Re: [Freeswitch-users] FS Rocks!!!!!!!!!

[Nik Middleton]

Looks good, but you've missed out billing and the key one, the event
socket which could be a chapter in it's self.

Do you have a publisher for it yet?

Regards

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Fred-145
Sent: 09 December 2009 19:55
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Rocks!!!!!!!!!



Nik Middleton wrote:
> I cannot imagine doing what I'm using FS for, with any other product.
Yes
> it's frustrating at times, but this is largely down to a lack
> documentation/samples. 

Speaking of which... would this layout be good for a book on Freeswitch?

Preface
1. VoIP, Freeswitch, FS vs. Asterisk, softswitch vs. PBX, etc.
2. Choosing hardware options (server, phones, gateways)
3. Setting up FS
4. Configuring FS (SIP, profiles/contexts, VoIP providers, SIP/POTS
gateways, etc.)
5. Administering FS (CLI and GUI)
6. Customizing dialplan (adding SIP accounts, voice-mail, etc.)
7. Performance, sound quality, other issues
8. Writing scripts (LUA, etc.), connecting to databases
9. Real-life examples (Gino's Pizza, etc.)
Conclusion
Index
-- 
View this message in context:
http://old.nabble.com/FS-Rocks%21%21%21%21%21%21%21%21%21-tp26708755p267
16612.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


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Re: [Freeswitch-users] no hang-up on B leg

[Nik Middleton]

I would have tended to agree with the glare, however, before I killed
both sides, I was back to my issue of the call not clearing down at all.
(rtp timeout eventually does it)

 

Thanks for the pointer to the source.

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Jerris
Sent: 09 December 2009 14:01
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] no hang-up on B leg

 

src/switch_ivr_bridge.c

 

This could just as well be a glare condition when the call is in process
of tearing down.

 

Mike

 

 

On Dec 8, 2009, at 6:48 PM, Nik Middleton wrote:





No doubt, but that's a little difficult as this only happens
occasionally and I have 200 calls going on at the time.  It's needle in
the haystack stuff.

 

Here's what I know.

 

I have an external process listening for DTMF events.  If I detect '*' I
do a kill uuid on the B leg.  On a number of occasions I get an error
saying the B leg doesn't exist, so I now do a double kill on the
associated leg which I get from the event.  I do not get a 'doesn't
exist' message for the A leg, which leads me to believe that process of
tearing down both bridged legs is flawed.

 

The kluge clears the B leg hang issue, so the pressure's off for me, but
when I get a few nano seconds, I'll look at the code to see if there's
anything obvious.

 

Can anyone give me a hint on what module handles bridged calls? (sorry,
being lazy and suffering from a lack of sleep)

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Jerris
Sent: 08 December 2009 16:16
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] no hangup on B leg

 

We will really need debug logs and sip traces to be able to figure out
what exactly is going on here.

 

Mike

 

On Dec 7, 2009, at 4:06 PM, Nik Middleton wrote:






Sorry no, apart from the fact that I was seeing the hangup.

 

 

I'm wondering if this a bandwidth congestion issue.  Is there anyway on
a bridged call I could trap on dtmf like look for '*' and force a
hangup?  I don't seem to able to see this tone on the B leg though.

 

Regards,



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 07 December 2009 19:12
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] no hangup on B leg

 

 

On Mon, Dec 7, 2009 at 11:01 AM, Nik Middleton
 wrote:

Hi all,

 

I'll slowly pulling my hair out on this one.  I had FS successfully
hanging up both legs on a bridge, now today, with nothing changed, I'm
not seeing a hangup of the b leg at all.

 

FS is behind a PIX, so it might be a weird NAT issue, but A leg calls
hangup just fine.  Before when I had an issue with the B leg not closing
the bridge, I was at least getting a hangup event, now it's not being
fired.  Does anyone have an idea what might be causing this?

 

Regards,

 

Time for SIP traces and debug logs. Also, do you have any logs from when
things seemed to be working so that you can compare?
-MC

 

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Re: [Freeswitch-users] no hang-up on B leg

[Nik Middleton]

No doubt, but that's a little difficult as this only happens
occasionally and I have 200 calls going on at the time.  It's needle in
the haystack stuff.

 

Here's what I know.

 

I have an external process listening for DTMF events.  If I detect '*' I
do a kill uuid on the B leg.  On a number of occasions I get an error
saying the B leg doesn't exist, so I now do a double kill on the
associated leg which I get from the event.  I do not get a 'doesn't
exist' message for the A leg, which leads me to believe that process of
tearing down both bridged legs is flawed.

 

The kluge clears the B leg hang issue, so the pressure's off for me, but
when I get a few nano seconds, I'll look at the code to see if there's
anything obvious.

 

Can anyone give me a hint on what module handles bridged calls? (sorry,
being lazy and suffering from a lack of sleep)

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Jerris
Sent: 08 December 2009 16:16
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] no hangup on B leg

 

We will really need debug logs and sip traces to be able to figure out
what exactly is going on here.

 

Mike

 

On Dec 7, 2009, at 4:06 PM, Nik Middleton wrote:





Sorry no, apart from the fact that I was seeing the hangup.

 

 

I'm wondering if this a bandwidth congestion issue.  Is there anyway on
a bridged call I could trap on dtmf like look for '*' and force a
hangup?  I don't seem to able to see this tone on the B leg though.

 

Regards,



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 07 December 2009 19:12
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] no hangup on B leg

 

 

On Mon, Dec 7, 2009 at 11:01 AM, Nik Middleton
 wrote:

Hi all,

 

I'll slowly pulling my hair out on this one.  I had FS successfully
hanging up both legs on a bridge, now today, with nothing changed, I'm
not seeing a hangup of the b leg at all.

 

FS is behind a PIX, so it might be a weird NAT issue, but A leg calls
hangup just fine.  Before when I had an issue with the B leg not closing
the bridge, I was at least getting a hangup event, now it's not being
fired.  Does anyone have an idea what might be causing this?

 

Regards,

 

Time for SIP traces and debug logs. Also, do you have any logs from when
things seemed to be working so that you can compare?
-MC

 

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[Freeswitch-users] FS Rocks!!!!!!!!!

[Nik Middleton]

Thought I'd send this little hurrah!  As there seems to have been a lot
of negativity on this list lately.

 

>From my point of view, having looked at many solutions out there, FS is
still number one with regards to flexibility and performance.  I cannot
imagine doing what I'm using FS for, with any other product.  Yes it's
frustrating at times, but this is largely down to a lack
documentation/samples.  

 

So, if you have a solution to a problem, share it by adding an entry on
the WIKI.

 

Kudos to AM and all the other dev's, as someone said once 'Don't let the
bastards grind you down'

 

Regards,

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

[Nik Middleton]

Yes I did, is it possible mod_vmd is interering?  It's stopped before I
call the start_dtmf function

 

 

session:setHangupHook("myHangupHook", "blah")

session:setInputCallback("onInput");

session:execute("vmd","start");

 

 

if (session:ready() == false) then

freeswitch.consoleLog("info", " : Call Failed!!!\n");

end

 

session:answer();

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 07 December 2009 23:21
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call

 

did you set the inputcallback too?



On Mon, Dec 7, 2009 at 4:59 PM, Nik Middleton
 wrote:

Can this be done in an lua script?

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 07 December 2009 22:18


To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call

 

 

On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton
 wrote:

Hi

 

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

Perhaps you could use bind-meta-app to bind a key combo like *1 to
whatever you want to have happen. Check it out:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app

The Local_Extension in the default.xml dialplan file has a few examples
of using this tool.
-MC

 


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Anthony Minessale II

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

AIM: anthm
MSN:anthony_miness...@hotmail.com
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IRC: irc.freenode.net #freeswitch

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sip:8...@conference.freeswitch.org
<mailto:sip%3a...@conference.freeswitch.org> 
iax:gu...@conference.freeswitch.org/888
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Re: [Freeswitch-users] Trapping dtmf on bridged call

[Nik Middleton]

Once the call is bridged, while I can see an inband DTMF event being
generated, it doesn't call my hook unfortuneately

 

function onInput(session, type, obj)

if type == "dtmf" and obj['digit'] == '*'  then

session:hangup();

return  true;

 end

 

 

session:execute("start_dtmf");

session:execute("bridge",bridgestring );

 

Am I missing something?  Before the bridge, the oninput function works
fine

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 07 December 2009 22:15
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call

 

session:execute("start_dtmf");

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



On Mon, Dec 7, 2009 at 4:02 PM, Nik Middleton
 wrote:

Hi

 

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

 

Regards,


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Anthony Minessale II

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ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_miness...@hotmail.com
<mailto:msn%3aanthony_miness...@hotmail.com> 
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FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org
<mailto:sip%3a...@conference.freeswitch.org> 
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.org
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Re: [Freeswitch-users] Trapping dtmf on bridged call

[Nik Middleton]

Can this be done in an lua script?

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 07 December 2009 22:18
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call

 

 

On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton
 wrote:

Hi

 

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

Perhaps you could use bind-meta-app to bind a key combo like *1 to
whatever you want to have happen. Check it out:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app

The Local_Extension in the default.xml dialplan file has a few examples
of using this tool.
-MC

 

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[Freeswitch-users] Trapping dtmf on bridged call

[Nik Middleton]

Hi

 

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

 

Regards,

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Re: [Freeswitch-users] no hangup on B leg

[Nik Middleton]

Sorry no, apart from the fact that I was seeing the hangup.

 

 

I'm wondering if this a bandwidth congestion issue.  Is there anyway on
a bridged call I could trap on dtmf like look for '*' and force a
hangup?  I don't seem to able to see this tone on the B leg though.

 

Regards,



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 07 December 2009 19:12
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] no hangup on B leg

 

 

On Mon, Dec 7, 2009 at 11:01 AM, Nik Middleton
 wrote:

Hi all,

 

I'll slowly pulling my hair out on this one.  I had FS successfully
hanging up both legs on a bridge, now today, with nothing changed, I'm
not seeing a hangup of the b leg at all.

 

FS is behind a PIX, so it might be a weird NAT issue, but A leg calls
hangup just fine.  Before when I had an issue with the B leg not closing
the bridge, I was at least getting a hangup event, now it's not being
fired.  Does anyone have an idea what might be causing this?

 

Regards,

 

Time for SIP traces and debug logs. Also, do you have any logs from when
things seemed to be working so that you can compare?
-MC

 

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[Freeswitch-users] no hangup on B leg

[Nik Middleton]

Hi all,

 

I'll slowly pulling my hair out on this one.  I had FS successfully
hanging up both legs on a bridge, now today, with nothing changed, I'm
not seeing a hangup of the b leg at all.

 

FS is behind a PIX, so it might be a weird NAT issue, but A leg calls
hangup just fine.  Before when I had an issue with the B leg not closing
the bridge, I was at least getting a hangup event, now it's not being
fired.  Does anyone have an idea what might be causing this?

 

Regards,

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Re: [Freeswitch-users] how to disable hook flash hold

[Nik Middleton]

It's a pots phone at the end of a VoIP trunk provided by my ISP.  I have
not control over it.

 

The only think I have found so far is:

 



 

Which is what I presume I add to my provider's conf file.

 

Regards,



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Tihomir Culjaga
Sent: 05 December 2009 19:01
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] how to disable hook flash hold

 

The POTS phone is attached to something... (ZAP channel or an ATA or a
gateway). It is there you configure this behaviour.


T.

