Re: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel!

[Michael Collins]

On Thu, Sep 17, 2009 at 5:53 PM, Michael Giagnocavo wrote:

> Oh, weird. Seems to work in other languages.
>
> Yet another reason to use Lua instead of JS. :)
-MC
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[Freeswitch-users] mod_lcr and indexes

[Marcelo Sosa - LST]

Hello all,

This is my first message on the list, i´m pretty new to FS.
I was playing a bit with mod_lcr and found that the sql query for fetching the 
lowest rate can be changed to a better use of indexes, at least on mysql. 
Anyone can do some test using other DBs?

The change i've made was simple, the original query was something about "... 
AND digits IN (12345, 1234, 123, 12, 1) ..." and using EXPLAIN i saw that it 
was using carrier_id as key for the biggest table and not digits. I've changed 
the code so the query is " AND (digits='12345' OR digits='1234' OR digits='123' 
OR digits='12' OR digits='1') " and mysql uses the index from the digits row, 
reducing the returned resultset of the subquery from all the digits from a 
carrier to the number of "OR" in the query (in my case, from 19850+ to 14).

Anyone think that this may be a nice change? or it is just a bad use of indexes 
by mysql?

Regards,
Marcelo Sosa___
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Re: [Freeswitch-users] skill-based ACD

[Pete Mueller]

I've worked with a few of these commercial systems, so let me provide my perspective.  Or just skip to the end for my solution.  Disclaimer: I am not an expert with all things mod_fifo can do.
 
First, I don't like the idea of using the work "skill" because I am one of those in the camp that the work to predict the specific needs of a incoming call is not worth it.  However, the general concept of queueing callers in buckets is very useful.
 
In general "skill"-based ACD has two parts, a series of buckets that callers are funneled into based on their issue, and a collection of agents that have knowledge in or can assist with issues in one or more buckets.  From what I've seen, a true ACD needs to go beyond this.  Some common issues are:
 
How the caller is treated while on hold
What happens while a caller is waiting to speak to an attendant.  Popular things are to play music, announce their position in the queue or estimate their time before being answered, and interrupt music with one or more advertisements for products.  
 
using mod_fifo: There is of course the ability to play music or other audio.  However, I remember reading once that FS loads the entire audio file in memory during playback (my knowledge may be outdated here)  which may be a resource issue with lots of callers and large audio files.  Regardless, if I want to play some music, then play an ad, more music, ad, etc.  I have two options.  either string all the audio together in one large file and loop it, or break the caller out of the queue and re-add them with the next audio clip.  The latter option risks losing their position in the queue (again, not sure, never tried the option).
 
How the caller is prioritized into the queue
Many customers have simple straight-forward requirements as to allocating callers into buckets.  Some, however have complex decision processes that may take into account the customer's support SLA, time of day, number of agents on duty, how many times they've called in the last X days, the particular issues they are having, the state of the company/systems.  In practice every customer's requirement is simple logic, but different from every other customer.
 
using mod_fifo: the 10 buckets that you can allocate callers to will be great for the simple cases.  Although one could make an arguement for more buckets.  The more complex cases could be handled by pre-calculating the priority using a scripiting language then placing them in a bucket, however that does not account for the situation where the caller's priority might change as theiy are waiting (One SLA required that if a caller waited more that 10 minutes, they were immediately bumped to a level 2 tech support engineer).
 
Which queues the agents can accept calls from
This is easier to deal with, in most of the cases the agents have a set knowledge set, and will take calls from a set number of buckets.  But there are a few gotchas.  One example is the level 2 support agent that knows product A and B.  Does he pull calls buckets A-2, B-2, A-1, B-1 or from buckets A-2, A-1, B-2, B-1  The answer may depend on how many calls are in each of the buckets at the time 
 
using mod_fifo: Again, the 10 buckets and assigning agents to multiple buckets works great for the simple cases, the more advanced cases however are not supported directly, and will require scripting.
 
Number of agents active so all calls are answered
Agents-to-buckets is one issue, Agent availablity is also a question.  Part of this is actually a staff forcasting problem.  But many commercial ACDs will provide reporting, if not out-right do this activity.  I've even seen some systems that will tell the agents when to take lunches/breaks based on active call volume and historical information.
 
using mod_fifo: There are a few variable that will asist with this type of reporting but nothing that will actual assist in doing the work of determining staffing and scheduling breaks.  One can make a very valid arguement that this really isn't the switch's problem.  But, customers do ask for it, and the more you can do with the least number of systems, the more you are likely to win a bid.
 
My Solution 
I sat down to answer this thread because it touches on a larger question.  Where does the switch stop and the application begin.  FS can do a great many things, but in my opinion, some of the modules and capabilties go beyond what a switch would do and are more of what a telephony application should do.  Looking at the architecture of FS I see four methods for implementing a feature:
1. write it into the switch core 
2. write a module to be loaded into FS address space
3. write a script to be executed on a channel
4. write an external application that uses ESL to control FS and it's channels
 
Some features have very obvious choices for which method to use, like bridge two channels (#1) or CDR geneation (#2).  Others, like IVR systems can work equally well as 2, 3, or 4.
 
In my view FS is a switch, and anyth

Re: [Freeswitch-users] Build problems with Shoutcast module under Debian

[Jason White]

It turns out that Debian recently removed the libogg.la file, deliberately,
from the package.


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[Freeswitch-users] Build problems with Shoutcast module under Debian

[Jason White]

While trying to build FreeSWITCH rev. 14913, compilation failed with the
following.

the operating system is Debian Sid. Ogg development files are installed, but
libogg.la does not exist anywhere. I'm still using libtool 1.5.26, because the
build problems with FreeSWITCH and libtool 2 under Debian haven't been
resolved.

As soon as someone takes over the Debian packaging I'll gladly help out with
testing and fixes - I'm far too busy at the moment to work on it intensively.

ranlib .libs/libshout.a
rm -fr .libs/libshout.lax
creating libshout.la
/bin/sed: can't read /usr/lib/libogg.la: No such file or directory
libtool: link: `/usr/lib/libogg.la' is not a valid libtool archive
make[10]: *** [libshout.la] Error 1


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Re: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel!

[Michael Giagnocavo]

Oh, weird. Seems to work in other languages. 

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Giovanni 
Maruzzelli
Sent: Thursday, September 17, 2009 5:27 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Callback in Javascript, session.destroy() does 
not free the channel!

On Fri, Sep 18, 2009 at 12:08 AM, Michael Giagnocavo  
wrote:
> Dispose is a .NET only thing. But I think you are right - with anthm's
> changes, any way you kill your session, if you're on the right thread, it
> should really hangup.
>

Problem is, we are trying to *not answer* the incoming call, get the
callid from the ring, destroy the session, create another session (on
the same, monoline interface), and make an outbound call.

Javascript (last svn) give us a

2009-09-18 01:18:49.291721 [ERR] inline:1  Session is not answered!

if we try to session.hangup() a session that was not answered (by the
way, it makes sense).

-giovanni




-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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Re: [Freeswitch-users] Simple call waiting question

[Mark Campbell-Smith]

Thanks Brian,  (I bought a dud phone.. and its a new DECT! - crazy)

I am using the 5900 and 5901 for parking/unparking.  That
functionality works fine and I can park/unpark the B leg as I wish.

The problem is that if I park the B-leg, the A-leg then gets a busy
signal.  If the A leg is then hung up, a user-busy signal is sent to
the C-leg, so the call goes to voicemail.

What I want to happen is park B and answer C directly.

Is this possible?


On Thu, Sep 17, 2009 at 11:21 PM, Brian West  wrote:
> Personally I would throw the phone in the trash.  :P
>
> In the default dialplan look at 5900 for park and 5901 for unpark.
>
> /b
>
>
> On Sep 17, 2009, at 7:58 AM, Mark Campbell-Smith wrote:
>
>> I am trying to create a simple call waiting dialplan as my phone does
>> not have Recall button.
>
>
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Re: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel!

[Giovanni Maruzzelli]

On Fri, Sep 18, 2009 at 12:08 AM, Michael Giagnocavo  
wrote:
> Dispose is a .NET only thing. But I think you are right – with anthm’s
> changes, any way you kill your session, if you’re on the right thread, it
> should really hangup.
>

Problem is, we are trying to *not answer* the incoming call, get the
callid from the ring, destroy the session, create another session (on
the same, monoline interface), and make an outbound call.

Javascript (last svn) give us a

2009-09-18 01:18:49.291721 [ERR] inline:1  Session is not answered!

if we try to session.hangup() a session that was not answered (by the
way, it makes sense).

-giovanni




-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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Re: [Freeswitch-users] skill-based ACD

[Michael Collins]

On Thu, Sep 17, 2009 at 4:13 PM, Phillip Jones  wrote:

> I would be interested in this too
>
> Concerning mod_fifo - can you restrict an agent to a slot. So lets say DNIS
> A is for product A and DNIS B product B - some agents know both - some know
> just 1 product  -  would that be possible?
>

Per Tony's post:

when you assign an agent to take calls off hook you can set a fifo_pop_order
variable that tells you which array indexes to service and in what order.
so if you pretend slot 1 is for general problems and slot 2 is for hard
problems you can put one agent in 1,2 and a more stupid agent in just 1

So it's a matter of setting up the routing on the calls and the priorities
that the agents are allowed to answer.
-MC
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Re: [Freeswitch-users] skill-based ACD

[Phillip Jones]

I would be interested in this too

Concerning mod_fifo - can you restrict an agent to a slot. So lets say DNIS
A is for product A and DNIS B product B - some agents know both - some know
just 1 product  -  would that be possible?

On Thu, Sep 17, 2009 at 3:17 PM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:

> I can tell you from years of painful experience, don't use asterisk for
> queues.
> see http://www.freeswitch.org/node/117
>
> You don't have to use FS, but please don't let the asterisk siren lure you
> to the rocks.
>
> mod_fifo is like a tool with basic functions you can exploit however you
> wish, it does not try to do high level
> features because those are best left in external logic.
>
>
> mod_fifo has priorities which means each individual fifo is really an array
> of 10 fifos
> when you set the priority you are choosing which index in the array to
> insert the caller.
> when an agent belongs to a queue he drills down the array from 0-9 so you
> could for instance put everyone in 5 by default and put more
> important people in 0 so they always go to the front
>
> when you assign an agent to take calls off hook you can set a
> fifo_pop_order variable that tells you which array indexes to service and in
> what order.
> so if you pretend slot 1 is for general problems and slot 2 is for hard
> problems you can put one agent in 1,2 and a more stupid agent in just 1
>
> *shrug*
>
>
>
> On Thu, Sep 17, 2009 at 1:56 PM, Christian Jensen <
> christian.jen...@teligence.net> wrote:
>
>>  This would be a fantastic addition – my company is currently looking to
>> Asterisk as a potential candidate for this if FS can’t do it.
>>
>>
>>
>> I want FS to win of course J
>>
>>
>>
>> *Christian Jensen*
>> Software Development Manager
>>
>> Back Office
>>   --
>>
>> *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
>> freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Michael
>> Collins
>> *Sent:* Thursday, September 17, 2009 11:20 AM
>> *To:* freeswitch-users@lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] skill-based ACD
>>
>>
>>
>>
>>
>> On Sun, Sep 13, 2009 at 8:01 AM, mark morreny 
>> wrote:
>>
>> Hello
>>
>>
>>
>> Has any tried setting up an ACD based on skillset?  The current out-of-box
>> version of fifo does not seem to support acd based on agent skillset.  Does
>> anyone have any experience in doing it with some external scripting using
>> lua or javascript?
>>
>>
>>
>> I am interested in hearing how others may have done it as I am trying to
>> implement one myself.
>>
>>
>>
>> thx,
>>
>>
>>
>> mark
>>
>>
>>
>>
>> I was curious about this myself. Even if someone has built a non-free
>> skills-based ACD using FS I'd like to know about it.
>> -MC
>>
>>
>>
>> ___
>> FreeSWITCH-users mailing list
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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> 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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>
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Re: [Freeswitch-users] skill-based ACD

[João Mesquita]

I would be very interested in getting my poor programming skills into
getting some decent real skill based routing working and shut those Avaya
bastards up.

