Minessale]

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  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 | ]
>
> I simply dialed 881010, trying to eavesdrop on extension 1010. Is this
> incorrect?
>
> http://pastebin.freeswitch.org/11363
>
> Thanks Lars
>
>
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Re: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted

[Anthony Minessale]

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
>  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] change the remote RTP port after sample rate doesnot match

[Anthony Minessale]

that was make hd-sounds-install sorrry

you should also update to SVN trunk because based on the line number in your
log
its clear you are using a much older version of FS


On Wed, Dec 2, 2009 at 2:08 PM, Erwin Davis  wrote:

> 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,
>
>
>
>
>
> On 12/2/09, Anthony Minessale  wrote:
>>
>> you must only have 8k sounds so the resample is when it's playing files
>>
>> try make hd-sounds-install to install 16k sounds too
>>
>>
>>
>>
>>
>> On Wed, Dec 2, 2009 at 12:41 PM, Erwin Davis 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,
>>>
>>>
>>> ___
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users@lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/

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

[Anthony Minessale]

you must only have 8k sounds so the resample is when it's playing files

try make hd-sounds-install to install 16k sounds too





On Wed, Dec 2, 2009 at 12:41 PM, Erwin Davis  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,
>
>
> ___
> FreeSWITCH-users mailing list
> FreeSWITCH-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

[Anthony Minessale]

nal/2001 [7db554a6-861e-4492-a87b-78b6dfec6488]
> 2009-12-02 13:31:59.624818 [NOTICE] sofia.c:3353 Ring-Ready
> sofia/internal/2001!
> 2009-12-02 13:32:00.130814 [NOTICE] sofia.c:3794 Channel
> [sofia/internal/2001] has
> been answered
> Dec 2, 2009 1:32:00 PM org.starpound.fs2agi.Translator tellAllWeCrashed
> INFO: AGI application agi://localhost:4573/hello.agi?callId=932 for session
> 7c37369b-ffb2-4436-9288-a640047d0e5e 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 13:32:00.149080 [INFO] switch_cpp.cpp:1130 AGI application
> agi://localhost:4573/hello.agi?callId=932 crashed. See FS2AGI log for
> details.
> 2009-12-02 13:32:00.388838 [INFO] switch_rtp.c:1869 Auto Changing port from
> 172.26.10.39:26402 to 91.190.120.190:26402
>
>
>
> Suggestions?
>
>
>
>
>
>
>
>
>
>
>
> On Sat, Nov 21, 2009 at 12:58 PM, Artem Shiyanov wrote:
>
>> Anthony,
>>
>> >>As soon as you call uuid_bridge you are transferring both legs of the
>> call to bridge to each other.
>> >>This means your java app must exit so the channels can connect to each
>> other.
>>
>> I didn't know that. Now my java app is exiting upon the onHangup() call so
>> everything has become "ok". Thank you much.
>> I'll add note to the wiki about this issue.
>>
>> Artem
>>
>>
>>
>>
>> On Fri, Nov 20, 2009 at 5:49 AM, Anthony Minessale <
>> anthony.miness...@gmail.com> wrote:
>>
>>> Your "annoying behaviour" is the exact behavior you should be getting
>>> considering what you told FS to do.
>>>
>>> As soon as you call uuid_bridge you are transferring both legs of the
>>> call to bridge to each other.
>>> This means your java app must exit so the channels can connect to each
>>> other.
>>>
>>> remember that you hangup hook can be called when the channel is
>>> transferred not only when it hangs up.
>>> you have to test which is happening based on the input to your callback.
>>>
>>>
>>> On Thu, Nov 19, 2009 at 1:46 PM, Artem Shiyanov wrote:
>>>
>>>> Hi there!
>>>>
>>>> I've got annoying FS behavior:
>>>> There are 2 channels executing the same Java application (application
>>>> itself is an IVR). If I try to bridge them with uuid_bridged then both
>>>> channels are killed. Here is a log from FS console:
>>>> uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2
>>>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de
>>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165
>>>> (sofia/internal/1...@192.168.147.130) State Change CS_EXECUTE ->
>>>> CS_HIBERNATE
>>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook
>>>> called
>>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done
>>>> playing file
>>>> 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done
>>>> playing file
>>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send signal
>>>> sofia/internal/1...@192.168.147.130 [BREAK]
>>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167
>>>> (sofia/internal/1...@master.agent.starpoundtech.net) State Change
>>>> CS_EXECUTE -> CS_HIBERNATE
>>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook
>>>> called
>>>> API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2
>>>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output:
>>>> +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de
>>>>
>>>> freeswi...@localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG]
>>>> switch_core_session.c:933 Send signal
>>>> sofia/internal/1...@master.agent.starpoundtec
>>>> 2009-07-09 0