On Sat, Dec 5, 2009 at 6:20 PM, Nik Middleton
 wrote:

Sorry, I meant from a POTS phone

Regards



-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Milena
Sent: 05 December 2009 16:05
To: freeswitch-users@lists.freeswitch.org

Subject: Re: [Freeswitch-users] how to disable hook flash hold

It can be done from the phone itself; for example on a Grandstream
phone it is done with the option "Onhook Threshold:" setting it to
"hookflash OFF"


2009/12/5 Nik Middleton 
>
> Hi,  Is it possible to disable being able to put a call on hold using
hook flash?
>
>
>
> Regards
>
>
>
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Re: [Freeswitch-users] how to disable hook flash hold

[Nik Middleton]

Sorry, I meant from a POTS phone

Regards


-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Milena
Sent: 05 December 2009 16:05
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] how to disable hook flash hold

It can be done from the phone itself; for example on a Grandstream
phone it is done with the option "Onhook Threshold:" setting it to
"hookflash OFF"


2009/12/5 Nik Middleton 
>
> Hi,  Is it possible to disable being able to put a call on hold using hook 
> flash?
>
>
>
> Regards
>
>
>
> ___
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[Freeswitch-users] how to disable hook flash hold

[Nik Middleton]

Hi,  Is it possible to disable being able to put a call on hold using
hook flash?

 

Regards

 

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Re: [Freeswitch-users] Option to hang-up both legs in a bridge

[Nik Middleton]

Thanks for that, no didn't see the message, there seems to be a big
delay in the messages getting turned around on the list.

 

Yup, works great thanks.  Script doesn't get events, so there was no way
to check for the b leg hang-up.

 

Regards,



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 04 December 2009 20:57
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Option to hang-up both legs in a bridge

 

did you see my reply to the other thread?

set the channel variable hangup_after_bridge=true on the a leg

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



On Fri, Dec 4, 2009 at 2:03 PM, Nik Middleton
 wrote:

Hi,

 

Is there an option to hang-up both call legs in a bridge when one leg
hangs up?

 

In my lua script I only ever see the hang-up for the call I'm in, not
for the bridged b leg.  That said, I can see both a hang-up and un
bridge event being fired for the B leg.  However my issue is that the A
leg is still up, and if I've called 2 Pots numbers, the phone network
will maintain the bridge.

 

Is my only option to subscribe to the unbridge event and fire a hang-up
event using the 'other leg' UID?

 

Regards,


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[Freeswitch-users] Option to hang-up both legs in a bridge

[Nik Middleton]

Hi,

 

Is there an option to hang-up both call legs in a bridge when one leg
hangs up?

 

In my lua script I only ever see the hang-up for the call I'm in, not
for the bridged b leg.  That said, I can see both a hang-up and un
bridge event being fired for the B leg.  However my issue is that the A
leg is still up, and if I've called 2 Pots numbers, the phone network
will maintain the bridge.

 

Is my only option to subscribe to the unbridge event and fire a hang-up
event using the 'other leg' UID?

 

Regards,

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[Freeswitch-users] B Leg on bridged call is not hanging up

[Nik Middleton]

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

[Nik Middleton]

Check out this range

http://www.noblesolutions.co.uk/shop/index.php?main_page=index&manufactu
rers_id=16

You should be able to find a local supplier

We've used them for a couple of years now.  They just plug into your
network.

Regards,

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Fred-145
Sent: 19 September 2009 11:34
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Affordable GSM gateway for one cellphone?


Hello

I'm selling a basic solution for SOHO customers (FS is installed on
their
work computer running Windows or Macs) to handle an analog phone line.
When they're on the road, in addition or instead of getting a
notification
by e-mail when someone calls their office, some users might want to have
the
Freeswitch server actually ring their cellphone so they can take calls.

Besides taking a subscription with a VoIP provider that the Freeswitch
server will use to ring their cellphone, I'd like to know what my
options
are when it comes to setting up a GSM gateway on the customer's
premises, in
case they don't want to depend on the Internet.

Are there Freeswitch-compatible, affordable solutions to handle a single
GSM
subscription? I guess all it takes is having them take a second
subscription
with their GSM provider and inserting the SIM chip inside the gateway to
have Freeswitch ring their cellphone, but I've never used those things.

Thank you.
-- 
View this message in context:
http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp255204
04p25520404.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


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Re: [Freeswitch-users] Getting core dump from last night's build

[Nik Middleton]

Yes I did a make current, and make sure,

 

I've now reverted back to the latest release and all's well.  

 

Is there anything else I could try so as to ensure I've not got any
issue like you suggest?

 

Regards,

 

 

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 10 September 2009 17:54
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Getting core dump from last night's
build

 

I wonder if maybe you have a build issue with an older mod_lua with a
newer FreeSWITCH
did you update via make current?



On Thu, Sep 10, 2009 at 11:11 AM, Nik Middleton
 wrote:

Hi Guys,

 

I'm getting a core dump when running an lua script that's been fine for
months

 

In Freeswitch_lua.cpp line 92 is being called, but it's not clear what
exactly this is doing

 

 

lua_State *Session::getLUA()

{

if (!L) {

switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_ERROR, "Doh!\n");

}

return L;

}

 

Anyone shed a light on this?

 

Regards

 


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Anthony Minessale II

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[Freeswitch-users] Getting core dump from last night's build

[Nik Middleton]

Hi Guys,

 

I'm getting a core dump when running an lua script that's been fine for
months

 

In Freeswitch_lua.cpp line 92 is being called, but it's not clear what
exactly this is doing

 

 

lua_State *Session::getLUA()

{

if (!L) {

switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_ERROR, "Doh!\n");

}

return L;

}

 

Anyone shed a light on this?

 

Regards

 

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Re: [Freeswitch-users] freepbx for freeswitch

[Nik Middleton]

I'd heard rumours that this was going to happen and it's great news and
good news for FS as well.  With a user friendly front end, FS is sure to
fly.  I have no doubt that this will be the first of many.

Regards,

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Giovanni Maruzzelli
Sent: 05 August 2009 22:00
To: freeswitch-users@lists.freeswitch.org;
freeswitch-...@lists.freeswitch.org
Subject: [Freeswitch-users] freepbx for freeswitch

Yay!

http://freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-fut
ure

Darren Schreiber has made the announcement and is doinng a
presentation of FreePBX V3 right now at www.cluecon.com.

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[Freeswitch-users] Outbound socket question

[Nik Middleton]

Hi Guys,

 

I'm using an outbound socket to control calls, and it works a charm.
However, what I'd like to do is send a custom event regarding the call
on hang-up.  The way I see things happening at the moment, and I could
be wrong, is that the socket is closed when a hang-up occurs, so am I
taking a chance trying to send the event then? (try to sneak out the
event before socket closure happens)  The other option is of course to
open an inbound socket and send the event, but I'd rather not do that if
possible.

 

Regards

  

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Re: [Freeswitch-users] Help Regarding memory leak with freeswitch

[Nik Middleton]

If this is Linux, there's nothing wrong with it using most of the
memory, if it starts to use the swap, then there might be an issue.
Utilizing the memory does not mean there is a memory leak

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Rajagopal, Sridhar (Sridhar)
Sent: 13 July 2009 08:07
To: 'freeswitch-users@lists.freeswitch.org'
Subject: [Freeswitch-users] Help Regarding memory leak with freeswitch

 

Hi all,

 

I am running freeswitch on powerpc processor. I see memory being
allocated  for each subsequent REGISTER requests coming to freeswitch.
But not all the memory allocated is not freed. If I run the code for two
days the system is running out of memory (RAM available to me is very
less).

The same memory issue is happening even for calls.

 

Please let me know if any body has seen this issue. Please let me know
how I can go ahead and debug this issue.

 

Thanks in advance for the help.

 

Regards,

Sridhar

 

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Re: [Freeswitch-users] Setting channel variables using event socket

[Nik Middleton]

As in 

 

call-command: set  joe=out_to_lunch ?

 

 

 

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Brian West
Sent: 12 July 2009 23:24
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting channel variables using event
socket

 

If you're going to do it that way you can just use set.

 

uuid_setvar is an api call... 

 

/b

 

On Jul 12, 2009, at 5:10 PM, Nik Middleton wrote:





HI Guys,

 

Can't seem to get this to work

 

call-command: execute

execute-app-name: uuid_setvar

execute-app-arg: cad8eb99-cdcd-4d0d-9453-20b8d9d71859 fred=out_to_lunch

 

Tried various permutations, but still nothing stored when the channel is
hung up

 

Can anyone tell me what I'm doing wrong?

 

Regards,

 

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Re: [Freeswitch-users] Setting channel variables using event socket

[Nik Middleton]

HI Guys,

 

Can't seem to get this to work

 

call-command: execute

execute-app-name: uuid_setvar

execute-app-arg: cad8eb99-cdcd-4d0d-9453-20b8d9d71859 fred=out_to_lunch

 

Tried various permutations, but still nothing stored when the channel is
hung up

 

Can anyone tell me what I'm doing wrong?

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 11 July 2009 22:51
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting channel variables using event
socket

 

I think also sendmsg with command execute works in this case if you are
using async socket but uuid_setvar always works in all cases

On Jul 11, 2009 4:27 PM, "Brian West" 
wrote:

I think you do ... 

 

/b

On Jul 11, 2009, at 4:08 PM, Nik Middleton wrote: > Excellent.
Do I need to supply uuid on an out...


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Re: [Freeswitch-users] Setting channel variables using event socket

[Nik Middleton]

Excellent. Do I need to supply uuid on an outbound socket?

 

Regards

 

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Brian West
Sent: 11 July 2009 20:51
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting channel variables using event
socket

 

uuid_setvar

 

/b

 

On Jul 11, 2009, at 2:19 PM, Nik Middleton wrote:





Hi Guys,

 

Is it possible to set a channel variable while a call is in progress
using an outbound event socket?  I have a listening process that
examines the hang-up events and it would be neat if it could also get
some variables that I have set mid call as well.  Note:  I know it's
possible to set them in the originate but that's not what I'm after

 

Regards,



 

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[Freeswitch-users] Setting channel variables using event socket

[Nik Middleton]

Hi Guys,

 

Is it possible to set a channel variable while a call is in progress
using an outbound event socket?  I have a listening process that
examines the hang-up events and it would be neat if it could also get
some variables that I have set mid call as well.  Note:  I know it's
possible to set them in the originate but that's not what I'm after

 

Regards,

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Re: [Freeswitch-users] freeswitch support PCMU only?

[Nik Middleton]

I'm ONLY use PCMA, so I would agree with Brian

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Brian West
Sent: 01 July 2009 20:35
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] freeswitch support PCMU only?

Thats bull... I just did PCMA all morning testing!  Your config is  
wrong.

/b

On Jul 1, 2009, at 1:33 PM, qian ma wrote:

> yes,PCMA enabled in x-lite.
>
> doesn't work.
>
> FS accept PCMU only.


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Re: [Freeswitch-users] bridge call from outbound socket

[Nik Middleton]

One little annoyance though,  I cannot for the life of me get a ringback
tone while the B leg is ringing,

 

I've tried putting ringback=${us-ring} in the originate params, but no
deal, just silence until the call is answered.  Anyone care to shed some
light on this?

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 29 June 2009 21:49
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] bridge call from outbound socket

 

 

On Mon, Jun 29, 2009 at 1:43 PM, Nik Middleton
 wrote:

Thanks, but I got it sorted.  The bridge application along with the
event-lock sorted.  Works a treat, yippee!  I'll write it up on the wiki

 

Cool deal.

 

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Re: [Freeswitch-users] bridge call from outbound socket

[Nik Middleton]

Thanks, but I got it sorted.  The bridge application along with the
event-lock sorted.  Works a treat, yippee!  I'll write it up on the wiki

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 29 June 2009 21:22
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] bridge call from outbound socket

 

would you mind doing a pb of your script that is handling the OB event
socket connection?
-MC

On Mon, Jun 29, 2009 at 12:47 PM, Nik Middleton
 wrote:

Hi Guys,

 

Is it possible to bridge to another destination while controlling a call
via the outbound socket?

 

In other words, I'm controlling a call using an outbound socket and at
some point want to originate a new call leg and bridge the two.

 

If it can't be done that way, I'm thinking I could originate the call
using an inbound socket, grab the uuid and then call  " api uuid_bridge
 " ?

 

 

Regards,


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[Freeswitch-users] bridge call from outbound socket

[Nik Middleton]

Hi Guys,

 

Is it possible to bridge to another destination while controlling a call
via the outbound socket?

 

In other words, I'm controlling a call using an outbound socket and at
some point want to originate a new call leg and bridge the two.