Functional model? Get it to me and I will try to make it happen as time lets
me.

jmesquita

On Thu, Sep 17, 2009 at 7:17 PM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:

> I can tell you from years of painful experience, don't use asterisk for
> queues.
> see http://www.freeswitch.org/node/117
>
> You don't have to use FS, but please don't let the asterisk siren lure you
> to the rocks.
>
> mod_fifo is like a tool with basic functions you can exploit however you
> wish, it does not try to do high level
> features because those are best left in external logic.
>
>
> mod_fifo has priorities which means each individual fifo is really an array
> of 10 fifos
> when you set the priority you are choosing which index in the array to
> insert the caller.
> when an agent belongs to a queue he drills down the array from 0-9 so you
> could for instance put everyone in 5 by default and put more
> important people in 0 so they always go to the front
>
> when you assign an agent to take calls off hook you can set a
> fifo_pop_order variable that tells you which array indexes to service and in
> what order.
> so if you pretend slot 1 is for general problems and slot 2 is for hard
> problems you can put one agent in 1,2 and a more stupid agent in just 1
>
> *shrug*
>
>
>
> On Thu, Sep 17, 2009 at 1:56 PM, Christian Jensen <
> christian.jen...@teligence.net> wrote:
>
>>  This would be a fantastic addition – my company is currently looking to
>> Asterisk as a potential candidate for this if FS can’t do it.
>>
>>
>>
>> I want FS to win of course J
>>
>>
>>
>> *Christian Jensen*
>> Software Development Manager
>>
>> Back Office
>>   --
>>
>> *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
>> freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Michael
>> Collins
>> *Sent:* Thursday, September 17, 2009 11:20 AM
>> *To:* freeswitch-users@lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] skill-based ACD
>>
>>
>>
>>
>>
>> On Sun, Sep 13, 2009 at 8:01 AM, mark morreny 
>> wrote:
>>
>> Hello
>>
>>
>>
>> Has any tried setting up an ACD based on skillset?  The current out-of-box
>> version of fifo does not seem to support acd based on agent skillset.  Does
>> anyone have any experience in doing it with some external scripting using
>> lua or javascript?
>>
>>
>>
>> I am interested in hearing how others may have done it as I am trying to
>> implement one myself.
>>
>>
>>
>> thx,
>>
>>
>>
>> mark
>>
>>
>>
>>
>> I was curious about this myself. Even if someone has built a non-free
>> skills-based ACD using FS I'd like to know about it.
>> -MC
>>
>>
>>
>> ___
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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_miness...@hotmail.com 
> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
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>
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Re: [Freeswitch-users] Event socket library bug on hangup?

[Anthony Minessale]

it's a race between getting the event or getting the disconnect from the
socket where if you need to get the channel hangup event instead of just
getting the socket disconnect you can send linger which makes the socket
connection wait for the hangup event to disconnect.


On Thu, Sep 17, 2009 at 2:08 PM, Shameem Shiek  wrote:

> What does "linger" do? I do not see it documented anywhere.
>
> Why would I get a SERVER_DISCONNECTED instead of CHANNEL_HANGUP ? By the
> way the BUG does not happen every time and normally I *DO* see the
> CHANNEL_HANGUP come through in most cases.
>
> Thanks for the help.
>
>
> On Thu, Sep 17, 2009 at 2:56 PM, Anthony Minessale <
> anthony.miness...@gmail.com> wrote:
>
>> try sending the linger command
>>
>> linger\n\n
>>
>>
>>
>>
>> On Thu, Sep 17, 2009 at 1:40 PM, Shameem Shiek wrote:
>>
>>> When a call is hangup, I get SERVER_DISCONNECTED Event over and over
>>> again instead of CHANNEL_HANGUP event. Has anyone else experienced this? I
>>> am using freeswitch 1.04.
>>>
>>>
>>>
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>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_miness...@hotmail.com 
>> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
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>>
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-- 
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MSN:anthony_miness...@hotmail.com 
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Re: [Freeswitch-users] fax detection

[Tihomir Culjaga]

Hi Michael, thanks for your response.

i think it will be enough to check the call capability... we always know the
call is fax. We just need to apply the correct protocol :P


let's suppose you have 2 incoming calls:

   1. SDP containing G711, gsm, T.38 caps
   2. SDP containing G711, gsm caps


the caps will be known on within INVITE message and FS can act accordingly.

   - If there is no T.38 support within SDP, start fax application
   (SpanDSP).
   - If there is T.38 suport within the SDP, route the call to some
   predefined gateway meant for T.38 fax receiving.

In both cases, the fax should be received :P.

So, any chance to route the call according to T.38 caps within SDP message?



Now, for sending faxes there is some challenge we still have 2 calls:

we start sending a fax inband (SpanDSP) as this is the only thing we know...
and:

   - if we receive 200 OK with SDP containing G711, anyCaps => continue with
   InBand Fax
   - if we receive 200 OK with DSP containing anyCompressedCaps, T.38 =>
   drop the call without sending ACK to 200 OK ... and move the fax to be sent
   into a different directory.

well, as i said, while receiving faxes will work 100%, sending is tricky...
but it might work.

what do you guys think?

T.




On Thu, Sep 17, 2009 at 10:51 PM, Michael Collins wrote:

>
>
> On Tue, Sep 15, 2009 at 8:45 AM, Tihomir Culjaga wrote:
>
>> Hi,
>>
>> is there any way to route fax calls according to the call capability?
>>
>> I mean .. if the fax call supports T.38 i'd like to route it to a T.38
>> capable gateway. All other fax calls (meaning inband) should be handled by
>> FS/SpanDSP.
>> Of course, I know that every fax call starts as a voice call and upon fax
>> tone detection additional capabilities are being negotiated(T.38 or G711).
>> Can it be done in early media, before the call is even answered?
>>
>> I don't claim to be an expert in all this, especially T.38, but if I
> understand correctly, in both cases the call needs to be answered first. I'm
> pretty sure that the sending fax machine won't start emitting the 1100Hz
> tone until the receiving end answers. Also, with T.38 doesn't the call have
> to come up and then T.38 gets negotiated? (I don't know, I have only read
> about it.)
> -MC
>
>>
>>
>> So, here the goal is to have a T.38 capable GW handling T.38 calls while
>> SpanDSP handling T.30...
>>
>>
>> Any chance to do that with FS?
>>
>>
>>
>> ___
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Re: [Freeswitch-users] skill-based ACD

[Anthony Minessale]

I can tell you from years of painful experience, don't use asterisk for
queues.
see http://www.freeswitch.org/node/117

You don't have to use FS, but please don't let the asterisk siren lure you
to the rocks.

mod_fifo is like a tool with basic functions you can exploit however you
wish, it does not try to do high level
features because those are best left in external logic.


mod_fifo has priorities which means each individual fifo is really an array
of 10 fifos
when you set the priority you are choosing which index in the array to
insert the caller.
when an agent belongs to a queue he drills down the array from 0-9 so you
could for instance put everyone in 5 by default and put more
important people in 0 so they always go to the front

when you assign an agent to take calls off hook you can set a fifo_pop_order
variable that tells you which array indexes to service and in what order.
so if you pretend slot 1 is for general problems and slot 2 is for hard
problems you can put one agent in 1,2 and a more stupid agent in just 1

*shrug*



On Thu, Sep 17, 2009 at 1:56 PM, Christian Jensen <
christian.jen...@teligence.net> wrote:

>  This would be a fantastic addition – my company is currently looking to
> Asterisk as a potential candidate for this if FS can’t do it.
>
>
>
> I want FS to win of course J
>
>
>
> *Christian Jensen*
> Software Development Manager
>
> Back Office
>   --
>
> *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
> freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, September 17, 2009 11:20 AM
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] skill-based ACD
>
>
>
>
>
> On Sun, Sep 13, 2009 at 8:01 AM, mark morreny 
> wrote:
>
> Hello
>
>
>
> Has any tried setting up an ACD based on skillset?  The current out-of-box
> version of fifo does not seem to support acd based on agent skillset.  Does
> anyone have any experience in doing it with some external scripting using
> lua or javascript?
>
>
>
> I am interested in hearing how others may have done it as I am trying to
> implement one myself.
>
>
>
> thx,
>
>
>
> mark
>
>
>
>
> I was curious about this myself. Even if someone has built a non-free
> skills-based ACD using FS I'd like to know about it.
> -MC
>
>
>
> ___
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>
>


-- 
Anthony Minessale II

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

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MSN:anthony_miness...@hotmail.com 
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Re: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel!

[Michael Giagnocavo]

Dispose is a .NET only thing. But I think you are right - with anthm's changes, 
any way you kill your session, if you're on the right thread, it should really 
hangup.

-Michael

From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Phillip 
Jones
Sent: Thursday, September 17, 2009 3:38 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Callback in Javascript, session.destroy() does 
not free the channel!

session.dispose();

???
On Thu, Sep 17, 2009 at 5:20 PM, Alberto Escudero 
mailto:aep.li...@it46.se>> wrote:
We are trying to create a callback application in Javascript. We get the
callerid from the unanswered call and after destroying the session, we
initiate a callback to the user to conenct it to a local extension in the
dialplan.

Although we have tried to destroy the first session, or even invoke a
second script using apiExecute("jsrun",dialer.js"), tried session.hangup()
or exit()... the first session does not seem to close properly until the
whole chain of scripts are completed.

Here is a piece of code that shows the concept (yes!, the sleep function
is far from ideal. CPU loves it! )

function sleep(milliseconds) {
 var start = new Date().getTime();
 for (var i = 0; i < 1e7; i++) {
   if ((new Date().getTime() - start) > milliseconds){
 break;
   }
 }
}

if (session.ready()) {
   //We catch the caller_id
   caller_id_num = session.caller_id_num;

   console_log("Now we got your Caller ID\n");

   //How long we want to wait to trigger a call back
   session.execute("sleep",5000);

   console_log("We have waited a while... time to create the
callback\n");

   //apiExecute("jsrun", "callback.js");
   }

//Destroy the session...
session.destroy();
session=undefined;

sleep(1);

//Preparing callback
session2 = new
Session('{ignore_early_media=true}celliax/interface1/600464646');
session2.setAutoHangup(false);
session2.answer();
exit();

++
Wisdom thoughts?

