Re: [Freeswitch-users] Choosing a Codec.

[David Knell]

On Tue, 2009-12-22 at 23:52 -0500, Michael Jerris wrote:
> That being said, ulaw l16 alaw will cause degredation and any other  
> modifications such as volume adjustment in this path will make it  
> worse.

Indeed.  Storing prompts 
as 8k, 16-bit WAVs
makes a lot of sense.

[I am inordinately pleased with the above]

--Dave



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Re: [Freeswitch-users] Choosing a Codec.

[David Knell]

On the other hand, a u-law WAV turned into L16 and then back to u-law to
be sent down the line shouldn't suffer any alteration at all - if it
does, the there's something wrong with the translation.

The quality dropping over time is almost certainly down to something
else.  Vinuth -can you get a recording to compare with the original?

--Dave


> If its degrading like that you have bigger issues... the sound files played 
> from wav files vs raw PCM files is NO different on a land line and I speak 
> from very many years of experience... your wav files are ulaw in wav 
> containers thus will never play native which might just be part of your 
> problem.  You would have to have raw headerless data in a .PCMU file for it 
> to play native.  
> 
> Can you elaborate on your setup a bit more?
> 
> /b
> 
> On Dec 22, 2009, at 4:54 PM, Vinuth Madinur wrote:
> 
> > The audio quality is a lot different when it plays on the landline. And the 
> > quality degrades a bit when the message played is lengthy >30s. So I 
> > thought it would be better if I have the file in mu-law and play it as is..
> > 
> > Thanks,
> > Vinuth.
> 
> 
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Re: [Freeswitch-users] mod_conference scalability

[David Knell]

Hi Brian,

Have a look at this:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop
- I took a quick look through the code and couldn't see any reason why
you shouldn't have a bunch of eavesdroppers listening to a single
caller.  I'd be surprised if this didn't perform a lot better for your
application.

Cheers --

Dave

> 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.
> 
>  
> 
> 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.
> 
>  
> 
> I work with other open source projects and fund enhancements or fixes
> I need. FreeSWITCH would be no different. 
> 
>  
> 
> Brian.
> 
>  
> 
>  
> 
> From: Anthony Minessale [mailto:anthony.miness...@gmail.com] 
> Sent: Friday, December 18, 2009 11:34 AM
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] mod_conference scalability
> 
> 
>  
> 
> 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.org This 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 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

Re: [Freeswitch-users] mod_conference scalability

[David Knell]

Hi Brian,

I imagine that one of the issues is that you're using a complex
sledgehammer (mod_conference) to crack a simple nut - that of having
multiple listeners listening to a single speaker.

As far as I am aware, FreeSWITCH doesn't have anything built in which
will allow this kind of simple audio path switching - maybe someone more
knowledgeable than me will correct me if I'm wrong?

I presented some stuff at ClueCon which would address this kind of
simple application and ought to scale well beyond what you've seen with
FS or Asterisk.  It's still pretty basic [I'd do more with it if I
wasn't so busy joshing with the other Brian on Facebook], and has never
been deployed in anger but, if you're interested, drop me a note
off-list.

--Dave

> 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:
> 
> 
>  
> 
> 
> Scena

Re: [Freeswitch-users] SIP Re-invite

[David Knell]

Can you post the full packets with Ethernet, IP, UDP headers as well, or
upload a pcap file?

I'll add the change in 'Max-Forwards' from 70 to 69 between the two
packets to my things to be suspicious of list.

--Dave

> The trace that I pasted on the pastebin was from our
> analyzer,Tektronix spectra2 that was sitting between FS and customer.
> I also had the FS sip trace on and compare with the trace from Spectra
> when I found out about the 3rd re-invite was missing from FS.
>  
> Thank you.
> 
> 
> 
> __
> From: Anthony Minessale 
> To: freeswitch-users@lists.freeswitch.org
> Sent: Thu, December 17, 2009 7:57:42 AM
> Subject: Re: [Freeswitch-users] SIP Re-invite
> 
> 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.
> 
> 
> 
> 
> 
> 
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> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_miness...@hotmail.com
> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:8...@conference.freeswitch.org
> iax:gu...@conference.freeswitch.org/888
> googletalk:conf+...@conference.freeswitch.org
> pstn: +19193869900  +19193869900 
> 
> 
> 
> 
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> 
> 
> -- 
> Anthony Mine

Re: [Freeswitch-users] SIP Re-invite

[David Knell]

I'd be suspicious of:
(a) the CSeq going backwards from 4 to 3 between re-invites 2 and 3;
(b) the branch on the Via tag changing
(c) (not sure about this one) the SDP session ID and version changing
for what's the same session.

--Dave


> 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.
> 
> 
> 
> 
> 
> 
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> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_miness...@hotmail.com
> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:8...@conference.freeswitch.org
> iax:gu...@conference.freeswitch.org/888
> googletalk:conf+...@conference.freeswitch.org
> pstn:+19193869900
> 
> 
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Re: [Freeswitch-users] FS Rocks!!!!!!!!!

[David Knell]

On Thu, 2009-12-10 at 10:26 -0800, Michael Collins wrote:

> Update to latest
> Did you type make current yet?
> Tony hates build skew

Brilliant.

Michael Collins-san
Shrinks all usual advice
Into one Haiku.

--Dave


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Re: [Freeswitch-users] Hardware echo cancellation.

[David Knell]

On Fri, 2009-11-20 at 09:57 +0800, Steve Underwood wrote:
> On 11/20/2009 05:15 AM, David Knell wrote:
> > On Fri, 2009-11-20 at 00:15 +0800, Steve Underwood wrote:
> >
> >
> >> The audio path between kernel and user space is not stable with any
> >> current PC based telephony system. At some point in the day the odd
> >> chunk of data is lost here and there, whether you use asterisk,
> >> callweaver, yate or FS, with dahdi or sangoma. This is the key problem
> >> for user space echo cancellation. When the path hiccups, the EC goes
> >> crazy, and howls. So far kernel space EC has been the only way to keep
> >> the path length rock solid.
> >>  
> > Why do you think this is?  Getting data from kernel space to user space
> > isn't something which it's difficult to do reliably: the disk system
> > manages it.  Even if data is being lost, buffer overruns can be dealt
> > with by using bigger buffers, or timestamping blocks of data on their
> > way in so that missing blocks can be detected.
> >
> Disk isn't audio. Audio is real time, and real time constraints are a 
> harsh mistress. Big buffers are out of the question, due to latency. 

Not necessarily.  A decent-sized FIFO, mostly run empty, but there to
buffer data in the case of the user-side not being able to accept it for
a short period wouldn't necessarily add to latency unless it were
needed.  The user side could then make a decision as to how to deal with
the queued data - dump it or handle it - according to its requirements.

> Some mitigation could be provided if you can detect where missing chunks 
> occur and their exact size. Right now, the I/O schemes do not provide 
> for that, and incorporating support would be tough. You'd need some out 
> of band indication, like an ioctl or something, which would lead to more 
> user space/kernel space exchanges, further increasing the problem.

I don't think it'd be all that hard.  Were I to do this, I'd probably:
- define an error return (ESLIP, EDATALOST, something like that) which
might be returned by read/write
- add an ioctl to enable and disable it
- maybe add an ioctl to indicate how much data's been lost

Doesn't break existing stuff, doesn't add any overhead under normal
conditions, would be handy for better reliability with EC, DTMF, fax,
etc.

--Dave


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Re: [Freeswitch-users] Hardware echo cancellation.

[David Knell]

On Fri, 2009-11-20 at 00:15 +0800, Steve Underwood wrote:

> The audio path between kernel and user space is not stable with any 
> current PC based telephony system. At some point in the day the odd 
> chunk of data is lost here and there, whether you use asterisk, 
> callweaver, yate or FS, with dahdi or sangoma. This is the key problem 
> for user space echo cancellation. When the path hiccups, the EC goes 
> crazy, and howls. So far kernel space EC has been the only way to keep 
> the path length rock solid.

Why do you think this is?  Getting data from kernel space to user space
isn't something which it's difficult to do reliably: the disk system
manages it.  Even if data is being lost, buffer overruns can be dealt
with by using bigger buffers, or timestamping blocks of data on their
way in so that missing blocks can be detected.

--Dave


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Re: [Freeswitch-users] Hardware echo cancellation.

[David Knell]

Hi Brian,

> It just doesn't belong in user space or kernel space in the machine  
> for true performance you should do it in hardware... I'm pretty sure  
> the poor box would die if you tried it on 32 E1's at the same time.

Disagree somewhat.  The challenge that echo cancellers further from the
hardware face is having some idea of the size of the buffers between the
canceller and the wire; provided that this is known, or is small in
comparison to the canceller's tail length, it can, in principle, go
anywhere.  All other things being equal, the right place for a software
EC is in user space: can be done in a cross-platform way, can use
FPU/MMX/SSE without guilt and voodoo, etc.  And there is no reason why
the same algorithm would perform differently if implemented in
"hardware" or on the host CPU.

And the OP only needed four E1s..

--Dave


> 
> /b
> 
> On Nov 18, 2009, at 5:39 PM, David Knell wrote:
> 
> > For the sort of box you're talking about (quad core++), this isn't  
> > lots;
> > it's hardly any..
> >
> > --Dave
> 
> 
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Re: [Freeswitch-users] Hardware echo cancellation.

[David Knell]

Hi Tim,

> > In (very) brief: maybe, no, and depends on the definition of 'lots'.
> >
> 
> By lots I mean somewhere between 50 to a 100 but it's mostly an IVR
> application so all it will be doing is either playing prompts or
> recording messages. Almost no live conversations.

For the sort of box you're talking about (quad core++), this isn't lots;
it's hardly any..

--Dave


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Re: [Freeswitch-users] Hardware echo cancellation.

[David Knell]

Hi Tim,

Here you go:
http://old.nabble.com/echo-cancellation-on-PRI-cards-td22552605.html

> I am about to build a new machine as a VOIP server. I am going to get
> either a quad core intel or a six core AMD processor with at least
> eight gigabytes of RAM in it.   Given that much horsepower I am
> wondering if there is any need to purchase hardware with echo
> cancellation (I am thinking about redfone devices)..  I can save some
> money by not getting the echo cancellation.
> 
> So is it worth saving that money? Is it always better to have hardware
> echo cancellation? Is a quad core capable of dealing with echo
> cancellation needs of an IVR which is going to take lots of
> simultaneous calls?

In (very) brief: maybe, no, and depends on the definition of 'lots'.

--Dave




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Re: [Freeswitch-users] Detecting Forwarding Device Caller ID (Off Topic)

[David Knell]

Hi Max,

Some operators (e.g. Orange in the UK) will allow you to have two
numbers attached to one SIM, which might get you 4/10ths of the way to
your goal.

I'm struggling to see why purchasing DIDs won't help, unless your FS box
has no internet connectivity.  A bit more background on the problem
might help with finding a solution?

--Dave

> Hey Guys,
> 
> Pleas help me answer the following question. Sorry for being a bit
> off-topic.  I have a GSM-VoiP modem with just 4 SIM Cards. Is there a
> known mechanism that allows me to use this modem for more than 4
> (albeit not at the same time) GSM numbers?  Let's say the phone
> numbers of the SIM cards inserted in the modem are call real phone
> numbers. My goal is to have some 10 additional virtual phone numbers
> served by this very same modem.
> 
> One way i thought this could be done is by forwarding the virtual
> phone numbers the 4 real phone numbers. Then, when somebody calls one
> of the virtual numbers, the Telco will forward it to the real phone
> number and it will end on Freeswitch box. My only problem is now that
> on the Freeswitch box I have no idea what virtual phone number was
> actually dialed. Any idea how to recover this phone number? I looked
> at the GSM specification and couldn't find a way.
> 
> PS. Purchasing DIDs won't solve my underlying problem.
> 
> Thanks,
> Max.
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Re: [Freeswitch-users] Gateways, configuration or directory with mod_xml_curl

[David Knell]

Hi Eric -

The way that we do it is to keep each gateway in its own Sofia profile.
Issuing 
api sofia profile  start reloadxml
does one call to the web server for that profile's XML, which can be
pretty compact if it just contains one gateway.

--Dave

> Hello,
> 
> We are looking at Freeswitch to solve a problem of ours, we have  
> thousands of users with individual gateway information.  The user's  
> will not register with us, but we need to register with their gateways  
> on their behalf.
> 
> Since users will be constantly adding/changing/deleting gateways, we  
> figure mox_xml_curl is the best way to maintain the configuration.
> 
> What we are not sure of is if we should be putting the gateway  
> information in the directory entry for each user (gateways are not  
> shared between users) or if the gateway should go in the configuration.
> 
> If the gateway configuration is in the directory, how does sofia know  
> when the gateway configuration changes?  The reloadxml documentation  
> doesn't talk about gateways in the directory, does it work with the  
> directory.  If so, is it one curl call or one call per user?  Is there  
> a way to tell sofia that only one user's gateway configuration has  
> changed?
> 
> Likewise with gateways in the configuration.  Does reloadxml generate  
> one curl call or many?
> 
> In both cases, is there a way to minimize parsing the data that has  
> not changed?
> 
> --
> Eric Chamberlain
> 
> 
> 
> 
> 
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Re: [Freeswitch-users] More than One DID

[David Knell]

Hi Mike,

You might want to try putting them in different profiles (maybe one on
port 5080, one on 5082?) so that the provider sees them as coming from
distinct places - that way they should let you use both at once, rather
than just seeing whichever was the last to register.

Cheers --

Dave

> Here they are, mildly modified:
> 
> ~/conf/sip_profiles/external/gateway1212.xml (name changed):
> 
> 
>   
> 
> 
> >
> 
> 
> 
> 
> 
> 
> 
> 
> 
>   
> 
> 
> ~/conf/sip_profiles/external/gateway1234.xml (name changed for pseudo
> tn):
> 
> 
>   
> 
> 
> >
> 
> 
> 
> 
> 
> 
> 
> 
> 
>   
> 
> 
> The gateway name is the same for both. Note that each registers and
> passes calls both ways on their own. I'd like to use 'em at the same
> time. 
> 
> Thanks for having a look.
> 
> Mike G.
> 
> On Tue, Sep 29, 2009 at 11:21 AM, Frank Carmickle
>  wrote:
> On Tue, Sep 29, Michael Gende wrote:
> > Say, could someone please direct me to information on
> registering more than
> > one DID with a SIP provider?
> >
> > When I try this using two XML files in
> ~/conf/sip_profiles/external, I find
> > only one or the other registers (both work fine when I use
> them separately).
> 
> 
> Please post your configs with out passwords.  Thanks.
> 
> --FC
>   
> 
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Re: [Freeswitch-users] Automagic Phone Provisioning

[David Knell]

I don't think it's trivially possible, unless you can stick the PBX
between the DHCP server and the rest of the network.  The reason is that
DHCP reply packets are not broadcast, but sent back to the MAC address
of the originator, so your Ethernet switch won't even let your PBX see
the replies.

Even if it could see them, add something to them and retransmit, the
client will almost certainly already have seen the original reply and
would be likely just to ignore the later one.

Any reason why you can't just ask them to add that option to their
existing DHCP server?  

And how do you know it's done, and, if you've figured out how to do it,
could you share?

--Dave

> Hello all,
>   I know this is done and I think I figured out how to do it but I don't
> want to reinvent the wheel so here goes.  I am looking for a program
> that will sit on the PBX.  This program will intercept DHCP reply
> packets destined for phones, inject "option 66" into the packet and
> release it back onto the network.
> 
> Some of you might be wondering why I want a program like this.  Simple.
> Lazy clients.  They don't want to mess with their network infrastructure
> to assist us with automated deployment of SIP devices.  They also don't
> want 50-100 devices connecting to an off site server downloading 20-40MB
> of firmware on a reboot.
> 
> The PBX is not hard coded with an IP address.  It's DHCP.  They were
> willing to allow the PBX on the network and assign it a static DHCP
> address.
> 
> Is what I am looking for not possible?  Does someone have a sensible
> solution that doesn't involve dropping the client (yes someone suggested
> that)?
> 
> Thanks in advance for any help you can give.
> 
> DigiLord
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Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing

[David Knell]

 see only one ip where as internally all fs boxes
> >> refer db for call states, db again is under replication.
> >> This in the thioery can be written, but I am sure if we think bit more on
> >> this direction the problem seem to be getting addressed.
> >> Other guys also chip in their 2 cents, we just need 50 of em to make a full
> >> dollar.
> >> 
> >> Thanks & Regards,
> >> Mitul Limbani,
> >> Founder & CEO,
> >> Enterux Solutions Pvt. Ltd.,
> >> The Enterprise Linux Company (r),
> >> http://www.enterux.com
> >> http://www.entVoice.com
> >> 
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> >> 
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> >> 
> >> 
> > 
> > 
> > 
> > -- 
> > Jim Burke
> > Director Evolutiontel.
> > http://www.evolutiontel.net
> > 
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> 
> 
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Re: [Freeswitch-users] Caller Authentication

[David Knell]

Hi Max,

I think that the answer is generally no - for example, I'm trivially
easily able to place a call to my GSM phone with any CLI that I choose.
If you can work around that, then consider the case where I divert my
GSM phone to you - the call to you will come from the GSM network, but I
can still set an arbitrary CLI.

