from:"Nandy Dagondon"

Re: [Freeswitch-users] continue_on_fail

2009-12-08 Thread Nandy Dagondon

this action can be accomplished using Group Dialing (Sequential). this may
not answer your problem but have you considered it?
-nandy


On Tue, Dec 8, 2009 at 2:31 AM, Peter P GMX  wrote:

> I have a Problem with continue_on_fail.
>
>
> I have setup a hunt group
> 
>  data="sofia/external/2...@10.11.12.243,sofia/external/2...@10.11.12.234
> ,sofia/external/2...@10.11.12.188,sofia/external/1...@10.11.12.245"/>
> 
> I want the fallback user to be called whenever none of the previously
> called 3 gateway numbers picks up or if they are all busy.
> Therefore continue_on_fail=NO_ANSWER,USER_BUSY
>
> The fallback user is called, however if any of the previously called
> gateways picks up and then hangs up, the fallback user is called
> afterwards.
> Means: The fallback user is always called.
>
> I had expected, that continue_on_fail=NO_ANSWER,USER_BUSY would not fire
> the next bridge if it gets a NORMAL_CLEARING.
>
> Am I thinking wrongly about this?
>
> I have added
>
> and this works, but I would like to specify more in detail the
> conditions when to follow the next hunt group entry.
>
> Best regards
> Peter
>
>
>
>
>
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Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through

2009-12-08 Thread Nandy Dagondon

have you created Extension 1002?
-nandy


On Wed, Dec 9, 2009 at 3:20 AM, mailinglist  wrote:

>  Hi All
>
> Ok, after reading a bit more I think I see what I've done wrong, but I
> don't know how to fix it properly.
> Looking in the Dialplan directory I see the following:
> default (dir)
> default.xml
> features.xml
> public (dir)
> public.xml
>
> Under the default dir the webinterface has created the 001_musimi.dk.xml
> file that I've created.
> But as I understand it, it doesn't use it.
>
> How do I make it use it, I would very much like to keep the webinterface
> editor, and not have to do it via ssh and vi all the time.
>
> >>> 08-12-2009 kl. 07:05 skrev "mailinglist"  i
> meddelelsen <4b1dfabc02e10...@mail.fribert.dk>:
>Hi Mark
>
> Ok, thanks.
> Yes I have a gateway placed in external called musimi.dk (or should it be
> in public?), and I'll just create the empty XML's in lan to get rid of that
> error.
>
> I'll remove the second part of the dialplan, my idea was that it was needed
> for calls between sip phones hooked up to the freeswitch.
>
> Now the remaining problem:
> When I call ext 1002 from ext 1001 I see this message and get an error, the
> same goes for dialing 0 to get an external number:
>
> 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel
> sofia/internal/1...@10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb]
> 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing
> 1001->1002 in context default
> 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel
> sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb]
> 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1
> [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
> 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed.
> Cause: NO_ROUTE_DESTINATION
> 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup
> sofia/internal/1...@10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION]
> 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2
> (sofia/external/$1) Ended
> 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close
> Channel sofia/external/$1 [CS_DESTROY]
> 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 (
> sofia/internal/1...@10.11.12.25) Ended
> 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close
> Channel sofia/internal/1...@10.11.12.25 [CS_DESTROY]
> I don't see any mention of the statements in the Dialplan, so for me it
> looks like it haven't registered the Dialplan?
>
> Best regards
> Kenneth
>
> >>> 08-12-2009 kl. 03:05 skrev Mark Crane  i meddelelsen
> 659603.29094...@web56408.mail.re3.yahoo.com:
>
> Question --
> If I do a reloadxml it gives me this output on the console:
> freeswi...@firewall.fribert.dk>
> reloadxml
> 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open
> /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No
> such file or directory)
> Error including
> /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No
> such file or directory)
>
> I'm not sure if it's a genuine problem,as I can see it, it just complains
> that I haven't created any sip_profiles in /lan, but is that necessary?
>
> Response: --
> This isn't really a problem. To get rid of the error simply put a blank xml
> file into each folder as in the internal and external directories. Dump the
> lan directory and lan profile as mentioned earlier.
>
> Question --
>
> Extension Name  musimi.dk
> Enabled true
> Order 001
> Description  ...
>
> condition ^0(.\d+)$
> action bridge sofia/gateway/musimi.dk/$1
>
> Response: --
>
> This is correct as long as you have a gateway that is registered called
> musimi.dk
>
> Question --
> Extension Name 10.11.12.25
> Enabled true
> Order 002
> Description ...
>
> action bridge  sofia/internal/$
>
> Response: --
>
> No idea what this is for its not needed as far as I can tell.
>
>
> Now please summarize what you still need help on.
>
>
> Mark J Crane
> http://fusionpbx.com
> pfSense FreeSWITCH package developer
>
>
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Re: [Freeswitch-users] How to pick up someone's phone remotely.

2009-11-10 Thread Nandy Dagondon

just change the dialplan/default.xml as mentioned by brian but i think you
can't use # as the first key 'cuz it normally used as a Send key. you may
change # to * (star key).


On Wed, Nov 11, 2009 at 12:06 AM, Ognjen Seslija  wrote:

> Add the following:
>
>  .
>
> after
>
>data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>
> in local extensions default example, or change it globally previously than
> this extension. You can join us on IRC if you can any more questions
> (sekil).
>
> Regards,
> Ognjen
>
>
>
> On Tue, Nov 10, 2009 at 4:01 PM, Piotr Żurek wrote:
>
>> Hello.
>>
>> Thank You developers for Freeswitch.
>> I have installed it lately and it's working quite nicely, but I have one
>> problem:
>>
>> I need to mimic behavior of my current analogue PBX installation using
>> Freeswitch.
>>
>> This is the scenario:
>> In the office with a few desks (extensions 1000-1010) and only one person
>> behind one of desks (whatever extension - in example 1000).
>> 1. There's incoming call on _one_ of extensions 1001-1010
>> 2. The person on extension 1000 wants to answer this call on his phone so
>> dials #37 and this call is redirected to his phone.
>>
>> That's how it works on my office on analogue PBX system. Anyone can answer
>> a call from any other phone as long as it hasn't been answered already.
>>
>> I tried to use the intercept action (with global example in default
>> config) but it's not what I need because it intercepts the call even if it's
>> already answered. I need to intercept all but only unanswered calls. I tried
>> to use Redirect but it does not work on other's extensions call's (or does
>> it?).
>>
>> Please help.
>> Peter Żurek
>>
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Re: [Freeswitch-users] Preannouncing a message before making a call

2009-10-14 Thread Nandy Dagondon

hi simon, implementation is almost the same. here's my dialplan:


 
   
   
   
   ... add more files to play here ...
   
via " where  might
> be a country, or area and  may be the name of one of several
> SIP gateways I have configured.  Each phrase has been a prerecorded wav
> file played in sequence.
>
> [macro-X]
> exten => s,1,Playback(/etc/asterisk/wav/dialing)
> exten => s,2,Playback(/etc/asterisk/wav/${ARG2})# Name of destination
> exten => s,3,Playback(/etc/asterisk/wav/via)
> exten => s,4,Playback(/etc/asterisk/wav/X)  # Gateway name
> exten => s,5,Dial(SIP/X/${ARG1},60,tT)  # ARG1 = Destination
> number
>
>
> # actual usage
> exten => _0044.,1,Macro(X,${EXTEN},touk)
>
> How would I approach doing something like this in FreeSwtich?
>
> Thanks for any pointers.
>
> Simon
>
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Re: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Update: Valet Parking

2009-10-13 Thread Nandy Dagondon

i come across the valet_park application when i just finished an
improved-version of the call parking (using mod_fifo)  such that it is
parked to different ext'n numbers when the caller is att_xfer'd to ext 777.
i used  strftime(%s) to generate 700~759 parking numbers. i also added
feature that if the parked call is not picked up after xx secs timeout, it's
returned back to the transferer. the only glitch here is - when another
parking is done exactly 1 minute later (unless we'll limit the timeout to 59
secs). i hv to use att_xfer so that the transferer can hear the parking
number.

i'll submit the dialplan if it's worth seeing it.

On Tue, Oct 13, 2009 at 3:55 AM, William King  wrote:

> I don't know if this was mentioned yet. It would be useful to have a way
> to have the parking lot automatically find the next available spot and
> tts it to the person parking the call.
>
> Then the auto unpark would pop off the lowest numbered lot, or return
> fail if there is nobody in the parking lots etc.
>
> -William King
>
> Michael Collins wrote:
> >
> >
> > On Sat, Oct 10, 2009 at 8:18 PM, mayamatakeshi
> > mailto:mayamatake...@gmail.com>> wrote:
> >
> >
> >
> > On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins
> > mailto:m...@freeswitch.org>> wrote:
> >
> > FYI,
> >
> > The FreeSWITCH devs have added valet parking! Check it out:
> > http://www.freeswitch.org/node/207
> >
> > Let us know what you think.
> >
> >
> > Very nice.
> >
> > But I think a valet_unpark app is missing.
> > If the intention of the person sent to the valet lot is to
> > retrieve a call there, the person can assume the call was already
> > retrieved by someone else or that the caller hung up if he/she
> > hears MOH. But it would be nicer to have a valet_unpark app to
> > fail and let the dialplan play a message.
> >
> > I understand what you are saying. I'm not sure I agree, but we'll kick
> > the idea around when we have a few minutes and let you know what we
> > decide.
> > -MC
> >
> > 
> >
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Re: [Freeswitch-users] wav files compression

2009-10-04 Thread Nandy Dagondon

agree that WAV/PCMA/PCMU formats are best for performance. you can use
mp3/ogg ONLY to archive recorded files.
/nandy

On Sun, Oct 4, 2009 at 7:38 AM, Tihomir Culjaga  wrote:

> also, you can store files in PCMA/PCMU format and avoid transcoding at
> all... and as said disk space is cheap.. go get some...
>
>
> On Sat, Oct 3, 2009 at 7:07 PM, Diego Viola  wrote:
>
>> Why is not recommended?
>>
>>
>> On Sat, Oct 3, 2009 at 2:52 PM, Brian West  wrote:
>>
>>> MP3 is NOT recommend and if WAV files are too large you can mosey on
>>> down to the local Best Buy and snag 1.5TB of disk for like $119
>>> dollars.  Disk is cheap.
>>>
>>> /b
>>>
>>> On Oct 3, 2009, at 1:44 AM, Keith Wood wrote:
>>>
>>> >
>>> > I am working on an implementation for managing thousands of IVR
>>> > within an organization.  Right now, I am storing all audio files in
>>> > wav format, but it quickly become unmanagable because the size of
>>> > these wav files ( 8 bits mono ) quickly consuming a lot of the disk
>>> > space.
>>> >
>>> > Is there anyway I can store those audio files and still have high
>>> > quality audio for IVR?  I know mp3 is smaller but freeswitch does
>>> > not support it.
>>> >
>>> > any ideas?
>>> >
>>> > keith
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Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey

2009-09-28 Thread Nandy Dagondon

please check Redfone's foneBridge. i come across an article about this
before.