Re: [Freeswitch-users] CLIP on FXS channels with mod_openzap

[Anthony Minessale]

Did you also update your wanpipe drivers and rebuild openzap again after you
upgraded it?


On Wed, Dec 2, 2009 at 2:12 AM, François Legal  wrote:

> So I did some tests and still I can not see CLIP on a phone connected to an
> FXS port. Whether the call is bridged from SIP UA or from an incoming call
> on FXO port does not change anything. Whether the parameter
> enable-caller-id=true is present or not in openzap.conf.xml does not change
> anything too.
>
> On that subject, sangoma support team says it must be freeswitch as this
> feature is supported and has been tested working.
>
>
>
> However, the good point is that I did not experience cuts in my call
> bridged from FXS to FXO with that new release.
>
>
>
> François
>
>
>
> On Tue, 1 Dec 2009 19:02:11 -0600, Anthony Minessale <
> anthony.miness...@gmail.com> wrote:
>
> upgrading always helps *something* not sure.  but that is where we have to
> start because we have changed that code alot.
>
>
> On Tue, Dec 1, 2009 at 2:37 AM, François Legal wrote:
>
>> Sure, I'll try that. I'm just building freeswitch-snapshot that I
>> downloaded from files.freeswitch.org
>>
>> I also experience, when bridging a call from an FXS to FXO the call is cut
>> after a random time (this does not appear when bridging SIP to FXO). Might
>> this upgrade fix this problem also ?
>>
>>
>>
>> François
>>
>>
>>
>> On Mon, 30 Nov 2009 10:48:26 -0600, Anthony Minessale wrote:
>>
>> can you test svn trunk or latest pre release of 1.0.5
>>
>>
>> On Mon, Nov 30, 2009 at 9:36 AM, François Legal wrote:
>>
>>> Hello,
>>>
>>>
>>>
>>> I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP
>>> problems on the FXS ports.
>>>
>>> When I ring on FXS ports, the connected phone does not display
>>> callerid/callerid-name.
>>>
>>> I tried turning the stuff of in openzap.conf.xml () but it did not help.
>>>
>>>
>>>
>>> As a side note, turning this on on the FXO ports drops the callerid
>>> information on incoming calls.
>>>
>>>
>>>
>>> Running freeswitch 1.0.4 on linux 2.6.27.
>>>
>>>
>>>
>>> François
>>>
>>> ___
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users@lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> 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
>>
>>
>>
>> ___
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users@lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> 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
>
>
>
> ___
> FreeSWITCH-users mailing list
> FreeSWITCH-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
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 
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Re: [Freeswitch-users] Choppy sound with PCMU

[Anthony Minessale]

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.