Cheers --

Dave

> Guys,
> 
> This question might be a bit off-topic, so please bear with me as i
> could hardly find a better place to solicit some help.
> I suspect caller authentication can be challenging in the broader
> context of VoIP/SIP. Now, if we restrict the set of calls to those
> coming from the GSM network, is there a way to tell if the Caller Id
> is coming from the right source?  It doesn't need to be a
> bulletproofed solution. For instance, if i can already say that a
> given call is certainly coming from a GSM network (and that the call
> metadata weren't tampered with), i could maybe rely on the GSM
> authentication mechanism. 
> 
> Any idea?
> Max.
> 
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Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing

[David Knell]

On Sun, 2009-08-30 at 11:17 +0800, Steve Underwood wrote:
> This sounds like so many "redundancy" projects that will probably offer 
> nothing in the real world.
> 
> On 08/30/2009 05:52 AM, Pete Mueller wrote:
> > I guess I should also mention that Xen is a side-project.
> >
> > When considering this issue for an existing production systems, we 
> > chose to put as much HA into hardware as we can.  We are not concerned 
> > with FS crashing, as so far we've never seen that happen (except when 
> > our module caused it :)  So for each of our systems:
> > - We have dual NIC cards (onboad NIC + PCI card) both bridged together 
> > in case one fails
> NICs hardly ever fail. Its the wiring which is the vulnerable area. How 
> independent can you make the two wiring paths, when they come from the 
> same box?

This is one area where you can do quite well.  A simple setup:
two machines (1, 2), two NICs (A, B) in each, two switches (S1, S2)
- wire up 1A <-> S1 <-> 2A, 1B <-> S2 <-> 2B
- run OSPF across the links
allows you to unplug any cable or any switch without interrupting
communications for more than a second or two if the OSPF timers are
suitably set.

This generalises nicely - we used to run two machines as web servers,
each advertising the same IP address via OSPF to the routers via a setup
like the one above.  Unplug any one thing, and the whole still worked.

Three complete power outages in the data center we were in in 18 months,
one of which took out a number of power supplies, neatly illustrated
Steve's point: our real-world reliability was determined elsewhere.

--Dave

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Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing

[David Knell]

; Now you have a 3 server solution (1 mediaprox, 2 loadbalanced /
> failover PBXes) out of the first box you bought, without headaches,
> because the system was built for it from the beginning!
> 
> 
> The Database drains to much?
> * Buy new machine
> * Setup Hardware Node
> * Livemigrate database VirtCnt4 (no downtime)
> 
> 
> You want to upgrade Hardware/Kernel in Hardware node 1?
> * Livemigrate VirtCnt2 to a hotstandby machine, or to the other PBX
> machine, upgrade the hardware, Re-Livemigrate the containers. (no
> downtime)
> * OR just break the loadbalancing, wait until all current calls are
> teared down correctly, upgrade machine, reenable the loadbalancer
> 
> 
> You want an exact copy of the first server for Hardware HA?
> * Buy new machine
> * Setup Hardware node
> * Buy hardware PRI switchover box
> * Clone VirtCnt1 - VirtCnt4 to the new machine
> * Make basic failover configuration 
> 
> 
> 
> 
> -> the sky's the limit, as the saying goes ...
> 
> 
> 
> 
> So, I can do all the openvz stuff and the integration with database /
> memcached / heartbeat / whatever is needed here, someone there to be
> willing to work with me on this on the FreeSWITCH side? or at least
> provide me with the necessary information about what's needed / how to
> talk / what states from FreeSWITCH?
> 
> 
> I know this seems very ambitious but if this could be made in a rather
> relativly easy to setup package, with good documentation, it would be
> a boost for FreeSWITCH, i am sure, because after all this is what
> everyone is grown accustomed to from good old phone companys and the
> good old pbx's: carrier grade uptimes ...
> 
> 
> Thanks for everyone reading up until here,
> all the best,
> 
> 
> Ray
> 
> 
> 
> 
> 
> 
> -- 
> Raimund Sacherer
> -
> RunSolutions
> Open Source It Consulting
> -
> 
> Parc Bit - Centro Empresarial Son Espanyol
> Edificio Estel - Local 3D
> 07121 -  Palma de Mallorca
> Baleares
> 
> On Aug 29, 2009, at 3:17 PM, Raimund Sacherer wrote:
> 
> > Oh yeah, that would be so helpfull for my situation, as my client
> > *demands* now a solution where he can press a big red button and all
> > fails over to another box. Hi es totally scared because of the
> > Lockups in Asterisk which under specific situations including AMI,
> > Automated Call Setup, and murphy led to a lockup of the entire
> > machine, no console was working anymore, only cold-reset could do
> > it.
> > 
> > 
> > So, IF there is the possibility for life-takeover, / failover etc. I
> > would love to here how has been done.
> > 
> > 
> > I am very experienced with openvz and use for about two years now
> > only openvz virtualization servers for anything because of
> > live-migration etc. But as I am new in this company we could not
> > adopt this until now.
> > 
> > 
> > So Please Ken, if you can, describe what need's to be done to get a
> > failover / takeover working (an outline would be enough)
> > 
> > 
> > Thanks in Advance
> > 
> > -- 
> > Raimund Sacherer
> > -
> > RunSolutions
> > Open Source It Consulting
> > -
> > 
> > Parc Bit - Centro Empresarial Son Espanyol
> > Edificio Estel - Local 3D
> > 07121 -  Palma de Mallorca
> > Baleares
> > 
> > On Aug 29, 2009, at 11:58 AM, Steve Kurzeja wrote:
> > 
> > > On Sat, Aug 29, 2009 at 2:34 PM, Diego Viola
> > >  wrote:
> > > Yes, FreeSWITCH is a system that you can trust 100%. I
> > > have switched my Asterisk servers to FreeSWITCH and have
> > > peace now.
> > > 
> > > If I were you I would get rid of Asterisk and use
> > > FreeSWITCH, FS will handle all what you want very well.
> > > 
> > > And I agree with David, fail-over is kinda irrelevant
> > > since the FS doesn't crash like Asterisk does.
> > > 
> > > 
> > >  
> > > You still have hardware failures and fail-over is also useful for
> > > hit-less maintenance on boxes.
> > > 
> > > I'd be interested to know how Brian West was approaching his live
> > > migration work. 
> > > 
> > > Steve
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Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing

[David Knell]

Hi Diego,

We didn't use mod_fifo; we built our own queues using an application
hanging off the event socket.  This was partly down to typical
programmer hubris, and partly to allow us to do things that mod_fifo
might not without either (a) trawling through mod_fifo, or (b) pestering
Anthony.

As a brief outline:
- operators each live in their own conference
- supervisors can add themselves to these conferences for monitoring or
coaching
- callers sit in queues for the various skills that the operators have.

When an operator is free, look at each queue for the skills that the
operator has, and choose the caller who's been waiting longest.

I used a one-thread-per-call model, so there's a little bit of thought
needed to get the access to shared structures right but, essentially,
given the low frequency of access to them and the fact that nothing was
time-critical at the sub-second level, I just wrapped all of the
accesses in a single global mutex, which was easy and is pretty
foolproof.  The latter is important, given that it was me writing the
stuff ;-)

Cheers --

Dave

> Hi David,
> 
> What have you used on FS for call center, mod_fifo?
> 
> Can you describe your experience with that, I'm currently interested
> in call center + FS scenario.
> 
> Diego
> 
> On Fri, Aug 28, 2009 at 9:01 PM, David Knell  wrote:
> Hi Raimund,
> 
> One FreeSWITCH box will be quite enough to handle the call
> volumes that
> you're talking about, and it ought to be much more stable than
> the
> Asterisk solution which you've outlined below.
> 
> It's probably best to forget about live failover without calls
> dropping
> - this isn't something that's supported, and there'd be a lot
> of work to
> do to develop code to keep two boxed in sync.  Once you get
> used to a
> stable solution - i.e. something which doesn't crash - then
> live
> failover, HA, etc., will seem somewhat irrelevant.
> 
> I recently did some work on an FS-based call center solution -
> drop me a
> note if I could be of any help with yours.
> 
> Cheers --
> 
> Dave
> 
> 
> > Hello List,
> >
> > I have read the current thread about scalability and I would
> need some
> > advice about a callcenter setup:
> >
> > First where I come from:
> > I have lot's of problems with an asterisk solution. I have
> regular
> > crash's and lock-ups, with downgrading and other stuff i got
> it
> > somewhat stable, but have nevertheless regular hickups. I am
> desperate
> > and want to get rid of asterisk and I hope that freeSwitch
> will
> > provide me with a more stable solution.
> >
> >
> > Our Setup (really nothing special):
> > * 1 Asterisk box, New IBM Hardware (3 month old), 2 HE rack
> server, 3
> > GIG of RAM, Xircom analog switch connected to mobile
> stations for 4
> > different providers, Digium 4port cards TP400
> > * 8 queues
> > * ~60 agents (which logon, logoff, pause, unpause), not more
> than 40
> > concurrently online
> > * ~ 7K - 9K calls (well, CDR entries) a day (not that much
> for a bpx)
> > * Music on Hold in the call-queues
> > * No special announcement
> > * Transfers between calls in queues and different agents as
> well as
> > non agents (i mention this because we have transfer related
> chrashes
> > in asterisk)
> >
> > The current Problems:
> > * Lockups with different causes (ranging from calls not
> terminated to
> > heavy thread locking through the AMI interface)
> > * Crashes and library aborts (pthread, libc, crashes related
> to music
> > on hold, app_queue, transfers)
> >
> > We used Asterisk: 1.4.23, 1.4.24, 1.4.26rc3, 1.4.26rc5,
> 1.4.26 and are
> > now back to 1.4.21.2 (stock debian) as anything beyond that
> is for
> > whatever reason highly unstable for our szenario. Maybe we
> should have
> > been segmenting the box into one asterisk dedicted to
> talking to the
> > hardware, one especial for queue/sip handling, i do not
> know. (all

Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing

[David Knell]

Hi Raimund,

One FreeSWITCH box will be quite enough to handle the call volumes that
you're talking about, and it ought to be much more stable than the
Asterisk solution which you've outlined below.

It's probably best to forget about live failover without calls dropping
- this isn't something that's supported, and there'd be a lot of work to
do to develop code to keep two boxed in sync.  Once you get used to a
stable solution - i.e. something which doesn't crash - then live
failover, HA, etc., will seem somewhat irrelevant.

I recently did some work on an FS-based call center solution - drop me a
note if I could be of any help with yours.

Cheers --

Dave

> Hello List,
> 
> I have read the current thread about scalability and I would need some  
> advice about a callcenter setup:
> 
> First where I come from:
> I have lot's of problems with an asterisk solution. I have regular  
> crash's and lock-ups, with downgrading and other stuff i got it  
> somewhat stable, but have nevertheless regular hickups. I am desperate  
> and want to get rid of asterisk and I hope that freeSwitch will  
> provide me with a more stable solution.
> 
> 
> Our Setup (really nothing special):
> * 1 Asterisk box, New IBM Hardware (3 month old), 2 HE rack server, 3  
> GIG of RAM, Xircom analog switch connected to mobile stations for 4  
> different providers, Digium 4port cards TP400
> * 8 queues
> * ~60 agents (which logon, logoff, pause, unpause), not more than 40  
> concurrently online
> * ~ 7K - 9K calls (well, CDR entries) a day (not that much for a bpx)
> * Music on Hold in the call-queues
> * No special announcement
> * Transfers between calls in queues and different agents as well as  
> non agents (i mention this because we have transfer related chrashes  
> in asterisk)
> 
> The current Problems:
> * Lockups with different causes (ranging from calls not terminated to  
> heavy thread locking through the AMI interface)
> * Crashes and library aborts (pthread, libc, crashes related to music  
> on hold, app_queue, transfers)
> 
> We used Asterisk: 1.4.23, 1.4.24, 1.4.26rc3, 1.4.26rc5, 1.4.26 and are  
> now back to 1.4.21.2 (stock debian) as anything beyond that is for  
> whatever reason highly unstable for our szenario. Maybe we should have  
> been segmenting the box into one asterisk dedicted to talking to the  
> hardware, one especial for queue/sip handling, i do not know. (all  
> issues are well documented in issues.asterisk.org, but it seems to be  
> very, very difficult to get to the bottom of them as they exist since  
> 1.4.23 as it seems and are open until know and not fixable since month.)
> 
> 
> Now, I really would appreciate some success-stories on how you guys  
> managed to get a stable pbx system with freeSWITCH in regard of HA and  
> scalability:
> 
> * How to segment freeSWITCH? Or is it stable enough to handle all in  
> one for such a szenario as outlined above?
> * What would be the best strategie for High Availability / Failover?
>   -> I read in the WIKI (featurelist) that Livemigration of calls from  
> one box to another should be possible?
>   -> I was thinking about using memcached for storing all state  
> information so another freeswitch box can takeover calls from the  
> first box if it dies, is this possible? If so, how?
>   -> Is there anotherway to somehow configure freeSWITCH that in the  
> event of a crash i do not loose the current established calls?
> 
> Basically I just want a stable PBX where I do not have to fear every  
> day it will core-dump or abort or Lock up. Is freeSWITCH mature enough  
> so i can sleep at night for at least 3 month without a crash?
> 
> 
> 
> Thank you for your Time and help in advance, and I am more than  
> willing to take all the information gathered here and create a wiki  
> page to help other people with the same questions/problems.
> 
> best
> Ray
> 
> 
-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: d...@3c.co.uk
W: http://www.3c.co.uk


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Re: [Freeswitch-users] Odd sonus warning

[David Knell]

On Thu, 2009-08-27 at 11:06 -0400, Greg Thoen wrote:
> Hi, I have some DIDs from Bandwidth.com and when they call in I see
> this in the console:
> 
> 
> 2009-08-27 11:01:47 [WARNING] sofia_glue.c:2701
> sofia_glue_negotiate_sdp() Hello,
> I see you have a Sonus!
> FYI, Sonus cannot follow the RFC on the proper way to send DTMF.
> Sadly, my creator had to spend several hours figuring this out so I
> thought you'd like to know that!
> Don't worry, DTMF will work but you may want to ask them to fix
> it..
> 
> 
> Anything I should worry about?

Not unless you're a Sonus shareholder - see
http://messages.finance.yahoo.com/Stocks_(A_to_Z)/Stocks_S/threadview?m=te&bn=16942&tid=827769&mid=-1&tof=15&rt=1&frt=2
for an amusing read ;-)

--Dave


-- 
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T: +44 20 3298 2000
E: d...@3c.co.uk
W: http://www.3c.co.uk


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Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message

[David Knell]

BYE, CANCEL, OPTIONS, PRACK,
> MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER,
> INFO, PUBLISH
>Supported: timer, precondition, path, replaces
>Allow-Events: talk, presence, dialog, call-info,
> sla, include-session-description, presence.winfo,
> message-summary, refer
>Content-Length: 0
> 
> 
> Tihomir.
> 
> 
> 
> 
> 
> 
> ___
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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 Conference
> sip:8...@conference.freeswitch.org
> iax:gu...@conference.freeswitch.org/888
> googletalk:conf+...@conference.freeswitch.org
> pstn:213-799-1400
> 
> 
> -- Forwarded message --
> From: "Raffaele P. Guidi" 
> To: freeswitch-users@lists.freeswitch.org
> Date: Mon, 24 Aug 2009 20:24:28 +0200
> Subject: Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows
> installer - great but I have a little problem
> Actually I did that and it worked fine. I had the problem the
> SECOND time I run FS and freepbx. And (@Brian) mod_sofia was
> loaded but sip_profiles were not
> 
> On Sun, Aug 16, 2009 at 16:04, Carlos Talbot
>  wrote:
> When you configure FreePBX for the first time it wipes
> out the sip_profiles directory. If you follow the
> FreePBX shortcut on your desktop it'll mention this on
> the last screen of the configuration. You might see
> something such as the following below. If you plan to
> use FreePBX you'll need to define trunk groups,
> trunks, etc in order to have the sip_profiles
> directory populated.
> 
> 
> regards,
> 
> 
> Carlos
> 
> 
> 
> 
>   Incompatible Configuration
> WARNING: THE FOLLOWING FILES WILL BE DELETED!
> 
>   * D:/FreeSWITCH/conf/sip_profiles/external.xml
>   * D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml
>   * D:/FreeSWITCH/conf/sip_profiles/internal.xml
> 
> 
> On Sun, Aug 16, 2009 at 4:43 AM, Raffaele P. Guidi
>  wrote:
> 
> 
> I had the sweet surprise to find the installer
> packaged with FreePBX... really great! Why it
> has not been advertised as it deserves? It
> worked like a breeze once launched, with the
> automatic configuration and all of that., Only
> thing: once stopped I cannot get it to load
> sofia profiles anymore - issueing sofia status
> doesn't show anything. I had to copy
> internal.xml and default.xml from a previous
> installation and everything got to work again
> - but no FreePBX anymore :( I'm sure I'm
> missing something important.
> 
> 
> Can you give me a hint? Should sofia profiles
> be served by curl or something?
> 
>     
> Thanks,
>Raf

Re: [Freeswitch-users] Choppy voive in conference from fritzbox to freeswitch (ptime:30 msecs)

[David Knell]

On Thu, 2009-08-20 at 14:35 -0700, Michael Collins wrote:


> Just curious - if it seems to be working with Asterisk but not
> FreeSWITCH then could you do some tcpdumps of working vs. non-working
> calls and then analyze them with Wireshark? I think Jason Garland's
> ClueCon presentation(s) might be applicable here. 