On Tue, Sep 29, 2009 at 5:26 AM, SP  wrote:

> Yes, with OpenSER/Kamailio/OpenSIPs/SER (you pick a name).
>
> On Mon, Sep 28, 2009 at 15:12, Mike van Lammeren 
> wrote:
> > Hello!
> >
> > I followed a tutorial that describes load-balancing Asterisk with
> > Ultramonkey, but cannot get it to work with FreeSWITCH:
> > http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf
> > My X-Lite client fails to register with the server. I have looked at the
> > packets with wireshark, and found that when X-Lite sends a SIP Register
> > packet, it gets an ICMP response: Destination Unreachable (Port
> > Unreachable).
> > Has anyone got a load-balanced FreeSWITCH setup working?
> > Mike van Lammeren
> >
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> >
>
>
>
> --
> Shannon
>
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Re: [Freeswitch-users] session record does not for very short calls

2009-09-16 Thread Nandy Dagondon

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?
> >
> >
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Re: [Freeswitch-users] FS 1.0.4 erl configure error

2009-09-16 Thread Nandy Dagondon

hi mike, i download the tarball file to check the configure script. it's
clean. so, there must be an error during my first download or build. - nandy


On Wed, Sep 16, 2009 at 3:54 PM, Michael Jerris  wrote:

> Are those in the Tarball?
>
>
> On Sep 15, 2009, at 11:47 PM, Nandy Dagondon  wrote:
>
> it's working now. the problem? it's the configure script itself. some ^M
> characters somehow crept into the line containing ac_config_files.  tks for
> the tip Andrew!
>
> /nandy
>
> On Wed, Sep 16, 2009 at 11:26 AM, Nandy Dagondon < 
> nandy1...@gmail.com> wrote:
>
>> is the Erlang source needed in the FS source directory?
>>
>> /nandy
>>
>>
>>
>> On Wed, Sep 16, 2009 at 11:21 AM, Nandy Dagondon < 
>> nandy1...@gmail.com> wrote:
>>
>>> the ./configure script aborts after the last error message. any hint
>>> where to look for the problem? tks.
>>>
>>> /nandy
>>>
>>>
>>>
>>> On Wed, Sep 16, 2009 at 10:53 AM, Andrew Thompson < 
>>> and...@hijacked.us> wrote:
>>>
>>>> On Wed, Sep 16, 2009 at 10:04:14AM +0800, Nandy Dagondon wrote:
>>>> > hi folks, anyone encountered this problem? tks.
>>>>
>>>> I don't think this has anything to do with erlang or the freeswitch
>>>> erlang module, it's simply that that module's config checks are run
>>>> shortly before the real failure occurs.
>>>>
>>>> Andrew
>>>>
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Re: [Freeswitch-users] FS 1.0.4 erl configure error

2009-09-15 Thread Nandy Dagondon

mike,  got it from tarball.  - nandy


On Wed, Sep 16, 2009 at 11:51 AM, Michael Jerris  wrote:

> something is messed up in your build environment, it has nothing to do with
> erlang.  Is this with a fresh svn checkout or tarball?
> Mike
>
> On Sep 13, 2009, at 10:27 AM, Nandy Dagondon wrote:
>
> hi,
>
> i want to enable odbc support which is required in mod_lcr feature.
> however, i encounter ./configure problem after installing Erlang R13B01.
> this is the portion of the error messages:
>
> ...
> checking for erl... /usr/local/bin/erl
> checking erlang version... 5.7.2
> checking erlang libdir... /usr/local/lib/erlang/lib/erl_interface-3.6.2/lib
> checking erlang incdir...
> /usr/local/lib/erlang/lib/erl_interface-3.6.2/include
> checking ei.h usability... yes
> checking ei.h presence... no
> configure: WARNING: ei.h: accepted by the compiler, rejected by the
> preprocessor!
> configure: WARNING: ei.h: proceeding with the compiler's result
> checking for ei.h... yes
> checking for ei_encode_version in -lei... yes
> checking for ei_link_unlink in -lei... no
> configure: Your erlang seems OK, do not forget to enable mod_erlang_event
> in modules.conf
> configure: creating ./config.status
> config.status: creating src/include/switch_version.h.in
> .infig.status: error: cannot find input file: Makefile
>  END 
>
> i set ERL_TOP environment variable to the source directory. has anyone
> encountered this problem? can anyone give me a hint what's wrong. i'm
> compiling FS 1.0.4.
>
> thank you,
> /nandy
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Re: [Freeswitch-users] FS 1.0.4 erl configure error

2009-09-15 Thread Nandy Dagondon

it's working now. the problem? it's the configure script itself. some ^M
characters somehow crept into the line containing ac_config_files.  tks for
the tip Andrew!

/nandy

On Wed, Sep 16, 2009 at 11:26 AM, Nandy Dagondon wrote:

> is the Erlang source needed in the FS source directory?
>
> /nandy
>
>
>
> On Wed, Sep 16, 2009 at 11:21 AM, Nandy Dagondon wrote:
>
>> the ./configure script aborts after the last error message. any hint where
>> to look for the problem? tks.
>>
>> /nandy
>>
>>
>>
>> On Wed, Sep 16, 2009 at 10:53 AM, Andrew Thompson wrote:
>>
>>> On Wed, Sep 16, 2009 at 10:04:14AM +0800, Nandy Dagondon wrote:
>>> > hi folks, anyone encountered this problem? tks.
>>>
>>> I don't think this has anything to do with erlang or the freeswitch
>>> erlang module, it's simply that that module's config checks are run
>>> shortly before the real failure occurs.
>>>
>>> Andrew
>>>
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Re: [Freeswitch-users] FS 1.0.4 erl configure error

2009-09-15 Thread Nandy Dagondon

is the Erlang source needed in the FS source directory?

/nandy


On Wed, Sep 16, 2009 at 11:21 AM, Nandy Dagondon wrote:

> the ./configure script aborts after the last error message. any hint where
> to look for the problem? tks.
>
> /nandy
>
>
>
> On Wed, Sep 16, 2009 at 10:53 AM, Andrew Thompson wrote:
>
>> On Wed, Sep 16, 2009 at 10:04:14AM +0800, Nandy Dagondon wrote:
>> > hi folks, anyone encountered this problem? tks.
>>
>> I don't think this has anything to do with erlang or the freeswitch
>> erlang module, it's simply that that module's config checks are run
>> shortly before the real failure occurs.
>>
>> Andrew
>>
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Re: [Freeswitch-users] FS 1.0.4 erl configure error

2009-09-15 Thread Nandy Dagondon

the ./configure script aborts after the last error message. any hint where
to look for the problem? tks.

/nandy


On Wed, Sep 16, 2009 at 10:53 AM, Andrew Thompson wrote:

> On Wed, Sep 16, 2009 at 10:04:14AM +0800, Nandy Dagondon wrote:
> > hi folks, anyone encountered this problem? tks.
>
> I don't think this has anything to do with erlang or the freeswitch
> erlang module, it's simply that that module's config checks are run
> shortly before the real failure occurs.
>
> Andrew
>
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Re: [Freeswitch-users] FS 1.0.4 erl configure error

2009-09-15 Thread Nandy Dagondon

hi folks, anyone encountered this problem? tks.
/nandy

On Mon, Sep 14, 2009 at 2:20 PM, Nandy Dagondon  wrote:

> meftah,
>
> i disabled mod_erlang_event in modules.conf. unixodbc is installed already.
> still ... the same error message. tks for your input.
>
> /nandy
>
>
> On Sun, Sep 13, 2009 at 11:56 PM, Meftah Tayeb wrote:
>
>>  hello,
>> i think you enabled mod_erlang_event in the modules.conf
>> install unixodbc if is not installed
>> thanks
>>
>> Nandy Dagondon a écrit :
>>
>> hi,
>>
>> i want to enable odbc support which is required in mod_lcr feature.
>> however, i encounter ./configure problem after installing Erlang R13B01.
>> this is the portion of the error messages:
>>
>> ...
>> checking for erl... /usr/local/bin/erl
>> checking erlang version... 5.7.2
>> checking erlang libdir...
>> /usr/local/lib/erlang/lib/erl_interface-3.6.2/lib
>> checking erlang incdir...
>> /usr/local/lib/erlang/lib/erl_interface-3.6.2/include
>> checking ei.h usability... yes
>> checking ei.h presence... no
>> configure: WARNING: ei.h: accepted by the compiler, rejected by the
>> preprocessor!
>> configure: WARNING: ei.h: proceeding with the compiler's result
>> checking for ei.h... yes
>> checking for ei_encode_version in -lei... yes
>> checking for ei_link_unlink in -lei... no
>> configure: Your erlang seems OK, do not forget to enable mod_erlang_event
>> in modules.conf
>> configure: creating ./config.status
>> config.status: creating src/include/switch_version.h.in
>> .infig.status: error: cannot find input file: Makefile
>>  END 
>>
>> i set ERL_TOP environment variable to the source directory. has anyone
>> encountered this problem? can anyone give me a hint what's wrong. i'm
>> compiling FS 1.0.4.
>>
>> thank you,
>> /nandy
>>
>> --
>>
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Re: [Freeswitch-users] FS 1.0.4 erl configure error

2009-09-13 Thread Nandy Dagondon

meftah,

i disabled mod_erlang_event in modules.conf. unixodbc is installed already.
still ... the same error message. tks for your input.

/nandy

On Sun, Sep 13, 2009 at 11:56 PM, Meftah Tayeb wrote:

>  hello,
> i think you enabled mod_erlang_event in the modules.conf
> install unixodbc if is not installed
> thanks
>
> Nandy Dagondon a écrit :
>
> hi,
>
> i want to enable odbc support which is required in mod_lcr feature.
> however, i encounter ./configure problem after installing Erlang R13B01.
> this is the portion of the error messages:
>
> ...
> checking for erl... /usr/local/bin/erl
> checking erlang version... 5.7.2
> checking erlang libdir... /usr/local/lib/erlang/lib/erl_interface-3.6.2/lib
> checking erlang incdir...
> /usr/local/lib/erlang/lib/erl_interface-3.6.2/include
> checking ei.h usability... yes
> checking ei.h presence... no
> configure: WARNING: ei.h: accepted by the compiler, rejected by the
> preprocessor!
> configure: WARNING: ei.h: proceeding with the compiler's result
> checking for ei.h... yes
> checking for ei_encode_version in -lei... yes
> checking for ei_link_unlink in -lei... no
> configure: Your erlang seems OK, do not forget to enable mod_erlang_event
> in modules.conf
> configure: creating ./config.status
> config.status: creating src/include/switch_version.h.in
> .infig.status: error: cannot find input file: Makefile
>  END 
>
> i set ERL_TOP environment variable to the source directory. has anyone
> encountered this problem? can anyone give me a hint what's wrong. i'm
> compiling FS 1.0.4.
>
> thank you,
> /nandy
>
> --
>
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[Freeswitch-users] FS 1.0.4 erl configure error

2009-09-13 Thread Nandy Dagondon

hi,

i want to enable odbc support which is required in mod_lcr feature. however,
i encounter ./configure problem after installing Erlang R13B01. this is the
portion of the error messages:

...
checking for erl... /usr/local/bin/erl
checking erlang version... 5.7.2
checking erlang libdir... /usr/local/lib/erlang/lib/erl_interface-3.6.2/lib
checking erlang incdir...
/usr/local/lib/erlang/lib/erl_interface-3.6.2/include
checking ei.h usability... yes
checking ei.h presence... no
configure: WARNING: ei.h: accepted by the compiler, rejected by the
preprocessor!
configure: WARNING: ei.h: proceeding with the compiler's result
checking for ei.h... yes
checking for ei_encode_version in -lei... yes
checking for ei_link_unlink in -lei... no
configure: Your erlang seems OK, do not forget to enable mod_erlang_event in
modules.conf
configure: creating ./config.status
config.status: creating src/include/switch_version.h.in
.infig.status: error: cannot find input file: Makefile
 END 

i set ERL_TOP environment variable to the source directory. has anyone
encountered this problem? can anyone give me a hint what's wrong. i'm
compiling FS 1.0.4.

thank you,
/nandy
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Re: [Freeswitch-users] example configs for FS outside of NAT?