--
Stopping junk mailers is good for the environment




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Re: [Freeswitch-users] "barge in" implementation with mod_socket and eavesdrop

[Anthony Minessale]

to do that you would have to transfer all the parties into a conference

On Thu, Sep 17, 2009 at 3:57 PM, Artem Shiyanov  wrote:

> Anthony,
> thank you much, "three_way" is a powerful app!
> I've added small description in wiki, if you don't mind.
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_three_way
>
> Meanwhile, another question has been evolved:
> In words of the scenario below, now if A hangs up the call then all other
> (B, C) channels also being hanged up automatically by FreeSwitch. Is there
> any way to save the call B <--> C if A has hanged up the phone? Again, the
> whole scenario is the same as it is described in my first message.
>
>
> Artem
>
>
>
> On Tue, Sep 15, 2009 at 7:08 PM, Anthony Minessale <
> anthony.miness...@gmail.com> wrote:
>
>> yes call the app as "three_way" like i said in the other thread.
>>
>>
>> On Tue, Sep 15, 2009 at 9:22 AM, Artem Shiyanov wrote:
>>
>>> Hello!
>>>
>>> I'm trying to implement "barge in" functionality (see
>>> http://www.yourdictionary.com/telecom/barge-in) with "eavesdrop" but
>>> still with no success.
>>>
>>> The situation is:
>>> - Person A calls to the extension:
>>> 
>>>   
>>> 
>>> 
>>>   
>>> 
>>>
>>> - I bridge him with person B with help of mod_socket:
>>> SendMsg 
>>> call-command: execute
>>> execute-app-name: bridge
>>> execute-app-arg: 
>>>
>>> - A and B talks
>>>
>>> - Person C decides to barge in the call A<-->B (to become a third
>>> participator in the call)
>>> a) I send (mod_socket):
>>> SendMsg 
>>> call-command: execute
>>> execute-app-name: eavesdrop
>>> execute-app-arg: 
>>>
>>> b) Then, as the spec says (
>>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop)
>>> I sent DTMF "3" with
>>> api uuid_send_dtmf  3
>>> but it doesn't work. I mean that A can hear B and vice verse, but both A
>>> and B can't hear C. C also doesn't hear neither A nor B.
>>>
>>> If I press "3" on the C's softphone (latest X-Lite) then, really, C
>>> becomes a full-capabilities participator of the call.
>>> Instead of "uuid_send_dtmf" I tried:
>>> 1)
>>> sendevent DTMF
>>> Unique-ID: 
>>> DTMF-Digit: 3
>>> DTMF-Duration: 2000
>>>
>>> 2) first make queue_dtmf for the , and then eavesdrop
>>>
>>> 3)
>>> SendMsg 
>>> call-command: execute
>>> execute-app-name: gentones
>>> execute-app-arg: 3
>>>
>>> 4)
>>> SendMsg 
>>> call-command: execute
>>> execute-app-name: send_dtmf
>>> execute-app-arg: 3
>>>
>>> And none of these methods leads to the "barged in" call.
>>>
>>> Anyone knows how to press "3" programmatically on behalf of the given
>>> channel with mod_socket?!
>>>
>>>
>>> Artem
>>>
>>> ___
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users@lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> 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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>>
>>
>
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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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Re: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel!

[Phillip Jones]

session.dispose();

???

On Thu, Sep 17, 2009 at 5:20 PM, Alberto Escudero  wrote:

> We are trying to create a callback application in Javascript. We get the
> callerid from the unanswered call and after destroying the session, we
> initiate a callback to the user to conenct it to a local extension in the
> dialplan.
>
> Although we have tried to destroy the first session, or even invoke a
> second script using apiExecute("jsrun",dialer.js"), tried session.hangup()
> or exit()... the first session does not seem to close properly until the
> whole chain of scripts are completed.
>
> Here is a piece of code that shows the concept (yes!, the sleep function
> is far from ideal. CPU loves it! )
>
> function sleep(milliseconds) {
>  var start = new Date().getTime();
>  for (var i = 0; i < 1e7; i++) {
>if ((new Date().getTime() - start) > milliseconds){
>  break;
>}
>  }
> }
>
> if (session.ready()) {
>//We catch the caller_id
>caller_id_num = session.caller_id_num;
>
>console_log("Now we got your Caller ID\n");
>
>//How long we want to wait to trigger a call back
>session.execute("sleep",5000);
>
>console_log("We have waited a while... time to create the
> callback\n");
>
>//apiExecute("jsrun", "callback.js");
>}
>
> //Destroy the session...
> session.destroy();
> session=undefined;
>
> sleep(1);
>
> //Preparing callback
> session2 = new
> Session('{ignore_early_media=true}celliax/interface1/600464646');
> session2.setAutoHangup(false);
> session2.answer();
> exit();
>
> ++
> Wisdom thoughts?
>
> --
> Stopping junk mailers is good for the environment
>
>
>
>
> ___
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Re: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel!

[Michael Giagnocavo]

So, what happens is that when you are executing an app, the state is 
CS_EXECUTE. Even if the session is hungup, the state machine doesn't go through 
all the hangup code until your app executes. 

The easiest workaround is probably to start a background api (bgapi?) call to a 
script. This will happen on another thread, then allow your current thread to 
execute and the hangup code will execute. This should work just fine, I think. 
(You can stop reading here.)

But wait, there's even more fun! anthm recently checked in a change a couple 
days that lets you work around this. Don't call destroy, call hangup on the 
session, on that session's thread. This will perform a hangup, then progress 
the state machine. Then the session will truly be hungup. Maybe you need update 
your freeswitch code, if this is not happening for you.

If you updated and hangup still isn't hanging up, you might want to ask 
specifically about that. Or, you may need to call 
switch_core_session_hangup_state directly -- just hangup alone might not do the 
trick. This is a C function, and not exposed to languages by default - you can 
either patch javascript plugin to expose this safely (and I have no idea what 
this means for the javascript runtime), or use a more capable plugin like 
mod_managed which _does_ expose all the C functions, and lets you call in and 
out of them as you please.

And now, someone who knows what they're talking about will chime in and point 
out what I got wrong.

Thanks,
-Michael

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Alberto 
Escudero
Sent: Thursday, September 17, 2009 3:20 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Callback in Javascript, session.destroy() does not 
free the channel!

We are trying to create a callback application in Javascript. We get the
callerid from the unanswered call and after destroying the session, we
initiate a callback to the user to conenct it to a local extension in the
dialplan.

Although we have tried to destroy the first session, or even invoke a
second script using apiExecute("jsrun",dialer.js"), tried session.hangup()
or exit()... the first session does not seem to close properly until the
whole chain of scripts are completed.

Here is a piece of code that shows the concept (yes!, the sleep function
is far from ideal. CPU loves it! )

function sleep(milliseconds) {
  var start = new Date().getTime();
  for (var i = 0; i < 1e7; i++) {
if ((new Date().getTime() - start) > milliseconds){
  break;
}
  }
}

if (session.ready()) {
//We catch the caller_id
caller_id_num = session.caller_id_num;

console_log("Now we got your Caller ID\n");

//How long we want to wait to trigger a call back
session.execute("sleep",5000);

console_log("We have waited a while... time to create the
callback\n");

//apiExecute("jsrun", "callback.js");
}

//Destroy the session...
session.destroy();
session=undefined;

sleep(1);

//Preparing callback
session2 = new
Session('{ignore_early_media=true}celliax/interface1/600464646');
session2.setAutoHangup(false);
session2.answer();
exit();

++
Wisdom thoughts?

-- 
Stopping junk mailers is good for the environment




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[Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel!

[Alberto Escudero]

We are trying to create a callback application in Javascript. We get the
callerid from the unanswered call and after destroying the session, we
initiate a callback to the user to conenct it to a local extension in the
dialplan.

Although we have tried to destroy the first session, or even invoke a
second script using apiExecute("jsrun",dialer.js"), tried session.hangup()
or exit()... the first session does not seem to close properly until the
whole chain of scripts are completed.

Here is a piece of code that shows the concept (yes!, the sleep function
is far from ideal. CPU loves it! )

function sleep(milliseconds) {
  var start = new Date().getTime();
  for (var i = 0; i < 1e7; i++) {
if ((new Date().getTime() - start) > milliseconds){
  break;
}
  }
}

if (session.ready()) {
//We catch the caller_id
caller_id_num = session.caller_id_num;

console_log("Now we got your Caller ID\n");

//How long we want to wait to trigger a call back
session.execute("sleep",5000);

console_log("We have waited a while... time to create the
callback\n");

//apiExecute("jsrun", "callback.js");
}

//Destroy the session...
session.destroy();
session=undefined;

sleep(1);

//Preparing callback
session2 = new
Session('{ignore_early_media=true}celliax/interface1/600464646');
session2.setAutoHangup(false);
session2.answer();
exit();

++
Wisdom thoughts?

-- 
Stopping junk mailers is good for the environment




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Re: [Freeswitch-users] IAX Origination error

[Michael Collins]

On Thu, Sep 17, 2009 at 3:01 PM, Meftah Tayeb wrote:

>  hello,
> i have a problem while originating a call using mod_iax:
> originate IAX/gu...@pbx.digium.com/s 1000
> freeswitch will crach and exit automatikaly
> anthm updated the mod_iax but same problem
> see the trace in the
> Pastebin 
> (http://pastebin:freeswi...@pastebin.freeswitch.org/1040)
>
FYI, Meftah reports that the correct pastebin is:
 http://pastebin.freeswitch.org/10408

-MC
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Re: [Freeswitch-users] IAX Origination error

[Meftah Tayeb]

Meftah Tayeb a écrit :

hello,
i have a problem while originating a call using mod_iax:
originate IAX/gu...@pbx.digium.com/s 1000
freeswitch will crach and exit automatikaly
anthm updated the mod_iax but same problem
see the trace in the
Pastebin <http://pastebin:freeswi...@pastebin.freeswitch.org/1040>
(http://pastebin:freeswi...@pastebin.freeswitch.org/1040)
this is the asterisk definition of this extension:
exten => 500,n,Dial(IAX2/gu...@pbx.digium.com/s...@default); Call the 
Asterisk demo

any help?
thanks


__ Information from ESET NOD32 Antivirus, version of virus 
signature database 4435 (20090917) __


The message was checked by ESET NOD32 Antivirus.

http://www.eset.com


__ Information from ESET NOD32 Antivirus, version of virus 
signature database 4435 (20090917) __


The message was checked by ESET NOD32 Antivirus.

http://www.eset.com



__ Information from ESET NOD32 Antivirus, version of virus 
signature database 4435 (20090917) __


The message was checked by ESET NOD32 Antivirus.

http://www.eset.com


__ Information from ESET NOD32 Antivirus, version of virus 
signature database 4435 (20090917) __


The message was checked by ESET NOD32 Antivirus.

http://www.eset.com


hello,
sory for the ERROR in the URL
the pb URL is:
http://pastebin.freeswitch.org/10408
thanks



__ Information from ESET NOD32 Antivirus, version of virus signature 
database 4435 (20090917) __

The message was checked by ESET NOD32 Antivirus.

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Re: [Freeswitch-users] [Freeswitch-dev] Please Help: TDM Hardware For Testing

[Shelby Ramsey]

MC,

Just sent you the message offline ...

SDR

Michael Collins wrote:
> Hello all!
>
> We could use your help with something. We've had several volunteers 
> willing to assist with testing and debugging various scenarios with 
> TDM circuits. However, there is a need to get TDM hardware into the 
> hands of those volunteers. If you have old TDM hardware, be it analog 
> or digital, that is sitting around collecting dust and you'd like to 
> donate it (or loan it) to the cause then please email me off list.
>
> Thanks!
> -MC
> 
>
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[Freeswitch-users] Please Help: TDM Hardware For Testing

[Michael Collins]

Hello all!