Just to deepen the mystery a little, we have a FRITZ!Box here in Greece,
and it works like a little champ for us.  Firmware's 06.04.49, it's
talking to a FreeSWITCH box in London.  It's set to pick its own codec
(but the other end only supports G.711), VAD's off.

Cheers --

Dave




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Re: [Freeswitch-users] Recommendations for an interface between FreeSwitch and ANSI C code

[David Knell]

On Thu, 2009-08-20 at 21:35 +1200, Matt Riddell wrote:
> We have an application written in ANSI C which currently talks to the 
> Asterisk Manager to make phone calls.
> 
> We're possibly looking at converting this to FreeSwitch.
> 
> At the moment it has an abstraction layer from Asterisk and speaks to 
> between 1 and 80 Asterisk machines using round robin for distribution.
> 
> Would you recommend that I use the mod_event socket and basically work 
> with FreeSwitch in the same way as I work with Asterisk or am I 
> overlooking a possibly different way of doing things?

Just to echo the sound advice from others, the event socket is almost
certainly the right place to start.  We've used it exclusively for our
stuff for getting on for two years, and I've no regrets.

--Dave

-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: d...@3c.co.uk
W: http://www.3c.co.uk


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Re: [Freeswitch-users] Loopback and bypass_media

[David Knell]

Just to add my $0.02-worth (if you're feeling generous..) - I don't
think that the dialplan is expressive enough to do what's needed here,
and that's where the trouble's coming from.  It's not enormously tricky
to build a generic "dial this set of numbers according to these rules"
service using something hanging off the event socket - there's a writeup
here: http://www.softivr.com/wiki/index.php/Find_me showing how it could
be done on SoftIVR.

To roll something similar yourself using the event socket, you'd need to
map the dial function to 'originate', bridge to (IIRC) 'uuid_bridge',
and have some way of passing messages around between the threads
handling the different call legs, assuming that you're using one thread
per leg.

--Dave


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Re: [Freeswitch-users] VoiceMail transcription

[David Knell]

Hi Pete,

I'm afraid that the answer's still the same: use a human.  Here's an
article describing the state of the art:
http://www.theregister.co.uk/2009/08/05/spinvox_demo_day/
- the links to previous stories at the bottom provide good background.

--Dave

> I apologize, I should have been more clear.  We will be using humans
> to scan the translated results.  But we are looking for a system to
> perform the "first pass" on the audio to hopefully help the human type
> less.
> 
> Although the question has been raised if it's faster to have a human
> just transcribe the whole thing, or fix up what the computer spit out.
> If you have any insights on this, that would be great.
> 
> -pete
> 
>  Original Message 
> Subject: Re: [Freeswitch-users] VoiceMail transcription
> From: David Knell 
> Date: Mon, August 10, 2009 11:51 am
> To: freeswitch-users@lists.freeswitch.org
> 
> Good evening Pete,
> 
> The only way to do this is, I'm afraid, to use a human. We use
> Amazon's
> Mechanical Turk to good effect.
> 
> Cheers --
> 
> Dave
> 
> > Good morning all,
> > 
> > I realize this is slightly off the FS topic, but I am
> wondering if
> > anyone out there has experience with software packages
> designed for
> > the transcription of voicemails to text. I've used
> pocketsphinx with
> > FS to handle IVR menus, but now have the task of figuring
> out how to
> > convert recorded phone conversations (voicemails mostly) to
> text.
> > 
> > This does not have to be a real-time process, I can store
> the audio
> > files and process them over time. This would need to be a
> software
> > (preferable open source) solution. ASPs like VoiceCloud
> would not
> > work for this application.
> > 
> > Thanks for any help
> > -pete
> > ___
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> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> -- 
> David Knell, Director, 3C Limited
> T: +44 20 3298 2000
> E: d...@3c.co.uk
> W: http://www.3c.co.uk
> 
> 
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Re: [Freeswitch-users] VoiceMail transcription

[David Knell]

Good evening Pete,

The only way to do this is, I'm afraid, to use a human.  We use Amazon's
Mechanical Turk to good effect.

Cheers --

Dave

> Good morning all,
> 
> I realize this is slightly off the FS topic, but I am wondering if
> anyone out there has experience with software packages designed for
> the transcription of voicemails to text.  I've used pocketsphinx  with
> FS to handle IVR menus, but now have the task of figuring out how to
> convert recorded phone conversations (voicemails mostly) to text.
> 
> This does not have to be a real-time process, I can store the audio
> files and process them over time.  This would need to be a software
> (preferable open source) solution.  ASPs like VoiceCloud would not
> work for this application.
> 
> Thanks for any help
> -pete
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[Freeswitch-users] Cluecon 2009

[David Knell]

Just a quick note to say thanks to Cluecon's organisers for putting
together such a useful, informative and packed three days.  I've come
away with a head full of ideas, a bunch of new contacts and a collection
of things to do; I'd thoroughly recommend that anyone interested in IP
telephony blocks out the first week of August 2010, right now..!

Cheers --

Dave

-- 
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T: +44 20 3298 2000
E: d...@3c.co.uk
W: http://www.3c.co.uk


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Re: [Freeswitch-users] arriving today for ClueCon

[David Knell]

-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: d...@3c.co.uk
W: http://www.3c.co.uk



On 3 Aug 2009, at 13:41, Brian West  wrote:

> Just look for large groups of people with laptops.  I'm sure you can't
> miss us.
>
> /b
>
> On Aug 3, 2009, at 5:46 AM, Chad Phillips -- Apartment Lines wrote:
>
>> i'm arriving around 12:30PM -- are there any pre-ClueCon meetups this
>> afternoon?  anybody need help with setup?
>
>
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Re: [Freeswitch-users] FS beats Aculab Prosody S on subjective test on lay users for conference quality

[David Knell]

On Thu, 2009-07-30 at 09:21 +0800, Steve Underwood wrote:
>  
> High quality conferencing is a difficult task, and still a research 
> topic. No two conferencing systems perform alike. The interesting thing 
> about this and other reports is that the conferencing in Freeswitch is 
> not very clever right now, yet people are already saying it beats 
> various other offerings, including long time commercial offerings.

It may well be that a simplistic implementation (noise gate, add them
all up) is all that's required for dealing with small groups or, more
generally, groups of any size which have a small number of active
speakers at any one time: it's predictable and unlikely to introduce
unpleasant side effects.  

--Dave


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Re: [Freeswitch-users] FS beats Aculab Prosody S on subjective test on lay users for conference quality

[David Knell]

Hi Fernando,

Greetings from Rio..!

It'd be interesting to understand more about these results - roughly
speaking, two conferencing systems using the same codecs, etc., ought to
perform pretty much identically, particularly with just a few callers.

I'd be interested to see a network packet dump of a conference on each
of the machines, if you're able to make one available.

Cheers --

Dave


> Hello freeswitchers!
> 
> 
> I thought you're be pleased to know: FS beated Prosody S on subjective
> test on lay users. 
> After 3 users testing the two conferences and reporting far better
> quality on FS, we decided to make a more rigorous test.
> Then we made a blind test by asking 4 lay users to dial two urls for
> 10 minutes from their x-lite and report the results. These users are
> non-techie and never heard about FS at all.
> The first url was Aculab Prosody S test app, then, a second call to FS
> on default conf (extension 3000).
> All users preferred FS. Some users were almost enthusiastic after
> entering FS conf. Some even started to blame ProsodyS after hearing
> FS. The average report told 50% more satisfaction on FS. Questions on
> report were simple ones, with results from 1(worse) to 5(best) :
> 1-Overall opinion(1-crap; 5-awesome);
> 2-Delay (1-more delay, 5-no delay at all);
> 3-Audio quality (1-noisy, choppy, etc; 5-cristal clear);
> 
> 
> Machines used:
> Prosody S: Windows Vista Business, Intel Core 2 Duo E7400 @ 2.8
> GHz, 4Gb de RAM
> FreeSwitch: CentOS 5.4, Intel(R) Pentium(R) 4 CPU 2.40GHz, 1Gb RAM
> 
> 
> Congratulations! You made a great product!
> -- 
> Fernando Gregianin Testa
> Voice Technology Ltda
> +55 11 35882166
> 
> 
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Re: [Freeswitch-users] Application to Record Calls - Out of Band

[David Knell]

Hi Matt,

FreeSWITCH probably isn't what you want.  A quick Google for 'sip call
sniffer' found this: http://www.enderunix.org/voipong/ which might well
be a more appropriate starting point.

A SPAN port is just a port on a network switch which has the traffic
going to/from another port (or ports) replicated to it.

Cheers --

Dave

> Hi,
> 
> 
> I'm trying to build an application that provides statistics of calls
> and call recording. Someone told me this could be done out of band
> with a SPAN (?) port that would replicate SIP and media packets to a
> separate NIC without having to actually pass the real-calls thru
> FreeSWITCH. It was explained that this SPAN port would in the SBC
> would replicate data received.
> 
> 
> If this is done, is there a way I can utilize FreeSWITCH to interpret
> these packets without actually having any control of the calls? If so
> how? Sorry, I'm new to telco, so hopefully this post makes sense to
> someone.
> 
> 
> --matt
> 
> 
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Re: [Freeswitch-users] Barge on on prompts

[David Knell]

Hi Phillip,

You need to call FreeSWITCH's break function - I'd guess
Session.Break(); might do it for you, but no guarantees.

--Dave

> Hi there,
> 
> Thanks for the reply. That information is extremely useful.
> 
> Given the code below though - when if I press '1' when the phrase is
> playing - playing does not stop. It continues. I am looking for a
> method to barge in and collect & react to digits immediately.
> 
> 
> Session.DtmfReceivedFunction = (d, t) =>
> {
> Log.WriteLine(LogLevel.Info, "Received {0} for
> {1}.", d, t);
> Session.StreamFile("", 0);
> CollectedDigits = d.ToString().Trim();
> return "";
> 
> };
> 
> Session.SayPhrase("msgcount", "187346", "en");
> 
> 
> Any ideas? I am sure I must be missing something simple.
> 
> Thanks a lot.
> 
> 
> Phillip Jones
> 
> 
> 
> 
> 
> On Thu, Jul 23, 2009 at 2:40 PM, Michael Collins 
> wrote:
> I think you might want to check out phrase macros...
> http://wiki.freeswitch.org/wiki/Speech_Phrase_Management
> -MC
> 
> 
> On Thu, Jul 23, 2009 at 11:13 AM, Phillip Jones
>  wrote:
> 
> 
> Hi there,
> 
> Very simple scenario:
> 
> Session.DtmfReceivedFunction = (d, t) =>
> {
> Log.WriteLine(LogLevel.Info, "Received {0} for
> {1}.", d, t);
> CollectedDigits = d.ToString().Trim();
> return "";
> };
> 
> 
> Session.flushDigits();
> Session.StreamFile(VoicemailPromptsDirectory +
> "abigfile.wav", 0);
> 
> Question is, it there a way to kill the streaming when
> the a digit is pressed?
> 
> I would use the Session.PlayAndGetDigits()
> 
> but that does not help when want to string things
> together like:
> 
> Session.StreamFile(VoicemailPromptsDirectory +
> "vm-to_delete_the_message.wav", 0);
> Session.StreamFile(VoicemailPromptsDirectory +
> "vm-press.wav", 0);
> Session.Say("7", "en", "number", "pronounced");
> 
> Any help would be appreciated.
> 
> 
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Re: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid

[David Knell]

Really?  That line (+/- the IP address) came directly out of a working
dialplan.  To be fair, the box is running a faintly prehistoric
FreeSWITCH - you crazy cats haven't been chewing on the tail of my
cherished mouse of backwards compatibility again, have you?!

What has been incorrect in this discussion is the name of the header:
it's P-Asserted-Identity, not P-Asserted-ID.  The fact that it's usually
shortened to PAID doesn't help; nor does the fact that Remote-Party-ID
(which is deprecated, but still widely used for the same job as
P-Asserted-Identity) is about as well.

--Dave

> Kinda wrong there!
> 
> 
> Gotta use CDATA because it has < and > in the data you're setting.
>  And you'll wanna export it I suspect.
> 
> 
> 
> 
> 
> /b
> 
> 
> 
> 
> 
> On Jul 16, 2009, at 9:51 AM, David Knell wrote:
> 
> > Hi Dale,
> > 
> > You can set the header to anything you like by including something
> > along
> > the lines of
> > 
> > in your dialplan.
> > 
> > Cheers --
> 
> 
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Re: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid

[David Knell]

Hi Dale,

You can set the header to anything you like by including something along
the lines of

in your dialplan.

Cheers --

Dave

> Hello again,
> 
> I wanted to first say thanks to Brain for helping me fix my from  
> domain issue the other day. It helped quite a bit.
> 
> Now with more testing and talking with the vendor (please don't shoot  
> the messenger :) )
> 
> They want the caller id info in the from and the charge number/ 
> screening number in the P-Asserted-ID.
> 
> I have tested this and verified that this does work like they say it  
> does by setting the callerid number to my charge number and setting  
> the from user in the gateway config to the callerid I want displayed.  
> But this solution doesn't scale very well.
> 
> I know I can set the gateway option caller-id-in-from to get that part  
> done. But is there a way to set the P-Asserted-ID to something other  
> than the callerid?
> 
> Any hints would be welcomed.
> 
> Thanks,
> 
> -Dale
> 
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Re: [Freeswitch-users] Even socket packets/chunks

[David Knell]

Hi -

You can't assume that 1 packet=1 command/event - it's true often enough
to lull you in to a false sense of security, but false often enough that
you'll end up with odd problems unless you do things properly.

In any case, it's not hard to get it right - there's plenty of other
instances where applications have to read from a socket until they hit
\n\n, then possibly read content-length bytes: HTTP for a starter.

--Dave

> Hi,
> 
> 
> You confirm if FS ever sends partial or incomplete commands over the
> event socket? I have heard from other developers that the commands I
> am listening for may come across incomplete.
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Re: [Freeswitch-users] Contemplating a jump from Asterisk to Freeswitch

[David Knell]

Hi Geoff,

> One of the benefits of our architecture is that our business logic is
> completely abstracted from the asterisk system. We use a combination
> of FastAGI and AMI to control channels on the asterisk server. We have
> a Java based server which interfaces with the higher level call
> routing engines. It looks to me like the Mod_event_socket would
> probably satisfy my requirements for controlling the calls via an
> external process, although it doesn't look as cut/dry as the FastAGI
> model. I haven't seen anything which would let me know the equivalent
> of the FastAGI 'script' being requested.

Three possibilities spring to mind:-
* have each distinct 'script' listen on a different socket;
* set a variable in the dialplan to a script name or other identifier
before making the outbound socket connection;
* have your event socket handler work out what to do itself based on the
dialled number, or whatever other criteria you'd use.

> The other thing I haven't seen is how to dynamically create
> conferences on the fly and redirect channels into them. We use
> app_conference on asterisk to avoid the ztdummy issue. Once the higher
> level intelligence engine determines two channels need to speak with
> each other, they are both redirected via AMI Redirect into a dynamic
> Conference created just for that particular call.

Choose a (unique) conference ID, and execute
conference 
on each of the channels.

> Also - what is the status of call progress on FreeSwitch? Some things
> that are important to me are answering machine detection as well as
> detecting SIT intercept tones in the early media stream... any love
> here?

Not sure on these, but I'm *am* sure that someone else will be ;-)

Cheers --

Dave

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Re: [Freeswitch-users] Baby Update!

[David Knell]

On Fri, 2009-07-03 at 17:43 +0600, Muhammad Shahzad wrote:
> Congratulations to Ray and Samantha. Lets see what new features and
> bug fixes we will get in their "new version"..! ;-)

Bug fixes..?!  I'd refer you to Philip Larkin (went to my school, a bit
before my time, poet, deceased, recently voted "Britain's favourite
poet") whose "This Be The Verse" suggests otherwise:
http://www.artofeurope.com/larkin/lar2.htm

[as a recent father myself, I'm trying to prove him wrong..]

--Dave

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Re: [Freeswitch-users] Language Handling: call for assistance

[David Knell]

On Thu, 2009-07-02 at 15:47 -0300, Raul Fragoso wrote:
> On Fri, 2009-07-03 at 01:29 +0800, Steve Underwood wrote:
> > Michael Collins wrote:
> > >
> > >
> > > On Thu, Jul 2, 2009 at 3:01 AM, Igor Neves  > > <mailto:i...@3gnt.net>> wrote:
> > >
> > > Hi,
> > >
> > >
> > > Michael Collins wrote:
> > >> Hello all!
> > >>
> > >> There's been some discussion lately on how to handle multiple
> > >> languages, specifically with the *say* application. We would like
> > >> some input from the community on how to handle multiple languages
> > >> and sound files. Anthony notes that the say application needs to
> > >> build the path to the sound files by using the ${sound_prefix}
> > >> and ${lang} variables. Some have asked about countries or
> > >> language variants, like European Portugese vs. Brazilian
> > >> Portugese. These are good questions.
> > >
> > > What it's the problem about Portuguese VS Brazilian?
> > >
> > > Can't we just use "PT_pt" and "PT_br" in ${lang}, just like a lot
> > > of others softwares do?
> > >
> > > What about ${sound_prefix} = ${lang}, since ${lang} should always
> > > be unique, and you make the path's automatically language organized?
> > >
> > >
> > > This is reasonable to me, but it would be nice to have a consensus, 
> > > just to be sure.