 

If it can't be done that way, I'm thinking I could originate the call
using an inbound socket, grab the uuid and then call  " api uuid_bridge
 " ?

 

 

Regards,

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Re: [Freeswitch-users] Myevents in outbound socket

[Nik Middleton]

Nope.

 

Can't find much on the Wiki on how to interface with ESL using C++.  I
want to control the outbound socket from a windows 2003 server only
because that's what I'm familiar with.  Is there some portable C++ or C
code?

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Brian West
Sent: 29 June 2009 00:00
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Myevents in outbound socket

 

Are you using ESL?

 

/b

 

On Jun 28, 2009, at 5:55 PM, Nik Middleton wrote:





Hi Guys,

 

I've almost got my c++ outbound socket control prog running, however
even though the filter works, it would be truly great to just subscribe
to myevents as even with the filter in place I get lots of channel
Execute and complete events which I don't really need.  Problem is, is
that mod_VMD isn't included in those events, even though it is channel
specific.  Is there any chance that this will be included?  If not, can
someone point me to where myevents is defined and I'll have a go at it
myself.

 

Regards,

 

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[Freeswitch-users] Myevents in outbound socket

[Nik Middleton]

Hi Guys,

 

I've almost got my c++ outbound socket control prog running, however
even though the filter works, it would be truly great to just subscribe
to myevents as even with the filter in place I get lots of channel
Execute and complete events which I don't really need.  Problem is, is
that mod_VMD isn't included in those events, even though it is channel
specific.  Is there any chance that this will be included?  If not, can
someone point me to where myevents is defined and I'll have a go at it
myself.

 

Regards,

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Re: [Freeswitch-users] Confused with event content lengths

[Nik Middleton]

OK, finally figured it out.

Have updated the Wiki to remove ambiguity and posted some SUDO code

Regards,

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Nik 
Middleton
Sent: 28 June 2009 20:38
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Confused with event content lengths

But from where? After the double LF of the header as one part of the wiki says 
or after the line containing the content-length that another part of the wiki 
says?


-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West
Sent: 28 June 2009 20:23
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Confused with event content lengths

Yes it says 264 bytes read exactly 264 bytes or die trying.

/b

On Jun 28, 2009, at 1:57 PM, João Mesquita wrote:

> If I am not mistaken, you are always safe reading the amount data  
> expressed on Content-Length since it is calculated based on the  
> total message length before it is sent out of FS.
>
> From a protocol point of view, it would indeed be much better to  
> rely on something such as Content-Length then \n\n termination  
> string. As I get to know more and more the core developers, I doubt  
> they would rely on the latter.
>
> Hope it helps...
>
> jmesquita


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Re: [Freeswitch-users] Confused with event content lengths

[Nik Middleton]

But from where? After the double LF of the header as one part of the wiki says 
or after the line containing the content-length that another part of the wiki 
says?


-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West
Sent: 28 June 2009 20:23
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Confused with event content lengths

Yes it says 264 bytes read exactly 264 bytes or die trying.

/b

On Jun 28, 2009, at 1:57 PM, João Mesquita wrote:

> If I am not mistaken, you are always safe reading the amount data  
> expressed on Content-Length since it is calculated based on the  
> total message length before it is sent out of FS.
>
> From a protocol point of view, it would indeed be much better to  
> rely on something such as Content-Length then \n\n termination  
> string. As I get to know more and more the core developers, I doubt  
> they would rely on the latter.
>
> Hope it helps...
>
> jmesquita


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[Freeswitch-users] Confused with event content lengths

[Nik Middleton]

Hi Guys,

 

I'm trying to parse events in C++ for an outbound socket.  The docs are
a little contradictory, so I wonder if someone could help me out.

 

As I understand it and event is terminated with double LF's (\n\n)
However if there is a Content-Length header the wiki very confusingly
says 

 

'Content-Length is the length of the event beginning AFTER the very next
LF only line ("\n") and inclusive the trailing LF/LF pair ("\n\n")'

 

BUT the example says it's after the \n\n in the header!! Which is it?

 

In addition, it also looks like the event body is also terminated by a
\n\n.  If this is the case, why do I care about content length value,
can't I simply read until I get the termination sequence?

 

Regards, 

 

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Re: [Freeswitch-users] Sound file or lua script not played underload

[Nik Middleton]

They're reading an audio file from a ram disk.  Wouldn't have thought
that this would cause a problem or am I wrong.  Running at around 400
concurrent calls

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 23 June 2009 19:21
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Sound file or lua script not played
underload

 

Are you making many calls share a single local_stream?
This error usually means a handle open to a local_stream is not reading
from that stream source, such as if you paused during playback of a
local_stream.
They are only a real issue if you are getting them with no calls up.



On Tue, Jun 23, 2009 at 11:54 AM, Nik Middleton
 wrote:

Hmm,

 

Looking at console I'm seeing this, does this offer any additional clues
to anyone?

 

2009-06-23 17:44:29.856574 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:44:33.620372 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:44:43.747518 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:44:45.452340 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:44:55.582892 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:44:57.285124 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:45:05.217018 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:45:07.136734 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:45:16.934136 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:45:17.751749 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:45:29.489415 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:45:30.877783 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:45:42.711411 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:45:53.924364 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:46:07.826509 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:46:17.623225 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Nik
Middleton
Sent: 23 June 2009 17:46
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Sound file or lua script not played under
load

 

Hi Guys,

 

Scratching my head on this one, under load FS is not playing an audio
file, OR and lua script is not getting executed.  Not all the time, just
some.  I've changed ulimit -n to 9 but no diff, and ideas where else
I might look?

 

Regards,


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-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_miness...@hotmail.com
<mailto:msn%3aanthony_miness...@hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org
<mailto:sip%3a...@conference.freeswitch.org> 
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.org
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pstn:213-799-1400

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Re: [Freeswitch-users] Sound file or lua script not played under load

[Nik Middleton]

Hmm,

 

Looking at console I'm seeing this, does this offer any additional clues
to anyone?

 

2009-06-23 17:44:29.856574 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:44:33.620372 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:44:43.747518 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:44:45.452340 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:44:55.582892 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:44:57.285124 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:45:05.217018 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:45:07.136734 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:45:16.934136 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:45:17.751749 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:45:29.489415 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:45:30.877783 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:45:42.711411 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:45:53.924364 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:46:07.826509 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

2009-06-23 17:46:17.623225 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Nik
Middleton
Sent: 23 June 2009 17:46
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Sound file or lua script not played under
load

 

Hi Guys,

 

Scratching my head on this one, under load FS is not playing an audio
file, OR and lua script is not getting executed.  Not all the time, just
some.  I've changed ulimit -n to 9 but no diff, and ideas where else
I might look?

 

Regards,

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[Freeswitch-users] Sound file or lua script not played under load

[Nik Middleton]

Hi Guys,

 

Scratching my head on this one, under load FS is not playing an audio
file, OR and lua script is not getting executed.  Not all the time, just
some.  I've changed ulimit -n to 9 but no diff, and ideas where else
I might look?

 

Regards,

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Re: [Freeswitch-users] high cpu utilization

[Nik Middleton]

You are indeed correct, it's the 64bit server that performs well, not
the 32bit PAE version.  I'm hoping that's the cause.  I need to dig
around and find out if it's possible to change the kernel remotely and
see it sorts the issue.  Ultimately I'll update it to 64 bit anyway, but
that's a 500 mile trek.

Regards,

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
David Burgess
Sent: 19 June 2009 16:20
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] high cpu utilization

On Fri, Jun 19, 2009 at 2:23 AM, Nik
Middleton wrote:
> I'm running Linux 2.6.18-128.1.10.el5PAE on i686 on the server, so
sadly I
> don't think that's the issue

I could be wrong, but I think PAE is a 32-bit kernel adapted for
hardware with >4GB RAM. This can create a lot of overhead compared to
running a true 64-bit kernel or a 32-bit kernel without PAE.

Confirm this with 'uname -a'

db

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Re: [Freeswitch-users] high cpu utilization

[Nik Middleton]

I'm running Linux 2.6.18-128.1.10.el5PAE on i686 on the server, so sadly
I don't think that's the issue

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Knight
Sent: 18 June 2009 21:31
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] high cpu utilization

 

Is this possibly an issue to do with a newer tickless kernel?

 

see
http://www.nabble.com/FreeSWITCH-under-the-Linux-2.6.29-kernel-td2324855
9.html

 

Tony

On Thu, Jun 18, 2009 at 3:54 PM, Nik Middleton
 wrote:

Hi Guys,

 

This one has me a little baffled.  If have a recent build (in the last
week) of FS installed on two near identical HP servers.  One happily
runs 400 concurrent calls at around 50% CPU.  The other can only run
around 50 calls without the CPU going to 98%.  Identical configs and lua
script.

 

Only diff is that the server having problems is running latest centos
64bit, where the other is 32bit.  Any suggestions of where I might start
looking?

 

Regards,


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[Freeswitch-users] high cpu utilization

[Nik Middleton]

Hi Guys,

 

This one has me a little baffled.  If have a recent build (in the last
week) of FS installed on two near identical HP servers.  One happily
runs 400 concurrent calls at around 50% CPU.  The other can only run
around 50 calls without the CPU going to 98%.  Identical configs and lua
script.

 

Only diff is that the server having problems is running latest centos
64bit, where the other is 32bit.  Any suggestions of where I might start
looking?

 

Regards,

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Re: [Freeswitch-users] When to use OpenSIPS/Kamailio in front of FS?

[Nik Middleton]

Anything that's dedicated undoubtedly has less load that something
that's multifunctioned.  However the lack of any conversations on front
ending a SIP server to FS would likely indicate that no one's found a
requirement for it at this time.

I would truly hate to see discussions of theoretical performance
advantages of one SIP server over another, when in my view, I have yet
to reach any real world limit with FS.  My FS servers are handling
100,000+ calls/day per server and are probably only at 50% capacity. (I
see no point in beating a server to pulp when it's relatively cheap to
add another if required)

Regards


-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Gavin Henry
Sent: 14 June 2009 21:34
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] When to use OpenSIPS/Kamailio in front of
FS?

Hi,

I'm excited reading all the threads about how FS blows Asterisk away
so that you don't need  OpenSIPS/Kamailio in front of FS. Surely there
must be a point when it would be advisable to do that though, as
mod_sofia can't be as good as a dedicated SIP proxy?

Thanks.

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com

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Re: [Freeswitch-users] mod_php needed

[Nik Middleton]

I couldn't agree more.  We're working with a group that are developing a 
massive PHP based music application.  They are experts in PHP and MySQL but not 
in VOIP/Telephony.  By tuning an abstraction layer that uses PHP to communicate 
with the FS event socket, allows them to work on the areas they know best and 
not worry about the telephony side too much.  We went the lua route, and don't 
use the dial plan at all.  My view is to keep all db access and processing out 
of FS as much as possible. With the event socket you simply don't need to embed 
anything apart from the essentials.

 

We are now processing 100,000+ call setups a day (4 hours) per server all using 
php scripts to drive the application.  We may well ultimately use C++ instead 
of PHP for the event socket comms, but right now PHP does just fine.

 

 

Regards

 

 



From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael 
Collins
Sent: 13 June 2009 21:57
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_php needed

 

Perhaps you should look at controlling calls via the FreeSWITCH event socket 
instead of from the dialplan. The nice thing about the event socket is that 
your call control can happen on a separate machine. There is a PHP module for 
the ESL (event socket library) and it would be relatively easy for you to get 
going. Here are some links to get you started:

http://wiki.freeswitch.org/wiki/PHP_Event_Socket
http://wiki.freeswitch.org/wiki/Event_Socket

If you absolutely MUST have call control with scripts inside of the dialplan 
then there simply is no better choice than Lua. You can learn Lua in a few 
hours, but getting mod_php finished and debugged will take time, money, and 
other resources that no one seems willing to spend. Here is some information to 
consider:

http://wiki.freeswitch.org/wiki/Mod_lua

Come join us on IRC (#freeswitch on irc.freenode.net) if you want to discuss 
this further.