We could use your help with something. We've had several volunteers willing
to assist with testing and debugging various scenarios with TDM circuits.
However, there is a need to get TDM hardware into the hands of those
volunteers. If you have old TDM hardware, be it analog or digital, that is
sitting around collecting dust and you'd like to donate it (or loan it) to
the cause then please email me off list.

Thanks!
-MC
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[Freeswitch-users] IAX Origination error

[Meftah Tayeb]

hello,
i have a problem while originating a call using mod_iax:
originate IAX/gu...@pbx.digium.com/s 1000
freeswitch will crach and exit automatikaly
anthm updated the mod_iax but same problem
see the trace in the
Pastebin <http://pastebin:freeswi...@pastebin.freeswitch.org/1040>
(http://pastebin:freeswi...@pastebin.freeswitch.org/1040)
this is the asterisk definition of this extension:
exten => 500,n,Dial(IAX2/gu...@pbx.digium.com/s...@default); Call the 
Asterisk demo

any help?
thanks


__ Information from ESET NOD32 Antivirus, version of virus 
signature database 4435 (20090917) __


The message was checked by ESET NOD32 Antivirus.

http://www.eset.com


__ Information from ESET NOD32 Antivirus, version of virus 
signature database 4435 (20090917) __


The message was checked by ESET NOD32 Antivirus.

http://www.eset.com




__ Information from ESET NOD32 Antivirus, version of virus signature 
database 4435 (20090917) __

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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Re: [Freeswitch-users] "barge in" implementation with mod_socket and eavesdrop

[Artem Shiyanov]

Anthony,
thank you much, "three_way" is a powerful app!
I've added small description in wiki, if you don't mind.
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_three_way

Meanwhile, another question has been evolved:
In words of the scenario below, now if A hangs up the call then all other
(B, C) channels also being hanged up automatically by FreeSwitch. Is there
any way to save the call B <--> C if A has hanged up the phone? Again, the
whole scenario is the same as it is described in my first message.


Artem



On Tue, Sep 15, 2009 at 7:08 PM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:

> yes call the app as "three_way" like i said in the other thread.
>
>
> On Tue, Sep 15, 2009 at 9:22 AM, Artem Shiyanov wrote:
>
>> Hello!
>>
>> I'm trying to implement "barge in" functionality (see
>> http://www.yourdictionary.com/telecom/barge-in) with "eavesdrop" but
>> still with no success.
>>
>> The situation is:
>> - Person A calls to the extension:
>> 
>>   
>> 
>> 
>>   
>> 
>>
>> - I bridge him with person B with help of mod_socket:
>> SendMsg 
>> call-command: execute
>> execute-app-name: bridge
>> execute-app-arg: 
>>
>> - A and B talks
>>
>> - Person C decides to barge in the call A<-->B (to become a third
>> participator in the call)
>> a) I send (mod_socket):
>> SendMsg 
>> call-command: execute
>> execute-app-name: eavesdrop
>> execute-app-arg: 
>>
>> b) Then, as the spec says (
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop)
>> I sent DTMF "3" with
>> api uuid_send_dtmf  3
>> but it doesn't work. I mean that A can hear B and vice verse, but both A
>> and B can't hear C. C also doesn't hear neither A nor B.
>>
>> If I press "3" on the C's softphone (latest X-Lite) then, really, C
>> becomes a full-capabilities participator of the call.
>> Instead of "uuid_send_dtmf" I tried:
>> 1)
>> sendevent DTMF
>> Unique-ID: 
>> DTMF-Digit: 3
>> DTMF-Duration: 2000
>>
>> 2) first make queue_dtmf for the , and then eavesdrop
>>
>> 3)
>> SendMsg 
>> call-command: execute
>> execute-app-name: gentones
>> execute-app-arg: 3
>>
>> 4)
>> SendMsg 
>> call-command: execute
>> execute-app-name: send_dtmf
>> execute-app-arg: 3
>>
>> And none of these methods leads to the "barged in" call.
>>
>> Anyone knows how to press "3" programmatically on behalf of the given
>> channel with mod_socket?!
>>
>>
>> Artem
>>
>> ___
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>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
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> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
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>
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Re: [Freeswitch-users] skill-based ACD

[Michael Collins]

On Thu, Sep 17, 2009 at 12:13 PM, Remko Kloosterman
wrote:

>  I have been working on several voice projects in the past with ACD
> features, mostly based on TDM technology. It's all commercial stuff, but I
> have the experience and I am willing to share that. If anyone wishes to
> start such a development I'm sure I can dig up a functional model and help
> with the design.
>
>
I would like to see the functional model. That sounds interesting. We could
take it from there. Perhaps the FS community will have a few members willing
to help out.
-MC
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Re: [Freeswitch-users] skill-based ACD

[Michael Collins]

2009/9/17 Paweł Pierścionek 

> Michael Collins pisze:
> >
> >
> > On Sun, Sep 13, 2009 at 8:01 AM, mark morreny  > > wrote:
> >
> > Hello
> >
> > Has any tried setting up an ACD based on skillset?  The current
> > out-of-box version of fifo does not seem to support acd based on
> > agent skillset.  Does anyone have any experience in doing it with
> > some external scripting using lua or javascript?
> >
> > I am interested in hearing how others may have done it as I am
> > trying to implement one myself.
> >
> > thx,
> >
> > mark
> >
> >
> > I was curious about this myself. Even if someone has built a non-free
> > skills-based ACD using FS I'd like to know about it.
> > -MC
> >
> What is a skills-based ACD ?
> My FS based ACD allows agents to log-in to multiple queues at once and
> have different priority setting in each of the queues - does this count
> as skills-based ?
> It does not allow to limit the number of calls an agent can answer from
> a queue daily nor does it allow to set distribution of calls to an agent
> from his queues (other the by priority).
>
> Pawel,
>

I would describe this scenario as "poor man's skills-based routing" - it's a
bit of a hack but if it works then great. It isn't truly skills-based
routing but it emulates some of the functionality of skills-based routing.
-MC
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Re: [Freeswitch-users] Event socket library bug on hangup?

[Michael Collins]

On Thu, Sep 17, 2009 at 1:48 PM, Shameem Shiek  wrote:

> I found the linger command on a old freeswitch user's email thread. Updated
> the Event socket outbound wiki with the command.
>
>
> http://wiki.freeswitch.org/wiki/Event_socket_outbound#Events

Look for "linger" in the events section of that page...
-MC
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Re: [Freeswitch-users] skill-based ACD

[Phillip Jones]

see here: *http://en.wikipedia.org/wiki/Skills_based_routing*

On Thu, Sep 17, 2009 at 4:42 PM, Phillip Jones  wrote:

> >> My FS based ACD allows agents to log-in to multiple queues at once
>
> What is the difference between those queues? Does each q require that
> agents have a different skill? ie Agent A has skills x,y,z and queue 1,2,3
> have calls about x,y,z - there agent A logins to queue 1,2,3.
>
> 2009/9/17 Paweł Pierścionek 
>
> Michael Collins pisze:
>> >
>> >
>> > On Sun, Sep 13, 2009 at 8:01 AM, mark morreny > > > wrote:
>> >
>> > Hello
>> >
>> > Has any tried setting up an ACD based on skillset?  The current
>> > out-of-box version of fifo does not seem to support acd based on
>> > agent skillset.  Does anyone have any experience in doing it with
>> > some external scripting using lua or javascript?
>> >
>> > I am interested in hearing how others may have done it as I am
>> > trying to implement one myself.
>> >
>> > thx,
>> >
>> > mark
>> >
>> >
>> > I was curious about this myself. Even if someone has built a non-free
>> > skills-based ACD using FS I'd like to know about it.
>> > -MC
>> >
>> What is a skills-based ACD ?
>> My FS based ACD allows agents to log-in to multiple queues at once and
>> have different priority setting in each of the queues - does this count
>> as skills-based ?
>> It does not allow to limit the number of calls an agent can answer from
>> a queue daily nor does it allow to set distribution of calls to an agent
>> from his queues (other the by priority).
>>
>> Pawel,
>>
>>
>> ___
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>
>
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Re: [Freeswitch-users] Event socket library bug on hangup?

[Shameem Shiek]

I found the linger command on a old freeswitch user's email thread. Updated
the Event socket outbound wiki with the command.

On Thu, Sep 17, 2009 at 3:08 PM, Shameem Shiek  wrote:

> What does "linger" do? I do not see it documented anywhere.
>
> Why would I get a SERVER_DISCONNECTED instead of CHANNEL_HANGUP ? By the
> way the BUG does not happen every time and normally I *DO* see the
> CHANNEL_HANGUP come through in most cases.
>
> Thanks for the help.
>
>
> On Thu, Sep 17, 2009 at 2:56 PM, Anthony Minessale <
> anthony.miness...@gmail.com> wrote:
>
>> try sending the linger command
>>
>> linger\n\n
>>
>>
>>
>>
>> On Thu, Sep 17, 2009 at 1:40 PM, Shameem Shiek wrote:
>>
>>> When a call is hangup, I get SERVER_DISCONNECTED Event over and over
>>> again instead of CHANNEL_HANGUP event. Has anyone else experienced this? I
>>> am using freeswitch 1.04.
>>>
>>>
>>>
>>> ___
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>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_miness...@hotmail.com 
>> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:8...@conference.freeswitch.org 
>> iax:gu...@conference.freeswitch.org/888
>> googletalk:conf+...@conference.freeswitch.org
>> pstn:213-799-1400
>>
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Re: [Freeswitch-users] fax detection

[Michael Collins]

On Tue, Sep 15, 2009 at 8:45 AM, Tihomir Culjaga  wrote:

> Hi,
>
> is there any way to route fax calls according to the call capability?
>
> I mean .. if the fax call supports T.38 i'd like to route it to a T.38
> capable gateway. All other fax calls (meaning inband) should be handled by
> FS/SpanDSP.
> Of course, I know that every fax call starts as a voice call and upon fax
> tone detection additional capabilities are being negotiated(T.38 or G711).
> Can it be done in early media, before the call is even answered?
>
> I don't claim to be an expert in all this, especially T.38, but if I
understand correctly, in both cases the call needs to be answered first. I'm
pretty sure that the sending fax machine won't start emitting the 1100Hz
tone until the receiving end answers. Also, with T.38 doesn't the call have
to come up and then T.38 gets negotiated? (I don't know, I have only read
about it.)
-MC

>
>
> So, here the goal is to have a T.38 capable GW handling T.38 calls while
> SpanDSP handling T.30...
>
>
> Any chance to do that with FS?
>
>
>
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Re: [Freeswitch-users] skill-based ACD

[Phillip Jones]

>> My FS based ACD allows agents to log-in to multiple queues at once

What is the difference between those queues? Does each q require that agents
have a different skill? ie Agent A has skills x,y,z and queue 1,2,3 have
calls about x,y,z - there agent A logins to queue 1,2,3.