The - thing would appear to be the obvious choice; what's
slightly less obvious is what to do about fallback.  Three choices:
- None - i.e. I say en-gb, I either get en-gb or nothing;
- Best guess - I want en-gb, but I'm quite happy with en-us or en-au.  I
could specify en-gb and, if en-gb's not available but en-something else
is, whatever's available gets chosen for me;
- Preference list - I specify a (possibly) wildcarded list of what I
want - e.g. en-gb,en-us,en* - and whichever match comes first is what I
get.

--Dave

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Re: [Freeswitch-users] Is there any license G729?

[David Knell]

On Tue, 2009-06-30 at 01:31 +0800, Steve Underwood wrote:
> Video calls are a really really bad idea. People who think otherwise 
> really haven't thought about it at all. They are available here, and 
> people desperately don't want them to be.

Video calls between 3G phones have been available in the UK for some
years.  I've only ever made two using one of these to real people, both
across a table in a pub to show them what it looked like; I have made
goodness knows how many to an IVR while trying to pick 3G-324M apart.

However, there are some instances where they're very useful.  For
example, my family is geographically quite dispersed, and we use Skype
with video a lot - particularly for grandparents to keep up with
grandchildren.  The usefulness and appropriateness of video calling
depends very much on the target market; it's not, of itself, a bad
thing.

--Dave

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Re: [Freeswitch-users] Announcement: Howler-optimised G.729A Solution for FreeSWITCH

[David Knell]

On Fri, 2009-06-26 at 10:43 -0500, Will Boyce wrote:
> Hey David,

[duplicate post snipped]

-- don't suppose your product includes an echo canceller, does it ;-)

Cheers --

Dave

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Re: [Freeswitch-users] Announcement: Howler-optimised G.729A Solution for FreeSWITCH

[David Knell]

> There doesn't seem to be any direct link between FreeSWITCH and Howler.  
> How does this benefit the project? Can we assume that Howler is "giving 
> something back"?

Well, they're providing additional functionality which a lot of folk
want, which requires substantial investment in money (visit sipro.com
and have a look at the up-front license fees if you wish) and time (to
get the thing working across the various platforms with the required
licensing) and they've presumably taken a commercial risk so to do.

Hats off to them: they've taken a risk, and the project benefits
immensely from it.  Expecting them to "give something back" in addition
would be entirely unreasonable.

--Dave

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Re: [Freeswitch-users] FS as a Class 5 switch

[David Knell]

Hi Nandy.

On Sun, 2009-06-21 at 08:58 +0800, Nandy Dagondon wrote:
> i'm interested to know if anyone employed FS as a local exchange
> switch. i'm confident FS can handle several calls using RTP by-pass
> mode. however, i'm more concerned on handling the large dialplan with
> hundreds (or even a few thousand) exchange prefixes nationwide during
> call setup.

We have probably ~100k prefixes in our LCR.  We don't put these in the
dialplan directly; instead, they live in a database and we have an
external application which routes calls.  FreeSWITCH has mod_lcr which I
would imagine will do the same sort of thing; we don't use it because it
wasn't around when we started.

I'd caution against trying to put thousands of prefixes in the dialplan:
I'd guess that matching each call against some thousands of regexes
during call setup might get expensive.

> i'd be glad to hear experiences and suggestions esp on the hardware
> dimensioning. we're talking a small exchange up to about 1,100 lines
> only, mostly linked to the main exchange via MFC-R2.

That'd depend on the number of concurrent calls you need to budget for -
taking it that 1,100 lines implies maybe 1-200 simultaneous calls, then
one low-end modern server (Core 2 Duo, etc.) ought to do just fine.

Cheers --

Dave

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Re: [Freeswitch-users] FreeSWITCH to sell Content, tones, MMS, etc ?

[David Knell]

Hi Edwin,

Rather than using a GSM/3G card, you might do better to find a mobile
services aggregator which covers the locations you're interested in -
MBlox or Sybase365 would be two places to start - and use them.  You'll
get scalability, better reliability, etc.; be warned that MMS is *still*
a pain.

--Dave

> Hello, I am planning to build a plataform to sell content, pictures,
> tones, MMS, etc.   
> 
>  
> 
> Do you know wich GSM 3G boards should work?  Anyone has done this?
> 
>  
> 
> Greetings!
> 
> Edwin
> 
> 
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Re: [Freeswitch-users] Sample rate and recordFile

[David Knell]

- plus UDP/RTP overhead.  Budget 10 calls/megabit for G.711 and you'll
have a bit of headroom available.

--Dave

> Most calls are at 8kHz. The formula for bandwidth is sampling rate *
> bit rate * channels, so 8000 * 8 * 1 = 64000 = 64kbps (per way).
> 
> 
> Math
> 
> On 18-Jun-09, at 1:01 PM, Andy wrote:
> 
> > Thanks Brian, 
> >  
> > So, just to calrify will the base call always be 8kHz? 
> >  
> > On a related note, do you happen to know the bitrate of each open
> > channel/live call? Is it 16 kilobits per second like the recorded
> > audio? I need to do some calculations on the badwidth required to
> > handle a certain number of concurrent calls.
> >  
> > Many thanks
> > Andy
> > 
> > 
> > 
> > From: freeswitch-users-boun...@lists.freeswitch.org
> > [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
> > Brian West
> > Sent: 18 June 2009 18:11
> > To: freeswitch-users@lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Sample rate and recordFile
> > 
> > 
> > 
> > 
> > On Jun 18, 2009, at 11:54 AM, Andy wrote:
> > 
> > > 1) I notice that when I change the sample rate it automatically
> > > changes the bit rate too. I understand why this is the case but
> > > wondered if it was just as easy to be able to control the bitrate
> > > as well as the sample rate.
> > 
> > 
> > If you're talking about mod_shout, NO.  You'll end up picking an
> > invalid bitrate and asking why it doesn't work... been there done
> > that... I changed it a few months back to pick the optimal bitrate
> > for the sample rate.
> > 
> > > 2) When I use a sample rate other than 8000 I get a warning
> > > 'Sample rate doesn't match'. I guess this puts some extra load on
> > > the server. If all my calls are being recorded and all at 11025
> > > can/should I alter the sample rate of the base call to 11025?
> > 
> > NO.  Your phone call is running at 8kHz, Your sound file is 11025
> > and they don't match, If you were to play this file into an 8k
> > channel without a resample it would sound a little like satan.  or a
> > dragging tape deck.  The file has to be resampled to match the
> > current session rate. 
> > 
> > 
> > /b
> > 
> > 
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Re: [Freeswitch-users] funny effect after minimizing xml files

[David Knell]

You've probably deleted the start/end markers from your dialplan
matches..?  It might be easier to help if you posted or pastebinned your
dialplan.

--Dave

> Hello I've minimized de xml files where possible to make a dialplan
> that is as short as possible. Now do I've this funny effect to dial my
> extensions who are running from 200 to 207. It seams that I'm able to
> dial an extension in closed in a number. So for instants if I dial
> 120275 extension 202 will ring even tried it whit two extensions in a
> number like 202205 . This results in the first extension ringing so
> 202205, 202 will ring 205202, 205 will ring. At this time I'm unable
> to pinpoint the cause of this behaviour. Could someone point me to the
> cause of this effect
> 
> 
> /d
> 
> 
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Re: [Freeswitch-users] Is Freeswitch ready for prime time?

[David Knell]

On Wed, 2009-06-10 at 10:02 -0700, Paul Mahler wrote:
> What is the current status of Freeswitch? Can I safely use it in a  
> large scale commercial environment? How active is the Freeswitch  
> developer community?

Hi Paul -

We've used FS over the last 18 months or so to handle millions of calls
- some wholesale in/out, some IVR, some calling card, some callthrough -
with a total value in the millions of dollars; we have no complaints.

--Dave

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Re: [Freeswitch-users] Outbound socket question

[David Knell]

Hi Nik,

Yes and no, respectively.

Cheers --

Dave

  - Original Message - 
  From: Nik Middleton 
  To: freeswitch-users@lists.freeswitch.org 
  Sent: Tuesday, June 02, 2009 6:04 PM
  Subject: [Freeswitch-users] Outbound socket question


  Hi Guys,

   

  I'm going some work with outbound socket, and have a few questions.

   

  When each call is answered, I get a connection to my server socket.

   

  Is it right to assume that this connection will remain for the duration of 
the call?

   

  If so, do I still need to pass the UUID when I call an application such as 
playfile?

   

  Regards

   



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Re: [Freeswitch-users] Make current fails (build 13537)

[David Knell]

At the risk of evisceration (but with the intention of helping avoid future 
brain dead build vs. idiot admin debates), I'd suggest that, when significant 
new bits are added to the switch core, they should default to being off and 
require a configuration option to turn them on.  Such config options can be 
added to the default config; that way new installs will have the new 
functionality enabled by default, but those upgrading from an older install 
will need to enable them manually, reducing the risk of stuff breaking.

--Dave
  - Original Message - 
  From: Brian West 
  To: freeswitch-users@lists.freeswitch.org 
  Sent: Monday, June 01, 2009 11:33 PM
  Subject: Re: [Freeswitch-users] Make current fails (build 13537)


  NO its not a bad one at all.  Its switch_nat_init(); in switch_core.c since 
your network must be eating the packets its sending out to detect if you're 
behind nat or not... and not getting an ICMP unreachable like it should be 
getting... the joys of admins that block all ICMP like idiots.  ICMP has many 
uses... and outright blocking it is stupid. (This is my assumption cuz its what 
makes sense in this case)


  So you're getting hit by the nice retry/timeout loop in the natpmp software 
we just added and possibly the upnp lib too.


  So for now edit switch_core.c and comment out switch_nat_init();


  I'm working my ass off to ensure that our users that do have to live in these 
insane nat scenarios can do so without much if any pain. Most of which uses 
SMB/Consumer grade routers which these two libs we added would allow us to poke 
holes and setup stuff and make it painless as possible. 


  Soon you'll have an option in switch.conf.xml to turn it off.


  Please next time don't be so demanding and calling builds brain dead .. when 
in fact its trying to become more aware of its network config without much user 
input.


  /b


  On Jun 1, 2009, at 5:24 PM, Nik Middleton wrote:


Well I can only assume build 13537 is brain dead.  Surely I shouldn’t have 
to edit a whole bunch of configs to get it working. FS now takes 3 minutes to 
start, with no indication as to what it’s looking for in the logs. That said, 
to date ‘make current’ has always worked well for me.  Guess I was bound to hit 
a bad one eventually. 

Still, it’s very frustrating.



  Brian West
  br...@freeswitch.org


  -- Meet us at ClueCon!  http://www.cluecon.com











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Re: [Freeswitch-users] g729 support

[David Knell]

> If you live in patent-free country, you can try this:
> http://github.com/Deepwalker/fs_itu_g729/tree/master

I've a friend who says he knows of someone who's tried it in non-
patent-free countries, and it works fine there too.

Alternatively, an Asterisk box makes a perfectly good G.729 to
G.711 transcoder.

--Dave

> Regards,
> Mindaugas Kezys
> http://www.kolmisoft.com
> VoIP Billing and Routing Solutions
>
>
> -Original Message-
> From: freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of 
> FERNANDO
> VILLARROEL
> Sent: 2009 m. gegužės 31 d. 06:24
> To: freeswitch-users@lists.freeswitch.org
> Subject: [Freeswitch-users] g729 support
>
>
> Dear All,
>
> In this moment i am using Asterisk for my services VoIP; i am testing FS 
> and
> i am very interest on change Asterisk for FS, my motivation is improve my
> qualify mainly ACD statics.
>
> But i have problem, G729 codec is only supported in passthrough mode.
> FreeSwitch does not do any transcoding with G729 yet, hence I can't fully
> move to FS without G729 support.
>
> I am receiving traffic in g729 in Asterisk(does very bad) i need a FS to
> receive this traffic and convert to ulaw and forward to other SIP 
> provider,
> how i can do or anyone help me?
>
> I hope your comments or idea how i can do,
>
> Fernando.
>
>
>
>
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Re: [Freeswitch-users] Double-dtmf detection in IVR when a call isrouted through FreeSWITCH

[David Knell]

 Hi Drew,

When you say that the problem goes away if you don't use start_dtmf, do you 
mean that you get one tone recognised per tone or none?  If the former, then 
you've got DTMF being signalled out of band as well; in that case, why do you 
need inband detection?

--Dave
  - Original Message - 
  From: Drew Ozier 
  To: freeswitch-users@lists.freeswitch.org 
  Sent: Tuesday, May 26, 2009 9:36 PM
  Subject: [Freeswitch-users] Double-dtmf detection in IVR when a call isrouted 
through FreeSWITCH


  I've got a configuration where I receive inbound calls and dial out to a 
pre-determined 800-number based on the DNIS of the call. I set  and have everything set up so that DTMF only comes 
to me via inband. When I'm providing DTMF data to the IVR, it will recognize a 
single keypress as a double-tap. My FreeSWITCH logs only contain one DTMF 
entry, but when I listen to the receiving end of the call, I can hear a hiccup 
in the DTMF tone that is getting played. When I do not use 'start_dtmf', this 
problem goes away. I need inband DTMF detection, but I can't have it messing up 
the audio stream. Any thoughts?

  -Drew Ozier



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Re: [Freeswitch-users] Unicast isn't working

[David Knell]

Hi Artem,

Please to see that some of the stuff I wrote is useful to someone..!

I've written an FS module which will send the audio over - it's more efficient 
than using unicast.  Let me know if you'd like a copy.

Cheers --

Dave
  - Original Message - 
  From: ?  
  To: freeswitch-users@lists.freeswitch.org 
  Sent: Tuesday, May 26, 2009 9:27 AM
  Subject: [Freeswitch-users] Unicast isn't working


  Hi,

  I am trying to setup ASR in FreeSwitch using Nuance ASR server and MRCP. Both 
FreeSwitch and Nuance installed on Windows Server 2003. FreeSwitch version is 
1.0.3 (12567M)

  I found an example in Perl at 
http://www.softivr.com/wiki/index.php/FreeSWITCH_MRCP_in_Perl and decided to do 
the same in C#.
  It establishes connection with Nuance and loads the grammar, everything works 
fine. The next step is to capture audio from FS and tramsmit it to ASR. This 
can be done with unicast. We must create an outbound socket and issue "unicast" 
command. Here it goes:

  private void SetupAudioTransmission()
  {
   EventSocket = new Socket(AddressFamily.InterNetwork, SocketType.Stream, 
ProtocolType.Tcp);
   EventSocket.SendTimeout = Config.Timeout;
   int ESPort = SearchESPort(EventSocket);

   Thread thrESListener = new Thread(new ThreadStart(ListenerThreadStart));
   thrESListener.Start();
   
   WriteLog(LogLevel.Info, "Creating outbound event socket");
   Session.Execute("socket", "127.0.0.1:" + ESPort); //main thread stops, 
listener thread listens for outbound socket connection.
   WriteLog(LogLevel.Info, "Outbound event socket disconnected");

   EventSocket.Close();
  }

  //here we accept outbound socket and transmit unicast command through it
  private void ListenerThreadStart()
  {
   Socket sockHandler = EventSocket.Accept();
   WriteLog(LogLevel.Info, "Incoming connection");
   sockHandler.Send(MessageEncoding.GetBytes("Connect\n\n"));
 
   WriteLog(LogLevel.Info, GetServerResponse(sockHandler));

   int rtpPort = (RTPSocket.RemoteEndPoint as IPEndPoint).Port;
   string command = string.Format("sendmsg\r\ncall-command: 
unicast\r\nlocal-ip: {0}\r\nlocal-port: {1}\r\nremote-ip: {2}\r\n" +
"remote-port: {3}\r\ntransport: udp\r\nflags: native\r\n\r\n", 
Config.LocalIP, rtpPort + 1, Config.LocalIP, rtpPort);

   WriteLog(LogLevel.Info, command);

   sockHandler.Send(MessageEncoding.GetBytes(command));
   WriteLog(LogLevel.Info, GetServerResponse(sockHandler));
   
   sockHandler.Disconnect(false);  
   sockHandler.Close();  
  }

  After this, FS writes that unicast has been created on corresponding IPs and 
ports. It really creates an UDP socket, but doesn't transmit any data through 
it. I tested it with Wireshark and from my application, nothing was detected.
  Also, if we specify "transport:tcp" in unicast command, it uses UDP anyway, 
that's strange.

  Here is how I listen UDP packets.
  private void DetectSpeech()
  {
   WriteLog(LogLevel.Info, "Reading audio");
   byte[] FSRecvBuf = new byte[2048];
   
   IPEndPoint epFS = new IPEndPoint(IPAddress.Loopback, 
(RTPSocket.RemoteEndPoint as IPEndPoint).Port);
   FSSocket = new Socket(AddressFamily.InterNetwork, SocketType.Dgram, 
ProtocolType.Udp);
   FSSocket.ReceiveTimeout = 10;
   FSSocket.SendTimeout = 10;
   WriteLog(LogLevel.Info, "Binding socket to " + epFS.Port);

   FSSocket.Bind(epFS);
   FSSocket.Connect(new IPEndPoint(IPAddress.Loopback, epFS.Port + 1));
 
   while (Session.Ready())
   {
int recvCount = FSSocket.Receive(FSRecvBuf);
WriteLog(LogLevel.Info, "Received bytes: " + recvCount);
   }
  }

  Can someone help me to solve this? Do I do something wrong or I forgot 
something or it doesn't work at all?