On Tue, Dec 1, 2009 at 9:19 PM,  wrote:

> Wow... Thinking about this timer setting and about how it converted
> send()/recv() from non-blocking to blocking, I straced freeswitch when it
> was
> supposed to be idle. It never pauses! It keeps going in and out of select()
> every millisecond! Why??
>
> -- Original Message --
> Received: Tue, 01 Dec 2009 08:31:46 PM EST
> From: erandr-j...@usa.net
> To: 
> Subject: Re: [Freeswitch-users] Choppy sound with PCMU
>
> > Thanks. I tried that... Just forcing SPA to 20ms didn't change anything.
> Just
> > installing SVN trunk didn't fix it either, but setting that option
> afterwards
> > surely did the trick.
> >
> > One thing I've noticed while staring at the console is that it *looks
> like*
> > that w/o the new setting the stuttering happens when FS either
> re-registers
> > itself with the provider or one of the SPA's port re-registers with FS.
> >
> > -- Original Message --
> > Received: Tue, 01 Dec 2009 05:33:26 PM EST
> > From: Anthony Minessale 
> > To: freeswitch-users@lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Choppy sound with PCMU
> >
> > > linksys has had a bug for eons that can be fixed by setting the ptime
> (or
> > > rtp packet size in their terms)
> > > in it's firmware to .20 instead of .30
> > >
> > > Asterisk does not use async RTP like we do so it's never a problem
> > > you can disable the timer by setting the channel var
> rtp_timer_name=none
> or
> > > sofia param rtp-timer-name to none in the sofia profile.
> > >
> > > You should also test this on latest SVN trunk or wait for pre8
> > >
> > >
> > >
> > > On Tue, Dec 1, 2009 at 3:52 PM, eaf  wrote:
> > >
> > > >
> > > > I should also add, after browsing through some topics here, that my
> SIP
> > > > provider sends 172-byte RTP frames, which is in accordance with
> ptime:20
> > > > that it gives to FreeSWITCH.
> > > >
> > > >
> > > > eaf wrote:
> > > > >
> > > > > Hi,
> > > > >
> > > > > I'm trying to migrate from Asterisk to FreeSWITCH (really like the
> way
> > > > how
> > > > > it can be programmed), but ran into one issue with sound quality
> that
> I
> > > > > just cannot workaround by myself. I would describe the sound
> problem
> as
> > > > > being "choppy". From time to time small portions of the other
> party's
> > > > > voice are dropped, so the voice kind of stutters. This is not too
> bad,
> > > > but
> > > > > is really noticeable, happens in every call and I don't experience
> the
> > > > > same with Asterisk running on the same box. I attached two files:
> > > > > freeswitch.wav and asterisk.mp3 to illustrate my point.
> > > > >
> > > > > Issue completely goes away, if I set inbound-proxy-media to true.
> > > > >
> > > > > The way how I test is to connect SPA-2000 via 10mbps LAN to the box
> > > > > directly exposed to internet, and then dial a toll-free via
> FutureNine
> > (a
> > > > > SIP provider).
> > > > >
> > > > > The codec in use is PCMU. Can't really try PCMA or anything else
> with
> > > > this
> > > > > provider. Only PCMU. Tried to match ptime of provider (30) with
> ptime
> > of
> > > > > the SPA, didn't get any improvement. Tried turning off recording,
> no
> > > > > change either.
> > > > >
> > > > > What puzzles me is that even with greedy codec negotiations and
> with
> > PCMU
> > > > > on both sides of  FreeSWITCH, it's still saying that
> > > > > TRANSCODING_NECESSARY. I'm attaching relevant portion of
> freeswitch.log
> > > > to
> > > > > illustrate.
> > > > >
> > > > > The box isn't particularly fast: Linux (Debian 4), CPU - AMD Geode
> > LX800
> > > > > with 997 bogomips. 256MB RAM. Only on

Re: [Freeswitch-users] Remote fetching of voicemail

[Anthony Minessale]

bind to the transfer app so that it transfers the call to the vm extension
that way the current application is always interrupted and replaced.