2009/9/17 Paweł Pierścionek 

> Michael Collins pisze:
> >
> >
> > On Sun, Sep 13, 2009 at 8:01 AM, mark morreny  > > wrote:
> >
> > Hello
> >
> > Has any tried setting up an ACD based on skillset?  The current
> > out-of-box version of fifo does not seem to support acd based on
> > agent skillset.  Does anyone have any experience in doing it with
> > some external scripting using lua or javascript?
> >
> > I am interested in hearing how others may have done it as I am
> > trying to implement one myself.
> >
> > thx,
> >
> > mark
> >
> >
> > I was curious about this myself. Even if someone has built a non-free
> > skills-based ACD using FS I'd like to know about it.
> > -MC
> >
> What is a skills-based ACD ?
> My FS based ACD allows agents to log-in to multiple queues at once and
> have different priority setting in each of the queues - does this count
> as skills-based ?
> It does not allow to limit the number of calls an agent can answer from
> a queue daily nor does it allow to set distribution of calls to an agent
> from his queues (other the by priority).
>
> Pawel,
>
>
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Re: [Freeswitch-users] skill-based ACD

[Paweł Pierścionek]

Michael Collins pisze:
>
>
> On Sun, Sep 13, 2009 at 8:01 AM, mark morreny  > wrote:
>
> Hello
>  
> Has any tried setting up an ACD based on skillset?  The current
> out-of-box version of fifo does not seem to support acd based on
> agent skillset.  Does anyone have any experience in doing it with
> some external scripting using lua or javascript? 
>  
> I am interested in hearing how others may have done it as I am
> trying to implement one myself.
>  
> thx,
>  
> mark
>
>
> I was curious about this myself. Even if someone has built a non-free 
> skills-based ACD using FS I'd like to know about it.
> -MC
>
What is a skills-based ACD ?
My FS based ACD allows agents to log-in to multiple queues at once and 
have different priority setting in each of the queues - does this count 
as skills-based ?
It does not allow to limit the number of calls an agent can answer from 
a queue daily nor does it allow to set distribution of calls to an agent 
from his queues (other the by priority).

Pawel,


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Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max Sessions (was 502 Bad Gateway: Destination out of order error)

[email lists]

Solaris OS is at 64 bit.  Freeswitch was compiled as 32 bit however.

Our sysadmin will attempt to compile it at 64 bit.

 

In the meantime, I have moved Freeswitch over to the new server and am
in the process of testing.

 

Vladimir

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Thursday, September 17, 2009 2:58 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max
Sessions (was 502 Bad Gateway: Destination out of order error)

 

You never answered the question if you are using the 32 bit version of
the OS
If you are, make sure you do not, this has a huge impact on your
performance.



On Thu, Sep 17, 2009 at 12:48 PM, email lists
 wrote:

I'm running on a (2) Dual Core, 16G RAM machine.  Tried 5 calls per
second with 3 minute duration -- noticed same issues.

 

I throttled down to 5 calls per second, with a 30 second duration and
it seems quite a bit more stable now, though this doesn't meet our
requirements.

 

Looking at moving Freeswitch to a Quad-core 2.8ghz server now, and
will re-test.

 

Vladimir

 

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Thursday, September 17, 2009 10:44 AM


To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max
Sessions (was 502 Bad Gateway: Destination out of order error)

 

You should get a more powerful machine if you intend to sustain
upwards of 3600 calls at 20cps

We recommend a 64 bit multi-core machine running 64bit Centos5.3

On Thu, Sep 17, 2009 at 9:31 AM, email lists
 wrote:

Yes.


-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org

[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Brian West
Sent: Thursday, September 17, 2009 10:07 AM
To: freeswitch-users@lists.freeswitch.org

Subject: Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max
Sessions (was 502 Bad Gateway: Destination out of order error)

Are you using SIPP?

/b

On Sep 17, 2009, at 8:33 AM, email lists wrote:

> Now, after a few minutes of running calls at 20 calls per second
with
> a 3 min. duration, I start seeing an influx of "480 Temporarily
> Unavailable" SIP error messages.


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Re: [Freeswitch-users] mod_conference performance

[Dmitry Kadantsev]

Hi Brian,

Robert has left to Germany. He will be again online on next Monday.


--
Best regards,
Dmitry Kadantsev




On Thu, Sep 17, 2009 at 9:53 PM, Brian West  wrote:

> also was this 260 people in a single conference or multiple smaller
> conferences?
> /b
>
> On Sep 17, 2009, at 1:58 PM, RobertT wrote:
>
> Okay, I've performed some additional tests and this is what I've found:
> *
> codec**max calls
> *speex (8kHz)  50
> iLBC(8kHz) 50
> PCMU(8kHz)  260(approx*)
> GSM(8kHz)150(approx*)
> speex(16kHz) 50
> G722(16kHz)  90(approx*)
>
> * - couldn't trace till the total load of CPU 'cause RDP was timedout due
> to channel load. Calculated by trend.
>
> Still I think Linux tests are necessary.
>
> Cheers, Robert.
>
>
>
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Re: [Freeswitch-users] mod_conference performance

[Brian West]

also was this 260 people in a single conference or multiple smaller  
conferences?


/b

On Sep 17, 2009, at 1:58 PM, RobertT wrote:

Okay, I've performed some additional tests and this is what I've  
found:


codecmax calls
speex (8kHz)  50
iLBC(8kHz) 50
PCMU(8kHz)  260(approx*)
GSM(8kHz)150(approx*)
speex(16kHz) 50
G722(16kHz)  90(approx*)

* - couldn't trace till the total load of CPU 'cause RDP was  
timedout due to channel load. Calculated by trend.


Still I think Linux tests are necessary.

Cheers, Robert.


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Re: [Freeswitch-users] Event socket library bug on hangup?

[Shameem Shiek]

What does "linger" do? I do not see it documented anywhere.

Why would I get a SERVER_DISCONNECTED instead of CHANNEL_HANGUP ? By the way
the BUG does not happen every time and normally I *DO* see the
CHANNEL_HANGUP come through in most cases.

Thanks for the help.

On Thu, Sep 17, 2009 at 2:56 PM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:

> try sending the linger command
>
> linger\n\n
>
>
>
>
> On Thu, Sep 17, 2009 at 1:40 PM, Shameem Shiek wrote:
>
>> When a call is hangup, I get SERVER_DISCONNECTED Event over and over again
>> instead of CHANNEL_HANGUP event. Has anyone else experienced this? I am
>> using freeswitch 1.04.
>>
>>
>>
>> ___
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>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_miness...@hotmail.com 
> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
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>
> FreeSWITCH Developer Conference
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> iax:gu...@conference.freeswitch.org/888
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>
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Re: [Freeswitch-users] skill-based ACD

[Remko Kloosterman]

I have been working on several voice projects in the past with ACD
features, mostly based on TDM technology. It's all commercial stuff, but
I have the experience and I am willing to share that. If anyone wishes
to start such a development I'm sure I can dig up a functional model and
help with the design. 
 
Regards,
Remko



Van: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] Namens Christian
Jensen
Verzonden: donderdag 17 september 2009 20:57
Aan: freeswitch-users@lists.freeswitch.org
Onderwerp: Re: [Freeswitch-users] skill-based ACD



This would be a fantastic addition - my company is currently looking to
Asterisk as a potential candidate for this if FS can't do it.

 

I want FS to win of course :-)

 

Christian Jensen
Software Development Manager

Back Office



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: Thursday, September 17, 2009 11:20 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] skill-based ACD

 

 

On Sun, Sep 13, 2009 at 8:01 AM, mark morreny 
wrote:

Hello

 

Has any tried setting up an ACD based on skillset?  The current
out-of-box version of fifo does not seem to support acd based on agent
skillset.  Does anyone have any experience in doing it with some
external scripting using lua or javascript?  

 

I am interested in hearing how others may have done it as I am trying to
implement one myself.

 

thx,

 

mark

 


I was curious about this myself. Even if someone has built a non-free
skills-based ACD using FS I'd like to know about it.
-MC

 

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Re: [Freeswitch-users] Event socket library bug on hangup?

[Anthony Minessale]

try sending the linger command

linger\n\n




On Thu, Sep 17, 2009 at 1:40 PM, Shameem Shiek  wrote:

> When a call is hangup, I get SERVER_DISCONNECTED Event over and over again
> instead of CHANNEL_HANGUP event. Has anyone else experienced this? I am
> using freeswitch 1.04.
>
>
>
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>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_miness...@hotmail.com 
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
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sip:8...@conference.freeswitch.org 
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Re: [Freeswitch-users] skill-based ACD

[Christian Jensen]

This would be a fantastic addition - my company is currently looking to
Asterisk as a potential candidate for this if FS can't do it.

 

I want FS to win of course :-)

 

Christian Jensen
Software Development Manager

Back Office



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: Thursday, September 17, 2009 11:20 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] skill-based ACD

 

 

On Sun, Sep 13, 2009 at 8:01 AM, mark morreny 
wrote:

Hello

 

Has any tried setting up an ACD based on skillset?  The current
out-of-box version of fifo does not seem to support acd based on agent
skillset.  Does anyone have any experience in doing it with some
external scripting using lua or javascript?  

 

I am interested in hearing how others may have done it as I am trying to
implement one myself.

 

thx,

 

mark

 


I was curious about this myself. Even if someone has built a non-free
skills-based ACD using FS I'd like to know about it.
-MC

 

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Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max Sessions (was 502 Bad Gateway: Destination out of order error)

[Anthony Minessale]

You never answered the question if you are using the 32 bit version of the
OS
If you are, make sure you do not, this has a huge impact on your
performance.


On Thu, Sep 17, 2009 at 12:48 PM, email lists <
email.list.subscri...@gmail.com> wrote:

>  I'm running on a (2) Dual Core, 16G RAM machine.  Tried 5 calls per
> second with 3 minute duration -- noticed same issues.
>
>
>
> I throttled down to 5 calls per second, with a 30 second duration and it
> seems quite a bit more stable now, though this doesn’t meet our
> requirements.
>
>
>
> Looking at moving Freeswitch to a Quad-core 2.8ghz server now, and will
> re-test.
>
>
>
> Vladimir
>
>
>
>
>
> *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
> freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Thursday, September 17, 2009 10:44 AM
>
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max
> Sessions (was 502 Bad Gateway: Destination out of order error)
>
>
>
> *You should get a more powerful machine if you intend to sustain upwards
> of 3600 calls at 20cps*
>
> We recommend a 64 bit multi-core machine running 64bit Centos5.3
>
>  On Thu, Sep 17, 2009 at 9:31 AM, email lists <
> email.list.subscri...@gmail.com> wrote:
>
> Yes.
>
>
> -Original Message-
> From: freeswitch-users-boun...@lists.freeswitch.org
>
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
> Brian West
> Sent: Thursday, September 17, 2009 10:07 AM
> To: freeswitch-users@lists.freeswitch.org
>
> Subject: Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max
> Sessions (was 502 Bad Gateway: Destination out of order error)
>
> Are you using SIPP?
>
> /b
>
> On Sep 17, 2009, at 8:33 AM, email lists wrote:
>
> > Now, after a few minutes of running calls at 20 calls per second
> with
> > a 3 min. duration, I start seeing an influx of "480 Temporarily
> > Unavailable" SIP error messages.
>
>
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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
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>
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> sip:8...@conference.freeswitch.org 
> iax:gu...@conference.freeswitch.org/888
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> pstn:213-799-1400
>
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>
>


-- 
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Re: [Freeswitch-users] mod_conference performance

[RobertT]

Okay, I've performed some additional tests and this is what I've found:
*
codec**max calls
*speex (8kHz)  50
iLBC(8kHz) 50
PCMU(8kHz)  260(approx*)
GSM(8kHz)150(approx*)
speex(16kHz) 50
G722(16kHz)  90(approx*)

* - couldn't trace till the total load of CPU 'cause RDP was timedout due to
channel load. Calculated by trend.