2009-09-10 Thread Nandy Dagondon

hi jason,

yes, we're aware of the external profile. but the sample profile shows only
how to register FS to SIP gateways - not external clients registering to FS.
the directory/default/*xml belongs to the internal profile. how can we
create another directory for external clients? we like to see sample configs
in the distribution.

tks for the clarification,
/nandy

On Thu, Sep 10, 2009 at 5:52 PM, Jason White  wrote:

> Nandy Dagondon  wrote:
> > for outside clients to register w/ the internal profile, the router has
> to
> > forward port 5060 to FS. am i correct?
>
> Yes, but by default the internal profile doesn't handle nat, which is why
> (if
> I recall correctly) it has been recommended that the external profile be
> used
> to register clients that are not on the local network when nat is involved.
>
>
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Re: [Freeswitch-users] example configs for FS outside of NAT?

2009-09-10 Thread Nandy Dagondon

hi brian,

for outside clients to register w/ the internal profile, the router has to
forward port 5060 to FS. am i correct?

/nandy

On Wed, Sep 9, 2009 at 10:28 PM, Brian West  wrote:

> Those configs will still work.
> /b
>
> On Sep 9, 2009, at 6:16 AM, Jörg Hartmann wrote:
>
> Hi there,
>
> the internal.xml and external.xml examples are for situations where FS is
> running inside a company's private network, behind a NAT router. So
> internal.xml connects the clients to FS without crossing a NAT, within the
> same private network, while external.xml connects SIP providers through the
> NAT router.
>
> But what if FS is running with a public IP (and DNS entry) outside the
> private network, so that the clients have to pass the NAT router to connect
> with FS, while FS can connect to SIP providers directly? Are there any
> example configs for such a configuration?
>
> Thanks in advance,
> Cheers,
> JH
>
>
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Re: [Freeswitch-users] Call Forwarding Question

2009-09-06 Thread Nandy Dagondon

nik,

please try the "legs" variable
http://www.nabble.com/CDR-accounting-question-td19212516.html

/nandy


On Sun, Sep 6, 2009 at 6:40 PM, Nikolai Geordzhev wrote:

> Hi,
> I`m trying to implement Call Forwarding in my FS setup. I set a user
> variable managing the type of forwarding (busy,no answer,unconditional) and
> the destination the phone is forwarded to:
>
> 
>  
>  
>

Re: [Freeswitch-users] Set disable-transcoding in dialplan

2009-09-03 Thread Nandy Dagondon

rod,

it looks more complicated now when PEER C comes to the picture. i think
we'll have to wait for the availability of g729 on FS, as per Anthony's
post.

/nandy


On Fri, Sep 4, 2009 at 1:54 PM, rod  wrote:

> Hi Nandy,
>
> yes already tried this, but if I use proxy_media=true, FS makes no
> control on the content of the RTP stream. But the pbm is that I need to
> use this:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf_generate
> This function enables transcoding of SIP_INFO or RFC2833 to Inband DTMF
> in G711
>
> But this feature doesn't work if I'm using proxy_media=true.
>
> In fact my setup is the following:
>
> CPE using G711A, G729 and SIP INFO for DTMF
> PEER_A using G729 only and RFC_2833
> PEER_B using G711 and SIP INFO
>
> I have been able to make this works, with proxy_media=true for PEER_B
> cause I don't need transcoding of DTMF (SIP INFO to SIP INFO).
> For PEER_A, proxy_media is set to false (default) cause  I need
> transcoding SIP INFO to RFC2833. I'm able to use G729 using
> codec_negotiation=greedy and setting G729 with highest priority on my
> internal profile.
>
> But the pbm is that I need to add PEER_C.
> PEER_C needs G711 with transcoding DTMF from SIP_INFO to Inband.
>
> And this is where I'm stuck, cause using "greedy settings and G729 with
> priority 1 in my codec list and proxy_media=false" force FS to negotiate
> G729 on leg A. But Leg B is willing to use G711 and FS is unable to
> transcode G729 <---> G711.
>
> I was wondering if there is a way for FS to force the codec order on Leg
> A with some knowledge of the preferred codec on Leg B, ie I know that
> Leg B will always use G711 so that I want to biase the SDP answer on Leg
> A based on this fact.
>
> regards,
> rod
>
> Nandy Dagondon a écrit :
> > rod,
> >
> > have you tried this?
> >
> http://lists.freeswitch.org/pipermail/freeswitch-users/2008-March/002199.html
> >
> > /nandy
> >
> >
> > On Thu, Sep 3, 2009 at 2:50 PM, rod  > <mailto:kawa...@laposte.net>> wrote:
> >
> > Hi Michael,
> >
> > I did some tests but I haven't been successful, so there is what I'm
> > trying to achieve:
> >
> > On A leg, my phone is using: PCMA and G729 (in this priority order)
> >
> > With PEER A, I want to use only G729 (thats is the only codec that
> > this
> > PEER support), so that the RTP flow will be:
> >Phone-G729FS-G729-PEER_A
> >
> > With PEER B, I want to use only G711, so:
> >Phone-G711FS-G711-PEER_B
> >
> > In fact, I'd like to force FS announcing the codec list priority
> based
> > on the priority of the codec announced by the PEER, cause FS is
> unable
> > to transcode G729 <--> G711.
> >
> > Tried a lot of things (greedy for codec-negociation, late_codec,
> > disable_transcoding, codec-prefs) without success.
> >
> > If you have some clue.
> >
> > regards,
> > rod
> >
> > Michael Collins a écrit :
> > > Check out this page:
> > > http://wiki.freeswitch.org/wiki/Codec_negotiation
> > >
> > > Late negotiation will probably let you handle all the cases you
> > need.
> > > -MC
> > >
> > > On Mon, Aug 31, 2009 at 8:00 AM, rod  > <mailto:kawa...@laposte.net>
> > > <mailto:kawa...@laposte.net <mailto:kawa...@laposte.net>>> wrote:
> > >
> > > Hi all,
> > >
> > > I'm wondering if I can do something like this:
> > >- in my internal profile, I have this because of some PEER
> > > using G729:
> > >  - 
> > >
> > > But for a specific PEER, I'd like to activate transcoding:
> > >  - for this PEER, only G711 is used
> > >  - I'd like to transcode DTMF SIP INFO or RFC2833 to
> > INBAND
> > >
> > > So in my dialplan, I tried before bridging:
> > >
> > >-  > data="disable-transcoding=false"/>
> > >- 
> > >
> > > But I still see RFC2833 events between my FS and PEER and
> > the DTMF are
> > > not working.
> > >
> > > So 2 questi

Re: [Freeswitch-users] Set disable-transcoding in dialplan

2009-09-03 Thread Nandy Dagondon

rod,

have you tried this?
http://lists.freeswitch.org/pipermail/freeswitch-users/2008-March/002199.html

/nandy


On Thu, Sep 3, 2009 at 2:50 PM, rod  wrote:

> Hi Michael,
>
> I did some tests but I haven't been successful, so there is what I'm
> trying to achieve:
>
> On A leg, my phone is using: PCMA and G729 (in this priority order)
>
> With PEER A, I want to use only G729 (thats is the only codec that this
> PEER support), so that the RTP flow will be:
>Phone-G729FS-G729-PEER_A
>
> With PEER B, I want to use only G711, so:
>Phone-G711FS-G711-PEER_B
>
> In fact, I'd like to force FS announcing the codec list priority based
> on the priority of the codec announced by the PEER, cause FS is unable
> to transcode G729 <--> G711.
>
> Tried a lot of things (greedy for codec-negociation, late_codec,
> disable_transcoding, codec-prefs) without success.
>
> If you have some clue.
>
> regards,
> rod
>
> Michael Collins a écrit :
> > Check out this page:
> > http://wiki.freeswitch.org/wiki/Codec_negotiation
> >
> > Late negotiation will probably let you handle all the cases you need.
> > -MC
> >
> > On Mon, Aug 31, 2009 at 8:00 AM, rod  > > wrote:
> >
> > Hi all,
> >
> > I'm wondering if I can do something like this:
> >- in my internal profile, I have this because of some PEER
> > using G729:
> >  - 
> >
> > But for a specific PEER, I'd like to activate transcoding:
> >  - for this PEER, only G711 is used
> >  - I'd like to transcode DTMF SIP INFO or RFC2833 to INBAND
> >
> > So in my dialplan, I tried before bridging:
> >
> >- 
> >- 
> >
> > But I still see RFC2833 events between my FS and PEER and the DTMF
> are
> > not working.
> >
> > So 2 questions:
> >- does application "start_dtmf_generate" requires transcoding
> >- if yes, can I set the variable disable-transcoding in my
> dialplan
> >
> > regards,
> > rod
> >
> > ___
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> > FreeSWITCH-users@lists.freeswitch.org
> > 
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> >
> >
> > 
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Re: [Freeswitch-users] FS - external users

2009-08-30 Thread Nandy Dagondon

no need to modify anything on the directory entries - unless you add more
extension numbers. FS has a uPNP feature already. if your router has uPNP,
the external phone only needs the ff:
1. public IP of the FS (or hostname)
2. username
3. password

Items 2 & 3 are found in the directory entries.


On Sun, Aug 30, 2009 at 8:49 AM, tom  wrote:

> hi,
>
> as newbie here a simple one:
>
> based on the std-installation, what do i need to do to have a user
> connected from the outside. lets assume i want to leave 1000-1020 as they
> are, i make a copy of 1020, call it 1021 and then?
> should i leave it where it is? do i have to move the file? and what about
> he conetxt?
>
> thx
>
>
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Re: [Freeswitch-users] Auto Nat

2009-08-07 Thread Nandy Dagondon

r 13612 is after 1.0.3. you better get 1.0.4 recently released.