  Best regards.
  Artem




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Re: [Freeswitch-users] Testimonials

[David Knell]

Hi Maxim,

> David, thank you for information!
> 
> What is your experience of this solution - was it easy to support it and to
> develop new services?

There was a reasonably steep learning curve but, once that's been
topped, it's a walk in the park to develop new services.  Have a look at
some SoftIVR samples: http://www.softivr.com/wiki/index.php/Howtos
- by and large, each Javascript function there is implemented using a
few (or, at most, a few tens of) lines of Perl which talk to FS over the
event socket interface.

There's easier ways to get started - FS has a number of built-in
scripting languages.

> What was the highest load on your solution (in number of calls)?

Over 400 bridged calls per box with media (that's 800 call legs, or
about 80Mbits/sec of RTP in and out using - as we do - G.711) - other
figures that we've reached are 130K calls/day/box and $500k/month/box in
turnover :-)

Cheers --

Dave

> 
> 
> David Knell wrote:
> > 
> > Hi Maxim -
> > 
> > We've used FreeSWITCH for switching large volumes of wholesale traffic
> > and for a variety of IVR services; we no longer use anything else.  See
> > http://www.softivr.com for something which we've built on it.
> > 
> > Cheers --
> > 
> > Dave
> > 
> >> Hello,
> >> 
> >> Our company want to use Freeswitch and now we testing this solution in
> >> the
> >> lab.
> >> 
> >> If someone already using Freeswitch as office pbx, ivr or any other
> >> commercial purposes
> >> could you please share you experience (and if it is not a secret -
> >> company
> >> name)?
> >> 
> >> Regards,
> >> Maxim Tsvetov
> > -- 
> > David Knell, Director, 3C Limited
> > T: +44 20 3298 2000
> > E: d...@3c.co.uk
> > W: http://www.3c.co.uk
> > 
> > 
> > _______
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> > 
> > 
> 
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Re: [Freeswitch-users] Testimonials

[David Knell]

Hi Maxim -

We've used FreeSWITCH for switching large volumes of wholesale traffic
and for a variety of IVR services; we no longer use anything else.  See
http://www.softivr.com for something which we've built on it.

Cheers --

Dave

> Hello,
> 
> Our company want to use Freeswitch and now we testing this solution in the
> lab.
> 
> If someone already using Freeswitch as office pbx, ivr or any other
> commercial purposes
> could you please share you experience (and if it is not a secret - company
> name)?
> 
> Regards,
> Maxim Tsvetov
-- 
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T: +44 20 3298 2000
E: d...@3c.co.uk
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Re: [Freeswitch-users] text to speech IVRs and MOH

[David Knell]

Just to add one:

Cepstral - quite a lot cheaper than the non-free ones Pete mentioned;
voice quality quite adequate for what we want it for, which is short
prompts and development.  Not tried it for reading e-mail or the like.
FS' mod_cepstral isn't wholly compatible with their 5.x release (unless
someone's fixed it), but that's easily worked around.

--Dave

> I've spent the last 2-3 months on researching TTS and ASR for FS for a
> project.  Best TTS depends on what you consider important.  Also, how
> do you plan on using it.
> 
> 
> Here's some of the TTS engines I've run across with some pros/cons:
> 
> 
> Festivate Lite (flite)
> Pros:
> - Free (comes with FS)
> - simple to use
> - 16K voice sounds decent
> - Completely customizable
> Cons:
> - 8K voice sounds horrible over cell phone
> 
> 
> NeoSpeech (VoiceWare) (around $300/port for 1 voice + $75 each addl
> voice)
> Pros:
> - My selection for best soundig voices
> - Recently select by Stephen Hawkings for his voice (geek points!)
> - Lots of Languages supported
> - Free trial available
> Cons:
> - Custom C-Based API (FS interface coming soon)
> - Large file size (Engine + SDK + 1 Voice = 900MB)
> - Support is lacking (Company beed in Korean, time zone issues, etc)
> 
> 
> Nuance ($500/port for 1 voice)
> Pros:
> - Wide Variety of Products
> - Support MRCP
> - Supports ASR as well (add'l fees)
> - Excellent support
> - Free trial
> - Decent sounding voices at 8K and 16K
> - Wide range of tuning parameters
> Cons:
> - Pricey
> - Limited voice selection
> - Limited support for 64-bit linux
> 
> 
> AT&T (NaturalVoice) (no pricing info available)
> Pros:
> - Big company (solid in marketplace)
> - Good suppport (user and developer)
> - ASP model means no software to maintain
> Cons:
> - ASP model incurs delay
> - Voices sound too digitized
> - Limited support for 64-bit linux
> 
> 
> 
> Loquendo ($500/port for 1 voice + 15% addl voice)
> Pros:
> - Good sounding voices (almost as good as NeoSpeech)
> - Wide variety of languages
> - Excellent support
> - Has free 30 day trial
> - Supports MRCP
> - Support ASR and Voice Recognition as well. (add'l fees)
> - Small footprint (< 150MB)
> Cons:
> - Pricey
> - Complicated install process
> - Limited management/tuning capabilities
> 
> 
> In the end, it was down to NeoSpeech or Loquendo for our application.
>  We are currently running tests with NeoSpeech and assuming all goes
> well, we will select them.  Though don't let that color your opinion
> too much after several "focus groups" we discovered the most important
> element in the equation is does your customer/boss like the sound of
> the voices, and that is a completely subjective decision.
> 
> 
> 
> 
> -pete
> 
> 
>  Original Message 
> Subject: [Freeswitch-users] text to speech IVRs and MOH
> From: Saeed Ahmad 
> Date: Tue, May 19, 2009 12:40 am
> To: freeswitch-users@lists.freeswitch.org
> 
> Hi all,
>  
> Could you guys recommend me any online text to speech IVR
> software which works OK with FS. i am using AT&T site and for
> some IVRs i get sample rate errors. Also some resource to
> download more MOH wav files.
> 
> Many thanks
> 
> __
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Re: [Freeswitch-users] help with mod_conference stability

[David Knell]

On Thu, 2009-05-14 at 12:31 -0500, Brian West wrote:
> Its obvious if you look at the size of the JS VM vs the lua VM.. it
> would clearly scale better not needing megs and megs of ram per call
> vs about 160kb for lua. 

Isn't this a bit of a non sequitur, given that there ought to be just
one copy of the JS interpreter in RAM which is shared across all of the
calls/threads which need to use it?

--Dave


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Re: [Freeswitch-users] mod_event_socket 183 early media

[David Knell]

Gotcha - but in the case where the call hasn't yet got to a point where
there's a 183 been sent then I guess this wouldn't apply - there
shouldn't be any audio from the far end at this point, nor would the far
end be expecting any.

I'd suggest (and would volunteer to knock together a patch) adding a
'nopark' option to the socket command, which doesn't park the call - nor
would it change existing behaviour.  Obviously, in a situation like that
outlined by Ibrahim where the socket app handles all aspects of the
call, then it'll need to make sure that it signals ringing, answer or
whatever to make the call state flow work.

--Dave

> The park state is where the channel goes to be kept alive while you
> hold the socket.
> 
> The channel has to be in constant motion reading and writing audio or
> the audio would build up, akin to a gui app blocking and you can't
> press any of the buttons.
> 
> it may be possible for the park loop to tighten up when the call is
> not answered or in early media in advance
> but it would not be able to do anything besides a few things like get
> and set variables until it was meda-enabled
> 
> 
> On Tue, May 12, 2009 at 12:49 PM, David Knell  wrote:
> This is something I've been wondering about as well.  What's
> the reason
> for the channel being parked?
> 
> Cheers --
> 
> Dave
> 
> 
> > the socket app will always open early media, it's not
> currently
> > possible to park a channel that does not have media, you
> could try to
> > post a bounty for such a feature but it would have to be an
> elegant
> > solution
> >
> >
> > 2009/5/12 İbrahim TUNALI 
> > Hi,
> > I am trying to build a SBC but a kind of different
> described
> > on wiki.
> >
> > I immediately send incoming leg A to a socket app
> and connect
> > bridge with my socket python application. I will
> handle all
> > business logic on python side.
> >
> > My dialplan is so simple;
> >
> > 
> >  > expression="^1000$">
> >  data="127.0.0.1:1905
> > async full"/>
> > 
> > 
> >
> > Freeswitch send calls to parking and 183 early media
> through
> > mod_sofia. This is unacceptable on my SBC scenario.
> Because I
> > not receive 183 or 180 from leg B yet. I want to
> relay early
> > media or ringing when I received from B leg
> >
> > I realize this is not a bug. It is a feature but i
> need to
> > know is this possible.
> >
> > I would appreciate all suggestions and clues.
> >
> > Regards,
> > ibrahim
> > ___
> > 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/
> >
> > 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
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> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
>

Re: [Freeswitch-users] mod_event_socket 183 early media

[David Knell]

This is something I've been wondering about as well.  What's the reason
for the channel being parked?

Cheers --

Dave

> the socket app will always open early media, it's not currently
> possible to park a channel that does not have media, you could try to
> post a bounty for such a feature but it would have to be an elegant
> solution
> 
> 
> 2009/5/12 İbrahim TUNALI 
> Hi,
> I am trying to build a SBC but a kind of different described
> on wiki.
> 
> I immediately send incoming leg A to a socket app and connect
> bridge with my socket python application. I will handle all
> business logic on python side.
> 
> My dialplan is so simple;
> 
> 
>  expression="^1000$">
> 
> 
> 
> 
> Freeswitch send calls to parking and 183 early media through
> mod_sofia. This is unacceptable on my SBC scenario. Because I
> not receive 183 or 180 from leg B yet. I want to relay early
> media or ringing when I received from B leg
> 
> I realize this is not a bug. It is a feature but i need to
> know is this possible.
> 
> I would appreciate all suggestions and clues.
> 
> Regards,
> ibrahim
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> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 
> AIM: anthm
> MSN:anthony_miness...@hotmail.com
> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:8...@conference.freeswitch.org
> iax:gu...@conference.freeswitch.org/888
> googletalk:conf+...@conference.freeswitch.org
> pstn:213-799-1400
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Re: [Freeswitch-users] no RTP send during Voice Mail recording

[David Knell]

Add something like
memset(write_buf, 0, SWITCH_RECOMMENDED_BUFFER_SIZE);
after 
char write_buf[SWITCH_RECOMMENDED_BUFFER_SIZE];
in switch_ivr_play_say.c (line 395)

--Dave

> Thank you very much, dujinfang, for your help!
> 
> When I use
> 
> 
> the FreeSWITCH really sends back RTP stream during recording, but
> instead of (faked) silence it is full of completely regular load noise :-)
> I have tested it with different devices (Linskys, Snom, FritzBOX,
> Nokia...) with the same result (even pcap files looks similar).
> 
> Dialplan snipped looks like:
> 
> 
> 
> 
> 
> 
> Do you (or anybody else :-) know what I'm doing wrong?
> 
> Thanks once more, dujinfang, for your help!
> 
> Best regards,
> 
> kokoska.rokoska
> 
> 
> 
> dujinfang napsal(a):
> > I haven't tested but I guess it's just like other variables and I  
> > documented to here:
> > 
> > http://wiki.freeswitch.org/wiki/Channel_Variables#record_waste_resources
> > 
> > 
> > On Apr 25, 2009, at 11:56 PM, kokoska rokoska wrote:
> > 
> >> Thank you very much, Anthony, for such fast solution!
> >>
> >> May I ask you - How should I activate this feature?
> >> I have tried to "grep" through sources for new NDLB variable but I
> >> didn't find one...
> >>
> >> Best regards,
> >>
> >> kokoska.rokoska
> >>
> >> Anthony Minessale napsal(a):
> >>> sigh,
> >>>
> >>> see r13144
> >>>
> >>>
> >>> On Fri, Apr 24, 2009 at 3:43 AM, kokoska rokoska
> >>> mailto:kokoska.roko...@post.cz>> wrote:
> >>>
> >>>
> >>>
> >>>
> >>>seven napsal(a):
>  You are not alone, I vote 1.
> 
>  And there's a similer variable in conference:
> 
>    
>    
> 
> >>>Thank you very much, seven, for your support :-)
> >>>
> >>>Best regards,
> >>>
> >>>kokoska.rokoska
> >>>
> >>>
> >>>___
> >>>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/
> >>>
> >>> 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
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> >>
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> > 
> > 
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> > 
> 
> 
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Re: [Freeswitch-users] VoiceXML

[David Knell]

Just sticking 'voice' in front of something doesn't automatically make
it a good tool for developing voice applications - there's more
marketing here than anything else.  And it's not like adding extensions
to an existing language to provide IVR control is anything new: it's
exactly what you get if you develop for FS in Javascript, Lua or any of
its other supported languages.

>From my point of view, as a programmer, VoiceXML is the wrong idiom for
development of IVR/telephony services; a procedural language works just
fine.  I suspect that I'm not alone, and I further suspect that that's
why there's no real push to get VoiceXML supported.

--Dave

> If you don't like vxml then here is a post on voicePHP
> 
> http://www.speechtechblog.com/2009/04/22/voicexml-to-go-down-in-the-third-says-voicephp
> 
> It's from a vendor but there might be some good ideas to get from what they
> are doing.
> 
> FreeSWITCH needs demand to get vxml and it's not there yet. For now, it
> looks like the FS community is waiting for demand instead of trying to
> create it. 
> 
> 
> 
> 
> David Knell wrote:
> > 
> > On Tue, 2009-04-21 at 14:35 -0400, mszla...@aol.com wrote:
> >> Great Idea. 
> >> Try setting up the exact same dialogue with say Voxeo's VoiceXML
> >> system and then with Javascript/Lua and pocketsphinx. It's an order of
> >> magnitude faster with VoiceXML.
> > 
> > Out of interest, is that using some RAD tool or coding directly in 
> > VoiceXML?  I ask because VoiceXML strikes me as being a bastard
> > abomination of the highest order, whose sole saving grace is that
> > it's a standardised bastard abomination.
> > 
> > Or is Pocketsphinx the problem?
> > 
> > Cheers --
> > 
> > Dave
> > 
> > 
> > 
> > 
> > ___
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> > http://www.freeswitch.org
> > 
> > 
> 


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Re: [Freeswitch-users] Compact, fanless appliance?

[David Knell]

You might want to take a look at this:
http://www.amazon.com/IEEE802-11N-Wireless-Broadband-MZK-W04NU-Designed/dp/B000YDS0YG

- twice as much everything as the NSLU2, and is supposed to run OpenWRT
just fine.  I've one sat in front of me right now, although I've not yet
plugged it in - have to work out how to take it apart first ;-)

--Dave

> BTW, at 85€, the Linksys NSLU2 looks like a bargain:
> 
> http://en.wikipedia.org/wiki/NSLU2
> 
> Has someone successfully ran Freeswitch on this to handle a couple of
> simultaneous SIP conversations?
> 
> What about the more expensive but very tiny Gumstix?
> 
> http://en.wikipedia.org/wiki/Gumstix


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Re: [Freeswitch-users] Help with mod_managed under Windows

[David Knell]

Google can:
http://tinyurl.com/dyvknb

- the bit that's being complained about is
"FreeSWITCH.Native.freeswitch" has caused an exception. -- 
-> System.EntryPointNotFoundException: The entry point
"CSharp_SWITCH_READ_TERMINATOR_USED_VARIABLE_get" was not found in the
DLL "mod_managed".