Still I think Linux tests are necessary.

Cheers, Robert.
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[Freeswitch-users] Event socket library bug on hangup?

[Shameem Shiek]

When a call is hangup, I get SERVER_DISCONNECTED Event over and over again
instead of CHANNEL_HANGUP event. Has anyone else experienced this? I am
using freeswitch 1.04.
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Re: [Freeswitch-users] session record does not for very short calls

[Michael Collins]

On Thu, Sep 17, 2009 at 6:21 AM, Brian West  wrote:

> I think if you remove data="RECORD_ANSWER_REQ=true"/>
>
> it'll do what you want.
>
> FYI
http://wiki.freeswitch.org/wiki/Channel_Variables#RECORD_ANSWER_REQ

Hopefully that will help fill in the details.
-MC
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Re: [Freeswitch-users] skill-based ACD

[Michael Collins]

On Sun, Sep 13, 2009 at 8:01 AM, mark morreny  wrote:

> Hello
>
> Has any tried setting up an ACD based on skillset?  The current out-of-box
> version of fifo does not seem to support acd based on agent skillset.  Does
> anyone have any experience in doing it with some external scripting using
> lua or javascript?
>
> I am interested in hearing how others may have done it as I am trying to
> implement one myself.
>
> thx,
>
> mark
>
>
I was curious about this myself. Even if someone has built a non-free
skills-based ACD using FS I'd like to know about it.
-MC
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Re: [Freeswitch-users] How to process s extension in FS

[Michael Collins]

On Thu, Sep 17, 2009 at 4:25 AM, Ahmed Munir wrote:

>
> Hi,
>
> How can I process s extension in FS? Is there other way around of doing it?
> Kindly advice me.
>
> Like Brian said, we need a little more information. Where is the call
coming from? In some cases you will handle the incoming call in the public
context. See conf/dialplan/public.xml and
conf/dialplan/public/00_inbound_did.xml for an example of how to handle an
incoming DID from a SIP provider.
-MC
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Re: [Freeswitch-users] FreeSWITCH and OpenBTS integration

[Michael Collins]

On Thu, Sep 17, 2009 at 8:39 AM, Alberto Escudero  wrote:

>
> Done!
> http://wiki.freeswitch.org/wiki/OpenBTS
>
>
AEP,

Thanks for the update! :)
-MC
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Re: [Freeswitch-users] DB table sip_dialogs is always empty

[Michael Collins]

Could you pastebin a minimal config that someone could put on their own
system in order to try to simulate what you're doing? It might help if
someone can try your setup on a different machine to see what happens.
-MC

On Thu, Sep 17, 2009 at 4:08 AM, kokoska.rokoska wrote:

> I'm sorry to resend this post, but even after few hours I can't see it
> in the mailing-list...
> Thanks.
>
>
> Best regards,
>
> kokoska.rokoska
>
>
> kokoska rokoska napsal(a):
> > Hello,
> >
> > I have setted-up odbc-dsn on all my FreeSWITCH sofia profiles and based
> > on logs FS connected to the dsns correctly. But during the calls I can't
> > see any rows in sip_dialogs table. When I run "show channels" from
> > console it works fine.
> > All other tables are "populated and maintained" as I expect :-)
> >
> > I'm running FreeSWITCH on 64bit Centos 5.3 with Postgresql 8.1 DB...
> >
> > Could you, please, point what I'm missing, or where I'm wrong? Thank you
> > very much!
> >
> >
> > Best regards,
> >
> > kokoska.rokoska
> >
>
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Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max Sessions (was 502 Bad Gateway: Destination out of order error)

[email lists]

I'm running on a (2) Dual Core, 16G RAM machine.  Tried 5 calls per
second with 3 minute duration -- noticed same issues.

 

I throttled down to 5 calls per second, with a 30 second duration and
it seems quite a bit more stable now, though this doesn't meet our
requirements.

 

Looking at moving Freeswitch to a Quad-core 2.8ghz server now, and
will re-test.

 

Vladimir

 

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Thursday, September 17, 2009 10:44 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max
Sessions (was 502 Bad Gateway: Destination out of order error)

 

You should get a more powerful machine if you intend to sustain
upwards of 3600 calls at 20cps

We recommend a 64 bit multi-core machine running 64bit Centos5.3



On Thu, Sep 17, 2009 at 9:31 AM, email lists
 wrote:

Yes.


-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org

[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Brian West
Sent: Thursday, September 17, 2009 10:07 AM
To: freeswitch-users@lists.freeswitch.org

Subject: Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max
Sessions (was 502 Bad Gateway: Destination out of order error)

Are you using SIPP?

/b

On Sep 17, 2009, at 8:33 AM, email lists wrote:

> Now, after a few minutes of running calls at 20 calls per second
with
> a 3 min. duration, I start seeing an influx of "480 Temporarily
> Unavailable" SIP error messages.


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Re: [Freeswitch-users] DB table sip_dialogs is always empty

[Anthony Minessale]

that table is specific to the manage-presence option


On Thu, Sep 17, 2009 at 6:08 AM, kokoska.rokoska wrote:

> I'm sorry to resend this post, but even after few hours I can't see it
> in the mailing-list...
> Thanks.
>
>
> Best regards,
>
> kokoska.rokoska
>
>
> kokoska rokoska napsal(a):
> > Hello,
> >
> > I have setted-up odbc-dsn on all my FreeSWITCH sofia profiles and based
> > on logs FS connected to the dsns correctly. But during the calls I can't
> > see any rows in sip_dialogs table. When I run "show channels" from
> > console it works fine.
> > All other tables are "populated and maintained" as I expect :-)
> >
> > I'm running FreeSWITCH on 64bit Centos 5.3 with Postgresql 8.1 DB...
> >
> > Could you, please, point what I'm missing, or where I'm wrong? Thank you
> > very much!
> >
> >
> > Best regards,
> >
> > kokoska.rokoska
> >
>
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Re: [Freeswitch-users] Delay when transferring call

[Michael Collins]

Turn on debug (press F8) level logging and capture the output, put in
pastebin.freeswitch.org. Hopefully the debug output will shed some light on
where the delay is occurring.
Also, see this page for some tips on how to collect information for
debugging purposes:
http://wiki.freeswitch.org/wiki/Reporting_Bugs

It will give you handy tips on collecting information, posting to pastebin,
asking community for help, etc. etc. In short, it will make your life
easier. :)
-MC


On Thu, Sep 17, 2009 at 3:23 AM, Sias Mey  wrote:

> Hi,
>
> Im having a strange issue with a api triggered call transfer.
>
> There seems to be a long delay between when the transfer is triggered and
> when it actually happens.
>
> 2009-09-17 11:36:26.995001 [NOTICE] switch_ivr.c:1350 Transfer
> sofia/internal/1...@192.168.0.10 to xml
> [incust-camp=lucidlive-call=78-conf...@default]
> Error in my_thread_global_end(): 26 threads didn't exit
> 2009-09-17 11:36:31.997191 [INFO] mod_dialplan_xml.c:315 Processing
> 1004->incust-camp=lucidlive-call=7
> 8-conf=41 in context default
> 2009-09-17 11:36:31.997191 [INFO] regin.js:1 Execution start
> 2009-09-17 11:36:31.997191 [INFO] regin.js:1 Connecting to Ringback to add
> call
> 2009-09-17 11:36:31.997191 [INFO] regin.js:1 Finished adding calls
> 2009-09-17 11:36:31.997191 [INFO] regin.js:1 Connecting to database
> lucidlive to update call
> 2009-09-17 11:36:32.37433 [INFO] regin.js:1 Finished updateing call
>
> I though it was my regin.js script causing the delay since it runs a couple
> of database queries and other things, but the output above show that runs
> fine.
>
> My question is about the delay between 11:36:26 -> 11:36:31. The call is
> being transfered out of a fifo, but for those 5 seconds there is no MOH or
> anything else. Just silence.
>
> The transfer is triggered via a xml rpc call. But since the delay is
> between the switch_ivr and mod_dialplan_xml somewhere I doubt that that has
> much to do with it.
>
> Any clues or other places I can go look?.
>
> Cheers,
> Sias
>
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Re: [Freeswitch-users] faxrx error 13 Unexpected message received

[Travis Stutsman]

Thanks Steve.

More tests today. Consistent results.

The Fax machine is in the same building as the FS server.  It goes out
over the PSTN to my carrier, then carrier to me over IP to the server.

I need to try more fax machines and more locations.

-- Travis

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Re: [Freeswitch-users] FreeSWITCH and OpenBTS integration

[Alberto Escudero]

Done!
http://wiki.freeswitch.org/wiki/OpenBTS

-- 
Stopping junk mailers is good for the environment

> Just create an OpenBTS page on our wiki.
>
> /b
>
> On Sep 17, 2009, at 7:46 AM, Alberto Escudero wrote:
>
>> Sorry, just realized that the sourceforge page is protected by
>> password. I
>> am happy to put the info in FreeSWITCH wiki, where does it  make
>> sense to
>> add this project info?
>
>
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>



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Re: [Freeswitch-users] FreeSWITCH and OpenBTS integration

[Alberto Escudero]

The real challenge at the moment is to find adequate regulatory scenarios
to run this technology. In many parts of the world where we works
operators/governments have literally locked the spectrum so others can not
run anything on it.

Community networks are left with garbage bands to operate 802.11 devices.
We welcome any scenarios where to operate an open gsm infrastructure
connected to IP.

/aep




-- 
Stopping junk mailers is good for the environment

> Am Thursday 17 September 2009 schrieb Steve Underwood:
>> On 09/17/2009 09:19 PM, Brian West wrote:
>> > Just create an OpenBTS page on our wiki.
>> >
>> > /b
>> >
>> > On Sep 17, 2009, at 7:46 AM, Alberto Escudero wrote:
>> >
>> >
>> >> Sorry, just realized that the sourceforge page is protected by
>> >> password. I
>> >> am happy to put the info in FreeSWITCH wiki, where does it  make
>> >> sense to
>> >> add this project info?
>> >>
>> >
>> Isn't there still some legal wrangling over openBTS?
>>
>> Steve
>
>
> nope, that was resolved:
>
> http://openbts.blogspot.com/2009/07/three-quotes.html
>
>
> --
> ---
> Stefan Knoblich| Web:   http://www.axsentis.de/
> axsentis GmbH  |http://oss.axsentis.de/
> Eupener Str. 74, 50933 Koeln, Germany  |
> Amtsgericht Koeln: HR B 56238  | Email: s.knobl...@axsentis.de
> UST-ID: DE244977565| JID:
> s.knobl...@jabber.axsentis.de
> ---
> Web:   http://stkn.techmage.de/
> Email: s...@freeswitch.org
> IRC:   #freeswitch-de @ irc.freenode.net
>
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Re: [Freeswitch-users] FreeSWITCH and OpenBTS integration

[Stefan Knoblich]

Am Thursday 17 September 2009 schrieb Steve Underwood:
> On 09/17/2009 09:19 PM, Brian West wrote:
> > Just create an OpenBTS page on our wiki.
> >
> > /b
> >
> > On Sep 17, 2009, at 7:46 AM, Alberto Escudero wrote:
> >
> >
> >> Sorry, just realized that the sourceforge page is protected by
> >> password. I
> >> am happy to put the info in FreeSWITCH wiki, where does it  make
> >> sense to
> >> add this project info?
> >>  
> >
> Isn't there still some legal wrangling over openBTS?
> 
> Steve


nope, that was resolved:

http://openbts.blogspot.com/2009/07/three-quotes.html


-- 
---
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axsentis GmbH  |http://oss.axsentis.de/
Eupener Str. 74, 50933 Koeln, Germany  |
Amtsgericht Koeln: HR B 56238  | Email: s.knobl...@axsentis.de
UST-ID: DE244977565| JID:   s.knobl...@jabber.axsentis.de
---
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Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max Sessions (was 502 Bad Gateway: Destination out of order error)

[Anthony Minessale]

You should get a more powerful machine if you intend to sustain upwards of
3600 calls at 20cps

We recommend a 64 bit multi-core machine running 64bit Centos5.3


On Thu, Sep 17, 2009 at 9:31 AM, email lists <
email.list.subscri...@gmail.com> wrote:

> Yes.
>
> -Original Message-
> From: freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
> Brian West
> Sent: Thursday, September 17, 2009 10:07 AM
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max
> Sessions (was 502 Bad Gateway: Destination out of order error)
>
> Are you using SIPP?
>
> /b
>
> On Sep 17, 2009, at 8:33 AM, email lists wrote:
>
> > Now, after a few minutes of running calls at 20 calls per second
> with
> > a 3 min. duration, I start seeing an influx of "480 Temporarily
> > Unavailable" SIP error messages.
>
>
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-- 
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Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max Sessions (was 502 Bad Gateway: Destination out of order error)

[email lists]

Yes.