-nandy

On Fri, Aug 7, 2009 at 8:19 PM,  wrote:

> I sent my first e-mail to the list this morning (about 4 hours ago) but it
> does not seem to have arrived back, even though I have received other,
> later posts.
>
> I have another question related to the first (about how to set everything
> up in a double nat environment) - so if I see this and not the other, I
> will send the first again.
>
> I am currently running the stable version 1.0.3 of freeswitch.  The wiki
> page says that auto-nat is introduced at r 13612.  Is this before or after
> that revision? (I don't want to have to download and rebuild the entire
> thing if I don't have to).
>
>
>
>
>
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Re: [Freeswitch-users] /etc/openzap/tones.conf for UK

2009-08-07 Thread Nandy Dagondon

you can create your tones.conf using call progress tones  found at
http://www.3amsystems.com/wireline/tone-search.htm


On Fri, Aug 7, 2009 at 7:17 PM, Merul Patel  wrote:

> Where can I find a sample tones.conf file for the UK? Am trying to
> configure a USBFXO device for outbound calls.
>
> Thanks in advance,
>
> Merul
>
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Re: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix

2009-07-27 Thread Nandy Dagondon

ed,

i mean you use separate extension names:



   




   


btw, you should also use separate gateway names "sip1" and "sip2". so
differentiate them in the bridge application.

On Mon, Jul 27, 2009 at 4:16 PM, Jason White  wrote:

> Edmar Cruz  wrote:
> >
> > Not working just the same both of them are running
>
> Do you have them as separate extensions in the dial plan?
>
>
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Re: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix

2009-07-27 Thread Nandy Dagondon

in my implementation, i would use 2 separate conditions that looks like
this:


   


   


On Mon, Jul 27, 2009 at 2:42 PM, Edmar Cruz  wrote:

>
> Hi FS Users,
>
> I just want to try multiple gateways. It works actually like this...
>
>
>
>But I test call like 513 at 222.333.444.555, it also calls the
> second bridge 111.222.333.333.
>
>   It there any way to determine which prefix will call to a bridge
> specified.
>
>E.g.
>
>for bridge 1: with prefix of 51 the call with run to 222.333.444.555 not
> at the second bridge and vice versa. Please help..
> --
> View this message in context:
> http://www.nabble.com/Multiple-gateways%3A-Priority-the-first-bridge-with-prefix-tp24674280p24674280.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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Re: [Freeswitch-users] Compact, fanless appliance?

2009-07-05 Thread Nandy Dagondon

ok. w/ my apologies. - nandy


On Sun, Jul 5, 2009 at 10:49 AM, Ken Rice  wrote:

>  No need to bump these things as this is a mailing list and it annoys
> quite a few people when you do that
>
>
> --
> *From: *Nandy Dagondon 
> *Reply-To: *
> *Date: *Sun, 5 Jul 2009 10:41:18 +0800
> *To: *
> *Subject: *Re: [Freeswitch-users] Compact, fanless appliance?
>
> just bumping this topic.
> -nandy
>
> On Fri, May 8, 2009 at 12:44 AM, Fred-145  wrote:
>
>
>
> Antonio Gallo wrote:
> > Alix cases are like 6/9 € from their shop site. I think its easy to find
> > someone who work with aluminium that can make for you custom boxes for
> > like like 6/20 € at pcs
>
> Unfortunately, none of the PCEngines cases (
> www.pcengines.ch/order1.php?c=2 
> <http://www.pcengines.ch/order1.php?c=2><http://www.pcengines.ch/order1.php?c=2>)
> allow for a PCI slot, either on top of the mobo, or away from it :-/
>
> I'll see if I can get those from Soekris (
> http://soekris.eu/shop/cases_en/) <http://soekris.eu/shop/cases_en/%29>
> allow this, and if I can get a good price for a case + PSU.
>
> Thank you.
> --
> View this message in context:
> http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23430873.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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Re: [Freeswitch-users] Compact, fanless appliance?

2009-07-04 Thread Nandy Dagondon

just bumping this topic.
-nandy

On Fri, May 8, 2009 at 12:44 AM, Fred-145  wrote:

>
>
> Antonio Gallo wrote:
> > Alix cases are like 6/9 € from their shop site. I think its easy to find
> > someone who work with aluminium that can make for you custom boxes for
> > like like 6/20 € at pcs
>
> Unfortunately, none of the PCEngines cases (
> www.pcengines.ch/order1.php?c=2)
> allow for a PCI slot, either on top of the mobo, or away from it :-/
>
> I'll see if I can get those from Soekris (http://soekris.eu/shop/cases_en/
> )
> allow this, and if I can get a good price for a case + PSU.
>
> Thank you.
> --
> View this message in context:
> http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23430873.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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Re: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance?

2009-07-04 Thread Nandy Dagondon

we have a forum on compact,fanless last may.


On Sun, Jul 5, 2009 at 6:21 AM, William Suffill
wrote:

> Ya I have a SheevaPlug but yet to have anything interesting to report
> about making it do anything. The potential is there tho.
>
> -- W
>
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Re: [Freeswitch-users] freeswitch support PCMU only?

2009-07-01 Thread Nandy Dagondon

is PCMA enabled in X-Lite, too?

On Wed, Jul 1, 2009 at 9:25 PM, qian ma  wrote:

>
> inbound_codec_negotiation is generous
> and the xlite PCMU is enabled.
>
> my var.xml.conf:
> 
> 
>
>
>
> 2009/7/1 Nandy Dagondon 
>
>> sorry. i mean check the x-lite client if PCMA is enabled?
>>
>>
>> On Wed, Jul 1, 2009 at 9:14 PM, Nandy Dagondon  wrote:
>>
>>> check the value of "inbound_codec_negotiation" in  the sip_profiles/*.xml
>>> files. is it "generous" or "greedy"? you should also check if the endpoint
>>> is offering PCMU.
>>>
>>>
>>>
>>> On Wed, Jul 1, 2009 at 8:27 PM, qian ma  wrote:
>>>
>>>> i want the fs accept the PCMA not PCMU.
>>>> i add "PCMA" in global_codec_prefs and outbound_codec_prefs, it doesn't
>>>> work. FS only accept PCMU.
>>>> why??
>>>>
>>>>
>>>>
>>>>
>>>> 2009/7/1 Nandy Dagondon 
>>>>
>>>> you FS doesn't accept PCMU. try to add "PCMU" on both variables.
>>>>>
>>>>>
>>>>> On Wed, Jul 1, 2009 at 3:44 PM, qian ma  wrote:
>>>>>
>>>>>> thanks for your replies.
>>>>>>
>>>>>  my var.xml:
>>>>>> 
>>>>>> 
>>>>>>
>>>>>>
>>>>>> below is the sip trace:
>>>>>>  recv 781 bytes from udp/[58.212.219.104]:40508 at 07:42:37.086711:
>>>>>>
>>>>>> 
>>>>>>INVITE sip:123...@58.212.219.104 SIP/2.0
>>>>>>Via: SIP/2.0/UDP 192.168.1.241:8422
>>>>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport
>>>>>>Max-Forwards: 70
>>>>>>Contact: 
>>>>>>To: "123456"
>>>>>> >
>>>>>>From: "9876"
>>>>>> >;tag=057de365
>>>>>>Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
>>>>>>CSeq: 1 INVITE
>>>>>>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
>>>>>> SUBSCRIBE, INFO
>>>>>>Content-Type: application/sdp
>>>>>>User-Agent: eyeBeam release 1102u stamp 52345
>>>>>>Content-Length: 237
>>>>>>
>>>>>>v=0
>>>>>>o=- 6 2 IN IP4 192.168.1.241
>>>>>>s=CounterPath eyeBeam 1.5
>>>>>>c=IN IP4 192.168.1.241
>>>>>>t=0 0
>>>>>>m=audio 57862 RTP/AVP 8 101
>>>>>>a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862
>>>>>>a=fmtp:101 0-15
>>>>>>a=rtpmap:101 telephone-event/8000
>>>>>>a=sendrecv
>>>>>>
>>>>>> 
>>>>>> send 380 bytes to udp/[58.212.219.104]:40508 at 07:42:37.087505:
>>>>>>
>>>>>> 
>>>>>>SIP/2.0 100 Trying
>>>>>>Via: SIP/2.0/UDP 192.168.1.241:8422
>>>>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104
>>>>>>From: "9876"
>>>>>> >;tag=057de365
>>>>>>To: "123456"
>>>>>> >
>>>>>>Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
>>>>>>CSeq: 1 INVITE
>>>>>>User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M
>>>>>>Content-Length: 0
>>>>>>
>>>>>>
>>>>>> 
>>>>>> 2009-07-01 15:42:37.87800 [NOTICE] switch_channel.c:602 New Channel
>>>>>> sofia/maq/9...@58.212.219.104 [574766e1-e71f-4a62-a659-1d43bea135bd]
>>>>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3214 Channel
>>>>>> sofia/maq/9...@58.212.219.104 entering state [received][100]
>>>>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3221 Remote SDP:
>>>>>> v=0
>>>>>> o=- 6 2 IN IP4 192.168.1.241
>>>>>> s=CounterPath eyeBeam 1.5
>>>>>> c=

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

2009-07-01 Thread Nandy Dagondon

sorry. i mean check the x-lite client if PCMA is enabled?


On Wed, Jul 1, 2009 at 9:14 PM, Nandy Dagondon  wrote:

> check the value of "inbound_codec_negotiation" in  the sip_profiles/*.xml
> files. is it "generous" or "greedy"? you should also check if the endpoint
> is offering PCMU.
>
>
>
> On Wed, Jul 1, 2009 at 8:27 PM, qian ma  wrote:
>
>> i want the fs accept the PCMA not PCMU.
>> i add "PCMA" in global_codec_prefs and outbound_codec_prefs, it doesn't
>> work. FS only accept PCMU.
>> why??
>>
>>
>>
>>
>> 2009/7/1 Nandy Dagondon 
>>
>> you FS doesn't accept PCMU. try to add "PCMU" on both variables.
>>>
>>>
>>> On Wed, Jul 1, 2009 at 3:44 PM, qian ma  wrote:
>>>
>>>> thanks for your replies.
>>>>
>>>  my var.xml:
>>>> 
>>>> 
>>>>
>>>>
>>>> below is the sip trace:
>>>>  recv 781 bytes from udp/[58.212.219.104]:40508 at 07:42:37.086711:
>>>>
>>>> 
>>>>INVITE sip:123...@58.212.219.104 SIP/2.0
>>>>Via: SIP/2.0/UDP 192.168.1.241:8422
>>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport
>>>>Max-Forwards: 70
>>>>Contact: 
>>>>To: "123456"
>>>> >
>>>>From: "9876"
>>>> >;tag=057de365
>>>>Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
>>>>CSeq: 1 INVITE
>>>>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
>>>> SUBSCRIBE, INFO
>>>>Content-Type: application/sdp
>>>>User-Agent: eyeBeam release 1102u stamp 52345
>>>>Content-Length: 237
>>>>
>>>>v=0
>>>>o=- 6 2 IN IP4 192.168.1.241
>>>>s=CounterPath eyeBeam 1.5
>>>>c=IN IP4 192.168.1.241
>>>>t=0 0
>>>>m=audio 57862 RTP/AVP 8 101
>>>>a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862
>>>>a=fmtp:101 0-15
>>>>a=rtpmap:101 telephone-event/8000
>>>>a=sendrecv
>>>>
>>>> 
>>>> send 380 bytes to udp/[58.212.219.104]:40508 at 07:42:37.087505:
>>>>
>>>> 
>>>>SIP/2.0 100 Trying
>>>>Via: SIP/2.0/UDP 192.168.1.241:8422
>>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104
>>>>From: "9876"
>>>> >;tag=057de365
>>>>To: "123456"
>>>> >
>>>>Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
>>>>CSeq: 1 INVITE
>>>>User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M
>>>>Content-Length: 0
>>>>
>>>>
>>>> 
>>>> 2009-07-01 15:42:37.87800 [NOTICE] switch_channel.c:602 New Channel
>>>> sofia/maq/9...@58.212.219.104 [574766e1-e71f-4a62-a659-1d43bea135bd]
>>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3214 Channel
>>>> sofia/maq/9...@58.212.219.104 entering state [received][100]
>>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3221 Remote SDP:
>>>> v=0
>>>> o=- 6 2 IN IP4 192.168.1.241
>>>> s=CounterPath eyeBeam 1.5
>>>> c=IN IP4 192.168.1.241
>>>> t=0 0
>>>> m=audio 57862 RTP/AVP 8 101
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-15
>>>> a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862
>>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare
>>>> [PCMA:8:8000:0]/[PCMU:0:8000:20]
>>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf
>>>> payload to 101
>>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare
>>>> [telephone-event:101:8000:0]/[PCMU:0:8000:20]
>>>> 2009-07-01 15:42:37.87800 [NOTICE] sofia.c:3423 Hangup
>>>> sofia/maq/9...@58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>>>> 2009-07-01 15:42:37.87800 [DEBUG] switch_channel.c:1683 Send signal
>>>> sofia/maq/9...@58.212.219.104 [KILL]
>>>> 2009-07-01 15:42:37.87800 [DEBUG] switch_cor

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

2009-07-01 Thread Nandy Dagondon

check the value of "inbound_codec_negotiation" in  the sip_profiles/*.xml
files. is it "generous" or "greedy"? you should also check if the endpoint
is offering PCMU.


On Wed, Jul 1, 2009 at 8:27 PM, qian ma  wrote:

> i want the fs accept the PCMA not PCMU.
> i add "PCMA" in global_codec_prefs and outbound_codec_prefs, it doesn't
> work. FS only accept PCMU.
> why??
>
>
>
>
> 2009/7/1 Nandy Dagondon 
>
> you FS doesn't accept PCMU. try to add "PCMU" on both variables.
>>
>>
>> On Wed, Jul 1, 2009 at 3:44 PM, qian ma  wrote:
>>
>>> thanks for your replies.
>>>
>>  my var.xml:
>>> 
>>> 
>>>
>>>
>>> below is the sip trace:
>>>  recv 781 bytes from udp/[58.212.219.104]:40508 at 07:42:37.086711:
>>>
>>> 
>>>INVITE sip:123...@58.212.219.104 SIP/2.0
>>>Via: SIP/2.0/UDP 192.168.1.241:8422
>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport
>>>Max-Forwards: 70
>>>Contact: 
>>>To: "123456">
>>>From: "9876"
>>> >;tag=057de365
>>>Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
>>>CSeq: 1 INVITE
>>>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
>>> SUBSCRIBE, INFO
>>>Content-Type: application/sdp
>>>User-Agent: eyeBeam release 1102u stamp 52345
>>>Content-Length: 237
>>>
>>>v=0
>>>o=- 6 2 IN IP4 192.168.1.241
>>>s=CounterPath eyeBeam 1.5
>>>c=IN IP4 192.168.1.241
>>>t=0 0
>>>m=audio 57862 RTP/AVP 8 101
>>>a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862
>>>a=fmtp:101 0-15
>>>a=rtpmap:101 telephone-event/8000
>>>a=sendrecv
>>>
>>> 
>>> send 380 bytes to udp/[58.212.219.104]:40508 at 07:42:37.087505:
>>>
>>> 
>>>SIP/2.0 100 Trying
>>>Via: SIP/2.0/UDP 192.168.1.241:8422
>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104
>>>From: "9876"
>>> >;tag=057de365
>>>To: "123456">
>>>Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
>>>CSeq: 1 INVITE
>>>User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M
>>>Content-Length: 0
>>>
>>>
>>> 
>>> 2009-07-01 15:42:37.87800 [NOTICE] switch_channel.c:602 New Channel
>>> sofia/maq/9...@58.212.219.104 [574766e1-e71f-4a62-a659-1d43bea135bd]
>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3214 Channel
>>> sofia/maq/9...@58.212.219.104 entering state [received][100]
>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3221 Remote SDP:
>>> v=0
>>> o=- 6 2 IN IP4 192.168.1.241
>>> s=CounterPath eyeBeam 1.5
>>> c=IN IP4 192.168.1.241
>>> t=0 0
>>> m=audio 57862 RTP/AVP 8 101
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862
>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare
>>> [PCMA:8:8000:0]/[PCMU:0:8000:20]
>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload
>>> to 101
>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare
>>> [telephone-event:101:8000:0]/[PCMU:0:8000:20]
>>> 2009-07-01 15:42:37.87800 [NOTICE] sofia.c:3423 Hangup
>>> sofia/maq/9...@58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>>> 2009-07-01 15:42:37.87800 [DEBUG] switch_channel.c:1683 Send signal
>>> sofia/maq/9...@58.212.219.104 [KILL]
>>> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_session.c:933 Send signal
>>> sofia/maq/9...@58.212.219.104 [BREAK]
>>> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_state_machine.c:397 (
>>> sofia/maq/9...@58.212.219.104) Running State Change CS_HANGUP
>>> 2009-07-01 15:42:37.88976 [DEBUG] switch_core_state_machine.c:433 (
>>> sofia/maq/9...@58.212.219.104) State HANGUP
>>> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:338 Channel
>>> sofia/maq/9...@58.212.219.104 hanging up, cause:
>>> INCOMPATIBLE_DESTINATION
&

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

2009-07-01 Thread Nandy Dagondon

you FS doesn't accept PCMU. try to add "PCMU" on both variables.

On Wed, Jul 1, 2009 at 3:44 PM, qian ma  wrote:

> thanks for your replies.
>
my var.xml:
> 
> 
>
>
> below is the sip trace:
>  recv 781 bytes from udp/[58.212.219.104]:40508 at 07:42:37.086711:
>
>INVITE sip:123...@58.212.219.104  SIP/2.0
>Via: SIP/2.0/UDP 192.168.1.241:8422
> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport
>Max-Forwards: 70
>Contact: 
>To: "123456">
>From: "9876"
> >;tag=057de365
>Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
>CSeq: 1 INVITE
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE, INFO
>Content-Type: application/sdp
>User-Agent: eyeBeam release 1102u stamp 52345
>Content-Length: 237
>
>v=0
>o=- 6 2 IN IP4 192.168.1.241
>s=CounterPath eyeBeam 1.5
>c=IN IP4 192.168.1.241
>t=0 0
>m=audio 57862 RTP/AVP 8 101
>a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862
>a=fmtp:101 0-15
>a=rtpmap:101 telephone-event/8000
>a=sendrecv
>
> send 380 bytes to udp/[58.212.219.104]:40508 at 07:42:37.087505:
>
>SIP/2.0 100 Trying
>Via: SIP/2.0/UDP 192.168.1.241:8422
> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104
>From: "9876"
> >;tag=057de365
>To: "123456">
>Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
>CSeq: 1 INVITE
>User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M
>Content-Length: 0
>
>
> 2009-07-01 15:42:37.87800 [NOTICE] switch_channel.c:602 New Channel
> sofia/maq/9...@58.212.219.104 [574766e1-e71f-4a62-a659-1d43bea135bd]
> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3214 Channel
> sofia/maq/9...@58.212.219.104 entering state [received][100]
> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3221 Remote SDP:
> v=0
> o=- 6 2 IN IP4 192.168.1.241
> s=CounterPath eyeBeam 1.5
> c=IN IP4 192.168.1.241
> t=0 0
> m=audio 57862 RTP/AVP 8 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862
> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare
> [PCMA:8:8000:0]/[PCMU:0:8000:20]
> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload
> to 101
> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare
> [telephone-event:101:8000:0]/[PCMU:0:8000:20]
> 2009-07-01 15:42:37.87800 [NOTICE] sofia.c:3423 Hangup
> sofia/maq/9...@58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION]
> 2009-07-01 15:42:37.87800 [DEBUG] switch_channel.c:1683 Send signal
> sofia/maq/9...@58.212.219.104 [KILL]
> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_session.c:933 Send signal
> sofia/maq/9...@58.212.219.104 [BREAK]
> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_state_machine.c:397 (
> sofia/maq/9...@58.212.219.104) Running State Change CS_HANGUP
> 2009-07-01 15:42:37.88976 [DEBUG] switch_core_state_machine.c:433 (
> sofia/maq/9...@58.212.219.104) State HANGUP
> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:338 Channel
> sofia/maq/9...@58.212.219.104 hanging up, cause: INCOMPATIBLE_DESTINATION
> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:414 Responding to INVITE
> with: 488
> send 676 bytes to udp/[58.212.219.104]:40508 at 07:42:37.089766:
>
>SIP/2.0 488 Not Acceptable Here
>Via: SIP/2.0/UDP 192.168.1.241:8422
> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104
>From: "9876"
> >;tag=057de365
>To: "123456" 
> >;tag=28Q0QB73Bm35K
>Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
>CSeq: 1 INVITE
>User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M
>Accept: application/sdp
>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO
>Supported: timer, precondition, path, replaces
>Allow-Events: talk, refer
>Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
>Content-Length: 0
>
>
> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:46
> sofia/maq/9...@58.212.219.104 Standard HANGUP, cause:
> INCOMPATIBLE_DESTINATION
> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:433 (
> sofia/maq/9...@58.212.219.104) State HANGUP going to sleep
> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:475 (
> sofia/maq/9...@58.212.219.104) State Change CS_HANGUP -> CS_REPORTING
> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_session.c:933 Send signal
> sofia/maq/9...@58.212.219.104 [BREAK]
> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state

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

2009-07-01 Thread Nandy Dagondon

seven,

of course, codec negotiation depends on the order of codecs in the
*_codec_prefs variables. but, the opposite end has also it's own codecs
prefs, too. fs can accept the other end's prefs
(inbound_codec_negotiation=generous) or imposes it's own prefs (=greedy).
you must include the codec in the *_codec_prefs to activate it. is this
correct?