-MC (IRC: mercutioviz)



2009/6/13 Christian Löschenkohl 

hello

i am working for an austrian voip carrier.
for a few months i work with freeswitch and it is simply great.
it solves our needs in many places (high volume, flexible, stable).
the only thing i really miss is the avalibilty of php as a call control 
language.
mod_php or mod_freehp are not compiling anymore and my c++ knowledge isn't
that good (or even there :-) ).
i know there is perl, i also implemented some applications (conference system 
with provisioning,
inbound call routing to our application servers, some tests as pbx), but what 
should i say -
perl and me aren't compatible in the end.

the greatest thing for us would be that mod_php comes alive again with the 
functional state
of mod_perl (http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes).
there is also a bounty entry about mod_php, to pay for this would also be an 
option and
could be discussed.

keep on with the excellent work and greetings from austria

--
Ing. Christian Löschenkohl
Technische Leitung, Forschung&  Entwicklung VoIP

xpirio
Telekommunikation&  Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenk...@xpirio.com


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Re: [Freeswitch-users] Orphaned calls

[Nik Middleton]

Will do

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Jerris
Sent: 11 June 2009 22:51
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Orphaned calls

 

If they were still showing in status, can you use gcore to dump a core
next time this happens, leave it running somewhere we can get to it and
post a thread apply all bt to Jira.

 

Mike

On Jun 11, 2009, at 5:40 PM, "Nik Middleton"
 wrote:

It was the output from show channels.  I've rebooted the server
now, so I can't run show calls.  I'll see what happens tomorrow.
Certainly running status showed 6 sessions

 

All calls are initiated using and 'Originate' from an inbound
socket

 

Regards

 

 

 





From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 11 June 2009 22:20
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Orphaned calls

 

 

    On Thu, Jun 11, 2009 at 2:07 PM, Nik Middleton <
<mailto:nik.middle...@noblesolutions.co.uk>
nik.middle...@noblesolutions.co.uk> wrote:

 

Ok, so I did a mere 86,000 calls today, but when it was all
over, I had 6 sessions remaining like the one below (number and ISP
changed)

 

Anyone have an idea why these 6 sessions remain?   I also had
120 calls that I didn't get a hang-up for, but that might be me not
processing the events fast enough.

 

Do they show on "show calls"? Or do they show up on "show
channels" only? Just curious to see if they were bridged or not.
-MC 

That said, FS was handling a steady concurrent call
level of around 350 which was awesome !!

 

Regards

 

 

UP 0 years, 0 days, 9 hours, 39 minutes, 10 seconds, 533
milliseconds, 400 microseconds

86913 session(s) since startup

 

f47e08a2-04b7-4faf-86bc-a57410f07676,inbound,2009-06-11
18:38:23,1244741903,sofia/external/ <mailto:0x...@gk.myisp.net>
0x...@gk.myisp.net,CS_NEW,


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[Freeswitch-users] Status Event

[Nik Middleton]

Not sure where enhancement requests should be posted, but here it is
anyway

 

 

I would dearly love to be able to send a status event that returns an
event style output that provides machine readable output rather than the
wordy human readable response. (I hate parsing)

 

Is there such an event already?

 

Regards

 

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Re: [Freeswitch-users] Orphaned calls

[Nik Middleton]

It was the output from show channels.  I've rebooted the server now, so
I can't run show calls.  I'll see what happens tomorrow.  Certainly
running status showed 6 sessions

 

All calls are initiated using and 'Originate' from an inbound socket

 

Regards

 

 

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 11 June 2009 22:20
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Orphaned calls

 

 

On Thu, Jun 11, 2009 at 2:07 PM, Nik Middleton
 wrote:

 

Ok, so I did a mere 86,000 calls today, but when it was all over, I had
6 sessions remaining like the one below (number and ISP changed)

 

Anyone have an idea why these 6 sessions remain?   I also had 120 calls
that I didn't get a hang-up for, but that might be me not processing the
events fast enough.

 

Do they show on "show calls"? Or do they show up on "show channels"
only? Just curious to see if they were bridged or not.
-MC 

That said, FS was handling a steady concurrent call level of
around 350 which was awesome !!

 

Regards

 

 

UP 0 years, 0 days, 9 hours, 39 minutes, 10 seconds, 533
milliseconds, 400 microseconds

86913 session(s) since startup

 

f47e08a2-04b7-4faf-86bc-a57410f07676,inbound,2009-06-11
18:38:23,1244741903,sofia/external/0x...@gk.myisp.net,CS_NEW
,


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[Freeswitch-users] Orphaned calls

[Nik Middleton]

 

Ok, so I did a mere 86,000 calls today, but when it was all over, I had
6 sessions remaining like the one below (number and ISP changed)

 

Anyone have an idea why these 6 sessions remain?   I also had 120 calls
that I didn't get a hang-up for, but that might be me not processing the
events fast enough.

 

That said, FS was handling a steady concurrent call level of around 350
which was awesome !!

 

Regards

 

 

UP 0 years, 0 days, 9 hours, 39 minutes, 10 seconds, 533 milliseconds,
400 microseconds

86913 session(s) since startup

 

f47e08a2-04b7-4faf-86bc-a57410f07676,inbound,2009-06-11
18:38:23,1244741903,sofia/external/0x...@gk.myisp.net,CS_NEW
,

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Re: [Freeswitch-users] Problems with make current

[Nik Middleton]

Thanks, that did the trick

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Jerris
Sent: 10 June 2009 19:50
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problems with make current

 

your svn update failed, 

 

rm -rf libs/pcre && svn update && ./bootstrap.sh && ./configure && make
current

 

 

On Jun 10, 2009, at 2:30 PM, Nik Middleton wrote:





Hi Guys,

 

Ran make current today, and am getting the following errors.  I ran
bootstrap and configure, but still get these messages.

 

Any ideas ?  Looks like I'm now missing some libraries

 

Regards,

 

configure: configuring in libs/pcre

configure: running /bin/sh './configure.gnu'
--prefix=/usr/local/freeswitch  --cache-file=/dev/null --srcdir=.

configure: error: cannot find sources (pcre.h.in) in .

configure: error: /bin/sh './configure.gnu' failed for libs/pcre

 

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[Freeswitch-users] Problems with make current

[Nik Middleton]

Hi Guys,

 

Ran make current today, and am getting the following errors.  I ran
bootstrap and configure, but still get these messages.

 

Any ideas ?  Looks like I'm now missing some libraries

 

Regards,

 

configure: configuring in libs/pcre

configure: running /bin/sh './configure.gnu'
--prefix=/usr/local/freeswitch  --cache-file=/dev/null --srcdir=.

configure: error: cannot find sources (pcre.h.in) in .

configure: error: /bin/sh './configure.gnu' failed for libs/pcre

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[Freeswitch-users] mod vmd and lua - Solved

[Nik Middleton]

I finally got around to looking at why mod vmd didn't appear to run when
using LUA.  Turned out that the example in the wiki was wrong.  

 

It should have been session:execute("vmd","start");

 

And not session:execute("vmd");
 
I've updated the wiki
 
Regards,

 

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[Freeswitch-users] Problems subscribing to outbound socket events

[Nik Middleton]

I've put some c++ test code together to let the outbound socket control
the call, all works as expected, apart from the event subscription

 

Sending myevents\n\ngives the channel events

 

However sending event text all\n\n doesn't give me any events apart from
the channel events.
 
 
Anyone care to suggest what I might be doing wrong?
 
Regards,

 

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

[Nik Middleton]

As Anthony comments later, using SVN for updates is usually a risky
business for most projects.  We all have been blessed by fantastic
coding to date with this project, that has lulled us into believing that
using the latest snapshot will be OK.  This is the first time that I've
had problems.  

 

I have no doubt that the DEV's have taken this onboard, but it can
sometimes be a reality check to realize that the subscribed based has
grown to such a size that regression testing now becomes mandatory if
the project is to move onto the next stage.  

 

A very valid comment was made on this thread that new features should be
disabled by default until thoroughly tested. It's all part of the
learning cycle.  In my view the trunk needs to be updated more
frequently and this should be what us mere mortals use.  To often I see
messages saying you're using a 2 week old version, that bug's been
fixed. 

 

FS, is coming to a level where code has to be managed in a more
structured way, but I have now doubt this will be addressed fairly
rapidly.

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Lars
Zeb
Sent: 02 June 2009 22:54
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Make current fails (build 13537)

 

Brian,

 

I'm probably not the only one here, but much of what I have to do to get
Freeswitch going is new to me. Never installed or really worked with
Linux and scripting; just a little xml. It is challenging. Freeswitch is
interesting, appealing and challenging. The work your group has done is
amazing. Given this, interacting with you can be intimidating.

 

I am experiencing the slow start with build 13532. I assume that "block
all ICMP" refers to the firewall/gateway. If this is correct, why is it
that I can ping the firewall from the Freeswitch box? Can you explain in
more detail what it might be on my network that is blocking ICMP? All my
clients and Freeswitch itself are behind a NAT firewall.

 

Thanks Lars

 

Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686
i686 i386 GNU/Linux

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Brian West
Sent: Monday, June 01, 2009 3:33 PM
To: freeswitch-users@lists.freeswitch.org
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 <http://www.cluecon.com/>


 

 

 

 

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[Freeswitch-users] Outbound socket question

[Nik Middleton]

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] Freeswitch taking too long to start up

[Nik Middleton]

As I understand it, a new 'feature' was added over the weekend to
resolve NAT.  If you're firewall is not allowing ICMP then FS waits
until it times out.  At this time there is no option to disable it.

 

Regards

 

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Muhammad Shahzad
Sent: 02 June 2009 11:40
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Freeswitch taking too long to start up

 

Hi,

I have just upgraded Freeswitch from svn revision 12432 to 13544. I am
using 32bit CentOS 5.3, "make current" command completes successfully
without any errors but when i start freeswitch it take considerable time
(roughly 90 - 120 seconds) to start up. During this time no message is
display on console. Once successfully started, it works fine. However,
this initial delay is really annoying. Is there anyway to reduce/remove
this delay?

Thank you.


-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com

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

[Nik Middleton]

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.

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Brian West
Sent: 01 June 2009 22:29
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Make current fails

 

I can tell you how to fix it but it'll cost ya!  :P

 

/b

 





Spoke too soon.

 

Clean compile and install, but now FS hangs for about 5 mins on startup

 

Error [unterminated ${var}] in file
/usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xm
l line 12

Error including
/usr/local/freeswitch/conf/autoload_configs/../mrcp_profiles/*.xml
(Invalid or incomplete multibyte or wide character)

 

The first error is a typo in the sample, but the second error, I don't
have that DIR at all.  I presume that this dir has been added, but how
to I create these without overwriting my working configs?

 

 

Regards

 

 

Brian West

br...@freeswitch.org

 

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


 

 

 

 

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Re: [Freeswitch-users] Make current fails

[Nik Middleton]

Spoke too soon.

 

Clean compile and install, but now FS hangs for about 5 mins on startup

 

Error [unterminated ${var}] in file
/usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xm
l line 12

Error including
/usr/local/freeswitch/conf/autoload_configs/../mrcp_profiles/*.xml
(Invalid or incomplete multibyte or wide character)

 

The first error is a typo in the sample, but the second error, I don't
have that DIR at all.  I presume that this dir has been added, but how
to I create these without overwriting my working configs?

 

 

Regards

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Nik
Middleton
Sent: 01 June 2009 21:52
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Make current fails

 

Thanks for that

 

./ bootstrap.sh

./configure

 

Did the trick

 

Regards

 

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Brian West
Sent: 01 June 2009 20:36
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Make current fails

 

Reboot strap.

 

/b

 

On Jun 1, 2009, at 2:33 PM, Nik Middleton wrote:

 

Hi Guys,

 

Been running 'make current' and appropriate intervals over the last few
months and all's been well until today

 

Now I get the following,  obviously mod_sndfile isn't happy, but I'm not
sure what to do to fix it

 

 

 

Brian West

br...@freeswitch.org

 

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


 

 

 

 

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Re: [Freeswitch-users] Make current fails

[Nik Middleton]

Thanks for that

 

./ bootstrap.sh

./configure

 

Did the trick

 

Regards

 

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Brian West
Sent: 01 June 2009 20:36
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Make current fails

 

Reboot strap.