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Brian West
Sent: Thursday, September 17, 2009 10:07 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max
Sessions (was 502 Bad Gateway: Destination out of order error)

Are you using SIPP?

/b

On Sep 17, 2009, at 8:33 AM, email lists wrote:

> Now, after a few minutes of running calls at 20 calls per second
with
> a 3 min. duration, I start seeing an influx of "480 Temporarily
> Unavailable" SIP error messages.


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Re: [Freeswitch-users] FreeSWITCH and OpenBTS integration

[Steve Underwood]

On 09/17/2009 09:19 PM, Brian West wrote:
> Just create an OpenBTS page on our wiki.
>
> /b
>
> On Sep 17, 2009, at 7:46 AM, Alberto Escudero wrote:
>
>
>> Sorry, just realized that the sourceforge page is protected by
>> password. I
>> am happy to put the info in FreeSWITCH wiki, where does it  make
>> sense to
>> add this project info?
>>  
>
Isn't there still some legal wrangling over openBTS?

Steve


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Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max Sessions (was 502 Bad Gateway: Destination out of order error)

[Brian West]

Are you using SIPP?

/b

On Sep 17, 2009, at 8:33 AM, email lists wrote:

> Now, after a few minutes of running calls at 20 calls per second with
> a 3 min. duration, I start seeing an influx of "480 Temporarily
> Unavailable" SIP error messages.


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Re: [Freeswitch-users] Limit_Hash

[Timur Irmatov]

Hi,

On Tue, Sep 15, 2009 at 4:10 PM, Matt Riddell  wrote:
> My question then becomes, how do I set a hash for an originated call?

You set a limit by executing limit_hash application at appropriate
place in your dial plan.

> It seems that limit_hash is an application rather than a channel
> variable, and so far I've been doing most things without touching the
> dialplan.

Correct, limit_hash is an application.

> So, say I want to originate 9 calls, 3 from 3 customers.
>
> I would like to mark the calls with my_customer_group_1 through 3, and
> then use the limit_hash_usage command to verify the count of channels in
> each group.
>
> I therefore have a few questions:
>
> 1. Can I mark a call in the originate statement?

Somehow you need an ability to distinguish one customer from another.
For example, if each customer has its own number and originates calls
only from this number, you can use this number for limiting their
usage.

> 2. How do I use the limit_hash_usage command?
> The wiki states:
> You can verify the usage of any resource with the limit_hash_usage api call.
> limit_hash_usage  
> Is realm the same as a SIP realm?
> Is id the hash that I have used to mark the call with?

No,  is just an arbitrary string to allow you have same s
in different realms. For example, realm can be a name of your
customer, and  can be a direction of call, so you can set a limit
of 5 outbound and 3 inbound calls per customer:







Of course, those two lines should be placed to appropriate (and most
probably separate) places of your dialplan.

So, back to your example. Three customers, 3 outbound calls per each.
Let's suppose customer 1, 2 and 3 have numbers 1001, 1002, 1003
assigned to them, and customer1 originates calls from 1001, and so
on.. You place the limit like this:


  


  


You should see messages in console about limits during live calls.
Also, bear in mind that if you place a limit in one dialplan context
and then transfer your call to another, limit will be reset. Does
anyone know, why it is like this? Is it possible to allow limits
persist across dialplan contexts?

-- 
Timur Irmatov, xmpp:irma...@jabber.ru

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Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max Sessions (was 502 Bad Gateway: Destination out of order error)

[email lists]

Anthony,

I increased the ulimit parameters as suggested.  I then ran into a few
other Freeswitch errors that required me to edit switch.conf.xml to
increase the max sessions, and max sessions per second parameters (to
10, and 100, respectively).

Now, after a few minutes of running calls at 20 calls per second with
a 3 min. duration, I start seeing an influx of "480 Temporarily
Unavailable" SIP error messages.

Here is a snip of the log:


---
2009-09-16 19:35:19.18798 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/2015692...@72.245.84.150 [BREAK]
2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:434
(sofia/internal/9726074...@72.245.84.24:5060) State HANGUP
2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:398
(sofia/external/2015692...@72.245.84.150) Running State Change
CS_REPORTING
2009-09-16 19:35:19.18798 [DEBUG] mod_sofia.c:338 Channel
sofia/internal/9726074...@72.245.84.24:5060 hanging up, cause:
NORMAL_CLEARING
2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:612
(sofia/external/2015692...@72.245.84.150) State REPORTING
2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:53
sofia/external/2015692...@72.245.84.150 Standard REPORTING, cause:
NO_ANSWER
2009-09-16 19:35:19.18798 [DEBUG] mod_sofia.c:417 Responding to INVITE
with: 480
2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:612
(sofia/external/2015692...@72.245.84.150) State REPORTING going to
sleep
2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:411
(sofia/external/2015692...@72.245.84.150) State Change CS_REPORTING ->
CS_DESTROY
2009-09-16 19:35:19.18798 [DEBUG] switch_core_session.c:1068 Session
3395 (sofia/external/2015692...@72.245.84.150) Locked, Waiting on
external entities
2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:46
sofia/internal/9726074...@72.245.84.24:5060 Standard HANGUP, cause:
NORMAL_CLEARING
2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:434
(sofia/internal/9726074...@72.245.84.24:5060) State HANGUP going to
sleep
2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:434
(sofia/external/2015692...@72.245.84.150) State HANGUP
2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:46
sofia/external/2015692...@72.245.84.150 Standard HANGUP, cause:
NO_ANSWER
2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:434
(sofia/external/2015692...@72.245.84.150) State HANGUP going to sleep
2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:476
(sofia/internal/9726074...@72.245.84.24:5060) State Change CS_HANGUP
-> CS_REPORTING
2009-09-16 19:35:19.18798 [DEBUG] switch_core_session.c:932 Send
signal sofia/internal/9726074...@72.245.84.24:5060 [BREAK]
2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/9726074...@72.245.84.24:5060) Running State Change
CS_REPORTING
2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:612
(sofia/internal/9726074...@72.245.84.24:5060) State REPORTING
2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:53
sofia/internal/9726074...@72.245.84.24:5060 Standard REPORTING, cause:
NORMAL_CLEARING
2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:612
(sofia/internal/9726074...@72.245.84.24:5060) State REPORTING going to
sleep
2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:411
(sofia/internal/9726074...@72.245.84.24:5060) State Change
CS_REPORTING -> CS_DESTROY
2009-09-16 19:35:19.18798 [DEBUG] switch_core_session.c:1068 Session
3392 (sofia/internal/9726074...@72.245.84.24:5060) Locked, Waiting on
external entities
2009-09-16 19:35:19.18798 [NOTICE] switch_core_session.c:1086 Session
3392 (sofia/internal/9726074...@72.245.84.24:5060) Ended
2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:434
(sofia/external/2015692...@72.245.84.150) State HANGUP
2009-09-16 19:35:19.18798 [NOTICE] switch_core_session.c:1088 Close
Channel sofia/internal/9726074...@72.245.84.24:5060 [CS_DESTROY]
2009-09-16 19:35:19.18798 [DEBUG] mod_sofia.c:338 Channel
sofia/external/2015692...@72.245.84.150 hanging up, cause: NO_ANSWER
2009-09-16 19:35:19.18798 [DEBUG] mod_sofia.c:406 Sending CANCEL to
sofia/external/2015692...@72.245.84.150
2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:46
sofia/external/2015692...@72.245.84.150 Standard HANGUP, cause:
NO_ANSWER
2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:434
(sofia/external/2015692...@72.245.84.150) State HANGUP going to sleep
2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:564
(sofia/internal/9726074...@72.245.84.24:5060) State DESTROY
2009-09-16 19:35:19.18798 [DEBUG] mod_sofia.c:255
sofia/internal/9726074...@72.245.84.24:5060 SOFIA DESTROY
2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:60
sofia/internal/9726074...@72.245.84.24:5060 Standard DESTROY
2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:564
(sofia/internal/9726074...@72.245.84.24:

Re: [Freeswitch-users] FreeSWITCH and OpenBTS integration

[Brian West]

Just create an OpenBTS page on our wiki.

/b

On Sep 17, 2009, at 7:46 AM, Alberto Escudero wrote:

> Sorry, just realized that the sourceforge page is protected by  
> password. I
> am happy to put the info in FreeSWITCH wiki, where does it  make  
> sense to
> add this project info?


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Re: [Freeswitch-users] Simple call waiting question

[Brian West]

Personally I would throw the phone in the trash.  :P

In the default dialplan look at 5900 for park and 5901 for unpark.

/b


On Sep 17, 2009, at 7:58 AM, Mark Campbell-Smith wrote:

> I am trying to create a simple call waiting dialplan as my phone does
> not have Recall button.


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Re: [Freeswitch-users] session record does not for very short calls

[Brian West]

I think if you remove

it'll do what you want.