-nandy

On Wed, Jul 1, 2009 at 3:19 PM, seven  wrote:

> absolutely not.
> codec negotiate depending on your conf. do you have a sip trace?
>
> On Jul 1, 2009, at 2:48 PM, qian ma wrote:
>
> hi all
>freeswitch support PCMU only?
>i follow the http://wiki.freeswitch.org/wiki/Codecs#sofia.conf.xml,
> but freeswitch still support PCMU only,
>below is the trace:
>
>2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec
> Compare [PCMA:8:8000:0]/[PCMU:0:8000:20]
> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload
> to 101
> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec Compare
> [telephone-event:101:8000:0]/[PCMU:0:8000:20]
> 2009-07-01 14:39:35.571364 [NOTICE] sofia.c:3423 Hangup
> sofia/maq/9...@58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION]
> 2009-07-01 14:39:35.571364 [DEBUG] switch_channel.c:1683 Send signal
> sofia/maq/9...@58.212.219.104 [KILL]
> 2009-07-01 14:39:35.571364 [DEBUG] switch_core_session.c:933 Send signal
> sofia/maq/9...@58.212.219.104 [BREAK]
> 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:397 (
> sofia/maq/9...@58.212.219.104) Running State Change CS_HANGUP
> 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:433 (
> sofia/maq/9...@58.212.219.104) State HANGUP
>
>
>
>   how to configure the freeswitch??
>   support more codecs???
>
>   thx!
>
> m.q
> ___
> Freeswitch-users mailing list
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>
>
>
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Re: [Freeswitch-users] freeswitch support PCMU only?

2009-07-01 Thread Nandy Dagondon

fs can support lots of codecs. you can find the ff variables defined in
vars.xml:

global_codec_prefs
outbound_codec_prefs

then look for "inbound_codec_negotiation" in
sip_profiles/internal.xml,sip_profiles/external.xml if you want your
codec_prefs to set priority or not.

-nandy

On Wed, Jul 1, 2009 at 2:48 PM, qian ma  wrote:

> hi all
>freeswitch support PCMU only?
>i follow the http://wiki.freeswitch.org/wiki/Codecs#sofia.conf.xml,
> but freeswitch still support PCMU only,
>below is the trace:
>
>2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec
> Compare [PCMA:8:8000:0]/[PCMU:0:8000:20]
> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload
> to 101
> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec Compare
> [telephone-event:101:8000:0]/[PCMU:0:8000:20]
> 2009-07-01 14:39:35.571364 [NOTICE] sofia.c:3423 Hangup
> sofia/maq/9...@58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION]
> 2009-07-01 14:39:35.571364 [DEBUG] switch_channel.c:1683 Send signal
> sofia/maq/9...@58.212.219.104 [KILL]
> 2009-07-01 14:39:35.571364 [DEBUG] switch_core_session.c:933 Send signal
> sofia/maq/9...@58.212.219.104 [BREAK]
> 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:397 (
> sofia/maq/9...@58.212.219.104) Running State Change CS_HANGUP
> 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:433 (
> sofia/maq/9...@58.212.219.104) State HANGUP
>
>
>
>   how to configure the freeswitch??
>   support more codecs???
>
>   thx!
>
> m.q
>
> ___
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> Freeswitch-users@lists.freeswitch.org
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Re: [Freeswitch-users] multiple gateways not working?

2009-06-25 Thread Nandy Dagondon

you can combine the 2 gateways into one bridge app. pls see
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall

/nandy

On Fri, Jun 26, 2009 at 1:48 PM, Edmar Cruz  wrote:

>
> 
>  
>
>  
>   data="effective_caller_id_name=${effective_caller_id_name}"/>
>   data="effective_caller_id_number=${effective_caller_id_number}"/>
>  -->
>  
>
>
> 
>  
>   data="effective_caller_id_name=${effective_caller_id_name}"/>
>   data="effective_caller_id_number=${effective_caller_id_number}"/>
>  -->
>  
>
>
>  
> 
>
>
> Is this correct for multiple gateways? When I try this the first gateway
> works but the second gateway does not work?
>
>
> What is the solution for this can u help me?
>
>
> Thanks
>
> --
> View this message in context:
> http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215324.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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Re: [Freeswitch-users] sofia external profile: external IP problem

2009-06-21 Thread Nandy Dagondon

it's working now, i mean the Auto NAT feature - after i enabled UPNP feature
on my router. it's based on external IP addresses of Ext-SIP-IP and
Ext-RTP-IP when performing "sofia status profile [internal|external]" on the
cli.

however, "sofia status" still shows internal IP address on the external
profile. it should display the external IP address instead.


On Sun, Jun 21, 2009 at 2:20 PM, Nandy Dagondon  wrote:

> the default setting is "auto-nat".
>
> i changed ext-sip-ip=$${external_sip_ip} and
> ext-rtp-ip=$${external_rtp_ip}. both of them are set in vars.xml as stun:
> stun.freeswitch.org. result: same problem
>
> i tried your suggestion. still the same problem.
>
>
>
> On Sun, Jun 21, 2009 at 1:45 PM, Jason White  wrote:
>
>> Nandy Dagondon  wrote:
>> > hi,
>> >
>> > i tested the latest SVN build (13884) using the sample configuration
>> files
>> > ... no modifications whatsoever. but in sofia external profile, the IP
>> > address is my internal address instead of my external IP address.
>> >
>> > did i miss something here?
>>
>> Try setting ext-sip-ip and ext-rtp-ip in the external profile to
>> stun:stun.freeswitch.org
>>
>> This can alternatively be set using global variables in vars.xml in the
>> supplied configuration.
>>
>>
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>>
>
>
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Re: [Freeswitch-users] sofia external profile: external IP problem

2009-06-21 Thread Nandy Dagondon

i just come across the Auto NAT feature in the Wiki. i'm testing if my
router UPNP works w/ FS. STUN works before i updated to SVN.

2009/6/21 João Mesquita 

> What I would guess is the the STUN lookup failed. Do you have anything on
> this box that would prevent FS from doing DNS lookup?
>
> jmesquita
>
>
> On Sun, Jun 21, 2009 at 3:20 AM, Nandy Dagondon  wrote:
>
>> the default setting is "auto-nat".
>>
>> i changed ext-sip-ip=$${external_sip_ip} and
>> ext-rtp-ip=$${external_rtp_ip}. both of them are set in vars.xml as stun:
>> stun.freeswitch.org. result: same problem
>>
>> i tried your suggestion. still the same problem.
>>
>>
>>
>> On Sun, Jun 21, 2009 at 1:45 PM, Jason White  wrote:
>>
>>> Nandy Dagondon  wrote:
>>> > hi,
>>> >
>>> > i tested the latest SVN build (13884) using the sample configuration
>>> files
>>> > ... no modifications whatsoever. but in sofia external profile, the IP
>>> > address is my internal address instead of my external IP address.
>>> >
>>> > did i miss something here?
>>>
>>> Try setting ext-sip-ip and ext-rtp-ip in the external profile to
>>> stun:stun.freeswitch.org
>>>
>>> This can alternatively be set using global variables in vars.xml in the
>>> supplied configuration.
>>>
>>>
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>>
>>
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>
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Re: [Freeswitch-users] sofia external profile: external IP problem

2009-06-20 Thread Nandy Dagondon

the default setting is "auto-nat".

i changed ext-sip-ip=$${external_sip_ip} and ext-rtp-ip=$${external_rtp_ip}.
both of them are set in vars.xml as stun:stun.freeswitch.org. result: same
problem

i tried your suggestion. still the same problem.


On Sun, Jun 21, 2009 at 1:45 PM, Jason White  wrote:

> Nandy Dagondon  wrote:
> > hi,
> >
> > i tested the latest SVN build (13884) using the sample configuration
> files
> > ... no modifications whatsoever. but in sofia external profile, the IP
> > address is my internal address instead of my external IP address.
> >
> > did i miss something here?
>
> Try setting ext-sip-ip and ext-rtp-ip in the external profile to
> stun:stun.freeswitch.org
>
> This can alternatively be set using global variables in vars.xml in the
> supplied configuration.
>
>
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[Freeswitch-users] sofia external profile: external IP problem

2009-06-20 Thread Nandy Dagondon

hi,

i tested the latest SVN build (13884) using the sample configuration files
... no modifications whatsoever. but in sofia external profile, the IP
address is my internal address instead of my external IP address.

did i miss something here? tks.

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

2009-06-20 Thread Nandy Dagondon

hi dave,

tks for sharing us this info. i don't think we can reach 10k prefixes but
your deployment to use external database or mod_lcr is the way to go. re
hardware, i think core2 platform would be enough cuz it will be in a rural
installation. i'm sure it wont reach 200 simultaneous calls.

FS community is really great!

tks once again,
nandy

On Sun, Jun 21, 2009 at 11:50 AM, David Knell  wrote:

> Hi Nandy.
>
> On Sun, 2009-06-21 at 08:58 +0800, Nandy Dagondon wrote:
> > i'm interested to know if anyone employed FS as a local exchange
> > switch. i'm confident FS can handle several calls using RTP by-pass
> > mode. however, i'm more concerned on handling the large dialplan with
> > hundreds (or even a few thousand) exchange prefixes nationwide during
> > call setup.
>
> We have probably ~100k prefixes in our LCR.  We don't put these in the
> dialplan directly; instead, they live in a database and we have an
> external application which routes calls.  FreeSWITCH has mod_lcr which I
> would imagine will do the same sort of thing; we don't use it because it
> wasn't around when we started.
>

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


>
> > i'd be glad to hear experiences and suggestions esp on the hardware
> > dimensioning. we're talking a small exchange up to about 1,100 lines
> > only, mostly linked to the main exchange via MFC-R2.
>
> That'd depend on the number of concurrent calls you need to budget for -
> taking it that 1,100 lines implies maybe 1-200 simultaneous calls, then
> one low-end modern server (Core 2 Duo, etc.) ought to do just fine.


> Cheers --
>
> Dave
>
> --
> David Knell, Director, 3C Limited
> T: +44 20 3298 2000
> E: d...@3c.co.uk
> W: http://www.3c.co.uk
>
>
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[Freeswitch-users] FS as a Class 5 switch

2009-06-20 Thread Nandy Dagondon

hi everybody,

i'm interested to know if anyone employed FS as a local exchange switch. i'm
confident FS can handle several calls using RTP by-pass mode. however, i'm
more concerned on handling the large dialplan with hundreds (or even a few
thousand) exchange prefixes nationwide during call setup.

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

tks,
nandy
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Re: [Freeswitch-users] Is there anyone who is connected to PCCW?