 

/b

 

On Jun 1, 2009, at 2:33 PM, Nik Middleton wrote:





Hi Guys,

 

Been running 'make current' and appropriate intervals over the last few
months and all's been well until today

 

Now I get the following,  obviously mod_sndfile isn't happy, but I'm not
sure what to do to fix it

 

 

 

Brian West

br...@freeswitch.org

 

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


 

 

 

 

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[Freeswitch-users] Make current fails

[Nik Middleton]

Hi Guys,

 

Been running 'make current' and appropriate intervals over the last few
months and all's been well until today

 

Now I get the following,  obviously mod_sndfile isn't happy, but I'm not
sure what to do to fix it

 

 

Regards,

 

making all mod_sndfile

 cd . && /bin/sh /usr/src/trunk/libs/libsndfile/missing --run
automake-1.9 --gnu

configure.ac: required file `Cfg/install-sh' not found

configure.ac: required file `Cfg/missing' not found

examples/Makefile.am: required file `Cfg/depcomp' not found

programs/Makefile.am: required file `Cfg/compile' not found

configure.ac:12: required file `Cfg/config.guess' not found

configure.ac:12: required file `Cfg/config.sub' not found

configure.ac:49: required file `Cfg/ltmain.sh' not found

make[6]: *** [Makefile.in] Error 1

make[5]: *** [../../../../libs/libsndfile/src/libsndfile.la] Error 2

make[4]: *** [all] Error 1

make[3]: *** [mod_sndfile-all] Error 1

make[2]: *** [all-recursive] Error 1

Making all in build

 + FreeSWITCH Build Complete ---+

 + FreeSWITCH has been successfully built.  +

 + Install by running:  +

 +  +

 +   make install   +

 +--+

make[1]: *** [all-recursive] Error 1

make: *** [all] Error 2

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Re: [Freeswitch-users] DTMF recognition flaky

[Nik Middleton]

Hi Jay,

 

Have to say my DTMF works flawlessly on thousands of calls.  (SVN trunk
from a couple of days ago.  We handle around 100,000 calls/day via FS)

 

That said, I've found it depends on your SIP trunk provider.That
doesn't mean to say there isn't a problem; it's just that I haven't come
across it.  

 

Know it's not helpful, but there you go.

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Jay
Austad
Sent: 06 May 2009 19:57
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] DTMF recognition flaky

 

I'm running 1.0.4pre3.  Haven't gotten a chance to upgrade to pre7 yet.


 

2833 is the default right?  I haven't changed anything.  I'm using
voicepulse for my SIP trunks.  Is there an option I can add to that
definition to force RFC2833?

 

--

jay austad  |  612.423.1433  |  aus...@signal15.com

 

 

 

 

On May 6, 2009, at 1:46 PM, Brian West wrote:





Well it depends.. first off are you doing inband dtmf or RFC2833?
Secondly what SVN rev are you running?

 

/b

 

On May 6, 2009, at 1:44 PM, Jay Austad wrote:





Using the default installation, I've noticed that when I (or someone  
else) calls in on my SIP trunk and keys in an extension, not all of  
the numbers are recognized unless they hold the key down for at least  
1/2 second to a second.

Is there a way to improve DTMF recognition so people can just type in  
stuff without having to hold the keys down?



 

Brian West

br...@freeswitch.org

 

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


 

 

 

 

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Re: [Freeswitch-users] Hang-up event - Alternative?

[Nik Middleton]

That won't work unless I'm mistaken.  Well it will if the call is
answered, but if it fails, the lua script will not be called.  So if the
result is BUSY, the script won't be called.

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Matthew Fong
Sent: 02 May 2009 22:22
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Hang-up event - Alternative?

 

You can always have your lua script fire a custom event on
api_hangup...this will only send the data you specify in your event.

On Sat, May 2, 2009 at 1:36 PM, Nik Middleton
 wrote:

Hi Guys,

 

Is there an alternative to the hang-up event that doesn't send quite as
much data?  This event is HUGE! 

 

All I'm looking for this the result of the call, duration, dialed number
and the ability to pass variables.  The hang-up event does all of this I
know, but I also get everything including the stock market reports (just
kidding)

 

Right now I'm using custom events for successful calls and the
BACKGROUND_JOB for call fails as my application is running an lua script
on call answer, but this doesn't get called if the call fails

 

 

Regards

 

 


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[Freeswitch-users] Hang-up event - Alternative?

[Nik Middleton]

Hi Guys,

 

Is there an alternative to the hang-up event that doesn't send quite as
much data?  This event is HUGE! 

 

All I'm looking for this the result of the call, duration, dialed number
and the ability to pass variables.  The hang-up event does all of this I
know, but I also get everything including the stock market reports (just
kidding)

 

Right now I'm using custom events for successful calls and the
BACKGROUND_JOB for call fails as my application is running an lua script
on call answer, but this doesn't get called if the call fails

 

 

Regards

 

 

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Re: [Freeswitch-users] Phones become unreachable after some time

[Nik Middleton]

Xlite may be working on the timeout FS is sending.

See the following from the wiki and see if that helps, but I'm not sure

 
In domain, set


-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
paul.degt
Sent: 30 April 2009 14:35
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Phones become unreachable after some
time

Worked for Grandstream, but not for X-Lite.

Nik Middleton wrote:
> Don't know where the setting is in FS, but force them to register
every
> 120 seconds and see if that helps
>
> Regards,
>
> -Original Message-
> From: freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
> paul.degt
> Sent: 29 April 2009 20:50
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Phones become unreachable after some
> time
>
> They do, but all necessary ports for FS are open. If that is fw issue,

> are there ways to fight with it?
>
> Nik Middleton wrote:
>   
>> Do the phones and FS have a firewall between them?  If so, sounds
like
>> the pin hole in the fw is being closed.  Alot only stay open for 4
>> 
> mins
>   
>> Regards,
>>
>> -Original Message-
>> From: freeswitch-users-boun...@lists.freeswitch.org
>> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
>> paul.degt
>> Sent: 29 April 2009 20:15
>> To: freeswitch-users@lists.freeswitch.org
>> Subject: [Freeswitch-users] Phones become unreachable after some time
>>
>> I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS
binds
>> 
>
>   
>> to Mysql DB for SIP registrations, presence etc.
>> I noticed that after some time probably >30 min. phones which have
>> 
> been 
>   
>> registered but without making calls become unreachable. Meaning that
>> 
> any
>   
>> call to such extension gets forwarded to VM as if it was offline,
>> 
> until 
>   
>> I reload such phone.
>> I did try to make the phones to register every 5 min. but it does not

>> help. I also see valid registration information in sip_registrations 
>> table. X-Lite has r-port and keep alive settings on.
>> Would appreciate any hints on what can be the issue here.
>>
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Re: [Freeswitch-users] Phones become unreachable after some time

[Nik Middleton]

Don't know where the setting is in FS, but force them to register every
120 seconds and see if that helps

Regards,

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
paul.degt
Sent: 29 April 2009 20:50
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Phones become unreachable after some
time

They do, but all necessary ports for FS are open. If that is fw issue, 
are there ways to fight with it?

Nik Middleton wrote:
> Do the phones and FS have a firewall between them?  If so, sounds like
> the pin hole in the fw is being closed.  Alot only stay open for 4
mins
>
> Regards,
>
> -Original Message-
> From: freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
> paul.degt
> Sent: 29 April 2009 20:15
> To: freeswitch-users@lists.freeswitch.org
> Subject: [Freeswitch-users] Phones become unreachable after some time
>
> I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS binds

> to Mysql DB for SIP registrations, presence etc.
> I noticed that after some time probably >30 min. phones which have
been 
> registered but without making calls become unreachable. Meaning that
any
>
> call to such extension gets forwarded to VM as if it was offline,
until 
> I reload such phone.
> I did try to make the phones to register every 5 min. but it does not 
> help. I also see valid registration information in sip_registrations 
> table. X-Lite has r-port and keep alive settings on.
> Would appreciate any hints on what can be the issue here.
>
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Re: [Freeswitch-users] Phones become unreachable after some time

[Nik Middleton]

Do the phones and FS have a firewall between them?  If so, sounds like
the pin hole in the fw is being closed.  Alot only stay open for 4 mins

Regards,

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
paul.degt
Sent: 29 April 2009 20:15
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Phones become unreachable after some time

I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS binds 
to Mysql DB for SIP registrations, presence etc.
I noticed that after some time probably >30 min. phones which have been 
registered but without making calls become unreachable. Meaning that any

call to such extension gets forwarded to VM as if it was offline, until 
I reload such phone.
I did try to make the phones to register every 5 min. but it does not 
help. I also see valid registration information in sip_registrations 
table. X-Lite has r-port and keep alive settings on.
Would appreciate any hints on what can be the issue here.

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[Freeswitch-users] RECOVERY_ON_TIMER_EXPIRE

[Nik Middleton]

Can anyone tell me what would or cause the above hang-up cause?  I'm
using latest svn and get loads of these above 10 Concurrent calls

 

Regards

 

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Re: [Freeswitch-users] Optimum sound file format

[Nik Middleton]

Thanks for this.  One of the servers is using sata and the other scsii
drives, so that may be the problem, I'll give it a go.  Problem seems to
escalate past 200 active calls.  Below that all is well.

 

That said, it could also be a db issue, so I've changed my log tables to
innodb (I'm hoping that now I have row level locking as opposed to table
level it will help)

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 16 April 2009 23:25
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Optimum sound file format

 

Looking at your post, You are already using the best format.
If you do not have a fast filesystem try making a ram disk and play the
files from there instead.

if you *really* want you can use sox to turn them all into raw alaw
files and rename them with a .PCMA extension
to avoid the g711 transconding but g711 to PCM is pretty trivial. it's
more likely a file i/o distress you see.



On Thu, Apr 16, 2009 at 5:04 PM, Nik Middleton
 wrote:

Hi Guys,

 

I'm looking for the optimum audio format when using streamfile in a lua
script.

 

I've found CPU load increases rapidly with the number of threads playing
a .wav file.  Can anyone tell me the optimum when using g711a?

 

Right now the the .wav files are 

 

Audio format: PCM

Sample rate : 8 kHz

Mono

Sample Size: 16 bit

Bit rate  :128kbps

 

Will it help CPU load if I resample to a bit rate of 64kbps and sample
size of 8 bit?

 

I have read that the sample size needs to be 13-14bit  +1 for alaw/ulaw
though

 

Regards,


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-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_miness...@hotmail.com
<mailto:msn%3aanthony_miness...@hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
<mailto:paypal%3aanthony.miness...@gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org
<mailto:sip%3a...@conference.freeswitch.org> 
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.org
<mailto:googletalk%3aconf%2b...@conference.freeswitch.org> 
pstn:213-799-1400

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[Freeswitch-users] Optimum sound file format

[Nik Middleton]

Hi Guys,

 

I'm looking for the optimum audio format when using streamfile in a lua
script.

 

I've found CPU load increases rapidly with the number of threads playing
a .wav file.  Can anyone tell me the optimum when using g711a?

 

Right now the the .wav files are 

 

Audio format: PCM

Sample rate : 8 kHz

Mono

Sample Size: 16 bit

Bit rate  :128kbps

 

Will it help CPU load if I resample to a bit rate of 64kbps and sample
size of 8 bit?

 

I have read that the sample size needs to be 13-14bit  +1 for alaw/ulaw
though

 

Regards,

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[Freeswitch-users] RTP errors

[Nik Middleton]

Hi Guys,

 

I'm getting a few of these errors below

 

sofia.c:3247 sofia_handle_sip_i_state() Reinvite RTP Error!

 

Are these caused by a fax machine?  Or am I barking up the wrong tree?