The special "inline" dialplan lets you transfer calls right to an
application

use "inline" as the dp name and voicemail: as the extension


On Wed, Dec 2, 2009 at 4:57 AM, François Legal  wrote:

> Hello,
>
>
>
> I created an extension in my dialplan so that when an incoming call
> arrives, it rings a group of lines and then fallback to the voicemail if no
> line is answered.
>
> I wanted then that when voicemail starts, the calling party could dial some
> numbers to fetch the voicemail. I used bind_meta_app for this. My problem
> is, when using bind_meta_app, the voicemail continues, and I sometimes
> experience freeswitch hanging after the call is over, depending on when the
> bind_meta_app is activated.
>
> How can I make freeswitch terminate the first voicemail instance when
> activating the bind_meta_app.
>
>
>
> Here's my extension :
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> Thanks
>
>
>
> François
>
> ___
> FreeSWITCH-users mailing list
> FreeSWITCH-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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MSN:anthony_miness...@hotmail.com 
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Re: [Freeswitch-users] Dictation System

[Anthony Minessale]

Yes, I'm familiar with that application, check the src code for the author
=p
There has not been much of a demand for such an application but it's of
course entirely possible to develop one.


On Wed, Dec 2, 2009 at 9:00 AM, David Laperle  wrote:

>  Hi Freeswitch users,
>
> i'm new into the PBX world. I just installed FreeSwitch and made work
> great, but one of my goal with the PBX system is to use it as a dictation
> system. We were using Callweaver, and there's a Dictation module for CW and
> one for Asterisk, but i can't find one for FreeSwitch so far. Is there
> anything in the trunk that i could find or any work in progress?
>
> I'm willing to develop a bit to help the work in progress, i have the
> programming knowledge but not the VOIP/PBX knowledge, so a work in progress
> could be enough for me to start with and complete the work!
>
> If any of you have an idea or a hint for me i would be very grateful!
>
> 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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-- 
Anthony Minessale II

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

[Anthony Minessale]

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.


On Wed, Dec 2, 2009 at 10:35 AM, Kristian Kielhofner <
kristian.kielhof...@gmail.com> wrote:

> As always, you are correct.
>
> The scenario now is:
>
> - If the caller places the callee on hold, the callee will get hold music
> - If the callee places the caller on hold, the caller will not get hold
> music
>
> I've uploaded a fresh pastebin here:
>
> http://pastebin.freeswitch.org/11356
>
> On Fri, Nov 20, 2009 at 10:34 PM, Anthony Minessale
>  wrote:
> > results cant possibly be the same
> > there is not even any broadcast involved in uuid_transfer ?
> >
> > you need to attach a console trace with debug log up
> >
> >
>
> --
> 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] CLIP on FXS channels with mod_openzap

[Anthony Minessale]

upgrading always helps *something* not sure.  but that is where we have to
start because we have changed that code alot.


On Tue, Dec 1, 2009 at 2:37 AM, François Legal  wrote:

> Sure, I'll try that. I'm just building freeswitch-snapshot that I
> downloaded from files.freeswitch.org
>
> I also experience, when bridging a call from an FXS to FXO the call is cut
> after a random time (this does not appear when bridging SIP to FXO). Might
> this upgrade fix this problem also ?
>
>
>
> François
>
>
>
> On Mon, 30 Nov 2009 10:48:26 -0600, Anthony Minessale wrote:
>
> can you test svn trunk or latest pre release of 1.0.5
>
>
> On Mon, Nov 30, 2009 at 9:36 AM, François Legal wrote:
>
>> Hello,
>>
>>
>>
>> I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP
>> problems on the FXS ports.
>>
>> When I ring on FXS ports, the connected phone does not display
>> callerid/callerid-name.
>>
>> I tried turning the stuff of in openzap.conf.xml () but it did not help.
>>
>>
>>
>> As a side note, turning this on on the FXO ports drops the callerid
>> information on incoming calls.
>>
>>
>>
>> Running freeswitch 1.0.4 on linux 2.6.27.
>>
>>
>>
>> François
>>
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>>
>
>
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