2009-06-16 Thread Nandy Dagondon

what is PCCW? could you please fill in more details what you like to do. to
connect mobile phones w/ FS, the mobile phone has to have SIP feature. pls
search the Wiki for some models.
-nandy
===
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Lapulapu City, Philippines 6015
Mobile: +63-920-6373450
Phone: +63-32-3401807
USA:   +1-360-8122281
http://sites.google.com/site/lanvoxphils



On Tue, Jun 16, 2009 at 3:44 PM, Edmar Cruz  wrote:

>
> PCCW is use for making calls through IP connected through cellphone just
> enter the areacode for example
>
> 900639274522123
>
> 900-prefix
> 63-areacode
> 9274522123 - number?
>
> Has anyone has tried it?
>
> Please help me how to connect to it
> --
> View this message in context:
> http://www.nabble.com/Is-there-anyone-who-is-connected-to-PCCW--tp24049302p24049302.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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Re: [Freeswitch-users] Reducing record_session load

2009-06-06 Thread Nandy Dagondon

i'm using version build 13245M on an Intel D945GCLF2 Atom Dual-core mobo w/
2GB ram.
-nandy

On Sat, Jun 6, 2009 at 11:02 AM, Brian West  wrote:

> You shouldn't be having problems... what version are you using?
> /b
>
> On Jun 5, 2009, at 9:58 PM, Nandy Dagondon wrote:
>
> there 10 client seats so at max. 10 simultaneous calls. however, the number
> of clients may be increased.
> -nandy
>
>
> Brian West
> br...@freeswitch.org
>
> -- Meet us at ClueCon!  http://www.cluecon.com
>
>
>
>
>
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Re: [Freeswitch-users] busy tone detect issue

2009-06-05 Thread Nandy Dagondon

dujinfang,

hv u tried OSLEC? it's really reduced echo even on the cheapy X100P card on
*. oslec works w/ FS, too.

-nandy

On Sat, Jun 6, 2009 at 10:11 AM, dujinfang  wrote:

>
> On Jun 5, 2009, at 1:14 PM, Michael Collins wrote:
>
>
>
> On Thu, Jun 4, 2009 at 6:59 PM, seven  wrote:
>
>> I'm using openzap analog with tone_detect, it works(conference not
>> tested). however, according to the asterisk book, Kewlstart can detect the
>> busy tone and disconnect the circuit. does anyone knows how to configure
>> kewlstart with freeswitch/openzap? guess we don't need tone_detect then.
>>
>
> Dujinfang,
>
> Your telco must support "kewlstart" signaling for this to be effective. The
> telco probably calls it something different, like "disconnect supervision"
> or "drop in loop current" or "battery reversal" or something like that. In
> any case, if the signaling is supported then you need to set up your
> zaptel.conf with the appropriate signaling type, which is either fxoks or
> fxsks. (I can never remember because zaptel does it backwards where if you
> have an FXO port then it uses FXS signaling but if you have an FXS port it
> uses FXO signaling. Stupidity, to be sure, so be aware of it.)
>
>
> 1) Don't know why but the similar zaptel.conf works on asterisk. I guess
> tone_detect in FS is equivalent to busydetect=yes in
> Asterisk(zapata.conf) .
>
>
> zaptel.conf
> # Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" (MASTER)
> fxsks=1
> fxsks=2
> fxsks=3
> fxsks=4
>
> # Span 2: WCTDM/1 "Wildcard TDM400P REV I Board 2"
> fxoks=5
> fxoks=6
> fxoks=7
> fxoks=8
>
> # Global data
>
> loadzone= us
> defaultzone = us
>
>
> zapata.conf
>
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=no
> ;echotraining=800
> rxgain=0.0
> txgain=0.0
> group=0
> callgroup=1
> pickupgroup=1
> immediate=no
> busydetect=yes
>
> I agree the FXO and FXS signaling is weird, why not they just match the
> care name and reverse that internally?
>
> 2) Another essue is if I dial out from a FXO port from a local
> extension(sip and zap), I can hear much echo on FreeSWITCH but not much echo
> on asterisk. the zt.conf as below and I tried to change the
> echo_cancel_level to 32 or 128 got no much difference. Is there any
> equivalent configuration in FS like echocanccelwhenbridged=no in asterisk?
> can I set busydetect and echocancelwhenbridged and other options like
> this  ?
>
> [defaults]
> codec_ms => 20
> wink_ms => 150
> flash_ms => 750
> echo_cancel_level => 64
>
>
> Find the sample zaptel.conf that comes with the zaptel package and search
> it for fxsks or fxoks and you'll see some notes on how to set it up for your
> analog trunks.
>
> -MC
>
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Re: [Freeswitch-users] Reducing record_session load

2009-06-05 Thread Nandy Dagondon

there 10 client seats so at max. 10 simultaneous calls. however, the number
of clients may be increased.
-nandy

On Sat, Jun 6, 2009 at 10:32 AM, Brian West  wrote:

>
> On Jun 5, 2009, at 9:24 PM, Nandy Dagondon wrote:
>
> we experience some latency in the recording files even with PCMU-PCMU
> session to a stereo WAV file. i want to reduce the CPU load hoping to reduce
> this problem. would it help if do the ff?
> 1. save it in PCMU file. i can use sox at the end of the shift.
>
>
> You shouldn't be experiencing this at all... how many are you doing at
> once?
>
> 2. record in mono. does it help?
>
>
> No.
>
> 3. will record_session work w/ proxy_media=true?
>
>
> Nope.
>
>
> Brian West
> br...@freeswitch.org
>
> -- Meet us at ClueCon!  http://www.cluecon.com
>
>
>
>
>
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[Freeswitch-users] Reducing record_session load

2009-06-05 Thread Nandy Dagondon

we experience some latency in the recording files even with PCMU-PCMU
session to a stereo WAV file. i want to reduce the CPU load hoping to reduce
this problem. would it help if do the ff?
1. save it in PCMU file. i can use sox at the end of the shift.
2. record in mono. does it help?
3. will record_session work w/ proxy_media=true?

tks for your help.
-nandy
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Re: [Freeswitch-users] Softphone configuration

2009-06-03 Thread Nandy Dagondon

what are the error messages in the FS CLI output?


On Thu, Jun 4, 2009 at 3:10 AM, Matthew Lockwood  wrote:

> Okay, I did that. I first ran it using X-Lite (which does connect on the
> second try; the first one always times out) - there was lots of output. When
> I tried it with Zoiper, YakaPhone, and Adore Softphone ... nothing. They
> just time out.
>
> M
>
>
> On Wed, Jun 3, 2009 at 10:39 AM, Matthew Lockwood <
> matthew.lockw...@gmail.com> wrote:
>
>> It's installing now. I'll get back with the results shortly.
>>
>> On Wed, Jun 3, 2009 at 10:34 AM, Brian West  wrote:
>>
>>> It usually is... if you pop on IRC people can tell you one way or
>>> another.  Issues never hang around for very long!
>>> /b
>>>
>>> On Jun 3, 2009, at 12:16 PM, Raymond Chandler wrote:
>>>
>>> Matthew Lockwood wrote:
>>>
>>> Is that stable enough to use in production?
>>>
>>> it's more stable than "not working"
>>> -Ray
>>>
>>>
>>>   Brian West
>>> br...@freeswitch.org
>>>
>>> -- Meet us at ClueCon!  http://www.cluecon.com
>>>
>>>
>>>
>>>
>>>
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Re: [Freeswitch-users] Default IVR action

2009-05-28 Thread Nandy Dagondon

tks brian. it worked. i changed the max-timeouts=1 for 1-pass announcement.

-nandy
===
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Lapulapu City, Philippines 6015
Mobile: +63-920-6373450
Phone: +63-32-3401807
USA:   +1-360-8122281
http://sites.google.com/site/lanvoxphils



On Fri, May 29, 2009 at 10:05 AM, Brian West  wrote:

> Set the app after ivr to transfer and set the exit sound to "please stay on
> the line to be connected"/b
>
> On May 28, 2009, at 8:58 PM, Nandy Dagondon wrote:
>
> hi to all,
>
> i'm looking for a default action in the IVR in case the caller doesn't
> press any key. is this option available? with this option, we can add this
> prompt "...  please stay on the line to be connected.".  i know this can be
> done using scripts but it's better to have this feature on the app itself.
>
>
> rgds,
> -nandy
>
>
> Brian West
> br...@freeswitch.org
>
> -- Meet us at ClueCon!  http://www.cluecon.com
>
>
>
>
>
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[Freeswitch-users] Default IVR action

2009-05-28 Thread Nandy Dagondon

hi to all,

i'm looking for a default action in the IVR in case the caller doesn't press
any key. is this option available? with this option, we can add this prompt
"...  please stay on the line to be connected.".  i know this can be done
using scripts but it's better to have this feature on the app itself.

rgds,
-nandy
===
LanVox Systems
Lapulapu City, Philippines 6015
Mobile: +63-920-6373450
Phone: +63-32-3401807
USA:   +1-360-8122281
http://sites.google.com/site/lanvoxphils
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Re: [Freeswitch-users] FS PABX experiences?

2009-05-27 Thread Nandy Dagondon

IMHO, you have tons of features w/ FS. i've setup FS on a low-power
consumption Intel D945GCLF2 motherboard (Atom dual-core CPU) ideal for 24/7
operation on a 10-seat contact center w/ default conversation recording. no
problem.

another cool feature. you can route the call based on the Caller ID. so u hv
to consider the selection of the telco (FXO) gateway.

one advantage over key system - you can turn PCs into extension phones using
free softphones. just use USB phones instead of  headsets.

re maintenance, just provide remote access to the FS box. in my home FS, i
create dialplan to reboot or shutdown my FS. it helps when problems occur
(not encountered so far).

-nandy
===
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Lapulapu City, Philippines 6015
Mobile: +63-920-6373450
Phone: +63-32-3401807
USA:   +1-360-8122281
http://sites.google.com/site/lanvoxphils

On Thu, May 28, 2009 at 6:51 AM, Neale Banks  wrote:

> Hi,
>
> We're considering deploying FS instead of a traditional PABX/Key-System in
> a small office environment (i.e. primarily non-technical users, 15-20
> handsets).
>
> Anyone have any experiences (good/bad/whatever) in this sort of scenario?
>
> Thanks,
> Neale.
>
>
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Re: [Freeswitch-users] Cool names for my VoIP company

2009-05-22 Thread Nandy Dagondon

how about InterTalk or InterMedia?

-nandy
===
LanVox Systems
Lapulapu City, Philippines 6015
Mobile: +63-920-6373450
Phone: +63-32-3401807
USA:   +1-360-8122281
http://sites.google.com/site/lanvoxphils



On Fri, May 22, 2009 at 11:52 PM, Brian West  wrote:

> I say bkw_
> On May 22, 2009, at 10:45 AM, SP wrote:
>
> Dasbus
>
>
> Brian West
> br...@freeswitch.org
>
> -- Meet us at ClueCon!  http://www.cluecon.com
>
>
>
>
>
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Re: [Freeswitch-users] Compact, fanless appliance?

2009-04-30 Thread Nandy Dagondon

rhino used the dual-core atom mobo d945gclf2 but it requires
downloading/building the linux r8168 LAN driver.

-nandy


On Fri, May 1, 2009 at 11:40 AM, Brian West  wrote:

> I have two intel atom boxes sitting on a shelf above my desk ... works like
> a charm!
> /b
>
> On Apr 30, 2009, at 10:31 PM, Mitch Capper wrote:
>
> You may want to look at the Intel Atom combo machines  you can get a 1.6
> ghz machine probably for around $100-150 USD in a very small form factor and
> very powerful.
>
> ~Mitch
>
>
> Brian West
> br...@freeswitch.org
>
> -- Meet us at ClueCon!  http://www.cluecon.com
>
>
>
>
>
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Re: [Freeswitch-users] Compact, fanless appliance?