 

Regards,

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[Freeswitch-users] Hi Load, but calls still perfect

[Nik Middleton]

Hi Guys,

 

I'm no Linux guru, but today I inadvertently had 1000+ call attempts
going through FS, load according to TOP was 16.5.  Calls were still
absolutely perfect.  Can I throw out the rule book on load ?  CPU was
~45% on each core. (dual)

 

Regards,

 

 

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Re: [Freeswitch-users] Another FreeSWITCH First!

[Nik Middleton]

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

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 01 April 2009 16:31
To: Freeswitch-users
Subject: [Freeswitch-users] Another FreeSWITCH First!

 

The FreeSWITCH team is excited to announce that FreeSWITCH is the first
telephony application to support the new SIP 4.1 protocol specification.

Unlike its predecessors, SIP 4.1 has been created with the collaboration
of both the jabber foundation and the IETF.  With this match made in
heaven, one can now encapsulate an xml representation of a sip message,
which in turn can encapsulate a standard SIP 2.0 message making it
possible to do more than ever before.
Other exciting features include: 

*) RFC 4109 support: (SIP over SMTP), allowing packets to traverse NAT
with ease. 

*) Full circle presence: endpoints must subscribe to each character in
the printable ASCII range that may be used to indicate presence and the
server will send an xml notification to the client for each character
that is enabled whenever a call takes place which in turn can be used to
build a SIP 4.1 FYI packet that can be sent to all the neighboring SIP
devices so they may send themselves a NOTIFY telling them that the light
should blink if the same packet happens to be sent from a neighbor.
Then when the neighbor wants to send a presence packet it establishes a
dialog with the Third Party Presence Agent TPPA and leaves the message
there.  Then it sends the server a PRESENCE packet, which is then,
relayed to the subscribers with the TPPA id so all the subscribers can
connect to the TPPA server to make the little light blink. 

*) Retirement of SDP:  SDP is deprecated in favor of a list of URL's
describing the desired codec.  The UA can then request this URL and get
the full details of the media requirements.  The media port is
negotiated through trial and error where the calling UA asks the called
UA if the port it has guessed randomly is correct via direct TCP
connection and an exchange of XML PORT MARKUP LANGUGE XPML

INVITE b...@alice.com SIP 4.1
Content-type: sip-xml-encapsulated

  

  
  
  


 



-- 
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
 
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Re: [Freeswitch-users] FS and Skypiax on Windows Video How To

[Nik Middleton]

Worked for me, just needed to add the missing codec for media player


-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Giovanni Maruzzelli
Sent: 31 March 2009 21:09
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS and Skypiax on Windows Video How To

Ciao Bipin,
there is both video and audio.

Use vlc (http://www.videolan.org/vlc/) or mplayer
(http://www.mplayerhq.hu/design7/dload.html), and you'll be ok :-).

Sincerely,

Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




2009/3/31 xbipin :
>
> hi,
>
> is it just audio or is it that im having broken codecs so cant view
any
> video?
>
> Regards,
> Bipin
>
>
>
> Giovanni Maruzzelli-3 wrote:
>>
>> Kulwinder Singh contributed this HOW TO: Freeswitch & Skype- OS
>> Microsoft Windows
>> Download 118MB HD: http://www.celliax.org/final.avi
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> =
>> www.celliax.org
>> via Pierlombardo 9, 20135 Milano
>> Italy
>> gmaruzz at celliax dot org
>> Cell : +39-347-2665618
>> Fax : +39-02-87390039
>>
>> ___
>> Freeswitch-users mailing list
>> Freeswitch-users@lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> --
> View this message in context:
http://www.nabble.com/FS-and-Skypiax-on-Windows-Video-How-To-tp22799792p
22800505.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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[Freeswitch-users] Injecting audio into live call

[Nik Middleton]

Hi Guys,

 

I know this sounds an odd question, but I need to inject audio into an
outbound call.  The reason for this is that for a pre-paid billing app,
I need to let the call initiator know they are running out of credit.
Is there a facility to do this?  Ideally I only want to let the
subscriber, I.e. the one paying for the call to hear this. In other
words 'you have 2 minutes left for this call'

 

Regards,

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Re: [Freeswitch-users] Is mod vmd working?

[Nik Middleton]

Hmm,

Well We're connected direct to E1's and it doesn't work reliably here.
That said, DTMF detect does recognise the beeps most of the time.
Perhaps there's a regional variation.  I wonder if it's country
specific.  The code looks logical.  When I get some time I'll have a
look at it and see how it can be improved.  

The concept is great and is much better that sniffing out human voice as
that's prone to false positives.  Much better to assume human and
machine.  Nothing worse than a silent call.

Regards,

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 18 March 2009 17:24
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Is mod vmd working?

2009/3/18  :
> I added a voicemail tag in  to a default extension 1001, I hear
the
> voicemail beep but still don't see vmd_detect.
>
> Mark

FYI, I've used mod_vmd but only in a TDM environment on outbound calls
via a PRI. It worked very well on for detecting answering  machine
beeps and vm beeps on cell phone voice mails.
-MC

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Re: [Freeswitch-users] Is mod vmd working?

[Nik Middleton]

Another issue with this module is the resources it consumes.  We had it
running on 50 calls yesterday and the cpu's all went to 90+%

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Shelby Ramsey
Sent: 18 March 2009 13:08
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Is mod vmd working?

 

Mark,

Because it didn't detect a "beep".  It will be be there as
vmd_detect=true if it does.  I'm not sure exactly how reliable it's
"beep" detection is.  

SDR

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Re: [Freeswitch-users] Start FreeSWITCH without any SQL but atthesame time have all info available on realtime/runtime

[Nik Middleton]

Yup, my mistake, got the wrong end of the stick.

 

However while we're on the subject of show channels, is it possible to
get a formatted response back from the event socket?  It would be nice
to interrogate an event style response. Perhaps even including some
other goodies such as load etc.

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 16 March 2009 00:16
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Start FreeSWITCH without any SQL but
atthesame time have all info available on realtime/runtime

 

Perhaps a misunderstanding?

We do not suffer from any problem at all regarding "show channels".

The reason for the link was to demonstrate the issue we are familiar
with from our asterisk days (3-4 years ago) and to help explain
how we solved it by storing the calls states in a separate table to
avoid locking the channels.

You can type show channel in FS all you want and all you are doing is
selecting from the SQLite db.

The Original question by the poster was if we can find a way to turn off
SQL and still allow show channels to work and the
answer is, sorry no.




On Sun, Mar 15, 2009 at 7:01 PM, Nik Middleton
 wrote:

To be fair, most of these messages are 4-5 years old.  That said to
date, I can crash * by repeatedly doing a 'show channels'.  All the same
FS should be robust enough to suffer this abuse.  If it's not,. the
issue needs to be investigated.

 

 

Regards,



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 15 March 2009 13:30
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Start FreeSWITCH without any SQL but at
thesame time have all info available on realtime/runtime

 

search google for bugs related to crash and show channels 
http://www.google.com/search?q=asterisk+crash+show+channels&ie=utf-8&oe=
utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a

On Sat, Mar 14, 2009 at 7:29 PM, Diego Viola 
wrote:

So how Asterisk does that "show channels" without SQL? I don't think
they use SQLite internally.

Just being curious.

Diego


On Sat, Mar 14, 2009 at 9:20 AM, Michael Jerris  wrote:
> To clarify, -nosql turns on and off only the collecting of data for
> the show commands, and can now handle higher load than the sip stack
> can.  The only thing your doing by saying -nosql is turning off the
> exact functionality you say you want.  Its similar to saying I would
> like to support sip but don't want to load mod_sofia.  There should be
> no reasons to use that command anymore, if you encounter any I would
> be interested in knowing what is going on.
>
> Mike
>
>
> On Mar 14, 2009, at 12:12 AM, Diego Viola wrote:
>
>> Yeah, but still, it would be nice to see the channels with -nosql :)
>>
>> I don't want to be a pain in the ass, just giving some user feedback.
>>
>> Regards,
>>
>> Diego
>>
>> On Fri, Mar 13, 2009 at 3:02 PM, Brian West 
>> wrote:
>>> Since we added indexes to the SQLite DB its not so bad.
>>> /b
>>> On Mar 13, 2009, at 9:36 AM, Diego Viola wrote:
>>>
>>> Oh, I thought that SQLite wasn't that great on performance and that
>>> people wanted to replace/remove it from the core.
>>>
>>> "On of the most interesting things about FreeSWITCH to me has been
>>> the
>>> fact that most data in the system such as registrations are kept in
a
>>> SQL database. The default installation uses SQLite internally though
>>> you can easily point FreeSWITCH at one of a number of other SQL
>>> servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite has
>>> become somewhat of a bottleneck in the core so future versions of
>>> FreeSWITCH will use less of it. For example, doing a "show channels"
>>> with over 500 channels in use starts to show issues. While I'm sad
to
>>> see SQLite go in these cases, I am anxious to see how Minessale
>>> replaces it."
>>>
>>> http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale
>>>
>>> I was just being curious about it :-)
>>>
>>> Regards,
>>>
>>> Diego
>>>
>
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Re: [Freeswitch-users] Start FreeSWITCH without any SQL but at thesame time have all info available on realtime/runtime

[Nik Middleton]

To be fair, most of these messages are 4-5 years old.  That said to
date, I can crash * by repeatedly doing a 'show channels'.  All the same
FS should be robust enough to suffer this abuse.  If it's not,. the
issue needs to be investigated.

 

 

Regards,



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 15 March 2009 13:30
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Start FreeSWITCH without any SQL but at
thesame time have all info available on realtime/runtime

 

search google for bugs related to crash and show channels 
http://www.google.com/search?q=asterisk+crash+show+channels&ie=utf-8&oe=
utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a



On Sat, Mar 14, 2009 at 7:29 PM, Diego Viola 
wrote:

So how Asterisk does that "show channels" without SQL? I don't think
they use SQLite internally.

Just being curious.

Diego


On Sat, Mar 14, 2009 at 9:20 AM, Michael Jerris  wrote:
> To clarify, -nosql turns on and off only the collecting of data for
> the show commands, and can now handle higher load than the sip stack
> can.  The only thing your doing by saying -nosql is turning off the
> exact functionality you say you want.  Its similar to saying I would
> like to support sip but don't want to load mod_sofia.  There should be
> no reasons to use that command anymore, if you encounter any I would
> be interested in knowing what is going on.
>
> Mike
>
>
> On Mar 14, 2009, at 12:12 AM, Diego Viola wrote:
>
>> Yeah, but still, it would be nice to see the channels with -nosql :)
>>
>> I don't want to be a pain in the ass, just giving some user feedback.
>>
>> Regards,
>>
>> Diego
>>
>> On Fri, Mar 13, 2009 at 3:02 PM, Brian West 
>> wrote:
>>> Since we added indexes to the SQLite DB its not so bad.
>>> /b
>>> On Mar 13, 2009, at 9:36 AM, Diego Viola wrote:
>>>
>>> Oh, I thought that SQLite wasn't that great on performance and that
>>> people wanted to replace/remove it from the core.
>>>
>>> "On of the most interesting things about FreeSWITCH to me has been
>>> the
>>> fact that most data in the system such as registrations are kept in
a
>>> SQL database. The default installation uses SQLite internally though
>>> you can easily point FreeSWITCH at one of a number of other SQL
>>> servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite has
>>> become somewhat of a bottleneck in the core so future versions of
>>> FreeSWITCH will use less of it. For example, doing a "show channels"
>>> with over 500 channels in use starts to show issues. While I'm sad
to
>>> see SQLite go in these cases, I am anxious to see how Minessale
>>> replaces it."
>>>
>>> http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale
>>>
>>> I was just being curious about it :-)
>>>
>>> Regards,
>>>
>>> Diego
>>>
>
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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] Getting a sip trace on the console

[Nik Middleton]

That's exactly what I was looking for, many thanks

Regards,

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Peter P GMX
Sent: 08 March 2009 12:58
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Getting a sip trace on the console

I use the ngrep tool on the OS console and write the output to a file:
ngrep -d any port 5060 -W byline >outfile.txt

Best regards
Peter

Nik Middleton schrieb:
>
> Hi Guys,
>
>  
>
> I'm trying to debug some SIP messaging issues.  Is there a way of
> doing the Asterisk equivalent of SIP Debug so I can see what's being
sent?
>
>  
>
> Regards,
>
>

>
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[Freeswitch-users] Freeswitch IAX support

[Nik Middleton]

Hi Guys,

 

Now that IAX  has been published as an RFC
(http://www.rfc-editor.org/authors/rfc5456.txt) are there any plans to
support registrations?  