--Dave

> Could you translate these into english?
> 
> On Apr 21, 2009, at 7:08 AM, Guido Kuth wrote:
> 
> > I am playing around with FS (Windows) for one month now. First I
> > tried using FreeSwitch.NET which is a good class library for inbound
> > event socket. Unfortunatley it can't be used for outbound event
> > socket. So I read the wiki back ond forth and also searched the net
> > and found that I should use mod_managed.
> >  
> > So I downloaded mod_managed in source from svn and compiled it with
> > C# 2008 Express Edition. After that I got a dll.
> > FreeSwitch.Managed.dll and copied it to the mod dir of FS. The
> > Problem is that I get and error when FS loads mod_managed and I
> > don't know what I should do with that.
> >  
> > 2009-04-21 11:26:44 [INFO] mod_managed.cpp:314 mod_managed_load()
> > Loading mod_ma
> > naged (Common Language Infrastructure), Microsoft CLR Version
> > 2009-04-21 11:26:44 [ERR] mod_managed.cpp:333 mod_managed_load()
> > Load did not re
> > turn true. System.Reflection.TargetInvocationException: Ein
> > Aufrufziel hat einen
> >  Ausnahmefehler verursacht. ---> System.TypeInitializationException:
> > Der Typenin
> > itialisierer f³r "FreeSWITCH.Native.freeswitch" hat eine Ausnahme
> > verursacht. --
> > -> System.EntryPointNotFoundException: Der Einstiegspunkt
> > "CSharp_SWITCH_READ_TE
> > RMINATOR_USED_VARIABLE_get" wurde nicht in der DLL "mod_managed"
> > gefunden.
> >bei
> > FreeSWITCH.Native.freeswitchPINVOKE.SWITCH_READ_TERMINATOR_USED_VARIABLE_
> > get()
> >bei FreeSWITCH.Native.freeswitch..cctor()
> >--- Ende der internen Ausnahmestapel³berwachung ---
> >bei FreeSWITCH.Native.freeswitch.get_SWITCH_GLOBAL_dirs()
> >bei FreeSWITCH.Loader.Load()
> >--- Ende der internen Ausnahmestapel³berwachung ---
> >bei System.RuntimeMethodHandle._InvokeMethodFast(Object target,
> > Object[] argu
> > ments, SignatureStruct& sig, MethodAttributes methodAttributes,
> > RuntimeTypeHandl
> > e typeOwner)
> >bei System.RuntimeMethodHandle.InvokeMethodFast(Object target,
> > Object[] argum
> > ents, Signature sig, MethodAttributes methodAttributes,
> > RuntimeTypeHandle typeOw
> > ner)
> >bei System.Reflection.RuntimeMethodInfo.Invoke(Object obj,
> > BindingFlags invok
> > eAttr, Binder binder, Object[] parameters, CultureInfo culture,
> > Boolean skipVisi
> > bilityChecks)
> >bei System.Reflection.RuntimeMethodInfo.Invoke(Object obj,
> > BindingFlags invok
> > eAttr, Binder binder, Object[] parameters, CultureInfo culture)
> >bei mod_managed_load(switch_loadable_module_interface**
> > module_interface, apr
> > _pool_t* pool)
> > 2009-04-21 11:26:44 [CRIT] switch_loadable_module.c:845
> > switch_loadable_module_l
> > oad_file() Error Loading module C:\Programme\FreeSWITCH\mod
> > \mod_managed.dll
> > **Module load routine returned an error**
> > 
> > Please help me with that.
> >  
> > Thanks...Guido
> 
> 
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Re: [Freeswitch-users] VoiceXML

[David Knell]

On Tue, 2009-04-21 at 14:35 -0400, mszla...@aol.com wrote:
> Great Idea. 
> Try setting up the exact same dialogue with say Voxeo's VoiceXML
> system and then with Javascript/Lua and pocketsphinx. It's an order of
> magnitude faster with VoiceXML.

Out of interest, is that using some RAD tool or coding directly in 
VoiceXML?  I ask because VoiceXML strikes me as being a bastard
abomination of the highest order, whose sole saving grace is that
it's a standardised bastard abomination.

Or is Pocketsphinx the problem?

Cheers --

Dave




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[Freeswitch-users] Event sockets and IPC

[David Knell]

Hi -

Consider a situation where I've two call legs up, each talking to a
controlling process using an outbound event socket connection from FS.
They each know the other's UUID.

I'd like to send an arbitrary message from one to the other.  I was
going to build some sort of IPC thing, but it struck me that the elegant
way to do it would be via FS' event mechanism - that way the messages
come in to the processes in exactly the same way as, for example, DTMF
indications.

Problem I'm having is working out how to (and, indeed, whether one can)
use sendevent to send a CUSTOM event to a specific UUID, and I was
wondering if anyone would be kind enough to put me out of my misery one
way or the other - or have I missed another mechanism for doing this?

Cheers --

Dave



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Re: [Freeswitch-users] Entire Wiki.FreeSwitch.org on Single PDF ?

[David Knell]

> If someone has a way to make true mirrors that support read/write
> this 
> would be interesting.

Do it robustly, transparently and in real time and that's the problem of
distributed source code revision control mostly sorted as well.
Although I'm not sure I'd really want to use "the kernel anyone can
edit"..

http://meta.wikimedia.org/wiki/Help:Export details a process for
exporting one or more pages, which'd be pretty trivially implementable
using WWW::Mechanize, as would the import.  Obviously this is just a
one-way solution.

Alternatively, wget --mirror http://wiki.freeswitch.org would be likely
to provide a straightforward starting point for anyone wanting to mirror
the thing..?

--Dave



> Mike
> 
> On Apr 17, 2009, at 4:06 AM, Will Boyce wrote:
> 
> > Special:Export will export a page to an XML format that can, in  
> > turn, be imported.
> >
> > There must be a way to automate that process (export extire wiki to  
> > XML, rsync and import and mirrors).
> > -- 
> > Regards,
> >
> > Will Boyce 
> > tel: 07933 515 987
> > url: http://willboyce.com
> >
> > - "Jason White"  wrote:
> >
> > | From: "Jason White" 
> > | To: freeswitch-users@lists.freeswitch.org
> > | Sent: Friday, 17 April, 2009 07:54:13 GMT +00:00 GMT Britain,  
> > Ireland, Portugal
> > | Subject: Re: [Freeswitch-users] Entire Wiki.FreeSwitch.org on  
> > Single PDF ?
> > |
> > | Mitul Limbani  wrote:
> > | >  Another idea would be to write simple rsync method, and post a
> > | page
> > | > on the same on the Wiki so all those people who have their own
> > | server
> > | > and willing to spare some bandwidth can mirror the entire Wiki
> > | > locally.
> > |
> > | MediaWiki uses a database, as I understand it. However, there  
> > might be
> > | a way
> > | to have it write out all of the content to a file of some sort.
> > |
> 
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Re: [Freeswitch-users] Call bridge in free switch

[David Knell]

Take out the brackets -
originate sofia/profile/1001...
(and you might want to replace profile with the name of the profile to
use)

There's documentation here which might help:
http://wiki.freeswitch.org/wiki/Mod_commands#originate

--Dave

> Hi Mike,
> 
> I tried the following command per ur advice.. but getting the error
> CHAN_NOT_IMPLEMENTED
> 
> originate (sofia/profile/1...@192.168.1.108) &
> conference(3085-192.168.1.102);
> 
> 
> freeswi...@internal> originate (sofia/profile/1...@192.168.1.102) &
> conference(3085-192.168.1.102);
> -ERR CHAN_NOT_IMPLEMENTED
> 
> freeswi...@internal> 2009-04-16 20:28:30 [ERR]
> switch_core_session.c:303 switch_core_session_outgoing_channel() Could
> not locate channel type (sofia
> 2009-04-16 20:28:30 [ERR] switch_ivr_originate.c:1486
> switch_ivr_originate() Cannot create outgoing channel of type [(sofia]
> cause: [CHAN_NOT_IMPLEMENTED]
> 2009-04-16 20:28:30 [DEBUG] switch_ivr_originate.c:2084
> switch_ivr_originate() Originate Resulted in Error Cause: 66
> [CHAN_NOT_IMPLEMENTED]
> 
> Thanks
> prabhu
> 
> On Thu, Apr 16, 2009 at 4:29 PM, Michael Collins 
> wrote:
> Do you need to monitor the possible failure of one of these
> calls? Just curious. You can call them individually and drop
> them into a conference right at the FS cmd line:
> 
> originate (sofia/profile/1...@192.168.1.108) &
> conference(myconfname);
> originate (sofia/profile/1...@192.168.1.108) &
> conference(myconfname);
> originate (sofia/profile/1...@192.168.1.108) &
> conference(myconfname);
> 
> You can control the conference behavior with numerous options.
> See http://wiki.freeswitch.org/wiki/Mod_conference for lots of
> great information.
> 
> -MC
> 
> 
> On Thu, Apr 16, 2009 at 1:09 AM, Prabhuram Mohan
>  wrote:
> 
> 
> Hello,
> 
> I am trying to find a way to this through
> fs_cli
> 1) call out to ClientA (1...@192.168.1.108),
> ClientB (1...@192.168.1.108) & ClientC
> (1...@192.168.1.108)
> 2) Bridge all the 3 legs together into one
> call
> 
> Thanks
> Prabhu
> 
> 
> 
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Re: [Freeswitch-users] Re-2: FreeSwitch Complex IVR System

[David Knell]

Hi Guido,

The event socket interface will give you DTMF events for bridged calls -
just tried it and it works fine.  There's one mild snag, which is that
outbound sockets (which are easier for inbound call handling) will only
give you events relating to the specific call leg that's attached to
that socket - i.e. you can use an outbound socket app to bridge that leg
to an outbound call leg just fine, but you won't get events related to
that outbound call.

So what we do is use an outbound socket app for call control and
scripting, and have a separate inbound socket app which listens for call
state changes and DTMF on all call legs, and a database table which
glues the two together. 

Cheers --

Dave


> Hi Dave,
> 
> thanks for the answer. I am playing around with FS and Event Socket Library 
> for .NET. I get pretty much to run with this, but the reason why I came from 
> Asterisk to FS is that I cannot get DTMF in a bridged call. I thought that I 
> get an Event as soon one dtmf digit is recognized. Unfortunately this isn't 
> the case.
> 
> If I use the default config files and map the keys with bind_meta_app the 
> dtmf tones are recognized and the function behind the bound app is executed. 
> Is this maybe a bug.
> 
> I have read about mod_managed and that I should use it, but I haven't found 
> anything about the usage of it.
> 
> Any suggestions would help
> 
> thanks...Guido
> 
>  Original Message 
> Subject: Re: [Freeswitch-users] FreeSwitch Complex IVR System (16-Apr-2009 
> 17:35)
> From:David Knell 
> To:  g...@exram.de
> 
> > Hi Guido,
> > 
> > My preferred way is to talk to FS through its event socket
> > interface.  This allows you fully to control FS, whilst giving
> > you the power to write the code in whatever language and on
> > whatever platform you choose.
> > 
> > The documentation starts here:
> > http://wiki.freeswitch.org/wiki/Mod_event_socket
> > 
> > Cheers --
> > 
> > Dave
> > 
> > > Hi @all
> > >  
> > > I have a question about a project I want to realize with FreeSwitch. I
> > > want to do a complex IVR System which takes a call, do many things in
> > > a MSSQL DB, send some Informations to one or many Middleware Servers
> > > via TCP/IP, call one or more mobile phones, the first is able to take
> > > the call, it can be that he must be able to hear a prompt before he is
> > > actually connected to the first caller, then the conversation must be
> > > recorded automatically and during the conversation it must be possible
> > > for the called party to redirect the call by dtmf. I know that this is
> > > all possible, but I want to know which way is the best to do all this?
> > > 
> > > 
> > > ___
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> > 
> > 
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> 
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Re: [Freeswitch-users] FreeSwitch Complex IVR System

[David Knell]

Hi Guido,

My preferred way is to talk to FS through its event socket
interface.  This allows you fully to control FS, whilst giving
you the power to write the code in whatever language and on
whatever platform you choose.

The documentation starts here:
http://wiki.freeswitch.org/wiki/Mod_event_socket

Cheers --

Dave

> Hi @all
>  
> I have a question about a project I want to realize with FreeSwitch. I
> want to do a complex IVR System which takes a call, do many things in
> a MSSQL DB, send some Informations to one or many Middleware Servers
> via TCP/IP, call one or more mobile phones, the first is able to take
> the call, it can be that he must be able to hear a prompt before he is
> actually connected to the first caller, then the conversation must be
> recorded automatically and during the conversation it must be possible
> for the called party to redirect the call by dtmf. I know that this is
> all possible, but I want to know which way is the best to do all this?
> 
> 
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Re: [Freeswitch-users] mod_vmd--can it do more than simply find VoiceMail Beeping?

[David Knell]

On Tue, 2009-04-07 at 17:17 +1000, Jason White wrote:
> Matthew Fong  wrote:
> > My question is, is there a way to use mod_vmd to detect if an answering
> > machine or human has picked up within the first 1-2 seconds after being
> > answered? 
> 
> Probably not. If you have an algorithm in mind that would achieve this with a
> high degree of reliability, I'm sure the FreeSWITCH developers would be
> interested in it. However, as far as I know, there is no reliable way to
> distinguish, for example, my voice as recorded in a voicemail greeting from my
> voice giving a live greeting after answering a phone call. Think about it.

The usual way is to measure how long the person who answers the phone
speaks for.  A person might say "Hello?", "Hello, this is Alice",
"Thank you for calling XYZ.  How may I direct your call?"  Voicemail 
will usually be longer - "Hi, this is Bob.  I'm sorry I can't take
your call right now, so please leave me a message after the tone and
I'll get back to you as soon as I can."

In the first couple of cases above, this would give you an answer - 
"human" - within the first few seconds of the call.

FreeSWITCH will give you TALK (start of speech (or noise)) and NOTALK
(end of) events if you enable VAD.

--Dave


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Re: [Freeswitch-users] How to call multi gateways for failover with early media?

[David Knell]

On Mon, 2009-04-06 at 00:08 -0400, Kristian Kielhofner wrote:

> Actually using 180 w/o SDP provides for enhanced call handing
> functionality while only requiring (in many cases) one additional test
> scenario.  Consider the current example (all 180s are actually 180s
> w/o SDP and 183 is 183 w/ SDP):
> 
> Bridging a call to multiple destinations (A, B, and C).
> 
> A: 100,180
> B: 100,180,200
> C: 100,183
> 
>   We could have implemented proper forking if it weren't for C who
> insisted on sending media early (for whatever reason).  While I could
> see many scenarios where this might happen even with the configuration
> I suggest, consider what would happen in the ideal scenario:
> 
> A: 100,180
> B: 100,180,200
> C: 100,180

> Ah, B won because it was the first endpoint to actually /answer/ the
> call and begin playing media.  Nice and clean.

Hang on - if you want to bridge the call on *answer*, then bridge it on
answer, not when one leg starts sending you early media.  I've no idea
if FS supports this behaviour for its forked dialling, but it's easy
to do with a bunch of originates, and uuid_bridge the inbound leg to the
first one which answers.

> People poke at SIP all the time for this one but this is where the
> PSTN even seems a bit ambiguous.  We have ISDN cause codes AND inband
> audio messages?

Yes.  A clearing code is used when the call's cleared; inband audio
can be used to give the caller more information than a simple clearing
code might allow - for example, "The number you are calling has been
changed.  Please redial on whatever the new number might be."  It 
makes eminent sense - simple, common causes (e.g. user busy) get dealt
with as part of the call clearing and it's the responsibility of the
originating switch to tell the user; more (indeed arbitrarily) complex
ones are dealt with by the far end.

--Dave


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Re: [Freeswitch-users] How to call multi gateways for failover with early media?

[David Knell]

On Sat, 2009-04-04 at 03:32 -0400, Kristian Kielhofner wrote:

> 
>   180/183 with SDP is a bit ambiguous.  I always get frustrated when
> various people /insist/ on using 183 w/ SDP just for ringback.  Have
> they never heard of 180 w/o SDP?  Let me generate my own local
> ringback and/or handle the situation accordingly!

Ah, well, that's where you're trying to change the way that things
have been done for some decades.  Ringback has historically been
generated close to the called party, which is why you hear different
ringback if you call people in different countries.

Furthermore, that audio path is used to convey all sorts of messages:
"the number you have called has been changed", "the cellphone you have
called has not responded", "calls to 1-800 numbers are not free if 
made from overseas.."  Lastly, there's no guarantee that it'll be
possible to differentiate between one of these and ringback from the
signalling alone and, in many cases, there is simply no mechanism 
available to provide such differentiation.

You're probably best advised to swim with the tide on this one..!

Cheers --

Dave  


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Re: [Freeswitch-users] Another FreeSWITCH First!

[David Knell]

Here's a sample SIP/SMTP INVITE (responses omitted for clarity)
MAIL FROM: 
RCPT TO: 
DATA
Call me
.

--Dave

Sent from my iPhone

On 1 Apr 2009, at 09:15, Brian West  wrote:

You know you could write a transport plugin for Sofia that would do  
SIP over SMTP  :P


/b

On Apr 1, 2009, at 11:07 AM, Nik Middleton wrote:


Well you almost had me there, but SIP over SMTP?  That was too much.

Regards,



Brian West
br...@freeswitch.org

-- Meet us a ClueCon!  http://www.cluecon.com



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Re: [Freeswitch-users] freeswitch as a session border controller

[David Knell]

Steve Underwood wrote:

Anthony Minessale wrote:
  

I'm really starting to feel like we're playing musical threads here.


Just avoid playing them through low bit rate codecs. :-)
  

I think we need an echo canceller ;-)

--Dave
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Re: [Freeswitch-users] echo cancellation on PRI cards

[David Knell]

Raymond Chandler wrote:
> What's interesting to me is everyone on this thread except you has 
> said that in real-world scenarios, they need the EC for reliability.  
> One of which, does signal processing programming professionally. It 
> seems to me that if you "build a better mouse trap" you must know what's 
> involved in making it work properly.  I'm not sure what your background 
> really is, but you'd be hard pressed to match up to Steve's reputation 
> and/or experience.
>   
Public willy-waving is undignified but, in brief, I've built and sold 
IVRs since
1997, wrote a CAPI-based soft IVR in 1999 (which required software for, 
inter
alia, DTMF detection), developed a software fax modem (V.29, V.27ter, T.30,
etc.) which I sold to a CTI card vendor and so on.

I've collected some data, of which it is commonly said that the plural 
of anecdote -
which is what we've had so far - is not.  The IVR collects a 16 digit 
DTMF string,
terminated by #.  TDM->IP conversion was performed by an Asterisk box with
an el-cheapo quad E1 card (no EC) for half the calls, and an AS5400 
(with EC)
for the other half.