/b

On Sep 16, 2009, at 7:05 PM, Nandy Dagondon wrote:

> this makes sense. a workaround would be to provide an optional  
> variable to delete recording file if it's less than N seconds.  
> otherwise, it defaults to a preset duration.
>
> /nandy
>
>
> On Thu, Sep 17, 2009 at 7:46 AM, Seven Du  wrote:
> I think the file was there but deleted by FreeSWITCH if it thinks it
> was too short (like 3 seconds?). If I'm not wrong, someone requested
> this feature becuase FreeSWITCH left too many small recordings.
>
>
> On Sep 17, 2009, at 1:27 AM, João Mesquita wrote:
> > I think you need to upgrade your version before we even take a look
> > at that... You are so far behind trunk right now and lots of things
> > have been changed since then.
> >
> > Not sure if this would solve your problem but not a lot of ppl will
> > look at your problem when you run this version.
> >
> > jmesquita
> >
> > On Wed, Sep 16, 2009 at 2:00 PM, Frank @ Impact
> >  wrote:
> > FreeSWITCH Version 1.0.trunk (12790M)
> >
> >
> > I have this in my DP
> >
> >   
> >
> >   
> >
> >   
> >
> >
> > works fine as long as the call is long enough.  But if the call is
> > only, say, 3-4 seconds long (or something very short like that),
> > then the wav file is never created with the audio in it.
> >
> >
> > Is there a work around for this?
> >
> >
> > ___
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users@lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> > http://www.freeswitch.org
> >
> >
> > ___
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> > FreeSWITCH-users@lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> > http://www.freeswitch.org
>
>
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Re: [Freeswitch-users] How to process s extension in FS

[Brian West]

Since we have no 's' extension or anything similar maybe if you tell  
me what you're trying to do I can tell you how to do it.a

/b

On Sep 17, 2009, at 6:25 AM, Ahmed Munir wrote:

>
> Hi,
>
> How can I process s extension in FS? Is there other way around of  
> doing it? Kindly advice me.
>
> -- 
> Regards,
>
> Ahmed Munir


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Re: [Freeswitch-users] Compile error

[Brian West]

NO you must not.  The issue has been fixed in svn already please start  
with a fresh tree.


/b
PS: end users should NEVER have to reswig.

On Sep 17, 2009, at 12:42 AM, Frank Carmickle wrote:


On Thu, Sep 17, Luis M. Zuccolo wrote:

Hi:

Since svn version 13523 to current I get this error:

make[5]: swig: Command not found


You must install swig.  If your on debian apt-get install swig.  If  
your not see http://www.swig.org/


HTH
--FC


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Re: [Freeswitch-users] How can I configure TO and FROM in the invite message

[Brian West]

You can do that in your dialplan.  Not sure it'll fix all your options  
unless you do both the gateway and the sip_invite_domain.


/b

On Sep 17, 2009, at 4:19 AM, Tzury Bar Yochay wrote:

> in which file under which section should I specify this
>
>  data="{sip_invite_domain=${sip_from_host}}sofia/gateway/ 
> gw1/$...@domain.org"/>


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Re: [Freeswitch-users] A real puzzler for you guys... (perpetual registration causes denial of service.)

[Brian West]

Its a bug in 2.6.26 thru 2.6.28 kernels that impact the performance of  
SQLite.  He was specifically running SUSE.

/b

On Sep 16, 2009, at 11:07 PM, Jason White wrote:

> Please take this up with your Linux distribution as a bug report  
> related to
> the kernel, and persist with it until it's sorted out.
>
> The more that users do this, the more kernel bugs will get fixed.
>
> We're all responsible to some extent for the quality of our free/ 
> open-source
> operating systems.


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Re: [Freeswitch-users] DB table sip_dialogs always empty

[Brian West]

These are not sip_dialogs as you think they are.  These are used for  
dialing-info dialogs for sip subscriptions.

/b

On Sep 17, 2009, at 2:51 AM, kokoska rokoska wrote:

>
> Hello,
>
> I have setted-up odbc-dsn on all my FreeSWITCH sofia profiles and  
> based
> on logs FS connected to the dsns correctly. But during the calls I  
> can't
> see any rows in sip_dialogs table. When I run "show channels" from
> console it works fine.
> All other tables are "populated and maintained" as I expect :-)
>
> I'm running FreeSWITCH on 64bit Centos 5.3 with Postgresql 8.1 DB...
>
> Could you, please, point what I'm missing, or where I'm wrong? Thank  
> you
> very much!
>
>
> Best regards,
>
> kokoska.rokoska


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[Freeswitch-users] Simple call waiting question

[Mark Campbell-Smith]

HI All,

I am trying to create a simple call waiting dialplan as my phone does
not have Recall button.

The simple scenario is:
1. B calls user A and is answered
2. C calls user A
3. A puts B on hold
4. A answers C
4. A then recalls first call from B

I was going to use fifo for step 3.  I am doing:

and then
 
  
  
  

My question is then, how do I get A to answer the call from C?  Step 4
and then step 5?

On another note, if I put  I get the following in the console:
Transfer sofia/internal_nat/sip:1...@xx.xx.xx.xx:5060 to 5900[-a...@xml]

Shouldn't that be xml[5...@default] ?

Thanks!

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Re: [Freeswitch-users] FreeSWITCH and OpenBTS integration

[Alberto Escudero]

Sorry, just realized that the sourceforge page is protected by password. I
am happy to put the info in FreeSWITCH wiki, where does it  make sense to
add this project info?

-aep
-- 
Stopping junk mailers is good for the environment

> I am happy to let you know that FreeSWITCH route calls from OpenBTS, the
> open base station based on the Universal Software Radio USRP. Yes! Calls
> from a standard handset to a GSM base station connected to FreeSWITCH
>
> If you want to read more about the idea check:
> http://openbts.sourceforge.net/
> http://www.it46.se/entry/380 (our effort to deploy the technology in a
> developing region)
>
> I have put a few notes for others to give it a try available here:
> https://sourceforge.net/apps/trac/openbts/wiki/OpenBTS/SettingUpFreeSWITCH
>
> Let me know what is the best place in FreeSWITCH wiki to add and keep
> updated this information
>
> --
> Stopping junk mailers is good for the environment
>
>
>
>
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[Freeswitch-users] FreeSWITCH and OpenBTS integration

[Alberto Escudero]

I am happy to let you know that FreeSWITCH route calls from OpenBTS, the
open base station based on the Universal Software Radio USRP. Yes! Calls
from a standard handset to a GSM base station connected to FreeSWITCH

If you want to read more about the idea check:
http://openbts.sourceforge.net/
http://www.it46.se/entry/380 (our effort to deploy the technology in a
developing region)

I have put a few notes for others to give it a try available here:
https://sourceforge.net/apps/trac/openbts/wiki/OpenBTS/SettingUpFreeSWITCH

Let me know what is the best place in FreeSWITCH wiki to add and keep
updated this information

-- 
Stopping junk mailers is good for the environment




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[Freeswitch-users] How to process s extension in FS

[Ahmed Munir]

Hi,

How can I process s extension in FS? Is there other way around of doing it?
Kindly advice me.

-- 
Regards,

Ahmed Munir
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[Freeswitch-users] DB table sip_dialogs is always empty

[kokoska.rokoska]

I'm sorry to resend this post, but even after few hours I can't see it
in the mailing-list...
Thanks.


Best regards,

kokoska.rokoska


kokoska rokoska napsal(a):
> Hello,
>
> I have setted-up odbc-dsn on all my FreeSWITCH sofia profiles and based
> on logs FS connected to the dsns correctly. But during the calls I can't
> see any rows in sip_dialogs table. When I run "show channels" from
> console it works fine.
> All other tables are "populated and maintained" as I expect :-)
>
> I'm running FreeSWITCH on 64bit Centos 5.3 with Postgresql 8.1 DB...
>
> Could you, please, point what I'm missing, or where I'm wrong? Thank you
> very much!
>
>
> Best regards,
>
> kokoska.rokoska
>

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[Freeswitch-users] Delay when transferring call

[Sias Mey]

Hi, 

Im having a strange issue with a api triggered call transfer.

There seems to be a long delay between when the transfer is triggered and when 
it actually happens.

2009-09-17 11:36:26.995001 [NOTICE] switch_ivr.c:1350 Transfer 
sofia/internal/1...@192.168.0.10 to xml
[incust-camp=lucidlive-call=78-conf...@default]
Error in my_thread_global_end(): 26 threads didn't exit
2009-09-17 11:36:31.997191 [INFO] mod_dialplan_xml.c:315 Processing 
1004->incust-camp=lucidlive-call=7
8-conf=41 in context default
2009-09-17 11:36:31.997191 [INFO] regin.js:1 Execution start
2009-09-17 11:36:31.997191 [INFO] regin.js:1 Connecting to Ringback to add call
2009-09-17 11:36:31.997191 [INFO] regin.js:1 Finished adding calls
2009-09-17 11:36:31.997191 [INFO] regin.js:1 Connecting to database lucidlive 
to update call
2009-09-17 11:36:32.37433 [INFO] regin.js:1 Finished updateing call

I though it was my regin.js script causing the delay since it runs a couple of 
database queries and other things, but the output above show that runs fine.

My question is about the delay between 11:36:26 -> 11:36:31. The call is being 
transfered out of a fifo, but for those 5 seconds there is no MOH or anything 
else. Just silence.

The transfer is triggered via a xml rpc call. But since the delay is between 
the switch_ivr and mod_dialplan_xml somewhere I doubt that that has much to do 
with it.

Any clues or other places I can go look?. 

Cheers,
Sias

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Re: [Freeswitch-users] How can I configure TO and FROM in the invite message

[Tzury Bar Yochay]

in which file under which section should I specify this



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Re: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects

[Giovanni Maruzzelli]

I'm gmaruzz on IRC, for GSM, Skype, Italian language, audio stuff, etc...

-giovanni


On Thu, Sep 17, 2009 at 10:21 AM, Tristan Mahé  wrote:
> Hi Michael,
>
> I'm gled on IRC, always connected so ping me if you wanna talk ;)
>
> Michael Collins a écrit :
>
> On Wed, Sep 16, 2009 at 1:53 AM, Tristan Mahé  wrote:
>>
>> Hi,
>>
>> Count on me for answering questions on IRC when I'm in, and for
>> subprojects I'm in too as you know ;)
>
> Merci!
>
> Okay, what's your IRC nick and when are you generally on line? Also, I'm
> pretty sure that you're fluent in French which is good because we need more
> multilingual people out there. Last question: what are your areas of
> expertise? I'd like to keep a list of people and what they're good at so we
> know whom to ask first when questions come up.
>
> Thanks again!
> -MC
>
> 
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>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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Re: [Freeswitch-users] switch_core_io.c:118 sofia/internal/XXXXX has no read codec.

[Yehavi Bourvine]

I've solved the problem: I am running it on a Fedora-10 system. Once I've
installed a vanilla kernel (from kernel.org) the problem went away.

BTW, can someone shed the light on the kernel's bug which I see mentions
of it in this list?

 Thanks! __Yehavi:
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Re: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects

[Tristan Mahé]

Hi Michael,

I'm gled on IRC, always connected so ping me if you wanna talk ;)

Michael Collins a écrit :
>
>
> On Wed, Sep 16, 2009 at 1:53 AM, Tristan Mahé  > wrote:
>
> Hi,
>
> Count on me for answering questions on IRC when I'm in, and for
> subprojects I'm in too as you know ;)
>
> Merci!
>
> Okay, what's your IRC nick and when are you generally on line? Also,
> I'm pretty sure that you're fluent in French which is good because we
> need more multilingual people out there. Last question: what are your
> areas of expertise? I'd like to keep a list of people and what they're
> good at so we know whom to ask first when questions come up.
>
> Thanks again!
> -MC
>
> 
>
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[Freeswitch-users] DB table sip_dialogs always empty

[kokoska rokoska]

Hello,

I have setted-up odbc-dsn on all my FreeSWITCH sofia profiles and based
on logs FS connected to the dsns correctly. But during the calls I can't
see any rows in sip_dialogs table. When I run "show channels" from
console it works fine.
All other tables are "populated and maintained" as I expect :-)

I'm running FreeSWITCH on 64bit Centos 5.3 with Postgresql 8.1 DB...

Could you, please, point what I'm missing, or where I'm wrong? Thank you
very much!


Best regards,

kokoska.rokoska


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