2009-04-30 Thread Nandy Dagondon

hi guys,

i've installed FreeNas using CF-to-IDE adaptor and SanDisk 128MB CF. it's
working fine. but i want to try FS on a 16GB Kingston CF. anyone tried this?
if none, i can also settle down for 8GB. pls mention which brand/size works.

tks,
nandy



On Thu, Apr 30, 2009 at 10:46 PM, Fred-145  wrote:

>
> Thanks guys for the links on CF-to-IDE adaptors.
> --
> View this message in context:
> http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23317579.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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Re: [Freeswitch-users] Getting a free Did number for my FS

2009-03-15 Thread Nandy Dagondon

perhaps you're referring to VPN (Virtual Phone Number). you can visit
http://www.ipkall.com that offers free Washington state numbers.


On Sun, Mar 15, 2009 at 12:47 AM, Meftah Tayeb wrote:

> hello,
> please ho to get a free did number ?
> also, is it pocible to link it to my FS ?
> thanks
>
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Re: [Freeswitch-users] Lunchbox-type PC as small server?

2009-03-06 Thread Nandy Dagondon

take a look at intel atom mobo d945gclf2 (dual-core). it has one PCI slot
and an S-Video and VGA video ports. rhino is using this platform.

On Sat, Mar 7, 2009 at 12:30 AM, Fred  wrote:

> Hello
>
>I'm looking for a small, lunchbox-like PC to build a small-form
> factor CRM server to sell to small companies. Ideally, it should have
> one PCI slot so that I can stick a voice card to connect to an analog
> phone line and run FreeSwitch as well.
>
> I like Asus' EeeBox (www.asus.com/products.aspx?l1=24&l2=165) but it
> doesn't have room for a PCI slot, and I'm concerned about its performance.
>
> I also like stuff from MiniITX (www.mini-itx.com) , but they're a bit
> pricey, and might also not be fast enough to act as a server.
>
> Are there brands/models you think I should look at?
>
> Thank you.
>
>
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[Freeswitch-users] Default IVR action

2009-02-19 Thread Nandy Dagondon

hi  everybody,

i'm looking for a default action in an IVR if the caller doesn't press any
key. for example, the caller will be transferred to the operator (or fifo)
if no key is received after, let's say 5 seconds. is this available in the
IVR? pls show a sample.

tks,
-nandy
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Re: [Freeswitch-users] TDM400 FXO can dialout only once

2009-01-25 Thread Nandy Dagondon

i monitored the line using another phone. there's indeed dialtone in all
attempts.
i see TONE_DETECTED in the first call but i wonder there's a WARNING message
immediately following [WARNING] mod_openzap.c:1196 on_fxo_signal() Unhandled
type for channel 2:1.
the dialtone freq should be okay since it's detected in the first call.could
the WARNING message gives us a hint of a possible problem other than the
dialtone freq?

okay, i'll try the SVN version next.


On Mon, Jan 26, 2009 at 10:18 AM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:

> Its not detecting a dial tone on the failure case.
> Before dialing it waits until it picks up dialtone.
> Try the svn trunk version to see if it works any better or verify there is
> a dialtone on the line.
>
> On Jan 25, 2009 6:19 PM, "Nandy Dagondon"  wrote:
>
> hi everybody,
>
> i installed FS 1.0.2 to an Atom mobo (Intel D945GCLF2). it's working using
> IP phones, softphones and digium FXS port. but there's a problem in dialing
> out to PSTN using digium tdm400 fxo - it works fine on the first attempt
> (after starting FS) but it fails on the subsequent attempts. i tested to
> call using the FXS port and IP phone. same problem.
>
> before i place any call,  i checked >oz dump 2 1 (show current state =
> DOWN, last state = DOWN)
>
> in the first call, there's this message:
> [WARNING] mod_openzap.c:1196 on_fxo_signal() Unhandled type for channel 2:1
> but
>
> then i hangup. checked >oz dump 21 (show current state=DOWN, last
> state=HANGUP)
>
> in the 2nd (and subsequent) attempts, the fxo just goes off-hook but
> doesn't send the dtmf tones.
> >oz dump 2 1 (shows current state = DIALING, last state = DOWN)
>
> has anyone encountered this problem before? i appreciate for any help to
> correct this problem.
>
> tks,
> nandy
>
>
> Environment:
> ==
> kernel 2.6.18-92.1.22.el5
> FS 1.0.2
> zaptel 1.4.11
> oslec
> digium TDM400P Rev. I (1-FXS, 2-FXO,3-4 vacant)
>
> zaptel.conf
> 
> loadzone = us
> defaultzone=us
> channels=1-2
> alaw=1-4
> fxsks=2
> fxoks=1
>
>
> openzap.conf.xml:
> ===
> 
>   
> 
> 
> 
> 
>   
>   
>   
> 
>   
>   
>   
>   
>   
>   
>   
>   
>   
>   
>   
>   
>   
> 
> 
>   
>   
>   
>   
>   
>   
>   
>   
>   
>   
>   
>   
>   
> 
>   
> 
>
> openzap.conf
> ==
> [span zt]
> name => OpenZAP FXS
> number => 1
> fxs-channel => 1
>
> [span zt]
> name => OpenZAP FXO
> number => 2
> fxo-channel => 2
>
> tones.conf  (the dialtone and ring tone is set to Philipping tones)
> 
> [us]
> generate-dial => v=-7;%(1000,0,425)
> detect-dial => 425
>
> generate-ring => v=-7;%(1000,4000,425,480)
> detect-ring => 425,480
>
> generate-busy => v=-7;%(500,500,480,620)
> detect-busy => 480,620
>
> generate-attn => v=0;%(200,300,1400,1800)
> detect-attn => 1400,1800
>
> generate-callwaiting-sas => v=0;%(300,1,440)
> detect-callwaiting-sas => 440
>
> generate-callwaiting-cas => v=0;%(80,0,2750,2130)
> detect-callwaiting-cas => 2750,2130
>
> detect-fail1 => 913.8
> detect-fail2 => 1370.6
> detect-fail3 => 776.7
>
> LOG OF FIRST CALL (OK)
> 
> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:152
> switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute
> bridge(openzap/2/1/3400534)
> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:340 tech_init() Set codec PCMU
> 20ms
> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1017 channel_outgoing_channel()
> Connect outbound channel OpenZAP/2:1/3400534
> 2009-01-25 10:35:58 [NOTICE] switch_channel.c:565 switch_channel_set_name()
> New Channel OpenZAP/2:1/3400534 [e5f12114-ea88-11dd-9f5c-290fb4a527a4]
> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1026 channel_outgoing_channel()
> (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT
> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807
> switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534
> [BREAK]
> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:52 analog_fxo_outgoing_call()
> Changing state on 2:1 from DOWN to DIALING
> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:239 zap_analog_channel_run()
> ANALOG CHANNEL thread starting.
> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379
> switch_core_session_run() (OpenZAP/2:1/340053

[Freeswitch-users] TDM400 FXO can dialout only once

2009-01-25 Thread Nandy Dagondon

hi everybody,

i installed FS 1.0.2 to an Atom mobo (Intel D945GCLF2). it's working using
IP phones, softphones and digium FXS port. but there's a problem in dialing
out to PSTN using digium tdm400 fxo - it works fine on the first attempt
(after starting FS) but it fails on the subsequent attempts. i tested to
call using the FXS port and IP phone. same problem.

before i place any call,  i checked >oz dump 2 1 (show current state = DOWN,
last state = DOWN)

in the first call, there's this message:
[WARNING] mod_openzap.c:1196 on_fxo_signal() Unhandled type for channel 2:1
but

then i hangup. checked >oz dump 21 (show current state=DOWN, last
state=HANGUP)

in the 2nd (and subsequent) attempts, the fxo just goes off-hook but doesn't
send the dtmf tones.
>oz dump 2 1 (shows current state = DIALING, last state = DOWN)

has anyone encountered this problem before? i appreciate for any help to
correct this problem.

tks,
nandy


Environment:
==
kernel 2.6.18-92.1.22.el5
FS 1.0.2
zaptel 1.4.11
oslec
digium TDM400P Rev. I (1-FXS, 2-FXO,3-4 vacant)

zaptel.conf

loadzone = us
defaultzone=us
channels=1-2
alaw=1-4
fxsks=2
fxoks=1


openzap.conf.xml:
===

  




  
  
  

  
  
  
  
  
  
  
  
  
  
  
  
  


  
  
  
  
  
  
  
  
  
  
  
  
  

  


openzap.conf
==
[span zt]
name => OpenZAP FXS
number => 1
fxs-channel => 1

[span zt]
name => OpenZAP FXO
number => 2
fxo-channel => 2

tones.conf  (the dialtone and ring tone is set to Philipping tones)

[us]
generate-dial => v=-7;%(1000,0,425)
detect-dial => 425

generate-ring => v=-7;%(1000,4000,425,480)
detect-ring => 425,480

generate-busy => v=-7;%(500,500,480,620)
detect-busy => 480,620

generate-attn => v=0;%(200,300,1400,1800)
detect-attn => 1400,1800

generate-callwaiting-sas => v=0;%(300,1,440)
detect-callwaiting-sas => 440

generate-callwaiting-cas => v=0;%(80,0,2750,2130)
detect-callwaiting-cas => 2750,2130

detect-fail1 => 913.8
detect-fail2 => 1370.6
detect-fail3 => 776.7

LOG OF FIRST CALL (OK)

2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:152
switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute
bridge(openzap/2/1/3400534)
2009-01-25 10:35:58 [DEBUG] mod_openzap.c:340 tech_init() Set codec PCMU
20ms
2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1017 channel_outgoing_channel()
Connect outbound channel OpenZAP/2:1/3400534
2009-01-25 10:35:58 [NOTICE] switch_channel.c:565 switch_channel_set_name()
New Channel OpenZAP/2:1/3400534 [e5f12114-ea88-11dd-9f5c-290fb4a527a4]
2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1026 channel_outgoing_channel()
(OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT
2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807
switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534
[BREAK]
2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:52 analog_fxo_outgoing_call()
Changing state on 2:1 from DOWN to DIALING
2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:239 zap_analog_channel_run()
ANALOG CHANNEL thread starting.
2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379
switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change CS_INIT
2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run()
Executing state handler on 2:1 for DIALING
2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444
switch_core_session_run() (OpenZAP/2:1/3400534) State INIT
2009-01-25 10:35:58 [DEBUG] mod_openzap.c:364 channel_on_init()
(OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING
2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807
switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534
[BREAK]
2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444
switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to sleep
2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379
switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change
CS_ROUTING
2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447
switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING
2009-01-25 10:35:58 [DEBUG] mod_openzap.c:387 channel_on_routing()
OpenZAP/2:1/3400534 CHANNEL ROUTING
2009-01-25 10:35:58 [DEBUG] switch_ivr_originate.c:58 originate_on_routing()
(OpenZAP/2:1/3400534) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807
switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534
[BREAK]
2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447
switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going to sleep
2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379
switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change
CS_CONSUME_MEDIA
2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:466
switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA
2009-0

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