The proportion of entries missing one or more digit was 3.1% (Asterisk) 
and 3.3%
(AS5400); this is not a statistically significant difference given the 
sample size.
The reason for looking at this criterion is (a) that it's easy to 
measure, and (b) the
most likely way that a DTMF detector will fail in the presence of excess 
noise,
which includes echo, would be to miss a digit.  This error rate is the 
sum of
human error + detector error, and I've no measurements to show how this 
might
be split; I would expect it's almost all human. Note that this is a 
digit error rate of
about 1 in 500.

This is, of course, only data from one site, but it's a start; it's only 
by collecting
data such as this that one can understand how well one's mouse trap works
and whether it needs improvement or not.
> That said, it might be a good idea to just agree to disagree as this is 
> starting to sound like the faxing over IP talks I hear a lot. (i.e. 
> "faxing over g.711u with no t.38 works fine for me") Where it might work 
> for some people by some mysterious phenomena, it won't work for the 
> general public. And telling people that they don't need EC, where so 
> many have already said that they obviously do, is just as irresponsible, 
> IMHO, as you claiming Steve was for telling them that they don't need it.
>   
That's a simplification.  Simple IVR (record, replay, collect DTMF) probably
doesn't need EC; if you're trying to do ASR with barge-in, bridge callers to
other callers or operators, etc., then you probably do.

I am interested that the recommended solution is 'buy Sangoma' - expensive
and proprietary - when Oslec, a FOSS echo cancellers which, by all accounts,
works extremely well, is out there and has been for some time.

--Dave



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Re: [Freeswitch-users] Freeswitch-users Digest, Vol 33, Issue 102

[David Knell]

Hi Brian,
> It's more sane to have the phone to NOT send them both in the first  
> place because it is WRONG to send both info and 2833 and NOT totally  
> expect the far end to make heads or tails of it.
>
> How about actually have the phone manufacture fix their broken phone?
>   
In an ideal world, of course; however:-
(a) the quick hack is probably a path of lesser resistance to getting 
Zhao up
and running with FS;
(b) he said it was an inbound SIP provider, rather than a phone, that he 
was
using, so he'd need to get them to fix their end: might be trivial, 
might not.

--Dave

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Re: [Freeswitch-users] Freeswitch-users Digest, Vol 33, Issue 102

[David Knell]

Sorry - my bad - dtmf-type looks like it just controls what's sent,  
not what's received.
Brian's advice is sound, or you can probably work around things right  
now by editing

src/mod/endpoints/mod_sofia/sofia.c - at around line 3838 you'll find:
if (dtmf.digit) {
  /* queue it up */
  switch_channel_queue_dtmf(channel, &dtmf);
..more code..
  /* Send 200 OK response */
  nua_respond(nh, SIP_200_OK, NUTAG_WITH_THIS(nua), TAG_END());

- lose the bit which handles the SIP INFO DTMF by adding a couple of  
lines thusly:

if (dtmf.digit) {
#if 0
  /* queue it up */
  switch_channel_queue_dtmf(channel, &dtmf);
..more code..
#endif
  /* Send 200 OK response */
  nua_respond(nh, SIP_200_OK, NUTAG_WITH_THIS(nua), TAG_END());

It's a nasty hack, but it just might work.

--Dave


Tell your phone to stop sending INFO and 2833 at the same time and  
the problem will stop.


/b

On Mar 23, 2009, at 4:17 AM, zhaoxxqq wrote:


HI, friend,
I added  to my sip  
profile in external , like below.

 
   -->
  
  
  
  
  
  

but. the problem is still exist. Can you help me.

Zhao Xiaoqiang


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Re: [Freeswitch-users] Problem about always Double accept DTMFs

[David Knell]

Hi -

It looks like you're getting digits both in the RTP stream and as SIP INFO.

Try adding  to the SIP profile 
you're using for inbound calls.


--Dave
I'm newbie of FS. I have setup the FS. Most is perfect. I connect FS 
to PSTN with DID numbers. For inband I connect it to FS's demo_ivr. 
When I press the key, the FS accept always DOUBLE of key. The debug 
information like below.
 
2009-03-22 17:50:26 [DEBUG] switch_rtp.c:1767 switch_rtp_dequeue_dtmf() RTP RECV DTMF 1:2000

2009-03-22 17:50:26 [DEBUG] switch_ivr_play_say.c:1279 switch_ivr_play_file() 
done playing file
2009-03-22 17:50:26 [DEBUG] switch_ivr_menu.c:308 play_and_collect() waiting 
for 3/4 digits t/o 2000
2009-03-22 17:50:26 [DEBUG] sofia.c:3753 sofia_handle_sip_i_info() INFO DTMF(1)
2009-03-22 17:50:28 [DEBUG] switch_ivr_menu.c:353 play_and_collect() digits '11'
2009-03-22 17:50:28 [DEBUG] switch_ivr_menu.c:523 switch_ivr_menu_execute() IVR 
menu 'jtq_greating' caught invalid input '11'
2009-03-22 17:50:28 [DEBUG] switch_ivr_play_say.c:989 switch_ivr_play_file() 
Codec Activated l...@8000hz 1 channels 20ms
2009-03-22 17:50:28 [DEBUG] switch_core_io.c:652 
switch_core_session_write_frame() sofia/external/13323015
 
 
Can any friend can help me?
 
Zhao Xiaoqiang
 



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Re: [Freeswitch-users] Cepstral and RSS feeds

[David Knell]

It's a book, Brian, and a long one at that, often printed in black and 
white - oops,

there I go again ;-)

I once referred to "oxymoronic hip-hop culture", only to be firmly told 
by a listener

that there was nothing moronic about hip-hop..

Cheers --

Dave

Oxymoron!?!?!  :)

/b

On Mar 20, 2009, at 8:53 PM, David Knell wrote:

  

War and Peace.




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Re: [Freeswitch-users] Cepstral and RSS feeds

[David Knell]

In the meantime, you can work around this by using the swift executable
to turn text in to WAV files, and then just play them back.  Works fine for
short(ish) texts - there might be a bit of a delay if you wanted the 
thing to

read back War and Peace.

--Dave

http://jira.freeswitch.org/browse/MODASRTTS-11

Might wanna know about that issue also :)

/b

On Mar 20, 2009, at 12:02 AM, Carlos Talbot wrote:


I wrote this wiki page a while back. Did it help?

http://wiki.freeswitch.org/wiki/Mod_rss




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Re: [Freeswitch-users] echo cancellation on PRI cards

[David Knell]

Hi Arnaldo,

That's interesting - Brasil was my first proper IVR installation: one 
with Embratel in Sao Paulo, and then a couple with TeleRJ.  I remember 
landing at Sao Paulo airport for the first time at 7 a.m. with 
instructions to "meet a fat man called Ferrari" unsure as to whether I 
was in some sort of elaborate hoax (I wasn't, and he was), and learning 
my first three words of Portuguese as we left the car park: filho da 
puta, of course.

Those had no EC.  DTMF detection worked fine, and the audio quality of 
the IVR recordings was perfect, which is what you'd expect: EC doesn't 
alter the IVR->caller audio at all.  A TDM->SIP->TDM type application is 
a different animal: you've got the added latency of packetisation/jitter 
buffering/etc. which pretty much makes echo cancellation a must.

--Dave
> Sharing my humble experience: in Brazil we usually need echo 
> cancellation to have reliable DTMF detection _and_ voice quality over 
> E1 lines (be it on MFC/R2 - r2d - or ISDN PRI lines), either for 
> sip/tdm gateway devices or IVR applications.
>
> Usually there's no need for echo cancellation on links from some 
> Telcos, in some specific places. But we need it in the majority of 
> cases, even when my box is just a gateway between legacy pbxes.
>
> This represents just a subset of the available E1s in the world and 
> it's just a practical experience, but it's a fact for me. If I don't 
> have a card with echo cancellation, I don't offer reliability to my 
> customer; I've done that in the past and didn't work out.
>
> I'm not theoretically discussing anything, just sharing what I've been 
> through in the last 4 or 5 years.


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Re: [Freeswitch-users] echo cancellation on PRI cards

[David Knell]

Steve Underwood wrote:

David Knell wrote:
  

Steve Underwood wrote:


[whopping big snip]
  
  

The first bit of that's a tad patronising, isn't it,



You are the one who started out being offensive.
  
  
I'm sorry if you find disagreement offensive; you might not wish to 
read beyond this

point if so.


and, in the case of the decade-old Aculab
cards which which I'm most familiar, is also untrue.


I can't find too much about the old cards on the web now, but I found 
http://www.amdevcomm.com/voice-mail-products/voice-mail-components/dialogic/dti_sc.html 
which is pretty much a copy and paste from the old Dialogic web pages, 
and you'll see it says "Cut through : Local echo cancellation permits 
100% detection with a >4.5 dB return loss line". The Aculabs did the 
same thing for sure. They just couldn't work without cancellation. There 
were some very early Dialogic cards, using DTMF receiver chips and OKI 
ADPCM chips, and had no general purpose DSPs. They performed really 
badly because of the lack of cancellation, and were quickly replaced 
with cards that put the OKI ADPCM, DTMF anf echo cancellation algorithms 
into a Motorola 56k DSP chips.
  
  

The same document, under the bit which you've quoted, says:
"(E-1) Digital trunks use separate transmit and receive paths to network.
Performance dependent on far end handset's match to local analog loop."
- i.e. the card does no echo cancellation. 

Your messages are starting to looked deranged. Why would they only apply 
echo cancellation to T1s? Its a bizarre idea, and you must realise its 
wrong. Are you so desperate to support a wrong answer you'll clutch at 
straws? :-\
  
More insults.  Answer me this: if there were echo cancellation in use, 
why would
DTMF detection performance depend on the far-end handset's match to the 
loop?


And the follow-up question (which you've already pretty much asked) - if the
card doesn't echo cancel for E1s, why would it for T1s?

As an aside, I'm not convinced that the document's not talking about 
return loss
on the T1 line itself, the implication being that the T1 is being 
carried on a single
pair, which makes the first sentence about E1s make a bit more sense.  
But that's

just a guess.
Aculab didn't even offer echo cancellation on Prosody for years and, 
when they did, it
consumed prodigious amounts of DSP.  Nonetheless, the DTMF detection 
worked
perfectly well, even across 120 channels per 40MHz SHARC - there's 
just no way
that those DSPs had enough horsepower to do echo cancellation across 
that manychannels.

This page 
http://www.aculab.com/support/pdf_documents/v6_solaris/ting/pubdoc/an-dtmf-det-issues.html 
seems to support what you say. It also implies DTMF detection sucks 
unless you echo cancel. The statement "If the outgoing signal is a tone 
of some sort (e.g. a 'beep'), ensure that its frequency is below 600Hz" 
is telling you to keep your outgoing signal in the same frequency range 
as dial-tone where the dial-tone filter on the DTMF receiver will 
obviate the need for an echo canceller. They are freely admitting 
exactly what I have said. If you want a normal IVR with cut-through to 
work you better turn that echo canceller on.


My only experience with Aculab was fitting a box designed by other 
people into a system. That one definitely echo cancelled, as it worked 
as well as the Dialogic based boxes we developed ourselves.
  
That only holds true if your premise - that you need echo cancellation 
for good

DTMF detection - is correct, which I don't believe it is.
An Asterisk box with an el-cheapo quad E1 card in that I use for 
TDM-SIP gatewaying

detects DTMF perfectly well with no echo cancellation.


You must have very low standards for "works well".
  

Nothing like a good old ad hominem attack.  Beats reasoned argument any day.
You just don't need echo cancellation to achieve perfectly acceptable 
DTMF detection.

Well, not if you expect people to wait for silence before entering DTMF, 
but who would tolerate that these days? Cut through has been de rigeur 
since the late 80s.
  
Oh, for pity's sake, you get perfectly good cut through without echo 
cancellation.

Humour me and draw a quick mental picture of the spectrum of a random bit of
speech at -20dBm; now add tones at -10dBm and -7dBm.  They stick out like
a pair of sore thumbs.

I'm sure it's quite possible to come up with a pathological case - e.g. 
cut-through
against a 1kHz milliwatt tone, but that sort of thing just doesn't 
happen in real-

life IVR applications.
ASR - yes, maybe, but surely only in the case where the application 
requires barge-in;
even then, I'd be interested to see some test results, particuarly 
where the outbound prompt

is killed the moment the ASR reports 

Re: [Freeswitch-users] echo cancellation on PRI cards

[David Knell]

Steve Underwood wrote:

[whopping big snip]
  

The first bit of that's a tad patronising, isn't it,


You are the one who started out being offensive.
  
I'm sorry if you find disagreement offensive; you might not wish to read 
beyond this

point if so.

and, in the case of the decade-old Aculab
cards which which I'm most familiar, is also untrue.

I can't find too much about the old cards on the web now, but I found 
http://www.amdevcomm.com/voice-mail-products/voice-mail-components/dialogic/dti_sc.html 
which is pretty much a copy and paste from the old Dialogic web pages, 
and you'll see it says "Cut through : Local echo cancellation permits 
100% detection with a >4.5 dB return loss line". The Aculabs did the 
same thing for sure. They just couldn't work without cancellation. There 
were some very early Dialogic cards, using DTMF receiver chips and OKI 
ADPCM chips, and had no general purpose DSPs. They performed really 
badly because of the lack of cancellation, and were quickly replaced 
with cards that put the OKI ADPCM, DTMF anf echo cancellation algorithms 
into a Motorola 56k DSP chips.
  

The same document, under the bit which you've quoted, says:
"(E-1) Digital trunks use separate transmit and receive paths to network.
Performance dependent on far end handset's match to local analog loop."
- i.e. the card does no echo cancellation. 

Aculab didn't even offer echo cancellation on Prosody for years and, 
when they did, it

consumed prodigious amounts of DSP.  Nonetheless, the DTMF detection worked
perfectly well, even across 120 channels per 40MHz SHARC - there's just 
no way
that those DSPs had enough horsepower to do echo cancellation across 
that many

channels.

An Asterisk box with an el-cheapo quad E1 card in that I use for TDM-SIP 
gatewaying

detects DTMF perfectly well with no echo cancellation.

You just don't need echo cancellation to achieve perfectly acceptable 
DTMF detection.


ASR - yes, maybe, but surely only in the case where the application 
requires barge-in;
even then, I'd be interested to see some test results, particuarly where 
the outbound prompt

is killed the moment the ASR reports start of speech.

I'm afraid that your original bald claim - that "IVRs badly need echo 
cancellation" is simply
wrong, misleading and irresponsible: those believing it will end up 
spending large sums

of money on technology which they probably do not need.

--Dave

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Re: [Freeswitch-users] echo cancellation on PRI cards

[David Knell]

Steve Underwood wrote:

David Knell wrote:
  

Steve Underwood wrote:

When there is Echo being generated from the far end, usually in a 
bridged call. If you application is just an IVR, with no far end 
connectivity, then you shouldn't need an echo can. If you are bridging 
calls, then at some point you may need it, depending on what else is 
in the loop.


This is VERY VERY WRONG. IVRs badly need echo cancellation. Without it 
they give very poor reliability detecting DTMF while the prompts are 
playing. If the system uses voice recognition, its reliability will be 
even worse.
  
  
With respect, this is at best half true.  DTMF detection has always 
worked just fine
without echo cancellation - the Dialogic, Aculab and Rhetorex cards 
which I used
in the late 1990s managed it perfectly well; if the DTMF detection 
code in * and FS

can't, then maybe that's something for its author to look at ;-)

Try reading the Dialogic and Aculab documentation. Those cards used 
quite a bit of their DSP capability to remove the spillback of outgoing 
voice into their DTMF receivers. You'll find the DTMF detector in 
spandsp (not necessarily the ones in * or FS, which have been altered a 
bit) is superior to either Dialogic or Aculab's.
  
The first bit of that's a tad patronising, isn't it, and, in the case of 
the decade-old Aculab

cards which which I'm most familiar, is also untrue.

As for the second, do you have any test results to back that up?  I'm 
more curious than

setting out for an argument..
ASR - yes, maybe, but L&H's ASR1500 used to work perfectly well on the 
same
hardware above back in the day.  I'd be interested to see results of 
testing an ASR
engine in with echo; unfortunately, most vendors appear to prohibit 
the publication

of test results in their licensing.

L&H used to work fine with the J series Dialogic cards. The Dialogic 
documents go into considerable details about the echo cancellation 
arrangements to make that happen.


  
You've missed the point I was trying to make.  It used to work fine with 
no echo cancellation

at all.

--Dave

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Re: [Freeswitch-users] echo cancellation on PRI cards

[David Knell]

Steve Underwood wrote:
When there is Echo being generated from the far end, usually in a 
bridged call. If you application is just an IVR, with no far end 
connectivity, then you shouldn't need an echo can. If you are bridging 
calls, then at some point you may need it, depending on what else is 
in the loop.

This is VERY VERY WRONG. IVRs badly need echo cancellation. Without it 
they give very poor reliability detecting DTMF while the prompts are 
playing. If the system uses voice recognition, its reliability will be 
even worse.
  
With respect, this is at best half true.  DTMF detection has always 
worked just fine
without echo cancellation - the Dialogic, Aculab and Rhetorex cards 
which I used
in the late 1990s managed it perfectly well; if the DTMF detection code 
in * and FS

can't, then maybe that's something for its author to look at ;-)

ASR - yes, maybe, but L&H's ASR1500 used to work perfectly well on the same
hardware above back in the day.  I'd be interested to see results of 
testing an ASR
engine in with echo; unfortunately, most vendors appear to prohibit the 
publication

of test results in their licensing.