 

Not a moan, just curious as to the road map.  

 

A lot of my users have Asterisk PBX's using IAX and I'd love to replace
my Asterisk central server with FS to better serve them. Yes I know I
could get them to move to using SIP, but there's a lot of them.

 

Regards

 

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[Freeswitch-users] Getting a sip trace on the console

[Nik Middleton]

Hi Guys,

 

I'm trying to debug some SIP messaging issues.  Is there a way of doing
the Asterisk equivalent of SIP Debug so I can see what's being sent?

 

Regards,

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Re: [Freeswitch-users] Lunchbox-type PC as small server?

[Nik Middleton]

We use the VIA mini ITX boards.  Great for small offices and very stable
with various fan-less options

 

Regards

 

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Henry Huang
Sent: 06 March 2009 17:11
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Lunchbox-type PC as small server?

 

Check out Shuttle XPC, they have a room for video card and a PCI slot.
But you have to think a about reliability when deployed in business
environment. I am using this as my home server.



On Fri, Mar 6, 2009 at 8:30 AM, Fred  wrote:

Hello

   I'm looking for a small, lunchbox-like PC to build a small-form
factor CRM server to sell to small companies. Ideally, it should have
one PCI slot so that I can stick a voice card to connect to an analog
phone line and run FreeSwitch as well.

I like Asus' EeeBox (www.asus.com/products.aspx?l1=24&l2=165) but it
doesn't have room for a PCI slot, and I'm concerned about its
performance.

I also like stuff from MiniITX (www.mini-itx.com) , but they're a bit
pricey, and might also not be fast enough to act as a server.

Are there brands/models you think I should look at?

Thank you.


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-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant

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Re: [Freeswitch-users] Setting External IP

[Nik Middleton]

Well Here's my problem

 

From: "FreeSWITCH" ;tag=yge6eNm7a7B0r 

 

To:  

 

CSeq: 112019702 INVITE 

 

Contact:  

 

I need to change the external IP value in the contact field to a
specific IP for this gateway as I'm losing the BYE message.  Is there
some way of manipulating this value for a given GW?

 

Regards

 

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 06 March 2009 14:23
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting External IP

 

gateways are children of profiles so if you need them to be separate you
need to make 2 profiles and run the other one on another IP or another
port.



On Fri, Mar 6, 2009 at 5:04 AM, Nik Middleton
 wrote:

Hi Guys,

 

In External.xml in  sip profiles I have

 





 

Can I override these for a given gateway profile?  I have one gateway
that's expecting a local routed IP address due to the way that it's
routed, but the other one expects the public IP, hence the need to make
it gateway specific

 

Regards,


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-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_miness...@hotmail.com
<mailto:msn%3aanthony_miness...@hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
<mailto:paypal%3aanthony.miness...@gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org
<mailto:sip%3a...@conference.freeswitch.org> 
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.org
<mailto:googletalk%3aconf%2b...@conference.freeswitch.org> 
pstn:213-799-1400

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[Freeswitch-users] Setting External IP

[Nik Middleton]

Hi Guys,

 

In External.xml in  sip profiles I have

 





 

Can I override these for a given gateway profile?  I have one gateway
that's expecting a local routed IP address due to the way that it's
routed, but the other one expects the public IP, hence the need to make
it gateway specific

 

Regards,

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[Freeswitch-users] Prefered Linux Distro to run FS on

[Nik Middleton]

Just curious here.

 

I've always followed the fedora route but became disillusioned with the
focus on the desktop rather than the server mode.  Of late I've moved my
servers to Centos.  I felt the need for stable systems. 

 

Everyone seems to slate Centos, but to my surprise Anthony recommends
Centos 5.2 which is nice to hear.  Yes I know it's not bleeding edge,
but I don't want that.

 

Any reason why I should not be running Centos with FS? (I do plan on
running 64 bit in future though)

 

Regards,

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Re: [Freeswitch-users] Hung Channels (SVN Rev 10231)

[Nik Middleton]

Well if it's any consolation, I have a 4 day ish old copy of SVN and I
have around 200 of these hung calls, though after an hour or so they did
seem to clear.

That said, FS made 138,330 call attempts today, not too shabby, and
through out the call quality was as good as the first one.  Not sure how
to debug this one.

Version: FreeSWITCH Version 1.0.trunk (12276)

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Eric
Liedtke
Sent: 05 March 2009 23:23
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Hung Channels (SVN Rev 10231)

Yup, as I mentioned to brian didn't want to clog jira with a bug that's
been fixed or report against a rev 2k+ revs behind. I was trying to work
through it as a learning exercise. And yeah I actually added a bunch of
stuff to the list_sessions function to spit out a variety of associated
variables for each session looking for a pattern somewhere to clue me
into what might be happening.

No proxy or bypass media here, just defaults.

I will keep at it and once we update the production systems, if the
problem persists I will open a bug in jira with all the neccessary
goodies. 

Thanks
-e

It's seems fuzzy now but I think on Thu, Mar 05, 2009 at 05:55:33PM
-0500 , Mathieu Rene said:
> HI,
> 
> If you suspect a bug, the place to report it is JIRA. See:
http://wiki.freeswitch.org/wiki/Reporting_Bugs 
> .
> This gives the whole team a way of following up on issues.
> 
> Also can you upgrade to svn trunk? A lot of fixes gets committed  
> daily, so its good to stay up to date.
> 
> As you seem familiar with GDB, you may symlink the .gdbinit file in  
> the support-d/ folder to your home directory.
> This will give you some FS-specific macros such as "list_sessions"  
> which will dump a list of uuids to session pointers.
> 
> In your jira, make sure you include "thread apply all bt",  
> "list_sessions" and show channels (this one goes in FS) but PLEASE  
> update to svn trunk and test again to see if it still happens.
> 
> Also, are you using proxy/bypass media or just the default?
> 
> Math
> 
> On 5-Mar-09, at 5:38 PM, Eric Liedtke wrote:
> 
> > Greetings,
> >
> > I've been using FS in production on this rev (I realize it's pretty

> > far
> > behind current) and it's been running well, save 1 issue.
> >
> > The basic setup is an SBC , 2 GiG-E ports, 1 public , 1 private. I  
> > have
> > 2 sip profiles created , 1 per ip interface. This is being used to
> > terminate traffic to a provider so calls are only 1 direction. They

> > come
> > into the private side profile, get routed via dialplan to the
gateway
> > defined in the external profile and on to the vendor. Pretty simple.
> >
> > I have noticed that under load (50 or so cps with ~800-900 bridged  
> > calls up)
> > that over time some channels on the public side seem to get  
> > "stuck".  Due to
> > the nature of how this is being used , I would expect both sip  
> > profiles to show
> > the same number of channels in use any time i do a 'sofia  
> > status' ( or at least
> > be within a channel or 2 of each other). However after a day of  
> > heavy use I had
> > a disparity of ~250 channels. These extra channels also seem to put

> > some
> > continual load on the 'system cpu' as well , reported via top.
> >
> > Of course due to the load on the box I have to keep logging turned
way
> > down. So I've been trying to troubleshoot it as best I can.
> >
> > Last night I grabbed a core file and started in with GDB today. I  
> > found
> > the 120 or so threads that represented real active calls when I took

> > the
> > corefile, I also found ~250 threads that appeared to be stuck in the
> > CS_NEW state. The backtraces on all of them looks the same,  
> > annotated below.
> >
> > I walked through the code path by hand , based on the bt's and I  
> > don't see how
> > this could be happening  unless it's a locking issue. But as far as

> > I can tell
> > each  session  has it's own mutex defined in the  
> > switch_core_session_t struct,
> > so I wouldn't think they would be stepping on each other. I also  
> > would have expected
> > if it were something of a deadlock nature it would stop processing  
> > calls all
> > together.
> >
> > I grabbed the commands from the .gdbinit (super handy btw!!) and  
> > have been trolling
> > through the variables to try to ascertain something about why these

> > threads seem to
> > be stuck, but am not having much luck even coming up with a scenario

> > to try
> > to replicate the issue.
> >
> > If anyone has any pointers as to where I might look next it would be

> > greatly
> > appreciated.
> >
> > We will be updating to the newest release soon, however I was hoping

> > to nail down
> > what is going so I can systematically replicate it and verify by  
> > testing in the lab
> > that it is fixed , rather than just pushing the new release to  
> > produvction and hopin

Re: [Freeswitch-users] Orginate: getting status of call fail

[Nik Middleton]

That's what I was wondering, however, won't the response to the bagi
(not the initial) give me the info on the call result?

 

Regards

 

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 02 March 2009 14:00
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Orginate: getting status of call fail

 

The best way would be to add a few custom variables and add a secondary
system that monitors the CDR data and uses the 
custom variables to identify what you want to do with the failed calls.




On Sat, Feb 28, 2009 at 4:49 PM, Nik Middleton
 wrote:

Hi Guys,

 

I've been running a test script written in lua which now works very well
thanks to Anthony's fix to stream file.

 

Right now I'm using an event socket to initiate the call and passing the
name of the script along with originate thus:

 

$dialstring = "originate
{ignore_early_media=true,origination,originate_timeout=25}sofia/gateway/
Mygw/phonenum '&lua(helloworld.lua )'";

$result = $obj ->bgapi_command($dialstring);

 

The script gets fired (it would appear) on answer.  However, if the
number is invalid , timed out or was busy, I'm not sure the script gets
executed or am I wrong?

 

I want to be able to fire an event back on what happed to the call in
the event that it failed for whatever reason.

 

I know I can simply call the originate and pass the number as an
argument and execute the dial within the script but I'm led to believe
that's not very efficient, or am I completely wrong?

 

Looking for the most FS friendly way here

 

Regards,


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[Freeswitch-users] Orginate: getting status of call fail

[Nik Middleton]

Hi Guys,

 

I've been running a test script written in lua which now works very well
thanks to Anthony's fix to stream file.

 

Right now I'm using an event socket to initiate the call and passing the
name of the script along with originate thus:

 

$dialstring = "originate
{ignore_early_media=true,origination,originate_timeout=25}sofia/gateway/
Mygw/phonenum '&lua(helloworld.lua )'";

$result = $obj ->bgapi_command($dialstring);

 

The script gets fired (it would appear) on answer.  However, if the
number is invalid , timed out or was busy, I'm not sure the script gets
executed or am I wrong?

 

I want to be able to fire an event back on what happed to the call in
the event that it failed for whatever reason.

 

I know I can simply call the originate and pass the number as an
argument and execute the dial within the script but I'm led to believe
that's not very efficient, or am I completely wrong?

 

Looking for the most FS friendly way here

 

Regards,

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[Freeswitch-users] Console messages

[Nik Middleton]

Hi Guys, 

 

Is there a way of displaying a console message not related to a log
level?  I've got the console only reporting errors now, but it would be
nice to be able to display a message when a given condition exists.
Yes, I could set it as an error level message, but I'd rather not do
that.

 

Regards,

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Re: [Freeswitch-users] Cant get Disposition status in Javascript

[Nik Middleton]

Works for me, see snippet below

 

var first_session = new Session(dial_string);



// Trap for call failure

if (!first_session.ready()) {

consoleLog("err", "Disposition: " +
first_session.cause + "\n");

if (first_session.cause == "USER_BUSY") {

Disposition
= "BUSY";

} 

else if (first_session.cause ==
"NO_ROUTE_DESTINATION") {

Disposition
= "DCN";

}

 


else if (first_session.cause ==
"NO_ANSWER") {

Disposition
= "NA";

}

 



exit();  

}

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Baskar
Sent: 26 February 2009 14:56
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Cant get Disposition status in
Javascript

 

Hi Anthony Minessale,

I have added these lines in my javascript with your guidance. But still
i did not get any status like  busy , no answer, etc .