--Dave

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Re: [Freeswitch-users] Rewriting Remote Party ID

[David Knell]

Hi Rod,

You can set it directly:
application="set">



--Dave


using these functions like this did nothing on the SIP INVITE packet :'(

seven wrote:
  
try  
bridge 
({effective_caller_id_name 
="your_name",effective_caller_id_number=""}sofia/b-leg)


On Mar 5, 2009, at 9:00 PM, rod wrote:

  


the A leg invite looks like this:
From: "Anonymous"

it has been rewritten like this:
From: "Anonymous" 

rod

rod wrote:

  

Hi Brian,

if I use the function effective_caller_id_number with my INVITE, I  
get this:


From: "Anonymous" ;tag=17geyFjX5p0gS.

this is not exactly what I'm looking for :p

rod


Brian West wrote:

  


Well this depends on how you're placing the call.. if its a standard
bridge you can on the A-Leg set
"effective_caller_id_number=000${caller_id_number}" before you call
bridge.

Is the from already in the correct format?

/b

On Mar 5, 2009, at 6:12 AM, rod wrote:



  

Dear list,

I'd like to rewrite the number in the Remote Party ID header and  
only in

this header.

ex: I'd like to prefix the caller  ID with a prefix code (000 in  
this

example) in the  RPID header :

  From: Anonymous;tag=1208367
  Remote-Party-ID:
123456 
@10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling


should become:
  From: Anonymous;tag=1208367
  Remote-Party-ID:
000123456 
@10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling


But the From field should remain unchanged.

And how to strip this prefix:
  From: Anonymous;tag=1208367
  Remote-Party-ID:
000123456 
@10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling


should become:
  From: Anonymous;tag=1208367
  Remote-Party-ID:
123456 
@10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling



regards.

  




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Re: [Freeswitch-users] FS SIP audio quality?

[David Knell]

Paul D. wrote:
> I re-tested calls to VM replacing some of FS prompts with * ones, and it 
> appears that * sounds were recorded with a better quality/higher volume, 
> so FS itself has nothing to do with that. That's solved. :-)
>   
There's a long history of people in A/B listening tests reporting louder 
as sounding
better on the same source material - even if the additional volume isn't 
detectable
as such.

Which, I guess, explains my 25 years of going to Motorhead gigs.

--Dave

-- 
David Knell, Director, 3C Limited
T: 020 8114 5002  F: 020 3002 7257  M: 07773 800623
http://www.3c.co.uk 


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Re: [Freeswitch-users] BT IPExchange Interoperability Testing

[David Knell]

Hi John,

I think we had a chat at a show at Olympia(?) a couple of years back.

We did an IPX interconnect some 12 months ago - it all went pretty well.
I'll give you a call later on: depending on where you are in the  
process,
we might be able to save you a pound or two.

Cheers --

Dave

> Hi;
>
> We're looking to set up a CP which will interact with BT's 21CN  
> network
> using the IPX gateway.
>
> We're running through the test scenarios (which, unfortunately, we  
> have
> under NDA) now.
>
> Just wondering if anyone out there has already passed the test suite
> with Freeswitch ?
>
> jd
>
> -- 
> John Daragon   argv[0] limited
> Lambs Lawn Cottage,   Staple Fitzpaine,   Taunton,   TA3 5SL,   UK
> Registered in England  Company Number 02947782
> v +44 (0) 1460 234068  f +44 (0) 1460 234069 m +44 (0) 7836 576127
>
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Re: [Freeswitch-users] Thread hijacking and BT interop

[David Knell]

Oops - I did it again ;-)

--Dave

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Re: [Freeswitch-users] Struggling with Originate

[David Knell]

Hi Nik,

How do you need to modify it?

Cheers --

Dave


Hi Guys,

I’m placing calls ok by using the event socket.  However, I need to  
modify the To: Sip header prior to the call going out for routing  
purposes.  Is it possible to do this in the Originate action?


If not, can someone explain if it’s possible to trigger part of the  
dial plan externally?  I can then modify the headers and then place  
the call/



Regards,
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Re: [Freeswitch-users] Generating calls from external source

[David Knell]

Hi Nik,

Here's a snipped in Perl that launches an outbound call:

			if (my $sock = IO::Socket::INET->new(Proto =>'tcp', PeerAddr =>  
'127.0.0.1', PeerPort => 8021)) {

print $sock "auth XXX\n\n";
print $sock "api originate {softivr_id=$siid,src_softivr_id=$siid,softivr_outdial=true}sofia/frombt/$...@1.2.3.4 
 $service\n\n";

$sock->close();
}

- it does no error checking or anything, but (line by line) it:
 - opens a socket to the event socket interface
 - authenticates
 - issues an originate which dials out to the number in $ntd.  The  
bits in {} set a bunch of variables on the channel, which are used by  
the software which processes the call later on.  The call is linked to  
the extension in $service - FS looks this up in the dialplan - which  
handles our end.

 - closes the socket

Cheers --

Dave


Thanks for that, coming from a C++ background it’s a refreshing  
change to be looking at something that seems logical and efficient.


I’d briefly looked at the event socket and wondered if that was the  
way to go.  I presume that there’s some sort of event generation  
that can trigger and external process as well somewhere, though all  
I need to do is update mysql (hopefully using some sort of pooled  
connection)


I’m not using a TDM card, I have a direct interconnect with the PSTN  
breakout provider with 1,500 channels available to me.  I’m finding  
Asterisk proving to be less than stable at high call volumes and  
load values spike at more than 100 calls with billing/accounting in  
place, hence my interest in FS.  The only thing that’s concerning me  
is XML at the moment.  Lots of code and very wordy.  I’m sure I’ll  
appreciate why XML given time


Regards,

From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org 
] On Behalf Of Michael S Collins

Sent: 03 February 2009 01:17
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Generating calls from external source

Nik,

Welcome to FreeSWITCH! The short answer is "yes, FS can do that."  
The first thing that you should do is unlearn "the Asterisk way" of  
thinking. Usually there is an elegant way of doing things in FS that  
wasn't possible in Ast.


I would recommend that you start by looking at the event socket,  
which is somewhat analogous to the AMI only cooler. :) I have  
personally done something similar to this using the event socket and  
a Perl script. The key is to learn the syntax of the originate  
command. (definitely hit the wiki and IRC channel)

Are you using TDM cards for this? Just curious.

-MC (IRC nick: mercutioviz)

Sent from my iPhone

On Feb 2, 2009, at 3:35 PM, "Nik Middleton" > wrote:

Hi Guys,

As a long time Asterisk user, I’m looking into freeswitch as an  
alternative mainly due to (list multiple reasons here)


Can anyone give me a pointer as to how I would achieve the following?

I need to replicate an emergency broadcast system currently running  
under Asterisk.


At the moment, I run through a Mysql database and using the manager  
API, issues an Originate command to dial a number.


When the call is answered, a message is played, and the recipient  
has the option of hitting a digit to confirm receipt.  I then call  
an AGI script to update the database.


Is this fairly easy to do in Freeswitch?

Not looking for code, just some pointers as to what’s available to  
do the above /


Regards,
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Re: [Freeswitch-users] mod_g729

[David Knell]

Steve Underwood wrote:
Depends what you are after. Speex offers the quality of G.729 at around 
the same processing load. However, nobody seems to want to pay for the 
processing load of G.729. Almost everything uses G.729A. Half the 
processing load, but significantly poorer quality.


VoIP is mostly a race to the bottom, and people wonder why it makes no 
money for provides. :-\
  

And, at the wholesale level, it makes no sense whatsoever to compress calls
any more: bandwidth is so cheap (and has been for a while) that the loss in
call quality - especially from tandem compressions - and the increased
processing requirements and other bits of expense do not stack up.  Case in
point: we moved a route from G.711 to G.729, and saw the ACD drop from
over 10 to under 7 minutes.  It was a route to mobiles, so the audio was 
being

recompressed with the GSM codec on its way to the handsets.  Economically,
had we carried on using G.729, we'd have lost about 30% of our margin on
that route.

--Dave

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Re: [Freeswitch-users] How to bridge without Answer?

[David Knell]

There's a whole bunch of reasons why you might not want to answer an 
inbound call:

- intercept messages (e.g. "the cellphone you've called is switched off")
- cost reduction on 1-800 calls, although you won't get a forward audio 
path from the

caller until you do answer it
- in one case, a company for whom I'd provided some IVR (back in the 
1990s) had
someone mail out some tens of thousands of cards with "You owe us X - 
you must call
this (900) number now to avoid court proceedings" on; we were able to 
not answer the
inbound leg of the call, but still play a recorded message to the caller 
informing them
that they could just ignore it.  Had we had to answer the inbound leg, 
they'd have been

charged.

--Dave

Thanks Anthony,

There are some toll-free numbers I need to configure such that, originator
does not need to charge to its users, even though they are answered on
terminator side.




Anthony Minessale-2 wrote:
  

You can't.

Why would you need that?  Are you trying to forward inbound calls from the
pstn to an ivr without answering them?
That could get you in trouble FYI.


On Wed, Jan 21, 2009 at 7:40 AM, shehzad p  wrote:



Hi all,

When I dial a number from Originator Gateway, It will route to Freeswitch
Server and then FS will bridge the call to Terminator Gateway as below.
Terminator Answer the call (and runs playback, and look for DTMF).

|Originator Gateway|---> |FreeSwitch |-->
|Terminator Gateway|

I used bridge application to route call to Terminator.
But my requirement is that when Terminator answer the call (Respnd with
200OK) , Freeswitch should NOT Answer call for A leg (Originater
Gateway).

How can be this done?

Thanks in advance.
msp.
--
View this message in context:
http://www.nabble.com/How-to-bridge-without-Answer--tp21583334p21583334.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


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--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_miness...@hotmail.com 
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sip:8...@conference.freeswitch.org 
iax:gu...@conference.freeswitch.org/888
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Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl

[David Knell]

Hi Brian,
"With FreeSWITCH not having any supported ASR at the time of writing 
(with the exception of PocketSphinx), we needed something to allow us 
to connect it to an MRCP server to test SoftIVR's ASR functionality. 
After a few false starts, we implemented a simple MRCP connector using 
the outbound socket interface, unicast and a bit of Perl."


Was mod_openmrcp not enough :)  We really need someone to fund the 
writing of mod_unimrcp.
mod_openmrcp is (from our testing) badly broken: it segfaults on the 
second call; it is (in my opinion) unnecessarily baroque and it is no 
longer supported.


Cheers --

Dave

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Re: [Freeswitch-users] SBC configuration

[David Knell]

Hi Ash,

It's the former.  Here's a snippet from a dialplan of ours - this  
takes calls with a specific prefix from a specific IP address and  
forwards them to a particular carrier:

   
 
   
 
 
 
 
   
   

- you'll need something similar for each direction.

Cheers --

Dave

> Hi,
>
> I am interested in testing Freeswitch acting as an SBC. Is it simply  
> a matter of configuring the dialplan correctly, using RE's so that  
> inbound calls are just forwarded to our internal PBX and outbound  
> calls from the PBX are forwarded to the VOIP provider?
>
> Or do I need to create an application that specifcally creates a new  
> call and then joins the inbound and outbound calls?
>
> I haven't been able to find info on the wiki or google re. SBC setup.
>
>
> thanks
> Ash
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[Freeswitch-users] FreeSWITCH, MRCP and Perl

[David Knell]

Hi all -

In case anyone's interested, I've documented how we interfaced FS with 
Lumenvox via MRCP using FS' event socket and unicast interfaces and a 
bit of Perl here: 
http://www.softivr.com/wiki/index.php/FreeSWITCH_MRCP_in_Perl

Three surprises: that it worked at all, that it works quite well and 
that it was really quite easy to do.

One thing I'm looking for: has anyone written a module which attaches a 
bug to an audio stream and forwards the audio as RTP to a specified 
IP/port to just allow audio to be tapped off a call and sent somewhere 
else to be listened to?

Cheers --

Dave

-- 
David Knell, Director, 3C Limited
T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
http://www.3c.co.uk 


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Re: [Freeswitch-users] Sending SIP calls via outbound proxy

[David Knell]

Hi Erick,

Not sure if you've tried this (or if it'll help), but you can force 
routing in the dialplan like so:




Cheers --

Dave



i forgot to give you the pastebin URL
http://pastebin.freeswitch.org/6379

  

I'm running latest trunk - Revision: 10682

I've been doing an ngrep on my external freeswitch SIP port and FS
is not sending any SIP packets anywhere when I run the following command.
Bumping up TPORT_LOG to 9 also confirms this, as no SIP packets are 
logged.


originate
'sofia/external/erick at ejjohnson.org;fs_path=proxybeta.jnctn.net' 
&echo()


Also, just to be clear, when I remove ";fs_path=..." from the command
above a call
is set up normally to erick at ejjohnson.org and the SIP packets are 
logged

to console.

Thanks guys.



What SVN rev are you running? Also, could you do a SIP trace?
TPORT_LOG=1 && /usr/local/freeswitch/bin/freeswitch
Pastebin the output of that and we'll take it from there.
-MC

On Tue, Dec 9, 2008 at 11:41 AM, Erick Johnson
 wrote:
  

Both:

originate sofia/external/'erick at


ejjohnson.org;fs_path=proxybeta.foo.net'
  

&echo()
originate sofia/external/erick at


ejjohnson.org;fs_path=proxybeta.foo.net
  

&echo()

produce the exact same result & log

:(



* I think you need to '' the sofia uri /b
  




  



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Re: [Freeswitch-users] voicemail disk quota poll

[David Knell]

I still know some folk in the 900-number business.  They won't do 
anything without

profanity in it ;-)

--Dave

Actually GM Voices won't do anything with profanity in it.

/b

On Dec 4, 2008, at 11:55 AM, Michael Collins wrote:

  

I think GM Voices levies a "naughtiness surcharge" but I'll see what I
can find out. :)
-MC




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Re: [Freeswitch-users] Bridging from Event Socket API

[David Knell]

Hi Klaus,

There's two differences that I can see between what you're doing and 
what we do:-

1.  We're using the socket in async mode (shouldn't make any difference)
2.  You don't need to send the UUID in after the sendmsg - FS already 
knows which call you're controlling.


Cheers --

Dave

Hi All,

Thanks for your feedback. I must be doing something fundamentally wrong. 
Inbound socket is working without problems. But the exact things that i do on 
inbound socket, i'm not able to replcate them on outbound socket.

The global picture: I have on Xlite registered at extension 1002 and another 
one at extension 1003.  Then i have an extension 8998 in the default context. 
Here is the extension definition:

   


  


I use Xlite-1003 to call this extension (8998) and the call is properly notified to the remote Java server. 


Then on the Java side, after receiving the event, i send a CONNECT command: 
"Connect\n\n"
The answer from Freeswitch is the state of the channel ( a set of variable, 
value pair).

Up to this point everything seems normal to me. But then, i try to send an 
answer command:

sendmsg  b30a2d2e-c146-11dd-9b99-07347b46e4ea
call-command: execute
execute-app-name: answer
execute-app-arg:

Freswitch replies with: 


Content-Type: command/reply
Reply-Text: +OK

But the call is still not answered. Nothing happens on the freeswitch console 
(Log level DEBUG) and the dialing XLite is still in calling modus.

Then i try bridging the call to 1002:

sendmsg  b30a2d2e-c146-11dd-9b99-07347b46e4ea
call-command: execute
execute-app-name: bridge
execute-app-arg: sofia/internal/1002%192.168.50.94
   
Again Freeswitch does answer with:


Content-Type: command/reply
Reply-Text: +OK

And yet again, nothing is really happening.

What am i missing here?

Thanks,
Klaus.

 Original-Nachricht 
  

Datum: Wed, 03 Dec 2008 11:15:57 +
Von: David Knell <[EMAIL PROTECTED]>
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Bridging from Event Socket API



  

Hi Klaus,

Some Perl code snippets - we use:
call_command("bridge", "sofia/gateway/bt/$ntd");
which, in turn, is:
sub call_command($$) {
my $cmd = shift;
my $arg = shift;
print $sock "sendmsg\ncall-command: execute\nexecute-app-name: 
$cmd\nexecute-app-arg: $arg\n\n";

}

Cheers --

Dave



Hi Folks,

so far i could understand how to bridge calls with Javascript. I'm
  

trying to do the same with Java via the Socket Interface. My first trials
weren't successful. maybe you can help me understand what is goin on. 


What i want to do is to bridge an existing leg (Unique-ID is known) to a
  

party that wasn't yet dialed (Unique-ID unknown). With javascript it is
something like:


session.bridge("sofia/internal/1002");

How do i do this using the event socket interface? what
  

application/command would i use with which arguments?


One way i tried to do this is to orginate a call to
  

'sofia/internal/1002' and bridge the two existing legs using uuid_bridge. 
Unfortunately, it
wasn't successful. The only message i had on the FS console is: 


2008-12-02 16:57:34 [DEBUG] switch_core_session.c:693
  

switch_core_session_queue_private_event() Send signal sofia/internal/[EMAIL 
PROTECTED]
[BREAK]


Any idea what i'm missing?

Thanks,

Klaus.





  
  

--
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T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
http://www.3c.co.uk 



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