 

session.setVariable("cause_code", session.causecode);

session.setVariable("cause_name", session.cause);

 

I Get this output only for all the call:

 

variable_cause_code: [0]

variable_cause_name: [NONE]

 


-- 
Warm Regards,
N.Baskar

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[Freeswitch-users] Help debuging core dump

[Nik Middleton]

Hi Guys

 

I'm having problems with seg faults about every 10 mins with call loads
> 200.  I've processed the core dump
(http://pastebin.freeswitch.org/7436) but I'm unsure what I should be
looking for. I don't see the point where the crash occurred.  Can
someone point me to where I should be looking?

 

 

FreeSWITCH Version 1.0.trunk (12246)

 

Regards,  

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Re: [Freeswitch-users] Originate and bridge with lua

[Nik Middleton]

Done

 

Seems it had a spam score of 2 for some reason

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 18 February 2009 23:39
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Originate and bridge with lua

 

i replied to your last private message and it was returned as
undeliverable.  overzealous spam server? Can you add my account to your
whitelist?



On Wed, Feb 18, 2009 at 4:06 PM, Anthony Minessale
 wrote:

You want to make it even more efficient?
when they press 1,
session:execute("transfer", "");

Then, put an extension in your dialplan to match  and do the
bridge.
Then you can exit the script and only run the script when you need it.

Your problem with js was the same issue, you should have been doing
something similar there too.

BTW,
If you make another comparison to asterisk comment, I will never answer
another email from you again I don't have time for that crap.







On Wed, Feb 18, 2009 at 3:56 PM, Nik Middleton
 wrote:

Sorted now thanks, it needed to be in the format

session:execute("bridge", "{params}sofia/gateway/Mygateway/number");

key change was '"'

Now I've converted my js script to lua going to run some tests tomorrow.

I sincerely hope it'll handle more than the 10 calls js would break at.


Here's my current setup

External prog generates bgapi calls via socket and calls originate with
name of lua script also passed.

Lua does IVR and then bridges where required.  It also fires back an
event to show result of call.

Astererisk happily does around 200 calls, I'm hoping FS will do better
or I've just been wasting my time.  Is there a more efficient way of
doing this?


Regards,






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

Sent: 18 February 2009 21:43
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Originate and bridge with lua

> Everything is working perfectly, except the bridge to another number.
> Because of the nature of the beast the bridge needs to dial an
external
> number (ie  sofia/gateway/Mygateway/num)  What I'm getting is:
>
> attempt to perform arithmetic on global 'sofia' (a nil value)
>
Can you pastebin your Lua script?
-MC

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

AIM: anthm
MSN:anthony_miness...@hotmail.com
<mailto:msn%3aanthony_miness...@hotmail.com> 
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Re: [Freeswitch-users] Originate and bridge with lua

[Nik Middleton]

Sorted now thanks, it needed to be in the format

session:execute("bridge", "{params}sofia/gateway/Mygateway/number");

key change was '"'

Now I've converted my js script to lua going to run some tests tomorrow.

I sincerely hope it'll handle more than the 10 calls js would break at.


Here's my current setup

External prog generates bgapi calls via socket and calls originate with
name of lua script also passed.

Lua does IVR and then bridges where required.  It also fires back an
event to show result of call.

Astererisk happily does around 200 calls, I'm hoping FS will do better
or I've just been wasting my time.  Is there a more efficient way of
doing this?


Regards,





-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 18 February 2009 21:43
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Originate and bridge with lua

> Everything is working perfectly, except the bridge to another number.
> Because of the nature of the beast the bridge needs to dial an
external
> number (ie  sofia/gateway/Mygateway/num)  What I'm getting is:
>
> attempt to perform arithmetic on global 'sofia' (a nil value)
>
Can you pastebin your Lua script?
-MC

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Re: [Freeswitch-users] Originate and bridge with lua

[Nik Middleton]

Hi Michael,

Yes that's exactly what it boils down to, an outbound ivr.

Everything is working perfectly, except the bridge to another number.
Because of the nature of the beast the bridge needs to dial an external
number (ie  sofia/gateway/Mygateway/num)  What I'm getting is: 

attempt to perform arithmetic on global 'sofia' (a nil value)

regards


-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 18 February 2009 21:09
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Originate and bridge with lua

On Wed, Feb 18, 2009 at 11:53 AM, Nik Middleton
 wrote:
> I'm trying to build an emergency broadcasting solution.
>
> So I place a call, and have ivr in the lua script.  But I also want to
> give them the option of speaking to someone.
>
> If they hit the option to speak to someone, while I can fire an event
to
> originate a call, I'm not sure how I could bridge the 2 call legs.
>
> Regards,

So really, it's just an outbound IVR, no? Just for a specific purpose.

I would recommend using the event socket and bgapi originate commands
from a central program/script/controller thingy. Generate the calls
and then drop them into a dialplan or script that controls them. I
like to use the dialpan but it really does not matter. Using a script
lets you make changes without doing a reloadxml command.

In any case, your originate commands could be something like this:

bgapi originate
{myvar='myval',myvar2='myval2'}sofia/gateway/mygateway/u...@domain


Have extension  do the gruntwork of confirming that you actually
had a successful call, got a human on the line, etc. It can also
handle failures that are not handled by the originate itself. (Depends
on whether or not you ignore early media.) In any case, you've got a
single dp entry that handles the mundane call handling. Then, if there
is a human on the line, you can do something like this:



Now you can write a plain Lua script that only has to handle the
delivery of the message. You can handle a DTMF event and the callback
function could use session:execute("bridge","agent") to connect the
called party with your agent.

Hope that helps.
-MC

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Re: [Freeswitch-users] Originate and bridge with lua

[Nik Middleton]

I'm trying to build an emergency broadcasting solution.

So I place a call, and have ivr in the lua script.  But I also want to
give them the option of speaking to someone.

If they hit the option to speak to someone, while I can fire an event to
originate a call, I'm not sure how I could bridge the 2 call legs.

Regards,


-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 18 February 2009 19:41
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Originate and bridge with lua

Nik,

What are you building? I'm wondering if this is the correct approach
for your application. You might be better off using the even socket
and controlling your calls from a central point.

-MC

On Wed, Feb 18, 2009 at 11:26 AM, Nik Middleton
 wrote:
> Hi Guys,
>
>
>
> It's not clear from the docs how I can originate a call from within an
lua
> script
>
>
>
> This what works in js,
>
>
>
> Question. How do I instantiate a new session, do I use the execute to
dial,
> and same for bridge?
>
>
>
> Regards,
>
>
>
> if (!first_session.ready())
>
>
>
> var new_session = new Session(tdial-string);
>
>
>
> if (!first_session.ready()) {
>
> disp_call(DROP)
>
> exit();
>
>
>
>
>
>
>
> new_session.answer();
>
>
>
> if (new_session.ready()) {
>
> bridge(first_session, new_session);
>
> }
>
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>

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[Freeswitch-users] Originate and bridge with lua

[Nik Middleton]

Hi Guys,

 

It's not clear from the docs how I can originate a call from within an
lua script

 

This what works in js,  

 

Question. How do I instantiate a new session, do I use the execute to
dial, and same for bridge?

 

Regards,

 

if (!first_session.ready())

 

var new_session = new Session(tdial-string);

 

if (!first_session.ready()) {

disp_call(DROP)

exit();  

 





new_session.answer();

 

if (new_session.ready()) {

bridge(first_session, new_session);

}   

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Re: [Freeswitch-users] Anyone running FS from a Thumb Flash USB?

[Nik Middleton]

Kristian,

You're my hero, if I hadn't come across astlinux 3 years ago, I wouldn't
be doing this stuff right now.  Not too sure if that's a good thing
though ;)

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Kristian Kielhofner
Sent: 18 February 2009 00:19
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Anyone running FS from a Thumb Flash
USB?

Ah yes, the "pinnacle of online discussion"! ;)

On Tue, Feb 17, 2009 at 7:10 PM, Michael Collins 
wrote:
> On Tue, Feb 17, 2009 at 3:43 PM, Kristian Kielhofner
>  wrote:
>> FreeSWITCH now compiles in AsLinux:
>
> Nice work! I'll go tell our friends over in the Yahoo financial forums
> - I'm sure they're dying to hear about it! ;)
> -MC
>

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

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Re: [Freeswitch-users] AddBody to events in lua

[Nik Middleton]

No, js, I was trying to break the fs_sock.php, though I found the time
was dependant on how much I echoed to the screen.

I expect lua to be even faster

Regards,

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Brian West
Sent: 18 February 2009 00:15
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] AddBody to events in lua

And you ran this in lua?

/b

On Feb 17, 2009, at 6:07 PM, Nik Middleton wrote:

>
> I ran 10,000 events, which completed in around 20 seconds, all  
> received
> and processed flawlessly.  A new one on me was arrayshift. To think  
> that
> I messed around in C for ages with circular buffers, this is so  
> simple.


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Re: [Freeswitch-users] AddBody to events in lua

[Nik Middleton]

I'll shortly post some docs on the php fs_sock.  There's also a couple
of bugs in it that I've fixed.

I ran 10,000 events, which completed in around 20 seconds, all received
and processed flawlessly.  A new one on me was arrayshift. To think that
I messed around in C for ages with circular buffers, this is so simple.

Regards,

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Brian West
Sent: 17 February 2009 23:36
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] AddBody to events in lua

Good... keep up the good work adding more docs.  ;)

/b

On Feb 17, 2009, at 5:33 PM, Nik Middleton wrote:

> Err, that's what I just posted :)
>
> Regards,


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Re: [Freeswitch-users] AddBody to events in lua

[Nik Middleton]

Err, that's what I just posted :)

Regards,

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 17 February 2009 23:30
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] AddBody to events in lua

On Tue, Feb 17, 2009 at 3:25 PM, Anthony Minessale
 wrote:
> in lua you call methods with a colon :
>
> e:addBody(blah);
>
> calling with a . implies you are going to supply the obj too
>
> e.addBody(e, blah);
>

Also, there is an explicit example here:
http://wiki.freeswitch.org/wiki/Lua#event:addBody

It looks exactly like what you're trying to do.
-MC

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Re: [Freeswitch-users] AddBody to events in lua

[Nik Middleton]

I've got it working now thanks

I've also added a working example to the Wiki (lua/addBody) which was
empty

Regards,

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 17 February 2009 23:23
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] AddBody to events in lua

> local e =
freeswitch.Event("custom",
> "dialer::dialer-result");
>
> e.addBody(custom_msg);
>
> e:fire(e);
The wiki page (http://wiki.freeswitch.org/wiki/Lua#event:fire) shows
that you fire thusly:
e:fire(); --No "e" in the parens.

Can you try it and report back?
-MC

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[Freeswitch-users] AddBody to events in lua

[Nik Middleton]

Hi Guys,

 

I'm having real problems doing something trivial, and there doesn't seem
to be any docs on this issue

 

In js I do this

 

//Disposition = disp;

//Create Custom event



custom_msg =  

"call_disposition: " + Disposition +
"\n" +

"called_number: "+ dial_num +
"\n"  ;  

 

e = new Event("custom",
"dialer::dialer-result");

e.addBody(custom_msg);

e.fire();

 

And it works

 

In lua I try this

 

--Disposition = disp;

--Create Custom event



custom_msg = "call_disposition: " .. Disposition .. "\n" ..

"called_number: ".. dial_num
.."\n"  ;  

 

local e = freeswitch.Event("custom",
"dialer::dialer-result");

e.addBody(custom_msg);

e:fire(e);

 

This doesn't work, I get an error :  Error in addBody expected 2..2
args, got 1

 

What are the arguments?  It seems to be looking for a pointer for the
first one, but there's nothing on the wiki on this.  

 

 

Regards,

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